Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

webrtc/p2p/base/packettransportinterface.h

-/*
- *  Copyright 2016 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_P2P_BASE_PACKETTRANSPORTINTERFACE_H_
-#define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_
-
-#include <string>
-#include <vector>
-
-// This is included for PacketOptions.
-#include "webrtc/base/asyncpacketsocket.h"
-#include "webrtc/base/sigslot.h"
-#include "webrtc/base/socket.h"
-
-namespace cricket {
-class TransportChannel;
-}
-
-namespace rtc {
-struct PacketOptions;
-struct PacketTime;
-struct SentPacket;
-
-class PacketTransportInterface : public sigslot::has_slots<> {
- public:
-  virtual ~PacketTransportInterface() {}
-
-  // Identify the object for logging and debug purpose.
-  virtual std::string debug_name() const = 0;
-
-  // The transport has been established.
-  virtual bool writable() const = 0;
-
-  // The transport has received a packet in the last X milliseconds, here X is
-  // configured by each implementation.
-  virtual bool receiving() const = 0;
-
-  // Attempts to send the given packet.
-  // The return value is < 0 on failure. The return value in failure case is not
-  // descriptive. Depending on failure cause and implementation details
-  // GetError() returns an descriptive errno.h error value.
-  // This mimics posix socket send() or sendto() behavior.
-  // TODO(johan): Reliable, meaningful, consistent error codes for all
-  // implementations would be nice.
-  // TODO(johan): Remove the default argument once channel code is updated.
-  virtual int SendPacket(const char* data,
-                         size_t len,
-                         const rtc::PacketOptions& options,
-                         int flags = 0) = 0;
-
-  // Sets a socket option. Note that not all options are
-  // supported by all transport types.
-  virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
-
-  // TODO(pthatcher): Once Chrome's MockPacketTransportInterface implements
-  // this, remove the default implementation.
-  virtual bool GetOption(rtc::Socket::Option opt, int* value) { return false; }
-
-  // Returns the most recent error that occurred on this channel.
-  virtual int GetError() = 0;
-
-  // Emitted when the writable state, represented by |writable()|, changes.
-  sigslot::signal1<PacketTransportInterface*> SignalWritableState;
-
-  //  Emitted when the PacketTransportInterface is ready to send packets. "Ready
-  //  to send" is more sensitive than the writable state; a transport may be
-  //  writable, but temporarily not able to send packets. For example, the
-  //  underlying transport's socket buffer may be full, as indicated by
-  //  SendPacket's return code and/or GetError.
-  sigslot::signal1<PacketTransportInterface*> SignalReadyToSend;
-
-  // Emitted when receiving state changes to true.
-  sigslot::signal1<PacketTransportInterface*> SignalReceivingState;
-
-  // Signalled each time a packet is received on this channel.
-  sigslot::signal5<PacketTransportInterface*,
-                   const char*,
-                   size_t,
-                   const rtc::PacketTime&,
-                   int>
-      SignalReadPacket;
-
-  // Signalled each time a packet is sent on this channel.
-  sigslot::signal2<PacketTransportInterface*, const rtc::SentPacket&>
-      SignalSentPacket;
-};
-
-}  // namespace rtc
-
-#endif  // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_