Define Stream base classes

talk/media/webrtc/webrtcvideoengine2.cc

 /*
  * libjingle
  * Copyright 2014 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 1410 matching lines @@
     const rtc::PacketTime& packet_time) {
   if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
           reinterpret_cast<const uint8_t*>(packet->data()), packet->size()) !=
       webrtc::PacketReceiver::DELIVERY_OK) {
     LOG(LS_WARNING) << "Failed to deliver RTCP packet.";
   }
 }
 
 void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
   LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
-  call_->SignalNetworkState(ready ? webrtc::Call::kNetworkUp
-                                  : webrtc::Call::kNetworkDown);
+  call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
 bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) {
   LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
                   << (mute ? "mute" : "unmute");
   DCHECK(ssrc != 0);
   rtc::CritScope stream_lock(&stream_crit_);
   if (send_streams_.find(ssrc) == send_streams_.end()) {
     LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
     return false;
@@ skipping 1167 matching lines @@
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VIDEO