| 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, ());
|
|
|