From 175ee39e8f4053f95e1948afd75c74552b3a175c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:11 +0200 Subject: ASoC: Remove support for reg_access_defaults No users of reg_access_defaults are left and new drivers are going to use regmap for this, so support for it can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 24 ------------------------ 1 file changed, 24 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index d22cb0a06feb..447278a3b3e6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -403,12 +403,6 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg); int snd_soc_platform_read(struct snd_soc_platform *platform, unsigned int reg); int snd_soc_platform_write(struct snd_soc_platform *platform, @@ -541,22 +535,6 @@ int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -/** - * struct snd_soc_reg_access - Describes whether a given register is - * readable, writable or volatile. - * - * @reg: the register number - * @read: whether this register is readable - * @write: whether this register is writable - * @vol: whether this register is volatile - */ -struct snd_soc_reg_access { - u16 reg; - u16 read; - u16 write; - u16 vol; -}; - /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * @@ -760,8 +738,6 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; - short reg_access_size; - const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ -- cgit v1.2.3 From 2a1212a8342c469cee240cf69fe3001b898cda8e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:12 +0200 Subject: ASoC: Remove snd_soc_bulk_write_raw() No users of snd_soc_bulk_write_raw() are left and new drivers are going to use regmap directly for this, so the function can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 447278a3b3e6..3f7de6f992c0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -686,7 +686,6 @@ struct snd_soc_codec { unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - int (*bulk_write_raw)(struct snd_soc_codec *, unsigned int, const void *, size_t); void *reg_cache; const void *reg_def_copy; const struct snd_soc_cache_ops *cache_ops; @@ -1097,8 +1096,6 @@ struct soc_enum { unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); unsigned int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val); -unsigned int snd_soc_bulk_write_raw(struct snd_soc_codec *codec, - unsigned int reg, const void *data, size_t len); /* device driver data */ -- cgit v1.2.3 From b012aa619e50d22df0835b64a5dcebc221fb8053 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:13 +0200 Subject: ASoC: Remove reg_def_copy reg_def_copy was introduced in commit 3335ddca ("ASoC: soc-cache: Use reg_def_copy instead of reg_cache_default") to keep a copy of the register defaults around in case the register defaults where placed in the __devinitdata section. With the __devinitdata section gone we effectivly keep the same data around twice. This patch removes reg_def_copy and uses reg_cache_default directly instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 3f7de6f992c0..62f320f56644 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -687,7 +687,6 @@ struct snd_soc_codec { unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); void *reg_cache; - const void *reg_def_copy; const struct snd_soc_cache_ops *cache_ops; struct mutex cache_rw_mutex; int val_bytes; -- cgit v1.2.3 From a94ed23436fb28bdcdd66e7fcf68ca5f7967e456 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:14 +0200 Subject: ASoC: Remove 'reg_size' field from snd_soc_codec struct The reg_size field is calculated in snd_soc_register_codec() and then used exactly once in snd_soc_flat_cache_init(). Since it is calculated based on other fields from the codec struct just move the calculation to snd_soc_flat_cache_init() and remove the 'reg_size' field from the codec struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 62f320f56644..577212629d0f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -662,7 +662,6 @@ struct snd_soc_codec { struct list_head card_list; int num_dai; enum snd_soc_compress_type compress_type; - size_t reg_size; /* reg_cache_size * reg_word_size */ int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); int (*writable_register)(struct snd_soc_codec *, unsigned int); -- cgit v1.2.3 From f90fb3f778042b0b9f9aa1fd48cb76047a25eac0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 20:31:15 +0200 Subject: ASoC: Remove infrastructure for supporting multiple cache types The only cache type left is the flat cache and new other cache types won't be added since new drivers are supposed to use regmap directly for IO and caching. This patch removes the snd_soc_cache_ops indirection that was added to support multiple cache types and modifies the code to always use the flat cache directly. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 27 --------------------------- 1 file changed, 27 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 577212629d0f..a72af6327987 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -330,7 +330,6 @@ struct soc_enum; struct snd_soc_jack; struct snd_soc_jack_zone; struct snd_soc_jack_pin; -struct snd_soc_cache_ops; #include #include @@ -348,10 +347,6 @@ enum snd_soc_control_type { SND_SOC_REGMAP, }; -enum snd_soc_compress_type { - SND_SOC_FLAT_COMPRESSION = 1, -}; - enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_PCM = 0, SND_SOC_PCM_CLASS_BE = 1, @@ -635,19 +630,6 @@ struct snd_soc_compr_ops { int (*trigger)(struct snd_compr_stream *); }; -/* SoC cache ops */ -struct snd_soc_cache_ops { - const char *name; - enum snd_soc_compress_type id; - int (*init)(struct snd_soc_codec *codec); - int (*exit)(struct snd_soc_codec *codec); - int (*read)(struct snd_soc_codec *codec, unsigned int reg, - unsigned int *value); - int (*write)(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value); - int (*sync)(struct snd_soc_codec *codec); -}; - /* SoC Audio Codec device */ struct snd_soc_codec { const char *name; @@ -661,7 +643,6 @@ struct snd_soc_codec { struct list_head list; struct list_head card_list; int num_dai; - enum snd_soc_compress_type compress_type; int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); int (*writable_register)(struct snd_soc_codec *, unsigned int); @@ -686,7 +667,6 @@ struct snd_soc_codec { unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); void *reg_cache; - const struct snd_soc_cache_ops *cache_ops; struct mutex cache_rw_mutex; int val_bytes; @@ -735,7 +715,6 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; - enum snd_soc_compress_type compress_type; /* codec bias level */ int (*set_bias_level)(struct snd_soc_codec *, @@ -917,12 +896,6 @@ struct snd_soc_codec_conf { * associated per device */ const char *name_prefix; - - /* - * set this to the desired compression type if you want to - * override the one supplied in codec->driver->compress_type - */ - enum snd_soc_compress_type compress_type; }; struct snd_soc_aux_dev { -- cgit v1.2.3 From 56fb7421d23c63cf22ac885d2db2302cefc9e1f1 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 13 Sep 2013 18:09:46 +0100 Subject: ASoC: trace: Make sure trace header doesnt depend on any headers Fix build so that asoc trace event header doesn't depend on other headers. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/trace/events/asoc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 5fc2dcdd21cd..03996b2bb04f 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -14,6 +14,7 @@ struct snd_soc_codec; struct snd_soc_platform; struct snd_soc_card; struct snd_soc_dapm_widget; +struct snd_soc_dapm_path; /* * Log register events -- cgit v1.2.3 From a0b03a616b08cf9d709812ff5cf7e9c0958d6807 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Sep 2013 20:37:34 +0100 Subject: ASoC: core: Implement devm_snd_soc_register_component() Since with the wider use of devres many drivers are now only calling snd_soc_unregister_component() in their remove functions providing a managed version will save a reasonable amount of code. Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index d22cb0a06feb..b970f019b452 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -386,6 +386,9 @@ 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); +int devm_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); -- cgit v1.2.3 From 0e4ff5c806263bf40ee5409ac283b776f0c11e41 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Sep 2013 18:02:05 +0100 Subject: ASoC: core: Add devm_snd_soc_register_card() Simplify error handling and remove repetitive (and rarely executed) code for unregistration by providing a devm_snd_soc_register() card. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index b970f019b452..d44728ab2be0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -369,6 +369,7 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); int snd_soc_poweroff(struct device *dev); -- cgit v1.2.3 From e54cf76ba2c9ec071a68e98f2830226c0cac8086 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 16 Sep 2013 13:01:46 +0100 Subject: ASoC: core: Add API for configuration of DAI BCLK ratio Some codec drivers when running in slave mode require that BCLK to sample rate ratio is explicitly set by the machine driver as it may not be exactly rate * frame size. Extend the DAI API by adding :- int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227d35d3..d8acf0ca77be 