Added an empty AudioTransportProxy to AudioState.

webrtc/audio/audio_transport_proxy.h

Index: webrtc/audio/audio_transport_proxy.h
diff --git a/webrtc/audio/audio_transport_proxy.h b/webrtc/audio/audio_transport_proxy.h
new file mode 100644
index 0000000000000000000000000000000000000000..35cb4c5eeae043c6d3ea292ccf0d1227c612fc67
--- /dev/null
+++ b/webrtc/audio/audio_transport_proxy.h
@@ -0,0 +1,129 @@
+/*
+ *  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,

[the sun, 2016/11/09 09:41:52]

Make a .cc file for this - this is too much implementation to keep in the
header. 

[aleloi, 2016/11/09 16:37:00]

I've already done it in the dependent CL
https://codereview.webrtc.org/2436033002 that connects the mixer.

I moved the change here, because the whole point was to divide that CL into
several smaller ones so that turning on the new mixer is as simple as possible.

+                      AudioProcessing* apm,

[the sun, 2016/11/09 09:41:53]

So, hooking up APM and mixer will be in a follow-up CL?

Could you at least DCHECK the pointers in the ctor here?

[aleloi, 2016/11/09 16:37:01]

I have a check for the mixer in the next dependent CL
https://codereview.webrtc.org/2469743002.

I've added a check for apm and transport in this one.

+                      AudioMixer* mixer)
+      : voe_audio_transport_(voe_audio_transport) {}
+
+  ~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.
+    if (voe_audio_transport_) {

[the sun, 2016/11/09 09:41:53]

So this is to not have to set up the transport in unit tests? I have an idea of
how we can avoid clutter in the tests and make this code unambiguous - see
mock_voice_engine.h.

[aleloi, 2016/11/09 16:37:00]

Acknowledged.

+      return voe_audio_transport_->RecordedDataIsAvailable(
+          audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
+          totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
+    }
+
+    RTC_NOTREACHED()
+        << "AudioTransport proxy doesn't know where to send recorded data: "
+           "no Audio Transport provided.";
+    return -1;
+  }
+
+  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 {
+    RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
+    RTC_DCHECK_GE(nChannels, 1u);
+    RTC_DCHECK_LE(nChannels, 2u);
+    RTC_DCHECK_GE(
+        samplesPerSec,
+        static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
+    RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
+    RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
+                  sizeof(AudioFrame::data_));
+
+    // Pass call through to original audio transport instance.
+    if (voe_audio_transport_) {
+      return voe_audio_transport_->NeedMorePlayData(
+          nSamples, nBytesPerSample, nChannels, samplesPerSec, audioSamples,
+          nSamplesOut, elapsed_time_ms, ntp_time_ms);
+    }
+
+    RTC_NOTREACHED()
+        << "AudioTransport proxy doesn't know from where to get play data: "
+           "no Audio Transport provided.";
+    return -1;
+  }
+
+  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();
+  }
+
+  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 {
+    RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 8 * sizeof(int16_t));
+    RTC_DCHECK_GE(number_of_channels, 1u);
+    RTC_DCHECK_LE(number_of_channels, 2u);
+    RTC_DCHECK_GE(static_cast<int>(sample_rate),
+                  AudioProcessing::NativeRate::kSampleRate8kHz);
+    RTC_DCHECK_EQ(static_cast<int>(number_of_frames * 100), sample_rate);
+    RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
+                  sizeof(AudioFrame::data_));
+    if (voe_audio_transport_) {
+      return voe_audio_transport_->PullRenderData(
+          bits_per_sample, sample_rate, number_of_channels, number_of_frames,
+          audio_data, elapsed_time_ms, ntp_time_ms);
+    }
+
+    RTC_NOTREACHED()
+        << "AudioTransport proxy doesn't know from where to get play data: "
+           "no Audio Transport provided.";
+  }
+
+ private:
+  AudioTransport* voe_audio_transport_;
+  AudioFrame frame_for_mixing_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportProxy);
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_