Mercurial > libavcodec.hg
annotate atrac3.c @ 9440:daee921fb6bb libavcodec
VC1: add and use avg_no_rnd chroma MC functions
author | conrad |
---|---|
date | Tue, 14 Apr 2009 23:56:10 +0000 |
parents | 0dce4fe6e6f3 |
children | 67a20f0eb42c |
rev | line source |
---|---|
4856 | 1 /* |
2 * Atrac 3 compatible decoder | |
6844 | 3 * Copyright (c) 2006-2008 Maxim Poliakovski |
4 * Copyright (c) 2006-2008 Benjamin Larsson | |
4856 | 5 * |
6 * This file is part of FFmpeg. | |
7 * | |
8 * FFmpeg is free software; you can redistribute it and/or | |
9 * modify it under the terms of the GNU Lesser General Public | |
10 * License as published by the Free Software Foundation; either | |
11 * version 2.1 of the License, or (at your option) any later version. | |
12 * | |
13 * FFmpeg is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
16 * Lesser General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU Lesser General Public | |
19 * License along with FFmpeg; if not, write to the Free Software | |
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
21 */ | |
22 | |
23 /** | |
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24 * @file libavcodec/atrac3.c |
4856 | 25 * Atrac 3 compatible decoder. |
6844 | 26 * This decoder handles Sony's ATRAC3 data. |
27 * | |
28 * Container formats used to store atrac 3 data: | |
29 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3). | |
4856 | 30 * |
31 * To use this decoder, a calling application must supply the extradata | |
6844 | 32 * bytes provided in the containers above. |
4856 | 33 */ |
34 | |
35 #include <math.h> | |
36 #include <stddef.h> | |
37 #include <stdio.h> | |
38 | |
39 #include "avcodec.h" | |
9428 | 40 #include "get_bits.h" |
4856 | 41 #include "dsputil.h" |
42 #include "bytestream.h" | |
43 | |
44 #include "atrac3data.h" | |
45 | |
46 #define JOINT_STEREO 0x12 | |
47 #define STEREO 0x2 | |
48 | |
49 | |
50 /* These structures are needed to store the parsed gain control data. */ | |
51 typedef struct { | |
52 int num_gain_data; | |
53 int levcode[8]; | |
54 int loccode[8]; | |
55 } gain_info; | |
56 | |
57 typedef struct { | |
58 gain_info gBlock[4]; | |
59 } gain_block; | |
60 | |
61 typedef struct { | |
62 int pos; | |
63 int numCoefs; | |
64 float coef[8]; | |
65 } tonal_component; | |
66 | |
67 typedef struct { | |
68 int bandsCoded; | |
69 int numComponents; | |
70 tonal_component components[64]; | |
71 float prevFrame[1024]; | |
72 int gcBlkSwitch; | |
73 gain_block gainBlock[2]; | |
74 | |
75 DECLARE_ALIGNED_16(float, spectrum[1024]); | |
76 DECLARE_ALIGNED_16(float, IMDCT_buf[1024]); | |
77 | |
78 float delayBuf1[46]; ///<qmf delay buffers | |
79 float delayBuf2[46]; | |
80 float delayBuf3[46]; | |
81 } channel_unit; | |
82 | |
83 typedef struct { | |
84 GetBitContext gb; | |
85 //@{ | |
86 /** stream data */ | |
87 int channels; | |
88 int codingMode; | |
89 int bit_rate; | |
90 int sample_rate; | |
91 int samples_per_channel; | |
92 int samples_per_frame; | |
93 | |
94 int bits_per_frame; | |
95 int bytes_per_frame; | |
96 int pBs; | |
97 channel_unit* pUnits; | |
98 //@} | |
99 //@{ | |
100 /** joint-stereo related variables */ | |
101 int matrix_coeff_index_prev[4]; | |
102 int matrix_coeff_index_now[4]; | |
103 int matrix_coeff_index_next[4]; | |
104 int weighting_delay[6]; | |
105 //@} | |
106 //@{ | |
107 /** data buffers */ | |
108 float outSamples[2048]; | |
109 uint8_t* decoded_bytes_buffer; | |
110 float tempBuf[1070]; | |
111 //@} | |
112 //@{ | |
113 /** extradata */ | |
114 int atrac3version; | |
115 int delay; | |
116 int scrambled_stream; | |
117 int frame_factor; | |
118 //@} | |
119 } ATRAC3Context; | |
120 | |
121 static DECLARE_ALIGNED_16(float,mdct_window[512]); | |
122 static float qmf_window[48]; | |
123 static VLC spectral_coeff_tab[7]; | |
124 static float SFTable[64]; | |
125 static float gain_tab1[16]; | |
126 static float gain_tab2[31]; | |
127 static MDCTContext mdct_ctx; | |
128 static DSPContext dsp; | |
129 | |
130 | |
131 /* quadrature mirror synthesis filter */ | |
132 | |
133 /** | |
134 * Quadrature mirror synthesis filter. | |
135 * | |
136 * @param inlo lower part of spectrum | |
137 * @param inhi higher part of spectrum | |
138 * @param nIn size of spectrum buffer | |
139 * @param pOut out buffer | |
140 * @param delayBuf delayBuf buffer | |
141 * @param temp temp buffer | |
142 */ | |
143 | |
144 | |
145 static void iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp) | |
146 { | |
147 int i, j; | |
148 float *p1, *p3; | |
149 | |
150 memcpy(temp, delayBuf, 46*sizeof(float)); | |
151 | |
152 p3 = temp + 46; | |
153 | |
154 /* loop1 */ | |
155 for(i=0; i<nIn; i+=2){ | |
156 p3[2*i+0] = inlo[i ] + inhi[i ]; | |
157 p3[2*i+1] = inlo[i ] - inhi[i ]; | |
158 p3[2*i+2] = inlo[i+1] + inhi[i+1]; | |
159 p3[2*i+3] = inlo[i+1] - inhi[i+1]; | |
160 } | |
161 | |
162 /* loop2 */ | |
163 p1 = temp; | |
164 for (j = nIn; j != 0; j--) { | |
165 float s1 = 0.0; | |
166 float s2 = 0.0; | |
167 | |
168 for (i = 0; i < 48; i += 2) { | |
169 s1 += p1[i] * qmf_window[i]; | |
170 s2 += p1[i+1] * qmf_window[i+1]; | |
171 } | |
172 | |
173 pOut[0] = s2; | |
174 pOut[1] = s1; | |
175 | |
176 p1 += 2; | |
177 pOut += 2; | |
178 } | |
179 | |
180 /* Update the delay buffer. */ | |
181 memcpy(delayBuf, temp + nIn*2, 46*sizeof(float)); | |
182 } | |
183 | |
184 /** | |
185 * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands | |
186 * caused by the reverse spectra of the QMF. | |
187 * | |
188 * @param pInput float input | |
189 * @param pOutput float output | |
190 * @param odd_band 1 if the band is an odd band | |
191 */ | |
192 | |
7546 | 193 static void IMLT(float *pInput, float *pOutput, int odd_band) |
4856 | 194 { |
195 int i; | |
196 | |
197 if (odd_band) { | |
198 /** | |
199 * Reverse the odd bands before IMDCT, this is an effect of the QMF transform | |
200 * or it gives better compression to do it this way. | |
201 * FIXME: It should be possible to handle this in ff_imdct_calc | |
202 * for that to happen a modification of the prerotation step of | |
203 * all SIMD code and C code is needed. | |
204 * Or fix the functions before so they generate a pre reversed spectrum. | |
205 */ | |
206 | |
207 for (i=0; i<128; i++) | |
208 FFSWAP(float, pInput[i], pInput[255-i]); | |
209 } | |
210 | |
7547 | 211 ff_imdct_calc(&mdct_ctx,pOutput,pInput); |
4856 | 212 |
213 /* Perform windowing on the output. */ | |
214 dsp.vector_fmul(pOutput,mdct_window,512); | |
215 | |
216 } | |
217 | |
218 | |
219 /** | |
220 * Atrac 3 indata descrambling, only used for data coming from the rm container | |
221 * | |
222 * @param in pointer to 8 bit array of indata | |
223 * @param bits amount of bits | |
224 * @param out pointer to 8 bit array of outdata | |
225 */ | |
226 | |
6228 | 227 static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){ |
4856 | 228 int i, off; |
229 uint32_t c; | |
6228 | 230 const uint32_t* buf; |
4856 | 231 uint32_t* obuf = (uint32_t*) out; |
232 | |
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233 off = (intptr_t)inbuffer & 3; |
6228 | 234 buf = (const uint32_t*) (inbuffer - off); |
4856 | 235 c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8)))); |
236 bytes += 3 + off; | |
237 for (i = 0; i < bytes/4; i++) | |
238 obuf[i] = c ^ buf[i]; | |
239 | |
240 if (off) | |
241 av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off); | |
242 | |
243 return off; | |
244 } | |
245 | |
246 | |
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247 static av_cold void init_atrac3_transforms(ATRAC3Context *q) { |
4856 | 248 float enc_window[256]; |
249 float s; | |
250 int i; | |
251 | |
252 /* Generate the mdct window, for details see | |
253 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */ | |
254 for (i=0 ; i<256; i++) | |
255 enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5; | |
256 | |
257 if (!mdct_window[0]) | |
258 for (i=0 ; i<256; i++) { | |
259 mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]); | |
260 mdct_window[511-i] = mdct_window[i]; | |
261 } | |
262 | |
263 /* Generate the QMF window. */ | |
264 for (i=0 ; i<24; i++) { | |
265 s = qmf_48tap_half[i] * 2.0; | |
266 qmf_window[i] = s; | |
267 qmf_window[47 - i] = s; | |
268 } | |
269 | |
270 /* Initialize the MDCT transform. */ | |
271 ff_mdct_init(&mdct_ctx, 9, 1); | |
272 } | |
273 | |
274 /** | |
275 * Atrac3 uninit, free all allocated memory | |
276 */ | |
277 | |
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278 static av_cold int atrac3_decode_close(AVCodecContext *avctx) |
4856 | 279 { |
280 ATRAC3Context *q = avctx->priv_data; | |
281 | |
282 av_free(q->pUnits); | |
283 av_free(q->decoded_bytes_buffer); | |
284 | |
285 return 0; | |
286 } | |
287 | |
288 /** | |
289 / * Mantissa decoding | |
290 * | |
291 * @param gb the GetBit context | |
292 * @param selector what table is the output values coded with | |
293 * @param codingFlag constant length coding or variable length coding | |
294 * @param mantissas mantissa output table | |
295 * @param numCodes amount of values to get | |
296 */ | |
297 | |
298 static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes) | |
299 { | |
300 int numBits, cnt, code, huffSymb; | |
301 | |
302 if (selector == 1) | |
303 numCodes /= 2; | |
304 | |
305 if (codingFlag != 0) { | |
306 /* constant length coding (CLC) */ | |
307 numBits = CLCLengthTab[selector]; | |
308 | |
309 if (selector > 1) { | |
310 for (cnt = 0; cnt < numCodes; cnt++) { | |
311 if (numBits) | |
312 code = get_sbits(gb, numBits); | |
313 else | |
314 code = 0; | |
315 mantissas[cnt] = code; | |
316 } | |
317 } else { | |
318 for (cnt = 0; cnt < numCodes; cnt++) { | |
319 if (numBits) | |
320 code = get_bits(gb, numBits); //numBits is always 4 in this case | |
321 else | |
322 code = 0; | |
323 mantissas[cnt*2] = seTab_0[code >> 2]; | |
324 mantissas[cnt*2+1] = seTab_0[code & 3]; | |
325 } | |
326 } | |
327 } else { | |
328 /* variable length coding (VLC) */ | |
329 if (selector != 1) { | |
330 for (cnt = 0; cnt < numCodes; cnt++) { | |
331 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3); | |
332 huffSymb += 1; | |
333 code = huffSymb >> 1; | |
334 if (huffSymb & 1) | |
335 code = -code; | |
336 mantissas[cnt] = code; | |
337 } | |
338 } else { | |
339 for (cnt = 0; cnt < numCodes; cnt++) { | |
340 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3); | |
341 mantissas[cnt*2] = decTable1[huffSymb*2]; | |
342 mantissas[cnt*2+1] = decTable1[huffSymb*2+1]; | |
343 } | |
344 } | |
345 } | |
346 } | |
347 | |
348 /** | |
349 * Restore the quantized band spectrum coefficients | |
350 * | |
351 * @param gb the GetBit context | |
352 * @param pOut decoded band spectrum | |
353 * @return outSubbands subband counter, fix for broken specification/files | |
354 */ | |
355 | |
356 static int decodeSpectrum (GetBitContext *gb, float *pOut) | |
357 { | |
358 int numSubbands, codingMode, cnt, first, last, subbWidth, *pIn; | |
359 int subband_vlc_index[32], SF_idxs[32]; | |
360 int mantissas[128]; | |
361 float SF; | |
362 | |
363 numSubbands = get_bits(gb, 5); // number of coded subbands | |
5513 | 364 codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC |
4856 | 365 |
366 /* Get the VLC selector table for the subbands, 0 means not coded. */ | |
367 for (cnt = 0; cnt <= numSubbands; cnt++) | |
368 subband_vlc_index[cnt] = get_bits(gb, 3); | |
369 | |
370 /* Read the scale factor indexes from the stream. */ | |
371 for (cnt = 0; cnt <= numSubbands; cnt++) { | |
372 if (subband_vlc_index[cnt] != 0) | |
373 SF_idxs[cnt] = get_bits(gb, 6); | |
374 } | |
375 | |
376 for (cnt = 0; cnt <= numSubbands; cnt++) { | |
377 first = subbandTab[cnt]; | |
378 last = subbandTab[cnt+1]; | |
379 | |
380 subbWidth = last - first; | |
381 | |
382 if (subband_vlc_index[cnt] != 0) { | |
383 /* Decode spectral coefficients for this subband. */ | |
384 /* TODO: This can be done faster is several blocks share the | |
385 * same VLC selector (subband_vlc_index) */ | |
386 readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth); | |
387 | |
388 /* Decode the scale factor for this subband. */ | |
389 SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]]; | |
390 | |
391 /* Inverse quantize the coefficients. */ | |
392 for (pIn=mantissas ; first<last; first++, pIn++) | |
393 pOut[first] = *pIn * SF; | |
394 } else { | |
395 /* This subband was not coded, so zero the entire subband. */ | |
396 memset(pOut+first, 0, subbWidth*sizeof(float)); | |
397 } | |
398 } | |
399 | |
400 /* Clear the subbands that were not coded. */ | |
401 first = subbandTab[cnt]; | |
402 memset(pOut+first, 0, (1024 - first) * sizeof(float)); | |
403 return numSubbands; | |
404 } | |
405 | |
406 /** | |
407 * Restore the quantized tonal components | |
408 * | |
409 * @param gb the GetBit context | |
410 * @param pComponent tone component | |
411 * @param numBands amount of coded bands | |
412 */ | |
413 | |
4865 | 414 static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands) |
4856 | 415 { |
416 int i,j,k,cnt; | |
4865 | 417 int components, coding_mode_selector, coding_mode, coded_values_per_component; |
4856 | 418 int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components; |
419 int band_flags[4], mantissa[8]; | |
420 float *pCoef; | |
421 float scalefactor; | |
4865 | 422 int component_count = 0; |
4856 | 423 |
424 components = get_bits(gb,5); | |
425 | |
426 /* no tonal components */ | |
427 if (components == 0) | |
428 return 0; | |
429 | |
430 coding_mode_selector = get_bits(gb,2); | |
431 if (coding_mode_selector == 2) | |
432 return -1; | |
433 | |
434 coding_mode = coding_mode_selector & 1; | |
435 | |
436 for (i = 0; i < components; i++) { | |
437 for (cnt = 0; cnt <= numBands; cnt++) | |
438 band_flags[cnt] = get_bits1(gb); | |
439 | |
440 coded_values_per_component = get_bits(gb,3); | |
441 | |
442 quant_step_index = get_bits(gb,3); | |
443 if (quant_step_index <= 1) | |
444 return -1; | |
445 | |
446 if (coding_mode_selector == 3) | |
447 coding_mode = get_bits1(gb); | |
448 | |
449 for (j = 0; j < (numBands + 1) * 4; j++) { | |
450 if (band_flags[j >> 2] == 0) | |
451 continue; | |
452 | |
453 coded_components = get_bits(gb,3); | |
454 | |
455 for (k=0; k<coded_components; k++) { | |
456 sfIndx = get_bits(gb,6); | |
457 pComponent[component_count].pos = j * 64 + (get_bits(gb,6)); | |
458 max_coded_values = 1024 - pComponent[component_count].pos; | |
459 coded_values = coded_values_per_component + 1; | |
460 coded_values = FFMIN(max_coded_values,coded_values); | |
461 | |
462 scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index]; | |
463 | |
464 readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values); | |
465 | |
466 pComponent[component_count].numCoefs = coded_values; | |
467 | |
468 /* inverse quant */ | |
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469 pCoef = pComponent[component_count].coef; |
4856 | 470 for (cnt = 0; cnt < coded_values; cnt++) |
471 pCoef[cnt] = mantissa[cnt] * scalefactor; | |
472 | |
473 component_count++; | |
474 } | |
475 } | |
476 } | |
477 | |
4865 | 478 return component_count; |
4856 | 479 } |
480 | |
481 /** | |
482 * Decode gain parameters for the coded bands | |
483 * | |
484 * @param gb the GetBit context | |
485 * @param pGb the gainblock for the current band | |
486 * @param numBands amount of coded bands | |
487 */ | |
488 | |
489 static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands) | |
490 { | |
491 int i, cf, numData; | |
492 int *pLevel, *pLoc; | |
493 | |
494 gain_info *pGain = pGb->gBlock; | |
495 | |
496 for (i=0 ; i<=numBands; i++) | |
497 { | |
498 numData = get_bits(gb,3); | |
499 pGain[i].num_gain_data = numData; | |
500 pLevel = pGain[i].levcode; | |
501 pLoc = pGain[i].loccode; | |
502 | |
503 for (cf = 0; cf < numData; cf++){ | |
504 pLevel[cf]= get_bits(gb,4); | |
505 pLoc [cf]= get_bits(gb,5); | |
506 if(cf && pLoc[cf] <= pLoc[cf-1]) | |
507 return -1; | |
508 } | |
509 } | |
510 | |
511 /* Clear the unused blocks. */ | |
512 for (; i<4 ; i++) | |
513 pGain[i].num_gain_data = 0; | |
514 | |
515 return 0; | |
516 } | |
517 | |
518 /** | |
519 * Apply gain parameters and perform the MDCT overlapping part | |
520 * | |
521 * @param pIn input float buffer | |
522 * @param pPrev previous float buffer to perform overlap against | |
523 * @param pOut output float buffer | |
524 * @param pGain1 current band gain info | |
525 * @param pGain2 next band gain info | |
526 */ | |
527 | |
528 static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2) | |
529 { | |
530 /* gain compensation function */ | |
531 float gain1, gain2, gain_inc; | |
532 int cnt, numdata, nsample, startLoc, endLoc; | |
533 | |
534 | |
535 if (pGain2->num_gain_data == 0) | |
536 gain1 = 1.0; | |
537 else | |
538 gain1 = gain_tab1[pGain2->levcode[0]]; | |
539 | |
540 if (pGain1->num_gain_data == 0) { | |
541 for (cnt = 0; cnt < 256; cnt++) | |
542 pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt]; | |
543 } else { | |
544 numdata = pGain1->num_gain_data; | |
545 pGain1->loccode[numdata] = 32; | |
546 pGain1->levcode[numdata] = 4; | |
547 | |
548 nsample = 0; // current sample = 0 | |
549 | |
550 for (cnt = 0; cnt < numdata; cnt++) { | |
551 startLoc = pGain1->loccode[cnt] * 8; | |
552 endLoc = startLoc + 8; | |
553 | |
554 gain2 = gain_tab1[pGain1->levcode[cnt]]; | |
555 gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15]; | |
556 | |
557 /* interpolate */ | |
558 for (; nsample < startLoc; nsample++) | |
559 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; | |
560 | |
561 /* interpolation is done over eight samples */ | |
562 for (; nsample < endLoc; nsample++) { | |
563 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; | |
564 gain2 *= gain_inc; | |
565 } | |
566 } | |
567 | |
568 for (; nsample < 256; nsample++) | |
569 pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample]; | |
570 } | |
571 | |
572 /* Delay for the overlapping part. */ | |
573 memcpy(pPrev, &pIn[256], 256*sizeof(float)); | |
574 } | |
575 | |
576 /** | |
577 * Combine the tonal band spectrum and regular band spectrum | |
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578 * Return position of the last tonal coefficient |
