Atrac3 decoder.
[libav.git] / libavcodec / atrac3.c
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1/*
2 * Atrac 3 compatible decoder
3 * Copyright (c) 2006-2007 Maxim Poliakovski
4 * Copyright (c) 2006-2007 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 RealNetworks, RealAudio atrc data.
27 * Atrac 3 is identified by the codec name atrc in RealMedia files.
28 *
29 * To use this decoder, a calling application must supply the extradata
30 * bytes provided from the RealMedia container: 10 bytes or 14 bytes
31 * from the WAV container.
32 */
33
34#include <math.h>
35#include <stddef.h>
36#include <stdio.h>
37
38#include "avcodec.h"
39#include "bitstream.h"
40#include "dsputil.h"
41#include "bytestream.h"
42
43#include "atrac3data.h"
44
45#define JOINT_STEREO 0x12
46#define STEREO 0x2
47
48
49/* These structures are needed to store the parsed gain control data. */
50typedef struct {
51 int num_gain_data;
52 int levcode[8];
53 int loccode[8];
54} gain_info;
55
56typedef struct {
57 gain_info gBlock[4];
58} gain_block;
59
60typedef struct {
61 int pos;
62 int numCoefs;
63 float coef[8];
64} tonal_component;
65
66typedef struct {
67 int bandsCoded;
68 int numComponents;
69 tonal_component components[64];
70 float prevFrame[1024];
71 int gcBlkSwitch;
72 gain_block gainBlock[2];
73
74 DECLARE_ALIGNED_16(float, spectrum[1024]);
75 DECLARE_ALIGNED_16(float, IMDCT_buf[1024]);
76
77 float delayBuf1[46]; ///<qmf delay buffers
78 float delayBuf2[46];
79 float delayBuf3[46];
80} channel_unit;
81
82typedef struct {
83 GetBitContext gb;
84 //@{
85 /** stream data */
86 int channels;
87 int codingMode;
88 int bit_rate;
89 int sample_rate;
90 int samples_per_channel;
91 int samples_per_frame;
92
93 int bits_per_frame;
94 int bytes_per_frame;
95 int pBs;
96 channel_unit* pUnits;
97 //@}
98 //@{
99 /** joint-stereo related variables */
100 int matrix_coeff_index_prev[4];
101 int matrix_coeff_index_now[4];
102 int matrix_coeff_index_next[4];
103 int weighting_delay[6];
104 //@}
105 //@{
106 /** data buffers */
107 float outSamples[2048];
108 uint8_t* decoded_bytes_buffer;
109 float tempBuf[1070];
110 DECLARE_ALIGNED_16(float,mdct_tmp[512]);
111 //@}
112 //@{
113 /** extradata */
114 int atrac3version;
115 int delay;
116 int scrambled_stream;
117 int frame_factor;
118 //@}
119} ATRAC3Context;
120
121static DECLARE_ALIGNED_16(float,mdct_window[512]);
122static float qmf_window[48];
123static VLC spectral_coeff_tab[7];
124static float SFTable[64];
125static float gain_tab1[16];
126static float gain_tab2[31];
127static MDCTContext mdct_ctx;
128static 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
145static 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 * @param mdct_tmp aligned temporary buffer for the mdct
192 */
193
194static void IMLT(float *pInput, float *pOutput, int odd_band, float* mdct_tmp)
195{
196 int i;
197
198 if (odd_band) {
199 /**
200 * Reverse the odd bands before IMDCT, this is an effect of the QMF transform
201 * or it gives better compression to do it this way.
202 * FIXME: It should be possible to handle this in ff_imdct_calc
203 * for that to happen a modification of the prerotation step of
204 * all SIMD code and C code is needed.
205 * Or fix the functions before so they generate a pre reversed spectrum.
206 */
207
208 for (i=0; i<128; i++)
209 FFSWAP(float, pInput[i], pInput[255-i]);
210 }
211
212 mdct_ctx.fft.imdct_calc(&mdct_ctx,pOutput,pInput,mdct_tmp);
213
214 /* Perform windowing on the output. */
215 dsp.vector_fmul(pOutput,mdct_window,512);
216
217}
218
219
220/**
221 * Atrac 3 indata descrambling, only used for data coming from the rm container
222 *
223 * @param in pointer to 8 bit array of indata
224 * @param bits amount of bits
225 * @param out pointer to 8 bit array of outdata
226 */
227
228static int decode_bytes(uint8_t* inbuffer, uint8_t* out, int bytes){
229 int i, off;
230 uint32_t c;
231 uint32_t* buf;
232 uint32_t* obuf = (uint32_t*) out;
233
234 off = (int)((long)inbuffer & 3);
235 buf = (uint32_t*) (inbuffer - off);
236 c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
237 bytes += 3 + off;
238 for (i = 0; i < bytes/4; i++)
239 obuf[i] = c ^ buf[i];
240
241 if (off)
242 av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);
243
244 return off;
245}
246
247
248static void init_atrac3_transforms(ATRAC3Context *q) {
249 float enc_window[256];
250 float s;
251 int i;
252
253 /* Generate the mdct window, for details see
254 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
255 for (i=0 ; i<256; i++)
256 enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
257
258 if (!mdct_window[0])
259 for (i=0 ; i<256; i++) {
260 mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
261 mdct_window[511-i] = mdct_window[i];
262 }
263
264 /* Generate the QMF window. */
265 for (i=0 ; i<24; i++) {
266 s = qmf_48tap_half[i] * 2.0;
267 qmf_window[i] = s;
268 qmf_window[47 - i] = s;
269 }
270
271 /* Initialize the MDCT transform. */
272 ff_mdct_init(&mdct_ctx, 9, 1);
273}
274
275/**
276 * Atrac3 uninit, free all allocated memory
277 */
278
279static int atrac3_decode_close(AVCodecContext *avctx)
280{
281 ATRAC3Context *q = avctx->priv_data;
282
283 av_free(q->pUnits);
284 av_free(q->decoded_bytes_buffer);
285
286 return 0;
287}
288
289/**
290/ * Mantissa decoding
291 *
292 * @param gb the GetBit context
293 * @param selector what table is the output values coded with
294 * @param codingFlag constant length coding or variable length coding
295 * @param mantissas mantissa output table
296 * @param numCodes amount of values to get
297 */
298
299static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
300{
301 int numBits, cnt, code, huffSymb;
302
303 if (selector == 1)
304 numCodes /= 2;
305
306 if (codingFlag != 0) {
307 /* constant length coding (CLC) */
308 //FIXME we don't have any samples coded in CLC mode
309 numBits = CLCLengthTab[selector];
310
311 if (selector > 1) {
312 for (cnt = 0; cnt < numCodes; cnt++) {
313 if (numBits)
314 code = get_sbits(gb, numBits);
315 else
316 code = 0;
317 mantissas[cnt] = code;
318 }
319 } else {
320 for (cnt = 0; cnt < numCodes; cnt++) {
321 if (numBits)
322 code = get_bits(gb, numBits); //numBits is always 4 in this case
323 else
324 code = 0;
325 mantissas[cnt*2] = seTab_0[code >> 2];
326 mantissas[cnt*2+1] = seTab_0[code & 3];
327 }
328 }
329 } else {
330 /* variable length coding (VLC) */
331 if (selector != 1) {
332 for (cnt = 0; cnt < numCodes; cnt++) {
333 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
334 huffSymb += 1;
335 code = huffSymb >> 1;
336 if (huffSymb & 1)
337 code = -code;
338 mantissas[cnt] = code;
339 }
340 } else {
341 for (cnt = 0; cnt < numCodes; cnt++) {
342 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
343 mantissas[cnt*2] = decTable1[huffSymb*2];
344 mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
345 }
346 }
347 }
348}
349
350/**
351 * Restore the quantized band spectrum coefficients
352 *
353 * @param gb the GetBit context
354 * @param pOut decoded band spectrum
355 * @return outSubbands subband counter, fix for broken specification/files
356 */
357
358static int decodeSpectrum (GetBitContext *gb, float *pOut)
359{
360 int numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
