TPEPlatform/WebRTC B2G/Workweek note

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  1. Audio latency
  2. Audio latency from microphone / network to speaker.
  3. SDP: Negotiation of Generic Image Attributes in the Session Description Protocol
  4. SCTP has flow control, should not test network performance above the protocol layer. Measurement between RTP end pionts will be a good one.
  5. The jitter buffer in GIPS will resample, generate or drop the incoming audio frames.
    • When audio is delayed but not timeout, jitter buffer generates fake audio (according to current received data).
    • When the delayed audio comes and the audio is not timeout, jitter buffer resamples the incoming audio with current received data. That makes the audio sounds a lillte sharp.
    • When the delayed audio comes and audio is timeout, jitter buffer drops the incoming audio.
  6. Linux scheduling
    • Only root can set real-time priority.
    • Task with real-time priority leave CPU only when there is other real-time prioriy task
    • In conclusion, if the process runs as root and there are threads set to real-time priority. Those threads with SCHED_OTHER can be ran only when real-time threads are sleeping.This causes problems. Ex, MSG's thread callback to send video frame to encoder. Just one memory copy and few mutexes, and it costs about 16ms for every call.
  7. ICE: handle dynamic network interface change
  8. Release workload in RTCP process