WIP: Move the creation of AudioCodecFactory into PeerConnectionFactory.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index ed67ce84ba00cc520872b1a17e0ea8bdbb579160..a58193c98e356cf58ce474d9777bea1942160b59 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -520,14 +520,20 @@ bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
   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)
-    : adm_(adm), voe_wrapper_(voe_wrapper) {
+WebRtcVoiceEngine::WebRtcVoiceEngine(
+    webrtc::AudioDeviceModule* adm,
+    std::shared_ptr<webrtc::AudioDecoderFactory> decoder_factory,
+    VoEWrapper* voe_wrapper)
+    : adm_(adm),
+      decoder_factory_(std::move(decoder_factory)),
+      voe_wrapper_(voe_wrapper) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
   RTC_DCHECK(voe_wrapper);
@@ -547,7 +553,8 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm,
   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,
+                                             decoder_factory_));
   webrtc::Trace::set_level_filter(kDefaultTraceFilter);
 
   // No ADM supplied? Get the default one from VoE.
@@ -1253,13 +1260,15 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
 
 class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
  public:
-  WebRtcAudioReceiveStream(int ch,
-                           uint32_t remote_ssrc,
-                           uint32_t local_ssrc,
-                           bool use_transport_cc,
-                           const std::string& sync_group,
-                           const std::vector<webrtc::RtpExtension>& extensions,
-                           webrtc::Call* call)
+  WebRtcAudioReceiveStream(
+      int ch,
+      uint32_t remote_ssrc,
+      uint32_t local_ssrc,
+      bool use_transport_cc,
+      const std::string& sync_group,
+      const std::vector<webrtc::RtpExtension>& extensions,
+      webrtc::Call* call,
+      std::shared_ptr<webrtc::AudioDecoderFactory> decoder_factory)
       : call_(call), config_() {
     RTC_DCHECK_GE(ch, 0);
     RTC_DCHECK(call);
@@ -1267,6 +1276,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
     config_.rtp.local_ssrc = local_ssrc;
     config_.voe_channel_id = ch;
     config_.sync_group = sync_group;
+    config_.decoder_factory = std::move(decoder_factory);
     RecreateAudioReceiveStream(use_transport_cc, extensions);
   }
 
@@ -2096,7 +2106,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
       ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
                                          recv_transport_cc_enabled_,
                                          sp.sync_label, recv_rtp_extensions_,
-                                         call_)));
+                                         call_, engine_->decoder_factory_)));
 
   SetNack(channel, send_codec_spec_.nack_enabled);
   SetPlayout(channel, playout_);