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/*
* 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 <cstddef>
#include <cstdint>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/strings/string_view.h"
#include "api/field_trials_view.h"
#include "api/make_ref_counted.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/rtc_event_log_output_file.h"
#include "api/rtp_parameters.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/metrics/global_metrics_logger_and_exporter.h"
#include "api/test/metrics/metric.h"
#include "api/test/simulated_network.h"
#include "api/transport/bitrate_settings.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/video/video_codec_type.h"
#include "api/video_codecs/sdp_video_format.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/flexfec_receive_stream.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "rtc_base/checks.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "test/call_test.h"
#include "test/encoder_settings.h"
#include "test/gtest.h"
#include "test/rtp_rtcp_observer.h"
#include "test/video_test_constants.h"
#include "video/config/video_encoder_config.h"
ABSL_FLAG(std::string,
ramp_dump_name,
"",
"Filename for dumped received RTP stream.");
namespace webrtc {
namespace {
using ::webrtc::test::GetGlobalMetricsLogger;
using ::webrtc::test::ImprovementDirection;
using ::webrtc::test::Unit;
constexpr TimeDelta kPollInterval = TimeDelta::Millis(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,
bool rtx,
bool red,
bool report_perf_stats,
TaskQueueBase* task_queue)
: EndToEndTest(test::VideoTestConstants::kLongTimeout),
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),
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() = default;
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<VideoReceiveStreamInterface*>& receive_streams) {
send_stream_ = send_stream;
}
BuiltInNetworkBehaviorConfig RampUpTester::GetSendTransportConfig() const {
return forward_transport_config_;
}
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(
const FieldTrialsView& /*field_trials*/,
int frame_width,
int frame_height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(frame_width, frame_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<VideoReceiveStreamInterface::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 =
rtc::make_ref_counted<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::VideoTestConstants::kDefaultWidth,
test::VideoTestConstants::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.nack.rtp_history_ms =
test::VideoTestConstants::kNackRtpHistoryMs;
send_config->rtp.ssrcs = video_ssrcs_;
if (rtx_) {
send_config->rtp.rtx.payload_type =
test::VideoTestConstants::kSendRtxPayloadType;
send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_;
}
if (red_) {
send_config->rtp.ulpfec.ulpfec_payload_type =
test::VideoTestConstants::kUlpfecPayloadType;
send_config->rtp.ulpfec.red_payload_type =
test::VideoTestConstants::kRedPayloadType;
if (rtx_) {
send_config->rtp.ulpfec.red_rtx_payload_type =
test::VideoTestConstants::kRtxRedPayloadType;
}
}
size_t i = 0;
for (VideoReceiveStreamInterface::Config& recv_config : *receive_configs) {
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::VideoTestConstants::kFlexfecPayloadType;
send_config->rtp.flexfec.ssrc = test::VideoTestConstants::kFlexfecSendSsrc;
send_config->rtp.flexfec.protected_media_ssrcs = {video_ssrcs_[0]};
}
}
void RampUpTester::ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>* receive_configs) {
if (num_audio_streams_ == 0)
return;
send_config->rtp.ssrc = audio_ssrcs_[0];
send_config->min_bitrate_bps = 6000;
send_config->max_bitrate_bps = 60000;
for (AudioReceiveStreamInterface::Config& recv_config : *receive_configs) {
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::VideoTestConstants::kFlexfecPayloadType;
(*receive_configs)[0].rtp.remote_ssrc =
test::VideoTestConstants::kFlexfecSendSsrc;
(*receive_configs)[0].protected_media_ssrcs = {video_ssrcs_[0]};
(*receive_configs)[0].rtp.local_ssrc = video_ssrcs_[0];
}
void RampUpTester::OnCallsCreated(Call* sender_call, Call* receiver_call) {
RTC_DCHECK(sender_call);
sender_call_ = sender_call;
pending_task_ = RepeatingTaskHandle::Start(task_queue_, [this] {
PollStats();
return kPollInterval;
});
}
void RampUpTester::OnTransportCreated(
test::PacketTransport* to_receiver,
SimulatedNetworkInterface* sender_network,
test::PacketTransport* to_sender,
SimulatedNetworkInterface* receiver_network) {
RTC_DCHECK_RUN_ON(task_queue_);
send_transport_ = to_receiver;
send_simulated_network_ = sender_network;
}
void RampUpTester::PollStats() {
RTC_DCHECK_RUN_ON(task_queue_);
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();
pending_task_.Stop();
}
}
void RampUpTester::ReportResult(
absl::string_view measurement,
size_t value,
Unit unit,
ImprovementDirection improvement_direction) const {
GetGlobalMetricsLogger()->LogSingleValueMetric(
measurement,
::testing::UnitTest::GetInstance()->current_test_info()->name(), value,
unit, improvement_direction);
}
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);
// Stop polling stats.
// Corner case for webrtc_quick_perf_test
SendTask(task_queue_, [this] { pending_task_.Stop(); });
// TODO(holmer): Add audio send stats here too when those APIs are available.
if (!send_stream_)
return;
VideoSendStream::Stats send_stats;
SendTask(task_queue_, [&] { 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, Unit::kBytes,
ImprovementDirection::kBiggerIsBetter);
ReportResult("ramp-up-padding-sent", padding_sent, Unit::kBytes,
ImprovementDirection::kSmallerIsBetter);
ReportResult("ramp-up-rtx-media-sent", rtx_media_sent, Unit::kBytes,
ImprovementDirection::kBiggerIsBetter);
ReportResult("ramp-up-rtx-padding-sent", rtx_padding_sent, Unit::kBytes,
ImprovementDirection::kSmallerIsBetter);
if (ramp_up_finished_ms_ >= 0) {
ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_,
Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter);
}
ReportResult("ramp-up-average-network-latency",
send_transport_->GetAverageDelayMs(), Unit::kMilliseconds,
ImprovementDirection::kSmallerIsBetter);
}
}
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,
bool rtx,
bool red,
const std::vector<int>& loss_rates,
bool report_perf_stats,
TaskQueueBase* task_queue)
: RampUpTester(num_video_streams,
num_audio_streams,
num_flexfec_streams,
start_bitrate_bps,
0,
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 =
DataRate::KilobitsPerSec(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() {
if (test_state_ == kTestEnd) {
pending_task_.Stop();
}
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);
}
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_) {
GetGlobalMetricsLogger()->LogSingleValueMetric(
"ramp_up_down_up" + GetModifierString(), "first_rampup",
now - state_start_ms_, Unit::kMilliseconds,
ImprovementDirection::kSmallerIsBetter);
}
// 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_) {
GetGlobalMetricsLogger()->LogSingleValueMetric(
"ramp_up_down_up" + GetModifierString(), "rampdown",
now - state_start_ms_, Unit::kMilliseconds,
ImprovementDirection::kSmallerIsBetter);
}
// 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_) {
GetGlobalMetricsLogger()->LogSingleValueMetric(
"ramp_up_down_up" + GetModifierString(), "second_rampup",
now - state_start_ms_, Unit::kMilliseconds,
ImprovementDirection::kSmallerIsBetter);
ReportResult("ramp-up-down-up-average-network-latency",
send_transport_->GetAverageDelayMs(),
Unit::kMilliseconds,
ImprovementDirection::kSmallerIsBetter);
}
// 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 =
DataRate::KilobitsPerSec(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() {
std::string dump_name(absl::GetFlag(FLAGS_ramp_dump_name));
if (!dump_name.empty()) {
std::unique_ptr<RtcEventLog> send_event_log =
rtc_event_log_factory_.Create(env());
std::unique_ptr<RtcEventLog> recv_event_log =
rtc_event_log_factory_.Create(env());
bool event_log_started =
send_event_log->StartLogging(
std::make_unique<RtcEventLogOutputFile>(
dump_name + ".send.rtc.dat", RtcEventLog::kUnlimitedOutput),
RtcEventLog::kImmediateOutput) &&
recv_event_log->StartLogging(
std::make_unique<RtcEventLogOutputFile>(
dump_name + ".recv.rtc.dat", RtcEventLog::kUnlimitedOutput),
RtcEventLog::kImmediateOutput);
RTC_DCHECK(event_log_started);
SetSendEventLog(std::move(send_event_log));
SetRecvEventLog(std::move(recv_event_log));
}
}
private:
RtcEventLogFactory rtc_event_log_factory_;
};
static const uint32_t kStartBitrateBps = 60000;
TEST_F(RampUpTest, UpDownUpAbsSendTimeSimulcastRedRtx) {
std::vector<int> loss_rates = {0, 0, 0, 0};
RegisterRtpExtension(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
RampUpDownUpTester test(3, 0, 0, kStartBitrateBps, 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};
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
RampUpDownUpTester test(3, 0, 0, kStartBitrateBps, 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};
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
RampUpDownUpTester test(1, 0, 1, kStartBitrateBps, 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) {
std::vector<int> loss_rates = {0, 0, 0, 0};
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
RampUpDownUpTester test(3, 1, 0, kStartBitrateBps, true, false, loss_rates,
false, task_queue());
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpAudioTransportSequenceNumberRtx) {
std::vector<int> loss_rates = {0, 0, 0, 0};
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
RampUpDownUpTester test(0, 1, 0, kStartBitrateBps, true, false, loss_rates,
false, task_queue());
RunBaseTest(&test);
}
TEST_F(RampUpTest, TOffsetSimulcastRedRtx) {
RegisterRtpExtension(RtpExtension(RtpExtension::kTimestampOffsetUri,
kTransmissionTimeOffsetExtensionId));
RampUpTester test(3, 0, 0, 0, 0, true, true, true, task_queue());
RunBaseTest(&test);
}
TEST_F(RampUpTest, AbsSendTime) {
RegisterRtpExtension(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
RampUpTester test(1, 0, 0, 0, 0, false, false, false, task_queue());
RunBaseTest(&test);
}
TEST_F(RampUpTest, AbsSendTimeSimulcastRedRtx) {
RegisterRtpExtension(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
RampUpTester test(3, 0, 0, 0, 0, true, true, true, task_queue());
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumber) {
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
RampUpTester test(1, 0, 0, 0, 0, false, false, false, task_queue());
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSimulcast) {
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
RampUpTester test(3, 0, 0, 0, 0, false, false, false, task_queue());
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) {
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
RampUpTester test(3, 0, 0, 0, 0, true, true, true, task_queue());
RunBaseTest(&test);
}
TEST_F(RampUpTest, AudioTransportSequenceNumber) {
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
RampUpTester test(0, 1, 0, 300000, 10000, false, false, false, task_queue());
RunBaseTest(&test);
}
} // namespace webrtc