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webrtc/call/call_perf_tests.cc
Sebastian Jansson 8f83b42946 Moved bitrate config interface from Call class. 
Moving usage of bitrate configuration related interface from Call
interface to the corresponding methods in the RtpSendTransportController
interface.
SetBitrateConfig was replaced with SetSdpBitrateParameters
SetBitrateConfigMask was replaced with SetClientBitratePreferences
OnNetworkRouteChanged was replaced with OnNetworkRouteChanged

This makes it more clear that RtpSendTransportController owns bitrate
configuration and fits a longer term ambition to reduce the scope of
the Call class.

Bug: webrtc:8415
Change-Id: I6d04eaad22a54ecd5ed60096e01689b0c67e9c65
Reviewed-on: https://webrtc-review.googlesource.com/54365
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22131}
2018-02-21 15:03:45 +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 <algorithm>
#include <limits>
#include <memory>
#include <string>
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "call/call.h"
#include "call/video_config.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "rtc_base/bitrateallocationstrategy.h"
#include "rtc_base/checks.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/metrics_default.h"
#include "test/call_test.h"
#include "test/direct_transport.h"
#include "test/drifting_clock.h"
#include "test/encoder_settings.h"
#include "test/fake_audio_device.h"
#include "test/fake_encoder.h"
#include "test/field_trial.h"
#include "test/frame_generator.h"
#include "test/frame_generator_capturer.h"
#include "test/gtest.h"
#include "test/rtp_rtcp_observer.h"
#include "test/single_threaded_task_queue.h"
#include "test/testsupport/fileutils.h"
#include "test/testsupport/perf_test.h"
#include "video/transport_adapter.h"
using webrtc::test::DriftingClock;
using webrtc::test::FakeAudioDevice;
namespace webrtc {
class CallPerfTest : public test::CallTest {
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 FakeNetworkPipe::Config& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms);
void TestMinAudioVideoBitrate(bool use_bitrate_allocation_strategy,
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(Clock* clock, const std::string& test_label)
: test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs),
clock_(clock),
test_label_(test_label),
creation_time_ms_(clock_->TimeInMilliseconds()),
first_time_in_sync_(-1),
receive_stream_(nullptr) {}
void OnFrame(const VideoFrame& video_frame) override {
VideoReceiveStream::Stats stats;
{
rtc::CritScope lock(&crit_);
if (receive_stream_)
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::CritScope lock(&crit_);
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_;
rtc::CriticalSection crit_;
VideoReceiveStream* receive_stream_ RTC_GUARDED_BY(crit_);
std::vector<double> sync_offset_ms_list_;
};
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;
FakeNetworkPipe::Config audio_net_config;
audio_net_config.queue_delay_ms = 500;
audio_net_config.loss_percent = 5;
VideoRtcpAndSyncObserver observer(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;
AudioSendStream* audio_send_stream;
AudioReceiveStream* audio_receive_stream;
std::unique_ptr<DriftingClock> drifting_clock;
task_queue_.SendTask([&]() {
metrics::Reset();
rtc::scoped_refptr<FakeAudioDevice> fake_audio_device =
new rtc::RefCountedObject<FakeAudioDevice>(
FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000),
FakeAudioDevice::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(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(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 = rtc::MakeUnique<test::PacketTransport>(
&task_queue_, sender_call_.get(), &observer,
test::PacketTransport::kSender, audio_pt_map, audio_net_config);
audio_send_transport->SetReceiver(receiver_call_->Receiver());
video_send_transport = rtc::MakeUnique<test::PacketTransport>(
&task_queue_, sender_call_.get(), &observer,
test::PacketTransport::kSender, video_pt_map,
FakeNetworkPipe::Config());
video_send_transport->SetReceiver(receiver_call_->Receiver());
receive_transport = rtc::MakeUnique<test::PacketTransport>(
&task_queue_, receiver_call_.get(), &observer,
test::PacketTransport::kReceiver, payload_type_map_,
FakeNetworkPipe::Config());
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);
video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (fec == FecMode::kOn) {
video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType;
video_send_config_.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;
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.sync_group = kSyncGroup;
audio_recv_config.decoder_factory = 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 = rtc::MakeUnique<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.";
task_queue_.SendTask([&]() {
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")) {
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
}
}
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 FakeNetworkPipe::Config& 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 FakeNetworkPipe::Config& 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:
test::PacketTransport* CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) override {
return new test::PacketTransport(task_queue, sender_call, this,
test::PacketTransport::kSender,
payload_type_map_, net_config_);
}
test::PacketTransport* CreateReceiveTransport(
test::SingleThreadedTaskQueueForTesting* task_queue) override {
return new test::PacketTransport(task_queue, nullptr, this,
test::PacketTransport::kReceiver,
payload_type_map_, net_config_);
}
void OnFrame(const VideoFrame& video_frame) override {
rtc::CritScope lock(&crit_);
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 {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
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_ = header.timestamp - first_capture_timestamp;
rtp_start_timestamp_set_ = true;
}
uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_;
capture_time_list_.insert(
capture_time_list_.end(),
std::make_pair(header.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);
}
rtc::CriticalSection crit_;
const FakeNetworkPipe::Config 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(&crit_);
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 due to webrtc:8291.
#if !(defined(WEBRTC_MAC))
TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
FakeNetworkPipe::Config 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) {
FakeNetworkPipe::Config 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::kStart) {}
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 {
// First expect CPU overuse. Then expect CPU underuse when the encoder
// delay has been decreased.
switch (test_phase_) {
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 { 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.size() > 0) {
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_.SetMaxBitrate(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;
static const uint32_t kPermittedReconfiguredBitrateDiffKbps = 100;
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()),
time_to_reconfigure_(false, false),
encoder_inits_(0),
last_set_bitrate_kbps_(0),
send_stream_(nullptr),
frame_generator_(nullptr) {}
int32_t InitEncode(const VideoCodec* config,
int32_t number_of_cores,
size_t max_payload_size) 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,
last_set_bitrate_kbps_ - kPermittedReconfiguredBitrateDiffKbps)
<< "Encoder reconfigured with bitrate too far away from last set.";
observation_complete_.Set();
}
return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
}
int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
uint32_t framerate) override {
last_set_bitrate_kbps_ = rate_allocation.get_sum_kbps();
if (encoder_inits_ == 1 &&
rate_allocation.get_sum_kbps() > kReconfigureThresholdKbps) {
time_to_reconfigure_.Set();
}
return FakeEncoder::SetRateAllocation(rate_allocation, framerate);
}
Call::Config GetSenderCallConfig() override {
Call::Config config = EndToEndTest::GetSenderCallConfig();
config.event_log = event_log_.get();
config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
return config;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = this;
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_;
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.
//
// |use_bitrate_allocation_strategy| use AudioPriorityBitrateAllocationStrategy
// |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(
bool use_bitrate_allocation_strategy,
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 uint32_t kSufficientAudioBitrateBps = 16000;
static constexpr int kOpusMinBitrateBps = 6000;
static constexpr int kOpusBitrateFbBps = 32000;
static constexpr int kBitrateStabilizationMs = 10000;
static constexpr int kBitrateMeasurements = 10;
static constexpr int kBitrateMeasurementMs = 1000;
static constexpr int kMinGoodRttMs = 400;
class MinVideoAndAudioBitrateTester : public test::EndToEndTest {
public:
MinVideoAndAudioBitrateTester(bool use_bitrate_allocation_strategy,
int test_bitrate_from,
int test_bitrate_to,
int test_bitrate_step,
int min_bwe,
int start_bwe,
int max_bwe)
: EndToEndTest(),
allocation_strategy_(new rtc::AudioPriorityBitrateAllocationStrategy(
kAudioTrackId,
kSufficientAudioBitrateBps)),
use_bitrate_allocation_strategy_(use_bitrate_allocation_strategy),
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) {}
protected:
FakeNetworkPipe::Config GetFakeNetworkPipeConfig() {
FakeNetworkPipe::Config pipe_config;
pipe_config.link_capacity_kbps = test_bitrate_from_;
return pipe_config;
}
test::PacketTransport* CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) override {
return send_transport_ = new test::PacketTransport(
task_queue, sender_call, this, test::PacketTransport::kSender,
test::CallTest::payload_type_map_, GetFakeNetworkPipeConfig());
}
test::PacketTransport* CreateReceiveTransport(
test::SingleThreadedTaskQueueForTesting* task_queue) override {
return receive_transport_ = new test::PacketTransport(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
test::CallTest::payload_type_map_, GetFakeNetworkPipeConfig());
}
void PerformTest() override {
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_) {
FakeNetworkPipe::Config pipe_config;
pipe_config.link_capacity_kbps = test_bitrate;
send_transport_->SetConfig(pipe_config);
receive_transport_->SetConfig(pipe_config);
rtc::ThreadManager::Instance()->CurrentThread()->SleepMs(
kBitrateStabilizationMs);
int64_t avg_rtt = 0;
for (int i = 0; i < kBitrateMeasurements; i++) {
Call::Stats call_stats = sender_call_->GetStats();
avg_rtt += call_stats.rtt_ms;
rtc::ThreadManager::Instance()->CurrentThread()->SleepMs(
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_",
use_bitrate_allocation_strategy_ ? "with_allocation_strategy"
: "no_allocation_strategy",
"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);
if (use_bitrate_allocation_strategy_) {
sender_call->SetBitrateAllocationStrategy(
std::move(allocation_strategy_));
}
}
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 {
if (use_bitrate_allocation_strategy_) {
send_config->track_id = kAudioTrackId;
send_config->min_bitrate_bps = kOpusMinBitrateBps;
send_config->max_bitrate_bps = kOpusBitrateFbBps;
} else {
send_config->send_codec_spec->target_bitrate_bps =
rtc::Optional<int>(kOpusBitrateFbBps);
}
}
private:
std::unique_ptr<rtc::BitrateAllocationStrategy> allocation_strategy_;
const bool use_bitrate_allocation_strategy_;
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_;
test::PacketTransport* send_transport_;
test::PacketTransport* receive_transport_;
Call* sender_call_;
} test(use_bitrate_allocation_strategy, test_bitrate_from, test_bitrate_to,
test_bitrate_step, min_bwe, start_bwe, max_bwe);
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(false, 110, 40, -10, 10000, 70000, 200000);
}
TEST_F(CallPerfTest, MinVideoAndAudioBitrateWStrategy) {
TestMinAudioVideoBitrate(true, 110, 40, -10, 10000, 70000, 200000);
}
} // namespace webrtc
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