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ffmpeg / libavcodec / ra288.c @ 2912e87a

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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 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 "avcodec.h"
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#define ALT_BITSTREAM_READER_LE
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#include "get_bits.h"
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#include "ra288.h"
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#include "lpc.h"
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#include "celp_math.h"
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#include "celp_filters.h"
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#define MAX_BACKWARD_FILTER_ORDER  36
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#define MAX_BACKWARD_FILTER_LEN    40
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#define MAX_BACKWARD_FILTER_NONREC 35
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typedef struct {
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    float sp_lpc[36];      ///< LPC coefficients for speech data (spec: A)
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    float gain_lpc[10];    ///< LPC coefficients for gain        (spec: GB)
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    /** speech data history                                      (spec: SB).
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     *  Its first 70 coefficients are updated only at backward filtering.
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     */
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    float sp_hist[111];
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    /// speech part of the gain autocorrelation                  (spec: REXP)
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    float sp_rec[37];
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    /** log-gain history                                         (spec: SBLG).
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     *  Its first 28 coefficients are updated only at backward filtering.
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     */
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    float gain_hist[38];
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    /// recursive part of the gain autocorrelation               (spec: REXPLG)
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    float gain_rec[11];
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} RA288Context;
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static av_cold int ra288_decode_init(AVCodecContext *avctx)
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{
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    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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    return 0;
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}
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static void apply_window(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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static void convolve(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] = ff_dot_productf(src, src - n, len);
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}
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static void decode(RA288Context *ractx, float gain, int cb_coef)
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{
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    int i;
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    double sumsum;
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    float sum, buffer[5];
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    float *block = ractx->sp_hist + 70 + 36; // current block
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    float *gain_block = ractx->gain_hist + 28;
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    memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
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    /* block 46 of G.728 spec */
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    sum = 32.;
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    for (i=0; i < 10; i++)
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        sum -= gain_block[9-i] * ractx->gain_lpc[i];
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    /* block 47 of G.728 spec */
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    sum = av_clipf(sum, 0, 60);
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    /* block 48 of G.728 spec */
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    /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
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    sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
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    for (i=0; i < 5; i++)
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        buffer[i] = codetable[cb_coef][i] * sumsum;
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    sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
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    sum = FFMAX(sum, 1);
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    /* shift and store */
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    memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
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    gain_block[9] = 10 * log10(sum) - 32;
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    ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
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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   filter order
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 * @param n       input length
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 * @param non_rec number of non-recursive samples
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 * @param out     filter output
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 * @param hist    pointer to the input history of the filter
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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, float *out,
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                             float *hist, float *out2, const float *window)
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{
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    int i;
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    float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
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    float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
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    float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC];
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    apply_window(work, window, hist, order + n + non_rec);
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    convolve(buffer1, work + order    , n      , order);
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    convolve(buffer2, work + order + n, non_rec, order);
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    for (i=0; i <= order; i++) {
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        out2[i] = out2[i] * 0.5625 + buffer1[i];
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        out [i] = out2[i]          + buffer2[i];
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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(float *hist, float *rec, const float *window,
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                            float *lpc, const float *tab,
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                            int order, int n, int non_rec, int move_size)
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{
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    float temp[MAX_BACKWARD_FILTER_ORDER+1];
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    do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
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    if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
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        apply_window(lpc, lpc, tab, order);
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    memmove(hist, hist + n, move_size*sizeof(*hist));
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}
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static int ra288_decode_frame(AVCodecContext * avctx, void *data,
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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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    float *out = data;
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    int i, j;
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    RA288Context *ractx = 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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    if (*data_size < 32*5*4)
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        return -1;
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    init_get_bits(&gb, buf, avctx->block_align * 8);
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    for (i=0; i < 32; i++) {
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        float gain = amptable[get_bits(&gb, 3)];
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        int cb_coef = get_bits(&gb, 6 + (i&1));
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        decode(ractx, gain, cb_coef);
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        for (j=0; j < 5; j++)
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            *(out++) = ractx->sp_hist[70 + 36 + j];
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        if ((i & 7) == 3) {
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            backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
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                            ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
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            backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
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                            ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
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        }
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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 ff_ra_288_decoder =
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{
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    "real_288",
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    AVMEDIA_TYPE_AUDIO,
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    CODEC_ID_RA_288,
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    sizeof(RA288Context),
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    ra288_decode_init,
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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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};