webrtc_m130/talk/app/webrtc/datachannel.h
jiayl@webrtc.org b43c99de29 Limits the send and receive buffer by bytes, not by packets.
The new limit is 16MB for each buffer.
Also refactors the code to handle send failure more consistently.

BUG=3429
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21559005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6511 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 17:11:14 +00:00

274 lines
9.3 KiB
C++

/*
* 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.
*/
#ifndef TALK_APP_WEBRTC_DATACHANNEL_H_
#define TALK_APP_WEBRTC_DATACHANNEL_H_
#include <string>
#include <deque>
#include "talk/app/webrtc/datachannelinterface.h"
#include "talk/app/webrtc/proxy.h"
#include "talk/base/messagehandler.h"
#include "talk/base/scoped_ref_ptr.h"
#include "talk/base/sigslot.h"
#include "talk/media/base/mediachannel.h"
#include "talk/session/media/channel.h"
namespace webrtc {
class DataChannel;
class DataChannelProviderInterface {
public:
// Sends the data to the transport.
virtual bool SendData(const cricket::SendDataParams& params,
const talk_base::Buffer& payload,
cricket::SendDataResult* result) = 0;
// Connects to the transport signals.
virtual bool ConnectDataChannel(DataChannel* data_channel) = 0;
// Disconnects from the transport signals.
virtual void DisconnectDataChannel(DataChannel* data_channel) = 0;
// Adds the data channel SID to the transport for SCTP.
virtual void AddSctpDataStream(uint32 sid) = 0;
// Removes the data channel SID from the transport for SCTP.
virtual void RemoveSctpDataStream(uint32 sid) = 0;
// Returns true if the transport channel is ready to send data.
virtual bool ReadyToSendData() const = 0;
protected:
virtual ~DataChannelProviderInterface() {}
};
struct InternalDataChannelInit : public DataChannelInit {
enum OpenHandshakeRole {
kOpener,
kAcker,
kNone
};
// The default role is kOpener because the default |negotiated| is false.
InternalDataChannelInit() : open_handshake_role(kOpener) {}
explicit InternalDataChannelInit(const DataChannelInit& base)
: DataChannelInit(base), open_handshake_role(kOpener) {
// If the channel is externally negotiated, do not send the OPEN message.
if (base.negotiated) {
open_handshake_role = kNone;
}
}
OpenHandshakeRole open_handshake_role;
};
// DataChannel is a an implementation of the DataChannelInterface based on
// libjingle's data engine. It provides an implementation of unreliable or
// reliabledata channels. Currently this class is specifically designed to use
// both RtpDataEngine and SctpDataEngine.
// DataChannel states:
// kConnecting: The channel has been created the transport might not yet be
// ready.
// kOpen: The channel have a local SSRC set by a call to UpdateSendSsrc
// and a remote SSRC set by call to UpdateReceiveSsrc and the transport
// has been writable once.
// kClosing: DataChannelInterface::Close has been called or UpdateReceiveSsrc
// has been called with SSRC==0
// kClosed: Both UpdateReceiveSsrc and UpdateSendSsrc has been called with
// SSRC==0.
class DataChannel : public DataChannelInterface,
public sigslot::has_slots<>,
public talk_base::MessageHandler {
public:
static talk_base::scoped_refptr<DataChannel> Create(
DataChannelProviderInterface* provider,
cricket::DataChannelType dct,
const std::string& label,
const InternalDataChannelInit& config);
virtual void RegisterObserver(DataChannelObserver* observer);
virtual void UnregisterObserver();
virtual std::string label() const { return label_; }
virtual bool reliable() const;
virtual bool ordered() const { return config_.ordered; }
virtual uint16 maxRetransmitTime() const {
return config_.maxRetransmitTime;
}
virtual uint16 maxRetransmits() const {
return config_.maxRetransmits;
}
virtual std::string protocol() const { return config_.protocol; }
virtual bool negotiated() const { return config_.negotiated; }
virtual int id() const { return config_.id; }
virtual uint64 buffered_amount() const;
virtual void Close();
