diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2013-05-03 09:10:23 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2013-05-03 09:10:23 -0700 |
| commit | 9992ba72327fa0d8bdc9fb624e80f5cce338a711 (patch) | |
| tree | e0bf31ae53cb19c44674df7e0d0343a26037ad34 | |
| parent | 00fdffb5131125dce0702bf61e24a806ec3aed80 (diff) | |
| parent | 4ca231b2e6ed171107c5b21f9e92d1965fd6fd9e (diff) | |
Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone
mic and headset mic support, jack_modes hint consolidation, proper
beep attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
ALSA: pcm_format_to_bits strong-typed conversion
ALSA: compress: fix the states to check for allowing read
ALSA: hda - Move Thinkpad X220 to use auto parser
ALSA: USB: adjust for changed 3.8 USB API
ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
sound: oss/dmabuf: use dma_map_single
ALSA: ali5451: use mdelay instead of large udelay constants
ALSA: hda - Add the support for ALC286 codec
ALSA: usb-audio: USB quirk for Yamaha THR10C
ALSA: usb-audio: USB quirk for Yamaha THR5A
ALSA: usb-audio: USB quirk for Yamaha THR10
ALSA: usb-audio: Fix autopm error during probing
ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
ALSA: sound kconfig typo
ALSA: emu10k1: Fix dock firmware loading
ASoC: ux500: forward declare msp_i2s_platform_data
ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
...
267 files changed, 7241 insertions, 3721 deletions
diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index bd6fee22c4dd..06741e925985 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -6164,14 +6164,12 @@ struct _snd_pcm_runtime { <para> The macro takes an conditional expression to evaluate. - When <constant>CONFIG_SND_DEBUG</constant>, is set, the - expression is actually evaluated. If it's non-zero, it shows - the warning message such as + When <constant>CONFIG_SND_DEBUG</constant>, is set, if the + expression is non-zero, it shows the warning message such as <computeroutput>BUG? (xxx)</computeroutput> - normally followed by stack trace. It returns the evaluated - value. - When no <constant>CONFIG_SND_DEBUG</constant> is set, this - macro always returns zero. + normally followed by stack trace. + + In both cases it returns the evaluated value. </para> </section> diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt new file mode 100644 index 000000000000..dc3914fe6ce8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -0,0 +1,19 @@ +AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC + +This device has no control interface. + +Required properties: + + - compatible : "asahi-kasei,ak5386" + +Optional properties: + + - reset-gpio : a GPIO spec for the reset/power down pin. + If specified, it will be deasserted at probe time. + +Example: + +spdif: ak5386@0 { + compatible = "asahi-kasei,ak5386"; + reset-gpio = <&gpio0 23>; +}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt index 1ac7b1642186..0e5c12c66523 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt @@ -1,12 +1,22 @@ NVIDIA Tegra30 AHUB (Audio Hub) Required properties: -- compatible : "nvidia,tegra30-ahub" +- compatible : "nvidia,tegra30-ahub", "nvidia,tegra114-ahub", etc. - reg : Should contain the register physical address and length for each of - the AHUB's APBIF registers and the AHUB's own registers. + the AHUB's register blocks. + - Tegra30 requires 2 entries, for the APBIF and AHUB/AUDIO register blocks. + - Tegra114 requires an additional entry, for the APBIF2 register block. - interrupts : Should contain AHUB interrupt -- nvidia,dma-request-selector : The Tegra DMA controller's phandle and - request selector for the first APBIF channel. +- nvidia,dma-request-selector : A list of the DMA channel specifiers. Each + entry contains the Tegra DMA controller's phandle and request selector. + If a single entry is present, the request selectors for the channels are + assumed to be contiguous, and increment from this value. + If multiple values are given, one value must be given per channel. +- clocks : Must contain an entry for each required entry in clock-names. +- clock-names : Must include the following entries: + - Tegra30: Requires d_audio, apbif, i2s0, i2s1, i2s2, i2s3, i2s4, dam0, + dam1, dam2, spdif_in. + - Tegra114: Additionally requires amx, adx. - ranges : The bus address mapping for the configlink register bus. Can be empty since the mapping is 1:1. - #address-cells : For the configlink bus. Should be <1>; @@ -25,7 +35,13 @@ ahub@70080000 { reg = <0x70080000 0x200 0x70080200 0x100>; interrupts = < 0 103 0x04 >; nvidia,dma-request-selector = <&apbdma 1>; - + clocks = <&tegra_car 106>, <&tegra_car 107>, <&tegra_car 30>, + <&tegra_car 11>, <&tegra_car 18>, <&tegra_car 101>, + <&tegra_car 102>, <&tegra_car 108>, <&tegra_car 109>, + <&tegra_car 110>, <&tegra_car 162>; + clock-names = "d_audio", "apbif", "i2s0", "i2s1", "i2s2", + "i2s3", "i2s4", "dam0", "dam1", "dam2", + "spdif_in"; ranges; #address-cells = <1>; #size-cells = <1>; diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt new file mode 100644 index 000000000000..8ea4f5b4818d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -0,0 +1,32 @@ +Texas Instruments TAS5086 6-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,tas5086". + - reg: The i2c address. Should contain <0x1b>. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + + - ti,charge-period: This property should contain the time in microseconds + that closely matches the external single-ended + split-capacitor charge period. The hardware chip + waits for this period of time before starting the + PWM signals. This helps reduce pops and clicks. + + When not specified, the hardware default of 1300ms + is retained. + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + }; + }; diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index d4faa63ff352..c3c912d023cc 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -461,11 +461,13 @@ The generic parser supports the following hints: the corresponding mixer control, if available - add_stereo_mix_input (bool): add the stereo mix (analog-loopback mix) to the input mux if available -- add_out_jack_modes (bool): add "xxx Jack Mode" enum controls to each - output jack for allowing to change the headphone amp capability -- add_in_jack_modes (bool): add "xxx Jack Mode" enum controls to each - input jack for allowing to change the mic bias vref +- add_jack_modes (bool): add "xxx Jack Mode" enum controls to each + I/O jack for allowing to change the headphone amp and mic bias VREF + capabilities - power_down_unused (bool): power down the unused widgets +- add_hp_mic (bool): add the headphone to capture source if possible +- hp_mic_detect (bool): enable/disable the hp/mic shared input for a + single built-in mic case; default true - mixer_nid (int): specifies the widget NID of the analog-loopback mixer diff --git a/arch/arm/mach-s3c24xx/dma-s3c2410.c b/arch/arm/mach-s3c24xx/dma-s3c2410.c index a6c94b820954..30aa53ff07a6 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2410.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2410.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> static struct s3c24xx_dma_map __initdata s3c2410_dma_mappings[] = { diff --git a/arch/arm/mach-s3c24xx/dma-s3c2412.c b/arch/arm/mach-s3c24xx/dma-s3c2412.c index c0e8c3f5057e..ab1700ec8e64 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2412.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2412.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> #define MAP(x) { (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID } diff --git a/arch/arm/mach-s3c24xx/dma-s3c2440.c b/arch/arm/mach-s3c24xx/dma-s3c2440.c index 1c08eccd9425..cd25de28804c 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2440.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2440.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> static struct s3c24xx_dma_map __initdata s3c2440_dma_mappings[] = { diff --git a/arch/arm/mach-s3c24xx/dma-s3c2443.c b/arch/arm/mach-s3c24xx/dma-s3c2443.c index 000e4c69fce9..5fe3539dc2b5 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2443.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2443.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> #define MAP(x) { \ diff --git a/arch/arm/plat-samsung/devs.c b/arch/arm/plat-samsung/devs.c index 8f456bed4fa0..33ad3f32c2b9 100644 --- a/arch/arm/plat-samsung/devs.c +++ b/arch/arm/plat-samsung/devs.c @@ -146,14 +146,20 @@ struct platform_device s3c_device_camif = { /* ASOC DMA */ +#ifdef CONFIG_PLAT_S5P +static struct resource samsung_asoc_idma_resource = DEFINE_RES_IRQ(IRQ_I2S0); + struct platform_device samsung_asoc_idma = { .name = "samsung-idma", .id = -1, + .num_resources = 1, + .resource = &samsung_asoc_idma_resource, .dev = { .dma_mask = &samsung_device_dma_mask, .coherent_dma_mask = DMA_BIT_MASK(32), } }; +#endif /* FB */ diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index ca2aed6bc830..f70c4956ff9d 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -290,12 +290,14 @@ static const struct reg_default wm5102_reg_default[] = { { 0x00000176, 0x0000 }, /* R374 - FLL1 Control 6 */ { 0x00000177, 0x0181 }, /* R375 - FLL1 Loop Filter Test 1 */ { 0x00000178, 0x0000 }, /* R376 - FLL1 NCO Test 0 */ + { 0x00000179, 0x0000 }, /* R377 - FLL1 Control 7 */ { 0x00000181, 0x0000 }, /* R385 - FLL1 Synchroniser 1 */ { 0x00000182, 0x0000 }, /* R386 - FLL1 Synchroniser 2 */ { 0x00000183, 0x0000 }, /* R387 - FLL1 Synchroniser 3 */ { 0x00000184, 0x0000 }, /* R388 - FLL1 Synchroniser 4 */ { 0x00000185, 0x0000 }, /* R389 - FLL1 Synchroniser 5 */ { 0x00000186, 0x0000 }, /* R390 - FLL1 Synchroniser 6 */ + { 0x00000187, 0x0001 }, /* R391 - FLL1 Synchroniser 7 */ { 0x00000189, 0x0000 }, /* R393 - FLL1 Spread Spectrum */ { 0x0000018A, 0x0004 }, /* R394 - FLL1 GPIO Clock */ { 0x00000191, 0x0000 }, /* R401 - FLL2 Control 1 */ @@ -306,12 +308,14 @@ static const struct reg_default wm5102_reg_default[] = { { 0x00000196, 0x0000 }, /* R406 - FLL2 Control 6 */ { 0x00000197, 0x0000 }, /* R407 - FLL2 Loop Filter Test 1 */ { 0x00000198, 0x0000 }, /* R408 - FLL2 NCO Test 0 */ + { 0x00000199, 0x0000 }, /* R409 - FLL2 Control 7 */ { 0x000001A1, 0x0000 }, /* R417 - FLL2 Synchroniser 1 */ { 0x000001A2, 0x0000 }, /* R418 - FLL2 Synchroniser 2 */ { 0x000001A3, 0x0000 }, /* R419 - FLL2 Synchroniser 3 */ { 0x000001A4, 0x0000 }, /* R420 - FLL2 Synchroniser 4 */ { 0x000001A5, 0x0000 }, /* R421 - FLL2 Synchroniser 5 */ { 0x000001A6, 0x0000 }, /* R422 - FLL2 Synchroniser 6 */ + { 0x000001A7, 0x0001 }, /* R423 - FLL2 Synchroniser 7 */ { 0x000001A9, 0x0000 }, /* R425 - FLL2 Spread Spectrum */ { 0x000001AA, 0x0004 }, /* R426 - FLL2 GPIO Clock */ { 0x00000200, 0x0006 }, /* R512 - Mic Charge Pump 1 */ @@ -1055,12 +1059,14 @@ static bool wm5102_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL1_CONTROL_6: case ARIZONA_FLL1_LOOP_FILTER_TEST_1: case ARIZONA_FLL1_NCO_TEST_0: + case ARIZONA_FLL1_CONTROL_7: case ARIZONA_FLL1_SYNCHRONISER_1: case ARIZONA_FLL1_SYNCHRONISER_2: case ARIZONA_FLL1_SYNCHRONISER_3: case ARIZONA_FLL1_SYNCHRONISER_4: case ARIZONA_FLL1_SYNCHRONISER_5: case ARIZONA_FLL1_SYNCHRONISER_6: + case ARIZONA_FLL1_SYNCHRONISER_7: case ARIZONA_FLL1_SPREAD_SPECTRUM: case ARIZONA_FLL1_GPIO_CLOCK: case ARIZONA_FLL2_CONTROL_1: @@ -1071,12 +1077,14 @@ static bool wm5102_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL2_CONTROL_6: case ARIZONA_FLL2_LOOP_FILTER_TEST_1: case ARIZONA_FLL2_NCO_TEST_0: + case ARIZONA_FLL2_CONTROL_7: case ARIZONA_FLL2_SYNCHRONISER_1: case ARIZONA_FLL2_SYNCHRONISER_2: case ARIZONA_FLL2_SYNCHRONISER_3: case ARIZONA_FLL2_SYNCHRONISER_4: case ARIZONA_FLL2_SYNCHRONISER_5: case ARIZONA_FLL2_SYNCHRONISER_6: + case ARIZONA_FLL2_SYNCHRONISER_7: case ARIZONA_FLL2_SPREAD_SPECTRUM: case ARIZONA_FLL2_GPIO_CLOCK: case ARIZONA_MIC_CHARGE_PUMP_1: @@ -1169,6 +1177,8 @@ static bool wm5102_readable_register(struct device *dev, unsigned int reg) case ARIZONA_NOISE_GATE_CONTROL: case ARIZONA_PDM_SPK1_CTRL_1: case ARIZONA_PDM_SPK1_CTRL_2: + case ARIZONA_SPK_CTRL_2: + case ARIZONA_SPK_CTRL_3: case ARIZONA_DAC_COMP_1: case ARIZONA_DAC_COMP_2: case ARIZONA_DAC_COMP_3: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index f43aa7c8d040..715b6ba3d52a 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -85,12 +85,14 @@ #define ARIZONA_FLL1_CONTROL_6 0x176 #define ARIZONA_FLL1_LOOP_FILTER_TEST_1 0x177 #define ARIZONA_FLL1_NCO_TEST_0 0x178 +#define ARIZONA_FLL1_CONTROL_7 0x179 #define ARIZONA_FLL1_SYNCHRONISER_1 0x181 #define ARIZONA_FLL1_SYNCHRONISER_2 0x182 #define ARIZONA_FLL1_SYNCHRONISER_3 0x183 #define ARIZONA_FLL1_SYNCHRONISER_4 0x184 #define ARIZONA_FLL1_SYNCHRONISER_5 0x185 #define ARIZONA_FLL1_SYNCHRONISER_6 0x186 +#define ARIZONA_FLL1_SYNCHRONISER_7 0x187 #define ARIZONA_FLL1_SPREAD_SPECTRUM 0x189 #define ARIZONA_FLL1_GPIO_CLOCK 0x18A #define ARIZONA_FLL2_CONTROL_1 0x191 @@ -101,12 +103,14 @@ #define ARIZONA_FLL2_CONTROL_6 0x196 #define ARIZONA_FLL2_LOOP_FILTER_TEST_1 0x197 #define ARIZONA_FLL2_NCO_TEST_0 0x198 +#define ARIZONA_FLL2_CONTROL_7 0x199 #define ARIZONA_FLL2_SYNCHRONISER_1 0x1A1 #define ARIZONA_FLL2_SYNCHRONISER_2 0x1A2 #define ARIZONA_FLL2_SYNCHRONISER_3 0x1A3 #define ARIZONA_FLL2_SYNCHRONISER_4 0x1A4 #define ARIZONA_FLL2_SYNCHRONISER_5 0x1A5 #define ARIZONA_FLL2_SYNCHRONISER_6 0x1A6 +#define ARIZONA_FLL2_SYNCHRONISER_7 0x1A7 #define ARIZONA_FLL2_SPREAD_SPECTRUM 0x1A9 #define ARIZONA_FLL2_GPIO_CLOCK 0x1AA #define ARIZONA_MIC_CHARGE_PUMP_1 0x200 @@ -217,6 +221,8 @@ #define ARIZONA_PDM_SPK1_CTRL_2 0x491 #define ARIZONA_PDM_SPK2_CTRL_1 0x492 #define ARIZONA_PDM_SPK2_CTRL_2 0x493 +#define ARIZONA_SPK_CTRL_2 0x4B5 +#define ARIZONA_SPK_CTRL_3 0x4B6 #define ARIZONA_DAC_COMP_1 0x4DC #define ARIZONA_DAC_COMP_2 0x4DD #define ARIZONA_DAC_COMP_3 0x4DE @@ -1682,6 +1688,13 @@ #define ARIZONA_FLL1_FRC_INTEG_VAL_WIDTH 12 /* FLL1_FRC_INTEG_VAL - [11:0] */ /* + * R377 (0x179) - FLL1 Control 7 + */ +#define ARIZONA_FLL1_GAIN_MASK 0x003c /* FLL1_GAIN */ +#define ARIZONA_FLL1_GAIN_SHIFT 2 /* FLL1_GAIN */ +#define ARIZONA_FLL1_GAIN_WIDTH 4 /* FLL1_GAIN */ + +/* * R385 (0x181) - FLL1 Synchroniser 1 */ #define ARIZONA_FLL1_SYNC_ENA 0x0001 /* FLL1_SYNC_ENA */ @@ -1728,6 +1741,17 @@ #define ARIZONA_FLL1_CLK_SYNC_SRC_WIDTH 4 /* FLL1_CLK_SYNC_SRC - [3:0] */ /* + * R391 (0x187) - FLL1 Synchroniser 7 + */ +#define ARIZONA_FLL1_SYNC_GAIN_MASK 0x003c /* FLL1_SYNC_GAIN */ +#define ARIZONA_FLL1_SYNC_GAIN_SHIFT 2 /* FLL1_SYNC_GAIN */ +#define ARIZONA_FLL1_SYNC_GAIN_WIDTH 4 /* FLL1_SYNC_GAIN */ +#define ARIZONA_FLL1_SYNC_BW 0x0001 /* FLL1_SYNC_BW */ +#define ARIZONA_FLL1_SYNC_BW_MASK 0x0001 /* FLL1_SYNC_BW */ +#define ARIZONA_FLL1_SYNC_BW_SHIFT 0 /* FLL1_SYNC_BW */ +#define ARIZONA_FLL1_SYNC_BW_WIDTH 1 /* FLL1_SYNC_BW */ + +/* * R393 (0x189) - FLL1 Spread Spectrum */ #define ARIZONA_FLL1_SS_AMPL_MASK 0x0030 /* FLL1_SS_AMPL - [5:4] */ @@ -1820,6 +1844,13 @@ #define ARIZONA_FLL2_FRC_INTEG_VAL_WIDTH 12 /* FLL2_FRC_INTEG_VAL - [11:0] */ /* + * R409 (0x199) - FLL2 Control 7 + */ +#define ARIZONA_FLL2_GAIN_MASK 0x003c /* FLL2_GAIN */ +#define ARIZONA_FLL2_GAIN_SHIFT 2 /* FLL2_GAIN */ +#define ARIZONA_FLL2_GAIN_WIDTH 4 /* FLL2_GAIN */ + +/* * R417 (0x1A1) - FLL2 Synchroniser 1 */ #define ARIZONA_FLL2_SYNC_ENA 0x0001 /* FLL2_SYNC_ENA */ @@ -1866,6 +1897,17 @@ #define ARIZONA_FLL2_CLK_SYNC_SRC_WIDTH 4 /* FLL2_CLK_SYNC_SRC - [3:0] */ /* + * R423 (0x1A7) - FLL2 Synchroniser 7 + */ +#define ARIZONA_FLL2_SYNC_GAIN_MASK 0x003c /* FLL2_SYNC_GAIN */ +#define ARIZONA_FLL2_SYNC_GAIN_SHIFT 2 /* FLL2_SYNC_GAIN */ +#define ARIZONA_FLL2_SYNC_GAIN_WIDTH 4 /* FLL2_SYNC_GAIN */ +#define ARIZONA_FLL2_SYNC_BW_MASK 0x0001 /* FLL2_SYNC_BW */ +#define ARIZONA_FLL2_SYNC_BW_MASK 0x0001 /* FLL2_SYNC_BW */ +#define ARIZONA_FLL2_SYNC_BW_SHIFT 0 /* FLL2_SYNC_BW */ +#define ARIZONA_FLL2_SYNC_BW_WIDTH 1 /* FLL2_SYNC_BW */ + +/* * R425 (0x1A9) - FLL2 Spread Spectrum */ #define ARIZONA_FLL2_SS_AMPL_MASK 0x0030 /* FLL2_SS_AMPL - [5:4] */ diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 8e21a094836d..68e776594889 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -17,6 +17,7 @@ #define WM8994_NUM_LDO 2 #define WM8994_NUM_GPIO 11 +#define WM8994_NUM_AIF 3 struct wm8994_ldo_pdata { /** GPIOs to enable regulator, 0 or less if not available */ @@ -215,6 +216,13 @@ struct wm8994_pdata { * system. */ bool spkmode_pu; + + /** + * Maximum number of channels clocks will be generated for, + * useful for systems where and I2S bus with multiple data + * lines is mastered. + */ + int max_channels_clocked[WM8994_NUM_AIF]; }; #endif diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index ed13053153f4..c5f2158ab00e 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -170,6 +170,8 @@ struct uac2_as_header_descriptor { __u8 iChannelNames; } __attribute__((packed)); +#define UAC2_FORMAT_TYPE_I_RAW_DATA (1 << 31) + /* 4.10.1.2 Class-Specific AS Isochronous Audio Data Endpoint Descriptor */ struct uac2_iso_endpoint_descriptor { diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index ff6c74153fa1..9031a26249b5 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -56,8 +56,6 @@ struct snd_compr_runtime { u64 buffer_size; u32 fragment_size; u32 fragments; - u64 hw_pointer; - u64 app_pointer; u64 total_bytes_available; u64 total_bytes_transferred; wait_queue_head_t sleep; @@ -121,7 +119,7 @@ struct snd_compr_ops { int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); - int (*copy)(struct snd_compr_stream *stream, const char __user *buf, + int (*copy)(struct snd_compr_stream *stream, char __user *buf, size_t count); int (*mmap)(struct snd_compr_stream *stream, struct vm_area_struct *vma); diff --git a/include/sound/control.h b/include/sound/control.h index 8332e865c759..34bc93d80d55 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -189,7 +189,6 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, * * Add a virtual slave control to the given master element created via * snd_ctl_create_virtual_master() beforehand. - * Returns zero if successful or a negative error code. * * All slaves must be the same type (returning the same information * via info callback). The function doesn't check it, so it's your @@ -199,6 +198,8 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, * at most two channels, * logarithmic volume control (dB level) thus no linear volume, * master can only attenuate the volume without gain + * + * Return: Zero if successful or a negative error code. */ static inline int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) @@ -219,6 +220,8 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) * When the control peeks the hardware values directly and the value * can be changed by other means than the put callback of the element, * this function should be used to keep the value always up-to-date. + * + * Return: Zero if successful or a negative error code. */ static inline int snd_ctl_add_slave_uncached(struct snd_kcontrol *master, diff --git a/include/sound/core.h b/include/sound/core.h index 7cede2d6aa86..5bfe5136441c 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -229,7 +229,7 @@ int snd_register_device_for_dev(int type, struct snd_card *card, * This function uses the card's device pointer to link to the * correct &struct device. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ static inline int snd_register_device(int type, struct snd_card *card, int dev, const struct file_operations *f_ops, @@ -379,18 +379,10 @@ void __snd_printk(unsigned int level, const char *file, int line, * snd_BUG_ON - debugging check macro * @cond: condition to evaluate * - * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition, - * and call WARN() and returns the value if it's non-zero. - * - * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given - * condition is ignored. - * - * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n. - * Thus, don't put any statement that influences on the code behavior, - * such as pre/post increment, to the argument of this macro. - * If you want to evaluate and give a warning, use standard WARN_ON(). + * Has the same behavior as WARN_ON when CONFIG_SND_DEBUG is set, + * otherwise just evaluates the conditional and returns the value. */ -#define snd_BUG_ON(cond) WARN((cond), "BUG? (%s)\n", __stringify(cond)) +#define snd_BUG_ON(cond) WARN_ON((cond)) #else /* !CONFIG_SND_DEBUG */ @@ -400,11 +392,11 @@ __printf(2, 3) static inline void _snd_printd(int level, const char *format, ...) {} #define snd_BUG() do { } while (0) -static inline int __snd_bug_on(int cond) -{ - return 0; -} -#define snd_BUG_ON(cond) __snd_bug_on(0 && (cond)) /* always false */ + +#define snd_BUG_ON(condition) ({ \ + int __ret_warn_on = !!(condition); \ + unlikely(__ret_warn_on); \ +}) #endif /* CONFIG_SND_DEBUG */ diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b877334bbb0f..f11c35cd5532 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -16,6 +16,7 @@ #define __SOUND_DMAENGINE_PCM_H__ #include <sound/pcm.h> +#include <sound/soc.h> #include <linux/dmaengine.h> /** @@ -32,9 +33,6 @@ snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream) return DMA_DEV_TO_MEM; } -void snd_dmaengine_pcm_set_data(struct snd_pcm_substream *substream, void *data); -void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream); - 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); @@ -42,9 +40,100 @@ 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); + struct dma_chan *chan); int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream); +int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, + dma_filter_fn filter_fn, void *filter_data); +int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream); + +struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, + void *filter_data); struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream); +/** + * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data + * @addr: Address of the DAI data source or destination register. + * @addr_width: Width of the DAI data source or destination register. + * @maxburst: Maximum number of words(note: words, as in units of the + * src_addr_width member, not bytes) that can be send to or received from the + * DAI in one burst. + * @slave_id: Slave requester id for the DMA channel. + * @filter_data: Custom DMA channel filter data, this will usually be used when + * requesting the DMA channel. + */ +struct snd_dmaengine_dai_dma_data { + dma_addr_t addr; + enum dma_slave_buswidth addr_width; + u32 maxburst; + unsigned int slave_id; + void *filter_data; +}; + +void snd_dmaengine_pcm_set_config_from_dai_data( + const struct snd_pcm_substream *substream, + const struct snd_dmaengine_dai_dma_data *dma_data, + struct dma_slave_config *config); + + +/* + * Try to request the DMA channel using compat_request_channel or + * compat_filter_fn if it couldn't be requested through devicetree. + */ +#define SND_DMAENGINE_PCM_FLAG_COMPAT BIT(0) +/* + * Don't try to request the DMA channels through devicetree. This flag only + * makes sense if SND_DMAENGINE_PCM_FLAG_COMPAT is set as well. + */ +#define SND_DMAENGINE_PCM_FLAG_NO_DT BIT(1) +/* + * The platforms dmaengine driver does not support reporting the amount of + * bytes that are still left to transfer. + */ +#define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2) +/* + * The PCM is half duplex and the DMA channel is shared between capture and + * playback. + */ +#define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) + +/** + * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM + * @prepare_slave_config: Callback used to fill in the DMA slave_config for a + * PCM substream. Will be called from the PCM drivers hwparams callback. + * @compat_request_channel: Callback to request a DMA channel for platforms + * which do not use devicetree. + * @compat_filter_fn: Will be used as the filter function when requesting a + * channel for platforms which do not use devicetree. The filter parameter + * will be the DAI's DMA data. + * @pcm_hardware: snd_pcm_hardware struct to be used for the PCM. + * @prealloc_buffer_size: Size of the preallocated audio buffer. + * + * Note: If both compat_request_channel and compat_filter_fn are set + * compat_request_channel will be used to request the channel and + * compat_filter_fn will be ignored. Otherwise the channel will be requested + * using dma_request_channel with compat_filter_fn as the filter function. + */ +struct snd_dmaengine_pcm_config { + int (*prepare_slave_config)(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config); + struct dma_chan *(*compat_request_channel)( + struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream); + dma_filter_fn compat_filter_fn; + + const struct snd_pcm_hardware *pcm_hardware; + unsigned int prealloc_buffer_size; +}; + +int snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, + unsigned int flags); +void snd_dmaengine_pcm_unregister(struct device *dev); + +int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config); + #endif diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index f841ba4bacb8..dfb42ca6d043 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1787,6 +1787,7 @@ struct snd_emu10k1 { unsigned int next_free_voice; const struct firmware *firmware; + const struct firmware *dock_fw; #ifdef CONFIG_PM_SLEEP unsigned int *saved_ptr; diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 5ec42dbd2308..b48792fe386b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -181,6 +181,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_G723_24_1B _SNDRV_PCM_FMTBIT(G723_24_1B) #define SNDRV_PCM_FMTBIT_G723_40 _SNDRV_PCM_FMTBIT(G723_40) #define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B) +#define SNDRV_PCM_FMTBIT_DSD_U8 _SNDRV_PCM_FMTBIT(DSD_U8) +#define SNDRV_PCM_FMTBIT_DSD_U16_LE _SNDRV_PCM_FMTBIT(DSD_U16_LE) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE @@ -659,7 +661,7 @@ static inline snd_pcm_sframes_t snd_pcm_capture_hw_avail(struct snd_pcm_runtime * * Checks whether enough free space is available on the playback buffer. * - * Returns non-zero if available, or zero if not. + * Return: Non-zero if available, or zero if not. */ static inline int snd_pcm_playback_ready(struct snd_pcm_substream *substream) { @@ -673,7 +675,7 @@ static inline int snd_pcm_playback_ready(struct snd_pcm_substream *substream) * * Checks whether enough capture data is available on the capture buffer. * - * Returns non-zero if available, or zero if not. + * Return: Non-zero if available, or zero if not. */ static inline int snd_pcm_capture_ready(struct snd_pcm_substream *substream) { @@ -685,10 +687,10 @@ static inline int snd_pcm_capture_ready(struct snd_pcm_substream *substream) * snd_pcm_playback_data - check whether any data exists on the playback buffer * @substream: the pcm substream instance * - * Checks whether any data exists on the playback buffer. If stop_threshold - * is bigger or equal to boundary, then this function returns always non-zero. + * Checks whether any data exists on the playback buffer. * - * Returns non-zero if exists, or zero if not. + * Return: Non-zero if any data exists, or zero if not. If stop_threshold + * is bigger or equal to boundary, then this function returns always non-zero. */ static inline int snd_pcm_playback_data(struct snd_pcm_substream *substream) { @@ -705,7 +707,7 @@ static inline int snd_pcm_playback_data(struct snd_pcm_substream *substream) * * Checks whether the playback buffer is empty. * - * Returns non-zero if empty, or zero if not. + * Return: Non-zero if empty, or zero if not. */ static inline int snd_pcm_playback_empty(struct snd_pcm_substream *substream) { @@ -719,7 +721,7 @@ static inline int snd_pcm_playback_empty(struct snd_pcm_substream *substream) * * Checks whether the capture buffer is empty. * - * Returns non-zero if empty, or zero if not. + * Return: Non-zero if empty, or zero if not. */ static inline int snd_pcm_capture_empty(struct snd_pcm_substream *substream) { @@ -852,7 +854,7 @@ int snd_pcm_format_big_endian(snd_pcm_format_t format); * snd_pcm_format_cpu_endian - Check the PCM format is CPU-endian * @format: the format to check * - * Returns 1 if the given PCM format is CPU-endian, 0 if + * Return: 1 if the given PCM format is CPU-endian, 0 if * opposite, or a negative error code if endian not specified. */ int snd_pcm_format_cpu_endian(snd_pcm_format_t format); @@ -963,7 +965,7 @@ struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, * contiguous in kernel virtual space, but not in physical memory. Use this * if the buffer is accessed by kernel code but not by device DMA. * - * Returns 1 if the buffer was changed, 0 if not changed, or a negative error + * Return: 1 if the buffer was changed, 0 if not changed, or a negative error * code. */ static int snd_pcm_lib_alloc_vmalloc_buffer @@ -975,6 +977,9 @@ static int snd_pcm_lib_alloc_vmalloc_buffer * * This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses * vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory. + * + * Return: 1 if the buffer was changed, 0 if not changed, or a negative error + * code. */ static int snd_pcm_lib_alloc_vmalloc_32_buffer (struct snd_pcm_substream *substream, size_t size); @@ -1070,6 +1075,8 @@ const char *snd_pcm_format_name(snd_pcm_format_t format); /** * snd_pcm_stream_str - Get a string naming the direction of a stream * @substream: the pcm substream instance + * + * Return: A string naming the direction of the stream. */ static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) { @@ -1126,4 +1133,10 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, unsigned long private_value, struct snd_pcm_chmap **info_ret); +/* Strong-typed conversion of pcm_format to bitwise */ +static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format) +{ + return 1ULL << (__force int) pcm_format; +} + #endif /* __SOUND_PCM_H */ diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3d84808952b9..ae9a227d35d3 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -95,14 +95,6 @@ struct snd_soc_dai_driver; struct snd_soc_dai; struct snd_ac97_bus_ops; -/* Digital Audio Interface registration */ -int snd_soc_register_dai(struct device *dev, - struct snd_soc_dai_driver *dai_drv); -void snd_soc_unregister_dai(struct device *dev); -int snd_soc_register_dais(struct device *dev, - struct snd_soc_dai_driver *dai_drv, size_t count); -void snd_soc_unregister_dais(struct device *dev, size_t count); - /* Digital Audio Interface clocking API.*/ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir); diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 44a30b108683..d4609029f014 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -566,7 +566,6 @@ struct snd_soc_dapm_update { /* DAPM context */ struct snd_soc_dapm_context { - int n_widgets; /* number of widgets in this context */ enum snd_soc_bias_level bias_level; enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059ca3874..85c15226103b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -324,6 +324,8 @@ struct snd_soc_dai_link; struct snd_soc_platform_driver; struct snd_soc_codec; struct snd_soc_codec_driver; +struct snd_soc_component; +struct snd_soc_component_driver; struct soc_enum; struct snd_soc_jack; struct snd_soc_jack_zone; @@ -371,12 +373,20 @@ int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); int snd_soc_poweroff(struct device *dev); int snd_soc_register_platform(struct device *dev, - struct snd_soc_platform_driver *platform_drv); + const struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); +int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, + const struct snd_soc_platform_driver *platform_drv); +void snd_soc_remove_platform(struct snd_soc_platform *platform); +struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev); int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_codec(struct device *dev); +int snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); +void snd_soc_unregister_component(struct device *dev); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_codec_readable_register(struct snd_soc_codec *codec, @@ -801,10 +811,10 @@ struct snd_soc_platform_driver { struct snd_soc_dai *); /* platform stream pcm ops */ - struct snd_pcm_ops *ops; + const struct snd_pcm_ops *ops; /* platform stream compress ops */ - struct snd_compr_ops *compr_ops; + const struct snd_compr_ops *compr_ops; /* platform stream completion event */ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); @@ -823,7 +833,7 @@ struct snd_soc_platform { const char *name; int id; struct device *dev; - struct snd_soc_platform_driver *driver; + const struct snd_soc_platform_driver *driver; struct mutex mutex; unsigned int suspended:1; /* platform is suspended */ @@ -841,6 +851,20 @@ struct snd_soc_platform { #endif }; +struct snd_soc_component_driver { + const char *name; +}; + +struct snd_soc_component { + const char *name; + int id; + int num_dai; + struct device *dev; + struct list_head list; + + const struct snd_soc_component_driver *driver; +}; + struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ @@ -1086,7 +1110,6 @@ struct soc_enum { unsigned int mask; const char * const *texts; const unsigned int *values; - void *dapm; }; /* codec IO */ diff --git a/include/sound/tas5086.h b/include/sound/tas5086.h new file mode 100644 index 000000000000..aac481b7db8f --- /dev/null +++ b/include/sound/tas5086.h @@ -0,0 +1,7 @@ +#ifndef _SND_SOC_CODEC_TAS5086_H_ +#define _SND_SOC_CODEC_TAS5086_H_ + +#define TAS5086_CLK_IDX_MCLK 0 +#define TAS5086_CLK_IDX_SCLK 1 + +#endif /* _SND_SOC_CODEC_TAS5086_H_ */ diff --git a/include/sound/tegra_wm8903.h b/include/sound/tegra_wm8903.h deleted file mode 100644 index 57b202ee97c3..000000000000 --- a/include/sound/tegra_wm8903.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * Copyright 2011 NVIDIA, Inc. - * - * This software is licensed under the terms of the GNU General Public - * License version 2, as published by the Free Software Foundation, and - * may be copied, distributed, and modified under those terms. - * - * 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. - * - */ - -#ifndef __SOUND_TEGRA_WM38903_H -#define __SOUND_TEGRA_WM38903_H - -struct tegra_wm8903_platform_data { - int gpio_spkr_en; - int gpio_hp_det; - int gpio_hp_mute; - int gpio_int_mic_en; - int gpio_ext_mic_en; -}; - -#endif diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1774a5c3ef10..e3983d508272 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,7 +214,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B +#define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ +#define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U16_LE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 19491ed9292f..7b74a4ba75f8 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -179,7 +179,7 @@ static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in) */ if (other->active) { /* FIXME: is this guaranteed by the alsa api? */ - hw->formats &= (1ULL << i2sdev->format); + hw->formats &= pcm_format_to_bits(i2sdev->format); /* see above, restrict rates to the one we already have */ hw->rate_min = i2sdev->rate; hw->rate_max = i2sdev->rate; diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 79d6bda58753..6b7e2b5a72de 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -182,7 +182,7 @@ static int atmel_ac97c_playback_open(struct snd_pcm_substream *substream) runtime->hw.rate_max = chip->cur_rate; } if (chip->cur_format) - runtime->hw.formats = (1ULL << chip->cur_format); + runtime->hw.formats = pcm_format_to_bits(chip->cur_format); mutex_unlock(&opened_mutex); chip->playback_substream = substream; return 0; @@ -201,7 +201,7 @@ static int atmel_ac97c_capture_open(struct snd_pcm_substream *substream) runtime->hw.rate_max = chip->cur_rate; } if (chip->cur_format) - runtime->hw.formats = (1ULL << chip->cur_format); + runtime->hw.formats = pcm_format_to_bits(chip->cur_format); mutex_unlock(&opened_mutex); chip->capture_substream = substream; return 0; diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c84abc886e90..99db892d7299 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -28,11 +28,13 @@ #include <linux/file.h> #include <linux/fs.h> #include <linux/list.h> +#include <linux/math64.h> #include <linux/mm.h> #include <linux/mutex.h> #include <linux/poll.h> #include <linux/slab.h> #include <linux/sched.h> +#include <linux/types.h> #include <linux/uio.h> #include <linux/uaccess.h> #include <linux/module.h> @@ -152,26 +154,23 @@ static int snd_compr_update_tstamp(struct snd_compr_stream *stream, stream->ops->pointer(stream, tstamp); pr_debug("dsp consumed till %d total %d bytes\n", tstamp->byte_offset, tstamp->copied_total); - stream->runtime->hw_pointer = tstamp->byte_offset; - stream->runtime->total_bytes_transferred = tstamp->copied_total; + if (stream->direction == SND_COMPRESS_PLAYBACK) + stream->runtime->total_bytes_transferred = tstamp->copied_total; + else + stream->runtime->total_bytes_available = tstamp->copied_total; return 0; } static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, struct snd_compr_avail *avail) { - long avail_calc; /*this needs to be signed variable */ - memset(avail, 0, sizeof(*avail)); snd_compr_update_tstamp(stream, &avail->tstamp); /* Still need to return avail even if tstamp can't be filled in */ - /* FIXME: This needs to be different for capture stream, - available is # of compressed data, for playback it's - remainder of buffer */ - if (stream->runtime->total_bytes_available == 0 && - stream->runtime->state == SNDRV_PCM_STATE_SETUP) { + stream->runtime->state == SNDRV_PCM_STATE_SETUP && + stream->direction == SND_COMPRESS_PLAYBACK) { pr_debug("detected init and someone forgot to do a write\n"); return stream->runtime->buffer_size; } @@ -180,26 +179,22 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, stream->runtime->total_bytes_transferred); if (stream->runtime->total_bytes_available == stream->runtime->total_bytes_transferred) { - pr_debug("both pointers are same, returning full avail\n"); - return stream->runtime->buffer_size; + if (stream->direction == SND_COMPRESS_PLAYBACK) { + pr_debug("both pointers are same, returning full avail\n"); + return stream->runtime->buffer_size; + } else { + pr_debug("both pointers are same, returning no avail\n"); + return 0; + } } - /* FIXME: this routine isn't consistent, in one test we use - * cumulative values and in the other byte offsets. Do we - * really need the byte offsets if the cumulative values have - * been updated? In the PCM interface app_ptr and hw_ptr are - * already cumulative */ + avail->avail = stream->runtime->total_bytes_available - + stream->runtime->total_bytes_transferred; + if (stream->direction == SND_COMPRESS_PLAYBACK) + avail->avail = stream->runtime->buffer_size - avail->avail; - avail_calc = stream->runtime->buffer_size - - (stream->runtime->app_pointer - stream->runtime->hw_pointer); - pr_debug("calc avail as %ld, app_ptr %lld, hw+ptr %lld\n", avail_calc, - stream->runtime->app_pointer, - stream->runtime->hw_pointer); - if (avail_calc >= stream->runtime->buffer_size) - avail_calc -= stream->runtime->buffer_size; - pr_debug("ret avail as %ld\n", avail_calc); - avail->avail = avail_calc; - return avail_calc; + pr_debug("ret avail as %lld\n", avail->avail); + return avail->avail; } static inline size_t snd_compr_get_avail(struct snd_compr_stream *stream) @@ -230,21 +225,24 @@ static int snd_compr_write_data(struct snd_compr_stream *stream, void *dstn; size_t copy; struct snd_compr_runtime *runtime = stream->runtime; + /* 64-bit Modulus */ + u64 app_pointer = div64_u64(runtime->total_bytes_available, + runtime->buffer_size); + app_pointer = runtime->total_bytes_available - + (app_pointer * runtime->buffer_size); - dstn = runtime->buffer + runtime->app_pointer; + dstn = runtime->buffer + app_pointer; pr_debug("copying %ld at %lld\n", - (unsigned long)count, runtime->app_pointer); - if (count < runtime->buffer_size - runtime->app_pointer) { + (unsigned long)count, app_pointer); + if (count < runtime->buffer_size - app_pointer) { if (copy_from_user(dstn, buf, count)) return -EFAULT; - runtime->app_pointer += count; } else { - copy = runtime->buffer_size - runtime->app_pointer; + copy = runtime->buffer_size - app_pointer; if (copy_from_user(dstn, buf, copy)) return -EFAULT; if (copy_from_user(runtime->buffer, buf + copy, count - copy)) return -EFAULT; - runtime->app_pointer = count - copy; } /* if DSP cares, let it know data has been written */ if (stream->ops->ack) @@ -278,10 +276,12 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, if (avail > count) avail = count; - if (stream->ops->copy) - retval = stream->ops->copy(stream, buf, avail); - else + if (stream->ops->copy) { + char __user* cbuf = (char __user*)buf; + retval = stream->ops->copy(stream, cbuf, avail); + } else { retval = snd_compr_write_data(stream, buf, avail); + } if (retval > 0) stream->runtime->total_bytes_available += retval; @@ -300,7 +300,49 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, static ssize_t snd_compr_read(struct file *f, char __user *buf, size_t count, loff_t *offset) { - return -ENXIO; + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + size_t avail; + int retval; + + if (snd_BUG_ON(!data)) + return -EFAULT; + + stream = &data->stream; + mutex_lock(&stream->device->lock); + + /* read is allowed when stream is running, paused, draining and setup + * (yes setup is state which we transition to after stop, so if user + * wants to read data after stop we allow that) + */ + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_XRUN: + case SNDRV_PCM_STATE_SUSPENDED: + case SNDRV_PCM_STATE_DISCONNECTED: + retval = -EBADFD; + goto out; + } + + avail = snd_compr_get_avail(stream); + pr_debug("avail returned %ld\n", (unsigned long)avail); + /* calculate how much we can read from buffer */ + if (avail > count) + avail = count; + + if (stream->ops->copy) { + retval = stream->ops->copy(stream, buf, avail); + } else { + retval = -ENXIO; + goto out; + } + if (retval > 0) + stream->runtime->total_bytes_transferred += retval; + +out: + mutex_unlock(&stream->device->lock); + return retval; } static int snd_compr_mmap(struct file *f, struct vm_area_struct *vma) @@ -375,6 +417,7 @@ snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg) if (!stream->ops->get_caps) return -ENXIO; + memset(&caps, 0, sizeof(caps)); retval = stream->ops->get_caps(stream, &caps); if (retval) goto out; @@ -393,7 +436,7 @@ snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg) if (!stream->ops->get_codec_caps) return -ENXIO; - caps = kmalloc(sizeof(*caps), GFP_KERNEL); + caps = kzalloc(sizeof(*caps), GFP_KERNEL); if (!caps) return -ENOMEM; @@ -485,9 +528,14 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) retval = stream->ops->set_params(stream, params); if (retval) goto out; - stream->runtime->state = SNDRV_PCM_STATE_SETUP; + stream->metadata_set = false; stream->next_track = false; + + if (stream->direction == SND_COMPRESS_PLAYBACK) + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + else + stream->runtime->state = SNDRV_PCM_STATE_PREPARED; } else { return -EPERM; } @@ -505,7 +553,7 @@ snd_compr_get_params(struct snd_compr_stream *stream, unsigned long arg) if (!stream->ops->get_params) return -EBADFD; - params = kmalloc(sizeof(*params), GFP_KERNEL); + params = kzalloc(sizeof(*params), GFP_KERNEL); if (!params) return -ENOMEM; retval = stream->ops->get_params(stream, params); @@ -622,8 +670,6 @@ static int snd_compr_stop(struct snd_compr_stream *stream) if (!retval) { stream->runtime->state = SNDRV_PCM_STATE_SETUP; wake_up(&stream->runtime->sleep); - stream->runtime->hw_pointer = 0; - stream->runtime->app_pointer = 0; stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; } diff --git a/sound/core/control.c b/sound/core/control.c index 8c7c2c9bba61..d8aa206e8bde 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -190,7 +190,7 @@ EXPORT_SYMBOL(snd_ctl_notify); * Allocates a new struct snd_kcontrol instance and copies the given template * to the new instance. It does not copy volatile data (access). * - * Returns the pointer of the new instance, or NULL on failure. + * Return: The pointer of the new instance, or %NULL on failure. */ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, unsigned int access) @@ -224,7 +224,7 @@ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, * template. When the access field of ncontrol is 0, it's assumed as * READWRITE access. When the count field is 0, it's assumes as one. * - * Returns the pointer of the newly generated instance, or NULL on failure. + * Return: The pointer of the newly generated instance, or %NULL on failure. */ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, void *private_data) @@ -322,9 +322,10 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) * snd_ctl_new1() to the given card. Assigns also an unique * numid used for fast search. * - * Returns zero if successful, or a negative error code on failure. - * * It frees automatically the control which cannot be added. + * + * Return: Zero if successful, or a negative error code on failure. + * */ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) { @@ -380,9 +381,9 @@ EXPORT_SYMBOL(snd_ctl_add); * and the add_on_replace flag is set, the control is added. If the * control exists, it is destroyed first. * - * Returns zero if successful, or a negative error code on failure. - * * It frees automatically the control which cannot be added or replaced. + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, bool add_on_replace) @@ -442,8 +443,8 @@ EXPORT_SYMBOL(snd_ctl_replace); * Removes the control from the card and then releases the instance. * You don't need to call snd_ctl_free_one(). You must be in * the write lock - down_write(&card->controls_rwsem). - * - * Returns 0 if successful, or a negative error code on failure. + * + * Return: 0 if successful, or a negative error code on failure. */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { @@ -470,8 +471,8 @@ EXPORT_SYMBOL(snd_ctl_remove); * * Finds the control instance with the given id, removes it from the * card list and releases it. - * - * Returns 0 if successful, or a negative error code on failure. + * + * Return: 0 if successful, or a negative error code on failure. */ int snd_ctl_remove_id(struct snd_card *card, struct snd_ctl_elem_id *id) { @@ -498,8 +499,8 @@ EXPORT_SYMBOL(snd_ctl_remove_id); * * Finds the control instance with the given id, removes it from the * card list and releases it. - * - * Returns 0 if successful, or a negative error code on failure. + * + * Return: 0 if successful, or a negative error code on failure. */ static int snd_ctl_remove_user_ctl(struct snd_ctl_file * file, struct snd_ctl_elem_id *id) @@ -541,7 +542,7 @@ error: * Finds the control instance with the given id, and activate or * inactivate the control together with notification, if changed. * - * Returns 0 if unchanged, 1 if changed, or a negative error code on failure. + * Return: 0 if unchanged, 1 if changed, or a negative error code on failure. */ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, int active) @@ -587,7 +588,7 @@ EXPORT_SYMBOL_GPL(snd_ctl_activate_id); * Finds the control with the old id from the card, and replaces the * id with the new one. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id) @@ -616,10 +617,11 @@ EXPORT_SYMBOL(snd_ctl_rename_id); * * Finds the control instance with the given number-id from the card. * - * Returns the pointer of the instance if found, or NULL if not. - * * The caller must down card->controls_rwsem before calling this function * (if the race condition can happen). + * + * Return: The pointer of the instance if found, or %NULL if not. + * */ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { @@ -643,10 +645,11 @@ EXPORT_SYMBOL(snd_ctl_find_numid); * * Finds the control instance with the given id from the card. * - * Returns the pointer of the instance if found, or NULL if not. - * * The caller must down card->controls_rwsem before calling this function * (if the race condition can happen). + * + * Return: The pointer of the instance if found, or %NULL if not. + * */ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, struct snd_ctl_elem_id *id) @@ -1710,6 +1713,8 @@ EXPORT_SYMBOL(snd_ctl_boolean_stereo_info); * Sets all required fields in @info to their appropriate values. * If the control's accessibility is not the default (readable and writable), * the caller has to fill @info->access. + * + * Return: Zero. */ int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, unsigned int items, const char *const names[]) diff --git a/sound/core/device.c b/sound/core/device.c index f03cb5444a5a..df88defed176 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -39,7 +39,7 @@ * The data pointer plays a role as the identifier, too, so the * pointer address must be unique and unchanged. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_device_new(struct snd_card *card, snd_device_type_t type, void *device_data, struct snd_device_ops *ops) @@ -73,7 +73,7 @@ EXPORT_SYMBOL(snd_device_new); * callbacks, dev_disconnect and dev_free, corresponding to the state. * Then release the device. * - * Returns zero if successful, or a negative error code on failure or if the + * Return: Zero if successful, or a negative error code on failure or if the * device not found. */ int snd_device_free(struct snd_card *card, void *device_data) @@ -116,7 +116,7 @@ EXPORT_SYMBOL(snd_device_free); * * Usually called from snd_card_disconnect(). * - * Returns zero if successful, or a negative error code on failure or if the + * Return: Zero if successful, or a negative error code on failure or if the * device not found. */ int snd_device_disconnect(struct snd_card *card, void *device_data) @@ -151,7 +151,7 @@ int snd_device_disconnect(struct snd_card *card, void *device_data) * but it can be called later if any new devices are created after * invocation of snd_card_register(). * - * Returns zero if successful, or a negative error code on failure or if the + * Return: Zero if successful, or a negative error code on failure or if the * device not found. */ int snd_device_register(struct snd_card *card, void *device_data) diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 3f7f6628cf7b..d105073298cb 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -356,7 +356,7 @@ static const struct file_operations snd_hwdep_f_ops = * The callbacks (hwdep->ops) must be set on the returned instance * after this call manually by the caller. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_hwdep_new(struct snd_card *card, char *id, int device, struct snd_hwdep **rhwdep) diff --git a/sound/core/info.c b/sound/core/info.c index 3c9bd6b10a96..e79baa11b60e 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -89,7 +89,7 @@ static int resize_info_buffer(struct snd_info_buffer *buffer, char *nbuf; nsize = PAGE_ALIGN(nsize); - nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL); + nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL | __GFP_ZERO); if (! nbuf) return -ENOMEM; @@ -105,7 +105,7 @@ static int resize_info_buffer(struct snd_info_buffer *buffer, * * Outputs the string on the procfs buffer just like printf(). * - * Returns the size of output string. + * Return: The size of output string, or a negative error code. */ int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...) { @@ -344,7 +344,7 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) goto __nomem; data->rbuffer = buffer; buffer->len = PAGE_SIZE; - buffer->buffer = kmalloc(buffer->len, GFP_KERNEL); + buffer->buffer = kzalloc(buffer->len, GFP_KERNEL); if (buffer->buffer == NULL) goto __nomem; } @@ -683,32 +683,27 @@ int snd_info_card_free(struct snd_card *card) * * Reads one line from the buffer and stores the string. * - * Returns zero if successful, or 1 if error or EOF. + * Return: Zero if successful, or 1 if error or EOF. */ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { int c = -1; + if (snd_BUG_ON(!buffer || !buffer->buffer)) + return 1; if (len <= 0 || buffer->stop || buffer->error) return 1; - while (--len > 0) { + while (!buffer->stop) { c = buffer->buffer[buffer->curr++]; - if (c == '\n') { - if (buffer->curr >= buffer->size) - buffer->stop = 1; - break; - } - *line++ = c; - if (buffer->curr >= buffer->size) { + if (buffer->curr >= buffer->size) buffer->stop = 1; + if (c == '\n') break; + if (len) { + len--; + *line++ = c; } } - while (c != '\n' && !buffer->stop) { - c = buffer->buffer[buffer->curr++]; - if (buffer->curr >= buffer->size) - buffer->stop = 1; - } *line = '\0'; return 0; } @@ -724,7 +719,7 @@ EXPORT_SYMBOL(snd_info_get_line); * Parses the original string and copy a token to the given * string buffer. * - * Returns the updated pointer of the original string so that + * Return: The updated pointer of the original string so that * it can be used for the next call. */ const char *snd_info_get_str(char *dest, const char *src, int len) @@ -763,7 +758,7 @@ EXPORT_SYMBOL(snd_info_get_str); * Usually called from other functions such as * snd_info_create_card_entry(). * - * Returns the pointer of the new instance, or NULL on failure. + * Return: The pointer of the new instance, or %NULL on failure. */ static struct snd_info_entry *snd_info_create_entry(const char *name) { @@ -792,7 +787,7 @@ static struct snd_info_entry *snd_info_create_entry(const char *name) * * Creates a new info entry and assigns it to the given module. * - * Returns the pointer of the new instance, or NULL on failure. + * Return: The pointer of the new instance, or %NULL on failure. */ struct snd_info_entry *snd_info_create_module_entry(struct module * module, const char *name, @@ -816,7 +811,7 @@ EXPORT_SYMBOL(snd_info_create_module_entry); * * Creates a new info entry and assigns it to the given card. * - * Returns the pointer of the new instance, or NULL on failure. + * Return: The pointer of the new instance, or %NULL on failure. */ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, @@ -882,7 +877,7 @@ static int snd_info_dev_register_entry(struct snd_device *device) * For releasing this entry, use snd_device_free() instead of * snd_info_free_entry(). * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp) @@ -938,7 +933,7 @@ EXPORT_SYMBOL(snd_info_free_entry); * * Registers the proc info entry. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_info_register(struct snd_info_entry * entry) { diff --git a/sound/core/init.c b/sound/core/init.c index 7b012d15c2cf..6ef06400dfc8 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -144,7 +144,7 @@ static inline int init_info_for_card(struct snd_card *card) * space for the driver to use freely. The allocated struct is stored * in the given card_ret pointer. * - * Returns zero if successful or a negative error code. + * Return: Zero if successful or a negative error code. */ int snd_card_create(int idx, const char *xid, struct module *module, int extra_size, @@ -337,7 +337,7 @@ static const struct file_operations snd_shutdown_f_ops = * * Disconnects all APIs from the file-operations (user space). * - * Returns zero, otherwise a negative error code. + * Return: Zero, otherwise a negative error code. * * Note: The current implementation replaces all active file->f_op with special * dummy file operations (they do nothing except release). @@ -415,7 +415,7 @@ EXPORT_SYMBOL(snd_card_disconnect); * devices automatically. That is, you don't have to release the devices * by yourself. * - * Returns zero. Frees all associated devices and frees the control + * Return: Zero. Frees all associated devices and frees the control * interface associated to given soundcard. */ static int snd_card_do_free(struct snd_card *card) @@ -677,7 +677,7 @@ static struct device_attribute card_number_attrs = * external accesses. Thus, you should call this function at the end * of the initialization of the card. * - * Returns zero otherwise a negative error code if the registration failed. + * Return: Zero otherwise a negative error code if the registration failed. */ int snd_card_register(struct snd_card *card) { @@ -849,7 +849,7 @@ int __exit snd_card_info_done(void) * This function adds the component id string to the supported list. * The component can be referred from the alsa-lib. * - * Returns zero otherwise a negative error code. + * Return: Zero otherwise a negative error code. */ int snd_component_add(struct snd_card *card, const char *component) @@ -883,7 +883,7 @@ EXPORT_SYMBOL(snd_component_add); * This linked-list is used to keep tracking the connection state, * and to avoid the release of busy resources by hotplug. * - * Returns zero or a negative error code. + * Return: zero or a negative error code. */ int snd_card_file_add(struct snd_card *card, struct file *file) { @@ -920,7 +920,7 @@ EXPORT_SYMBOL(snd_card_file_add); * called beforehand, it processes the pending release of * resources. * - * Returns zero or a negative error code. + * Return: Zero or a negative error code. */ int snd_card_file_remove(struct snd_card *card, struct file *file) { @@ -959,6 +959,8 @@ EXPORT_SYMBOL(snd_card_file_remove); * * Waits until the power-state is changed. * + * Return: Zero if successful, or a negative error code. + * * Note: the power lock must be active before call. */ int snd_power_wait(struct snd_card *card, unsigned int power_state) diff --git a/sound/core/isadma.c b/sound/core/isadma.c index c0f1208bb7df..e2b386156a4c 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -81,7 +81,7 @@ EXPORT_SYMBOL(snd_dma_disable); * @dma: the dma number * @size: the dma transfer size * - * Returns the current pointer in DMA tranfer buffer in bytes + * Return: The current pointer in DMA transfer buffer in bytes. */ unsigned int snd_dma_pointer(unsigned long dma, unsigned int size) { diff --git a/sound/core/jack.c b/sound/core/jack.c index a06b1651fcba..b35fe7345c20 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -98,8 +98,8 @@ static int snd_jack_dev_register(struct snd_device *device) * * Creates a new jack object. * - * Returns zero if successful, or a negative error code on failure. - * On success jjack will be initialised. + * Return: Zero if successful, or a negative error code on failure. + * On success @jjack will be initialised. */ int snd_jack_new(struct snd_card *card, const char *id, int type, struct snd_jack **jjack) @@ -189,6 +189,8 @@ EXPORT_SYMBOL(snd_jack_set_parent); * using this abstraction. * * This function may only be called prior to registration of the jack. + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_jack_set_key(struct snd_jack *jack, enum snd_jack_types type, int keytype) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 691569238435..bdf826f4fe0c 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -81,7 +81,7 @@ static inline void dec_snd_pages(int order) * * Allocates the physically contiguous pages with the given size. * - * Returns the pointer of the buffer, or NULL if no enoguh memory. + * Return: The pointer of the buffer, or %NULL if no enough memory. */ void *snd_malloc_pages(size_t size, gfp_t gfp_flags) { @@ -175,9 +175,9 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, * * Calls the memory-allocator function for the corresponding * buffer type. - * - * Returns zero if the buffer with the given size is allocated successfully, - * other a negative value at error. + * + * Return: Zero if the buffer with the given size is allocated successfully, + * otherwise a negative value on error. */ int snd_dma_alloc_pages(int type, struct device *device, size_t size, struct snd_dma_buffer *dmab) @@ -229,9 +229,9 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, * buffer type. When no space is left, this function reduces the size and * tries to allocate again. The size actually allocated is stored in * res_size argument. - * - * Returns zero if the buffer with the given size is allocated successfully, - * other a negative value at error. + * + * Return: Zero if the buffer with the given size is allocated successfully, + * otherwise a negative value on error. */ int snd_dma_alloc_pages_fallback(int type, struct device *device, size_t size, struct snd_dma_buffer *dmab) @@ -292,7 +292,7 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) * Looks for the reserved-buffer list and re-uses if the same buffer * is found in the list. When the buffer is found, it's removed from the free list. * - * Returns the size of buffer if the buffer is found, or zero if not found. + * Return: The size of buffer if the buffer is found, or zero if not found. */ size_t snd_dma_get_reserved_buf(struct snd_dma_buffer *dmab, unsigned int id) { @@ -326,8 +326,8 @@ size_t snd_dma_get_reserved_buf(struct snd_dma_buffer *dmab, unsigned int id) * @id: the buffer id * * Reserves the given buffer as a reserved buffer. - * - * Returns zero if successful, or a negative code at error. + * + * Return: Zero if successful, or a negative code on error. */ int snd_dma_reserve_buf(struct snd_dma_buffer *dmab, unsigned int id) { diff --git a/sound/core/memory.c b/sound/core/memory.c index 66a278d0b04e..36c0f1a2e189 100644 --- a/sound/core/memory.c +++ b/sound/core/memory.c @@ -33,7 +33,7 @@ * * Copies the data from mmio-space to user-space. * - * Returns zero if successful, or non-zero on failure. + * Return: Zero if successful, or non-zero on failure. */ int copy_to_user_fromio(void __user *dst, const volatile void __iomem *src, size_t count) { @@ -66,7 +66,7 @@ EXPORT_SYMBOL(copy_to_user_fromio); * * Copies the data from user-space to mmio-space. * - * Returns zero if successful, or non-zero on failure. + * Return: Zero if successful, or non-zero on failure. */ int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size_t count) { diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 61798f85d030..17f45e8aa89c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -209,6 +209,8 @@ static char *snd_pcm_format_names[] = { FORMAT(G723_24_1B), FORMAT(G723_40), FORMAT(G723_40_1B), + FORMAT(DSD_U8), + FORMAT(DSD_U16_LE), }; const char *snd_pcm_format_name(snd_pcm_format_t format) @@ -637,7 +639,7 @@ static inline int snd_pcm_substream_proc_done(struct snd_pcm_substream *substrea * calling this, i.e. zero must be given to the argument of * snd_pcm_new(). * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) { @@ -759,7 +761,7 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, * The pcm operators have to be set afterwards to the new instance * via snd_pcm_set_ops(). * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_new(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, struct snd_pcm **rpcm) @@ -787,7 +789,7 @@ EXPORT_SYMBOL(snd_pcm_new); * The pcm operators have to be set afterwards to the new instance * via snd_pcm_set_ops(). * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index c4840ff75d00..41b3dfe68698 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -666,7 +666,8 @@ static inline unsigned int muldiv32(unsigned int a, unsigned int b, * The interval is changed to the range satisfying both intervals. * The interval status (min, max, integer, etc.) are evaluated. * - * Returns non-zero if the value is changed, zero if not changed. + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. */ int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v) { @@ -865,7 +866,8 @@ void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, * @nump: pointer to store the resultant numerator * @denp: pointer to store the resultant denominator * - * Returns non-zero if the value is changed, zero if not changed. + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. */ int snd_interval_ratnum(struct snd_interval *i, unsigned int rats_count, struct snd_ratnum *rats, @@ -983,7 +985,8 @@ EXPORT_SYMBOL(snd_interval_ratnum); * @nump: pointer to store the resultant numerator * @denp: pointer to store the resultant denominator * - * Returns non-zero if the value is changed, zero if not changed. + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. */ static int snd_interval_ratden(struct snd_interval *i, unsigned int rats_count, struct snd_ratden *rats, @@ -1082,7 +1085,8 @@ static int snd_interval_ratden(struct snd_interval *i, * When mask is non-zero, only the elements corresponding to bit 1 are * evaluated. * - * Returns non-zero if the value is changed, zero if not changed. + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. */ int snd_interval_list(struct snd_interval *i, unsigned int count, const unsigned int *list, unsigned int mask) @@ -1142,7 +1146,7 @@ static int snd_interval_step(struct snd_interval *i, unsigned int min, unsigned * @private: the private data pointer passed to function * @dep: the dependent variables * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, int var, @@ -1200,6 +1204,8 @@ EXPORT_SYMBOL(snd_pcm_hw_rule_add); * @mask: the bitmap mask * * Apply the constraint of the given bitmap mask to a 32-bit mask parameter. + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int32_t mask) @@ -1220,6 +1226,8 @@ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param * @mask: the 64bit bitmap mask * * Apply the constraint of the given bitmap mask to a 64-bit mask parameter. + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int64_t mask) @@ -1240,6 +1248,9 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par * @var: hw_params variable to apply the integer constraint * * Apply the constraint of integer to an interval parameter. + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. */ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var) { @@ -1257,6 +1268,9 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); * @max: the maximal value * * Apply the min/max range constraint to an interval parameter. + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. */ int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, unsigned int min, unsigned int max) @@ -1288,6 +1302,8 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, * @l: list * * Apply the list of constraints to an interval parameter. + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, @@ -1322,6 +1338,8 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, * @cond: condition bits * @var: hw_params variable to apply the ratnums constraint * @r: struct snd_ratnums constriants + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, @@ -1355,6 +1373,8 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, * @cond: condition bits * @var: hw_params variable to apply the ratdens constraint * @r: struct snd_ratdens constriants + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, @@ -1386,6 +1406,8 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, * @cond: condition bits * @width: sample bits width * @msbits: msbits width + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, unsigned int cond, @@ -1414,6 +1436,8 @@ static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params, * @cond: condition bits * @var: hw_params variable to apply the step constraint * @step: step size + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, unsigned int cond, @@ -1444,6 +1468,8 @@ static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the power-of-2 constraint + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, unsigned int cond, @@ -1470,6 +1496,8 @@ static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params, * snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling * @runtime: PCM runtime instance * @base_rate: the rate at which the hardware does not resample + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, unsigned int base_rate) @@ -1519,8 +1547,8 @@ EXPORT_SYMBOL(_snd_pcm_hw_params_any); * @var: parameter to retrieve * @dir: pointer to the direction (-1,0,1) or %NULL * - * Return the value for field @var if it's fixed in configuration space - * defined by @params. Return -%EINVAL otherwise. + * Return: The value for field @var if it's fixed in configuration space + * defined by @params. -%EINVAL otherwise. */ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) @@ -1591,7 +1619,8 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params, * * Inside configuration space defined by @params remove from @var all * values > minimum. Reduce configuration space accordingly. - * Return the minimum. + * + * Return: The minimum, or a negative error code on failure. */ int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, @@ -1637,7 +1666,8 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params, * * Inside configuration space defined by @params remove from @var all * values < maximum. Reduce configuration space accordingly. - * Return the maximum. + * + * Return: The maximum, or a negative error code on failure. */ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, @@ -1665,6 +1695,8 @@ EXPORT_SYMBOL(snd_pcm_hw_param_last); * The configuration chosen is that obtained fixing in this order: * first access, first format, first subformat, min channels, * min rate, min period time, max buffer size, min tick time + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params) @@ -1771,7 +1803,7 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, * Processes the generic ioctl commands for PCM. * Can be passed as the ioctl callback for PCM ops. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) @@ -2510,7 +2542,7 @@ static void pcm_chmap_ctl_private_free(struct snd_kcontrol *kcontrol) * @info_ret: store struct snd_pcm_chmap instance if non-NULL * * Create channel-mapping control elements assigned to the given PCM stream(s). - * Returns zero if succeed, or a negative error value. + * Return: Zero if successful, or a negative error value. */ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, const struct snd_pcm_chmap_elem *chmap, diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 69e01c4fc32d..0af622c34e19 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -95,7 +95,7 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream * * Releases the pre-allocated buffer of the given substream. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { @@ -115,7 +115,7 @@ int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) * * Releases all the pre-allocated buffers on the given pcm. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_lib_preallocate_free_for_all(struct snd_pcm *pcm) { @@ -265,7 +265,7 @@ static int snd_pcm_lib_preallocate_pages1(struct snd_pcm_substream *substream, * destruction time. The dma_buf_id must be unique for all systems * (in the same DMA buffer type) e.g. using snd_dma_pci_buf_id(). * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream, int type, struct device *data, @@ -289,7 +289,7 @@ EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages); * Do pre-allocation to all substreams of the given pcm for the * specified DMA type. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int type, void *data, @@ -313,8 +313,9 @@ EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all); * @substream: the pcm substream instance * @offset: the buffer offset * - * Returns the page struct at the given buffer offset. * Used as the page callback of PCM ops. + * + * Return: The page struct at the given buffer offset. %NULL on failure. */ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigned long offset) { @@ -337,7 +338,7 @@ EXPORT_SYMBOL(snd_pcm_sgbuf_ops_page); * Allocates the DMA buffer on the BUS type given earlier to * snd_pcm_lib_preallocate_xxx_pages(). * - * Returns 1 if the buffer is changed, 0 if not changed, or a negative + * Return: 1 if the buffer is changed, 0 if not changed, or a negative * code on failure. */ int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size) @@ -390,7 +391,7 @@ EXPORT_SYMBOL(snd_pcm_lib_malloc_pages); * * Releases the DMA buffer allocated via snd_pcm_lib_malloc_pages(). * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream) { @@ -437,6 +438,8 @@ EXPORT_SYMBOL(_snd_pcm_lib_alloc_vmalloc_buffer); * snd_pcm_lib_free_vmalloc_buffer - free vmalloc buffer * @substream: the substream with a buffer allocated by * snd_pcm_lib_alloc_vmalloc_buffer() + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream) { @@ -458,6 +461,8 @@ EXPORT_SYMBOL(snd_pcm_lib_free_vmalloc_buffer); * @offset: offset in the buffer * * This function is to be used as the page callback in the PCM ops. + * + * Return: The page struct, or %NULL on failure. */ struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, unsigned long offset) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index d4fc1bfbe457..43f24cce3dec 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -140,6 +140,14 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { .width = 5, .phys = 5, .le = -1, .signd = -1, .silence = {}, }, + [SNDRV_PCM_FORMAT_DSD_U8] = { + .width = 8, .phys = 8, .le = 1, .signd = 0, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_DSD_U16_LE] = { + .width = 16, .phys = 16, .le = 1, .signd = 0, + .silence = {}, + }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, @@ -213,7 +221,7 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { * snd_pcm_format_signed - Check the PCM format is signed linear * @format: the format to check * - * Returns 1 if the given PCM format is signed linear, 0 if unsigned + * Return: 1 if the given PCM format is signed linear, 0 if unsigned * linear, and a negative error code for non-linear formats. */ int snd_pcm_format_signed(snd_pcm_format_t format) @@ -232,7 +240,7 @@ EXPORT_SYMBOL(snd_pcm_format_signed); * snd_pcm_format_unsigned - Check the PCM format is unsigned linear * @format: the format to check * - * Returns 1 if the given PCM format is unsigned linear, 0 if signed + * Return: 1 if the given PCM format is unsigned linear, 0 if signed * linear, and a negative error code for non-linear formats. */ int snd_pcm_format_unsigned(snd_pcm_format_t format) @@ -251,7 +259,7 @@ EXPORT_SYMBOL(snd_pcm_format_unsigned); * snd_pcm_format_linear - Check the PCM format is linear * @format: the format to check * - * Returns 1 if the given PCM format is linear, 0 if not. + * Return: 1 if the given PCM format is linear, 0 if not. */ int snd_pcm_format_linear(snd_pcm_format_t format) { @@ -264,7 +272,7 @@ EXPORT_SYMBOL(snd_pcm_format_linear); * snd_pcm_format_little_endian - Check the PCM format is little-endian * @format: the format to check * - * Returns 1 if the given PCM format is little-endian, 0 if + * Return: 1 if the given PCM format is little-endian, 0 if * big-endian, or a negative error code if endian not specified. */ int snd_pcm_format_little_endian(snd_pcm_format_t format) @@ -283,7 +291,7 @@ EXPORT_SYMBOL(snd_pcm_format_little_endian); * snd_pcm_format_big_endian - Check the PCM format is big-endian * @format: the format to check * - * Returns 1 if the given PCM format is big-endian, 0 if + * Return: 1 if the given PCM format is big-endian, 0 if * little-endian, or a negative error code if endian not specified. */ int snd_pcm_format_big_endian(snd_pcm_format_t format) @@ -302,7 +310,7 @@ EXPORT_SYMBOL(snd_pcm_format_big_endian); * snd_pcm_format_width - return the bit-width of the format * @format: the format to check * - * Returns the bit-width of the format, or a negative error code + * Return: The bit-width of the format, or a negative error code * if unknown format. */ int snd_pcm_format_width(snd_pcm_format_t format) @@ -321,7 +329,7 @@ EXPORT_SYMBOL(snd_pcm_format_width); * snd_pcm_format_physical_width - return the physical bit-width of the format * @format: the format to check * - * Returns the physical bit-width of the format, or a negative error code + * Return: The physical bit-width of the format, or a negative error code * if unknown format. */ int snd_pcm_format_physical_width(snd_pcm_format_t format) @@ -341,7 +349,7 @@ EXPORT_SYMBOL(snd_pcm_format_physical_width); * @format: the format to check * @samples: sampling rate * - * Returns the byte size of the given samples for the format, or a + * Return: The byte size of the given samples for the format, or a * negative error code if unknown format. */ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) @@ -358,7 +366,7 @@ EXPORT_SYMBOL(snd_pcm_format_size); * snd_pcm_format_silence_64 - return the silent data in 8 bytes array * @format: the format to check * - * Returns the format pattern to fill or NULL if error. + * Return: The format pattern to fill or %NULL if error. */ const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format) { @@ -379,7 +387,7 @@ EXPORT_SYMBOL(snd_pcm_format_silence_64); * * Sets the silence data on the buffer for the given samples. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int samples) { @@ -449,7 +457,7 @@ EXPORT_SYMBOL(snd_pcm_format_set_silence); * Determines the rate_min and rate_max fields from the rates bits of * the given runtime->hw. * - * Returns zero if successful. + * Return: Zero if successful. */ int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) { @@ -475,7 +483,7 @@ EXPORT_SYMBOL(snd_pcm_limit_hw_rates); * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit * @rate: the sample rate to convert * - * Returns the SNDRV_PCM_RATE_xxx flag that corresponds to the given rate, or + * Return: The SNDRV_PCM_RATE_xxx flag that corresponds to the given rate, or * SNDRV_PCM_RATE_KNOT for an unknown rate. */ unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate) @@ -493,8 +501,8 @@ 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 + * Return: 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) { diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index eb560fa32321..23e3c46cd0a4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -898,6 +898,8 @@ static struct action_ops snd_pcm_action_start = { /** * snd_pcm_start - start all linked streams * @substream: the PCM substream instance + * + * Return: Zero if successful, or a negative error code. */ int snd_pcm_start(struct snd_pcm_substream *substream) { @@ -951,6 +953,8 @@ static struct action_ops snd_pcm_action_stop = { * @state: PCM state after stopping the stream * * The state of each stream is then changed to the given state unconditionally. + * + * Return: Zero if succesful, or a negative error code. */ int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) { @@ -965,6 +969,8 @@ EXPORT_SYMBOL(snd_pcm_stop); * * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. + * + * Return: Zero if succesful, or a negative error code. */ int snd_pcm_drain_done(struct snd_pcm_substream *substream) { @@ -1098,6 +1104,9 @@ static struct action_ops snd_pcm_action_suspend = { * @substream: the PCM substream * * After this call, all streams are changed to SUSPENDED state. + * + * Return: Zero if successful (or @substream is %NULL), or a negative error + * code. */ int snd_pcm_suspend(struct snd_pcm_substream *substream) { @@ -1120,6 +1129,8 @@ EXPORT_SYMBOL(snd_pcm_suspend); * @pcm: the PCM instance * * After this call, all streams are changed to SUSPENDED state. + * + * Return: Zero if successful (or @pcm is %NULL), or a negative error code. */ int snd_pcm_suspend_all(struct snd_pcm *pcm) { @@ -1343,6 +1354,8 @@ static struct action_ops snd_pcm_action_prepare = { * snd_pcm_prepare - prepare the PCM substream to be triggerable * @substream: the PCM substream instance * @file: file to refer f_flags + * + * Return: Zero if successful, or a negative error code. */ static int snd_pcm_prepare(struct snd_pcm_substream *substream, struct file *file) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 1bb95aeea084..7b596b5751db 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -863,7 +863,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, * * Reads the data from the internal buffer. * - * Returns the size of read data, or a negative error code on failure. + * Return: The size of read data, or a negative error code on failure. */ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) @@ -1024,8 +1024,8 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun /** * snd_rawmidi_transmit_empty - check whether the output buffer is empty * @substream: the rawmidi substream - * - * Returns 1 if the internal output buffer is empty, 0 if not. + * + * Return: 1 if the internal output buffer is empty, 0 if not. */ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) { @@ -1055,7 +1055,7 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) * and call snd_rawmidi_transmit_ack() after the transmission is * finished. * - * Returns the size of copied data, or a negative error code on failure. + * Return: The size of copied data, or a negative error code on failure. */ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) @@ -1107,7 +1107,7 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, * the given size and updates the condition. * Call after the transmission is finished. * - * Returns the advanced size if successful, or a negative error code on failure. + * Return: The advanced size if successful, or a negative error code on failure. */ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) { @@ -1140,7 +1140,7 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) * * Copies data from the buffer to the device and advances the pointer. * - * Returns the copied size if successful, or a negative error code on failure. + * Return: The copied size if successful, or a negative error code on failure. */ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) @@ -1438,7 +1438,7 @@ static int snd_rawmidi_alloc_substreams(struct snd_rawmidi *rmidi, * Creates a new rawmidi instance. * Use snd_rawmidi_set_ops() to set the operators to the new instance. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_rawmidi_new(struct snd_card *card, char *id, int device, int output_count, int input_count, diff --git a/sound/core/sound.c b/sound/core/sound.c index 70ccdab74153..f002bd911dae 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -102,6 +102,9 @@ static void snd_request_other(int minor) * This function increments the reference counter of the card instance * if an associated instance with the given minor number and type is found. * The caller must call snd_card_unref() appropriately later. + * + * Return: The user data pointer if the specified device is found. %NULL + * otherwise. */ void *snd_lookup_minor_data(unsigned int minor, int type) { @@ -261,7 +264,7 @@ static int snd_kernel_minor(int type, struct snd_card *card, int dev) * Registers an ALSA device file for the given card. * The operators have to be set in reg parameter. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, const struct file_operations *f_ops, @@ -339,7 +342,7 @@ static int find_snd_minor(int type, struct snd_card *card, int dev) * Unregisters the device file already registered via * snd_register_device(). * - * Returns zero if sucecessful, or a negative error code on failure + * Return: Zero if successful, or a negative error code on failure. */ int snd_unregister_device(int type, struct snd_card *card, int dev) { diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 0097f3619faa..02f90b4f8b86 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -365,8 +365,7 @@ static void master_free(struct snd_kcontrol *kcontrol) * @name: name string of the control element to create * @tlv: optional TLV int array for dB information * - * Creates a virtual matster control with the given name string. - * Returns the created control element, or NULL for errors (ENOMEM). + * Creates a virtual master control with the given name string. * * After creating a vmaster element, you can add the slave controls * via snd_ctl_add_slave() or snd_ctl_add_slave_uncached(). @@ -375,6 +374,8 @@ static void master_free(struct snd_kcontrol *kcontrol) * for dB scale of the master control. It should be a single element * with #SNDRV_CTL_TLVT_DB_SCALE, #SNDRV_CTL_TLV_DB_MINMAX or * #SNDRV_CTL_TLVT_DB_MINMAX_MUTE type, and should be the max 0dB. + * + * Return: The created control element, or %NULL for errors (ENOMEM). */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv) @@ -426,6 +427,8 @@ EXPORT_SYMBOL(snd_ctl_make_virtual_master); * * Adds the given hook to the vmaster control element so that it's called * at each time when the value is changed. + * + * Return: Zero. */ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol, void (*hook)(void *private_data, int), diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 7d02c322ed93..8545da99b183 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -87,7 +87,7 @@ config SND_ALOOP configured number of substreams (see the pcm_substreams module parameter). - The looback device allow time sychronization with an external + The loopback device allows time sychronization with an external timing source using the time shift universal control (+-20% of system time). diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 64d534710b51..6f78de9c6fb6 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -325,7 +325,7 @@ static void params_change(struct snd_pcm_substream *substream) struct loopback_pcm *dpcm = runtime->private_data; struct loopback_cable *cable = dpcm->cable; - cable->hw.formats = (1ULL << runtime->format); + cable->hw.formats = pcm_format_to_bits(runtime->format); cable->hw.rate_min = runtime->rate; cable->hw.rate_max = runtime->rate; cable->hw.channels_min = runtime->channels; diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 4608c2ca43f8..e3a90d043f03 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -129,6 +129,8 @@ static void _snd_mpu401_uart_interrupt(struct snd_mpu401 *mpu) * @dev_id: mpu401 instance * * Processes the interrupt for MPU401-UART i/o. + * + * Return: %IRQ_HANDLED if the interrupt was handled. %IRQ_NONE otherwise. */ irqreturn_t snd_mpu401_uart_interrupt(int irq, void *dev_id) { @@ -148,6 +150,8 @@ EXPORT_SYMBOL(snd_mpu401_uart_interrupt); * @dev_id: mpu401 instance * * Processes the interrupt for MPU401-UART output. + * + * Return: %IRQ_HANDLED if the interrupt was handled. %IRQ_NONE otherwise. */ irqreturn_t snd_mpu401_uart_interrupt_tx(int irq, void *dev_id) { @@ -519,7 +523,7 @@ static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi) * not the mpu401 instance itself. To access to the mpu401 instance, * cast from rawmidi->private_data (with struct snd_mpu401 magic-cast). * - * Returns zero if successful, or a negative error code. + * Return: Zero if successful, or a negative error code. */ int snd_mpu401_uart_new(struct snd_card *card, int device, unsigned short hardware, diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index bcc3e8e07122..a59c88818f48 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -114,7 +114,7 @@ static int sound_alloc_dmap(struct dma_buffparms *dmap) } } dmap->raw_buf = start_addr; - dmap->raw_buf_phys = virt_to_bus(start_addr); + dmap->raw_buf_phys = dma_map_single(NULL, start_addr, dmap->buffsize, DMA_BIDIRECTIONAL); for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) SetPageReserved(page); @@ -139,6 +139,7 @@ static void sound_free_dmap(struct dma_buffparms *dmap) for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) ClearPageReserved(page); + dma_unmap_single(NULL, dmap->raw_buf_phys, dmap->buffsize, DMA_BIDIRECTIONAL); free_pages((unsigned long) dmap->raw_buf, sz); dmap->raw_buf = NULL; } diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index 7d42c5418d1b..851a1da46be1 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -626,13 +626,12 @@ int sb_dsp_detect(struct address_info *hw_config, int pci, int pciio, struct sb_ */ - detected_devc = kmalloc(sizeof(sb_devc), GFP_KERNEL); + detected_devc = kmemdup(devc, sizeof(sb_devc), GFP_KERNEL); if (detected_devc == NULL) { printk(KERN_ERR "sb: Can't allocate memory for device information\n"); return 0; } - memcpy(detected_devc, devc, sizeof(sb_devc)); MDB(printk(KERN_INFO "SB %d.%02d detected OK (%x)\n", devc->major, devc->minor, hw_config->io_base)); return 1; } diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index 8e514a676a0d..5433c6f5eca2 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -352,23 +352,26 @@ int probe_uart401(struct address_info *hw_config, struct module *owner) goto cleanup_irq; } conf_printf(name, hw_config); - midi_devs[devc->my_dev] = kmalloc(sizeof(struct midi_operations), GFP_KERNEL); + midi_devs[devc->my_dev] = kmemdup(&uart401_operations, + sizeof(struct midi_operations), + GFP_KERNEL); if (!midi_devs[devc->my_dev]) { printk(KERN_ERR "uart401: Failed to allocate memory\n"); goto cleanup_unload_mididev; } - memcpy(midi_devs[devc->my_dev], &uart401_operations, sizeof(struct midi_operations)); if (owner) midi_devs[devc->my_dev]->owner = owner; midi_devs[devc->my_dev]->devc = devc; - midi_devs[devc->my_dev]->converter = kmalloc(sizeof(struct synth_operations), GFP_KERNEL); + midi_devs[devc->my_dev]->converter = kmemdup(&std_midi_synth, + sizeof(struct synth_operations), + GFP_KERNEL); + if (!midi_devs[devc->my_dev]->converter) { printk(KERN_WARNING "uart401: Failed to allocate memory\n"); goto cleanup_midi_devs; } - memcpy(midi_devs[devc->my_dev]->converter, &std_midi_synth, sizeof(struct synth_operations)); strcpy(midi_devs[devc->my_dev]->info.name, name); midi_devs[devc->my_dev]->converter->id = "UART401"; midi_devs[devc->my_dev]->converter->midi_dev = devc->my_dev; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 8b0f99688303..d37c683cfd7a 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -299,7 +299,7 @@ EXPORT_SYMBOL(snd_ac97_write); * Reads a value from the given register. This will invoke the read * callback directly after the register check. * - * Returns the read value. + * Return: The read value. */ unsigned short snd_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { @@ -352,7 +352,7 @@ EXPORT_SYMBOL(snd_ac97_write_cache); * Compares the value with the register cache and updates the value * only when the value is changed. * - * Returns 1 if the value is changed, 0 if no change, or a negative + * Return: 1 if the value is changed, 0 if no change, or a negative * code on failure. */ int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short value) @@ -384,7 +384,7 @@ EXPORT_SYMBOL(snd_ac97_update); * Updates the masked-bits on the given register only when the value * is changed. * - * Returns 1 if the bits are changed, 0 if no change, or a negative + * Return: 1 if the bits are changed, 0 if no change, or a negative * code on failure. */ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value) @@ -1836,7 +1836,7 @@ void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name, int m * snd_ac97_get_short_name - retrieve codec name * @ac97: the codec instance * - * Returns the short identifying name of the codec. + * Return: The short identifying name of the codec. */ const char *snd_ac97_get_short_name(struct snd_ac97 *ac97) { @@ -1910,7 +1910,7 @@ static int ac97_reset_wait(struct snd_ac97 *ac97, int timeout, int with_modem) * The AC97 bus instance is registered as a low-level device, so you don't * have to release it manually. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops, void *private_data, struct snd_ac97_bus **rbus) @@ -2006,7 +2006,7 @@ static void do_update_power(struct work_struct *work) * The ac97 instance is registered as a low-level device, so you don't * have to release it manually. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, struct snd_ac97 **rac97) { @@ -2373,6 +2373,8 @@ static struct ac97_power_reg power_regs[PWIDX_SIZE] = { * @powerup: non-zero when power up the part * * Update the AC97 powerdown register bits of the given part. + * + * Return: Zero. */ int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup) { @@ -2885,7 +2887,7 @@ static int apply_quirk_str(struct snd_ac97 *ac97, const char *typestr) * headphone (true line-out) control as "Master". * The quirk-list must be terminated with a zero-filled entry. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override) diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index f1488fc176d5..eab0fc9ff2e0 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -253,7 +253,7 @@ static int set_spdif_rate(struct snd_ac97 *ac97, unsigned short rate) * AC97_SPDIF is accepted as a pseudo register to modify the SPDIF * status bits. * - * Returns zero if successful, or a negative error code on failure. + * Return: Zero if successful, or a negative error code on failure. */ int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate) { @@ -440,6 +440,8 @@ static unsigned int get_rates(struct ac97_pcm *pcm, unsigned int cidx, unsigned * It assigns available AC97 slots for given PCMs. If none or only * some slots are available, pcm->xxx.slots and pcm->xxx.rslots[] members * are reduced and might be zero. + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_ac97_pcm_assign(struct snd_ac97_bus *bus, unsigned short pcms_count, @@ -562,6 +564,8 @@ EXPORT_SYMBOL(snd_ac97_pcm_assign); * @slots: a subset of allocated slots (snd_ac97_pcm_assign) for this pcm * * It locks the specified slots and sets the given rate to AC97 registers. + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate, enum ac97_pcm_cfg cfg, unsigned short slots) @@ -644,6 +648,8 @@ EXPORT_SYMBOL(snd_ac97_pcm_open); * @pcm: the ac97 pcm instance * * It frees the locked AC97 slots. + * + * Return: Zero. */ int snd_ac97_pcm_close(struct ac97_pcm *pcm) { @@ -718,6 +724,8 @@ static int double_rate_hw_constraint_channels(struct snd_pcm_hw_params *params, * * Installs the hardware constraint rules to prevent using double rates and * more than two channels at the same time. + * + * Return: Zero if successful, or a negative error code on failure. */ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime) { diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index e760af9d1fb6..53754f5edeb1 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -451,10 +451,10 @@ static int snd_ali_reset_5451(struct snd_ali *codec) if (pci_dev) { pci_read_config_dword(pci_dev, 0x7c, &dwVal); pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000); - udelay(5000); + mdelay(5); pci_read_config_dword(pci_dev, 0x7c, &dwVal); pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff); - udelay(5000); + mdelay(5); } pci_dev = codec->pci; @@ -463,14 +463,14 @@ static int snd_ali_reset_5451(struct snd_ali *codec) udelay(500); pci_read_config_dword(pci_dev, 0x44, &dwVal); pci_write_config_dword(pci_dev, 0x44, dwVal & 0xfffbffff); - udelay(5000); + mdelay(5); wCount = 200; while(wCount--) { wReg = snd_ali_codec_peek(codec, 0, AC97_POWERDOWN); if ((wReg & 0x000f) == 0x000f) return 0; - udelay(5000); + mdelay(5); } /* non-fatal if you have a non PM capable codec */ diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 0aabfedeecba..fbc17203613c 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -966,7 +966,7 @@ static u64 snd_card_asihpi_playback_formats(struct snd_card_asihpi *asihpi, if (!err) err = hpi_outstream_query_format(h_stream, &hpi_format); if (!err && (hpi_to_alsa_formats[format] != -1)) - formats |= (1ULL << hpi_to_alsa_formats[format]); + formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]); } return formats; } @@ -1141,8 +1141,8 @@ static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi, format, sample_rate, 128000, 0); if (!err) err = hpi_instream_query_format(h_stream, &hpi_format); - if (!err) - formats |= (1ULL << hpi_to_alsa_formats[format]); + if (!err && (hpi_to_alsa_formats[format] != -1)) + formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]); } return formats; } diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index e6b016693240..bdd888ec9a84 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -657,14 +657,14 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu) return 0; } -static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu) +static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, + const struct firmware *fw_entry) { int n, i; int reg; int value; unsigned int write_post; unsigned long flags; - const struct firmware *fw_entry = emu->firmware; if (!fw_entry) return -EIO; @@ -725,9 +725,34 @@ static int emu1010_firmware_thread(void *data) /* Return to Audio Dock programming mode */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK); - err = snd_emu1010_load_firmware(emu); - if (err != 0) - continue; + + if (!emu->dock_fw) { + const char *filename = NULL; + switch (emu->card_capabilities->emu_model) { + case EMU_MODEL_EMU1010: + filename = DOCK_FILENAME; + break; + case EMU_MODEL_EMU1010B: + filename = MICRO_DOCK_FILENAME; + break; + case EMU_MODEL_EMU1616: + filename = MICRO_DOCK_FILENAME; + break; + } + if (filename) { + err = request_firmware(&emu->dock_fw, + filename, + &emu->pci->dev); + if (err) + continue; + } + } + + if (emu->dock_fw) { + err = snd_emu1010_load_firmware(emu, emu->dock_fw); + if (err) + continue; + } snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0); snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ®); @@ -862,7 +887,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) filename, emu->firmware->size); } - err = snd_emu1010_load_firmware(emu); + err = snd_emu1010_load_firmware(emu, emu->firmware); if (err != 0) { snd_printk(KERN_INFO "emu1010: Loading Firmware failed\n"); return err; @@ -1253,6 +1278,8 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) kthread_stop(emu->emu1010.firmware_thread); if (emu->firmware) release_firmware(emu->firmware); + if (emu->dock_fw) + release_firmware(emu->dock_fw); if (emu->irq >= 0) free_irq(emu->irq, emu); /* remove reserved page */ diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index a3ea76a4c9d2..7c11d46b84d3 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -119,6 +119,32 @@ static bool check_pincap_validity(struct hda_codec *codec, hda_nid_t pin, } } +static bool can_be_headset_mic(struct hda_codec *codec, + struct auto_pin_cfg_item *item, + int seq_number) +{ + int attr; + unsigned int def_conf; + if (item->type != AUTO_PIN_MIC) + return false; + + if (item->is_headset_mic || item->is_headphone_mic) + return false; /* Already assigned */ + + def_conf = snd_hda_codec_get_pincfg(codec, item->pin); + attr = snd_hda_get_input_pin_attr(def_conf); + if (attr <= INPUT_PIN_ATTR_DOCK) + return false; + + if (seq_number >= 0) { + int seq = get_defcfg_sequence(def_conf); + if (seq != seq_number) + return false; + } + + return true; +} + /* * Parse all pin widgets and store the useful pin nids to cfg * @@ -260,6 +286,38 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, } } + /* Find a pin that could be a headset or headphone mic */ + if (cond_flags & HDA_PINCFG_HEADSET_MIC || cond_flags & HDA_PINCFG_HEADPHONE_MIC) { + bool hsmic = !!(cond_flags & HDA_PINCFG_HEADSET_MIC); + bool hpmic = !!(cond_flags & HDA_PINCFG_HEADPHONE_MIC); + for (i = 0; (hsmic || hpmic) && (i < cfg->num_inputs); i++) + if (hsmic && can_be_headset_mic(codec, &cfg->inputs[i], 0xc)) { + cfg->inputs[i].is_headset_mic = 1; + hsmic = false; + } else if (hpmic && can_be_headset_mic(codec, &cfg->inputs[i], 0xd)) { + cfg->inputs[i].is_headphone_mic = 1; + hpmic = false; + } + + /* If we didn't find our sequence number mark, fall back to any sequence number */ + for (i = 0; (hsmic || hpmic) && (i < cfg->num_inputs); i++) { + if (!can_be_headset_mic(codec, &cfg->inputs[i], -1)) + continue; + if (hsmic) { + cfg->inputs[i].is_headset_mic = 1; + hsmic = false; + } else if (hpmic) { + cfg->inputs[i].is_headphone_mic = 1; + hpmic = false; + } + } + + if (hsmic) + snd_printdd("Told to look for a headset mic, but didn't find any.\n"); + if (hpmic) + snd_printdd("Told to look for a headphone mic, but didn't find any.\n"); + } + /* FIX-UP: * If no line-out is defined but multiple HPs are found, * some of them might be the real line-outs. @@ -388,6 +446,7 @@ EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); */ static const char *hda_get_input_pin_label(struct hda_codec *codec, + const struct auto_pin_cfg_item *item, hda_nid_t pin, bool check_location) { unsigned int def_conf; @@ -400,6 +459,10 @@ static const char *hda_get_input_pin_label(struct hda_codec *codec, switch (get_defcfg_device(def_conf)) { case AC_JACK_MIC_IN: + if (item && item->is_headset_mic) + return "Headset Mic"; + if (item && item->is_headphone_mic) + return "Headphone Mic"; if (!check_location) return "Mic"; attr = snd_hda_get_input_pin_attr(def_conf); @@ -480,7 +543,8 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, has_multiple_pins = 1; if (has_multiple_pins && type == AUTO_PIN_MIC) has_multiple_pins &= check_mic_location_need(codec, cfg, input); - return hda_get_input_pin_label(codec, cfg->inputs[input].pin, + return hda_get_input_pin_label(codec, &cfg->inputs[input], + cfg->inputs[input].pin, has_multiple_pins); } EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); @@ -649,7 +713,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, } } if (!name) - name = hda_get_input_pin_label(codec, nid, true); + name = hda_get_input_pin_label(codec, NULL, nid, true); break; } if (!name) diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h index f74807138b49..e941f604f5e5 100644 --- a/sound/pci/hda/hda_auto_parser.h +++ b/sound/pci/hda/hda_auto_parser.h @@ -36,6 +36,8 @@ enum { struct auto_pin_cfg_item { hda_nid_t pin; int type; + unsigned int is_headset_mic:1; + unsigned int is_headphone_mic:1; /* Mic-only in headphone jack */ }; struct auto_pin_cfg; @@ -78,8 +80,10 @@ struct auto_pin_cfg { }; /* bit-flags for snd_hda_parse_pin_def_config() behavior */ -#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ -#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ +#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ +#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ +#define HDA_PINCFG_HEADSET_MIC (1 << 2) /* Try to find headset mic; mark seq number as 0xc to trigger */ +#define HDA_PINCFG_HEADPHONE_MIC (1 << 3) /* Try to find headphone mic; mark seq number as 0xd to trigger */ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, struct auto_pin_cfg *cfg, @@ -90,4 +94,25 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, #define snd_hda_parse_pin_def_config(codec, cfg, ignore) \ snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0) +static inline int auto_cfg_hp_outs(const struct auto_pin_cfg *cfg) +{ + return (cfg->line_out_type == AUTO_PIN_HP_OUT) ? + cfg->line_outs : cfg->hp_outs; +} +static inline const hda_nid_t *auto_cfg_hp_pins(const struct auto_pin_cfg *cfg) +{ + return (cfg->line_out_type == AUTO_PIN_HP_OUT) ? + cfg->line_out_pins : cfg->hp_pins; +} +static inline int auto_cfg_speaker_outs(const struct auto_pin_cfg *cfg) +{ + return (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) ? + cfg->line_outs : cfg->speaker_outs; +} +static inline const hda_nid_t *auto_cfg_speaker_pins(const struct auto_pin_cfg *cfg) +{ + return (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) ? + cfg->line_out_pins : cfg->speaker_pins; +} + #endif /* __SOUND_HDA_AUTO_PARSER_H */ diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0849aac449f2..63c99090a4ec 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -39,13 +39,23 @@ static void snd_hda_generate_beep(struct work_struct *work) struct hda_beep *beep = container_of(work, struct hda_beep, beep_work); struct hda_codec *codec = beep->codec; + int tone; if (!beep->enabled) return; + tone = beep->tone; + if (tone && !beep->playing) { + snd_hda_power_up(codec); + beep->playing = 1; + } /* generate tone */ snd_hda_codec_write(codec, beep->nid, 0, - AC_VERB_SET_BEEP_CONTROL, beep->tone); + AC_VERB_SET_BEEP_CONTROL, tone); + if (!tone && beep->playing) { + beep->playing = 0; + snd_hda_power_down(codec); + } } /* (non-standard) Linear beep tone calculation for IDT/STAC codecs @@ -115,14 +125,23 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, return 0; } +static void turn_off_beep(struct hda_beep *beep) +{ + cancel_work_sync(&beep->beep_work); + if (beep->playing) { + /* turn off beep */ + snd_hda_codec_write(beep->codec, beep->nid, 0, + AC_VERB_SET_BEEP_CONTROL, 0); + beep->playing = 0; + snd_hda_power_down(beep->codec); + } +} + static void snd_hda_do_detach(struct hda_beep *beep) { input_unregister_device(beep->dev); beep->dev = NULL; - cancel_work_sync(&beep->beep_work); - /* turn off beep for sure */ - snd_hda_codec_write(beep->codec, beep->nid, 0, - AC_VERB_SET_BEEP_CONTROL, 0); + turn_off_beep(beep); } static int snd_hda_do_attach(struct hda_beep *beep) @@ -170,12 +189,8 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) 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 (!enable) + turn_off_beep(beep); return 1; } return 0; @@ -198,7 +213,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) snprintf(beep->phys, sizeof(beep->phys), "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); /* enable linear scale */ - snd_hda_codec_write(codec, nid, 0, + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, 0x01); beep->nid = nid; diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 4dc6933bc655..cb88464676b6 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -36,6 +36,7 @@ struct hda_beep { hda_nid_t nid; unsigned int enabled:1; unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ + unsigned int playing:1; 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 4aba7646dd9c..6f9b64700f6e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1065,8 +1065,14 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, { struct hda_pincfg *pin; + /* the check below may be invalid when pins are added by a fixup + * dynamically (e.g. via snd_hda_codec_update_widgets()), so disabled + * for now + */ + /* if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) return -EINVAL; + */ pin = look_up_pincfg(codec, list, nid); if (!pin) { @@ -1300,8 +1306,6 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, static unsigned int hda_set_power_state(struct hda_codec *codec, unsigned int power_state); -static unsigned int default_power_filter(struct hda_codec *codec, hda_nid_t nid, - unsigned int power_state); /** * snd_hda_codec_new - create a HDA codec @@ -1422,7 +1426,6 @@ int snd_hda_codec_new(struct hda_bus *bus, #endif codec->epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); - codec->power_filter = default_power_filter; /* power-up all before initialization */ hda_set_power_state(codec, AC_PWRST_D0); @@ -2787,6 +2790,11 @@ void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook) { if (!hook->hook || !hook->codec) return; + /* don't call vmaster hook in the destructor since it might have + * been already destroyed + */ + if (hook->codec->bus->shutdown) + return; switch (hook->mute_mode) { case HDA_VMUTE_FOLLOW_MASTER: snd_ctl_sync_vmaster_hook(hook->sw_kctl); @@ -3770,8 +3778,9 @@ static unsigned int hda_sync_power_state(struct hda_codec *codec, } /* don't power down the widget if it controls eapd and EAPD_BTLENABLE is set */ -static unsigned int default_power_filter(struct hda_codec *codec, hda_nid_t nid, - unsigned int power_state) +unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state) { if (power_state == AC_PWRST_D3 && get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_PIN && @@ -3783,6 +3792,7 @@ static unsigned int default_power_filter(struct hda_codec *codec, hda_nid_t nid, } return power_state; } +EXPORT_SYMBOL_HDA(snd_hda_codec_eapd_power_filter); /* * set power state of the codec, and return the power state @@ -3827,8 +3837,8 @@ static void sync_power_up_states(struct hda_codec *codec) hda_nid_t nid = codec->start_nid; int i; - /* don't care if no or standard filter is used */ - if (!codec->power_filter || codec->power_filter == default_power_filter) + /* don't care if no filter is used */ + if (!codec->power_filter) return; for (i = 0; i < codec->num_nodes; i++, nid++) { @@ -5546,14 +5556,12 @@ void *snd_array_new(struct snd_array *array) if (array->used >= array->alloced) { int num = array->alloced + array->alloc_align; int size = (num + 1) * array->elem_size; - int oldsize = array->alloced * array->elem_size; void *nlist; if (snd_BUG_ON(num >= 4096)) return NULL; - nlist = krealloc(array->list, size, GFP_KERNEL); + nlist = krealloc(array->list, size, GFP_KERNEL | __GFP_ZERO); if (!nlist) return NULL; - memset(nlist + oldsize, 0, size - oldsize); array->list = nlist; array->alloced = num; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 23ca1722aff1..c93f9021f452 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -757,6 +757,9 @@ struct hda_pcm_ops { struct snd_pcm_substream *substream); int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, struct snd_pcm_substream *substream); + unsigned int (*get_delay)(struct hda_pcm_stream *info, + struct hda_codec *codec, + struct snd_pcm_substream *substream); }; /* PCM information for each substream */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2dbe767be16b..ac079f93c535 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -34,6 +34,7 @@ #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" +#include "hda_beep.h" #include "hda_generic.h" @@ -150,15 +151,25 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "add_stereo_mix_input"); if (val >= 0) spec->add_stereo_mix_input = !!val; + /* the following two are just for compatibility */ val = snd_hda_get_bool_hint(codec, "add_out_jack_modes"); if (val >= 0) - spec->add_out_jack_modes = !!val; + spec->add_jack_modes = !!val; val = snd_hda_get_bool_hint(codec, "add_in_jack_modes"); if (val >= 0) - spec->add_in_jack_modes = !!val; + spec->add_jack_modes = !!val; + val = snd_hda_get_bool_hint(codec, "add_jack_modes"); + if (val >= 0) + spec->add_jack_modes = !!val; val = snd_hda_get_bool_hint(codec, "power_down_unused"); if (val >= 0) spec->power_down_unused = !!val; + val = snd_hda_get_bool_hint(codec, "add_hp_mic"); + if (val >= 0) + spec->hp_mic = !!val; + val = snd_hda_get_bool_hint(codec, "hp_mic_detect"); + if (val >= 0) + spec->suppress_hp_mic_detect = !val; if (!snd_hda_get_int_hint(codec, "mixer_nid", &val)) spec->mixer_nid = val; @@ -996,7 +1007,7 @@ enum { /* Primary DAC shared with main surrounds */ BAD_SHARED_SURROUND = 0x100, /* No independent HP possible */ - BAD_NO_INDEP_HP = 0x40, + BAD_NO_INDEP_HP = 0x10, /* Primary DAC shared with main CLFE */ BAD_SHARED_CLFE = 0x10, /* Primary DAC shared with extra surrounds */ @@ -1051,16 +1062,7 @@ static int assign_out_path_ctls(struct hda_codec *codec, struct nid_path *path) return badness; } -struct badness_table { - int no_primary_dac; /* no primary DAC */ - int no_dac; /* no secondary DACs */ - int shared_primary; /* primary DAC is shared with main output */ - int shared_surr; /* secondary DAC shared with main or primary */ - int shared_clfe; /* third DAC shared with main or primary */ - int shared_surr_main; /* secondary DAC sahred with main/DAC0 */ -}; - -static struct badness_table main_out_badness = { +const struct badness_table hda_main_out_badness = { .no_primary_dac = BAD_NO_PRIMARY_DAC, .no_dac = BAD_NO_DAC, .shared_primary = BAD_NO_PRIMARY_DAC, @@ -1068,8 +1070,9 @@ static struct badness_table main_out_badness = { .shared_clfe = BAD_SHARED_CLFE, .shared_surr_main = BAD_SHARED_SURROUND, }; +EXPORT_SYMBOL_HDA(hda_main_out_badness); -static struct badness_table extra_out_badness = { +const struct badness_table hda_extra_out_badness = { .no_primary_dac = BAD_NO_DAC, .no_dac = BAD_NO_DAC, .shared_primary = BAD_NO_EXTRA_DAC, @@ -1077,6 +1080,7 @@ static struct badness_table extra_out_badness = { .shared_clfe = BAD_SHARED_EXTRA_SURROUND, .shared_surr_main = BAD_NO_EXTRA_SURR_DAC, }; +EXPORT_SYMBOL_HDA(hda_extra_out_badness); /* get the DAC of the primary output corresponding to the given array index */ static hda_nid_t get_primary_out(struct hda_codec *codec, int idx) @@ -1367,22 +1371,25 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) { struct hda_gen_spec *spec = codec->spec; struct nid_path *path; - hda_nid_t dac, pin; + hda_nid_t path_dac, dac, pin; path = snd_hda_get_path_from_idx(codec, path_idx); if (!path || !path->depth || is_nid_contained(path, spec->mixer_nid)) return 0; - dac = path->path[0]; + path_dac = path->path[0]; + dac = spec->private_dac_nids[0]; pin = path->path[path->depth - 1]; path = snd_hda_add_new_path(codec, dac, pin, spec->mixer_nid); if (!path) { - if (dac != spec->multiout.dac_nids[0]) - dac = spec->multiout.dac_nids[0]; + if (dac != path_dac) + dac = path_dac; else if (spec->multiout.hp_out_nid[0]) dac = spec->multiout.hp_out_nid[0]; else if (spec->multiout.extra_out_nid[0]) dac = spec->multiout.extra_out_nid[0]; + else + dac = 0; if (dac) path = snd_hda_add_new_path(codec, dac, pin, spec->mixer_nid); @@ -1507,7 +1514,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec, badness += try_assign_dacs(codec, cfg->line_outs, cfg->line_out_pins, spec->private_dac_nids, spec->out_paths, - &main_out_badness); + spec->main_out_badness); if (fill_mio_first && cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { @@ -1522,7 +1529,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec, err = try_assign_dacs(codec, cfg->hp_outs, cfg->hp_pins, spec->multiout.hp_out_nid, spec->hp_paths, - &extra_out_badness); + spec->extra_out_badness); if (err < 0) return err; badness += err; @@ -1532,7 +1539,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec, cfg->speaker_pins, spec->multiout.extra_out_nid, spec->speaker_paths, - &extra_out_badness); + spec->extra_out_badness); if (err < 0) return err; badness += err; @@ -1926,6 +1933,17 @@ static int create_speaker_out_ctls(struct hda_codec *codec) * independent HP controls */ +/* update HP auto-mute state too */ +static void update_hp_automute_hook(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + + if (spec->hp_automute_hook) + spec->hp_automute_hook(codec, NULL); + else + snd_hda_gen_hp_automute(codec, NULL); +} + static int indep_hp_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1986,12 +2004,7 @@ static int indep_hp_put(struct snd_kcontrol *kcontrol, else *dacp = spec->alt_dac_nid; - /* update HP auto-mute state too */ - if (spec->hp_automute_hook) - spec->hp_automute_hook(codec, NULL); - else - snd_hda_gen_hp_automute(codec, NULL); - + update_hp_automute_hook(codec); ret = 1; } unlock: @@ -2072,6 +2085,14 @@ get_multiio_path(struct hda_codec *codec, int idx) static void update_automute_all(struct hda_codec *codec); +/* Default value to be passed as aamix argument for snd_hda_activate_path(); + * used for output paths + */ +static bool aamix_default(struct hda_gen_spec *spec) +{ + return !spec->have_aamix_ctl || spec->aamix_mode; +} + static int set_multi_io(struct hda_codec *codec, int idx, bool output) { struct hda_gen_spec *spec = codec->spec; @@ -2087,11 +2108,11 @@ static int set_multi_io(struct hda_codec *codec, int idx, bool output) if (output) { set_pin_target(codec, nid, PIN_OUT, true); - snd_hda_activate_path(codec, path, true, true); + snd_hda_activate_path(codec, path, true, aamix_default(spec)); set_pin_eapd(codec, nid, true); } else { set_pin_eapd(codec, nid, false); - snd_hda_activate_path(codec, path, false, true); + snd_hda_activate_path(codec, path, false, aamix_default(spec)); set_pin_target(codec, nid, spec->multi_io[idx].ctl_in, true); path_power_down_sync(codec, path); } @@ -2182,8 +2203,8 @@ static void update_aamix_paths(struct hda_codec *codec, bool do_mix, snd_hda_activate_path(codec, mix_path, true, true); path_power_down_sync(codec, nomix_path); } else { - snd_hda_activate_path(codec, mix_path, false, true); - snd_hda_activate_path(codec, nomix_path, true, true); + snd_hda_activate_path(codec, mix_path, false, false); + snd_hda_activate_path(codec, nomix_path, true, false); path_power_down_sync(codec, mix_path); } } @@ -2240,63 +2261,95 @@ static int create_loopback_mixing_ctl(struct hda_codec *codec) static void call_update_outputs(struct hda_codec *codec); /* for shared I/O, change the pin-control accordingly */ -static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic) +static void update_hp_mic(struct hda_codec *codec, int adc_mux, bool force) { struct hda_gen_spec *spec = codec->spec; + bool as_mic; 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. - */ + hda_nid_t pin; - val = snd_hda_get_default_vref(codec, pin); + pin = spec->hp_mic_pin; + as_mic = spec->cur_mux[adc_mux] == spec->hp_mic_mux_idx; + + if (!force) { + val = snd_hda_codec_get_pin_target(codec, pin); + if (as_mic) { + if (val & PIN_IN) + return; + } else { + if (val & PIN_OUT) + return; + } + } - /* This pin does not have vref caps - let's enable vref on pin 0x18 - instead, as suggested by Realtek */ + val = snd_hda_get_default_vref(codec, pin); + /* if the HP pin doesn't support VREF and the codec driver gives an + * alternative pin, set up the VREF on that pin instead + */ if (val == AC_PINCTL_VREF_HIZ && spec->shared_mic_vref_pin) { const hda_nid_t vref_pin = spec->shared_mic_vref_pin; unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin); if (vref_val != AC_PINCTL_VREF_HIZ) snd_hda_set_pin_ctl_cache(codec, vref_pin, - PIN_IN | (set_as_mic ? vref_val : 0)); + PIN_IN | (as_mic ? vref_val : 0)); } - val = set_as_mic ? val | PIN_IN : PIN_HP; - set_pin_target(codec, pin, val, true); - - spec->automute_speaker = !set_as_mic; - call_update_outputs(codec); + if (!spec->hp_mic_jack_modes) { + if (as_mic) + val |= PIN_IN; + else + val = PIN_HP; + set_pin_target(codec, pin, val, true); + update_hp_automute_hook(codec); + } } /* create a shared input with the headphone out */ -static int create_shared_input(struct hda_codec *codec) +static int create_hp_mic(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int defcfg; hda_nid_t nid; - /* only one internal input pin? */ - if (cfg->num_inputs != 1) - return 0; - defcfg = snd_hda_codec_get_pincfg(codec, cfg->inputs[0].pin); - if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT) + if (!spec->hp_mic) { + if (spec->suppress_hp_mic_detect) + return 0; + /* automatic detection: only if no input or a single internal + * input pin is found, try to detect the shared hp/mic + */ + if (cfg->num_inputs > 1) + return 0; + else if (cfg->num_inputs == 1) { + defcfg = snd_hda_codec_get_pincfg(codec, cfg->inputs[0].pin); + if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT) + return 0; + } + } + + spec->hp_mic = 0; /* clear once */ + if (cfg->num_inputs >= AUTO_CFG_MAX_INS) return 0; - if (cfg->hp_outs == 1 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) - nid = cfg->hp_pins[0]; /* OK, we have a single HP-out */ - else if (cfg->line_outs == 1 && cfg->line_out_type == AUTO_PIN_HP_OUT) - nid = cfg->line_out_pins[0]; /* OK, we have a single line-out */ - else - return 0; /* both not available */ + nid = 0; + if (cfg->line_out_type == AUTO_PIN_HP_OUT && cfg->line_outs > 0) + nid = cfg->line_out_pins[0]; + else if (cfg->hp_outs > 0) + nid = cfg->hp_pins[0]; + if (!nid) + return 0; if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_IN)) return 0; /* no input */ - cfg->inputs[1].pin = nid; - cfg->inputs[1].type = AUTO_PIN_MIC; - cfg->num_inputs = 2; - spec->shared_mic_hp = 1; + cfg->inputs[cfg->num_inputs].pin = nid; + cfg->inputs[cfg->num_inputs].type = AUTO_PIN_MIC; + cfg->inputs[cfg->num_inputs].is_headphone_mic = 1; + cfg->num_inputs++; + spec->hp_mic = 1; + spec->hp_mic_pin = nid; + /* we can't handle auto-mic together with HP-mic */ + spec->suppress_auto_mic = 1; snd_printdd("hda-codec: Enable shared I/O jack on NID 0x%x\n", nid); return 0; } @@ -2304,13 +2357,17 @@ static int create_shared_input(struct hda_codec *codec) /* * output jack mode */ + +static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin); + +static const char * const out_jack_texts[] = { + "Line Out", "Headphone Out", +}; + static int out_jack_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char * const texts[] = { - "Line Out", "Headphone Out", - }; - return snd_hda_enum_helper_info(kcontrol, uinfo, 2, texts); + return snd_hda_enum_helper_info(kcontrol, uinfo, 2, out_jack_texts); } static int out_jack_mode_get(struct snd_kcontrol *kcontrol, @@ -2372,6 +2429,17 @@ static void get_jack_mode_name(struct hda_codec *codec, hda_nid_t pin, ; } +static int get_out_jack_num_items(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + if (spec->add_jack_modes) { + unsigned int pincap = snd_hda_query_pin_caps(codec, pin); + if ((pincap & AC_PINCAP_OUT) && (pincap & AC_PINCAP_HP_DRV)) + return 2; + } + return 1; +} + static int create_out_jack_modes(struct hda_codec *codec, int num_pins, hda_nid_t *pins) { @@ -2380,8 +2448,13 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins, for (i = 0; i < num_pins; i++) { hda_nid_t pin = pins[i]; - unsigned int pincap = snd_hda_query_pin_caps(codec, pin); - if ((pincap & AC_PINCAP_OUT) && (pincap & AC_PINCAP_HP_DRV)) { + if (pin == spec->hp_mic_pin) { + int ret = create_hp_mic_jack_mode(codec, pin); + if (ret < 0) + return ret; + continue; + } + if (get_out_jack_num_items(codec, pin) > 1) { struct snd_kcontrol_new *knew; char name[44]; get_jack_mode_name(codec, pin, name, sizeof(name)); @@ -2502,12 +2575,24 @@ static const struct snd_kcontrol_new in_jack_mode_enum = { .put = in_jack_mode_put, }; +static int get_in_jack_num_items(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + int nitems = 0; + if (spec->add_jack_modes) + nitems = hweight32(get_vref_caps(codec, pin)); + return nitems ? nitems : 1; +} + static int create_in_jack_mode(struct hda_codec *codec, hda_nid_t pin) { struct hda_gen_spec *spec = codec->spec; - unsigned int defcfg; struct snd_kcontrol_new *knew; char name[44]; + unsigned int defcfg; + + if (pin == spec->hp_mic_pin) + return 0; /* already done in create_out_jack_mode() */ /* no jack mode for fixed pins */ defcfg = snd_hda_codec_get_pincfg(codec, pin); @@ -2515,7 +2600,7 @@ static int create_in_jack_mode(struct hda_codec *codec, hda_nid_t pin) return 0; /* no multiple vref caps? */ - if (hweight32(get_vref_caps(codec, pin)) <= 1) + if (get_in_jack_num_items(codec, pin) <= 1) return 0; get_jack_mode_name(codec, pin, name, sizeof(name)); @@ -2526,6 +2611,132 @@ static int create_in_jack_mode(struct hda_codec *codec, hda_nid_t pin) return 0; } +/* + * HP/mic shared jack mode + */ +static int hp_mic_jack_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value; + int out_jacks = get_out_jack_num_items(codec, nid); + int in_jacks = get_in_jack_num_items(codec, nid); + const char *text = NULL; + int idx; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = out_jacks + in_jacks; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; + idx = uinfo->value.enumerated.item; + if (idx < out_jacks) { + if (out_jacks > 1) + text = out_jack_texts[idx]; + else + text = "Headphone Out"; + } else { + idx -= out_jacks; + if (in_jacks > 1) { + unsigned int vref_caps = get_vref_caps(codec, nid); + text = vref_texts[get_vref_idx(vref_caps, idx)]; + } else + text = "Mic In"; + } + + strcpy(uinfo->value.enumerated.name, text); + return 0; +} + +static int get_cur_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t nid) +{ + int out_jacks = get_out_jack_num_items(codec, nid); + int in_jacks = get_in_jack_num_items(codec, nid); + unsigned int val = snd_hda_codec_get_pin_target(codec, nid); + int idx = 0; + + if (val & PIN_OUT) { + if (out_jacks > 1 && val == PIN_HP) + idx = 1; + } else if (val & PIN_IN) { + idx = out_jacks; + if (in_jacks > 1) { + unsigned int vref_caps = get_vref_caps(codec, nid); + val &= AC_PINCTL_VREFEN; + idx += cvt_from_vref_idx(vref_caps, val); + } + } + return idx; +} + +static int hp_mic_jack_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value; + ucontrol->value.enumerated.item[0] = + get_cur_hp_mic_jack_mode(codec, nid); + return 0; +} + +static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value; + int out_jacks = get_out_jack_num_items(codec, nid); + int in_jacks = get_in_jack_num_items(codec, nid); + unsigned int val, oldval, idx; + + oldval = get_cur_hp_mic_jack_mode(codec, nid); + idx = ucontrol->value.enumerated.item[0]; + if (oldval == idx) + return 0; + + if (idx < out_jacks) { + if (out_jacks > 1) + val = idx ? PIN_HP : PIN_OUT; + else + val = PIN_HP; + } else { + idx -= out_jacks; + if (in_jacks > 1) { + unsigned int vref_caps = get_vref_caps(codec, nid); + val = snd_hda_codec_get_pin_target(codec, nid); + val &= ~(AC_PINCTL_VREFEN | PIN_HP); + val |= get_vref_idx(vref_caps, idx) | PIN_IN; + } else + val = snd_hda_get_default_vref(codec, nid); + } + snd_hda_set_pin_ctl_cache(codec, nid, val); + update_hp_automute_hook(codec); + + return 1; +} + +static const struct snd_kcontrol_new hp_mic_jack_mode_enum = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = hp_mic_jack_mode_info, + .get = hp_mic_jack_mode_get, + .put = hp_mic_jack_mode_put, +}; + +static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + struct snd_kcontrol_new *knew; + + if (get_out_jack_num_items(codec, pin) <= 1 && + get_in_jack_num_items(codec, pin) <= 1) + return 0; /* no need */ + knew = snd_hda_gen_add_kctl(spec, "Headphone Mic Jack Mode", + &hp_mic_jack_mode_enum); + if (!knew) + return -ENOMEM; + knew->private_value = pin; + spec->hp_mic_jack_modes = 1; + return 0; +} /* * Parse input paths @@ -2648,7 +2859,6 @@ static int check_dyn_adc_switch(struct hda_codec *codec) unsigned int ok_bits; int i, n, nums; - again: nums = 0; ok_bits = 0; for (n = 0; n < spec->num_adc_nids; n++) { @@ -2663,12 +2873,6 @@ static int check_dyn_adc_switch(struct hda_codec *codec) } if (!ok_bits) { - if (spec->shared_mic_hp) { - spec->shared_mic_hp = 0; - imux->num_items = 1; - goto again; - } - /* check whether ADC-switch is possible */ for (i = 0; i < imux->num_items; i++) { for (n = 0; n < spec->num_adc_nids; n++) { @@ -2701,7 +2905,8 @@ static int check_dyn_adc_switch(struct hda_codec *codec) spec->num_adc_nids = nums; } - if (imux->num_items == 1 || spec->shared_mic_hp) { + if (imux->num_items == 1 || + (imux->num_items == 2 && spec->hp_mic)) { snd_printdd("hda-codec: reducing to a single ADC\n"); spec->num_adc_nids = 1; /* reduce to a single ADC */ } @@ -2738,6 +2943,8 @@ static int parse_capture_source(struct hda_codec *codec, hda_nid_t pin, snd_hda_get_path_idx(codec, path); if (!imux_added) { + if (spec->hp_mic_pin == pin) + spec->hp_mic_mux_idx = imux->num_items; spec->imux_pins[imux->num_items] = pin; snd_hda_add_imux_item(imux, label, cfg_idx, NULL); imux_added = true; @@ -2812,7 +3019,8 @@ static int create_input_ctls(struct hda_codec *codec) val = PIN_IN; if (cfg->inputs[i].type == AUTO_PIN_MIC) val |= snd_hda_get_default_vref(codec, pin); - set_pin_target(codec, pin, val, false); + if (pin != spec->hp_mic_pin) + set_pin_target(codec, pin, val, false); if (mixer) { if (is_reachable_path(codec, pin, mixer)) { @@ -2830,7 +3038,7 @@ static int create_input_ctls(struct hda_codec *codec) if (err < 0) return err; - if (spec->add_in_jack_modes) { + if (spec->add_jack_modes) { err = create_in_jack_mode(codec, pin); if (err < 0) return err; @@ -3462,8 +3670,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, spec->cur_mux[adc_idx] = idx; - if (spec->shared_mic_hp) - update_shared_mic_hp(codec, spec->cur_mux[adc_idx]); + if (spec->hp_mic) + update_hp_mic(codec, adc_idx, false); if (spec->dyn_adc_switch) dyn_adc_pcm_resetup(codec, idx); @@ -3511,18 +3719,21 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, for (i = 0; i < num_pins; i++) { hda_nid_t nid = pins[i]; - unsigned int val; + unsigned int val, oldval; if (!nid) break; + oldval = snd_hda_codec_get_pin_target(codec, nid); + if (oldval & PIN_IN) + continue; /* no mute for inputs */ /* don't reset VREF value in case it's controlling * the amp (see alc861_fixup_asus_amp_vref_0f()) */ if (spec->keep_vref_in_automute) - val = snd_hda_codec_get_pin_target(codec, nid) & ~PIN_HP; + val = oldval & ~PIN_HP; else val = 0; if (!mute) - val |= snd_hda_codec_get_pin_target(codec, nid); + val |= oldval; /* here we call update_pin_ctl() so that the pinctl is changed * without changing the pinctl target value; * the original target value will be still referred at the @@ -3543,8 +3754,7 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec) * in general, HP pins/amps control should be enabled in all cases, * but currently set only for master_mute, just to be safe */ - if (!spec->shared_mic_hp) /* don't change HP-pin when shared with mic */ - do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), + do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins, spec->master_mute); if (!spec->automute_speaker) @@ -3649,10 +3859,7 @@ static void update_automute_all(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; - if (spec->hp_automute_hook) - spec->hp_automute_hook(codec, NULL); - else - snd_hda_gen_hp_automute(codec, NULL); + update_hp_automute_hook(codec); if (spec->line_automute_hook) spec->line_automute_hook(codec, NULL); else @@ -3978,6 +4185,11 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, cfg = &spec->autocfg; } + if (!spec->main_out_badness) + spec->main_out_badness = &hda_main_out_badness; + if (!spec->extra_out_badness) + spec->extra_out_badness = &hda_extra_out_badness; + fill_all_dac_nids(codec); if (!cfg->line_outs) { @@ -4024,7 +4236,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, err = create_loopback_mixing_ctl(codec); if (err < 0) return err; - err = create_shared_input(codec); + err = create_hp_mic(codec); if (err < 0) return err; err = create_input_ctls(codec); @@ -4050,11 +4262,9 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; - if (!spec->shared_mic_hp) { - err = check_auto_mic_availability(codec); - if (err < 0) - return err; - } + err = check_auto_mic_availability(codec); + if (err < 0) + return err; err = create_capture_mixers(codec); if (err < 0) @@ -4064,7 +4274,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; - if (spec->add_out_jack_modes) { + if (spec->add_jack_modes) { if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = create_out_jack_modes(codec, cfg->line_outs, cfg->line_out_pins); @@ -4085,6 +4295,12 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (spec->power_down_unused) codec->power_filter = snd_hda_gen_path_power_filter; + if (!spec->no_analog && spec->beep_nid) { + err = snd_hda_attach_beep_device(codec, spec->beep_nid); + if (err < 0) + return err; + } + return 1; } EXPORT_SYMBOL_HDA(snd_hda_gen_parse_auto_config); @@ -4161,17 +4377,6 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) free_kctls(spec); /* no longer needed */ - 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; - } - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); if (err < 0) return err; @@ -4729,7 +4934,8 @@ static void set_output_and_unmute(struct hda_codec *codec, int path_idx) return; pin = path->path[path->depth - 1]; restore_pin_ctl(codec, pin); - snd_hda_activate_path(codec, path, path->active, true); + snd_hda_activate_path(codec, path, path->active, + aamix_default(codec->spec)); set_pin_eapd(codec, pin, path->active); } @@ -4779,7 +4985,8 @@ static void init_multi_io(struct hda_codec *codec) if (!spec->multi_io[i].ctl_in) spec->multi_io[i].ctl_in = snd_hda_codec_get_pin_target(codec, pin); - snd_hda_activate_path(codec, path, path->active, true); + snd_hda_activate_path(codec, path, path->active, + aamix_default(spec)); } } @@ -4826,11 +5033,10 @@ static void init_input_src(struct hda_codec *codec) snd_hda_activate_path(codec, path, active, false); } } + if (spec->hp_mic) + update_hp_mic(codec, c, true); } - if (spec->shared_mic_hp) - update_shared_mic_hp(codec, spec->cur_mux[0]); - if (spec->cap_sync_hook) spec->cap_sync_hook(codec, NULL); } @@ -4911,6 +5117,7 @@ EXPORT_SYMBOL_HDA(snd_hda_gen_init); */ void snd_hda_gen_free(struct hda_codec *codec) { + snd_hda_detach_beep_device(codec); snd_hda_gen_spec_free(codec->spec); kfree(codec->spec); codec->spec = NULL; diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 009b57be96d3..54e665160379 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -76,6 +76,19 @@ enum { HDA_GEN_PCM_ACT_CLOSE, }; +/* DAC assignment badness table */ +struct badness_table { + int no_primary_dac; /* no primary DAC */ + int no_dac; /* no secondary DACs */ + int shared_primary; /* primary DAC is shared with main output */ + int shared_surr; /* secondary DAC shared with main or primary */ + int shared_clfe; /* third DAC shared with main or primary */ + int shared_surr_main; /* secondary DAC sahred with main/DAC0 */ +}; + +extern const struct badness_table hda_main_out_badness; +extern const struct badness_table hda_extra_out_badness; + struct hda_gen_spec { char stream_name_analog[32]; /* analog PCM stream */ const struct hda_pcm_stream *stream_analog_playback; @@ -145,7 +158,10 @@ struct hda_gen_spec { hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS]; unsigned int dyn_adc_idx[HDA_MAX_NUM_INPUTS]; + /* shared hp/mic */ hda_nid_t shared_mic_vref_pin; + hda_nid_t hp_mic_pin; + int hp_mic_mux_idx; /* DAC/ADC lists */ int num_all_dacs; @@ -200,7 +216,8 @@ struct hda_gen_spec { /* other parse behavior flags */ unsigned int need_dac_fix:1; /* need to limit DACs for multi channels */ - unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */ + unsigned int hp_mic:1; /* Allow HP as a mic-in */ + unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */ unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */ unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */ unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */ @@ -209,8 +226,7 @@ struct hda_gen_spec { unsigned int indep_hp:1; /* independent HP supported */ unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */ unsigned int add_stereo_mix_input:1; /* add aamix as a capture src */ - unsigned int add_out_jack_modes:1; /* add output jack mode enum ctls */ - unsigned int add_in_jack_modes:1; /* add input jack mode enum ctls */ + unsigned int add_jack_modes:1; /* add i/o jack mode enum ctls */ unsigned int power_down_unused:1; /* power down unused widgets */ /* other internal flags */ @@ -218,10 +234,18 @@ struct hda_gen_spec { unsigned int dyn_adc_switch:1; /* switch ADCs (for ALC275) */ unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; + unsigned int hp_mic_jack_modes:1; + + /* badness tables for output path evaluations */ + const struct badness_table *main_out_badness; + const struct badness_table *extra_out_badness; /* loopback mixing mode */ bool aamix_mode; + /* digital beep */ + hda_nid_t beep_nid; + /* for virtual master */ hda_nid_t vmaster_nid; unsigned int vmaster_tlv[4]; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bcd40ee488e3..7b213d589ef6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,6 +1889,26 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static u64 azx_adjust_codec_delay(struct snd_pcm_substream *substream, + u64 nsec) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + u64 codec_frames, codec_nsecs; + + if (!hinfo->ops.get_delay) + return nsec; + + codec_frames = hinfo->ops.get_delay(hinfo, apcm->codec, substream); + codec_nsecs = div_u64(codec_frames * 1000000000LL, + substream->runtime->rate); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return nsec + codec_nsecs; + + return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; +} + static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, struct timespec *ts) { @@ -1897,6 +1917,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ + nsec = azx_adjust_codec_delay(substream, nsec); *ts = ns_to_timespec(nsec); @@ -2349,8 +2370,11 @@ static unsigned int azx_get_position(struct azx *chip, struct azx_dev *azx_dev, bool with_check) { + struct snd_pcm_substream *substream = azx_dev->substream; + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); unsigned int pos; - int stream = azx_dev->substream->stream; + int stream = substream->stream; + struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; int delay = 0; switch (chip->position_fix[stream]) { @@ -2381,7 +2405,7 @@ static unsigned int azx_get_position(struct azx *chip, pos = 0; /* calculate runtime delay from LPIB */ - if (azx_dev->substream->runtime && + if (substream->runtime && chip->position_fix[stream] == POS_FIX_POSBUF && (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)) { unsigned int lpib_pos = azx_sd_readl(azx_dev, SD_LPIB); @@ -2399,9 +2423,16 @@ static unsigned int azx_get_position(struct azx *chip, delay = 0; chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; } - azx_dev->substream->runtime->delay = - bytes_to_frames(azx_dev->substream->runtime, delay); + delay = bytes_to_frames(substream->runtime, delay); } + + if (substream->runtime) { + if (hinfo->ops.get_delay) + delay += hinfo->ops.get_delay(hinfo, apcm->codec, + substream); + substream->runtime->delay = delay; + } + trace_azx_get_position(chip, azx_dev, pos, delay); return pos; } diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 1d035efeff4f..9e0a95288f46 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -394,7 +394,8 @@ static int get_unique_index(struct hda_codec *codec, const char *name, int idx) } static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg) + const struct auto_pin_cfg *cfg, + const char *base_name) { unsigned int def_conf, conn; char name[44]; @@ -410,7 +411,11 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, 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 (base_name) { + strlcpy(name, base_name, sizeof(name)); + idx = 0; + } else + snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); if (phantom_jack) /* Example final name: "Internal Mic Phantom Jack" */ strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); @@ -433,39 +438,51 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const hda_nid_t *p; int i, err; + for (i = 0; i < cfg->num_inputs; i++) { + /* If we have headphone mics; make sure they get the right name + before grabbed by output pins */ + if (cfg->inputs[i].is_headphone_mic) { + if (auto_cfg_hp_outs(cfg) == 1) + err = add_jack_kctl(codec, auto_cfg_hp_pins(cfg)[0], + cfg, "Headphone Mic"); + else + err = add_jack_kctl(codec, cfg->inputs[i].pin, + cfg, "Headphone Mic"); + } else + err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, + NULL); + if (err < 0) + return err; + } + for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, NULL); 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); + err = add_jack_kctl(codec, *p, cfg, NULL); 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); - if (err < 0) - return err; - } - for (i = 0; i < cfg->num_inputs; i++) { - err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg); + err = add_jack_kctl(codec, *p, cfg, NULL); 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); + err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; } - err = add_jack_kctl(codec, cfg->dig_in_pin, cfg); + err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, NULL); if (err < 0) return err; - err = add_jack_kctl(codec, cfg->mono_out_pin, cfg); + err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, NULL); if (err < 0) return err; return 0; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 83b7486c8eff..e0bf7534fa1f 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -670,6 +670,10 @@ snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid, return (state != target_state); } +unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state); + /* * AMP control callbacks */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index df8014b27596..977b0d878dae 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -43,7 +43,6 @@ struct ad198x_spec { hda_nid_t eapd_nid; unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - hda_nid_t beep_dev_nid; #ifdef ENABLE_AD_STATIC_QUIRKS const struct snd_kcontrol_new *mixers[6]; @@ -609,7 +608,7 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = { .build_controls = ad198x_auto_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, - .free = ad198x_free, + .free = snd_hda_gen_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, @@ -638,12 +637,6 @@ static int ad198x_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - if (spec->beep_dev_nid) { - err = snd_hda_attach_beep_device(codec, spec->beep_dev_nid); - if (err < 0) - return err; - } - codec->patch_ops = ad198x_auto_patch_ops; return 0; @@ -1240,7 +1233,7 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec) codec->inv_eapd = 1; spec->gen.mixer_nid = 0x07; - spec->beep_dev_nid = 0x19; + spec->gen.beep_nid = 0x19; set_beep_amp(spec, 0x18, 0, HDA_OUTPUT); /* AD1986A has a hardware problem that it can't share a stream @@ -1256,7 +1249,7 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec) err = ad198x_parse_auto_config(codec); if (err < 0) { - ad198x_free(codec); + snd_hda_gen_free(codec); return err; } @@ -1673,7 +1666,7 @@ static int ad1983_parse_auto_config(struct hda_codec *codec) return err; spec = codec->spec; - spec->beep_dev_nid = 0x10; + spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); err = ad198x_parse_auto_config(codec); if (err < 0) @@ -1684,7 +1677,7 @@ static int ad1983_parse_auto_config(struct hda_codec *codec) return 0; error: - ad198x_free(codec); + snd_hda_gen_free(codec); return err; } @@ -2187,7 +2180,7 @@ static int ad1981_parse_auto_config(struct hda_codec *codec) spec = codec->spec; spec->gen.mixer_nid = 0x0e; - spec->beep_dev_nid = 0x10; + spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT); snd_hda_pick_fixup(codec, NULL, ad1981_fixup_tbl, ad1981_fixups); @@ -2205,7 +2198,7 @@ static int ad1981_parse_auto_config(struct hda_codec *codec) return 0; error: - ad198x_free(codec); + snd_hda_gen_free(codec); return err; } @@ -3236,7 +3229,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) spec->gen.mixer_nid = 0x20; spec->gen.mixer_merge_nid = 0x21; - spec->beep_dev_nid = 0x10; + spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); err = ad198x_parse_auto_config(codec); if (err < 0) @@ -3247,7 +3240,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) return 0; error: - ad198x_free(codec); + snd_hda_gen_free(codec); return err; } @@ -3653,7 +3646,7 @@ static int ad1884_parse_auto_config(struct hda_codec *codec) spec = codec->spec; spec->gen.mixer_nid = 0x20; - spec->beep_dev_nid = 0x10; + spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); snd_hda_pick_fixup(codec, NULL, ad1884_fixup_tbl, ad1884_fixups); @@ -3671,7 +3664,7 @@ static int ad1884_parse_auto_config(struct hda_codec *codec) return 0; error: - ad198x_free(codec); + snd_hda_gen_free(codec); return err; } @@ -5155,7 +5148,7 @@ static int ad1882_parse_auto_config(struct hda_codec *codec) spec->gen.mixer_nid = 0x20; spec->gen.mixer_merge_nid = 0x21; - spec->beep_dev_nid = 0x10; + spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); err = ad198x_parse_auto_config(codec); if (err < 0) @@ -5166,7 +5159,7 @@ static int ad1882_parse_auto_config(struct hda_codec *codec) return 0; error: - ad198x_free(codec); + snd_hda_gen_free(codec); return err; } diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0792b5725f9c..90ff7a3f72df 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -131,6 +131,13 @@ enum { /* Effects values size*/ #define EFFECT_VALS_MAX_COUNT 12 +/* Latency introduced by DSP blocks in milliseconds. */ +#define DSP_CAPTURE_INIT_LATENCY 0 +#define DSP_CRYSTAL_VOICE_LATENCY 124 +#define DSP_PLAYBACK_INIT_LATENCY 13 +#define DSP_PLAY_ENHANCEMENT_LATENCY 30 +#define DSP_SPEAKER_OUT_LATENCY 7 + struct ct_effect { char name[44]; hda_nid_t nid; @@ -741,6 +748,9 @@ struct ca0132_spec { long voicefx_val; long cur_mic_boost; + struct hda_codec *codec; + struct delayed_work unsol_hp_work; + #ifdef ENABLE_TUNING_CONTROLS long cur_ctl_vals[TUNING_CTLS_COUNT]; #endif @@ -2740,6 +2750,31 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } +static unsigned int ca0132_playback_pcm_delay(struct hda_pcm_stream *info, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int latency = DSP_PLAYBACK_INIT_LATENCY; + struct snd_pcm_runtime *runtime = substream->runtime; + + if (spec->dsp_state != DSP_DOWNLOADED) + return 0; + + /* Add latency if playback enhancement and either effect is enabled. */ + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) { + if ((spec->effects_switch[SURROUND - EFFECT_START_NID]) || + (spec->effects_switch[DIALOG_PLUS - EFFECT_START_NID])) + latency += DSP_PLAY_ENHANCEMENT_LATENCY; + } + + /* Applying Speaker EQ adds latency as well. */ + if (spec->cur_out_type == SPEAKER_OUT) + latency += DSP_SPEAKER_OUT_LATENCY; + + return (latency * runtime->rate) / 1000; +} + /* * Digital out */ @@ -2808,6 +2843,23 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } +static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int latency = DSP_CAPTURE_INIT_LATENCY; + struct snd_pcm_runtime *runtime = substream->runtime; + + if (spec->dsp_state != DSP_DOWNLOADED) + return 0; + + if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) + latency += DSP_CRYSTAL_VOICE_LATENCY; + + return (latency * runtime->rate) / 1000; +} + /* * Controls stuffs. */ @@ -3227,6 +3279,14 @@ exit: return err < 0 ? err : 0; } +static void ca0132_unsol_hp_delayed(struct work_struct *work) +{ + struct ca0132_spec *spec = container_of( + to_delayed_work(work), struct ca0132_spec, unsol_hp_work); + ca0132_select_out(spec->codec); + snd_hda_jack_report_sync(spec->codec); +} + static void ca0132_set_dmic(struct hda_codec *codec, int enable); static int ca0132_mic_boost_set(struct hda_codec *codec, long val); static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val); @@ -3991,7 +4051,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = { .channels_max = 6, .ops = { .prepare = ca0132_playback_pcm_prepare, - .cleanup = ca0132_playback_pcm_cleanup + .cleanup = ca0132_playback_pcm_cleanup, + .get_delay = ca0132_playback_pcm_delay, }, }; @@ -4001,7 +4062,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = { .channels_max = 2, .ops = { .prepare = ca0132_capture_pcm_prepare, - .cleanup = ca0132_capture_pcm_cleanup + .cleanup = ca0132_capture_pcm_cleanup, + .get_delay = ca0132_capture_pcm_delay, }, }; @@ -4399,8 +4461,7 @@ static void ca0132_process_dsp_response(struct hda_codec *codec) static void ca0132_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd(KERN_INFO "ca0132_unsol_event: 0x%x\n", res); - + struct ca0132_spec *spec = codec->spec; if (((res >> AC_UNSOL_RES_TAG_SHIFT) & 0x3f) == UNSOL_TAG_DSP) { ca0132_process_dsp_response(codec); @@ -4412,8 +4473,13 @@ static void ca0132_unsol_event(struct hda_codec *codec, unsigned int res) switch (res) { case UNSOL_TAG_HP: - ca0132_select_out(codec); - snd_hda_jack_report_sync(codec); + /* Delay enabling the HP amp, to let the mic-detection + * state machine run. + */ + cancel_delayed_work_sync(&spec->unsol_hp_work); + queue_delayed_work(codec->bus->workq, + &spec->unsol_hp_work, + msecs_to_jiffies(500)); break; case UNSOL_TAG_AMIC1: ca0132_select_mic(codec); @@ -4588,6 +4654,7 @@ static void ca0132_free(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; + cancel_delayed_work_sync(&spec->unsol_hp_work); snd_hda_power_up(codec); snd_hda_sequence_write(codec, spec->base_exit_verbs); ca0132_exit_chip(codec); @@ -4653,6 +4720,7 @@ static int patch_ca0132(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + spec->codec = codec; spec->num_mixers = 1; spec->mixers[0] = ca0132_mixer; @@ -4663,6 +4731,8 @@ static int patch_ca0132(struct hda_codec *codec) spec->init_verbs[1] = ca0132_init_verbs1; spec->num_init_verbs = 2; + INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); + ca0132_init_chip(codec); ca0132_config(codec); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0d9c58f13560..bd8d46cca2b3 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -68,6 +68,7 @@ enum { enum { CS421X_CDB4210, CS421X_SENSE_B, + CS421X_STUMPY, }; /* Vendor-specific processing widget */ @@ -538,6 +539,7 @@ static int patch_cs420x(struct hda_codec *codec) /* CS4210 board names */ static const struct hda_model_fixup cs421x_models[] = { { .id = CS421X_CDB4210, .name = "cdb4210" }, + { .id = CS421X_STUMPY, .name = "stumpy" }, {} }; @@ -559,6 +561,17 @@ static const struct hda_pintbl cdb4210_pincfgs[] = { {} /* terminator */ }; +/* Stumpy ChromeBox */ +static const struct hda_pintbl stumpy_pincfgs[] = { + { 0x05, 0x022120f0 }, + { 0x06, 0x901700f0 }, + { 0x07, 0x02a120f0 }, + { 0x08, 0x77a70037 }, + { 0x09, 0x77a6003e }, + { 0x0a, 0x434510f0 }, + {} /* terminator */ +}; + /* Setup GPIO/SENSE for each board (if used) */ static void cs421x_fixup_sense_b(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -578,7 +591,11 @@ static const struct hda_fixup cs421x_fixups[] = { [CS421X_SENSE_B] = { .type = HDA_FIXUP_FUNC, .v.func = cs421x_fixup_sense_b, - } + }, + [CS421X_STUMPY] = { + .type = HDA_FIXUP_PINS, + .v.pins = stumpy_pincfgs, + }, }; static const struct hda_verb cs421x_coef_init_verbs[] = { diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2a89d1eefeb6..84b81c874a4a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -139,8 +139,12 @@ struct conexant_spec { #ifdef CONFIG_SND_HDA_INPUT_BEEP -#define set_beep_amp(spec, nid, idx, dir) \ - ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) +static inline void set_beep_amp(struct conexant_spec *spec, hda_nid_t nid, + int idx, int dir) +{ + spec->gen.beep_nid = nid; + spec->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); +} /* additional beep mixers; the actual parameters are overwritten at build */ static const struct snd_kcontrol_new cxt_beep_mixer[] = { HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT), @@ -2942,7 +2946,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), @@ -3191,17 +3194,11 @@ static int cx_auto_build_controls(struct hda_codec *codec) return 0; } -static void cx_auto_free(struct hda_codec *codec) -{ - snd_hda_detach_beep_device(codec); - snd_hda_gen_free(codec); -} - static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, - .free = cx_auto_free, + .free = snd_hda_gen_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, @@ -3310,6 +3307,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), @@ -3356,7 +3354,6 @@ static int patch_conexant_auto(struct hda_codec *codec) switch (codec->vendor_id) { case 0x14f15045: codec->single_adc_amp = 1; - codec->power_filter = NULL; /* Needs speaker amp to D3 to avoid click */ break; case 0x14f15047: codec->pin_amp_workaround = 1; @@ -3396,8 +3393,6 @@ static int patch_conexant_auto(struct hda_codec *codec) goto error; codec->patch_ops = cx_auto_patch_ops; - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); /* Some laptops with Conexant chips show stalls in S3 resume, * which falls into the single-cmd mode. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index de8ac5c07fd0..32930e668854 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -44,16 +44,6 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); -/* - * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device - * could support N independent pipes, each of them can be connected to one or - * more ports (DVI, HDMI or DisplayPort). - * - * The HDA correspondence of pipes/ports are converter/pin nodes. - */ -#define MAX_HDMI_CVTS 8 -#define MAX_HDMI_PINS 8 - struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -80,16 +70,17 @@ struct hdmi_spec_per_pin { bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ + char pcm_name[8]; /* filled in build_pcm callbacks */ }; struct hdmi_spec { int num_cvts; - struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; - hda_nid_t cvt_nids[MAX_HDMI_CVTS]; + struct snd_array cvts; /* struct hdmi_spec_per_cvt */ + hda_nid_t cvt_nids[4]; /* only for haswell fix */ int num_pins; - struct hdmi_spec_per_pin pins[MAX_HDMI_PINS]; - struct hda_pcm pcm_rec[MAX_HDMI_PINS]; + struct snd_array pins; /* struct hdmi_spec_per_pin */ + struct snd_array pcm_rec; /* struct hda_pcm */ unsigned int channels_max; /* max over all cvts */ struct hdmi_eld temp_eld; @@ -304,12 +295,19 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { * HDMI routines */ +#define get_pin(spec, idx) \ + ((struct hdmi_spec_per_pin *)snd_array_elem(&spec->pins, idx)) +#define get_cvt(spec, idx) \ + ((struct hdmi_spec_per_cvt *)snd_array_elem(&spec->cvts, idx)) +#define get_pcm_rec(spec, idx) \ + ((struct hda_pcm *)snd_array_elem(&spec->pcm_rec, idx)) + static int pin_nid_to_pin_index(struct hdmi_spec *spec, hda_nid_t pin_nid) { int pin_idx; for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) - if (spec->pins[pin_idx].pin_nid == pin_nid) + if (get_pin(spec, pin_idx)->pin_nid == pin_nid) return pin_idx; snd_printk(KERN_WARNING "HDMI: pin nid %d not registered\n", pin_nid); @@ -322,7 +320,7 @@ static int hinfo_to_pin_index(struct hdmi_spec *spec, int pin_idx; for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) - if (&spec->pcm_rec[pin_idx].stream[0] == hinfo) + if (get_pcm_rec(spec, pin_idx)->stream == hinfo) return pin_idx; snd_printk(KERN_WARNING "HDMI: hinfo %p not registered\n", hinfo); @@ -334,7 +332,7 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) int cvt_idx; for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) - if (spec->cvts[cvt_idx].cvt_nid == cvt_nid) + if (get_cvt(spec, cvt_idx)->cvt_nid == cvt_nid) return cvt_idx; snd_printk(KERN_WARNING "HDMI: cvt nid %d not registered\n", cvt_nid); @@ -352,7 +350,7 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; pin_idx = kcontrol->private_value; - eld = &spec->pins[pin_idx].sink_eld; + eld = &get_pin(spec, pin_idx)->sink_eld; mutex_lock(&eld->lock); uinfo->count = eld->eld_valid ? eld->eld_size : 0; @@ -370,7 +368,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, int pin_idx; pin_idx = kcontrol->private_value; - eld = &spec->pins[pin_idx].sink_eld; + eld = &get_pin(spec, pin_idx)->sink_eld; mutex_lock(&eld->lock); if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data)) { @@ -410,11 +408,11 @@ static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, kctl->private_value = pin_idx; kctl->id.device = device; - err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl); + err = snd_hda_ctl_add(codec, get_pin(spec, pin_idx)->pin_nid, kctl); if (err < 0) return err; - spec->pins[pin_idx].eld_ctl = kctl; + get_pin(spec, pin_idx)->eld_ctl = kctl; return 0; } @@ -875,14 +873,14 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; int channels = substream->runtime->channels; struct hdmi_eld *eld; int ca; union audio_infoframe ai; - eld = &spec->pins[pin_idx].sink_eld; + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -977,7 +975,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; - hdmi_present_sense(&spec->pins[pin_idx], 1); + hdmi_present_sense(get_pin(spec, pin_idx), 1); snd_hda_jack_report_sync(codec); } @@ -1020,6 +1018,41 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } +static void haswell_verify_pin_D0(struct hda_codec *codec, hda_nid_t nid) +{ + int pwr, lamp, ramp; + + pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); + pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; + if (pwr != AC_PWRST_D0) { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + msleep(40); + pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); + pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; + snd_printd("Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); + } + + lamp = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); + ramp = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); + if (lamp != ramp) { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT | lamp); + + lamp = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); + ramp = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); + snd_printd("Haswell HDMI audio: Mute after set on pin 0x%x: [0x%x 0x%x]\n", nid, lamp, ramp); + } +} + /* * Callbacks */ @@ -1034,6 +1067,9 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int pinctl; int new_pinctl = 0; + if (codec->vendor_id == 0x80862807) + haswell_verify_pin_D0(codec, pin_nid); + if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); @@ -1083,12 +1119,12 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, pin_idx = hinfo_to_pin_index(spec, hinfo); if (snd_BUG_ON(pin_idx < 0)) return -EINVAL; - per_pin = &spec->pins[pin_idx]; + per_pin = get_pin(spec, pin_idx); eld = &per_pin->sink_eld; /* Dynamically assign converter to stream */ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { - per_cvt = &spec->cvts[cvt_idx]; + per_cvt = get_cvt(spec, cvt_idx); /* Must not already be assigned */ if (per_cvt->assigned) @@ -1151,7 +1187,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { @@ -1275,14 +1311,13 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) return 0; - if (snd_BUG_ON(spec->num_pins >= MAX_HDMI_PINS)) - return -E2BIG; - if (codec->vendor_id == 0x80862807) intel_haswell_fixup_connect_list(codec, pin_nid); pin_idx = spec->num_pins; - per_pin = &spec->pins[pin_idx]; + per_pin = snd_array_new(&spec->pins); + if (!per_pin) + return -ENOMEM; per_pin->pin_nid = pin_nid; per_pin->non_pcm = false; @@ -1299,19 +1334,16 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) { struct hdmi_spec *spec = codec->spec; - int cvt_idx; struct hdmi_spec_per_cvt *per_cvt; unsigned int chans; int err; - if (snd_BUG_ON(spec->num_cvts >= MAX_HDMI_CVTS)) - return -E2BIG; - chans = get_wcaps(codec, cvt_nid); chans = get_wcaps_channels(chans); - cvt_idx = spec->num_cvts; - per_cvt = &spec->cvts[cvt_idx]; + per_cvt = snd_array_new(&spec->cvts); + if (!per_cvt) + return -ENOMEM; per_cvt->cvt_nid = cvt_nid; per_cvt->channels_min = 2; @@ -1328,7 +1360,9 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) if (err < 0) return err; - spec->cvt_nids[spec->num_cvts++] = cvt_nid; + if (spec->num_cvts < ARRAY_SIZE(spec->cvt_nids)) + spec->cvt_nids[spec->num_cvts] = cvt_nid; + spec->num_cvts++; return 0; } @@ -1384,13 +1418,6 @@ static int hdmi_parse_codec(struct hda_codec *codec) /* */ -static char *get_hdmi_pcm_name(int idx) -{ - static char names[MAX_HDMI_PINS][8]; - sprintf(&names[idx][0], "HDMI %d", idx); - return &names[idx][0]; -} - static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) { struct hda_spdif_out *spdif; @@ -1417,7 +1444,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hda_nid_t cvt_nid = hinfo->nid; 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; + hda_nid_t pin_nid = get_pin(spec, pin_idx)->pin_nid; bool non_pcm; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); @@ -1450,7 +1477,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid); if (snd_BUG_ON(cvt_idx < 0)) return -EINVAL; - per_cvt = &spec->cvts[cvt_idx]; + per_cvt = get_cvt(spec, cvt_idx); snd_BUG_ON(!per_cvt->assigned); per_cvt->assigned = 0; @@ -1459,7 +1486,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, pin_idx = hinfo_to_pin_index(spec, hinfo); if (snd_BUG_ON(pin_idx < 0)) return -EINVAL; - per_pin = &spec->pins[pin_idx]; + per_pin = get_pin(spec, pin_idx); snd_hda_spdif_ctls_unassign(codec, pin_idx); per_pin->chmap_set = false; @@ -1553,7 +1580,7 @@ static int hdmi_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct hda_codec *codec = info->private_data; struct hdmi_spec *spec = codec->spec; int pin_idx = kcontrol->private_value; - struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); int i; for (i = 0; i < ARRAY_SIZE(per_pin->chmap); i++) @@ -1568,7 +1595,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = info->private_data; struct hdmi_spec *spec = codec->spec; int pin_idx = kcontrol->private_value; - struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); unsigned int ctl_idx; struct snd_pcm_substream *substream; unsigned char chmap[8]; @@ -1613,9 +1640,14 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hda_pcm *info; struct hda_pcm_stream *pstr; - - info = &spec->pcm_rec[pin_idx]; - info->name = get_hdmi_pcm_name(pin_idx); + struct hdmi_spec_per_pin *per_pin; + + per_pin = get_pin(spec, pin_idx); + sprintf(per_pin->pcm_name, "HDMI %d", pin_idx); + info = snd_array_new(&spec->pcm_rec); + if (!info) + return -ENOMEM; + info->name = per_pin->pcm_name; info->pcm_type = HDA_PCM_TYPE_HDMI; info->own_chmap = true; @@ -1626,7 +1658,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) } codec->num_pcms = spec->num_pins; - codec->pcm_info = spec->pcm_rec; + codec->pcm_info = spec->pcm_rec.list; return 0; } @@ -1635,8 +1667,8 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) { char hdmi_str[32] = "HDMI/DP"; struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; - int pcmdev = spec->pcm_rec[pin_idx].device; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + int pcmdev = get_pcm_rec(spec, pin_idx)->device; if (pcmdev > 0) sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); @@ -1654,7 +1686,7 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) 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]; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); err = generic_hdmi_build_jack(codec, pin_idx); if (err < 0) @@ -1669,9 +1701,8 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) snd_hda_spdif_ctls_unassign(codec, pin_idx); /* add control for ELD Bytes */ - err = hdmi_create_eld_ctl(codec, - pin_idx, - spec->pcm_rec[pin_idx].device); + err = hdmi_create_eld_ctl(codec, pin_idx, + get_pcm_rec(spec, pin_idx)->device); if (err < 0) return err; @@ -1709,7 +1740,7 @@ static int generic_hdmi_init_per_pins(struct hda_codec *codec) 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]; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); struct hdmi_eld *eld = &per_pin->sink_eld; per_pin->codec = codec; @@ -1726,7 +1757,7 @@ static int generic_hdmi_init(struct hda_codec *codec) 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]; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; hdmi_init_pin(codec, pin_nid); @@ -1735,13 +1766,27 @@ static int generic_hdmi_init(struct hda_codec *codec) return 0; } +static void hdmi_array_init(struct hdmi_spec *spec, int nums) +{ + snd_array_init(&spec->pins, sizeof(struct hdmi_spec_per_pin), nums); + snd_array_init(&spec->cvts, sizeof(struct hdmi_spec_per_cvt), nums); + snd_array_init(&spec->pcm_rec, sizeof(struct hda_pcm), nums); +} + +static void hdmi_array_free(struct hdmi_spec *spec) +{ + snd_array_free(&spec->pins); + snd_array_free(&spec->cvts); + snd_array_free(&spec->pcm_rec); +} + static void generic_hdmi_free(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]; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); struct hdmi_eld *eld = &per_pin->sink_eld; cancel_delayed_work(&per_pin->work); @@ -1749,6 +1794,7 @@ static void generic_hdmi_free(struct hda_codec *codec) } flush_workqueue(codec->bus->workq); + hdmi_array_free(spec); kfree(spec); } @@ -1775,6 +1821,7 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, /* override pins connection list */ snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); + nconns = max(spec->num_cvts, 4); snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } @@ -1855,6 +1902,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + hdmi_array_init(spec, 4); snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -1882,24 +1930,30 @@ static int patch_generic_hdmi(struct hda_codec *codec) static int simple_playback_build_pcms(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; + struct hda_pcm *info; unsigned int chans; struct hda_pcm_stream *pstr; + struct hdmi_spec_per_cvt *per_cvt; - codec->num_pcms = 1; - codec->pcm_info = info; - - chans = get_wcaps(codec, spec->cvts[0].cvt_nid); + per_cvt = get_cvt(spec, 0); + chans = get_wcaps(codec, per_cvt->cvt_nid); chans = get_wcaps_channels(chans); - info->name = get_hdmi_pcm_name(0); + info = snd_array_new(&spec->pcm_rec); + if (!info) + return -ENOMEM; + info->name = get_pin(spec, 0)->pcm_name; + sprintf(info->name, "HDMI 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; + pstr->nid = per_cvt->cvt_nid; if (pstr->channels_max <= 2 && chans && chans <= 16) pstr->channels_max = chans; + codec->num_pcms = 1; + codec->pcm_info = info; + return 0; } @@ -1919,11 +1973,12 @@ static void simple_hdmi_unsol_event(struct hda_codec *codec, static int simple_playback_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_cvt *per_cvt; int err; - err = snd_hda_create_spdif_out_ctls(codec, - spec->cvts[0].cvt_nid, - spec->cvts[0].cvt_nid); + per_cvt = get_cvt(spec, 0); + err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid, + per_cvt->cvt_nid); if (err < 0) return err; return simple_hdmi_build_jack(codec, 0); @@ -1932,7 +1987,8 @@ static int simple_playback_build_controls(struct hda_codec *codec) static int simple_playback_init(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - hda_nid_t pin = spec->pins[0].pin_nid; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, 0); + hda_nid_t pin = per_pin->pin_nid; snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); @@ -1948,6 +2004,7 @@ static void simple_playback_free(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; + hdmi_array_free(spec); kfree(spec); } @@ -2111,20 +2168,29 @@ static int patch_simple_hdmi(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t pin_nid) { struct hdmi_spec *spec; + struct hdmi_spec_per_cvt *per_cvt; + struct hdmi_spec_per_pin *per_pin; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; codec->spec = spec; + hdmi_array_init(spec, 1); 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; + per_pin = snd_array_new(&spec->pins); + per_cvt = snd_array_new(&spec->cvts); + if (!per_pin || !per_cvt) { + simple_playback_free(codec); + return -ENOMEM; + } + per_cvt->cvt_nid = cvt_nid; + per_pin->pin_nid = pin_nid; spec->pcm_playback = simple_pcm_playback; codec->patch_ops = simple_hdmi_patch_ops; @@ -2201,9 +2267,11 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, int i; struct hdmi_spec *spec = codec->spec; struct hda_spdif_out *spdif; + struct hdmi_spec_per_cvt *per_cvt; mutex_lock(&codec->spdif_mutex); - spdif = snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid); + per_cvt = get_cvt(spec, 0); + spdif = snd_hda_spdif_out_of_nid(codec, per_cvt->cvt_nid); chs = substream->runtime->channels; @@ -2325,13 +2393,17 @@ static int nvhdmi_7x_8ch_build_pcms(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; int err = simple_playback_build_pcms(codec); - spec->pcm_rec[0].own_chmap = true; + if (!err) { + struct hda_pcm *info = get_pcm_rec(spec, 0); + info->own_chmap = true; + } return err; } static int nvhdmi_7x_8ch_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; + struct hda_pcm *info; struct snd_pcm_chmap *chmap; int err; @@ -2340,7 +2412,8 @@ static int nvhdmi_7x_8ch_build_controls(struct hda_codec *codec) return err; /* add channel maps */ - err = snd_pcm_add_chmap_ctls(spec->pcm_rec[0].pcm, + info = get_pcm_rec(spec, 0); + err = snd_pcm_add_chmap_ctls(info->pcm, SNDRV_PCM_STREAM_PLAYBACK, snd_pcm_alt_chmaps, 8, 0, &chmap); if (err < 0) @@ -2395,6 +2468,7 @@ static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_cvt *per_cvt = get_cvt(spec, 0); int chans = substream->runtime->channels; int i, err; @@ -2402,11 +2476,11 @@ static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, substream); if (err < 0) return err; - snd_hda_codec_write(codec, spec->cvts[0].cvt_nid, 0, + snd_hda_codec_write(codec, per_cvt->cvt_nid, 0, AC_VERB_SET_CVT_CHAN_COUNT, chans - 1); /* FIXME: XXX */ for (i = 0; i < chans; i++) { - snd_hda_codec_write(codec, spec->cvts[0].cvt_nid, 0, + snd_hda_codec_write(codec, per_cvt->cvt_nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, (i << 4) | i); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f15c36bde540..6bf47f7326ad 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -34,7 +34,6 @@ #include "hda_codec.h" #include "hda_local.h" #include "hda_auto_parser.h" -#include "hda_beep.h" #include "hda_jack.h" #include "hda_generic.h" @@ -53,6 +52,20 @@ enum { ALC_INIT_GPIO3, }; +enum { + ALC_HEADSET_MODE_UNKNOWN, + ALC_HEADSET_MODE_UNPLUGGED, + ALC_HEADSET_MODE_HEADSET, + ALC_HEADSET_MODE_MIC, + ALC_HEADSET_MODE_HEADPHONE, +}; + +enum { + ALC_HEADSET_TYPE_UNKNOWN, + ALC_HEADSET_TYPE_CTIA, + ALC_HEADSET_TYPE_OMTP, +}; + struct alc_customize_define { unsigned int sku_cfg; unsigned char port_connectivity; @@ -86,6 +99,13 @@ struct alc_spec { int mute_led_polarity; hda_nid_t mute_led_nid; + unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */ + + hda_nid_t headset_mic_pin; + hda_nid_t headphone_mic_pin; + int current_headset_mode; + int current_headset_type; + /* hooks */ void (*init_hook)(struct hda_codec *codec); #ifdef CONFIG_PM @@ -805,17 +825,7 @@ static inline void alc_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } -static void alc_free(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - if (!spec) - return; - - snd_hda_gen_spec_free(&spec->gen); - snd_hda_detach_beep_device(codec); - kfree(spec); -} +#define alc_free snd_hda_gen_free #ifdef CONFIG_PM static void alc_power_eapd(struct hda_codec *codec) @@ -1401,6 +1411,7 @@ static int patch_alc880(struct hda_codec *codec) spec = codec->spec; spec->gen.need_dac_fix = 1; + spec->gen.beep_nid = 0x01; snd_hda_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl, alc880_fixups); @@ -1411,12 +1422,8 @@ static int patch_alc880(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) { - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) - goto error; + if (!spec->gen.no_analog) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - } codec->patch_ops = alc_patch_ops; codec->patch_ops.unsol_event = alc880_unsol_event; @@ -1455,6 +1462,7 @@ enum { ALC260_FIXUP_HP_B1900, ALC260_FIXUP_KN1, ALC260_FIXUP_FSC_S7020, + ALC260_FIXUP_FSC_S7020_JWSE, }; static void alc260_gpio1_automute(struct hda_codec *codec) @@ -1516,14 +1524,17 @@ static void alc260_fixup_fsc_s7020(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - spec->gen.add_out_jack_modes = 1; - break; - case HDA_FIXUP_ACT_PROBE: + if (action == HDA_FIXUP_ACT_PROBE) spec->init_amp = ALC_INIT_NONE; - break; +} + +static void alc260_fixup_fsc_s7020_jwse(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.add_jack_modes = 1; + spec->gen.hp_mic = 1; } } @@ -1586,6 +1597,12 @@ static const struct hda_fixup alc260_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc260_fixup_fsc_s7020, }, + [ALC260_FIXUP_FSC_S7020_JWSE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc260_fixup_fsc_s7020_jwse, + .chained = true, + .chain_id = ALC260_FIXUP_FSC_S7020, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -1602,6 +1619,14 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { {} }; +static const struct hda_model_fixup alc260_fixup_models[] = { + {.id = ALC260_FIXUP_GPIO1, .name = "gpio1"}, + {.id = ALC260_FIXUP_COEF, .name = "coef"}, + {.id = ALC260_FIXUP_FSC_S7020, .name = "fujitsu"}, + {.id = ALC260_FIXUP_FSC_S7020_JWSE, .name = "fujitsu-jwse"}, + {} +}; + /* */ static int patch_alc260(struct hda_codec *codec) @@ -1619,8 +1644,10 @@ static int patch_alc260(struct hda_codec *codec) * it's almost harmless. */ spec->gen.prefer_hp_amp = 1; + spec->gen.beep_nid = 0x01; - snd_hda_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); + snd_hda_pick_fixup(codec, alc260_fixup_models, alc260_fixup_tbl, + alc260_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); /* automatic parse from the BIOS config */ @@ -1628,12 +1655,8 @@ static int patch_alc260(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) { - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) - goto error; + if (!spec->gen.no_analog) set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); - } codec->patch_ops = alc_patch_ops; spec->shutup = alc_eapd_shutup; @@ -2132,17 +2155,16 @@ static int patch_alc882(struct hda_codec *codec) alc_auto_parse_customize_define(codec); + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x01; + /* automatic parse from the BIOS config */ err = alc882_parse_auto_config(codec); if (err < 0) goto error; - if (!spec->gen.no_analog && has_cdefine_beep(codec)) { - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) - goto error; + if (!spec->gen.no_analog && spec->gen.beep_nid) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - } codec->patch_ops = alc_patch_ops; @@ -2295,17 +2317,16 @@ static int patch_alc262(struct hda_codec *codec) alc_auto_parse_customize_define(codec); + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x01; + /* automatic parse from the BIOS config */ err = alc262_parse_auto_config(codec); if (err < 0) goto error; - if (!spec->gen.no_analog && has_cdefine_beep(codec)) { - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) - goto error; + if (!spec->gen.no_analog && spec->gen.beep_nid) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - } codec->patch_ops = alc_patch_ops; spec->shutup = alc_eapd_shutup; @@ -2386,16 +2407,7 @@ static const struct snd_pci_quirk alc268_fixup_tbl[] = { static int alc268_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc268_ssids[] = { 0x15, 0x1b, 0x14, 0 }; - struct alc_spec *spec = codec->spec; - int err = alc_parse_auto_config(codec, NULL, alc268_ssids); - if (err > 0) { - if (!spec->gen.no_analog && - spec->gen.autocfg.speaker_pins[0] != 0x1d) { - add_mixer(spec, alc268_beep_mixer); - snd_hda_add_verbs(codec, alc268_beep_init_verbs); - } - } - return err; + return alc_parse_auto_config(codec, NULL, alc268_ssids); } /* @@ -2403,7 +2415,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) static int patch_alc268(struct hda_codec *codec) { struct alc_spec *spec; - int i, has_beep, err; + int err; /* ALC268 has no aa-loopback mixer */ err = alc_alloc_spec(codec, 0); @@ -2411,6 +2423,7 @@ static int patch_alc268(struct hda_codec *codec) return err; spec = codec->spec; + spec->gen.beep_nid = 0x01; snd_hda_pick_fixup(codec, alc268_fixup_models, alc268_fixup_tbl, alc268_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -2420,18 +2433,10 @@ static int patch_alc268(struct hda_codec *codec) if (err < 0) goto error; - has_beep = 0; - for (i = 0; i < spec->num_mixers; i++) { - if (spec->mixers[i] == alc268_beep_mixer) { - has_beep = 1; - break; - } - } - - if (has_beep) { - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) - goto error; + if (err > 0 && !spec->gen.no_analog && + spec->gen.autocfg.speaker_pins[0] != 0x1d) { + add_mixer(spec, alc268_beep_mixer); + snd_hda_add_verbs(codec, alc268_beep_init_verbs); if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) /* override the amp caps for beep generator */ snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, @@ -2515,6 +2520,7 @@ enum { ALC269_TYPE_ALC280, ALC269_TYPE_ALC282, ALC269_TYPE_ALC284, + ALC269_TYPE_ALC286, }; /* @@ -2538,6 +2544,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC269VB: case ALC269_TYPE_ALC269VD: case ALC269_TYPE_ALC282: + case ALC269_TYPE_ALC286: ssids = alc269_ssids; break; default: @@ -2631,7 +2638,8 @@ static void alc271_fixup_dmic(struct hda_codec *codec, }; unsigned int cfg; - if (strcmp(codec->chip_name, "ALC271X")) + if (strcmp(codec->chip_name, "ALC271X") && + strcmp(codec->chip_name, "ALC269VB")) return; cfg = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED) @@ -2693,6 +2701,34 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->gen.automute_hook = alc269_quanta_automute; } +static void alc269_x101_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + msleep(200); + snd_hda_gen_hp_automute(codec, jack); + + vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0; + msleep(100); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); + msleep(500); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + spec->gen.hp_automute_hook = alc269_x101_hp_automute_hook; + } +} + + /* update mute-LED according to the speaker mute state via mic VREF pin */ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) { @@ -2757,6 +2793,356 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, } } +/* turn on/off mute LED per vmaster hook */ +static void alc269_fixup_hp_gpio_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct alc_spec *spec = codec->spec; + unsigned int oldval = spec->gpio_led; + + if (enabled) + spec->gpio_led &= ~0x08; + else + spec->gpio_led |= 0x08; + if (spec->gpio_led != oldval) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); +} + +/* turn on/off mic-mute LED per capture hook */ +static void alc269_fixup_hp_gpio_mic_mute_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + struct alc_spec *spec = codec->spec; + unsigned int oldval = spec->gpio_led; + + if (!ucontrol) + return; + + if (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]) + spec->gpio_led &= ~0x10; + else + spec->gpio_led |= 0x10; + if (spec->gpio_led != oldval) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); +} + +static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x18 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 }, + {} + }; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook; + spec->gen.cap_sync_hook = alc269_fixup_hp_gpio_mic_mute_hook; + spec->gpio_led = 0; + snd_hda_add_verbs(codec, gpio_init); + } +} + +static void alc_headset_mode_unplugged(struct hda_codec *codec) +{ + int val; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x1b, 0x0c0b); + alc_write_coef_idx(codec, 0x45, 0xc429); + val = alc_read_coef_idx(codec, 0x35); + alc_write_coef_idx(codec, 0x35, val & 0xbfff); + alc_write_coef_idx(codec, 0x06, 0x2104); + alc_write_coef_idx(codec, 0x1a, 0x0001); + alc_write_coef_idx(codec, 0x26, 0x0004); + alc_write_coef_idx(codec, 0x32, 0x42a3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x76, 0x000e); + alc_write_coef_idx(codec, 0x6c, 0x2400); + alc_write_coef_idx(codec, 0x18, 0x7308); + alc_write_coef_idx(codec, 0x6b, 0xc429); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d40); + alc_write_coef_idx(codec, 0xb7, 0x802b); + break; + } + snd_printdd("Headset jack set to unplugged mode.\n"); +} + + +static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, + hda_nid_t mic_pin) +{ + int val; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xc429); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + val = alc_read_coef_idx(codec, 0x35); + alc_write_coef_idx(codec, 0x35, val | 1<<14); + alc_write_coef_idx(codec, 0x06, 0x2100); + alc_write_coef_idx(codec, 0x1a, 0x0021); + alc_write_coef_idx(codec, 0x26, 0x008c); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; + case 0x10ec0292: + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_write_coef_idx(codec, 0x19, 0xa208); + alc_write_coef_idx(codec, 0x2e, 0xacf0); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_write_coef_idx(codec, 0xb7, 0x802b); + alc_write_coef_idx(codec, 0xb5, 0x1040); + val = alc_read_coef_idx(codec, 0xc3); + alc_write_coef_idx(codec, 0xc3, val | 1<<12); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; + } + snd_printdd("Headset jack set to mic-in mode.\n"); +} + +static void alc_headset_mode_default(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x06, 0x2100); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x76, 0x000e); + alc_write_coef_idx(codec, 0x6c, 0x2400); + alc_write_coef_idx(codec, 0x6b, 0xc429); + alc_write_coef_idx(codec, 0x18, 0x7308); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0041); + alc_write_coef_idx(codec, 0x15, 0x0d40); + alc_write_coef_idx(codec, 0xb7, 0x802b); + break; + } + snd_printdd("Headset jack set to headphone (default) mode.\n"); +} + +/* Iphone type */ +static void alc_headset_mode_ctia(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xd429); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xd429); + alc_write_coef_idx(codec, 0x76, 0x0008); + alc_write_coef_idx(codec, 0x18, 0x7388); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d60); + alc_write_coef_idx(codec, 0xc3, 0x0000); + break; + } + snd_printdd("Headset jack set to iPhone-style headset mode.\n"); +} + +/* Nokia type */ +static void alc_headset_mode_omtp(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xe429); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xe429); + alc_write_coef_idx(codec, 0x76, 0x0008); + alc_write_coef_idx(codec, 0x18, 0x7388); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d50); + alc_write_coef_idx(codec, 0xc3, 0x0000); + break; + } + snd_printdd("Headset jack set to Nokia-style headset mode.\n"); +} + +static void alc_determine_headset_type(struct hda_codec *codec) +{ + int val; + bool is_ctia = false; + struct alc_spec *spec = codec->spec; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xd029); + msleep(300); + val = alc_read_coef_idx(codec, 0x46); + is_ctia = (val & 0x0070) == 0x0070; + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xd429); + msleep(300); + val = alc_read_coef_idx(codec, 0x6c); + is_ctia = (val & 0x001c) == 0x001c; + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); + alc_write_coef_idx(codec, 0xb7, 0x802b); + alc_write_coef_idx(codec, 0x15, 0x0d60); + alc_write_coef_idx(codec, 0xc3, 0x0c00); + msleep(300); + val = alc_read_coef_idx(codec, 0xbe); + is_ctia = (val & 0x1c02) == 0x1c02; + break; + } + + snd_printdd("Headset jack detected iPhone-style headset: %s\n", + is_ctia ? "yes" : "no"); + spec->current_headset_type = is_ctia ? ALC_HEADSET_TYPE_CTIA : ALC_HEADSET_TYPE_OMTP; +} + +static void alc_update_headset_mode(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + + int new_headset_mode; + + if (!snd_hda_jack_detect(codec, hp_pin)) + new_headset_mode = ALC_HEADSET_MODE_UNPLUGGED; + else if (mux_pin == spec->headset_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_HEADSET; + else if (mux_pin == spec->headphone_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_MIC; + else + new_headset_mode = ALC_HEADSET_MODE_HEADPHONE; + + if (new_headset_mode == spec->current_headset_mode) + return; + + switch (new_headset_mode) { + case ALC_HEADSET_MODE_UNPLUGGED: + alc_headset_mode_unplugged(codec); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADSET: + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; + break; + case ALC_HEADSET_MODE_MIC: + alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADPHONE: + alc_headset_mode_default(codec); + spec->gen.hp_jack_present = true; + break; + } + if (new_headset_mode != ALC_HEADSET_MODE_MIC) { + snd_hda_set_pin_ctl_cache(codec, hp_pin, + AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + if (spec->headphone_mic_pin) + snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, + PIN_VREFHIZ); + } + spec->current_headset_mode = new_headset_mode; + + snd_hda_gen_update_outputs(codec); +} + +static void alc_update_headset_mode_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + alc_update_headset_mode(codec); +} + +static void alc_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + spec->current_headset_type = ALC_HEADSET_MODE_UNKNOWN; + snd_hda_gen_hp_automute(codec, jack); +} + +static void alc_probe_headset_mode(struct hda_codec *codec) +{ + int i; + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + + /* Find mic pins */ + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].is_headset_mic && !spec->headset_mic_pin) + spec->headset_mic_pin = cfg->inputs[i].pin; + if (cfg->inputs[i].is_headphone_mic && !spec->headphone_mic_pin) + spec->headphone_mic_pin = cfg->inputs[i].pin; + } + + spec->gen.cap_sync_hook = alc_update_headset_mode_hook; + spec->gen.automute_hook = alc_update_headset_mode; + spec->gen.hp_automute_hook = alc_update_headset_jack_cb; +} + +static void alc_fixup_headset_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC | HDA_PINCFG_HEADPHONE_MIC; + break; + case HDA_FIXUP_ACT_PROBE: + alc_probe_headset_mode(codec); + break; + case HDA_FIXUP_ACT_INIT: + spec->current_headset_mode = 0; + alc_update_headset_mode(codec); + break; + } +} + +static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + } + else + alc_fixup_headset_mode(codec, fix, action); +} + +static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + int val; + alc_write_coef_idx(codec, 0xc4, 0x8000); + val = alc_read_coef_idx(codec, 0xc2); + alc_write_coef_idx(codec, 0xc2, val & 0xfe); + snd_hda_set_pin_ctl_cache(codec, 0x18, 0); + } + alc_fixup_headset_mode(codec, fix, action); +} + static void alc271_hp_gate_mic_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -2772,6 +3158,38 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + int i; + + /* The mic boosts on level 2 and 3 are too noisy + on the internal mic input. + Therefore limit the boost to 0 or 1. */ + + if (action != HDA_FIXUP_ACT_PROBE) + return; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + unsigned int defcfg; + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + defcfg = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT) + continue; + + snd_hda_override_amp_caps(codec, nid, HDA_INPUT, + (0x00 << AC_AMPCAP_OFFSET_SHIFT) | + (0x01 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x2f << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -2792,11 +3210,21 @@ enum { ALC269_FIXUP_HP_MUTE_LED, ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC269_FIXUP_HP_MUTE_LED_MIC2, + ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, + ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_HEADSET_MODE, + ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, + ALC269_FIXUP_ASUS_X101_FUNC, + ALC269_FIXUP_ASUS_X101_VERB, + ALC269_FIXUP_ASUS_X101, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, + ALC269_FIXUP_ACER_AC700, + ALC269_FIXUP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -2931,6 +3359,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic2, }, + [ALC269_FIXUP_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_gpio_led, + }, [ALC269_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, @@ -2949,6 +3381,59 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, }, + [ALC269_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, + [ALC269_FIXUP_DELL2_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x16, 0x21014020 }, /* dock line out */ + { 0x19, 0x21a19030 }, /* dock mic */ + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC269_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode, + }, + [ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_no_hp_mic, + }, + [ALC269_FIXUP_ASUS_X101_FUNC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_x101_headset_mic, + }, + [ALC269_FIXUP_ASUS_X101_VERB] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0310}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_ASUS_X101_FUNC + }, + [ALC269_FIXUP_ASUS_X101] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x04a1182c }, /* Headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_ASUS_X101_VERB + }, [ALC271_FIXUP_AMIC_MIC2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -2965,29 +3450,72 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_AMIC_MIC2, }, + [ALC269_FIXUP_ACER_AC700] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x99a3092f }, /* int-mic */ + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x03a11c20 }, /* mic */ + { 0x1e, 0x0346101e }, /* SPDIF1 */ + { 0x21, 0x0321101f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC271_FIXUP_DMIC, + }, + [ALC269_FIXUP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + }, }; 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(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c7, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ec, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ed, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ee, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f3, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), + SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), + SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_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, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), @@ -3063,6 +3591,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, + {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {} }; @@ -3131,6 +3660,9 @@ static int patch_alc269(struct hda_codec *codec) alc_auto_parse_customize_define(codec); + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x01; + switch (codec->vendor_id) { case 0x10ec0269: spec->codec_variant = ALC269_TYPE_ALC269VA; @@ -3172,6 +3704,9 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0292: spec->codec_variant = ALC269_TYPE_ALC284; break; + case 0x10ec0286: + spec->codec_variant = ALC269_TYPE_ALC286; + break; } /* automatic parse from the BIOS config */ @@ -3179,12 +3714,8 @@ static int patch_alc269(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog && has_cdefine_beep(codec)) { - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) - goto error; + if (!spec->gen.no_analog && spec->gen.beep_nid) set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); - } codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM @@ -3292,6 +3823,7 @@ static int patch_alc861(struct hda_codec *codec) return err; spec = codec->spec; + spec->gen.beep_nid = 0x23; snd_hda_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -3301,12 +3833,8 @@ static int patch_alc861(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) { - err = snd_hda_attach_beep_device(codec, 0x23); - if (err < 0) - goto error; + if (!spec->gen.no_analog) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); - } codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM @@ -3387,6 +3915,7 @@ static int patch_alc861vd(struct hda_codec *codec) return err; spec = codec->spec; + spec->gen.beep_nid = 0x23; snd_hda_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -3396,12 +3925,8 @@ static int patch_alc861vd(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) { - err = snd_hda_attach_beep_device(codec, 0x23); - if (err < 0) - goto error; + if (!spec->gen.no_analog) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - } codec->patch_ops = alc_patch_ops; @@ -3480,6 +4005,8 @@ enum { ALC662_FIXUP_NO_JACK_DETECT, ALC662_FIXUP_ZOTAC_Z68, ALC662_FIXUP_INV_DMIC, + ALC668_FIXUP_DELL_MIC_NO_PRESENCE, + ALC668_FIXUP_HEADSET_MODE, }; static const struct hda_fixup alc662_fixups[] = { @@ -3640,6 +4167,20 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { 0x1b, 0x03a1113c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE + }, + [ALC668_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc668, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -3648,6 +4189,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { 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(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), @@ -3784,10 +4327,14 @@ static int patch_alc662(struct hda_codec *codec) alc_auto_parse_customize_define(codec); + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x01; + if ((alc_get_coef0(codec) & (1 << 14)) && codec->bus->pci->subsystem_vendor == 0x1025 && spec->cdefine.platform_type == 1) { - if (alc_codec_rename(codec, "ALC272X") < 0) + err = alc_codec_rename(codec, "ALC272X"); + if (err < 0) goto error; } @@ -3796,10 +4343,7 @@ static int patch_alc662(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog && has_cdefine_beep(codec)) { - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) - goto error; + if (!spec->gen.no_analog && spec->gen.beep_nid) { switch (codec->vendor_id) { case 0x10ec0662: set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); @@ -3878,6 +4422,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 }, { .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 }, { .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 }, + { .id = 0x10ec0286, .name = "ALC286", .patch = patch_alc269 }, { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 }, { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dafe04ae8c72..1d9d6427e0bf 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -211,7 +211,6 @@ struct sigmatel_spec { /* beep widgets */ hda_nid_t anabeep_nid; - hda_nid_t digbeep_nid; /* SPDIF-out mux */ const char * const *spdif_labels; @@ -3529,8 +3528,12 @@ static int stac_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; int err; + int flags = 0; - err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + if (spec->headset_jack) + flags |= HDA_PINCFG_HEADSET_MIC; + + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, flags); if (err < 0) return err; @@ -3560,16 +3563,13 @@ static int stac_parse_auto_config(struct hda_codec *codec) /* setup digital beep controls and input device */ #ifdef CONFIG_SND_HDA_INPUT_BEEP - if (spec->digbeep_nid > 0) { - hda_nid_t nid = spec->digbeep_nid; + if (spec->gen.beep_nid) { + hda_nid_t nid = spec->gen.beep_nid; unsigned int caps; err = stac_auto_create_beep_ctls(codec, nid); if (err < 0) return err; - err = snd_hda_attach_beep_device(codec, nid); - if (err < 0) - return err; if (codec->beep) { /* IDT/STAC codecs have linear beep tone parameter */ codec->beep->linear_tone = spec->linear_tone_beep; @@ -3657,17 +3657,7 @@ static void stac_shutup(struct hda_codec *codec) ~spec->eapd_mask); } -static void stac_free(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - if (!spec) - return; - - snd_hda_gen_spec_free(&spec->gen); - kfree(spec); - snd_hda_detach_beep_device(codec); -} +#define stac_free snd_hda_gen_free #ifdef CONFIG_PROC_FS static void stac92hd_proc_hook(struct snd_info_buffer *buffer, @@ -3797,6 +3787,7 @@ static int patch_stac9200(struct hda_codec *codec) spec->gen.own_eapd_ctl = 1; codec->patch_ops = stac_patch_ops; + codec->power_filter = snd_hda_codec_eapd_power_filter; snd_hda_add_verbs(codec, stac9200_eapd_init); @@ -3884,7 +3875,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) spec->aloopback_mask = 0x01; spec->aloopback_shift = 8; - spec->digbeep_nid = 0x1c; + spec->gen.beep_nid = 0x1c; /* digital beep */ /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; @@ -3968,7 +3959,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->gen.power_down_unused = 1; spec->gen.mixer_nid = 0x1b; - spec->digbeep_nid = 0x21; + spec->gen.beep_nid = 0x21; /* digital beep */ spec->pwr_nids = stac92hd83xxx_pwr_nids; spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->default_polarity = -1; /* no default cfg */ @@ -4016,7 +4007,7 @@ static int patch_stac92hd95(struct hda_codec *codec) spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; - spec->digbeep_nid = 0x19; + spec->gen.beep_nid = 0x19; /* digital beep */ spec->pwr_nids = stac92hd95_pwr_nids; spec->num_pwrs = ARRAY_SIZE(stac92hd95_pwr_nids); spec->default_polarity = -1; /* no default cfg */ @@ -4091,7 +4082,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->aloopback_shift = 0; spec->powerdown_adcs = 1; - spec->digbeep_nid = 0x26; + spec->gen.beep_nid = 0x26; /* digital beep */ spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; @@ -4173,7 +4164,7 @@ static int patch_stac927x(struct hda_codec *codec) spec->have_spdif_mux = 1; spec->spdif_labels = stac927x_spdif_labels; - spec->digbeep_nid = 0x23; + spec->gen.beep_nid = 0x23; /* digital beep */ /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = 0x01; @@ -4232,7 +4223,7 @@ static int patch_stac9205(struct hda_codec *codec) spec->gen.own_eapd_ctl = 1; spec->have_spdif_mux = 1; - spec->digbeep_nid = 0x23; + spec->gen.beep_nid = 0x23; /* digital beep */ snd_hda_add_verbs(codec, stac9205_core_init); spec->aloopback_ctl = &stac9205_loopback; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c35338a8771d..e0dadcf2030d 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -626,11 +626,31 @@ static void via_set_jack_unsol_events(struct hda_codec *codec) } } +static const struct badness_table via_main_out_badness = { + .no_primary_dac = 0x10000, + .no_dac = 0x4000, + .shared_primary = 0x10000, + .shared_surr = 0x20, + .shared_clfe = 0x20, + .shared_surr_main = 0x20, +}; +static const struct badness_table via_extra_out_badness = { + .no_primary_dac = 0x4000, + .no_dac = 0x4000, + .shared_primary = 0x12, + .shared_surr = 0x20, + .shared_clfe = 0x20, + .shared_surr_main = 0x10, +}; + static int via_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; int err; + spec->gen.main_out_badness = &via_main_out_badness; + spec->gen.extra_out_badness = &via_extra_out_badness; + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); if (err < 0) return err; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 223c3d9cc69e..9ea05e956474 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -969,6 +969,7 @@ static int snd_hdspm_create_pcm(struct snd_card *card, struct hdspm *hdspm); static inline void snd_hdspm_initialize_midi_flush(struct hdspm *hdspm); +static inline int hdspm_get_pll_freq(struct hdspm *hdspm); static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm); static int hdspm_autosync_ref(struct hdspm *hdspm); static int snd_hdspm_set_defaults(struct hdspm *hdspm); @@ -1075,6 +1076,20 @@ static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm) return ret; } +/* round arbitary sample rates to commonly known rates */ +static int hdspm_round_frequency(int rate) +{ + if (rate < 38050) + return 32000; + if (rate < 46008) + return 44100; + else + return 48000; +} + +static int hdspm_tco_sync_check(struct hdspm *hdspm); +static int hdspm_sync_in_sync_check(struct hdspm *hdspm); + /* check for external sample rate */ static int hdspm_external_sample_rate(struct hdspm *hdspm) { @@ -1216,22 +1231,45 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) break; } - /* QS and DS rates normally can not be detected - * automatically by the card. Only exception is MADI - * in 96k frame mode. - * - * So if we read SS values (32 .. 48k), check for - * user-provided DS/QS bits in the control register - * and multiply the base frequency accordingly. - */ - if (rate <= 48000) { - if (hdspm->control_register & HDSPM_QuadSpeed) - rate *= 4; - else if (hdspm->control_register & - HDSPM_DoubleSpeed) - rate *= 2; + } /* endif HDSPM_madiLock */ + + /* check sample rate from TCO or SYNC_IN */ + { + bool is_valid_input = 0; + bool has_sync = 0; + + syncref = hdspm_autosync_ref(hdspm); + if (HDSPM_AUTOSYNC_FROM_TCO == syncref) { + is_valid_input = 1; + has_sync = (HDSPM_SYNC_CHECK_SYNC == + hdspm_tco_sync_check(hdspm)); + } else if (HDSPM_AUTOSYNC_FROM_SYNC_IN == syncref) { + is_valid_input = 1; + has_sync = (HDSPM_SYNC_CHECK_SYNC == + hdspm_sync_in_sync_check(hdspm)); + } + + if (is_valid_input && has_sync) { + rate = hdspm_round_frequency( + hdspm_get_pll_freq(hdspm)); } } + + /* QS and DS rates normally can not be detected + * automatically by the card. Only exception is MADI + * in 96k frame mode. + * + * So if we read SS values (32 .. 48k), check for + * user-provided DS/QS bits in the control register + * and multiply the base frequency accordingly. + */ + if (rate <= 48000) { + if (hdspm->control_register & HDSPM_QuadSpeed) + rate *= 4; + else if (hdspm->control_register & + HDSPM_DoubleSpeed) + rate *= 2; + } break; } @@ -1979,16 +2017,25 @@ static void hdspm_midi_tasklet(unsigned long arg) /* get the system sample rate which is set */ +static inline int hdspm_get_pll_freq(struct hdspm *hdspm) +{ + unsigned int period, rate; + + period = hdspm_read(hdspm, HDSPM_RD_PLL_FREQ); + rate = hdspm_calc_dds_value(hdspm, period); + + return rate; +} + /** * Calculate the real sample rate from the * current DDS value. **/ static int hdspm_get_system_sample_rate(struct hdspm *hdspm) { - unsigned int period, rate; + unsigned int rate; - period = hdspm_read(hdspm, HDSPM_RD_PLL_FREQ); - rate = hdspm_calc_dds_value(hdspm, period); + rate = hdspm_get_pll_freq(hdspm); if (rate > 207000) { /* Unreasonable high sample rate as seen on PCI MADI cards. */ @@ -2128,6 +2175,16 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx) return (status >> (idx*4)) & 0xF; } +#define ENUMERATED_CTL_INFO(info, texts) \ +{ \ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; \ + uinfo->count = 1; \ + uinfo->value.enumerated.items = ARRAY_SIZE(texts); \ + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) \ + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; \ + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); \ +} + #define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \ @@ -2143,14 +2200,7 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx) static int snd_hdspm_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 10; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts_freq[uinfo->value.enumerated.item]); + ENUMERATED_CTL_INFO(uinfo, texts_freq); return 0; } @@ -2316,15 +2366,7 @@ static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "Master", "AutoSync" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -2888,6 +2930,112 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol, return 0; } + + +#define HDSPM_TCO_VIDEO_INPUT_FORMAT(xname, xindex) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READ |\ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .info = snd_hdspm_info_tco_video_input_format, \ + .get = snd_hdspm_get_tco_video_input_format, \ +} + +static int snd_hdspm_info_tco_video_input_format(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = {"No video", "NTSC", "PAL"}; + ENUMERATED_CTL_INFO(uinfo, texts); + return 0; +} + +static int snd_hdspm_get_tco_video_input_format(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status; + int ret = 0; + + struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); + status = hdspm_read(hdspm, HDSPM_RD_TCO + 4); + switch (status & (HDSPM_TCO1_Video_Input_Format_NTSC | + HDSPM_TCO1_Video_Input_Format_PAL)) { + case HDSPM_TCO1_Video_Input_Format_NTSC: + /* ntsc */ + ret = 1; + break; + case HDSPM_TCO1_Video_Input_Format_PAL: + /* pal */ + ret = 2; + break; + default: + /* no video */ + ret = 0; + break; + } + ucontrol->value.enumerated.item[0] = ret; + return 0; +} + + + +#define HDSPM_TCO_LTC_FRAMES(xname, xindex) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READ |\ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .info = snd_hdspm_info_tco_ltc_frames, \ + .get = snd_hdspm_get_tco_ltc_frames, \ +} + +static int snd_hdspm_info_tco_ltc_frames(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps", + "30 fps"}; + ENUMERATED_CTL_INFO(uinfo, texts); + return 0; +} + +static int hdspm_tco_ltc_frames(struct hdspm *hdspm) +{ + u32 status; + int ret = 0; + + status = hdspm_read(hdspm, HDSPM_RD_TCO + 4); + if (status & HDSPM_TCO1_LTC_Input_valid) { + switch (status & (HDSPM_TCO1_LTC_Format_LSB | + HDSPM_TCO1_LTC_Format_MSB)) { + case 0: + /* 24 fps */ + ret = 1; + break; + case HDSPM_TCO1_LTC_Format_LSB: + /* 25 fps */ + ret = 2; + break; + case HDSPM_TCO1_LTC_Format_MSB: + /* 25 fps */ + ret = 3; + break; + default: + /* 30 fps */ + ret = 4; + break; + } + } + + return ret; +} + +static int snd_hdspm_get_tco_ltc_frames(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = hdspm_tco_ltc_frames(hdspm); + return 0; +} + #define HDSPM_TOGGLE_SETTING(xname, xindex) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ @@ -2974,17 +3122,7 @@ static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "optical", "coaxial" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3046,17 +3184,7 @@ static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "Single", "Double" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3129,17 +3257,7 @@ static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "Single", "Double", "Quad" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3215,17 +3333,7 @@ static int snd_hdspm_info_madi_speedmode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "Single", "Double", "Quad" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3445,19 +3553,30 @@ static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol, .get = snd_hdspm_get_sync_check \ } +#define HDSPM_TCO_LOCK_CHECK(xname, xindex) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .private_value = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .info = snd_hdspm_tco_info_lock_check, \ + .get = snd_hdspm_get_sync_check \ +} + + static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "No Lock", "Lock", "Sync", "N/A" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); + ENUMERATED_CTL_INFO(uinfo, texts); + return 0; +} + +static int snd_hdspm_tco_info_lock_check(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { "No Lock", "Lock" }; + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3590,6 +3709,14 @@ static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx) return 0; } +static int hdspm_tco_input_check(struct hdspm *hdspm, u32 mask) +{ + u32 status; + status = hdspm_read(hdspm, HDSPM_RD_TCO + 4); + + return (status & mask) ? 1 : 0; +} + static int hdspm_tco_sync_check(struct hdspm *hdspm) { @@ -3697,6 +3824,22 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, } + if (hdspm->tco) { + switch (kcontrol->private_value) { + case 11: + /* Check TCO for lock state of its current input */ + val = hdspm_tco_input_check(hdspm, HDSPM_TCO1_TCO_lock); + break; + case 12: + /* Check TCO for valid time code on LTC input. */ + val = hdspm_tco_input_check(hdspm, + HDSPM_TCO1_LTC_Input_valid); + break; + default: + break; + } + } + if (-1 == val) val = 3; @@ -3813,17 +3956,7 @@ static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "44.1 kHz", "48 kHz" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3869,17 +4002,7 @@ static int snd_hdspm_info_tco_pull(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "0", "+ 0.1 %", "- 0.1 %", "+ 4 %", "- 4 %" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 5; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3924,17 +4047,7 @@ static int snd_hdspm_info_tco_wck_conversion(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3981,17 +4094,7 @@ static int snd_hdspm_info_tco_frame_rate(struct snd_kcontrol *kcontrol, { static char *texts[] = { "24 fps", "25 fps", "29.97fps", "29.97 dfps", "30 fps", "30 dfps" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 6; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4037,17 +4140,7 @@ static int snd_hdspm_info_tco_sync_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "LTC", "Video", "WCK" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - + ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4145,6 +4238,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_madi[] = { HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 3), HDSPM_TOGGLE_SETTING("Line Out", HDSPM_LineOut), HDSPM_TOGGLE_SETTING("TX 64 channels mode", HDSPM_TX_64ch), + HDSPM_TOGGLE_SETTING("Disable 96K frames", HDSPM_SMUX), HDSPM_TOGGLE_SETTING("Clear Track Marker", HDSPM_clr_tms), HDSPM_TOGGLE_SETTING("Safe Mode", HDSPM_AutoInp), HDSPM_INPUT_SELECT("Input Select", 0), @@ -4272,7 +4366,11 @@ static struct snd_kcontrol_new snd_hdspm_controls_tco[] = { HDSPM_TCO_WCK_CONVERSION("TCO WCK Conversion", 0), HDSPM_TCO_FRAME_RATE("TCO Frame Rate", 0), HDSPM_TCO_SYNC_SOURCE("TCO Sync Source", 0), - HDSPM_TCO_WORD_TERM("TCO Word Term", 0) + HDSPM_TCO_WORD_TERM("TCO Word Term", 0), + HDSPM_TCO_LOCK_CHECK("TCO Input Check", 11), + HDSPM_TCO_LOCK_CHECK("TCO LTC Valid", 12), + HDSPM_TCO_LTC_FRAMES("TCO Detected Frame Rate", 0), + HDSPM_TCO_VIDEO_INPUT_FORMAT("Video Input Format", 0) }; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5da8ca7aee05..9e675c76436c 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -29,6 +29,10 @@ config SND_SOC_AC97_BUS config SND_SOC_DMAENGINE_PCM bool +config SND_SOC_GENERIC_DMAENGINE_PCM + bool + select SND_SOC_DMAENGINE_PCM + # All the supported SoCs source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 99f32f7c0692..197b6ae54c8d 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -5,6 +5,10 @@ ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),) snd-soc-core-objs += soc-dmaengine-pcm.o endif +ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) +snd-soc-core-objs += soc-generic-dmaengine-pcm.o +endif + obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 30184a4a147a..1d38fd0bc4e2 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -67,9 +67,10 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = { static void atmel_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pcm_dma_params *prtd; - prtd = snd_dmaengine_pcm_get_data(substream); + prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if (ssc_sr & prtd->mask->ssc_error) { if (snd_pcm_running(substream)) @@ -104,15 +105,13 @@ static bool filter(struct dma_chan *chan, void *slave) } static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, struct atmel_pcm_dma_params *prtd) { - struct atmel_pcm_dma_params *prtd; struct ssc_device *ssc; struct dma_chan *dma_chan; struct dma_slave_config slave_config; int ret; - prtd = snd_dmaengine_pcm_get_data(substream); ssc = prtd->ssc; ret = snd_hwparams_to_dma_slave_config(substream, params, @@ -130,8 +129,6 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, slave_config.src_maxburst = 1; } - slave_config.device_fc = false; - dma_chan = snd_dmaengine_pcm_get_chan(substream); if (dmaengine_slave_config(dma_chan, &slave_config)) { pr_err("atmel-pcm: failed to configure dma channel\n"); @@ -158,15 +155,13 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, if (ssc->pdev) sdata = ssc->pdev->dev.platform_data; - ret = snd_dmaengine_pcm_open(substream, filter, sdata); + ret = snd_dmaengine_pcm_open_request_chan(substream, filter, sdata); if (ret) { pr_err("atmel-pcm: dmaengine pcm open failed\n"); return -EINVAL; } - snd_dmaengine_pcm_set_data(substream, prtd); - - ret = atmel_pcm_configure_dma(substream, params); + ret = atmel_pcm_configure_dma(substream, params, prtd); if (ret) { pr_err("atmel-pcm: failed to configure dmai\n"); goto err; @@ -176,15 +171,16 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, return 0; err: - snd_dmaengine_pcm_close(substream); + snd_dmaengine_pcm_close_release_chan(substream); return ret; } static int atmel_pcm_dma_prepare(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pcm_dma_params *prtd; - prtd = snd_dmaengine_pcm_get_data(substream); + prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ssc_writex(prtd->ssc->regs, SSC_IER, prtd->mask->ssc_error); ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_enable); @@ -199,16 +195,9 @@ static int atmel_pcm_open(struct snd_pcm_substream *substream) return 0; } -static int atmel_pcm_close(struct snd_pcm_substream *substream) -{ - snd_dmaengine_pcm_close(substream); - - return 0; -} - static struct snd_pcm_ops atmel_pcm_ops = { .open = atmel_pcm_open, - .close = atmel_pcm_close, + .close = snd_dmaengine_pcm_close_release_chan, .ioctl = snd_pcm_lib_ioctl, .hw_params = atmel_pcm_hw_params, .prepare = atmel_pcm_dma_prepare, diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e13580d6c476..f3fdfa07fcb9 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -533,6 +533,49 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + /* + * DSP/PCM Mode A format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * Data is transferred on first BCLK after LRC pulse rising + * edge.If stereo, the right channel data is contiguous with + * the left channel data. + */ + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_PIN); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + default: printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", ssc_p->daifmt); @@ -707,13 +750,18 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .ops = &atmel_ssc_dai_ops, }; +static const struct snd_soc_component_driver atmel_ssc_component = { + .name = "atmel-ssc", +}; + static int asoc_ssc_init(struct device *dev) { struct platform_device *pdev = to_platform_device(dev); struct ssc_device *ssc = platform_get_drvdata(pdev); int ret; - ret = snd_soc_register_dai(dev, &atmel_ssc_dai); + ret = snd_soc_register_component(dev, &atmel_ssc_component, + &atmel_ssc_dai, 1); if (ret) { dev_err(dev, "Could not register DAI: %d\n", ret); goto err; @@ -732,7 +780,7 @@ static int asoc_ssc_init(struct device *dev) return 0; err_unregister_dai: - snd_soc_unregister_dai(dev); + snd_soc_unregister_component(dev); err: return ret; } @@ -747,7 +795,7 @@ static void asoc_ssc_exit(struct device *dev) else atmel_pcm_pdc_platform_unregister(dev); - snd_soc_unregister_dai(dev); + snd_soc_unregister_component(dev); } /** diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index ea7d9d157022..44b8dcecf571 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -223,6 +223,10 @@ static struct snd_soc_dai_driver au1xac97c_dai_driver = { .ops = &alchemy_ac97c_ops, }; +static const struct snd_soc_component_driver au1xac97c_component = { + .name = "au1xac97c", +}; + static int au1xac97c_drvprobe(struct platform_device *pdev) { int ret; @@ -268,7 +272,8 @@ static int au1xac97c_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); - ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver); + ret = snd_soc_register_component(&pdev->dev, &au1xac97c_component, + &au1xac97c_dai_driver, 1); if (ret) return ret; @@ -280,7 +285,7 @@ static int au1xac97c_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */ diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 072448afc219..b3f37f6edbcb 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -225,6 +225,10 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = { .ops = &au1xi2s_dai_ops, }; +static const struct snd_soc_component_driver au1xi2s_component = { + .name = "au1xi2s", +}; + static int au1xi2s_drvprobe(struct platform_device *pdev) { struct resource *iores, *dmares; @@ -260,14 +264,15 @@ static int au1xi2s_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); - return snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); + return snd_soc_register_component(&pdev->dev, &au1xi2s_component, + &au1xi2s_dai_driver, 1); } static int au1xi2s_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */ diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 6ba07e365967..8f1862aa7333 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -361,6 +361,10 @@ static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { .ops = &au1xpsc_ac97_dai_ops, }; +static const struct snd_soc_component_driver au1xpsc_ac97_component = { + .name = "au1xpsc-ac97", +}; + static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) { int ret; @@ -419,7 +423,8 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, wd); - ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); + ret = snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component, + &wd->dai_drv, 1); if (ret) return ret; @@ -431,7 +436,7 @@ static int au1xpsc_ac97_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); /* disable PSC completely */ au_writel(0, AC97_CFG(wd)); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 360b4e50d7c8..fe923a7bdc39 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -288,6 +288,10 @@ static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { .ops = &au1xpsc_i2s_dai_ops, }; +static const struct snd_soc_component_driver au1xpsc_i2s_component = { + .name = "au1xpsc-i2s", +}; + static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) { struct resource *iores, *dmares; @@ -350,14 +354,15 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, wd); - return snd_soc_register_dai(&pdev->dev, &wd->dai_drv); + return snd_soc_register_component(&pdev->dev, &au1xpsc_i2s_component, + &wd->dai_drv, 1); } static int au1xpsc_i2s_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); au_writel(0, I2S_CFG(wd)); au_sync(); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 8e41bcb020eb..490217325975 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -282,6 +282,10 @@ static struct snd_soc_dai_driver bfin_ac97_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; +static const struct snd_soc_component_driver bfin_ac97_component = { + .name = "bfin-ac97", +}; + static int asoc_bfin_ac97_probe(struct platform_device *pdev) { struct sport_device *sport_handle; @@ -331,7 +335,8 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) goto sport_config_err; } - ret = snd_soc_register_dai(&pdev->dev, &bfin_ac97_dai); + ret = snd_soc_register_component(&pdev->dev, &bfin_ac97_component, + &bfin_ac97_dai, 1); if (ret) { pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; @@ -356,7 +361,7 @@ static int asoc_bfin_ac97_remove(struct platform_device *pdev) { struct sport_device *sport_handle = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 168d88bccb41..dd0c2a4f83a3 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -245,6 +245,10 @@ static struct snd_soc_dai_driver bf5xx_i2s_dai = { .ops = &bf5xx_i2s_dai_ops, }; +static const struct snd_soc_component_driver bf5xx_i2s_component = { + .name = "bf5xx-i2s", +}; + static int bf5xx_i2s_probe(struct platform_device *pdev) { struct sport_device *sport_handle; @@ -257,7 +261,8 @@ static int bf5xx_i2s_probe(struct platform_device *pdev) return -ENODEV; /* register with the ASoC layers */ - ret = snd_soc_register_dai(&pdev->dev, &bf5xx_i2s_dai); + ret = snd_soc_register_component(&pdev->dev, &bf5xx_i2s_component, + &bf5xx_i2s_dai, 1); if (ret) { pr_err("Failed to register DAI: %d\n", ret); sport_done(sport_handle); @@ -273,7 +278,7 @@ static int bf5xx_i2s_remove(struct platform_device *pdev) pr_debug("%s enter\n", __func__); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); return 0; diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index c1e516ec53ad..69e9a3e935bd 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -249,6 +249,10 @@ static struct snd_soc_dai_driver bf5xx_tdm_dai = { .ops = &bf5xx_tdm_dai_ops, }; +static const struct snd_soc_component_driver bf5xx_tdm_component = { + .name = "bf5xx-tdm", +}; + static int bfin_tdm_probe(struct platform_device *pdev) { struct sport_device *sport_handle; @@ -282,7 +286,8 @@ static int bfin_tdm_probe(struct platform_device *pdev) goto sport_config_err; } - ret = snd_soc_register_dai(&pdev->dev, &bf5xx_tdm_dai); + ret = snd_soc_register_component(&pdev->dev, &bf5xx_tdm_component, + &bf5xx_tdm_dai, 1); if (ret) { pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; @@ -299,7 +304,7 @@ static int bfin_tdm_remove(struct platform_device *pdev) { struct sport_device *sport_handle = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); return 0; diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c index 8f337972f438..c02405cc007d 100644 --- a/sound/soc/blackfin/bf6xx-i2s.c +++ b/sound/soc/blackfin/bf6xx-i2s.c @@ -186,6 +186,10 @@ static struct snd_soc_dai_driver bfin_i2s_dai = { .ops = &bfin_i2s_dai_ops, }; +static const struct snd_soc_component_driver bfin_i2s_component = { + .name = "bfin-i2s", +}; + static int bfin_i2s_probe(struct platform_device *pdev) { struct sport_device *sport; @@ -197,7 +201,8 @@ static int bfin_i2s_probe(struct platform_device *pdev) return -ENODEV; /* register with the ASoC layers */ - ret = snd_soc_register_dai(dev, &bfin_i2s_dai); + ret = snd_soc_register_component(dev, &bfin_i2s_component, + &bfin_i2s_dai, 1); if (ret) { dev_err(dev, "Failed to register DAI: %d\n", ret); sport_delete(sport); @@ -212,7 +217,7 @@ static int bfin_i2s_remove(struct platform_device *pdev) { struct sport_device *sport = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); sport_delete(sport); return 0; diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index 5db68cf7b281..c43fb214558a 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -27,7 +27,6 @@ #include <sound/soc.h> #include <asm/mach-types.h> #include <mach/hardware.h> -#include "ep93xx-pcm.h" static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 1738d28fb04f..7798fbd5e81d 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -23,7 +23,6 @@ #include <sound/soc.h> #include <linux/platform_data/dma-ep93xx.h> -#include "ep93xx-pcm.h" /* * Per channel (1-4) registers. @@ -101,14 +100,16 @@ struct ep93xx_ac97_info { /* currently ALSA only supports a single AC97 device */ static struct ep93xx_ac97_info *ep93xx_ac97_info; -static struct ep93xx_pcm_dma_params ep93xx_ac97_pcm_out = { +static struct ep93xx_dma_data ep93xx_ac97_pcm_out = { .name = "ac97-pcm-out", .dma_port = EP93XX_DMA_AAC1, + .direction = DMA_MEM_TO_DEV, }; -static struct ep93xx_pcm_dma_params ep93xx_ac97_pcm_in = { +static struct ep93xx_dma_data ep93xx_ac97_pcm_in = { .name = "ac97-pcm-in", .dma_port = EP93XX_DMA_AAC1, + .direction = DMA_DEV_TO_MEM, }; static inline unsigned ep93xx_ac97_read_reg(struct ep93xx_ac97_info *info, @@ -316,7 +317,7 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, static int ep93xx_ac97_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct ep93xx_pcm_dma_params *dma_data; + struct ep93xx_dma_data *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dma_data = &ep93xx_ac97_pcm_out; @@ -353,6 +354,10 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = { .ops = &ep93xx_ac97_dai_ops, }; +static const struct snd_soc_component_driver ep93xx_ac97_component = { + .name = "ep93xx-ac97", +}; + static int ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; @@ -390,7 +395,8 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) ep93xx_ac97_info = info; platform_set_drvdata(pdev, info); - ret = snd_soc_register_dai(&pdev->dev, &ep93xx_ac97_dai); + ret = snd_soc_register_component(&pdev->dev, &ep93xx_ac97_component, + &ep93xx_ac97_dai, 1); if (ret) goto fail; @@ -407,7 +413,7 @@ static int ep93xx_ac97_remove(struct platform_device *pdev) { struct ep93xx_ac97_info *info = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); /* disable the AC97 controller */ ep93xx_ac97_write_reg(info, AC97GCR, 0); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 323ed69b7975..5c1102e9e159 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -30,8 +30,6 @@ #include <mach/ep93xx-regs.h> #include <linux/platform_data/dma-ep93xx.h> -#include "ep93xx-pcm.h" - #define EP93XX_I2S_TXCLKCFG 0x00 #define EP93XX_I2S_RXCLKCFG 0x04 #define EP93XX_I2S_GLCTRL 0x0C @@ -62,18 +60,20 @@ struct ep93xx_i2s_info { struct clk *mclk; struct clk *sclk; struct clk *lrclk; - struct ep93xx_pcm_dma_params *dma_params; + struct ep93xx_dma_data *dma_data; void __iomem *regs; }; -struct ep93xx_pcm_dma_params ep93xx_i2s_dma_params[] = { +struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { [SNDRV_PCM_STREAM_PLAYBACK] = { .name = "i2s-pcm-out", - .dma_port = EP93XX_DMA_I2S1, + .port = EP93XX_DMA_I2S1, + .direction = DMA_MEM_TO_DEV, }, [SNDRV_PCM_STREAM_CAPTURE] = { .name = "i2s-pcm-in", - .dma_port = EP93XX_DMA_I2S1, + .port = EP93XX_DMA_I2S1, + .direction = DMA_DEV_TO_MEM, }, }; @@ -147,7 +147,7 @@ static int ep93xx_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; snd_soc_dai_set_dma_data(cpu_dai, substream, - &info->dma_params[substream->stream]); + &info->dma_data[substream->stream]); return 0; } @@ -366,6 +366,10 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = { .ops = &ep93xx_i2s_dai_ops, }; +static const struct snd_soc_component_driver ep93xx_i2s_component = { + .name = "ep93xx-i2s", +}; + static int ep93xx_i2s_probe(struct platform_device *pdev) { struct ep93xx_i2s_info *info; @@ -403,9 +407,10 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, info); - info->dma_params = ep93xx_i2s_dma_params; + info->dma_data = ep93xx_i2s_dma_data; - err = snd_soc_register_dai(&pdev->dev, &ep93xx_i2s_dai); + err = snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component, + &ep93xx_i2s_dai, 1); if (err) goto fail_put_lrclk; @@ -426,7 +431,7 @@ static int ep93xx_i2s_remove(struct platform_device *pdev) { struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); clk_put(info->sclk); diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index 72eb7a49e16a..488032690378 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -29,8 +29,6 @@ #include <mach/hardware.h> #include <mach/ep93xx-regs.h> -#include "ep93xx-pcm.h" - static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -68,40 +66,12 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) static int ep93xx_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct ep93xx_pcm_dma_params *dma_params; - struct ep93xx_dma_data *dma_data; - int ret; snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware); - dma_data = kmalloc(sizeof(*dma_data), GFP_KERNEL); - if (!dma_data) - return -ENOMEM; - - dma_params = snd_soc_dai_get_dma_data(cpu_dai, substream); - dma_data->port = dma_params->dma_port; - dma_data->name = dma_params->name; - dma_data->direction = snd_pcm_substream_to_dma_direction(substream); - - ret = snd_dmaengine_pcm_open(substream, ep93xx_pcm_dma_filter, dma_data); - if (ret) { - kfree(dma_data); - return ret; - } - - snd_dmaengine_pcm_set_data(substream, dma_data); - - return 0; -} - -static int ep93xx_pcm_close(struct snd_pcm_substream *substream) -{ - struct dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); - - snd_dmaengine_pcm_close(substream); - kfree(dma_data); - return 0; + return snd_dmaengine_pcm_open_request_chan(substream, + ep93xx_pcm_dma_filter, + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); } static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream, @@ -131,7 +101,7 @@ static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream, static struct snd_pcm_ops ep93xx_pcm_ops = { .open = ep93xx_pcm_open, - .close = ep93xx_pcm_close, + .close = snd_dmaengine_pcm_close_release_chan, .ioctl = snd_pcm_lib_ioctl, .hw_params = ep93xx_pcm_hw_params, .hw_free = ep93xx_pcm_hw_free, diff --git a/sound/soc/cirrus/ep93xx-pcm.h b/sound/soc/cirrus/ep93xx-pcm.h deleted file mode 100644 index 111e1121ecb8..000000000000 --- a/sound/soc/cirrus/ep93xx-pcm.h +++ /dev/null @@ -1,20 +0,0 @@ -/* - * sound/soc/ep93xx/ep93xx-pcm.h - EP93xx ALSA PCM interface - * - * Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.org> - * Copyright (C) 2006 Applied Data Systems - * - * 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 _EP93XX_SND_SOC_PCM_H -#define _EP93XX_SND_SOC_PCM_H - -struct ep93xx_pcm_dma_params { - char *name; - int dma_port; -}; - -#endif /* _EP93XX_SND_SOC_PCM_H */ diff --git a/sound/soc/cirrus/simone.c b/sound/soc/cirrus/simone.c index a397bb0d8179..4d094d00c34a 100644 --- a/sound/soc/cirrus/simone.c +++ b/sound/soc/cirrus/simone.c @@ -21,8 +21,6 @@ #include <asm/mach-types.h> #include <mach/hardware.h> -#include "ep93xx-pcm.h" - static struct snd_soc_dai_link simone_dai = { .name = "AC97", .stream_name = "AC97 HiFi", diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 9d77fe28dfcc..69041074f2c1 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -21,7 +21,6 @@ #include <mach/hardware.h> #include "../codecs/tlv320aic23.h" -#include "ep93xx-pcm.h" #define CODEC_CLOCK 5644800 diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b72561c615..2f45f00e31b0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC @@ -63,6 +64,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C @@ -203,6 +205,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_AK5386 + tristate + config SND_SOC_ALC5623 tristate config SND_SOC_ALC5632 @@ -320,11 +325,14 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS5086 + tristate + config SND_SOC_TLV320AIC23 tristate config SND_SOC_TLV320AIC26 - tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE + tristate depends on SPI config SND_SOC_TLV320AIC32X4 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3b8b41..b9e41c9a1f4c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +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-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o @@ -55,6 +56,7 @@ 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-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -137,6 +139,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o @@ -177,6 +180,7 @@ obj-$(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_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 068b3ae56a17..1aa10ddf3a61 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -133,6 +133,8 @@ struct adau1373 { #define ADAU1373_DAI_FORMAT_DSP 0x3 #define ADAU1373_BCLKDIV_SOURCE BIT(5) +#define ADAU1373_BCLKDIV_SR_MASK (0x07 << 2) +#define ADAU1373_BCLKDIV_BCLK_MASK 0x03 #define ADAU1373_BCLKDIV_32 0x03 #define ADAU1373_BCLKDIV_64 0x02 #define ADAU1373_BCLKDIV_128 0x01 @@ -937,7 +939,8 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, adau1373_dai->enable_src = (div != 0); snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), - ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); + ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, + (div << 2) | ADAU1373_BCLKDIV_64); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 6f6c335a5baa..c7cfdf957e4d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -55,6 +55,7 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int val = 0; int ret; @@ -77,9 +78,9 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) return -EINVAL; - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, - val); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, + val); if (ret < 0) return ret; @@ -91,11 +92,12 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - int val = 0; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + int ret, val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ val |= IEC958_AES0_CON_NOT_COPYRIGHT; - snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val); + regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(0), val); val = 0; @@ -132,11 +134,33 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val); + ret = regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val); + if (ret < 0) + return ret; + + /* enable transmitter */ + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, AK4104_TX_TXE); + if (ret < 0) + return ret; + + return 0; +} + +static int ak4104_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + + /* disable transmitter */ + return regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, 0); } static const struct snd_soc_dai_ops ak4101_dai_ops = { .hw_params = ak4104_hw_params, + .hw_free = ak4104_hw_free, .set_fmt = ak4104_set_dai_fmt, }; @@ -160,20 +184,17 @@ static int ak4104_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ak4104->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) - return ret; /* set power-up and non-reset bits */ - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); if (ret < 0) return ret; /* enable transmitter */ - ret = snd_soc_update_bits(codec, AK4104_REG_TX, - AK4104_TX_TXE, AK4104_TX_TXE); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, AK4104_TX_TXE); if (ret < 0) return ret; @@ -182,8 +203,10 @@ static int ak4104_probe(struct snd_soc_codec *codec) static int ak4104_remove(struct snd_soc_codec *codec) { - snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); return 0; } diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c new file mode 100644 index 000000000000..1f303983ae02 --- /dev/null +++ b/sound/soc/codecs/ak5386.c @@ -0,0 +1,152 @@ +/* + * ALSA SoC driver for + * Asahi Kasei AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC + * + * (c) 2013 Daniel Mack <zonque@gmail.