diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2012-07-24 13:37:37 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2012-07-24 13:37:37 -0700 |
| commit | dbf7b5915b39bfff548e4c6a3a753fc291a60e25 (patch) | |
| tree | 55c457a22aa869d2ab558317877138369ae5f9bb | |
| parent | d14b7a419a664cd7c1c585c9e7fffee9e9051d53 (diff) | |
| parent | c1b623d9e4117d18d244e9b7fb30d2c27aeaf074 (diff) | |
Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound update from Takashi Iwai:
"This is a fairly quiet release in all sound area. Only a little bit
of changes in the core side while most of changes are seen in the
drivers.
HD-audio:
- A few new codec additions for Nvidia, Realtek and VIA
- Intel Haswell audio support
- Support for "phantom" jacks for consistent jack reporting
- Major clean-ups in HDMI/DP driver codes
- A workaround for inverted digital-mic pins with Realtek codecs
- Removal of beep_mode=2 option
ASoC:
- Added the ability to add and remove DAPM paths dynamically, mostly
for reparenting on clock changes
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500,
TI Isabelle and Wolfson Microelectronics WM5102 and WM5110
- DAPM fixes for the recent locking changes
- Fix for _PRE and _POST widgets (which have been broken for a few
releases now)
- A couple of minor driver updates
Misc
- Conversion to new dev_pm_ops in platform and PCI drivers
- LTC support and some fixes in PCXHR driver
- A few fixes and PM support for ISA OPti9xx and WSS cards
- Some TLV code cleanup
- Move driver-specific headers from include/sound to local dirs"
* tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (212 commits)
ASoC: dapm: Fix _PRE and _POST events for DAPM performance improvements
ALSA: hda - add dock support for Thinkpad X230 Tablet
ALSA: hda - Turn on PIN_OUT from hdmi playback prepare.
ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define
ASoC: littlemill: Add userspace control of the WM1250 I/O
ASoC: wm8994: Update micdet for irqdomain conversion
ALSA: hda - make sure alc268 does not OOPS on codec parse
ALSA: hda - Add support for Realtek ALC282
ALSA: hda - Fix index number conflicts of phantom jacks
ALSA: opti9xx: Fix section mismatch by PM support
ALSA: snd-opti9xx: Implement suspend/resume
ALSA: hda - Add new GPU codec ID to snd-hda
ALSA: hda - Fix driver type of Haswell controller to AZX_DRIVER_SCH
ALSA: hda - add Haswell HDMI codec id
ALSA: hda - Add DeviceID for Haswell HDA
ALSA: wss_lib: Fix resume on Yamaha OPL3-SAx
ALSA: wss_lib: fix suspend/resume
ALSA: es1938: replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE
ALSA: tlv: add DECLARE_TLV_DB_RANGE()
ALSA: tlv: add DECLARE_TLV_CONTAINER()
...
239 files changed, 18347 insertions, 2642 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 221b81016dba..4e4d0bc9816f 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -875,8 +875,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. setup before initializing the codecs. This option is available only when CONFIG_SND_HDA_PATCH_LOADER=y is set. See HD-Audio.txt for details. - beep_mode - Selects the beep registration mode (0=off, 1=on, 2= - dynamic registration via mute switch on/off); the default + beep_mode - Selects the beep registration mode (0=off, 1=on); default value is set via CONFIG_SND_HDA_INPUT_BEEP_MODE kconfig. [Single (global) options] diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 03f7897c6414..7456360e161c 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -15,19 +15,24 @@ ALC260 ALC262 ====== - N/A + inv-dmic Inverted internal mic workaround ALC267/268 ========== - N/A + inv-dmic Inverted internal mic workaround -ALC269 +ALC269/270/275/276/280/282 ====== laptop-amic Laptops with analog-mic input laptop-dmic Laptops with digital-mic input + alc269-dmic Enable ALC269(VA) digital mic workaround + alc271-dmic Enable ALC271X digital mic workaround + inv-dmic Inverted internal mic workaround + lenovo-dock Enables docking station I/O for some Lenovos ALC662/663/272 ============== + mario Chromebook mario model fixup asus-mode1 ASUS asus-mode2 ASUS asus-mode3 ASUS @@ -36,6 +41,7 @@ ALC662/663/272 asus-mode6 ASUS asus-mode7 ASUS asus-mode8 ASUS + inv-dmic Inverted internal mic workaround ALC680 ====== @@ -46,6 +52,7 @@ ALC882/883/885/888/889 acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G acer-aspire-8930g Acer Aspire 8330G/6935G acer-aspire Acer Aspire others + inv-dmic Inverted internal mic workaround ALC861/660 ========== diff --git a/MAINTAINERS b/MAINTAINERS index 0ed7048352b5..8ae601c431d6 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6765,9 +6765,11 @@ F: include/linux/tifm.h TI LM49xxx FAMILY ASoC CODEC DRIVERS M: M R Swami Reddy <mr.swami.reddy@ti.com> +M: Vishwas A Deshpande <vishwas.a.deshpande@ti.com> L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Maintained F: sound/soc/codecs/lm49453* +F: sound/soc/codecs/isabelle* TI TWL4030 SERIES SOC CODEC DRIVER M: Peter Ujfalusi <peter.ujfalusi@ti.com> diff --git a/arch/arm/mach-ux500/board-mop500.c b/arch/arm/mach-ux500/board-mop500.c index 84461fa2a3ba..a310222951da 100644 --- a/arch/arm/mach-ux500/board-mop500.c +++ b/arch/arm/mach-ux500/board-mop500.c @@ -25,6 +25,7 @@ #include <linux/mfd/tc3589x.h> #include <linux/mfd/tps6105x.h> #include <linux/mfd/abx500/ab8500-gpio.h> +#include <linux/mfd/abx500/ab8500-codec.h> #include <linux/leds-lp5521.h> #include <linux/input.h> #include <linux/smsc911x.h> @@ -97,6 +98,18 @@ static struct ab8500_gpio_platform_data ab8500_gpio_pdata = { 0x7A, 0x00, 0x00}, }; +/* ab8500-codec */ +static struct ab8500_codec_platform_data ab8500_codec_pdata = { + .amics = { + .mic1_type = AMIC_TYPE_DIFFERENTIAL, + .mic2_type = AMIC_TYPE_DIFFERENTIAL, + .mic1a_micbias = AMIC_MICBIAS_VAMIC1, + .mic1b_micbias = AMIC_MICBIAS_VAMIC1, + .mic2_micbias = AMIC_MICBIAS_VAMIC2 + }, + .ear_cmv = EAR_CMV_0_95V +}; + static struct gpio_keys_button snowball_key_array[] = { { .gpio = 32, @@ -195,6 +208,7 @@ static struct ab8500_platform_data ab8500_platdata = { .regulator = ab8500_regulators, .num_regulator = ARRAY_SIZE(ab8500_regulators), .gpio = &ab8500_gpio_pdata, + .codec = &ab8500_codec_pdata, }; static struct resource ab8500_resources[] = { diff --git a/include/linux/ac97_codec.h b/include/linux/ac97_codec.h deleted file mode 100644 index 0260c3e79fdd..000000000000 --- a/include/linux/ac97_codec.h +++ /dev/null @@ -1,362 +0,0 @@ -#ifndef _AC97_CODEC_H_ -#define _AC97_CODEC_H_ - -#include <linux/types.h> -#include <linux/soundcard.h> - -/* AC97 1.0 */ -#define AC97_RESET 0x0000 // -#define AC97_MASTER_VOL_STEREO 0x0002 // Line Out -#define AC97_HEADPHONE_VOL 0x0004 // -#define AC97_MASTER_VOL_MONO 0x0006 // TAD Output -#define AC97_MASTER_TONE 0x0008 // -#define AC97_PCBEEP_VOL 0x000a // none -#define AC97_PHONE_VOL 0x000c // TAD Input (mono) -#define AC97_MIC_VOL 0x000e // MIC Input (mono) -#define AC97_LINEIN_VOL 0x0010 // Line Input (stereo) -#define AC97_CD_VOL 0x0012 // CD Input (stereo) -#define AC97_VIDEO_VOL 0x0014 // none -#define AC97_AUX_VOL 0x0016 // Aux Input (stereo) -#define AC97_PCMOUT_VOL 0x0018 // Wave Output (stereo) -#define AC97_RECORD_SELECT 0x001a // -#define AC97_RECORD_GAIN 0x001c -#define AC97_RECORD_GAIN_MIC 0x001e -#define AC97_GENERAL_PURPOSE 0x0020 -#define AC97_3D_CONTROL 0x0022 -#define AC97_MODEM_RATE 0x0024 -#define AC97_POWER_CONTROL 0x0026 - -/* AC'97 2.0 */ -#define AC97_EXTENDED_ID 0x0028 /* Extended Audio ID */ -#define AC97_EXTENDED_STATUS 0x002A /* Extended Audio Status */ -#define AC97_PCM_FRONT_DAC_RATE 0x002C /* PCM Front DAC Rate */ -#define AC97_PCM_SURR_DAC_RATE 0x002E /* PCM Surround DAC Rate */ -#define AC97_PCM_LFE_DAC_RATE 0x0030 /* PCM LFE DAC Rate */ -#define AC97_PCM_LR_ADC_RATE 0x0032 /* PCM LR ADC Rate */ -#define AC97_PCM_MIC_ADC_RATE 0x0034 /* PCM MIC ADC Rate */ -#define AC97_CENTER_LFE_MASTER 0x0036 /* Center + LFE Master Volume */ -#define AC97_SURROUND_MASTER 0x0038 /* Surround (Rear) Master Volume */ -#define AC97_RESERVED_3A 0x003A /* Reserved in AC '97 < 2.2 */ - -/* AC'97 2.2 */ -#define AC97_SPDIF_CONTROL 0x003A /* S/PDIF Control */ - -/* range 0x3c-0x58 - MODEM */ -#define AC97_EXTENDED_MODEM_ID 0x003C -#define AC97_EXTEND_MODEM_STAT 0x003E -#define AC97_LINE1_RATE 0x0040 -#define AC97_LINE2_RATE 0x0042 -#define AC97_HANDSET_RATE 0x0044 -#define AC97_LINE1_LEVEL 0x0046 -#define AC97_LINE2_LEVEL 0x0048 -#define AC97_HANDSET_LEVEL 0x004A -#define AC97_GPIO_CONFIG 0x004C -#define AC97_GPIO_POLARITY 0x004E -#define AC97_GPIO_STICKY 0x0050 -#define AC97_GPIO_WAKE_UP 0x0052 -#define AC97_GPIO_STATUS 0x0054 -#define AC97_MISC_MODEM_STAT 0x0056 -#define AC97_RESERVED_58 0x0058 - -/* registers 0x005a - 0x007a are vendor reserved */ - -#define AC97_VENDOR_ID1 0x007c -#define AC97_VENDOR_ID2 0x007e - -/* volume control bit defines */ -#define AC97_MUTE 0x8000 -#define AC97_MICBOOST 0x0040 -#define AC97_LEFTVOL 0x3f00 -#define AC97_RIGHTVOL 0x003f - -/* record mux defines */ -#define AC97_RECMUX_MIC 0x0000 -#define AC97_RECMUX_CD 0x0101 -#define AC97_RECMUX_VIDEO 0x0202 -#define AC97_RECMUX_AUX 0x0303 -#define AC97_RECMUX_LINE 0x0404 -#define AC97_RECMUX_STEREO_MIX 0x0505 -#define AC97_RECMUX_MONO_MIX 0x0606 -#define AC97_RECMUX_PHONE 0x0707 - -/* general purpose register bit defines */ -#define AC97_GP_LPBK 0x0080 /* Loopback mode */ -#define AC97_GP_MS 0x0100 /* Mic Select 0=Mic1, 1=Mic2 */ -#define AC97_GP_MIX 0x0200 /* Mono output select 0=Mix, 1=Mic */ -#define AC97_GP_RLBK 0x0400 /* Remote Loopback - Modem line codec */ -#define AC97_GP_LLBK 0x0800 /* Local Loopback - Modem Line codec */ -#define AC97_GP_LD 0x1000 /* Loudness 1=on */ -#define AC97_GP_3D 0x2000 /* 3D Enhancement 1=on */ -#define AC97_GP_ST 0x4000 /* Stereo Enhancement 1=on */ -#define AC97_GP_POP 0x8000 /* Pcm Out Path, 0=pre 3D, 1=post 3D */ - -/* extended audio status and control bit defines */ -#define AC97_EA_VRA 0x0001 /* Variable bit rate enable bit */ -#define AC97_EA_DRA 0x0002 /* Double-rate audio enable bit */ -#define AC97_EA_SPDIF 0x0004 /* S/PDIF Enable bit */ -#define AC97_EA_VRM 0x0008 /* Variable bit rate for MIC enable bit */ -#define AC97_EA_CDAC 0x0040 /* PCM Center DAC is ready (Read only) */ -#define AC97_EA_SDAC 0x0040 /* PCM Surround DACs are ready (Read only) */ -#define AC97_EA_LDAC 0x0080 /* PCM LFE DAC is ready (Read only) */ -#define AC97_EA_MDAC 0x0100 /* MIC ADC is ready (Read only) */ -#define AC97_EA_SPCV 0x0400 /* S/PDIF configuration valid (Read only) */ -#define AC97_EA_PRI 0x0800 /* Turns the PCM Center DAC off */ -#define AC97_EA_PRJ 0x1000 /* Turns the PCM Surround DACs off */ -#define AC97_EA_PRK 0x2000 /* Turns the PCM LFE DAC off */ -#define AC97_EA_PRL 0x4000 /* Turns the MIC ADC off */ -#define AC97_EA_SLOT_MASK 0xffcf /* Mask for slot assignment bits */ -#define AC97_EA_SPSA_3_4 0x0000 /* Slot assigned to 3 & 4 */ -#define AC97_EA_SPSA_7_8 0x0010 /* Slot assigned to 7 & 8 */ -#define AC97_EA_SPSA_6_9 0x0020 /* Slot assigned to 6 & 9 */ -#define AC97_EA_SPSA_10_11 0x0030 /* Slot assigned to 10 & 11 */ - -/* S/PDIF control bit defines */ -#define AC97_SC_PRO 0x0001 /* Professional status */ -#define AC97_SC_NAUDIO 0x0002 /* Non audio stream */ -#define AC97_SC_COPY 0x0004 /* Copyright status */ -#define AC97_SC_PRE 0x0008 /* Preemphasis status */ -#define AC97_SC_CC_MASK 0x07f0 /* Category Code mask */ -#define AC97_SC_L 0x0800 /* Generation Level status */ -#define AC97_SC_SPSR_MASK 0xcfff /* S/PDIF Sample Rate bits */ -#define AC97_SC_SPSR_44K 0x0000 /* Use 44.1kHz Sample rate */ -#define AC97_SC_SPSR_48K 0x2000 /* Use 48kHz Sample rate */ -#define AC97_SC_SPSR_32K 0x3000 /* Use 32kHz Sample rate */ -#define AC97_SC_DRS 0x4000 /* Double Rate S/PDIF */ -#define AC97_SC_V 0x8000 /* Validity status */ - -/* powerdown control and status bit defines */ - -/* status */ -#define AC97_PWR_MDM 0x0010 /* Modem section ready */ -#define AC97_PWR_REF 0x0008 /* Vref nominal */ -#define AC97_PWR_ANL 0x0004 /* Analog section ready */ -#define AC97_PWR_DAC 0x0002 /* DAC section ready */ -#define AC97_PWR_ADC 0x0001 /* ADC section ready */ - -/* control */ -#define AC97_PWR_PR0 0x0100 /* ADC and Mux powerdown */ -#define AC97_PWR_PR1 0x0200 /* DAC powerdown */ -#define AC97_PWR_PR2 0x0400 /* Output mixer powerdown (Vref on) */ -#define AC97_PWR_PR3 0x0800 /* Output mixer powerdown (Vref off) */ -#define AC97_PWR_PR4 0x1000 /* AC-link powerdown */ -#define AC97_PWR_PR5 0x2000 /* Internal Clk disable */ -#define AC97_PWR_PR6 0x4000 /* HP amp powerdown */ -#define AC97_PWR_PR7 0x8000 /* Modem off - if supported */ - -/* extended audio ID register bit defines */ -#define AC97_EXTID_VRA 0x0001 -#define AC97_EXTID_DRA 0x0002 -#define AC97_EXTID_SPDIF 0x0004 -#define AC97_EXTID_VRM 0x0008 -#define AC97_EXTID_DSA0 0x0010 -#define AC97_EXTID_DSA1 0x0020 -#define AC97_EXTID_CDAC 0x0040 -#define AC97_EXTID_SDAC 0x0080 -#define AC97_EXTID_LDAC 0x0100 -#define AC97_EXTID_AMAP 0x0200 -#define AC97_EXTID_REV0 0x0400 -#define AC97_EXTID_REV1 0x0800 -#define AC97_EXTID_ID0 0x4000 -#define AC97_EXTID_ID1 0x8000 - -/* extended status register bit defines */ -#define AC97_EXTSTAT_VRA 0x0001 -#define AC97_EXTSTAT_DRA 0x0002 -#define AC97_EXTSTAT_SPDIF 0x0004 -#define AC97_EXTSTAT_VRM 0x0008 -#define AC97_EXTSTAT_SPSA0 0x0010 -#define AC97_EXTSTAT_SPSA1 0x0020 -#define AC97_EXTSTAT_CDAC 0x0040 -#define AC97_EXTSTAT_SDAC 0x0080 -#define AC97_EXTSTAT_LDAC 0x0100 -#define AC97_EXTSTAT_MADC 0x0200 -#define AC97_EXTSTAT_SPCV 0x0400 -#define AC97_EXTSTAT_PRI 0x0800 -#define AC97_EXTSTAT_PRJ 0x1000 -#define AC97_EXTSTAT_PRK 0x2000 -#define AC97_EXTSTAT_PRL 0x4000 - -/* extended audio ID register bit defines */ -#define AC97_EXTID_VRA 0x0001 -#define AC97_EXTID_DRA 0x0002 -#define AC97_EXTID_SPDIF 0x0004 -#define AC97_EXTID_VRM 0x0008 -#define AC97_EXTID_DSA0 0x0010 -#define AC97_EXTID_DSA1 0x0020 -#define AC97_EXTID_CDAC 0x0040 -#define AC97_EXTID_SDAC 0x0080 -#define AC97_EXTID_LDAC 0x0100 -#define AC97_EXTID_AMAP 0x0200 -#define AC97_EXTID_REV0 0x0400 -#define AC97_EXTID_REV1 0x0800 -#define AC97_EXTID_ID0 0x4000 -#define AC97_EXTID_ID1 0x8000 - -/* extended status register bit defines */ -#define AC97_EXTSTAT_VRA 0x0001 -#define AC97_EXTSTAT_DRA 0x0002 -#define AC97_EXTSTAT_SPDIF 0x0004 -#define AC97_EXTSTAT_VRM 0x0008 -#define AC97_EXTSTAT_SPSA0 0x0010 -#define AC97_EXTSTAT_SPSA1 0x0020 -#define AC97_EXTSTAT_CDAC 0x0040 -#define AC97_EXTSTAT_SDAC 0x0080 -#define AC97_EXTSTAT_LDAC 0x0100 -#define AC97_EXTSTAT_MADC 0x0200 -#define AC97_EXTSTAT_SPCV 0x0400 -#define AC97_EXTSTAT_PRI 0x0800 -#define AC97_EXTSTAT_PRJ 0x1000 -#define AC97_EXTSTAT_PRK 0x2000 -#define AC97_EXTSTAT_PRL 0x4000 - -/* useful power states */ -#define AC97_PWR_D0 0x0000 /* everything on */ -#define AC97_PWR_D1 AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR4 -#define AC97_PWR_D2 AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4 -#define AC97_PWR_D3 AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4 -#define AC97_PWR_ANLOFF AC97_PWR_PR2|AC97_PWR_PR3 /* analog section off */ - -/* Total number of defined registers. */ -#define AC97_REG_CNT 64 - - -/* OSS interface to the ac97s.. */ -#define AC97_STEREO_MASK (SOUND_MASK_VOLUME|SOUND_MASK_PCM|\ - SOUND_MASK_LINE|SOUND_MASK_CD|\ - SOUND_MASK_ALTPCM|SOUND_MASK_IGAIN|\ - SOUND_MASK_LINE1|SOUND_MASK_VIDEO) - -#define AC97_SUPPORTED_MASK (AC97_STEREO_MASK | \ - SOUND_MASK_BASS|SOUND_MASK_TREBLE|\ - SOUND_MASK_SPEAKER|SOUND_MASK_MIC|\ - SOUND_MASK_PHONEIN|SOUND_MASK_PHONEOUT) - -#define AC97_RECORD_MASK (SOUND_MASK_MIC|\ - SOUND_MASK_CD|SOUND_MASK_IGAIN|SOUND_MASK_VIDEO|\ - SOUND_MASK_LINE1| SOUND_MASK_LINE|\ - SOUND_MASK_PHONEIN) - -/* original check is not good enough in case FOO is greater than - * SOUND_MIXER_NRDEVICES because the supported_mixers has exactly - * SOUND_MIXER_NRDEVICES elements. - * before matching the given mixer against the bitmask in supported_mixers we - * check if mixer number exceeds maximum allowed size which is as mentioned - * above SOUND_MIXER_NRDEVICES */ -#define supported_mixer(CODEC,FOO) ((FOO >= 0) && \ - (FOO < SOUND_MIXER_NRDEVICES) && \ - (CODEC)->supported_mixers & (1<<FOO) ) - -struct ac97_codec { - /* Linked list of codecs */ - struct list_head list; - - /* AC97 controller connected with */ - void *private_data; - - char *name; - int id; - int dev_mixer; - int type; - u32 model; - - unsigned int modem:1; - - struct ac97_ops *codec_ops; - - /* controller specific lower leverl ac97 accessing routines. - must be re-entrant safe */ - u16 (*codec_read) (struct ac97_codec *codec, u8 reg); - void (*codec_write) (struct ac97_codec *codec, u8 reg, u16 val); - - /* Wait for codec-ready. Ok to sleep here. */ - void (*codec_wait) (struct ac97_codec *codec); - - /* callback used by helper drivers for interesting ac97 setups */ - void (*codec_unregister) (struct ac97_codec *codec); - - struct ac97_driver *driver; - void *driver_private; /* Private data for the driver */ - - spinlock_t lock; - - /* OSS mixer masks */ - int modcnt; - int supported_mixers; - int stereo_mixers; - int record_sources; - - /* Property flags */ - int flags; - - int bit_resolution; - - /* OSS mixer interface */ - int (*read_mixer) (struct ac97_codec *codec, int oss_channel); - void (*write_mixer)(struct ac97_codec *codec, int oss_channel, - unsigned int left, unsigned int right); - int (*recmask_io) (struct ac97_codec *codec, int rw, int mask); - int (*mixer_ioctl)(struct ac97_codec *codec, unsigned int cmd, unsigned long arg); - - /* saved OSS mixer states */ - unsigned int mixer_state[SOUND_MIXER_NRDEVICES]; - - /* Software Modem interface */ - int (*modem_ioctl)(struct ac97_codec *codec, unsigned int cmd, unsigned long arg); -}; - -/* - * Operation structures for each known AC97 chip - */ - -struct ac97_ops -{ - /* Initialise */ - int (*init)(struct ac97_codec *c); - /* Amplifier control */ - int (*amplifier)(struct ac97_codec *codec, int on); - /* Digital mode control */ - int (*digital)(struct ac97_codec *codec, int slots, int rate, int mode); -#define AUDIO_DIGITAL 0x8000 -#define AUDIO_PRO 0x4000 -#define AUDIO_DRS 0x2000 -#define AUDIO_CCMASK 0x003F - -#define AC97_DELUDED_MODEM 1 /* Audio codec reports its a modem */ -#define AC97_NO_PCM_VOLUME 2 /* Volume control is missing */ -#define AC97_DEFAULT_POWER_OFF 4 /* Needs warm reset to power up */ -}; - -extern int ac97_probe_codec(struct ac97_codec *); - -extern struct ac97_codec *ac97_alloc_codec(void); -extern void ac97_release_codec(struct ac97_codec *codec); - -struct ac97_driver { - struct list_head list; - char *name; - u32 codec_id; - u32 codec_mask; - int (*probe) (struct ac97_codec *codec, struct ac97_driver *driver); - void (*remove) (struct ac97_codec *codec, struct ac97_driver *driver); -}; - -/* quirk types */ -enum { - AC97_TUNE_DEFAULT = -1, /* use default from quirk list (not valid in list) */ - AC97_TUNE_NONE = 0, /* nothing extra to do */ - AC97_TUNE_HP_ONLY, /* headphone (true line-out) control as master only */ - AC97_TUNE_SWAP_HP, /* swap headphone and master controls */ - AC97_TUNE_SWAP_SURROUND, /* swap master and surround controls */ - AC97_TUNE_AD_SHARING, /* for AD1985, turn on OMS bit and use headphone */ - AC97_TUNE_ALC_JACK, /* for Realtek, enable JACK detection */ -}; - -struct ac97_quirk { - unsigned short vendor; /* PCI vendor id */ - unsigned short device; /* PCI device id */ - unsigned short mask; /* device id bit mask, 0 = accept all */ - const char *name; /* name shown as info */ - int type; /* quirk type above */ -}; - -#endif /* _AC97_CODEC_H_ */ diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 56377df39124..cc0756a35ae3 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -670,6 +670,12 @@ static inline int dmaengine_resume(struct dma_chan *chan) return dmaengine_device_control(chan, DMA_RESUME, 0); } +static inline enum dma_status dmaengine_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *state) +{ + return chan->device->device_tx_status(chan, cookie, state); +} + static inline dma_cookie_t dmaengine_submit(struct dma_async_tx_descriptor *desc) { return desc->tx_submit(desc); diff --git a/include/linux/mfd/abx500/ab8500-codec.h b/include/linux/mfd/abx500/ab8500-codec.h new file mode 100644 index 000000000000..dc6529202cdd --- /dev/null +++ b/include/linux/mfd/abx500/ab8500-codec.h @@ -0,0 +1,52 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef AB8500_CORE_CODEC_H +#define AB8500_CORE_CODEC_H + +/* Mic-types */ +enum amic_type { + AMIC_TYPE_SINGLE_ENDED, + AMIC_TYPE_DIFFERENTIAL +}; + +/* Mic-biases */ +enum amic_micbias { + AMIC_MICBIAS_VAMIC1, + AMIC_MICBIAS_VAMIC2 +}; + +/* Bias-voltage */ +enum ear_cm_voltage { + EAR_CMV_0_95V, + EAR_CMV_1_10V, + EAR_CMV_1_27V, + EAR_CMV_1_58V +}; + +/* Analog microphone settings */ +struct amic_settings { + enum amic_type mic1_type; + enum amic_type mic2_type; + enum amic_micbias mic1a_micbias; + enum amic_micbias mic1b_micbias; + enum amic_micbias mic2_micbias; +}; + +/* Platform data structure for the audio-parts of the AB8500 */ +struct ab8500_codec_platform_data { + struct amic_settings amics; + enum ear_cm_voltage ear_cmv; +}; + +#endif diff --git a/include/linux/mfd/abx500/ab8500.h b/include/linux/mfd/abx500/ab8500.h index 91dd3ef63e99..bc9b84b60ec6 100644 --- a/include/linux/mfd/abx500/ab8500.h +++ b/include/linux/mfd/abx500/ab8500.h @@ -266,6 +266,7 @@ struct ab8500 { struct regulator_reg_init; struct regulator_init_data; struct ab8500_gpio_platform_data; +struct ab8500_codec_platform_data; /** * struct ab8500_platform_data - AB8500 platform data @@ -284,6 +285,7 @@ struct ab8500_platform_data { int num_regulator; struct regulator_init_data *regulator; struct ab8500_gpio_platform_data *gpio; + struct ab8500_codec_platform_data *codec; }; extern int __devinit ab8500_init(struct ab8500 *ab8500, diff --git a/include/linux/platform_data/mmp_audio.h b/include/linux/platform_data/mmp_audio.h new file mode 100644 index 000000000000..0f25d165abd6 --- /dev/null +++ b/include/linux/platform_data/mmp_audio.h @@ -0,0 +1,22 @@ +/* + * MMP Platform AUDIO Management + * + * Copyright (c) 2011 Marvell Semiconductors Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef MMP_AUDIO_H +#define MMP_AUDIO_H + +struct mmp_audio_platdata { + u32 period_max_capture; + u32 buffer_max_capture; + u32 period_max_playback; + u32 buffer_max_playback; +}; + +#endif /* MMP_AUDIO_H */ diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h new file mode 100644 index 000000000000..26f406e0f673 --- /dev/null +++ b/include/sound/designware_i2s.h @@ -0,0 +1,69 @@ +/* + * Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#ifndef __SOUND_DESIGNWARE_I2S_H +#define __SOUND_DESIGNWARE_I2S_H + +#include <linux/dmaengine.h> +#include <linux/types.h> + +/* + * struct i2s_clk_config_data - represent i2s clk configuration data + * @chan_nr: number of channel + * @data_width: number of bits per sample (8/16/24/32 bit) + * @sample_rate: sampling frequency (8Khz, 16Khz, 32Khz, 44Khz, 48Khz) + */ +struct i2s_clk_config_data { + int chan_nr; + u32 data_width; + u32 sample_rate; +}; + +struct i2s_platform_data { + #define DWC_I2S_PLAY (1 << 0) + #define DWC_I2S_RECORD (1 << 1) + unsigned int cap; + int channel; + u32 snd_fmts; + u32 snd_rates; + + void *play_dma_data; + void *capture_dma_data; + bool (*filter)(struct dma_chan *chan, void *slave); + int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); +}; + +struct i2s_dma_data { + void *data; + dma_addr_t addr; + u32 max_burst; + enum dma_slave_buswidth addr_width; + bool (*filter)(struct dma_chan *chan, void *slave); +}; + +/* I2S DMA registers */ +#define I2S_RXDMA 0x01C0 +#define I2S_TXDMA 0x01C8 + +#define TWO_CHANNEL_SUPPORT 2 /* up to 2.0 */ +#define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */ +#define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */ +#define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */ + +#endif /* __SOUND_DESIGNWARE_I2S_H */ diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index a8fcaa6d531f..b877334bbb0f 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -39,6 +39,7 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd); snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream); +snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream); int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data); diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0d1112815be3..c75c0d1a85e2 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -810,7 +810,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_list *l); + const struct snd_pcm_hw_constraint_list *l); int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, @@ -893,6 +893,7 @@ extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates; int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); +unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit); static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream, struct snd_dma_buffer *bufp) @@ -1073,4 +1074,15 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) const char *snd_pcm_format_name(snd_pcm_format_t format); +/** + * Get a string naming the direction of a stream + */ +static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + #endif /* __SOUND_PCM_H */ diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index f494f1e3c900..37ae12e0ab06 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -22,6 +22,8 @@ * */ +#include <sound/pcm.h> + int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir); diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e3833d9f1914..abe373d57adc 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -229,6 +229,10 @@ struct device; { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \ +{ .id = snd_soc_dapm_clock_supply, .name = wname, \ + .reg = SND_SOC_NOPM, .event = dapm_clock_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } /* generic widgets */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ @@ -245,6 +249,7 @@ struct device; .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -327,6 +332,8 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -367,6 +374,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm); int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); +int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route, int num); int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); @@ -432,6 +441,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ snd_soc_dapm_regulator_supply, /* external regulator */ + snd_soc_dapm_clock_supply, /* external clock */ snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ @@ -537,6 +547,8 @@ struct snd_soc_dapm_widget { struct list_head dirty; int inputs; int outputs; + + struct clk *clk; }; struct snd_soc_dapm_update { diff --git a/include/sound/soc.h b/include/sound/soc.h index c703871f5f65..e063380f63a2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -42,11 +42,22 @@ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ .max = xmax, .platform_max = xmax, .invert = xinvert}) +#define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ + .min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ + .put = snd_soc_put_volsw_range, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = xshift, .min = xmin,\ + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -67,6 +78,16 @@ {.reg = xreg, .rreg = xreg, \ .shift = xshift, .rshift = xshift, \ .max = xmax, .min = xmin} } +#define SOC_SINGLE_RANGE_TLV(xname, xreg, xshift, xmin, xmax, xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = xshift, .min = xmin,\ + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ @@ -79,6 +100,13 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -97,6 +125,16 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE_TLV(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -460,6 +498,12 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, @@ -785,13 +829,36 @@ struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ const char *stream_name; /* Stream name */ - const char *codec_name; /* for multi-codec */ - const struct device_node *codec_of_node; - const char *platform_name; /* for multi-platform */ - const struct device_node *platform_of_node; + /* + * You MAY specify the link's CPU-side device, either by device name, + * or by DT/OF node, but not both. If this information is omitted, + * the CPU-side DAI is matched using .cpu_dai_name only, which hence + * must be globally unique. These fields are currently typically used + * only for codec to codec links, or systems using device tree. + */ + const char *cpu_name; + const struct device_node *cpu_of_node; + /* + * You MAY specify the DAI name of the CPU DAI. If this information is + * omitted, the CPU-side DAI is matched using .cpu_name/.cpu_of_node + * only, which only works well when that device exposes a single DAI. + */ const char *cpu_dai_name; - const struct device_node *cpu_dai_of_node; + /* + * You MUST specify the link's codec, either by device name, or by + * DT/OF node, but not both. + */ + const char *codec_name; + const struct device_node *codec_of_node; + /* You MUST specify the DAI name within the codec */ const char *codec_dai_name; + /* + * You MAY specify the link's platform/PCM/DMA driver, either by + * device name, or by DT/OF node, but not both. Some forms of link + * do not need a platform. + */ + const char *platform_name; + const struct device_node *platform_of_node; int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h new file mode 100644 index 000000000000..1b365bfdfb37 --- /dev/null +++ b/include/sound/spear_dma.h @@ -0,0 +1,35 @@ +/* +* linux/spear_dma.h +* +* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com) +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License as published by +* the Free Software Foundation; either version 2 of the License, or +* (at your option) any later version. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +*/ + +#ifndef SPEAR_DMA_H +#define SPEAR_DMA_H + +#include <linux/dmaengine.h> + +struct spear_dma_data { + void *data; + dma_addr_t addr; + u32 max_burst; + enum dma_slave_buswidth addr_width; + bool (*filter)(struct dma_chan *chan, void *slave); +}; + +#endif /* SPEAR_DMA_H */ diff --git a/include/sound/spear_spdif.h b/include/sound/spear_spdif.h new file mode 100644 index 000000000000..a12f39695610 --- /dev/null +++ b/include/sound/spear_spdif.h @@ -0,0 +1,29 @@ +/* + * Copyright (ST) 2012 Vipin Kumar (vipin.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __SOUND_SPDIF_H +#define __SOUND_SPDIF_H + +struct spear_spdif_platform_data { + /* DMA params */ + void *dma_params; + bool (*filter)(struct dma_chan *chan, void *slave); + void (*reset_perip)(void); +}; + +#endif /* SOUND_SPDIF_H */ diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 7067e2dfb0b9..a64d8fe3f855 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -38,21 +38,31 @@ #define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ #define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ +#define TLV_ITEM(type, ...) \ + (type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__ +#define TLV_LENGTH(...) \ + ((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ })) + +#define TLV_CONTAINER_ITEM(...) \ + TLV_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__) +#define DECLARE_TLV_CONTAINER(name, ...) \ + unsigned int name[] = { TLV_CONTAINER_ITEM(__VA_ARGS__) } + #define TLV_DB_SCALE_MASK 0xffff #define TLV_DB_SCALE_MUTE 0x10000 #define TLV_DB_SCALE_ITEM(min, step, mute) \ - SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \ - (min), ((step) & TLV_DB_SCALE_MASK) | ((mute) ? TLV_DB_SCALE_MUTE : 0) + TLV_ITEM(SNDRV_CTL_TLVT_DB_SCALE, \ + (min), \ + ((step) & TLV_DB_SCALE_MASK) | \ + ((mute) ? TLV_DB_SCALE_MUTE : 0)) #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) } /* dB scale specified with min/max values instead of step */ #define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \ - SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int), \ - (min_dB), (max_dB) + TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB)) #define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \ - SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int), \ - (min_dB), (max_dB) + TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX_MUTE, (min_dB), (max_dB)) #define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \ unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) } #define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \ @@ -60,13 +70,16 @@ /* linear volume between min_dB and max_dB (.01dB unit) */ #define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \ - SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \ - (min_dB), (max_dB) + TLV_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB)) #define DECLARE_TLV_DB_LINEAR(name, min_dB, max_dB) \ unsigned int name[] = { TLV_DB_LINEAR_ITEM(min_dB, max_dB) } /* dB range container */ /* Each item is: <min> <max> <TLV> */ +#define TLV_DB_RANGE_ITEM(...) \ + TLV_ITEM(SNDRV_CTL_TLVT_DB_RANGE, __VA_ARGS__) +#define DECLARE_TLV_DB_RANGE(name, ...) \ + unsigned int name[] = { TLV_DB_RANGE_ITEM(__VA_ARGS__) } /* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */ #define TLV_DB_RANGE_HEAD(num) \ SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int) diff --git a/include/sound/vx_core.h b/include/sound/vx_core.h index 5456343ebe4c..4f67c762cd74 100644 --- a/include/sound/vx_core.h +++ b/include/sound/vx_core.h @@ -341,7 +341,7 @@ int vx_change_frequency(struct vx_core *chip); /* * PM */ -int snd_vx_suspend(struct vx_core *card, pm_message_t state); +int snd_vx_suspend(struct vx_core *card); int snd_vx_resume(struct vx_core *card); /* diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index afef72c4f0d3..0d7b25e81643 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -108,7 +108,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { #ifdef CONFIG_PM -static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state) +static int pxa2xx_ac97_do_suspend(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; @@ -144,7 +144,7 @@ static int pxa2xx_ac97_suspend(struct device *dev) int ret = 0; if (card) - ret = pxa2xx_ac97_do_suspend(card, PMSG_SUSPEND); + ret = pxa2xx_ac97_do_suspend(card); return ret; } @@ -160,10 +160,7 @@ static int pxa2xx_ac97_resume(struct device *dev) return ret; } -static const struct dev_pm_ops pxa2xx_ac97_pm_ops = { - .suspend = pxa2xx_ac97_suspend, - .resume = pxa2xx_ac97_resume, -}; +static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); #endif static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index f7c2bb08055d..eb4ceb71123e 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -535,9 +535,9 @@ out_put_pclk: } #ifdef CONFIG_PM -static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg) +static int atmel_abdac_suspend(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct atmel_abdac *dac = card->private_data; dw_dma_cyclic_stop(dac->dma.chan); @@ -547,9 +547,9 @@ static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg) return 0; } -static int atmel_abdac_resume(struct platform_device *pdev) +static int atmel_abdac_resume(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct atmel_abdac *dac = card->private_data; clk_enable(dac->pclk); @@ -559,9 +559,11 @@ static int atmel_abdac_resume(struct platform_device *pdev) return 0; } + +static SIMPLE_DEV_PM_OPS(atmel_abdac_pm, atmel_abdac_suspend, atmel_abdac_resume); +#define ATMEL_ABDAC_PM_OPS &atmel_abdac_pm #else -#define atmel_abdac_suspend NULL -#define atmel_abdac_resume NULL +#define ATMEL_ABDAC_PM_OPS NULL #endif static int __devexit atmel_abdac_remove(struct platform_device *pdev) @@ -589,9 +591,9 @@ static struct platform_driver atmel_abdac_driver = { .remove = __devexit_p(atmel_abdac_remove), .driver = { .name = "atmel_abdac", + .owner = THIS_MODULE, + .pm = ATMEL_ABDAC_PM_OPS, }, - .suspend = atmel_abdac_suspend, - .resume = atmel_abdac_resume, }; static int __init atmel_abdac_init(void) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index f5ded640b395..bf47025bdf45 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1135,9 +1135,9 @@ err_snd_card_new: } #ifdef CONFIG_PM -static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) +static int atmel_ac97c_suspend(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; if (cpu_is_at32ap7000()) { @@ -1151,9 +1151,9 @@ static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) return 0; } -static int atmel_ac97c_resume(struct platform_device *pdev) +static int atmel_ac97c_resume(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; clk_enable(chip->pclk); @@ -1165,9 +1165,11 @@ static int atmel_ac97c_resume(struct platform_device *pdev) } return 0; } + +static SIMPLE_DEV_PM_OPS(atmel_ac97c_pm, atmel_ac97c_suspend, atmel_ac97c_resume); +#define ATMEL_AC97C_PM_OPS &atmel_ac97c_pm #else -#define atmel_ac97c_suspend NULL -#define atmel_ac97c_resume NULL +#define ATMEL_AC97C_PM_OPS NULL #endif static int __devexit atmel_ac97c_remove(struct platform_device *pdev) @@ -1210,9 +1212,9 @@ static struct platform_driver atmel_ac97c_driver = { .remove = __devexit_p(atmel_ac97c_remove), .driver = { .name = "atmel_ac97c", + .owner = THIS_MODULE, + .pm = ATMEL_AC97C_PM_OPS, }, - .suspend = atmel_ac97c_suspend, - .resume = atmel_ac97c_resume, }; static int __init atmel_ac97c_init(void) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8f312fa6c282..7ae671923393 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1250,10 +1250,10 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_list *l) + const struct snd_pcm_hw_constraint_list *l) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_list, l, + snd_pcm_hw_rule_list, (void *)l, var, -1); } diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9c9eff9afbac..d4fc1bfbe457 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -488,3 +488,21 @@ unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate) return SNDRV_PCM_RATE_KNOT; } EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit); + +/** + * snd_pcm_rate_bit_to_rate - converts SNDRV_PCM_RATE_xxx bit to sample rate + * @rate_bit: the rate bit to convert + * + * Returns the sample rate that corresponds to the given SNDRV_PCM_RATE_xxx flag + * or 0 for an unknown rate bit + */ +unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit) +{ + unsigned int i; + + for (i = 0; i < snd_pcm_known_rates.count; i++) + if ((1u << i) == rate_bit) + return snd_pcm_known_rates.list[i]; + return 0; +} +EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate); diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 8b5c36f4d303..1128b35b2b05 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1177,10 +1177,9 @@ static int __devexit loopback_remove(struct platform_device *devptr) } #ifdef CONFIG_PM -static int loopback_suspend(struct platform_device *pdev, - pm_message_t state) +static int loopback_suspend(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct loopback *loopback = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -1190,13 +1189,18 @@ static int loopback_suspend(struct platform_device *pdev, return 0; } -static int loopback_resume(struct platform_device *pdev) +static int loopback_resume(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(loopback_pm, loopback_suspend, loopback_resume); +#define LOOPBACK_PM_OPS &loopback_pm +#else +#define LOOPBACK_PM_OPS NULL #endif #define SND_LOOPBACK_DRIVER "snd_aloop" @@ -1204,12 +1208,10 @@ static int loopback_resume(struct platform_device *pdev) static struct platform_driver loopback_driver = { .probe = loopback_probe, .remove = __devexit_p(loopback_remove), -#ifdef CONFIG_PM - .suspend = loopback_suspend, - .resume = loopback_resume, -#endif .driver = { - .name = SND_LOOPBACK_DRIVER + .name = SND_LOOPBACK_DRIVER, + .owner = THIS_MODULE, + .pm = LOOPBACK_PM_OPS, }, }; diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index ad9434fd6370..f7d3bfc6bca8 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1065,9 +1065,9 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr) } #ifdef CONFIG_PM -static int snd_dummy_suspend(struct platform_device *pdev, pm_message_t state) +static int snd_dummy_suspend(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct snd_dummy *dummy = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -1075,13 +1075,18 @@ static int snd_dummy_suspend(struct platform_device *pdev, pm_message_t state) return 0; } -static int snd_dummy_resume(struct platform_device *pdev) +static int snd_dummy_resume(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_dummy_pm, snd_dummy_suspend, snd_dummy_resume); +#define SND_DUMMY_PM_OPS &snd_dummy_pm +#else +#define SND_DUMMY_PM_OPS NULL #endif #define SND_DUMMY_DRIVER "snd_dummy" @@ -1089,12 +1094,10 @@ static int snd_dummy_resume(struct platform_device *pdev) static struct platform_driver snd_dummy_driver = { .probe = snd_dummy_probe, .remove = __devexit_p(snd_dummy_remove), -#ifdef CONFIG_PM - .suspend = snd_dummy_suspend, - .resume = snd_dummy_resume, -#endif .driver = { - .name = SND_DUMMY_DRIVER + .name = SND_DUMMY_DRIVER, + .owner = THIS_MODULE, + .pm = SND_DUMMY_PM_OPS, }, }; diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 86f5fbc2da72..bc03a2046c9c 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -139,7 +139,8 @@ static struct platform_driver snd_mpu401_driver = { .probe = snd_mpu401_probe, .remove = __devexit_p(snd_mpu401_remove), .driver = { - .name = SND_MPU401_DRIVER + .name = SND_MPU401_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 76930793fb69..cad73af3860c 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -759,7 +759,8 @@ static struct platform_driver snd_mtpav_driver = { .probe = snd_mtpav_probe, .remove = __devexit_p(snd_mtpav_remove), .driver = { - .name = SND_MTPAV_DRIVER + .name = SND_MTPAV_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 621e60e2029f..2d5514b0a290 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -1040,7 +1040,8 @@ static struct platform_driver snd_mts64_driver = { .probe = snd_mts64_probe, .remove = __devexit_p(snd_mts64_remove), .driver = { - .name = PLATFORM_DRIVER + .name = PLATFORM_DRIVER, + .owner = THIS_MODULE, } }; diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 99704e6a2e26..6ca59fc6dcb9 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -200,15 +200,18 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) } #ifdef CONFIG_PM -static int pcsp_suspend(struct platform_device *dev, pm_message_t state) +static int pcsp_suspend(struct device *dev) { - struct snd_pcsp *chip = platform_get_drvdata(dev); + struct snd_pcsp *chip = dev_get_drvdata(dev); pcsp_stop_beep(chip); snd_pcm_suspend_all(chip->pcm); return 0; } + +static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL); +#define PCSP_PM_OPS &pcsp_pm #else -#define pcsp_suspend NULL +#define PCSP_PM_OPS NULL #endif /* CONFIG_PM */ static void pcsp_shutdown(struct platform_device *dev) @@ -221,10 +224,10 @@ static struct platform_driver pcsp_platform_driver = { .driver = { .name = "pcspkr", .owner = THIS_MODULE, + .pm = PCSP_PM_OPS, }, .probe = pcsp_probe, .remove = __devexit_p(pcsp_remove), - .suspend = pcsp_suspend, .shutdown = pcsp_shutdown, }; diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 3e32bd3d95d9..8364855ed14f 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -829,7 +829,8 @@ static struct platform_driver snd_portman_driver = { .probe = snd_portman_probe, .remove = __devexit_p(snd_portman_remove), .driver = { - .name = PLATFORM_DRIVER + .name = PLATFORM_DRIVER, + .owner = THIS_MODULE, } }; diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index b2d0e8e49bed..86700671d1ac 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -995,7 +995,8 @@ static struct platform_driver snd_serial_driver = { .probe = snd_serial_probe, .remove = __devexit_p( snd_serial_remove), .driver = { - .name = SND_SERIAL_DRIVER + .name = SND_SERIAL_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 9d97478a18b3..d7d514df9058 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -142,7 +142,8 @@ static struct platform_driver snd_virmidi_driver = { .probe = snd_virmidi_probe, .remove = __devexit_p(snd_virmidi_remove), .driver = { - .name = SND_VIRMIDI_DRIVER + .name = SND_VIRMIDI_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index b8e515999bc2..de5055a3b0d0 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -725,7 +725,7 @@ EXPORT_SYMBOL(snd_vx_dsp_load); /* * suspend */ -int snd_vx_suspend(struct vx_core *chip, pm_message_t state) +int snd_vx_suspend(struct vx_core *chip) { unsigned int i; diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d7ccf28bd66a..f8fbe22515c9 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -135,10 +135,9 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; - unsigned long mc_indir_size; struct resource *res_mc_indir; - struct snd_wss *codec; #endif /* OPTi93X */ + struct snd_wss *codec; unsigned long pwd_reg; spinlock_t lock; @@ -245,10 +244,8 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d; - if (!chip->mc_indir_index) { + if (!chip->mc_indir_index) chip->mc_indir_index = 0xe0e; - chip->mc_indir_size = 2; - } chip->password = 0xe4; chip->pwd_reg = 0; break; @@ -351,7 +348,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) -static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, +static int snd_opti9xx_configure(struct snd_opti9xx *chip, long port, int irq, int dma1, int dma2, long mpu_port, int mpu_irq) @@ -403,7 +400,9 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, #else /* OPTi93X */ case OPTi9XX_HW_82C931: - case OPTi9XX_HW_82C933: + /* disable 3D sound (set GPIO1 as output, low) */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c); + case OPTi9XX_HW_82C933: /* FALL THROUGH */ /* * The BTC 1817DW has QS1000 wavetable which is connected * to the serial digital input of the OPTI931. @@ -696,8 +695,7 @@ static int __devinit snd_opti9xx_read_check(struct snd_opti9xx *chip) if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) return 0; #else /* OPTi93X */ - chip->res_mc_indir = request_region(chip->mc_indir_index, - chip->mc_indir_size, + chip->res_mc_indir = request_region(chip->mc_indir_index, 2, "OPTi93x MC"); if (chip->res_mc_indir == NULL) return -EBUSY; @@ -770,8 +768,9 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, #ifdef OPTi93X port = pnp_port_start(pdev, 0) - 4; fm_port = pnp_port_start(pdev, 1) + 8; - chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; - chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; + /* adjust mc_indir_index - some cards report it at 0xe?d, + other at 0xe?c but it really is always at 0xe?e */ + chip->mc_indir_index = (pnp_port_start(pdev, 3) & ~0xf) | 0xe; #else devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); if (devmc == NULL) @@ -871,9 +870,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) &codec); if (error < 0) return error; -#ifdef OPTi93X chip->codec = codec; -#endif error = snd_wss_pcm(codec, 0, &pcm); if (error < 0) return error; @@ -1054,11 +1051,55 @@ static int __devexit snd_opti9xx_isa_remove(struct device *devptr, return 0; } +#ifdef CONFIG_PM +static int snd_opti9xx_suspend(struct snd_card *card) +{ + struct snd_opti9xx *chip = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + chip->codec->suspend(chip->codec); + return 0; +} + +static int snd_opti9xx_resume(struct snd_card *card) +{ + struct snd_opti9xx *chip = card->private_data; + int error, xdma2; +#if defined(CS4231) || defined(OPTi93X) + xdma2 = dma2; +#else + xdma2 = -1; +#endif + + error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2, + mpu_port, mpu_irq); + if (error) + return error; + chip->codec->resume(chip->codec); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static int snd_opti9xx_isa_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + return snd_opti9xx_suspend(dev_get_drvdata(dev)); +} + +static int snd_opti9xx_isa_resume(struct device *dev, unsigned int n) +{ + return snd_opti9xx_resume(dev_get_drvdata(dev)); +} +#endif + static struct isa_driver snd_opti9xx_driver = { .match = snd_opti9xx_isa_match, .probe = snd_opti9xx_isa_probe, .remove = __devexit_p(snd_opti9xx_isa_remove), - /* FIXME: suspend/resume */ +#ifdef CONFIG_PM + .suspend = snd_opti9xx_isa_suspend, + .resume = snd_opti9xx_isa_resume, +#endif .driver = { .name = DEV_NAME }, @@ -1124,12 +1165,29 @@ static void __devexit snd_opti9xx_pnp_remove(struct pnp_card_link * pcard) snd_opti9xx_pnp_is_probed = 0; } +#ifdef CONFIG_PM +static int snd_opti9xx_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + return snd_opti9xx_suspend(pnp_get_card_drvdata(pcard)); +} + +static int snd_opti9xx_pnp_resume(struct pnp_card_link *pcard) +{ + return snd_opti9xx_resume(pnp_get_card_drvdata(pcard)); +} +#endif + static struct pnp_card_driver opti9xx_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, .name = "opti9xx", .id_table = snd_opti9xx_pnpids, .probe = snd_opti9xx_pnp_probe, .remove = __devexit_p(snd_opti9xx_pnp_remove), +#ifdef CONFIG_PM + .suspend = snd_opti9xx_pnp_suspend, + .resume = snd_opti9xx_pnp_resume, +#endif }; #endif diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 49c8a0c2442c..360b08b03e1d 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1456,7 +1456,6 @@ static struct snd_pcm_hardware snd_wss_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_SYNC_START), .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), @@ -1657,6 +1656,10 @@ static void snd_wss_resume(struct snd_wss *chip) break; } } + /* Yamaha needs this to resume properly */ + if (chip->hardware == WSS_HW_OPL3SA2) + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, + chip->image[CS4231_PLAYBK_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); #if 1 snd_wss_mce_down(chip); diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 09d46484bc1a..7d8803a00b79 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -69,7 +69,6 @@ #include <linux/sound.h> #include <linux/slab.h> #include <linux/soundcard.h> -#include <linux/ac97_codec.h> #include <linux/pci.h> #include <linux/bitops.h> #include <linux/interrupt.h> @@ -199,6 +198,22 @@ static const char invalid_magic[] = } \ }) +/* AC97 registers */ +#define AC97_MASTER_VOL_STEREO 0x0002 /* Line Out */ +#define AC97_PCBEEP_VOL 0x000a /* none */ +#define AC97_PHONE_VOL 0x000c /* TAD Input (mono) */ +#define AC97_MIC_VOL 0x000e /* MIC Input (mono) */ +#define AC97_LINEIN_VOL 0x0010 /* Line Input (stereo) */ +#define AC97_CD_VOL 0x0012 /* CD Input (stereo) */ +#define AC97_AUX_VOL 0x0016 /* Aux Input (stereo) */ +#define AC97_PCMOUT_VOL 0x0018 /* Wave Output (stereo) */ +#define AC97_RECORD_SELECT 0x001a /* */ +#define AC97_RECORD_GAIN 0x001c +#define AC97_GENERAL_PURPOSE 0x0020 +#define AC97_3D_CONTROL 0x0022 +#define AC97_POWER_CONTROL 0x0026 +#define AC97_VENDOR_ID1 0x007c + struct list_head cs4297a_devs = { &cs4297a_devs, &cs4297a_devs }; typedef struct serdma_descr_s { diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 9dfc27bf6cc6..ee895f3c8605 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1884,9 +1884,10 @@ static int __devinit snd_ali_mixer(struct snd_ali * codec) } #ifdef CONFIG_PM -static int ali_suspend(struct pci_dev *pci, pm_message_t state) +static int ali_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ali *chip = card->private_data; struct snd_ali_image *im; int i, j; @@ -1929,13 +1930,14 @@ static int ali_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int ali_resume(struct pci_dev *pci) +static int ali_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ali *chip = card->private_data; struct snd_ali_image *im; int i, j; @@ -1982,6 +1984,11 @@ static int ali_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(ali_pm, ali_suspend, ali_resume); +#define ALI_PM_OPS &ali_pm +#else +#define ALI_PM_OPS NULL #endif /* CONFIG_PM */ static int snd_ali_free(struct snd_ali * codec) @@ -2299,10 +2306,9 @@ static struct pci_driver ali5451_driver = { .id_table = snd_ali_ids, .probe = snd_ali_probe, .remove = __devexit_p(snd_ali_remove), -#ifdef CONFIG_PM - .suspend = ali_suspend, - .resume = ali_resume, -#endif + .driver = { + .pm = ALI_PM_OPS, + }, }; module_pci_driver(ali5451_driver); diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 59d65388faf5..68c4469c6d19 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -766,9 +766,10 @@ static int __devinit snd_als300_create(struct snd_card *card, } #ifdef CONFIG_PM -static int snd_als300_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_als300_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_als300 *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -777,13 +778,14 @@ static int snd_als300_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_als300_resume(struct pci_dev *pci) +static int snd_als300_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_als300 *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -802,6 +804,11 @@ static int snd_als300_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_als300_pm, snd_als300_suspend, snd_als300_resume); +#define SND_ALS300_PM_OPS &snd_als300_pm +#else +#define SND_ALS300_PM_OPS NULL #endif static int __devinit snd_als300_probe(struct pci_dev *pci, @@ -857,10 +864,9 @@ static struct pci_driver als300_driver = { .id_table = snd_als300_ids, .probe = snd_als300_probe, .remove = __devexit_p(snd_als300_remove), -#ifdef CONFIG_PM - .suspend = snd_als300_suspend, - .resume = snd_als300_resume, -#endif + .driver = { + .pm = SND_ALS300_PM_OPS, + }, }; module_pci_driver(als300_driver); diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 7d7f2598c748..0eeca49c5754 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -988,9 +988,10 @@ static void __devexit snd_card_als4000_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int snd_als4000_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_als4000_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_card_als4000 *acard = card->private_data; struct snd_sb *chip = acard->chip; @@ -1001,13 +1002,14 @@ static int snd_als4000_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_als4000_resume(struct pci_dev *pci) +static int snd_als4000_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_card_als4000 *acard = card->private_data; struct snd_sb *chip = acard->chip; @@ -1033,18 +1035,21 @@ static int snd_als4000_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif /* CONFIG_PM */ +static SIMPLE_DEV_PM_OPS(snd_als4000_pm, snd_als4000_suspend, snd_als4000_resume); +#define SND_ALS4000_PM_OPS &snd_als4000_pm +#else +#define SND_ALS4000_PM_OPS NULL +#endif /* CONFIG_PM */ static struct pci_driver als4000_driver = { .name = KBUILD_MODNAME, .id_table = snd_als4000_ids, .probe = snd_card_als4000_probe, .remove = __devexit_p(snd_card_als4000_remove), -#ifdef CONFIG_PM - .suspend = snd_als4000_suspend, - .resume = snd_als4000_resume, -#endif + .driver = { + .pm = SND_ALS4000_PM_OPS, + }, }; module_pci_driver(als4000_driver); diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 156a94f8a123..31020d2a868b 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1462,9 +1462,10 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp *chip, int clock, /* * power management */ -static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_atiixp_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct atiixp *chip = card->private_data; int i; @@ -1484,13 +1485,14 @@ static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_atiixp_resume(struct pci_dev *pci) +static int snd_atiixp_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct atiixp *chip = card->private_data; int i; @@ -1526,6 +1528,11 @@ static int snd_atiixp_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); +#define SND_ATIIXP_PM_OPS &snd_atiixp_pm +#else +#define SND_ATIIXP_PM_OPS NULL #endif /* CONFIG_PM */ @@ -1705,10 +1712,9 @@ static struct pci_driver atiixp_driver = { .id_table = snd_atiixp_ids, .probe = snd_atiixp_probe, .remove = __devexit_p(snd_atiixp_remove), -#ifdef CONFIG_PM - .suspend = snd_atiixp_suspend, - .resume = snd_atiixp_resume, -#endif + .driver = { + .pm = SND_ATIIXP_PM_OPS, + }, }; module_pci_driver(atiixp_driver); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 30a4fd96ce73..79e204ec623f 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1117,9 +1117,10 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp_modem *chip, int clock) /* * power management */ -static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_atiixp_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct atiixp_modem *chip = card->private_data; int i; @@ -1133,13 +1134,14 @@ static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_atiixp_resume(struct pci_dev *pci) +static int snd_atiixp_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct atiixp_modem *chip = card->private_data; int i; @@ -1162,8 +1164,12 @@ static int snd_atiixp_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif /* CONFIG_PM */ +static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); +#define SND_ATIIXP_PM_OPS &snd_atiixp_pm +#else +#define SND_ATIIXP_PM_OPS NULL +#endif /* CONFIG_PM */ #ifdef CONFIG_PROC_FS /* @@ -1336,10 +1342,9 @@ static struct pci_driver atiixp_modem_driver = { .id_table = snd_atiixp_ids, .probe = snd_atiixp_probe, .remove = __devexit_p(snd_atiixp_remove), -#ifdef CONFIG_PM - .suspend = snd_atiixp_suspend, - .resume = snd_atiixp_resume, -#endif + .driver = { + .pm = SND_ATIIXP_PM_OPS, + }, }; module_pci_driver(atiixp_modem_driver); diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c index 557c782ae4fc..fa13efbebdaf 100644 --- a/sound/pci/au88x0/au88x0_mixer.c +++ b/sound/pci/au88x0/au88x0_mixer.c @@ -10,6 +10,15 @@ #include <sound/core.h> #include "au88x0.h" +static int remove_ctl(struct snd_card *card, const char *name) +{ + struct snd_ctl_elem_id id; + memset(&id, 0, sizeof(id)); + strcpy(id.name, name); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + return snd_ctl_remove_id(card, &id); +} + static int __devinit snd_vortex_mixer(vortex_t * vortex) { struct snd_ac97_bus *pbus; @@ -28,5 +37,7 @@ static int __devinit snd_vortex_mixer(vortex_t * vortex) ac97.scaps = AC97_SCAP_NO_SPDIF; err = snd_ac97_mixer(pbus, &ac97, &vortex->codec); vortex->isquad = ((vortex->codec == NULL) ? 0 : (vortex->codec->ext_id&0x80)); + remove_ctl(vortex->card, "Master Mono Playback Volume"); + remove_ctl(vortex->card, "Master Mono Playback Switch"); return err; } diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index f0b4d7493af5..4dddd871548b 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2794,9 +2794,10 @@ snd_azf3328_resume_ac97(const struct snd_azf3328 *chip) } static int -snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) +snd_azf3328_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_azf3328 *chip = card->private_data; u16 *saved_regs_ctrl_u16; @@ -2824,14 +2825,15 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } static int -snd_azf3328_resume(struct pci_dev *pci) +snd_azf3328_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); const struct snd_azf3328 *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -2859,18 +2861,21 @@ snd_azf3328_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif /* CONFIG_PM */ +static SIMPLE_DEV_PM_OPS(snd_azf3328_pm, snd_azf3328_suspend, snd_azf3328_resume); +#define SND_AZF3328_PM_OPS &snd_azf3328_pm +#else +#define SND_AZF3328_PM_OPS NULL +#endif /* CONFIG_PM */ static struct pci_driver azf3328_driver = { .name = KBUILD_MODNAME, .id_table = snd_azf3328_ids, .probe = snd_azf3328_probe, .remove = __devexit_p(snd_azf3328_remove), -#ifdef CONFIG_PM - .suspend = snd_azf3328_suspend, - .resume = snd_azf3328_resume, -#endif + .driver = { + .pm = SND_AZF3328_PM_OPS, + }, }; module_pci_driver(azf3328_driver); diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index e76d68a7081f..83277b747b36 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1872,9 +1872,10 @@ static void __devexit snd_ca0106_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_ca0106_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ca0106 *chip = card->private_data; int i; @@ -1889,13 +1890,14 @@ static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_ca0106_resume(struct pci_dev *pci) +static int snd_ca0106_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ca0106 *chip = card->private_data; int i; @@ -1922,6 +1924,11 @@ static int snd_ca0106_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_ca0106_pm, snd_ca0106_suspend, snd_ca0106_resume); +#define SND_CA0106_PM_OPS &snd_ca0106_pm +#else +#define SND_CA0106_PM_OPS NULL #endif // PCI IDs @@ -1937,10 +1944,9 @@ static struct pci_driver ca0106_driver = { .id_table = snd_ca0106_ids, .probe = snd_ca0106_probe, .remove = __devexit_p(snd_ca0106_remove), -#ifdef CONFIG_PM - .suspend = snd_ca0106_suspend, - .resume = snd_ca0106_resume, -#endif + .driver = { + .pm = SND_CA0106_PM_OPS, + }, }; module_pci_driver(ca0106_driver); diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 3815bd4c6779..b7d6f2b886ef 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3338,9 +3338,10 @@ static unsigned char saved_mixers[] = { SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, }; -static int snd_cmipci_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_cmipci_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cmipci *cm = card->private_data; int i; @@ -3361,13 +3362,14 @@ static int snd_cmipci_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_cmipci_resume(struct pci_dev *pci) +static int snd_cmipci_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cmipci *cm = card->private_data; int i; @@ -3396,6 +3398,11 @@ static int snd_cmipci_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_cmipci_pm, snd_cmipci_suspend, snd_cmipci_resume); +#define SND_CMIPCI_PM_OPS &snd_cmipci_pm +#else +#define SND_CMIPCI_PM_OPS NULL #endif /* CONFIG_PM */ static struct pci_driver cmipci_driver = { @@ -3403,10 +3410,9 @@ static struct pci_driver cmipci_driver = { .id_table = snd_cmipci_ids, .probe = snd_cmipci_probe, .remove = __devexit_p(snd_cmipci_remove), -#ifdef CONFIG_PM - .suspend = snd_cmipci_suspend, - .resume = snd_cmipci_resume, -#endif + .driver = { + .pm = SND_CMIPCI_PM_OPS, + }, }; module_pci_driver(cmipci_driver); diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 33506ee569bd..45a8317085f4 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1997,9 +1997,10 @@ static int saved_regs[SUSPEND_REGISTERS] = { #define CLKCR1_CKRA 0x00010000L -static int cs4281_suspend(struct pci_dev *pci, pm_message_t state) +static int cs4281_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cs4281 *chip = card->private_data; u32 ulCLK; unsigned int i; @@ -2040,13 +2041,14 @@ static int cs4281_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int cs4281_resume(struct pci_dev *pci) +static int cs4281_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cs4281 *chip = card->private_data; unsigned int i; u32 ulCLK; @@ -2082,6 +2084,11 @@ static int cs4281_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(cs4281_pm, cs4281_suspend, cs4281_resume); +#define CS4281_PM_OPS &cs4281_pm +#else +#define CS4281_PM_OPS NULL #endif /* CONFIG_PM */ static struct pci_driver cs4281_driver = { @@ -2089,10 +2096,9 @@ static struct pci_driver cs4281_driver = { .id_table = snd_cs4281_ids, .probe = snd_cs4281_probe, .remove = __devexit_p(snd_cs4281_remove), -#ifdef CONFIG_PM - .suspend = cs4281_suspend, - .resume = cs4281_resume, -#endif + .driver = { + .pm = CS4281_PM_OPS, + }, }; module_pci_driver(cs4281_driver); diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 6cc7404e0e8f..1e007c736a8b 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -30,7 +30,7 @@ #include <linux/init.h> #include <linux/module.h> #include <sound/core.h> -#include <sound/cs46xx.h> +#include "cs46xx.h" #include <sound/initval.h> MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); @@ -167,8 +167,9 @@ static struct pci_driver cs46xx_driver = { .probe = snd_card_cs46xx_probe, .remove = __devexit_p(snd_card_cs46xx_remove), #ifdef CONFIG_PM - .suspend = snd_cs46xx_suspend, - .resume = snd_cs46xx_resume, + .driver = { + .pm = &snd_cs46xx_pm, + }, #endif }; diff --git a/include/sound/cs46xx.h b/sound/pci/cs46xx/cs46xx.h index e3005a674a24..29d8a8da1ba7 100644 --- a/include/sound/cs46xx.h +++ b/sound/pci/cs46xx/cs46xx.h @@ -23,10 +23,10 @@ * */ -#include "pcm.h" -#include "pcm-indirect.h" -#include "rawmidi.h" -#include "ac97_codec.h" +#include <sound/pcm.h> +#include <sound/pcm-indirect.h> +#include <sound/rawmidi.h> +#include <sound/ac97_codec.h> #include "cs46xx_dsp_spos.h" /* @@ -1730,8 +1730,7 @@ int snd_cs46xx_create(struct snd_card *card, struct pci_dev *pci, int external_amp, int thinkpad, struct snd_cs46xx **rcodec); -int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state); -int snd_cs46xx_resume(struct pci_dev *pci); +extern const struct dev_pm_ops snd_cs46xx_pm; int snd_cs46xx_pcm(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm); int snd_cs46xx_pcm_rear(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm); diff --git a/include/sound/cs46xx_dsp_scb_types.h b/sound/pci/cs46xx/cs46xx_dsp_scb_types.h index 080857ad0ca2..080857ad0ca2 100644 --- a/include/sound/cs46xx_dsp_scb_types.h +++ b/sound/pci/cs46xx/cs46xx_dsp_scb_types.h diff --git a/include/sound/cs46xx_dsp_spos.h b/sound/pci/cs46xx/cs46xx_dsp_spos.h index 8008c59288a6..8008c59288a6 100644 --- a/include/sound/cs46xx_dsp_spos.h +++ b/sound/pci/cs46xx/cs46xx_dsp_spos.h diff --git a/include/sound/cs46xx_dsp_task_types.h b/sound/pci/cs46xx/cs46xx_dsp_task_types.h index 5cf920bfda27..5cf920bfda27 100644 --- a/include/sound/cs46xx_dsp_task_types.h +++ b/sound/pci/cs46xx/cs46xx_dsp_task_types.h diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 4fa53161b094..f75f5ffdfdfb 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -61,7 +61,7 @@ #include <sound/info.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -#include <sound/cs46xx.h> +#include "cs46xx.h" #include <asm/io.h> @@ -3599,9 +3599,10 @@ static unsigned int saved_regs[] = { BA1_CVOL, }; -int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_cs46xx_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_cs46xx *chip = card->private_data; int i, amp_saved; @@ -3628,13 +3629,14 @@ int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -int snd_cs46xx_resume(struct pci_dev *pci) +static int snd_cs46xx_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_cs46xx *chip = card->private_data; int amp_saved; #ifdef CONFIG_SND_CS46XX_NEW_DSP @@ -3707,6 +3709,8 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +SIMPLE_DEV_PM_OPS(snd_cs46xx_pm, snd_cs46xx_suspend, snd_cs46xx_resume); #endif /* CONFIG_PM */ diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index e377287192aa..56fec0bc0efb 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -32,7 +32,7 @@ #include <sound/control.h> #include <sound/info.h> #include <sound/asoundef.h> -#include <sound/cs46xx.h> +#include "cs46xx.h" #include "cs46xx_lib.h" #include "dsp_spos.h" diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 00b148a10239..c2c695b07f8c 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -31,7 +31,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/info.h> -#include <sound/cs46xx.h> +#include "cs46xx.h" #include "cs46xx_lib.h" #include "dsp_spos.h" diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 2c9697cf0a1a..51f64ba5facf 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -400,8 +400,9 @@ static struct pci_driver cs5535audio_driver = { .probe = snd_cs5535audio_probe, .remove = __devexit_p(snd_cs5535audio_remove), #ifdef CONFIG_PM - .suspend = snd_cs5535audio_suspend, - .resume = snd_cs5535audio_resume, + .driver = { + .pm = &snd_cs5535audio_pm, + }, #endif }; diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h index 51966d782a3c..bb3cc641130c 100644 --- a/sound/pci/cs5535audio/cs5535audio.h +++ b/sound/pci/cs5535audio/cs5535audio.h @@ -94,10 +94,7 @@ struct cs5535audio { struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS]; }; -#ifdef CONFIG_PM -int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state); -int snd_cs5535audio_resume(struct pci_dev *pci); -#endif +extern const struct dev_pm_ops snd_cs5535audio_pm; #ifdef CONFIG_OLPC void __devinit olpc_prequirks(struct snd_card *card, diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c index 185b00088320..6c34def5986d 100644 --- a/sound/pci/cs5535audio/cs5535audio_pm.c +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -55,9 +55,10 @@ static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au) } -int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_cs5535audio_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cs5535audio *cs5535au = card->private_data; int i; @@ -77,13 +78,14 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state) return -EIO; } pci_disable_device(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -int snd_cs5535audio_resume(struct pci_dev *pci) +static int snd_cs5535audio_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cs5535audio *cs5535au = card->private_data; u32 tmp; int timeout; @@ -129,3 +131,4 @@ int snd_cs5535audio_resume(struct pci_dev *pci) return 0; } +SIMPLE_DEV_PM_OPS(snd_cs5535audio_pm, snd_cs5535audio_suspend, snd_cs5535audio_resume); diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index d8a4423539ce..8e40262d4117 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1537,7 +1537,7 @@ static void atc_connect_resources(struct ct_atc *atc) } #ifdef CONFIG_PM -static int atc_suspend(struct ct_atc *atc, pm_message_t state) +static int atc_suspend(struct ct_atc *atc) { int i; struct hw *hw = atc->hw; @@ -1553,7 +1553,7 @@ static int atc_suspend(struct ct_atc *atc, pm_message_t state) atc_release_resources(atc); - hw->suspend(hw, state); + hw->suspend(hw); return 0; } diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index 3a0def656af0..653e813ad142 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -144,7 +144,7 @@ struct ct_atc { struct ct_timer *timer; #ifdef CONFIG_PM - int (*suspend)(struct ct_atc *atc, pm_message_t state); + int (*suspend)(struct ct_atc *atc); int (*resume)(struct ct_atc *atc); #define NUM_PCMS (NUM_CTALSADEVS - 1) struct snd_pcm *pcms[NUM_PCMS]; diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index 908315bec3b4..c56fe533b3f3 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -73,7 +73,7 @@ struct hw { int (*card_stop)(struct hw *hw); int (*pll_init)(struct hw *hw, unsigned int rsr); #ifdef CONFIG_PM - int (*suspend)(struct hw *hw, pm_message_t state); + int (*suspend)(struct hw *hw); int (*resume)(struct hw *hw, struct card_conf *info); #endif int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source); diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index a7df19791f5a..dc1969bc67d4 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -2086,7 +2086,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) } #ifdef CONFIG_PM -static int hw_suspend(struct hw *hw, pm_message_t state) +static int hw_suspend(struct hw *hw) { struct pci_dev *pci = hw->pci; @@ -2099,7 +2099,7 @@ static int hw_suspend(struct hw *hw, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index d6c54b524bfa..9d1231dc4ae2 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -2202,7 +2202,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) } #ifdef CONFIG_PM -static int hw_suspend(struct hw *hw, pm_message_t state) +static int hw_suspend(struct hw *hw) { struct pci_dev *pci = hw->pci; @@ -2210,7 +2210,7 @@ static int hw_suspend(struct hw *hw, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 75aa2c338410..e002183ef8b2 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -126,21 +126,26 @@ static void __devexit ct_card_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int ct_card_suspend(struct pci_dev *pci, pm_message_t state) +static int ct_card_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct snd_card *card = dev_get_drvdata(dev); struct ct_atc *atc = card->private_data; - return atc->suspend(atc, state); + return atc->suspend(atc); } -static int ct_card_resume(struct pci_dev *pci) +static int ct_card_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct snd_card *card = dev_get_drvdata(dev); struct ct_atc *atc = card->private_data; return atc->resume(atc); } + +static SIMPLE_DEV_PM_OPS(ct_card_pm, ct_card_suspend, ct_card_resume); +#define CT_CARD_PM_OPS &ct_card_pm +#else +#define CT_CARD_PM_OPS NULL #endif static struct pci_driver ct_driver = { @@ -148,10 +153,9 @@ static struct pci_driver ct_driver = { .id_table = ct_pci_dev_ids, .probe = ct_card_probe, .remove = __devexit_p(ct_card_remove), -#ifdef CONFIG_PM - .suspend = ct_card_suspend, - .resume = ct_card_resume, -#endif + .driver = { + .pm = CT_CARD_PM_OPS, + }, }; module_pci_driver(ct_driver); diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 0f8eda1dafdb..0ff754f180d0 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2205,9 +2205,10 @@ ctl_error: #if defined(CONFIG_PM) -static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_echo_suspend(struct device *dev) { - struct echoaudio *chip = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct echoaudio *chip = dev_get_drvdata(dev); DE_INIT(("suspend start\n")); snd_pcm_suspend_all(chip->analog_pcm); @@ -2242,9 +2243,10 @@ static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) -static int snd_echo_resume(struct pci_dev *pci) +static int snd_echo_resume(struct device *dev) { - struct echoaudio *chip = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct echoaudio *chip = dev_get_drvdata(dev); struct comm_page *commpage, *commpage_bak; u32 pipe_alloc_mask; int err; @@ -2307,10 +2309,13 @@ static int snd_echo_resume(struct pci_dev *pci) return 0; } +static SIMPLE_DEV_PM_OPS(snd_echo_pm, snd_echo_suspend, snd_echo_resume); +#define SND_ECHO_PM_OPS &snd_echo_pm +#else +#define SND_ECHO_PM_OPS NULL #endif /* CONFIG_PM */ - static void __devexit snd_echo_remove(struct pci_dev *pci) { struct echoaudio *chip; @@ -2333,10 +2338,9 @@ static struct pci_driver echo_driver = { .id_table = snd_echo_ids, .probe = snd_echo_probe, .remove = __devexit_p(snd_echo_remove), -#ifdef CONFIG_PM - .suspend = snd_echo_suspend, - .resume = snd_echo_resume, -#endif /* CONFIG_PM */ + .driver = { + .pm = SND_ECHO_PM_OPS, + }, }; module_pci_driver(echo_driver); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 7fdbbe4d9965..ddac4e6d660d 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -207,9 +207,10 @@ static void __devexit snd_card_emu10k1_remove(struct pci_dev *pci) #ifdef CONFIG_PM -static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_emu10k1_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_emu10k1 *emu = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -231,13 +232,14 @@ static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_emu10k1_resume(struct pci_dev *pci) +static int snd_emu10k1_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_emu10k1 *emu = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -261,17 +263,21 @@ static int snd_emu10k1_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif + +static SIMPLE_DEV_PM_OPS(snd_emu10k1_pm, snd_emu10k1_suspend, snd_emu10k1_resume); +#define SND_EMU10K1_PM_OPS &snd_emu10k1_pm +#else +#define SND_EMU10K1_PM_OPS NULL +#endif /* CONFIG_PM */ static struct pci_driver emu10k1_driver = { .name = KBUILD_MODNAME, .id_table = snd_emu10k1_ids, .probe = snd_card_emu10k1_probe, .remove = __devexit_p(snd_card_emu10k1_remove), -#ifdef CONFIG_PM - .suspend = snd_emu10k1_suspend, - .resume = snd_emu10k1_resume, -#endif + .driver = { + .pm = SND_EMU10K1_PM_OPS, + }, }; module_pci_driver(emu10k1_driver); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 3821c81d1c99..f7e6f73186e1 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2033,9 +2033,10 @@ static void snd_ensoniq_chip_init(struct ensoniq *ensoniq) } #ifdef CONFIG_PM -static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_ensoniq_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct ensoniq *ensoniq = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -2058,13 +2059,14 @@ static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_ensoniq_resume(struct pci_dev *pci) +static int snd_ensoniq_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct ensoniq *ensoniq = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -2087,8 +2089,12 @@ static int snd_ensoniq_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif /* CONFIG_PM */ +static SIMPLE_DEV_PM_OPS(snd_ensoniq_pm, snd_ensoniq_suspend, snd_ensoniq_resume); +#define SND_ENSONIQ_PM_OPS &snd_ensoniq_pm +#else +#define SND_ENSONIQ_PM_OPS NULL +#endif /* CONFIG_PM */ static int __devinit snd_ensoniq_create(struct snd_card *card, struct pci_dev *pci, @@ -2493,10 +2499,9 @@ static struct pci_driver ens137x_driver = { .id_table = snd_audiopci_ids, .probe = snd_audiopci_probe, .remove = __devexit_p(snd_audiopci_remove), -#ifdef CONFIG_PM - .suspend = snd_ensoniq_suspend, - .resume = snd_ensoniq_resume, -#endif + .driver = { + .pm = SND_ENSONIQ_PM_OPS, + }, }; module_pci_driver(ens137x_driver); diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 82c8d8c5c52a..dbb81807bc1a 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1321,35 +1321,30 @@ static int snd_es1938_put_double(struct snd_kcontrol *kcontrol, return change; } -static unsigned int db_scale_master[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_master, 0, 54, TLV_DB_SCALE_ITEM(-3600, 50, 1), 54, 63, TLV_DB_SCALE_ITEM(-900, 100, 0), -}; +); -static unsigned int db_scale_audio1[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_audio1, 0, 8, TLV_DB_SCALE_ITEM(-3300, 300, 1), 8, 15, TLV_DB_SCALE_ITEM(-900, 150, 0), -}; +); -static unsigned int db_scale_audio2[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_audio2, 0, 8, TLV_DB_SCALE_ITEM(-3450, 300, 1), 8, 15, TLV_DB_SCALE_ITEM(-1050, 150, 0), -}; +); -static unsigned int db_scale_mic[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_mic, 0, 8, TLV_DB_SCALE_ITEM(-2400, 300, 1), 8, 15, TLV_DB_SCALE_ITEM(0, 150, 0), -}; +); -static unsigned int db_scale_line[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_line, 0, 8, TLV_DB_SCALE_ITEM(-3150, 300, 1), 8, 15, TLV_DB_SCALE_ITEM(-750, 150, 0), -}; +); static const DECLARE_TLV_DB_SCALE(db_scale_capture, 0, 150, 0); @@ -1474,9 +1469,10 @@ static unsigned char saved_regs[SAVED_REG_SIZE+1] = { }; -static int es1938_suspend(struct pci_dev *pci, pm_message_t state) +static int es1938_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct es1938 *chip = card->private_data; unsigned char *s, *d; @@ -1494,13 +1490,14 @@ static int es1938_suspend(struct pci_dev *pci, pm_message_t state) } pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int es1938_resume(struct pci_dev *pci) +static int es1938_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct es1938 *chip = card->private_data; unsigned char *s, *d; @@ -1534,6 +1531,11 @@ static int es1938_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(es1938_pm, es1938_suspend, es1938_resume); +#define ES1938_PM_OPS &es1938_pm +#else +#define ES1938_PM_OPS NULL #endif /* CONFIG_PM */ #ifdef SUPPORT_JOYSTICK @@ -1887,10 +1889,9 @@ static struct pci_driver es1938_driver = { .id_table = snd_es1938_ids, .probe = snd_es1938_probe, .remove = __devexit_p(snd_es1938_remove), -#ifdef CONFIG_PM - .suspend = es1938_suspend, - .resume = es1938_resume, -#endif + .driver = { + .pm = ES1938_PM_OPS, + }, }; module_pci_driver(es1938_driver); diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 52b5c0bf90c1..fb4c90b99c00 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2381,9 +2381,10 @@ static void snd_es1968_start_irq(struct es1968 *chip) /* * PM support */ -static int es1968_suspend(struct pci_dev *pci, pm_message_t state) +static int es1968_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct es1968 *chip = card->private_data; if (! chip->do_pm) @@ -2398,13 +2399,14 @@ static int es1968_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int es1968_resume(struct pci_dev *pci) +static int es1968_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct es1968 *chip = card->private_data; struct esschan *es; @@ -2454,6 +2456,11 @@ static int es1968_resume(struct pci_dev *pci) chip->in_suspend = 0; return 0; } + +static SIMPLE_DEV_PM_OPS(es1968_pm, es1968_suspend, es1968_resume); +#define ES1968_PM_OPS &es1968_pm +#else +#define ES1968_PM_OPS NULL #endif /* CONFIG_PM */ #ifdef SUPPORT_JOYSTICK @@ -2903,10 +2910,9 @@ static struct pci_driver es1968_driver = { .id_table = snd_es1968_ids, .probe = snd_es1968_probe, .remove = __devexit_p(snd_es1968_remove), -#ifdef CONFIG_PM - .suspend = es1968_suspend, - .resume = es1968_resume, -#endif + .driver = { + .pm = ES1968_PM_OPS, + }, }; module_pci_driver(es1968_driver); diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index b32e8024ea86..522c8706f244 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1369,9 +1369,10 @@ static unsigned char saved_regs[] = { FM801_CODEC_CTRL, FM801_I2S_MODE, FM801_VOLUME, FM801_GEN_CTRL, }; -static int snd_fm801_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_fm801_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct fm801 *chip = card->private_data; int i; @@ -1385,13 +1386,14 @@ static int snd_fm801_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_fm801_resume(struct pci_dev *pci) +static int snd_fm801_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct fm801 *chip = card->private_data; int i; @@ -1414,17 +1416,21 @@ static int snd_fm801_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif + +static SIMPLE_DEV_PM_OPS(snd_fm801_pm, snd_fm801_suspend, snd_fm801_resume); +#define SND_FM801_PM_OPS &snd_fm801_pm +#else +#define SND_FM801_PM_OPS NULL +#endif /* CONFIG_PM */ static struct pci_driver fm801_driver = { .name = KBUILD_MODNAME, .id_table = snd_fm801_ids, .probe = snd_card_fm801_probe, .remove = __devexit_p(snd_card_fm801_remove), -#ifdef CONFIG_PM - .suspend = snd_fm801_suspend, - .resume = snd_fm801_resume, -#endif + .driver = { + .pm = SND_FM801_PM_OPS, + }, }; module_pci_driver(fm801_driver); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index d03079764189..194d625c1f83 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -53,15 +53,14 @@ config SND_HDA_INPUT_BEEP driver. This interface is used to generate digital beeps. config SND_HDA_INPUT_BEEP_MODE - int "Digital beep registration mode (0=off, 1=on, 2=mute sw on/off)" + int "Digital beep registration mode (0=off, 1=on)" depends on SND_HDA_INPUT_BEEP=y default "1" - range 0 2 + range 0 1 help Set 0 to disable the digital beep interface for HD-audio by default. Set 1 to always enable the digital beep interface for HD-audio by - default. Set 2 to control the beep device registration to input - layer using a "Beep Switch" in mixer applications. + default. config SND_HDA_INPUT_JACK bool "Support jack plugging notification via input layer" diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index f7520b9f909c..647218d69f68 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -727,7 +727,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, models++; } } - if (id < 0) { + if (id < 0 && quirk) { q = snd_pci_quirk_lookup(codec->bus->pci, quirk); if (q) { id = q->value; @@ -736,7 +736,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, #endif } } - if (id < 0) { + if (id < 0 && quirk) { for (q = quirk; q->subvendor; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 60738e52b8f9..0bc2315b181d 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -162,50 +162,20 @@ static int snd_hda_do_attach(struct hda_beep *beep) return 0; } -static void snd_hda_do_register(struct work_struct *work) -{ - struct hda_beep *beep = - container_of(work, struct hda_beep, register_work); - - mutex_lock(&beep->mutex); - if (beep->enabled && !beep->dev) - snd_hda_do_attach(beep); - mutex_unlock(&beep->mutex); -} - -static void snd_hda_do_unregister(struct work_struct *work) -{ - struct hda_beep *beep = - container_of(work, struct hda_beep, unregister_work.work); - - mutex_lock(&beep->mutex); - if (!beep->enabled && beep->dev) - snd_hda_do_detach(beep); - mutex_unlock(&beep->mutex); -} - int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) { struct hda_beep *beep = codec->beep; - enable = !!enable; - if (beep == NULL) + if (!beep) return 0; + enable = !!enable; if (beep->enabled != enable) { beep->enabled = enable; if (!enable) { + cancel_work_sync(&beep->beep_work); /* turn off beep */ snd_hda_codec_write(beep->codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, 0); } - if (beep->mode == HDA_BEEP_MODE_SWREG) { - if (enable) { - cancel_delayed_work(&beep->unregister_work); - schedule_work(&beep->register_work); - } else { - schedule_delayed_work(&beep->unregister_work, - HZ); - } - } return 1; } return 0; @@ -215,6 +185,7 @@ EXPORT_SYMBOL_HDA(snd_hda_enable_beep_device); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { struct hda_beep *beep; + int err; if (!snd_hda_get_bool_hint(codec, "beep")) return 0; /* disabled explicitly by hints */ @@ -232,21 +203,16 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) beep->nid = nid; beep->codec = codec; - beep->mode = codec->beep_mode; codec->beep = beep; - INIT_WORK(&beep->register_work, &snd_hda_do_register); - INIT_DELAYED_WORK(&beep->unregister_work, &snd_hda_do_unregister); INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); mutex_init(&beep->mutex); - if (beep->mode == HDA_BEEP_MODE_ON) { - int err = snd_hda_do_attach(beep); - if (err < 0) { - kfree(beep); - codec->beep = NULL; - return err; - } + err = snd_hda_do_attach(beep); + if (err < 0) { + kfree(beep); + codec->beep = NULL; + return err; } return 0; @@ -257,8 +223,6 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) { struct hda_beep *beep = codec->beep; if (beep) { - cancel_work_sync(&beep->register_work); - cancel_delayed_work(&beep->unregister_work); if (beep->dev) snd_hda_do_detach(beep); codec->beep = NULL; @@ -266,3 +230,31 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); + +/* get/put callbacks for beep mute mixer switches */ +int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_beep *beep = codec->beep; + if (beep) { + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[1] = + beep->enabled; + return 0; + } + return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); +} +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get_beep); + +int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_beep *beep = codec->beep; + if (beep) + snd_hda_enable_beep_device(codec, + *ucontrol->value.integer.value); + return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); +} +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 55f0647458c7..4dc6933bc655 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -26,21 +26,16 @@ #define HDA_BEEP_MODE_OFF 0 #define HDA_BEEP_MODE_ON 1 -#define HDA_BEEP_MODE_SWREG 2 /* beep information */ struct hda_beep { struct input_dev *dev; struct hda_codec *codec; - unsigned int mode; char phys[32]; int tone; hda_nid_t nid; unsigned int enabled:1; - unsigned int request_enable:1; unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ - struct work_struct register_work; /* registration work */ - struct delayed_work unregister_work; /* unregistration work */ struct work_struct beep_work; /* scheduled task for beep event */ struct mutex mutex; }; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 51cb2a2e4fce..88a9c20eb7a2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2676,25 +2676,6 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put); -#ifdef CONFIG_SND_HDA_INPUT_BEEP -/** - * snd_hda_mixer_amp_switch_put_beep - Put callback for a beep AMP switch - * - * This function calls snd_hda_enable_beep_device(), which behaves differently - * depending on beep_mode option. - */ -int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - - snd_hda_enable_beep_device(codec, *valp); - return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); -} -EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -#endif /* CONFIG_SND_HDA_INPUT_BEEP */ - /* * bound volume controls * @@ -3509,22 +3490,52 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all); /* + * supported power states check + */ +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); + + if (sup < 0) + return false; + if (sup & power_state) + return true; + else + return false; +} + +/* * set power state of the codec */ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { + int count; + unsigned int state; + if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; } /* this delay seems necessary to avoid click noise at power-down */ - if (power_state == AC_PWRST_D3) - msleep(100); - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - power_state); - snd_hda_codec_set_power_to_all(codec, fg, power_state, true); + if (power_state == AC_PWRST_D3) { + /* transition time less than 10ms for power down */ + bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); + msleep(epss ? 