2 * COOK compatible decoder
3 * Copyright (c) 2003 Sascha Sommer
4 * Copyright (c) 2005 Benjamin Larsson
6 * This file is part of Libav.
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.
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.
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
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
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
45 #include "libavutil/lfg.h"
49 #include "bytestream.h"
51 #include "libavutil/audioconvert.h"
56 /* the different Cook versions */
57 #define MONO 0x1000001
58 #define STEREO 0x1000002
59 #define JOINT_STEREO 0x1000003
60 #define MC_COOK 0x2000000 // multichannel Cook, not supported
62 #define SUBBAND_SIZE 20
63 #define MAX_SUBPACKETS 5
75 int samples_per_frame
;
79 int samples_per_channel
;
80 int log2_numvector_size
;
81 unsigned int channel_mask
;
82 VLC ccpl
; ///< channel coupling
84 int bits_per_subpacket
;
87 int numvector_size
; ///< 1 << log2_numvector_size;
89 float mono_previous_buffer1
[1024];
90 float mono_previous_buffer2
[1024];
100 typedef struct cook
{
102 * The following 5 functions provide the lowlevel arithmetic on
103 * the internal audio buffers.
105 void (*scalar_dequant
)(struct cook
*q
, int index
, int quant_index
,
106 int *subband_coef_index
, int *subband_coef_sign
,
109 void (*decouple
)(struct cook
*q
,
113 float *decode_buffer
,
114 float *mlt_buffer1
, float *mlt_buffer2
);
116 void (*imlt_window
)(struct cook
*q
, float *buffer1
,
117 cook_gains
*gains_ptr
, float *previous_buffer
);
119 void (*interpolate
)(struct cook
*q
, float *buffer
,
120 int gain_index
, int gain_index_next
);
122 void (*saturate_output
)(struct cook
*q
, int chan
, float *out
);
124 AVCodecContext
* avctx
;
132 int samples_per_channel
;
135 int discarded_packets
;
142 VLC envelope_quant_index
[13];
143 VLC sqvh
[7]; // scalar quantization
145 /* generatable tables and related variables */
146 int gain_size_factor
;
147 float gain_table
[23];
151 uint8_t* decoded_bytes_buffer
;
152 DECLARE_ALIGNED(32, float, mono_mdct_output
)[2048];
153 float decode_buffer_1
[1024];
154 float decode_buffer_2
[1024];
155 float decode_buffer_0
[1060]; /* static allocation for joint decode */
157 const float *cplscales
[5];
159 COOKSubpacket subpacket
[MAX_SUBPACKETS
];
162 static float pow2tab
[127];
163 static float rootpow2tab
[127];
165 /*************** init functions ***************/
167 /* table generator */
168 static av_cold
void init_pow2table(void)
171 for (i
= -63; i
< 64; i
++) {
172 pow2tab
[63 + i
] = pow(2, i
);
173 rootpow2tab
[63 + i
] = sqrt(pow(2, i
));
177 /* table generator */
178 static av_cold
void init_gain_table(COOKContext
*q
)
181 q
->gain_size_factor
= q
->samples_per_channel
/ 8;
182 for (i
= 0; i
< 23; i
++)
183 q
->gain_table
[i
] = pow(pow2tab
[i
+ 52],
184 (1.0 / (double) q
->gain_size_factor
));
188 static av_cold
int init_cook_vlc_tables(COOKContext
*q
)
193 for (i
= 0; i
< 13; i
++) {
194 result
|= init_vlc(&q
->envelope_quant_index
[i
], 9, 24,
195 envelope_quant_index_huffbits
[i
], 1, 1,
196 envelope_quant_index_huffcodes
[i
], 2, 2, 0);
198 av_log(q
->avctx
, AV_LOG_DEBUG
, "sqvh VLC init\n");
199 for (i
= 0; i
< 7; i
++) {
200 result
|= init_vlc(&q
->sqvh
[i
], vhvlcsize_tab
[i
], vhsize_tab
[i
],
201 cvh_huffbits
[i
], 1, 1,
202 cvh_huffcodes
[i
], 2, 2, 0);
205 for (i
= 0; i
< q
->num_subpackets
; i
++) {
206 if (q
->subpacket
[i
].joint_stereo
== 1) {
207 result
|= init_vlc(&q
->subpacket
[i
].ccpl
, 6, (1 << q
->subpacket
[i
].js_vlc_bits
) - 1,
208 ccpl_huffbits
[q
->subpacket
[i
].js_vlc_bits
- 2], 1, 1,
209 ccpl_huffcodes
[q
->subpacket
[i
].js_vlc_bits
- 2], 2, 2, 0);
210 av_log(q
->avctx
, AV_LOG_DEBUG
, "subpacket %i Joint-stereo VLC used.\n", i
);
214 av_log(q
->avctx
, AV_LOG_DEBUG
, "VLC tables initialized.\n");
218 static av_cold
int init_cook_mlt(COOKContext
*q
)
221 int mlt_size
= q
->samples_per_channel
;
223 if ((q
->mlt_window
= av_malloc(mlt_size
* sizeof(*q
->mlt_window
))) == 0)
224 return AVERROR(ENOMEM
);
226 /* Initialize the MLT window: simple sine window. */
227 ff_sine_window_init(q
->mlt_window
, mlt_size
);
228 for (j
= 0; j
< mlt_size
; j
++)
229 q
->mlt_window
[j
] *= sqrt(2.0 / q
->samples_per_channel
);
231 /* Initialize the MDCT. */
232 if ((ret
= ff_mdct_init(&q
->mdct_ctx
, av_log2(mlt_size
) + 1, 1, 1.0 / 32768.0))) {
233 av_free(q
->mlt_window
);
236 av_log(q
->avctx
, AV_LOG_DEBUG
, "MDCT initialized, order = %d.\n",
237 av_log2(mlt_size
) + 1);
242 static const float *maybe_reformat_buffer32(COOKContext
*q
, const float *ptr
, int n
)
248 static av_cold
void init_cplscales_table(COOKContext
*q
)
251 for (i
= 0; i
< 5; i
++)
252 q
->cplscales
[i
] = maybe_reformat_buffer32(q
, cplscales
[i
], (1 << (i
+ 2)) - 1);
255 /*************** init functions end ***********/
257 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
258 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
261 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
262 * Why? No idea, some checksum/error detection method maybe.
