webrtc_m130/call/rampup_tests.h
Artem Titov 46c4e60939 Introduce SimulatedNetworkReceiverInterface.
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport
on this interface. Also switch part of related users on
DefaultNetworkSimulationConfig.

This two changes united into single CL to prevent work duplication.
Most changes were done because of stop including fake_network_pipe.h
into direct_transport.h, so splitting this into 2 CLs will require
first fix all imports of fake_network_pipe.h and then replace them
on new API imports again.

Bug: webrtc:9630
Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6
Reviewed-on: https://webrtc-review.googlesource.com/94762
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24336}
2018-08-20 07:23:41 +00:00

161 lines
4.9 KiB
C++

/*
* Copyright (c) 2014 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 CALL_RAMPUP_TESTS_H_
#define CALL_RAMPUP_TESTS_H_
#include <map>
#include <string>
#include <vector>
#include "api/test/simulated_network.h"
#include "call/call.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/event.h"
#include "test/call_test.h"
namespace webrtc {
static const int kTransmissionTimeOffsetExtensionId = 6;
static const int kAbsSendTimeExtensionId = 7;
static const int kTransportSequenceNumberExtensionId = 8;
static const unsigned int kSingleStreamTargetBps = 1000000;
class Clock;
class RampUpTester : public test::EndToEndTest {
public:
RampUpTester(size_t num_video_streams,
size_t num_audio_streams,
size_t num_flexfec_streams,
unsigned int start_bitrate_bps,
int64_t min_run_time_ms,
const std::string& extension_type,
bool rtx,
bool red,
bool report_perf_stats);
~RampUpTester() override;
size_t GetNumVideoStreams() const override;
size_t GetNumAudioStreams() const override;
size_t GetNumFlexfecStreams() const override;
void PerformTest() override;
protected:
virtual void PollStats();
void AccumulateStats(const VideoSendStream::StreamStats& stream,
size_t* total_packets_sent,
size_t* total_sent,
size_t* padding_sent,
size_t* media_sent) const;
void ReportResult(const std::string& measurement,
size_t value,
const std::string& units) const;
void TriggerTestDone();
rtc::Event stop_event_;
Clock* const clock_;
DefaultNetworkSimulationConfig forward_transport_config_;
const size_t num_video_streams_;
const size_t num_audio_streams_;
const size_t num_flexfec_streams_;
const bool rtx_;
const bool red_;
const bool report_perf_stats_;
Call* sender_call_;
VideoSendStream* send_stream_;
test::PacketTransport* send_transport_;
private:
typedef std::map<uint32_t, uint32_t> SsrcMap;
class VideoStreamFactory;
void ModifySenderCallConfig(Call::Config* config) override;
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override;
test::PacketTransport* CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) override;
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override;
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override;
void ModifyFlexfecConfigs(
std::vector<FlexfecReceiveStream::Config>* receive_configs) override;
void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
static void BitrateStatsPollingThread(void* obj);
const int start_bitrate_bps_;
const int64_t min_run_time_ms_;
int expected_bitrate_bps_;
int64_t test_start_ms_;
int64_t ramp_up_finished_ms_;
const std::string extension_type_;
std::vector<uint32_t> video_ssrcs_;
std::vector<uint32_t> video_rtx_ssrcs_;
std::vector<uint32_t> audio_ssrcs_;
rtc::PlatformThread poller_thread_;
};
class RampUpDownUpTester : public RampUpTester {
public:
RampUpDownUpTester(size_t num_video_streams,
size_t num_audio_streams,
size_t num_flexfec_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
bool rtx,
bool red,
const std::vector<int>& loss_rates,
bool report_perf_stats);
~RampUpDownUpTester() override;
protected:
void PollStats() override;
private:
enum TestStates {
kFirstRampup = 0,
kLowRate,
kSecondRampup,
kTestEnd,
kTransitionToNextState,
};
void ModifyReceiverCallConfig(Call::Config* config);
std::string GetModifierString() const;
int GetExpectedHighBitrate() const;
int GetHighLinkCapacity() const;
size_t GetFecBytes() const;
bool ExpectingFec() const;
void EvolveTestState(int bitrate_bps, bool suspended);
const std::vector<int> link_rates_;
TestStates test_state_;
TestStates next_state_;
int64_t state_start_ms_;
int64_t interval_start_ms_;
int sent_bytes_;
std::vector<int> loss_rates_;
};
} // namespace webrtc
#endif // CALL_RAMPUP_TESTS_H_