Introduce struct MediaConfig, with construction-time settings.

webrtc/media/webrtc/webrtcvoiceengine.cc

Index: webrtc/media/webrtc/webrtcvoiceengine.cc
diff --git a/webrtc/media/webrtc/webrtcvoiceengine.cc b/webrtc/media/webrtc/webrtcvoiceengine.cc
index afdc6783ace9434805574425bf6e2ecaa46b9af1..5d264fe828fa6a9a9c7ea35b7d2c0ff84aa8d735 100644
--- a/webrtc/media/webrtc/webrtcvoiceengine.cc
+++ b/webrtc/media/webrtc/webrtcvoiceengine.cc
@@ -666,10 +666,12 @@ rtc::scoped_refptr<webrtc::AudioState>
   return audio_state_;
 }
 
-VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call,
+VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
+    webrtc::Call* call,
+    const webrtc::MediaConfig& config,
     const AudioOptions& options) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return new WebRtcVoiceMediaChannel(this, options, call);
+  return new WebRtcVoiceMediaChannel(this, config, options, call);
 }
 
 bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
@@ -1382,10 +1384,12 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
 };
 
-WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
-                                                 const AudioOptions& options,
-                                                 webrtc::Call* call)
-    : engine_(engine), call_(call) {
+WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(
+    WebRtcVoiceEngine* engine,
+    const webrtc::MediaConfig& config,
+    const AudioOptions& options,
+    webrtc::Call* call)
+    : VoiceMediaChannel(config), engine_(engine), call_(call) {
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
   RTC_DCHECK(call);
   engine->RegisterChannel(this);
@@ -1407,6 +1411,10 @@ WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
   engine()->UnregisterChannel(this);
 }
 
+rtc::DiffServCodePoint WebRtcVoiceMediaChannel::MediaTypeDscpValue() const {
+  return kAudioDscpValue;
+}
+
 bool WebRtcVoiceMediaChannel::SetSendParameters(
     const AudioSendParameters& params) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ -1470,9 +1478,6 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
   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.
@@ -1482,17 +1487,6 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
         "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";
-    }
-  }
-
   LOG(LS_INFO) << "Set voice channel options.  Current options: "
                << options_.ToString();
   return true;