Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / atrac3.c
CommitLineData
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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 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
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#include "libavutil/libm.h"
42#include "avcodec.h"
43#include "bytestream.h"
44#include "fft.h"
45#include "fmtconvert.h"
46#include "get_bits.h"
47#include "internal.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
58typedef struct GainBlock {
59 AtracGainInfo g_block[4];
60} GainBlock;
61
62typedef struct TonalComponent {
63 int pos;
64 int num_coefs;
65 float coef[8];
66} TonalComponent;
67
68typedef 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
84typedef struct ATRAC3Context {
85 GetBitContext gb;
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 FmtConvertContext fmt_conv;
f6fa7814 112 AVFloatDSPContext *fdsp;
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113} ATRAC3Context;
114
115static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
116static VLC_TYPE atrac3_vlc_table[4096][2];
117static VLC spectral_coeff_tab[7];
118
119/**
120 * Regular 512 points IMDCT without overlapping, with the exception of the
121 * swapping of odd bands caused by the reverse spectra of the QMF.
122 *
123 * @param odd_band 1 if the band is an odd band
124 */
125static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
126{
127 int i;
128
129 if (odd_band) {
130 /**
131 * Reverse the odd bands before IMDCT, this is an effect of the QMF
132 * transform or it gives better compression to do it this way.
133 * FIXME: It should be possible to handle this in imdct_calc
134 * for that to happen a modification of the prerotation step of
135 * all SIMD code and C code is needed.
136 * Or fix the functions before so they generate a pre reversed spectrum.
137 */
138 for (i = 0; i < 128; i++)
139 FFSWAP(float, input[i], input[255 - i]);
140 }
141
142 q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
143
144 /* Perform windowing on the output. */
f6fa7814 145 q->fdsp->vector_fmul(output, output, mdct_window, MDCT_SIZE);
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146}
147
148/*
149 * indata descrambling, only used for data coming from the rm container
150 */
151static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
152{
153 int i, off;
154 uint32_t c;
155 const uint32_t *buf;
156 uint32_t *output = (uint32_t *)out;
157
158 off = (intptr_t)input & 3;
159 buf = (const uint32_t *)(input - off);
160 if (off)
161 c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
162 else
163 c = av_be2ne32(0x537F6103U);
164 bytes += 3 + off;
165 for (i = 0; i < bytes / 4; i++)
166 output[i] = c ^ buf[i];
167
168 if (off)
169 avpriv_request_sample(NULL, "Offset of %d", off);
170
171 return off;
172}
173
174static av_cold void init_imdct_window(void)
175{
176 int i, j;
177
178 /* generate the mdct window, for details see
179 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
180 for (i = 0, j = 255; i < 128; i++, j--) {
181 float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
182 float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
183 float w = 0.5 * (wi * wi + wj * wj);
184 mdct_window[i] = mdct_window[511 - i] = wi / w;
185 mdct_window[j] = mdct_window[511 - j] = wj / w;
186 }
187}
188
189static av_cold int atrac3_decode_close(AVCodecContext *avctx)
190{
191 ATRAC3Context *q = avctx->priv_data;
192
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193 av_freep(&q->units);
194 av_freep(&q->decoded_bytes_buffer);
195 av_freep(&q->fdsp);
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196
197 ff_mdct_end(&q->mdct_ctx);
198
199 return 0;
200}
201
202/**
203 * Mantissa decoding
204 *
205 * @param selector which table the output values are coded with
206 * @param coding_flag constant length coding or variable length coding
207 * @param mantissas mantissa output table
208 * @param num_codes number of values to get
209 */
210static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
211 int coding_flag, int *mantissas,
212 int num_codes)
213{
214 int i, code, huff_symb;
215
216 if (selector == 1)