4856 | 579 * |
580 * @param pSpectrum output spectrum buffer | |
581 * @param numComponents amount of tonal components | |
582 * @param pComponent tonal components for this band | |
583 */ | |
584 | |
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585 static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent) |
4856 | 586 { |
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587 int cnt, i, lastPos = -1; |
4856 | 588 float *pIn, *pOut; |
589 | |
590 for (cnt = 0; cnt < numComponents; cnt++){ | |
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591 lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos); |
4856 | 592 pIn = pComponent[cnt].coef; |
593 pOut = &(pSpectrum[pComponent[cnt].pos]); | |
594 | |
595 for (i=0 ; i<pComponent[cnt].numCoefs ; i++) | |
596 pOut[i] += pIn[i]; | |
597 } | |
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598 |
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599 return lastPos; |
4856 | 600 } |
601 | |
602 | |
603 #define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old))) | |
604 | |
605 static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode) | |
606 { | |
607 int i, band, nsample, s1, s2; | |
608 float c1, c2; | |
609 float mc1_l, mc1_r, mc2_l, mc2_r; | |
610 | |
611 for (i=0,band = 0; band < 4*256; band+=256,i++) { | |
612 s1 = pPrevCode[i]; | |
613 s2 = pCurrCode[i]; | |
614 nsample = 0; | |
615 | |
616 if (s1 != s2) { | |
617 /* Selector value changed, interpolation needed. */ | |
618 mc1_l = matrixCoeffs[s1*2]; | |
619 mc1_r = matrixCoeffs[s1*2+1]; | |
620 mc2_l = matrixCoeffs[s2*2]; | |
621 mc2_r = matrixCoeffs[s2*2+1]; | |
622 | |
623 /* Interpolation is done over the first eight samples. */ | |
624 for(; nsample < 8; nsample++) { | |
625 c1 = su1[band+nsample]; | |
626 c2 = su2[band+nsample]; | |
627 c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample); | |
628 su1[band+nsample] = c2; | |
629 su2[band+nsample] = c1 * 2.0 - c2; | |
630 } | |
631 } | |
632 | |
633 /* Apply the matrix without interpolation. */ | |
634 switch (s2) { | |
635 case 0: /* M/S decoding */ | |
636 for (; nsample < 256; nsample++) { | |
637 c1 = su1[band+nsample]; | |
638 c2 = su2[band+nsample]; | |
639 su1[band+nsample] = c2 * 2.0; | |
640 su2[band+nsample] = (c1 - c2) * 2.0; | |
641 } | |
642 break; | |
643 | |
644 case 1: | |
645 for (; nsample < 256; nsample++) { | |
646 c1 = su1[band+nsample]; | |
647 c2 = su2[band+nsample]; | |
648 su1[band+nsample] = (c1 + c2) * 2.0; | |
649 su2[band+nsample] = c2 * -2.0; | |
650 } | |
651 break; | |
652 case 2: | |
653 case 3: | |
654 for (; nsample < 256; nsample++) { | |
655 c1 = su1[band+nsample]; | |
656 c2 = su2[band+nsample]; | |
657 su1[band+nsample] = c1 + c2; | |
658 su2[band+nsample] = c1 - c2; | |
659 } | |
660 break; | |
661 default: | |
662 assert(0); | |
663 } | |
664 } | |
665 } | |
666 | |
667 static void getChannelWeights (int indx, int flag, float ch[2]){ | |
668 | |
669 if (indx == 7) { | |
670 ch[0] = 1.0; | |
671 ch[1] = 1.0; | |
672 } else { | |
673 ch[0] = (float)(indx & 7) / 7.0; | |
674 ch[1] = sqrt(2 - ch[0]*ch[0]); | |
675 if(flag) | |
676 FFSWAP(float, ch[0], ch[1]); | |
677 } | |
678 } | |
679 | |
680 static void channelWeighting (float *su1, float *su2, int *p3) | |
681 { | |
682 int band, nsample; | |
683 /* w[x][y] y=0 is left y=1 is right */ | |
684 float w[2][2]; | |
685 | |
686 if (p3[1] != 7 || p3[3] != 7){ | |
687 getChannelWeights(p3[1], p3[0], w[0]); | |
688 getChannelWeights(p3[3], p3[2], w[1]); | |
689 | |
690 for(band = 1; band < 4; band++) { | |
691 /* scale the channels by the weights */ | |
692 for(nsample = 0; nsample < 8; nsample++) { | |
693 su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample); | |
694 su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample); | |
695 } | |
696 | |
697 for(; nsample < 256; nsample++) { | |
698 su1[band*256+nsample] *= w[1][0]; | |
699 su2[band*256+nsample] *= w[1][1]; | |
700 } | |
701 } | |
702 } | |
703 } | |
704 | |
705 | |
706 /** | |
707 * Decode a Sound Unit | |
708 * | |
709 * @param gb the GetBit context | |
710 * @param pSnd the channel unit to be used | |
711 * @param pOut the decoded samples before IQMF in float representation | |
712 * @param channelNum channel number | |
713 * @param codingMode the coding mode (JOINT_STEREO or regular stereo/mono) | |
714 */ | |
715 | |
716 | |
717 static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode) | |
718 { | |
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719 int band, result=0, numSubbands, lastTonal, numBands; |
