Configure RTX at construction of the RTP module

The RTX mode and payload types were being set right after the RTP module
had been created. That kept RTPSender's RTX state mutable, which
requires a lock for every read of the RTX mode.

* Add rtx mode and payload types to RtpRtcpInterface::Configuration
* Apply them when constructing RTPSender.
* RtpVideoSender now passes the RTX configuration at creation.
* Remove SetRtxSendStatus() and SetRtxSendPayloadType() from
  RtpRtcpInterface and ModuleRtpRtcpImpl2.
* Deprecate RTPSender::SetRtxStatus() and SetRtxPayloadType(). Once
  direct users of RTPSender have migrated, they can be removed and the
  RTX state made const.

Invalid configurations still log an error and leave RTX off. Since all
RTX payload types are now applied at once, a single invalid one drops
the whole mapping. This keeps RTX off for a simulcast layer whose codec
has no RTX payload type, as before. Keeping just the RED mapping would
enable RTX without a way to retransmit media packets.

Bug: webrtc:42223727
Change-Id: I34be4f765293599fc2358c4ab52cebe1fcdb98d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507340
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48836}
11 files changed
tree: 141c78a5de459d82b87b8095a0b1d90c41d9f973
  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. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.mdformat
  39. .style.yapf
  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. vpython.toml
  64. vpython.toml.uv.lock
  65. WATCHLISTS
  66. webrtc.gni
  67. webrtc_lib_link_test.cc
  68. 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