From ed70f3a264e9f746eaf17c96ccc4c9b7eda742dc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 4 Jun 2014 10:11:06 +0100 Subject: ASoC: arizona: Implement TDM support for Arizona devices Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- include/linux/mfd/arizona/core.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 5cf8b91ce996..11783b511b9a 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -110,6 +110,9 @@ struct arizona { int clk32k_ref; struct snd_soc_dapm_context *dapm; + + int tdm_width[ARIZONA_MAX_AIF]; + int tdm_slots[ARIZONA_MAX_AIF]; }; int arizona_clk32k_enable(struct arizona *arizona); -- cgit v1.2.3 From cc9e92431ee9c7fe974266e0e6533a1a68e45539 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 6 Jun 2014 14:14:05 +0100 Subject: ASoC: wm5102: Add controls to allow shaping of ultrasonic response Add controls to allow custom shaping of the ultrasonic response. This custom shaping can be turned on/off at runtime, although, it should be noted that settings will not affect a currently open audio stream, they will be applied when the next audio stream is started. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- include/linux/mfd/arizona/core.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 11783b511b9a..55926517d50b 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -113,6 +113,9 @@ struct arizona { int tdm_width[ARIZONA_MAX_AIF]; int tdm_slots[ARIZONA_MAX_AIF]; + + uint16_t dac_comp_coeff; + uint8_t dac_comp_enabled; }; int arizona_clk32k_enable(struct arizona *arizona); -- cgit v1.2.3 From c4027faf1dcfc325663464b3f97847358b172c0b Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 11 Jun 2014 18:14:39 +0800 Subject: ASoC: atmel-ssc: distinguish whether SSC supports fslen ext Add compatible string to distinguish whether SSC supports frame sync length extension. Signed-off-by: Bo Shen Acked-by: Nicolas Ferre Signed-off-by: Mark Brown --- include/linux/atmel-ssc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h index 571a12ebb018..e8dd40873d55 100644 --- a/include/linux/atmel-ssc.h +++ b/include/linux/atmel-ssc.h @@ -7,6 +7,7 @@ struct atmel_ssc_platform_data { int use_dma; + int has_fslen_ext; }; struct ssc_device { -- cgit v1.2.3 From dfaf535665faa4b5aba4b59633f6b724a467c96e Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 11 Jun 2014 18:14:40 +0800 Subject: ASoC: atmel_ssc_dai: enable fslen extension feature When SSC work as master, it will generate the frame sync signal. On old SoCs, it only supports frame sync length less or equal to 16bits, on newer SoCs, it supports frame sync length extension, which can support frame size larger than 16 bits. So, add this to make it supports playback 24/32 bits audio clips. Signed-off-by: Bo Shen Acked-by: Nicolas Ferre Signed-off-by: Mark Brown --- include/linux/atmel-ssc.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'include') diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h index e8dd40873d55..7c0f6549898b 100644 --- a/include/linux/atmel-ssc.h +++ b/include/linux/atmel-ssc.h @@ -72,6 +72,12 @@ void ssc_free(struct ssc_device *ssc); #define SSC_RFMR_DATNB_OFFSET 8 #define SSC_RFMR_FSEDGE_SIZE 1 #define SSC_RFMR_FSEDGE_OFFSET 24 +/* + * The FSLEN_EXT exist on at91sam9rl, at91sam9g10, + * at91sam9g20, and at91sam9g45 and newer SoCs + */ +#define SSC_RFMR_FSLEN_EXT_SIZE 4 +#define SSC_RFMR_FSLEN_EXT_OFFSET 28 #define SSC_RFMR_FSLEN_SIZE 4 #define SSC_RFMR_FSLEN_OFFSET 16 #define SSC_RFMR_FSOS_SIZE 4 @@ -110,6 +116,12 @@ void ssc_free(struct ssc_device *ssc); #define SSC_TFMR_FSDEN_OFFSET 23 #define SSC_TFMR_FSEDGE_SIZE 1 #define SSC_TFMR_FSEDGE_OFFSET 24 +/* + * The FSLEN_EXT exist on at91sam9rl, at91sam9g10, + * at91sam9g20, and at91sam9g45 and newer SoCs + */ +#define SSC_TFMR_FSLEN_EXT_SIZE 4 +#define SSC_TFMR_FSLEN_EXT_OFFSET 28 #define SSC_TFMR_FSLEN_SIZE 4 #define SSC_TFMR_FSLEN_OFFSET 16 #define SSC_TFMR_FSOS_SIZE 3 -- cgit v1.2.3 From 94f99c875c109e51decf0d8c25ec2c946db20c56 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:01 +0200 Subject: ASoC: Move name_prefix from CODEC to component Move the name_prefix from the CODEC struct to the component struct. This will eventually allow to specify prefixes for all types of components. It is also necessary to make the DAPM code component type independent (i.e. a DAPM context does not need to know whether it belongs to a CODEC or a platform or something else). Signed-off-by: Lars-Peter Clausen 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 ed9e2d7e5fdc..e1cce0042f8a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -682,6 +682,7 @@ struct snd_soc_component_driver { struct snd_soc_component { const char *name; int id; + const char *name_prefix; struct device *dev; unsigned int active; @@ -710,7 +711,6 @@ struct snd_soc_component { /* SoC Audio Codec device */ struct snd_soc_codec { const char *name; - const char *name_prefix; int id; struct device *dev; const struct snd_soc_codec_driver *driver; -- cgit v1.2.3 From f4333203ec933f9272c90c7add01774ec2cf94d3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:02 +0200 Subject: ASoC: Move name and id from CODEC/platform to component The component struct already has a name and id field which are initialized to the same values as the same fields in the CODEC and platform structs. So remove them from the CODEC and platform structs and used the ones from the component struct instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ---- include/trace/events/asoc.h | 6 +++--- 2 files changed, 3 insertions(+), 7 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index e1cce0042f8a..f64bf9452466 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -710,8 +710,6 @@ struct snd_soc_component { /* SoC Audio Codec device */ struct snd_soc_codec { - const char *name; - int id; struct device *dev; const struct snd_soc_codec_driver *driver; @@ -848,8 +846,6 @@ struct snd_soc_platform_driver { }; struct snd_soc_platform { - const char *name; - int id; struct device *dev; const struct snd_soc_platform_driver *driver; diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index c75c795a377b..0194a641e4e2 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -296,17 +296,17 @@ TRACE_EVENT(snd_soc_cache_sync, TP_ARGS(codec, type, status), TP_STRUCT__entry( - __string( name, codec->name ) + __string( name, codec->component.name) __string( status, status ) __string( type, type ) __field( int, id ) ), TP_fast_assign( - __assign_str(name, codec->name); + __assign_str(name, codec->component.name); __assign_str(status, status); __assign_str(type, type); - __entry->id = codec->id; + __entry->id = codec->component.id; ), TP_printk("codec=%s.%d type=%s status=%s", __get_str(name), -- cgit v1.2.3 From 7df3788410e674423375b88e2d95c46e4015f5f5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:04 +0200 Subject: ASoC: Auto disconnect pins from all DAPM contexts Currently only pins in CODEC DAPM contexts are automatically marked as non-connected if the card has the fully_routed flag set. This makes sense since widgets which qualify for auto-disconnection are only found in CODEC DAPM contexts. But with componentisation this is going to change, so consider all widgets for auto-disconnection. Also it is probably faster to walk the widgets list only once rather than once for each CODEC. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6b59471cdf44..8db627cc2fbe 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -431,7 +431,7 @@ int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, const char *pin); int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); -void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); +void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card); /* Mostly internal - should not normally be used */ void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); -- cgit v1.2.3 From 68f831c2724ab72c0088471b2ed1dc99e81948ef Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:05 +0200 Subject: ASoC: Add a set_bias_level() callback to the DAPM context struct Currently the DAPM code directly looks at the CODEC driver struct to get a handle to the set_bias_level() callback. This patch adds a new set_bias_level() callback to the DAPM context struct. The DAPM code will use this new callback instead of the CODEC callback. For CODECs the new callback is set up to call the CODEC specific set_bias_level callback(). Not looking directly at the CODEC driver struct will allow non CODEC DAPM contexts to implement a set_bias_level() callback. This is also similar to how the seq_notifier() and stream_event() callbacks are currently handled. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8db627cc2fbe..3a5c4f969c04 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -601,6 +601,8 @@ struct snd_soc_dapm_context { struct list_head list; int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + int (*set_bias_level)(struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level); #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; -- cgit v1.2.3 From ce0fc93ae56e2ba50ff8c220d69e4e860e889320 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:06 +0200 Subject: ASoC: Add DAPM support at the component level This patch adds full DAPM support at the component level. Previously there was only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor MUXs) for platforms. Having DAPM support at the component level will allow all types of components to use DAPM and also help in consolidating the DAPM support between CODECs and platforms. Since the DAPM context is directly embedded into the snd_soc_codec and snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of drivers moving the field just right now is not possible without causing code churn. The approach this patch takes is to add two new fields to the component struct. One field which is the pointer to the actual DAPM context used by the component and one DAPM context that will be used as the default if no other context was specified. For CODECs and platforms the pointer is initialized to point to the CODEC or platform DAPM context. All generic code when referencing a component's DAPM struct will go via the pointer. This will make it possible to eventually seamlessly move the DAPM context from snd_soc_codec and snd_soc_platform struct over once all direct references have been eliminated. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 ++ include/sound/soc.h | 30 ++++++++++++++++++++++++++++++ 2 files changed, 32 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3a5c4f969c04..e292683ee694 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -441,6 +441,8 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list); struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol); +struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( + struct snd_kcontrol *kcontrol); /* dapm widget types */ enum snd_soc_dapm_type { diff --git a/include/sound/soc.h b/include/sound/soc.h index f64bf9452466..a21dfecba56b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -706,6 +706,10 @@ struct snd_soc_component { int val_bytes; struct mutex io_mutex; + + /* Don't use these, use snd_soc_component_get_dapm() */ + struct snd_soc_dapm_context dapm; + struct snd_soc_dapm_context *dapm_ptr; }; /* SoC Audio Codec device */ @@ -1160,6 +1164,21 @@ static inline struct snd_soc_platform *snd_soc_component_to_platform( return container_of(component, struct snd_soc_platform, component); } +/** + * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is + * embedded in + * @dapm: The DAPM context to cast to the component + * + * This function must only be used on DAPM contexts that are known to be part of + * a component (e.g. in a component driver). Otherwise the behavior is + * undefined. + */ +static inline struct snd_soc_component *snd_soc_dapm_to_component( + struct snd_soc_dapm_context *dapm) +{ + return container_of(dapm, struct snd_soc_component, dapm); +} + /** * snd_soc_dapm_to_codec() - Casts a DAPM context to the CODEC it is embedded in * @dapm: The DAPM context to cast to the CODEC @@ -1187,6 +1206,17 @@ static inline struct snd_soc_platform *snd_soc_dapm_to_platform( return container_of(dapm, struct snd_soc_platform, dapm); } +/** + * snd_soc_component_get_dapm() - Returns the DAPM context associated with a + * component + * @component: The component for which to get the DAPM context + */ +static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( + struct snd_soc_component *component) +{ + return component->dapm_ptr; +} + /* codec IO */ unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, -- cgit v1.2.3 From bc9af9fa9b89cb74eed020066d882abf238fad69 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:07 +0200 Subject: ASoC: Use component DAPM context for platforms The snd_soc_platform dapm field is not accessed outside of the ASoC core. Switch it over to using the snd_soc_component DAPM context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index a21dfecba56b..84ebc079c92f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -862,8 +862,6 @@ struct snd_soc_platform { struct snd_soc_component component; - struct snd_soc_dapm_context dapm; - #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_platform_root; #endif @@ -1203,7 +1201,7 @@ static inline struct snd_soc_codec *snd_soc_dapm_to_codec( static inline struct snd_soc_platform *snd_soc_dapm_to_platform( struct snd_soc_dapm_context *dapm) { - return container_of(dapm, struct snd_soc_platform, dapm); + return snd_soc_component_to_platform(snd_soc_dapm_to_component(dapm)); } /** -- cgit v1.2.3 From 14e8bdebfbc1d5c8804b3520233b2d4e516056bc Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:08 +0200 Subject: ASoC: Add component level stream_event() and seq_notifier() support This patch adds stream_event() and seq_notifier() callbacks similar to those found in the snd_soc_codec_driver and snd_soc_platform driver struct to the snd_soc_component_driver struct. This is meant to unify the handling of these callbacks across different types of components and will eventually allow their removal from the CODEC and platfrom driver structs. The new callbacks are slightly different from the old ones in that they take a snd_soc_component as a parameter rather than a snd_soc_dapm_context. This was done since otherwise casting from the DAPM context to the component would typically be the first thing to do in the callback. And the interface becomes slightly cleaner by passing a snd_soc_component to all callbacks in the snd_soc_component_driver struct. The patch also already removes the stream_event() callback from the snd_soc_codec_driver and snd_soc_platform_driver structs as it is currently unused. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 84ebc079c92f..9a5b4f6fe847 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -677,6 +677,9 @@ struct snd_soc_component_driver { int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name); + void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type, + int subseq); + int (*stream_event)(struct snd_soc_component *, int event); }; struct snd_soc_component { @@ -792,9 +795,6 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); - /* codec stream completion event */ - int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); - bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ /* probe ordering - for components with runtime dependencies */ @@ -836,9 +836,6 @@ struct snd_soc_platform_driver { /* platform stream compress ops */ const struct snd_compr_ops *compr_ops; - /* platform stream completion event */ - int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); - /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; -- cgit v1.2.3 From 9420d97b3f8672478696ae9c3e33051243e1f4a0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:10 +0200 Subject: ASoC: dapm: Remove DAI DAPM context The DAI DAPM context was added in commit be09ad90 ("ASoC: core: Add platform DAI widget mapping") and the only user was removed again in commit ae10e7e8f ("ASoC: core: Only add platform DAI widgets once."). Now that we have a per component DAPM context it is unlikely that we'll need the DAI DAPM context again. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 688f2ba8009f..031be2ab75d0 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -257,7 +257,6 @@ struct snd_soc_dai { struct snd_soc_dapm_widget *playback_widget; struct snd_soc_dapm_widget *capture_widget; - struct snd_soc_dapm_context dapm; /* DAI DMA data */ void *playback_dma_data; -- cgit v1.2.3 From 88a8fe3df65fa0229b04f1c03411062230091cdd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:11 +0200 Subject: ASoC: dapm: Remove platform field from widget and dapm context struct The platform field in the snd_soc_dapm_widget and snd_soc_dapm_context structs is now unused can be removed. New code that wants to get the platform for a widget or dapm context should use snd_soc_dapm_to_platform(w->dapm) or snd_soc_dapm_to_platform(dapm). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e292683ee694..aac04ff84eea 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -526,7 +526,6 @@ struct snd_soc_dapm_widget { const char *name; /* widget name */ const char *sname; /* stream name */ struct snd_soc_codec *codec; - struct snd_soc_platform *platform; struct list_head list; struct snd_soc_dapm_context *dapm; @@ -595,7 +594,6 @@ struct snd_soc_dapm_context { struct device *dev; /* from parent - for debug */ struct snd_soc_component *component; /* parent component */ struct snd_soc_codec *codec; /* parent codec */ - struct snd_soc_platform *platform; /* parent platform */ struct snd_soc_card *card; /* parent card */ /* used during DAPM updates */ -- cgit v1.2.3 From d9288d0ba12de1b5efb830b9128e4cc6877318fc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:56:23 -0700 Subject: ASoC: rsnd: SSI + DMA can select BUSIF Sound data needs to be sent to R-Car sound SSI when playback. But, there are 2 interfaces for it. 1st is SSITDR/SSIRDR which are mapped on SSI. 2nd is SSIn_BUSIF which are mapped on SSIU. 2nd SSIn_BUSIF is used when DMA transfer, and it is always used if sound data came from via SRC. But, we can use it when SSI+DMA case too. (Current driver is assuming 1st SSITDR/SSIRDR for it) 2nd SSIn_BUSIF can be used as FIFO. This is very helpful/useful for SSI+DMA. But DMA address / DMA ID are not same between 1st/2nd cases. This patch care about these settings. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index f4a706f82cb7..d76412b84b48 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -34,6 +34,7 @@ * 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, _pio_irq, _flags) \ { .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags } -- cgit v1.2.3 From 5e8351de740d9eff26cc146a6591a4e7517496b0 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 30 Jun 2014 20:31:13 +0800 Subject: ASoC: add RT5670 CODEC driver This patch adds a minimum support of Realtek ALC5670 codec. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- include/sound/rt5670.h | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) create mode 100644 include/sound/rt5670.h (limited to 'include') diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h new file mode 100644 index 000000000000..bd311197a3b5 --- /dev/null +++ b/include/sound/rt5670.h @@ -0,0 +1,27 @@ +/* + * linux/sound/rt5670.h -- Platform data for RT5670 + * + * Copyright 2014 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5670_H +#define __LINUX_SND_RT5670_H + +struct rt5670_platform_data { + int jd_mode; + bool in2_diff; + + bool dmic_en; + unsigned int dmic1_data_pin; + /* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/ + unsigned int dmic2_data_pin; + /* 0 = GPIO8; 1 = IN3N; */ + unsigned int dmic3_data_pin; + /* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/ +}; + +#endif -- cgit v1.2.3 From 07cf7cbadb4d97a78be61119a406de8fe446467e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 20 Jun 2014 14:41:13 +0800 Subject: ASoC: add RT286 CODEC driver This patch adds Realtek ALC286 codec driver. ALC286 is a dual mode codec, which can run as HD-A or I2S mode. It is controlled by HD-A verb commands via I2C protocol. The following is the I/O difference between ALC286 and general I2S codecs. 1. A HD-A verb command contains three parts, NID, VID, and PID. And an I2S command contains only two parts: address and data. 2. Not only the register address is written, but the read command also includes the entire write command. 