From 0052b7dcf9d9ec6be4fc3fe815a2ceda623bb9d1 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 13 Sep 2015 19:00:05 +0900 Subject: ALSA: pcm: remove structure member of 'struct snd_pcm_hwptr_log *' type because this structure had been removed This structure was added by 4d96eb255c53 ('ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions') to store PCM pointers information of latest 10 pointer movements (=XRUN_LOG_CNT). When CONFIG_SND_PCM_XRUN_DEBUG is configured, 'struct snd_pcm_runtime' has 'hwptr_log' member with a pointer to the structure. When calling xrun_log() in pcm_lib.c, the structure was allocated to the pointer. When calling snd_pcm_detach_substream() in pcm.c, the allocated pointer is released. In f5914908a5b7 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints'), the pointer logging is replaced with using Linux Kernel Tracepoints. The structure was also removed, while it's just declared. The member and kfree still remains. This commit removes the member and related codes. I think this was overlooked because it brings no errors/warnings to C compilers. Fixes: f5914908a5b7 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints') Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 6 ------ 1 file changed, 6 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee0a510..a4fcc9456194 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -285,8 +285,6 @@ struct snd_pcm_hw_constraint_ranges { unsigned int mask; }; -struct snd_pcm_hwptr_log; - /* * userspace-provided audio timestamp config to kernel, * structure is for internal use only and filled with dedicated unpack routine @@ -428,10 +426,6 @@ struct snd_pcm_runtime { /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif - -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - struct snd_pcm_hwptr_log *hwptr_log; -#endif }; struct snd_pcm_group { /* keep linked substreams */ -- cgit v1.2.3 From 7486d80f7d853f50088124824bf62d9a4d14de75 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Sep 2015 06:49:29 +0000 Subject: ASoC: rsnd: remove unneeded sh_clk header sh_clk header is not needed, and it will create confusion. Let's remove it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index bb7b2ebfee7b..d8e33d38da43 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -12,7 +12,6 @@ #ifndef RCAR_SND_H #define RCAR_SND_H -#include #define RSND_GEN1_SRU 0 #define RSND_GEN1_ADG 1 -- cgit v1.2.3 From 6131084a0bc966107021d8c89489f9cd1663b902 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 9 Sep 2015 21:27:43 +0300 Subject: ASoC: simple-card: Add tdm slot mask support to simple-card Adds DT binding for explicitly choosing a tdm mask for DAI and uses it in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been changed. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- include/sound/simple_card.h | 2 ++ include/sound/soc.h | 2 ++ 2 files changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index b9b4f289fe6b..0399352f3a62 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -19,6 +19,8 @@ struct asoc_simple_dai { unsigned int sysclk; int slots; int slot_width; + unsigned int tx_slot_mask; + unsigned int rx_slot_mask; struct clk *clk; }; diff --git a/include/sound/soc.h b/include/sound/soc.h index 884e728b09d9..a76622d7bb2f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1601,6 +1601,8 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_tdm_slot(struct device_node *np, + unsigned int *tx_mask, + unsigned int *rx_mask, unsigned int *slots, unsigned int *slot_width); void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, -- cgit v1.2.3 From a54e22f40480eb63533d3f0520e9c84b102dd09f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 16 Sep 2015 13:59:42 +0100 Subject: ASoC: Add SOC_SINGLE_RANGE_EXT_TLV macro Add a version of the SOC_SINGLE_RANGE_TLV macro that allows a custom get and put to be specified. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- include/sound/soc.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 884e728b09d9..9ffa28514fa7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -226,6 +226,18 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } +#define SOC_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ -- cgit v1.2.3 From 917536aeb88d34e06c1353b0dd144f0987bb66bd Mon Sep 17 00:00:00 2001 From: John Lin Date: Mon, 21 Sep 2015 14:12:01 +0800 Subject: ASoC: rt5645: Add jd_invert for Broadwell Broadwell can not triger the IRQ falling and rising simultaneously, so it can not detect jack-in and jack-out simultaneously. We add a flag "jd_invert" to platform data. If this flag is set, codec IRQ will be set to invert that forces IRQ as pulse when jack-in and jack-out. Signed-off-by: John Lin Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- include/sound/rt5645.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index 22734bc3ffd4..a5cf6152e778 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -21,6 +21,8 @@ struct rt5645_platform_data { /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ unsigned int jd_mode; + /* Invert JD when jack insert */ + bool jd_invert; }; #endif -- cgit v1.2.3 From 5334240c30cb0058fa8c373bf0d653337833230d Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Wed, 2 Sep 2015 14:11:38 +0800 Subject: drm/i915: Add audio sync_audio_rate callback Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang Reviewed-by: Jani Nikula Signed-off-by: Takashi Iwai --- include/drm/i915_component.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index b2d56dd483d9..e6d35d7239c0 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,13 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; const struct i915_audio_component_audio_ops { -- cgit v1.2.3 From 7e8275c2f2bbb384e18af37066b8b2f32b7d092f Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 25 Sep 2015 09:36:12 +0800 Subject: drm/i915: set proper N/CTS in modeset When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang Reviewed-by: Jani Nikula Signed-off-by: Takashi Iwai --- include/drm/i915_component.h | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'include') diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7239c0..89dc7d6bc1cc 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,18 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- cgit v1.2.3 From b7631a12e726597cffde1b29cc3bcf811981c1fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Sep 2015 12:19:08 +0200 Subject: ALSA: hda - Fix typos in snd_hdac_regmap_*() documents Fixes the wrong reference names to regmap amp functions. Signed-off-by: Takashi Iwai --- include/sound/hda_regmap.