Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.

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 76 matching lines @@
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(congestion_controller);
   RTC_DCHECK(rtp_header_parser_);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
 
+  // This is where we'd like to set the decoder factory to use. However, since

[the sun, 2016/05/30 19:09:53]

A TODO(ossu) seems appropriate here I think.

[ossu, 2016/06/02 15:47:47]

Acknowledged.

+  // it needs to be included when constructing Channel, we cannot do that until
+  // we're able to move Channel ownership into the Audio{Send,Receive}Streams.
+  // The best we can do is check that we're not trying to use two different
+  // factories using the different interfaces.
+  RTC_CHECK(config.decoder_factory);
+  RTC_CHECK(config.decoder_factory == channel_proxy_->GetAudioDecoderFactory());

[the sun, 2016/05/30 19:09:53]

RTC_CHECK_EQ

[ossu, 2016/06/02 15:47:47]

Acknowledged.

+
   channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
 
   for (const auto& extension : config.rtp.extensions) {
     if (extension.name == RtpExtension::kAudioLevel) {
       channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, extension.id);
       RTC_DCHECK(registered);
     } else if (extension.name == RtpExtension::kAbsSendTime) {
       channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
@@ skipping 136 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc