/* * 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 #include #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(frame_size_bits) / ideal_frame_size_bits); } else { utilization_factor = 1.0 + (static_cast(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 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 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( (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(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( -max_underrun_seconds * target_bitrate_.bps(), 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: case VideoCodecType::kVideoCodecMultiplex: break; } } } // namespace webrtc