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 568 matching lines @@
   // 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");
   }
+  RemixAndResample(frame, &audioproc_resampler_, &_audioFrame);

[Andrew MacDonald, 2015/07/24 23:50:40]

This is a big deal ;) First, you can't use the same resampler here, as it should
be operating on the same continuous stream.

I think the right way to do this is remove the pre-RemixAndResample and let APM
handle it. In the "modern" float interface that would be no problem.
Unfortunately, this AudioFrame interface doesn't handle resampling in
DeinterleaveFrom, as you've seen. So we'd need to add support for it there.

Another, and possibly superior, option is to remove support for IE through the
AudioFrame interface in this CL.

[ekm, 2015/07/29 00:37:19]

Done (1st option): added support in InterleaveTo in audio_buffer.

 }
 
 int
 OutputMixer::InsertInbandDtmfTone()
 {
     uint16_t sampleRate(0);
     _dtmfGenerator.GetSampleRate(sampleRate);
     if (sampleRate != _audioFrame.sample_rate_hz_)
     {
         // Update sample rate of Dtmf tone since the mixing frequency changed.
@@ skipping 28 matching lines @@
             _audioFrame.data_[2 * i + 1] = 0;
         }
     }
     assert(_audioFrame.samples_per_channel_ == toneSamples);
 
     return 0;
 }
 
 }  // namespace voe
 }  // namespace webrtc