Integrate Intelligibility with APM

talk/media/webrtc/fakewebrtcvoiceengine.h

Index: talk/media/webrtc/fakewebrtcvoiceengine.h
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 50cdd144ee248a1a468c7a9e7ab92064d7e29085..8db632417bd2d234c20d94f7b330ce9710f7f397 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -144,14 +144,22 @@ 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(ProcessReverseStream,
+              (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,
+              (float* const* data, const webrtc::StreamConfig& reverse_config));
   WEBRTC_STUB(set_stream_delay_ms, (int delay));
   WEBRTC_STUB_CONST(stream_delay_ms, ());
   WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());