| /* |
| * 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 "video/video_analyzer.h" |
| |
| #include <algorithm> |
| #include <utility> |
| |
| #include "common_video/libyuv/include/webrtc_libyuv.h" |
| #include "modules/rtp_rtcp/source/rtp_format.h" |
| #include "modules/rtp_rtcp/source/rtp_utility.h" |
| #include "rtc_base/cpu_time.h" |
| #include "rtc_base/flags.h" |
| #include "rtc_base/format_macros.h" |
| #include "rtc_base/memory_usage.h" |
| #include "system_wrappers/include/cpu_info.h" |
| #include "test/call_test.h" |
| #include "test/testsupport/fileutils.h" |
| #include "test/testsupport/frame_writer.h" |
| #include "test/testsupport/perf_test.h" |
| #include "test/testsupport/test_artifacts.h" |
| |
| WEBRTC_DEFINE_bool( |
| save_worst_frame, |
| false, |
| "Enable saving a frame with the lowest PSNR to a jpeg file in the " |
| "test_artifacts_dir"); |
| |
| namespace webrtc { |
| namespace { |
| constexpr int kSendStatsPollingIntervalMs = 1000; |
| constexpr size_t kMaxComparisons = 10; |
| |
| bool IsFlexfec(int payload_type) { |
| return payload_type == test::CallTest::kFlexfecPayloadType; |
| } |
| } // namespace |
| |
| VideoAnalyzer::VideoAnalyzer(test::LayerFilteringTransport* transport, |
| const std::string& test_label, |
| double avg_psnr_threshold, |
| double avg_ssim_threshold, |
| int duration_frames, |
| FILE* graph_data_output_file, |
| const std::string& graph_title, |
| uint32_t ssrc_to_analyze, |
| uint32_t rtx_ssrc_to_analyze, |
| size_t selected_stream, |
| int selected_sl, |
| int selected_tl, |
| bool is_quick_test_enabled, |
| Clock* clock, |
| std::string rtp_dump_name) |
| : transport_(transport), |
| receiver_(nullptr), |
| call_(nullptr), |
| send_stream_(nullptr), |
| receive_stream_(nullptr), |
| audio_receive_stream_(nullptr), |
| captured_frame_forwarder_(this, clock), |
| test_label_(test_label), |
| graph_data_output_file_(graph_data_output_file), |
| graph_title_(graph_title), |
| ssrc_to_analyze_(ssrc_to_analyze), |
| rtx_ssrc_to_analyze_(rtx_ssrc_to_analyze), |
| selected_stream_(selected_stream), |
| selected_sl_(selected_sl), |
| selected_tl_(selected_tl), |
| last_fec_bytes_(0), |
| frames_to_process_(duration_frames), |
| frames_recorded_(0), |
| frames_processed_(0), |
| dropped_frames_(0), |
| dropped_frames_before_first_encode_(0), |
| dropped_frames_before_rendering_(0), |
| last_render_time_(0), |
| last_render_delta_ms_(0), |
| last_unfreeze_time_ms_(0), |
| rtp_timestamp_delta_(0), |
| total_media_bytes_(0), |
| first_sending_time_(0), |
| last_sending_time_(0), |
| cpu_time_(0), |
| wallclock_time_(0), |
| avg_psnr_threshold_(avg_psnr_threshold), |
| avg_ssim_threshold_(avg_ssim_threshold), |
| is_quick_test_enabled_(is_quick_test_enabled), |
| stats_polling_thread_(&PollStatsThread, this, "StatsPoller"), |
| done_(true, false), |
| clock_(clock), |
| start_ms_(clock->TimeInMilliseconds()) { |
| // Create thread pool for CPU-expensive PSNR/SSIM calculations. |
| |
| // Try to use about as many threads as cores, but leave kMinCoresLeft alone, |
| // so that we don't accidentally starve "real" worker threads (codec etc). |
| // Also, don't allocate more than kMaxComparisonThreads, even if there are |
| // spare cores. |
| |
| uint32_t num_cores = CpuInfo::DetectNumberOfCores(); |
| RTC_DCHECK_GE(num_cores, 1); |
| static const uint32_t kMinCoresLeft = 4; |
| static const uint32_t kMaxComparisonThreads = 8; |
| |
| if (num_cores <= kMinCoresLeft) { |
| num_cores = 1; |
| } else { |
| num_cores -= kMinCoresLeft; |
| num_cores = std::min(num_cores, kMaxComparisonThreads); |
| } |
| |
| for (uint32_t i = 0; i < num_cores; ++i) { |
| rtc::PlatformThread* thread = |
| new rtc::PlatformThread(&FrameComparisonThread, this, "Analyzer"); |
| thread->Start(); |
| comparison_thread_pool_.push_back(thread); |
| } |
| |
| if (!rtp_dump_name.empty()) { |
| fprintf(stdout, "Writing rtp dump to %s\n", rtp_dump_name.c_str()); |
| rtp_file_writer_.reset(test::RtpFileWriter::Create( |
| test::RtpFileWriter::kRtpDump, rtp_dump_name)); |
| } |
| } |
| |
| VideoAnalyzer::~VideoAnalyzer() { |
| for (rtc::PlatformThread* thread : comparison_thread_pool_) { |
| thread->Stop(); |
| delete thread; |
| } |
| } |
| |
| void VideoAnalyzer::SetReceiver(PacketReceiver* receiver) { |
| receiver_ = receiver; |
| } |
| |
| void VideoAnalyzer::SetSource( |
| rtc::VideoSourceInterface<VideoFrame>* video_source, |
| bool respect_sink_wants) { |
| if (respect_sink_wants) |
| captured_frame_forwarder_.SetSource(video_source); |
| rtc::VideoSinkWants wants; |
| video_source->AddOrUpdateSink(InputInterface(), wants); |
| } |
| |
| void VideoAnalyzer::SetCall(Call* call) { |
| rtc::CritScope lock(&crit_); |
| RTC_DCHECK(!call_); |
| call_ = call; |
| } |
| |
| void VideoAnalyzer::SetSendStream(VideoSendStream* stream) { |
| rtc::CritScope lock(&crit_); |
| RTC_DCHECK(!send_stream_); |
| send_stream_ = stream; |
| } |
| |
| void VideoAnalyzer::SetReceiveStream(VideoReceiveStream* stream) { |
| rtc::CritScope lock(&crit_); |
| RTC_DCHECK(!receive_stream_); |
| receive_stream_ = stream; |
| } |
| |
| void VideoAnalyzer::SetAudioReceiveStream(AudioReceiveStream* recv_stream) { |
| rtc::CritScope lock(&crit_); |
| RTC_CHECK(!audio_receive_stream_); |
| audio_receive_stream_ = recv_stream; |
| } |
| |
| rtc::VideoSinkInterface<VideoFrame>* VideoAnalyzer::InputInterface() { |
| return &captured_frame_forwarder_; |
| } |
| |
| rtc::VideoSourceInterface<VideoFrame>* VideoAnalyzer::OutputInterface() { |
| return &captured_frame_forwarder_; |
| } |
| |
| PacketReceiver::DeliveryStatus VideoAnalyzer::DeliverPacket( |
| MediaType media_type, |
| rtc::CopyOnWriteBuffer packet, |
| int64_t packet_time_us) { |
| // Ignore timestamps of RTCP packets. They're not synchronized with |
| // RTP packet timestamps and so they would confuse wrap_handler_. |
| if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) { |
| return receiver_->DeliverPacket(media_type, std::move(packet), |
| packet_time_us); |
| } |
| |
| if (rtp_file_writer_) { |
| test::RtpPacket p; |
| memcpy(p.data, packet.cdata(), packet.size()); |
| p.length = packet.size(); |
| p.original_length = packet.size(); |
| p.time_ms = clock_->TimeInMilliseconds() - start_ms_; |
| rtp_file_writer_->WritePacket(&p); |
| } |
| |
| RtpUtility::RtpHeaderParser parser(packet.cdata(), packet.size()); |
| RTPHeader header; |
| parser.Parse(&header); |
| if (!IsFlexfec(header.payloadType) && (header.ssrc == ssrc_to_analyze_ || |
| header.ssrc == rtx_ssrc_to_analyze_)) { |
| // Ignore FlexFEC timestamps, to avoid collisions with media timestamps. |
| // (FlexFEC and media are sent on different SSRCs, which have different |
| // timestamps spaces.) |
| // Also ignore packets from wrong SSRC, but include retransmits. |
| rtc::CritScope lock(&crit_); |
| int64_t timestamp = |
| wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_); |
| recv_times_[timestamp] = |
| Clock::GetRealTimeClock()->CurrentNtpInMilliseconds(); |
| } |
| |
| return receiver_->DeliverPacket(media_type, std::move(packet), |
| packet_time_us); |
| } |
| |
| void VideoAnalyzer::PreEncodeOnFrame(const VideoFrame& video_frame) { |
| rtc::CritScope lock(&crit_); |
| if (!first_encoded_timestamp_) { |
| while (frames_.front().timestamp() != video_frame.timestamp()) { |
| ++dropped_frames_before_first_encode_; |
| frames_.pop_front(); |
| RTC_CHECK(!frames_.empty()); |
| } |
| first_encoded_timestamp_ = video_frame.timestamp(); |
| } |
| } |
| |
| void VideoAnalyzer::PostEncodeOnFrame(size_t stream_id, uint32_t timestamp) { |
| rtc::CritScope lock(&crit_); |
| if (!first_sent_timestamp_ && stream_id == selected_stream_) { |
