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webrtc / src / webrtc / 708d0cd42cc7d99a4c2564824e1c3e8f0e39e25c
commit708d0cd42cc7d99a4c2564824e1c3e8f0e39e25c[log] [tgz]
authordanilchap <danilchap@webrtc.org>Fri Jul 28 15:56:04 2017
committerCommit Bot <commit-bot@chromium.org>Fri Jul 28 15:56:04 2017
treed0546fc7d63e6062d40e1dddd887a44caec4f3ad
parent3b5bf18b450478e6ea76309c966f6fe9bf663d68 [diff]
Simplify FakeReceiveStatistics in video send stream tests

Rtcp sender now take smaller interface making it possible to simplify the fake

BUG=webrtc:8016

Review-Url: https://codereview.webrtc.org/2984283002
Cr-Original-Commit-Position: refs/heads/master@{#19181}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 901b2df431d9834910be5747df69960ce9830b70
1 file changed
tree: d0546fc7d63e6062d40e1dddd887a44caec4f3ad
  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. rtc_base/
  15. rtc_tools/
  16. sdk/
  17. stats/
  18. system_wrappers/
  19. test/
  20. video/
  21. voice_engine/
  22. .gitignore
  23. BUILD.gn
  24. codereview.settings
  25. common_types.cc
  26. common_types.h
  27. config.cc
  28. config.h
  29. config_unittest.cc
  30. DEPS
  31. LICENSE
  32. LICENSE_THIRD_PARTY
  33. OWNERS
  34. PATENTS
  35. PRESUBMIT.py
  36. README.chromium
  37. typedefs.h
  38. video_receive_stream.h
  39. video_send_stream.h
  40. webrtc.gni
  41. whitespace.txt