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/audio_state.h"
 #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 {
-AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config,
-                                 VoiceEngine* voice_engine)
-    : config_(config),
-      voice_engine_(voice_engine),
-      voe_base_(voice_engine) {
+
+AudioSendStream::AudioSendStream(
+    const webrtc::AudioSendStream::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
+    : config_(config), audio_state_(audio_state) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
-  RTC_DCHECK_NE(config.voe_channel_id, -1);
-  RTC_DCHECK(voice_engine_);
+  RTC_DCHECK_NE(config_.voe_channel_id, -1);
+  RTC_DCHECK(audio_state_.get());
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
 }
 
 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_);
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
+  VoiceEngine* voice_engine = audio_state->voice_engine();
+  ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine);
+  ScopedVoEInterface<VoECodec> codec(voice_engine);
+  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine);
+  ScopedVoEInterface<VoEVolumeControl> volume(voice_engine);
   unsigned int ssrc = 0;
   webrtc::CallStatistics call_stats = {0};
   if (rtp->GetLocalSSRC(config_.voe_channel_id, ssrc) == -1 ||
       rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1) {
     return stats;
   }
 
   stats.bytes_sent = call_stats.bytesSent;
   stats.packets_sent = call_stats.packetsSent;
 
@@ skipping 57 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 = audio_state->typing_noise_detected();
 
   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