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 "libavutil/mem.h"
32 #include "libavutil/mem_internal.h"
33 #include "libavutil/thread.h"
34 
35 #include "avcodec.h"
36 #include "codec_internal.h"
37 #include "decode.h"
38 #include "get_bits.h"
39 #include "mpegaudiodsp.h"
40 
41 #include "mpc.h"
42 #include "mpc7data.h"
43 
48 
49 static av_cold void mpc7_init_static(void)
50 {
51  static VLCElem quant_tables[7224];
53  const uint8_t *raw_quant_table = mpc7_quant_vlcs;
54 
56  &mpc7_scfi[1], 2,
57  &mpc7_scfi[0], 2, 1, 0, 0);
59  &mpc7_dscf[1], 2,
60  &mpc7_dscf[0], 2, 1, -7, 0);
62  &mpc7_hdr[1], 2,
63  &mpc7_hdr[0], 2, 1, -5, 0);
64  for (int i = 0; i < MPC7_QUANT_VLC_TABLES; i++) {
65  for (int j = 0; j < 2; j++) {
66  quant_vlc[i][j] =
68  &raw_quant_table[1], 2,
69  &raw_quant_table[0], 2, 1,
70  mpc7_quant_vlc_off[i], 0);
71  raw_quant_table += 2 * mpc7_quant_vlc_sizes[i];
72  }
73  }
75 }
76 
78 {
79  static AVOnce init_static_once = AV_ONCE_INIT;
80  MPCContext *c = avctx->priv_data;
81  GetBitContext gb;
82  LOCAL_ALIGNED_16(uint8_t, buf, [16]);
83 
84  /* Musepack SV7 is always stereo */
85  if (avctx->ch_layout.nb_channels != 2) {
86  avpriv_request_sample(avctx, "%d channels", avctx->ch_layout.nb_channels);
87  return AVERROR_PATCHWELCOME;
88  }
89 
90  if(avctx->extradata_size < 16){
91  av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
92  return AVERROR_INVALIDDATA;
93  }
94  memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
95  av_lfg_init(&c->rnd, 0xDEADBEEF);
96  ff_bswapdsp_init(&c->bdsp);
97  ff_mpadsp_init(&c->mpadsp);
98  c->bdsp.bswap_buf((uint32_t *) buf, (const uint32_t *) avctx->extradata, 4);
99  init_get_bits(&gb, buf, 128);
100 
101  c->IS = get_bits1(&gb);
102  c->MSS = get_bits1(&gb);
103  c->maxbands = get_bits(&gb, 6);
104  if(c->maxbands >= BANDS){
105  av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->maxbands);
106  return AVERROR_INVALIDDATA;
107  }
108  skip_bits_long(&gb, 88);
109  c->gapless = get_bits1(&gb);
110  c->lastframelen = get_bits(&gb, 11);
111  av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n",
112  c->IS, c->MSS, c->gapless, c->lastframelen, c->maxbands);
113  c->frames_to_skip = 0;
114 
118 
119  ff_thread_once(&init_static_once, mpc7_init_static);
120 
121  return 0;
122 }
123 
124 /**
125  * Fill samples for given subband
126  */
127 static inline void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
128 {
129  int i, i1, t;
130  switch(idx){
131  case -1:
132  for(i = 0; i < SAMPLES_PER_BAND; i++){
133  *dst++ = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
134  }
135  break;
136  case 1:
137  i1 = get_bits1(gb);
138  for(i = 0; i < SAMPLES_PER_BAND/3; i++){
139  t = get_vlc2(gb, quant_vlc[0][i1], 9, 2);
140  *dst++ = mpc7_idx30[t];
141  *dst++ = mpc7_idx31[t];
142  *dst++ = mpc7_idx32[t];
143  }
144  break;
145  case 2:
146  i1 = get_bits1(gb);
147  for(i = 0; i < SAMPLES_PER_BAND/2; i++){
148  t = get_vlc2(gb, quant_vlc[1][i1], 9, 2);
149  *dst++ = mpc7_idx50[t];
150  *dst++ = mpc7_idx51[t];
151  }
152  break;
153  case 3: case 4: case 5: case 6: case 7:
154  i1 = get_bits1(gb);
155  for(i = 0; i < SAMPLES_PER_BAND; i++)
156  *dst++ = get_vlc2(gb, quant_vlc[idx-1][i1], 9, 2);
157  break;
158  case 8: case 9: case 10: case 11: case 12:
159  case 13: case 14: case 15: case 16: case 17:
160  t = (1 << (idx - 2)) - 1;
161  for(i = 0; i < SAMPLES_PER_BAND; i++)
162  *dst++ = get_bits(gb, idx - 1) - t;
163  break;
164  default: // case 0 and -2..-17
