vlc: Add header #include when the types are used
[libav.git] / libavcodec / atrac3.c
1 /*
2 * ATRAC3 compatible decoder
3 * Copyright (c) 2006-2008 Maxim Poliakovski
4 * Copyright (c) 2006-2008 Benjamin Larsson
5 *
6 * This file is part of Libav.
7 *
8 * Libav 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 * Libav 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 Libav; 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
25 * ATRAC3 compatible decoder.
26 * This decoder handles Sony's ATRAC3 data.
27 *
28 * Container formats used to store ATRAC3 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 "libavutil/attributes.h"
40 #include "libavutil/float_dsp.h"
41
42 #include "avcodec.h"
43 #include "bitstream.h"
44 #include "bytestream.h"
45 #include "fft.h"
46 #include "internal.h"
47 #include "vlc.h"
48
49 #include "atrac.h"
50 #include "atrac3data.h"
51
52 #define JOINT_STEREO 0x12
53 #define STEREO 0x2
54
55 #define SAMPLES_PER_FRAME 1024
56 #define MDCT_SIZE 512
57
58 typedef struct GainBlock {
59 AtracGainInfo g_block[4];
60 } GainBlock;
61
62 typedef struct TonalComponent {
63 int pos;
64 int num_coefs;
65 float coef[8];
66 } TonalComponent;
67
68 typedef struct ChannelUnit {
69 int bands_coded;
70 int num_components;
71 float prev_frame[SAMPLES_PER_FRAME];
72 int gc_blk_switch;
73 TonalComponent components[64];
74 GainBlock gain_block[2];
75
76 DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
77 DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
78
79 float delay_buf1[46]; ///<qmf delay buffers
80 float delay_buf2[46];
81 float delay_buf3[46];
82 } ChannelUnit;
83
84 typedef struct ATRAC3Context {
85 BitstreamContext bc;
86 //@{
87 /** stream data */
88 int coding_mode;
89
90 ChannelUnit *units;
91 //@}
92 //@{
93 /** joint-stereo related variables */
94 int matrix_coeff_index_prev[4];
95 int matrix_coeff_index_now[4];
96 int matrix_coeff_index_next[4];
97 int weighting_delay[6];
98 //@}
99 //@{
100 /** data buffers */
101 uint8_t *decoded_bytes_buffer;
102 float temp_buf[1070];
103 //@}
104 //@{
105 /** extradata */
106 int scrambled_stream;
107 //@}
108
109 AtracGCContext gainc_ctx;
110 FFTContext mdct_ctx;
111 AVFloatDSPContext fdsp;
112 } ATRAC3Context;
113
114 static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
115 static VLC_TYPE atrac3_vlc_table[4096][2];
116 static VLC spectral_coeff_tab[7];
117
118 /**
119 * Regular 512 points IMDCT without overlapping, with the exception of the
120 * swapping of odd bands caused by the reverse spectra of the QMF.
121 *
122 * @param odd_band 1 if the band is an odd band
123 */
124 static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
125 {
126 int i;
127
128 if (odd_band) {
129 /**
130 * Reverse the odd bands before IMDCT, this is an effect of the QMF
131 * transform or it gives better compression to do it this way.
132 * FIXME: It should be possible to handle this in imdct_calc
133 * for that to happen a modification of the prerotation step of
134 * all SIMD code and C code is needed.
135 * Or fix the functions before so they generate a pre reversed spectrum.
136 */
137 for (i = 0; i < 128; i++)
138 FFSWAP(float, input[i], input[255 - i]);
139 }
140
141 q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
142
143 /* Perform windowing on the output. */
144 q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE);
145 }
146
147 /*
148 * indata descrambling, only used for data coming from the rm container
149 */
150 static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
151 {
152 int i, off;
153 uint32_t c;
154 const uint32_t *buf;
155 uint32_t *output = (uint32_t *)out;
156
157 off = (intptr_t)input & 3;
158 buf = (const uint32_t *)(input - off);
159 if (off)
160 c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
161 else
162 c = av_be2ne32(0x537F6103U);
163 bytes += 3 + off;
164 for (i = 0; i < bytes / 4; i++)
165 output[i] = c ^ buf[i];
166
167 if (off)
168 avpriv_request_sample(NULL, "Offset of %d", off);
169
170 return off;
171 }
172
173 static av_cold void init_imdct_window(void)
174 {
175 int i, j;
176
177 /* generate the mdct window, for details see
178 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
179 for (i = 0, j = 255; i < 128; i++, j--) {
180 float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
181 float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
182 float w = 0.5 * (wi * wi + wj * wj);
183 mdct_window[i] = mdct_window[511 - i] = wi / w;
184 mdct_window[j] = mdct_window[511 - j] = wj / w;
185 }
186 }
187
188 static av_cold int atrac3_decode_close(AVCodecContext *avctx)
189 {
190 ATRAC3Context *q = avctx->priv_data;
191
192 av_free(q->units);
193 av_free(q->decoded_bytes_buffer);
194
195 ff_mdct_end(&q->mdct_ctx);
196
197 return 0;
198 }
199
200 /**
201 * Mantissa decoding
202 *
203 * @param selector which table the output values are coded with
204 * @param coding_flag constant length coding or variable length coding
205 * @param mantissas mantissa output table
206 * @param num_codes number of values to get
207 */
208 static void read_quant_spectral_coeffs(BitstreamContext *bc, int selector,
209 int coding_flag, int *mantissas,
210 int num_codes)
211 {
212 int i, code, huff_symb;
213
214 if (selector == 1)
215 num_codes /= 2;
216
217 if (coding_flag != 0) {
218 /* constant length coding (CLC) */
219 int num_bits = clc_length_tab[selector];
220
221 if (selector > 1) {
222 for (i = 0; i < num_codes; i++) {
223 if (num_bits)
224 code = bitstream_read_signed(bc, num_bits);
225 else
226 code = 0;
227 mantissas[i] = code;
228 }
229 } else {
230 for (i = 0; i < num_codes; i++) {
231 if (num_bits)
232 code = bitstream_read(bc, num_bits); // num_bits is always 4 in this case
233 else
234 code = 0;
235 mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
236 mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
237 }
238 }
239 } else {
240 /* variable length coding (VLC) */
241 if (selector != 1) {
242 for (i = 0; i < num_codes; i++) {
243 huff_symb = bitstream_read_vlc(bc, spectral_coeff_tab[selector-1].table,
244 spectral_coeff_tab[selector-1].bits, 3);
245 huff_symb += 1;
246 code = huff_symb >> 1;
247 if (huff_symb & 1)
248 code = -code;
249 mantissas[i] = code;
250 }
251 } else {
252 for (i = 0; i < num_codes; i++) {
253 huff_symb = bitstream_read_vlc(bc, spectral_coeff_tab[selector - 1].table,
254 spectral_coeff_tab[selector - 1].bits, 3);
255 mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
256 mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
257 }
258 }
259 }
260 }
261
262 /**
263 * Restore the quantized band spectrum coefficients
264 *
265 * @return subband count, fix for broken specification/files
266 */
267 static int decode_spectrum(BitstreamContext *bc, float *output)
268 {
269 int num_subbands, coding_mode, i, j, first, last, subband_size;
270 int subband_vlc_index[32], sf_index[32];
271 int mantissas[128];
272 float scale_factor;
273
274 num_subbands = bitstream_read(bc, 5); // number of coded subbands
275 coding_mode = bitstream_read_bit(bc); // coding Mode: 0 - VLC/ 1 - CLC
276
