Move ready to send logic from BaseChannel to RtpTransport.

webrtc/pc/rtptransport.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_PC_RTPTRANSPORT_H_
 #define WEBRTC_PC_RTPTRANSPORT_H_
 
 #include "webrtc/api/ortc/rtptransportinterface.h"
+#include "webrtc/base/sigslot.h"
 
 namespace rtc {
 
 class PacketTransportInternal;
 
 }  // namespace rtc
 
 namespace webrtc {
 
-class RtpTransport : public RtpTransportInterface {
+class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> {
  public:
   RtpTransport(const RtpTransport&) = delete;
   RtpTransport& operator=(const RtpTransport&) = delete;
 
   explicit RtpTransport(bool rtcp_mux_required)
       : rtcp_mux_required_(rtcp_mux_required) {}
 
   bool rtcp_mux_required() const { return rtcp_mux_required_; }
 
   rtc::PacketTransportInternal* rtp_packet_transport() const {
     return rtp_packet_transport_;
   }
   void set_rtp_packet_transport(rtc::PacketTransportInternal* rtp);
 
   rtc::PacketTransportInternal* rtcp_packet_transport() const {
     return rtcp_packet_transport_;
   }
   void set_rtcp_packet_transport(rtc::PacketTransportInternal* rtcp);
 
+  void Connect(bool rtcp);
+  void Disconnect(bool rtcp);
+
+  void OnReadyToSend(rtc::PacketTransportInternal* transport);

[Taylor Brandstetter, 2017/04/14 18:05:33]

This should be private.

[Zach Stein, 2017/04/18 23:39:29]

Done.

+  sigslot::signal1<bool> SignalReadyToSend;
+
+  void SetReadyToSend(bool rtcp, bool ready);
+
   PacketTransportInterface* GetRtpPacketTransport() const override;
   PacketTransportInterface* GetRtcpPacketTransport() const override;
 
   // TODO(zstein): Use these RtcpParameters for configuration elsewhere.
   RTCError SetRtcpParameters(const RtcpParameters& parameters) override;
   RtcpParameters GetRtcpParameters() const override;
 
  protected:
   // TODO(zstein): Remove this when we remove RtpTransportAdapter.
   RtpTransportAdapter* GetInternal() override;
 
  private:
   // True if RTCP-multiplexing is required. rtcp_packet_transport_ should
   // always be null in this case.
   const bool rtcp_mux_required_;
 
   rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
   rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr;
 
+  bool rtp_ready_to_send_ = false;
+  bool rtcp_ready_to_send_ = false;
+
   RtcpParameters rtcp_parameters_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_RTPTRANSPORT_H_