Added an empty AudioTransportProxy to AudioState.

webrtc/audio/audio_state.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include "webrtc/audio/audio_state.h"
 
 #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 {
 namespace internal {
 
 AudioState::AudioState(const AudioState::Config& config)
-    : config_(config), voe_base_(config.voice_engine) {
+    : config_(config),
+      voe_base_(config.voice_engine),
+      audio_transport_proxy_(voe_base_->audio_transport(),
+                             voe_base_->audio_processing(),
+                             config_.audio_mixer) {
   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);
+  device->RegisterAudioCallback(nullptr);

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

Is this really necessary? In AudioDeviceBuffer::RegisterAudioCallback() it
doesn't look like that, but perhaps other platforms implement it in different
ways?

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

It's needed in Chrome.
WebRtcAudioDeviceImpl::RegisterAudioCallback contains this line:
DCHECK_EQ(audio_transport_callback_ == NULL, audio_callback != NULL);

+  device->RegisterAudioCallback(&audio_transport_proxy_);
 }
 
 AudioState::~AudioState() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   voe_base_->DeRegisterVoiceEngineObserver();
 }
 
 VoiceEngine* AudioState::voice_engine() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_.voice_engine;
 }
 
+AudioDeviceModule* AudioState::audio_device() {

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

Only used in one place. Please make the call directly in the ctor instead.

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

Done.

+  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_);
   return typing_noise_detected_;
 }
 
 // Reference count; implementation copied from rtc::RefCountedObject.
 int AudioState::AddRef() const {
   return rtc::AtomicOps::Increment(&ref_count_);
 }
@@ skipping 22 matching lines @@
     typing_noise_detected_ = false;
   }
 }
 }  // namespace internal
 
 rtc::scoped_refptr<AudioState> AudioState::Create(
     const AudioState::Config& config) {
   return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
 }
 }  // namespace webrtc