277 /* get the VLC selector table for the subbands, 0 means not coded */
278 for (i = 0; i <= num_subbands; i++)
279 subband_vlc_index[i] = bitstream_read(bc, 3);
280
281 /* read the scale factor indexes from the stream */
282 for (i = 0; i <= num_subbands; i++) {
283 if (subband_vlc_index[i] != 0)
284 sf_index[i] = bitstream_read(bc, 6);
285 }
286
287 for (i = 0; i <= num_subbands; i++) {
288 first = subband_tab[i ];
289 last = subband_tab[i + 1];
290
291 subband_size = last - first;
292
293 if (subband_vlc_index[i] != 0) {
294 /* decode spectral coefficients for this subband */
295 /* TODO: This can be done faster is several blocks share the
296 * same VLC selector (subband_vlc_index) */
297 read_quant_spectral_coeffs(bc, subband_vlc_index[i], coding_mode,
298 mantissas, subband_size);
299
300 /* decode the scale factor for this subband */
301 scale_factor = ff_atrac_sf_table[sf_index[i]] *
302 inv_max_quant[subband_vlc_index[i]];
303
304 /* inverse quantize the coefficients */
305 for (j = 0; first < last; first++, j++)
306 output[first] = mantissas[j] * scale_factor;
307 } else {
308 /* this subband was not coded, so zero the entire subband */
309 memset(output + first, 0, subband_size * sizeof(*output));
310 }
311 }
312
313 /* clear the subbands that were not coded */
314 first = subband_tab[i];
315 memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
316 return num_subbands;
317 }
318
319 /**
320 * Restore the quantized tonal components
321 *
322 * @param components tonal components
323 * @param num_bands number of coded bands
324 */
325 static int decode_tonal_components(BitstreamContext *bc,
326 TonalComponent *components, int num_bands)
327 {
328 int i, b, c, m;
329 int nb_components, coding_mode_selector, coding_mode;
330 int band_flags[4], mantissa[8];
331 int component_count = 0;
332
333 nb_components = bitstream_read(bc, 5);
334
335 /* no tonal components */
336 if (nb_components == 0)
337 return 0;
338
339 coding_mode_selector = bitstream_read(bc, 2);
340 if (coding_mode_selector == 2)
341 return AVERROR_INVALIDDATA;
342
343 coding_mode = coding_mode_selector & 1;
344
345 for (i = 0; i < nb_components; i++) {
346 int coded_values_per_component, quant_step_index;
347
348 for (b = 0; b <= num_bands; b++)
349 band_flags[b] = bitstream_read_bit(bc);
350
351 coded_values_per_component = bitstream_read(bc, 3);
352
353 quant_step_index = bitstream_read(bc, 3);
354 if (quant_step_index <= 1)
355 return AVERROR_INVALIDDATA;
356
357 if (coding_mode_selector == 3)
358 coding_mode = bitstream_read_bit(bc);
359
360 for (b = 0; b < (num_bands + 1) * 4; b++) {
361 int coded_components;
362
363 if (band_flags[b >> 2] == 0)
364 continue;
365
366 coded_components = bitstream_read(bc, 3);
367
368 for (c = 0; c < coded_components; c++) {
369 TonalComponent *cmp = &components[component_count];
370 int sf_index, coded_values, max_coded_values;
371 float scale_factor;
372
373 sf_index = bitstream_read(bc, 6);
374 if (component_count >= 64)
375 return AVERROR_INVALIDDATA;
376
377 cmp->pos = b * 64 + bitstream_read(bc, 6);
378
379 max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
380 coded_values = coded_values_per_component + 1;
381 coded_values = FFMIN(max_coded_values, coded_values);
382
383 scale_factor = ff_atrac_sf_table[sf_index] *
384 inv_max_quant[quant_step_index];
385
386 read_quant_spectral_coeffs(bc, quant_step_index, coding_mode,
387 mantissa, coded_values);
388
389 cmp->num_coefs = coded_values;
390
391 /* inverse quant */
392 for (m = 0; m < coded_values; m++)
393 cmp->coef[m] = mantissa[m] * scale_factor;
394
395 component_count++;
396 }
397 }
398 }
399
400 return component_count;
401 }
402
403 /**
404 * Decode gain parameters for the coded bands
405 *
406 * @param block the gainblock for the current band
407 * @param num_bands amount of coded bands
408 */
409 static int decode_gain_control(BitstreamContext *bc, GainBlock *block,
410 int num_bands)
411 {
412 int i, j;
413 int *level, *loc;
414
415 AtracGainInfo *gain = block->g_block;
416
417 for (i = 0; i <= num_bands; i++) {
418 gain[i].num_points = bitstream_read(bc, 3);
419 level = gain[i].lev_code;
420 loc = gain[i].loc_code;
421
422 for (j = 0; j < gain[i].num_points; j++) {
423 level[j] = bitstream_read(bc, 4);
424 loc[j] = bitstream_read(bc, 5);
425 if (j && loc[j] <= loc[j - 1])
426 return AVERROR_INVALIDDATA;
427 }
428 }
429
430 /* Clear the unused blocks. */
431 for (; i < 4 ; i++)
432 gain[i].num_points = 0;
433
434 return 0;
435 }
436
437 /**
438 * Combine the tonal band spectrum and regular band spectrum
439 *
440 * @param spectrum output spectrum buffer
441 * @param num_components number of tonal components
442 * @param components tonal components for this band
443 * @return position of the last tonal coefficient
444 */
445 static int add_tonal_components(float *spectrum, int num_components,
446 TonalComponent *components)
447 {
448 int i, j, last_pos = -1;
449 float *input, *output;
450
451 for (i = 0; i < num_components; i++) {
452 last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
453 input = components[i].coef;
454 output = &spectrum[components[i].pos];
455
456 for (j = 0; j < components[i].num_coefs; j++)
457 output[j] += input[j];
458 }
459
460 return last_pos;
461 }
462
463 #define INTERPOLATE(old, new, nsample) \
464 ((old) + (nsample) * 0.125 * ((new) - (old)))
465
466 static void reverse_matrixing(float *su1, float *su2, int *prev_code,
467 int *curr_code)
468 {
469 int i, nsample, band;
470 float mc1_l, mc1_r, mc2_l, mc2_r;
471
472 for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
473 int s1 = prev_code[i];
474 int s2 = curr_code[i];
475 nsample = band;
476
477 if (s1 != s2) {
478 /* Selector value changed, interpolation needed. */
479 mc1_l = matrix_coeffs[s1 * 2 ];
480 mc1_r = matrix_coeffs[s1 * 2 + 1];
481 mc2_l = matrix_coeffs[s2 * 2 ];
482 mc2_r = matrix_coeffs[s2 * 2 + 1];
483
484 /* Interpolation is done over the first eight samples. */
485 for (; nsample < band + 8; nsample++) {
486 float c1 = su1[nsample];
487 float c2 = su2[nsample];
488 c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
489 c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
490 su1[nsample] = c2;
491 su2[nsample] = c1 * 2.0 - c2;
492 }
493 }
494
495 /* Apply the matrix without interpolation. */
496 switch (s2) {
497 case 0: /* M/S decoding */
498 for (; nsample < band + 256; nsample++) {
499 float c1 = su1[nsample];
500 float c2 = su2[nsample];
501 su1[nsample] = c2 * 2.0;
502 su2[nsample] = (c1 - c2) * 2.0;
503 }
504 break;
505 case 1:
506 for (; nsample < band + 256; nsample++) {
507 float c1 = su1[nsample];
508 float c2 = su2[nsample];
509 su1[nsample] = (c1 + c2) * 2.0;
510 su2[nsample] = c2 * -2.0;
511 }
512 break;
513 case 2:
514 case 3:
515 for (; nsample < band + 256; nsample++) {
516 float c1 = su1[nsample];
517 float c2 = su2[nsample];
518 su1[nsample] = c1 + c2;
519 su2[nsample] = c1 - c2;
520 }
521 break;
522 default:
523 assert(0);
524 }
525 }
526 }
527
528 static void get_channel_weights(int index, int flag, float ch[2])
529 {
530 if (index == 7) {
531 ch[0] = 1.0;
532 ch[1] = 1.0;
533 } else {
534 ch[0] = (index & 7) / 7.0;
535 ch[1] = sqrt(2 - ch[0] * ch[0]);
536 if (flag)
537 FFSWAP(float, ch[0], ch[1]);
538 }
539 }
540
541 static void channel_weighting(float *su1, float *su2, int *p3)
542 {
543 int band, nsample;
544 /* w[x][y] y=0 is left y=1 is right */
545 float w[2][2];
546
547 if (p3[1] != 7 || p3[3] != 7) {
548 get_channel_weights(p3[1], p3[0], w[0]);
549 get_channel_weights(p3[3], p3[2], w[1]);
550
551 for (band = 256; band < 4 * 256; band += 256) {
552 for (nsample = band; nsample < band + 8; nsample++) {
553 su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
554 su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
555 }
556 for(; nsample < band + 256; nsample++) {
557 su1[nsample] *= w[1][0];
558 su2[nsample] *= w[1][1];
559 }
560 }
561 }
562 }
563
564 /**
565 * Decode a Sound Unit
566 *
