webrtc/talk/session/tunnel/pseudotcpchannel.h
henrike@webrtc.org 269fb4bc90 move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00

141 lines
5.4 KiB
C++

/*
* libjingle
* Copyright 2004--2006, 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.
*/
#ifndef TALK_SESSION_TUNNEL_PSEUDOTCPCHANNEL_H_
#define TALK_SESSION_TUNNEL_PSEUDOTCPCHANNEL_H_
#include "webrtc/p2p/base/pseudotcp.h"
#include "webrtc/p2p/base/session.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/messagequeue.h"
#include "webrtc/base/stream.h"
namespace rtc {
class Thread;
}
namespace cricket {
class Candidate;
class TransportChannel;
///////////////////////////////////////////////////////////////////////////////
// PseudoTcpChannel
// Note: The PseudoTcpChannel must persist until both of:
// 1) The StreamInterface provided via GetStream has been closed.
// This is tracked via non-null stream_.
// 2) The PseudoTcp session has completed.
// This is tracked via non-null worker_thread_. When PseudoTcp is done,
// the TransportChannel is signalled to tear-down. Once the channel is
// torn down, the worker thread is purged.
// These indicators are checked by CheckDestroy, invoked whenever one of them
// changes.
///////////////////////////////////////////////////////////////////////////////
// PseudoTcpChannel::GetStream
// Note: The stream pointer returned by GetStream is owned by the caller.
// They can close & immediately delete the stream while PseudoTcpChannel still
// has cleanup work to do. They can also close the stream but not delete it
// until long after PseudoTcpChannel has finished. We must cope with both.
///////////////////////////////////////////////////////////////////////////////
class PseudoTcpChannel
: public IPseudoTcpNotify,
public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
// Signal thread methods
PseudoTcpChannel(rtc::Thread* stream_thread,
Session* session);
bool Connect(const std::string& content_name,
const std::string& channel_name,
int component);
rtc::StreamInterface* GetStream();
sigslot::signal1<PseudoTcpChannel*> SignalChannelClosed;
// Call this when the Session used to create this channel is being torn
// down, to ensure that things get cleaned up properly.
void OnSessionTerminate(Session* session);
// See the PseudoTcp class for available options.
void GetOption(PseudoTcp::Option opt, int* value);
void SetOption(PseudoTcp::Option opt, int value);
private:
class InternalStream;
friend class InternalStream;
virtual ~PseudoTcpChannel();
// Stream thread methods
rtc::StreamState GetState() const;
rtc::StreamResult Read(void* buffer, size_t buffer_len,
size_t* read, int* error);
rtc::StreamResult Write(const void* data, size_t data_len,
size_t* written, int* error);
void Close();
// Multi-thread methods
void OnMessage(rtc::Message* pmsg);
void AdjustClock(bool clear = true);
void CheckDestroy();
// Signal thread methods
void OnChannelDestroyed(TransportChannel* channel);
// Worker thread methods
void OnChannelWritableState(TransportChannel* channel);
void OnChannelRead(TransportChannel* channel, const char* data, size_t size,
const rtc::PacketTime& packet_time, int flags);
void OnChannelConnectionChanged(TransportChannel* channel,
const Candidate& candidate);
virtual void OnTcpOpen(PseudoTcp* ptcp);
virtual void OnTcpReadable(PseudoTcp* ptcp);
virtual void OnTcpWriteable(PseudoTcp* ptcp);
virtual void OnTcpClosed(PseudoTcp* ptcp, uint32 nError);
virtual IPseudoTcpNotify::WriteResult TcpWritePacket(PseudoTcp* tcp,
const char* buffer,
size_t len);
rtc::Thread* signal_thread_, * worker_thread_, * stream_thread_;
Session* session_;
TransportChannel* channel_;
std::string content_name_;
std::string channel_name_;
PseudoTcp* tcp_;
InternalStream* stream_;
bool stream_readable_, pending_read_event_;
bool ready_to_connect_;
mutable rtc::CriticalSection cs_;
};
} // namespace cricket
#endif // TALK_SESSION_TUNNEL_PSEUDOTCPCHANNEL_H_