4856 | 720 |
721 if (codingMode == JOINT_STEREO && channelNum == 1) { | |
722 if (get_bits(gb,2) != 3) { | |
723 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n"); | |
724 return -1; | |
725 } | |
726 } else { | |
727 if (get_bits(gb,6) != 0x28) { | |
728 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n"); | |
729 return -1; | |
730 } | |
731 } | |
732 | |
733 /* number of coded QMF bands */ | |
734 pSnd->bandsCoded = get_bits(gb,2); | |
735 | |
736 result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded); | |
737 if (result) return result; | |
738 | |
4865 | 739 pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded); |
740 if (pSnd->numComponents == -1) return -1; | |
4856 | 741 |
742 numSubbands = decodeSpectrum (gb, pSnd->spectrum); | |
743 | |
744 /* Merge the decoded spectrum and tonal components. */ | |
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745 lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components); |
4856 | 746 |
747 | |
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748 /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */ |
4856 | 749 numBands = (subbandTab[numSubbands] - 1) >> 8; |
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750 if (lastTonal >= 0) |
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751 numBands = FFMAX((lastTonal + 256) >> 8, numBands); |
4856 | 752 |
753 | |
754 /* Reconstruct time domain samples. */ | |
755 for (band=0; band<4; band++) { | |
756 /* Perform the IMDCT step without overlapping. */ | |
757 if (band <= numBands) { | |
7546 | 758 IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1); |
4856 | 759 } else |
760 memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float)); | |
761 | |
762 /* gain compensation and overlapping */ | |
763 gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]), | |
764 &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]), | |
765 &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band])); | |
766 } | |
767 | |
768 /* Swap the gain control buffers for the next frame. */ | |
769 pSnd->gcBlkSwitch ^= 1; | |
770 | |
771 return 0; | |
772 } | |
773 | |
774 /** | |
775 * Frame handling | |
776 * | |
777 * @param q Atrac3 private context | |
778 * @param databuf the input data | |
779 */ | |
780 | |
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781 static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf) |
4856 | 782 { |
783 int result, i; | |
784 float *p1, *p2, *p3, *p4; | |
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785 uint8_t *ptr1; |
4856 | 786 |
787 if (q->codingMode == JOINT_STEREO) { | |
788 | |
789 /* channel coupling mode */ | |
790 /* decode Sound Unit 1 */ | |
791 init_get_bits(&q->gb,databuf,q->bits_per_frame); | |
792 | |
793 result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO); | |
794 if (result != 0) | |
795 return (result); | |
796 | |
797 /* Framedata of the su2 in the joint-stereo mode is encoded in | |
798 * reverse byte order so we need to swap it first. */ | |
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799 if (databuf == q->decoded_bytes_buffer) { |
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800 uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1; |
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801 ptr1 = q->decoded_bytes_buffer; |
7987 | 802 for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) { |
803 FFSWAP(uint8_t,*ptr1,*ptr2); | |
804 } | |
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805 } else { |
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806 const uint8_t *ptr2 = databuf+q->bytes_per_frame-1; |
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807 for (i = 0; i < q->bytes_per_frame; i++) |
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808 q->decoded_bytes_buffer[i] = *ptr2--; |
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809 } |
4856 | 810 |
811 /* Skip the sync codes (0xF8). */ | |
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812 ptr1 = q->decoded_bytes_buffer; |
4856 | 813 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) { |
814 if (i >= q->bytes_per_frame) | |
815 return -1; | |
816 } | |
817 | |
818 | |
819 /* set the bitstream reader at the start of the second Sound Unit*/ | |
820 init_get_bits(&q->gb,ptr1,q->bits_per_frame); | |
821 | |
822 /* Fill the Weighting coeffs delay buffer */ | |
823 memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int)); | |
5513 | 824 q->weighting_delay[4] = get_bits1(&q->gb); |
4856 | 825 q->weighting_delay[5] = get_bits(&q->gb,3); |
