| Index: talk/media/webrtc/fakewebrtcvoiceengine.h | 
| diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h | 
| index 50cdd144ee248a1a468c7a9e7ab92064d7e29085..3ac2f3bff3808657594d85323534001cefb96d2a 100644 | 
| --- a/talk/media/webrtc/fakewebrtcvoiceengine.h | 
| +++ b/talk/media/webrtc/fakewebrtcvoiceengine.h | 
| @@ -144,14 +144,17 @@ class FakeAudioProcessing : public webrtc::AudioProcessing { | 
| const webrtc::StreamConfig& output_config, | 
| float* const* dest)); | 
| WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame)); | 
| +  WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame)); | 
| WEBRTC_STUB(AnalyzeReverseStream, ( | 
| const float* const* data, | 
| int samples_per_channel, | 
| int sample_rate_hz, | 
| webrtc::AudioProcessing::ChannelLayout layout)); | 
| -  WEBRTC_STUB(AnalyzeReverseStream, ( | 
| -      const float* const* data, | 
| -      const webrtc::StreamConfig& reverse_config)); | 
| +  WEBRTC_STUB(ProcessReverseStream, | 
| +              (const float* const* src, | 
| +               const webrtc::StreamConfig& reverse_input_config, | 
| +               const webrtc::StreamConfig& reverse_output_config, | 
| +               float* const* dest)); | 
| WEBRTC_STUB(set_stream_delay_ms, (int delay)); | 
| WEBRTC_STUB_CONST(stream_delay_ms, ()); | 
| WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); | 
|  |