/* * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include #include "config.h" #include "libavutil/attributes.h" #include "libavutil/common.h" #include "apv.h" #include "apv_dsp.h" static const int8_t apv_trans_matrix[8][8] = { { 64, 64, 64, 64, 64, 64, 64, 64 }, { 89, 75, 50, 18, -18, -50, -75, -89 }, { 84, 35, -35, -84, -84, -35, 35, 84 }, { 75, -18, -89, -50, 50, 89, 18, -75 }, { 64, -64, -64, 64, 64, -64, -64, 64 }, { 50, -89, 18, 75, -75, -18, 89, -50 }, { 35, -84, 84, -35, -35, 84, -84, 35 }, { 18, -50, 75, -89, 89, -75, 50, -18 }, }; static void apv_decode_transquant_c(void *output, ptrdiff_t pitch, const int16_t *input_flat, const int16_t *qmatrix_flat, int bit_depth, int qp_shift) { const int16_t (*input)[8] = (const int16_t(*)[8])input_flat; const int16_t (*qmatrix)[8] = (const int16_t(*)[8])qmatrix_flat; int16_t scaled_coeff[8][8]; int32_t recon_sample[8][8]; // Dequant. { // Note that level_scale was already combined into qmatrix // before we got here. int bd_shift = bit_depth + 3 - 5; for (int y = 0; y < 8; y++) { for (int x = 0; x < 8; x++) { int coeff = ((int)(input[y][x] * qmatrix[y][x] * (1U << qp_shift) + (1 << (bd_shift - 1)))) >> bd_shift; scaled_coeff[y][x] = av_clip(coeff, APV_MIN_TRANS_COEFF, APV_MAX_TRANS_COEFF); } } } // Transform. { int32_t tmp[8][8]; // Vertical transform of columns. for (int x = 0; x < 8; x++) { for (int i = 0; i < 8; i++) { int sum = 0; for (int j = 0; j < 8; j++) sum += apv_trans_matrix[j][i] * scaled_coeff[j][x]; tmp[i][x] = sum; } } // Renormalise. for (int x = 0; x < 8; x++) { for (int y = 0; y < 8; y++) tmp[y][x] = (tmp[y][x] + 64) >> 7; } // Horizontal transform of rows. for (int y = 0; y < 8; y++) { for (int i = 0; i < 8; i++) { int sum = 0; for (int j = 0; j < 8; j++) sum += apv_trans_matrix[j][i] * tmp[y][j]; recon_sample[y][i] = sum; } } } // Output. if (bit_depth == 8) { uint8_t *ptr = output; int bd_shift = 20 - bit_depth; for (int y = 0; y < 8; y++) { for (int x = 0; x < 8; x++) { int sample = ((recon_sample[y][x] + (1 << (bd_shift - 1))) >> bd_shift) + (1 << (bit_depth - 1)); ptr[x] = av_clip_uintp2(sample, bit_depth); } ptr += pitch; } } else { uint16_t *ptr = output; int bd_shift = 20 - bit_depth; pitch /= 2; // Pitch was in bytes, 2 bytes per sample. for (int y = 0; y < 8; y++) { for (int x = 0; x < 8; x++) { int sample = ((recon_sample[y][x] + (1 << (bd_shift - 1))) >> bd_shift) + (1 << (bit_depth - 1)); ptr[x] = av_clip_uintp2(sample, bit_depth); } ptr += pitch; } } } av_cold void ff_apv_dsp_init(APVDSPContext *dsp) { dsp->decode_transquant = apv_decode_transquant_c; #if ARCH_X86_64 ff_apv_dsp_init_x86_64(dsp); #endif }