264 * Out buffer size: extra bytes are needed to cope with
265 * padding/misalignment.
266 * Subpackets passed to the decoder can contain two, consecutive
267 * half-subpackets, of identical but arbitrary size.
268 * 1234 1234 1234 1234 extraA extraB
269 * Case 1: AAAA BBBB 0 0
270 * Case 2: AAAA ABBB BB-- 3 3
271 * Case 3: AAAA AABB BBBB 2 2
272 * Case 4: AAAA AAAB BBBB BB-- 1 5
274 * Nice way to waste CPU cycles.
276 * @param inbuffer pointer to byte array of indata
277 * @param out pointer to byte array of outdata
278 * @param bytes number of bytes
280 static inline int decode_bytes(const uint8_t *inbuffer
, uint8_t *out
, int bytes
)
282 static const uint32_t tab
[4] = {
283 AV_BE2NE32C(0x37c511f2u
), AV_BE2NE32C(0xf237c511u
),
284 AV_BE2NE32C(0x11f237c5u
), AV_BE2NE32C(0xc511f237u
),
289 uint32_t *obuf
= (uint32_t *) out
;
290 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
291 * I'm too lazy though, should be something like
292 * for (i = 0; i < bitamount / 64; i++)
293 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
294 * Buffer alignment needs to be checked. */
296 off
= (intptr_t) inbuffer
& 3;
297 buf
= (const uint32_t *) (inbuffer
- off
);
300 for (i
= 0; i
< bytes
/ 4; i
++)
301 obuf
[i
] = c
^ buf
[i
];
309 static av_cold
int cook_decode_close(AVCodecContext
*avctx
)
312 COOKContext
*q
= avctx
->priv_data
;
313 av_log(avctx
, AV_LOG_DEBUG
, "Deallocating memory.\n");
315 /* Free allocated memory buffers. */
316 av_free(q
->mlt_window
);
317 av_free(q
->decoded_bytes_buffer
);
319 /* Free the transform. */
320 ff_mdct_end(&q
->mdct_ctx
);
322 /* Free the VLC tables. */
323 for (i
= 0; i
< 13; i
++)
324 ff_free_vlc(&q
->envelope_quant_index
[i
]);
325 for (i
= 0; i
< 7; i
++)
326 ff_free_vlc(&q
->sqvh
[i
]);
327 for (i
= 0; i
< q
->num_subpackets
; i
++)
328 ff_free_vlc(&q
->subpacket
[i
].ccpl
);
330 av_log(avctx
, AV_LOG_DEBUG
, "Memory deallocated.\n");
336 * Fill the gain array for the timedomain quantization.
338 * @param gb pointer to the GetBitContext
339 * @param gaininfo array[9] of gain indexes
341 static void decode_gain_info(GetBitContext
*gb
, int *gaininfo
)
345 while (get_bits1(gb
)) {
349 n
= get_bits_count(gb
) - 1; // amount of elements*2 to update
353 int index
= get_bits(gb
, 3);
354 int gain
= get_bits1(gb
) ?
get_bits(gb
, 4) - 7 : -1;
357 gaininfo
[i
++] = gain
;
364 * Create the quant index table needed for the envelope.
366 * @param q pointer to the COOKContext
367 * @param quant_index_table pointer to the array
369 static int decode_envelope(COOKContext
*q
, COOKSubpacket
*p
,
370 int *quant_index_table
)
374 quant_index_table
[0] = get_bits(&q
->gb
, 6) - 6; // This is used later in categorize
376 for (i
= 1; i
< p
->total_subbands
; i
++) {
378 if (i
>= p
->js_subband_start
* 2) {
379 vlc_index
-= p
->js_subband_start
;
386 vlc_index
= 13; // the VLC tables >13 are identical to No. 13
388 j
= get_vlc2(&q
->gb
, q
->envelope_quant_index
[vlc_index
- 1].table
,
389 q
->envelope_quant_index
[vlc_index
- 1].bits
, 2);
390 quant_index_table
[i
] = quant_index_table
[i
- 1] + j
- 12; // differential encoding
391 if (quant_index_table
[i
] > 63 || quant_index_table
[i
] < -63) {
392 av_log(q
->avctx
, AV_LOG_ERROR
,
393 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
394 quant_index_table
[i
], i
);
395 return AVERROR_INVALIDDATA
;
403 * Calculate the category and category_index vector.
