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);
73  ff_bswapdsp_init(&c->bdsp);
74  ff_mpadsp_init(&c->mpadsp);
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;
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;
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;
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 };
MPC7_SCFI_BITS
#define MPC7_SCFI_BITS
Definition: mpc7data.h:34
dscf_vlc
static VLC dscf_vlc
Definition: mpc7.c:39
mpc7_scfi
static const uint8_t mpc7_scfi[MPC7_SCFI_SIZE *2]
Definition: mpc7data.h:35
AVCodec
AVCodec.
Definition: avcodec.h:3481
skip_bits_long
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
scfi_vlc
static VLC scfi_vlc
Definition: mpc7.c:39
AVERROR
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
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2276
hdr_vlc
static VLC hdr_vlc
Definition: mpc7.c:39
av_lfg_init
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
Definition: lfg.c:32
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:686
ch
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
get_bits_count
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
ff_mpadsp_init
av_cold void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:31
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
internal.h
AVPacket::data
uint8_t * data
Definition: avcodec.h:1477
quant_vlc
static VLC quant_vlc[MPC7_QUANT_VLC_TABLES][2]
Definition: mpc7.c:39
init_vlc
#define init_vlc(vlc, nb_bits, nb_codes, bits, bits_wrap, bits_size, codes, codes_wrap, codes_size, flags)
Definition: vlc.h:38
quant_offsets
static const uint16_t quant_offsets[MPC7_QUANT_VLC_TABLES *2+1]
Definition: mpc7.c:41
table
static const uint16_t table[]
Definition: prosumer.c:206
data
const char data[16]
Definition: mxf.c:91
MPCContext
Definition: mpc.h:52
mpc7_decode_close
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
Definition: mpc7.c:323
get_vlc2
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
MPC7_HDR_BITS
#define MPC7_HDR_BITS
Definition: mpc7data.h:47
init_get_bits
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
mpc7_hdr
static const uint8_t mpc7_hdr[MPC7_HDR_SIZE *2]
Definition: mpc7data.h:48
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
VLC_TYPE
#define VLC_TYPE
Definition: vlc.h:24
GetBitContext
Definition: get_bits.h:61
ff_mpc_dequantize_and_synth
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
Definition: mpc.c:61
ff_mpc7_decoder
AVCodec ff_mpc7_decoder
Definition: mpc7.c:331
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:86
mpc7_quant_vlc
static const uint16_t mpc7_quant_vlc[MPC7_QUANT_VLC_TABLES][2][64 *2]
Definition: mpc7data.h:62
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
buf
void * buf
Definition: avisynth_c.h:766
av_cold
#define av_cold
Definition: attributes.h:84
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:1667
av_lfg_get
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
Definition: lfg.h:53
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
mpc7_idx32
static const int8_t mpc7_idx32[]
Definition: mpc7data.h:29
lfg.h
mpc7_decode_flush
static void mpc7_decode_flush(AVCodecContext *avctx)
Definition: mpc7.c:315
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
ff_mpc_init
av_cold void ff_mpc_init(void)
Definition: mpc.c:37
get_bits.h
bands
static const float bands[]
Definition: af_superequalizer.c:56
mpc.h
ff_bswapdsp_init
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
Definition: bswapdsp.c:49
flush
static void flush(AVCodecContext *avctx)
Definition: aacdec_template.c:500
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
mpc7_idx51
static const int8_t mpc7_idx51[]
Definition: mpc7data.h:31
MPC_FRAME_SIZE
#define MPC_FRAME_SIZE
Definition: mpc.h:41
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
mpc7_decode_frame
static int mpc7_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: mpc7.c:192
INIT_VLC_USE_NEW_STATIC
#define INIT_VLC_USE_NEW_STATIC
Definition: vlc.h:55
MPC7_HDR_SIZE
#define MPC7_HDR_SIZE
Definition: mpc7data.h:46
c
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
MPC7_QUANT_VLC_TABLES
#define MPC7_QUANT_VLC_TABLES
Definition: mpc7data.h:53
VLC::table_allocated
int table_allocated
Definition: vlc.h:29
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1965
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:981
AVPacket::size
int size
Definition: avcodec.h:1478
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:188
SAMPLES_PER_BAND
#define SAMPLES_PER_BAND
Definition: mpc.h:40
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2233
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
MPC7_SCFI_SIZE
#define MPC7_SCFI_SIZE
Definition: mpc7data.h:33
mb
#define mb
Definition: vf_colormatrix.c:101
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:2226
get_scale_idx
static int get_scale_idx(GetBitContext *gb, int ref)
Definition: mpc7.c:184
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
mpc7_idx31
static const int8_t mpc7_idx31[]
Definition: mpc7data.h:28
mpc7data.h
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1666
internal.h
BANDS
#define BANDS
Definition: imc.c:52
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
av_fast_padded_malloc
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
mpc7_decode_init
static av_cold int mpc7_decode_init(AVCodecContext *avctx)
Definition: mpc7.c:48
uint8_t
uint8_t
Definition: audio_convert.c:194
AVCodec::name
const char * name
Name of the codec implementation.
Definition: avcodec.h:3488
avcodec.h
mpc7_idx30
static const int8_t mpc7_idx30[]
Definition: mpc7data.h:27
ret
ret
Definition: filter_design.txt:187
frame
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
Definition: filter_design.txt:264
mpc7_dscf
static const uint8_t mpc7_dscf[MPC7_DSCF_SIZE *2]
Definition: mpc7data.h:41
mpc7_quant_vlc_sizes
static const uint8_t mpc7_quant_vlc_sizes[MPC7_QUANT_VLC_TABLES *2]
Definition: mpc7data.h:54
AVCodecContext
main external API structure.
Definition: avcodec.h:1565
mpc7_idx50
static const int8_t mpc7_idx50[]
Definition: mpc7data.h:30
channel_layout.h
Band
Subband structure - hold all variables for each subband.
Definition: mpc.h:44
VLC
Definition: vlc.h:26
MPC7_DSCF_BITS
#define MPC7_DSCF_BITS
Definition: mpc7data.h:40
MPC7_DSCF_SIZE
#define MPC7_DSCF_SIZE
Definition: mpc7data.h:39
ref
static int ref[MAX_W *MAX_W]
Definition: jpeg2000dwt.c:107
mpegaudiodsp.h
avpriv_request_sample
#define avpriv_request_sample(...)
Definition: tableprint_vlc.h:39
LOCAL_ALIGNED_16
#define LOCAL_ALIGNED_16(t, v,...)
Definition: internal.h:131
AVPacket
This structure stores compressed data.
Definition: avcodec.h:1454
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:1592
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
idx_to_quant
static void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
Fill samples for given subband.
Definition: mpc7.c:142
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
AV_CODEC_ID_MUSEPACK7
@ AV_CODEC_ID_MUSEPACK7
Definition: avcodec.h:592
mpc7_quant_vlc_off
static const uint8_t mpc7_quant_vlc_off[MPC7_QUANT_VLC_TABLES]
Definition: mpc7data.h:58
VLC::table
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28