826 | |
827 for (i = 0; i < 4; i++) { | |
828 q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i]; | |
829 q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i]; | |
830 q->matrix_coeff_index_next[i] = get_bits(&q->gb,2); | |
831 } | |
832 | |
833 /* Decode Sound Unit 2. */ | |
834 result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO); | |
835 if (result != 0) | |
836 return (result); | |
837 | |
838 /* Reconstruct the channel coefficients. */ | |
839 reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now); | |
840 | |
841 channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay); | |
842 | |
843 } else { | |
844 /* normal stereo mode or mono */ | |
845 /* Decode the channel sound units. */ | |
846 for (i=0 ; i<q->channels ; i++) { | |
847 | |
848 /* Set the bitstream reader at the start of a channel sound unit. */ | |
849 init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels); | |
850 | |
851 result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode); | |
852 if (result != 0) | |
853 return (result); | |
854 } | |
855 } | |
856 | |
857 /* Apply the iQMF synthesis filter. */ | |
858 p1= q->outSamples; | |
859 for (i=0 ; i<q->channels ; i++) { | |
860 p2= p1+256; | |
861 p3= p2+256; | |
862 p4= p3+256; | |
863 iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf); | |
864 iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf); | |
865 iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf); | |
866 p1 +=1024; | |
867 } | |
868 | |
869 return 0; | |
870 } | |
871 | |
872 | |
873 /** | |
874 * Atrac frame decoding | |
875 * | |
876 * @param avctx pointer to the AVCodecContext | |
877 */ | |
878 | |
879 static int atrac3_decode_frame(AVCodecContext *avctx, | |
880 void *data, int *data_size, | |
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881 AVPacket *avpkt) { |
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882 const uint8_t *buf = avpkt->data; |
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883 int buf_size = avpkt->size; |
4856 | 884 ATRAC3Context *q = avctx->priv_data; |
885 int result = 0, i; | |
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886 const uint8_t* databuf; |
4856 | 887 int16_t* samples = data; |
888 | |
889 if (buf_size < avctx->block_align) | |
890 return buf_size; | |
891 | |
892 /* Check if we need to descramble and what buffer to pass on. */ | |
893 if (q->scrambled_stream) { | |
894 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align); | |
895 databuf = q->decoded_bytes_buffer; | |
896 } else { | |
897 databuf = buf; | |
898 } | |
899 | |
900 result = decodeFrame(q, databuf); | |
901 | |
902 if (result != 0) { | |
903 av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n"); | |
904 return -1; | |
905 } | |
906 | |
907 if (q->channels == 1) { | |
908 /* mono */ | |
909 for (i = 0; i<1024; i++) | |
5523 | 910 samples[i] = av_clip_int16(round(q->outSamples[i])); |
4856 | 911 *data_size = 1024 * sizeof(int16_t); |
912 } else { | |
913 /* stereo */ | |
914 for (i = 0; i < 1024; i++) { | |
5523 | 915 samples[i*2] = av_clip_int16(round(q->outSamples[i])); |
916 samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i])); | |
4856 | 917 } |
918 *data_size = 2048 * sizeof(int16_t); | |
919 } | |
920 | |
921 return avctx->block_align; | |
922 } | |
923 | |
924 | |
925 /** | |
926 * Atrac3 initialization | |
927 * | |
928 * @param avctx pointer to the AVCodecContext | |
929 */ | |
930 | |
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931 static av_cold int atrac3_decode_init(AVCodecContext *avctx) |
4856 | 932 { |
933 int i; | |
6228 | 934 const uint8_t *edata_ptr = avctx->extradata; |
4856 | 935 ATRAC3Context *q = avctx->priv_data; |
936 | |
937 /* Take data from the AVCodecContext (RM container). */ | |
938 q->sample_rate = avctx->sample_rate; | |
939 q->channels = avctx->channels; | |
940 q->bit_rate = avctx->bit_rate; | |
941 q->bits_per_frame = avctx->block_align * 8; | |
942 q->bytes_per_frame = avctx->block_align; | |
943 | |
944 /* Take care of the codec-specific extradata. */ | |
945 if (avctx->extradata_size == 14) { | |
946 /* Parse the extradata, WAV format */ | |
947 av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1 | |
948 q->samples_per_channel = bytestream_get_le32(&edata_ptr); | |
949 q->codingMode = bytestream_get_le16(&edata_ptr); | |
950 av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode | |
951 q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1 | |
952 av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0 | |
