blob: 143b9616b70f1535084619a41b464ed403f1b5c1 [file] [log] [blame]
/*
* Copyright 2018 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 "absl/types/optional.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "rtc_base/synchronization/mutex.h"
#include "system_wrappers/include/metrics.h"
#include "test/call_test.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kTransportSequenceNumberExtensionId = 1,
kVideoContentTypeExtensionId,
};
} // namespace
class HistogramTest : public test::CallTest {
public:
HistogramTest() {
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
RegisterRtpExtension(RtpExtension(RtpExtension::kVideoContentTypeUri,
kVideoContentTypeExtensionId));
}
protected:
void VerifyHistogramStats(bool use_rtx, bool use_fec, bool screenshare);
};
void HistogramTest::VerifyHistogramStats(bool use_rtx,
bool use_fec,
bool screenshare) {
class FrameObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
FrameObserver(bool use_rtx, bool use_fec, bool screenshare)
: EndToEndTest(kLongTimeoutMs),
use_rtx_(use_rtx),
use_fec_(use_fec),
screenshare_(screenshare),
// This test uses NACK, so to send FEC we can't use a fake encoder.
encoder_factory_([]() { return VP8Encoder::Create(); }),
num_frames_received_(0) {}
private:
void OnFrame(const VideoFrame& video_frame) override {
// The RTT is needed to estimate `ntp_time_ms` which is used by
// end-to-end delay stats. Therefore, start counting received frames once
// `ntp_time_ms` is valid.
if (video_frame.ntp_time_ms() > 0 &&
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >=
video_frame.ntp_time_ms()) {
MutexLock lock(&mutex_);
++num_frames_received_;
}
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (MinMetricRunTimePassed() && MinNumberOfFramesReceived())
observation_complete_.Set();
return SEND_PACKET;
}
bool MinMetricRunTimePassed() {
int64_t now_ms = Clock::GetRealTimeClock()->TimeInMilliseconds();
if (!start_runtime_ms_)
start_runtime_ms_ = now_ms;
int64_t elapsed_sec = (now_ms - *start_runtime_ms_) / 1000;
return elapsed_sec > metrics::kMinRunTimeInSeconds * 2;
}
bool MinNumberOfFramesReceived() const {
const int kMinRequiredHistogramSamples = 200;
MutexLock lock(&mutex_);
return num_frames_received_ > kMinRequiredHistogramSamples;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// NACK
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].renderer = this;
// FEC
if (use_fec_) {
send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
send_config->encoder_settings.encoder_factory = &encoder_factory_;
send_config->rtp.payload_name = "VP8";
encoder_config->codec_type = kVideoCodecVP8;
(*receive_configs)[0].decoders[0].video_format = SdpVideoFormat("VP8");
(*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
}
// RTX
if (use_rtx_) {
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
(*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
(*receive_configs)[0]
.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
kFakeVideoSendPayloadType;
if (use_fec_) {
send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
(*receive_configs)[0]
.rtp.rtx_associated_payload_types[kRtxRedPayloadType] =
kSendRtxPayloadType;
}
}
// RTT needed for RemoteNtpTimeEstimator for the receive stream.
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
encoder_config->content_type =
screenshare_ ? VideoEncoderConfig::ContentType::kScreen
: VideoEncoderConfig::ContentType::kRealtimeVideo;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out waiting for min frames to be received.";
}
mutable Mutex mutex_;
const bool use_rtx_;
const bool use_fec_;
const bool screenshare_;
test::FunctionVideoEncoderFactory encoder_factory_;
absl::optional<int64_t> start_runtime_ms_;
int num_frames_received_ RTC_GUARDED_BY(&mutex_);
} test(use_rtx, use_fec, screenshare);
metrics::Reset();
RunBaseTest(&test);
const std::string video_prefix =
screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video.";
// The content type extension is disabled in non screenshare test,
// therefore no slicing on simulcast id should be present.
const std::string video_suffix = screenshare ? ".S0" : "";
// Verify that stats have been updated once.
EXPECT_METRIC_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(
"WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Call.PacerBitrateInKbps"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.SendStreamLifetimeInSeconds"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.NackPacketsSentPerMinute"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples(video_prefix + "NackPacketsReceivedPerMinute"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.FirPacketsSentPerMinute"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples(video_prefix + "FirPacketsReceivedPerMinute"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.PliPacketsSentPerMinute"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples(video_prefix + "PliPacketsReceivedPerMinute"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples(video_prefix + "KeyFramesSentInPermille"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples(video_prefix + "SentPacketsLostInPercent"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.ReceivedPacketsLostInPercent"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "InputWidthInPixels"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples(video_prefix + "InputHeightInPixels"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "SentWidthInPixels"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "SentHeightInPixels"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples(video_prefix + "ReceivedWidthInPixels"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples(video_prefix + "ReceivedHeightInPixels"));
EXPECT_METRIC_EQ(1, metrics::NumEvents(video_prefix + "InputWidthInPixels",
kDefaultWidth));
EXPECT_METRIC_EQ(1, metrics::NumEvents(video_prefix + "InputHeightInPixels",
kDefaultHeight));
EXPECT_METRIC_EQ(
1, metrics::NumEvents(video_prefix + "SentWidthInPixels", kDefaultWidth));
EXPECT_METRIC_EQ(1, metrics::NumEvents(video_prefix + "SentHeightInPixels",
kDefaultHeight));
EXPECT_METRIC_EQ(1, metrics::NumEvents(video_prefix + "ReceivedWidthInPixels",
kDefaultWidth));
EXPECT_METRIC_EQ(1,
metrics::NumEvents(video_prefix + "ReceivedHeightInPixels",
kDefaultHeight));
EXPECT_METRIC_EQ(1,
metrics::NumSamples(video_prefix + "InputFramesPerSecond"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples(video_prefix + "SentFramesPerSecond"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.RenderFramesPerSecond"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.JitterBufferDelayInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.TargetDelayInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.CurrentDelayInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.OnewayDelayInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs" +
video_suffix));
EXPECT_METRIC_EQ(1,
metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs" +
video_suffix));
EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayInMs" +
video_suffix));
EXPECT_METRIC_EQ(1,
metrics::NumSamples(video_prefix + "InterframeDelayMaxInMs" +
video_suffix));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "EncodeTimeInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.DecodeTimeInMs"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples(video_prefix + "NumberOfPauseEvents"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples(video_prefix + "PausedTimeInPercent"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "BitrateSentInKbps"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.BitrateReceivedInKbps"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples(video_prefix + "MediaBitrateSentInKbps"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.MediaBitrateReceivedInKbps"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples(video_prefix + "PaddingBitrateSentInKbps"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.PaddingBitrateReceivedInKbps"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples(video_prefix + "RetransmittedBitrateSentInKbps"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(
"WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.SendDelayInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayInMs"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples(video_prefix + "SendSideDelayMaxInMs"));
int num_rtx_samples = use_rtx ? 1 : 0;
EXPECT_METRIC_EQ(num_rtx_samples,
metrics::NumSamples("WebRTC.Video.RtxBitrateSentInKbps"));
EXPECT_METRIC_EQ(
num_rtx_samples,
metrics::NumSamples("WebRTC.Video.RtxBitrateReceivedInKbps"));
int num_red_samples = use_fec ? 1 : 0;
EXPECT_METRIC_EQ(num_red_samples,
metrics::NumSamples("WebRTC.Video.FecBitrateSentInKbps"));
EXPECT_METRIC_EQ(
num_red_samples,
metrics::NumSamples("WebRTC.Video.FecBitrateReceivedInKbps"));
EXPECT_METRIC_EQ(
num_red_samples,
metrics::NumSamples("WebRTC.Video.ReceivedFecPacketsInPercent"));
}
TEST_F(HistogramTest, VerifyStatsWithRtx) {
const bool kEnabledRtx = true;
const bool kEnabledRed = false;
const bool kScreenshare = false;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
TEST_F(HistogramTest, VerifyStatsWithRed) {
const bool kEnabledRtx = false;
const bool kEnabledRed = true;
const bool kScreenshare = false;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
TEST_F(HistogramTest, VerifyStatsWithScreenshare) {
const bool kEnabledRtx = false;
const bool kEnabledRed = false;
const bool kScreenshare = true;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
} // namespace webrtc