Move suspend_below_min_bitrate from VideoOptions to MediaConfig.

webrtc/api/webrtcsession.cc

 /*
  *  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 579 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 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);
 
   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);
@@ skipping 1500 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