10 : 100); + } + + /* repeat power states setting at most 10 times*/ + for (count = 0; count < 10; count++) { + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, + power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state, true); + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (!(state & AC_PWRST_ERROR)) + break; + } } #ifdef CONFIG_SND_HDA_HWDEP @@ -3545,7 +3556,7 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} static void hda_call_codec_suspend(struct hda_codec *codec) { if (codec->patch_ops.suspend) - codec->patch_ops.suspend(codec, PMSG_SUSPEND); + codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, @@ -4418,6 +4429,13 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) cancel_delayed_work_sync(&codec->power_work); spin_lock(&codec->power_lock); + /* If the power down delayed work was cancelled above before starting, + * then there is no need to go through power up here. + */ + if (codec->power_on) { + spin_unlock(&codec->power_lock); + return; + } trace_hda_power_up(codec); snd_hda_update_power_acct(codec); codec->power_on = 1; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2fdaadbb4326..c422d330ca54 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -323,6 +323,9 @@ enum { #define AC_PWRST_D1 0x01 #define AC_PWRST_D2 0x02 #define AC_PWRST_D3 0x03 +#define AC_PWRST_ERROR (1<<8) +#define AC_PWRST_CLK_STOP_OK (1<<9) +#define AC_PWRST_SETTING_RESET (1<<10) /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) @@ -703,7 +706,7 @@ struct hda_codec_ops { void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); #ifdef CONFIG_PM - int (*suspend)(struct hda_codec *codec, pm_message_t state); + int (*suspend)(struct hda_codec *codec); int (*resume)(struct hda_codec *codec); #endif #ifdef CONFIG_SND_HDA_POWER_SAVE diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7757536b9d5f..c8aced182fd1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -72,7 +72,7 @@ static int enable_msi = -1; static char *patch[SNDRV_CARDS]; #endif #ifdef CONFIG_SND_HDA_INPUT_BEEP -static int beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = +static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif @@ -103,9 +103,9 @@ module_param_array(patch, charp, NULL, 0444); MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface."); #endif #ifdef CONFIG_SND_HDA_INPUT_BEEP -module_param_array(beep_mode, int, NULL, 0444); +module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " - "(0=off, 1=on, 2=mute switch on/off) (default=1)."); + "(0=off, 1=on) (default=1)."); #endif #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," @@ -535,6 +536,7 @@ enum { #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ +#define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as default */ /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -2403,9 +2405,10 @@ static void azx_power_notify(struct hda_bus *bus) * power management */ -static int azx_suspend(struct pci_dev *pci, pm_message_t state) +static int azx_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct azx_pcm *p; @@ -2424,13 +2427,14 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_msi(chip->pci); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int azx_resume(struct pci_dev *pci) +static int azx_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -2455,6 +2459,12 @@ static int azx_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } +static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); +#define AZX_PM_OPS &azx_pm +#else +#define azx_suspend(dev) +#define azx_resume(dev) +#define AZX_PM_OPS NULL #endif /* CONFIG_PM */ @@ -2521,13 +2531,13 @@ static void azx_vs_set_state(struct pci_dev *pci, disabled ? "Disabling" : "Enabling", pci_name(chip->pci)); if (disabled) { - azx_suspend(pci, PMSG_FREEZE); + azx_suspend(&pci->dev); chip->disabled = true; snd_hda_lock_devices(chip->bus); } else { snd_hda_unlock_devices(chip->bus); chip->disabled = false; - azx_resume(pci); + azx_resume(&pci->dev); } } } @@ -2731,6 +2741,10 @@ static int __devinit check_position_fix(struct azx *chip, int fix) snd_printd(SFX "Using LPIB position fix\n"); return POS_FIX_LPIB; } + if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) { + snd_printd(SFX "Using COMBO position fix\n"); + return POS_FIX_COMBO; + } return POS_FIX_AUTO; } @@ -3243,7 +3257,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | @@ -3251,11 +3265,15 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Haswell */ + { PCI_DEVICE(0x8086, 0x0c0c), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | @@ -3341,6 +3359,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, + /* VIA GFX VT7122/VX900 */ + { PCI_DEVICE(0x1106, 0x9170), .driver_data = AZX_DRIVER_GENERIC }, + /* VIA GFX VT6122/VX11 */ + { PCI_DEVICE(0x1106, 0x9140), .driver_data = AZX_DRIVER_GENERIC }, /* SIS966 */ { PCI_DEVICE(0x1039, 0x7502), .driver_data = AZX_DRIVER_SIS }, /* ULI M5461 */ @@ -3398,10 +3420,9 @@ static struct pci_driver azx_driver = { .id_table = azx_ids, .probe = azx_probe, .remove = __devexit_p(azx_remove), -#ifdef CONFIG_PM - .suspend = azx_suspend, - .resume = azx_resume, -#endif + .driver = { + .pm = AZX_PM_OPS, + }, }; module_pci_driver(azx_driver); diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 2dd1c113a4c1..aaccc0236bda 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -127,10 +127,15 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) static void jack_detect_update(struct hda_codec *codec, struct hda_jack_tbl *jack) { - if (jack->jack_dirty || !jack->jack_detect) { + if (!jack->jack_dirty) + return; + + if (jack->phantom_jack) + jack->pin_sense = AC_PINSENSE_PRESENCE; + else jack->pin_sense = read_pin_sense(codec, jack->nid); - jack->jack_dirty = 0; - } + + jack->jack_dirty = 0; } /** @@ -264,8 +269,8 @@ static void hda_free_jack_priv(struct snd_jack *jack) * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. */ -int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, int idx) +static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx, bool phantom_jack) { struct hda_jack_tbl *jack; struct snd_kcontrol *kctl; @@ -283,47 +288,81 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, if (err < 0) return err; jack->kctl = kctl; + jack->phantom_jack = !!phantom_jack; + state = snd_hda_jack_detect(codec, nid); snd_kctl_jack_report(codec->bus->card, kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK - jack->type = get_input_jack_type(codec, nid); - err = snd_jack_new(codec->bus->card, name, jack->type, &jack->jack); - if (err < 0) - return err; - jack->jack->private_data = jack; - jack->jack->private_free = hda_free_jack_priv; - snd_jack_report(jack->jack, state ? jack->type : 0); + if (!phantom_jack) { + jack->type = get_input_jack_type(codec, nid); + err = snd_jack_new(codec->bus->card, name, jack->type, + &jack->jack); + if (err < 0) + return err; + jack->jack->private_data = jack; + jack->jack->private_free = hda_free_jack_priv; + snd_jack_report(jack->jack, state ? jack->type : 0); + } #endif return 0; } + +int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx) +{ + return __snd_hda_jack_add_kctl(codec, nid, name, idx, false); +} EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); +/* get the unique index number for the given kctl name */ +static int get_unique_index(struct hda_codec *codec, const char *name, int idx) +{ + struct hda_jack_tbl *jack; + int i, len = strlen(name); + again: + jack = codec->jacktbl.list; + for (i = 0; i < codec->jacktbl.used; i++, jack++) { + /* jack->kctl.id contains "XXX Jack" name string with index */ + if (jack->kctl && + !strncmp(name, jack->kctl->id.name, len) && + !strcmp(" Jack", jack->kctl->id.name + len) && + jack->kctl->id.index == idx) { + idx++; + goto again; + } + } + return idx; +} + static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg, - char *lastname, int *lastidx) + const struct auto_pin_cfg *cfg) { unsigned int def_conf, conn; char name[44]; int idx, err; + bool phantom_jack; if (!nid) return 0; - if (!is_jack_detectable(codec, nid)) - return 0; def_conf = snd_hda_codec_get_pincfg(codec, nid); conn = get_defcfg_connect(def_conf); - if (conn != AC_JACK_PORT_COMPLEX) + if (conn == AC_JACK_PORT_NONE) return 0; + phantom_jack = (conn != AC_JACK_PORT_COMPLEX) || + !is_jack_detectable(codec, nid); snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); - if (!strcmp(name, lastname) && idx == *lastidx) - idx++; - strncpy(lastname, name, 44); - *lastidx = idx; - err = snd_hda_jack_add_kctl(codec, nid, name, idx); + if (phantom_jack) + /* Example final name: "Internal Mic Phantom Jack" */ + strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); + idx = get_unique_index(codec, name, idx); + err = __snd_hda_jack_add_kctl(codec, nid, name, idx, phantom_jack); if (err < 0) return err; - return snd_hda_jack_detect_enable(codec, nid, 0); + + if (!phantom_jack) + return snd_hda_jack_detect_enable(codec, nid, 0); + return 0; } /** @@ -333,42 +372,41 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; - int i, err, lastidx = 0; - char lastname[44] = ""; + int i, err; for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } for (i = 0; i < cfg->num_inputs; i++) { - err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg); if (err < 0) return err; } for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } - err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, cfg->dig_in_pin, cfg); if (err < 0) return err; - err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, cfg->mono_out_pin, cfg); if (err < 0) return err; return 0; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 8ae52465ec5d..a9803da633c0 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -23,6 +23,7 @@ struct hda_jack_tbl { unsigned int pin_sense; /* cached pin-sense value */ unsigned int jack_detect:1; /* capable of jack-detection? */ unsigned int jack_dirty:1; /* needs to update? */ + unsigned int phantom_jack:1; /* a fixed, always present port? */ struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ #ifdef CONFIG_SND_HDA_INPUT_JACK int type; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 9a096a8e0fc5..1b4c12941baa 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -89,7 +89,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ .subdevice = HDA_SUBDEV_AMP_FLAG, \ .info = snd_hda_mixer_amp_switch_info, \ - .get = snd_hda_mixer_amp_switch_get, \ + .get = snd_hda_mixer_amp_switch_get_beep, \ .put = snd_hda_mixer_amp_switch_put_beep, \ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) } #else @@ -121,6 +121,8 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); #ifdef CONFIG_SND_HDA_INPUT_BEEP +int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); #endif diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e59e2f059b6e..7e46258fc700 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -426,10 +426,10 @@ static void print_digital_conv(struct snd_info_buffer *buffer, static const char *get_pwr_state(u32 state) { - static const char * const buf[4] = { - "D0", "D1", "D2", "D3" + static const char * const buf[] = { + "D0", "D1", "D2", "D3", "D3cold" }; - if (state < 4) + if (state < ARRAY_SIZE(buf)) return buf[state]; return "UNKNOWN"; } @@ -451,14 +451,21 @@ static void print_power_state(struct snd_info_buffer *buffer, int sup = snd_hda_param_read(codec, nid, AC_PAR_POWER_STATE); int pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); - if (sup) + if (sup != -1) snd_iprintf(buffer, " Power states: %s\n", bits_names(sup, names, ARRAY_SIZE(names))); - snd_iprintf(buffer, " Power: setting=%s, actual=%s\n", + snd_iprintf(buffer, " Power: setting=%s, actual=%s", get_pwr_state(pwr & AC_PWRST_SETTING), get_pwr_state((pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT)); + if (pwr & AC_PWRST_ERROR) + snd_iprintf(buffer, ", Error"); + if (pwr & AC_PWRST_CLK_STOP_OK) + snd_iprintf(buffer, ", Clock-stop-OK"); + if (pwr & AC_PWRST_SETTING_RESET) + snd_iprintf(buffer, ", Setting-reset"); + snd_iprintf(buffer, "\n"); } static void print_unsol_cap(struct snd_info_buffer *buffer, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d8b2d6dee986..0208fa121e5a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -642,7 +642,7 @@ static void ad198x_free(struct hda_codec *codec) } #ifdef CONFIG_PM -static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) +static int ad198x_suspend(struct hda_codec *codec) { ad198x_shutup(codec); return 0; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 9647ed4d7929..0c4c1a61b378 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1892,7 +1892,7 @@ static int cs421x_parse_auto_config(struct hda_codec *codec) Manage PDREF, when transitioning to D3hot (DAC,ADC) -> D3, PDREF=1, AFG->D3 */ -static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +static int cs421x_suspend(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; unsigned int coef; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2bf99fc1cbf2..14361184ae1e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -554,7 +554,7 @@ static int conexant_build_controls(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static int conexant_suspend(struct hda_codec *codec, pm_message_t state) +static int conexant_suspend(struct hda_codec *codec) { snd_hda_shutup_pins(codec); return 0; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ad319d4dc32f..641408dc28c0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -85,7 +85,7 @@ struct hdmi_spec { * Non-generic ATI/NVIDIA specific */ struct hda_multi_out multiout; - const struct hda_pcm_stream *pcm_playback; + struct hda_pcm_stream pcm_playback; }; @@ -787,7 +787,7 @@ static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); printk(KERN_INFO - "HDMI CP event: CODEC=%d PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + "HDMI CP event: CODEC=%d TAG=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", codec->addr, tag, subtag, @@ -876,7 +876,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; - int pinctl; /* Validate hinfo */ pin_idx = hinfo_to_pin_index(spec, hinfo); @@ -912,11 +911,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); - pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write(codec, per_pin->pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl | PIN_OUT); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ @@ -1153,11 +1147,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; + int pinctl; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); hdmi_setup_audio_infoframe(codec, pin_idx, substream); + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1277,23 +1277,34 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) return 0; } -static int generic_hdmi_init(struct hda_codec *codec) +static int generic_hdmi_init_per_pins(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; int pin_idx; for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; - hda_nid_t pin_nid = per_pin->pin_nid; struct hdmi_eld *eld = &per_pin->sink_eld; - hdmi_init_pin(codec, pin_nid); - snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); - per_pin->codec = codec; INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld); snd_hda_eld_proc_new(codec, eld, pin_idx); } + return 0; +} + +static int generic_hdmi_init(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + hda_nid_t pin_nid = per_pin->pin_nid; + + hdmi_init_pin(codec, pin_nid); + snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); + } snd_hda_jack_report_sync(codec); return 0; } @@ -1338,6 +1349,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); @@ -1352,45 +1364,65 @@ static int simple_playback_build_pcms(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; - int i; + unsigned int chans; + struct hda_pcm_stream *pstr; - codec->num_pcms = spec->num_cvts; + codec->num_pcms = 1; codec->pcm_info = info; - for (i = 0; i < codec->num_pcms; i++, info++) { - unsigned int chans; - struct hda_pcm_stream *pstr; - - chans = get_wcaps(codec, spec->cvts[i].cvt_nid); - chans = get_wcaps_channels(chans); + chans = get_wcaps(codec, spec->cvts[0].cvt_nid); + chans = get_wcaps_channels(chans); - info->name = get_hdmi_pcm_name(i); - info->pcm_type = HDA_PCM_TYPE_HDMI; - pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; - snd_BUG_ON(!spec->pcm_playback); - *pstr = *spec->pcm_playback; - pstr->nid = spec->cvts[i].cvt_nid; - if (pstr->channels_max <= 2 && chans && chans <= 16) - pstr->channels_max = chans; - } + info->name = get_hdmi_pcm_name(0); + info->pcm_type = HDA_PCM_TYPE_HDMI; + pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; + *pstr = spec->pcm_playback; + pstr->nid = spec->cvts[0].cvt_nid; + if (pstr->channels_max <= 2 && chans && chans <= 16) + pstr->channels_max = chans; return 0; } +/* unsolicited event for jack sensing */ +static void simple_hdmi_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_report_sync(codec); +} + +/* generic_hdmi_build_jack can be used for simple_hdmi, too, + * as long as spec->pins[] is set correctly + */ +#define simple_hdmi_build_jack generic_hdmi_build_jack + static int simple_playback_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; int err; - int i; - for (i = 0; i < codec->num_pcms; i++) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->cvts[i].cvt_nid, - spec->cvts[i].cvt_nid); - if (err < 0) - return err; - } + err = snd_hda_create_spdif_out_ctls(codec, + spec->cvts[0].cvt_nid, + spec->cvts[0].cvt_nid); + if (err < 0) + return err; + return simple_hdmi_build_jack(codec, 0); +} +static int simple_playback_init(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + hda_nid_t pin = spec->pins[0].pin_nid; + + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* some codecs require to unmute the pin */ + if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + snd_hda_jack_detect_enable(codec, pin, pin); + snd_hda_jack_report_sync(codec); return 0; } @@ -1418,7 +1450,15 @@ static const hda_nid_t nvhdmi_con_nids_7x[4] = { 0x6, 0x8, 0xa, 0xc, }; -static const struct hda_verb nvhdmi_basic_init_7x[] = { +static const struct hda_verb nvhdmi_basic_init_7x_2ch[] = { + /* set audio protect on */ + { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, + /* enable digital output on pin widget */ + { 0x5, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + {} /* terminator */ +}; + +static const struct hda_verb nvhdmi_basic_init_7x_8ch[] = { /* set audio protect on */ { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, /* enable digital output on pin widget */ @@ -1446,9 +1486,15 @@ static const struct hda_verb nvhdmi_basic_init_7x[] = { (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) #endif -static int nvhdmi_7x_init(struct hda_codec *codec) +static int nvhdmi_7x_init_2ch(struct hda_codec *codec) { - snd_hda_sequence_write(codec, nvhdmi_basic_init_7x); + snd_hda_sequence_write(codec, nvhdmi_basic_init_7x_2ch); + return 0; +} + +static int nvhdmi_7x_init_8ch(struct hda_codec *codec) +{ + snd_hda_sequence_write(codec, nvhdmi_basic_init_7x_8ch); return 0; } @@ -1524,6 +1570,50 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static const struct hda_pcm_stream simple_pcm_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .open = simple_playback_pcm_open, + .close = simple_playback_pcm_close, + .prepare = simple_playback_pcm_prepare + }, +}; + +static const struct hda_codec_ops simple_hdmi_patch_ops = { + .build_controls = simple_playback_build_controls, + .build_pcms = simple_playback_build_pcms, + .init = simple_playback_init, + .free = simple_playback_free, + .unsol_event = simple_hdmi_unsol_event, +}; + +static int patch_simple_hdmi(struct hda_codec *codec, + hda_nid_t cvt_nid, hda_nid_t pin_nid) +{ + struct hdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + + codec->spec = spec; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 2; + spec->multiout.dig_out_nid = cvt_nid; + spec->num_cvts = 1; + spec->num_pins = 1; + spec->cvts[0].cvt_nid = cvt_nid; + spec->pins[0].pin_nid = pin_nid; + spec->pcm_playback = simple_pcm_playback; + + codec->patch_ops = simple_hdmi_patch_ops; + + return 0; +} + static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec, int channels) { @@ -1696,54 +1786,20 @@ static const struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = { }, }; -static const struct hda_pcm_stream nvhdmi_pcm_playback_2ch = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = nvhdmi_master_con_nid_7x, - .rates = SUPPORTED_RATES, - .maxbps = SUPPORTED_MAXBPS, - .formats = SUPPORTED_FORMATS, - .ops = { - .open = simple_playback_pcm_open, - .close = simple_playback_pcm_close, - .prepare = simple_playback_pcm_prepare - }, -}; - -static const struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = { - .build_controls = simple_playback_build_controls, - .build_pcms = simple_playback_build_pcms, - .init = nvhdmi_7x_init, - .free = simple_playback_free, -}; - -static const struct hda_codec_ops nvhdmi_patch_ops_2ch = { - .build_controls = simple_playback_build_controls, - .build_pcms = simple_playback_build_pcms, - .init = nvhdmi_7x_init, - .free = simple_playback_free, -}; - static int patch_nvhdmi_2ch(struct hda_codec *codec) { struct hdmi_spec *spec; + int err = patch_simple_hdmi(codec, nvhdmi_master_con_nid_7x, + nvhdmi_master_pin_nid_7x); + if (err < 0) + return err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->multiout.num_dacs = 0; /* no analog */ - spec->multiout.max_channels = 2; - spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x; - spec->num_cvts = 1; - spec->cvts[0].cvt_nid = nvhdmi_master_con_nid_7x; - spec->pcm_playback = &nvhdmi_pcm_playback_2ch; - - codec->patch_ops = nvhdmi_patch_ops_2ch; - + codec->patch_ops.init = nvhdmi_7x_init_2ch; + /* override the PCM rates, etc, as the codec doesn't give full list */ + spec = codec->spec; + spec->pcm_playback.rates = SUPPORTED_RATES; + spec->pcm_playback.maxbps = SUPPORTED_MAXBPS; + spec->pcm_playback.formats = SUPPORTED_FORMATS; return 0; } @@ -1751,13 +1807,12 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) { struct hdmi_spec *spec; int err = patch_nvhdmi_2ch(codec); - if (err < 0) return err; spec = codec->spec; spec->multiout.max_channels = 8; - spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x; - codec->patch_ops = nvhdmi_patch_ops_8ch_7x; + spec->pcm_playback = nvhdmi_pcm_playback_8ch_7x; + codec->patch_ops.init = nvhdmi_7x_init_8ch; /* Initialize the audio infoframe channel mask and checksum to something * valid */ @@ -1801,69 +1856,26 @@ static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return 0; } -static const struct hda_pcm_stream atihdmi_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = ATIHDMI_CVT_NID, - .ops = { - .open = simple_playback_pcm_open, - .close = simple_playback_pcm_close, - .prepare = atihdmi_playback_pcm_prepare - }, -}; - -static const struct hda_verb atihdmi_basic_init[] = { - /* enable digital output on pin widget */ - { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - {} /* terminator */ -}; - -static int atihdmi_init(struct hda_codec *codec) +static int patch_atihdmi(struct hda_codec *codec) { - struct hdmi_spec *spec = codec->spec; - - snd_hda_sequence_write(codec, atihdmi_basic_init); - /* SI codec requires to unmute the pin */ - if (get_wcaps(codec, spec->pins[0].pin_nid) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, spec->pins[0].pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + struct hdmi_spec *spec; + int err = patch_simple_hdmi(codec, ATIHDMI_CVT_NID, ATIHDMI_PIN_NID); + if (err < 0) + return err; + spec = codec->spec; + spec->pcm_playback.ops.prepare = atihdmi_playback_pcm_prepare; return 0; } -static const struct hda_codec_ops atihdmi_patch_ops = { - .build_controls = simple_playback_build_controls, - .build_pcms = simple_playback_build_pcms, - .init = atihdmi_init, - .free = simple_playback_free, -}; +/* VIA HDMI Implementation */ +#define VIAHDMI_CVT_NID 0x02 /* audio converter1 */ +#define VIAHDMI_PIN_NID 0x03 /* HDMI output pin1 */ - -static int patch_atihdmi(struct hda_codec *codec) +static int patch_via_hdmi(struct hda_codec *codec) { - struct hdmi_spec *spec; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->multiout.num_dacs = 0; /* no analog */ - spec->multiout.max_channels = 2; - spec->multiout.dig_out_nid = ATIHDMI_CVT_NID; - spec->num_cvts = 1; - spec->cvts[0].cvt_nid = ATIHDMI_CVT_NID; - spec->pins[0].pin_nid = ATIHDMI_PIN_NID; - spec->pcm_playback = &atihdmi_pcm_digital_playback; - - codec->patch_ops = atihdmi_patch_ops; - - return 0; + return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } - /* * patch entries */ @@ -1902,8 +1914,13 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, +{ .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, +{ .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, +{ .id = 0x11069f84, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x11069f85, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x80860054, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862801, .name = "Bearlake HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862802, .name = "Cantiga HDMI", .patch = patch_generic_hdmi }, @@ -1911,6 +1928,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ @@ -1948,8 +1966,13 @@ MODULE_ALIAS("snd-hda-codec-id:10de0041"); MODULE_ALIAS("snd-hda-codec-id:10de0042"); MODULE_ALIAS("snd-hda-codec-id:10de0043"); MODULE_ALIAS("snd-hda-codec-id:10de0044"); +MODULE_ALIAS("snd-hda-codec-id:10de0051"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); +MODULE_ALIAS("snd-hda-codec-id:11069f80"); +MODULE_ALIAS("snd-hda-codec-id:11069f81"); +MODULE_ALIAS("snd-hda-codec-id:11069f84"); +MODULE_ALIAS("snd-hda-codec-id:11069f85"); MODULE_ALIAS("snd-hda-codec-id:17e80047"); MODULE_ALIAS("snd-hda-codec-id:80860054"); MODULE_ALIAS("snd-hda-codec-id:80862801"); @@ -1958,6 +1981,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862803"); MODULE_ALIAS("snd-hda-codec-id:80862804"); MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); +MODULE_ALIAS("snd-hda-codec-id:80862807"); MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aa4c25e0f327..f141395dfee6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -170,10 +170,10 @@ struct alc_spec { hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS]; unsigned int dyn_adc_idx[HDA_MAX_NUM_INPUTS]; int int_mic_idx, ext_mic_idx, dock_mic_idx; /* for auto-mic */ + hda_nid_t inv_dmic_pin; /* hooks */ void (*init_hook)(struct hda_codec *codec); - void (*unsol_event)(struct hda_codec *codec, unsigned int res); #ifdef CONFIG_SND_HDA_POWER_SAVE void (*power_hook)(struct hda_codec *codec); #endif @@ -201,6 +201,8 @@ struct alc_spec { unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */ unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */ unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */ + unsigned int inv_dmic_fixup:1; /* has inverted digital-mic workaround */ + unsigned int inv_dmic_muted:1; /* R-ch of inv d-mic is muted? */ /* auto-mute control */ int automute_mode; @@ -298,6 +300,39 @@ static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx) } static void call_update_outputs(struct hda_codec *codec); +static void alc_inv_dmic_sync(struct hda_codec *codec, bool force); + +/* for shared I/O, change the pin-control accordingly */ +static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + hda_nid_t pin = spec->autocfg.inputs[1].pin; + /* NOTE: this assumes that there are only two inputs, the + * first is the real internal mic and the second is HP/mic jack. + */ + + val = snd_hda_get_default_vref(codec, pin); + + /* This pin does not have vref caps - let's enable vref on pin 0x18 + instead, as suggested by Realtek */ + if (val == AC_PINCTL_VREF_HIZ) { + const hda_nid_t vref_pin = 0x18; + /* Sanity check pin 0x18 */ + if (get_wcaps_type(get_wcaps(codec, vref_pin)) == AC_WID_PIN && + get_defcfg_connect(snd_hda_codec_get_pincfg(codec, vref_pin)) == AC_JACK_PORT_NONE) { + unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin); + if (vref_val != AC_PINCTL_VREF_HIZ) + snd_hda_set_pin_ctl(codec, vref_pin, PIN_IN | (set_as_mic ? vref_val : 0)); + } + } + + val = set_as_mic ? val | PIN_IN : PIN_HP; + snd_hda_set_pin_ctl(codec, pin, val); + + spec->automute_speaker = !set_as_mic; + call_update_outputs(codec); +} /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, @@ -325,21 +360,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; spec->cur_mux[adc_idx] = idx; - /* for shared I/O, change the pin-control accordingly */ - if (spec->shared_mic_hp) { - unsigned int val; - hda_nid_t pin = spec->autocfg.inputs[1].pin; - /* NOTE: this assumes that there are only two inputs, the - * first is the real internal mic and the second is HP jack. - */ - if (spec->cur_mux[adc_idx]) - val = snd_hda_get_default_vref(codec, pin) | PIN_IN; - else - val = PIN_HP; - snd_hda_set_pin_ctl(codec, pin, val); - spec->automute_speaker = !spec->cur_mux[adc_idx]; - call_update_outputs(codec); - } + if (spec->shared_mic_hp) + update_shared_mic_hp(codec, spec->cur_mux[adc_idx]); if (spec->dyn_adc_switch) { alc_dyn_adc_pcm_resetup(codec, idx); @@ -368,6 +390,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, AC_VERB_SET_CONNECT_SEL, imux->items[idx].index); } + alc_inv_dmic_sync(codec, true); return 1; } @@ -664,7 +687,7 @@ static void alc_update_knob_master(struct hda_codec *codec, hda_nid_t nid) } /* unsolicited event for HP jack sensing */ -static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +static void alc_unsol_event(struct hda_codec *codec, unsigned int res) { int action; @@ -1000,11 +1023,9 @@ static void alc_init_automute(struct hda_codec *codec) spec->automute_lo = spec->automute_lo_possible; spec->automute_speaker = spec->automute_speaker_possible; - if (spec->automute_speaker_possible || spec->automute_lo_possible) { + if (spec->automute_speaker_possible || spec->automute_lo_possible) /* create a control for automute mode */ alc_add_automute_mode_enum(codec); - spec->unsol_event = alc_sku_unsol_event; - } } /* return the position of NID in the list, or -1 if not found */ @@ -1167,7 +1188,6 @@ static void alc_init_auto_mic(struct hda_codec *codec) snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x/0x%x\n", ext, fixed, dock); - spec->unsol_event = alc_sku_unsol_event; } /* check the availabilities of auto-mute and auto-mic switches */ @@ -1556,14 +1576,14 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol, static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol, - getput_call_t func, bool check_adc_switch) + getput_call_t func, bool is_put) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; int i, err = 0; mutex_lock(&codec->control_mutex); - if (check_adc_switch && spec->dyn_adc_switch) { + if (is_put && spec->dyn_adc_switch) { for (i = 0; i < spec->num_adc_nids; i++) { kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[i], @@ -1584,6 +1604,8 @@ static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, 3, 0, HDA_INPUT); err = func(kcontrol, ucontrol); } + if (err >= 0 && is_put) + alc_inv_dmic_sync(codec, false); error: mutex_unlock(&codec->control_mutex); return err; @@ -1676,6 +1698,116 @@ DEFINE_CAPMIX_NOSRC(2); DEFINE_CAPMIX_NOSRC(3); /* + * Inverted digital-mic handling + * + * First off, it's a bit tricky. The "Inverted Internal Mic Capture Switch" + * gives the additional mute only to the right channel of the digital mic + * capture stream. This is a workaround for avoiding the almost silence + * by summing the stereo stream from some (known to be ForteMedia) + * digital mic unit. + * + * The logic is to call alc_inv_dmic_sync() after each action (possibly) + * modifying ADC amp. When the mute flag is set, it mutes the R-channel + * without caching so that the cache can still keep the original value. + * The cached value is then restored when the flag is set off or any other + * than d-mic is used as the current input source. + */ +static void alc_inv_dmic_sync(struct hda_codec *codec, bool force) +{ + struct alc_spec *spec = codec->spec; + int i; + + if (!spec->inv_dmic_fixup) + return; + if (!spec->inv_dmic_muted && !force) + return; + for (i = 0; i < spec->num_adc_nids; i++) { + int src = spec->dyn_adc_switch ? 0 : i; + bool dmic_fixup = false; + hda_nid_t nid; + int parm, dir, v; + + if (spec->inv_dmic_muted && + spec->imux_pins[spec->cur_mux[src]] == spec->inv_dmic_pin) + dmic_fixup = true; + if (!dmic_fixup && !force) + continue; + if (spec->vol_in_capsrc) { + nid = spec->capsrc_nids[i]; + parm = AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT; + dir = HDA_OUTPUT; + } else { + nid = spec->adc_nids[i]; + parm = AC_AMP_SET_RIGHT | AC_AMP_SET_INPUT; + dir = HDA_INPUT; + } + /* we care only right channel */ + v = snd_hda_codec_amp_read(codec, nid, 1, dir, 0); + if (v & 0x80) /* if already muted, we don't need to touch */ + continue; + if (dmic_fixup) /* add mute for d-mic */ + v |= 0x80; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + parm | v); + } +} + +static int alc_inv_dmic_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = !spec->inv_dmic_muted; + return 0; +} + +static int alc_inv_dmic_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + unsigned int val = !ucontrol->value.integer.value[0]; + + if (val == spec->inv_dmic_muted) + return 0; + spec->inv_dmic_muted = val; + alc_inv_dmic_sync(codec, true); + return 0; +} + +static const struct snd_kcontrol_new alc_inv_dmic_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_ctl_boolean_mono_info, + .get = alc_inv_dmic_sw_get, + .put = alc_inv_dmic_sw_put, +}; + +static int alc_add_inv_dmic_mixer(struct hda_codec *codec, hda_nid_t nid) +{ + struct alc_spec *spec = codec->spec; + struct snd_kcontrol_new *knew = alc_kcontrol_new(spec); + if (!knew) + return -ENOMEM; + *knew = alc_inv_dmic_sw; + knew->name = kstrdup("Inverted Internal Mic Capture Switch", GFP_KERNEL); + if (!knew->name) + return -ENOMEM; + spec->inv_dmic_fixup = 1; + spec->inv_dmic_muted = 0; + spec->inv_dmic_pin = nid; + return 0; +} + +/* typically the digital mic is put at node 0x12 */ +static void alc_fixup_inv_dmic_0x12(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PROBE) + alc_add_inv_dmic_mixer(codec, 0x12); +} + +/* * virtual master controls */ @@ -1865,13 +1997,31 @@ static int __alc_build_controls(struct hda_codec *codec) return 0; } -static int alc_build_controls(struct hda_codec *codec) +static int alc_build_jacks(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + + if (spec->shared_mic_hp) { + int err; + int nid = spec->autocfg.inputs[1].pin; + err = snd_hda_jack_add_kctl(codec, nid, "Headphone Mic", 0); + if (err < 0) + return err; + err = snd_hda_jack_detect_enable(codec, nid, 0); + if (err < 0) + return err; + } + + return snd_hda_jack_add_kctls(codec, &spec->autocfg); +} + +static int alc_build_controls(struct hda_codec *codec) +{ int err = __alc_build_controls(codec); if (err < 0) return err; - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + + err = alc_build_jacks(codec); if (err < 0) return err; alc_apply_fixup(codec, ALC_FIXUP_ACT_BUILD); @@ -1908,14 +2058,6 @@ static int alc_init(struct hda_codec *codec) return 0; } -static void alc_unsol_event(struct hda_codec *codec, unsigned int res) -{ - struct alc_spec *spec = codec->spec; - - if (spec->unsol_event) - spec->unsol_event(codec, res); -} - #ifdef CONFIG_SND_HDA_POWER_SAVE static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -2300,7 +2442,7 @@ static void alc_power_eapd(struct hda_codec *codec) alc_auto_setup_eapd(codec, false); } -static int alc_suspend(struct hda_codec *codec, pm_message_t state) +static int alc_suspend(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; alc_shutup(codec); @@ -2317,6 +2459,7 @@ static int alc_resume(struct hda_codec *codec) codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); + alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); return 0; } @@ -4116,14 +4259,12 @@ static void set_capture_mixer(struct hda_codec *codec) */ static void alc_auto_init_std(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; alc_auto_init_multi_out(codec); alc_auto_init_extra_out(codec); alc_auto_init_analog_input(codec); alc_auto_init_input_src(codec); alc_auto_init_digital(codec); - if (spec->unsol_event) - alc_inithook(codec); + alc_inithook(codec); } /* @@ -4724,7 +4865,6 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, spec->automute_speaker = 1; spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT); - spec->unsol_event = alc_sku_unsol_event; snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs); } } @@ -4909,6 +5049,7 @@ enum { ALC889_FIXUP_DAC_ROUTE, ALC889_FIXUP_MBP_VREF, ALC889_FIXUP_IMAC91_VREF, + ALC882_FIXUP_INV_DMIC, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -5212,6 +5353,10 @@ static const struct alc_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC882_FIXUP_GPIO1, }, + [ALC882_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -5286,6 +5431,7 @@ static const struct alc_model_fixup alc882_fixup_models[] = { {.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"}, {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, + {.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"}, {} }; @@ -5373,6 +5519,7 @@ enum { ALC262_FIXUP_LENOVO_3000, ALC262_FIXUP_BENQ, ALC262_FIXUP_BENQ_T31, + ALC262_FIXUP_INV_DMIC, }; static const struct alc_fixup alc262_fixups[] = { @@ -5424,6 +5571,10 @@ static const struct alc_fixup alc262_fixups[] = { {} } }, + [ALC262_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, }; static const struct snd_pci_quirk alc262_fixup_tbl[] = { @@ -5438,6 +5589,10 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { {} }; +static const struct alc_model_fixup alc262_fixup_models[] = { + {.id = ALC262_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {} +}; /* */ @@ -5466,7 +5621,8 @@ static int patch_alc262(struct hda_codec *codec) #endif alc_fix_pll_init(codec, 0x20, 0x0a, 10); - alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups); + alc_pick_fixup(codec, alc262_fixup_models, alc262_fixup_tbl, + alc262_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); @@ -5522,6 +5678,22 @@ static const struct hda_verb alc268_beep_init_verbs[] = { { } }; +enum { + ALC268_FIXUP_INV_DMIC, +}; + +static const struct alc_fixup alc268_fixups[] = { + [ALC268_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, +}; + +static const struct alc_model_fixup alc268_fixup_models[] = { + {.id = ALC268_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {} +}; + /* * BIOS auto configuration */ @@ -5553,6 +5725,9 @@ static int patch_alc268(struct hda_codec *codec) spec = codec->spec; + alc_pick_fixup(codec, alc268_fixup_models, NULL, alc268_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + /* automatic parse from the BIOS config */ err = alc268_parse_auto_config(codec); if (err < 0) @@ -5582,6 +5757,8 @@ static int patch_alc268(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; spec->shutup = alc_eapd_shutup; + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: @@ -5704,6 +5881,15 @@ static int alc269_resume(struct hda_codec *codec) } #endif /* CONFIG_PM */ +static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == ALC_FIXUP_ACT_PRE_PROBE) + spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; +} + static void alc269_fixup_hweq(struct hda_codec *codec, const struct alc_fixup *fix, int action) { @@ -5810,6 +5996,7 @@ static void alc269_fixup_mic2_mute(struct hda_codec *codec, } } + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5828,6 +6015,9 @@ enum { ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, ALC269_FIXUP_MIC2_MUTE_LED, + ALC269_FIXUP_INV_DMIC, + ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, }; static const struct alc_fixup alc269_fixups[] = { @@ -5952,12 +6142,33 @@ static const struct alc_fixup alc269_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, }, + [ALC269_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, + [ALC269_FIXUP_LENOVO_DOCK] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x19, 0x23a11040 }, /* dock mic */ + { 0x1b, 0x2121103f }, /* dock headphone */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT + }, + [ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_pincfg_no_hp_to_lineout, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), + SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), @@ -5975,6 +6186,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), @@ -6033,6 +6245,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { static const struct alc_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_AMIC, .name = "laptop-amic"}, {.id = ALC269_FIXUP_DMIC, .name = "laptop-dmic"}, + {.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"}, + {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, + {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {} }; @@ -6329,12 +6545,6 @@ static const struct snd_pci_quirk alc861vd_fixup_tbl[] = { {} }; -static const struct hda_verb alc660vd_eapd_verbs[] = { - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - /* */ static int patch_alc861vd(struct hda_codec *codec) @@ -6356,11 +6566,6 @@ static int patch_alc861vd(struct hda_codec *codec) if (err < 0) goto error; - if (codec->vendor_id == 0x10ec0660) { - /* always turn on EAPD */ - snd_hda_gen_add_verbs(&spec->gen, alc660vd_eapd_verbs); - } - if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x23); if (err < 0) @@ -6443,6 +6648,7 @@ enum { ALC662_FIXUP_ASUS_MODE8, ALC662_FIXUP_NO_JACK_DETECT, ALC662_FIXUP_ZOTAC_Z68, + ALC662_FIXUP_INV_DMIC, }; static const struct alc_fixup alc662_fixups[] = { @@ -6599,12 +6805,17 @@ static const struct alc_fixup alc662_fixups[] = { { } } }, + [ALC662_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), @@ -6685,6 +6896,7 @@ static const struct alc_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"}, {.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"}, {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, + {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, {} }; @@ -6831,6 +7043,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 }, { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 }, + { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 07675282015a..a1596a3b171c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4997,7 +4997,7 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } -static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) +static int stac92xx_suspend(struct hda_codec *codec) { stac92xx_shutup(codec); return 0; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 82b368068e08..90645560ed39 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1748,7 +1748,7 @@ static void via_unsol_event(struct hda_codec *codec, } #ifdef CONFIG_PM -static int via_suspend(struct hda_codec *codec, pm_message_t state) +static int via_suspend(struct hda_codec *codec) { struct via_spec *spec = codec->spec; vt1708_stop_hp_work(spec); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index a01a00d1cf4d..bed9f34f4efe 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2793,9 +2793,10 @@ static void __devexit snd_vt1724_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_vt1724_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ice1712 *ice = card->private_data; if (!ice->pm_suspend_enabled) @@ -2820,13 +2821,14 @@ static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_vt1724_resume(struct pci_dev *pci) +static int snd_vt1724_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ice1712 *ice = card->private_data; if (!ice->pm_suspend_enabled) @@ -2871,17 +2873,21 @@ static int snd_vt1724_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif + +static SIMPLE_DEV_PM_OPS(snd_vt1724_pm, snd_vt1724_suspend, snd_vt1724_resume); +#define SND_VT1724_PM_OPS &snd_vt1724_pm +#else +#define SND_VT1724_PM_OPS NULL +#endif /* CONFIG_PM */ static struct pci_driver vt1724_driver = { .name = KBUILD_MODNAME, .id_table = snd_vt1724_ids, .probe = snd_vt1724_probe, .remove = __devexit_p(snd_vt1724_remove), -#ifdef CONFIG_PM - .suspend = snd_vt1724_suspend, - .resume = snd_vt1724_resume, -#endif + .driver = { + .pm = SND_VT1724_PM_OPS, + }, }; module_pci_driver(vt1724_driver); diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index f4e2dd4da8cf..cd553f592e2d 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2624,9 +2624,10 @@ static int snd_intel8x0_free(struct intel8x0 *chip) /* * power management */ -static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state) +static int intel8x0_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct intel8x0 *chip = card->private_data; int i; @@ -2658,13 +2659,14 @@ static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state) /* The call below may disable built-in speaker on some laptops * after S2RAM. So, don't touch it. */ - /* pci_set_power_state(pci, pci_choose_state(pci, state)); */ + /* pci_set_power_state(pci, PCI_D3hot); */ return 0; } -static int intel8x0_resume(struct pci_dev *pci) +static int intel8x0_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct intel8x0 *chip = card->private_data; int i; @@ -2734,6 +2736,11 @@ static int intel8x0_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(intel8x0_pm, intel8x0_suspend, intel8x0_resume); +#define INTEL8X0_PM_OPS &intel8x0_pm +#else +#define INTEL8X0_PM_OPS NULL #endif /* CONFIG_PM */ #define INTEL8X0_TESTBUF_SIZE 32768 /* enough large for one shot */ @@ -3343,10 +3350,9 @@ static struct pci_driver intel8x0_driver = { .id_table = snd_intel8x0_ids, .probe = snd_intel8x0_probe, .remove = __devexit_p(snd_intel8x0_remove), -#ifdef CONFIG_PM - .suspend = intel8x0_suspend, - .resume = intel8x0_resume, -#endif + .driver = { + .pm = INTEL8X0_PM_OPS, + }, }; module_pci_driver(intel8x0_driver); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index fc27a6a69e77..da44bb3f8e7a 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1012,9 +1012,10 @@ static int snd_intel8x0m_free(struct intel8x0m *chip) /* * power management */ -static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state) +static int intel8x0m_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct intel8x0m *chip = card->private_data; int i; @@ -1028,13 +1029,14 @@ static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state) } pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int intel8x0m_resume(struct pci_dev *pci) +static int intel8x0m_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct intel8x0m *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -1060,6 +1062,11 @@ static int intel8x0m_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(intel8x0m_pm, intel8x0m_suspend, intel8x0m_resume); +#define INTEL8X0M_PM_OPS &intel8x0m_pm +#else +#define INTEL8X0M_PM_OPS NULL #endif /* CONFIG_PM */ #ifdef CONFIG_PROC_FS @@ -1329,10 +1336,9 @@ static struct pci_driver intel8x0m_driver = { .id_table = snd_intel8x0m_ids, .probe = snd_intel8x0m_probe, .remove = __devexit_p(snd_intel8x0m_remove), -#ifdef CONFIG_PM - .suspend = intel8x0m_suspend, - .resume = intel8x0m_resume, -#endif + .driver = { + .pm = INTEL8X0M_PM_OPS, + }, }; module_pci_driver(intel8x0m_driver); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index deef21399586..c85d1ffcc955 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -361,74 +361,6 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)"); #define DSP2HOST_REQ_I2SRATE 0x02 #define DSP2HOST_REQ_TIMER 0x04 -/* AC97 registers */ -/* XXX fix this crap up */ -/*#define AC97_RESET 0x00*/ - -#define AC97_VOL_MUTE_B 0x8000 -#define AC97_VOL_M 0x1F -#define AC97_LEFT_VOL_S 8 - -#define AC97_MASTER_VOL 0x02 -#define AC97_LINE_LEVEL_VOL 0x04 -#define AC97_MASTER_MONO_VOL 0x06 -#define AC97_PC_BEEP_VOL 0x0A -#define AC97_PC_BEEP_VOL_M 0x0F -#define AC97_SROUND_MASTER_VOL 0x38 -#define AC97_PC_BEEP_VOL_S 1 - -/*#define AC97_PHONE_VOL 0x0C -#define AC97_MIC_VOL 0x0E*/ -#define AC97_MIC_20DB_ENABLE 0x40 - -/*#define AC97_LINEIN_VOL 0x10 -#define AC97_CD_VOL 0x12 -#define AC97_VIDEO_VOL 0x14 -#define AC97_AUX_VOL 0x16*/ -#define AC97_PCM_OUT_VOL 0x18 -/*#define AC97_RECORD_SELECT 0x1A*/ -#define AC97_RECORD_MIC 0x00 -#define AC97_RECORD_CD 0x01 -#define AC97_RECORD_VIDEO 0x02 -#define AC97_RECORD_AUX 0x03 -#define AC97_RECORD_MONO_MUX 0x02 -#define AC97_RECORD_DIGITAL 0x03 -#define AC97_RECORD_LINE 0x04 -#define AC97_RECORD_STEREO 0x05 -#define AC97_RECORD_MONO 0x06 -#define AC97_RECORD_PHONE 0x07 - -/*#define AC97_RECORD_GAIN 0x1C*/ -#define AC97_RECORD_VOL_M 0x0F - -/*#define AC97_GENERAL_PURPOSE 0x20*/ -#define AC97_POWER_DOWN_CTRL 0x26 -#define AC97_ADC_READY 0x0001 -#define AC97_DAC_READY 0x0002 -#define AC97_ANALOG_READY 0x0004 -#define AC97_VREF_ON 0x0008 -#define AC97_PR0 0x0100 -#define AC97_PR1 0x0200 -#define AC97_PR2 0x0400 -#define AC97_PR3 0x0800 -#define AC97_PR4 0x1000 - -#define AC97_RESERVED1 0x28 - -#define AC97_VENDOR_TEST 0x5A - -#define AC97_CLOCK_DELAY 0x5C -#define AC97_LINEOUT_MUX_SEL 0x0001 -#define AC97_MONO_MUX_SEL 0x0002 -#define AC97_CLOCK_DELAY_SEL 0x1F -#define AC97_DAC_CDS_SHIFT 6 -#define AC97_ADC_CDS_SHIFT 11 - -#define AC97_MULTI_CHANNEL_SEL 0x74 - -/*#define AC97_VENDOR_ID1 0x7C -#define AC97_VENDOR_ID2 0x7E*/ - /* * ASSP control regs */ @@ -2459,9 +2391,10 @@ static int snd_m3_free(struct snd_m3 *chip) * APM support */ #ifdef CONFIG_PM -static int m3_suspend(struct pci_dev *pci, pm_message_t state) +static int m3_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_m3 *chip = card->private_data; int i, dsp_index; @@ -2489,13 +2422,14 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int m3_resume(struct pci_dev *pci) +static int m3_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_m3 *chip = card->private_data; int i, dsp_index; @@ -2546,6 +2480,11 @@ static int m3_resume(struct pci_dev *pci) chip->in_suspend = 0; return 0; } + +static SIMPLE_DEV_PM_OPS(m3_pm, m3_suspend, m3_resume); +#define M3_PM_OPS &m3_pm +#else +#define M3_PM_OPS NULL #endif /* CONFIG_PM */ #ifdef CONFIG_SND_MAESTRO3_INPUT @@ -2842,10 +2781,9 @@ static struct pci_driver m3_driver = { .id_table = snd_m3_ids, .probe = snd_m3_probe, .remove = __devexit_p(snd_m3_remove), -#ifdef CONFIG_PM - .suspend = m3_suspend, - .resume = m3_resume, -#endif + .driver = { + .pm = M3_PM_OPS, + }, }; module_pci_driver(m3_driver); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 8159b05ee94d..465cff25b146 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1382,9 +1382,10 @@ snd_nm256_peek_for_sig(struct nm256 *chip) * APM event handler, so the card is properly reinitialized after a power * event. */ -static int nm256_suspend(struct pci_dev *pci, pm_message_t state) +static int nm256_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct nm256 *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -1393,13 +1394,14 @@ static int nm256_suspend(struct pci_dev *pci, pm_message_t state) chip->coeffs_current = 0; pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int nm256_resume(struct pci_dev *pci) +static int nm256_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct nm256 *chip = card->private_data; int i; @@ -1434,6 +1436,11 @@ static int nm256_resume(struct pci_dev *pci) chip->in_resume = 0; return 0; } + +static SIMPLE_DEV_PM_OPS(nm256_pm, nm256_suspend, nm256_resume); +#define NM256_PM_OPS &nm256_pm +#else +#define NM256_PM_OPS NULL #endif /* CONFIG_PM */ static int snd_nm256_free(struct nm256 *chip) @@ -1747,10 +1754,9 @@ static struct pci_driver nm256_driver = { .id_table = snd_nm256_ids, .probe = snd_nm256_probe, .remove = __devexit_p(snd_nm256_remove), -#ifdef CONFIG_PM - .suspend = nm256_suspend, - .resume = nm256_resume, -#endif + .driver = { + .pm = NM256_PM_OPS, + }, }; module_pci_driver(nm256_driver); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 610275bfbaeb..37520a2b4dcf 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -873,8 +873,9 @@ static struct pci_driver oxygen_driver = { .probe = generic_oxygen_probe, .remove = __devexit_p(oxygen_pci_remove), #ifdef CONFIG_PM - .suspend = oxygen_pci_suspend, - .resume = oxygen_pci_resume, + .driver = { + .pm = &oxygen_pci_pm, + }, #endif }; diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index f53897a708b4..7112a89fb8bd 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -162,8 +162,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, ); void oxygen_pci_remove(struct pci_dev *pci); #ifdef CONFIG_PM -int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); -int oxygen_pci_resume(struct pci_dev *pci); +extern const struct dev_pm_ops oxygen_pci_pm; #endif void oxygen_pci_shutdown(struct pci_dev *pci); diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 92e2d67f16a1..ab8738e21ad1 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -727,9 +727,10 @@ void oxygen_pci_remove(struct pci_dev *pci) EXPORT_SYMBOL(oxygen_pci_remove); #ifdef CONFIG_PM -int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state) +static int oxygen_pci_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct oxygen *chip = card->private_data; unsigned int i, saved_interrupt_mask; @@ -756,10 +757,9 @@ int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -EXPORT_SYMBOL(oxygen_pci_suspend); static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = { 0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff, @@ -787,9 +787,10 @@ static void oxygen_restore_ac97(struct oxygen *chip, unsigned int codec) chip->saved_ac97_registers[codec][i]); } -int oxygen_pci_resume(struct pci_dev *pci) +static int oxygen_pci_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct oxygen *chip = card->private_data; unsigned int i; @@ -820,7 +821,9 @@ int oxygen_pci_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -EXPORT_SYMBOL(oxygen_pci_resume); + +SIMPLE_DEV_PM_OPS(oxygen_pci_pm, oxygen_pci_suspend, oxygen_pci_resume); +EXPORT_SYMBOL(oxygen_pci_pm); #endif /* CONFIG_PM */ void oxygen_pci_shutdown(struct pci_dev *pci) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 19962c6d38c3..d3b606b69f3b 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -94,8 +94,9 @@ static struct pci_driver xonar_driver = { .probe = xonar_probe, .remove = __devexit_p(oxygen_pci_remove), #ifdef CONFIG_PM - .suspend = oxygen_pci_suspend, - .resume = oxygen_pci_resume, + .driver = { + .pm = &oxygen_pci_pm, + }, #endif .shutdown = oxygen_pci_shutdown, }; diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 0435f45e9513..e3ac1f768ff6 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1368,6 +1368,67 @@ static void pcxhr_proc_gpo_write(struct snd_info_entry *entry, } } +/* Access to the results of the CMD_GET_TIME_CODE RMH */ +#define TIME_CODE_VALID_MASK 0x00800000 +#define TIME_CODE_NEW_MASK 0x00400000 +#define TIME_CODE_BACK_MASK 0x00200000 +#define TIME_CODE_WAIT_MASK 0x00100000 + +/* Values for the CMD_MANAGE_SIGNAL RMH */ +#define MANAGE_SIGNAL_TIME_CODE 0x01 +#define MANAGE_SIGNAL_MIDI 0x02 + +/* linear time code read proc*/ +static void pcxhr_proc_ltc(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_pcxhr *chip = entry->private_data; + struct pcxhr_mgr *mgr = chip->mgr; + struct pcxhr_rmh rmh; + unsigned int ltcHrs, ltcMin, ltcSec, ltcFrm; + int err; + /* commands available when embedded DSP is running */ + if (!(mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX))) { + snd_iprintf(buffer, "no firmware loaded\n"); + return; + } + if (!mgr->capture_ltc) { + pcxhr_init_rmh(&rmh, CMD_MANAGE_SIGNAL); + rmh.cmd[0] |= MANAGE_SIGNAL_TIME_CODE; + err = pcxhr_send_msg(mgr, &rmh); + if (err) { + snd_iprintf(buffer, "ltc not activated (%d)\n", err); + return; + } + if (mgr->is_hr_stereo) + hr222_manage_timecode(mgr, 1); + else + pcxhr_write_io_num_reg_cont(mgr, REG_CONT_VALSMPTE, + REG_CONT_VALSMPTE, NULL); + mgr->capture_ltc = 1; + } + pcxhr_init_rmh(&rmh, CMD_GET_TIME_CODE); + err = pcxhr_send_msg(mgr, &rmh); + if (err) { + snd_iprintf(buffer, "ltc read error (err=%d)\n", err); + return ; + } + ltcHrs = 10*((rmh.stat[0] >> 8) & 0x3) + (rmh.stat[0] & 0xf); + ltcMin = 10*((rmh.stat[1] >> 16) & 0x7) + ((rmh.stat[1] >> 8) & 0xf); + ltcSec = 10*(rmh.stat[1] & 0x7) + ((rmh.stat[2] >> 16) & 0xf); + ltcFrm = 10*((rmh.stat[2] >> 8) & 0x3) + (rmh.stat[2] & 0xf); + + snd_iprintf(buffer, "timecode: %02u:%02u:%02u-%02u\n", + ltcHrs, ltcMin, ltcSec, ltcFrm); + snd_iprintf(buffer, "raw: 0x%04x%06x%06x\n", rmh.stat[0] & 0x00ffff, + rmh.stat[1] & 0xffffff, rmh.stat[2] & 0xffffff); + /*snd_iprintf(buffer, "dsp ref time: 0x%06x%06x\n", + rmh.stat[3] & 0xffffff, rmh.stat[4] & 0xffffff);*/ + if (!(rmh.stat[0] & TIME_CODE_VALID_MASK)) { + snd_iprintf(buffer, "warning: linear timecode not valid\n"); + } +} + static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) { struct snd_info_entry *entry; @@ -1383,6 +1444,8 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) entry->c.text.write = pcxhr_proc_gpo_write; entry->mode |= S_IWUSR; } + if (!snd_card_proc_new(chip->card, "ltc", &entry)) + snd_info_set_text_ops(entry, chip, pcxhr_proc_ltc); } /* end of proc interface */ diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index bda776c49884..a4c602c45173 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -103,6 +103,7 @@ struct pcxhr_mgr { unsigned int board_has_mic:1; /* if 1 the board has microphone input */ unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ unsigned int mono_capture:1; /* if 1 the board does mono capture */ + unsigned int capture_ltc:1; /* if 1 the board captures LTC input */ struct snd_dma_buffer hostport; diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 304411c1fe4b..b33db1e006e7 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -504,6 +504,8 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = { [CMD_FORMAT_STREAM_IN] = { 0x870000, 0, RMH_SSIZE_FIXED }, [CMD_STREAM_SAMPLE_COUNT] = { 0x902000, 2, RMH_SSIZE_FIXED }, [CMD_AUDIO_LEVEL_ADJUST] = { 0xc22000, 0, RMH_SSIZE_FIXED }, +[CMD_GET_TIME_CODE] = { 0x060000, 5, RMH_SSIZE_FIXED }, +[CMD_MANAGE_SIGNAL] = { 0x0f0000, 0, RMH_SSIZE_FIXED }, }; #ifdef CONFIG_SND_DEBUG_VERBOSE @@ -533,6 +535,8 @@ static char* cmd_names[] = { [CMD_FORMAT_STREAM_IN] = "CMD_FORMAT_STREAM_IN", [CMD_STREAM_SAMPLE_COUNT] = "CMD_STREAM_SAMPLE_COUNT", [CMD_AUDIO_LEVEL_ADJUST] = "CMD_AUDIO_LEVEL_ADJUST", +[CMD_GET_TIME_CODE] = "CMD_GET_TIME_CODE", +[CMD_MANAGE_SIGNAL] = "CMD_MANAGE_SIGNAL", }; #endif @@ -1133,13 +1137,12 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr, hw_sample_count = ((u_int64_t)rmh.stat[0]) << 24; hw_sample_count += (u_int64_t)rmh.stat[1]; - snd_printdd("stream %c%d : abs samples real(%ld) timer(%ld)\n", + snd_printdd("stream %c%d : abs samples real(%llu) timer(%llu)\n", stream->pipe->is_capture ? 'C' : 'P', stream->substream->number, - (long unsigned int)hw_sample_count, - (long unsigned int)(stream->timer_abs_periods + - stream->timer_period_frag + - mgr->granularity)); + hw_sample_count, + stream->timer_abs_periods + stream->timer_period_frag + + mgr->granularity); return hw_sample_count; } @@ -1243,10 +1246,18 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) if ((dsp_time_diff < 0) && (mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID)) { - snd_printdd("ERROR DSP TIME old(%d) new(%d) -> " - "resynchronize all streams\n", + /* handle dsp counter wraparound without resync */ + int tmp_diff = dsp_time_diff + PCXHR_DSP_TIME_MASK + 1; + snd_printdd("WARNING DSP timestamp old(%d) new(%d)", mgr->dsp_time_last, dsp_time_new); - mgr->dsp_time_err++; + if (tmp_diff > 0 && tmp_diff <= (2*mgr->granularity)) { + snd_printdd("-> timestamp wraparound OK: " + "diff=%d\n", tmp_diff); + dsp_time_diff = tmp_diff; + } else { + snd_printdd("-> resynchronize all streams\n"); + mgr->dsp_time_err++; + } } #ifdef CONFIG_SND_DEBUG_VERBOSE if (dsp_time_diff == 0) diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h index be0173796cdb..a81ab6b811e7 100644 --- a/sound/pci/pcxhr/pcxhr_core.h +++ b/sound/pci/pcxhr/pcxhr_core.h @@ -79,6 +79,8 @@ enum { CMD_FORMAT_STREAM_IN, /* cmd_len >= 4 stat_len = 0 */ CMD_STREAM_SAMPLE_COUNT, /* cmd_len = 2 stat_len = (2 * nb_stream) */ CMD_AUDIO_LEVEL_ADJUST, /* cmd_len = 3 stat_len = 0 */ + CMD_GET_TIME_CODE, /* cmd_len = 1 stat_len = 5 */ + CMD_MANAGE_SIGNAL, /* cmd_len = 1 stat_len = 0 */ CMD_LAST_INDEX }; @@ -116,7 +118,7 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh); #define IO_NUM_REG_OUT_ANA_LEVEL 20 #define IO_NUM_REG_IN_ANA_LEVEL 21 - +#define REG_CONT_VALSMPTE 0x000800 #define REG_CONT_UNMUTE_INPUTS 0x020000 /* parameters used with register IO_NUM_REG_STATUS */ diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c index 1cb82c0a9cb3..84fe57626eba 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.c +++ b/sound/pci/pcxhr/pcxhr_mix22.c @@ -53,6 +53,7 @@ #define PCXHR_DSP_RESET_DSP 0x01 #define PCXHR_DSP_RESET_MUTE 0x02 #define PCXHR_DSP_RESET_CODEC 0x08 +#define PCXHR_DSP_RESET_SMPTE 0x10 #define PCXHR_DSP_RESET_GPO_OFFSET 5 #define PCXHR_DSP_RESET_GPO_MASK 0x60 @@ -527,6 +528,16 @@ int hr222_write_gpo(struct pcxhr_mgr *mgr, int value) return 0; } +int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable) +{ + if (enable) + mgr->dsp_reset |= PCXHR_DSP_RESET_SMPTE; + else + mgr->dsp_reset &= ~PCXHR_DSP_RESET_SMPTE; + + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, mgr->dsp_reset); + return 0; +} int hr222_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel) diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h index 5a37a0007e8f..5971b9933f41 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.h +++ b/sound/pci/pcxhr/pcxhr_mix22.h @@ -34,6 +34,7 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr, int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value); int hr222_write_gpo(struct pcxhr_mgr *mgr, int value); +int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable); #define HR222_LINE_PLAYBACK_LEVEL_MIN 0 /* -25.5 dB */ #define HR222_LINE_PLAYBACK_ZERO_LEVEL 51 /* 0.0 dB */ diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index cbeb3f77350c..760ee467cd9a 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1151,9 +1151,10 @@ static void riptide_handleirq(unsigned long dev_id) } #ifdef CONFIG_PM -static int riptide_suspend(struct pci_dev *pci, pm_message_t state) +static int riptide_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_riptide *chip = card->private_data; chip->in_suspend = 1; @@ -1162,13 +1163,14 @@ static int riptide_suspend(struct pci_dev *pci, pm_message_t state) snd_ac97_suspend(chip->ac97); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int riptide_resume(struct pci_dev *pci) +static int riptide_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_riptide *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -1186,7 +1188,12 @@ static int riptide_resume(struct pci_dev *pci) chip->in_suspend = 0; return 0; } -#endif + +static SIMPLE_DEV_PM_OPS(riptide_pm, riptide_suspend, riptide_resume); +#define RIPTIDE_PM_OPS &riptide_pm +#else +#define RIPTIDE_PM_OPS NULL +#endif /* CONFIG_PM */ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) { @@ -2180,10 +2187,9 @@ static struct pci_driver driver = { .id_table = snd_riptide_ids, .probe = snd_card_riptide_probe, .remove = __devexit_p(snd_card_riptide_remove), -#ifdef CONFIG_PM - .suspend = riptide_suspend, - .resume = riptide_resume, -#endif + .driver = { + .pm = RIPTIDE_PM_OPS, + }, }; #ifdef SUPPORT_JOYSTICK diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 1552642765d6..512434efcc31 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1209,9 +1209,10 @@ static int sis_chip_init(struct sis7019 *sis) } #ifdef CONFIG_PM -static int sis_suspend(struct pci_dev *pci, pm_message_t state) +static int sis_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct sis7019 *sis = card->private_data; void __iomem *ioaddr = sis->ioaddr; int i; @@ -1241,13 +1242,14 @@ static int sis_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int sis_resume(struct pci_dev *pci) +static int sis_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct sis7019 *sis = card->private_data; void __iomem *ioaddr = sis->ioaddr; int i; @@ -1298,6 +1300,11 @@ error: snd_card_disconnect(card); return -EIO; } + +static SIMPLE_DEV_PM_OPS(sis_pm, sis_suspend, sis_resume); +#define SIS_PM_OPS &sis_pm +#else +#define SIS_PM_OPS NULL #endif /* CONFIG_PM */ static int sis_alloc_suspend(struct sis7019 *sis) @@ -1481,11 +1488,9 @@ static struct pci_driver sis7019_driver = { .id_table = snd_sis7019_ids, .probe = snd_sis7019_probe, .remove = __devexit_p(snd_sis7019_remove), - -#ifdef CONFIG_PM - .suspend = sis_suspend, - .resume = sis_resume, -#endif + .driver = { + .pm = SIS_PM_OPS, + }, }; module_pci_driver(sis7019_driver); diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 611983ec7321..d36e6ca147e1 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -26,7 +26,7 @@ #include <linux/time.h> #include <linux/module.h> #include <sound/core.h> -#include <sound/trident.h> +#include "trident.h" #include <sound/initval.h> MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, <audio@tridentmicro.com>"); @@ -178,8 +178,9 @@ static struct pci_driver trident_driver = { .probe = snd_trident_probe, .remove = __devexit_p(snd_trident_remove), #ifdef CONFIG_PM - .suspend = snd_trident_suspend, - .resume = snd_trident_resume, + .driver = { + .pm = &snd_trident_pm, + }, #endif }; diff --git a/include/sound/trident.h b/sound/pci/trident/trident.h index 9f191a0a1e19..5f110eb56e47 100644 --- a/include/sound/trident.h +++ b/sound/pci/trident/trident.h @@ -23,10 +23,10 @@ * */ -#include "pcm.h" -#include "mpu401.h" -#include "ac97_codec.h" -#include "util_mem.h" +#include <sound/pcm.h> +#include <sound/mpu401.h> +#include <sound/ac97_codec.h> +#include <sound/util_mem.h> #define TRIDENT_DEVICE_ID_DX ((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_DX) #define TRIDENT_DEVICE_ID_NX ((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_NX) @@ -430,8 +430,7 @@ void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voi void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice); void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice); void snd_trident_write_voice_regs(struct snd_trident * trident, struct snd_trident_voice *voice); -int snd_trident_suspend(struct pci_dev *pci, pm_message_t state); -int snd_trident_resume(struct pci_dev *pci); +extern const struct dev_pm_ops snd_trident_pm; /* TLB memory allocation */ struct snd_util_memblk *snd_trident_alloc_pages(struct snd_trident *trident, diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 61d3c0e8d4ce..94011dcae731 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -41,7 +41,7 @@ #include <sound/info.h> #include <sound/control.h> #include <sound/tlv.h> -#include <sound/trident.h> +#include "trident.h" #include <sound/asoundef.h> #include <asm/io.h> @@ -3920,9 +3920,10 @@ static void snd_trident_clear_voices(struct snd_trident * trident, unsigned shor } #ifdef CONFIG_PM -int snd_trident_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_trident_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_trident *trident = card->private_data; trident->in_suspend = 1; @@ -3936,13 +3937,14 @@ int snd_trident_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -int snd_trident_resume(struct pci_dev *pci) +static int snd_trident_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_trident *trident = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -3979,4 +3981,6 @@ int snd_trident_resume(struct pci_dev *pci) trident->in_suspend = 0; return 0; } + +SIMPLE_DEV_PM_OPS(snd_trident_pm, snd_trident_suspend, snd_trident_resume); #endif /* CONFIG_PM */ diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index f9779e23fe57..3102a579660b 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -29,7 +29,7 @@ #include <linux/mutex.h> #include <sound/core.h> -#include <sound/trident.h> +#include "trident.h" /* page arguments of these two macros are Trident page (4096 bytes), not like * aligned pages in others diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index b5afab48943e..0eb7245dd362 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2242,9 +2242,10 @@ static int snd_via82xx_chip_init(struct via82xx *chip) /* * power management */ -static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_via82xx_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct via82xx *chip = card->private_data; int i; @@ -2265,13 +2266,14 @@ static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_via82xx_resume(struct pci_dev *pci) +static int snd_via82xx_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct via82xx *chip = card->private_data; int i; @@ -2306,6 +2308,11 @@ static int snd_via82xx_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume); +#define SND_VIA82XX_PM_OPS &snd_via82xx_pm +#else +#define SND_VIA82XX_PM_OPS NULL #endif /* CONFIG_PM */ static int snd_via82xx_free(struct via82xx *chip) @@ -2624,10 +2631,9 @@ static struct pci_driver via82xx_driver = { .id_table = snd_via82xx_ids, .probe = snd_via82xx_probe, .remove = __devexit_p(snd_via82xx_remove), -#ifdef CONFIG_PM - .suspend = snd_via82xx_suspend, - .resume = snd_via82xx_resume, -#endif + .driver = { + .pm = SND_VIA82XX_PM_OPS, + }, }; module_pci_driver(via82xx_driver); diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 59fd47ed0a31..e886bc16999d 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1023,9 +1023,10 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) /* * power management */ -static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_via82xx_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct via82xx_modem *chip = card->private_data; int i; @@ -1039,13 +1040,14 @@ static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_via82xx_resume(struct pci_dev *pci) +static int snd_via82xx_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct via82xx_modem *chip = card->private_data; int i; @@ -1069,6 +1071,11 @@ static int snd_via82xx_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume); +#define SND_VIA82XX_PM_OPS &snd_via82xx_pm +#else +#define SND_VIA82XX_PM_OPS NULL #endif /* CONFIG_PM */ static int snd_via82xx_free(struct via82xx_modem *chip) @@ -1228,10 +1235,9 @@ static struct pci_driver via82xx_modem_driver = { .id_table = snd_via82xx_modem_ids, .probe = snd_via82xx_probe, .remove = __devexit_p(snd_via82xx_remove), -#ifdef CONFIG_PM - .suspend = snd_via82xx_suspend, - .resume = snd_via82xx_resume, -#endif + .driver = { + .pm = SND_VIA82XX_PM_OPS, + }, }; module_pci_driver(via82xx_modem_driver); diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index 1ea1f656a5dc..b89e7a86e9d8 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -258,22 +258,24 @@ static void __devexit snd_vx222_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int snd_vx222_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_vx222_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_vx222 *vx = card->private_data; int err; - err = snd_vx_suspend(&vx->core, state); + err = snd_vx_suspend(&vx->core); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return err; } -static int snd_vx222_resume(struct pci_dev *pci) +static int snd_vx222_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_vx222 *vx = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -287,6 +289,11 @@ static int snd_vx222_resume(struct pci_dev *pci) pci_set_master(pci); return snd_vx_resume(&vx->core); } + +static SIMPLE_DEV_PM_OPS(snd_vx222_pm, snd_vx222_suspend, snd_vx222_resume); +#define SND_VX222_PM_OPS &snd_vx222_pm +#else +#define SND_VX222_PM_OPS NULL #endif static struct pci_driver vx222_driver = { @@ -294,10 +301,9 @@ static struct pci_driver vx222_driver = { .id_table = snd_vx222_ids, .probe = snd_vx222_probe, .remove = __devexit_p(snd_vx222_remove), -#ifdef CONFIG_PM - .suspend = snd_vx222_suspend, - .resume = snd_vx222_resume, -#endif + .driver = { + .pm = SND_VX222_PM_OPS, + }, }; module_pci_driver(vx222_driver); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 9a1d01d653a7..4810356b97ba 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -24,7 +24,7 @@ #include <linux/time.h> #include <linux/module.h> #include <sound/core.h> -#include <sound/ymfpci.h> +#include "ymfpci.h" #include <sound/mpu401.h> #include <sound/opl3.h> #include <sound/initval.h> @@ -356,8 +356,9 @@ static struct pci_driver ymfpci_driver = { .probe = snd_card_ymfpci_probe, .remove = __devexit_p(snd_card_ymfpci_remove), #ifdef CONFIG_PM - .suspend = snd_ymfpci_suspend, - .resume = snd_ymfpci_resume, + .driver = { + .pm = &snd_ymfpci_pm, + }, #endif }; diff --git a/include/sound/ymfpci.h b/sound/pci/ymfpci/ymfpci.h index 41199664666b..bddc4052286b 100644 --- a/include/sound/ymfpci.h +++ b/sound/pci/ymfpci/ymfpci.h @@ -22,10 +22,10 @@ * */ -#include "pcm.h" -#include "rawmidi.h" -#include "ac97_codec.h" -#include "timer.h" +#include <sound/pcm.h> +#include <sound/rawmidi.h> +#include <sound/ac97_codec.h> +#include <sound/timer.h> #include <linux/gameport.h> /* @@ -377,8 +377,7 @@ int snd_ymfpci_create(struct snd_card *card, struct snd_ymfpci ** rcodec); void snd_ymfpci_free_gameport(struct snd_ymfpci *chip); -int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state); -int snd_ymfpci_resume(struct pci_dev *pci); +extern const struct dev_pm_ops snd_ymfpci_pm; int snd_ymfpci_pcm(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); int snd_ymfpci_pcm2(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index a8159b81e9c4..62b23635b754 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -33,7 +33,7 @@ #include <sound/control.h> #include <sound/info.h> #include <sound/tlv.h> -#include <sound/ymfpci.h> +#include "ymfpci.h" #include <sound/asoundef.h> #include <sound/mpu401.h> @@ -2302,9 +2302,10 @@ static int saved_regs_index[] = { }; #define YDSXGR_NUM_SAVED_REGS ARRAY_SIZE(saved_regs_index) -int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_ymfpci_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ymfpci *chip = card->private_data; unsigned int i; @@ -2326,13 +2327,14 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) snd_ymfpci_disable_dsp(chip); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -int snd_ymfpci_resume(struct pci_dev *pci) +static int snd_ymfpci_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ymfpci *chip = card->private_data; unsigned int i; @@ -2370,6 +2372,8 @@ int snd_ymfpci_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +SIMPLE_DEV_PM_OPS(snd_ymfpci_pm, snd_ymfpci_suspend, snd_ymfpci_resume); #endif /* CONFIG_PM */ int __devinit snd_ymfpci_create(struct snd_card *card, diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 830839a874b6..f9b5229b2723 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -251,7 +251,7 @@ static int pdacf_suspend(struct pcmcia_device *link) snd_printdd(KERN_DEBUG "SUSPEND\n"); if (chip) { snd_printdd(KERN_DEBUG "snd_pdacf_suspend calling\n"); - snd_pdacf_suspend(chip, PMSG_SUSPEND); + snd_pdacf_suspend(chip); } return 0; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h index 6ce9ad700290..ea41e57d7179 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.h +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h @@ -131,7 +131,7 @@ struct snd_pdacf *snd_pdacf_create(struct snd_card *card); int snd_pdacf_ak4117_create(struct snd_pdacf *pdacf); void snd_pdacf_powerdown(struct snd_pdacf *chip); #ifdef CONFIG_PM -int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state); +int snd_pdacf_suspend(struct snd_pdacf *chip); int snd_pdacf_resume(struct snd_pdacf *chip); #endif int snd_pdacf_pcm_new(struct snd_pdacf *chip); diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index 9dce0bde5c05..ea0adfb984ad 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -262,7 +262,7 @@ void snd_pdacf_powerdown(struct snd_pdacf *chip) #ifdef CONFIG_PM -int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state) +int snd_pdacf_suspend(struct snd_pdacf *chip) { u16 val; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 512f0b472375..8f9350475c7b 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -260,7 +260,7 @@ static int vxp_suspend(struct pcmcia_device *link) snd_printdd(KERN_DEBUG "SUSPEND\n"); if (chip) { snd_printdd(KERN_DEBUG "snd_vx_suspend calling\n"); - snd_vx_suspend(chip, PMSG_SUSPEND); + snd_vx_suspend(chip); } return 0; diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 5a4e263b5b0f..f5ceb6f282de 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -144,19 +144,24 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr) } #ifdef CONFIG_PM -static int snd_pmac_driver_suspend(struct platform_device *devptr, pm_message_t state) +static int snd_pmac_driver_suspend(struct device *dev) { - struct snd_card *card = platform_get_drvdata(devptr); + struct snd_card *card = dev_get_drvdata(dev); snd_pmac_suspend(card->private_data); return 0; } -static int snd_pmac_driver_resume(struct platform_device *devptr) +static int snd_pmac_driver_resume(struct device *dev) { - struct snd_card *card = platform_get_drvdata(devptr); + struct snd_card *card = dev_get_drvdata(dev); snd_pmac_resume(card->private_data); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_pmac_pm, snd_pmac_driver_suspend, snd_pmac_driver_resume); +#define SND_PMAC_PM_OPS &snd_pmac_pm +#else +#define SND_PMAC_PM_OPS NULL #endif #define SND_PMAC_DRIVER "snd_powermac" @@ -164,12 +169,10 @@ static int snd_pmac_driver_resume(struct platform_device *devptr) static struct platform_driver snd_pmac_driver = { .probe = snd_pmac_probe, .remove = __devexit_p(snd_pmac_remove), -#ifdef CONFIG_PM - .suspend = snd_pmac_driver_suspend, - .resume = snd_pmac_driver_resume, -#endif .driver = { - .name = SND_PMAC_DRIVER + .name = SND_PMAC_DRIVER, + .owner = THIS_MODULE, + .pm = SND_PMAC_PM_OPS, }, }; diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 391a38ca58bc..d48b523207eb 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -654,7 +654,9 @@ static struct platform_driver snd_aica_driver = { .probe = snd_aica_probe, .remove = __devexit_p(snd_aica_remove), .driver = { - .name = SND_AICA_DRIVER}, + .name = SND_AICA_DRIVER, + .owner = THIS_MODULE, + }, }; static int __init aica_init(void) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index f8b01c77b298..0a3394751ed2 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -438,6 +438,7 @@ static struct platform_driver sh_dac_driver = { .remove = snd_sh_dac_remove, .driver = { .name = "dac_audio", + .owner = THIS_MODULE, }, }; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 40b2ad1bb1cd..c5de0a84566f 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,6 +33,7 @@ source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/davinci/Kconfig" +source "sound/soc/dwc/Kconfig" source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/jz4740/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 70990f4017f4..00a555a743b6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -11,6 +11,7 @@ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += davinci/ +obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += jz4740/ diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 9f6bc55fc399..16b88f5c26e2 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,7 +1,8 @@ config SND_BF5XX_I2S - tristate "SoC I2S Audio for the ADI BF5xx chip" + tristate "SoC I2S Audio for the ADI Blackfin chip" depends on BLACKFIN - select SND_BF5XX_SOC_SPORT + select SND_BF5XX_SOC_SPORT if !BF60x + select SND_BF6XX_SOC_SPORT if BF60x help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -9,12 +10,14 @@ config SND_BF5XX_I2S You will also need to select the audio interfaces to support below. config SND_BF5XX_SOC_SSM2602 - tristate "SoC SSM2602 Audio support for BF52x ezkit" + tristate "SoC SSM2602 Audio Codec Add-On Card support" depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) - select SND_BF5XX_SOC_I2S + select SND_BF5XX_SOC_I2S if !BF60x + select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_SSM2602 help - Say Y if you want to add support for SoC audio on BF527-EZKIT. + Say Y if you want to add support for the Analog Devices + SSM2602 Audio Codec Add-On Card. config SND_SOC_BFIN_EVAL_ADAU1701 tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" @@ -162,9 +165,15 @@ config SND_BF5XX_SOC_AD1980 config SND_BF5XX_SOC_SPORT tristate +config SND_BF6XX_SOC_SPORT + tristate + config SND_BF5XX_SOC_I2S tristate +config SND_BF6XX_SOC_I2S + tristate + config SND_BF5XX_SOC_TDM tristate @@ -173,7 +182,7 @@ config SND_BF5XX_SOC_AC97 config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) + depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT) range 0 3 if BF54x range 0 1 if !BF54x default 0 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 1bf86ccaa8de..6fea1f4cbee2 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -3,16 +3,20 @@ snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o +snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o +snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o +obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o +obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c new file mode 100644 index 000000000000..c3c2466d3a42 --- /dev/null +++ b/sound/soc/blackfin/bf6xx-i2s.c @@ -0,0 +1,234 @@ +/* + * bf6xx-i2s.c - Analog Devices BF6XX i2s interface driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include <linux/device.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "bf6xx-sport.h" + +struct sport_params param; + +static int bfin_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sport->pdev->dev; + int ret = 0; + + param.spctl &= ~(SPORT_CTL_OPMODE | SPORT_CTL_CKRE | SPORT_CTL_FSR + | SPORT_CTL_LFS | SPORT_CTL_LAFS); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_CKRE + | SPORT_CTL_LFS; + break; + case SND_SOC_DAIFMT_DSP_A: + param.spctl |= SPORT_CTL_FSR; + break; + case SND_SOC_DAIFMT_LEFT_J: + param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_LFS + | SPORT_CTL_LAFS; + break; + default: + dev_err(dev, "%s: Unknown DAI format type\n", __func__); + ret = -EINVAL; + break; + } + + param.spctl &= ~(SPORT_CTL_ICLK | SPORT_CTL_IFS); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + dev_err(dev, "%s: Unknown DAI master type\n", __func__); + ret = -EINVAL; + break; + } + + return ret; +} + +static int bfin_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + struct device *dev = &sport->pdev->dev; + int ret = 0; + + param.spctl &= ~SPORT_CTL_SLEN; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + param.spctl |= 0x70; + sport->wdsize = 1; + case SNDRV_PCM_FORMAT_S16_LE: + param.spctl |= 0xf0; + sport->wdsize = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + param.spctl |= 0x170; + sport->wdsize = 3; + break; + case SNDRV_PCM_FORMAT_S32_LE: + param.spctl |= 0x1f0; + sport->wdsize = 4; + break; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = sport_set_tx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT tx is busy!\n"); + return ret; + } + } else { + ret = sport_set_rx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT rx is busy!\n"); + return ret; + } + } + return 0; +} + +#ifdef CONFIG_PM +static int bfin_i2s_suspend(struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + + if (dai->capture_active) + sport_rx_stop(sport); + if (dai->playback_active) + sport_tx_stop(sport); + return 0; +} + +static int bfin_i2s_resume(struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + struct device *dev = &sport->pdev->dev; + int ret; + + ret = sport_set_tx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT tx is busy!\n"); + return ret; + } + ret = sport_set_rx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT rx is busy!\n"); + return ret; + } + + return 0; +} + +#else +#define bfin_i2s_suspend NULL +#define bfin_i2s_resume NULL +#endif + +#define BFIN_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define BFIN_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops bfin_i2s_dai_ops = { + .hw_params = bfin_i2s_hw_params, + .set_fmt = bfin_i2s_set_dai_fmt, +}; + +static struct snd_soc_dai_driver bfin_i2s_dai = { + .suspend = bfin_i2s_suspend, + .resume = bfin_i2s_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = BFIN_I2S_RATES, + .formats = BFIN_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = BFIN_I2S_RATES, + .formats = BFIN_I2S_FORMATS, + }, + .ops = &bfin_i2s_dai_ops, +}; + +static int __devinit bfin_i2s_probe(struct platform_device *pdev) +{ + struct sport_device *sport; + struct device *dev = &pdev->dev; + int ret; + + sport = sport_create(pdev); + if (!sport) + return -ENODEV; + + /* register with the ASoC layers */ + ret = snd_soc_register_dai(dev, &bfin_i2s_dai); + if (ret) { + dev_err(dev, "Failed to register DAI: %d\n", ret); + sport_delete(sport); + return ret; + } + platform_set_drvdata(pdev, sport); + + return 0; +} + +static int __devexit bfin_i2s_remove(struct platform_device *pdev) +{ + struct sport_device *sport = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&pdev->dev); + sport_delete(sport); + + return 0; +} + +static struct platform_driver bfin_i2s_driver = { + .probe = bfin_i2s_probe, + .remove = __devexit_p(bfin_i2s_remove), + .driver = { + .name = "bfin-i2s", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(bfin_i2s_driver); + +MODULE_DESCRIPTION("Analog Devices BF6XX i2s interface driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c new file mode 100644 index 000000000000..318c5ba5360f --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -0,0 +1,422 @@ +/* + * bf6xx_sport.c Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <asm/blackfin.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "bf6xx-sport.h" + +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN; + sport->tx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_tx_params); + +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN; + sport->rx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_rx_params); + +static int compute_wdsize(size_t wdsize) +{ + switch (wdsize) { + case 1: + return WDSIZE_8 | PSIZE_8; + case 2: + return WDSIZE_16 | PSIZE_16; + default: + return WDSIZE_32 | PSIZE_32; + } +} + +void sport_tx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc); + set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->tx_dma_chan); + sport->tx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_tx_start); + +void sport_rx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc); + set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->rx_dma_chan); + sport->rx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_rx_start); + +void sport_tx_stop(struct sport_device *sport) +{ + sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->tx_dma_chan); +} +EXPORT_SYMBOL(sport_tx_stop); + +void sport_rx_stop(struct sport_device *sport) +{ + sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->rx_dma_chan); +} +EXPORT_SYMBOL(sport_rx_stop); + +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data) +{ + sport->tx_callback = tx_callback; + sport->tx_data = tx_data; +} +EXPORT_SYMBOL(sport_set_tx_callback); + +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data) +{ + sport->rx_callback = rx_callback; + sport->rx_data = rx_data; +} +EXPORT_SYMBOL(sport_set_rx_callback); + +static void setup_desc(struct dmasg *desc, void *buf, int fragcount, + size_t fragsize, unsigned int cfg, + unsigned int count, size_t wdsize) +{ + + int i; + + for (i = 0; i < fragcount; ++i) { + desc[i].next_desc_addr = &(desc[i + 1]); + desc[i].start_addr = (unsigned long)buf + i*fragsize; + desc[i].cfg = cfg; + desc[i].x_count = count; + desc[i].x_modify = wdsize; + desc[i].y_count = 0; + desc[i].y_modify = 0; + } + + /* make circular */ + desc[fragcount-1].next_desc_addr = desc; +} + +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + + sport->tx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->tx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->tx_desc) + return -ENOMEM; + + sport->tx_buf = buf; + sport->tx_fragsize = fragsize; + sport->tx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6; + + setup_desc(sport->tx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_tx_dma); + +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); + + sport->rx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->rx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->rx_desc) + return -ENOMEM; + + sport->rx_buf = buf; + sport->rx_fragsize = fragsize; + sport->rx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) + | WNR | NDSIZE_6; + + setup_desc(sport->rx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_rx_dma); + +unsigned long sport_curr_offset_tx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan); + + return (unsigned char *)curr - sport->tx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_tx); + +unsigned long sport_curr_offset_rx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan); + + return (unsigned char *)curr - sport->rx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_rx); + +static irqreturn_t sport_tx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + static unsigned long status; + + status = get_dma_curr_irqstat(sport->tx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->tx_dma_chan); + SSYNC(); + } + if (sport->tx_callback) + sport->tx_callback(sport->tx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_rx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + unsigned long status; + + status = get_dma_curr_irqstat(sport->rx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->rx_dma_chan); + SSYNC(); + } + if (sport->rx_callback) + sport->rx_callback(sport->rx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_err_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + struct device *dev = &sport->pdev->dev; + + if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) + dev_err(dev, "sport error: TUVF\n"); + if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) + dev_err(dev, "sport error: ROVF\n"); + + return IRQ_HANDLED; +} + +static int sport_get_resource(struct sport_device *sport) +{ + struct platform_device *pdev = sport->pdev; + struct device *dev = &pdev->dev; + struct bfin_snd_platform_data *pdata = dev->platform_data; + struct resource *res; + + if (!pdata) { + dev_err(dev, "No platform data\n"); + return -ENODEV; + } + sport->pin_req = pdata->pin_req; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "No tx MEM resource\n"); + return -ENODEV; + } + sport->tx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res) { + dev_err(dev, "No rx MEM resource\n"); + return -ENODEV; + } + sport->rx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(dev, "No tx DMA resource\n"); + return -ENODEV; + } + sport->tx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(dev, "No rx DMA resource\n"); + return -ENODEV; + } + sport->rx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(dev, "No tx error irq resource\n"); + return -ENODEV; + } + sport->tx_err_irq = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 1); + if (!res) { + dev_err(dev, "No rx error irq resource\n"); + return -ENODEV; + } + sport->rx_err_irq = res->start; + + return 0; +} + +static int sport_request_resource(struct sport_device *sport) +{ + struct device *dev = &sport->pdev->dev; + int ret; + + ret = peripheral_request_list(sport->pin_req, "soc-audio"); + if (ret) { + dev_err(dev, "Unable to request sport pin\n"); + return ret; + } + + ret = request_dma(sport->tx_dma_chan, "SPORT TX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport tx\n"); + goto err_tx_dma; + } + set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport); + + ret = request_dma(sport->rx_dma_chan, "SPORT RX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport rx\n"); + goto err_rx_dma; + } + set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport); + + ret = request_irq(sport->tx_err_irq, sport_err_irq, + 0, "SPORT TX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate tx error IRQ for sport\n"); + goto err_tx_irq; + } + + ret = request_irq(sport->rx_err_irq, sport_err_irq, + 0, "SPORT RX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate rx error IRQ for sport\n"); + goto err_rx_irq; + } + + return 0; +err_rx_irq: + free_irq(sport->tx_err_irq, sport); +err_tx_irq: + free_dma(sport->rx_dma_chan); +err_rx_dma: + free_dma(sport->tx_dma_chan); +err_tx_dma: + peripheral_free_list(sport->pin_req); + return ret; +} + +static void sport_free_resource(struct sport_device *sport) +{ + free_irq(sport->rx_err_irq, sport); + free_irq(sport->tx_err_irq, sport); + free_dma(sport->rx_dma_chan); + free_dma(sport->tx_dma_chan); + peripheral_free_list(sport->pin_req); +} + +struct sport_device *sport_create(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct sport_device *sport; + int ret; + + sport = kzalloc(sizeof(*sport), GFP_KERNEL); + if (!sport) { + dev_err(dev, "Unable to allocate memory for sport device\n"); + return NULL; + } + sport->pdev = pdev; + + ret = sport_get_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + ret = sport_request_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + dev_dbg(dev, "SPORT create success\n"); + return sport; +} +EXPORT_SYMBOL(sport_create); + +void sport_delete(struct sport_device *sport) +{ + sport_free_resource(sport); +} +EXPORT_SYMBOL(sport_delete); + +MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h new file mode 100644 index 000000000000..307d193cfcef --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.h @@ -0,0 +1,82 @@ +/* + * bf6xx_sport - Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef _BF6XX_SPORT_H_ +#define _BF6XX_SPORT_H_ + +#include <linux/platform_device.h> +#include <asm/bfin_sport3.h> + +struct sport_device { + struct platform_device *pdev; + const unsigned short *pin_req; + struct sport_register *tx_regs; + struct sport_register *rx_regs; + int tx_dma_chan; + int rx_dma_chan; + int tx_err_irq; + int rx_err_irq; + + void (*tx_callback)(void *data); + void *tx_data; + void (*rx_callback)(void *data); + void *rx_data; + + struct dmasg *tx_desc; + struct dmasg *rx_desc; + unsigned int tx_desc_size; + unsigned int rx_desc_size; + unsigned char *tx_buf; + unsigned char *rx_buf; + unsigned int tx_fragsize; + unsigned int rx_fragsize; + unsigned int tx_frags; + unsigned int rx_frags; + unsigned int wdsize; +}; + +struct sport_params { + u32 spctl; + u32 div; +}; + +struct sport_device *sport_create(struct platform_device *pdev); +void sport_delete(struct sport_device *sport); +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params); +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params); +void sport_tx_start(struct sport_device *sport); +void sport_rx_start(struct sport_device *sport); +void sport_tx_stop(struct sport_device *sport); +void sport_rx_stop(struct sport_device *sport); +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data); +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data); +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +unsigned long sport_curr_offset_tx(struct sport_device *sport); +unsigned long sport_curr_offset_rx(struct sport_device *sport); + + + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1e1613a438dd..9f8e8594aeb9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -12,6 +12,7 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 + select SND_SOC_AB8500_CODEC if ABX500_CORE select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI @@ -35,7 +36,9 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C + select SND_SOC_DA732X if I2C select SND_SOC_DFBMCS320 + select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C @@ -54,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C + select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -70,6 +74,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM2000 if I2C select SND_SOC_WM2200 if I2C select SND_SOC_WM5100 if I2C + select SND_SOC_WM5102 if MFD_WM5102 + select SND_SOC_WM5110 if MFD_WM5110 select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI @@ -126,11 +132,21 @@ config SND_SOC_ALL_CODECS config SND_SOC_88PM860X tristate +config SND_SOC_ARIZONA + tristate + default y if SND_SOC_WM5102=y + default y if SND_SOC_WM5110=y + default m if SND_SOC_WM5102=m + default m if SND_SOC_WM5110=m + config SND_SOC_WM_HUBS tristate default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m +config SND_SOC_AB8500_CODEC + tristate + config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC @@ -219,12 +235,18 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate +config SND_SOC_DA732X + tristate + config SND_SOC_DFBMCS320 tristate config SND_SOC_DMIC tristate +config SND_SOC_ISABELLE + tristate + config SND_SOC_LM49453 tristate @@ -266,6 +288,9 @@ config SND_SOC_SSM2602 config SND_SOC_STA32X tristate +config SND_SOC_STA529 + tristate + config SND_SOC_STAC9766 tristate @@ -313,6 +338,12 @@ config SND_SOC_WM2200 config SND_SOC_WM5100 tristate +config SND_SOC_WM5102 + tristate + +config SND_SOC_WM5110 + tristate + config SND_SOC_WM8350 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fc27fec39487..34148bb59c68 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-88pm860x-objs := 88pm860x-codec.o +snd-soc-ab8500-codec-objs := ab8500-codec.o snd-soc-ac97-objs := ac97.o snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o @@ -13,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l52-objs := cs42l52.o @@ -21,8 +23,10 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o +snd-soc-da732x-objs := da732x.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o +snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o @@ -41,9 +45,11 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-sn95031-objs := sn95031.o -snd-soc-spdif-objs := spdif_transciever.o +snd-soc-spdif-tx-objs := spdif_transciever.o +snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -59,6 +65,8 @@ snd-soc-wm1250-ev1-objs := wm1250-ev1.o snd-soc-wm2000-objs := wm2000.o snd-soc-wm2200-objs := wm2200.o snd-soc-wm5100-objs := wm5100.o wm5100-tables.o +snd-soc-wm5102-objs := wm5102.o +snd-soc-wm5110-objs := wm5110.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o @@ -108,6 +116,7 @@ snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o +obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o @@ -124,6 +133,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o +obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o @@ -132,8 +142,10 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o +obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o @@ -150,9 +162,10 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o -obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o @@ -168,6 +181,8 @@ obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o +obj-$(CONFIG_SND_SOC_WM5102) += snd-soc-wm5102.o +obj-$(CONFIG_SND_SOC_WM5110) += snd-soc-wm5110.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c new file mode 100644 index 000000000000..3c795921c5f6 --- /dev/null +++ b/sound/soc/codecs/ab8500-codec.c @@ -0,0 +1,2522 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com>, + * for ST-Ericsson. + * + * Based on the early work done by: + * Mikko J. Lehto <mikko.lehto@symbio.com>, + * Mikko Sarmanne <mikko.sarmanne@symbio.com>, + * Jarmo K. Kuronen <jarmo.kuronen@symbio.com>, + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/mutex.h> +#include <linux/mfd/abx500/ab8500.h> +#include <linux/mfd/abx500.h> +#include <linux/mfd/abx500/ab8500-sysctrl.h> +#include <linux/mfd/abx500/ab8500-codec.h> +#include <linux/regulator/consumer.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "ab8500-codec.h" + +/* Macrocell value definitions */ +#define CLK_32K_OUT2_DISABLE 0x01 +#define INACTIVE_RESET_AUDIO 0x02 +#define ENABLE_AUDIO_CLK_TO_AUDIO_BLK 0x10 +#define ENABLE_VINTCORE12_SUPPLY 0x04 +#define GPIO27_DIR_OUTPUT 0x04 +#define GPIO29_DIR_OUTPUT 0x10 +#define GPIO31_DIR_OUTPUT 0x40 + +/* Macrocell register definitions */ +#define AB8500_CTRL3_REG 0x0200 +#define AB8500_GPIO_DIR4_REG 0x1013 + +/* Nr of FIR/IIR-coeff banks in ANC-block */ +#define AB8500_NR_OF_ANC_COEFF_BANKS 2 + +/* Minimum duration to keep ANC IIR Init bit high or +low before proceeding with the configuration sequence */ +#define AB8500_ANC_SM_DELAY 2000 + +#define AB8500_FILTER_CONTROL(xname, xcount, xmin, xmax) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = filter_control_info, \ + .get = filter_control_get, .put = filter_control_put, \ + .private_value = (unsigned long)&(struct filter_control) \ + {.count = xcount, .min = xmin, .max = xmax} } + +struct filter_control { + long min, max; + unsigned int count; + long value[128]; +}; + +/* Sidetone states */ +static const char * const enum_sid_state[] = { + "Unconfigured", + "Apply FIR", + "FIR is configured", +}; +enum sid_state { + SID_UNCONFIGURED = 0, + SID_APPLY_FIR = 1, + SID_FIR_CONFIGURED = 2, +}; + +static const char * const enum_anc_state[] = { + "Unconfigured", + "Apply FIR and IIR", + "FIR and IIR are configured", + "Apply FIR", + "FIR is configured", + "Apply IIR", + "IIR is configured" +}; +enum anc_state { + ANC_UNCONFIGURED = 0, + ANC_APPLY_FIR_IIR = 1, + ANC_FIR_IIR_CONFIGURED = 2, + ANC_APPLY_FIR = 3, + ANC_FIR_CONFIGURED = 4, + ANC_APPLY_IIR = 5, + ANC_IIR_CONFIGURED = 6 +}; + +/* Analog microphones */ +enum amic_idx { + AMIC_IDX_1A, + AMIC_IDX_1B, + AMIC_IDX_2 +}; + +struct ab8500_codec_drvdata_dbg { + struct regulator *vaud; + struct regulator *vamic1; + struct regulator *vamic2; + struct regulator *vdmic; +}; + +/* Private data for AB8500 device-driver */ +struct ab8500_codec_drvdata { + /* Sidetone */ + long *sid_fir_values; + enum sid_state sid_status; + + /* ANC */ + struct mutex anc_lock; + long *anc_fir_values; + long *anc_iir_values; + enum anc_state anc_status; +}; + +static inline const char *amic_micbias_str(enum amic_micbias micbias) +{ + switch (micbias) { + case AMIC_MICBIAS_VAMIC1: + return "VAMIC1"; + case AMIC_MICBIAS_VAMIC2: + return "VAMIC2"; + default: + return "Unknown"; + } +} + +static inline const char *amic_type_str(enum amic_type type) +{ + switch (type) { + case AMIC_TYPE_DIFFERENTIAL: + return "DIFFERENTIAL"; + case AMIC_TYPE_SINGLE_ENDED: + return "SINGLE ENDED"; + default: + return "Unknown"; + } +} + +/* + * Read'n'write functions + */ + +/* Read a register from the audio-bank of AB8500 */ +static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, + unsigned int reg) +{ + int status; + unsigned int value = 0; + + u8 value8; + status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, + reg, &value8); + if (status < 0) { + dev_err(codec->dev, + "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", + __func__, (u8)AB8500_AUDIO, (u8)reg, status); + } else { + dev_dbg(codec->dev, + "%s: Read 0x%02x from register 0x%02x:0x%02x\n", + __func__, value8, (u8)AB8500_AUDIO, (u8)reg); + value = (unsigned int)value8; + } + + return value; +} + +/* Write to a register in the audio-bank of AB8500 */ +static int ab8500_codec_write_reg(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int status; + + status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, + reg, value); + if (status < 0) + dev_err(codec->dev, + "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", + __func__, (u8)AB8500_AUDIO, (u8)reg, status); + else + dev_dbg(codec->dev, + "%s: Wrote 0x%02x into register %02x:%02x\n", + __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); + + return status; +} + +/* + * Controls - DAPM + */ + +/* Earpiece */ + +/* Earpiece source selector */ +static const char * const enum_ear_lineout_source[] = {"Headset Left", + "Speaker Left"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ear_lineout_source, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DA3TOEAR, enum_ear_lineout_source); +static const struct snd_kcontrol_new dapm_ear_lineout_source = + SOC_DAPM_ENUM("Earpiece or LineOut Mono Source", + dapm_enum_ear_lineout_source); + +/* LineOut */ + +/* LineOut source selector */ +static const char * const enum_lineout_source[] = {"Mono Path", "Stereo Path"}; +static SOC_ENUM_DOUBLE_DECL(dapm_enum_lineout_source, AB8500_ANACONF5, + AB8500_ANACONF5_HSLDACTOLOL, + AB8500_ANACONF5_HSRDACTOLOR, enum_lineout_source); +static const struct snd_kcontrol_new dapm_lineout_source[] = { + SOC_DAPM_ENUM("LineOut Source", dapm_enum_lineout_source), +}; + +/* Handsfree */ + +/* Speaker Left - ANC selector */ +static const char * const enum_HFx_sel[] = {"Audio Path", "ANC"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_HFl_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_HFLSEL, enum_HFx_sel); +static const struct snd_kcontrol_new dapm_HFl_select[] = { + SOC_DAPM_ENUM("Speaker Left Source", dapm_enum_HFl_sel), +}; + +/* Speaker Right - ANC selector */ +static SOC_ENUM_SINGLE_DECL(dapm_enum_HFr_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_HFRSEL, enum_HFx_sel); +static const struct snd_kcontrol_new dapm_HFr_select[] = { + SOC_DAPM_ENUM("Speaker Right Source", dapm_enum_HFr_sel), +}; + +/* Mic 1 */ + +/* Mic 1 - Mic 1a or 1b selector */ +static const char * const enum_mic1ab_sel[] = {"Mic 1b", "Mic 1a"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_mic1ab_sel, AB8500_ANACONF3, + AB8500_ANACONF3_MIC1SEL, enum_mic1ab_sel); +static const struct snd_kcontrol_new dapm_mic1ab_mux[] = { + SOC_DAPM_ENUM("Mic 1a or 1b Select", dapm_enum_mic1ab_sel), +}; + +/* Mic 1 - AD3 - Mic 1 or DMic 3 selector */ +static const char * const enum_ad3_sel[] = {"Mic 1", "DMic 3"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad3_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD3SEL, enum_ad3_sel); +static const struct snd_kcontrol_new dapm_ad3_select[] = { + SOC_DAPM_ENUM("AD3 Source Select", dapm_enum_ad3_sel), +}; + +/* Mic 1 - AD6 - Mic 1 or DMic 6 selector */ +static const char * const enum_ad6_sel[] = {"Mic 1", "DMic 6"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad6_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD6SEL, enum_ad6_sel); +static const struct snd_kcontrol_new dapm_ad6_select[] = { + SOC_DAPM_ENUM("AD6 Source Select", dapm_enum_ad6_sel), +}; + +/* Mic 2 */ + +/* Mic 2 - AD5 - Mic 2 or DMic 5 selector */ +static const char * const enum_ad5_sel[] = {"Mic 2", "DMic 5"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad5_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD5SEL, enum_ad5_sel); +static const struct snd_kcontrol_new dapm_ad5_select[] = { + SOC_DAPM_ENUM("AD5 Source Select", dapm_enum_ad5_sel), +}; + +/* LineIn */ + +/* LineIn left - AD1 - LineIn Left or DMic 1 selector */ +static const char * const enum_ad1_sel[] = {"LineIn Left", "DMic 1"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad1_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD1SEL, enum_ad1_sel); +static const struct snd_kcontrol_new dapm_ad1_select[] = { + SOC_DAPM_ENUM("AD1 Source Select", dapm_enum_ad1_sel), +}; + +/* LineIn right - Mic 2 or LineIn Right selector */ +static const char * const enum_mic2lr_sel[] = {"Mic 2", "LineIn Right"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_mic2lr_sel, AB8500_ANACONF3, + AB8500_ANACONF3_LINRSEL, enum_mic2lr_sel); +static const struct snd_kcontrol_new dapm_mic2lr_select[] = { + SOC_DAPM_ENUM("Mic 2 or LINR Select", dapm_enum_mic2lr_sel), +}; + +/* LineIn right - AD2 - LineIn Right or DMic2 selector */ +static const char * const enum_ad2_sel[] = {"LineIn Right", "DMic 2"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad2_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD2SEL, enum_ad2_sel); +static const struct snd_kcontrol_new dapm_ad2_select[] = { + SOC_DAPM_ENUM("AD2 Source Select", dapm_enum_ad2_sel), +}; + + +/* ANC */ + +static const char * const enum_anc_in_sel[] = {"Mic 1 / DMic 6", + "Mic 2 / DMic 5"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_anc_in_sel, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_ANCINSEL, enum_anc_in_sel); +static const struct snd_kcontrol_new dapm_anc_in_select[] = { + SOC_DAPM_ENUM("ANC Source", dapm_enum_anc_in_sel), +}; + +/* ANC - Enable/Disable */ +static const struct snd_kcontrol_new dapm_anc_enable[] = { + SOC_DAPM_SINGLE("Switch", AB8500_ANCCONF1, + AB8500_ANCCONF1_ENANC, 0, 0), +}; + +/* ANC to Earpiece - Mute */ +static const struct snd_kcontrol_new dapm_anc_ear_mute[] = { + SOC_DAPM_SINGLE("Switch", AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_ANCSEL, 1, 0), +}; + + + +/* Sidetone left */ + +/* Sidetone left - Input selector */ +static const char * const enum_stfir1_in_sel[] = { + "LineIn Left", "LineIn Right", "Mic 1", "Headset Left" +}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir1_in_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_FIRSID1SEL, enum_stfir1_in_sel); +static const struct snd_kcontrol_new dapm_stfir1_in_select[] = { + SOC_DAPM_ENUM("Sidetone Left Source", dapm_enum_stfir1_in_sel), +}; + +/* Sidetone right path */ + +/* Sidetone right - Input selector */ +static const char * const enum_stfir2_in_sel[] = { + "LineIn Right", "Mic 1", "DMic 4", "Headset Right" +}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir2_in_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_FIRSID2SEL, enum_stfir2_in_sel); +static const struct snd_kcontrol_new dapm_stfir2_in_select[] = { + SOC_DAPM_ENUM("Sidetone Right Source", dapm_enum_stfir2_in_sel), +}; + +/* Vibra */ + +static const char * const enum_pwm2vibx[] = {"Audio Path", "PWM Generator"}; + +static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib1, AB8500_PWMGENCONF1, + AB8500_PWMGENCONF1_PWMTOVIB1, enum_pwm2vibx); + +static const struct snd_kcontrol_new dapm_pwm2vib1[] = { + SOC_DAPM_ENUM("Vibra 1 Controller", dapm_enum_pwm2vib1), +}; + +static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib2, AB8500_PWMGENCONF1, + AB8500_PWMGENCONF1_PWMTOVIB2, enum_pwm2vibx); + +static const struct snd_kcontrol_new dapm_pwm2vib2[] = { + SOC_DAPM_ENUM("Vibra 2 Controller", dapm_enum_pwm2vib2), +}; + +/* + * DAPM-widgets + */ + +static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = { + + /* Clocks */ + SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"), + + /* Regulators */ + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0), + + /* Power */ + SND_SOC_DAPM_SUPPLY("Audio Power", + AB8500_POWERUP, AB8500_POWERUP_POWERUP, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Audio Analog Power", + AB8500_POWERUP, AB8500_POWERUP_ENANA, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* Main supply node */ + SND_SOC_DAPM_SUPPLY("Main Supply", SND_SOC_NOPM, 0, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* DA/AD */ + + SND_SOC_DAPM_INPUT("ADC Input"), + SND_SOC_DAPM_ADC("ADC", "ab8500_0c", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("DAC Output"), + + SND_SOC_DAPM_AIF_IN("DA_IN1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN5", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN6", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT57", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT68", NULL, 0, SND_SOC_NOPM, 0, 0), + + /* Headset path */ + + SND_SOC_DAPM_SUPPLY("Charge Pump", AB8500_ANACONF5, + AB8500_ANACONF5_ENCPHS, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DA1 Enable", "ab8500_0p", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA1, 0), + SND_SOC_DAPM_DAC("DA2 Enable", "ab8500_0p", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA2, 0), + + SND_SOC_DAPM_PGA("HSL Digital Volume", SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_PGA("HSR Digital Volume", SND_SOC_NOPM, 0, 0, + NULL, 0), + + SND_SOC_DAPM_DAC("HSL DAC", "ab8500_0p", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSL, 0), + SND_SOC_DAPM_DAC("HSR DAC", "ab8500_0p", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSR, 0), + SND_SOC_DAPM_MIXER("HSL DAC Mute", AB8500_MUTECONF, + AB8500_MUTECONF_MUTDACHSL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("HSR DAC Mute", AB8500_MUTECONF, + AB8500_MUTECONF_MUTDACHSR, 1, + NULL, 0), + SND_SOC_DAPM_DAC("HSL DAC Driver", "ab8500_0p", + AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSL, 0), + SND_SOC_DAPM_DAC("HSR DAC Driver", "ab8500_0p", + AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSR, 0), + + SND_SOC_DAPM_MIXER("HSL Mute", + AB8500_MUTECONF, AB8500_MUTECONF_MUTHSL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("HSR Mute", + AB8500_MUTECONF, AB8500_MUTECONF_MUTHSR, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("HSL Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHSL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HSR Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHSR, 0, + NULL, 0), + SND_SOC_DAPM_PGA("HSL Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_PGA("HSR Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Headset Left"), + SND_SOC_DAPM_OUTPUT("Headset Right"), + + /* LineOut path */ + + SND_SOC_DAPM_MUX("LineOut Source", + SND_SOC_NOPM, 0, 0, dapm_lineout_source), + + SND_SOC_DAPM_MIXER("LOL Disable HFL", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("LOR Disable HFR", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 1, + NULL, 0), + + SND_SOC_DAPM_MIXER("LOL Enable", + AB8500_ANACONF5, AB8500_ANACONF5_ENLOL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LOR Enable", + AB8500_ANACONF5, AB8500_ANACONF5_ENLOR, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("LineOut Left"), + SND_SOC_DAPM_OUTPUT("LineOut Right"), + + /* Earpiece path */ + + SND_SOC_DAPM_MUX("Earpiece or LineOut Mono Source", + SND_SOC_NOPM, 0, 0, &dapm_ear_lineout_source), + SND_SOC_DAPM_MIXER("EAR DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACEAR, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("EAR Mute", + AB8500_MUTECONF, AB8500_MUTECONF_MUTEAR, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("EAR Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENEAR, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Earpiece"), + + /* Handsfree path */ + + SND_SOC_DAPM_MIXER("DA3 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA3, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA4 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA4, 0, + NULL, 0), + SND_SOC_DAPM_MUX("Speaker Left Source", + SND_SOC_NOPM, 0, 0, dapm_HFl_select), + SND_SOC_DAPM_MUX("Speaker Right Source", + SND_SOC_NOPM, 0, 0, dapm_HFr_select), + SND_SOC_DAPM_MIXER("HFL DAC", AB8500_DAPATHCONF, + AB8500_DAPATHCONF_ENDACHFL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HFR DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHFR, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA4 or ANC path to HfR", + AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFREN, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA3 or ANC path to HfL", + AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFLEN, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HFL Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HFR Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Speaker Left"), + SND_SOC_DAPM_OUTPUT("Speaker Right"), + + /* Vibrator path */ + + SND_SOC_DAPM_INPUT("PWMGEN1"), + SND_SOC_DAPM_INPUT("PWMGEN2"), + + SND_SOC_DAPM_MIXER("DA5 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA5, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA6 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA6, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("VIB1 DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("VIB2 DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB2, 0, + NULL, 0), + SND_SOC_DAPM_MUX("Vibra 1 Controller", + SND_SOC_NOPM, 0, 0, dapm_pwm2vib1), + SND_SOC_DAPM_MUX("Vibra 2 Controller", + SND_SOC_NOPM, 0, 0, dapm_pwm2vib2), + SND_SOC_DAPM_MIXER("VIB1 Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENVIB1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("VIB2 Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENVIB2, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Vibra 1"), + SND_SOC_DAPM_OUTPUT("Vibra 2"), + + /* Mic 1 */ + + SND_SOC_DAPM_INPUT("Mic 1"), + + SND_SOC_DAPM_MUX("Mic 1a or 1b Select", + SND_SOC_NOPM, 0, 0, dapm_mic1ab_mux), + SND_SOC_DAPM_MIXER("MIC1 Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC1, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC1A V-AMICx Enable", + AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC1B V-AMICx Enable", + AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC1 ADC", + AB8500_ANACONF3, AB8500_ANACONF3_ENADCMIC, 0, + NULL, 0), + SND_SOC_DAPM_MUX("AD3 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad3_select), + SND_SOC_DAPM_MIXER("AD3 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD3 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34, 0, + NULL, 0), + + /* Mic 2 */ + + SND_SOC_DAPM_INPUT("Mic 2"), + + SND_SOC_DAPM_MIXER("MIC2 Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC2, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC2 V-AMICx Enable", AB8500_ANACONF2, + AB8500_ANACONF2_ENMIC2, 0, + NULL, 0), + + /* LineIn */ + + SND_SOC_DAPM_INPUT("LineIn Left"), + SND_SOC_DAPM_INPUT("LineIn Right"), + + SND_SOC_DAPM_MIXER("LINL Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTLINL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTLINR, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("LINL Enable", AB8500_ANACONF2, + AB8500_ANACONF2_ENLINL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR Enable", AB8500_ANACONF2, + AB8500_ANACONF2_ENLINR, 0, + NULL, 0), + + /* LineIn Bypass path */ + SND_SOC_DAPM_MIXER("LINL to HSL Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR to HSR Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + + /* LineIn, Mic 2 */ + SND_SOC_DAPM_MUX("Mic 2 or LINR Select", + SND_SOC_NOPM, 0, 0, dapm_mic2lr_select), + SND_SOC_DAPM_MIXER("LINL ADC", AB8500_ANACONF3, + AB8500_ANACONF3_ENADCLINL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR ADC", AB8500_ANACONF3, + AB8500_ANACONF3_ENADCLINR, 0, + NULL, 0), + SND_SOC_DAPM_MUX("AD1 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad1_select), + SND_SOC_DAPM_MUX("AD2 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad2_select), + SND_SOC_DAPM_MIXER("AD1 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD2 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + + SND_SOC_DAPM_MIXER("AD12 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD12, 0, + NULL, 0), + + /* HD Capture path */ + + SND_SOC_DAPM_MUX("AD5 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad5_select), + SND_SOC_DAPM_MUX("AD6 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad6_select), + SND_SOC_DAPM_MIXER("AD5 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD6 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD57 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD68 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0, + NULL, 0), + + /* Digital Microphone path */ + + SND_SOC_DAPM_INPUT("DMic 1"), + SND_SOC_DAPM_INPUT("DMic 2"), + SND_SOC_DAPM_INPUT("DMic 3"), + SND_SOC_DAPM_INPUT("DMic 4"), + SND_SOC_DAPM_INPUT("DMic 5"), + SND_SOC_DAPM_INPUT("DMic 6"), + + SND_SOC_DAPM_MIXER("DMIC1", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC2", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC2, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC3", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC3, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC4", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC4, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC5", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC5, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC6", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC6, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD4 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD4 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34, + 0, NULL, 0), + + /* Acoustical Noise Cancellation path */ + + SND_SOC_DAPM_INPUT("ANC Configure Input"), + SND_SOC_DAPM_OUTPUT("ANC Configure Output"), + + SND_SOC_DAPM_MUX("ANC Source", + SND_SOC_NOPM, 0, 0, + dapm_anc_in_select), + SND_SOC_DAPM_SWITCH("ANC", + SND_SOC_NOPM, 0, 0, + dapm_anc_enable), + SND_SOC_DAPM_SWITCH("ANC to Earpiece", + SND_SOC_NOPM, 0, 0, + dapm_anc_ear_mute), + + /* Sidetone Filter path */ + + SND_SOC_DAPM_MUX("Sidetone Left Source", + SND_SOC_NOPM, 0, 0, + dapm_stfir1_in_select), + SND_SOC_DAPM_MUX("Sidetone Right Source", + SND_SOC_NOPM, 0, 0, + dapm_stfir2_in_select), + SND_SOC_DAPM_MIXER("STFIR1 Control", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("STFIR2 Control", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("STFIR1 Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("STFIR2 Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), +}; + +/* + * DAPM-routes + */ +static const struct snd_soc_dapm_route ab8500_dapm_routes[] = { + /* Power AB8500 audio-block when AD/DA is active */ + {"Main Supply", NULL, "V-AUD"}, + {"Main Supply", NULL, "audioclk"}, + {"Main Supply", NULL, "Audio Power"}, + {"Main Supply", NULL, "Audio Analog Power"}, + + {"DAC", NULL, "ab8500_0p"}, + {"DAC", NULL, "Main Supply"}, + {"ADC", NULL, "ab8500_0c"}, + {"ADC", NULL, "Main Supply"}, + + /* ANC Configure */ + {"ANC Configure Input", NULL, "Main Supply"}, + {"ANC Configure Output", NULL, "ANC Configure Input"}, + + /* AD/DA */ + {"ADC", NULL, "ADC Input"}, + {"DAC Output", NULL, "DAC"}, + + /* Powerup charge pump if DA1/2 is in use */ + + {"DA_IN1", NULL, "ab8500_0p"}, + {"DA_IN1", NULL, "Charge Pump"}, + {"DA_IN2", NULL, "ab8500_0p"}, + {"DA_IN2", NULL, "Charge Pump"}, + + /* Headset path */ + + {"DA1 Enable", NULL, "DA_IN1"}, + {"DA2 Enable", NULL, "DA_IN2"}, + + {"HSL Digital Volume", NULL, "DA1 Enable"}, + {"HSR Digital Volume", NULL, "DA2 Enable"}, + + {"HSL DAC", NULL, "HSL Digital Volume"}, + {"HSR DAC", NULL, "HSR Digital Volume"}, + + {"HSL DAC Mute", NULL, "HSL DAC"}, + {"HSR DAC Mute", NULL, "HSR DAC"}, + + {"HSL DAC Driver", NULL, "HSL DAC Mute"}, + {"HSR DAC Driver", NULL, "HSR DAC Mute"}, + + {"HSL Mute", NULL, "HSL DAC Driver"}, + {"HSR Mute", NULL, "HSR DAC Driver"}, + + {"HSL Enable", NULL, "HSL Mute"}, + {"HSR Enable", NULL, "HSR Mute"}, + + {"HSL Volume", NULL, "HSL Enable"}, + {"HSR Volume", NULL, "HSR Enable"}, + + {"Headset Left", NULL, "HSL Volume"}, + {"Headset Right", NULL, "HSR Volume"}, + + /* HF or LineOut path */ + + {"DA_IN3", NULL, "ab8500_0p"}, + {"DA3 Channel Volume", NULL, "DA_IN3"}, + {"DA_IN4", NULL, "ab8500_0p"}, + {"DA4 Channel Volume", NULL, "DA_IN4"}, + + {"Speaker Left Source", "Audio Path", "DA3 Channel Volume"}, + {"Speaker Right Source", "Audio Path", "DA4 Channel Volume"}, + + {"DA3 or ANC path to HfL", NULL, "Speaker Left Source"}, + {"DA4 or ANC path to HfR", NULL, "Speaker Right Source"}, + + /* HF path */ + + {"HFL DAC", NULL, "DA3 or ANC path to HfL"}, + {"HFR DAC", NULL, "DA4 or ANC path to HfR"}, + + {"HFL Enable", NULL, "HFL DAC"}, + {"HFR Enable", NULL, "HFR DAC"}, + + {"Speaker Left", NULL, "HFL Enable"}, + {"Speaker Right", NULL, "HFR Enable"}, + + /* Earpiece path */ + + {"Earpiece or LineOut Mono Source", "Headset Left", + "HSL Digital Volume"}, + {"Earpiece or LineOut Mono Source", "Speaker Left", + "DA3 or ANC path to HfL"}, + + {"EAR DAC", NULL, "Earpiece or LineOut Mono Source"}, + + {"EAR Mute", NULL, "EAR DAC"}, + + {"EAR Enable", NULL, "EAR Mute"}, + + {"Earpiece", NULL, "EAR Enable"}, + + /* LineOut path stereo */ + + {"LineOut Source", "Stereo Path", "HSL DAC Driver"}, + {"LineOut Source", "Stereo Path", "HSR DAC Driver"}, + + /* LineOut path mono */ + + {"LineOut Source", "Mono Path", "EAR DAC"}, + + /* LineOut path */ + + {"LOL Disable HFL", NULL, "LineOut Source"}, + {"LOR Disable HFR", NULL, "LineOut Source"}, + + {"LOL Enable", NULL, "LOL Disable HFL"}, + {"LOR Enable", NULL, "LOR Disable HFR"}, + + {"LineOut Left", NULL, "LOL Enable"}, + {"LineOut Right", NULL, "LOR Enable"}, + + /* Vibrator path */ + + {"DA_IN5", NULL, "ab8500_0p"}, + {"DA5 Channel Volume", NULL, "DA_IN5"}, + {"DA_IN6", NULL, "ab8500_0p"}, + {"DA6 Channel Volume", NULL, "DA_IN6"}, + + {"VIB1 DAC", NULL, "DA5 Channel Volume"}, + {"VIB2 DAC", NULL, "DA6 Channel Volume"}, + + {"Vibra 1 Controller", "Audio Path", "VIB1 DAC"}, + {"Vibra 2 Controller", "Audio Path", "VIB2 DAC"}, + {"Vibra 1 Controller", "PWM Generator", "PWMGEN1"}, + {"Vibra 2 Controller", "PWM Generator", "PWMGEN2"}, + + {"VIB1 Enable", NULL, "Vibra 1 Controller"}, + {"VIB2 Enable", NULL, "Vibra 2 Controller"}, + + {"Vibra 1", NULL, "VIB1 Enable"}, + {"Vibra 2", NULL, "VIB2 Enable"}, + + + /* Mic 2 */ + + {"MIC2 V-AMICx Enable", NULL, "Mic 2"}, + + /* LineIn */ + {"LINL Mute", NULL, "LineIn Left"}, + {"LINR Mute", NULL, "LineIn Right"}, + + {"LINL Enable", NULL, "LINL Mute"}, + {"LINR Enable", NULL, "LINR Mute"}, + + /* LineIn, Mic 2 */ + {"Mic 2 or LINR Select", "LineIn Right", "LINR Enable"}, + {"Mic 2 or LINR Select", "Mic 2", "MIC2 V-AMICx Enable"}, + + {"LINL ADC", NULL, "LINL Enable"}, + {"LINR ADC", NULL, "Mic 2 or LINR Select"}, + + {"AD1 Source Select", "LineIn Left", "LINL ADC"}, + {"AD2 Source Select", "LineIn Right", "LINR ADC"}, + + {"AD1 Channel Volume", NULL, "AD1 Source Select"}, + {"AD2 Channel Volume", NULL, "AD2 Source Select"}, + + {"AD12 Enable", NULL, "AD1 Channel Volume"}, + {"AD12 Enable", NULL, "AD2 Channel Volume"}, + + {"AD_OUT1", NULL, "ab8500_0c"}, + {"AD_OUT1", NULL, "AD12 Enable"}, + {"AD_OUT2", NULL, "ab8500_0c"}, + {"AD_OUT2", NULL, "AD12 Enable"}, + + /* Mic 1 */ + + {"MIC1 Mute", NULL, "Mic 1"}, + + {"MIC1A V-AMICx Enable", NULL, "MIC1 Mute"}, + {"MIC1B V-AMICx Enable", NULL, "MIC1 Mute"}, + + {"Mic 1a or 1b Select", "Mic 1a", "MIC1A V-AMICx Enable"}, + {"Mic 1a or 1b Select", "Mic 1b", "MIC1B V-AMICx Enable"}, + + {"MIC1 ADC", NULL, "Mic 1a or 1b Select"}, + + {"AD3 Source Select", "Mic 1", "MIC1 ADC"}, + + {"AD3 Channel Volume", NULL, "AD3 Source Select"}, + + {"AD3 Enable", NULL, "AD3 Channel Volume"}, + + {"AD_OUT3", NULL, "ab8500_0c"}, + {"AD_OUT3", NULL, "AD3 Enable"}, + + /* HD Capture path */ + + {"AD5 Source Select", "Mic 2", "LINR ADC"}, + {"AD6 Source Select", "Mic 1", "MIC1 ADC"}, + + {"AD5 Channel Volume", NULL, "AD5 Source Select"}, + {"AD6 Channel Volume", NULL, "AD6 Source Select"}, + + {"AD57 Enable", NULL, "AD5 Channel Volume"}, + {"AD68 Enable", NULL, "AD6 Channel Volume"}, + + {"AD_OUT57", NULL, "ab8500_0c"}, + {"AD_OUT57", NULL, "AD57 Enable"}, + {"AD_OUT68", NULL, "ab8500_0c"}, + {"AD_OUT68", NULL, "AD68 Enable"}, + + /* Digital Microphone path */ + + {"DMic 1", NULL, "V-DMIC"}, + {"DMic 2", NULL, "V-DMIC"}, + {"DMic 3", NULL, "V-DMIC"}, + {"DMic 4", NULL, "V-DMIC"}, + {"DMic 5", NULL, "V-DMIC"}, + {"DMic 6", NULL, "V-DMIC"}, + + {"AD1 Source Select", NULL, "DMic 1"}, + {"AD2 Source Select", NULL, "DMic 2"}, + {"AD3 Source Select", NULL, "DMic 3"}, + {"AD5 Source Select", NULL, "DMic 5"}, + {"AD6 Source Select", NULL, "DMic 6"}, + + {"AD4 Channel Volume", NULL, "DMic 4"}, + {"AD4 Enable", NULL, "AD4 Channel Volume"}, + + {"AD_OUT4", NULL, "ab8500_0c"}, + {"AD_OUT4", NULL, "AD4 Enable"}, + + /* LineIn Bypass path */ + + {"LINL to HSL Volume", NULL, "LINL Enable"}, + {"LINR to HSR Volume", NULL, "LINR Enable"}, + + {"HSL DAC Driver", NULL, "LINL to HSL Volume"}, + {"HSR DAC Driver", NULL, "LINR to HSR Volume"}, + + /* ANC path (Acoustic Noise Cancellation) */ + + {"ANC Source", "Mic 2 / DMic 5", "AD5 Channel Volume"}, + {"ANC Source", "Mic 1 / DMic 6", "AD6 Channel Volume"}, + + {"ANC", "Switch", "ANC Source"}, + + {"Speaker Left Source", "ANC", "ANC"}, + {"Speaker Right Source", "ANC", "ANC"}, + {"ANC to Earpiece", "Switch", "ANC"}, + + {"HSL Digital Volume", NULL, "ANC to Earpiece"}, + + /* Sidetone Filter path */ + + {"Sidetone Left Source", "LineIn Left", "AD12 Enable"}, + {"Sidetone Left Source", "LineIn Right", "AD12 Enable"}, + {"Sidetone Left Source", "Mic 1", "AD3 Enable"}, + {"Sidetone Left Source", "Headset Left", "DA_IN1"}, + {"Sidetone Right Source", "LineIn Right", "AD12 Enable"}, + {"Sidetone Right Source", "Mic 1", "AD3 Enable"}, + {"Sidetone Right Source", "DMic 4", "AD4 Enable"}, + {"Sidetone Right Source", "Headset Right", "DA_IN2"}, + + {"STFIR1 Control", NULL, "Sidetone Left Source"}, + {"STFIR2 Control", NULL, "Sidetone Right Source"}, + + {"STFIR1 Volume", NULL, "STFIR1 Control"}, + {"STFIR2 Volume", NULL, "STFIR2 Control"}, + + {"DA1 Enable", NULL, "STFIR1 Volume"}, + {"DA2 Enable", NULL, "STFIR2 Volume"}, +}; + +static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1a_vamicx[] = { + {"MIC1A V-AMICx Enable", NULL, "V-AMIC1"}, + {"MIC1A V-AMICx Enable", NULL, "V-AMIC2"}, +}; + +static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1b_vamicx[] = { + {"MIC1B V-AMICx Enable", NULL, "V-AMIC1"}, + {"MIC1B V-AMICx Enable", NULL, "V-AMIC2"}, +}; + +static const struct snd_soc_dapm_route ab8500_dapm_routes_mic2_vamicx[] = { + {"MIC2 V-AMICx Enable", NULL, "V-AMIC1"}, + {"MIC2 V-AMICx Enable", NULL, "V-AMIC2"}, +}; + +/* ANC FIR-coefficients configuration sequence */ +static void anc_fir(struct snd_soc_codec *codec, + unsigned int bnk, unsigned int par, unsigned int val) +{ + if (par == 0 && bnk == 0) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCFIRUPDATE), + BIT(AB8500_ANCCONF1_ANCFIRUPDATE)); + + snd_soc_write(codec, AB8500_ANCCONF5, val >> 8 & 0xff); + snd_soc_write(codec, AB8500_ANCCONF6, val & 0xff); + + if (par == AB8500_ANC_FIR_COEFFS - 1 && bnk == 1) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCFIRUPDATE), 0); +} + +/* ANC IIR-coefficients configuration sequence */ +static void anc_iir(struct snd_soc_codec *codec, unsigned int bnk, + unsigned int par, unsigned int val) +{ + if (par == 0) { + if (bnk == 0) { + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRINIT), + BIT(AB8500_ANCCONF1_ANCIIRINIT)); + usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY); + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRINIT), 0); + usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY); + } else { + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRUPDATE), + BIT(AB8500_ANCCONF1_ANCIIRUPDATE)); + } + } else if (par > 3) { + snd_soc_write(codec, AB8500_ANCCONF7, 0); + snd_soc_write(codec, AB8500_ANCCONF8, val >> 16 & 0xff); + } + + snd_soc_write(codec, AB8500_ANCCONF7, val >> 8 & 0xff); + snd_soc_write(codec, AB8500_ANCCONF8, val & 0xff); + + if (par == AB8500_ANC_IIR_COEFFS - 1 && bnk == 1) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRUPDATE), 0); +} + +/* ANC IIR-/FIR-coefficients configuration sequence */ +static void anc_configure(struct snd_soc_codec *codec, + bool apply_fir, bool apply_iir) +{ + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + unsigned int bnk, par, val; + + dev_dbg(codec->dev, "%s: Enter.\n", __func__); + + if (apply_fir) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ENANC), 0); + + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ENANC), BIT(AB8500_ANCCONF1_ENANC)); + + if (apply_fir) + for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++) + for (par = 0; par < AB8500_ANC_FIR_COEFFS; par++) { + val = snd_soc_read(codec, + drvdata->anc_fir_values[par]); + anc_fir(codec, bnk, par, val); + } + + if (apply_iir) + for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++) + for (par = 0; par < AB8500_ANC_IIR_COEFFS; par++) { + val = snd_soc_read(codec, + drvdata->anc_iir_values[par]); + anc_iir(codec, bnk, par, val); + } + + dev_dbg(codec->dev, "%s: Exit.\n", __func__); +} + +/* + * Control-events + */ + +static int sid_status_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = drvdata->sid_status; + mutex_unlock(&codec->mutex); + + return 0; +} + +/* Write sidetone FIR-coefficients configuration sequence */ +static int sid_status_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + unsigned int param, sidconf, val; + int status = 1; + + dev_dbg(codec->dev, "%s: Enter\n", __func__); + + if (ucontrol->value.integer.value[0] != SID_APPLY_FIR) { + dev_err(codec->dev, + "%s: ERROR: This control supports '%s' only!\n", + __func__, enum_sid_state[SID_APPLY_FIR]); + return -EIO; + } + + mutex_lock(&codec->mutex); + + sidconf = snd_soc_read(codec, AB8500_SIDFIRCONF); + if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) { + if ((sidconf & BIT(AB8500_SIDFIRCONF_ENFIRSIDS)) == 0) { + dev_err(codec->dev, "%s: Sidetone busy while off!\n", + __func__); + status = -EPERM; + } else { + status = -EBUSY; + } + goto out; + } + + snd_soc_write(codec, AB8500_SIDFIRADR, 0); + + for (param = 0; param < AB8500_SID_FIR_COEFFS; param++) { + val = snd_soc_read(codec, drvdata->sid_fir_values[param]); + snd_soc_write(codec, AB8500_SIDFIRCOEF1, val >> 8 & 0xff); + snd_soc_write(codec, AB8500_SIDFIRCOEF2, val & 0xff); + } + + snd_soc_update_bits(codec, AB8500_SIDFIRADR, + BIT(AB8500_SIDFIRADR_FIRSIDSET), + BIT(AB8500_SIDFIRADR_FIRSIDSET)); + snd_soc_update_bits(codec, AB8500_SIDFIRADR, + BIT(AB8500_SIDFIRADR_FIRSIDSET), 0); + + drvdata->sid_status = SID_FIR_CONFIGURED; + +out: + mutex_unlock(&codec->mutex); + + dev_dbg(codec->dev, "%s: Exit\n", __func__); + + return status; +} + +static int anc_status_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = drvdata->anc_status; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int anc_status_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + struct device *dev = codec->dev; + bool apply_fir, apply_iir; + int req, status; + + dev_dbg(dev, "%s: Enter.\n", __func__); + + mutex_lock(&drvdata->anc_lock); + + req = ucontrol->value.integer.value[0]; + if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && + req != ANC_APPLY_IIR) { + dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", + __func__, enum_anc_state[req]); + status = -EINVAL; + goto cleanup; + } + apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR; + apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR; + + status = snd_soc_dapm_force_enable_pin(&codec->dapm, + "ANC Configure Input"); + if (status < 0) { + dev_err(dev, + "%s: ERROR: Failed to enable power (status = %d)!\n", + __func__, status); + goto cleanup; + } + snd_soc_dapm_sync(&codec->dapm); + + mutex_lock(&codec->mutex); + anc_configure(codec, apply_fir, apply_iir); + mutex_unlock(&codec->mutex); + + if (apply_fir) { + if (drvdata->anc_status == ANC_IIR_CONFIGURED) + drvdata->anc_status = ANC_FIR_IIR_CONFIGURED; + else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED) + drvdata->anc_status = ANC_FIR_CONFIGURED; + } + if (apply_iir) { + if (drvdata->anc_status == ANC_FIR_CONFIGURED) + drvdata->anc_status = ANC_FIR_IIR_CONFIGURED; + else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED) + drvdata->anc_status = ANC_IIR_CONFIGURED; + } + + status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input"); + snd_soc_dapm_sync(&codec->dapm); + +cleanup: + mutex_unlock(&drvdata->anc_lock); + + if (status < 0) + dev_err(dev, "%s: Unable to configure ANC! (status = %d)\n", + __func__, status); + + dev_dbg(dev, "%s: Exit.\n", __func__); + + return (status < 0) ? status : 1; +} + +static int filter_control_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct filter_control *fc = + (struct filter_control *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = fc->count; + uinfo->value.integer.min = fc->min; + uinfo->value.integer.max = fc->max; + + return 0; +} + +static int filter_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct filter_control *fc = + (struct filter_control *)kcontrol->private_value; + unsigned int i; + + mutex_lock(&codec->mutex); + for (i = 0; i < fc->count; i++) + ucontrol->value.integer.value[i] = fc->value[i]; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int filter_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct filter_control *fc = + (struct filter_control *)kcontrol->private_value; + unsigned int i; + + mutex_lock(&codec->mutex); + for (i = 0; i < fc->count; i++) + fc->value[i] = ucontrol->value.integer.value[i]; + mutex_unlock(&codec->mutex); + + return 0; +} + +/* + * Controls - Non-DAPM ASoC + */ + +static DECLARE_TLV_DB_SCALE(adx_dig_gain_tlv, -3200, 100, 1); +/* -32dB = Mute */ + +static DECLARE_TLV_DB_SCALE(dax_dig_gain_tlv, -6300, 100, 1); +/* -63dB = Mute */ + +static DECLARE_TLV_DB_SCALE(hs_ear_dig_gain_tlv, -100, 100, 1); +/* -1dB = Mute */ + +static const unsigned int hs_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-3200, 400, 0), + 4, 15, TLV_DB_SCALE_ITEM(-1800, 200, 0), +}; + +static DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0); + +static DECLARE_TLV_DB_SCALE(lin_gain_tlv, -1000, 200, 0); + +static DECLARE_TLV_DB_SCALE(lin2hs_gain_tlv, -3800, 200, 1); +/* -38dB = Mute */ + +static const char * const enum_hsfadspeed[] = {"2ms", "0.5ms", "10.6ms", + "5ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_hsfadspeed, + AB8500_DIGMICCONF, AB8500_DIGMICCONF_HSFADSPEED, enum_hsfadspeed); + +static const char * const enum_envdetthre[] = { + "250mV", "300mV", "350mV", "400mV", + "450mV", "500mV", "550mV", "600mV", + "650mV", "700mV", "750mV", "800mV", + "850mV", "900mV", "950mV", "1.00V" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_envdeththre, + AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETHTHRE, enum_envdetthre); +static SOC_ENUM_SINGLE_DECL(soc_enum_envdetlthre, + AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETLTHRE, enum_envdetthre); +static const char * const enum_envdettime[] = { + "26.6us", "53.2us", "106us", "213us", + "426us", "851us", "1.70ms", "3.40ms", + "6.81ms", "13.6ms", "27.2ms", "54.5ms", + "109ms", "218ms", "436ms", "872ms" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_envdettime, + AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETTIME, enum_envdettime); + +static const char * const enum_sinc31[] = {"Sinc 3", "Sinc 1"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_hsesinc, AB8500_HSLEARDIGGAIN, + AB8500_HSLEARDIGGAIN_HSSINC1, enum_sinc31); + +static const char * const enum_fadespeed[] = {"1ms", "4ms", "8ms", "16ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_fadespeed, AB8500_HSRDIGGAIN, + AB8500_HSRDIGGAIN_FADESPEED, enum_fadespeed); + +/* Earpiece */ + +static const char * const enum_lowpow[] = {"Normal", "Low Power"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_eardaclowpow, AB8500_ANACONF1, + AB8500_ANACONF1_EARDACLOWPOW, enum_lowpow); +static SOC_ENUM_SINGLE_DECL(soc_enum_eardrvlowpow, AB8500_ANACONF1, + AB8500_ANACONF1_EARDRVLOWPOW, enum_lowpow); + +static const char * const enum_av_mode[] = {"Audio", "Voice"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad12voice, AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD1VOICE, AB8500_ADFILTCONF_AD2VOICE, enum_av_mode); +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad34voice, AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD3VOICE, AB8500_ADFILTCONF_AD4VOICE, enum_av_mode); + +/* DA */ + +static SOC_ENUM_SINGLE_DECL(soc_enum_da12voice, + AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DA12VOICE, + enum_av_mode); +static SOC_ENUM_SINGLE_DECL(soc_enum_da34voice, + AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DA34VOICE, + enum_av_mode); +static SOC_ENUM_SINGLE_DECL(soc_enum_da56voice, + AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DA56VOICE, + enum_av_mode); + +static const char * const enum_da2hslr[] = {"Sidetone", "Audio Path"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_da2hslr, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_DATOHSLEN, + AB8500_DIGMULTCONF1_DATOHSREN, enum_da2hslr); + +static const char * const enum_sinc53[] = {"Sinc 5", "Sinc 3"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic12sinc, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DMIC1SINC3, + AB8500_DMICFILTCONF_DMIC2SINC3, enum_sinc53); +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic34sinc, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DMIC3SINC3, + AB8500_DMICFILTCONF_DMIC4SINC3, enum_sinc53); +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic56sinc, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DMIC5SINC3, + AB8500_DMICFILTCONF_DMIC6SINC3, enum_sinc53); + +/* Digital interface - DA from slot mapping */ +static const char * const enum_da_from_slot_map[] = {"SLOT0", + "SLOT1", + "SLOT2", + "SLOT3", + "SLOT4", + "SLOT5", + "SLOT6", + "SLOT7", + "SLOT8", + "SLOT9", + "SLOT10", + "SLOT11", + "SLOT12", + "SLOT13", + "SLOT14", + "SLOT15", + "SLOT16", + "SLOT17", + "SLOT18", + "SLOT19", + "SLOT20", + "SLOT21", + "SLOT22", + "SLOT23", + "SLOT24", + "SLOT25", + "SLOT26", + "SLOT27", + "SLOT28", + "SLOT29", + "SLOT30", + "SLOT31"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_da1slotmap, + AB8500_DASLOTCONF1, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da2slotmap, + AB8500_DASLOTCONF2, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da3slotmap, + AB8500_DASLOTCONF3, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da4slotmap, + AB8500_DASLOTCONF4, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da5slotmap, + AB8500_DASLOTCONF5, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da6slotmap, + AB8500_DASLOTCONF6, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da7slotmap, + AB8500_DASLOTCONF7, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da8slotmap, + AB8500_DASLOTCONF8, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); + +/* Digital interface - AD to slot mapping */ +static const char * const enum_ad_to_slot_map[] = {"AD_OUT1", + "AD_OUT2", + "AD_OUT3", + "AD_OUT4", + "AD_OUT5", + "AD_OUT6", + "AD_OUT7", + "AD_OUT8", + "zeroes", + "tristate"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot0map, + AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot1map, + AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot2map, + AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot3map, + AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot4map, + AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot5map, + AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot6map, + AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot7map, + AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot8map, + AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot9map, + AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot10map, + AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot11map, + AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot12map, + AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot13map, + AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot14map, + AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot15map, + AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot16map, + AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot17map, + AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot18map, + AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot19map, + AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot20map, + AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot21map, + AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot22map, + AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot23map, + AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot24map, + AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot25map, + AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot26map, + AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot27map, + AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot28map, + AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot29map, + AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot30map, + AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot31map, + AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); + +/* Digital interface - Burst mode */ +static const char * const enum_mask[] = {"Unmasked", "Masked"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomask, + AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOMASK, + enum_mask); +static const char * const enum_bitclk0[] = {"19_2_MHz", "38_4_MHz"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifo19m2, + AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFO19M2, + enum_bitclk0); +static const char * const enum_slavemaster[] = {"Slave", "Master"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomast, + AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOMAST_SHIFT, + enum_slavemaster); + +/* Sidetone */ +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_sidstate, enum_sid_state); + +/* ANC */ +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_ancstate, enum_anc_state); + +static struct snd_kcontrol_new ab8500_ctrls[] = { + /* Charge pump */ + SOC_ENUM("Charge Pump High Threshold For Low Voltage", + soc_enum_envdeththre), + SOC_ENUM("Charge Pump Low Threshold For Low Voltage", + soc_enum_envdetlthre), + SOC_SINGLE("Charge Pump Envelope Detection Switch", + AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETCPEN, + 1, 0), + SOC_ENUM("Charge Pump Envelope Detection Decay Time", + soc_enum_envdettime), + + /* Headset */ + SOC_ENUM("Headset Mode", soc_enum_da12voice), + SOC_SINGLE("Headset High Pass Switch", + AB8500_ANACONF1, AB8500_ANACONF1_HSHPEN, + 1, 0), + SOC_SINGLE("Headset Low Power Switch", + AB8500_ANACONF1, AB8500_ANACONF1_HSLOWPOW, + 1, 0), + SOC_SINGLE("Headset DAC Low Power Switch", + AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW1, + 1, 0), + SOC_SINGLE("Headset DAC Drv Low Power Switch", + AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW0, + 1, 0), + SOC_ENUM("Headset Fade Speed", soc_enum_hsfadspeed), + SOC_ENUM("Headset Source", soc_enum_da2hslr), + SOC_ENUM("Headset Filter", soc_enum_hsesinc), + SOC_DOUBLE_R_TLV("Headset Master Volume", + AB8500_DADIGGAIN1, AB8500_DADIGGAIN2, + 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv), + SOC_DOUBLE_R_TLV("Headset Digital Volume", + AB8500_HSLEARDIGGAIN, AB8500_HSRDIGGAIN, + 0, AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX, 1, hs_ear_dig_gain_tlv), + SOC_DOUBLE_TLV("Headset Volume", + AB8500_ANAGAIN3, + AB8500_ANAGAIN3_HSLGAIN, AB8500_ANAGAIN3_HSRGAIN, + AB8500_ANAGAIN3_HSXGAIN_MAX, 1, hs_gain_tlv), + + /* Earpiece */ + SOC_ENUM("Earpiece DAC Mode", + soc_enum_eardaclowpow), + SOC_ENUM("Earpiece DAC Drv Mode", + soc_enum_eardrvlowpow), + + /* HandsFree */ + SOC_ENUM("HF Mode", soc_enum_da34voice), + SOC_SINGLE("HF and Headset Swap Switch", + AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_SWAPDA12_34, + 1, 0), + SOC_DOUBLE("HF Low EMI Mode Switch", + AB8500_CLASSDCONF1, + AB8500_CLASSDCONF1_HFLSWAPEN, AB8500_CLASSDCONF1_HFRSWAPEN, + 1, 0), + SOC_DOUBLE("HF FIR Bypass Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_FIRBYP0, AB8500_CLASSDCONF2_FIRBYP1, + 1, 0), + SOC_DOUBLE("HF High Volume Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_HIGHVOLEN0, AB8500_CLASSDCONF2_HIGHVOLEN1, + 1, 0), + SOC_SINGLE("HF L and R Bridge Switch", + AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLHF, + 1, 0), + SOC_DOUBLE_R_TLV("HF Master Volume", + AB8500_DADIGGAIN3, AB8500_DADIGGAIN4, + 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv), + + /* Vibra */ + SOC_DOUBLE("Vibra High Volume Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_HIGHVOLEN2, AB8500_CLASSDCONF2_HIGHVOLEN3, + 1, 0), + SOC_DOUBLE("Vibra Low EMI Mode Switch", + AB8500_CLASSDCONF1, + AB8500_CLASSDCONF1_VIB1SWAPEN, AB8500_CLASSDCONF1_VIB2SWAPEN, + 1, 0), + SOC_DOUBLE("Vibra FIR Bypass Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_FIRBYP2, AB8500_CLASSDCONF2_FIRBYP3, + 1, 0), + SOC_ENUM("Vibra Mode", soc_enum_da56voice), + SOC_DOUBLE_R("Vibra PWM Duty Cycle N", + AB8500_PWMGENCONF3, AB8500_PWMGENCONF5, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0), + SOC_DOUBLE_R("Vibra PWM Duty Cycle P", + AB8500_PWMGENCONF2, AB8500_PWMGENCONF4, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0), + SOC_SINGLE("Vibra 1 and 2 Bridge Switch", + AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLVIB, + 1, 0), + SOC_DOUBLE_R_TLV("Vibra Master Volume", + AB8500_DADIGGAIN5, AB8500_DADIGGAIN6, + 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv), + + /* HandsFree, Vibra */ + SOC_SINGLE("ClassD High Pass Volume", + AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHHPGAIN, + AB8500_CLASSDCONF3_DITHHPGAIN_MAX, 0), + SOC_SINGLE("ClassD White Volume", + AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHWGAIN, + AB8500_CLASSDCONF3_DITHWGAIN_MAX, 0), + + /* Mic 1, Mic 2, LineIn */ + SOC_DOUBLE_R_TLV("Mic Master Volume", + AB8500_ADDIGGAIN3, AB8500_ADDIGGAIN4, + 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv), + + /* Mic 1 */ + SOC_SINGLE_TLV("Mic 1", + AB8500_ANAGAIN1, + AB8500_ANAGAINX_MICXGAIN, + AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv), + SOC_SINGLE("Mic 1 Low Power Switch", + AB8500_ANAGAIN1, AB8500_ANAGAINX_LOWPOWMICX, + 1, 0), + + /* Mic 2 */ + SOC_DOUBLE("Mic High Pass Switch", + AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD3NH, AB8500_ADFILTCONF_AD4NH, + 1, 1), + SOC_ENUM("Mic Mode", soc_enum_ad34voice), + SOC_ENUM("Mic Filter", soc_enum_dmic34sinc), + SOC_SINGLE_TLV("Mic 2", + AB8500_ANAGAIN2, + AB8500_ANAGAINX_MICXGAIN, + AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv), + SOC_SINGLE("Mic 2 Low Power Switch", + AB8500_ANAGAIN2, AB8500_ANAGAINX_LOWPOWMICX, + 1, 0), + + /* LineIn */ + SOC_DOUBLE("LineIn High Pass Switch", + AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD1NH, AB8500_ADFILTCONF_AD2NH, + 1, 1), + SOC_ENUM("LineIn Filter", soc_enum_dmic12sinc), + SOC_ENUM("LineIn Mode", soc_enum_ad12voice), + SOC_DOUBLE_R_TLV("LineIn Master Volume", + AB8500_ADDIGGAIN1, AB8500_ADDIGGAIN2, + 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv), + SOC_DOUBLE_TLV("LineIn", + AB8500_ANAGAIN4, + AB8500_ANAGAIN4_LINLGAIN, AB8500_ANAGAIN4_LINRGAIN, + AB8500_ANAGAIN4_LINXGAIN_MAX, 0, lin_gain_tlv), + SOC_DOUBLE_R_TLV("LineIn to Headset Volume", + AB8500_DIGLINHSLGAIN, AB8500_DIGLINHSRGAIN, + AB8500_DIGLINHSXGAIN_LINTOHSXGAIN, + AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX, + 1, lin2hs_gain_tlv), + + /* DMic */ + SOC_ENUM("DMic Filter", soc_enum_dmic56sinc), + SOC_DOUBLE_R_TLV("DMic Master Volume", + AB8500_ADDIGGAIN5, AB8500_ADDIGGAIN6, + 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv), + + /* Digital gains */ + SOC_ENUM("Digital Gain Fade Speed", soc_enum_fadespeed), + + /* Analog loopback */ + SOC_DOUBLE_R_TLV("Analog Loopback Volume", + AB8500_ADDIGLOOPGAIN1, AB8500_ADDIGLOOPGAIN2, + 0, AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX, 1, dax_dig_gain_tlv), + + /* Digital interface - DA from slot mapping */ + SOC_ENUM("Digital Interface DA 1 From Slot Map", soc_enum_da1slotmap), + SOC_ENUM("Digital Interface DA 2 From Slot Map", soc_enum_da2slotmap), + SOC_ENUM("Digital Interface DA 3 From Slot Map", soc_enum_da3slotmap), + SOC_ENUM("Digital Interface DA 4 From Slot Map", soc_enum_da4slotmap), + SOC_ENUM("Digital Interface DA 5 From Slot Map", soc_enum_da5slotmap), + SOC_ENUM("Digital Interface DA 6 From Slot Map", soc_enum_da6slotmap), + SOC_ENUM("Digital Interface DA 7 From Slot Map", soc_enum_da7slotmap), + SOC_ENUM("Digital Interface DA 8 From Slot Map", soc_enum_da8slotmap), + + /* Digital interface - AD to slot mapping */ + SOC_ENUM("Digital Interface AD To Slot 0 Map", soc_enum_adslot0map), + SOC_ENUM("Digital Interface AD To Slot 1 Map", soc_enum_adslot1map), + SOC_ENUM("Digital Interface AD To Slot 2 Map", soc_enum_adslot2map), + SOC_ENUM("Digital Interface AD To Slot 3 Map", soc_enum_adslot3map), + SOC_ENUM("Digital Interface AD To Slot 4 Map", soc_enum_adslot4map), + SOC_ENUM("Digital Interface AD To Slot 5 Map", soc_enum_adslot5map), + SOC_ENUM("Digital Interface AD To Slot 6 Map", soc_enum_adslot6map), + SOC_ENUM("Digital Interface AD To Slot 7 Map", soc_enum_adslot7map), + SOC_ENUM("Digital Interface AD To Slot 8 Map", soc_enum_adslot8map), + SOC_ENUM("Digital Interface AD To Slot 9 Map", soc_enum_adslot9map), + SOC_ENUM("Digital Interface AD To Slot 10 Map", soc_enum_adslot10map), + SOC_ENUM("Digital Interface AD To Slot 11 Map", soc_enum_adslot11map), + SOC_ENUM("Digital Interface AD To Slot 12 Map", soc_enum_adslot12map), + SOC_ENUM("Digital Interface AD To Slot 13 Map", soc_enum_adslot13map), + SOC_ENUM("Digital Interface AD To Slot 14 Map", soc_enum_adslot14map), + SOC_ENUM("Digital Interface AD To Slot 15 Map", soc_enum_adslot15map), + SOC_ENUM("Digital Interface AD To Slot 16 Map", soc_enum_adslot16map), + SOC_ENUM("Digital Interface AD To Slot 17 Map", soc_enum_adslot17map), + SOC_ENUM("Digital Interface AD To Slot 18 Map", soc_enum_adslot18map), + SOC_ENUM("Digital Interface AD To Slot 19 Map", soc_enum_adslot19map), + SOC_ENUM("Digital Interface AD To Slot 20 Map", soc_enum_adslot20map), + SOC_ENUM("Digital Interface AD To Slot 21 Map", soc_enum_adslot21map), + SOC_ENUM("Digital Interface AD To Slot 22 Map", soc_enum_adslot22map), + SOC_ENUM("Digital Interface AD To Slot 23 Map", soc_enum_adslot23map), + SOC_ENUM("Digital Interface AD To Slot 24 Map", soc_enum_adslot24map), + SOC_ENUM("Digital Interface AD To Slot 25 Map", soc_enum_adslot25map), + SOC_ENUM("Digital Interface AD To Slot 26 Map", soc_enum_adslot26map), + SOC_ENUM("Digital Interface AD To Slot 27 Map", soc_enum_adslot27map), + SOC_ENUM("Digital Interface AD To Slot 28 Map", soc_enum_adslot28map), + SOC_ENUM("Digital Interface AD To Slot 29 Map", soc_enum_adslot29map), + SOC_ENUM("Digital Interface AD To Slot 30 Map", soc_enum_adslot30map), + SOC_ENUM("Digital Interface AD To Slot 31 Map", soc_enum_adslot31map), + + /* Digital interface - Loopback */ + SOC_SINGLE("Digital Interface AD 1 Loopback Switch", + AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DAI7TOADO1, + 1, 0), + SOC_SINGLE("Digital Interface AD 2 Loopback Switch", + AB8500_DASLOTCONF2, AB8500_DASLOTCONF2_DAI8TOADO2, + 1, 0), + SOC_SINGLE("Digital Interface AD 3 Loopback Switch", + AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DAI7TOADO3, + 1, 0), + SOC_SINGLE("Digital Interface AD 4 Loopback Switch", + AB8500_DASLOTCONF4, AB8500_DASLOTCONF4_DAI8TOADO4, + 1, 0), + SOC_SINGLE("Digital Interface AD 5 Loopback Switch", + AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DAI7TOADO5, + 1, 0), + SOC_SINGLE("Digital Interface AD 6 Loopback Switch", + AB8500_DASLOTCONF6, AB8500_DASLOTCONF6_DAI8TOADO6, + 1, 0), + SOC_SINGLE("Digital Interface AD 7 Loopback Switch", + AB8500_DASLOTCONF7, AB8500_DASLOTCONF7_DAI8TOADO7, + 1, 0), + SOC_SINGLE("Digital Interface AD 8 Loopback Switch", + AB8500_DASLOTCONF8, AB8500_DASLOTCONF8_DAI7TOADO8, + 1, 0), + + /* Digital interface - Burst FIFO */ + SOC_SINGLE("Digital Interface 0 FIFO Enable Switch", + AB8500_DIGIFCONF3, AB8500_DIGIFCONF3_IF0BFIFOEN, + 1, 0), + SOC_ENUM("Burst FIFO Mask", soc_enum_bfifomask), + SOC_ENUM("Burst FIFO Bit-clock Frequency", soc_enum_bfifo19m2), + SOC_SINGLE("Burst FIFO Threshold", + AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOINT_SHIFT, + AB8500_FIFOCONF1_BFIFOINT_MAX, 0), + SOC_SINGLE("Burst FIFO Length", + AB8500_FIFOCONF2, AB8500_FIFOCONF2_BFIFOTX_SHIFT, + AB8500_FIFOCONF2_BFIFOTX_MAX, 0), + SOC_SINGLE("Burst FIFO EOS Extra Slots", + AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOEXSL_SHIFT, + AB8500_FIFOCONF3_BFIFOEXSL_MAX, 0), + SOC_SINGLE("Burst FIFO FS Extra Bit-clocks", + AB8500_FIFOCONF3, AB8500_FIFOCONF3_PREBITCLK0_SHIFT, + AB8500_FIFOCONF3_PREBITCLK0_MAX, 0), + SOC_ENUM("Burst FIFO Interface Mode", soc_enum_bfifomast), + + SOC_SINGLE("Burst FIFO Interface Switch", + AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFORUN_SHIFT, + 1, 0), + SOC_SINGLE("Burst FIFO Switch Frame Number", + AB8500_FIFOCONF4, AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT, + AB8500_FIFOCONF4_BFIFOFRAMSW_MAX, 0), + SOC_SINGLE("Burst FIFO Wake Up Delay", + AB8500_FIFOCONF5, AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT, + AB8500_FIFOCONF5_BFIFOWAKEUP_MAX, 0), + SOC_SINGLE("Burst FIFO Samples In FIFO", + AB8500_FIFOCONF6, AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT, + AB8500_FIFOCONF6_BFIFOSAMPLE_MAX, 0), + + /* ANC */ + SOC_ENUM_EXT("ANC Status", soc_enum_ancstate, + anc_status_control_get, anc_status_control_put), + SOC_SINGLE_XR_SX("ANC Warp Delay Shift", + AB8500_ANCCONF2, 1, AB8500_ANCCONF2_SHIFT, + AB8500_ANCCONF2_MIN, AB8500_ANCCONF2_MAX, 0), + SOC_SINGLE_XR_SX("ANC FIR Output Shift", + AB8500_ANCCONF3, 1, AB8500_ANCCONF3_SHIFT, + AB8500_ANCCONF3_MIN, AB8500_ANCCONF3_MAX, 0), + SOC_SINGLE_XR_SX("ANC IIR Output Shift", + AB8500_ANCCONF4, 1, AB8500_ANCCONF4_SHIFT, + AB8500_ANCCONF4_MIN, AB8500_ANCCONF4_MAX, 0), + SOC_SINGLE_XR_SX("ANC Warp Delay", + AB8500_ANCCONF9, 2, AB8500_ANC_WARP_DELAY_SHIFT, + AB8500_ANC_WARP_DELAY_MIN, AB8500_ANC_WARP_DELAY_MAX, 0), + + /* Sidetone */ + SOC_ENUM_EXT("Sidetone Status", soc_enum_sidstate, + sid_status_control_get, sid_status_control_put), + SOC_SINGLE_STROBE("Sidetone Reset", + AB8500_SIDFIRADR, AB8500_SIDFIRADR_FIRSIDSET, 0), +}; + +static struct snd_kcontrol_new ab8500_filter_controls[] = { + AB8500_FILTER_CONTROL("ANC FIR Coefficients", AB8500_ANC_FIR_COEFFS, + AB8500_ANC_FIR_COEFF_MIN, AB8500_ANC_FIR_COEFF_MAX), + AB8500_FILTER_CONTROL("ANC IIR Coefficients", AB8500_ANC_IIR_COEFFS, + AB8500_ANC_IIR_COEFF_MIN, AB8500_ANC_IIR_COEFF_MAX), + AB8500_FILTER_CONTROL("Sidetone FIR Coefficients", + AB8500_SID_FIR_COEFFS, AB8500_SID_FIR_COEFF_MIN, + AB8500_SID_FIR_COEFF_MAX) +}; +enum ab8500_filter { + AB8500_FILTER_ANC_FIR = 0, + AB8500_FILTER_ANC_IIR = 1, + AB8500_FILTER_SID_FIR = 2, +}; + +/* + * Extended interface for codec-driver + */ + +static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec) +{ + int status; + + dev_dbg(codec->dev, "%s: Enter.\n", __func__); + + /* Reset audio-registers and disable 32kHz-clock output 2 */ + status = ab8500_sysctrl_write(AB8500_STW4500CTRL3, + AB8500_STW4500CTRL3_CLK32KOUT2DIS | + AB8500_STW4500CTRL3_RESETAUDN, + AB8500_STW4500CTRL3_RESETAUDN); + if (status < 0) + return status; + + return 0; +} + +static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, + struct amic_settings *amics) +{ + u8 value8; + unsigned int value; + int status; + const struct snd_soc_dapm_route *route; + + dev_dbg(codec->dev, "%s: Enter.\n", __func__); + + /* Set DMic-clocks to outputs */ + status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC, + (u8)AB8500_GPIO_DIR4_REG, + &value8); + if (status < 0) + return status; + value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | + GPIO31_DIR_OUTPUT; + status = abx500_set_register_interruptible(codec->dev, + (u8)AB8500_MISC, + (u8)AB8500_GPIO_DIR4_REG, + value); + if (status < 0) + return status; + + /* Attach regulators to AMic DAPM-paths */ + dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__, + amic_micbias_str(amics->mic1a_micbias)); + route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias]; + status = snd_soc_dapm_add_routes(&codec->dapm, route, 1); + dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__, + amic_micbias_str(amics->mic1b_micbias)); + route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias]; + status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1); + dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__, + amic_micbias_str(amics->mic2_micbias)); + route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias]; + status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1); + if (status < 0) { + dev_err(codec->dev, + "%s: Failed to add AMic-regulator DAPM-routes (%d).\n", + __func__, status); + return status; + } + + /* Set AMic-configuration */ + dev_dbg(codec->dev, "%s: Mic 1 mic-type: %s\n", __func__, + amic_type_str(amics->mic1_type)); + snd_soc_update_bits(codec, AB8500_ANAGAIN1, AB8500_ANAGAINX_ENSEMICX, + amics->mic1_type == AMIC_TYPE_DIFFERENTIAL ? + 0 : AB8500_ANAGAINX_ENSEMICX); + dev_dbg(codec->dev, "%s: Mic 2 mic-type: %s\n", __func__, + amic_type_str(amics->mic2_type)); + snd_soc_update_bits(codec, AB8500_ANAGAIN2, AB8500_ANAGAINX_ENSEMICX, + amics->mic2_type == AMIC_TYPE_DIFFERENTIAL ? + 0 : AB8500_ANAGAINX_ENSEMICX); + + return 0; +} +EXPORT_SYMBOL_GPL(ab8500_audio_setup_mics); + +static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec, + enum ear_cm_voltage ear_cmv) +{ + char *cmv_str; + + switch (ear_cmv) { + case EAR_CMV_0_95V: + cmv_str = "0.95V"; + break; + case EAR_CMV_1_10V: + cmv_str = "1.10V"; + break; + case EAR_CMV_1_27V: + cmv_str = "1.27V"; + break; + case EAR_CMV_1_58V: + cmv_str = "1.58V"; + break; + default: + dev_err(codec->dev, + "%s: Unknown earpiece CM-voltage (%d)!\n", + __func__, (int)ear_cmv); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: Earpiece CM-voltage: %s\n", __func__, + cmv_str); + snd_soc_update_bits(codec, AB8500_ANACONF1, AB8500_ANACONF1_EARSELCM, + ear_cmv); + + return 0; +} +EXPORT_SYMBOL_GPL(ab8500_audio_set_ear_cmv); + +static int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai, + unsigned int delay) +{ + unsigned int mask, val; + struct snd_soc_codec *codec = dai->codec; + + mask = BIT(AB8500_DIGIFCONF2_IF0DEL); + val = 0; + + switch (delay) { + case 0: + break; + case 1: + val |= BIT(AB8500_DIGIFCONF2_IF0DEL); + break; + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported bit-delay (0x%x)!\n", + __func__, delay); + return -EINVAL; + } + + dev_dbg(dai->codec->dev, "%s: IF0 Bit-delay: %d bits.\n", + __func__, delay); + snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val); + + return 0; +} + +/* Gates clocking according format mask */ +static int ab8500_codec_set_dai_clock_gate(struct snd_soc_codec *codec, + unsigned int fmt) +{ + unsigned int mask; + unsigned int val; + + mask = BIT(AB8500_DIGIFCONF1_ENMASTGEN) | + BIT(AB8500_DIGIFCONF1_ENFSBITCLK0); + + val = BIT(AB8500_DIGIFCONF1_ENMASTGEN); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: /* continuous clock */ + dev_dbg(codec->dev, "%s: IF0 Clock is continuous.\n", + __func__); + val |= BIT(AB8500_DIGIFCONF1_ENFSBITCLK0); + break; + case SND_SOC_DAIFMT_GATED: /* clock is gated */ + dev_dbg(codec->dev, "%s: IF0 Clock is gated.\n", + __func__); + break; + default: + dev_err(codec->dev, + "%s: ERROR: Unsupported clock mask (0x%x)!\n", + __func__, fmt & SND_SOC_DAIFMT_CLOCK_MASK); + return -EINVAL; + } + + snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val); + + return 0; +} + +static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + unsigned int mask; + unsigned int val; + struct snd_soc_codec *codec = dai->codec; + int status; + + dev_dbg(codec->dev, "%s: Enter (fmt = 0x%x)\n", __func__, fmt); + + mask = BIT(AB8500_DIGIFCONF3_IF1DATOIF0AD) | + BIT(AB8500_DIGIFCONF3_IF1CLKTOIF0CLK) | + BIT(AB8500_DIGIFCONF3_IF0BFIFOEN) | + BIT(AB8500_DIGIFCONF3_IF0MASTER); + val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & FRM master */ + dev_dbg(dai->codec->dev, + "%s: IF0 Master-mode: AB8500 master.\n", __func__); + val |= BIT(AB8500_DIGIFCONF3_IF0MASTER); + break; + case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & FRM slave */ + dev_dbg(dai->codec->dev, + "%s: IF0 Master-mode: AB8500 slave.\n", __func__); + break; + case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & FRM master */ + case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + dev_err(dai->codec->dev, + "%s: ERROR: The device is either a master or a slave.\n", + __func__); + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupporter master mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + break; + } + + snd_soc_update_bits(codec, AB8500_DIGIFCONF3, mask, val); + + /* Set clock gating */ + status = ab8500_codec_set_dai_clock_gate(codec, fmt); + if (status) { + dev_err(dai->codec->dev, + "%s: ERRROR: Failed to set clock gate (%d).\n", + __func__, status); + return status; + } + + /* Setting data transfer format */ + + mask = BIT(AB8500_DIGIFCONF2_IF0FORMAT0) | + BIT(AB8500_DIGIFCONF2_IF0FORMAT1) | + BIT(AB8500_DIGIFCONF2_FSYNC0P) | + BIT(AB8500_DIGIFCONF2_BITCLK0P); + val = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: /* I2S mode */ + dev_dbg(dai->codec->dev, "%s: IF0 Protocol: I2S\n", __func__); + val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT1); + ab8500_audio_set_bit_delay(dai, 0); + break; + + case SND_SOC_DAIFMT_DSP_A: /* L data MSB after FRM LRC */ + dev_dbg(dai->codec->dev, + "%s: IF0 Protocol: DSP A (TDM)\n", __func__); + val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0); + ab8500_audio_set_bit_delay(dai, 1); + break; + + case SND_SOC_DAIFMT_DSP_B: /* L data MSB during FRM LRC */ + dev_dbg(dai->codec->dev, + "%s: IF0 Protocol: DSP B (TDM)\n", __func__); + val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0); + ab8500_audio_set_bit_delay(dai, 0); + break; + + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported format (0x%x)!\n", + __func__, fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */ + dev_dbg(dai->codec->dev, + "%s: IF0: Normal bit clock, normal frame\n", + __func__); + break; + case SND_SOC_DAIFMT_NB_IF: /* normal BCLK + inv FRM */ + dev_dbg(dai->codec->dev, + "%s: IF0: Normal bit clock, inverted frame\n", + __func__); + val |= BIT(AB8500_DIGIFCONF2_FSYNC0P); + break; + case SND_SOC_DAIFMT_IB_NF: /* invert BCLK + nor FRM */ + dev_dbg(dai->codec->dev, + "%s: IF0: Inverted bit clock, normal frame\n", + __func__); + val |= BIT(AB8500_DIGIFCONF2_BITCLK0P); + break; + case SND_SOC_DAIFMT_IB_IF: /* invert BCLK + FRM */ + dev_dbg(dai->codec->dev, + "%s: IF0: Inverted bit clock, inverted frame\n", + __func__); + val |= BIT(AB8500_DIGIFCONF2_FSYNC0P); + val |= BIT(AB8500_DIGIFCONF2_BITCLK0P); + break; + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported INV mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val); + + return 0; +} + +static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val, mask, slots_active; + + mask = BIT(AB8500_DIGIFCONF2_IF0WL0) | + BIT(AB8500_DIGIFCONF2_IF0WL1); + val = 0; + + switch (slot_width) { + case 16: + break; + case 20: + val |= BIT(AB8500_DIGIFCONF2_IF0WL0); + break; + case 24: + val |= BIT(AB8500_DIGIFCONF2_IF0WL1); + break; + case 32: + val |= BIT(AB8500_DIGIFCONF2_IF0WL1) | + BIT(AB8500_DIGIFCONF2_IF0WL0); + break; + default: + dev_err(dai->codec->dev, "%s: Unsupported slot-width 0x%x\n", + __func__, slot_width); + return -EINVAL; + } + + dev_dbg(dai->codec->dev, "%s: IF0 slot-width: %d bits.\n", + __func__, slot_width); + snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val); + + /* Setup TDM clocking according to slot count */ + dev_dbg(dai->codec->dev, "%s: Slots, total: %d\n", __func__, slots); + mask = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) | + BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1); + switch (slots) { + case 2: + val = AB8500_MASK_NONE; + break; + case 4: + val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0); + break; + case 8: + val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1); + break; + case 16: + val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) | + BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1); + break; + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported number of slots (%d)!\n", + __func__, slots); + return -EINVAL; + } + snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val); + + /* Setup TDM DA according to active tx slots */ + mask = AB8500_DASLOTCONFX_SLTODAX_MASK; + slots_active = hweight32(tx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, TX: %d\n", __func__, + slots_active); + switch (slots_active) { + case 0: + break; + case 1: + /* Slot 9 -> DA_IN1 & DA_IN3 */ + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); + break; + case 2: + /* Slot 9 -> DA_IN1 & DA_IN3, Slot 11 -> DA_IN2 & DA_IN4 */ + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 9); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 9); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); + + break; + case 8: + dev_dbg(dai->codec->dev, + "%s: In 8-channel mode DA-from-slot mapping is set manually.", + __func__); + break; + default: + dev_err(dai->codec->dev, + "%s: Unsupported number of active TX-slots (%d)!\n", + __func__, slots_active); + return -EINVAL; + } + + /* Setup TDM AD according to active RX-slots */ + slots_active = hweight32(rx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, RX: %d\n", __func__, + slots_active); + switch (slots_active) { + case 0: + break; + case 1: + /* AD_OUT3 -> slot 0 & 1 */ + snd_soc_update_bits(codec, AB8500_ADSLOTSEL1, AB8500_MASK_ALL, + AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | + AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD); + break; + case 2: + /* AD_OUT3 -> slot 0, AD_OUT2 -> slot 1 */ + snd_soc_update_bits(codec, + AB8500_ADSLOTSEL1, + AB8500_MASK_ALL, + AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | + AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD); + break; + case 8: + dev_dbg(dai->codec->dev, + "%s: In 8-channel mode AD-to-slot mapping is set manually.", + __func__); + break; + default: + dev_err(dai->codec->dev, + "%s: Unsupported number of active RX-slots (%d)!\n", + __func__, slots_active); + return -EINVAL; + } + + return 0; +} + +struct snd_soc_dai_driver ab8500_codec_dai[] = { + { + .name = "ab8500-codec-dai.0", + .id = 0, + .playback = { + .stream_name = "ab8500_0p", + .channels_min = 1, + .channels_max = 8, + .rates = AB8500_SUPPORTED_RATE, + .formats = AB8500_SUPPORTED_FMT, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, + .set_fmt = ab8500_codec_set_dai_fmt, + } + }, + .symmetric_rates = 1 + }, + { + .name = "ab8500-codec-dai.1", + .id = 1, + .capture = { + .stream_name = "ab8500_0c", + .channels_min = 1, + .channels_max = 8, + .rates = AB8500_SUPPORTED_RATE, + .formats = AB8500_SUPPORTED_FMT, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, + .set_fmt = ab8500_codec_set_dai_fmt, + } + }, + .symmetric_rates = 1 + } +}; + +static int ab8500_codec_probe(struct snd_soc_codec *codec) +{ + struct device *dev = codec->dev; + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); + struct ab8500_platform_data *pdata; + struct filter_control *fc; + int status; + + dev_dbg(dev, "%s: Enter.\n", __func__); + + /* Setup AB8500 according to board-settings */ + pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); + if (status < 0) { + pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); + return status; + } + status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv); + if (status < 0) { + pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n", + __func__, status); + return status; + } + + status = ab8500_audio_init_audioblock(codec); + if (status < 0) { + dev_err(dev, "%s: failed to init audio-block (%d)!\n", + __func__, status); + return status; + } + + /* Override HW-defaults */ + ab8500_codec_write_reg(codec, + AB8500_ANACONF5, + BIT(AB8500_ANACONF5_HSAUTOEN)); + ab8500_codec_write_reg(codec, + AB8500_SHORTCIRCONF, + BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); + + /* Add filter controls */ + status = snd_soc_add_codec_controls(codec, ab8500_filter_controls, + ARRAY_SIZE(ab8500_filter_controls)); + if (status < 0) { + dev_err(dev, + "%s: failed to add ab8500 filter controls (%d).\n", + __func__, status); + return status; + } + fc = (struct filter_control *) + &ab8500_filter_controls[AB8500_FILTER_ANC_FIR].private_value; + drvdata->anc_fir_values = (long *)fc->value; + fc = (struct filter_control *) + &ab8500_filter_controls[AB8500_FILTER_ANC_IIR].private_value; + drvdata->anc_iir_values = (long *)fc->value; + fc = (struct filter_control *) + &ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value; + drvdata->sid_fir_values = (long *)fc->value; + + (void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input"); + + mutex_init(&drvdata->anc_lock); + + return status; +} + +static struct snd_soc_codec_driver ab8500_codec_driver = { + .probe = ab8500_codec_probe, + .read = ab8500_codec_read_reg, + .write = ab8500_codec_write_reg, + .reg_word_size = sizeof(u8), + .controls = ab8500_ctrls, + .num_controls = ARRAY_SIZE(ab8500_ctrls), + .dapm_widgets = ab8500_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ab8500_dapm_widgets), + .dapm_routes = ab8500_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ab8500_dapm_routes), +}; + +static int __devinit ab8500_codec_driver_probe(struct platform_device *pdev) +{ + int status; + struct ab8500_codec_drvdata *drvdata; + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + /* Create driver private-data struct */ + drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ab8500_codec_drvdata), + GFP_KERNEL); + drvdata->sid_status = SID_UNCONFIGURED; + drvdata->anc_status = ANC_UNCONFIGURED; + dev_set_drvdata(&pdev->dev, drvdata); + + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); + status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, + ab8500_codec_dai, + ARRAY_SIZE(ab8500_codec_dai)); + if (status < 0) + dev_err(&pdev->dev, + "%s: Error: Failed to register codec (%d).\n", + __func__, status); + + return status; +} + +static int __devexit ab8500_codec_driver_remove(struct platform_device *pdev) +{ + dev_info(&pdev->dev, "%s Enter.\n", __func__); + + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver ab8500_codec_platform_driver = { + .driver = { + .name = "ab8500-codec", + .owner = THIS_MODULE, + }, + .probe = ab8500_codec_driver_probe, + .remove = __devexit_p(ab8500_codec_driver_remove), + .suspend = NULL, + .resume = NULL, +}; +module_platform_driver(ab8500_codec_platform_driver); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h new file mode 100644 index 000000000000..114f69a0c629 --- /dev/null +++ b/sound/soc/codecs/ab8500-codec.h @@ -0,0 +1,590 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com>, + * for ST-Ericsson. + * + * Based on the early work done by: + * Mikko J. Lehto <mikko.lehto@symbio.com>, + * Mikko Sarmanne <mikko.sarmanne@symbio.com>, + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef AB8500_CODEC_REGISTERS_H +#define AB8500_CODEC_REGISTERS_H + +#define AB8500_SUPPORTED_RATE (SNDRV_PCM_RATE_48000) +#define AB8500_SUPPORTED_FMT (SNDRV_PCM_FMTBIT_S16_LE) + +/* AB8500 audio bank (0x0d) register definitions */ + +#define AB8500_POWERUP 0x00 +#define AB8500_AUDSWRESET 0x01 +#define AB8500_ADPATHENA 0x02 +#define AB8500_DAPATHENA 0x03 +#define AB8500_ANACONF1 0x04 +#define AB8500_ANACONF2 0x05 +#define AB8500_DIGMICCONF 0x06 +#define AB8500_ANACONF3 0x07 +#define AB8500_ANACONF4 0x08 +#define AB8500_DAPATHCONF 0x09 +#define AB8500_MUTECONF 0x0A +#define AB8500_SHORTCIRCONF 0x0B +#define AB8500_ANACONF5 0x0C +#define AB8500_ENVCPCONF 0x0D +#define AB8500_SIGENVCONF 0x0E +#define AB8500_PWMGENCONF1 0x0F +#define AB8500_PWMGENCONF2 0x10 +#define AB8500_PWMGENCONF3 0x11 +#define AB8500_PWMGENCONF4 0x12 +#define AB8500_PWMGENCONF5 0x13 +#define AB8500_ANAGAIN1 0x14 +#define AB8500_ANAGAIN2 0x15 +#define AB8500_ANAGAIN3 0x16 +#define AB8500_ANAGAIN4 0x17 +#define AB8500_DIGLINHSLGAIN 0x18 +#define AB8500_DIGLINHSRGAIN 0x19 +#define AB8500_ADFILTCONF 0x1A +#define AB8500_DIGIFCONF1 0x1B +#define AB8500_DIGIFCONF2 0x1C +#define AB8500_DIGIFCONF3 0x1D +#define AB8500_DIGIFCONF4 0x1E +#define AB8500_ADSLOTSEL1 0x1F +#define AB8500_ADSLOTSEL2 0x20 +#define AB8500_ADSLOTSEL3 0x21 +#define AB8500_ADSLOTSEL4 0x22 +#define AB8500_ADSLOTSEL5 0x23 +#define AB8500_ADSLOTSEL6 0x24 +#define AB8500_ADSLOTSEL7 0x25 +#define AB8500_ADSLOTSEL8 0x26 +#define AB8500_ADSLOTSEL9 0x27 +#define AB8500_ADSLOTSEL10 0x28 +#define AB8500_ADSLOTSEL11 0x29 +#define AB8500_ADSLOTSEL12 0x2A +#define AB8500_ADSLOTSEL13 0x2B +#define AB8500_ADSLOTSEL14 0x2C +#define AB8500_ADSLOTSEL15 0x2D +#define AB8500_ADSLOTSEL16 0x2E +#define AB8500_ADSLOTHIZCTRL1 0x2F +#define AB8500_ADSLOTHIZCTRL2 0x30 +#define AB8500_ADSLOTHIZCTRL3 0x31 +#define AB8500_ADSLOTHIZCTRL4 0x32 +#define AB8500_DASLOTCONF1 0x33 +#define AB8500_DASLOTCONF2 0x34 +#define AB8500_DASLOTCONF3 0x35 +#define AB8500_DASLOTCONF4 0x36 +#define AB8500_DASLOTCONF5 0x37 +#define AB8500_DASLOTCONF6 0x38 +#define AB8500_DASLOTCONF7 0x39 +#define AB8500_DASLOTCONF8 0x3A +#define AB8500_CLASSDCONF1 0x3B +#define AB8500_CLASSDCONF2 0x3C +#define AB8500_CLASSDCONF3 0x3D +#define AB8500_DMICFILTCONF 0x3E +#define AB8500_DIGMULTCONF1 0x3F +#define AB8500_DIGMULTCONF2 0x40 +#define AB8500_ADDIGGAIN1 0x41 +#define AB8500_ADDIGGAIN2 0x42 +#define AB8500_ADDIGGAIN3 0x43 +#define AB8500_ADDIGGAIN4 0x44 +#define AB8500_ADDIGGAIN5 0x45 +#define AB8500_ADDIGGAIN6 0x46 +#define AB8500_DADIGGAIN1 0x47 +#define AB8500_DADIGGAIN2 0x48 +#define AB8500_DADIGGAIN3 0x49 +#define AB8500_DADIGGAIN4 0x4A +#define AB8500_DADIGGAIN5 0x4B +#define AB8500_DADIGGAIN6 0x4C +#define AB8500_ADDIGLOOPGAIN1 0x4D +#define AB8500_ADDIGLOOPGAIN2 0x4E +#define AB8500_HSLEARDIGGAIN 0x4F +#define AB8500_HSRDIGGAIN 0x50 +#define AB8500_SIDFIRGAIN1 0x51 +#define AB8500_SIDFIRGAIN2 0x52 +#define AB8500_ANCCONF1 0x53 +#define AB8500_ANCCONF2 0x54 +#define AB8500_ANCCONF3 0x55 +#define AB8500_ANCCONF4 0x56 +#define AB8500_ANCCONF5 0x57 +#define AB8500_ANCCONF6 0x58 +#define AB8500_ANCCONF7 0x59 +#define AB8500_ANCCONF8 0x5A +#define AB8500_ANCCONF9 0x5B +#define AB8500_ANCCONF10 0x5C +#define AB8500_ANCCONF11 0x5D +#define AB8500_ANCCONF12 0x5E +#define AB8500_ANCCONF13 0x5F +#define AB8500_ANCCONF14 0x60 +#define AB8500_SIDFIRADR 0x61 +#define AB8500_SIDFIRCOEF1 0x62 +#define AB8500_SIDFIRCOEF2 0x63 +#define AB8500_SIDFIRCONF 0x64 +#define AB8500_AUDINTMASK1 0x65 +#define AB8500_AUDINTSOURCE1 0x66 +#define AB8500_AUDINTMASK2 0x67 +#define AB8500_AUDINTSOURCE2 0x68 +#define AB8500_FIFOCONF1 0x69 +#define AB8500_FIFOCONF2 0x6A +#define AB8500_FIFOCONF3 0x6B +#define AB8500_FIFOCONF4 0x6C +#define AB8500_FIFOCONF5 0x6D +#define AB8500_FIFOCONF6 0x6E +#define AB8500_AUDREV 0x6F + +#define AB8500_FIRST_REG AB8500_POWERUP +#define AB8500_LAST_REG AB8500_AUDREV +#define AB8500_CACHEREGNUM (AB8500_LAST_REG + 1) + +#define AB8500_MASK_ALL 0xFF +#define AB8500_MASK_NONE 0x00 + +/* AB8500_POWERUP */ +#define AB8500_POWERUP_POWERUP 7 +#define AB8500_POWERUP_ENANA 3 + +/* AB8500_AUDSWRESET */ +#define AB8500_AUDSWRESET_SWRESET 7 + +/* AB8500_ADPATHENA */ +#define AB8500_ADPATHENA_ENAD12 7 +#define AB8500_ADPATHENA_ENAD34 5 +#define AB8500_ADPATHENA_ENAD5768 3 + +/* AB8500_DAPATHENA */ +#define AB8500_DAPATHENA_ENDA1 7 +#define AB8500_DAPATHENA_ENDA2 6 +#define AB8500_DAPATHENA_ENDA3 5 +#define AB8500_DAPATHENA_ENDA4 4 +#define AB8500_DAPATHENA_ENDA5 3 +#define AB8500_DAPATHENA_ENDA6 2 + +/* AB8500_ANACONF1 */ +#define AB8500_ANACONF1_HSLOWPOW 7 +#define AB8500_ANACONF1_DACLOWPOW1 6 +#define AB8500_ANACONF1_DACLOWPOW0 5 +#define AB8500_ANACONF1_EARDACLOWPOW 4 +#define AB8500_ANACONF1_EARSELCM 2 +#define AB8500_ANACONF1_HSHPEN 1 +#define AB8500_ANACONF1_EARDRVLOWPOW 0 + +/* AB8500_ANACONF2 */ +#define AB8500_ANACONF2_ENMIC1 7 +#define AB8500_ANACONF2_ENMIC2 6 +#define AB8500_ANACONF2_ENLINL 5 +#define AB8500_ANACONF2_ENLINR 4 +#define AB8500_ANACONF2_MUTMIC1 3 +#define AB8500_ANACONF2_MUTMIC2 2 +#define AB8500_ANACONF2_MUTLINL 1 +#define AB8500_ANACONF2_MUTLINR 0 + +/* AB8500_DIGMICCONF */ +#define AB8500_DIGMICCONF_ENDMIC1 7 +#define AB8500_DIGMICCONF_ENDMIC2 6 +#define AB8500_DIGMICCONF_ENDMIC3 5 +#define AB8500_DIGMICCONF_ENDMIC4 4 +#define AB8500_DIGMICCONF_ENDMIC5 3 +#define AB8500_DIGMICCONF_ENDMIC6 2 +#define AB8500_DIGMICCONF_HSFADSPEED 0 + +/* AB8500_ANACONF3 */ +#define AB8500_ANACONF3_MIC1SEL 7 +#define AB8500_ANACONF3_LINRSEL 6 +#define AB8500_ANACONF3_ENDRVHSL 5 +#define AB8500_ANACONF3_ENDRVHSR 4 +#define AB8500_ANACONF3_ENADCMIC 2 +#define AB8500_ANACONF3_ENADCLINL 1 +#define AB8500_ANACONF3_ENADCLINR 0 + +/* AB8500_ANACONF4 */ +#define AB8500_ANACONF4_DISPDVSS 7 +#define AB8500_ANACONF4_ENEAR 6 +#define AB8500_ANACONF4_ENHSL 5 +#define AB8500_ANACONF4_ENHSR 4 +#define AB8500_ANACONF4_ENHFL 3 +#define AB8500_ANACONF4_ENHFR 2 +#define AB8500_ANACONF4_ENVIB1 1 +#define AB8500_ANACONF4_ENVIB2 0 + +/* AB8500_DAPATHCONF */ +#define AB8500_DAPATHCONF_ENDACEAR 6 +#define AB8500_DAPATHCONF_ENDACHSL 5 +#define AB8500_DAPATHCONF_ENDACHSR 4 +#define AB8500_DAPATHCONF_ENDACHFL 3 +#define AB8500_DAPATHCONF_ENDACHFR 2 +#define AB8500_DAPATHCONF_ENDACVIB1 1 +#define AB8500_DAPATHCONF_ENDACVIB2 0 + +/* AB8500_MUTECONF */ +#define AB8500_MUTECONF_MUTEAR 6 +#define AB8500_MUTECONF_MUTHSL 5 +#define AB8500_MUTECONF_MUTHSR 4 +#define AB8500_MUTECONF_MUTDACEAR 2 +#define AB8500_MUTECONF_MUTDACHSL 1 +#define AB8500_MUTECONF_MUTDACHSR 0 + +/* AB8500_SHORTCIRCONF */ +#define AB8500_SHORTCIRCONF_ENSHORTPWD 7 +#define AB8500_SHORTCIRCONF_EARSHORTDIS 6 +#define AB8500_SHORTCIRCONF_HSSHORTDIS 5 +#define AB8500_SHORTCIRCONF_HSPULLDEN 4 +#define AB8500_SHORTCIRCONF_HSOSCEN 2 +#define AB8500_SHORTCIRCONF_HSFADDIS 1 +#define AB8500_SHORTCIRCONF_HSZCDDIS 0 +/* Zero cross should be disabled */ + +/* AB8500_ANACONF5 */ +#define AB8500_ANACONF5_ENCPHS 7 +#define AB8500_ANACONF5_HSLDACTOLOL 5 +#define AB8500_ANACONF5_HSRDACTOLOR 4 +#define AB8500_ANACONF5_ENLOL 3 +#define AB8500_ANACONF5_ENLOR 2 +#define AB8500_ANACONF5_HSAUTOEN 0 + +/* AB8500_ENVCPCONF */ +#define AB8500_ENVCPCONF_ENVDETHTHRE 4 +#define AB8500_ENVCPCONF_ENVDETLTHRE 0 +#define AB8500_ENVCPCONF_ENVDETHTHRE_MAX 0x0F +#define AB8500_ENVCPCONF_ENVDETLTHRE_MAX 0x0F + +/* AB8500_SIGENVCONF */ +#define AB8500_SIGENVCONF_CPLVEN 5 +#define AB8500_SIGENVCONF_ENVDETCPEN 4 +#define AB8500_SIGENVCONF_ENVDETTIME 0 +#define AB8500_SIGENVCONF_ENVDETTIME_MAX 0x0F + +/* AB8500_PWMGENCONF1 */ +#define AB8500_PWMGENCONF1_PWMTOVIB1 7 +#define AB8500_PWMGENCONF1_PWMTOVIB2 6 +#define AB8500_PWMGENCONF1_PWM1CTRL 5 +#define AB8500_PWMGENCONF1_PWM2CTRL 4 +#define AB8500_PWMGENCONF1_PWM1NCTRL 3 +#define AB8500_PWMGENCONF1_PWM1PCTRL 2 +#define AB8500_PWMGENCONF1_PWM2NCTRL 1 +#define AB8500_PWMGENCONF1_PWM2PCTRL 0 + +/* AB8500_PWMGENCONF2 */ +/* AB8500_PWMGENCONF3 */ +/* AB8500_PWMGENCONF4 */ +/* AB8500_PWMGENCONF5 */ +#define AB8500_PWMGENCONFX_PWMVIBXPOL 7 +#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC 0 +#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX 0x64 + +/* AB8500_ANAGAIN1 */ +/* AB8500_ANAGAIN2 */ +#define AB8500_ANAGAINX_ENSEMICX 7 +#define AB8500_ANAGAINX_LOWPOWMICX 6 +#define AB8500_ANAGAINX_MICXGAIN 0 +#define AB8500_ANAGAINX_MICXGAIN_MAX 0x1F + +/* AB8500_ANAGAIN3 */ +#define AB8500_ANAGAIN3_HSLGAIN 4 +#define AB8500_ANAGAIN3_HSRGAIN 0 +#define AB8500_ANAGAIN3_HSXGAIN_MAX 0x0F + +/* AB8500_ANAGAIN4 */ +#define AB8500_ANAGAIN4_LINLGAIN 4 +#define AB8500_ANAGAIN4_LINRGAIN 0 +#define AB8500_ANAGAIN4_LINXGAIN_MAX 0x0F + +/* AB8500_DIGLINHSLGAIN */ +/* AB8500_DIGLINHSRGAIN */ +#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN 0 +#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX 0x13 + +/* AB8500_ADFILTCONF */ +#define AB8500_ADFILTCONF_AD1NH 7 +#define AB8500_ADFILTCONF_AD2NH 6 +#define AB8500_ADFILTCONF_AD3NH 5 +#define AB8500_ADFILTCONF_AD4NH 4 +#define AB8500_ADFILTCONF_AD1VOICE 3 +#define AB8500_ADFILTCONF_AD2VOICE 2 +#define AB8500_ADFILTCONF_AD3VOICE 1 +#define AB8500_ADFILTCONF_AD4VOICE 0 + +/* AB8500_DIGIFCONF1 */ +#define AB8500_DIGIFCONF1_ENMASTGEN 7 +#define AB8500_DIGIFCONF1_IF1BITCLKOS1 6 +#define AB8500_DIGIFCONF1_IF1BITCLKOS0 5 +#define AB8500_DIGIFCONF1_ENFSBITCLK1 4 +#define AB8500_DIGIFCONF1_IF0BITCLKOS1 2 +#define AB8500_DIGIFCONF1_IF0BITCLKOS0 1 +#define AB8500_DIGIFCONF1_ENFSBITCLK0 0 + +/* AB8500_DIGIFCONF2 */ +#define AB8500_DIGIFCONF2_FSYNC0P 6 +#define AB8500_DIGIFCONF2_BITCLK0P 5 +#define AB8500_DIGIFCONF2_IF0DEL 4 +#define AB8500_DIGIFCONF2_IF0FORMAT1 3 +#define AB8500_DIGIFCONF2_IF0FORMAT0 2 +#define AB8500_DIGIFCONF2_IF0WL1 1 +#define AB8500_DIGIFCONF2_IF0WL0 0 + +/* AB8500_DIGIFCONF3 */ +#define AB8500_DIGIFCONF3_IF0DATOIF1AD 7 +#define AB8500_DIGIFCONF3_IF0CLKTOIF1CLK 6 +#define AB8500_DIGIFCONF3_IF1MASTER 5 +#define AB8500_DIGIFCONF3_IF1DATOIF0AD 3 +#define AB8500_DIGIFCONF3_IF1CLKTOIF0CLK 2 +#define AB8500_DIGIFCONF3_IF0MASTER 1 +#define AB8500_DIGIFCONF3_IF0BFIFOEN 0 + +/* AB8500_DIGIFCONF4 */ +#define AB8500_DIGIFCONF4_FSYNC1P 6 +#define AB8500_DIGIFCONF4_BITCLK1P 5 +#define AB8500_DIGIFCONF4_IF1DEL 4 +#define AB8500_DIGIFCONF4_IF1FORMAT1 3 +#define AB8500_DIGIFCONF4_IF1FORMAT0 2 +#define AB8500_DIGIFCONF4_IF1WL1 1 +#define AB8500_DIGIFCONF4_IF1WL0 0 + +/* AB8500_ADSLOTSELX */ +#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00 +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x01 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x02 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x03 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x04 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x05 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x06 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x07 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x08 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0x0F +#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00 +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x10 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x20 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x30 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x40 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x50 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x60 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x70 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x80 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0xF0 +#define AB8500_ADSLOTSELX_EVEN_SHIFT 0 +#define AB8500_ADSLOTSELX_ODD_SHIFT 4 + +/* AB8500_ADSLOTHIZCTRL1 */ +/* AB8500_ADSLOTHIZCTRL2 */ +/* AB8500_ADSLOTHIZCTRL3 */ +/* AB8500_ADSLOTHIZCTRL4 */ +/* AB8500_DASLOTCONF1 */ +#define AB8500_DASLOTCONF1_DA12VOICE 7 +#define AB8500_DASLOTCONF1_SWAPDA12_34 6 +#define AB8500_DASLOTCONF1_DAI7TOADO1 5 + +/* AB8500_DASLOTCONF2 */ +#define AB8500_DASLOTCONF2_DAI8TOADO2 5 + +/* AB8500_DASLOTCONF3 */ +#define AB8500_DASLOTCONF3_DA34VOICE 7 +#define AB8500_DASLOTCONF3_DAI7TOADO3 5 + +/* AB8500_DASLOTCONF4 */ +#define AB8500_DASLOTCONF4_DAI8TOADO4 5 + +/* AB8500_DASLOTCONF5 */ +#define AB8500_DASLOTCONF5_DA56VOICE 7 +#define AB8500_DASLOTCONF5_DAI7TOADO5 5 + +/* AB8500_DASLOTCONF6 */ +#define AB8500_DASLOTCONF6_DAI8TOADO6 5 + +/* AB8500_DASLOTCONF7 */ +#define AB8500_DASLOTCONF7_DAI8TOADO7 5 + +/* AB8500_DASLOTCONF8 */ +#define AB8500_DASLOTCONF8_DAI7TOADO8 5 + +#define AB8500_DASLOTCONFX_SLTODAX_SHIFT 0 +#define AB8500_DASLOTCONFX_SLTODAX_MASK 0x1F + +/* AB8500_CLASSDCONF1 */ +#define AB8500_CLASSDCONF1_PARLHF 7 +#define AB8500_CLASSDCONF1_PARLVIB 6 +#define AB8500_CLASSDCONF1_VIB1SWAPEN 3 +#define AB8500_CLASSDCONF1_VIB2SWAPEN 2 +#define AB8500_CLASSDCONF1_HFLSWAPEN 1 +#define AB8500_CLASSDCONF1_HFRSWAPEN 0 + +/* AB8500_CLASSDCONF2 */ +#define AB8500_CLASSDCONF2_FIRBYP3 7 +#define AB8500_CLASSDCONF2_FIRBYP2 6 +#define AB8500_CLASSDCONF2_FIRBYP1 5 +#define AB8500_CLASSDCONF2_FIRBYP0 4 +#define AB8500_CLASSDCONF2_HIGHVOLEN3 3 +#define AB8500_CLASSDCONF2_HIGHVOLEN2 2 +#define AB8500_CLASSDCONF2_HIGHVOLEN1 1 +#define AB8500_CLASSDCONF2_HIGHVOLEN0 0 + +/* AB8500_CLASSDCONF3 */ +#define AB8500_CLASSDCONF3_DITHHPGAIN 4 +#define AB8500_CLASSDCONF3_DITHHPGAIN_MAX 0x0A +#define AB8500_CLASSDCONF3_DITHWGAIN 0 +#define AB8500_CLASSDCONF3_DITHWGAIN_MAX 0x0A + +/* AB8500_DMICFILTCONF */ +#define AB8500_DMICFILTCONF_ANCINSEL 7 +#define AB8500_DMICFILTCONF_DA3TOEAR 6 +#define AB8500_DMICFILTCONF_DMIC1SINC3 5 +#define AB8500_DMICFILTCONF_DMIC2SINC3 4 +#define AB8500_DMICFILTCONF_DMIC3SINC3 3 +#define AB8500_DMICFILTCONF_DMIC4SINC3 2 +#define AB8500_DMICFILTCONF_DMIC5SINC3 1 +#define AB8500_DMICFILTCONF_DMIC6SINC3 0 + +/* AB8500_DIGMULTCONF1 */ +#define AB8500_DIGMULTCONF1_DATOHSLEN 7 +#define AB8500_DIGMULTCONF1_DATOHSREN 6 +#define AB8500_DIGMULTCONF1_AD1SEL 5 +#define AB8500_DIGMULTCONF1_AD2SEL 4 +#define AB8500_DIGMULTCONF1_AD3SEL 3 +#define AB8500_DIGMULTCONF1_AD5SEL 2 +#define AB8500_DIGMULTCONF1_AD6SEL 1 +#define AB8500_DIGMULTCONF1_ANCSEL 0 + +/* AB8500_DIGMULTCONF2 */ +#define AB8500_DIGMULTCONF2_DATOHFREN 7 +#define AB8500_DIGMULTCONF2_DATOHFLEN 6 +#define AB8500_DIGMULTCONF2_HFRSEL 5 +#define AB8500_DIGMULTCONF2_HFLSEL 4 +#define AB8500_DIGMULTCONF2_FIRSID1SEL 2 +#define AB8500_DIGMULTCONF2_FIRSID2SEL 0 + +/* AB8500_ADDIGGAIN1 */ +/* AB8500_ADDIGGAIN2 */ +/* AB8500_ADDIGGAIN3 */ +/* AB8500_ADDIGGAIN4 */ +/* AB8500_ADDIGGAIN5 */ +/* AB8500_ADDIGGAIN6 */ +#define AB8500_ADDIGGAINX_FADEDISADX 6 +#define AB8500_ADDIGGAINX_ADXGAIN_MAX 0x3F + +/* AB8500_DADIGGAIN1 */ +/* AB8500_DADIGGAIN2 */ +/* AB8500_DADIGGAIN3 */ +/* AB8500_DADIGGAIN4 */ +/* AB8500_DADIGGAIN5 */ +/* AB8500_DADIGGAIN6 */ +#define AB8500_DADIGGAINX_FADEDISDAX 6 +#define AB8500_DADIGGAINX_DAXGAIN_MAX 0x3F + +/* AB8500_ADDIGLOOPGAIN1 */ +/* AB8500_ADDIGLOOPGAIN2 */ +#define AB8500_ADDIGLOOPGAINX_FADEDISADXL 6 +#define AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX 0x3F + +/* AB8500_HSLEARDIGGAIN */ +#define AB8500_HSLEARDIGGAIN_HSSINC1 7 +#define AB8500_HSLEARDIGGAIN_FADEDISHSL 4 +#define AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX 0x09 + +/* AB8500_HSRDIGGAIN */ +#define AB8500_HSRDIGGAIN_FADESPEED 6 +#define AB8500_HSRDIGGAIN_FADEDISHSR 4 +#define AB8500_HSRDIGGAIN_HSRDGAIN_MAX 0x09 + +/* AB8500_SIDFIRGAIN1 */ +/* AB8500_SIDFIRGAIN2 */ +#define AB8500_SIDFIRGAINX_FIRSIDXGAIN_MAX 0x1F + +/* AB8500_ANCCONF1 */ +#define AB8500_ANCCONF1_ANCIIRUPDATE 3 +#define AB8500_ANCCONF1_ENANC 2 +#define AB8500_ANCCONF1_ANCIIRINIT 1 +#define AB8500_ANCCONF1_ANCFIRUPDATE 0 + +/* AB8500_ANCCONF2 */ +#define AB8500_ANCCONF2_SHIFT 5 +#define AB8500_ANCCONF2_MIN -0x10 +#define AB8500_ANCCONF2_MAX 0xF + +/* AB8500_ANCCONF3 */ +#define AB8500_ANCCONF3_SHIFT 5 +#define AB8500_ANCCONF3_MIN -0x10 +#define AB8500_ANCCONF3_MAX 0xF + +/* AB8500_ANCCONF4 */ +#define AB8500_ANCCONF4_SHIFT 5 +#define AB8500_ANCCONF4_MIN -0x10 +#define AB8500_ANCCONF4_MAX 0xF + +/* AB8500_ANC_FIR_COEFFS */ +#define AB8500_ANC_FIR_COEFF_MIN -0x8000 +#define AB8500_ANC_FIR_COEFF_MAX 0x7FFF +#define AB8500_ANC_FIR_COEFFS 15 + +/* AB8500_ANC_IIR_COEFFS */ +#define AB8500_ANC_IIR_COEFF_MIN -0x800000 +#define AB8500_ANC_IIR_COEFF_MAX 0x7FFFFF +#define AB8500_ANC_IIR_COEFFS 24 +/* AB8500_ANC_WARP_DELAY */ +#define AB8500_ANC_WARP_DELAY_SHIFT 16 +#define AB8500_ANC_WARP_DELAY_MIN 0x0000 +#define AB8500_ANC_WARP_DELAY_MAX 0xFFFF + +/* AB8500_ANCCONF11 */ +/* AB8500_ANCCONF12 */ +/* AB8500_ANCCONF13 */ +/* AB8500_ANCCONF14 */ + +/* AB8500_SIDFIRADR */ +#define AB8500_SIDFIRADR_FIRSIDSET 7 +#define AB8500_SIDFIRADR_ADDRESS_SHIFT 0 +#define AB8500_SIDFIRADR_ADDRESS_MAX 0x7F + +/* AB8500_SIDFIRCOEF1 */ +/* AB8500_SIDFIRCOEF2 */ +#define AB8500_SID_FIR_COEFF_MIN 0 +#define AB8500_SID_FIR_COEFF_MAX 0xFFFF +#define AB8500_SID_FIR_COEFFS 128 + +/* AB8500_SIDFIRCONF */ +#define AB8500_SIDFIRCONF_ENFIRSIDS 2 +#define AB8500_SIDFIRCONF_FIRSIDSTOIF1 1 +#define AB8500_SIDFIRCONF_FIRSIDBUSY 0 + +/* AB8500_AUDINTMASK1 */ +/* AB8500_AUDINTSOURCE1 */ +/* AB8500_AUDINTMASK2 */ +/* AB8500_AUDINTSOURCE2 */ + +/* AB8500_FIFOCONF1 */ +#define AB8500_FIFOCONF1_BFIFOMASK 0x80 +#define AB8500_FIFOCONF1_BFIFO19M2 0x40 +#define AB8500_FIFOCONF1_BFIFOINT_SHIFT 0 +#define AB8500_FIFOCONF1_BFIFOINT_MAX 0x3F + +/* AB8500_FIFOCONF2 */ +#define AB8500_FIFOCONF2_BFIFOTX_SHIFT 0 +#define AB8500_FIFOCONF2_BFIFOTX_MAX 0xFF + +/* AB8500_FIFOCONF3 */ +#define AB8500_FIFOCONF3_BFIFOEXSL_SHIFT 5 +#define AB8500_FIFOCONF3_BFIFOEXSL_MAX 0x5 +#define AB8500_FIFOCONF3_PREBITCLK0_SHIFT 2 +#define AB8500_FIFOCONF3_PREBITCLK0_MAX 0x7 +#define AB8500_FIFOCONF3_BFIFOMAST_SHIFT 1 +#define AB8500_FIFOCONF3_BFIFORUN_SHIFT 0 + +/* AB8500_FIFOCONF4 */ +#define AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT 0 +#define AB8500_FIFOCONF4_BFIFOFRAMSW_MAX 0xFF + +/* AB8500_FIFOCONF5 */ +#define AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT 0 +#define AB8500_FIFOCONF5_BFIFOWAKEUP_MAX 0xFF + +/* AB8500_FIFOCONF6 */ +#define AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT 0 +#define AB8500_FIFOCONF6_BFIFOSAMPLE_MAX 0xFF + +/* AB8500_AUDREV */ + +#endif diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 2023c749f232..ea06b834a7de 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -91,11 +91,6 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) return 0; } -static int ac97_soc_remove(struct snd_soc_codec *codec) -{ - return 0; -} - #ifdef CONFIG_PM static int ac97_soc_suspend(struct snd_soc_codec *codec) { @@ -119,7 +114,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .write = ac97_write, .read = ac97_read, .probe = ac97_soc_probe, - .remove = ac97_soc_remove, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, }; diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c new file mode 100644 index 000000000000..5c9cacaf2d52 --- /dev/null +++ b/sound/soc/codecs/arizona.c @@ -0,0 +1,937 @@ +/* + * arizona.c - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gcd.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" + +#define ARIZONA_AIF_BCLK_CTRL 0x00 +#define ARIZONA_AIF_TX_PIN_CTRL 0x01 +#define ARIZONA_AIF_RX_PIN_CTRL 0x02 +#define ARIZONA_AIF_RATE_CTRL 0x03 +#define ARIZONA_AIF_FORMAT 0x04 +#define ARIZONA_AIF_TX_BCLK_RATE 0x05 +#define ARIZONA_AIF_RX_BCLK_RATE 0x06 +#define ARIZONA_AIF_FRAME_CTRL_1 0x07 +#define ARIZONA_AIF_FRAME_CTRL_2 0x08 +#define ARIZONA_AIF_FRAME_CTRL_3 0x09 +#define ARIZONA_AIF_FRAME_CTRL_4 0x0A +#define ARIZONA_AIF_FRAME_CTRL_5 0x0B +#define ARIZONA_AIF_FRAME_CTRL_6 0x0C +#define ARIZONA_AIF_FRAME_CTRL_7 0x0D +#define ARIZONA_AIF_FRAME_CTRL_8 0x0E +#define ARIZONA_AIF_FRAME_CTRL_9 0x0F +#define ARIZONA_AIF_FRAME_CTRL_10 0x10 +#define ARIZONA_AIF_FRAME_CTRL_11 0x11 +#define ARIZONA_AIF_FRAME_CTRL_12 0x12 +#define ARIZONA_AIF_FRAME_CTRL_13 0x13 +#define ARIZONA_AIF_FRAME_CTRL_14 0x14 +#define ARIZONA_AIF_FRAME_CTRL_15 0x15 +#define ARIZONA_AIF_FRAME_CTRL_16 0x16 +#define ARIZONA_AIF_FRAME_CTRL_17 0x17 +#define ARIZONA_AIF_FRAME_CTRL_18 0x18 +#define ARIZONA_AIF_TX_ENABLES 0x19 +#define ARIZONA_AIF_RX_ENABLES 0x1A +#define ARIZONA_AIF_FORCE_WRITE 0x1B + +#define arizona_fll_err(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_warn(_fll, fmt, ...) \ + dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_dbg(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + +#define arizona_aif_err(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_warn(_dai, fmt, ...) \ + dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_dbg(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + +const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { + "None", + "Tone Generator 1", + "Tone Generator 2", + "Haptics", + "AEC", + "Mic Mute Mixer", + "Noise Generator", + "IN1L", + "IN1R", + "IN2L", + "IN2R", + "IN3L", + "IN3R", + "IN4L", + "IN4R", + "AIF1RX1", + "AIF1RX2", + "AIF1RX3", + "AIF1RX4", + "AIF1RX5", + "AIF1RX6", + "AIF1RX7", + "AIF1RX8", + "AIF2RX1", + "AIF2RX2", + "AIF3RX1", + "AIF3RX2", + "SLIMRX1", + "SLIMRX2", + "SLIMRX3", + "SLIMRX4", + "SLIMRX5", + "SLIMRX6", + "SLIMRX7", + "SLIMRX8", + "EQ1", + "EQ2", + "EQ3", + "EQ4", + "DRC1L", + "DRC1R", + "DRC2L", + "DRC2R", + "LHPF1", + "LHPF2", + "LHPF3", + "LHPF4", + "DSP1.1", + "DSP1.2", + "DSP1.3", + "DSP1.4", + "DSP1.5", + "DSP1.6", + "ASRC1L", + "ASRC1R", + "ASRC2L", + "ASRC2R", +}; +EXPORT_SYMBOL_GPL(arizona_mixer_texts); + +int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { + 0x00, /* None */ + 0x04, /* Tone */ + 0x05, + 0x06, /* Haptics */ + 0x08, /* AEC */ + 0x0c, /* Noise mixer */ + 0x0d, /* Comfort noise */ + 0x10, /* IN1L */ + 0x11, + 0x12, + 0x13, + 0x14, + 0x15, + 0x16, + 0x17, + 0x20, /* AIF1RX1 */ + 0x21, + 0x22, + 0x23, + 0x24, + 0x25, + 0x26, + 0x27, + 0x28, /* AIF2RX1 */ + 0x29, + 0x30, /* AIF3RX1 */ + 0x31, + 0x38, /* SLIMRX1 */ + 0x39, + 0x3a, + 0x3b, + 0x3c, + 0x3d, + 0x3e, + 0x3f, + 0x50, /* EQ1 */ + 0x51, + 0x52, + 0x53, + 0x58, /* DRC1L */ + 0x59, + 0x5a, + 0x5b, + 0x60, /* LHPF1 */ + 0x61, + 0x62, + 0x63, + 0x68, /* DSP1.1 */ + 0x69, + 0x6a, + 0x6b, + 0x6c, + 0x6d, + 0x90, /* ASRC1L */ + 0x91, + 0x92, + 0x93, +}; +EXPORT_SYMBOL_GPL(arizona_mixer_values); + +const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); +EXPORT_SYMBOL_GPL(arizona_mixer_tlv); + +static const char *arizona_lhpf_mode_text[] = { + "Low-pass", "High-pass" +}; + +const struct soc_enum arizona_lhpf1_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf1_mode); + +const struct soc_enum arizona_lhpf2_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf2_mode); + +const struct soc_enum arizona_lhpf3_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf3_mode); + +const struct soc_enum arizona_lhpf4_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); + +int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_in_ev); + +int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_out_ev); + +int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + char *name; + unsigned int reg; + unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK; + unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT; + unsigned int *clk; + + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + name = "SYSCLK"; + reg = ARIZONA_SYSTEM_CLOCK_1; + clk = &priv->sysclk; + mask |= ARIZONA_SYSCLK_FRAC; + break; + case ARIZONA_CLK_ASYNCCLK: + name = "ASYNCCLK"; + reg = ARIZONA_ASYNC_CLOCK_1; + clk = &priv->asyncclk; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 6144000: + break; + case 11289600: + case 12288000: + val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 22579200: + case 24576000: + val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 45158400: + case 49152000: + val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + default: + return -EINVAL; + } + + *clk = freq; + + if (freq % 6144000) + val |= ARIZONA_SYSCLK_FRAC; + + dev_dbg(arizona->dev, "%s set to %uHz", name, freq); + + return regmap_update_bits(arizona->regmap, reg, mask, val); +} +EXPORT_SYMBOL_GPL(arizona_set_sysclk); + +static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int lrclk, bclk, mode, base; + + base = dai->driver->base; + + lrclk = 0; + bclk = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + mode = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + break; + case SND_SOC_DAIFMT_I2S: + mode = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode = 3; + break; + default: + arizona_aif_err(dai, "Unsupported DAI format %d\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + default: + arizona_aif_err(dai, "Unsupported master mode %d\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= ARIZONA_AIF1_BCLK_INV; + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= ARIZONA_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR, + bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL, + ARIZONA_AIF1TX_LRCLK_INV | + ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL, + ARIZONA_AIF1RX_LRCLK_INV | + ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT, + ARIZONA_AIF1_FMT_MASK, mode); + + return 0; +} + +static const int arizona_48k_bclk_rates[] = { + -1, + 48000, + 64000, + 96000, + 128000, + 192000, + 256000, + 384000, + 512000, + 768000, + 1024000, + 1536000, + 2048000, + 3072000, + 4096000, + 6144000, + 8192000, + 12288000, + 24576000, +}; + +static const unsigned int arizona_48k_rates[] = { + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = { + .count = ARRAY_SIZE(arizona_48k_rates), + .list = arizona_48k_rates, +}; + +static const int arizona_44k1_bclk_rates[] = { + -1, + 44100, + 58800, + 88200, + 117600, + 177640, + 235200, + 352800, + 470400, + 705600, + 940800, + 1411200, + 1881600, + 2882400, + 3763200, + 5644800, + 7526400, + 11289600, + 22579200, +}; + +static const unsigned int arizona_44k1_rates[] = { + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, +}; + +static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = { + .count = ARRAY_SIZE(arizona_44k1_rates), + .list = arizona_44k1_rates, +}; + +static int arizona_sr_vals[] = { + 0, + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 0, + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static int arizona_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + const struct snd_pcm_hw_constraint_list *constraint; + unsigned int base_rate; + + switch (dai_priv->clk) { + case ARIZONA_CLK_SYSCLK: + base_rate = priv->sysclk; + break; + case ARIZONA_CLK_ASYNCCLK: + base_rate = priv->asyncclk; + break; + default: + return 0; + } + + if (base_rate % 8000) + constraint = &arizona_44k1_constraint; + else + constraint = &arizona_48k_constraint; + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + constraint); +} + +static int arizona_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + int base = dai->driver->base; + const int *rates; + int i; + int bclk, lrclk, wl, frame, sr_val; + + if (params_rate(params) % 8000) + rates = &arizona_44k1_bclk_rates[0]; + else + rates = &arizona_48k_bclk_rates[0]; + + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { + if (rates[i] >= snd_soc_params_to_bclk(params) && + rates[i] % params_rate(params) == 0) { + bclk = i; + break; + } + } + if (i == ARRAY_SIZE(arizona_44k1_bclk_rates)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) + if (arizona_sr_vals[i] == params_rate(params)) + break; + if (i == ARRAY_SIZE(arizona_sr_vals)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + sr_val = i; + + lrclk = snd_soc_params_to_bclk(params) / params_rate(params); + + arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", + rates[bclk], rates[bclk] / lrclk); + + wl = snd_pcm_format_width(params_format(params)); + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for SYSCLK. + */ + switch (dai_priv->clk) { + case ARIZONA_CLK_SYSCLK: + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); + break; + case ARIZONA_CLK_ASYNCCLK: + snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, + ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 8); + break; + default: + arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); + return -EINVAL; + } + + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + + return 0; +} + +static const char *arizona_dai_clk_str(int clk_id) +{ + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + return "SYSCLK"; + case ARIZONA_CLK_ASYNCCLK: + return "ASYNCCLK"; + default: + return "Unknown clock"; + } +} + +static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + struct snd_soc_dapm_route routes[2]; + + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + case ARIZONA_CLK_ASYNCCLK: + break; + default: + return -EINVAL; + } + + if (clk_id == dai_priv->clk) + return 0; + + if (dai->active) { + dev_err(codec->dev, "Can't change clock on active DAI %d\n", + dai->id); + return -EBUSY; + } + + memset(&routes, 0, sizeof(routes)); + routes[0].sink = dai->driver->capture.stream_name; + routes[1].sink = dai->driver->playback.stream_name; + + routes[0].source = arizona_dai_clk_str(dai_priv->clk); + routes[1].source = arizona_dai_clk_str(dai_priv->clk); + snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + + routes[0].source = arizona_dai_clk_str(clk_id); + routes[1].source = arizona_dai_clk_str(clk_id); + snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + + return snd_soc_dapm_sync(&codec->dapm); +} + +const struct snd_soc_dai_ops arizona_dai_ops = { + .startup = arizona_startup, + .set_fmt = arizona_set_fmt, + .hw_params = arizona_hw_params, + .set_sysclk = arizona_dai_set_sysclk, +}; +EXPORT_SYMBOL_GPL(arizona_dai_ops); + +int arizona_init_dai(struct arizona_priv *priv, int id) +{ + struct arizona_dai_priv *dai_priv = &priv->dai[id]; + + dai_priv->clk = ARIZONA_CLK_SYSCLK; + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_dai); + +static irqreturn_t arizona_fll_lock(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "Locked\n"); + + complete(&fll->lock); + + return IRQ_HANDLED; +} + +static irqreturn_t arizona_fll_clock_ok(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "clock OK\n"); + + complete(&fll->ok); + + return IRQ_HANDLED; +} + +static struct { + unsigned int min; + unsigned int max; + u16 fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +struct arizona_fll_cfg { + int n; + int theta; + int lambda; + int refdiv; + int outdiv; + int fratio; +}; + +static int arizona_calc_fll(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int Fref, + unsigned int Fout) +{ + unsigned int target, div, gcd_fll; + int i, ratio; + + arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout); + + /* Fref must be <=13.5MHz */ + div = 1; + cfg->refdiv = 0; + while ((Fref / div) > 13500000) { + div *= 2; + cfg->refdiv++; + + if (div > 8) { + arizona_fll_err(fll, + "Can't scale %dMHz in to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be over the targt; don't check the upper bound */ + div = 1; + while (Fout * div < 90000000 * fll->vco_mult) { + div++; + if (div > 7) { + arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div / fll->vco_mult; + cfg->outdiv = div; + + arizona_fll_dbg(fll, "Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + cfg->fratio = fll_fratios[i].fratio; + ratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", + Fref); + return -EINVAL; + } + + cfg->n = target / (ratio * Fref); + + if (target % Fref) { + gcd_fll = gcd(target, ratio * Fref); + arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll); + + cfg->theta = (target - (cfg->n * ratio * Fref)) + / gcd_fll; + cfg->lambda = (ratio * Fref) / gcd_fll; + } else { + cfg->theta = 0; + cfg->lambda = 0; + } + + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", + cfg->n, cfg->theta, cfg->lambda); + arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", + cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); + + return 0; + +} + +static void arizona_apply_fll(struct arizona *arizona, unsigned int base, + struct arizona_fll_cfg *cfg, int source) +{ + regmap_update_bits(arizona->regmap, base + 3, + ARIZONA_FLL1_THETA_MASK, cfg->theta); + regmap_update_bits(arizona->regmap, base + 4, + ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); + regmap_update_bits(arizona->regmap, base + 5, + ARIZONA_FLL1_FRATIO_MASK, + cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); + regmap_update_bits(arizona->regmap, base + 6, + ARIZONA_FLL1_CLK_REF_DIV_MASK | + ARIZONA_FLL1_CLK_REF_SRC_MASK, + cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | + source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + + regmap_update_bits(arizona->regmap, base + 2, + ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, + ARIZONA_FLL1_CTRL_UPD | cfg->n); +} + +int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout) +{ + struct arizona *arizona = fll->arizona; + struct arizona_fll_cfg cfg, sync; + unsigned int reg, val; + int syncsrc; + bool ena; + int ret; + + ret = regmap_read(arizona->regmap, fll->base + 1, ®); + if (ret != 0) { + arizona_fll_err(fll, "Failed to read current state: %d\n", + ret); + return ret; + } + ena = reg & ARIZONA_FLL1_ENA; + + if (Fout) { + /* Do we have a 32kHz reference? */ + regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); + switch (val & ARIZONA_CLK_32K_SRC_MASK) { + case ARIZONA_CLK_SRC_MCLK1: + case ARIZONA_CLK_SRC_MCLK2: + syncsrc = val & ARIZONA_CLK_32K_SRC_MASK; + break; + default: + syncsrc = -1; + } + + if (source == syncsrc) + syncsrc = -1; + + if (syncsrc >= 0) { + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; + + ret = arizona_calc_fll(fll, &cfg, 32768, Fout); + if (ret != 0) + return ret; + } else { + ret = arizona_calc_fll(fll, &cfg, Fref, Fout); + if (ret != 0) + return ret; + } + } else { + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, 0); + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); + + if (ena) + pm_runtime_put_autosuspend(arizona->dev); + + return 0; + } + + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + if (syncsrc >= 0) { + arizona_apply_fll(arizona, fll->base, &cfg, syncsrc); + arizona_apply_fll(arizona, fll->base + 0x10, &sync, source); + } else { + arizona_apply_fll(arizona, fll->base, &cfg, source); + } + + if (!ena) + pm_runtime_get(arizona->dev); + + /* Clear any pending completions */ + try_wait_for_completion(&fll->ok); + + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + if (syncsrc >= 0) + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); + + ret = wait_for_completion_timeout(&fll->ok, + msecs_to_jiffies(25)); + if (ret == 0) + arizona_fll_warn(fll, "Timed out waiting for lock\n"); + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_set_fll); + +int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, + int ok_irq, struct arizona_fll *fll) +{ + int ret; + + init_completion(&fll->lock); + init_completion(&fll->ok); + + fll->id = id; + fll->base = base; + fll->arizona = arizona; + + snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id); + snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), + "FLL%d clock OK", id); + + ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, + arizona_fll_lock, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", + id, ret); + } + + ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, + arizona_fll_clock_ok, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", + id, ret); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_fll); + +MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h new file mode 100644 index 000000000000..59caca8865e8 --- /dev/null +++ b/sound/soc/codecs/arizona.h @@ -0,0 +1,159 @@ +/* + * arizona.h - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ASOC_ARIZONA_H +#define _ASOC_ARIZONA_H + +#include <linux/completion.h> + +#include <sound/soc.h> + +#define ARIZONA_CLK_SYSCLK 1 +#define ARIZONA_CLK_ASYNCCLK 2 + +#define ARIZONA_CLK_SRC_MCLK1 0x0 +#define ARIZONA_CLK_SRC_MCLK2 0x1 +#define ARIZONA_CLK_SRC_FLL1 0x4 +#define ARIZONA_CLK_SRC_FLL2 0x5 +#define ARIZONA_CLK_SRC_AIF1BCLK 0x8 +#define ARIZONA_CLK_SRC_AIF2BCLK 0x9 +#define ARIZONA_CLK_SRC_AIF3BCLK 0xa + +#define ARIZONA_FLL_SRC_MCLK1 0 +#define ARIZONA_FLL_SRC_MCLK2 1 +#define ARIZONA_FLL_SRC_SLIMCLK 2 +#define ARIZONA_FLL_SRC_FLL1 3 +#define ARIZONA_FLL_SRC_FLL2 4 +#define ARIZONA_FLL_SRC_AIF1BCLK 5 +#define ARIZONA_FLL_SRC_AIF2BCLK 6 +#define ARIZONA_FLL_SRC_AIF3BCLK 7 +#define ARIZONA_FLL_SRC_AIF1LRCLK 8 +#define ARIZONA_FLL_SRC_AIF2LRCLK 9 +#define ARIZONA_FLL_SRC_AIF3LRCLK 10 + +#define ARIZONA_MIXER_VOL_MASK 0x00FE +#define ARIZONA_MIXER_VOL_SHIFT 1 +#define ARIZONA_MIXER_VOL_WIDTH 7 + +#define ARIZONA_MAX_DAI 3 + +struct arizona; + +struct arizona_dai_priv { + int clk; +}; + +struct arizona_priv { + struct arizona *arizona; + int sysclk; + int asyncclk; + struct arizona_dai_priv dai[ARIZONA_MAX_DAI]; +}; + +#define ARIZONA_NUM_MIXER_INPUTS 57 + +extern const unsigned int arizona_mixer_tlv[]; +extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; +extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; + +#define ARIZONA_MIXER_CONTROLS(name, base) \ + SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 2 Volume", base + 3, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 3 Volume", base + 5, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv) + +#define ARIZONA_MUX_ENUM_DECL(name, reg) \ + SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \ + arizona_mixer_texts, arizona_mixer_values) + +#define ARIZONA_MUX_CTL_DECL(name) \ + const struct snd_kcontrol_new name##_mux = \ + SOC_DAPM_VALUE_ENUM("Route", name##_enum) + +#define ARIZONA_MIXER_ENUMS(name, base_reg) \ + static ARIZONA_MUX_ENUM_DECL(name##_in1_enum, base_reg); \ + static ARIZONA_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \ + static ARIZONA_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \ + static ARIZONA_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \ + static ARIZONA_MUX_CTL_DECL(name##_in1); \ + static ARIZONA_MUX_CTL_DECL(name##_in2); \ + static ARIZONA_MUX_CTL_DECL(name##_in3); \ + static ARIZONA_MUX_CTL_DECL(name##_in4) + +#define ARIZONA_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +#define ARIZONA_MIXER_WIDGETS(name, name_str) \ + ARIZONA_MUX(name_str " Input 1", &name##_in1_mux), \ + ARIZONA_MUX(name_str " Input 2", &name##_in2_mux), \ + ARIZONA_MUX(name_str " Input 3", &name##_in3_mux), \ + ARIZONA_MUX(name_str " Input 4", &name##_in4_mux), \ + SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0) + +#define ARIZONA_MIXER_ROUTES(widget, name) \ + { widget, NULL, name " Mixer" }, \ + { name " Mixer", NULL, name " Input 1" }, \ + { name " Mixer", NULL, name " Input 2" }, \ + { name " Mixer", NULL, name " Input 3" }, \ + { name " Mixer", NULL, name " Input 4" }, \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 1"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 2"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 3"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 4") + +extern const struct soc_enum arizona_lhpf1_mode; +extern const struct soc_enum arizona_lhpf2_mode; +extern const struct soc_enum arizona_lhpf3_mode; +extern const struct soc_enum arizona_lhpf4_mode; + +extern int arizona_in_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); +extern int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); + +extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); + +extern const struct snd_soc_dai_ops arizona_dai_ops; + +#define ARIZONA_FLL_NAME_LEN 20 + +struct arizona_fll { + struct arizona *arizona; + int id; + unsigned int base; + unsigned int vco_mult; + struct completion lock; + struct completion ok; + + char lock_name[ARIZONA_FLL_NAME_LEN]; + char clock_ok_name[ARIZONA_FLL_NAME_LEN]; +}; + +extern int arizona_init_fll(struct arizona *arizona, int id, int base, + int lock_irq, int ok_irq, struct arizona_fll *fll); +extern int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout); + +extern int arizona_init_dai(struct arizona_priv *priv, int dai); + +#endif diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index a7109413aef1..628daf6a1d97 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -14,7 +14,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> @@ -1217,11 +1216,11 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; cs42l52->dev = &i2c_client->dev; - cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap); + cs42l52->regmap = devm_regmap_init_i2c(i2c_client, &cs42l52_regmap); if (IS_ERR(cs42l52->regmap)) { ret = PTR_ERR(cs42l52->regmap); dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); - goto err; + return ret; } i2c_set_clientdata(i2c_client, cs42l52); @@ -1243,7 +1242,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L52 Device ID (%X). Expected %X\n", devid, CS42L52_CHIP_ID); - goto err_regmap; + return ret; } regcache_cache_only(cs42l52->regmap, true); @@ -1251,23 +1250,13 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_cs42l52, &cs42l52_dai, 1); if (ret < 0) - goto err_regmap; + return ret; return 0; - -err_regmap: - regmap_exit(cs42l52->regmap); - -err: - return ret; } static int cs42l52_i2c_remove(struct i2c_client *client) { - struct cs42l52_private *cs42l52 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(cs42l52->regmap); - return 0; } diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index e0d45fdaa750..2c08c4cb465a 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1362,11 +1362,11 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, i2c_set_clientdata(i2c_client, cs42l73); - cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap); + cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap); if (IS_ERR(cs42l73->regmap)) { ret = PTR_ERR(cs42l73->regmap); dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); - goto err; + return ret; } /* initialize codec */ ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, ®); @@ -1384,13 +1384,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L73 Device ID (%X). Expected %X\n", devid, CS42L73_DEVID); - goto err_regmap; + return ret; } ret = regmap_read(cs42l73->regmap, CS42L73_REVID, ®); if (ret < 0) { dev_err(&i2c_client->dev, "Get Revision ID failed\n"); - goto err_regmap; + return ret;; } dev_info(&i2c_client->dev, @@ -1402,23 +1402,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, &soc_codec_dev_cs42l73, cs42l73_dai, ARRAY_SIZE(cs42l73_dai)); if (ret < 0) - goto err_regmap; + return ret; return 0; - -err_regmap: - regmap_exit(cs42l73->regmap); - -err: - return ret; } static __devexit int cs42l73_i2c_remove(struct i2c_client *client) { - struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(cs42l73->regmap); - return 0; } diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c new file mode 100644 index 000000000000..01be2a320e21 --- /dev/null +++ b/sound/soc/codecs/da732x.c @@ -0,0 +1,1627 @@ +/* + * da732x.c --- Dialog DA732X ALSA SoC Audio Driver + * + * Copyright (C) 2012 Dialog Semiconductor GmbH + * + * Author: Michal Hajduk <Michal.Hajduk@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/sysfs.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> + +#include "da732x.h" +#include "da732x_reg.h" + + +struct da732x_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + + unsigned int sysclk; + bool pll_en; +}; + +/* + * da732x register cache - default settings + */ +static struct reg_default da732x_reg_cache[] = { + { DA732X_REG_REF1 , 0x02 }, + { DA732X_REG_BIAS_EN , 0x80 }, + { DA732X_REG_BIAS1 , 0x00 }, + { DA732X_REG_BIAS2 , 0x00 }, + { DA732X_REG_BIAS3 , 0x00 }, + { DA732X_REG_BIAS4 , 0x00 }, + { DA732X_REG_MICBIAS2 , 0x00 }, + { DA732X_REG_MICBIAS1 , 0x00 }, + { DA732X_REG_MICDET , 0x00 }, + { DA732X_REG_MIC1_PRE , 0x01 }, + { DA732X_REG_MIC1 , 0x40 }, + { DA732X_REG_MIC2_PRE , 0x01 }, + { DA732X_REG_MIC2 , 0x40 }, + { DA732X_REG_AUX1L , 0x75 }, + { DA732X_REG_AUX1R , 0x75 }, + { DA732X_REG_MIC3_PRE , 0x01 }, + { DA732X_REG_MIC3 , 0x40 }, + { DA732X_REG_INP_PINBIAS , 0x00 }, + { DA732X_REG_INP_ZC_EN , 0x00 }, + { DA732X_REG_INP_MUX , 0x50 }, + { DA732X_REG_HP_DET , 0x00 }, + { DA732X_REG_HPL_DAC_OFFSET , 0x00 }, + { DA732X_REG_HPL_DAC_OFF_CNTL , 0x00 }, + { DA732X_REG_HPL_OUT_OFFSET , 0x00 }, + { DA732X_REG_HPL , 0x40 }, + { DA732X_REG_HPL_VOL , 0x0F }, + { DA732X_REG_HPR_DAC_OFFSET , 0x00 }, + { DA732X_REG_HPR_DAC_OFF_CNTL , 0x00 }, + { DA732X_REG_HPR_OUT_OFFSET , 0x00 }, + { DA732X_REG_HPR , 0x40 }, + { DA732X_REG_HPR_VOL , 0x0F }, + { DA732X_REG_LIN2 , 0x4F }, + { DA732X_REG_LIN3 , 0x4F }, + { DA732X_REG_LIN4 , 0x4F }, + { DA732X_REG_OUT_ZC_EN , 0x00 }, + { DA732X_REG_HP_LIN1_GNDSEL , 0x00 }, + { DA732X_REG_CP_HP1 , 0x0C }, + { DA732X_REG_CP_HP2 , 0x03 }, + { DA732X_REG_CP_CTRL1 , 0x00 }, + { DA732X_REG_CP_CTRL2 , 0x99 }, + { DA732X_REG_CP_CTRL3 , 0x25 }, + { DA732X_REG_CP_LEVEL_MASK , 0x3F }, + { DA732X_REG_CP_DET , 0x00 }, + { DA732X_REG_CP_STATUS , 0x00 }, + { DA732X_REG_CP_THRESH1 , 0x00 }, + { DA732X_REG_CP_THRESH2 , 0x00 }, + { DA732X_REG_CP_THRESH3 , 0x00 }, + { DA732X_REG_CP_THRESH4 , 0x00 }, + { DA732X_REG_CP_THRESH5 , 0x00 }, + { DA732X_REG_CP_THRESH6 , 0x00 }, + { DA732X_REG_CP_THRESH7 , 0x00 }, + { DA732X_REG_CP_THRESH8 , 0x00 }, + { DA732X_REG_PLL_DIV_LO , 0x00 }, + { DA732X_REG_PLL_DIV_MID , 0x00 }, + { DA732X_REG_PLL_DIV_HI , 0x00 }, + { DA732X_REG_PLL_CTRL , 0x02 }, + { DA732X_REG_CLK_CTRL , 0xaa }, + { DA732X_REG_CLK_DSP , 0x07 }, + { DA732X_REG_CLK_EN1 , 0x00 }, + { DA732X_REG_CLK_EN2 , 0x00 }, + { DA732X_REG_CLK_EN3 , 0x00 }, + { DA732X_REG_CLK_EN4 , 0x00 }, + { DA732X_REG_CLK_EN5 , 0x00 }, + { DA732X_REG_AIF_MCLK , 0x00 }, + { DA732X_REG_AIFA1 , 0x02 }, + { DA732X_REG_AIFA2 , 0x00 }, + { DA732X_REG_AIFA3 , 0x08 }, + { DA732X_REG_AIFB1 , 0x02 }, + { DA732X_REG_AIFB2 , 0x00 }, + { DA732X_REG_AIFB3 , 0x08 }, + { DA732X_REG_PC_CTRL , 0xC0 }, + { DA732X_REG_DATA_ROUTE , 0x00 }, + { DA732X_REG_DSP_CTRL , 0x00 }, + { DA732X_REG_CIF_CTRL2 , 0x00 }, + { DA732X_REG_HANDSHAKE , 0x00 }, + { DA732X_REG_SPARE1_OUT , 0x00 }, + { DA732X_REG_SPARE2_OUT , 0x00 }, + { DA732X_REG_SPARE1_IN , 0x00 }, + { DA732X_REG_ADC1_PD , 0x00 }, + { DA732X_REG_ADC1_HPF , 0x00 }, + { DA732X_REG_ADC1_SEL , 0x00 }, + { DA732X_REG_ADC1_EQ12 , 0x00 }, + { DA732X_REG_ADC1_EQ34 , 0x00 }, + { DA732X_REG_ADC1_EQ5 , 0x00 }, + { DA732X_REG_ADC2_PD , 0x00 }, + { DA732X_REG_ADC2_HPF , 0x00 }, + { DA732X_REG_ADC2_SEL , 0x00 }, + { DA732X_REG_ADC2_EQ12 , 0x00 }, + { DA732X_REG_ADC2_EQ34 , 0x00 }, + { DA732X_REG_ADC2_EQ5 , 0x00 }, + { DA732X_REG_DAC1_HPF , 0x00 }, + { DA732X_REG_DAC1_L_VOL , 0x00 }, + { DA732X_REG_DAC1_R_VOL , 0x00 }, + { DA732X_REG_DAC1_SEL , 0x00 }, + { DA732X_REG_DAC1_SOFTMUTE , 0x00 }, + { DA732X_REG_DAC1_EQ12 , 0x00 }, + { DA732X_REG_DAC1_EQ34 , 0x00 }, + { DA732X_REG_DAC1_EQ5 , 0x00 }, + { DA732X_REG_DAC2_HPF , 0x00 }, + { DA732X_REG_DAC2_L_VOL , 0x00 }, + { DA732X_REG_DAC2_R_VOL , 0x00 }, + { DA732X_REG_DAC2_SEL , 0x00 }, + { DA732X_REG_DAC2_SOFTMUTE , 0x00 }, + { DA732X_REG_DAC2_EQ12 , 0x00 }, + { DA732X_REG_DAC2_EQ34 , 0x00 }, + { DA732X_REG_DAC2_EQ5 , 0x00 }, + { DA732X_REG_DAC3_HPF , 0x00 }, + { DA732X_REG_DAC3_VOL , 0x00 }, + { DA732X_REG_DAC3_SEL , 0x00 }, + { DA732X_REG_DAC3_SOFTMUTE , 0x00 }, + { DA732X_REG_DAC3_EQ12 , 0x00 }, + { DA732X_REG_DAC3_EQ34 , 0x00 }, + { DA732X_REG_DAC3_EQ5 , 0x00 }, + { DA732X_REG_BIQ_BYP , 0x00 }, + { DA732X_REG_DMA_CMD , 0x00 }, + { DA732X_REG_DMA_ADDR0 , 0x00 }, + { DA732X_REG_DMA_ADDR1 , 0x00 }, + { DA732X_REG_DMA_DATA0 , 0x00 }, + { DA732X_REG_DMA_DATA1 , 0x00 }, + { DA732X_REG_DMA_DATA2 , 0x00 }, + { DA732X_REG_DMA_DATA3 , 0x00 }, + { DA732X_REG_UNLOCK , 0x00 }, +}; + +static inline int da732x_get_input_div(struct snd_soc_codec *codec, int sysclk) +{ + int val; + int ret; + + if (sysclk < DA732X_MCLK_10MHZ) { + val = DA732X_MCLK_RET_0_10MHZ; + ret = DA732X_MCLK_VAL_0_10MHZ; + } else if ((sysclk >= DA732X_MCLK_10MHZ) && + (sysclk < DA732X_MCLK_20MHZ)) { + val = DA732X_MCLK_RET_10_20MHZ; + ret = DA732X_MCLK_VAL_10_20MHZ; + } else if ((sysclk >= DA732X_MCLK_20MHZ) && + (sysclk < DA732X_MCLK_40MHZ)) { + val = DA732X_MCLK_RET_20_40MHZ; + ret = DA732X_MCLK_VAL_20_40MHZ; + } else if ((sysclk >= DA732X_MCLK_40MHZ) && + (sysclk <= DA732X_MCLK_54MHZ)) { + val = DA732X_MCLK_RET_40_54MHZ; + ret = DA732X_MCLK_VAL_40_54MHZ; + } else { + return -EINVAL; + } + + snd_soc_write(codec, DA732X_REG_PLL_CTRL, val); + + return ret; +} + +static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state) +{ + switch (state) { + case DA732X_ENABLE_CP: + snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_EN); + snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_EN | + DA732X_HP_CP_REG | DA732X_HP_CP_PULSESKIP); + snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA732X_CP_EN | + DA732X_CP_CTRL_CPVDD1); + snd_soc_write(codec, DA732X_REG_CP_CTRL2, + DA732X_CP_MANAGE_MAGNITUDE | DA732X_CP_BOOST); + snd_soc_write(codec, DA732X_REG_CP_CTRL3, DA732X_CP_1MHZ); + break; + case DA732X_DISABLE_CP: + snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_DIS); + snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_DIS); + snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS); + break; + default: + pr_err(KERN_ERR "Wrong charge pump state\n"); + break; + } +} + +static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, DA732X_MIC_PRE_VOL_DB_MIN, + DA732X_MIC_PRE_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, DA732X_MIC_VOL_DB_MIN, + DA732X_MIC_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(aux_pga_tlv, DA732X_AUX_VOL_DB_MIN, + DA732X_AUX_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(hp_pga_tlv, DA732X_HP_VOL_DB_MIN, + DA732X_AUX_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(lin2_pga_tlv, DA732X_LIN2_VOL_DB_MIN, + DA732X_LIN2_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(lin3_pga_tlv, DA732X_LIN3_VOL_DB_MIN, + DA732X_LIN3_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(lin4_pga_tlv, DA732X_LIN4_VOL_DB_MIN, + DA732X_LIN4_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(adc_pga_tlv, DA732X_ADC_VOL_DB_MIN, + DA732X_ADC_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(dac_pga_tlv, DA732X_DAC_VOL_DB_MIN, + DA732X_DAC_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(eq_band_pga_tlv, DA732X_EQ_BAND_VOL_DB_MIN, + DA732X_EQ_BAND_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(eq_overall_tlv, DA732X_EQ_OVERALL_VOL_DB_MIN, + DA732X_EQ_OVERALL_VOL_DB_INC, 0); + +/* High Pass Filter */ +static const char *da732x_hpf_mode[] = { + "Disable", "Music", "Voice", +}; + +static const char *da732x_hpf_music[] = { + "1.8Hz", "3.75Hz", "7.5Hz", "15Hz", +}; + +static const char *da732x_hpf_voice[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", + "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da732x_dac1_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_dac2_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_dac3_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_adc1_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_adc2_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_dac1_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_dac2_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_dac3_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_adc1_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_adc2_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_dac1_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_dac2_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_dac3_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_adc1_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_adc2_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + + +static int da732x_hpf_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; + unsigned int reg = enum_ctrl->reg; + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int bits; + + switch (sel) { + case DA732X_HPF_DISABLED: + bits = DA732X_HPF_DIS; + break; + case DA732X_HPF_VOICE: + bits = DA732X_HPF_VOICE_EN; + break; + case DA732X_HPF_MUSIC: + bits = DA732X_HPF_MUSIC_EN; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, reg, DA732X_HPF_MASK, bits); + + return 0; +} + +static int da732x_hpf_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; + unsigned int reg = enum_ctrl->reg; + int val; + + val = snd_soc_read(codec, reg) & DA732X_HPF_MASK; + + switch (val) { + case DA732X_HPF_VOICE_EN: + ucontrol->value.integer.value[0] = DA732X_HPF_VOICE; + break; + case DA732X_HPF_MUSIC_EN: + ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC; + break; + default: + ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED; + break; + } + + return 0; +} + +static const struct snd_kcontrol_new da732x_snd_controls[] = { + /* Input PGAs */ + SOC_SINGLE_RANGE_TLV("MIC1 Boost Volume", DA732X_REG_MIC1_PRE, + DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN, + DA732X_MICBOOST_MAX, 0, mic_boost_tlv), + SOC_SINGLE_RANGE_TLV("MIC2 Boost Volume", DA732X_REG_MIC2_PRE, + DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN, + DA732X_MICBOOST_MAX, 0, mic_boost_tlv), + SOC_SINGLE_RANGE_TLV("MIC3 Boost Volume", DA732X_REG_MIC3_PRE, + DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN, + DA732X_MICBOOST_MAX, 0, mic_boost_tlv), + + /* MICs */ + SOC_SINGLE("MIC1 Switch", DA732X_REG_MIC1, DA732X_MIC_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_RANGE_TLV("MIC1 Volume", DA732X_REG_MIC1, + DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN, + DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv), + SOC_SINGLE("MIC2 Switch", DA732X_REG_MIC2, DA732X_MIC_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_RANGE_TLV("MIC2 Volume", DA732X_REG_MIC2, + DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN, + DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv), + SOC_SINGLE("MIC3 Switch", DA732X_REG_MIC3, DA732X_MIC_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_RANGE_TLV("MIC3 Volume", DA732X_REG_MIC3, + DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN, + DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv), + + /* AUXs */ + SOC_SINGLE("AUX1L Switch", DA732X_REG_AUX1L, DA732X_AUX_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("AUX1L Volume", DA732X_REG_AUX1L, + DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX, + DA732X_NO_INVERT, aux_pga_tlv), + SOC_SINGLE("AUX1R Switch", DA732X_REG_AUX1R, DA732X_AUX_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("AUX1R Volume", DA732X_REG_AUX1R, + DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX, + DA732X_NO_INVERT, aux_pga_tlv), + + /* ADCs */ + SOC_DOUBLE_TLV("ADC1 Volume", DA732X_REG_ADC1_SEL, + DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT, + DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv), + + SOC_DOUBLE_TLV("ADC2 Volume", DA732X_REG_ADC2_SEL, + DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT, + DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv), + + /* DACs */ + SOC_DOUBLE("Digital Playback DAC12 Switch", DA732X_REG_DAC1_SEL, + DA732X_DACL_MUTE_SHIFT, DA732X_DACR_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_DOUBLE_R_TLV("Digital Playback DAC12 Volume", DA732X_REG_DAC1_L_VOL, + DA732X_REG_DAC1_R_VOL, DA732X_DAC_VOL_SHIFT, + DA732X_DAC_VOL_VAL_MAX, DA732X_INVERT, dac_pga_tlv), + SOC_SINGLE("Digital Playback DAC3 Switch", DA732X_REG_DAC2_SEL, + DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Digital Playback DAC3 Volume", DA732X_REG_DAC2_L_VOL, + DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX, + DA732X_INVERT, dac_pga_tlv), + SOC_SINGLE("Digital Playback DAC4 Switch", DA732X_REG_DAC2_SEL, + DA732X_DACR_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Digital Playback DAC4 Volume", DA732X_REG_DAC2_R_VOL, + DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX, + DA732X_INVERT, dac_pga_tlv), + SOC_SINGLE("Digital Playback DAC5 Switch", DA732X_REG_DAC3_SEL, + DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Digital Playback DAC5 Volume", DA732X_REG_DAC3_VOL, + DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX, + DA732X_INVERT, dac_pga_tlv), + + /* High Pass Filters */ + SOC_ENUM_EXT("DAC1 High Pass Filter Mode", + da732x_dac1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("DAC1 High Pass Filter", da732x_dac1_hp_filter_enum), + SOC_ENUM("DAC1 Voice Filter", da732x_dac1_voice_filter_enum), + + SOC_ENUM_EXT("DAC2 High Pass Filter Mode", + da732x_dac2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("DAC2 High Pass Filter", da732x_dac2_hp_filter_enum), + SOC_ENUM("DAC2 Voice Filter", da732x_dac2_voice_filter_enum), + + SOC_ENUM_EXT("DAC3 High Pass Filter Mode", + da732x_dac3_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("DAC3 High Pass Filter", da732x_dac3_hp_filter_enum), + SOC_ENUM("DAC3 Filter Mode", da732x_dac3_voice_filter_enum), + + SOC_ENUM_EXT("ADC1 High Pass Filter Mode", + da732x_adc1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("ADC1 High Pass Filter", da732x_adc1_hp_filter_enum), + SOC_ENUM("ADC1 Voice Filter", da732x_adc1_voice_filter_enum), + + SOC_ENUM_EXT("ADC2 High Pass Filter Mode", + da732x_adc2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("ADC2 High Pass Filter", da732x_adc2_hp_filter_enum), + SOC_ENUM("ADC2 Voice Filter", da732x_adc2_voice_filter_enum), + + /* Equalizers */ + SOC_SINGLE("ADC1 EQ Switch", DA732X_REG_ADC1_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("ADC1 EQ Band 1 Volume", DA732X_REG_ADC1_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 2 Volume", DA732X_REG_ADC1_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 3 Volume", DA732X_REG_ADC1_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 4 Volume", DA732X_REG_ADC1_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 5 Volume", DA732X_REG_ADC1_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Overall Volume", DA732X_REG_ADC1_EQ5, + DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX, + DA732X_INVERT, eq_overall_tlv), + + SOC_SINGLE("ADC2 EQ Switch", DA732X_REG_ADC2_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("ADC2 EQ Band 1 Volume", DA732X_REG_ADC2_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC2 EQ Band 2 Volume", DA732X_REG_ADC2_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC2 EQ Band 3 Volume", DA732X_REG_ADC2_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ACD2 EQ Band 4 Volume", DA732X_REG_ADC2_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ACD2 EQ Band 5 Volume", DA732X_REG_ADC2_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC2 EQ Overall Volume", DA732X_REG_ADC1_EQ5, + DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX, + DA732X_INVERT, eq_overall_tlv), + + SOC_SINGLE("DAC1 EQ Switch", DA732X_REG_DAC1_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("DAC1 EQ Band 1 Volume", DA732X_REG_DAC1_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 2 Volume", DA732X_REG_DAC1_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 3 Volume", DA732X_REG_DAC1_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 4 Volume", DA732X_REG_DAC1_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 5 Volume", DA732X_REG_DAC1_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + + SOC_SINGLE("DAC2 EQ Switch", DA732X_REG_DAC2_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("DAC2 EQ Band 1 Volume", DA732X_REG_DAC2_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 2 Volume", DA732X_REG_DAC2_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 3 Volume", DA732X_REG_DAC2_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 4 Volume", DA732X_REG_DAC2_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 5 Volume", DA732X_REG_DAC2_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + + SOC_SINGLE("DAC3 EQ Switch", DA732X_REG_DAC3_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("DAC3 EQ Band 1 Volume", DA732X_REG_DAC3_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 2 Volume", DA732X_REG_DAC3_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 3 Volume", DA732X_REG_DAC3_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 4 Volume", DA732X_REG_DAC3_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 5 Volume", DA732X_REG_DAC3_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + + /* Lineout 2 Reciever*/ + SOC_SINGLE("Lineout 2 Switch", DA732X_REG_LIN2, DA732X_LOUT_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Lineout 2 Volume", DA732X_REG_LIN2, + DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX, + DA732X_NO_INVERT, lin2_pga_tlv), + + /* Lineout 3 SPEAKER*/ + SOC_SINGLE("Lineout 3 Switch", DA732X_REG_LIN3, DA732X_LOUT_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Lineout 3 Volume", DA732X_REG_LIN3, + DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX, + DA732X_NO_INVERT, lin3_pga_tlv), + + /* Lineout 4 */ + SOC_SINGLE("Lineout 4 Switch", DA732X_REG_LIN4, DA732X_LOUT_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Lineout 4 Volume", DA732X_REG_LIN4, + DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX, + DA732X_NO_INVERT, lin4_pga_tlv), + + /* Headphones */ + SOC_DOUBLE_R("Headphone Switch", DA732X_REG_HPR, DA732X_REG_HPL, + DA732X_HP_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_DOUBLE_R_TLV("Headphone Volume", DA732X_REG_HPL_VOL, + DA732X_REG_HPR_VOL, DA732X_HP_VOL_SHIFT, + DA732X_HP_VOL_VAL_MAX, DA732X_NO_INVERT, hp_pga_tlv), +}; + +static int da732x_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + switch (w->reg) { + case DA732X_REG_ADC1_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCA_BB_CLK_EN, + DA732X_ADCA_BB_CLK_EN); + break; + case DA732X_REG_ADC2_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCC_BB_CLK_EN, + DA732X_ADCC_BB_CLK_EN); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK, + DA732X_ADC_SET_ACT); + snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK, + DA732X_ADC_ON); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK, + DA732X_ADC_OFF); + snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK, + DA732X_ADC_SET_RST); + + switch (w->reg) { + case DA732X_REG_ADC1_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCA_BB_CLK_EN, 0); + break; + case DA732X_REG_ADC2_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCC_BB_CLK_EN, 0); + break; + default: + return -EINVAL; + } + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int da732x_out_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, w->reg, + (1 << w->shift) | DA732X_OUT_HIZ_EN, + (1 << w->shift) | DA732X_OUT_HIZ_EN); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, + (1 << w->shift) | DA732X_OUT_HIZ_EN, + (1 << w->shift) | DA732X_OUT_HIZ_DIS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char *adcl_text[] = { + "AUX1L", "MIC1" +}; + +static const char *adcr_text[] = { + "AUX1R", "MIC2", "MIC3" +}; + +static const char *enable_text[] = { + "Disabled", + "Enabled" +}; + +/* ADC1LMUX */ +static const struct soc_enum adc1l_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT, + DA732X_ADCL_MUX_MAX, adcl_text); +static const struct snd_kcontrol_new adc1l_mux = + SOC_DAPM_ENUM("ADC Route", adc1l_enum); + +/* ADC1RMUX */ +static const struct soc_enum adc1r_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT, + DA732X_ADCR_MUX_MAX, adcr_text); +static const struct snd_kcontrol_new adc1r_mux = + SOC_DAPM_ENUM("ADC Route", adc1r_enum); + +/* ADC2LMUX */ +static const struct soc_enum adc2l_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT, + DA732X_ADCL_MUX_MAX, adcl_text); +static const struct snd_kcontrol_new adc2l_mux = + SOC_DAPM_ENUM("ADC Route", adc2l_enum); + +/* ADC2RMUX */ +static const struct soc_enum adc2r_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT, + DA732X_ADCR_MUX_MAX, adcr_text); + +static const struct snd_kcontrol_new adc2r_mux = + SOC_DAPM_ENUM("ADC Route", adc2r_enum); + +static const struct soc_enum da732x_hp_left_output = + SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new hpl_mux = + SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output); + +static const struct soc_enum da732x_hp_right_output = + SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new hpr_mux = + SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output); + +static const struct soc_enum da732x_speaker_output = + SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new spk_mux = + SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output); + +static const struct soc_enum da732x_lout4_output = + SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new lout4_mux = + SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output); + +static const struct soc_enum da732x_lout2_output = + SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new lout2_mux = + SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output); + +static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = { + /* Supplies */ + SND_SOC_DAPM_SUPPLY("ADC1 Supply", DA732X_REG_ADC1_PD, 0, + DA732X_NO_INVERT, da732x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("ADC2 Supply", DA732X_REG_ADC2_PD, 0, + DA732X_NO_INVERT, da732x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DAC1 CLK", DA732X_REG_CLK_EN4, + DA732X_DACA_BB_CLK_SHIFT, DA732X_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC2 CLK", DA732X_REG_CLK_EN4, + DA732X_DACC_BB_CLK_SHIFT, DA732X_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC3 CLK", DA732X_REG_CLK_EN5, + DA732X_DACE_BB_CLK_SHIFT, DA732X_NO_INVERT, + NULL, 0), + + /* Micbias */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", DA732X_REG_MICBIAS1, + DA732X_MICBIAS_EN_SHIFT, + DA732X_NO_INVERT, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", DA732X_REG_MICBIAS2, + DA732X_MICBIAS_EN_SHIFT, + DA732X_NO_INVERT, NULL, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("ClassD"), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1L", NULL, DA732X_REG_ADC1_SEL, + DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_ADC("ADC1R", NULL, DA732X_REG_ADC1_SEL, + DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_ADC("ADC2L", NULL, DA732X_REG_ADC2_SEL, + DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_ADC("ADC2R", NULL, DA732X_REG_ADC2_SEL, + DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC1L", NULL, DA732X_REG_DAC1_SEL, + DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC1R", NULL, DA732X_REG_DAC1_SEL, + DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC2L", NULL, DA732X_REG_DAC2_SEL, + DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC2R", NULL, DA732X_REG_DAC2_SEL, + DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC3", NULL, DA732X_REG_DAC3_SEL, + DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT), + + /* Input Pgas */ + SND_SOC_DAPM_PGA("MIC1 PGA", DA732X_REG_MIC1, DA732X_MIC_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC2 PGA", DA732X_REG_MIC2, DA732X_MIC_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC3 PGA", DA732X_REG_MIC3, DA732X_MIC_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX1L PGA", DA732X_REG_AUX1L, DA732X_AUX_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX1R PGA", DA732X_REG_AUX1R, DA732X_AUX_EN_SHIFT, + 0, NULL, 0), + + SND_SOC_DAPM_PGA_E("HP Left", DA732X_REG_HPL, DA732X_HP_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("HP Right", DA732X_REG_HPR, DA732X_HP_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LIN2", DA732X_REG_LIN2, DA732X_LIN_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LIN3", DA732X_REG_LIN3, DA732X_LIN_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LIN4", DA732X_REG_LIN4, DA732X_LIN_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* MUXs */ + SND_SOC_DAPM_MUX("ADC1 Left MUX", SND_SOC_NOPM, 0, 0, &adc1l_mux), + SND_SOC_DAPM_MUX("ADC1 Right MUX", SND_SOC_NOPM, 0, 0, &adc1r_mux), + SND_SOC_DAPM_MUX("ADC2 Left MUX", SND_SOC_NOPM, 0, 0, &adc2l_mux), + SND_SOC_DAPM_MUX("ADC2 Right MUX", SND_SOC_NOPM, 0, 0, &adc2r_mux), + + SND_SOC_DAPM_MUX("HP Left MUX", SND_SOC_NOPM, 0, 0, &hpl_mux), + SND_SOC_DAPM_MUX("HP Right MUX", SND_SOC_NOPM, 0, 0, &hpr_mux), + SND_SOC_DAPM_MUX("Speaker MUX", SND_SOC_NOPM, 0, 0, &spk_mux), + SND_SOC_DAPM_MUX("LOUT2 MUX", SND_SOC_NOPM, 0, 0, &lout2_mux), + SND_SOC_DAPM_MUX("LOUT4 MUX", SND_SOC_NOPM, 0, 0, &lout4_mux), + + /* AIF interfaces */ + SND_SOC_DAPM_AIF_OUT("AIFA Output", "AIFA Capture", 0, DA732X_REG_AIFA3, + DA732X_AIF_EN_SHIFT, 0), + SND_SOC_DAPM_AIF_IN("AIFA Input", "AIFA Playback", 0, DA732X_REG_AIFA3, + DA732X_AIF_EN_SHIFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFB Output", "AIFB Capture", 0, DA732X_REG_AIFB3, + DA732X_AIF_EN_SHIFT, 0), + SND_SOC_DAPM_AIF_IN("AIFB Input", "AIFB Playback", 0, DA732X_REG_AIFB3, + DA732X_AIF_EN_SHIFT, 0), +}; + +static const struct snd_soc_dapm_route da732x_dapm_routes[] = { + /* Inputs */ + {"AUX1L PGA", "NULL", "AUX1L"}, + {"AUX1R PGA", "NULL", "AUX1R"}, + {"MIC1 PGA", NULL, "MIC1"}, + {"MIC2 PGA", "NULL", "MIC2"}, + {"MIC3 PGA", "NULL", "MIC3"}, + + /* Capture Path */ + {"ADC1 Left MUX", "MIC1", "MIC1 PGA"}, + {"ADC1 Left MUX", "AUX1L", "AUX1L PGA"}, + + {"ADC1 Right MUX", "AUX1R", "AUX1R PGA"}, + {"ADC1 Right MUX", "MIC2", "MIC2 PGA"}, + {"ADC1 Right MUX", "MIC3", "MIC3 PGA"}, + + {"ADC2 Left MUX", "AUX1L", "AUX1L PGA"}, + {"ADC2 Left MUX", "MIC1", "MIC1 PGA"}, + + {"ADC2 Right MUX", "AUX1R", "AUX1R PGA"}, + {"ADC2 Right MUX", "MIC2", "MIC2 PGA"}, + {"ADC2 Right MUX", "MIC3", "MIC3 PGA"}, + + {"ADC1L", NULL, "ADC1 Supply"}, + {"ADC1R", NULL, "ADC1 Supply"}, + {"ADC2L", NULL, "ADC2 Supply"}, + {"ADC2R", NULL, "ADC2 Supply"}, + + {"ADC1L", NULL, "ADC1 Left MUX"}, + {"ADC1R", NULL, "ADC1 Right MUX"}, + {"ADC2L", NULL, "ADC2 Left MUX"}, + {"ADC2R", NULL, "ADC2 Right MUX"}, + + {"AIFA Output", NULL, "ADC1L"}, + {"AIFA Output", NULL, "ADC1R"}, + {"AIFB Output", NULL, "ADC2L"}, + {"AIFB Output", NULL, "ADC2R"}, + + {"HP Left MUX", "Enabled", "AIFA Input"}, + {"HP Right MUX", "Enabled", "AIFA Input"}, + {"Speaker MUX", "Enabled", "AIFB Input"}, + {"LOUT2 MUX", "Enabled", "AIFB Input"}, + {"LOUT4 MUX", "Enabled", "AIFB Input"}, + + {"DAC1L", NULL, "DAC1 CLK"}, + {"DAC1R", NULL, "DAC1 CLK"}, + {"DAC2L", NULL, "DAC2 CLK"}, + {"DAC2R", NULL, "DAC2 CLK"}, + {"DAC3", NULL, "DAC3 CLK"}, + + {"DAC1L", NULL, "HP Left MUX"}, + {"DAC1R", NULL, "HP Right MUX"}, + {"DAC2L", NULL, "Speaker MUX"}, + {"DAC2R", NULL, "LOUT4 MUX"}, + {"DAC3", NULL, "LOUT2 MUX"}, + + /* Output Pgas */ + {"HP Left", NULL, "DAC1L"}, + {"HP Right", NULL, "DAC1R"}, + {"LIN3", NULL, "DAC2L"}, + {"LIN4", NULL, "DAC2R"}, + {"LIN2", NULL, "DAC3"}, + + /* Outputs */ + {"ClassD", NULL, "LIN3"}, + {"LOUTL", NULL, "LIN2"}, + {"LOUTR", NULL, "LIN4"}, + {"HPL", NULL, "HP Left"}, + {"HPR", NULL, "HP Right"}, +}; + +static int da732x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u32 aif = 0; + u32 reg_aif; + u32 fs; + + reg_aif = dai->driver->base; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + aif |= DA732X_AIF_WORD_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif |= DA732X_AIF_WORD_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif |= DA732X_AIF_WORD_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif |= DA732X_AIF_WORD_32; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + fs = DA732X_SR_8KHZ; + break; + case 11025: + fs = DA732X_SR_11_025KHZ; + break; + case 12000: + fs = DA732X_SR_12KHZ; + break; + case 16000: + fs = DA732X_SR_16KHZ; + break; + case 22050: + fs = DA732X_SR_22_05KHZ; + break; + case 24000: + fs = DA732X_SR_24KHZ; + break; + case 32000: + fs = DA732X_SR_32KHZ; + break; + case 44100: + fs = DA732X_SR_44_1KHZ; + break; + case 48000: + fs = DA732X_SR_48KHZ; + break; + case 88100: + fs = DA732X_SR_88_1KHZ; + break; + case 96000: + fs = DA732X_SR_96KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, reg_aif, DA732X_AIF_WORD_MASK, aif); + snd_soc_update_bits(codec, DA732X_REG_CLK_CTRL, DA732X_SR1_MASK, fs); + + return 0; +} + +static int da732x_set_dai_fmt(struct snd_soc_dai *dai, u32 fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u32 aif_mclk, pc_count; + u32 reg_aif1, aif1; + u32 reg_aif3, aif3; + + switch (dai->id) { + case DA732X_DAI_ID1: + reg_aif1 = DA732X_REG_AIFA1; + reg_aif3 = DA732X_REG_AIFA3; + pc_count = DA732X_PC_PULSE_AIFA | DA732X_PC_RESYNC_NOT_AUT | + DA732X_PC_SAME; + break; + case DA732X_DAI_ID2: + reg_aif1 = DA732X_REG_AIFB1; + reg_aif3 = DA732X_REG_AIFB3; + pc_count = DA732X_PC_PULSE_AIFB | DA732X_PC_RESYNC_NOT_AUT | + DA732X_PC_SAME; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif1 = DA732X_AIF_SLAVE; + aif_mclk = DA732X_AIFM_FRAME_64 | DA732X_AIFM_SRC_SEL_AIFA; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif1 = DA732X_AIF_CLK_FROM_SRC; + aif_mclk = DA732X_CLK_GENERATION_AIF_A; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif3 = DA732X_AIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif3 = DA732X_AIF_RIGHT_J_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif3 = DA732X_AIF_LEFT_J_MODE; + break; + case SND_SOC_DAIFMT_DSP_B: + aif3 = DA732X_AIF_DSP_MODE; + break; + default: + return -EINVAL; + } + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif3 |= DA732X_AIF_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif3 |= DA732X_AIF_BCLK_INV | DA732X_AIF_WCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif3 |= DA732X_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif3 |= DA732X_AIF_WCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, DA732X_REG_AIF_MCLK, aif_mclk); + snd_soc_update_bits(codec, reg_aif1, DA732X_AIF1_CLK_MASK, aif1); + snd_soc_update_bits(codec, reg_aif3, DA732X_AIF_BCLK_INV | + DA732X_AIF_WCLK_INV | DA732X_AIF_MODE_MASK, aif3); + snd_soc_write(codec, DA732X_REG_PC_CTRL, pc_count); + + return 0; +} + + + +static int da732x_set_dai_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + int fref, indiv; + u8 div_lo, div_mid, div_hi; + u64 frac_div; + + /* Disable PLL */ + if (freq_out == 0) { + snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, + DA732X_PLL_EN, 0); + da732x->pll_en = false; + return 0; + } + + if (da732x->pll_en) + return -EBUSY; + + if (source == DA732X_SRCCLK_MCLK) { + /* Validate Sysclk rate */ + switch (da732x->sysclk) { + case 11290000: + case 12288000: + case 22580000: + case 24576000: + case 45160000: + case 49152000: + snd_soc_write(codec, DA732X_REG_PLL_CTRL, + DA732X_PLL_BYPASS); + return 0; + default: + dev_err(codec->dev, + "Cannot use PLL Bypass, invalid SYSCLK rate\n"); + return -EINVAL; + } + } + + indiv = da732x_get_input_div(codec, da732x->sysclk); + if (indiv < 0) + return indiv; + + fref = (da732x->sysclk / indiv); + div_hi = freq_out / fref; + frac_div = (u64)(freq_out % fref) * 8192ULL; + do_div(frac_div, fref); + div_mid = (frac_div >> DA732X_1BYTE_SHIFT) & DA732X_U8_MASK; + div_lo = (frac_div) & DA732X_U8_MASK; + + snd_soc_write(codec, DA732X_REG_PLL_DIV_LO, div_lo); + snd_soc_write(codec, DA732X_REG_PLL_DIV_MID, div_mid); + snd_soc_write(codec, DA732X_REG_PLL_DIV_HI, div_hi); + + snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, DA732X_PLL_EN, + DA732X_PLL_EN); + + da732x->pll_en = true; + + return 0; +} + +static int da732x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + + da732x->sysclk = freq; + + return 0; +} + +#define DA732X_RATES SNDRV_PCM_RATE_8000_96000 + +#define DA732X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops da732x_dai1_ops = { + .hw_params = da732x_hw_params, + .set_fmt = da732x_set_dai_fmt, + .set_sysclk = da732x_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops da732x_dai2_ops = { + .hw_params = da732x_hw_params, + .set_fmt = da732x_set_dai_fmt, + .set_sysclk = da732x_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver da732x_dai[] = { + { + .name = "DA732X_AIFA", + .id = DA732X_DAI_ID1, + .base = DA732X_REG_AIFA1, + .playback = { + .stream_name = "AIFA Playback", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .capture = { + .stream_name = "AIFA Capture", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .ops = &da732x_dai1_ops, + }, + { + .name = "DA732X_AIFB", + .id = DA732X_DAI_ID2, + .base = DA732X_REG_AIFB1, + .playback = { + .stream_name = "AIFB Playback", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .capture = { + .stream_name = "AIFB Capture", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .ops = &da732x_dai2_ops, + }, +}; + +static const struct regmap_config da732x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DA732X_MAX_REG, + .reg_defaults = da732x_reg_cache, + .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), + .cache_type = REGCACHE_RBTREE, +}; + + +static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) +{ + u8 offset[DA732X_HP_DACS]; + u8 sign[DA732X_HP_DACS]; + u8 step = DA732X_DAC_OFFSET_STEP; + + /* Initialize DAC offset calibration circuits and registers */ + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET, + DA732X_HP_DAC_OFFSET_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET, + DA732X_HP_DAC_OFFSET_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_CALIBRATION | + DA732X_HP_DAC_OFF_SCALE_STEPS); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_CALIBRATION | + DA732X_HP_DAC_OFF_SCALE_STEPS); + + /* Wait for voltage stabilization */ + msleep(DA732X_WAIT_FOR_STABILIZATION); + + /* Check DAC offset sign */ + sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO); + sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO); + + /* Binary search DAC offset values (both channels at once) */ + offset[DA732X_HPL_DAC] = sign[DA732X_HPL_DAC] << DA732X_HP_DAC_COMPO_SHIFT; + offset[DA732X_HPR_DAC] = sign[DA732X_HPR_DAC] << DA732X_HP_DAC_COMPO_SHIFT; + + do { + offset[DA732X_HPL_DAC] |= step; + offset[DA732X_HPR_DAC] |= step; + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET, + ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET, + ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK); + + msleep(DA732X_WAIT_FOR_STABILIZATION); + + if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC]) + offset[DA732X_HPL_DAC] &= ~step; + if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC]) + offset[DA732X_HPR_DAC] &= ~step; + + step >>= 1; + } while (step); + + /* Write final DAC offsets to registers */ + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET, + ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET, + ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK); + + /* End DAC calibration mode */ + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_SCALE_STEPS); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_SCALE_STEPS); +} + +static void da732x_output_offset_adjust(struct snd_soc_codec *codec) +{ + u8 offset[DA732X_HP_AMPS]; + u8 sign[DA732X_HP_AMPS]; + u8 step = DA732X_OUTPUT_OFFSET_STEP; + + offset[DA732X_HPL_AMP] = DA732X_HP_OUT_TRIM_VAL; + offset[DA732X_HPR_AMP] = DA732X_HP_OUT_TRIM_VAL; + + /* Initialize output offset calibration circuits and registers */ + snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPL, + DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN); + snd_soc_write(codec, DA732X_REG_HPR, + DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN); + + /* Wait for voltage stabilization */ + msleep(DA732X_WAIT_FOR_STABILIZATION); + + /* Check output offset sign */ + sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) & + DA732X_HP_OUT_COMPO; + sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) & + DA732X_HP_OUT_COMPO; + + snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP | + (sign[DA732X_HPL_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) | + DA732X_HP_OUT_EN); + snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_COMP | + (sign[DA732X_HPR_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) | + DA732X_HP_OUT_EN); + + /* Binary search output offset values (both channels at once) */ + do { + offset[DA732X_HPL_AMP] |= step; + offset[DA732X_HPR_AMP] |= step; + snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, + offset[DA732X_HPL_AMP]); + snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, + offset[DA732X_HPR_AMP]); + + msleep(DA732X_WAIT_FOR_STABILIZATION); + + if ((codec->hw_read(codec, DA732X_REG_HPL) & + DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP]) + offset[DA732X_HPL_AMP] &= ~step; + if ((codec->hw_read(codec, DA732X_REG_HPR) & + DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP]) + offset[DA732X_HPR_AMP] &= ~step; + + step >>= 1; + } while (step); + + /* Write final DAC offsets to registers */ + snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, offset[DA732X_HPL_AMP]); + snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, offset[DA732X_HPR_AMP]); +} + +static void da732x_hp_dc_offset_cancellation(struct snd_soc_codec *codec) +{ + /* Make sure that we have Soft Mute enabled */ + snd_soc_write(codec, DA732X_REG_DAC1_SOFTMUTE, DA732X_SOFTMUTE_EN | + DA732X_GAIN_RAMPED | DA732X_16_SAMPLES); + snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACL_EN | + DA732X_DACR_EN | DA732X_DACL_SDM | DA732X_DACR_SDM | + DA732X_DACL_MUTE | DA732X_DACR_MUTE); + snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN | + DA732X_HP_OUT_MUTE | DA732X_HP_OUT_EN); + snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_EN | + DA732X_HP_OUT_MUTE | DA732X_HP_OUT_DAC_EN); + + da732x_dac_offset_adjust(codec); + da732x_output_offset_adjust(codec); + + snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACS_DIS); + snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_DIS); + snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_DIS); +} + +static int da732x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, + DA732X_BIAS_BOOST_MASK, + DA732X_BIAS_BOOST_100PC); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* Init Codec */ + snd_soc_write(codec, DA732X_REG_REF1, + DA732X_VMID_FASTCHG); + snd_soc_write(codec, DA732X_REG_BIAS_EN, + DA732X_BIAS_EN); + + mdelay(DA732X_STARTUP_DELAY); + + /* Disable Fast Charge and enable DAC ref voltage */ + snd_soc_write(codec, DA732X_REG_REF1, + DA732X_REFBUFX2_EN); + + /* Enable bypass DSP routing */ + snd_soc_write(codec, DA732X_REG_DATA_ROUTE, + DA732X_BYPASS_DSP); + + /* Enable Digital subsystem */ + snd_soc_write(codec, DA732X_REG_DSP_CTRL, + DA732X_DIGITAL_EN); + + snd_soc_write(codec, DA732X_REG_SPARE1_OUT, + DA732X_HP_DRIVER_EN | + DA732X_HP_GATE_LOW | + DA732X_HP_LOOP_GAIN_CTRL); + snd_soc_write(codec, DA732X_REG_HP_LIN1_GNDSEL, + DA732X_HP_OUT_GNDSEL); + + da732x_set_charge_pump(codec, DA732X_ENABLE_CP); + + snd_soc_write(codec, DA732X_REG_CLK_EN1, + DA732X_SYS3_CLK_EN | DA732X_PC_CLK_EN); + + /* Enable Zero Crossing */ + snd_soc_write(codec, DA732X_REG_INP_ZC_EN, + DA732X_MIC1_PRE_ZC_EN | + DA732X_MIC1_ZC_EN | + DA732X_MIC2_PRE_ZC_EN | + DA732X_MIC2_ZC_EN | + DA732X_AUXL_ZC_EN | + DA732X_AUXR_ZC_EN | + DA732X_MIC3_PRE_ZC_EN | + DA732X_MIC3_ZC_EN); + snd_soc_write(codec, DA732X_REG_OUT_ZC_EN, + DA732X_HPL_ZC_EN | DA732X_HPR_ZC_EN | + DA732X_LIN2_ZC_EN | DA732X_LIN3_ZC_EN | + DA732X_LIN4_ZC_EN); + + da732x_hp_dc_offset_cancellation(codec); + + regcache_cache_only(codec->control_data, false); + regcache_sync(codec->control_data); + } else { + snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, + DA732X_BIAS_BOOST_MASK, + DA732X_BIAS_BOOST_50PC); + snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, + DA732X_PLL_EN, 0); + da732x->pll_en = false; + } + break; + case SND_SOC_BIAS_OFF: + regcache_cache_only(codec->control_data, true); + da732x_set_charge_pump(codec, DA732X_DISABLE_CP); + snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN, + DA732X_BIAS_DIS); + da732x->pll_en = false; + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int da732x_probe(struct snd_soc_codec *codec) +{ + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret = 0; + + da732x->codec = codec; + + dapm->idle_bias_off = false; + + codec->control_data = da732x->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec.\n"); + goto err; + } + + da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +err: + return ret; +} + +static int da732x_remove(struct snd_soc_codec *codec) +{ + + da732x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da732x = { + .probe = da732x_probe, + .remove = da732x_remove, + .set_bias_level = da732x_set_bias_level, + .controls = da732x_snd_controls, + .num_controls = ARRAY_SIZE(da732x_snd_controls), + .dapm_widgets = da732x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da732x_dapm_widgets), + .dapm_routes = da732x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes), + .set_pll = da732x_set_dai_pll, + .reg_cache_size = ARRAY_SIZE(da732x_reg_cache), +}; + +static __devinit int da732x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da732x_priv *da732x; + unsigned int reg; + int ret; + + da732x = devm_kzalloc(&i2c->dev, sizeof(struct da732x_priv), + GFP_KERNEL); + if (!da732x) + return -ENOMEM; + + i2c_set_clientdata(i2c, da732x); + + da732x->regmap = devm_regmap_init_i2c(i2c, &da732x_regmap); + if (IS_ERR(da732x->regmap)) { + ret = PTR_ERR(da732x->regmap); + dev_err(&i2c->dev, "Failed to initialize regmap\n"); + goto err; + } + + ret = regmap_read(da732x->regmap, DA732X_REG_ID, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); + goto err; + } + + dev_info(&i2c->dev, "Revision: %d.%d\n", + (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK)); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x, + da732x_dai, ARRAY_SIZE(da732x_dai)); + if (ret != 0) + dev_err(&i2c->dev, "Failed to register codec.\n"); + +err: + return ret; +} + +static __devexit int da732x_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id da732x_i2c_id[] = { + { "da7320", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, da732x_i2c_id); + +static struct i2c_driver da732x_i2c_driver = { + .driver = { + .name = "da7320", + .owner = THIS_MODULE, + }, + .probe = da732x_i2c_probe, + .remove = __devexit_p(da732x_i2c_remove), + .id_table = da732x_i2c_id, +}; + +module_i2c_driver(da732x_i2c_driver); + + +MODULE_DESCRIPTION("ASoC DA732X driver"); +MODULE_AUTHOR("Michal Hajduk <michal.hajduk@diasemi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h new file mode 100644 index 000000000000..c8ce5475de22 --- /dev/null +++ b/sound/soc/codecs/da732x.h @@ -0,0 +1,133 @@ +/* + * da732x.h -- Dialog DA732X ALSA SoC Audio Driver Header File + * + * Copyright (C) 2012 Dialog Semiconductor GmbH + * + * Author: Michal Hajduk <Michal.Hajduk@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __DA732X_H_ +#define __DA732X_H + +#include <sound/soc.h> + +/* General */ +#define DA732X_U8_MASK 0xFF +#define DA732X_4BYTES 4 +#define DA732X_3BYTES 3 +#define DA732X_2BYTES 2 +#define DA732X_1BYTE 1 +#define DA732X_1BYTE_SHIFT 8 +#define DA732X_2BYTES_SHIFT 16 +#define DA732X_3BYTES_SHIFT 24 +#define DA732X_4BYTES_SHIFT 32 + +#define DA732X_DACS_DIS 0x0 +#define DA732X_HP_DIS 0x0 +#define DA732X_CLEAR_REG 0x0 + +/* Calibration */ +#define DA732X_DAC_OFFSET_STEP 0x20 +#define DA732X_OUTPUT_OFFSET_STEP 0x80 +#define DA732X_HP_OUT_TRIM_VAL 0x0 +#define DA732X_WAIT_FOR_STABILIZATION 1 +#define DA732X_HPL_DAC 0 +#define DA732X_HPR_DAC 1 +#define DA732X_HP_DACS 2 +#define DA732X_HPL_AMP 0 +#define DA732X_HPR_AMP 1 +#define DA732X_HP_AMPS 2 + +/* Clock settings */ +#define DA732X_STARTUP_DELAY 100 +#define DA732X_PLL_OUT_196608 196608000 +#define DA732X_PLL_OUT_180634 180633600 +#define DA732X_PLL_OUT_SRM 188620800 +#define DA732X_MCLK_10MHZ 10000000 +#define DA732X_MCLK_20MHZ 20000000 +#define DA732X_MCLK_40MHZ 40000000 +#define DA732X_MCLK_54MHZ 54000000 +#define DA732X_MCLK_RET_0_10MHZ 0 +#define DA732X_MCLK_VAL_0_10MHZ 1 +#define DA732X_MCLK_RET_10_20MHZ 1 +#define DA732X_MCLK_VAL_10_20MHZ 2 +#define DA732X_MCLK_RET_20_40MHZ 2 +#define DA732X_MCLK_VAL_20_40MHZ 4 +#define DA732X_MCLK_RET_40_54MHZ 3 +#define DA732X_MCLK_VAL_40_54MHZ 8 +#define DA732X_DAI_ID1 0 +#define DA732X_DAI_ID2 1 +#define DA732X_SRCCLK_PLL 0 +#define DA732X_SRCCLK_MCLK 1 + +#define DA732X_LIN_LP_VOL 0x4F +#define DA732X_LP_VOL 0x40 + +/* Kcontrols */ +#define DA732X_DAC_EN_MAX 2 +#define DA732X_ADCL_MUX_MAX 2 +#define DA732X_ADCR_MUX_MAX 3 +#define DA732X_HPF_MODE_MAX 3 +#define DA732X_HPF_MODE_SHIFT 4 +#define DA732X_HPF_MUSIC_SHIFT 0 +#define DA732X_HPF_MUSIC_MAX 4 +#define DA732X_HPF_VOICE_SHIFT 4 +#define DA732X_HPF_VOICE_MAX 8 +#define DA732X_EQ_EN_MAX 1 +#define DA732X_HPF_VOICE 1 +#define DA732X_HPF_MUSIC 2 +#define DA732X_HPF_DISABLED 0 +#define DA732X_NO_INVERT 0 +#define DA732X_INVERT 1 +#define DA732X_SWITCH_MAX 1 +#define DA732X_ENABLE_CP 1 +#define DA732X_DISABLE_CP 0 +#define DA732X_DISABLE_ALL_CLKS 0 +#define DA732X_RESET_ADCS 0 + +/* dB values */ +#define DA732X_MIC_VOL_DB_MIN 0 +#define DA732X_MIC_VOL_DB_INC 50 +#define DA732X_MIC_PRE_VOL_DB_MIN 0 +#define DA732X_MIC_PRE_VOL_DB_INC 600 +#define DA732X_AUX_VOL_DB_MIN -6000 +#define DA732X_AUX_VOL_DB_INC 150 +#define DA732X_HP_VOL_DB_MIN -2250 +#define DA732X_HP_VOL_DB_INC 150 +#define DA732X_LIN2_VOL_DB_MIN -1650 +#define DA732X_LIN2_VOL_DB_INC 150 +#define DA732X_LIN3_VOL_DB_MIN -1650 +#define DA732X_LIN3_VOL_DB_INC 150 +#define DA732X_LIN4_VOL_DB_MIN -2250 +#define DA732X_LIN4_VOL_DB_INC 150 +#define DA732X_EQ_BAND_VOL_DB_MIN -1050 +#define DA732X_EQ_BAND_VOL_DB_INC 150 +#define DA732X_DAC_VOL_DB_MIN -7725 +#define DA732X_DAC_VOL_DB_INC 75 +#define DA732X_ADC_VOL_DB_MIN 0 +#define DA732X_ADC_VOL_DB_INC -1 +#define DA732X_EQ_OVERALL_VOL_DB_MIN -1800 +#define DA732X_EQ_OVERALL_VOL_DB_INC 600 + +#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \ + {.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext} + +enum da732x_sysctl { + DA732X_SR_8KHZ = 0x1, + DA732X_SR_11_025KHZ = 0x2, + DA732X_SR_12KHZ = 0x3, + DA732X_SR_16KHZ = 0x5, + DA732X_SR_22_05KHZ = 0x6, + DA732X_SR_24KHZ = 0x7, + DA732X_SR_32KHZ = 0x9, + DA732X_SR_44_1KHZ = 0xA, + DA732X_SR_48KHZ = 0xB, + DA732X_SR_88_1KHZ = 0xE, + DA732X_SR_96KHZ = 0xF, +}; + +#endif /* __DA732X_H_ */ diff --git a/sound/soc/codecs/da732x_reg.h b/sound/soc/codecs/da732x_reg.h new file mode 100644 index 000000000000..bdd03ca4b2de --- /dev/null +++ b/sound/soc/codecs/da732x_reg.h @@ -0,0 +1,654 @@ +/* + * da732x_reg.h --- Dialog DA732X ALSA SoC Audio Registers Header File + * + * Copyright (C) 2012 Dialog Semiconductor GmbH + * + * Author: Michal Hajduk <Michal.Hajduk@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __DA732X_REG_H_ +#define __DA732X_REG_H_ + +/* DA732X registers */ +#define DA732X_REG_STATUS_EXT 0x00 +#define DA732X_REG_STATUS 0x01 +#define DA732X_REG_REF1 0x02 +#define DA732X_REG_BIAS_EN 0x03 +#define DA732X_REG_BIAS1 0x04 +#define DA732X_REG_BIAS2 0x05 +#define DA732X_REG_BIAS3 0x06 +#define DA732X_REG_BIAS4 0x07 +#define DA732X_REG_MICBIAS2 0x0F +#define DA732X_REG_MICBIAS1 0x10 +#define DA732X_REG_MICDET 0x11 +#define DA732X_REG_MIC1_PRE 0x12 +#define DA732X_REG_MIC1 0x13 +#define DA732X_REG_MIC2_PRE 0x14 +#define DA732X_REG_MIC2 0x15 +#define DA732X_REG_AUX1L 0x16 +#define DA732X_REG_AUX1R 0x17 +#define DA732X_REG_MIC3_PRE 0x18 +#define DA732X_REG_MIC3 0x19 +#define DA732X_REG_INP_PINBIAS 0x1A +#define DA732X_REG_INP_ZC_EN 0x1B +#define DA732X_REG_INP_MUX 0x1D +#define DA732X_REG_HP_DET 0x20 +#define DA732X_REG_HPL_DAC_OFFSET 0x21 +#define DA732X_REG_HPL_DAC_OFF_CNTL 0x22 +#define DA732X_REG_HPL_OUT_OFFSET 0x23 +#define DA732X_REG_HPL 0x24 +#define DA732X_REG_HPL_VOL 0x25 +#define DA732X_REG_HPR_DAC_OFFSET 0x26 +#define DA732X_REG_HPR_DAC_OFF_CNTL 0x27 +#define DA732X_REG_HPR_OUT_OFFSET 0x28 +#define DA732X_REG_HPR 0x29 +#define DA732X_REG_HPR_VOL 0x2A +#define DA732X_REG_LIN2 0x2B +#define DA732X_REG_LIN3 0x2C +#define DA732X_REG_LIN4 0x2D +#define DA732X_REG_OUT_ZC_EN 0x2E +#define DA732X_REG_HP_LIN1_GNDSEL 0x37 +#define DA732X_REG_CP_HP1 0x3A +#define DA732X_REG_CP_HP2 0x3B +#define DA732X_REG_CP_CTRL1 0x40 +#define DA732X_REG_CP_CTRL2 0x41 +#define DA732X_REG_CP_CTRL3 0x42 +#define DA732X_REG_CP_LEVEL_MASK 0x43 +#define DA732X_REG_CP_DET 0x44 +#define DA732X_REG_CP_STATUS 0x45 +#define DA732X_REG_CP_THRESH1 0x46 +#define DA732X_REG_CP_THRESH2 0x47 +#define DA732X_REG_CP_THRESH3 0x48 +#define DA732X_REG_CP_THRESH4 0x49 +#define DA732X_REG_CP_THRESH5 0x4A +#define DA732X_REG_CP_THRESH6 0x4B +#define DA732X_REG_CP_THRESH7 0x4C +#define DA732X_REG_CP_THRESH8 0x4D +#define DA732X_REG_PLL_DIV_LO 0x50 +#define DA732X_REG_PLL_DIV_MID 0x51 +#define DA732X_REG_PLL_DIV_HI 0x52 +#define DA732X_REG_PLL_CTRL 0x53 +#define DA732X_REG_CLK_CTRL 0x54 +#define DA732X_REG_CLK_DSP 0x5A +#define DA732X_REG_CLK_EN1 0x5B +#define DA732X_REG_CLK_EN2 0x5C +#define DA732X_REG_CLK_EN3 0x5D +#define DA732X_REG_CLK_EN4 0x5E +#define DA732X_REG_CLK_EN5 0x5F +#define DA732X_REG_AIF_MCLK 0x60 +#define DA732X_REG_AIFA1 0x61 +#define DA732X_REG_AIFA2 0x62 +#define DA732X_REG_AIFA3 0x63 +#define DA732X_REG_AIFB1 0x64 +#define DA732X_REG_AIFB2 0x65 +#define DA732X_REG_AIFB3 0x66 +#define DA732X_REG_PC_CTRL 0x6A +#define DA732X_REG_DATA_ROUTE 0x70 +#define DA732X_REG_DSP_CTRL 0x71 +#define DA732X_REG_CIF_CTRL2 0x74 +#define DA732X_REG_HANDSHAKE 0x75 +#define DA732X_REG_MBOX0 0x76 +#define DA732X_REG_MBOX1 0x77 +#define DA732X_REG_MBOX2 0x78 +#define DA732X_REG_MBOX_STATUS 0x79 +#define DA732X_REG_SPARE1_OUT 0x7D +#define DA732X_REG_SPARE2_OUT 0x7E +#define DA732X_REG_SPARE1_IN 0x7F +#define DA732X_REG_ID 0x81 +#define DA732X_REG_ADC1_PD 0x90 +#define DA732X_REG_ADC1_HPF 0x93 +#define DA732X_REG_ADC1_SEL 0x94 +#define DA732X_REG_ADC1_EQ12 0x95 +#define DA732X_REG_ADC1_EQ34 0x96 +#define DA732X_REG_ADC1_EQ5 0x97 +#define DA732X_REG_ADC2_PD 0x98 +#define DA732X_REG_ADC2_HPF 0x9B +#define DA732X_REG_ADC2_SEL 0x9C +#define DA732X_REG_ADC2_EQ12 0x9D +#define DA732X_REG_ADC2_EQ34 0x9E +#define DA732X_REG_ADC2_EQ5 0x9F +#define DA732X_REG_DAC1_HPF 0xA0 +#define DA732X_REG_DAC1_L_VOL 0xA1 +#define DA732X_REG_DAC1_R_VOL 0xA2 +#define DA732X_REG_DAC1_SEL 0xA3 +#define DA732X_REG_DAC1_SOFTMUTE 0xA4 +#define DA732X_REG_DAC1_EQ12 0xA5 +#define DA732X_REG_DAC1_EQ34 0xA6 +#define DA732X_REG_DAC1_EQ5 0xA7 +#define DA732X_REG_DAC2_HPF 0xB0 +#define DA732X_REG_DAC2_L_VOL 0xB1 +#define DA732X_REG_DAC2_R_VOL 0xB2 +#define DA732X_REG_DAC2_SEL 0xB3 +#define DA732X_REG_DAC2_SOFTMUTE 0xB4 +#define DA732X_REG_DAC2_EQ12 0xB5 +#define DA732X_REG_DAC2_EQ34 0xB6 +#define DA732X_REG_DAC2_EQ5 0xB7 +#define DA732X_REG_DAC3_HPF 0xC0 +#define DA732X_REG_DAC3_VOL 0xC1 +#define DA732X_REG_DAC3_SEL 0xC3 +#define DA732X_REG_DAC3_SOFTMUTE 0xC4 +#define DA732X_REG_DAC3_EQ12 0xC5 +#define DA732X_REG_DAC3_EQ34 0xC6 +#define DA732X_REG_DAC3_EQ5 0xC7 +#define DA732X_REG_BIQ_BYP 0xD2 +#define DA732X_REG_DMA_CMD 0xD3 +#define DA732X_REG_DMA_ADDR0 0xD4 +#define DA732X_REG_DMA_ADDR1 0xD5 +#define DA732X_REG_DMA_DATA0 0xD6 +#define DA732X_REG_DMA_DATA1 0xD7 +#define DA732X_REG_DMA_DATA2 0xD8 +#define DA732X_REG_DMA_DATA3 0xD9 +#define DA732X_REG_DMA_STATUS 0xDA +#define DA732X_REG_BROWNOUT 0xDF +#define DA732X_REG_UNLOCK 0xE0 + +#define DA732X_MAX_REG DA732X_REG_UNLOCK +/* + * Bits + */ + +/* DA732X_REG_STATUS_EXT (addr=0x00) */ +#define DA732X_STATUS_EXT_DSP (1 << 4) +#define DA732X_STATUS_EXT_CLEAR (0 << 0) + +/* DA732X_REG_STATUS (addr=0x01) */ +#define DA732X_STATUS_PLL_LOCK (1 << 0) +#define DA732X_STATUS_PLL_MCLK_DET (1 << 1) +#define DA732X_STATUS_HPDET_OUT (1 << 2) +#define DA732X_STATUS_INP_MIXDET_1 (1 << 3) +#define DA732X_STATUS_INP_MIXDET_2 (1 << 4) +#define DA732X_STATUS_BO_STATUS (1 << 5) + +/* DA732X_REG_REF1 (addr=0x02) */ +#define DA732X_VMID_FASTCHG (1 << 1) +#define DA732X_VMID_FASTDISCHG (1 << 2) +#define DA732X_REFBUFX2_EN (1 << 6) +#define DA732X_REFBUFX2_DIS (0 << 6) + +/* DA732X_REG_BIAS_EN (addr=0x03) */ +#define DA732X_BIAS_BOOST_MASK (3 << 0) +#define DA732X_BIAS_BOOST_100PC (0 << 0) +#define DA732X_BIAS_BOOST_133PC (1 << 0) +#define DA732X_BIAS_BOOST_88PC (2 << 0) +#define DA732X_BIAS_BOOST_50PC (3 << 0) +#define DA732X_BIAS_EN (1 << 7) +#define DA732X_BIAS_DIS (0 << 7) + +/* DA732X_REG_BIAS1 (addr=0x04) */ +#define DA732X_BIAS1_HP_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS1_HP_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_BIAS2 (addr=0x05) */ +#define DA732X_BIAS2_LINE2_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS2_LINE2_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_BIAS3 (addr=0x06) */ +#define DA732X_BIAS3_LINE3_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS3_LINE3_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_BIAS4 (addr=0x07) */ +#define DA732X_BIAS4_LINE4_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS4_LINE4_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_SIF_VDD_SEL (addr=0x08) */ +#define DA732X_SIF_VDD_SEL_AIFA_VDD2 (1 << 0) +#define DA732X_SIF_VDD_SEL_AIFB_VDD2 (1 << 1) +#define DA732X_SIF_VDD_SEL_CIFA_VDD2 (1 << 4) + +/* DA732X_REG_MICBIAS2/1 (addr=0x0F/0x10) */ +#define DA732X_MICBIAS_VOLTAGE_MASK (0x0F << 0) +#define DA732X_MICBIAS_VOLTAGE_2V (0x00 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V05 (0x01 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V1 (0x02 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V15 (0x03 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V2 (0x04 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V25 (0x05 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V3 (0x06 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V35 (0x07 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V4 (0x08 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V45 (0x09 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V5 (0x0A << 0) +#define DA732X_MICBIAS_EN (1 << 7) +#define DA732X_MICBIAS_EN_SHIFT 7 +#define DA732X_MICBIAS_VOLTAGE_SHIFT 0 +#define DA732X_MICBIAS_VOLTAGE_MAX 0x0B + +/* DA732X_REG_MICDET (addr=0x11) */ +#define DA732X_MICDET_INP_MICRES (1 << 0) +#define DA732X_MICDET_INP_MICHOOK (1 << 1) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_8MS (0 << 0) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_16MS (1 << 0) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_32MS (2 << 0) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_64MS (3 << 0) +#define DA732X_MICDET_INP_MICDET_EN (1 << 7) + +/* DA732X_REG_MIC1/2/3_PRE (addr=0x11/0x14/0x18) */ +#define DA732X_MICBOOST_MASK 0x7 +#define DA732X_MICBOOST_SHIFT 0 +#define DA732X_MICBOOST_MIN 0x1 +#define DA732X_MICBOOST_MAX DA732X_MICBOOST_MASK + +/* DA732X_REG_MIC1/2/3 (addr=0x13/0x15/0x19) */ +#define DA732X_MIC_VOL_SHIFT 0 +#define DA732X_MIC_VOL_VAL_MASK 0x1F +#define DA732X_MIC_MUTE_SHIFT 6 +#define DA732X_MIC_EN_SHIFT 7 +#define DA732X_MIC_VOL_VAL_MIN 0x7 +#define DA732X_MIC_VOL_VAL_MAX DA732X_MIC_VOL_VAL_MASK + +/* DA732X_REG_AUX1L/R (addr=0x16/0x17) */ +#define DA732X_AUX_VOL_SHIFT 0 +#define DA732X_AUX_VOL_MASK 0x7 +#define DA732X_AUX_MUTE_SHIFT 6 +#define DA732X_AUX_EN_SHIFT 7 +#define DA732X_AUX_VOL_VAL_MAX DA732X_AUX_VOL_MASK + +/* DA732X_REG_INP_PINBIAS (addr=0x1A) */ +#define DA732X_INP_MICL_PINBIAS_EN (1 << 0) +#define DA732X_INP_MICR_PINBIAS_EN (1 << 1) +#define DA732X_INP_AUX1L_PINBIAS_EN (1 << 2) +#define DA732X_INP_AUX1R_PINBIAS_EN (1 << 3) +#define DA732X_INP_AUX2_PINBIAS_EN (1 << 4) + +/* DA732X_REG_INP_ZC_EN (addr=0x1B) */ +#define DA732X_MIC1_PRE_ZC_EN (1 << 0) +#define DA732X_MIC1_ZC_EN (1 << 1) +#define DA732X_MIC2_PRE_ZC_EN (1 << 2) +#define DA732X_MIC2_ZC_EN (1 << 3) +#define DA732X_AUXL_ZC_EN (1 << 4) +#define DA732X_AUXR_ZC_EN (1 << 5) +#define DA732X_MIC3_PRE_ZC_EN (1 << 6) +#define DA732X_MIC3_ZC_EN (1 << 7) + +/* DA732X_REG_INP_MUX (addr=0x1D) */ +#define DA732X_INP_ADC1L_MUX_SEL_AUX1L (0 << 0) +#define DA732X_INP_ADC1L_MUX_SEL_MIC1 (1 << 0) +#define DA732X_INP_ADC1R_MUX_SEL_MASK (3 << 2) +#define DA732X_INP_ADC1R_MUX_SEL_AUX1R (0 << 2) +#define DA732X_INP_ADC1R_MUX_SEL_MIC2 (1 << 2) +#define DA732X_INP_ADC1R_MUX_SEL_MIC3 (2 << 2) +#define DA732X_INP_ADC2L_MUX_SEL_AUX1L (0 << 4) +#define DA732X_INP_ADC2L_MUX_SEL_MICL (1 << 4) +#define DA732X_INP_ADC2R_MUX_SEL_MASK (3 << 6) +#define DA732X_INP_ADC2R_MUX_SEL_AUX1R (0 << 6) +#define DA732X_INP_ADC2R_MUX_SEL_MICR (1 << 6) +#define DA732X_INP_ADC2R_MUX_SEL_AUX2 (2 << 6) +#define DA732X_ADC1L_MUX_SEL_SHIFT 0 +#define DA732X_ADC1R_MUX_SEL_SHIFT 2 +#define DA732X_ADC2L_MUX_SEL_SHIFT 4 +#define DA732X_ADC2R_MUX_SEL_SHIFT 6 + +/* DA732X_REG_HP_DET (addr=0x20) */ +#define DA732X_HP_DET_AZ (1 << 0) +#define DA732X_HP_DET_SEL1 (1 << 1) +#define DA732X_HP_DET_IS_MASK (3 << 2) +#define DA732X_HP_DET_IS_0_5UA (0 << 2) +#define DA732X_HP_DET_IS_1UA (1 << 2) +#define DA732X_HP_DET_IS_2UA (2 << 2) +#define DA732X_HP_DET_IS_4UA (3 << 2) +#define DA732X_HP_DET_RS_MASK (3 << 4) +#define DA732X_HP_DET_RS_INFINITE (0 << 4) +#define DA732X_HP_DET_RS_100KOHM (1 << 4) +#define DA732X_HP_DET_RS_10KOHM (2 << 4) +#define DA732X_HP_DET_RS_1KOHM (3 << 4) +#define DA732X_HP_DET_EN (1 << 7) + +/* DA732X_REG_HPL_DAC_OFFSET (addr=0x21/0x26) */ +#define DA732X_HP_DAC_OFFSET_TRIM_MASK (0x3F << 0) +#define DA732X_HP_DAC_OFFSET_DAC_SIGN (1 << 6) + +/* DA732X_REG_HPL_DAC_OFF_CNTL (addr=0x22/0x27) */ +#define DA732X_HP_DAC_OFF_CNTL_CONT_MASK (7 << 0) +#define DA732X_HP_DAC_OFF_CNTL_COMPO (1 << 3) +#define DA732X_HP_DAC_OFF_CALIBRATION (1 << 0) +#define DA732X_HP_DAC_OFF_SCALE_STEPS (1 << 1) +#define DA732X_HP_DAC_OFF_MASK 0x7F +#define DA732X_HP_DAC_COMPO_SHIFT 3 + +/* DA732X_REG_HPL_OUT_OFFSET (addr=0x23/0x28) */ +#define DA732X_HP_OUT_OFFSET_MASK (0xFF << 0) +#define DA732X_HP_DAC_OFFSET_TRIM_VAL 0x7F + +/* DA732X_REG_HPL/R (addr=0x24/0x29) */ +#define DA732X_HP_OUT_SIGN (1 << 0) +#define DA732X_HP_OUT_COMP (1 << 1) +#define DA732X_HP_OUT_RESERVED (1 << 2) +#define DA732X_HP_OUT_COMPO (1 << 3) +#define DA732X_HP_OUT_DAC_EN (1 << 4) +#define DA732X_HP_OUT_HIZ_EN (1 << 5) +#define DA732X_HP_OUT_HIZ_DIS (0 << 5) +#define DA732X_HP_OUT_MUTE (1 << 6) +#define DA732X_HP_OUT_EN (1 << 7) +#define DA732X_HP_OUT_COMPO_SHIFT 3 +#define DA732X_HP_OUT_DAC_EN_SHIFT 4 +#define DA732X_HP_HIZ_SHIFT 5 +#define DA732X_HP_MUTE_SHIFT 6 +#define DA732X_HP_OUT_EN_SHIFT 7 + +#define DA732X_OUT_HIZ_EN (1 << 5) +#define DA732X_OUT_HIZ_DIS (0 << 5) + +/* DA732X_REG_HPL/R_VOL (addr=0x25/0x2A) */ +#define DA732X_HP_VOL_VAL_MASK 0xF +#define DA732X_HP_VOL_SHIFT 0 +#define DA732X_HP_VOL_VAL_MAX DA732X_HP_VOL_VAL_MASK + +/* DA732X_REG_LIN2/3/4 (addr=0x2B/0x2C/0x2D) */ +#define DA732X_LOUT_VOL_SHIFT 0 +#define DA732X_LOUT_VOL_MASK 0x0F +#define DA732X_LOUT_DAC_OFF (0 << 4) +#define DA732X_LOUT_DAC_EN (1 << 4) +#define DA732X_LOUT_HIZ_N_DIS (0 << 5) +#define DA732X_LOUT_HIZ_N_EN (1 << 5) +#define DA732X_LOUT_UNMUTED (0 << 6) +#define DA732X_LOUT_MUTED (1 << 6) +#define DA732X_LOUT_EN (0 << 7) +#define DA732X_LOUT_DIS (1 << 7) +#define DA732X_LOUT_DAC_EN_SHIFT 4 +#define DA732X_LOUT_MUTE_SHIFT 6 +#define DA732X_LIN_OUT_EN_SHIFT 7 +#define DA732X_LOUT_VOL_VAL_MAX DA732X_LOUT_VOL_MASK + +/* DA732X_REG_OUT_ZC_EN (addr=0x2E) */ +#define DA732X_HPL_ZC_EN_SHIFT 0 +#define DA732X_HPR_ZC_EN_SHIFT 1 +#define DA732X_HPL_ZC_EN (1 << 0) +#define DA732X_HPL_ZC_DIS (0 << 0) +#define DA732X_HPR_ZC_EN (1 << 1) +#define DA732X_HPR_ZC_DIS (0 << 1) +#define DA732X_LIN2_ZC_EN (1 << 2) +#define DA732X_LIN2_ZC_DIS (0 << 2) +#define DA732X_LIN3_ZC_EN (1 << 3) +#define DA732X_LIN3_ZC_DIS (0 << 3) +#define DA732X_LIN4_ZC_EN (1 << 4) +#define DA732X_LIN4_ZC_DIS (0 << 4) + +/* DA732X_REG_HP_LIN1_GNDSEL (addr=0x37) */ +#define DA732X_HP_OUT_GNDSEL (1 << 0) + +/* DA732X_REG_CP_HP2 (addr=0x3a) */ +#define DA732X_HP_CP_PULSESKIP (1 << 0) +#define DA732X_HP_CP_REG (1 << 1) +#define DA732X_HP_CP_EN (1 << 3) +#define DA732X_HP_CP_DIS (0 << 3) + +/* DA732X_REG_CP_CTRL1 (addr=0x40) */ +#define DA732X_CP_MODE_MASK (7 << 1) +#define DA732X_CP_CTRL_STANDBY (0 << 1) +#define DA732X_CP_CTRL_CPVDD6 (2 << 1) +#define DA732X_CP_CTRL_CPVDD5 (3 << 1) +#define DA732X_CP_CTRL_CPVDD4 (4 << 1) +#define DA732X_CP_CTRL_CPVDD3 (5 << 1) +#define DA732X_CP_CTRL_CPVDD2 (6 << 1) +#define DA732X_CP_CTRL_CPVDD1 (7 << 1) +#define DA723X_CP_DIS (0 << 7) +#define DA732X_CP_EN (1 << 7) + +/* DA732X_REG_CP_CTRL2 (addr=0x41) */ +#define DA732X_CP_BOOST (1 << 0) +#define DA732X_CP_MANAGE_MAGNITUDE (2 << 2) + +/* DA732X_REG_CP_CTRL3 (addr=0x42) */ +#define DA732X_CP_1MHZ (0 << 0) +#define DA732X_CP_500KHZ (1 << 0) +#define DA732X_CP_250KHZ (2 << 0) +#define DA732X_CP_125KHZ (3 << 0) +#define DA732X_CP_63KHZ (4 << 0) +#define DA732X_CP_0KHZ (5 << 0) + +/* DA732X_REG_PLL_CTRL (addr=0x53) */ +#define DA732X_PLL_INDIV_MASK (3 << 0) +#define DA732X_PLL_SRM_EN (1 << 2) +#define DA732X_PLL_EN (1 << 7) +#define DA732X_PLL_BYPASS (0 << 0) + +/* DA732X_REG_CLK_CTRL (addr=0x54) */ +#define DA732X_SR1_MASK (0xF) +#define DA732X_SR2_MASK (0xF0) + +/* DA732X_REG_CLK_DSP (addr=0x5A) */ +#define DA732X_DSP_FREQ_MASK (7 << 0) +#define DA732X_DSP_FREQ_12MHZ (0 << 0) +#define DA732X_DSP_FREQ_24MHZ (1 << 0) +#define DA732X_DSP_FREQ_36MHZ (2 << 0) +#define DA732X_DSP_FREQ_48MHZ (3 << 0) +#define DA732X_DSP_FREQ_60MHZ (4 << 0) +#define DA732X_DSP_FREQ_72MHZ (5 << 0) +#define DA732X_DSP_FREQ_84MHZ (6 << 0) +#define DA732X_DSP_FREQ_96MHZ (7 << 0) + +/* DA732X_REG_CLK_EN1 (addr=0x5B) */ +#define DA732X_DSP_CLK_EN (1 << 0) +#define DA732X_SYS3_CLK_EN (1 << 1) +#define DA732X_DSP12_CLK_EN (1 << 2) +#define DA732X_PC_CLK_EN (1 << 3) +#define DA732X_MCLK_SQR_EN (1 << 7) + +/* DA732X_REG_CLK_EN2 (addr=0x5C) */ +#define DA732X_UART_CLK_EN (1 << 1) +#define DA732X_CP_CLK_EN (1 << 2) +#define DA732X_CP_CLK_DIS (0 << 2) + +/* DA732X_REG_CLK_EN3 (addr=0x5D) */ +#define DA732X_ADCA_BB_CLK_EN (1 << 0) +#define DA732X_ADCC_BB_CLK_EN (1 << 4) + +/* DA732X_REG_CLK_EN4 (addr=0x5E) */ +#define DA732X_DACA_BB_CLK_EN (1 << 0) +#define DA732X_DACC_BB_CLK_EN (1 << 4) +#define DA732X_DACA_BB_CLK_SHIFT 0 +#define DA732X_DACC_BB_CLK_SHIFT 4 + +/* DA732X_REG_CLK_EN5 (addr=0x5F) */ +#define DA732X_DACE_BB_CLK_EN (1 << 0) +#define DA732X_DACE_BB_CLK_SHIFT 0 + +/* DA732X_REG_AIF_MCLK (addr=0x60) */ +#define DA732X_AIFM_FRAME_64 (1 << 2) +#define DA732X_AIFM_SRC_SEL_AIFA (1 << 6) +#define DA732X_CLK_GENERATION_AIF_A (1 << 4) +#define DA732X_NO_CLK_GENERATION 0x0 + +/* DA732X_REG_AIFA1 (addr=0x61) */ +#define DA732X_AIF_WORD_MASK (0x3 << 0) +#define DA732X_AIF_WORD_16 (0 << 0) +#define DA732X_AIF_WORD_20 (1 << 0) +#define DA732X_AIF_WORD_24 (2 << 0) +#define DA732X_AIF_WORD_32 (3 << 0) +#define DA732X_AIF_TDM_MONO_SHIFT (1 << 6) +#define DA732X_AIF1_CLK_MASK (1 << 7) +#define DA732X_AIF_SLAVE (0 << 7) +#define DA732X_AIF_CLK_FROM_SRC (1 << 7) + +/* DA732X_REG_AIFA3 (addr=0x63) */ +#define DA732X_AIF_MODE_SHIFT 0 +#define DA732X_AIF_MODE_MASK 0x3 +#define DA732X_AIF_I2S_MODE (0 << 0) +#define DA732X_AIF_LEFT_J_MODE (1 << 0) +#define DA732X_AIF_RIGHT_J_MODE (2 << 0) +#define DA732X_AIF_DSP_MODE (3 << 0) +#define DA732X_AIF_WCLK_INV (1 << 4) +#define DA732X_AIF_BCLK_INV (1 << 5) +#define DA732X_AIF_EN (1 << 7) +#define DA732X_AIF_EN_SHIFT 7 + +/* DA732X_REG_PC_CTRL (addr=0x6a) */ +#define DA732X_PC_PULSE_AIFA (0 << 0) +#define DA732X_PC_PULSE_AIFB (1 << 0) +#define DA732X_PC_RESYNC_AUT (1 << 6) +#define DA732X_PC_RESYNC_NOT_AUT (0 << 6) +#define DA732X_PC_SAME (1 << 7) + +/* DA732X_REG_DATA_ROUTE (addr=0x70) */ +#define DA732X_ADC1_TO_AIFA (0 << 0) +#define DA732X_DSP_TO_AIFA (1 << 0) +#define DA732X_ADC2_TO_AIFB (0 << 1) +#define DA732X_DSP_TO_AIFB (1 << 1) +#define DA732X_AIFA_TO_DAC1L (0 << 2) +#define DA732X_DSP_TO_DAC1L (1 << 2) +#define DA732X_AIFA_TO_DAC1R (0 << 3) +#define DA732X_DSP_TO_DAC1R (1 << 3) +#define DA732X_AIFB_TO_DAC2L (0 << 4) +#define DA732X_DSP_TO_DAC2L (1 << 4) +#define DA732X_AIFB_TO_DAC2R (0 << 5) +#define DA732X_DSP_TO_DAC2R (1 << 5) +#define DA732X_AIFB_TO_DAC3 (0 << 6) +#define DA732X_DSP_TO_DAC3 (1 << 6) +#define DA732X_BYPASS_DSP (0 << 0) +#define DA732X_ALL_TO_DSP (0x7F << 0) + +/* DA732X_REG_DSP_CTRL (addr=0x71) */ +#define DA732X_DIGITAL_EN (1 << 0) +#define DA732X_DIGITAL_RESET (0 << 0) +#define DA732X_DSP_CORE_EN (1 << 1) +#define DA732X_DSP_CORE_RESET (0 << 1) + +/* DA732X_REG_SPARE1_OUT (addr=0x7D)*/ +#define DA732X_HP_DRIVER_EN (1 << 0) +#define DA732X_HP_GATE_LOW (1 << 2) +#define DA732X_HP_LOOP_GAIN_CTRL (1 << 3) + +/* DA732X_REG_ID (addr=0x81)*/ +#define DA732X_ID_MINOR_MASK (0xF << 0) +#define DA732X_ID_MAJOR_MASK (0xF << 4) + +/* DA732X_REG_ADC1/2_PD (addr=0x90/0x98) */ +#define DA732X_ADC_RST_MASK (0x3 << 0) +#define DA732X_ADC_PD_MASK (0x3 << 2) +#define DA732X_ADC_SET_ACT (0x3 << 0) +#define DA732X_ADC_SET_RST (0x0 << 0) +#define DA732X_ADC_ON (0x3 << 2) +#define DA732X_ADC_OFF (0x0 << 2) + +/* DA732X_REG_ADC1/2_SEL (addr=0x94/0x9C) */ +#define DA732X_ADC_VOL_VAL_MASK 0x7 +#define DA732X_ADCL_VOL_SHIFT 0 +#define DA732X_ADCR_VOL_SHIFT 4 +#define DA732X_ADCL_EN_SHIFT 2 +#define DA732X_ADCR_EN_SHIFT 3 +#define DA732X_ADCL_EN (1 << 2) +#define DA732X_ADCR_EN (1 << 3) +#define DA732X_ADC_VOL_VAL_MAX DA732X_ADC_VOL_VAL_MASK + +/* + * DA732X_REG_ADC1/2_HPF (addr=0x93/0x9b) + * DA732x_REG_DAC1/2/3_HPG (addr=0xA5/0xB5/0xC5) + */ +#define DA732X_HPF_MUSIC_EN (1 << 3) +#define DA732X_HPF_VOICE_EN ((1 << 3) | (1 << 7)) +#define DA732X_HPF_MASK ((1 << 3) | (1 << 7)) +#define DA732X_HPF_DIS ((0 << 3) | (0 << 7)) + +/* DA732X_REG_DAC1/2/3_VOL */ +#define DA732X_DAC_VOL_VAL_MASK 0x7F +#define DA732X_DAC_VOL_SHIFT 0 +#define DA732X_DAC_VOL_VAL_MAX DA732X_DAC_VOL_VAL_MASK + +/* DA732X_REG_DAC1/2/3_SEL (addr=0xA3/0xB3/0xC3) */ +#define DA732X_DACL_EN_SHIFT 3 +#define DA732X_DACR_EN_SHIFT 7 +#define DA732X_DACL_MUTE_SHIFT 2 +#define DA732X_DACR_MUTE_SHIFT 6 +#define DA732X_DACL_EN (1 << 3) +#define DA732X_DACR_EN (1 << 7) +#define DA732X_DACL_SDM (1 << 0) +#define DA732X_DACR_SDM (1 << 4) +#define DA732X_DACL_MUTE (1 << 2) +#define DA732X_DACR_MUTE (1 << 6) + +/* DA732X_REG_DAC_SOFTMUTE (addr=0xA4/0xB4/0xC4) */ +#define DA732X_SOFTMUTE_EN (1 << 7) +#define DA732X_GAIN_RAMPED (1 << 6) +#define DA732X_16_SAMPLES (4 << 0) +#define DA732X_SOFTMUTE_MASK (1 << 7) +#define DA732X_SOFTMUTE_SHIFT 7 + +/* + * DA732x_REG_ADC1/2_EQ12 (addr=0x95/0x9D) + * DA732x_REG_ADC1/2_EQ34 (addr=0x96/0x9E) + * DA732x_REG_ADC1/2_EQ5 (addr=0x97/0x9F) + * DA732x_REG_DAC1/2/3_EQ12 (addr=0xA5/0xB5/0xC5) + * DA732x_REG_DAC1/2/3_EQ34 (addr=0xA6/0xB6/0xC6) + * DA732x_REG_DAC1/2/3_EQ5 (addr=0xA7/0xB7/0xB7) + */ +#define DA732X_EQ_VOL_VAL_MASK 0xF +#define DA732X_EQ_BAND1_SHIFT 0 +#define DA732X_EQ_BAND2_SHIFT 4 +#define DA732X_EQ_BAND3_SHIFT 0 +#define DA732X_EQ_BAND4_SHIFT 4 +#define DA732X_EQ_BAND5_SHIFT 0 +#define DA732X_EQ_OVERALL_SHIFT 4 +#define DA732X_EQ_OVERALL_VOL_VAL_MASK 0x3 +#define DA732X_EQ_DIS (0 << 7) +#define DA732X_EQ_EN (1 << 7) +#define DA732X_EQ_EN_SHIFT 7 +#define DA732X_EQ_VOL_VAL_MAX DA732X_EQ_VOL_VAL_MASK +#define DA732X_EQ_OVERALL_VOL_VAL_MAX DA732X_EQ_OVERALL_VOL_VAL_MASK + +/* DA732X_REG_DMA_CMD (addr=0xD3) */ +#define DA732X_SEL_DSP_DMA_MASK (3 << 0) +#define DA732X_SEL_DSP_DMA_DIS (0 << 0) +#define DA732X_SEL_DSP_DMA_PMEM (1 << 0) +#define DA732X_SEL_DSP_DMA_XMEM (2 << 0) +#define DA732X_SEL_DSP_DMA_YMEM (3 << 0) +#define DA732X_DSP_RW_MASK (1 << 4) +#define DA732X_DSP_DMA_WRITE (0 << 4) +#define DA732X_DSP_DMA_READ (1 << 4) + +/* DA732X_REG_DMA_STATUS (addr=0xDA) */ +#define DA732X_DSP_DMA_FREE (0 << 0) +#define DA732X_DSP_DMA_BUSY (1 << 0) + +#endif /* __DA732X_REG_H_ */ diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c new file mode 100644 index 000000000000..5d8f39e32978 --- /dev/null +++ b/sound/soc/codecs/isabelle.c @@ -0,0 +1,1176 @@ +/* + * isabelle.c - Low power high fidelity audio codec driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * + * Initially based on sound/soc/codecs/twl6040.c + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <asm/div64.h> +#include "isabelle.h" + + +/* Register default values for ISABELLE driver. */ +static struct reg_default isabelle_reg_defs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x00 }, + { 15, 0x00 }, + { 16, 0x00 }, + { 17, 0x00 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x02 }, + { 22, 0x02 }, + { 23, 0x02 }, + { 24, 0x02 }, + { 25, 0x0F }, + { 26, 0x8F }, + { 27, 0x0F }, + { 28, 0x8F }, + { 29, 0x00 }, + { 30, 0x00 }, + { 31, 0x00 }, + { 32, 0x00 }, + { 33, 0x00 }, + { 34, 0x00 }, + { 35, 0x00 }, + { 36, 0x00 }, + { 37, 0x00 }, + { 38, 0x00 }, + { 39, 0x00 }, + { 40, 0x00 }, + { 41, 0x00 }, + { 42, 0x00 }, + { 43, 0x00 }, + { 44, 0x00 }, + { 45, 0x00 }, + { 46, 0x00 }, + { 47, 0x00 }, + { 48, 0x00 }, + { 49, 0x00 }, + { 50, 0x00 }, + { 51, 0x00 }, + { 52, 0x00 }, + { 53, 0x00 }, + { 54, 0x00 }, + { 55, 0x00 }, + { 56, 0x00 }, + { 57, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x90 }, + { 70, 0x90 }, + { 71, 0x90 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0x00 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 84, 0x00 }, + { 85, 0x07 }, + { 86, 0x00 }, + { 87, 0x00 }, + { 88, 0x00 }, + { 89, 0x07 }, + { 90, 0x80 }, + { 91, 0x07 }, + { 92, 0x07 }, + { 93, 0x00 }, + { 94, 0x00 }, + { 95, 0x00 }, + { 96, 0x00 }, + { 97, 0x00 }, + { 98, 0x00 }, + { 99, 0x00 }, +}; + +static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"}; +static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"}; + +static const struct soc_enum isabelle_rx1_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts), +}; + +static const struct soc_enum isabelle_rx2_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts), +}; + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new rx1_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_rx1_enum); + +static const struct snd_kcontrol_new rx2_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_rx2_enum); + +/* TX input selection */ +static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"}; +static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"}; + +static const struct soc_enum isabelle_atx_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts), +}; + +static const struct soc_enum isabelle_vtx_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts), +}; + +static const struct snd_kcontrol_new atx_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_atx_enum); + +static const struct snd_kcontrol_new vtx_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_vtx_enum); + +/* Left analog microphone selection */ +static const char *isabelle_amic1_texts[] = { + "Main Mic", "Headset Mic", "Aux/FM Left"}; + +/* Left analog microphone selection */ +static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"}; + +static const struct soc_enum isabelle_amic1_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5, + ARRAY_SIZE(isabelle_amic1_texts), + isabelle_amic1_texts), +}; + +static const struct soc_enum isabelle_amic2_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4, + ARRAY_SIZE(isabelle_amic2_texts), + isabelle_amic2_texts), +}; + +static const struct snd_kcontrol_new amic1_control = + SOC_DAPM_ENUM("Route", isabelle_amic1_enum); + +static const struct snd_kcontrol_new amic2_control = + SOC_DAPM_ENUM("Route", isabelle_amic2_enum); + +static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"}; + +static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"}; + +static const struct soc_enum isabelle_st_audio_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1, + isabelle_st_audio_texts), + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1, + isabelle_st_audio_texts), +}; + +static const struct soc_enum isabelle_st_voice_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1, + isabelle_st_voice_texts), + SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1, + isabelle_st_voice_texts), +}; + +static const struct snd_kcontrol_new st_audio_control = + SOC_DAPM_ENUM("Route", isabelle_st_audio_enum); + +static const struct snd_kcontrol_new st_voice_control = + SOC_DAPM_ENUM("Route", isabelle_st_voice_enum); + +/* Mixer controls */ +static const struct snd_kcontrol_new isabelle_hs_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC1L Playback Switch", ISABELLE_HSDRV_CFG1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hs_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC1R Playback Switch", ISABELLE_HSDRV_CFG1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hf_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_HFLPGA_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HFLPGA_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hf_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2R Playback Switch", ISABELLE_HFRPGA_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HFRPGA_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_ep_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_EARDRV_CFG1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_EARDRV_CFG1_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_aux_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC3L Playback Switch", ISABELLE_LINEAMP_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_aux_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC3R Playback Switch", ISABELLE_LINEAMP_CFG_REG, 5, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga1_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 5, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga1_right_mixer_controls[] = { +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga2_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 5, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 4, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 2, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga2_right_mixer_controls[] = { +SOC_DAPM_SINGLE("USNC Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga3_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 5, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga3_right_mixer_controls[] = { +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx1_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DL1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx2_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 5, 1, 0), +SOC_DAPM_SINGLE("DL2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx3_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 3, 1, 0), +SOC_DAPM_SINGLE("DL3 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 2, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx4_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DL4 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 0, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx5_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DL5 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx6_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("DL6 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new ep_path_enable_control = + SOC_DAPM_SINGLE("Switch", ISABELLE_EARDRV_CFG2_REG, 0, 1, 0); + +/* TLV Declarations */ +static const DECLARE_TLV_DB_SCALE(mic_amp_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(afm_amp_tlv, -3300, 300, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -1200, 200, 0); +static const DECLARE_TLV_DB_SCALE(hf_tlv, -5000, 200, 0); + +/* from -63 to 0 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dpga_tlv, -6300, 100, 1); + +/* from -63 to 9 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(rx_tlv, -6300, 100, 1); + +static const DECLARE_TLV_DB_SCALE(st_tlv, -2700, 300, 1); +static const DECLARE_TLV_DB_SCALE(tx_tlv, -600, 100, 0); + +static const struct snd_kcontrol_new isabelle_snd_controls[] = { + SOC_DOUBLE_TLV("Headset Playback Volume", ISABELLE_HSDRV_GAIN_REG, + 4, 0, 0xF, 0, dac_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + ISABELLE_HFLPGA_CFG_REG, ISABELLE_HFRPGA_CFG_REG, + 0, 0x1F, 0, hf_tlv), + SOC_DOUBLE_TLV("Aux Playback Volume", ISABELLE_LINEAMP_GAIN_REG, + 4, 0, 0xF, 0, dac_tlv), + SOC_SINGLE_TLV("Earpiece Playback Volume", ISABELLE_EARDRV_CFG1_REG, + 0, 0xF, 0, dac_tlv), + + SOC_DOUBLE_TLV("Aux FM Volume", ISABELLE_APGA_GAIN_REG, 4, 0, 0xF, 0, + afm_amp_tlv), + SOC_SINGLE_TLV("Mic1 Capture Volume", ISABELLE_MIC1_GAIN_REG, 3, 0x1F, + 0, mic_amp_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", ISABELLE_MIC2_GAIN_REG, 3, 0x1F, + 0, mic_amp_tlv), + + SOC_DOUBLE_R_TLV("DPGA1 Volume", ISABELLE_DPGA1L_GAIN_REG, + ISABELLE_DPGA1R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + SOC_DOUBLE_R_TLV("DPGA2 Volume", ISABELLE_DPGA2L_GAIN_REG, + ISABELLE_DPGA2R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + SOC_DOUBLE_R_TLV("DPGA3 Volume", ISABELLE_DPGA3L_GAIN_REG, + ISABELLE_DPGA3R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + + SOC_SINGLE_TLV("Sidetone Audio TX1 Volume", + ISABELLE_ATX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Audio TX2 Volume", + ISABELLE_ATX_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Voice TX1 Volume", + ISABELLE_VTX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Voice TX2 Volume", + ISABELLE_VTX2_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv), + + SOC_SINGLE_TLV("Audio TX1 Volume", ISABELLE_ATX1_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Audio TX2 Volume", ISABELLE_ATX2_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Voice TX1 Volume", ISABELLE_VTX1_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Voice TX2 Volume", ISABELLE_VTX2_DPGA_REG, 4, 0xF, 0, + tx_tlv), + + SOC_SINGLE_TLV("RX1 DPGA Volume", ISABELLE_RX1_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX2 DPGA Volume", ISABELLE_RX2_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX3 DPGA Volume", ISABELLE_RX3_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX4 DPGA Volume", ISABELLE_RX4_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX5 DPGA Volume", ISABELLE_RX5_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX6 DPGA Volume", ISABELLE_RX6_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + + SOC_SINGLE("Headset Noise Gate", ISABELLE_HS_NG_CFG1_REG, 7, 1, 0), + SOC_SINGLE("Handsfree Noise Gate", ISABELLE_HF_NG_CFG1_REG, 7, 1, 0), + + SOC_SINGLE("ATX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("ATX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 6, 1, 0), + SOC_SINGLE("ARX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 5, 1, 0), + SOC_SINGLE("ARX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 4, 1, 0), + SOC_SINGLE("ARX3 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 3, 1, 0), + SOC_SINGLE("ARX4 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 2, 1, 0), + SOC_SINGLE("ARX5 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 1, 1, 0), + SOC_SINGLE("ARX6 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 0, 1, 0), + SOC_SINGLE("VRX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 3, 1, 0), + SOC_SINGLE("VRX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 2, 1, 0), + + SOC_SINGLE("ATX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 7, 1, 0), + SOC_SINGLE("ATX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 6, 1, 0), + SOC_SINGLE("VTX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 5, 1, 0), + SOC_SINGLE("VTX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 4, 1, 0), + SOC_SINGLE("RX1 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 5, 1, 0), + SOC_SINGLE("RX2 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 4, 1, 0), + SOC_SINGLE("RX3 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 3, 1, 0), + SOC_SINGLE("RX4 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 2, 1, 0), + SOC_SINGLE("RX5 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 1, 1, 0), + SOC_SINGLE("RX6 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 0, 1, 0), + + SOC_SINGLE("ULATX12 Capture Switch", ISABELLE_ULATX12_INTF_CFG_REG, + 7, 1, 0), + + SOC_SINGLE("DL12 Playback Switch", ISABELLE_DL12_INTF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("DL34 Playback Switch", ISABELLE_DL34_INTF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("DL56 Playback Switch", ISABELLE_DL56_INTF_CFG_REG, + 7, 1, 0), + + /* DMIC Switch */ + SOC_SINGLE("DMIC Switch", ISABELLE_DMIC_CFG_REG, 0, 1, 0), +}; + +static const struct snd_soc_dapm_widget isabelle_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("LINEIN1"), + SND_SOC_DAPM_INPUT("LINEIN2"), + SND_SOC_DAPM_INPUT("DMICDAT"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + SND_SOC_DAPM_OUTPUT("EP"), + SND_SOC_DAPM_OUTPUT("LINEOUT1"), + SND_SOC_DAPM_OUTPUT("LINEOUT2"), + + SND_SOC_DAPM_PGA("DL1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL6", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amic1_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amic2_control), + + SND_SOC_DAPM_MUX("Sidetone Audio Playback", SND_SOC_NOPM, 0, 0, + &st_audio_control), + SND_SOC_DAPM_MUX("Sidetone Voice Playback", SND_SOC_NOPM, 0, 0, + &st_voice_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("INTF1_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 7, 0), + SND_SOC_DAPM_AIF_IN("INTF2_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("INTF1_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 5, 0), + SND_SOC_DAPM_AIF_OUT("INTF2_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 4, 0), + + SND_SOC_DAPM_OUT_DRV("ULATX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULATX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULVTX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULVTX2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog Capture PGAs */ + SND_SOC_DAPM_PGA("MicAmp1", ISABELLE_AMIC_CFG_REG, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmp2", ISABELLE_AMIC_CFG_REG, 4, 0, NULL, 0), + + /* Auxiliary FM PGAs */ + SND_SOC_DAPM_PGA("APGA1", ISABELLE_APGA_CFG_REG, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("APGA2", ISABELLE_APGA_CFG_REG, 6, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1", "Left Front Capture", + ISABELLE_AMIC_CFG_REG, 7, 0), + SND_SOC_DAPM_ADC("ADC2", "Right Front Capture", + ISABELLE_AMIC_CFG_REG, 6, 0), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("Headset Mic Bias", ISABELLE_ABIAS_CFG_REG, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Main Mic Bias", ISABELLE_ABIAS_CFG_REG, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias", + ISABELLE_DBIAS_CFG_REG, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias", + ISABELLE_DBIAS_CFG_REG, 2, 0, NULL, 0), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Headset Left Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hs_left_mixer_controls, + ARRAY_SIZE(isabelle_hs_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Headset Right Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hs_right_mixer_controls, + ARRAY_SIZE(isabelle_hs_right_mixer_controls)), + SND_SOC_DAPM_MIXER("Handsfree Left Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hf_left_mixer_controls, + ARRAY_SIZE(isabelle_hf_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Handsfree Right Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hf_right_mixer_controls, + ARRAY_SIZE(isabelle_hf_right_mixer_controls)), + SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_aux_left_mixer_controls, + ARRAY_SIZE(isabelle_aux_left_mixer_controls)), + SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_aux_right_mixer_controls, + ARRAY_SIZE(isabelle_aux_right_mixer_controls)), + SND_SOC_DAPM_MIXER("Earphone Mixer", SND_SOC_NOPM, 0, 0, + isabelle_ep_mixer_controls, + ARRAY_SIZE(isabelle_ep_mixer_controls)), + + SND_SOC_DAPM_MIXER("DPGA1L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga1_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga1_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA1R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga1_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga1_right_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA2L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga2_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga2_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA2R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga2_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga2_right_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA3L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga3_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga3_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA3R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga3_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga3_right_mixer_controls)), + + SND_SOC_DAPM_MIXER("RX1 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx1_mixer_controls, + ARRAY_SIZE(isabelle_rx1_mixer_controls)), + SND_SOC_DAPM_MIXER("RX2 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx2_mixer_controls, + ARRAY_SIZE(isabelle_rx2_mixer_controls)), + SND_SOC_DAPM_MIXER("RX3 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx3_mixer_controls, + ARRAY_SIZE(isabelle_rx3_mixer_controls)), + SND_SOC_DAPM_MIXER("RX4 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx4_mixer_controls, + ARRAY_SIZE(isabelle_rx4_mixer_controls)), + SND_SOC_DAPM_MIXER("RX5 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx5_mixer_controls, + ARRAY_SIZE(isabelle_rx5_mixer_controls)), + SND_SOC_DAPM_MIXER("RX6 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx6_mixer_controls, + ARRAY_SIZE(isabelle_rx6_mixer_controls)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC1L", "Headset Playback", ISABELLE_DAC_CFG_REG, + 5, 0), + SND_SOC_DAPM_DAC("DAC1R", "Headset Playback", ISABELLE_DAC_CFG_REG, + 4, 0), + SND_SOC_DAPM_DAC("DAC2L", "Handsfree Playback", ISABELLE_DAC_CFG_REG, + 3, 0), + SND_SOC_DAPM_DAC("DAC2R", "Handsfree Playback", ISABELLE_DAC_CFG_REG, + 2, 0), + SND_SOC_DAPM_DAC("DAC3L", "Lineout Playback", ISABELLE_DAC_CFG_REG, + 1, 0), + SND_SOC_DAPM_DAC("DAC3R", "Lineout Playback", ISABELLE_DAC_CFG_REG, + 0, 0), + + /* Analog Playback PGAs */ + SND_SOC_DAPM_PGA("Sidetone Audio PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sidetone Voice PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HF Left PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HF Right PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA1L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA1R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA2L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA2R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA3L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA3R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog Playback Mux */ + SND_SOC_DAPM_MUX("RX1 Playback", ISABELLE_ALU_RX_EN_REG, 5, 0, + &rx1_mux_controls), + SND_SOC_DAPM_MUX("RX2 Playback", ISABELLE_ALU_RX_EN_REG, 4, 0, + &rx2_mux_controls), + + /* TX Select */ + SND_SOC_DAPM_MUX("ATX Select", ISABELLE_TX_INPUT_CFG_REG, + 7, 0, &atx_mux_controls), + SND_SOC_DAPM_MUX("VTX Select", ISABELLE_TX_INPUT_CFG_REG, + 6, 0, &vtx_mux_controls), + + SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0, + &ep_path_enable_control), + + /* Output Drivers */ + SND_SOC_DAPM_OUT_DRV("HS Left Driver", ISABELLE_HSDRV_CFG2_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Right Driver", ISABELLE_HSDRV_CFG2_REG, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("LINEOUT1 Left Driver", ISABELLE_LINEAMP_CFG_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("LINEOUT2 Right Driver", ISABELLE_LINEAMP_CFG_REG, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earphone Driver", ISABELLE_EARDRV_CFG2_REG, + 1, 0, NULL, 0), + + SND_SOC_DAPM_OUT_DRV("HF Left Driver", ISABELLE_HFDRV_CFG_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HF Right Driver", ISABELLE_HFDRV_CFG_REG, + 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route isabelle_intercon[] = { + /* Interface mapping */ + { "DL1", "DL12 Playback Switch", "INTF1_SDI" }, + { "DL2", "DL12 Playback Switch", "INTF1_SDI" }, + { "DL3", "DL34 Playback Switch", "INTF1_SDI" }, + { "DL4", "DL34 Playback Switch", "INTF1_SDI" }, + { "DL5", "DL56 Playback Switch", "INTF1_SDI" }, + { "DL6", "DL56 Playback Switch", "INTF1_SDI" }, + + { "DL1", "DL12 Playback Switch", "INTF2_SDI" }, + { "DL2", "DL12 Playback Switch", "INTF2_SDI" }, + { "DL3", "DL34 Playback Switch", "INTF2_SDI" }, + { "DL4", "DL34 Playback Switch", "INTF2_SDI" }, + { "DL5", "DL56 Playback Switch", "INTF2_SDI" }, + { "DL6", "DL56 Playback Switch", "INTF2_SDI" }, + + /* Input side mapping */ + { "Sidetone Audio PGA", NULL, "Sidetone Audio Playback" }, + { "Sidetone Voice PGA", NULL, "Sidetone Voice Playback" }, + + { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Audio PGA" }, + + { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX1 Mixer", "DL1 Playback Switch", "DL1" }, + + { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Audio PGA" }, + + { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX2 Mixer", "DL2 Playback Switch", "DL2" }, + + { "RX3 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX3 Mixer", "DL3 Playback Switch", "DL3" }, + + { "RX4 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX4 Mixer", "DL4 Playback Switch", "DL4" }, + + { "RX5 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX5 Mixer", "DL5 Playback Switch", "DL5" }, + + { "RX6 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX6 Mixer", "DL6 Playback Switch", "DL6" }, + + /* Capture path */ + { "Analog Left Capture Route", "Headset Mic", "HSMIC" }, + { "Analog Left Capture Route", "Main Mic", "MAINMIC" }, + { "Analog Left Capture Route", "Aux/FM Left", "LINEIN1" }, + + { "Analog Right Capture Route", "Sub Mic", "SUBMIC" }, + { "Analog Right Capture Route", "Aux/FM Right", "LINEIN2" }, + + { "MicAmp1", NULL, "Analog Left Capture Route" }, + { "MicAmp2", NULL, "Analog Right Capture Route" }, + + { "ADC1", NULL, "MicAmp1" }, + { "ADC2", NULL, "MicAmp2" }, + + { "ATX Select", "AMIC1", "ADC1" }, + { "ATX Select", "DMIC", "DMICDAT" }, + { "ATX Select", "AMIC2", "ADC2" }, + + { "VTX Select", "AMIC1", "ADC1" }, + { "VTX Select", "DMIC", "DMICDAT" }, + { "VTX Select", "AMIC2", "ADC2" }, + + { "ULATX1", "ATX1 Filter Enable Switch", "ATX Select" }, + { "ULATX1", "ATX1 Filter Bypass Switch", "ATX Select" }, + { "ULATX2", "ATX2 Filter Enable Switch", "ATX Select" }, + { "ULATX2", "ATX2 Filter Bypass Switch", "ATX Select" }, + + { "ULVTX1", "VTX1 Filter Enable Switch", "VTX Select" }, + { "ULVTX1", "VTX1 Filter Bypass Switch", "VTX Select" }, + { "ULVTX2", "VTX2 Filter Enable Switch", "VTX Select" }, + { "ULVTX2", "VTX2 Filter Bypass Switch", "VTX Select" }, + + { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX1" }, + { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX2" }, + { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX1" }, + { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX2" }, + + { "INTF1_SDO", NULL, "ULVTX1" }, + { "INTF1_SDO", NULL, "ULVTX2" }, + { "INTF2_SDO", NULL, "ULVTX1" }, + { "INTF2_SDO", NULL, "ULVTX2" }, + + /* AFM Path */ + { "APGA1", NULL, "LINEIN1" }, + { "APGA2", NULL, "LINEIN2" }, + + { "RX1 Playback", "VRX1 Filter Bypass Switch", "RX1 Mixer" }, + { "RX1 Playback", "ARX1 Filter Bypass Switch", "RX1 Mixer" }, + { "RX1 Playback", "RX1 Filter Enable Switch", "RX1 Mixer" }, + + { "RX2 Playback", "VRX2 Filter Bypass Switch", "RX2 Mixer" }, + { "RX2 Playback", "ARX2 Filter Bypass Switch", "RX2 Mixer" }, + { "RX2 Playback", "RX2 Filter Enable Switch", "RX2 Mixer" }, + + { "RX3 Playback", "ARX3 Filter Bypass Switch", "RX3 Mixer" }, + { "RX3 Playback", "RX3 Filter Enable Switch", "RX3 Mixer" }, + + { "RX4 Playback", "ARX4 Filter Bypass Switch", "RX4 Mixer" }, + { "RX4 Playback", "RX4 Filter Enable Switch", "RX4 Mixer" }, + + { "RX5 Playback", "ARX5 Filter Bypass Switch", "RX5 Mixer" }, + { "RX5 Playback", "RX5 Filter Enable Switch", "RX5 Mixer" }, + + { "RX6 Playback", "ARX6 Filter Bypass Switch", "RX6 Mixer" }, + { "RX6 Playback", "RX6 Filter Enable Switch", "RX6 Mixer" }, + + { "DPGA1L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA1L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA1L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + + { "DPGA1R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA1R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA1R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA1L", NULL, "DPGA1L Mixer" }, + { "DPGA1R", NULL, "DPGA1R Mixer" }, + + { "DAC1L", NULL, "DPGA1L" }, + { "DAC1R", NULL, "DPGA1R" }, + + { "DPGA2L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA2L Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA2L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA2L Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA2L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + { "DPGA2L Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA2R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA2R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA2R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA2L", NULL, "DPGA2L Mixer" }, + { "DPGA2R", NULL, "DPGA2R Mixer" }, + + { "DAC2L", NULL, "DPGA2L" }, + { "DAC2R", NULL, "DPGA2R" }, + + { "DPGA3L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA3L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA3L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + + { "DPGA3R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA3R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA3R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA3L", NULL, "DPGA3L Mixer" }, + { "DPGA3R", NULL, "DPGA3R Mixer" }, + + { "DAC3L", NULL, "DPGA3L" }, + { "DAC3R", NULL, "DPGA3R" }, + + { "Headset Left Mixer", "DAC1L Playback Switch", "DAC1L" }, + { "Headset Left Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Headset Right Mixer", "DAC1R Playback Switch", "DAC1R" }, + { "Headset Right Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "HS Left Driver", NULL, "Headset Left Mixer" }, + { "HS Right Driver", NULL, "Headset Right Mixer" }, + + { "HSOL", NULL, "HS Left Driver" }, + { "HSOR", NULL, "HS Right Driver" }, + + /* Earphone playback path */ + { "Earphone Mixer", "DAC2L Playback Switch", "DAC2L" }, + { "Earphone Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Earphone Playback", "Switch", "Earphone Mixer" }, + { "Earphone Driver", NULL, "Earphone Playback" }, + { "EP", NULL, "Earphone Driver" }, + + { "Handsfree Left Mixer", "DAC2L Playback Switch", "DAC2L" }, + { "Handsfree Left Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Handsfree Right Mixer", "DAC2R Playback Switch", "DAC2R" }, + { "Handsfree Right Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "HF Left PGA", NULL, "Handsfree Left Mixer" }, + { "HF Right PGA", NULL, "Handsfree Right Mixer" }, + + { "HF Left Driver", NULL, "HF Left PGA" }, + { "HF Right Driver", NULL, "HF Right PGA" }, + + { "HFL", NULL, "HF Left Driver" }, + { "HFR", NULL, "HF Right Driver" }, + + { "LINEOUT1 Mixer", "DAC3L Playback Switch", "DAC3L" }, + { "LINEOUT1 Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "LINEOUT2 Mixer", "DAC3R Playback Switch", "DAC3R" }, + { "LINEOUT2 Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "LINEOUT1 Driver", NULL, "LINEOUT1 Mixer" }, + { "LINEOUT2 Driver", NULL, "LINEOUT2 Mixer" }, + + { "LINEOUT1", NULL, "LINEOUT1 Driver" }, + { "LINEOUT2", NULL, "LINEOUT2 