From 1ef6ab75c7deef931d6308af282ed7d8e480e77f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:31:55 +0200 Subject: [ALSA] ASoC: Make CPU and codec DAI operations have same type The CPU and codec DAI operations differ only in the presence of the digital mute operation for the codec so they may as well be the same type. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 30 ++++++------------------------ 1 file changed, 6 insertions(+), 24 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index d3c8c033dff8..73accbcfbd2d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -272,9 +272,9 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -/* ASoC codec DAI ops */ -struct snd_soc_codec_ops { - /* codec DAI clocking configuration */ +/* ASoC DAI ops */ +struct snd_soc_dai_ops { + /* DAI clocking configuration */ int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai, int clk_id, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_codec_dai *codec_dai, @@ -282,7 +282,7 @@ struct snd_soc_codec_ops { int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai, int div_id, int div); - /* CPU DAI format configuration */ + /* DAI format configuration */ int (*set_fmt)(struct snd_soc_codec_dai *codec_dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai, @@ -293,24 +293,6 @@ struct snd_soc_codec_ops { int (*digital_mute)(struct snd_soc_codec_dai *, int mute); }; -/* ASoC cpu DAI ops */ -struct snd_soc_cpu_ops { - /* CPU DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_cpu_dai *cpu_dai, - int clk_id, unsigned int freq, int dir); - int (*set_clkdiv)(struct snd_soc_cpu_dai *cpu_dai, - int div_id, int div); - int (*set_pll)(struct snd_soc_cpu_dai *cpu_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out); - - /* CPU DAI format configuration */ - int (*set_fmt)(struct snd_soc_cpu_dai *cpu_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_cpu_dai *cpu_dai, - unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_cpu_dai *, int tristate); -}; - /* SoC Codec DAI */ struct snd_soc_codec_dai { char *name; @@ -328,7 +310,7 @@ struct snd_soc_codec_dai { /* ops */ struct snd_soc_ops ops; - struct snd_soc_codec_ops dai_ops; + struct snd_soc_dai_ops dai_ops; /* DAI private data */ void *private_data; @@ -352,7 +334,7 @@ struct snd_soc_cpu_dai { /* ops */ struct snd_soc_ops ops; - struct snd_soc_cpu_ops dai_ops; + struct snd_soc_dai_ops dai_ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; -- cgit v1.2.1 From 0be9898adb6f58fee44f0fec0bbc0eac997ea9eb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:31:28 +0200 Subject: [ALSA] ASoC: Clarify API for bias configuration Currently the ASoC core configures the bias levels in the system using a callback on codecs and machines called 'dapm_event', passing it PCI style power levels as SNDRV_CTL_POWER_ constants. This is more obscure than it needs to be and has caused confusion to driver authors, especially given that DAPM is also performing power management. Address this by renaming the callback function to 'set_bias_level' and using constants explicitly representing the off, standby, pre-on and on states which DAPM transitions through. Also unexport the API for setting bias level: there are currently no in-tree users of this API other than the core itself and it is likely that the core would need to be extended to cater for any users. Signed-off-by: Mark Brown Cc: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 28 ++++++++++++++++++++++++---- 1 file changed, 24 insertions(+), 4 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 73accbcfbd2d..bca9538d9e50 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -102,6 +102,24 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } +/* + * Bias levels + * + * @ON: Bias is fully on for audio playback and capture operations. + * @PREPARE: Prepare for audio operations. Called before DAPM switching for + * stream start and stop operations. + * @STANDBY: Low power standby state when no playback/capture operations are + * in progress. NOTE: The transition time between STANDBY and ON + * should be as fast as possible and no longer than 10ms. + * @OFF: Power Off. No restrictions on transition times. + */ +enum snd_soc_bias_level { + SND_SOC_BIAS_ON, + SND_SOC_BIAS_PREPARE, + SND_SOC_BIAS_STANDBY, + SND_SOC_BIAS_OFF, +}; + /* * Digital Audio Interface (DAI) types */ @@ -356,7 +374,8 @@ struct snd_soc_codec { struct mutex mutex; /* callbacks */ - int (*dapm_event)(struct snd_soc_codec *codec, int event); + int (*set_bias_level)(struct snd_soc_codec *, + enum snd_soc_bias_level level); /* runtime */ struct snd_card *card; @@ -378,8 +397,8 @@ struct snd_soc_codec { /* dapm */ struct list_head dapm_widgets; struct list_head dapm_paths; - unsigned int dapm_state; - unsigned int suspend_dapm_state; + enum snd_soc_bias_level bias_level; + enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; /* codec DAI's */ @@ -449,7 +468,8 @@ struct snd_soc_machine { int (*resume_post)(struct platform_device *pdev); /* callbacks */ - int (*dapm_event)(struct snd_soc_machine *, int event); + int (*set_bias_level)(struct snd_soc_machine *, + enum snd_soc_bias_level level); /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; -- cgit v1.2.1 From e13ac2e9b18bde51cf32c69c2209df25791ab3e5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 May 2008 17:58:05 +0100 Subject: [ALSA] ASoC: Add SOC_DOUBLE_S8_TLV control type The SOC_DOUBLE_S8_TLV control type was originally implemented in the UDA1380 driver by Philipp Zabel and was moved into the core by me. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- include/sound/soc.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index bca9538d9e50..9fa2093e74eb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -73,6 +73,15 @@ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (reg_left) | ((shift) << 8) | \ ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } +#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, 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_s8, .get = snd_soc_get_volsw_s8, \ + .put = snd_soc_put_volsw_s8, \ + .private_value = (reg) | (((signed char)max) << 16) | \ + (((signed char)min) << 24) } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .texts = xtexts } @@ -267,6 +276,12 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /* SoC PCM stream information */ struct snd_soc_pcm_stream { -- cgit v1.2.1 From bdb92876f0a9d2b431199e385732ede89ff0b97d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:10 +0100 Subject: ALSA: ASoC: Pass the DAI being configured into CPU DAI probe and remove This allows per-DAI initialisation to be done by the CPU DAI drivers. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 9fa2093e74eb..56d2224c2c07 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -358,8 +358,10 @@ struct snd_soc_cpu_dai { unsigned char type; /* DAI callbacks */ - int (*probe)(struct platform_device *pdev); - void (*remove)(struct platform_device *pdev); + int (*probe)(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai); + void (*remove)(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai); int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); int (*resume)(struct platform_device *pdev, -- cgit v1.2.1 From 10144c09a0d6a62e1d56e25f142743c7a00e5dba Mon Sep 17 00:00:00 2001 From: Mike Montour Date: Wed, 11 Jun 2008 13:47:13 +0100 Subject: ALSA: ASoC: Add SOC_SINGLE_EXT_TLV control type Signed-off-by: Mike Montour Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 56d2224c2c07..1f5c62181002 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -100,6 +100,15 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } +#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmask, 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, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ -- cgit v1.2.1 From 6ed2597883b1b03ca94f62f0cfe908314cba6d6b Mon Sep 17 00:00:00 2001 From: Andy Green Date: Fri, 13 Jun 2008 16:24:05 +0100 Subject: ALSA: ASoC: Don't block system resume On OpenMoko soc-audio resume is taking 700ms of the whole resume time of 1.3s, dominated by writes to the codec over I2C. This patch shunts the resume guts into a workqueue which then is done asynchronously. The "card" is locked using the ALSA power state APIs as suggested by Mark Brown. [Added fix for race with resume to suspend and fixed a couple of nits from checkpatch -- broonie.] Signed-off-by: Andy Green Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f5c62181002..340223a8f24c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -510,6 +510,7 @@ struct snd_soc_device { struct snd_soc_codec *codec; struct snd_soc_codec_device *codec_dev; struct delayed_work delayed_work; + struct work_struct deferred_resume_work; void *codec_data; }; -- cgit v1.2.1 From 3c4b266fe642bcaebe2b95edb56c3f8802924ff9 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:17 +0100 Subject: ALSA: asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai in preparation for further ASoC v2 patches. This merger removes duplication in both DAI structures and simplifies the API for other users. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 71 +++++++++++++++++------------------------------------ 1 file changed, 23 insertions(+), 48 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 340223a8f24c..778e57e74dc8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -221,8 +221,7 @@ struct snd_soc_pcm_stream; struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; -struct snd_soc_codec_dai; -struct snd_soc_cpu_dai; +struct snd_soc_dai; struct snd_soc_codec; struct snd_soc_machine_config; struct soc_enum; @@ -317,50 +316,24 @@ struct snd_soc_ops { /* ASoC DAI ops */ struct snd_soc_dai_ops { /* DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai, + int (*set_sysclk)(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_codec_dai *codec_dai, + int (*set_pll)(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out); - int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai, - int div_id, int div); + int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); /* DAI format configuration */ - int (*set_fmt)(struct snd_soc_codec_dai *codec_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai, + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_codec_dai *, int tristate); + int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* digital mute */ - int (*digital_mute)(struct snd_soc_codec_dai *, int mute); + int (*digital_mute)(struct snd_soc_dai *dai, int mute); }; -/* SoC Codec DAI */ -struct snd_soc_codec_dai { - char *name; - int id; - unsigned char type; - - /* DAI capabilities */ - struct snd_soc_pcm_stream playback; - struct snd_soc_pcm_stream capture; - - /* DAI runtime info */ - struct snd_soc_codec *codec; - unsigned int active; - unsigned char pop_wait:1; - - /* ops */ - struct snd_soc_ops ops; - struct snd_soc_dai_ops dai_ops; - - /* DAI private data */ - void *private_data; -}; - -/* SoC CPU DAI */ -struct snd_soc_cpu_dai { - +/* SoC DAI (Digital Audio Interface) */ +struct snd_soc_dai { /* DAI description */ char *name; unsigned int id; @@ -368,13 +341,13 @@ struct snd_soc_cpu_dai { /* DAI callbacks */ int (*probe)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); void (*remove)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* ops */ struct snd_soc_ops ops; @@ -386,7 +359,9 @@ struct snd_soc_cpu_dai { /* DAI runtime info */ struct snd_pcm_runtime *runtime; - unsigned char active:1; + struct snd_soc_codec *codec; + unsigned int active; + unsigned char pop_wait:1; void *dma_data; /* DAI private data */ @@ -428,7 +403,7 @@ struct snd_soc_codec { struct delayed_work delayed_work; /* codec DAI's */ - struct snd_soc_codec_dai *dai; + struct snd_soc_dai *dai; unsigned int num_dai; }; @@ -447,12 +422,12 @@ struct snd_soc_platform { int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, + int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); @@ -466,8 +441,8 @@ struct snd_soc_dai_link { char *stream_name; /* Stream name */ /* DAI */ - struct snd_soc_codec_dai *codec_dai; - struct snd_soc_cpu_dai *cpu_dai; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; /* machine stream operations */ struct snd_soc_ops *ops; -- cgit v1.2.1 From 8c6529dbf881303920a415c2d14a500218661949 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:19:13 +0100 Subject: ALSA: asoc: core - add Digital Audio Interface (DAI) control functions. This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 778e57e74dc8..1890d87c5204 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -256,6 +256,27 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); +/* Digital Audio Interface clocking API.*/ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir); + +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div); + +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + +/* Digital Audio interface formatting */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); + +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots); + +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); + +/* Digital Audio Interface mute */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); + /* *Controls */ -- cgit v1.2.1