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 "bitstream.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
68 typedef struct cook_gains
{
73 typedef struct COOKSubpacket
{
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 /* generate 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(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 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_free(q
->mlt_window
);
308 av_free(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 bc pointer to the BitstreamContext
330 * @param gaininfo array[9] of gain indexes
332 static void decode_gain_info(BitstreamContext
*bc
, int *gaininfo
)
336 while (bitstream_read_bit(bc
)) {
340 n
= bitstream_tell(bc
) - 1; // amount of elements * 2 to update
344 int index
= bitstream_read(bc
, 3);
345 int gain
= bitstream_read_bit(bc
) ?
bitstream_read(bc
, 4) - 7 : -1;
348 gaininfo
[i
++] = gain
;
355 * Create the quant index table needed for the envelope.
357 * @param q pointer to the COOKContext
358 * @param quant_index_table pointer to the array
360 static int decode_envelope(COOKContext
*q
, COOKSubpacket
*p
,
361 int *quant_index_table
)
365 quant_index_table
[0] = bitstream_read(&q
->bc
, 6) - 6; // This is used later in categorize
367 for (i
= 1; i
< p
->total_subbands
; i
++) {
369 if (i
>= p
->js_subband_start
* 2) {
370 vlc_index
-= p
->js_subband_start
;
377 vlc_index
= 13; // the VLC tables >13 are identical to No. 13
379 j
= bitstream_read_vlc(&q
->bc
, q
->envelope_quant_index
[vlc_index
- 1].table
,
380 q
->envelope_quant_index
[vlc_index
- 1].bits
, 2);
381 quant_index_table
[i
] = quant_index_table
[i
- 1] + j
- 12; // differential encoding
382 if (quant_index_table
[i
] > 63 || quant_index_table
[i
] < -63) {
383 av_log(q
->avctx
, AV_LOG_ERROR
,
384 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
385 quant_index_table
[i
], i
);
386 return AVERROR_INVALIDDATA
;
394 * Calculate the category and category_index vector.
396 * @param q pointer to the COOKContext
397 * @param quant_index_table pointer to the array
398 * @param category pointer to the category array
399 * @param category_index pointer to the category_index array
401 static void categorize(COOKContext
*q
, COOKSubpacket
*p
, int *quant_index_table
,
402 int *category
, int *category_index
)
404 int exp_idx
, bias
, tmpbias1
, tmpbias2
, bits_left
, num_bits
, index
, v
, i
, j
;
405 int exp_index2
[102] = { 0 };
406 int exp_index1
[102] = { 0 };
408 int tmp_categorize_array
[128 * 2] = { 0 };
409 int tmp_categorize_array1_idx
= p
->numvector_size
;
410 int tmp_categorize_array2_idx
= p
->numvector_size
;
412 bits_left
= p
->bits_per_subpacket
- bitstream_tell(&q
->bc
);
414 if (bits_left
> q
->samples_per_channel
)
415 bits_left
= q
->samples_per_channel
+
416 ((bits_left
- q
->samples_per_channel
) * 5) / 8;
421 for (i
= 32; i
> 0; i
= i
/ 2) {
424 for (j
= p
->total_subbands
; j
> 0; j
--) {
425 exp_idx
= av_clip((i
- quant_index_table
[index
] + bias
) / 2, 0, 7);
427 num_bits
+= expbits_tab
[exp_idx
];
429 if (num_bits
>= bits_left
- 32)
433 /* Calculate total number of bits. */
435 for (i
= 0; i
< p
->total_subbands
; i
++) {
436 exp_idx
= av_clip((bias
- quant_index_table
[i
]) / 2, 0, 7);
437 num_bits
+= expbits_tab
[exp_idx
];
438 exp_index1
[i
] = exp_idx
;
439 exp_index2
[i
] = exp_idx
;
441 tmpbias1
= tmpbias2
= num_bits
;
443 for (j
= 1; j
< p
->numvector_size
; j
++) {
444 if (tmpbias1
+ tmpbias2
> 2 * bits_left
) { /* ---> */
447 for (i
= 0; i
< p
->total_subbands
; i
++) {
448 if (exp_index1
[i
] < 7) {
449 v
= (-2 * exp_index1
[i
]) - quant_index_table
[i
] + bias
;
458 tmp_categorize_array
[tmp_categorize_array1_idx
++] = index
;
459 tmpbias1
-= expbits_tab
[exp_index1
[index
]] -
460 expbits_tab
[exp_index1
[index
] + 1];
465 for (i
= 0; i
< p
->total_subbands
; i
++) {
466 if (exp_index2
[i
] > 0) {
467 v
= (-2 * exp_index2
[i
]) - quant_index_table
[i
] + bias
;
476 tmp_categorize_array
[--tmp_categorize_array2_idx
] = index
;
477 tmpbias2
-= expbits_tab
[exp_index2
[index
]] -
478 expbits_tab
[exp_index2
[index
] - 1];
483 for (i
= 0; i
< p
->total_subbands
; i
++)
484 category
[i
] = exp_index2
[i
];
486 for (i
= 0; i
< p
->numvector_size
- 1; i
++)
487 category_index
[i
] = tmp_categorize_array
[tmp_categorize_array2_idx
++];
492 * Expand the category vector.
