atrac3dec: cosmetics: indentation
[libav.git] / libavcodec / atrac3.c
1 /*
2 * Atrac 3 compatible decoder
3 * Copyright (c) 2006-2008 Maxim Poliakovski
4 * Copyright (c) 2006-2008 Benjamin Larsson
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 /**
24 * @file atrac3.c
25 * Atrac 3 compatible decoder.
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).
30 *
31 * To use this decoder, a calling application must supply the extradata
32 * bytes provided in the containers above.
33 */
34
35 #include <math.h>
36 #include <stddef.h>
37 #include <stdio.h>
38
39 #include "avcodec.h"
40 #include "bitstream.h"
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
193 static void IMLT(float *pInput, float *pOutput, int odd_band)
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
211 ff_imdct_calc(&mdct_ctx,pOutput,pInput);
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
227 static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
228 int i, off;
229 uint32_t c;
230 const uint32_t* buf;
231 uint32_t* obuf = (uint32_t*) out;
232
233 off = (int)((long)inbuffer & 3);
234 buf = (const uint32_t*) (inbuffer - off);
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
247 static void init_atrac3_transforms(ATRAC3Context *q) {
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
278 static int atrac3_decode_close(AVCodecContext *avctx)
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
364 codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
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
414 static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
415 {
416 int i,j,k,cnt;
417 int components, coding_mode_selector, coding_mode, coded_values_per_component;
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;
422 int component_count = 0;
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 */
469 pCoef = pComponent[component_count].coef;
470 for (cnt = 0; cnt < coded_values; cnt++)
471 pCoef[cnt] = mantissa[cnt] * scalefactor;
472
473 component_count++;
474 }
475 }
476 }
477
478 return component_count;
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
578 * Return position of the last tonal coefficient
579 *
580 * @param pSpectrum output spectrum buffer
581 * @param numComponents amount of tonal components
582 * @param pComponent tonal components for this band
583 */
584
585 static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
586 {
587 int cnt, i, lastPos = -1;
588 float *pIn, *pOut;
589
590 for (cnt = 0; cnt < numComponents; cnt++){
591 lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);
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 }
598
599 return lastPos;
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 {
719 int band, result=0, numSubbands, lastTonal, numBands;
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
739 pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
740 if (pSnd->numComponents == -1) return -1;
741
742 numSubbands = decodeSpectrum (gb, pSnd->spectrum);
743
744 /* Merge the decoded spectrum and tonal components. */
745 lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
746
747
748 /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
749 numBands = (subbandTab[numSubbands] - 1) >> 8;
750 if (lastTonal >= 0)
751 numBands = FFMAX((lastTonal + 256) >> 8, numBands);
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) {
758 IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
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
781 static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
782 {
783 int result, i;
784 float *p1, *p2, *p3, *p4;
785 uint8_t *ptr1;
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. */
799 if (databuf == q->decoded_bytes_buffer) {
800 uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1;
801 ptr1 = q->decoded_bytes_buffer;
802 for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
803 FFSWAP(uint8_t,*ptr1,*ptr2);
804 }
805 } else {
806 const uint8_t *ptr2 = databuf+q->bytes_per_frame-1;
807 for (i = 0; i < q->bytes_per_frame; i++)
808 q->decoded_bytes_buffer[i] = *ptr2--;
809 }
810
811 /* Skip the sync codes (0xF8). */
812 ptr1 = q->decoded_bytes_buffer;
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));
824 q->weighting_delay[4] = get_bits1(&q->gb);
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,
881 const uint8_t *buf, int buf_size) {
882 ATRAC3Context *q = avctx->priv_data;
883 int result = 0, i;
884 const uint8_t* databuf;
885 int16_t* samples = data;
886
887 if (buf_size < avctx->block_align)
888 return buf_size;
889
890 /* Check if we need to descramble and what buffer to pass on. */
891 if (q->scrambled_stream) {
892 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
893 databuf = q->decoded_bytes_buffer;
894 } else {
895 databuf = buf;
896 }
897
898 result = decodeFrame(q, databuf);
899
900 if (result != 0) {
901 av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
902 return -1;
903 }
904
905 if (q->channels == 1) {
906 /* mono */
907 for (i = 0; i<1024; i++)
908 samples[i] = av_clip_int16(round(q->outSamples[i]));
909 *data_size = 1024 * sizeof(int16_t);
910 } else {
911 /* stereo */
912 for (i = 0; i < 1024; i++) {
913 samples[i*2] = av_clip_int16(round(q->outSamples[i]));
914 samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
915 }
916 *data_size = 2048 * sizeof(int16_t);
917 }
918
919 return avctx->block_align;
920 }
921
922
923 /**
924 * Atrac3 initialization
925 *
926 * @param avctx pointer to the AVCodecContext
927 */
928
929 static int atrac3_decode_init(AVCodecContext *avctx)
930 {
931 int i;
932 const uint8_t *edata_ptr = avctx->extradata;
933 ATRAC3Context *q = avctx->priv_data;
934
935 /* Take data from the AVCodecContext (RM container). */
936 q->sample_rate = avctx->sample_rate;
937 q->channels = avctx->channels;
938 q->bit_rate = avctx->bit_rate;
939 q->bits_per_frame = avctx->block_align * 8;
940 q->bytes_per_frame = avctx->block_align;
941
942 /* Take care of the codec-specific extradata. */
943 if (avctx->extradata_size == 14) {
944 /* Parse the extradata, WAV format */
945 av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1
946 q->samples_per_channel = bytestream_get_le32(&edata_ptr);
947 q->codingMode = bytestream_get_le16(&edata_ptr);
948 av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
949 q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1
950 av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0
951
952 /* setup */
953 q->samples_per_frame = 1024 * q->channels;
954 q->atrac3version = 4;
955 q->delay = 0x88E;
956 if (q->codingMode)
957 q->codingMode = JOINT_STEREO;
958 else
959 q->codingMode = STEREO;
960
961 q->scrambled_stream = 0;
962
963 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)) {
964 } else {
965 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);
966 return -1;
967 }
968
969 } else if (avctx->extradata_size == 10) {
970 /* Parse the extradata, RM format. */
971 q->atrac3version = bytestream_get_be32(&edata_ptr);
972 q->samples_per_frame = bytestream_get_be16(&edata_ptr);
973 q->delay = bytestream_get_be16(&edata_ptr);
974 q->codingMode = bytestream_get_be16(&edata_ptr);
975
976 q->samples_per_channel = q->samples_per_frame / q->channels;
977 q->scrambled_stream = 1;
978
979 } else {
980 av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
981 }
982 /* Check the extradata. */
983
984 if (q->atrac3version != 4) {
985 av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
986 return -1;
987 }
988
989 if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
990 av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
991 return -1;
992 }
993
994 if (q->delay != 0x88E) {
995 av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
996 return -1;
997 }
998
999 if (q->codingMode == STEREO) {
1000 av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
1001 } else if (q->codingMode == JOINT_STEREO) {
1002 av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
1003 } else {
1004 av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
1005 return -1;
1006 }
1007
1008 if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
1009 av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
1010 return -1;
1011 }
1012
1013
1014 if(avctx->block_align >= UINT_MAX/2)
1015 return -1;
1016
1017 /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
1018 * this is for the bitstream reader. */
1019 if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE))) == NULL)
1020 return AVERROR(ENOMEM);
1021
1022
1023 /* Initialize the VLC tables. */
1024 for (i=0 ; i<7 ; i++) {
1025 init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
1026 huff_bits[i], 1, 1,
1027 huff_codes[i], 1, 1, INIT_VLC_USE_STATIC);
1028 }
1029
1030 init_atrac3_transforms(q);
1031
1032 /* Generate the scale factors. */
1033 for (i=0 ; i<64 ; i++)
1034 SFTable[i] = pow(2.0, (i - 15) / 3.0);
1035
1036 /* Generate gain tables. */
1037 for (i=0 ; i<16 ; i++)
1038 gain_tab1[i] = powf (2.0, (4 - i));
1039
1040 for (i=-15 ; i<16 ; i++)
1041 gain_tab2[i+15] = powf (2.0, i * -0.125);
1042
1043 /* init the joint-stereo decoding data */
1044 q->weighting_delay[0] = 0;
1045 q->weighting_delay[1] = 7;
1046 q->weighting_delay[2] = 0;
1047 q->weighting_delay[3] = 7;
1048 q->weighting_delay[4] = 0;
1049 q->weighting_delay[5] = 7;
1050
1051 for (i=0; i<4; i++) {
1052 q->matrix_coeff_index_prev[i] = 3;
1053 q->matrix_coeff_index_now[i] = 3;
1054 q->matrix_coeff_index_next[i] = 3;
1055 }
1056
1057 dsputil_init(&dsp, avctx);
1058
1059 q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
1060 if (!q->pUnits) {
1061 av_free(q->decoded_bytes_buffer);
1062 return AVERROR(ENOMEM);
1063 }
1064
1065 avctx->sample_fmt = SAMPLE_FMT_S16;
1066 return 0;
1067 }
1068
1069
1070 AVCodec atrac3_decoder =
1071 {
1072 .name = "atrac3",
1073 .type = CODEC_TYPE_AUDIO,
1074 .id = CODEC_ID_ATRAC3,
1075 .priv_data_size = sizeof(ATRAC3Context),
1076 .init = atrac3_decode_init,
1077 .close = atrac3_decode_close,
1078 .decode = atrac3_decode_frame,
1079 .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
1080 };