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mpc7.c
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1 /*
2  * Musepack SV7 decoder
3  * Copyright (c) 2006 Konstantin Shishkov
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  * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
25  * divided into 32 subbands.
26  */
27 
29 #include "libavutil/internal.h"
30 #include "libavutil/lfg.h"
31 #include "avcodec.h"
32 #include "get_bits.h"
33 #include "internal.h"
34 #include "mpegaudiodsp.h"
35 
36 #include "mpc.h"
37 #include "mpc7data.h"
38 
40 
41 static const uint16_t quant_offsets[MPC7_QUANT_VLC_TABLES*2 + 1] =
42 {
43  0, 512, 1024, 1536, 2052, 2564, 3076, 3588, 4100, 4612, 5124,
44  5636, 6164, 6676, 7224
45 };
46 
47 
49 {
50  int i, j, ret;
51  MPCContext *c = avctx->priv_data;
52  GetBitContext gb;
54  static int vlc_initialized = 0;
55 
56  static VLC_TYPE scfi_table[1 << MPC7_SCFI_BITS][2];
57  static VLC_TYPE dscf_table[1 << MPC7_DSCF_BITS][2];
58  static VLC_TYPE hdr_table[1 << MPC7_HDR_BITS][2];
59  static VLC_TYPE quant_tables[7224][2];
60 
61  /* Musepack SV7 is always stereo */
62  if (avctx->channels != 2) {
63  avpriv_request_sample(avctx, "%d channels", avctx->channels);
64  return AVERROR_PATCHWELCOME;
65  }
66 
67  if(avctx->extradata_size < 16){
68  av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
69  return AVERROR_INVALIDDATA;
70  }
71  memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
72  av_lfg_init(&c->rnd, 0xDEADBEEF);
75  c->bdsp.bswap_buf((uint32_t *) buf, (const uint32_t *) avctx->extradata, 4);
76  ff_mpc_init();
77  init_get_bits(&gb, buf, 128);
78 
79  c->IS = get_bits1(&gb);
80  c->MSS = get_bits1(&gb);
81  c->maxbands = get_bits(&gb, 6);
82  if(c->maxbands >= BANDS){
83  av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->maxbands);
84  return AVERROR_INVALIDDATA;
85  }
86  skip_bits_long(&gb, 88);
87  c->gapless = get_bits1(&gb);
88  c->lastframelen = get_bits(&gb, 11);
89  av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n",
90  c->IS, c->MSS, c->gapless, c->lastframelen, c->maxbands);
91  c->frames_to_skip = 0;
92 
95 
96  if(vlc_initialized) return 0;
97  av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
98  scfi_vlc.table = scfi_table;
99  scfi_vlc.table_allocated = 1 << MPC7_SCFI_BITS;
100  if ((ret = init_vlc(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
101  &mpc7_scfi[1], 2, 1,
102  &mpc7_scfi[0], 2, 1, INIT_VLC_USE_NEW_STATIC))) {
103  av_log(avctx, AV_LOG_ERROR, "Cannot init SCFI VLC\n");
104  return ret;
105  }
106  dscf_vlc.table = dscf_table;
107  dscf_vlc.table_allocated = 1 << MPC7_DSCF_BITS;
108  if ((ret = init_vlc(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
109  &mpc7_dscf[1], 2, 1,
110  &mpc7_dscf[0], 2, 1, INIT_VLC_USE_NEW_STATIC))) {
111  av_log(avctx, AV_LOG_ERROR, "Cannot init DSCF VLC\n");
112  return ret;
113  }
114  hdr_vlc.table = hdr_table;
115  hdr_vlc.table_allocated = 1 << MPC7_HDR_BITS;
116  if ((ret = init_vlc(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
117  &mpc7_hdr[1], 2, 1,
118  &mpc7_hdr[0], 2, 1, INIT_VLC_USE_NEW_STATIC))) {
119  av_log(avctx, AV_LOG_ERROR, "Cannot init HDR VLC\n");
120  return ret;
121  }
122  for(i = 0; i < MPC7_QUANT_VLC_TABLES; i++){
123  for(j = 0; j < 2; j++){
124  quant_vlc[i][j].table = &quant_tables[quant_offsets[i*2 + j]];
125  quant_vlc[i][j].table_allocated = quant_offsets[i*2 + j + 1] - quant_offsets[i*2 + j];
126  if ((ret = init_vlc(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i],
127  &mpc7_quant_vlc[i][j][1], 4, 2,
128  &mpc7_quant_vlc[i][j][0], 4, 2, INIT_VLC_USE_NEW_STATIC))) {
129  av_log(avctx, AV_LOG_ERROR, "Cannot init QUANT VLC %i,%i\n",i,j);
130  return ret;
131  }
132  }
133  }
134  vlc_initialized = 1;
135 
136  return 0;
137 }
138 
139 /**
140  * Fill samples for given subband
141  */
142 static inline void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
143 {
144  int i, i1, t;
145  switch(idx){
146  case -1:
147  for(i = 0; i < SAMPLES_PER_BAND; i++){
148  *dst++ = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
149  }
150  break;
151  case 1:
152  i1 = get_bits1(gb);
153  for(i = 0; i < SAMPLES_PER_BAND/3; i++){
154  t = get_vlc2(gb, quant_vlc[0][i1].table, 9, 2);
155  *dst++ = mpc7_idx30[t];
156  *dst++ = mpc7_idx31[t];
157  *dst++ = mpc7_idx32[t];
158  }
159  break;
160  case 2:
161  i1 = get_bits1(gb);
162  for(i = 0; i < SAMPLES_PER_BAND/2; i++){
163  t = get_vlc2(gb, quant_vlc[1][i1].table, 9, 2);
164  *dst++ = mpc7_idx50[t];
165  *dst++ = mpc7_idx51[t];
166  }
167  break;
168  case 3: case 4: case 5: case 6: case 7:
169  i1 = get_bits1(gb);
170  for(i = 0; i < SAMPLES_PER_BAND; i++)
171  *dst++ = get_vlc2(gb, quant_vlc[idx-1][i1].table, 9, 2) - mpc7_quant_vlc_off[idx-1];
172  break;
173  case 8: case 9: case 10: case 11: case 12:
174  case 13: case 14: case 15: case 16: case 17:
175  t = (1 << (idx - 2)) - 1;
176  for(i = 0; i < SAMPLES_PER_BAND; i++)
177  *dst++ = get_bits(gb, idx - 1) - t;
178  break;
179  default: // case 0 and -2..-17
180  return;
181  }
182 }
183 
184 static int get_scale_idx(GetBitContext *gb, int ref)
185 {
186  int t = get_vlc2(gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
187  if (t == 8)
188  return get_bits(gb, 6);
189  return ref + t;
190 }
191 
192 static int mpc7_decode_frame(AVCodecContext * avctx, void *data,
193  int *got_frame_ptr, AVPacket *avpkt)
194 {
195  AVFrame *frame = data;
196  const uint8_t *buf = avpkt->data;
197  int buf_size;
198  MPCContext *c = avctx->priv_data;
199  GetBitContext gb;
200  int i, ch;
201  int mb = -1;
202  Band *bands = c->bands;
203  int off, ret, last_frame, skip;
204  int bits_used, bits_avail;
205 
206  memset(bands, 0, sizeof(*bands) * (c->maxbands + 1));
207 
208  buf_size = avpkt->size & ~3;
209  if (buf_size <= 0) {
210  av_log(avctx, AV_LOG_ERROR, "packet size is too small (%i bytes)\n",
211  avpkt->size);
212  return AVERROR_INVALIDDATA;
213  }
214  if (buf_size != avpkt->size) {
215  av_log(avctx, AV_LOG_WARNING, "packet size is not a multiple of 4. "
216  "extra bytes at the end will be skipped.\n");
217  }
218 
219  skip = buf[0];
220  last_frame = buf[1];
221  buf += 4;
222  buf_size -= 4;
223 
224  /* get output buffer */
225  frame->nb_samples = MPC_FRAME_SIZE;
226  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
227  return ret;
228 
229  av_fast_padded_malloc(&c->bits, &c->buf_size, buf_size);
230  if (!c->bits)
231  return AVERROR(ENOMEM);
232  c->bdsp.bswap_buf((uint32_t *) c->bits, (const uint32_t *) buf,
233  buf_size >> 2);
234  if ((ret = init_get_bits8(&gb, c->bits, buf_size)) < 0)
235  return ret;
236  skip_bits_long(&gb, skip);
237 
238  /* read subband indexes */
239  for(i = 0; i <= c->maxbands; i++){
240  for(ch = 0; ch < 2; ch++){
241  int t = 4;
242  if(i) t = get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) - 5;
243  if(t == 4) bands[i].res[ch] = get_bits(&gb, 4);
244  else bands[i].res[ch] = bands[i-1].res[ch] + t;
245  if (bands[i].res[ch] < -1 || bands[i].res[ch] > 17) {
246  av_log(avctx, AV_LOG_ERROR, "subband index invalid\n");
247  return AVERROR_INVALIDDATA;
248  }
249  }
250 
251  if(bands[i].res[0] || bands[i].res[1]){
252  mb = i;
253  if(c->MSS) bands[i].msf = get_bits1(&gb);
254  }
255  }
256  /* get scale indexes coding method */
257  for(i = 0; i <= mb; i++)
