Remove redundant code
[FFMpeg-mirror/ordered_chapters.git] / libavcodec / cook.c
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1 /*
2 * COOK compatible decoder
3 * Copyright (c) 2003 Sascha Sommer
4 * Copyright (c) 2005 Benjamin Larsson
6 * This file is part of FFmpeg.
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.
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.
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
23 /**
24 * @file libavcodec/cook.c
25 * Cook compatible decoder. Bastardization of the G.722.1 standard.
26 * This decoder handles RealNetworks, RealAudio G2 data.
27 * Cook is identified by the codec name cook in RM files.
29 * To use this decoder, a calling application must supply the extradata
30 * bytes provided from the RM container; 8+ bytes for mono streams and
31 * 16+ for stereo streams (maybe more).
33 * Codec technicalities (all this assume a buffer length of 1024):
34 * Cook works with several different techniques to achieve its compression.
35 * In the timedomain the buffer is divided into 8 pieces and quantized. If
36 * two neighboring pieces have different quantization index a smooth
37 * quantization curve is used to get a smooth overlap between the different
38 * pieces.
39 * To get to the transformdomain Cook uses a modulated lapped transform.
40 * The transform domain has 50 subbands with 20 elements each. This
41 * means only a maximum of 50*20=1000 coefficients are used out of the 1024
42 * available.
45 #include <math.h>
46 #include <stddef.h>
47 #include <stdio.h>
49 #include "libavutil/lfg.h"
50 #include "libavutil/random_seed.h"
51 #include "avcodec.h"
52 #include "get_bits.h"
53 #include "dsputil.h"
54 #include "bytestream.h"
56 #include "cookdata.h"
58 /* the different Cook versions */
59 #define MONO 0x1000001
60 #define STEREO 0x1000002
61 #define JOINT_STEREO 0x1000003
62 #define MC_COOK 0x2000000 //multichannel Cook, not supported
64 #define SUBBAND_SIZE 20
65 #define MAX_SUBPACKETS 5
66 //#define COOKDEBUG
68 typedef struct {
69 int *now;
70 int *previous;
71 } cook_gains;
73 typedef struct {
74 int ch_idx;
75 int size;
76 int num_channels;
77 int cookversion;
78 int samples_per_frame;
79 int subbands;
80 int js_subband_start;
81 int js_vlc_bits;
82 int samples_per_channel;
83 int log2_numvector_size;
84 unsigned int channel_mask;
85 VLC ccpl; ///< channel coupling
86 int joint_stereo;
87 int bits_per_subpacket;
88 int bits_per_subpdiv;
89 int total_subbands;
90 int numvector_size; ///< 1 << log2_numvector_size;
92 float mono_previous_buffer1[1024];
93 float mono_previous_buffer2[1024];
94 /** gain buffers */
95 cook_gains gains1;
96 cook_gains gains2;
97 int gain_1[9];
98 int gain_2[9];
99 int gain_3[9];
100 int gain_4[9];
101 } COOKSubpacket;
103 typedef struct cook {
105 * The following 5 functions provide the lowlevel arithmetic on
106 * the internal audio buffers.
108 void (* scalar_dequant)(struct cook *q, int index, int quant_index,
109 int* subband_coef_index, int* subband_coef_sign,
110 float* mlt_p);
112 void (* decouple) (struct cook *q,
113 COOKSubpacket *p,
114 int subband,
115 float f1, float f2,
116 float *decode_buffer,
117 float *mlt_buffer1, float *mlt_buffer2);
119 void (* imlt_window) (struct cook *q, float *buffer1,
120 cook_gains *gains_ptr, float *previous_buffer);
122 void (* interpolate) (struct cook *q, float* buffer,
123 int gain_index, int gain_index_next);
125 void (* saturate_output) (struct cook *q, int chan, int16_t *out);
127 AVCodecContext* avctx;
128 GetBitContext gb;
129 /* stream data */
130 int nb_channels;
131 int bit_rate;
132 int sample_rate;
133 int num_vectors;
134 int samples_per_channel;
135 /* states */
136 AVLFG random_state;
138 /* transform data */
139 MDCTContext mdct_ctx;
140 float* mlt_window;
142 /* VLC data */
143 VLC envelope_quant_index[13];
144 VLC sqvh[7]; //scalar quantization
146 /* generatable tables and related variables */
147 int gain_size_factor;
148 float gain_table[23];
150 /* data buffers */
152 uint8_t* decoded_bytes_buffer;
153 DECLARE_ALIGNED_16(float,mono_mdct_output[2048]);
154 float decode_buffer_1[1024];
155 float decode_buffer_2[1024];
156 float decode_buffer_0[1060]; /* static allocation for joint decode */
158 const float *cplscales[5];
159 int num_subpackets;
160 COOKSubpacket subpacket[MAX_SUBPACKETS];
161 } COOKContext;
163 static float pow2tab[127];
164 static float rootpow2tab[127];
166 /* debug functions */
168 #ifdef COOKDEBUG
169 static void dump_float_table(float* table, int size, int delimiter) {
170 int i=0;
171 av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
172 for (i=0 ; i<size ; i++) {
173 av_log(NULL, AV_LOG_ERROR, "%5.1f, ", table[i]);
174 if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
178 static void dump_int_table(int* table, int size, int delimiter) {
179 int i=0;
180 av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
181 for (i=0 ; i<size ; i++) {
182 av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
183 if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
187 static void dump_short_table(short* table, int size, int delimiter) {
188 int i=0;
189 av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
190 for (i=0 ; i<size ; i++) {
191 av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
192 if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
196 #endif
198 /*************** init functions ***************/
200 /* table generator */
201 static av_cold void init_pow2table(void){
202 int i;
203 for (i=-63 ; i<64 ; i++){
204 pow2tab[63+i]= pow(2, i);
205 rootpow2tab[63+i]=sqrt(pow(2, i));
209 /* table generator */
210 static av_cold void init_gain_table(COOKContext *q) {
211 int i;
212 q->gain_size_factor = q->samples_per_channel/8;
213 for (i=0 ; i<23 ; i++) {
214 q->gain_table[i] = pow(pow2tab[i+52] ,
215 (1.0/(double)q->gain_size_factor));
220 static av_cold int init_cook_vlc_tables(COOKContext *q) {
221 int i, result;
223 result = 0;
224 for (i=0 ; i<13 ; i++) {
225 result |= init_vlc (&q->envelope_quant_index[i], 9, 24,
226 envelope_quant_index_huffbits[i], 1, 1,
227 envelope_quant_index_huffcodes[i], 2, 2, 0);
229 av_log(q->avctx,AV_LOG_DEBUG,"sqvh VLC init\n");
230 for (i=0 ; i<7 ; i++) {
231 result |= init_vlc (&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
232 cvh_huffbits[i], 1, 1,
233 cvh_huffcodes[i], 2, 2, 0);
236 for(i=0;i<q->num_subpackets;i++){
237 if (q->subpacket[i].joint_stereo==1){
238 result |= init_vlc (&q->subpacket[i].ccpl, 6, (1<<q->subpacket[i].js_vlc_bits)-1,
239 ccpl_huffbits[q->subpacket[i].js_vlc_bits-2], 1, 1,
240 ccpl_huffcodes[q->subpacket[i].js_vlc_bits-2], 2, 2, 0);
241 av_log(q->avctx,AV_LOG_DEBUG,"subpacket %i Joint-stereo VLC used.\n",i);
245 av_log(q->avctx,AV_LOG_DEBUG,"VLC tables initialized.\n");
246 return result;
249 static av_cold int init_cook_mlt(COOKContext *q) {
250 int j;
251 int mlt_size = q->samples_per_channel;
253 if ((q->mlt_window = av_malloc(sizeof(float)*mlt_size)) == 0)
254 return -1;
256 /* Initialize the MLT window: simple sine window. */
257 ff_sine_window_init(q->mlt_window, mlt_size);
258 for(j=0 ; j<mlt_size ; j++)
259 q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
261 /* Initialize the MDCT. */
262 if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1)) {
263 av_free(q->mlt_window);
264 return -1;
266 av_log(q->avctx,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
267 av_log2(mlt_size)+1);
269 return 0;
272 static const float *maybe_reformat_buffer32 (COOKContext *q, const float *ptr, int n)
274 if (1)
275 return ptr;
278 static av_cold void init_cplscales_table (COOKContext *q) {
279 int i;
280 for (i=0;i<5;i++)
281 q->cplscales[i] = maybe_reformat_buffer32 (q, cplscales[i], (1<<(i+2))-1);
284 /*************** init functions end ***********/
287 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
288 * Why? No idea, some checksum/error detection method maybe.
290 * Out buffer size: extra bytes are needed to cope with
291 * padding/misalignment.
292 * Subpackets passed to the decoder can contain two, consecutive
293 * half-subpackets, of identical but arbitrary size.
294 * 1234 1234 1234 1234 extraA extraB
295 * Case 1: AAAA BBBB 0 0
296 * Case 2: AAAA ABBB BB-- 3 3
297 * Case 3: AAAA AABB BBBB 2 2
298 * Case 4: AAAA AAAB BBBB BB-- 1 5
300 * Nice way to waste CPU cycles.
302 * @param inbuffer pointer to byte array of indata
303 * @param out pointer to byte array of outdata
304 * @param bytes number of bytes
306 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes)+3) % 4)
307 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
309 static inline int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
310 int i, off;
311 uint32_t c;
312 const uint32_t* buf;
313 uint32_t* obuf = (uint32_t*) out;
314 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
315 * I'm too lazy though, should be something like
316 * for(i=0 ; i<bitamount/64 ; i++)
317 * (int64_t)out[i] = 0x37c511f237c511f2^be2me_64(int64_t)in[i]);
318 * Buffer alignment needs to be checked. */
320 off = (intptr_t)inbuffer & 3;
321 buf = (const uint32_t*) (inbuffer - off);
322 c = be2me_32((0x37c511f2 >> (off*8)) | (0x37c511f2 << (32-(off*8))));
323 bytes += 3 + off;
324 for (i = 0; i < bytes/4; i++)
325 obuf[i] = c ^ buf[i];
327 return off;
331 * Cook uninit
334 static av_cold int cook_decode_close(AVCodecContext *avctx)
336 int i;
337 COOKContext *q = avctx->priv_data;
338 av_log(avctx,AV_LOG_DEBUG, "Deallocating memory.\n");
340 /* Free allocated memory buffers. */
341 av_free(q->mlt_window);
342 av_free(q->decoded_bytes_buffer);
344 /* Free the transform. */
345 ff_mdct_end(&q->mdct_ctx);
347 /* Free the VLC tables. */
348 for (i=0 ; i<13 ; i++) {
349 free_vlc(&q->envelope_quant_index[i]);
351 for (i=0 ; i<7 ; i++) {
352 free_vlc(&q->sqvh[i]);
354 for (i=0 ; i<q->num_subpackets ; i++) {
355 free_vlc(&q->subpacket[i].ccpl);
358 av_log(avctx,AV_LOG_DEBUG,"Memory deallocated.\n");
360 return 0;
364 * Fill the gain array for the timedomain quantization.
