Reland "Fix threading model of video quality test with audio enabled"
This is a reland of f537da6c194d2c021709a255563c27b261e92488 Original change's description: > Fix threading model of video quality test with audio enabled > > Bug: None > Change-Id: Ifb7fc57df54ec4d0a6f8c7f0504f3c06de6ac756 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130514 > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27413} Bug: None Change-Id: I4fb793a5a5f636103159ed537847d6f2deb60108 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132797 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27621}
This commit is contained in:
@ -49,21 +49,23 @@ bool IsFlexfec(int payload_type) {
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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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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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VideoAnalyzer::VideoAnalyzer(
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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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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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test::SingleThreadedTaskQueueForTesting* task_queue)
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: transport_(transport),
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receiver_(nullptr),
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call_(nullptr),
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@ -99,10 +101,10 @@ VideoAnalyzer::VideoAnalyzer(test::LayerFilteringTransport* transport,
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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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stats_polling_thread_(&PollStatsThread, this, "StatsPoller"),
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done_(true, false),
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clock_(clock),
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start_ms_(clock->TimeInMilliseconds()) {
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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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@ -336,7 +338,12 @@ void VideoAnalyzer::Wait() {
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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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stats_polling_thread_.Start();
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{
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rtc::CritScope lock(&comparison_lock_);
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stop_stats_poller_ = false;
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stats_polling_task_id_ = task_queue_->PostDelayedTask(
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[this]() { PollStats(); }, kSendStatsPollingIntervalMs);
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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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@ -381,11 +388,15 @@ void VideoAnalyzer::Wait() {
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if (iteration > 0)
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printf("- Farewell, sweet Concorde!\n");
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{
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rtc::CritScope lock(&comparison_lock_);
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stop_stats_poller_ = true;
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task_queue_->CancelTask(stats_polling_task_id_);
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}
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PrintResults();
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if (graph_data_output_file_)
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PrintSamplesToFile();
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stats_polling_thread_.Stop();
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}
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void VideoAnalyzer::StartMeasuringCpuProcessTime() {
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@ -456,57 +467,56 @@ bool VideoAnalyzer::IsInSelectedSpatialAndTemporalLayer(
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}
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}
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void VideoAnalyzer::PollStatsThread(void* obj) {
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static_cast<VideoAnalyzer*>(obj)->PollStats();
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}
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void VideoAnalyzer::PollStats() {
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while (!done_.Wait(kSendStatsPollingIntervalMs)) {
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rtc::CritScope crit(&comparison_lock_);
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Call::Stats call_stats = call_->GetStats();
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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);
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if (send_stats.media_bitrate_bps > 0)
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media_bitrate_bps_.AddSample(send_stats.media_bitrate_bps);
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size_t fec_bytes = 0;
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for (const auto& kv : send_stats.substreams) {
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fec_bytes += kv.second.rtp_stats.fec.payload_bytes +
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kv.second.rtp_stats.fec.padding_bytes;
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}
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fec_bitrate_bps_.AddSample((fec_bytes - last_fec_bytes_) * 8);
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last_fec_bytes_ = fec_bytes;
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if (receive_stream_ != nullptr) {
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VideoReceiveStream::Stats receive_stats = receive_stream_->GetStats();
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if (receive_stats.decode_ms > 0)
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decode_time_ms_.AddSample(receive_stats.decode_ms);
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if (receive_stats.max_decode_ms > 0)
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decode_time_max_ms_.AddSample(receive_stats.max_decode_ms);
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if (receive_stats.width > 0 && receive_stats.height > 0) {
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pixels_.AddSample(receive_stats.width * receive_stats.height);
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}
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}
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if (audio_receive_stream_ != nullptr) {
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AudioReceiveStream::Stats receive_stats =
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audio_receive_stream_->GetStats();
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audio_expand_rate_.AddSample(receive_stats.expand_rate);
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audio_accelerate_rate_.AddSample(receive_stats.accelerate_rate);
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audio_jitter_buffer_ms_.AddSample(receive_stats.jitter_buffer_ms);
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}
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memory_usage_.AddSample(rtc::GetProcessResidentSizeBytes());
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rtc::CritScope crit(&comparison_lock_);
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if (stop_stats_poller_) {
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return;
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}
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Call::Stats call_stats = call_->GetStats();
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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);
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if (send_stats.media_bitrate_bps > 0)
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media_bitrate_bps_.AddSample(send_stats.media_bitrate_bps);
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size_t fec_bytes = 0;
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for (const auto& kv : send_stats.substreams) {
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fec_bytes += kv.second.rtp_stats.fec.payload_bytes +
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kv.second.rtp_stats.fec.padding_bytes;
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}
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fec_bitrate_bps_.AddSample((fec_bytes - last_fec_bytes_) * 8);
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last_fec_bytes_ = fec_bytes;
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if (receive_stream_ != nullptr) {
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VideoReceiveStream::Stats receive_stats = receive_stream_->GetStats();
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if (receive_stats.decode_ms > 0)
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decode_time_ms_.AddSample(receive_stats.decode_ms);
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if (receive_stats.max_decode_ms > 0)
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decode_time_max_ms_.AddSample(receive_stats.max_decode_ms);
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if (receive_stats.width > 0 && receive_stats.height > 0) {
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pixels_.AddSample(receive_stats.width * receive_stats.height);
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}
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}
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if (audio_receive_stream_ != nullptr) {
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AudioReceiveStream::Stats receive_stats = audio_receive_stream_->GetStats();
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audio_expand_rate_.AddSample(receive_stats.expand_rate);
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audio_accelerate_rate_.AddSample(receive_stats.accelerate_rate);
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audio_jitter_buffer_ms_.AddSample(receive_stats.jitter_buffer_ms);
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}
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memory_usage_.AddSample(rtc::GetProcessResidentSizeBytes());
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stats_polling_task_id_ = task_queue_->PostDelayedTask(
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[this]() { PollStats(); }, kSendStatsPollingIntervalMs);
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}
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bool VideoAnalyzer::FrameComparisonThread(void* obj) {
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@ -44,7 +44,8 @@ class VideoAnalyzer : public PacketReceiver,
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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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std::string rtp_dump_name,
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test::SingleThreadedTaskQueueForTesting* task_queue);
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~VideoAnalyzer();
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virtual void SetReceiver(PacketReceiver* receiver);
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@ -178,7 +179,6 @@ class VideoAnalyzer : public PacketReceiver,
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int64_t render_time_ms)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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static void PollStatsThread(void* obj);
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void PollStats();
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static bool FrameComparisonThread(void* obj);
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bool CompareFrames();
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@ -273,14 +273,17 @@ class VideoAnalyzer : public PacketReceiver,
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bool is_quick_test_enabled_;
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std::vector<rtc::PlatformThread*> comparison_thread_pool_;
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rtc::PlatformThread stats_polling_thread_;
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rtc::Event comparison_available_event_;
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std::deque<FrameComparison> comparisons_ RTC_GUARDED_BY(comparison_lock_);
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rtc::Event done_;
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test::SingleThreadedTaskQueueForTesting::TaskId stats_polling_task_id_
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RTC_GUARDED_BY(comparison_lock_);
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bool stop_stats_poller_ RTC_GUARDED_BY(comparison_lock_);
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std::unique_ptr<test::RtpFileWriter> rtp_file_writer_;
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Clock* const clock_;
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const int64_t start_ms_;
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test::SingleThreadedTaskQueueForTesting* task_queue_;
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};
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} // namespace webrtc
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@ -1203,13 +1203,11 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
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recv_event_log_ = RtcEventLog::CreateNull();
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}
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Call::Config send_call_config(send_event_log_.get());
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Call::Config recv_call_config(recv_event_log_.get());
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send_call_config.bitrate_config = params.call.call_bitrate_config;
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recv_call_config.bitrate_config = params.call.call_bitrate_config;
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task_queue_.SendTask([this, &send_call_config, &recv_call_config,
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&send_transport, &recv_transport]() {
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task_queue_.SendTask([this, ¶ms, &send_transport, &recv_transport]() {
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Call::Config send_call_config(send_event_log_.get());
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Call::Config recv_call_config(recv_event_log_.get());
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send_call_config.bitrate_config = params.call.call_bitrate_config;
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recv_call_config.bitrate_config = params.call.call_bitrate_config;
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if (params_.audio.enabled)
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InitializeAudioDevice(&send_call_config, &recv_call_config,
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params_.audio.use_real_adm);
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@ -1234,7 +1232,8 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
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kSendRtxSsrcs[params_.ss[0].selected_stream],
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static_cast<size_t>(params_.ss[0].selected_stream),
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params.ss[0].selected_sl, params_.video[0].selected_tl,
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is_quick_test_enabled, clock_, params_.logging.rtp_dump_name);
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is_quick_test_enabled, clock_, params_.logging.rtp_dump_name,
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&task_queue_);
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task_queue_.SendTask([&]() {
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analyzer_->SetCall(sender_call_.get());
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