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/channel_layout.h"
46 #include "libavutil/lfg.h"
50 #include "bytestream.h"
57 /* the different Cook versions */
58 #define MONO 0x1000001
59 #define STEREO 0x1000002
60 #define JOINT_STEREO 0x1000003
61 #define MC_COOK 0x2000000 // multichannel Cook, not supported
63 #define SUBBAND_SIZE 20
64 #define MAX_SUBPACKETS 5
79 int samples_per_channel
;
80 int log2_numvector_size
;
81 unsigned int channel_mask
;
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
, float *out
);
124 AVCodecContext
* avctx
;
129 int samples_per_channel
;
132 int discarded_packets
;
139 VLC envelope_quant_index
[13];
140 VLC sqvh
[7]; // scalar quantization
142 /* generatable tables and related variables */
143 int gain_size_factor
;
144 float gain_table
[23];
148 uint8_t* decoded_bytes_buffer
;
149 DECLARE_ALIGNED(32, float, mono_mdct_output
)[2048];
150 float decode_buffer_1
[1024];
151 float decode_buffer_2
[1024];
152 float decode_buffer_0
[1060]; /* static allocation for joint decode */
154 const float *cplscales
[5];
156 COOKSubpacket subpacket
[MAX_SUBPACKETS
];
159 static float pow2tab
[127];
160 static float rootpow2tab
[127];
162 /*************** init functions ***************/
164 /* table generator */
165 static av_cold
void init_pow2table(void)
168 for (i
= -63; i
< 64; i
++) {
169 pow2tab
[63 + i
] = pow(2, i
);
170 rootpow2tab
[63 + i
] = sqrt(pow(2, i
));
174 /* table generator */
175 static av_cold
void init_gain_table(COOKContext
*q
)
178 q
->gain_size_factor
= q
->samples_per_channel
/ 8;
179 for (i
= 0; i
< 23; i
++)
180 q
->gain_table
[i
] = pow(pow2tab
[i
+ 52],
181 (1.0 / (double) q
->gain_size_factor
));
185 static av_cold
int init_cook_vlc_tables(COOKContext
*q
)
190 for (i
= 0; i
< 13; i
++) {
191 result
|= init_vlc(&q
->envelope_quant_index
[i
], 9, 24,
192 envelope_quant_index_huffbits
[i
], 1, 1,
193 envelope_quant_index_huffcodes
[i
], 2, 2, 0);
195 av_log(q
->avctx
, AV_LOG_DEBUG
, "sqvh VLC init\n");
196 for (i
= 0; i
< 7; i
++) {
197 result
|= init_vlc(&q
->sqvh
[i
], vhvlcsize_tab
[i
], vhsize_tab
[i
],
198 cvh_huffbits
[i
], 1, 1,
199 cvh_huffcodes
[i
], 2, 2, 0);
202 for (i
= 0; i
< q
->num_subpackets
; i
++) {
203 if (q
->subpacket
[i
].joint_stereo
== 1) {
204 result
|= init_vlc(&q
->subpacket
[i
].channel_coupling
, 6,
205 (1 << q
->subpacket
[i
].js_vlc_bits
) - 1,
206 ccpl_huffbits
[q
->subpacket
[i
].js_vlc_bits
- 2], 1, 1,
207 ccpl_huffcodes
[q
->subpacket
[i
].js_vlc_bits
- 2], 2, 2, 0);
208 av_log(q
->avctx
, AV_LOG_DEBUG
, "subpacket %i Joint-stereo VLC used.\n", i
);
212 av_log(q
->avctx
, AV_LOG_DEBUG
, "VLC tables initialized.\n");
216 static av_cold
int init_cook_mlt(COOKContext
*q
)
219 int mlt_size
= q
->samples_per_channel
;
221 if ((q
->mlt_window
= av_malloc(mlt_size
* sizeof(*q
->mlt_window
))) == 0)
222 return AVERROR(ENOMEM
);
224 /* Initialize the MLT window: simple sine window. */
225 ff_sine_window_init(q
->mlt_window
, mlt_size
);
226 for (j
= 0; j
< mlt_size
; j
++)
227 q
->mlt_window
[j
] *= sqrt(2.0 / q
->samples_per_channel
);
229 /* Initialize the MDCT. */
230 if ((ret
= ff_mdct_init(&q
->mdct_ctx
, av_log2(mlt_size
) + 1, 1, 1.0 / 32768.0))) {
231 av_free(q
->mlt_window
);
234 av_log(q
->avctx
, AV_LOG_DEBUG
, "MDCT initialized, order = %d.\n",
235 av_log2(mlt_size
) + 1);
240 static av_cold
void init_cplscales_table(COOKContext
*q
)
243 for (i
= 0; i
< 5; i
++)
244 q
->cplscales
[i
] = cplscales
[i
];
247 /*************** init functions end ***********/
249 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
250 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
253 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
254 * Why? No idea, some checksum/error detection method maybe.
256 * Out buffer size: extra bytes are needed to cope with
257 * padding/misalignment.
258 * Subpackets passed to the decoder can contain two, consecutive
259 * half-subpackets, of identical but arbitrary size.
260 * 1234 1234 1234 1234 extraA extraB
261 * Case 1: AAAA BBBB 0 0
262 * Case 2: AAAA ABBB BB-- 3 3
263 * Case 3: AAAA AABB BBBB 2 2
264 * Case 4: AAAA AAAB BBBB BB-- 1 5
266 * Nice way to waste CPU cycles.
