Files
platform-external-webrtc/test/scenario/scenario_config.cc
Sebastian Jansson 71a091e24e Adds simulated time scenario client.
Adds SimulatedTimeClient, a class that simulates time so congestion
controllers can be tested using the Scenario test framework without
running in real time.

This allows using simplified scenario tests as unit tests, narrowing
the gap between end to end tests and unit tests.

Bug: webrtc:9510
Change-Id: I61ab388bd610f636b926675b1f14b8d85e3c1114
Reviewed-on: https://webrtc-review.googlesource.com/99801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24890}
2018-09-28 12:30:44 +00:00

61 lines
2.3 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.
*/
#include "test/scenario/scenario_config.h"
namespace webrtc {
namespace test {
TransportControllerConfig::Rates::Rates() = default;
TransportControllerConfig::Rates::Rates(
const TransportControllerConfig::Rates&) = default;
TransportControllerConfig::Rates::~Rates() = default;
PacketStreamConfig::PacketStreamConfig() = default;
PacketStreamConfig::PacketStreamConfig(const PacketStreamConfig&) = default;
PacketStreamConfig::~PacketStreamConfig() = default;
VideoStreamConfig::Encoder::Encoder() = default;
VideoStreamConfig::Encoder::Encoder(const VideoStreamConfig::Encoder&) =
default;
VideoStreamConfig::Encoder::~Encoder() = default;
VideoStreamConfig::Stream::Stream() = default;
VideoStreamConfig::Stream::Stream(const VideoStreamConfig::Stream&) = default;
VideoStreamConfig::Stream::~Stream() = default;
AudioStreamConfig::AudioStreamConfig() = default;
AudioStreamConfig::AudioStreamConfig(const AudioStreamConfig&) = default;
AudioStreamConfig::~AudioStreamConfig() = default;
AudioStreamConfig::Encoder::Encoder() = default;
AudioStreamConfig::Encoder::Encoder(const AudioStreamConfig::Encoder&) =
default;
AudioStreamConfig::Encoder::~Encoder() = default;
AudioStreamConfig::Stream::Stream() = default;
AudioStreamConfig::Stream::Stream(const AudioStreamConfig::Stream&) = default;
AudioStreamConfig::Stream::~Stream() = default;
NetworkNodeConfig::NetworkNodeConfig() = default;
NetworkNodeConfig::NetworkNodeConfig(const NetworkNodeConfig&) = default;
NetworkNodeConfig::~NetworkNodeConfig() = default;
NetworkNodeConfig::Simulation::Simulation() = default;
NetworkNodeConfig::Simulation::Simulation(
const NetworkNodeConfig::Simulation&) = default;
NetworkNodeConfig::Simulation::~Simulation() = default;
CrossTrafficConfig::CrossTrafficConfig() = default;
CrossTrafficConfig::CrossTrafficConfig(const CrossTrafficConfig&) = default;
CrossTrafficConfig::~CrossTrafficConfig() = default;
} // namespace test
} // namespace webrtc