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() { |