view dtsdec.c @ 4630:1416371d4a6c libavcodec

add avcodec_get_context_defaults2() / avcodec_alloc_context2() which take CodecType as an additional parameter also mark them as NOT part of the public API yet, so we can change their argument to CodecID if we decide to do so

/*
 * dtsdec.c : free DTS Coherent Acoustics stream decoder.
 * Copyright (C) 2004 Benjamin Zores <[e-mail]>
 *
 * 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

#define CONVERT_LEVEL 1
#define CONVERT_BIAS 0

typedef struct DTSContext {
    dts_state_t *state;
    uint8_t buf[BUFFER_SIZE];
    uint8_t *bufptr;
    uint8_t *bufpos;
} DTSContext;

static inline int16_t
convert(sample_t s)
{
    return s * 0x7fff;
}

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

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

static int
channels_multi(int flags)
{
    switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){
    case DTS_CHANNEL:
    case DTS_STEREO:
    case DTS_DOLBY:
        return 2;
    case DTS_2F2R:
        return 4;
    case DTS_MONO:
    case DTS_3F:
    case DTS_3F2R:
        return 5;
    case DTS_MONO | DTS_LFE:
    case DTS_CHANNEL | DTS_LFE:
    case DTS_STEREO | DTS_LFE:
    case DTS_DOLBY | DTS_LFE:
    case DTS_3F | DTS_LFE:
    case DTS_2F2R | DTS_LFE:
    case DTS_3F2R | DTS_LFE:
        return 6;
    }

    return -1;
}

static int
dts_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
                 uint8_t * buff, int buff_size)
{
    DTSContext *s = avctx->priv_data;
    uint8_t *start = buff;
    uint8_t *end = buff + buff_size;
    int16_t *out_samples = data;
    int sample_rate;
    int frame_length;
    int flags;
    int bit_rate;
    int len;
    level_t level;
    sample_t bias;
    int nblocks;
    int i;

    *data_size = 0;

    while(1) {
        int length;

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

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

    level = CONVERT_LEVEL;
    bias = CONVERT_BIAS;

    flags |= DTS_ADJUST_LEVEL;
    if(dts_frame(s->state, s->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;

    nblocks = dts_blocks_num(s->state);

    for(i = 0; i < nblocks; i++) {
        if(dts_block(s->state)) {
            av_log(avctx, AV_LOG_ERROR, "dts_block() failed\n");
            goto end;
        }

        convert2s16_multi(dts_samples(s->state), out_samples, flags);

        out_samples += 256 * avctx->channels;
        *data_size += 256 * sizeof(int16_t) * avctx->channels;
    }

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

static int
dts_decode_init(AVCodecContext * avctx)
{
    DTSContext *s = avctx->priv_data;
    s->bufptr = s->buf;
    s->bufpos = s->buf + HEADER_SIZE;
    s->state = dts_init(0);
    if(s->state == NULL)
        return -1;

    return 0;
}

static int
dts_decode_end(AVCodecContext * avctx)
{
    DTSContext *s = avctx->priv_data;
    dts_free(s->state);
    return 0;
}

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