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Nagram/TMessagesProj/jni/voip/webrtc/call/call_perf_tests.cc

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Update to 7.0.0 (2060)
2020-08-14 16:58:22 +00:00
/*
* 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 <algorithm>
#include <limits>
#include <memory>
#include <string>
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/video_encoder_config.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "rtc_base/checks.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/metrics.h"
#include "test/call_test.h"
#include "test/direct_transport.h"
#include "test/drifting_clock.h"
#include "test/encoder_settings.h"
#include "test/fake_encoder.h"
#include "test/field_trial.h"
#include "test/frame_generator_capturer.h"
#include "test/gtest.h"
#include "test/null_transport.h"
#include "test/rtp_header_parser.h"
#include "test/rtp_rtcp_observer.h"
#include "test/testsupport/file_utils.h"
#include "test/testsupport/perf_test.h"
#include "test/video_encoder_proxy_factory.h"
#include "video/transport_adapter.h"
using webrtc::test::DriftingClock;
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kTransportSequenceNumberExtensionId = 1,
};
} // namespace
class CallPerfTest : public test::CallTest {
public:
CallPerfTest() {
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
}
protected:
enum class FecMode { kOn, kOff };
enum class CreateOrder { kAudioFirst, kVideoFirst };
void TestAudioVideoSync(FecMode fec,
CreateOrder create_first,
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed,
const std::string& test_label);
void TestMinTransmitBitrate(bool pad_to_min_bitrate);
void TestCaptureNtpTime(const BuiltInNetworkBehaviorConfig& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms);
void TestMinAudioVideoBitrate(int test_bitrate_from,
int test_bitrate_to,
int test_bitrate_step,
int min_bwe,
int start_bwe,
int max_bwe);
};
class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
public rtc::VideoSinkInterface<VideoFrame> {
static const int kInSyncThresholdMs = 50;
static const int kStartupTimeMs = 2000;
static const int kMinRunTimeMs = 30000;
public:
explicit VideoRtcpAndSyncObserver(TaskQueueBase* task_queue,
Clock* clock,
const std::string& test_label)
: test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs),
clock_(clock),
test_label_(test_label),
creation_time_ms_(clock_->TimeInMilliseconds()),
task_queue_(task_queue) {}
void OnFrame(const VideoFrame& video_frame) override {
task_queue_->PostTask(ToQueuedTask([this]() { CheckStats(); }));
}
void CheckStats() {
if (!receive_stream_)
return;
VideoReceiveStream::Stats stats = receive_stream_->GetStats();
if (stats.sync_offset_ms == std::numeric_limits<int>::max())
return;
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t time_since_creation = now_ms - creation_time_ms_;
// During the first couple of seconds audio and video can falsely be
// estimated as being synchronized. We don't want to trigger on those.
if (time_since_creation < kStartupTimeMs)
return;
if (std::abs(stats.sync_offset_ms) < kInSyncThresholdMs) {
if (first_time_in_sync_ == -1) {
first_time_in_sync_ = now_ms;
webrtc::test::PrintResult("sync_convergence_time", test_label_,
"synchronization", time_since_creation, "ms",
false);
}
if (time_since_creation > kMinRunTimeMs)
observation_complete_.Set();
}
if (first_time_in_sync_ != -1)
sync_offset_ms_list_.push_back(stats.sync_offset_ms);
}
void set_receive_stream(VideoReceiveStream* receive_stream) {
RTC_DCHECK_EQ(task_queue_, TaskQueueBase::Current());
// Note that receive_stream may be nullptr.