268 * @param inbuffer pointer to byte array of indata
269 * @param out pointer to byte array of outdata
270 * @param bytes number of bytes
272 static inline int decode_bytes(const uint8_t *inbuffer
, uint8_t *out
, int bytes
)
274 static const uint32_t tab
[4] = {
275 AV_BE2NE32C(0x37c511f2u
), AV_BE2NE32C(0xf237c511u
),
276 AV_BE2NE32C(0x11f237c5u
), AV_BE2NE32C(0xc511f237u
),
281 uint32_t *obuf
= (uint32_t *) out
;
282 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
283 * I'm too lazy though, should be something like
284 * for (i = 0; i < bitamount / 64; i++)
285 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
286 * Buffer alignment needs to be checked. */
288 off
= (intptr_t) inbuffer
& 3;
289 buf
= (const uint32_t *) (inbuffer
- off
);
292 for (i
= 0; i
< bytes
/ 4; i
++)
293 obuf
[i
] = c
^ buf
[i
];
298 static av_cold
int cook_decode_close(AVCodecContext
*avctx
)
301 COOKContext
*q
= avctx
->priv_data
;
302 av_log(avctx
, AV_LOG_DEBUG
, "Deallocating memory.\n");
304 /* Free allocated memory buffers. */
305 av_free(q
->mlt_window
);
306 av_free(q
->decoded_bytes_buffer
);
308 /* Free the transform. */
309 ff_mdct_end(&q
->mdct_ctx
);
311 /* Free the VLC tables. */
312 for (i
= 0; i
< 13; i
++)
313 ff_free_vlc(&q
->envelope_quant_index
[i
]);
314 for (i
= 0; i
< 7; i
++)
315 ff_free_vlc(&q
->sqvh
[i
]);
316 for (i
= 0; i
< q
->num_subpackets
; i
++)
317 ff_free_vlc(&q
->subpacket
[i
].channel_coupling
);
319 av_log(avctx
, AV_LOG_DEBUG
, "Memory deallocated.\n");
325 * Fill the gain array for the timedomain quantization.
327 * @param gb pointer to the GetBitContext
328 * @param gaininfo array[9] of gain indexes
330 static void decode_gain_info(GetBitContext
*gb
, int *gaininfo
)
334 while (get_bits1(gb
)) {
338 n
= get_bits_count(gb
) - 1; // amount of elements*2 to update
342 int index
= get_bits(gb
, 3);
343 int gain
= get_bits1(gb
) ? get_bits(gb
, 4) - 7 : -1;
346 gaininfo
[i
++] = gain
;
353 * Create the quant index table needed for the envelope.
355 * @param q pointer to the COOKContext
356 * @param quant_index_table pointer to the array
358 static int decode_envelope(COOKContext
*q
, COOKSubpacket
*p
,
359 int *quant_index_table
)
363 quant_index_table
[0] = get_bits(&q
->gb
, 6) - 6; // This is used later in categorize
365 for (i
= 1; i
< p
->total_subbands
; i
++) {
367 if (i
>= p
->js_subband_start
* 2) {
368 vlc_index
-= p
->js_subband_start
;
375 vlc_index
= 13; // the VLC tables >13 are identical to No. 13
377 j
= get_vlc2(&q
->gb
, q
->envelope_quant_index
[vlc_index
- 1].table
,
378 q
->envelope_quant_index
[vlc_index
- 1].bits
, 2);
379 quant_index_table
[i
] = quant_index_table
[i
- 1] + j
- 12; // differential encoding
380 if (quant_index_table
[i
] > 63 || quant_index_table
[i
] < -63) {
381 av_log(q
->avctx
, AV_LOG_ERROR
,
382 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
383 quant_index_table
[i
], i
);
384 return AVERROR_INVALIDDATA
;
392 * Calculate the category and category_index vector.
394 * @param q pointer to the COOKContext
395 * @param quant_index_table pointer to the array
396 * @param category pointer to the category array
397 * @param category_index pointer to the category_index array
399 static void categorize(COOKContext
*q
, COOKSubpacket
*p
, int *quant_index_table
,
400 int *category
, int *category_index
)
402 int exp_idx
, bias
, tmpbias1
, tmpbias2
, bits_left
, num_bits
, index
, v
, i
, j
;
403 int exp_index2
[102] = { 0 };
404 int exp_index1
[102] = { 0 };
406 int tmp_categorize_array
[128 * 2] = { 0 };
407 int tmp_categorize_array1_idx
= p
->numvector_size
;
408 int tmp_categorize_array2_idx
= p
->numvector_size
;
410 bits_left
= p
->bits_per_subpacket
- get_bits_count(&q
->gb
);
412 if (bits_left
> q
->samples_per_channel
)
413 bits_left
= q
->samples_per_channel
+
414 ((bits_left
- q
->samples_per_channel
) * 5) / 8;
419 for (i
= 32; i
> 0; i
= i
/ 2) {
422 for (j
= p
->total_subbands
; j
> 0; j
--) {
423 exp_idx
= av_clip((i
- quant_index_table
[index
] + bias
) / 2, 0, 7);
425 num_bits
+= expbits_tab
[exp_idx
];
427 if (num_bits
>= bits_left
- 32)
431 /* Calculate total number of bits. */
433 for (i
= 0; i
< p
->total_subbands
; i
++) {
434 exp_idx
= av_clip((bias
- quant_index_table
[i
]) / 2, 0, 7);
435 num_bits
+= expbits_tab
[exp_idx
];
436 exp_index1
[i
] = exp_idx
;
437 exp_index2
[i
] = exp_idx
;
439 tmpbias1
= tmpbias2
= num_bits
;
441 for (j
= 1; j
< p
->numvector_size
; j
++) {
442 if (tmpbias1
+ tmpbias2
> 2 * bits_left
) { /* ---> */
445 for (i
= 0; i
< p
->total_subbands
; i
++) {
446 if (exp_index1
[i
] < 7) {
447 v
= (-2 * exp_index1
[i
]) - quant_index_table
[i
] + bias
;
456 tmp_categorize_array
[tmp_categorize_array1_idx
++] = index
;
457 tmpbias1
-= expbits_tab
[exp_index1
[index
]] -
458 expbits_tab
[exp_index1
[index
] + 1];
463 for (i
= 0; i
< p
->total_subbands
; i
++) {
464 if (exp_index2
[i
] > 0) {
465 v
= (-2 * exp_index2
[i
]) - quant_index_table
[i
] + bias
;
474 tmp_categorize_array
[--tmp_categorize_array2_idx
] = index
;
475 tmpbias2
-= expbits_tab
[exp_index2
[index
]] -
476 expbits_tab
[exp_index2
[index
] - 1];
481 for (i
= 0; i
< p
->total_subbands
; i
++)
482 category
[i
] = exp_index2
[i
];
484 for (i
= 0; i
< p
->numvector_size
- 1; i
++)
485 category_index
[i
] = tmp_categorize_array
[tmp_categorize_array2_idx
++];
490 * Expand the category vector.
492 * @param q pointer to the COOKContext
493 * @param category pointer to the category array
494 * @param category_index pointer to the category_index array
496 static inline void expand_category(COOKContext
*q
, int *category
,
500 for (i
= 0; i
< q
->num_vectors
; i
++)
502 int idx
= category_index
[i
];
503 if (++category
[idx
] >= FF_ARRAY_ELEMS(dither_tab
))
509 * The real requantization of the mltcoefs
511 * @param q pointer to the COOKContext
513 * @param quant_index quantisation index
514 * @param subband_coef_index array of indexes to quant_centroid_tab
515 * @param subband_coef_sign signs of coefficients
516 * @param mlt_p pointer into the mlt buffer
518 static void scalar_dequant_float(COOKContext
*q
, int index
, int quant_index
,
519 int *subband_coef_index
, int *subband_coef_sign
,
525 for (i
= 0; i
< SUBBAND_SIZE
; i
++) {
526 if (subband_coef_index
[i
]) {
527 f1
= quant_centroid_tab
[index
][subband_coef_index
[i
]];
528 if (subband_coef_sign
[i
])
531 /* noise coding if subband_coef_index[i] == 0 */
532 f1
= dither_tab
[index
];
533 if (av_lfg_get(&q
->random_state
) < 0x80000000)
536 mlt_p
[i
] = f1
* rootpow2tab
[quant_index
+ 63];
540 * Unpack the subband_coef_index and subband_coef_sign vectors.
542 * @param q pointer to the COOKContext
543 * @param category pointer to the category array
544 * @param subband_coef_index array of indexes to quant_centroid_tab
545 * @param subband_coef_sign signs of coefficients
547 static int unpack_SQVH(COOKContext
*q
, COOKSubpacket
*p
, int category
,
548 int *subband_coef_index
, int *subband_coef_sign
)
551 int vlc
, vd
, tmp
, result
;
553 vd
= vd_tab
[category
];
555 for (i
= 0; i
< vpr_tab
[category
]; i
++) {
556 vlc
= get_vlc2(&q
->gb
, q
->sqvh
[category
].table
, q
->sqvh
[category
].bits
, 3);
557 if (p
->bits_per_subpacket
< get_bits_count(&q
->gb
)) {
561 for (j
= vd
- 1; j
>= 0; j
--) {
562 tmp
= (vlc
* invradix_tab
[category
]) / 0x100000;
563 subband_coef_index
[vd
* i
+ j
] = vlc
- tmp
* (kmax_tab
[category
] + 1);
566 for (j
= 0; j
< vd
; j
++) {
567 if (subband_coef_index
[i
* vd
+ j
]) {
568 if (get_bits_count(&q
->gb
) < p
->bits_per_subpacket
) {
569 subband_coef_sign
[i
* vd
+ j
] = get_bits1(&q
->gb
);
572 subband_coef_sign
[i
* vd
+ j
] = 0;
575 subband_coef_sign
[i
* vd
+ j
] = 0;
584 * Fill the mlt_buffer with mlt coefficients.