405 * @param q pointer to the COOKContext
406 * @param quant_index_table pointer to the array
407 * @param category pointer to the category array
408 * @param category_index pointer to the category_index array
410 static void categorize(COOKContext
*q
, COOKSubpacket
*p
, int *quant_index_table
,
411 int *category
, int *category_index
)
413 int exp_idx
, bias
, tmpbias1
, tmpbias2
, bits_left
, num_bits
, index
, v
, i
, j
;
414 int exp_index2
[102] = { 0 };
415 int exp_index1
[102] = { 0 };
417 int tmp_categorize_array
[128 * 2] = { 0 };
418 int tmp_categorize_array1_idx
= p
->numvector_size
;
419 int tmp_categorize_array2_idx
= p
->numvector_size
;
421 bits_left
= p
->bits_per_subpacket
- get_bits_count(&q
->gb
);
423 if (bits_left
> q
->samples_per_channel
) {
424 bits_left
= q
->samples_per_channel
+
425 ((bits_left
- q
->samples_per_channel
) * 5) / 8;
426 //av_log(q->avctx, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
432 for (i
= 32; i
> 0; i
= i
/ 2) {
435 for (j
= p
->total_subbands
; j
> 0; j
--) {
436 exp_idx
= av_clip((i
- quant_index_table
[index
] + bias
) / 2, 0, 7);
438 num_bits
+= expbits_tab
[exp_idx
];
440 if (num_bits
>= bits_left
- 32)
444 /* Calculate total number of bits. */
446 for (i
= 0; i
< p
->total_subbands
; i
++) {
447 exp_idx
= av_clip((bias
- quant_index_table
[i
]) / 2, 0, 7);
448 num_bits
+= expbits_tab
[exp_idx
];
449 exp_index1
[i
] = exp_idx
;
450 exp_index2
[i
] = exp_idx
;
452 tmpbias1
= tmpbias2
= num_bits
;
454 for (j
= 1; j
< p
->numvector_size
; j
++) {
455 if (tmpbias1
+ tmpbias2
> 2 * bits_left
) { /* ---> */
458 for (i
= 0; i
< p
->total_subbands
; i
++) {
459 if (exp_index1
[i
] < 7) {
460 v
= (-2 * exp_index1
[i
]) - quant_index_table
[i
] + bias
;
469 tmp_categorize_array
[tmp_categorize_array1_idx
++] = index
;
470 tmpbias1
-= expbits_tab
[exp_index1
[index
]] -
471 expbits_tab
[exp_index1
[index
] + 1];
476 for (i
= 0; i
< p
->total_subbands
; i
++) {
477 if (exp_index2
[i
] > 0) {
478 v
= (-2 * exp_index2
[i
]) - quant_index_table
[i
] + bias
;
487 tmp_categorize_array
[--tmp_categorize_array2_idx
] = index
;
488 tmpbias2
-= expbits_tab
[exp_index2
[index
]] -
489 expbits_tab
[exp_index2
[index
] - 1];
494 for (i
= 0; i
< p
->total_subbands
; i
++)
495 category
[i
] = exp_index2
[i
];
497 for (i
= 0; i
< p
->numvector_size
- 1; i
++)
498 category_index
[i
] = tmp_categorize_array
[tmp_categorize_array2_idx
++];
503 * Expand the category vector.
505 * @param q pointer to the COOKContext
506 * @param category pointer to the category array
507 * @param category_index pointer to the category_index array
509 static inline void expand_category(COOKContext
*q
, int *category
,
513 for (i
= 0; i
< q
->num_vectors
; i
++)
515 int idx
= category_index
[i
];
516 if (++category
[idx
] >= FF_ARRAY_ELEMS(dither_tab
))
522 * The real requantization of the mltcoefs
524 * @param q pointer to the COOKContext
526 * @param quant_index quantisation index
527 * @param subband_coef_index array of indexes to quant_centroid_tab
528 * @param subband_coef_sign signs of coefficients
529 * @param mlt_p pointer into the mlt buffer
531 static void scalar_dequant_float(COOKContext
*q
, int index
, int quant_index
,
532 int *subband_coef_index
, int *subband_coef_sign
,
538 for (i
= 0; i
< SUBBAND_SIZE
; i
++) {
539 if (subband_coef_index
[i
]) {
540 f1
= quant_centroid_tab
[index
][subband_coef_index
[i
]];
541 if (subband_coef_sign
[i
])
544 /* noise coding if subband_coef_index[i] == 0 */
545 f1
= dither_tab
[index
];
546 if (av_lfg_get(&q
->random_state
) < 0x80000000)
549 mlt_p
[i
] = f1
* rootpow2tab
[quant_index
+ 63];
553 * Unpack the subband_coef_index and subband_coef_sign vectors.
555 * @param q pointer to the COOKContext
556 * @param category pointer to the category array
557 * @param subband_coef_index array of indexes to quant_centroid_tab
558 * @param subband_coef_sign signs of coefficients
560 static int unpack_SQVH(COOKContext
*q
, COOKSubpacket
*p
, int category
,
561 int *subband_coef_index
, int *subband_coef_sign
)
564 int vlc
, vd
, tmp
, result
;
566 vd
= vd_tab
[category
];
568 for (i
= 0; i
< vpr_tab
[category
]; i
++) {
569 vlc
= get_vlc2(&q
->gb
, q
->sqvh
[category
].table
, q
->sqvh
[category
].bits
, 3);
570 if (p
->bits_per_subpacket
< get_bits_count(&q
->gb
)) {
574 for (j
= vd
- 1; j
>= 0; j
--) {
575 tmp
= (vlc
* invradix_tab
[category
]) / 0x100000;
576 subband_coef_index
[vd
* i
+ j
] = vlc
- tmp
* (kmax_tab
[category
] + 1);
579 for (j
= 0; j
< vd
; j
++) {
580 if (subband_coef_index
[i
* vd
+ j
]) {
581 if (get_bits_count(&q
->gb
) < p
->bits_per_subpacket
) {
582 subband_coef_sign
[i
* vd
+ j
] = get_bits1(&q
->gb
);
585 subband_coef_sign
[i
* vd
+ j
] = 0;
588 subband_coef_sign
[i
* vd
+ j
] = 0;
597 * Fill the mlt_buffer with mlt coefficients.
599 * @param q pointer to the COOKContext
600 * @param category pointer to the category array
601 * @param quant_index_table pointer to the array
602 * @param mlt_buffer pointer to mlt coefficients
604 static void decode_vectors(COOKContext
*q
, COOKSubpacket
*p
, int *category
,
605 int *quant_index_table
, float *mlt_buffer
)
607 /* A zero in this table means that the subband coefficient is
608 random noise coded. */
609 int subband_coef_index
[SUBBAND_SIZE
];
610 /* A zero in this table means that the subband coefficient is a
611 positive multiplicator. */
612 int subband_coef_sign
[SUBBAND_SIZE
];
616 for (band
= 0; band
< p
->total_subbands
; band
++) {
617 index
= category
[band
];
618 if (category
[band
] < 7) {
619 if (unpack_SQVH(q
, p
, category
[band
], subband_coef_index
, subband_coef_sign
)) {
621 for (j
= 0; j
< p
->total_subbands
; j
++)
622 category
[band
+ j
] = 7;
626 memset(subband_coef_index
, 0, sizeof(subband_coef_index
));
627 memset(subband_coef_sign
, 0, sizeof(subband_coef_sign
));
629 q
->scalar_dequant(q
, index
, quant_index_table
[band
],
630 subband_coef_index
, subband_coef_sign
,
631 &mlt_buffer
[band
* SUBBAND_SIZE
]);
634 /* FIXME: should this be removed, or moved into loop above? */
635 if (p
->total_subbands
* SUBBAND_SIZE
>= q
->samples_per_channel
)
641 * function for decoding mono data
643 * @param q pointer to the COOKContext
644 * @param mlt_buffer pointer to mlt coefficients
646 static int mono_decode(COOKContext
*q
, COOKSubpacket
*p
, float *mlt_buffer
)
648 int category_index
[128] = { 0 };
649 int category
[128] = { 0 };
650 int quant_index_table
[102];
653 if ((res
= decode_envelope(q
, p
, quant_index_table
)) < 0)
655 q
->num_vectors
= get_bits(&q
->gb
, p
->log2_numvector_size
);
656 categorize(q
, p
, quant_index_table
, category
, category_index
);
657 expand_category(q
, category
, category_index
);
658 decode_vectors(q
, p
, category
, quant_index_table
, mlt_buffer
);
665 * the actual requantization of the timedomain samples
667 * @param q pointer to the COOKContext
668 * @param buffer pointer to the timedomain buffer
669 * @param gain_index index for the block multiplier
670 * @param gain_index_next index for the next block multiplier
672 static void interpolate_float(COOKContext
*q
, float *buffer
,
673 int gain_index
, int gain_index_next
)
677 fc1
= pow2tab
[gain_index
+ 63];
679 if (gain_index
== gain_index_next
) { // static gain
680 for (i
= 0; i
< q
->gain_size_factor
; i
++)
682 } else { // smooth gain
683 fc2
= q
->gain_table
[11 + (gain_index_next
- gain_index
)];
684 for (i
= 0; i
< q
->gain_size_factor
; i
++) {
692 * Apply transform window, overlap buffers.
694 * @param q pointer to the COOKContext
695 * @param inbuffer pointer to the mltcoefficients
696 * @param gains_ptr current and previous gains
697 * @param previous_buffer pointer to the previous buffer to be used for overlapping
699 static void imlt_window_float(COOKContext
*q
, float *inbuffer
,
700 cook_gains
*gains_ptr
, float *previous_buffer
)
702 const float fc
= pow2tab
[gains_ptr
->previous
[0] + 63];
704 /* The weird thing here, is that the two halves of the time domain
705 * buffer are swapped. Also, the newest data, that we save away for
706 * next frame, has the wrong sign. Hence the subtraction below.
707 * Almost sounds like a complex conjugate/reverse data/FFT effect.
710 /* Apply window and overlap */
711 for (i
= 0; i
< q
->samples_per_channel
; i
++)
712 inbuffer
[i
] = inbuffer
[i
] * fc
* q
->mlt_window
[i
] -
713 previous_buffer
[i
] * q
->mlt_window
[q
->samples_per_channel
- 1 - i
];
717 * The modulated lapped transform, this takes transform coefficients
718 * and transforms them into timedomain samples.
719 * Apply transform window, overlap buffers, apply gain profile
720 * and buffer management.
722 * @param q pointer to the COOKContext
723 * @param inbuffer pointer to the mltcoefficients
724 * @param gains_ptr current and previous gains
725 * @param previous_buffer pointer to the previous buffer to be used for overlapping
727 static void imlt_gain(COOKContext
*q
, float *inbuffer
,
728 cook_gains
*gains_ptr
, float *previous_buffer
)
730 float *buffer0
= q
->mono_mdct_output
;
731 float *buffer1
= q
->mono_mdct_output
+ q
->samples_per_channel
;
734 /* Inverse modified discrete cosine transform */
735 q
->mdct_ctx
.imdct_calc(&q
->mdct_ctx
, q
->mono_mdct_output
, inbuffer
);
737 q
->imlt_window(q
, buffer1
, gains_ptr
, previous_buffer
);
739 /* Apply gain profile */
740 for (i
= 0; i
< 8; i
++)
741 if (gains_ptr
->now
[i
] || gains_ptr
->now
[i
+ 1])
742 q
->interpolate(q
, &buffer1
[q
->gain_size_factor
* i
],
743 gains_ptr
->now
[i
], gains_ptr
->now
[i
+ 1]);
745 /* Save away the current to be previous block. */
746 memcpy(previous_buffer
, buffer0
,
747 q
->samples_per_channel
* sizeof(*previous_buffer
));
752 * function for getting the jointstereo coupling information
754 * @param q pointer to the COOKContext
755 * @param decouple_tab decoupling array
758 static void decouple_info(COOKContext
*q
, COOKSubpacket
*p
, int *decouple_tab
)
761 int vlc
= get_bits1(&q
->gb
);
762 int start
= cplband
[p
->js_subband_start
];
763 int end
= cplband
[p
->subbands
- 1];
764 int length
= end
- start
+ 1;
770 for (i
= 0; i
< length
; i
++)
771 decouple_tab
[start
+ i
] = get_vlc2(&q
->gb
, p
->ccpl
.table
, p
->ccpl
.bits
, 2);
773 for (i
= 0; i
< length
; i
++)
774 decouple_tab
[start
+ i
] = get_bits(&q
->gb
, p
->js_vlc_bits
);
778 * function decouples a pair of signals from a single signal via multiplication.
