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
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"
51 #include "bytestream.h"
59 /* the different Cook versions */
60 #define MONO 0x1000001
61 #define STEREO 0x1000002
62 #define JOINT_STEREO 0x1000003
63 #define MC_COOK 0x2000000 // multichannel Cook, not supported
65 #define SUBBAND_SIZE 20
66 #define MAX_SUBPACKETS 5
81 int samples_per_channel
;
82 int log2_numvector_size
;
83 unsigned int channel_mask
;
86 int bits_per_subpacket
;
89 int numvector_size
; // 1 << log2_numvector_size;
91 float mono_previous_buffer1
[1024];
92 float mono_previous_buffer2
[1024];
102 typedef struct cook
{
104 * The following 5 functions provide the lowlevel arithmetic on
105 * the internal audio buffers.
107 void (*scalar_dequant
)(struct cook
*q
, int index
, int quant_index
,
108 int *subband_coef_index
, int *subband_coef_sign
,
111 void (*decouple
)(struct cook
*q
,
115 float *decode_buffer
,
116 float *mlt_buffer1
, float *mlt_buffer2
);
118 void (*imlt_window
)(struct cook
*q
, float *buffer1
,
119 cook_gains
*gains_ptr
, float *previous_buffer
);
121 void (*interpolate
)(struct cook
*q
, float *buffer
,
122 int gain_index
, int gain_index_next
);
124 void (*saturate_output
)(struct cook
*q
, float *out
);
126 AVCodecContext
* avctx
;
127 AudioDSPContext adsp
;
131 int samples_per_channel
;
134 int discarded_packets
;
141 VLC envelope_quant_index
[13];
142 VLC sqvh
[7]; // scalar quantization
144 /* generatable tables and related variables */
145 int gain_size_factor
;
146 float gain_table
[23];
150 uint8_t* decoded_bytes_buffer
;
151 DECLARE_ALIGNED(32, float, mono_mdct_output
)[2048];
152 float decode_buffer_1
[1024];
153 float decode_buffer_2
[1024];
154 float decode_buffer_0
[1060]; /* static allocation for joint decode */
156 const float *cplscales
[5];
158 COOKSubpacket subpacket
[MAX_SUBPACKETS
];
161 static float pow2tab
[127];
162 static float rootpow2tab
[127];
164 /*************** init functions ***************/
166 /* table generator */
167 static av_cold
void init_pow2table(void)
170 for (i
= -63; i
< 64; i
++) {
171 pow2tab
[63 + i
] = pow(2, i
);
172 rootpow2tab
[63 + i
] = sqrt(pow(2, i
));
176 /* table generator */
177 static av_cold
void init_gain_table(COOKContext
*q
)
180 q
->gain_size_factor
= q
->samples_per_channel
/ 8;
181 for (i
= 0; i
< 23; i
++)
182 q
->gain_table
[i
] = pow(pow2tab
[i
+ 52],
183 (1.0 / (double) q
->gain_size_factor
));
187 static av_cold
int init_cook_vlc_tables(COOKContext
*q
)
192 for (i
= 0; i
< 13; i
++) {
193 result
|= init_vlc(&q
->envelope_quant_index
[i
], 9, 24,
194 envelope_quant_index_huffbits
[i
], 1, 1,
195 envelope_quant_index_huffcodes
[i
], 2, 2, 0);
197 av_log(q
->avctx
, AV_LOG_DEBUG
, "sqvh VLC init\n");
198 for (i
= 0; i
< 7; i
++) {
199 result
|= init_vlc(&q
->sqvh
[i
], vhvlcsize_tab
[i
], vhsize_tab
[i
],
200 cvh_huffbits
[i
], 1, 1,
201 cvh_huffcodes
[i
], 2, 2, 0);
204 for (i
= 0; i
< q
->num_subpackets
; i
++) {
205 if (q
->subpacket
[i
].joint_stereo
== 1) {
206 result
|= init_vlc(&q
->subpacket
[i
].channel_coupling
, 6,
207 (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_array(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_freep(&q
->mlt_window
);
236 av_log(q
->avctx
, AV_LOG_DEBUG
, "MDCT initialized, order = %d.\n",
237 av_log2(mlt_size
) + 1);
242 static av_cold
void init_cplscales_table(COOKContext
*q
)
245 for (i
= 0; i
< 5; i
++)
246 q
->cplscales
[i
] = cplscales
[i
];
249 /*************** init functions end ***********/
251 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
252 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
255 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
256 * Why? No idea, some checksum/error detection method maybe.
258 * Out buffer size: extra bytes are needed to cope with
259 * padding/misalignment.
260 * Subpackets passed to the decoder can contain two, consecutive
261 * half-subpackets, of identical but arbitrary size.
262 * 1234 1234 1234 1234 extraA extraB
263 * Case 1: AAAA BBBB 0 0
264 * Case 2: AAAA ABBB BB-- 3 3
265 * Case 3: AAAA AABB BBBB 2 2
266 * Case 4: AAAA AAAB BBBB BB-- 1 5
268 * Nice way to waste CPU cycles.
270 * @param inbuffer pointer to byte array of indata
271 * @param out pointer to byte array of outdata
272 * @param bytes number of bytes
274 static inline int decode_bytes(const uint8_t *inbuffer
, uint8_t *out
, int bytes
)
276 static const uint32_t tab
[4] = {
277 AV_BE2NE32C(0x37c511f2u
), AV_BE2NE32C(0xf237c511u
),
278 AV_BE2NE32C(0x11f237c5u
), AV_BE2NE32C(0xc511f237u
),
283 uint32_t *obuf
= (uint32_t *) out
;
284 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
285 * I'm too lazy though, should be something like
286 * for (i = 0; i < bitamount / 64; i++)
287 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
288 * Buffer alignment needs to be checked. */
290 off
= (intptr_t) inbuffer
& 3;
291 buf
= (const uint32_t *) (inbuffer
- off
);
294 for (i
= 0; i
< bytes
/ 4; i
++)
295 obuf
[i
] = c
^ buf
[i
];
300 static av_cold
int cook_decode_close(AVCodecContext
*avctx
)
303 COOKContext
*q
= avctx
->priv_data
;
304 av_log(avctx
, AV_LOG_DEBUG
, "Deallocating memory.\n");
306 /* Free allocated memory buffers. */
307 av_freep(&q
->mlt_window
);
308 av_freep(&q
->decoded_bytes_buffer
);
310 /* Free the transform. */
311 ff_mdct_end(&q
->mdct_ctx
);
313 /* Free the VLC tables. */
314 for (i
= 0; i
< 13; i
++)
315 ff_free_vlc(&q
->envelope_quant_index
[i
]);
316 for (i
= 0; i
< 7; i
++)
317 ff_free_vlc(&q
->sqvh
[i
]);
318 for (i
= 0; i
< q
->num_subpackets
; i
++)
319 ff_free_vlc(&q
->subpacket
[i
].channel_coupling
);
321 av_log(avctx
, AV_LOG_DEBUG
, "Memory deallocated.\n");
327 * Fill the gain array for the timedomain quantization.
329 * @param gb pointer to the GetBitContext
330 * @param gaininfo array[9] of gain indexes
332 static void decode_gain_info(GetBitContext
*gb
, int *gaininfo
)
336 n
= get_unary(gb
, 0, get_bits_left(gb
)); // amount of elements*2 to update
340 int index
= get_bits(gb
, 3);
341 int gain
= get_bits1(gb
) ? get_bits(gb
, 4) - 7 : -1;
344 gaininfo
[i
++] = gain
;
351 * Create the quant index table needed for the envelope.
353 * @param q pointer to the COOKContext
354 * @param quant_index_table pointer to the array
356 static int decode_envelope(COOKContext
*q
, COOKSubpacket
*p
,
357 int *quant_index_table
)
361 quant_index_table
[0] = get_bits(&q
->gb
, 6) - 6; // This is used later in categorize
363 for (i
= 1; i
< p
->total_subbands
; i
++) {
365 if (i
>= p
->js_subband_start
* 2) {
366 vlc_index
-= p
->js_subband_start
;
373 vlc_index
= 13; // the VLC tables >13 are identical to No. 13
375 j
= get_vlc2(&q
->gb
, q
->envelope_quant_index
[vlc_index
- 1].table
,
376 q
->envelope_quant_index
[vlc_index
- 1].bits
, 2);
377 quant_index_table
[i
] = quant_index_table
[i
- 1] + j
- 12; // differential encoding
378 if (quant_index_table
[i
] > 63 || quant_index_table
[i
] < -63) {
379 av_log(q
->avctx
, AV_LOG_ERROR
,
380 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
381 quant_index_table
[i
], i
);
382 return AVERROR_INVALIDDATA
;
390 * Calculate the category and category_index vector.
392 * @param q pointer to the COOKContext
393 * @param quant_index_table pointer to the array
394 * @param category pointer to the category array
395 * @param category_index pointer to the category_index array
397 static void categorize(COOKContext
*q
, COOKSubpacket
*p
, const int *quant_index_table
,
398 int *category
, int *category_index
)
400 int exp_idx
, bias
, tmpbias1
, tmpbias2
, bits_left
, num_bits
, index
, v
, i
, j
;
401 int exp_index2
[102] = { 0 };
402 int exp_index1
[102] = { 0 };
404 int tmp_categorize_array
[128 * 2] = { 0 };
405 int tmp_categorize_array1_idx
= p
->numvector_size
;
406 int tmp_categorize_array2_idx
= p
->numvector_size
;
408 bits_left
= p
->bits_per_subpacket
- get_bits_count(&q
->gb
);
410 if (bits_left
> q
->samples_per_channel
)
411 bits_left
= q
->samples_per_channel
+
412 ((bits_left
- q
->samples_per_channel
) * 5) / 8;
417 for (i
= 32; i
> 0; i
= i
/ 2) {
420 for (j
= p
->total_subbands
; j
> 0; j
--) {
421 exp_idx
= av_clip((i
- quant_index_table
[index
] + bias
) / 2, 0, 7);
423 num_bits
+= expbits_tab
[exp_idx
];
425 if (num_bits
>= bits_left
- 32)
429 /* Calculate total number of bits. */
431 for (i
= 0; i
< p
->total_subbands
; i
++) {
432 exp_idx
= av_clip((bias
- quant_index_table
[i
]) / 2, 0, 7);
433 num_bits
+= expbits_tab
[exp_idx
];
434 exp_index1
[i
] = exp_idx
;
435 exp_index2
[i
] = exp_idx
;
437 tmpbias1
= tmpbias2
= num_bits
;
439 for (j
= 1; j
< p
->numvector_size
; j
++) {
440 if (tmpbias1
+ tmpbias2
> 2 * bits_left
) { /* ---> */
443 for (i
= 0; i
< p
->total_subbands
; i
++) {
444 if (exp_index1
[i
] < 7) {
445 v
= (-2 * exp_index1
[i
]) - quant_index_table
[i
] + bias
;
454 tmp_categorize_array
[tmp_categorize_array1_idx
++] = index
;
455 tmpbias1
-= expbits_tab
[exp_index1
[index
]] -
456 expbits_tab
[exp_index1
[index
] + 1];
461 for (i
= 0; i
< p
->total_subbands
; i
++) {
462 if (exp_index2
[i
] > 0) {
463 v
= (-2 * exp_index2
[i
]) - quant_index_table
[i
] + bias
;
472 tmp_categorize_array
[--tmp_categorize_array2_idx
] = index
;
473 tmpbias2
-= expbits_tab
[exp_index2
[index
]] -
474 expbits_tab
[exp_index2
[index
] - 1];
479 for (i
= 0; i
< p
->total_subbands
; i
++)
480 category
[i
] = exp_index2
[i
];
482 for (i
= 0; i
< p
->numvector_size
- 1; i
++)
483 category_index
[i
] = tmp_categorize_array
[tmp_categorize_array2_idx
++];
488 * Expand the category vector.
490 * @param q pointer to the COOKContext
491 * @param category pointer to the category array
492 * @param category_index pointer to the category_index array
494 static inline void expand_category(COOKContext
*q
, int *category
,
498 for (i
= 0; i
< q
->num_vectors
; i
++)
500 int idx
= category_index
[i
];
501 if (++category
[idx
] >= FF_ARRAY_ELEMS(dither_tab
))
507 * The real requantization of the mltcoefs
509 * @param q pointer to the COOKContext
511 * @param quant_index quantisation index
512 * @param subband_coef_index array of indexes to quant_centroid_tab
513 * @param subband_coef_sign signs of coefficients
514 * @param mlt_p pointer into the mlt buffer
516 static void scalar_dequant_float(COOKContext
*q
, int index
, int quant_index
,
517 int *subband_coef_index
, int *subband_coef_sign
,
523 for (i
= 0; i
< SUBBAND_SIZE
; i
++) {
524 if (subband_coef_index
[i
]) {
525 f1
= quant_centroid_tab
[index
][subband_coef_index
[i
]];
526 if (subband_coef_sign
[i
])
529 /* noise coding if subband_coef_index[i] == 0 */
530 f1
= dither_tab
[index
];
531 if (av_lfg_get(&q
->random_state
) < 0x80000000)
534 mlt_p
[i
] = f1
* rootpow2tab
[quant_index
+ 63];
538 * Unpack the subband_coef_index and subband_coef_sign vectors.
540 * @param q pointer to the COOKContext
541 * @param category pointer to the category array
542 * @param subband_coef_index array of indexes to quant_centroid_tab
543 * @param subband_coef_sign signs of coefficients
545 static int unpack_SQVH(COOKContext
*q
, COOKSubpacket
*p
, int category
,
546 int *subband_coef_index
, int *subband_coef_sign
)
549 int vlc
, vd
, tmp
, result
;
551 vd
= vd_tab
[category
];
553 for (i
= 0; i
< vpr_tab
[category
]; i
++) {
554 vlc
= get_vlc2(&q
->gb
, q
->sqvh
[category
].table
, q
->sqvh
[category
].bits
, 3);
555 if (p
->bits_per_subpacket
< get_bits_count(&q
->gb
)) {
559 for (j
= vd
- 1; j
>= 0; j
--) {
560 tmp
= (vlc
* invradix_tab
[category
]) / 0x100000;
561 subband_coef_index
[vd
* i
+ j
] = vlc
- tmp
* (kmax_tab
[category
] + 1);
564 for (j
= 0; j
< vd
; j
++) {
565 if (subband_coef_index
[i
* vd
+ j
]) {
566 if (get_bits_count(&q
->gb
) < p
->bits_per_subpacket
) {
567 subband_coef_sign
[i
* vd
+ j
] = get_bits1(&q
->gb
);
570 subband_coef_sign
[i
* vd
+ j
] = 0;
573 subband_coef_sign
[i
* vd
+ j
] = 0;
582 * Fill the mlt_buffer with mlt coefficients.
584 * @param q pointer to the COOKContext
585 * @param category pointer to the category array
586 * @param quant_index_table pointer to the array
587 * @param mlt_buffer pointer to mlt coefficients
589 static void decode_vectors(COOKContext
*q
, COOKSubpacket
*p
, int *category
,
590 int *quant_index_table
, float *mlt_buffer
)
592 /* A zero in this table means that the subband coefficient is
593 random noise coded. */
594 int subband_coef_index
[SUBBAND_SIZE
];
595 /* A zero in this table means that the subband coefficient is a
596 positive multiplicator. */
597 int subband_coef_sign
[SUBBAND_SIZE
];
601 for (band
= 0; band
< p
->total_subbands
; band
++) {
602 index
= category
[band
];
603 if (category
[band
] < 7) {
604 if (unpack_SQVH(q
, p
, category
[band
], subband_coef_index
, subband_coef_sign
)) {
606 for (j
= 0; j
< p
->total_subbands
; j
++)
607 category
[band
+ j
] = 7;
611 memset(subband_coef_index
, 0, sizeof(subband_coef_index
));
612 memset(subband_coef_sign
, 0, sizeof(subband_coef_sign
));
614 q
->scalar_dequant(q
, index
, quant_index_table
[band
],
615 subband_coef_index
, subband_coef_sign
,
616 &mlt_buffer
[band
* SUBBAND_SIZE
]);
619 /* FIXME: should this be removed, or moved into loop above? */
620 if (p
->total_subbands
* SUBBAND_SIZE
>= q
->samples_per_channel
)
625 static int mono_decode(COOKContext
*q
, COOKSubpacket
*p
, float *mlt_buffer
)
627 int category_index
[128] = { 0 };
628 int category
[128] = { 0 };
629 int quant_index_table
[102];
632 if ((res
= decode_envelope(q
, p
, quant_index_table
)) < 0)
634 q
->num_vectors
= get_bits(&q
->gb
, p
->log2_numvector_size
);
635 categorize(q
, p
, quant_index_table
, category
, category_index
);
636 expand_category(q
, category
, category_index
);
637 for (i
=0; i
<p
->total_subbands
; i
++) {
639 return AVERROR_INVALIDDATA
;
641 decode_vectors(q
, p
, category
, quant_index_table
, mlt_buffer
);
648 * the actual requantization of the timedomain samples
650 * @param q pointer to the COOKContext
651 * @param buffer pointer to the timedomain buffer
652 * @param gain_index index for the block multiplier
653 * @param gain_index_next index for the next block multiplier
655 static void interpolate_float(COOKContext
*q
, float *buffer
,
656 int gain_index
, int gain_index_next
)
660 fc1
= pow2tab
[gain_index
+ 63];
662 if (gain_index
== gain_index_next
) { // static gain
663 for (i
= 0; i
< q
->gain_size_factor
; i
++)
665 } else { // smooth gain
666 fc2
= q
->gain_table
[11 + (gain_index_next
- gain_index
)];
667 for (i
= 0; i
< q
->gain_size_factor
; i
++) {
675 * Apply transform window, overlap buffers.
677 * @param q pointer to the COOKContext
678 * @param inbuffer pointer to the mltcoefficients
679 * @param gains_ptr current and previous gains
680 * @param previous_buffer pointer to the previous buffer to be used for overlapping
682 static void imlt_window_float(COOKContext
*q
, float *inbuffer
,
683 cook_gains
*gains_ptr
, float *previous_buffer
)
685 const float fc
= pow2tab
[gains_ptr
->previous
[0] + 63];
687 /* The weird thing here, is that the two halves of the time domain
688 * buffer are swapped. Also, the newest data, that we save away for
689 * next frame, has the wrong sign. Hence the subtraction below.
690 * Almost sounds like a complex conjugate/reverse data/FFT effect.
693 /* Apply window and overlap */
694 for (i
= 0; i
< q
->samples_per_channel
; i
++)
695 inbuffer
[i
] = inbuffer
[i
] * fc
* q
->mlt_window
[i
] -
696 previous_buffer
[i
] * q
->mlt_window
[q
->samples_per_channel
- 1 - i
];
700 * The modulated lapped transform, this takes transform coefficients
701 * and transforms them into timedomain samples.
702 * Apply transform window, overlap buffers, apply gain profile
703 * and buffer management.
705 * @param q pointer to the COOKContext
706 * @param inbuffer pointer to the mltcoefficients
707 * @param gains_ptr current and previous gains
708 * @param previous_buffer pointer to the previous buffer to be used for overlapping
710 static void imlt_gain(COOKContext
*q
, float *inbuffer
,
711 cook_gains
*gains_ptr
, float *previous_buffer
)
713 float *buffer0
= q
->mono_mdct_output
;
714 float *buffer1
= q
->mono_mdct_output
+ q
->samples_per_channel
;
717 /* Inverse modified discrete cosine transform */
718 q
->mdct_ctx
.imdct_calc(&q
->mdct_ctx
, q
->mono_mdct_output
, inbuffer
);
720 q
->imlt_window(q
, buffer1
, gains_ptr
, previous_buffer
);
722 /* Apply gain profile */
723 for (i
= 0; i
< 8; i
++)
724 if (gains_ptr
->now
[i
] || gains_ptr
->now
[i
+ 1])
725 q
->interpolate(q
, &buffer1
[q
->gain_size_factor
* i
],
726 gains_ptr
->now
[i
], gains_ptr
->now
[i
+ 1]);
728 /* Save away the current to be previous block. */
729 memcpy(previous_buffer
, buffer0
,
730 q
->samples_per_channel
* sizeof(*previous_buffer
));
735 * function for getting the jointstereo coupling information
737 * @param q pointer to the COOKContext
738 * @param decouple_tab decoupling array
740 static int decouple_info(COOKContext
*q
, COOKSubpacket
*p
, int *decouple_tab
)
743 int vlc
= get_bits1(&q
->gb
);
744 int start
= cplband
[p
->js_subband_start
];
745 int end
= cplband
[p
->subbands
- 1];
746 int length
= end
- start
+ 1;
752 for (i
= 0; i
< length
; i
++)
753 decouple_tab
[start
+ i
] = get_vlc2(&q
->gb
,
754 p
->channel_coupling
.table
,
755 p
->channel_coupling
.bits
, 2);
757 for (i
= 0; i
< length
; i
++) {
758 int v
= get_bits(&q
->gb
, p
->js_vlc_bits
);
759 if (v
== (1<<p
->js_vlc_bits
)-1) {
760 av_log(q
->avctx
, AV_LOG_ERROR
, "decouple value too large\n");
761 return AVERROR_INVALIDDATA
;
763 decouple_tab
[start
+ i
] = v
;
769 * function decouples a pair of signals from a single signal via multiplication.