Driver" }, +}; + +static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC1_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC2_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_line_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC3_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG, + ISABELLE_CHIP_EN, BIT(0)); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG, + ISABELLE_CHIP_EN, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int isabelle_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + u16 aif = 0; + unsigned int fs_val = 0; + + switch (params_rate(params)) { + case 8000: + fs_val = ISABELLE_FS_RATE_8; + break; + case 11025: + fs_val = ISABELLE_FS_RATE_11; + break; + case 12000: + fs_val = ISABELLE_FS_RATE_12; + break; + case 16000: + fs_val = ISABELLE_FS_RATE_16; + break; + case 22050: + fs_val = ISABELLE_FS_RATE_22; + break; + case 24000: + fs_val = ISABELLE_FS_RATE_24; + break; + case 32000: + fs_val = ISABELLE_FS_RATE_32; + break; + case 44100: + fs_val = ISABELLE_FS_RATE_44; + break; + case 48000: + fs_val = ISABELLE_FS_RATE_48; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_FS_RATE_CFG_REG, + ISABELLE_FS_RATE_MASK, fs_val); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S20_3LE: + aif |= ISABELLE_AIF_LENGTH_20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif |= ISABELLE_AIF_LENGTH_32; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG, + ISABELLE_AIF_LENGTH_MASK, aif); + + return 0; +} + +static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int aif_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val &= ~ISABELLE_AIF_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val |= ISABELLE_AIF_MS; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_val |= ISABELLE_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_val |= ISABELLE_LEFT_J_MODE; + break; + case SND_SOC_DAIFMT_PDM: + aif_val |= ISABELLE_PDM_MODE; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG, + (ISABELLE_AIF_MS | ISABELLE_AIF_FMT_MASK), aif_val); + + return 0; +} + +/* Rates supported by Isabelle driver */ +#define ISABELLE_RATES SNDRV_PCM_RATE_8000_48000 + +/* Formates supported by Isabelle driver. */ +#define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops isabelle_hs_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_hs_mute, +}; + +static struct snd_soc_dai_ops isabelle_hf_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_hf_mute, +}; + +static struct snd_soc_dai_ops isabelle_line_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_line_mute, +}; + +static struct snd_soc_dai_ops isabelle_ul_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, +}; + +/* ISABELLE dai structure */ +static struct snd_soc_dai_driver isabelle_dai[] = { + { + .name = "isabelle-dl1", + .playback = { + .stream_name = "Headset Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_hs_dai_ops, + }, + { + .name = "isabelle-dl2", + .playback = { + .stream_name = "Handsfree Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_hf_dai_ops, + }, + { + .name = "isabelle-lineout", + .playback = { + .stream_name = "Lineout Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_line_dai_ops, + }, + { + .name = "isabelle-ul", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_ul_dai_ops, + }, +}; + +static int isabelle_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + + codec->control_data = dev_get_regmap(codec->dev, NULL); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_isabelle = { + .probe = isabelle_probe, + .set_bias_level = isabelle_set_bias_level, + .controls = isabelle_snd_controls, + .num_controls = ARRAY_SIZE(isabelle_snd_controls), + .dapm_widgets = isabelle_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(isabelle_dapm_widgets), + .dapm_routes = isabelle_intercon, + .num_dapm_routes = ARRAY_SIZE(isabelle_intercon), + .idle_bias_off = true, +}; + +static const struct regmap_config isabelle_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = ISABELLE_MAX_REGISTER, + .reg_defaults = isabelle_reg_defs, + .num_reg_defaults = ARRAY_SIZE(isabelle_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *isabelle_regmap; + int ret = 0; + + isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config); + if (IS_ERR(isabelle_regmap)) { + ret = PTR_ERR(isabelle_regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + i2c_set_clientdata(i2c, isabelle_regmap); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_isabelle, isabelle_dai, + ARRAY_SIZE(isabelle_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + return ret; +} + +static int __devexit isabelle_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id isabelle_i2c_id[] = { + { "isabelle", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id); + +static struct i2c_driver isabelle_i2c_driver = { + .driver = { + .name = "isabelle", + .owner = THIS_MODULE, + }, + .probe = isabelle_i2c_probe, + .remove = __devexit_p(isabelle_i2c_remove), + .id_table = isabelle_i2c_id, +}; + +module_i2c_driver(isabelle_i2c_driver); + +MODULE_DESCRIPTION("ASoC ISABELLE driver"); +MODULE_AUTHOR("Vishwas A Deshpande <vishwas.a.deshpande@ti.com>"); +MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/isabelle.h b/sound/soc/codecs/isabelle.h new file mode 100644 index 000000000000..96d839a8c956 --- /dev/null +++ b/sound/soc/codecs/isabelle.h @@ -0,0 +1,143 @@ +/* + * isabelle.h - Low power high fidelity audio codec driver header file + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + */ + +#ifndef _ISABELLE_H +#define _ISABELLE_H + +#include <linux/bitops.h> + +/* ISABELLE REGISTERS */ + +#define ISABELLE_PWR_CFG_REG 0x01 +#define ISABELLE_PWR_EN_REG 0x02 +#define ISABELLE_PS_EN1_REG 0x03 +#define ISABELLE_INT1_STATUS_REG 0x04 +#define ISABELLE_INT1_MASK_REG 0x05 +#define ISABELLE_INT2_STATUS_REG 0x06 +#define ISABELLE_INT2_MASK_REG 0x07 +#define ISABELLE_HKCTL1_REG 0x08 +#define ISABELLE_HKCTL2_REG 0x09 +#define ISABELLE_HKCTL3_REG 0x0A +#define ISABELLE_ACCDET_STATUS_REG 0x0B +#define ISABELLE_BUTTON_ID_REG 0x0C +#define ISABELLE_PLL_CFG_REG 0x10 +#define ISABELLE_PLL_EN_REG 0x11 +#define ISABELLE_FS_RATE_CFG_REG 0x12 +#define ISABELLE_INTF_CFG_REG 0x13 +#define ISABELLE_INTF_EN_REG 0x14 +#define ISABELLE_ULATX12_INTF_CFG_REG 0x15 +#define ISABELLE_DL12_INTF_CFG_REG 0x16 +#define ISABELLE_DL34_INTF_CFG_REG 0x17 +#define ISABELLE_DL56_INTF_CFG_REG 0x18 +#define ISABELLE_ATX_STPGA1_CFG_REG 0x19 +#define ISABELLE_ATX_STPGA2_CFG_REG 0x1A +#define ISABELLE_VTX_STPGA1_CFG_REG 0x1B +#define ISABELLE_VTX2_STPGA2_CFG_REG 0x1C +#define ISABELLE_ATX1_DPGA_REG 0x1D +#define ISABELLE_ATX2_DPGA_REG 0x1E +#define ISABELLE_VTX1_DPGA_REG 0x1F +#define ISABELLE_VTX2_DPGA_REG 0x20 +#define ISABELLE_TX_INPUT_CFG_REG 0x21 +#define ISABELLE_RX_INPUT_CFG_REG 0x22 +#define ISABELLE_RX_INPUT_CFG2_REG 0x23 +#define ISABELLE_VOICE_HPF_CFG_REG 0x24 +#define ISABELLE_AUDIO_HPF_CFG_REG 0x25 +#define ISABELLE_RX1_DPGA_REG 0x26 +#define ISABELLE_RX2_DPGA_REG 0x27 +#define ISABELLE_RX3_DPGA_REG 0x28 +#define ISABELLE_RX4_DPGA_REG 0x29 +#define ISABELLE_RX5_DPGA_REG 0x2A +#define ISABELLE_RX6_DPGA_REG 0x2B +#define ISABELLE_ALU_TX_EN_REG 0x2C +#define ISABELLE_ALU_RX_EN_REG 0x2D +#define ISABELLE_IIR_RESYNC_REG 0x2E +#define ISABELLE_ABIAS_CFG_REG 0x30 +#define ISABELLE_DBIAS_CFG_REG 0x31 +#define ISABELLE_MIC1_GAIN_REG 0x32 +#define ISABELLE_MIC2_GAIN_REG 0x33 +#define ISABELLE_AMIC_CFG_REG 0x34 +#define ISABELLE_DMIC_CFG_REG 0x35 +#define ISABELLE_APGA_GAIN_REG 0x36 +#define ISABELLE_APGA_CFG_REG 0x37 +#define ISABELLE_TX_GAIN_DLY_REG 0x38 +#define ISABELLE_RX_GAIN_DLY_REG 0x39 +#define ISABELLE_RX_PWR_CTRL_REG 0x3A +#define ISABELLE_DPGA1LR_IN_SEL_REG 0x3B +#define ISABELLE_DPGA1L_GAIN_REG 0x3C +#define ISABELLE_DPGA1R_GAIN_REG 0x3D +#define ISABELLE_DPGA2L_IN_SEL_REG 0x3E +#define ISABELLE_DPGA2R_IN_SEL_REG 0x3F +#define ISABELLE_DPGA2L_GAIN_REG 0x40 +#define ISABELLE_DPGA2R_GAIN_REG 0x41 +#define ISABELLE_DPGA3LR_IN_SEL_REG 0x42 +#define ISABELLE_DPGA3L_GAIN_REG 0x43 +#define ISABELLE_DPGA3R_GAIN_REG 0x44 +#define ISABELLE_DAC1_SOFTRAMP_REG 0x45 +#define ISABELLE_DAC2_SOFTRAMP_REG 0x46 +#define ISABELLE_DAC3_SOFTRAMP_REG 0x47 +#define ISABELLE_DAC_CFG_REG 0x48 +#define ISABELLE_EARDRV_CFG1_REG 0x49 +#define ISABELLE_EARDRV_CFG2_REG 0x4A +#define ISABELLE_HSDRV_GAIN_REG 0x4B +#define ISABELLE_HSDRV_CFG1_REG 0x4C +#define ISABELLE_HSDRV_CFG2_REG 0x4D +#define ISABELLE_HS_NG_CFG1_REG 0x4E +#define ISABELLE_HS_NG_CFG2_REG 0x4F +#define ISABELLE_LINEAMP_GAIN_REG 0x50 +#define ISABELLE_LINEAMP_CFG_REG 0x51 +#define ISABELLE_HFL_VOL_CTRL_REG 0x52 +#define ISABELLE_HFL_SFTVOL_CTRL_REG 0x53 +#define ISABELLE_HFL_LIM_CTRL_1_REG 0x54 +#define ISABELLE_HFL_LIM_CTRL_2_REG 0x55 +#define ISABELLE_HFR_VOL_CTRL_REG 0x56 +#define ISABELLE_HFR_SFTVOL_CTRL_REG 0x57 +#define ISABELLE_HFR_LIM_CTRL_1_REG 0x58 +#define ISABELLE_HFR_LIM_CTRL_2_REG 0x59 +#define ISABELLE_HF_MODE_REG 0x5A +#define ISABELLE_HFLPGA_CFG_REG 0x5B +#define ISABELLE_HFRPGA_CFG_REG 0x5C +#define ISABELLE_HFDRV_CFG_REG 0x5D +#define ISABELLE_PDMOUT_CFG1_REG 0x5E +#define ISABELLE_PDMOUT_CFG2_REG 0x5F +#define ISABELLE_PDMOUT_L_WM_REG 0x60 +#define ISABELLE_PDMOUT_R_WM_REG 0x61 +#define ISABELLE_HF_NG_CFG1_REG 0x62 +#define ISABELLE_HF_NG_CFG2_REG 0x63 + +/* ISABELLE_PWR_EN_REG (0x02h) */ +#define ISABELLE_CHIP_EN BIT(0) + +/* ISABELLE DAI FORMATS */ +#define ISABELLE_AIF_FMT_MASK 0x70 +#define ISABELLE_I2S_MODE 0x0 +#define ISABELLE_LEFT_J_MODE 0x1 +#define ISABELLE_PDM_MODE 0x2 + +#define ISABELLE_AIF_LENGTH_MASK 0x30 +#define ISABELLE_AIF_LENGTH_20 0x00 +#define ISABELLE_AIF_LENGTH_32 0x10 + +#define ISABELLE_AIF_MS 0x80 + +#define ISABELLE_FS_RATE_MASK 0xF +#define ISABELLE_FS_RATE_8 0x0 +#define ISABELLE_FS_RATE_11 0x1 +#define ISABELLE_FS_RATE_12 0x2 +#define ISABELLE_FS_RATE_16 0x4 +#define ISABELLE_FS_RATE_22 0x5 +#define ISABELLE_FS_RATE_24 0x6 +#define ISABELLE_FS_RATE_32 0x8 +#define ISABELLE_FS_RATE_44 0x9 +#define ISABELLE_FS_RATE_48 0xA + +#define ISABELLE_MAX_REGISTER 0xFF + +#endif diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 802b9f176b16..99b0a9dcff34 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -12,7 +12,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> @@ -1358,7 +1357,7 @@ static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { }; /* LM49453 dai structure. */ -static const struct snd_soc_dai_driver lm49453_dai[] = { +static struct snd_soc_dai_driver lm49453_dai[] = { { .name = "LM49453 Headset", .playback = { diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 35179e2c23c9..7cd508e16a5c 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2216,7 +2216,7 @@ static irqreturn_t max98095_report_jack(int irq, void *data) return IRQ_HANDLED; } -int max98095_jack_detect_enable(struct snd_soc_codec *codec) +static int max98095_jack_detect_enable(struct snd_soc_codec *codec) { struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); int ret = 0; @@ -2245,7 +2245,7 @@ int max98095_jack_detect_enable(struct snd_soc_codec *codec) return ret; } -int max98095_jack_detect_disable(struct snd_soc_codec *codec) +static int max98095_jack_detect_disable(struct snd_soc_codec *codec) { int ret = 0; @@ -2286,6 +2286,7 @@ int max98095_jack_detect(struct snd_soc_codec *codec, max98095_report_jack(client->irq, codec); return 0; } +EXPORT_SYMBOL_GPL(max98095_jack_detect); #ifdef CONFIG_PM static int max98095_suspend(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 22cb5bf59273..96aa5fa05160 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -638,7 +638,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, priv); - priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap); + priv->regmap = devm_regmap_init_i2c(i2c, &ml26124_i2c_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); @@ -651,10 +651,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, static __devexit int ml26124_i2c_remove(struct i2c_client *client) { - struct ml26124_priv *priv = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(priv->regmap); return 0; } diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c new file mode 100644 index 000000000000..dd8d856053fc --- /dev/null +++ b/sound/soc/codecs/spdif_receiver.c @@ -0,0 +1,67 @@ +/* + * ALSA SoC SPDIF DIR (Digital Interface Reciever) driver + * + * Based on ALSA SoC SPDIF DIT driver + * + * This driver is used by controllers which can operate in DIR (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. SPEAr SPDIF IN Audio controller uses this driver. + * + * Author: Vipin Kumar, <vipin.kumar@st.com> + * Copyright: (C) 2012 ST Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/initval.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) + +static struct snd_soc_codec_driver soc_codec_spdif_dir; + +static struct snd_soc_dai_driver dir_stub_dai = { + .name = "dir-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int spdif_dir_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_spdif_dir, + &dir_stub_dai, 1); +} + +static int spdif_dir_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver spdif_dir_driver = { + .probe = spdif_dir_probe, + .remove = spdif_dir_remove, + .driver = { + .name = "spdif-dir", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(spdif_dir_driver); + +MODULE_DESCRIPTION("ASoC SPDIF DIR driver"); +MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c new file mode 100644 index 000000000000..0c225cd569d2 --- /dev/null +++ b/sound/soc/codecs/sta529.c @@ -0,0 +1,442 @@ +/* + * ASoC codec driver for spear platform + * + * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver + * + * Copyright (C) 2012 ST Microelectronics + * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +/* STA529 Register offsets */ +#define STA529_FFXCFG0 0x00 +#define STA529_FFXCFG1 0x01 +#define STA529_MVOL 0x02 +#define STA529_LVOL 0x03 +#define STA529_RVOL 0x04 +#define STA529_TTF0 0x05 +#define STA529_TTF1 0x06 +#define STA529_TTP0 0x07 +#define STA529_TTP1 0x08 +#define STA529_S2PCFG0 0x0A +#define STA529_S2PCFG1 0x0B +#define STA529_P2SCFG0 0x0C +#define STA529_P2SCFG1 0x0D +#define STA529_PLLCFG0 0x14 +#define STA529_PLLCFG1 0x15 +#define STA529_PLLCFG2 0x16 +#define STA529_PLLCFG3 0x17 +#define STA529_PLLPFE 0x18 +#define STA529_PLLST 0x19 +#define STA529_ADCCFG 0x1E /*mic_select*/ +#define STA529_CKOCFG 0x1F +#define STA529_MISC 0x20 +#define STA529_PADST0 0x21 +#define STA529_PADST1 0x22 +#define STA529_FFXST 0x23 +#define STA529_PWMIN1 0x2D +#define STA529_PWMIN2 0x2E +#define STA529_POWST 0x32 + +#define STA529_MAX_REGISTER 0x32 + +#define STA529_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define STA529_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) +#define S2PC_VALUE 0x98 +#define CLOCK_OUT 0x60 +#define LEFT_J_DATA_FORMAT 0x10 +#define I2S_DATA_FORMAT 0x12 +#define RIGHT_J_DATA_FORMAT 0x14 +#define CODEC_MUTE_VAL 0x80 + +#define POWER_CNTLMSAK 0x40 +#define POWER_STDBY 0x40 +#define FFX_MASK 0x80 +#define FFX_OFF 0x80 +#define POWER_UP 0x00 +#define FFX_CLK_ENB 0x01 +#define FFX_CLK_DIS 0x00 +#define FFX_CLK_MSK 0x01 +#define PLAY_FREQ_RANGE_MSK 0x70 +#define CAP_FREQ_RANGE_MSK 0x0C +#define PDATA_LEN_MSK 0xC0 +#define BCLK_TO_FS_MSK 0x30 +#define AUDIO_MUTE_MSK 0x80 + +static const struct reg_default sta529_reg_defaults[] = { + { 0, 0x35 }, /* R0 - FFX Configuration reg 0 */ + { 1, 0xc8 }, /* R1 - FFX Configuration reg 1 */ + { 2, 0x50 }, /* R2 - Master Volume */ + { 3, 0x00 }, /* R3 - Left Volume */ + { 4, 0x00 }, /* R4 - Right Volume */ + { 10, 0xb2 }, /* R10 - S2P Config Reg 0 */ + { 11, 0x41 }, /* R11 - S2P Config Reg 1 */ + { 12, 0x92 }, /* R12 - P2S Config Reg 0 */ + { 13, 0x41 }, /* R13 - P2S Config Reg 1 */ + { 30, 0xd2 }, /* R30 - ADC Config Reg */ + { 31, 0x40 }, /* R31 - clock Out Reg */ + { 32, 0x21 }, /* R32 - Misc Register */ +}; + +struct sta529 { + struct regmap *regmap; +}; + +static bool sta529_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + + case STA529_FFXCFG0: + case STA529_FFXCFG1: + case STA529_MVOL: + case STA529_LVOL: + case STA529_RVOL: + case STA529_S2PCFG0: + case STA529_S2PCFG1: + case STA529_P2SCFG0: + case STA529_P2SCFG1: + case STA529_ADCCFG: + case STA529_CKOCFG: + case STA529_MISC: + return true; + default: + return false; + } +} + + +static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary", + "Phase-shift"}; + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0); +static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0); +static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text); + +static const struct snd_kcontrol_new sta529_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0, + 127, 0, out_gain_tlv), + SOC_SINGLE_TLV("Master Playback Volume", STA529_MVOL, 0, 127, 1, + master_vol_tlv), + SOC_ENUM("PWM Select", pwm_src), +}; + +static int sta529_set_bias_level(struct snd_soc_codec *codec, enum + snd_soc_bias_level level) +{ + struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, STA529_FFXCFG0, POWER_CNTLMSAK, + POWER_UP); + snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK, + FFX_CLK_ENB); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(sta529->regmap); + snd_soc_update_bits(codec, STA529_FFXCFG0, + POWER_CNTLMSAK, POWER_STDBY); + /* Making FFX output to zero */ + snd_soc_update_bits(codec, STA529_FFXCFG0, FFX_MASK, + FFX_OFF); + snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK, + FFX_CLK_DIS); + break; + case SND_SOC_BIAS_OFF: + break; + } + + /* + * store the label for powers down audio subsystem for suspend.This is + * used by soc core layer + */ + codec->dapm.bias_level = level; + + return 0; + +} + +static int sta529_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + int pdata, play_freq_val, record_freq_val; + int bclk_to_fs_ratio; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + pdata = 1; + bclk_to_fs_ratio = 0; + break; + case SNDRV_PCM_FORMAT_S24_LE: + pdata = 2; + bclk_to_fs_ratio = 1; + break; + case SNDRV_PCM_FORMAT_S32_LE: + pdata = 3; + bclk_to_fs_ratio = 2; + break; + default: + dev_err(codec->dev, "Unsupported format\n"); + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + case 11025: + play_freq_val = 0; + record_freq_val = 2; + break; + case 16000: + case 22050: + play_freq_val = 1; + record_freq_val = 0; + break; + + case 32000: + case 44100: + case 48000: + play_freq_val = 2; + record_freq_val = 0; + break; + default: + dev_err(codec->dev, "Unsupported rate\n"); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_soc_update_bits(codec, STA529_S2PCFG1, PDATA_LEN_MSK, + pdata << 6); + snd_soc_update_bits(codec, STA529_S2PCFG1, BCLK_TO_FS_MSK, + bclk_to_fs_ratio << 4); + snd_soc_update_bits(codec, STA529_MISC, PLAY_FREQ_RANGE_MSK, + play_freq_val << 4); + } else { + snd_soc_update_bits(codec, STA529_P2SCFG1, PDATA_LEN_MSK, + pdata << 6); + snd_soc_update_bits(codec, STA529_P2SCFG1, BCLK_TO_FS_MSK, + bclk_to_fs_ratio << 4); + snd_soc_update_bits(codec, STA529_MISC, CAP_FREQ_RANGE_MSK, + record_freq_val << 2); + } + + return 0; +} + +static int sta529_mute(struct snd_soc_dai *dai, int mute) +{ + u8 val = 0; + + if (mute) + val |= CODEC_MUTE_VAL; + + snd_soc_update_bits(dai->codec, STA529_FFXCFG0, AUDIO_MUTE_MSK, val); + + return 0; +} + +static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + mode = LEFT_J_DATA_FORMAT; + break; + case SND_SOC_DAIFMT_I2S: + mode = I2S_DATA_FORMAT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + mode = RIGHT_J_DATA_FORMAT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + + return 0; +} + +static const struct snd_soc_dai_ops sta529_dai_ops = { + .hw_params = sta529_hw_params, + .set_fmt = sta529_set_dai_fmt, + .digital_mute = sta529_mute, +}; + +static struct snd_soc_dai_driver sta529_dai = { + .name = "sta529-audio", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA529_RATES, + .formats = STA529_FORMAT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = STA529_RATES, + .formats = STA529_FORMAT, + }, + .ops = &sta529_dai_ops, +}; + +static int sta529_probe(struct snd_soc_codec *codec) +{ + struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = sta529->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static int sta529_remove(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sta529_suspend(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sta529_resume(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +struct snd_soc_codec_driver sta529_codec_driver = { + .probe = sta529_probe, + .remove = sta529_remove, + .set_bias_level = sta529_set_bias_level, + .suspend = sta529_suspend, + .resume = sta529_resume, + .controls = sta529_snd_controls, + .num_controls = ARRAY_SIZE(sta529_snd_controls), +}; + +static const struct regmap_config sta529_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = STA529_MAX_REGISTER, + .readable_reg = sta529_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = sta529_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(sta529_reg_defaults), +}; + +static __devinit int sta529_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta529 *sta529; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL); + if (sta529 == NULL) { + dev_err(&i2c->dev, "Can not allocate memory\n"); + return -ENOMEM; + } + + sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap); + if (IS_ERR(sta529->regmap)) { + ret = PTR_ERR(sta529->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, sta529); + + ret = snd_soc_register_codec(&i2c->dev, + &sta529_codec_driver, &sta529_dai, 1); + if (ret != 0) + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + + return ret; +} + +static int __devexit sta529_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id sta529_i2c_id[] = { + { "sta529", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta529_i2c_id); + +static struct i2c_driver sta529_i2c_driver = { + .driver = { + .name = "sta529", + .owner = THIS_MODULE, + }, + .probe = sta529_i2c_probe, + .remove = __devexit_p(sta529_i2c_remove), + .id_table = sta529_i2c_id, +}; + +module_i2c_driver(sta529_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA529 codec driver"); +MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e9b62b5ea637..dc78f5a4bcbf 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -118,7 +118,9 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x00, 0x00, /* 88 */ 0x00, 0x00, 0x00, 0x00, /* 92 */ 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x02, 0x00, /* 100 */ + 0x00, 0x00, 0x00, 0x00, /* 104 */ + 0x00, 0x00, /* 108 */ }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ @@ -229,6 +231,25 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const struct soc_enum aic3x_agc_level_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), +}; + +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const struct soc_enum aic3x_agc_attack_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), +}; + +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; +static const struct soc_enum aic3x_agc_decay_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -353,6 +374,15 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]), + SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]), + SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]), + SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]), + SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]), + SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]), + + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, @@ -368,7 +398,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0); static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl = - SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); + SOC_DOUBLE_TLV("Class-D Playback Volume", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = @@ -970,6 +1000,12 @@ static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + /* set clock on MCLK or GPIO2 or BCLK */ + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK, + clk_id << PLLCLK_IN_SHIFT); + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK, + clk_id << CLKDIV_IN_SHIFT); + aic3x->sysclk = freq; return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 08c7f6685ff0..6db3c41b0163 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0 @@ -179,6 +196,14 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_MASK 0x30 +#define PLLCLK_IN_SHIFT 4 +#define CLKDIV_IN_MASK 0xc0 +#define CLKDIV_IN_SHIFT 6 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 /* Software reset register bits */ #define SOFT_RESET 0x80 diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index a36e9fcdf184..0ff1e70b7770 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -553,7 +553,7 @@ static const struct snd_kcontrol_new vibrar_mux_controls = /* Headset power mode */ static const char *twl6040_power_mode_texts[] = { - "Low-Power", "High-Perfomance", + "Low-Power", "High-Performance", }; static const struct soc_enum twl6040_power_mode_enum = diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index e0b51e9f8b12..951d7b49476a 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -121,20 +121,23 @@ static const struct snd_soc_dai_ops wm1250_ev1_ops = { .hw_params = wm1250_ev1_hw_params, }; +#define WM1250_EV1_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000) + static struct snd_soc_dai_driver wm1250_ev1_dai = { .name = "wm1250-ev1", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &wm1250_ev1_ops, diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 0418fa11e6bd..3fd5b29dc933 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -1,7 +1,7 @@ /* * wm2000.c -- WM2000 ALSA Soc Audio driver * - * Copyright 2008-2010 Wolfson Microelectronics PLC. + * Copyright 2008-2011 Wolfson Microelectronics PLC. * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -674,9 +674,39 @@ static int wm2000_resume(struct snd_soc_codec *codec) #define wm2000_resume NULL #endif +static bool wm2000_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM2000_REG_SYS_START: + case WM2000_REG_SPEECH_CLARITY: + case WM2000_REG_SYS_WATCHDOG: + case WM2000_REG_ANA_VMID_PD_TIME: + case WM2000_REG_ANA_VMID_PU_TIME: + case WM2000_REG_CAT_FLTR_INDX: + case WM2000_REG_CAT_GAIN_0: + case WM2000_REG_SYS_STATUS: + case WM2000_REG_SYS_MODE_CNTRL: + case WM2000_REG_SYS_START0: + case WM2000_REG_SYS_START1: + case WM2000_REG_ID1: + case WM2000_REG_ID2: + case WM2000_REG_REVISON: + case WM2000_REG_SYS_CTL1: + case WM2000_REG_SYS_CTL2: + case WM2000_REG_ANC_STAT: + case WM2000_REG_IF_CTL: + return true; + default: + return false; + } +} + static const struct regmap_config wm2000_regmap = { .reg_bits = 8, .val_bits = 8, + + .max_register = WM2000_REG_IF_CTL, + .readable_reg = wm2000_readable_reg, }; static int wm2000_probe(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index e167207a19cc..e239f4bf2460 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -1,7 +1,7 @@ /* * wm5100-tables.c -- WM5100 ALSA SoC Audio driver data * - * Copyright 2011 Wolfson Microelectronics plc + * Copyright 2011-2 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index cb6d5372103a..f4817292ef45 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1,7 +1,7 @@ /* * wm5100.c -- WM5100 ALSA SoC Audio driver * - * Copyright 2011 Wolfson Microelectronics plc + * Copyright 2011-2 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -2378,13 +2378,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) return 0; } -static int wm5100_soc_volatile(struct snd_soc_codec *codec, - unsigned int reg) -{ - return true; -} - - static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .probe = wm5100_probe, .remove = wm5100_remove, @@ -2392,8 +2385,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .set_sysclk = wm5100_set_sysclk, .set_pll = wm5100_set_fll, .idle_bias_off = 1, - .reg_cache_size = WM5100_MAX_REGISTER, - .volatile_register = wm5100_soc_volatile, .seq_notifier = wm5100_seq_notifier, .controls = wm5100_snd_controls, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c new file mode 100644 index 000000000000..6537f16d383e --- /dev/null +++ b/sound/soc/codecs/wm5102.c @@ -0,0 +1,903 @@ +/* + * wm5102.c -- WM5102 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" +#include "wm5102.h" + +struct wm5102_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new wm5102_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1R_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2R_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3R_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), +SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, + ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 1, 0), +SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUT2_OSR_SHIFT, 1, 0), +SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 1, 0), +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUTPUT_PATH_CONFIG_1R, + ARIZONA_OUT1L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUTPUT_PATH_CONFIG_2R, + ARIZONA_OUT2L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), +ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), +ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"), +ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"), +ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"), +ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"), +ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2L"), +SND_SOC_DAPM_OUTPUT("HPOUT2R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTLN"), +SND_SOC_DAPM_OUTPUT("SPKOUTLP"), +SND_SOC_DAPM_OUTPUT("SPKOUTRN"), +SND_SOC_DAPM_OUTPUT("SPKOUTRP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "IN3L", "IN3L PGA" }, \ + { name, "IN3R", "IN3R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "DRC2L", "DRC2L" }, \ + { name, "DRC2R", "DRC2R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ASRC1L", "ASRC1L" }, \ + { name, "ASRC1R", "ASRC1R" }, \ + { name, "ASRC2L", "ASRC2L" }, \ + { name, "ASRC2R", "ASRC2R" } + +static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "AIF3 Capture", NULL, "DBVDD3" }, + { "AIF3 Playback", NULL, "DBVDD3" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT2L", NULL, "CPVDD" }, + { "OUT2R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDDL" }, + { "OUT4R", NULL, "SPKVDDR" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT4R", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), + ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"), + ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), + ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"), + + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "HPOUT2L", NULL, "OUT2L" }, + { "HPOUT2R", NULL, "OUT2R" }, + + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "SPKOUTLN", NULL, "OUT4L" }, + { "SPKOUTLP", NULL, "OUT4L" }, + + { "SPKOUTRN", NULL, "OUT4R" }, + { "SPKOUTRP", NULL, "OUT4R" }, + + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, +}; + +static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM5102_FLL1: + return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout); + case WM5102_FLL2: + return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout); + default: + return -EINVAL; + } +} + +#define WM5102_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm5102_dai[] = { + { + .name = "wm5102-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5102-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5102-aif3", + .id = 3, + .base = ARIZONA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int wm5102_codec_probe(struct snd_soc_codec *codec) +{ + struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + + codec->control_data = priv->core.arizona->regmap; + return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); +} + +#define WM5102_DIG_VU 0x0200 + +static unsigned int wm5102_digital_vu[] = { + ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, + + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_3R, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm5102 = { + .probe = wm5102_codec_probe, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm5102_set_fll, + + .controls = wm5102_snd_controls, + .num_controls = ARRAY_SIZE(wm5102_snd_controls), + .dapm_widgets = wm5102_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm5102_dapm_widgets), + .dapm_routes = wm5102_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm5102_dapm_routes), +}; + +static int __devinit wm5102_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm5102_priv *wm5102; + int i; + + wm5102 = devm_kzalloc(&pdev->dev, sizeof(struct wm5102_priv), + GFP_KERNEL); + if (wm5102 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm5102); + + wm5102->core.arizona = arizona; + + for (i = 0; i < ARRAY_SIZE(wm5102->fll); i++) + wm5102->fll[i].vco_mult = 1; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm5102->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm5102->fll[1]); + + for (i = 0; i < ARRAY_SIZE(wm5102_dai); i++) + arizona_init_dai(&wm5102->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm5102_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], + WM5102_DIG_VU, WM5102_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5102, + wm5102_dai, ARRAY_SIZE(wm5102_dai)); +} + +static int __devexit wm5102_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm5102_codec_driver = { + .driver = { + .name = "wm5102-codec", + .owner = THIS_MODULE, + }, + .probe = wm5102_probe, + .remove = __devexit_p(wm5102_remove), +}; + +module_platform_driver(wm5102_codec_driver); + +MODULE_DESCRIPTION("ASoC WM5102 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm5102-codec"); diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h new file mode 100644 index 000000000000..d30477f3070c --- /dev/null +++ b/sound/soc/codecs/wm5102.h @@ -0,0 +1,21 @@ +/* + * wm5102.h -- WM5102 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM5102_H +#define _WM5102_H + +#include "arizona.h" + +#define WM5102_FLL1 1 +#define WM5102_FLL2 2 + +#endif diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c new file mode 100644 index 000000000000..8033f7065189 --- /dev/null +++ b/sound/soc/codecs/wm5110.c @@ -0,0 +1,950 @@ +/* + * wm5110.c -- WM5110 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" +#include "wm5110.h" + +struct wm5110_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new wm5110_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, + ARIZONA_IN4_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1R_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2R_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3R_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), +SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, + ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 1, 0), +SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUT2_OSR_SHIFT, 1, 0), +SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 1, 0), +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, + ARIZONA_OUT6_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L, + ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L, + ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUTPUT_PATH_CONFIG_1R, + ARIZONA_OUT1L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUTPUT_PATH_CONFIG_2R, + ARIZONA_OUT2L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), +SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, + ARIZONA_SPK2R_MUTE_SHIFT, 1, 1), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT2L, ARIZONA_OUT6LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT2R, ARIZONA_OUT6RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("IN4L"), +SND_SOC_DAPM_INPUT("IN4R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT6L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT6L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT6R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT6R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), +ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), +ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"), +ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), +ARIZONA_MIXER_WIDGETS(SPKDAT2L, "SPKDAT2L"), +ARIZONA_MIXER_WIDGETS(SPKDAT2R, "SPKDAT2R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"), +ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"), +ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"), +ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2L"), +SND_SOC_DAPM_OUTPUT("HPOUT2R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTLN"), +SND_SOC_DAPM_OUTPUT("SPKOUTLP"), +SND_SOC_DAPM_OUTPUT("SPKOUTRN"), +SND_SOC_DAPM_OUTPUT("SPKOUTRP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), +SND_SOC_DAPM_OUTPUT("SPKDAT2L"), +SND_SOC_DAPM_OUTPUT("SPKDAT2R"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "IN3L", "IN3L PGA" }, \ + { name, "IN3R", "IN3R PGA" }, \ + { name, "IN4L", "IN4L PGA" }, \ + { name, "IN4R", "IN4R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "DRC2L", "DRC2L" }, \ + { name, "DRC2R", "DRC2R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ASRC1L", "ASRC1L" }, \ + { name, "ASRC1R", "ASRC1R" }, \ + { name, "ASRC2L", "ASRC2L" }, \ + { name, "ASRC2R", "ASRC2R" } + +static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "AIF3 Capture", NULL, "DBVDD3" }, + { "AIF3 Playback", NULL, "DBVDD3" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT2L", NULL, "CPVDD" }, + { "OUT2R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDDL" }, + { "OUT4R", NULL, "SPKVDDR" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT4R", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + { "OUT6L", NULL, "SYSCLK" }, + { "OUT6R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), + ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"), + ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + ARIZONA_MIXER_ROUTES("OUT6L", "SPKDAT2L"), + ARIZONA_MIXER_ROUTES("OUT6R", "SPKDAT2R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), + ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"), + + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "HPOUT2L", NULL, "OUT2L" }, + { "HPOUT2R", NULL, "OUT2R" }, + + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "SPKOUTLN", NULL, "OUT4L" }, + { "SPKOUTLP", NULL, "OUT4L" }, + + { "SPKOUTRN", NULL, "OUT4R" }, + { "SPKOUTRP", NULL, "OUT4R" }, + + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "SPKDAT2L", NULL, "OUT6L" }, + { "SPKDAT2R", NULL, "OUT6R" }, +}; + +static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM5110_FLL1: + return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout); + case WM5110_FLL2: + return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout); + default: + return -EINVAL; + } +} + +#define WM5110_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm5110_dai[] = { + { + .name = "wm5110-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5110-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5110-aif3", + .id = 3, + .base = ARIZONA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int wm5110_codec_probe(struct snd_soc_codec *codec) +{ + struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + + codec->control_data = priv->core.arizona->regmap; + return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); +} + +#define WM5110_DIG_VU 0x0200 + +static unsigned int wm5110_digital_vu[] = { + ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, + + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_3R, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm5110 = { + .probe = wm5110_codec_probe, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm5110_set_fll, + + .controls = wm5110_snd_controls, + .num_controls = ARRAY_SIZE(wm5110_snd_controls), + .dapm_widgets = wm5110_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm5110_dapm_widgets), + .dapm_routes = wm5110_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm5110_dapm_routes), +}; + +static int __devinit wm5110_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm5110_priv *wm5110; + int i; + + wm5110 = devm_kzalloc(&pdev->dev, sizeof(struct wm5110_priv), + GFP_KERNEL); + if (wm5110 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm5110); + + wm5110->core.arizona = arizona; + + for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++) + wm5110->fll[i].vco_mult = 3; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm5110->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm5110->fll[1]); + + for (i = 0; i < ARRAY_SIZE(wm5110_dai); i++) + arizona_init_dai(&wm5110->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm5110_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm5110_digital_vu[i], + WM5110_DIG_VU, WM5110_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110, + wm5110_dai, ARRAY_SIZE(wm5110_dai)); +} + +static int __devexit wm5110_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm5110_codec_driver = { + .driver = { + .name = "wm5110-codec", + .owner = THIS_MODULE, + }, + .probe = wm5110_probe, + .remove = __devexit_p(wm5110_remove), +}; + +module_platform_driver(wm5110_codec_driver); + +MODULE_DESCRIPTION("ASoC WM5110 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm5110-codec"); diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h new file mode 100644 index 000000000000..75e9351ccab0 --- /dev/null +++ b/sound/soc/codecs/wm5110.h @@ -0,0 +1,21 @@ +/* + * wm5110.h -- WM5110 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM5110_H +#define _WM5110_H + +#include "arizona.h" + +#define WM5110_FLL1 1 +#define WM5110_FLL2 2 + +#endif diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 555ee146ae0d..d26c8ae4e6d9 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1,7 +1,7 @@ /* * wm8350.c -- WM8350 ALSA SoC audio driver * - * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. + * Copyright (C) 2007-12 Wolfson Microelectronics PLC. * * Author: Liam Girdwood <lrg@slimlogic.co.uk> * @@ -71,20 +71,6 @@ struct wm8350_data { int fll_freq_in; }; -static unsigned int wm8350_codec_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct wm8350 *wm8350 = codec->control_data; - return wm8350_reg_read(wm8350, reg); -} - -static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct wm8350 *wm8350 = codec->control_data; - return wm8350_reg_write(wm8350, reg, value); -} - /* * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown. */ @@ -1519,7 +1505,9 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - codec->control_data = wm8350; + codec->control_data = wm8350->regmap; + + snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); @@ -1629,8 +1617,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8350 = { .remove = wm8350_codec_remove, .suspend = wm8350_suspend, .resume = wm8350_resume, - .read = wm8350_codec_read, - .write = wm8350_codec_write, .set_bias_level = wm8350_set_bias_level, .controls = wm8350_snd_controls, diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 5dc31ebcd0e7..5d277a915f81 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1,7 +1,7 @@ /* * wm8400.c -- WM8400 ALSA Soc Audio driver * - * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Copyright 2008-11 Wolfson Microelectronics PLC. * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 211285164d70..7c68226376e4 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,7 +1,7 @@ /* * wm8580.c -- WM8580 ALSA Soc Audio driver * - * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Copyright 2008-11 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 9d1b9b0271f1..bb1d26919b10 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -2,6 +2,7 @@ * wm8731.c -- WM8731 ALSA SoC Audio driver * * Copyright 2005 Openedhand Ltd. + * Copyright 2006-12 Wolfson Microelectronics, plc * * Author: Richard Purdie <richard@openedhand.com> * diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 6e849cb04243..35f3d23200e0 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -1,7 +1,7 @@ /* * wm8741.c -- WM8741 ALSA SoC Audio driver * - * Copyright 2010 Wolfson Microelectronics plc + * Copyright 2010-1 Wolfson Microelectronics plc * * Author: Ian Lartey <ian@opensource.wolfsonmicro.com> * diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a26482cd7654..13bff87ddcf5 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1,7 +1,7 @@ /* * wm8753.c -- WM8753 ALSA Soc Audio driver * - * Copyright 2003 Wolfson Microelectronics PLC. + * Copyright 2003-11 Wolfson Microelectronics PLC. * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index a19db5a0a17a..879c356a9045 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -1,7 +1,7 @@ /* * wm8776.c -- WM8776 ALSA SoC Audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6bd1b767b138..c088020172ab 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -1,7 +1,7 @@ /* * wm8804.c -- WM8804 S/PDIF transceiver driver * - * Copyright 2010 Wolfson Microelectronics plc + * Copyright 2010-11 Wolfson Microelectronics plc * * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> * diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 86b8a2926591..73f1c8d7bafb 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1,8 +1,8 @@ /* * wm8903.c -- WM8903 ALSA SoC Audio driver * - * Copyright 2008 Wolfson Microelectronics - * Copyright 2011 NVIDIA, Inc. + * Copyright 2008-12 Wolfson Microelectronics + * Copyright 2011-2012 NVIDIA, Inc. * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -116,6 +116,7 @@ static const struct reg_default wm8903_reg_defaults[] = { struct wm8903_priv { struct wm8903_platform_data *pdata; + struct device *dev; struct snd_soc_codec *codec; struct regmap *regmap; @@ -1635,17 +1636,27 @@ EXPORT_SYMBOL_GPL(wm8903_mic_detect); static irqreturn_t wm8903_irq(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int mic_report; - int int_pol; - int int_val = 0; - int mask = ~snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1_MASK); + struct wm8903_priv *wm8903 = data; + int mic_report, ret; + unsigned int int_val, mask, int_pol; - int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask; + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1_MASK, + &mask); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to read IRQ mask: %d\n", ret); + return IRQ_NONE; + } + + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1, &int_val); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to read IRQ status: %d\n", ret); + return IRQ_NONE; + } + + int_val &= ~mask; if (int_val & WM8903_WSEQ_BUSY_EINT) { - dev_warn(codec->dev, "Write sequencer done\n"); + dev_warn(wm8903->dev, "Write sequencer done\n"); } /* @@ -1656,22 +1667,28 @@ static irqreturn_t wm8903_irq(int irq, void *data) * the polarity register. */ mic_report = wm8903->mic_last_report; - int_pol = snd_soc_read(codec, WM8903_INTERRUPT_POLARITY_1); + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1, + &int_pol); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to read interrupt polarity: %d\n", + ret); + return IRQ_HANDLED; + } #ifndef CONFIG_SND_SOC_WM8903_MODULE if (int_val & (WM8903_MICSHRT_EINT | WM8903_MICDET_EINT)) - trace_snd_soc_jack_irq(dev_name(codec->dev)); + trace_snd_soc_jack_irq(dev_name(wm8903->dev)); #endif if (int_val & WM8903_MICSHRT_EINT) { - dev_dbg(codec->dev, "Microphone short (pol=%x)\n", int_pol); + dev_dbg(wm8903->dev, "Microphone short (pol=%x)\n", int_pol); mic_report ^= wm8903->mic_short; int_pol ^= WM8903_MICSHRT_INV; } if (int_val & WM8903_MICDET_EINT) { - dev_dbg(codec->dev, "Microphone detect (pol=%x)\n", int_pol); + dev_dbg(wm8903->dev, "Microphone detect (pol=%x)\n", int_pol); mic_report ^= wm8903->mic_det; int_pol ^= WM8903_MICDET_INV; @@ -1679,8 +1696,8 @@ static irqreturn_t wm8903_irq(int irq, void *data) msleep(wm8903->mic_delay); } - snd_soc_update_bits(codec, WM8903_INTERRUPT_POLARITY_1, - WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol); + regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1, + WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol); snd_soc_jack_report(wm8903->mic_jack, mic_report, wm8903->mic_short | wm8903->mic_det); @@ -1774,7 +1791,6 @@ static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; int ret; @@ -1782,8 +1798,8 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | WM8903_GP1_DIR; - ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = regmap_update_bits(wm8903->regmap, + WM8903_GPIO_CONTROL_1 + offset, mask, val); if (ret < 0) return ret; @@ -1793,10 +1809,9 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; - int reg; + unsigned int reg; - reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset); + regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, ®); return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT; } @@ -1805,7 +1820,6 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; int ret; @@ -1813,8 +1827,8 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | (value << WM8903_GP2_LVL_SHIFT); - ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = regmap_update_bits(wm8903->regmap, + WM8903_GPIO_CONTROL_1 + offset, mask, val); if (ret < 0) return ret; @@ -1824,11 +1838,10 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; - snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - WM8903_GP1_LVL_MASK, - !!value << WM8903_GP1_LVL_SHIFT); + regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, + WM8903_GP1_LVL_MASK, + !!value << WM8903_GP1_LVL_SHIFT); } static struct gpio_chip wm8903_template_chip = { @@ -1842,15 +1855,14 @@ static struct gpio_chip wm8903_template_chip = { .can_sleep = 1, }; -static void wm8903_init_gpio(struct snd_soc_codec *codec) +static void wm8903_init_gpio(struct wm8903_priv *wm8903) { - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); struct wm8903_platform_data *pdata = wm8903->pdata; int ret; wm8903->gpio_chip = wm8903_template_chip; wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; - wm8903->gpio_chip.dev = codec->dev; + wm8903->gpio_chip.dev = wm8903->dev; if (pdata->gpio_base) wm8903->gpio_chip.base = pdata->gpio_base; @@ -1859,24 +1871,23 @@ static void wm8903_init_gpio(struct snd_soc_codec *codec) ret = gpiochip_add(&wm8903->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); + dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret); } -static void wm8903_free_gpio(struct snd_soc_codec *codec) +static void wm8903_free_gpio(struct wm8903_priv *wm8903) { - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int ret; ret = gpiochip_remove(&wm8903->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret); } #else -static void wm8903_init_gpio(struct snd_soc_codec *codec) +static void wm8903_init_gpio(struct wm8903_priv *wm8903) { } -static void wm8903_free_gpio(struct snd_soc_codec *codec) +static void wm8903_free_gpio(struct wm8903_priv *wm8903) { } #endif @@ -1884,11 +1895,7 @@ static void wm8903_free_gpio(struct snd_soc_codec *codec) static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - struct wm8903_platform_data *pdata = wm8903->pdata; - int ret, i; - int trigger, irq_pol; - u16 val; - bool mic_gpio = false; + int ret; wm8903->codec = codec; codec->control_data = wm8903->regmap; @@ -1899,121 +1906,16 @@ static int wm8903_probe(struct snd_soc_codec *codec) return ret; } - /* Set up GPIOs, detect if any are MIC detect outputs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if ((!pdata->gpio_cfg[i]) || - (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) - continue; - - snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, - pdata->gpio_cfg[i] & 0x7fff); - - val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) - >> WM8903_GP1_FN_SHIFT; - - switch (val) { - case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: - case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: - mic_gpio = true; - break; - default: - break; - } - } - - /* Set up microphone detection */ - snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, - pdata->micdet_cfg); - - /* Microphone detection needs the WSEQ clock */ - if (pdata->micdet_cfg) - snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, - WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); - - /* If microphone detection is enabled by pdata but - * detected via IRQ then interrupts can be lost before - * the machine driver has set up microphone detection - * IRQs as the IRQs are clear on read. The detection - * will be enabled when the machine driver configures. - */ - WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); - - wm8903->mic_delay = pdata->micdet_delay; - - if (wm8903->irq) { - if (pdata->irq_active_low) { - trigger = IRQF_TRIGGER_LOW; - irq_pol = WM8903_IRQ_POL; - } else { - trigger = IRQF_TRIGGER_HIGH; - irq_pol = 0; - } - - snd_soc_update_bits(codec, WM8903_INTERRUPT_CONTROL, - WM8903_IRQ_POL, irq_pol); - - ret = request_threaded_irq(wm8903->irq, NULL, wm8903_irq, - trigger | IRQF_ONESHOT, - "wm8903", codec); - if (ret != 0) { - dev_err(codec->dev, "Failed to request IRQ: %d\n", - ret); - return ret; - } - - /* Enable write sequencer interrupts */ - snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, - WM8903_IM_WSEQ_BUSY_EINT, 0); - } - /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Latch volume update bits */ - val = snd_soc_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT); - val |= WM8903_ADCVU; - snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val); - snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val); - - val = snd_soc_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT); - val |= WM8903_DACVU; - snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val); - snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val); - - val = snd_soc_read(codec, WM8903_ANALOGUE_OUT1_LEFT); - val |= WM8903_HPOUTVU; - snd_soc_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val); - snd_soc_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val); - - val = snd_soc_read(codec, WM8903_ANALOGUE_OUT2_LEFT); - val |= WM8903_LINEOUTVU; - snd_soc_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val); - snd_soc_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val); - - val = snd_soc_read(codec, WM8903_ANALOGUE_OUT3_LEFT); - val |= WM8903_SPKVU; - snd_soc_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val); - snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val); - - /* Enable DAC soft mute by default */ - snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1, - WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, - WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); - - wm8903_init_gpio(codec); - return ret; } /* power down chip */ static int wm8903_remove(struct snd_soc_codec *codec) { - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - - wm8903_free_gpio(codec); wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm8903->irq) - free_irq(wm8903->irq, codec); return 0; } @@ -2123,15 +2025,18 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, { struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev); struct wm8903_priv *wm8903; - unsigned int val; - int ret; + int trigger; + bool mic_gpio = false; + unsigned int val, irq_pol; + int ret, i; wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), GFP_KERNEL); if (wm8903 == NULL) return -ENOMEM; + wm8903->dev = &i2c->dev; - wm8903->regmap = regmap_init_i2c(i2c, &wm8903_regmap); + wm8903->regmap = devm_regmap_init_i2c(i2c, &wm8903_regmap); if (IS_ERR(wm8903->regmap)) { ret = PTR_ERR(wm8903->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2140,7 +2045,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, wm8903); - wm8903->irq = i2c->irq; /* If no platform data was supplied, create storage for defaults */ if (pdata) { @@ -2167,6 +2071,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, } } + pdata = wm8903->pdata; + ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); if (ret != 0) { dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); @@ -2189,6 +2095,107 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, /* Reset the device */ regmap_write(wm8903->regmap, WM8903_SW_RESET_AND_ID, 0x8903); + wm8903_init_gpio(wm8903); + + /* Set up GPIO pin state, detect if any are MIC detect outputs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { + if ((!pdata->gpio_cfg[i]) || + (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) + continue; + + regmap_write(wm8903->regmap, WM8903_GPIO_CONTROL_1 + i, + pdata->gpio_cfg[i] & 0x7fff); + + val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + >> WM8903_GP1_FN_SHIFT; + + switch (val) { + case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: + case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: + mic_gpio = true; + break; + default: + break; + } + } + + /* Set up microphone detection */ + regmap_write(wm8903->regmap, WM8903_MIC_BIAS_CONTROL_0, + pdata->micdet_cfg); + + /* Microphone detection needs the WSEQ clock */ + if (pdata->micdet_cfg) + regmap_update_bits(wm8903->regmap, WM8903_WRITE_SEQUENCER_0, + WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + + /* If microphone detection is enabled by pdata but + * detected via IRQ then interrupts can be lost before + * the machine driver has set up microphone detection + * IRQs as the IRQs are clear on read. The detection + * will be enabled when the machine driver configures. + */ + WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + + wm8903->mic_delay = pdata->micdet_delay; + + if (i2c->irq) { + if (pdata->irq_active_low) { + trigger = IRQF_TRIGGER_LOW; + irq_pol = WM8903_IRQ_POL; + } else { + trigger = IRQF_TRIGGER_HIGH; + irq_pol = 0; + } + + regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_CONTROL, + WM8903_IRQ_POL, irq_pol); + + ret = request_threaded_irq(i2c->irq, NULL, wm8903_irq, + trigger | IRQF_ONESHOT, + "wm8903", wm8903); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to request IRQ: %d\n", + ret); + return ret; + } + + /* Enable write sequencer interrupts */ + regmap_update_bits(wm8903->regmap, + WM8903_INTERRUPT_STATUS_1_MASK, + WM8903_IM_WSEQ_BUSY_EINT, 0); + } + + /* Latch volume update bits */ + regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_LEFT, + WM8903_ADCVU, WM8903_ADCVU); + regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_RIGHT, + WM8903_ADCVU, WM8903_ADCVU); + + regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_LEFT, + WM8903_DACVU, WM8903_DACVU); + regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_RIGHT, + WM8903_DACVU, WM8903_DACVU); + + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_LEFT, + WM8903_HPOUTVU, WM8903_HPOUTVU); + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_RIGHT, + WM8903_HPOUTVU, WM8903_HPOUTVU); + + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_LEFT, + WM8903_LINEOUTVU, WM8903_LINEOUTVU); + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_RIGHT, + WM8903_LINEOUTVU, WM8903_LINEOUTVU); + + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_LEFT, + WM8903_SPKVU, WM8903_SPKVU); + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_RIGHT, + WM8903_SPKVU, WM8903_SPKVU); + + /* Enable DAC soft mute by default */ + regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_1, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); if (ret != 0) @@ -2196,7 +2203,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, return 0; err: - regmap_exit(wm8903->regmap); return ret; } @@ -2204,7 +2210,9 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) { struct wm8903_priv *wm8903 = i2c_get_clientdata(client); - regmap_exit(wm8903->regmap); + if (client->irq) + free_irq(client->irq, wm8903); + wm8903_free_gpio(wm8903); snd_soc_unregister_codec(&client->dev); return 0; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 812acd83fb48..0013afe48e66 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1,7 +1,7 @@ /* * wm8904.c -- WM8904 ALSA SoC Audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -314,11 +314,6 @@ static bool wm8904_readable_register(struct device *dev, unsigned int reg) } } -static int wm8904_reset(struct snd_soc_codec *codec) -{ - return snd_soc_write(codec, WM8904_SW_RESET_AND_ID, 0); -} - static int wm8904_configure_clocking(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); @@ -1945,25 +1940,6 @@ static struct snd_soc_dai_driver wm8904_dai = { .symmetric_rates = 1, }; -#ifdef CONFIG_PM -static int wm8904_suspend(struct snd_soc_codec *codec) -{ - wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int wm8904_resume(struct snd_soc_codec *codec) -{ - wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define wm8904_suspend NULL -#define wm8904_resume NULL -#endif - static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); @@ -2078,8 +2054,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - struct wm8904_pdata *pdata = wm8904->pdata; - int ret, i; + int ret; codec->control_data = wm8904->regmap; @@ -2101,127 +2076,17 @@ static int wm8904_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++) - wm8904->supplies[i].supply = wm8904_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8904->supplies), - wm8904->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), - wm8904->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - - ret = snd_soc_read(codec, WM8904_SW_RESET_AND_ID); - if (ret < 0) { - dev_err(codec->dev, "Failed to read ID register\n"); - goto err_enable; - } - if (ret != 0x8904) { - dev_err(codec->dev, "Device is not a WM8904, ID is %x\n", ret); - ret = -EINVAL; - goto err_enable; - } - - ret = snd_soc_read(codec, WM8904_REVISION); - if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", - ret); - goto err_enable; - } - dev_info(codec->dev, "revision %c\n", ret + 'A'); - - ret = wm8904_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_enable; - } - - regcache_cache_only(wm8904->regmap, true); - /* Change some default settings - latch VU and enable ZC */ - snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, - WM8904_ADC_VU, WM8904_ADC_VU); - snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT, - WM8904_ADC_VU, WM8904_ADC_VU); - snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT, - WM8904_DAC_VU, WM8904_DAC_VU); - snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT, - WM8904_DAC_VU, WM8904_DAC_VU); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT, - WM8904_HPOUT_VU | WM8904_HPOUTLZC, - WM8904_HPOUT_VU | WM8904_HPOUTLZC); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT, - WM8904_HPOUT_VU | WM8904_HPOUTRZC, - WM8904_HPOUT_VU | WM8904_HPOUTRZC); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT, - WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, - WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT, - WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, - WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); - snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, - WM8904_SR_MODE, 0); - - /* Apply configuration from the platform data. */ - if (wm8904->pdata) { - for (i = 0; i < WM8904_GPIO_REGS; i++) { - if (!pdata->gpio_cfg[i]) - continue; - - regmap_update_bits(wm8904->regmap, - WM8904_GPIO_CONTROL_1 + i, - 0xffff, - pdata->gpio_cfg[i]); - } - - /* Zero is the default value for these anyway */ - for (i = 0; i < WM8904_MIC_REGS; i++) - regmap_update_bits(wm8904->regmap, - WM8904_MIC_BIAS_CONTROL_0 + i, - 0xffff, - pdata->mic_cfg[i]); - } - - /* Set Class W by default - this will be managed by the Class - * G widget at runtime where bypass paths are available. - */ - snd_soc_update_bits(codec, WM8904_CLASS_W_0, - WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); - - /* Use normal bias source */ - snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, - WM8904_POBCTRL, 0); - - wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* Bias level configuration will have done an extra enable */ - regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); - wm8904_handle_pdata(codec); wm8904_add_widgets(codec); return 0; - -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); - return ret; } static int wm8904_remove(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); kfree(wm8904->retune_mobile_texts); kfree(wm8904->drc_texts); @@ -2231,8 +2096,6 @@ static int wm8904_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8904 = { .probe = wm8904_probe, .remove = wm8904_remove, - .suspend = wm8904_suspend, - .resume = wm8904_resume, .set_bias_level = wm8904_set_bias_level, .idle_bias_off = true, }; @@ -2254,14 +2117,15 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8904_priv *wm8904; - int ret; + unsigned int val; + int ret, i; wm8904 = devm_kzalloc(&i2c->dev, sizeof(struct wm8904_priv), GFP_KERNEL); if (wm8904 == NULL) return -ENOMEM; - wm8904->regmap = regmap_init_i2c(i2c, &wm8904_regmap); + wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap); if (IS_ERR(wm8904->regmap)) { ret = PTR_ERR(wm8904->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2273,23 +2137,121 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8904); wm8904->pdata = i2c->dev.platform_data; + for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++) + wm8904->supplies[i].supply = wm8904_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); + goto err_enable; + } + if (val != 0x8904) { + dev_err(&i2c->dev, "Device is not a WM8904, ID is %x\n", val); + ret = -EINVAL; + goto err_enable; + } + + ret = regmap_read(wm8904->regmap, WM8904_REVISION, &val); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read device revision: %d\n", + ret); + goto err_enable; + } + dev_info(&i2c->dev, "revision %c\n", val + 'A'); + + ret = regmap_write(wm8904->regmap, WM8904_SW_RESET_AND_ID, 0); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + goto err_enable; + } + + /* Change some default settings - latch VU and enable ZC */ + regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_VU, WM8904_ADC_VU); + regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_RIGHT, + WM8904_ADC_VU, WM8904_ADC_VU); + regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_VU, WM8904_DAC_VU); + regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_RIGHT, + WM8904_DAC_VU, WM8904_DAC_VU); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_LEFT, + WM8904_HPOUT_VU | WM8904_HPOUTLZC, + WM8904_HPOUT_VU | WM8904_HPOUTLZC); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_RIGHT, + WM8904_HPOUT_VU | WM8904_HPOUTRZC, + WM8904_HPOUT_VU | WM8904_HPOUTRZC); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_LEFT, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_RIGHT, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); + regmap_update_bits(wm8904->regmap, WM8904_CLOCK_RATES_0, + WM8904_SR_MODE, 0); + + /* Apply configuration from the platform data. */ + if (wm8904->pdata) { + for (i = 0; i < WM8904_GPIO_REGS; i++) { + if (!wm8904->pdata->gpio_cfg[i]) + continue; + + regmap_update_bits(wm8904->regmap, + WM8904_GPIO_CONTROL_1 + i, + 0xffff, + wm8904->pdata->gpio_cfg[i]); + } + + /* Zero is the default value for these anyway */ + for (i = 0; i < WM8904_MIC_REGS; i++) + regmap_update_bits(wm8904->regmap, + WM8904_MIC_BIAS_CONTROL_0 + i, + 0xffff, + wm8904->pdata->mic_cfg[i]); + } + + /* Set Class W by default - this will be managed by the Class + * G widget at runtime where bypass paths are available. + */ + regmap_update_bits(wm8904->regmap, WM8904_CLASS_W_0, + WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); + + /* Use normal bias source */ + regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0, + WM8904_POBCTRL, 0); + + /* Can leave the device powered off until we need it */ + regcache_cache_only(wm8904->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8904, &wm8904_dai, 1); if (ret != 0) - goto err; + return ret; return 0; -err: - regmap_exit(wm8904->regmap); +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); return ret; } static __devexit int wm8904_i2c_remove(struct i2c_client *client) { - struct wm8904_priv *wm8904 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regmap_exit(wm8904->regmap); return 0; } @@ -2311,23 +2273,7 @@ static struct i2c_driver wm8904_i2c_driver = { .id_table = wm8904_i2c_id, }; -static int __init wm8904_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8904_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8904 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8904_modinit); - -static void __exit wm8904_exit(void) -{ - i2c_del_driver(&wm8904_i2c_driver); -} -module_exit(wm8904_exit); +module_i2c_driver(wm8904_i2c_driver); MODULE_DESCRIPTION("ASoC WM8904 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 8bc659d8dd2e..96518ac8e24c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1,6 +1,8 @@ /* * wm8960.c -- WM8960 ALSA SoC Audio driver * + * Copyright 2007-11 Wolfson Microelectronics, plc + * * Author: Liam Girdwood * * This program is free software; you can redistribute it and/or modify diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 05ea7c274093..01edbcc754d2 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1,6 +1,8 @@ /* * wm8961.c -- WM8961 ALSA SoC Audio driver * + * Copyright 2009-10 Wolfson Microelectronics, plc + * * Author: Mark Brown * * This program is free software; you can redistribute it and/or modify diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0cfce9999c89..eaf65863ec21 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1,7 +1,7 @@ /* * wm8962.c -- WM8962 ALSA SoC Audio driver * - * Copyright 2010 Wolfson Microelectronics plc + * Copyright 2010-2 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -2580,6 +2580,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, WM8962_SAMPLE_RATE_INT_MODE | WM8962_SAMPLE_RATE_MASK, adctl3); + dev_dbg(codec->dev, "hw_params set BCLK %dHz LRCLK %dHz\n", + wm8962->bclk, wm8962->lrclk); + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) wm8962_configure_bclk(codec); @@ -3722,6 +3725,9 @@ static int wm8962_runtime_resume(struct device *dev) } regcache_cache_only(wm8962->regmap, false); + + wm8962_reset(wm8962); + regcache_sync(wm8962->regmap); regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 36acfccab999..9fd80d688979 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1,7 +1,7 @@ /* * wm8993.c -- WM8993 ALSA SoC audio driver * - * Copyright 2009, 2010 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1436b6ce74d1..bb62f4b3d563 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1,7 +1,7 @@ /* * wm8994.c -- WM8994 ALSA SoC Audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -2967,23 +2967,8 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_codec_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; - switch (control->type) { - case WM8994: - snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); - break; - case WM1811: - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, 0); - /* Fall through */ - case WM8958: - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, 0); - break; - } - for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -3033,28 +3018,6 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) i + 1, ret); } - switch (control->type) { - case WM8994: - if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) - snd_soc_update_bits(codec, WM8994_MICBIAS, - WM8994_MICD_ENA, WM8994_MICD_ENA); - break; - case WM1811: - if (wm8994->jackdet && wm8994->jack_cb) { - /* Restart from idle */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, - WM1811_JACKDET_MODE_JACK); - break; - } - break; - case WM8958: - if (wm8994->jack_cb) - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, WM8958_MICD_ENA); - break; - } - return 0; } #else @@ -3729,9 +3692,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) if (wm8994->pdata && wm8994->pdata->micdet_irq) wm8994->micdet_irq = wm8994->pdata->micdet_irq; - else if (wm8994->pdata && wm8994->pdata->irq_base) - wm8994->micdet_irq = wm8994->pdata->irq_base + - WM8994_IRQ_MIC1_DET; pm_runtime_enable(codec->dev); pm_runtime_idle(codec->dev); @@ -3870,6 +3830,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) dev_warn(codec->dev, "Failed to request Mic detect IRQ: %d\n", ret); + } else { + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET, + wm8958_mic_irq, "Mic detect", + wm8994); } } diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index dc9b42b7fc4d..00f183dfa454 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1,7 +1,7 @@ /* * wm8996.c - WM8996 audio codec interface * - * Copyright 2011 Wolfson Microelectronics PLC. + * Copyright 2011-2 Wolfson Microelectronics PLC. * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify it @@ -296,184 +296,6 @@ static struct reg_default wm8996_reg[] = { { WM8996_RIGHT_PDM_SPEAKER, 0x1 }, { WM8996_PDM_SPEAKER_MUTE_SEQUENCE, 0x69 }, { WM8996_PDM_SPEAKER_VOLUME, 0x66 }, - { WM8996_WRITE_SEQUENCER_0, 0x1 }, - { WM8996_WRITE_SEQUENCER_1, 0x1 }, - { WM8996_WRITE_SEQUENCER_3, 0x6 }, - { WM8996_WRITE_SEQUENCER_4, 0x40 }, - { WM8996_WRITE_SEQUENCER_5, 0x1 }, - { WM8996_WRITE_SEQUENCER_6, 0xf }, - { WM8996_WRITE_SEQUENCER_7, 0x6 }, - { WM8996_WRITE_SEQUENCER_8, 0x1 }, - { WM8996_WRITE_SEQUENCER_9, 0x3 }, - { WM8996_WRITE_SEQUENCER_10, 0x104 }, - { WM8996_WRITE_SEQUENCER_12, 0x60 }, - { WM8996_WRITE_SEQUENCER_13, 0x11 }, - { WM8996_WRITE_SEQUENCER_14, 0x401 }, - { WM8996_WRITE_SEQUENCER_16, 0x50 }, - { WM8996_WRITE_SEQUENCER_17, 0x3 }, - { WM8996_WRITE_SEQUENCER_18, 0x100 }, - { WM8996_WRITE_SEQUENCER_20, 0x51 }, - { WM8996_WRITE_SEQUENCER_21, 0x3 }, - { WM8996_WRITE_SEQUENCER_22, 0x104 }, - { WM8996_WRITE_SEQUENCER_23, 0xa }, - { WM8996_WRITE_SEQUENCER_24, 0x60 }, - { WM8996_WRITE_SEQUENCER_25, 0x3b }, - { WM8996_WRITE_SEQUENCER_26, 0x502 }, - { WM8996_WRITE_SEQUENCER_27, 0x100 }, - { WM8996_WRITE_SEQUENCER_28, 0x2fff }, - { WM8996_WRITE_SEQUENCER_32, 0x2fff }, - { WM8996_WRITE_SEQUENCER_36, 0x2fff }, - { WM8996_WRITE_SEQUENCER_40, 0x2fff }, - { WM8996_WRITE_SEQUENCER_44, 0x2fff }, - { WM8996_WRITE_SEQUENCER_48, 0x2fff }, - { WM8996_WRITE_SEQUENCER_52, 0x2fff }, - { WM8996_WRITE_SEQUENCER_56, 0x2fff }, - { WM8996_WRITE_SEQUENCER_60, 0x2fff }, - { WM8996_WRITE_SEQUENCER_64, 0x1 }, - { WM8996_WRITE_SEQUENCER_65, 0x1 }, - { WM8996_WRITE_SEQUENCER_67, 0x6 }, - { WM8996_WRITE_SEQUENCER_68, 0x40 }, - { WM8996_WRITE_SEQUENCER_69, 0x1 }, - { WM8996_WRITE_SEQUENCER_70, 0xf }, - { WM8996_WRITE_SEQUENCER_71, 0x6 }, - { WM8996_WRITE_SEQUENCER_72, 0x1 }, - { WM8996_WRITE_SEQUENCER_73, 0x3 }, - { WM8996_WRITE_SEQUENCER_74, 0x104 }, - { WM8996_WRITE_SEQUENCER_76, 0x60 }, - { WM8996_WRITE_SEQUENCER_77, 0x11 }, - { WM8996_WRITE_SEQUENCER_78, 0x401 }, - { WM8996_WRITE_SEQUENCER_80, 0x50 }, - { WM8996_WRITE_SEQUENCER_81, 0x3 }, - { WM8996_WRITE_SEQUENCER_82, 0x100 }, - { WM8996_WRITE_SEQUENCER_84, 0x60 }, - { WM8996_WRITE_SEQUENCER_85, 0x3b }, - { WM8996_WRITE_SEQUENCER_86, 0x502 }, - { WM8996_WRITE_SEQUENCER_87, 0x100 }, - { WM8996_WRITE_SEQUENCER_88, 0x2fff }, - { WM8996_WRITE_SEQUENCER_92, 0x2fff }, - { WM8996_WRITE_SEQUENCER_96, 0x2fff }, - { WM8996_WRITE_SEQUENCER_100, 0x2fff }, - { WM8996_WRITE_SEQUENCER_104, 0x2fff }, - { WM8996_WRITE_SEQUENCER_108, 0x2fff }, - { WM8996_WRITE_SEQUENCER_112, 0x2fff }, - { WM8996_WRITE_SEQUENCER_116, 0x2fff }, - { WM8996_WRITE_SEQUENCER_120, 0x2fff }, - { WM8996_WRITE_SEQUENCER_124, 0x2fff }, - { WM8996_WRITE_SEQUENCER_128, 0x1 }, - { WM8996_WRITE_SEQUENCER_129, 0x1 }, - { WM8996_WRITE_SEQUENCER_131, 0x6 }, - { WM8996_WRITE_SEQUENCER_132, 0x40 }, - { WM8996_WRITE_SEQUENCER_133, 0x1 }, - { WM8996_WRITE_SEQUENCER_134, 0xf }, - { WM8996_WRITE_SEQUENCER_135, 0x6 }, - { WM8996_WRITE_SEQUENCER_136, 0x1 }, - { WM8996_WRITE_SEQUENCER_137, 0x3 }, - { WM8996_WRITE_SEQUENCER_138, 0x106 }, - { WM8996_WRITE_SEQUENCER_140, 0x61 }, - { WM8996_WRITE_SEQUENCER_141, 0x11 }, - { WM8996_WRITE_SEQUENCER_142, 0x401 }, - { WM8996_WRITE_SEQUENCER_144, 0x50 }, - { WM8996_WRITE_SEQUENCER_145, 0x3 }, - { WM8996_WRITE_SEQUENCER_146, 0x102 }, - { WM8996_WRITE_SEQUENCER_148, 0x51 }, - { WM8996_WRITE_SEQUENCER_149, 0x3 }, - { WM8996_WRITE_SEQUENCER_150, 0x106 }, - { WM8996_WRITE_SEQUENCER_151, 0xa }, - { WM8996_WRITE_SEQUENCER_152, 0x61 }, - { WM8996_WRITE_SEQUENCER_153, 0x3b }, - { WM8996_WRITE_SEQUENCER_154, 0x502 }, - { WM8996_WRITE_SEQUENCER_155, 0x100 }, - { WM8996_WRITE_SEQUENCER_156, 0x2fff }, - { WM8996_WRITE_SEQUENCER_160, 0x2fff }, - { WM8996_WRITE_SEQUENCER_164, 0x2fff }, - { WM8996_WRITE_SEQUENCER_168, 0x2fff }, - { WM8996_WRITE_SEQUENCER_172, 0x2fff }, - { WM8996_WRITE_SEQUENCER_176, 0x2fff }, - { WM8996_WRITE_SEQUENCER_180, 0x2fff }, - { WM8996_WRITE_SEQUENCER_184, 0x2fff }, - { WM8996_WRITE_SEQUENCER_188, 0x2fff }, - { WM8996_WRITE_SEQUENCER_192, 0x1 }, - { WM8996_WRITE_SEQUENCER_193, 0x1 }, - { WM8996_WRITE_SEQUENCER_195, 0x6 }, - { WM8996_WRITE_SEQUENCER_196, 0x40 }, - { WM8996_WRITE_SEQUENCER_197, 0x1 }, - { WM8996_WRITE_SEQUENCER_198, 0xf }, - { WM8996_WRITE_SEQUENCER_199, 0x6 }, - { WM8996_WRITE_SEQUENCER_200, 0x1 }, - { WM8996_WRITE_SEQUENCER_201, 0x3 }, - { WM8996_WRITE_SEQUENCER_202, 0x106 }, - { WM8996_WRITE_SEQUENCER_204, 0x61 }, - { WM8996_WRITE_SEQUENCER_205, 0x11 }, - { WM8996_WRITE_SEQUENCER_206, 0x401 }, - { WM8996_WRITE_SEQUENCER_208, 0x50 }, - { WM8996_WRITE_SEQUENCER_209, 0x3 }, - { WM8996_WRITE_SEQUENCER_210, 0x102 }, - { WM8996_WRITE_SEQUENCER_212, 0x61 }, - { WM8996_WRITE_SEQUENCER_213, 0x3b }, - { WM8996_WRITE_SEQUENCER_214, 0x502 }, - { WM8996_WRITE_SEQUENCER_215, 0x100 }, - { WM8996_WRITE_SEQUENCER_216, 0x2fff }, - { WM8996_WRITE_SEQUENCER_220, 0x2fff }, - { WM8996_WRITE_SEQUENCER_224, 0x2fff }, - { WM8996_WRITE_SEQUENCER_228, 0x2fff }, - { WM8996_WRITE_SEQUENCER_232, 0x2fff }, - { WM8996_WRITE_SEQUENCER_236, 0x2fff }, - { WM8996_WRITE_SEQUENCER_240, 0x2fff }, - { WM8996_WRITE_SEQUENCER_244, 0x2fff }, - { WM8996_WRITE_SEQUENCER_248, 0x2fff }, - { WM8996_WRITE_SEQUENCER_252, 0x2fff }, - { WM8996_WRITE_SEQUENCER_256, 0x60 }, - { WM8996_WRITE_SEQUENCER_258, 0x601 }, - { WM8996_WRITE_SEQUENCER_260, 0x50 }, - { WM8996_WRITE_SEQUENCER_262, 0x100 }, - { WM8996_WRITE_SEQUENCER_264, 0x1 }, - { WM8996_WRITE_SEQUENCER_266, 0x104 }, - { WM8996_WRITE_SEQUENCER_267, 0x100 }, - { WM8996_WRITE_SEQUENCER_268, 0x2fff }, - { WM8996_WRITE_SEQUENCER_272, 0x2fff }, - { WM8996_WRITE_SEQUENCER_276, 0x2fff }, - { WM8996_WRITE_SEQUENCER_280, 0x2fff }, - { WM8996_WRITE_SEQUENCER_284, 0x2fff }, - { WM8996_WRITE_SEQUENCER_288, 0x2fff }, - { WM8996_WRITE_SEQUENCER_292, 0x2fff }, - { WM8996_WRITE_SEQUENCER_296, 0x2fff }, - { WM8996_WRITE_SEQUENCER_300, 0x2fff }, - { WM8996_WRITE_SEQUENCER_304, 0x2fff }, - { WM8996_WRITE_SEQUENCER_308, 0x2fff }, - { WM8996_WRITE_SEQUENCER_312, 0x2fff }, - { WM8996_WRITE_SEQUENCER_316, 0x2fff }, - { WM8996_WRITE_SEQUENCER_320, 0x61 }, - { WM8996_WRITE_SEQUENCER_322, 0x601 }, - { WM8996_WRITE_SEQUENCER_324, 0x50 }, - { WM8996_WRITE_SEQUENCER_326, 0x102 }, - { WM8996_WRITE_SEQUENCER_328, 0x1 }, - { WM8996_WRITE_SEQUENCER_330, 0x106 }, - { WM8996_WRITE_SEQUENCER_331, 0x100 }, - { WM8996_WRITE_SEQUENCER_332, 0x2fff }, - { WM8996_WRITE_SEQUENCER_336, 0x2fff }, - { WM8996_WRITE_SEQUENCER_340, 0x2fff }, - { WM8996_WRITE_SEQUENCER_344, 0x2fff }, - { WM8996_WRITE_SEQUENCER_348, 0x2fff }, - { WM8996_WRITE_SEQUENCER_352, 0x2fff }, - { WM8996_WRITE_SEQUENCER_356, 0x2fff }, - { WM8996_WRITE_SEQUENCER_360, 0x2fff }, - { WM8996_WRITE_SEQUENCER_364, 0x2fff }, - { WM8996_WRITE_SEQUENCER_368, 0x2fff }, - { WM8996_WRITE_SEQUENCER_372, 0x2fff }, - { WM8996_WRITE_SEQUENCER_376, 0x2fff }, - { WM8996_WRITE_SEQUENCER_380, 0x2fff }, - { WM8996_WRITE_SEQUENCER_384, 0x60 }, - { WM8996_WRITE_SEQUENCER_386, 0x601 }, - { WM8996_WRITE_SEQUENCER_388, 0x61 }, - { WM8996_WRITE_SEQUENCER_390, 0x601 }, - { WM8996_WRITE_SEQUENCER_392, 0x50 }, - { WM8996_WRITE_SEQUENCER_394, 0x300 }, - { WM8996_WRITE_SEQUENCER_396, 0x1 }, - { WM8996_WRITE_SEQUENCER_398, 0x304 }, - { WM8996_WRITE_SEQUENCER_400, 0x40 }, - { WM8996_WRITE_SEQUENCER_402, 0xf }, - { WM8996_WRITE_SEQUENCER_404, 0x1 }, - { WM8996_WRITE_SEQUENCER_407, 0x100 }, }; static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0); @@ -1706,18 +1528,6 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) } } -static int wm8996_reset(struct wm8996_priv *wm8996) -{ - if (wm8996->pdata.ldo_ena > 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); - return 0; - } else { - return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, - 0x8915); - } -} - static const int bclk_divs[] = { 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 }; @@ -1809,8 +1619,10 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: regcache_cache_only(codec->control_data, true); - if (wm8996->pdata.ldo_ena >= 0) + if (wm8996->pdata.ldo_ena >= 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + regcache_cache_only(codec->control_data, true); + } regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); break; @@ -2807,7 +2619,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) int ret; struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - int i, irq_flags; + int irq_flags; wm8996->codec = codec; @@ -2822,177 +2634,12 @@ static int wm8996_probe(struct snd_soc_codec *codec) goto err; } - wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; - wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; - wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; - - /* This should really be moved into the regulator core */ - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { - ret = regulator_register_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); - if (ret != 0) { - dev_err(codec->dev, - "Failed to register regulator notifier: %d\n", - ret); - } - } - - /* Apply platform data settings */ - snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL, - WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, - wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT | - wm8996->pdata.inr_mode); - - for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) { - if (!wm8996->pdata.gpio_default[i]) - continue; - - snd_soc_write(codec, WM8996_GPIO_1 + i, - wm8996->pdata.gpio_default[i] & 0xffff); - } - - if (wm8996->pdata.spkmute_seq) - snd_soc_update_bits(codec, WM8996_PDM_SPEAKER_MUTE_SEQUENCE, - WM8996_SPK_MUTE_ENDIAN | - WM8996_SPK_MUTE_SEQ1_MASK, - wm8996->pdata.spkmute_seq); - - snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_2, - WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC | - WM8996_MICD_SRC, wm8996->pdata.micdet_def); - - /* Latch volume update bits */ - snd_soc_update_bits(codec, WM8996_LEFT_LINE_INPUT_VOLUME, - WM8996_IN1_VU, WM8996_IN1_VU); - snd_soc_update_bits(codec, WM8996_RIGHT_LINE_INPUT_VOLUME, - WM8996_IN1_VU, WM8996_IN1_VU); - - snd_soc_update_bits(codec, WM8996_DAC1_LEFT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_DAC1_RIGHT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_DAC2_LEFT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - snd_soc_update_bits(codec, WM8996_DAC2_RIGHT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - - snd_soc_update_bits(codec, WM8996_OUTPUT1_LEFT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_OUTPUT1_RIGHT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_OUTPUT2_LEFT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - snd_soc_update_bits(codec, WM8996_OUTPUT2_RIGHT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - - snd_soc_update_bits(codec, WM8996_DSP1_TX_LEFT_VOLUME, - WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); - snd_soc_update_bits(codec, WM8996_DSP1_TX_RIGHT_VOLUME, - WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_TX_LEFT_VOLUME, - WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_TX_RIGHT_VOLUME, - WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); - - snd_soc_update_bits(codec, WM8996_DSP1_RX_LEFT_VOLUME, - WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); - snd_soc_update_bits(codec, WM8996_DSP1_RX_RIGHT_VOLUME, - WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_RX_LEFT_VOLUME, - WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_RX_RIGHT_VOLUME, - WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); - - /* No support currently for the underclocked TDM modes and - * pick a default TDM layout with each channel pair working with - * slots 0 and 1. */ - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_0_CONFIGURATION, - WM8996_AIF1RX_CHAN0_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_1_CONFIGURATION, - WM8996_AIF1RX_CHAN1_SLOTS_MASK | - WM8996_AIF1RX_CHAN1_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_2_CONFIGURATION, - WM8996_AIF1RX_CHAN2_SLOTS_MASK | - WM8996_AIF1RX_CHAN2_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_3_CONFIGURATION, - WM8996_AIF1RX_CHAN3_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_4_CONFIGURATION, - WM8996_AIF1RX_CHAN4_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_5_CONFIGURATION, - WM8996_AIF1RX_CHAN5_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1); - - snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_0_CONFIGURATION, - WM8996_AIF2RX_CHAN0_SLOTS_MASK | - WM8996_AIF2RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_1_CONFIGURATION, - WM8996_AIF2RX_CHAN1_SLOTS_MASK | - WM8996_AIF2RX_CHAN1_START_SLOT_MASK, - 1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1); - - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_0_CONFIGURATION, - WM8996_AIF1TX_CHAN0_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, - WM8996_AIF1TX_CHAN1_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_2_CONFIGURATION, - WM8996_AIF1TX_CHAN2_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_3_CONFIGURATION, - WM8996_AIF1TX_CHAN3_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_4_CONFIGURATION, - WM8996_AIF1TX_CHAN4_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_5_CONFIGURATION, - WM8996_AIF1TX_CHAN5_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1); - - snd_soc_update_bits(codec, WM8996_AIF2TX_CHANNEL_0_CONFIGURATION, - WM8996_AIF2TX_CHAN0_SLOTS_MASK | - WM8996_AIF2TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, - WM8996_AIF2TX_CHAN1_SLOTS_MASK | - WM8996_AIF2TX_CHAN1_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); - if (wm8996->pdata.num_retune_mobile_cfgs) wm8996_retune_mobile_pdata(codec); else snd_soc_add_codec_controls(codec, wm8996_eq_controls, ARRAY_SIZE(wm8996_eq_controls)); - /* If the TX LRCLK pins are not in LRCLK mode configure the - * AIFs to source their clocks from the RX LRCLKs. - */ - if ((snd_soc_read(codec, WM8996_GPIO_1))) - snd_soc_update_bits(codec, WM8996_AIF1_TX_LRCLK_2, - WM8996_AIF1TX_LRCLK_MODE, - WM8996_AIF1TX_LRCLK_MODE); - - if ((snd_soc_read(codec, WM8996_GPIO_2))) - snd_soc_update_bits(codec, WM8996_AIF2_TX_LRCLK_2, - WM8996_AIF2TX_LRCLK_MODE, - WM8996_AIF2TX_LRCLK_MODE); - if (i2c->irq) { if (wm8996->pdata.irq_flags) irq_flags = wm8996->pdata.irq_flags; @@ -3036,9 +2683,7 @@ err: static int wm8996_remove(struct snd_soc_codec *codec) { - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - int i; snd_soc_update_bits(codec, WM8996_INTERRUPT_CONTROL, WM8996_IM_IRQ, WM8996_IM_IRQ); @@ -3046,10 +2691,6 @@ static int wm8996_remove(struct snd_soc_codec *codec) if (i2c->irq) free_irq(i2c->irq, codec); - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) - regulator_unregister_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); - return 0; } @@ -3163,6 +2804,21 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, goto err_gpio; } + wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; + wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; + wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { + ret = regulator_register_notifier(wm8996->supplies[i].consumer, + &wm8996->disable_nb[i]); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); if (ret != 0) { @@ -3175,7 +2831,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, msleep(5); } - wm8996->regmap = regmap_init_i2c(i2c, &wm8996_regmap); + wm8996->regmap = devm_regmap_init_i2c(i2c, &wm8996_regmap); if (IS_ERR(wm8996->regmap)) { ret = PTR_ERR(wm8996->regmap); dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); @@ -3203,15 +2859,199 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "revision %c\n", (reg & WM8996_CHIP_REV_MASK) + 'A'); - ret = wm8996_reset(wm8996); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to issue reset\n"); - goto err_regmap; + if (wm8996->pdata.ldo_ena > 0) { + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + regcache_cache_only(wm8996->regmap, true); + } else { + ret = regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, + 0x8915); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + goto err_regmap; + } } - regcache_cache_only(wm8996->regmap, true); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); + /* Apply platform data settings */ + regmap_update_bits(wm8996->regmap, WM8996_LINE_INPUT_CONTROL, + WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, + wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT | + wm8996->pdata.inr_mode); + + for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) { + if (!wm8996->pdata.gpio_default[i]) + continue; + + regmap_write(wm8996->regmap, WM8996_GPIO_1 + i, + wm8996->pdata.gpio_default[i] & 0xffff); + } + + if (wm8996->pdata.spkmute_seq) + regmap_update_bits(wm8996->regmap, + WM8996_PDM_SPEAKER_MUTE_SEQUENCE, + WM8996_SPK_MUTE_ENDIAN | + WM8996_SPK_MUTE_SEQ1_MASK, + wm8996->pdata.spkmute_seq); + + regmap_update_bits(wm8996->regmap, WM8996_ACCESSORY_DETECT_MODE_2, + WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC | + WM8996_MICD_SRC, wm8996->pdata.micdet_def); + + /* Latch volume update bits */ + regmap_update_bits(wm8996->regmap, WM8996_LEFT_LINE_INPUT_VOLUME, + WM8996_IN1_VU, WM8996_IN1_VU); + regmap_update_bits(wm8996->regmap, WM8996_RIGHT_LINE_INPUT_VOLUME, + WM8996_IN1_VU, WM8996_IN1_VU); + + regmap_update_bits(wm8996->regmap, WM8996_DAC1_LEFT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_DAC1_RIGHT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_DAC2_LEFT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + regmap_update_bits(wm8996->regmap, WM8996_DAC2_RIGHT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_LEFT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_RIGHT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_LEFT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_RIGHT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + + regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_LEFT_VOLUME, + WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_RIGHT_VOLUME, + WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_LEFT_VOLUME, + WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_RIGHT_VOLUME, + WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); + + regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_LEFT_VOLUME, + WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_RIGHT_VOLUME, + WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_LEFT_VOLUME, + WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_RIGHT_VOLUME, + WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); + + /* No support currently for the underclocked TDM modes and + * pick a default TDM layout with each channel pair working with + * slots 0 and 1. */ + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_0_CONFIGURATION, + WM8996_AIF1RX_CHAN0_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_1_CONFIGURATION, + WM8996_AIF1RX_CHAN1_SLOTS_MASK | + WM8996_AIF1RX_CHAN1_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_2_CONFIGURATION, + WM8996_AIF1RX_CHAN2_SLOTS_MASK | + WM8996_AIF1RX_CHAN2_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_3_CONFIGURATION, + WM8996_AIF1RX_CHAN3_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_4_CONFIGURATION, + WM8996_AIF1RX_CHAN4_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_5_CONFIGURATION, + WM8996_AIF1RX_CHAN5_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1); + + regmap_update_bits(wm8996->regmap, + WM8996_AIF2RX_CHANNEL_0_CONFIGURATION, + WM8996_AIF2RX_CHAN0_SLOTS_MASK | + WM8996_AIF2RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF2RX_CHANNEL_1_CONFIGURATION, + WM8996_AIF2RX_CHAN1_SLOTS_MASK | + WM8996_AIF2RX_CHAN1_START_SLOT_MASK, + 1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1); + + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_0_CONFIGURATION, + WM8996_AIF1TX_CHAN0_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, + WM8996_AIF1TX_CHAN1_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_2_CONFIGURATION, + WM8996_AIF1TX_CHAN2_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_3_CONFIGURATION, + WM8996_AIF1TX_CHAN3_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_4_CONFIGURATION, + WM8996_AIF1TX_CHAN4_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_5_CONFIGURATION, + WM8996_AIF1TX_CHAN5_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1); + + regmap_update_bits(wm8996->regmap, + WM8996_AIF2TX_CHANNEL_0_CONFIGURATION, + WM8996_AIF2TX_CHAN0_SLOTS_MASK | + WM8996_AIF2TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, + WM8996_AIF2TX_CHAN1_SLOTS_MASK | + WM8996_AIF2TX_CHAN1_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); + + /* If the TX LRCLK pins are not in LRCLK mode configure the + * AIFs to source their clocks from the RX LRCLKs. + */ + ret = regmap_read(wm8996->regmap, WM8996_GPIO_1, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read GPIO1: %d\n", ret); + goto err_regmap; + } + + if (reg & WM8996_GP1_FN_MASK) + regmap_update_bits(wm8996->regmap, WM8996_AIF1_TX_LRCLK_2, + WM8996_AIF1TX_LRCLK_MODE, + WM8996_AIF1TX_LRCLK_MODE); + + ret = regmap_read(wm8996->regmap, WM8996_GPIO_2, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read GPIO2: %d\n", ret); + goto err_regmap; + } + + if (reg & WM8996_GP2_FN_MASK) + regmap_update_bits(wm8996->regmap, WM8996_AIF2_TX_LRCLK_2, + WM8996_AIF2TX_LRCLK_MODE, + WM8996_AIF2TX_LRCLK_MODE); + wm8996_init_gpio(wm8996); ret = snd_soc_register_codec(&i2c->dev, @@ -3225,7 +3065,6 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, err_gpiolib: wm8996_free_gpio(wm8996); err_regmap: - regmap_exit(wm8996->regmap); err_enable: if (wm8996->pdata.ldo_ena > 0) gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); @@ -3241,14 +3080,18 @@ err: static __devexit int wm8996_i2c_remove(struct i2c_client *client) { struct wm8996_priv *wm8996 = i2c_get_clientdata(client); + int i; snd_soc_unregister_codec(&client->dev); wm8996_free_gpio(wm8996); - regmap_exit(wm8996->regmap); if (wm8996->pdata.ldo_ena > 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); gpio_free(wm8996->pdata.ldo_ena); } + for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) + regulator_unregister_notifier(wm8996->supplies[i].consumer, + &wm8996->disable_nb[i]); + return 0; } diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 9328270df16c..2de74e1ea225 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -3,7 +3,7 @@ * * Author: Mark Brown * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4b263b6edf13..2c2346fdd637 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -1,7 +1,7 @@ /* * ALSA SoC WM9090 driver * - * Copyright 2009, 2010 Wolfson Microelectronics + * Copyright 2009-12 Wolfson Microelectronics * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index a1541414d904..099e6ec32125 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -1,7 +1,7 @@ /* * wm9712.c -- ALSA Soc WM9712 codec support * - * Copyright 2006 Wolfson Microelectronics PLC. + * Copyright 2006-12 Wolfson Microelectronics PLC. * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2d22cc70d536..3eb19fb71d17 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1,7 +1,7 @@ /* * wm9713.c -- ALSA Soc WM9713 codec support * - * Copyright 2006 Wolfson Microelectronics PLC. + * Copyright 2006-10 Wolfson Microelectronics PLC. * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index dfe957a47f29..61baa48823cb 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1,7 +1,7 @@ /* * wm_hubs.c -- WM8993/4 common code * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig new file mode 100644 index 000000000000..e334900cf0b8 --- /dev/null +++ b/sound/soc/dwc/Kconfig @@ -0,0 +1,9 @@ +config SND_DESIGNWARE_I2S + tristate "Synopsys I2S Device Driver" + depends on CLKDEV_LOOKUP + help + Say Y or M if you want to add support for I2S driver for + Synopsys desigwnware I2S device. The device supports upto + maximum of 8 channels each for play and record. + + diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile new file mode 100644 index 000000000000..319371f690f4 --- /dev/null +++ b/sound/soc/dwc/Makefile @@ -0,0 +1,3 @@ +# SYNOPSYS Platform Support +obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o + diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c new file mode 100644 index 000000000000..1aa51300c564 --- /dev/null +++ b/sound/soc/dwc/designware_i2s.c @@ -0,0 +1,455 @@ +/* + * ALSA SoC Synopsys I2S Audio Layer + * + * sound/soc/spear/designware_i2s.c + * + * Copyright (C) 2010 ST Microelectronics + * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <sound/designware_i2s.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +/* common register for all channel */ +#define IER 0x000 +#define IRER 0x004 +#define ITER 0x008 +#define CER 0x00C +#define CCR 0x010 +#define RXFFR 0x014 +#define TXFFR 0x018 + +/* I2STxRxRegisters for all channels */ +#define LRBR_LTHR(x) (0x40 * x + 0x020) +#define RRBR_RTHR(x) (0x40 * x + 0x024) +#define RER(x) (0x40 * x + 0x028) +#define TER(x) (0x40 * x + 0x02C) +#define RCR(x) (0x40 * x + 0x030) +#define TCR(x) (0x40 * x + 0x034) +#define ISR(x) (0x40 * x + 0x038) +#define IMR(x) (0x40 * x + 0x03C) +#define ROR(x) (0x40 * x + 0x040) +#define TOR(x) (0x40 * x + 0x044) +#define RFCR(x) (0x40 * x + 0x048) +#define TFCR(x) (0x40 * x + 0x04C) +#define RFF(x) (0x40 * x + 0x050) +#define TFF(x) (0x40 * x + 0x054) + +/* I2SCOMPRegisters */ +#define I2S_COMP_PARAM_2 0x01F0 +#define I2S_COMP_PARAM_1 0x01F4 +#define I2S_COMP_VERSION 0x01F8 +#define I2S_COMP_TYPE 0x01FC + +#define MAX_CHANNEL_NUM 8 +#define MIN_CHANNEL_NUM 2 + +struct dw_i2s_dev { + void __iomem *i2s_base; + struct clk *clk; + int active; + unsigned int capability; + struct device *dev; + + /* data related to DMA transfers b/w i2s and DMAC */ + struct i2s_dma_data play_dma_data; + struct i2s_dma_data capture_dma_data; + struct i2s_clk_config_data config; + int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); +}; + +static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val) +{ + writel(val, io_base + reg); +} + +static inline u32 i2s_read_reg(void __iomem *io_base, int reg) +{ + return readl(io_base + reg); +} + +static inline void i2s_disable_channels(struct dw_i2s_dev *dev, u32 stream) +{ + u32 i = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, TER(i), 0); + } else { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, RER(i), 0); + } +} + +static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) +{ + u32 i = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, TOR(i), 0); + } else { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, ROR(i), 0); + } +} + +static void i2s_start(struct dw_i2s_dev *dev, + struct snd_pcm_substream *substream) +{ + + i2s_write_reg(dev->i2s_base, IER, 1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s_write_reg(dev->i2s_base, ITER, 1); + else + i2s_write_reg(dev->i2s_base, IRER, 1); + + i2s_write_reg(dev->i2s_base, CER, 1); +} + +static void i2s_stop(struct dw_i2s_dev *dev, + struct snd_pcm_substream *substream) +{ + u32 i = 0, irq; + + i2s_clear_irqs(dev, substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, ITER, 0); + + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x30); + } + } else { + i2s_write_reg(dev->i2s_base, IRER, 0); + + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x03); + } + } + + if (!dev->active) { + i2s_write_reg(dev->i2s_base, CER, 0); + i2s_write_reg(dev->i2s_base, IER, 0); + } +} + +static int dw_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + struct i2s_dma_data *dma_data = NULL; + + if (!(dev->capability & DWC_I2S_RECORD) && + (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) + return -EINVAL; + + if (!(dev->capability & DWC_I2S_PLAY) && + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &dev->play_dma_data; + else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + dma_data = &dev->capture_dma_data; + + snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data); + + return 0; +} + +static int dw_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + struct i2s_clk_config_data *config = &dev->config; + u32 ccr, xfer_resolution, ch_reg, irq; + int ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + config->data_width = 16; + ccr = 0x00; + xfer_resolution = 0x02; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + config->data_width = 24; + ccr = 0x08; + xfer_resolution = 0x04; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + config->data_width = 32; + ccr = 0x10; + xfer_resolution = 0x05; + break; + + default: + dev_err(dev->dev, "designware-i2s: unsuppted PCM fmt"); + return -EINVAL; + } + + config->chan_nr = params_channels(params); + + switch (config->chan_nr) { + case EIGHT_CHANNEL_SUPPORT: + ch_reg = 3; + case SIX_CHANNEL_SUPPORT: + ch_reg = 2; + case FOUR_CHANNEL_SUPPORT: + ch_reg = 1; + case TWO_CHANNEL_SUPPORT: + ch_reg = 0; + break; + default: + dev_err(dev->dev, "channel not supported\n"); + } + + i2s_disable_channels(dev, substream->stream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); + i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + } else { + i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); + i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + } + + i2s_write_reg(dev->i2s_base, CCR, ccr); + + config->sample_rate = params_rate(params); + + if (!dev->i2s_clk_cfg) + return -EINVAL; + + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + + return 0; +} + +static void dw_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int dw_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dev->active++; + i2s_start(dev, substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev->active--; + i2s_stop(dev, substream); + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +static struct snd_soc_dai_ops dw_i2s_dai_ops = { + .startup = dw_i2s_startup, + .shutdown = dw_i2s_shutdown, + .hw_params = dw_i2s_hw_params, + .trigger = dw_i2s_trigger, +}; + +#ifdef CONFIG_PM + +static int dw_i2s_suspend(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + clk_disable(dev->clk); + return 0; +} + +static int dw_i2s_resume(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + clk_enable(dev->clk); + return 0; +} + +#else +#define dw_i2s_suspend NULL +#define dw_i2s_resume NULL +#endif + +static int dw_i2s_probe(struct platform_device *pdev) +{ + const struct i2s_platform_data *pdata = pdev->dev.platform_data; + struct dw_i2s_dev *dev; + struct resource *res; + int ret; + unsigned int cap; + struct snd_soc_dai_driver *dw_i2s_dai; + + if (!pdata) { + dev_err(&pdev->dev, "Invalid platform data\n"); + return -EINVAL; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no i2s resource defined\n"); + return -ENODEV; + } + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_err(&pdev->dev, "i2s region already claimed\n"); + return -EBUSY; + } + + dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); + if (!dev) { + dev_warn(&pdev->dev, "kzalloc fail\n"); + return -ENOMEM; + } + + dev->i2s_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!dev->i2s_base) { + dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + return -ENOMEM; + } + + cap = pdata->cap; + dev->capability = cap; + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + + /* Set DMA slaves info */ + + dev->play_dma_data.data = pdata->play_dma_data; + dev->capture_dma_data.data = pdata->capture_dma_data; + dev->play_dma_data.addr = res->start + I2S_TXDMA; + dev->capture_dma_data.addr = res->start + I2S_RXDMA; + dev->play_dma_data.max_burst = 16; + dev->capture_dma_data.max_burst = 16; + dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->play_dma_data.filter = pdata->filter; + dev->capture_dma_data.filter = pdata->filter; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + ret = clk_enable(dev->clk); + if (ret < 0) + goto err_clk_put; + + dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); + if (!dw_i2s_dai) { + dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); + ret = -ENOMEM; + goto err_clk_disable; + } + + if (cap & DWC_I2S_PLAY) { + dev_dbg(&pdev->dev, " SPEAr: play supported\n"); + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = pdata->channel; + dw_i2s_dai->playback.formats = pdata->snd_fmts; + dw_i2s_dai->playback.rates = pdata->snd_rates; + } + + if (cap & DWC_I2S_RECORD) { + dev_dbg(&pdev->dev, "SPEAr: record supported\n"); + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = pdata->channel; + dw_i2s_dai->capture.formats = pdata->snd_fmts; + dw_i2s_dai->capture.rates = pdata->snd_rates; + } + + dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->suspend = dw_i2s_suspend; + dw_i2s_dai->resume = dw_i2s_resume; + + dev->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, dev); + ret = snd_soc_register_dai(&pdev->dev, dw_i2s_dai); + if (ret != 0) { + dev_err(&pdev->dev, "not able to register dai\n"); + goto err_set_drvdata; + } + + return 0; + +err_set_drvdata: + dev_set_drvdata(&pdev->dev, NULL); +err_clk_disable: + clk_disable(dev->clk); +err_clk_put: + clk_put(dev->clk); + return ret; +} + +static int dw_i2s_remove(struct platform_device *pdev) +{ + struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + clk_put(dev->clk); + + return 0; +} + +static struct platform_driver dw_i2s_driver = { + .probe = dw_i2s_probe, + .remove = dw_i2s_remove, + .driver = { + .name = "designware-i2s", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(dw_i2s_driver); + +MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:designware_i2s"); diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 162dbb74f4cc..4eea98b42bc8 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -136,7 +136,7 @@ static struct snd_pcm_ops ep93xx_pcm_ops = { .hw_params = ep93xx_pcm_hw_params, .hw_free = ep93xx_pcm_hw_free, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = ep93xx_pcm_mmap, }; diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 080327414c6b..e7c800ebbd75 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -156,7 +156,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { + for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index 04ebbab8d7b9..b8ff44b9dafa 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -14,6 +14,7 @@ #define MX31_AUDMUX_PORT4_SSI_PINS_4 3 #define MX31_AUDMUX_PORT5_SSI_PINS_5 4 #define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 #define MX51_AUDMUX_PORT1_SSI0 0 #define MX51_AUDMUX_PORT2_SSI1 1 diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index f59c34943662..549b31fdc9dd 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -111,22 +111,39 @@ static int __devinit imx_mc13783_probe(struct platform_device *pdev) return ret; } - imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4, - IMX_AUDMUX_V2_PTCR_SYN, - IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) | - IMX_AUDMUX_V2_PDCR_MODE(1) | - IMX_AUDMUX_V2_PDCR_INMMASK(0xfc)); - imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | - IMX_AUDMUX_V2_PTCR_TCLKDIR | - IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | - IMX_AUDMUX_V2_PTCR_RFSDIR | - IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | - IMX_AUDMUX_V2_PTCR_RCLKDIR | - IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4), - IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4)); + if (machine_is_mx31_3ds()) { + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) | + IMX_AUDMUX_V2_PDCR_MODE(1) | + IMX_AUDMUX_V2_PDCR_INMMASK(0xfc)); + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4), + IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4)); + } else if (machine_is_mx27_3ds()) { + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_TFSDIR | + IMX_AUDMUX_V1_PCR_TCLKDIR | + IMX_AUDMUX_V1_PCR_RFSDIR | + IMX_AUDMUX_V1_PCR_RCLKDIR | + IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) + ); + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); + } return ret; } diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index f3c0a5ef35c8..48f9d886f020 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -141,7 +141,7 @@ static struct snd_pcm_ops imx_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_imx_pcm_hw_params, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = snd_imx_pcm_mmap, }; diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 3a729caeb8c8..fb21b17f17f5 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -95,8 +95,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) return ret; } imx_audmux_v2_configure_port(ext_port, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TCSEL(int_port), + IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { dev_err(&pdev->dev, "audmux external port setup failed\n"); diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 373dec90579f..f82d766cbf9e 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -141,7 +141,7 @@ static struct snd_pcm_ops mxs_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_mxs_pcm_hw_params, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = snd_mxs_pcm_mmap, }; diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 3e6e8764b2e6..215113b05f7d 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -133,7 +133,7 @@ static int __devinit mxs_sgtl5000_probe_dt(struct platform_device *pdev) mxs_sgtl5000_dai[i].codec_name = NULL; mxs_sgtl5000_dai[i].codec_of_node = codec_np; mxs_sgtl5000_dai[i].cpu_dai_name = NULL; - mxs_sgtl5000_dai[i].cpu_dai_of_node = saif_np[i]; + mxs_sgtl5000_dai[i].cpu_of_node = saif_np[i]; mxs_sgtl5000_dai[i].platform_name = NULL; mxs_sgtl5000_dai[i].platform_of_node = saif_np[i]; } diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 59d47ab5b15d..2c66e2498a45 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -527,6 +527,7 @@ static struct platform_driver asoc_mcpdm_driver = { module_platform_driver(asoc_mcpdm_driver); +MODULE_ALIAS("platform:omap-mcpdm"); MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>"); MODULE_DESCRIPTION("OMAP PDM SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index a0f7d3cfa470..4d2e46fae77c 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -8,6 +8,15 @@ config SND_PXA2XX_SOC the PXA2xx AC97, I2S or SSP interface. You will also need to select the audio interfaces to support below. +config SND_MMP_SOC + bool "Soc Audio for Marvell MMP chips" + depends on ARCH_MMP + select SND_SOC_DMAENGINE_PCM + select SND_ARM + help + Say Y if you want to add support for codecs attached to + the MMP SSPA interface. + config SND_PXA2XX_AC97 tristate select SND_AC97_CODEC @@ -26,6 +35,9 @@ config SND_PXA_SOC_SSP tristate select PXA_SSP +config SND_MMP_SOC_SSPA + tristate + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx @@ -138,6 +150,26 @@ config SND_SOC_TAVOREVB3 Say Y if you want to add support for SoC audio on the Marvell Saarb reference platform. +config SND_PXA910_SOC + tristate "SoC Audio for Marvell PXA910 chip" + depends on ARCH_MMP && SND + select SND_PCM + help + Say Y if you want to add support for SoC audio on the + Marvell PXA910 reference platform. + +config SND_SOC_TTC_DKB + bool "SoC Audio support for TTC DKB" + depends on SND_PXA910_SOC && MACH_TTC_DKB + select PXA_SSP + select SND_PXA_SOC_SSP + select SND_MMP_SOC + select MFD_88PM860X + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on TTC DKB + + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE @@ -194,3 +226,13 @@ config SND_PXA2XX_SOC_IMOTE2 help Say Y if you want to add support for SoC audio on the IMote 2. + +config SND_MMP_SOC_BROWNSTONE + tristate "SoC Audio support for Marvell Brownstone" + depends on SND_MMP_SOC && MACH_BROWNSTONE + select SND_MMP_SOC_SSPA + select MFD_WM8994 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on the + Marvell Brownstone reference platform. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index af357623be9d..d8a265d2d5d7 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -3,11 +3,15 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa-ssp-objs := pxa-ssp.o +snd-soc-mmp-objs := mmp-pcm.o +snd-soc-mmp-sspa-objs := mmp-sspa.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o +obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o +obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o # PXA Machine Support snd-soc-corgi-objs := corgi.o @@ -28,6 +32,8 @@ snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-z2-objs := z2.o snd-soc-imote2-objs := imote2.o snd-soc-raumfeld-objs := raumfeld.o +snd-soc-brownstone-objs := brownstone.o +snd-soc-ttc-dkb-objs := ttc-dkb.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -47,3 +53,5 @@ obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o +obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o +obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c new file mode 100644 index 000000000000..5e666e03d333 --- /dev/null +++ b/sound/soc/pxa/brownstone.c @@ -0,0 +1,174 @@ +/* + * linux/sound/soc/pxa/brownstone.c + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "../codecs/wm8994.h" +#include "mmp-sspa.h" + +static const struct snd_kcontrol_new brownstone_dapm_control[] = { + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), +}; + +static const struct snd_soc_dapm_route brownstone_audio_map[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + {"Ext Spk", NULL, "SPKOUTRP"}, + {"Ext Spk", NULL, "SPKOUTRN"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + + {"DMIC1DAT", NULL, "MICBIAS1"}, + {"MICBIAS1", NULL, "Main Mic"}, +}; + +static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Main Mic"); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(dapm, "HPOUT2P"); + snd_soc_dapm_nc_pin(dapm, "HPOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); + snd_soc_dapm_nc_pin(dapm, "IN1LN"); + snd_soc_dapm_nc_pin(dapm, "IN1LP"); + snd_soc_dapm_nc_pin(dapm, "IN1RP"); + snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(dapm, "IN2RN"); + snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(dapm, "IN2LN"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int freq_out, sspa_mclk, sysclk; + int sspa_div; + + if (params_rate(params) > 11025) { + freq_out = params_rate(params) * 512; + sysclk = params_rate(params) * 256; + sspa_mclk = params_rate(params) * 64; + } else { + freq_out = params_rate(params) * 1024; + sysclk = params_rate(params) * 512; + sspa_mclk = params_rate(params) * 64; + } + sspa_div = freq_out; + do_div(sspa_div, sspa_mclk); + + snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); + snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk); + snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk); + + /* set wm8994 sysclk */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0); + + return 0; +} + +/* machine stream operations */ +static struct snd_soc_ops brownstone_ops = { + .hw_params = brownstone_wm8994_hw_params, +}; + +static struct snd_soc_dai_link brownstone_wm8994_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_name = "mmp-sspa-dai.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "mmp-pcm-audio", + .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &brownstone_ops, + .init = brownstone_wm8994_init, +}, +}; + +/* audio machine driver */ +static struct snd_soc_card brownstone = { + .name = "brownstone", + .dai_link = brownstone_wm8994_dai, + .num_links = ARRAY_SIZE(brownstone_wm8994_dai), + + .controls = brownstone_dapm_control, + .num_controls = ARRAY_SIZE(brownstone_dapm_control), + .dapm_widgets = brownstone_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets), + .dapm_routes = brownstone_audio_map, + .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map), +}; + +static int __devinit brownstone_probe(struct platform_device *pdev) +{ + int ret; + + brownstone.dev = &pdev->dev; + ret = snd_soc_register_card(&brownstone); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; +} + +static int __devexit brownstone_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&brownstone); + return 0; +} + +static struct platform_driver mmp_driver = { + .driver = { + .name = "brownstone-audio", + .owner = THIS_MODULE, + }, + .probe = brownstone_probe, + .remove = __devexit_p(brownstone_remove), +}; + +module_platform_driver(mmp_driver); + +MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC Brownstone"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 9c585af59b5f..8687c1c65d29 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -186,36 +186,27 @@ static struct snd_soc_card mioa701 = { .num_links = ARRAY_SIZE(mioa701_dai), }; -static struct platform_device *mioa701_snd_device; - -static int mioa701_wm9713_probe(struct platform_device *pdev) +static int __devinit mioa701_wm9713_probe(struct platform_device *pdev) { - int ret; + int rc; if (!machine_is_mioa701()) return -ENODEV; - dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" - "lead to overheating and possible destruction of your device." - "Do not use without a good knowledge of mio's board design!\n"); - - mioa701_snd_device = platform_device_alloc("soc-audio", -1); - if (!mioa701_snd_device) - return -ENOMEM; - - platform_set_drvdata(mioa701_snd_device, &mioa701); - - ret = platform_device_add(mioa701_snd_device); - if (!ret) - return 0; - - platform_device_put(mioa701_snd_device); - return ret; + mioa701.dev = &pdev->dev; + rc = snd_soc_register_card(&mioa701); + if (!rc) + dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" + "lead to overheating and possible destruction of your device." + " Do not use without a good knowledge of mio's board design!\n"); + return rc; } static int __devexit mioa701_wm9713_remove(struct platform_device *pdev) { - platform_device_unregister(mioa701_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c new file mode 100644 index 000000000000..73ac5463c9e4 --- /dev/null +++ b/sound/soc/pxa/mmp-pcm.c @@ -0,0 +1,297 @@ +/* + * linux/sound/soc/pxa/mmp-pcm.c + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/platform_data/mmp_audio.h> +#include <sound/pxa2xx-lib.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <mach/sram.h> +#include <sound/dmaengine_pcm.h> + +struct mmp_dma_data { + int ssp_id; + struct resource *dma_res; +}; + +#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \ + SNDRV_PCM_INFO_MMAP_VALID | \ + SNDRV_PCM_INFO_INTERLEAVED | \ + SNDRV_PCM_INFO_PAUSE | \ + SNDRV_PCM_INFO_RESUME) + +#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_pcm_hardware mmp_pcm_hardware[] = { + { + .info = MMP_PCM_INFO, + .formats = MMP_PCM_FORMATS, + .period_bytes_min = 1024, + .period_bytes_max = 2048, + .periods_min = 2, + .periods_max = 32, + .buffer_bytes_max = 4096, + .fifo_size = 32, + }, + { + .info = MMP_PCM_INFO, + .formats = MMP_PCM_FORMATS, + .period_bytes_min = 1024, + .period_bytes_max = 2048, + .periods_min = 2, + .periods_max = 32, + .buffer_bytes_max = 4096, + .fifo_size = 32, + }, +}; + +static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct pxa2xx_pcm_dma_params *dma_params; + struct dma_slave_config slave_config; + int ret; + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; + + ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + if (ret) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.dst_addr = dma_params->dev_addr; + slave_config.dst_maxburst = 4; + } else { + slave_config.src_addr = dma_params->dev_addr; + slave_config.src_maxburst = 4; + } + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct mmp_dma_data *dma_data = param; + bool found = false; + char *devname; + + devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name, + dma_data->ssp_id); + if ((strcmp(dev_name(chan->device->dev), devname) == 0) && + (chan->chan_id == dma_data->dma_res->start)) { + found = true; + } + + kfree(devname); + return found; +} + +static int mmp_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct platform_device *pdev = to_platform_device(rtd->platform->dev); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct mmp_dma_data *dma_data; + struct resource *r; + int ret; + + r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream); + if (!r) + return -EBUSY; + + snd_soc_set_runtime_hwparams(substream, + &mmp_pcm_hardware[substream->stream]); + dma_data = devm_kzalloc(&pdev->dev, + sizeof(struct mmp_dma_data), GFP_KERNEL); + if (dma_data == NULL) + return -ENOMEM; + + dma_data->dma_res = r; + dma_data->ssp_id = cpu_dai->id; + + ret = snd_dmaengine_pcm_open(substream, filter, dma_data); + if (ret) { + devm_kfree(&pdev->dev, dma_data); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_data); + return 0; +} + +static int mmp_pcm_close(struct snd_pcm_substream *substream) +{ + struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct platform_device *pdev = to_platform_device(rtd->platform->dev); + + snd_dmaengine_pcm_close(substream); + devm_kfree(&pdev->dev, dma_data); + return 0; +} + +static int mmp_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long off = vma->vm_pgoff; + + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + return remap_pfn_range(vma, vma->vm_start, + __phys_to_pfn(runtime->dma_addr) + off, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops mmp_pcm_ops = { + .open = mmp_pcm_open, + .close = mmp_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = mmp_pcm_hw_params, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = mmp_pcm_mmap, +}; + +static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + struct gen_pool *gpool; + + gpool = sram_get_gpool("asram"); + if (!gpool) + return; + + for (stream = 0; stream < 2; stream++) { + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; + + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + gen_pool_free(gpool, (unsigned long)buf->area, size); + buf->area = NULL; + } + + return; +} + +static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, + int stream) +{ + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; + struct gen_pool *gpool; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = substream->pcm->card->dev; + buf->private_data = NULL; + + gpool = sram_get_gpool("asram"); + if (!gpool) + return -ENOMEM; + + buf->area = (unsigned char *)gen_pool_alloc(gpool, size); + if (!buf->area) + return -ENOMEM; + buf->addr = gen_pool_virt_to_phys(gpool, (unsigned long)buf->area); + buf->bytes = size; + return 0; +} + +int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm_substream *substream; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0, stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + + ret = mmp_pcm_preallocate_dma_buffer(substream, stream); + if (ret) + goto err; + } + + return 0; + +err: + mmp_pcm_free_dma_buffers(pcm); + return ret; +} + +struct snd_soc_platform_driver mmp_soc_platform = { + .ops = &mmp_pcm_ops, + .pcm_new = mmp_pcm_new, + .pcm_free = mmp_pcm_free_dma_buffers, +}; + +static __devinit int mmp_pcm_probe(struct platform_device *pdev) +{ + struct mmp_audio_platdata *pdata = pdev->dev.platform_data; + + if (pdata) { + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max = + pdata->buffer_max_playback; + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max = + pdata->period_max_playback; + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max = + pdata->buffer_max_capture; + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max = + pdata->period_max_capture; + } + return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform); +} + +static int __devexit mmp_pcm_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver mmp_pcm_driver = { + .driver = { + .name = "mmp-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = mmp_pcm_probe, + .remove = __devexit_p(mmp_pcm_remove), +}; + +module_platform_driver(mmp_pcm_driver); + +MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); +MODULE_DESCRIPTION("MMP Soc Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c new file mode 100644 index 000000000000..4d6cb8a30fc8 --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.c @@ -0,0 +1,480 @@ +/* + * linux/sound/soc/pxa/mmp-sspa.c + * Base on pxa2xx-ssp.c + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/slab.h> +#include <linux/pxa2xx_ssp.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> +#include "mmp-sspa.h" + +/* + * SSPA audio private data + */ +struct sspa_priv { + struct ssp_device *sspa; + struct pxa2xx_pcm_dma_params *dma_params; + struct clk *audio_clk; + struct clk *sysclk; + int dai_fmt; + int running_cnt; +}; + +static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val) +{ + __raw_writel(val, sspa->mmio_base + reg); +} + +static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg) +{ + return __raw_readl(sspa->mmio_base + reg); +} + +static void mmp_sspa_tx_enable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP); + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); +} + +static void mmp_sspa_tx_disable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP); + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); +} + +static void mmp_sspa_rx_enable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP); + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); +} + +static void mmp_sspa_rx_disable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP); + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); +} + +static int mmp_sspa_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai); + + clk_enable(priv->sysclk); + clk_enable(priv->sspa->clk); + + return 0; +} + +static void mmp_sspa_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable(priv->sspa->clk); + clk_disable(priv->sysclk); + + return; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (clk_id) { + case MMP_SSPA_CLK_AUDIO: + ret = clk_set_rate(priv->audio_clk, freq); + if (ret) + return ret; + break; + case MMP_SSPA_CLK_PLL: + case MMP_SSPA_CLK_VCXO: + /* not support yet */ + return -EINVAL; + default: + return -EINVAL; + } + + return 0; +} + +static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (pll_id) { + case MMP_SYSCLK: + ret = clk_set_rate(priv->sysclk, freq_out); + if (ret) + return ret; + break; + case MMP_SSPA_CLK: + ret = clk_set_rate(priv->sspa->clk, freq_out); + if (ret) + return ret; + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Set up the sspa dai format. The sspa port must be inactive + * before calling this function as the physical + * interface format is changed. + */ +static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *sspa = sspa_priv->sspa; + u32 sspa_sp, sspa_ctrl; + + /* check if we need to change anything at all */ + if (sspa_priv->dai_fmt == fmt) + return 0; + + /* we can only change the settings if the port is not in use */ + if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) || + (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) { + dev_err(&sspa->pdev->dev, + "can't change hardware dai format: stream is in use\n"); + return -EINVAL; + } + + /* reset port settings */ + sspa_sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH; + sspa_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + sspa_sp |= SSPA_SP_MSL; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspa_sp |= SSPA_SP_FSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sspa_sp |= SSPA_TXSP_FPER(63); + sspa_sp |= SSPA_SP_FWID(31); + sspa_ctrl |= SSPA_CTL_XDATDLY(1); + break; + default: + return -EINVAL; + } + + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); + + sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH); + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); + + /* + * FIXME: hw issue, for the tx serial port, + * can not config the master/slave mode; + * so must clean this bit. + * The master/slave mode has been set in the + * rx port. + */ + sspa_sp &= ~SSPA_SP_MSL; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + + mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + sspa_priv->dai_fmt = fmt; + return 0; +} + +/* + * Set the SSPA audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); + struct ssp_device *sspa = sspa_priv->sspa; + struct pxa2xx_pcm_dma_params *dma_params; + u32 sspa_ctrl; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL); + else + sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL); + + sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK; + sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1); + sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK; + sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS); + sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS); + break; + case SNDRV_PCM_FORMAT_S16_LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS); + break; + case SNDRV_PCM_FORMAT_S24_3LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS); + break; + case SNDRV_PCM_FORMAT_S32_LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS); + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1); + } else { + mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0); + } + + dma_params = &sspa_priv->dma_params[substream->stream]; + dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + (sspa->phys_base + SSPA_TXD) : + (sspa->phys_base + SSPA_RXD); + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params); + return 0; +} + +static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); + struct ssp_device *sspa = sspa_priv->sspa; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * whatever playback or capture, must enable rx. + * this is a hw issue, so need check if rx has been + * enabled or not; if has been enabled by another + * stream, do not enable again. + */ + if (!sspa_priv->running_cnt) + mmp_sspa_rx_enable(sspa); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_enable(sspa); + + sspa_priv->running_cnt++; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sspa_priv->running_cnt--; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_disable(sspa); + + /* have no capture stream, disable rx port */ + if (!sspa_priv->running_cnt) + mmp_sspa_rx_disable(sspa); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static int mmp_sspa_probe(struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, priv); + return 0; + +} + +#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000 +#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops mmp_sspa_dai_ops = { + .startup = mmp_sspa_startup, + .shutdown = mmp_sspa_shutdown, + .trigger = mmp_sspa_trigger, + .hw_params = mmp_sspa_hw_params, + .set_sysclk = mmp_sspa_set_dai_sysclk, + .set_pll = mmp_sspa_set_dai_pll, + .set_fmt = mmp_sspa_set_dai_fmt, +}; + +struct snd_soc_dai_driver mmp_sspa_dai = { + .probe = mmp_sspa_probe, + .playback = { + .channels_min = 1, + .channels_max = 128, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .ops = &mmp_sspa_dai_ops, +}; + +static __devinit int asoc_mmp_sspa_probe(struct platform_device *pdev) +{ + struct sspa_priv *priv; + struct resource *res; + + priv = devm_kzalloc(&pdev->dev, + sizeof(struct sspa_priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->sspa = devm_kzalloc(&pdev->dev, + sizeof(struct ssp_device), GFP_KERNEL); + if (priv->sspa == NULL) + return -ENOMEM; + + priv->dma_params = devm_kzalloc(&pdev->dev, + 2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL); + if (priv->dma_params == NULL) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) + return -ENOMEM; + + priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res); + if (priv->sspa->mmio_base == NULL) + return -ENODEV; + + priv->sspa->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(priv->sspa->clk)) + return PTR_ERR(priv->sspa->clk); + + priv->audio_clk = clk_get(NULL, "mmp-audio"); + if (IS_ERR(priv->audio_clk)) + return PTR_ERR(priv->audio_clk); + + priv->sysclk = clk_get(NULL, "mmp-sysclk"); + if (IS_ERR(priv->sysclk)) { + clk_put(priv->audio_clk); + return PTR_ERR(priv->sysclk); + } + clk_enable(priv->audio_clk); + priv->dai_fmt = (unsigned int) -1; + platform_set_drvdata(pdev, priv); + + return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai); +} + +static int __devexit asoc_mmp_sspa_remove(struct platform_device *pdev) +{ + struct sspa_priv *priv = platform_get_drvdata(pdev); + + clk_disable(priv->audio_clk); + clk_put(priv->audio_clk); + clk_put(priv->sysclk); + snd_soc_unregister_dai(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_mmp_sspa_driver = { + .driver = { + .name = "mmp-sspa-dai", + .owner = THIS_MODULE, + }, + .probe = asoc_mmp_sspa_probe, + .remove = __devexit_p(asoc_mmp_sspa_remove), +}; + +module_platform_driver(asoc_mmp_sspa_driver); + +MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); +MODULE_DESCRIPTION("MMP SSPA SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h new file mode 100644 index 000000000000..ea365cb9e784 --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.h @@ -0,0 +1,92 @@ +/* + * linux/sound/soc/pxa/mmp-sspa.h + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef _MMP_SSPA_H +#define _MMP_SSPA_H + +/* + * SSPA Registers + */ +#define SSPA_RXD (0x00) +#define SSPA_RXID (0x04) +#define SSPA_RXCTL (0x08) +#define SSPA_RXSP (0x0c) +#define SSPA_RXFIFO_UL (0x10) +#define SSPA_RXINT_MASK (0x14) +#define SSPA_RXC (0x18) +#define SSPA_RXFIFO_NOFS (0x1c) +#define SSPA_RXFIFO_SIZE (0x20) + +#define SSPA_TXD (0x80) +#define SSPA_TXID (0x84) +#define SSPA_TXCTL (0x88) +#define SSPA_TXSP (0x8c) +#define SSPA_TXFIFO_LL (0x90) +#define SSPA_TXINT_MASK (0x94) +#define SSPA_TXC (0x98) +#define SSPA_TXFIFO_NOFS (0x9c) +#define SSPA_TXFIFO_SIZE (0xa0) + +/* SSPA Control Register */ +#define SSPA_CTL_XPH (1 << 31) /* Read Phase */ +#define SSPA_CTL_XFIG (1 << 15) /* Transmit Zeros when FIFO Empty */ +#define SSPA_CTL_JST (1 << 3) /* Audio Sample Justification */ +#define SSPA_CTL_XFRLEN2_MASK (7 << 24) +#define SSPA_CTL_XFRLEN2(x) ((x) << 24) /* Transmit Frame Length in Phase 2 */ +#define SSPA_CTL_XWDLEN2_MASK (7 << 21) +#define SSPA_CTL_XWDLEN2(x) ((x) << 21) /* Transmit Word Length in Phase 2 */ +#define SSPA_CTL_XDATDLY(x) ((x) << 19) /* Tansmit Data Delay */ +#define SSPA_CTL_XSSZ2_MASK (7 << 16) +#define SSPA_CTL_XSSZ2(x) ((x) << 16) /* Transmit Sample Audio Size */ +#define SSPA_CTL_XFRLEN1_MASK (7 << 8) +#define SSPA_CTL_XFRLEN1(x) ((x) << 8) /* Transmit Frame Length in Phase 1 */ +#define SSPA_CTL_XWDLEN1_MASK (7 << 5) +#define SSPA_CTL_XWDLEN1(x) ((x) << 5) /* Transmit Word Length in Phase 1 */ +#define SSPA_CTL_XSSZ1_MASK (7 << 0) +#define SSPA_CTL_XSSZ1(x) ((x) << 0) /* XSSZ1 */ + +#define SSPA_CTL_8_BITS (0x0) /* Sample Size */ +#define SSPA_CTL_12_BITS (0x1) +#define SSPA_CTL_16_BITS (0x2) +#define SSPA_CTL_20_BITS (0x3) +#define SSPA_CTL_24_BITS (0x4) +#define SSPA_CTL_32_BITS (0x5) + +/* SSPA Serial Port Register */ +#define SSPA_SP_WEN (1 << 31) /* Write Configuration Enable */ +#define SSPA_SP_MSL (1 << 18) /* Master Slave Configuration */ +#define SSPA_SP_CLKP (1 << 17) /* CLKP Polarity Clock Edge Select */ +#define SSPA_SP_FSP (1 << 16) /* FSP Polarity Clock Edge Select */ +#define SSPA_SP_FFLUSH (1 << 2) /* FIFO Flush */ +#define SSPA_SP_S_RST (1 << 1) /* Active High Reset Signal */ +#define SSPA_SP_S_EN (1 << 0) /* Serial Clock Domain Enable */ +#define SSPA_SP_FWID(x) ((x) << 20) /* Frame-Sync Width */ +#define SSPA_TXSP_FPER(x) ((x) << 4) /* Frame-Sync Active */ + +/* sspa clock sources */ +#define MMP_SSPA_CLK_PLL 0 +#define MMP_SSPA_CLK_VCXO 1 +#define MMP_SSPA_CLK_AUDIO 3 + +/* sspa pll id */ +#define MMP_SYSCLK 0 +#define MMP_SSPA_CLK 1 + +#endif /* _MMP_SSPA_H */ diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c new file mode 100644 index 000000000000..935491a8a770 --- /dev/null +++ b/sound/soc/pxa/ttc-dkb.c @@ -0,0 +1,173 @@ +/* + * linux/sound/soc/pxa/ttc_dkb.c + * + * Copyright (C) 2012 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <sound/pcm_params.h> +#include "../codecs/88pm860x-codec.h" + +static struct snd_soc_jack hs_jack, mic_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, +}; + +static struct snd_soc_jack_pin mic_jack_pins[] = { + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* ttc machine dapm widgets */ +static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Ext Speaker", NULL), + SND_SOC_DAPM_MIC("Ext Mic 1", NULL), + SND_SOC_DAPM_MIC("Headset Mic 2", NULL), + SND_SOC_DAPM_MIC("Ext Mic 3", NULL), +}; + +/* ttc machine audio map */ +static const struct snd_soc_dapm_route ttc_audio_map[] = { + {"Headset Stereophone", NULL, "HS1"}, + {"Headset Stereophone", NULL, "HS2"}, + + {"Ext Speaker", NULL, "LSP"}, + {"Ext Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Ext Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Headset Mic 2"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Ext Mic 3"}, +}; + +static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + /* connected pins */ + snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); + snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE + | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack); + snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack); + snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), + mic_jack_pins); + + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); + pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); + + return 0; +} + +/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = { +{ + .name = "88pm860x i2s", + .stream_name = "audio playback", + .codec_name = "88pm860x-codec", + .platform_name = "mmp-pcm-audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .init = ttc_pm860x_init, +}, +}; + +/* ttc/td audio machine driver */ +static struct snd_soc_card ttc_dkb_card = { + .name = "ttc-dkb-hifi", + .dai_link = ttc_pm860x_hifi_dai, + .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai), + + .dapm_widgets = ttc_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets), + .dapm_routes = ttc_audio_map, + .num_dapm_routes = ARRAY_SIZE(ttc_audio_map), +}; + +static int __devinit ttc_dkb_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &ttc_dkb_card; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + + return ret; +} + +static int __devexit ttc_dkb_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver ttc_dkb_driver = { + .driver = { + .name = "ttc-dkb-audio", + .owner = THIS_MODULE, + }, + .probe = ttc_dkb_probe, + .remove = __devexit_p(ttc_dkb_remove), +}; + +module_platform_driver(ttc_dkb_driver); + +/* Module information */ +MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC TTC DKB"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ttc-dkb-audio"); diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index c82c646b8a08..ee52c8a00779 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -211,6 +211,11 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("WM1250 Input"), + SOC_DAPM_PIN_SWITCH("WM1250 Output"), +}; + static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), @@ -282,6 +287,8 @@ static struct snd_soc_card littlemill = { .set_bias_level = littlemill_set_bias_level, .set_bias_level_post = littlemill_set_bias_level_post, + .controls = controls, + .num_controls = ARRAY_SIZE(controls), .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 79fbeea99d46..ac7701b3c5dc 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -25,7 +25,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/regs-gpio.h> #include <mach/dma.h> #include "dma.h" @@ -83,12 +82,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); return 0; } diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index c4aa4d412fbf..0aae3a3883dc 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -23,7 +23,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/regs-gpio.h> #include <mach/dma.h> #include <plat/regs-iis.h> @@ -391,12 +390,9 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) } clk_enable(s3c24xx_i2s.iis_clk); - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 8eb309f23d18..48dd4dd9ee08 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -149,31 +149,41 @@ static struct snd_soc_card smdk = { .num_links = ARRAY_SIZE(smdk_dai), }; -static struct platform_device *smdk_snd_device; -static int __init smdk_audio_init(void) +static int __devinit smdk_audio_probe(struct platform_device *pdev) { int ret; + struct snd_soc_card *card = &smdk; - smdk_snd_device = platform_device_alloc("soc-audio", -1); - if (!smdk_snd_device) - return -ENOMEM; + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); - platform_set_drvdata(smdk_snd_device, &smdk); - - ret = platform_device_add(smdk_snd_device); if (ret) - platform_device_put(smdk_snd_device); + dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); return ret; } -module_init(smdk_audio_init); -static void __exit smdk_audio_exit(void) +static int __devexit smdk_audio_remove(struct platform_device *pdev) { - platform_device_unregister(smdk_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; } -module_exit(smdk_audio_exit); + +static struct platform_driver smdk_audio_driver = { + .driver = { + .name = "smdk-audio", + .owner = THIS_MODULE, + }, + .probe = smdk_audio_probe, + .remove = __devexit_p(smdk_audio_remove), +}; + +module_platform_driver(smdk_audio_driver); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:smdk-audio"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 2ef98536f1da..53486ff9c2af 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -247,7 +247,7 @@ struct fsi_priv { struct fsi_stream_handler { int (*init)(struct fsi_priv *fsi, struct fsi_stream *io); int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io); - int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io); + int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, @@ -571,16 +571,16 @@ static int fsi_stream_transfer(struct fsi_stream *io) #define fsi_stream_stop(fsi, io)\ fsi_stream_handler_call(io, start_stop, fsi, io, 0) -static int fsi_stream_probe(struct fsi_priv *fsi) +static int fsi_stream_probe(struct fsi_priv *fsi, struct device *dev) { struct fsi_stream *io; int ret1, ret2; io = &fsi->playback; - ret1 = fsi_stream_handler_call(io, probe, fsi, io); + ret1 = fsi_stream_handler_call(io, probe, fsi, io, dev); io = &fsi->capture; - ret2 = fsi_stream_handler_call(io, probe, fsi, io); + ret2 = fsi_stream_handler_call(io, probe, fsi, io, dev); if (ret1 < 0) return ret1; @@ -1089,13 +1089,10 @@ static void fsi_dma_do_tasklet(unsigned long data) { struct fsi_stream *io = (struct fsi_stream *)data; struct fsi_priv *fsi = fsi_stream_to_priv(io); - struct dma_chan *chan; struct snd_soc_dai *dai; struct dma_async_tx_descriptor *desc; - struct scatterlist sg; struct snd_pcm_runtime *runtime; enum dma_data_direction dir; - dma_cookie_t cookie; int is_play = fsi_stream_is_play(fsi, io); int len; dma_addr_t buf; @@ -1104,7 +1101,6 @@ static void fsi_dma_do_tasklet(unsigned long data) return; dai = fsi_get_dai(io->substream); - chan = io->chan; runtime = io->substream->runtime; dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; len = samples_to_bytes(runtime, io->period_samples); @@ -1112,14 +1108,8 @@ static void fsi_dma_do_tasklet(unsigned long data) dma_sync_single_for_device(dai->dev, buf, len, dir); - sg_init_table(&sg, 1); - sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf)), - len , offset_in_page(buf)); - sg_dma_address(&sg) = buf; - sg_dma_len(&sg) = len; - - desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; @@ -1128,13 +1118,12 @@ static void fsi_dma_do_tasklet(unsigned long data) desc->callback = fsi_dma_complete; desc->callback_param = io; - cookie = desc->tx_submit(desc); - if (cookie < 0) { + if (dmaengine_submit(desc) < 0) { dev_err(dai->dev, "tx_submit() fail\n"); return; } - dma_async_issue_pending(chan); + dma_async_issue_pending(io->chan); /* * FIXME @@ -1184,7 +1173,7 @@ static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } -static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) +static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { dma_cap_mask_t mask; @@ -1192,8 +1181,19 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) dma_cap_set(DMA_SLAVE, mask); io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave); - if (!io->chan) - return -EIO; + if (!io->chan) { + + /* switch to PIO handler */ + if (fsi_stream_is_play(fsi, io)) + fsi->playback.handler = &fsi_pio_push_handler; + else + fsi->capture.handler = &fsi_pio_pop_handler; + + dev_info(dev, "switch handler (dma => pio)\n"); + + /* probe again */ + return fsi_stream_probe(fsi, dev); + } tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); @@ -1683,7 +1683,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.master = master; master->fsia.info = &info->port_a; fsi_handler_init(&master->fsia); - ret = fsi_stream_probe(&master->fsia); + ret = fsi_stream_probe(&master->fsia, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIA stream probe failed\n"); goto exit_iounmap; @@ -1694,7 +1694,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsib.master = master; master->fsib.info = &info->port_b; fsi_handler_init(&master->fsib); - ret = fsi_stream_probe(&master->fsib); + ret = fsi_stream_probe(&master->fsib, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIB stream probe failed\n"); goto exit_fsia; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b37ee8077ed1..f219b2f7ee68 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -812,13 +812,15 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* Find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { - if (dai_link->cpu_dai_of_node) { - if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) - continue; - } else { - if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; - } + if (dai_link->cpu_of_node && + (cpu_dai->dev->of_node != dai_link->cpu_of_node)) + continue; + if (dai_link->cpu_name && + strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name)) + continue; + if (dai_link->cpu_dai_name && + strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; rtd->cpu_dai = cpu_dai; } @@ -896,6 +898,28 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return 0; } +static int soc_remove_platform(struct snd_soc_platform *platform) +{ + int ret; + + if (platform->driver->remove) { + ret = platform->driver->remove(platform); + if (ret < 0) + pr_err("asoc: failed to remove %s: %d\n", + platform->name, ret); + } + + /* Make sure all DAPM widgets are freed */ + snd_soc_dapm_free(&platform->dapm); + + soc_cleanup_platform_debugfs(platform); + platform->probed = 0; + list_del(&platform->card_list); + module_put(platform->dev->driver->owner); + + return 0; +} + static void soc_remove_codec(struct snd_soc_codec *codec) { int err; @@ -917,11 +941,9 @@ static void soc_remove_codec(struct snd_soc_codec *codec) module_put(codec->dev->driver->owner); } -static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) +static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; int err; @@ -946,30 +968,6 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) list_del(&codec_dai->card_list); } - /* remove the platform */ - if (platform && platform->probed && - platform->driver->remove_order == order) { - if (platform->driver->remove) { - err = platform->driver->remove(platform); - if (err < 0) - pr_err("asoc: failed to remove %s: %d\n", - platform->name, err); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->dapm); - - soc_cleanup_platform_debugfs(platform); - platform->probed = 0; - list_del(&platform->card_list); - module_put(platform->dev->driver->owner); - } - - /* remove the CODEC */ - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); - /* remove the cpu_dai */ if (cpu_dai && cpu_dai->probed && cpu_dai->driver->remove_order == order) { @@ -981,7 +979,43 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) } cpu_dai->probed = 0; list_del(&cpu_dai->card_list); - module_put(cpu_dai->dev->driver->owner); + + if (!cpu_dai->codec) { + snd_soc_dapm_free(&cpu_dai->dapm); + module_put(cpu_dai->dev->driver->owner); + } + } +} + +static void soc_remove_link_components(struct snd_soc_card *card, int num, + int order) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_codec *codec; + + /* remove the platform */ + if (platform && platform->probed && + platform->driver->remove_order == order) { + soc_remove_platform(platform); + } + + /* remove the CODEC-side CODEC */ + if (codec_dai) { + codec = codec_dai->codec; + if (codec && codec->probed && + codec->driver->remove_order == order) + soc_remove_codec(codec); + } + + /* remove any CPU-side CODEC */ + if (cpu_dai) { + codec = cpu_dai->codec; + if (codec && codec->probed && + codec->driver->remove_order == order) + soc_remove_codec(codec); } } @@ -992,8 +1026,15 @@ static void soc_remove_dai_links(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { for (dai = 0; dai < card->num_rtd; dai++) - soc_remove_dai_link(card, dai, order); + soc_remove_link_dais(card, dai, order); } + + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (dai = 0; dai < card->num_rtd; dai++) + soc_remove_link_components(card, dai, order); + } + card->num_rtd = 0; } @@ -1054,6 +1095,10 @@ static int soc_probe_codec(struct snd_soc_card *card, } } + /* If the driver didn't set I/O up try regmap */ + if (!codec->control_data) + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (driver->controls) snd_soc_add_codec_controls(codec, driver->controls, driver->num_controls); @@ -1230,7 +1275,44 @@ out: return 0; } -static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) +static int soc_probe_link_components(struct snd_soc_card *card, int num, + int order) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + int ret; + + /* probe the CPU-side component, if it is a CODEC */ + if (cpu_dai->codec && + !cpu_dai->codec->probed && + cpu_dai->codec->driver->probe_order == order) { + ret = soc_probe_codec(card, cpu_dai->codec); + if (ret < 0) + return ret; + } + + /* probe the CODEC-side component */ + if (!codec_dai->codec->probed && + codec_dai->codec->driver->probe_order == order) { + ret = soc_probe_codec(card, codec_dai->codec); + if (ret < 0) + return ret; + } + + /* probe the platform */ + if (!platform->probed && + platform->driver->probe_order == order) { + ret = soc_probe_platform(card, platform); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; @@ -1255,11 +1337,14 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - cpu_dai->dapm.card = card; - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; + if (!cpu_dai->codec) { + cpu_dai->dapm.card = card; + if (!try_module_get(cpu_dai->dev->driver->owner)) + return -ENODEV; - snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + list_add(&cpu_dai->dapm.list, &card->dapm_list); + snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + } if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); @@ -1275,22 +1360,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) list_add(&cpu_dai->card_list, &card->dai_dev_list); } - /* probe the CODEC */ - if (!codec->probed && - codec->driver->probe_order == order) { - ret = soc_probe_codec(card, codec); - if (ret < 0) - return ret; - } - - /* probe the platform */ - if (!platform->probed && - platform->driver->probe_order == order) { - ret = soc_probe_platform(card, platform); - if (ret < 0) - return ret; - } - /* probe the CODEC DAI */ if (!codec_dai->probed && codec_dai->driver->probe_order == order) { if (codec_dai->driver->probe) { @@ -1565,14 +1634,27 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto card_probe_error; } - /* early DAI link probe */ + /* probe all components used by DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { for (i = 0; i < card->num_links; i++) { - ret = soc_probe_dai_link(card, i, order); + ret = soc_probe_link_components(card, i, order); if (ret < 0) { pr_err("asoc: failed to instantiate card %s: %d\n", - card->name, ret); + card->name, ret); + goto probe_dai_err; + } + } + } + + /* probe all DAI links on this card */ + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_link_dais(card, i, order); + if (ret < 0) { + pr_err("asoc: failed to instantiate card %s: %d\n", + card->name, ret); goto probe_dai_err; } } @@ -2790,6 +2872,104 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); /** + * snd_soc_info_volsw_range - single mixer info callback with range. + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information, within a range, about a single + * mixer control. + * + * returns 0 for success. + */ +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + int min = mc->min; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); + +/** + * snd_soc_put_volsw_range - single mixer put value callback with range. + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value, within a range, for a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val, val_mask; + + val = ((ucontrol->value.integer.value[0] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + return snd_soc_update_bits_locked(codec, reg, val_mask, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); + +/** + * snd_soc_get_volsw_range - single mixer get callback with range + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value, within a range, of a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + + ucontrol->value.integer.value[0] = + (snd_soc_read(codec, reg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[0] = + max - ucontrol->value.integer.value[0]; + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[0] - min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); + +/** * snd_soc_limit_volume - Set new limit to an existing volume control. * * @codec: where to look for the control @@ -3346,6 +3526,12 @@ int snd_soc_register_card(struct snd_soc_card *card) link->name); return -EINVAL; } + /* Codec DAI name must be specified */ + if (!link->codec_dai_name) { + dev_err(card->dev, "codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } /* * Platform may be specified by either name or OF node, but @@ -3358,12 +3544,24 @@ int snd_soc_register_card(struct snd_soc_card *card) } /* - * CPU DAI must be specified by 1 of name or OF node, - * not both or neither. + * CPU device may be specified by either name or OF node, but + * can be left unspecified, and will be matched based on DAI + * name alone.. + */ + if (link->cpu_name && link->cpu_of_node) { + dev_err(card->dev, + "Neither/both cpu name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* + * At least one of CPU DAI name or CPU device name/node must be + * specified */ - if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { + if (!link->cpu_dai_name && + !(link->cpu_name || link->cpu_of_node)) { dev_err(card->dev, - "Neither/both cpu_dai name/of_node are set for %s\n", + "Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", link->name); return -EINVAL; } @@ -3938,6 +4136,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); + kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -3946,6 +4145,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); + kfree(routes); return -EINVAL; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 89eae93445cf..4d181df95dc3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -35,6 +35,7 @@ #include <linux/debugfs.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/clk.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -51,6 +52,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, + [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai] = 3, @@ -92,6 +94,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_aif_out] = 10, [snd_soc_dapm_dai] = 10, [snd_soc_dapm_dai_link] = 11, + [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, [snd_soc_dapm_supply] = 12, [snd_soc_dapm_post] = 13, @@ -288,9 +291,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, if (dapm->codec->driver->set_bias_level) ret = dapm->codec->driver->set_bias_level(dapm->codec, level); - else - dapm->bias_level = level; - } + } else + dapm->bias_level = level; + if (ret != 0) goto out; @@ -321,11 +324,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, val = soc_widget_read(w, reg); val = (val >> shift) & mask; + if (invert) + val = max - val; - if ((invert && !val) || (!invert && val)) - p->connect = 1; - else - p->connect = 0; + p->connect = !!val; } break; case snd_soc_dapm_mux: { @@ -391,6 +393,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_vmid: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -764,6 +767,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -850,6 +854,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -996,6 +1001,27 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_regulator_event); +/* + * Handler for clock supply widget. + */ +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (!w->clk) + return -EIO; + +#ifdef CONFIG_HAVE_CLK + if (SND_SOC_DAPM_EVENT_ON(event)) { + return clk_enable(w->clk); + } else { + clk_disable(w->clk); + return 0; + } +#endif + return 0; +} +EXPORT_SYMBOL_GPL(dapm_clock_event); + static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) { if (w->power_checked) @@ -1487,6 +1513,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, switch (w->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -1570,7 +1597,15 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } list_for_each_entry(w, &card->widgets, list) { - list_del_init(&w->dirty); + switch (w->id) { + case snd_soc_dapm_pre: + case snd_soc_dapm_post: + /* These widgets always need to be powered */ + break; + default: + list_del_init(&w->dirty); + break; + } if (w->power) { d = w->dapm; @@ -1587,6 +1622,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1941,6 +1977,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -2187,6 +2224,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_post: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -2221,6 +2259,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, path->connect = 0; return 0; } + + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + return 0; err: @@ -2230,6 +2272,59 @@ err: return ret; } +static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route) +{ + struct snd_soc_dapm_path *path, *p; + const char *sink; + const char *source; + char prefixed_sink[80]; + char prefixed_source[80]; + + if (route->control) { + dev_err(dapm->dev, + "Removal of routes with controls not supported\n"); + return -EINVAL; + } + + if (dapm->codec && dapm->codec->name_prefix) { + snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", + dapm->codec->name_prefix, route->sink); + sink = prefixed_sink; + snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", + dapm->codec->name_prefix, route->source); + source = prefixed_source; + } else { + sink = route->sink; + source = route->source; + } + + path = NULL; + list_for_each_entry(p, &dapm->card->paths, list) { + if (strcmp(p->source->name, source) != 0) + continue; + if (strcmp(p->sink->name, sink) != 0) + continue; + path = p; + break; + } + + if (path) { + dapm_mark_dirty(path->source, "Route removed"); + dapm_mark_dirty(path->sink, "Route removed"); + + list_del(&path->list); + list_del(&path->list_sink); + list_del(&path->list_source); + kfree(path); + } else { + dev_warn(dapm->dev, "Route %s->%s does not exist\n", + source, sink); + } + + return 0; +} + /** * snd_soc_dapm_add_routes - Add routes between DAPM widgets * @dapm: DAPM context @@ -2246,15 +2341,15 @@ err: int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret = 0; + int i, r, ret = 0; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(dapm, route); - if (ret < 0) { + r = snd_soc_dapm_add_route(dapm, route); + if (r < 0) { dev_err(dapm->dev, "Failed to add route %s->%s\n", route->source, route->sink); - break; + ret = r; } route++; } @@ -2264,6 +2359,30 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); +/** + * snd_soc_dapm_del_routes - Remove routes between DAPM widgets + * @dapm: DAPM context + * @route: audio routes + * @num: number of routes + * + * Removes routes from the DAPM context. + */ +int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route, int num) +{ + int i, ret = 0; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + for (i = 0; i < num; i++) { + snd_soc_dapm_del_route(dapm, route); + route++; + } + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_del_routes); + static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route) { @@ -2434,23 +2553,20 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; int max = mc->max; - unsigned int invert = mc->invert; unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + + if (snd_soc_volsw_is_stereo(mc)) + dev_warn(widget->dapm->dev, + "Control '%s' is stereo, which is not supported\n", + kcontrol->id.name); ucontrol->value.integer.value[0] = (snd_soc_read(widget->codec, reg) >> shift) & mask; - if (shift != rshift) - ucontrol->value.integer.value[1] = - (snd_soc_read(widget->codec, reg) >> rshift) & mask; - if (invert) { + if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; - if (shift != rshift) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - } return 0; } @@ -2484,20 +2600,19 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_update update; int wi; + if (snd_soc_volsw_is_stereo(mc)) + dev_warn(widget->dapm->dev, + "Control '%s' is stereo, which is not supported\n", + kcontrol->id.name); + val = (ucontrol->value.integer.value[0] & mask); + connect = !!val; if (invert) val = max - val; mask = mask << shift; val = val << shift; - if (val) - /* new connection */ - connect = invert ? 