Added an empty AudioTransportProxy to AudioState.

webrtc/audio/audio_state.cc

Index: webrtc/audio/audio_state.cc
diff --git a/webrtc/audio/audio_state.cc b/webrtc/audio/audio_state.cc
index e63f97af2d4fb9fd3123f37532f0553382715984..bb80b2573dcf47071a2261d6bd5c83b8916c4dd3 100644
--- a/webrtc/audio/audio_state.cc
+++ b/webrtc/audio/audio_state.cc
@@ -13,6 +13,7 @@
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 
 namespace webrtc {
@@ -23,6 +24,13 @@ AudioState::AudioState(const AudioState::Config& config)
   process_thread_checker_.DetachFromThread();
   // Only one AudioState should be created per VoiceEngine.
   RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
+
+  auto* const device = audio_device();
+  RTC_DCHECK(device);
+  audio_transport_proxy_.reset(new AudioTransportProxy(
+      voe_base_->audio_transport(), voe_base_->audio_processing(), mixer()));
+  device->RegisterAudioCallback(nullptr);
+  device->RegisterAudioCallback(audio_transport_proxy_.get());
 }
 
 AudioState::~AudioState() {
@@ -35,6 +43,15 @@ VoiceEngine* AudioState::voice_engine() {
   return config_.voice_engine;
 }
 
+AudioDeviceModule* AudioState::audio_device() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return voe_base_->audio_device_module();
+}
+
+rtc::scoped_refptr<AudioMixer> AudioState::mixer() const {
+  return config_.audio_mixer;
+}
+
 bool AudioState::typing_noise_detected() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   rtc::CritScope lock(&crit_sect_);