Introduce struct MediaConfig, with construction-time settings.

talk/app/webrtc/webrtcsession.cc

 /*
  * libjingle
  * Copyright 2012 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 583 matching lines @@
   LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
 }
 
 bool WebRtcSession::Initialize(
     const PeerConnectionFactoryInterface::Options& options,
     const MediaConstraintsInterface* constraints,
     rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
   bundle_policy_ = rtc_configuration.bundle_policy;
   rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
-  video_options_.disable_prerenderer_smoothing =
-      rtc::Optional<bool>(rtc_configuration.disable_prerenderer_smoothing);

[perkj_webrtc, 2016/02/09 15:35:37]

where do you set disable_prerenderer_smoothing now? It make sence to me to have
this kind of settings in PeerConnectionInterface::RTCConfiguration .

[nisse-webrtc, 2016/02/10 08:53:00]

It's copied from RTCConfiguration to MediaConfig. But it is now done by
PeerConnection instead of WebRtcSession.

+
   transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
 
   // Obtain a certificate from RTCConfiguration if any were provided (optional).
   rtc::scoped_refptr<rtc::RTCCertificate> certificate;
   if (!rtc_configuration.certificates.empty()) {
     // TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
     // just picking the first one. The decision should be made based on the DTLS
     // handshake. The DTLS negotiations need to know about all certificates.
     certificate = rtc_configuration.certificates[0];
   }
@@ skipping 26 matching lines @@
     LOG(LS_INFO) << "Allowing RTP data engine.";
     data_channel_type_ = cricket::DCT_RTP;
   } else {
     // DTLS has to be enabled to use SCTP.
     if (!options.disable_sctp_data_channels && dtls_enabled_) {
       LOG(LS_INFO) << "Allowing SCTP data engine.";
       data_channel_type_ = cricket::DCT_SCTP;
     }
   }
 
-  // Find DSCP constraint.
-  if (FindConstraint(
-        constraints,
-        MediaConstraintsInterface::kEnableDscp,
-        &value, NULL)) {
-    audio_options_.dscp = rtc::Optional<bool>(value);
-    video_options_.dscp = rtc::Optional<bool>(value);
-  }
-
   // Find Suspend Below Min Bitrate constraint.
   if (FindConstraint(
           constraints,
           MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
           &value,
           NULL)) {
     video_options_.suspend_below_min_bitrate = rtc::Optional<bool>(value);
   }
 
   SetOptionFromOptionalConstraint(constraints,
       MediaConstraintsInterface::kScreencastMinBitrate,
       &video_options_.screencast_min_bitrate_kbps);
 
-  // Find constraints for cpu overuse detection.
-  SetOptionFromOptionalConstraint(constraints,

[perkj_webrtc, 2016/02/09 15:35:37]

Add test for this as well?

-      MediaConstraintsInterface::kCpuOveruseDetection,
-      &video_options_.cpu_overuse_detection);
-
   SetOptionFromOptionalConstraint(constraints,
       MediaConstraintsInterface::kCombinedAudioVideoBwe,
       &audio_options_.combined_audio_video_bwe);
 
   audio_options_.audio_jitter_buffer_max_packets =
       rtc::Optional<int>(rtc_configuration.audio_jitter_buffer_max_packets);
 
   audio_options_.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(
       rtc_configuration.audio_jitter_buffer_fast_accelerate);
 
@@ skipping 1488 matching lines @@
   }
 }
 
 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
                                    const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc