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webrtc / src / f68df0b95c9f8d2f56baa3c850d1f49e6471db02
commitf68df0b95c9f8d2f56baa3c850d1f49e6471db02[log] [tgz]
authorPhilipp Hancke <phancke@meta.com>Thu Jan 30 18:46:09 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Fri Jan 31 14:41:50 2025
tree28906fdad7c3e192fe12cb994234b2311c6f0679
parentc58a767a23ab620a78fd92aa9d869881ba8afd25 [diff]
Restore primary/rtx payload type assignment logic

Changes the order of payload type assignment to keep the rtx_pt =
primary_pt+1 pattern (even if not required by the specification)

On https://webrtc.github.io/samples/src/content/peerconnection/pc1/ this
changes PTs as follows:

M132: m=video 9 UDP/TLS/RTP/SAVPF 96 97 102 103
104 105 106 107 108 109 127 125 39 40 45 46 98 99 100 101 112 113 114
(121 more lines) mid=1

M134: m=video 9 UDP/TLS/RTP/SAVPF 96 109 99 118 100 119 101 120 103 121
104 122 37 123 40 127 97 114 98 115 107 44 108 (121 more lines) mid=1

M134 with this patch: m=video 9 UDP/TLS/RTP/SAVPF 96 97 107 108 109 114
115 116 117 118 119 120 37 121 40 124 98 99 100 101 127 42 43 (121 more
lines) mid=1

Note that this pushes red and ulpfec into lower range but those codecs
are not widely used and it is possible to get them back into the upper
range e.g. by using setCodecPreferences to disable H264 (where ulpfec is
not supported)

BUG=chromium:391132280

Change-Id: I892f00a2f276728d16c37e8ba5f76d01f6322a7d
No-Iwyu: single include missing, keep it more merge-able
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375847
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43833}
2 files changed
tree: 28906fdad7c3e192fe12cb994234b2311c6f0679
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. OWNERS_INFRA
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. pylintrc_old_style
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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