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ffmpeg / libavcodec / ra144dec.c @ 2912e87a

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/*
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 * Real Audio 1.0 (14.4K)
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 *
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 * Copyright (c) 2008 Vitor Sessak
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 * Copyright (c) 2003 Nick Kurshev
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 *     Based on public domain decoder at http://www.honeypot.net/audio
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 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with Libav; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "libavutil/intmath.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "ra144.h"
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static av_cold int ra144_decode_init(AVCodecContext * avctx)
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{
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    RA144Context *ractx = avctx->priv_data;
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    ractx->avctx = avctx;
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    ractx->lpc_coef[0] = ractx->lpc_tables[0];
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    ractx->lpc_coef[1] = ractx->lpc_tables[1];
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    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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    return 0;
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}
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static void do_output_subblock(RA144Context *ractx, const uint16_t  *lpc_coefs,
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                               int gval, GetBitContext *gb)
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{
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    int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
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    int gain    = get_bits(gb, 8);
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    int cb1_idx = get_bits(gb, 7);
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    int cb2_idx = get_bits(gb, 7);
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    ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, gval,
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                          gain);
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}
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/** Uncompress one block (20 bytes -> 160*2 bytes). */
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static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
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                              int *data_size, AVPacket *avpkt)
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{
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    const uint8_t *buf = avpkt->data;
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    int buf_size = avpkt->size;
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    static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
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    unsigned int refl_rms[4];    // RMS of the reflection coefficients
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    uint16_t block_coefs[4][10]; // LPC coefficients of each sub-block
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    unsigned int lpc_refl[10];   // LPC reflection coefficients of the frame
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    int i, j;
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    int16_t *data = vdata;
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    unsigned int energy;
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    RA144Context *ractx = avctx->priv_data;
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    GetBitContext gb;
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    if (*data_size < 2*160)
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        return -1;
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    if(buf_size < 20) {
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        av_log(avctx, AV_LOG_ERROR,
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               "Frame too small (%d bytes). Truncated file?\n", buf_size);
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        *data_size = 0;
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        return buf_size;
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    }
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    init_get_bits(&gb, buf, 20 * 8);
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    for (i=0; i<10; i++)
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        lpc_refl[i] = ff_lpc_refl_cb[i][get_bits(&gb, sizes[i])];
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    ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
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    ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
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    energy = ff_energy_tab[get_bits(&gb, 5)];
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    refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
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    refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
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                            energy <= ractx->old_energy,
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                            ff_t_sqrt(energy*ractx->old_energy) >> 12);
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    refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
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    refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
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    ff_int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
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    for (i=0; i < 4; i++) {
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        do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);
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        for (j=0; j < BLOCKSIZE; j++)
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            *data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
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    }
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    ractx->old_energy = energy;
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    ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
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    FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
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    *data_size = 2*160;
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    return 20;
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}
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AVCodec ff_ra_144_decoder =
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{
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    "real_144",
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    AVMEDIA_TYPE_AUDIO,
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    CODEC_ID_RA_144,
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    sizeof(RA144Context),
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    ra144_decode_init,
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    NULL,
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    NULL,
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    ra144_decode_frame,
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    .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
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};