webrtc/call/rampup_tests.h
Per Kjellander 89870ffa95 Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
This reverts commit 3e61f881cd.

Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104

Original change's description:
> Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
>
> This reverts commit 3b96f2c770.
>
> Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and  CallPerfTest.Min_Bitrate_VideoAndAudio 
>
>
> Original change's description:
> > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp
> >
> > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
> > Therefore DirectTransport is provided with the extension mapping.
> >
> > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
> >
> >
> > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
> > Bug: webrtc:7135, webrtc:14795
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
> > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39137}
>
> Bug: webrtc:7135, webrtc:14795, webrtc:14833
> Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Auto-Submit: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#39146}

Bug: webrtc:7135, webrtc:14795, webrtc:14833
Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-20 06:32:29 +00:00

173 lines
5.6 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 <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/metrics/metric.h"
#include "api/test/simulated_network.h"
#include "call/call.h"
#include "call/simulated_network.h"
#include "rtc_base/event.h"
#include "rtc_base/task_utils/repeating_task.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,
absl::string_view extension_type,
bool rtx,
bool red,
bool report_perf_stats,
TaskQueueBase* task_queue);
~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(absl::string_view measurement,
size_t value,
test::Unit unit,
test::ImprovementDirection improvement_direction) const;
void TriggerTestDone();
Clock* const clock_;
BuiltInNetworkBehaviorConfig 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_;
SimulatedNetworkInterface* send_simulated_network_;
private:
typedef std::map<uint32_t, uint32_t> SsrcMap;
class VideoStreamFactory;
void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) override;
void OnVideoStreamsCreated(VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>&
receive_streams) override;
BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override;
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override;
void ModifyAudioConfigs(AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>*
receive_configs) override;
void ModifyFlexfecConfigs(
std::vector<FlexfecReceiveStream::Config>* receive_configs) override;
void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
void OnTransportCreated(test::PacketTransport* to_receiver,
SimulatedNetworkInterface* sender_network,
test::PacketTransport* to_sender,
SimulatedNetworkInterface* receiver_network) override;
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_;
protected:
TaskQueueBase* const task_queue_;
RepeatingTaskHandle pending_task_;
};
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,
absl::string_view extension_type,
bool rtx,
bool red,
const std::vector<int>& loss_rates,
bool report_perf_stats,
TaskQueueBase* task_queue);
~RampUpDownUpTester() override;
protected:
void PollStats() override;
private:
enum TestStates {
kFirstRampup = 0,
kLowRate,
kSecondRampup,
kTestEnd,
kTransitionToNextState,
};
void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) override;
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_