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webrtc / src / ca90a552e9f0be327fb5543a6b4f63ca4145bd4b
commitca90a552e9f0be327fb5543a6b4f63ca4145bd4b[log] [tgz]
authorAlessio Bazzica <alessiob@webrtc.org>Wed Sep 27 09:51:10 2017
committerCommit Bot <commit-bot@chromium.org>Wed Sep 27 10:27:56 2017
tree26d1784d915d46cbc49ca477de5f5f43b2016732
parent06319b7830437646d694d6c1934d7e79e1a9fb50 [diff]
audioproc_f with simulated mic analog gain

The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic.
The simulation is done by applying digital gain to the input samples.
This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal.

This CL has been ported from https://codereview.webrtc.org/2834643002/.

Bug: webrtc:7494
Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48
Reviewed-on: https://webrtc-review.googlesource.com/2685
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19992}
11 files changed
tree: 26d1784d915d46cbc49ca477de5f5f43b2016732
  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. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. typedefs.h
  51. WATCHLISTS
  52. webrtc.gni
  53. 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.

More info