| 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, ());
|
|
|