Repo created

TMessagesProj/jni/voip/webrtc/video/encoder_overshoot_detector.h

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+/*
+ *  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"
+#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.
+  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;
+  };
+  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_