virtual DataState state() const { return state_; }
virtual bool Send(const DataBuffer& buffer);
// talk_base::MessageHandler override.
virtual void OnMessage(talk_base::Message* msg);
// Called if the underlying data engine is closing.
void OnDataEngineClose();
// Called when the channel's ready to use. That can happen when the
// underlying DataMediaChannel becomes ready, or when this channel is a new
// stream on an existing DataMediaChannel, and we've finished negotiation.
void OnChannelReady(bool writable);
// Sigslots from cricket::DataChannel
void OnDataReceived(cricket::DataChannel* channel,
const cricket::ReceiveDataParams& params,
const talk_base::Buffer& payload);
// The remote peer request that this channel should be closed.
void RemotePeerRequestClose();
// The following methods are for SCTP only.
// Sets the SCTP sid and adds to transport layer if not set yet. Should only
// be called once.
void SetSctpSid(int sid);
// Called when the transport channel is created.
void OnTransportChannelCreated();
// The following methods are for RTP only.
// Set the SSRC this channel should use to send data on the
// underlying data engine. |send_ssrc| == 0 means that the channel is no
// longer part of the session negotiation.
void SetSendSsrc(uint32 send_ssrc);
// Set the SSRC this channel should use to receive data from the
// underlying data engine.
void SetReceiveSsrc(uint32 receive_ssrc);
cricket::DataChannelType data_channel_type() const {
return data_channel_type_;
}
protected:
DataChannel(DataChannelProviderInterface* client,
cricket::DataChannelType dct,
const std::string& label);
virtual ~DataChannel();
private:
// A packet queue which tracks the total queued bytes. Queued packets are
// owned by this class.
class PacketQueue {
public:
PacketQueue();
~PacketQueue();
size_t byte_count() const {
return byte_count_;
}
bool Empty() const;
DataBuffer* Front();
void Pop();
void Push(DataBuffer* packet);
void Clear();
void Swap(PacketQueue* other);
private:
std::deque<DataBuffer*> packets_;
size_t byte_count_;
};
bool Init(const InternalDataChannelInit& config);
void DoClose();
void UpdateState();
void SetState(DataState state);
void DisconnectFromTransport();
void DeliverQueuedReceivedData();
void SendQueuedDataMessages();
bool SendDataMessage(const DataBuffer& buffer);
bool QueueSendDataMessage(const DataBuffer& buffer);
void SendQueuedControlMessages();
void QueueControlMessage(const talk_base::Buffer& buffer);
bool SendControlMessage(const talk_base::Buffer& buffer);
std::string label_;
InternalDataChannelInit config_;
DataChannelObserver* observer_;
DataState state_;
cricket::DataChannelType data_channel_type_;
DataChannelProviderInterface* provider_;
bool waiting_for_open_ack_;
bool was_ever_writable_;
bool connected_to_provider_;
bool send_ssrc_set_;
bool receive_ssrc_set_;
uint32 send_ssrc_;
uint32 receive_ssrc_;
// Control messages that always have to get sent out before any queued
// data.
PacketQueue queued_control_data_;
PacketQueue queued_received_data_;
PacketQueue queued_send_data_;
};
class DataChannelFactory {
public:
virtual talk_base::scoped_refptr<DataChannel> CreateDataChannel(
const std::string& label,
const InternalDataChannelInit* config) = 0;
protected:
virtual ~DataChannelFactory() {}
};
// Define proxy for DataChannelInterface.
BEGIN_PROXY_MAP(DataChannel)
PROXY_METHOD1(void, RegisterObserver, DataChannelObserver*)
PROXY_METHOD0(void, UnregisterObserver)
PROXY_CONSTMETHOD0(std::string, label)
PROXY_CONSTMETHOD0(bool, reliable)
PROXY_CONSTMETHOD0(bool, ordered)
PROXY_CONSTMETHOD0(uint16, maxRetransmitTime)
PROXY_CONSTMETHOD0(uint16, maxRetransmits)
PROXY_CONSTMETHOD0(std::string, protocol)
PROXY_CONSTMETHOD0(bool, negotiated)
PROXY_CONSTMETHOD0(int, id)
PROXY_CONSTMETHOD0(DataState, state)
PROXY_CONSTMETHOD0(uint64, buffered_amount)
PROXY_METHOD0(void, Close)
PROXY_METHOD1(bool, Send, const DataBuffer&)
END_PROXY()
} // namespace webrtc
#endif // TALK_APP_WEBRTC_DATACHANNEL_H_