/* * 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 #include #include #include #include #include #include "api/rtc_event_log_output.h" #include "api/units/data_rate.h" #include "api/units/data_size.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "call/audio_receive_stream.h" #include "call/call.h" #include "call/video_receive_stream.h" #include "call/video_send_stream.h" #include "common_video/libyuv/include/webrtc_libyuv.h" #include "rtc_base/checks.h" #include "rtc_base/memory_usage.h" #include "rtc_base/thread.h" #include "system_wrappers/include/clock.h" #include "test/logging/log_writer.h" #include "test/scenario/performance_stats.h" namespace webrtc { namespace test { VideoQualityAnalyzer::VideoQualityAnalyzer( VideoQualityAnalyzerConfig config, std::unique_ptr 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 VideoQualityAnalyzer::Handler( Clock* clock) { return [this, clock](VideoFramePair pair) { HandleFramePair(pair, clock->CurrentTime()); }; } void VideoQualityAnalyzer::HandleFramePair(VideoFramePair sample, double psnr, Timestamp at_time) { layer_analyzers_[sample.layer_id].HandleFramePair(sample, psnr, writer_.get(), at_time); cached_.reset(); } void VideoQualityAnalyzer::HandleFramePair(VideoFramePair sample, Timestamp at_time) { double psnr = NAN; if (sample.decoded) psnr = I420PSNR(*sample.captured->ToI420(), *sample.decoded->ToI420()); if (config_.thread) { config_.thread->PostTask([this, sample, psnr, at_time] { HandleFramePair(std::move(sample), psnr, at_time); }); } else { HandleFramePair(std::move(sample), psnr, at_time); } } std::vector VideoQualityAnalyzer::layer_stats() const { std::vector 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, Timestamp at_time) { 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({.value = psnr, .time = at_time}); if (sample.repeated) { ++stats_.freeze_count; ++skip_count_; } else { stats_.psnr.AddSample({.value = psnr, .time = at_time}); HandleRenderedFrame(sample); } } if (writer) { LogWriteFormat(writer, "%.3f %.3f %.3f %i %i %i %i %.3f\n", sample.capture_time.seconds(), sample.render_time.seconds(), 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, sample.capture_time); stats_.end_to_end_delay.AddSample(sample.render_time - sample.capture_time, sample.capture_time); stats_.render.AddFrameInfo(*sample.decoded, sample.render_time); stats_.skipped_between_rendered.AddSample( {.value = static_cast(skip_count_), .time = sample.render_time}); 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, sample.capture_time); stats_.time_between_freezes.AddSample( last_render_time_ - last_freeze_time_, sample.capture_time); last_freeze_time_ = sample.render_time; } } last_render_time_ = sample.render_time; } void CallStatsCollector::AddStats(Call::Stats sample, Timestamp at_time) { if (sample.send_bandwidth_bps > 0) stats_.target_rate.AddSample( DataRate::BitsPerSec(sample.send_bandwidth_bps), at_time); if (sample.pacer_delay_ms > 0) stats_.pacer_delay.AddSample(TimeDelta::Millis(sample.pacer_delay_ms), at_time); if (sample.rtt_ms > 0) stats_.round_trip_time.AddSample(TimeDelta::Millis(sample.rtt_ms), at_time); stats_.memory_usage.AddSample( {.value = static_cast(GetProcessResidentSizeBytes()), .time = at_time}); } void AudioReceiveStatsCollector::AddStats( AudioReceiveStreamInterface::Stats sample, Timestamp at_time) { stats_.expand_rate.AddSample({.value = sample.expand_rate, .time = at_time}); stats_.accelerate_rate.AddSample( {.value = sample.accelerate_rate, .time = at_time}); stats_.jitter_buffer.AddSample(TimeDelta::Millis(sample.jitter_buffer_ms), at_time); } 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( {.value = static_cast(sample.encode_frame_rate), .time = at_time}); stats_.encode_time.AddSample(TimeDelta::Millis(sample.avg_encode_time_ms), at_time); stats_.encode_usage.AddSample( {.value = sample.encode_usage_percent / 100.0, .time = at_time}); stats_.media_bitrate.AddSample(DataRate::BitsPerSec(sample.media_bitrate_bps), at_time); 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, at_time); } last_fec_bytes_ = fec_bytes; last_update_ = at_time; } void VideoReceiveStatsCollector::AddStats( VideoReceiveStreamInterface::Stats sample, Timestamp at_time) { if (sample.decode_ms > 0) stats_.decode_time.AddSample(TimeDelta::Millis(sample.decode_ms), at_time); if (sample.max_decode_ms > 0) stats_.decode_time_max.AddSample(TimeDelta::Millis(sample.max_decode_ms), at_time); if (sample.width > 0 && sample.height > 0) { stats_.decode_pixels.AddSample( {.value = static_cast(sample.width * sample.height), .time = at_time}); stats_.resolution.AddSample( {.value = static_cast(sample.height), .time = at_time}); } } } // namespace test } // namespace webrtc