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webrtc / src / 1bca65bdc9014fa274fe7aee5c331755c71504bc
commit1bca65bdc9014fa274fe7aee5c331755c71504bc[log] [tgz]
authorSebastian Jansson <srte@webrtc.org>Wed Oct 10 07:58:08 2018
committerCommit Bot <commit-bot@chromium.org>Wed Oct 10 08:28:34 2018
treeeef0d8e958dc71d26294b9063261b52d87db2efa
parent81125f0abacb5b1c2a2c4606caf8682309d6e58f [diff]
Makes RtpSender indicate allocation and feedback status on packets.

Streams that are part of transport feedback are assumed to be part of
allocation. A SetAsPartOfAllocation method is introduced to be used by
media streams that are part of bitrate allocation but not included in
feedback.

This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.

Bug: webrtc:9796
Change-Id: If7ac1ad3e6f3c28b2ada2aef1607a42689d899b2
Reviewed-on: https://webrtc-review.googlesource.com/c/104881
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25079}
7 files changed
tree: eef0d8e958dc71d26294b9063261b52d87db2efa
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. style-guide/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. abseil-in-webrtc.md
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. common_types.h
  37. DEPS
  38. ENG_REVIEW_OWNERS
  39. LICENSE
  40. license_template.txt
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. WATCHLISTS
  52. webrtc.gni
  53. 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 http://www.webrtc.org/native-code/development 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