586 * @param q pointer to the COOKContext
587 * @param category pointer to the category array
588 * @param quant_index_table pointer to the array
589 * @param mlt_buffer pointer to mlt coefficients
591 static void decode_vectors(COOKContext
*q
, COOKSubpacket
*p
, int *category
,
592 int *quant_index_table
, float *mlt_buffer
)
594 /* A zero in this table means that the subband coefficient is
595 random noise coded. */
596 int subband_coef_index
[SUBBAND_SIZE
];
597 /* A zero in this table means that the subband coefficient is a
598 positive multiplicator. */
599 int subband_coef_sign
[SUBBAND_SIZE
];
603 for (band
= 0; band
< p
->total_subbands
; band
++) {
604 index
= category
[band
];
605 if (category
[band
] < 7) {
606 if (unpack_SQVH(q
, p
, category
[band
], subband_coef_index
, subband_coef_sign
)) {
608 for (j
= 0; j
< p
->total_subbands
; j
++)
609 category
[band
+ j
] = 7;
613 memset(subband_coef_index
, 0, sizeof(subband_coef_index
));
614 memset(subband_coef_sign
, 0, sizeof(subband_coef_sign
));
616 q
->scalar_dequant(q
, index
, quant_index_table
[band
],
617 subband_coef_index
, subband_coef_sign
,
618 &mlt_buffer
[band
* SUBBAND_SIZE
]);
621 /* FIXME: should this be removed, or moved into loop above? */
622 if (p
->total_subbands
* SUBBAND_SIZE
>= q
->samples_per_channel
)
627 static int mono_decode(COOKContext
*q
, COOKSubpacket
*p
, float *mlt_buffer
)
629 int category_index
[128] = { 0 };
630 int category
[128] = { 0 };
631 int quant_index_table
[102];
634 if ((res
= decode_envelope(q
, p
, quant_index_table
)) < 0)
636 q
->num_vectors
= get_bits(&q
->gb
, p
->log2_numvector_size
);
637 categorize(q
, p
, quant_index_table
, category
, category_index
);
638 expand_category(q
, category
, category_index
);
639 decode_vectors(q
, p
, category
, quant_index_table
, mlt_buffer
);
646 * the actual requantization of the timedomain samples
648 * @param q pointer to the COOKContext
649 * @param buffer pointer to the timedomain buffer
650 * @param gain_index index for the block multiplier
651 * @param gain_index_next index for the next block multiplier
653 static void interpolate_float(COOKContext
*q
, float *buffer
,
654 int gain_index
, int gain_index_next
)
658 fc1
= pow2tab
[gain_index
+ 63];
660 if (gain_index
== gain_index_next
) { // static gain
661 for (i
= 0; i
< q
->gain_size_factor
; i
++)
663 } else { // smooth gain
664 fc2
= q
->gain_table
[11 + (gain_index_next
- gain_index
)];
665 for (i
= 0; i
< q
->gain_size_factor
; i
++) {
673 * Apply transform window, overlap buffers.
675 * @param q pointer to the COOKContext
676 * @param inbuffer pointer to the mltcoefficients
677 * @param gains_ptr current and previous gains
678 * @param previous_buffer pointer to the previous buffer to be used for overlapping
680 static void imlt_window_float(COOKContext
*q
, float *inbuffer
,
681 cook_gains
*gains_ptr
, float *previous_buffer
)
683 const float fc
= pow2tab
[gains_ptr
->previous
[0] + 63];
685 /* The weird thing here, is that the two halves of the time domain
686 * buffer are swapped. Also, the newest data, that we save away for
687 * next frame, has the wrong sign. Hence the subtraction below.
688 * Almost sounds like a complex conjugate/reverse data/FFT effect.
691 /* Apply window and overlap */
692 for (i
= 0; i
< q
->samples_per_channel
; i
++)
693 inbuffer
[i
] = inbuffer
[i
] * fc
* q
->mlt_window
[i
] -
694 previous_buffer
[i
] * q
->mlt_window
[q
->samples_per_channel
- 1 - i
];
698 * The modulated lapped transform, this takes transform coefficients
699 * and transforms them into timedomain samples.
700 * Apply transform window, overlap buffers, apply gain profile
701 * and buffer management.
703 * @param q pointer to the COOKContext
704 * @param inbuffer pointer to the mltcoefficients
705 * @param gains_ptr current and previous gains
706 * @param previous_buffer pointer to the previous buffer to be used for overlapping
708 static void imlt_gain(COOKContext
*q
, float *inbuffer
,
709 cook_gains
*gains_ptr
, float *previous_buffer
)
711 float *buffer0
= q
->mono_mdct_output
;
712 float *buffer1
= q
->mono_mdct_output
+ q
->samples_per_channel
;
715 /* Inverse modified discrete cosine transform */
716 q
->mdct_ctx
.imdct_calc(&q
->mdct_ctx
, q
->mono_mdct_output
, inbuffer
);
718 q
->imlt_window(q
, buffer1
, gains_ptr
, previous_buffer
);
720 /* Apply gain profile */
721 for (i
= 0; i
< 8; i
++)
722 if (gains_ptr
->now
[i
] || gains_ptr
->now
[i
+ 1])
723 q
->interpolate(q
, &buffer1
[q
->gain_size_factor
* i
],
724 gains_ptr
->now
[i
], gains_ptr
->now
[i
+ 1]);
726 /* Save away the current to be previous block. */
727 memcpy(previous_buffer
, buffer0
,
728 q
->samples_per_channel
* sizeof(*previous_buffer
));
733 * function for getting the jointstereo coupling information
735 * @param q pointer to the COOKContext
736 * @param decouple_tab decoupling array
738 static void decouple_info(COOKContext
*q
, COOKSubpacket
*p
, int *decouple_tab
)
741 int vlc
= get_bits1(&q
->gb
);
742 int start
= cplband
[p
->js_subband_start
];
743 int end
= cplband
[p
->subbands
- 1];
744 int length
= end
- start
+ 1;
750 for (i
= 0; i
< length
; i
++)
751 decouple_tab
[start
+ i
] = get_vlc2(&q
->gb
,
752 p
->channel_coupling
.table
,
753 p
->channel_coupling
.bits
, 2);
755 for (i
= 0; i
< length
; i
++)
756 decouple_tab
[start
+ i
] = get_bits(&q
->gb
, p
->js_vlc_bits
);
760 * function decouples a pair of signals from a single signal via multiplication.
