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8svx.c
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1 /*
2  * Copyright (C) 2008 Jaikrishnan Menon
3  * Copyright (C) 2011 Stefano Sabatini
4  *
5  * This file is part of FFmpeg.
6  *
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.
11  *
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.
16  *
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
20  */
21 
22 /**
23  * @file
24  * 8svx audio decoder
25  * @author Jaikrishnan Menon
26  *
27  * supports: fibonacci delta encoding
28  * : exponential encoding
29  *
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
35  *
36  * Samples can be found here:
37  * http://aminet.net/mods/smpl/
38  */
39 
40 #include "libavutil/avassert.h"
41 #include "avcodec.h"
42 #include "internal.h"
43 #include "libavutil/common.h"
44 
45 /** decoder context */
46 typedef struct EightSvxContext {
48  const int8_t *table;
49 
50  /* buffer used to store the whole first packet.
51  data is only sent as one large packet */
53  int data_size;
54  int data_idx;
56 
57 static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
58 static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
59 
60 #define MAX_FRAME_SIZE 2048
61 
62 /**
63  * Delta decode the compressed values in src, and put the resulting
64  * decoded samples in dst.
65  *
66  * @param[in,out] state starting value. it is saved for use in the next call.
67  * @param table delta sequence table
68  */
69 static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size,
70  uint8_t *state, const int8_t *table)
71 {
72  uint8_t val = *state;
73 
74  while (src_size--) {
75  uint8_t d = *src++;
76  val = av_clip_uint8(val + table[d & 0xF]);
77  *dst++ = val;
78  val = av_clip_uint8(val + table[d >> 4]);
79  *dst++ = val;
80  }
81 
82  *state = val;
83 }
84 
85 /** decode a frame */
86 static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
87  int *got_frame_ptr, AVPacket *avpkt)
88 {
89  EightSvxContext *esc = avctx->priv_data;
90  AVFrame *frame = data;
91  int buf_size;
92  int ch, ret;
93  int hdr_size = 2;
94 
95  /* decode and interleave the first packet */
96  if (!esc->data[0] && avpkt) {
97  int chan_size = avpkt->size / avctx->channels - hdr_size;
98 
99  if (avpkt->size % avctx->channels) {
100  av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n");
101  }
102  if (avpkt->size < (hdr_size + 1) * avctx->channels) {
103  av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
104  return AVERROR_INVALIDDATA;
105  }
106 
107  esc->fib_acc[0] = avpkt->data[1] + 128;
108  if (avctx->channels == 2)
109  esc->fib_acc[1] = avpkt->data[2+chan_size+1] + 128;
110 
111  esc->data_idx = 0;
112  esc->data_size = chan_size;
113  if (!(esc->data[0] = av_malloc(chan_size)))
114  return AVERROR(ENOMEM);
115  if (avctx->channels == 2) {
116  if (!(esc->data[1] = av_malloc(chan_size))) {
117  av_freep(&esc->data[0]);
118  return AVERROR(ENOMEM);
119  }
120  }
121  memcpy(esc->data[0], &avpkt->data[hdr_size], chan_size);
122  if (avctx->channels == 2)
123  memcpy(esc->data[1], &avpkt->data[2*hdr_size+chan_size], chan_size);
124  }
125  if (!esc->data[0]) {
126  av_log(avctx, AV_LOG_ERROR, "unexpected empty packet\n");
127  return AVERROR_INVALIDDATA;
128  }
129 
130  /* decode next piece of data from the buffer */
131  buf_size = FFMIN(MAX_FRAME_SIZE, esc->data_size - esc->data_idx);
132  if (buf_size <= 0) {
133  *got_frame_ptr = 0;
134  return avpkt->size;
135  }
136 
137  /* get output buffer */
138  frame->nb_samples = buf_size * 2;
139  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
140  return ret;
141 
142  for (ch = 0; ch < avctx->channels; ch++) {
143  delta_decode(frame->data[ch], &esc->data[ch][esc->data_idx],
144  buf_size, &esc->fib_acc[ch], esc->table);
145  }
146 
147  esc->data_idx += buf_size;
148 
149  *got_frame_ptr = 1;
150 
151  return ((avctx->frame_number == 0)*hdr_size + buf_size)*avctx->channels;
152 }
153 
155 {
156  EightSvxContext *esc = avctx->priv_data;
157 
158  if (avctx->channels < 1 || avctx->channels > 2) {
159  av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
160  return AVERROR_INVALIDDATA;
161  }
162 
163  switch (avctx->codec->id) {
164  case AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
165  case AV_CODEC_ID_8SVX_EXP: esc->table = exponential; break;
166  default:
167  av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
168  return AVERROR_INVALIDDATA;
169  }
170  avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
171 
172  return 0;
173 }
174 
176 {
177  EightSvxContext *esc = avctx->priv_data;
178 
179  av_freep(&esc->data[0]);
180  av_freep(&esc->data[1]);
181  esc->data_size = 0;
182  esc->data_idx = 0;
183 
184  return 0;
185 }
186 
187 #if CONFIG_EIGHTSVX_FIB_DECODER
188 AVCodec ff_eightsvx_fib_decoder = {
189  .name = "8svx_fib",
190  .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
191  .type = AVMEDIA_TYPE_AUDIO,
192  .id = AV_CODEC_ID_8SVX_FIB,
193  .priv_data_size = sizeof (EightSvxContext),
196  .close = eightsvx_decode_close,
197  .capabilities = AV_CODEC_CAP_DR1,
198  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
200 };
201 #endif
202 #if CONFIG_EIGHTSVX_EXP_DECODER
203 AVCodec ff_eightsvx_exp_decoder = {
204  .name = "8svx_exp",
205  .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
206  .type = AVMEDIA_TYPE_AUDIO,
207  .id = AV_CODEC_ID_8SVX_EXP,
208  .priv_data_size = sizeof (EightSvxContext),
211  .close = eightsvx_decode_close,
212  .capabilities = AV_CODEC_CAP_DR1,
213  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
215 };
216 #endif
const struct AVCodec * codec
Definition: avcodec.h:1691
const char const char void * val
Definition: avisynth_c.h:771
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
static const int8_t fibonacci[16]
Definition: 8svx.c:57
This structure describes decoded (raw) audio or video data.
Definition: frame.h:190
#define MAX_FRAME_SIZE
Definition: 8svx.c:60
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
int size
Definition: avcodec.h:1608
#define src
Definition: vp8dsp.c:254
AVCodec.
Definition: avcodec.h:3607
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2452
uint8_t
#define av_cold
Definition: attributes.h:82
#define av_malloc(s)
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
decode a frame
Definition: 8svx.c:86
static AVFrame * frame
uint8_t * data
Definition: avcodec.h:1607
static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
Definition: 8svx.c:175
static const int8_t exponential[16]
Definition: 8svx.c:58
#define av_log(a,...)
enum AVCodecID id
Definition: avcodec.h:3621
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define AVERROR(e)
Definition: error.h:43
static const struct endianess table[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size, uint8_t *state, const int8_t *table)
Delta decode the compressed values in src, and put the resulting decoded samples in dst...
Definition: 8svx.c:69
simple assert() macros that are a bit more flexible than ISO C assert().
const char * name
Name of the codec implementation.
Definition: avcodec.h:3614
#define FFMIN(a, b)
Definition: common.h:96
const int8_t * table
Definition: 8svx.c:48
unsigned 8 bits, planar
Definition: samplefmt.h:66
int data_idx
Definition: 8svx.c:54
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
main external API structure.
Definition: avcodec.h:1682
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:948
static struct @246 state
uint8_t fib_acc[2]
Definition: 8svx.c:47
int data_size
Definition: 8svx.c:53
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:204
decoder context
Definition: 8svx.c:46
common internal api header.
common internal and external API header
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
Definition: 8svx.c:154
void * priv_data
Definition: avcodec.h:1724
uint8_t * data[2]
Definition: 8svx.c:52
int channels
number of audio channels
Definition: avcodec.h:2445
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
int frame_number
Frame counter, set by libavcodec.
Definition: avcodec.h:2475
#define av_freep(p)
static int decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame, AVPacket *pkt)
Definition: ffmpeg.c:2045
This structure stores compressed data.
Definition: avcodec.h:1584
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:247
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:959