commit | 1309c77a4890c1a2d48854725c5f59268745e4eb | [log] [tgz] |
---|---|---|
author | Alessio Bazzica <alessiob@webrtc.org> | Wed Sep 30 22:53:08 2020 +0200 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Sep 30 22:10:53 2020 +0000 |
tree | efc975d40848a397fb87a4e3185ee8e0f57e0a0a | |
parent | 87b86acde990a0288b2c75be4f03d8bd5e1be74b [diff] [blame] |
AGC2 adaptive digital controller config: new param This CL adds and wires up a parameter (namely, adjacent speech frames threshold) used in `AdaptiveDigitalGainApplier`. Tested: bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: I751cd91f08a6e98ee20f767c8df0ed121c8d4b68 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186049 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32264}
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h index 43c40ff..f16f0ad 100644 --- a/modules/audio_processing/include/audio_processing.h +++ b/modules/audio_processing/include/audio_processing.h
@@ -348,6 +348,7 @@ LevelEstimator level_estimator = kRms; bool use_saturation_protector = true; float extra_saturation_margin_db = 2.f; + int gain_applier_adjacent_speech_frames_threshold = 1; } adaptive_digital; } gain_controller2;