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 * @author Jaikrishnan Menon
27 * supports: fibonacci delta encoding
28 * : exponential encoding
30 * For more information about the 8SVX format:
31 * http://netghost.narod.ru/gff/vendspec/iff/iff.txt
32 * http://sox.sourceforge.net/AudioFormats-11.html
33 * http://aminet.net/package/mus/misc/wavepak
34 * http://amigan.1emu.net/reg/8SVX.txt
36 * Samples can be found here:
37 * http://aminet.net/mods/smpl/
42 /** decoder context */
43 typedef struct EightSvxContext {
46 /* buffer used to store the whole audio decoded/interleaved chunk,
47 * which is sent with the first packet */
53 static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
54 static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
56 #define MAX_FRAME_SIZE 2048
59 * Interleave samples in buffer containing all left channel samples
60 * at the beginning, and right channel samples at the end.
61 * Each sample is assumed to be in signed 8-bit format.
63 * @param size the size in bytes of the dst and src buffer
65 static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
67 uint8_t *dst_end = dst + size;
70 while (dst < dst_end) {
78 * Delta decode the compressed values in src, and put the resulting
79 * decoded n samples in dst.
81 * @param val starting value assumed by the delta sequence
82 * @param table delta sequence table
83 * @return size in bytes of the decoded data, must be src_size*2
85 static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
86 int8_t val, const int8_t *table)
93 val = av_clip(val + table[d & 0x0f], -127, 128);
95 val = av_clip(val + table[d >> 4] , -127, 128);
102 static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
105 EightSvxContext *esc = avctx->priv_data;
106 int out_data_size, n;
109 /* decode and interleave the first packet */
110 if (!esc->samples && avpkt) {
111 uint8_t *deinterleaved_samples;
113 esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR?
114 avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
115 if (!(esc->samples = av_malloc(esc->samples_size)))
116 return AVERROR(ENOMEM);
119 if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
120 const uint8_t *buf = avpkt->data;
121 int buf_size = avpkt->size;
122 int n = esc->samples_size;
125 av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
126 return AVERROR(EINVAL);
128 if (!(deinterleaved_samples = av_mallocz(n)))
129 return AVERROR(ENOMEM);
131 /* the uncompressed starting value is contained in the first byte */
132 if (avctx->channels == 2) {
133 delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table);
135 delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
137 delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table);
139 deinterleaved_samples = avpkt->data;
142 if (avctx->channels == 2)
143 interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
145 memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
148 /* return single packed with fixed size */
149 out_data_size = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx);
150 if (*data_size < out_data_size) {
151 av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n", *data_size);
152 return AVERROR(EINVAL);
155 *data_size = out_data_size;
157 src = esc->samples + esc->samples_idx;
158 for (n = out_data_size; n > 0; n--)
159 *dst++ = *src++ + 128;
160 esc->samples_idx += *data_size;
162 return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
163 (avctx->frame_number == 0)*2 + out_data_size / 2 :
167 static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
169 EightSvxContext *esc = avctx->priv_data;
171 if (avctx->channels < 1 || avctx->channels > 2) {
172 av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
173 return AVERROR_INVALIDDATA;
176 switch (avctx->codec->id) {
177 case CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
178 case CODEC_ID_8SVX_EXP: esc->table = exponential; break;
179 case CODEC_ID_PCM_S8_PLANAR:
180 case CODEC_ID_8SVX_RAW: esc->table = NULL; break;
182 av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
183 return AVERROR_INVALIDDATA;
185 avctx->sample_fmt = AV_SAMPLE_FMT_U8;
190 static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
192 EightSvxContext *esc = avctx->priv_data;
194 av_freep(&esc->samples);
195 esc->samples_size = 0;
196 esc->samples_idx = 0;
201 AVCodec ff_eightsvx_fib_decoder = {
203 .type = AVMEDIA_TYPE_AUDIO,
204 .id = CODEC_ID_8SVX_FIB,
205 .priv_data_size = sizeof (EightSvxContext),
206 .init = eightsvx_decode_init,
207 .decode = eightsvx_decode_frame,
208 .close = eightsvx_decode_close,
209 .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
212 AVCodec ff_eightsvx_exp_decoder = {
214 .type = AVMEDIA_TYPE_AUDIO,
215 .id = CODEC_ID_8SVX_EXP,
216 .priv_data_size = sizeof (EightSvxContext),
217 .init = eightsvx_decode_init,
218 .decode = eightsvx_decode_frame,
219 .close = eightsvx_decode_close,
220 .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
223 AVCodec ff_pcm_s8_planar_decoder = {
224 .name = "pcm_s8_planar",
225 .type = AVMEDIA_TYPE_AUDIO,
226 .id = CODEC_ID_PCM_S8_PLANAR,
227 .priv_data_size = sizeof(EightSvxContext),
228 .init = eightsvx_decode_init,
229 .close = eightsvx_decode_close,
230 .decode = eightsvx_decode_frame,
231 .long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),