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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 "test/layer_filtering_transport.h"
#include "test/rtp_file_writer.h"
#include "test/statistics.h"
#include "test/vcm_capturer.h"
namespace webrtc {
class VideoAnalyzer : public PacketReceiver,
public Transport,
public rtc::VideoSinkInterface<VideoFrame>,
public EncodedFrameObserver {
public:
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();
virtual void SetReceiver(PacketReceiver* receiver);
void SetSource(test::VideoCapturer* video_capturer, 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);
// EncodedFrameObserver implementation, wired to post_encode_callback.
void EncodedFrameCallback(const EncodedFrame& encoded_frame) override;
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();
rtc::VideoSinkInterface<VideoFrame>* pre_encode_proxy();
void StartMeasuringCpuProcessTime();
void StopMeasuringCpuProcessTime();
void StartExcludingCpuThreadTime();
void StopExcludingCpuThreadTime();
double GetCpuUsagePercent();
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;
};
// This class receives the send-side OnFrame callback and is provided to not
// conflict with the receiver-side renderer callback.
class PreEncodeProxy : public rtc::VideoSinkInterface<VideoFrame> {
public:
explicit PreEncodeProxy(VideoAnalyzer* parent);
void OnFrame(const VideoFrame& video_frame) override;
private:
VideoAnalyzer* const parent_;
};
// 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:
explicit CapturedFrameForwarder(VideoAnalyzer* analyzer, Clock* clock);
void SetSource(test::VideoCapturer* video_capturer);
private:
void OnFrame(const VideoFrame& video_frame) override;
// Called when |send_stream_.SetSource()| is called.
void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override;
// Called by |send_stream_| when |send_stream_.SetSource()| is called.
void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override;
VideoAnalyzer* const analyzer_;
rtc::CriticalSection crit_;
rtc::VideoSinkInterface<VideoFrame>* send_stream_input_
RTC_GUARDED_BY(crit_);
test::VideoCapturer* video_capturer_;
Clock* clock_;
};
struct FrameWithPsnr {
double psnr;
VideoFrame frame;
};
bool IsInSelectedSpatialAndTemporalLayer(const uint8_t* packet,
size_t length,
const RTPHeader& header);
void AddFrameComparison(const VideoFrame& reference,
const VideoFrame& render,
bool dropped,
int64_t render_time_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
static void PollStatsThread(void* obj);
void PollStats();
static bool FrameComparisonThread(void* obj);
bool CompareFrames();
bool PopComparison(FrameComparison* comparison);
// Increment counter for number of frames received for comparison.
void FrameRecorded();
// Returns true if all frames to be compared have been taken from the queue.
bool AllFramesRecorded();
// Increase count of number of frames processed. Returns true if this was the
// last frame to be processed.
bool FrameProcessed();
void PrintResults();
void PerformFrameComparison(const FrameComparison& comparison);
void PrintResult(const char* result_type,
test::Statistics stats,
const char* unit);
void PrintSamplesToFile(void);
double GetAverageMediaBitrateBps();
void AddCapturedFrameForComparison(const VideoFrame& video_frame);
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_;
PreEncodeProxy pre_encode_proxy_;
rtc::CriticalSection comparison_lock_;
std::vector<Sample> samples_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics sender_time_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics receiver_time_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics network_time_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics psnr_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics ssim_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics end_to_end_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics rendered_delta_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics encoded_frame_size_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics encode_frame_rate_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics encode_time_ms_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics encode_usage_percent_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics decode_time_ms_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics decode_time_max_ms_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics media_bitrate_bps_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics fec_bitrate_bps_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics send_bandwidth_bps_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics memory_usage_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics time_between_freezes_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics audio_expand_rate_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics audio_accelerate_rate_ RTC_GUARDED_BY(comparison_lock_);
test::Statistics audio_jitter_buffer_ms_ 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_);
size_t last_fec_bytes_;
const int frames_to_process_;
int frames_recorded_;
int frames_processed_;
int dropped_frames_;
int dropped_frames_before_first_encode_;
int dropped_frames_before_rendering_;
int64_t last_render_time_;
int64_t last_render_delta_ms_;
int64_t last_unfreeze_time_ms_;
uint32_t rtp_timestamp_delta_;
int64_t total_media_bytes_;
int64_t first_sending_time_;
int64_t last_sending_time_;
rtc::CriticalSection cpu_measurement_lock_;
int64_t cpu_time_ RTC_GUARDED_BY(cpu_measurement_lock_);
int64_t wallclock_time_ RTC_GUARDED_BY(cpu_measurement_lock_);
rtc::CriticalSection crit_;
std::deque<VideoFrame> frames_ RTC_GUARDED_BY(crit_);
absl::optional<VideoFrame> last_rendered_frame_ RTC_GUARDED_BY(crit_);
rtc::TimestampWrapAroundHandler wrap_handler_ RTC_GUARDED_BY(crit_);
std::map<int64_t, int64_t> send_times_ RTC_GUARDED_BY(crit_);
std::map<int64_t, int64_t> recv_times_ RTC_GUARDED_BY(crit_);
std::map<int64_t, size_t> encoded_frame_sizes_ RTC_GUARDED_BY(crit_);
absl::optional<uint32_t> first_encoded_timestamp_ RTC_GUARDED_BY(crit_);
absl::optional<uint32_t> first_sent_timestamp_ RTC_GUARDED_BY(crit_);
const double avg_psnr_threshold_;
const double avg_ssim_threshold_;
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_;
std::unique_ptr<test::RtpFileWriter> rtp_file_writer_;
Clock* const clock_;
const int64_t start_ms_;
};
} // namespace webrtc
#endif // VIDEO_VIDEO_ANALYZER_H_