Configure VoE NACK through AudioSendStream::Config.

webrtc/audio/audio_send_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 21 matching lines @@
   std::stringstream ss;
   ss << "{ssrc: " << ssrc;
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
     if (i != extensions.size() - 1) {
       ss << ", ";
     }
   }
   ss << ']';
+  ss << ", nack: " << nack.ToString();
   ss << ", c_name: " << c_name;
   ss << '}';
   return ss.str();
 }
 
 std::string AudioSendStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
   ss << ", voe_channel_id: " << voe_channel_id;
   // TODO(solenberg): Encoder config.
@@ skipping 16 matching lines @@
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->RegisterSenderCongestionControlObjects(
       congestion_controller->pacer(),
       congestion_controller->GetTransportFeedbackObserver(),
       congestion_controller->packet_router());
   channel_proxy_->SetRTCPStatus(true);
   channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
   channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
+                                config_.rtp.nack.rtp_history_ms / 20);
 
   channel_proxy_->RegisterExternalTransport(config.send_transport);
 
   for (const auto& extension : config.rtp.extensions) {
     if (extension.uri == RtpExtension::kAbsSendTimeUri) {
       channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kAudioLevelUri) {
       channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
@@ skipping 137 matching lines @@
 
 VoiceEngine* AudioSendStream::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