| /* |
| * 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; |
| } |
| |
| std::optional<double> EncoderOvershootDetector::GetNetworkRateUtilizationFactor( |
| int64_t time_ms) { |
| CullOldUpdates(time_ms); |
| |
| // No data points within window, return. |
| if (utilization_factors_.empty()) { |
| return std::nullopt; |
| } |
| |
| // TODO(sprang): Consider changing from arithmetic mean to some other |
| // function such as 90th percentile. |
| return sum_network_utilization_factors_ / utilization_factors_.size(); |
| } |
| |
| std::optional<double> EncoderOvershootDetector::GetMediaRateUtilizationFactor( |
| int64_t time_ms) { |
| CullOldUpdates(time_ms); |
| |
| // No data points within window, return. |
| if (utilization_factors_.empty()) { |
| return std::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 |