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-rw-r--r--Documentation/sound/alsa/timestamping.txt200
-rw-r--r--include/sound/pcm.h60
-rw-r--r--include/uapi/sound/asound.h34
3 files changed, 290 insertions, 4 deletions
diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt
new file mode 100644
index 000000000000..0b191a23f534
--- /dev/null
+++ b/Documentation/sound/alsa/timestamping.txt
@@ -0,0 +1,200 @@
+The ALSA API can provide two different system timestamps:
+
+- Trigger_tstamp is the system time snapshot taken when the .trigger
+callback is invoked. This snapshot is taken by the ALSA core in the
+general case, but specific hardware may have synchronization
+capabilities or conversely may only be able to provide a correct
+estimate with a delay. In the latter two cases, the low-level driver
+is responsible for updating the trigger_tstamp at the most appropriate
+and precise moment. Applications should not rely solely on the first
+trigger_tstamp but update their internal calculations if the driver
+provides a refined estimate with a delay.
+
+- tstamp is the current system timestamp updated during the last
+event or application query.
+The difference (tstamp - trigger_tstamp) defines the elapsed time.
+
+The ALSA API provides reports two basic pieces of information, avail
+and delay, which combined with the trigger and current system
+timestamps allow for applications to keep track of the 'fullness' of
+the ring buffer and the amount of queued samples.
+
+The use of these different pointers and time information depends on
+the application needs:
+
+- 'avail' reports how much can be written in the ring buffer
+- 'delay' reports the time it will take to hear a new sample after all
+queued samples have been played out.
+
+When timestamps are enabled, the avail/delay information is reported
+along with a snapshot of system time. Applications can select from
+CLOCK_REALTIME (NTP corrections including going backwards),
+CLOCK_MONOTONIC (NTP corrections but never going backwards),
+CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode
+dynamically with sw_params
+
+
+The ALSA API also provide an audio_tstamp which reflects the passage
+of time as measured by different components of audio hardware. In
+ascii-art, this could be represented as follows (for the playback
+case):
+
+
+--------------------------------------------------------------> time
+ ^ ^ ^ ^ ^
+ | | | | |
+ analog link dma app FullBuffer
+ time time time time time
+ | | | | |
+ |< codec delay >|<--hw delay-->|<queued samples>|<---avail->|
+ |<----------------- delay---------------------->| |
+ |<----ring buffer length---->|
+
+The analog time is taken at the last stage of the playback, as close
+as possible to the actual transducer
+
+The link time is taken at the output of the SOC/chipset as the samples
+are pushed on a link. The link time can be directly measured if
+supported in hardware by sample counters or wallclocks (e.g. with
+HDAudio 24MHz or PTP clock for networked solutions) or indirectly
+estimated (e.g. with the frame counter in USB).
+
+The DMA time is measured using counters - typically the least reliable
+of all measurements due to the bursty natured of DMA transfers.
+
+The app time corresponds to the time tracked by an application after
+writing in the ring buffer.
+
+The application can query what the hardware supports, define which
+audio time it wants reported by selecting the relevant settings in
+audio_tstamp_config fields, get an estimate of the timestamp
+accuracy. It can also request the delay-to-analog be included in the
+measurement. Direct access to the link time is very interesting on
+platforms that provide an embedded DSP; measuring directly the link
+time with dedicated hardware, possibly synchronized with system time,
+removes the need to keep track of internal DSP processing times and
+latency.
+
+In case the application requests an audio tstamp that is not supported
+in hardware/low-level driver, the type is overridden as DEFAULT and the
+timestamp will report the DMA time based on the hw_pointer value.
+
+For backwards compatibility with previous implementations that did not
+provide timestamp selection, with a zero-valued COMPAT timestamp type
+the results will default to the HDAudio wall clock for playback
+streams and to the DMA time (hw_ptr) in all other cases.
+
+The audio timestamp accuracy can be returned to user-space, so that
+appropriate decisions are made:
+
+- for dma time (default), the granularity of the transfers can be
+ inferred from the steps between updates and in turn provide
+ information on how much the application pointer can be rewound
+ safely.
