[cherry pick] AGC2: retuning and large refactoring

- Bug fix: the desired initial gain quickly dropped to 0 dB hence
  starting a call with a too low level
- New tuning to make AGC2 more robust to VAD mistakes
- Smarter max gain increase speed: to deal with an increased threshold
  of adjacent speech frames, the gain applier temporarily allows a
  faster gain increase to deal with a longer time spent waiting for
  enough speech frames in a row to be observed
- Saturation protector isolated from `AdaptiveModeLevelEstimator` to
  simplify the unit tests for the latter (non bit-exact change)
- AGC2 adaptive digital config: unnecessary params deprecated
- Code readability improvements
- Data dumps clean-up and better naming

(cherry picked from commit 980c4601e188a80bb435edf0c3e345899cd18c59)

Bug: chromium:1199259,webrtc:7494
Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#33736}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215782
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/branch-heads/4472@{#7}
Cr-Branched-From: 3e0c60ba4ef28a9f26fe991e5eec3150402c7dd3-refs/heads/master@{#33644}
29 files changed
tree: 95c3aa0bb772a625ec729b22ebb3278f3bcbb70a
  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
  34. BUILD.gn
  36. codereview.settings
  37. DEPS
  40. g3doc.lua
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. style-guide.md
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. whitespace.txt

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.


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