1036 lines
38 KiB
C++
1036 lines
38 KiB
C++
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "video/video_analyzer.h"
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#include <algorithm>
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#include <utility>
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#include "absl/algorithm/container.h"
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#include "absl/flags/flag.h"
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#include "absl/flags/parse.h"
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#include "common_video/libyuv/include/webrtc_libyuv.h"
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#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
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#include "modules/rtp_rtcp/source/rtp_packet.h"
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#include "rtc_base/cpu_time.h"
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#include "rtc_base/format_macros.h"
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#include "rtc_base/memory_usage.h"
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#include "rtc_base/task_queue_for_test.h"
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#include "rtc_base/task_utils/repeating_task.h"
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#include "rtc_base/time_utils.h"
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#include "system_wrappers/include/cpu_info.h"
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#include "test/call_test.h"
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#include "test/testsupport/file_utils.h"
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#include "test/testsupport/frame_writer.h"
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#include "test/testsupport/perf_test.h"
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#include "test/testsupport/test_artifacts.h"
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ABSL_FLAG(bool,
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save_worst_frame,
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false,
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"Enable saving a frame with the lowest PSNR to a jpeg file in the "
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"test_artifacts_dir");
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namespace webrtc {
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namespace {
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constexpr TimeDelta kSendStatsPollingInterval = TimeDelta::Seconds(1);
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constexpr size_t kMaxComparisons = 10;
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// How often is keep alive message printed.
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constexpr int kKeepAliveIntervalSeconds = 30;
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// Interval between checking that the test is over.
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constexpr int kProbingIntervalMs = 500;
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constexpr int kKeepAliveIntervalIterations =
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kKeepAliveIntervalSeconds * 1000 / kProbingIntervalMs;
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bool IsFlexfec(int payload_type) {
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return payload_type == test::CallTest::kFlexfecPayloadType;
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}
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} // namespace
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VideoAnalyzer::VideoAnalyzer(test::LayerFilteringTransport* transport,
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const std::string& test_label,
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double avg_psnr_threshold,
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double avg_ssim_threshold,
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int duration_frames,
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TimeDelta test_duration,
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FILE* graph_data_output_file,
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const std::string& graph_title,
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uint32_t ssrc_to_analyze,
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uint32_t rtx_ssrc_to_analyze,
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size_t selected_stream,
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int selected_sl,
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int selected_tl,
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bool is_quick_test_enabled,
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Clock* clock,
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std::string rtp_dump_name,
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TaskQueueBase* task_queue)
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: transport_(transport),
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receiver_(nullptr),
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call_(nullptr),
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send_stream_(nullptr),
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receive_stream_(nullptr),
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audio_receive_stream_(nullptr),
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captured_frame_forwarder_(this, clock, duration_frames, test_duration),
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test_label_(test_label),
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graph_data_output_file_(graph_data_output_file),
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graph_title_(graph_title),
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ssrc_to_analyze_(ssrc_to_analyze),
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rtx_ssrc_to_analyze_(rtx_ssrc_to_analyze),
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selected_stream_(selected_stream),
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selected_sl_(selected_sl),
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selected_tl_(selected_tl),
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mean_decode_time_ms_(0.0),
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freeze_count_(0),
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total_freezes_duration_ms_(0),
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total_frames_duration_ms_(0),
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sum_squared_frame_durations_(0),
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decode_frame_rate_(0),
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render_frame_rate_(0),
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last_fec_bytes_(0),
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frames_to_process_(duration_frames),
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test_end_(clock->CurrentTime() + test_duration),
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frames_recorded_(0),
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frames_processed_(0),
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captured_frames_(0),
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dropped_frames_(0),
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dropped_frames_before_first_encode_(0),
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dropped_frames_before_rendering_(0),
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last_render_time_(0),
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last_render_delta_ms_(0),
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last_unfreeze_time_ms_(0),
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rtp_timestamp_delta_(0),
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cpu_time_(0),
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wallclock_time_(0),
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avg_psnr_threshold_(avg_psnr_threshold),
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avg_ssim_threshold_(avg_ssim_threshold),
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is_quick_test_enabled_(is_quick_test_enabled),
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quit_(false),
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done_(true, false),
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vp8_depacketizer_(CreateVideoRtpDepacketizer(kVideoCodecVP8)),
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vp9_depacketizer_(CreateVideoRtpDepacketizer(kVideoCodecVP9)),
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clock_(clock),
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start_ms_(clock->TimeInMilliseconds()),
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task_queue_(task_queue) {
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// Create thread pool for CPU-expensive PSNR/SSIM calculations.
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// Try to use about as many threads as cores, but leave kMinCoresLeft alone,
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// so that we don't accidentally starve "real" worker threads (codec etc).
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// Also, don't allocate more than kMaxComparisonThreads, even if there are
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// spare cores.
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uint32_t num_cores = CpuInfo::DetectNumberOfCores();
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RTC_DCHECK_GE(num_cores, 1);
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static const uint32_t kMinCoresLeft = 4;
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static const uint32_t kMaxComparisonThreads = 8;
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if (num_cores <= kMinCoresLeft) {
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num_cores = 1;
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} else {
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num_cores -= kMinCoresLeft;
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num_cores = std::min(num_cores, kMaxComparisonThreads);
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}
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for (uint32_t i = 0; i < num_cores; ++i) {
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comparison_thread_pool_.push_back(rtc::PlatformThread::SpawnJoinable(
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[this] {
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while (CompareFrames()) {
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}
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},
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"Analyzer"));
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}
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if (!rtp_dump_name.empty()) {
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fprintf(stdout, "Writing rtp dump to %s\n", rtp_dump_name.c_str());
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rtp_file_writer_.reset(test::RtpFileWriter::Create(
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test::RtpFileWriter::kRtpDump, rtp_dump_name));
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}
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}
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VideoAnalyzer::~VideoAnalyzer() {
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{
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MutexLock lock(&comparison_lock_);
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quit_ = true;
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}
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// Joins all threads.
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comparison_thread_pool_.clear();
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}
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void VideoAnalyzer::SetReceiver(PacketReceiver* receiver) {
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receiver_ = receiver;
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}
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void VideoAnalyzer::SetSource(
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rtc::VideoSourceInterface<VideoFrame>* video_source,
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bool respect_sink_wants) {
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if (respect_sink_wants)
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captured_frame_forwarder_.SetSource(video_source);
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rtc::VideoSinkWants wants;
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video_source->AddOrUpdateSink(InputInterface(), wants);
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}
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void VideoAnalyzer::SetCall(Call* call) {
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MutexLock lock(&lock_);
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RTC_DCHECK(!call_);
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call_ = call;
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}
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void VideoAnalyzer::SetSendStream(VideoSendStream* stream) {
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MutexLock lock(&lock_);
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RTC_DCHECK(!send_stream_);
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send_stream_ = stream;
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}
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void VideoAnalyzer::SetReceiveStream(VideoReceiveStream* stream) {
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MutexLock lock(&lock_);
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RTC_DCHECK(!receive_stream_);
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receive_stream_ = stream;
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}
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void VideoAnalyzer::SetAudioReceiveStream(AudioReceiveStream* recv_stream) {
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MutexLock lock(&lock_);
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RTC_CHECK(!audio_receive_stream_);
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audio_receive_stream_ = recv_stream;
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}
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rtc::VideoSinkInterface<VideoFrame>* VideoAnalyzer::InputInterface() {
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return &captured_frame_forwarder_;
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}
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rtc::VideoSourceInterface<VideoFrame>* VideoAnalyzer::OutputInterface() {
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return &captured_frame_forwarder_;
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}
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PacketReceiver::DeliveryStatus VideoAnalyzer::DeliverPacket(
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MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) {
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// Ignore timestamps of RTCP packets. They're not synchronized with
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// RTP packet timestamps and so they would confuse wrap_handler_.
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if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) {
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return receiver_->DeliverPacket(media_type, std::move(packet),
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packet_time_us);
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}
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if (rtp_file_writer_) {
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test::RtpPacket p;
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memcpy(p.data, packet.cdata(), packet.size());
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p.length = packet.size();
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p.original_length = packet.size();
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p.time_ms = clock_->TimeInMilliseconds() - start_ms_;
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rtp_file_writer_->WritePacket(&p);
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}
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RtpPacket rtp_packet;
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rtp_packet.Parse(packet);
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if (!IsFlexfec(rtp_packet.PayloadType()) &&
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(rtp_packet.Ssrc() == ssrc_to_analyze_ ||
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rtp_packet.Ssrc() == rtx_ssrc_to_analyze_)) {
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// Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
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// (FlexFEC and media are sent on different SSRCs, which have different
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// timestamps spaces.)
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// Also ignore packets from wrong SSRC, but include retransmits.
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MutexLock lock(&lock_);
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int64_t timestamp =
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wrap_handler_.Unwrap(rtp_packet.Timestamp() - rtp_timestamp_delta_);
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recv_times_[timestamp] = clock_->CurrentNtpInMilliseconds();
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}
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return receiver_->DeliverPacket(media_type, std::move(packet),
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packet_time_us);
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}
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void VideoAnalyzer::PreEncodeOnFrame(const VideoFrame& video_frame) {
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MutexLock lock(&lock_);
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if (!first_encoded_timestamp_) {
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while (frames_.front().timestamp() != video_frame.timestamp()) {
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++dropped_frames_before_first_encode_;
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frames_.pop_front();
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RTC_CHECK(!frames_.empty());
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}
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first_encoded_timestamp_ = video_frame.timestamp();
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}
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}
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void VideoAnalyzer::PostEncodeOnFrame(size_t stream_id, uint32_t timestamp) {
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MutexLock lock(&lock_);
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if (!first_sent_timestamp_ && stream_id == selected_stream_) {
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first_sent_timestamp_ = timestamp;
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}
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}
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bool VideoAnalyzer::SendRtp(const uint8_t* packet,
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size_t length,
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const PacketOptions& options) {
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RtpPacket rtp_packet;
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rtp_packet.Parse(packet, length);
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int64_t current_time = clock_->CurrentNtpInMilliseconds();
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bool result = transport_->SendRtp(packet, length, options);
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{
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MutexLock lock(&lock_);
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if (rtp_timestamp_delta_ == 0 && rtp_packet.Ssrc() == ssrc_to_analyze_) {
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RTC_CHECK(static_cast<bool>(first_sent_timestamp_));
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rtp_timestamp_delta_ = rtp_packet.Timestamp() - *first_sent_timestamp_;
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}
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if (!IsFlexfec(rtp_packet.PayloadType()) &&
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rtp_packet.Ssrc() == ssrc_to_analyze_) {
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// Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
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// (FlexFEC and media are sent on different SSRCs, which have different
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// timestamps spaces.)
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// Also ignore packets from wrong SSRC and retransmits.
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int64_t timestamp =
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wrap_handler_.Unwrap(rtp_packet.Timestamp() - rtp_timestamp_delta_);
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send_times_[timestamp] = current_time;
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if (IsInSelectedSpatialAndTemporalLayer(rtp_packet)) {
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encoded_frame_sizes_[timestamp] += rtp_packet.payload_size();
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}
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}
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}
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return result;
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}
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bool VideoAnalyzer::SendRtcp(const uint8_t* packet, size_t length) {
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return transport_->SendRtcp(packet, length);
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}
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void VideoAnalyzer::OnFrame(const VideoFrame& video_frame) {
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int64_t render_time_ms = clock_->CurrentNtpInMilliseconds();
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MutexLock lock(&lock_);
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StartExcludingCpuThreadTime();
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int64_t send_timestamp =
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wrap_handler_.Unwrap(video_frame.timestamp() - rtp_timestamp_delta_);
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while (wrap_handler_.Unwrap(frames_.front().timestamp()) < send_timestamp) {
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if (!last_rendered_frame_) {
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// No previous frame rendered, this one was dropped after sending but
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// before rendering.
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++dropped_frames_before_rendering_;
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} else {
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AddFrameComparison(frames_.front(), *last_rendered_frame_, true,
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render_time_ms);
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}
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frames_.pop_front();
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RTC_DCHECK(!frames_.empty());
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}
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VideoFrame reference_frame = frames_.front();
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frames_.pop_front();
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int64_t reference_timestamp =
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wrap_handler_.Unwrap(reference_frame.timestamp());
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if (send_timestamp == reference_timestamp - 1) {
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// TODO(ivica): Make this work for > 2 streams.
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// Look at RTPSender::BuildRTPHeader.
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++send_timestamp;
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}
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ASSERT_EQ(reference_timestamp, send_timestamp);
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AddFrameComparison(reference_frame, video_frame, false, render_time_ms);
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last_rendered_frame_ = video_frame;
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StopExcludingCpuThreadTime();
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}
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void VideoAnalyzer::Wait() {
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// Frame comparisons can be very expensive. Wait for test to be done, but
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// at time-out check if frames_processed is going up. If so, give it more
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// time, otherwise fail. Hopefully this will reduce test flakiness.
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RepeatingTaskHandle stats_polling_task = RepeatingTaskHandle::DelayedStart(
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task_queue_, kSendStatsPollingInterval, [this] {
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PollStats();
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return kSendStatsPollingInterval;
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});
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int last_frames_processed = -1;
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int last_frames_captured = -1;
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int iteration = 0;
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while (!done_.Wait(kProbingIntervalMs)) {
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int frames_processed;
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int frames_captured;
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{
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MutexLock lock(&comparison_lock_);
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frames_processed = frames_processed_;
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frames_captured = captured_frames_;
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}
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// Print some output so test infrastructure won't think we've crashed.
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const char* kKeepAliveMessages[3] = {
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"Uh, I'm-I'm not quite dead, sir.",
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"Uh, I-I think uh, I could pull through, sir.",
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"Actually, I think I'm all right to come with you--"};
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if (++iteration % kKeepAliveIntervalIterations == 0) {
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printf("- %s\n", kKeepAliveMessages[iteration % 3]);
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}
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if (last_frames_processed == -1) {
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last_frames_processed = frames_processed;
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last_frames_captured = frames_captured;
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continue;
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}
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if (frames_processed == last_frames_processed &&
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last_frames_captured == frames_captured &&
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clock_->CurrentTime() > test_end_) {
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done_.Set();
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break;
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}
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last_frames_processed = frames_processed;
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last_frames_captured = frames_captured;
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}
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if (iteration > 0)
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printf("- Farewell, sweet Concorde!\n");
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SendTask(RTC_FROM_HERE, task_queue_, [&] { stats_polling_task.Stop(); });
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PrintResults();
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if (graph_data_output_file_)
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PrintSamplesToFile();
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}
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void VideoAnalyzer::StartMeasuringCpuProcessTime() {
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MutexLock lock(&cpu_measurement_lock_);
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cpu_time_ -= rtc::GetProcessCpuTimeNanos();
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wallclock_time_ -= rtc::SystemTimeNanos();
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}
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void VideoAnalyzer::StopMeasuringCpuProcessTime() {
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MutexLock lock(&cpu_measurement_lock_);
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cpu_time_ += rtc::GetProcessCpuTimeNanos();
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wallclock_time_ += rtc::SystemTimeNanos();
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}
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void VideoAnalyzer::StartExcludingCpuThreadTime() {
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MutexLock lock(&cpu_measurement_lock_);
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cpu_time_ += rtc::GetThreadCpuTimeNanos();
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}
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void VideoAnalyzer::StopExcludingCpuThreadTime() {
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MutexLock lock(&cpu_measurement_lock_);
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cpu_time_ -= rtc::GetThreadCpuTimeNanos();
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}
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double VideoAnalyzer::GetCpuUsagePercent() {
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MutexLock lock(&cpu_measurement_lock_);
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return static_cast<double>(cpu_time_) / wallclock_time_ * 100.0;
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}
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bool VideoAnalyzer::IsInSelectedSpatialAndTemporalLayer(
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const RtpPacket& rtp_packet) {
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if (rtp_packet.PayloadType() == test::CallTest::kPayloadTypeVP8) {
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auto parsed_payload = vp8_depacketizer_->Parse(rtp_packet.PayloadBuffer());
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RTC_DCHECK(parsed_payload);
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const auto& vp8_header = absl::get<RTPVideoHeaderVP8>(
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parsed_payload->video_header.video_type_header);
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int temporal_idx = vp8_header.temporalIdx;
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return selected_tl_ < 0 || temporal_idx == kNoTemporalIdx ||
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temporal_idx <= selected_tl_;
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}
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if (rtp_packet.PayloadType() == test::CallTest::kPayloadTypeVP9) {
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auto parsed_payload = vp9_depacketizer_->Parse(rtp_packet.PayloadBuffer());
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RTC_DCHECK(parsed_payload);
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const auto& vp9_header = absl::get<RTPVideoHeaderVP9>(
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parsed_payload->video_header.video_type_header);
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int temporal_idx = vp9_header.temporal_idx;
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int spatial_idx = vp9_header.spatial_idx;
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return (selected_tl_ < 0 || temporal_idx == kNoTemporalIdx ||
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temporal_idx <= selected_tl_) &&
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(selected_sl_ < 0 || spatial_idx == kNoSpatialIdx ||
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spatial_idx <= selected_sl_);
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}
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return true;
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}
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void VideoAnalyzer::PollStats() {
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// Do not grab |comparison_lock_|, before |GetStats()| completes.
