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1723cf9fa2
This feature is not needed in video codec testing framework. In WebRTC video codecs never deal with packet loss. Packet loss is handled by jitter buffer which prevents passing of incomplete frames to decoder. Bug: webrtc:8768 Change-Id: I211cf51d913bec6a1f935e30691661d428ebd3b6 Reviewed-on: https://webrtc-review.googlesource.com/40740 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21722}
88 lines
2.3 KiB
C++
88 lines
2.3 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 <map>
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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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FrameStatistic(size_t frame_number, size_t rtp_timestamp)
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: frame_number(frame_number), rtp_timestamp(rtp_timestamp) {}
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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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// 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(size_t timestamp);
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// Returns the FrameStatistic corresponding to |frame_number| or |timestamp|.
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FrameStatistic* GetFrame(size_t frame_number);
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FrameStatistic* GetFrameWithTimestamp(size_t timestamp);
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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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std::map<size_t, size_t> rtp_timestamp_to_frame_num_;
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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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