780 * @param q pointer to the COOKContext
781 * @param subband index of the current subband
782 * @param f1 multiplier for channel 1 extraction
783 * @param f2 multiplier for channel 2 extraction
784 * @param decode_buffer input buffer
785 * @param mlt_buffer1 pointer to left channel mlt coefficients
786 * @param mlt_buffer2 pointer to right channel mlt coefficients
788 static void decouple_float(COOKContext
*q
,
792 float *decode_buffer
,
793 float *mlt_buffer1
, float *mlt_buffer2
)
796 for (j
= 0; j
< SUBBAND_SIZE
; j
++) {
797 tmp_idx
= ((p
->js_subband_start
+ subband
) * SUBBAND_SIZE
) + j
;
798 mlt_buffer1
[SUBBAND_SIZE
* subband
+ j
] = f1
* decode_buffer
[tmp_idx
];
799 mlt_buffer2
[SUBBAND_SIZE
* subband
+ j
] = f2
* decode_buffer
[tmp_idx
];
804 * function for decoding joint stereo data
806 * @param q pointer to the COOKContext
807 * @param mlt_buffer1 pointer to left channel mlt coefficients
808 * @param mlt_buffer2 pointer to right channel mlt coefficients
810 static int joint_decode(COOKContext
*q
, COOKSubpacket
*p
, float *mlt_buffer1
,
814 int decouple_tab
[SUBBAND_SIZE
] = { 0 };
815 float *decode_buffer
= q
->decode_buffer_0
;
818 const float *cplscale
;
820 memset(decode_buffer
, 0, sizeof(q
->decode_buffer_0
));
822 /* Make sure the buffers are zeroed out. */
823 memset(mlt_buffer1
, 0, 1024 * sizeof(*mlt_buffer1
));
824 memset(mlt_buffer2
, 0, 1024 * sizeof(*mlt_buffer2
));
825 decouple_info(q
, p
, decouple_tab
);
826 if ((res
= mono_decode(q
, p
, decode_buffer
)) < 0)
829 /* The two channels are stored interleaved in decode_buffer. */
830 for (i
= 0; i
< p
->js_subband_start
; i
++) {
831 for (j
= 0; j
< SUBBAND_SIZE
; j
++) {
832 mlt_buffer1
[i
* 20 + j
] = decode_buffer
[i
* 40 + j
];
833 mlt_buffer2
[i
* 20 + j
] = decode_buffer
[i
* 40 + 20 + j
];
837 /* When we reach js_subband_start (the higher frequencies)
838 the coefficients are stored in a coupling scheme. */
839 idx
= (1 << p
->js_vlc_bits
) - 1;
840 for (i
= p
->js_subband_start
; i
< p
->subbands
; i
++) {
841 cpl_tmp
= cplband
[i
];
842 idx
-= decouple_tab
[cpl_tmp
];
843 cplscale
= q
->cplscales
[p
->js_vlc_bits
- 2]; // choose decoupler table
844 f1
= cplscale
[decouple_tab
[cpl_tmp
] + 1];
846 q
->decouple(q
, p
, i
, f1
, f2
, decode_buffer
, mlt_buffer1
, mlt_buffer2
);
847 idx
= (1 << p
->js_vlc_bits
) - 1;
854 * First part of subpacket decoding:
855 * decode raw stream bytes and read gain info.
857 * @param q pointer to the COOKContext
858 * @param inbuffer pointer to raw stream data
859 * @param gains_ptr array of current/prev gain pointers
861 static inline void decode_bytes_and_gain(COOKContext
*q
, COOKSubpacket
*p
,
862 const uint8_t *inbuffer
,
863 cook_gains
*gains_ptr
)
867 offset
= decode_bytes(inbuffer
, q
->decoded_bytes_buffer
,
868 p
->bits_per_subpacket
/ 8);
869 init_get_bits(&q
->gb
, q
->decoded_bytes_buffer
+ offset
,
870 p
->bits_per_subpacket
);
871 decode_gain_info(&q
->gb
, gains_ptr
->now
);
873 /* Swap current and previous gains */
874 FFSWAP(int *, gains_ptr
->now
, gains_ptr
->previous
);
878 * Saturate the output signal and interleave.
880 * @param q pointer to the COOKContext
881 * @param chan channel to saturate
882 * @param out pointer to the output vector
884 static void saturate_output_float(COOKContext
*q
, int chan
, float *out
)
887 float *output
= q
->mono_mdct_output
+ q
->samples_per_channel
;
888 for (j
= 0; j
< q
->samples_per_channel
; j
++) {
889 out
[chan
+ q
->nb_channels
* j
] = av_clipf(output
[j
], -1.0, 1.0);
894 * Final part of subpacket decoding:
895 * Apply modulated lapped transform, gain compensation,
896 * clip and convert to integer.
