ref: c0ae72652fc9619e8b1e8f365ab977614179779a
dir: /libfaad/output.c/
/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003 M. Bakker, Ahead Software AG, http://www.nero.com
**
** 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.
**
** Any non-GPL usage of this software or parts of this software is strictly
** forbidden.
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Ahead Software through [email protected].
**
** $Id: output.c,v 1.29 2003/11/12 20:47:58 menno Exp $
**/
#include "common.h"
#include "structs.h"
#include "output.h"
#include "decoder.h"
#ifndef FIXED_POINT
#define FLOAT_SCALE (1.0f/(1<<15))
#define DM_MUL ((real_t)1.0/((real_t)1.0+(real_t)sqrt(2.0)))
static INLINE real_t get_sample(real_t **input, uint8_t channel, uint16_t sample,
uint8_t downMatrix, uint8_t *internal_channel)
{
if (!downMatrix)
return input[internal_channel[channel]][sample];
if (channel == 0)
{
return DM_MUL * (input[internal_channel[1]][sample] +
input[internal_channel[0]][sample]/(real_t)sqrt(2.) +
input[internal_channel[3]][sample]/(real_t)sqrt(2.));
} else {
return DM_MUL * (input[internal_channel[2]][sample] +
input[internal_channel[0]][sample]/(real_t)sqrt(2.) +
input[internal_channel[4]][sample]/(real_t)sqrt(2.));
}
}
void* output_to_PCM(faacDecHandle hDecoder,
real_t **input, void *sample_buffer, uint8_t channels,
uint16_t frame_len, uint8_t format)
{
uint8_t ch;
uint16_t i, j = 0;
uint8_t internal_channel;
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;
double *double_sample_buffer = (double*)sample_buffer;
/* Copy output to a standard PCM buffer */
for (ch = 0; ch < channels; ch++)
{
internal_channel = hDecoder->internal_channel[ch];
switch (format)
{
case FAAD_FMT_16BIT:
for(i = 0; i < frame_len; i++)
{
real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel);
if (inp >= 0.0f)
{
#ifndef HAS_LRINTF
inp += 0.5f;
#endif
if (inp >= 32768.0f)
{
inp = 32767.0f;
}
} else {
#ifndef HAS_LRINTF
inp += -0.5f;
#endif
if (inp <= -32769.0f)
{
inp = -32768.0f;
}
}
short_sample_buffer[(i*channels)+ch] = (int16_t)lrintf(inp);
}
break;
case FAAD_FMT_24BIT:
for(i = 0; i < frame_len; i++)
{
real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel);
inp *= 256.0f;
if (inp >= 0.0f)
{
#ifndef HAS_LRINTF
inp += 0.5f;
#endif
if (inp >= 8388608.0f)
{
inp = 8388607.0f;
}
} else {
#ifndef HAS_LRINTF
inp += -0.5f;
#endif
if (inp <= -8388609.0f)
{
inp = -8388608.0f;
}
}
int_sample_buffer[(i*channels)+ch] = lrintf(inp);
}
break;
case FAAD_FMT_32BIT:
for(i = 0; i < frame_len; i++)
{
real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel);
inp *= 65536.0f;
if (inp >= 0.0f)
{
#ifndef HAS_LRINTF
inp += 0.5f;
#endif
if (inp >= 2147483648.0f)
{
inp = 2147483647.0f;
}
} else {
#ifndef HAS_LRINTF
inp += -0.5f;
#endif
if (inp <= -2147483649.0f)
{
inp = -2147483648.0f;
}
}
int_sample_buffer[(i*channels)+ch] = lrintf(inp);
}
break;
case FAAD_FMT_FLOAT:
for(i = 0; i < frame_len; i++)
{
//real_t inp = input[internal_channel][i];
real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel);
float_sample_buffer[(i*channels)+ch] = inp*FLOAT_SCALE;
}
break;
case FAAD_FMT_DOUBLE:
for(i = 0; i < frame_len; i++)
{
//real_t inp = input[internal_channel][i];
real_t inp = get_sample(input, ch, i, hDecoder->downMatrix, hDecoder->internal_channel);
double_sample_buffer[(i*channels)+ch] = (double)inp*FLOAT_SCALE;
}
break;
}
}
return sample_buffer;
}
#else
void* output_to_PCM(faacDecHandle hDecoder,
real_t **input, void *sample_buffer, uint8_t channels,
uint16_t frame_len, uint8_t format)
{
uint8_t ch;
uint16_t i;
int16_t *short_sample_buffer = (int16_t*)sample_buffer;
int32_t *int_sample_buffer = (int32_t*)sample_buffer;
/* Copy output to a standard PCM buffer */
for (ch = 0; ch < channels; ch++)
{
switch (format)
{
case FAAD_FMT_16BIT:
for(i = 0; i < frame_len; i++)
{
int32_t tmp = input[ch][i];
if (tmp >= 0)
{
tmp += (1 << (REAL_BITS-1));
if (tmp >= REAL_CONST(32768))
{
tmp = REAL_CONST(32767);
}
} else {
tmp += -(1 << (REAL_BITS-1));
if (tmp <= REAL_CONST(-32769))
{
tmp = REAL_CONST(-32768);
}
}
tmp >>= REAL_BITS;
short_sample_buffer[(i*channels)+ch] = (int16_t)tmp;
}
break;
case FAAD_FMT_24BIT:
for(i = 0; i < frame_len; i++)
{
int32_t tmp = input[ch][i];
if (tmp >= 0)
{
tmp += (1 << (REAL_BITS-9));
tmp >>= (REAL_BITS-8);
if (tmp >= 8388608)
{
tmp = 8388607;
}
} else {
tmp += -(1 << (REAL_BITS-9));
tmp >>= (REAL_BITS-8);
if (tmp <= -8388609)
{
tmp = -8388608;
}
}
int_sample_buffer[(i*channels)+ch] = (int32_t)tmp;
}
break;
case FAAD_FMT_32BIT:
for(i = 0; i < frame_len; i++)
{
int32_t tmp = input[ch][i];
if (tmp >= 0)
{
tmp += (1 << (16-REAL_BITS-1));
tmp <<= (16-REAL_BITS);
} else {
tmp += -(1 << (16-REAL_BITS-1));
tmp <<= (16-REAL_BITS);
}
int_sample_buffer[(i*channels)+ch] = (int32_t)tmp;
}
break;
}
}
return sample_buffer;
}
#endif