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index df705908480a..2767c55a641e 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -67,7 +67,7 @@ int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, * @reg: verb to write * @val: value to write * - * For writing an amp value, use snd_hda_regmap_amp_update(). + * For writing an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, @@ -85,7 +85,7 @@ snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, * @mask: bit mask to update * @val: value to update * - * For updating an amp value, use snd_hda_regmap_amp_update(). + * For updating an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid, -- cgit v1.2.3 From 660dd3d52ead45b8e60dcf966daf304de2121a28 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Sep 2015 09:39:21 +0900 Subject: ALSA: firewire-digi00x: add hwdep interface This commit adds hwdep interface so as the other sound drivers for units on IEEE 1394 bus have. This interface is designed for mixer/control applications. By using this interface, an application can get information about firewire node, can lock/unlock kernel streaming and can get notification at starting/stopping kernel streaming. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 ++- include/uapi/sound/firewire.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index a45be6bdcf5b..aa329132f6c4 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -100,9 +100,10 @@ enum { SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */ SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */ + SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_OXFW + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_DIGI00X }; struct snd_hwdep_info { diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index 49122df3b56b..f67d228f731b 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -56,6 +56,7 @@ union snd_firewire_event { #define SNDRV_FIREWIRE_TYPE_FIREWORKS 2 #define SNDRV_FIREWIRE_TYPE_BEBOB 3 #define SNDRV_FIREWIRE_TYPE_OXFW 4 +#define SNDRV_FIREWIRE_TYPE_DIGI00X 5 /* RME, MOTU, ... */ struct snd_firewire_get_info { -- cgit v1.2.3 From 44b7308871ac6fd85fc840bfa3ddb466fe7aff23 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Sep 2015 09:39:22 +0900 Subject: ALSA: firewire-digi00x: add support for asynchronous messaging Digi 002/003 family uses asynchronous transaction for messaging. The address to transmit this message is stored on a certain register. This commit allocates a range of address on OHCI 1394 host controller to handle the messaging. As long as I know, the purpose of this message seems to notify lost of synchronization. While, the meaning of content of the message is not clear. Actual examples of this messaging: * When clock source is set as internal: - 0x00007051 - 0x00007052 - 0x00007054 - 0x00007057 - 0x00007058 * When clock source is set as somewhat external: - 0x00009000 - 0x00009010 - 0x00009020 - 0x00009021 - 0x00009022 The lost often occurs when using internal clock source. In this case, users hear sounds with quite short gap every several minutes. In fact, the lost is recovered temporarily. When using with external clock source, the lost seems not to occur. The mechanism is not clear yet. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/uapi/sound/firewire.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index f67d228f731b..deb041cb9af0 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -9,6 +9,7 @@ #define SNDRV_FIREWIRE_EVENT_LOCK_STATUS 0x000010cc #define SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION 0xd1ce004e #define SNDRV_FIREWIRE_EVENT_EFW_RESPONSE 0x4e617475 +#define SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE 0x746e736c struct snd_firewire_event_common { unsigned int type; /* SNDRV_FIREWIRE_EVENT_xxx */ @@ -40,11 +41,17 @@ struct snd_firewire_event_efw_response { __be32 response[0]; /* some responses */ }; +struct snd_firewire_event_digi00x_message { + unsigned int type; + __u32 message; /* Digi00x-specific message */ +}; + union snd_firewire_event { struct snd_firewire_event_common common; struct snd_firewire_event_lock_status lock_status; struct snd_firewire_event_dice_notification dice_notification; struct snd_firewire_event_efw_response efw_response; + struct snd_firewire_event_digi00x_message digi00x_message; }; -- cgit v1.2.3 From c25c79b468a61ad8a54f764553056e2e2a427ea8 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 29 Sep 2015 16:43:56 +0100 Subject: ASoC: Add SOC_DOUBLE_R_EXT _EXT version of SOC_DOUBLE_R required to allow for custom handlers. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/soc.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 884e728b09d9..2d79cb55aec5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -217,6 +217,13 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = \ SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert, 0) } +#define SOC_DOUBLE_R_EXT(xname, reg_left, reg_right, xshift, xmax, xinvert,\ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ + xmax, xinvert) } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -- cgit v1.2.3 From 58ceb57ec1be928bec2faeca11fe0752f930669d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 30 Sep 2015 22:51:52 +0200 Subject: ASoC: pxa: pxa-pcm-lib: switch over to snd-soc-dmaengine-pcm This patch removes the old PXA DMA API usage and switches over to generic functions provided by snd-soc-dmaengine-pcm. More cleanups may be done on top of this, and some function stubs can now be removed completetly. However, the intention here was to keep the transition as small as possible. This was tested on the mioa701 pxa27x board. Signed-off-by: Daniel Mack [trivial change from mmp-dma to pxa-dma] Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- include/sound/pxa2xx-lib.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 56e818e4a1cb..6ef629bde164 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -12,7 +12,6 @@ extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd); extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream); extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); -extern void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id); extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream); extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, -- cgit v1.2.3 From e5e0c3dd257bf34cf001e10422943f90437f0f1b Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 1 Oct 2015 22:02:17 +0900 Subject: ALSA: firewire-tascam: add hwdep interface This commit adds hwdep interface so as the other IEEE 1394 sound devices has. This interface is designed for mixer/control applications. By using this interface, an application can get information about firewire node, can lock/unlock kernel streaming and can get notification at starting/stopping kernel streaming. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 ++- include/uapi/sound/firewire.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index aa329132f6c4..a82108e5d1c0 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -101,9 +101,10 @@ enum { SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */ SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ + SNDRV_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_DIGI00X + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_TASCAM }; struct snd_hwdep_info { diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index deb041cb9af0..db79a12fcc78 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -64,6 +64,7 @@ union snd_firewire_event { #define SNDRV_FIREWIRE_TYPE_BEBOB 3 #define SNDRV_FIREWIRE_TYPE_OXFW 4 #define SNDRV_FIREWIRE_TYPE_DIGI00X 5 +#define SNDRV_FIREWIRE_TYPE_TASCAM 6 /* RME, MOTU, ... */ struct snd_firewire_get_info { -- cgit v1.2.3 From 5b2688a59af686f7c0a80edc49d7f190365ac090 Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:28:47 +0800 Subject: ASoC: topology: ABI - Add PCM Support and bump ABI version to 4 The struct snd_soc_tplg_pcm_dai is renamed to snd_soc_tplg_pcm. This struct will now be used to handle data related to PCMs (FE DAI & DAI links). It's not for BE, because BE DAI mappings will be provided by ACPI/FDT data. Remove the unused struct snd_soc_tplg_pcm_cfg_caps. We are using snd_soc_tplg_stream and snd_soc_stream_caps instead. Bump ABI version to 4. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 31 +++++++++++++++---------------- 1 file changed, 15 insertions(+), 16 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 247c50bd60f0..2aa081ca95c1 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -83,7 +83,7 @@ #define SND_SOC_TPLG_NUM_TEXTS 16 /* ABI version */ -#define SND_SOC_TPLG_ABI_VERSION 0x3 +#define SND_SOC_TPLG_ABI_VERSION 0x4 /* Max size of TLV data */ #define SND_SOC_TPLG_TLV_SIZE 32 @@ -378,30 +378,29 @@ struct snd_soc_tplg_dapm_widget { */ } __attribute__((packed)); -struct snd_soc_tplg_pcm_cfg_caps { - struct snd_soc_tplg_stream_caps caps; - struct snd_soc_tplg_stream_config configs[SND_SOC_TPLG_STREAM_CONFIG_MAX]; - __le32 num_configs; /* number of configs */ -} __attribute__((packed)); /* - * Describes SW/FW specific features of PCM or DAI link. + * Describes SW/FW specific features of PCM (FE DAI & DAI link). * - * File block representation for PCM/DAI-Link :- + * File block representation for PCM :- * +-----------------------------------+-----+ * | struct snd_soc_tplg_hdr | 1 | * +-----------------------------------+-----+ - * | struct snd_soc_tplg_dapm_pcm_dai | N | + * | struct snd_soc_tplg_pcm | N | * +-----------------------------------+-----+ */ -struct snd_soc_tplg_pcm_dai { +struct snd_soc_tplg_pcm { __le32 size; /* in bytes of this structure */ - char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - __le32 id; /* unique ID - used to match */ - __le32 playback; /* supports playback mode */ - __le32 capture; /* supports capture mode */ - __le32 compress; /* 1 = compressed; 0 = PCM */ - struct snd_soc_tplg_pcm_cfg_caps capconf[2]; /* capabilities and configs */ + char pcm_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 pcm_id; /* unique ID - used to match */ + __le32 dai_id; /* unique ID - used to match */ + __le32 playback; /* supports playback mode */ + __le32 capture; /* supports capture mode */ + __le32 compress; /* 1 = compressed; 0 = PCM */ + struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */ + __le32 num_streams; /* number of streams */ + struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */ } __attribute__((packed)); #endif -- cgit v1.2.3 From 7654855ef84d78079109bb38195a8db6b188117b Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:29:47 +0800 Subject: ASoC: topology: ABI - Remove unused struct snd_soc_tplg_stream_config The struct snd_soc_tplg_stream_config is no longer used in the ABI. We are using snd_soc_tplg_stream instead. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 2aa081ca95c1..4bef63f5e1ff 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -226,16 +226,6 @@ struct snd_soc_tplg_stream { __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ } __attribute__((packed)); -/* - * Duplex stream configuration supported by SW/FW. - */ -struct snd_soc_tplg_stream_config { - __le32 size; /* in bytes of this structure */ - char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - struct snd_soc_tplg_stream playback; - struct snd_soc_tplg_stream capture; -} __attribute__((packed)); - /* * Manifest. List totals for each payload type. Not used in parsing, but will * be passed to the component driver before any other objects in order for any -- cgit v1.2.3 From e8b3fe8f383bbf055cbd69b776fcbbb5ed3befcd Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:30:05 +0800 Subject: ASoC: topology: ABI - Use __le32 instead of __u32 in snd_soc_tplg_dapm_widget This fixes the endianness of the ABI parameters in the struct. The field 'num_kcontrols' is also extended from 16 bits to 32 bits. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 4bef63f5e1ff..88210a8e450f 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -356,11 +356,11 @@ struct snd_soc_tplg_dapm_widget { __le32 shift; /* bits to shift */ __le32 mask; /* non-shifted mask */ __le32 subseq; /* sort within widget type */ - __u32 invert; /* invert the power bit */ - __u32 ignore_suspend; /* kept enabled over suspend */ - __u16 event_flags; - __u16 event_type; - __u16 num_kcontrols; + __le32 invert; /* invert the power bit */ + __le32 ignore_suspend; /* kept enabled over suspend */ + __le16 event_flags; + __le16 event_type; + __le32 num_kcontrols; struct snd_soc_tplg_private priv; /* * kcontrols that relate to this widget -- cgit v1.2.3 From 731324f5cee3caf230427f754701212961fe0bb1 Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:31:34 +0800 Subject: ASoC: topology: ABI - Add name element to snd_soc_tplg_stream For codec-codec links, this struct will be mapped to the DAI links's params, which is struct snd_soc_pcm_stream and it needs a stream name. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 88210a8e450f..218148000187 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -217,6 +217,7 @@ struct snd_soc_tplg_stream_caps { */ struct snd_soc_tplg_stream { __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* Name of the stream */ __le64 format; /* SNDRV_PCM_FMTBIT_* */ __le32 rate; /* SNDRV_PCM_RATE_* */ __le32 period_bytes; /* size of period in bytes */ -- cgit v1.2.3 From 478d0d39719127c42f21ac85b8719a87b8fa7bd7 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 30 Sep 2015 17:31:44 +0800 Subject: ASoC: topology: ABI - Change stream formats to a bitwise flag The toplogy user space tool will generate this bitwise flag by using SNDRV_PCM_FORMAT_* exposed by asound.h, and the topology core will copy this flag when generating DAI streams. Signed-off-by: Mengdong Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 218148000187..eeb74a9cc2f8 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -198,7 +198,7 @@ struct snd_soc_tplg_ctl_hdr { struct snd_soc_tplg_stream_caps { __le32 size; /* in bytes of this structure */ char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le64 formats; /* supported formats SNDRV_PCM_FMTBIT_* */ __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ __le32 rate_min; /* min rate */ __le32 rate_max; /* max rate */ -- cgit v1.2.3 From 7c545b327d54cdc3f693093f744459f6e0d8ce58 Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:32:04 +0800 Subject: ASoC: topology: ABI - Add the type for BE DAI link Define the topology type for BE DAI link: SND_SOC_TPLG_TYPE_BACKEND_LINK. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index eeb74a9cc2f8..f75fd29bf580 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -103,7 +103,8 @@ #define SND_SOC_TPLG_TYPE_PCM 7 #define SND_SOC_TPLG_TYPE_MANIFEST 8 #define SND_SOC_TPLG_TYPE_CODEC_LINK 9 -#define SND_SOC_TPLG_TYPE_PDATA 10 +#define SND_SOC_TPLG_TYPE_BACKEND_LINK 10 +#define SND_SOC_TPLG_TYPE_PDATA 11 #define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_PDATA /* vendor block IDs - please add new vendor types to end */ -- cgit v1.2.3 From e2a9df656f28e42538562d090e2b6f3dec5b41f2 Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:32:22 +0800 Subject: ASoC: topology: ABI - Add configuration for BE & Codec-Codec DAI Links struct snd_soc_tplg_link_config is defined to configure BE & CC links. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index f75fd29bf580..538ea1308008 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -395,4 +395,21 @@ struct snd_soc_tplg_pcm { struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */ } __attribute__((packed)); + +/* + * Describes the BE or CC link runtime supported configs or params + * + * File block representation for BE/CC link config :- + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_link_config | N | + * +-----------------------------------+-----+ + */ +struct snd_soc_tplg_link_config { + __le32 size; /* in bytes of this structure */ + __le32 id; /* unique ID - used to match */ + struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */ + __le32 num_streams; /* number of streams */ +} __attribute__((packed)); #endif -- cgit v1.2.3 From 76a822a6ae9c0c67309318bb60a4117329252fc4 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 30 Sep 2015 17:32:40 +0800 Subject: ASoC: topology: ABI - Remove tdm_slot & dai_fmt from snd_soc_tplg_stream These two fields are line parameters for BE/CC links and should not be from toplogy but from ACPI. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 538ea1308008..26539a7e4880 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -224,8 +224,6 @@ struct snd_soc_tplg_stream { __le32 period_bytes; /* size of period in bytes */ __le32 buffer_bytes; /* size of buffer in bytes */ __le32 channels; /* channels */ - __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */ - __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ } __attribute__((packed)); /* -- cgit v1.2.3 From 6d817c0e9fd7536be76690bfdee88e8a81c16f7d Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 29 Sep 2015 16:44:01 +0100 Subject: ASoC: codecs: Add da7219 codec driver This adds support for the DA7219 audio codec with built-in advanced accessory detect features. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/da7219-aad.h | 99 ++++++++++++++++++++++++++++++++++++++++++++++ include/sound/da7219.h | 55 ++++++++++++++++++++++++++ 2 files changed, 154 insertions(+) create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h (limited to 'include') diff --git a/include/sound/da7219-aad.h b/include/sound/da7219-aad.h new file mode 100644 index 000000000000..17802fb86ec4 --- /dev/null +++ b/include/sound/da7219-aad.h @@ -0,0 +1,99 @@ +/* + * da7219-aad.h - DA7322 ASoC Codec AAD Driver Platform Data + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson + * + * 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. + */ + +#ifndef __DA7219_AAD_PDATA_H +#define __DA7219_AAD_PDATA_H + +enum da7219_aad_micbias_pulse_lvl { + DA7219_AAD_MICBIAS_PULSE_LVL_OFF = 0, + DA7219_AAD_MICBIAS_PULSE_LVL_2_8V = 6, + DA7219_AAD_MICBIAS_PULSE_LVL_2_9V, +}; + +enum da7219_aad_btn_cfg { + DA7219_AAD_BTN_CFG_2MS = 1, + DA7219_AAD_BTN_CFG_5MS, + DA7219_AAD_BTN_CFG_10MS, + DA7219_AAD_BTN_CFG_50MS, + DA7219_AAD_BTN_CFG_100MS, + DA7219_AAD_BTN_CFG_200MS, + DA7219_AAD_BTN_CFG_500MS, +}; + +enum da7219_aad_mic_det_thr { + DA7219_AAD_MIC_DET_THR_200_OHMS = 0, + DA7219_AAD_MIC_DET_THR_500_OHMS, + DA7219_AAD_MIC_DET_THR_750_OHMS, + DA7219_AAD_MIC_DET_THR_1000_OHMS, +}; + +enum da7219_aad_jack_ins_deb { + DA7219_AAD_JACK_INS_DEB_5MS = 0, + DA7219_AAD_JACK_INS_DEB_10MS, + DA7219_AAD_JACK_INS_DEB_20MS, + DA7219_AAD_JACK_INS_DEB_50MS, + DA7219_AAD_JACK_INS_DEB_100MS, + DA7219_AAD_JACK_INS_DEB_200MS, + DA7219_AAD_JACK_INS_DEB_500MS, + DA7219_AAD_JACK_INS_DEB_1S, +}; + +enum da7219_aad_jack_det_rate { + DA7219_AAD_JACK_DET_RATE_32_64MS = 0, + DA7219_AAD_JACK_DET_RATE_64_128MS, + DA7219_AAD_JACK_DET_RATE_128_256MS, + DA7219_AAD_JACK_DET_RATE_256_512MS, +}; + +enum da7219_aad_jack_rem_deb { + DA7219_AAD_JACK_REM_DEB_1MS = 0, + DA7219_AAD_JACK_REM_DEB_5MS, + DA7219_AAD_JACK_REM_DEB_10MS, + DA7219_AAD_JACK_REM_DEB_20MS, +}; + +enum da7219_aad_btn_avg { + DA7219_AAD_BTN_AVG_1 = 0, + DA7219_AAD_BTN_AVG_2, + DA7219_AAD_BTN_AVG_4, + DA7219_AAD_BTN_AVG_8, +}; + +enum da7219_aad_adc_1bit_rpt { + DA7219_AAD_ADC_1BIT_RPT_1 = 0, + DA7219_AAD_ADC_1BIT_RPT_2, + DA7219_AAD_ADC_1BIT_RPT_4, + DA7219_AAD_ADC_1BIT_RPT_8, +}; + +struct da7219_aad_pdata { + int irq; + + enum da7219_aad_micbias_pulse_lvl micbias_pulse_lvl; + u32 micbias_pulse_time; + enum da7219_aad_btn_cfg btn_cfg; + enum da7219_aad_mic_det_thr mic_det_thr; + enum da7219_aad_jack_ins_deb jack_ins_deb; + enum da7219_aad_jack_det_rate jack_det_rate; + enum da7219_aad_jack_rem_deb jack_rem_deb; + + u8 a_d_btn_thr; + u8 d_b_btn_thr; + u8 b_c_btn_thr; + u8 c_mic_btn_thr; + + enum da7219_aad_btn_avg btn_avg; + enum da7219_aad_adc_1bit_rpt adc_1bit_rpt; +}; + +#endif /* __DA7219_AAD_PDATA_H */ diff --git a/include/sound/da7219.h b/include/sound/da7219.h new file mode 100644 index 000000000000..3f39e135312d --- /dev/null +++ b/include/sound/da7219.h @@ -0,0 +1,55 @@ +/* + * da7219.h - DA7219 ASoC Codec Driver Platform Data + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson + * + * 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. + */ + +#ifndef __DA7219_PDATA_H +#define __DA7219_PDATA_H + +/* LDO */ +enum da7219_ldo_lvl_sel { + DA7219_LDO_LVL_SEL_1_05V = 0, + DA7219_LDO_LVL_SEL_1_10V, + DA7219_LDO_LVL_SEL_1_20V, + DA7219_LDO_LVL_SEL_1_40V, +}; + +/* Mic Bias */ +enum da7219_micbias_voltage { + DA7219_MICBIAS_1_8V = 1, + DA7219_MICBIAS_2_0V, + DA7219_MICBIAS_2_2V, + DA7219_MICBIAS_2_4V, + DA7219_MICBIAS_2_6V, +}; + +/* Mic input type */ +enum da7219_mic_amp_in_sel { + DA7219_MIC_AMP_IN_SEL_DIFF = 0, + DA7219_MIC_AMP_IN_SEL_SE_P, + DA7219_MIC_AMP_IN_SEL_SE_N, +}; + +struct da7219_aad_pdata; + +struct da7219_pdata { + /* Internal LDO */ + enum da7219_ldo_lvl_sel ldo_lvl_sel; + + /* Mic */ + enum da7219_micbias_voltage micbias_lvl; + enum da7219_mic_amp_in_sel mic_amp_in_sel; + + /* AAD */ + struct da7219_aad_pdata *aad_pdata; +}; + +#endif /* __DA7219_PDATA_H */ -- cgit v1.2.3 From 1d957d862ac782eaf5803d4d4cf167708e4dc147 Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu Date: Fri, 25 Sep 2015 17:48:22 -0400 Subject: ASoC: dwc: support dw i2s in slave mode dw i2s controller can work in slave mode, codec being master. dw i2s is made to support master/slave operation, by reading dwc register. Signed-off-by: Maruthi Bayyavarapu Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- include/sound/designware_i2s.