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Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."

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This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.

http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457


> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
> 
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12199004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
perkj@webrtc.org
2014-05-13 08:15:48 +00:00
parent 3a5825909d
commit e9a604accd
386 changed files with 6 additions and 84973 deletions
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140
webrtc/base/asyncpacketsocket.h
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@ -1,140 +0,0 @@
/*
* Copyright 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.
*/
#ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_
#define WEBRTC_BASE_ASYNCPACKETSOCKET_H_
#include "webrtc/base/dscp.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/socket.h"
#include "webrtc/base/timeutils.h"
namespace rtc {
// This structure holds the info needed to update the packet send time header
// extension, including the information needed to update the authentication tag
// after changing the value.
struct PacketTimeUpdateParams {
PacketTimeUpdateParams()
: rtp_sendtime_extension_id(-1), srtp_auth_tag_len(-1),
srtp_packet_index(-1) {
}
int rtp_sendtime_extension_id; // extension header id present in packet.
std::vector<char> srtp_auth_key; // Authentication key.
int srtp_auth_tag_len; // Authentication tag length.
int64 srtp_packet_index; // Required for Rtp Packet authentication.
};
// This structure holds meta information for the packet which is about to send
// over network.
struct PacketOptions {
PacketOptions() : dscp(DSCP_NO_CHANGE) {}
explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp) {}
DiffServCodePoint dscp;
PacketTimeUpdateParams packet_time_params;
};
// This structure will have the information about when packet is actually
// received by socket.
struct PacketTime {
PacketTime() : timestamp(-1), not_before(-1) {}
PacketTime(int64 timestamp, int64 not_before)
: timestamp(timestamp), not_before(not_before) {
}
int64 timestamp; // Receive time after socket delivers the data.
int64 not_before; // Earliest possible time the data could have arrived,
// indicating the potential error in the |timestamp| value,
// in case the system, is busy. For example, the time of
// the last select() call.
// If unknown, this value will be set to zero.
};
inline PacketTime CreatePacketTime(int64 not_before) {
return PacketTime(TimeMicros(), not_before);
}
// Provides the ability to receive packets asynchronously. Sends are not
// buffered since it is acceptable to drop packets under high load.
class AsyncPacketSocket : public sigslot::has_slots<> {
public:
enum State {
STATE_CLOSED,
STATE_BINDING,
STATE_BOUND,
STATE_CONNECTING,
STATE_CONNECTED
};
AsyncPacketSocket() { }
virtual ~AsyncPacketSocket() { }
// Returns current local address. Address may be set to NULL if the
// socket is not bound yet (GetState() returns STATE_BINDING).
virtual SocketAddress GetLocalAddress() const = 0;
// Returns remote address. Returns zeroes if this is not a client TCP socket.
virtual SocketAddress GetRemoteAddress() const = 0;
// Send a packet.
virtual int Send(const void *pv, size_t cb, const PacketOptions& options) = 0;
virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
const PacketOptions& options) = 0;
// Close the socket.
virtual int Close() = 0;
// Returns current state of the socket.
virtual State GetState() const = 0;
// Get/set options.
virtual int GetOption(Socket::Option opt, int* value) = 0;
virtual int SetOption(Socket::Option opt, int value) = 0;
// Get/Set current error.
// TODO: Remove SetError().
virtual int GetError() const = 0;
virtual void SetError(int error) = 0;
// Emitted each time a packet is read. Used only for UDP and
// connected TCP sockets.
sigslot::signal5<AsyncPacketSocket*, const char*, size_t,
const SocketAddress&,
const PacketTime&> SignalReadPacket;
// Emitted when the socket is currently able to send.
sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend;
// Emitted after address for the socket is allocated, i.e. binding
// is finished. State of the socket is changed from BINDING to BOUND
// (for UDP and server TCP sockets) or CONNECTING (for client TCP
// sockets).
sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady;
// Emitted for client TCP sockets when state is changed from
// CONNECTING to CONNECTED.
sigslot::signal1<AsyncPacketSocket*> SignalConnect;
// Emitted for client TCP sockets when state is changed from
// CONNECTED to CLOSED.
sigslot::signal2<AsyncPacketSocket*, int> SignalClose;
// Used only for listening TCP sockets.
sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection;
private:
DISALLOW_EVIL_CONSTRUCTORS(AsyncPacketSocket);
};
} // namespace rtc
#endif // WEBRTC_BASE_ASYNCPACKETSOCKET_H_