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ffmpeg / libavcodec / ra288.c @ 2477d609

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/*
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 * RealAudio 2.0 (28.8K)
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 * Copyright (c) 2003 the ffmpeg project
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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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#define ALT_BITSTREAM_READER_LE
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#include "bitstream.h"
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#include "ra288.h"
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typedef struct {
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    float history[8];
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    float output[40];
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    float pr1[36];
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    float pr2[10];
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    int   phase;
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    float st1a[111], st1b[37], st1[37];
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    float st2a[38], st2b[11], st2[11];
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    float sb[41];
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    float lhist[10];
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} Real288_internal;
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static inline float scalar_product_float(const float * v1, const float * v2,
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                                         int size)
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{
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    float res = 0.;
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    while (size--)
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        res += *v1++ * *v2++;
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    return res;
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}
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/* Decode and produce output */
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static void decode(Real288_internal *glob, float gain, int cb_coef)
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{
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    int x, y;
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    double sumsum;
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    float sum, buffer[5];
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    memmove(glob->sb + 5, glob->sb, 36 * sizeof(*glob->sb));
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    for (x=4; x >= 0; x--)
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        glob->sb[x] = -scalar_product_float(glob->sb + x + 1, glob->pr1, 36);
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    /* convert log and do rms */
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    sum = 32. - scalar_product_float(glob->pr2, glob->lhist, 10);
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    sum = av_clipf(sum, 0, 60);
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    sumsum = exp(sum * 0.1151292546497) * gain;    /* pow(10.0,sum/20)*f */
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    for (x=0; x < 5; x++)
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        buffer[x] = codetable[cb_coef][x] * sumsum;
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    sum = scalar_product_float(buffer, buffer, 5) / 5;
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    sum = FFMAX(sum, 1);
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    /* shift and store */
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    memmove(glob->lhist, glob->lhist - 1, 10 * sizeof(*glob->lhist));
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    *glob->lhist = glob->history[glob->phase] = 10 * log10(sum) - 32;
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    for (x=1; x < 5; x++)
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        for (y=x-1; y >= 0; y--)
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            buffer[x] -= glob->pr1[x-y-1] * buffer[y];
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    /* output */
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    for (x=0; x < 5; x++) {
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        glob->output[glob->phase*5+x] = glob->sb[4-x] =
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            av_clipf(glob->sb[4-x] + buffer[x], -4095, 4095);
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    }
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}
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/* column multiply */
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static void colmult(float *tgt, const float *m1, const float *m2, int n)
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{
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    while (n--)
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        *(tgt++) = (*(m1++)) * (*(m2++));
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}
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/**
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 * Converts autocorrelation coefficients to LPC coefficients using the
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 * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification.
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 *
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 * @return 0 if success, -1 if fail
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 */
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static int eval_lpc_coeffs(const float *in, float *tgt, int n)
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{
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    int x, y;
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    double f0, f1, f2;
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    if (in[n] == 0)
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        return -1;
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    if ((f0 = *in) <= 0)
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        return -1;
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    in--; // To avoid a -1 subtraction in the inner loop
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    for (x=1; x <= n; x++) {
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        f1 = in[x+1];
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        for (y=0; y < x - 1; y++)
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            f1 += in[x-y]*tgt[y];
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        tgt[x-1] = f2 = -f1/f0;
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        for (y=0; y < x >> 1; y++) {
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            float temp = tgt[y] + tgt[x-y-2]*f2;
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            tgt[x-y-2] += tgt[y]*f2;
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            tgt[y] = temp;
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        }
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        if ((f0 += f1*f2) < 0)
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            return -1;
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    }
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    return 0;
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}
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/* product sum (lsf) */
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static void prodsum(float *tgt, const float *src, int len, int n)
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{
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    for (; n >= 0; n--)
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        tgt[n] = scalar_product_float(src, src - n, len);
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}
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/**
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 * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
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 *
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 * @param order   the order of the filter
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 * @param n       the length of the input
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 * @param non_rec the number of non recursive samples
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 * @param out     the filter output
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 * @param in      pointer to the input of the filter
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 * @param hist    pointer to the input history of the filter. It is updated by
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 *                this function.
