3be2a55e7f
This is a reland of 1880c7162bd3637c433f9421c798808cd6eacaf7 Original change's description: > Updated analysis in videoprocessor. > > - Run analysis after all frames are processed. Before part of it was > done at bitrate change points; > - Analysis is done for whole stream as well as for each rate update > interval; > - Changed units from number of frames to time units for some metrics > and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to > 'time to reach target bitrate, sec'; > - Changed data type of FrameStatistic::max_nalu_length (renamed to > max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to > use such advanced data type in such low level data structure. > > Bug: webrtc:8524 > Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f > Reviewed-on: https://webrtc-review.googlesource.com/31901 > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21653} TBR=brandtr@webrtc.org, stefan@webrtc.org Bug: webrtc:8524 Change-Id: Ie0ad7790689422ffa61da294967fc492a13b75ae Reviewed-on: https://webrtc-review.googlesource.com/40202 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21668}
51 lines
1.4 KiB
C++
51 lines
1.4 KiB
C++
/*
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* Copyright (c) 2012 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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#include "modules/video_coding/codecs/test/stats.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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std::string FrameStatistic::ToString() const {
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std::stringstream ss;
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ss << "frame " << frame_number;
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ss << " " << decoded_width << "x" << decoded_height;
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ss << " sl " << simulcast_svc_idx;
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ss << " tl " << temporal_layer_idx;
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ss << " type " << frame_type;
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ss << " length " << encoded_frame_size_bytes;
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ss << " qp " << qp;
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ss << " psnr " << psnr;
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ss << " ssim " << ssim;
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ss << " enc_time_us " << encode_time_us;
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ss << " dec_time_us " << decode_time_us;
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ss << " rtp_ts " << rtp_timestamp;
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ss << " bitrate_kbps " << target_bitrate_kbps;
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return ss.str();
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}
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FrameStatistic* Stats::AddFrame() {
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stats_.emplace_back(stats_.size());
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return &stats_.back();
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}
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FrameStatistic* Stats::GetFrame(size_t frame_number) {
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RTC_CHECK_LT(frame_number, stats_.size());
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return &stats_[frame_number];
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}
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size_t Stats::size() const {
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return stats_.size();
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}
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} // namespace test
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} // namespace webrtc
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