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