WIP: Move the creation of AudioCodecFactory into PeerConnectionFactory.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 502 matching lines @@
     {kRedCodecName, 8000, 1, 127, false, {}},
     {kDtmfCodecName, 8000, 1, 126, false, {}},
 };
 } // namespace {
 
 bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
                                     webrtc::CodecInst* out) {
   return WebRtcVoiceCodecs::ToCodecInst(in, out);
 }
 
-WebRtcVoiceEngine::WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm)
-    : WebRtcVoiceEngine(adm, new VoEWrapper()) {
+WebRtcVoiceEngine::WebRtcVoiceEngine(
+    webrtc::AudioDeviceModule* adm,
+    std::shared_ptr<webrtc::AudioDecoderFactory> decoder_factory)
+    : WebRtcVoiceEngine(adm, decoder_factory, new VoEWrapper()) {
   audio_state_ = webrtc::AudioState::Create(MakeAudioStateConfig(voe()));
 }
 
-WebRtcVoiceEngine::WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm,
-                                     VoEWrapper* voe_wrapper)
+WebRtcVoiceEngine::WebRtcVoiceEngine(
+    webrtc::AudioDeviceModule* adm,
+    std::shared_ptr<webrtc::AudioDecoderFactory> decoder_factory,
+    VoEWrapper* voe_wrapper)
     : adm_(adm), voe_wrapper_(voe_wrapper) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
   RTC_DCHECK(voe_wrapper);
 
   signal_thread_checker_.DetachFromThread();
 
   // Load our audio codec list.
   LOG(LS_INFO) << "Supported codecs in order of preference:";
   codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
   for (const AudioCodec& codec : codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
 
   voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
 
   // Temporarily turn logging level up for the Init() call.
   webrtc::Trace::SetTraceCallback(this);
   webrtc::Trace::set_level_filter(kElevatedTraceFilter);
   LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
-  RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm_.get()));
+  RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm_.get(), nullptr,
+                                             std::move(decoder_factory)));
   webrtc::Trace::set_level_filter(kDefaultTraceFilter);
 
   // No ADM supplied? Get the default one from VoE.
   if (!adm_) {
     adm_ = voe_wrapper_->base()->audio_device_module();
   }
   RTC_DCHECK(adm_);
 
   // Save the default AGC configuration settings. This must happen before
   // calling ApplyOptions or the default will be overwritten.
@@ skipping 1995 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE