From 72496edcf85e048b4c5373d518e4f27938d9594e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2015 22:39:51 +0100 Subject: ALSA: seq: Don't compile snd_seq_device_load_drivers() for built-in Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index 2b5f24cc7548..d52433563da2 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -67,7 +67,11 @@ struct snd_seq_dev_ops { /* * prototypes */ +#ifdef CONFIG_MODULES void snd_seq_device_load_drivers(void); +#else +#define snd_seq_device_load_drivers() +#endif int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result); int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize); int snd_seq_device_unregister_driver(char *id); -- cgit v1.2.3 From 7c37ae5c625aaa4836466cfaea829a3199dfc571 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 10:51:59 +0100 Subject: ALSA: seq: Rewrite sequencer device binding with standard bus We've used the old house-made code for binding the sequencer device and driver. This can be far better implemented with the standard bus nowadays. This patch refactors the whole sequencer binding code with the bus /sys/bus/snd_seq. The devices appear as id-card-device on this bus and are bound with the drivers corresponding to the given id like the former implementation. The module autoload is also kept like before. There is no change in API functions by this patch, and almost all transitions are kept inside seq_device.c. The proc file output will change slightly but kept compatible as much as possible. Further integration works will follow in later patches. Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index d52433563da2..ea256b825146 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -43,8 +43,11 @@ struct snd_seq_device { void *private_data; /* private data for the caller */ void (*private_free)(struct snd_seq_device *device); struct list_head list; /* link to next device */ + struct device dev; }; +#define to_seq_dev(_dev) \ + container_of(_dev, struct snd_seq_device, dev) /* driver operators * init_device: -- cgit v1.2.3 From af03c243a1f014145dae34368fe975b2f08ed964 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 13:40:50 +0100 Subject: ALSA: seq: Clean up device and driver structs Use const string pointer instead of copying the id string to each object. Also drop the status and list fields of snd_seq_device struct that are no longer used. Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) (limited to 'include') diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index ea256b825146..b13cd2930d32 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -25,24 +25,16 @@ * registered device information */ -#define ID_LEN 32 - -/* status flag */ -#define SNDRV_SEQ_DEVICE_FREE 0 -#define SNDRV_SEQ_DEVICE_REGISTERED 1 - struct snd_seq_device { /* device info */ struct snd_card *card; /* sound card */ int device; /* device number */ - char id[ID_LEN]; /* driver id */ + const char *id; /* driver id */ char name[80]; /* device name */ int argsize; /* size of the argument */ void *driver_data; /* private data for driver */ - int status; /* flag - read only */ void *private_data; /* private data for the caller */ void (*private_free)(struct snd_seq_device *device); - struct list_head list; /* link to next device */ struct device dev; }; @@ -75,9 +67,11 @@ void snd_seq_device_load_drivers(void); #else #define snd_seq_device_load_drivers() #endif -int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result); -int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize); -int snd_seq_device_unregister_driver(char *id); +int snd_seq_device_new(struct snd_card *card, int device, const char *id, + int argsize, struct snd_seq_device **result); +int snd_seq_device_register_driver(const char *id, + struct snd_seq_dev_ops *entry, int argsize); +int snd_seq_device_unregister_driver(const char *id); #define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device)) -- cgit v1.2.3 From 056622053b8ae02978678ac1321b5bd956e7c812 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 13:43:22 +0100 Subject: ALSA: seq: Define driver object in each driver This patch moves the driver object initialization and allocation to each driver's module init/exit code like other normal drivers. The snd_seq_driver struct is now published in seq_device.h, and each driver is responsible to define it with proper driver attributes (name, probe and remove) with snd_seq_driver specific attributes as id and argsize fields. The helper functions snd_seq_driver_register(), snd_seq_driver_unregister() and module_snd_seq_driver() are used for simplifying codes. Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 27 +++++++++++++++++++-------- 1 file changed, 19 insertions(+), 8 deletions(-) (limited to 'include') diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index b13cd2930d32..ddc0d504cf39 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -41,8 +41,10 @@ struct snd_seq_device { #define to_seq_dev(_dev) \ container_of(_dev, struct snd_seq_device, dev) +/* sequencer driver */ + /* driver operators - * init_device: + * probe: * Initialize the device with given parameters. * Typically, * 1. call snd_hwdep_new @@ -50,15 +52,19 @@ struct snd_seq_device { * 3. call snd_hwdep_register * 4. store the instance to dev->driver_data pointer. * - * free_device: + * remove: * Release the private data. * Typically, call snd_device_free(dev->card, dev->driver_data) */ -struct snd_seq_dev_ops { - int (*init_device)(struct snd_seq_device *dev); - int (*free_device)(struct snd_seq_device *dev); +struct snd_seq_driver { + struct device_driver driver; + char *id; + int argsize; }; +#define to_seq_drv(_drv) \ + container_of(_drv, struct snd_seq_driver, driver) + /* * prototypes */ @@ -69,12 +75,17 @@ void snd_seq_device_load_drivers(void); #endif int snd_seq_device_new(struct snd_card *card, int device, const char *id, int argsize, struct snd_seq_device **result); -int snd_seq_device_register_driver(const char *id, - struct snd_seq_dev_ops *entry, int argsize); -int snd_seq_device_unregister_driver(const char *id); #define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device)) +int __must_check __snd_seq_driver_register(struct snd_seq_driver *drv, + struct module *mod); +#define snd_seq_driver_register(drv) \ + __snd_seq_driver_register(drv, THIS_MODULE) +void snd_seq_driver_unregister(struct snd_seq_driver *drv); + +#define module_snd_seq_driver(drv) \ + module_driver(drv, snd_seq_driver_register, snd_seq_driver_unregister) /* * id strings for generic devices -- cgit v1.2.3 From 54a721abd7953a58e5479065c0cfdd8679d785c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 14:20:24 +0100 Subject: ALSA: seq: Drop snd_seq_autoload_lock() and _unlock() The autoload lock became already superfluous due to the recent rework of autoload code. Let's drop them now. This allows us to simplify a few codes nicely. Signed-off-by: Takashi Iwai --- include/sound/seq_kernel.h | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'include') diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index 18a2ac58b88f..feb58d455560 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -99,13 +99,9 @@ int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp, int snd_seq_event_port_detach(int client, int port); #ifdef CONFIG_MODULES -void snd_seq_autoload_lock(void); -void snd_seq_autoload_unlock(void); void snd_seq_autoload_init(void); -#define snd_seq_autoload_exit() snd_seq_autoload_lock() +void snd_seq_autoload_exit(void); #else -#define snd_seq_autoload_lock() -#define snd_seq_autoload_unlock() #define snd_seq_autoload_init() #define snd_seq_autoload_exit() #endif -- cgit v1.2.3 From 76a3aeac2f6c02ecf065fa9baa279dd54bf2d819 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:27 +0100 Subject: hdspm.h: include stdint.h in userspace MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes compilation error: sound/hdspm.h:43:2: error: unknown type name ‘uint32_t’ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/hdspm.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index b357f1a5e29c..5737332d38f2 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,6 +20,12 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef __KERNEL__ +#include +#else +#include +#endif + /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 -- cgit v1.2.3 From 4bebf7091aa15ec60edf0dcbc654410a87ca21fe Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:38 +0100 Subject: include/uapi/sound/asound.h: include stdlib.h in userspace MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes compiler errors like: error: field ‘trigger_tstamp’ has incomplete type error: invalid application of ‘sizeof’ to incomplete t ype ‘struct timespec’ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd635957..1fe3f4f3d696 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -25,6 +25,9 @@ #include +#ifndef __KERNEL__ +#include +#endif /* * protocol version -- cgit v1.2.3 From bbf91c1c5bfc00c2961f15657359ee7e87de4269 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:42 +0100 Subject: include/uapi/sound/asequencer.h: include sound/asound.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes userspace compilation error: error: unknown type name ‘snd_seq_client_type_t’ snd_seq_client_type_t type; /* client type */ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/asequencer.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 09c8a00ea503..5a5fa4956ebd 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -22,6 +22,7 @@ #ifndef _UAPI__SOUND_ASEQUENCER_H #define _UAPI__SOUND_ASEQUENCER_H +#include /** version of the sequencer */ #define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) -- cgit v1.2.3 From b9956409c281931c74ba8d0a2b61a98076a58602 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:43 +0100 Subject: include/uapi/sound/emu10k1.h: include sound/asound.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes userspace compilation errors like: error: field ‘id’ has incomplete type struct snd_ctl_elem_id id; /* full control ID definition */ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/emu10k1.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index d1bbaf78457a..ec1535bb6aed 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -23,8 +23,7 @@ #define _UAPI__SOUND_EMU10K1_H #include - - +#include /* * ---- FX8010 ---- -- cgit v1.2.3 From 229d043096ea8e58829d37d35767afeac15997f5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:03 -0600 Subject: ALSA: core: selection of audio_tstamp type and accuracy reports Audio timestamps can be extracted from sample counters, wall clocks, PHC clocks (Ethernet AVB), on-demand synchronized snapshots. This patch provides the ability to report timestamping capabilities, select timestamp types and retrieve timestamp accuracy, if supported. Details can be found in Documentations/sound/alsa/timestamping.txt This functionality is introduced by reclaiming the reserved_aligned field introduced by commit9c7066aef4a5eb8e4063de28f06c508bf6f2963a in snd_pcm_status to provide userspace with selection/query capabilities. Additional driver_tstamp and audio_tstamp_accuracy fields are also added. snd_pcm_mmap_status remains a read-only structure with only the audio timestamp value accessible from user space. The selection of audio timestamp type is done through snd_pcm_status only This commit does not impact ABI and does not impact the default behavior. By default audio timestamp is aligned with hw_pointer and reports the DMA position. Backwards compatibility is handled by using the HDAudio wall clock for playback and the hw_ptr for all other cases. For timestamp selection a new STATUS_EXT ioctl is introduced with read/write parameters. Alsa-lib will be modified to make use of STATUS_EXT. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 60 +++++++++++++++++++++++++++++++++++++++++++++ include/uapi/sound/asound.h | 34 ++++++++++++++++++++++--- 2 files changed, 90 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c0ddb7e69c28..60f0e48f7905 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -60,6 +60,9 @@ struct snd_pcm_hardware { struct snd_pcm_substream; +struct snd_pcm_audio_tstamp_config; /* definitions further down */ +struct snd_pcm_audio_tstamp_report; + struct snd_pcm_ops { int (*open)(struct snd_pcm_substream *substream); int (*close)(struct snd_pcm_substream *substream); @@ -281,6 +284,58 @@ struct snd_pcm_hw_constraint_ranges { struct snd_pcm_hwptr_log; +/* + * userspace-provided audio timestamp config to kernel, + * structure is for internal use only and filled with dedicated unpack routine + */ +struct snd_pcm_audio_tstamp_config { + /* 5 of max 16 bits used */ + u32 type_requested:4; + u32 report_delay:1; /* add total delay to A/D or D/A */ +}; + +static inline void snd_pcm_unpack_audio_tstamp_config(__u32 data, + struct snd_pcm_audio_tstamp_config *config) +{ + config->type_requested = data & 0xF; + config->report_delay = (data >> 4) & 1; +} + +/* + * kernel-provided audio timestamp report to user-space + * structure is for internal use only and read by dedicated pack routine + */ +struct snd_pcm_audio_tstamp_report { + /* 6 of max 16 bits used for bit-fields */ + + /* for backwards compatibility */ + u32 valid:1; + + /* actual type if hardware could not support requested timestamp */ + u32 actual_type:4; + + /* accuracy represented in ns units */ + u32 accuracy_report:1; /* 0 if accuracy unknown, 1 if accuracy field is valid */ + u32 accuracy; /* up to 4.29s, will be packed in separate field */ +}; + +static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy, + const struct snd_pcm_audio_tstamp_report *report) +{ + u32 tmp; + + tmp = report->accuracy_report; + tmp <<= 4; + tmp |= report->actual_type; + tmp <<= 1; + tmp |= report->valid; + + *data &= 0xffff; /* zero-clear MSBs */ + *data |= (tmp << 16); + *accuracy = report->accuracy; +} + + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -361,6 +416,11 @@ struct snd_pcm_runtime { struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + /* -- audio timestamp config -- */ + struct snd_pcm_audio_tstamp_config audio_tstamp_config; + struct snd_pcm_audio_tstamp_report audio_tstamp_report; + struct timespec driver_tstamp; + #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 0e88e7a0f0eb..acef4e4d2735 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -267,10 +267,17 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ -#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ +#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ + #define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ + + typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ #define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */ @@ -408,6 +415,22 @@ struct snd_pcm_channel_info { unsigned int step; /* samples distance in bits */ }; +enum { + /* + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + + /* timestamp definitions */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +}; + struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ @@ -419,9 +442,11 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - __u32 reserved_alignment; /* must be filled with zero */ - struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */ + struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */ + __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -534,6 +559,7 @@ enum { #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) #define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) +#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) #define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40) #define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41) -- cgit v1.2.3 From 3179f62001880e588e229db3006a59ad87b7792a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:06 -0600 Subject: ALSA: core: add .get_time_info Introduce more generic .get_time_info to retrieve system timestamp and audio timestamp in single routine. Backwards compatibility is preserved with same functionality as with .wall_clock method (to be removed in following commits to avoid breaking git bisect) Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 60f0e48f7905..04f2d492ae57 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -76,6 +76,10 @@ struct snd_pcm_ops { snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); int (*wall_clock)(struct snd_pcm_substream *substream, struct timespec *audio_ts); + int (*get_time_info)(struct snd_pcm_substream *substream, + struct timespec *system_ts, struct timespec *audio_ts, + struct snd_pcm_audio_tstamp_config *audio_tstamp_config, + struct snd_pcm_audio_tstamp_report *audio_tstamp_report); int (*copy)(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *buf, snd_pcm_uframes_t count); -- cgit v1.2.3 From 2d52a5abd5be35340296a251092c8a594679df54 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:08 -0600 Subject: ALSA: core: remove .wall_clock can be removed without breaking git-bisect now Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 04f2d492ae57..0cb7f3f5df7b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -74,8 +74,6 @@ struct snd_pcm_ops { int (*prepare)(struct snd_pcm_substream *substream); int (*trigger)(struct snd_pcm_substream *substream, int cmd); snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); - int (*wall_clock)(struct snd_pcm_substream *substream, - struct timespec *audio_ts); int (*get_time_info)(struct snd_pcm_substream *substream, struct timespec *system_ts, struct timespec *audio_ts, struct snd_pcm_audio_tstamp_config *audio_tstamp_config, -- cgit v1.2.3 From c72638bdaabe9ea4b09003b9db7e1754f472fbed Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:09 -0600 Subject: ALSA: bump PCM protocol to 2.0.13 Bump PCM protocol to enable use of STATUS_EXT ioctls, older apps will still use STATUS and audio timestamp configuration is not supported (backwards compatible behavior). Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index acef4e4d2735..3d46e9a0cd2e 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -140,7 +140,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; -- cgit v1.2.3 From 052a9f698268e606ca01eb1ce2a672e548f2ce11 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Mon, 9 Feb 2015 00:18:11 -0800 Subject: ALSA: Add params_set_format helper Add a helper to set pcm format directly from params Signed-off-by: Fang, Yang A Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/pcm_params.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 3c45f3924ba7..c704357775fc 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -366,4 +366,11 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p) return snd_pcm_format_physical_width(params_format(p)); } +static inline void +params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt) +{ + snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), + (__force int)fmt); +} + #endif /* __SOUND_PCM_PARAMS_H */ -- cgit v1.2.3 From 48c7699fb2c799d084ce490bceea14fe04ad12a1 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:53 +0530 Subject: ASoC: core: allow pcms to be registered as nonatomic ALSA core with commit 257f8cce5d40 - "ALSA: pcm: Allow nonatomic trigger operations" allows trigger ops to implemented as nonatomic. For ASoC, we can specify this in dailinks and is updated while snd_pcm is created Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Cc: Takashi Iwai 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 0d1ade195628..76bc944dcb5c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -954,6 +954,9 @@ struct snd_soc_dai_link { unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; + /* Mark this pcm with non atomic ops */ + bool nonatomic; + /* Do not create a PCM for this DAI link (Backend link) */ unsigned int no_pcm:1; -- cgit v1.2.3 From f23e860edbb3f2208c0ab3448e756689bb4a3760 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Sat, 14 Feb 2015 17:22:49 -0800 Subject: ASoC: core: Add extra dapm properties for Device Tree The current helper functions, snd_soc_of_parse_audio_simple_widgets() and snd_soc_of_parse_audio_routing(), set dapm_widgets and dapm_routes without caring if they are already set by using build-in widgets and routes in the card driver. So there could be one of them, build-in one or Device Tree one, overrided by the other depending on which one was assigned later. This patch adds an extra pair of dapm_widgets and dapm_routes for DT use only so as to prevent unexpected overriding. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..f66a1ef98a40 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1071,11 +1071,16 @@ struct snd_soc_card { /* * Card-specific routes and widgets. + * Note: of_dapm_xxx for Device Tree; Otherwise for driver build-in. */ const struct snd_soc_dapm_widget *dapm_widgets; int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + const struct snd_soc_dapm_widget *of_dapm_widgets; + int num_of_dapm_widgets; + const struct snd_soc_dapm_route *of_dapm_routes; + int num_of_dapm_routes; bool fully_routed; struct work_struct deferred_resume_work; -- cgit v1.2.3 From e086e3035e0691b362755d1b5e24df631eee335a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 18:01:22 +0100 Subject: ALSA: core: Re-add snd_device_disconnect() Revive snd_device_disconnect() again so that it can be called from the individual driver. This time, HD-audio will need it. Signed-off-by: Takashi Iwai --- include/sound/core.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index da5748289968..b12931f513f4 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -278,7 +278,8 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, void *device_data, struct snd_device_ops *ops); int snd_device_register(struct snd_card *card, void *device_data); int snd_device_register_all(struct snd_card *card); -int snd_device_disconnect_all(struct snd_card *card); +void snd_device_disconnect(struct snd_card *card, void *device_data); +void snd_device_disconnect_all(struct snd_card *card); void snd_device_free(struct snd_card *card, void *device_data); void snd_device_free_all(struct snd_card *card); -- cgit v1.2.3 From 1a6ab46fa9c2bc9399694b4856ab7ea19c036485 Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Wed, 4 Mar 2015 10:56:13 +0900 Subject: ALSA: Fix spelling typo in Documentation/DocBook/alsa-driver-api.xml This patch fix spelling typo found in alsa-driver-api.xml. It is because this file is generated from comments in source files, I have to fix source files. Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 4 ++-- include/sound/control.h | 2 +- include/sound/soc.h | 2 +- include/uapi/sound/compress_offload.h | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) (limited to 'include') diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f48089d364c5..fa1d05512c09 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -70,7 +70,7 @@ struct snd_compr_runtime { * @device: device pointer * @direction: stream direction, playback/recording * @metadata_set: metadata set flag, true when set - * @next_track: has userspace signall next track transistion, true when set + * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -95,7 +95,7 @@ struct snd_compr_stream { * and the stream properties * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream - * @get_metadata: retreives the requested metadata values from stream + * @get_metadata: retrieves the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054023f7..95aad6d3fd1a 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -227,7 +227,7 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) * Add a virtual slave control to the given master. * Unlike snd_ctl_add_slave(), the element added via this function * is supposed to have volatile values, and get callback is called - * at each time quried from the master. + * at each time queried from the master. * * When the control peeks the hardware values directly and the value * can be changed by other means than the put callback of the element, diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..cf0bb156d6da 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1469,7 +1469,7 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec( } /** - * snd_soc_kcontrol_platform() - Returns the platform that registerd the control + * snd_soc_kcontrol_platform() - Returns the platform that registered the control * @kcontrol: The control for which to get the platform * * Note: This function will only work correctly if the control has been diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 22ed8cb7800b..e00d8cbfc628 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -75,7 +75,7 @@ struct snd_compr_tstamp { /** * struct snd_compr_avail - avail descriptor * @avail: Number of bytes available in ring buffer for writing/reading - * @tstamp: timestamp infomation + * @tstamp: timestamp information */ struct snd_compr_avail { __u64 avail; -- cgit v1.2.3 From 970939964c26db4643f58d4e84487821962e0b65 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:17 +0100 Subject: ASoC: Allow to register jacks at the card level Jacks are typically card level elements, but are currently registered with a CODEC. When it was originally introduced snd_soc_jack_new() took a snd_soc_card as its parameter, but at that time DAPM was only implemented at the CODEC level and there was only one CODEC per card. This made it clear which CODEC to use for the jack DAPM operations. But the multi-component patchset added support for having multiple CODECs per card and with it the API was updated to register jacks with a specific CODEC instance instead. Subsequently DAPM support at the card level has been introduced, but the snd_soc_jack_new() API has so remained unchanged. This leaves us with the issue that the DAPM pins that are managed by the jack detection logic usually are part of the card DAPM context but are accessed through a CODEC DAPM context. Currently this works fine, but might break in the future if we take a more hierarchical approach to DAPM contexts. Furthermore with componentization progressing systems that do not register a snd_soc_codec might appear, while these system may still want to able to register a jack. This patch addresses these issues by adding a new function called snd_soc_card_jack_new() that can be used to register jacks with the card rather than a CODEC. This new function is mostly identical to snd_soc_jack_new() except that it additionally allows to directly specify the DAPM pins associated with the jack. This was done since most users of snd_soc_jack_new() typically call snd_soc_jack_add_pins() right after it, which is not necessary with the new API and allows to reduce the amount of boiler plate code. The old snd_soc_jack_new() is re-implemented as a wrapper around snd_soc_card_jack_new(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 28 +++++++++++++++++++++++++--- 1 file changed, 25 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..99d9ce272209 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -450,8 +450,10 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, 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); +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins); + void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); @@ -659,7 +661,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack { struct mutex mutex; struct snd_jack *jack; - struct snd_soc_codec *codec; + struct snd_soc_card *card; struct list_head pins; int status; struct blocking_notifier_head notifier; @@ -1482,6 +1484,26 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform( return snd_soc_component_to_platform(snd_soc_kcontrol_component(kcontrol)); } +/** + * snd_soc_jack_new - Create a new jack + * @codec: ASoC CODEC + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jack: structure to use for the jack + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jack will be initialised. + */ +static inline int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, + int type, struct snd_soc_jack *jack) +{ + return snd_soc_card_jack_new(codec->component.card, id, type, jack, + NULL, 0); +} + int snd_soc_util_init(void); void snd_soc_util_exit(void); -- cgit v1.2.3 From 77c71765ef78f87dd901fcb4c751908e835a3b2e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:45 +0100 Subject: ASoC: Remove snd_soc_jack_new() There are no users of snd_soc_jack_new() left and new users should use snd_soc_card_jack_new() instead. So remove the function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 20 -------------------- 1 file changed, 20 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 99d9ce272209..40b3ee96f317 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1484,26 +1484,6 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform( return snd_soc_component_to_platform(snd_soc_kcontrol_component(kcontrol)); } -/** - * snd_soc_jack_new - Create a new jack - * @codec: ASoC CODEC - * @id: an identifying string for this jack - * @type: a bitmask of enum snd_jack_type values that can be detected by - * this jack - * @jack: structure to use for the jack - * - * Creates a new jack object. - * - * Returns zero if successful, or a negative error code on failure. - * On success jack will be initialised. - */ -static inline int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, - int type, struct snd_soc_jack *jack) -{ - return snd_soc_card_jack_new(codec->component.card, id, type, jack, - NULL, 0); -} - int snd_soc_util_init(void); void snd_soc_util_exit(void); -- cgit v1.2.3