771 * @param q pointer to the COOKContext
772 * @param subband index of the current subband
773 * @param f1 multiplier for channel 1 extraction
774 * @param f2 multiplier for channel 2 extraction
775 * @param decode_buffer input buffer
776 * @param mlt_buffer1 pointer to left channel mlt coefficients
777 * @param mlt_buffer2 pointer to right channel mlt coefficients
779 static void decouple_float(COOKContext
*q
,
783 float *decode_buffer
,
784 float *mlt_buffer1
, float *mlt_buffer2
)
787 for (j
= 0; j
< SUBBAND_SIZE
; j
++) {
788 tmp_idx
= ((p
->js_subband_start
+ subband
) * SUBBAND_SIZE
) + j
;
789 mlt_buffer1
[SUBBAND_SIZE
* subband
+ j
] = f1
* decode_buffer
[tmp_idx
];
790 mlt_buffer2
[SUBBAND_SIZE
* subband
+ j
] = f2
* decode_buffer
[tmp_idx
];
795 * function for decoding joint stereo data
797 * @param q pointer to the COOKContext
798 * @param mlt_buffer1 pointer to left channel mlt coefficients
799 * @param mlt_buffer2 pointer to right channel mlt coefficients
801 static int joint_decode(COOKContext
*q
, COOKSubpacket
*p
,
802 float *mlt_buffer_left
, float *mlt_buffer_right
)
805 int decouple_tab
[SUBBAND_SIZE
] = { 0 };
806 float *decode_buffer
= q
->decode_buffer_0
;
809 const float *cplscale
;
811 memset(decode_buffer
, 0, sizeof(q
->decode_buffer_0
));
813 /* Make sure the buffers are zeroed out. */
814 memset(mlt_buffer_left
, 0, 1024 * sizeof(*mlt_buffer_left
));
815 memset(mlt_buffer_right
, 0, 1024 * sizeof(*mlt_buffer_right
));
816 if ((res
= decouple_info(q
, p
, decouple_tab
)) < 0)
818 if ((res
= mono_decode(q
, p
, decode_buffer
)) < 0)
820 /* The two channels are stored interleaved in decode_buffer. */
821 for (i
= 0; i
< p
->js_subband_start
; i
++) {
822 for (j
= 0; j
< SUBBAND_SIZE
; j
++) {
823 mlt_buffer_left
[i
* 20 + j
] = decode_buffer
[i
* 40 + j
];
824 mlt_buffer_right
[i
* 20 + j
] = decode_buffer
[i
* 40 + 20 + j
];
828 /* When we reach js_subband_start (the higher frequencies)
829 the coefficients are stored in a coupling scheme. */
830 idx
= (1 << p
->js_vlc_bits
) - 1;
831 for (i
= p
->js_subband_start
; i
< p
->subbands
; i
++) {
832 cpl_tmp
= cplband
[i
];
833 idx
-= decouple_tab
[cpl_tmp
];
834 cplscale
= q
->cplscales
[p
->js_vlc_bits
- 2]; // choose decoupler table
835 f1
= cplscale
[decouple_tab
[cpl_tmp
] + 1];
837 q
->decouple(q
, p
, i
, f1
, f2
, decode_buffer
,
838 mlt_buffer_left
, mlt_buffer_right
);
839 idx
= (1 << p
->js_vlc_bits
) - 1;
846 * First part of subpacket decoding:
847 * decode raw stream bytes and read gain info.
849 * @param q pointer to the COOKContext
850 * @param inbuffer pointer to raw stream data
851 * @param gains_ptr array of current/prev gain pointers
853 static inline void decode_bytes_and_gain(COOKContext
*q
, COOKSubpacket
*p
,
854 const uint8_t *inbuffer
,
855 cook_gains
*gains_ptr
)
859 offset
= decode_bytes(inbuffer
, q
->decoded_bytes_buffer
,
860 p
->bits_per_subpacket
/ 8);
861 init_get_bits(&q
->gb
, q
->decoded_bytes_buffer
+ offset
,
862 p
->bits_per_subpacket
);
863 decode_gain_info(&q
->gb
, gains_ptr
->now
);
865 /* Swap current and previous gains */
866 FFSWAP(int *, gains_ptr
->now
, gains_ptr
->previous
);
870 * Saturate the output signal and interleave.
872 * @param q pointer to the COOKContext
873 * @param out pointer to the output vector
875 static void saturate_output_float(COOKContext
*q
, float *out
)
877 q
->adsp
.vector_clipf(out
, q
->mono_mdct_output
+ q
->samples_per_channel
,
878 -1.0f
, 1.0f
, FFALIGN(q
->samples_per_channel
, 8));
883 * Final part of subpacket decoding:
884 * Apply modulated lapped transform, gain compensation,
885 * clip and convert to integer.
887 * @param q pointer to the COOKContext
888 * @param decode_buffer pointer to the mlt coefficients
889 * @param gains_ptr array of current/prev gain pointers
890 * @param previous_buffer pointer to the previous buffer to be used for overlapping
891 * @param out pointer to the output buffer
893 static inline void mlt_compensate_output(COOKContext
*q
, float *decode_buffer
,
894 cook_gains
*gains_ptr
, float *previous_buffer
,
897 imlt_gain(q
, decode_buffer
, gains_ptr
, previous_buffer
);
899 q
->saturate_output(q
, out
);
904 * Cook subpacket decoding. This function returns one decoded subpacket,
905 * usually 1024 samples per channel.