361 int subband_vlc_index[32], SF_idxs[32];
362 int mantissas[128];
363 float SF;
364
365 numSubbands = get_bits(gb, 5); // number of coded subbands
366 codingMode = get_bits(gb, 1); // coding Mode: 0 - VLC/ 1-CLC
367
368 /* Get the VLC selector table for the subbands, 0 means not coded. */
369 for (cnt = 0; cnt <= numSubbands; cnt++)
370 subband_vlc_index[cnt] = get_bits(gb, 3);
371
372 /* Read the scale factor indexes from the stream. */
373 for (cnt = 0; cnt <= numSubbands; cnt++) {
374 if (subband_vlc_index[cnt] != 0)
375 SF_idxs[cnt] = get_bits(gb, 6);
376 }
377
378 for (cnt = 0; cnt <= numSubbands; cnt++) {
379 first = subbandTab[cnt];
380 last = subbandTab[cnt+1];
381
382 subbWidth = last - first;
383
384 if (subband_vlc_index[cnt] != 0) {
385 /* Decode spectral coefficients for this subband. */
386 /* TODO: This can be done faster is several blocks share the
387 * same VLC selector (subband_vlc_index) */
388 readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
389
390 /* Decode the scale factor for this subband. */
391 SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
392
393 /* Inverse quantize the coefficients. */
394 for (pIn=mantissas ; first<last; first++, pIn++)
395 pOut[first] = *pIn * SF;
396 } else {
397 /* This subband was not coded, so zero the entire subband. */
398 memset(pOut+first, 0, subbWidth*sizeof(float));
399 }
400 }
401
402 /* Clear the subbands that were not coded. */
403 first = subbandTab[cnt];
404 memset(pOut+first, 0, (1024 - first) * sizeof(float));
405 return numSubbands;
406}
407
408/**
409 * Restore the quantized tonal components
410 *
411 * @param gb the GetBit context
412 * @param numComponents tonal components to report back
413 * @param pComponent tone component
414 * @param numBands amount of coded bands
415 */
416
417static int decodeTonalComponents (GetBitContext *gb, int *numComponents, tonal_component *pComponent, int numBands)
418{
419 int i,j,k,cnt;
420 int component_count, components, coding_mode_selector, coding_mode, coded_values_per_component;
421 int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
422 int band_flags[4], mantissa[8];
423 float *pCoef;
424 float scalefactor;
425
426 component_count = 0;
427 *numComponents = 0;
428
429 components = get_bits(gb,5);
430
431 /* no tonal components */
432 if (components == 0)
433 return 0;
434
435 coding_mode_selector = get_bits(gb,2);
436 if (coding_mode_selector == 2)
437 return -1;
438
439 coding_mode = coding_mode_selector & 1;
440
441 for (i = 0; i < components; i++) {
442 for (cnt = 0; cnt <= numBands; cnt++)
443 band_flags[cnt] = get_bits1(gb);
444
445 coded_values_per_component = get_bits(gb,3);
446
447 quant_step_index = get_bits(gb,3);
448 if (quant_step_index <= 1)
449 return -1;
450
451 if (coding_mode_selector == 3)
452 coding_mode = get_bits1(gb);
453
454 for (j = 0; j < (numBands + 1) * 4; j++) {
455 if (band_flags[j >> 2] == 0)
456 continue;
457
458 coded_components = get_bits(gb,3);
459
460 for (k=0; k<coded_components; k++) {
461 sfIndx = get_bits(gb,6);
462 pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
463 max_coded_values = 1024 - pComponent[component_count].pos;
464 coded_values = coded_values_per_component + 1;
465 coded_values = FFMIN(max_coded_values,coded_values);
466
467 scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index];
468
469 readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
470
471 pComponent[component_count].numCoefs = coded_values;
472
473 /* inverse quant */
474 pCoef = pComponent[k].coef;
475 for (cnt = 0; cnt < coded_values; cnt++)
476 pCoef[cnt] = mantissa[cnt] * scalefactor;
477
478 component_count++;
479 }
480 }
481 }
482
483 *numComponents = component_count;
484
485 return 0;
486}
487
488/**
489 * Decode gain parameters for the coded bands
490 *
491 * @param gb the GetBit context
492 * @param pGb the gainblock for the current band
493 * @param numBands amount of coded bands
494 */
495