366 * @param q pointer to the COOKContext
367 * @param gaininfo[9] array of gain indexes
370 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
372 int i, n;
374 while (get_bits1(gb)) {}
375 n = get_bits_count(gb) - 1; //amount of elements*2 to update
377 i = 0;
378 while (n--) {
379 int index = get_bits(gb, 3);
380 int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
382 while (i <= index) gaininfo[i++] = gain;
384 while (i <= 8) gaininfo[i++] = 0;
388 * Create the quant index table needed for the envelope.
390 * @param q pointer to the COOKContext
391 * @param quant_index_table pointer to the array
394 static void decode_envelope(COOKContext *q, COOKSubpacket *p, int* quant_index_table) {
395 int i,j, vlc_index;
397 quant_index_table[0]= get_bits(&q->gb,6) - 6; //This is used later in categorize
399 for (i=1 ; i < p->total_subbands ; i++){
400 vlc_index=i;
401 if (i >= p->js_subband_start * 2) {
402 vlc_index-=p->js_subband_start;
403 } else {
404 vlc_index/=2;
405 if(vlc_index < 1) vlc_index = 1;
407 if (vlc_index>13) vlc_index = 13; //the VLC tables >13 are identical to No. 13
409 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index-1].table,
410 q->envelope_quant_index[vlc_index-1].bits,2);
411 quant_index_table[i] = quant_index_table[i-1] + j - 12; //differential encoding
416 * Calculate the category and category_index vector.
418 * @param q pointer to the COOKContext
419 * @param quant_index_table pointer to the array
420 * @param category pointer to the category array
421 * @param category_index pointer to the category_index array
424 static void categorize(COOKContext *q, COOKSubpacket *p, int* quant_index_table,
425 int* category, int* category_index){
426 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
427 int exp_index2[102];
428 int exp_index1[102];
430 int tmp_categorize_array[128*2];
431 int tmp_categorize_array1_idx=p->numvector_size;
432 int tmp_categorize_array2_idx=p->numvector_size;
434 bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
436 if(bits_left > q->samples_per_channel) {
437 bits_left = q->samples_per_channel +
438 ((bits_left - q->samples_per_channel)*5)/8;
439 //av_log(q->avctx, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
442 memset(&exp_index1,0,102*sizeof(int));
443 memset(&exp_index2,0,102*sizeof(int));
444 memset(&tmp_categorize_array,0,128*2*sizeof(int));
446 bias=-32;
448 /* Estimate bias. */
449 for (i=32 ; i>0 ; i=i/2){
450 num_bits = 0;
451 index = 0;
452 for (j=p->total_subbands ; j>0 ; j--){
453 exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
454 index++;
455 num_bits+=expbits_tab[exp_idx];
457 if(num_bits >= bits_left - 32){
458 bias+=i;
462 /* Calculate total number of bits. */
463 num_bits=0;
464 for (i=0 ; i<p->total_subbands ; i++) {
465 exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
466 num_bits += expbits_tab[exp_idx];
467 exp_index1[i] = exp_idx;
468 exp_index2[i] = exp_idx;
470 tmpbias1 = tmpbias2 = num_bits;
472 for (j = 1 ; j < p->numvector_size ; j++) {
473 if (tmpbias1 + tmpbias2 > 2*bits_left) { /* ---> */
474 int max = -999999;
475 index=-1;
476 for (i=0 ; i<p->total_subbands ; i++){
477 if (exp_index1[i] < 7) {
478 v = (-2*exp_index1[i]) - quant_index_table[i] + bias;
479 if ( v >= max) {
480 max = v;
481 index = i;
485 if(index==-1)break;
486 tmp_categorize_array[tmp_categorize_array1_idx++] = index;
487 tmpbias1 -= expbits_tab[exp_index1[index]] -
488 expbits_tab[exp_index1[index]+1];
489 ++exp_index1[index];
490 } else { /* <--- */
491 int min = 999999;
492 index=-1;
493 for (i=0 ; i<p->total_subbands ; i++){
494 if(exp_index2[i] > 0){
495 v = (-2*exp_index2[i])-quant_index_table[i]+bias;
496 if ( v < min) {
497 min = v;
498 index = i;
502 if(index == -1)break;
503 tmp_categorize_array[--tmp_categorize_array2_idx] = index;
504 tmpbias2 -= expbits_tab[exp_index2[index]] -
505 expbits_tab[exp_index2[index]-1];
506 --exp_index2[index];
510 for(i=0 ; i<p->total_subbands ; i++)
511 category[i] = exp_index2[i];
513 for(i=0 ; i<p->numvector_size-1 ; i++)
514 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
520 * Expand the category vector.
522 * @param q pointer to the COOKContext
523 * @param category pointer to the category array
524 * @param category_index pointer to the category_index array
527 static inline void expand_category(COOKContext *q, int* category,
528 int* category_index){
529 int i;
530 for(i=0 ; i<q->num_vectors ; i++){
531 ++category[category_index[i]];
536 * The real requantization of the mltcoefs
538 * @param q pointer to the COOKContext
539 * @param index index
540 * @param quant_index quantisation index
541 * @param subband_coef_index array of indexes to quant_centroid_tab
542 * @param subband_coef_sign signs of coefficients
543 * @param mlt_p pointer into the mlt buffer
546 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
547 int* subband_coef_index, int* subband_coef_sign,
548 float* mlt_p){
549 int i;
550 float f1;
552 for(i=0 ; i<SUBBAND_SIZE ; i++) {
553 if (subband_coef_index[i]) {
554 f1 = quant_centroid_tab[index][subband_coef_index[i]];
555 if (subband_coef_sign[i]) f1 = -f1;
556 } else {
557 /* noise coding if subband_coef_index[i] == 0 */
558 f1 = dither_tab[index];
559 if (av_lfg_get(&q->random_state) < 0x80000000) f1 = -f1;
561 mlt_p[i] = f1 * rootpow2tab[quant_index+63];
565 * Unpack the subband_coef_index and subband_coef_sign vectors.
567 * @param q pointer to the COOKContext
568 * @param category pointer to the category array
569 * @param subband_coef_index array of indexes to quant_centroid_tab
570 * @param subband_coef_sign signs of coefficients
573 static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category, int* subband_coef_index,
574 int* subband_coef_sign) {
575 int i,j;
576 int vlc, vd ,tmp, result;
578 vd = vd_tab[category];
579 result = 0;
580 for(i=0 ; i<vpr_tab[category] ; i++){
581 vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
582 if (p->bits_per_subpacket < get_bits_count(&q->gb)){
583 vlc = 0;
584 result = 1;
586 for(j=vd-1 ; j>=0 ; j--){
587 tmp = (vlc * invradix_tab[category])/0x100000;
588 subband_coef_index[vd*i+j] = vlc - tmp * (kmax_tab[category]+1);
589 vlc = tmp;
591 for(j=0 ; j<vd ; j++){
592 if (subband_coef_index[i*vd + j]) {
593 if(get_bits_count(&q->gb) < p->bits_per_subpacket){
594 subband_coef_sign[i*vd+j] = get_bits1(&q->gb);
595 } else {
596 result=1;
597 subband_coef_sign[i*vd+j]=0;
599 } else {
600 subband_coef_sign[i*vd+j]=0;
604 return result;
609 * Fill the mlt_buffer with mlt coefficients.
611 * @param q pointer to the COOKContext
612 * @param category pointer to the category array
613 * @param quant_index_table pointer to the array
614 * @param mlt_buffer pointer to mlt coefficients
618 static void decode_vectors(COOKContext* q, COOKSubpacket* p, int* category,
619 int *quant_index_table, float* mlt_buffer){
620 /* A zero in this table means that the subband coefficient is
621 random noise coded. */
622 int subband_coef_index[SUBBAND_SIZE];
623 /* A zero in this table means that the subband coefficient is a
624 positive multiplicator. */
625 int subband_coef_sign[SUBBAND_SIZE];
626 int band, j;
627 int index=0;
629 for(band=0 ; band<p->total_subbands ; band++){
630 index = category[band];
631 if(category[band] < 7){
632 if(unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)){
633 index=7;
634 for(j=0 ; j<p->total_subbands ; j++) category[band+j]=7;
637 if(index>=7) {
638 memset(subband_coef_index, 0, sizeof(subband_coef_index));
639 memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
641 q->scalar_dequant(q, index, quant_index_table[band],
642 subband_coef_index, subband_coef_sign,
643 &mlt_buffer[band * SUBBAND_SIZE]);
646 if(p->total_subbands*SUBBAND_SIZE >= q->samples_per_channel){
647 return;
648 } /* FIXME: should this be removed, or moved into loop above? */
653 * function for decoding mono data
655 * @param q pointer to the COOKContext
656 * @param mlt_buffer pointer to mlt coefficients
659 static void mono_decode(COOKContext *q, COOKSubpacket *p, float* mlt_buffer) {
661 int category_index[128];
662 int quant_index_table[102];
663 int category[128];
665 memset(&category, 0, 128*sizeof(int));
666 memset(&category_index, 0, 128*sizeof(int));
668 decode_envelope(q, p, quant_index_table);
669 q->num_vectors = get_bits(&q->gb,p->log2_numvector_size);
670 categorize(q, p, quant_index_table, category, category_index);
671 expand_category(q, category, category_index);
672 decode_vectors(q, p, category, quant_index_table, mlt_buffer);
677 * the actual requantization of the timedomain samples
679 * @param q pointer to the COOKContext
680 * @param buffer pointer to the timedomain buffer
681 * @param gain_index index for the block multiplier
682 * @param gain_index_next index for the next block multiplier
685 static void interpolate_float(COOKContext *q, float* buffer,
686 int gain_index, int gain_index_next){
687 int i;
688 float fc1, fc2;
689 fc1 = pow2tab[gain_index+63];
691 if(gain_index == gain_index_next){ //static gain
692 for(i=0 ; i<q->gain_size_factor ; i++){
693 buffer[i]*=fc1;
695 return;
696 } else { //smooth gain
697 fc2 = q->gain_table[11 + (gain_index_next-gain_index)];
698 for(i=0 ; i<q->gain_size_factor ; i++){
699 buffer[i]*=fc1;
700 fc1*=fc2;
702 return;
707 * Apply transform window, overlap buffers.
