Split out dscp option from VideoOptions to new struct MediaChannelOptions.

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 574 matching lines @@
 
   voe_wrapper_->base()->Terminate();
 }
 
 rtc::scoped_refptr<webrtc::AudioState>
     WebRtcVoiceEngine::GetAudioState() const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return audio_state_;
 }
 
-VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call,
-    const AudioOptions& options) {
+VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
+    webrtc::Call* call,
+    const MediaChannelOptions& options,
+    const AudioOptions& audio_options) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return new WebRtcVoiceMediaChannel(this, options, call);
+  return new WebRtcVoiceMediaChannel(this, options, audio_options, call);
 }
 
 bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString();
   AudioOptions options = options_in;  // The options are modified below.
 
   // kEcConference is AEC with high suppression.
   webrtc::EcModes ec_mode = webrtc::kEcConference;
   webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone;
@@ skipping 673 matching lines @@
   rtc::ThreadChecker worker_thread_checker_;
   webrtc::Call* call_ = nullptr;
   webrtc::AudioReceiveStream::Config config_;
   // The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
   // configuration changes.
   webrtc::AudioReceiveStream* stream_ = nullptr;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
 };
 
-WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
-                                                 const AudioOptions& options,
-                                                 webrtc::Call* call)
-    : engine_(engine), call_(call) {
+WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(
+    WebRtcVoiceEngine* engine,
+    const MediaChannelOptions& options,
+    const AudioOptions& audio_options,
+    webrtc::Call* call)
+    : VoiceMediaChannel(options), engine_(engine), call_(call) {
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
   RTC_DCHECK(call);
   engine->RegisterChannel(this);
-  SetOptions(options);
+  SetOptions(audio_options);
 }
 
 WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";
   // TODO(solenberg): Should be able to delete the streams directly, without
   //                  going through RemoveNnStream(), once stream objects handle
   //                  all (de)configuration.
   while (!send_streams_.empty()) {
     RemoveSendStream(send_streams_.begin()->first);
   }
   while (!recv_streams_.empty()) {
     RemoveRecvStream(recv_streams_.begin()->first);
   }
   engine()->UnregisterChannel(this);
 }
 
+rtc::DiffServCodePoint WebRtcVoiceMediaChannel::MediaTypeDscpValue() const {
+  return kAudioDscpValue;
+}
+
 bool WebRtcVoiceMediaChannel::SetSendParameters(
     const AudioSendParameters& params) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
                << params.ToString();
   // TODO(pthatcher): Refactor this to be more clean now that we have
   // all the information at once.
 
   if (!SetSendCodecs(params.codecs)) {
     return false;
@@ skipping 44 matching lines @@
   }
 
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "Setting voice channel options: "
                << options.ToString();
 
-  // Check if DSCP value is changed from previous.
-  bool dscp_option_changed = (options_.dscp != options.dscp);
-
   // We retain all of the existing options, and apply the given ones
   // on top.  This means there is no way to "clear" options such that
   // they go back to the engine default.
   options_.SetAll(options);
   if (!engine()->ApplyOptions(options_)) {
     LOG(LS_WARNING) <<
         "Failed to apply engine options during channel SetOptions.";
     return false;
   }
 
-  if (dscp_option_changed) {
-    rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
-    if (options_.dscp.value_or(false)) {
-      dscp = kAudioDscpValue;
-    }
-    if (MediaChannel::SetDscp(dscp) != 0) {
-      LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
-    }
-  }
-
   // TODO(solenberg): Don't recreate unless options changed.
   for (auto& it : recv_streams_) {
     it.second->RecreateAudioReceiveStream(
         options_.combined_audio_video_bwe.value_or(false));
   }
 
   LOG(LS_INFO) << "Set voice channel options.  Current options: "
                << options_.ToString();
   return true;
 }
@@ skipping 1116 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE