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webrtc / src / 1788dcb506c5a28a10db07d5a9139a29ce72f350
commit1788dcb506c5a28a10db07d5a9139a29ce72f350[log] [tgz]
authorNiels Moller <nisse@webrtc.org>Thu Aug 09 06:18:57 2018
committerCommit Bot <commit-bot@chromium.org>Thu Aug 09 06:19:14 2018
treea338feecc1c893056b5641a6bdbf2ad11c8e736b
parentb2db53998dc4226d41e6c101bf24ff38a8188a32 [diff]
Revert "Refactor RtpVideoStreamReceiver without RtpReceiver."

This reverts commit 0b9e01d605a174a52635626c885738a222abff46.

Reason for revert: Appears to breaks AV sync, failing perftests: 
CallPerfTest.PlaysOutAudioAndVideoInSyncWithVideoNtpDrift
CallPerfTest.PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift
CallPerfTest.PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift



Original change's description:
> Refactor RtpVideoStreamReceiver without RtpReceiver.
> 
> Bug: webrtc:7135
> Change-Id: Iabf3330e579b892efc160683f9f90efbf6ff9a40
> Reviewed-on: https://webrtc-review.googlesource.com/92398
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24232}

TBR=danilchap@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,sprang@webrtc.org

Change-Id: I70a1dcb519c51937e35e04ac82b2ab495bec0a3d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/93260
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24235}
5 files changed
tree: a338feecc1c893056b5641a6bdbf2ad11c8e736b
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. style-guide/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. whitespace.txt
README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info