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webrtc / src / / 8b60e8bc34deb1a9f7fee04d153956e9663f881d
commit8b60e8bc34deb1a9f7fee04d153956e9663f881d[log] [tgz]
authorElad Alon <eladalon@webrtc.org>Mon Apr 08 12:14:05 2019
committerCommit Bot <commit-bot@chromium.org>Mon Apr 08 14:29:38 2019
tree35c970c05a9a3d620c558f309d8648da4c20e9ba
parent0cfa4cba5cae5e942f5d8e0e4e93b94982d0bfc3 [diff]
Give VideoSendStreamImpl access to RTP timestamps

When a LossNotification RTCP message is received, the sequence numbers
it refers to must be converted to timestamps before passing the message
down to the encoder. This CL gives VideoSendStreamImpl access to that
information via VideoSendStreamImpl::rtp_video_sender_.

TBR=sprang@webrtc.org

Bug: webrtc:10501
Change-Id: If207f0b6d2fb344da35b525cc104e8ba5cc614ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131323
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27489}
6 files changed
tree: 35c970c05a9a3d620c558f309d8648da4c20e9ba
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. WATCHLISTS
  51. webrtc.gni
  52. 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