FFmpeg
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 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:268
static const int8_t mpc7_idx32[]
Definition: mpc7data.h:29
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
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:379
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
#define avpriv_request_sample(...)
int msf
mid-stereo flag
Definition: mpc.h:45
int Q[2][MPC_FRAME_SIZE]
Definition: mpc.h:61
int size
Definition: avcodec.h:1478
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8: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);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=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){int planes=out->planar?out->ch_count:1;unsigned m=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){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
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:70
#define AV_CH_LAYOUT_STEREO
static VLC dscf_vlc
Definition: mpc7.c:39
AVCodec.
Definition: avcodec.h:3477
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
int gapless
Definition: mpc.h:55
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2229
uint8_t
#define av_cold
Definition: attributes.h:82
#define mb
void(* bswap_buf)(uint32_t *dst, const uint32_t *src, int w)
Definition: bswapdsp.h:25
#define MPC7_DSCF_BITS
Definition: mpc7data.h:40
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1666
int scf_idx[2][3]
Definition: mpc.h:48
uint8_t * data
Definition: avcodec.h:1477
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
bitstream reader API header.
static const int8_t mpc7_idx50[]
Definition: mpc7data.h:30
#define av_log(a,...)
static const uint16_t table[]
Definition: prosumer.c:206
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:260
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 NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
#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:3484
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:2272
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
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
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:797
#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:677
int scfi[2]
Definition: mpc.h:47
main external API structure.
Definition: avcodec.h:1565
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:1964
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:766
int extradata_size
Definition: avcodec.h:1667
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
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:659
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
#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:1592
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:2222
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:131
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:67
#define VLC_TYPE
Definition: vlc.h:24
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:315
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:1454
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:334
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:981
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