remove preprocessor cruft

[libav.git] / libavcodec / dtsdec.c

/*
 * dtsdec.c : free DTS Coherent Acoustics stream decoder.
 * Copyright (C) 2004 Benjamin Zores <ben@geexbox.org>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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.
 *
 * 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 General Public License for more details.
 *
 * You should have received a copy of the GNU 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 "avcodec.h"
#include <dts.h>

#include <stdlib.h>
#include <string.h>

#define BUFFER_SIZE 18726
#define HEADER_SIZE 14

#ifdef LIBDTS_FIXED
#define CONVERT_LEVEL (1 << 26)
#define CONVERT_BIAS 0
#else
#define CONVERT_LEVEL 1
#define CONVERT_BIAS 384
#endif

static inline int16_t
convert(int32_t i)
{
#ifdef LIBDTS_FIXED
    i >>= 15;
#else
    i -= 0x43c00000;
#endif
    return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
}

static void
convert2s16_2(sample_t * _f, int16_t * s16)
{
    int i;
    int32_t *f = (int32_t *) _f;

    for(i = 0; i < 256; i++) {
        s16[2 * i] = convert(f[i]);
        s16[2 * i + 1] = convert(f[i + 256]);
    }
}

static void
convert2s16_4(sample_t * _f, int16_t * s16)
{
    int i;
    int32_t *f = (int32_t *) _f;

    for(i = 0; i < 256; i++) {
        s16[4 * i] = convert(f[i]);
        s16[4 * i + 1] = convert(f[i + 256]);
        s16[4 * i + 2] = convert(f[i + 512]);
        s16[4 * i + 3] = convert(f[i + 768]);
    }
}

static void
convert2s16_5(sample_t * _f, int16_t * s16)
{
    int i;
    int32_t *f = (int32_t *) _f;

    for(i = 0; i < 256; i++) {
        s16[5 * i] = convert(f[i]);
        s16[5 * i + 1] = convert(f[i + 256]);
        s16[5 * i + 2] = convert(f[i + 512]);
        s16[5 * i + 3] = convert(f[i + 768]);
        s16[5 * i + 4] = convert(f[i + 1024]);
    }
}

static void
convert2s16_multi(sample_t * _f, int16_t * s16, int flags)
{
    int i;
    int32_t *f = (int32_t *) _f;

    switch (flags) {
    case DTS_MONO:
        for(i = 0; i < 256; i++) {
            s16[5 * i] = s16[5 * i + 1] = s16[5 * i + 2] = s16[5 * i + 3] =
                0;
            s16[5 * i + 4] = convert(f[i]);
        }
        break;
    case DTS_CHANNEL:
    case DTS_STEREO:
    case DTS_DOLBY:
        convert2s16_2(_f, s16);
        break;
    case DTS_3F:
        for(i = 0; i < 256; i++) {
            s16[5 * i] = convert(f[i]);
            s16[5 * i + 1] = convert(f[i + 512]);
            s16[5 * i + 2] = s16[5 * i + 3] = 0;
            s16[5 * i + 4] = convert(f[i + 256]);
        }
        break;
    case DTS_2F2R:
        convert2s16_4(_f, s16);
        break;
    case DTS_3F2R:
        convert2s16_5(_f, s16);
        break;
    case DTS_MONO | DTS_LFE:
        for(i = 0; i < 256; i++) {
            s16[6 * i] = s16[6 * i + 1] = s16[6 * i + 2] = s16[6 * i + 3] =
                0;
            s16[6 * i + 4] = convert(f[i + 256]);
            s16[6 * i + 5] = convert(f[i]);
        }
        break;
    case DTS_CHANNEL | DTS_LFE:
    case DTS_STEREO | DTS_LFE:
    case DTS_DOLBY | DTS_LFE:
        for(i = 0; i < 256; i++) {
            s16[6 * i] = convert(f[i + 256]);
            s16[6 * i + 1] = convert(f[i + 512]);
            s16[6 * i + 2] = s16[6 * i + 3] = s16[6 * i + 4] = 0;
            s16[6 * i + 5] = convert(f[i]);
        }
        break;
    case DTS_3F | DTS_LFE:
        for(i = 0; i < 256; i++) {
            s16[6 * i] = convert(f[i + 256]);
            s16[6 * i + 1] = convert(f[i + 768]);
            s16[6 * i + 2] = s16[6 * i + 3] = 0;
            s16[6 * i + 4] = convert(f[i + 512]);
            s16[6 * i + 5] = convert(f[i]);
        }
        break;
    case DTS_2F2R | DTS_LFE:
        for(i = 0; i < 256; i++) {
            s16[6 * i] = convert(f[i + 256]);
            s16[6 * i + 1] = convert(f[i + 512]);
            s16[6 * i + 2] = convert(f[i + 768]);
            s16[6 * i + 3] = convert(f[i + 1024]);
            s16[6 * i + 4] = 0;
            s16[6 * i + 5] = convert(f[i]);
        }
        break;
    case DTS_3F2R | DTS_LFE:
        for(i = 0; i < 256; i++) {
            s16[6 * i] = convert(f[i + 256]);
            s16[6 * i + 1] = convert(f[i + 768]);
            s16[6 * i + 2] = convert(f[i + 1024]);
            s16[6 * i + 3] = convert(f[i + 1280]);
            s16[6 * i + 4] = convert(f[i + 512]);
            s16[6 * i + 5] = convert(f[i]);
        }
        break;
    }
}

static int
channels_multi(int flags)
{
    if(flags & DTS_LFE)
        return 6;
    else if(flags & 1)          /* center channel */
        return 5;
    else if((flags & DTS_CHANNEL_MASK) == DTS_2F2R)
        return 4;
    else
        return 2;
}

static int
dts_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
                 uint8_t * buff, int buff_size)
{
    uint8_t *start = buff;
    uint8_t *end = buff + buff_size;
    static uint8_t buf[BUFFER_SIZE];
    static uint8_t *bufptr = buf;
    static uint8_t *bufpos = buf + HEADER_SIZE;
    int16_t *out_samples = data;
    static int sample_rate;
    static int frame_length;
    static int flags;
    int bit_rate;
    int len;
    dts_state_t *state = avctx->priv_data;
    level_t level;
    sample_t bias;
    int i;

    *data_size = 0;

    while(1) {
        int length;

        len = end - start;
        if(!len)
            break;
        if(len > bufpos - bufptr)
            len = bufpos - bufptr;
        memcpy(bufptr, start, len);
        bufptr += len;
        start += len;
        if(bufptr != bufpos)
            return start - buff;
        if(bufpos != buf + HEADER_SIZE)
            break;

        length = dts_syncinfo(state, buf, &flags, &sample_rate, &bit_rate,
                              &frame_length);
        if(!length) {
            av_log(NULL, AV_LOG_INFO, "skip\n");
            for(bufptr = buf; bufptr < buf + HEADER_SIZE - 1; bufptr++)
                bufptr[0] = bufptr[1];
            continue;
        }
        bufpos = buf + length;
    }

    flags = 2;              /* ???????????? */
    level = CONVERT_LEVEL;
    bias = CONVERT_BIAS;

    flags |= DTS_ADJUST_LEVEL;
    if(dts_frame(state, buf, &flags, &level, bias)) {
        av_log(avctx, AV_LOG_ERROR, "dts_frame() failed\n");
        goto end;
    }

    avctx->sample_rate = sample_rate;
    avctx->channels = channels_multi(flags);
    avctx->bit_rate = bit_rate;

    for(i = 0; i < dts_blocks_num(state); i++) {
        int chans;

        if(dts_block(state)) {
            av_log(avctx, AV_LOG_ERROR, "dts_block() failed\n");
            goto end;
        }

        chans = channels_multi(flags);
        convert2s16_multi(dts_samples(state), out_samples,
                          flags & (DTS_CHANNEL_MASK | DTS_LFE));

        out_samples += 256 * chans;
        *data_size += 256 * sizeof(int16_t) * chans;
    }

end:
    bufptr = buf;
    bufpos = buf + HEADER_SIZE;
    return start - buff;
}

static int
dts_decode_init(AVCodecContext * avctx)
{
    avctx->priv_data = dts_init(0);
    if(avctx->priv_data == NULL)
        return -1;

    return 0;
}

static int
dts_decode_end(AVCodecContext * s)
{
    return 0;
}

AVCodec dts_decoder = {
    "dts",
    CODEC_TYPE_AUDIO,
    CODEC_ID_DTS,
    sizeof(dts_state_t *),
    dts_decode_init,
    NULL,
    dts_decode_end,
    dts_decode_frame,
};