3. rt286 uses different registers for read and write the same bits. We map verb command to regmap structure. However, we read most registers from cache to prevent the asymmetry read/write issue in rt286. Signed-off-by: Bard Liao Signed-off-by: Gustaw Lewandowski Signed-off-by: Mark Brown --- include/sound/rt286.h | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) create mode 100644 include/sound/rt286.h (limited to 'include') diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 000000000000..eb773d1485f2 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif -- cgit v1.2.3 From 93c6ee94c140eefb6f9d5b6e2ad1acc2e138e44c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 3 Jul 2014 07:51:52 +0300 Subject: dma: Support for 3 bytes word size Add DMA_SLAVE_BUSWIDTH_3_BYTES to dma_slave_buswidth for engines and users to select 3 bytes as bus width. For example eDMA can be configured to use 3bytes mode and in audio we have formats stored on 3bytes in memory (_XXX_3LE) where this new bus width can be used. Signed-off-by: Peter Ujfalusi Acked-by: Vinod Koul Acked-by: Takashi Iwai Signed-off-by: Mark Brown --- include/linux/dmaengine.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index d2c5cc7c583c..3d1c2aa51530 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -299,6 +299,7 @@ enum dma_slave_buswidth { DMA_SLAVE_BUSWIDTH_UNDEFINED = 0, DMA_SLAVE_BUSWIDTH_1_BYTE = 1, DMA_SLAVE_BUSWIDTH_2_BYTES = 2, + DMA_SLAVE_BUSWIDTH_3_BYTES = 3, DMA_SLAVE_BUSWIDTH_4_BYTES = 4, DMA_SLAVE_BUSWIDTH_8_BYTES = 8, }; -- cgit v1.2.3 From 1e4c0d7c9a2b44e18fe9e93712672741f70e36da Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 7 Jul 2014 13:16:54 +0100 Subject: ASoC: add SOC_VALUE_ENUM_EXT Adds an equivalent of SOC_ENUM_EXT for value enums Strictly speaking SOC_ENUM_EXT can also be used to define a value enum since the only difference is the get and set functions. But this doesn't look good in code because it is inconsistent with the normal control definitions. Adding a specific SOC_VALUE_ENUM_EXT is better for code clarity. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7e5fdc..5d812cfe9427 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -248,6 +248,8 @@ .info = snd_soc_info_enum_double, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } +#define SOC_VALUE_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \ + SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) #define SND_SOC_BYTES(xname, xbase, xregs) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -- cgit v1.2.3 From 0ac8a52d4541adc284a4f3e8a1f6847236de1d8a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Jul 2014 16:51:49 +0200 Subject: ALSA: Provide a CLOCK_MONOTONIC_RAW timestamp type For applications which need to synchronise with external timebases such as broadcast TV applications the kernel monotonic time is not optimal as it includes adjustments from NTP and so may still include discontinuities due to that. A raw monotonic time which does not include any adjustments is available in the kernel from getrawmonotonic() so provide userspace with a new timestamp type SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW which provides timestamps based on this as an option. [dropped tstamp_type assignment code, as it's no longer needed -- tiwai] Reported-by: Daniel Thompson Signed-off-by: Mark Brown Acked-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 11 +++++++++-- include/uapi/sound/asound.h | 3 ++- 2 files changed, 11 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index d854fb31c000..6f3e10ca0e32 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -931,10 +931,17 @@ void snd_pcm_timer_done(struct snd_pcm_substream *substream); static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime, struct timespec *tv) { - if (runtime->tstamp_type == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC) + switch (runtime->tstamp_type) { + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC: ktime_get_ts(tv); - else + break; + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: + getrawmonotonic(tv); + break; + default: getnstimeofday(tv); + break; + } } /* diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 224948342f14..cbf7dc850a46 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -462,7 +462,8 @@ struct snd_xfern { enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ - SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; /* channel positions */ -- cgit v1.2.3 From 5646eda5851e6cfdfa22d41895e3f5daffa643d3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Jul 2014 09:50:19 +0200 Subject: ALSA: pcm: Add timestamp type to sw_params For allowing adjusting the timestamp type on the fly, add it to sw_params. The existing ioctl is still kept for compatibility. Along with this, increment the PCM protocol version. The extension was suggested by Clemens Ladisch. Acked-by: Jaroslav Kysela Reviewed-by: Mark Brown Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index cbf7dc850a46..a7e062f91f39 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -139,7 +139,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -391,7 +391,9 @@ struct snd_pcm_sw_params { snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */ snd_pcm_uframes_t silence_size; /* silence block size */ snd_pcm_uframes_t boundary; /* pointers wrap point */ - unsigned char reserved[64]; /* reserved for future */ + unsigned int tstamp_type; /* timestamp type */ + int pads; /* alignment, reserved */ + unsigned char reserved[56]; /* reserved for future */ }; struct snd_pcm_channel_info { -- cgit v1.2.3 From d788cbd3f9065d829351746f94417d469f14eaaf Mon Sep 17 00:00:00 2001 From: Kukjin Kim Date: Tue, 1 Jul 2014 06:32:27 +0900 Subject: ASoC: samsung: remove s5pc100 related codes This patch removes s5pc100 related codes in . Signed-off-by: Kukjin