summaryrefslogtreecommitdiffstats
path: root/libavcodec/ra288.c
blob: 215786c803689d986a56b258ab6ba759ff038c09 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
/*
 * RealAudio 2.0 (28.8K)
 * Copyright (c) 2003 the ffmpeg project
 *
 * This file is part of Libav.
 *
 * Libav is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * Libav is distributed in the hope that it will be useful,
 * 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
 * License along with Libav; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/internal.h"
#include "avcodec.h"
#include "internal.h"
#define BITSTREAM_READER_LE
#include "get_bits.h"
#include "ra288.h"
#include "lpc.h"
#include "celp_filters.h"

#define MAX_BACKWARD_FILTER_ORDER  36
#define MAX_BACKWARD_FILTER_LEN    40
#define MAX_BACKWARD_FILTER_NONREC 35

#define RA288_BLOCK_SIZE        5
#define RA288_BLOCKS_PER_FRAME 32

typedef struct {
    AVFloatDSPContext fdsp;
    DECLARE_ALIGNED(32, float,   sp_lpc)[FFALIGN(36, 16)];   ///< LPC coefficients for speech data (spec: A)
    DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)];   ///< LPC coefficients for gain        (spec: GB)

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

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

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

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

static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
    RA288Context *ractx = avctx->priv_data;

    avctx->channels       = 1;
    avctx->channel_layout = AV_CH_LAYOUT_MONO;
    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;

    avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

    return 0;
}

static void convolve(float *tgt, const float *src, int len, int n)
{
    for (; n >= 0; n--)
        tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);

}

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

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

    /* block 46 of G.728 spec */
    sum = 32.0;
    for (i=0; i < 10; i++)
        sum -= gain_block[9-i] * ractx->gain_lpc[i];

    /* block 47 of G.728 spec */
    sum = av_clipf(sum, 0, 60);

    /* block 48 of G.728 spec */
    /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
    sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));

    for (i=0; i < 5; i++)
        buffer[i] = codetable[cb_coef][i] * sumsum;

    sum = avpriv_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.0);

    sum = FFMAX(sum, 1);

    /* shift and store */
    memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));

    gain_block[9] = 10 * log10(sum) - 32;

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

/**
 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
 *
 * @param order   filter order
 * @param n       input length
 * @param non_rec number of non-recursive samples
 * @param out     filter output
 * @param hist    pointer to the input history of the filter
 * @param out     pointer to the non-recursive part of the output
 * @param out2    pointer to the recursive part of the output
 * @param window  pointer to the windowing function table
 */
static void do_hybrid_window(RA288Context *ractx,
                             int order, int n, int non_rec, float *out,
                             float *hist, float *out2, const float *window)
{
    int i;
    float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
    float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
    LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
                                            MAX_BACKWARD_FILTER_LEN   +
                                            MAX_BACKWARD_FILTER_NONREC, 16)]);

    ractx->fdsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));

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

    for (i=0; i <= order; i++) {
        out2[i] = out2[i] * 0.5625 + buffer1[i];
        out [i] = out2[i]          + buffer2[i];
    }

    /* Multiply by the white noise correcting factor (WNCF). */
    *out *= 257.0 / 256.0;
}

/**
 * Backward synthesis filter, find the LPC coefficients from past speech data.
 */
static void backward_filter(RA288Context *ractx,
                            float *hist, float *rec, const float *window,
                            float *lpc, const float *tab,
                            int order, int n, int non_rec, int move_size)
{
    float temp[MAX_BACKWARD_FILTER_ORDER+1];

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

    if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
        ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));

    memmove(hist, hist + n, move_size*sizeof(*hist));
}

static int ra288_decode_frame(AVCodecContext * avctx, void *data,
                              int *got_frame_ptr, AVPacket *avpkt)
{
    AVFrame *frame     = data;
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    float *out;
    int i, ret;
    RA288Context *ractx = avctx->priv_data;
    GetBitContext gb;

    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 AVERROR_INVALIDDATA;
    }

    /* get output buffer */
    frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
    }
    out = (float *)frame->data[0];

    init_get_bits(&gb, buf, avctx->block_align * 8);

    for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
        float gain = amptable[get_bits(&gb, 3)];
        int cb_coef = get_bits(&gb, 6 + (i&1));

        decode(ractx, gain, cb_coef);

        memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
        out += RA288_BLOCK_SIZE;

        if ((i & 7) == 3) {
            backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
                            ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);

            backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
                            ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
        }
    }

    *got_frame_ptr = 1;

    return avctx->block_align;
}

AVCodec ff_ra_288_decoder = {
    .name           = "real_288",
    .long_name      = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = AV_CODEC_ID_RA_288,
    .priv_data_size = sizeof(RA288Context),
    .init           = ra288_decode_init,
    .decode         = ra288_decode_frame,
    .capabilities   = CODEC_CAP_DR1,
};
OpenPOWER on IntegriCloud