Revert of Separating SCTP code from BaseChannel/MediaChannel.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 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 1059 matching lines @@
   DMT_CONTROL = 1,
   DMT_BINARY = 2,
   DMT_TEXT = 3,
 };
 
 // Info about data received in DataMediaChannel.  For use in
 // DataMediaChannel::SignalDataReceived and in all of the signals that
 // signal fires, on up the chain.
 struct ReceiveDataParams {
   // The in-packet stream indentifier.
-  // RTP data channels use SSRCs, SCTP data channels use SIDs.
-  union {
-    uint32_t ssrc;
-    int sid;
-  };
+  // For SCTP, this is really SID, not SSRC.
+  uint32_t ssrc;
   // The type of message (binary, text, or control).
   DataMessageType type;
   // A per-stream value incremented per packet in the stream.
   int seq_num;
   // A per-stream value monotonically increasing with time.
   int timestamp;
 
-  ReceiveDataParams() : sid(0), type(DMT_TEXT), seq_num(0), timestamp(0) {}
+  ReceiveDataParams() :
+      ssrc(0),
+      type(DMT_TEXT),
+      seq_num(0),
+      timestamp(0) {
+  }
 };
 
 struct SendDataParams {
   // The in-packet stream indentifier.
-  // RTP data channels use SSRCs, SCTP data channels use SIDs.
-  union {
-    uint32_t ssrc;
-    int sid;
-  };
+  // For SCTP, this is really SID, not SSRC.
+  uint32_t ssrc;
   // The type of message (binary, text, or control).
   DataMessageType type;
 
   // For SCTP, whether to send messages flagged as ordered or not.
   // If false, messages can be received out of order.
   bool ordered;
   // For SCTP, whether the messages are sent reliably or not.
   // If false, messages may be lost.
   bool reliable;
   // For SCTP, if reliable == false, provide partial reliability by
   // resending up to this many times.  Either count or millis
   // is supported, not both at the same time.
   int max_rtx_count;
   // For SCTP, if reliable == false, provide partial reliability by
   // resending for up to this many milliseconds.  Either count or millis
   // is supported, not both at the same time.
   int max_rtx_ms;
 
-  SendDataParams()
-      : sid(0),
-        type(DMT_TEXT),
-        // TODO(pthatcher): Make these true by default?
-        ordered(false),
-        reliable(false),
-        max_rtx_count(0),
-        max_rtx_ms(0) {}
+  SendDataParams() :
+      ssrc(0),
+      type(DMT_TEXT),
+      // TODO(pthatcher): Make these true by default?
+      ordered(false),
+      reliable(false),
+      max_rtx_count(0),
+      max_rtx_ms(0) {
+  }
 };
 
 enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK };
 
 struct DataSendParameters : RtpSendParameters<DataCodec> {
   std::string ToString() const {
     std::ostringstream ost;
     // Options and extensions aren't used.
     ost << "{";
     ost << "codecs: " << VectorToString(codecs) << ", ";
@@ skipping 38 matching lines @@
       const SendDataParams& params,
       const rtc::CopyOnWriteBuffer& payload,
       SendDataResult* result = NULL) = 0;
   // Signals when data is received (params, data, len)
   sigslot::signal3<const ReceiveDataParams&,
                    const char*,
                    size_t> SignalDataReceived;
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
+  // Signal for notifying that the remote side has closed the DataChannel.
+  sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_