762 * @param q pointer to the COOKContext
763 * @param subband index of the current subband
764 * @param f1 multiplier for channel 1 extraction
765 * @param f2 multiplier for channel 2 extraction
766 * @param decode_buffer input buffer
767 * @param mlt_buffer1 pointer to left channel mlt coefficients
768 * @param mlt_buffer2 pointer to right channel mlt coefficients
770 static void decouple_float(COOKContext
*q
,
774 float *decode_buffer
,
775 float *mlt_buffer1
, float *mlt_buffer2
)
778 for (j
= 0; j
< SUBBAND_SIZE
; j
++) {
779 tmp_idx
= ((p
->js_subband_start
+ subband
) * SUBBAND_SIZE
) + j
;
780 mlt_buffer1
[SUBBAND_SIZE
* subband
+ j
] = f1
* decode_buffer
[tmp_idx
];
781 mlt_buffer2
[SUBBAND_SIZE
* subband
+ j
] = f2
* decode_buffer
[tmp_idx
];
786 * function for decoding joint stereo data
788 * @param q pointer to the COOKContext
789 * @param mlt_buffer1 pointer to left channel mlt coefficients
790 * @param mlt_buffer2 pointer to right channel mlt coefficients
792 static int joint_decode(COOKContext
*q
, COOKSubpacket
*p
,
793 float *mlt_buffer_left
, float *mlt_buffer_right
)
796 int decouple_tab
[SUBBAND_SIZE
] = { 0 };
797 float *decode_buffer
= q
->decode_buffer_0
;
800 const float *cplscale
;
802 memset(decode_buffer
, 0, sizeof(q
->decode_buffer_0
));
804 /* Make sure the buffers are zeroed out. */
805 memset(mlt_buffer_left
, 0, 1024 * sizeof(*mlt_buffer_left
));
806 memset(mlt_buffer_right
, 0, 1024 * sizeof(*mlt_buffer_right
));
807 decouple_info(q
, p
, decouple_tab
);
808 if ((res
= mono_decode(q
, p
, decode_buffer
)) < 0)
811 /* The two channels are stored interleaved in decode_buffer. */
812 for (i
= 0; i
< p
->js_subband_start
; i
++) {
813 for (j
= 0; j
< SUBBAND_SIZE
; j
++) {
814 mlt_buffer_left
[i
* 20 + j
] = decode_buffer
[i
* 40 + j
];
815 mlt_buffer_right
[i
* 20 + j
] = decode_buffer
[i
* 40 + 20 + j
];
819 /* When we reach js_subband_start (the higher frequencies)
820 the coefficients are stored in a coupling scheme. */
821 idx
= (1 << p
->js_vlc_bits
) - 1;
822 for (i
= p
->js_subband_start
; i
< p
->subbands
; i
++) {
823 cpl_tmp
= cplband
[i
];
824 idx
-= decouple_tab
[cpl_tmp
];
825 cplscale
= q
->cplscales
[p
->js_vlc_bits
- 2]; // choose decoupler table
826 f1
= cplscale
[decouple_tab
[cpl_tmp
] + 1];
828 q
->decouple(q
, p
, i
, f1
, f2
, decode_buffer
,
829 mlt_buffer_left
, mlt_buffer_right
);
830 idx
= (1 << p
->js_vlc_bits
) - 1;
837 * First part of subpacket decoding:
838 * decode raw stream bytes and read gain info.
840 * @param q pointer to the COOKContext
841 * @param inbuffer pointer to raw stream data
842 * @param gains_ptr array of current/prev gain pointers
844 static inline void decode_bytes_and_gain(COOKContext
*q
, COOKSubpacket
*p
,
845 const uint8_t *inbuffer
,
846 cook_gains
*gains_ptr
)
850 offset
= decode_bytes(inbuffer
, q
->decoded_bytes_buffer
,
851 p
->bits_per_subpacket
/ 8);
852 init_get_bits(&q
->gb
, q
->decoded_bytes_buffer
+ offset
,
853 p
->bits_per_subpacket
);
854 decode_gain_info(&q
->gb
, gains_ptr
->now
);
856 /* Swap current and previous gains */
857 FFSWAP(int *, gains_ptr
->now
, gains_ptr
->previous
);
861 * Saturate the output signal and interleave.
863 * @param q pointer to the COOKContext
864 * @param out pointer to the output vector
866 static void saturate_output_float(COOKContext
*q
, float *out
)
868 q
->dsp
.vector_clipf(out
, q
->mono_mdct_output
+ q
->samples_per_channel
,
869 -1.0f
, 1.0f
, FFALIGN(q
->samples_per_channel
, 8));
874 * Final part of subpacket decoding:
875 * Apply modulated lapped transform, gain compensation,
876 * clip and convert to integer.
878 * @param q pointer to the COOKContext
879 * @param decode_buffer pointer to the mlt coefficients
880 * @param gains_ptr array of current/prev gain pointers
881 * @param previous_buffer pointer to the previous buffer to be used for overlapping
882 * @param out pointer to the output buffer
884 static inline void mlt_compensate_output(COOKContext
*q
, float *decode_buffer
,
885 cook_gains
*gains_ptr
, float *previous_buffer
,
888 imlt_gain(q
, decode_buffer
, gains_ptr
, previous_buffer
);
890 q
->saturate_output(q
, out
);
895 * Cook subpacket decoding. This function returns one decoded subpacket,
896 * usually 1024 samples per channel.