494 * @param q pointer to the COOKContext
495 * @param category pointer to the category array
496 * @param category_index pointer to the category_index array
498 static inline void expand_category(COOKContext
*q
, int *category
,
502 for (i
= 0; i
< q
->num_vectors
; i
++)
504 int idx
= category_index
[i
];
505 if (++category
[idx
] >= FF_ARRAY_ELEMS(dither_tab
))
511 * The real requantization of the mltcoefs
513 * @param q pointer to the COOKContext
515 * @param quant_index quantisation index
516 * @param subband_coef_index array of indexes to quant_centroid_tab
517 * @param subband_coef_sign signs of coefficients
518 * @param mlt_p pointer into the mlt buffer
520 static void scalar_dequant_float(COOKContext
*q
, int index
, int quant_index
,
521 int *subband_coef_index
, int *subband_coef_sign
,
527 for (i
= 0; i
< SUBBAND_SIZE
; i
++) {
528 if (subband_coef_index
[i
]) {
529 f1
= quant_centroid_tab
[index
][subband_coef_index
[i
]];
530 if (subband_coef_sign
[i
])
533 /* noise coding if subband_coef_index[i] == 0 */
534 f1
= dither_tab
[index
];
535 if (av_lfg_get(&q
->random_state
) < 0x80000000)
538 mlt_p
[i
] = f1
* rootpow2tab
[quant_index
+ 63];
542 * Unpack the subband_coef_index and subband_coef_sign vectors.
544 * @param q pointer to the COOKContext
545 * @param category pointer to the category array
546 * @param subband_coef_index array of indexes to quant_centroid_tab
547 * @param subband_coef_sign signs of coefficients
549 static int unpack_SQVH(COOKContext
*q
, COOKSubpacket
*p
, int category
,
550 int *subband_coef_index
, int *subband_coef_sign
)
553 int vlc
, vd
, tmp
, result
;
555 vd
= vd_tab
[category
];
557 for (i
= 0; i
< vpr_tab
[category
]; i
++) {
558 vlc
= bitstream_read_vlc(&q
->bc
, q
->sqvh
[category
].table
, q
->sqvh
[category
].bits
, 3);
559 if (p
->bits_per_subpacket
< bitstream_tell(&q
->bc
)) {
563 for (j
= vd
- 1; j
>= 0; j
--) {
564 tmp
= (vlc
* invradix_tab
[category
]) / 0x100000;
565 subband_coef_index
[vd
* i
+ j
] = vlc
- tmp
* (kmax_tab
[category
] + 1);
568 for (j
= 0; j
< vd
; j
++) {
569 if (subband_coef_index
[i
* vd
+ j
]) {
570 if (bitstream_tell(&q
->bc
) < p
->bits_per_subpacket
) {
571 subband_coef_sign
[i
* vd
+ j
] = bitstream_read_bit(&q
->bc
);
574 subband_coef_sign
[i
* vd
+ j
] = 0;
577 subband_coef_sign
[i
* vd
+ j
] = 0;
586 * Fill the mlt_buffer with mlt coefficients.