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 3a8fca9409a7..8966ba7c9629 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -38,6 +38,8 @@ struct i2s_clk_config_data { struct i2s_platform_data { #define DWC_I2S_PLAY (1 << 0) #define DWC_I2S_RECORD (1 << 1) + #define DW_I2S_SLAVE (1 << 2) + #define DW_I2S_MASTER (1 << 3) unsigned int cap; int channel; u32 snd_fmts; -- cgit v1.2.3 From dcc448e619194098b24477a6d56af50c57f26f1d Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Wed, 7 Oct 2015 12:42:22 +0200 Subject: ASoC: rsnd: Remove obsolete platform data support Since commit 3d7608e4c169af03 ("ARM: shmobile: bockw: remove legacy board file and config"), Renesas R-Car SoCs are only supported in generic DT-only ARM multi-platform builds. The driver doesn't need to use platform data anymore, hence remove platform data configuration. Move to sound/soc/sh/rcar/, as it's no longer needed by platform code. Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 117 ----------------------------------------------- 1 file changed, 117 deletions(-) delete mode 100644 include/sound/rcar_snd.h (limited to 'include') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h deleted file mode 100644 index d8e33d38da43..000000000000 --- a/include/sound/rcar_snd.h +++ /dev/null @@ -1,117 +0,0 @@ -/* - * Renesas R-Car SRU/SCU/SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * 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 RCAR_SND_H -#define RCAR_SND_H - - -#define RSND_GEN1_SRU 0 -#define RSND_GEN1_ADG 1 -#define RSND_GEN1_SSI 2 - -#define RSND_GEN2_SCU 0 -#define RSND_GEN2_ADG 1 -#define RSND_GEN2_SSIU 2 -#define RSND_GEN2_SSI 3 - -#define RSND_BASE_MAX 4 - -/* - * flags - * - * 0xAB000000 - * - * A : clock sharing settings - * B : SSI direction - */ -#define RSND_SSI_CLK_PIN_SHARE (1 << 31) -#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */ - -#define RSND_SSI(_dma_id, _irq, _flags) \ -{ .dma_id = _dma_id, .irq = _irq, .flags = _flags } -#define RSND_SSI_UNUSED \ -{ .dma_id = -1, .irq = -1, .flags = 0 } - -struct rsnd_ssi_platform_info { - int dma_id; - int irq; - u32 flags; -}; - -#define RSND_SRC(rate, _dma_id) \ -{ .convert_rate = rate, .dma_id = _dma_id, } -#define RSND_SRC_UNUSED \ -{ .convert_rate = 0, .dma_id = -1, } - -struct rsnd_src_platform_info { - u32 convert_rate; /* sampling rate convert */ - int dma_id; /* for Gen2 SCU */ - int irq; -}; - -/* - * flags - */ -struct rsnd_ctu_platform_info { - u32 flags; -}; - -struct rsnd_mix_platform_info { - u32 flags; -}; - -struct rsnd_dvc_platform_info { - u32 flags; -}; - -struct rsnd_dai_path_info { - struct rsnd_ssi_platform_info *ssi; - struct rsnd_src_platform_info *src; - struct rsnd_ctu_platform_info *ctu; - struct rsnd_mix_platform_info *mix; - struct rsnd_dvc_platform_info *dvc; -}; - -struct rsnd_dai_platform_info { - struct rsnd_dai_path_info playback; - struct rsnd_dai_path_info capture; -}; - -/* - * flags - * - * 0x0000000A - * - * A : generation - */ -#define RSND_GEN_MASK (0xF << 0) -#define RSND_GEN1 (1 << 0) /* fixme */ -#define RSND_GEN2 (2 << 0) /* fixme */ - -struct rcar_snd_info { - u32 flags; - struct rsnd_ssi_platform_info *ssi_info; - int ssi_info_nr; - struct rsnd_src_platform_info *src_info; - int src_info_nr; - struct rsnd_ctu_platform_info *ctu_info; - int ctu_info_nr; - struct rsnd_mix_platform_info *mix_info; - int mix_info_nr; - struct rsnd_dvc_platform_info *dvc_info; - int dvc_info_nr; - struct rsnd_dai_platform_info *dai_info; - int dai_info_nr; - int (*start)(int id); - int (*stop)(int id); -}; - -#endif -- cgit v1.2.3 From 6e7c444318699496e6e6f30c875cf67534aeccc6 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Wed, 7 Oct 2015 14:27:11 +0100 Subject: ASoC: da7213: Add support to handle mclk data provided to driver Driver now can make use of mclk data, if provided, to set, enable and disable the clock source. As part of this, the choice to enable clock squaring is dealt with as part of dai_sysclk() call rather than as platform data. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/da7213.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'include') diff --git a/include/sound/da7213.h b/include/sound/da7213.h index 673f5c39cbf2..e7eac8979995 100644 --- a/include/sound/da7213.h +++ b/include/sound/da7213.h @@ -44,9 +44,6 @@ struct da7213_platform_data { enum da7213_dmic_data_sel dmic_data_sel; enum da7213_dmic_samplephase dmic_samplephase; enum da7213_dmic_clk_rate dmic_clk_rate; - - /* MCLK squaring config */ - bool mclk_squaring; }; #endif /* _DA7213_PDATA_H */ -- cgit v1.2.3 From 1b5e6167c27e1d3be33155baf9660768ac74aae0 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 8 Oct 2015 09:48:05 +0100 Subject: ALSA: hdac: Copy codec helpers to core The current codec helpers are local to hda code and needs to be moved to core so that other users can use it. The helpers to read/write the codec and to check the power state of widgets is copied Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 49bc836fcd84..26e956f4b7c6 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -147,6 +147,12 @@ int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid, unsigned int format); +int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm); +int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm); +bool snd_hdac_check_power_state(struct hdac_device *hdac, + hda_nid_t nid, unsigned int target_state); /** * snd_hdac_read_parm - read a codec parameter * @codec: the codec object -- cgit v1.2.3 From a82d24f83de2c63199acead488259fcdf947e90e Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Thu, 15 Oct 2015 07:55:55 +0200 Subject: ALSA: emu10k1: added EMU10K1 version of DECLARE_BITMAP macro for UAPI MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes userspace compilation error: error: expected specifier-qualifier-list before ‘DECLARE_BITMAP’ DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ DECLARE_BITMAP macro is not meant for userspace headers and thus added here as private copy for emu10k.h. Fix was suggested by Arnd Bergmann in message <2168807.4Yxh5gl11Q@wuerfel> and Takashi Iwai in message on lkml. Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/emu10k1.h | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index ec1535bb6aed..5175e166987d 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -34,6 +34,14 @@ #define EMU10K1_FX8010_PCM_COUNT 8 +/* + * Following definition is copied from linux/types.h to support compiling + * this header file in userspace since they are not generally available for + * uapi headers. + */ +#define __EMU10K1_DECLARE_BITMAP(name,bits) \ + unsigned long name[(bits) / (sizeof(unsigned long) * 8)] + /* instruction set */ #define iMAC0 0x00 /* R = A + (X * Y >> 31) ; saturation */ #define iMAC1 0x01 /* R = A + (-X * Y >> 31) ; saturation */ @@ -300,7 +308,7 @@ struct snd_emu10k1_fx8010_control_old_gpr { struct snd_emu10k1_fx8010_code { char name[128]; - DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ __u32 __user *gpr_map; /* initializers */ unsigned int gpr_add_control_count; /* count of GPR controls to add/replace */ @@ -313,11 +321,11 @@ struct snd_emu10k1_fx8010_code { unsigned int gpr_list_control_total; /* total count of GPR controls */ struct snd_emu10k1_fx8010_control_gpr __user *gpr_list_controls; /* listed GPR controls */ - DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ __u32 __user *tram_data_map; /* data initializers */ __u32 __user *tram_addr_map; /* map initializers */ - DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ + __EMU10K1_DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ __u32 __user *code; /* one instruction - 64 bits */ }; -- cgit v1.2.3 From ffc287c89169705d9a01d48e05453ab0eda631e4 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Thu, 15 Oct 2015 07:56:06 +0200 Subject: ALSA: hdspm: use __u8, __u32 and __u64 from linux/types.h instead of stdint.h Kernel headers should use linux/types.h based definitions. Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/hdspm.h | 40 ++++++++++++++++++---------------------- 1 file changed, 18 insertions(+), 22 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index 5737332d38f2..c4db6f5b306e 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,11 +20,7 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ -#ifdef __KERNEL__ #include -#else -#include -#endif /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 @@ -46,15 +42,15 @@ enum hdspm_speed { /* -------------------- IOCTL Peak/RMS Meters -------------------- */ struct hdspm_peak_rms { - uint32_t input_peaks[64]; - uint32_t playback_peaks[64]; - uint32_t output_peaks[64]; + __u32 input_peaks[64]; + __u32 playback_peaks[64]; + __u32 output_peaks[64]; - uint64_t input_rms[64]; - uint64_t playback_rms[64]; - uint64_t output_rms[64]; + __u64 input_rms[64]; + __u64 playback_rms[64]; + __u64 output_rms[64]; - uint8_t speed; /* enum {ss, ds, qs} */ + __u8 speed; /* enum {ss, ds, qs} */ int status2; }; @@ -155,21 +151,21 @@ enum hdspm_syncsource { }; struct hdspm_status { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ enum hdspm_syncsource autosync_source; - uint64_t card_clock; - uint32_t master_period; + __u64 card_clock; + __u32 master_period; union { struct { - uint8_t sync_wc; /* enum hdspm_sync */ - uint8_t sync_madi; /* enum hdspm_sync */ - uint8_t sync_tco; /* enum hdspm_sync */ - uint8_t sync_in; /* enum hdspm_sync */ - uint8_t madi_input; /* enum hdspm_madi_input */ - uint8_t channel_format; /* enum hdspm_madi_channel_format */ - uint8_t frame_format; /* enum hdspm_madi_frame_format */ + __u8 sync_wc; /* enum hdspm_sync */ + __u8 sync_madi; /* enum hdspm_sync */ + __u8 sync_tco; /* enum hdspm_sync */ + __u8 sync_in; /* enum hdspm_sync */ + __u8 madi_input; /* enum hdspm_madi_input */ + __u8 channel_format; /* enum hdspm_madi_channel_format */ + __u8 frame_format; /* enum hdspm_madi_frame_format */ } madi; } card_specific; }; @@ -184,7 +180,7 @@ struct hdspm_status { #define HDSPM_ADDON_TCO 1 struct hdspm_version { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ char cardname[20]; unsigned int serial; unsigned short firmware_rev; -- cgit v1.2.3 From ded255be2276d365a91af2de7c7f8e2c233d4fa2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Oct 2015 17:59:43 +0200 Subject: ALSA: hda - consolidate chip rename functions A few multiple codec drivers do renaming the chip_name string but all these are open-coded and some of them have even no error check. Let's make common helpers to do it properly. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 26e956f4b7c6..49df61c7afdc 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -117,6 +117,7 @@ int snd_hdac_device_init(struct hdac_device *dev, struct hdac_bus *bus, void snd_hdac_device_exit(struct hdac_device *dev); int snd_hdac_device_register(struct hdac_device *codec); void snd_hdac_device_unregister(struct hdac_device *codec); +int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name); int snd_hdac_refresh_widgets(struct hdac_device *codec); int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec); -- cgit v1.2.3 From 90bbaf66ee7b946952f1e82a0069639dea5fd893 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Fri, 16 Oct 2015 17:57:46 +0800 Subject: ALSA: timer: add config item to export PCM timer disabling for expert PCM timer is not always used. For embedded device, we need an interface to disable it when it is not needed, to shrink the kernel size and memory footprint, here add CONFIG_SND_PCM_TIMER for it. When both CONFIG_SND_PCM_TIMER and CONFIG_SND_TIMER is unselected, about 25KB saving bonus we can get. Please be noted that when disabled, those stubs who using pcm timer (e.g. dmix, dsnoop & co) may work incorrectlly. Suggested-by: Takashi Iwai Signed-off-by: Jie Yang Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index a4fcc9456194..2882dddfc91c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1111,10 +1111,16 @@ static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substrea * Timer interface */ +#ifdef CONFIG_SND_PCM_TIMER void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream); void snd_pcm_timer_init(struct snd_pcm_substream *substream); void snd_pcm_timer_done(struct snd_pcm_substream *substream); - +#else +static inline void +snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} +#endif /** * snd_pcm_gettime - Fill the timespec depending on the timestamp mode * @runtime: PCM runtime instance -- cgit v1.2.3 From 93ed8560e98afc486df94f5a6238c1f0894b38b8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Oct 2015 21:35:53 +0200 Subject: ALSA: hda - Add api_version to hda_device_id struct For distinguishing the difference between HDA legacy and ext codec driver entries, we need to expose the value corresponding to type field. This patch adds a new field, api_version, to hda_device_id struct, so that this information is embedded in modalias string. Although the information is basically redundant (struct hdac_device already has type field), the helper that extracts from MODULE_DEVICE_TABLE() won't take it account except for the exported table entries themselves. So we need to put the same information in the table, too. Reviewed-by: Vinod Koul Tested-by: Subhransu S Prusty Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 49df61c7afdc..b35bf59a1ecc 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -33,6 +33,7 @@ extern struct bus_type snd_hda_bus_type; struct hda_device_id { __u32 vendor_id; __u32 rev_id; + __u8 api_version; const char *name; unsigned long driver_data; }; -- cgit v1.2.3 From da23ac1e40ce844d1a9553906bdacce160af76f6 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 29 Sep 2015 13:56:10 +0530 Subject: ALSA: hda - Add hduadio support to DEVTABLE For generating modalias entries automatically, move the definition of struct hda_device_id to linux/mod_devicetable.h and add the handling of this record in file2alias helper. The new modalias is represented with combination of vendor id, device id, and api version as "hdaudio:vNrNaN". This patch itself doesn't convert the existing modaliases. Since they were added manually, this patch won't give any regression by itself at this point. [Modified the modalias format to adapt the api_version field, and drop invalid ANY_ID definition by tiwai] Signed-off-by: Subhransu S. Prusty Reviewed-by: Vinod Koul Tested-by: Subhransu S Prusty Signed-off-by: Takashi Iwai --- include/linux/mod_devicetable.h | 8 ++++++++ include/sound/hdaudio.h | 12 +----------- 2 files changed, 9 insertions(+), 11 deletions(-) (limited to 'include') diff --git a/include/linux/mod_devicetable.h b/include/linux/mod_devicetable.h index 688997a24aad..00825672d256 100644 --- a/include/linux/mod_devicetable.h +++ b/include/linux/mod_devicetable.h @@ -219,6 +219,14 @@ struct serio_device_id { __u8 proto; }; +struct hda_device_id { + __u32 vendor_id; + __u32 rev_id; + __u8 api_version; + const char *name; + unsigned long driver_data; +}; + /* * Struct used for matching a device */ diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b35bf59a1ecc..ddca48eb02e0 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -21,23 +21,13 @@ struct hdac_stream; struct hdac_device; struct hdac_driver; struct hdac_widget_tree; +struct hda_device_id; /* * exported bus type */ extern struct bus_type snd_hda_bus_type; -/* - * HDA device table - */ -struct hda_device_id { - __u32 vendor_id; - __u32 rev_id; - __u8 api_version; - const char *name; - unsigned long driver_data; -}; - /* * generic arrays */ -- cgit v1.2.3 From 4f9e0c38c5e991e2d050d13e28be74b93ab704c0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 Oct 2015 11:35:49 +0200 Subject: ALSA: hda - Add a common helper to give the codec modalias string This patch provide a new common helper function, snd_hdac_codec_modalias(), to give the codec modalias name string. This function will be used by multiple places in the later patches. Reviewed-by: Vinod Koul Tested-by: Subhransu S Prusty Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index ddca48eb02e0..e2b712c90d3f 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -109,6 +109,7 @@ void snd_hdac_device_exit(struct hdac_device *dev); int snd_hdac_device_register(struct hdac_device *codec); void snd_hdac_device_unregister(struct hdac_device *codec); int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name); +int snd_hdac_codec_modalias(struct hdac_device *hdac, char *buf, size_t size); int snd_hdac_refresh_widgets(struct hdac_device *codec); int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec); -- cgit v1.2.3 From b6e84c99b121fcba34166842987be96956148bb8 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Mon, 19 Oct 2015 16:58:46 +0530 Subject: ALSA: hdac: Add macro for hda ext devices entry With the new modalias infrastructure support added for hda, create a macro for ext devices similar to legacy to add the device entry. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/hdaudio_ext.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 94210dcdb6ea..a4cadd9c297a 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -40,6 +40,13 @@ void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus); #define hbus_to_ebus(_bus) \ container_of(_bus, struct hdac_ext_bus, bus) +#define HDA_CODEC_REV_EXT_ENTRY(_vid, _rev, _name, drv_data) \ + { .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \ + .api_version = HDA_DEV_ASOC, \ + .driver_data = (unsigned long)(drv_data) } +#define HDA_CODEC_EXT_ENTRY(_vid, _revid, _name, _drv_data) \ + HDA_CODEC_REV_EXT_ENTRY(_vid, _revid, _name, _drv_data) + int snd_hdac_ext_bus_parse_capabilities(struct hdac_ext_bus *sbus); void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *chip, bool enable); void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *chip, bool enable); -- cgit v1.2.3 From bc1043cdcd84cb441d80cfd79ae4325218f6f9ba Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 18 Oct 2015 15:39:16 +0200 Subject: ALSA: Add helper function to add single value constraint The recommended and most efficient way to constraint a configuration parameter to a single value is to set the minimum and maximum allowed values to the same value, i.e. calling snd_pcm_hw_constraint_minmax() with the same value for min and max. It is not necessarily obvious though that this is the approach that should be taken and some drivers have come up with other ways of solving this problem, e.g. installing a list constraint with a single item. List constraints are dynamic constraints though and hence less efficient than the static min-max constraint. This patch introduces a new helper function called snd_pcm_hw_constraint_single() which only takes a single value has the same effect as calling snd_pcm_hw_constraint_minmax() with the same values for min and max. But it is hopefully semantically more expressive, making it clear that this is the preferred way of setting a single value constraint. Signed-off-by: Lars-Peter Clausen Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee0a510..04fbcba9290b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1034,6 +1034,22 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, snd_pcm_hw_rule_func_t func, void *private, int dep, ...); +/** + * snd_pcm_hw_constraint_single() - Constrain parameter to a single value + * @runtime: PCM runtime instance + * @var: The hw_params variable to constrain + * @val: The value to constrain to + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +static inline int snd_pcm_hw_constraint_single( + struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + unsigned int val) +{ + return snd_pcm_hw_constraint_minmax(runtime, var, val, val); +} + int snd_pcm_format_signed(snd_pcm_format_t format); int snd_pcm_format_unsigned(snd_pcm_format_t format); int snd_pcm_format_linear(snd_pcm_format_t format); -- cgit v1.2.3 From 16566e47098211e30b3d8a0bc6a3576871ada8e8 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 21 Oct 2015 09:46:05 +0800 Subject: ASoC: rt5640: Fill up the IN3's support Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- include/sound/rt5640.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/rt5640.h b/include/sound/rt5640.h index 59d26dd81e45..e3c84b92ff70 100644 --- a/include/sound/rt5640.h +++ b/include/sound/rt5640.h @@ -12,9 +12,10 @@ #define __LINUX_SND_RT5640_H struct rt5640_platform_data { - /* IN1 & IN2 can optionally be differential */ + /* IN1 & IN2 & IN3 can optionally be differential */ bool in1_diff; bool in2_diff; + bool in3_diff; bool dmic_en; bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */ -- cgit v1.2.3 From 26d9ca3462df8f7e83fc372b23c8da5ed2b1c4f3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 18 Oct 2015 17:04:33 +0200 Subject: ASoC: Let snd_soc_limit_volume() take a snd_soc_card snd_soc_limit_volume() operates on a card and the CODEC that is passed in is only used to look up the card. Let it directly take the card instead. This makes it possible to use it when no snd_soc_codec is available. Signed-off-by: Lars-Peter Clausen Acked-by: Jarkko Nikula Tested-by: Jarkko Nikula Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 884e728b09d9..71e0c0566b6e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -591,7 +591,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_soc_limit_volume(struct snd_soc_codec *codec, +int snd_soc_limit_volume(struct snd_soc_card *card, const char *name, int max); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -- cgit v1.2.3 From 6f0c42269f000b1e346c84d9a589f17aa94c96d8 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 13 Oct 2015 23:41:00 +0800 Subject: ASoC: compress: add config item for soc-compress to make it compiled only when needed We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, when nobody select it, the soc-compress will not be compiled. Here also change Kconfig to 'select SND_SOC_COMPRESS' for drivers that needed soc-compress. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 +- include/sound/soc.h | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1384c7..238200ffba5b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -214,7 +214,7 @@ struct snd_soc_dai_driver { int (*suspend)(struct snd_soc_dai *dai); int (*resume)(struct snd_soc_dai *dai); /* compress dai */ - bool compress_dai; + int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); /* DAI is also used for the control bus */ bool bus_control; diff --git a/include/sound/soc.h b/include/sound/soc.h index 71e0c0566b6e..a7bc82b08cd4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -440,7 +440,9 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); -int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#ifdef CONFIG_SND_SOC_COMPRESS +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); -- cgit v1.2.3 From 93e39a11520c000c9086215460bf27b35b09c724 Mon Sep 17 00:00:00 2001 From: Mythri P K Date: Tue, 20 Oct 2015 22:30:08 +0530 Subject: ASoC: dapm: Add snd_soc_dapm_kcontrol_widget() Given a kcontrol, we may want to access the parent widget and it's associated data. So export function to return it. Signed-off-by: Mythri P K Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5abba037d245..7855cfe46b69 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -451,6 +451,9 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); +struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_widget( + struct snd_kcontrol *kcontrol); + int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level); -- cgit v1.2.3 From 53e597b1d194910bef53ed0632da329fef497904 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 22 Oct 2015 13:11:56 +0200 Subject: ALSA: Remove transfer_ack_{begin,end} callbacks from struct snd_pcm_runtime While there is nothing wrong with the transfer_ack_begin and transfer_ack_end callbacks per-se, the last documented user was part of the alsa-driver 0.5.12a package, which was released 14 years ago and even predates the upstream integration of the ALSA core and has subsequently been superseded by newer alsa-driver releases. This seems to indicate that there is no need for having these callbacks and they are just cruft that can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 ---- 1 file changed, 4 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2882dddfc91c..3e0ffd21901f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -402,10 +402,6 @@ struct snd_pcm_runtime { struct snd_pcm_hardware hw; struct snd_pcm_hw_constraints hw_constraints; - /* -- interrupt callbacks -- */ - void (*transfer_ack_begin)(struct snd_pcm_substream *substream); - void (*transfer_ack_end)(struct snd_pcm_substream *substream); - /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ int tstamp_type; /* timestamp type */ -- cgit v1.2.3 From 1d387a3fd86f2acf70803262c5a5a5a89df0e097 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Thu, 8 Oct 2015 09:37:51 -0700 Subject: ASoC: Document DAI signal polarity Currently there is no clear definition of what FSYNC polarity is. Different drivers use its own definition of what is "normal" and what is "inverted" fsync. This leads to compatibility problems between drivers. For example TegraX1 driver assumes that DSP-A format with frames starting at rising FSYNC edge has "inverted" polarity, while RT5677 assumes it is "normal" polarity. Explicitly specify meaning of BCLK/FSYNC polarity to avoid future compatibility problems. Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1384c7..91e2e61d584c 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,25 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * * Specifies whether the DAI can also support inverted clocks for the specified * format. + * + * BCLK: + * - "normal" polarity means signal is available at rising edge of BCLK + * - "inverted" polarity means signal is available at falling edge of BCLK + * + * FSYNC "normal" polarity depends on the frame format: + * - I2S: frame consists of left then right channel data. Left channel starts + * with falling FSYNC edge, right channel starts with rising FSYNC edge. + * - Left/Right Justified: frame consists of left then right channel data. + * Left channel starts with rising FSYNC edge, right channel starts with + * falling FSYNC edge. + * - DSP A/B: Frame starts with rising FSYNC edge. + * - AC97: Frame starts with rising FSYNC edge. + * + * "Negative" FSYNC polarity is the one opposite of "normal" polarity. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- cgit v1.2.3 From e5e113cf0d19392f26c6b63e63ad4680ee4ec5da Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 28 Oct 2015 11:37:53 +0100 Subject: ALSA: Constify ratden/ratnum constraints The ALSA core does not modify the constraints provided by a driver. Most constraint helper functions already take a const pointer to the constraint description, the exception at the moment being the ratden and ratnum constraints. Make those const as well, this allows a driver to declare them as const. Signed-off-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 4e6ad74fd8a2..b0be09279943 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -265,12 +265,12 @@ struct snd_ratden { struct snd_pcm_hw_constraint_ratnums { int nrats; - struct snd_ratnum *rats; + const struct snd_ratnum *rats; }; struct snd_pcm_hw_constraint_ratdens { int nrats; - struct snd_ratden *rats; + const struct snd_ratden *rats; }; struct snd_pcm_hw_constraint_list { @@ -970,7 +970,7 @@ int snd_interval_list(struct snd_interval *i, unsigned int count, int snd_interval_ranges(struct snd_interval *i, unsigned int count, const struct snd_interval *list, unsigned int mask); int snd_interval_ratnum(struct snd_interval *i, - unsigned int rats_count, struct snd_ratnum *rats, + unsigned int rats_count, const struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp); void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params); @@ -1000,11 +1000,11 @@ int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratnums *r); + const struct snd_pcm_hw_constraint_ratnums *r); int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratdens *r); + const struct snd_pcm_hw_constraint_ratdens *r); int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, unsigned int cond, unsigned int width, -- cgit v1.2.3