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webrtc / src.git / 1ad210876f233e38c4f210e5707be1fee85908e9
commit1ad210876f233e38c4f210e5707be1fee85908e9[log] [tgz]
authorSergey Silkin <ssilkin@webrtc.org>Sun Apr 08 21:14:33 2018
committerCommit Bot <commit-bot@chromium.org>Mon Apr 09 10:01:13 2018
tree478ab10f90665768f2ea7c7c6aad4b87a8d2bf7b
parentbb19fcf3bdbbf3752bed2290df0d43475fb2eb08 [diff]
Revert "Storing frame if encoder is paused."

This reverts commit dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8.

Reason for revert: breaks downstream projects

Original change's description:
> Storing frame if encoder is paused.
> 
> Adds a pending frame to VideoStreamEncoder that is used to store frames
> that are not sent because the encoder is paused. If the encoder is
> resumed within 200 ms, the pending frame will be encoded and sent. This
> ensures that resuming a stream instantly starts sending frames if it is
> possible.
> 
> This also protects against a race between submitting the first frame
> and enabling the encoder that caused flakiness in end to end tests
> when using the task queue based congestion controller.
> 
> Bug: webrtc:8415
> Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
> Reviewed-on: https://webrtc-review.googlesource.com/67141
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22781}

TBR=sprang@webrtc.org,srte@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8415
Change-Id: I4449eca65a64e2bc2fb25d866e5775e9a085cee9
Reviewed-on: https://webrtc-review.googlesource.com/68280
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22788}
4 files changed
tree: 478ab10f90665768f2ea7c7c6aad4b87a8d2bf7b
  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.cc
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. typedefs.h
  52. WATCHLISTS
  53. webrtc.gni
  54. 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