Introduce SetSsrcTask and batch SetSsrc in ApplyLocalDescription

The existing SetSsrc implementation in RtpSender required multiple
synchronous blocking calls to the worker thread. In scenarios with many
transceivers this was a significant performance bottleneck.

This change introduces SetSsrcTask, which allows SetSsrc operations to
be decomposed into asynchronous tasks. By using ScopedOperationsBatcher,
these tasks are now batched and executed efficiently on the worker
thread. The existing SetSsrc will be for PlanB only.

Key modifications:
* Added SetSsrcTask to the RtpSenderInternal interface to return a
  batchable task containing both worker and signaling thread logic.
* Updated SdpOfferAnswer::ApplyLocalDescription to use
  ScopedOperationsBatcher, reducing the number of thread context
  switches.
* Moved media channel configurations (frame encryptors,
  transformers, and selectors) into the batched worker task.
* Marked the legacy SetSsrc method as Plan B only.

Bug: webrtc:42222117
Change-Id: I49cfbeef4764d8423d678eec4f5c8017374e3187
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/465020
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47507}
10 files changed
tree: f2c7cbdaa407fdccc5bf4ae97c5e9165f52372d0
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. sdk/
  25. stats/
  26. system_wrappers/
  27. test/
  28. tools_webrtc/
  29. video/
  30. .clang-format
  31. .clang-tidy
  32. .git-blame-ignore-revs
  33. .gitignore
  34. .gn
  35. .mailmap
  36. .rustfmt.toml
  37. .style.mdformat
  38. .style.yapf
  39. .vpython3
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. WATCHLISTS
  64. webrtc.gni
  65. webrtc_lib_link_test.cc
  66. 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