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8037fc6ffa
Bug: webrtc:342905193 No-Try: True Change-Id: Icc968be43b8830038ea9a1f5f604307220457807 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021 Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42911}
87 lines
3.3 KiB
C++
87 lines
3.3 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef VIDEO_ENCODER_OVERSHOOT_DETECTOR_H_
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#define VIDEO_ENCODER_OVERSHOOT_DETECTOR_H_
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#include <deque>
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#include <optional>
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#include "api/units/data_rate.h"
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#include "api/video_codecs/video_codec.h"
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namespace webrtc {
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class EncoderOvershootDetector {
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public:
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explicit EncoderOvershootDetector(int64_t window_size_ms,
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VideoCodecType codec,
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bool is_screenshare);
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~EncoderOvershootDetector();
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void SetTargetRate(DataRate target_bitrate,
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double target_framerate_fps,
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int64_t time_ms);
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// A frame has been encoded or dropped. `bytes` == 0 indicates a drop.
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void OnEncodedFrame(size_t bytes, int64_t time_ms);
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// This utilization factor reaches 1.0 only if the encoder produces encoded
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// frame in such a way that they can be sent onto the network at
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// `target_bitrate` without building growing queues.
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std::optional<double> GetNetworkRateUtilizationFactor(int64_t time_ms);
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// This utilization factor is based just on actual encoded frame sizes in
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// relation to ideal sizes. An undershoot may be compensated by an
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// overshoot so that the average over time is close to `target_bitrate`.
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std::optional<double> GetMediaRateUtilizationFactor(int64_t time_ms);
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void Reset();
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private:
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int64_t IdealFrameSizeBits() const;
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void LeakBits(int64_t time_ms);
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void CullOldUpdates(int64_t time_ms);
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// Updates provided buffer and checks if overuse ensues, returns
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// the calculated utilization factor for this frame.
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double HandleEncodedFrame(size_t frame_size_bits,
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int64_t ideal_frame_size_bits,
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int64_t time_ms,
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int64_t* buffer_level_bits) const;
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const int64_t window_size_ms_;
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int64_t time_last_update_ms_;
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struct BitrateUpdate {
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BitrateUpdate(double network_utilization_factor,
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double media_utilization_factor,
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int64_t update_time_ms)
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: network_utilization_factor(network_utilization_factor),
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media_utilization_factor(media_utilization_factor),
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update_time_ms(update_time_ms) {}
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// The utilization factor based on strict network rate.
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double network_utilization_factor;
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// The utilization based on average media rate.
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double media_utilization_factor;
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int64_t update_time_ms;
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};
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void UpdateHistograms();
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std::deque<BitrateUpdate> utilization_factors_;
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double sum_network_utilization_factors_;
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double sum_media_utilization_factors_;
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DataRate target_bitrate_;
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double target_framerate_fps_;
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int64_t network_buffer_level_bits_;
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int64_t media_buffer_level_bits_;
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VideoCodecType codec_;
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bool is_screenshare_;
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int64_t frame_count_;
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int64_t sum_diff_kbps_squared_;
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int64_t sum_overshoot_percent_;
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};
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} // namespace webrtc
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#endif // VIDEO_ENCODER_OVERSHOOT_DETECTOR_H_
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