Google Git
Sign in
webrtc / src / webrtc / 5d1102317e766a95c2514f7e617f9980b53cf4c2
commit5d1102317e766a95c2514f7e617f9980b53cf4c2[log] [tgz]
authorilnik <ilnik@webrtc.org>Thu Feb 23 13:07:56 2017
committerCommit bot <commit-bot@chromium.org>Thu Feb 23 13:07:56 2017
treec9acd5900e5f0a6f0c320e99713b27818aef7ffa
parent9b4df223d5f9f108f1006009d5daa778df1ce53a [diff]
Fixed Full stack tests to correctly process selected TL and SL while
calculating frame sizes. Added actual_bitrate metric which also accounts
for TL and SL info. Metric encoded_frame_size calculation is cleaned up. Perf alerts should be ignored.

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2709483009
Cr-Original-Commit-Position: refs/heads/master@{#16800}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 1e7732c3d9c1b8d4d3fa7f7c2c94ba383c50a1e5
1 file changed
tree: c9acd5900e5f0a6f0c320e99713b27818aef7ffa
  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. no_op_function.cc
  33. OWNERS
  34. PATENTS
  35. PRESUBMIT.py
  36. README.chromium
  37. typedefs.h
  38. video_decoder.h
  39. video_encoder.h
  40. video_frame.h
  41. video_receive_stream.h
  42. video_send_stream.h
  43. webrtc.gni
  44. whitespace.txt