| first_sent_timestamp_ = timestamp; |
| } |
| } |
| |
| bool VideoAnalyzer::SendRtp(const uint8_t* packet, |
| size_t length, |
| const PacketOptions& options) { |
| RtpUtility::RtpHeaderParser parser(packet, length); |
| RTPHeader header; |
| parser.Parse(&header); |
| |
| int64_t current_time = Clock::GetRealTimeClock()->CurrentNtpInMilliseconds(); |
| |
| bool result = transport_->SendRtp(packet, length, options); |
| { |
| rtc::CritScope lock(&crit_); |
| if (rtp_timestamp_delta_ == 0 && header.ssrc == ssrc_to_analyze_) { |
| RTC_CHECK(static_cast<bool>(first_sent_timestamp_)); |
| rtp_timestamp_delta_ = header.timestamp - *first_sent_timestamp_; |
| } |
| |
| if (!IsFlexfec(header.payloadType) && header.ssrc == ssrc_to_analyze_) { |
| // Ignore FlexFEC timestamps, to avoid collisions with media timestamps. |
| // (FlexFEC and media are sent on different SSRCs, which have different |
| // timestamps spaces.) |
| // Also ignore packets from wrong SSRC and retransmits. |
| int64_t timestamp = |
| wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_); |
| send_times_[timestamp] = current_time; |
| |
| if (IsInSelectedSpatialAndTemporalLayer(packet, length, header)) { |
| encoded_frame_sizes_[timestamp] += |
| length - (header.headerLength + header.paddingLength); |
| total_media_bytes_ += |
| length - (header.headerLength + header.paddingLength); |
| } |
| if (first_sending_time_ == 0) |
| first_sending_time_ = current_time; |
| last_sending_time_ = current_time; |
| } |
| } |
| return result; |
| } |
| |
| bool VideoAnalyzer::SendRtcp(const uint8_t* packet, size_t length) { |
| return transport_->SendRtcp(packet, length); |
| } |
| |
| void VideoAnalyzer::OnFrame(const VideoFrame& video_frame) { |
| int64_t render_time_ms = |
| Clock::GetRealTimeClock()->CurrentNtpInMilliseconds(); |
| |
| rtc::CritScope lock(&crit_); |
| |
| StartExcludingCpuThreadTime(); |
| |
| int64_t send_timestamp = |
| wrap_handler_.Unwrap(video_frame.timestamp() - rtp_timestamp_delta_); |
| |
| while (wrap_handler_.Unwrap(frames_.front().timestamp()) < send_timestamp) { |
| if (!last_rendered_frame_) { |
| // No previous frame rendered, this one was dropped after sending but |
| // before rendering. |
| ++dropped_frames_before_rendering_; |
| } else { |
| AddFrameComparison(frames_.front(), *last_rendered_frame_, true, |
| render_time_ms); |
| } |
| frames_.pop_front(); |
| RTC_DCHECK(!frames_.empty()); |
| } |
| |
| VideoFrame reference_frame = frames_.front(); |
| frames_.pop_front(); |
| int64_t reference_timestamp = |
| wrap_handler_.Unwrap(reference_frame.timestamp()); |
| if (send_timestamp == reference_timestamp - 1) { |
| // TODO(ivica): Make this work for > 2 streams. |
| // Look at RTPSender::BuildRTPHeader. |
| ++send_timestamp; |
| } |
| ASSERT_EQ(reference_timestamp, send_timestamp); |
| |
| AddFrameComparison(reference_frame, video_frame, false, render_time_ms); |
| |
| last_rendered_frame_ = video_frame; |
| |
| StopExcludingCpuThreadTime(); |
| } |
| |
| void VideoAnalyzer::Wait() { |
| // Frame comparisons can be very expensive. Wait for test to be done, but |
| // at time-out check if frames_processed is going up. If so, give it more |
| // time, otherwise fail. Hopefully this will reduce test flakiness. |
| |
| stats_polling_thread_.Start(); |
| |
| int last_frames_processed = -1; |
| int iteration = 0; |
| while (!done_.Wait(test::CallTest::kDefaultTimeoutMs)) { |
| int frames_processed; |
| { |
| rtc::CritScope crit(&comparison_lock_); |
| frames_processed = frames_processed_; |
| } |
| |
| // Print some output so test infrastructure won't think we've crashed. |
| const char* kKeepAliveMessages[3] = { |
| "Uh, I'm-I'm not quite dead, sir.", |
| "Uh, I-I think uh, I could pull through, sir.", |
| "Actually, I think I'm all right to come with you--"}; |
| printf("- %s\n", kKeepAliveMessages[iteration++ % 3]); |
| |
| if (last_frames_processed == -1) { |
| last_frames_processed = frames_processed; |
| continue; |
| } |
| if (frames_processed == last_frames_processed) { |
| EXPECT_GT(frames_processed, last_frames_processed) |
| << "Analyzer stalled while waiting for test to finish."; |
| done_.Set(); |
| break; |
| } |
| last_frames_processed = frames_processed; |
| } |
| |
| if (iteration > 0) |
| printf("- Farewell, sweet Concorde!\n"); |
| |
| stats_polling_thread_.Stop(); |
| } |
| |
| void VideoAnalyzer::StartMeasuringCpuProcessTime() { |
| rtc::CritScope lock(&cpu_measurement_lock_); |
| cpu_time_ -= rtc::GetProcessCpuTimeNanos(); |
| wallclock_time_ -= rtc::SystemTimeNanos(); |
| } |
| |
| void VideoAnalyzer::StopMeasuringCpuProcessTime() { |
| rtc::CritScope lock(&cpu_measurement_lock_); |
| cpu_time_ += rtc::GetProcessCpuTimeNanos(); |
| wallclock_time_ += rtc::SystemTimeNanos(); |
| } |
| |
| void VideoAnalyzer::StartExcludingCpuThreadTime() { |
| rtc::CritScope lock(&cpu_measurement_lock_); |
| cpu_time_ += rtc::GetThreadCpuTimeNanos(); |
| } |
| |
| void VideoAnalyzer::StopExcludingCpuThreadTime() { |
| rtc::CritScope lock(&cpu_measurement_lock_); |
| cpu_time_ -= rtc::GetThreadCpuTimeNanos(); |
| } |
| |
| double VideoAnalyzer::GetCpuUsagePercent() { |
| rtc::CritScope lock(&cpu_measurement_lock_); |
| return static_cast<double>(cpu_time_) / wallclock_time_ * 100.0; |
| } |
| |
| bool VideoAnalyzer::IsInSelectedSpatialAndTemporalLayer( |
| const uint8_t* packet, |
| size_t length, |
| const RTPHeader& header) { |
| if (header.payloadType != test::CallTest::kPayloadTypeVP9 && |
| header.payloadType != test::CallTest::kPayloadTypeVP8) { |
| return true; |
| } else { |
| // Get VP8 and VP9 specific header to check layers indexes. |
| const uint8_t* payload = packet + header.headerLength; |
| const size_t payload_length = length - header.headerLength; |
| const size_t payload_data_length = payload_length - header.paddingLength; |
| const bool is_vp8 = header.payloadType == test::CallTest::kPayloadTypeVP8; |
| std::unique_ptr<RtpDepacketizer> depacketizer( |
| RtpDepacketizer::Create(is_vp8 ? kVideoCodecVP8 : kVideoCodecVP9)); |
| RtpDepacketizer::ParsedPayload parsed_payload; |
| bool result = |
| depacketizer->Parse(&parsed_payload, payload, payload_data_length); |
| RTC_DCHECK(result); |
| |
| int temporal_idx; |
| int spatial_idx; |
| if (is_vp8) { |
| temporal_idx = absl::get<RTPVideoHeaderVP8>( |
| parsed_payload.video_header().video_type_header) |
| .temporalIdx; |
| spatial_idx = kNoTemporalIdx; |
| } else { |
| const auto& vp9_header = absl::get<RTPVideoHeaderVP9>( |
| parsed_payload.video_header().video_type_header); |
| temporal_idx = vp9_header.temporal_idx; |
| spatial_idx = vp9_header.spatial_idx; |
| } |
| |
| return (selected_tl_ < 0 || temporal_idx == kNoTemporalIdx || |
| temporal_idx <= selected_tl_) && |
| (selected_sl_ < 0 || spatial_idx == kNoSpatialIdx || |
| spatial_idx <= selected_sl_); |
| } |
| } |
| |
| void VideoAnalyzer::PollStatsThread(void* obj) { |
| static_cast<VideoAnalyzer*>(obj)->PollStats(); |
| } |
| |
| void VideoAnalyzer::PollStats() { |
| while (!done_.Wait(kSendStatsPollingIntervalMs)) { |
| rtc::CritScope crit(&comparison_lock_); |
| |
| Call::Stats call_stats = call_->GetStats(); |
| send_bandwidth_bps_.AddSample(call_stats.send_bandwidth_bps); |
| |
| VideoSendStream::Stats send_stats = send_stream_->GetStats(); |
| // It's not certain that we yet have estimates for any of these stats. |
| // Check that they are positive before mixing them in. |
| if (send_stats.encode_frame_rate > 0) |
| encode_frame_rate_.AddSample(send_stats.encode_frame_rate); |
| if (send_stats.avg_encode_time_ms > 0) |
| encode_time_ms_.AddSample(send_stats.avg_encode_time_ms); |
| if (send_stats.encode_usage_percent > 0) |
| encode_usage_percent_.AddSample(send_stats.encode_usage_percent); |
| if (send_stats.media_bitrate_bps > 0) |
| media_bitrate_bps_.AddSample(send_stats.media_bitrate_bps); |
| size_t fec_bytes = 0; |
| for (auto kv : send_stats.substreams) { |
| fec_bytes += kv.second.rtp_stats.fec.payload_bytes + |
| kv.second.rtp_stats.fec.padding_bytes; |
| } |
| fec_bitrate_bps_.AddSample((fec_bytes - last_fec_bytes_) * 8); |
| last_fec_bytes_ = fec_bytes; |
| |
| if (receive_stream_ != nullptr) { |
| VideoReceiveStream::Stats receive_stats = receive_stream_->GetStats(); |
| if (receive_stats.decode_ms > 0) |
| decode_time_ms_.AddSample(receive_stats.decode_ms); |
| if (receive_stats.max_decode_ms > 0) |
| decode_time_max_ms_.AddSample(receive_stats.max_decode_ms); |
| } |
| |
| if (audio_receive_stream_ != nullptr) { |
| AudioReceiveStream::Stats receive_stats = |
| audio_receive_stream_->GetStats(); |
| audio_expand_rate_.AddSample(receive_stats.expand_rate); |
| audio_accelerate_rate_.AddSample(receive_stats.accelerate_rate); |
| audio_jitter_buffer_ms_.AddSample(receive_stats.jitter_buffer_ms); |
| } |
| |
| memory_usage_.AddSample(rtc::GetProcessResidentSizeBytes()); |
| } |
| } |
| |
| bool VideoAnalyzer::FrameComparisonThread(void* obj) { |
| return static_cast<VideoAnalyzer*>(obj)->CompareFrames(); |
| } |
| |
| bool VideoAnalyzer::CompareFrames() { |
| if (AllFramesRecorded()) |
| return false; |
| |
| FrameComparison comparison; |
| |
| if (!PopComparison(&comparison)) { |
| // Wait until new comparison task is available, or test is done. |
| // If done, wake up remaining threads waiting. |
| comparison_available_event_.Wait(1000); |
| if (AllFramesRecorded()) { |
| comparison_available_event_.Set(); |
| return false; |
| } |
| return true; // Try again. |
| } |
| |
| StartExcludingCpuThreadTime(); |
| |
| PerformFrameComparison(comparison); |
| |
| StopExcludingCpuThreadTime(); |
| |
| if (FrameProcessed()) { |
| PrintResults(); |
| if (graph_data_output_file_) |
| PrintSamplesToFile(); |
| done_.Set(); |
| comparison_available_event_.Set(); |
| return false; |
| } |
| |
| return true; |
| } |
| |
| bool VideoAnalyzer::PopComparison(VideoAnalyzer::FrameComparison* comparison) { |
| rtc::CritScope crit(&comparison_lock_); |
| // If AllFramesRecorded() is true, it means we have already popped |
| // frames_to_process_ frames from comparisons_, so there is no more work |
| // for this thread to be done. frames_processed_ might still be lower if |
| // all comparisons are not done, but those frames are currently being |
| // worked on by other threads. |
| if (comparisons_.empty() || AllFramesRecorded()) |
| return false; |
| |
| *comparison = comparisons_.front(); |
| comparisons_.pop_front(); |
| |
| FrameRecorded(); |
| return true; |
| } |
| |
| void VideoAnalyzer::FrameRecorded() { |
| rtc::CritScope crit(&comparison_lock_); |
| ++frames_recorded_; |
| } |
| |
| bool VideoAnalyzer::AllFramesRecorded() { |
| rtc::CritScope crit(&comparison_lock_); |
| assert(frames_recorded_ <= frames_to_process_); |
| return frames_recorded_ == frames_to_process_; |
| } |
| |
| bool VideoAnalyzer::FrameProcessed() { |
| rtc::CritScope crit(&comparison_lock_); |
| ++frames_processed_; |
| assert(frames_processed_ <= frames_to_process_); |
| return frames_processed_ == frames_to_process_; |
| } |
| |
| void VideoAnalyzer::PrintResults() { |
| StopMeasuringCpuProcessTime(); |
| rtc::CritScope crit(&comparison_lock_); |
| // Record the time from the last freeze until the last rendered frame to |
| // ensure we cover the full timespan of the session. Otherwise the metric |
| // would penalize an early freeze followed by no freezes until the end. |
| time_between_freezes_.AddSample(last_render_time_ - last_unfreeze_time_ms_); |
| PrintResult("psnr", psnr_, " dB"); |
| PrintResult("ssim", ssim_, " score"); |