709 * @param q pointer to the COOKContext
710 * @param inbuffer pointer to the mltcoefficients
711 * @param gains_ptr current and previous gains
712 * @param previous_buffer pointer to the previous buffer to be used for overlapping
715 static void imlt_window_float (COOKContext *q, float *buffer1,
716 cook_gains *gains_ptr, float *previous_buffer)
718 const float fc = pow2tab[gains_ptr->previous[0] + 63];
719 int i;
720 /* The weird thing here, is that the two halves of the time domain
721 * buffer are swapped. Also, the newest data, that we save away for
722 * next frame, has the wrong sign. Hence the subtraction below.
723 * Almost sounds like a complex conjugate/reverse data/FFT effect.
726 /* Apply window and overlap */
727 for(i = 0; i < q->samples_per_channel; i++){
728 buffer1[i] = buffer1[i] * fc * q->mlt_window[i] -
729 previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
734 * The modulated lapped transform, this takes transform coefficients
735 * and transforms them into timedomain samples.
736 * Apply transform window, overlap buffers, apply gain profile
737 * and buffer management.
739 * @param q pointer to the COOKContext
740 * @param inbuffer pointer to the mltcoefficients
741 * @param gains_ptr current and previous gains
742 * @param previous_buffer pointer to the previous buffer to be used for overlapping
745 static void imlt_gain(COOKContext *q, float *inbuffer,
746 cook_gains *gains_ptr, float* previous_buffer)
748 float *buffer0 = q->mono_mdct_output;
749 float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
750 int i;
752 /* Inverse modified discrete cosine transform */
753 ff_imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
755 q->imlt_window (q, buffer1, gains_ptr, previous_buffer);
757 /* Apply gain profile */
758 for (i = 0; i < 8; i++) {
759 if (gains_ptr->now[i] || gains_ptr->now[i + 1])
760 q->interpolate(q, &buffer1[q->gain_size_factor * i],
761 gains_ptr->now[i], gains_ptr->now[i + 1]);
764 /* Save away the current to be previous block. */
765 memcpy(previous_buffer, buffer0, sizeof(float)*q->samples_per_channel);
770 * function for getting the jointstereo coupling information
772 * @param q pointer to the COOKContext
773 * @param decouple_tab decoupling array
777 static void decouple_info(COOKContext *q, COOKSubpacket *p, int* decouple_tab){
778 int length, i;
780 if(get_bits1(&q->gb)) {
781 if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;
783 length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
784 for (i=0 ; i<length ; i++) {
785 decouple_tab[cplband[p->js_subband_start] + i] = get_vlc2(&q->gb, p->ccpl.table, p->ccpl.bits, 2);
787 return;
790 if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;
792 length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
793 for (i=0 ; i<length ; i++) {
794 decouple_tab[cplband[p->js_subband_start] + i] = get_bits(&q->gb, p->js_vlc_bits);
796 return;
800 * function decouples a pair of signals from a single signal via multiplication.
802 * @param q pointer to the COOKContext
803 * @param subband index of the current subband
804 * @param f1 multiplier for channel 1 extraction
805 * @param f2 multiplier for channel 2 extraction
806 * @param decode_buffer input buffer
807 * @param mlt_buffer1 pointer to left channel mlt coefficients
808 * @param mlt_buffer2 pointer to right channel mlt coefficients
810 static void decouple_float (COOKContext *q,
811 COOKSubpacket *p,
812 int subband,
813 float f1, float f2,
814 float *decode_buffer,
815 float *mlt_buffer1, float *mlt_buffer2)
817 int j, tmp_idx;
818 for (j=0 ; j<SUBBAND_SIZE ; j++) {
819 tmp_idx = ((p->js_subband_start + subband)*SUBBAND_SIZE)+j;
820 mlt_buffer1[SUBBAND_SIZE*subband + j] = f1 * decode_buffer[tmp_idx];
821 mlt_buffer2[SUBBAND_SIZE*subband + j] = f2 * decode_buffer[tmp_idx];
826 * function for decoding joint stereo data
828 * @param q pointer to the COOKContext
829 * @param mlt_buffer1 pointer to left channel mlt coefficients
830 * @param mlt_buffer2 pointer to right channel mlt coefficients
833 static void joint_decode(COOKContext *q, COOKSubpacket *p, float* mlt_buffer1,
834 float* mlt_buffer2) {
835 int i,j;
836 int decouple_tab[SUBBAND_SIZE];
837 float *decode_buffer = q->decode_buffer_0;
838 int idx, cpl_tmp;
839 float f1,f2;
840 const float* cplscale;
842 memset(decouple_tab, 0, sizeof(decouple_tab));
843 memset(decode_buffer, 0, sizeof(decode_buffer));
845 /* Make sure the buffers are zeroed out. */
846 memset(mlt_buffer1,0, 1024*sizeof(float));
847 memset(mlt_buffer2,0, 1024*sizeof(float));
848 decouple_info(q, p, decouple_tab);
849 mono_decode(q, p, decode_buffer);
851 /* The two channels are stored interleaved in decode_buffer. */
852 for (i=0 ; i<p->js_subband_start ; i++) {
853 for (j=0 ; j<SUBBAND_SIZE ; j++) {
854 mlt_buffer1[i*20+j] = decode_buffer[i*40+j];
855 mlt_buffer2[i*20+j] = decode_buffer[i*40+20+j];
859 /* When we reach js_subband_start (the higher frequencies)
860 the coefficients are stored in a coupling scheme. */
861 idx = (1 << p->js_vlc_bits) - 1;
862 for (i=p->js_subband_start ; i<p->subbands ; i++) {
863 cpl_tmp = cplband[i];
864 idx -=decouple_tab[cpl_tmp];
865 cplscale = q->cplscales[p->js_vlc_bits-2]; //choose decoupler table
866 f1 = cplscale[decouple_tab[cpl_tmp]];
867 f2 = cplscale[idx-1];
868 q->decouple (q, p, i, f1, f2, decode_buffer, mlt_buffer1, mlt_buffer2);
869 idx = (1 << p->js_vlc_bits) - 1;
874 * First part of subpacket decoding:
875 * decode raw stream bytes and read gain info.
877 * @param q pointer to the COOKContext
878 * @param inbuffer pointer to raw stream data
879 * @param gain_ptr array of current/prev gain pointers
882 static inline void
883 decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer,
884 cook_gains *gains_ptr)
886 int offset;
888 offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
889 p->bits_per_subpacket/8);
890 init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
891 p->bits_per_subpacket);
892 decode_gain_info(&q->gb, gains_ptr->now);
894 /* Swap current and previous gains */
895 FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
899 * Saturate the output signal to signed 16bit integers.
901 * @param q pointer to the COOKContext
902 * @param chan channel to saturate
903 * @param out pointer to the output vector
905 static void
906 saturate_output_float (COOKContext *q, int chan, int16_t *out)
908 int j;
909 float *output = q->mono_mdct_output + q->samples_per_channel;
910 /* Clip and convert floats to 16 bits.
912 for (j = 0; j < q->samples_per_channel; j++) {
913 out[chan + q->nb_channels * j] =
914 av_clip_int16(lrintf(output[j]));
919 * Final part of subpacket decoding:
920 * Apply modulated lapped transform, gain compensation,
921 * clip and convert to integer.
923 * @param q pointer to the COOKContext
924 * @param decode_buffer pointer to the mlt coefficients
925 * @param gain_ptr array of current/prev gain pointers
926 * @param previous_buffer pointer to the previous buffer to be used for overlapping
927 * @param out pointer to the output buffer
928 * @param chan 0: left or single channel, 1: right channel
931 static inline void
932 mlt_compensate_output(COOKContext *q, float *decode_buffer,
933 cook_gains *gains, float *previous_buffer,
934 int16_t *out, int chan)
936 imlt_gain(q, decode_buffer, gains, previous_buffer);
937 q->saturate_output (q, chan, out);
942 * Cook subpacket decoding. This function returns one decoded subpacket,
943 * usually 1024 samples per channel.
945 * @param q pointer to the COOKContext
946 * @param inbuffer pointer to the inbuffer
947 * @param sub_packet_size subpacket size
948 * @param outbuffer pointer to the outbuffer
952 static void decode_subpacket(COOKContext *q, COOKSubpacket* p, const uint8_t *inbuffer, int16_t *outbuffer) {
953 int sub_packet_size = p->size;
954 /* packet dump */
955 // for (i=0 ; i<sub_packet_size ; i++) {
956 // av_log(q->avctx, AV_LOG_ERROR, "%02x", inbuffer[i]);
957 // }
958 // av_log(q->avctx, AV_LOG_ERROR, "\n");
959 memset(q->decode_buffer_1,0,sizeof(q->decode_buffer_1));
960 decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
962 if (p->joint_stereo) {
