gstreamer/gst/mpegtsdemux/TODO

tsdemux/tsparse TODO
--------------------

* Perfomance
  * Bufferlist : Creating/Destroying very small buffers is too
  costly. Switch to pre-/re-allocating outgoing buffers in which we
  copy the data. 
  * Adapter : Use gst_adapter_peek()/_flush() instead of constantly
  creating buffers.

* Latency
  * Calculate the actual latency instead of returning a fixed
  value. The latency (for live streams) is the difference between the
  currently inputted buffer timestamp (can be stored in the
  packetizer) and the buffer we're pushing out.
  This value should be reported/updated (leave a bit of extra margin
  in addition to the calculated value).

* mpegtsparser
  * SERIOUS room for improvement performance-wise (see callgrind),
  mostly related to performance issues mentionned above.

* Random-access seeking
  * Do minimal parsing of video headers to detect keyframes and use
  that to compute the keyframe intervals. Use that interval to offset
  the seek position in order to maximize the chance of pushing out the
  requested frames. 


Synchronization, Scheduling and Timestamping
--------------------------------------------

  A mpeg-ts demuxer can be used in a variety of situations:
  * lives streaming over DVB, UDP, RTP,..
  * play-as-you-download like HTTP Live Streaming or UPNP/DLNA
  * random-access local playback, file, Bluray, ...

  Those use-cases can be categorized in 3 different categories:
  * Push-based scheduling with live sources [0]
  * Push-based scheduling with non-live sources
  * Pull-based scheduling with fast random-access

  Due to the nature of timing within the mpeg-ts format, we need to
pay extra attention to the outgoing NEWSEGMENT event and buffer
timestamps in order to guarantee proper playback and synchronization
of the stream.

  In the following, 'timestamps' correspond to GStreamer
  buffer/segment values. The mpeg-ts PCR/DTS/PTS values are indicated
  with their actual name.

 1) Live push-based scheduling

  The NEWSEGMENT event will be in time format and is forwarded as is,
  and the values are cached locally.

  Since the clock is running when the upstream buffers are captured,
  the outgoing buffer timestamps need to correspond to the incoming
  buffer timestamp values.

    => mpegtspacketizer keeps track of PCR and input timestamp and
       extrapolates a clock skew using the EPTLA algorithm.

    => The outgoing buffers will be timestamped with their PTS values
       (overflow corrected) corrected by that calculated clock skew.

  A latency is introduced between the time the buffer containing the
  first bit of a Access Unit is received in the demuxer and the moment
  the demuxer pushed out the buffer corresponding to that Access Unit.

    => That latency needs to be reported.

  According to the ISO/IEC 13818-1:2007 specifications, D.0.1 Timing
  mode, the "coded audio and video that represent sound and pictures
  that are to be presented simultaneously may be separated in time
  within the coded bit stream by ==>as much as one second<=="

    => The algorithm to calculate the latency should take that into
       account. 


 2) Non-live push-based scheduling

  If the upstream NEWSEGMENT is in time format, the NEWSEGMENT event
  is forwarded as is, and the values are cached locally.

  If upstream does provide a NEWSEGMENT in another format, we need to
  compute one by taking the default values:
    start : 0
    stop  : GST_CLOCK_TIME_NONE
    time  : 0

  Since no prerolling is happening downstream and the incoming buffers
  do not have capture timestamps, we need to ensure the first buffer
  we push out corresponds to the base segment start runing time.

    => The packetizer keeps track of PCR locations and offsets in
       addition to the clock skew (in the case of upstream buffers
       being timestamped, which is the case for HLS).

    => The demuxer indicates to the packetizer when he sees the
       'beginning' of the program (i.e. the first valid PAT/PMT
       combination). The packetizer will then use that location as
       "timestamp 0", or "reference position/PCR".

    => The lowest DTS is passed to the packetizer to be converted to
       timestamp. That value is computed in the same way as live
       streams if upstream buffers have timestamps, or will be
       subtracted from the reference PCR.

    => The outgoing buffers will be timestamped with their PTS values
       (overflow corrected) adjusted by the packetizer.

  Latency is reported just as with the live use-case.


 3) Random access pull-mode

  We do not get a NEWSEGMENT event from upstream, we therefore need to
  compute the outgoing values.

    => The outgoing values for the newsegment are calculated like for
       the non-live push-based mode when upstream doesn't provide
       timestamp'ed buffers.

    => The outgoing buffer timestamps are timestamped with their PTS
       values (overflow corrected) adjusted by the packetizer.



[0] When talking about live sources, we mean this in the GStreamer
definition of live sources, which is to say sources where if we miss
the capture, we will miss the data to be captured. Sources which do
internal buffering (like TCP connections or file descriptors) are
*NOT* live sources.