496static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
497{
498 int i, cf, numData;
499 int *pLevel, *pLoc;
500
501 gain_info *pGain = pGb->gBlock;
502
503 for (i=0 ; i<=numBands; i++)
504 {
505 numData = get_bits(gb,3);
506 pGain[i].num_gain_data = numData;
507 pLevel = pGain[i].levcode;
508 pLoc = pGain[i].loccode;
509
510 for (cf = 0; cf < numData; cf++){
511 pLevel[cf]= get_bits(gb,4);
512 pLoc [cf]= get_bits(gb,5);
513 if(cf && pLoc[cf] <= pLoc[cf-1])
514 return -1;
515 }
516 }
517
518 /* Clear the unused blocks. */
519 for (; i<4 ; i++)
520 pGain[i].num_gain_data = 0;
521
522 return 0;
523}
524
525/**
526 * Apply gain parameters and perform the MDCT overlapping part
527 *
528 * @param pIn input float buffer
529 * @param pPrev previous float buffer to perform overlap against
530 * @param pOut output float buffer
531 * @param pGain1 current band gain info
532 * @param pGain2 next band gain info
533 */
534
535static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
536{
537 /* gain compensation function */
538 float gain1, gain2, gain_inc;
539 int cnt, numdata, nsample, startLoc, endLoc;
540
541
542 if (pGain2->num_gain_data == 0)
543 gain1 = 1.0;
544 else
545 gain1 = gain_tab1[pGain2->levcode[0]];
546
547 if (pGain1->num_gain_data == 0) {
548 for (cnt = 0; cnt < 256; cnt++)
549 pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
550 } else {
551 numdata = pGain1->num_gain_data;
552 pGain1->loccode[numdata] = 32;
553 pGain1->levcode[numdata] = 4;
554
555 nsample = 0; // current sample = 0
556
557 for (cnt = 0; cnt < numdata; cnt++) {
558 startLoc = pGain1->loccode[cnt] * 8;
559 endLoc = startLoc + 8;
560
561 gain2 = gain_tab1[pGain1->levcode[cnt]];
562 gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
563
564 /* interpolate */
565 for (; nsample < startLoc; nsample++)
566 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
567
568 /* interpolation is done over eight samples */
569 for (; nsample < endLoc; nsample++) {
570 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
571 gain2 *= gain_inc;
572 }
573 }
574
575 for (; nsample < 256; nsample++)
576 pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
577 }
578
579 /* Delay for the overlapping part. */
580 memcpy(pPrev, &pIn[256], 256*sizeof(float));
581}
582
583/**
584 * Combine the tonal band spectrum and regular band spectrum
585 *
586 * @param pSpectrum output spectrum buffer
587 * @param numComponents amount of tonal components
588 * @param pComponent tonal components for this band
589 */
590
591static void addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
592{
593 int cnt, i;
594 float *pIn, *pOut;
595
596 for (cnt = 0; cnt < numComponents; cnt++){
597 pIn = pComponent[cnt].coef;
598 pOut = &(pSpectrum[pComponent[cnt].pos]);
599
600 for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
601 pOut[i] += pIn[i];
602 }
603}
604
605
606#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
607
608static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
609{
610 int i, band, nsample, s1, s2;
611 float c1, c2;
612 float mc1_l, mc1_r, mc2_l, mc2_r;
613
614 for (i=0,band = 0; band < 4*256; band+=256,i++) {
615 s1 = pPrevCode[i];
616 s2 = pCurrCode[i];
617 nsample = 0;
618
619 if (s1 != s2) {
620 /* Selector value changed, interpolation needed. */
621 mc1_l = matrixCoeffs[s1*2];
622 mc1_r = matrixCoeffs[s1*2+1];
623 mc2_l = matrixCoeffs[s2*2];
624 mc2_r = matrixCoeffs[s2*2+1];
625
626 /* Interpolation is done over the first eight samples. */
627 for(; nsample < 8; nsample++) {
628 c1 = su1[band+nsample];
629 c2 = su2[band+nsample];
630 c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
631 su1[band+nsample] = c2;
632 su2[band+nsample] = c1 * 2.0 - c2;
633 }
634 }
635
636 /* Apply the matrix without interpolation. */
637 switch (s2) {
638 case 0: /* M/S decoding */
639 for (; nsample < 256; nsample++) {
640 c1 = su1[band+nsample];
641 c2 = su2[band+nsample];
642 su1[band+nsample] = c2 * 2.0;
