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webrtc / src.git / e6256055e7db560da560829a3e249898f9685d1f
commite6256055e7db560da560829a3e249898f9685d1f[log] [tgz]
authorSebastian Jansson <srte@webrtc.org>Fri May 04 12:08:15 2018
committerCommit Bot <commit-bot@chromium.org>Fri May 04 12:44:44 2018
treea98bab558c35abefdb483b1d3895186d0293d083
parent9d96e923169dc237b056f3e2b432a6759155bf2d [diff]
Moving task queue from Call to transport controller.

Moving ownership of worker task queue in Call to
RtpTransportControllerSend. This CL also ensures that the task queue
is not destroyed until the process thread running
SendSideCongestionController is stopped.

The worker queue should be owned by RtpTransportControllerSend since
it is mainly used for rtp and transport related tasks such as bitrate
allocation and signaling network state.

Bug: webrtc:9232
Change-Id: I211edf1a3b9f9b2572875d5584cb788cb2449ef9
Reviewed-on: https://webrtc-review.googlesource.com/63023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23119}
6 files changed
tree: a98bab558c35abefdb483b1d3895186d0293d083
  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. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  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. typedefs.h
  52. WATCHLISTS
  53. webrtc.gni
  54. 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