1880c7162b
- 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}
90 lines
2.2 KiB
C++
90 lines
2.2 KiB
C++
/*
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* Copyright (c) 2011 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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#ifndef MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
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#define MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
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#include <string>
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#include <vector>
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#include "common_types.h" // NOLINT(build/include)
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namespace webrtc {
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namespace test {
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// Statistics for one processed frame.
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struct FrameStatistic {
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explicit FrameStatistic(size_t frame_number) : frame_number(frame_number) {}
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std::string ToString() const;
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size_t frame_number = 0;
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size_t rtp_timestamp = 0;
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// Encoding.
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int64_t encode_start_ns = 0;
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int encode_return_code = 0;
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bool encoding_successful = false;
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size_t encode_time_us = 0;
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size_t target_bitrate_kbps = 0;
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size_t encoded_frame_size_bytes = 0;
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webrtc::FrameType frame_type = kVideoFrameDelta;
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// Layering.
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size_t temporal_layer_idx = 0;
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size_t simulcast_svc_idx = 0;
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// H264 specific.
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size_t max_nalu_size_bytes = 0;
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// Decoding.
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int64_t decode_start_ns = 0;
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int decode_return_code = 0;
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bool decoding_successful = false;
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size_t decode_time_us = 0;
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size_t decoded_width = 0;
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size_t decoded_height = 0;
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// Quantization.
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int qp = -1;
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// How many packets were discarded of the encoded frame data (if any).
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size_t packets_dropped = 0;
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size_t total_packets = 0;
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size_t manipulated_length = 0;
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// Quality.
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float psnr = 0.0;
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float ssim = 0.0;
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};
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// Statistics for a sequence of processed frames. This class is not thread safe.
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class Stats {
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public:
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Stats() = default;
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~Stats() = default;
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// Creates a FrameStatistic for the next frame to be processed.
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FrameStatistic* AddFrame();
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// Returns the FrameStatistic corresponding to |frame_number|.
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FrameStatistic* GetFrame(size_t frame_number);
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size_t size() const;
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private:
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std::vector<FrameStatistic> stats_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
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