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/slab.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/initval.h> + +struct ak5386_priv { + int reset_gpio; +}; + +static struct snd_soc_codec_driver soc_codec_ak5386; + +static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + format &= SND_SOC_DAIFMT_FORMAT_MASK; + if (format != SND_SOC_DAIFMT_LEFT_J && + format != SND_SOC_DAIFMT_I2S) { + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + return 0; +} + +static int ak5386_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 ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * From the datasheet: + * + * All external clocks (MCLK, SCLK and LRCK) must be present unless + * PDN pin = “L”. If these clocks are not provided, the AK5386 may + * draw excess current due to its use of internal dynamically + * refreshed logic. If the external clocks are not present, place + * the AK5386 in power-down mode (PDN pin = “L”). + */ + + if (gpio_is_valid(priv->reset_gpio)) + gpio_set_value(priv->reset_gpio, 1); + + return 0; +} + +static int ak5386_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->reset_gpio)) + gpio_set_value(priv->reset_gpio, 0); + + return 0; +} + +static const struct snd_soc_dai_ops ak5386_dai_ops = { + .set_fmt = ak5386_set_dai_fmt, + .hw_params = ak5386_hw_params, + .hw_free = ak5386_hw_free, +}; + +static struct snd_soc_dai_driver ak5386_dai = { + .name = "ak5386-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + }, + .ops = &ak5386_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id ak5386_dt_ids[] = { + { .compatible = "asahi-kasei,ak5386", }, + { } +}; +MODULE_DEVICE_TABLE(of, ak5386_dt_ids); +#endif + +static int ak5386_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct ak5386_priv *priv; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->reset_gpio = -EINVAL; + dev_set_drvdata(dev, priv); + + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) + priv->reset_gpio = of_get_named_gpio(dev->of_node, + "reset-gpio", 0); + + if (gpio_is_valid(priv->reset_gpio)) + if (devm_gpio_request_one(dev, priv->reset_gpio, + GPIOF_OUT_INIT_LOW, + "AK5386 Reset")) + priv->reset_gpio = -EINVAL; + + return snd_soc_register_codec(dev, &soc_codec_ak5386, + &ak5386_dai, 1); +} + +static int ak5386_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ak5386_driver = { + .probe = ak5386_probe, + .remove = ak5386_remove, + .driver = { + .name = "ak5386", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ak5386_dt_ids), + }, +}; + +module_platform_driver(ak5386_driver); + +MODULE_DESCRIPTION("ASoC driver for AK5386 ADC"); +MODULE_AUTHOR("Daniel Mack <zonque@gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e7d34711412c..389f23253831 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -10,6 +10,7 @@ * published by the Free Software Foundation. */ +#include <linux/delay.h> #include <linux/gcd.h> #include <linux/module.h> #include <linux/pm_runtime.h> @@ -65,6 +66,163 @@ #define arizona_aif_dbg(_dai, fmt, ...) \ dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +static int arizona_spk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + bool manual_ena = false; + int val; + + switch (arizona->type) { + case WM5102: + switch (arizona->rev) { + case 0: + break; + default: + manual_ena = true; + break; + } + default: + break; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (!priv->spk_ena && manual_ena) { + snd_soc_write(codec, 0x4f5, 0x25a); + priv->spk_ena_pending = true; + } + break; + case SND_SOC_DAPM_POST_PMU: + val = snd_soc_read(codec, ARIZONA_INTERRUPT_RAW_STATUS_3); + if (val & ARIZONA_SPK_SHUTDOWN_STS) { + dev_crit(arizona->dev, + "Speaker not enabled due to temperature\n"); + return -EBUSY; + } + + snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 1 << w->shift); + + if (priv->spk_ena_pending) { + msleep(75); + snd_soc_write(codec, 0x4f5, 0xda); + priv->spk_ena_pending = false; + priv->spk_ena++; + } + break; + case SND_SOC_DAPM_PRE_PMD: + if (manual_ena) { + priv->spk_ena--; + if (!priv->spk_ena) + snd_soc_write(codec, 0x4f5, 0x25a); + } + + snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 0); + break; + case SND_SOC_DAPM_POST_PMD: + if (manual_ena) { + if (!priv->spk_ena) + snd_soc_write(codec, 0x4f5, 0x0da); + } + break; + } + + return 0; +} + +static irqreturn_t arizona_thermal_warn(int irq, void *data) +{ + struct arizona *arizona = data; + unsigned int val; + int ret; + + ret = regmap_read(arizona->regmap, ARIZONA_INTERRUPT_RAW_STATUS_3, + &val); + if (ret != 0) { + dev_err(arizona->dev, "Failed to read thermal status: %d\n", + ret); + } else if (val & ARIZONA_SPK_SHUTDOWN_WARN_STS) { + dev_crit(arizona->dev, "Thermal warning\n"); + } + + return IRQ_HANDLED; +} + +static irqreturn_t arizona_thermal_shutdown(int irq, void *data) +{ + struct arizona *arizona = data; + unsigned int val; + int ret; + + ret = regmap_read(arizona->regmap, ARIZONA_INTERRUPT_RAW_STATUS_3, + &val); + if (ret != 0) { + dev_err(arizona->dev, "Failed to read thermal status: %d\n", + ret); + } else if (val & ARIZONA_SPK_SHUTDOWN_STS) { + dev_crit(arizona->dev, "Thermal shutdown\n"); + ret = regmap_update_bits(arizona->regmap, + ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4L_ENA | + ARIZONA_OUT4R_ENA, 0); + if (ret != 0) + dev_crit(arizona->dev, + "Failed to disable speaker outputs: %d\n", + ret); + } + + return IRQ_HANDLED; +} + +static const struct snd_soc_dapm_widget arizona_spkl = + SND_SOC_DAPM_PGA_E("OUT4L", SND_SOC_NOPM, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU); + +static const struct snd_soc_dapm_widget arizona_spkr = + SND_SOC_DAPM_PGA_E("OUT4R", SND_SOC_NOPM, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU); + +int arizona_init_spk(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int ret; + + ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkl, 1); + if (ret != 0) + return ret; + + ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1); + if (ret != 0) + return ret; + + ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN, + "Thermal warning", arizona_thermal_warn, + arizona); + if (ret != 0) + dev_err(arizona->dev, + "Failed to get thermal warning IRQ: %d\n", + ret); + + ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN, + "Thermal shutdown", arizona_thermal_shutdown, + arizona); + if (ret != 0) + dev_err(arizona->dev, + "Failed to get thermal shutdown IRQ: %d\n", + ret); + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_spk); + const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", "Tone Generator 1", @@ -274,6 +432,33 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values); const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); EXPORT_SYMBOL_GPL(arizona_mixer_tlv); +const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = { + "SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate", +}; +EXPORT_SYMBOL_GPL(arizona_rate_text); + +const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { + 0, 1, 2, 8, +}; +EXPORT_SYMBOL_GPL(arizona_rate_val); + + +const struct soc_enum arizona_isrc_fsl[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_2, + ARIZONA_ISRC1_FSL_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_2_CTRL_2, + ARIZONA_ISRC2_FSL_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_3_CTRL_2, + ARIZONA_ISRC3_FSL_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; +EXPORT_SYMBOL_GPL(arizona_isrc_fsl); + static const char *arizona_vol_ramp_text[] = { "0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB", "15ms/6dB", "30ms/6dB", @@ -332,9 +517,27 @@ const struct soc_enum arizona_ng_hold = 4, arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); +static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + if (ena) + val = ARIZONA_IN_VU; + else + val = 0; + + for (i = 0; i < priv->num_inputs; i++) + snd_soc_update_bits(codec, + ARIZONA_ADC_DIGITAL_VOLUME_1L + (i * 4), + ARIZONA_IN_VU, val); +} + int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); unsigned int reg; if (w->shift % 2) @@ -343,13 +546,29 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, reg = ARIZONA_ADC_DIGITAL_VOLUME_1R + ((w->shift / 2) * 8); switch (event) { + case SND_SOC_DAPM_PRE_PMU: + priv->in_pending++; + break; case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0); + + /* If this is the last input pending then allow VU */ + priv->in_pending--; + if (priv->in_pending == 0) { + msleep(1); + arizona_in_set_vu(w->codec, 1); + } break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, - ARIZONA_IN1L_MUTE); + snd_soc_update_bits(w->codec, reg, + ARIZONA_IN1L_MUTE | ARIZONA_IN_VU, + ARIZONA_IN1L_MUTE | ARIZONA_IN_VU); break; + case SND_SOC_DAPM_POST_PMD: + /* Disable volume updates if no inputs are enabled */ + reg = snd_soc_read(w->codec, ARIZONA_INPUT_ENABLES); + if (reg == 0) + arizona_in_set_vu(w->codec, 0); } return 0; @@ -360,6 +579,24 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + switch (event) { + case SND_SOC_DAPM_POST_PMU: + switch (w->shift) { + case ARIZONA_OUT1L_ENA_SHIFT: + case ARIZONA_OUT1R_ENA_SHIFT: + case ARIZONA_OUT2L_ENA_SHIFT: + case ARIZONA_OUT2R_ENA_SHIFT: + case ARIZONA_OUT3L_ENA_SHIFT: + case ARIZONA_OUT3R_ENA_SHIFT: + msleep(17); + break; + + default: + break; + } + break; + } + return 0; } EXPORT_SYMBOL_GPL(arizona_out_ev); @@ -502,27 +739,27 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, break; case 11289600: case 12288000: - val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_12MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 22579200: case 24576000: - val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_24MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 45158400: case 49152000: - val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_49MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 67737600: case 73728000: - val |= 4 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_73MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 90316800: case 98304000: - val |= 5 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_98MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 135475200: case 147456000: - val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_147MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 0: dev_dbg(arizona->dev, "%s cleared\n", name); @@ -816,7 +1053,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct arizona *arizona = priv->arizona; int base = dai->driver->base; const int *rates; - int i, ret; + int i, ret, val; int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; @@ -832,6 +1069,13 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, bclk_target *= chan_limit; } + /* Force stereo for I2S mode */ + val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); + if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + arizona_aif_dbg(dai, "Forcing stereo mode\n"); + bclk_target *= 2; + } + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { if (rates[i] >= bclk_target && rates[i] % params_rate(params) == 0) { @@ -988,6 +1232,16 @@ static struct { { 1000000, 13500000, 0, 1 }, }; +static struct { + unsigned int min; + unsigned int max; + u16 gain; +} fll_gains[] = { + { 0, 256000, 0 }, + { 256000, 1000000, 2 }, + { 1000000, 13500000, 4 }, +}; + struct arizona_fll_cfg { int n; int theta; @@ -995,6 +1249,7 @@ struct arizona_fll_cfg { int refdiv; int outdiv; int fratio; + int gain; }; static int arizona_calc_fll(struct arizona_fll *fll, @@ -1054,6 +1309,18 @@ static int arizona_calc_fll(struct arizona_fll *fll, return -EINVAL; } + for (i = 0; i < ARRAY_SIZE(fll_gains); i++) { + if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) { + cfg->gain = fll_gains[i].gain; + break; + } + } + if (i == ARRAY_SIZE(fll_gains)) { + arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n", + Fref); + return -EINVAL; + } + cfg->n = target / (ratio * Fref); if (target % (ratio * Fref)) { @@ -1081,13 +1348,15 @@ static int arizona_calc_fll(struct arizona_fll *fll, 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); + arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain); return 0; } static void arizona_apply_fll(struct arizona *arizona, unsigned int base, - struct arizona_fll_cfg *cfg, int source) + struct arizona_fll_cfg *cfg, int source, + bool sync) { regmap_update_bits(arizona->regmap, base + 3, ARIZONA_FLL1_THETA_MASK, cfg->theta); @@ -1102,87 +1371,84 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + if (sync) + regmap_update_bits(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + else + regmap_update_bits(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_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) +static bool arizona_is_enabled_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; - struct arizona_fll_cfg cfg, sync; - unsigned int reg, val; - int syncsrc; - bool ena; + unsigned int reg; int ret; - if (fll->fref == Fref && fll->fout == Fout) - return 0; - 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; - } + return reg & ARIZONA_FLL1_ENA; +} - if (source == syncsrc) - syncsrc = -1; +static void arizona_enable_fll(struct arizona_fll *fll, + struct arizona_fll_cfg *ref, + struct arizona_fll_cfg *sync) +{ + struct arizona *arizona = fll->arizona; + int ret; - if (syncsrc >= 0) { - ret = arizona_calc_fll(fll, &sync, Fref, Fout); - if (ret != 0) - return ret; + /* + * If we have both REFCLK and SYNCCLK then enable both, + * otherwise apply the SYNCCLK settings to REFCLK. + */ + if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, + false); + if (fll->sync_src >= 0) + arizona_apply_fll(arizona, fll->base + 0x10, sync, + fll->sync_src, true); + } else if (fll->sync_src >= 0) { + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + arizona_apply_fll(arizona, fll->base, sync, + fll->sync_src, false); - 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); - - fll->fref = Fref; - fll->fout = Fout; - - 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); + arizona_fll_err(fll, "No clocks provided\n"); + return; } - if (!ena) + /* + * Increase the bandwidth if we're not using a low frequency + * sync source. + */ + if (fll->sync_src >= 0 && fll->sync_freq > 100000) + regmap_update_bits(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, 0); + else + regmap_update_bits(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); + + if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); /* Clear any pending completions */ @@ -1190,7 +1456,8 @@ int arizona_set_fll(struct arizona_fll *fll, int source, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (syncsrc >= 0) + if (fll->ref_src >= 0 && fll->sync_src >= 0 && + fll->ref_src != fll->sync_src) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1199,10 +1466,88 @@ int arizona_set_fll(struct arizona_fll *fll, int source, msecs_to_jiffies(250)); if (ret == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); +} + +static void arizona_disable_fll(struct arizona_fll *fll) +{ + struct arizona *arizona = fll->arizona; + bool change; + + regmap_update_bits_check(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, 0, &change); + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); + + if (change) + pm_runtime_put_autosuspend(arizona->dev); +} + +int arizona_set_fll_refclk(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout) +{ + struct arizona_fll_cfg ref, sync; + int ret; - fll->fref = Fref; + if (fll->ref_src == source && fll->ref_freq == Fref) + return 0; + + if (fll->fout && Fref > 0) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); + if (ret != 0) + return ret; + + if (fll->sync_src >= 0) { + ret = arizona_calc_fll(fll, &sync, fll->sync_freq, + fll->fout); + if (ret != 0) + return ret; + } + } + + fll->ref_src = source; + fll->ref_freq = Fref; + + if (fll->fout && Fref > 0) { + arizona_enable_fll(fll, &ref, &sync); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_set_fll_refclk); + +int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout) +{ + struct arizona_fll_cfg ref, sync; + int ret; + + if (fll->sync_src == source && + fll->sync_freq == Fref && fll->fout == Fout) + return 0; + + if (Fout) { + if (fll->ref_src >= 0) { + ret = arizona_calc_fll(fll, &ref, fll->ref_freq, + Fout); + if (ret != 0) + return ret; + } + + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; + } + + fll->sync_src = source; + fll->sync_freq = Fref; fll->fout = Fout; + if (Fout) { + arizona_enable_fll(fll, &ref, &sync); + } else { + arizona_disable_fll(fll); + } + return 0; } EXPORT_SYMBOL_GPL(arizona_set_fll); @@ -1211,12 +1556,26 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, int ok_irq, struct arizona_fll *fll) { int ret; + unsigned int val; init_completion(&fll->ok); fll->id = id; fll->base = base; fll->arizona = arizona; + fll->sync_src = ARIZONA_FLL_SRC_NONE; + + /* Configure default refclk to 32kHz if we have one */ + 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: + fll->ref_src = val & ARIZONA_CLK_32K_SRC_MASK; + break; + default: + fll->ref_src = ARIZONA_FLL_SRC_NONE; + } + fll->ref_freq = 32768; snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id); snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 13dd2916b721..af39f1006427 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -32,6 +32,7 @@ #define ARIZONA_CLK_SRC_AIF2BCLK 0x9 #define ARIZONA_CLK_SRC_AIF3BCLK 0xa +#define ARIZONA_FLL_SRC_NONE -1 #define ARIZONA_FLL_SRC_MCLK1 0 #define ARIZONA_FLL_SRC_MCLK2 1 #define ARIZONA_FLL_SRC_SLIMCLK 3 @@ -48,6 +49,14 @@ #define ARIZONA_MIXER_VOL_SHIFT 1 #define ARIZONA_MIXER_VOL_WIDTH 7 +#define ARIZONA_CLK_6MHZ 0 +#define ARIZONA_CLK_12MHZ 1 +#define ARIZONA_CLK_24MHZ 2 +#define ARIZONA_CLK_49MHZ 3 +#define ARIZONA_CLK_73MHZ 4 +#define ARIZONA_CLK_98MHZ 5 +#define ARIZONA_CLK_147MHZ 6 + #define ARIZONA_MAX_DAI 4 #define ARIZONA_MAX_ADSP 4 @@ -64,6 +73,12 @@ struct arizona_priv { int sysclk; int asyncclk; struct arizona_dai_priv dai[ARIZONA_MAX_DAI]; + + int num_inputs; + unsigned int in_pending; + + unsigned int spk_ena:2; + unsigned int spk_ena_pending:1; }; #define ARIZONA_NUM_MIXER_INPUTS 99 @@ -165,6 +180,12 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_ROUTES(name, name "L"), \ ARIZONA_MIXER_ROUTES(name, name "R") +#define ARIZONA_RATE_ENUM_SIZE 4 +extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; +extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; + +extern const struct soc_enum arizona_isrc_fsl[]; + extern const struct soc_enum arizona_in_vi_ramp; extern const struct soc_enum arizona_in_vd_ramp; @@ -201,8 +222,12 @@ struct arizona_fll { unsigned int base; unsigned int vco_mult; struct completion ok; - unsigned int fref; + unsigned int fout; + int sync_src; + unsigned int sync_freq; + int ref_src; + unsigned int ref_freq; char lock_name[ARIZONA_FLL_NAME_LEN]; char clock_ok_name[ARIZONA_FLL_NAME_LEN]; @@ -210,9 +235,13 @@ struct arizona_fll { 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_refclk(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout); extern int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout); +extern int arizona_init_spk(struct snd_soc_codec *codec); + extern int arizona_init_dai(struct arizona_priv *priv, int dai); int arizona_set_output_mode(struct snd_soc_codec *codec, int output, diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 2415a4118dbd..03036b326732 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -39,17 +39,15 @@ /* * CS4271 registers - * High byte represents SPI chip address (0x10) + write command (0) - * Low byte - codec register address */ -#define CS4271_MODE1 0x2001 /* Mode Control 1 */ -#define CS4271_DACCTL 0x2002 /* DAC Control */ -#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ -#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ -#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ -#define CS4271_ADCCTL 0x2006 /* ADC Control */ -#define CS4271_MODE2 0x2007 /* Mode Control 2 */ -#define CS4271_CHIPID 0x2008 /* Chip ID */ +#define CS4271_MODE1 0x01 /* Mode Control 1 */ +#define CS4271_DACCTL 0x02 /* DAC Control */ +#define CS4271_DACVOL 0x03 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x04 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x05 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x06 /* ADC Control */ +#define CS4271_MODE2 0x07 /* Mode Control 2 */ +#define CS4271_CHIPID 0x08 /* Chip ID */ #define CS4271_FIRSTREG CS4271_MODE1 #define CS4271_LASTREG CS4271_MODE2 @@ -144,23 +142,27 @@ * Array do not include Chip ID, as codec driver does not use * registers read operations at all */ -static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { - 0, - 0, - CS4271_DACCTL_AMUTE, - CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, - 0, - 0, - 0, - 0, +static const struct reg_default cs4271_reg_defaults[] = { + { CS4271_MODE1, 0, }, + { CS4271_DACCTL, CS4271_DACCTL_AMUTE, }, + { CS4271_DACVOL, CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, }, + { CS4271_VOLA, 0, }, + { CS4271_VOLB, 0, }, + { CS4271_ADCCTL, 0, }, + { CS4271_MODE2, 0, }, }; +static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == CS4271_CHIPID; +} + struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ - enum snd_soc_control_type bus_type; unsigned int mclk; bool master; bool deemph; + struct regmap *regmap; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -210,14 +212,14 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); if (ret < 0) return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); if (ret < 0) return ret; @@ -227,7 +229,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); if (ret < 0) return ret; @@ -252,7 +254,7 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - ret = snd_soc_update_bits(codec, CS4271_DACCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); if (ret < 0) return ret; @@ -341,14 +343,14 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, !dai->capture_active) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && !dai->playback_active)) { - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; } @@ -378,7 +380,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, val |= cs4271_clk_tab[i].ratio_mask; - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); if (ret < 0) return ret; @@ -386,22 +388,29 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, return cs4271_set_deemph(codec); } -static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) +static int cs4271_mute_stream(struct snd_soc_dai *dai, int mute, int stream) { struct snd_soc_codec *codec = dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int ret; int val_a = 0; int val_b = 0; + if (stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + if (mute) { val_a = CS4271_VOLA_MUTE; val_b = CS4271_VOLB_MUTE; } - ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLA, + CS4271_VOLA_MUTE, val_a); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLB, + CS4271_VOLB_MUTE, val_b); if (ret < 0) return ret; @@ -436,7 +445,7 @@ static const struct snd_soc_dai_ops cs4271_dai_ops = { .hw_params = cs4271_hw_params, .set_sysclk = cs4271_set_dai_sysclk, .set_fmt = cs4271_set_dai_fmt, - .digital_mute = cs4271_digital_mute, + .mute_stream = cs4271_mute_stream, }; static struct snd_soc_dai_driver cs4271_dai = { @@ -463,25 +472,33 @@ static struct snd_soc_dai_driver cs4271_dai = { static int cs4271_soc_suspend(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, CS4271_MODE2_PDN); if (ret < 0) return ret; + return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Restore codec state */ - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(cs4271->regmap); if (ret < 0) return ret; + /* then disable the power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; + return 0; } #else @@ -542,40 +559,22 @@ static int cs4271_probe(struct snd_soc_codec *codec) cs4271->gpio_nreset = gpio_nreset; - /* - * In case of I2C, chip address specified in board data. - * So cache IO operations use 8 bit codec register address. - * In case of SPI, chip address and register address - * passed together as 16 bit value. - * Anyway, register address is masked with 0xFF inside - * soc-cache code. - */ - if (cs4271->bus_type == SND_SOC_SPI) - ret = snd_soc_codec_set_cache_io(codec, 16, 8, - cs4271->bus_type); - else - ret = snd_soc_codec_set_cache_io(codec, 8, 8, - cs4271->bus_type); - if (ret) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; /* Power-up sequence requires 85 uS */ udelay(85); if (amutec_eq_bmutec) - snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_MUTECAEQUB, - CS4271_MODE2_MUTECAEQUB); + regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); @@ -597,13 +596,24 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, - .reg_cache_default = cs4271_dflt_reg, - .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), - .reg_word_size = sizeof(cs4271_dflt_reg[0]), - .compress_type = SND_SOC_FLAT_COMPRESSION, }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; @@ -613,7 +623,9 @@ static int cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->bus_type = SND_SOC_SPI; + cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); @@ -643,6 +655,18 @@ static const struct i2c_device_id cs4271_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); +static const struct regmap_config cs4271_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = CS4271_LASTREG, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -653,7 +677,9 @@ static int cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->bus_type = SND_SOC_I2C; + cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 6361dab48bd1..3b20c86cdb01 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1180,7 +1180,11 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].mmcc &= 0xC0; priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; priv->config[id].spc &= 0xFC; - priv->config[id].spc |= MCK_SCLK_MCLK; + /* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */ + if (priv->mclk >= 6400000) + priv->config[id].spc |= MCK_SCLK_64FS; + else + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index a4c16fd70f77..3eeada57e87d 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -739,14 +739,32 @@ static const unsigned int max98088_micboost_tlv[] = { 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), }; +static const unsigned int max98088_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98088_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + static const struct snd_kcontrol_new max98088_snd_controls[] = { - SOC_DOUBLE_R("Headphone Volume", M98088_REG_39_LVL_HP_L, - M98088_REG_3A_LVL_HP_R, 0, 31, 0), - SOC_DOUBLE_R("Speaker Volume", M98088_REG_3D_LVL_SPK_L, - M98088_REG_3E_LVL_SPK_R, 0, 31, 0), - SOC_DOUBLE_R("Receiver Volume", M98088_REG_3B_LVL_REC_L, - M98088_REG_3C_LVL_REC_R, 0, 31, 0), + SOC_DOUBLE_R_TLV("Headphone Volume", M98088_REG_39_LVL_HP_L, + M98088_REG_3A_LVL_HP_R, 0, 31, 0, max98088_hp_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", M98088_REG_3D_LVL_SPK_L, + M98088_REG_3E_LVL_SPK_R, 0, 31, 0, max98088_spk_tlv), + SOC_DOUBLE_R_TLV("Receiver Volume", M98088_REG_3B_LVL_REC_L, + M98088_REG_3C_LVL_REC_R, 0, 31, 0, max98088_spk_tlv), SOC_DOUBLE_R("Headphone Switch", M98088_REG_39_LVL_HP_L, M98088_REG_3A_LVL_HP_R, 7, 1, 1), @@ -2006,7 +2024,7 @@ static int max98088_probe(struct snd_soc_codec *codec) ret); goto err_access; } - dev_info(codec->dev, "revision %c\n", ret + 'A'); + dev_info(codec->dev, "revision %c\n", ret - 0x40 + 'A'); snd_soc_write(codec, M98088_REG_51_PWR_SYS, M98088_PWRSV); diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc176044994d..ce0d36412c97 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -23,8 +23,6 @@ #include <sound/max98090.h> #include "max98090.h" -#include <linux/version.h> - #define DEBUG #define EXTMIC_METHOD #define EXTMIC_METHOD_TEST @@ -509,16 +507,16 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, return 0; } -static const char * max98090_perf_pwr_text[] = +static const char *max98090_perf_pwr_text[] = { "High Performance", "Low Power" }; -static const char * max98090_pwr_perf_text[] = +static const char *max98090_pwr_perf_text[] = { "Low Power", "High Performance" }; static const struct soc_enum max98090_vcmbandgap_enum = SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); -static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; +static const char *max98090_osr128_text[] = { "64*fs", "128*fs" }; static const struct soc_enum max98090_osr128_enum = SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, @@ -535,28 +533,28 @@ static const struct soc_enum max98090_filter_dmic34mode_enum = M98090_FLT_DMIC34MODE_SHIFT, ARRAY_SIZE(max98090_mode_text), max98090_mode_text); -static const char * max98090_drcatk_text[] = +static const char *max98090_drcatk_text[] = { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; static const struct soc_enum max98090_drcatk_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); -static const char * max98090_drcrls_text[] = +static const char *max98090_drcrls_text[] = { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; static const struct soc_enum max98090_drcrls_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); -static const char * max98090_alccmp_text[] = +static const char *max98090_alccmp_text[] = { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; static const struct soc_enum max98090_alccmp_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); -static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; +static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; static const struct soc_enum max98090_drcexp_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, @@ -859,7 +857,7 @@ static const struct soc_enum mic2_mux_enum = static const struct snd_kcontrol_new max98090_mic2_mux = SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); -static const char * max98090_micpre_text[] = { "Off", "On" }; +static const char *max98090_micpre_text[] = { "Off", "On" }; static const struct soc_enum max98090_pa1en_enum = SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, @@ -1703,9 +1701,8 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, * seen for the case of TDM mode. The remaining cases have * normal logic. */ - if (max98090->tdm_slots > 1) { + if (max98090->tdm_slots > 1) regval ^= M98090_BCI_MASK; - } snd_soc_write(codec, M98090_REG_INTERFACE_FORMAT, regval); @@ -2059,17 +2056,14 @@ static irqreturn_t max98090_interrupt(int irq, void *data) if (!active) return IRQ_NONE; - if (active & M98090_CLD_MASK) { + if (active & M98090_CLD_MASK) dev_err(codec->dev, "M98090_CLD_MASK\n"); - } - if (active & M98090_SLD_MASK) { + if (active & M98090_SLD_MASK) dev_dbg(codec->dev, "M98090_SLD_MASK\n"); - } - if (active & M98090_ULK_MASK) { + if (active & M98090_ULK_MASK) dev_err(codec->dev, "M98090_ULK_MASK\n"); - } if (active & M98090_JDET_MASK) { dev_dbg(codec->dev, "M98090_JDET_MASK\n"); @@ -2080,13 +2074,11 @@ static irqreturn_t max98090_interrupt(int irq, void *data) msecs_to_jiffies(100)); } - if (active & M98090_DRCACT_MASK) { + if (active & M98090_DRCACT_MASK) dev_dbg(codec->dev, "M98090_DRCACT_MASK\n"); - } - if (active & M98090_DRCCLP_MASK) { + if (active & M98090_DRCCLP_MASK) dev_err(codec->dev, "M98090_DRCCLP_MASK\n"); - } return IRQ_HANDLED; } @@ -2324,7 +2316,7 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->pdata = i2c->dev.platform_data; max98090->irq = i2c->irq; - max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); if (IS_ERR(max98090->regmap)) { ret = PTR_ERR(max98090->regmap); dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); @@ -2334,18 +2326,13 @@ static int max98090_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98090, max98090_dai, ARRAY_SIZE(max98090_dai)); - if (ret < 0) - regmap_exit(max98090->regmap); - err_enable: return ret; } static int max98090_i2c_remove(struct i2c_client *client) { - struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); snd_soc_unregister_codec(&client->dev); - regmap_exit(max98090->regmap); return 0; } @@ -2369,7 +2356,7 @@ static int max98090_runtime_suspend(struct device *dev) return 0; } -static struct dev_pm_ops max98090_pm = { +static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, max98090_runtime_resume, NULL) }; diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 566ea3256e2d..721587c9cd84 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -1,3 +1,22 @@ +/* + * sound/soc/codecs/si476x.c -- Codec driver for SI476X chips + * + * Copyright (C) 2012 Innovative Converged Devices(ICD) + * Copyright (C) 2013 Andrey Smirnov + * + * Author: Andrey Smirnov <andrew.smirnov@gmail.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; version 2 of the License. + * + * 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. + * + */ + #include <linux/module.h> #include <linux/slab.h> #include <sound/pcm.h> @@ -45,13 +64,23 @@ static unsigned int si476x_codec_read(struct snd_soc_codec *codec, unsigned int reg) { int err; + unsigned int val; struct si476x_core *core = codec->control_data; si476x_core_lock(core); - err = si476x_core_cmd_get_property(core, reg); + if (!si476x_core_is_powered_up(core)) + regcache_cache_only(core->regmap, true); + + err = regmap_read(core->regmap, reg, &val); + + if (!si476x_core_is_powered_up(core)) + regcache_cache_only(core->regmap, false); si476x_core_unlock(core); - return err; + if (err < 0) + return err; + + return val; } static int si476x_codec_write(struct snd_soc_codec *codec, @@ -61,7 +90,13 @@ static int si476x_codec_write(struct snd_soc_codec *codec, struct si476x_core *core = codec->control_data; si476x_core_lock(core); - err = si476x_core_cmd_set_property(core, reg, val); + if (!si476x_core_is_powered_up(core)) + regcache_cache_only(core->regmap, true); + + err = regmap_write(core->regmap, reg, val); + + if (!si476x_core_is_powered_up(core)) + regcache_cache_only(core->regmap, false); si476x_core_unlock(core); return err; @@ -140,7 +175,7 @@ static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, dev_err(codec_dai->codec->dev, "Failed to set output format\n"); return err; } - + return 0; } @@ -182,7 +217,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK, - (width << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | + (width << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | (width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)); if (err < 0) { dev_err(dai->codec->dev, "Failed to set output width\n"); @@ -251,6 +286,6 @@ static struct platform_driver si476x_platform_driver = { }; module_platform_driver(si476x_platform_driver); -MODULE_AUTHOR("Andrey Smirnov <andrey.smirnov@convergeddevices.net>"); +MODULE_AUTHOR("Andrey Smirnov <andrew.smirnov@gmail.com>"); MODULE_DESCRIPTION("ASoC Si4761/64 codec driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c new file mode 100644 index 000000000000..d447c4aa1d5e --- /dev/null +++ b/sound/soc/codecs/tas5086.c @@ -0,0 +1,591 @@ +/* + * TAS5086 ASoC codec driver + * + * Copyright (c) 2013 Daniel Mack <zonque@gmail.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. + * + * TODO: + * - implement DAPM and input muxing + * - implement modulation limit + * - implement non-default PWM start + * + * Note that this chip has a very unusual register layout, specifically + * because the registers are of unequal size, and multi-byte registers + * require bulk writes to take effect. Regmap does not support that kind + * of devices. + * + * Currently, the driver does not touch any of the registers >= 0x20, so + * it doesn't matter because the entire map can be accessed as 8-bit + * array. In case more features will be added in the future + * that require access to higher registers, the entire regmap H/W I/O + * routines have to be open-coded. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas5086.h> + +#define TAS5086_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE) + +#define TAS5086_PCM_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +/* + * TAS5086 registers + */ +#define TAS5086_CLOCK_CONTROL 0x00 /* Clock control register */ +#define TAS5086_CLOCK_RATE(val) (val << 5) +#define TAS5086_CLOCK_RATE_MASK (0x7 << 5) +#define TAS5086_CLOCK_RATIO(val) (val << 2) +#define TAS5086_CLOCK_RATIO_MASK (0x7 << 2) +#define TAS5086_CLOCK_SCLK_RATIO_48 (1 << 1) +#define TAS5086_CLOCK_VALID (1 << 0) + +#define TAS5086_DEEMPH_MASK 0x03 +#define TAS5086_SOFT_MUTE_ALL 0x3f + +#define TAS5086_DEV_ID 0x01 /* Device ID register */ +#define TAS5086_ERROR_STATUS 0x02 /* Error status register */ +#define TAS5086_SYS_CONTROL_1 0x03 /* System control register 1 */ +#define TAS5086_SERIAL_DATA_IF 0x04 /* Serial data interface register */ +#define TAS5086_SYS_CONTROL_2 0x05 /* System control register 2 */ +#define TAS5086_SOFT_MUTE 0x06 /* Soft mute register */ +#define TAS5086_MASTER_VOL 0x07 /* Master volume */ +#define TAS5086_CHANNEL_VOL(X) (0x08 + (X)) /* Channel 1-6 volume */ +#define TAS5086_VOLUME_CONTROL 0x09 /* Volume control register */ +#define TAS5086_MOD_LIMIT 0x10 /* Modulation limit register */ +#define TAS5086_PWM_START 0x18 /* PWM start register */ +#define TAS5086_SURROUND 0x19 /* Surround register */ +#define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */ +#define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */ +#define TAS5086_BKNDERR 0x1c + +/* + * Default TAS5086 power-up configuration + */ +static const struct reg_default tas5086_reg_defaults[] = { + { 0x00, 0x6c }, + { 0x01, 0x03 }, + { 0x02, 0x00 }, + { 0x03, 0xa0 }, + { 0x04, 0x05 }, + { 0x05, 0x60 }, + { 0x06, 0x00 }, + { 0x07, 0xff }, + { 0x08, 0x30 }, + { 0x09, 0x30 }, + { 0x0a, 0x30 }, + { 0x0b, 0x30 }, + { 0x0c, 0x30 }, + { 0x0d, 0x30 }, + { 0x0e, 0xb1 }, + { 0x0f, 0x00 }, + { 0x10, 0x02 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x3f }, + { 0x19, 0x00 }, + { 0x1a, 0x18 }, + { 0x1b, 0x82 }, + { 0x1c, 0x05 }, +}; + +static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); +} + +static bool tas5086_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5086_DEV_ID: + case TAS5086_ERROR_STATUS: + return true; + } + + return false; +} + +static bool tas5086_writeable_reg(struct device *dev, unsigned int reg) +{ + return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID); +} + +struct tas5086_private { + struct regmap *regmap; + unsigned int mclk, sclk; + unsigned int format; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; +}; + +static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; + +static int tas5086_set_deemph(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int i, val = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) + if (tas5086_deemph[i] == priv->rate) + val = i; + + return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1, + TAS5086_DEEMPH_MASK, val); +} + +static int tas5086_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int tas5086_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return tas5086_set_deemph(codec); +} + + +static int tas5086_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case TAS5086_CLK_IDX_MCLK: + priv->mclk = freq; + break; + case TAS5086_CLK_IDX_SCLK: + priv->sclk = freq; + break; + } + + return 0; +} + +static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The TAS5086 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + /* we need to refer to the data format from hw_params() */ + priv->format = format; + + return 0; +} + +static const int tas5086_sample_rates[] = { + 32000, 38000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const int tas5086_ratios[] = { + 64, 128, 192, 256, 384, 512 +}; + +static int index_in_array(const int *array, int len, int needle) +{ + int i; + + for (i = 0; i < len; i++) + if (array[i] == needle) + return i; + + return -ENOENT; +} + +static int tas5086_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 tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int val; + int ret; + + priv->rate = params_rate(params); + + /* Look up the sample rate and refer to the offset in the list */ + val = index_in_array(tas5086_sample_rates, + ARRAY_SIZE(tas5086_sample_rates), priv->rate); + + if (val < 0) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATE_MASK, + TAS5086_CLOCK_RATE(val)); + if (ret < 0) + return ret; + + /* MCLK / Fs ratio */ + val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), + priv->mclk / priv->rate); + if (val < 0) { + dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATIO_MASK, + TAS5086_CLOCK_RATIO(val)); + if (ret < 0) + return ret; + + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_SCLK_RATIO_48, + (priv->sclk == 48 * priv->rate) ? + TAS5086_CLOCK_SCLK_RATIO_48 : 0); + if (ret < 0) + return ret; + + /* + * The chip has a very unituitive register mapping and muxes information + * about data format and sample depth into the same register, but not on + * a logical bit-boundary. Hence, we have to refer to the format passed + * in the set_dai_fmt() callback and set up everything from here. + * + * First, determine the 'base' value, using the format ... + */ + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x03; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x06; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* ... then add the offset for the sample bit depth. */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val += 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val += 1; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + val += 2; + break; + default: + dev_err(codec->dev, "Invalid bit width\n"); + return -EINVAL; + }; + + ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); + if (ret < 0) + return ret; + + /* clock is considered valid now */ + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_VALID, TAS5086_CLOCK_VALID); + if (ret < 0) + return ret; + + return tas5086_set_deemph(codec); +} + +static int tas5086_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + if (mute) + val = TAS5086_SOFT_MUTE_ALL; + + return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, val); +} + +/* TAS5086 controls */ +static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new tas5086_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", TAS5086_MASTER_VOL, + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + TAS5086_CHANNEL_VOL(0), TAS5086_CHANNEL_VOL(1), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + TAS5086_CHANNEL_VOL(2), TAS5086_CHANNEL_VOL(3), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + TAS5086_CHANNEL_VOL(4), TAS5086_CHANNEL_VOL(5), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + tas5086_get_deemph, tas5086_put_deemph), +}; + +static const struct snd_soc_dai_ops tas5086_dai_ops = { + .hw_params = tas5086_hw_params, + .set_sysclk = tas5086_set_dai_sysclk, + .set_fmt = tas5086_set_dai_fmt, + .mute_stream = tas5086_mute_stream, +}; + +static struct snd_soc_dai_driver tas5086_dai = { + .name = "tas5086-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = TAS5086_PCM_RATES, + .formats = TAS5086_PCM_FORMATS, + }, + .ops = &tas5086_dai_ops, +}; + +#ifdef CONFIG_PM +static int tas5086_soc_resume(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* Restore codec state */ + return regcache_sync(priv->regmap); +} +#else +#define tas5086_soc_resume NULL +#endif /* CONFIG_PM */ + +#ifdef CONFIG_OF +static const struct of_device_id tas5086_dt_ids[] = { + { .compatible = "ti,tas5086", }, + { } +}; +MODULE_DEVICE_TABLE(of, tas5086_dt_ids); +#endif + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_probe(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int charge_period = 1300000; /* hardware default is 1300 ms */ + int i, ret; + + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { + struct device_node *of_node = codec->dev->of_node; + of_property_read_u32(of_node, "ti,charge-period", &charge_period); + } + + /* lookup and set split-capacitor charge period */ + if (charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(codec->dev, + "Invalid split-cap charge period of %d ns.\n", + charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* set master volume to 0 dB */ + ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); + if (ret < 0) + return ret; + + /* mute all channels for now */ + ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + TAS5086_SOFT_MUTE_ALL); + if (ret < 0) + return ret; + + return 0; +} + +static int tas5086_remove(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->gpio_nreset)) + /* Set codec to the reset state */ + gpio_set_value(priv->gpio_nreset, 0); + + return 0; +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { + .probe = tas5086_probe, + .remove = tas5086_remove, + .resume = tas5086_soc_resume, + .controls = tas5086_controls, + .num_controls = ARRAY_SIZE(tas5086_controls), +}; + +static const struct i2c_device_id tas5086_i2c_id[] = { + { "tas5086", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); + +static const struct regmap_config tas5086_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(tas5086_reg_defaults), + .reg_defaults = tas5086_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5086_volatile_reg, + .writeable_reg = tas5086_writeable_reg, + .readable_reg = tas5086_accessible_reg, +}; + +static int tas5086_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tas5086_private *priv; + struct device *dev = &i2c->dev; + int gpio_nreset = -EINVAL; + int i, ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, priv); + + if (of_match_device(of_match_ptr(tas5086_dt_ids), dev)) { + struct device_node *of_node = dev->of_node; + gpio_nreset = of_get_named_gpio(of_node, "reset-gpio", 0); + } + + if (gpio_is_valid(gpio_nreset)) + if (devm_gpio_request(dev, gpio_nreset, "TAS5086 Reset")) + gpio_nreset = -EINVAL; + + if (gpio_is_valid(gpio_nreset)) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } + + priv->gpio_nreset = gpio_nreset; + + /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ + ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); + if (ret < 0) + return ret; + + if (i != 0x3) { + dev_err(dev, + "Failed to identify TAS5086 codec (got %02x)\n", i); + return -ENODEV; + } + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); +} + +static int tas5086_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver tas5086_i2c_driver = { + .driver = { + .name = "tas5086", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas5086_dt_ids), + }, + .id_table = tas5086_i2c_id, + .probe = tas5086_i2c_probe, + .remove = tas5086_i2c_remove, +}; + +module_i2c_driver(tas5086_i2c_driver); + +MODULE_AUTHOR("Daniel Mack <zonque@gmail.com>"); +MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index ad2fee4bb4cd..8df2b6e1a1a6 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -342,7 +342,7 @@ static void byte_swap_64(u64 *data_in, u64 *data_out, u32 len) data_out[i] = cpu_to_be64(le64_to_cpu(data_in[i])); } -static int wm0010_firmware_load(char *name, struct snd_soc_codec *codec) +static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) { struct spi_device *spi = to_spi_device(codec->dev); struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); @@ -361,8 +361,8 @@ static int wm0010_firmware_load(char *name, struct snd_soc_codec *codec) ret = request_firmware(&fw, name, codec->dev); if (ret != 0) { - dev_err(codec->dev, "Failed to request application: %d\n", - ret); + dev_err(codec->dev, "Failed to request application(%s): %d\n", + name, ret); return ret; } diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index f2ac38b61a1b..7fefd766b582 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -761,6 +761,8 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg) case WM2000_REG_SYS_CTL2: case WM2000_REG_ANC_STAT: case WM2000_REG_IF_CTL: + case WM2000_REG_ANA_MIC_CTL: + case WM2000_REG_SPK_CTL: return true; default: return false; @@ -771,7 +773,7 @@ static const struct regmap_config wm2000_regmap = { .reg_bits = 16, .val_bits = 8, - .max_register = WM2000_REG_IF_CTL, + .max_register = WM2000_REG_SPK_CTL, .readable_reg = wm2000_readable_reg, }; diff --git a/sound/soc/codecs/wm2000.h b/sound/soc/codecs/wm2000.h index fb812cd9e77d..3870c0e1d246 100644 --- a/sound/soc/codecs/wm2000.h +++ b/sound/soc/codecs/wm2000.h @@ -30,6 +30,8 @@ #define WM2000_REG_SYS_CTL2 0xf004 #define WM2000_REG_ANC_STAT 0xf005 #define WM2000_REG_IF_CTL 0xf006 +#define WM2000_REG_ANA_MIC_CTL 0xf028 +#define WM2000_REG_SPK_CTL 0xf034 /* SPEECH_CLARITY */ #define WM2000_SPEECH_CLARITY 0x01 diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index ddc98f02ecbd..57ba315d0c84 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1565,7 +1565,7 @@ static int wm2200_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); + ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 15bc31f1abb1..e895d3939eef 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -36,9 +36,6 @@ struct wm5102_priv { struct arizona_priv core; struct arizona_fll fll[2]; - - unsigned int spk_ena:2; - unsigned int spk_ena_pending:1; }; static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); @@ -615,6 +612,26 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static const char *wm5102_osr_text[] = { + "Low power", "Normal", "High performance", +}; + +static const unsigned int wm5102_osr_val[] = { + 0x0, 0x3, 0x5, +}; + +static const struct soc_enum wm5102_hpout_osr[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + wm5102_osr_text, wm5102_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUT2_OSR_SHIFT, 0x7, 3, + wm5102_osr_text, wm5102_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + wm5102_osr_text, wm5102_osr_val), +}; + #define WM5102_NG_SRC(name, base) \ SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ @@ -745,6 +762,9 @@ SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), + ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), @@ -761,6 +781,8 @@ 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("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), @@ -790,6 +812,10 @@ 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_VALUE_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), +SOC_VALUE_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), +SOC_VALUE_ENUM("HPOUT3 OSR", wm5102_hpout_osr[2]), + SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), @@ -828,47 +854,6 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), }; -static int wm5102_spk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, - int event) -{ - struct snd_soc_codec *codec = w->codec; - struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec); - - if (arizona->rev < 1) - return 0; - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - if (!wm5102->spk_ena) { - snd_soc_write(codec, 0x4f5, 0x25a); - wm5102->spk_ena_pending = true; - } - break; - case SND_SOC_DAPM_POST_PMU: - if (wm5102->spk_ena_pending) { - msleep(75); - snd_soc_write(codec, 0x4f5, 0xda); - wm5102->spk_ena_pending = false; - wm5102->spk_ena++; - } - break; - case SND_SOC_DAPM_PRE_PMD: - wm5102->spk_ena--; - if (!wm5102->spk_ena) - snd_soc_write(codec, 0x4f5, 0x25a); - break; - case SND_SOC_DAPM_POST_PMD: - if (!wm5102->spk_ena) - snd_soc_write(codec, 0x4f5, 0x0da); - break; - } - - return 0; -} - - 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); @@ -984,22 +969,28 @@ 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), @@ -1146,12 +1137,6 @@ SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, 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, wm5102_spk_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, wm5102_spk_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), @@ -1494,6 +1479,12 @@ static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout); case WM5102_FLL2: return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout); + case WM5102_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm5102->fll[0], source, Fref, + Fout); + case WM5102_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm5102->fll[1], source, Fref, + Fout); default: return -EINVAL; } @@ -1581,10 +1572,12 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 1); + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; + arizona_init_spk(codec); + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; @@ -1604,13 +1597,6 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) #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, @@ -1653,6 +1639,7 @@ static int wm5102_probe(struct platform_device *pdev) platform_set_drvdata(pdev, wm5102); wm5102->core.arizona = arizona; + wm5102->core.num_inputs = 6; wm5102->core.adsp[0].part = "wm5102"; wm5102->core.adsp[0].num = 1; @@ -1677,6 +1664,12 @@ static int wm5102_probe(struct platform_device *pdev) ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, &wm5102->fll[1]); + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + for (i = 0; i < ARRAY_SIZE(wm5102_dai); i++) arizona_init_dai(&wm5102->core, i); diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h index d30477f3070c..adb38040f661 100644 --- a/sound/soc/codecs/wm5102.h +++ b/sound/soc/codecs/wm5102.h @@ -15,7 +15,9 @@ #include "arizona.h" -#define WM5102_FLL1 1 -#define WM5102_FLL2 2 +#define WM5102_FLL1 1 +#define WM5102_FLL2 2 +#define WM5102_FLL1_REFCLK 3 +#define WM5102_FLL2_REFCLK 4 #endif diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 7841b42a819c..731884e04776 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -416,28 +416,36 @@ 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | 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_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), @@ -569,12 +577,6 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, SND_SOC_DAPM_PGA_E("OUT3R", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3R_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), @@ -880,6 +882,12 @@ static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout); case WM5110_FLL2: return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout); + case WM5110_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm5110->fll[0], source, Fref, + Fout); + case WM5110_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm5110->fll[1], source, Fref, + Fout); default: return -EINVAL; } @@ -987,15 +995,6 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec) #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_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, - ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_DAC_DIGITAL_VOLUME_2L, @@ -1040,6 +1039,7 @@ static int wm5110_probe(struct platform_device *pdev) platform_set_drvdata(pdev, wm5110); wm5110->core.arizona = arizona; + wm5110->core.num_inputs = 8; for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++) wm5110->fll[i].vco_mult = 3; diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h index 75e9351ccab0..e6c0cd4235c5 100644 --- a/sound/soc/codecs/wm5110.h +++ b/sound/soc/codecs/wm5110.h @@ -15,7 +15,9 @@ #include "arizona.h" -#define WM5110_FLL1 1 -#define WM5110_FLL2 2 +#define WM5110_FLL1 1 +#define WM5110_FLL2 2 +#define WM5110_FLL1_REFCLK 3 +#define WM5110_FLL2_REFCLK 4 #endif diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index f8a31ad0b203..9d88437cdcd1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -478,6 +478,8 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol, /* ALSA can only do steps of .01dB */ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0); + static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); @@ -698,6 +700,8 @@ SOC_ENUM("DAC Mute Mode", mute_mode), SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0), SOC_ENUM("DAC Companding Mode", dac_companding), SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0), +SOC_SINGLE_TLV("DAC Boost Volume", WM8903_AUDIO_INTERFACE_0, 9, 3, 0, + dac_boost_tlv), SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0, wm8903_get_deemph, wm8903_put_deemph), diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index a64b93425ae3..0a4ffdd1d2a7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -204,6 +204,7 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1); static const struct snd_kcontrol_new wm8960_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, @@ -213,6 +214,15 @@ SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL, SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL, 7, 1, 0), +SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume", + WM8960_INBMIX1, 4, 7, 0, boost_tlv), +SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume", + WM8960_INBMIX1, 1, 7, 0, boost_tlv), +SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume", + WM8960_INBMIX2, 4, 7, 0, boost_tlv), +SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume", + WM8960_INBMIX2, 1, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC, 0, 255, 0, dac_tlv), diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c9bd445c4976..14094f558e03 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2209,7 +2209,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, vmid_reference(codec); break; case WM8958: - if (wm8994->revision < 1) + if (control->revision < 1) vmid_reference(codec); break; default: @@ -2244,7 +2244,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, vmid_dereference(codec); break; case WM8958: - if (wm8994->revision < 1) + if (control->revision < 1) vmid_dereference(codec); break; default: @@ -2268,10 +2268,26 @@ out: */ if (max(wm8994->aifclk[0], wm8994->aifclk[1]) < 50000) { dev_dbg(codec->dev, "Configuring AIFs for 128fs\n"); + + wm8994->aifdiv[0] = snd_soc_read(codec, WM8994_AIF1_RATE) + & WM8994_AIF1CLK_RATE_MASK; + wm8994->aifdiv[1] = snd_soc_read(codec, WM8994_AIF2_RATE) + & WM8994_AIF1CLK_RATE_MASK; + snd_soc_update_bits(codec, WM8994_AIF1_RATE, WM8994_AIF1CLK_RATE_MASK, 0x1); snd_soc_update_bits(codec, WM8994_AIF2_RATE, WM8994_AIF2CLK_RATE_MASK, 0x1); + } else if (wm8994->aifdiv[0]) { + snd_soc_update_bits(codec, WM8994_AIF1_RATE, + WM8994_AIF1CLK_RATE_MASK, + wm8994->aifdiv[0]); + snd_soc_update_bits(codec, WM8994_AIF2_RATE, + WM8994_AIF2CLK_RATE_MASK, + wm8994->aifdiv[1]); + + wm8994->aifdiv[0] = 0; + wm8994->aifdiv[1] = 0; } return 0; @@ -2368,10 +2384,26 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, */ if (max(wm8994->aifclk[0], wm8994->aifclk[1]) < 50000) { dev_dbg(codec->dev, "Configuring AIFs for 128fs\n"); + + wm8994->aifdiv[0] = snd_soc_read(codec, WM8994_AIF1_RATE) + & WM8994_AIF1CLK_RATE_MASK; + wm8994->aifdiv[1] = snd_soc_read(codec, WM8994_AIF2_RATE) + & WM8994_AIF1CLK_RATE_MASK; + snd_soc_update_bits(codec, WM8994_AIF1_RATE, WM8994_AIF1CLK_RATE_MASK, 0x1); snd_soc_update_bits(codec, WM8994_AIF2_RATE, WM8994_AIF2CLK_RATE_MASK, 0x1); + } else if (wm8994->aifdiv[0]) { + snd_soc_update_bits(codec, WM8994_AIF1_RATE, + WM8994_AIF1CLK_RATE_MASK, + wm8994->aifdiv[0]); + snd_soc_update_bits(codec, WM8994_AIF2_RATE, + WM8994_AIF2CLK_RATE_MASK, + wm8994->aifdiv[1]); + + wm8994->aifdiv[0] = 0; + wm8994->aifdiv[1] = 0; } return 0; @@ -2411,7 +2443,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { switch (control->type) { case WM8958: - if (wm8994->revision == 0) { + if (control->revision == 0) { /* Optimise performance for rev A */ snd_soc_update_bits(codec, WM8958_CHARGE_PUMP_2, @@ -2656,6 +2688,8 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = wm8994->wm8994; + struct wm8994_pdata *pdata = &control->pdata; int aif1_reg; int aif2_reg; int bclk_reg; @@ -2723,7 +2757,14 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, } wm8994->channels[id] = params_channels(params); - switch (params_channels(params)) { + if (pdata->max_channels_clocked[id] && + wm8994->channels[id] > pdata->max_channels_clocked[id]) { + dev_dbg(dai->dev, "Constraining channels to %d from %d\n", + pdata->max_channels_clocked[id], wm8994->channels[id]); + wm8994->channels[id] = pdata->max_channels_clocked[id]; + } + + switch (wm8994->channels[id]) { case 1: case 2: bclk_rate *= 2; @@ -2745,7 +2786,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "AIF%dCLK is %dHz, target BCLK %dHz\n", dai->id, wm8994->aifclk[id], bclk_rate); - if (params_channels(params) == 1 && + if (wm8994->channels[id] == 1 && (snd_soc_read(codec, aif1_reg) & 0x18) == 0x18) aif2 |= WM8994_AIF1_MONO; @@ -3053,7 +3094,7 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) int i, ret; unsigned int val, mask; - if (wm8994->revision < 4) { + if (control->revision < 4) { /* force a HW read */ ret = regmap_read(control->regmap, WM8994_POWER_MANAGEMENT_5, &val); @@ -3870,7 +3911,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) codec->dapm.idle_bias_off = 1; /* Set revision-specific configuration */ - wm8994->revision = snd_soc_read(codec, WM8994_CHIP_REVISION); switch (control->type) { case WM8994: /* Single ended line outputs should have VMID on. */ @@ -3878,7 +3918,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) !control->pdata.lineout2_diff) codec->dapm.idle_bias_off = 0; - switch (wm8994->revision) { + switch (control->revision) { case 2: case 3: wm8994->hubs.dcs_codes_l = -5; @@ -3897,7 +3937,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; - switch (wm8994->revision) { + switch (control->revision) { case 0: break; default: @@ -4000,7 +4040,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM1811: - if (control->cust_id > 1 || wm8994->revision > 1) { + if (control->cust_id > 1 || control->revision > 1) { ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_GPIO(6), wm1811_jackdet_irq, "JACKDET", @@ -4114,7 +4154,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); - if (wm8994->revision < 4) { + if (control->revision < 4) { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, ARRAY_SIZE(wm8994_lateclk_revd_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, @@ -4135,7 +4175,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8958_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); - if (wm8994->revision < 1) { + if (control->revision < 1) { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, ARRAY_SIZE(wm8994_lateclk_revd_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, @@ -4174,7 +4214,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); - if (wm8994->revision < 4) { + if (control->revision < 4) { snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, @@ -4185,7 +4225,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } break; case WM8958: - if (wm8994->revision < 1) { + if (control->revision < 1) { snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 45f192702024..55ddf4d57d9b 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -79,6 +79,7 @@ struct wm8994_priv { int sysclk_rate[2]; int mclk[2]; int aifclk[2]; + int aifdiv[2]; int channels[2]; struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; @@ -146,8 +147,6 @@ struct wm8994_priv { wm1811_mic_id_cb mic_id_cb; void *mic_id_cb_data; - int revision; - unsigned int aif1clk_enable:1; unsigned int aif2clk_enable:1; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9af1bddc4c62..3470b649c0b2 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -31,6 +31,7 @@ #include <linux/mfd/arizona/registers.h> +#include "arizona.h" #include "wm_adsp.h" #define adsp_crit(_dsp, fmt, ...) \ @@ -193,17 +194,25 @@ static void wm_adsp_buf_free(struct list_head *list) #define WM_ADSP_NUM_FW 4 +#define WM_ADSP_FW_MBC_VSS 0 +#define WM_ADSP_FW_TX 1 +#define WM_ADSP_FW_TX_SPK 2 +#define WM_ADSP_FW_RX_ANC 3 + static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { - "MBC/VSS", "Tx", "Tx Speaker", "Rx ANC" + [WM_ADSP_FW_MBC_VSS] = "MBC/VSS", + [WM_ADSP_FW_TX] = "Tx", + [WM_ADSP_FW_TX_SPK] = "Tx Speaker", + [WM_ADSP_FW_RX_ANC] = "Rx ANC", }; static struct { const char *file; } wm_adsp_fw[WM_ADSP_NUM_FW] = { - { .file = "mbc-vss" }, - { .file = "tx" }, - { .file = "tx-spk" }, - { .file = "rx-anc" }, + [WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" }, + [WM_ADSP_FW_TX] = { .file = "tx" }, + [WM_ADSP_FW_TX_SPK] = { .file = "tx-spk" }, + [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, }; static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, @@ -246,17 +255,52 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 3, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; -const struct snd_kcontrol_new wm_adsp_fw_controls[] = { +const struct snd_kcontrol_new wm_adsp1_fw_controls[] = { + SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], + wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], + wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], + wm_adsp_fw_get, wm_adsp_fw_put), +}; +EXPORT_SYMBOL_GPL(wm_adsp1_fw_controls); + +#if IS_ENABLED(CONFIG_SND_SOC_ARIZONA) +static const struct soc_enum wm_adsp2_rate_enum[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP2_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; + +const struct snd_kcontrol_new wm_adsp2_fw_controls[] = { SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM("DSP1 Rate", wm_adsp2_rate_enum[0]), SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM("DSP2 Rate", wm_adsp2_rate_enum[1]), SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM("DSP3 Rate", wm_adsp2_rate_enum[2]), SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM("DSP4 Rate", wm_adsp2_rate_enum[3]), }; -EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); +EXPORT_SYMBOL_GPL(wm_adsp2_fw_controls); +#endif static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -549,13 +593,30 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) buf_size = sizeof(adsp1_id); algs = be32_to_cpu(adsp1_id.algs); + dsp->fw_id = be32_to_cpu(adsp1_id.fw.id); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", - be32_to_cpu(adsp1_id.fw.id), + dsp->fw_id, (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp1_id.fw.ver) & 0xff, algs); + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_ZM; + region->alg = be32_to_cpu(adsp1_id.fw.id); + region->base = be32_to_cpu(adsp1_id.zm); + list_add_tail(®ion->list, &dsp->alg_regions); + + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_DM; + region->alg = be32_to_cpu(adsp1_id.fw.id); + region->base = be32_to_cpu(adsp1_id.dm); + list_add_tail(®ion->list, &dsp->alg_regions); + pos = sizeof(adsp1_id) / 2; term = pos + ((sizeof(*adsp1_alg) * algs) / 2); break; @@ -573,13 +634,38 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) buf_size = sizeof(adsp2_id); algs = be32_to_cpu(adsp2_id.algs); + dsp->fw_id = be32_to_cpu(adsp2_id.fw.id); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", - be32_to_cpu(adsp2_id.fw.id), + dsp->fw_id, (be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp2_id.fw.ver) & 0xff, algs); + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_XM; + region->alg = be32_to_cpu(adsp2_id.fw.id); + region->base = be32_to_cpu(adsp2_id.xm); + list_add_tail(®ion->list, &dsp->alg_regions); + + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_YM; + region->alg = be32_to_cpu(adsp2_id.fw.id); + region->base = be32_to_cpu(adsp2_id.ym); + list_add_tail(®ion->list, &dsp->alg_regions); + + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_ZM; + region->alg = be32_to_cpu(adsp2_id.fw.id); + region->base = be32_to_cpu(adsp2_id.zm); + list_add_tail(®ion->list, &dsp->alg_regions); + pos = sizeof(adsp2_id) / 2; term = pos + ((sizeof(*adsp2_alg) * algs) / 2); break; @@ -781,8 +867,24 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) case (WMFW_INFO_TEXT << 8): break; case (WMFW_ABSOLUTE << 8): - region_name = "register"; - reg = offset; + /* + * Old files may use this for global + * coefficients. + */ + if (le32_to_cpu(blk->id) == dsp->fw_id && + offset == 0) { + region_name = "global coefficients"; + mem = wm_adsp_find_region(dsp, type); + if (!mem) { + adsp_err(dsp, "No ZM\n"); + break; + } + reg = wm_adsp_region_to_reg(mem, 0); + + } else { + region_name = "register"; + reg = offset; + } break; case WMFW_ADSP1_DM: diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index cb8871a3ec00..fea514627526 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -46,6 +46,8 @@ struct wm_adsp { struct list_head alg_regions; + int fw_id; + const struct wm_adsp_region *mem; int num_mems; @@ -65,7 +67,8 @@ struct wm_adsp { .shift = num, .event = wm_adsp2_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } -extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; +extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; +extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *adsp); int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 867ae97ddcec..f5d81b948759 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -199,11 +199,12 @@ static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) list_add_tail(&cache->list, &hubs->dcs_cache); } -static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec, +static int wm_hubs_read_dc_servo(struct snd_soc_codec *codec, u16 *reg_l, u16 *reg_r) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); u16 dcs_reg, reg; + int ret = 0; switch (hubs->dcs_readback_mode) { case 2: @@ -236,8 +237,9 @@ static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec, break; default: WARN(1, "Unknown DCS readback method\n"); - return; + ret = -1; } + return ret; } /* @@ -286,7 +288,8 @@ static void enable_dc_servo(struct snd_soc_codec *codec) WM8993_DCS_TRIG_STARTUP_1); } - wm_hubs_read_dc_servo(codec, ®_l, ®_r); + if (wm_hubs_read_dc_servo(codec, ®_l, ®_r) < 0) + return; dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 821831207180..ebe82947bab3 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -645,6 +645,10 @@ static struct snd_soc_dai_driver davinci_i2s_dai = { }; +static const struct snd_soc_component_driver davinci_i2s_component = { + .name = "davinci-i2s", +}; + static int davinci_i2s_probe(struct platform_device *pdev) { struct snd_platform_data *pdata = pdev->dev.platform_data; @@ -727,20 +731,21 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, dev); - ret = snd_soc_register_dai(&pdev->dev, &davinci_i2s_dai); + ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component, + &davinci_i2s_dai, 1); if (ret != 0) goto err_release_clk; ret = davinci_soc_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - goto err_unregister_dai; + goto err_unregister_component; } return 0; -err_unregister_dai: - snd_soc_unregister_dai(&pdev->dev); +err_unregister_component: + snd_soc_unregister_component(&pdev->dev); err_release_clk: clk_disable(dev->clk); clk_put(dev->clk); @@ -751,7 +756,7 @@ static int davinci_i2s_remove(struct platform_device *pdev) { struct davinci_mcbsp_dev *dev = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); davinci_soc_platform_unregister(&pdev->dev); clk_disable(dev->clk); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c9d8c1..8b85049daab0 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,8 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -634,35 +636,43 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, * callback, take it into account here. That allows us to for example * send 32 bits per channel to the codec, while only 16 of them carry * audio payload. - * The clock ratio is given for a full period of data (both left and - * right channels), so it has to be divided by 2. + * The clock ratio is given for a full period of data (for I2S format + * both left and right channels), so it has to be divided by number of + * tdm-slots (for I2S - divided by 2). */ if (dev->bclk_lrclk_ratio) - word_length = dev->bclk_lrclk_ratio / 2; + word_length = dev->bclk_lrclk_ratio / dev->tdm_slots; /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); + if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(rotate), TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXROT(rotate), RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, + mask); + } + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -682,17 +692,33 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { if (dev->txnumevt * tx_ser > 64) dev->txnumevt = 1; @@ -729,6 +755,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -772,12 +800,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - - /* TXMASK for 24 bits */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); - /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, @@ -812,12 +834,21 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + u8 active_serializers; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; + + active_serializers = (channels + slots - 1) / slots; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_level = dev->txnumevt; + fifo_level = dev->txnumevt * active_serializers; else - fifo_level = dev->rxnumevt; + fifo_level = dev->rxnumevt * active_serializers; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -936,13 +967,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, @@ -962,6 +993,10 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { }; +static const struct snd_soc_component_driver davinci_mcasp_component = { + .name = "davinci-mcasp", +}; + static const struct of_device_id mcasp_dt_ids[] = { { .compatible = "ti,dm646x-mcasp-audio", @@ -1015,8 +1050,16 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata->op_mode = val; ret = of_property_read_u32(np, "tdm-slots", &val); - if (ret >= 0) + if (ret >= 0) { + if (val < 2 || val > 32) { + dev_err(&pdev->dev, + "tdm-slots must be in rage [2-32]\n"); + ret = -EINVAL; + goto nodata; + } + pdata->tdm_slots = val; + } ret = of_property_read_u32(np, "num-serializer", &val); if (ret >= 0) @@ -1170,7 +1213,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; dev_set_drvdata(&pdev->dev, dev); - ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]); + ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, + &davinci_mcasp_dai[pdata->op_mode], 1); if (ret != 0) goto err_release_clk; @@ -1178,13 +1222,13 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = davinci_soc_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - goto err_unregister_dai; + goto err_unregister_component; } return 0; -err_unregister_dai: - snd_soc_unregister_dai(&pdev->dev); +err_unregister_component: + snd_soc_unregister_component(&pdev->dev); err_release_clk: pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); @@ -1194,7 +1238,7 @@ err_release_clk: static int davinci_mcasp_remove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); davinci_soc_platform_unregister(&pdev->dev); pm_runtime_put_sync(&pdev->dev); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 0edd3b5a37fd..a9ac0c11da71 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -38,7 +38,7 @@ struct davinci_audio_dev { u8 num_serializer; u8 *serial_dir; u8 version; - u8 bclk_lrclk_ratio; + u16 bclk_lrclk_ratio; /* McASP FIFO related */ u8 txnumevt; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index afab81f844ae..b2f27c2e5fdc 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -200,7 +200,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; + dst_bidx = 4; src_cidx = data_type * fifo_level; dst_cidx = 0; } else { @@ -223,9 +223,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level, + count, fifo_level, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 07bde2e6f84e..30587c0cdbd2 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -204,6 +204,10 @@ static struct snd_soc_dai_driver davinci_vcif_dai = { }; +static const struct snd_soc_component_driver davinci_vcif_component = { + .name = "davinci-vcif", +}; + static int davinci_vcif_probe(struct platform_device *pdev) { struct davinci_vc *davinci_vc = pdev->dev.platform_data; @@ -234,7 +238,8 @@ static int davinci_vcif_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, davinci_vcif_dev); - ret = snd_soc_register_dai(&pdev->dev, &davinci_vcif_dai); + ret = snd_soc_register_component(&pdev->dev, &davinci_vcif_component, + &davinci_vcif_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "could not register dai\n"); return ret; @@ -243,7 +248,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) ret = davinci_soc_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return ret; } @@ -252,7 +257,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) static int davinci_vcif_remove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); davinci_soc_platform_unregister(&pdev->dev); return 0; diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index deb30d59965e..593a3ea12d4c 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -297,6 +297,10 @@ static struct snd_soc_dai_ops dw_i2s_dai_ops = { .trigger = dw_i2s_trigger, }; +static const struct snd_soc_component_driver dw_i2s_component = { + .name = "dw-i2s", +}; + #ifdef CONFIG_PM static int dw_i2s_suspend(struct snd_soc_dai *dai) @@ -413,7 +417,8 @@ static int dw_i2s_probe(struct platform_device *pdev) dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); - ret = snd_soc_register_dai(&pdev->dev, dw_i2s_dai); + ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, + dw_i2s_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "not able to register dai\n"); goto err_set_drvdata; @@ -434,7 +439,7 @@ 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); + snd_soc_unregister_component(&pdev->dev); dev_set_drvdata(&pdev->dev, NULL); clk_put(dev->clk); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3b98159d9645..3843a18d4e56 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -118,7 +118,7 @@ config SND_SOC_IMX_PCM_FIQ config SND_SOC_IMX_PCM_DMA bool - select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7decbd9b2340..0f0bed6def9e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -27,6 +27,7 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> #include "fsl_ssi.h" #include "imx-pcm.h" @@ -122,8 +123,10 @@ struct fsl_ssi_private { bool ssi_on_imx; struct clk *clk; struct platform_device *imx_pcm_pdev; - struct imx_pcm_dma_params dma_params_tx; - struct imx_pcm_dma_params dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct imx_dma_data filter_data_tx; + struct imx_dma_data filter_data_rx; struct { unsigned int rfrc; @@ -422,12 +425,6 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, ssi_private->second_stream = substream; } - if (ssi_private->ssi_on_imx) - snd_soc_dai_set_dma_data(dai, substream, - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - &ssi_private->dma_params_tx : - &ssi_private->dma_params_rx); - return 0; } @@ -549,6 +546,18 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, } } +static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai); + + if (ssi_private->ssi_on_imx) { + dai->playback_dma_data = &ssi_private->dma_params_tx; + dai->capture_dma_data = &ssi_private->dma_params_rx; + } + + return 0; +} + static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, @@ -558,6 +567,7 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { /* Template for the CPU dai driver structure */ static struct snd_soc_dai_driver fsl_ssi_dai_template = { + .probe = fsl_ssi_dai_probe, .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, @@ -574,6 +584,10 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .ops = &fsl_ssi_dai_ops, }; +static const struct snd_soc_component_driver fsl_ssi_component = { + .name = "fsl-ssi", +}; + /* Show the statistics of a flag only if its interrupt is enabled. The * compiler will optimze this code to a no-op if the interrupt is not * enabled. @@ -649,6 +663,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) const uint32_t *iprop; struct resource res; char name[64]; + bool shared; /* SSIs that are not connected on the board should have a * status = "disabled" @@ -737,14 +752,18 @@ static int fsl_ssi_probe(struct platform_device *pdev) * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). */ - ssi_private->dma_params_tx.burstsize = + ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - 2; - ssi_private->dma_params_rx.burstsize = + ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - 2; - ssi_private->dma_params_tx.dma_addr = + ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + offsetof(struct ccsr_ssi, stx0); - ssi_private->dma_params_rx.dma_addr = + ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + offsetof(struct ccsr_ssi, srx0); + ssi_private->dma_params_tx.filter_data = + &ssi_private->filter_data_tx; + ssi_private->dma_params_rx.filter_data = + &ssi_private->filter_data_rx; /* * TODO: This is a temporary solution and should be changed * to use generic DMA binding later when the helplers get in. @@ -755,14 +774,14 @@ static int fsl_ssi_probe(struct platform_device *pdev) dev_err(&pdev->dev, "could not get dma events\n"); goto error_clk; } - ssi_private->dma_params_tx.dma = dma_events[0]; - ssi_private->dma_params_rx.dma = dma_events[1]; - ssi_private->dma_params_tx.shared_peripheral = - of_device_is_compatible(of_get_parent(np), - "fsl,spba-bus"); - ssi_private->dma_params_rx.shared_peripheral = - ssi_private->dma_params_tx.shared_peripheral; + shared = of_device_is_compatible(of_get_parent(np), + "fsl,spba-bus"); + + imx_pcm_dma_params_init_data(&ssi_private->filter_data_tx, + dma_events[0], shared); + imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx, + dma_events[1], shared); } /* Initialize the the device_attribute structure */ @@ -782,7 +801,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Register with ASoC */ dev_set_drvdata(&pdev->dev, ssi_private); - ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv); + ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component, + &ssi_private->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); goto error_dev; @@ -835,7 +855,7 @@ done: error_dai: if (ssi_private->ssi_on_imx) platform_device_unregister(ssi_private->imx_pcm_pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); error_dev: dev_set_drvdata(&pdev->dev, NULL); @@ -873,7 +893,7 @@ static int fsl_ssi_remove(struct platform_device *pdev) clk_disable_unprepare(ssi_private->clk); clk_put(ssi_private->clk); } - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); free_irq(ssi_private->irq, ssi_private); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 217300029b5b..e6b9a69e2a68 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -196,5 +196,13 @@ struct ccsr_ssi { #define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT) #define CCSR_SSI_SOR_SYNRST 0x00000001 +#define CCSR_SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) +#define CCSR_SSI_SACNT_WR 0x00000010 +#define CCSR_SSI_SACNT_RD 0x00000008 +#define CCSR_SSI_SACNT_RDWR_MASK 0x00000018 +#define CCSR_SSI_SACNT_TIF 0x00000004 +#define CCSR_SSI_SACNT_FV 0x00000002 +#define CCSR_SSI_SACNT_AC97EN 0x00000001 + #endif diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 3f333e5b4673..47f046a8fdab 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -262,7 +262,7 @@ static int imx_audmux_probe(struct platform_device *pdev) return PTR_ERR(pinctrl); } - audmux_clk = clk_get(&pdev->dev, "audmux"); + audmux_clk = devm_clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", PTR_ERR(audmux_clk)); @@ -282,7 +282,6 @@ static int imx_audmux_remove(struct platform_device *pdev) { if (audmux_type == IMX31_AUDMUX) audmux_debugfs_remove(); - clk_put(audmux_clk); return 0; } diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 500f8ce55d78..c246fb514930 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -11,74 +11,30 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. */ -#include <linux/clk.h> -#include <linux/delay.h> -#include <linux/device.h> -#include <linux/dma-mapping.h> -#include <linux/init.h> -#include <linux/interrupt.h> -#include <linux/module.h> #include <linux/platform_device.h> -#include <linux/slab.h> #include <linux/dmaengine.h> #include <linux/types.h> #include <sound/core.h> -#include <sound/initval.h> #include <sound/pcm.h> -#include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include <linux/platform_data/dma-imx.h> - #include "imx-pcm.h" static bool filter(struct dma_chan *chan, void *param) { + struct snd_dmaengine_dai_dma_data *dma_data = param; + if (!imx_dma_is_general_purpose(chan)) return false; - chan->private = param; + chan->private = dma_data->filter_data; return true; } -static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - struct imx_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); - - ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); - if (ret) - return ret; - - slave_config.device_fc = false; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = dma_params->dma_addr; - slave_config.dst_maxburst = dma_params->burstsize; - } else { - slave_config.src_addr = dma_params->dma_addr; - slave_config.src_maxburst = dma_params->burstsize; - } - - ret = dmaengine_slave_config(chan, &slave_config); - if (ret) - return ret; - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static struct snd_pcm_hardware snd_imx_hardware = { +static const struct snd_pcm_hardware imx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -97,64 +53,22 @@ static struct snd_pcm_hardware snd_imx_hardware = { .fifo_size = 0, }; -static int snd_imx_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params; - struct imx_dma_data *dma_data; - int ret; - - snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); - - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); - if (!dma_data) - return -ENOMEM; - - dma_data->peripheral_type = dma_params->shared_peripheral ? - IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI; - dma_data->priority = DMA_PRIO_HIGH; - dma_data->dma_request = dma_params->dma; - - ret = snd_dmaengine_pcm_open(substream, filter, dma_data); - if (ret) { - kfree(dma_data); - return ret; - } - - snd_dmaengine_pcm_set_data(substream, dma_data); - - return 0; -} +static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { + .pcm_hardware = &imx_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .compat_filter_fn = filter, + .prealloc_buffer_size = IMX_SSI_DMABUF_SIZE, +}; -static int snd_imx_close(struct snd_pcm_substream *substream) +int imx_pcm_dma_init(struct platform_device *pdev) { - struct imx_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); - - snd_dmaengine_pcm_close(substream); - kfree(dma_data); - - return 0; + return snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT); } -static struct snd_pcm_ops imx_pcm_ops = { - .open = snd_imx_open, - .close = snd_imx_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_imx_pcm_hw_params, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer_no_residue, - .mmap = snd_imx_pcm_mmap, -}; - -static struct snd_soc_platform_driver imx_soc_platform_mx2 = { - .ops = &imx_pcm_ops, - .pcm_new = imx_pcm_new, - .pcm_free = imx_pcm_free, -}; - -int imx_pcm_dma_init(struct platform_device *pdev) +void imx_pcm_dma_exit(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2); + snd_dmaengine_pcm_unregister(&pdev->dev); } diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 920f945cb2f4..670b96b0ce2f 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -34,7 +34,7 @@ #include "imx-ssi.h" struct imx_pcm_runtime_data { - int period; + unsigned int period; int periods; unsigned long offset; unsigned long last_offset; @@ -299,8 +299,8 @@ int imx_pcm_fiq_init(struct platform_device *pdev) imx_ssi_fiq_base = (unsigned long)ssi->base; - ssi->dma_params_tx.burstsize = 4; - ssi->dma_params_rx.burstsize = 6; + ssi->dma_params_tx.maxburst = 4; + ssi->dma_params_rx.maxburst = 6; ret = snd_soc_register_platform(&pdev->dev, &imx_soc_platform_fiq); if (ret) diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index 0d0625bfcb65..c49896442d8e 100644 --- a/sound/soc/fsl/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c @@ -114,7 +114,11 @@ static int imx_pcm_probe(struct platform_device *pdev) static int imx_pcm_remove(struct platform_device *pdev) { - snd_soc_unregister_platform(&pdev->dev); + if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0) + snd_soc_unregister_platform(&pdev->dev); + else + imx_pcm_dma_exit(pdev); + return 0; } diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 5ae13a13a353..b7fa0d75c687 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -13,17 +13,24 @@ #ifndef _IMX_PCM_H #define _IMX_PCM_H +#include <linux/platform_data/dma-imx.h> + /* * Do not change this as the FIQ handler depends on this size */ #define IMX_SSI_DMABUF_SIZE (64 * 1024) -struct imx_pcm_dma_params { - int dma; - unsigned long dma_addr; - int burstsize; - bool shared_peripheral; /* The peripheral is on SPBA bus */ -}; +static inline void +imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, + int dma, bool shared) +{ + dma_data->dma_request = dma; + dma_data->priority = DMA_PRIO_HIGH; + if (shared) + dma_data->peripheral_type = IMX_DMATYPE_SSI_SP; + else + dma_data->peripheral_type = IMX_DMATYPE_SSI; +} int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); @@ -32,11 +39,16 @@ void imx_pcm_free(struct snd_pcm *pcm); #ifdef CONFIG_SND_SOC_IMX_PCM_DMA int imx_pcm_dma_init(struct platform_device *pdev); +void imx_pcm_dma_exit(struct platform_device *pdev); #else static inline int imx_pcm_dma_init(struct platform_device *pdev) { return -ENODEV; } + +static inline void imx_pcm_dma_exit(struct platform_device *pdev) +{ +} #endif #ifdef CONFIG_SND_SOC_IMX_PCM_FIQ diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 424347e9b2d7..9584e78858df 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -148,7 +148,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->dai.stream_name = "HiFi"; data->dai.codec_dai_name = "sgtl5000"; data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.cpu_of_node = ssi_np; data->dai.platform_name = "imx-pcm-audio"; data->dai.init = &imx_sgtl5000_dai_init; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 810c7eeb7b03..902fab02b851 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -232,23 +232,6 @@ static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, return 0; } -static int imx_ssi_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); - struct imx_pcm_dma_params *dma_data; - - /* Tx/Rx config */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &ssi->dma_params_tx; - else - dma_data = &ssi->dma_params_rx; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - /* * Should only be called when port is inactive (i.e. SSIEN = 0), * although can be called multiple times by upper layers. @@ -353,7 +336,6 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, } static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { - .startup = imx_ssi_startup, .hw_params = imx_ssi_hw_params, .set_fmt = imx_ssi_set_dai_fmt, .set_clkdiv = imx_ssi_set_dai_clkdiv, @@ -369,10 +351,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) snd_soc_dai_set_drvdata(dai, ssi); - val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) | - SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); + val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.maxburst) | + SSI_SFCSR_RFWM0(ssi->dma_params_rx.maxburst); writel(val, ssi->base + SSI_SFCSR); + /* Tx/Rx config */ + dai->playback_dma_data = &ssi->dma_params_tx; + dai->capture_dma_data = &ssi->dma_params_rx; + return 0; } @@ -400,7 +386,7 @@ static struct snd_soc_dai_driver imx_ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, + .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { @@ -413,6 +399,10 @@ static struct snd_soc_dai_driver imx_ac97_dai = { .ops = &imx_ssi_pcm_dai_ops, }; +static const struct snd_soc_component_driver imx_component = { + .name = DRV_NAME, +}; + static void setup_channel_to_ac97(struct imx_ssi *imx_ssi) { void __iomem *base = imx_ssi->base; @@ -575,23 +565,31 @@ static int imx_ssi_probe(struct platform_device *pdev) writel(0x0, ssi->base + SSI_SIER); - ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0; - ssi->dma_params_tx.dma_addr = res->start + SSI_STX0; + ssi->dma_params_rx.addr = res->start + SSI_SRX0; + ssi->dma_params_tx.addr = res->start + SSI_STX0; + + ssi->dma_params_tx.maxburst = 6; + ssi->dma_params_rx.maxburst = 4; - ssi->dma_params_tx.burstsize = 6; - ssi->dma_params_rx.burstsize = 4; + ssi->dma_params_tx.filter_data = &ssi->filter_data_tx; + ssi->dma_params_rx.filter_data = &ssi->filter_data_rx; res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); - if (res) - ssi->dma_params_tx.dma = res->start; + if (res) { + imx_pcm_dma_params_init_data(&ssi->filter_data_tx, res->start, + false); + } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); - if (res) - ssi->dma_params_rx.dma = res->start; + if (res) { + imx_pcm_dma_params_init_data(&ssi->filter_data_rx, res->start, + false); + } platform_set_drvdata(pdev, ssi); - ret = snd_soc_register_dai(&pdev->dev, dai); + ret = snd_soc_register_component(&pdev->dev, &imx_component, + dai, 1); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); goto failed_register; @@ -632,7 +630,7 @@ failed_pdev_alloc: failed_pdev_fiq_add: platform_device_put(ssi->soc_platform_pdev_fiq); failed_pdev_fiq_alloc: - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); failed_register: release_mem_region(res->start, resource_size(res)); failed_get_resource: @@ -650,7 +648,7 @@ static int imx_ssi_remove(struct platform_device *pdev) platform_device_unregister(ssi->soc_platform_pdev); platform_device_unregister(ssi->soc_platform_pdev_fiq); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index dc114bdedce5..bb6b3dbb13fd 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -187,6 +187,7 @@ #include <linux/dmaengine.h> #include <linux/platform_data/dma-imx.h> +#include <sound/dmaengine_pcm.h> #include "imx-pcm.h" struct imx_ssi { @@ -204,8 +205,10 @@ struct imx_ssi { void (*ac97_reset) (struct snd_ac97 *ac97); void (*ac97_warm_reset)(struct snd_ac97 *ac97); - struct imx_pcm_dma_params dma_params_rx; - struct imx_pcm_dma_params dma_params_tx; + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct imx_dma_data filter_data_tx; + struct imx_dma_data filter_data_rx; int enabled; diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index a4aec0488dd3..4141b35ef0bb 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -270,6 +270,9 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = { .ops = &psc_ac97_digital_ops, } }; +static const struct snd_soc_component_driver psc_ac97_component = { + .name = DRV_NAME, +}; /* --------------------------------------------------------------------- @@ -287,7 +290,8 @@ static int psc_ac97_of_probe(struct platform_device *op) if (rc != 0) return rc; - rc = snd_soc_register_dais(&op->dev, psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + rc = snd_soc_register_component(&op->dev, &psc_ac97_component, + psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); if (rc != 0) { dev_err(&op->dev, "Failed to register DAI\n"); return rc; @@ -313,7 +317,7 @@ static int psc_ac97_of_probe(struct platform_device *op) static int psc_ac97_of_remove(struct platform_device *op) { mpc5200_audio_dma_destroy(op); - snd_soc_unregister_dais(&op->dev, ARRAY_SIZE(psc_ac97_dai)); + snd_soc_unregister_component(&op->dev); return 0; } diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index b95b966f25a0..f4efaadb80a2 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -148,6 +148,10 @@ static struct snd_soc_dai_driver psc_i2s_dai[] = {{ .ops = &psc_i2s_dai_ops, } }; +static const struct snd_soc_component_driver psc_i2s_component = { + .name = "mpc5200-i2s", +}; + /* --------------------------------------------------------------------- * OF platform bus binding code: * - Probe/remove operations @@ -163,7 +167,8 @@ static int psc_i2s_of_probe(struct platform_device *op) if (rc != 0) return rc; - rc = snd_soc_register_dais(&op->dev, psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai)); + rc = snd_soc_register_component(&op->dev, &psc_i2s_component, + psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai)); if (rc != 0) { pr_err("Failed to register DAI\n"); return rc; @@ -208,7 +213,7 @@ static int psc_i2s_of_probe(struct platform_device *op) static int psc_i2s_of_remove(struct platform_device *op) { mpc5200_audio_dma_destroy(op); - snd_soc_unregister_dais(&op->dev, ARRAY_SIZE(psc_i2s_dai)); + snd_soc_unregister_component(&op->dev); return 0; } diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 6cef491f4823..9a126441c5f3 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -425,6 +425,10 @@ static struct snd_soc_dai_driver jz4740_i2s_dai = { .resume = jz4740_i2s_resume, }; +static const struct snd_soc_component_driver jz4740_i2s_component = { + .name = "jz4740-i2s", +}; + static int jz4740_i2s_dev_probe(struct platform_device *pdev) { struct jz4740_i2s *i2s; @@ -469,7 +473,8 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) } platform_set_drvdata(pdev, i2s); - ret = snd_soc_register_dai(&pdev->dev, &jz4740_i2s_dai); + ret = snd_soc_register_component(&pdev->dev, &jz4740_i2s_component, + &jz4740_i2s_dai, 1); if (ret) { dev_err(&pdev->dev, "Failed to register DAI\n"); @@ -496,7 +501,7 @@ static int jz4740_i2s_dev_remove(struct platform_device *pdev) { struct jz4740_i2s *i2s = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); clk_put(i2s->clk_i2s); clk_put(i2s->clk_aic); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index c74c89065493..befe68f59285 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -451,6 +451,10 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .ops = &kirkwood_i2s_dai_ops, }; +static const struct snd_soc_component_driver kirkwood_i2s_component = { + .name = DRV_NAME, +}; + static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; @@ -524,10 +528,11 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_128; } - err = snd_soc_register_dai(&pdev->dev, soc_dai); + err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, + soc_dai, 1); if (!err) return 0; - dev_err(&pdev->dev, "snd_soc_register_dai failed\n"); + dev_err(&pdev->dev, "snd_soc_register_component failed\n"); if (!IS_ERR(priv->extclk)) { clk_disable_unprepare(priv->extclk); @@ -542,7 +547,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) { struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); if (!IS_ERR(priv->extclk)) { clk_disable_unprepare(priv->extclk); diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index a263cbed8624..392fc0b8f5b8 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -1,7 +1,7 @@ /* * sst_platform.c - Intel MID Platform driver * - * Copyright (C) 2010-2012 Intel Corp + * Copyright (C) 2010-2013 Intel Corp * Author: Vinod Koul <vinod.koul@intel.com> * Author: Harsha Priya <priya.harsha@intel.com> * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -165,6 +165,10 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { }, }; +static const struct snd_soc_component_driver sst_component = { + .name = "sst", +}; + /* helper functions */ static inline void sst_set_stream_status(struct sst_runtime_stream *stream, int state) @@ -652,11 +656,21 @@ static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, return stream->compr_ops->get_codec_caps(codec); } +static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->set_metadata(stream->id, metadata); +} + static struct snd_compr_ops sst_platform_compr_ops = { .open = sst_platform_compr_open, .free = sst_platform_compr_free, .set_params = sst_platform_compr_set_params, + .set_metadata = sst_platform_compr_set_metadata, .trigger = sst_platform_compr_trigger, .pointer = sst_platform_compr_pointer, .ack = sst_platform_compr_ack, @@ -683,7 +697,7 @@ static int sst_platform_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_register_dais(&pdev->dev, + ret = snd_soc_register_component(&pdev->dev, &sst_component, sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); if (ret) { pr_err("registering cpu dais failed\n"); @@ -695,7 +709,7 @@ static int sst_platform_probe(struct platform_device *pdev) static int sst_platform_remove(struct platform_device *pdev) { - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai)); + snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); pr_debug("sst_platform_remove success\n"); return 0; diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h index d61c5d514ffa..cacc9066ec52 100644 --- a/sound/soc/mid-x86/sst_platform.h +++ b/sound/soc/mid-x86/sst_platform.h @@ -124,6 +124,8 @@ struct compress_sst_ops { int (*close) (unsigned int str_id); int (*get_caps) (struct snd_compr_caps *caps); int (*get_codec_caps) (struct snd_compr_codec_caps *codec); + int (*set_metadata) (unsigned int str_id, + struct snd_compr_metadata *mdata); }; diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index b6fa77678d97..78d321cbe8b4 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,7 +1,7 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" depends on ARCH_MXS - select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the MXS SAIF interface. diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 564b5b60319d..b41fffc056fb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -18,38 +18,24 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include <linux/clk.h> -#include <linux/delay.h> #include <linux/device.h> -#include <linux/dma-mapping.h> #include <linux/init.h> -#include <linux/interrupt.h> #include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dmaengine.h> -#include <linux/fsl/mxs-dma.h> #include <sound/core.h> -#include <sound/initval.h> #include <sound/pcm.h> -#include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> #include "mxs-pcm.h" -struct mxs_pcm_dma_data { - struct mxs_dma_data dma_data; - struct mxs_pcm_dma_params *dma_params; -}; - -static struct snd_pcm_hardware snd_mxs_hardware = { +static const struct snd_pcm_hardware snd_mxs_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_INTERLEAVED, + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, @@ -61,13 +47,11 @@ static struct snd_pcm_hardware snd_mxs_hardware = { .periods_max = 52, .buffer_bytes_max = 64 * 1024, .fifo_size = 32, - }; static bool filter(struct dma_chan *chan, void *param) { - struct mxs_pcm_dma_data *pcm_dma_data = param; - struct mxs_pcm_dma_params *dma_params = pcm_dma_data->dma_params; + struct mxs_pcm_dma_params *dma_params = param; if (!mxs_dma_is_apbx(chan)) return false; @@ -75,160 +59,30 @@ static bool filter(struct dma_chan *chan, void *param) if (chan->chan_id != dma_params->chan_num) return false; - chan->private = &pcm_dma_data->dma_data; + chan->private = &dma_params->dma_data; return true; } -static int snd_mxs_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 snd_mxs_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mxs_pcm_dma_data *pcm_dma_data; - int ret; - - pcm_dma_data = kzalloc(sizeof(*pcm_dma_data), GFP_KERNEL); - if (pcm_dma_data == NULL) - return -ENOMEM; - - pcm_dma_data->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - pcm_dma_data->dma_data.chan_irq = pcm_dma_data->dma_params->chan_irq; - - ret = snd_dmaengine_pcm_open(substream, filter, pcm_dma_data); - if (ret) { - kfree(pcm_dma_data); - return ret; - } - - snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); - - snd_dmaengine_pcm_set_data(substream, pcm_dma_data); - - return 0; -} - -static int snd_mxs_close(struct snd_pcm_substream *substream) -{ - struct mxs_pcm_dma_data *pcm_dma_data = snd_dmaengine_pcm_get_data(substream); - - snd_dmaengine_pcm_close(substream); - kfree(pcm_dma_data); - - return 0; -} - -static int snd_mxs_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 mxs_pcm_ops = { - .open = snd_mxs_open, - .close = snd_mxs_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_mxs_pcm_hw_params, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer_no_residue, - .mmap = snd_mxs_pcm_mmap, -}; - -static int mxs_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = snd_mxs_hardware.buffer_bytes_max; - - 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; - buf->bytes = size; - - return 0; -} - -static u64 mxs_pcm_dmamask = DMA_BIT_MASK(32); -static int mxs_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &mxs_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = mxs_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = mxs_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - -out: - return ret; -} - -static void mxs_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->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static struct snd_soc_platform_driver mxs_soc_platform = { - .ops = &mxs_pcm_ops, - .pcm_new = mxs_pcm_new, - .pcm_free = mxs_pcm_free, +static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { + .pcm_hardware = &snd_mxs_hardware, + .compat_filter_fn = filter, + .prealloc_buffer_size = 64 * 1024, }; int mxs_pcm_platform_register(struct device *dev) { - return snd_soc_register_platform(dev, &mxs_soc_platform); + return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT | + SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); void mxs_pcm_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(dev); + snd_dmaengine_pcm_unregister(dev); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 35ba2ca42384..3aa918f9ed3e 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,8 +19,10 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H +#include <linux/fsl/mxs-dma.h> + struct mxs_pcm_dma_params { - int chan_irq; + struct mxs_dma_data dma_data; int chan_num; }; diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 41a6136e3535..d31dc52fa862 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -370,7 +370,6 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); - snd_soc_dai_set_dma_data(cpu_dai, substream, &saif->dma_param); /* clear error status to 0 for each re-open */ saif->fifo_underrun = 0; @@ -606,6 +605,8 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) struct mxs_saif *saif = dev_get_drvdata(dai->dev); snd_soc_dai_set_drvdata(dai, saif); + dai->playback_dma_data = &saif->dma_param; + dai->capture_dma_data = &saif->dma_param; return 0; } @@ -628,6 +629,10 @@ static struct snd_soc_dai_driver mxs_saif_dai = { .ops = &mxs_saif_dai_ops, }; +static const struct snd_soc_component_driver mxs_saif_component = { + .name = "mxs-saif", +}; + static irqreturn_t mxs_saif_irq(int irq, void *dev_id) { struct mxs_saif *saif = dev_id; @@ -754,9 +759,9 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.chan_irq < 0) { - ret = saif->dma_param.chan_irq; + saif->dma_param.dma_data.chan_irq = platform_get_irq(pdev, 1); + if (saif->dma_param.dma_data.chan_irq < 0) { + ret = saif->dma_param.dma_data.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); return ret; @@ -764,7 +769,8 @@ static int mxs_saif_probe(struct platform_device *pdev) platform_set_drvdata(pdev, saif); - ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); + ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, + &mxs_saif_dai, 1); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); return ret; @@ -779,7 +785,7 @@ static int mxs_saif_probe(struct platform_device *pdev) return 0; failed_pdev_alloc: - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return ret; } @@ -787,7 +793,7 @@ failed_pdev_alloc: static int mxs_saif_remove(struct platform_device *pdev) { mxs_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 0418467a4848..fe3285ceaf5b 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -314,6 +314,10 @@ static struct snd_soc_dai_driver nuc900_ac97_dai = { .ops = &nuc900_ac97_dai_ops, }; +static const struct snd_soc_component_driver nuc900_ac97_component = { + .name = "nuc900-ac97", +}; + static int nuc900_ac97_drvprobe(struct platform_device *pdev) { struct nuc900_audio *nuc900_audio; @@ -361,7 +365,8 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) nuc900_ac97_data = nuc900_audio; - ret = snd_soc_register_dai(&pdev->dev, &nuc900_ac97_dai); + ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, + &nuc900_ac97_dai, 1); if (ret) goto out3; @@ -384,7 +389,7 @@ out0: static int nuc900_ac97_drvremove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); clk_put(nuc900_ac97_data->clk); iounmap(nuc900_ac97_data->mmio); diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index c1900b2a6f28..994dcf345975 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -28,7 +28,6 @@ #include <linux/platform_data/asoc-ti-mcbsp.h> #include "omap-mcbsp.h" -#include "omap-pcm.h" #include "../codecs/tlv320aic23.h" diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 2600447fa74f..629446482a91 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -36,7 +36,6 @@ #include <linux/platform_data/asoc-ti-mcbsp.h> #include "omap-mcbsp.h" -#include "omap-pcm.h" #include "../codecs/cx20442.h" diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 285c8368cb47..eb68c7db1cf3 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -1018,9 +1018,10 @@ int omap_mcbsp_init(struct platform_device *pdev) return -ENODEV; } /* RX DMA request number, and port address configuration */ - mcbsp->dma_data[1].name = "Audio Capture"; - mcbsp->dma_data[1].dma_req = res->start; - mcbsp->dma_data[1].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 1); + mcbsp->dma_req[1] = res->start; + mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; + mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1); + mcbsp->dma_data[1].maxburst = 4; res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); if (!res) { @@ -1028,9 +1029,10 @@ int omap_mcbsp_init(struct platform_device *pdev) return -ENODEV; } /* TX DMA request number, and port address configuration */ - mcbsp->dma_data[0].name = "Audio Playback"; - mcbsp->dma_data[0].dma_req = res->start; - mcbsp->dma_data[0].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 0); + mcbsp->dma_req[0] = res->start; + mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; + mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, 0); + mcbsp->dma_data[0].maxburst = 4; mcbsp->fclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index f93e0b0af303..96d1b086bcf8 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -24,14 +24,14 @@ #ifndef __ASOC_MCBSP_H #define __ASOC_MCBSP_H -#include "omap-pcm.h" - #ifdef CONFIG_ARCH_OMAP1 #define mcbsp_omap1() 1 #else #define mcbsp_omap1() 0 #endif +#include <sound/dmaengine_pcm.h> + /* McBSP register numbers. Register address offset = num * reg_step */ enum { /* Common registers */ @@ -312,7 +312,8 @@ struct omap_mcbsp { struct omap_mcbsp_platform_data *pdata; struct omap_mcbsp_st_data *st_data; struct omap_mcbsp_reg_cfg cfg_regs; - struct omap_pcm_dma_data dma_data[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + unsigned int dma_req[2]; int dma_op_mode; u16 max_tx_thres; u16 max_rx_thres; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index ee7cd53aa3ee..5e8d640d314f 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -34,7 +34,6 @@ #include <linux/platform_data/asoc-ti-mcbsp.h> #include "omap-mcbsp.h" -#include "omap-pcm.h" #define N810_HEADSET_AMP_GPIO 10 #define N810_SPEAKER_AMP_GPIO 101 diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index e7d93fa412a9..70cd5c7b2e14 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -34,7 +34,6 @@ #include "omap-dmic.h" #include "omap-mcpdm.h" -#include "omap-pcm.h" #include "../codecs/twl6040.h" struct abe_twl6040 { diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index ba49ccd9eed9..2ad0370146fd 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -39,8 +39,8 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> -#include "omap-pcm.h" #include "omap-dmic.h" struct omap_dmic { @@ -55,13 +55,9 @@ struct omap_dmic { u32 ch_enabled; bool active; struct mutex mutex; -}; -/* - * Stream DMA parameters - */ -static struct omap_pcm_dma_data omap_dmic_dai_dma_params = { - .name = "DMIC capture", + struct snd_dmaengine_dai_dma_data dma_data; + unsigned int dma_req; }; static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val) @@ -118,7 +114,7 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream, mutex_unlock(&dmic->mutex); - snd_soc_dai_set_dma_data(dai, substream, &omap_dmic_dai_dma_params); + snd_soc_dai_set_dma_data(dai, substream, &dmic->dma_data); return ret; } @@ -203,7 +199,7 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); - struct omap_pcm_dma_data *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; int channels; dmic->clk_div = omap_dmic_select_divider(dmic, params_rate(params)); @@ -230,7 +226,7 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, /* packet size is threshold * channels */ dma_data = snd_soc_dai_get_dma_data(dai, substream); - dma_data->packet_size = dmic->threshold * channels; + dma_data->maxburst = dmic->threshold * channels; return 0; } @@ -448,6 +444,10 @@ static struct snd_soc_dai_driver omap_dmic_dai = { .ops = &omap_dmic_dai_ops, }; +static const struct snd_soc_component_driver omap_dmic_component = { + .name = "omap-dmic", +}; + static int asoc_dmic_probe(struct platform_device *pdev) { struct omap_dmic *dmic; @@ -476,7 +476,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) ret = -ENODEV; goto err_put_clk; } - omap_dmic_dai_dma_params.port_addr = res->start + OMAP_DMIC_DATA_REG; + dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!res) { @@ -484,7 +484,9 @@ static int asoc_dmic_probe(struct platform_device *pdev) ret = -ENODEV; goto err_put_clk; } - omap_dmic_dai_dma_params.dma_req = res->start; + + dmic->dma_req = res->start; + dmic->dma_data.filter_data = &dmic->dma_req; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!res) { @@ -493,21 +495,12 @@ static int asoc_dmic_probe(struct platform_device *pdev) goto err_put_clk; } - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(dmic->dev, "memory region already claimed\n"); - ret = -ENODEV; - goto err_put_clk; - } - - dmic->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!dmic->io_base) { - ret = -ENOMEM; - goto err_put_clk; - } + dmic->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(dmic->io_base)) + return PTR_ERR(dmic->io_base); - ret = snd_soc_register_dai(&pdev->dev, &omap_dmic_dai); + ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component, + &omap_dmic_dai, 1); if (ret) goto err_put_clk; @@ -522,7 +515,7 @@ static int asoc_dmic_remove(struct platform_device *pdev) { struct omap_dmic *dmic = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); clk_put(dmic->fclk); return 0; diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index 32fa840c493e..ced3b88b44d4 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -32,15 +32,16 @@ #include <sound/soc.h> #include <sound/asound.h> #include <sound/asoundef.h> +#include <sound/dmaengine_pcm.h> #include <video/omapdss.h> -#include "omap-pcm.h" #include "omap-hdmi.h" #define DRV_NAME "omap-hdmi-audio-dai" struct hdmi_priv { - struct omap_pcm_dma_data dma_params; + struct snd_dmaengine_dai_dma_data dma_data; + unsigned int dma_req; struct omap_dss_audio dss_audio; struct snd_aes_iec958 iec; struct snd_cea_861_aud_if cea; @@ -68,7 +69,7 @@ static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream, return -ENODEV; } - snd_soc_dai_set_dma_data(dai, substream, &priv->dma_params); + snd_soc_dai_set_dma_data(dai, substream, &priv->dma_data); return 0; } @@ -88,25 +89,20 @@ static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream, struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai); struct snd_aes_iec958 *iec = &priv->iec; struct snd_cea_861_aud_if *cea = &priv->cea; - struct omap_pcm_dma_data *dma_data; int err = 0; - dma_data = snd_soc_dai_get_dma_data(dai, substream); - switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - dma_data->packet_size = 16; + priv->dma_data.maxburst = 16; break; case SNDRV_PCM_FORMAT_S24_LE: - dma_data->packet_size = 32; + priv->dma_data.maxburst = 32; break; default: dev_err(dai->dev, "format not supported!\n"); return -EINVAL; } - dma_data->data_type = 32; - /* * fill the IEC-60958 channel status word */ @@ -264,6 +260,10 @@ static struct snd_soc_dai_driver omap_hdmi_dai = { .ops = &omap_hdmi_dai_ops, }; +static const struct snd_soc_component_driver omap_hdmi_component = { + .name = DRV_NAME, +}; + static int omap_hdmi_probe(struct platform_device *pdev) { int ret; @@ -283,8 +283,7 @@ static int omap_hdmi_probe(struct platform_device *pdev) return -ENODEV; } - hdmi_data->dma_params.port_addr = hdmi_rsrc->start - + OMAP_HDMI_AUDIO_DMA_PORT; + hdmi_data->dma_data.addr = hdmi_rsrc->start + OMAP_HDMI_AUDIO_DMA_PORT; hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!hdmi_rsrc) { @@ -292,8 +291,9 @@ static int omap_hdmi_probe(struct platform_device *pdev) return -ENODEV; } - hdmi_data->dma_params.dma_req = hdmi_rsrc->start; - hdmi_data->dma_params.name = "HDMI playback"; + hdmi_data->dma_req = hdmi_rsrc->start; + hdmi_data->dma_data.filter_data = &hdmi_data->dma_req; + hdmi_data->dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; /* * TODO: We assume that there is only one DSS HDMI device. Future @@ -321,7 +321,8 @@ static int omap_hdmi_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, hdmi_data); - ret = snd_soc_register_dai(&pdev->dev, &omap_hdmi_dai); + ret = snd_soc_register_component(&pdev->dev, &omap_hdmi_component, + &omap_hdmi_dai, 1); return ret; } @@ -330,7 +331,7 @@ static int omap_hdmi_remove(struct platform_device *pdev) { struct hdmi_priv *hdmi_data = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); if (hdmi_data == NULL) { dev_err(&pdev->dev, "cannot obtain HDMi data\n"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8d2defd6fdbe..eadbfb6b5000 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -33,11 +33,11 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> #include <linux/platform_data/asoc-ti-mcbsp.h> #include "mcbsp.h" #include "omap-mcbsp.h" -#include "omap-pcm.h" #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) @@ -62,24 +62,22 @@ enum { * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs */ -static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) +static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream, + unsigned int packet_size) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_pcm_dma_data *dma_data; int words; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - /* * Configure McBSP threshold based on either: * packet_size, when the sDMA is in packet mode, or based on the * period size in THRESHOLD mode, otherwise use McBSP threshold = 1 * for mono streams. */ - if (dma_data->packet_size) - words = dma_data->packet_size; + if (packet_size) + words = packet_size; else words = 1; @@ -226,7 +224,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; - struct omap_pcm_dma_data *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; int wlen, channels, wpf; int pkt_size = 0; unsigned int format, div, framesize, master; @@ -245,7 +243,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } if (mcbsp->pdata->buffer_size) { - dma_data->set_threshold = omap_mcbsp_set_threshold; if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; int divider = 0; @@ -276,9 +273,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Use packet mode for non mono streams */ pkt_size = channels; } + omap_mcbsp_set_threshold(substream, pkt_size); } - dma_data->packet_size = pkt_size; + dma_data->maxburst = pkt_size; if (mcbsp->configured) { /* McBSP already configured by another stream */ @@ -586,6 +584,10 @@ static struct snd_soc_dai_driver omap_mcbsp_dai = { .ops = &mcbsp_dai_ops, }; +static const struct snd_soc_component_driver omap_mcbsp_component = { + .name = "omap-mcbsp", +}; + static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -793,7 +795,8 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) ret = omap_mcbsp_init(pdev); if (!ret) - return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai); + return snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, + &omap_mcbsp_dai, 1); return ret; } @@ -802,7 +805,7 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) { struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 5ca11bdac21e..eb05c7ed6d05 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -39,11 +39,14 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> #include "omap-mcpdm.h" -#include "omap-pcm.h" -#define OMAP44XX_MCPDM_L3_BASE 0x49032000 +struct mcpdm_link_config { + u32 link_mask; /* channel mask for the direction */ + u32 threshold; /* FIFO threshold */ +}; struct omap_mcpdm { struct device *dev; @@ -53,29 +56,22 @@ struct omap_mcpdm { struct mutex mutex; - /* channel data */ - u32 dn_channels; - u32 up_channels; - - /* McPDM FIFO thresholds */ - u32 dn_threshold; - u32 up_threshold; + /* Playback/Capture configuration */ + struct mcpdm_link_config config[2]; /* McPDM dn offsets for rx1, and 2 channels */ u32 dn_rx_offset; + + /* McPDM needs to be restarted due to runtime reconfiguration */ + bool restart; + + struct snd_dmaengine_dai_dma_data dma_data[2]; + unsigned int dma_req[2]; }; /* * Stream DMA parameters */ -static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { - { - .name = "Audio playback", - }, - { - .name = "Audio capture", - }, -}; static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val) { @@ -130,11 +126,12 @@ static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) {} static void omap_mcpdm_start(struct omap_mcpdm *mcpdm) { u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + u32 link_mask = mcpdm->config[0].link_mask | mcpdm->config[1].link_mask; ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); - ctrl |= mcpdm->dn_channels | mcpdm->up_channels; + ctrl |= link_mask; omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); @@ -148,11 +145,12 @@ static void omap_mcpdm_start(struct omap_mcpdm *mcpdm) static void omap_mcpdm_stop(struct omap_mcpdm *mcpdm) { u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + u32 link_mask = MCPDM_PDM_DN_MASK | MCPDM_PDM_UP_MASK; ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); - ctrl &= ~(mcpdm->dn_channels | mcpdm->up_channels); + ctrl &= ~(link_mask); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); @@ -188,8 +186,10 @@ static void omap_mcpdm_open_streams(struct omap_mcpdm *mcpdm) omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset); } - omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN, mcpdm->dn_threshold); - omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP, mcpdm->up_threshold); + omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN, + mcpdm->config[SNDRV_PCM_STREAM_PLAYBACK].threshold); + omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP, + mcpdm->config[SNDRV_PCM_STREAM_CAPTURE].threshold); omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_SET, MCPDM_DMA_DN_ENABLE | MCPDM_DMA_UP_ENABLE); @@ -267,7 +267,7 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, mutex_unlock(&mcpdm->mutex); snd_soc_dai_set_dma_data(dai, substream, - &omap_mcpdm_dai_dma_params[substream->stream]); + &mcpdm->dma_data[substream->stream]); return 0; } @@ -283,6 +283,8 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, if (omap_mcpdm_active(mcpdm)) { omap_mcpdm_stop(mcpdm); omap_mcpdm_close_streams(mcpdm); + mcpdm->config[0].link_mask = 0; + mcpdm->config[1].link_mask = 0; } } @@ -295,7 +297,8 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int stream = substream->stream; - struct omap_pcm_dma_data *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; + u32 threshold; int channels; int link_mask = 0; @@ -325,16 +328,32 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, dma_data = snd_soc_dai_get_dma_data(dai, substream); + threshold = mcpdm->config[stream].threshold; /* Configure McPDM channels, and DMA packet size */ if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - mcpdm->dn_channels = link_mask << 3; - dma_data->packet_size = - (MCPDM_DN_THRES_MAX - mcpdm->dn_threshold) * channels; + link_mask <<= 3; + + /* If capture is not running assume a stereo stream to come */ + if (!mcpdm->config[!stream].link_mask) + mcpdm->config[!stream].link_mask = 0x3; + + dma_data->maxburst = + (MCPDM_DN_THRES_MAX - threshold) * channels; } else { - mcpdm->up_channels = link_mask << 0; - dma_data->packet_size = mcpdm->up_threshold * channels; + /* If playback is not running assume a stereo stream to come */ + if (!mcpdm->config[!stream].link_mask) + mcpdm->config[!stream].link_mask = (0x3 << 3); + + dma_data->maxburst = threshold * channels; } + /* Check if we need to restart McPDM with this stream */ + if (mcpdm->config[stream].link_mask && + mcpdm->config[stream].link_mask != link_mask) + mcpdm->restart = true; + + mcpdm->config[stream].link_mask = link_mask; + return 0; } @@ -346,6 +365,11 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, if (!omap_mcpdm_active(mcpdm)) { omap_mcpdm_start(mcpdm); omap_mcpdm_reg_dump(mcpdm); + } else if (mcpdm->restart) { + omap_mcpdm_stop(mcpdm); + omap_mcpdm_start(mcpdm); + mcpdm->restart = false; + omap_mcpdm_reg_dump(mcpdm); } return 0; @@ -369,7 +393,7 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(mcpdm->dev); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM", (void *)mcpdm); pm_runtime_put_sync(mcpdm->dev); @@ -380,8 +404,9 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) } /* Configure McPDM threshold values */ - mcpdm->dn_threshold = 2; - mcpdm->up_threshold = MCPDM_UP_THRES_MAX - 3; + mcpdm->config[SNDRV_PCM_STREAM_PLAYBACK].threshold = 2; + mcpdm->config[SNDRV_PCM_STREAM_CAPTURE].threshold = + MCPDM_UP_THRES_MAX - 3; return ret; } @@ -389,7 +414,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); return 0; @@ -420,6 +444,10 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = { .ops = &omap_mcpdm_dai_ops, }; +static const struct snd_soc_component_driver omap_mcpdm_component = { + .name = "omap-mcpdm", +}; + void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd, u8 rx1, u8 rx2) { @@ -446,33 +474,30 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) if (res == NULL) return -ENOMEM; - omap_mcpdm_dai_dma_params[0].port_addr = res->start + MCPDM_REG_DN_DATA; - omap_mcpdm_dai_dma_params[1].port_addr = res->start + MCPDM_REG_UP_DATA; + mcpdm->dma_data[0].addr = res->start + MCPDM_REG_DN_DATA; + mcpdm->dma_data[1].addr = res->start + MCPDM_REG_UP_DATA; res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link"); if (!res) return -ENODEV; - omap_mcpdm_dai_dma_params[0].dma_req = res->start; + mcpdm->dma_req[0] = res->start; + mcpdm->dma_data[0].filter_data = &mcpdm->dma_req[0]; res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "up_link"); if (!res) return -ENODEV; - omap_mcpdm_dai_dma_params[1].dma_req = res->start; + mcpdm->dma_req[1] = res->start; + mcpdm->dma_data[1].filter_data = &mcpdm->dma_req[1]; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (res == NULL) return -ENOMEM; - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), "McPDM")) - return -EBUSY; - - mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!mcpdm->io_base) - return -ENOMEM; + mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(mcpdm->io_base)) + return PTR_ERR(mcpdm->io_base); mcpdm->irq = platform_get_irq(pdev, 0); if (mcpdm->irq < 0) @@ -480,12 +505,13 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; - return snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); + return snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, + &omap_mcpdm_dai, 1); } static int asoc_mcpdm_remove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index c722c2ef9665..c28e042f2208 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -32,8 +32,6 @@ #include <sound/dmaengine_pcm.h> #include <sound/soc.h> -#include "omap-pcm.h" - #ifdef CONFIG_ARCH_OMAP1 #define pcm_omap1510() cpu_is_omap1510() #else @@ -56,25 +54,6 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { .buffer_bytes_max = 128 * 1024, }; -static int omap_pcm_get_dma_buswidth(int num_bits) -{ - int buswidth; - - switch (num_bits) { - case 16: - buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; - break; - case 32: - buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - break; - default: - buswidth = -EINVAL; - break; - } - return buswidth; -} - - /* this may get called several times by oss emulation */ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -105,20 +84,9 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, if (err) return err; - /* Override the *_dma addr_width if requested by the DAI driver */ - if (dma_data->data_type) { - int buswidth = omap_pcm_get_dma_buswidth(dma_data->data_type); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - config.dst_addr_width = buswidth; - else - config.src_addr_width = buswidth; - } - - config.src_addr = dma_data->port_addr; - config.dst_addr = dma_data->port_addr; - config.src_maxburst = dma_data->packet_size; - config.dst_maxburst = dma_data->packet_size; + snd_dmaengine_pcm_set_config_from_dai_data(substream, + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream), + &config); return dmaengine_slave_config(chan, &config); } @@ -129,37 +97,6 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct omap_pcm_dma_data *dma_data; - int ret = 0; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - /* Configure McBSP internal buffer usage */ - if (dma_data->set_threshold) - dma_data->set_threshold(substream); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - break; - default: - ret = -EINVAL; - } - - if (ret == 0) - ret = snd_dmaengine_pcm_trigger(substream, cmd); - - return ret; -} - static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) { snd_pcm_uframes_t offset; @@ -175,20 +112,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) static int omap_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct omap_pcm_dma_data *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware); dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - return snd_dmaengine_pcm_open(substream, omap_dma_filter_fn, - &dma_data->dma_req); -} - -static int omap_pcm_close(struct snd_pcm_substream *substream) -{ - snd_dmaengine_pcm_close(substream); - return 0; + return snd_dmaengine_pcm_open_request_chan(substream, + omap_dma_filter_fn, + dma_data->filter_data); } static int omap_pcm_mmap(struct snd_pcm_substream *substream, @@ -204,11 +136,11 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream, static struct snd_pcm_ops omap_pcm_ops = { .open = omap_pcm_open, - .close = omap_pcm_close, + .close = snd_dmaengine_pcm_close_release_chan, .ioctl = snd_pcm_lib_ioctl, .hw_params = omap_pcm_hw_params, .hw_free = omap_pcm_hw_free, - .trigger = omap_pcm_trigger, + .trigger = snd_dmaengine_pcm_trigger, .pointer = omap_pcm_pointer, .mmap = omap_pcm_mmap, }; diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h deleted file mode 100644 index cabe74c4068b..000000000000 --- a/sound/soc/omap/omap-pcm.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * omap-pcm.h - * - * Copyright (C) 2008 Nokia Corporation - * - * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com> - * Peter Ujfalusi <peter.ujfalusi@ti.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. - * - * 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., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __OMAP_PCM_H__ -#define __OMAP_PCM_H__ - -struct snd_pcm_substream; - -struct omap_pcm_dma_data { - char *name; /* stream identifier */ - int dma_req; /* DMA request line */ - unsigned long port_addr; /* transmit/receive register */ - void (*set_threshold)(struct snd_pcm_substream *substream); - int data_type; /* 8, 16, 32 (bits) or 0 to let omap-pcm - * to decide the sDMA data type */ - int packet_size; /* packet size only in PACKET mode */ -}; - -#endif diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index fd98509d0f49..2a9324f794d8 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -43,7 +43,6 @@ #include <sound/jack.h> #include "omap-mcbsp.h" -#include "omap-pcm.h" struct omap_twl4030 { int jack_detect; /* board can detect jack events */ diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 805512f2555a..cf604a2faa18 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -34,7 +34,6 @@ #include <linux/platform_data/asoc-ti-mcbsp.h> #include "omap-mcbsp.h" -#include "omap-pcm.h" #define OMAP3_PANDORA_DAC_POWER_GPIO 118 #define OMAP3_PANDORA_AMP_POWER_GPIO 14 @@ -80,12 +79,18 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream, static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { + int ret; + /* * The PCM1773 DAC datasheet requires 1ms delay between switching * VCC power on/off and /PD pin high/low */ if (SND_SOC_DAPM_EVENT_ON(event)) { - regulator_enable(omap3pandora_dac_reg); + ret = regulator_enable(omap3pandora_dac_reg); + if (ret) { + dev_err(w->dapm->dev, "Failed to power DAC: %d\n", ret); + return ret; + } mdelay(1); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); } else { diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 06ef8d67ed1c..d03e57da7708 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -33,7 +33,6 @@ #include <linux/platform_data/asoc-ti-mcbsp.h> #include "omap-mcbsp.h" -#include "omap-pcm.h" #include "../codecs/tlv320aic23.h" #define CODEC_CLOCK 12000000 diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 3cd525748975..249cd230ad8f 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -37,7 +37,6 @@ #include <asm/mach-types.h> #include "omap-mcbsp.h" -#include "omap-pcm.h" #define RX51_TVOUT_SEL_GPIO 40 #define RX51_JACK_DETECT_GPIO 177 diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 190eb0bccf5f..349930015264 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -118,9 +118,8 @@ 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 mmp_dma_data dma_data; struct resource *r; - int ret; r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream); if (!r) @@ -128,33 +127,12 @@ static int mmp_pcm_open(struct snd_pcm_substream *substream) 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; + 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; + return snd_dmaengine_pcm_open_request_chan(substream, filter, + &dma_data); } static int mmp_pcm_mmap(struct snd_pcm_substream *substream, @@ -171,7 +149,7 @@ static int mmp_pcm_mmap(struct snd_pcm_substream *substream, struct snd_pcm_ops mmp_pcm_ops = { .open = mmp_pcm_open, - .close = mmp_pcm_close, + .close = snd_dmaengine_pcm_close_release_chan, .ioctl = snd_pcm_lib_ioctl, .hw_params = mmp_pcm_hw_params, .trigger = snd_dmaengine_pcm_trigger, diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 9140c4abafbc..a64779980177 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -405,6 +405,10 @@ struct snd_soc_dai_driver mmp_sspa_dai = { .ops = &mmp_sspa_dai_ops, }; +static const struct snd_soc_component_driver mmp_sspa_component = { + .name = "mmp-sspa", +}; + static int asoc_mmp_sspa_probe(struct platform_device *pdev) { struct sspa_priv *priv; @@ -450,7 +454,8 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) priv->dai_fmt = (unsigned int) -1; platform_set_drvdata(pdev, priv); - return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai); + return snd_soc_register_component(&pdev->dev, &mmp_sspa_component, + &mmp_sspa_dai, 1); } static int asoc_mmp_sspa_remove(struct platform_device *pdev) @@ -460,7 +465,7 @@ static int asoc_mmp_sspa_remove(struct platform_device *pdev) clk_disable(priv->audio_clk); clk_put(priv->audio_clk); clk_put(priv->sysclk); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d3eb0c2eec77..6f4dd7543e82 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -794,14 +794,19 @@ static struct snd_soc_dai_driver pxa_ssp_dai = { .ops = &pxa_ssp_dai_ops, }; +static const struct snd_soc_component_driver pxa_ssp_component = { + .name = "pxa-ssp", +}; + static int asoc_ssp_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &pxa_ssp_dai); + return snd_soc_register_component(&pdev->dev, &pxa_ssp_component, + &pxa_ssp_dai, 1); } static int asoc_ssp_remove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 4b0a009bd683..57ea8e6c5488 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -47,6 +47,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = { .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; +EXPORT_SYMBOL_GPL(soc_ac97_ops); static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { .name = "AC97 PCM Stereo out", @@ -232,7 +233,9 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { }, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); +static const struct snd_soc_component_driver pxa_ac97_component = { + .name = "pxa-ac97", +}; static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { @@ -245,13 +248,13 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) * driver to do interesting things with the clocking to get us up * and running. */ - return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, - ARRAY_SIZE(pxa_ac97_dai_driver)); + return snd_soc_register_component(&pdev->dev, &pxa_ac97_component, + pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver)); } static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6b1a06f67564..f7ca71664112 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -360,14 +360,19 @@ static struct snd_soc_dai_driver pxa_i2s_dai = { .symmetric_rates = 1, }; +static const struct snd_soc_component_driver pxa_i2s_component = { + .name = "pxa-i2s", +}; + static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &pxa_i2s_dai); + return snd_soc_register_component(&pdev->dev, &pxa_i2s_component, + &pxa_i2s_dai, 1); } static int pxa2xx_i2s_drv_remove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index fee4d477a49c..73bb99f0109a 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -436,6 +436,10 @@ static struct snd_soc_dai_driver s6000_i2s_dai = { .ops = &s6000_i2s_dai_ops, }; +static const struct snd_soc_component_driver s6000_i2s_component = { + .name = "s6000-i2s", +}; + static int s6000_i2s_probe(struct platform_device *pdev) { struct s6000_i2s_dev *dev; @@ -543,7 +547,8 @@ static int s6000_i2s_probe(struct platform_device *pdev) S6_I2S_INT_UNDERRUN | S6_I2S_INT_OVERRUN); - ret = snd_soc_register_dai(&pdev->dev, &s6000_i2s_dai); + ret = snd_soc_register_component(&pdev->dev, &s6000_i2s_component, + &s6000_i2s_dai, 1); if (ret) goto err_release_dev; @@ -572,7 +577,7 @@ static void s6000_i2s_remove(struct platform_device *pdev) struct resource *region; void __iomem *mmio = dev->scbbase; - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); s6000_i2s_stop_channel(dev, 0); s6000_i2s_stop_channel(dev, 1); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 90e7e6653233..475fb0d8b3c6 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -35,11 +35,10 @@ config SND_SAMSUNG_I2S tristate config SND_SOC_SAMSUNG_NEO1973_WM8753 - tristate "Audio support for Openmoko Neo1973 Smartphones (GTA01/GTA02)" - depends on SND_SOC_SAMSUNG && (MACH_NEO1973_GTA01 || MACH_NEO1973_GTA02) + tristate "Audio support for Openmoko Neo1973 Smartphones (GTA02)" + depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 - select SND_SOC_LM4857 if MACH_NEO1973_GTA01 select SND_SOC_DFBMCS320 help Say Y here to enable audio support for the Openmoko Neo1973 diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 0df3c5644cfa..cb88ead98917 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -20,7 +20,7 @@ #include <sound/soc.h> #include <mach/dma.h> -#include <plat/regs-ac97.h> +#include "regs-ac97.h" #include <linux/platform_data/asoc-s3c.h> #include "dma.h" @@ -370,6 +370,10 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { }, }; +static const struct snd_soc_component_driver s3c_ac97_component = { + .name = "s3c-ac97", +}; + static int s3c_ac97_probe(struct platform_device *pdev) { struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; @@ -457,8 +461,8 @@ static int s3c_ac97_probe(struct platform_device *pdev) goto err4; } - ret = snd_soc_register_dais(&pdev->dev, s3c_ac97_dai, - ARRAY_SIZE(s3c_ac97_dai)); + ret = snd_soc_register_component(&pdev->dev, &s3c_ac97_component, + s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); if (ret) goto err5; @@ -470,7 +474,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) return 0; err6: - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(s3c_ac97_dai)); + snd_soc_unregister_component(&pdev->dev); err5: free_irq(irq_res->start, NULL); err4: @@ -490,7 +494,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) struct resource *mem_res, *irq_res; asoc_dma_platform_unregister(&pdev->dev); - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(s3c_ac97_dai)); + snd_soc_unregister_component(&pdev->dev); irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); if (irq_res) diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index d37ede58e0a8..415ad81999c4 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -218,6 +218,10 @@ static struct snd_soc_dai_driver voice_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE,}, }; +static const struct snd_soc_component_driver voice_component = { + .name = "goni-voice", +}; + static struct snd_soc_ops goni_voice_ops = { .hw_params = goni_voice_hw_params, }; @@ -270,7 +274,8 @@ static int __init goni_init(void) return -ENOMEM; /* register voice DAI here */ - ret = snd_soc_register_dai(&goni_snd_device->dev, &voice_dai); + ret = snd_soc_register_component(&goni_snd_device->dev, &voice_component, + &voice_dai, 1); if (ret) { platform_device_put(goni_snd_device); return ret; @@ -280,7 +285,7 @@ static int __init goni_init(void) ret = platform_device_add(goni_snd_device); if (ret) { - snd_soc_unregister_dai(&goni_snd_device->dev); + snd_soc_unregister_component(&goni_snd_device->dev); platform_device_put(goni_snd_device); } @@ -289,7 +294,7 @@ static int __init goni_init(void) static void __exit goni_exit(void) { - snd_soc_unregister_dai(&goni_snd_device->dev); + snd_soc_unregister_component(&goni_snd_device->dev); platform_device_unregister(goni_snd_device); } diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 15a3817aa5c8..fa91376e323d 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -20,7 +20,7 @@ #include <sound/soc.h> #include <sound/jack.