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webrtc / src / 5da581b564ee42ea0202ce74dd93c6073d83435e
commit5da581b564ee42ea0202ce74dd93c6073d83435e[log] [tgz]
authorAlessio Bazzica <alessiob@webrtc.org>Tue Sep 28 14:28:26 2021
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Tue Sep 28 16:52:16 2021
tree4a3a6b9bb7ac15c56d9ab123a4487fe02e6f8c1d
parent355495a0f280d4099cd9a348b6bcd2a06c45637b [diff]
AGC2: use only one headroom parameter

Instead of using two different headroom parameters, namely
`kHeadroomDbfs` and `kSaturationProtectorExtraHeadroomDb`, only use
the former that now also accounts for the deleted one - i.e., it equals
the sum of the two headrooms. In this way, tuning AGC2 will be easier.

This CL does *not* change the behavior of the AGC2 adaptive digital
controller - bitexactness verified with audioproc_f on a collection of
AEC dumps and Wav files (42 recordings in total).

The unit tests changes in agc2/saturation_protector_unittest.cc are
required since `extra_headroom_db` is removed and the changes in
agc2/adaptive_digital_gain_applier_unittest.cc are required because
`AdaptiveDigitalGainApplier` depends on `kHeadroomDbfs` which has been
updated as stated above.

Bug: webrtc:7494
Change-Id: I0a2a710bbede0caa53938090a004d185fdefaeb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232905
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35109}
7 files changed
tree: 4a3a6b9bb7ac15c56d9ab123a4487fe02e6f8c1d
  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. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. .vpython3
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. DEPS
  37. DIR_METADATA
  38. ENG_REVIEW_OWNERS
  39. g3doc.lua
  40. LICENSE
  41. license_template.txt
  42. native-api.md
  43. OWNERS
  44. PATENTS
  45. PRESUBMIT.py
  46. presubmit_test.py
  47. presubmit_test_mocks.py
  48. pylintrc
  49. README.chromium
  50. README.md
  51. WATCHLISTS
  52. webrtc.gni
  53. webrtc_lib_link_test.cc
  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 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