643 su2[band+nsample] = (c1 - c2) * 2.0;
644 }
645 break;
646
647 case 1:
648 for (; nsample < 256; nsample++) {
649 c1 = su1[band+nsample];
650 c2 = su2[band+nsample];
651 su1[band+nsample] = (c1 + c2) * 2.0;
652 su2[band+nsample] = c2 * -2.0;
653 }
654 break;
655 case 2:
656 case 3:
657 for (; nsample < 256; nsample++) {
658 c1 = su1[band+nsample];
659 c2 = su2[band+nsample];
660 su1[band+nsample] = c1 + c2;
661 su2[band+nsample] = c1 - c2;
662 }
663 break;
664 default:
665 assert(0);
666 }
667 }
668}
669
670static void getChannelWeights (int indx, int flag, float ch[2]){
671
672 if (indx == 7) {
673 ch[0] = 1.0;
674 ch[1] = 1.0;
675 } else {
676 ch[0] = (float)(indx & 7) / 7.0;
677 ch[1] = sqrt(2 - ch[0]*ch[0]);
678 if(flag)
679 FFSWAP(float, ch[0], ch[1]);
680 }
681}
682
683static void channelWeighting (float *su1, float *su2, int *p3)
684{
685 int band, nsample;
686 /* w[x][y] y=0 is left y=1 is right */
687 float w[2][2];
688
689 if (p3[1] != 7 || p3[3] != 7){
690 getChannelWeights(p3[1], p3[0], w[0]);
691 getChannelWeights(p3[3], p3[2], w[1]);
692
693 for(band = 1; band < 4; band++) {
694 /* scale the channels by the weights */
695 for(nsample = 0; nsample < 8; nsample++) {
696 su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
697 su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
698 }
699
700 for(; nsample < 256; nsample++) {
701 su1[band*256+nsample] *= w[1][0];
702 su2[band*256+nsample] *= w[1][1];
703 }
704 }
705 }
706}
707
708
709/**
710 * Decode a Sound Unit
711 *
712 * @param gb the GetBit context
713 * @param pSnd the channel unit to be used
714 * @param pOut the decoded samples before IQMF in float representation
715 * @param channelNum channel number
716 * @param codingMode the coding mode (JOINT_STEREO or regular stereo/mono)
717 */
718
719
720static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
721{
722 int band, result=0, numSubbands, numBands;
723
724 if (codingMode == JOINT_STEREO && channelNum == 1) {
725 if (get_bits(gb,2) != 3) {
726 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
727 return -1;
728 }
729 } else {
730 if (get_bits(gb,6) != 0x28) {
731 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
732 return -1;
733 }
734 }
735
736 /* number of coded QMF bands */
737 pSnd->bandsCoded = get_bits(gb,2);
738
739 result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
740 if (result) return result;
741
742 result = decodeTonalComponents (gb, &pSnd->numComponents, pSnd->components, pSnd->bandsCoded);
743 if (result) return result;
744
745 numSubbands = decodeSpectrum (gb, pSnd->spectrum);
746
747 /* Merge the decoded spectrum and tonal components. */
748 addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
749
750
751 /* Convert number of subbands into number of MLT/QMF bands */
752 numBands = (subbandTab[numSubbands] - 1) >> 8;
753
754
755 /* Reconstruct time domain samples. */
756 for (band=0; band<4; band++) {
757 /* Perform the IMDCT step without overlapping. */
758 if (band <= numBands) {
759 IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1,q->mdct_tmp);
760 } else
761 memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
762
763 /* gain compensation and overlapping */
764 gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
765 &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
766 &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
767 }
768
769 /* Swap the gain control buffers for the next frame. */
770 pSnd->gcBlkSwitch ^= 1;
771
772 return 0;
773}
774
775/**
776 * Frame handling
777 *
778 * @param q Atrac3 private context
779 * @param databuf the input data
780 */
781
782static int decodeFrame(ATRAC3Context *q, uint8_t* databuf)
783{
784 int result, i;
785 float *p1, *p2, *p3, *p4;
786 uint8_t *ptr1, *ptr2;
787
788 if (q->codingMode == JOINT_STEREO) {
789
790 /* channel coupling mode */
791 /* decode Sound Unit 1 */
792 init_get_bits(&q->gb,databuf,q->bits_per_frame);
793