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -105,6 +105,8 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); + /* Digital Audio interface formatting */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); @@ -131,6 +133,7 @@ struct snd_soc_dai_ops { int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); + int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); /* * DAI format configuration -- cgit v1.2.3 From d191bd8de8c61619563f2b19f1fdcc0944ff1a72 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 4 Sep 2013 19:39:03 -0700 Subject: ASoC: snd_soc_codec includes snd_soc_component Codec includes component by this patch, and component moved to upside of codec to avoid extra declaration. Codec dai will be registered via component by this patch. Current component register function is used for cpu, and it is using dai/dais functions properly to keep existing cpu dai name. And now, it will be used from codec also. But codec driver had been used dais function only even though it was single dai. This patch adds new flag which can selects dai/dais function on component register function to keep existing codec dai name. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 33 +++++++++++++++++++-------------- 1 file changed, 19 insertions(+), 14 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index d22cb0a06feb..9a81e2e7d661 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -670,6 +670,21 @@ struct snd_soc_cache_ops { int (*sync)(struct snd_soc_codec *codec); }; +/* component interface */ +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; +}; + /* SoC Audio Codec device */ struct snd_soc_codec { const char *name; @@ -715,6 +730,9 @@ struct snd_soc_codec { struct mutex cache_rw_mutex; int val_bytes; + /* component */ + struct snd_soc_component component; + /* dapm */ struct snd_soc_dapm_context dapm; unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ @@ -733,6 +751,7 @@ struct snd_soc_codec_driver { int (*remove)(struct snd_soc_codec *); int (*suspend)(struct snd_soc_codec *); int (*resume)(struct snd_soc_codec *); + struct snd_soc_component_driver component_driver; /* Default control and setup, added after probe() is run */ const struct snd_kcontrol_new *controls; @@ -849,20 +868,6 @@ 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 */ -- cgit v1.2.3 From cb470087669a3fab1958fec79dd7db280b33f178 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Sep 2013 17:39:56 -0700 Subject: ASoC: add .of_xlate_dai_name on snd_soc_component_driver ASoC sound driver requires CPU/CODEC drivers for probing, and each CPU/CODEC has some DAI on it. Then, "dai name matching" have been used to identify CPU-CODEC DAI pair on ASoC. But, the "dai port number matching" is now required from DeviceTree. The solution of this issue is to replace the dai port number into dai name. Now, CPU/CODEC are based on struct snd_soc_component, and it can care above as common issue. This patch adds .of_xlate_dai_name callback interface on struct snd_soc_component_driver, and snd_soc_of_get_dai_name() which is using .of_xlate_dai_name. Then, #sound-dai-cells which enables DAI specifier is required on CPU/CODEC device tree properties. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a81e2e7d661..1dd7dc5f7d52 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -13,6 +13,7 @@ #ifndef __LINUX_SND_SOC_H #define __LINUX_SND_SOC_H +#include #include #include #include @@ -673,6 +674,11 @@ struct snd_soc_cache_ops { /* component interface */ struct snd_soc_component_driver { const char *name; + + /* DT */ + int (*of_xlate_dai_name)(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name); }; struct snd_soc_component { @@ -1206,6 +1212,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix); +int snd_soc_of_get_dai_name(struct device_node *of_node, + const char **dai_name); #include -- cgit v1.2.3 From fd792f8fbcfa95674b6c417429f576ad1d808086 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:14:32 +0100 Subject: mfd: mc13xxx: Move SPI erratum workaround into SPI I/O function Move the workaround for double sending AUDIO_CODEC and AUDIO_DAC writes into the SPI core, aiding refactoring to eliminate the ASoC custom I/O functions and avoiding the extra writes for I2C. Signed-off-by: Mark Brown Signed-off-by: Lee Jones --- include/linux/mfd/mc13xxx.