Implement FrameBuffer3Proxy

This emulates behaviour from frame buffer 2, but does not handle stats.
In contrast to frame buffer 2, all work happens on the same task queue.
FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind
a field trial WebRTC-FrameBuffer3.

This separates frame scheduling behaviour into a few components,

VideoReceiveStreamTimeoutTracker
* Handles the stream timeouts.

FrameDecodeScheduler
* Manages the scheduling and cancelling of frames being sent to the
  decoder.

FrameDecodeTiming
* Handles the timing and ordering of frames to be decoded.

Other changes
* Adds CurrentSize() method to FrameBuffer3
* Move timing to a separate library
* Does a thread check for Receive statistics as this is now
on the worker thread.
* Adds `FlushImmediate` method to RunLoop so that
  video_receive_stream2_unittest can pass when scheduling is happening
  on the worker thread.

Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721
Bug: webrtc:13343
Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35847}
21 files changed
tree: 1d6ab3d9fda7c5c0dd4612ad742e3f52c2d182f9
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .style.yapf
  31. .vpython
  32. .vpython3
  33. AUTHORS
  34. BUILD.gn
  35. CODE_OF_CONDUCT.md
  36. codereview.settings
  37. DEPS
  38. DIR_METADATA
  39. ENG_REVIEW_OWNERS
  40. g3doc.lua
  41. LICENSE
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  45. PATENTS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. WATCHLISTS
  53. webrtc.gni
  54. webrtc_lib_link_test.cc
  55. 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 here 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