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webrtc / src / 9a0a17fb7a4a594825d7fb3c676e58476e892cfc
commit9a0a17fb7a4a594825d7fb3c676e58476e892cfc[log] [tgz]
authorerikvarga@webrtc.org <erikvarga@webrtc.org>Wed Dec 20 14:47:34 2017
committerCommit Bot <commit-bot@chromium.org>Mon Jan 08 15:50:02 2018
treeded494204a960ea0f422c7e9eb4c76289b76eb73
parent6206eef0eed380170b550c15032b388bf4fd109e [diff]
Make it possible to change the amplitude of the pulses generated by PulsedNoiseCapturer.

This adds a SetCapturer function to testing::FakeAudioDevice::PulsedNoiseCapturer
that can be used to update the volume of the generated audio mid-call. It also modifies
CreatePulsedNoiseCapturer to use PulsedNoiseCapturer's type directly so that its new
function is visible for the callers.

Bug: webrtc:8666
Change-Id: I47726e242ccf221f5511e2797b2954ce035ba371
Reviewed-on: https://webrtc-review.googlesource.com/34650
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21521}
3 files changed
tree: ded494204a960ea0f422c7e9eb4c76289b76eb73
  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