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// Otherwise a deadlock may occur:
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// 1) |comparison_lock_| is acquired after |lock_|
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// 2) |lock_| is acquired after internal pacer lock in SendRtp()
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// 3) internal pacer lock is acquired by GetStats().
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Call::Stats call_stats = call_->GetStats();
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MutexLock lock(&comparison_lock_);
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send_bandwidth_bps_.AddSample(call_stats.send_bandwidth_bps);
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VideoSendStream::Stats send_stats = send_stream_->GetStats();
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// It's not certain that we yet have estimates for any of these stats.
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// Check that they are positive before mixing them in.
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if (send_stats.encode_frame_rate > 0)
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encode_frame_rate_.AddSample(send_stats.encode_frame_rate);
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if (send_stats.avg_encode_time_ms > 0)
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encode_time_ms_.AddSample(send_stats.avg_encode_time_ms);
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if (send_stats.encode_usage_percent > 0)
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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 (const 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();
|
|
// |total_decode_time_ms| gives a good estimate of the mean decode time,
|
|
// |decode_ms| is used to keep track of the standard deviation.
|
|
if (receive_stats.frames_decoded > 0)
|
|
mean_decode_time_ms_ =
|
|
static_cast<double>(receive_stats.total_decode_time_ms) /
|
|
receive_stats.frames_decoded;
|
|
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 (receive_stats.width > 0 && receive_stats.height > 0) {
|
|
pixels_.AddSample(receive_stats.width * receive_stats.height);
|
|
}
|
|
|
|
// |frames_decoded| and |frames_rendered| are used because they are more
|
|
// accurate than |decode_frame_rate| and |render_frame_rate|.
|
|
// The latter two are calculated on a momentary basis.
|
|
const double total_frames_duration_sec_double =
|
|
static_cast<double>(receive_stats.total_frames_duration_ms) / 1000.0;
|
|
if (total_frames_duration_sec_double > 0) {
|
|
decode_frame_rate_ = static_cast<double>(receive_stats.frames_decoded) /
|
|
total_frames_duration_sec_double;
|
|
render_frame_rate_ = static_cast<double>(receive_stats.frames_rendered) /
|
|
total_frames_duration_sec_double;
|
|
}
|
|
|
|
// Freeze metrics.
|
|
freeze_count_ = receive_stats.freeze_count;
|
|
total_freezes_duration_ms_ = receive_stats.total_freezes_duration_ms;
|
|
total_frames_duration_ms_ = receive_stats.total_frames_duration_ms;
|
|
sum_squared_frame_durations_ = receive_stats.sum_squared_frame_durations;
|
|
}
|
|
|
|
if (audio_receive_stream_ != nullptr) {
|
|
AudioReceiveStream::Stats receive_stats =
|
|
audio_receive_stream_->GetStats(/*get_and_clear_legacy_stats=*/true);
|
|
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::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()) {
|
|
done_.Set();
|
|
comparison_available_event_.Set();
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool VideoAnalyzer::PopComparison(VideoAnalyzer::FrameComparison* comparison) {
|
|
MutexLock lock(&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() || AllFramesRecordedLocked())
|
|
return false;
|
|
|
|
*comparison = comparisons_.front();
|
|
comparisons_.pop_front();
|
|
|
|
FrameRecorded();
|
|
return true;
|
|
}
|
|
|
|
void VideoAnalyzer::FrameRecorded() {
|
|
++frames_recorded_;
|
|
}
|
|
|
|
bool VideoAnalyzer::AllFramesRecorded() {
|
|
MutexLock lock(&comparison_lock_);
|
|
return AllFramesRecordedLocked();
|
|
}
|
|
|
|
bool VideoAnalyzer::AllFramesRecordedLocked() {
|
|
RTC_DCHECK(frames_recorded_ <= frames_to_process_);
|
|
return frames_recorded_ == frames_to_process_ ||
|
|
(clock_->CurrentTime() > test_end_ && comparisons_.empty()) || quit_;
|
|
}
|
|
|
|
bool VideoAnalyzer::FrameProcessed() {
|
|
MutexLock lock(&comparison_lock_);
|
|
++frames_processed_;
|
|
assert(frames_processed_ <= frames_to_process_);
|
|
return frames_processed_ == frames_to_process_ ||
|
|
(clock_->CurrentTime() > test_end_ && comparisons_.empty());
|
|
}
|
|
|
|
void VideoAnalyzer::PrintResults() {
|
|
using ::webrtc::test::ImproveDirection;
|
|
|
|
StopMeasuringCpuProcessTime();
|
|
int dropped_frames_diff;
|
|
{
|
|
MutexLock lock(&lock_);
|
|
dropped_frames_diff = dropped_frames_before_first_encode_ +
|
|
dropped_frames_before_rendering_ + frames_.size();
|
|
}
|
|
MutexLock lock(&comparison_lock_);
|
|
PrintResult("psnr", psnr_, "dB", ImproveDirection::kBiggerIsBetter);
|
|
PrintResult("ssim", ssim_, "unitless", ImproveDirection::kBiggerIsBetter);
|
|
PrintResult("sender_time", sender_time_, "ms",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
PrintResult("receiver_time", receiver_time_, "ms",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
PrintResult("network_time", network_time_, "ms",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
PrintResult("total_delay_incl_network", end_to_end_, "ms",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
PrintResult("time_between_rendered_frames", rendered_delta_, "ms",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
PrintResult("encode_frame_rate", encode_frame_rate_, "fps",
|
|
ImproveDirection::kBiggerIsBetter);
|
|
PrintResult("encode_time", encode_time_ms_, "ms",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
PrintResult("media_bitrate", media_bitrate_bps_, "bps",
|
|
ImproveDirection::kNone);
|
|
PrintResult("fec_bitrate", fec_bitrate_bps_, "bps", ImproveDirection::kNone);
|
|
PrintResult("send_bandwidth", send_bandwidth_bps_, "bps",
|
|
ImproveDirection::kNone);
|
|
PrintResult("pixels_per_frame", pixels_, "count",
|
|
ImproveDirection::kBiggerIsBetter);
|
|
|
|
test::PrintResult("decode_frame_rate", "", test_label_.c_str(),
|
|
decode_frame_rate_, "fps", false,
|
|
ImproveDirection::kBiggerIsBetter);
|
|
test::PrintResult("render_frame_rate", "", test_label_.c_str(),
|
|
render_frame_rate_, "fps", false,
|
|
ImproveDirection::kBiggerIsBetter);
|
|
|
|
// 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_);
|
|
|
|
// Freeze metrics.
|
|
PrintResult("time_between_freezes", time_between_freezes_, "ms",
|
|
ImproveDirection::kBiggerIsBetter);
|
|
|
|
const double freeze_count_double = static_cast<double>(freeze_count_);
|
|
const double total_freezes_duration_ms_double =
|
|
static_cast<double>(total_freezes_duration_ms_);
|
|
const double total_frames_duration_ms_double =
|
|
static_cast<double>(total_frames_duration_ms_);
|
|
|
|
if (total_frames_duration_ms_double > 0) {
|
|
test::PrintResult(
|
|
"freeze_duration_ratio", "", test_label_.c_str(),
|
|
total_freezes_duration_ms_double / total_frames_duration_ms_double,
|
|
"unitless", false, ImproveDirection::kSmallerIsBetter);
|
|
RTC_DCHECK_LE(total_freezes_duration_ms_double,
|
|
total_frames_duration_ms_double);
|
|
|
|
constexpr double ms_per_minute = 60 * 1000;
|
|
const double total_frames_duration_min =
|
|
total_frames_duration_ms_double / ms_per_minute;
|
|
if (total_frames_duration_min > 0) {
|
|
test::PrintResult("freeze_count_per_minute", "", test_label_.c_str(),
|
|
freeze_count_double / total_frames_duration_min,
|
|
"unitless", false, ImproveDirection::kSmallerIsBetter);
|
|
}
|
|
}
|
|
|
|
test::PrintResult("freeze_duration_average", "", test_label_.c_str(),
|
|
freeze_count_double > 0
|
|
? total_freezes_duration_ms_double / freeze_count_double
|
|
: 0,
|
|
"ms", false, ImproveDirection::kSmallerIsBetter);
|
|
|
|
if (1000 * sum_squared_frame_durations_ > 0) {
|
|
test::PrintResult(
|
|
"harmonic_frame_rate", "", test_label_.c_str(),
|
|
total_frames_duration_ms_double / (1000 * sum_squared_frame_durations_),
|
|
"fps", false, ImproveDirection::kBiggerIsBetter);
|
|
}
|
|
|
|
if (worst_frame_) {
|
|
test::PrintResult("min_psnr", "", test_label_.c_str(), worst_frame_->psnr,
|
|
"dB", false, ImproveDirection::kBiggerIsBetter);
|
|
}
|
|
|
|
if (receive_stream_ != nullptr) {
|
|
PrintResultWithExternalMean("decode_time", mean_decode_time_ms_,
|
|
decode_time_ms_, "ms",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
}
|
|
dropped_frames_ += dropped_frames_diff;
|
|
test::PrintResult("dropped_frames", "", test_label_.c_str(), dropped_frames_,
|
|
"count", false, ImproveDirection::kSmallerIsBetter);
|
|
test::PrintResult("cpu_usage", "", test_label_.c_str(), GetCpuUsagePercent(),
|
|
"%", false, ImproveDirection::kSmallerIsBetter);
|
|
|
|
#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_, "sizeInBytes",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
#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 (absl::GetFlag(FLAGS_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_, "unitless",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
PrintResult("audio_accelerate_rate", audio_accelerate_rate_, "unitless",
|
|
ImproveDirection::kSmallerIsBetter);
|
|
PrintResult("audio_jitter_buffer", audio_jitter_buffer_ms_, "ms",
|
|
ImproveDirection::kNone);
|
|
}
|
|
|
|
// 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_.GetMean(), avg_psnr_threshold_);
|
|
EXPECT_GT(*ssim_.GetMean(), 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);
|
|
}
|
|
|
|
MutexLock lock(&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,
|
|
Statistics stats,
|
|
const char* unit,
|
|
webrtc::test::ImproveDirection improve_direction) {
|
|
test::PrintResultMeanAndError(
|
|
result_type, "", test_label_.c_str(), stats.GetMean().value_or(0),
|
|
stats.GetStandardDeviation().value_or(0), unit, false, improve_direction);
|
|
}
|
|
|
|
void VideoAnalyzer::PrintResultWithExternalMean(
|
|
const char* result_type,
|
|
double mean,
|
|
Statistics stats,
|
|
const char* unit,
|
|
webrtc::test::ImproveDirection improve_direction) {
|
|
// If the true mean is different than the sample mean, the sample variance is
|
|
// too low. The sample variance given a known mean is obtained by adding the
|
|
// squared error between the true mean and the sample mean.
