Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream"

This reverts commit 73f048daf07b157961c43e7dbc9d2c378e6457d8.

Reason for revert: Real culprit fixed here: https://chromium-review.googlesource.com/c/chromium/src/+/4417639

Original change's description:
> Revert "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream"
>
> This reverts commit dd557fdb1e300068c62c870d9dc5273b48c7b79d.
>
> Reason for revert: Looks like the Chromium FYI builders are failing.
>
> Original change's description:
> > [WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream
> >
> > This remove use of MaybeWorkerThread* rtp_transport_queue_ from
> > AudioSendStream.  The worker queue is alwauys assumed ot be used where
> > rtp_transport_queue_ was used.
> >
> > Bug: webrtc:14502
> > Change-Id: Ia516ce7340d712671e0ecb301bba9d66e7216973
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300400
> > Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39816}
>
> Bug: webrtc:14502
> Change-Id: I0547548032756fc579b76b6bb362f576aa06b8f7
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301020
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#39820}

Bug: webrtc:14502
Change-Id: I4db2560de3b21ee0c5c7c579af1891b2c7b2815f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300866
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39828}
4 files changed
tree: 2943f80ed2045705f352fdfe60e8ad0acf243c4c
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  34. .vpython
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. OWNERS_INFRA
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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