FFmpeg  4.4.6
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_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];
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);
93  ff_bswapdsp_init(&c->bdsp);
94  ff_mpadsp_init(&c->mpadsp);
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 */
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 
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 };
static void flush(AVCodecContext *avctx)
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
static const float bands[]
#define av_cold
Definition: attributes.h:88
uint8_t
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
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
audio channel layout utility functions
#define NULL
Definition: coverity.c:32
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1900
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
static AVFrame * frame
bitstream reader API header.
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
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
#define AV_CH_LAYOUT_STEREO
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
@ AV_CODEC_ID_MUSEPACK7
Definition: codec_id.h:452
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:50
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
#define BANDS
Definition: imc.c:55
int i
Definition: input.c:407
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
Definition: lfg.c:32
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
Definition: lfg.h:53
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
Definition: bswapdsp.c:49
#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:41
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
Definition: mpc.c:56
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define AVOnce
Definition: thread.h:172
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:175
#define AV_ONCE_INIT
Definition: thread.h:173
#define LOCAL_ALIGNED_16(t, v,...)
Definition: mem_internal.h:130
static int mpc7_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: mpc7.c:173
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
Definition: mpc7.c:303
static int get_scale_idx(GetBitContext *gb, int ref)
Definition: mpc7.c:165
static av_cold int mpc7_decode_init(AVCodecContext *avctx)
Definition: mpc7.c:74
static VLC quant_vlc[MPC7_QUANT_VLC_TABLES][2]
Definition: mpc7.c:42
static void mpc7_decode_flush(AVCodecContext *avctx)
Definition: mpc7.c:295
static VLC hdr_vlc
Definition: mpc7.c:42
AVCodec ff_mpc7_decoder
Definition: mpc7.c:311
static void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
Fill samples for given subband.
Definition: mpc7.c:123
static VLC scfi_vlc
Definition: mpc7.c:42
static VLC dscf_vlc
Definition: mpc7.c:42
static av_cold void mpc7_init_static(void)
Definition: mpc7.c:44
static const uint8_t mpc7_scfi[MPC7_SCFI_SIZE *2]
Definition: mpc7data.h:35
static const int8_t mpc7_idx32[]
Definition: mpc7data.h:29
#define MPC7_DSCF_SIZE
Definition: mpc7data.h:39
static const uint8_t mpc7_hdr[MPC7_HDR_SIZE *2]
Definition: mpc7data.h:48
static const uint8_t mpc7_dscf[MPC7_DSCF_SIZE *2]
Definition: mpc7data.h:41
static const uint8_t mpc7_quant_vlc_sizes[MPC7_QUANT_VLC_TABLES]
Definition: mpc7data.h:54
#define MPC7_QUANT_VLC_TABLES
Definition: mpc7data.h:53
static const uint8_t mpc7_quant_vlcs[177 *2 *2]
Definition: mpc7data.h:62
#define MPC7_SCFI_SIZE
Definition: mpc7data.h:33
#define MPC7_SCFI_BITS
Definition: mpc7data.h:34
static const int8_t mpc7_idx51[]
Definition: mpc7data.h:31
static const int8_t mpc7_idx30[]
Definition: mpc7data.h:27
static const int8_t mpc7_quant_vlc_off[MPC7_QUANT_VLC_TABLES]
Definition: mpc7data.h:58
#define MPC7_DSCF_BITS
Definition: mpc7data.h:40
#define MPC7_HDR_SIZE
Definition: mpc7data.h:46
#define MPC7_HDR_BITS
Definition: mpc7data.h:47
static const int8_t mpc7_idx50[]
Definition: mpc7data.h:30
static const int8_t mpc7_idx31[]
Definition: mpc7data.h:28
Musepack decoder MPEG Audio Layer 1/2 -like codec with frames of 1152 samples divided into 32 subband...
#define SAMPLES_PER_BAND
Definition: mpc.h:41
#define MPC_FRAME_SIZE
Definition: mpc.h:42
av_cold void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:81
void ff_mpa_synth_init_fixed(void)
const char data[16]
Definition: mxf.c:142
static const uint16_t table[]
Definition: prosumer.c:206
#define FF_ARRAY_ELEMS(a)
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:637
int channels
number of audio channels
Definition: avcodec.h:1197
int extradata_size
Definition: avcodec.h:638
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1247
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
Subband structure - hold all variables for each subband.
Definition: mpc.h:45
Definition: mpc.h:53
Definition: vlc.h:26
int table_size
Definition: vlc.h:29
int table_allocated
Definition: vlc.h:29
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
#define avpriv_request_sample(...)
#define av_freep(p)
#define av_log(a,...)
static int ref[MAX_W *MAX_W]
Definition: jpeg2000dwt.c:107
#define mb
static const uint8_t offset[127][2]
Definition: vf_spp.c:107
#define INIT_VLC_STATIC_OVERLONG
Definition: vlc.h:96
#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
#define VLC_TYPE
Definition: vlc.h:24
static double c[64]