blob: 8f689e715269580f72075ce2aaf0c0465f041b51 [file] [log] [blame]
/*
* Copyright (c) 2013 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 "call/rampup_tests.h"
#include <memory>
#include "absl/flags/flag.h"
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/rtc_event_log_output_file.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/task_queue_factory.h"
#include "call/fake_network_pipe.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/string_encode.h"
#include "test/encoder_settings.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/testsupport/perf_test.h"
ABSL_FLAG(std::string,
ramp_dump_name,
"",
"Filename for dumped received RTP stream.");
namespace webrtc {
namespace {
static const int64_t kPollIntervalMs = 20;
static const int kExpectedHighVideoBitrateBps = 80000;
static const int kExpectedHighAudioBitrateBps = 30000;
static const int kLowBandwidthLimitBps = 20000;
// Set target detected bitrate to slightly larger than the target bitrate to
// avoid flakiness.
static const int kLowBitrateMarginBps = 2000;
std::vector<uint32_t> GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) {
std::vector<uint32_t> ssrcs;
for (size_t i = 0; i != num_streams; ++i)
ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i));
return ssrcs;
}
} // namespace
RampUpTester::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,
test::SingleThreadedTaskQueueForTesting* task_queue)
: EndToEndTest(test::CallTest::kLongTimeoutMs),
clock_(Clock::GetRealTimeClock()),
num_video_streams_(num_video_streams),
num_audio_streams_(num_audio_streams),
num_flexfec_streams_(num_flexfec_streams),
rtx_(rtx),
red_(red),
report_perf_stats_(report_perf_stats),
sender_call_(nullptr),
send_stream_(nullptr),
send_transport_(nullptr),
send_simulated_network_(nullptr),
start_bitrate_bps_(start_bitrate_bps),
min_run_time_ms_(min_run_time_ms),
expected_bitrate_bps_(0),
test_start_ms_(-1),
ramp_up_finished_ms_(-1),
extension_type_(extension_type),
video_ssrcs_(GenerateSsrcs(num_video_streams_, 100)),
video_rtx_ssrcs_(GenerateSsrcs(num_video_streams_, 200)),
audio_ssrcs_(GenerateSsrcs(num_audio_streams_, 300)),
task_queue_(task_queue) {
if (red_)
EXPECT_EQ(0u, num_flexfec_streams_);
EXPECT_LE(num_audio_streams_, 1u);
}
RampUpTester::~RampUpTester() {
// Special case for WebRTC-QuickPerfTest/Enabled/
task_queue_->SendTask([this]() {
if (pending_task_ !=
static_cast<test::SingleThreadedTaskQueueForTesting::TaskId>(-1)) {
task_queue_->CancelTask(pending_task_);
pending_task_ = -1;
}
});
}
void RampUpTester::ModifySenderBitrateConfig(
BitrateConstraints* bitrate_config) {
if (start_bitrate_bps_ != 0) {
bitrate_config->start_bitrate_bps = start_bitrate_bps_;
}
bitrate_config->min_bitrate_bps = 10000;
}
void RampUpTester::OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) {
send_stream_ = send_stream;
}
test::PacketTransport* RampUpTester::CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) {
auto network = absl::make_unique<SimulatedNetwork>(forward_transport_config_);
send_simulated_network_ = network.get();
send_transport_ = new test::PacketTransport(
task_queue, sender_call, this, test::PacketTransport::kSender,
test::CallTest::payload_type_map_,
absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network)));
return send_transport_;
}
size_t RampUpTester::GetNumVideoStreams() const {
return num_video_streams_;
}
size_t RampUpTester::GetNumAudioStreams() const {
return num_audio_streams_;
}
size_t RampUpTester::GetNumFlexfecStreams() const {
return num_flexfec_streams_;
}
class RampUpTester::VideoStreamFactory
: public VideoEncoderConfig::VideoStreamFactoryInterface {
public:
VideoStreamFactory() {}
private:
std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(width, height, encoder_config);
if (encoder_config.number_of_streams == 1) {
streams[0].target_bitrate_bps = streams[0].max_bitrate_bps = 2000000;
}
return streams;
}
};
void RampUpTester::ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) {
send_config->suspend_below_min_bitrate = true;
encoder_config->number_of_streams = num_video_streams_;
encoder_config->max_bitrate_bps = 2000000;
encoder_config->video_stream_factory =
new rtc::RefCountedObject<RampUpTester::VideoStreamFactory>();
if (num_video_streams_ == 1) {
// For single stream rampup until 1mbps
expected_bitrate_bps_ = kSingleStreamTargetBps;
} else {
// To ensure simulcast rate allocation.
