Google Git
Sign in
webrtc / src / 8bc865bc70ad4620d89cae7e1824917837fd4c67
commit8bc865bc70ad4620d89cae7e1824917837fd4c67[log] [tgz]
authorAlessio Bazzica <alessiob@webrtc.org>Wed Apr 14 14:17:09 2021
committerCommit Bot <commit-bot@chromium.org>Tue Apr 20 20:27:58 2021
treeda4802170191e7ae2778b02756542ffa7b99caea
parentbcdf8f09214a414a76f8549612834b576c8214c6 [diff]
[cherry pick] AGC2 lightweight noise floor estimator

The current noise level estimator has a bug due to which the estimated
level decays to the lower bound in a few seconds when speech is observed.
Instead of fixing the current implementation, which is based on a
stationarity classifier, an alternative, lightweight, noise floor
estimator has been added and tuned for AGC2.

Tested on several AEC dumps including HW mute, music and fast talking.

(cherry picked from commit 61982a7f2d77a03000983b563e840b16a2211e02)
NOTRY=true

Bug: chromium:1199259,webrtc:7494
Change-Id: Iae4cff9fc955a716878f830957e893cd5bc59446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214133
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#33733}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215781
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/branch-heads/4472@{#6}
Cr-Branched-From: 3e0c60ba4ef28a9f26fe991e5eec3150402c7dd3-refs/heads/master@{#33644}
10 files changed
tree: da4802170191e7ae2778b02756542ffa7b99caea
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. style-guide/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. abseil-in-webrtc.md
  33. AUTHORS
  34. BUILD.gn
  35. CODE_OF_CONDUCT.md
  36. codereview.settings
  37. DEPS
  38. DIR_METADATA
  39. ENG_REVIEW_OWNERS
  40. g3doc.lua
  41. LICENSE
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  45. PATENTS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. style-guide.md
  53. WATCHLISTS
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. 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