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 "bitstream.h"
  25 #include "ra288.h"
  26 #include "lpc.h"
  27
  28 typedef struct {
  29     float sp_lpc[36];      ///< LPC coefficients for speech data (spec: A)
  30     float gain_lpc[10];    ///< LPC coefficients for gain        (spec: GB)
  31
  32     /** speech data history                                      (spec: SB).
  33      *  Its first 70 coefficients are updated only at backward filtering.
  34      */
  35     float sp_hist[111];
  36
  37     /// speech part of the gain autocorrelation                  (spec: REXP)
  38     float sp_rec[37];
  39
  40     /** log-gain history                                         (spec: SBLG).
  41      *  Its first 28 coefficients are updated only at backward filtering.
  42      */
  43     float gain_hist[38];
  44
  45     /// recursive part of the gain autocorrelation               (spec: REXPLG)
  46     float gain_rec[11];
  47 } RA288Context;
  48
  49 static av_cold int ra288_decode_init(AVCodecContext *avctx)
  50 {
  51     avctx->sample_fmt = SAMPLE_FMT_FLT;
  52     return 0;
  53 }
  54
  55 static inline float scalar_product_float(const float * v1, const float * v2,
  56                                          int size)
  57 {
  58     float res = 0.;
  59
  60     while (size--)
  61         res += *v1++ * *v2++;
  62
  63     return res;
  64 }
  65
  66 static void apply_window(float *tgt, const float *m1, const float *m2, int n)
  67 {
  68     while (n--)
  69         *tgt++ = *m1++ * *m2++;
  70 }
  71
  72 static void convolve(float *tgt, const float *src, int len, int n)
  73 {
  74     for (; n >= 0; n--)
  75         tgt[n] = scalar_product_float(src, src - n, len);
  76
  77 }
  78
  79 static void decode(RA288Context *ractx, float gain, int cb_coef)
  80 {
  81     int i, j;
  82     double sumsum;
  83     float sum, buffer[5];
  84     float *block = ractx->sp_hist + 70 + 36; // current block
  85     float *gain_block = ractx->gain_hist + 28;
  86
  87     memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
  88
  89     /* block 46 of G.728 spec */
  90     sum = 32.;
  91     for (i=0; i < 10; i++)
  92         sum -= gain_block[9-i] * ractx->gain_lpc[i];
  93
  94     /* block 47 of G.728 spec */
  95     sum = av_clipf(sum, 0, 60);
  96
  97     /* block 48 of G.728 spec */
  98     /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
  99     sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
 100
 101     for (i=0; i < 5; i++)
 102         buffer[i] = codetable[cb_coef][i] * sumsum;
 103
 104     sum = scalar_product_float(buffer, buffer, 5) * ((1<<24)/5.);
 105
 106     sum = FFMAX(sum, 1);
 107
 108     /* shift and store */
 109     memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
 110
 111     gain_block[9] = 10 * log10(sum) - 32;
 112
 113     for (i=0; i < 5; i++) {
 114         block[i] = buffer[i];
 115         for (j=0; j < 36; j++)
 116             block[i] -= block[i-1-j]*ractx->sp_lpc[j];
 117     }
 118
 119     /* output */
 120     for (i=0; i < 5; i++)
 121         block[i] = av_clipf(block[i], -4095./4096., 4095./4096.);
 122 }
 123
 124 /**
 125  * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
 126  *
 127  * @param order   filter order
 128  * @param n       input length
 129  * @param non_rec number of non-recursive samples
 130  * @param out     filter output
 131  * @param hist    pointer to the input history of the filter
 132  * @param out     pointer to the non-recursive part of the output
 133  * @param out2    pointer to the recursive part of the output
 134  * @param window  pointer to the windowing function table
 135  */
 136 static void do_hybrid_window(int order, int n, int non_rec, float *out,
 137                              float *hist, float *out2, const float *window)
 138 {
 139     int i;
 140     float buffer1[order + 1];
 141     float buffer2[order + 1];
 142     float work[order + n + non_rec];
 143
 144     apply_window(work, window, hist, order + n + non_rec);
 145
 146     convolve(buffer1, work + order    , n      , order);
 147     convolve(buffer2, work + order + n, non_rec, order);
 148
 149     for (i=0; i <= order; i++) {
 150         out2[i] = out2[i] * 0.5625 + buffer1[i];
 151         out [i] = out2[i]          + buffer2[i];
 152     }
 153
 154     /* Multiply by the white noise correcting factor (WNCF). */
 155     *out *= 257./256.;
 156 }
 157
 158 /**
 159  * Backward synthesis filter, find the LPC coefficients from past speech data.
 160  */
 161 static void backward_filter(float *hist, float *rec, const float *window,
 162                             float *lpc, const float *tab,
 163                             int order, int n, int non_rec, int move_size)
 164 {
 165     float temp[order+1];
 166
 167     do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
 168
 169     if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
 170         apply_window(lpc, lpc, tab, order);
 171
 172     memmove(hist, hist + n, move_size*sizeof(*hist));
 173 }
 174
 175 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
 176                               int *data_size, const uint8_t * buf,
 177                               int buf_size)
 178 {
 179     float *out = data;
 180     int i, j;
 181     RA288Context *ractx = avctx->priv_data;
 182     GetBitContext gb;
 183
 184     if (buf_size < avctx->block_align) {
 185         av_log(avctx, AV_LOG_ERROR,
 186                "Error! Input buffer is too small [%d<%d]\n",
 187                buf_size, avctx->block_align);
 188         return 0;
 189     }
 190
 191     if (*data_size < 32*5*4)
 192         return -1;
 193
 194     init_get_bits(&gb, buf, avctx->block_align * 8);
 195
 196     for (i=0; i < 32; i++) {
 197         float gain = amptable[get_bits(&gb, 3)];
 198         int cb_coef = get_bits(&gb, 6 + (i&1));
 199
 200         decode(ractx, gain, cb_coef);
 201
 202         for (j=0; j < 5; j++)
 203             *(out++) = ractx->sp_hist[70 + 36 + j];
 204
 205         if ((i & 7) == 3) {
 206             backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
 207                             ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
 208
 209             backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
 210                             ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
 211         }
 212     }
 213
 214     *data_size = (char *)out - (char *)data;
 215     return avctx->block_align;
 216 }
 217
 218 AVCodec ra_288_decoder =
 219 {
 220     "real_288",
 221     CODEC_TYPE_AUDIO,
 222     CODEC_ID_RA_288,
 223     sizeof(RA288Context),
 224     ra288_decode_init,
 225     NULL,
 226     NULL,
 227     ra288_decode_frame,
 228     .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
 229 };