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#include "celp_math.h"
28#include "celp_filters.h"
29
30typedef 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
51static av_cold int ra288_decode_init(AVCodecContext *avctx)
52{
53    avctx->sample_fmt = SAMPLE_FMT_FLT;
54    return 0;
55}
56
57static void apply_window(float *tgt, const float *m1, const float *m2, int n)
58{
59    while (n--)
60        *tgt++ = *m1++ * *m2++;
61}
62
63static 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
70static 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 */
123static 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 */
148static 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
162static int ra288_decode_frame(AVCodecContext * avctx, void *data,
163                              int *data_size, const uint8_t * buf,
164                              int buf_size)
165{
166    float *out = data;
167    int i, j;
168    RA288Context *ractx = avctx->priv_data;
169    GetBitContext gb;
170
171    if (buf_size < avctx->block_align) {
172        av_log(avctx, AV_LOG_ERROR,
173               "Error! Input buffer is too small [%d<%d]\n",
174               buf_size, avctx->block_align);
175        return 0;
176    }
177
178    if (*data_size < 32*5*4)
179        return -1;
180
181    init_get_bits(&gb, buf, avctx->block_align * 8);
182
183    for (i=0; i < 32; i++) {
184        float gain = amptable[get_bits(&gb, 3)];
185        int cb_coef = get_bits(&gb, 6 + (i&1));
186
187        decode(ractx, gain, cb_coef);
188
189        for (j=0; j < 5; j++)
190            *(out++) = ractx->sp_hist[70 + 36 + j];
191
192        if ((i & 7) == 3) {
193            backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
194                            ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
195
196            backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
197                            ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
198        }
199    }
200
201    *data_size = (char *)out - (char *)data;
202    return avctx->block_align;
203}
204
205AVCodec ra_288_decoder =
206{
207    "real_288",
208    CODEC_TYPE_AUDIO,
209    CODEC_ID_RA_288,
210    sizeof(RA288Context),
211    ra288_decode_init,
212    NULL,
213    NULL,
214    ra288_decode_frame,
215    .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
216};
217