567 * @param snd the channel unit to be used
568 * @param output the decoded samples before IQMF in float representation
569 * @param channel_num channel number
570 * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)
571 */
572 static int decode_channel_sound_unit(ATRAC3Context *q, BitstreamContext *bc,
573 ChannelUnit *snd, float *output,
574 int channel_num, int coding_mode)
575 {
576 int band, ret, num_subbands, last_tonal, num_bands;
577 GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
578 GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
579
580 if (coding_mode == JOINT_STEREO && channel_num == 1) {
581 if (bitstream_read(bc, 2) != 3) {
582 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
583 return AVERROR_INVALIDDATA;
584 }
585 } else {
586 if (bitstream_read(bc, 6) != 0x28) {
587 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
588 return AVERROR_INVALIDDATA;
589 }
590 }
591
592 /* number of coded QMF bands */
593 snd->bands_coded = bitstream_read(bc, 2);
594
595 ret = decode_gain_control(bc, gain2, snd->bands_coded);
596 if (ret)
597 return ret;
598
599 snd->num_components = decode_tonal_components(bc, snd->components,
600 snd->bands_coded);
601 if (snd->num_components < 0)
602 return snd->num_components;
603
604 num_subbands = decode_spectrum(bc, snd->spectrum);
605
606 /* Merge the decoded spectrum and tonal components. */
607 last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
608 snd->components);
609
610
611 /* calculate number of used MLT/QMF bands according to the amount of coded
612 spectral lines */
613 num_bands = (subband_tab[num_subbands] - 1) >> 8;
614 if (last_tonal >= 0)
615 num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
616
617
618 /* Reconstruct time domain samples. */
619 for (band = 0; band < 4; band++) {
620 /* Perform the IMDCT step without overlapping. */
621 if (band <= num_bands)
622 imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
623 else
624 memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
625
626 /* gain compensation and overlapping */
627 ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
628 &snd->prev_frame[band * 256],
629 &gain1->g_block[band], &gain2->g_block[band],
630 256, &output[band * 256]);
631 }
632
633 /* Swap the gain control buffers for the next frame. */
634 snd->gc_blk_switch ^= 1;
635
636 return 0;
637 }
638
639 static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
640 float **out_samples)
641 {
642 ATRAC3Context *q = avctx->priv_data;
643 int ret, i;
644 uint8_t *ptr1;
645
646 if (q->coding_mode == JOINT_STEREO) {
647 /* channel coupling mode */
648 /* decode Sound Unit 1 */
649 bitstream_init8(&q->bc, databuf, avctx->block_align);
650
651 ret = decode_channel_sound_unit(q, &q->bc, q->units, out_samples[0], 0,
652 JOINT_STEREO);
653 if (ret != 0)
654 return ret;
655
656 /* Framedata of the su2 in the joint-stereo mode is encoded in
657 * reverse byte order so we need to swap it first. */
658 if (databuf == q->decoded_bytes_buffer) {
659 uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
660 ptr1 = q->decoded_bytes_buffer;
661 for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
662 FFSWAP(uint8_t, *ptr1, *ptr2);
663 } else {
664 const uint8_t *ptr2 = databuf + avctx->block_align - 1;
665 for (i = 0; i < avctx->block_align; i++)
666 q->decoded_bytes_buffer[i] = *ptr2--;
667 }
668
669 /* Skip the sync codes (0xF8). */
670 ptr1 = q->decoded_bytes_buffer;
671 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
672 if (i >= avctx->block_align)
673 return AVERROR_INVALIDDATA;
674 }
675
676
677 /* set the bitstream reader at the start of the second Sound Unit*/
678 bitstream_init8(&q->bc, ptr1, avctx->block_align - i);
679
680 /* Fill the Weighting coeffs delay buffer */
681 memmove(q->weighting_delay, &q->weighting_delay[2],
682 4 * sizeof(*q->weighting_delay));
683 q->weighting_delay[4] = bitstream_read_bit(&q->bc);
684 q->weighting_delay[5] = bitstream_read(&q->bc, 3);
685
686 for (i = 0; i < 4; i++) {