258  for(ch = 0; ch < 2; ch++)
259  if(bands[i].res[ch]) bands[i].scfi[ch] = get_vlc2(&gb, scfi_vlc.table, MPC7_SCFI_BITS, 1);
260  /* get scale indexes */
261  for(i = 0; i <= mb; i++){
262  for(ch = 0; ch < 2; ch++){
263  if(bands[i].res[ch]){
264  bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i];
265  bands[i].scf_idx[ch][0] = get_scale_idx(&gb, bands[i].scf_idx[ch][2]);
266  switch(bands[i].scfi[ch]){
267  case 0:
268  bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
269  bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
270  break;
271  case 1:
272  bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
273  bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1];
274  break;
275  case 2:
276  bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
277  bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
278  break;
279  case 3:
280  bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
281  break;
282  }
283  c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2];
284  }
285  }
286  }
287  /* get quantizers */
288  memset(c->Q, 0, sizeof(c->Q));
289  off = 0;
290  for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
291  for(ch = 0; ch < 2; ch++)
292  idx_to_quant(c, &gb, bands[i].res[ch], c->Q[ch] + off);
293 
294  ff_mpc_dequantize_and_synth(c, mb, (int16_t **)frame->extended_data, 2);
295  if(last_frame)
296  frame->nb_samples = c->lastframelen;
297 
298  bits_used = get_bits_count(&gb);
299  bits_avail = buf_size * 8;
300  if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
301  av_log(avctx, AV_LOG_ERROR, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
302  return AVERROR_INVALIDDATA;
303  }
304  if(c->frames_to_skip){
305  c->frames_to_skip--;
306  *got_frame_ptr = 0;
307  return avpkt->size;
308  }
309 
310  *got_frame_ptr = 1;
311 
312  return avpkt->size;
313 }
314 
316 {
317  MPCContext *c = avctx->priv_data;
318 
319  memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
320  c->frames_to_skip = 32;
321 }
322 
324 {
325  MPCContext *c = avctx->priv_data;
326  av_freep(&c->bits);
327  c->buf_size = 0;
328  return 0;
329 }
330 
332  .name = "mpc7",
333  .long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
334  .type = AVMEDIA_TYPE_AUDIO,
335  .id = AV_CODEC_ID_MUSEPACK7,
336  .priv_data_size = sizeof(MPCContext),
338  .close = mpc7_decode_close,
341  .capabilities = AV_CODEC_CAP_DR1,
342  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
344 };
void(* bswap_buf)(uint32_t *dst, const uint32_t *src, int w)
Definition: bswapdsp.h:25
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
Definition: mpc.c:61
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:201
static const int8_t mpc7_idx32[]
Definition: mpc7data.h:29
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
static void flush(AVCodecContext *avctx)
MPADSPContext mpadsp
Definition: mpc.h:54
static const uint16_t mpc7_quant_vlc[MPC7_QUANT_VLC_TABLES][2][64 *2]
Definition: mpc7data.h:62
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:261
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static void skip_bits_long(GetBitContext *s, int n)
Definition: get_bits.h:204
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
int msf
mid-stereo flag
Definition: mpc.h:45
int Q[2][MPC_FRAME_SIZE]
Definition: mpc.h:61
int size
Definition: avcodec.h:1680
int res[2]
Definition: mpc.h:46
AVLFG rnd
Definition: mpc.h:65
void av_fast_padded_malloc(void *ptr, unsigned int *size, size_t min_size)
Same behaviour av_fast_malloc but the buffer has additional AV_INPUT_BUFFER_PADDING_SIZE at the end w...
Definition: utils.c:121
#define AV_CH_LAYOUT_STEREO
static VLC dscf_vlc
Definition: mpc7.c:39
AVCodec.
Definition: avcodec.h:3739
static const uint8_t mpc7_quant_vlc_sizes[MPC7_QUANT_VLC_TABLES *2]
Definition: mpc7data.h:54
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
int buf_size
Definition: mpc.h:64
int lastframelen
Definition: mpc.h:56
static int mpc7_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: mpc7.c:192
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
int gapless
Definition: mpc.h:55
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2531
uint8_t
#define av_cold
Definition: attributes.h:82
#define mb
#define MPC7_DSCF_BITS
Definition: mpc7data.h:40
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_X86ASM &&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
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1876
static AVFrame * frame
int scf_idx[2][3]
Definition: mpc.h:48
uint8_t * data
Definition: avcodec.h:1679
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:199
bitstream reader API header.
static const int8_t mpc7_idx50[]
Definition: mpc7data.h:30
#define av_log(a,...)
static const int8_t mpc7_idx31[]
Definition: mpc7data.h:28
#define MPC7_HDR_SIZE
Definition: mpc7data.h:46
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int IS
Definition: mpc.h:55
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
Definition: vlc.h:38
static const uint8_t mpc7_hdr[MPC7_HDR_SIZE *2]
Definition: mpc7data.h:48
#define MPC7_QUANT_VLC_TABLES
Definition: mpc7data.h:53
#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:179
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
static const uint8_t mpc7_quant_vlc_off[MPC7_QUANT_VLC_TABLES]
Definition: mpc7data.h:58
const char * name
Name of the codec implementation.
Definition: avcodec.h:3746
static const uint8_t mpc7_scfi[MPC7_SCFI_SIZE *2]
Definition: mpc7data.h:35
int maxbands
Definition: mpc.h:57
#define SAMPLES_PER_BAND
Definition: mpc.h:40
Definition: vlc.h:26
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2574
static av_cold int mpc7_decode_init(AVCodecContext *avctx)
Definition: mpc7.c:48
common internal API header
audio channel layout utility functions
uint8_t * bits
Definition: mpc.h:63
#define MPC_FRAME_SIZE
Definition: mpc.h:41
static void mpc7_decode_flush(AVCodecContext *avctx)
Definition: mpc7.c:315
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:554
#define MPC7_SCFI_BITS
Definition: mpc7data.h:34
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
int table_allocated
Definition: vlc.h:29
BswapDSPContext bdsp
Definition: mpc.h:53
Libavcodec external API header.
Musepack decoder MPEG Audio Layer 1/2 -like codec with frames of 1152 samples divided into 32 subband...
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
static const int8_t mpc7_idx51[]
Definition: mpc7data.h:31
av_cold void ff_mpc_init(void)
Definition: mpc.c:37
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:456
int scfi[2]
Definition: mpc.h:47
main external API structure.
Definition: avcodec.h:1761
static VLC hdr_vlc
Definition: mpc7.c:39
AVCodec ff_mpc7_decoder
Definition: mpc7.c:331
static const float bands[]
static void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
Fill samples for given subband.
Definition: mpc7.c:142
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1669
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
Definition: lfg.h:47
void * buf
Definition: avisynth_c.h:690
int extradata_size
Definition: avcodec.h:1877
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:313
int oldDSCF[2][BANDS]
Definition: mpc.h:59
#define MPC7_HDR_BITS
Definition: mpc7data.h:47
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:425
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
Definition: lfg.c:32
#define BANDS
Definition: imc.c:52
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
Definition: mpc7.c:323
#define MPC7_SCFI_SIZE
Definition: mpc7data.h:33
Band bands[BANDS]
Definition: mpc.h:60
common internal api header.
static int ref[MAX_W *MAX_W]
Definition: jpeg2000dwt.c:107
static double c[64]
#define INIT_VLC_USE_NEW_STATIC
Definition: vlc.h:55
static const uint16_t quant_offsets[MPC7_QUANT_VLC_TABLES *2+1]
Definition: mpc7.c:41
int MSS
Definition: mpc.h:55
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
Definition: bswapdsp.c:49
void * priv_data
Definition: avcodec.h:1803
static const int8_t mpc7_idx30[]
Definition: mpc7data.h:27
static VLC scfi_vlc
Definition: mpc7.c:39
int channels
number of audio channels
Definition: avcodec.h:2524
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define LOCAL_ALIGNED_16(t, v,...)
Definition: internal.h:124
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:67
#define VLC_TYPE
Definition: vlc.h:24
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:248
static const uint8_t mpc7_dscf[MPC7_DSCF_SIZE *2]
Definition: mpc7data.h:41
static int get_scale_idx(GetBitContext *gb, int ref)
Definition: mpc7.c:184
Subband structure - hold all variables for each subband.
Definition: mpc.h:44
This structure stores compressed data.
Definition: avcodec.h:1656
av_cold void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:31
#define MPC7_DSCF_SIZE
Definition: mpc7data.h:39
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:1002
static VLC quant_vlc[MPC7_QUANT_VLC_TABLES][2]
Definition: mpc7.c:39
Definition: mpc.h:52
int frames_to_skip
Definition: mpc.h:66