907 * @param q pointer to the COOKContext
908 * @param inbuffer pointer to the inbuffer
909 * @param outbuffer pointer to the outbuffer
911 static int decode_subpacket(COOKContext
*q
, COOKSubpacket
*p
,
912 const uint8_t *inbuffer
, float **outbuffer
)
914 int sub_packet_size
= p
->size
;
917 memset(q
->decode_buffer_1
, 0, sizeof(q
->decode_buffer_1
));
918 decode_bytes_and_gain(q
, p
, inbuffer
, &p
->gains1
);
920 if (p
->joint_stereo
) {
921 if ((res
= joint_decode(q
, p
, q
->decode_buffer_1
, q
->decode_buffer_2
)) < 0)
924 if ((res
= mono_decode(q
, p
, q
->decode_buffer_1
)) < 0)
927 if (p
->num_channels
== 2) {
928 decode_bytes_and_gain(q
, p
, inbuffer
+ sub_packet_size
/ 2, &p
->gains2
);
929 if ((res
= mono_decode(q
, p
, q
->decode_buffer_2
)) < 0)
934 mlt_compensate_output(q
, q
->decode_buffer_1
, &p
->gains1
,
935 p
->mono_previous_buffer1
,
936 outbuffer
? outbuffer
[p
->ch_idx
] : NULL
);
938 if (p
->num_channels
== 2) {
940 mlt_compensate_output(q
, q
->decode_buffer_2
, &p
->gains1
,
941 p
->mono_previous_buffer2
,
942 outbuffer
? outbuffer
[p
->ch_idx
+ 1] : NULL
);
944 mlt_compensate_output(q
, q
->decode_buffer_2
, &p
->gains2
,
945 p
->mono_previous_buffer2
,
946 outbuffer
? outbuffer
[p
->ch_idx
+ 1] : NULL
);
953 static int cook_decode_frame(AVCodecContext
*avctx
, void *data
,
954 int *got_frame_ptr
, AVPacket
*avpkt
)
956 AVFrame
*frame
= data
;
957 const uint8_t *buf
= avpkt
->data
;
958 int buf_size
= avpkt
->size
;
959 COOKContext
*q
= avctx
->priv_data
;
960 float **samples
= NULL
;
965 if (buf_size
< avctx
->block_align
)
968 /* get output buffer */
969 if (q
->discarded_packets
>= 2) {
970 frame
->nb_samples
= q
->samples_per_channel
;
971 if ((ret
= ff_get_buffer(avctx
, frame
, 0)) < 0)
973 samples
= (float **)frame
->extended_data
;
976 /* estimate subpacket sizes */
977 q
->subpacket
[0].size
= avctx
->block_align
;
979 for (i
= 1; i
< q
->num_subpackets
; i
++) {
980 q
->subpacket
[i
].size
= 2 * buf
[avctx
->block_align
- q
->num_subpackets
+ i
];
981 q
->subpacket
[0].size
-= q
->subpacket
[i
].size
+ 1;
982 if (q
->subpacket
[0].size
< 0) {
983 av_log(avctx
, AV_LOG_DEBUG
,
984 "frame subpacket size total > avctx->block_align!\n");
985 return AVERROR_INVALIDDATA
;
989 /* decode supbackets */
990 for (i
= 0; i
< q
->num_subpackets
; i
++) {
991 q
->subpacket
[i
].bits_per_subpacket
= (q
->subpacket
[i
].size
* 8) >>
992 q
->subpacket
[i
].bits_per_subpdiv
;
993 q
->subpacket
[i
].ch_idx
= chidx
;
994 av_log(avctx
, AV_LOG_DEBUG
,
995 "subpacket[%i] size %i js %i %i block_align %i\n",
996 i
, q
->subpacket
[i
].size
, q
->subpacket
[i
].joint_stereo
, offset
,
999 if ((ret
= decode_subpacket(q
, &q
->subpacket
[i
], buf
+ offset
, samples
)) < 0)
1001 offset
+= q
->subpacket
[i
].size
;
1002 chidx
+= q
->subpacket
[i
].num_channels
;
1003 av_log(avctx
, AV_LOG_DEBUG
, "subpacket[%i] %i %i\n",
1004 i
, q
->subpacket
[i
].size
* 8, get_bits_count(&q
->gb
));
1007 /* Discard the first two frames: no valid audio. */
1008 if (q
->discarded_packets
< 2) {
1009 q
->discarded_packets
++;
1011 return avctx
->block_align
;
1016 return avctx
->block_align
;
1020 static void dump_cook_context(COOKContext
*q
)
1023 #define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
1024 av_dlog(q
->avctx
, "COOKextradata\n");
1025 av_dlog(q
->avctx
, "cookversion=%x\n", q
->subpacket
[0].cookversion
);
1026 if (q
->subpacket
[0].cookversion
> STEREO
) {
1027 PRINT("js_subband_start", q
->subpacket
[0].js_subband_start
);
1028 PRINT("js_vlc_bits", q
->subpacket
[0].js_vlc_bits
);
1030 av_dlog(q
->avctx
, "COOKContext\n");
1031 PRINT("nb_channels", q
->avctx
->channels
);
1032 PRINT("bit_rate", q
->avctx
->bit_rate
);
1033 PRINT("sample_rate", q
->avctx
->sample_rate
);
1034 PRINT("samples_per_channel", q
->subpacket
[0].samples_per_channel
);
1035 PRINT("subbands", q
->subpacket
[0].subbands
);
1036 PRINT("js_subband_start", q
->subpacket
[0].js_subband_start
);
1037 PRINT("log2_numvector_size", q
->subpacket
[0].log2_numvector_size
);
1038 PRINT("numvector_size", q
->subpacket
[0].numvector_size
);
1039 PRINT("total_subbands", q
->subpacket
[0].total_subbands
);
1044 * Cook initialization
1046 * @param avctx pointer to the AVCodecContext
1048 static av_cold
int cook_decode_init(AVCodecContext
*avctx
)
1050 COOKContext
*q
= avctx
->priv_data
;
1051 const uint8_t *edata_ptr
= avctx
->extradata
;
1052 const uint8_t *edata_ptr_end
= edata_ptr
+ avctx
->extradata_size
;
1053 int extradata_size
= avctx
->extradata_size
;
1055 unsigned int channel_mask
= 0;
1056 int samples_per_frame
= 0;
1060 /* Take care of the codec specific extradata. */
1061 if (extradata_size
< 8) {
1062 av_log(avctx
, AV_LOG_ERROR
, "Necessary extradata missing!\n");
1063 return AVERROR_INVALIDDATA
;
1065 av_log(avctx
, AV_LOG_DEBUG
, "codecdata_length=%d\n", avctx
->extradata_size
);
1067 /* Take data from the AVCodecContext (RM container). */
1068 if (!avctx
->channels
) {
1069 av_log(avctx
, AV_LOG_ERROR
, "Invalid number of channels\n");
1070 return AVERROR_INVALIDDATA
;
1073 /* Initialize RNG. */
1074 av_lfg_init(&q
->random_state
, 0);
1076 ff_audiodsp_init(&q
->adsp
);
1078 while (edata_ptr
< edata_ptr_end
) {
1079 /* 8 for mono, 16 for stereo, ? for multichannel
1080 Swap to right endianness so we don't need to care later on. */
1081 if (extradata_size
>= 8) {
1082 q
->subpacket
[s
].cookversion
= bytestream_get_be32(&edata_ptr
);
1083 samples_per_frame
= bytestream_get_be16(&edata_ptr
);
1084 q
->subpacket
[s
].subbands
= bytestream_get_be16(&edata_ptr
);
1085 extradata_size
-= 8;
1087 if (extradata_size
>= 8) {
1088 bytestream_get_be32(&edata_ptr
); // Unknown unused
1089 q
->subpacket
[s
].js_subband_start
= bytestream_get_be16(&edata_ptr
);
1090 if (q
->subpacket
[s
].js_subband_start
>= 51) {
1091 av_log(avctx
, AV_LOG_ERROR
, "js_subband_start %d is too large\n", q
->subpacket
[s
].js_subband_start
);
1092 return AVERROR_INVALIDDATA
;
1095 q
->subpacket
[s
].js_vlc_bits
= bytestream_get_be16(&edata_ptr
);
1096 extradata_size
-= 8;
1099 /* Initialize extradata related variables. */
1100 q
->subpacket
[s
].samples_per_channel
= samples_per_frame
/ avctx
->channels
;
1101 q
->subpacket
[s
].bits_per_subpacket
= avctx
->block_align
* 8;
1103 /* Initialize default data states. */
1104 q
->subpacket
[s
].log2_numvector_size
= 5;
1105 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
;
1106 q
->subpacket
[s
].num_channels
= 1;
1108 /* Initialize version-dependent variables */
1110 av_log(avctx
, AV_LOG_DEBUG
, "subpacket[%i].cookversion=%x\n", s
,
1111 q
->subpacket
[s
].cookversion
);
1112 q
->subpacket
[s
].joint_stereo
= 0;
1113 switch (q
->subpacket
[s
].cookversion
) {
1115 if (avctx
->channels
!= 1) {
1116 avpriv_request_sample(avctx
, "Container channels != 1");
1117 return AVERROR_PATCHWELCOME
;
1119 av_log(avctx
, AV_LOG_DEBUG
, "MONO\n");
1122 if (avctx
->channels
!= 1) {
1123 q
->subpacket
[s
].bits_per_subpdiv
= 1;
1124 q
->subpacket
[s
].num_channels
= 2;
1126 av_log(avctx
, AV_LOG_DEBUG
, "STEREO\n");
1129 if (avctx
->channels
!= 2) {
1130 avpriv_request_sample(avctx
, "Container channels != 2");
1131 return AVERROR_PATCHWELCOME
;
1133 av_log(avctx
, AV_LOG_DEBUG
, "JOINT_STEREO\n");
1134 if (avctx
->extradata_size
>= 16) {
1135 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
+
1136 q
->subpacket
[s
].js_subband_start
;
1137 q
->subpacket
[s
].joint_stereo
= 1;
1138 q
->subpacket
[s
].num_channels
= 2;
1140 if (q
->subpacket
[s
].samples_per_channel
> 256) {
1141 q
->subpacket
[s
].log2_numvector_size
= 6;
1143 if (q
->subpacket
[s
].samples_per_channel
> 512) {
1144 q
->subpacket
[s
].log2_numvector_size
= 7;
1148 av_log(avctx
, AV_LOG_DEBUG
, "MULTI_CHANNEL\n");
1149 if (extradata_size
>= 4)
1150 channel_mask
|= q
->subpacket
[s
].channel_mask
