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webrtc / src / webrtc / 0d8805b69cbefee54800f73cedbdaaa105cf8254
commit0d8805b69cbefee54800f73cedbdaaa105cf8254[log] [tgz]
authorbrandtr <brandtr@webrtc.org>Tue May 30 09:32:12 2017
committerCommit Bot <commit-bot@chromium.org>Tue May 30 09:32:12 2017
tree452182a6a6e5d4e18cc446243ce66569d6c13722
parent7fb576fba9d08965a6ecb6c471a7c07b0aed77a5 [diff]
Persist RTP state for FlexFEC.

Before this CL, the RTP state would be re-randomized after a
recreation of VideoSendStream. That might lead to us sending
a non-compliant RTP stream, which is avoided after the
changes in this CL.

BUG=webrtc:5654
TBR=pbos@webrtc.org  # Trivial change to fuzzer.

Review-Url: https://codereview.webrtc.org/2912713002
Cr-Original-Commit-Position: refs/heads/master@{#18322}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 48d21a23c6663dc1d07c8a05f2922ad14bd12c8a
7 files changed
tree: 452182a6a6e5d4e18cc446243ce66569d6c13722
  1. api/
  2. audio/
  3. base/
  4. call/
  5. common_audio/
  6. common_video/
  7. examples/
  8. logging/
  9. media/
  10. modules/
  11. ortc/
  12. p2p/
  13. pc/
  14. sdk/
  15. stats/
  16. system_wrappers/
  17. test/
  18. tools/
  19. video/
  20. voice_engine/
  21. .gitignore
  22. BUILD.gn
  23. call.h
  24. codereview.settings
  25. common_types.cc
  26. common_types.h
  27. config.cc
  28. config.h
  29. DEPS
  30. LICENSE
  31. LICENSE_THIRD_PARTY
  32. OWNERS
  33. PATENTS
  34. PRESUBMIT.py
  35. README.chromium
  36. typedefs.h
  37. video_frame.h
  38. video_receive_stream.h
  39. video_send_stream.h
  40. webrtc.gni
  41. whitespace.txt