165  return;
166  }
167 }
168 
169 static int get_scale_idx(GetBitContext *gb, int ref)
170 {
171  int t = get_vlc2(gb, dscf_vlc, MPC7_DSCF_BITS, 1);
172  if (t == 8)
173  return get_bits(gb, 6);
174  return ref + t;
175 }
176 
178  int *got_frame_ptr, AVPacket *avpkt)
179 {
180  const uint8_t *buf = avpkt->data;
181  int buf_size;
182  MPCContext *c = avctx->priv_data;
183  GetBitContext gb;
184  int i, ch;
185  int mb = -1;
186  Band *bands = c->bands;
187  int off, ret, last_frame, skip;
188  int bits_used, bits_avail;
189 
190  memset(bands, 0, sizeof(*bands) * (c->maxbands + 1));
191 
192  buf_size = avpkt->size & ~3;
193  if (buf_size <= 0) {
194  av_log(avctx, AV_LOG_ERROR, "packet size is too small (%i bytes)\n",
195  avpkt->size);
196  return AVERROR_INVALIDDATA;
197  }
198  if (buf_size != avpkt->size) {
199  av_log(avctx, AV_LOG_WARNING, "packet size is not a multiple of 4. "
200  "extra bytes at the end will be skipped.\n");
201  }
202 
203  skip = buf[0];
204  last_frame = buf[1];
205  buf += 4;
206  buf_size -= 4;
207 
208  /* get output buffer */
209  frame->nb_samples = MPC_FRAME_SIZE;
210  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
211  return ret;
212 
213  av_fast_padded_malloc(&c->bits, &c->buf_size, buf_size);
214  if (!c->bits)
215  return AVERROR(ENOMEM);
216  c->bdsp.bswap_buf((uint32_t *) c->bits, (const uint32_t *) buf,
217  buf_size >> 2);
218  if ((ret = init_get_bits8(&gb, c->bits, buf_size)) < 0)
219  return ret;
220  skip_bits_long(&gb, skip);
221 
222  /* read subband indexes */
223  for(i = 0; i <= c->maxbands; i++){
224  for(ch = 0; ch < 2; ch++){
225  int t = i ? get_vlc2(&gb, hdr_vlc, MPC7_HDR_BITS, 1) : 4;
226  if(t == 4) bands[i].res[ch] = get_bits(&gb, 4);
227  else bands[i].res[ch] = bands[i-1].res[ch] + t;
228  if (bands[i].res[ch] < -1 || bands[i].res[ch] > 17) {
229  av_log(avctx, AV_LOG_ERROR, "subband index invalid\n");
230  return AVERROR_INVALIDDATA;
231  }
232  }
233 
234  if(bands[i].res[0] || bands[i].res[1]){
235  mb = i;
236  if(c->MSS) bands[i].msf = get_bits1(&gb);
237  }
238  }
239  /* get scale indexes coding method */
240  for(i = 0; i <= mb; i++)
241  for(ch = 0; ch < 2; ch++)
242  if (bands[i].res[ch])
243  bands[i].scfi[ch] = get_vlc2(&gb, scfi_vlc, MPC7_SCFI_BITS, 1);
244  /* get scale indexes */
245  for(i = 0; i <= mb; i++){
246  for(ch = 0; ch < 2; ch++){
247  if(bands[i].res[ch]){
248  bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i];
249  bands[i].scf_idx[ch][0] = get_scale_idx(&gb, bands[i].scf_idx[ch][2]);
250  switch(bands[i].scfi[ch]){
251  case 0:
252  bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
253  bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
254  break;
255  case 1:
256  bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
257  bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1];
258  break;
259  case 2:
260  bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
261  bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
262  break;
263  case 3:
264  bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
265  break;
266  }
267  c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2];
268  }
269  }
270  }
271  /* get quantizers */
272  memset(c->Q, 0, sizeof(c->Q));
273  off = 0;
274  for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
275  for(ch = 0; ch < 2; ch++)
276  idx_to_quant(c, &gb, bands[i].res[ch], c->Q[ch] + off);
277 
278  ff_mpc_dequantize_and_synth(c, mb, (int16_t **)frame->extended_data, 2);
