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![]() In order to experiment with AGC2 and TS at the same time, 3 field trials are removed and merged into `WebRTC-Audio-GainController2`, which is existing. New parameters for the `WebRTC-Audio-GainController2` field trial: - `switch_to_agc2`: true by default; when true, the gain control switches to AGC2 (both for the input volume and for the adaptive digital gain); - `min_input_volume`: minimum input volume enforced by the input volume controller when the applied input volume is not zero; - `disallow_transient_suppressor_usage`: when true, TS is never created. Removed field trials: - `WebRTC-Audio-Agc2-MinInputVolume`: now a parameter of `WebRTC-Audio-GainController2`; - `WebRTC-ApmTransientSuppressorKillSwitch`: now a parameter of `WebRTC-Audio-GainController2`; - `WebRTC-Audio-TransientSuppressorVadMode`: automatically inferred from `WebRTC-Audio-GainController2`. Bug: webrtc:7494 Change-Id: I452798c0862d71f9adae6d163fe841df05ca44d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287861 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38890} |
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.. | ||
async_audio_processing | ||
audio_coding | ||
audio_device | ||
audio_mixer | ||
audio_processing | ||
congestion_controller | ||
desktop_capture | ||
include | ||
pacing | ||
portal | ||
remote_bitrate_estimator | ||
rtp_rtcp | ||
third_party | ||
utility | ||
video_capture | ||
video_coding | ||
BUILD.gn | ||
module_common_types_unittest.cc |