libavcodec/ra144dec.c

/*
 * Real Audio 1.0 (14.4K)
 *
 * Copyright (c) 2008 Vitor Sessak
 * Copyright (c) 2003 Nick Kurshev
 *     Based on public domain decoder at http://www.honeypot.net/audio
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 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
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser 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 "libavutil/intmath.h"
#include "avcodec.h"
#include "get_bits.h"
#include "ra144.h"


static av_cold int ra144_decode_init(AVCodecContext * avctx)
{
    RA144Context *ractx = avctx->priv_data;

    ractx->avctx = avctx;

    ractx->lpc_coef[0] = ractx->lpc_tables[0];
    ractx->lpc_coef[1] = ractx->lpc_tables[1];

    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
    return 0;
}

static void do_output_subblock(RA144Context *ractx, const uint16_t  *lpc_coefs,
                               int gval, GetBitContext *gb)
{
    int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
    int gain    = get_bits(gb, 8);
    int cb1_idx = get_bits(gb, 7);
    int cb2_idx = get_bits(gb, 7);

    ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, gval,
                          gain);
}

/** Uncompress one block (20 bytes -> 160*2 bytes). */
static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
                              int *data_size, AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
    unsigned int refl_rms[4];    // RMS of the reflection coefficients
    uint16_t block_coefs[4][10]; // LPC coefficients of each sub-block
    unsigned int lpc_refl[10];   // LPC reflection coefficients of the frame
    int i, j;
    int16_t *data = vdata;
    unsigned int energy;

    RA144Context *ractx = avctx->priv_data;
    GetBitContext gb;

    if (*data_size < 2*160)
        return -1;

    if(buf_size < 20) {
        av_log(avctx, AV_LOG_ERROR,
               "Frame too small (%d bytes). Truncated file?\n", buf_size);
        *data_size = 0;
        return buf_size;
    }
    init_get_bits(&gb, buf, 20 * 8);

    for (i=0; i<10; i++)
        lpc_refl[i] = ff_lpc_refl_cb[i][get_bits(&gb, sizes[i])];

    ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
    ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);

    energy = ff_energy_tab[get_bits(&gb, 5)];

    refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
    refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
                            energy <= ractx->old_energy,
                            ff_t_sqrt(energy*ractx->old_energy) >> 12);
    refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
    refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);

    ff_int_to_int16(block_coefs[3], ractx->lpc_coef[0]);

    for (i=0; i < 4; i++) {
        do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);

        for (j=0; j < BLOCKSIZE; j++)
            *data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
    }

    ractx->old_energy = energy;
    ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];

    FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);

    *data_size = 2*160;
    return 20;
}

AVCodec ff_ra_144_decoder =
{
    "real_144",
    AVMEDIA_TYPE_AUDIO,
    CODEC_ID_RA_144,
    sizeof(RA144Context),
    ra144_decode_init,
    NULL,
    NULL,
    ra144_decode_frame,
    .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
};