Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 796 matching lines @@
       _RxVadDetection(false),
       _rxAgcIsEnabled(false),
       _rxNsIsEnabled(false),
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
       associate_send_channel_(ChannelOwner(nullptr)),
       pacing_enabled_(config.Get<VoicePacing>().enabled),
       feedback_observer_proxy_(new TransportFeedbackProxy()),
       seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
-      rtp_packet_sender_proxy_(new RtpPacketSenderProxy()) {
+      rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
+      decoder_factory_(decoder_factory) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
   AudioCodingModule::Config acm_config;
   acm_config.id = VoEModuleId(instanceId, channelId);
   if (config.Get<NetEqCapacityConfig>().enabled) {
     // Clamping the buffer capacity at 20 packets. While going lower will
     // probably work, it makes little sense.
     acm_config.neteq_config.max_packets_in_buffer =
         std::max(20, config.Get<NetEqCapacityConfig>().capacity);
   }
@@ skipping 235 matching lines @@
 int32_t Channel::UpdateLocalTimeStamp() {
   _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
   return 0;
 }
 
 void Channel::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   rtc::CritScope cs(&_callbackCritSect);
   audio_sink_ = std::move(sink);
 }
 
+const rtc::scoped_refptr<AudioDecoderFactory>&
+Channel::GetAudioDecoderFactory() const {
+  return decoder_factory_;
+}
 int32_t Channel::StartPlayout() {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::StartPlayout()");
   if (channel_state_.Get().playing) {
     return 0;
   }
 
   if (!_externalMixing) {
     // Add participant as candidates for mixing.
     if (_outputMixerPtr->SetMixabilityStatus(*this, true) != 0) {
@@ skipping 2480 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc