Chromium Code Reviews| Index: webrtc/modules/audio_processing/audio_processing_impl.cc |
| diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc |
| index 12b4c949046f08ddcc3f974d26024bea7e516b54..acbc80c9ddd5742c5065fd93d5c724f0f3853c05 100644 |
| --- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
| @@ -435,9 +435,6 @@ int AudioProcessingImpl::InitializeLocked() { |
| capture_audiobuffer_num_channels, |
| formats_.api_format.output_stream().num_frames())); |
| - public_submodules_->gain_control->Initialize(num_proc_channels(), |
| - proc_sample_rate_hz()); |
| - |
| public_submodules_->echo_cancellation->Initialize( |
| proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(), |
| num_proc_channels()); |
| @@ -446,6 +443,9 @@ int AudioProcessingImpl::InitializeLocked() { |
| public_submodules_->echo_control_mobile->Initialize( |
| proc_split_sample_rate_hz(), num_reverse_channels(), |
| num_output_channels()); |
| + |
|
peah-webrtc
2016/10/28 08:37:49
I moved the initialization of the gain_control to
|
| + public_submodules_->gain_control->Initialize(num_proc_channels(), |
| + proc_sample_rate_hz()); |
| if (constants_.use_experimental_agc) { |
| if (!private_submodules_->agc_manager.get()) { |
| private_submodules_->agc_manager.reset(new AgcManagerDirect( |
| @@ -456,6 +456,7 @@ int AudioProcessingImpl::InitializeLocked() { |
| private_submodules_->agc_manager->Initialize(); |
| private_submodules_->agc_manager->SetCaptureMuted( |
| capture_.output_will_be_muted); |
| + public_submodules_->gain_control_for_experimental_agc->Initialize(); |
|
peah-webrtc
2016/10/28 08:37:50
This is needed to ensure that a new set of data du
|
| } |
| InitializeTransient(); |
| InitializeBeamformer(); |