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/*
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* Copyright (c) 2007-2008 Ian Caulfield
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* 2009 Ramiro Polla
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config.h"
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#include "libavutil/attributes.h"
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#include "mlpdsp.h"
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#include "mlp.h"
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static void mlp_filter_channel(int32_t *state, const int32_t *coeff,
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int firorder, int iirorder,
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unsigned int filter_shift, int32_t mask,
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int blocksize, int32_t *sample_buffer)
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{
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int32_t *firbuf = state;
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int32_t *iirbuf = state + MAX_BLOCKSIZE + MAX_FIR_ORDER;
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const int32_t *fircoeff = coeff;
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const int32_t *iircoeff = coeff + MAX_FIR_ORDER;
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int i;
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for (i = 0; i < blocksize; i++) {
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int32_t residual = *sample_buffer;
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unsigned int order;
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int64_t accum = 0;
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int32_t result;
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for (order = 0; order < firorder; order++)
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accum += (int64_t) firbuf[order] * fircoeff[order];
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for (order = 0; order < iirorder; order++)
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accum += (int64_t) iirbuf[order] * iircoeff[order];
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accum = accum >> filter_shift;
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result = (accum + residual) & mask;
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*--firbuf = result;
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*--iirbuf = result - accum;
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*sample_buffer = result;
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sample_buffer += MAX_CHANNELS;
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}
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}
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void ff_mlp_rematrix_channel(int32_t *samples,
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const int32_t *coeffs,
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const uint8_t *bypassed_lsbs,
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const int8_t *noise_buffer,
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int index,
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unsigned int dest_ch,
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uint16_t blockpos,
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unsigned int maxchan,
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int matrix_noise_shift,
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int access_unit_size_pow2,
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int32_t mask)
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{
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unsigned int src_ch, i;
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int index2 = 2 * index + 1;
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for (i = 0; i < blockpos; i++) {
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int64_t accum = 0;
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for (src_ch = 0; src_ch <= maxchan; src_ch++)
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accum += (int64_t) samples[src_ch] * coeffs[src_ch];
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if (matrix_noise_shift) {
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index &= access_unit_size_pow2 - 1;
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accum += noise_buffer[index] << (matrix_noise_shift + 7);
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index += index2;
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}
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samples[dest_ch] = ((accum >> 14) & mask) + *bypassed_lsbs;
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bypassed_lsbs += MAX_CHANNELS;
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samples += MAX_CHANNELS;
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}
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}
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static int32_t (*mlp_select_pack_output(uint8_t *ch_assign,
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int8_t *output_shift,
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uint8_t max_matrix_channel,
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int is32))(int32_t, uint16_t, int32_t (*)[], void *, uint8_t*, int8_t *, uint8_t, int)
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{
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return ff_mlp_pack_output;
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}
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int32_t ff_mlp_pack_output(int32_t lossless_check_data,
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uint16_t blockpos,
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int32_t (*sample_buffer)[MAX_CHANNELS],
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void *data,
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uint8_t *ch_assign,
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int8_t *output_shift,
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uint8_t max_matrix_channel,
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int is32)
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{
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unsigned int i, out_ch = 0;
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int32_t *data_32 = data;
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int16_t *data_16 = data;
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for (i = 0; i < blockpos; i++) {
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for (out_ch = 0; out_ch <= max_matrix_channel; out_ch++) {
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int mat_ch = ch_assign[out_ch];
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int32_t sample = sample_buffer[i][mat_ch] *
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(1 << output_shift[mat_ch]);
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lossless_check_data ^= (sample & 0xffffff) << mat_ch;
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if (is32)
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*data_32++ = sample << 8;
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else
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*data_16++ = sample >> 8;
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}
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}
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return lossless_check_data;
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}
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av_cold void ff_mlpdsp_init(MLPDSPContext *c)
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{
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c->mlp_filter_channel = mlp_filter_channel;
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c->mlp_rematrix_channel = ff_mlp_rematrix_channel;
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c->mlp_select_pack_output = mlp_select_pack_output;
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c->mlp_pack_output = ff_mlp_pack_output;
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if (ARCH_ARM)
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ff_mlpdsp_init_arm(c);
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if (ARCH_X86)
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ff_mlpdsp_init_x86(c);
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}
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