898 * @param q pointer to the COOKContext
899 * @param inbuffer pointer to the inbuffer
900 * @param outbuffer pointer to the outbuffer
902 static int decode_subpacket(COOKContext
*q
, COOKSubpacket
*p
,
903 const uint8_t *inbuffer
, float **outbuffer
)
905 int sub_packet_size
= p
->size
;
908 memset(q
->decode_buffer_1
, 0, sizeof(q
->decode_buffer_1
));
909 decode_bytes_and_gain(q
, p
, inbuffer
, &p
->gains1
);
911 if (p
->joint_stereo
) {
912 if ((res
= joint_decode(q
, p
, q
->decode_buffer_1
, q
->decode_buffer_2
)) < 0)
915 if ((res
= mono_decode(q
, p
, q
->decode_buffer_1
)) < 0)
918 if (p
->num_channels
== 2) {
919 decode_bytes_and_gain(q
, p
, inbuffer
+ sub_packet_size
/ 2, &p
->gains2
);
920 if ((res
= mono_decode(q
, p
, q
->decode_buffer_2
)) < 0)
925 mlt_compensate_output(q
, q
->decode_buffer_1
, &p
->gains1
,
926 p
->mono_previous_buffer1
,
927 outbuffer
? outbuffer
[p
->ch_idx
] : NULL
);
929 if (p
->num_channels
== 2)
931 mlt_compensate_output(q
, q
->decode_buffer_2
, &p
->gains1
,
932 p
->mono_previous_buffer2
,
933 outbuffer
? outbuffer
[p
->ch_idx
+ 1] : NULL
);
935 mlt_compensate_output(q
, q
->decode_buffer_2
, &p
->gains2
,
936 p
->mono_previous_buffer2
,
937 outbuffer
? outbuffer
[p
->ch_idx
+ 1] : NULL
);
943 static int cook_decode_frame(AVCodecContext
*avctx
, void *data
,
944 int *got_frame_ptr
, AVPacket
*avpkt
)
946 AVFrame
*frame
= data
;
947 const uint8_t *buf
= avpkt
->data
;
948 int buf_size
= avpkt
->size
;
949 COOKContext
*q
= avctx
->priv_data
;
950 float **samples
= NULL
;
955 if (buf_size
< avctx
->block_align
)
958 /* get output buffer */
959 if (q
->discarded_packets
>= 2) {
960 frame
->nb_samples
= q
->samples_per_channel
;
961 if ((ret
= ff_get_buffer(avctx
, frame
, 0)) < 0) {
962 av_log(avctx
, AV_LOG_ERROR
, "get_buffer() failed\n");
965 samples
= (float **)frame
->extended_data
;
968 /* estimate subpacket sizes */
969 q
->subpacket
[0].size
= avctx
->block_align
;
971 for (i
= 1; i
< q
->num_subpackets
; i
++) {
972 q
->subpacket
[i
].size
= 2 * buf
[avctx
->block_align
- q
->num_subpackets
+ i
];
973 q
->subpacket
[0].size
-= q
->subpacket
[i
].size
+ 1;
974 if (q
->subpacket
[0].size
< 0) {
975 av_log(avctx
, AV_LOG_DEBUG
,
976 "frame subpacket size total > avctx->block_align!\n");
977 return AVERROR_INVALIDDATA
;
981 /* decode supbackets */
982 for (i
= 0; i
< q
->num_subpackets
; i
++) {
983 q
->subpacket
[i
].bits_per_subpacket
= (q
->subpacket
[i
].size
* 8) >>
984 q
->subpacket
[i
].bits_per_subpdiv
;
985 q
->subpacket
[i
].ch_idx
= chidx
;
986 av_log(avctx
, AV_LOG_DEBUG
,
987 "subpacket[%i] size %i js %i %i block_align %i\n",
988 i
, q
->subpacket
[i
].size
, q
->subpacket
[i
].joint_stereo
, offset
,
991 if ((ret
= decode_subpacket(q
, &q
->subpacket
[i
], buf
+ offset
, samples
)) < 0)
993 offset
+= q
->subpacket
[i
].size
;
994 chidx
+= q
->subpacket
[i
].num_channels
;
995 av_log(avctx
, AV_LOG_DEBUG
, "subpacket[%i] %i %i\n",
996 i
, q
->subpacket
[i
].size
* 8, get_bits_count(&q
->gb
));
999 /* Discard the first two frames: no valid audio. */
1000 if (q
->discarded_packets
< 2) {
1001 q
->discarded_packets
++;
1003 return avctx
->block_align
;
1008 return avctx
->block_align
;
1012 static void dump_cook_context(COOKContext
*q
)
1015 #define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
1016 av_dlog(q
->avctx
, "COOKextradata\n");
1017 av_dlog(q
->avctx
, "cookversion=%x\n", q
->subpacket
[0].cookversion
);
1018 if (q
->subpacket
[0].cookversion
> STEREO
) {
1019 PRINT("js_subband_start", q
->subpacket
[0].js_subband_start
);
1020 PRINT("js_vlc_bits", q
->subpacket
[0].js_vlc_bits
);
1022 av_dlog(q
->avctx
, "COOKContext\n");
1023 PRINT("nb_channels", q
->avctx
->channels
);
1024 PRINT("bit_rate", q
->avctx
->bit_rate
);
1025 PRINT("sample_rate", q
->avctx
->sample_rate
);
1026 PRINT("samples_per_channel", q
->subpacket
[0].samples_per_channel
);
1027 PRINT("subbands", q
->subpacket
[0].subbands
);