588 * @param q pointer to the COOKContext
589 * @param category pointer to the category array
590 * @param quant_index_table pointer to the array
591 * @param mlt_buffer pointer to mlt coefficients
593 static void decode_vectors(COOKContext
*q
, COOKSubpacket
*p
, int *category
,
594 int *quant_index_table
, float *mlt_buffer
)
596 /* A zero in this table means that the subband coefficient is
597 random noise coded. */
598 int subband_coef_index
[SUBBAND_SIZE
];
599 /* A zero in this table means that the subband coefficient is a
600 positive multiplicator. */
601 int subband_coef_sign
[SUBBAND_SIZE
];
605 for (band
= 0; band
< p
->total_subbands
; band
++) {
606 index
= category
[band
];
607 if (category
[band
] < 7) {
608 if (unpack_SQVH(q
, p
, category
[band
], subband_coef_index
, subband_coef_sign
)) {
610 for (j
= 0; j
< p
->total_subbands
; j
++)
611 category
[band
+ j
] = 7;
615 memset(subband_coef_index
, 0, sizeof(subband_coef_index
));
616 memset(subband_coef_sign
, 0, sizeof(subband_coef_sign
));
618 q
->scalar_dequant(q
, index
, quant_index_table
[band
],
619 subband_coef_index
, subband_coef_sign
,
620 &mlt_buffer
[band
* SUBBAND_SIZE
]);
623 /* FIXME: should this be removed, or moved into loop above? */
624 if (p
->total_subbands
* SUBBAND_SIZE
>= q
->samples_per_channel
)
629 static int mono_decode(COOKContext
*q
, COOKSubpacket
*p
, float *mlt_buffer
)
631 int category_index
[128] = { 0 };
632 int category
[128] = { 0 };
633 int quant_index_table
[102];
636 if ((res
= decode_envelope(q
, p
, quant_index_table
)) < 0)
638 q
->num_vectors
= bitstream_read(&q
->bc
, p
->log2_numvector_size
);
639 categorize(q
, p
, quant_index_table
, category
, category_index
);
640 expand_category(q
, category
, category_index
);
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 void decouple_info(COOKContext
*q
, COOKSubpacket
*p
, int *decouple_tab
)
743 int vlc
= bitstream_read_bit(&q
->bc
);
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
] =
754 bitstream_read_vlc(&q
->bc
,
755 p
->channel_coupling
.table
,
756 p
->channel_coupling
.bits
, 2);
758 for (i
= 0; i
< length
; i
++)
759 decouple_tab
[start
+ i
] = bitstream_read(&q
->bc
, p
->js_vlc_bits
);
763 * function decouples a pair of signals from a single signal via multiplication.
765 * @param q pointer to the COOKContext
766 * @param subband index of the current subband
767 * @param f1 multiplier for channel 1 extraction
768 * @param f2 multiplier for channel 2 extraction
769 * @param decode_buffer input buffer
770 * @param mlt_buffer1 pointer to left channel mlt coefficients
771 * @param mlt_buffer2 pointer to right channel mlt coefficients
773 static void decouple_float(COOKContext
*q
,
777 float *decode_buffer
,
778 float *mlt_buffer1
, float *mlt_buffer2
)
781 for (j
= 0; j
< SUBBAND_SIZE
; j
++) {
782 tmp_idx
= ((p
->js_subband_start
+ subband
) * SUBBAND_SIZE
) + j
;
783 mlt_buffer1
[SUBBAND_SIZE
* subband
+ j
] = f1
* decode_buffer
[tmp_idx
];
784 mlt_buffer2
[SUBBAND_SIZE
* subband
+ j
] = f2
* decode_buffer
[tmp_idx
];
789 * function for decoding joint stereo data
791 * @param q pointer to the COOKContext
792 * @param mlt_buffer1 pointer to left channel mlt coefficients
793 * @param mlt_buffer2 pointer to right channel mlt coefficients
795 static int joint_decode(COOKContext
*q
, COOKSubpacket
*p
,
796 float *mlt_buffer_left
, float *mlt_buffer_right
)
799 int decouple_tab
[SUBBAND_SIZE
] = { 0 };
800 float *decode_buffer
= q
->decode_buffer_0
;
803 const float *cplscale
;
805 memset(decode_buffer
, 0, sizeof(q
->decode_buffer_0
));
807 /* Make sure the buffers are zeroed out. */
808 memset(mlt_buffer_left
, 0, 1024 * sizeof(*mlt_buffer_left
));
809 memset(mlt_buffer_right
, 0, 1024 * sizeof(*mlt_buffer_right
));
810 decouple_info(q
, p
, decouple_tab
);
811 if ((res
= mono_decode(q
, p
, decode_buffer
)) < 0)
814 /* The two channels are stored interleaved in decode_buffer. */
815 for (i
= 0; i
< p
->js_subband_start
; i
++) {
816 for (j
= 0; j
< SUBBAND_SIZE
; j
++) {
817 mlt_buffer_left
[i
* 20 + j
] = decode_buffer
[i
* 40 + j
];
818 mlt_buffer_right
[i
* 20 + j
] = decode_buffer
[i
* 40 + 20 + j
];
822 /* When we reach js_subband_start (the higher frequencies)
823 the coefficients are stored in a coupling scheme. */
824 idx
= (1 << p
->js_vlc_bits
) - 1;
825 for (i
= p
->js_subband_start
; i
< p
->subbands
; i
++) {
826 cpl_tmp
= cplband
[i
];
827 idx
-= decouple_tab
[cpl_tmp
];
828 cplscale
= q
->cplscales
[p
->js_vlc_bits
- 2]; // choose decoupler table
829 f1
= cplscale
[decouple_tab
[cpl_tmp
] + 1];
831 q
->decouple(q
, p
, i
, f1
, f2
, decode_buffer
,
832 mlt_buffer_left
, mlt_buffer_right
);
833 idx
= (1 << p
->js_vlc_bits
) - 1;
840 * First part of subpacket decoding:
841 * decode raw stream bytes and read gain info.
