Add encoder switch request callback to VideoMediaSendChannel

Introduce a callback mechanism to intercept and handle encoder switch
requests within the video media engine. This makes it so that codec
switch/fallback requests first go through the the RtpTransceiver on the
signaling thread. That is consistent with other parameter related
changes and allows the transceiver to update the sender object(s) before
the codec change is applied.

Specifically, the changes include:
* Define EncoderSwitchRequestCallback and EncoderSwitchRequestAction
  in VideoMediaSendChannelInterface.
* Updating WebRtcVideoSendChannel to optionally use the callback
  instead of posting directly to the worker thread when an encoder
  switch is requested.
* Implementing GetEncoderSwitchRequestCallback in RtpTransceiver
  to ensure that when a switch occurs, cached RTP parameters on the
  signaling thread are cleared before the switch action is executed
  on the worker thread.
* Updating RtpSender to support clearing cached parameters via
  std::optional.
* Adding RTC_DCHECK to SafeTask and SafeInvocable to catch if
  the provided safety flag is invalid, early.

Bug: b/478050997
Change-Id: Id81a219a0f3dab6a167248c21513723504fa6e79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47148}
14 files changed
tree: 534ff42fe08de5a348ead26411b0646cf2efb183
  1. agents/
  2. api/
  3. audio/
  4. build_overrides/
  5. call/
  6. common_audio/
  7. common_video/
  8. data/
  9. docs/
  10. examples/
  11. experiments/
  12. g3doc/
  13. infra/
  14. logging/
  15. media/
  16. modules/
  17. net/
  18. p2p/
  19. pc/
  20. resources/
  21. rtc_base/
  22. rtc_tools/
  23. sdk/
  24. stats/
  25. system_wrappers/
  26. test/
  27. tools_webrtc/
  28. video/
  29. .clang-format
  30. .clang-tidy
  31. .git-blame-ignore-revs
  32. .gitignore
  33. .gn
  34. .mailmap
  35. .rustfmt.toml
  36. .style.yapf
  37. .vpython3
  38. AUTHORS
  39. BUILD.gn
  40. CODE_OF_CONDUCT.md
  41. codereview.settings
  42. DEPS
  43. DIR_METADATA
  44. ENG_REVIEW_OWNERS
  45. GEMINI.md
  46. LICENSE
  47. license_template.txt
  48. native-api.md
  49. OWNERS
  50. OWNERS_INFRA
  51. PATENTS
  52. PRESUBMIT.py
  53. presubmit_test.py
  54. presubmit_test_mocks.py
  55. pylintrc
  56. pylintrc_old_style
  57. README.chromium
  58. README.md
  59. unsafe_buffers_paths.txt
  60. WATCHLISTS
  61. webrtc.gni
  62. webrtc_lib_link_test.cc
  63. 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