| PrintResult("sender_time", sender_time_, " ms"); |
| PrintResult("receiver_time", receiver_time_, " ms"); |
| PrintResult("network_time", network_time_, " ms"); |
| PrintResult("total_delay_incl_network", end_to_end_, " ms"); |
| PrintResult("time_between_rendered_frames", rendered_delta_, " ms"); |
| PrintResult("encode_frame_rate", encode_frame_rate_, " fps"); |
| PrintResult("encode_time", encode_time_ms_, " ms"); |
| PrintResult("media_bitrate", media_bitrate_bps_, " bps"); |
| PrintResult("fec_bitrate", fec_bitrate_bps_, " bps"); |
| PrintResult("send_bandwidth", send_bandwidth_bps_, " bps"); |
| PrintResult("time_between_freezes", time_between_freezes_, " ms"); |
| |
| if (worst_frame_) { |
| test::PrintResult("min_psnr", "", test_label_.c_str(), worst_frame_->psnr, |
| "dB", false); |
| } |
| |
| if (receive_stream_ != nullptr) { |
| PrintResult("decode_time", decode_time_ms_, " ms"); |
| } |
| |
| test::PrintResult("dropped_frames", "", test_label_.c_str(), dropped_frames_, |
| "frames", false); |
| test::PrintResult("cpu_usage", "", test_label_.c_str(), GetCpuUsagePercent(), |
| "%", false); |
| |
| #if defined(WEBRTC_WIN) |
| // On Linux and Mac in Resident Set some unused pages may be counted. |
| // Therefore this metric will depend on order in which tests are run and |
| // will be flaky. |
| PrintResult("memory_usage", memory_usage_, " bytes"); |
| #endif |
| |
| // Saving only the worst frame for manual analysis. Intention here is to |
| // only detect video corruptions and not to track picture quality. Thus, |
| // jpeg is used here. |
| if (FLAG_save_worst_frame && worst_frame_) { |
| std::string output_dir; |
| test::GetTestArtifactsDir(&output_dir); |
| std::string output_path = |
| test::JoinFilename(output_dir, test_label_ + ".jpg"); |
| RTC_LOG(LS_INFO) << "Saving worst frame to " << output_path; |
| test::JpegFrameWriter frame_writer(output_path); |
| RTC_CHECK( |
| frame_writer.WriteFrame(worst_frame_->frame, 100 /*best quality*/)); |
| } |
| |
| if (audio_receive_stream_ != nullptr) { |
| PrintResult("audio_expand_rate", audio_expand_rate_, ""); |
| PrintResult("audio_accelerate_rate", audio_accelerate_rate_, ""); |
| PrintResult("audio_jitter_buffer", audio_jitter_buffer_ms_, " ms"); |
| } |
| |
| // Disable quality check for quick test, as quality checks may fail |
| // because too few samples were collected. |
| if (!is_quick_test_enabled_) { |
| EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_); |
| EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_); |
| } |
| } |
| |
| void VideoAnalyzer::PerformFrameComparison( |
| const VideoAnalyzer::FrameComparison& comparison) { |
| // Perform expensive psnr and ssim calculations while not holding lock. |
| double psnr = -1.0; |
| double ssim = -1.0; |
| if (comparison.reference && !comparison.dropped) { |
| psnr = I420PSNR(&*comparison.reference, &*comparison.render); |
| ssim = I420SSIM(&*comparison.reference, &*comparison.render); |
| } |
| |
| rtc::CritScope crit(&comparison_lock_); |
| |
| if (psnr >= 0.0 && (!worst_frame_ || worst_frame_->psnr > psnr)) { |
| worst_frame_.emplace(FrameWithPsnr{psnr, *comparison.render}); |
| } |
| |
| if (graph_data_output_file_) { |
| samples_.push_back(Sample(comparison.dropped, comparison.input_time_ms, |
| comparison.send_time_ms, comparison.recv_time_ms, |
| comparison.render_time_ms, |
| comparison.encoded_frame_size, psnr, ssim)); |
| } |
| if (psnr >= 0.0) |
| psnr_.AddSample(psnr); |
| if (ssim >= 0.0) |
| ssim_.AddSample(ssim); |
| |
| if (comparison.dropped) { |
| ++dropped_frames_; |
| return; |
| } |
| if (last_unfreeze_time_ms_ == 0) |
| last_unfreeze_time_ms_ = comparison.render_time_ms; |
| if (last_render_time_ != 0) { |
| const int64_t render_delta_ms = |
| comparison.render_time_ms - last_render_time_; |
| rendered_delta_.AddSample(render_delta_ms); |
