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 1d12fbf2ffbac76cf1fbc9c927923ecf32eae546..58612342877a96b02bd626158e94e52f209e6bc0 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -393,9 +393,7 @@ std::string GetEnableString(bool enable) {
 WebRtcVoiceEngine::WebRtcVoiceEngine()
     : voe_wrapper_(new VoEWrapper()),
       tracing_(new VoETraceWrapper()),
-      adm_(NULL),
-      log_filter_(SeverityToFilter(kDefaultLogSeverity)),
-      is_dumping_aec_(false) {
+      log_filter_(SeverityToFilter(kDefaultLogSeverity)) {
   Construct();
 }
 
@@ -403,25 +401,17 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
                                      VoETraceWrapper* tracing)
     : voe_wrapper_(voe_wrapper),
       tracing_(tracing),
-      adm_(NULL),
-      log_filter_(SeverityToFilter(kDefaultLogSeverity)),
-      is_dumping_aec_(false) {
+      log_filter_(SeverityToFilter(kDefaultLogSeverity)) {
   Construct();
 }
 
 void WebRtcVoiceEngine::Construct() {
   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 +499,6 @@ 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();
@@ -1201,28 +1188,15 @@ 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.
@@ -1434,20 +1408,9 @@ 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 +2615,8 @@ 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 +2656,6 @@ 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 +2738,6 @@ 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;