webrtc_m130/webrtc/call/rampup_tests.h
nisse e5ad5ca06a Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
Reason for revert:
Intend to fix perf failures and reland.

Original issue's description:
> Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
>
> Reason for revert:
> Reverting since this seems to break multiple WebRTC Perf buildbots
>
> Original issue's description:
> > Don't hardcode MediaType::ANY in FakeNetworkPipe.
> >
> > Instead let each test set the appropriate media type. This simplifies
> > demuxing in Call and later in RtpTransportController.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2774463003
> > Cr-Commit-Position: refs/heads/master@{#17418}
> > Committed: 9c47b00e24
>
> TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2784543002
> Cr-Commit-Position: refs/heads/master@{#17427}
> Committed: 3a3bd50610

TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2783853002
Cr-Commit-Position: refs/heads/master@{#17459}
2017-03-30 06:57:43 +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 WEBRTC_CALL_RAMPUP_TESTS_H_
#define WEBRTC_CALL_RAMPUP_TESTS_H_
#include <map>
#include <string>
#include <vector>
#include "webrtc/base/event.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/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();
webrtc::RtcEventLogNullImpl event_log_;
rtc::Event stop_event_;
Clock* const clock_;
FakeNetworkPipe::Config 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;
Call::Config GetSenderCallConfig() override;
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override;
MediaType SelectMediaType();
test::PacketTransport* CreateSendTransport(Call* sender_call) override;
test::PacketTransport* CreateReceiveTransport() 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,
};
Call::Config GetReceiverCallConfig() 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 // WEBRTC_CALL_RAMPUP_TESTS_H_