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include <asm/mach-types.h> #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 6bbeb0bf1a73..82ebb1a51479 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -963,6 +963,10 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = { .delay = i2s_delay, }; +static const struct snd_soc_component_driver samsung_i2s_component = { + .name = "samsung-i2s", +}; + #define SAMSUNG_I2S_RATES SNDRV_PCM_RATE_8000_96000 #define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ @@ -1114,8 +1118,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Unable to get drvdata\n"); return -EFAULT; } - snd_soc_register_dai(&sec_dai->pdev->dev, - &sec_dai->i2s_dai_drv); + snd_soc_register_component(&sec_dai->pdev->dev, + &samsung_i2s_component, + &sec_dai->i2s_dai_drv, 1); asoc_dma_platform_register(&pdev->dev); return 0; } @@ -1244,7 +1249,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) } } - snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv); + snd_soc_register_component(&pri_dai->pdev->dev, &samsung_i2s_component, + &pri_dai->i2s_dai_drv, 1); pm_runtime_enable(&pdev->dev); @@ -1283,7 +1289,7 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s->sec_dai = NULL; asoc_dma_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } @@ -1298,7 +1304,7 @@ static struct platform_device_id samsung_i2s_driver_ids[] = { }, {}, }; -MODULE_DEVICE_TABLE(platform, samsung-i2s-driver-ids); +MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids); #ifdef CONFIG_OF static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = { diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index a07950b0c8ce..6e5fed30aa27 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -68,6 +68,8 @@ static struct idma_info { dma_addr_t lp_tx_addr; } idma; +static int idma_irq; + static void idma_getpos(dma_addr_t *src) { *src = idma.lp_tx_addr + @@ -305,7 +307,7 @@ static int idma_open(struct snd_pcm_substream *substream) if (prtd == NULL) return -ENOMEM; - ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + ret = request_irq(idma_irq, iis_irq, 0, "i2s", prtd); if (ret < 0) { pr_err("fail to claim i2s irq , ret = %d\n", ret); kfree(prtd); @@ -324,7 +326,7 @@ static int idma_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct idma_ctrl *prtd = runtime->private_data; - free_irq(IRQ_I2S0, prtd); + free_irq(idma_irq, prtd); if (!prtd) pr_err("idma_close called with prtd == NULL\n"); @@ -409,6 +411,7 @@ void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) idma.regs = regs; idma.lp_tx_addr = addr; } +EXPORT_SYMBOL_GPL(idma_reg_addr_init); static struct snd_soc_platform_driver asoc_idma_platform = { .ops = &idma_ops, @@ -418,6 +421,10 @@ static struct snd_soc_platform_driver asoc_idma_platform = { static int asoc_idma_platform_probe(struct platform_device *pdev) { + idma_irq = platform_get_irq(pdev, 0); + if (idma_irq < 0) + return idma_irq; + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); } diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index a301d8cfaa34..e591c386917a 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -21,8 +21,7 @@ #include <sound/soc.h> #include <asm/mach-types.h> -#include <plat/regs-iis.h> -#include <mach/gta02.h> +#include "regs-iis.h" #include "../codecs/wm8753.h" #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 13bab79ad93d..1566afe9ef52 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -490,6 +490,10 @@ static struct snd_soc_dai_driver s3c_pcm_dai[] = { }, }; +static const struct snd_soc_component_driver s3c_pcm_component = { + .name = "s3c-pcm", +}; + static int s3c_pcm_dev_probe(struct platform_device *pdev) { struct s3c_pcm_info *pcm; @@ -583,7 +587,8 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); + ret = snd_soc_register_component(&pdev->dev, &s3c_pcm_component, + &s3c_pcm_dai[pdev->id], 1); if (ret != 0) { dev_err(&pdev->dev, "failed to get register DAI: %d\n", ret); goto err5; @@ -598,7 +603,7 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev) return 0; err6: - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); err5: clk_disable_unprepare(pcm->pclk); clk_put(pcm->pclk); @@ -619,7 +624,7 @@ static int s3c_pcm_dev_remove(struct platform_device *pdev) struct resource *mem_res; asoc_dma_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); pm_runtime_disable(&pdev->dev); diff --git a/arch/arm/plat-samsung/include/plat/regs-ac97.h b/sound/soc/samsung/regs-ac97.h index c3878f7acb83..c3878f7acb83 100644 --- a/arch/arm/plat-samsung/include/plat/regs-ac97.h +++ b/sound/soc/samsung/regs-ac97.h diff --git a/arch/arm/plat-samsung/include/plat/regs-iis.h b/sound/soc/samsung/regs-iis.h index a18d35e7a735..a18d35e7a735 100644 --- a/arch/arm/plat-samsung/include/plat/regs-iis.h +++ b/sound/soc/samsung/regs-iis.h diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index a5826ea9cad6..704460a37005 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -24,7 +24,7 @@ #include <sound/soc.h> #include <sound/jack.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include <asm/mach-types.h> #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 7a73380b3560..20e98d1dded2 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -731,8 +731,9 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai) #define s3c2412_i2s_resume NULL #endif -int s3c_i2sv2_register_dai(struct device *dev, int id, - struct snd_soc_dai_driver *drv) +int s3c_i2sv2_register_component(struct device *dev, int id, + struct snd_soc_component_driver *cmp_drv, + struct snd_soc_dai_driver *dai_drv) { struct snd_soc_dai_ops *ops = drv->ops; @@ -750,8 +751,8 @@ int s3c_i2sv2_register_dai(struct device *dev, int id, drv->suspend = s3c2412_i2s_suspend; drv->resume = s3c2412_i2s_resume; - return snd_soc_register_dai(dev, drv); + return snd_soc_register_component(dev, cmp_drv, dai_drv, 1); } -EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); +EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h index f8297d9bb8a3..90abab364b49 100644 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -92,7 +92,7 @@ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, unsigned long base); /** - * s3c_i2sv2_register_dai - register dai with soc core + * s3c_i2sv2_register_component - register component and dai with soc core * @dev: DAI device * @id: DAI ID * @drv: The driver structure to register @@ -100,7 +100,8 @@ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, * Fill in any missing fields and then register the given dai with the * soc core. */ -extern int s3c_i2sv2_register_dai(struct device *dev, int id, - struct snd_soc_dai_driver *drv); +extern int s3c_i2sv2_register_component(struct device *dev, int id, + struct snd_soc_component_driver *cmp_drv, + struct snd_soc_dai_driver *dai_drv); #endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 221337716393..47e23864ea72 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -160,11 +160,17 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { .ops = &s3c2412_i2s_dai_ops, }; +static const struct snd_soc_component_driver s3c2412_i2s_component = { + .name = "s3c2412-i2s", +}; + static int s3c2412_iis_dev_probe(struct platform_device *pdev) { int ret = 0; - ret = s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai); + ret = s3c_i2sv2_register_component(&pdev->dev, -1, + &s3c2412_i2s_component, + &s3c2412_i2s_dai); if (ret) { pr_err("failed to register the dai\n"); return ret; @@ -178,14 +184,14 @@ static int s3c2412_iis_dev_probe(struct platform_device *pdev) return 0; err: - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return ret; } static int s3c2412_iis_dev_remove(struct platform_device *pdev) { asoc_dma_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 13f6dd1ceb00..8b3414551a62 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -24,7 +24,7 @@ #include <sound/pcm_params.h> #include <mach/dma.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include "dma.h" #include "s3c24xx-i2s.h" @@ -465,11 +465,16 @@ static struct snd_soc_dai_driver s3c24xx_i2s_dai = { .ops = &s3c24xx_i2s_dai_ops, }; +static const struct snd_soc_component_driver s3c24xx_i2s_component = { + .name = "s3c24xx-i2s", +}; + static int s3c24xx_iis_dev_probe(struct platform_device *pdev) { int ret = 0; - ret = snd_soc_register_dai(&pdev->dev, &s3c24xx_i2s_dai); + ret = snd_soc_register_component(&pdev->dev, &s3c24xx_i2s_component, + &s3c24xx_i2s_dai, 1); if (ret) { pr_err("failed to register the dai\n"); return ret; @@ -483,14 +488,14 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev) return 0; err: - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return ret; } static int s3c24xx_iis_dev_remove(struct platform_device *pdev) { asoc_dma_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 333e1b7f06c7..1b7b52b0af97 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -18,7 +18,7 @@ #include <sound/soc.h> #include <sound/s3c24xx_uda134x.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 5008e5bd6ed8..2e5ebb2f1982 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -357,6 +357,10 @@ static struct snd_soc_dai_driver samsung_spdif_dai = { .resume = spdif_resume, }; +static const struct snd_soc_component_driver samsung_spdif_component = { + .name = "samsung-spdif", +}; + static int spdif_probe(struct platform_device *pdev) { struct s3c_audio_pdata *spdif_pdata; @@ -424,7 +428,8 @@ static int spdif_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, spdif); - ret = snd_soc_register_dai(&pdev->dev, &samsung_spdif_dai); + ret = snd_soc_register_component(&pdev->dev, &samsung_spdif_component, + &samsung_spdif_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "fail to register dai\n"); goto err4; @@ -445,7 +450,7 @@ static int spdif_probe(struct platform_device *pdev) return 0; err5: - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); err4: iounmap(spdif->regs); err3: @@ -466,7 +471,7 @@ static int spdif_remove(struct platform_device *pdev) struct resource *mem_res; asoc_dma_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); iounmap(spdif->regs); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index c724026a246f..f830c41f97dd 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -296,7 +296,6 @@ struct fsi_core { struct fsi_master { void __iomem *base; - int irq; struct fsi_priv fsia; struct fsi_priv fsib; const struct fsi_core *core; @@ -1886,6 +1885,10 @@ static struct snd_soc_platform_driver fsi_soc_platform = { .pcm_free = fsi_pcm_free, }; +static const struct snd_soc_component_driver fsi_soc_component = { + .name = "fsi", +}; + /* * platform function */ @@ -2002,7 +2005,6 @@ static int fsi_probe(struct platform_device *pdev) } /* master setting */ - master->irq = irq; master->core = core; spin_lock_init(&master->lock); @@ -2046,10 +2048,10 @@ static int fsi_probe(struct platform_device *pdev) goto exit_fsib; } - ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai, - ARRAY_SIZE(fsi_soc_dai)); + ret = snd_soc_register_component(&pdev->dev, &fsi_soc_component, + fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); if (ret < 0) { - dev_err(&pdev->dev, "cannot snd dai register\n"); + dev_err(&pdev->dev, "cannot snd component register\n"); goto exit_snd_soc; } @@ -2074,7 +2076,7 @@ static int fsi_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); + snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); fsi_stream_remove(&master->fsia); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 4cc2d64ef476..af19f77b7bf0 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -310,15 +310,19 @@ static struct snd_soc_dai_driver sh4_hac_dai[] = { #endif }; +static const struct snd_soc_component_driver sh4_hac_component = { + .name = "sh4-hac", +}; + static int hac_soc_platform_probe(struct platform_device *pdev) { - return snd_soc_register_dais(&pdev->dev, sh4_hac_dai, - ARRAY_SIZE(sh4_hac_dai)); + return snd_soc_register_component(&pdev->dev, &sh4_hac_component, + sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); } static int hac_soc_platform_remove(struct platform_device *pdev) { - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sh4_hac_dai)); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 8526e1edaf45..5014a884afee 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -153,7 +153,7 @@ static int migor_dai_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link migor_dai = { .name = "wm8978", .stream_name = "WM8978", - .cpu_dai_name = "siu-i2s-dai", + .cpu_dai_name = "siu-pcm-audio", .codec_dai_name = "wm8978-hifi", .platform_name = "siu-pcm-audio", .codec_name = "wm8978.0-001a", diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index 34facdc9e4ac..9dc24ffa892a 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -726,6 +726,10 @@ static struct snd_soc_dai_driver siu_i2s_dai = { .ops = &siu_dai_ops, }; +static const struct snd_soc_component_driver siu_i2s_component = { + .name = "siu-i2s", +}; + static int siu_probe(struct platform_device *pdev) { const struct firmware *fw_entry; @@ -783,7 +787,8 @@ static int siu_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, info); /* register using ARRAY version so we can keep dai name */ - ret = snd_soc_register_dais(&pdev->dev, &siu_i2s_dai, 1); + ret = snd_soc_register_component(&pdev->dev, &siu_i2s_component, + &siu_i2s_dai, 1); if (ret < 0) goto edaiinit; @@ -796,7 +801,7 @@ static int siu_probe(struct platform_device *pdev) return ret; esocregp: - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); edaiinit: iounmap(info->reg); emapreg: @@ -823,7 +828,7 @@ static int siu_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); iounmap(info->reg); iounmap(info->yram); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index c8e73a703934..e889405ebd38 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -379,15 +379,19 @@ static struct snd_soc_dai_driver sh4_ssi_dai[] = { #endif }; +static const struct snd_soc_component_driver sh4_ssi_component = { + .name = "sh4-ssi", +}; + static int sh4_soc_dai_probe(struct platform_device *pdev) { - return snd_soc_register_dais(&pdev->dev, sh4_ssi_dai, - ARRAY_SIZE(sh4_ssi_dai)); + return snd_soc_register_component(&pdev->dev, &sh4_ssi_component, + sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai)); } static int sh4_soc_dai_remove(struct platform_device *pdev) { - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai)); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ed0bfb0ddb96..3853f7eb3f28 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -315,7 +315,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, } static int soc_compr_copy(struct snd_compr_stream *cstream, - const char __user *buf, size_t count) + char __user *buf, size_t count) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; @@ -330,11 +330,38 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, return ret; } +static int sst_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->set_metadata) + ret = platform->driver->compr_ops->set_metadata(cstream, metadata); + + return ret; +} + +static int sst_compr_get_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->get_metadata) + ret = platform->driver->compr_ops->get_metadata(cstream, metadata); + + return ret; +} /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, + .set_metadata = sst_compr_set_metadata, + .get_metadata = sst_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, @@ -357,7 +384,14 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); - direction = SND_COMPRESS_PLAYBACK; + + if (codec_dai->driver->playback.channels_min) + direction = SND_COMPRESS_PLAYBACK; + else if (codec_dai->driver->capture.channels_min) + direction = SND_COMPRESS_CAPTURE; + else + return -EINVAL; + compr = kzalloc(sizeof(*compr), GFP_KERNEL); if (compr == NULL) { snd_printk(KERN_ERR "Cannot allocate compr\n"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ff4b45a5d796..d56bbea6e75e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -58,6 +58,7 @@ static DEFINE_MUTEX(client_mutex); static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); +static LIST_HEAD(component_list); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -3740,7 +3741,7 @@ static inline char *fmt_multiple_name(struct device *dev, * * @dai: DAI to register */ -int snd_soc_register_dai(struct device *dev, +static int snd_soc_register_dai(struct device *dev, struct snd_soc_dai_driver *dai_drv) { struct snd_soc_codec *codec; @@ -3787,14 +3788,13 @@ int snd_soc_register_dai(struct device *dev, return 0; } -EXPORT_SYMBOL_GPL(snd_soc_register_dai); /** * snd_soc_unregister_dai - Unregister a DAI from the ASoC core * * @dai: DAI to unregister */ -void snd_soc_unregister_dai(struct device *dev) +static void snd_soc_unregister_dai(struct device *dev) { struct snd_soc_dai *dai; @@ -3813,7 +3813,6 @@ found: kfree(dai->name); kfree(dai); } -EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); /** * snd_soc_register_dais - Register multiple DAIs with the ASoC core @@ -3821,7 +3820,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); * @dai: Array of DAIs to register * @count: Number of DAIs */ -int snd_soc_register_dais(struct device *dev, +static int snd_soc_register_dais(struct device *dev, struct snd_soc_dai_driver *dai_drv, size_t count) { struct snd_soc_codec *codec; @@ -3885,7 +3884,6 @@ err: return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_dais); /** * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core @@ -3893,31 +3891,23 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dais); * @dai: Array of DAIs to unregister * @count: Number of DAIs */ -void snd_soc_unregister_dais(struct device *dev, size_t count) +static void snd_soc_unregister_dais(struct device *dev, size_t count) { int i; for (i = 0; i < count; i++) snd_soc_unregister_dai(dev); } -EXPORT_SYMBOL_GPL(snd_soc_unregister_dais); /** - * snd_soc_register_platform - Register a platform with the ASoC core - * - * @platform: platform to register + * snd_soc_add_platform - Add a platform to the ASoC core + * @dev: The parent device for the platform + * @platform: The platform to add + * @platform_driver: The driver for the platform */ -int snd_soc_register_platform(struct device *dev, - struct snd_soc_platform_driver *platform_drv) +int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, + const struct snd_soc_platform_driver *platform_drv) { - struct snd_soc_platform *platform; - - dev_dbg(dev, "ASoC: platform register %s\n", dev_name(dev)); - - platform = kzalloc(sizeof(struct snd_soc_platform), GFP_KERNEL); - if (platform == NULL) - return -ENOMEM; - /* create platform component name */ platform->name = fmt_single_name(dev, &platform->id); if (platform->name == NULL) { @@ -3940,30 +3930,76 @@ int snd_soc_register_platform(struct device *dev, return 0; } -EXPORT_SYMBOL_GPL(snd_soc_register_platform); +EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** - * snd_soc_unregister_platform - Unregister a platform from the ASoC core + * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to unregister + * @platform: platform to register */ -void snd_soc_unregister_platform(struct device *dev) +int snd_soc_register_platform(struct device *dev, + const struct snd_soc_platform_driver *platform_drv) { struct snd_soc_platform *platform; + int ret; - list_for_each_entry(platform, &platform_list, list) { - if (dev == platform->dev) - goto found; - } - return; + dev_dbg(dev, "ASoC: platform register %s\n", dev_name(dev)); -found: + platform = kzalloc(sizeof(struct snd_soc_platform), GFP_KERNEL); + if (platform == NULL) + return -ENOMEM; + + ret = snd_soc_add_platform(dev, platform, platform_drv); + if (ret) + kfree(platform); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_register_platform); + +/** + * snd_soc_remove_platform - Remove a platform from the ASoC core + * @platform: the platform to remove + */ +void snd_soc_remove_platform(struct snd_soc_platform *platform) +{ mutex_lock(&client_mutex); list_del(&platform->list); mutex_unlock(&client_mutex); - dev_dbg(dev, "ASoC: Unregistered platform '%s'\n", platform->name); + dev_dbg(platform->dev, "ASoC: Unregistered platform '%s'\n", + platform->name); kfree(platform->name); +} +EXPORT_SYMBOL_GPL(snd_soc_remove_platform); + +struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev) +{ + struct snd_soc_platform *platform; + + list_for_each_entry(platform, &platform_list, list) { + if (dev == platform->dev) + return platform; + } + + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); + +/** + * snd_soc_unregister_platform - Unregister a platform from the ASoC core + * + * @platform: platform to unregister + */ +void snd_soc_unregister_platform(struct device *dev) +{ + struct snd_soc_platform *platform; + + platform = snd_soc_lookup_platform(dev); + if (!platform) + return; + + snd_soc_remove_platform(platform); kfree(platform); } EXPORT_SYMBOL_GPL(snd_soc_unregister_platform); @@ -4024,8 +4060,8 @@ int snd_soc_register_codec(struct device *dev, /* create CODEC component name */ codec->name = fmt_single_name(dev, &codec->id); if (codec->name == NULL) { - kfree(codec); - return -ENOMEM; + ret = -ENOMEM; + goto fail_codec; } if (codec_drv->compress_type) @@ -4064,7 +4100,7 @@ int snd_soc_register_codec(struct device *dev, reg_size, GFP_KERNEL); if (!codec->reg_def_copy) { ret = -ENOMEM; - goto fail; + goto fail_codec_name; } } } @@ -4088,18 +4124,22 @@ int snd_soc_register_codec(struct device *dev, mutex_unlock(&client_mutex); /* register any DAIs */ - if (num_dai) { - ret = snd_soc_register_dais(dev, dai_drv, num_dai); - if (ret < 0) - dev_err(codec->dev, "ASoC: Failed to regster" - " DAIs: %d\n", ret); + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) { + dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto fail_codec_name; } dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); return 0; -fail: +fail_codec_name: + mutex_lock(&client_mutex); + list_del(&codec->list); + mutex_unlock(&client_mutex); + kfree(codec->name); +fail_codec: kfree(codec); return ret; } @@ -4113,7 +4153,6 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); void snd_soc_unregister_codec(struct device *dev) { struct snd_soc_codec *codec; - int i; list_for_each_entry(codec, &codec_list, list) { if (dev == codec->dev) @@ -4122,9 +4161,7 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - if (codec->num_dai) - for (i = 0; i < codec->num_dai; i++) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(dev, codec->num_dai); mutex_lock(&client_mutex); list_del(&codec->list); @@ -4139,6 +4176,92 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); + +/** + * snd_soc_register_component - Register a component with the ASoC core + * + */ +int snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, + int num_dai) +{ + struct snd_soc_component *cmpnt; + int ret; + + dev_dbg(dev, "component register %s\n", dev_name(dev)); + + cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL); + if (!cmpnt) { + dev_err(dev, "ASoC: Failed to allocate memory\n"); + return -ENOMEM; + } + + cmpnt->name = fmt_single_name(dev, &cmpnt->id); + if (!cmpnt->name) { + dev_err(dev, "ASoC: Failed to simplifying name\n"); + return -ENOMEM; + } + + cmpnt->dev = dev; + cmpnt->driver = cmpnt_drv; + cmpnt->num_dai = num_dai; + + /* + * snd_soc_register_dai() uses fmt_single_name(), and + * snd_soc_register_dais() uses fmt_multiple_name() + * for dai->name which is used for name based matching + */ + if (1 == num_dai) + ret = snd_soc_register_dai(dev, dai_drv); + else + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto error_component_name; + } + + mutex_lock(&client_mutex); + list_add(&cmpnt->list, &component_list); + mutex_unlock(&client_mutex); + + dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name); + + return ret; + +error_component_name: + kfree(cmpnt->name); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_register_component); + +/** + * snd_soc_unregister_component - Unregister a component from the ASoC core + * + */ +void snd_soc_unregister_component(struct device *dev) +{ + struct snd_soc_component *cmpnt; + + list_for_each_entry(cmpnt, &component_list, list) { + if (dev == cmpnt->dev) + goto found; + } + return; + +found: + snd_soc_unregister_dais(dev, cmpnt->num_dai); + + mutex_lock(&client_mutex); + list_del(&cmpnt->list); + mutex_unlock(&client_mutex); + + dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); + kfree(cmpnt->name); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_component); + /* Retrieve a card's name from device tree */ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d6d9ba2e6916..21779a6a781a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -504,17 +504,27 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, return 0; } -/* create new dapm mixer control */ -static int dapm_new_mixer(struct snd_soc_dapm_widget *w) +/* + * Determine if a kcontrol is shared. If it is, look it up. If it isn't, + * create it. Either way, add the widget into the control's widget list + */ +static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, + int kci, struct snd_soc_dapm_path *path) { struct snd_soc_dapm_context *dapm = w->dapm; - int i, ret = 0; - size_t name_len, prefix_len; - struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; const char *prefix; + size_t prefix_len; + int shared; + struct snd_kcontrol *kcontrol; struct snd_soc_dapm_widget_list *wlist; + int wlistentries; size_t wlistsize; + bool wname_in_long_name, kcname_in_long_name; + size_t name_len; + char *long_name; + const char *name; + int ret; if (dapm->codec) prefix = dapm->codec->name_prefix; @@ -526,103 +536,141 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) else prefix_len = 0; - /* add kcontrol */ - for (i = 0; i < w->num_kcontrols; i++) { + shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[kci], + &kcontrol); - /* match name */ - list_for_each_entry(path, &w->sources, list_sink) { + if (kcontrol) { + wlist = kcontrol->private_data; + wlistentries = wlist->num_widgets + 1; + } else { + wlist = NULL; + wlistentries = 1; + } - /* mixer/mux paths name must match control name */ - if (path->name != (char *)w->kcontrol_news[i].name) - continue; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + wlist = krealloc(wlist, wlistsize, GFP_KERNEL); + if (wlist == NULL) { + dev_err(dapm->dev, "ASoC: can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist->num_widgets = wlistentries; + wlist->widgets[wlistentries - 1] = w; - if (w->kcontrols[i]) { - path->kcontrol = w->kcontrols[i]; - continue; + if (!kcontrol) { + if (shared) { + wname_in_long_name = false; + kcname_in_long_name = true; + } else { + switch (w->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + wname_in_long_name = true; + kcname_in_long_name = true; + break; + case snd_soc_dapm_mixer_named_ctl: + wname_in_long_name = false; + kcname_in_long_name = true; + break; + case snd_soc_dapm_mux: + case snd_soc_dapm_virt_mux: + case snd_soc_dapm_value_mux: + wname_in_long_name = true; + kcname_in_long_name = false; + break; + default: + kfree(wlist); + return -EINVAL; } + } + + if (wname_in_long_name && kcname_in_long_name) { + name_len = strlen(w->name) - prefix_len + 1 + + strlen(w->kcontrol_news[kci].name) + 1; - wlistsize = sizeof(struct snd_soc_dapm_widget_list) + - sizeof(struct snd_soc_dapm_widget *), - wlist = kzalloc(wlistsize, GFP_KERNEL); - if (wlist == NULL) { - dev_err(dapm->dev, - "ASoC: can't allocate widget list for %s\n", - w->name); + long_name = kmalloc(name_len, GFP_KERNEL); + if (long_name == NULL) { + kfree(wlist); return -ENOMEM; } - wlist->num_widgets = 1; - wlist->widgets[0] = w; - - /* add dapm control with long name. - * for dapm_mixer this is the concatenation of the - * mixer and kcontrol name. - * for dapm_mixer_named_ctl this is simply the - * kcontrol name. + + /* + * The control will get a prefix from the control + * creation process but we're also using the same + * prefix for widgets so cut the prefix off the + * front of the widget name. */ - name_len = strlen(w->kcontrol_news[i].name) + 1; - if (w->id != snd_soc_dapm_mixer_named_ctl) - name_len += 1 + strlen(w->name); + snprintf(long_name, name_len, "%s %s", + w->name + prefix_len, + w->kcontrol_news[kci].name); + long_name[name_len - 1] = '\0'; + + name = long_name; + } else if (wname_in_long_name) { + long_name = NULL; + name = w->name + prefix_len; + } else { + long_name = NULL; + name = w->kcontrol_news[kci].name; + } - path->long_name = kmalloc(name_len, GFP_KERNEL); + kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], wlist, name, + prefix); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, name, ret); + kfree(wlist); + kfree(long_name); + return ret; + } - if (path->long_name == NULL) { - kfree(wlist); - return -ENOMEM; - } + path->long_name = long_name; + } - switch (w->id) { - default: - /* The control will get a prefix from - * the control creation process but - * we're also using the same prefix - * for widgets so cut the prefix off - * the front of the widget name. - */ - snprintf((char *)path->long_name, name_len, - "%s %s", w->name + prefix_len, - w->kcontrol_news[i].name); - break; - case snd_soc_dapm_mixer_named_ctl: - snprintf((char *)path->long_name, name_len, - "%s", w->kcontrol_news[i].name); - break; - } + kcontrol->private_data = wlist; + w->kcontrols[kci] = kcontrol; + path->kcontrol = kcontrol; - ((char *)path->long_name)[name_len - 1] = '\0'; + return 0; +} - path->kcontrol = snd_soc_cnew(&w->kcontrol_news[i], - wlist, path->long_name, - prefix); - ret = snd_ctl_add(card, path->kcontrol); - if (ret < 0) { - dev_err(dapm->dev, "ASoC: failed to add widget" - " %s dapm kcontrol %s: %d\n", - w->name, path->long_name, ret); - kfree(wlist); - kfree(path->long_name); - path->long_name = NULL; - return ret; +/* create new dapm mixer control */ +static int dapm_new_mixer(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_soc_dapm_path *path; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + /* match name */ + list_for_each_entry(path, &w->sources, list_sink) { + /* mixer/mux paths name must match control name */ + if (path->name != (char *)w->kcontrol_news[i].name) + continue; + + if (w->kcontrols[i]) { + path->kcontrol = w->kcontrols[i]; + continue; } - w->kcontrols[i] = path->kcontrol; + + ret = dapm_create_or_share_mixmux_kcontrol(w, i, path); + if (ret < 0) + return ret; } } - return ret; + + return 0; } /* create new dapm mux control */ static int dapm_new_mux(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_dapm_path *path = NULL; - struct snd_kcontrol *kcontrol; - struct snd_card *card = dapm->card->snd_card; - const char *prefix; - size_t prefix_len; + struct snd_soc_dapm_path *path; int ret; - struct snd_soc_dapm_widget_list *wlist; - int shared, wlistentries; - size_t wlistsize; - const char *name; if (w->num_kcontrols != 1) { dev_err(dapm->dev, @@ -631,65 +679,19 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0], - &kcontrol); - if (kcontrol) { - wlist = kcontrol->private_data; - wlistentries = wlist->num_widgets + 1; - } else { - wlist = NULL; - wlistentries = 1; - } - wlistsize = sizeof(struct snd_soc_dapm_widget_list) + - wlistentries * sizeof(struct snd_soc_dapm_widget *), - wlist = krealloc(wlist, wlistsize, GFP_KERNEL); - if (wlist == NULL) { - dev_err(dapm->dev, - "ASoC: can't allocate widget list for %s\n", w->name); - return -ENOMEM; - } - wlist->num_widgets = wlistentries; - wlist->widgets[wlistentries - 1] = w; - - if (!kcontrol) { - if (dapm->codec) - prefix = dapm->codec->name_prefix; - else - prefix = NULL; - - if (shared) { - name = w->kcontrol_news[0].name; - prefix_len = 0; - } else { - name = w->name; - if (prefix) - prefix_len = strlen(prefix) + 1; - else - prefix_len = 0; - } - - /* - * The control will get a prefix from the control creation - * process but we're also using the same prefix for widgets so - * cut the prefix off the front of the widget name. - */ - kcontrol = snd_soc_cnew(&w->kcontrol_news[0], wlist, - name + prefix_len, prefix); - ret = snd_ctl_add(card, kcontrol); - if (ret < 0) { - dev_err(dapm->dev, "ASoC: failed to add kcontrol %s: %d\n", - w->name, ret); - kfree(wlist); - return ret; - } + path = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + if (!path) { + dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name); + return -EINVAL; } - kcontrol->private_data = wlist; - - w->kcontrols[0] = kcontrol; + ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); + if (ret < 0) + return ret; list_for_each_entry(path, &w->sources, list_sink) - path->kcontrol = kcontrol; + path->kcontrol = w->kcontrols[0]; return 0; } @@ -705,14 +707,33 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) } /* reset 'walked' bit for each dapm path */ -static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) +static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, + struct list_head *sink) { struct snd_soc_dapm_path *p; - list_for_each_entry(p, &dapm->card->paths, list) - p->walked = 0; + list_for_each_entry(p, sink, list_source) { + if (p->walked) { + p->walked = 0; + dapm_clear_walk_output(dapm, &p->sink->sinks); + } + } } +static void dapm_clear_walk_input(struct snd_soc_dapm_context *dapm, + struct list_head *source) +{ + struct snd_soc_dapm_path *p; + + list_for_each_entry(p, source, list_sink) { + if (p->walked) { + p->walked = 0; + dapm_clear_walk_input(dapm, &p->source->sources); + } + } +} + + /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. @@ -995,13 +1016,17 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); dapm_reset(card); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { paths = is_connected_output_ep(dai->playback_widget, list); - else + dapm_clear_walk_output(&card->dapm, + &dai->playback_widget->sinks); + } else { paths = is_connected_input_ep(dai->capture_widget, list); + dapm_clear_walk_input(&card->dapm, + &dai->capture_widget->sources); + } trace_snd_soc_dapm_connected(paths, stream); - dapm_clear_walk(&card->dapm); mutex_unlock(&card->dapm_mutex); return paths; @@ -1104,9 +1129,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); in = is_connected_input_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_input(w->dapm, &w->sources); out = is_connected_output_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_output(w->dapm, &w->sinks); return out != 0 && in != 0; } @@ -1129,7 +1154,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) if (w->active) { in = is_connected_input_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_input(w->dapm, &w->sources); return in != 0; } else { return dapm_generic_check_power(w); @@ -1145,7 +1170,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) if (w->active) { out = is_connected_output_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_output(w->dapm, &w->sinks); return out != 0; } else { return dapm_generic_check_power(w); @@ -1177,8 +1202,6 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) return 1; } - dapm_clear_walk(w->dapm); - return 0; } @@ -1759,9 +1782,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, return -ENOMEM; in = is_connected_input_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_input(w->dapm, &w->sources); out = is_connected_output_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_output(w->dapm, &w->sinks); ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", @@ -3137,7 +3160,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; } - dapm->n_widgets++; w->dapm = dapm; w->codec = dapm->codec; w->platform = dapm->platform; diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 111b7d921e89..aa924d9b7986 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -33,8 +33,6 @@ struct dmaengine_pcm_runtime_data { dma_cookie_t cookie; unsigned int pos; - - void *data; }; static inline struct dmaengine_pcm_runtime_data *substream_to_prtd( @@ -43,33 +41,6 @@ static inline struct dmaengine_pcm_runtime_data *substream_to_prtd( return substream->runtime->private_data; } -/** - * snd_dmaengine_pcm_set_data - Set dmaengine substream private data - * @substream: PCM substream - * @data: Data to set - */ -void snd_dmaengine_pcm_set_data(struct snd_pcm_substream *substream, void *data) -{ - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - - prtd->data = data; -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_data); - -/** - * snd_dmaengine_pcm_get_data - Get dmaeinge substream private data - * @substream: PCM substream - * - * Returns the data previously set with snd_dmaengine_pcm_set_data - */ -void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream) -{ - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - - return prtd->data; -} -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_get_data); - struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); @@ -118,10 +89,49 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, slave_config->src_addr_width = buswidth; } + slave_config->device_fc = false; + return 0; } EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); +/** + * snd_dmaengine_pcm_set_config_from_dai_data() - Initializes a dma slave config + * using DAI DMA data. + * @substream: PCM substream + * @dma_data: DAI DMA data + * @slave_config: DMA slave configuration + * + * Initializes the {dst,src}_addr, {dst,src}_maxburst, {dst,src}_addr_width and + * slave_id fields of the DMA slave config from the same fields of the DAI DMA + * data struct. The src and dst fields will be initialized depending on the + * direction of the substream. If the substream is a playback stream the dst + * fields will be initialized, if it is a capture stream the src fields will be + * initialized. The {dst,src}_addr_width field will only be initialized if the + * addr_width field of the DAI DMA data struct is not equal to + * DMA_SLAVE_BUSWIDTH_UNDEFINED. + */ +void snd_dmaengine_pcm_set_config_from_dai_data( + const struct snd_pcm_substream *substream, + const struct snd_dmaengine_dai_dma_data *dma_data, + struct dma_slave_config *slave_config) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config->dst_addr = dma_data->addr; + slave_config->dst_maxburst = dma_data->maxburst; + if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) + slave_config->dst_addr_width = dma_data->addr_width; + } else { + slave_config->src_addr = dma_data->addr; + slave_config->src_maxburst = dma_data->maxburst; + if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) + slave_config->src_addr_width = dma_data->addr_width; + } + + slave_config->slave_id = dma_data->slave_id; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); + static void dmaengine_pcm_dma_complete(void *arg) { struct snd_pcm_substream *substream = arg; @@ -244,44 +254,48 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); -static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd, - dma_filter_fn filter_fn, void *filter_data) +/** + * snd_dmaengine_pcm_request_channel - Request channel for the dmaengine PCM + * @filter_fn: Filter function used to request the DMA channel + * @filter_data: Data passed to the DMA filter function + * + * Returns NULL or the requested DMA channel. + * + * This function request a DMA channel for usage with dmaengine PCM. + */ +struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, + void *filter_data) { dma_cap_mask_t mask; dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); dma_cap_set(DMA_CYCLIC, mask); - prtd->dma_chan = dma_request_channel(mask, filter_fn, filter_data); - if (!prtd->dma_chan) - return -ENXIO; - - return 0; + return dma_request_channel(mask, filter_fn, filter_data); } +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); /** * snd_dmaengine_pcm_open - Open a dmaengine based PCM substream * @substream: PCM substream - * @filter_fn: Filter function used to request the DMA channel - * @filter_data: Data passed to the DMA filter function + * @chan: DMA channel to use for data transfers * * Returns 0 on success, a negative error code otherwise. * - * This function will request a DMA channel using the passed filter function and - * data. The function should usually be called from the pcm open callback. - * - * 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_dmaengine_pcm_{set,get}_data. + * The function should usually be called from the pcm open callback. Note that + * this function will use private_data field of the substream's runtime. So it + * is not availabe to your pcm driver implementation. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, - dma_filter_fn filter_fn, void *filter_data) + struct dma_chan *chan) { struct dmaengine_pcm_runtime_data *prtd; int ret; + if (!chan) + return -ENXIO; + ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) @@ -291,11 +305,7 @@ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, if (!prtd) return -ENOMEM; - ret = dmaengine_pcm_request_channel(prtd, filter_fn, filter_data); - if (ret < 0) { - kfree(prtd); - return ret; - } + prtd->dma_chan = chan; substream->runtime->private_data = prtd; @@ -304,6 +314,27 @@ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); /** + * snd_dmaengine_pcm_open_request_chan - Open a dmaengine based PCM substream and request channel + * @substream: PCM substream + * @filter_fn: Filter function used to request the DMA channel + * @filter_data: Data passed to the DMA filter function + * + * Returns 0 on success, a negative error code otherwise. + * + * This function will request a DMA channel using the passed filter function and + * data. The function should usually be called from the pcm open callback. Note + * that this function will use private_data field of the substream's runtime. So + * it is not availabe to your pcm driver implementation. + */ +int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, + dma_filter_fn filter_fn, void *filter_data) +{ + return snd_dmaengine_pcm_open(substream, + snd_dmaengine_pcm_request_channel(filter_fn, filter_data)); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan); + +/** * snd_dmaengine_pcm_close - Close a dmaengine based PCM substream * @substream: PCM substream */ @@ -311,11 +342,26 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - dma_release_channel(prtd->dma_chan); kfree(prtd); return 0; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); +/** + * snd_dmaengine_pcm_release_chan_close - Close a dmaengine based PCM substream and release channel + * @substream: PCM substream + * + * Releases the DMA channel associated with the PCM substream. + */ +int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + dma_release_channel(prtd->dma_chan); + + return snd_dmaengine_pcm_close(substream); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close_release_chan); + MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c new file mode 100644 index 000000000000..e29ec3cd84b1 --- /dev/null +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -0,0 +1,300 @@ +/* + * Copyright (C) 2013, Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * 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. + * + * 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/module.h> +#include <linux/init.h> +#include <linux/dmaengine.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/dma-mapping.h> +#include <linux/of.h> + +#include <sound/dmaengine_pcm.h> + +struct dmaengine_pcm { + struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; + const struct snd_dmaengine_pcm_config *config; + struct snd_soc_platform platform; + unsigned int flags; +}; + +static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) +{ + return container_of(p, struct dmaengine_pcm, platform); +} + +/** + * snd_dmaengine_pcm_prepare_slave_config() - Generic prepare_slave_config callback + * @substream: PCM substream + * @params: hw_params + * @slave_config: DMA slave config to prepare + * + * This function can be used as a generic prepare_slave_config callback for + * platforms which make use of the snd_dmaengine_dai_dma_data struct for their + * DAI DMA data. Internally the function will first call + * snd_hwparams_to_dma_slave_config to fill in the slave config based on the + * hw_params, followed by snd_dmaengine_set_config_from_dai_data to fill in the + * remaining fields based on the DAI DMA data. + */ +int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_dmaengine_dai_dma_data *dma_data; + int ret; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); + if (ret) + return ret; + + snd_dmaengine_pcm_set_config_from_dai_data(substream, dma_data, + slave_config); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_prepare_slave_config); + +static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct dma_slave_config slave_config; + int ret; + + if (pcm->config->prepare_slave_config) { + ret = pcm->config->prepare_slave_config(substream, params, + &slave_config); + if (ret) + return ret; + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret) + return ret; + } + + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); +} + +static int dmaengine_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct dma_chan *chan = pcm->chan[substream->stream]; + int ret; + + ret = snd_soc_set_runtime_hwparams(substream, + pcm->config->pcm_hardware); + if (ret) + return ret; + + return snd_dmaengine_pcm_open(substream, chan); +} + +static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, + struct snd_pcm_substream *substream) +{ + if (!pcm->chan[substream->stream]) + return NULL; + + return pcm->chan[substream->stream]->device->dev; +} + +static void dmaengine_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static struct dma_chan *dmaengine_pcm_compat_request_channel( + struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) + return pcm->chan[0]; + + if (pcm->config->compat_request_channel) + return pcm->config->compat_request_channel(rtd, substream); + + return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); +} + +static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + const struct snd_dmaengine_pcm_config *config = pcm->config; + struct snd_pcm_substream *substream; + unsigned int i; + int ret; + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + substream = rtd->pcm->streams[i].substream; + if (!substream) + continue; + + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { + pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, + substream); + } + + if (!pcm->chan[i]) { + dev_err(rtd->platform->dev, + "Missing dma channel for stream: %d\n", i); + ret = -EINVAL; + goto err_free; + } + + ret = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + dmaengine_dma_dev(pcm, substream), + config->prealloc_buffer_size, + config->pcm_hardware->buffer_bytes_max); + if (ret) + goto err_free; + } + + return 0; + +err_free: + dmaengine_pcm_free(rtd->pcm); + return ret; +} + +static const struct snd_pcm_ops dmaengine_pcm_ops = { + .open = dmaengine_pcm_open, + .close = snd_dmaengine_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = dmaengine_pcm_hw_params, + .hw_free = snd_pcm_lib_free_pages, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, +}; + +static const struct snd_soc_platform_driver dmaengine_pcm_platform = { + .ops = &dmaengine_pcm_ops, + .pcm_new = dmaengine_pcm_new, + .pcm_free = dmaengine_pcm_free, + .probe_order = SND_SOC_COMP_ORDER_LATE, +}; + +static const struct snd_pcm_ops dmaengine_no_residue_pcm_ops = { + .open = dmaengine_pcm_open, + .close = snd_dmaengine_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = dmaengine_pcm_hw_params, + .hw_free = snd_pcm_lib_free_pages, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer_no_residue, +}; + +static const struct snd_soc_platform_driver dmaengine_no_residue_pcm_platform = { + .ops = &dmaengine_no_residue_pcm_ops, + .pcm_new = dmaengine_pcm_new, + .pcm_free = dmaengine_pcm_free, + .probe_order = SND_SOC_COMP_ORDER_LATE, +}; + +static const char * const dmaengine_pcm_dma_channel_names[] = { + [SNDRV_PCM_STREAM_PLAYBACK] = "tx", + [SNDRV_PCM_STREAM_CAPTURE] = "rx", +}; + +static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, + struct device *dev) +{ + unsigned int i; + + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) + return; + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { + pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx"); + pcm->chan[1] = pcm->chan[0]; + } else { + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + pcm->chan[i] = dma_request_slave_channel(dev, + dmaengine_pcm_dma_channel_names[i]); + } + } +} + +/** + * snd_dmaengine_pcm_register - Register a dmaengine based PCM device + * @dev: The parent device for the PCM device + * @config: Platform specific PCM configuration + * @flags: Platform specific quirks + */ +int snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, unsigned int flags) +{ + struct dmaengine_pcm *pcm; + + pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->config = config; + pcm->flags = flags; + + dmaengine_pcm_request_chan_of(pcm, dev); + + if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) + return snd_soc_add_platform(dev, &pcm->platform, + &dmaengine_no_residue_pcm_platform); + else + return snd_soc_add_platform(dev, &pcm->platform, + &dmaengine_pcm_platform); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); + +/** + * snd_dmaengine_pcm_unregister - Removes a dmaengine based PCM device + * @dev: Parent device the PCM was register with + * + * Removes a dmaengine based PCM device previously registered with + * snd_dmaengine_pcm_register. + */ +void snd_dmaengine_pcm_unregister(struct device *dev) +{ + struct snd_soc_platform *platform; + struct dmaengine_pcm *pcm; + unsigned int i; + + platform = snd_soc_lookup_platform(dev); + if (!platform) + return; + + pcm = soc_platform_to_pcm(platform); + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + if (pcm->chan[i]) { + dma_release_channel(pcm->chan[i]); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; + } + } + + snd_soc_remove_platform(platform); + kfree(pcm); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 29183ef2b93d..8ca9ecc5ac57 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -158,10 +158,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return -EINVAL; } - if (IS_ERR(codec->control_data)) - return PTR_ERR(codec->control_data); - - return 0; + return PTR_RET(codec->control_data); } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); #else diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index fe4541df498c..4b3be6c3c91e 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -90,8 +90,33 @@ static struct snd_soc_platform_driver dummy_platform = { }; static struct snd_soc_codec_driver dummy_codec; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) static struct snd_soc_dai_driver dummy_dai = { .name = "snd-soc-dummy-dai", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, }; static int snd_soc_dummy_probe(struct platform_device *pdev) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index c7c4b20395bb..14d57e89bcba 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -170,6 +170,10 @@ struct snd_soc_dai_driver spdif_in_dai = { .ops = &spdif_in_dai_ops, }; +static const struct snd_soc_component_driver spdif_in_component = { + .name = "spdif-in", +}; + static irqreturn_t spdif_in_irq(int irq, void *arg) { struct spdif_in_dev *host = (struct spdif_in_dev *)arg; @@ -258,7 +262,8 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai); + ret = snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); if (ret != 0) { clk_put(host->clk); return ret; @@ -271,7 +276,7 @@ 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); + snd_soc_unregister_component(&pdev->dev); dev_set_drvdata(&pdev->dev, NULL); clk_put(host->clk); diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 5eac4cda2fd7..1e3c3dda3598 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -270,6 +270,10 @@ static struct snd_soc_dai_driver spdif_out_dai = { .ops = &spdif_out_dai_ops, }; +static const struct snd_soc_component_driver spdif_out_component = { + .name = "spdif-out", +}; + static int spdif_out_probe(struct platform_device *pdev) { struct spdif_out_dev *host; @@ -314,7 +318,8 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai); + ret = snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); if (ret != 0) { clk_put(host->clk); return ret; @@ -327,7 +332,7 @@ 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); + snd_soc_unregister_component(&pdev->dev); dev_set_drvdata(&pdev->dev, NULL); clk_put(host->clk); diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 5e7aebe1e664..2fbd4899d8ef 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -25,7 +25,7 @@ #include <sound/soc.h> #include <sound/spear_dma.h> -struct snd_pcm_hardware spear_pcm_hardware = { +static 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), @@ -64,21 +64,8 @@ static int spear_pcm_open(struct snd_pcm_substream *substream) 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; + return snd_dmaengine_pcm_open_request_chan(substream, dma_data->filter, + dma_data); } static int spear_pcm_mmap(struct snd_pcm_substream *substream, @@ -93,7 +80,7 @@ static int spear_pcm_mmap(struct snd_pcm_substream *substream, static struct snd_pcm_ops spear_pcm_ops = { .open = spear_pcm_open, - .close = spear_pcm_close, + .close = snd_dmaengine_pcm_close_release_chan, .ioctl = snd_pcm_lib_ioctl, .hw_params = spear_pcm_hw_params, .hw_free = spear_pcm_hw_free, @@ -178,7 +165,7 @@ static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -struct snd_soc_platform_driver spear_soc_platform = { +static struct snd_soc_platform_driver spear_soc_platform = { .ops = &spear_pcm_ops, .pcm_new = spear_pcm_new, .pcm_free = spear_pcm_free, diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index dbc27ce1d4de..b1c9d573da05 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -2,7 +2,7 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on ARCH_TEGRA && TEGRA20_APB_DMA select REGMAP_MMIO - select SND_SOC_DMAENGINE_PCM if TEGRA20_APB_DMA + select SND_SOC_GENERIC_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_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 336dcdd3e8a4..2f70ea7f6618 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -35,6 +35,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> #include "tegra_asoc_utils.h" #include "tegra20_ac97.h" @@ -248,6 +249,10 @@ static struct snd_soc_dai_driver tegra20_ac97_dai = { .ops = &tegra20_ac97_dai_ops, }; +static const struct snd_soc_component_driver tegra20_ac97_component = { + .name = DRV_NAME, +}; + static bool tegra20_ac97_wr_rd_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -389,16 +394,17 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } ac97->capture_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_RX1; - ac97->capture_dma_data.wrap = 4; - ac97->capture_dma_data.width = 32; - ac97->capture_dma_data.req_sel = of_dma[1]; + ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + ac97->capture_dma_data.maxburst = 4; + ac97->capture_dma_data.slave_id = of_dma[1]; ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1; - ac97->playback_dma_data.wrap = 4; - ac97->playback_dma_data.width = 32; - ac97->playback_dma_data.req_sel = of_dma[1]; + ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + ac97->capture_dma_data.maxburst = 4; + ac97->capture_dma_data.slave_id = of_dma[0]; - ret = snd_soc_register_dais(&pdev->dev, &tegra20_ac97_dai, 1); + ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, + &tegra20_ac97_dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -408,7 +414,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = tegra_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_dai; + goto err_unregister_component; } ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); @@ -434,8 +440,8 @@ err_asoc_utils_fini: tegra_asoc_utils_fini(&ac97->util_data); err_unregister_pcm: tegra_pcm_platform_unregister(&pdev->dev); -err_unregister_dai: - snd_soc_unregister_dai(&pdev->dev); +err_unregister_component: + snd_soc_unregister_component(&pdev->dev); err_clk_put: clk_put(ac97->clk_ac97); err: @@ -447,7 +453,7 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) struct tegra20_ac97 *ac97 = dev_get_drvdata(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); tegra_asoc_utils_fini(&ac97->util_data); diff --git a/sound/soc/tegra/tegra20_ac97.h b/sound/soc/tegra/tegra20_ac97.h index dddc6828004e..4acb3aaba29b 100644 --- a/sound/soc/tegra/tegra20_ac97.h +++ b/sound/soc/tegra/tegra20_ac97.h @@ -85,8 +85,8 @@ struct tegra20_ac97 { struct clk *clk_ac97; - struct tegra_pcm_dma_params capture_dma_data; - struct tegra_pcm_dma_params playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct snd_dmaengine_dai_dma_data playback_dma_data; struct regmap *regmap; int reset_gpio; int sync_gpio; diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index caa772de5a18..52af7f6fb37f 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -41,6 +41,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> #include "tegra20_i2s.h" @@ -276,6 +277,10 @@ static const struct snd_soc_dai_driver tegra20_i2s_dai_template = { .symmetric_rates = 1, }; +static const struct snd_soc_component_driver tegra20_i2s_component = { + .name = DRV_NAME, +}; + static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -403,14 +408,14 @@ static int tegra20_i2s_platform_probe(struct platform_device *pdev) } i2s->capture_dma_data.addr = mem->start + TEGRA20_I2S_FIFO2; - i2s->capture_dma_data.wrap = 4; - i2s->capture_dma_data.width = 32; - i2s->capture_dma_data.req_sel = dma_ch; + i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->capture_dma_data.maxburst = 4; + i2s->capture_dma_data.slave_id = dma_ch; i2s->playback_dma_data.addr = mem->start + TEGRA20_I2S_FIFO1; - i2s->playback_dma_data.wrap = 4; - i2s->playback_dma_data.width = 32; - i2s->playback_dma_data.req_sel = dma_ch; + i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->playback_dma_data.maxburst = 4; + i2s->playback_dma_data.slave_id = dma_ch; pm_runtime_enable(&pdev->dev); if (!pm_runtime_enabled(&pdev->dev)) { @@ -419,7 +424,8 @@ static int tegra20_i2s_platform_probe(struct platform_device *pdev) goto err_pm_disable; } - ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); + ret = snd_soc_register_component(&pdev->dev, &tegra20_i2s_component, + &i2s->dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -429,13 +435,13 @@ static int tegra20_i2s_platform_probe(struct platform_device *pdev) ret = tegra_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_dai; + goto err_unregister_component; } return 0; -err_unregister_dai: - snd_soc_unregister_dai(&pdev->dev); +err_unregister_component: + snd_soc_unregister_component(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) tegra20_i2s_runtime_suspend(&pdev->dev); @@ -456,7 +462,7 @@ static int tegra20_i2s_platform_remove(struct platform_device *pdev) tegra20_i2s_runtime_suspend(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); clk_put(i2s->clk_i2s); diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index 729958713cd4..fa6c29cc12b9 100644 --- a/sound/soc/tegra/tegra20_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -155,8 +155,8 @@ struct tegra20_i2s { struct snd_soc_dai_driver dai; struct clk *clk_i2s; - struct tegra_pcm_dma_params capture_dma_data; - struct tegra_pcm_dma_params playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct snd_dmaengine_dai_dma_data playback_dma_data; struct regmap *regmap; }; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 04771d14d343..5eaa12cdc6eb 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -32,6 +32,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> #include "tegra20_spdif.h" @@ -182,6 +183,10 @@ static struct snd_soc_dai_driver tegra20_spdif_dai = { .ops = &tegra20_spdif_dai_ops, }; +static const struct snd_soc_component_driver tegra20_spdif_component = { + .name = DRV_NAME, +}; + static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -318,9 +323,9 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; - spdif->playback_dma_data.wrap = 4; - spdif->playback_dma_data.width = 32; - spdif->playback_dma_data.req_sel = dmareq->start; + spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + spdif->capture_dma_data.maxburst = 4; + spdif->playback_dma_data.slave_id = dmareq->start; pm_runtime_enable(&pdev->dev); if (!pm_runtime_enabled(&pdev->dev)) { @@ -329,7 +334,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_pm_disable; } - ret = snd_soc_register_dai(&pdev->dev, &tegra20_spdif_dai); + ret = snd_soc_register_component(&pdev->dev, &tegra20_spdif_component, + &tegra20_spdif_dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -339,13 +345,13 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) ret = tegra_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_dai; + goto err_unregister_component; } return 0; -err_unregister_dai: - snd_soc_unregister_dai(&pdev->dev); +err_unregister_component: + snd_soc_unregister_component(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) tegra20_spdif_runtime_suspend(&pdev->dev); @@ -366,7 +372,7 @@ static int tegra20_spdif_platform_remove(struct platform_device *pdev) tegra20_spdif_runtime_suspend(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); clk_put(spdif->clk_spdif_out); diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h index b48d699fd583..85a9aefcc287 100644 --- a/sound/soc/tegra/tegra20_spdif.h +++ b/sound/soc/tegra/tegra20_spdif.h @@ -462,8 +462,8 @@ struct tegra20_spdif { struct clk *clk_spdif_out; - struct tegra_pcm_dma_params capture_dma_data; - struct tegra_pcm_dma_params playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct snd_dmaengine_dai_dma_data playback_dma_data; struct regmap *regmap; }; diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index e5cfb4ac41ba..23e592f453fa 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -95,8 +95,8 @@ static int tegra30_ahub_runtime_resume(struct device *dev) } int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, - unsigned long *fiforeg, - unsigned long *reqsel) + dma_addr_t *fiforeg, + unsigned int *reqsel) { int channel; u32 reg, val; @@ -178,8 +178,8 @@ int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif) EXPORT_SYMBOL_GPL(tegra30_ahub_free_rx_fifo); int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, - unsigned long *fiforeg, - unsigned long *reqsel) + dma_addr_t *fiforeg, + unsigned int *reqsel) { int channel; u32 reg, val; @@ -287,16 +287,27 @@ int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif) } EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); -static const char * const configlink_clocks[] = { - "i2s0", - "i2s1", - "i2s2", - "i2s3", - "i2s4", - "dam0", - "dam1", - "dam2", - "spdif_in", +#define CLK_LIST_MASK_TEGRA30 BIT(0) +#define CLK_LIST_MASK_TEGRA114 BIT(1) + +#define CLK_LIST_MASK_TEGRA30_OR_LATER \ + (CLK_LIST_MASK_TEGRA30 | CLK_LIST_MASK_TEGRA114) + +static const struct { + const char *clk_name; + u32 clk_list_mask; +} configlink_clocks[] = { + { "i2s0", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s1", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s2", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s3", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s4", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "dam0", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "dam1", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "dam2", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "spdif_in", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "amx", CLK_LIST_MASK_TEGRA114 }, + { "adx", CLK_LIST_MASK_TEGRA114 }, }; #define LAST_REG(name) \ @@ -424,8 +435,24 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { .cache_type = REGCACHE_RBTREE, }; +static struct tegra30_ahub_soc_data soc_data_tegra30 = { + .clk_list_mask = CLK_LIST_MASK_TEGRA30, +}; + +static struct tegra30_ahub_soc_data soc_data_tegra114 = { + .clk_list_mask = CLK_LIST_MASK_TEGRA114, +}; + +static const struct of_device_id tegra30_ahub_of_match[] = { + { .compatible = "nvidia,tegra114-ahub", .data = &soc_data_tegra114 }, + { .compatible = "nvidia,tegra30-ahub", .data = &soc_data_tegra30 }, + {}, +}; + static int tegra30_ahub_probe(struct platform_device *pdev) { + const struct of_device_id *match; + const struct tegra30_ahub_soc_data *soc_data; struct clk *clk; int i; struct resource *res0, *res1, *region; @@ -436,16 +463,24 @@ static int tegra30_ahub_probe(struct platform_device *pdev) if (ahub) return -ENODEV; + match = of_match_device(tegra30_ahub_of_match, &pdev->dev); + if (!match) + return -EINVAL; + soc_data = match->data; + /* * The AHUB hosts a register bus: the "configlink". For this to * operate correctly, all devices on this bus must be out of reset. * Ensure that here. */ for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { - clk = clk_get(&pdev->dev, configlink_clocks[i]); + if (!(configlink_clocks[i].clk_list_mask & + soc_data->clk_list_mask)) + continue; + clk = clk_get(&pdev->dev, configlink_clocks[i].clk_name); if (IS_ERR(clk)) { dev_err(&pdev->dev, "Can't get clock %s\n", - configlink_clocks[i]); + configlink_clocks[i].clk_name); ret = PTR_ERR(clk); goto err; } @@ -592,11 +627,6 @@ static int tegra30_ahub_remove(struct platform_device *pdev) return 0; } -static const struct of_device_id tegra30_ahub_of_match[] = { - { .compatible = "nvidia,tegra30-ahub", }, - {}, -}; - static const struct dev_pm_ops tegra30_ahub_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, tegra30_ahub_runtime_resume, NULL) diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h index e690e2eecc92..09766cdc45ca 100644 --- a/sound/soc/tegra/tegra30_ahub.h +++ b/sound/soc/tegra/tegra30_ahub.h @@ -451,15 +451,15 @@ enum tegra30_ahub_rxcif { }; extern int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, - unsigned long *fiforeg, - unsigned long *reqsel); + dma_addr_t *fiforeg, + unsigned int *reqsel); extern int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif); extern int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif); extern int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif); extern int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, - unsigned long *fiforeg, - unsigned long *reqsel); + dma_addr_t *fiforeg, + unsigned int *reqsel); extern int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif); extern int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif); extern int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif); @@ -468,7 +468,23 @@ extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, enum tegra30_ahub_txcif txcif); extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif); +struct tegra30_ahub_soc_data { + u32 clk_list_mask; + /* + * FIXME: There are many more differences in HW, such as: + * - More APBIF channels. + * - Extra separate chunks of register address space to represent + * the extra APBIF channels. + * - More units connected to the AHUB, so that tegra30_ahub_[rt]xcif + * need expansion, coupled with there being more defined bits in + * the AHUB routing registers. + * However, the driver doesn't support those new features yet, so we + * don't represent them here yet. + */ +}; + struct tegra30_ahub { + const struct tegra30_ahub_soc_data *soc_data; struct device *dev; struct clk *clk_d_audio; struct clk *clk_apbif; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index f4e1ce82750a..31d092d83c71 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -38,6 +38,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> #include "tegra30_ahub.h" #include "tegra30_i2s.h" @@ -80,17 +81,17 @@ static int tegra30_i2s_startup(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = tegra30_ahub_allocate_tx_fifo(&i2s->playback_fifo_cif, &i2s->playback_dma_data.addr, - &i2s->playback_dma_data.req_sel); - i2s->playback_dma_data.wrap = 4; - i2s->playback_dma_data.width = 32; + &i2s->playback_dma_data.slave_id); + i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->playback_dma_data.maxburst = 4; tegra30_ahub_set_rx_cif_source(i2s->playback_i2s_cif, i2s->playback_fifo_cif); } else { ret = tegra30_ahub_allocate_rx_fifo(&i2s->capture_fifo_cif, &i2s->capture_dma_data.addr, - &i2s->capture_dma_data.req_sel); - i2s->capture_dma_data.wrap = 4; - i2s->capture_dma_data.width = 32; + &i2s->capture_dma_data.slave_id); + i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->capture_dma_data.maxburst = 4; tegra30_ahub_set_rx_cif_source(i2s->capture_fifo_cif, i2s->capture_i2s_cif); } @@ -336,6 +337,10 @@ static const struct snd_soc_dai_driver tegra30_i2s_dai_template = { .symmetric_rates = 1, }; +static const struct snd_soc_component_driver tegra30_i2s_component = { + .name = DRV_NAME, +}; + static bool tegra30_i2s_wr_rd_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -464,7 +469,8 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) goto err_pm_disable; } - ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); + ret = snd_soc_register_component(&pdev->dev, &tegra30_i2s_component, + &i2s->dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -474,13 +480,13 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) ret = tegra_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_dai; + goto err_unregister_component; } return 0; -err_unregister_dai: - snd_soc_unregister_dai(&pdev->dev); +err_unregister_component: + snd_soc_unregister_component(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) tegra30_i2s_runtime_suspend(&pdev->dev); @@ -501,7 +507,7 @@ static int tegra30_i2s_platform_remove(struct platform_device *pdev) tegra30_i2s_runtime_suspend(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); clk_put(i2s->clk_i2s); diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index a294d942b9f7..bea23afe3b9f 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -231,10 +231,10 @@ struct tegra30_i2s { struct clk *clk_i2s; enum tegra30_ahub_txcif capture_i2s_cif; enum tegra30_ahub_rxcif capture_fifo_cif; - struct tegra_pcm_dma_params capture_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; enum tegra30_ahub_rxcif playback_i2s_cif; enum tegra30_ahub_txcif playback_fifo_cif; - struct tegra_pcm_dma_params playback_dma_data; + struct snd_dmaengine_dai_dma_data playback_dma_data; struct regmap *regmap; }; diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index c80adb9da472..48d05d9e1002 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -161,20 +161,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev) sizeof(struct tegra_alc5632), GFP_KERNEL); if (!alc5632) { dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, alc5632); - if (!