| /* |
| * 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: |
| 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::TestVideoCapturer* 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); |
| 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(); |
| 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; |
| }; |
| |
| // 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::TestVideoCapturer* 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::TestVideoCapturer* 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_; |
| |
| 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_ |