Now pass ADM as a constructor argument to audio_state.

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..cd5a7b88ca90eb98f73793409c9f446786c012ba 100644
--- a/webrtc/audio/audio_state.cc
+++ b/webrtc/audio/audio_state.cc
@@ -14,15 +14,22 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/modules/audio_device/include/audio_device_defines.h"
 
 namespace webrtc {
 namespace internal {
 
-AudioState::AudioState(const AudioState::Config& config)
-    : config_(config), voe_base_(config.voice_engine) {
+AudioState::AudioState(const AudioState::Config& config,
+                       webrtc::AudioDeviceModule* adm)
+    : config_(config), voe_base_(config.voice_engine), adm_(adm) {
   process_thread_checker_.DetachFromThread();
+
   // Only one AudioState should be created per VoiceEngine.
   RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
+  if (!adm_) {
+    adm_ = voe_base_->audio_device_module();
+  }
 }
 
 AudioState::~AudioState() {
@@ -35,6 +42,11 @@ VoiceEngine* AudioState::voice_engine() {
   return config_.voice_engine;
 }
 
+AudioDeviceModule* AudioState::audio_device() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return adm_;
+}
+
 bool AudioState::typing_noise_detected() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   rtc::CritScope lock(&crit_sect_);
@@ -73,7 +85,8 @@ void AudioState::CallbackOnError(int channel_id, int err_code) {
 }  // namespace internal
 
 rtc::scoped_refptr<AudioState> AudioState::Create(
-    const AudioState::Config& config) {
-  return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
+    const AudioState::Config& config,
+    webrtc::AudioDeviceModule* adm) {
+  return rtc::scoped_refptr<AudioState>(new internal::AudioState(config, adm));
 }
 }  // namespace webrtc