1028 PRINT("js_subband_start", q
->subpacket
[0].js_subband_start
);
1029 PRINT("log2_numvector_size", q
->subpacket
[0].log2_numvector_size
);
1030 PRINT("numvector_size", q
->subpacket
[0].numvector_size
);
1031 PRINT("total_subbands", q
->subpacket
[0].total_subbands
);
1036 * Cook initialization
1038 * @param avctx pointer to the AVCodecContext
1040 static av_cold
int cook_decode_init(AVCodecContext
*avctx
)
1042 COOKContext
*q
= avctx
->priv_data
;
1043 const uint8_t *edata_ptr
= avctx
->extradata
;
1044 const uint8_t *edata_ptr_end
= edata_ptr
+ avctx
->extradata_size
;
1045 int extradata_size
= avctx
->extradata_size
;
1047 unsigned int channel_mask
= 0;
1048 int samples_per_frame
;
1052 /* Take care of the codec specific extradata. */
1053 if (extradata_size
<= 0) {
1054 av_log(avctx
, AV_LOG_ERROR
, "Necessary extradata missing!\n");
1055 return AVERROR_INVALIDDATA
;
1057 av_log(avctx
, AV_LOG_DEBUG
, "codecdata_length=%d\n", avctx
->extradata_size
);
1059 /* Take data from the AVCodecContext (RM container). */
1060 if (!avctx
->channels
) {
1061 av_log(avctx
, AV_LOG_ERROR
, "Invalid number of channels\n");
1062 return AVERROR_INVALIDDATA
;
1065 /* Initialize RNG. */
1066 av_lfg_init(&q
->random_state
, 0);
1068 ff_dsputil_init(&q
->dsp
, avctx
);
1070 while (edata_ptr
< edata_ptr_end
) {
1071 /* 8 for mono, 16 for stereo, ? for multichannel
1072 Swap to right endianness so we don't need to care later on. */
1073 if (extradata_size
>= 8) {
1074 q
->subpacket
[s
].cookversion
= bytestream_get_be32(&edata_ptr
);
1075 samples_per_frame
= bytestream_get_be16(&edata_ptr
);
1076 q
->subpacket
[s
].subbands
= bytestream_get_be16(&edata_ptr
);
1077 extradata_size
-= 8;
1079 if (extradata_size
>= 8) {
1080 bytestream_get_be32(&edata_ptr
); // Unknown unused
1081 q
->subpacket
[s
].js_subband_start
= bytestream_get_be16(&edata_ptr
);
1082 q
->subpacket
[s
].js_vlc_bits
= bytestream_get_be16(&edata_ptr
);
1083 extradata_size
-= 8;
1086 /* Initialize extradata related variables. */
1087 q
->subpacket
[s
].samples_per_channel
= samples_per_frame
/ avctx
->channels
;
1088 q
->subpacket
[s
].bits_per_subpacket
= avctx
->block_align
* 8;
1090 /* Initialize default data states. */
1091 q
->subpacket
[s
].log2_numvector_size
= 5;
1092 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
;
1093 q
->subpacket
[s
].num_channels
= 1;
1095 /* Initialize version-dependent variables */
1097 av_log(avctx
, AV_LOG_DEBUG
, "subpacket[%i].cookversion=%x\n", s
,
1098 q
->subpacket
[s
].cookversion
);
1099 q
->subpacket
[s
].joint_stereo
= 0;
1100 switch (q
->subpacket
[s
].cookversion
) {
1102 if (avctx
->channels
!= 1) {
1103 av_log_ask_for_sample(avctx
, "Container channels != 1.\n");
1104 return AVERROR_PATCHWELCOME
;
1106 av_log(avctx
, AV_LOG_DEBUG
, "MONO\n");
1109 if (avctx
->channels
!= 1) {
1110 q
->subpacket
[s
].bits_per_subpdiv
= 1;
1111 q
->subpacket
[s
].num_channels
= 2;
1113 av_log(avctx
, AV_LOG_DEBUG
, "STEREO\n");
1116 if (avctx
->channels
!= 2) {
1117 av_log_ask_for_sample(avctx
, "Container channels != 2.\n");
1118 return AVERROR_PATCHWELCOME
;
1120 av_log(avctx
, AV_LOG_DEBUG
, "JOINT_STEREO\n");
1121 if (avctx
->extradata_size
>= 16) {
1122 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
+
1123 q
->subpacket
[s
].js_subband_start
;
1124 q
->subpacket
[s
].joint_stereo
= 1;
1125 q
->subpacket
[s
].num_channels
= 2;
1127 if (q
->subpacket
[s
].samples_per_channel
> 256) {
1128 q
->subpacket
[s
].log2_numvector_size
= 6;
1130 if (q
->subpacket
[s
].samples_per_channel
> 512) {
1131 q
->subpacket
[s
].log2_numvector_size
= 7;
1135 av_log(avctx
, AV_LOG_DEBUG
, "MULTI_CHANNEL\n");
1136 if (extradata_size
>= 4)
1137 channel_mask
|= q
->subpacket
[s
].channel_mask
= bytestream_get_be32(&edata_ptr
);
1139 if (av_get_channel_layout_nb_channels(q
->subpacket
[s
].channel_mask
) > 1) {
1140 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
+
1141 q
->subpacket
[s
].js_subband_start
