Pause/resume pacer from Call instead of via SendStreams.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 378 matching lines @@
   channel_group_->SetBweBitrates(bitrate_config.min_bitrate_bps,
                                  bitrate_config.start_bitrate_bps,
                                  bitrate_config.max_bitrate_bps);
 }
 
 void Call::SignalNetworkState(NetworkState state) {
   // Take crit for entire function, it needs to be held while updating streams
   // to guarantee a consistent state across streams.
   rtc::CritScope lock(&network_enabled_crit_);
   network_enabled_ = state == kNetworkUp;
+  channel_group_->SignalNetworkState(state);
   {
     ReadLockScoped write_lock(*send_crit_);
     for (auto& kv : video_send_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
   }
   {
     ReadLockScoped write_lock(*receive_crit_);
     for (auto& kv : video_receive_ssrcs_) {
       kv.second->SignalNetworkState(state);
@@ skipping 122 matching lines @@
     size_t length,
     const PacketTime& packet_time) {
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc