Move payload type demuxing management to RtpTransport

Shift the responsibility for enabling and disabling payload type
demuxing from the Channel and Transceiver layers to the RtpTransport and
RtpDemuxer layer. This centralizes demuxing logic within the transport
layer.

Moving this logic to RtpTransport allows the SdpOfferAnswerHandler to
apply state changes directly to the transport and not need to go through
the Channel classes.

Key modifications:
- Remove SetPayloadTypeDemuxingEnabled from BaseChannel,
  ChannelInterface, and RtpTransceiver.
- Implement SetActivePayloadTypeDemuxing in RtpTransport to manage
  demuxer criteria filtering and coordinate with RtpDemuxer.
- Update SdpOfferAnswerHandler to apply demuxing state changes to
  RtpTransport instances directly.
- Ensure ResetUnsignaledRecvStream executes on the worker thread
  after applying transport state, to clear legacy streams when
  demuxing is disabled.

Bug: webrtc:42222117
Change-Id: If02194417dc8e4c3dcf5fda5e969aa81116004e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456720
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47318}
17 files changed
tree: 6076016b30fd93e7183b83bdd7d7fed8d3608506
  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.yapf
  38. .vpython3
  39. AUTHORS
  40. BUILD.gn
  41. CODE_OF_CONDUCT.md
  42. codereview.settings
  43. DEPS
  44. DIR_METADATA
  45. ENG_REVIEW_OWNERS
  46. GEMINI.md
  47. LICENSE
  48. license_template.txt
  49. native-api.md
  50. OWNERS
  51. OWNERS_INFRA
  52. PATENTS
  53. PRESUBMIT.py
  54. presubmit_test.py
  55. presubmit_test_mocks.py
  56. pylintrc
  57. pylintrc_old_style
  58. README.chromium
  59. README.md
  60. unsafe_buffers_paths.txt
  61. WATCHLISTS
  62. webrtc.gni
  63. webrtc_lib_link_test.cc
  64. 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