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/*
** FAAD - Freeware Advanced Audio Decoder
** Copyright (C) 2002 M. Bakker
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU General Public License as published by
** the Free Software Foundation; either version 2 of the License, or
** (at your option) any later version.
**
** This program 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 General Public License for more details.
**
** You should have received a copy of the GNU General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
** $Id: output.c,v 1.1 2002/07/14 23:43:01 miguelfreitas Exp $
**/
#include "common.h"
#include "output.h"
#include "decoder.h"
#define ftol(A,B) {tmp = *(int32_t*) & A - 0x4B7F8000; \
B = (int16_t)((tmp==(int16_t)tmp) ? tmp : (tmp>>31)^0x7FFF);}
#define ROUND(x) ((x >= 0) ? (int32_t)floor((x) + 0.5) : (int32_t)ceil((x) + 0.5))
#define ROUND32(x) ROUND(x)
#define FLOAT_SCALE (1.0f/(1<<15))
void* output_to_PCM(real_t **input, void *sample_buffer, uint8_t channels,
uint16_t frame_len, uint8_t format)
{
uint8_t ch;
uint16_t i;
uint8_t *p = (uint8_t*)sample_buffer;
int16_t *short_sample_buffer = (int16_t*)sample_buffer;
int32_t *int_sample_buffer = (int32_t*)sample_buffer;
float32_t *float_sample_buffer = (float32_t*)sample_buffer;
/* Copy output to a standard PCM buffer */
switch (format)
{
case FAAD_FMT_16BIT:
for (ch = 0; ch < channels; ch++)
{
for(i = 0; i < frame_len; i++)
{
int32_t tmp;
real_t ftemp;
ftemp = input[ch][i] + 0xff8000;
ftol(ftemp, short_sample_buffer[(i*channels)+ch]);
}
}
break;
case FAAD_FMT_24BIT:
for (ch = 0; ch < channels; ch++)
{
for(i = 0; i < frame_len; i++)
{
if (input[ch][i] > (1<<15)-1)
input[ch][i] = (1<<15)-1;
else if (input[ch][i] < -(1<<15))
input[ch][i] = -(1<<15);
int_sample_buffer[(i*channels)+ch] = ROUND(input[ch][i]*(1<<8));
}
}
break;
case FAAD_FMT_32BIT:
for (ch = 0; ch < channels; ch++)
{
for(i = 0; i < frame_len; i++)
{
if (input[ch][i] > (1<<15)-1)
input[ch][i] = (1<<15)-1;
else if (input[ch][i] < -(1<<15))
input[ch][i] = -(1<<15);
int_sample_buffer[(i*channels)+ch] = ROUND32(input[ch][i]*(1<<16));
}
}
break;
case FAAD_FMT_FLOAT:
for (ch = 0; ch < channels; ch++)
{
for(i = 0; i < frame_len; i++)
{
if (input[ch][i] > (1<<15)-1)
input[ch][i] = (1<<15)-1;
else if (input[ch][i] < -(1<<15))
input[ch][i] = -(1<<15);
float_sample_buffer[(i*channels)+ch] = input[ch][i]*FLOAT_SCALE;
}
}
break;
}
return sample_buffer;
}
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