send_config->rtp.payload_name = "VP8";
encoder_config->codec_type = kVideoCodecVP8;
std::vector<VideoStream> streams = test::CreateVideoStreams(
test::CallTest::kDefaultWidth, test::CallTest::kDefaultHeight,
*encoder_config);
// For multi stream rampup until all streams are being sent. That means
// enough bitrate to send all the target streams plus the min bitrate of
// the last one.
expected_bitrate_bps_ = streams.back().min_bitrate_bps;
for (size_t i = 0; i < streams.size() - 1; ++i) {
expected_bitrate_bps_ += streams[i].target_bitrate_bps;
}
}
send_config->rtp.extensions.clear();
bool remb;
bool transport_cc;
if (extension_type_ == RtpExtension::kAbsSendTimeUri) {
remb = true;
transport_cc = false;
send_config->rtp.extensions.push_back(
RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
} else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) {
remb = false;
transport_cc = true;
send_config->rtp.extensions.push_back(RtpExtension(
extension_type_.c_str(), kTransportSequenceNumberExtensionId));
} else {
remb = true;
transport_cc = false;
send_config->rtp.extensions.push_back(RtpExtension(
extension_type_.c_str(), kTransmissionTimeOffsetExtensionId));
}
send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
send_config->rtp.ssrcs = video_ssrcs_;
if (rtx_) {
send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_;
}
if (red_) {
send_config->rtp.ulpfec.ulpfec_payload_type =
test::CallTest::kUlpfecPayloadType;
send_config->rtp.ulpfec.red_payload_type = test::CallTest::kRedPayloadType;
if (rtx_) {
send_config->rtp.ulpfec.red_rtx_payload_type =
test::CallTest::kRtxRedPayloadType;
}
}
size_t i = 0;
for (VideoReceiveStream::Config& recv_config : *receive_configs) {
recv_config.rtp.remb = remb;
recv_config.rtp.transport_cc = transport_cc;
recv_config.rtp.extensions = send_config->rtp.extensions;
recv_config.decoders.reserve(1);
recv_config.decoders[0].payload_type = send_config->rtp.payload_type;
recv_config.decoders[0].video_format =
SdpVideoFormat(send_config->rtp.payload_name);
recv_config.rtp.remote_ssrc = video_ssrcs_[i];
recv_config.rtp.nack.rtp_history_ms = send_config->rtp.nack.rtp_history_ms;
if (red_) {
recv_config.rtp.red_payload_type =
send_config->rtp.ulpfec.red_payload_type;
recv_config.rtp.ulpfec_payload_type =
send_config->rtp.ulpfec.ulpfec_payload_type;
if (rtx_) {
recv_config.rtp.rtx_associated_payload_types
[send_config->rtp.ulpfec.red_rtx_payload_type] =
send_config->rtp.ulpfec.red_payload_type;
}
}
if (rtx_) {
recv_config.rtp.rtx_ssrc = video_rtx_ssrcs_[i];
recv_config.rtp
.rtx_associated_payload_types[send_config->rtp.rtx.payload_type] =
send_config->rtp.payload_type;
}
++i;
}
RTC_DCHECK_LE(num_flexfec_streams_, 1);
if (num_flexfec_streams_ == 1) {
send_config->rtp.flexfec.payload_type = test::CallTest::kFlexfecPayloadType;
send_config->rtp.flexfec.ssrc = test::CallTest::kFlexfecSendSsrc;
send_config->rtp.flexfec.protected_media_ssrcs = {video_ssrcs_[0]};
}
}
void RampUpTester::ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) {
if (num_audio_streams_ == 0)
return;
EXPECT_NE(RtpExtension::kTimestampOffsetUri, extension_type_)
<< "Audio BWE not supported with toffset.";
EXPECT_NE(RtpExtension::kAbsSendTimeUri, extension_type_)
<< "Audio BWE not supported with abs-send-time.";