|
|
double compensated_variance =
|
|
stats.Size() > 0
|
|
? *stats.GetVariance() + pow(mean - *stats.GetMean(), 2.0)
|
|
: 0.0;
|
|
test::PrintResultMeanAndError(result_type, "", test_label_.c_str(), mean,
|
|
std::sqrt(compensated_variance), unit, false,
|
|
improve_direction);
|
|
}
|
|
|
|
void VideoAnalyzer::PrintSamplesToFile() {
|
|
FILE* out = graph_data_output_file_;
|
|
MutexLock lock(&comparison_lock_);
|
|
absl::c_sort(samples_, [](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, "%" RTC_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 " %" RTC_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);
|
|
}
|
|
}
|
|
|
|
void VideoAnalyzer::AddCapturedFrameForComparison(
|
|
const VideoFrame& video_frame) {
|
|
bool must_capture = false;
|
|
{
|
|
MutexLock lock(&comparison_lock_);
|
|
must_capture = captured_frames_ < frames_to_process_;
|
|
if (must_capture) {
|
|
++captured_frames_;
|
|
}
|
|
}
|
|
if (must_capture) {
|
|
MutexLock lock(&lock_);
|
|
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);
|
|
|
|
MutexLock lock(&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,
|
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const VideoFrame& render,
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bool dropped,
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int64_t input_time_ms,
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int64_t send_time_ms,
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int64_t recv_time_ms,
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int64_t render_time_ms,
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size_t encoded_frame_size)
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: reference(reference),
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render(render),
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dropped(dropped),
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input_time_ms(input_time_ms),
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send_time_ms(send_time_ms),
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recv_time_ms(recv_time_ms),
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render_time_ms(render_time_ms),
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encoded_frame_size(encoded_frame_size) {}
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VideoAnalyzer::FrameComparison::FrameComparison(bool dropped,
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int64_t input_time_ms,
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int64_t send_time_ms,
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int64_t recv_time_ms,
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int64_t render_time_ms,
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size_t encoded_frame_size)
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: dropped(dropped),
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input_time_ms(input_time_ms),
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send_time_ms(send_time_ms),
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recv_time_ms(recv_time_ms),
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render_time_ms(render_time_ms),
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encoded_frame_size(encoded_frame_size) {}
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VideoAnalyzer::Sample::Sample(int dropped,
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int64_t input_time_ms,
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int64_t send_time_ms,
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int64_t recv_time_ms,
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int64_t render_time_ms,
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size_t encoded_frame_size,
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double psnr,
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double ssim)
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: dropped(dropped),
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input_time_ms(input_time_ms),
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send_time_ms(send_time_ms),
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recv_time_ms(recv_time_ms),
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render_time_ms(render_time_ms),
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encoded_frame_size(encoded_frame_size),
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psnr(psnr),
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ssim(ssim) {}
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VideoAnalyzer::CapturedFrameForwarder::CapturedFrameForwarder(
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VideoAnalyzer* analyzer,
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Clock* clock,
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int frames_to_capture,
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TimeDelta test_duration)
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: analyzer_(analyzer),
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send_stream_input_(nullptr),
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video_source_(nullptr),
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clock_(clock),
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captured_frames_(0),
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frames_to_capture_(frames_to_capture),
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test_end_(clock->CurrentTime() + test_duration) {}
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void VideoAnalyzer::CapturedFrameForwarder::SetSource(
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VideoSourceInterface<VideoFrame>* video_source) {
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video_source_ = video_source;
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}
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void VideoAnalyzer::CapturedFrameForwarder::OnFrame(
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const VideoFrame& video_frame) {
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VideoFrame copy = video_frame;
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// Frames from the capturer does not have a rtp timestamp.
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// Create one so it can be used for comparison.
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RTC_DCHECK_EQ(0, video_frame.timestamp());
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if (video_frame.ntp_time_ms() == 0)
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copy.set_ntp_time_ms(clock_->CurrentNtpInMilliseconds());
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copy.set_timestamp(copy.ntp_time_ms() * 90);
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analyzer_->AddCapturedFrameForComparison(copy);
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MutexLock lock(&lock_);
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++captured_frames_;
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if (send_stream_input_ && clock_->CurrentTime() <= test_end_ &&
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captured_frames_ <= frames_to_capture_) {
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send_stream_input_->OnFrame(copy);
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}
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}
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void VideoAnalyzer::CapturedFrameForwarder::AddOrUpdateSink(
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rtc::VideoSinkInterface<VideoFrame>* sink,
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const rtc::VideoSinkWants& wants) {
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{
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MutexLock lock(&lock_);
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RTC_DCHECK(!send_stream_input_ || send_stream_input_ == sink);
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send_stream_input_ = sink;
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}
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if (video_source_) {
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video_source_->AddOrUpdateSink(this, wants);
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}
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}
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void VideoAnalyzer::CapturedFrameForwarder::RemoveSink(
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rtc::VideoSinkInterface<VideoFrame>* sink) {
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MutexLock lock(&lock_);
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RTC_DCHECK(sink == send_stream_input_);
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send_stream_input_ = nullptr;
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}
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} // namespace webrtc
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