jamosdev
/
rhubarb-lip-sync
mirror of https://github.com/DanielSWolf/rhubarb-lip-sync.git
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity
rhubarb-lip-sync/lib/webrtc-8d2248ff/webrtc/call/rampup_tests.cc

585 lines
20 KiB
C++
Raw Blame History

/*
* 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 "webrtc/call/rampup_tests.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/test/testsupport/perf_test.h"
namespace webrtc {
namespace {
static const int64_t kPollIntervalMs = 20;
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,
unsigned int start_bitrate_bps,
const std::string& extension_type,
bool rtx,
bool red)
: EndToEndTest(test::CallTest::kLongTimeoutMs),
event_(false, false),
clock_(Clock::GetRealTimeClock()),
num_video_streams_(num_video_streams),
num_audio_streams_(num_audio_streams),
rtx_(rtx),
red_(red),
send_stream_(nullptr),
start_bitrate_bps_(start_bitrate_bps),
start_bitrate_verified_(false),
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)),
poller_thread_(&BitrateStatsPollingThread,
this,
"BitrateStatsPollingThread"),
sender_call_(nullptr) {
EXPECT_LE(num_audio_streams_, 1u);
if (rtx_) {
for (size_t i = 0; i < video_ssrcs_.size(); ++i)
rtx_ssrc_map_[video_rtx_ssrcs_[i]] = video_ssrcs_[i];
}
}
RampUpTester::~RampUpTester() {
event_.Set();
}
Call::Config RampUpTester::GetSenderCallConfig() {
Call::Config call_config;
if (start_bitrate_bps_ != 0) {
call_config.bitrate_config.start_bitrate_bps = start_bitrate_bps_;
}
call_config.bitrate_config.min_bitrate_bps = 10000;
return call_config;
}
void RampUpTester::OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) {
send_stream_ = send_stream;
}
test::PacketTransport* RampUpTester::CreateSendTransport(Call* sender_call) {
send_transport_ = new test::PacketTransport(sender_call, this,
test::PacketTransport::kSender,
forward_transport_config_);
return send_transport_;
}
size_t RampUpTester::GetNumVideoStreams() const {
return num_video_streams_;
}
size_t RampUpTester::GetNumAudioStreams() const {
return num_audio_streams_;
}
void RampUpTester::ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) {
send_config->suspend_below_min_bitrate = true;
if (num_video_streams_ == 1) {
encoder_config->streams[0].target_bitrate_bps =
encoder_config->streams[0].max_bitrate_bps = 2000000;
// For single stream rampup until 1mbps
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.
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;
}
}
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.fec.ulpfec_payload_type =
test::CallTest::kUlpfecPayloadType;
send_config->rtp.fec.red_payload_type = test::CallTest::kRedPayloadType;
}
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.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.fec.red_payload_type =
send_config->rtp.fec.red_payload_type;
recv_config.rtp.fec.ulpfec_payload_type =
send_config->rtp.fec.ulpfec_payload_type;
}
if (rtx_) {
recv_config.rtp.rtx[send_config->encoder_settings.payload_type].ssrc =
video_rtx_ssrcs_[i];
recv_config.rtp.rtx[send_config->encoder_settings.payload_type]
.payload_type = send_config->rtp.rtx.payload_type;
}
++i;
}
}
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.";
send_config->rtp.ssrc = audio_ssrcs_[0];
send_config->rtp.extensions.clear();
bool transport_cc = false;
if (extension_type_ == RtpExtension::kAbsSendTimeUri) {
transport_cc = false;
send_config->rtp.extensions.push_back(
RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
} else 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::OnCallsCreated(Call* sender_call, Call* receiver_call) {
sender_call_ = sender_call;
}
bool RampUpTester::BitrateStatsPollingThread(void* obj) {
return static_cast<RampUpTester*>(obj)->PollStats();
}
bool RampUpTester::PollStats() {
if (sender_call_) {
Call::Stats stats = sender_call_->GetStats();
RTC_DCHECK_GT(expected_bitrate_bps_, 0);
if (!start_bitrate_verified_ && 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(stats.send_bandwidth_bps, 0.9 * start_bitrate_bps_);
start_bitrate_verified_ = true;
}
if (stats.send_bandwidth_bps >= expected_bitrate_bps_) {
ramp_up_finished_ms_ = clock_->TimeInMilliseconds();
observation_complete_.Set();
}
}
return !event_.Wait(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.
VideoSendStream::Stats send_stats = send_stream_->GetStats();
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);
}
ReportResult("ramp-up-total-packets-sent", total_packets_sent, "packets");
ReportResult("ramp-up-total-sent", total_sent, "bytes");
ReportResult("ramp-up-media-sent", media_sent, "bytes");
ReportResult("ramp-up-padding-sent", padding_sent, "bytes");
ReportResult("ramp-up-rtx-total-packets-sent", rtx_total_packets_sent,
"packets");
ReportResult("ramp-up-rtx-total-sent", rtx_total_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();
poller_thread_.Start();
EXPECT_TRUE(Wait()) << "Timed out while waiting for ramp-up to complete.";
TriggerTestDone();
poller_thread_.Stop();
}
RampUpDownUpTester::RampUpDownUpTester(size_t num_video_streams,
size_t num_audio_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
bool rtx,
bool red)
: RampUpTester(num_video_streams,
num_audio_streams,
start_bitrate_bps,
extension_type,
rtx,
red),
test_state_(kFirstRampup),
state_start_ms_(clock_->TimeInMilliseconds()),
interval_start_ms_(clock_->TimeInMilliseconds()),
sent_bytes_(0) {
forward_transport_config_.link_capacity_kbps = kHighBandwidthLimitBps / 1000;
}
RampUpDownUpTester::~RampUpDownUpTester() {}
bool RampUpDownUpTester::PollStats() {
if (send_stream_) {
webrtc::VideoSendStream::Stats stats = send_stream_->GetStats();
int transmit_bitrate_bps = 0;
for (auto it : stats.substreams) {
transmit_bitrate_bps += it.second.total_bitrate_bps;
}
EvolveTestState(transmit_bitrate_bps, stats.suspended);
}
return !event_.Wait(kPollIntervalMs);
}
Call::Config RampUpDownUpTester::GetReceiverCallConfig() {
Call::Config config;
config.bitrate_config.min_bitrate_bps = 10000;
return config;
}
std::string RampUpDownUpTester::GetModifierString() const {
std::string str("_");
if (num_video_streams_ > 0) {
std::ostringstream s;
s << num_video_streams_;
str += s.str();
str += "stream";
str += (num_video_streams_ > 1 ? "s" : "");
str += "_";
}
if (num_audio_streams_ > 0) {
std::ostringstream s;
s << num_audio_streams_;
str += s.str();
str += "stream";
str += (num_audio_streams_ > 1 ? "s" : "");
str += "_";
}
str += (rtx_ ? "" : "no");
str += "rtx";
return str;
}
void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
int64_t now = clock_->TimeInMilliseconds();
switch (test_state_) {
case kFirstRampup: {
EXPECT_FALSE(suspended);
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;
send_transport_->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) {
// 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;
send_transport_->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) {
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");
observation_complete_.Set();
}
break;
}
}
}
class RampUpTest : public test::CallTest {
public:
RampUpTest() {}
virtual ~RampUpTest() {
EXPECT_EQ(nullptr, video_send_stream_);
EXPECT_TRUE(video_receive_streams_.empty());
}
};
TEST_F(RampUpTest, SingleStream) {
RampUpTester test(1, 0, 0, RtpExtension::kTimestampOffsetUri, false, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, Simulcast) {
RampUpTester test(3, 0, 0, RtpExtension::kTimestampOffsetUri, false, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, SimulcastWithRtx) {
RampUpTester test(3, 0, 0, RtpExtension::kTimestampOffsetUri, true, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, SimulcastByRedWithRtx) {
RampUpTester test(3, 0, 0, RtpExtension::kTimestampOffsetUri, true, true);
RunBaseTest(&test);
}
TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) {
RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
RtpExtension::kTimestampOffsetUri, false, false);
RunBaseTest(&test);
}
static const uint32_t kStartBitrateBps = 60000;
// Disabled: https://bugs.chromium.org/p/webrtc/issues/detail?id=5576
TEST_F(RampUpTest, DISABLED_UpDownUpOneStream) {
RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTimeUri,
false, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpThreeStreams) {
RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTimeUri,
false, false);
RunBaseTest(&test);
}
// Disabled: https://bugs.chromium.org/p/webrtc/issues/detail?id=5576
TEST_F(RampUpTest, DISABLED_UpDownUpOneStreamRtx) {
RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTimeUri,
true, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) {
RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTimeUri,
true, false);
RunBaseTest(&test);
}
// Disabled: https://bugs.chromium.org/p/webrtc/issues/detail?id=5576
TEST_F(RampUpTest, DISABLED_UpDownUpOneStreamByRedRtx) {
RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTimeUri,
true, true);
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpThreeStreamsByRedRtx) {
RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTimeUri,
true, true);
RunBaseTest(&test);
}
TEST_F(RampUpTest, SendSideVideoUpDownUpRtx) {
RampUpDownUpTester test(3, 0, kStartBitrateBps,
RtpExtension::kTransportSequenceNumberUri, true,
false);
RunBaseTest(&test);
}
// TODO(holmer): Enable when audio bitrates are included in the bitrate
// allocation.
TEST_F(RampUpTest, DISABLED_SendSideAudioVideoUpDownUpRtx) {
RampUpDownUpTester test(3, 1, kStartBitrateBps,
RtpExtension::kTransportSequenceNumberUri, true,
false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, AbsSendTimeSingleStream) {
RampUpTester test(1, 0, 0, RtpExtension::kAbsSendTimeUri, false, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, AbsSendTimeSimulcast) {
RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTimeUri, false, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, AbsSendTimeSimulcastWithRtx) {
RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTimeUri, true, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, AbsSendTimeSimulcastByRedWithRtx) {
RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTimeUri, true, true);
RunBaseTest(&test);
}
TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
RtpExtension::kAbsSendTimeUri, false, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSingleStream) {
RampUpTester test(1, 0, 0, RtpExtension::kTransportSequenceNumberUri, false,
false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSimulcast) {
RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumberUri, false,
false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSimulcastWithRtx) {
RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumberUri, true,
false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, AudioVideoTransportSequenceNumberSimulcastWithRtx) {
RampUpTester test(3, 1, 0, RtpExtension::kTransportSequenceNumberUri, true,
false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSimulcastByRedWithRtx) {
RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumberUri, true,
true);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
RtpExtension::kTransportSequenceNumberUri, false, false);
RunBaseTest(&test);
}
} // namespace webrtc
Reference in New Issue View Git Blame Copy Permalink