Allow an external audio processing module to be used in WebRTC

webrtc/audio/audio_transport_proxy.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_
 #define WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 
 class AudioTransportProxy : public AudioTransport {
  public:
   AudioTransportProxy(AudioTransport* voe_audio_transport,
-                      AudioProcessing* apm,
+                      AudioProcessing* audio_processing,
                       AudioMixer* mixer);
 
   ~AudioTransportProxy() override;
 
   int32_t RecordedDataIsAvailable(const void* audioSamples,
                                   const size_t nSamples,
                                   const size_t nBytesPerSample,
                                   const size_t nChannels,
                                   const uint32_t samplesPerSec,
                                   const uint32_t totalDelayMS,
@@ skipping 21 matching lines @@
   void PullRenderData(int bits_per_sample,
                       int sample_rate,
                       size_t number_of_channels,
                       size_t number_of_frames,
                       void* audio_data,
                       int64_t* elapsed_time_ms,
                       int64_t* ntp_time_ms) override;
 
  private:
   AudioTransport* voe_audio_transport_;
-  AudioProcessing* apm_;
+  AudioProcessing* audio_processing_;
   rtc::scoped_refptr<AudioMixer> mixer_;
   AudioFrame mixed_frame_;
   // Converts mixed audio to the audio device output rate.
   PushResampler<int16_t> resampler_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportProxy);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_