Kim Signed-off-by: Mark Brown --- include/linux/platform_data/asoc-s3c.h | 9 --------- 1 file changed, 9 deletions(-) (limited to 'include') diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 709c6f7e2f8c..a6591c693ebb 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -15,15 +15,6 @@ #define S3C64XX_AC97_GPE 1 extern void s3c64xx_ac97_setup_gpio(int); -/* - * The machine init code calls s5p*_spdif_setup_gpio with - * one of these defines in order to select appropriate bank - * of GPIO for S/PDIF pins - */ -#define S5PC100_SPDIF_GPD 0 -#define S5PC100_SPDIF_GPG3 1 -extern void s5pc100_spdif_setup_gpio(int); - struct samsung_i2s { /* If the Primary DAI has 5.1 Channels */ #define QUIRK_PRI_6CHAN (1 << 0) -- cgit v1.2.3 From 0cea76f3393782d67ccea8f07e9abf341bc4f60e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Jul 2014 16:31:01 +0200 Subject: ALSA: control: Define SNDRV_CTL_TLV_OP_* constants Instead of hard-coded magic numbers, define constants for op_flag to tlv callbacks. Signed-off-by: Takashi Iwai --- include/sound/control.h | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/control.h b/include/sound/control.h index 5358892b1b39..042613938a1d 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -31,10 +31,15 @@ typedef int (snd_kcontrol_info_t) (struct snd_kcontrol * kcontrol, struct snd_ct typedef int (snd_kcontrol_get_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol); typedef int (snd_kcontrol_put_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol); typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol, - int op_flag, /* 0=read,1=write,-1=command */ + int op_flag, /* SNDRV_CTL_TLV_OP_XXX */ unsigned int size, unsigned int __user *tlv); +enum { + SNDRV_CTL_TLV_OP_READ = 0, + SNDRV_CTL_TLV_OP_WRITE = 1, + SNDRV_CTL_TLV_OP_CMD = -1, +}; struct snd_kcontrol_new { snd_ctl_elem_iface_t iface; /* interface identifier */ -- cgit v1.2.3 From 7523a271682fc1f35355da344249ce8f8c8b8cb9 Mon Sep 17 00:00:00 2001 From: Omair Mohammed Abdullah Date: Tue, 15 Jul 2014 21:34:48 +0530 Subject: ASoC: core: add a helper for extended byte controls using TLV ALSA supports arbitrary length TLVs for each kcontrol that can be used to pass metadata about the control (e.g. volumes, enum information). The same transport mechanism is now used for arbitrary length data by defining a new helper. Signed-off-by: Omair Mohammed Abdullah Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/sound/soc.h | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7e5fdc..688fa667dee5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -270,7 +270,14 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_bytes_ext) \ {.max = xcount} } - +#define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .tlv.c = (snd_soc_bytes_tlv_callback), \ + .info = snd_soc_info_bytes_ext, \ + .private_value = (unsigned long)&(struct soc_bytes_ext) \ + {.max = xcount, .get = xhandler_get, .put = xhandler_put, } } #define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -552,6 +559,8 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *ucontrol); +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv); int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, @@ -1119,6 +1128,9 @@ struct soc_bytes { struct soc_bytes_ext { int max; + /* used for TLV byte control */ + int (*get)(unsigned int __user *bytes, unsigned int size); + int (*put)(const unsigned int __user *bytes, unsigned int size); }; /* multi register control */ -- cgit v1.2.3 From 88bd870f02dff5c9445286e185f21873f25a977f Mon Sep 17 00:00:00 2001 From: Benoit Cousson Date: Tue, 8 Jul 2014 23:19:34 +0200 Subject: ASoC: core: Add initial support for DAI multicodec DAI link assumes a one to one mapping between CPU DAI and CODEC. In some cases, the same CPU DAI can be connected to several codecs. This is the case for example, if you connect two mono codecs to the same I2S link in order to have a stereo card. The current ASoC implementation does not allow such setup. Add support for DAI links composed of a single CPU DAI and multiple CODECs. Sound cards have to pass the CODECs array in the corresponding DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in this array is described in the same manner single CODEC DAIs are (either DT/OF node or codec_name). Multi-codec links are not supported in the case of CODEC to CODEC links. Just print a warning if it happens. Based on an original code done by Misael. Signed-off-by: Benoit Cousson Signed-off-by: Misael Lopez Cruz Signed-off-by: Fabien Parent Tested-by: Lars-Peter Clausen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 4 ++++ include/sound/soc.h | 13 +++++++++++++ 2 files changed, 17 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 031be2ab75d0..e8b3080d196a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -272,6 +272,10 @@ struct snd_soc_dai { struct snd_soc_codec *codec; struct snd_soc_component *component; + /* CODEC TDM slot masks and params (for fixup) */ + unsigned int tx_mask; + unsigned int rx_mask; + struct snd_soc_card *card; struct list_head list; diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a5b4f6fe847..f2142cf3f243 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -846,6 +846,12 @@ struct snd_soc_platform_driver { int (*bespoke_trigger)(struct