217 num_codes /= 2;
218
219 if (coding_flag != 0) {
220 /* constant length coding (CLC) */
221 int num_bits = clc_length_tab[selector];
222
223 if (selector > 1) {
224 for (i = 0; i < num_codes; i++) {
225 if (num_bits)
226 code = get_sbits(gb, num_bits);
227 else
228 code = 0;
229 mantissas[i] = code;
230 }
231 } else {
232 for (i = 0; i < num_codes; i++) {
233 if (num_bits)
234 code = get_bits(gb, num_bits); // num_bits is always 4 in this case
235 else
236 code = 0;
237 mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
238 mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
239 }
240 }
241 } else {
242 /* variable length coding (VLC) */
243 if (selector != 1) {
244 for (i = 0; i < num_codes; i++) {
245 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
246 spectral_coeff_tab[selector-1].bits, 3);
247 huff_symb += 1;
248 code = huff_symb >> 1;
249 if (huff_symb & 1)
250 code = -code;
251 mantissas[i] = code;
252 }
253 } else {
254 for (i = 0; i < num_codes; i++) {
255 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
256 spectral_coeff_tab[selector - 1].bits, 3);
257 mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
258 mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
259 }
260 }
261 }
262}
263
264/**
265 * Restore the quantized band spectrum coefficients
266 *
267 * @return subband count, fix for broken specification/files
268 */
269static int decode_spectrum(GetBitContext *gb, float *output)
270{
271 int num_subbands, coding_mode, i, j, first, last, subband_size;
272 int subband_vlc_index[32], sf_index[32];
273 int mantissas[128];
274 float scale_factor;
275
276 num_subbands = get_bits(gb, 5); // number of coded subbands
277 coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
278
279 /* get the VLC selector table for the subbands, 0 means not coded */
280 for (i = 0; i <= num_subbands; i++)
281 subband_vlc_index[i] = get_bits(gb, 3);
282
283 /* read the scale factor indexes from the stream */
284 for (i = 0; i <= num_subbands; i++) {
285 if (subband_vlc_index[i] != 0)
286 sf_index[i] = get_bits(gb, 6);
287 }
288
289 for (i = 0; i <= num_subbands; i++) {
290 first = subband_tab[i ];
291 last = subband_tab[i + 1];
292
293 subband_size = last - first;
294
295 if (subband_vlc_index[i] != 0) {
296 /* decode spectral coefficients for this subband */
297 /* TODO: This can be done faster is several blocks share the
298 * same VLC selector (subband_vlc_index) */
299 read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
300 mantissas, subband_size);
301
302 /* decode the scale factor for this subband */
303 scale_factor = ff_atrac_sf_table[sf_index[i]] *
304 inv_max_quant[subband_vlc_index[i]];
305
306 /* inverse quantize the coefficients */
307 for (j = 0; first < last; first++, j++)
308 output[first] = mantissas[j] * scale_factor;
309 } else {
310 /* this subband was not coded, so zero the entire subband */
311 memset(output + first, 0, subband_size * sizeof(*output));
312 }
313 }
314
315 /* clear the subbands that were not coded */
316 first = subband_tab[i];
317 memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
318 return num_subbands;
319}
320
321/**
322 * Restore the quantized tonal components
323 *
324 * @param components tonal components
325 * @param num_bands number of coded bands
326 */
327static int decode_tonal_components(GetBitContext *gb,
328 TonalComponent *components, int num_bands)
329{
330 int i, b, c, m;
331 int nb_components, coding_mode_selector, coding_mode;
332 int band_flags[4], mantissa[8];
333 int component_count = 0;
334
335 nb_components = get_bits(gb, 5);
336
337 /* no tonal components */
338 if (nb_components == 0)
339 return 0;
340
341 coding_mode_selector = get_bits(gb, 2);
342 if (coding_mode_selector == 2)
343 return AVERROR_INVALIDDATA;
344
345 coding_mode = coding_mode_selector & 1;
346
347 for (i = 0; i < nb_components; i++) {
348 int coded_values_per_component, quant_step_index;
349
350 for (b = 0; b <= num_bands; b++)
351 band_flags[b] = get_bits1(gb);
352
353 coded_values_per_component = get_bits(gb, 3);
354
355 quant_step_index = get_bits(gb, 3);
356 if (quant_step_index <= 1)
357 return AVERROR_INVALIDDATA;
358
359 if (coding_mode_selector == 3)
360 coding_mode = get_bits1(gb);
361
362 for (b = 0; b < (num_bands + 1) * 4; b++) {
363 int coded_components;
364
365 if (band_flags[b >> 2] == 0)
366 continue;
367
368 coded_components = get_bits(gb, 3);
369
370 for (c = 0; c < coded_components; c++) {
371 TonalComponent *cmp = &components[component_count];
372 int sf_index, coded_values, max_coded_values;
373 float scale_factor;
374
375 sf_index = get_bits(gb, 6);
376 if (component_count >= 64)
377 return AVERROR_INVALIDDATA;
378
379 cmp->pos = b * 64 + get_bits(gb, 6);
380
381 max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
382 coded_values = coded_values_per_component + 1;
383 coded_values = FFMIN(max_coded_values, coded_values);
384
385 scale_factor = ff_atrac_sf_table[sf_index] *
386 inv_max_quant[quant_step_index];
387
388 read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
389 mantissa, coded_values);
390
391 cmp->num_coefs = coded_values;
392
393 /* inverse quant */
394 for (m = 0; m < coded_values; m++)
395 cmp->coef[m] = mantissa[m] * scale_factor;
396
397 component_count++;
398 }
399 }
400 }
401
402 return component_count;
403}
404
405/**
406 * Decode gain parameters for the coded bands
407 *
408 * @param block the gainblock for the current band
409 * @param num_bands amount of coded bands
410 */
411static int decode_gain_control(GetBitContext *gb, GainBlock *block,
412 int num_bands)
413{
414 int b, j;
415 int *level, *loc;
416
417 AtracGainInfo *gain = block->g_block;
418
419 for (b = 0; b <= num_bands; b++) {
420 gain[b].num_points = get_bits(gb, 3);
421 level = gain[b].lev_code;
422 loc = gain[b].loc_code;
423
424 for (j = 0; j < gain[b].num_points; j++) {
425 level[j] = get_bits(gb, 4);
426 loc[j] = get_bits(gb, 5);
427 if (j && loc[j] <= loc[j - 1])
428 return AVERROR_INVALIDDATA;
429 }
430 }
431
432 /* Clear the unused blocks. */
433 for (; b < 4 ; b++)
434 gain[b].num_points = 0;
435
436 return 0;
437}
438
439/**
440 * Combine the tonal band spectrum and regular band spectrum
441 *
442 * @param spectrum output spectrum buffer
443 * @param num_components number of tonal components
444 * @param components tonal components for this band
445 * @return position of the last tonal coefficient
446 */
447static int add_tonal_components(float *spectrum, int num_components,
448 TonalComponent *components)
449{
450 int i, j, last_pos = -1;
451 float *input, *output;
452
453 for (i = 0; i < num_components; i++) {
454 last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
455 input = components[i].coef;
456 output = &spectrum[components[i].pos];
457
458 for (j = 0; j < components[i].num_coefs; j++)
459 output[j] += input[j];
460 }
461
462 return last_pos;
463}
464
465#define INTERPOLATE(old, new, nsample) \
466 ((old) + (nsample) * 0.125 * ((new) - (old)))
467
468static void reverse_matrixing(float *su1, float *su2, int *prev_code,
469 int *curr_code)
470{
471 int i, nsample, band;
472 float mc1_l, mc1_r, mc2_l, mc2_r;
473
474 for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
475 int s1 = prev_code[i];
476 int s2 = curr_code[i];
477 nsample = band;
478
479 if (s1 != s2) {
480 /* Selector value changed, interpolation needed. */
481 mc1_l = matrix_coeffs[s1 * 2 ];
482 mc1_r = matrix_coeffs[s1 * 2 + 1];
483 mc2_l = matrix_coeffs[s2 * 2 ];
484 mc2_r = matrix_coeffs[s2 * 2 + 1];
485
486 /* Interpolation is done over the first eight samples. */
487 for (; nsample < band + 8; nsample++) {
488 float c1 = su1[nsample];
489 float c2 = su2[nsample];
490 c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
491 c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
492 su1[nsample] = c2;
493 su2[nsample] = c1 * 2.0 - c2;
494 }
495 }
496
497 /* Apply the matrix without interpolation. */
498 switch (s2) {
499 case 0: /* M/S decoding */
500 for (; nsample < band + 256; nsample++) {
501 float c1 = su1[nsample];
502 float c2 = su2[nsample];
503 su1[nsample] = c2 * 2.0;
504 su2[nsample] = (c1 - c2) * 2.0;
505 }
506 break;
507 case 1:
508 for (; nsample < band + 256; nsample++) {
509 float c1 = su1[nsample];
510 float c2 = su2[nsample];
511 su1[nsample] = (c1 + c2) * 2.0;
512 su2[nsample] = c2 * -2.0;
513 }
514 break;
515 case 2:
516 case 3:
517 for (; nsample < band + 256; nsample++) {
518 float c1 = su1[nsample];
519 float c2 = su2[nsample];
520 su1[nsample] = c1 + c2;
521 su2[nsample] = c1 - c2;
522 }
523 break;
524 default:
525 av_assert1(0);
526 }
527 }
528}
529
530static void get_channel_weights(int index, int flag, float ch[2])
531{
532 if (index == 7) {
533 ch[0] = 1.0;
534 ch[1] = 1.0;
535 } else {
536 ch[0] = (index & 7) / 7.0;
537 ch[1] = sqrt(2 - ch[0] * ch[0]);
538 if (flag)
539 FFSWAP(float, ch[0], ch[1]);
540 }
541}
542
543static void channel_weighting(float *su1, float *su2, int *p3)
544{
545 int band, nsample;
546 /* w[x][y] y=0 is left y=1 is right */
547 float w[2][2];
548
549 if (p3[1] != 7 || p3[3] != 7) {
550 get_channel_weights(p3[1], p3[0], w[0]);
551 get_channel_weights(p3[3], p3[2], w[1]);
552
553 for (band = 256; band < 4 * 256; band += 256) {
554 for (nsample = band; nsample < band + 8; nsample++) {
555 su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
556 su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
557 }
558 for(; nsample < band + 256; nsample++) {
559 su1[nsample] *= w[1][0];
560 su2[nsample] *= w[1][1];
561 }
562 }
563 }
564}
565
566/**
567 * Decode a Sound Unit
568 *
569 * @param snd the channel unit to be used
570 * @param output the decoded samples before IQMF in float representation
571 * @param channel_num channel number
572 * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)
573 */
574static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
575 ChannelUnit *snd, float *output,
576 int channel_num, int coding_mode)
577{
578 int band, ret, num_subbands, last_tonal, num_bands;
579 GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
580 GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
581
582 if (coding_mode == JOINT_STEREO && channel_num == 1) {
583 if (get_bits(gb, 2) != 3) {
584 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
585 return AVERROR_INVALIDDATA;
586 }
587 } else {
588 if (get_bits(gb, 6) != 0x28) {
589 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
590 return AVERROR_INVALIDDATA;
591 }
592 }
593
594 /* number of coded QMF bands */
595 snd->bands_coded = get_bits(gb, 2);
596
597 ret = decode_gain_control(gb, gain2, snd->bands_coded);
598 if (ret)
599 return ret;
600
601 snd->num_components = decode_tonal_components(gb, snd->components,
602 snd->bands_coded);
603 if (snd->num_components < 0)
604 return snd->num_components;
605
606 num_subbands = decode_spectrum(gb, snd->spectrum);
607
608 /* Merge the decoded spectrum and tonal components. */
609 last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
610 snd->components);
611
612
613 /* calculate number of used MLT/QMF bands according to the amount of coded
614 spectral lines */
615 num_bands = (subband_tab[num_subbands] - 1) >> 8;
616 if (last_tonal >= 0)
617 num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
618
619
620 /* Reconstruct time domain samples. */
621 for (band = 0; band < 4; band++) {
622 /* Perform the IMDCT step without overlapping. */
623 if (band <= num_bands)
624 imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
625 else
626 memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
627
628 /* gain compensation and overlapping */
629 ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
630 &snd->prev_frame[band * 256],
631 &gain1->g_block[band], &gain2->g_block[band],
632 256, &output[band * 256]);
633 }
634
635 /* Swap the gain control buffers for the next frame. */
636 snd->gc_blk_switch ^= 1;
637
638 return 0;
639}
640
641static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
642 float **out_samples)
643{
644 ATRAC3Context *q = avctx->priv_data;
645 int ret, i;
646 uint8_t *ptr1;
647
648 if (q->coding_mode == JOINT_STEREO) {
649 /* channel coupling mode */
650 /* decode Sound Unit 1 */
651 init_get_bits(&q->gb, databuf, avctx->block_align * 8);
652
653 ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
654 JOINT_STEREO);
655 if (ret != 0)
656 return ret;
657
658 /* Framedata of the su2 in the joint-stereo mode is encoded in
659 * reverse byte order so we need to swap it first. */
660 if (databuf == q->decoded_bytes_buffer) {
661 uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
662 ptr1 = q->decoded_bytes_buffer;
663 for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
664 FFSWAP(uint8_t, *ptr1, *ptr2);
665 } else {
666 const uint8_t *ptr2 = databuf + avctx->block_align - 1;
667 for (i = 0; i < avctx->block_align; i++)
668 q->decoded_bytes_buffer[i] = *ptr2--;
669 }
670
671 /* Skip the sync codes (0xF8). */
672 ptr1 = q->decoded_bytes_buffer;
673 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
674 if (i >= avctx->block_align)
675 return AVERROR_INVALIDDATA;
676 }
677
678
679 /* set the bitstream reader at the start of the second Sound Unit*/
680 init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1);
681
682 /* Fill the Weighting coeffs delay buffer */
683 memmove(q->weighting_delay, &q->weighting_delay[2],
684 4 * sizeof(*q->weighting_delay));
685 q->weighting_delay[4] = get_bits1(&q->gb);
686 q->weighting_delay[5] = get_bits(&q->gb, 3);
687
688 for (i = 0; i < 4; i++) {
689 q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
690 q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
691 q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
692 }
693
694 /* Decode Sound Unit 2. */
695 ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
696 out_samples[1], 1, JOINT_STEREO);
697 if (ret != 0)
698 return ret;
699
700 /* Reconstruct the channel coefficients. */
701 reverse_matrixing(out_samples[0], out_samples[1],
702 q->matrix_coeff_index_prev,
703 q->matrix_coeff_index_now);
704
705 channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
706 } else {
707 /* normal stereo mode or mono */
708 /* Decode the channel sound units. */
709 for (i = 0; i < avctx->channels; i++) {
710 /* Set the bitstream reader at the start of a channel sound unit. */
711 init_get_bits(&q->gb,
712 databuf + i * avctx->block_align / avctx->channels,
713 avctx->block_align * 8 / avctx->channels);
714
715 ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
716 out_samples[i], i, q->coding_mode);
717 if (ret != 0)
718 return ret;
719 }
720 }
721
722 /* Apply the iQMF synthesis filter. */
723 for (i = 0; i < avctx->channels; i++) {
724 float *p1 = out_samples[i];
725 float *p2 = p1 + 256;
726 float *p3 = p2 + 256;
727 float *p4 = p3 + 256;
728 ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
729 ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
730 ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
731 }
732
733 return 0;
734}
735
736static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
737 int *got_frame_ptr, AVPacket *avpkt)
738{
739 AVFrame *frame = data;
740 const uint8_t *buf = avpkt->data;
741 int buf_size = avpkt->size;
742 ATRAC3Context *q = avctx->priv_data;
743 int ret;
744 const uint8_t *databuf;
745
746 if (buf_size < avctx->block_align) {
747 av_log(avctx, AV_LOG_ERROR,
748 "Frame too small (%d bytes). Truncated file?\n", buf_size);
749 return AVERROR_INVALIDDATA;
750 }
751
752 /* get output buffer */
753 frame->nb_samples = SAMPLES_PER_FRAME;
754 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
755 return ret;
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
776static av_cold void atrac3_init_static_data(void)
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
794static av_cold int atrac3_decode_init(AVCodecContext *avctx)
795{
796 static int static_init_done;
797 int i, ret;
798 int version, delay, samples_per_frame, frame_factor;
799 const uint8_t *edata_ptr = avctx->extradata;
800 ATRAC3Context *q = avctx->priv_data;
801
802 if (avctx->channels <= 0 || avctx->channels > 2) {
803 av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
804 return AVERROR(EINVAL);
805 }
806
807 if (!static_init_done)
808 atrac3_init_static_data();
809 static_init_done = 1;
810
811 /* Take care of the codec-specific extradata. */
812 if (avctx->extradata_size == 14) {
813 /* Parse the extradata, WAV format */
814 av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
815 bytestream_get_le16(&edata_ptr)); // Unknown value always 1
816 edata_ptr += 4; // samples per channel
817 q->coding_mode = bytestream_get_le16(&edata_ptr);
