Fix unsignalled ssrc race in WebRtcVideoChannel.

BaseChannel adds and removes receive streams on the worker thread
(UpdateRemoteStreams_w) and then posts a task to the network thread to
update the demuxer criteria. Until this happens, OnRtpPacket() keeps
forwarding "recently removed" ssrc packets to the WebRtcVideoChannel.
Furthermore WebRtcVideoChannel::OnPacketReceived() posts task from the
network thread to the worker thread, so even if the demuxer criteria was
instantly updated we would still have an issue of in-flight packets for
old ssrcs arriving late on the worker thread inside WebRtcVideoChannel.

The wrong ssrc could also arrive when the demuxer goes from forwarding
all packets to a single m= section to forwarding to different m=
sections. In this case we get packets with an ssrc for a recently
created m= section and the ssrc was never intended for our channel.

This is a problem because when WebRtcVideoChannel sees an unknown ssrc
it treats it as an unsignalled stream, creating and destroying default
streams which can be very expensive and introduce large delays when lots
of packets are queued up.

This CL addresses the issue with callbacks for when a demuxer criteria
update is pending and when it has completed. During this window of time,
WebRtcVideoChannel will drop packets for unknown ssrcs.

This approach fixes the race without introducing any new locks and
packets belonging to ssrcs that were not removed continue to be
forwarded even if a demuxer criteria update is pending. This should make
a=inactive for 50p receive streams a glitch-free experience.

Bug: webrtc:12258, chromium:1069603
Change-Id: I30d85f53d84e7eddf7d21380fb608631863aad21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214964
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33757}
15 files changed
tree: 648b9707a101b8ffc96d44964f2f415baa9806dd
  1. .clang-format
  2. .git-blame-ignore-revs
  3. .gitignore
  4. .gn
  5. .vpython
  6. AUTHORS
  7. BUILD.gn
  8. CODE_OF_CONDUCT.md
  9. DEPS
  10. DIR_METADATA
  11. ENG_REVIEW_OWNERS
  12. LICENSE
  13. OWNERS
  14. PATENTS
  15. PRESUBMIT.py
  16. README.chromium
  17. README.md
  18. WATCHLISTS
  19. abseil-in-webrtc.md
  20. api/
  21. audio/
  22. build_overrides/
  23. call/
  24. codereview.settings
  25. common_audio/
  26. common_video/
  27. data/
  28. docs/
  29. examples/
  30. g3doc.lua
  31. g3doc/
  32. license_template.txt
  33. logging/
  34. media/
  35. modules/
  36. native-api.md
  37. net/
  38. p2p/
  39. pc/
  40. presubmit_test.py
  41. presubmit_test_mocks.py
  42. pylintrc
  43. resources/
  44. rtc_base/
  45. rtc_tools/
  46. sdk/
  47. stats/
  48. style-guide.md
  49. style-guide/
  50. system_wrappers/
  51. test/
  52. tools_webrtc/
  53. video/
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. whitespace.txt
README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info