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)
-    : 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);
 
   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,
+                                             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 685 matching lines @@
   // goes away.
   AudioSource* source_ = nullptr;
   bool send_ = false;
   webrtc::RtpParameters rtp_parameters_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(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);
     config_.rtp.remote_ssrc = remote_ssrc;
     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);
   }
 
   ~WebRtcAudioReceiveStream() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     call_->DestroyAudioReceiveStream(stream_);
   }
 
   void RecreateAudioReceiveStream(
       const std::vector<webrtc::RtpExtension>& extensions) {
@@ skipping 809 matching lines @@
     // can obtain RTT from the send channel)
     engine()->voe()->base()->AssociateSendChannel(channel, send_channel);
     LOG(LS_INFO) << "VoiceEngine channel #" << channel
                  << " is associated with channel #" << send_channel << ".";
   }
 
   recv_streams_.insert(std::make_pair(
       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_);
 
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveRecvStream");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ skipping 446 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE