telegram/TMessagesProj/jni/voip/tgcalls/v2/RawTcpSocket.cpp
2025-11-22 14:04:28 +01:00

111 lines
3.2 KiB
C++

/*
* 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.
*/
#include "RawTcpSocket.h"
#include <stdint.h>
#include <string.h>
#include <algorithm>
#include <cstddef>
#include <cstdint>
#include <memory>
#include "api/array_view.h"
#include "rtc_base/byte_order.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/time_utils.h" // for TimeMillis
#if defined(WEBRTC_POSIX)
#include <errno.h>
#endif // WEBRTC_POSIX
namespace rtc {
static const size_t kMaxPacketSize = 64 * 1024;
static const size_t kBufSize = kMaxPacketSize + 4;
// RawTcpSocket
// Binds and connects `socket` and creates RawTcpSocket for
// it. Takes ownership of `socket`. Returns null if bind() or
// connect() fail (`socket` is destroyed in that case).
RawTcpSocket* RawTcpSocket::Create(Socket* socket,
const SocketAddress& bind_address,
const SocketAddress& remote_address) {
return new RawTcpSocket(
AsyncTCPSocketBase::ConnectSocket(socket, bind_address, remote_address));
}
RawTcpSocket::RawTcpSocket(Socket* socket)
: AsyncTCPSocketBase(socket, kBufSize) {}
int RawTcpSocket::Send(const void* pv,
size_t cb,
const rtc::PacketOptions& options) {
if (cb > kBufSize) {
SetError(EMSGSIZE);
return -1;
}
// If we are blocking on send, then silently drop this packet
if (!IsOutBufferEmpty())
return static_cast<int>(cb);
if (!did_send_mtproto_prologue_) {
did_send_mtproto_prologue_ = true;
uint32_t prologue = 0xeeeeeeee;
AppendToOutBuffer(&prologue, 4);
}
uint32_t pkt_len = (uint32_t)cb;
AppendToOutBuffer(&pkt_len, 4);
AppendToOutBuffer(pv, cb);
int res = FlushOutBuffer();
if (res <= 0) {
// drop packet if we made no progress
ClearOutBuffer();
return res;
}
rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(),
options.info_signaled_after_sent);
CopySocketInformationToPacketInfo(cb, *this, false, &sent_packet.info);
SignalSentPacket(this, sent_packet);
// We claim to have sent the whole thing, even if we only sent partial
return static_cast<int>(cb);
}
size_t RawTcpSocket::ProcessInput(rtc::ArrayView<const uint8_t> data) {
SocketAddress remote_addr(GetRemoteAddress());
size_t processed_bytes = 0;
while (true) {
size_t bytes_left = data.size() - processed_bytes;
if (bytes_left < 4)
return processed_bytes;
uint32_t pkt_len = rtc::GetLE32(data.data() + processed_bytes);
if (bytes_left < 4 + pkt_len)
return processed_bytes;
rtc::ReceivedPacket received_packet(
data.subview(processed_bytes + 4, pkt_len), remote_addr,
webrtc::Timestamp::Micros(rtc::TimeMicros()));
NotifyPacketReceived(received_packet);
processed_bytes += 4 + pkt_len;
}
}
} // namespace rtc