| if (last_render_delta_ms_ != 0 && |
| render_delta_ms - last_render_delta_ms_ > 150) { |
| time_between_freezes_.AddSample(last_render_time_ - |
| last_unfreeze_time_ms_); |
| last_unfreeze_time_ms_ = comparison.render_time_ms; |
| } |
| last_render_delta_ms_ = render_delta_ms; |
| } |
| last_render_time_ = comparison.render_time_ms; |
| |
| sender_time_.AddSample(comparison.send_time_ms - comparison.input_time_ms); |
| if (comparison.recv_time_ms > 0) { |
| // If recv_time_ms == 0, this frame consisted of a packets which were all |
| // lost in the transport. Since we were able to render the frame, however, |
| // the dropped packets were recovered by FlexFEC. The FlexFEC recovery |
| // happens internally in Call, and we can therefore here not know which |
| // FEC packets that protected the lost media packets. Consequently, we |
| // were not able to record a meaningful recv_time_ms. We therefore skip |
| // this sample. |
| // |
| // The reasoning above does not hold for ULPFEC and RTX, as for those |
| // strategies the timestamp of the received packets is set to the |
| // timestamp of the protected/retransmitted media packet. I.e., then |
| // recv_time_ms != 0, even though the media packets were lost. |
| receiver_time_.AddSample(comparison.render_time_ms - |
| comparison.recv_time_ms); |
| network_time_.AddSample(comparison.recv_time_ms - comparison.send_time_ms); |
| } |
| end_to_end_.AddSample(comparison.render_time_ms - comparison.input_time_ms); |
| encoded_frame_size_.AddSample(comparison.encoded_frame_size); |
| } |
| |
| void VideoAnalyzer::PrintResult(const char* result_type, |
| test::Statistics stats, |
| const char* unit) { |
| test::PrintResultMeanAndError(result_type, "", test_label_.c_str(), |
| stats.Mean(), stats.StandardDeviation(), unit, |
| false); |
| } |
| |
| void VideoAnalyzer::PrintSamplesToFile() { |
| FILE* out = graph_data_output_file_; |
| rtc::CritScope crit(&comparison_lock_); |
| std::sort(samples_.begin(), samples_.end(), |
| [](const Sample& A, const Sample& B) -> bool { |
| return A.input_time_ms < B.input_time_ms; |
| }); |
| |
| fprintf(out, "%s\n", graph_title_.c_str()); |
| fprintf(out, "%" PRIuS "\n", samples_.size()); |
| fprintf(out, |
| "dropped " |
| "input_time_ms " |
| "send_time_ms " |
| "recv_time_ms " |
| "render_time_ms " |
| "encoded_frame_size " |
| "psnr " |
| "ssim " |
| "encode_time_ms\n"); |
| for (const Sample& sample : samples_) { |
| fprintf(out, |
| "%d %" PRId64 " %" PRId64 " %" PRId64 " %" PRId64 " %" PRIuS |
| " %lf %lf\n", |
| sample.dropped, sample.input_time_ms, sample.send_time_ms, |
| sample.recv_time_ms, sample.render_time_ms, |
| sample.encoded_frame_size, sample.psnr, sample.ssim); |
| } |
| } |
| |
| double VideoAnalyzer::GetAverageMediaBitrateBps() { |
| if (last_sending_time_ == first_sending_time_) { |
| return 0; |
| } else { |
| return static_cast<double>(total_media_bytes_) * 8 / |
| (last_sending_time_ - first_sending_time_) * |
| rtc::kNumMillisecsPerSec; |
| } |
| } |
| |
| void VideoAnalyzer::AddCapturedFrameForComparison( |
| const VideoFrame& video_frame) { |
| rtc::CritScope lock(&crit_); |
| frames_.push_back(video_frame); |
| } |
| |
| void VideoAnalyzer::AddFrameComparison(const VideoFrame& reference, |
| const VideoFrame& render, |
| bool dropped, |
| int64_t render_time_ms) { |
| int64_t reference_timestamp = wrap_handler_.Unwrap(reference.timestamp()); |
| int64_t send_time_ms = send_times_[reference_timestamp]; |
| send_times_.erase(reference_timestamp); |
| int64_t recv_time_ms = recv_times_[reference_timestamp]; |
| recv_times_.erase(reference_timestamp); |
| |
| // TODO(ivica): Make this work for > 2 streams. |
| auto it = encoded_frame_sizes_.find(reference_timestamp); |
| if (it == encoded_frame_sizes_.end()) |
| it = encoded_frame_sizes_.find(reference_timestamp - 1); |