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/linux/mfd/mc13xxx.h b/include/linux/mfd/mc13xxx.h index 41ed59276c00..67c17b5a6f44 100644 --- a/include/linux/mfd/mc13xxx.h +++ b/include/linux/mfd/mc13xxx.h @@ -41,6 +41,13 @@ int mc13xxx_adc_do_conversion(struct mc13xxx *mc13xxx, unsigned int mode, unsigned int channel, u8 ato, bool atox, unsigned int *sample); +#define MC13783_AUDIO_RX0 36 +#define MC13783_AUDIO_RX1 37 +#define MC13783_AUDIO_TX 38 +#define MC13783_SSI_NETWORK 39 +#define MC13783_AUDIO_CODEC 40 +#define MC13783_AUDIO_DAC 41 + #define MC13XXX_IRQ_ADCDONE 0 #define MC13XXX_IRQ_ADCBISDONE 1 #define MC13XXX_IRQ_TS 2 -- cgit v1.2.3 From 249ce1387b7739dbea2ac1a697e4bf1e37ec06b7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 6 Oct 2013 13:43:49 +0200 Subject: ASoC: dapm: Add support for virtual mixer controls This patch adds support for virtual DAPM mixer controls. They are similar to virtual DAPM enums. There is no hardware register backing the control, so changing the control's value wont have any direct effect on the hardware. But it still influences the DAPM graph by causing the path it sits on to be connected or disconnected. This in turn can cause power changes for some of the widgets on the DAPM graph, which will then modify the hardware state. Signed-off-by: Lars-Peter Clausen Tested-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 4 ++++ include/sound/soc.h | 3 ++- 2 files changed, 6 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 27a72d5d4b00..2037c45adfe6 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -286,6 +286,8 @@ struct device; .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } +#define SOC_DAPM_SINGLE_VIRT(xname, max) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0) #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -300,6 +302,8 @@ struct device; .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array) #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ diff --git a/include/sound/soc.h b/include/sound/soc.h index d22cb0a06feb..b429dba57bf6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1088,7 +1088,8 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift; + int reg, rreg; + unsigned int shift, rshift; unsigned int invert:1; unsigned int autodisable:1; }; -- cgit v1.2.3 From 9f1614aae59033b79941220f65ad36f5fe54a579 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 11 Oct 2013 12:11:02 +0200 Subject: ASoC: snd_soc_dai_ops trigger function description Add a comment to the trigger function in snd_soc_dai_ops struct about possible command sequences. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227d35d3..0f2e5daa1451 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -166,6 +166,13 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); int (*prepare)(struct snd_pcm_substream *, struct snd_soc_dai *); + /* + * NOTE: Commands passed to the trigger function are not necessarily + * compatible with the current state of the dai. For example this + * sequence of commands is possible: START STOP STOP. + * So do not unconditionally use refcounting functions in the trigger + * function, e.g. clk_enable/disable. + */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); int (*bespoke_trigger)(struct snd_pcm_substream *, int, -- cgit v1.2.3 From 420f9739a62cdb027f5580d25c813501ff93aa6f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 16 Oct 2013 23:10:31 +0200 Subject: thinkpad-acpi: Add mute and mic-mute LED functionality The LEDs are currently not visible to userspace, for security reasons. They are exported through thinkpad_acpi.h for use by the snd-hda-intel driver. Thanks to Alex Hung and Takashi Iwai for writing parts of this patch. Signed-off-by: David Henningsson Acked-by: Henrique de Moraes Holschuh Signed-off-by: Takashi Iwai --- include/linux/thinkpad_acpi.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) create mode 100644 include/linux/thinkpad_acpi.h (limited to 'include') diff --git a/include/linux/thinkpad_acpi.h b/include/linux/thinkpad_acpi.h new file mode 100644 index 000000000000..361de59a2285 --- /dev/null +++ b/include/linux/thinkpad_acpi.h @@ -0,0 +1,15 @@ +#ifndef __THINKPAD_ACPI_H__ +#define __THINKPAD_ACPI_H__ + +/* These two functions return 0 if success, or negative error code + (e g -ENODEV if no led present) */ + +enum { + TPACPI_LED_MUTE, + TPACPI_LED_MICMUTE, + TPACPI_LED_MAX, +}; + +int tpacpi_led_set(int whichled, bool on); + +#endif -- cgit v1.2.3 From 82fbb4f7b47683077e0716474d4f1ce65a2146cb Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 4 Sep 2011 22:04:49 +0200 Subject: ALSA: add DICE driver As a start point for further development, this is an incomplete driver for DICE devices: - only playback (so no clock source except the bus clock) - only 44.1 kHz - no MIDI - recovery after bus reset is slow - hwdep device is created, but not actually implemented Contains compilation fixes