webrtc_m130/test/scenario/stats_collection.cc
Sebastian Jansson 7cbee84610 Reland "Adds PeerConnection scenario test framework."
This is a reland of ad5c4accad00e04de08e2b62d366cc1f8e0320a5

It was flaky due to starting ICE signaling before SDP negotiation
finished. This was solved by adding an helper for adding ice candidates
which will wait until the peer connection is ready if needed.

Original change's description:
> Adds PeerConnection scenario test framework.
>
> Bug: webrtc:10839
> Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28754}

Bug: webrtc:10839
Change-Id: I6eb8f482561c87e7b0f20d2431d21a41b26c91d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147877
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28777}
2019-08-06 16:12:12 +00:00

188 lines
6.6 KiB
C++

/*
* Copyright 2019 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 "test/scenario/stats_collection.h"
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "rtc_base/memory_usage.h"
namespace webrtc {
namespace test {
VideoQualityAnalyzer::VideoQualityAnalyzer(
VideoQualityAnalyzerConfig config,
std::unique_ptr<RtcEventLogOutput> writer)
: config_(config), writer_(std::move(writer)) {
if (writer_) {
PrintHeaders();
}
}
VideoQualityAnalyzer::~VideoQualityAnalyzer() = default;
void VideoQualityAnalyzer::PrintHeaders() {
writer_->Write(
"capture_time render_time capture_width capture_height render_width "
"render_height psnr\n");
}
std::function<void(const VideoFramePair&)> VideoQualityAnalyzer::Handler() {
return [this](VideoFramePair pair) { HandleFramePair(pair); };
}
void VideoQualityAnalyzer::HandleFramePair(VideoFramePair sample, double psnr) {
layer_analyzers_[sample.layer_id].HandleFramePair(sample, psnr,
writer_.get());
cached_.reset();
}
void VideoQualityAnalyzer::HandleFramePair(VideoFramePair sample) {
double psnr = NAN;
if (sample.decoded)
psnr = I420PSNR(*sample.captured->ToI420(), *sample.decoded->ToI420());
if (config_.thread) {
config_.thread->PostTask(RTC_FROM_HERE, [this, sample, psnr] {
HandleFramePair(std::move(sample), psnr);
});
} else {
HandleFramePair(std::move(sample), psnr);
}
}
std::vector<VideoQualityStats> VideoQualityAnalyzer::layer_stats() const {
std::vector<VideoQualityStats> res;
for (auto& layer : layer_analyzers_)
res.push_back(layer.second.stats_);
return res;
}
VideoQualityStats& VideoQualityAnalyzer::stats() {
if (!cached_) {
cached_ = VideoQualityStats();
for (auto& layer : layer_analyzers_)
cached_->AddStats(layer.second.stats_);
}
return *cached_;
}
void VideoLayerAnalyzer::HandleFramePair(VideoFramePair sample,
double psnr,
RtcEventLogOutput* writer) {
RTC_CHECK(sample.captured);
HandleCapturedFrame(sample);
if (!sample.decoded) {
// Can only happen in the beginning of a call or if the resolution is
// reduced. Otherwise we will detect a freeze.
++stats_.lost_count;
++skip_count_;
} else {
stats_.psnr_with_freeze.AddSample(psnr);
if (sample.repeated) {
++stats_.freeze_count;
++skip_count_;
} else {
stats_.psnr.AddSample(psnr);
HandleRenderedFrame(sample);
}
}
if (writer) {
LogWriteFormat(writer, "%.3f %.3f %.3f %i %i %i %i %.3f\n",
sample.capture_time.seconds<double>(),
sample.render_time.seconds<double>(),
sample.captured->width(), sample.captured->height(),
sample.decoded->width(), sample.decoded->height(), psnr);
}
}
void VideoLayerAnalyzer::HandleCapturedFrame(const VideoFramePair& sample) {
stats_.capture.AddFrameInfo(*sample.captured, sample.capture_time);
if (last_freeze_time_.IsInfinite())
last_freeze_time_ = sample.capture_time;
}
void VideoLayerAnalyzer::HandleRenderedFrame(const VideoFramePair& sample) {
stats_.capture_to_decoded_delay.AddSample(sample.decoded_time -
sample.capture_time);
stats_.end_to_end_delay.AddSample(sample.render_time - sample.capture_time);
stats_.render.AddFrameInfo(*sample.decoded, sample.render_time);
stats_.skipped_between_rendered.AddSample(skip_count_);
skip_count_ = 0;
if (last_render_time_.IsFinite()) {
RTC_DCHECK(sample.render_time.IsFinite());
TimeDelta render_interval = sample.render_time - last_render_time_;
TimeDelta mean_interval = stats_.render.frames.interval().Mean();
if (render_interval > TimeDelta::ms(150) + mean_interval ||
render_interval > 3 * mean_interval) {
stats_.freeze_duration.AddSample(render_interval);
stats_.time_between_freezes.AddSample(last_render_time_ -
last_freeze_time_);
last_freeze_time_ = sample.render_time;
}
}
last_render_time_ = sample.render_time;
}
void CallStatsCollector::AddStats(Call::Stats sample) {
if (sample.send_bandwidth_bps > 0)
stats_.target_rate.AddSampleBps(sample.send_bandwidth_bps);
if (sample.pacer_delay_ms > 0)
stats_.pacer_delay.AddSample(TimeDelta::ms(sample.pacer_delay_ms));
if (sample.rtt_ms > 0)
stats_.round_trip_time.AddSample(TimeDelta::ms(sample.rtt_ms));
stats_.memory_usage.AddSample(rtc::GetProcessResidentSizeBytes());
}
void AudioReceiveStatsCollector::AddStats(AudioReceiveStream::Stats sample) {
stats_.expand_rate.AddSample(sample.expand_rate);
stats_.accelerate_rate.AddSample(sample.accelerate_rate);
stats_.jitter_buffer.AddSampleMs(sample.jitter_buffer_ms);
}
void VideoSendStatsCollector::AddStats(VideoSendStream::Stats sample,
Timestamp at_time) {
// It's not certain that we yet have estimates for any of these stats.
// Check that they are positive before mixing them in.
if (sample.encode_frame_rate <= 0)
return;
stats_.encode_frame_rate.AddSample(sample.encode_frame_rate);
stats_.encode_time.AddSampleMs(sample.avg_encode_time_ms);
stats_.encode_usage.AddSample(sample.encode_usage_percent / 100.0);
stats_.media_bitrate.AddSampleBps(sample.media_bitrate_bps);
size_t fec_bytes = 0;
for (const auto& kv : sample.substreams) {
fec_bytes += kv.second.rtp_stats.fec.payload_bytes +
kv.second.rtp_stats.fec.padding_bytes;
}
if (last_update_.IsFinite()) {
auto fec_delta = DataSize::bytes(fec_bytes - last_fec_bytes_);
auto time_delta = at_time - last_update_;
stats_.fec_bitrate.AddSample(fec_delta / time_delta);
}
last_fec_bytes_ = fec_bytes;
last_update_ = at_time;
}
void VideoReceiveStatsCollector::AddStats(VideoReceiveStream::Stats sample) {
if (sample.decode_ms > 0)
stats_.decode_time.AddSampleMs(sample.decode_ms);
if (sample.max_decode_ms > 0)
stats_.decode_time_max.AddSampleMs(sample.max_decode_ms);
if (sample.width > 0 && sample.height > 0) {
stats_.decode_pixels.AddSample(sample.width * sample.height);
stats_.resolution.AddSample(sample.height);
}
}
} // namespace test
} // namespace webrtc