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

webrtc/audio/audio_send_stream.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_send_stream.h"
 
 #include <string>
 
 #include "webrtc/audio/conversion.h"
+#include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 
 namespace webrtc {
 std::string AudioSendStream::Config::Rtp::ToString() const {
   std::stringstream ss;
@@ skipping 15 matching lines @@
   ss << "{rtp: " << rtp.ToString();
   ss << ", voe_channel_id: " << voe_channel_id;
   // TODO(solenberg): Encoder config.
   ss << ", cng_payload_type: " << cng_payload_type;
   ss << ", red_payload_type: " << red_payload_type;
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
+
+VoiceEngines AudioSendStream::gVoiceEngines;
+
 AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config,
                                  VoiceEngine* voice_engine)
     : config_(config),
-      voice_engine_(voice_engine),
-      voe_base_(voice_engine) {
+      voice_engine_(voice_engine) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config.voe_channel_id, -1);
   RTC_DCHECK(voice_engine_);
+  gVoiceEngines.RegisterVoiceEngine(voice_engine);
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+  gVoiceEngines.DeregisterVoiceEngine(voice_engine_);
 }
 
 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioSendStream::Stats stats;
   stats.local_ssrc = config_.rtp.ssrc;
   ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine_);
   ScopedVoEInterface<VoECodec> codec(voice_engine_);
   ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
   ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
@@ skipping 67 matching lines @@
     int erl = -100;
     int erle = -100;
     int dummy1 = 0;
     int dummy2 = 0;
     if (processing->GetEchoMetrics(erl, erle, dummy1, dummy2) != -1) {
       stats.echo_return_loss = erl;
       stats.echo_return_loss_enhancement = erle;
     }
   }
 
-  // TODO(solenberg): Collect typing noise warnings here too!
-  // bool typing_noise_detected = typing_noise_detected_;
+  stats.typing_noise_detected =
+      gVoiceEngines.GetTypingNoiseDetected(voice_engine_);
 
   return stats;
 }
 
 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;
 }
 
 void AudioSendStream::Start() {
@@ skipping 10 matching lines @@
 
 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
   return false;
 }
 }  // namespace internal
 }  // namespace webrtc