| 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 50cdd144ee248a1a468c7a9e7ab92064d7e29085..3ac2f3bff3808657594d85323534001cefb96d2a 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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| @@ -144,14 +144,17 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
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|                 const webrtc::StreamConfig& output_config,
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|                 float* const* dest));
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|    WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame));
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| +  WEBRTC_STUB(ProcessReverseStream, (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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| -      const webrtc::StreamConfig& reverse_config));
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| +  WEBRTC_STUB(ProcessReverseStream,
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| +              (const float* const* src,
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| +               const webrtc::StreamConfig& reverse_input_config,
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| +               const webrtc::StreamConfig& reverse_output_config,
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| +               float* const* dest));
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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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