| 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 1756e8506d1fa977c02179f08057b8385e1f3375..a09c1c9acf3362f53cbd9f968ad5a4b6a5cf0932 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| @@ -837,17 +837,6 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
|
| int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
|
| TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
|
| rtc::CritScope cs(&crit_render_);
|
| - RETURN_ON_ERR(AnalyzeReverseStream(frame));
|
| - if (is_rev_processed()) {
|
| - render_.render_audio->InterleaveTo(frame, true);
|
| - }
|
| -
|
| - return kNoError;
|
| -}
|
| -
|
| -int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
|
| - TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_AudioFrame");
|
| - rtc::CritScope cs(&crit_render_);
|
| if (frame == nullptr) {
|
| return kNullPointerError;
|
| }
|
| @@ -893,7 +882,11 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
|
| }
|
| #endif
|
| render_.render_audio->DeinterleaveFrom(frame);
|
| - return ProcessReverseStreamLocked();
|
| + RETURN_ON_ERR(ProcessReverseStreamLocked());
|
| + if (is_rev_processed()) {
|
| + render_.render_audio->InterleaveTo(frame, true);
|
| + }
|
| + return kNoError;
|
| }
|
|
|
| int AudioProcessingImpl::ProcessReverseStreamLocked() {
|
|
|