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webrtc_m130/video/end_to_end_tests/histogram_tests.cc
Sebastian Jansson c5017136c7 Split end to end tests into more cohesive test sets. 
end_to_end_tests.cc was over 5000 lines and covered many different
areas in it's testing. In this change it is separated into multiple
smaller test sets separated by the functionality they are testing. The
reasoning behind this is that the fact that a test is working end to end
should be secondary to what functionality the test is actually testing.

A slight functional change is that for some of the tests the
parametrization over round robin pacing being controlled with a field
trial is removed since they are simple enough that they should not be
affected by the pacing method.

Bug: None
Change-Id: I4b7eba80fc142ecfc8fa642dab9b6f587d914048
Reviewed-on: https://webrtc-review.googlesource.com/46143
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21876}
2018-02-02 16:09:16 +00:00

278 lines
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/*
* 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 "modules/video_coding/codecs/vp8/include/vp8.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/metrics_default.h"
#include "test/call_test.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
namespace webrtc {
class HistogramTest : public test::CallTest {
protected:
void VerifyHistogramStats(bool use_rtx, bool use_fec, bool screenshare);
};
void HistogramTest::VerifyHistogramStats(bool use_rtx,
bool use_fec,
bool screenshare) {
class StatsObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
StatsObserver(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.
vp8_encoder_(use_fec ? VP8Encoder::Create() : nullptr),
sender_call_(nullptr),
receiver_call_(nullptr),
start_runtime_ms_(-1),
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()) {
rtc::CritScope lock(&crit_);
++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 = Clock::GetRealTimeClock()->TimeInMilliseconds();
if (start_runtime_ms_ == -1) {
start_runtime_ms_ = now;
return false;
}
int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
return elapsed_sec > metrics::kMinRunTimeInSeconds * 2;
}
bool MinNumberOfFramesReceived() const {
const int kMinRequiredHistogramSamples = 200;
rtc::CritScope lock(&crit_);
return num_frames_received_ > kMinRequiredHistogramSamples;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::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 = vp8_encoder_.get();
send_config->encoder_settings.payload_name = "VP8";
(*receive_configs)[0].decoders[0].payload_name = "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 OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
receiver_call_ = receiver_call;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
}
rtc::CriticalSection crit_;
const bool use_rtx_;
const bool use_fec_;
const bool screenshare_;
const std::unique_ptr<VideoEncoder> vp8_encoder_;
Call* sender_call_;
Call* receiver_call_;
int64_t start_runtime_ms_;
int num_frames_received_ RTC_GUARDED_BY(&crit_);
} test(use_rtx, use_fec, screenshare);
metrics::Reset();
RunBaseTest(&test);
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.
std::string video_suffix = screenshare ? ".S0" : "";
// Verify that stats have been updated once.
EXPECT_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
EXPECT_EQ(1, metrics::NumSamples(
"WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.PacerBitrateInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendStreamLifetimeInSeconds"));
EXPECT_EQ(1,
metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.NackPacketsSentPerMinute"));
EXPECT_EQ(1,
metrics::NumSamples(video_prefix + "NackPacketsReceivedPerMinute"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.FirPacketsSentPerMinute"));
EXPECT_EQ(1,
metrics::NumSamples(video_prefix + "FirPacketsReceivedPerMinute"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.PliPacketsSentPerMinute"));
EXPECT_EQ(1,
metrics::NumSamples(video_prefix + "PliPacketsReceivedPerMinute"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "KeyFramesSentInPermille"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentPacketsLostInPercent"));
EXPECT_EQ(1,
metrics::NumSamples("WebRTC.Video.ReceivedPacketsLostInPercent"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputWidthInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputHeightInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentWidthInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentHeightInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "ReceivedWidthInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "ReceivedHeightInPixels"));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "InputWidthInPixels",
kDefaultWidth));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "InputHeightInPixels",
kDefaultHeight));
EXPECT_EQ(
1, metrics::NumEvents(video_prefix + "SentWidthInPixels", kDefaultWidth));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "SentHeightInPixels",
kDefaultHeight));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "ReceivedWidthInPixels",
kDefaultWidth));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "ReceivedHeightInPixels",
kDefaultHeight));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputFramesPerSecond"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentFramesPerSecond"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderFramesPerSecond"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.JitterBufferDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.TargetDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.CurrentDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.OnewayDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs" +
video_suffix));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs" +
video_suffix));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayInMs" +
video_suffix));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayMaxInMs" +
video_suffix));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EncodeTimeInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodeTimeInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "NumberOfPauseEvents"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PausedTimeInPercent"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "BitrateSentInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.BitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "MediaBitrateSentInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.MediaBitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PaddingBitrateSentInKbps"));
EXPECT_EQ(1,
metrics::NumSamples("WebRTC.Video.PaddingBitrateReceivedInKbps"));
EXPECT_EQ(
1, metrics::NumSamples(video_prefix + "RetransmittedBitrateSentInKbps"));
EXPECT_EQ(1, metrics::NumSamples(
"WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayMaxInMs"));
int num_rtx_samples = use_rtx ? 1 : 0;
EXPECT_EQ(num_rtx_samples,
metrics::NumSamples("WebRTC.Video.RtxBitrateSentInKbps"));
EXPECT_EQ(num_rtx_samples,
metrics::NumSamples("WebRTC.Video.RtxBitrateReceivedInKbps"));
int num_red_samples = use_fec ? 1 : 0;
EXPECT_EQ(num_red_samples,
metrics::NumSamples("WebRTC.Video.FecBitrateSentInKbps"));
EXPECT_EQ(num_red_samples,
metrics::NumSamples("WebRTC.Video.FecBitrateReceivedInKbps"));
EXPECT_EQ(num_red_samples,
metrics::NumSamples("WebRTC.Video.ReceivedFecPacketsInPercent"));
}
TEST_F(HistogramTest, VerifyHistogramStatsWithRtx) {
const bool kEnabledRtx = true;
const bool kEnabledRed = false;
const bool kScreenshare = false;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
TEST_F(HistogramTest, VerifyHistogramStatsWithRed) {
const bool kEnabledRtx = false;
const bool kEnabledRed = true;
const bool kScreenshare = false;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
TEST_F(HistogramTest, VerifyHistogramStatsWithScreenshare) {
const bool kEnabledRtx = false;
const bool kEnabledRed = false;
const bool kScreenshare = true;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
} // namespace webrtc
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