= bytestream_get_be32(&edata_ptr
);
1152 if (av_get_channel_layout_nb_channels(q
->subpacket
[s
].channel_mask
) > 1) {
1153 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
+
1154 q
->subpacket
[s
].js_subband_start
;
1155 q
->subpacket
[s
].joint_stereo
= 1;
1156 q
->subpacket
[s
].num_channels
= 2;
1157 q
->subpacket
[s
].samples_per_channel
= samples_per_frame
>> 1;
1159 if (q
->subpacket
[s
].samples_per_channel
> 256) {
1160 q
->subpacket
[s
].log2_numvector_size
= 6;
1162 if (q
->subpacket
[s
].samples_per_channel
> 512) {
1163 q
->subpacket
[s
].log2_numvector_size
= 7;
1166 q
->subpacket
[s
].samples_per_channel
= samples_per_frame
;
1170 avpriv_request_sample(avctx
, "Cook version %d",
1171 q
->subpacket
[s
].cookversion
);
1172 return AVERROR_PATCHWELCOME
;
1175 if (s
> 1 && q
->subpacket
[s
].samples_per_channel
!= q
->samples_per_channel
) {
1176 av_log(avctx
, AV_LOG_ERROR
, "different number of samples per channel!\n");
1177 return AVERROR_INVALIDDATA
;
1179 q
->samples_per_channel
= q
->subpacket
[0].samples_per_channel
;
1182 /* Initialize variable relations */
1183 q
->subpacket
[s
].numvector_size
= (1 << q
->subpacket
[s
].log2_numvector_size
);
1185 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1186 if (q
->subpacket
[s
].total_subbands
> 53) {
1187 avpriv_request_sample(avctx
, "total_subbands > 53");
1188 return AVERROR_PATCHWELCOME
;
1191 if ((q
->subpacket
[s
].js_vlc_bits
> 6) ||
1192 (q
->subpacket
[s
].js_vlc_bits
< 2 * q
->subpacket
[s
].joint_stereo
)) {
1193 av_log(avctx
, AV_LOG_ERROR
, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1194 q
->subpacket
[s
].js_vlc_bits
, 2 * q
->subpacket
[s
].joint_stereo
);
1195 return AVERROR_INVALIDDATA
;
1198 if (q
->subpacket
[s
].subbands
> 50) {
1199 avpriv_request_sample(avctx
, "subbands > 50");
1200 return AVERROR_PATCHWELCOME
;
1202 if (q
->subpacket
[s
].subbands
== 0) {
1203 avpriv_request_sample(avctx
, "subbands = 0");
1204 return AVERROR_PATCHWELCOME
;
1206 q
->subpacket
[s
].gains1
.now
= q
->subpacket
[s
].gain_1
;
1207 q
->subpacket
[s
].gains1
.previous
= q
->subpacket
[s
].gain_2
;
1208 q
->subpacket
[s
].gains2
.now
= q
->subpacket
[s
].gain_3
;
1209 q
->subpacket
[s
].gains2
.previous
= q
->subpacket
[s
].gain_4
;
1211 if (q
->num_subpackets
+ q
->subpacket
[s
].num_channels
> q
->avctx
->channels
) {
1212 av_log(avctx
, AV_LOG_ERROR
, "Too many subpackets %d for channels %d\n", q
->num_subpackets
, q
->avctx
->channels
);
1213 return AVERROR_INVALIDDATA
;
1216 q
->num_subpackets
++;
1218 if (s
> FFMIN(MAX_SUBPACKETS
, avctx
->block_align
)) {
1219 avpriv_request_sample(avctx
, "subpackets > %d", FFMIN(MAX_SUBPACKETS
, avctx
->block_align
));
1220 return AVERROR_PATCHWELCOME
;
1223 /* Generate tables */
1226 init_cplscales_table(q
);
1228 if ((ret
= init_cook_vlc_tables(q
)))
1232 if (avctx
->block_align
>= UINT_MAX
/ 2)
1233 return AVERROR(EINVAL
);
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
)
1243 return AVERROR(ENOMEM
);
1245 /* Initialize transform. */
1246 if ((ret
= init_cook_mlt(q
)))
1249 /* Initialize COOK signal arithmetic handling */
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 &&
1260 q
->samples_per_channel
!= 1024) {
1261 avpriv_request_sample(avctx
, "samples_per_channel = %d",
1262 q
->samples_per_channel
);
1263 return AVERROR_PATCHWELCOME
;
1266 avctx
->sample_fmt
= AV_SAMPLE_FMT_FLTP
;
1268 avctx
->channel_layout
= channel_mask
;
1270 avctx
->channel_layout
= (avctx
->channels
== 2) ? AV_CH_LAYOUT_STEREO
: AV_CH_LAYOUT_MONO
;
1273 dump_cook_context(q
);
1278 AVCodec ff_cook_decoder
= {
1280 .long_name
= NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1281 .type
= AVMEDIA_TYPE_AUDIO
,
1282 .id
= AV_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 .capabilities
= CODEC_CAP_DR1
,
1288 .sample_fmts
= (const enum AVSampleFormat
[]) { AV_SAMPLE_FMT_FLTP
,
1289 AV_SAMPLE_FMT_NONE
},