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ffmpeg / libavcodec / ra144.c @ 4fdb41a8

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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 FFmpeg.
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 *
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 * FFmpeg 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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 * FFmpeg 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 FFmpeg; 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 "avcodec.h"
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#include "bitstream.h"
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#include "ra144.h"
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#include "acelp_filters.h"
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#define NBLOCKS         4       ///< number of subblocks within a block
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#define BLOCKSIZE       40      ///< subblock size in 16-bit words
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#define BUFFERSIZE      146     ///< the size of the adaptive codebook
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typedef struct {
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    unsigned int     old_energy;        ///< previous frame energy
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    unsigned int     lpc_tables[2][10];
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    /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
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     *  and lpc_coef[1] of the previous one */
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    unsigned int    *lpc_coef[2];
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    unsigned int     lpc_refl_rms[2];
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    /** the current subblock padded by the last 10 values of the previous one*/
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    int16_t curr_sblock[50];
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    /** adaptive codebook. Its size is two units bigger to avoid a
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     *  buffer overflow */
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    uint16_t adapt_cb[148];
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} RA144Context;
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static int ra144_decode_init(AVCodecContext * avctx)
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{
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    RA144Context *ractx = avctx->priv_data;
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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 = SAMPLE_FMT_S16;
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    return 0;
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}
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/**
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 * Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an
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 * odd way to make the output identical to the binary decoder.
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 */
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static int t_sqrt(unsigned int x)
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{
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    int s = 2;
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    while (x > 0xfff) {
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        s++;
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        x >>= 2;
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    }
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    return ff_sqrt(x << 20) << s;
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}
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/**
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 * Evaluate the LPC filter coefficients from the reflection coefficients.
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 * Does the inverse of the eval_refl() function.
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 */
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static void eval_coefs(int *coefs, const int *refl)
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{
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    int buffer[10];
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    int *b1 = buffer;
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    int *b2 = coefs;
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    int i, j;
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    for (i=0; i < 10; i++) {
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        b1[i] = refl[i] << 4;
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        for (j=0; j < i; j++)
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            b1[j] = ((refl[i] * b2[i-j-1]) >> 12) + b2[j];
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        FFSWAP(int *, b1, b2);
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    }
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    for (i=0; i < 10; i++)
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        coefs[i] >>= 4;
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}
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/**
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 * Copy the last offset values of *source to *target. If those values are not
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 * enough to fill the target buffer, fill it with another copy of those values.
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 */
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static void copy_and_dup(int16_t *target, const int16_t *source, int offset)
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{
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    source += BUFFERSIZE - offset;
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    if (offset > BLOCKSIZE) {
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        memcpy(target, source, BLOCKSIZE*sizeof(*target));
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    } else {
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        memcpy(target, source, offset*sizeof(*target));
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        memcpy(target + offset, source, (BLOCKSIZE - offset)*sizeof(*target));
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    }
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}
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/** inverse root mean square */
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static int irms(const int16_t *data)
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{
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    unsigned int i, sum = 0;
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    for (i=0; i < BLOCKSIZE; i++)
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        sum += data[i] * data[i];
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    if (sum == 0)
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        return 0; /* OOPS - division by zero */
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    return 0x20000000 / (t_sqrt(sum) >> 8);
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}
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static void add_wav(int16_t *dest, int n, int skip_first, int *m,
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                    const int16_t *s1, const int8_t *s2, const int8_t *s3)
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{
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    int i;
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    int v[3];
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    v[0] = 0;
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    for (i=!skip_first; i<3; i++)
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        v[i] = (gain_val_tab[n][i] * m[i]) >> (gain_exp_tab[n][i] + 1);
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    for (i=0; i < BLOCKSIZE; i++)
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        dest[i] = (s1[i]*v[0] + s2[i]*v[1] + s3[i]*v[2]) >> 12;
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}
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static unsigned int rescale_rms(unsigned int rms, unsigned int energy)
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{
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    return (rms * energy) >> 10;
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}
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static unsigned int rms(const int *data)
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{
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    int i;
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    unsigned int res = 0x10000;
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    int b = 10;
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    for (i=0; i < 10; i++) {
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        res = (((0x1000000 - data[i]*data[i]) >> 12) * res) >> 12;
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        if (res == 0)
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            return 0;
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        while (res <= 0x3fff) {
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            b++;
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            res <<= 2;
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        }
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    }
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    return t_sqrt(res) >> b;
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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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    uint16_t buffer_a[40];
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    uint16_t *block;
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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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    int m[3];
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    if (cba_idx) {
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        cba_idx += BLOCKSIZE/2 - 1;
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        copy_and_dup(buffer_a, ractx->adapt_cb, cba_idx);
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        m[0] = (irms(buffer_a) * gval) >> 12;
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    } else {
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        m[0] = 0;
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    }
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    m[1] = (cb1_base[cb1_idx] * gval) >> 8;
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    m[2] = (cb2_base[cb2_idx] * gval) >> 8;
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    memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE,
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            (BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb));
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    block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE;
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    add_wav(block, gain, cba_idx, m, buffer_a,
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            cb1_vects[cb1_idx], cb2_vects[cb2_idx]);
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    memcpy(ractx->curr_sblock, ractx->curr_sblock + 40,
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           10*sizeof(*ractx->curr_sblock));
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    if (ff_acelp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs,
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                                     block, BLOCKSIZE, 10, 1, 0xfff))
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        memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock));
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}
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static void int_to_int16(int16_t *out, const int *inp)
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{
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    int i;
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    for (i=0; i < 30; i++)
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        *out++ = *inp++;
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}
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/**
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 * Evaluate the reflection coefficients from the filter coefficients.
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 * Does the inverse of the eval_coefs() function.
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 *
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 * @return 1 if one of the reflection coefficients is of magnitude greater than
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 *         4095, 0 if not.
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 */
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static int eval_refl(int *refl, const int16_t *coefs, RA144Context *ractx)
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{
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    int b, i, j;
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    int buffer1[10];
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    int buffer2[10];
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    int *bp1 = buffer1;
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    int *bp2 = buffer2;
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    for (i=0; i < 10; i++)
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        buffer2[i] = coefs[i];
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    refl[9] = bp2[9];
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    if ((unsigned) bp2[9] + 0x1000 > 0x1fff) {
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        av_log(ractx, AV_LOG_ERROR, "Overflow. Broken sample?\n");
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        return 1;
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    }
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    for (i=8; i >= 0; i--) {
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        b = 0x1000-((bp2[i+1] * bp2[i+1]) >> 12);
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        if (!b)
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            b = -2;
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        for (j=0; j <= i; j++)
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            bp1[j] = ((bp2[j] - ((refl[i+1] * bp2[i-j]) >> 12)) * (0x1000000 / b)) >> 12;
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        if ((unsigned) bp1[i] + 0x1000 > 0x1fff)
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            return 1;
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        refl[i] = bp1[i];
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        FFSWAP(int *, bp1, bp2);
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    }
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    return 0;
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}
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static int interp(RA144Context *ractx, int16_t *out, int block_num,
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                  int copyold, int energy)
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{
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    int work[10];
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    int a = block_num + 1;
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    int b = NBLOCKS - a;
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    int i;
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    // Interpolate block coefficients from the this frame forth block and
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    // last frame forth block
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    for (i=0; i<30; i++)
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        out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2;
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    if (eval_refl(work, out, ractx)) {
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        // The interpolated coefficients are unstable, copy either new or old
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        // coefficients
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        int_to_int16(out, ractx->lpc_coef[copyold]);
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        return rescale_rms(ractx->lpc_refl_rms[copyold], energy);
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    } else {
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        return rescale_rms(rms(work), energy);
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    }
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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, const uint8_t *buf, int buf_size)
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{
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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][30]; // 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] = lpc_refl_cb[i][get_bits(&gb, sizes[i])];
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    eval_coefs(ractx->lpc_coef[0], lpc_refl);
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    ractx->lpc_refl_rms[0] = rms(lpc_refl);
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    energy = energy_tab[get_bits(&gb, 5)];
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    refl_rms[0] = interp(ractx, block_coefs[0], 0, 1, ractx->old_energy);
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    refl_rms[1] = interp(ractx, block_coefs[1], 1, energy <= ractx->old_energy,
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                    t_sqrt(energy*ractx->old_energy) >> 12);
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    refl_rms[2] = interp(ractx, block_coefs[2], 2, 0, energy);
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    refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy);
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    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 ra_144_decoder =
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{
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    "real_144",
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    CODEC_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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};