send_config->rtp.ssrc = audio_ssrcs_[0];
send_config->rtp.extensions.clear();
send_config->min_bitrate_bps = 6000;
send_config->max_bitrate_bps = 60000;
bool transport_cc = false;
if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) {
transport_cc = true;
send_config->rtp.extensions.push_back(RtpExtension(
extension_type_.c_str(), kTransportSequenceNumberExtensionId));
}
for (AudioReceiveStream::Config& recv_config : *receive_configs) {
recv_config.rtp.transport_cc = transport_cc;
recv_config.rtp.extensions = send_config->rtp.extensions;
recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
}
}
void RampUpTester::ModifyFlexfecConfigs(
std::vector<FlexfecReceiveStream::Config>* receive_configs) {
if (num_flexfec_streams_ == 0)
return;
RTC_DCHECK_EQ(1, num_flexfec_streams_);
(*receive_configs)[0].payload_type = test::CallTest::kFlexfecPayloadType;
(*receive_configs)[0].remote_ssrc = test::CallTest::kFlexfecSendSsrc;
(*receive_configs)[0].protected_media_ssrcs = {video_ssrcs_[0]};
(*receive_configs)[0].local_ssrc = video_ssrcs_[0];
if (extension_type_ == RtpExtension::kAbsSendTimeUri) {
(*receive_configs)[0].transport_cc = false;
(*receive_configs)[0].rtp_header_extensions.push_back(
RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
} else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) {
(*receive_configs)[0].transport_cc = true;
(*receive_configs)[0].rtp_header_extensions.push_back(RtpExtension(
extension_type_.c_str(), kTransportSequenceNumberExtensionId));
}
}
void RampUpTester::OnCallsCreated(Call* sender_call, Call* receiver_call) {
RTC_DCHECK(sender_call);
sender_call_ = sender_call;
pending_task_ = task_queue_->PostTask([this]() { PollStats(); });
}
void RampUpTester::PollStats() {
pending_task_ = -1;
Call::Stats stats = sender_call_->GetStats();
EXPECT_GE(expected_bitrate_bps_, 0);
if (stats.send_bandwidth_bps >= expected_bitrate_bps_ &&
(min_run_time_ms_ == -1 ||
clock_->TimeInMilliseconds() - test_start_ms_ >= min_run_time_ms_)) {
ramp_up_finished_ms_ = clock_->TimeInMilliseconds();
observation_complete_.Set();
} else {
pending_task_ = task_queue_->PostDelayedTask([this]() { PollStats(); },
kPollIntervalMs);
}
}
void RampUpTester::ReportResult(const std::string& measurement,
size_t value,
const std::string& units) const {
webrtc::test::PrintResult(
measurement, "",
::testing::UnitTest::GetInstance()->current_test_info()->name(), value,
units, false);
}
void RampUpTester::AccumulateStats(const VideoSendStream::StreamStats& stream,
size_t* total_packets_sent,
size_t* total_sent,
size_t* padding_sent,
size_t* media_sent) const {
*total_packets_sent += stream.rtp_stats.transmitted.packets +
stream.rtp_stats.retransmitted.packets +
stream.rtp_stats.fec.packets;
*total_sent += stream.rtp_stats.transmitted.TotalBytes() +
stream.rtp_stats.retransmitted.TotalBytes() +
stream.rtp_stats.fec.TotalBytes();
*padding_sent += stream.rtp_stats.transmitted.padding_bytes +
stream.rtp_stats.retransmitted.padding_bytes +
stream.rtp_stats.fec.padding_bytes;
*media_sent += stream.rtp_stats.MediaPayloadBytes();
}
void RampUpTester::TriggerTestDone() {
RTC_DCHECK_GE(test_start_ms_, 0);
// TODO(holmer): Add audio send stats here too when those APIs are available.
if (!send_stream_)
return;
// Stop polling stats.
