view Plugins/Input/aac/libfaad2/output.c @ 184:740c072f8e23 trunk

[svn] List sndfile extensions in the plugin configuration table.

/*
** 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 [e-mail].
**
** $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