Embed a cricket::MediaConfig in RTCConfiguration.

webrtc/api/peerconnectioninterface.h

 /*
  *  Copyright 2012 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 52 matching lines @@
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/api/umametrics.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/rtccertificate.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/sslstreamadapter.h"
+#include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/portallocator.h"
 
 namespace rtc {
 class SSLIdentity;
 class Thread;
 }
 
 namespace cricket {
 class WebRtcVideoDecoderFactory;
 class WebRtcVideoEncoderFactory;
@@ skipping 151 matching lines @@
     IceServers servers;
     BundlePolicy bundle_policy;
     RtcpMuxPolicy rtcp_mux_policy;
     TcpCandidatePolicy tcp_candidate_policy;
     int audio_jitter_buffer_max_packets;
     bool audio_jitter_buffer_fast_accelerate;
     int ice_connection_receiving_timeout;         // ms
     int ice_backup_candidate_pair_ping_interval;  // ms
     ContinualGatheringPolicy continual_gathering_policy;
     std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
-    bool disable_prerenderer_smoothing;
     bool prioritize_most_likely_ice_candidate_pairs;
+    struct cricket::MediaConfig media_config;

[perkj_webrtc, 2016/03/22 08:31:25]

Doesn't this change break chrome and other clients? 

Also- I am not so happy about moving parts of the config into a sub struct. Why
these 4 types and not many of the others such as screencast_min_bintrate or
audio_jitter_buffer_max_packets....  I am not so sure this is a step in the
right direction. 

     // Flags corresponding to values set by constraint flags.
     // rtc::Optional flags can be "missing", in which case the webrtc
     // default applies.
     bool disable_ipv6;
-    rtc::Optional<bool> enable_dscp;
     bool enable_rtp_data_channel;
-    rtc::Optional<bool> cpu_overuse_detection;
-    rtc::Optional<bool> suspend_below_min_bitrate;
     rtc::Optional<int> screencast_min_bitrate;
     rtc::Optional<bool> combined_audio_video_bwe;
     rtc::Optional<bool> enable_dtls_srtp;
     RTCConfiguration()
         : type(kAll),
           bundle_policy(kBundlePolicyBalanced),
           rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
           tcp_candidate_policy(kTcpCandidatePolicyEnabled),
           audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
           audio_jitter_buffer_fast_accelerate(false),
           ice_connection_receiving_timeout(kUndefined),
           ice_backup_candidate_pair_ping_interval(kUndefined),
           continual_gathering_policy(GATHER_ONCE),
-          disable_prerenderer_smoothing(false),
           prioritize_most_likely_ice_candidate_pairs(false),
           disable_ipv6(false),
           enable_rtp_data_channel(false) {}
   };
 
   struct RTCOfferAnswerOptions {
     static const int kUndefined = -1;
     static const int kMaxOfferToReceiveMedia = 1;
 
     // The default value for constraint offerToReceiveX:true.
@@ skipping 366 matching lines @@
 CreatePeerConnectionFactory(
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_