Allow an external audio processing module to be used in WebRTC

webrtc/audio/audio_transport_proxy.cc

 /*
  *  Copyright (c) 2016 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 16 matching lines @@
 
   // TODO(yujo): make resampler take an AudioFrame, and add special case
   // handling of muted frames.
   return resampler->Resample(
       frame.data(), frame.samples_per_channel_ * number_of_channels,
       destination, number_of_channels * target_number_of_samples_per_channel);
 }
 }  // namespace
 
 AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport,
-                                         AudioProcessing* apm,
+                                         AudioProcessing* audio_processing,
                                          AudioMixer* mixer)
-    : voe_audio_transport_(voe_audio_transport), apm_(apm), mixer_(mixer) {
+    : voe_audio_transport_(voe_audio_transport),
+      audio_processing_(audio_processing), mixer_(mixer) {
   RTC_DCHECK(voe_audio_transport);
-  RTC_DCHECK(apm);
+  RTC_DCHECK(audio_processing);
   RTC_DCHECK(mixer);
 }
 
 AudioTransportProxy::~AudioTransportProxy() {}
 
 int32_t AudioTransportProxy::RecordedDataIsAvailable(
     const void* audioSamples,
     const size_t nSamples,
     const size_t nBytesPerSample,
     const size_t nChannels,
@@ skipping 26 matching lines @@
 
   // 100 = 1 second / data duration (10 ms).
   RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
   RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
                 AudioFrame::kMaxDataSizeBytes);
 
   mixer_->Mix(nChannels, &mixed_frame_);
   *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
   *ntp_time_ms = mixed_frame_.ntp_time_ms_;
 
-  const auto error = apm_->ProcessReverseStream(&mixed_frame_);
+  const auto error = audio_processing_->ProcessReverseStream(&mixed_frame_);
   RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
 
   nSamplesOut = Resample(mixed_frame_, samplesPerSec, &resampler_,
                          static_cast<int16_t*>(audioSamples));
   RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
   return 0;
 }
 
 void AudioTransportProxy::PushCaptureData(int voe_channel,
                                           const void* audio_data,
@@ skipping 27 matching lines @@
   mixer_->Mix(number_of_channels, &mixed_frame_);
   *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
   *ntp_time_ms = mixed_frame_.ntp_time_ms_;
 
   const auto output_samples = Resample(mixed_frame_, sample_rate, &resampler_,
                                        static_cast<int16_t*>(audio_data));
   RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
 }
 
 }  // namespace webrtc