Move ready to send logic from BaseChannel to RtpTransport.

webrtc/pc/rtptransport.cc

 /*
  *  Copyright 2017 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.
  */
 
 #include "webrtc/pc/rtptransport.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/p2p/base/packettransportinterface.h"
 
 namespace webrtc {
 
 void RtpTransport::set_rtp_packet_transport(rtc::PacketTransportInternal* rtp) {
   rtp_packet_transport_ = rtp;
 }
 
 void RtpTransport::set_rtcp_packet_transport(
     rtc::PacketTransportInternal* rtcp) {
   RTC_DCHECK(!rtcp_mux_required_);
   rtcp_packet_transport_ = rtcp;
 }
 
+void RtpTransport::Connect(bool rtcp) {
+  rtc::PacketTransportInternal* transport =
+      rtcp ? rtcp_packet_transport_ : rtp_packet_transport_;
+  RTC_DCHECK(transport);
+  transport->SignalReadyToSend.connect(this, &RtpTransport::OnReadyToSend);
+}
+
+void RtpTransport::Disconnect(bool rtcp) {
+  rtc::PacketTransportInternal* transport =
+      rtcp ? rtcp_packet_transport_ : rtp_packet_transport_;
+  RTC_DCHECK(transport);
+  transport->SignalReadyToSend.disconnect(this);
+}
+
+void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
+  SetReadyToSend(transport == rtcp_packet_transport_, true);
+}
+
+void RtpTransport::SetReadyToSend(bool rtcp, bool ready) {
+  if (rtcp) {
+    rtcp_ready_to_send_ = ready;
+  } else {
+    rtp_ready_to_send_ = ready;
+  }
+
+  // In the case of rtcp mux |rtcp_packet_transport_| will be null.
+  // TODO(zstein): A better way to verify that rtcp mux is enabled.
+  bool ready_to_send =
+      rtp_ready_to_send_ && (rtcp_ready_to_send_ || !rtcp_packet_transport_);
+  SignalReadyToSend(ready_to_send);
+}
+
 PacketTransportInterface* RtpTransport::GetRtpPacketTransport() const {
   return rtp_packet_transport_;
 }
 
 PacketTransportInterface* RtpTransport::GetRtcpPacketTransport() const {
   return rtcp_packet_transport_;
 }
 
 RTCError RtpTransport::SetRtcpParameters(const RtcpParameters& parameters) {
   if (rtcp_parameters_.mux && !parameters.mux) {
@@ skipping 13 matching lines @@
 
 RtcpParameters RtpTransport::GetRtcpParameters() const {
   return rtcp_parameters_;
 }
 
 RtpTransportAdapter* RtpTransport::GetInternal() {
   return nullptr;
 }
 
 }  // namespace webrtc