687 q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
688 q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
689 q->matrix_coeff_index_next[i] = bitstream_read(&q->bc, 2);
690 }
691
692 /* Decode Sound Unit 2. */
693 ret = decode_channel_sound_unit(q, &q->bc, &q->units[1],
694 out_samples[1], 1, JOINT_STEREO);
695 if (ret != 0)
696 return ret;
697
698 /* Reconstruct the channel coefficients. */
699 reverse_matrixing(out_samples[0], out_samples[1],
700 q->matrix_coeff_index_prev,
701 q->matrix_coeff_index_now);
702
703 channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
704 } else {
705 /* normal stereo mode or mono */
706 /* Decode the channel sound units. */
707 for (i = 0; i < avctx->channels; i++) {
708 /* Set the bitstream reader at the start of a channel sound unit. */
709 bitstream_init8(&q->bc,
710 databuf + i * avctx->block_align / avctx->channels,
711 avctx->block_align / avctx->channels);
712
713 ret = decode_channel_sound_unit(q, &q->bc, &q->units[i],
714 out_samples[i], i, q->coding_mode);
715 if (ret != 0)
716 return ret;
717 }
718 }
719
720 /* Apply the iQMF synthesis filter. */
721 for (i = 0; i < avctx->channels; i++) {
722 float *p1 = out_samples[i];
723 float *p2 = p1 + 256;
724 float *p3 = p2 + 256;
725 float *p4 = p3 + 256;
726 ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
727 ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
728 ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
729 }
730
731 return 0;
732 }
733
734 static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
735 int *got_frame_ptr, AVPacket *avpkt)
736 {
737 AVFrame *frame = data;
738 const uint8_t *buf = avpkt->data;
739 int buf_size = avpkt->size;
740 ATRAC3Context *q = avctx->priv_data;
741 int ret;
742 const uint8_t *databuf;
743
744 if (buf_size < avctx->block_align) {
745 av_log(avctx, AV_LOG_ERROR,
746 "Frame too small (%d bytes). Truncated file?\n", buf_size);
747 return AVERROR_INVALIDDATA;
748 }
749
750 /* get output buffer */
751 frame->nb_samples = SAMPLES_PER_FRAME;
752 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
753 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
754 return ret;
755 }
756
757 /* Check if we need to descramble and what buffer to pass on. */
758 if (q->scrambled_stream) {
759 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
760 databuf = q->decoded_bytes_buffer;
761 } else {
762 databuf = buf;
763 }
764
765 ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
766 if (ret) {
767 av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
768 return ret;
769 }
770
771 *got_frame_ptr = 1;
772
773 return avctx->block_align;
774 }
775
776 static av_cold void atrac3_init_static_data(AVCodec *codec)
777 {
778 int i;
779
780 init_imdct_window();
781 ff_atrac_generate_tables();
782
783 /* Initialize the VLC tables. */
784 for (i = 0; i < 7; i++) {
785 spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
786 spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
787 atrac3_vlc_offs[i ];
788 init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
789 huff_bits[i], 1, 1,
790 huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
791 }
792 }
793
794 static av_cold int atrac3_decode_init(AVCodecContext *avctx)
795 {
796 int i, ret;
797 int version, delay, samples_per_frame, frame_factor;
798 const uint8_t *edata_ptr = avctx->extradata;
799 ATRAC3Context *q = avctx->priv_data;
800
801 if (avctx->channels <= 0 || avctx->channels > 2) {
802 av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
803 return AVERROR(EINVAL);
804 }
805
806 /* Take care of the codec-specific extradata. */
807 if (avctx->extradata_size == 14) {
808 /* Parse the extradata, WAV format */
809 av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
810 bytestream_get_le16(&edata_ptr)); // Unknown value always 1
811 edata_ptr += 4; // samples per channel
812 q->coding_mode = bytestream_get_le16(&edata_ptr);
813 av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