// Corner case for field_trials=WebRTC-QuickPerfTest/Enabled/
task_queue_->SendTask([this]() {
if (pending_task_ !=
static_cast<test::SingleThreadedTaskQueueForTesting::TaskId>(-1)) {
task_queue_->CancelTask(pending_task_);
pending_task_ = -1;
}
});
VideoSendStream::Stats send_stats = send_stream_->GetStats();
send_stream_ = nullptr; // To avoid dereferencing a bad pointer.
size_t total_packets_sent = 0;
size_t total_sent = 0;
size_t padding_sent = 0;
size_t media_sent = 0;
for (uint32_t ssrc : video_ssrcs_) {
AccumulateStats(send_stats.substreams[ssrc], &total_packets_sent,
&total_sent, &padding_sent, &media_sent);
}
size_t rtx_total_packets_sent = 0;
size_t rtx_total_sent = 0;
size_t rtx_padding_sent = 0;
size_t rtx_media_sent = 0;
for (uint32_t rtx_ssrc : video_rtx_ssrcs_) {
AccumulateStats(send_stats.substreams[rtx_ssrc], &rtx_total_packets_sent,
&rtx_total_sent, &rtx_padding_sent, &rtx_media_sent);
}
if (report_perf_stats_) {
ReportResult("ramp-up-media-sent", media_sent, "bytes");
ReportResult("ramp-up-padding-sent", padding_sent, "bytes");
ReportResult("ramp-up-rtx-media-sent", rtx_media_sent, "bytes");
ReportResult("ramp-up-rtx-padding-sent", rtx_padding_sent, "bytes");
if (ramp_up_finished_ms_ >= 0) {
ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_,
"milliseconds");
}
ReportResult("ramp-up-average-network-latency",
send_transport_->GetAverageDelayMs(), "milliseconds");
}
}
void RampUpTester::PerformTest() {
test_start_ms_ = clock_->TimeInMilliseconds();
EXPECT_TRUE(Wait()) << "Timed out while waiting for ramp-up to complete.";
TriggerTestDone();
}
RampUpDownUpTester::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,
test::SingleThreadedTaskQueueForTesting* task_queue)
: RampUpTester(num_video_streams,
num_audio_streams,
num_flexfec_streams,
start_bitrate_bps,
0,
extension_type,
rtx,
red,
report_perf_stats,
task_queue),
link_rates_({4 * GetExpectedHighBitrate() / (3 * 1000),
kLowBandwidthLimitBps / 1000,
4 * GetExpectedHighBitrate() / (3 * 1000), 0}),
test_state_(kFirstRampup),
next_state_(kTransitionToNextState),
state_start_ms_(clock_->TimeInMilliseconds()),
interval_start_ms_(clock_->TimeInMilliseconds()),
sent_bytes_(0),
loss_rates_(loss_rates) {
forward_transport_config_.link_capacity_kbps = link_rates_[test_state_];
forward_transport_config_.queue_delay_ms = 100;
forward_transport_config_.loss_percent = loss_rates_[test_state_];
}
RampUpDownUpTester::~RampUpDownUpTester() {}
void RampUpDownUpTester::PollStats() {
pending_task_ = -1;
bool last_round = (test_state_ == kTestEnd);
int transmit_bitrate_bps = 0;
bool suspended = false;
if (num_video_streams_ > 0 && send_stream_) {
webrtc::VideoSendStream::Stats stats = send_stream_->GetStats();
for (const auto& it : stats.substreams) {
transmit_bitrate_bps += it.second.total_bitrate_bps;
}
suspended = stats.suspended;
}
if (num_audio_streams_ > 0 && sender_call_) {
// An audio send stream doesn't have bitrate stats, so the call send BW is
// currently used instead.
transmit_bitrate_bps = sender_call_->GetStats().send_bandwidth_bps;
}
EvolveTestState(transmit_bitrate_bps, suspended);
if (!last_round) {
pending_task_ = task_queue_->PostDelayedTask([this]() { PollStats(); },
kPollIntervalMs);
}
}
void RampUpDownUpTester::ModifyReceiverBitrateConfig(
BitrateConstraints* bitrate_config) {
bitrate_config->min_bitrate_bps = 10000;
}
std::string RampUpDownUpTester::GetModifierString() const {
std::string str("_");
if (num_video_streams_ > 0) {
str += rtc::ToString(num_video_streams_);
str += "stream";
str += (num_video_streams_ > 1 ? "s" : "");
str += "_";
}
if (num_audio_streams_ > 0) {
str += rtc::ToString(num_audio_streams_);
str += "stream";
str += (num_audio_streams_ > 1 ? "s" : "");
str += "_";
}
str += (rtx_ ? "" : "no");
str += "rtx_";
str += (red_ ? "" : "no");
str += "red";
return str;
}
int RampUpDownUpTester::GetExpectedHighBitrate() const {
int expected_bitrate_bps = 0;
if (num_audio_streams_ > 0)
expected_bitrate_bps += kExpectedHighAudioBitrateBps;
if (num_video_streams_ > 0)
expected_bitrate_bps += kExpectedHighVideoBitrateBps;
return expected_bitrate_bps;
}
size_t RampUpDownUpTester::GetFecBytes() const {
size_t flex_fec_bytes = 0;
if (num_flexfec_streams_ > 0) {
webrtc::VideoSendStream::Stats stats = send_stream_->GetStats();
for (const auto& kv : stats.substreams)
flex_fec_bytes += kv.second.rtp_stats.fec.TotalBytes();
}
return flex_fec_bytes;
}
bool RampUpDownUpTester::ExpectingFec() const {
return num_flexfec_streams_ > 0 && forward_transport_config_.loss_percent > 0;
}
void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
int64_t now = clock_->TimeInMilliseconds();
switch (test_state_) {
case kFirstRampup:
EXPECT_FALSE(suspended);
if (bitrate_bps >= GetExpectedHighBitrate()) {
if (report_perf_stats_) {
webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
"first_rampup", now - state_start_ms_, "ms",
false);
}
// Apply loss during the transition between states if FEC is enabled.
forward_transport_config_.loss_percent = loss_rates_[test_state_];
test_state_ = kTransitionToNextState;
next_state_ = kLowRate;
}
break;
case kLowRate: {
// Audio streams are never suspended.
bool check_suspend_state = num_video_streams_ > 0;
if (bitrate_bps < kLowBandwidthLimitBps + kLowBitrateMarginBps &&
suspended == check_suspend_state) {
if (report_perf_stats_) {
webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
"rampdown", now - state_start_ms_, "ms",
false);
}
// Apply loss during the transition between states if FEC is enabled.
forward_transport_config_.loss_percent = loss_rates_[test_state_];
test_state_ = kTransitionToNextState;
next_state_ = kSecondRampup;
}
break;
}
case kSecondRampup:
if (bitrate_bps >= GetExpectedHighBitrate() && !suspended) {
if (report_perf_stats_) {
webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
"second_rampup", now - state_start_ms_,
"ms", false);
ReportResult("ramp-up-down-up-average-network-latency",
send_transport_->GetAverageDelayMs(), "milliseconds");
}
// Apply loss during the transition between states if FEC is enabled.
forward_transport_config_.loss_percent = loss_rates_[test_state_];
test_state_ = kTransitionToNextState;
next_state_ = kTestEnd;
}
break;
case kTestEnd:
observation_complete_.Set();
break;
case kTransitionToNextState:
if (!ExpectingFec() || GetFecBytes() > 0) {
test_state_ = next_state_;
forward_transport_config_.link_capacity_kbps = link_rates_[test_state_];
// No loss while ramping up and down as it may affect the BWE
// negatively, making the test flaky.
forward_transport_config_.loss_percent = 0;
state_start_ms_ = now;
interval_start_ms_ = now;
sent_bytes_ = 0;
send_simulated_network_->SetConfig(forward_transport_config_);
}
break;
}
}
class RampUpTest : public test::CallTest {
public:
RampUpTest()
: task_queue_factory_(CreateDefaultTaskQueueFactory()),
rtc_event_log_factory_(task_queue_factory_.get()) {
std::string dump_name(absl::GetFlag(FLAGS_ramp_dump_name));
if (!dump_name.empty()) {
send_event_log_ = rtc_event_log_factory_.CreateRtcEventLog(
RtcEventLog::EncodingType::Legacy);
recv_event_log_ = rtc_event_log_factory_.CreateRtcEventLog(
RtcEventLog::EncodingType::Legacy);
bool event_log_started =
send_event_log_->StartLogging(
absl::make_unique<RtcEventLogOutputFile>(
dump_name + ".send.rtc.dat", RtcEventLog::kUnlimitedOutput),
RtcEventLog::kImmediateOutput) &&
recv_event_log_->StartLogging(
absl::make_unique<RtcEventLogOutputFile>(
dump_name + ".recv.rtc.dat", RtcEventLog::kUnlimitedOutput),
RtcEventLog::kImmediateOutput);
RTC_DCHECK(event_log_started);
}
}
private:
const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
RtcEventLogFactory rtc_event_log_factory_;
};
static const uint32_t kStartBitrateBps = 60000;
TEST_F(RampUpTest, UpDownUpAbsSendTimeSimulcastRedRtx) {
std::vector<int> loss_rates = {0, 0, 0, 0};
RampUpDownUpTester test(3, 0, 0, kStartBitrateBps,
RtpExtension::kAbsSendTimeUri, true, true, loss_rates,
true, &task_queue_);
RunBaseTest(&test);
}
// TODO(bugs.webrtc.org/8878)
#if defined(WEBRTC_MAC)
#define MAYBE_UpDownUpTransportSequenceNumberRtx \
DISABLED_UpDownUpTransportSequenceNumberRtx
#else
#define MAYBE_UpDownUpTransportSequenceNumberRtx \
UpDownUpTransportSequenceNumberRtx
#endif
TEST_F(RampUpTest, MAYBE_UpDownUpTransportSequenceNumberRtx) {
std::vector<int> loss_rates = {0, 0, 0, 0};
RampUpDownUpTester test(3, 0, 0, kStartBitrateBps,
RtpExtension::kTransportSequenceNumberUri, true,
false, loss_rates, true, &task_queue_);
RunBaseTest(&test);
}
// TODO(holmer): Tests which don't report perf stats should be moved to a
// different executable since they per definition are not perf tests.
// This test is disabled because it crashes on Linux, and is flaky on other
// platforms. See: crbug.com/webrtc/7919
TEST_F(RampUpTest, DISABLED_UpDownUpTransportSequenceNumberPacketLoss) {
std::vector<int> loss_rates = {20, 0, 0, 0};
RampUpDownUpTester test(1, 0, 1, kStartBitrateBps,
RtpExtension::kTransportSequenceNumberUri, true,
false, loss_rates, false, &task_queue_);
RunBaseTest(&test);
}
// TODO(bugs.webrtc.org/8878)
#if defined(WEBRTC_MAC)
#define MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx \
DISABLED_UpDownUpAudioVideoTransportSequenceNumberRtx
#else
#define MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx \
UpDownUpAudioVideoTransportSequenceNumberRtx
#endif
TEST_F(RampUpTest, MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx) {
test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
std::vector<int> loss_rates = {0, 0, 0, 0};
RampUpDownUpTester test(3, 1, 0, kStartBitrateBps,
RtpExtension::kTransportSequenceNumberUri, true,
false, loss_rates, false, &task_queue_);
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpAudioTransportSequenceNumberRtx) {
test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
std::vector<int> loss_rates = {0, 0, 0, 0};
RampUpDownUpTester test(0, 1, 0, kStartBitrateBps,
RtpExtension::kTransportSequenceNumberUri, true,
false, loss_rates, false, &task_queue_);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TOffsetSimulcastRedRtx) {
RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTimestampOffsetUri, true,
true, true, &task_queue_);
RunBaseTest(&test);
}
TEST_F(RampUpTest, AbsSendTime) {
RampUpTester test(1, 0, 0, 0, 0, RtpExtension::kAbsSendTimeUri, false, false,
false, &task_queue_);
RunBaseTest(&test);
}
TEST_F(RampUpTest, AbsSendTimeSimulcastRedRtx) {
RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kAbsSendTimeUri, true, true,
true, &task_queue_);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumber) {
RampUpTester test(1, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri,
false, false, false, &task_queue_);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSimulcast) {
RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri,
false, false, false, &task_queue_);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) {
RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri,
true, true, true, &task_queue_);
RunBaseTest(&test);
}
TEST_F(RampUpTest, AudioTransportSequenceNumber) {
RampUpTester test(0, 1, 0, 300000, 10000,
RtpExtension::kTransportSequenceNumberUri, false, false,
false, &task_queue_);
RunBaseTest(&test);
}
} // namespace webrtc