by Stefan Richter. Signed-off-by: Clemens Ladisch --- include/uapi/sound/Kbuild | 1 + include/uapi/sound/asound.h | 3 ++- include/uapi/sound/firewire.h | 51 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 54 insertions(+), 1 deletion(-) create mode 100644 include/uapi/sound/firewire.h (limited to 'include') diff --git a/include/uapi/sound/Kbuild b/include/uapi/sound/Kbuild index 0f7d279ebde3..a7f27704f980 100644 --- a/include/uapi/sound/Kbuild +++ b/include/uapi/sound/Kbuild @@ -5,6 +5,7 @@ header-y += asound_fm.h header-y += compress_offload.h header-y += compress_params.h header-y += emu10k1.h +header-y += firewire.h header-y += hdsp.h header-y += hdspm.h header-y += sb16_csp.h diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 041203f20f6d..9fc6219d3848 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -93,9 +93,10 @@ enum { SNDRV_HWDEP_IFACE_SB_RC, /* SB Extigy/Audigy2NX remote control */ SNDRV_HWDEP_IFACE_HDA, /* HD-audio */ SNDRV_HWDEP_IFACE_USB_STREAM, /* direct access to usb stream */ + SNDRV_HWDEP_IFACE_FW_DICE, /* TC DICE FireWire device */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_USB_STREAM + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_DICE }; struct snd_hwdep_info { diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h new file mode 100644 index 000000000000..e86131ca49e5 --- /dev/null +++ b/include/uapi/sound/firewire.h @@ -0,0 +1,51 @@ +#ifndef UAPI_SOUND_FIREWIRE_H_INCLUDED +#define UAPI_SOUND_FIREWIRE_H_INCLUDED + +#include + +/* events can be read() from the hwdep device */ + +#define SNDRV_FIREWIRE_EVENT_LOCK_STATUS 0x000010cc +#define SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION 0xd1ce004e + +struct snd_firewire_event_common { + unsigned int type; /* SNDRV_FIREWIRE_EVENT_xxx */ +}; + +struct snd_firewire_event_lock_status { + unsigned int type; + unsigned int status; /* 0/1 = unlocked/locked */ +}; + +struct snd_firewire_event_dice_notification { + unsigned int type; + unsigned int notification; /* DICE-specific bits */ +}; + +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; +}; + + +#define SNDRV_FIREWIRE_IOCTL_GET_INFO _IOR('H', 0xf8, struct snd_firewire_get_info) +#define SNDRV_FIREWIRE_IOCTL_LOCK _IO('H', 0xf9) +#define SNDRV_FIREWIRE_IOCTL_UNLOCK _IO('H', 0xfa) + +#define SNDRV_FIREWIRE_TYPE_DICE 1 +/* Fireworks, AV/C, RME, MOTU, ... */ + +struct snd_firewire_get_info { + unsigned int type; /* SNDRV_FIREWIRE_TYPE_xxx */ + unsigned int card; /* same as fw_cdev_get_info.card */ + unsigned char guid[8]; + char device_name[16]; /* device node in /dev */ +}; + +/* + * SNDRV_FIREWIRE_IOCTL_LOCK prevents the driver from streaming. + * Returns -EBUSY if the driver is already streaming. + */ + +#endif -- cgit v1.2.3 From 3d8c8bc0250f7cb11f887691b7473b51adcd2bcb Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 17 Oct 2013 11:03:33 -0500 Subject: ASoC: cs42l73: Add platform data support for cs42l73 codec Add support for RST GPIO and Charge Pump Freq in platform data Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- include/sound/cs42l73.h | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) create mode 100644 include/sound/cs42l73.h (limited to 'include') diff --git a/include/sound/cs42l73.h b/include/sound/cs42l73.h new file mode 100644 index 000000000000..f354be4cdc9e --- /dev/null +++ b/include/sound/cs42l73.h @@ -0,0 +1,22 @@ +/* + * linux/sound/cs42l73.h -- Platform data for CS42L73 + * + * Copyright (c) 2012 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS42L73_H +#define __CS42L73_H + +struct cs42l73_platform_data { + /* RST GPIO */ + unsigned int reset_gpio; + unsigned int chgfreq; + int jack_detection; + unsigned int mclk_freq; +}; + +#endif /* __CS42L73_H */ -- cgit v1.2.3 From 6833c452c2fb47353566aa705d68541c6045c796 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 16 Oct 2013 22:05:26 -0700 Subject: ASoC: add snd_soc_of_get_dai_name() default of_xlate Current snd_soc_of_get_dai_name() needs .of_xlate_dai_name() callback on each component drivers. But required behavior on almost all these drivers is just returns its indexed driver's name. This patch adds this feature as default behavior. .of_xlate_dai_name() can overwrite it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index b13eecbaea78..6ed3dc0773cc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -644,10 +644,12 @@ struct snd_soc_component_driver { struct snd_soc_component { const char *name; int id; - int num_dai; struct device *dev; struct list_head list; + struct snd_soc_dai_driver *dai_drv; + int num_dai; + const struct snd_soc_component_driver *driver; }; -- cgit v1.2.3 