mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00

Bug: webrtc:9883 Change-Id: I2a687b46e3374db0dd08b0c02dfea1482e6fb89f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161229 Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30024}
149 lines
5.8 KiB
C++
149 lines
5.8 KiB
C++
/*
|
|
* Copyright (c) 2019 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 API_TEST_NETWORK_EMULATION_NETWORK_EMULATION_INTERFACES_H_
|
|
#define API_TEST_NETWORK_EMULATION_NETWORK_EMULATION_INTERFACES_H_
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "api/units/data_rate.h"
|
|
#include "api/units/data_size.h"
|
|
#include "api/units/timestamp.h"
|
|
#include "rtc_base/copy_on_write_buffer.h"
|
|
#include "rtc_base/ip_address.h"
|
|
#include "rtc_base/socket_address.h"
|
|
|
|
namespace webrtc {
|
|
|
|
struct EmulatedIpPacket {
|
|
public:
|
|
static constexpr int kUdpHeaderSize = 8;
|
|
|
|
EmulatedIpPacket(const rtc::SocketAddress& from,
|
|
const rtc::SocketAddress& to,
|
|
rtc::CopyOnWriteBuffer data,
|
|
Timestamp arrival_time);
|
|
~EmulatedIpPacket() = default;
|
|
// This object is not copyable or assignable.
|
|
EmulatedIpPacket(const EmulatedIpPacket&) = delete;
|
|
EmulatedIpPacket& operator=(const EmulatedIpPacket&) = delete;
|
|
// This object is only moveable.
|
|
EmulatedIpPacket(EmulatedIpPacket&&) = default;
|
|
EmulatedIpPacket& operator=(EmulatedIpPacket&&) = default;
|
|
|
|
size_t size() const { return data.size(); }
|
|
const uint8_t* cdata() const { return data.cdata(); }
|
|
|
|
size_t ip_packet_size() const {
|
|
return size() + kUdpHeaderSize + ip_header_size;
|
|
}
|
|
rtc::SocketAddress from;
|
|
rtc::SocketAddress to;
|
|
// Holds the UDP payload.
|
|
rtc::CopyOnWriteBuffer data;
|
|
int ip_header_size;
|
|
Timestamp arrival_time;
|
|
};
|
|
|
|
// Interface for handling IP packets from an emulated network. This is used with
|
|
// EmulatedEndpoint to receive packets on a specific port.
|
|
class EmulatedNetworkReceiverInterface {
|
|
public:
|
|
virtual ~EmulatedNetworkReceiverInterface() = default;
|
|
|
|
virtual void OnPacketReceived(EmulatedIpPacket packet) = 0;
|
|
};
|
|
|
|
struct EmulatedNetworkStats {
|
|
int64_t packets_sent = 0;
|
|
DataSize bytes_sent = DataSize::Zero();
|
|
// Total amount of packets received with or without destination.
|
|
int64_t packets_received = 0;
|
|
// Total amount of bytes in received packets.
|
|
DataSize bytes_received = DataSize::Zero();
|
|
// Total amount of packets that were received, but no destination was found.
|
|
int64_t packets_dropped = 0;
|
|
// Total amount of bytes in dropped packets.
|
|
DataSize bytes_dropped = DataSize::Zero();
|
|
|
|
DataSize first_received_packet_size = DataSize::Zero();
|
|
DataSize first_sent_packet_size = DataSize::Zero();
|
|
|
|
Timestamp first_packet_sent_time = Timestamp::PlusInfinity();
|
|
Timestamp last_packet_sent_time = Timestamp::PlusInfinity();
|
|
Timestamp first_packet_received_time = Timestamp::PlusInfinity();
|
|
Timestamp last_packet_received_time = Timestamp::PlusInfinity();
|
|
|
|
DataRate AverageSendRate() const {
|
|
RTC_DCHECK_GE(packets_sent, 2);
|
|
return (bytes_sent - first_sent_packet_size) /
|
|
(last_packet_sent_time - first_packet_sent_time);
|
|
}
|
|
DataRate AverageReceiveRate() const {
|
|
RTC_DCHECK_GE(packets_received, 2);
|
|
return (bytes_received - first_received_packet_size) /
|
|
(last_packet_received_time - first_packet_received_time);
|
|
}
|
|
};
|
|
|
|
// EmulatedEndpoint is an abstraction for network interface on device. Instances
|
|
// of this are created by NetworkEmulationManager::CreateEndpoint.
|
|
class EmulatedEndpoint : public EmulatedNetworkReceiverInterface {
|
|
public:
|
|
// Send packet into network.
|
|
// |from| will be used to set source address for the packet in destination
|
|
// socket.
|
|
// |to| will be used for routing verification and picking right socket by port
|
|
// on destination endpoint.
|
|
virtual void SendPacket(const rtc::SocketAddress& from,
|
|
const rtc::SocketAddress& to,
|
|
rtc::CopyOnWriteBuffer packet_data) = 0;
|
|
|
|
// Binds receiver to this endpoint to send and receive data.
|
|
// |desired_port| is a port that should be used. If it is equal to 0,
|
|
// endpoint will pick the first available port starting from
|
|
// |kFirstEphemeralPort|.
|
|
//
|
|
// Returns the port, that should be used (it will be equals to desired, if
|
|
// |desired_port| != 0 and is free or will be the one, selected by endpoint)
|
|
// or absl::nullopt if desired_port in used. Also fails if there are no more
|
|
// free ports to bind to.
|
|
virtual absl::optional<uint16_t> BindReceiver(
|
|
uint16_t desired_port,
|
|
EmulatedNetworkReceiverInterface* receiver) = 0;
|
|
virtual void UnbindReceiver(uint16_t port) = 0;
|
|
virtual rtc::IPAddress GetPeerLocalAddress() const = 0;
|
|
|
|
virtual EmulatedNetworkStats stats() = 0;
|
|
|
|
private:
|
|
// Ensure that there can be no other subclass than EmulatedEndpointImpl. This
|
|
// means that it's always safe to downcast EmulatedEndpoint instances to
|
|
// EmulatedEndpointImpl.
|
|
friend class EmulatedEndpointImpl;
|
|
EmulatedEndpoint() = default;
|
|
};
|
|
|
|
// Simulates a TCP connection, this roughly implements the Reno algorithm. In
|
|
// difference from TCP this only support sending messages with a fixed length,
|
|
// no streaming. This is useful to simulate signaling and cross traffic using
|
|
// message based protocols such as HTTP. It differs from UDP messages in that
|
|
// they are guranteed to be delivered eventually, even on lossy networks.
|
|
class TcpMessageRoute {
|
|
public:
|
|
// Sends a TCP message of the given |size| over the route, |on_received| is
|
|
// called when the message has been delivered. Note that the connection
|
|
// parameters are reset iff there's no currently pending message on the route.
|
|
virtual void SendMessage(size_t size, std::function<void()> on_received) = 0;
|
|
|
|
protected:
|
|
~TcpMessageRoute() = default;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // API_TEST_NETWORK_EMULATION_NETWORK_EMULATION_INTERFACES_H_
|