aosp12/external/webrtc/test/scenario/stats_collection.cc

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2023-01-09 17:11:35 +08:00
/*
* 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"
#include "rtc_base/thread.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 ? sample.decoded->width() : 0,
sample.decoded ? sample.decoded->height() : 0, 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::Millis(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::Millis(sample.pacer_delay_ms));
if (sample.rtt_ms > 0)
stats_.round_trip_time.AddSample(TimeDelta::Millis(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