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/
40 #include "libavutil/avassert.h"
42 #include "libavutil/common.h"
44 /** decoder context */
45 typedef struct EightSvxContext {
49 /* buffer used to store the whole audio decoded/interleaved chunk,
50 * which is sent with the first packet */
56 static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
57 static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
59 #define MAX_FRAME_SIZE 2048
62 * Interleave samples in buffer containing all left channel samples
63 * at the beginning, and right channel samples at the end.
64 * Each sample is assumed to be in signed 8-bit format.
66 * @param size the size in bytes of the dst and src buffer
68 static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
70 uint8_t *dst_end = dst + size;
73 while (dst < dst_end) {
81 * Delta decode the compressed values in src, and put the resulting
82 * decoded n samples in dst.
84 * @param val starting value assumed by the delta sequence
85 * @param table delta sequence table
86 * @return size in bytes of the decoded data, must be src_size*2
88 static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
89 int8_t val, const int8_t *table)
96 val = av_clip(val + table[d & 0x0f], -127, 128);
98 val = av_clip(val + table[d >> 4] , -127, 128);
105 /** decode a frame */
106 static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
107 int *got_frame_ptr, AVPacket *avpkt)
109 EightSvxContext *esc = avctx->priv_data;
110 int n, out_data_size, ret;
113 /* decode and interleave the first packet */
114 if (!esc->samples && avpkt) {
115 uint8_t *deinterleaved_samples, *p = NULL;
116 int packet_size = avpkt->size;
118 if (packet_size % avctx->channels) {
119 av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n");
120 if (packet_size < avctx->channels)
122 packet_size -= packet_size % avctx->channels;
124 esc->samples_size = !esc->table ?
125 packet_size : avctx->channels + (packet_size-avctx->channels) * 2;
126 if (!(esc->samples = av_malloc(esc->samples_size)))
127 return AVERROR(ENOMEM);
131 const uint8_t *buf = avpkt->data;
133 int buf_size = avpkt->size;
134 int i, n = esc->samples_size;
137 av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
138 return AVERROR(EINVAL);
140 if (!(deinterleaved_samples = av_mallocz(n)))
141 return AVERROR(ENOMEM);
142 dst = p = deinterleaved_samples;
144 /* the uncompressed starting value is contained in the first byte */
145 dst = deinterleaved_samples;
146 for (i = 0; i < avctx->channels; i++) {
147 delta_decode(dst, buf + 1, buf_size / avctx->channels - 1, buf[0], esc->table);
148 buf += buf_size / avctx->channels;
149 dst += n / avctx->channels - 1;
152 deinterleaved_samples = avpkt->data;
155 if (avctx->channels == 2)
156 interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
158 memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
162 /* get output buffer */
163 av_assert1(!(esc->samples_size % avctx->channels || esc->samples_idx % avctx->channels));
164 esc->frame.nb_samples = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) / avctx->channels;
165 if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) {
166 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
171 *(AVFrame *)data = esc->frame;
173 dst = esc->frame.data[0];
174 src = esc->samples + esc->samples_idx;
175 out_data_size = esc->frame.nb_samples * avctx->channels;
176 for (n = out_data_size; n > 0; n--)
177 *dst++ = *src++ + 128;
178 esc->samples_idx += out_data_size;
181 (avctx->frame_number == 0)*2 + out_data_size / 2 :
185 static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
187 EightSvxContext *esc = avctx->priv_data;
189 if (avctx->channels < 1 || avctx->channels > 2) {
190 av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
191 return AVERROR_INVALIDDATA;
194 switch (avctx->codec->id) {
195 case AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
196 case AV_CODEC_ID_8SVX_EXP: esc->table = exponential; break;
197 case AV_CODEC_ID_PCM_S8_PLANAR:
198 case AV_CODEC_ID_8SVX_RAW: esc->table = NULL; break;
200 av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
201 return AVERROR_INVALIDDATA;
203 avctx->sample_fmt = AV_SAMPLE_FMT_U8;
205 avcodec_get_frame_defaults(&esc->frame);
206 avctx->coded_frame = &esc->frame;
211 static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
213 EightSvxContext *esc = avctx->priv_data;
215 av_freep(&esc->samples);
216 esc->samples_size = 0;
217 esc->samples_idx = 0;
222 #if CONFIG_EIGHTSVX_FIB_DECODER
223 AVCodec ff_eightsvx_fib_decoder = {
225 .type = AVMEDIA_TYPE_AUDIO,
226 .id = AV_CODEC_ID_8SVX_FIB,
227 .priv_data_size = sizeof (EightSvxContext),
228 .init = eightsvx_decode_init,
229 .decode = eightsvx_decode_frame,
230 .close = eightsvx_decode_close,
231 .capabilities = CODEC_CAP_DR1,
232 .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
235 #if CONFIG_EIGHTSVX_EXP_DECODER
236 AVCodec ff_eightsvx_exp_decoder = {
238 .type = AVMEDIA_TYPE_AUDIO,
239 .id = AV_CODEC_ID_8SVX_EXP,
240 .priv_data_size = sizeof (EightSvxContext),
241 .init = eightsvx_decode_init,
242 .decode = eightsvx_decode_frame,
243 .close = eightsvx_decode_close,
244 .capabilities = CODEC_CAP_DR1,
245 .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
248 #if CONFIG_PCM_S8_PLANAR_DECODER
249 AVCodec ff_pcm_s8_planar_decoder = {
250 .name = "pcm_s8_planar",
251 .type = AVMEDIA_TYPE_AUDIO,
252 .id = AV_CODEC_ID_PCM_S8_PLANAR,
253 .priv_data_size = sizeof(EightSvxContext),
254 .init = eightsvx_decode_init,
255 .close = eightsvx_decode_close,
256 .decode = eightsvx_decode_frame,
257 .capabilities = CODEC_CAP_DR1,
258 .long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),