ALSA: doc: ReSTize timestamping document

A simple conversion from a plain text file.
Put to designs subdirectory.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

2 files changed, 216 insertions, 0 deletions

diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst
index f7ca11307033..798b1a44bbbd 100644
--- a/Documentation/sound/designs/index.rst
+++ b/Documentation/sound/designs/index.rst
@@ -7,6 +7,7 @@ Designs and Implementations
    control-names
    channel-mapping-api
    compress-offload
+   timestamping
    procfile
    powersave
    oss-emulation
diff --git a/Documentation/sound/designs/timestamping.rst b/Documentation/sound/designs/timestamping.rst
new file mode 100644
index 000000000000..2b0fff503415
--- /dev/null
+++ b/Documentation/sound/designs/timestamping.rst
@@ -0,0 +1,215 @@
+=====================
+ALSA PCM Timestamping
+=====================
+
+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 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 nature of DMA transfers.
+
+The app time corresponds to the time tracked by an application after
+writing in the ring buffer.
+
+The application can query the hardware capabilities, define which
+audio time it wants reported by selecting the relevant settings in
+audio_tstamp_config fields, thus 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 subsystem, 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 but
+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