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tsdemux: Update TODO
This commit is contained in:
Edward Hervey
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commit
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@ -1,35 +1,30 @@
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tsdemux/tsparse TODO
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--------------------
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* clock for live streams
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In order for playback to happen at the same rate as on the producer,
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we need to estimate the remote clock based on capture time and PCR
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values.
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For this estimation to be as accurate as possible, the capture time
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needs to happen on the sources.
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=> Ensure live sources actually timestamp their buffers
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Once we have accurate timestamps, we can use an algorithm to
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calculate the PCR/local-clock skew.
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=> Use the EPTLA algorithm as used in -good/rtp/rtpmanager/
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gstrtpjitterbuffer
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* Perfomance
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* Bufferlist : Creating/Destroying very small buffers is too
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costly. Switch to pre-/re-allocating outgoing buffers in which we
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copy the data.
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* Adapter : Use gst_adapter_peek()/_flush() instead of constantly
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creating buffers.
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* Seeking
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=> Split out in a separate file/object. It is polluting tsdemux for
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code readability/clarity.
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* Perfomance : Creation/Destruction of buffers is slow
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* => This is due to g_type_instance_create using a dogslow rwlock
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which take up to 50% of gst_adapter_take_buffer()
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=> Bugzilla #585375 (performance and contention problems)
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* mpegtspacketizer
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* offset/timestamp of incoming buffers need to be carried on to the
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sub-buffers in order for several demuxer features to work correctly.
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* Latency
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* Calculate the actual latency instead of returning a fixed
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value. The latency (for live streams) is the difference between the
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currently inputted buffer timestamp (can be stored in the
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packetizer) and the buffer we're pushing out.
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This value should be reported/updated (leave a bit of extra margin
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in addition to the calculated value).
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* mpegtsparser
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* SERIOUS room for improvement performance-wise (see callgrind)
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* SERIOUS room for improvement performance-wise (see callgrind),
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mostly related to performance issues mentionned above.
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* Random-access seeking
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* Do minimal parsing of video headers to detect keyframes and use
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that to compute the keyframe intervals. Use that interval to offset
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the seek position in order to maximize the chance of pushing out the
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requested frames.
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Synchronization, Scheduling and Timestamping
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@ -50,6 +45,9 @@ pay extra attention to the outgoing NEWSEGMENT event and buffer
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timestamps in order to guarantee proper playback and synchronization
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of the stream.
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In the following, 'timestamps' correspond to GStreamer
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buffer/segment values. The mpeg-ts PCR/DTS/PTS values are indicated
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with their actual name.
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1) Live push-based scheduling
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@ -60,26 +58,25 @@ of the stream.
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the outgoing buffer timestamps need to correspond to the incoming
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buffer timestamp values.
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=> A delta, DTS_delta between incoming buffer timestamp and
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DTS/PTS needs to be computed.
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=> mpegtspacketizer keeps track of PCR and input timestamp and
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extrapolates a clock skew using the EPTLA algorithm.
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=> The outgoing buffers will be timestamped with their PTS values
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(overflow corrected) offseted by that initial DTS_delta.
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(overflow corrected) corrected by that calculated clock skew.
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A latency is introduced between the time the buffer containing the
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first bit of a Access Unit is received in the demuxer and the moment
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the demuxer pushed out the buffer corresponding to that Access Unit.
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=> That latency needs to be reported. It corresponds to the
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biggest Access Unit spacing, in this case 1/video-framerate.
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=> That latency needs to be reported.
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According to the ISO/IEC 13818-1:2007 specifications, D.0.1 Timing
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mode, the "coded audio and video that represent sound and pictures
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that are to be presented simultaneously may be separated in time
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within the coded bit stream by ==>as much as one second<=="
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=> The demuxer will therefore report an added latency of 1s to
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handle this interleave.
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=> The algorithm to calculate the latency should take that into
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account.
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2) Non-live push-based scheduling
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@ -97,11 +94,22 @@ of the stream.
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do not have capture timestamps, we need to ensure the first buffer
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we push out corresponds to the base segment start runing time.
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=> A delta between the first DTS to output and the segment start
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position needs to be computed.
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=> The packetizer keeps track of PCR locations and offsets in
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addition to the clock skew (in the case of upstream buffers
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being timestamped, which is the case for HLS).
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=> The demuxer indicates to the packetizer when he sees the
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'beginning' of the program (i.e. the first valid PAT/PMT
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combination). The packetizer will then use that location as
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"timestamp 0", or "reference position/PCR".
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=> The lowest DTS is passed to the packetizer to be converted to
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timestamp. That value is computed in the same way as live
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streams if upstream buffers have timestamps, or will be
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subtracted from the reference PCR.
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=> The outgoing buffers will be timestamped with their PTS values
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(overflow corrected) offseted by that initial delta.
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(overflow corrected) adjusted by the packetizer.
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Latency is reported just as with the live use-case.
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@ -111,37 +119,13 @@ of the stream.
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We do not get a NEWSEGMENT event from upstream, we therefore need to
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compute the outgoing values.
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The base stream/running time corresponds to the DTS of the first
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buffer we will output. The DTS_delta becomes that earliest DTS.
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=> The outgoing values for the newsegment are calculated like for
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the non-live push-based mode when upstream doesn't provide
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timestamp'ed buffers.
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=> FILLME
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=> The outgoing buffer timestamps are timestamped with their PTS
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values (overflow corrected) adjusted by the packetizer.
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X) General notes
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It is assumed that PTS/DTS rollovers are detected and corrected such
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as the outgoing timestamps never rollover. This can be easily
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handled by correcting the DTS_delta when such rollovers are
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detected. The maximum value of a GstClockTimeDiff is almost 3
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centuries, we therefore have enough margin to handle a decent number
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of rollovers.
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The generic equation for calculating outgoing buffer timestamps
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therefore becomes:
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D = DTS_delta, with rollover corrections
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PTS = PTS of the buffer we are going to push out
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TS = Timestamp of the outgoing buffer
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==> TS = PTS + D
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If seeking is handled upstream for push-based cases, whether live or
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not, no extra modification is required.
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If seeking is handled by the demuxer in the non-live push-based
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cases (converting from TIME to BYTES), the demuxer will need to
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set the segment start/time values to the requested seek position.
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The DTS_delta will also have to be recomputed to take into account
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the seek position.
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[0] When talking about live sources, we mean this in the GStreamer
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