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webrtc / src / webrtc / 963832becfa34440de40171bb71330fada9c112a
commit963832becfa34440de40171bb71330fada9c112a[log] [tgz]
authorhenrika <henrika@webrtc.org>Tue Nov 17 13:54:48 2015
committerhenrika <henrika@webrtc.org>Tue Nov 17 13:54:58 2015
treef0f1f172eee9dfcd5eb78d11e74b9b6f97af5fd0
parent3c173a9bb653f64b71fd3db71c3d59bc63ac7b4e [diff]
Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant
since it is recommended for VoIP apps.

BUG=b/23356406
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1418483004 .

Cr-Original-Commit-Position: refs/heads/master@{#10673}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 5a71f03f8bb0bd1691729717ab7ef22b1c3f94a0
1 file changed
tree: f0f1f172eee9dfcd5eb78d11e74b9b6f97af5fd0
  1. audio/
  2. base/
  3. build/
  4. call/
  5. common_audio/
  6. common_video/
  7. examples/
  8. libjingle/
  9. modules/
  10. p2p/
  11. sound/
  12. system_wrappers/
  13. test/
  14. tools/
  15. video/
  16. video_engine/
  17. voice_engine/
  18. .gitignore
  19. audio_receive_stream.h
  20. audio_send_stream.h
  21. audio_state.h
  22. BUILD.gn
  23. call.h
  24. codereview.settings
  25. common.gyp
  26. common.h
  27. common_types.cc
  28. common_types.h
  29. config.cc
  30. config.h
  31. engine_configurations.h
  32. frame_callback.h
  33. LICENSE
  34. LICENSE_THIRD_PARTY
  35. OWNERS
  36. PATENTS
  37. PRESUBMIT.py
  38. README.chromium
  39. rtc_unittests.isolate
  40. stream.h
  41. supplement.gypi
  42. transport.h
  43. typedefs.h
  44. video_decoder.h
  45. video_encoder.h
  46. video_engine_tests.isolate
  47. video_frame.h
  48. video_receive_stream.h
  49. video_renderer.h
  50. video_send_stream.h
  51. webrtc.gyp
  52. webrtc_examples.gyp
  53. webrtc_nonparallel_tests.isolate
  54. webrtc_perf_tests.isolate
  55. webrtc_tests.gypi