Integrate Intelligibility with APM

webrtc/voice_engine/output_mixer.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 500 matching lines @@
         {
             // Pure stereo mode (we are receiving a stereo signal).
         }
 
         assert(_audioFrame.num_channels_ == 2);
         AudioFrameOperations::Scale(_panLeft, _panRight, _audioFrame);
     }
 
     // --- Far-end Voice Quality Enhancement (AudioProcessing Module)
     if (feed_data_to_apm)
-      APMAnalyzeReverseStream();
+      APMProcessReverseStream();
 
     // --- External media processing
     {
         CriticalSectionScoped cs(&_callbackCritSect);
         if (_externalMedia)
         {
             const bool is_stereo = (_audioFrame.num_channels_ == 2);
             if (_externalMediaCallbackPtr)
             {
                 _externalMediaCallbackPtr->Process(
@@ skipping 10 matching lines @@
     // --- Measure audio level (0-9) for the combined signal
     _audioLevel.ComputeLevel(_audioFrame);
 
     return 0;
 }
 
 // ----------------------------------------------------------------------------
 //                             Private methods
 // ----------------------------------------------------------------------------
 
-void OutputMixer::APMAnalyzeReverseStream() {
-  // Convert from mixing to AudioProcessing sample rate, determined by the send
-  // side. Downmix to mono.
-  AudioFrame frame;
-  frame.num_channels_ = 1;
-  frame.sample_rate_hz_ = _audioProcessingModulePtr->input_sample_rate_hz();
-  RemixAndResample(_audioFrame, &audioproc_resampler_, &frame);
-
-  if (_audioProcessingModulePtr->AnalyzeReverseStream(&frame) == -1) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
-                 "AudioProcessingModule::AnalyzeReverseStream() => error");
+void OutputMixer::APMProcessReverseStream() {
+  if (_audioProcessingModulePtr->ProcessReverseStream(&_audioFrame) == -1) {
+    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+                 "AudioProcessingModule::ProcessReverseStream() => error");
   }
 }
 
 int
 OutputMixer::InsertInbandDtmfTone()
 {
     uint16_t sampleRate(0);
     _dtmfGenerator.GetSampleRate(sampleRate);
     if (sampleRate != _audioFrame.sample_rate_hz_)
     {
@@ skipping 29 matching lines @@
             _audioFrame.data_[2 * i + 1] = 0;
         }
     }
     assert(_audioFrame.samples_per_channel_ == toneSamples);
 
     return 0;
 }
 
 }  // namespace voe
 }  // namespace webrtc