(pdev->dev.of_node)) { - dev_err(&pdev->dev, "Must be instantiated using device tree\n"); - ret = -EINVAL; - goto err; - } - alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); if (alc5632->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; @@ -197,11 +190,11 @@ static int tegra_alc5632_probe(struct platform_device *pdev) goto err; } - tegra_alc5632_dai.cpu_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,i2s-controller", 0); + tegra_alc5632_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); if (!tegra_alc5632_dai.cpu_of_node) { dev_err(&pdev->dev, - "Property 'nvidia,i2s-controller' missing or invalid\n"); + "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; goto err; } diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index ba419f86384d..24fb001be7f4 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -43,8 +43,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 88200: if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) new_baseclock = 56448000; - else + else if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA30) new_baseclock = 564480000; + else + new_baseclock = 282240000; break; case 8000: case 16000: @@ -54,8 +56,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 96000: if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) new_baseclock = 73728000; - else + else if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA30) new_baseclock = 552960000; + else + new_baseclock = 368640000; break; default: return -EINVAL; @@ -169,6 +173,7 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev) { int ret; + bool new_clocks = false; data->dev = dev; @@ -176,28 +181,37 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else if (of_machine_is_compatible("nvidia,tegra30")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; - else if (!dev->of_node) - /* non-DT is always Tegra20 */ - data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; - else - /* DT boot, but unknown SoC */ + else if (of_machine_is_compatible("nvidia,tegra114")) { + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA114; + new_clocks = true; + } else { + dev_err(data->dev, "SoC unknown to Tegra ASoC utils\n"); return -EINVAL; + } - data->clk_pll_a = clk_get_sys(NULL, "pll_a"); + if (new_clocks) + data->clk_pll_a = clk_get(dev, "pll_a"); + else + data->clk_pll_a = clk_get_sys(NULL, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); ret = PTR_ERR(data->clk_pll_a); goto err; } - data->clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); + if (new_clocks) + data->clk_pll_a_out0 = clk_get(dev, "pll_a_out0"); + else + data->clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); if (IS_ERR(data->clk_pll_a_out0)) { dev_err(data->dev, "Can't retrieve clk pll_a_out0\n"); ret = PTR_ERR(data->clk_pll_a_out0); goto err_put_pll_a; } - if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + if (new_clocks) + data->clk_cdev1 = clk_get(dev, "mclk"); + else if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); else data->clk_cdev1 = clk_get_sys("extern1", NULL); diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 974c9f8830f9..19fdcafed32f 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -29,6 +29,7 @@ struct device; enum tegra_asoc_utils_soc { TEGRA_ASOC_UTILS_SOC_TEGRA20, TEGRA_ASOC_UTILS_SOC_TEGRA30, + TEGRA_ASOC_UTILS_SOC_TEGRA114, }; struct tegra_asoc_utils_data { diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 5e2c55c5b255..f056f632557c 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -29,9 +29,7 @@ * */ -#include <linux/dma-mapping.h> #include <linux/module.h> -#include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -55,191 +53,24 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .fifo_size = 4, }; -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.dst_maxburst = 4; - } else { - slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - slave_config.src_addr = dmap->addr; - slave_config.src_maxburst = 4; - } - 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_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 = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, - .mmap = tegra_pcm_mmap, -}; - -static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = tegra_pcm_hardware.buffer_bytes_max; - - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->bytes = size; - - return 0; -} - -static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - - substream = pcm->streams[stream].substream; - if (!substream) - return; - - buf = &substream->dma_buffer; - if (!buf->area) - return; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; -} - -static u64 tegra_dma_mask = DMA_BIT_MASK(32); - -static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &tegra_dma_mask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto err; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto err_free_play; - } - - return 0; - -err_free_play: - tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); -err: - return ret; -} - -static void tegra_pcm_free(struct snd_pcm *pcm) -{ - tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); - tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); -} - -static struct snd_soc_platform_driver tegra_pcm_platform = { - .ops = &tegra_pcm_ops, - .pcm_new = tegra_pcm_new, - .pcm_free = tegra_pcm_free, +static const struct snd_dmaengine_pcm_config tegra_dmaengine_pcm_config = { + .pcm_hardware = &tegra_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .compat_filter_fn = NULL, + .prealloc_buffer_size = PAGE_SIZE * 8, }; int tegra_pcm_platform_register(struct device *dev) { - return snd_soc_register_platform(dev, &tegra_pcm_platform); + return snd_dmaengine_pcm_register(dev, &tegra_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT); } EXPORT_SYMBOL_GPL(tegra_pcm_platform_register); void tegra_pcm_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(dev); + return snd_dmaengine_pcm_unregister(dev); } EXPORT_SYMBOL_GPL(tegra_pcm_platform_unregister); diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h index bc8b46af928e..68ad901714a9 100644 --- a/sound/soc/tegra/tegra_pcm.h +++ b/sound/soc/tegra/tegra_pcm.h @@ -31,13 +31,6 @@ #ifndef __TEGRA_PCM_H__ #define __TEGRA_PCM_H__ -struct tegra_pcm_dma_params { - unsigned long addr; - unsigned long wrap; - unsigned long width; - unsigned long req_sel; -}; - 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 c8ef88a67c59..f87fc53e9b8c 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -124,6 +124,7 @@ static struct snd_soc_card snd_soc_tegra_wm8753 = { static int tegra_wm8753_driver_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &snd_soc_tegra_wm8753; struct tegra_wm8753 *machine; int ret; @@ -132,8 +133,7 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev) GFP_KERNEL); if (!machine) { dev_err(&pdev->dev, "Can't allocate tegra_wm8753 struct\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } card->dev = &pdev->dev; @@ -148,8 +148,8 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev) if (ret) goto err; - tegra_wm8753_dai.codec_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,audio-codec", 0); + tegra_wm8753_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); if (!tegra_wm8753_dai.codec_of_node) { dev_err(&pdev->dev, "Property 'nvidia,audio-codec' missing or invalid\n"); @@ -157,8 +157,8 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev) goto err; } - tegra_wm8753_dai.cpu_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,i2s-controller", 0); + tegra_wm8753_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); if (!tegra_wm8753_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); @@ -166,8 +166,7 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev) goto err; } - tegra_wm8753_dai.platform_of_node = - tegra_wm8753_dai.cpu_of_node; + tegra_wm8753_dai.platform_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 bbd79bf56303..4ac73730d79a 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -39,7 +39,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/tegra_wm8903.h> #include "../codecs/wm8903.h" @@ -48,7 +47,11 @@ #define DRV_NAME "tegra-snd-wm8903" struct tegra_wm8903 { - struct tegra_wm8903_platform_data pdata; + int gpio_spkr_en; + int gpio_hp_det; + int gpio_hp_mute; + int gpio_int_mic_en; + int gpio_ext_mic_en; struct tegra_asoc_utils_data util_data; }; @@ -129,12 +132,11 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (!gpio_is_valid(pdata->gpio_spkr_en)) + if (!gpio_is_valid(machine->gpio_spkr_en)) return 0; - gpio_set_value_cansleep(pdata->gpio_spkr_en, + gpio_set_value_cansleep(machine->gpio_spkr_en, SND_SOC_DAPM_EVENT_ON(event)); return 0; @@ -146,12 +148,11 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (!gpio_is_valid(pdata->gpio_hp_mute)) + if (!gpio_is_valid(machine->gpio_hp_mute)) return 0; - gpio_set_value_cansleep(pdata->gpio_hp_mute, + gpio_set_value_cansleep(machine->gpio_hp_mute, !SND_SOC_DAPM_EVENT_ON(event)); return 0; @@ -163,17 +164,6 @@ static const struct snd_soc_dapm_widget tegra_wm8903_dapm_widgets[] = { SND_SOC_DAPM_MIC("Mic Jack", NULL), }; -static const struct snd_soc_dapm_route harmony_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"}, - {"IN1L", NULL, "Mic Jack"}, -}; - static const struct snd_kcontrol_new tegra_wm8903_controls[] = { SOC_DAPM_PIN_SWITCH("Int Spk"), }; @@ -185,10 +175,9 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) 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; - if (gpio_is_valid(pdata->gpio_hp_det)) { - tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det; + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det; snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &tegra_wm8903_hp_jack); snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, @@ -226,9 +215,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) static struct snd_soc_dai_link tegra_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", - .codec_name = "wm8903.0-001a", - .platform_name = "tegra20-i2s.0", - .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "wm8903-hifi", .init = tegra_wm8903_init, .ops = &tegra_wm8903_ops, @@ -257,96 +243,25 @@ static 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) { - dev_err(&pdev->dev, "No platform data supplied\n"); - return -EINVAL; - } - machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8903), GFP_KERNEL); if (!machine) { dev_err(&pdev->dev, "Can't allocate tegra_wm8903 struct\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } - pdata = &machine->pdata; card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - 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 == -EPROBE_DEFER) - return -EPROBE_DEFER; - - pdata->gpio_hp_mute = of_get_named_gpio(np, - "nvidia,hp-mute-gpios", 0); - if (pdata->gpio_hp_mute == -EPROBE_DEFER) - return -EPROBE_DEFER; - - pdata->gpio_hp_det = of_get_named_gpio(np, - "nvidia,hp-det-gpios", 0); - if (pdata->gpio_hp_det == -EPROBE_DEFER) - 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 == -EPROBE_DEFER) - 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 == -EPROBE_DEFER) - return -EPROBE_DEFER; - } - - if (np) { - ret = snd_soc_of_parse_card_name(card, "nvidia,model"); - if (ret) - goto err; - - ret = snd_soc_of_parse_audio_routing(card, - "nvidia,audio-routing"); - if (ret) - goto err; - - tegra_wm8903_dai.codec_name = NULL; - 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"); - ret = -EINVAL; - goto err; - } - - tegra_wm8903_dai.cpu_dai_name = NULL; - 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; - goto err; - } - - tegra_wm8903_dai.platform_name = NULL; - tegra_wm8903_dai.platform_of_node = - tegra_wm8903_dai.cpu_of_node; - } else { - 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, + machine->gpio_spkr_en = of_get_named_gpio(np, "nvidia,spkr-en-gpios", + 0); + if (machine->gpio_spkr_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_spkr_en)) { + ret = devm_gpio_request_one(&pdev->dev, machine->gpio_spkr_en, GPIOF_OUT_INIT_LOW, "spkr_en"); if (ret) { dev_err(card->dev, "cannot get spkr_en gpio\n"); @@ -354,8 +269,12 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) } } - if (gpio_is_valid(pdata->gpio_hp_mute)) { - ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_hp_mute, + machine->gpio_hp_mute = of_get_named_gpio(np, "nvidia,hp-mute-gpios", + 0); + if (machine->gpio_hp_mute == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_hp_mute)) { + ret = devm_gpio_request_one(&pdev->dev, machine->gpio_hp_mute, GPIOF_OUT_INIT_HIGH, "hp_mute"); if (ret) { dev_err(card->dev, "cannot get hp_mute gpio\n"); @@ -363,9 +282,18 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) } } - if (gpio_is_valid(pdata->gpio_int_mic_en)) { + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_int_mic_en = of_get_named_gpio(np, + "nvidia,int-mic-en-gpios", 0); + if (machine->gpio_int_mic_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_int_mic_en)) { /* Disable int mic; enable signal is active-high */ - ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_int_mic_en, + ret = devm_gpio_request_one(&pdev->dev, + machine->gpio_int_mic_en, GPIOF_OUT_INIT_LOW, "int_mic_en"); if (ret) { dev_err(card->dev, "cannot get int_mic_en gpio\n"); @@ -373,9 +301,14 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) } } - if (gpio_is_valid(pdata->gpio_ext_mic_en)) { + machine->gpio_ext_mic_en = of_get_named_gpio(np, + "nvidia,ext-mic-en-gpios", 0); + if (machine->gpio_ext_mic_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_ext_mic_en)) { /* Enable ext mic; enable signal is active-low */ - ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_ext_mic_en, + ret = devm_gpio_request_one(&pdev->dev, + machine->gpio_ext_mic_en, GPIOF_OUT_INIT_LOW, "ext_mic_en"); if (ret) { dev_err(card->dev, "cannot get ext_mic_en gpio\n"); @@ -383,6 +316,34 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) } } + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + 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"); + ret = -EINVAL; + goto err; + } + + 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; + goto err; + } + + tegra_wm8903_dai.platform_of_node = tegra_wm8903_dai.cpu_of_node; + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) goto err; diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 68d42403d9b5..5e119630b0e0 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -55,7 +55,7 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", - .cpu_dai_name = "tegra-ac97-pcm", + .cpu_dai_name = "tegra20-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .init = tegra_wm9712_init, @@ -79,11 +79,6 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) struct tegra_wm9712 *machine; int ret; - if (!pdev->dev.of_node) { - dev_err(&pdev->dev, "No platform data supplied\n"); - return -EINVAL; - } - machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm9712), GFP_KERNEL); if (!machine) { diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 7fcf6c2297db..05c68aab5cf0 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -97,9 +97,6 @@ static const struct snd_soc_dapm_route trimslice_audio_map[] = { static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", - .codec_name = "tlv320aic23-codec.2-001a", - .platform_name = "tegra20-i2s.0", - .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "tlv320aic23-hifi", .ops = &trimslice_asoc_ops, .dai_fmt = SND_SOC_DAIFMT_I2S | @@ -122,6 +119,7 @@ static struct snd_soc_card snd_soc_trimslice = { static int tegra_snd_trimslice_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &snd_soc_trimslice; struct tegra_trimslice *trimslice; int ret; @@ -130,44 +128,38 @@ static int tegra_snd_trimslice_probe(struct platform_device *pdev) GFP_KERNEL); if (!trimslice) { dev_err(&pdev->dev, "Can't allocate tegra_trimslice\n"); - ret = -ENOMEM; + return -ENOMEM; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, trimslice); + + trimslice_tlv320aic23_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!trimslice_tlv320aic23_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; goto err; } - if (pdev->dev.of_node) { - trimslice_tlv320aic23_dai.codec_name = NULL; - trimslice_tlv320aic23_dai.codec_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,audio-codec", 0); - if (!trimslice_tlv320aic23_dai.codec_of_node) { - dev_err(&pdev->dev, - "Property 'nvidia,audio-codec' missing or invalid\n"); - ret = -EINVAL; - goto err; - } - - trimslice_tlv320aic23_dai.cpu_dai_name = NULL; - trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!trimslice_tlv320aic23_dai.cpu_of_node) { - dev_err(&pdev->dev, - "Property 'nvidia,i2s-controller' missing or invalid\n"); - ret = -EINVAL; - goto err; - } - - trimslice_tlv320aic23_dai.platform_name = NULL; - trimslice_tlv320aic23_dai.platform_of_node = - trimslice_tlv320aic23_dai.cpu_of_node; + trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!trimslice_tlv320aic23_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; } + trimslice_tlv320aic23_dai.platform_of_node = + trimslice_tlv320aic23_dai.cpu_of_node; + ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev); if (ret) goto err; - card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); - snd_soc_card_set_drvdata(card, trimslice); - ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 16ab69635e2e..8a2840304d28 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -170,6 +170,10 @@ static struct snd_soc_dai_driver txx9aclc_ac97_dai = { }, }; +static const struct snd_soc_component_driver txx9aclc_ac97_component = { + .name = "txx9aclc-ac97", +}; + static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) { struct txx9aclc_plat_drvdata *drvdata; @@ -205,12 +209,13 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (err < 0) return err; - return snd_soc_register_dai(&pdev->dev, &txx9aclc_ac97_dai); + return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, + &txx9aclc_ac97_dai, 1); } static int txx9aclc_ac97_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 6b799c0e88df..aa5011894c74 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_UX500_PLAT_MSP_I2S config SND_SOC_UX500_PLAT_DMA tristate "Platform - DB8500 (DMA)" depends on SND_SOC_UX500 - select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to enable the Ux500 platform-driver. diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 54028cf76511..7d5fc1328523 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -766,6 +766,11 @@ static struct snd_soc_dai_driver ux500_msp_dai_drv[UX500_NBR_OF_DAI] = { }, }; +static const struct snd_soc_component_driver ux500_msp_component = { + .name = "ux500-msp", +}; + + static int ux500_msp_drv_probe(struct platform_device *pdev) { struct ux500_msp_i2s_drvdata *drvdata; @@ -823,8 +828,8 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, drvdata); - ret = snd_soc_register_dai(&pdev->dev, - &ux500_msp_dai_drv[drvdata->msp->id]); + ret = snd_soc_register_component(&pdev->dev, &ux500_msp_component, + &ux500_msp_dai_drv[drvdata->msp->id], 1); if (ret < 0) { dev_err(&pdev->dev, "Error: %s: Failed to register MSP%d!\n", __func__, drvdata->msp->id); @@ -842,7 +847,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) return 0; err_reg_plat: - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(ux500_msp_dai_drv)); + snd_soc_unregister_component(&pdev->dev); err_init_msp: clk_put(drvdata->clk); err_clk: @@ -859,7 +864,7 @@ static int ux500_msp_drv_remove(struct platform_device *pdev) ux500_pcm_unregister_platform(pdev); - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(ux500_msp_dai_drv)); + snd_soc_unregister_component(&pdev->dev); devm_regulator_put(drvdata->reg_vape); prcmu_qos_remove_requirement(PRCMU_QOS_APE_OPP, "ux500_msp_i2s"); diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h index 9c778d9c3838..f53104359f15 100644 --- a/sound/soc/ux500/ux500_msp_dai.h +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -35,13 +35,8 @@ #define FRAME_PER_8_SLOTS 138 #define FRAME_PER_16_SLOTS 277 -#ifndef CONFIG_SND_SOC_UX500_AB5500 #define UX500_MSP_INTERNAL_CLOCK_FREQ 40000000 #define UX500_MSP1_INTERNAL_CLOCK_FREQ UX500_MSP_INTERNAL_CLOCK_FREQ -#else -#define UX500_MSP_INTERNAL_CLOCK_FREQ 13000000 -#define UX500_MSP1_INTERNAL_CLOCK_FREQ (UX500_MSP_INTERNAL_CLOCK_FREQ * 2) -#endif #define UX500_MSP_MIN_CHANNELS 1 #define UX500_MSP_MAX_CHANNELS 8 diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 437f0c032c58..e5cd105c90f9 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -541,6 +541,7 @@ struct ux500_msp_dma_params { struct stedma40_chan_cfg *dma_cfg; }; +struct msp_i2s_platform_data; int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct ux500_msp **msp_p, struct msp_i2s_platform_data *platform_data); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 846fa82a58d0..b6e5ae277299 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -28,28 +28,19 @@ #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, -}; +#define UX500_PLATFORM_MIN_RATE 8000 +#define UX500_PLATFORM_MAX_RATE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 -static struct snd_pcm_hardware ux500_pcm_hw_capture = { +#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) + +static const struct snd_pcm_hardware ux500_pcm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_RESUME | @@ -59,8 +50,8 @@ static struct snd_pcm_hardware ux500_pcm_hw_capture = { 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, + .rate_min = UX500_PLATFORM_MIN_RATE, + .rate_max = UX500_PLATFORM_MAX_RATE, .channels_min = UX500_PLATFORM_MIN_CHANNELS, .channels_max = UX500_PLATFORM_MAX_CHANNELS, .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, @@ -70,64 +61,23 @@ static struct snd_pcm_hardware ux500_pcm_hw_capture = { .periods_max = UX500_PLATFORM_PERIODS_MAX, }; -static void ux500_pcm_dma_hw_free(struct device *dev, - struct snd_pcm_substream *substream) +static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, + 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; + struct ux500_msp_dma_params *dma_params; 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; + dma_params = snd_soc_dai_get_dma_data(dai, substream); + dma_cfg = dma_params->dma_cfg; 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; @@ -140,13 +90,8 @@ static int ux500_pcm_open(struct snd_pcm_substream *substream) 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; @@ -155,137 +100,24 @@ static int ux500_pcm_open(struct snd_pcm_substream *substream) 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; + return snd_dmaengine_pcm_request_channel(stedma40_filter, dma_cfg); } -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 const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = { + .pcm_hardware = &ux500_pcm_hw, + .compat_request_channel = ux500_pcm_request_chan, + .prealloc_buffer_size = 128 * 1024, }; int ux500_pcm_register_platform(struct platform_device *pdev) { int ret; - ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + ret = snd_dmaengine_pcm_register(&pdev->dev, + &ux500_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_COMPAT | + SND_DMAENGINE_PCM_FLAG_NO_DT); if (ret < 0) { dev_err(&pdev->dev, "%s: ERROR: Failed to register platform '%s' (%d)!\n", @@ -299,8 +131,7 @@ EXPORT_SYMBOL_GPL(ux500_pcm_register_platform); int ux500_pcm_unregister_platform(struct platform_device *pdev) { - snd_soc_unregister_platform(&pdev->dev); - + snd_dmaengine_pcm_unregister(&pdev->dev); return 0; } EXPORT_SYMBOL_GPL(ux500_pcm_unregister_platform); diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h index 76d344476afc..d76e1aff6458 100644 --- a/sound/soc/ux500/ux500_pcm.h +++ b/sound/soc/ux500/ux500_pcm.h @@ -18,20 +18,6 @@ #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) - int ux500_pcm_register_platform(struct platform_device *pdev); int ux500_pcm_unregister_platform(struct platform_device *pdev); diff --git a/sound/sound_core.c b/sound/sound_core.c index bb23009edc8d..359753fc24e1 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -352,7 +352,9 @@ static struct sound_unit *chains[SOUND_STEP]; * @dev: device pointer * * Allocate a special sound device by minor number from the sound - * subsystem. The allocated number is returned on success. On failure + * subsystem. + * + * Return: The allocated number is returned on success. On failure, * a negative error code is returned. */ @@ -436,8 +438,10 @@ EXPORT_SYMBOL(register_sound_special); * @dev: Unit number to allocate * * Allocate a mixer device. Unit is the number of the mixer requested. - * Pass -1 to request the next free mixer unit. On success the allocated - * number is returned, on failure a negative error code is returned. + * Pass -1 to request the next free mixer unit. + * + * Return: On success, the allocated number is returned. On failure, + * a negative error code is returned. */ int register_sound_mixer(const struct file_operations *fops, int dev) @@ -454,8 +458,10 @@ EXPORT_SYMBOL(register_sound_mixer); * @dev: Unit number to allocate * * Allocate a midi device. Unit is the number of the midi device requested. - * Pass -1 to request the next free midi unit. On success the allocated - * number is returned, on failure a negative error code is returned. + * Pass -1 to request the next free midi unit. + * + * Return: On success, the allocated number is returned. On failure, + * a negative error code is returned. */ int register_sound_midi(const struct file_operations *fops, int dev) @@ -477,11 +483,13 @@ EXPORT_SYMBOL(register_sound_midi); * @dev: Unit number to allocate * * Allocate a DSP device. Unit is the number of the DSP requested. - * Pass -1 to request the next free DSP unit. On success the allocated - * number is returned, on failure a negative error code is returned. + * Pass -1 to request the next free DSP unit. * * This function allocates both the audio and dsp device entries together * and will always allocate them as a matching pair - eg dsp3/audio3 + * + * Return: On success, the allocated number is returned. On failure, + * a negative error code is returned. */ int register_sound_dsp(const struct file_operations *fops, int dev) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4dd60d8a4889..a1a24b979ed2 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1075,10 +1075,11 @@ out: return 0; } -#ifdef CONFIG_PM -static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) +#ifdef CONFIG_PM_SLEEP + +static int snd_at73c213_suspend(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); @@ -1087,9 +1088,9 @@ static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) return 0; } -static int snd_at73c213_resume(struct spi_device *spi) +static int snd_at73c213_resume(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; clk_enable(chip->board->dac_clk); @@ -1097,18 +1098,21 @@ static int snd_at73c213_resume(struct spi_device *spi) return 0; } + +static SIMPLE_DEV_PM_OPS(at73c213_pm_ops, snd_at73c213_suspend, + snd_at73c213_resume); +#define AT73C213_PM_OPS (&at73c213_pm_ops) + #else -#define snd_at73c213_suspend NULL -#define snd_at73c213_resume NULL +#define AT73C213_PM_OPS NULL #endif static struct spi_driver at73c213_driver = { .driver = { .name = "at73c213", + .pm = AT73C213_PM_OPS, }, .probe = snd_at73c213_probe, - .suspend = snd_at73c213_suspend, - .resume = snd_at73c213_resume, .remove = snd_at73c213_remove, }; diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index e2ca12fe92e9..40dd50a80f55 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -575,7 +575,6 @@ static void usb6fire_pcm_init_urb(struct pcm_urb *urb, urb->instance.pipe = in ? usb_rcvisocpipe(chip->dev, ep) : usb_sndisocpipe(chip->dev, ep); urb->instance.interval = 1; - urb->instance.transfer_flags = URB_ISO_ASAP; urb->instance.complete = handler; urb->instance.context = urb; urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB; diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index fde9a7a29cb6..c1916184e2e1 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/spinlock.h> #include <linux/slab.h> #include <linux/init.h> @@ -39,8 +40,8 @@ #define ENDPOINT_CAPTURE 2 #define ENDPOINT_PLAYBACK 6 -#define MAKE_CHECKBYTE(dev,stream,i) \ - (stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1) +#define MAKE_CHECKBYTE(cdev,stream,i) \ + (stream << 1) | (~(i / (cdev->n_streams * BYTES_PER_SAMPLE_USB)) & 1) static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | @@ -60,32 +61,32 @@ static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { }; static void -activate_substream(struct snd_usb_caiaqdev *dev, +activate_substream(struct snd_usb_caiaqdev *cdev, struct snd_pcm_substream *sub) { - spin_lock(&dev->spinlock); + spin_lock(&cdev->spinlock); if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - dev->sub_playback[sub->number] = sub; + cdev->sub_playback[sub->number] = sub; else - dev->sub_capture[sub->number] = sub; + cdev->sub_capture[sub->number] = sub; - spin_unlock(&dev->spinlock); + spin_unlock(&cdev->spinlock); } static void -deactivate_substream(struct snd_usb_caiaqdev *dev, +deactivate_substream(struct snd_usb_caiaqdev *cdev, struct snd_pcm_substream *sub) { unsigned long flags; - spin_lock_irqsave(&dev->spinlock, flags); + spin_lock_irqsave(&cdev->spinlock, flags); if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - dev->sub_playback[sub->number] = NULL; + cdev->sub_playback[sub->number] = NULL; else - dev->sub_capture[sub->number] = NULL; + cdev->sub_capture[sub->number] = NULL; - spin_unlock_irqrestore(&dev->spinlock, flags); + spin_unlock_irqrestore(&cdev->spinlock, flags); } static int @@ -98,28 +99,30 @@ all_substreams_zero(struct snd_pcm_substream **subs) return 1; } -static int stream_start(struct snd_usb_caiaqdev *dev) +static int stream_start(struct snd_usb_caiaqdev *cdev) { int i, ret; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, dev); + dev_dbg(dev, "%s(%p)\n", __func__, cdev); - if (dev->streaming) + if (cdev->streaming) return -EINVAL; - memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); - memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); - dev->input_panic = 0; - dev->output_panic = 0; - dev->first_packet = 4; - dev->streaming = 1; - dev->warned = 0; + memset(cdev->sub_playback, 0, sizeof(cdev->sub_playback)); + memset(cdev->sub_capture, 0, sizeof(cdev->sub_capture)); + cdev->input_panic = 0; + cdev->output_panic = 0; + cdev->first_packet = 4; + cdev->streaming = 1; + cdev->warned = 0; for (i = 0; i < N_URBS; i++) { - ret = usb_submit_urb(dev->data_urbs_in[i], GFP_ATOMIC); + ret = usb_submit_urb(cdev->data_urbs_in[i], GFP_ATOMIC); if (ret) { - log("unable to trigger read #%d! (ret %d)\n", i, ret); - dev->streaming = 0; + dev_err(dev, "unable to trigger read #%d! (ret %d)\n", + i, ret); + cdev->streaming = 0; return -EPIPE; } } @@ -127,46 +130,51 @@ static int stream_start(struct snd_usb_caiaqdev *dev) return 0; } -static void stream_stop(struct snd_usb_caiaqdev *dev) +static void stream_stop(struct snd_usb_caiaqdev *cdev) { int i; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, dev); - if (!dev->streaming) + dev_dbg(dev, "%s(%p)\n", __func__, cdev); + if (!cdev->streaming) return; - dev->streaming = 0; + cdev->streaming = 0; for (i = 0; i < N_URBS; i++) { - usb_kill_urb(dev->data_urbs_in[i]); + usb_kill_urb(cdev->data_urbs_in[i]); - if (test_bit(i, &dev->outurb_active_mask)) - usb_kill_urb(dev->data_urbs_out[i]); + if (test_bit(i, &cdev->outurb_active_mask)) + usb_kill_urb(cdev->data_urbs_out[i]); } - dev->outurb_active_mask = 0; + cdev->outurb_active_mask = 0; } static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream) { - struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(substream); - debug("%s(%p)\n", __func__, substream); - substream->runtime->hw = dev->pcm_info; + struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); + struct device *dev = caiaqdev_to_dev(cdev); + + dev_dbg(dev, "%s(%p)\n", __func__, substream); + substream->runtime->hw = cdev->pcm_info; snd_pcm_limit_hw_rates(substream->runtime); + return 0; } static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) { - struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(substream); + struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, substream); - if (all_substreams_zero(dev->sub_playback) && - all_substreams_zero(dev->sub_capture)) { + dev_dbg(dev, "%s(%p)\n", __func__, substream); + if (all_substreams_zero(cdev->sub_playback) && + all_substreams_zero(cdev->sub_capture)) { /* when the last client has stopped streaming, * all sample rates are allowed again */ - stream_stop(dev); - dev->pcm_info.rates = dev->samplerates; + stream_stop(cdev); + cdev->pcm_info.rates = cdev->samplerates; } return 0; @@ -175,15 +183,13 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, struct snd_pcm_hw_params *hw_params) { - debug("%s(%p)\n", __func__, sub); return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); } static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) { - struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); - debug("%s(%p)\n", __func__, sub); - deactivate_substream(dev, sub); + struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); + deactivate_substream(cdev, sub); return snd_pcm_lib_free_pages(sub); } @@ -199,15 +205,16 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) { int bytes_per_sample, bpp, ret, i; int index = substream->number; - struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(substream); + struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, substream); + dev_dbg(dev, "%s(%p)\n", __func__, substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { int out_pos; - switch (dev->spec.data_alignment) { + switch (cdev->spec.data_alignment) { case 0: case 2: out_pos = BYTES_PER_SAMPLE + 1; @@ -218,12 +225,12 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) break; } - dev->period_out_count[index] = out_pos; - dev->audio_out_buf_pos[index] = out_pos; + cdev->period_out_count[index] = out_pos; + cdev->audio_out_buf_pos[index] = out_pos; } else { int in_pos; - switch (dev->spec.data_alignment) { + switch (cdev->spec.data_alignment) { case 0: in_pos = BYTES_PER_SAMPLE + 2; break; @@ -236,44 +243,44 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) break; } - dev->period_in_count[index] = in_pos; - dev->audio_in_buf_pos[index] = in_pos; + cdev->period_in_count[index] = in_pos; + cdev->audio_in_buf_pos[index] = in_pos; } - if (dev->streaming) + if (cdev->streaming) return 0; /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ for (i=0; i < ARRAY_SIZE(rates); i++) if (runtime->rate == rates[i]) - dev->pcm_info.rates = 1 << i; + cdev->pcm_info.rates = 1 << i; snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; - if (dev->spec.data_alignment >= 2) + if (cdev->spec.data_alignment >= 2) bytes_per_sample++; bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) - * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams; + * bytes_per_sample * CHANNELS_PER_STREAM * cdev->n_streams; if (bpp > MAX_ENDPOINT_SIZE) bpp = MAX_ENDPOINT_SIZE; - ret = snd_usb_caiaq_set_audio_params(dev, runtime->rate, + ret = snd_usb_caiaq_set_audio_params(cdev, runtime->rate, runtime->sample_bits, bpp); if (ret) return ret; - ret = stream_start(dev); + ret = stream_start(cdev); if (ret) return ret; - dev->output_running = 0; - wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ); - if (!dev->output_running) { - stream_stop(dev); + cdev->output_running = 0; + wait_event_timeout(cdev->prepare_wait_queue, cdev->output_running, HZ); + if (!cdev->output_running) { + stream_stop(cdev); return -EPIPE; } @@ -282,18 +289,19 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_trigger(struct snd_pcm_substream *sub, int cmd) { - struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); + struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p) cmd %d\n", __func__, sub, cmd); + dev_dbg(dev, "%s(%p) cmd %d\n", __func__, sub, cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - activate_substream(dev, sub); + activate_substream(cdev, sub); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - deactivate_substream(dev, sub); + deactivate_substream(cdev, sub); break; default: return -EINVAL; @@ -306,25 +314,25 @@ static snd_pcm_uframes_t snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) { int index = sub->number; - struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); + struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); snd_pcm_uframes_t ptr; - spin_lock(&dev->spinlock); + spin_lock(&cdev->spinlock); - if (dev->input_panic || dev->output_panic) { + if (cdev->input_panic || cdev->output_panic) { ptr = SNDRV_PCM_POS_XRUN; goto unlock; } if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ptr = bytes_to_frames(sub->runtime, - dev->audio_out_buf_pos[index]); + cdev->audio_out_buf_pos[index]); else ptr = bytes_to_frames(sub->runtime, - dev->audio_in_buf_pos[index]); + cdev->audio_in_buf_pos[index]); unlock: - spin_unlock(&dev->spinlock); + spin_unlock(&cdev->spinlock); return ptr; } @@ -340,21 +348,21 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .pointer = snd_usb_caiaq_pcm_pointer }; -static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, +static void check_for_elapsed_periods(struct snd_usb_caiaqdev *cdev, struct snd_pcm_substream **subs) { int stream, pb, *cnt; struct snd_pcm_substream *sub; - for (stream = 0; stream < dev->n_streams; stream++) { + for (stream = 0; stream < cdev->n_streams; stream++) { sub = subs[stream]; if (!sub) continue; pb = snd_pcm_lib_period_bytes(sub); cnt = (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - &dev->period_out_count[stream] : - &dev->period_in_count[stream]; + &cdev->period_out_count[stream] : + &cdev->period_in_count[stream]; if (*cnt >= pb) { snd_pcm_period_elapsed(sub); @@ -363,7 +371,7 @@ static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, } } -static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, +static void read_in_urb_mode0(struct snd_usb_caiaqdev *cdev, const struct urb *urb, const struct usb_iso_packet_descriptor *iso) { @@ -371,27 +379,27 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub; int stream, i; - if (all_substreams_zero(dev->sub_capture)) + if (all_substreams_zero(cdev->sub_capture)) return; for (i = 0; i < iso->actual_length;) { - for (stream = 0; stream < dev->n_streams; stream++, i++) { - sub = dev->sub_capture[stream]; + for (stream = 0; stream < cdev->n_streams; stream++, i++) { + sub = cdev->sub_capture[stream]; if (sub) { struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] + audio_buf[cdev->audio_in_buf_pos[stream]++] = usb_buf[i]; - dev->period_in_count[stream]++; - if (dev->audio_in_buf_pos[stream] == sz) - dev->audio_in_buf_pos[stream] = 0; + cdev->period_in_count[stream]++; + if (cdev->audio_in_buf_pos[stream] == sz) + cdev->audio_in_buf_pos[stream] = 0; } } } } -static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, +static void read_in_urb_mode2(struct snd_usb_caiaqdev *cdev, const struct urb *urb, const struct usb_iso_packet_descriptor *iso) { @@ -401,48 +409,49 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, int stream, i; for (i = 0; i < iso->actual_length;) { - if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { + if (i % (cdev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { for (stream = 0; - stream < dev->n_streams; + stream < cdev->n_streams; stream++, i++) { - if (dev->first_packet) + if (cdev->first_packet) continue; - check_byte = MAKE_CHECKBYTE(dev, stream, i); + check_byte = MAKE_CHECKBYTE(cdev, stream, i); if ((usb_buf[i] & 0x3f) != check_byte) - dev->input_panic = 1; + cdev->input_panic = 1; if (usb_buf[i] & 0x80) - dev->output_panic = 1; + cdev->output_panic = 1; } } - dev->first_packet = 0; + cdev->first_packet = 0; - for (stream = 0; stream < dev->n_streams; stream++, i++) { - sub = dev->sub_capture[stream]; - if (dev->input_panic) + for (stream = 0; stream < cdev->n_streams; stream++, i++) { + sub = cdev->sub_capture[stream]; + if (cdev->input_panic) usb_buf[i] = 0; if (sub) { struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] = + audio_buf[cdev->audio_in_buf_pos[stream]++] = usb_buf[i]; - dev->period_in_count[stream]++; - if (dev->audio_in_buf_pos[stream] == sz) - dev->audio_in_buf_pos[stream] = 0; + cdev->period_in_count[stream]++; + if (cdev->audio_in_buf_pos[stream] == sz) + cdev->audio_in_buf_pos[stream] = 0; } } } } -static void read_in_urb_mode3(struct snd_usb_caiaqdev *dev, +static void read_in_urb_mode3(struct snd_usb_caiaqdev *cdev, const struct urb *urb, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; + struct device *dev = caiaqdev_to_dev(cdev); int stream, i; /* paranoia check */ @@ -450,12 +459,12 @@ static void read_in_urb_mode3(struct snd_usb_caiaqdev *dev, return; for (i = 0; i < iso->actual_length;) { - for (stream = 0; stream < dev->n_streams; stream++) { - struct snd_pcm_substream *sub = dev->sub_capture[stream]; + for (stream = 0; stream < cdev->n_streams; stream++) { + struct snd_pcm_substream *sub = cdev->sub_capture[stream]; char *audio_buf = NULL; int c, n, sz = 0; - if (sub && !dev->input_panic) { + if (sub && !cdev->input_panic) { struct snd_pcm_runtime *rt = sub->runtime; audio_buf = rt->dma_area; sz = frames_to_bytes(rt, rt->buffer_size); @@ -465,23 +474,23 @@ static void read_in_urb_mode3(struct snd_usb_caiaqdev *dev, /* 3 audio data bytes, followed by 1 check byte */ if (audio_buf) { for (n = 0; n < BYTES_PER_SAMPLE; n++) { - audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i+n]; + audio_buf[cdev->audio_in_buf_pos[stream]++] = usb_buf[i+n]; - if (dev->audio_in_buf_pos[stream] == sz) - dev->audio_in_buf_pos[stream] = 0; + if (cdev->audio_in_buf_pos[stream] == sz) + cdev->audio_in_buf_pos[stream] = 0; } - dev->period_in_count[stream] += BYTES_PER_SAMPLE; + cdev->period_in_count[stream] += BYTES_PER_SAMPLE; } i += BYTES_PER_SAMPLE; if (usb_buf[i] != ((stream << 1) | c) && - !dev->first_packet) { - if (!dev->input_panic) - printk(" EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", - ((stream << 1) | c), usb_buf[i], c, stream, i); - dev->input_panic = 1; + !cdev->first_packet) { + if (!cdev->input_panic) + dev_warn(dev, " EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", + ((stream << 1) | c), usb_buf[i], c, stream, i); + cdev->input_panic = 1; } i++; @@ -489,41 +498,43 @@ static void read_in_urb_mode3(struct snd_usb_caiaqdev *dev, } } - if (dev->first_packet > 0) - dev->first_packet--; + if (cdev->first_packet > 0) + cdev->first_packet--; } -static void read_in_urb(struct snd_usb_caiaqdev *dev, +static void read_in_urb(struct snd_usb_caiaqdev *cdev, const struct urb *urb, const struct usb_iso_packet_descriptor *iso) { - if (!dev->streaming) + struct device *dev = caiaqdev_to_dev(cdev); + + if (!cdev->streaming) return; - if (iso->actual_length < dev->bpp) + if (iso->actual_length < cdev->bpp) return; - switch (dev->spec.data_alignment) { + switch (cdev->spec.data_alignment) { case 0: - read_in_urb_mode0(dev, urb, iso); + read_in_urb_mode0(cdev, urb, iso); break; case 2: - read_in_urb_mode2(dev, urb, iso); + read_in_urb_mode2(cdev, urb, iso); break; case 3: - read_in_urb_mode3(dev, urb, iso); + read_in_urb_mode3(cdev, urb, iso); break; } - if ((dev->input_panic || dev->output_panic) && !dev->warned) { - debug("streaming error detected %s %s\n", - dev->input_panic ? "(input)" : "", - dev->output_panic ? "(output)" : ""); - dev->warned = 1; + if ((cdev->input_panic || cdev->output_panic) && !cdev->warned) { + dev_warn(dev, "streaming error detected %s %s\n", + cdev->input_panic ? "(input)" : "", + cdev->output_panic ? "(output)" : ""); + cdev->warned = 1; } } -static void fill_out_urb_mode_0(struct snd_usb_caiaqdev *dev, +static void fill_out_urb_mode_0(struct snd_usb_caiaqdev *cdev, struct urb *urb, const struct usb_iso_packet_descriptor *iso) { @@ -532,32 +543,32 @@ static void fill_out_urb_mode_0(struct snd_usb_caiaqdev *dev, int stream, i; for (i = 0; i < iso->length;) { - for (stream = 0; stream < dev->n_streams; stream++, i++) { - sub = dev->sub_playback[stream]; + for (stream = 0; stream < cdev->n_streams; stream++, i++) { + sub = cdev->sub_playback[stream]; if (sub) { struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); usb_buf[i] = - audio_buf[dev->audio_out_buf_pos[stream]]; - dev->period_out_count[stream]++; - dev->audio_out_buf_pos[stream]++; - if (dev->audio_out_buf_pos[stream] == sz) - dev->audio_out_buf_pos[stream] = 0; + audio_buf[cdev->audio_out_buf_pos[stream]]; + cdev->period_out_count[stream]++; + cdev->audio_out_buf_pos[stream]++; + if (cdev->audio_out_buf_pos[stream] == sz) + cdev->audio_out_buf_pos[stream] = 0; } else usb_buf[i] = 0; } /* fill in the check bytes */ - if (dev->spec.data_alignment == 2 && - i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == - (dev->n_streams * CHANNELS_PER_STREAM)) - for (stream = 0; stream < dev->n_streams; stream++, i++) - usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); + if (cdev->spec.data_alignment == 2 && + i % (cdev->n_streams * BYTES_PER_SAMPLE_USB) == + (cdev->n_streams * CHANNELS_PER_STREAM)) + for (stream = 0; stream < cdev->n_streams; stream++, i++) + usb_buf[i] = MAKE_CHECKBYTE(cdev, stream, i); } } -static void fill_out_urb_mode_3(struct snd_usb_caiaqdev *dev, +static void fill_out_urb_mode_3(struct snd_usb_caiaqdev *cdev, struct urb *urb, const struct usb_iso_packet_descriptor *iso) { @@ -565,8 +576,8 @@ static void fill_out_urb_mode_3(struct snd_usb_caiaqdev *dev, int stream, i; for (i = 0; i < iso->length;) { - for (stream = 0; stream < dev->n_streams; stream++) { - struct snd_pcm_substream *sub = dev->sub_playback[stream]; + for (stream = 0; stream < cdev->n_streams; stream++) { + struct snd_pcm_substream *sub = cdev->sub_playback[stream]; char *audio_buf = NULL; int c, n, sz = 0; @@ -579,17 +590,17 @@ static void fill_out_urb_mode_3(struct snd_usb_caiaqdev *dev, for (c = 0; c < CHANNELS_PER_STREAM; c++) { for (n = 0; n < BYTES_PER_SAMPLE; n++) { if (audio_buf) { - usb_buf[i+n] = audio_buf[dev->audio_out_buf_pos[stream]++]; + usb_buf[i+n] = audio_buf[cdev->audio_out_buf_pos[stream]++]; - if (dev->audio_out_buf_pos[stream] == sz) - dev->audio_out_buf_pos[stream] = 0; + if (cdev->audio_out_buf_pos[stream] == sz) + cdev->audio_out_buf_pos[stream] = 0; } else { usb_buf[i+n] = 0; } } if (audio_buf) - dev->period_out_count[stream] += BYTES_PER_SAMPLE; + cdev->period_out_count[stream] += BYTES_PER_SAMPLE; i += BYTES_PER_SAMPLE; @@ -600,17 +611,17 @@ static void fill_out_urb_mode_3(struct snd_usb_caiaqdev *dev, } } -static inline void fill_out_urb(struct snd_usb_caiaqdev *dev, +static inline void fill_out_urb(struct snd_usb_caiaqdev *cdev, struct urb *urb, const struct usb_iso_packet_descriptor *iso) { - switch (dev->spec.data_alignment) { + switch (cdev->spec.data_alignment) { case 0: case 2: - fill_out_urb_mode_0(dev, urb, iso); + fill_out_urb_mode_0(cdev, urb, iso); break; case 3: - fill_out_urb_mode_3(dev, urb, iso); + fill_out_urb_mode_3(cdev, urb, iso); break; } } @@ -618,7 +629,8 @@ static inline void fill_out_urb(struct snd_usb_caiaqdev *dev, static void read_completed(struct urb *urb) { struct snd_usb_caiaq_cb_info *info = urb->context; - struct snd_usb_caiaqdev *dev; + struct snd_usb_caiaqdev *cdev; + struct device *dev; struct urb *out = NULL; int i, frame, len, send_it = 0, outframe = 0; size_t offset = 0; @@ -626,20 +638,21 @@ static void read_completed(struct urb *urb) if (urb->status || !info) return; - dev = info->dev; + cdev = info->cdev; + dev = caiaqdev_to_dev(cdev); - if (!dev->streaming) + if (!cdev->streaming) return; /* find an unused output urb that is unused */ for (i = 0; i < N_URBS; i++) - if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) { - out = dev->data_urbs_out[i]; + if (test_and_set_bit(i, &cdev->outurb_active_mask) == 0) { + out = cdev->data_urbs_out[i]; break; } if (!out) { - log("Unable to find an output urb to use\n"); + dev_err(dev, "Unable to find an output urb to use\n"); goto requeue; } @@ -656,12 +669,12 @@ static void read_completed(struct urb *urb) offset += len; if (len > 0) { - spin_lock(&dev->spinlock); - fill_out_urb(dev, out, &out->iso_frame_desc[outframe]); - read_in_urb(dev, urb, &urb->iso_frame_desc[frame]); - spin_unlock(&dev->spinlock); - check_for_elapsed_periods(dev, dev->sub_playback); - check_for_elapsed_periods(dev, dev->sub_capture); + spin_lock(&cdev->spinlock); + fill_out_urb(cdev, out, &out->iso_frame_desc[outframe]); + read_in_urb(cdev, urb, &urb->iso_frame_desc[frame]); + spin_unlock(&cdev->spinlock); + check_for_elapsed_periods(cdev, cdev->sub_playback); + check_for_elapsed_periods(cdev, cdev->sub_capture); send_it = 1; } @@ -670,11 +683,10 @@ static void read_completed(struct urb *urb) if (send_it) { out->number_of_packets = outframe; - out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } else { struct snd_usb_caiaq_cb_info *oinfo = out->context; - clear_bit(oinfo->index, &dev->outurb_active_mask); + clear_bit(oinfo->index, &cdev->outurb_active_mask); } requeue: @@ -686,28 +698,28 @@ requeue: } urb->number_of_packets = FRAMES_PER_URB; - urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); } static void write_completed(struct urb *urb) { struct snd_usb_caiaq_cb_info *info = urb->context; - struct snd_usb_caiaqdev *dev = info->dev; + struct snd_usb_caiaqdev *cdev = info->cdev; - if (!dev->output_running) { - dev->output_running = 1; - wake_up(&dev->prepare_wait_queue); + if (!cdev->output_running) { + cdev->output_running = 1; + wake_up(&cdev->prepare_wait_queue); } - clear_bit(info->index, &dev->outurb_active_mask); + clear_bit(info->index, &cdev->outurb_active_mask); } -static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) +static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) { int i, frame; struct urb **urbs; - struct usb_device *usb_dev = dev->chip.dev; + struct usb_device *usb_dev = cdev->chip.dev; + struct device *dev = caiaqdev_to_dev(cdev); unsigned int pipe; pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? @@ -716,7 +728,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) urbs = kmalloc(N_URBS * sizeof(*urbs), GFP_KERNEL); if (!urbs) { - log("unable to kmalloc() urbs, OOM!?\n"); + dev_err(dev, "unable to kmalloc() urbs, OOM!?\n"); *ret = -ENOMEM; return NULL; } @@ -724,7 +736,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) for (i = 0; i < N_URBS; i++) { urbs[i] = usb_alloc_urb(FRAMES_PER_URB, GFP_KERNEL); if (!urbs[i]) { - log("unable to usb_alloc_urb(), OOM!?\n"); + dev_err(dev, "unable to usb_alloc_urb(), OOM!?\n"); *ret = -ENOMEM; return urbs; } @@ -732,7 +744,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { - log("unable to kmalloc() transfer buffer, OOM!?\n"); + dev_err(dev, "unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } @@ -749,9 +761,8 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) urbs[i]->pipe = pipe; urbs[i]->transfer_buffer_length = FRAMES_PER_URB * BYTES_PER_FRAME; - urbs[i]->context = &dev->data_cb_info[i]; + urbs[i]->context = &cdev->data_cb_info[i]; urbs[i]->interval = 1; - urbs[i]->transfer_flags = URB_ISO_ASAP; urbs[i]->number_of_packets = FRAMES_PER_URB; urbs[i]->complete = (dir == SNDRV_PCM_STREAM_CAPTURE) ? read_completed : write_completed; @@ -780,110 +791,113 @@ static void free_urbs(struct urb **urbs) kfree(urbs); } -int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) +int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) { int i, ret; + struct device *dev = caiaqdev_to_dev(cdev); - dev->n_audio_in = max(dev->spec.num_analog_audio_in, - dev->spec.num_digital_audio_in) / + cdev->n_audio_in = max(cdev->spec.num_analog_audio_in, + cdev->spec.num_digital_audio_in) / CHANNELS_PER_STREAM; - dev->n_audio_out = max(dev->spec.num_analog_audio_out, - dev->spec.num_digital_audio_out) / + cdev->n_audio_out = max(cdev->spec.num_analog_audio_out, + cdev->spec.num_digital_audio_out) / CHANNELS_PER_STREAM; - dev->n_streams = max(dev->n_audio_in, dev->n_audio_out); + cdev->n_streams = max(cdev->n_audio_in, cdev->n_audio_out); - debug("dev->n_audio_in = %d\n", dev->n_audio_in); - debug("dev->n_audio_out = %d\n", dev->n_audio_out); - debug("dev->n_streams = %d\n", dev->n_streams); + dev_dbg(dev, "cdev->n_audio_in = %d\n", cdev->n_audio_in); + dev_dbg(dev, "cdev->n_audio_out = %d\n", cdev->n_audio_out); + dev_dbg(dev, "cdev->n_streams = %d\n", cdev->n_streams); - if (dev->n_streams > MAX_STREAMS) { - log("unable to initialize device, too many streams.\n"); + if (cdev->n_streams > MAX_STREAMS) { + dev_err(dev, "unable to initialize device, too many streams.\n"); return -EINVAL; } - ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, - dev->n_audio_out, dev->n_audio_in, &dev->pcm); + ret = snd_pcm_new(cdev->chip.card, cdev->product_name, 0, + cdev->n_audio_out, cdev->n_audio_in, &cdev->pcm); if (ret < 0) { - log("snd_pcm_new() returned %d\n", ret); + dev_err(dev, "snd_pcm_new() returned %d\n", ret); return ret; } - dev->pcm->private_data = dev; - strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name)); + cdev->pcm->private_data = cdev; + strlcpy(cdev->pcm->name, cdev->product_name, sizeof(cdev->pcm->name)); - memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); - memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); + memset(cdev->sub_playback, 0, sizeof(cdev->sub_playback)); + memset(cdev->sub_capture, 0, sizeof(cdev->sub_capture)); - memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware, + memcpy(&cdev->pcm_info, &snd_usb_caiaq_pcm_hardware, sizeof(snd_usb_caiaq_pcm_hardware)); /* setup samplerates */ - dev->samplerates = dev->pcm_info.rates; - switch (dev->chip.usb_id) { + cdev->samplerates = cdev->pcm_info.rates; + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE): - dev->samplerates |= SNDRV_PCM_RATE_192000; + cdev->samplerates |= SNDRV_PCM_RATE_192000; /* fall thru */ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORAUDIO2): - dev->samplerates |= SNDRV_PCM_RATE_88200; + cdev->samplerates |= SNDRV_PCM_RATE_88200; break; } - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(cdev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usb_caiaq_ops); - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(cdev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); - snd_pcm_lib_preallocate_pages_for_all(dev->pcm, + snd_pcm_lib_preallocate_pages_for_all(cdev->pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); - dev->data_cb_info = + cdev->data_cb_info = kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); - if (!dev->data_cb_info) + if (!cdev->data_cb_info) return -ENOMEM; - dev->outurb_active_mask = 0; - BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8)); + cdev->outurb_active_mask = 0; + BUILD_BUG_ON(N_URBS > (sizeof(cdev->outurb_active_mask) * 8)); for (i = 0; i < N_URBS; i++) { - dev->data_cb_info[i].dev = dev; - dev->data_cb_info[i].index = i; + cdev->data_cb_info[i].cdev = cdev; + cdev->data_cb_info[i].index = i; } - dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret); + cdev->data_urbs_in = alloc_urbs(cdev, SNDRV_PCM_STREAM_CAPTURE, &ret); if (ret < 0) { - kfree(dev->data_cb_info); - free_urbs(dev->data_urbs_in); + kfree(cdev->data_cb_info); + free_urbs(cdev->data_urbs_in); return ret; } - dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret); + cdev->data_urbs_out = alloc_urbs(cdev, SNDRV_PCM_STREAM_PLAYBACK, &ret); if (ret < 0) { - kfree(dev->data_cb_info); - free_urbs(dev->data_urbs_in); - free_urbs(dev->data_urbs_out); + kfree(cdev->data_cb_info); + free_urbs(cdev->data_urbs_in); + free_urbs(cdev->data_urbs_out); return ret; } return 0; } -void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *dev) +void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *cdev) { - debug("%s(%p)\n", __func__, dev); - stream_stop(dev); - free_urbs(dev->data_urbs_in); - free_urbs(dev->data_urbs_out); - kfree(dev->data_cb_info); + struct device *dev = caiaqdev_to_dev(cdev); + + dev_dbg(dev, "%s(%p)\n", __func__, cdev); + stream_stop(cdev); + free_urbs(cdev->data_urbs_in); + free_urbs(cdev->data_urbs_out); + kfree(cdev->data_cb_info); } diff --git a/sound/usb/caiaq/audio.h b/sound/usb/caiaq/audio.h index 8ab1f8d9529e..bdf155300a8a 100644 --- a/sound/usb/caiaq/audio.h +++ b/sound/usb/caiaq/audio.h @@ -1,7 +1,7 @@ #ifndef CAIAQ_AUDIO_H #define CAIAQ_AUDIO_H -int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev); -void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *dev); +int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev); +void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *cdev); #endif /* CAIAQ_AUDIO_H */ diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index adb8d03267a0..ae6b50f9ed56 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -17,6 +17,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/init.h> #include <linux/usb.h> #include <sound/control.h> @@ -32,7 +33,7 @@ static int control_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); - struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); + struct snd_usb_caiaqdev *cdev = caiaqdev(chip->card); int pos = kcontrol->private_value; int is_intval = pos & CNT_INTVAL; int maxval = 63; @@ -40,7 +41,7 @@ static int control_info(struct snd_kcontrol *kcontrol, uinfo->count = 1; pos &= ~CNT_INTVAL; - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): if (pos == 0) { @@ -78,15 +79,15 @@ static int control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); - struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); + struct snd_usb_caiaqdev *cdev = caiaqdev(chip->card); int pos = kcontrol->private_value; if (pos & CNT_INTVAL) ucontrol->value.integer.value[0] - = dev->control_state[pos & ~CNT_INTVAL]; + = cdev->control_state[pos & ~CNT_INTVAL]; else ucontrol->value.integer.value[0] - = !!(dev->control_state[pos / 8] & (1 << pos % 8)); + = !!(cdev->control_state[pos / 8] & (1 << pos % 8)); return 0; } @@ -95,43 +96,43 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); - struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); + struct snd_usb_caiaqdev *cdev = caiaqdev(chip->card); int pos = kcontrol->private_value; int v = ucontrol->value.integer.value[0]; unsigned char cmd = EP1_CMD_WRITE_IO; - if (dev->chip.usb_id == + if (cdev->chip.usb_id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1)) cmd = EP1_CMD_DIMM_LEDS; if (pos & CNT_INTVAL) { int i = pos & ~CNT_INTVAL; - dev->control_state[i] = v; + cdev->control_state[i] = v; - if (dev->chip.usb_id == + if (cdev->chip.usb_id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4)) { int actual_len; - dev->ep8_out_buf[0] = i; - dev->ep8_out_buf[1] = v; + cdev->ep8_out_buf[0] = i; + cdev->ep8_out_buf[1] = v; - usb_bulk_msg(dev->chip.dev, - usb_sndbulkpipe(dev->chip.dev, 8), - dev->ep8_out_buf, sizeof(dev->ep8_out_buf), + usb_bulk_msg(cdev->chip.dev, + usb_sndbulkpipe(cdev->chip.dev, 8), + cdev->ep8_out_buf, sizeof(cdev->ep8_out_buf), &actual_len, 200); } else { - snd_usb_caiaq_send_command(dev, cmd, - dev->control_state, sizeof(dev->control_state)); + snd_usb_caiaq_send_command(cdev, cmd, + cdev->control_state, sizeof(cdev->control_state)); } } else { if (v) - dev->control_state[pos / 8] |= 1 << (pos % 8); + cdev->control_state[pos / 8] |= 1 << (pos % 8); else - dev->control_state[pos / 8] &= ~(1 << (pos % 8)); + cdev->control_state[pos / 8] &= ~(1 << (pos % 8)); - snd_usb_caiaq_send_command(dev, cmd, - dev->control_state, sizeof(dev->control_state)); + snd_usb_caiaq_send_command(cdev, cmd, + cdev->control_state, sizeof(cdev->control_state)); } return 1; @@ -490,7 +491,7 @@ static struct caiaq_controller kontrols4_controller[] = { }; static int add_controls(struct caiaq_controller *c, int num, - struct snd_usb_caiaqdev *dev) + struct snd_usb_caiaqdev *cdev) { int i, ret; struct snd_kcontrol *kc; @@ -498,8 +499,8 @@ static int add_controls(struct caiaq_controller *c, int num, for (i = 0; i < num; i++, c++) { kcontrol_template.name = c->name; kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - ret = snd_ctl_add(dev->chip.card, kc); + kc = snd_ctl_new1(&kcontrol_template, cdev); + ret = snd_ctl_add(cdev->chip.card, kc); if (ret < 0) return ret; } @@ -507,50 +508,50 @@ static int add_controls(struct caiaq_controller *c, int num, return 0; } -int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) +int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *cdev) { int ret = 0; - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): ret = add_controls(ak1_controller, - ARRAY_SIZE(ak1_controller), dev); + ARRAY_SIZE(ak1_controller), cdev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): ret = add_controls(rk2_controller, - ARRAY_SIZE(rk2_controller), dev); + ARRAY_SIZE(rk2_controller), cdev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): ret = add_controls(rk3_controller, - ARRAY_SIZE(rk3_controller), dev); + ARRAY_SIZE(rk3_controller), cdev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): ret = add_controls(kore_controller, - ARRAY_SIZE(kore_controller), dev); + ARRAY_SIZE(kore_controller), cdev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): ret = add_controls(a8dj_controller, - ARRAY_SIZE(a8dj_controller), dev); + ARRAY_SIZE(a8dj_controller), cdev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): ret = add_controls(a4dj_controller, - ARRAY_SIZE(a4dj_controller), dev); + ARRAY_SIZE(a4dj_controller), cdev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): ret = add_controls(kontrolx1_controller, - ARRAY_SIZE(kontrolx1_controller), dev); + ARRAY_SIZE(kontrolx1_controller), cdev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): ret = add_controls(kontrols4_controller, - ARRAY_SIZE(kontrols4_controller), dev); + ARRAY_SIZE(kontrols4_controller), cdev); break; } diff --git a/sound/usb/caiaq/control.h b/sound/usb/caiaq/control.h index 2e7ab1aa4fb3..501c4883aef6 100644 --- a/sound/usb/caiaq/control.h +++ b/sound/usb/caiaq/control.h @@ -1,6 +1,6 @@ #ifndef CAIAQ_CONTROL_H #define CAIAQ_CONTROL_H -int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev); +int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *cdev); #endif /* CAIAQ_CONTROL_H */ diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index e4d6dbb0342d..48b63ccc78c7 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -20,6 +20,7 @@ */ #include <linux/moduleparam.h> +#include <linux/device.h> #include <linux/interrupt.h> #include <linux/module.h> #include <linux/init.h> @@ -158,67 +159,68 @@ static struct usb_device_id snd_usb_id_table[] = { static void usb_ep1_command_reply_dispatch (struct urb* urb) { int ret; - struct snd_usb_caiaqdev *dev = urb->context; + struct device *dev = &urb->dev->dev; + struct snd_usb_caiaqdev *cdev = urb->context; unsigned char *buf = urb->transfer_buffer; - if (urb->status || !dev) { - log("received EP1 urb->status = %i\n", urb->status); + if (urb->status || !cdev) { + dev_warn(dev, "received EP1 urb->status = %i\n", urb->status); return; } switch(buf[0]) { case EP1_CMD_GET_DEVICE_INFO: - memcpy(&dev->spec, buf+1, sizeof(struct caiaq_device_spec)); - dev->spec.fw_version = le16_to_cpu(dev->spec.fw_version); - debug("device spec (firmware %d): audio: %d in, %d out, " + memcpy(&cdev->spec, buf+1, sizeof(struct caiaq_device_spec)); + cdev->spec.fw_version = le16_to_cpu(cdev->spec.fw_version); + dev_dbg(dev, "device spec (firmware %d): audio: %d in, %d out, " "MIDI: %d in, %d out, data alignment %d\n", - dev->spec.fw_version, - dev->spec.num_analog_audio_in, - dev->spec.num_analog_audio_out, - dev->spec.num_midi_in, - dev->spec.num_midi_out, - dev->spec.data_alignment); - - dev->spec_received++; - wake_up(&dev->ep1_wait_queue); + cdev->spec.fw_version, + cdev->spec.num_analog_audio_in, + cdev->spec.num_analog_audio_out, + cdev->spec.num_midi_in, + cdev->spec.num_midi_out, + cdev->spec.data_alignment); + + cdev->spec_received++; + wake_up(&cdev->ep1_wait_queue); break; case EP1_CMD_AUDIO_PARAMS: - dev->audio_parm_answer = buf[1]; - wake_up(&dev->ep1_wait_queue); + cdev->audio_parm_answer = buf[1]; + wake_up(&cdev->ep1_wait_queue); break; case EP1_CMD_MIDI_READ: - snd_usb_caiaq_midi_handle_input(dev, buf[1], buf + 3, buf[2]); + snd_usb_caiaq_midi_handle_input(cdev, buf[1], buf + 3, buf[2]); break; case EP1_CMD_READ_IO: - if (dev->chip.usb_id == + if (cdev->chip.usb_id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)) { - if (urb->actual_length > sizeof(dev->control_state)) - urb->actual_length = sizeof(dev->control_state); - memcpy(dev->control_state, buf + 1, urb->actual_length); - wake_up(&dev->ep1_wait_queue); + if (urb->actual_length > sizeof(cdev->control_state)) + urb->actual_length = sizeof(cdev->control_state); + memcpy(cdev->control_state, buf + 1, urb->actual_length); + wake_up(&cdev->ep1_wait_queue); break; } #ifdef CONFIG_SND_USB_CAIAQ_INPUT case EP1_CMD_READ_ERP: case EP1_CMD_READ_ANALOG: - snd_usb_caiaq_input_dispatch(dev, buf, urb->actual_length); + snd_usb_caiaq_input_dispatch(cdev, buf, urb->actual_length); #endif break; } - dev->ep1_in_urb.actual_length = 0; - ret = usb_submit_urb(&dev->ep1_in_urb, GFP_ATOMIC); + cdev->ep1_in_urb.actual_length = 0; + ret = usb_submit_urb(&cdev->ep1_in_urb, GFP_ATOMIC); if (ret < 0) - log("unable to submit urb. OOM!?\n"); + dev_err(dev, "unable to submit urb. OOM!?\n"); } -int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, +int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, unsigned char command, const unsigned char *buffer, int len) { int actual_len; - struct usb_device *usb_dev = dev->chip.dev; + struct usb_device *usb_dev = cdev->chip.dev; if (!usb_dev) return -EIO; @@ -227,18 +229,19 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, len = EP1_BUFSIZE - 1; if (buffer && len > 0) - memcpy(dev->ep1_out_buf+1, buffer, len); + memcpy(cdev->ep1_out_buf+1, buffer, len); - dev->ep1_out_buf[0] = command; + cdev->ep1_out_buf[0] = command; return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), - dev->ep1_out_buf, len+1, &actual_len, 200); + cdev->ep1_out_buf, len+1, &actual_len, 200); } -int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, +int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, int rate, int depth, int bpp) { int ret; char tmp[5]; + struct device *dev = caiaqdev_to_dev(cdev); switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; @@ -259,49 +262,50 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, tmp[3] = bpp >> 8; tmp[4] = 1; /* packets per microframe */ - debug("setting audio params: %d Hz, %d bits, %d bpp\n", + dev_dbg(dev, "setting audio params: %d Hz, %d bits, %d bpp\n", rate, depth, bpp); - dev->audio_parm_answer = -1; - ret = snd_usb_caiaq_send_command(dev, EP1_CMD_AUDIO_PARAMS, + cdev->audio_parm_answer = -1; + ret = snd_usb_caiaq_send_command(cdev, EP1_CMD_AUDIO_PARAMS, tmp, sizeof(tmp)); if (ret) return ret; - if (!wait_event_timeout(dev->ep1_wait_queue, - dev->audio_parm_answer >= 0, HZ)) + if (!wait_event_timeout(cdev->ep1_wait_queue, + cdev->audio_parm_answer >= 0, HZ)) return -EPIPE; - if (dev->audio_parm_answer != 1) - debug("unable to set the device's audio params\n"); + if (cdev->audio_parm_answer != 1) + dev_dbg(dev, "unable to set the device's audio params\n"); else - dev->bpp = bpp; + cdev->bpp = bpp; - return dev->audio_parm_answer == 1 ? 0 : -EINVAL; + return cdev->audio_parm_answer == 1 ? 0 : -EINVAL; } -int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, +int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *cdev, int digital, int analog, int erp) { char tmp[3] = { digital, analog, erp }; - return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG, + return snd_usb_caiaq_send_command(cdev, EP1_CMD_AUTO_MSG, tmp, sizeof(tmp)); } -static void setup_card(struct snd_usb_caiaqdev *dev) +static void setup_card(struct snd_usb_caiaqdev *cdev) { int ret; char val[4]; + struct device *dev = caiaqdev_to_dev(cdev); /* device-specific startup specials */ - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): /* RigKontrol2 - display centered dash ('-') */ val[0] = 0x00; val[1] = 0x00; val[2] = 0x01; - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, val, 3); + snd_usb_caiaq_send_command(cdev, EP1_CMD_WRITE_IO, val, 3); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): /* RigKontrol2 - display two centered dashes ('--') */ @@ -309,69 +313,69 @@ static void setup_card(struct snd_usb_caiaqdev *dev) val[1] = 0x40; val[2] = 0x40; val[3] = 0x00; - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, val, 4); + snd_usb_caiaq_send_command(cdev, EP1_CMD_WRITE_IO, val, 4); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): /* Audio Kontrol 1 - make USB-LED stop blinking */ val[0] = 0x00; - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, val, 1); + snd_usb_caiaq_send_command(cdev, EP1_CMD_WRITE_IO, val, 1); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): /* Audio 8 DJ - trigger read of current settings */ - dev->control_state[0] = 0xff; - snd_usb_caiaq_set_auto_msg(dev, 1, 0, 0); - snd_usb_caiaq_send_command(dev, EP1_CMD_READ_IO, NULL, 0); + cdev->control_state[0] = 0xff; + snd_usb_caiaq_set_auto_msg(cdev, 1, 0, 0); + snd_usb_caiaq_send_command(cdev, EP1_CMD_READ_IO, NULL, 0); - if (!wait_event_timeout(dev->ep1_wait_queue, - dev->control_state[0] != 0xff, HZ)) + if (!wait_event_timeout(cdev->ep1_wait_queue, + cdev->control_state[0] != 0xff, HZ)) return; /* fix up some defaults */ - if ((dev->control_state[1] != 2) || - (dev->control_state[2] != 3) || - (dev->control_state[4] != 2)) { - dev->control_state[1] = 2; - dev->control_state[2] = 3; - dev->control_state[4] = 2; - snd_usb_caiaq_send_command(dev, - EP1_CMD_WRITE_IO, dev->control_state, 6); + if ((cdev->control_state[1] != 2) || + (cdev->control_state[2] != 3) || + (cdev->control_state[4] != 2)) { + cdev->control_state[1] = 2; + cdev->control_state[2] = 3; + cdev->control_state[4] = 2; + snd_usb_caiaq_send_command(cdev, + EP1_CMD_WRITE_IO, cdev->control_state, 6); } break; } - if (dev->spec.num_analog_audio_out + - dev->spec.num_analog_audio_in + - dev->spec.num_digital_audio_out + - dev->spec.num_digital_audio_in > 0) { - ret = snd_usb_caiaq_audio_init(dev); + if (cdev->spec.num_analog_audio_out + + cdev->spec.num_analog_audio_in + + cdev->spec.num_digital_audio_out + + cdev->spec.num_digital_audio_in > 0) { + ret = snd_usb_caiaq_audio_init(cdev); if (ret < 0) - log("Unable to set up audio system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up audio system (ret=%d)\n", ret); } - if (dev->spec.num_midi_in + - dev->spec.num_midi_out > 0) { - ret = snd_usb_caiaq_midi_init(dev); + if (cdev->spec.num_midi_in + + cdev->spec.num_midi_out > 0) { + ret = snd_usb_caiaq_midi_init(cdev); if (ret < 0) - log("Unable to set up MIDI system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up MIDI system (ret=%d)\n", ret); } #ifdef CONFIG_SND_USB_CAIAQ_INPUT - ret = snd_usb_caiaq_input_init(dev); + ret = snd_usb_caiaq_input_init(cdev); if (ret < 0) - log("Unable to set up input system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up input system (ret=%d)\n", ret); #endif /* finally, register the card and all its sub-instances */ - ret = snd_card_register(dev->chip.card); + ret = snd_card_register(cdev->chip.card); if (ret < 0) { - log("snd_card_register() returned %d\n", ret); - snd_card_free(dev->chip.card); + dev_err(dev, "snd_card_register() returned %d\n", ret); + snd_card_free(cdev->chip.card); } - ret = snd_usb_caiaq_control_init(dev); + ret = snd_usb_caiaq_control_init(cdev); if (ret < 0) - log("Unable to set up control system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up control system (ret=%d)\n", ret); } static int create_card(struct usb_device *usb_dev, @@ -381,7 +385,7 @@ static int create_card(struct usb_device *usb_dev, int devnum; int err; struct snd_card *card; - struct snd_usb_caiaqdev *dev; + struct snd_usb_caiaqdev *cdev; for (devnum = 0; devnum < SNDRV_CARDS; devnum++) if (enable[devnum] && !snd_card_used[devnum]) @@ -395,65 +399,66 @@ static int create_card(struct usb_device *usb_dev, if (err < 0) return err; - dev = caiaqdev(card); - dev->chip.dev = usb_dev; - dev->chip.card = card; - dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor), + cdev = caiaqdev(card); + cdev->chip.dev = usb_dev; + cdev->chip.card = card; + cdev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor), le16_to_cpu(usb_dev->descriptor.idProduct)); - spin_lock_init(&dev->spinlock); + spin_lock_init(&cdev->spinlock); snd_card_set_dev(card, &intf->dev); *cardp = card; return 0; } -static int init_card(struct snd_usb_caiaqdev *dev) +static int init_card(struct snd_usb_caiaqdev *cdev) { char *c, usbpath[32]; - struct usb_device *usb_dev = dev->chip.dev; - struct snd_card *card = dev->chip.card; + struct usb_device *usb_dev = cdev->chip.dev; + struct snd_card *card = cdev->chip.card; + struct device *dev = caiaqdev_to_dev(cdev); int err, len; if (usb_set_interface(usb_dev, 0, 1) != 0) { - log("can't set alt interface.\n"); + dev_err(dev, "can't set alt interface.