video/encoder_overshoot_detector.cc - src/ - Git at Google

 /*
  *  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.
  */

 #include "video/encoder_overshoot_detector.h"

 #include <algorithm>
 #include <string>

 #include "system_wrappers/include/metrics.h"

 namespace webrtc {
 namespace {
 // The buffer level for media-rate utilization is allowed to go below zero,
 // down to
 // -(`kMaxMediaUnderrunFrames` / `target_framerate_fps_`) * `target_bitrate_`.
 static constexpr double kMaxMediaUnderrunFrames = 5.0;
 }  // namespace

 EncoderOvershootDetector::EncoderOvershootDetector(int64_t window_size_ms,
                                                    VideoCodecType codec,
                                                    bool is_screenshare)
     : window_size_ms_(window_size_ms),
       time_last_update_ms_(-1),
       sum_network_utilization_factors_(0.0),
       sum_media_utilization_factors_(0.0),
       target_bitrate_(DataRate::Zero()),
       target_framerate_fps_(0),
       network_buffer_level_bits_(0),
       media_buffer_level_bits_(0),
       codec_(codec),
       is_screenshare_(is_screenshare),
       frame_count_(0),
       sum_diff_kbps_squared_(0),
       sum_overshoot_percent_(0) {}

 EncoderOvershootDetector::~EncoderOvershootDetector() {
   UpdateHistograms();
 }

 void EncoderOvershootDetector::SetTargetRate(DataRate target_bitrate,
                                              double target_framerate_fps,
                                              int64_t time_ms) {
   // First leak bits according to the previous target rate.
   if (target_bitrate_ != DataRate::Zero()) {
     LeakBits(time_ms);
   } else if (target_bitrate != DataRate::Zero()) {
     // Stream was just enabled, reset state.
     time_last_update_ms_ = time_ms;
     utilization_factors_.clear();
     sum_network_utilization_factors_ = 0.0;
     sum_media_utilization_factors_ = 0.0;
     network_buffer_level_bits_ = 0;
     media_buffer_level_bits_ = 0;
   }

   target_bitrate_ = target_bitrate;
   target_framerate_fps_ = target_framerate_fps;
 }

 void EncoderOvershootDetector::OnEncodedFrame(size_t bytes, int64_t time_ms) {
   // Leak bits from the virtual pacer buffer, according to the current target
   // bitrate.
   LeakBits(time_ms);

   const int64_t frame_size_bits = bytes * 8;
   // Ideal size of a frame given the current rates.
   const int64_t ideal_frame_size_bits = IdealFrameSizeBits();
   if (ideal_frame_size_bits == 0) {
     // Frame without updated bitrate and/or framerate, ignore it.
     return;
   }

   const double network_utilization_factor =
       HandleEncodedFrame(frame_size_bits, ideal_frame_size_bits, time_ms,
                          &network_buffer_level_bits_);
   const double media_utilization_factor =
       HandleEncodedFrame(frame_size_bits, ideal_frame_size_bits, time_ms,
                          &media_buffer_level_bits_);

   sum_network_utilization_factors_ += network_utilization_factor;
   sum_media_utilization_factors_ += media_utilization_factor;

   // Calculate the bitrate diff in kbps
   int64_t diff_kbits = (frame_size_bits - ideal_frame_size_bits) / 1000;
   sum_diff_kbps_squared_ += diff_kbits * diff_kbits;
   sum_overshoot_percent_ += diff_kbits * 100 * 1000 / ideal_frame_size_bits;
   ++frame_count_;

   utilization_factors_.emplace_back(network_utilization_factor,
                                     media_utilization_factor, time_ms);
 }

 double EncoderOvershootDetector::HandleEncodedFrame(
     size_t frame_size_bits,
     int64_t ideal_frame_size_bits,
     int64_t time_ms,
     int64_t* buffer_level_bits) const {
   // Add new frame to the buffer level. If doing so exceeds the ideal buffer
   // size, penalize this frame but cap overshoot to current buffer level rather
   // than size of this frame. This is done so that a single large frame is not
   // penalized if the encoder afterwards compensates by dropping frames and/or
   // reducing frame size. If however a large frame is followed by more data,
   // we cannot pace that next frame out within one frame space.
   const int64_t bitsum = frame_size_bits + *buffer_level_bits;
   int64_t overshoot_bits = 0;
   if (bitsum > ideal_frame_size_bits) {
     overshoot_bits =
         std::min(*buffer_level_bits, bitsum - ideal_frame_size_bits);
   }

   // Add entry for the (over) utilization for this frame. Factor is capped
   // at 1.0 so that we don't risk overshooting on sudden changes.
   double utilization_factor;
   if (utilization_factors_.empty()) {
     // First frame, cannot estimate overshoot based on previous one so
     // for this particular frame, just like as size vs optimal size.
     utilization_factor = std::max(
         1.0, static_cast<double>(frame_size_bits) / ideal_frame_size_bits);
   } else {
     utilization_factor =
         1.0 + (static_cast<double>(overshoot_bits) / ideal_frame_size_bits);
   }

   // Remove the overshot bits from the virtual buffer so we don't penalize
   // those bits multiple times.
   *buffer_level_bits -= overshoot_bits;
   *buffer_level_bits += frame_size_bits;

   return utilization_factor;
 }

 absl::optional<double>
 EncoderOvershootDetector::GetNetworkRateUtilizationFactor(int64_t time_ms) {
   CullOldUpdates(time_ms);

   // No data points within window, return.
   if (utilization_factors_.empty()) {
     return absl::nullopt;
   }

   // TODO(sprang): Consider changing from arithmetic mean to some other
   // function such as 90th percentile.
   return sum_network_utilization_factors_ / utilization_factors_.size();
 }

 absl::optional<double> EncoderOvershootDetector::GetMediaRateUtilizationFactor(
     int64_t time_ms) {
   CullOldUpdates(time_ms);

   // No data points within window, return.
   if (utilization_factors_.empty()) {
     return absl::nullopt;
   }

   return sum_media_utilization_factors_ / utilization_factors_.size();
 }

 void EncoderOvershootDetector::Reset() {
   UpdateHistograms();
   sum_diff_kbps_squared_ = 0;
   frame_count_ = 0;
   sum_overshoot_percent_ = 0;
   time_last_update_ms_ = -1;
   utilization_factors_.clear();
   target_bitrate_ = DataRate::Zero();
   sum_network_utilization_factors_ = 0.0;
   sum_media_utilization_factors_ = 0.0;
   target_framerate_fps_ = 0.0;
   network_buffer_level_bits_ = 0;
   media_buffer_level_bits_ = 0;
 }

 int64_t EncoderOvershootDetector::IdealFrameSizeBits() const {
   if (target_framerate_fps_ <= 0 || target_bitrate_ == DataRate::Zero()) {
     return 0;
   }

   // Current ideal frame size, based on the current target bitrate.
   return static_cast<int64_t>(
       (target_bitrate_.bps() + target_framerate_fps_ / 2) /
       target_framerate_fps_);
 }

 void EncoderOvershootDetector::LeakBits(int64_t time_ms) {
   if (time_last_update_ms_ != -1 && target_bitrate_ > DataRate::Zero()) {
     int64_t time_delta_ms = time_ms - time_last_update_ms_;
     // Leak bits according to the current target bitrate.
     const int64_t leaked_bits = (target_bitrate_.bps() * time_delta_ms) / 1000;

     // Network buffer may not go below zero.
     network_buffer_level_bits_ =
         std::max<int64_t>(0, network_buffer_level_bits_ - leaked_bits);

     // Media buffer my go down to minus `kMaxMediaUnderrunFrames` frames worth
     // of data.
     const double max_underrun_seconds =
         std::min(kMaxMediaUnderrunFrames, target_framerate_fps_) /
         target_framerate_fps_;
     media_buffer_level_bits_ = std::max<int64_t>(
         -max_underrun_seconds * target_bitrate_.bps<int64_t>(),
         media_buffer_level_bits_ - leaked_bits);
   }
   time_last_update_ms_ = time_ms;
 }

 void EncoderOvershootDetector::CullOldUpdates(int64_t time_ms) {
   // Cull old data points.
   const int64_t cutoff_time_ms = time_ms - window_size_ms_;
   while (!utilization_factors_.empty() &&
          utilization_factors_.front().update_time_ms < cutoff_time_ms) {
     // Make sure sum is never allowed to become negative due rounding errors.
     sum_network_utilization_factors_ = std::max(
         0.0, sum_network_utilization_factors_ -
                  utilization_factors_.front().network_utilization_factor);
     sum_media_utilization_factors_ = std::max(
         0.0, sum_media_utilization_factors_ -
                  utilization_factors_.front().media_utilization_factor);
     utilization_factors_.pop_front();
   }
 }

 void EncoderOvershootDetector::UpdateHistograms() {
   if (frame_count_ == 0)
     return;

   int64_t bitrate_rmse = std::sqrt(sum_diff_kbps_squared_ / frame_count_);
   int64_t average_overshoot_percent = sum_overshoot_percent_ / frame_count_;
   const std::string rmse_histogram_prefix =
       is_screenshare_ ? "WebRTC.Video.Screenshare.RMSEOfEncodingBitrateInKbps."
                       : "WebRTC.Video.RMSEOfEncodingBitrateInKbps.";
   const std::string overshoot_histogram_prefix =
       is_screenshare_ ? "WebRTC.Video.Screenshare.EncodingBitrateOvershoot."
                       : "WebRTC.Video.EncodingBitrateOvershoot.";
   // index = 1 represents screensharing histograms recording.
   // index = 0 represents normal video histograms recording.
   const int index = is_screenshare_ ? 1 : 0;
   switch (codec_) {
     case VideoCodecType::kVideoCodecAV1:
       RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "Av1",
                                   bitrate_rmse);
       RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "Av1",
                                   average_overshoot_percent);
       break;
     case VideoCodecType::kVideoCodecVP9:
       RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "Vp9",
                                   bitrate_rmse);
       RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "Vp9",
                                   average_overshoot_percent);
       break;
     case VideoCodecType::kVideoCodecVP8:
       RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "Vp8",
                                   bitrate_rmse);
       RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "Vp8",
                                   average_overshoot_percent);
       break;
     case VideoCodecType::kVideoCodecH264:
       RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "H264",
                                   bitrate_rmse);
       RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "H264",
                                   average_overshoot_percent);
       break;
     case VideoCodecType::kVideoCodecH265:
       RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "H265",
                                   bitrate_rmse);
       RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "H265",
                                   average_overshoot_percent);
       break;
     case VideoCodecType::kVideoCodecGeneric:
       break;
   }
 }

 }  // namespace webrtc
	/*
	* 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.
	*/

	#include "video/encoder_overshoot_detector.h"

	#include <algorithm>
	#include <string>

	#include "system_wrappers/include/metrics.h"

	namespace webrtc {
	namespace {
	// The buffer level for media-rate utilization is allowed to go below zero,
	// down to
	// -(`kMaxMediaUnderrunFrames` / `target_framerate_fps_`) * `target_bitrate_`.
	static constexpr double kMaxMediaUnderrunFrames = 5.0;
	} // namespace

	EncoderOvershootDetector::EncoderOvershootDetector(int64_t window_size_ms,
	VideoCodecType codec,
	bool is_screenshare)
	: window_size_ms_(window_size_ms),
	time_last_update_ms_(-1),
	sum_network_utilization_factors_(0.0),
	sum_media_utilization_factors_(0.0),
	target_bitrate_(DataRate::Zero()),
	target_framerate_fps_(0),
	network_buffer_level_bits_(0),
	media_buffer_level_bits_(0),
	codec_(codec),
	is_screenshare_(is_screenshare),
	frame_count_(0),
	sum_diff_kbps_squared_(0),
	sum_overshoot_percent_(0) {}

	EncoderOvershootDetector::~EncoderOvershootDetector() {
	UpdateHistograms();
	}

	void EncoderOvershootDetector::SetTargetRate(DataRate target_bitrate,
	double target_framerate_fps,
	int64_t time_ms) {
	// First leak bits according to the previous target rate.
	if (target_bitrate_ != DataRate::Zero()) {
	LeakBits(time_ms);
	} else if (target_bitrate != DataRate::Zero()) {
	// Stream was just enabled, reset state.
	time_last_update_ms_ = time_ms;
	utilization_factors_.clear();
	sum_network_utilization_factors_ = 0.0;
	sum_media_utilization_factors_ = 0.0;
	network_buffer_level_bits_ = 0;
	media_buffer_level_bits_ = 0;
	}

	target_bitrate_ = target_bitrate;
	target_framerate_fps_ = target_framerate_fps;
	}

	void EncoderOvershootDetector::OnEncodedFrame(size_t bytes, int64_t time_ms) {
	// Leak bits from the virtual pacer buffer, according to the current target
	// bitrate.
	LeakBits(time_ms);

	const int64_t frame_size_bits = bytes * 8;
	// Ideal size of a frame given the current rates.
	const int64_t ideal_frame_size_bits = IdealFrameSizeBits();
	if (ideal_frame_size_bits == 0) {
	// Frame without updated bitrate and/or framerate, ignore it.
	return;
	}

	const double network_utilization_factor =
	HandleEncodedFrame(frame_size_bits, ideal_frame_size_bits, time_ms,
	&network_buffer_level_bits_);
	const double media_utilization_factor =
	HandleEncodedFrame(frame_size_bits, ideal_frame_size_bits, time_ms,
	&media_buffer_level_bits_);

	sum_network_utilization_factors_ += network_utilization_factor;
	sum_media_utilization_factors_ += media_utilization_factor;

	// Calculate the bitrate diff in kbps
	int64_t diff_kbits = (frame_size_bits - ideal_frame_size_bits) / 1000;
	sum_diff_kbps_squared_ += diff_kbits * diff_kbits;
	sum_overshoot_percent_ += diff_kbits * 100 * 1000 / ideal_frame_size_bits;
	++frame_count_;

	utilization_factors_.emplace_back(network_utilization_factor,
	media_utilization_factor, time_ms);
	}

	double EncoderOvershootDetector::HandleEncodedFrame(
	size_t frame_size_bits,
	int64_t ideal_frame_size_bits,
	int64_t time_ms,
	int64_t* buffer_level_bits) const {
	// Add new frame to the buffer level. If doing so exceeds the ideal buffer
	// size, penalize this frame but cap overshoot to current buffer level rather
	// than size of this frame. This is done so that a single large frame is not
	// penalized if the encoder afterwards compensates by dropping frames and/or
	// reducing frame size. If however a large frame is followed by more data,
	// we cannot pace that next frame out within one frame space.
	const int64_t bitsum = frame_size_bits + *buffer_level_bits;
	int64_t overshoot_bits = 0;
	if (bitsum > ideal_frame_size_bits) {
	overshoot_bits =
	std::min(*buffer_level_bits, bitsum - ideal_frame_size_bits);
	}

	// Add entry for the (over) utilization for this frame. Factor is capped
	// at 1.0 so that we don't risk overshooting on sudden changes.
	double utilization_factor;
	if (utilization_factors_.empty()) {
	// First frame, cannot estimate overshoot based on previous one so
	// for this particular frame, just like as size vs optimal size.
	utilization_factor = std::max(
	1.0, static_cast<double>(frame_size_bits) / ideal_frame_size_bits);
	} else {
	utilization_factor =
	1.0 + (static_cast<double>(overshoot_bits) / ideal_frame_size_bits);
	}

	// Remove the overshot bits from the virtual buffer so we don't penalize
	// those bits multiple times.
	*buffer_level_bits -= overshoot_bits;
	*buffer_level_bits += frame_size_bits;

	return utilization_factor;
	}

	absl::optional<double>
	EncoderOvershootDetector::GetNetworkRateUtilizationFactor(int64_t time_ms) {
	CullOldUpdates(time_ms);

	// No data points within window, return.
	if (utilization_factors_.empty()) {
	return absl::nullopt;
	}

	// TODO(sprang): Consider changing from arithmetic mean to some other
	// function such as 90th percentile.
	return sum_network_utilization_factors_ / utilization_factors_.size();
	}

	absl::optional<double> EncoderOvershootDetector::GetMediaRateUtilizationFactor(
	int64_t time_ms) {
	CullOldUpdates(time_ms);

	// No data points within window, return.
	if (utilization_factors_.empty()) {
	return absl::nullopt;
	}

	return sum_media_utilization_factors_ / utilization_factors_.size();
	}

	void EncoderOvershootDetector::Reset() {
	UpdateHistograms();
	sum_diff_kbps_squared_ = 0;
	frame_count_ = 0;
	sum_overshoot_percent_ = 0;
	time_last_update_ms_ = -1;
	utilization_factors_.clear();
	target_bitrate_ = DataRate::Zero();
	sum_network_utilization_factors_ = 0.0;
	sum_media_utilization_factors_ = 0.0;
	target_framerate_fps_ = 0.0;
	network_buffer_level_bits_ = 0;
	media_buffer_level_bits_ = 0;
	}

	int64_t EncoderOvershootDetector::IdealFrameSizeBits() const {
	if (target_framerate_fps_ <= 0 \|\| target_bitrate_ == DataRate::Zero()) {
	return 0;
	}

