Batch SetupMediaChannel blocking calls in rtp receiver classes.

The current implementation of media channel setup triggers individual
blocking calls to the worker thread for every transceiver during state
updates. When applying remote descriptions with multiple transceivers,
this results in frequent thread context switching and increased
signaling thread latency.

Modify RtpReceiverInternal and its implementations to return a deferred
task using absl::AnyInvocable instead of executing the setup
immediately. This allows the SdpOfferAnswerHandler to collect setup
requirements across all transceivers and execute them in a single
batched blocking call on the worker thread.

Key modifications:
- Update SetupMediaChannel and SetupUnsignaledMediaChannel to return
  a callable task.
- Refactor AudioRtpReceiver and VideoRtpReceiver to package their
  worker-thread logic into returnable objects.
- Implement task collection and batch execution logic within the
  SdpOfferAnswerHandler.
- Update associated mocks and unit tests to accommodate the new
  asynchronous execution pattern.

(using No-try because of `webrtc_linux_chromium` timeout issues)

Bug: webrtc:42222804
No-try: true
Change-Id: I4b4d114e3f750fd41eaa4ed29e5c8ee6976eca39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/447063
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46865}
12 files changed
tree: b53ddbf25c4979e70766982df0eecfa712c0f24a
  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