279  if(last_frame)
280  frame->nb_samples = c->lastframelen;
281 
282  bits_used = get_bits_count(&gb);
283  bits_avail = buf_size * 8;
284  if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
285  av_log(avctx, AV_LOG_ERROR, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
286  return AVERROR_INVALIDDATA;
287  }
288  if(c->frames_to_skip){
289  c->frames_to_skip--;
290  *got_frame_ptr = 0;
291  return avpkt->size;
292  }
293 
294  *got_frame_ptr = 1;
295 
296  return avpkt->size;
297 }
298 
300 {
301  MPCContext *c = avctx->priv_data;
302 
303  memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
304  c->frames_to_skip = 32;
305 }
306 
308 {
309  MPCContext *c = avctx->priv_data;
310  av_freep(&c->bits);
311  c->buf_size = 0;
312  return 0;
313 }
314 
316  .p.name = "mpc7",
317  CODEC_LONG_NAME("Musepack SV7"),
318  .p.type = AVMEDIA_TYPE_AUDIO,
319  .p.id = AV_CODEC_ID_MUSEPACK7,
320  .priv_data_size = sizeof(MPCContext),
322  .close = mpc7_decode_close,
324  .flush = mpc7_decode_flush,
325  .p.capabilities = AV_CODEC_CAP_DR1,
326  .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
328 };
MPC7_SCFI_BITS
#define MPC7_SCFI_BITS
Definition: mpc7data.h:34
mpc7_scfi
static const uint8_t mpc7_scfi[MPC7_SCFI_SIZE *2]
Definition: mpc7data.h:35
skip_bits_long
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:278
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:215
mpc7_decode_frame
static int mpc7_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition: mpc7.c:177
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
ff_mpc7_decoder
const FFCodec ff_mpc7_decoder
Definition: mpc7.c:315
mem_internal.h
AV_CHANNEL_LAYOUT_STEREO
#define AV_CHANNEL_LAYOUT_STEREO
Definition: channel_layout.h:394
av_lfg_init
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
Definition: lfg.c:32
thread.h
quant_vlc
static const VLCElem * quant_vlc[MPC7_QUANT_VLC_TABLES][2]
Definition: mpc7.c:47
get_bits_count
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:266
hdr_vlc
static VLCElem hdr_vlc[1<< MPC7_HDR_BITS]
Definition: mpc7.c:46
ff_mpadsp_init
av_cold void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:81
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:389
AVPacket::data
uint8_t * data
Definition: packet.h:539
MPCContext
Definition: mpc.h:54
FFCodec
Definition: codec_internal.h:127
mpc7_decode_close
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
Definition: mpc7.c:307
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:328
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:514
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:335
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:131
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:1079
GetBitContext
Definition: get_bits.h:108
ff_mpc_dequantize_and_synth
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
Definition: mpc.c:55
ff_thread_once
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:205
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:209
av_cold
#define av_cold
Definition: attributes.h:90
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:545
VLCInitState
For static VLCs, the number of bits can often be hardcoded at each get_vlc2() callsite.
Definition: vlc.h:212
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:538
FF_CODEC_DECODE_CB
#define FF_CODEC_DECODE_CB(func)
Definition: codec_internal.h:311
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
LOCAL_ALIGNED_16
#define LOCAL_ALIGNED_16(t, v,...)
