Move ready to send logic from BaseChannel to RtpTransport.

webrtc/pc/channel.h

 /*
  *  Copyright 2004 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 165 matching lines @@
   // Only public for unit tests.  Otherwise, consider private.
   DtlsTransportInternal* rtp_dtls_transport() const {
     return rtp_dtls_transport_;
   }
   DtlsTransportInternal* rtcp_dtls_transport() const {
     return rtcp_dtls_transport_;
   }
 
   bool NeedsRtcpTransport();
 
-  // Made public for easier testing.
-  //
-  // Updates "ready to send" for an individual channel, and informs the media
-  // channel that the transport is ready to send if each channel (in use) is
-  // ready to send. This is more specific than just "writable"; it means the
-  // last send didn't return ENOTCONN.
-  //
-  // This should be called whenever a channel's ready-to-send state changes,
-  // or when RTCP muxing becomes active/inactive.
-  void SetTransportChannelReadyToSend(bool rtcp, bool ready);
+  // From RtpTransport - public for testing only
+  void OnTransportReadyToSend(bool ready);
 
   // Only public for unit tests.  Otherwise, consider protected.
   int SetOption(SocketType type, rtc::Socket::Option o, int val)
       override;
   int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
 
   SrtpFilter* srtp_filter() { return &srtp_filter_; }
 
   virtual cricket::MediaType media_type() = 0;
 
@@ skipping 48 matching lines @@
   bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
                 const rtc::PacketOptions& options) override;
 
   // From TransportChannel
   void OnWritableState(rtc::PacketTransportInternal* transport);
   virtual void OnPacketRead(rtc::PacketTransportInternal* transport,
                             const char* data,
                             size_t len,
                             const rtc::PacketTime& packet_time,
                             int flags);
-  void OnReadyToSend(rtc::PacketTransportInternal* transport);
 
   void OnDtlsState(DtlsTransportInternal* transport, DtlsTransportState state);
 
   void OnSelectedCandidatePairChanged(
       IceTransportInternal* ice_transport,
       CandidatePairInterface* selected_candidate_pair,
       int last_sent_packet_id,
       bool ready_to_send);
 
   bool PacketIsRtcp(const rtc::PacketTransportInternal* transport,
@@ skipping 109 matching lines @@
   rtc::Thread* const network_thread_;
   rtc::Thread* const signaling_thread_;
   rtc::AsyncInvoker invoker_;
 
   const std::string content_name_;
   std::unique_ptr<ConnectionMonitor> connection_monitor_;
 
   // Won't be set when using raw packet transports. SDP-specific thing.
   std::string transport_name_;
 
+  const bool rtcp_mux_required_;
+
   // Separate DTLS/non-DTLS pointers to support using BaseChannel without DTLS.
   // Temporary measure until more refactoring is done.
   // If non-null, "X_dtls_transport_" will always equal "X_packet_transport_".
   DtlsTransportInternal* rtp_dtls_transport_ = nullptr;
   DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;
   webrtc::RtpTransport rtp_transport_;
   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
   SrtpFilter srtp_filter_;
   RtcpMuxFilter rtcp_mux_filter_;
   BundleFilter bundle_filter_;
-  bool rtp_ready_to_send_ = false;
-  bool rtcp_ready_to_send_ = false;
   bool writable_ = false;
   bool was_ever_writable_ = false;
   bool has_received_packet_ = false;
   bool dtls_keyed_ = false;
   const bool srtp_required_ = true;
   int rtp_abs_sendtime_extn_id_ = -1;
 
   // MediaChannel related members that should be accessed from the worker
   // thread.
   MediaChannel* const media_channel_;
@@ skipping 313 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNEL_H_