blob: e78b601d1dbb079e0a41c3b68d38bc5cff896ab2 [file] [log] [blame]
/*
* Copyright (c) 2019 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_ENCODER_OVERSHOOT_DETECTOR_H_
#define VIDEO_ENCODER_OVERSHOOT_DETECTOR_H_
#include <deque>
#include <optional>
#include "api/units/data_rate.h"
#include "api/video_codecs/video_codec.h"
namespace webrtc {
class EncoderOvershootDetector {
public:
explicit EncoderOvershootDetector(int64_t window_size_ms,
VideoCodecType codec,
bool is_screenshare);
~EncoderOvershootDetector();
void SetTargetRate(DataRate target_bitrate,
double target_framerate_fps,
int64_t time_ms);
// A frame has been encoded or dropped. `bytes` == 0 indicates a drop.
void OnEncodedFrame(size_t bytes, int64_t time_ms);
// This utilization factor reaches 1.0 only if the encoder produces encoded
// frame in such a way that they can be sent onto the network at
// `target_bitrate` without building growing queues.
std::optional<double> GetNetworkRateUtilizationFactor(int64_t time_ms);
// This utilization factor is based just on actual encoded frame sizes in
// relation to ideal sizes. An undershoot may be compensated by an
// overshoot so that the average over time is close to `target_bitrate`.
std::optional<double> GetMediaRateUtilizationFactor(int64_t time_ms);
void Reset();
private:
int64_t IdealFrameSizeBits() const;
void LeakBits(int64_t time_ms);
void CullOldUpdates(int64_t time_ms);
// Updates provided buffer and checks if overuse ensues, returns
// the calculated utilization factor for this frame.
double HandleEncodedFrame(size_t frame_size_bits,
int64_t ideal_frame_size_bits,
int64_t time_ms,
int64_t* buffer_level_bits) const;
const int64_t window_size_ms_;
int64_t time_last_update_ms_;
struct BitrateUpdate {
BitrateUpdate(double network_utilization_factor,
double media_utilization_factor,
int64_t update_time_ms)
: network_utilization_factor(network_utilization_factor),
media_utilization_factor(media_utilization_factor),
update_time_ms(update_time_ms) {}
// The utilization factor based on strict network rate.
double network_utilization_factor;
// The utilization based on average media rate.
double media_utilization_factor;
int64_t update_time_ms;
};
void UpdateHistograms();
std::deque<BitrateUpdate> utilization_factors_;
double sum_network_utilization_factors_;
double sum_media_utilization_factors_;
DataRate target_bitrate_;
double target_framerate_fps_;
int64_t network_buffer_level_bits_;
int64_t media_buffer_level_bits_;
VideoCodecType codec_;
bool is_screenshare_;
int64_t frame_count_;
int64_t sum_diff_kbps_squared_;
int64_t sum_overshoot_percent_;
};
} // namespace webrtc
#endif // VIDEO_ENCODER_OVERSHOOT_DETECTOR_H_