963 joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2);
964 } else {
965 mono_decode(q, p, q->decode_buffer_1);
967 if (p->num_channels == 2) {
968 decode_bytes_and_gain(q, p, inbuffer + sub_packet_size/2, &p->gains2);
969 mono_decode(q, p, q->decode_buffer_2);
973 mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
974 p->mono_previous_buffer1, outbuffer, p->ch_idx);
976 if (p->num_channels == 2) {
977 if (p->joint_stereo) {
978 mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
979 p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
980 } else {
981 mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
982 p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
990 * Cook frame decoding
992 * @param avctx pointer to the AVCodecContext
995 static int cook_decode_frame(AVCodecContext *avctx,
996 void *data, int *data_size,
997 AVPacket *avpkt) {
998 const uint8_t *buf = avpkt->data;
999 int buf_size = avpkt->size;
1000 COOKContext *q = avctx->priv_data;
1001 int i;
1002 int offset = 0;
1003 int chidx = 0;
1005 if (buf_size < avctx->block_align)
1006 return buf_size;
1008 /* estimate subpacket sizes */
1009 q->subpacket[0].size = avctx->block_align;
1011 for(i=1;i<q->num_subpackets;i++){
1012 q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
1013 q->subpacket[0].size -= (q->subpacket[i].size + 1);
1016 /* decode supbackets */
1017 *data_size = 0;
1018 for(i=0;i<q->num_subpackets;i++){
1019 q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv;
1020 q->subpacket[i].ch_idx = chidx;
1021 av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align);
1022 decode_subpacket(q, &q->subpacket[i], buf + offset, (int16_t*)data);
1023 offset += q->subpacket[i].size;
1024 chidx += q->subpacket[i].num_channels;
1025 av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb));
1027 *data_size = sizeof(int16_t) * q->nb_channels * q->samples_per_channel;
1029 /* Discard the first two frames: no valid audio. */
1030 if (avctx->frame_number < 2) *data_size = 0;
1032 return avctx->block_align;
1035 #ifdef COOKDEBUG
1036 static void dump_cook_context(COOKContext *q)
1038 //int i=0;
1039 #define PRINT(a,b) av_log(q->avctx,AV_LOG_ERROR," %s = %d\n", a, b);
1040 av_log(q->avctx,AV_LOG_ERROR,"COOKextradata\n");
1041 av_log(q->avctx,AV_LOG_ERROR,"cookversion=%x\n",q->subpacket[0].cookversion);
1042 if (q->subpacket[0].cookversion > STEREO) {
1043 PRINT("js_subband_start",q->subpacket[0].js_subband_start);
1044 PRINT("js_vlc_bits",q->subpacket[0].js_vlc_bits);
1046 av_log(q->avctx,AV_LOG_ERROR,"COOKContext\n");
1047 PRINT("nb_channels",q->nb_channels);
1048 PRINT("bit_rate",q->bit_rate);
1049 PRINT("sample_rate",q->sample_rate);
1050 PRINT("samples_per_channel",q->subpacket[0].samples_per_channel);
1051 PRINT("samples_per_frame",q->subpacket[0].samples_per_frame);
1052 PRINT("subbands",q->subpacket[0].subbands);
1053 PRINT("random_state",q->random_state);
1054 PRINT("js_subband_start",q->subpacket[0].js_subband_start);
1055 PRINT("log2_numvector_size",q->subpacket[0].log2_numvector_size);
1056 PRINT("numvector_size",q->subpacket[0].numvector_size);
1057 PRINT("total_subbands",q->subpacket[0].total_subbands);
1059 #endif
1061 static av_cold int cook_count_channels(unsigned int mask){
1062 int i;
1063 int channels = 0;
1064 for(i = 0;i<32;i++){
1065 if(mask & (1<<i))
1066 ++channels;
1068 return channels;
1072 * Cook initialization
1074 * @param avctx pointer to the AVCodecContext
1077 static av_cold int cook_decode_init(AVCodecContext *avctx)
1079 COOKContext *q = avctx->priv_data;
1080 const uint8_t *edata_ptr = avctx->extradata;
1081 const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
1082 int extradata_size = avctx->extradata_size;
1083 int s = 0;
1084 unsigned int channel_mask = 0;
1085 q->avctx = avctx;
1087 /* Take care of the codec specific extradata. */
1088 if (extradata_size <= 0) {
1089 av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n");
1090 return -1;
1092 av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
1094 /* Take data from the AVCodecContext (RM container). */
1095 q->sample_rate = avctx->sample_rate;
1096 q->nb_channels = avctx->channels;
1097 q->bit_rate = avctx->bit_rate;
1099 /* Initialize RNG. */
1100 av_lfg_init(&q->random_state, ff_random_get_seed());
1102 while(edata_ptr < edata_ptr_end){
1103 /* 8 for mono, 16 for stereo, ? for multichannel
1104 Swap to right endianness so we don't need to care later on. */
1105 if (extradata_size >= 8){
1106 q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
1107 q->subpacket[s].samples_per_frame = bytestream_get_be16(&edata_ptr);
1108 q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
1109 extradata_size -= 8;
1111 if (avctx->extradata_size >= 8){
1112 bytestream_get_be32(&edata_ptr); //Unknown unused
1113 q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
1114 q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
1115 extradata_size -= 8;
1118 /* Initialize extradata related variables. */
1119 q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame / q->nb_channels;
1120 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1122 /* Initialize default data states. */
1123 q->subpacket[s].log2_numvector_size = 5;
1124 q->subpacket[s].total_subbands = q->subpacket[s].subbands;
1125 q->subpacket[s].num_channels = 1;
1127 /* Initialize version-dependent variables */
1129 av_log(avctx,AV_LOG_DEBUG,"subpacket[%i].cookversion=%x\n",s,q->subpacket[s].cookversion);
1130 q->subpacket[s].joint_stereo = 0;
1131 switch (q->subpacket[s].cookversion) {
1132 case MONO:
1133 if (q->nb_channels != 1) {
1134 av_log(avctx,AV_LOG_ERROR,"Container channels != 1, report sample!\n");
1135 return -1;
1137 av_log(avctx,AV_LOG_DEBUG,"MONO\n");
1138 break;
1139 case STEREO:
1140 if (q->nb_channels != 1)
1141 q->subpacket[s].bits_per_subpdiv = 1;
1142 q->subpacket[s].num_channels = 2;
1143 av_log(avctx,AV_LOG_DEBUG,"STEREO\n");
1144 break;
1145 case JOINT_STEREO:
1146 if (q->nb_channels != 2) {
1147 av_log(avctx,AV_LOG_ERROR,"Container channels != 2, report sample!\n");
1148 return -1;
1150 av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n");
1151 if (avctx->extradata_size >= 16){
1152 q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start;
1153 q->subpacket[s].joint_stereo = 1;
1154 q->subpacket[s].num_channels = 2;
1156 if (q->subpacket[s].samples_per_channel > 256) {
1157 q->subpacket[s].log2_numvector_size = 6;
1159 if (q->subpacket[s].samples_per_channel > 512) {
1160 q->subpacket[s].log2_numvector_size = 7;
1162 break;
1163 case MC_COOK:
1164 av_log(avctx,AV_LOG_DEBUG,"MULTI_CHANNEL\n");
1165 if(extradata_size >= 4)
1166 channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
1168 if(cook_count_channels(q->subpacket[s].channel_mask) > 1){
1169 q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start;
1170 q->subpacket[s].joint_stereo = 1;
1171 q->subpacket[s].num_channels = 2;
1172 q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame >> 1;
1174 if (q->subpacket[s].samples_per_channel > 256) {
1175 q->subpacket[s].log2_numvector_size = 6;
1177 if (q->subpacket[s].samples_per_channel > 512) {
1178 q->subpacket[s].log2_numvector_size = 7;
1180 }else
1181 q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame;
1183 break;
1184 default:
1185 av_log(avctx,AV_LOG_ERROR,"Unknown Cook version, report sample!\n");
1186 return -1;
1187 break;
1190 if(s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1191 av_log(avctx,AV_LOG_ERROR,"different number of samples per channel!\n");
1192 return -1;
1193 } else
1194 q->samples_per_channel = q->subpacket[0].samples_per_channel;
1197 /* Initialize variable relations */
1198 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
1200 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1201 if (q->subpacket[s].total_subbands > 53) {
1202 av_log(avctx,AV_LOG_ERROR,"total_subbands > 53, report sample!\n");
1203 return -1;
1206 if ((q->subpacket[s].js_vlc_bits > 6) || (q->subpacket[s].js_vlc_bits < 0)) {
1207 av_log(avctx,AV_LOG_ERROR,"js_vlc_bits = %d, only >= 0 and <= 6 allowed!\n",q->subpacket[s].js_vlc_bits);
1208 return -1;
1211 if (q->subpacket[s].subbands > 50) {
1212 av_log(avctx,AV_LOG_ERROR,"subbands > 50, report sample!\n");
1213 return -1;
1215 q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1216 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1217 q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1218 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1220 q->num_subpackets++;
1221 s++;
1223 /* Generate tables */
1224 init_pow2table();
1225 init_gain_table(q);
1226 init_cplscales_table(q);
1228 if (init_cook_vlc_tables(q) != 0)
1229 return -1;
1232 if(avctx->block_align >= UINT_MAX/2)
1233 return -1;
1235 /* Pad the databuffer with:
1236 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1237 FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1238 q->decoded_bytes_buffer =
1239 av_mallocz(avctx->block_align
1240 + DECODE_BYTES_PAD1(avctx->block_align)
1241 + FF_INPUT_BUFFER_PADDING_SIZE);
1242 if (q->decoded_bytes_buffer == NULL)
1243 return -1;
1245 /* Initialize transform. */
1246 if ( init_cook_mlt(q) != 0 )
1247 return -1;
1249 /* Initialize COOK signal arithmetic handling */
1250 if (1) {
1251 q->scalar_dequant = scalar_dequant_float;
1252 q->decouple = decouple_float;
1253 q->imlt_window = imlt_window_float;
1254 q->interpolate = interpolate_float;
1255 q->saturate_output = saturate_output_float;
1258 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1259 if ((q->samples_per_channel == 256) || (q->samples_per_channel == 512) || (q->samples_per_channel == 1024)) {
1260 } else {
1261 av_log(avctx,AV_LOG_ERROR,"unknown amount of samples_per_channel = %d, report sample!\n",q->samples_per_channel);
1262 return -1;
1265 avctx->sample_fmt = SAMPLE_FMT_S16;
1266 if (channel_mask)
1267 avctx->channel_layout = channel_mask;
1268 else
1269 avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
1271 #ifdef COOKDEBUG
1272 dump_cook_context(q);
1273 #endif
1274 return 0;
1278 AVCodec cook_decoder =
1280 .name = "cook",
1281 .type = CODEC_TYPE_AUDIO,
1282 .id = CODEC_ID_COOK,
1283 .priv_data_size = sizeof(COOKContext),
1284 .init = cook_decode_init,
1285 .close = cook_decode_close,
1286 .decode = cook_decode_frame,
1287 .long_name = NULL_IF_CONFIG_SMALL("COOK"),