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webrtc_m130/video/video_analyzer.h
Markus Handell ad5037b4a8 Reland "Refactor the PlatformThread API." 
This reverts commit 793bac569fdf1be16cbf24d7871d20d00bbec81b.

Reason for revert: rare compilation error fixed

Original change's description:
> Revert "Refactor the PlatformThread API."
>
> This reverts commit c89fdd716c4c8af608017c76f75bf27e4c3d602e.
>
> Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
> See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
>
> Original change's description:
> > Refactor the PlatformThread API.
> >
> > PlatformThread's API is using old style function pointers, causes
> > casting, is unintuitive and forces artificial call sequences, and
> > is additionally possible to misuse in release mode.
> >
> > Fix this by an API face lift:
> > 1. The class is turned into a handle, which can be empty.
> > 2. The only way of getting a non-empty PlatformThread is by calling
> > SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> > code reader.
> > 3. Handles can be Finalized, which works differently for joinable and
> > detached threads:
> >   a) Handles for detached threads are simply closed where applicable.
> >   b) Joinable threads are joined before handles are closed.
> > 4. The destructor finalizes handles. No explicit call is needed.
> >
> > Fixed: webrtc:12727
> > Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33923}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=handellm@webrtc.org
>
> Bug: webrtc:12727
> Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33936}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:12727
Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33950}
2021-05-07 14:14:43 +00:00

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/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef VIDEO_VIDEO_ANALYZER_H_
#define VIDEO_VIDEO_ANALYZER_H_
#include <deque>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/task_queue/task_queue_base.h"
#include "api/video/video_source_interface.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/event.h"
#include "rtc_base/numerics/running_statistics.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/time_utils.h"
#include "test/layer_filtering_transport.h"
#include "test/rtp_file_writer.h"
#include "test/testsupport/perf_test.h"
namespace webrtc {
class VideoAnalyzer : public PacketReceiver,
public Transport,
public rtc::VideoSinkInterface<VideoFrame> {
public:
using Statistics = webrtc_impl::RunningStatistics<double>;
VideoAnalyzer(test::LayerFilteringTransport* transport,
const std::string& test_label,
double avg_psnr_threshold,
double avg_ssim_threshold,
int duration_frames,
TimeDelta test_duration,
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,
TaskQueueBase* task_queue);
~VideoAnalyzer();
virtual void SetReceiver(PacketReceiver* receiver);
void SetSource(rtc::VideoSourceInterface<VideoFrame>* video_source,
bool respect_sink_wants);
void SetCall(Call* call);
void SetSendStream(VideoSendStream* stream);
void SetReceiveStream(VideoReceiveStream* stream);
void SetAudioReceiveStream(AudioReceiveStream* recv_stream);
rtc::VideoSinkInterface<VideoFrame>* InputInterface();
rtc::VideoSourceInterface<VideoFrame>* OutputInterface();
DeliveryStatus DeliverPacket(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
void PreEncodeOnFrame(const VideoFrame& video_frame);
void PostEncodeOnFrame(size_t stream_id, uint32_t timestamp);
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override;
bool SendRtcp(const uint8_t* packet, size_t length) override;
void OnFrame(const VideoFrame& video_frame) override;
void Wait();
void StartMeasuringCpuProcessTime();
void StopMeasuringCpuProcessTime();
void StartExcludingCpuThreadTime() RTC_LOCKS_EXCLUDED(cpu_measurement_lock_);
void StopExcludingCpuThreadTime() RTC_LOCKS_EXCLUDED(cpu_measurement_lock_);
double GetCpuUsagePercent() RTC_LOCKS_EXCLUDED(cpu_measurement_lock_);
test::LayerFilteringTransport* const transport_;
PacketReceiver* receiver_;
private:
struct FrameComparison {
FrameComparison();
FrameComparison(const VideoFrame& reference,
const VideoFrame& render,
bool dropped,
int64_t input_time_ms,
int64_t send_time_ms,
int64_t recv_time_ms,
int64_t render_time_ms,
size_t encoded_frame_size);
FrameComparison(bool dropped,
int64_t input_time_ms,
int64_t send_time_ms,
int64_t recv_time_ms,
int64_t render_time_ms,
size_t encoded_frame_size);
absl::optional<VideoFrame> reference;
absl::optional<VideoFrame> render;
bool dropped;
int64_t input_time_ms;
int64_t send_time_ms;
int64_t recv_time_ms;
int64_t render_time_ms;
size_t encoded_frame_size;
};
struct Sample {
Sample(int dropped,
int64_t input_time_ms,
int64_t send_time_ms,
int64_t recv_time_ms,
int64_t render_time_ms,
size_t encoded_frame_size,
double psnr,
double ssim);
int dropped;
int64_t input_time_ms;
int64_t send_time_ms;
int64_t recv_time_ms;
int64_t render_time_ms;
size_t encoded_frame_size;
double psnr;
double ssim;
};
// Implements VideoSinkInterface to receive captured frames from a
// FrameGeneratorCapturer. Implements VideoSourceInterface to be able to act
// as a source to VideoSendStream.
// It forwards all input frames to the VideoAnalyzer for later comparison and
// forwards the captured frames to the VideoSendStream.
class CapturedFrameForwarder : public rtc::VideoSinkInterface<VideoFrame>,
public rtc::VideoSourceInterface<VideoFrame> {
public:
CapturedFrameForwarder(VideoAnalyzer* analyzer,
Clock* clock,
int frames_to_capture,
TimeDelta test_duration);
void SetSource(rtc::VideoSourceInterface<VideoFrame>* video_source);
private:
void OnFrame(const VideoFrame& video_frame)
RTC_LOCKS_EXCLUDED(lock_) override;
// Called when |send_stream_.SetSource()| is called.
void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
const rtc::VideoSinkWants& wants)
RTC_LOCKS_EXCLUDED(lock_) override;
// Called by |send_stream_| when |send_stream_.SetSource()| is called.
void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink)
RTC_LOCKS_EXCLUDED(lock_) override;
VideoAnalyzer* const analyzer_;
Mutex lock_;
rtc::VideoSinkInterface<VideoFrame>* send_stream_input_
RTC_GUARDED_BY(lock_);
VideoSourceInterface<VideoFrame>* video_source_;
Clock* clock_;
int captured_frames_ RTC_GUARDED_BY(lock_);
const int frames_to_capture_;
const Timestamp test_end_;
};
struct FrameWithPsnr {
double psnr;
VideoFrame frame;
};
bool IsInSelectedSpatialAndTemporalLayer(const RtpPacket& rtp_packet);
void AddFrameComparison(const VideoFrame& reference,
const VideoFrame& render,
bool dropped,
int64_t render_time_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
void PollStats() RTC_LOCKS_EXCLUDED(comparison_lock_);
static void FrameComparisonThread(void* obj);
bool CompareFrames();
bool PopComparison(FrameComparison* comparison);
// Increment counter for number of frames received for comparison.
void FrameRecorded() RTC_EXCLUSIVE_LOCKS_REQUIRED(comparison_lock_);
// Returns true if all frames to be compared have been taken from the queue.
bool AllFramesRecorded() RTC_LOCKS_EXCLUDED(comparison_lock_);
bool AllFramesRecordedLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(comparison_lock_);
// Increase count of number of frames processed. Returns true if this was the
// last frame to be processed.
bool FrameProcessed() RTC_LOCKS_EXCLUDED(comparison_lock_);
void PrintResults() RTC_LOCKS_EXCLUDED(lock_, comparison_lock_);
void PerformFrameComparison(const FrameComparison& comparison)
RTC_LOCKS_EXCLUDED(comparison_lock_);
void PrintResult(const char* result_type,
Statistics stats,
const char* unit,
webrtc::test::ImproveDirection improve_direction);
void PrintResultWithExternalMean(
const char* result_type,
double mean,
Statistics stats,
const char* unit,
webrtc::test::ImproveDirection improve_direction);
void PrintSamplesToFile(void) RTC_LOCKS_EXCLUDED(comparison_lock_);
void AddCapturedFrameForComparison(const VideoFrame& video_frame)
RTC_LOCKS_EXCLUDED(lock_, comparison_lock_);
Call* call_;
VideoSendStream* send_stream_;
VideoReceiveStream* receive_stream_;
AudioReceiveStream* audio_receive_stream_;
CapturedFrameForwarder captured_frame_forwarder_;
const std::string test_label_;
FILE* const graph_data_output_file_;
const std::string graph_title_;
const uint32_t ssrc_to_analyze_;
const uint32_t rtx_ssrc_to_analyze_;
const size_t selected_stream_;
const int selected_sl_;
const int selected_tl_;
Mutex comparison_lock_;
std::vector<Sample> samples_ RTC_GUARDED_BY(comparison_lock_);
Statistics sender_time_ RTC_GUARDED_BY(comparison_lock_);
Statistics receiver_time_ RTC_GUARDED_BY(comparison_lock_);
Statistics network_time_ RTC_GUARDED_BY(comparison_lock_);
Statistics psnr_ RTC_GUARDED_BY(comparison_lock_);
Statistics ssim_ RTC_GUARDED_BY(comparison_lock_);
Statistics end_to_end_ RTC_GUARDED_BY(comparison_lock_);
Statistics rendered_delta_ RTC_GUARDED_BY(comparison_lock_);
Statistics encoded_frame_size_ RTC_GUARDED_BY(comparison_lock_);
Statistics encode_frame_rate_ RTC_GUARDED_BY(comparison_lock_);
Statistics encode_time_ms_ RTC_GUARDED_BY(comparison_lock_);
Statistics encode_usage_percent_ RTC_GUARDED_BY(comparison_lock_);
double mean_decode_time_ms_ RTC_GUARDED_BY(comparison_lock_);
Statistics decode_time_ms_ RTC_GUARDED_BY(comparison_lock_);
Statistics decode_time_max_ms_ RTC_GUARDED_BY(comparison_lock_);
Statistics media_bitrate_bps_ RTC_GUARDED_BY(comparison_lock_);
Statistics fec_bitrate_bps_ RTC_GUARDED_BY(comparison_lock_);
Statistics send_bandwidth_bps_ RTC_GUARDED_BY(comparison_lock_);
Statistics memory_usage_ RTC_GUARDED_BY(comparison_lock_);
Statistics audio_expand_rate_ RTC_GUARDED_BY(comparison_lock_);
Statistics audio_accelerate_rate_ RTC_GUARDED_BY(comparison_lock_);
Statistics audio_jitter_buffer_ms_ RTC_GUARDED_BY(comparison_lock_);
Statistics pixels_ RTC_GUARDED_BY(comparison_lock_);
// Rendered frame with worst PSNR is saved for further analysis.
