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webrtc / src / dd3768ef7266e0e4840e883a0f652e3c75887cad
commitdd3768ef7266e0e4840e883a0f652e3c75887cad[log] [tgz]
authorHelmer Nylen <helmern@google.com>Wed Jun 04 07:39:05 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Jun 04 09:57:46 2025
treeca2ee4f34d4a0b193e9dff79afc2295ba3d07cf7
parent8b93e6dcd801175d3611e4d3ab6564c848f7fef4 [diff]
Add support for setting CSRCs on audio and video senders

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.

This CL is part of a chain, with the next being
https://webrtc-review.googlesource.com/c/src/+/392961. Ultimately, the
point is to support setting the CSRC list via RtpEncodingParameters.
This is done in https://webrtc-review.googlesource.com/c/src/+/392980.

Bug: b/410811496
Change-Id: I2b9c430c6b19b423f2f29cf8e81b04ad04c2b915
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392940
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@{#44824}
19 files changed
tree: ca2ee4f34d4a0b193e9dff79afc2295ba3d07cf7
  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