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streamsynchronizer: Add documentation
I didn't find the behavior and purpose of streamsynchronizer documented or intuitive. Eventually I got Edward to explain it to me, which was very helpful. Now I'm contributing some docs so that the next person doesn't have to figure it out by asking around and hoping for an answer. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7084>
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* Boston, MA 02110-1301, USA.
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* Boston, MA 02110-1301, USA.
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*/
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*/
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/**
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* SECTION:element-streamsynchronizer
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* @title: streamsynchronizer
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*
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* Enables gapless playback of heterogenous streams groups. This element is
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* used inside #playsink.
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*
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* #streamsynchronizer ensures only one stream group is active downstream at any
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* given time. Whenever a pad receives a `STREAM_START` with a group-id
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* different than the current one the pad is blocked until all other pads also
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* receive a `STREAM_START` with the same group-id.
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*
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* Once all pads have received a `STREAM_START` with the same group-id, the
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* previous group is completed and all the pads unblocked.
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*
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* When a group is completed, a patched #GstSegment is emitted downstream so
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* that the running time 0 from upstream of the new group becomes the running
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* time of the end of the previous group.
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*
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* ### Warning: deadlocks with multiple pads with different group-ids
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*
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* All pads connected to a streamsynchronizer are expected to share a group-id.
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* Therefore, sending `STREAM_START` in two or more sinkpads of
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* #streamsynchronizer with different group-ids will not allow for playback.
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*
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* As #streamsynchronizer is part of #playsink, this can easily happen
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* accidentally when mixing streams from different elements. The following is a
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* minimal naive pipeline exhibiting this problem:
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*
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* |[
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* # Will get stuck! The streams from audiotestsrc and videotestsrc don't
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* # share a group-id.
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* gst-launch-1.0 \
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* playsink name=myplaysink \
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* audiotestsrc ! myplaysink.audio_sink \
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* videotestsrc ! myplaysink.video_sink
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* ]|
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*
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* ## What are stream groups and group-ids
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*
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* A stream group is a group of streams that are meant to be played together.
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* Two streams belong to the same group if they set the same group-id in their
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* `STREAM_START` event (see #gst_event_set_group_id).
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*
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* The most common example is video and audio from the same .mp4 file. Demuxers
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* will ensure that the streams they output from their srcpads share the same
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* group-id.
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*
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* Another example is an out-of-band .srt subtitles file. In this case, despite
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* being a separate file, the subtitle stream should use the same group-id as
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* the video its meant to be used with, as they both need to be played together.
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* uridecodebin3 takes care of this automatically.
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*
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* ## Why do we need stream groups for gapless playback
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*
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* In theory, a player could implement gapless playback by performing some kind
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* of auto-plugging, keeping track of the playlists themselves and adding probes
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* to patch the #GstSegment. This is a lot of work, especially to get it done
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* correctly.
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*
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* The goal of #streamsynchronizer and group-ids is to standardize a solution
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* for gapless playback so that applications don't have to implement it from the
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* ground up, and for that solution to be generalizable to the worst possible
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* cases.
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*
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* Every new stream group received upstream of streamsynchronizer is expected
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* to start at running-time of zero. Hence, once the previous file is drained,
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* you can even remove a demuxer element and replace it by a demuxer for a
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* different format, and streamsynchronizer will take care of adjusting the
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* segment so that the data of the second file plays after the second.
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* If there is more than one stream within the same group (e.g. audio and video)
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* the shift will be done by the duration of the longest stream.
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*
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* ## Gapless playback caveats
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*
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* Given the substantial number of edge cases that need to be handled across a
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* large code surface, bugs in gapless playback are not uncommon.
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*
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* In general (not only in GStreamer) gapless playback requires several things
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* to work:
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*
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* - Buffering of the new file done well ahead of the previous file.
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* For this purpose #urisourcebin emits (and #decodebin3, #uridecodebin3,
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* #playbin and #playbin3 propagate) the #urisourcebin::about-to-finish
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* signal to notify what is the optimal time at which to provide the next uri
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* to play.
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* - Note that this requirement applies to the entire pipeline, including
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* decoders and audio sinks.
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*
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* - Extremely well bounded audio files. This is particularly hard for
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* compressed audio formats. See: codec delay, audio frame sizes. Achieving
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* arbitrary length of audio in many compressed formats will require either:
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* - Container features like MP4 edit lists to clip the priming and padding
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* PCM samples at the beginning and the end of the file, plus elements
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* correctly clipping the data accordingly.
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* - Features of the specific audio format used to perform the same clipping
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* at the decoder level.
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*
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* - All autoplugging during the switch must be done without pausing the
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* pipeline.
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*
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* - Playback must be possible without resetting the audio device (e.g. to
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* select a different sampling rate or audio format), or such a switch to be
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* done gapless.
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*
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* ## Example command line
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*
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* gst-play supports the `--gapless` flag which can be used to test gapless
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* playback. It will instantiate a playbin/playbin3 with a #playsink, which
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* itself will contain a #streamsynchronizer.
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*
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* |[
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* # BEWARE: high-pitch audio sweep, lower your volume.
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* wget https://www2.iis.fraunhofer.de/AAC/gapless-sweep_part1_iis.m4a
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* wget https://www2.iis.fraunhofer.de/AAC/gapless-sweep_part2_iis.m4a
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* gst-play-1.0 --gapless \
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* gapless-sweep_part1_iis.m4a \
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* gapless-sweep_part2_iis.m4a \
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* ]|
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*/
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#ifdef HAVE_CONFIG_H
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#include "config.h"
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#endif
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#endif
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