Repo created

TMessagesProj/jni/voip/webrtc/rtc_base/network/BUILD.gn

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+# Copyright (c) 2018 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.
+
+import("../../webrtc.gni")
+
+rtc_library("sent_packet") {
+  sources = [
+    "sent_packet.cc",
+    "sent_packet.h",
+  ]
+  deps = [ "../system:rtc_export" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("received_packet") {
+  visibility = [ "*" ]
+  sources = [
+    "received_packet.cc",
+    "received_packet.h",
+  ]
+  deps = [
+    "..:socket_address",
+    "../../api:array_view",
+    "../../api/units:timestamp",
+    "../system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

TMessagesProj/jni/voip/webrtc/rtc_base/network/received_packet.cc

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+/*
+ *  Copyright 2023 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 "rtc_base/network/received_packet.h"
+
+#include <utility>
+
+#include "absl/types/optional.h"
+
+namespace rtc {
+
+ReceivedPacket::ReceivedPacket(rtc::ArrayView<const uint8_t> payload,
+                               const SocketAddress& source_address,
+                               absl::optional<webrtc::Timestamp> arrival_time)
+    : payload_(payload),
+      arrival_time_(std::move(arrival_time)),
+      source_address_(source_address) {}
+
+// static
+ReceivedPacket ReceivedPacket::CreateFromLegacy(
+    const uint8_t* data,
+    size_t size,
+    int64_t packet_time_us,
+    const rtc::SocketAddress& source_address) {
+  RTC_DCHECK(packet_time_us == -1 || packet_time_us >= 0);
+  return ReceivedPacket(rtc::MakeArrayView(data, size), source_address,
+                        (packet_time_us >= 0)
+                            ? absl::optional<webrtc::Timestamp>(
+                                  webrtc::Timestamp::Micros(packet_time_us))
+                            : absl::nullopt);
+}
+
+}  // namespace rtc

TMessagesProj/jni/voip/webrtc/rtc_base/network/received_packet.h

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+/*
+ *  Copyright 2023 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_NETWORK_RECEIVED_PACKET_H_
+#define RTC_BASE_NETWORK_RECEIVED_PACKET_H_
+
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace rtc {
+
+// ReceivedPacket repressent a received IP packet.
+// It contains a payload and metadata.
+// ReceivedPacket itself does not put constraints on what payload contains. For
+// example it may contains STUN, SCTP, SRTP, RTP, RTCP.... etc.
+class RTC_EXPORT ReceivedPacket {
+ public:
+  // Caller must keep memory pointed to by payload and address valid for the
+  // lifetime of this ReceivedPacket.
+  ReceivedPacket(
+      rtc::ArrayView<const uint8_t> payload,
+      const SocketAddress& source_address,
+      absl::optional<webrtc::Timestamp> arrival_time = absl::nullopt);
+
+  // Address/port of the packet sender.
+  const SocketAddress& source_address() const { return source_address_; }
+  rtc::ArrayView<const uint8_t> payload() const { return payload_; }
+
+  // Timestamp when this packet was received. Not available on all socket
+  // implementations.
+  absl::optional<webrtc::Timestamp> arrival_time() const {
+    return arrival_time_;
+  }
+
+  static ReceivedPacket CreateFromLegacy(
+      const char* data,
+      size_t size,
+      int64_t packet_time_us,
+      const rtc::SocketAddress& addr = rtc::SocketAddress()) {
+    return CreateFromLegacy(reinterpret_cast<const uint8_t*>(data), size,
+                            packet_time_us, addr);
+  }
+
+  static ReceivedPacket CreateFromLegacy(
+      const uint8_t* data,
+      size_t size,
+      int64_t packet_time_us,
+      const rtc::SocketAddress& = rtc::SocketAddress());
+
+ private:
+  rtc::ArrayView<const uint8_t> payload_;
+  absl::optional<webrtc::Timestamp> arrival_time_;
+  const SocketAddress& source_address_;
+};
+
+}  // namespace rtc
+#endif  // RTC_BASE_NETWORK_RECEIVED_PACKET_H_

TMessagesProj/jni/voip/webrtc/rtc_base/network/sent_packet.cc

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+/*
+ *  Copyright 2018 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 "rtc_base/network/sent_packet.h"
+
+namespace rtc {
+
+PacketInfo::PacketInfo() = default;
+PacketInfo::PacketInfo(const PacketInfo& info) = default;
+PacketInfo::~PacketInfo() = default;
+
+SentPacket::SentPacket() = default;
+SentPacket::SentPacket(int64_t packet_id, int64_t send_time_ms)
+    : packet_id(packet_id), send_time_ms(send_time_ms) {}
+SentPacket::SentPacket(int64_t packet_id,
+                       int64_t send_time_ms,
+                       const rtc::PacketInfo& info)
+    : packet_id(packet_id), send_time_ms(send_time_ms), info(info) {}
+
+}  // namespace rtc

TMessagesProj/jni/voip/webrtc/rtc_base/network/sent_packet.h

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+/*
+ *  Copyright 2018 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_NETWORK_SENT_PACKET_H_
+#define RTC_BASE_NETWORK_SENT_PACKET_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace rtc {
+
+enum class PacketType {
+  kUnknown,
+  kData,
+  kIceConnectivityCheck,
+  kIceConnectivityCheckResponse,
+  kStunMessage,
+  kTurnMessage,
+};
+
+enum class PacketInfoProtocolType {
+  kUnknown,
+  kUdp,
+  kTcp,
+  kSsltcp,
+  kTls,
+};
+
+struct RTC_EXPORT PacketInfo {
+  PacketInfo();
+  PacketInfo(const PacketInfo& info);
+  ~PacketInfo();
+
+  bool included_in_feedback = false;
+  bool included_in_allocation = false;
+  PacketType packet_type = PacketType::kUnknown;
+  PacketInfoProtocolType protocol = PacketInfoProtocolType::kUnknown;
+  // A unique id assigned by the network manager, and absl::nullopt if not set.
+  absl::optional<uint16_t> network_id;
+  size_t packet_size_bytes = 0;
+  size_t turn_overhead_bytes = 0;
+  size_t ip_overhead_bytes = 0;
+};
+
+struct RTC_EXPORT SentPacket {
+  SentPacket();
+  SentPacket(int64_t packet_id, int64_t send_time_ms);
+  SentPacket(int64_t packet_id,
+             int64_t send_time_ms,
+             const rtc::PacketInfo& info);
+
+  int64_t packet_id = -1;
+  int64_t send_time_ms = -1;
+  rtc::PacketInfo info;
+};
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_NETWORK_SENT_PACKET_H_