+
+- the link time can be used to track long-term drifts between audio
+ and system time using the (tstamp-trigger_tstamp)/audio_tstamp
+ ratio, the precision helps define how much smoothing/low-pass
+ filtering is required. The link time can be either reset on startup
+ or reported as is (the latter being useful to compare progress of
+ different streams - but may require the wallclock to be always
+ running and not wrap-around during idle periods). If supported in
+ hardware, the absolute link time could also be used to define a
+ precise start time (patches WIP)
+
+- including the delay in the audio timestamp may
+ counter-intuitively not increase the precision of timestamps, e.g. if a
+ codec includes variable-latency DSP processing or a chain of
+ hardware components the delay is typically not known with precision.
+
+The accuracy is reported in nanosecond units (using an unsigned 32-bit
+word), which gives a max precision of 4.29s, more than enough for
+audio applications...
+
+Due to the varied nature of timestamping needs, even for a single
+application, the audio_tstamp_config can be changed dynamically. In
+the STATUS ioctl, the parameters are read-only and do not allow for
+any application selection. To work around this limitation without
+impacting legacy applications, a new STATUS_EXT ioctl is introduced
+with read/write parameters. ALSA-lib will be modified to make use of
+STATUS_EXT and effectively deprecate STATUS.
+
+The ALSA API only allows for a single audio timestamp to be reported
+at a time. This is a conscious design decision, reading the audio
+timestamps from hardware registers or from IPC takes time, the more
+timestamps are read the more imprecise the combined measurements
+are. To avoid any interpretation issues, a single (system, audio)
+timestamp is reported. Applications that need different timestamps
+will be required to issue multiple queries and perform an
+interpolation of the results
+
+In some hardware-specific configuration, the system timestamp is
+latched by a low-level audio subsytem, and the information provided
+back to the driver. Due to potential delays in the communication with
+the hardware, there is a risk of misalignment with the avail and delay
+information. To make sure applications are not confused, a
+driver_timestamp field is added in the snd_pcm_status structure; this
+timestamp shows when the information is put together by the driver
+before returning from the STATUS and STATUS_EXT ioctl. in most cases
+this driver_timestamp will be identical to the regular system tstamp.
+
+Examples of typestamping with HDaudio:
+
+1. DMA timestamp, no compensation for DMA+analog delay
+$ ./audio_time -p --ts_type=1
+playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662
+playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837
+playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420
+playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051
+playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751
+playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822
+
+2. DMA timestamp, compensation for DMA+analog delay
+$ ./audio_time -p --ts_type=1 -d
+playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153
+playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947
+playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685
+playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349
+playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694
+
+3. link timestamp, compensation for DMA+analog delay
+$ ./audio_time -p --ts_type=2 -d
+playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787
+playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801
+playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591
+playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779
+playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687
+playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146
+
+Example 1 shows that the timestamp at the DMA level is close to 1ms
+ahead of the actual playback time (as a side time this sort of
+measurement can help define rewind safeguards). Compensating for the
+DMA-link delay in example 2 helps remove the hardware buffering abut
+the information is still very jittery, with up to one sample of
+error. In example 3 where the timestamps are measured with the link
+wallclock, the timestamps show a monotonic behavior and a lower
+dispersion.
+
+Example 3 and 4 are with USB audio class. Example 3 shows a high
+offset between audio time and system time due to buffering. Example 4
+shows how compensating for the delay exposes a 1ms accuracy (due to
+the use of the frame counter by the driver)
+
+Example 3: DMA timestamp, no compensation for delay, delta of ~5ms
+$ ./audio_time -p -Dhw:1 -t1
+playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981
+playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864
+playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912
+playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935
+playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821
+playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259
+playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664
+
+Example 4: DMA timestamp, compensation for delay, delay of ~1ms
+$ ./audio_time -p -Dhw:1 -t1 -d
+playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520
+playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740
+playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081
+playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907
+playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824
+playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847
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)