From c0de42bf595238e9dd593405ebc2992cc8470732 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 8 Oct 2013 15:07:59 +0200 Subject: ASoC: dmaengine-pcm: Add support for querying DMA capabilities Currently each platform making use the the generic dmaengine PCM driver still needs to provide a custom snd_pcm_hardware struct which specifies the capabilities of the DMA controller, e.g. the maximum period size that can be supported. This patch adds code which uses the newly introduced dma_get_slave_caps() API to query this information from the dmaengine driver. The new code path will only be taken if the 'pcm_hardware' field of the snd_dmaengine_pcm_config struct is NULL. The patch also introduces a new 'fifo_size' field to the snd_dmaengine_dai_dma_data struct which is used to initialize the snd_pcm_hardware 'fifo_size' field and needs to be set by the DAI driver. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f11c35cd5532..83b2c3e95a1a 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,6 +61,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @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. + * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -68,6 +69,7 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; + unsigned int fifo_size; }; void snd_dmaengine_pcm_set_config_from_dai_data( -- cgit v1.2.3 From ecfc0c04f236f1e2a95094792ec10cf27be39f7c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Oct 2013 21:13:19 +0100 Subject: ASoC: dai: Provide interface for setting DMA data at probe time Allow DMA data to be set at probe time for devices that can do that, avoiding the need to do it every time we start a stream and supporting non-DT dmaengine users using the helpers. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227d35d3..97943fd397cc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -276,6 +276,13 @@ static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, dai->capture_dma_data = data; } +static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, + void *playback, void *capture) +{ + dai->playback_dma_data = playback; + dai->capture_dma_data = capture; +} + static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, void *data) { -- cgit v1.2.3 From cdcfcac968a1ec648434892b6addd80e66a5a892 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 17 Oct 2013 22:50:59 -0700 Subject: ASoC: rcar: add rsnd_scu_hpbif_is_enable() Current SSI needs RSND_SSI_DEPENDENT flag to decide dependent/independent mode. And SCU needs RSND_SCU_USE_HPBIF flag to decide HPBIF is enable/disable. But these 2 means same things. This patch adds new rsnd_scu_hpbif_is_enable() function, and merges above methods. 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 fe66533e9b7a..6b4211c256b3 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -36,7 +36,6 @@ #define RSND_SSI_CLK_PIN_SHARE (1 << 31) #define RSND_SSI_CLK_FROM_ADG (1 << 30) /* clock parent is master */ #define RSND_SSI_SYNC (1 << 29) /* SSI34_sync etc */ -#define RSND_SSI_DEPENDENT (1 << 28) /* SSI needs SRU/SCU */ #define RSND_SSI_PLAY (1 << 24) -- cgit v1.2.3 From 4023fe6ff2192d6050647571ea54f5497b2ec8f6 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 18 Oct 2013 18:37:43 +0300 Subject: ASoC: davinci-mcasp: Extract DMA channels directly from DT Extract DMA channels directly from DT as they can not be found from platform resources anymore. This is a work-around until davinci audio driver is updated to use dmaengine. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- include/linux/platform_data/davinci_asp.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 8db5ae03b6e3..689a856b86f9 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -84,6 +84,8 @@ struct snd_platform_data { u8 version; u8 txnumevt; u8 rxnumevt; + int tx_dma_channel; + int rx_dma_channel; }; enum { -- cgit v1.2.3 From 055032142c42d2821c4aa617915292d6a08d56fc Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 23 Oct 2013 11:47:43 +0800 Subject: ALSA: Add SoC on-chip internal ram support for DMA buffer allocation Now it's quite common that an SoC contains its on-chip internal RAM. By using this RAM space for DMA buffer during audio playback/record, we can shutdown the voltage for external RAM to save power. So add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen Reviewed-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b8213df7..510aec437f72 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ /* * info for buffer allocation -- cgit v1.2.3 From ea73b7ddf13548afd666373dc5e26ee7c812a3fe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 19 Oct 2013 17:43:51 +0100 Subject: ASoC: dmaengine: Support custom channel names Some devices have more than just simple TX and RX DMA channels, for example modern Samsung I2S IPs support a secondary transmit DMA stream which is mixed into the primary stream during playback. Allow such devices to specify the names of the channels to be requested in their dma_data. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- include/sound/dmaengine_pcm.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 83b2c3e95a1a..15017311f2e9 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,6 +61,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @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. + * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { @@ -69,6 +70,7 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; + const char *chan_name; unsigned int fifo_size; }; @@ -98,6 +100,10 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) +/* + * The PCM streams have custom channel names specified. + */ +#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM -- cgit v1.2.3 From a5606f85611267047206d8ba055bc0e4ba166ad3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Oct 2013 14:25:32 +0200 Subject: ALSA: Add ifdef CONFIG_GENERIC_ALLOCATOR for SNDRV_DMA_TYPE_IRAM code It turned out that we can't use gen_pool_*() functions on archs without CONFIG_GENERIC_ALLOCATOR (resulting in missing symbols), since linux/genalloc.h doesn't provide dummy functions for all. We'd be able to fix linux/genalloc.h size, but I take an easier path for now... Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 510aec437f72..af9983970417 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,7 +52,11 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#ifdef CONFIG_GENERIC_ALLOCATOR #define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ +#else +#define SNDRV_DMA_TYPE_DEV_IRAM SNDRV_DMA_TYPE_DEV +#endif /* * info for buffer allocation -- cgit v1.2.3 From 917f4b5cba78980a527098a910d94139d3e82c8d Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 24 Oct 2013 16:37:31 +0530 Subject: ALSA: compress: fix drain calls blocking other compress functions The drain and drain_notify callback were blocked by low level driver untill the draining was complete. Due to this being invoked with big fat mutex held, others ops like reading timestamp, calling pause, drop were blocked. So to fix this we add a new snd_compr_drain_notify() API. This would be required to be invoked by low level driver when drain or partial drain has been completed by the DSP. Thus we make the drain and partial_drain callback as non blocking and driver returns immediately after notifying DSP. The waiting is done while relasing the lock so that other ops can go ahead. Signed-off-by: Vinod Koul CC: stable@vger.kernel.org Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'include') diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 9031a26249b5..175ab3237b58 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -48,6 +48,8 @@ struct snd_compr_ops; * the ring buffer * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep + * @wait: drain wait queue + * @drain_wake: condition for drain wake */ struct snd_compr_runtime { snd_pcm_state_t state; @@ -59,6 +61,8 @@ struct snd_compr_runtime { u64 total_bytes_available; u64 total_bytes_transferred; wait_queue_head_t sleep; + wait_queue_head_t wait; + unsigned int drain_wake; void *private_data; }; @@ -171,4 +175,12 @@ static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream) wake_up(&stream->runtime->sleep); } +static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) +{ + snd_BUG_ON(!stream); + + stream->runtime->drain_wake = 1; + wake_up(&stream->runtime->wait); +} + #endif -- cgit v1.2.3 From 6dd17757927ba9d23c604fee6fe72b4755c7ea7f Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 25 Oct 2013 10:01:14 -0500 Subject: ASoC: cs42l52: Add platform data for reset gpio This patch adds platform data support for a reset GPIO. Also uses reset_gpio to toggle reset of the CODEC Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- include/sound/cs42l52.