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 * @param out     pointer to the non recursive part of the output
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 * @param out2    pointer to the recursive part of the output
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 * @param window  pointer to the windowing function table
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 */
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static void do_hybrid_window(int order, int n, int non_rec, const float *in,
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                             float *out, float *hist, float *out2,
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                             const float *window)
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{
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    unsigned int x;
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    float buffer1[37];
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    float buffer2[37];
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    float work[111];
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    /* update history */
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    memmove(hist                  , hist + n, (order + non_rec)*sizeof(*hist));
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    memcpy (hist + order + non_rec, in      , n                *sizeof(*hist));
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    colmult(work, window, hist, order + n + non_rec);
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    prodsum(buffer1, work + order    , n      , order);
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    prodsum(buffer2, work + order + n, non_rec, order);
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    for (x=0; x <= order; x++) {
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        out2[x] = out2[x] * 0.5625 + buffer1[x];
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        out [x] = out2[x]          + buffer2[x];
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    }
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    /* Multiply by the white noise correcting factor (WNCF) */
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    *out *= 257./256.;
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}
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/**
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 * Backward synthesis filter. Find the LPC coefficients from past speech data.
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 */
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static void backward_filter(Real288_internal *glob)
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{
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    float buffer1[40], temp1[37];
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    float buffer2[8], temp2[11];
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    memcpy(buffer1     , glob->output + 20, 20*sizeof(*buffer1));
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    memcpy(buffer1 + 20, glob->output     , 20*sizeof(*buffer1));
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    do_hybrid_window(36, 40, 35, buffer1, temp1, glob->st1a, glob->st1b,
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                     syn_window);
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    if (!eval_lpc_coeffs(temp1, glob->st1, 36))
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        colmult(glob->pr1, glob->st1, syn_bw_tab, 36);
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    memcpy(buffer2    , glob->history + 4, 4*sizeof(*buffer2));
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    memcpy(buffer2 + 4, glob->history    , 4*sizeof(*buffer2));
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    do_hybrid_window(10, 8, 20, buffer2, temp2, glob->st2a, glob->st2b,
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                     gain_window);
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    if (!eval_lpc_coeffs(temp2, glob->st2, 10))
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        colmult(glob->pr2, glob->st2, gain_bw_tab, 10);
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}
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/* Decode a block (celp) */
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static int ra288_decode_frame(AVCodecContext * avctx, void *data,
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                              int *data_size, const uint8_t * buf,
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                              int buf_size)
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{
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    int16_t *out = data;
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    int x, y;
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    Real288_internal *glob = avctx->priv_data;
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    GetBitContext gb;
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    if (buf_size < avctx->block_align) {
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        av_log(avctx, AV_LOG_ERROR,
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               "Error! Input buffer is too small [%d<%d]\n",
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               buf_size, avctx->block_align);
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        return 0;
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    }
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    init_get_bits(&gb, buf, avctx->block_align * 8);
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    for (x=0; x < 32; x++) {
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        float gain = amptable[get_bits(&gb, 3)];
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        int cb_coef = get_bits(&gb, 6 + (x&1));
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        glob->phase = x & 7;
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        decode(glob, gain, cb_coef);
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        for (y=0; y < 5; y++)
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            *(out++) = 8 * glob->output[glob->phase*5 + y];
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        if (glob->phase == 3)
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            backward_filter(glob);
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    }
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    *data_size = (char *)out - (char *)data;
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    return avctx->block_align;
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}
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AVCodec ra_288_decoder =
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{
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    "real_288",
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    CODEC_TYPE_AUDIO,
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    CODEC_ID_RA_288,
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    sizeof(Real288_internal),
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    NULL,
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    NULL,
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    NULL,
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    ra288_decode_frame,
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    .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
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};