;
1142 q
->subpacket
[s
].joint_stereo
= 1;
1143 q
->subpacket
[s
].num_channels
= 2;
1144 q
->subpacket
[s
].samples_per_channel
= samples_per_frame
>> 1;
1146 if (q
->subpacket
[s
].samples_per_channel
> 256) {
1147 q
->subpacket
[s
].log2_numvector_size
= 6;
1149 if (q
->subpacket
[s
].samples_per_channel
> 512) {
1150 q
->subpacket
[s
].log2_numvector_size
= 7;
1153 q
->subpacket
[s
].samples_per_channel
= samples_per_frame
;
1157 av_log_ask_for_sample(avctx
, "Unknown Cook version.\n");
1158 return AVERROR_PATCHWELCOME
;
1161 if (s
> 1 && q
->subpacket
[s
].samples_per_channel
!= q
->samples_per_channel
) {
1162 av_log(avctx
, AV_LOG_ERROR
, "different number of samples per channel!\n");
1163 return AVERROR_INVALIDDATA
;
1165 q
->samples_per_channel
= q
->subpacket
[0].samples_per_channel
;
1168 /* Initialize variable relations */
1169 q
->subpacket
[s
].numvector_size
= (1 << q
->subpacket
[s
].log2_numvector_size
);
1171 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1172 if (q
->subpacket
[s
].total_subbands
> 53) {
1173 av_log_ask_for_sample(avctx
, "total_subbands > 53\n");
1174 return AVERROR_PATCHWELCOME
;
1177 if ((q
->subpacket
[s
].js_vlc_bits
> 6) ||
1178 (q
->subpacket
[s
].js_vlc_bits
< 2 * q
->subpacket
[s
].joint_stereo
)) {
1179 av_log(avctx
, AV_LOG_ERROR
, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1180 q
->subpacket
[s
].js_vlc_bits
, 2 * q
->subpacket
[s
].joint_stereo
);
1181 return AVERROR_INVALIDDATA
;
1184 if (q
->subpacket
[s
].subbands
> 50) {
1185 av_log_ask_for_sample(avctx
, "subbands > 50\n");
1186 return AVERROR_PATCHWELCOME
;
1188 q
->subpacket
[s
].gains1
.now
= q
->subpacket
[s
].gain_1
;
1189 q
->subpacket
[s
].gains1
.previous
= q
->subpacket
[s
].gain_2
;
1190 q
->subpacket
[s
].gains2
.now
= q
->subpacket
[s
].gain_3
;
1191 q
->subpacket
[s
].gains2
.previous
= q
->subpacket
[s
].gain_4
;
1193 q
->num_subpackets
++;
1195 if (s
> MAX_SUBPACKETS
) {
1196 av_log_ask_for_sample(avctx
, "Too many subpackets > 5\n");
1197 return AVERROR_PATCHWELCOME
;
1200 /* Generate tables */
1203 init_cplscales_table(q
);
1205 if ((ret
= init_cook_vlc_tables(q
)))
1209 if (avctx
->block_align
>= UINT_MAX
/ 2)
1210 return AVERROR(EINVAL
);
1212 /* Pad the databuffer with:
1213 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1214 FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1215 q
->decoded_bytes_buffer
=
1216 av_mallocz(avctx
->block_align
1217 + DECODE_BYTES_PAD1(avctx
->block_align
)
1218 + FF_INPUT_BUFFER_PADDING_SIZE
);
1219 if (q
->decoded_bytes_buffer
== NULL
)
1220 return AVERROR(ENOMEM
);
1222 /* Initialize transform. */
1223 if ((ret
= init_cook_mlt(q
)))
1226 /* Initialize COOK signal arithmetic handling */
1228 q
->scalar_dequant
= scalar_dequant_float
;
1229 q
->decouple
= decouple_float
;
1230 q
->imlt_window
= imlt_window_float
;
1231 q
->interpolate
= interpolate_float
;
1232 q
->saturate_output
= saturate_output_float
;
1235 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1236 if (q
->samples_per_channel
!= 256 && q
->samples_per_channel
!= 512 &&
1237 q
->samples_per_channel
!= 1024) {
1238 av_log_ask_for_sample(avctx
,
1239 "unknown amount of samples_per_channel = %d\n",
1240 q
->samples_per_channel
);
1241 return AVERROR_PATCHWELCOME
;
1244 avctx
->sample_fmt
= AV_SAMPLE_FMT_FLTP
;
1246 avctx
->channel_layout
= channel_mask
;
1248 avctx
->channel_layout
= (avctx
->channels
== 2) ? AV_CH_LAYOUT_STEREO
: AV_CH_LAYOUT_MONO
;
1251 dump_cook_context(q
);
1256 AVCodec ff_cook_decoder
= {
1258 .type
= AVMEDIA_TYPE_AUDIO
,
1259 .id
= AV_CODEC_ID_COOK
,
1260 .priv_data_size
= sizeof(COOKContext
),
1261 .init
= cook_decode_init
,
1262 .close
= cook_decode_close
,
1263 .decode
= cook_decode_frame
,
1264 .capabilities
= CODEC_CAP_DR1
,
1265 .long_name
= NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1266 .sample_fmts
= (const enum AVSampleFormat
[]) { AV_SAMPLE_FMT_FLTP
,
1267 AV_SAMPLE_FMT_NONE
},