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 "libavutil/mem_internal.h"
32 #include "libavutil/thread.h"
33 
34 #include "avcodec.h"
35 #include "get_bits.h"
36 #include "internal.h"
37 #include "mpegaudiodsp.h"
38 
39 #include "mpc.h"
40 #include "mpc7data.h"
41 
43 
44 static av_cold void mpc7_init_static(void)
45 {
46  static VLC_TYPE quant_tables[7224][2];
47  const uint8_t *raw_quant_table = mpc7_quant_vlcs;
48 
50  &mpc7_scfi[1], 2,
51  &mpc7_scfi[0], 2, 1, 0, 0, 1 << MPC7_SCFI_BITS);
53  &mpc7_dscf[1], 2,
54  &mpc7_dscf[0], 2, 1, -7, 0, 1 << MPC7_DSCF_BITS);
56  &mpc7_hdr[1], 2,
57  &mpc7_hdr[0], 2, 1, -5, 0, 1 << MPC7_HDR_BITS);
58  for (unsigned i = 0, offset = 0; i < MPC7_QUANT_VLC_TABLES; i++){
59  for (int j = 0; j < 2; j++) {
60  quant_vlc[i][j].table = &quant_tables[offset];
61  quant_vlc[i][j].table_allocated = FF_ARRAY_ELEMS(quant_tables) - offset;
63  &raw_quant_table[1], 2,
64  &raw_quant_table[0], 2, 1,
67  raw_quant_table += 2 * mpc7_quant_vlc_sizes[i];
68  offset += quant_vlc[i][j].table_size;
69  }
70  }
72 }
73 
75 {
76  static AVOnce init_static_once = AV_ONCE_INIT;
77  MPCContext *c = avctx->priv_data;
78  GetBitContext gb;
79  LOCAL_ALIGNED_16(uint8_t, buf, [16]);
80 
81  /* Musepack SV7 is always stereo */
82  if (avctx->channels != 2) {
83  avpriv_request_sample(avctx, "%d channels", avctx->channels);
84  return AVERROR_PATCHWELCOME;
85  }
86 
87  if(avctx->extradata_size < 16){
88  av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
89  return AVERROR_INVALIDDATA;
90  }
91  memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
92  av_lfg_init(&c->rnd, 0xDEADBEEF);
95  c->bdsp.bswap_buf((uint32_t *) buf, (const uint32_t *) avctx->extradata, 4);
96  init_get_bits(&gb, buf, 128);
97 
98  c->IS = get_bits1(&gb);
99  c->MSS = get_bits1(&gb);
100  c->maxbands = get_bits(&gb, 6);
101  if(c->maxbands >= BANDS){
102  av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->maxbands);
103  return AVERROR_INVALIDDATA;
104  }
105  skip_bits_long(&gb, 88);
106  c->gapless = get_bits1(&gb);
107  c->lastframelen = get_bits(&gb, 11);
108  av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n",
109  c->IS, c->MSS, c->gapless, c->lastframelen, c->maxbands);
110  c->frames_to_skip = 0;
111 
114 
115  ff_thread_once(&init_static_once, mpc7_init_static);
116 
117  return 0;
118 }
119 
120 /**
121  * Fill samples for given subband
122  */
123 static inline void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
124 {
125  int i, i1, t;
126  switch(idx){
127  case -1:
128  for(i = 0; i < SAMPLES_PER_BAND; i++){
129  *dst++ = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
130  }
131  break;
132  case 1:
133  i1 = get_bits1(gb);
134  for(i = 0; i < SAMPLES_PER_BAND/3; i++){
135  t = get_vlc2(gb, quant_vlc[0][i1].table, 9, 2);
136  *dst++ = mpc7_idx30[t];
137  *dst++ = mpc7_idx31[t];
138  *dst++ = mpc7_idx32[t];
139  }
140  break;
141  case 2:
142  i1 = get_bits1(gb);
143  for(i = 0; i < SAMPLES_PER_BAND/2; i++){
144  t = get_vlc2(gb, quant_vlc[1][i1].table, 9, 2);
145  *dst++ = mpc7_idx50[t];
146  *dst++ = mpc7_idx51[t];
147  }
148  break;
149  case 3: case 4: case 5: case 6: case 7:
150  i1 = get_bits1(gb);
151  for(i = 0; i < SAMPLES_PER_BAND; i++)
152  *dst++ = get_vlc2(gb, quant_vlc[idx-1][i1].table, 9, 2);
153  break;
154  case 8: case 9: case 10: case 11: case 12:
155  case 13: case 14: case 15: case 16: case 17:
156  t = (1 << (idx - 2)) - 1;
157  for(i = 0; i < SAMPLES_PER_BAND; i++)
158  *dst++ = get_bits(gb, idx - 1) - t;
159  break;
160  default: // case 0 and -2..-17
161  return;
162  }
163 }
164 
165 static int get_scale_idx(GetBitContext *gb, int ref)
166 {
167  int t = get_vlc2(gb, dscf_vlc.table, MPC7_DSCF_BITS, 1);
168  if (t == 8)
169  return get_bits(gb, 6);
170  return ref + t;
171 }
172 
173 static int mpc7_decode_frame(AVCodecContext * avctx, void *data,
174  int *got_frame_ptr, AVPacket *avpkt)
175 {
176  AVFrame *frame = data;
177  const uint8_t *buf = avpkt->data;
178  int buf_size;
179  MPCContext *c = avctx->priv_data;
180  GetBitContext gb;
181  int i, ch;
182  int mb = -1;
183  Band *bands = c->bands;
184  int off, ret, last_frame, skip;
185  int bits_used, bits_avail;
186 
187  memset(bands, 0, sizeof(*bands) * (c->maxbands + 1));
188 
189  buf_size = avpkt->size & ~3;
190  if (buf_size <= 0) {
