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• ffmpeg_mux_init : use strtoll() to parse stream and group indexes

  19 décembre 2023, par James Almer

  ffmpeg_mux_init : use strtoll() to parse stream and group indexes
  Long is 32 bits signed on Windows, and nb_streams,_groups are both unsigned
  
  int. In a realistic scenario this wont make a difference, but it's still
  
  proper.
  Also ensure the parsed string is an integer while at it.
  Signed-off-by : James Almer <jamrial@gmail.com>
  

  • [DH] fftools/ffmpeg_mux_init.c

• h264 via WebRTC latency issue

  18 septembre 2024, par Lucas

  I am trying to send a video stream encoded with h264 (hardware accelerated with nvidia encoder) via WebRTC for low latency display on a browser.

  


  More precisely, I have a thread that encodes an opengl framebuffer at a fixed frame rate, the resulting AVPacket's data (I encode using ffmpeg's C api) is then forwarded via WebRTC to the client (using aiortc)

  


  The problem is that I observe significant delays, that seem to depend on the frame rate I use.
For example, running it locally, I get around 160ms delay when running at 30fps, and around 30ms when encoding at 90fps.

  


  The delay here is the measured time to encode + transmit + decode, and I have the strong impression that the issue happen when presenting the video frame, like the browser is not immediately presenting the frame... (encoding is fast, I would expect the transmission to be also rather fast on a local setup, and decoding seems to be fine as well, as reported by the RTP stats in the browser).

  


  I tried to play with RTP timestamps, but that did not change anything, the only variable that seems to impact the latency is the encoding thread 'frequency'.

  


  Any idea on what could be creating this latency ? Am I missing a parameter ?

  


  Also, here are the codec options I use : (they do not influence the latency that much from what I experimented)

  


  profile = high
preset = llhq  # low latency, high quality
tune = zerolatency
zerolatency = 1
g = 2 * FRAME_PER_SECOND  # key frame every 2s
strict-gop = 1


  


  UPDATE

  


  I have the impression that the jitter buffer on Chrome's side is kind of preventing the rtp packets to be decoded immediately, is that possible ?

  


  UPDATE 2

  


  

  • Using RTP playout-delay header extension slightly reduced the latency.

  


  • Setting playoudDelayHint in browser also seemed to help a bit

  


  


  UPDATE 3

  


  After further investigations, I came to the conclusion that it was not possible to get a lower latency by going through the standard webrtc for video streams, as there is little to no control on the video buffering, which I believe to be responsible of the observed latency.

  


  On a side note, I tried to check how google stadia is doing it, as they seem to use WebRTC as well, but they use some in-house frameworks... (plus Chrome is the only supported browser)

  


• avfilter/framesync : fix shortest with eof_action=pass

  5 juin 2019, par Gyan Doshi

  avfilter/framesync : fix shortest with eof_action=pass
  Shifted check of shortest to after repeatlast,
  
  to ensure shortest=1 is always honoured.
  

  • [DH] libavfilter/framesync.c