snd_pcm_substream *, int); }; +struct snd_soc_dai_link_component { + const char *name; + const struct device_node *of_node; + const char *dai_name; +}; + struct snd_soc_platform { struct device *dev; const struct snd_soc_platform_driver *driver; @@ -891,6 +897,10 @@ struct snd_soc_dai_link { const struct device_node *codec_of_node; /* You MUST specify the DAI name within the codec */ const char *codec_dai_name; + + struct snd_soc_dai_link_component *codecs; + unsigned int num_codecs; + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link @@ -1089,6 +1099,9 @@ struct snd_soc_pcm_runtime { struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; + struct snd_soc_dai **codec_dais; + unsigned int num_codecs; + struct delayed_work delayed_work; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; -- cgit v1.2.3 From 93e6958a3674d2fa42e2c24ad5156e65da1d8621 Mon Sep 17 00:00:00 2001 From: Benoit Cousson Date: Tue, 8 Jul 2014 23:19:38 +0200 Subject: ASoC: pcm: Add soc_dai_hw_params helper Add a function helper to factorize the hw_params code. Suggested by Lars-Peter Clausen Signed-off-by: Benoit Cousson Tested-by: Lars-Peter Clausen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index f2142cf3f243..98555f833ab4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -436,6 +436,10 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, int snd_soc_platform_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_platform *platform); +int soc_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); + /* Jack reporting */ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack); -- cgit v1.2.3 From 5df7f71d5cdfbcbfd7e1b68df9994609d33f7e58 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Mon, 14 Jul 2014 15:10:45 -0500 Subject: ASoC: tas2552: Support TI TAS2552 Amplifier Support the TI TAS2552 Class D amplifier. The TAS2552 is a high efficiency Class-D audio power amplifier with advanced battery current management and an integrated Class-G boost The device constantly measures the current and voltage across the load and provides a digital stream of this information. Signed-off-by: Dan Murphy Signed-off-by: Mark Brown --- include/sound/tas2552-plat.h | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) create mode 100644 include/sound/tas2552-plat.h (limited to 'include') diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 000000000000..65e7627ba38e --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int enable_gpio; +}; + +#endif -- cgit v1.2.3 From 589008106b9b64153554c8f3f120ee512b134bb5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 16 Jul 2014 17:45:27 +0200 Subject: ALSA: pcm: Introduce protocol version field to sw_params For controlling the new fields more strictly, add sw_params.proto field indicating the protocol version of the user-space. User-space should fill the SNDRV_PCM_VERSION value it's built with, then kernel can know whether the new fields should be evaluated or not. And now tstamp_type field is evaluated only when the valid value is set there. This avoids the wrong override of tstamp_type to zero, which is SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY. Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index a7e062f91f39..32168f7ffce3 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -391,8 +391,8 @@ struct snd_pcm_sw_params { snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */ snd_pcm_uframes_t silence_size; /* silence block size */ snd_pcm_uframes_t boundary; /* pointers wrap point */ - unsigned int tstamp_type; /* timestamp type */ - int pads; /* alignment, reserved */ + unsigned int proto; /* protocol version */ + unsigned int tstamp_type; /* timestamp type (req. proto >= 2.0.12) */ unsigned char reserved[56]; /* reserved for future */ }; -- cgit v1.2.3 From 9898e1ccf5af70ad2d03d5b77c591fb243c0f021 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 17 Jul 2014 22:01:05 +0200 Subject: ASoC: Remove per card platform list The platform_dev_list was added in commit f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component Support") and while platforms are added and remove from that list it is otherwise unused. This patch removes it again. 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 98555f833ab4..1f5b4901415e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -865,7 +865,6 @@ struct snd_soc_platform { struct snd_soc_card *card; struct list_head list; - struct list_head card_list; struct snd_soc_component component; @@ -1056,7 +1055,6 @@ struct snd_soc_card { /* lists of probed devices belonging to this card */ struct list_head codec_dev_list; - struct list_head platform_dev_list; struct list_head widgets; struct list_head paths; @@ -1299,7 +1297,6 @@ static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) { INIT_LIST_HEAD(&card->codec_dev_list); - INIT_LIST_HEAD(&card->platform_dev_list); INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); -- cgit v1.2.3 From 00200107a296cad3a950049a5ad7134a0d962b8d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 17 Jul 2014 22:01:07 +0200 Subject: ASoC: Move card field form platform/codec to component Both the snd_soc_codec and snd_soc_platform struct do have a pointer to the parent card and both handle this pointer in mostly the