webrtc_m130/video/video_quality_observer.h
Niels Möller 9a9f18a736 Get WebRTC.Video.ReceivedPacketsLostInPercent from ReceiveStatistics
Old way to produce this histogram was based on RtcpStatisticsCallback
reporting sent RTCP messages, with some additional processing by the
ReportBlockStats class. After this cl, to grand average fraction loss
is computed by StreamStatistician, queried by VideoReceiveStream when
the stream is closed down, and passed to ReceiveStatisticsProxy which
produces histograms.

This is a preparation for deleting the RtcpStatisticsCallback from
ReceiveStatistics.

Bug: webrtc:10679
Change-Id: Ie37062c1ae590fd92d3bd0f94c510e135ab93e8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147722
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28747}
2019-08-02 12:38:34 +00:00

100 lines
3.1 KiB
C++

/*
* Copyright (c) 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_QUALITY_OBSERVER_H_
#define VIDEO_VIDEO_QUALITY_OBSERVER_H_
#include <stdint.h>
#include <set>
#include <vector>
#include "absl/types/optional.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
#include "rtc_base/numerics/moving_average.h"
#include "rtc_base/numerics/sample_counter.h"
namespace webrtc {
// Calculates spatial and temporal quality metrics and reports them to UMA
// stats.
class VideoQualityObserver {
public:
// Use either VideoQualityObserver::kBlockyQpThresholdVp8 or
// VideoQualityObserver::kBlockyQpThresholdVp9.
explicit VideoQualityObserver(VideoContentType content_type);
~VideoQualityObserver() = default;
void OnDecodedFrame(const VideoFrame& frame,
absl::optional<uint8_t> qp,
VideoCodecType codec);
void OnRenderedFrame(const VideoFrame& frame, int64_t now_ms);
void OnStreamInactive();
uint32_t NumFreezes() const;
uint32_t NumPauses() const;
uint32_t TotalFreezesDurationMs() const;
uint32_t TotalPausesDurationMs() const;
uint32_t TotalFramesDurationMs() const;
double SumSquaredFrameDurationsSec() const;
void UpdateHistograms();
static const uint32_t kMinFrameSamplesToDetectFreeze;
static const uint32_t kMinIncreaseForFreezeMs;
static const uint32_t kAvgInterframeDelaysWindowSizeFrames;
private:
enum Resolution {
Low = 0,
Medium = 1,
High = 2,
};
int64_t last_frame_rendered_ms_;
int64_t num_frames_rendered_;
int64_t first_frame_rendered_ms_;
int64_t last_frame_pixels_;
bool is_last_frame_blocky_;
// Decoded timestamp of the last delayed frame.
int64_t last_unfreeze_time_ms_;
rtc::MovingAverage render_interframe_delays_;
double sum_squared_interframe_delays_secs_;
// An inter-frame delay is counted as a freeze if it's significantly longer
// than average inter-frame delay.
rtc::SampleCounter freezes_durations_;
rtc::SampleCounter pauses_durations_;
// Time between freezes.
rtc::SampleCounter smooth_playback_durations_;
// Counters for time spent in different resolutions. Time between each two
// Consecutive frames is counted to bin corresponding to the first frame
// resolution.
std::vector<int64_t> time_in_resolution_ms_;
// Resolution of the last decoded frame. Resolution enum is used as an index.
Resolution current_resolution_;
int num_resolution_downgrades_;
// Similar to resolution, time spent in high-QP video.
int64_t time_in_blocky_video_ms_;
// Content type of the last decoded frame.
VideoContentType content_type_;
bool is_paused_;
// Set of decoded frames with high QP value.
std::set<int64_t> blocky_frames_;
};
} // namespace webrtc
#endif // VIDEO_VIDEO_QUALITY_OBSERVER_H_