Definition: mem_internal.h:130
mpc7_decode_flush
static void mpc7_decode_flush(AVCodecContext *avctx)
Definition: mpc7.c:299
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:230
decode.h
get_bits.h
bands
static const float bands[]
Definition: af_superequalizer.c:56
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:296
mpc.h
dscf_vlc
static VLCElem dscf_vlc[1<< MPC7_DSCF_BITS]
Definition: mpc7.c:45
AV_ONCE_INIT
#define AV_ONCE_INIT
Definition: thread.h:203
ff_bswapdsp_init
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
Definition: bswapdsp.c:49
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:64
ff_mpa_synth_init_fixed
void ff_mpa_synth_init_fixed(void)
mpc7_idx51
static const int8_t mpc7_idx51[]
Definition: mpc7data.h:31
mpc7_quant_vlcs
static const uint8_t mpc7_quant_vlcs[177 *2 *2]
Definition: mpc7data.h:62
MPC_FRAME_SIZE
#define MPC_FRAME_SIZE
Definition: mpc.h:43
state
static struct @466 state
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:388
MPC7_HDR_SIZE
#define MPC7_HDR_SIZE
Definition: mpc7data.h:46
mpc7_init_static
static av_cold void mpc7_init_static(void)
Definition: mpc7.c:49
get_vlc2
static av_always_inline int get_vlc2(GetBitContext *s, const VLCElem *table, int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:652
AVOnce
#define AVOnce
Definition: thread.h:202
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
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1697
init
int(* init)(AVBSFContext *ctx)
Definition: dts2pts.c:368
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
AVPacket::size
int size
Definition: packet.h:540
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:318
codec_internal.h
dst
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
Definition: dsp.h:83
SAMPLES_PER_BAND
#define SAMPLES_PER_BAND
Definition: mpc.h:42
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1071
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
VLCElem
Definition: vlc.h:32
MPC7_SCFI_SIZE
#define MPC7_SCFI_SIZE
Definition: mpc7data.h:33
mpc7_quant_vlc_sizes
static const uint8_t mpc7_quant_vlc_sizes[MPC7_QUANT_VLC_TABLES]
Definition: mpc7data.h:54
mb
#define mb
Definition: vf_colormatrix.c:99
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:64
get_scale_idx
static int get_scale_idx(GetBitContext *gb, int ref)
Definition: mpc7.c:169
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
mpc7_idx31
static const int8_t mpc7_idx31[]
Definition: mpc7data.h:28
mpc7data.h
AVCodecContext::extradata
uint8_t * extradata
Out-of-band global headers that may be used by some codecs.
Definition: avcodec.h:537
internal.h
BANDS
#define BANDS
Definition: imc.c:57
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
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:52
mpc7_decode_init
static av_cold int mpc7_decode_init(AVCodecContext *avctx)
Definition: mpc7.c:77
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:194
scfi_vlc
static VLCElem scfi_vlc[1<< MPC7_SCFI_BITS]
Definition: mpc7.c:44
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
AVCodecContext
main external API structure.
Definition: avcodec.h:451
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:46
av_channel_layout_uninit
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
Definition: channel_layout.c:441
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:117
ff_vlc_init_tables_from_lengths
const av_cold VLCElem * ff_vlc_init_tables_from_lengths(VLCInitState *state, 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)
Definition: vlc.c:366
mpegaudiodsp.h
mem.h
avpriv_request_sample
#define avpriv_request_sample(...)
Definition: tableprint_vlc.h:36
VLC_INIT_STATIC_TABLE_FROM_LENGTHS
#define VLC_INIT_STATIC_TABLE_FROM_LENGTHS(vlc_table, nb_bits, nb_codes, lens, lens_wrap, syms, syms_wrap, syms_size, offset, flags)
Definition: vlc.h:280
VLC_INIT_STATE
#define VLC_INIT_STATE(_table)
Definition: vlc.h:217
AVPacket
This structure stores compressed data.
Definition: packet.h:516
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:478
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
quant_tables
static const uint8_t quant_tables[]
Definition: vorbis_enc_data.h:395
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
idx_to_quant
static void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
Fill samples for given subband.
Definition: mpc7.c:127
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:61
AV_CODEC_ID_MUSEPACK7
@ AV_CODEC_ID_MUSEPACK7
Definition: codec_id.h:474
skip
static void BS_FUNC() skip(BSCTX *bc, unsigned int n)
Skip n bits in the buffer.
Definition: bitstream_template.h:375
mpc7_quant_vlc_off
static const int8_t mpc7_quant_vlc_off[MPC7_QUANT_VLC_TABLES]
Definition: mpc7data.h:58