FFmpeg
mpc7.c
Go to the documentation of this file.
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_sizes[MPC7_QUANT_VLC_TABLES*2] =
42 {
43  512, 512, 512, 516, 512, 512, 512, 512, 512, 512, 512, 528, 512, 548
44 };
45 
46 
48 {
49  int i, j;
50  MPCContext *c = avctx->priv_data;
51  GetBitContext gb;
52  LOCAL_ALIGNED_16(uint8_t, buf, [16]);
53  static int vlc_initialized = 0;
54 
55  static VLC_TYPE quant_tables[7224][2];
56  VLC_TYPE (*quant_table)[2] = quant_tables;
57  const uint16_t *raw_quant_table = mpc7_quant_vlcs;
58 
59  /* Musepack SV7 is always stereo */
60  if (avctx->channels != 2) {
61  avpriv_request_sample(avctx, "%d channels", avctx->channels);
62  return AVERROR_PATCHWELCOME;
63  }
64 
65  if(avctx->extradata_size < 16){
66  av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
67  return AVERROR_INVALIDDATA;
68  }
69  memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
70  av_lfg_init(&c->rnd, 0xDEADBEEF);
73  c->bdsp.bswap_buf((uint32_t *) buf, (const uint32_t *) avctx->extradata, 4);
74  ff_mpc_init();
75  init_get_bits(&gb, buf, 128);
76 
77  c->IS = get_bits1(&gb);
78  c->MSS = get_bits1(&gb);
79  c->maxbands = get_bits(&gb, 6);
80  if(c->maxbands >= BANDS){
81  av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->maxbands);
82  return AVERROR_INVALIDDATA;
83  }
84  skip_bits_long(&gb, 88);
85  c->gapless = get_bits1(&gb);
86  c->lastframelen = get_bits(&gb, 11);
87  av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n",
88  c->IS, c->MSS, c->gapless, c->lastframelen, c->maxbands);
89  c->frames_to_skip = 0;
90 
93 
94  if(vlc_initialized) return 0;
95  av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
97  &mpc7_scfi[1], 2, 1,
98  &mpc7_scfi[0], 2, 1, 1 << MPC7_SCFI_BITS);
100  &mpc7_dscf[1], 2, 1,
101  &mpc7_dscf[0], 2, 1, 1 << MPC7_DSCF_BITS);
103  &mpc7_hdr[1], 2, 1,
104  &mpc7_hdr[0], 2, 1, 1 << MPC7_HDR_BITS);
105  for(i = 0; i < MPC7_QUANT_VLC_TABLES; i++){
106  for(j = 0; j < 2; j++){
107  quant_vlc[i][j].table = quant_table;
108  quant_vlc[i][j].table_allocated = quant_sizes[i * 2 + j];
109  quant_table += quant_sizes[i * 2 + j];
110  init_vlc(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i],
111  raw_quant_table + 1, 4, 2,
112  raw_quant_table, 4, 2, INIT_VLC_USE_NEW_STATIC);
113  raw_quant_table += 2 * mpc7_quant_vlc_sizes[i];
114  }
115  }
116  vlc_initialized = 1;
117 
118  return 0;
119 }
120 
121 /**
122  * Fill samples for given subband
123  */
124 static inline void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
125 {
126  int i, i1, t;
127  switch(idx){
128  case -1:
129  for(i = 0; i < SAMPLES_PER_BAND; i++){
130  *dst++ = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
131  }
132  break;
133  case 1:
134  i1 = get_bits1(gb);
135  for(i = 0; i < SAMPLES_PER_BAND/3; i++){
136  t = get_vlc2(gb, quant_vlc[0][i1].table, 9, 2);
137  *dst++ = mpc7_idx30[t];
138  *dst++ = mpc7_idx31[t];
139  *dst++ = mpc7_idx32[t];
140  }
141  break;
142  case 2:
143  i1 = get_bits1(gb);
144  for(i = 0; i < SAMPLES_PER_BAND/2; i++){
145  t = get_vlc2(gb, quant_vlc[1][i1].table, 9, 2);
146  *dst++ = mpc7_idx50[t];
147  *dst++ = mpc7_idx51[t];
148  }
149  break;
150  case 3: case 4: case 5: case 6: case 7:
151  i1 = get_bits1(gb);
152  for(i = 0; i < SAMPLES_PER_BAND; i++)
153  *dst++ = get_vlc2(gb, quant_vlc[idx-1][i1].table, 9, 2) - mpc7_quant_vlc_off[idx-1];
154  break;
155  case 8: case 9: case 10: case 11: case 12:
156  case 13: case 14: case 15: case 16: case 17:
157  t = (1 << (idx - 2)) - 1;
158  for(i = 0; i < SAMPLES_PER_BAND; i++)
159  *dst++ = get_bits(gb, idx - 1) - t;
160  break;
161  default: // case 0 and -2..-17
162  return;
163  }
164 }
165 
166 static int get_scale_idx(GetBitContext *gb, int ref)
167 {
168  int t = get_vlc2(gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
169  if (t == 8)
170  return get_bits(gb, 6);
171  return ref + t;
172 }
173 
174 static int mpc7_decode_frame(AVCodecContext * avctx, void *data,
