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}diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc
index 11d2c7b..d74b79a 100644
--- a/video/video_analyzer.cc
+++ b/video/video_analyzer.cc
@@ -49,21 +49,23 @@
}
} // 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)
+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,
+ test::SingleThreadedTaskQueueForTesting* task_queue)
: transport_(transport),
receiver_(nullptr),
call_(nullptr),
@@ -99,10 +101,10 @@
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()) {
+ start_ms_(clock->TimeInMilliseconds()),
+ task_queue_(task_queue) {
// Create thread pool for CPU-expensive PSNR/SSIM calculations.
// Try to use about as many threads as cores, but leave kMinCoresLeft alone,
@@ -336,7 +338,12 @@
// 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();
+ {
+ rtc::CritScope lock(&comparison_lock_);
+ stop_stats_poller_ = false;
+ stats_polling_task_id_ = task_queue_->PostDelayedTask(
+ [this]() { PollStats(); }, kSendStatsPollingIntervalMs);
+ }
int last_frames_processed = -1;
int last_frames_captured = -1;
@@ -381,11 +388,15 @@
if (iteration > 0)
printf("- Farewell, sweet Concorde!\n");
+ {
+ rtc::CritScope lock(&comparison_lock_);
+ stop_stats_poller_ = true;
+ task_queue_->CancelTask(stats_polling_task_id_);
+ }
+
PrintResults();
if (graph_data_output_file_)
PrintSamplesToFile();
-
- stats_polling_thread_.Stop();
}
void VideoAnalyzer::StartMeasuringCpuProcessTime() {
@@ -456,57 +467,56 @@
}
}
-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 (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();
- 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);
- }
- }
-
- 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());
+ rtc::CritScope crit(&comparison_lock_);
+ if (stop_stats_poller_) {
+ return;
}
+
+ 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 (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();
+ 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);
+ }
+ }
+
+ 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());
+
+ stats_polling_task_id_ = task_queue_->PostDelayedTask(
+ [this]() { PollStats(); }, kSendStatsPollingIntervalMs);
}
bool VideoAnalyzer::FrameComparisonThread(void* obj) {
diff --git a/video/video_analyzer.h b/video/video_analyzer.h
index 36ed03e..8550345 100644
--- a/video/video_analyzer.h
+++ b/video/video_analyzer.h
@@ -44,7 +44,8 @@
int selected_tl,
bool is_quick_test_enabled,
Clock* clock,
- std::string rtp_dump_name);
+ std::string rtp_dump_name,
+ test::SingleThreadedTaskQueueForTesting* task_queue);
~VideoAnalyzer();
virtual void SetReceiver(PacketReceiver* receiver);
@@ -178,7 +179,6 @@
int64_t render_time_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
- static void PollStatsThread(void* obj);
void PollStats();
static bool FrameComparisonThread(void* obj);
bool CompareFrames();
@@ -273,14 +273,17 @@
bool is_quick_test_enabled_;
std::vector<rtc::PlatformThread*> comparison_thread_pool_;
- rtc::PlatformThread stats_polling_thread_;
rtc::Event comparison_available_event_;
std::deque<FrameComparison> comparisons_ RTC_GUARDED_BY(comparison_lock_);
rtc::Event done_;
+ test::SingleThreadedTaskQueueForTesting::TaskId stats_polling_task_id_
+ RTC_GUARDED_BY(comparison_lock_);
+ bool stop_stats_poller_ RTC_GUARDED_BY(comparison_lock_);
std::unique_ptr<test::RtpFileWriter> rtp_file_writer_;
Clock* const clock_;
const int64_t start_ms_;
+ test::SingleThreadedTaskQueueForTesting* task_queue_;
};
} // namespace webrtc
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index 139be61..10f9699 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -1203,13 +1203,11 @@
recv_event_log_ = RtcEventLog::CreateNull();
}
- Call::Config send_call_config(send_event_log_.get());
- Call::Config recv_call_config(recv_event_log_.get());
- send_call_config.bitrate_config = params.call.call_bitrate_config;
- recv_call_config.bitrate_config = params.call.call_bitrate_config;
-
- task_queue_.SendTask([this, &send_call_config, &recv_call_config,
- &send_transport, &recv_transport]() {
+ task_queue_.SendTask([this, ¶ms, &send_transport, &recv_transport]() {
+ Call::Config send_call_config(send_event_log_.get());
+ Call::Config recv_call_config(recv_event_log_.get());
+ send_call_config.bitrate_config = params.call.call_bitrate_config;
+ recv_call_config.bitrate_config = params.call.call_bitrate_config;
if (params_.audio.enabled)
InitializeAudioDevice(&send_call_config, &recv_call_config,
params_.audio.use_real_adm);
@@ -1234,7 +1232,8 @@
kSendRtxSsrcs[params_.ss[0].selected_stream],
static_cast<size_t>(params_.ss[0].selected_stream),
params.ss[0].selected_sl, params_.video[0].selected_tl,
- is_quick_test_enabled, clock_, params_.logging.rtp_dump_name);
+ is_quick_test_enabled, clock_, params_.logging.rtp_dump_name,
+ &task_queue_);
task_queue_.SendTask([&]() {
analyzer_->SetCall(sender_call_.get());
