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webrtc_m130/webrtc/video/rampup_tests.cc
pbos@webrtc.org 6ae48c6609 Make VideoSendStream/VideoReceiveStream configs const. 
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.

CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.

This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.

R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
BUG=3260

Review URL: https://webrtc-codereview.appspot.com/20409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 10:49:19 +00:00

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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 <assert.h>
#include <map>
#include <string>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/call.h"
#include "webrtc/common.h"
#include "webrtc/experiments.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/video/transport_adapter.h"
namespace webrtc {
namespace {
static const int kAbsoluteSendTimeExtensionId = 7;
static const int kMaxPacketSize = 1500;
class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
public:
typedef std::map<uint32_t, int> BytesSentMap;
typedef std::map<uint32_t, uint32_t> SsrcMap;
StreamObserver(const SsrcMap& rtx_media_ssrcs,
newapi::Transport* feedback_transport,
Clock* clock)
: clock_(clock),
test_done_(EventWrapper::Create()),
rtp_parser_(RtpHeaderParser::Create()),
feedback_transport_(feedback_transport),
receive_stats_(ReceiveStatistics::Create(clock)),
payload_registry_(
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
crit_(CriticalSectionWrapper::CreateCriticalSection()),
expected_bitrate_bps_(0),
rtx_media_ssrcs_(rtx_media_ssrcs),
total_sent_(0),
padding_sent_(0),
rtx_media_sent_(0),
total_packets_sent_(0),
padding_packets_sent_(0),
rtx_media_packets_sent_(0) {
// Ideally we would only have to instantiate an RtcpSender, an
// RtpHeaderParser and a RemoteBitrateEstimator here, but due to the current
// state of the RTP module we need a full module and receive statistics to
// be able to produce an RTCP with REMB.
RtpRtcp::Configuration config;
config.receive_statistics = receive_stats_.get();
feedback_transport_.Enable();
config.outgoing_transport = &feedback_transport_;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetREMBStatus(true);
rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId);
AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
remote_bitrate_estimator_.reset(
rbe_factory.Create(this, clock, kMimdControl,
kRemoteBitrateEstimatorMinBitrateBps));
}
void set_expected_bitrate_bps(unsigned int expected_bitrate_bps) {
CriticalSectionScoped lock(crit_.get());
expected_bitrate_bps_ = expected_bitrate_bps;
}
virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
unsigned int bitrate) OVERRIDE {
CriticalSectionScoped lock(crit_.get());
assert(expected_bitrate_bps_ > 0);
if (bitrate >= expected_bitrate_bps_) {
// Just trigger if there was any rtx padding packet.
if (rtx_media_ssrcs_.empty() || rtx_media_sent_ > 0) {
TriggerTestDone();
}
}
rtp_rtcp_->SetREMBData(
bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]);
rtp_rtcp_->Process();
}
virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE {
CriticalSectionScoped lock(crit_.get());
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
receive_stats_->IncomingPacket(header, length, false);
payload_registry_->SetIncomingPayloadType(header);
remote_bitrate_estimator_->IncomingPacket(
clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header);
if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
remote_bitrate_estimator_->Process();
}
total_sent_ += length;
padding_sent_ += header.paddingLength;
++total_packets_sent_;
if (header.paddingLength > 0)
++padding_packets_sent_;
if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end()) {
rtx_media_sent_ += length - header.headerLength - header.paddingLength;
if (header.paddingLength == 0)
++rtx_media_packets_sent_;
uint8_t restored_packet[kMaxPacketSize];
uint8_t* restored_packet_ptr = restored_packet;
int restored_length = static_cast<int>(length);
payload_registry_->RestoreOriginalPacket(&restored_packet_ptr,
packet,
&restored_length,
rtx_media_ssrcs_[header.ssrc],
header);
length = restored_length;
EXPECT_TRUE(rtp_parser_->Parse(
restored_packet, static_cast<int>(length), &header));
} else {
rtp_rtcp_->SetRemoteSSRC(header.ssrc);
}
return true;
}
virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
return true;
}
EventTypeWrapper Wait() { return test_done_->Wait(120 * 1000); }
private:
void ReportResult(const std::string& measurement,
size_t value,
const std::string& units) {
webrtc::test::PrintResult(
measurement, "",
::testing::UnitTest::GetInstance()->current_test_info()->name(),
value, units, false);
}
void TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_) {
ReportResult("total-sent", total_sent_, "bytes");
ReportResult("padding-sent", padding_sent_, "bytes");
ReportResult("rtx-media-sent", rtx_media_sent_, "bytes");
ReportResult("total-packets-sent", total_packets_sent_, "packets");
ReportResult("padding-packets-sent", padding_packets_sent_, "packets");
ReportResult("rtx-packets-sent", rtx_media_packets_sent_, "packets");
test_done_->Set();
}
Clock* const clock_;
const scoped_ptr<EventWrapper> test_done_;
const scoped_ptr<RtpHeaderParser> rtp_parser_;
scoped_ptr<RtpRtcp> rtp_rtcp_;
internal::TransportAdapter feedback_transport_;
const scoped_ptr<ReceiveStatistics> receive_stats_;
const scoped_ptr<RTPPayloadRegistry> payload_registry_;
scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
const scoped_ptr<CriticalSectionWrapper> crit_;
unsigned int expected_bitrate_bps_ GUARDED_BY(crit_);
SsrcMap rtx_media_ssrcs_ GUARDED_BY(crit_);
size_t total_sent_ GUARDED_BY(crit_);
size_t padding_sent_ GUARDED_BY(crit_);
size_t rtx_media_sent_ GUARDED_BY(crit_);
int total_packets_sent_ GUARDED_BY(crit_);
int padding_packets_sent_ GUARDED_BY(crit_);
int rtx_media_packets_sent_ GUARDED_BY(crit_);
};
class LowRateStreamObserver : public test::DirectTransport,
public RemoteBitrateObserver,
public PacketReceiver {
public:
LowRateStreamObserver(newapi::Transport* feedback_transport,
Clock* clock,
size_t number_of_streams,
bool rtx_used)
: clock_(clock),
number_of_streams_(number_of_streams),
rtx_used_(rtx_used),
test_done_(EventWrapper::Create()),
rtp_parser_(RtpHeaderParser::Create()),
feedback_transport_(feedback_transport),
receive_stats_(ReceiveStatistics::Create(clock)),
crit_(CriticalSectionWrapper::CreateCriticalSection()),
send_stream_(NULL),
test_state_(kFirstRampup),
state_start_ms_(clock_->TimeInMilliseconds()),
interval_start_ms_(state_start_ms_),
last_remb_bps_(0),
sent_bytes_(0),
total_overuse_bytes_(0),
suspended_in_stats_(false) {
RtpRtcp::Configuration config;
config.receive_statistics = receive_stats_.get();
feedback_transport_.Enable();
config.outgoing_transport = &feedback_transport_;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetREMBStatus(true);
rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId);
AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 10000;
remote_bitrate_estimator_.reset(
rbe_factory.Create(this, clock, kMimdControl,
kRemoteBitrateEstimatorMinBitrateBps));
forward_transport_config_.link_capacity_kbps =
kHighBandwidthLimitBps / 1000;
forward_transport_config_.queue_length = 100; // Something large.
test::DirectTransport::SetConfig(forward_transport_config_);
test::DirectTransport::SetReceiver(this);
}
virtual void SetSendStream(const VideoSendStream* send_stream) {
CriticalSectionScoped lock(crit_.get());
send_stream_ = send_stream;
}
virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
unsigned int bitrate) {
CriticalSectionScoped lock(crit_.get());
rtp_rtcp_->SetREMBData(
bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]);
rtp_rtcp_->Process();
last_remb_bps_ = bitrate;
}
virtual bool SendRtp(const uint8_t* data, size_t length) OVERRIDE {
CriticalSectionScoped lock(crit_.get());
sent_bytes_ += length;
int64_t now_ms = clock_->TimeInMilliseconds();
if (now_ms > interval_start_ms_ + 1000) { // Let at least 1 second pass.