898 * @param q pointer to the COOKContext
899 * @param decode_buffer pointer to the mlt coefficients
900 * @param gains_ptr array of current/prev gain pointers
901 * @param previous_buffer pointer to the previous buffer to be used for overlapping
902 * @param out pointer to the output buffer
903 * @param chan 0: left or single channel, 1: right channel
905 static inline void mlt_compensate_output(COOKContext
*q
, float *decode_buffer
,
906 cook_gains
*gains_ptr
, float *previous_buffer
,
907 float *out
, int chan
)
909 imlt_gain(q
, decode_buffer
, gains_ptr
, previous_buffer
);
911 q
->saturate_output(q
, chan
, out
);
916 * Cook subpacket decoding. This function returns one decoded subpacket,
917 * usually 1024 samples per channel.
919 * @param q pointer to the COOKContext
920 * @param inbuffer pointer to the inbuffer
921 * @param outbuffer pointer to the outbuffer
923 static int decode_subpacket(COOKContext
*q
, COOKSubpacket
*p
,
924 const uint8_t *inbuffer
, float *outbuffer
)
926 int sub_packet_size
= p
->size
;
929 // for (i = 0; i < sub_packet_size ; i++)
930 // av_log(q->avctx, AV_LOG_ERROR, "%02x", inbuffer[i]);
931 // av_log(q->avctx, AV_LOG_ERROR, "\n");
932 memset(q
->decode_buffer_1
, 0, sizeof(q
->decode_buffer_1
));
933 decode_bytes_and_gain(q
, p
, inbuffer
, &p
->gains1
);
935 if (p
->joint_stereo
) {
936 if ((res
= joint_decode(q
, p
, q
->decode_buffer_1
, q
->decode_buffer_2
)) < 0)
939 if ((res
= mono_decode(q
, p
, q
->decode_buffer_1
)) < 0)
942 if (p
->num_channels
== 2) {
943 decode_bytes_and_gain(q
, p
, inbuffer
+ sub_packet_size
/ 2, &p
->gains2
);
944 if ((res
= mono_decode(q
, p
, q
->decode_buffer_2
)) < 0)
949 mlt_compensate_output(q
, q
->decode_buffer_1
, &p
->gains1
,
950 p
->mono_previous_buffer1
, outbuffer
, p
->ch_idx
);
952 if (p
->num_channels
== 2)
954 mlt_compensate_output(q
, q
->decode_buffer_2
, &p
->gains1
,
955 p
->mono_previous_buffer2
, outbuffer
, p
->ch_idx
+ 1);
957 mlt_compensate_output(q
, q
->decode_buffer_2
, &p
->gains2
,
958 p
->mono_previous_buffer2
, outbuffer
, p
->ch_idx
+ 1);
965 * Cook frame decoding
967 * @param avctx pointer to the AVCodecContext
969 static int cook_decode_frame(AVCodecContext
*avctx
, void *data
,
970 int *got_frame_ptr
, AVPacket
*avpkt
)
972 const uint8_t *buf
= avpkt
->data
;
973 int buf_size
= avpkt
->size
;
974 COOKContext
*q
= avctx
->priv_data
;
975 float *samples
= NULL
;
980 if (buf_size
< avctx
->block_align
)
983 /* get output buffer */
984 if (q
->discarded_packets
>= 2) {
985 q
->frame
.nb_samples
= q
->samples_per_channel
;
986 if ((ret
= avctx
->get_buffer(avctx
, &q
->frame
)) < 0) {
987 av_log(avctx
, AV_LOG_ERROR
, "get_buffer() failed\n");
990 samples
= (float *) q
->frame
.data
[0];
993 /* estimate subpacket sizes */
994 q
->subpacket
[0].size
= avctx
->block_align
;
996 for (i
= 1; i
< q
->num_subpackets
; i
++) {
997 q
->subpacket
[i
].size
= 2 * buf
[avctx
->block_align
- q
->num_subpackets
+ i
];
998 q
->subpacket
[0].size
-= q
->subpacket
[i
].size
+ 1;
999 if (q
->subpacket
[0].size
< 0) {
1000 av_log(avctx
, AV_LOG_DEBUG
,
1001 "frame subpacket size total > avctx->block_align!\n");
1002 return AVERROR_INVALIDDATA
;
1006 /* decode supbackets */
1007 for (i
= 0; i
< q
->num_subpackets
; i
++) {
1008 q
->subpacket
[i
].bits_per_subpacket
= (q
->subpacket
[i
].size
* 8) >>
1009 q
->subpacket
[i
].bits_per_subpdiv
;
1010 q
->subpacket
[i
].ch_idx
= chidx
;
1011 av_log(avctx
, AV_LOG_DEBUG
,
1012 "subpacket[%i] size %i js %i %i block_align %i\n",
1013 i
, q
->subpacket
[i
].size
, q
->subpacket
[i
].joint_stereo
, offset
,
1014 avctx
->block_align
);
1016 if ((ret
= decode_subpacket(q
, &q
->subpacket
[i
], buf
+ offset
, samples
)) < 0)
1018 offset
+= q
->subpacket
[i
].size
;
1019 chidx
+= q
->subpacket
[i
].num_channels
;
1020 av_log(avctx
, AV_LOG_DEBUG
, "subpacket[%i] %i %i\n",
1021 i
, q
->subpacket
[i
].size
* 8, get_bits_count(&q
->gb
));
1024 /* Discard the first two frames: no valid audio. */
1025 if (q
->discarded_packets
< 2) {
1026 q
->discarded_packets
++;
1028 return avctx
->block_align
;
1032 *(AVFrame
*) data
= q
->frame
;
1034 return avctx
->block_align
;
1038 static void dump_cook_context(COOKContext
*q
)
1041 #define PRINT(a, b) av_log(q->avctx, AV_LOG_ERROR, " %s = %d\n", a, b);