	// Current ideal frame size, based on the current target bitrate.
	return static_cast<int64_t>(
	(target_bitrate_.bps() + target_framerate_fps_ / 2) /
	target_framerate_fps_);
	}

	void EncoderOvershootDetector::LeakBits(int64_t time_ms) {
	if (time_last_update_ms_ != -1 && target_bitrate_ > DataRate::Zero()) {
	int64_t time_delta_ms = time_ms - time_last_update_ms_;
	// Leak bits according to the current target bitrate.
	const int64_t leaked_bits = (target_bitrate_.bps() * time_delta_ms) / 1000;

	// Network buffer may not go below zero.
	network_buffer_level_bits_ =
	std::max<int64_t>(0, network_buffer_level_bits_ - leaked_bits);

	// Media buffer my go down to minus `kMaxMediaUnderrunFrames` frames worth
	// of data.
	const double max_underrun_seconds =
	std::min(kMaxMediaUnderrunFrames, target_framerate_fps_) /
	target_framerate_fps_;
	media_buffer_level_bits_ = std::max<int64_t>(
	-max_underrun_seconds * target_bitrate_.bps<int64_t>(),
	media_buffer_level_bits_ - leaked_bits);
	}
	time_last_update_ms_ = time_ms;
	}

	void EncoderOvershootDetector::CullOldUpdates(int64_t time_ms) {
	// Cull old data points.
	const int64_t cutoff_time_ms = time_ms - window_size_ms_;
	while (!utilization_factors_.empty() &&
	utilization_factors_.front().update_time_ms < cutoff_time_ms) {
	// Make sure sum is never allowed to become negative due rounding errors.
	sum_network_utilization_factors_ = std::max(
	0.0, sum_network_utilization_factors_ -
	utilization_factors_.front().network_utilization_factor);
	sum_media_utilization_factors_ = std::max(
	0.0, sum_media_utilization_factors_ -
	utilization_factors_.front().media_utilization_factor);
	utilization_factors_.pop_front();
	}
	}

	void EncoderOvershootDetector::UpdateHistograms() {
	if (frame_count_ == 0)
	return;

	int64_t bitrate_rmse = std::sqrt(sum_diff_kbps_squared_ / frame_count_);
	int64_t average_overshoot_percent = sum_overshoot_percent_ / frame_count_;
	const std::string rmse_histogram_prefix =
	is_screenshare_ ? "WebRTC.Video.Screenshare.RMSEOfEncodingBitrateInKbps."
	: "WebRTC.Video.RMSEOfEncodingBitrateInKbps.";
	const std::string overshoot_histogram_prefix =
	is_screenshare_ ? "WebRTC.Video.Screenshare.EncodingBitrateOvershoot."
	: "WebRTC.Video.EncodingBitrateOvershoot.";
	// index = 1 represents screensharing histograms recording.
	// index = 0 represents normal video histograms recording.
	const int index = is_screenshare_ ? 1 : 0;
	switch (codec_) {
	case VideoCodecType::kVideoCodecAV1:
	RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "Av1",
	bitrate_rmse);
	RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "Av1",
	average_overshoot_percent);
	break;
	case VideoCodecType::kVideoCodecVP9:
	RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "Vp9",
	bitrate_rmse);
	RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "Vp9",
	average_overshoot_percent);
	break;
	case VideoCodecType::kVideoCodecVP8:
	RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "Vp8",
	bitrate_rmse);
	RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "Vp8",
	average_overshoot_percent);
	break;
	case VideoCodecType::kVideoCodecH264:
	RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "H264",
	bitrate_rmse);
	RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "H264",
	average_overshoot_percent);
	break;
	case VideoCodecType::kVideoCodecH265:
	RTC_HISTOGRAMS_COUNTS_10000(index, rmse_histogram_prefix + "H265",
	bitrate_rmse);
	RTC_HISTOGRAMS_COUNTS_10000(index, overshoot_histogram_prefix + "H265",
	average_overshoot_percent);
	break;
	case VideoCodecType::kVideoCodecGeneric:
	break;
	}
	}

	} // namespace webrtc