Index: talk/media/webrtc/fakewebrtcvoiceengine.h |
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h |
index 419170b24dc471bdb2c1a8c2ec9ee7fe6fde6a5c..9a6ce4389c7b2590aebc4bfba64707e6d372acc1 100644 |
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h |
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h |
@@ -137,11 +137,16 @@ class FakeAudioProcessing : public webrtc::AudioProcessing { |
webrtc::AudioProcessing::ChannelLayout output_layout, |
float* const* dest)); |
WEBRTC_STUB(AnalyzeReverseStream, (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, |
+ int samples_per_channel, |
+ int sample_rate_hz, |
+ webrtc::AudioProcessing::ChannelLayout layout)); |
+ WEBRTC_STUB(ProcessReverseStream, |
+ (float* const* data, |
+ int samples_per_channel, |
+ int sample_rate_hz, |
+ webrtc::AudioProcessing::ChannelLayout layout)); |
WEBRTC_STUB(set_stream_delay_ms, (int delay)); |
WEBRTC_STUB_CONST(stream_delay_ms, ()); |
WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); |