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Moving src/webrtc into src/.

Browse Source 
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
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788
call/call_perf_tests.cc Normal file
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@ -0,0 +1,788 @@
/*
* 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 "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
#include "webrtc/call/call.h"
#include "webrtc/call/video_config.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/thread_annotations.h"
#include "webrtc/system_wrappers/include/metrics_default.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/drifting_clock.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/frame_generator.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/rtp_rtcp_observer.h"
#include "webrtc/test/single_threaded_task_queue.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/video/transport_adapter.h"
#include "webrtc/voice_engine/include/voe_base.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);
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);
};
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)
: test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs),
clock_(clock),
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",
"",
"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", "", "synchronization",
test::ValuesToString(sync_offset_ms_list_), "ms",
false);
}
private:
Clock* const clock_;
const int64_t creation_time_ms_;
int64_t first_time_in_sync_;
rtc::CriticalSection crit_;
VideoReceiveStream* receive_stream_ RTC_GUARDED_BY(crit_);
std::vector<int> 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 char* kSyncGroup = "av_sync";
const uint32_t kAudioSendSsrc = 1234;
const uint32_t kAudioRecvSsrc = 5678;
int send_channel_id;
int recv_channel_id;
FakeNetworkPipe::Config audio_net_config;
audio_net_config.queue_delay_ms = 500;
audio_net_config.loss_percent = 5;
rtc::scoped_refptr<AudioProcessing> audio_processing;
VoiceEngine* voice_engine;
VoEBase* voe_base;
std::unique_ptr<FakeAudioDevice> fake_audio_device;
VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock());
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();
audio_processing = AudioProcessing::Create();
voice_engine = VoiceEngine::Create();
voe_base = VoEBase::GetInterface(voice_engine);
fake_audio_device = rtc::MakeUnique<FakeAudioDevice>(
FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000),
FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed);
EXPECT_EQ(0, voe_base->Init(fake_audio_device.get(), audio_processing.get(),
decoder_factory_));
VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
send_channel_id = voe_base->CreateChannel(config);
recv_channel_id = voe_base->CreateChannel();
AudioState::Config send_audio_state_config;
send_audio_state_config.voice_engine = voice_engine;
send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
send_audio_state_config.audio_processing = audio_processing;
Call::Config sender_config(event_log_.get());
sender_config.audio_state = AudioState::Create(send_audio_state_config);
Call::Config receiver_config(event_log_.get());
receiver_config.audio_state = sender_config.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.voe_channel_id = send_channel_id;
audio_send_config.rtp.ssrc = kAudioSendSsrc;
audio_send_config.send_codec_spec =
rtc::Optional<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.ulpfec.red_payload_type = kRedPayloadType;
video_receive_configs_[0].rtp.ulpfec.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.voe_channel_id = recv_channel_id;
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);
voe_base->DeleteChannel(send_channel_id);
voe_base->DeleteChannel(recv_channel_id);
voe_base->Release();
DestroyCalls();
VoiceEngine::Delete(voice_engine);
fake_audio_device.reset();
});
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);
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::kNoDrift,
DriftingClock::PercentsSlower(30.0f),
DriftingClock::PercentsFaster(30.0f));
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift) {
TestAudioVideoSync(FecMode::kOn, CreateOrder::kVideoFirst,
DriftingClock::kNoDrift,
DriftingClock::PercentsFaster(30.0f),
DriftingClock::PercentsSlower(30.0f));
}
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",
test::ValuesToString(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<int> time_offset_ms_list_;
} test(net_config, threshold_ms, start_time_ms, run_time_ms);
RunBaseTest(&test);
}
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);
}
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", test::ValuesToString(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<size_t> 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);
}
TEST_F(CallPerfTest, 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);
}
} // namespace webrtc