// Verify that the send rate was about right.
unsigned int average_rate_bps = static_cast<unsigned int>(sent_bytes_) *
8 * 1000 / (now_ms - interval_start_ms_);
// TODO(holmer): Why is this failing?
// EXPECT_LT(average_rate_bps, last_remb_bps_ * 1.1);
if (average_rate_bps > last_remb_bps_ * 1.1) {
total_overuse_bytes_ +=
sent_bytes_ -
last_remb_bps_ / 8 * (now_ms - interval_start_ms_) / 1000;
}
EvolveTestState(average_rate_bps);
interval_start_ms_ = now_ms;
sent_bytes_ = 0;
}
return test::DirectTransport::SendRtp(data, length);
}
virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
size_t length) OVERRIDE {
CriticalSectionScoped lock(crit_.get());
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
receive_stats_->IncomingPacket(header, length, false);
remote_bitrate_estimator_->IncomingPacket(
clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header);
if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
remote_bitrate_estimator_->Process();
}
suspended_in_stats_ = send_stream_->GetStats().suspended;
return DELIVERY_OK;
}
virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
return true;
}
// Produces a string similar to "1stream_nortx", depending on the values of
// number_of_streams_ and rtx_used_;
std::string GetModifierString() {
std::string str("_");
char temp_str[5];
sprintf(temp_str, "%i", static_cast<int>(number_of_streams_));
str += std::string(temp_str);
str += "stream";
str += (number_of_streams_ > 1 ? "s" : "");
str += "_";
str += (rtx_used_ ? "" : "no");
str += "rtx";
return str;
}
// This method defines the state machine for the ramp up-down-up test.
void EvolveTestState(unsigned int bitrate_bps) {
int64_t now = clock_->TimeInMilliseconds();
CriticalSectionScoped lock(crit_.get());
assert(send_stream_ != NULL);
switch (test_state_) {
case kFirstRampup: {
EXPECT_FALSE(suspended_in_stats_);
if (bitrate_bps > kExpectedHighBitrateBps) {
// The first ramp-up has reached the target bitrate. Change the
// channel limit, and move to the next test state.
forward_transport_config_.link_capacity_kbps =
kLowBandwidthLimitBps / 1000;
test::DirectTransport::SetConfig(forward_transport_config_);
test_state_ = kLowRate;
webrtc::test::PrintResult("ramp_up_down_up",
GetModifierString(),
"first_rampup",
now - state_start_ms_,
"ms",
false);
state_start_ms_ = now;
interval_start_ms_ = now;
sent_bytes_ = 0;
}
break;
}
case kLowRate: {
if (bitrate_bps < kExpectedLowBitrateBps && suspended_in_stats_) {
// The ramp-down was successful. Change the channel limit back to a
// high value, and move to the next test state.
