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/*
* 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 RTC_BASE_ASYNC_PACKET_SOCKET_H_
#define RTC_BASE_ASYNC_PACKET_SOCKET_H_
#include <vector>
#include "api/sequence_checker.h"
#include "rtc_base/callback_list.h"
#include "rtc_base/dscp.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/socket.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/system/rtc_export.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/time_utils.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();
PacketTimeUpdateParams(const PacketTimeUpdateParams& other);
~PacketTimeUpdateParams();
int rtp_sendtime_extension_id = -1; // extension header id present in packet.
std::vector<char> srtp_auth_key; // Authentication key.
int srtp_auth_tag_len = -1; // Authentication tag length.
int64_t srtp_packet_index = -1; // Required for Rtp Packet authentication.
};
// This structure holds meta information for the packet which is about to send
// over network.
struct RTC_EXPORT PacketOptions {
PacketOptions();
explicit PacketOptions(DiffServCodePoint dscp);
PacketOptions(const PacketOptions& other);
~PacketOptions();
DiffServCodePoint dscp = DSCP_NO_CHANGE;
// When used with RTP packets (for example, webrtc::PacketOptions), the value
// should be 16 bits. A value of -1 represents "not set".
int64_t packet_id = -1;
PacketTimeUpdateParams packet_time_params;
// PacketInfo is passed to SentPacket when signaling this packet is sent.
PacketInfo info_signaled_after_sent;
};
// Provides the ability to receive packets asynchronously. Sends are not
// buffered since it is acceptable to drop packets under high load.
class RTC_EXPORT AsyncPacketSocket : public sigslot::has_slots<> {
public:
enum State {
STATE_CLOSED,
STATE_BINDING,
STATE_BOUND,
STATE_CONNECTING,
STATE_CONNECTED
};
AsyncPacketSocket();
~AsyncPacketSocket() override;
AsyncPacketSocket(const AsyncPacketSocket&) = delete;
AsyncPacketSocket& operator=(const AsyncPacketSocket&) = delete;
// 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;
// Register a callback to be called when the socket is closed.
void SubscribeClose(const void* removal_tag,
std::function<void(AsyncPacketSocket*, int)> callback);
void UnsubscribeClose(const void* removal_tag);
// Emitted each time a packet is read. Used only for UDP and
// connected TCP sockets.
sigslot::signal5<AsyncPacketSocket*,
const char*,
size_t,
const SocketAddress&,
// TODO(bugs.webrtc.org/9584): Change to passing the int64_t
// timestamp by value.
const int64_t&>
SignalReadPacket;
// Emitted each time a packet is sent.
sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket;
// 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 sockets).
sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady;
// Emitted for client TCP sockets when state is changed from
// CONNECTING to CONNECTED.
sigslot::signal1<AsyncPacketSocket*> SignalConnect;
void NotifyClosedForTest(int err) { NotifyClosed(err); }
protected:
// TODO(bugs.webrtc.org/11943): Remove after updating downstream code.
void SignalClose(AsyncPacketSocket* s, int err) {
RTC_DCHECK_EQ(s, this);
NotifyClosed(err);
}
void NotifyClosed(int err) {
RTC_DCHECK_RUN_ON(&network_checker_);
on_close_.Send(this, err);
}
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker network_checker_;
private:
webrtc::CallbackList<AsyncPacketSocket*, int> on_close_
RTC_GUARDED_BY(&network_checker_);
};
// Listen socket, producing an AsyncPacketSocket when a peer connects.
class RTC_EXPORT AsyncListenSocket : public sigslot::has_slots<> {
public:
enum class State {
kClosed,
kBound,
};
// Returns current state of the socket.
virtual State GetState() const = 0;
// Returns current local address. Address may be set to null if the
// socket is not bound yet (GetState() returns kBinding).
virtual SocketAddress GetLocalAddress() const = 0;
sigslot::signal2<AsyncListenSocket*, AsyncPacketSocket*> SignalNewConnection;
};
void CopySocketInformationToPacketInfo(size_t packet_size_bytes,
const AsyncPacketSocket& socket_from,
bool is_connectionless,
rtc::PacketInfo* info);
} // namespace rtc
#endif // RTC_BASE_ASYNC_PACKET_SOCKET_H_
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