0 : 1; - else - /* old connection must be powered down */ - connect = invert ? 1 : 0; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, reg, mask, val); @@ -2873,6 +2988,19 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, return NULL; } break; + case snd_soc_dapm_clock_supply: +#ifdef CONFIG_CLKDEV_LOOKUP + w->clk = devm_clk_get(dapm->dev, w->name); + if (IS_ERR(w->clk)) { + ret = PTR_ERR(w->clk); + dev_err(dapm->dev, "Failed to request %s: %d\n", + w->name, ret); + return NULL; + } +#else + return NULL; +#endif + break; default: break; } @@ -2924,6 +3052,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; case snd_soc_dapm_dai: @@ -3538,10 +3667,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_free); static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) { + struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; LIST_HEAD(down_list); int powerdown = 0; + mutex_lock(&card->dapm_mutex); + list_for_each_entry(w, &dapm->card->widgets, list) { if (w->dapm != dapm) continue; @@ -3564,6 +3696,8 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); } + + mutex_unlock(&card->dapm_mutex); } /* diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 475695234b3d..5df529eda251 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -30,6 +30,7 @@ struct dmaengine_pcm_runtime_data { struct dma_chan *dma_chan; + dma_cookie_t cookie; unsigned int pos; @@ -153,7 +154,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) desc->callback = dmaengine_pcm_dma_complete; desc->callback_param = substream; - dmaengine_submit(desc); + prtd->cookie = dmaengine_submit(desc); return 0; } @@ -200,6 +201,20 @@ int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); /** + * snd_dmaengine_pcm_pointer_no_residue - dmaengine based PCM pointer implementation + * @substream: PCM substream + * + * This function is deprecated and should not be used by new drivers, as its + * results may be unreliable. + */ +snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + return bytes_to_frames(substream->runtime, prtd->pos); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); + +/** * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation * @substream: PCM substream * @@ -209,7 +224,19 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - return bytes_to_frames(substream->runtime, prtd->pos); + struct dma_tx_state state; + enum dma_status status; + unsigned int buf_size; + unsigned int pos = 0; + + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); + if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { + buf_size = snd_pcm_lib_buffer_bytes(substream); + if (state.residue > 0 && state.residue <= buf_size) + pos = buf_size - state.residue; + } + + return bytes_to_frames(substream->runtime, pos); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); @@ -243,7 +270,7 @@ static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd * Note that this function will use private_data field of the substream's * runtime. So it is not availabe to your pcm driver implementation. If you need * to keep additional data attached to a substream use - * snd_dmaeinge_pcm_{set,get}_data. + * snd_dmaengine_pcm_{set,get}_data. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 4d8dc6a27d4d..29183ef2b93d 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -142,11 +142,16 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ codec->using_regmap = true; - - ret = regmap_get_val_bytes(codec->control_data); - /* Errors are legitimate for non-integer byte multiples */ - if (ret > 0) - codec->val_bytes = ret; + if (!codec->control_data) + codec->control_data = dev_get_regmap(codec->dev, NULL); + + if (codec->control_data) { + ret = regmap_get_val_bytes(codec->control_data); + /* Errors are legitimate for non-integer byte + * multiples */ + if (ret > 0) + codec->val_bytes = ret; + } break; default: diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 48fd15b312c1..ef22d0bd9e9e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1955,10 +1955,8 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { - dev_warn(fe->dev, "asoc: %s no valid %s route\n", + dev_dbg(fe->dev, "asoc: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - mutex_unlock(&fe->card->mutex); - return -EINVAL; } /* calculate valid and active FE <-> BE dpcms */ @@ -2003,7 +2001,6 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; @@ -2042,7 +2039,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture, &pcm); } if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + dev_err(rtd->card->dev, "can't create pcm for %s\n", + rtd->dai_link->name); return ret; } dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name); @@ -2099,14 +2097,14 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); + dev_err(platform->dev, "pcm constructor failed\n"); return ret; } } pcm->private_free = platform->driver->pcm_free; out: - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c new file mode 100644 index 000000000000..c7c4b20395bb --- /dev/null +++ b/sound/soc/spear/spdif_in.c @@ -0,0 +1,297 @@ +/* + * ALSA SoC SPDIF In Audio Layer for spear processors + * + * Copyright (C) 2012 ST Microelectronics + * Vipin Kumar <vipin.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/ioport.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/spear_dma.h> +#include <sound/spear_spdif.h> +#include "spdif_in_regs.h" + +struct spdif_in_params { + u32 format; +}; + +struct spdif_in_dev { + struct clk *clk; + struct spear_dma_data dma_params; + struct spdif_in_params saved_params; + void *io_base; + struct device *dev; + void (*reset_perip)(void); + int irq; +}; + +static void spdif_in_configure(struct spdif_in_dev *host) +{ + u32 ctrl = SPDIF_IN_PRTYEN | SPDIF_IN_STATEN | SPDIF_IN_USREN | + SPDIF_IN_VALEN | SPDIF_IN_BLKEN; + ctrl |= SPDIF_MODE_16BIT | SPDIF_FIFO_THRES_16; + + writel(ctrl, host->io_base + SPDIF_IN_CTRL); + writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK); +} + +static int spdif_in_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); + return 0; +} + +static void spdif_in_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return; + + writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK); + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static void spdif_in_format(struct spdif_in_dev *host, u32 format) +{ + u32 ctrl = readl(host->io_base + SPDIF_IN_CTRL); + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl |= SPDIF_XTRACT_16BIT; + break; + + case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE: + ctrl &= ~SPDIF_XTRACT_16BIT; + break; + } + + writel(ctrl, host->io_base + SPDIF_IN_CTRL); +} + +static int spdif_in_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); + u32 format; + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + format = params_format(params); + host->saved_params.format = format; + + return 0; +} + +static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); + u32 ctrl; + int ret = 0; + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + clk_enable(host->clk); + spdif_in_configure(host); + spdif_in_format(host, host->saved_params.format); + + ctrl = readl(host->io_base + SPDIF_IN_CTRL); + ctrl |= SPDIF_IN_SAMPLE | SPDIF_IN_ENB; + writel(ctrl, host->io_base + SPDIF_IN_CTRL); + writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ctrl = readl(host->io_base + SPDIF_IN_CTRL); + ctrl &= ~(SPDIF_IN_SAMPLE | SPDIF_IN_ENB); + writel(ctrl, host->io_base + SPDIF_IN_CTRL); + writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK); + + if (host->reset_perip) + host->reset_perip(); + clk_disable(host->clk); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static struct snd_soc_dai_ops spdif_in_dai_ops = { + .startup = spdif_in_startup, + .shutdown = spdif_in_shutdown, + .trigger = spdif_in_trigger, + .hw_params = spdif_in_hw_params, +}; + +struct snd_soc_dai_driver spdif_in_dai = { + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000), + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE, + }, + .ops = &spdif_in_dai_ops, +}; + +static irqreturn_t spdif_in_irq(int irq, void *arg) +{ + struct spdif_in_dev *host = (struct spdif_in_dev *)arg; + + u32 irq_status = readl(host->io_base + SPDIF_IN_IRQ); + + if (!irq_status) + return IRQ_NONE; + + if (irq_status & SPDIF_IRQ_FIFOWRITE) + dev_err(host->dev, "spdif in: fifo write error"); + if (irq_status & SPDIF_IRQ_EMPTYFIFOREAD) + dev_err(host->dev, "spdif in: empty fifo read error"); + if (irq_status & SPDIF_IRQ_FIFOFULL) + dev_err(host->dev, "spdif in: fifo full error"); + if (irq_status & SPDIF_IRQ_OUTOFRANGE) + dev_err(host->dev, "spdif in: out of range error"); + + writel(0, host->io_base + SPDIF_IN_IRQ); + + return IRQ_HANDLED; +} + +static int spdif_in_probe(struct platform_device *pdev) +{ + struct spdif_in_dev *host; + struct spear_spdif_platform_data *pdata; + struct resource *res, *res_fifo; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -EINVAL; + + res_fifo = platform_get_resource(pdev, IORESOURCE_IO, 0); + if (!res_fifo) + return -EINVAL; + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_warn(&pdev->dev, "Failed to get memory resourse\n"); + return -ENOENT; + } + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) { + dev_warn(&pdev->dev, "kzalloc fail\n"); + return -ENOMEM; + } + + host->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!host->io_base) { + dev_warn(&pdev->dev, "ioremap failed\n"); + return -ENOMEM; + } + + host->irq = platform_get_irq(pdev, 0); + if (host->irq < 0) + return -EINVAL; + + host->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(host->clk)) + return PTR_ERR(host->clk); + + pdata = dev_get_platdata(&pdev->dev); + + if (!pdata) + return -EINVAL; + + host->dma_params.data = pdata->dma_params; + host->dma_params.addr = res_fifo->start; + host->dma_params.max_burst = 16; + host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + host->dma_params.filter = pdata->filter; + host->reset_perip = pdata->reset_perip; + + host->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, host); + + ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0, + "spdif-in", host); + if (ret) { + clk_put(host->clk); + dev_warn(&pdev->dev, "request_irq failed\n"); + return ret; + } + + ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai); + if (ret != 0) { + clk_put(host->clk); + return ret; + } + + return 0; +} + +static int spdif_in_remove(struct platform_device *pdev) +{ + struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + clk_put(host->clk); + + return 0; +} + + +static struct platform_driver spdif_in_driver = { + .probe = spdif_in_probe, + .remove = spdif_in_remove, + .driver = { + .name = "spdif-in", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(spdif_in_driver); + +MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>"); +MODULE_DESCRIPTION("SPEAr SPDIF IN SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spdif_in"); diff --git a/sound/soc/spear/spdif_in_regs.h b/sound/soc/spear/spdif_in_regs.h new file mode 100644 index 000000000000..37af7bc66b7f --- /dev/null +++ b/sound/soc/spear/spdif_in_regs.h @@ -0,0 +1,60 @@ +/* + * SPEAr SPDIF IN controller header file + * + * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef SPDIF_IN_REGS_H +#define SPDIF_IN_REGS_H + +#define SPDIF_IN_CTRL 0x00 + #define SPDIF_IN_PRTYEN (1 << 20) + #define SPDIF_IN_STATEN (1 << 19) + #define SPDIF_IN_USREN (1 << 18) + #define SPDIF_IN_VALEN (1 << 17) + #define SPDIF_IN_BLKEN (1 << 16) + + #define SPDIF_MODE_24BIT (8 << 12) + #define SPDIF_MODE_23BIT (7 << 12) + #define SPDIF_MODE_22BIT (6 << 12) + #define SPDIF_MODE_21BIT (5 << 12) + #define SPDIF_MODE_20BIT (4 << 12) + #define SPDIF_MODE_19BIT (3 << 12) + #define SPDIF_MODE_18BIT (2 << 12) + #define SPDIF_MODE_17BIT (1 << 12) + #define SPDIF_MODE_16BIT (0 << 12) + #define SPDIF_MODE_MASK (0x0F << 12) + + #define SPDIF_IN_VALID (1 << 11) + #define SPDIF_IN_SAMPLE (1 << 10) + #define SPDIF_DATA_SWAP (1 << 9) + #define SPDIF_IN_ENB (1 << 8) + #define SPDIF_DATA_REVERT (1 << 7) + #define SPDIF_XTRACT_16BIT (1 << 6) + #define SPDIF_FIFO_THRES_16 (16 << 0) + +#define SPDIF_IN_IRQ_MASK 0x04 +#define SPDIF_IN_IRQ 0x08 + #define SPDIF_IRQ_FIFOWRITE (1 << 0) + #define SPDIF_IRQ_EMPTYFIFOREAD (1 << 1) + #define SPDIF_IRQ_FIFOFULL (1 << 2) + #define SPDIF_IRQ_OUTOFRANGE (1 << 3) + +#define SPDIF_IN_STA 0x0C + #define SPDIF_IN_LOCK (0x1 << 0) + +#endif /* SPDIF_IN_REGS_H */ diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c new file mode 100644 index 000000000000..5eac4cda2fd7 --- /dev/null +++ b/sound/soc/spear/spdif_out.c @@ -0,0 +1,389 @@ +/* + * ALSA SoC SPDIF Out Audio Layer for spear processors + * + * Copyright (C) 2012 ST Microelectronics + * Vipin Kumar <vipin.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/ioport.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/spear_dma.h> +#include <sound/spear_spdif.h> +#include "spdif_out_regs.h" + +struct spdif_out_params { + u32 rate; + u32 core_freq; + u32 mute; +}; + +struct spdif_out_dev { + struct clk *clk; + struct spear_dma_data dma_params; + struct spdif_out_params saved_params; + u32 running; + void __iomem *io_base; +}; + +static void spdif_out_configure(struct spdif_out_dev *host) +{ + writel(SPDIF_OUT_RESET, host->io_base + SPDIF_OUT_SOFT_RST); + mdelay(1); + writel(readl(host->io_base + SPDIF_OUT_SOFT_RST) & ~SPDIF_OUT_RESET, + host->io_base + SPDIF_OUT_SOFT_RST); + + writel(SPDIF_OUT_FDMA_TRIG_16 | SPDIF_OUT_MEMFMT_16_16 | + SPDIF_OUT_VALID_HW | SPDIF_OUT_USER_HW | + SPDIF_OUT_CHNLSTA_HW | SPDIF_OUT_PARITY_HW, + host->io_base + SPDIF_OUT_CFG); + + writel(0x7F, host->io_base + SPDIF_OUT_INT_STA_CLR); + writel(0x7F, host->io_base + SPDIF_OUT_INT_EN_CLR); +} + +static int spdif_out_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + int ret; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); + + ret = clk_enable(host->clk); + if (ret) + return ret; + + host->running = true; + spdif_out_configure(host); + + return 0; +} + +static void spdif_out_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return; + + clk_disable(host->clk); + host->running = false; + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static void spdif_out_clock(struct spdif_out_dev *host, u32 core_freq, + u32 rate) +{ + u32 divider, ctrl; + + clk_set_rate(host->clk, core_freq); + divider = DIV_ROUND_CLOSEST(clk_get_rate(host->clk), (rate * 128)); + + ctrl = readl(host->io_base + SPDIF_OUT_CTRL); + ctrl &= ~SPDIF_DIVIDER_MASK; + ctrl |= (divider << SPDIF_DIVIDER_SHIFT) & SPDIF_DIVIDER_MASK; + writel(ctrl, host->io_base + SPDIF_OUT_CTRL); +} + +static int spdif_out_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + u32 rate, core_freq; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + rate = params_rate(params); + + switch (rate) { + case 8000: + case 16000: + case 32000: + case 64000: + /* + * The clock is multiplied by 10 to bring it to feasible range + * of frequencies for sscg + */ + core_freq = 64000 * 128 * 10; /* 81.92 MHz */ + break; + case 5512: + case 11025: + case 22050: + case 44100: + case 88200: + case 176400: + core_freq = 176400 * 128; /* 22.5792 MHz */ + break; + case 48000: + case 96000: + case 192000: + default: + core_freq = 192000 * 128; /* 24.576 MHz */ + break; + } + + spdif_out_clock(host, core_freq, rate); + host->saved_params.core_freq = core_freq; + host->saved_params.rate = rate; + + return 0; +} + +static int spdif_out_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + u32 ctrl; + int ret = 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ctrl = readl(host->io_base + SPDIF_OUT_CTRL); + ctrl &= ~SPDIF_OPMODE_MASK; + if (!host->saved_params.mute) + ctrl |= SPDIF_OPMODE_AUD_DATA | + SPDIF_STATE_NORMAL; + else + ctrl |= SPDIF_OPMODE_MUTE_PCM; + writel(ctrl, host->io_base + SPDIF_OUT_CTRL); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ctrl = readl(host->io_base + SPDIF_OUT_CTRL); + ctrl &= ~SPDIF_OPMODE_MASK; + ctrl |= SPDIF_OPMODE_OFF; + writel(ctrl, host->io_base + SPDIF_OUT_CTRL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int spdif_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + u32 val; + + host->saved_params.mute = mute; + val = readl(host->io_base + SPDIF_OUT_CTRL); + val &= ~SPDIF_OPMODE_MASK; + + if (mute) + val |= SPDIF_OPMODE_MUTE_PCM; + else { + if (host->running) + val |= SPDIF_OPMODE_AUD_DATA | SPDIF_STATE_NORMAL; + else + val |= SPDIF_OPMODE_OFF; + } + + writel(val, host->io_base + SPDIF_OUT_CTRL); + return 0; +} + +static int spdif_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = codec->card; + struct snd_soc_pcm_runtime *rtd = card->rtd; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + ucontrol->value.integer.value[0] = host->saved_params.mute; + return 0; +} + +static int spdif_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = codec->card; + struct snd_soc_pcm_runtime *rtd = card->rtd; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + if (host->saved_params.mute == ucontrol->value.integer.value[0]) + return 0; + + spdif_digital_mute(cpu_dai, ucontrol->value.integer.value[0]); + + return 1; +} +static const struct snd_kcontrol_new spdif_out_controls[] = { + SOC_SINGLE_BOOL_EXT("IEC958 Playback Switch", 0, + spdif_mute_get, spdif_mute_put), +}; + +int spdif_soc_dai_probe(struct snd_soc_dai *dai) +{ + return snd_soc_add_dai_controls(dai, spdif_out_controls, + ARRAY_SIZE(spdif_out_controls)); +} + +static const struct snd_soc_dai_ops spdif_out_dai_ops = { + .digital_mute = spdif_digital_mute, + .startup = spdif_out_startup, + .shutdown = spdif_out_shutdown, + .trigger = spdif_out_trigger, + .hw_params = spdif_out_hw_params, +}; + +static struct snd_soc_dai_driver spdif_out_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .probe = spdif_soc_dai_probe, + .ops = &spdif_out_dai_ops, +}; + +static int spdif_out_probe(struct platform_device *pdev) +{ + struct spdif_out_dev *host; + struct spear_spdif_platform_data *pdata; + struct resource *res; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -EINVAL; + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_warn(&pdev->dev, "Failed to get memory resourse\n"); + return -ENOENT; + } + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) { + dev_warn(&pdev->dev, "kzalloc fail\n"); + return -ENOMEM; + } + + host->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!host->io_base) { + dev_warn(&pdev->dev, "ioremap failed\n"); + return -ENOMEM; + } + + host->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(host->clk)) + return PTR_ERR(host->clk); + + pdata = dev_get_platdata(&pdev->dev); + + host->dma_params.data = pdata->dma_params; + host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA; + host->dma_params.max_burst = 16; + host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + host->dma_params.filter = pdata->filter; + + dev_set_drvdata(&pdev->dev, host); + + ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai); + if (ret != 0) { + clk_put(host->clk); + return ret; + } + + return 0; +} + +static int spdif_out_remove(struct platform_device *pdev) +{ + struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + clk_put(host->clk); + + return 0; +} + +#ifdef CONFIG_PM +static int spdif_out_suspend(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) + clk_disable(host->clk); + + return 0; +} + +static int spdif_out_resume(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) { + clk_enable(host->clk); + spdif_out_configure(host); + spdif_out_clock(host, host->saved_params.core_freq, + host->saved_params.rate); + } + return 0; +} + +static SIMPLE_DEV_PM_OPS(spdif_out_dev_pm_ops, spdif_out_suspend, \ + spdif_out_resume); + +#define SPDIF_OUT_DEV_PM_OPS (&spdif_out_dev_pm_ops) + +#else +#define SPDIF_OUT_DEV_PM_OPS NULL + +#endif + +static struct platform_driver spdif_out_driver = { + .probe = spdif_out_probe, + .remove = spdif_out_remove, + .driver = { + .name = "spdif-out", + .owner = THIS_MODULE, + .pm = SPDIF_OUT_DEV_PM_OPS, + }, +}; + +module_platform_driver(spdif_out_driver); + +MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>"); +MODULE_DESCRIPTION("SPEAr SPDIF OUT SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spdif_out"); diff --git a/sound/soc/spear/spdif_out_regs.h b/sound/soc/spear/spdif_out_regs.h new file mode 100644 index 000000000000..a5e53324b452 --- /dev/null +++ b/sound/soc/spear/spdif_out_regs.h @@ -0,0 +1,79 @@ +/* + * SPEAr SPDIF OUT controller header file + * + * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef SPDIF_OUT_REGS_H +#define SPDIF_OUT_REGS_H + +#define SPDIF_OUT_SOFT_RST 0x00 + #define SPDIF_OUT_RESET (1 << 0) +#define SPDIF_OUT_FIFO_DATA 0x04 +#define SPDIF_OUT_INT_STA 0x08 +#define SPDIF_OUT_INT_STA_CLR 0x0C + #define SPDIF_INT_UNDERFLOW (1 << 0) + #define SPDIF_INT_EODATA (1 << 1) + #define SPDIF_INT_EOBLOCK (1 << 2) + #define SPDIF_INT_EOLATENCY (1 << 3) + #define SPDIF_INT_EOPD_DATA (1 << 4) + #define SPDIF_INT_MEMFULLREAD (1 << 5) + #define SPDIF_INT_EOPD_PAUSE (1 << 6) + +#define SPDIF_OUT_INT_EN 0x10 +#define SPDIF_OUT_INT_EN_SET 0x14 +#define SPDIF_OUT_INT_EN_CLR 0x18 +#define SPDIF_OUT_CTRL 0x1C + #define SPDIF_OPMODE_MASK (7 << 0) + #define SPDIF_OPMODE_OFF (0 << 0) + #define SPDIF_OPMODE_MUTE_PCM (1 << 0) + #define SPDIF_OPMODE_MUTE_PAUSE (2 << 0) + #define SPDIF_OPMODE_AUD_DATA (3 << 0) + #define SPDIF_OPMODE_ENCODE (4 << 0) + #define SPDIF_STATE_NORMAL (1 << 3) + #define SPDIF_DIVIDER_MASK (0xff << 5) + #define SPDIF_DIVIDER_SHIFT (5) + #define SPDIF_SAMPLEREAD_MASK (0x1ffff << 15) + #define SPDIF_SAMPLEREAD_SHIFT (15) +#define SPDIF_OUT_STA 0x20 +#define SPDIF_OUT_PA_PB 0x24 +#define SPDIF_OUT_PC_PD 0x28 +#define SPDIF_OUT_CL1 0x2C +#define SPDIF_OUT_CR1 0x30 +#define SPDIF_OUT_CL2_CR2_UV 0x34 +#define SPDIF_OUT_PAUSE_LAT 0x38 +#define SPDIF_OUT_FRMLEN_BRST 0x3C +#define SPDIF_OUT_CFG 0x40 + #define SPDIF_OUT_MEMFMT_16_0 (0 << 5) + #define SPDIF_OUT_MEMFMT_16_16 (1 << 5) + #define SPDIF_OUT_VALID_DMA (0 << 3) + #define SPDIF_OUT_VALID_HW (1 << 3) + #define SPDIF_OUT_USER_DMA (0 << 2) + #define SPDIF_OUT_USER_HW (1 << 2) + #define SPDIF_OUT_CHNLSTA_DMA (0 << 1) + #define SPDIF_OUT_CHNLSTA_HW (1 << 1) + #define SPDIF_OUT_PARITY_HW (0 << 0) + #define SPDIF_OUT_PARITY_DMA (1 << 0) + #define SPDIF_OUT_FDMA_TRIG_2 (2 << 8) + #define SPDIF_OUT_FDMA_TRIG_6 (6 << 8) + #define SPDIF_OUT_FDMA_TRIG_8 (8 << 8) + #define SPDIF_OUT_FDMA_TRIG_10 (10 << 8) + #define SPDIF_OUT_FDMA_TRIG_12 (12 << 8) + #define SPDIF_OUT_FDMA_TRIG_16 (16 << 8) + #define SPDIF_OUT_FDMA_TRIG_18 (18 << 8) + +#endif /* SPDIF_OUT_REGS_H */ diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c new file mode 100644 index 000000000000..97c2cac8e92c --- /dev/null +++ b/sound/soc/spear/spear_pcm.c @@ -0,0 +1,214 @@ +/* + * ALSA PCM interface for ST SPEAr Processors + * + * sound/soc/spear/spear_pcm.c + * + * Copyright (C) 2012 ST Microelectronics + * Rajeev Kumar<rajeev-dlh.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/module.h> +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/scatterlist.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/spear_dma.h> + +struct snd_pcm_hardware spear_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + .buffer_bytes_max = 16 * 1024, /* max buffer size */ + .period_bytes_min = 2 * 1024, /* 1 msec data minimum period size */ + .period_bytes_max = 2 * 1024, /* maximum period size */ + .periods_min = 1, /* min # periods */ + .periods_max = 8, /* max # of periods */ + .fifo_size = 0, /* fifo size in bytes */ +}; + +static int spear_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int spear_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int spear_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + struct spear_dma_data *dma_data = (struct spear_dma_data *) + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + int ret; + + ret = snd_soc_set_runtime_hwparams(substream, &spear_pcm_hardware); + if (ret) + return ret; + + ret = snd_dmaengine_pcm_open(substream, dma_data->filter, dma_data); + if (ret) + return ret; + + snd_dmaengine_pcm_set_data(substream, dma_data); + + return 0; +} + +static int spear_pcm_close(struct snd_pcm_substream *substream) +{ + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int spear_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops spear_pcm_ops = { + .open = spear_pcm_open, + .close = spear_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = spear_pcm_hw_params, + .hw_free = spear_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = spear_pcm_mmap, +}; + +static int +spear_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, + size_t size) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + dev_info(buf->dev.dev, + " preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *)buf->area, (void *)buf->addr, size); + + buf->bytes = size; + return 0; +} + +static void spear_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf && !buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); + +static int spear_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &spear_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK, + spear_pcm_hardware.buffer_bytes_max); + if (ret) + return ret; + } + + if (dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE, + spear_pcm_hardware.buffer_bytes_max); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_platform_driver spear_soc_platform = { + .ops = &spear_pcm_ops, + .pcm_new = spear_pcm_new, + .pcm_free = spear_pcm_free, +}; + +static int __devinit spear_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &spear_soc_platform); +} + +static int __devexit spear_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver spear_pcm_driver = { + .driver = { + .name = "spear-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = spear_soc_platform_probe, + .remove = __devexit_p(spear_soc_platform_remove), +}; + +module_platform_driver(spear_pcm_driver); + +MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_DESCRIPTION("SPEAr PCM DMA module"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spear-pcm-audio"); diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 76dc230f2bb0..02bcd308c189 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,7 +1,8 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" - depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA + depends on ARCH_TEGRA && (TEGRA_SYSTEM_DMA || TEGRA20_APB_DMA) select REGMAP_MMIO + select SND_SOC_DMAENGINE_PCM if TEGRA20_APB_DMA help Say Y or M here if you want support for SoC audio on Tegra. diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 1647dbfe74b5..0832e8afd73c 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -46,18 +46,6 @@ #define DRV_NAME "tegra20-i2s" -static inline void tegra20_i2s_write(struct tegra20_i2s *i2s, u32 reg, u32 val) -{ - regmap_write(i2s->regmap, reg, val); -} - -static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg) -{ - u32 val; - regmap_read(i2s->regmap, reg, &val); - return val; -} - static int tegra20_i2s_runtime_suspend(struct device *dev) { struct tegra20_i2s *i2s = dev_get_drvdata(dev); @@ -85,6 +73,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -93,10 +82,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_MASTER_ENABLE; + mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_MASTER_ENABLE; + val = TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; @@ -104,33 +93,35 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~(TEGRA20_I2S_CTRL_BIT_FORMAT_MASK | - TEGRA20_I2S_CTRL_LRCK_MASK); + mask |= TEGRA20_I2S_CTRL_BIT_FORMAT_MASK | + TEGRA20_I2S_CTRL_LRCK_MASK; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_DSP_B: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_R_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + val |= TEGRA20_I2S_CTRL_LRCK_R_LOW; break; case SND_SOC_DAIFMT_I2S: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_RIGHT_J: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_LEFT_J: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; default: return -EINVAL; } + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val); + return 0; } @@ -138,29 +129,34 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct device *dev = substream->pcm->card->dev; + struct device *dev = dai->dev; struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); - u32 reg; + unsigned int mask, val; int ret, sample_size, srate, i2sclock, bitcnt; - i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_BIT_SIZE_MASK; + mask = TEGRA20_I2S_CTRL_BIT_SIZE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_16; + val = TEGRA20_I2S_CTRL_BIT_SIZE_16; sample_size = 16; break; case SNDRV_PCM_FORMAT_S24_LE: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_24; + val = TEGRA20_I2S_CTRL_BIT_SIZE_24; sample_size = 24; break; case SNDRV_PCM_FORMAT_S32_LE: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_32; + val = TEGRA20_I2S_CTRL_BIT_SIZE_32; sample_size = 32; break; default: return -EINVAL; } + mask |= TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK; + val |= TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED; + + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val); + srate = params_rate(params); /* Final "* 2" required by Tegra hardware */ @@ -175,42 +171,44 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream, bitcnt = (i2sclock / (2 * srate)) - 1; if (bitcnt < 0 || bitcnt > TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) return -EINVAL; - reg = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + val = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; if (i2sclock % (2 * srate)) - reg |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE; + val |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_TIMING, reg); + regmap_write(i2s->regmap, TEGRA20_I2S_TIMING, val); - tegra20_i2s_write(i2s, TEGRA20_I2S_FIFO_SCR, - TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | - TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); + regmap_write(i2s->regmap, TEGRA20_I2S_FIFO_SCR, + TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | + TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); return 0; } static void tegra20_i2s_start_playback(struct tegra20_i2s *i2s) { - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO1_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO1_ENABLE, + TEGRA20_I2S_CTRL_FIFO1_ENABLE); } static void tegra20_i2s_stop_playback(struct tegra20_i2s *i2s) { - i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO1_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO1_ENABLE, 0); } static void tegra20_i2s_start_capture(struct tegra20_i2s *i2s) { - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO2_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO2_ENABLE, + TEGRA20_I2S_CTRL_FIFO2_ENABLE); } static void tegra20_i2s_stop_capture(struct tegra20_i2s *i2s) { - i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO2_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO2_ENABLE, 0); } static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd, @@ -261,12 +259,14 @@ static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = { static const struct snd_soc_dai_driver tegra20_i2s_dai_template = { .probe = tegra20_i2s_probe, .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, @@ -412,8 +412,6 @@ static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) i2s->playback_dma_data.width = 32; i2s->playback_dma_data.req_sel = dma_ch; - i2s->reg_ctrl = TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED; - pm_runtime_enable(&pdev->dev); if (!pm_runtime_enabled(&pdev->dev)) { ret = tegra20_i2s_runtime_resume(&pdev->dev); diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index a57efc6a597e..c27069d24d77 100644 --- a/sound/soc/tegra/tegra20_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -158,7 +158,6 @@ struct tegra20_i2s { struct tegra_pcm_dma_params capture_dma_data; struct tegra_pcm_dma_params playback_dma_data; struct regmap *regmap; - u32 reg_ctrl; }; #endif diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 2262e4fdec2a..3ebc8670ba00 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -37,19 +37,6 @@ #define DRV_NAME "tegra20-spdif" -static inline void tegra20_spdif_write(struct tegra20_spdif *spdif, u32 reg, - u32 val) -{ - regmap_write(spdif->regmap, reg, val); -} - -static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg) -{ - u32 val; - regmap_read(spdif->regmap, reg, &val); - return val; -} - static int tegra20_spdif_runtime_suspend(struct device *dev) { struct tegra20_spdif *spdif = dev_get_drvdata(dev); @@ -77,21 +64,24 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct device *dev = substream->pcm->card->dev; + struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; int ret, spdifclock; - spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_PACK; - spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + mask = TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_PACK; - spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + val = TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; } + regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, mask, val); + switch (params_rate(params)) { case 32000: spdifclock = 4096000; @@ -129,14 +119,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, static void tegra20_spdif_start_playback(struct tegra20_spdif *spdif) { - spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_TX_EN; - tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl); + regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, + TEGRA20_SPDIF_CTRL_TX_EN, + TEGRA20_SPDIF_CTRL_TX_EN); } static void tegra20_spdif_stop_playback(struct tegra20_spdif *spdif) { - spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_TX_EN; - tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl); + regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, + TEGRA20_SPDIF_CTRL_TX_EN, 0); } static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd, @@ -181,6 +172,7 @@ static struct snd_soc_dai_driver tegra20_spdif_dai = { .name = DRV_NAME, .probe = tegra20_spdif_probe, .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h index ed756527efea..b48d699fd583 100644 --- a/sound/soc/tegra/tegra20_spdif.h +++ b/sound/soc/tegra/tegra20_spdif.h @@ -465,7 +465,6 @@ struct tegra20_spdif { struct tegra_pcm_dma_params capture_dma_data; struct tegra_pcm_dma_params playback_dma_data; struct regmap *regmap; - u32 reg_ctrl; }; #endif diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index d308faaae148..44184228d1f0 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -44,18 +44,6 @@ #define DRV_NAME "tegra30-i2s" -static inline void tegra30_i2s_write(struct tegra30_i2s *i2s, u32 reg, u32 val) -{ - regmap_write(i2s->regmap, reg, val); -} - -static inline u32 tegra30_i2s_read(struct tegra30_i2s *i2s, u32 reg) -{ - u32 val; - regmap_read(i2s->regmap, reg, &val); - return val; -} - static int tegra30_i2s_runtime_suspend(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); @@ -128,6 +116,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -136,10 +125,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_MASTER_ENABLE; + mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_MASTER_ENABLE; + val = TEGRA30_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; @@ -147,33 +136,37 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~(TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK | - TEGRA30_I2S_CTRL_LRCK_MASK); + mask |= TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK | + TEGRA30_I2S_CTRL_LRCK_MASK; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_DSP_B: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_R_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + val |= TEGRA30_I2S_CTRL_LRCK_R_LOW; break; case SND_SOC_DAIFMT_I2S: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_RIGHT_J: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_LEFT_J: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; break; default: return -EINVAL; } + pm_runtime_get_sync(dai->dev); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val); + pm_runtime_put(dai->dev); + return 0; } @@ -181,24 +174,26 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct device *dev = substream->pcm->card->dev; + struct device *dev = dai->dev; struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); - u32 val; + unsigned int mask, val, reg; int ret, sample_size, srate, i2sclock, bitcnt; if (params_channels(params) != 2) return -EINVAL; - i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_BIT_SIZE_MASK; + mask = TEGRA30_I2S_CTRL_BIT_SIZE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_BIT_SIZE_16; + val = TEGRA30_I2S_CTRL_BIT_SIZE_16; sample_size = 16; break; default: return -EINVAL; } + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val); + srate = params_rate(params); /* Final "* 2" required by Tegra hardware */ @@ -219,7 +214,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, if (i2sclock % (2 * srate)) val |= TEGRA30_I2S_TIMING_NON_SYM_ENABLE; - tegra30_i2s_write(i2s, TEGRA30_I2S_TIMING, val); + regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val); val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | @@ -229,15 +224,17 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_RX_CTRL, val); + reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_TX_CTRL, val); + reg = TEGRA30_I2S_CIF_RX_CTRL; } + regmap_write(i2s->regmap, reg, val); + val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) | (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT); - tegra30_i2s_write(i2s, TEGRA30_I2S_OFFSET, val); + regmap_write(i2s->regmap, TEGRA30_I2S_OFFSET, val); return 0; } @@ -245,29 +242,31 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, static void tegra30_i2s_start_playback(struct tegra30_i2s *i2s) { tegra30_ahub_enable_tx_fifo(i2s->playback_fifo_cif); - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_TX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_TX, + TEGRA30_I2S_CTRL_XFER_EN_TX); } static void tegra30_i2s_stop_playback(struct tegra30_i2s *i2s) { tegra30_ahub_disable_tx_fifo(i2s->playback_fifo_cif); - i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_TX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_TX, 0); } static void tegra30_i2s_start_capture(struct tegra30_i2s *i2s) { tegra30_ahub_enable_rx_fifo(i2s->capture_fifo_cif); - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_RX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_RX, + TEGRA30_I2S_CTRL_XFER_EN_RX); } static void tegra30_i2s_stop_capture(struct tegra30_i2s *i2s) { tegra30_ahub_disable_rx_fifo(i2s->capture_fifo_cif); - i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_RX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_RX, 0); } static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd, @@ -320,12 +319,14 @@ static struct snd_soc_dai_ops tegra30_i2s_dai_ops = { static const struct snd_soc_dai_driver tegra30_i2s_dai_template = { .probe = tegra30_i2s_probe, .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 91adf29c7a87..34dc47b9581c 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -236,7 +236,6 @@ struct tegra30_i2s { enum tegra30_ahub_txcif playback_fifo_cif; struct tegra_pcm_dma_params playback_dma_data; struct regmap *regmap; - u32 reg_ctrl; }; #endif diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 32de7006daf0..d684df294c0c 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -1,5 +1,5 @@ /* - * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver +* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver * * Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net> * Copyright (C) 2012 - NVIDIA, Inc. @@ -33,11 +33,8 @@ #define DRV_NAME "tegra-alc5632" -#define GPIO_HP_DET BIT(0) - struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; - int gpio_requested; int gpio_hp_det; }; @@ -46,7 +43,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -108,9 +105,9 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = { static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct device_node *np = codec->card->dev->of_node; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, @@ -119,14 +116,11 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(tegra_alc5632_hs_jack_pins), tegra_alc5632_hs_jack_pins); - machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, 1, &tegra_alc5632_hp_jack_gpio); - machine->gpio_requested |= GPIO_HP_DET; } snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); @@ -159,6 +153,7 @@ static struct snd_soc_card snd_soc_tegra_alc5632 = { static __devinit int tegra_alc5632_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &snd_soc_tegra_alc5632; struct tegra_alc5632 *alc5632; int ret; @@ -181,6 +176,10 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) goto err; } + alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (alc5632->gpio_hp_det == -ENODEV) + return -EPROBE_DEFER; + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); if (ret) goto err; @@ -199,16 +198,16 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) goto err; } - tegra_alc5632_dai.cpu_dai_of_node = of_parse_phandle( + tegra_alc5632_dai.cpu_of_node = of_parse_phandle( pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!tegra_alc5632_dai.cpu_dai_of_node) { + if (!tegra_alc5632_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; goto err; } - tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_dai_of_node; + tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) @@ -234,11 +233,8 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); - if (machine->gpio_requested & GPIO_HP_DET) - snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, - 1, - &tegra_alc5632_hp_jack_gpio); - machine->gpio_requested = 0; + snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1, + &tegra_alc5632_hp_jack_gpio); snd_soc_unregister_card(card); diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 127348dc09b1..5658bcec1931 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -36,6 +36,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> #include "tegra_pcm.h" @@ -56,6 +57,7 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .fifo_size = 4, }; +#if defined(CONFIG_TEGRA_SYSTEM_DMA) static void tegra_pcm_queue_dma(struct tegra_runtime_data *prtd) { struct snd_pcm_substream *substream = prtd->substream; @@ -285,6 +287,119 @@ static struct snd_pcm_ops tegra_pcm_ops = { .pointer = tegra_pcm_pointer, .mmap = tegra_pcm_mmap, }; +#else +static int tegra_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + int ret; + + /* Set HW params now that initialization is complete */ + snd_soc_set_runtime_hwparams(substream, &tegra_pcm_hardware); + + ret = snd_dmaengine_pcm_open(substream, NULL, NULL); + if (ret) { + dev_err(dev, "dmaengine pcm open failed with err %d\n", ret); + return ret; + } + + return 0; +} + +static int tegra_pcm_close(struct snd_pcm_substream *substream) +{ + snd_dmaengine_pcm_close(substream); + return 0; +} + +static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct tegra_pcm_dma_params *dmap; + struct dma_slave_config slave_config; + int ret; + + dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + ret = snd_hwparams_to_dma_slave_config(substream, params, + &slave_config); + if (ret) { + dev_err(dev, "hw params config failed with err %d\n", ret); + return ret; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + slave_config.dst_addr = dmap->addr; + slave_config.src_maxburst = 0; + } else { + slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + slave_config.src_addr = dmap->addr; + slave_config.dst_maxburst = 0; + } + slave_config.slave_id = dmap->req_sel; + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret < 0) { + dev_err(dev, "dma slave config failed with err %d\n", ret); + return ret; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + return 0; +} + +static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + return snd_dmaengine_pcm_trigger(substream, + SNDRV_PCM_TRIGGER_START); + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + return snd_dmaengine_pcm_trigger(substream, + SNDRV_PCM_TRIGGER_STOP); + default: + return -EINVAL; + } + return 0; +} + +static int tegra_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops tegra_pcm_ops = { + .open = tegra_pcm_open, + .close = tegra_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = tegra_pcm_hw_params, + .hw_free = tegra_pcm_hw_free, + .trigger = tegra_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = tegra_pcm_mmap, +}; +#endif static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h index 985d418a35e7..a3a450352dcf 100644 --- a/sound/soc/tegra/tegra_pcm.h +++ b/sound/soc/tegra/tegra_pcm.h @@ -40,6 +40,7 @@ struct tegra_pcm_dma_params { unsigned long req_sel; }; +#if defined(CONFIG_TEGRA_SYSTEM_DMA) struct tegra_runtime_data { struct snd_pcm_substream *substream; spinlock_t lock; @@ -51,6 +52,7 @@ struct tegra_runtime_data { struct tegra_dma_req dma_req[2]; struct tegra_dma_channel *dma_chan; }; +#endif int tegra_pcm_platform_register(struct device *dev); void tegra_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index 4e77026807a2..ea9166d5c4eb 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -57,7 +57,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -157,9 +157,9 @@ static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev) goto err; } - tegra_wm8753_dai.cpu_dai_of_node = of_parse_phandle( + tegra_wm8753_dai.cpu_of_node = of_parse_phandle( pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!tegra_wm8753_dai.cpu_dai_of_node) { + if (!tegra_wm8753_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; @@ -167,7 +167,7 @@ static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev) } tegra_wm8753_dai.platform_of_node = - tegra_wm8753_dai.cpu_dai_of_node; + tegra_wm8753_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 3b6da91188a9..0c5bb33d258e 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -28,8 +28,6 @@ * */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -50,16 +48,9 @@ #define DRV_NAME "tegra-snd-wm8903" -#define GPIO_SPKR_EN BIT(0) -#define GPIO_HP_MUTE BIT(1) -#define GPIO_INT_MIC_EN BIT(2) -#define GPIO_EXT_MIC_EN BIT(3) -#define GPIO_HP_DET BIT(4) - struct tegra_wm8903 { struct tegra_wm8903_platform_data pdata; struct tegra_asoc_utils_data util_data; - int gpio_requested; }; static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, @@ -67,8 +58,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -95,24 +85,6 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, return err; } - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "codec_dai fmt not set\n"); - return err; - } - - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "cpu_dai fmt not set\n"); - return err; - } - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (err < 0) { @@ -160,7 +132,7 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w, struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (!(machine->gpio_requested & GPIO_SPKR_EN)) + if (!gpio_is_valid(pdata->gpio_spkr_en)) return 0; gpio_set_value_cansleep(pdata->gpio_spkr_en, @@ -177,7 +149,7 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w, struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (!(machine->gpio_requested & GPIO_HP_MUTE)) + if (!gpio_is_valid(pdata->gpio_hp_mute)) return 0; gpio_set_value_cansleep(pdata->gpio_hp_mute, @@ -203,122 +175,18 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = { {"IN1L", NULL, "Mic Jack"}, }; -static const struct snd_soc_dapm_route seaboard_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "ROP"}, - {"Int Spk", NULL, "RON"}, - {"Int Spk", NULL, "LOP"}, - {"Int Spk", NULL, "LON"}, - {"Mic Jack", NULL, "MICBIAS"}, - {"IN1R", NULL, "Mic Jack"}, -}; - -static const struct snd_soc_dapm_route kaen_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "ROP"}, - {"Int Spk", NULL, "RON"}, - {"Int Spk", NULL, "LOP"}, - {"Int Spk", NULL, "LON"}, - {"Mic Jack", NULL, "MICBIAS"}, - {"IN2R", NULL, "Mic Jack"}, -}; - -static const struct snd_soc_dapm_route aebl_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "LINEOUTR"}, - {"Int Spk", NULL, "LINEOUTL"}, - {"Mic Jack", NULL, "MICBIAS"}, - {"IN1R", NULL, "Mic Jack"}, -}; - static const struct snd_kcontrol_new tegra_wm8903_controls[] = { SOC_DAPM_PIN_SWITCH("Int Spk"), }; static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = &machine->pdata; - struct device_node *np = card->dev->of_node; - int ret; - - if (card->dev->platform_data) { - memcpy(pdata, card->dev->platform_data, sizeof(*pdata)); - } else if (np) { - /* - * This part must be in init() rather than probe() in order to - * guarantee that the WM8903 has been probed, and hence its - * GPIO controller registered, which is a pre-condition for - * of_get_named_gpio() to be able to map the phandles in the - * properties to the controller node. Given this, all - * pdata handling is in init() for consistency. - */ - pdata->gpio_spkr_en = of_get_named_gpio(np, - "nvidia,spkr-en-gpios", 0); - pdata->gpio_hp_mute = of_get_named_gpio(np, - "nvidia,hp-mute-gpios", 0); - pdata->gpio_hp_det = of_get_named_gpio(np, - "nvidia,hp-det-gpios", 0); - pdata->gpio_int_mic_en = of_get_named_gpio(np, - "nvidia,int-mic-en-gpios", 0); - pdata->gpio_ext_mic_en = of_get_named_gpio(np, - "nvidia,ext-mic-en-gpios", 0); - } else { - dev_err(card->dev, "No platform data supplied\n"); - return -EINVAL; - } - - if (gpio_is_valid(pdata->gpio_spkr_en)) { - ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); - if (ret) { - dev_err(card->dev, "cannot get spkr_en gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_SPKR_EN; - - gpio_direction_output(pdata->gpio_spkr_en, 0); - } - - if (gpio_is_valid(pdata->gpio_hp_mute)) { - ret = gpio_request(pdata->gpio_hp_mute, "hp_mute"); - if (ret) { - dev_err(card->dev, "cannot get hp_mute gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_HP_MUTE; - - gpio_direction_output(pdata->gpio_hp_mute, 1); - } - - if (gpio_is_valid(pdata->gpio_int_mic_en)) { - ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get int_mic_en gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_INT_MIC_EN; - - /* Disable int mic; enable signal is active-high */ - gpio_direction_output(pdata->gpio_int_mic_en, 0); - } - - if (gpio_is_valid(pdata->gpio_ext_mic_en)) { - ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get ext_mic_en gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_EXT_MIC_EN; - - /* Enable ext mic; enable signal is active-low */ - gpio_direction_output(pdata->gpio_ext_mic_en, 0); - } if (gpio_is_valid(pdata->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det; @@ -330,7 +198,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); - machine->gpio_requested |= GPIO_HP_DET; } snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, @@ -366,6 +233,9 @@ static struct snd_soc_dai_link tegra_wm8903_dai = { .codec_dai_name = "wm8903-hifi", .init = tegra_wm8903_init, .ops = &tegra_wm8903_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, }; static struct snd_soc_card snd_soc_tegra_wm8903 = { @@ -385,8 +255,10 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &snd_soc_tegra_wm8903; struct tegra_wm8903 *machine; + struct tegra_wm8903_platform_data *pdata; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -401,12 +273,42 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) ret = -ENOMEM; goto err; } + pdata = &machine->pdata; card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - if (pdev->dev.of_node) { + if (pdev->dev.platform_data) { + memcpy(pdata, card->dev->platform_data, sizeof(*pdata)); + } else if (np) { + pdata->gpio_spkr_en = of_get_named_gpio(np, + "nvidia,spkr-en-gpios", 0); + if (pdata->gpio_spkr_en == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_hp_mute = of_get_named_gpio(np, + "nvidia,hp-mute-gpios", 0); + if (pdata->gpio_hp_mute == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_hp_det = of_get_named_gpio(np, + "nvidia,hp-det-gpios", 0); + if (pdata->gpio_hp_det == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_int_mic_en = of_get_named_gpio(np, + "nvidia,int-mic-en-gpios", 0); + if (pdata->gpio_int_mic_en == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_ext_mic_en = of_get_named_gpio(np, + "nvidia,ext-mic-en-gpios", 0); + if (pdata->gpio_ext_mic_en == -ENODEV) + return -EPROBE_DEFER; + } + + if (np) { ret = snd_soc_of_parse_card_name(card, "nvidia,model"); if (ret) goto err; @@ -417,8 +319,8 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) goto err; tegra_wm8903_dai.codec_name = NULL; - tegra_wm8903_dai.codec_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,audio-codec", 0); + tegra_wm8903_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); if (!tegra_wm8903_dai.codec_of_node) { dev_err(&pdev->dev, "Property 'nvidia,audio-codec' missing or invalid\n"); @@ -427,9 +329,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } tegra_wm8903_dai.cpu_dai_name = NULL; - tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!tegra_wm8903_dai.cpu_dai_of_node) { + tegra_wm8903_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_wm8903_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; @@ -438,20 +340,47 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) tegra_wm8903_dai.platform_name = NULL; tegra_wm8903_dai.platform_of_node = - tegra_wm8903_dai.cpu_dai_of_node; + tegra_wm8903_dai.cpu_of_node; } else { - if (machine_is_harmony()) { - card->dapm_routes = harmony_audio_map; - card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); - } else if (machine_is_seaboard()) { - card->dapm_routes = seaboard_audio_map; - card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map); - } else if (machine_is_kaen()) { - card->dapm_routes = kaen_audio_map; - card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map); - } else { - card->dapm_routes = aebl_audio_map; - card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map); + card->dapm_routes = harmony_audio_map; + card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); + } + + if (gpio_is_valid(pdata->gpio_spkr_en)) { + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_spkr_en, + GPIOF_OUT_INIT_LOW, "spkr_en"); + if (ret) { + dev_err(card->dev, "cannot get spkr_en gpio\n"); + return ret; + } + } + + if (gpio_is_valid(pdata->gpio_hp_mute)) { + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_hp_mute, + GPIOF_OUT_INIT_HIGH, "hp_mute"); + if (ret) { + dev_err(card->dev, "cannot get hp_mute gpio\n"); + return ret; + } + } + + if (gpio_is_valid(pdata->gpio_int_mic_en)) { + /* Disable int mic; enable signal is active-high */ + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_int_mic_en, + GPIOF_OUT_INIT_LOW, "int_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get int_mic_en gpio\n"); + return ret; + } + } + + if (gpio_is_valid(pdata->gpio_ext_mic_en)) { + /* Enable ext mic; enable signal is active-low */ + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_ext_mic_en, + GPIOF_OUT_INIT_LOW, "ext_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get ext_mic_en gpio\n"); + return ret; } } @@ -478,21 +407,9 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (machine->gpio_requested & GPIO_HP_DET) - snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, - 1, - &tegra_wm8903_hp_jack_gpio); - if (machine->gpio_requested & GPIO_EXT_MIC_EN) - gpio_free(pdata->gpio_ext_mic_en); - if (machine->gpio_requested & GPIO_INT_MIC_EN) - gpio_free(pdata->gpio_int_mic_en); - if (machine->gpio_requested & GPIO_HP_MUTE) - gpio_free(pdata->gpio_hp_mute); - if (machine->gpio_requested & GPIO_SPKR_EN) - gpio_free(pdata->gpio_spkr_en); - machine->gpio_requested = 0; + snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1, + &tegra_wm8903_hp_jack_gpio); snd_soc_unregister_card(card); diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 4a8d5b672c9f..e69a4f7000d6 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -52,8 +52,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -68,24 +67,6 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, return err; } - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "codec_dai fmt not set\n"); - return err; - } - - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "cpu_dai fmt not set\n"); - return err; - } - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (err < 0) { @@ -121,6 +102,9 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "tlv320aic23-hifi", .ops = &trimslice_asoc_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, }; static struct snd_soc_card snd_soc_trimslice = { @@ -162,9 +146,9 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) } trimslice_tlv320aic23_dai.cpu_dai_name = NULL; - trimslice_tlv320aic23_dai.cpu_dai_of_node = of_parse_phandle( + trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle( pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!trimslice_tlv320aic23_dai.cpu_dai_of_node) { + if (!trimslice_tlv320aic23_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; @@ -173,7 +157,7 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) trimslice_tlv320aic23_dai.platform_name = NULL; trimslice_tlv320aic23_dai.platform_of_node = - trimslice_tlv320aic23_dai.cpu_dai_of_node; + trimslice_tlv320aic23_dai.cpu_of_node; } ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev); diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 44cf43404cd9..069330d82be5 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -12,3 +12,21 @@ menuconfig SND_SOC_UX500 config SND_SOC_UX500_PLAT_MSP_I2S tristate depends on SND_SOC_UX500 + +config SND_SOC_UX500_PLAT_DMA + tristate "Platform - DB8500 (DMA)" + depends on SND_SOC_UX500 + select SND_SOC_DMAENGINE_PCM + help + Say Y if you want to enable the Ux500 platform-driver. + ++config SND_SOC_UX500_MACH_MOP500 ++ tristate "Machine - MOP500 (Ux500 + AB8500)" + depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500 + select SND_SOC_AB8500_CODEC + select SND_SOC_UX500_PLAT_MSP_I2S + select SND_SOC_UX500_PLAT_DMA + help + Select this to enable the MOP500 machine-driver. + This will enable platform-drivers for: Ux500 + This will enable codec-drivers for: AB8500 diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 19974c5a2ea1..cce0c11a4d86 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -2,3 +2,9 @@ snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o + +snd-soc-ux500-plat-dma-objs := ux500_pcm.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o + +snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o +obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c new file mode 100644 index 000000000000..31c4d26d0359 --- /dev/null +++ b/sound/soc/ux500/mop500.c @@ -0,0 +1,113 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja (ola.o.lilja@stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/io.h> +#include <linux/spi/spi.h> + +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#include <mop500_ab8500.h> + +/* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link mop500_dai_links[] = { + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = mop500_ab8500_machine_init, + .ops = mop500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = mop500_ab8500_ops, + }, +}; + +static struct snd_soc_card mop500_card = { + .name = "MOP500-card", + .probe = NULL, + .dai_link = mop500_dai_links, + .num_links = ARRAY_SIZE(mop500_dai_links), +}; + +static int __devinit mop500_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + mop500_card.dev = &pdev->dev; + + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", + __func__, mop500_card.name); + platform_set_drvdata(pdev, &mop500_card); + + snd_soc_card_set_drvdata(&mop500_card, NULL); + + dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n", + __func__, mop500_card.name, mop500_card.num_links); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n", + __func__, mop500_card.name, mop500_card.dai_link[0].name); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, mop500_card.name, + mop500_card.dai_link[0].stream_name); + + ret = snd_soc_register_card(&mop500_card); + if (ret) + dev_err(&pdev->dev, + "Error: snd_soc_register_card failed (%d)!\n", + ret); + + return ret; +} + +static int __devexit mop500_remove(struct platform_device *pdev) +{ + struct snd_soc_card *mop500_card = platform_get_drvdata(pdev); + + pr_debug("%s: Enter.\n", __func__); + + snd_soc_unregister_card(mop500_card); + mop500_ab8500_remove(mop500_card); + + return 0; +} + +static struct platform_driver snd_soc_mop500_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-soc-mop500", + }, + .probe = mop500_probe, + .remove = __devexit_p(mop500_remove), +}; + +module_platform_driver(snd_soc_mop500_driver); diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c new file mode 100644 index 000000000000..78cce236693e --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.c @@ -0,0 +1,431 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <mach/hardware.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500-codec.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +/* Clocks */ +static const char * const enum_mclk[] = { + "SYSCLK", + "ULPCLK" +}; +enum mclk { + MCLK_SYSCLK, + MCLK_ULPCLK, +}; + +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_mclk, enum_mclk); + +/* Private data for machine-part MOP500<->AB8500 */ +struct mop500_ab8500_drvdata { + /* Clocks */ + enum mclk mclk_sel; + struct clk *clk_ptr_intclk; + struct clk *clk_ptr_sysclk; + struct clk *clk_ptr_ulpclk; +}; + +static inline const char *get_mclk_str(enum mclk mclk_sel) +{ + switch (mclk_sel) { + case MCLK_SYSCLK: + return "SYSCLK"; + case MCLK_ULPCLK: + return "ULPCLK"; + default: + return "Unknown"; + } +} + +static int mop500_ab8500_set_mclk(struct device *dev, + struct mop500_ab8500_drvdata *drvdata) +{ + int status; + struct clk *clk_ptr; + + if (IS_ERR(drvdata->clk_ptr_intclk)) { + dev_err(dev, + "%s: ERROR: intclk not initialized!\n", __func__); + return -EIO; + } + + switch (drvdata->mclk_sel) { + case MCLK_SYSCLK: + clk_ptr = drvdata->clk_ptr_sysclk; + break; + case MCLK_ULPCLK: + clk_ptr = drvdata->clk_ptr_ulpclk; + break; + default: + return -EINVAL; + } + + if (IS_ERR(clk_ptr)) { + dev_err(dev, "%s: ERROR: %s not initialized!\n", __func__, + get_mclk_str(drvdata->mclk_sel)); + return -EIO; + } + + status = clk_set_parent(drvdata->clk_ptr_intclk, clk_ptr); + if (status) + dev_err(dev, + "%s: ERROR: Setting intclk parent to %s failed (ret = %d)!", + __func__, get_mclk_str(drvdata->mclk_sel), status); + else + dev_dbg(dev, + "%s: intclk parent changed to %s.\n", + __func__, get_mclk_str(drvdata->mclk_sel)); + + return status; +} + +/* + * Control-events + */ + +static int mclk_input_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + + ucontrol->value.enumerated.item[0] = drvdata->mclk_sel; + + return 0; +} + +static int mclk_input_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + unsigned int val = ucontrol->value.enumerated.item[0]; + + if (val > (unsigned int)MCLK_ULPCLK) + return -EINVAL; + if (drvdata->mclk_sel == val) + return 0; + + drvdata->mclk_sel = val; + + return 1; +} + +/* + * Controls + */ + +static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { + SOC_ENUM_EXT("Master Clock Select", + soc_enum_mclk, + mclk_input_control_get, mclk_input_control_put), + /* Digital interface - Clocks */ + SOC_SINGLE("Digital Interface Master Generator Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN, + 1, 0), + SOC_SINGLE("Digital Interface 0 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0, + 1, 0), + SOC_SINGLE("Digital Interface 1 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1, + 1, 0), + SOC_DAPM_PIN_SWITCH("Headset Left"), + SOC_DAPM_PIN_SWITCH("Headset Right"), + SOC_DAPM_PIN_SWITCH("Earpiece"), + SOC_DAPM_PIN_SWITCH("Speaker Left"), + SOC_DAPM_PIN_SWITCH("Speaker Right"), + SOC_DAPM_PIN_SWITCH("LineOut Left"), + SOC_DAPM_PIN_SWITCH("LineOut Right"), + SOC_DAPM_PIN_SWITCH("Vibra 1"), + SOC_DAPM_PIN_SWITCH("Vibra 2"), + SOC_DAPM_PIN_SWITCH("Mic 1"), + SOC_DAPM_PIN_SWITCH("Mic 2"), + SOC_DAPM_PIN_SWITCH("LineIn Left"), + SOC_DAPM_PIN_SWITCH("LineIn Right"), + SOC_DAPM_PIN_SWITCH("DMic 1"), + SOC_DAPM_PIN_SWITCH("DMic 2"), + SOC_DAPM_PIN_SWITCH("DMic 3"), + SOC_DAPM_PIN_SWITCH("DMic 4"), + SOC_DAPM_PIN_SWITCH("DMic 5"), + SOC_DAPM_PIN_SWITCH("DMic 6"), +}; + +/* ASoC */ + +int mop500_ab8500_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set audio-clock source */ + return mop500_ab8500_set_mclk(rtd->card->dev, + snd_soc_card_get_drvdata(rtd->card)); +} + +void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + + dev_dbg(dev, "%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; +} + +int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct device *dev = rtd->card->dev; + unsigned int fmt; + int channels, ret = 0, driver_mode, slots; + unsigned int sw_codec, sw_cpu; + bool is_playback; + + dev_dbg(dev, "%s: Enter\n", __func__); + + dev_dbg(dev, "%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + sw_cpu = 32; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + sw_cpu = 16; + break; + + default: + return -EINVAL; + } + + /* Setup codec depending on driver-mode */ + if (channels == 8) + driver_mode = DRIVERMODE_CODEC_ONLY; + else + driver_mode = DRIVERMODE_NORMAL; + dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + /* Setup format */ + + if (driver_mode == DRIVERMODE_NORMAL) { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Setup TDM-slots */ + + is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_MONO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_STEREO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_8CH : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + if (driver_mode == DRIVERMODE_NORMAL) + sw_codec = sw_cpu; + else + sw_codec = 20; + + dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, + sw_cpu); + if (ret) + return ret; + + dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, + sw_codec); + if (ret) + return ret; + + return 0; +} + +struct snd_soc_ops mop500_ab8500_ops[] = { + { + .hw_params = mop500_ab8500_hw_params, + .startup = mop500_ab8500_startup, + .shutdown = mop500_ab8500_shutdown, + } +}; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct device *dev = rtd->card->dev; + struct mop500_ab8500_drvdata *drvdata; + int ret; + + dev_dbg(dev, "%s Enter.\n", __func__); + + /* Create driver private-data struct */ + drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata), + GFP_KERNEL); + snd_soc_card_set_drvdata(rtd->card, drvdata); + + /* Setup clocks */ + + drvdata->clk_ptr_sysclk = clk_get(dev, "sysclk"); + if (IS_ERR(drvdata->clk_ptr_sysclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'sysclk'!\n", + __func__); + drvdata->clk_ptr_ulpclk = clk_get(dev, "ulpclk"); + if (IS_ERR(drvdata->clk_ptr_ulpclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'ulpclk'!\n", + __func__); + drvdata->clk_ptr_intclk = clk_get(dev, "intclk"); + if (IS_ERR(drvdata->clk_ptr_intclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'intclk'!\n", + __func__); + + /* Set intclk default parent to ulpclk */ + drvdata->mclk_sel = MCLK_ULPCLK; + ret = mop500_ab8500_set_mclk(dev, drvdata); + if (ret < 0) + dev_warn(dev, "%s: WARNING: mop500_ab8500_set_mclk!\n", + __func__); + + drvdata->mclk_sel = MCLK_ULPCLK; + + /* Add controls */ + ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ARRAY_SIZE(mop500_ab8500_ctrls)); + if (ret < 0) { + pr_err("%s: Failed to add machine-controls (%d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + + return ret; +} + +void mop500_ab8500_remove(struct snd_soc_card *card) +{ + struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card); + + if (drvdata->clk_ptr_sysclk != NULL) + clk_put(drvdata->clk_ptr_sysclk); + if (drvdata->clk_ptr_ulpclk != NULL) + clk_put(drvdata->clk_ptr_ulpclk); + if (drvdata->clk_ptr_intclk != NULL) + clk_put(drvdata->clk_ptr_intclk); + + snd_soc_card_set_drvdata(card, drvdata); +} diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h new file mode 100644 index 000000000000..cca5b33964b6 --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.h @@ -0,0 +1,22 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef MOP500_AB8500_H +#define MOP500_AB8500_H + +extern struct snd_soc_ops mop500_ab8500_ops[]; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime); +void mop500_ab8500_remove(struct snd_soc_card *card); + +#endif diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 93c6c40e724c..62ac0285bfaf 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -840,4 +840,4 @@ static struct platform_driver msp_i2s_driver = { }; module_platform_driver(msp_i2s_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 496dec10c96e..ee14d2dac2f5 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -739,4 +739,4 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, devm_kfree(&pdev->dev, msp); } -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c new file mode 100644 index 000000000000..1a04e248453c --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.c @@ -0,0 +1,318 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/page.h> + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/slab.h> + +#include <plat/ste_dma40.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "ux500_msp_i2s.h" +#include "ux500_pcm.h" + +static struct snd_pcm_hardware ux500_pcm_hw_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK, + .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static struct snd_pcm_hardware ux500_pcm_hw_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE, + .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static void ux500_pcm_dma_hw_free(struct device *dev, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + + if (runtime->dma_area == NULL) + return; + + if (buf != &substream->dma_buffer) { + dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, + buf->addr); + kfree(runtime->dma_buffer_p); + } + + snd_pcm_set_runtime_buffer(substream, NULL); +} + +static int ux500_pcm_open(struct snd_pcm_substream *substream) +{ + int stream_id = substream->pstr->stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct device *dev = dai->dev; + int ret; + struct ux500_msp_dma_params *dma_params; + u16 per_data_width, mem_data_width; + struct stedma40_chan_cfg *dma_cfg; + + dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__); + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_playback); + else + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_capture); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n", + __func__, ret); + return ret; + } + + dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__, + snd_pcm_stream_str(substream)); + runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + ux500_pcm_hw_playback : ux500_pcm_hw_capture; + + mem_data_width = STEDMA40_HALFWORD_WIDTH; + + dma_params = snd_soc_dai_get_dma_data(dai, substream); + switch (dma_params->data_size) { + case 32: + per_data_width = STEDMA40_WORD_WIDTH; + break; + case 16: + per_data_width = STEDMA40_HALFWORD_WIDTH; + break; + case 8: + per_data_width = STEDMA40_BYTE_WIDTH; + break; + default: + per_data_width = STEDMA40_WORD_WIDTH; + dev_warn(rtd->platform->dev, + "%s: Unknown data-size (%d)! Assuming 32 bits.\n", + __func__, dma_params->data_size); + } + + dma_cfg = dma_params->dma_cfg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_cfg->src_info.data_width = mem_data_width; + dma_cfg->dst_info.data_width = per_data_width; + } else { + dma_cfg->src_info.data_width = per_data_width; + dma_cfg->dst_info.data_width = mem_data_width; + } + + + ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg); + if (ret) { + dev_dbg(dai->dev, + "%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n", + __func__, ret); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_cfg); + + return 0; +} + +static int ux500_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + + dev_dbg(dai->dev, "%s: Enter\n", __func__); + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int ux500_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + int size; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + size = params_buffer_bytes(hw_params); + + if (buf) { + if (buf->bytes >= size) + goto out; + ux500_pcm_dma_hw_free(NULL, substream); + } + + if (substream->dma_buffer.area != NULL && + substream->dma_buffer.bytes >= size) { + buf = &substream->dma_buffer; + } else { + buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); + if (!buf) + goto nomem; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = NULL; + buf->area = dma_alloc_coherent(NULL, size, &buf->addr, + GFP_KERNEL); + buf->bytes = size; + buf->private_data = NULL; + + if (!buf->area) + goto free; + } + snd_pcm_set_runtime_buffer(substream, buf); + ret = 1; + out: + runtime->dma_bytes = size; + return ret; + + free: + kfree(buf); + nomem: + return -ENOMEM; +} + +static int ux500_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + ux500_pcm_dma_hw_free(NULL, substream); + + return 0; +} + +static int ux500_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + return dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); +} + +static struct snd_pcm_ops ux500_pcm_ops = { + .open = ux500_pcm_open, + .close = ux500_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ux500_pcm_hw_params, + .hw_free = ux500_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer_no_residue, + .mmap = ux500_pcm_mmap +}; + +int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + dev_dbg(rtd->platform->dev, "%s: Enter (id = '%s').\n", __func__, + pcm->id); + + pcm->info_flags = 0; + + return 0; +} + +static struct snd_soc_platform_driver ux500_pcm_soc_drv = { + .ops = &ux500_pcm_ops, + .pcm_new = ux500_pcm_new, +}; + +static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +{ + int ret; + + ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + if (ret < 0) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to register platform '%s' (%d)!\n", + __func__, pdev->name, ret); + return ret; + } + + return 0; +} + +static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver ux500_pcm_driver = { + .driver = { + .name = "ux500-pcm", + .owner = THIS_MODULE, + }, + + .probe = ux500_pcm_drv_probe, + .remove = __devexit_p(ux500_pcm_drv_remove), +}; +module_platform_driver(ux500_pcm_driver); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h new file mode 100644 index 000000000000..77ed44d371e9 --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.h @@ -0,0 +1,35 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ +#ifndef UX500_PCM_H +#define UX500_PCM_H + +#include <asm/page.h> + +#include <linux/workqueue.h> + +#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000 +#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000 +#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000 +#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 + +#define UX500_PLATFORM_PERIODS_BYTES_MIN 128 +#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) +#define UX500_PLATFORM_PERIODS_MIN 2 +#define UX500_PLATFORM_PERIODS_MAX 48 +#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) + +#endif diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 64aed432ae22..7da0d0aa72cb 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -485,7 +485,7 @@ static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; - struct snd_card *card; + struct snd_card *card = NULL; struct usb_device *device = interface_to_usbdev(intf); ret = create_card(device, intf, &card); diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 41f4b6911920..690000db0ec0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -42,6 +42,13 @@ extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; +struct std_mono_table { + unsigned int unitid, control, cmask; + int val_type; + const char *name; + snd_kcontrol_tlv_rw_t *tlv_callback; +}; + /* private_free callback */ static void usb_mixer_elem_free(struct snd_kcontrol *kctl) { @@ -114,6 +121,25 @@ static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer, } /* + * Create a set of standard UAC controls from a table + */ +static int snd_create_std_mono_table(struct usb_mixer_interface *mixer, + struct std_mono_table *t) +{ + int err; + + while (t->name != NULL) { + err = snd_create_std_mono_ctl(mixer, t->unitid, t->control, + t->cmask, t->val_type, t->name, t->tlv_callback); + if (err < 0) + return err; + t++; + } + + return 0; +} + +/* * Sound Blaster remote control configuration * * format of remote control data: @@ -916,61 +942,6 @@ static int snd_ftu_create_mixer(struct usb_mixer_interface *mixer) return 0; } - -/* - * Create mixer for Electrix Ebox-44 - * - * The mixer units from this device are corrupt, and even where they - * are valid they presents mono controls as L and R channels of - * stereo. So we create a good mixer in code. - */ - -static int snd_ebox44_create_mixer(struct usb_mixer_interface *mixer) -{ - int err; - - err = snd_create_std_mono_ctl(mixer, 4, 1, 0x0, USB_MIXER_INV_BOOLEAN, - "Headphone Playback Switch", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 4, 2, 0x1, USB_MIXER_S16, - "Headphone A Mix Playback Volume", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 4, 2, 0x2, USB_MIXER_S16, - "Headphone B Mix Playback Volume", NULL); - if (err < 0) - return err; - - err = snd_create_std_mono_ctl(mixer, 7, 1, 0x0, USB_MIXER_INV_BOOLEAN, - "Output Playback Switch", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 7, 2, 0x1, USB_MIXER_S16, - "Output A Playback Volume", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 7, 2, 0x2, USB_MIXER_S16, - "Output B Playback Volume", NULL); - if (err < 0) - return err; - - err = snd_create_std_mono_ctl(mixer, 10, 1, 0x0, USB_MIXER_INV_BOOLEAN, - "Input Capture Switch", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 10, 2, 0x1, USB_MIXER_S16, - "Input A Capture Volume", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 10, 2, 0x2, USB_MIXER_S16, - "Input B Capture Volume", NULL); - if (err < 0) - return err; - - return 0; -} - void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, unsigned char samplerate_id) { @@ -990,6 +961,81 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } +/* + * The mixer units for Ebox-44 are corrupt, and even where they + * are valid they presents mono controls as L and R channels of + * stereo. So we provide a good mixer here. + */ +struct std_mono_table ebox44_table[] = { + { + .unitid = 4, + .control = 1, + .cmask = 0x0, + .val_type = USB_MIXER_INV_BOOLEAN, + .name = "Headphone Playback Switch" + }, + { + .unitid = 4, + .control = 2, + .cmask = 0x1, + .val_type = USB_MIXER_S16, + .name = "Headphone A Mix Playback Volume" + }, + { + .unitid = 4, + .control = 2, + .cmask = 0x2, + .val_type = USB_MIXER_S16, + .name = "Headphone B Mix Playback Volume" + }, + + { + .unitid = 7, + .control = 1, + .cmask = 0x0, + .val_type = USB_MIXER_INV_BOOLEAN, + .name = "Output Playback Switch" + }, + { + .unitid = 7, + .control = 2, + .cmask = 0x1, + .val_type = USB_MIXER_S16, + .name = "Output A Playback Volume" + }, + { + .unitid = 7, + .control = 2, + .cmask = 0x2, + .val_type = USB_MIXER_S16, + .name = "Output B Playback Volume" + }, + + { + .unitid = 10, + .control = 1, + .cmask = 0x0, + .val_type = USB_MIXER_INV_BOOLEAN, + .name = "Input Capture Switch" + }, + { + .unitid = 10, + .control = 2, + .cmask = 0x1, + .val_type = USB_MIXER_S16, + .name = "Input A Capture Volume" + }, + { + .unitid = 10, + .control = 2, + .cmask = 0x2, + .val_type = USB_MIXER_S16, + .name = "Input B Capture Volume" + }, + + {} +}; + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1035,7 +1081,8 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */ - err = snd_ebox44_create_mixer(mixer); + /* detection is disabled in mixer_maps.c */ + err = snd_create_std_mono_table(mixer, ebox44_table); break; } |