/* * Copyright (c) 2011 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. */ #ifndef MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_ #define MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_ #include #include #include #include "common_types.h" // NOLINT(build/include) namespace webrtc { namespace test { // Statistics for one processed frame. struct FrameStatistic { FrameStatistic(size_t frame_number, size_t rtp_timestamp) : frame_number(frame_number), rtp_timestamp(rtp_timestamp) {} std::string ToString() const; size_t frame_number = 0; size_t rtp_timestamp = 0; // Encoding. int64_t encode_start_ns = 0; int encode_return_code = 0; bool encoding_successful = false; size_t encode_time_us = 0; size_t target_bitrate_kbps = 0; size_t encoded_frame_size_bytes = 0; webrtc::FrameType frame_type = kVideoFrameDelta; // Layering. size_t temporal_layer_idx = 0; size_t simulcast_svc_idx = 0; // H264 specific. size_t max_nalu_size_bytes = 0; // Decoding. int64_t decode_start_ns = 0; int decode_return_code = 0; bool decoding_successful = false; size_t decode_time_us = 0; size_t decoded_width = 0; size_t decoded_height = 0; // Quantization. int qp = -1; // Quality. float psnr = 0.0; float ssim = 0.0; }; // Statistics for a sequence of processed frames. This class is not thread safe. class Stats { public: Stats() = default; ~Stats() = default; // Creates a FrameStatistic for the next frame to be processed. FrameStatistic* AddFrame(size_t timestamp); // Returns the FrameStatistic corresponding to |frame_number| or |timestamp|. FrameStatistic* GetFrame(size_t frame_number); FrameStatistic* GetFrameWithTimestamp(size_t timestamp); size_t size() const; private: std::vector stats_; std::map rtp_timestamp_to_frame_num_; }; } // namespace test } // namespace webrtc #endif // MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_