Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i…

talk/media/webrtc/webrtcvoiceengine.cc

Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 27ca1deb2d41ed2a94edea8502ee916766e39def..f3124bf8b135e0d1448761d1898456321aac4735 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -388,40 +388,41 @@ AudioOptions GetDefaultEngineOptions() {
 std::string GetEnableString(bool enable) {
   return enable ? "enable" : "disable";
 }
+
+webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) {
+  webrtc::AudioState::Config config;
+  config.voice_engine = voe_wrapper->engine();
+  return config;
+}
+
 } // namespace {
 
 WebRtcVoiceEngine::WebRtcVoiceEngine()
     : voe_wrapper_(new VoEWrapper()),
       tracing_(new VoETraceWrapper()),
-      adm_(NULL),
-      log_filter_(SeverityToFilter(kDefaultLogSeverity)),
-      is_dumping_aec_(false) {
+      audio_state_(webrtc::AudioState::Create(MakeAudioStateConfig(voe()))),
+      log_filter_(SeverityToFilter(kDefaultLogSeverity)) {
   Construct();
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
-                                     VoETraceWrapper* tracing)
+                                     VoETraceWrapper* tracing,
+                                     webrtc::AudioState* audio_state)
     : voe_wrapper_(voe_wrapper),
       tracing_(tracing),
-      adm_(NULL),
-      log_filter_(SeverityToFilter(kDefaultLogSeverity)),
-      is_dumping_aec_(false) {
+      audio_state_(audio_state),
+      log_filter_(SeverityToFilter(kDefaultLogSeverity)) {
   Construct();
 }
 
 void WebRtcVoiceEngine::Construct() {
+  std::memset(&default_agc_config_, 0, sizeof(default_agc_config_));
   SetTraceFilter(log_filter_);
-  initialized_ = false;
   LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
   SetTraceOptions("");
   if (tracing_->SetTraceCallback(this) == -1) {
     LOG_RTCERR0(SetTraceCallback);
   }
-  if (voe_wrapper_->base()->RegisterVoiceEngineObserver(*this) == -1) {
-    LOG_RTCERR0(RegisterVoiceEngineObserver);
-  }
-  // Clear the default agc state.
-  memset(&default_agc_config_, 0, sizeof(default_agc_config_));
 
   // Load our audio codec list.
   ConstructCodecs();
@@ -509,9 +510,6 @@ bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst* codec) {
 
 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
   LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
-  if (voe_wrapper_->base()->DeRegisterVoiceEngineObserver() == -1) {
-    LOG_RTCERR0(DeRegisterVoiceEngineObserver);
-  }
   if (adm_) {
     voe_wrapper_.reset();
     adm_->Release();
@@ -521,8 +519,8 @@ WebRtcVoiceEngine::~WebRtcVoiceEngine() {
   tracing_->SetTraceCallback(NULL);
 }
 
-bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
-  RTC_DCHECK(worker_thread == rtc::Thread::Current());
+bool WebRtcVoiceEngine::Init(rtc::Thread* thread) {
+  RTC_DCHECK(thread == rtc::Thread::Current());
   LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
   bool res = InitInternal();
   if (res) {
@@ -1201,28 +1199,15 @@ void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
   }
 }
 
-void WebRtcVoiceEngine::CallbackOnError(int channel_id, int err_code) {
-  RTC_DCHECK(channel_id == -1);
-  LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
-                  << channel_id << ".";
-  rtc::CritScope lock(&channels_cs_);
-  for (WebRtcVoiceMediaChannel* channel : channels_) {
-    channel->OnError(err_code);
-  }
-}
-
 void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
-  RTC_DCHECK(channel != NULL);
-  rtc::CritScope lock(&channels_cs_);
+  RTC_DCHECK(channel);
   channels_.push_back(channel);
 }
 
 void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
-  rtc::CritScope lock(&channels_cs_);
   auto it = std::find(channels_.begin(), channels_.end(), channel);
-  if (it != channels_.end()) {
-    channels_.erase(it);
-  }
+  RTC_DCHECK(it != channels_.end());
+  channels_.erase(it);
 }
 
 // Adjusts the default AGC target level by the specified delta.
@@ -1333,7 +1318,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     RTC_DCHECK(stream_);
   }
   ~WebRtcAudioSendStream() override {
-    RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     Stop();
     call_->DestroyAudioSendStream(stream_);
   }
@@ -1343,7 +1328,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // This method is called on the libjingle worker thread.
   // TODO(xians): Make sure Start() is called only once.
   void Start(AudioRenderer* renderer) {
-    RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     RTC_DCHECK(renderer);
     if (renderer_) {
       RTC_DCHECK(renderer_ == renderer);
@@ -1354,7 +1339,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   }
 
   webrtc::AudioSendStream::Stats GetStats() const {
-    RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return stream_->GetStats();
   }
 
