libavcodec/ra288.c

   1 /*
   2  * RealAudio 2.0 (28.8K)
   3  * Copyright (c) 2003 the ffmpeg project
   4  *
   5  * This file is part of FFmpeg.
   6  *
   7  * FFmpeg is free software; you can redistribute it and/or
   8  * modify it under the terms of the GNU Lesser General Public
   9  * License as published by the Free Software Foundation; either
  10  * version 2.1 of the License, or (at your option) any later version.
  11  *
  12  * FFmpeg is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15  * Lesser General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU Lesser General Public
  18  * License along with FFmpeg; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 #include "avcodec.h"
  23 #define ALT_BITSTREAM_READER_LE
  24 #include "get_bits.h"
  25 #include "ra288.h"
  26 #include "lpc.h"
  27 #include "celp_math.h"
  28 #include "celp_filters.h"
  29
  30 typedef struct {
  31     float sp_lpc[36];      ///< LPC coefficients for speech data (spec: A)
  32     float gain_lpc[10];    ///< LPC coefficients for gain        (spec: GB)
  33
  34     /** speech data history                                      (spec: SB).
  35      *  Its first 70 coefficients are updated only at backward filtering.
  36      */
  37     float sp_hist[111];
  38
  39     /// speech part of the gain autocorrelation                  (spec: REXP)
  40     float sp_rec[37];
  41
  42     /** log-gain history                                         (spec: SBLG).
  43      *  Its first 28 coefficients are updated only at backward filtering.
  44      */
  45     float gain_hist[38];
  46
  47     /// recursive part of the gain autocorrelation               (spec: REXPLG)
  48     float gain_rec[11];
  49 } RA288Context;
  50
  51 static av_cold int ra288_decode_init(AVCodecContext *avctx)
  52 {
  53     avctx->sample_fmt = SAMPLE_FMT_FLT;
  54     return 0;
  55 }
  56
  57 static void apply_window(float *tgt, const float *m1, const float *m2, int n)
  58 {
  59     while (n--)
  60         *tgt++ = *m1++ * *m2++;
  61 }
  62
  63 static void convolve(float *tgt, const float *src, int len, int n)
  64 {
  65     for (; n >= 0; n--)
  66         tgt[n] = ff_dot_productf(src, src - n, len);
  67
  68 }
  69
  70 static void decode(RA288Context *ractx, float gain, int cb_coef)
  71 {
  72     int i;
  73     double sumsum;
  74     float sum, buffer[5];
  75     float *block = ractx->sp_hist + 70 + 36; // current block
  76     float *gain_block = ractx->gain_hist + 28;
  77
  78     memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
  79
  80     /* block 46 of G.728 spec */
  81     sum = 32.;
  82     for (i=0; i < 10; i++)
  83         sum -= gain_block[9-i] * ractx->gain_lpc[i];
  84
  85     /* block 47 of G.728 spec */
  86     sum = av_clipf(sum, 0, 60);
  87
  88     /* block 48 of G.728 spec */
  89     /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
  90     sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
  91
  92     for (i=0; i < 5; i++)
  93         buffer[i] = codetable[cb_coef][i] * sumsum;
  94
  95     sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
  96
  97     sum = FFMAX(sum, 1);
  98
  99     /* shift and store */
 100     memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
 101
 102     gain_block[9] = 10 * log10(sum) - 32;
 103
 104     ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
 105
 106     /* output */
 107     for (i=0; i < 5; i++)
 108         block[i] = av_clipf(block[i], -4095./4096., 4095./4096.);
 109 }
 110
 111 /**
 112  * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
 113  *
 114  * @param order   filter order
 115  * @param n       input length
 116  * @param non_rec number of non-recursive samples
 117  * @param out     filter output
 118  * @param hist    pointer to the input history of the filter
 119  * @param out     pointer to the non-recursive part of the output
 120  * @param out2    pointer to the recursive part of the output
 121  * @param window  pointer to the windowing function table
 122  */
 123 static void do_hybrid_window(int order, int n, int non_rec, float *out,
 124                              float *hist, float *out2, const float *window)
 125 {
 126     int i;
 127     float buffer1[order + 1];
 128     float buffer2[order + 1];
 129     float work[order + n + non_rec];
 130
 131     apply_window(work, window, hist, order + n + non_rec);
 132
 133     convolve(buffer1, work + order    , n      , order);
 134     convolve(buffer2, work + order + n, non_rec, order);
 135
 136     for (i=0; i <= order; i++) {
 137         out2[i] = out2[i] * 0.5625 + buffer1[i];
 138         out [i] = out2[i]          + buffer2[i];
 139     }
 140
 141     /* Multiply by the white noise correcting factor (WNCF). */
 142     *out *= 257./256.;
 143 }
 144
 145 /**
 146  * Backward synthesis filter, find the LPC coefficients from past speech data.
 147  */
 148 static void backward_filter(float *hist, float *rec, const float *window,
 149                             float *lpc, const float *tab,
 150                             int order, int n, int non_rec, int move_size)
 151 {
 152     float temp[order+1];
 153
 154     do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
 155
 156     if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
 157         apply_window(lpc, lpc, tab, order);
 158
 159     memmove(hist, hist + n, move_size*sizeof(*hist));
 160 }
 161
 162 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
 163                               int *data_size, AVPacket *avpkt)
 164 {
 165     const uint8_t *buf = avpkt->data;
 166     int buf_size = avpkt->size;
 167     float *out = data;
 168     int i, j;
 169     RA288Context *ractx = avctx->priv_data;
 170     GetBitContext gb;
 171
 172     if (buf_size < avctx->block_align) {
 173         av_log(avctx, AV_LOG_ERROR,
 174                "Error! Input buffer is too small [%d<%d]\n",
 175                buf_size, avctx->block_align);
 176         return 0;
 177     }
 178
 179     if (*data_size < 32*5*4)
 180         return -1;
 181
 182     init_get_bits(&gb, buf, avctx->block_align * 8);
 183
 184     for (i=0; i < 32; i++) {
 185         float gain = amptable[get_bits(&gb, 3)];
 186         int cb_coef = get_bits(&gb, 6 + (i&1));
 187
 188         decode(ractx, gain, cb_coef);
 189
 190         for (j=0; j < 5; j++)
 191             *(out++) = ractx->sp_hist[70 + 36 + j];
 192
 193         if ((i & 7) == 3) {
 194             backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
 195                             ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
 196
 197             backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
 198                             ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
 199         }
 200     }
 201
 202     *data_size = (char *)out - (char *)data;
 203     return avctx->block_align;
 204 }
 205
 206 AVCodec ra_288_decoder =
 207 {
 208     "real_288",
 209     CODEC_TYPE_AUDIO,
 210     CODEC_ID_RA_288,
 211     sizeof(RA288Context),
 212     ra288_decode_init,
 213     NULL,
 214     NULL,
 215     ra288_decode_frame,
 216     .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
 217 };