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 {
47 /* buffer used to store the whole audio decoded/interleaved chunk,
48 * which is sent with the first packet */
54 static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
55 static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
57 #define MAX_FRAME_SIZE 2048
60 * Interleave samples in buffer containing all left channel samples
61 * at the beginning, and right channel samples at the end.
62 * Each sample is assumed to be in signed 8-bit format.
64 * @param size the size in bytes of the dst and src buffer
66 static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
68 uint8_t *dst_end = dst + size;
71 while (dst < dst_end) {
79 * Delta decode the compressed values in src, and put the resulting
80 * decoded n samples in dst.
82 * @param val starting value assumed by the delta sequence
83 * @param table delta sequence table
84 * @return size in bytes of the decoded data, must be src_size*2
86 static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
87 int8_t val, const int8_t *table)
94 val = av_clip(val + table[d & 0x0f], -127, 128);
96 val = av_clip(val + table[d >> 4] , -127, 128);
103 /** decode a frame */
104 static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
105 int *got_frame_ptr, AVPacket *avpkt)
107 EightSvxContext *esc = avctx->priv_data;
108 int n, out_data_size, ret;
111 /* decode and interleave the first packet */
112 if (!esc->samples && avpkt) {
113 uint8_t *deinterleaved_samples, *p = NULL;
115 esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR?
116 avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
117 if (!(esc->samples = av_malloc(esc->samples_size)))
118 return AVERROR(ENOMEM);
121 if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
122 const uint8_t *buf = avpkt->data;
123 int buf_size = avpkt->size;
124 int n = esc->samples_size;
127 av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
128 return AVERROR(EINVAL);
130 if (!(deinterleaved_samples = av_mallocz(n)))
131 return AVERROR(ENOMEM);
132 p = deinterleaved_samples;
134 /* the uncompressed starting value is contained in the first byte */
135 if (avctx->channels == 2) {
136 delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table);
138 delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
140 delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table);
142 deinterleaved_samples = avpkt->data;
145 if (avctx->channels == 2)
146 interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
148 memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
152 /* get output buffer */
153 esc->frame.nb_samples = (FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) +avctx->channels-1) / avctx->channels;
154 if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) {
155 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
160 *(AVFrame *)data = esc->frame;
162 dst = esc->frame.data[0];
163 src = esc->samples + esc->samples_idx;
164 out_data_size = esc->frame.nb_samples * avctx->channels;
165 for (n = out_data_size; n > 0; n--)
166 *dst++ = *src++ + 128;
167 esc->samples_idx += out_data_size;
169 return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
170 (avctx->frame_number == 0)*2 + out_data_size / 2 :
174 static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
176 EightSvxContext *esc = avctx->priv_data;
178 if (avctx->channels < 1 || avctx->channels > 2) {
179 av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
180 return AVERROR_INVALIDDATA;
183 switch (avctx->codec->id) {
184 case CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
185 case CODEC_ID_8SVX_EXP: esc->table = exponential; break;
186 case CODEC_ID_PCM_S8_PLANAR:
187 case CODEC_ID_8SVX_RAW: esc->table = NULL; break;
189 av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
190 return AVERROR_INVALIDDATA;
192 avctx->sample_fmt = AV_SAMPLE_FMT_U8;
194 avcodec_get_frame_defaults(&esc->frame);
195 avctx->coded_frame = &esc->frame;
200 static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
202 EightSvxContext *esc = avctx->priv_data;
204 av_freep(&esc->samples);
205 esc->samples_size = 0;
206 esc->samples_idx = 0;
211 AVCodec ff_eightsvx_fib_decoder = {
213 .type = AVMEDIA_TYPE_AUDIO,
214 .id = CODEC_ID_8SVX_FIB,
215 .priv_data_size = sizeof (EightSvxContext),
216 .init = eightsvx_decode_init,
217 .decode = eightsvx_decode_frame,
218 .close = eightsvx_decode_close,
219 .capabilities = CODEC_CAP_DR1,
220 .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
223 AVCodec ff_eightsvx_exp_decoder = {
225 .type = AVMEDIA_TYPE_AUDIO,
226 .id = CODEC_ID_8SVX_EXP,
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 exponential"),
235 AVCodec ff_pcm_s8_planar_decoder = {
236 .name = "pcm_s8_planar",
237 .type = AVMEDIA_TYPE_AUDIO,
238 .id = CODEC_ID_PCM_S8_PLANAR,
239 .priv_data_size = sizeof(EightSvxContext),
240 .init = eightsvx_decode_init,
241 .close = eightsvx_decode_close,
242 .decode = eightsvx_decode_frame,
243 .capabilities = CODEC_CAP_DR1,
244 .long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),