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webrtc / src / d7719ab4d8a016ee2ea4eab90b3deb079d2bc152
commitd7719ab4d8a016ee2ea4eab90b3deb079d2bc152[log] [tgz]
authorHelmer Nylen <helmern@google.com>Mon Jun 09 12:18:54 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Jun 11 16:01:32 2025
treee55d504a178247300bb51182db04a070d9c1e243
parentcc1bc9861ce00e5a0f31d5fb370b82f502587ee8 [diff]
Add support for setting CSRCs on audio and video senders

This is a modified version of
https://webrtc-review.googlesource.com/c/src/+/392940 to avoid breaking
downstream dependencies.

With this change, CSRCs can be added to video packets sent via
RTPSenderVideo::SendEncodedImage. This is implemented by keeping a list
of CSRCs in the calling class RtpVideoSender, which is included in all
calls to SendEncodedImage. Similarly, a list of CSRCs for audio packets
is kept on ChannelSend and added to calls to SendRtpAudio. CSRCs are
also propagated to frame transformers for audio and video.

Bug: b/410811496
Change-Id: I728934f8c190120211672e2d6dc5940bc8f83838
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Helmer Nylén <helmern@google.com>
Cr-Commit-Position: refs/heads/main@{#44907}
19 files changed
tree: e55d504a178247300bb51182db04a070d9c1e243
  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. .rustfmt.toml
  34. .style.yapf
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. OWNERS_INFRA
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  53. pylintrc_old_style
  54. README.chromium
  55. README.md
  56. WATCHLISTS
  57. webrtc.gni
  58. webrtc_lib_link_test.cc
  59. 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