79 lines
3.0 KiB
C++
79 lines
3.0 KiB
C++
/*
|
|
* 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 "absl/types/optional.h"
|
|
#include "api/units/data_rate.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class EncoderOvershootDetector {
|
|
public:
|
|
explicit EncoderOvershootDetector(int64_t window_size_ms);
|
|
~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.
|
|
absl::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|.
|
|
absl::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;
|
|
};
|
|
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_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // VIDEO_ENCODER_OVERSHOOT_DETECTOR_H_
|