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webrtc / src / / 593e32551c7353d6e4ca67da1552874c07a7febb
commit593e32551c7353d6e4ca67da1552874c07a7febb[log] [tgz]
authorSteve Anton <steveanton@webrtc.org>Fri Dec 15 19:44:48 2017
committerCommit Bot <commit-bot@chromium.org>Mon Dec 18 18:55:23 2017
treebaa25f92fc07664cb706d197a6f3747dd2566ac3
parent8ca5a446b9748ad1d3e971c4cbea3ae1f6c2f1f6 [diff]
Change RTCStatsCollector to only access channels from signaling thread

Previously, the RTCStatsCollector needed to ask the voice/video
channel for its transport name in order to generate transport
level stats. That would happen on the networking thread which was
unsafe because the voice/video channel could have disappeared in
the duration of the asynchronous thread hop from the signaling
thread to the networking thread. This changes the networking stats
code to check a saved map that tracks the transport name for each
voice/video channel.

Bug: None
Change-Id: I1f03ba8c0526eaa4419f660f18b8b9da62c3f932
Reviewed-on: https://webrtc-review.googlesource.com/33660
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21332}
4 files changed
tree: baa25f92fc07664cb706d197a6f3747dd2566ac3
  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. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. voice_engine/
  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