814 bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
815 frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
816 av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
817 bytestream_get_le16(&edata_ptr)); // Unknown always 0
818
819 /* setup */
820 samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
821 version = 4;
822 delay = 0x88E;
823 q->coding_mode = q->coding_mode ? JOINT_STEREO : STEREO;
824 q->scrambled_stream = 0;
825
826 if (avctx->block_align != 96 * avctx->channels * frame_factor &&
827 avctx->block_align != 152 * avctx->channels * frame_factor &&
828 avctx->block_align != 192 * avctx->channels * frame_factor) {
829 av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
830 "configuration %d/%d/%d\n", avctx->block_align,
831 avctx->channels, frame_factor);
832 return AVERROR_INVALIDDATA;
833 }
834 } else if (avctx->extradata_size == 10) {
835 /* Parse the extradata, RM format. */
836 version = bytestream_get_be32(&edata_ptr);
837 samples_per_frame = bytestream_get_be16(&edata_ptr);
838 delay = bytestream_get_be16(&edata_ptr);
839 q->coding_mode = bytestream_get_be16(&edata_ptr);
840 q->scrambled_stream = 1;
841
842 } else {
843 av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
844 avctx->extradata_size);
845 return AVERROR(EINVAL);
846 }
847
848 /* Check the extradata */
849
850 if (version != 4) {
851 av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
852 return AVERROR_INVALIDDATA;
853 }
854
855 if (samples_per_frame != SAMPLES_PER_FRAME &&
856 samples_per_frame != SAMPLES_PER_FRAME * 2) {
857 av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
858 samples_per_frame);
859 return AVERROR_INVALIDDATA;
860 }
861
862 if (delay != 0x88E) {
863 av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
864 delay);
865 return AVERROR_INVALIDDATA;
866 }
867
868 if (q->coding_mode == STEREO)
869 av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
870 else if (q->coding_mode == JOINT_STEREO) {
871 if (avctx->channels != 2)
872 return AVERROR_INVALIDDATA;
873 av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
874 } else {
875 av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
876 q->coding_mode);
877 return AVERROR_INVALIDDATA;
878 }
879
880 if (avctx->block_align >= UINT_MAX / 2)
881 return AVERROR(EINVAL);
882
883 q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
884 AV_INPUT_BUFFER_PADDING_SIZE);
885 if (!q->decoded_bytes_buffer)
886 return AVERROR(ENOMEM);
887
888 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
889
890 /* initialize the MDCT transform */
891 if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
892 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
893 av_freep(&q->decoded_bytes_buffer);
894 return ret;
895 }
896
897 /* init the joint-stereo decoding data */
898 q->weighting_delay[0] = 0;
899 q->weighting_delay[1] = 7;
900 q->weighting_delay[2] = 0;
901 q->weighting_delay[3] = 7;
902 q->weighting_delay[4] = 0;
903 q->weighting_delay[5] = 7;
904
905 for (i = 0; i < 4; i++) {
906 q->matrix_coeff_index_prev[i] = 3;
907 q->matrix_coeff_index_now[i] = 3;
908 q->matrix_coeff_index_next[i] = 3;
909 }
910
911 ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
912 avpriv_float_dsp_init(&q->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
913
914 q->units = av_mallocz(sizeof(*q->units) * avctx->channels);
915 if (!q->units) {
916 atrac3_decode_close(avctx);
917 return AVERROR(ENOMEM);
918 }
919
920 return 0;
921 }
922
923 AVCodec ff_atrac3_decoder = {
924 .name = "atrac3",
925 .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
926 .type = AVMEDIA_TYPE_AUDIO,
927 .id = AV_CODEC_ID_ATRAC3,
928 .priv_data_size = sizeof(ATRAC3Context),
929 .init = atrac3_decode_init,
930 .init_static_data = atrac3_init_static_data,
931 .close = atrac3_decode_close,
932 .decode = atrac3_decode_frame,
933 .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
934 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
935 AV_SAMPLE_FMT_NONE },
936 };