ra288.c 6.49 KB
Newer Older
1 2 3 4
/*
 * RealAudio 2.0 (28.8K)
 * Copyright (c) 2003 the ffmpeg project
 *
5
 * This file is part of Libav.
6
 *
7
 * Libav is free software; you can redistribute it and/or
8 9
 * modify it under the terms of the GNU Lesser General Public
 * 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
 * Libav is distributed in the hope that it will be useful,
13 14 15 16 17
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
18
 * License along with Libav; 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 31 32 33
#define MAX_BACKWARD_FILTER_ORDER  36
#define MAX_BACKWARD_FILTER_LEN    40
#define MAX_BACKWARD_FILTER_NONREC 35

34
typedef struct {
Vitor Sessak's avatar
Vitor Sessak committed
35
    float sp_lpc[36];      ///< LPC coefficients for speech data (spec: A)
36
    float gain_lpc[10];    ///< LPC coefficients for gain        (spec: GB)
Vitor Sessak's avatar
Vitor Sessak committed
37

38 39 40 41
    /** speech data history                                      (spec: SB).
     *  Its first 70 coefficients are updated only at backward filtering.
     */
    float sp_hist[111];
42

43
    /// speech part of the gain autocorrelation                  (spec: REXP)
44 45
    float sp_rec[37];

46 47 48 49
    /** log-gain history                                         (spec: SBLG).
     *  Its first 28 coefficients are updated only at backward filtering.
     */
    float gain_hist[38];
50

51
    /// recursive part of the gain autocorrelation               (spec: REXPLG)
52
    float gain_rec[11];
53
} RA288Context;
54

55 56
static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
57
    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
58 59 60
    return 0;
}

61
static void apply_window(float *tgt, const float *m1, const float *m2, int n)
62 63
{
    while (n--)
64
        *tgt++ = *m1++ * *m2++;
65 66
}

67 68 69
static void convolve(float *tgt, const float *src, int len, int n)
{
    for (; n >= 0; n--)
70
        tgt[n] = ff_dot_productf(src, src - n, len);
71 72 73

}

74
static void decode(RA288Context *ractx, float gain, int cb_coef)
75
{
76
    int i;
77 78
    double sumsum;
    float sum, buffer[5];
79 80
    float *block = ractx->sp_hist + 70 + 36; // current block
    float *gain_block = ractx->gain_hist + 28;
81

82
    memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
83

84
    /* block 46 of G.728 spec */
85 86
    sum = 32.;
    for (i=0; i < 10; i++)
87
        sum -= gain_block[9-i] * ractx->gain_lpc[i];
88

89
    /* block 47 of G.728 spec */
Vitor Sessak's avatar
Vitor Sessak committed
90
    sum = av_clipf(sum, 0, 60);
Vitor Sessak's avatar
Vitor Sessak committed
91

92
    /* block 48 of G.728 spec */
93
    /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
94
    sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
Vitor Sessak's avatar
Vitor Sessak committed
95

96
    for (i=0; i < 5; i++)
97
        buffer[i] = codetable[cb_coef][i] * sumsum;
Vitor Sessak's avatar
Vitor Sessak committed
98

99
    sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
100 101

    sum = FFMAX(sum, 1);
Vitor Sessak's avatar
Vitor Sessak committed
102 103

    /* shift and store */
104
    memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
Vitor Sessak's avatar
Vitor Sessak committed
105

106
    gain_block[9] = 10 * log10(sum) - 32;
Vitor Sessak's avatar
Vitor Sessak committed
107

108
    ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
109 110
}

111
/**
112
 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
113
 *
114 115 116 117
 * @param order   filter order
 * @param n       input length
 * @param non_rec number of non-recursive samples
 * @param out     filter output
118
 * @param hist    pointer to the input history of the filter
Vitor Sessak's avatar
Vitor Sessak committed
119
 * @param out     pointer to the non-recursive part of the output
120 121 122
 * @param out2    pointer to the recursive part of the output
 * @param window  pointer to the windowing function table
 */
Vitor Sessak's avatar
Vitor Sessak committed
123 124
static void do_hybrid_window(int order, int n, int non_rec, float *out,
                             float *hist, float *out2, const float *window)
125
{
126
    int i;
127 128 129
    float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
    float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
    float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC];
Vitor Sessak's avatar
Vitor Sessak committed
130

131
    apply_window(work, window, hist, order + n + non_rec);
132

133 134
    convolve(buffer1, work + order    , n      , order);
    convolve(buffer2, work + order + n, non_rec, order);
135

136 137 138
    for (i=0; i <= order; i++) {
        out2[i] = out2[i] * 0.5625 + buffer1[i];
        out [i] = out2[i]          + buffer2[i];
Vitor Sessak's avatar
Vitor Sessak committed
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
 */
148 149 150
static void backward_filter(float *hist, float *rec, const float *window,
                            float *lpc, const float *tab,
                            int order, int n, int non_rec, int move_size)
151
{
152
    float temp[MAX_BACKWARD_FILTER_ORDER+1];
153

154
    do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
155

156 157
    if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
        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)
Nick Kurshev's avatar
Nick Kurshev committed
164
{
165 166
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
167
    float *out = data;
168
    int i, j;
169
    RA288Context *ractx = avctx->priv_data;
170
    GetBitContext gb;
Vitor Sessak's avatar
Vitor Sessak committed
171

172 173 174 175 176 177 178
    if (buf_size < avctx->block_align) {
        av_log(avctx, AV_LOG_ERROR,
               "Error! Input buffer is too small [%d<%d]\n",
               buf_size, avctx->block_align);
        return 0;
    }

179
    if (*data_size < 32*5*4)
180 181
        return -1;

182
    init_get_bits(&gb, buf, avctx->block_align * 8);
Vitor Sessak's avatar
Vitor Sessak committed
183

184
    for (i=0; i < 32; i++) {
Vitor Sessak's avatar
Vitor Sessak committed
185
        float gain = amptable[get_bits(&gb, 3)];
186
        int cb_coef = get_bits(&gb, 6 + (i&1));
Vitor Sessak's avatar
Vitor Sessak committed
187

188
        decode(ractx, gain, cb_coef);
Vitor Sessak's avatar
Vitor Sessak committed
189

190
        for (j=0; j < 5; j++)
191
            *(out++) = ractx->sp_hist[70 + 36 + j];
Vitor Sessak's avatar
Vitor Sessak committed
192

193 194 195 196 197 198 199
        if ((i & 7) == 3) {
            backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
                            ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);

            backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
                            ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
        }
Vitor Sessak's avatar
Vitor Sessak committed
200 201
    }

202
    *data_size = (char *)out - (char *)data;
203
    return avctx->block_align;
204 205
}

206
AVCodec ff_ra_288_decoder =
207 208
{
    "real_288",
209
    AVMEDIA_TYPE_AUDIO,
210
    CODEC_ID_RA_288,
211
    sizeof(RA288Context),
212
    ra288_decode_init,
213 214 215
    NULL,
    NULL,
    ra288_decode_frame,
216
    .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
217
};