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 bbfb771182cc7196f0e8e0ca9fb63860617d6b8f..81d6c70be9131a4a3e263f6738e95369acdd3327 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -489,6 +489,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src, |
int output_sample_rate_hz, |
ChannelLayout output_layout, |
float* const* dest) { |
+ CriticalSectionScoped crit_scoped(crit_); |
Andrew MacDonald
2015/08/11 18:56:36
FYI: Looks like the webrtc mutexes are recursive,
|
StreamConfig input_stream = api_format_.input_stream(); |
input_stream.set_sample_rate_hz(input_sample_rate_hz); |
input_stream.set_num_channels(ChannelsFromLayout(input_layout)); |