175  int *got_frame_ptr, AVPacket *avpkt)
176 {
177  AVFrame *frame = data;
178  const uint8_t *buf = avpkt->data;
179  int buf_size;
180  MPCContext *c = avctx->priv_data;
181  GetBitContext gb;
182  int i, ch;
183  int mb = -1;
184  Band *bands = c->bands;
185  int off, ret, last_frame, skip;
186  int bits_used, bits_avail;
187 
188  memset(bands, 0, sizeof(*bands) * (c->maxbands + 1));
189 
190  buf_size = avpkt->size & ~3;
191  if (buf_size <= 0) {
192  av_log(avctx, AV_LOG_ERROR, "packet size is too small (%i bytes)\n",
193  avpkt->size);
194  return AVERROR_INVALIDDATA;
195  }
196  if (buf_size != avpkt->size) {
197  av_log(avctx, AV_LOG_WARNING, "packet size is not a multiple of 4. "
198  "extra bytes at the end will be skipped.\n");
199  }
200 
201  skip = buf[0];
202  last_frame = buf[1];
203  buf += 4;
204  buf_size -= 4;
205 
206  /* get output buffer */
207  frame->nb_samples = MPC_FRAME_SIZE;
208  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
209  return ret;
210 
211  av_fast_padded_malloc(&c->bits, &c->buf_size, buf_size);
212  if (!c->bits)
213  return AVERROR(ENOMEM);
214  c->bdsp.bswap_buf((uint32_t *) c->bits, (const uint32_t *) buf,
215  buf_size >> 2);
216  if ((ret = init_get_bits8(&gb, c->bits, buf_size)) < 0)
217  return ret;
218  skip_bits_long(&gb, skip);
219 
220  /* read subband indexes */
221  for(i = 0; i <= c->maxbands; i++){
222  for(ch = 0; ch < 2; ch++){
223  int t = 4;
224  if(i) t = get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) - 5;
225  if(t == 4) bands[i].res[ch] = get_bits(&gb, 4);
226  else bands[i].res[ch] = bands[i-1].res[ch] + t;
227  if (bands[i].res[ch] < -1 || bands[i].res[ch] > 17) {
228  av_log(avctx, AV_LOG_ERROR, "subband index invalid\n");
229  return AVERROR_INVALIDDATA;
230  }
231  }
232 
233  if(bands[i].res[0] || bands[i].res[1]){
234  mb = i;
235  if(c->MSS) bands[i].msf = get_bits1(&gb);
236  }
237  }
238  /* get scale indexes coding method */
239  for(i = 0; i <= mb; i++)
240  for(ch = 0; ch < 2; ch++)
241  if(bands[i].res[ch]) bands[i].scfi[ch] = get_vlc2(&gb, scfi_vlc.table, MPC7_SCFI_BITS, 1);
242  /* get scale indexes */
243  for(i = 0; i <= mb; i++){
244  for(ch = 0; ch < 2; ch++){
245  if(bands[i].res[ch]){
246  bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i];
247  bands[i].scf_idx[ch][0] = get_scale_idx(&gb, bands[i].scf_idx[ch][2]);
248  switch(bands[i].scfi[ch]){
249  case 0:
250  bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
251  bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
252  break;
253  case 1:
254  bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
255  bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1];
256  break;
257  case 2:
258  bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
259  bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
260  break;
261  case 3:
262  bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
263  break;
264  }
265  c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2];
266  }
267  }
268  }
269  /* get quantizers */
270  memset(c->Q, 0, sizeof(c->Q));
271  off = 0;
272  for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
273  for(ch = 0; ch < 2; ch++)
274  idx_to_quant(c, &gb, bands[i].res[ch], c->Q[ch] + off);
275 
276  ff_mpc_dequantize_and_synth(c, mb, (int16_t **)frame->extended_data, 2);
277  if(last_frame)
278  frame->nb_samples = c->lastframelen;
279 
280  bits_used = get_bits_count(&gb);
281  bits_avail = buf_size * 8;
282  if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
283  av_log(avctx, AV_LOG_ERROR, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
284  return AVERROR_INVALIDDATA;
285  }
286  if(c->frames_to_skip){
287  c->frames_to_skip--;
288  *got_frame_ptr = 0;
289  return avpkt->size;
290  }
291 
292  *got_frame_ptr = 1;
293 
294  return avpkt->size;
295 }
296 
298 {
299  MPCContext *c = avctx->priv_data;
300 
301  memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
302  c->frames_to_skip = 32;
303 }
304 
306 {
307  MPCContext *c = avctx->priv_data;
308  av_freep(&c->bits);
309  c->buf_size = 0;
310  return 0;
311 }
312 
314  .name = "mpc7",
315  .long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
316  .type = AVMEDIA_TYPE_AUDIO,
317  .id = AV_CODEC_ID_MUSEPACK7,
318  .priv_data_size = sizeof(MPCContext),
320  .close = mpc7_decode_close,
323  .capabilities = AV_CODEC_CAP_DR1,
324  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
326 };
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:308
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 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:200
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 INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
Definition: vlc.h:75
#define avpriv_request_sample(...)
int msf
mid-stereo flag
Definition: mpc.h:45
static const uint8_t mpc7_quant_vlc_sizes[MPC7_QUANT_VLC_TABLES]
Definition: mpc7data.h:54
int Q[2][MPC_FRAME_SIZE]
Definition: mpc.h:61
int size
Definition: packet.h:364
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:72
#define AV_CH_LAYOUT_STEREO
static VLC dscf_vlc
Definition: mpc7.c:39
AVCodec.
Definition: codec.h:190
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
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:174
int gapless
Definition: mpc.h:55
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
uint8_t
#define av_cold
Definition: attributes.h:88
#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:627
int scf_idx[2][3]
Definition: mpc.h:48
uint8_t * data
Definition: packet.h:363
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
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:194
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:153
static const int16_t quant_table[64]
Definition: intrax8.c:556
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
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: codec.h:197
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
static const uint16_t mpc7_quant_vlcs[177 *2 *2]
Definition: mpc7data.h:62
Definition: vlc.h:26
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
static av_cold int mpc7_decode_init(AVCodecContext *avctx)
Definition: mpc7.c:47
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:297
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:526
static VLC hdr_vlc
Definition: mpc7.c:39
AVCodec ff_mpc7_decoder
Definition: mpc7.c:313
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:124
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1872
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
Definition: lfg.h:53
int extradata_size
Definition: avcodec.h:628
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
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/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(UINT64_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 *(UINT64_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 *(UINT64_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
#define BANDS
Definition: imc.c:52
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
Definition: mpc7.c:305
#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
int MSS
Definition: mpc.h:55
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
Definition: bswapdsp.c:49
void * priv_data
Definition: avcodec.h:553
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:1187
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:836
#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:355
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:166
Subband structure - hold all variables for each subband.
Definition: mpc.h:44
This structure stores compressed data.
Definition: packet.h:340
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:374
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
static VLC quant_vlc[MPC7_QUANT_VLC_TABLES][2]
Definition: mpc7.c:39
int i
Definition: input.c:407
static const uint16_t quant_sizes[MPC7_QUANT_VLC_TABLES *2]
Definition: mpc7.c:41
Definition: mpc.h:52
int frames_to_skip
Definition: mpc.h:66