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Ensure test capturer does not set ntp time

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This aligns with chromium capturer.
https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/platform/peerconnection/webrtc_video_track_source.cc;l=390-401;drc=c0265133106c7647e90f9aaa4377d28190b1a6a9?q=webrtc_video_track_source.cc&ss=chromium

Bug: webrtc:42223979
Change-Id: Ibc60297e49e44b1d55a3869d68b20feba7aa38f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42401}
This commit is contained in:
Per K 2024-05-29 08:41:34 +00:00 committed by WebRTC LUCI CQ
parent 19c51ea537
commit f4ff3f34e0
3 changed files with 0 additions and 166 deletions
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156
call/call_perf_tests.cc
View file

@ -405,162 +405,6 @@ TEST_F(CallPerfTest,
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(test::VideoTestConstants::kLongTimeout),
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:
BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
return net_config_;
}
BuiltInNetworkBehaviorConfig GetReceiveTransportConfig() const override {
return 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.rtp_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_.AddSample(time_offset_ms);
EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
}
Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
MutexLock lock(&mutex_);
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet));
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<VideoReceiveStreamInterface::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.";
GetGlobalMetricsLogger()->LogMetric(
"capture_ntp_time", "real - estimated", time_offset_ms_list_,
Unit::kMilliseconds, ImprovementDirection::kNeitherIsBetter);
}
Mutex mutex_;
const BuiltInNetworkBehaviorConfig net_config_;
Clock* const clock_;
const int threshold_ms_;
const int start_time_ms_;
const int run_time_ms_;
const 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_);
SamplesStatsCounter 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, Real_Estimated_CaptureNtpTimeWithNetworkDelay) {
BuiltInNetworkBehaviorConfig net_config;
net_config.queue_delay_ms = 100;
// TODO(wu): lower the threshold as the calculation/estimation becomes more
// accurate.
const int kThresholdMs = 100;
const int kStartTimeMs = 10000;
const int kRunTimeMs = 20000;
TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
}
TEST_F(CallPerfTest, Real_Estimated_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/estimation 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.

6
test/frame_generator_capturer.cc
View file

@ -47,7 +47,6 @@ FrameGeneratorCapturer::FrameGeneratorCapturer(
frame_generator_(std::move(frame_generator)),
source_fps_(target_fps),
target_capture_fps_(target_fps),
first_frame_capture_time_(-1),
task_queue_(task_queue_factory.CreateTaskQueue(
"FrameGenCapQ",
TaskQueueFactory::Priority::HIGH)) {
@ -106,14 +105,9 @@ void FrameGeneratorCapturer::InsertFrame() {
.set_video_frame_buffer(frame_data.buffer)
.set_rotation(fake_rotation_)
.set_timestamp_us(clock_->TimeInMicroseconds())
.set_ntp_time_ms(clock_->CurrentNtpInMilliseconds())
.set_update_rect(frame_data.update_rect)
.set_color_space(fake_color_space_)
.build();
if (first_frame_capture_time_ == -1) {
first_frame_capture_time_ = frame.ntp_time_ms();
}
TestVideoCapturer::OnFrame(frame);
}
}

4
test/frame_generator_capturer.h
View file

@ -80,8 +80,6 @@ class FrameGeneratorCapturer : public TestVideoCapturer {
void SetFakeRotation(VideoRotation rotation);
void SetFakeColorSpace(absl::optional<ColorSpace> color_space);
int64_t first_frame_capture_time() const { return first_frame_capture_time_; }
bool Init();
private:
@ -104,8 +102,6 @@ class FrameGeneratorCapturer : public TestVideoCapturer {
VideoRotation fake_rotation_ = kVideoRotation_0;
absl::optional<ColorSpace> fake_color_space_ RTC_GUARDED_BY(&lock_);
int64_t first_frame_capture_time_;
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
};
} // namespace test