\n"); return -EIO; } - usb_init_urb(&dev->ep1_in_urb); - usb_init_urb(&dev->midi_out_urb); + usb_init_urb(&cdev->ep1_in_urb); + usb_init_urb(&cdev->midi_out_urb); - usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, + usb_fill_bulk_urb(&cdev->ep1_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x1), - dev->ep1_in_buf, EP1_BUFSIZE, - usb_ep1_command_reply_dispatch, dev); + cdev->ep1_in_buf, EP1_BUFSIZE, + usb_ep1_command_reply_dispatch, cdev); - usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, + usb_fill_bulk_urb(&cdev->midi_out_urb, usb_dev, usb_sndbulkpipe(usb_dev, 0x1), - dev->midi_out_buf, EP1_BUFSIZE, - snd_usb_caiaq_midi_output_done, dev); + cdev->midi_out_buf, EP1_BUFSIZE, + snd_usb_caiaq_midi_output_done, cdev); - init_waitqueue_head(&dev->ep1_wait_queue); - init_waitqueue_head(&dev->prepare_wait_queue); + init_waitqueue_head(&cdev->ep1_wait_queue); + init_waitqueue_head(&cdev->prepare_wait_queue); - if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0) + if (usb_submit_urb(&cdev->ep1_in_urb, GFP_KERNEL) != 0) return -EIO; - err = snd_usb_caiaq_send_command(dev, EP1_CMD_GET_DEVICE_INFO, NULL, 0); + err = snd_usb_caiaq_send_command(cdev, EP1_CMD_GET_DEVICE_INFO, NULL, 0); if (err) return err; - if (!wait_event_timeout(dev->ep1_wait_queue, dev->spec_received, HZ)) + if (!wait_event_timeout(cdev->ep1_wait_queue, cdev->spec_received, HZ)) return -ENODEV; usb_string(usb_dev, usb_dev->descriptor.iManufacturer, - dev->vendor_name, CAIAQ_USB_STR_LEN); + cdev->vendor_name, CAIAQ_USB_STR_LEN); usb_string(usb_dev, usb_dev->descriptor.iProduct, - dev->product_name, CAIAQ_USB_STR_LEN); + cdev->product_name, CAIAQ_USB_STR_LEN); strlcpy(card->driver, MODNAME, sizeof(card->driver)); - strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); - strlcpy(card->mixername, dev->product_name, sizeof(card->mixername)); + strlcpy(card->shortname, cdev->product_name, sizeof(card->shortname)); + strlcpy(card->mixername, cdev->product_name, sizeof(card->mixername)); /* if the id was not passed as module option, fill it with a shortened * version of the product string which does not contain any @@ -473,11 +478,10 @@ static int init_card(struct snd_usb_caiaqdev *dev) } usb_make_path(usb_dev, usbpath, sizeof(usbpath)); - snprintf(card->longname, sizeof(card->longname), - "%s %s (%s)", - dev->vendor_name, dev->product_name, usbpath); + snprintf(card->longname, sizeof(card->longname), "%s %s (%s)", + cdev->vendor_name, cdev->product_name, usbpath); - setup_card(dev); + setup_card(cdev); return 0; } @@ -486,9 +490,9 @@ static int snd_probe(struct usb_interface *intf, { int ret; struct snd_card *card = NULL; - struct usb_device *device = interface_to_usbdev(intf); + struct usb_device *usb_dev = interface_to_usbdev(intf); - ret = create_card(device, intf, &card); + ret = create_card(usb_dev, intf, &card); if (ret < 0) return ret; @@ -496,7 +500,7 @@ static int snd_probe(struct usb_interface *intf, usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { - log("unable to init card! (ret=%d)\n", ret); + dev_err(&usb_dev->dev, "unable to init card! (ret=%d)\n", ret); snd_card_free(card); return ret; } @@ -506,24 +510,25 @@ static int snd_probe(struct usb_interface *intf, static void snd_disconnect(struct usb_interface *intf) { - struct snd_usb_caiaqdev *dev; struct snd_card *card = usb_get_intfdata(intf); - - debug("%s(%p)\n", __func__, intf); + struct device *dev = intf->usb_dev; + struct snd_usb_caiaqdev *cdev; if (!card) return; - dev = caiaqdev(card); + cdev = caiaqdev(card); + dev_dbg(dev, "%s(%p)\n", __func__, intf); + snd_card_disconnect(card); #ifdef CONFIG_SND_USB_CAIAQ_INPUT - snd_usb_caiaq_input_free(dev); + snd_usb_caiaq_input_free(cdev); #endif - snd_usb_caiaq_audio_free(dev); + snd_usb_caiaq_audio_free(cdev); - usb_kill_urb(&dev->ep1_in_urb); - usb_kill_urb(&dev->midi_out_urb); + usb_kill_urb(&cdev->ep1_in_urb); + usb_kill_urb(&cdev->midi_out_urb); snd_card_free(card); usb_reset_device(interface_to_usbdev(intf)); @@ -539,4 +544,3 @@ static struct usb_driver snd_usb_driver = { }; module_usb_driver(snd_usb_driver); - diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index 562b0bff9c41..ad102fac6942 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -25,16 +25,7 @@ #define CAIAQ_USB_STR_LEN 0xff #define MAX_STREAMS 32 -//#define SND_USB_CAIAQ_DEBUG - #define MODNAME "snd-usb-caiaq" -#define log(x...) snd_printk(KERN_WARNING MODNAME" log: " x) - -#ifdef SND_USB_CAIAQ_DEBUG -#define debug(x...) snd_printk(KERN_WARNING MODNAME " debug: " x) -#else -#define debug(x...) do { } while(0) -#endif #define EP1_CMD_GET_DEVICE_INFO 0x1 #define EP1_CMD_READ_ERP 0x2 @@ -124,15 +115,16 @@ struct snd_usb_caiaqdev { }; struct snd_usb_caiaq_cb_info { - struct snd_usb_caiaqdev *dev; + struct snd_usb_caiaqdev *cdev; int index; }; #define caiaqdev(c) ((struct snd_usb_caiaqdev*)(c)->private_data) +#define caiaqdev_to_dev(d) (d->chip.card->dev) -int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, int rate, int depth, int bbp); -int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, int digital, int analog, int erp); -int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, +int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, int rate, int depth, int bbp); +int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *cdev, int digital, int analog, int erp); +int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, unsigned char command, const unsigned char *buffer, int len); diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index 26a121b42c3c..4b3fb91deecd 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/gfp.h> #include <linux/init.h> #include <linux/usb.h> @@ -199,55 +200,55 @@ static unsigned int decode_erp(unsigned char a, unsigned char b) #undef HIGH_PEAK #undef LOW_PEAK -static inline void snd_caiaq_input_report_abs(struct snd_usb_caiaqdev *dev, +static inline void snd_caiaq_input_report_abs(struct snd_usb_caiaqdev *cdev, int axis, const unsigned char *buf, int offset) { - input_report_abs(dev->input_dev, axis, + input_report_abs(cdev->input_dev, axis, (buf[offset * 2] << 8) | buf[offset * 2 + 1]); } -static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, +static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *cdev, const unsigned char *buf, unsigned int len) { - struct input_dev *input_dev = dev->input_dev; + struct input_dev *input_dev = cdev->input_dev; - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): - snd_caiaq_input_report_abs(dev, ABS_X, buf, 2); - snd_caiaq_input_report_abs(dev, ABS_Y, buf, 0); - snd_caiaq_input_report_abs(dev, ABS_Z, buf, 1); + snd_caiaq_input_report_abs(cdev, ABS_X, buf, 2); + snd_caiaq_input_report_abs(cdev, ABS_Y, buf, 0); + snd_caiaq_input_report_abs(cdev, ABS_Z, buf, 1); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): - snd_caiaq_input_report_abs(dev, ABS_X, buf, 0); - snd_caiaq_input_report_abs(dev, ABS_Y, buf, 1); - snd_caiaq_input_report_abs(dev, ABS_Z, buf, 2); + snd_caiaq_input_report_abs(cdev, ABS_X, buf, 0); + snd_caiaq_input_report_abs(cdev, ABS_Y, buf, 1); + snd_caiaq_input_report_abs(cdev, ABS_Z, buf, 2); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): - snd_caiaq_input_report_abs(dev, ABS_HAT0X, buf, 4); - snd_caiaq_input_report_abs(dev, ABS_HAT0Y, buf, 2); - snd_caiaq_input_report_abs(dev, ABS_HAT1X, buf, 6); - snd_caiaq_input_report_abs(dev, ABS_HAT1Y, buf, 1); - snd_caiaq_input_report_abs(dev, ABS_HAT2X, buf, 7); - snd_caiaq_input_report_abs(dev, ABS_HAT2Y, buf, 0); - snd_caiaq_input_report_abs(dev, ABS_HAT3X, buf, 5); - snd_caiaq_input_report_abs(dev, ABS_HAT3Y, buf, 3); + snd_caiaq_input_report_abs(cdev, ABS_HAT0X, buf, 4); + snd_caiaq_input_report_abs(cdev, ABS_HAT0Y, buf, 2); + snd_caiaq_input_report_abs(cdev, ABS_HAT1X, buf, 6); + snd_caiaq_input_report_abs(cdev, ABS_HAT1Y, buf, 1); + snd_caiaq_input_report_abs(cdev, ABS_HAT2X, buf, 7); + snd_caiaq_input_report_abs(cdev, ABS_HAT2Y, buf, 0); + snd_caiaq_input_report_abs(cdev, ABS_HAT3X, buf, 5); + snd_caiaq_input_report_abs(cdev, ABS_HAT3Y, buf, 3); break; } input_sync(input_dev); } -static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, +static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *cdev, const char *buf, unsigned int len) { - struct input_dev *input_dev = dev->input_dev; + struct input_dev *input_dev = cdev->input_dev; int i; - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): i = decode_erp(buf[0], buf[1]); input_report_abs(input_dev, ABS_X, i); @@ -299,10 +300,10 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, } } -static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, +static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *cdev, unsigned char *buf, unsigned int len) { - struct input_dev *input_dev = dev->input_dev; + struct input_dev *input_dev = cdev->input_dev; unsigned short *keycode = input_dev->keycode; int i; @@ -317,17 +318,17 @@ static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, input_report_key(input_dev, keycode[i], buf[i / 8] & (1 << (i % 8))); - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): - input_report_abs(dev->input_dev, ABS_MISC, 255 - buf[4]); + input_report_abs(cdev->input_dev, ABS_MISC, 255 - buf[4]); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): /* rotary encoders */ - input_report_abs(dev->input_dev, ABS_X, buf[5] & 0xf); - input_report_abs(dev->input_dev, ABS_Y, buf[5] >> 4); - input_report_abs(dev->input_dev, ABS_Z, buf[6] & 0xf); - input_report_abs(dev->input_dev, ABS_MISC, buf[6] >> 4); + input_report_abs(cdev->input_dev, ABS_X, buf[5] & 0xf); + input_report_abs(cdev->input_dev, ABS_Y, buf[5] >> 4); + input_report_abs(cdev->input_dev, ABS_Z, buf[6] & 0xf); + input_report_abs(cdev->input_dev, ABS_MISC, buf[6] >> 4); break; } @@ -336,10 +337,12 @@ static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, #define TKS4_MSGBLOCK_SIZE 16 -static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *dev, +static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *cdev, const unsigned char *buf, unsigned int len) { + struct device *dev = caiaqdev_to_dev(cdev); + while (len) { unsigned int i, block_id = (buf[0] << 8) | buf[1]; @@ -347,126 +350,126 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *dev, case 0: /* buttons */ for (i = 0; i < KONTROLS4_BUTTONS; i++) - input_report_key(dev->input_dev, KONTROLS4_BUTTON(i), + input_report_key(cdev->input_dev, KONTROLS4_BUTTON(i), (buf[4 + (i / 8)] >> (i % 8)) & 1); break; case 1: /* left wheel */ - input_report_abs(dev->input_dev, KONTROLS4_ABS(36), buf[9] | ((buf[8] & 0x3) << 8)); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(36), buf[9] | ((buf[8] & 0x3) << 8)); /* right wheel */ - input_report_abs(dev->input_dev, KONTROLS4_ABS(37), buf[13] | ((buf[12] & 0x3) << 8)); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(37), buf[13] | ((buf[12] & 0x3) << 8)); /* rotary encoders */ - input_report_abs(dev->input_dev, KONTROLS4_ABS(38), buf[3] & 0xf); - input_report_abs(dev->input_dev, KONTROLS4_ABS(39), buf[4] >> 4); - input_report_abs(dev->input_dev, KONTROLS4_ABS(40), buf[4] & 0xf); - input_report_abs(dev->input_dev, KONTROLS4_ABS(41), buf[5] >> 4); - input_report_abs(dev->input_dev, KONTROLS4_ABS(42), buf[5] & 0xf); - input_report_abs(dev->input_dev, KONTROLS4_ABS(43), buf[6] >> 4); - input_report_abs(dev->input_dev, KONTROLS4_ABS(44), buf[6] & 0xf); - input_report_abs(dev->input_dev, KONTROLS4_ABS(45), buf[7] >> 4); - input_report_abs(dev->input_dev, KONTROLS4_ABS(46), buf[7] & 0xf); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(38), buf[3] & 0xf); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(39), buf[4] >> 4); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(40), buf[4] & 0xf); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(41), buf[5] >> 4); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(42), buf[5] & 0xf); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(43), buf[6] >> 4); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(44), buf[6] & 0xf); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(45), buf[7] >> 4); + input_report_abs(cdev->input_dev, KONTROLS4_ABS(46), buf[7] & 0xf); break; case 2: /* Volume Fader Channel D */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(0), buf, 1); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(0), buf, 1); /* Volume Fader Channel B */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(1), buf, 2); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(1), buf, 2); /* Volume Fader Channel A */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(2), buf, 3); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(2), buf, 3); /* Volume Fader Channel C */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(3), buf, 4); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(3), buf, 4); /* Loop Volume */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(4), buf, 6); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(4), buf, 6); /* Crossfader */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(7), buf, 7); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(7), buf, 7); break; case 3: /* Tempo Fader R */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(6), buf, 3); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(6), buf, 3); /* Tempo Fader L */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(5), buf, 4); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(5), buf, 4); /* Mic Volume */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(8), buf, 6); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(8), buf, 6); /* Cue Mix */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(9), buf, 7); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(9), buf, 7); break; case 4: /* Wheel distance sensor L */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(10), buf, 1); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(10), buf, 1); /* Wheel distance sensor R */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(11), buf, 2); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(11), buf, 2); /* Channel D EQ - Filter */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(12), buf, 3); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(12), buf, 3); /* Channel D EQ - Low */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(13), buf, 4); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(13), buf, 4); /* Channel D EQ - Mid */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(14), buf, 5); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(14), buf, 5); /* Channel D EQ - Hi */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(15), buf, 6); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(15), buf, 6); /* FX2 - dry/wet */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(16), buf, 7); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(16), buf, 7); break; case 5: /* FX2 - 1 */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(17), buf, 1); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(17), buf, 1); /* FX2 - 2 */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(18), buf, 2); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(18), buf, 2); /* FX2 - 3 */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(19), buf, 3); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(19), buf, 3); /* Channel B EQ - Filter */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(20), buf, 4); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(20), buf, 4); /* Channel B EQ - Low */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(21), buf, 5); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(21), buf, 5); /* Channel B EQ - Mid */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(22), buf, 6); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(22), buf, 6); /* Channel B EQ - Hi */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(23), buf, 7); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(23), buf, 7); break; case 6: /* Channel A EQ - Filter */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(24), buf, 1); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(24), buf, 1); /* Channel A EQ - Low */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(25), buf, 2); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(25), buf, 2); /* Channel A EQ - Mid */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(26), buf, 3); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(26), buf, 3); /* Channel A EQ - Hi */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(27), buf, 4); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(27), buf, 4); /* Channel C EQ - Filter */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(28), buf, 5); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(28), buf, 5); /* Channel C EQ - Low */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(29), buf, 6); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(29), buf, 6); /* Channel C EQ - Mid */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(30), buf, 7); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(30), buf, 7); break; case 7: /* Channel C EQ - Hi */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(31), buf, 1); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(31), buf, 1); /* FX1 - wet/dry */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(32), buf, 2); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(32), buf, 2); /* FX1 - 1 */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(33), buf, 3); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(33), buf, 3); /* FX1 - 2 */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(34), buf, 4); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(34), buf, 4); /* FX1 - 3 */ - snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(35), buf, 5); + snd_caiaq_input_report_abs(cdev, KONTROLS4_ABS(35), buf, 5); break; default: - debug("%s(): bogus block (id %d)\n", + dev_dbg(dev, "%s(): bogus block (id %d)\n", __func__, block_id); return; } @@ -475,81 +478,82 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *dev, buf += TKS4_MSGBLOCK_SIZE; } - input_sync(dev->input_dev); + input_sync(cdev->input_dev); } #define MASCHINE_MSGBLOCK_SIZE 2 -static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *dev, +static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, const unsigned char *buf, unsigned int len) { unsigned int i, pad_id; - uint16_t pressure; + __le16 *pressure = (__le16 *) buf; for (i = 0; i < MASCHINE_PADS; i++) { - pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); - pad_id = pressure >> 12; - - input_report_abs(dev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + pad_id = le16_to_cpu(*pressure) >> 12; + input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), + le16_to_cpu(*pressure) & 0xfff); + pressure++; } - input_sync(dev->input_dev); + input_sync(cdev->input_dev); } static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) { - struct snd_usb_caiaqdev *dev = urb->context; + struct snd_usb_caiaqdev *cdev = urb->context; unsigned char *buf = urb->transfer_buffer; + struct device *dev = &urb->dev->dev; int ret; - if (urb->status || !dev || urb != dev->ep4_in_urb) + if (urb->status || !cdev || urb != cdev->ep4_in_urb) return; - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): if (urb->actual_length < 24) goto requeue; if (buf[0] & 0x3) - snd_caiaq_input_read_io(dev, buf + 1, 7); + snd_caiaq_input_read_io(cdev, buf + 1, 7); if (buf[0] & 0x4) - snd_caiaq_input_read_analog(dev, buf + 8, 16); + snd_caiaq_input_read_analog(cdev, buf + 8, 16); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): - snd_usb_caiaq_tks4_dispatch(dev, buf, urb->actual_length); + snd_usb_caiaq_tks4_dispatch(cdev, buf, urb->actual_length); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): if (urb->actual_length < (MASCHINE_PADS * MASCHINE_MSGBLOCK_SIZE)) goto requeue; - snd_usb_caiaq_maschine_dispatch(dev, buf, urb->actual_length); + snd_usb_caiaq_maschine_dispatch(cdev, buf, urb->actual_length); break; } requeue: - dev->ep4_in_urb->actual_length = 0; - ret = usb_submit_urb(dev->ep4_in_urb, GFP_ATOMIC); + cdev->ep4_in_urb->actual_length = 0; + ret = usb_submit_urb(cdev->ep4_in_urb, GFP_ATOMIC); if (ret < 0) - log("unable to submit urb. OOM!?\n"); + dev_err(dev, "unable to submit urb. OOM!?\n"); } static int snd_usb_caiaq_input_open(struct input_dev *idev) { - struct snd_usb_caiaqdev *dev = input_get_drvdata(idev); + struct snd_usb_caiaqdev *cdev = input_get_drvdata(idev); - if (!dev) + if (!cdev) return -EINVAL; - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): - if (usb_submit_urb(dev->ep4_in_urb, GFP_KERNEL) != 0) + if (usb_submit_urb(cdev->ep4_in_urb, GFP_KERNEL) != 0) return -EIO; break; } @@ -559,43 +563,43 @@ static int snd_usb_caiaq_input_open(struct input_dev *idev) static void snd_usb_caiaq_input_close(struct input_dev *idev) { - struct snd_usb_caiaqdev *dev = input_get_drvdata(idev); + struct snd_usb_caiaqdev *cdev = input_get_drvdata(idev); - if (!dev) + if (!cdev) return; - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): - usb_kill_urb(dev->ep4_in_urb); + usb_kill_urb(cdev->ep4_in_urb); break; } } -void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *dev, +void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *cdev, char *buf, unsigned int len) { - if (!dev->input_dev || len < 1) + if (!cdev->input_dev || len < 1) return; switch (buf[0]) { case EP1_CMD_READ_ANALOG: - snd_caiaq_input_read_analog(dev, buf + 1, len - 1); + snd_caiaq_input_read_analog(cdev, buf + 1, len - 1); break; case EP1_CMD_READ_ERP: - snd_caiaq_input_read_erp(dev, buf + 1, len - 1); + snd_caiaq_input_read_erp(cdev, buf + 1, len - 1); break; case EP1_CMD_READ_IO: - snd_caiaq_input_read_io(dev, buf + 1, len - 1); + snd_caiaq_input_read_io(cdev, buf + 1, len - 1); break; } } -int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) +int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *cdev) { - struct usb_device *usb_dev = dev->chip.dev; + struct usb_device *usb_dev = cdev->chip.dev; struct input_dev *input; int i, ret = 0; @@ -603,49 +607,49 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) if (!input) return -ENOMEM; - usb_make_path(usb_dev, dev->phys, sizeof(dev->phys)); - strlcat(dev->phys, "/input0", sizeof(dev->phys)); + usb_make_path(usb_dev, cdev->phys, sizeof(cdev->phys)); + strlcat(cdev->phys, "/input0", sizeof(cdev->phys)); - input->name = dev->product_name; - input->phys = dev->phys; + input->name = cdev->product_name; + input->phys = cdev->phys; usb_to_input_id(usb_dev, &input->id); input->dev.parent = &usb_dev->dev; - input_set_drvdata(input, dev); + input_set_drvdata(input, cdev); - switch (dev->chip.usb_id) { + switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); input->absbit[0] = BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | BIT_MASK(ABS_Z); - BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_rk2)); - memcpy(dev->keycode, keycode_rk2, sizeof(keycode_rk2)); + BUILD_BUG_ON(sizeof(cdev->keycode) < sizeof(keycode_rk2)); + memcpy(cdev->keycode, keycode_rk2, sizeof(keycode_rk2)); input->keycodemax = ARRAY_SIZE(keycode_rk2); input_set_abs_params(input, ABS_X, 0, 4096, 0, 10); input_set_abs_params(input, ABS_Y, 0, 4096, 0, 10); input_set_abs_params(input, ABS_Z, 0, 4096, 0, 10); - snd_usb_caiaq_set_auto_msg(dev, 1, 10, 0); + snd_usb_caiaq_set_auto_msg(cdev, 1, 10, 0); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); input->absbit[0] = BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | BIT_MASK(ABS_Z); - BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_rk3)); - memcpy(dev->keycode, keycode_rk3, sizeof(keycode_rk3)); + BUILD_BUG_ON(sizeof(cdev->keycode) < sizeof(keycode_rk3)); + memcpy(cdev->keycode, keycode_rk3, sizeof(keycode_rk3)); input->keycodemax = ARRAY_SIZE(keycode_rk3); input_set_abs_params(input, ABS_X, 0, 1024, 0, 10); input_set_abs_params(input, ABS_Y, 0, 1024, 0, 10); input_set_abs_params(input, ABS_Z, 0, 1024, 0, 10); - snd_usb_caiaq_set_auto_msg(dev, 1, 10, 0); + snd_usb_caiaq_set_auto_msg(cdev, 1, 10, 0); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); input->absbit[0] = BIT_MASK(ABS_X); - BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_ak1)); - memcpy(dev->keycode, keycode_ak1, sizeof(keycode_ak1)); + BUILD_BUG_ON(sizeof(cdev->keycode) < sizeof(keycode_ak1)); + memcpy(cdev->keycode, keycode_ak1, sizeof(keycode_ak1)); input->keycodemax = ARRAY_SIZE(keycode_ak1); input_set_abs_params(input, ABS_X, 0, 999, 0, 10); - snd_usb_caiaq_set_auto_msg(dev, 1, 0, 5); + snd_usb_caiaq_set_auto_msg(cdev, 1, 0, 5); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): @@ -657,8 +661,8 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | BIT_MASK(ABS_Z); input->absbit[BIT_WORD(ABS_MISC)] |= BIT_MASK(ABS_MISC); - BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_kore)); - memcpy(dev->keycode, keycode_kore, sizeof(keycode_kore)); + BUILD_BUG_ON(sizeof(cdev->keycode) < sizeof(keycode_kore)); + memcpy(cdev->keycode, keycode_kore, sizeof(keycode_kore)); input->keycodemax = ARRAY_SIZE(keycode_kore); input_set_abs_params(input, ABS_HAT0X, 0, 999, 0, 10); input_set_abs_params(input, ABS_HAT0Y, 0, 999, 0, 10); @@ -672,7 +676,7 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) input_set_abs_params(input, ABS_Y, 0, 4096, 0, 10); input_set_abs_params(input, ABS_Z, 0, 4096, 0, 10); input_set_abs_params(input, ABS_MISC, 0, 255, 0, 1); - snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + snd_usb_caiaq_set_auto_msg(cdev, 1, 10, 5); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); @@ -683,9 +687,9 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | BIT_MASK(ABS_Z); input->absbit[BIT_WORD(ABS_MISC)] |= BIT_MASK(ABS_MISC); - BUILD_BUG_ON(sizeof(dev->keycode) < KONTROLX1_INPUTS); + BUILD_BUG_ON(sizeof(cdev->keycode) < KONTROLX1_INPUTS); for (i = 0; i < KONTROLX1_INPUTS; i++) - dev->keycode[i] = BTN_MISC + i; + cdev->keycode[i] = BTN_MISC + i; input->keycodemax = KONTROLX1_INPUTS; /* analog potentiometers */ @@ -704,26 +708,26 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) input_set_abs_params(input, ABS_Z, 0, 0xf, 0, 1); input_set_abs_params(input, ABS_MISC, 0, 0xf, 0, 1); - dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); - if (!dev->ep4_in_urb) { + cdev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!cdev->ep4_in_urb) { ret = -ENOMEM; goto exit_free_idev; } - usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev, + usb_fill_bulk_urb(cdev->ep4_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x4), - dev->ep4_in_buf, EP4_BUFSIZE, - snd_usb_caiaq_ep4_reply_dispatch, dev); + cdev->ep4_in_buf, EP4_BUFSIZE, + snd_usb_caiaq_ep4_reply_dispatch, cdev); - snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + snd_usb_caiaq_set_auto_msg(cdev, 1, 10, 5); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); - BUILD_BUG_ON(sizeof(dev->keycode) < KONTROLS4_BUTTONS); + BUILD_BUG_ON(sizeof(cdev->keycode) < KONTROLS4_BUTTONS); for (i = 0; i < KONTROLS4_BUTTONS; i++) - dev->keycode[i] = KONTROLS4_BUTTON(i); + cdev->keycode[i] = KONTROLS4_BUTTON(i); input->keycodemax = KONTROLS4_BUTTONS; for (i = 0; i < KONTROLS4_AXIS; i++) { @@ -743,18 +747,18 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) for (i = 0; i < 9; i++) input_set_abs_params(input, KONTROLS4_ABS(38+i), 0, 0xf, 0, 1); - dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); - if (!dev->ep4_in_urb) { + cdev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!cdev->ep4_in_urb) { ret = -ENOMEM; goto exit_free_idev; } - usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev, + usb_fill_bulk_urb(cdev->ep4_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x4), - dev->ep4_in_buf, EP4_BUFSIZE, - snd_usb_caiaq_ep4_reply_dispatch, dev); + cdev->ep4_in_buf, EP4_BUFSIZE, + snd_usb_caiaq_ep4_reply_dispatch, cdev); - snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + snd_usb_caiaq_set_auto_msg(cdev, 1, 10, 5); break; @@ -767,8 +771,8 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) BIT_MASK(ABS_RX) | BIT_MASK(ABS_RY) | BIT_MASK(ABS_RZ); - BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_maschine)); - memcpy(dev->keycode, keycode_maschine, sizeof(keycode_maschine)); + BUILD_BUG_ON(sizeof(cdev->keycode) < sizeof(keycode_maschine)); + memcpy(cdev->keycode, keycode_maschine, sizeof(keycode_maschine)); input->keycodemax = ARRAY_SIZE(keycode_maschine); for (i = 0; i < MASCHINE_PADS; i++) { @@ -788,18 +792,18 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) input_set_abs_params(input, ABS_RY, 0, 999, 0, 10); input_set_abs_params(input, ABS_RZ, 0, 999, 0, 10); - dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); - if (!dev->ep4_in_urb) { + cdev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!cdev->ep4_in_urb) { ret = -ENOMEM; goto exit_free_idev; } - usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev, + usb_fill_bulk_urb(cdev->ep4_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x4), - dev->ep4_in_buf, EP4_BUFSIZE, - snd_usb_caiaq_ep4_reply_dispatch, dev); + cdev->ep4_in_buf, EP4_BUFSIZE, + snd_usb_caiaq_ep4_reply_dispatch, cdev); - snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + snd_usb_caiaq_set_auto_msg(cdev, 1, 10, 5); break; default: @@ -809,12 +813,12 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) input->open = snd_usb_caiaq_input_open; input->close = snd_usb_caiaq_input_close; - input->keycode = dev->keycode; + input->keycode = cdev->keycode; input->keycodesize = sizeof(unsigned short); for (i = 0; i < input->keycodemax; i++) - __set_bit(dev->keycode[i], input->keybit); + __set_bit(cdev->keycode[i], input->keybit); - dev->input_dev = input; + cdev->input_dev = input; ret = input_register_device(input); if (ret < 0) @@ -824,19 +828,19 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) exit_free_idev: input_free_device(input); - dev->input_dev = NULL; + cdev->input_dev = NULL; return ret; } -void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev) +void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *cdev) { - if (!dev || !dev->input_dev) + if (!cdev || !cdev->input_dev) return; - usb_kill_urb(dev->ep4_in_urb); - usb_free_urb(dev->ep4_in_urb); - dev->ep4_in_urb = NULL; + usb_kill_urb(cdev->ep4_in_urb); + usb_free_urb(cdev->ep4_in_urb); + cdev->ep4_in_urb = NULL; - input_unregister_device(dev->input_dev); - dev->input_dev = NULL; + input_unregister_device(cdev->input_dev); + cdev->input_dev = NULL; } diff --git a/sound/usb/caiaq/input.h b/sound/usb/caiaq/input.h index ced535577864..6014e2713a60 100644 --- a/sound/usb/caiaq/input.h +++ b/sound/usb/caiaq/input.h @@ -1,8 +1,8 @@ #ifndef CAIAQ_INPUT_H #define CAIAQ_INPUT_H -void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *dev, char *buf, unsigned int len); -int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev); -void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev); +void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *cdev, char *buf, unsigned int len); +int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *cdev); +void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *cdev); #endif diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index a1a47088fd0c..2d7588461b33 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/usb.h> #include <linux/gfp.h> #include <sound/rawmidi.h> @@ -37,12 +38,12 @@ static int snd_usb_caiaq_midi_input_close(struct snd_rawmidi_substream *substrea static void snd_usb_caiaq_midi_input_trigger(struct snd_rawmidi_substream *substream, int up) { - struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; + struct snd_usb_caiaqdev *cdev = substream->rmidi->private_data; - if (!dev) + if (!cdev) return; - dev->midi_receive_substream = up ? substream : NULL; + cdev->midi_receive_substream = up ? substream : NULL; } @@ -53,49 +54,50 @@ static int snd_usb_caiaq_midi_output_open(struct snd_rawmidi_substream *substrea static int snd_usb_caiaq_midi_output_close(struct snd_rawmidi_substream *substream) { - struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - if (dev->midi_out_active) { - usb_kill_urb(&dev->midi_out_urb); - dev->midi_out_active = 0; + struct snd_usb_caiaqdev *cdev = substream->rmidi->private_data; + if (cdev->midi_out_active) { + usb_kill_urb(&cdev->midi_out_urb); + cdev->midi_out_active = 0; } return 0; } -static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, +static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *cdev, struct snd_rawmidi_substream *substream) { int len, ret; + struct device *dev = caiaqdev_to_dev(cdev); - dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; - dev->midi_out_buf[1] = 0; /* port */ - len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, + cdev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; + cdev->midi_out_buf[1] = 0; /* port */ + len = snd_rawmidi_transmit(substream, cdev->midi_out_buf + 3, EP1_BUFSIZE - 3); if (len <= 0) return; - dev->midi_out_buf[2] = len; - dev->midi_out_urb.transfer_buffer_length = len+3; + cdev->midi_out_buf[2] = len; + cdev->midi_out_urb.transfer_buffer_length = len+3; - ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); + ret = usb_submit_urb(&cdev->midi_out_urb, GFP_ATOMIC); if (ret < 0) - log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," - "ret=%d, len=%d\n", - substream, ret, len); + dev_err(dev, + "snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," + "ret=%d, len=%d\n", substream, ret, len); else - dev->midi_out_active = 1; + cdev->midi_out_active = 1; } static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { - struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; + struct snd_usb_caiaqdev *cdev = substream->rmidi->private_data; if (up) { - dev->midi_out_substream = substream; - if (!dev->midi_out_active) - snd_usb_caiaq_midi_send(dev, substream); + cdev->midi_out_substream = substream; + if (!cdev->midi_out_active) + snd_usb_caiaq_midi_send(cdev, substream); } else { - dev->midi_out_substream = NULL; + cdev->midi_out_substream = NULL; } } @@ -114,13 +116,13 @@ static struct snd_rawmidi_ops snd_usb_caiaq_midi_input = .trigger = snd_usb_caiaq_midi_input_trigger, }; -void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, +void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *cdev, int port, const char *buf, int len) { - if (!dev->midi_receive_substream) + if (!cdev->midi_receive_substream) return; - snd_rawmidi_receive(dev->midi_receive_substream, buf, len); + snd_rawmidi_receive(cdev->midi_receive_substream, buf, len); } int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) @@ -160,15 +162,14 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { - struct snd_usb_caiaqdev *dev = urb->context; + struct snd_usb_caiaqdev *cdev = urb->context; - dev->midi_out_active = 0; + cdev->midi_out_active = 0; if (urb->status != 0) return; - if (!dev->midi_out_substream) + if (!cdev->midi_out_substream) return; - snd_usb_caiaq_midi_send(dev, dev->midi_out_substream); + snd_usb_caiaq_midi_send(cdev, cdev->midi_out_substream); } - diff --git a/sound/usb/caiaq/midi.h b/sound/usb/caiaq/midi.h index 380f984babc9..60bf3442b283 100644 --- a/sound/usb/caiaq/midi.h +++ b/sound/usb/caiaq/midi.h @@ -1,8 +1,9 @@ #ifndef CAIAQ_MIDI_H #define CAIAQ_MIDI_H -int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *dev); -void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len); +int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *cdev); +void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *cdev, + int port, const char *buf, int len); void snd_usb_caiaq_midi_output_done(struct urb *urb); #endif /* CAIAQ_MIDI_H */ diff --git a/sound/usb/card.c b/sound/usb/card.c index 2da8ad75fd96..1a033177b83f 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -82,6 +82,7 @@ static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; static int nrpacks = 8; /* max. number of packets per urb */ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; +static bool autoclock = true; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); @@ -100,6 +101,8 @@ MODULE_PARM_DESC(device_setup, "Specific device setup (if needed)."); module_param(ignore_ctl_error, bool, 0444); MODULE_PARM_DESC(ignore_ctl_error, "Ignore errors from USB controller for mixer interfaces."); +module_param(autoclock, bool, 0444); +MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); /* * we keep the snd_usb_audio_t instances by ourselves for merging @@ -354,6 +357,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, chip->card = card; chip->setup = device_setup[idx]; chip->nrpacks = nrpacks; + chip->autoclock = autoclock; chip->probing = 1; chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), @@ -627,7 +631,9 @@ int snd_usb_autoresume(struct snd_usb_audio *chip) int err = -ENODEV; down_read(&chip->shutdown_rwsem); - if (!chip->shutdown && !chip->probing) + if (chip->probing) + err = 0; + else if (!chip->shutdown) err = usb_autopm_get_interface(chip->pm_intf); up_read(&chip->shutdown_rwsem); @@ -645,7 +651,6 @@ void snd_usb_autosuspend(struct snd_usb_audio *chip) static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) { struct snd_usb_audio *chip = usb_get_intfdata(intf); - struct list_head *p; struct snd_usb_stream *as; struct usb_mixer_interface *mixer; @@ -655,8 +660,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) if (!PMSG_IS_AUTO(message)) { snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); if (!chip->num_suspended_intf++) { - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); + list_for_each_entry(as, &chip->pcm_list, list) { snd_pcm_suspend_all(as->pcm); as->substream[0].need_setup_ep = as->substream[1].need_setup_ep = true; @@ -716,8 +720,7 @@ static struct usb_device_id usb_audio_ids [] = { .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { } /* Terminating entry */ }; - -MODULE_DEVICE_TABLE (usb, usb_audio_ids); +MODULE_DEVICE_TABLE(usb, usb_audio_ids); /* * entry point for linux usb interface diff --git a/sound/usb/card.h b/sound/usb/card.h index 8a751b4887ea..bf2889a2cae5 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -28,6 +28,8 @@ struct audioformat { unsigned int *rate_table; /* rate table */ unsigned char clock; /* associated clock */ struct snd_pcm_chmap_elem *chmap; /* (optional) channel map */ + bool dsd_dop; /* add DOP headers in case of DSD samples */ + bool dsd_bitrev; /* reverse the bits of each DSD sample */ }; struct snd_usb_substream; @@ -116,6 +118,7 @@ struct snd_usb_substream { unsigned int altset_idx; /* USB data format: index of alternate setting */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ + unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */ unsigned int running: 1; /* running status */ @@ -138,6 +141,12 @@ struct snd_usb_substream { int last_frame_number; /* stored frame number */ int last_delay; /* stored delay */ + + struct { + int marker; + int channel; + int byte_idx; + } dsd_dop; }; struct snd_usb_stream { diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 9e2703a25156..3a2ce390e278 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -32,6 +32,7 @@ #include "card.h" #include "helper.h" #include "clock.h" +#include "quirks.h" static struct uac_clock_source_descriptor * snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, @@ -99,6 +100,41 @@ static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_i return buf; } +static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_id, + unsigned char pin) +{ + int ret; + + ret = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), + UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + UAC2_CX_CLOCK_SELECTOR << 8, + snd_usb_ctrl_intf(chip) | (selector_id << 8), + &pin, sizeof(pin)); + if (ret < 0) + return ret; + + if (ret != sizeof(pin)) { + snd_printk(KERN_ERR + "usb-audio:%d: setting selector (id %d) unexpected length %d\n", + chip->dev->devnum, selector_id, ret); + return -EINVAL; + } + + ret = uac_clock_selector_get_val(chip, selector_id); + if (ret < 0) + return ret; + + if (ret != pin) { + snd_printk(KERN_ERR + "usb-audio:%d: setting selector (id %d) to %x failed (current: %d)\n", + chip->dev->devnum, selector_id, pin, ret); + return -EINVAL; + } + + return ret; +} + static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) { int err; @@ -131,7 +167,8 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) } static int __uac_clock_find_source(struct snd_usb_audio *chip, - int entity_id, unsigned long *visited) + int entity_id, unsigned long *visited, + bool validate) { struct uac_clock_source_descriptor *source; struct uac_clock_selector_descriptor *selector; @@ -148,12 +185,19 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, /* first, see if the ID we're looking for is a clock source already */ source = snd_usb_find_clock_source(chip->ctrl_intf, entity_id); - if (source) - return source->bClockID; + if (source) { + entity_id = source->bClockID; + if (validate && !uac_clock_source_is_valid(chip, entity_id)) { + snd_printk(KERN_ERR "usb-audio:%d: clock source %d is not valid, cannot use\n", + chip->dev->devnum, entity_id); + return -ENXIO; + } + return entity_id; + } selector = snd_usb_find_clock_selector(chip->ctrl_intf, entity_id); if (selector) { - int ret; + int ret, i, cur; /* the entity ID we are looking for is a selector. * find out what it currently selects */ @@ -164,22 +208,49 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, /* Selector values are one-based */ if (ret > selector->bNrInPins || ret < 1) { - printk(KERN_ERR + snd_printk(KERN_ERR "%s(): selector reported illegal value, id %d, ret %d\n", __func__, selector->bClockID, ret); return -EINVAL; } - return __uac_clock_find_source(chip, selector->baCSourceID[ret-1], - visited); + cur = ret; + ret = __uac_clock_find_source(chip, selector->baCSourceID[ret - 1], + visited, validate); + if (!validate || ret > 0 || !chip->autoclock) + return ret; + + /* The current clock source is invalid, try others. */ + for (i = 1; i <= selector->bNrInPins; i++) { + int err; + + if (i == cur) + continue; + + ret = __uac_clock_find_source(chip, selector->baCSourceID[i - 1], + visited, true); + if (ret < 0) + continue; + + err = uac_clock_selector_set_val(chip, entity_id, i); + if (err < 0) + continue; + + snd_printk(KERN_INFO + "usb-audio:%d: found and selected valid clock source %d\n", + chip->dev->devnum, ret); + return ret; + } + + return -ENXIO; } /* FIXME: multipliers only act as pass-thru element for now */ multiplier = snd_usb_find_clock_multiplier(chip->ctrl_intf, entity_id); if (multiplier) return __uac_clock_find_source(chip, multiplier->bCSourceID, - visited); + visited, validate); return -EINVAL; } @@ -195,11 +266,12 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, * * Returns the clock source UnitID (>=0) on success, or an error. */ -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int entity_id) +int snd_usb_clock_find_source(struct snd_usb_audio *chip, int entity_id, + bool validate) { DECLARE_BITMAP(visited, 256); memset(visited, 0, sizeof(visited)); - return __uac_clock_find_source(chip, entity_id, visited); + return __uac_clock_find_source(chip, entity_id, visited, validate); } static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, @@ -247,66 +319,73 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, return 0; } +static int get_sample_rate_v2(struct snd_usb_audio *chip, int iface, + int altsetting, int clock) +{ + struct usb_device *dev = chip->dev; + __le32 data; + int err; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + &data, sizeof(data)); + if (err < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2): err %d\n", + dev->devnum, iface, altsetting, err); + return 0; + } + + return le32_to_cpu(data); +} + static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate) { struct usb_device *dev = chip->dev; - unsigned char data[4]; + __le32 data; int err, cur_rate, prev_rate; - int clock = snd_usb_clock_find_source(chip, fmt->clock); + int clock; + bool writeable; + struct uac_clock_source_descriptor *cs_desc; + clock = snd_usb_clock_find_source(chip, fmt->clock, true); if (clock < 0) return clock; - if (!uac_clock_source_is_valid(chip, clock)) { - /* TODO: should we try to find valid clock setups by ourself? */ - snd_printk(KERN_ERR "%d:%d:%d: clock source %d is not valid, cannot use\n", - dev->devnum, iface, fmt->altsetting, clock); - return -ENXIO; - } - - err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data)); - if (err < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", - dev->devnum, iface, fmt->altsetting); - prev_rate = 0; - } else { - prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - } + prev_rate = get_sample_rate_v2(chip, iface, fmt->altsetting, clock); + if (prev_rate == rate) + return 0; - data[0] = rate; - data[1] = rate >> 8; - data[2] = rate >> 16; - data[3] = rate >> 24; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, - UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data))) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", - dev->devnum, iface, fmt->altsetting, rate); - return err; - } + cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, clock); + writeable = uac2_control_is_writeable(cs_desc->bmControls, UAC2_CS_CONTROL_SAM_FREQ - 1); + if (writeable) { + data = cpu_to_le32(rate); + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + &data, sizeof(data)); + if (err < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2): err %d\n", + dev->devnum, iface, fmt->altsetting, rate, err); + return err; + } - err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data)); - if (err < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", - dev->devnum, iface, fmt->altsetting); - cur_rate = 0; + cur_rate = get_sample_rate_v2(chip, iface, fmt->altsetting, clock); } else { - cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + cur_rate = prev_rate; } if (cur_rate != rate) { + if (!writeable) { + snd_printk(KERN_WARNING + "%d:%d:%d: freq mismatch (RO clock): req %d, clock runs @%d\n", + dev->devnum, iface, fmt->altsetting, rate, cur_rate); + return -ENXIO; + } snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", cur_rate, rate); @@ -316,7 +395,9 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, * interface is active. */ if (rate != prev_rate) { usb_set_interface(dev, iface, 0); + snd_usb_set_interface_quirk(dev); usb_set_interface(dev, iface, fmt->altsetting); + snd_usb_set_interface_quirk(dev); } return 0; diff --git a/sound/usb/clock.h b/sound/usb/clock.h index 46630936d31f..d592e4a29856 100644 --- a/sound/usb/clock.h +++ b/sound/usb/clock.h @@ -5,6 +5,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate); -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int entity_id); +int snd_usb_clock_find_source(struct snd_usb_audio *chip, int entity_id, + bool validate); #endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 21049b882ee6..7a444b5501d9 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -128,7 +128,7 @@ static const char *usb_error_string(int err) * Determine whether an endpoint is driven by an implicit feedback * data endpoint source. */ -int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) { return ep->sync_master && ep->sync_master->type == SND_USB_ENDPOINT_TYPE_DATA && @@ -363,7 +363,7 @@ static void snd_complete_urb(struct urb *urb) if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) goto exit_clear; - if (snd_usb_endpoint_implict_feedback_sink(ep)) { + if (snd_usb_endpoint_implicit_feedback_sink(ep)) { unsigned long flags; spin_lock_irqsave(&ep->lock, flags); @@ -415,14 +415,12 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, struct usb_host_interface *alts, int ep_num, int direction, int type) { - struct list_head *p; struct snd_usb_endpoint *ep; int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; mutex_lock(&chip->mutex); - list_for_each(p, &chip->ep_list) { - ep = list_entry(p, struct snd_usb_endpoint, list); + list_for_each_entry(ep, &chip->ep_list, list) { if (ep->ep_num == ep_num && ep->iface == alts->desc.bInterfaceNumber && ep->alt_idx == alts->desc.bAlternateSetting) { @@ -580,6 +578,15 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, int is_playback = usb_pipeout(ep->pipe); int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; + if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { + /* + * When operating in DSD DOP mode, the size of a sample frame + * in hardware differs from the actual physical format width + * because we need to make room for the DOP markers. + */ + frame_bits += channels << 3; + } + ep->datainterval = fmt->datainterval; ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; @@ -607,7 +614,7 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, else packs_per_ms = 1; - if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) { + if (is_playback && !snd_usb_endpoint_implicit_feedback_sink(ep)) { urb_packs = max(ep->chip->nrpacks, 1); urb_packs = min(urb_packs, (unsigned int) MAX_PACKS); } else { @@ -616,11 +623,11 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, urb_packs *= packs_per_ms; - if (sync_ep && !snd_usb_endpoint_implict_feedback_sink(ep)) + if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) urb_packs = min(urb_packs, 1U << sync_ep->syncinterval); /* decide how many packets to be used */ - if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) { + if (is_playback && !snd_usb_endpoint_implicit_feedback_sink(ep)) { unsigned int minsize, maxpacks; /* determine how small a packet can be */ minsize = (ep->freqn >> (16 - ep->datainterval)) @@ -677,7 +684,7 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, if (!u->urb->transfer_buffer) goto out_of_memory; u->urb->pipe = ep->pipe; - u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; + u->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP; u->urb->interval = 1 << ep->datainterval; u->urb->context = u; u->urb->complete = snd_complete_urb; @@ -716,8 +723,7 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep, u->urb->transfer_dma = ep->sync_dma + i * 4; u->urb->transfer_buffer_length = 4; u->urb->pipe = ep->pipe; - u->urb->transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; + u->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP; u->urb->number_of_packets = 1; u->urb->interval = 1 << ep->syncinterval; u->urb->context = u; @@ -847,7 +853,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep) set_bit(EP_FLAG_RUNNING, &ep->flags); - if (snd_usb_endpoint_implict_feedback_sink(ep)) { + if (snd_usb_endpoint_implicit_feedback_sink(ep)) { for (i = 0; i < ep->nurbs; i++) { struct snd_urb_ctx *ctx = ep->urb + i; list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); @@ -990,7 +996,7 @@ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, * and add it to the list of pending urbs. queue_pending_output_urbs() * will take care of them later. */ - if (snd_usb_endpoint_implict_feedback_sink(ep) && + if (snd_usb_endpoint_implicit_feedback_sink(ep) && ep->use_count != 0) { /* implicit feedback case */ diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 447902dd8a4a..2287adf5ca59 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -23,7 +23,7 @@ int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct list_head *head); -int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/format.c b/sound/usb/format.c index e831ee4238bb..99299ffb33ac 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -43,11 +43,11 @@ */ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *_fmt, + unsigned int format, void *_fmt, int protocol) { int sample_width, sample_bytes; - u64 pcm_formats; + u64 pcm_formats = 0; switch (protocol) { case UAC_VERSION_1: @@ -63,14 +63,17 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct uac_format_type_i_ext_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubslotSize; + + if (format & UAC2_FORMAT_TYPE_I_RAW_DATA) + pcm_formats |= SNDRV_PCM_FMTBIT_SPECIAL; + format <<= 1; break; } } - pcm_formats = 0; - - if (format == 0 || format == (1 << UAC_FORMAT_TYPE_I_UNDEFINED)) { + if ((pcm_formats == 0) && + (format == 0 || format == (1 << UAC_FORMAT_TYPE_I_UNDEFINED))) { /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", chip->dev->devnum, fp->iface, fp->altsetting); @@ -133,6 +136,9 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, snd_printk(KERN_INFO "%d:%u:%d : unsupported format bits %#x\n", chip->dev->devnum, fp->iface, fp->altsetting, format); } + + pcm_formats |= snd_usb_interface_dsd_format_quirks(chip, fp, sample_bytes); + return pcm_formats; } @@ -277,7 +283,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; int nr_triplets, data_size, ret = 0; - int clock = snd_usb_clock_find_source(chip, fp->clock); + int clock = snd_usb_clock_find_source(chip, fp->clock, false); if (clock < 0) { snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n", @@ -353,13 +359,14 @@ err: * parse the format type I and III descriptors */ static int parse_audio_format_i(struct snd_usb_audio *chip, - struct audioformat *fp, int format, + struct audioformat *fp, unsigned int format, struct uac_format_type_i_continuous_descriptor *fmt, struct usb_host_interface *iface) { struct usb_interface_descriptor *altsd = get_iface_desc(iface); int protocol = altsd->bInterfaceProtocol; - int pcm_format, ret; + snd_pcm_format_t pcm_format; + int ret; if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx @@ -378,7 +385,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, default: pcm_format = SNDRV_PCM_FORMAT_S16_LE; } - fp->formats = 1uLL << pcm_format; + fp->formats = pcm_format_to_bits(pcm_format); } else { fp->formats = parse_audio_format_i_type(chip, fp, format, fmt, protocol); @@ -473,8 +480,9 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, return ret; } -int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, struct uac_format_type_i_continuous_descriptor *fmt, +int snd_usb_parse_audio_format(struct snd_usb_audio *chip, + struct audioformat *fp, unsigned int format, + struct uac_format_type_i_continuous_descriptor *fmt, int stream, struct usb_host_interface *iface) { int err; diff --git a/sound/usb/format.h b/sound/usb/format.h index 387924f0af85..6f315226f320 100644 --- a/sound/usb/format.h +++ b/sound/usb/format.h @@ -2,7 +2,7 @@ #define __USBAUDIO_FORMAT_H int snd_usb_parse_audio_format(struct snd_usb_audio *chip, - struct audioformat *fp, int format, + struct audioformat *fp, unsigned int format, struct uac_format_type_i_continuous_descriptor *fmt, int stream, struct usb_host_interface *iface); diff --git a/sound/usb/helper.c b/sound/usb/helper.c index c1db28f874c2..620902463c6e 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -86,14 +86,22 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, { int err; void *buf = NULL; + int timeout; if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) return -ENOMEM; } + + if (requesttype & USB_DIR_IN) + timeout = USB_CTRL_GET_TIMEOUT; + else + timeout = USB_CTRL_SET_TIMEOUT; + err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, 1000); + value, index, buf, size, timeout); + if (size > 0) { memcpy(data, buf, size); kfree(buf); diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 34b9bb7fe87c..8e01fa4991c5 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -126,7 +126,6 @@ struct snd_usb_midi { struct snd_usb_midi_in_endpoint *in; } endpoints[MIDI_MAX_ENDPOINTS]; unsigned long input_triggered; - bool autopm_reference; unsigned int opened[2]; unsigned char disconnected; unsigned char input_running; @@ -1040,7 +1039,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, { struct snd_usb_midi* umidi = substream->rmidi->private_data; struct snd_kcontrol *ctl; - int err; down_read(&umidi->disc_rwsem); if (umidi->disconnected) { @@ -1051,13 +1049,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, mutex_lock(&umidi->mutex); if (open) { if (!umidi->opened[0] && !umidi->opened[1]) { - err = usb_autopm_get_interface(umidi->iface); - umidi->autopm_reference = err >= 0; - if (err < 0 && err != -EACCES) { - mutex_unlock(&umidi->mutex); - up_read(&umidi->disc_rwsem); - return -EIO; - } if (umidi->roland_load_ctl) { ctl = umidi->roland_load_ctl; ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -1080,8 +1071,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, snd_ctl_notify(umidi->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); } - if (umidi->autopm_reference) - usb_autopm_put_interface(umidi->iface); } } mutex_unlock(&umidi->mutex); @@ -1455,6 +1444,7 @@ void snd_usbmidi_disconnect(struct list_head* p) } del_timer_sync(&umidi->error_timer); } +EXPORT_SYMBOL(snd_usbmidi_disconnect); static void snd_usbmidi_rawmidi_free(struct snd_rawmidi *rmidi) { @@ -1465,10 +1455,9 @@ static void snd_usbmidi_rawmidi_free(struct snd_rawmidi *rmidi) static struct snd_rawmidi_substream *snd_usbmidi_find_substream(struct snd_usb_midi* umidi, int stream, int number) { - struct list_head* list; + struct snd_rawmidi_substream *substream; - list_for_each(list, &umidi->rmidi->streams[stream].substreams) { - struct snd_rawmidi_substream *substream = list_entry(list, struct snd_rawmidi_substream, list); + list_for_each_entry(substream, &umidi->rmidi->streams[stream].substreams, list) { if (substream->number == number) return substream; } @@ -2091,6 +2080,7 @@ void snd_usbmidi_input_stop(struct list_head* p) } umidi->input_running = 0; } +EXPORT_SYMBOL(snd_usbmidi_input_stop); static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep) { @@ -2120,6 +2110,7 @@ void snd_usbmidi_input_start(struct list_head* p) snd_usbmidi_input_start_ep(umidi->endpoints[i].in); umidi->input_running = 1; } +EXPORT_SYMBOL(snd_usbmidi_input_start); /* * Creates and registers everything needed for a MIDI streaming interface. @@ -2256,11 +2247,9 @@ int snd_usbmidi_create(struct snd_card *card, return err; } + usb_autopm_get_interface_no_resume(umidi->iface); + list_add_tail(&umidi->list, midi_list); return 0; } - EXPORT_SYMBOL(snd_usbmidi_create); -EXPORT_SYMBOL(snd_usbmidi_input_stop); -EXPORT_SYMBOL(snd_usbmidi_input_start); -EXPORT_SYMBOL(snd_usbmidi_disconnect); diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 8b81cb54026f..6ad617b94732 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -1120,8 +1120,7 @@ static int alloc_stream_urbs(struct ua101 *ua, struct ua101_stream *stream, usb_init_urb(&urb->urb); urb->urb.dev = ua->dev; urb->urb.pipe = stream->usb_pipe; - urb->urb.transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; + urb->urb.transfer_flags = URB_NO_TRANSFER_DMA_MAP; urb->urb.transfer_buffer = addr; urb->urb.transfer_dma = dma; urb->urb.transfer_buffer_length = max_packet_size; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f94397b42aa5..93b6e32cfead 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/slab.h> +#include <linux/bitrev.h> #include <linux/ratelimit.h> #include <linux/usb.h> #include <linux/usb/audio.h> @@ -94,14 +95,12 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream */ static struct audioformat *find_format(struct snd_usb_substream *subs) { - struct list_head *p; + struct audioformat *fp; struct audioformat *found = NULL; int cur_attr = 0, attr; - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); - if (!(fp->formats & (1uLL << subs->pcm_format))) + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!(fp->formats & pcm_format_to_bits(subs->pcm_format))) continue; if (fp->channels != subs->channels) continue; @@ -350,6 +349,8 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) fmt->iface, fmt->altsetting); subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; + + snd_usb_set_interface_quirk(dev); } subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, @@ -477,7 +478,7 @@ static int match_endpoint_audioformats(struct audioformat *fp, return 0; } - if (!(fp->formats & (1ULL << pcm_format))) { + if (!(fp->formats & pcm_format_to_bits(pcm_format))) { snd_printdd("%s: (fmt @%p) no match for format %d\n", __func__, fp, pcm_format); return 0; @@ -802,7 +803,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_usb_substream *subs = rule->private; - struct list_head *p; + struct audioformat *fp; struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); unsigned int rmin, rmax; int changed; @@ -810,9 +811,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, hwc_debug("hw_rule_rate: (%d,%d)\n", it->min, it->max); changed = 0; rmin = rmax = 0; - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); + list_for_each_entry(fp, &subs->fmt_list, list) { if (!hw_check_valid_format(subs, params, fp)) continue; if (changed++) { @@ -856,7 +855,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_usb_substream *subs = rule->private; - struct list_head *p; + struct audioformat *fp; struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); unsigned int rmin, rmax; int changed; @@ -864,9 +863,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, hwc_debug("hw_rule_channels: (%d,%d)\n", it->min, it->max); changed = 0; rmin = rmax = 0; - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); + list_for_each_entry(fp, &subs->fmt_list, list) { if (!hw_check_valid_format(subs, params, fp)) continue; if (changed++) { @@ -909,7 +906,7 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_usb_substream *subs = rule->private; - struct list_head *p; + struct audioformat *fp; struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); u64 fbits; u32 oldbits[2]; @@ -917,9 +914,7 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); fbits = 0; - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); + list_for_each_entry(fp, &subs->fmt_list, list) { if (!hw_check_valid_format(subs, params, fp)) continue; fbits |= fp->formats; @@ -1027,7 +1022,7 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { - struct list_head *p; + struct audioformat *fp; unsigned int pt, ptmin; int param_period_time_if_needed; int err; @@ -1041,9 +1036,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.rates = 0; ptmin = UINT_MAX; /* check min/max rates and channels */ - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); + list_for_each_entry(fp, &subs->fmt_list, list) { runtime->hw.rates |= fp->rates; if (runtime->hw.rate_min > fp->rate_min) runtime->hw.rate_min = fp->rate_min; @@ -1128,6 +1121,12 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) runtime->private_data = subs; subs->pcm_substream = substream; /* runtime PM is also done there */ + + /* initialize DSD/DOP context */ + subs->dsd_dop.byte_idx = 0; + subs->dsd_dop.channel = 0; + subs->dsd_dop.marker = 1; + return setup_hw_info(runtime, subs); } @@ -1170,7 +1169,7 @@ static void retire_capture_urb(struct snd_usb_substream *subs, stride = runtime->frame_bits >> 3; for (i = 0; i < urb->number_of_packets; i++) { - cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; + cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset + subs->pkt_offset_adj; if (urb->iso_frame_desc[i].status && printk_ratelimit()) { snd_printdd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; @@ -1222,6 +1221,61 @@ static void retire_capture_urb(struct snd_usb_substream *subs, snd_pcm_period_elapsed(subs->pcm_substream); } +static inline void fill_playback_urb_dsd_dop(struct snd_usb_substream *subs, + struct urb *urb, unsigned int bytes) +{ + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + unsigned int stride = runtime->frame_bits >> 3; + unsigned int dst_idx = 0; + unsigned int src_idx = subs->hwptr_done; + unsigned int wrap = runtime->buffer_size * stride; + u8 *dst = urb->transfer_buffer; + u8 *src = runtime->dma_area; + u8 marker[] = { 0x05, 0xfa }; + + /* + * The DSP DOP format defines a way to transport DSD samples over + * normal PCM data endpoints. It requires stuffing of marker bytes + * (0x05 and 0xfa, alternating per sample frame), and then expects + * 2 additional bytes of actual payload. The whole frame is stored + * LSB. + * + * Hence, for a stereo transport, the buffer layout looks like this, + * where L refers to left channel samples and R to right. + * + * L1 L2 0x05 R1 R2 0x05 L3 L4 0xfa R3 R4 0xfa + * L5 L6 0x05 R5 R6 0x05 L7 L8 0xfa R7 R8 0xfa + * ..... + * + */ + + while (bytes--) { + if (++subs->dsd_dop.byte_idx == 3) { + /* frame boundary? */ + dst[dst_idx++] = marker[subs->dsd_dop.marker]; + src_idx += 2; + subs->dsd_dop.byte_idx = 0; + + if (++subs->dsd_dop.channel % runtime->channels == 0) { + /* alternate the marker */ + subs->dsd_dop.marker++; + subs->dsd_dop.marker %= ARRAY_SIZE(marker); + subs->dsd_dop.channel = 0; + } + } else { + /* stuff the DSD payload */ + int idx = (src_idx + subs->dsd_dop.byte_idx - 1) % wrap; + + if (subs->cur_audiofmt->dsd_bitrev) + dst[dst_idx++] = bitrev8(src[idx]); + else + dst[dst_idx++] = src[idx]; + + subs->hwptr_done++; + } + } +} + static void prepare_playback_urb(struct snd_usb_substream *subs, struct urb *urb) { @@ -1244,8 +1298,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, counts = snd_usb_endpoint_next_packet_size(ep); /* set up descriptor */ - urb->iso_frame_desc[i].offset = frames * stride; - urb->iso_frame_desc[i].length = counts * stride; + urb->iso_frame_desc[i].offset = frames * ep->stride; + urb->iso_frame_desc[i].length = counts * ep->stride; frames += counts; urb->number_of_packets++; subs->transfer_done += counts; @@ -1259,14 +1313,14 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, frames -= subs->transfer_done; counts -= subs->transfer_done; urb->iso_frame_desc[i].length = - counts * stride; + counts * ep->stride; subs->transfer_done = 0; } i++; if (i < ctx->packets) { /* add a transfer delimiter */ urb->iso_frame_desc[i].offset = - frames * stride; + frames * ep->stride; urb->iso_frame_desc[i].length = 0; urb->number_of_packets++; } @@ -1274,23 +1328,43 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, } } if (period_elapsed && - !snd_usb_endpoint_implict_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */ + !snd_usb_endpoint_implicit_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */ break; } - bytes = frames * stride; - if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { - /* err, the transferred area goes over buffer boundary. */ - unsigned int bytes1 = - runtime->buffer_size * stride - subs->hwptr_done; - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes1); - memcpy(urb->transfer_buffer + bytes1, - runtime->dma_area, bytes - bytes1); + bytes = frames * ep->stride; + + if (unlikely(subs->pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && + subs->cur_audiofmt->dsd_dop)) { + fill_playback_urb_dsd_dop(subs, urb, bytes); + } else if (unlikely(subs->pcm_format == SNDRV_PCM_FORMAT_DSD_U8 && + subs->cur_audiofmt->dsd_bitrev)) { + /* bit-reverse the bytes */ + u8 *buf = urb->transfer_buffer; + for (i = 0; i < bytes; i++) { + int idx = (subs->hwptr_done + i) + % (runtime->buffer_size * stride); + buf[i] = bitrev8(runtime->dma_area[idx]); + } + + subs->hwptr_done += bytes; } else { - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes); + /* usual PCM */ + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { + /* err, the transferred area goes over buffer boundary. */ + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); + } else { + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes); + } + + subs->hwptr_done += bytes; } - subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; @@ -1318,8 +1392,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs, { unsigned long flags; struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; - int stride = runtime->frame_bits >> 3; - int processed = urb->transfer_buffer_length / stride; + struct snd_usb_endpoint *ep = subs->data_endpoint; + int processed = urb->transfer_buffer_length / ep->stride; int est_delay; /* ignore the delay accounting when procssed=0 is given, i.e. diff --git a/sound/usb/proc.c b/sound/usb/proc.c index d218f763501f..135c76871063 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -73,20 +73,19 @@ void snd_usb_audio_create_proc(struct snd_usb_audio *chip) */ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct snd_info_buffer *buffer) { - struct list_head *p; + struct audioformat *fp; static char *sync_types[4] = { "NONE", "ASYNC", "ADAPTIVE", "SYNC" }; - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; + list_for_each_entry(fp, &subs->fmt_list, list) { snd_pcm_format_t fmt; - fp = list_entry(p, struct audioformat, list); + snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); snd_iprintf(buffer, " Format:"); for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt) - if (fp->formats & (1uLL << fmt)) + if (fp->formats & pcm_format_to_bits(fmt)) snd_iprintf(buffer, " %s", snd_pcm_format_name(fmt)); snd_iprintf(buffer, "\n"); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c39f898b15d2..7f1722f82c89 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -353,6 +353,84 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x1507), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "THR10", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = -1 + } + } + } +}, +{ + USB_DEVICE(0x0499, 0x150a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "THR5A", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = -1 + } + } + } +}, +{ + USB_DEVICE(0x0499, 0x150c), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "THR10C", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), @@ -2748,6 +2826,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE(0x1235, 0x0018), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Novation", + .product_name = "Twitch", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, +{ USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Novation", @@ -2996,7 +3114,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x02, .ep_attr = 0x01, - .maxpacksize = 0x130, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .rate_min = 44100, @@ -3044,7 +3161,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .attributes = 0x00, .endpoint = 0x03, .ep_attr = USB_ENDPOINT_SYNC_ASYNC, - .maxpacksize = 0x128, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, .rate_max = 48000, @@ -3070,7 +3186,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x85, .ep_attr = USB_ENDPOINT_SYNC_SYNC, - .maxpacksize = 0x128, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, .rate_max = 48000, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9c5ab22358b1..3879eae7e874 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -165,8 +165,10 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; - fp->datainterval = snd_usb_parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + if (fp->datainterval == 0) + fp->datainterval = snd_usb_parse_datainterval(chip, alts); + if (fp->maxpacksize == 0) + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); usb_set_interface(chip->dev, fp->iface, 0); snd_usb_init_pitch(chip, fp->iface, alts, fp); snd_usb_init_sample_rate(chip, fp->iface, alts, fp, fp->rate_max); @@ -446,6 +448,17 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) } /* + * Novation Twitch DJ controller + */ +static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +{ + /* preemptively set up the device because otherwise the + * raw MIDI endpoints are not active */ + usb_set_interface(dev, 0, 1); + return 0; +} + +/* * This call will put the synth in "USB send" mode, i.e it will send MIDI * messages through USB (this is disabled at startup). The synth will * acknowledge by sending a sysex on endpoint 0x85 and by displaying a USB @@ -746,6 +759,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); + case USB_ID(0x1235, 0x0018): + /* Focusrite Novation Twitch */ + return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */ return snd_usb_accessmusic_boot_quirk(dev); @@ -837,6 +854,7 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, break; } snd_emuusb_set_samplerate(subs->stream->chip, emu_samplerate_id); + subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } void snd_usb_set_format_quirk(struct snd_usb_substream *subs, @@ -875,6 +893,16 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->skip_packets = 16; } +void snd_usb_set_interface_quirk(struct usb_device *dev) +{ + /* + * "Playback Design" products need a 50ms delay after setting the + * USB interface. + */ + if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) + mdelay(50); +} + void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) @@ -888,3 +916,31 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, mdelay(20); } +/* + * snd_usb_interface_dsd_format_quirks() is called from format.c to + * augment the PCM format bit-field for DSD types. The UAC standards + * don't have a designated bit field to denote DSD-capable interfaces, + * hence all hardware that is known to support this format has to be + * listed here. + */ +u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int sample_bytes) +{ + /* Playback Designs */ + if (le16_to_cpu(chip->dev->descriptor.idVendor) == 0x23ba) { + switch (fp->altsetting) { + case 1: + fp->dsd_dop = true; + return SNDRV_PCM_FMTBIT_DSD_U16_LE; + case 2: + fp->dsd_bitrev = true; + return SNDRV_PCM_FMTBIT_DSD_U8; + case 3: + fp->dsd_bitrev = true; + return SNDRV_PCM_FMTBIT_DSD_U16_LE; + } + } + + return 0; +} diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 0ca9e91067a6..665e972a1b40 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -26,8 +26,13 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep); +void snd_usb_set_interface_quirk(struct usb_device *dev); void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); +u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int sample_bytes); + #endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ad181d538bd9..7db2f8958e79 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -42,12 +42,11 @@ */ static void free_substream(struct snd_usb_substream *subs) { - struct list_head *p, *n; + struct audioformat *fp, *n; if (!subs->num_formats) return; /* not initialized */ - list_for_each_safe(p, n, &subs->fmt_list) { - struct audioformat *fp = list_entry(p, struct audioformat, list); + list_for_each_entry_safe(fp, n, &subs->fmt_list, list) { kfree(fp->rate_table); kfree(fp->chmap); kfree(fp); @@ -94,6 +93,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->dev = as->chip->dev; subs->txfr_quirk = as->chip->txfr_quirk; subs->speed = snd_usb_get_speed(subs->dev); + subs->pkt_offset_adj = 0; snd_usb_set_pcm_ops(as->pcm, stream); @@ -313,14 +313,12 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, int stream, struct audioformat *fp) { - struct list_head *p; struct snd_usb_stream *as; struct snd_usb_substream *subs; struct snd_pcm *pcm; int err; - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); + list_for_each_entry(as, &chip->pcm_list, list) { if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; @@ -332,8 +330,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, } } /* look for an empty stream */ - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); + list_for_each_entry(as, &chip->pcm_list, list) { if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; @@ -396,6 +393,14 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, if (!csep && altsd->bNumEndpoints >= 2) csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); + /* + * If we can't locate the USB_DT_CS_ENDPOINT descriptor in the extra + * bytes after the first endpoint, go search the entire interface. + * Some devices have it directly *before* the standard endpoint. + */ + if (!csep) + csep = snd_usb_find_desc(alts->extra, alts->extralen, NULL, USB_DT_CS_ENDPOINT); + if (!csep || csep->bLength < 7 || csep->bDescriptorSubtype != UAC_EP_GENERAL) { snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" @@ -463,7 +468,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; int i, altno, err, stream; - int format = 0, num_channels = 0; + unsigned int format = 0, num_channels = 0; struct audioformat *fp = NULL; int num, protocol, clock = 0; struct uac_format_type_i_continuous_descriptor *fmt; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1ac3fd9cc5a6..bc43bcaddf4d 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -56,6 +56,7 @@ struct snd_usb_audio { int setup; /* from the 'device_setup' module param */ int nrpacks; /* from the 'nrpacks' module param */ + bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ }; diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 1e7a47a86605..bf618e1500ac 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -69,7 +69,6 @@ static void init_pipe_urbs(struct usb_stream_kernel *sk, unsigned use_packsize, ++u, transfer += transfer_length) { struct urb *urb = urbs[u]; struct usb_iso_packet_descriptor *desc; - urb->transfer_flags = URB_ISO_ASAP; urb->transfer_buffer = transfer; urb->dev = dev; urb->pipe = pipe; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 520ef96d7c75..b37653247ef4 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -503,7 +503,6 @@ static int usX2Y_urbs_start(struct snd_usX2Y_substream *subs) if (0 == i) atomic_set(&subs->state, state_STARTING3); urb->dev = usX2Y->dev; - urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * pack; urb->iso_frame_desc[pack].length = subs->maxpacksize; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index cc56007791e0..f2a1acdc4d83 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -443,7 +443,6 @@ static int usX2Y_usbpcm_urbs_start(struct snd_usX2Y_substream *subs) if (0 == u) atomic_set(&subs->state, state_STARTING3); urb->dev = usX2Y->dev; - urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * (pack + u * nr_of_packs()); urb->iso_frame_desc[pack].length = subs->maxpacksize; |