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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-28 14:29:10 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-28 14:29:10 +0000 |
commit | 2aa4a82499d4becd2284cdb482213d541b8804dd (patch) | |
tree | b80bf8bf13c3766139fbacc530efd0dd9d54394c /third_party/libwebrtc/webrtc/rtc_base/asyncpacketsocket.h | |
parent | Initial commit. (diff) | |
download | firefox-upstream.tar.xz firefox-upstream.zip |
Adding upstream version 86.0.1.upstream/86.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/webrtc/rtc_base/asyncpacketsocket.h')
-rw-r--r-- | third_party/libwebrtc/webrtc/rtc_base/asyncpacketsocket.h | 143 |
1 files changed, 143 insertions, 0 deletions
diff --git a/third_party/libwebrtc/webrtc/rtc_base/asyncpacketsocket.h b/third_party/libwebrtc/webrtc/rtc_base/asyncpacketsocket.h new file mode 100644 index 0000000000..16f4de0822 --- /dev/null +++ b/third_party/libwebrtc/webrtc/rtc_base/asyncpacketsocket.h @@ -0,0 +1,143 @@ +/* + * 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_ASYNCPACKETSOCKET_H_ +#define RTC_BASE_ASYNCPACKETSOCKET_H_ + +#include "rtc_base/constructormagic.h" +#include "rtc_base/dscp.h" +#include "rtc_base/sigslot.h" +#include "rtc_base/socket.h" +#include "rtc_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(); + ~PacketTimeUpdateParams(); + + 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_t 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), packet_id(-1) {} + explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp), packet_id(-1) {} + + DiffServCodePoint dscp; + int packet_id; // 16 bits, -1 represents "not set". + 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_t timestamp, int64_t not_before) + : timestamp(timestamp), not_before(not_before) {} + + int64_t timestamp; // Receive time after socket delivers the data. + + // 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. + int64_t not_before; +}; + +inline PacketTime CreatePacketTime(int64_t 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(); + ~AsyncPacketSocket() override; + + // 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 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 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: + RTC_DISALLOW_COPY_AND_ASSIGN(AsyncPacketSocket); +}; + +} // namespace rtc + +#endif // RTC_BASE_ASYNCPACKETSOCKET_H_ |