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webrtc / src / webrtc / 87fc68475fa4614367243b84e88bef5e9590bdb7
commit87fc68475fa4614367243b84e88bef5e9590bdb7[log] [tgz]
authorErik Språng <sprang@webrtc.org>Thu Nov 26 15:26:12 2015
committerErik Språng <sprang@webrtc.org>Thu Nov 26 15:26:25 2015
tree9dd80563a3a54cd5eb57d47b892a36b67e8e0bc5
parent9780212dec7696c9865b3082a1e482dc7cf73807 [diff]
Fix bug in calculation of averge queue time in paced sender.

Also work around a flaw in fake encoder which caused bogus perf
regression in rampup tests.

BUG=560434
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1474533006 .

Cr-Original-Commit-Position: refs/heads/master@{#10811}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: ad113e50d251c95adcf501ed29f8312ad1193a35
4 files changed
tree: 9dd80563a3a54cd5eb57d47b892a36b67e8e0bc5
  1. audio/
  2. base/
  3. build/
  4. call/
  5. common_audio/
  6. common_video/
  7. examples/
  8. libjingle/
  9. modules/
  10. p2p/
  11. sound/
  12. system_wrappers/
  13. test/
  14. tools/
  15. video/
  16. video_engine/
  17. voice_engine/
  18. .gitignore
  19. audio_receive_stream.h
  20. audio_send_stream.h
  21. audio_state.h
  22. BUILD.gn
  23. call.h
  24. codereview.settings
  25. common.gyp
  26. common.h
  27. common_types.cc
  28. common_types.h
  29. config.cc
  30. config.h
  31. engine_configurations.h
  32. frame_callback.h
  33. LICENSE
  34. LICENSE_THIRD_PARTY
  35. OWNERS
  36. PATENTS
  37. PRESUBMIT.py
  38. README.chromium
  39. rtc_unittests.isolate
  40. stream.h
  41. supplement.gypi
  42. transport.h
  43. typedefs.h
  44. video_decoder.h
  45. video_encoder.h
  46. video_engine_tests.isolate
  47. video_frame.h
  48. video_receive_stream.h
  49. video_renderer.h
  50. video_send_stream.h
  51. webrtc.gyp
  52. webrtc_examples.gyp
  53. webrtc_nonparallel_tests.isolate
  54. webrtc_perf_tests.isolate
  55. webrtc_tests.gypi