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 CopyOnWriteBuffer;
+struct PacketOptions;
 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) {}
+  explicit RtpTransport(bool rtcp_mux_enabled)
+      : rtcp_mux_enabled_(rtcp_mux_enabled) {}
 
-  bool rtcp_mux_required() const { return rtcp_mux_required_; }
+  bool rtcp_mux_enabled() const { return rtcp_mux_enabled_; }
+  void set_rtcp_mux_enabled(bool enable) { rtcp_mux_enabled_ = enable; }
 
   rtc::PacketTransportInternal* rtp_packet_transport() const {
     return rtp_packet_transport_;
   }
-  void set_rtp_packet_transport(rtc::PacketTransportInternal* rtp);
+  void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp);
 
   rtc::PacketTransportInternal* rtcp_packet_transport() const {
     return rtcp_packet_transport_;
   }
-  void set_rtcp_packet_transport(rtc::PacketTransportInternal* rtcp);
+  void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp);
 
   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;
 
+  // Called whenever a transport's ready-to-send state changes. The argument
+  // is true if all used transports are ready to send. This is more specific
+  // than just "writable"; it means the last send didn't return ENOTCONN.
+  sigslot::signal1<bool> SignalReadyToSend;
+
+  bool IsWritable(bool send_on_rtcp) const;

[Taylor Brandstetter, 2017/04/19 05:56:33]

As mentioned in other comments, I think "send_on_rtcp" can change to just "rtcp"
(meaning, whether an RTP or RTCP packet needs to be sent).

[Zach Stein, 2017/04/20 19:59:11]

Done.

+
+  bool SendPacket(bool send_on_rtcp,
+                  const rtc::CopyOnWriteBuffer* packet,
+                  const rtc::PacketOptions& options,
+                  int flags);
+
  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_;
+  void OnReadyToSend(rtc::PacketTransportInternal* transport);
+
+  // Updates "ready to send" for an individual channel and fires
+  // SignalReadyToSend.
+  void SetReadyToSend(bool rtcp, bool ready);
+
+  bool rtcp_mux_enabled_;
 
   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_