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index 4c68955f7330..7c2be4a51894 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -31,6 +31,8 @@ struct cs42l52_platform_data { /* Charge Pump Freq. Check datasheet Pg73 */ unsigned int chgfreq; + /* Reset GPIO */ + unsigned int reset_gpio; }; #endif /* __CS42L52_H */ -- cgit v1.2.3 From e12483e0f3dbc32dad8fa1dc97efac22b6aee94f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Oct 2013 16:37:11 +0100 Subject: ALSA: ak4114: Fix wrong register array size The size of the register cache array is actually 6 instead of 7, as it caches up to AK4114_REG_INT1_MASK. This resulted in unexpected access out of array range, although most of them aren't so serious (just reading one more byte on the stack at snd_ak4114_create()). Also, the check of cache size was wrongly done by checking with sizeof() instead of ARRAY_SIZE(). Fixed this together. (And yes, hardcoded numbers are bad, but I keep the coding style as is for making it clear what this patch actually does.) Spotted by coverity among several CIDs, e.g. 711621. Signed-off-by: Takashi Iwai --- include/sound/ak4114.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/ak4114.h b/include/sound/ak4114.h index 3ce69fd92523..52f02a60dba7 100644 --- a/include/sound/ak4114.h +++ b/include/sound/ak4114.h @@ -170,7 +170,7 @@ struct ak4114 { void * private_data; unsigned int init: 1; spinlock_t lock; - unsigned char regmap[7]; + unsigned char regmap[6]; unsigned char txcsb[5]; struct snd_kcontrol *kctls[AK4114_CONTROLS]; struct snd_pcm_substream *playback_substream; @@ -189,7 +189,7 @@ struct ak4114 { int snd_ak4114_create(struct snd_card *card, ak4114_read_t *read, ak4114_write_t *write, - const unsigned char pgm[7], const unsigned char txcsb[5], + const unsigned char pgm[6], const unsigned char txcsb[5], void *private_data, struct ak4114 **r_ak4114); void snd_ak4114_reg_write(struct ak4114 *ak4114, unsigned char reg, unsigned char mask, unsigned char val); void snd_ak4114_reinit(struct ak4114 *ak4114); -- cgit v1.2.3 From f44f2a5417b2968a8724b352cc0b2545a6bcb1f4 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 7 Nov 2013 10:08:22 +0100 Subject: ALSA: compress: fix drain calls blocking other compress functions (v6) The drain and drain_notify callback were blocked by low level driver until the draining was complete. Due to this being invoked with big fat mutex held, others ops like reading timestamp, calling pause, drop were blocked. So to fix this we add a new snd_compr_drain_notify() API. This would be required to be invoked by low level driver when drain or partial drain has been completed by the DSP. Thus we make the drain and partial_drain callback as non blocking and driver returns immediately after notifying DSP. The waiting is done while releasing the lock so that other ops can go ahead. [ The commit 917f4b5cba78 was wrongly applied from the preliminary patch. This commit corrects to the final version. Sorry for inconvenience! -- tiwai ] Signed-off-by: Vinod Koul CC: stable@vger.kernel.org Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) (limited to 'include') diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 175ab3237b58..ae6c3b8ed2f5 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -48,8 +48,6 @@ struct snd_compr_ops; * the ring buffer * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep - * @wait: drain wait queue - * @drain_wake: condition for drain wake */ struct snd_compr_runtime { snd_pcm_state_t state; @@ -61,8 +59,6 @@ struct snd_compr_runtime { u64 total_bytes_available; u64 total_bytes_transferred; wait_queue_head_t sleep; - wait_queue_head_t wait; - unsigned int drain_wake; void *private_data; }; @@ -177,10 +173,11 @@ static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream) static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) { - snd_BUG_ON(!stream); + if (snd_BUG_ON(!stream)) + return; - stream->runtime->drain_wake = 1; - wake_up(&stream->runtime->wait); + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + wake_up(&stream->runtime->sleep); } #endif -- cgit v1.2.3 From 3b098eb486868d57d7f2666d05b86c19a07df71b Mon Sep 17 00:00:00 2001 From: Chen Gang Date: Thu, 7 Nov 2013 11:14:29 +0800 Subject: ALSA: include/uapi/sound/firewire.h: use "_UAPI" instead of "UAPI" When installing, "scripts/headers_install.sh" will strip guard macro' "_UAPI" to prevent from appearing it to users. And also, all another files which need uapi prefix always use "_UAPI", not "UAPI". So use "_UAPI" instead of "UAPI" on the guard macro, and also give a comment for "#endif". Signed-off-by: Chen Gang Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/uapi/sound/firewire.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index e86131ca49e5..59f5961302bf 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -1,5 +1,5 @@ -#ifndef UAPI_SOUND_FIREWIRE_H_INCLUDED -#define UAPI_SOUND_FIREWIRE_H_INCLUDED +#ifndef _UAPI_SOUND_FIREWIRE_H_INCLUDED +#define _UAPI_SOUND_FIREWIRE_H_INCLUDED #include @@ -48,4 +48,4 @@ struct snd_firewire_get_info { * Returns -EBUSY if the driver is already streaming. */ -#endif +#endif /* _UAPI_SOUND_FIREWIRE_H_INCLUDED */ -- cgit v1.2.3