@@ -1362,7 +1347,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // callback will be received after this method.
   // This method is called on the libjingle worker thread.
   void Stop() {
-    RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (renderer_) {
       renderer_->SetSink(nullptr);
       renderer_ = nullptr;
@@ -1376,6 +1361,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
               int sample_rate,
               int number_of_channels,
               size_t number_of_frames) override {
+    RTC_DCHECK(!thread_checker_.CalledOnValidThread());
     RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(voe_audio_transport_);
     voe_audio_transport_->OnData(channel_,
@@ -1389,7 +1375,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // Callback from the |renderer_| when it is going away. In case Start() has
   // never been called, this callback won't be triggered.
   void OnClose() override {
-    RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     // Set |renderer_| to nullptr to make sure no more callback will get into
     // the renderer.
     renderer_ = nullptr;
@@ -1397,12 +1383,12 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
 
   // Accessor to the VoE channel ID.
   int channel() const {
-    RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return channel_;
   }
 
  private:
-  rtc::ThreadChecker signal_thread_checker_;
+  rtc::ThreadChecker thread_checker_;
   rtc::ThreadChecker audio_capture_thread_checker_;
   const int channel_ = -1;
   webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
@@ -1434,20 +1420,9 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
 WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
                                                  const AudioOptions& options,
                                                  webrtc::Call* call)
-    : engine_(engine),
-      send_bitrate_setting_(false),
-      send_bitrate_bps_(0),
-      options_(),
-      dtmf_allowed_(false),
-      desired_playout_(false),
-      nack_enabled_(false),
-      playout_(false),
-      typing_noise_detected_(false),
-      desired_send_(SEND_NOTHING),
-      send_(SEND_NOTHING),
-      call_(call) {
+    : engine_(engine), call_(call) {
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
-  RTC_DCHECK(nullptr != call);
+  RTC_DCHECK(call);
   engine->RegisterChannel(this);
   SetOptions(options);
 }
@@ -2652,9 +2627,8 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     sinfo.echo_delay_std_ms = stats.echo_delay_std_ms;
     sinfo.echo_return_loss = stats.echo_return_loss;
     sinfo.echo_return_loss_enhancement = stats.echo_return_loss_enhancement;
-    sinfo.typing_noise_detected = typing_noise_detected_;
-    // TODO(solenberg): Move to AudioSendStream.
-    //  sinfo.typing_noise_detected = stats.typing_noise_detected;
+    sinfo.typing_noise_detected =
+        (send_ == SEND_NOTHING ? false : stats.typing_noise_detected);
     info->senders.push_back(sinfo);
   }
 
@@ -2694,17 +2668,6 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   return true;
 }
 
-void WebRtcVoiceMediaChannel::OnError(int error) {
-  if (send_ == SEND_NOTHING) {
-    return;
-  }
-  if (error == VE_TYPING_NOISE_WARNING) {
-    typing_noise_detected_ = true;
-  } else if (error == VE_TYPING_NOISE_OFF_WARNING) {
-    typing_noise_detected_ = false;
-  }
-}
-
 int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
   unsigned int ulevel = 0;
   int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
@@ -2787,39 +2750,6 @@ bool WebRtcVoiceMediaChannel::SetPlayout(int channel, bool playout) {
   return true;
 }
 
-// Convert VoiceEngine error code into VoiceMediaChannel::Error enum.
-VoiceMediaChannel::Error
-    WebRtcVoiceMediaChannel::WebRtcErrorToChannelError(int err_code) {
-  switch (err_code) {
-    case 0:
-      return ERROR_NONE;
-    case VE_CANNOT_START_RECORDING:
-    case VE_MIC_VOL_ERROR:
-    case VE_GET_MIC_VOL_ERROR:
-    case VE_CANNOT_ACCESS_MIC_VOL:
-      return ERROR_REC_DEVICE_OPEN_FAILED;
-    case VE_SATURATION_WARNING:
-      return ERROR_REC_DEVICE_SATURATION;
-    case VE_REC_DEVICE_REMOVED:
-      return ERROR_REC_DEVICE_REMOVED;
-    case VE_RUNTIME_REC_WARNING:
-    case VE_RUNTIME_REC_ERROR:
-      return ERROR_REC_RUNTIME_ERROR;
-    case VE_CANNOT_START_PLAYOUT:
-    case VE_SPEAKER_VOL_ERROR:
-    case VE_GET_SPEAKER_VOL_ERROR:
-    case VE_CANNOT_ACCESS_SPEAKER_VOL:
-      return ERROR_PLAY_DEVICE_OPEN_FAILED;
-    case VE_RUNTIME_PLAY_WARNING:
-    case VE_RUNTIME_PLAY_ERROR:
-      return ERROR_PLAY_RUNTIME_ERROR;
-    case VE_TYPING_NOISE_WARNING:
-      return ERROR_REC_TYPING_NOISE_DETECTED;
-    default:
-      return VoiceMediaChannel::ERROR_OTHER;
-  }
-}
-
 bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
     int channel_id, const RtpHeaderExtension* extension) {
   bool enable = false;