| /* |
| * 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_ |