webrtc/test/scenario/network_node.h
Artem Titov 6cae2d5513 Reland "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"
This reverts commit 3f87250a4f.

Reason for revert: Downstream is fixed

Original change's description:
> Revert "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"
>
> This reverts commit 5f0eb93d2a.
>
> Reason for revert: Breaks downstream project. I'm going to fix that one and create a reland of this CL after.
>
> Original change's description:
> > Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely
> >
> > Bug: webrtc:13555, webrtc:13082
> > Change-Id: Iff2cda6f516739419e97e975e03f77a98f74be03
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249260
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Artem Titov <titovartem@webrtc.org>
> > Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
> > Cr-Commit-Position: refs/heads/main@{#35805}
>
> TBR=hta@webrtc.org,titovartem@webrtc.org,daniel.l@hpcnt.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I33d497f1132adfe6d151023195a388d9b7d548f9
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:13555, webrtc:13082
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249364
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Owners-Override: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35807}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:13555, webrtc:13082
Change-Id: I7ef1ef3b6e3c41b1a96014aa75f003c0fcf33949
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249365
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35814}
2022-01-27 12:55:44 +00:00

84 lines
2.7 KiB
C++

/*
* 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 TEST_SCENARIO_NETWORK_NODE_H_
#define TEST_SCENARIO_NETWORK_NODE_H_
#include <deque>
#include <map>
#include <memory>
#include <utility>
#include <vector>
#include "api/call/transport.h"
#include "api/units/timestamp.h"
#include "call/call.h"
#include "call/simulated_network.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue.h"
#include "test/network/network_emulation.h"
#include "test/scenario/column_printer.h"
#include "test/scenario/scenario_config.h"
namespace webrtc {
namespace test {
class SimulationNode {
public:
SimulationNode(NetworkSimulationConfig config,
SimulatedNetwork* behavior,
EmulatedNetworkNode* network_node);
static std::unique_ptr<SimulatedNetwork> CreateBehavior(
NetworkSimulationConfig config);
void UpdateConfig(std::function<void(NetworkSimulationConfig*)> modifier);
void PauseTransmissionUntil(Timestamp until);
ColumnPrinter ConfigPrinter() const;
EmulatedNetworkNode* node() { return network_node_; }
private:
NetworkSimulationConfig config_;
SimulatedNetwork* const simulation_;
EmulatedNetworkNode* const network_node_;
};
class NetworkNodeTransport : public Transport {
public:
NetworkNodeTransport(Clock* sender_clock, Call* sender_call);
~NetworkNodeTransport() override;
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override;
bool SendRtcp(const uint8_t* packet, size_t length) override;
void Connect(EmulatedEndpoint* endpoint,
const rtc::SocketAddress& receiver_address,
DataSize packet_overhead);
void Disconnect();
DataSize packet_overhead() {
MutexLock lock(&mutex_);
return packet_overhead_;
}
private:
Mutex mutex_;
Clock* const sender_clock_;
Call* const sender_call_;
EmulatedEndpoint* endpoint_ RTC_GUARDED_BY(mutex_) = nullptr;
rtc::SocketAddress local_address_ RTC_GUARDED_BY(mutex_);
rtc::SocketAddress remote_address_ RTC_GUARDED_BY(mutex_);
DataSize packet_overhead_ RTC_GUARDED_BY(mutex_) = DataSize::Zero();
rtc::NetworkRoute current_network_route_ RTC_GUARDED_BY(mutex_);
};
} // namespace test
} // namespace webrtc
#endif // TEST_SCENARIO_NETWORK_NODE_H_