1042 av_log(q
->avctx
, AV_LOG_ERROR
, "COOKextradata\n");
1043 av_log(q
->avctx
, AV_LOG_ERROR
, "cookversion=%x\n", q
->subpacket
[0].cookversion
);
1044 if (q
->subpacket
[0].cookversion
> STEREO
) {
1045 PRINT("js_subband_start", q
->subpacket
[0].js_subband_start
);
1046 PRINT("js_vlc_bits", q
->subpacket
[0].js_vlc_bits
);
1048 av_log(q
->avctx
, AV_LOG_ERROR
, "COOKContext\n");
1049 PRINT("nb_channels", q
->nb_channels
);
1050 PRINT("bit_rate", q
->bit_rate
);
1051 PRINT("sample_rate", q
->sample_rate
);
1052 PRINT("samples_per_channel", q
->subpacket
[0].samples_per_channel
);
1053 PRINT("samples_per_frame", q
->subpacket
[0].samples_per_frame
);
1054 PRINT("subbands", q
->subpacket
[0].subbands
);
1055 PRINT("js_subband_start", q
->subpacket
[0].js_subband_start
);
1056 PRINT("log2_numvector_size", q
->subpacket
[0].log2_numvector_size
);
1057 PRINT("numvector_size", q
->subpacket
[0].numvector_size
);
1058 PRINT("total_subbands", q
->subpacket
[0].total_subbands
);
1062 static av_cold
int cook_count_channels(unsigned int mask
)
1066 for (i
= 0; i
< 32; i
++)
1067 if (mask
& (1 << i
))
1073 * Cook initialization
1075 * @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
;
1084 unsigned int channel_mask
= 0;
1088 /* Take care of the codec specific extradata. */
1089 if (extradata_size
<= 0) {
1090 av_log(avctx
, AV_LOG_ERROR
, "Necessary extradata missing!\n");
1091 return AVERROR_INVALIDDATA
;
1093 av_log(avctx
, AV_LOG_DEBUG
, "codecdata_length=%d\n", avctx
->extradata_size
);
1095 /* Take data from the AVCodecContext (RM container). */
1096 q
->sample_rate
= avctx
->sample_rate
;
1097 q
->nb_channels
= avctx
->channels
;
1098 q
->bit_rate
= avctx
->bit_rate
;
1099 if (!q
->nb_channels
) {
1100 av_log(avctx
, AV_LOG_ERROR
, "Invalid number of channels\n");
1101 return AVERROR_INVALIDDATA
;
1104 /* Initialize RNG. */
1105 av_lfg_init(&q
->random_state
, 0);
1107 while (edata_ptr
< edata_ptr_end
) {
1108 /* 8 for mono, 16 for stereo, ? for multichannel
1109 Swap to right endianness so we don't need to care later on. */
1110 if (extradata_size
>= 8) {
1111 q
->subpacket
[s
].cookversion
= bytestream_get_be32(&edata_ptr
);
1112 q
->subpacket
[s
].samples_per_frame
= bytestream_get_be16(&edata_ptr
);
1113 q
->subpacket
[s
].subbands
= bytestream_get_be16(&edata_ptr
);
1114 extradata_size
-= 8;
1116 if (extradata_size
>= 8) {
1117 bytestream_get_be32(&edata_ptr
); // Unknown unused
1118 q
->subpacket
[s
].js_subband_start
= bytestream_get_be16(&edata_ptr
);
1119 q
->subpacket
[s
].js_vlc_bits
= bytestream_get_be16(&edata_ptr
);
1120 extradata_size
-= 8;
1123 /* Initialize extradata related variables. */
1124 q
->subpacket
[s
].samples_per_channel
= q
->subpacket
[s
].samples_per_frame
/ q
->nb_channels
;
1125 q
->subpacket
[s
].bits_per_subpacket
= avctx
->block_align
* 8;
1127 /* Initialize default data states. */
1128 q
->subpacket
[s
].log2_numvector_size
= 5;
1129 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
;
1130 q
->subpacket
[s
].num_channels
= 1;
1132 /* Initialize version-dependent variables */
1134 av_log(avctx
, AV_LOG_DEBUG
, "subpacket[%i].cookversion=%x\n", s
,
1135 q
->subpacket
[s
].cookversion
);
1136 q
->subpacket
[s
].joint_stereo
= 0;
1137 switch (q
->subpacket
[s
].cookversion
) {
1139 if (q
->nb_channels
!= 1) {
1140 av_log_ask_for_sample(avctx
, "Container channels != 1.\n");
1141 return AVERROR_PATCHWELCOME
;
1143 av_log(avctx
, AV_LOG_DEBUG
, "MONO\n");
1146 if (q
->nb_channels
!= 1) {
1147 q
->subpacket
[s
].bits_per_subpdiv
= 1;
1148 q
->subpacket
[s
].num_channels
= 2;
1150 av_log(avctx
, AV_LOG_DEBUG
, "STEREO\n");
1153 if (q
->nb_channels
!= 2) {
1154 av_log_ask_for_sample(avctx
, "Container channels != 2.\n");
1155 return AVERROR_PATCHWELCOME
;
1157 av_log(avctx
, AV_LOG_DEBUG
, "JOINT_STEREO\n");
1158 if (avctx
->extradata_size
>= 16) {
1159 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
+
1160 q
->subpacket
[s
].js_subband_start
;
1161 q
->subpacket
[s
].joint_stereo
= 1;
1162 q
->subpacket
[s
].num_channels
= 2;
1164 if (q
->subpacket
[s
].samples_per_channel
> 256) {
1165 q
->subpacket
[s
].log2_numvector_size
= 6;
1167 if (q
->subpacket
[s
].samples_per_channel
> 512) {
1168 q
->subpacket
[s
].log2_numvector_size
= 7;
1172 av_log(avctx
, AV_LOG_DEBUG