absl::optional<FrameWithPsnr> worst_frame_ RTC_GUARDED_BY(comparison_lock_);
// Freeze metrics.
Statistics time_between_freezes_ RTC_GUARDED_BY(comparison_lock_);
uint32_t freeze_count_ RTC_GUARDED_BY(comparison_lock_);
uint32_t total_freezes_duration_ms_ RTC_GUARDED_BY(comparison_lock_);
uint32_t total_frames_duration_ms_ RTC_GUARDED_BY(comparison_lock_);
double sum_squared_frame_durations_ RTC_GUARDED_BY(comparison_lock_);
double decode_frame_rate_ RTC_GUARDED_BY(comparison_lock_);
double render_frame_rate_ RTC_GUARDED_BY(comparison_lock_);
size_t last_fec_bytes_;
Mutex lock_ RTC_ACQUIRED_BEFORE(comparison_lock_)
RTC_ACQUIRED_BEFORE(cpu_measurement_lock_);
const int frames_to_process_;
const Timestamp test_end_;
int frames_recorded_ RTC_GUARDED_BY(comparison_lock_);
int frames_processed_ RTC_GUARDED_BY(comparison_lock_);
int captured_frames_ RTC_GUARDED_BY(comparison_lock_);
int dropped_frames_ RTC_GUARDED_BY(comparison_lock_);
int dropped_frames_before_first_encode_ RTC_GUARDED_BY(lock_);
int dropped_frames_before_rendering_ RTC_GUARDED_BY(lock_);
int64_t last_render_time_ RTC_GUARDED_BY(comparison_lock_);
int64_t last_render_delta_ms_ RTC_GUARDED_BY(comparison_lock_);
int64_t last_unfreeze_time_ms_ RTC_GUARDED_BY(comparison_lock_);
uint32_t rtp_timestamp_delta_ RTC_GUARDED_BY(lock_);
Mutex cpu_measurement_lock_;
int64_t cpu_time_ RTC_GUARDED_BY(cpu_measurement_lock_);
int64_t wallclock_time_ RTC_GUARDED_BY(cpu_measurement_lock_);
std::deque<VideoFrame> frames_ RTC_GUARDED_BY(lock_);
absl::optional<VideoFrame> last_rendered_frame_ RTC_GUARDED_BY(lock_);
rtc::TimestampWrapAroundHandler wrap_handler_ RTC_GUARDED_BY(lock_);
std::map<int64_t, int64_t> send_times_ RTC_GUARDED_BY(lock_);
std::map<int64_t, int64_t> recv_times_ RTC_GUARDED_BY(lock_);
std::map<int64_t, size_t> encoded_frame_sizes_ RTC_GUARDED_BY(lock_);
absl::optional<uint32_t> first_encoded_timestamp_ RTC_GUARDED_BY(lock_);
absl::optional<uint32_t> first_sent_timestamp_ RTC_GUARDED_BY(lock_);
const double avg_psnr_threshold_;
const double avg_ssim_threshold_;
bool is_quick_test_enabled_;
std::vector<rtc::PlatformThread> comparison_thread_pool_;
rtc::Event comparison_available_event_;
std::deque<FrameComparison> comparisons_ RTC_GUARDED_BY(comparison_lock_);
bool quit_ RTC_GUARDED_BY(comparison_lock_);
rtc::Event done_;
std::unique_ptr<VideoRtpDepacketizer> vp8_depacketizer_;
std::unique_ptr<VideoRtpDepacketizer> vp9_depacketizer_;
std::unique_ptr<test::RtpFileWriter> rtp_file_writer_;
Clock* const clock_;
const int64_t start_ms_;
TaskQueueBase* task_queue_;
};
} // namespace webrtc
#endif // VIDEO_VIDEO_ANALYZER_H_
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