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webrtc / src / 14137a10641dc7e18d24c1d8bedbf2cf5c0e4a92
commit14137a10641dc7e18d24c1d8bedbf2cf5c0e4a92[log] [tgz]
authorhenrika <henrika@webrtc.org>Thu Sep 19 11:01:29 2019
committerCommit Bot <commit-bot@chromium.org>Thu Sep 19 11:35:10 2019
tree1b08fa823dca9f5bb7b3afc06f53004f85bda27c
parent86873f0cd3754e21818fe53d0bc60c3d48813db3 [diff]
Adds logging of audio sessions status on the recording side in ADM for Android.

Goal is to be able to retrieve more details about possible microphone conflicts in
cases where Init/Start of audio recording fails.

Only supported on Android N and higher.

Also adds new boolean UMA histogram called WebRTC.Audio.SourceMatchesRecordingSession.
Its value is stored after the recording session has been stopped.

Does not affect the media flow or functionality of the ADM. Time to start audio should
not be affected either since the new check and logging takes place on a separate
ExecutorService thread.

See go/webrtc-adm-android for more details and examples.

Bug: webrtc:10971
Change-Id: Ia80c1534e326907a1582824225d5f58caa016922
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150793
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29236}
4 files changed
tree: 1b08fa823dca9f5bb7b3afc06f53004f85bda27c
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  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