953 | |
954 /* setup */ | |
955 q->samples_per_frame = 1024 * q->channels; | |
956 q->atrac3version = 4; | |
957 q->delay = 0x88E; | |
958 if (q->codingMode) | |
959 q->codingMode = JOINT_STEREO; | |
960 else | |
961 q->codingMode = STEREO; | |
962 | |
963 q->scrambled_stream = 0; | |
964 | |
965 if ((q->bytes_per_frame == 96*q->channels*q->frame_factor) || (q->bytes_per_frame == 152*q->channels*q->frame_factor) || (q->bytes_per_frame == 192*q->channels*q->frame_factor)) { | |
966 } else { | |
967 av_log(avctx,AV_LOG_ERROR,"Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor); | |
968 return -1; | |
969 } | |
970 | |
971 } else if (avctx->extradata_size == 10) { | |
972 /* Parse the extradata, RM format. */ | |
973 q->atrac3version = bytestream_get_be32(&edata_ptr); | |
974 q->samples_per_frame = bytestream_get_be16(&edata_ptr); | |
975 q->delay = bytestream_get_be16(&edata_ptr); | |
976 q->codingMode = bytestream_get_be16(&edata_ptr); | |
977 | |
978 q->samples_per_channel = q->samples_per_frame / q->channels; | |
979 q->scrambled_stream = 1; | |
980 | |
981 } else { | |
982 av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size); | |
983 } | |
984 /* Check the extradata. */ | |
985 | |
986 if (q->atrac3version != 4) { | |
987 av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version); | |
988 return -1; | |
989 } | |
990 | |
991 if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) { | |
992 av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame); | |
993 return -1; | |
994 } | |
995 | |
996 if (q->delay != 0x88E) { | |
997 av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay); | |
998 return -1; | |
999 } | |
1000 | |
1001 if (q->codingMode == STEREO) { | |
1002 av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n"); | |
1003 } else if (q->codingMode == JOINT_STEREO) { | |
1004 av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n"); | |
1005 } else { | |
1006 av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode); | |
1007 return -1; | |
1008 } | |
1009 | |
1010 if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) { | |
1011 av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n"); | |
1012 return -1; | |
1013 } | |
1014 | |
1015 | |
1016 if(avctx->block_align >= UINT_MAX/2) | |
1017 return -1; | |
1018 | |
1019 /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE, | |
1020 * this is for the bitstream reader. */ | |
1021 if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE))) == NULL) | |
5407 | 1022 return AVERROR(ENOMEM); |
4856 | 1023 |
1024 | |
1025 /* Initialize the VLC tables. */ | |
1026 for (i=0 ; i<7 ; i++) { | |
1027 init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i], | |
1028 huff_bits[i], 1, 1, | |
1029 huff_codes[i], 1, 1, INIT_VLC_USE_STATIC); | |
1030 } | |
1031 | |
1032 init_atrac3_transforms(q); | |
1033 | |
1034 /* Generate the scale factors. */ | |
1035 for (i=0 ; i<64 ; i++) | |
1036 SFTable[i] = pow(2.0, (i - 15) / 3.0); | |
1037 | |
1038 /* Generate gain tables. */ | |
1039 for (i=0 ; i<16 ; i++) | |
1040 gain_tab1[i] = powf (2.0, (4 - i)); | |
1041 | |
1042 for (i=-15 ; i<16 ; i++) | |
1043 gain_tab2[i+15] = powf (2.0, i * -0.125); | |
1044 | |
1045 /* init the joint-stereo decoding data */ | |
1046 q->weighting_delay[0] = 0; | |
1047 q->weighting_delay[1] = 7; | |
1048 q->weighting_delay[2] = 0; | |
1049 q->weighting_delay[3] = 7; | |
1050 q->weighting_delay[4] = 0; | |
1051 q->weighting_delay[5] = 7; | |
1052 | |
1053 for (i=0; i<4; i++) { | |
1054 q->matrix_coeff_index_prev[i] = 3; | |
1055 q->matrix_coeff_index_now[i] = 3; | |
1056 q->matrix_coeff_index_next[i] = 3; | |
1057 } | |
1058 | |
1059 dsputil_init(&dsp, avctx); | |
1060 | |
1061 q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels); | |
5423 | 1062 if (!q->pUnits) { |
1063 av_free(q->decoded_bytes_buffer); | |
1064 return AVERROR(ENOMEM); | |
1065 } | |
4856 | 1066 |
7451
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
pross
parents:
7040
diff
changeset
|
1067 avctx->sample_fmt = SAMPLE_FMT_S16; |
4856 | 1068 return 0; |
1069 } | |
1070 | |
1071 | |
1072 AVCodec atrac3_decoder = | |
1073 { | |
6716 | 1074 .name = "atrac3", |
4856 | 1075 .type = CODEC_TYPE_AUDIO, |
1076 .id = CODEC_ID_ATRAC3, | |
1077 .priv_data_size = sizeof(ATRAC3Context), | |
1078 .init = atrac3_decode_init, | |
1079 .close = atrac3_decode_close, | |
1080 .decode = atrac3_decode_frame, | |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
6997
diff
changeset
|
1081 .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"), |
4856 | 1082 }; |