forward_transport_config_.link_capacity_kbps =
kHighBandwidthLimitBps / 1000;
test::DirectTransport::SetConfig(forward_transport_config_);
test_state_ = kSecondRampup;
webrtc::test::PrintResult("ramp_up_down_up",
GetModifierString(),
"rampdown",
now - state_start_ms_,
"ms",
false);
state_start_ms_ = now;
interval_start_ms_ = now;
sent_bytes_ = 0;
}
break;
}
case kSecondRampup: {
if (bitrate_bps > kExpectedHighBitrateBps && !suspended_in_stats_) {
webrtc::test::PrintResult("ramp_up_down_up",
GetModifierString(),
"second_rampup",
now - state_start_ms_,
"ms",
false);
webrtc::test::PrintResult("ramp_up_down_up",
GetModifierString(),
"total_overuse",
total_overuse_bytes_,
"bytes",
false);
test_done_->Set();
}
break;
}
}
}
EventTypeWrapper Wait() { return test_done_->Wait(120 * 1000); }
private:
static const unsigned int kHighBandwidthLimitBps = 80000;
static const unsigned int kExpectedHighBitrateBps = 60000;
static const unsigned int kLowBandwidthLimitBps = 20000;
static const unsigned int kExpectedLowBitrateBps = 20000;
enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
Clock* const clock_;
const size_t number_of_streams_;
const bool rtx_used_;
const scoped_ptr<EventWrapper> test_done_;
const scoped_ptr<RtpHeaderParser> rtp_parser_;
scoped_ptr<RtpRtcp> rtp_rtcp_;
internal::TransportAdapter feedback_transport_;
const scoped_ptr<ReceiveStatistics> receive_stats_;
scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
scoped_ptr<CriticalSectionWrapper> crit_;
const VideoSendStream* send_stream_ GUARDED_BY(crit_);
FakeNetworkPipe::Config forward_transport_config_ GUARDED_BY(crit_);
TestStates test_state_ GUARDED_BY(crit_);
int64_t state_start_ms_ GUARDED_BY(crit_);
int64_t interval_start_ms_ GUARDED_BY(crit_);
unsigned int last_remb_bps_ GUARDED_BY(crit_);
size_t sent_bytes_ GUARDED_BY(crit_);
size_t total_overuse_bytes_ GUARDED_BY(crit_);
bool suspended_in_stats_ GUARDED_BY(crit_);
};
}
class RampUpTest : public ::testing::Test {
public:
virtual void SetUp() { reserved_ssrcs_.clear(); }
protected:
void RunRampUpTest(bool pacing, bool rtx, size_t num_streams) {
std::vector<uint32_t> ssrcs(GenerateSsrcs(num_streams, 100));
std::vector<uint32_t> rtx_ssrcs(GenerateSsrcs(num_streams, 200));
StreamObserver::SsrcMap rtx_ssrc_map;
if (rtx) {
for (size_t i = 0; i < ssrcs.size(); ++i)
rtx_ssrc_map[rtx_ssrcs[i]] = ssrcs[i];
}
test::DirectTransport receiver_transport;
StreamObserver stream_observer(rtx_ssrc_map,
&receiver_transport,
Clock::GetRealTimeClock());
Call::Config call_config(&stream_observer);
webrtc::Config webrtc_config;
call_config.webrtc_config = &webrtc_config;
webrtc_config.Set<PaddingStrategy>(new PaddingStrategy(rtx));
scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = call->GetDefaultSendConfig();
receiver_transport.SetReceiver(call->Receiver());
test::FakeEncoder encoder(Clock::GetRealTimeClock());
send_config.encoder_settings.encoder = &encoder;
send_config.encoder_settings.payload_type = 125;
send_config.encoder_settings.payload_name = "FAKE";
std::vector<VideoStream> video_streams =
test::CreateVideoStreams(num_streams);
if (num_streams == 1) {
video_streams[0].target_bitrate_bps = 2000000;
video_streams[0].max_bitrate_bps = 2000000;
}
send_config.pacing = pacing;
send_config.rtp.nack.rtp_history_ms = 1000;
send_config.rtp.ssrcs = ssrcs;
if (rtx) {
send_config.rtp.rtx.payload_type = 96;
send_config.rtp.rtx.ssrcs = rtx_ssrcs;
}
send_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsoluteSendTimeExtensionId));
if (num_streams == 1) {
// For single stream rampup until 1mbps
stream_observer.set_expected_bitrate_bps(1000000);
} else {
// For multi stream rampup until all streams are being sent. That means
// enough birate to sent all the target streams plus the min bitrate of
// the last one.
int expected_bitrate_bps = video_streams.back().min_bitrate_bps;
for (size_t i = 0; i < video_streams.size() - 1; ++i) {
expected_bitrate_bps += video_streams[i].target_bitrate_bps;
}
stream_observer.set_expected_bitrate_bps(expected_bitrate_bps);
}
VideoSendStream* send_stream =
call->CreateVideoSendStream(send_config, video_streams, NULL);
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(send_stream->Input(),
video_streams.back().width,
video_streams.back().height,
video_streams.back().max_framerate,
Clock::GetRealTimeClock()));
send_stream->Start();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, stream_observer.Wait());
frame_generator_capturer->Stop();
send_stream->Stop();
call->DestroyVideoSendStream(send_stream);
}
void RunRampUpDownUpTest(size_t number_of_streams, bool rtx) {
std::vector<uint32_t> ssrcs;
for (size_t i = 0; i < number_of_streams; ++i)
ssrcs.push_back(static_cast<uint32_t>(i + 1));
test::DirectTransport receiver_transport;
LowRateStreamObserver stream_observer(
&receiver_transport, Clock::GetRealTimeClock(), number_of_streams, rtx);
Call::Config call_config(&stream_observer);
webrtc::Config webrtc_config;
call_config.webrtc_config = &webrtc_config;
webrtc_config.Set<PaddingStrategy>(new PaddingStrategy(rtx));
scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = call->GetDefaultSendConfig();
receiver_transport.SetReceiver(call->Receiver());
test::FakeEncoder encoder(Clock::GetRealTimeClock());
send_config.encoder_settings.encoder = &encoder;
send_config.encoder_settings.payload_type = 125;
send_config.encoder_settings.payload_name = "FAKE";
std::vector<VideoStream> video_streams =
test::CreateVideoStreams(number_of_streams);
send_config.rtp.nack.rtp_history_ms = 1000;
send_config.rtp.ssrcs.insert(
send_config.rtp.ssrcs.begin(), ssrcs.begin(), ssrcs.end());
send_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsoluteSendTimeExtensionId));
send_config.suspend_below_min_bitrate = true;
send_config.pacing = true;
VideoSendStream* send_stream =
call->CreateVideoSendStream(send_config, video_streams, NULL);
stream_observer.SetSendStream(send_stream);
size_t width = 0;
size_t height = 0;
for (size_t i = 0; i < video_streams.size(); ++i) {
size_t stream_width = video_streams[i].width;
size_t stream_height = video_streams[i].height;
if (stream_width > width)
width = stream_width;
if (stream_height > height)
height = stream_height;
}
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(send_stream->Input(),
width,
height,
30,
Clock::GetRealTimeClock()));
send_stream->Start();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, stream_observer.Wait());
stream_observer.StopSending();
receiver_transport.StopSending();
frame_generator_capturer->Stop();
send_stream->Stop();
call->DestroyVideoSendStream(send_stream);
}
private:
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;
}
std::map<uint32_t, bool> reserved_ssrcs_;
};
TEST_F(RampUpTest, SingleStreamWithoutPacing) {
RunRampUpTest(false, false, 1);
}
TEST_F(RampUpTest, SingleStreamWithPacing) {
RunRampUpTest(true, false, 1);
}
TEST_F(RampUpTest, SimulcastWithoutPacing) {
RunRampUpTest(false, false, 3);
}
TEST_F(RampUpTest, SimulcastWithPacing) {
RunRampUpTest(true, false, 3);
}
// TODO(pbos): Re-enable, webrtc:2992.
TEST_F(RampUpTest, DISABLED_SimulcastWithPacingAndRtx) {
RunRampUpTest(true, true, 3);
}
TEST_F(RampUpTest, UpDownUpOneStream) { RunRampUpDownUpTest(1, false); }
TEST_F(RampUpTest, UpDownUpThreeStreams) { RunRampUpDownUpTest(3, false); }
TEST_F(RampUpTest, UpDownUpOneStreamRtx) { RunRampUpDownUpTest(1, true); }
TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) { RunRampUpDownUpTest(3, true); }
} // namespace webrtc
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