/*
* 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 "testing/gtest/include/gtest/gtest.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/test/testsupport/perf_test.h"
#include "webrtc/video/rampup_tests.h"
namespace webrtc {
namespace {
static const int kMaxPacketSize = 1500;
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
StreamObserver::StreamObserver(const SsrcMap& rtx_media_ssrcs,
newapi::Transport* feedback_transport,
Clock* clock,
RemoteBitrateEstimatorFactory* rbe_factory,
RateControlType control_type)
: 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),
start_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),
test_start_ms_(clock_->TimeInMilliseconds()),
ramp_up_finished_ms_(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,
kAbsSendTimeExtensionId);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
remote_bitrate_estimator_.reset(
rbe_factory->Create(this, clock, control_type,
kRemoteBitrateEstimatorMinBitrateBps));
}
void StreamObserver::set_expected_bitrate_bps(
unsigned int expected_bitrate_bps) {
CriticalSectionScoped lock(crit_.get());
expected_bitrate_bps_ = expected_bitrate_bps;
}
void StreamObserver::set_start_bitrate_bps(unsigned int start_bitrate_bps) {
CriticalSectionScoped lock(crit_.get());
start_bitrate_bps_ = start_bitrate_bps;
}
void StreamObserver::OnReceiveBitrateChanged(
const std::vector<unsigned int>& ssrcs, unsigned int bitrate) {
CriticalSectionScoped lock(crit_.get());
assert(expected_bitrate_bps_ > 0);
if (start_bitrate_bps_ != 0) {
// For tests with an explicitly set start bitrate, verify the first
// bitrate estimate is close to the start bitrate and lower than the
// test target bitrate. This is to verify a call respects the configured
// start bitrate, but due to the BWE implementation we can't guarantee the
// first estimate really is as high as the start bitrate.
EXPECT_GT(bitrate, 0.9 * start_bitrate_bps_);
start_bitrate_bps_ = 0;
}
if (bitrate >= expected_bitrate_bps_) {
ramp_up_finished_ms_ = clock_->TimeInMilliseconds();
// 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();
}
bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
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;
}
bool StreamObserver::SendRtcp(const uint8_t* packet, size_t length) {
return true;
}
EventTypeWrapper StreamObserver::Wait() { return test_done_->Wait(120 * 1000); }
void StreamObserver::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 StreamObserver::TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_) {
ReportResult("ramp-up-total-sent", total_sent_, "bytes");
ReportResult("ramp-up-padding-sent", padding_sent_, "bytes");
ReportResult("ramp-up-rtx-media-sent", rtx_media_sent_, "bytes");
ReportResult("ramp-up-total-packets-sent", total_packets_sent_, "packets");
ReportResult("ramp-up-padding-packets-sent",
padding_packets_sent_,
"packets");
ReportResult("ramp-up-rtx-packets-sent",
rtx_media_packets_sent_,
"packets");
ReportResult("ramp-up-time",
ramp_up_finished_ms_ - test_start_ms_,
"milliseconds");
test_done_->Set();
}
LowRateStreamObserver::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(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
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_packets = 100; // Something large.
test::DirectTransport::SetConfig(forward_transport_config_);
test::DirectTransport::SetReceiver(this);
}
void LowRateStreamObserver::SetSendStream(const VideoSendStream* send_stream) {
CriticalSectionScoped lock(crit_.get());
send_stream_ = send_stream;
}
void LowRateStreamObserver::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;
}
bool LowRateStreamObserver::SendRtp(const uint8_t* data, size_t length) {
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);
}
PacketReceiver::DeliveryStatus LowRateStreamObserver::DeliverPacket(
const uint8_t* packet, size_t length) {
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;
}
bool LowRateStreamObserver::SendRtcp(const uint8_t* packet, size_t length) {
return true;
}
std::string LowRateStreamObserver::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;
}
void LowRateStreamObserver::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 LowRateStreamObserver::Wait() {
return test_done_->Wait(test::CallTest::kLongTimeoutMs);
}
void RampUpTest::RunRampUpTest(bool rtx,
size_t num_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type) {
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];
}
CreateSendConfig(num_streams);
scoped_ptr<RemoteBitrateEstimatorFactory> rbe_factory;
RateControlType control_type;
if (extension_type == RtpExtension::kAbsSendTime) {
control_type = kAimdControl;
rbe_factory.reset(new AbsoluteSendTimeRemoteBitrateEstimatorFactory);
send_config_.rtp.extensions.push_back(RtpExtension(
extension_type.c_str(), kAbsSendTimeExtensionId));
} else {
control_type = kMimdControl;
rbe_factory.reset(new RemoteBitrateEstimatorFactory);
send_config_.rtp.extensions.push_back(RtpExtension(
extension_type.c_str(), kTransmissionTimeOffsetExtensionId));
}
test::DirectTransport receiver_transport;
StreamObserver stream_observer(rtx_ssrc_map,
&receiver_transport,
Clock::GetRealTimeClock(),
rbe_factory.get(),
control_type);
Call::Config call_config(&stream_observer);
if (start_bitrate_bps != 0) {
call_config.start_bitrate_bps = start_bitrate_bps;
stream_observer.set_start_bitrate_bps(start_bitrate_bps);
}
CreateSenderCall(call_config);
receiver_transport.SetReceiver(sender_call_->Receiver());
if (num_streams == 1) {
encoder_config_.streams[0].target_bitrate_bps = 2000000;
encoder_config_.streams[0].max_bitrate_bps = 2000000;
}
send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config_.rtp.ssrcs = ssrcs;
if (rtx) {
send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
send_config_.rtp.rtx.ssrcs = rtx_ssrcs;
send_config_.rtp.rtx.pad_with_redundant_payloads = true;
}
if (num_streams == 1) {
// For single stream rampup until 1mbps
stream_observer.set_expected_bitrate_bps(kSingleStreamTargetBps);
} else {
// For multi stream rampup until all streams are being sent. That means
// enough birate to send all the target streams plus the min bitrate of
// the last one.
int expected_bitrate_bps = encoder_config_.streams.back().min_bitrate_bps;
for (size_t i = 0; i < encoder_config_.streams.size() - 1; ++i) {
expected_bitrate_bps += encoder_config_.streams[i].target_bitrate_bps;
}
stream_observer.set_expected_bitrate_bps(expected_bitrate_bps);
}
CreateStreams();
CreateFrameGeneratorCapturer();
Start();
EXPECT_EQ(kEventSignaled, stream_observer.Wait());
Stop();
DestroyStreams();
}
void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams, bool rtx) {
test::DirectTransport receiver_transport;
LowRateStreamObserver stream_observer(
&receiver_transport, Clock::GetRealTimeClock(), number_of_streams, rtx);
Call::Config call_config(&stream_observer);
CreateSenderCall(call_config);
receiver_transport.SetReceiver(sender_call_->Receiver());
CreateSendConfig(number_of_streams);
send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kTOffset, kTransmissionTimeOffsetExtensionId));
send_config_.suspend_below_min_bitrate = true;
if (rtx) {
send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
send_config_.rtp.rtx.ssrcs = GenerateSsrcs(number_of_streams, 200);
send_config_.rtp.rtx.pad_with_redundant_payloads = true;
}
CreateStreams();
stream_observer.SetSendStream(send_stream_);
CreateFrameGeneratorCapturer();
Start();
EXPECT_EQ(kEventSignaled, stream_observer.Wait());
Stop();
DestroyStreams();
}
TEST_F(RampUpTest, SingleStream) {
RunRampUpTest(false, 1, 0, RtpExtension::kTOffset);
}
TEST_F(RampUpTest, Simulcast) {
RunRampUpTest(false, 3, 0, RtpExtension::kTOffset);
}
TEST_F(RampUpTest, SimulcastWithRtx) {
RunRampUpTest(true, 3, 0, RtpExtension::kTOffset);
}
TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) {
RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps, RtpExtension::kTOffset);
}
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