191  av_log(avctx, AV_LOG_ERROR, "packet size is too small (%i bytes)\n",
192  avpkt->size);
193  return AVERROR_INVALIDDATA;
194  }
195  if (buf_size != avpkt->size) {
196  av_log(avctx, AV_LOG_WARNING, "packet size is not a multiple of 4. "
197  "extra bytes at the end will be skipped.\n");
198  }
199 
200  skip = buf[0];
201  last_frame = buf[1];
202  buf += 4;
203  buf_size -= 4;
204 
205  /* get output buffer */
206  frame->nb_samples = MPC_FRAME_SIZE;
207  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
208  return ret;
209 
210  av_fast_padded_malloc(&c->bits, &c->buf_size, buf_size);
211  if (!c->bits)
212  return AVERROR(ENOMEM);
213  c->bdsp.bswap_buf((uint32_t *) c->bits, (const uint32_t *) buf,
214  buf_size >> 2);
215  if ((ret = init_get_bits8(&gb, c->bits, buf_size)) < 0)
216  return ret;
217  skip_bits_long(&gb, skip);
218 
219  /* read subband indexes */
220  for(i = 0; i <= c->maxbands; i++){
221  for(ch = 0; ch < 2; ch++){
222  int t = i ? get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) : 4;
223  if(t == 4) bands[i].res[ch] = get_bits(&gb, 4);
224  else bands[i].res[ch] = bands[i-1].res[ch] + t;
225  if (bands[i].res[ch] < -1 || bands[i].res[ch] > 17) {
226  av_log(avctx, AV_LOG_ERROR, "subband index invalid\n");
227  return AVERROR_INVALIDDATA;
228  }
229  }
230 
231  if(bands[i].res[0] || bands[i].res[1]){
232  mb = i;
233  if(c->MSS) bands[i].msf = get_bits1(&gb);
234  }
235  }
236  /* get scale indexes coding method */
237  for(i = 0; i <= mb; i++)
238  for(ch = 0; ch < 2; ch++)
239  if(bands[i].res[ch]) bands[i].scfi[ch] = get_vlc2(&gb, scfi_vlc.table, MPC7_SCFI_BITS, 1);
240  /* get scale indexes */
241  for(i = 0; i <= mb; i++){
242  for(ch = 0; ch < 2; ch++){
243  if(bands[i].res[ch]){
244  bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i];
245  bands[i].scf_idx[ch][0] = get_scale_idx(&gb, bands[i].scf_idx[ch][2]);
246  switch(bands[i].scfi[ch]){
247  case 0:
248  bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
249  bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
250  break;
251  case 1:
252  bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
253  bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1];
254  break;
255  case 2:
256  bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
257  bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
258  break;
259  case 3:
260  bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
261  break;
262  }
263  c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2];
264  }
265  }
266  }
267  /* get quantizers */
268  memset(c->Q, 0, sizeof(c->Q));
269  off = 0;
270  for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
271  for(ch = 0; ch < 2; ch++)
272  idx_to_quant(c, &gb, bands[i].res[ch], c->Q[ch] + off);
273 
274  ff_mpc_dequantize_and_synth(c, mb, (int16_t **)frame->extended_data, 2);
275  if(last_frame)
276  frame->nb_samples = c->lastframelen;
277 
278  bits_used = get_bits_count(&gb);
279  bits_avail = buf_size * 8;
280  if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
281  av_log(avctx, AV_LOG_ERROR, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
282  return AVERROR_INVALIDDATA;
283  }
284  if(c->frames_to_skip){
285  c->frames_to_skip--;
286  *got_frame_ptr = 0;
287  return avpkt->size;
288  }
289 
290  *got_frame_ptr = 1;
291 
292  return avpkt->size;
293 }
294 
296 {
297  MPCContext *c = avctx->priv_data;
298 
299  memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
300  c->frames_to_skip = 32;
301 }
302 
304 {
305  MPCContext *c = avctx->priv_data;
306  av_freep(&c->bits);
307  c->buf_size = 0;
308  return 0;
309 }
310 
312  .name = "mpc7",
313  .long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
314  .type = AVMEDIA_TYPE_AUDIO,
315  .id = AV_CODEC_ID_MUSEPACK7,
316  .priv_data_size = sizeof(MPCContext),
318  .close = mpc7_decode_close,
321  .capabilities = AV_CODEC_CAP_DR1,
322  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
324  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
325 };
#define NULL
Definition: coverity.c:32
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
Definition: mpc.c:56
int table_size
Definition: vlc.h:29
#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:314
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:55
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:33
#define avpriv_request_sample(...)
int msf
mid-stereo flag
Definition: mpc.h:46
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:62
int size
Definition: packet.h:364
int res[2]
Definition: mpc.h:47
AVLFG rnd
Definition: mpc.h:66
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 FF_ARRAY_ELEMS(a)
#define AV_CH_LAYOUT_STEREO
static VLC dscf_vlc
Definition: mpc7.c:42
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:65
static const uint8_t mpc7_quant_vlcs[177 *2 *2]
Definition: mpc7data.h:62
#define LOCAL_ALIGNED_16(t, v,...)
Definition: mem_internal.h:130
int lastframelen
Definition: mpc.h:57
static int mpc7_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: mpc7.c:173
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:40
int gapless
Definition: mpc.h:56
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1199
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
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
#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:632
int scf_idx[2][3]
Definition: mpc.h:49
static const int8_t mpc7_quant_vlc_off[MPC7_QUANT_VLC_TABLES]
Definition: mpc7data.h:58
uint8_t * data
Definition: packet.h:363
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
bitstream reader API header.
#define AVOnce
Definition: thread.h:172
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:56
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:115
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
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:58
#define SAMPLES_PER_BAND
Definition: mpc.h:41
Definition: vlc.h:26
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1242
static av_cold int mpc7_decode_init(AVCodecContext *avctx)
Definition: mpc7.c:74
common internal API header
void ff_mpa_synth_init_fixed(void)
audio channel layout utility functions
uint8_t * bits
Definition: mpc.h:64
#define MPC_FRAME_SIZE
Definition: mpc.h:42
static void mpc7_decode_flush(AVCodecContext *avctx)
Definition: mpc7.c:295
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
#define INIT_VLC_STATIC_FROM_LENGTHS(vlc, bits, nb_codes, lens, len_wrap,symbols, symbols_wrap, symbols_size,offset, flags, static_size)
Definition: vlc.h:126
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:54
#define AV_ONCE_INIT
Definition: thread.h:173
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
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
static av_cold void mpc7_init_static(void)
Definition: mpc7.c:44
int scfi[2]
Definition: mpc.h:48
main external API structure.
Definition: avcodec.h:531
static VLC hdr_vlc
Definition: mpc7.c:42
AVCodec ff_mpc7_decoder
Definition: mpc7.c:311
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:123
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1884
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:633
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
int oldDSCF[2][BANDS]
Definition: mpc.h:60
#define MPC7_HDR_BITS
Definition: mpc7data.h:47
int ff_init_vlc_from_lengths(VLC *vlc_arg, int nb_bits, int nb_codes, const int8_t *lens, int lens_wrap, const void *symbols, int symbols_wrap, int symbols_size, int offset, int flags, void *logctx)
Build VLC decoding tables suitable for use with get_vlc2()
Definition: bitstream.c:381
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:55
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
Definition: mpc7.c:303
#define MPC7_SCFI_SIZE
Definition: mpc7data.h:33
Band bands[BANDS]
Definition: mpc.h:61
common internal api header.
static int ref[MAX_W *MAX_W]
Definition: jpeg2000dwt.c:107
int MSS
Definition: mpc.h:56
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
Definition: bswapdsp.c:49
void * priv_data
Definition: avcodec.h:558
static const int8_t mpc7_idx30[]
Definition: mpc7data.h:27
static VLC scfi_vlc
Definition: mpc7.c:42
int channels
number of audio channels
Definition: avcodec.h:1192
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:175
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:940
#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:361
static const uint8_t mpc7_dscf[MPC7_DSCF_SIZE *2]
Definition: mpc7data.h:41
#define INIT_VLC_STATIC_OVERLONG
Definition: vlc.h:96
static int get_scale_idx(GetBitContext *gb, int ref)
Definition: mpc7.c:165
Subband structure - hold all variables for each subband.
Definition: mpc.h:45
This structure stores compressed data.
Definition: packet.h:340
av_cold void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:81
#define MPC7_DSCF_SIZE
Definition: mpc7data.h:39
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:380
#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:42
int i
Definition: input.c:407
Definition: mpc.h:53
int frames_to_skip
Definition: mpc.h:67