2 * Copyright (C) 2008 Jaikrishnan Menon
3 * Copyright (C) 2011 Stefano Sabatini
5 * This file is part of FFmpeg.
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.
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.
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
25 * supports: fibonacci delta encoding
26 * : exponential encoding
28 * For more information about the 8SVX format:
29 * http://netghost.narod.ru/gff/vendspec/iff/iff.txt
30 * http://sox.sourceforge.net/AudioFormats-11.html
31 * http://aminet.net/package/mus/misc/wavepak
32 * http://amigan.1emu.net/reg/8SVX.txt
34 * Samples can be found here:
35 * http://aminet.net/mods/smpl/
40 /** decoder context */
41 typedef struct EightSvxContext {
44 /* buffer used to store the whole audio decoded/interleaved chunk,
45 * which is sent with the first packet */
51 static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
52 static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
54 #define MAX_FRAME_SIZE 2048
57 * Interleave samples in buffer containing all left channel samples
58 * at the beginning, and right channel samples at the end.
59 * Each sample is assumed to be in signed 8-bit format.
61 * @param size the size in bytes of the dst and src buffer
63 static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
65 uint8_t *dst_end = dst + size;
68 while (dst < dst_end) {
76 * Delta decode the compressed values in src, and put the resulting
77 * decoded n samples in dst.
79 * @param val starting value assumed by the delta sequence
80 * @param table delta sequence table
81 * @return size in bytes of the decoded data, must be src_size*2
83 static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
84 int8_t val, const int8_t *table)
91 val = av_clip(val + table[d & 0x0f], -127, 128);
93 val = av_clip(val + table[d >> 4] , -127, 128);
100 static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
103 EightSvxContext *esc = avctx->priv_data;
104 int out_data_size, n;
107 /* decode and interleave the first packet */
108 if (!esc->samples && avpkt) {
109 uint8_t *deinterleaved_samples;
111 esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW ?
112 avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
113 if (!(esc->samples = av_malloc(esc->samples_size)))
114 return AVERROR(ENOMEM);
117 if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
118 const uint8_t *buf = avpkt->data;
119 int buf_size = avpkt->size;
120 int n = esc->samples_size;
122 if (!(deinterleaved_samples = av_mallocz(n)))
123 return AVERROR(ENOMEM);
125 /* the uncompressed starting value is contained in the first byte */
126 if (avctx->channels == 2) {
127 delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table);
129 delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
131 delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table);
133 deinterleaved_samples = avpkt->data;
136 if (avctx->channels == 2)
137 interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
139 memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
142 /* return single packed with fixed size */
143 out_data_size = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx);
144 if (*data_size < out_data_size) {
145 av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n", *data_size);
146 return AVERROR(EINVAL);
149 *data_size = out_data_size;
151 src = esc->samples + esc->samples_idx;
152 for (n = out_data_size; n > 0; n--)
153 *dst++ = *src++ + 128;
154 esc->samples_idx += *data_size;
156 return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
157 (avctx->frame_number == 0)*2 + out_data_size / 2 :
161 static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
163 EightSvxContext *esc = avctx->priv_data;
165 if (avctx->channels > 2) {
166 av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
167 return AVERROR_INVALIDDATA;
170 switch (avctx->codec->id) {
171 case CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
172 case CODEC_ID_8SVX_EXP: esc->table = exponential; break;
173 case CODEC_ID_8SVX_RAW: esc->table = NULL; break;
175 av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
176 return AVERROR_INVALIDDATA;
178 avctx->sample_fmt = AV_SAMPLE_FMT_U8;
183 static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
185 EightSvxContext *esc = avctx->priv_data;
187 av_freep(&esc->samples);
188 esc->samples_size = 0;
189 esc->samples_idx = 0;
194 AVCodec ff_eightsvx_fib_decoder = {
196 .type = AVMEDIA_TYPE_AUDIO,
197 .id = CODEC_ID_8SVX_FIB,
198 .priv_data_size = sizeof (EightSvxContext),
199 .init = eightsvx_decode_init,
200 .decode = eightsvx_decode_frame,
201 .close = eightsvx_decode_close,
202 .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
205 AVCodec ff_eightsvx_exp_decoder = {
207 .type = AVMEDIA_TYPE_AUDIO,
208 .id = CODEC_ID_8SVX_EXP,
209 .priv_data_size = sizeof (EightSvxContext),
210 .init = eightsvx_decode_init,
211 .decode = eightsvx_decode_frame,
212 .close = eightsvx_decode_close,
213 .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
216 AVCodec ff_eightsvx_raw_decoder = {
218 .type = AVMEDIA_TYPE_AUDIO,
219 .id = CODEC_ID_8SVX_RAW,
220 .priv_data_size = sizeof(EightSvxContext),
221 .init = eightsvx_decode_init,
222 .decode = eightsvx_decode_frame,
223 .close = eightsvx_decode_close,
224 .long_name = NULL_IF_CONFIG_SMALL("8SVX rawaudio"),