Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_receive_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.
  */
 
@@ skipping 123 matching lines @@
   // Configure bandwidth estimation.
   channel_proxy_->RegisterReceiverCongestionControlObjects(
       congestion_controller->packet_router());
   if (UseSendSideBwe(config)) {
     remote_bitrate_estimator_ =
         congestion_controller->GetRemoteBitrateEstimator(true);
   }
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
-  Stop();
-  channel_proxy_->DisassociateSendChannel();

[the sun, 2016/11/16 20:22:57]

Bad merge

[aleloi, 2016/11/17 18:12:26]

Good spot! Thanks!

+  if (playing_) {
+    Stop();
+  }
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
   if (remote_bitrate_estimator_) {
     remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
   }
 }
 
 void AudioReceiveStream::Start() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (playing_) {
+    return;
+  }
+
   ScopedVoEInterface<VoEBase> base(voice_engine());
   int error = base->StartPlayout(config_.voe_channel_id);
   if (error != 0) {
     LOG(LS_ERROR) << "AudioReceiveStream::Start failed with error: " << error;
+    return;
   }
+
+  auto* const the_audio_state = audio_state();
+
+  // TODO(aleloi): rethink handling of error flags and the
+  // AudioMixer::AddSource()/::RemoveSource() interface.
+  if (!the_audio_state->mixer()->AddSource(this)) {
+    LOG(LS_ERROR) << "Failed to add source to mixer.";
+    return;
+  }
+
+  playing_ = true;
 }
 
 void AudioReceiveStream::Stop() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+

[the sun, 2016/11/16 20:22:57]

dd

+  if (!playing_) {
+    return;
+  }
+  playing_ = false;
+
+  auto* const the_audio_state = audio_state();

[the sun, 2016/11/16 20:22:57]

Don't need this local

[aleloi, 2016/11/17 18:12:26]

Acknowledged.

+  if (!the_audio_state->mixer()->RemoveSource(this)) {
+    LOG(LS_ERROR) << "Failed to remove stream from mixer.";
+  }
+
   ScopedVoEInterface<VoEBase> base(voice_engine());
   base->StopPlayout(config_.voe_channel_id);
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
   webrtc::AudioReceiveStream::Stats stats;
   stats.remote_ssrc = config_.rtp.remote_ssrc;
   ScopedVoEInterface<VoECodec> codec(voice_engine());
 
   webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
   webrtc::CodecInst codec_inst = {0};
   if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
     return stats;
   }
 
@@ skipping 28 matching lines @@
   stats.decoding_normal = ds.decoded_normal;
   stats.decoding_plc = ds.decoded_plc;
   stats.decoding_cng = ds.decoded_cng;
   stats.decoding_plc_cng = ds.decoded_plc_cng;
   stats.decoding_muted_output = ds.decoded_muted_output;
 
   return stats;
 }
 
 void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
   channel_proxy_->SetSink(std::move(sink));
 }
 
 void AudioReceiveStream::SetGain(float gain) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
   channel_proxy_->SetChannelOutputVolumeScaling(gain);
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
   return config_;
 }
 
 void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (send_stream) {
     VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
     std::unique_ptr<voe::ChannelProxy> send_channel_proxy =
         voe_impl->GetChannelProxy(send_stream->config().voe_channel_id);
     channel_proxy_->AssociateSendChannel(*send_channel_proxy.get());
   } else {
     channel_proxy_->DisassociateSendChannel();
   }
 }
 
 void AudioReceiveStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
 }
 
 bool AudioReceiveStream::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 channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
@@ skipping 32 matching lines @@
 }
 
 int AudioReceiveStream::PreferredSampleRate() const {
   return channel_proxy_->NeededFrequency();
 }
 
 int AudioReceiveStream::Ssrc() const {
   return config_.rtp.local_ssrc;
 }
 
+internal::AudioState* AudioReceiveStream::audio_state() const {
+  auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
+  RTC_DCHECK(audio_state);
+  return audio_state;
+}
+
 VoiceEngine* AudioReceiveStream::voice_engine() const {
-  internal::AudioState* audio_state =
-      static_cast<internal::AudioState*>(audio_state_.get());
-  VoiceEngine* voice_engine = audio_state->voice_engine();
+  auto* voice_engine = audio_state()->voice_engine();

[the sun, 2016/11/16 20:22:57]

Yes, nicer! :)

   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc