Repo created

TMessagesProj/jni/voip/tgcalls/v2/RawTcpSocket.cpp

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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.
+ */
+
+#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