Google Git
Sign in
webrtc / src / 42bf2c670caa67ba5853267ac1d57adbf7eac4ea
commit42bf2c670caa67ba5853267ac1d57adbf7eac4ea[log] [tgz]
authorSergey Silkin <ssilkin@webrtc.org>Tue Jan 18 09:34:25 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Tue Jan 18 14:36:33 2022
tree94ffff1d5ff4673eacd18dcd5145a867ce1e5f4a
parent9defabbac50dce810a0f927c49b8b8947b743d6c [diff]
Put current send codec to front of codecs list in RTP sender parameters

WebRTC can switch encoder on-fly when encoder fails or by request from
encoder selector. Putting the current send codec to the front of the
codecs list provides a simple way for apps to know what is actually
used without retrieving stats.

Bug: webrtc:13572
Change-Id: Iaaa5f7ad8667f59016dc92bff9e9a57a7425ef44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246500
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35723}
2 files changed
tree: 94ffff1d5ff4673eacd18dcd5145a867ce1e5f4a
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .style.yapf
  31. .vpython
  32. .vpython3
  33. AUTHORS
  34. BUILD.gn
  35. CODE_OF_CONDUCT.md
  36. codereview.settings
  37. DEPS
  38. DIR_METADATA
  39. ENG_REVIEW_OWNERS
  40. g3doc.lua
  41. LICENSE
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  45. PATENTS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. WATCHLISTS
  53. webrtc.gni
  54. webrtc_lib_link_test.cc
  55. 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