, "MULTI_CHANNEL\n");
1173 if (extradata_size
>= 4)
1174 channel_mask
|= q
->subpacket
[s
].channel_mask
= bytestream_get_be32(&edata_ptr
);
1176 if (cook_count_channels(q
->subpacket
[s
].channel_mask
) > 1) {
1177 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
+
1178 q
->subpacket
[s
].js_subband_start
;
1179 q
->subpacket
[s
].joint_stereo
= 1;
1180 q
->subpacket
[s
].num_channels
= 2;
1181 q
->subpacket
[s
].samples_per_channel
= q
->subpacket
[s
].samples_per_frame
>> 1;
1183 if (q
->subpacket
[s
].samples_per_channel
> 256) {
1184 q
->subpacket
[s
].log2_numvector_size
= 6;
1186 if (q
->subpacket
[s
].samples_per_channel
> 512) {
1187 q
->subpacket
[s
].log2_numvector_size
= 7;
1190 q
->subpacket
[s
].samples_per_channel
= q
->subpacket
[s
].samples_per_frame
;
1194 av_log_ask_for_sample(avctx
, "Unknown Cook version.\n");
1195 return AVERROR_PATCHWELCOME
;
1198 if (s
> 1 && q
->subpacket
[s
].samples_per_channel
!= q
->samples_per_channel
) {
1199 av_log(avctx
, AV_LOG_ERROR
, "different number of samples per channel!\n");
1200 return AVERROR_INVALIDDATA
;
1202 q
->samples_per_channel
= q
->subpacket
[0].samples_per_channel
;
1205 /* Initialize variable relations */
1206 q
->subpacket
[s
].numvector_size
= (1 << q
->subpacket
[s
].log2_numvector_size
);
1208 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1209 if (q
->subpacket
[s
].total_subbands
> 53) {
1210 av_log_ask_for_sample(avctx
, "total_subbands > 53\n");
1211 return AVERROR_PATCHWELCOME
;
1214 if ((q
->subpacket
[s
].js_vlc_bits
> 6) ||
1215 (q
->subpacket
[s
].js_vlc_bits
< 2 * q
->subpacket
[s
].joint_stereo
)) {
1216 av_log(avctx
, AV_LOG_ERROR
, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1217 q
->subpacket
[s
].js_vlc_bits
, 2 * q
->subpacket
[s
].joint_stereo
);
1218 return AVERROR_INVALIDDATA
;
1221 if (q
->subpacket
[s
].subbands
> 50) {
1222 av_log_ask_for_sample(avctx
, "subbands > 50\n");
1223 return AVERROR_PATCHWELCOME
;
1225 q
->subpacket
[s
].gains1
.now
= q
->subpacket
[s
].gain_1
;
1226 q
->subpacket
[s
].gains1
.previous
= q
->subpacket
[s
].gain_2
;
1227 q
->subpacket
[s
].gains2
.now
= q
->subpacket
[s
].gain_3
;
1228 q
->subpacket
[s
].gains2
.previous
= q
->subpacket
[s
].gain_4
;
1230 q
->num_subpackets
++;
1232 if (s
> MAX_SUBPACKETS
) {
1233 av_log_ask_for_sample(avctx
, "Too many subpackets > 5\n");
1234 return AVERROR_PATCHWELCOME
;
1237 /* Generate tables */
1240 init_cplscales_table(q
);
1242 if ((ret
= init_cook_vlc_tables(q
)))
1246 if (avctx
->block_align
>= UINT_MAX
/ 2)
1247 return AVERROR(EINVAL
);
1249 /* Pad the databuffer with:
1250 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1251 FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1252 q
->decoded_bytes_buffer
=
1253 av_mallocz(avctx
->block_align
1254 + DECODE_BYTES_PAD1(avctx
->block_align
)
1255 + FF_INPUT_BUFFER_PADDING_SIZE
);
1256 if (q
->decoded_bytes_buffer
== NULL
)
1257 return AVERROR(ENOMEM
);
1259 /* Initialize transform. */
1260 if ((ret
= init_cook_mlt(q
)))
1263 /* Initialize COOK signal arithmetic handling */
1265 q
->scalar_dequant
= scalar_dequant_float
;
1266 q
->decouple
= decouple_float
;
1267 q
->imlt_window
= imlt_window_float
;
1268 q
->interpolate
= interpolate_float
;
1269 q
->saturate_output
= saturate_output_float
;
1272 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1273 if ((q
->samples_per_channel
== 256) || (q
->samples_per_channel
== 512)
1274 || (q
->samples_per_channel
== 1024)) {
1276 av_log_ask_for_sample(avctx
,
1277 "unknown amount of samples_per_channel = %d\n",
1278 q
->samples_per_channel
);
1279 return AVERROR_PATCHWELCOME
;
1282 avctx
->sample_fmt
= AV_SAMPLE_FMT_FLT
;
1284 avctx
->channel_layout
= channel_mask
;
1286 avctx
->channel_layout
= (avctx
->channels
== 2) ? AV_CH_LAYOUT_STEREO
: AV_CH_LAYOUT_MONO
;
1288 avcodec_get_frame_defaults(&q
->frame
);
1289 avctx
->coded_frame
= &q
->frame
;
1292 dump_cook_context(q
);
1297 AVCodec ff_cook_decoder
= {
1299 .type
= AVMEDIA_TYPE_AUDIO
,
1300 .id
= AV_CODEC_ID_COOK
,
1301 .priv_data_size
= sizeof(COOKContext
),
1302 .init
= cook_decode_init
,
1303 .close
= cook_decode_close
,
1304 .decode
= cook_decode_frame
,
1305 .capabilities
= CODEC_CAP_DR1
,
1306 .long_name
= NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),