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webrtc / src / refs/heads/main
commit7d3af2ee95f292d148cf606f17a8129f8e7177b0[log] [tgz]
authorRasmus Brandt <brandtr@webrtc.org>Wed Jan 14 11:43:37 2026
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Jan 14 14:37:59 2026
tree83060f32e8192d34897c206fd1892c6bc8c2ab64
parenta917e7ab9a7805d91410bde0e00ed206bf1391f5 [diff]
Log RTX OSN by decentralizing packet event logging to streams.

This change decentralizes the logging of `RtcEventRtpPacketIncoming` to
the streams, to allow logging of RTX original sequence number (OSN) for
valid RTX streams.

Why not keep logging centralized? We tried that in
https://webrtc-review.googlesource.com/c/src/+/434300, but that solution
turned out to be non-ideal: 1) it is complicated and hacky [duplicating
SSRC demuxing and RTX decapsulation], 2) it is not safe [we cannot
guarantee that the first two bytes are indeed _always_ RTX OSNs]

This decentralized logging solution is neither complicated/hacky nor
unsafe. It does however have the drawback that a single log call is
replaced by one of five calls (one per stream type + undemuxable). So
there is a risk that future refactorings might break the logging. We
protect against that with a set of tests.

Bug: b/305890678, b/423646186
Change-Id: Ic97a954d277b7088886bd95230e9b93ebd4f41c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432440
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46664}
17 files changed
tree: 83060f32e8192d34897c206fd1892c6bc8c2ab64
  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. WATCHLISTS
  60. webrtc.gni
  61. webrtc_lib_link_test.cc
  62. 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