aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/8svx.c
blob: 5c68d347e24bf680739c48514bd9bca43e406345 (plain) (blame)
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
/*
 * Copyright (C) 2008 Jaikrishnan Menon
 * Copyright (C) 2011 Stefano Sabatini
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file
 * 8svx audio decoder
 * @author Jaikrishnan Menon
 *
 * supports: fibonacci delta encoding
 *         : exponential encoding
 *
 * For more information about the 8SVX format:
 * http://netghost.narod.ru/gff/vendspec/iff/iff.txt
 * http://sox.sourceforge.net/AudioFormats-11.html
 * http://aminet.net/package/mus/misc/wavepak
 * http://amigan.1emu.net/reg/8SVX.txt
 *
 * Samples can be found here:
 * http://aminet.net/mods/smpl/
 */

#include "avcodec.h"

/** decoder context */
typedef struct EightSvxContext {
    const int8_t *table;

    /* buffer used to store the whole audio decoded/interleaved chunk,
     * which is sent with the first packet */
    uint8_t *samples;
    size_t samples_size;
    int samples_idx;
} EightSvxContext;

static const int8_t fibonacci[16]   = { -34,  -21, -13,  -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8,  13, 21 };
static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };

#define MAX_FRAME_SIZE 2048

/**
 * Interleave samples in buffer containing all left channel samples
 * at the beginning, and right channel samples at the end.
 * Each sample is assumed to be in signed 8-bit format.
 *
 * @param size the size in bytes of the dst and src buffer
 */
static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
{
    uint8_t *dst_end = dst + size;
    size = size>>1;

    while (dst < dst_end) {
        *dst++ = *src;
        *dst++ = *(src+size);
        src++;
    }
}

/**
 * Delta decode the compressed values in src, and put the resulting
 * decoded n samples in dst.
 *
 * @param val starting value assumed by the delta sequence
 * @param table delta sequence table
 * @return size in bytes of the decoded data, must be src_size*2
 */
static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
                        int8_t val, const int8_t *table)
{
    int n = src_size;
    int8_t *dst0 = dst;

    while (n--) {
        uint8_t d = *src++;
        val = av_clip(val + table[d & 0x0f], -127, 128);
        *dst++ = val;
        val = av_clip(val + table[d >> 4]  , -127, 128);
        *dst++ = val;
    }

    return dst-dst0;
}

static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
                                 AVPacket *avpkt)
{
    EightSvxContext *esc = avctx->priv_data;
    int out_data_size, n;
    uint8_t *src, *dst;

    /* decode and interleave the first packet */
    if (!esc->samples && avpkt) {
        uint8_t *deinterleaved_samples;

        esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW ?
            avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
        if (!(esc->samples = av_malloc(esc->samples_size)))
            return AVERROR(ENOMEM);

        /* decompress */
        if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
            const uint8_t *buf = avpkt->data;
            int buf_size = avpkt->size;
            int n = esc->samples_size;

            if (buf_size < 2) {
                av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
                return AVERROR(EINVAL);
            }
            if (!(deinterleaved_samples = av_mallocz(n)))
                return AVERROR(ENOMEM);

            /* the uncompressed starting value is contained in the first byte */
            if (avctx->channels == 2) {
                delta_decode(deinterleaved_samples      , buf+1, buf_size/2-1, buf[0], esc->table);
                buf += buf_size/2;
                delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
            } else
                delta_decode(deinterleaved_samples      , buf+1, buf_size-1  , buf[0], esc->table);
        } else {
            deinterleaved_samples = avpkt->data;
        }

        if (avctx->channels == 2)
            interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
        else
            memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
    }

    /* return single packed with fixed size */
    out_data_size = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx);
    if (*data_size < out_data_size) {
        av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n", *data_size);
        return AVERROR(EINVAL);
    }

    *data_size = out_data_size;
    dst = data;
    src = esc->samples + esc->samples_idx;
    for (n = out_data_size; n > 0; n--)
        *dst++ = *src++ + 128;
    esc->samples_idx += *data_size;

    return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
        (avctx->frame_number == 0)*2 + out_data_size / 2 :
        out_data_size;
}

static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
{
    EightSvxContext *esc = avctx->priv_data;

    if (avctx->channels > 2) {
        av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
        return AVERROR_INVALIDDATA;
    }

    switch (avctx->codec->id) {
    case CODEC_ID_8SVX_FIB: esc->table = fibonacci;    break;
    case CODEC_ID_8SVX_EXP: esc->table = exponential;  break;
    case CODEC_ID_8SVX_RAW: esc->table = NULL;         break;
    default:
        av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
        return AVERROR_INVALIDDATA;
    }
    avctx->sample_fmt = AV_SAMPLE_FMT_U8;

    return 0;
}

static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
{
    EightSvxContext *esc = avctx->priv_data;

    av_freep(&esc->samples);
    esc->samples_size = 0;
    esc->samples_idx = 0;

    return 0;
}

AVCodec ff_eightsvx_fib_decoder = {
  .name           = "8svx_fib",
  .type           = AVMEDIA_TYPE_AUDIO,
  .id             = CODEC_ID_8SVX_FIB,
  .priv_data_size = sizeof (EightSvxContext),
  .init           = eightsvx_decode_init,
  .decode         = eightsvx_decode_frame,
  .close          = eightsvx_decode_close,
  .long_name      = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
};

AVCodec ff_eightsvx_exp_decoder = {
  .name           = "8svx_exp",
  .type           = AVMEDIA_TYPE_AUDIO,
  .id             = CODEC_ID_8SVX_EXP,
  .priv_data_size = sizeof (EightSvxContext),
  .init           = eightsvx_decode_init,
  .decode         = eightsvx_decode_frame,
  .close          = eightsvx_decode_close,
  .long_name      = NULL_IF_CONFIG_SMALL("8SVX exponential"),
};

AVCodec ff_eightsvx_raw_decoder = {
  .name           = "8svx_raw",
  .type           = AVMEDIA_TYPE_AUDIO,
  .id             = CODEC_ID_8SVX_RAW,
  .priv_data_size = sizeof(EightSvxContext),
  .init           = eightsvx_decode_init,
  .decode         = eightsvx_decode_frame,
  .close          = eightsvx_decode_close,
  .long_name      = NULL_IF_CONFIG_SMALL("8SVX rawaudio"),
};