818 av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
819 bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
820 frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
821 av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
822 bytestream_get_le16(&edata_ptr)); // Unknown always 0
823
824 /* setup */
825 samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
826 version = 4;
827 delay = 0x88E;
828 q->coding_mode = q->coding_mode ? JOINT_STEREO : STEREO;
829 q->scrambled_stream = 0;
830
831 if (avctx->block_align != 96 * avctx->channels * frame_factor &&
832 avctx->block_align != 152 * avctx->channels * frame_factor &&
833 avctx->block_align != 192 * avctx->channels * frame_factor) {
834 av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
835 "configuration %d/%d/%d\n", avctx->block_align,
836 avctx->channels, frame_factor);
837 return AVERROR_INVALIDDATA;
838 }
839 } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
840 /* Parse the extradata, RM format. */
841 version = bytestream_get_be32(&edata_ptr);
842 samples_per_frame = bytestream_get_be16(&edata_ptr);
843 delay = bytestream_get_be16(&edata_ptr);
844 q->coding_mode = bytestream_get_be16(&edata_ptr);
845 q->scrambled_stream = 1;
846
847 } else {
848 av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
849 avctx->extradata_size);
850 return AVERROR(EINVAL);
851 }
852
853 /* Check the extradata */
854
855 if (version != 4) {
856 av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
857 return AVERROR_INVALIDDATA;
858 }
859
860 if (samples_per_frame != SAMPLES_PER_FRAME &&
861 samples_per_frame != SAMPLES_PER_FRAME * 2) {
862 av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
863 samples_per_frame);
864 return AVERROR_INVALIDDATA;
865 }
866
867 if (delay != 0x88E) {
868 av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
869 delay);
870 return AVERROR_INVALIDDATA;
871 }
872
873 if (q->coding_mode == STEREO)
874 av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
875 else if (q->coding_mode == JOINT_STEREO) {
876 if (avctx->channels != 2) {
877 av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");
878 return AVERROR_INVALIDDATA;
879 }
880 av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
881 } else {
882 av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
883 q->coding_mode);
884 return AVERROR_INVALIDDATA;
885 }
886
887 if (avctx->block_align >= UINT_MAX / 2)
888 return AVERROR(EINVAL);
889
890 q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
891 FF_INPUT_BUFFER_PADDING_SIZE);
892 if (!q->decoded_bytes_buffer)
893 return AVERROR(ENOMEM);
894
895 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
896
897 /* initialize the MDCT transform */
898 if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
899 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
900 av_freep(&q->decoded_bytes_buffer);
901 return ret;
902 }
903
904 /* init the joint-stereo decoding data */
905 q->weighting_delay[0] = 0;
906 q->weighting_delay[1] = 7;
907 q->weighting_delay[2] = 0;
908 q->weighting_delay[3] = 7;
909 q->weighting_delay[4] = 0;
910 q->weighting_delay[5] = 7;
911
912 for (i = 0; i < 4; i++) {
913 q->matrix_coeff_index_prev[i] = 3;
914 q->matrix_coeff_index_now[i] = 3;
915 q->matrix_coeff_index_next[i] = 3;
916 }
917
918 ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
f6fa7814 919 q->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
2ba45a60
DM
920 ff_fmt_convert_init(&q->fmt_conv, avctx);
921
922 q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
f6fa7814 923 if (!q->units || !q->fdsp) {
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DM
924 atrac3_decode_close(avctx);
925 return AVERROR(ENOMEM);
926 }
927
928 return 0;
929}
930
931AVCodec ff_atrac3_decoder = {
932 .name = "atrac3",
933 .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
934 .type = AVMEDIA_TYPE_AUDIO,
935 .id = AV_CODEC_ID_ATRAC3,
936 .priv_data_size = sizeof(ATRAC3Context),
937 .init = atrac3_decode_init,
938 .close = atrac3_decode_close,
939 .decode = atrac3_decode_frame,
940 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
941 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
942 AV_SAMPLE_FMT_NONE },
943};