receive_stream_ = receive_stream;
}
void PrintResults() {
test::PrintResultList("stream_offset", test_label_, "synchronization",
sync_offset_ms_list_, "ms", false);
}
private:
Clock* const clock_;
std::string test_label_;
const int64_t creation_time_ms_;
int64_t first_time_in_sync_ = -1;
VideoReceiveStream* receive_stream_ = nullptr;
std::vector<double> sync_offset_ms_list_;
TaskQueueBase* const task_queue_;
};
void CallPerfTest::TestAudioVideoSync(FecMode fec,
CreateOrder create_first,
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed,
const std::string& test_label) {
const char* kSyncGroup = "av_sync";
const uint32_t kAudioSendSsrc = 1234;
const uint32_t kAudioRecvSsrc = 5678;
BuiltInNetworkBehaviorConfig audio_net_config;
audio_net_config.queue_delay_ms = 500;
audio_net_config.loss_percent = 5;
auto observer = std::make_unique<VideoRtcpAndSyncObserver>(
task_queue(), Clock::GetRealTimeClock(), test_label);
std::map<uint8_t, MediaType> audio_pt_map;
std::map<uint8_t, MediaType> video_pt_map;
std::unique_ptr<test::PacketTransport> audio_send_transport;
std::unique_ptr<test::PacketTransport> video_send_transport;
std::unique_ptr<test::PacketTransport> receive_transport;
test::NullTransport rtcp_send_transport;
AudioSendStream* audio_send_stream;
AudioReceiveStream* audio_receive_stream;
std::unique_ptr<DriftingClock> drifting_clock;
SendTask(RTC_FROM_HERE, task_queue(), [&]() {
metrics::Reset();
rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device =
TestAudioDeviceModule::Create(
task_queue_factory_.get(),
TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000),
TestAudioDeviceModule::CreateDiscardRenderer(48000),
audio_rtp_speed);
EXPECT_EQ(0, fake_audio_device->Init());
AudioState::Config send_audio_state_config;
send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
send_audio_state_config.audio_processing =
AudioProcessingBuilder().Create();
send_audio_state_config.audio_device_module = fake_audio_device;
Call::Config sender_config(send_event_log_.get());
auto audio_state = AudioState::Create(send_audio_state_config);
fake_audio_device->RegisterAudioCallback(audio_state->audio_transport());
sender_config.audio_state = audio_state;
Call::Config receiver_config(recv_event_log_.get());
receiver_config.audio_state = audio_state;
CreateCalls(sender_config, receiver_config);
std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_),
std::inserter(audio_pt_map, audio_pt_map.end()),
[](const std::pair<const uint8_t, MediaType>& pair) {
return pair.second == MediaType::AUDIO;
});
std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_),
std::inserter(video_pt_map, video_pt_map.end()),
[](const std::pair<const uint8_t, MediaType>& pair) {
return pair.second == MediaType::VIDEO;
});
audio_send_transport = std::make_unique<test::PacketTransport>(
task_queue(), sender_call_.get(), observer.get(),
test::PacketTransport::kSender, audio_pt_map,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(audio_net_config)));
audio_send_transport->SetReceiver(receiver_call_->Receiver());
video_send_transport = std::make_unique<test::PacketTransport>(
task_queue(), sender_call_.get(), observer.get(),
test::PacketTransport::kSender, video_pt_map,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())));
video_send_transport->SetReceiver(receiver_call_->Receiver());
receive_transport = std::make_unique<test::PacketTransport>(
task_queue(), receiver_call_.get(), observer.get(),
test::PacketTransport::kReceiver, payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())));
receive_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, video_send_transport.get());
CreateMatchingReceiveConfigs(receive_transport.get());
AudioSendStream::Config audio_send_config(audio_send_transport.get());
audio_send_config.rtp.ssrc = kAudioSendSsrc;
audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
kAudioSendPayloadType, {"ISAC", 16000, 1});
audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config);
GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (fec == FecMode::kOn) {
GetVideoSendConfig()->rtp.ulpfec.red_payload_type = kRedPayloadType;
GetVideoSendConfig()->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
video_receive_configs_[0].rtp.red_payload_type = kRedPayloadType;
video_receive_configs_[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
}
video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
video_receive_configs_[0].renderer = observer.get();
video_receive_configs_[0].sync_group = kSyncGroup;
AudioReceiveStream::Config audio_recv_config;
audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
audio_recv_config.rtcp_send_transport = &rtcp_send_transport;
audio_recv_config.sync_group = kSyncGroup;
audio_recv_config.decoder_factory = audio_decoder_factory_;
audio_recv_config.decoder_map = {
{kAudioSendPayloadType, {"ISAC", 16000, 1}}};
if (create_first == CreateOrder::kAudioFirst) {
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
CreateVideoStreams();
} else {
CreateVideoStreams();
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
}
EXPECT_EQ(1u, video_receive_streams_.size());
observer->set_receive_stream(video_receive_streams_[0]);
drifting_clock = std::make_unique<DriftingClock>(clock_, video_ntp_speed);
CreateFrameGeneratorCapturerWithDrift(drifting_clock.get(), video_rtp_speed,
kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
audio_send_stream->Start();
audio_receive_stream->Start();
});
EXPECT_TRUE(observer->Wait())
<< "Timed out while waiting for audio and video to be synchronized.";
SendTask(RTC_FROM_HERE, task_queue(), [&]() {
// Clear the pointer to the receive stream since it will now be deleted.
observer->set_receive_stream(nullptr);
audio_send_stream->Stop();
audio_receive_stream->Stop();
Stop();
DestroyStreams();
video_send_transport.reset();
audio_send_transport.reset();
receive_transport.reset();
sender_call_->DestroyAudioSendStream(audio_send_stream);
receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
DestroyCalls();
});
observer->PrintResults();
// In quick test synchronization may not be achieved in time.
if (!field_trial::IsEnabled("WebRTC-QuickPerfTest")) {
// TODO(bugs.webrtc.org/10417): Reenable this for iOS
#if !defined(WEBRTC_IOS)
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
#endif
}
task_queue()->PostTask(
ToQueuedTask([to_delete = observer.release()]() { delete to_delete; }));
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithoutClockDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::kNoDrift, DriftingClock::kNoDrift,
DriftingClock::kNoDrift, "_video_no_drift");
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::PercentsFaster(10.0f),
DriftingClock::kNoDrift, DriftingClock::kNoDrift,
"_video_ntp_drift");
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::kNoDrift,
DriftingClock::PercentsSlower(30.0f),
DriftingClock::PercentsFaster(30.0f), "_audio_faster");
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift) {
TestAudioVideoSync(FecMode::kOn, CreateOrder::kVideoFirst,
DriftingClock::kNoDrift,
DriftingClock::PercentsFaster(30.0f),
DriftingClock::PercentsSlower(30.0f), "_video_faster");
}
void CallPerfTest::TestCaptureNtpTime(
const BuiltInNetworkBehaviorConfig& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms) {
class CaptureNtpTimeObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
CaptureNtpTimeObserver(const BuiltInNetworkBehaviorConfig& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms)
: EndToEndTest(kLongTimeoutMs),
net_config_(net_config),
clock_(Clock::GetRealTimeClock()),
threshold_ms_(threshold_ms),
start_time_ms_(start_time_ms),
run_time_ms_(run_time_ms),
creation_time_ms_(clock_->TimeInMilliseconds()),
capturer_(nullptr),
rtp_start_timestamp_set_(false),
rtp_start_timestamp_(0) {}
private:
std::unique_ptr<test::PacketTransport> CreateSendTransport(
TaskQueueBase* task_queue,
Call* sender_call) override {
return std::make_unique<test::PacketTransport>(
task_queue, sender_call, this, test::PacketTransport::kSender,
payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(net_config_)));
}
std::unique_ptr<test::PacketTransport> CreateReceiveTransport(
TaskQueueBase* task_queue) override {
return std::make_unique<test::PacketTransport>(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(net_config_)));
}
void OnFrame(const VideoFrame& video_frame) override {
MutexLock lock(&mutex_);
if (video_frame.ntp_time_ms() <= 0) {
// Haven't got enough RTCP SR in order to calculate the capture ntp
// time.
return;
}
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t time_since_creation = now_ms - creation_time_ms_;
if (time_since_creation < start_time_ms_) {
// Wait for |start_time_ms_| before start measuring.
return;
}
if (time_since_creation > run_time_ms_) {
observation_complete_.Set();
}
FrameCaptureTimeList::iterator iter =
capture_time_list_.find(video_frame.timestamp());
EXPECT_TRUE(iter != capture_time_list_.end());
// The real capture time has been wrapped to uint32_t before converted
// to rtp timestamp in the sender side. So here we convert the estimated
// capture time to a uint32_t 90k timestamp also for comparing.
uint32_t estimated_capture_timestamp =
90 * static_cast<uint32_t>(video_frame.ntp_time_ms());
uint32_t real_capture_timestamp = iter->second;
int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
time_offset_ms = time_offset_ms / 90;
time_offset_ms_list_.push_back(time_offset_ms);
EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
if (!rtp_start_timestamp_set_) {
// Calculate the rtp timestamp offset in order to calculate the real
// capture time.
uint32_t first_capture_timestamp =
90 * static_cast<uint32_t>(capturer_->first_frame_capture_time());
rtp_start_timestamp_ = rtp_packet.Timestamp() - first_capture_timestamp;
rtp_start_timestamp_set_ = true;
}
uint32_t capture_timestamp =
rtp_packet.Timestamp() - rtp_start_timestamp_;
capture_time_list_.insert(
capture_time_list_.end(),
std::make_pair(rtp_packet.Timestamp(), capture_timestamp));
return SEND_PACKET;
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
capturer_ = frame_generator_capturer;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
(*receive_configs)[0].renderer = this;
// Enable the receiver side rtt calculation.
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for "
"estimated capture NTP time to be "
"within bounds.";
test::PrintResultList("capture_ntp_time", "", "real - estimated",
time_offset_ms_list_, "ms", true);
}
Mutex mutex_;
const BuiltInNetworkBehaviorConfig net_config_;
Clock* const clock_;
int threshold_ms_;
int start_time_ms_;
int run_time_ms_;
int64_t creation_time_ms_;
test::FrameGeneratorCapturer* capturer_;
bool rtp_start_timestamp_set_;
uint32_t rtp_start_timestamp_;
typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
FrameCaptureTimeList capture_time_list_ RTC_GUARDED_BY(&mutex_);
std::vector<double> time_offset_ms_list_;
} test(net_config, threshold_ms, start_time_ms, run_time_ms);
RunBaseTest(&test);
}
// Flaky tests, disabled on Mac and Windows due to webrtc:8291.
#if !(defined(WEBRTC_MAC) || defined(WEBRTC_WIN))
TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
BuiltInNetworkBehaviorConfig net_config;
net_config.queue_delay_ms = 100;
// TODO(wu): lower the threshold as the calculation/estimatation becomes more
// accurate.
const int kThresholdMs = 100;
const int kStartTimeMs = 10000;
const int kRunTimeMs = 20000;
TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
}
TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
BuiltInNetworkBehaviorConfig net_config;
net_config.queue_delay_ms = 100;
net_config.delay_standard_deviation_ms = 10;
// TODO(wu): lower the threshold as the calculation/estimatation becomes more
// accurate.
const int kThresholdMs = 100;
const int kStartTimeMs = 10000;
const int kRunTimeMs = 20000;
TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
}
#endif
TEST_F(CallPerfTest, ReceivesCpuOveruseAndUnderuse) {
// Minimal normal usage at the start, then 30s overuse to allow filter to
// settle, and then 80s underuse to allow plenty of time for rampup again.
test::ScopedFieldTrials fake_overuse_settings(
"WebRTC-ForceSimulatedOveruseIntervalMs/1-30000-80000/");
class LoadObserver : public test::SendTest,
public test::FrameGeneratorCapturer::SinkWantsObserver {
public:
LoadObserver() : SendTest(kLongTimeoutMs), test_phase_(TestPhase::kInit) {}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
frame_generator_capturer->SetSinkWantsObserver(this);
// Set a high initial resolution to be sure that we can scale down.
frame_generator_capturer->ChangeResolution(1920, 1080);
}
// OnSinkWantsChanged is called when FrameGeneratorCapturer::AddOrUpdateSink
// is called.
// TODO(sprang): Add integration test for maintain-framerate mode?
void OnSinkWantsChanged(rtc::VideoSinkInterface<VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override {
// At kStart expect CPU overuse. Then expect CPU underuse when the encoder
// delay has been decreased.
switch (test_phase_) {
case TestPhase::kInit:
// Max framerate should be set initially.
if (wants.max_framerate_fps != std::numeric_limits<int>::max() &&
wants.max_pixel_count == std::numeric_limits<int>::max()) {
test_phase_ = TestPhase::kStart;
} else {
ADD_FAILURE() << "Got unexpected adaptation request, max res = "
<< wants.max_pixel_count << ", target res = "
<< wants.target_pixel_count.value_or(-1)
<< ", max fps = " << wants.max_framerate_fps;
}
break;
case TestPhase::kStart:
if (wants.max_pixel_count < std::numeric_limits<int>::max()) {
// On adapting down, VideoStreamEncoder::VideoSourceProxy will set
// only the max pixel count, leaving the target unset.
test_phase_ = TestPhase::kAdaptedDown;
} else {
ADD_FAILURE() << "Got unexpected adaptation request, max res = "
<< wants.max_pixel_count << ", target res = "
<< wants.target_pixel_count.value_or(-1)
<< ", max fps = " << wants.max_framerate_fps;
}
break;
case TestPhase::kAdaptedDown:
// On adapting up, the adaptation counter will again be at zero, and
// so all constraints will be reset.
if (wants.max_pixel_count == std::numeric_limits<int>::max() &&
!wants.target_pixel_count) {
test_phase_ = TestPhase::kAdaptedUp;
observation_complete_.Set();
} else {
ADD_FAILURE() << "Got unexpected adaptation request, max res = "
<< wants.max_pixel_count << ", target res = "
<< wants.target_pixel_count.value_or(-1)
<< ", max fps = " << wants.max_framerate_fps;
}
break;
case TestPhase::kAdaptedUp:
ADD_FAILURE() << "Got unexpected adaptation request, max res = "
<< wants.max_pixel_count << ", target res = "
<< wants.target_pixel_count.value_or(-1)
<< ", max fps = " << wants.max_framerate_fps;
}
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
}
enum class TestPhase {
kInit,
kStart,
kAdaptedDown,
kAdaptedUp
} test_phase_;
} test;
RunBaseTest(&test);
}
void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
static const int kMaxEncodeBitrateKbps = 30;
static const int kMinTransmitBitrateBps = 150000;
static const int kMinAcceptableTransmitBitrate = 130;
static const int kMaxAcceptableTransmitBitrate = 170;
static const int kNumBitrateObservationsInRange = 100;
static const int kAcceptableBitrateErrorMargin = 15; // +- 7
class BitrateObserver : public test::EndToEndTest {
public:
explicit BitrateObserver(bool using_min_transmit_bitrate)
: EndToEndTest(kLongTimeoutMs),
send_stream_(nullptr),
converged_(false),
pad_to_min_bitrate_(using_min_transmit_bitrate),
min_acceptable_bitrate_(using_min_transmit_bitrate
? kMinAcceptableTransmitBitrate
: (kMaxEncodeBitrateKbps -
kAcceptableBitrateErrorMargin / 2)),
max_acceptable_bitrate_(using_min_transmit_bitrate
? kMaxAcceptableTransmitBitrate
: (kMaxEncodeBitrateKbps +
kAcceptableBitrateErrorMargin / 2)),
num_bitrate_observations_in_range_(0) {}
private:
// TODO(holmer): Run this with a timer instead of once per packet.
Action OnSendRtp(const uint8_t* packet, size_t length) override {
VideoSendStream::Stats stats = send_stream_->GetStats();
if (!stats.substreams.empty()) {
RTC_DCHECK_EQ(1, stats.substreams.size());
int bitrate_kbps =
stats.substreams.begin()->second.total_bitrate_bps / 1000;
if (bitrate_kbps > min_acceptable_bitrate_ &&
bitrate_kbps < max_acceptable_bitrate_) {
converged_ = true;
++num_bitrate_observations_in_range_;
if (num_bitrate_observations_in_range_ ==
kNumBitrateObservationsInRange)
observation_complete_.Set();
}
if (converged_)
bitrate_kbps_list_.push_back(bitrate_kbps);
}
return SEND_PACKET;
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (pad_to_min_bitrate_) {
encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
} else {
RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
}
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
test::PrintResultList(
"bitrate_stats_",
(pad_to_min_bitrate_ ? "min_transmit_bitrate"
: "without_min_transmit_bitrate"),
"bitrate_kbps", bitrate_kbps_list_, "kbps", false);
}
VideoSendStream* send_stream_;
bool converged_;
const bool pad_to_min_bitrate_;
const int min_acceptable_bitrate_;
const int max_acceptable_bitrate_;
int num_bitrate_observations_in_range_;
std::vector<double> bitrate_kbps_list_;
} test(pad_to_min_bitrate);
fake_encoder_max_bitrate_ = kMaxEncodeBitrateKbps;
RunBaseTest(&test);
}
TEST_F(CallPerfTest, PadsToMinTransmitBitrate) {
TestMinTransmitBitrate(true);
}
TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) {
TestMinTransmitBitrate(false);
}
// TODO(bugs.webrtc.org/8878)
#if defined(WEBRTC_MAC)
#define MAYBE_KeepsHighBitrateWhenReconfiguringSender \
DISABLED_KeepsHighBitrateWhenReconfiguringSender
#else
#define MAYBE_KeepsHighBitrateWhenReconfiguringSender \
KeepsHighBitrateWhenReconfiguringSender
#endif
TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
static const uint32_t kInitialBitrateKbps = 400;
static const uint32_t kReconfigureThresholdKbps = 600;
class 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);
streams[0].min_bitrate_bps = 50000;
streams[0].target_bitrate_bps = streams[0].max_bitrate_bps = 2000000;
return streams;
}
};
class BitrateObserver : public test::EndToEndTest, public test::FakeEncoder {
public:
BitrateObserver()
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
encoder_inits_(0),
last_set_bitrate_kbps_(0),
send_stream_(nullptr),
frame_generator_(nullptr),
encoder_factory_(this),
bitrate_allocator_factory_(
CreateBuiltinVideoBitrateAllocatorFactory()) {}
int32_t InitEncode(const VideoCodec* config,
const VideoEncoder::Settings& settings) override {
++encoder_inits_;
if (encoder_inits_ == 1) {
// First time initialization. Frame size is known.
// |expected_bitrate| is affected by bandwidth estimation before the
// first frame arrives to the encoder.
uint32_t expected_bitrate = last_set_bitrate_kbps_ > 0
? last_set_bitrate_kbps_
: kInitialBitrateKbps;
EXPECT_EQ(expected_bitrate, config->startBitrate)
<< "Encoder not initialized at expected bitrate.";
EXPECT_EQ(kDefaultWidth, config->width);
EXPECT_EQ(kDefaultHeight, config->height);
} else if (encoder_inits_ == 2) {
EXPECT_EQ(2 * kDefaultWidth, config->width);
EXPECT_EQ(2 * kDefaultHeight, config->height);
EXPECT_GE(last_set_bitrate_kbps_, kReconfigureThresholdKbps);
EXPECT_GT(config->startBitrate, kReconfigureThresholdKbps)
<< "Encoder reconfigured with bitrate too far away from last set.";
observation_complete_.Set();
}
return FakeEncoder::InitEncode(config, settings);
}
void SetRates(const RateControlParameters& parameters) override {
last_set_bitrate_kbps_ = parameters.bitrate.get_sum_kbps();
if (encoder_inits_ == 1 &&
parameters.bitrate.get_sum_kbps() > kReconfigureThresholdKbps) {
time_to_reconfigure_.Set();
}
FakeEncoder::SetRates(parameters);
}
void ModifySenderBitrateConfig(
BitrateConstraints* bitrate_config) override {
bitrate_config->start_bitrate_bps = kInitialBitrateKbps * 1000;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder_factory = &encoder_factory_;
send_config->encoder_settings.bitrate_allocator_factory =
bitrate_allocator_factory_.get();
encoder_config->max_bitrate_bps = 2 * kReconfigureThresholdKbps * 1000;
encoder_config->video_stream_factory =
new rtc::RefCountedObject<VideoStreamFactory>();
encoder_config_ = encoder_config->Copy();
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
frame_generator_ = frame_generator_capturer;
}
void PerformTest() override {
ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
<< "Timed out before receiving an initial high bitrate.";
frame_generator_->ChangeResolution(kDefaultWidth * 2, kDefaultHeight * 2);
send_stream_->ReconfigureVideoEncoder(encoder_config_.Copy());
EXPECT_TRUE(Wait())
<< "Timed out while waiting for a couple of high bitrate estimates "
"after reconfiguring the send stream.";
}
private:
rtc::Event time_to_reconfigure_;
int encoder_inits_;
uint32_t last_set_bitrate_kbps_;
VideoSendStream* send_stream_;
test::FrameGeneratorCapturer* frame_generator_;
test::VideoEncoderProxyFactory encoder_factory_;
std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
VideoEncoderConfig encoder_config_;
} test;
RunBaseTest(&test);
}
// Discovers the minimal supported audio+video bitrate. The test bitrate is
// considered supported if Rtt does not go above 400ms with the network
// contrained to the test bitrate.
//
// |test_bitrate_from test_bitrate_to| bitrate constraint range
// |test_bitrate_step| bitrate constraint update step during the test
// |min_bwe max_bwe| BWE range
// |start_bwe| initial BWE
void CallPerfTest::TestMinAudioVideoBitrate(int test_bitrate_from,
int test_bitrate_to,
int test_bitrate_step,
int min_bwe,
int start_bwe,
int max_bwe) {
static const std::string kAudioTrackId = "audio_track_0";
static constexpr int kOpusBitrateFbBps = 32000;
static constexpr int kBitrateStabilizationMs = 10000;
static constexpr int kBitrateMeasurements = 10;
static constexpr int kBitrateMeasurementMs = 1000;
static constexpr int kShortDelayMs = 10;
static constexpr int kMinGoodRttMs = 400;
class MinVideoAndAudioBitrateTester : public test::EndToEndTest {
public:
MinVideoAndAudioBitrateTester(int test_bitrate_from,
int test_bitrate_to,
int test_bitrate_step,
int min_bwe,
int start_bwe,
int max_bwe,
TaskQueueBase* task_queue)
: EndToEndTest(),
test_bitrate_from_(test_bitrate_from),
test_bitrate_to_(test_bitrate_to),
test_bitrate_step_(test_bitrate_step),
min_bwe_(min_bwe),
start_bwe_(start_bwe),
max_bwe_(max_bwe),
task_queue_(task_queue) {}
protected:
BuiltInNetworkBehaviorConfig GetFakeNetworkPipeConfig() {
BuiltInNetworkBehaviorConfig pipe_config;
pipe_config.link_capacity_kbps = test_bitrate_from_;
return pipe_config;
}
std::unique_ptr<test::PacketTransport> CreateSendTransport(
TaskQueueBase* task_queue,
Call* sender_call) override {
auto network =
std::make_unique<SimulatedNetwork>(GetFakeNetworkPipeConfig());
send_simulated_network_ = network.get();
return std::make_unique<test::PacketTransport>(
task_queue, sender_call, this, test::PacketTransport::kSender,
test::CallTest::payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network)));
}
std::unique_ptr<test::PacketTransport> CreateReceiveTransport(
TaskQueueBase* task_queue) override {
auto network =
std::make_unique<SimulatedNetwork>(GetFakeNetworkPipeConfig());
receive_simulated_network_ = network.get();
return std::make_unique<test::PacketTransport>(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
test::CallTest::payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network)));
}
void PerformTest() override {
// Quick test mode, just to exercise all the code paths without actually
// caring about performance measurements.
const bool quick_perf_test =
field_trial::IsEnabled("WebRTC-QuickPerfTest");
int last_passed_test_bitrate = -1;
for (int test_bitrate = test_bitrate_from_;
test_bitrate_from_ < test_bitrate_to_
? test_bitrate <= test_bitrate_to_
: test_bitrate >= test_bitrate_to_;
test_bitrate += test_bitrate_step_) {
BuiltInNetworkBehaviorConfig pipe_config;
pipe_config.link_capacity_kbps = test_bitrate;
send_simulated_network_->SetConfig(pipe_config);
receive_simulated_network_->SetConfig(pipe_config);
rtc::Thread::SleepMs(quick_perf_test ? kShortDelayMs
: kBitrateStabilizationMs);
int64_t avg_rtt = 0;
for (int i = 0; i < kBitrateMeasurements; i++) {
Call::Stats call_stats;
SendTask(RTC_FROM_HERE, task_queue_, [this, &call_stats]() {
call_stats = sender_call_->GetStats();
});
avg_rtt += call_stats.rtt_ms;
rtc::Thread::SleepMs(quick_perf_test ? kShortDelayMs
: kBitrateMeasurementMs);
}
avg_rtt = avg_rtt / kBitrateMeasurements;
if (avg_rtt > kMinGoodRttMs) {
break;
} else {
last_passed_test_bitrate = test_bitrate;
}
}
EXPECT_GT(last_passed_test_bitrate, -1)
<< "Minimum supported bitrate out of the test scope";
webrtc::test::PrintResult("min_test_bitrate_", "", "min_bitrate",
last_passed_test_bitrate, "kbps", false);
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
BitrateConstraints bitrate_config;
bitrate_config.min_bitrate_bps = min_bwe_;
bitrate_config.start_bitrate_bps = start_bwe_;
bitrate_config.max_bitrate_bps = max_bwe_;
sender_call->GetTransportControllerSend()->SetSdpBitrateParameters(
bitrate_config);
}
size_t GetNumVideoStreams() const override { return 1; }
size_t GetNumAudioStreams() const override { return 1; }
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->send_codec_spec->target_bitrate_bps =
absl::optional<int>(kOpusBitrateFbBps);
}
private:
const int test_bitrate_from_;
const int test_bitrate_to_;
const int test_bitrate_step_;
const int min_bwe_;
const int start_bwe_;
const int max_bwe_;
SimulatedNetwork* send_simulated_network_;
SimulatedNetwork* receive_simulated_network_;
Call* sender_call_;
TaskQueueBase* const task_queue_;
} test(test_bitrate_from, test_bitrate_to, test_bitrate_step, min_bwe,
start_bwe, max_bwe, task_queue());
RunBaseTest(&test);
}
// TODO(bugs.webrtc.org/8878)
#if defined(WEBRTC_MAC)
#define MAYBE_MinVideoAndAudioBitrate DISABLED_MinVideoAndAudioBitrate
#else
#define MAYBE_MinVideoAndAudioBitrate MinVideoAndAudioBitrate
#endif
TEST_F(CallPerfTest, MAYBE_MinVideoAndAudioBitrate) {
TestMinAudioVideoBitrate(110, 40, -10, 10000, 70000, 200000);
}
} // namespace webrtc
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