same way. This patch moves the card field to the component level which will allow further code consolidation between platforms and CODECS. Since there are only a handful of users of the snd_soc_codec struct's card field (and none of the snd_soc_platform's) these are update in this patch as well, which allows it to be removed from the snd_soc_codec struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f5b4901415e..5ee6ddde4831 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -691,6 +691,7 @@ struct snd_soc_component { int id; const char *name_prefix; struct device *dev; + struct snd_soc_card *card; unsigned int active; @@ -725,7 +726,6 @@ struct snd_soc_codec { const struct snd_soc_codec_driver *driver; struct mutex mutex; - struct snd_soc_card *card; struct list_head list; struct list_head card_list; @@ -863,7 +863,6 @@ struct snd_soc_platform { unsigned int suspended:1; /* platform is suspended */ unsigned int probed:1; - struct snd_soc_card *card; struct list_head list; struct snd_soc_component component; -- cgit v1.2.3 From 0f2780ad4c2a398528c7bb1572158d6e894e5dd2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 17 Jul 2014 22:01:08 +0200 Subject: ASoC: Add function to register component controls We have now everything in place to actual let a component register controls. Add a function which allows to do so. Also update snd_soc_add_codec_controls() and snd_soc_platform_controls() to use this new function internally. And while we are at it also change the num_controls parameter of those two functions from int to unsigned int. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 5ee6ddde4831..50ff6f3459aa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -507,10 +507,12 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, const char *prefix); struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); +int snd_soc_add_component_controls(struct snd_soc_component *component, + const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_codec_controls(struct snd_soc_codec *codec, - const struct snd_kcontrol_new *controls, int num_controls); + const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_platform_controls(struct snd_soc_platform *platform, - const struct snd_kcontrol_new *controls, int num_controls); + const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_card_controls(struct snd_soc_card *soc_card, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_dai_controls(struct snd_soc_dai *dai, -- cgit v1.2.3 From 94b912e42829b25d97b6b1f2be66c6aa81ac125f Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 29 Jul 2014 18:08:52 +0800 Subject: ARM: imx: Add the secondary request into the structure for imx-sdma SDMA supports device to device (per_2_per) scripts to handle DMA transfering between two peripheral devices. The per_2_per script, however, needs two dma requests from two sides while the current structure only defined one request. So this patch just simply adds the secondary request so as to let SDMA and its user to add its implementation later. [ Both change in the SDMA driver and its users like Freescale ASRC ASoC driver should be taken along with this change in order to truly support per_2_per sciprts. However, we here make an expediency by adding this first so that we can add either side later since this patch won't break any function and meanwhile it can make merge window more smoothly: we don't need to apply the change inside dmaengine branch via ASoC tree any more. -- Nicolin ] Signed-off-by: Nicolin Chen Acked-by: Shawn Guo Signed-off-by: Mark Brown --- include/linux/platform_data/dma-imx.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/linux/platform_data/dma-imx.h b/include/linux/platform_data/dma-imx.h index bcbc6c3c14c0..d05542aafa3e 100644 --- a/include/linux/platform_data/dma-imx.h +++ b/include/linux/platform_data/dma-imx.h @@ -50,6 +50,7 @@ enum imx_dma_prio { struct imx_dma_data { int dma_request; /* DMA request line */ + int dma_request2; /* secondary DMA request line */ enum sdma_peripheral_type peripheral_type; int priority; }; -- cgit v1.2.3 From d7821953cfe9803c593a682320468ce2de862803 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 29 Jul 2014 18:38:39 +0800 Subject: ASoC: wm8962: Let CODEC driver enable and disable its own MCLK snd_soc_open() will trigger pm_runtime resume() which will then enable the regulator and initialization. So we should make sure the MCLK is enabled before this resume(). Previously we let the machine driver get the clock and enable it in its probe(). However, considering about power saving, it'll be better to enable it when it's going to be used and disable it after using. So this patch just simply adds clk_get() and clk_enable() to WM8962 driver. Meanwhile, it marks clock pointer to NULL if no clock assigned to it so it will not break any current function. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- include/sound/wm8962.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/wm8962.h b/include/sound/wm8962.h index 79e6d427b858..0af7c1674cbf 100644 --- a/include/sound/wm8962.h +++ b/include/sound/wm8962.h @@ -37,6 +37,7 @@ #define WM8962_GPIO_FN_MICSCD 22 struct wm8962_pdata { + struct clk *mclk; int gpio_base; u32 gpio_init[WM8962_MAX_GPIO]; -- cgit v1.2.3