794 result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);
795 if (result != 0)
796 return (result);
797
798 /* Framedata of the su2 in the joint-stereo mode is encoded in
799 * reverse byte order so we need to swap it first. */
800 ptr1 = databuf;
801 ptr2 = databuf+q->bytes_per_frame-1;
802 for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
803 FFSWAP(uint8_t,*ptr1,*ptr2);
804 }
805
806 /* Skip the sync codes (0xF8). */
807 ptr1 = databuf;
808 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
809 if (i >= q->bytes_per_frame)
810 return -1;
811 }
812
813
814 /* set the bitstream reader at the start of the second Sound Unit*/
815 init_get_bits(&q->gb,ptr1,q->bits_per_frame);
816
817 /* Fill the Weighting coeffs delay buffer */
818 memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
819 q->weighting_delay[4] = get_bits(&q->gb,1);
820 q->weighting_delay[5] = get_bits(&q->gb,3);
821
822 for (i = 0; i < 4; i++) {
823 q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
824 q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
825 q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
826 }
827
828 /* Decode Sound Unit 2. */
829 result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
830 if (result != 0)
831 return (result);
832
833 /* Reconstruct the channel coefficients. */
834 reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
835
836 channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);
837
838 } else {
839 /* normal stereo mode or mono */
840 /* Decode the channel sound units. */
841 for (i=0 ; i<q->channels ; i++) {
842
843 /* Set the bitstream reader at the start of a channel sound unit. */
844 init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
845
846 result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);
847 if (result != 0)
848 return (result);
849 }
850 }
851
852 /* Apply the iQMF synthesis filter. */
853 p1= q->outSamples;
854 for (i=0 ; i<q->channels ; i++) {
855 p2= p1+256;
856 p3= p2+256;
857 p4= p3+256;
858 iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
859 iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
860 iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
861 p1 +=1024;
862 }
863
864 return 0;
865}
866
867
868/**
869 * Atrac frame decoding
870 *
871 * @param avctx pointer to the AVCodecContext
872 */
873
874static int atrac3_decode_frame(AVCodecContext *avctx,
875 void *data, int *data_size,
876 uint8_t *buf, int buf_size) {
877 ATRAC3Context *q = avctx->priv_data;
878 int result = 0, i;
879 uint8_t* databuf;
880 int16_t* samples = data;
881
882 if (buf_size < avctx->block_align)
883 return buf_size;
884
885 /* Check if we need to descramble and what buffer to pass on. */
886 if (q->scrambled_stream) {
887 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
888 databuf = q->decoded_bytes_buffer;
889 } else {
890 databuf = buf;
891 }
892
893 result = decodeFrame(q, databuf);
894
895 if (result != 0) {
896 av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
897 return -1;
898 }
899
900 if (q->channels == 1) {
901 /* mono */
902 for (i = 0; i<1024; i++)
903 samples[i] = av_clip(round(q->outSamples[i]), -32768, 32767);
904 *data_size = 1024 * sizeof(int16_t);
905 } else {
906 /* stereo */
907 for (i = 0; i < 1024; i++) {
908 samples[i*2] = av_clip(round(q->outSamples[i]), -32768, 32767);
909 samples[i*2+1] = av_clip(round(q->outSamples[1024+i]), -32768, 32767);
910 }
911 *data_size = 2048 * sizeof(int16_t);
912 }
913
914 return avctx->block_align;
915}
916
917
918/**
919 * Atrac3 initialization
920 *
921 * @param avctx pointer to the AVCodecContext
922 */
923
924static int atrac3_decode_init(AVCodecContext *avctx)
925{
926 int i;
927 uint8_t *edata_ptr = avctx->extradata;
928 ATRAC3Context *q = avctx->priv_data;
929
930 /* Take data from the AVCodecContext (RM container). */
931 q->sample_rate = avctx->sample_rate;
932 q->channels = avctx->channels;
933 q->bit_rate = avctx->bit_rate;
934 q->bits_per_frame = avctx->block_align * 8;
935 q->bytes_per_frame = avctx->block_align;
936
937 /* Take care of the codec-specific extradata. */
938 if (avctx->extradata_size == 14) {
939 /* Parse the extradata, WAV format */
940 av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1
941 q->samples_per_channel = bytestream_get_le32(&edata_ptr);
942 q->codingMode = bytestream_get_le16(&edata_ptr);
943 av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
944 q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1
945 av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0
946
947 /* setup */
948 q->samples_per_frame = 1024 * q->channels;
949 q->atrac3version = 4;
950 q->delay = 0x88E;
951 if (q->codingMode)
952 q->codingMode = JOINT_STEREO;
953 else
954 q->codingMode = STEREO;
955
956 q->scrambled_stream = 0;
957
958 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)) {
959 } else {
960 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);
961 return -1;
962 }
963
964 } else if (avctx->extradata_size == 10) {
965 /* Parse the extradata, RM format. */
966 q->atrac3version = bytestream_get_be32(&edata_ptr);
967 q->samples_per_frame = bytestream_get_be16(&edata_ptr);
968 q->delay = bytestream_get_be16(&edata_ptr);
969 q->codingMode = bytestream_get_be16(&edata_ptr);
970
971 q->samples_per_channel = q->samples_per_frame / q->channels;
972 q->scrambled_stream = 1;
973
974 } else {
975 av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
976 }
977 /* Check the extradata. */
978
979 if (q->atrac3version != 4) {
980 av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
981 return -1;
982 }
983
984 if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
985 av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
986 return -1;
987 }
988
989 if (q->delay != 0x88E) {
990 av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
991 return -1;
992 }
993
994 if (q->codingMode == STEREO) {
995 av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
996 } else if (q->codingMode == JOINT_STEREO) {
997 av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
998 } else {
999 av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
1000 return -1;
1001 }
1002
1003 if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
1004 av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
1005 return -1;
1006 }
1007
1008
1009 if(avctx->block_align >= UINT_MAX/2)
1010 return -1;
1011
1012 /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
1013 * this is for the bitstream reader. */
1014 if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE))) == NULL)
1015 return -1;
1016
1017
1018 /* Initialize the VLC tables. */
1019 for (i=0 ; i<7 ; i++) {
1020 init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
1021 huff_bits[i], 1, 1,
1022 huff_codes[i], 1, 1, INIT_VLC_USE_STATIC);
1023 }
1024
1025 init_atrac3_transforms(q);
1026
1027 /* Generate the scale factors. */
1028 for (i=0 ; i<64 ; i++)
1029 SFTable[i] = pow(2.0, (i - 15) / 3.0);
1030
1031 /* Generate gain tables. */
1032 for (i=0 ; i<16 ; i++)
1033 gain_tab1[i] = powf (2.0, (4 - i));
1034
1035 for (i=-15 ; i<16 ; i++)
1036 gain_tab2[i+15] = powf (2.0, i * -0.125);
1037
1038 /* init the joint-stereo decoding data */
1039 q->weighting_delay[0] = 0;
1040 q->weighting_delay[1] = 7;
1041 q->weighting_delay[2] = 0;
1042 q->weighting_delay[3] = 7;
1043 q->weighting_delay[4] = 0;
1044 q->weighting_delay[5] = 7;
1045
1046 for (i=0; i<4; i++) {
1047 q->matrix_coeff_index_prev[i] = 3;
1048 q->matrix_coeff_index_now[i] = 3;
1049 q->matrix_coeff_index_next[i] = 3;
1050 }
1051
1052 dsputil_init(&dsp, avctx);
1053
1054 q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
1055
1056 return 0;
1057}
1058
1059
1060AVCodec atrac3_decoder =
1061{
1062 .name = "atrac 3",
1063 .type = CODEC_TYPE_AUDIO,
1064 .id = CODEC_ID_ATRAC3,
1065 .priv_data_size = sizeof(ATRAC3Context),
1066 .init = atrac3_decode_init,
1067 .close = atrac3_decode_close,
1068 .decode = atrac3_decode_frame,
1069};