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, ()); |