Added an empty AudioTransportProxy to AudioState.

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/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,
+                      AudioMixer* mixer)
+      : voe_audio_transport_(voe_audio_transport) {

[the sun, 2016/11/14 13:50:09]

Move to .cc

[aleloi, 2016/11/14 14:24:43]

For clarity to have all defs in the same place? (Done).

+    RTC_DCHECK(voe_audio_transport);
+    RTC_DCHECK(apm);
+  }
+
+  ~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,
+                                  const int32_t clockDrift,
+                                  const uint32_t currentMicLevel,
+                                  const bool keyPressed,
+                                  uint32_t& newMicLevel) override {
+    // Pass call through to original audio transport instance.
+    return voe_audio_transport_->RecordedDataIsAvailable(

[the sun, 2016/11/14 13:50:09]

Move to .cc

[aleloi, 2016/11/14 14:24:42]

Done.

+        audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
+        totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
+  }
+
+  int32_t NeedMorePlayData(const size_t nSamples,
+                           const size_t nBytesPerSample,
+                           const size_t nChannels,
+                           const uint32_t samplesPerSec,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override;
+
+  void PushCaptureData(int voe_channel,
+                       const void* audio_data,
+                       int bits_per_sample,
+                       int sample_rate,
+                       size_t number_of_channels,
+                       size_t number_of_frames) override {
+    // This is part of deprecated VoE interface operating on specific
+    // VoE channels. It should not be used.
+    RTC_NOTREACHED();

[the sun, 2016/11/14 13:50:08]

Here as well

[aleloi, 2016/11/14 14:24:43]

Done.

+  }
+
+  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_;
+  AudioFrame frame_for_mixing_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportProxy);
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_