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webrtc / src / webrtc / bab6178db8eb1a26e5bac76e0e9115ad80c16eda
commitbab6178db8eb1a26e5bac76e0e9115ad80c16eda[log] [tgz]
authorasapersson@webrtc.org <asapersson@webrtc.org>Mon Feb 23 08:14:07 2015
committerasapersson@webrtc.org <asapersson@webrtc.org>Mon Feb 23 08:14:46 2015
tree17f9a7e95264154db543ccee1b2e19c35f86672a
parent8a734d1fce5a80a43787772434ae3516b0dc8886 [diff]
Fix for flaky test: VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent.

Only compare media bytes sent if number of sent packets in rtcp packet are equal to sent rtp packets.

BUG=4327
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34299004

Cr-Original-Commit-Position: refs/heads/master@{#8454}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 146742164664f9e35ed57575854e14f8011f02a8
1 file changed
tree: 17f9a7e95264154db543ccee1b2e19c35f86672a
  1. base/
  2. build/
  3. common_audio/
  4. common_video/
  5. examples/
  6. libjingle/
  7. modules/
  8. overrides/
  9. p2p/
  10. sound/
  11. system_wrappers/
  12. test/
  13. tools/
  14. video/
  15. video_engine/
  16. voice_engine/
  17. .gitignore
  18. BUILD.gn
  19. call.h
  20. codereview.settings
  21. common.gyp
  22. common.h
  23. common_types.h
  24. config.cc
  25. config.h
  26. engine_configurations.h
  27. experiments.h
  28. frame_callback.h
  29. LICENSE
  30. LICENSE_THIRD_PARTY
  31. OWNERS
  32. PATENTS
  33. PRESUBMIT.py
  34. README.chromium
  35. rtc_unittests.isolate
  36. supplement.gypi
  37. transport.h
  38. typedefs.h
  39. video_decoder.h
  40. video_encoder.h
  41. video_engine_tests.isolate
  42. video_frame.h
  43. video_receive_stream.h
  44. video_renderer.h
  45. video_send_stream.h
  46. webrtc.gyp
  47. webrtc_examples.gyp
  48. webrtc_perf_tests.isolate
  49. webrtc_tests.gypi