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/*
* Copyright (c) 2007-2008 Ian Caulfield
* 2009 Ramiro Polla
*
* 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 "config.h"
#include "libavutil/attributes.h"
#include "mlpdsp.h"
#include "mlp.h"
static void mlp_filter_channel(int32_t *state, const int32_t *coeff,
int firorder, int iirorder,
unsigned int filter_shift, int32_t mask,
int blocksize, int32_t *sample_buffer)
{
int32_t *firbuf = state;
int32_t *iirbuf = state + MAX_BLOCKSIZE + MAX_FIR_ORDER;
const int32_t *fircoeff = coeff;
const int32_t *iircoeff = coeff + MAX_FIR_ORDER;
int i;
for (i = 0; i < blocksize; i++) {
int32_t residual = *sample_buffer;
unsigned int order;
int64_t accum = 0;
int32_t result;
for (order = 0; order < firorder; order++)
accum += (int64_t) firbuf[order] * fircoeff[order];
for (order = 0; order < iirorder; order++)
accum += (int64_t) iirbuf[order] * iircoeff[order];
accum = accum >> filter_shift;
result = (accum + residual) & mask;
*--firbuf = result;
*--iirbuf = result - accum;
*sample_buffer = result;
sample_buffer += MAX_CHANNELS;
}
}
void ff_mlp_rematrix_channel(int32_t *samples,
const int32_t *coeffs,
const uint8_t *bypassed_lsbs,
const int8_t *noise_buffer,
int index,
unsigned int dest_ch,
uint16_t blockpos,
unsigned int maxchan,
int matrix_noise_shift,
int access_unit_size_pow2,
int32_t mask)
{
unsigned int src_ch, i;
int index2 = 2 * index + 1;
for (i = 0; i < blockpos; i++) {
int64_t accum = 0;
for (src_ch = 0; src_ch <= maxchan; src_ch++)
accum += (int64_t) samples[src_ch] * coeffs[src_ch];
if (matrix_noise_shift) {
index &= access_unit_size_pow2 - 1;
accum += noise_buffer[index] << (matrix_noise_shift + 7);
index += index2;
}
samples[dest_ch] = ((accum >> 14) & mask) + *bypassed_lsbs;
bypassed_lsbs += MAX_CHANNELS;
samples += MAX_CHANNELS;
}
}
static int32_t (*mlp_select_pack_output(uint8_t *ch_assign,
int8_t *output_shift,
uint8_t max_matrix_channel,
int is32))(int32_t, uint16_t, int32_t (*)[], void *, uint8_t*, int8_t *, uint8_t, int)
{
return ff_mlp_pack_output;
}
int32_t ff_mlp_pack_output(int32_t lossless_check_data,
uint16_t blockpos,
int32_t (*sample_buffer)[MAX_CHANNELS],
void *data,
uint8_t *ch_assign,
int8_t *output_shift,
uint8_t max_matrix_channel,
int is32)
{
unsigned int i, out_ch = 0;
int32_t *data_32 = data;
int16_t *data_16 = data;
for (i = 0; i < blockpos; i++) {
for (out_ch = 0; out_ch <= max_matrix_channel; out_ch++) {
int mat_ch = ch_assign[out_ch];
int32_t sample = sample_buffer[i][mat_ch] *
(1U << output_shift[mat_ch]);
lossless_check_data ^= (sample & 0xffffff) << mat_ch;
if (is32)
*data_32++ = sample << 8;
else
*data_16++ = sample >> 8;
}
}
return lossless_check_data;
}
av_cold void ff_mlpdsp_init(MLPDSPContext *c)
{
c->mlp_filter_channel = mlp_filter_channel;
c->mlp_rematrix_channel = ff_mlp_rematrix_channel;
c->mlp_select_pack_output = mlp_select_pack_output;
c->mlp_pack_output = ff_mlp_pack_output;
if (ARCH_ARM)
ff_mlpdsp_init_arm(c);
if (ARCH_X86)
ff_mlpdsp_init_x86(c);
}
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