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=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/base/asyncpacketsocket.h b/webrtc/base/asyncpacketsocket.h
new file mode 100644
index 0000000..dd91ea1
--- /dev/null
+++ b/webrtc/base/asyncpacketsocket.h
@@ -0,0 +1,140 @@
+/*
+ * 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_