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webrtc / src / / 6a18f06bd09fdeaad6e6e00d098fc50ab946ed40
commit6a18f06bd09fdeaad6e6e00d098fc50ab946ed40[log] [tgz]
authorAlessio Bazzica <alessiob@webrtc.org>Wed Oct 26 13:23:46 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Oct 26 13:29:27 2022
tree5dc15da55951d8f1d06358073eeb10b5659e25ed
parent48a0c1a86040d70bf187a53f9a8e53c7dc95cd7d [diff]
Revert "APM: rename `AnalogGainStatsReporter` to `InputVolumeStatsReporter`"

This reverts commit b5319fabeeda4ffbf58f28f4ee3d5c7c3868fb3b.

Reason for revert: audioproc_f crash 

Original change's description:
> APM: rename `AnalogGainStatsReporter` to `InputVolumeStatsReporter`
>
> Adopt the new naming convention, which replaces "analog gain" and
> "mic level" with "input volume", in the input volume stats reporter.
>
> Bug: webrtc:7494
> Change-Id: Ia24876151f51dd1dcc4e4f9db56c64d11ae3b442
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279840
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38467}

Bug: webrtc:7494
Change-Id: Ia943a57c93fc77eb8450fab17961e60774e10f02
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280600
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38478}
5 files changed
tree: 5dc15da55951d8f1d06358073eeb10b5659e25ed
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  34. .vpython
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. g3doc.lua
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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 here 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