Remove use of asyncinvoker from WebRtcVideoSendStream.

This turned out to be a bit complicated, mostly
related to the tests, but here's what's changed:

* No AsyncInvoker (and avoid ClearInternal) in
  WebRtcVideoSendStream (WVSS)
* The reason it was there is due to a "design leak" from
  VideoSourceSinkController/VideoStreamEncoder where the former uses
  locks in all methods and is unaware of a threading model. That design
  affected downstream objects, pushed the need for an async hop into
  WVSS and added a lock.
  A suggestion was made to address this in a follow-up change, here:
  https://webrtc-review.googlesource.com/c/src/+/165684
* All methods in VideoSourceSinkController are now called on a known
  and checked sequence and this CL removes the lock. This also makes
  checking state consistent (i.e. calling a getter twice in a row on the
  same sequence, will always return the same value, avoiding race with
  other threads).
* Handling of reporting state changes from the encoder queue to the
  VSSC, is done by VideoStreamEncoder.
* VideoSendStreamImpl is still instantiated on the incorrect thread [1]
  but has two initialization steps [2]. The second one already runs on
  the right thread. Addressing that TODO [1] is something we should do
  but it has side effects to consider. For the purposes of this CL
  the steps relating to the encoder (setting the sink pointer) have
  been moved to [2].

[1] https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/video/video_send_stream.cc;l=94
[2] https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/video/video_send_stream.cc;drc=f4a9991cce74a37d006438ec0e366313ed33162e;l=115

Bug: webrtc:11222, webrtc:11908
Change-Id: Ie46d46e3a52bbe225951b4bd580ecb8cc9cad873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184508
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32150}
9 files changed
tree: 8531d17f553eda73fcbf2c78eea6d9ad1aa0f75a
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. DEPS
  36. ENG_REVIEW_OWNERS
  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. webrtc_lib_link_test.cc
  52. 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