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-rw-r--r--third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.h2
-rw-r--r--third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc2
2 files changed, 2 insertions, 2 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.h b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.h
index 21405542dc..0bec7af450 100644
--- a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.h
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.h
@@ -50,7 +50,7 @@ class InputVolumeController final {
// Limited to values higher than 0.
int clipped_wait_frames = 300;
// Enables clipping prediction functionality.
- bool enable_clipping_predictor = false;
+ bool enable_clipping_predictor = true;
// Speech level target range (dBFS). If the speech level is in the range
// [`target_range_min_dbfs`, `target_range_max_dbfs`], no input volume
// adjustments are done based on the speech level. For speech levels below
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
index 4ac074526c..5f6dd59d02 100644
--- a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
@@ -2382,7 +2382,7 @@ void AudioProcessingImpl::InitializeGainController2() {
!UseApmVadSubModule(config_, gain_controller2_experiment_params_);
submodules_.gain_controller2 = std::make_unique<GainController2>(
config_.gain_controller2, input_volume_controller_config,
- proc_fullband_sample_rate_hz(), num_proc_channels(), use_internal_vad);
+ proc_fullband_sample_rate_hz(), num_output_channels(), use_internal_vad);
submodules_.gain_controller2->SetCaptureOutputUsed(
capture_.capture_output_used);
}