Move talk/app/webrtc to webrtc/api

talk/app/webrtc/datachannelinterface.h

-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-// This file contains interfaces for DataChannels
-// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel
-
-#ifndef TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_
-#define TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_
-
-#include <string>
-
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/refcount.h"
-
-
-namespace webrtc {
-
-struct DataChannelInit {
-  DataChannelInit()
-      : reliable(false),
-        ordered(true),
-        maxRetransmitTime(-1),
-        maxRetransmits(-1),
-        negotiated(false),
-        id(-1) {
-  }
-
-  bool reliable;           // Deprecated.
-  bool ordered;            // True if ordered delivery is required.
-  int maxRetransmitTime;   // The max period of time in milliseconds in which
-                           // retransmissions will be sent.  After this time, no
-                           // more retransmissions will be sent. -1 if unset.
-  int maxRetransmits;      // The max number of retransmissions. -1 if unset.
-  std::string protocol;    // This is set by the application and opaque to the
-                           // WebRTC implementation.
-  bool negotiated;         // True if the channel has been externally negotiated
-                           // and we do not send an in-band signalling in the
-                           // form of an "open" message.
-  int id;                  // The stream id, or SID, for SCTP data channels. -1
-                           // if unset.
-};
-
-struct DataBuffer {
-  DataBuffer(const rtc::Buffer& data, bool binary)
-      : data(data),
-        binary(binary) {
-  }
-  // For convenience for unit tests.
-  explicit DataBuffer(const std::string& text)
-      : data(text.data(), text.length()),
-        binary(false) {
-  }
-  size_t size() const { return data.size(); }
-
-  rtc::Buffer data;
-  // Indicates if the received data contains UTF-8 or binary data.
-  // Note that the upper layers are left to verify the UTF-8 encoding.
-  // TODO(jiayl): prefer to use an enum instead of a bool.
-  bool binary;
-};
-
-class DataChannelObserver {
- public:
-  // The data channel state have changed.
-  virtual void OnStateChange() = 0;
-  //  A data buffer was successfully received.
-  virtual void OnMessage(const DataBuffer& buffer) = 0;
-  // The data channel's buffered_amount has changed.
-  virtual void OnBufferedAmountChange(uint64_t previous_amount){};
-
- protected:
-  virtual ~DataChannelObserver() {}
-};
-
-class DataChannelInterface : public rtc::RefCountInterface {
- public:
-  // Keep in sync with DataChannel.java:State and
-  // RTCDataChannel.h:RTCDataChannelState.
-  enum DataState {
-    kConnecting,
-    kOpen,  // The DataChannel is ready to send data.
-    kClosing,
-    kClosed
-  };
-
-  static const char* DataStateString(DataState state) {
-    switch (state) {
-      case kConnecting:
-        return "connecting";
-      case kOpen:
-        return "open";
-      case kClosing:
-        return "closing";
-      case kClosed:
-        return "closed";
-    }
-    RTC_CHECK(false) << "Unknown DataChannel state: " << state;
-    return "";
-  }
-
-  virtual void RegisterObserver(DataChannelObserver* observer) = 0;
-  virtual void UnregisterObserver() = 0;
-  // The label attribute represents a label that can be used to distinguish this
-  // DataChannel object from other DataChannel objects.
-  virtual std::string label() const = 0;
-  virtual bool reliable() const = 0;
-
-  // TODO(tommyw): Remove these dummy implementations when all classes have
-  // implemented these APIs. They should all just return the values the
-  // DataChannel was created with.
-  virtual bool ordered() const { return false; }
-  virtual uint16_t maxRetransmitTime() const { return 0; }
-  virtual uint16_t maxRetransmits() const { return 0; }
-  virtual std::string protocol() const { return std::string(); }
-  virtual bool negotiated() const { return false; }
-
-  virtual int id() const = 0;
-  virtual DataState state() const = 0;
-  // The buffered_amount returns the number of bytes of application data
-  // (UTF-8 text and binary data) that have been queued using SendBuffer but
-  // have not yet been transmitted to the network.
-  virtual uint64_t buffered_amount() const = 0;
-  virtual void Close() = 0;
-  // Sends |data| to the remote peer.
-  virtual bool Send(const DataBuffer& buffer) = 0;
-
- protected:
-  virtual ~DataChannelInterface() {}
-};
-
-}  // namespace webrtc
-
-#endif  // TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_