| size_t encoded_size = it == encoded_frame_sizes_.end() ? 0 : it->second; |
| if (it != encoded_frame_sizes_.end()) |
| encoded_frame_sizes_.erase(it); |
| |
| rtc::CritScope crit(&comparison_lock_); |
| if (comparisons_.size() < kMaxComparisons) { |
| comparisons_.push_back(FrameComparison( |
| reference, render, dropped, reference.ntp_time_ms(), send_time_ms, |
| recv_time_ms, render_time_ms, encoded_size)); |
| } else { |
| comparisons_.push_back(FrameComparison(dropped, reference.ntp_time_ms(), |
| send_time_ms, recv_time_ms, |
| render_time_ms, encoded_size)); |
| } |
| comparison_available_event_.Set(); |
| } |
| |
| VideoAnalyzer::FrameComparison::FrameComparison() |
| : dropped(false), |
| input_time_ms(0), |
| send_time_ms(0), |
| recv_time_ms(0), |
| render_time_ms(0), |
| encoded_frame_size(0) {} |
| |
| VideoAnalyzer::FrameComparison::FrameComparison(const VideoFrame& reference, |
| const VideoFrame& render, |
| bool dropped, |
| int64_t input_time_ms, |
| int64_t send_time_ms, |
| int64_t recv_time_ms, |
| int64_t render_time_ms, |
| size_t encoded_frame_size) |
| : reference(reference), |
| render(render), |
| dropped(dropped), |
| input_time_ms(input_time_ms), |
| send_time_ms(send_time_ms), |
| recv_time_ms(recv_time_ms), |
| render_time_ms(render_time_ms), |
| encoded_frame_size(encoded_frame_size) {} |
| |
| VideoAnalyzer::FrameComparison::FrameComparison(bool dropped, |
| int64_t input_time_ms, |
| int64_t send_time_ms, |
| int64_t recv_time_ms, |
| int64_t render_time_ms, |
| size_t encoded_frame_size) |
| : dropped(dropped), |
| input_time_ms(input_time_ms), |
| send_time_ms(send_time_ms), |
| recv_time_ms(recv_time_ms), |
| render_time_ms(render_time_ms), |
| encoded_frame_size(encoded_frame_size) {} |
| |
| VideoAnalyzer::Sample::Sample(int dropped, |
| int64_t input_time_ms, |
| int64_t send_time_ms, |
| int64_t recv_time_ms, |
| int64_t render_time_ms, |
| size_t encoded_frame_size, |
| double psnr, |
| double ssim) |
| : dropped(dropped), |
| input_time_ms(input_time_ms), |
| send_time_ms(send_time_ms), |
| recv_time_ms(recv_time_ms), |
| render_time_ms(render_time_ms), |
| encoded_frame_size(encoded_frame_size), |
| psnr(psnr), |
| ssim(ssim) {} |
| |
| VideoAnalyzer::CapturedFrameForwarder::CapturedFrameForwarder( |
| VideoAnalyzer* analyzer, |
| Clock* clock) |
| : analyzer_(analyzer), |
| send_stream_input_(nullptr), |
| video_source_(nullptr), |
| clock_(clock) {} |
| |
| void VideoAnalyzer::CapturedFrameForwarder::SetSource( |
| VideoSourceInterface<VideoFrame>* video_source) { |
| video_source_ = video_source; |
| } |
| |
| void VideoAnalyzer::CapturedFrameForwarder::OnFrame( |
| const VideoFrame& video_frame) { |
| VideoFrame copy = video_frame; |
| // Frames from the capturer does not have a rtp timestamp. |
| // Create one so it can be used for comparison. |
| RTC_DCHECK_EQ(0, video_frame.timestamp()); |
| if (video_frame.ntp_time_ms() == 0) |
| copy.set_ntp_time_ms(clock_->CurrentNtpInMilliseconds()); |
| copy.set_timestamp(copy.ntp_time_ms() * 90); |
| analyzer_->AddCapturedFrameForComparison(copy); |
| rtc::CritScope lock(&crit_); |
| if (send_stream_input_) |
| send_stream_input_->OnFrame(copy); |
| } |
| |
| void VideoAnalyzer::CapturedFrameForwarder::AddOrUpdateSink( |
| rtc::VideoSinkInterface<VideoFrame>* sink, |
| const rtc::VideoSinkWants& wants) { |
| { |
| rtc::CritScope lock(&crit_); |
| RTC_DCHECK(!send_stream_input_ || send_stream_input_ == sink); |
| send_stream_input_ = sink; |
| } |
| if (video_source_) { |
| video_source_->AddOrUpdateSink(this, wants); |
| } |
| } |
| |
| void VideoAnalyzer::CapturedFrameForwarder::RemoveSink( |
| rtc::VideoSinkInterface<VideoFrame>* sink) { |
| rtc::CritScope lock(&crit_); |
| RTC_DCHECK(sink == send_stream_input_); |
| send_stream_input_ = nullptr; |
| } |
| |
| } // namespace webrtc |