843 * @param q pointer to the COOKContext
844 * @param inbuffer pointer to raw stream data
845 * @param gains_ptr array of current/prev gain pointers
847 static inline void decode_bytes_and_gain(COOKContext
*q
, COOKSubpacket
*p
,
848 const uint8_t *inbuffer
,
849 cook_gains
*gains_ptr
)
853 offset
= decode_bytes(inbuffer
, q
->decoded_bytes_buffer
,
854 p
->bits_per_subpacket
/ 8);
855 bitstream_init(&q
->bc
, q
->decoded_bytes_buffer
+ offset
,
856 p
->bits_per_subpacket
);
857 decode_gain_info(&q
->bc
, gains_ptr
->now
);
859 /* Swap current and previous gains */
860 FFSWAP(int *, gains_ptr
->now
, gains_ptr
->previous
);
864 * Saturate the output signal and interleave.
866 * @param q pointer to the COOKContext
867 * @param out pointer to the output vector
869 static void saturate_output_float(COOKContext
*q
, float *out
)
871 q
->adsp
.vector_clipf(out
, q
->mono_mdct_output
+ q
->samples_per_channel
,
872 FFALIGN(q
->samples_per_channel
, 8), -1.0f
, 1.0f
);
877 * Final part of subpacket decoding:
878 * Apply modulated lapped transform, gain compensation,
879 * clip and convert to integer.
881 * @param q pointer to the COOKContext
882 * @param decode_buffer pointer to the mlt coefficients
883 * @param gains_ptr array of current/prev gain pointers
884 * @param previous_buffer pointer to the previous buffer to be used for overlapping
885 * @param out pointer to the output buffer
887 static inline void mlt_compensate_output(COOKContext
*q
, float *decode_buffer
,
888 cook_gains
*gains_ptr
, float *previous_buffer
,
891 imlt_gain(q
, decode_buffer
, gains_ptr
, previous_buffer
);
893 q
->saturate_output(q
, out
);
898 * Cook subpacket decoding. This function returns one decoded subpacket,
899 * usually 1024 samples per channel.
901 * @param q pointer to the COOKContext
902 * @param inbuffer pointer to the inbuffer
903 * @param outbuffer pointer to the outbuffer
905 static int decode_subpacket(COOKContext
*q
, COOKSubpacket
*p
,
906 const uint8_t *inbuffer
, float **outbuffer
)
908 int sub_packet_size
= p
->size
;
911 memset(q
->decode_buffer_1
, 0, sizeof(q
->decode_buffer_1
));
912 decode_bytes_and_gain(q
, p
, inbuffer
, &p
->gains1
);
914 if (p
->joint_stereo
) {
915 if ((res
= joint_decode(q
, p
, q
->decode_buffer_1
, q
->decode_buffer_2
)) < 0)
918 if ((res
= mono_decode(q
, p
, q
->decode_buffer_1
)) < 0)
921 if (p
->num_channels
== 2) {
922 decode_bytes_and_gain(q
, p
, inbuffer
+ sub_packet_size
/ 2, &p
->gains2
);
923 if ((res
= mono_decode(q
, p
, q
->decode_buffer_2
)) < 0)
928 mlt_compensate_output(q
, q
->decode_buffer_1
, &p
->gains1
,
929 p
->mono_previous_buffer1
,
930 outbuffer ? outbuffer
[p
->ch_idx
] : NULL
);
932 if (p
->num_channels
== 2)
934 mlt_compensate_output(q
, q
->decode_buffer_2
, &p
->gains1
,
935 p
->mono_previous_buffer2
,
936 outbuffer ? outbuffer
[p
->ch_idx
+ 1] : NULL
);
938 mlt_compensate_output(q
, q
->decode_buffer_2
, &p
->gains2
,
939 p
->mono_previous_buffer2
,
940 outbuffer ? outbuffer
[p
->ch_idx
+ 1] : NULL
);
946 static int cook_decode_frame(AVCodecContext
*avctx
, void *data
,
947 int *got_frame_ptr
, AVPacket
*avpkt
)
949 AVFrame
*frame
= data
;
950 const uint8_t *buf
= avpkt
->data
;
951 int buf_size
= avpkt
->size
;
952 COOKContext
*q
= avctx
->priv_data
;
953 float **samples
= NULL
;
958 if (buf_size
< avctx
->block_align
)
961 /* get output buffer */
962 if (q
->discarded_packets
>= 2) {
963 frame
->nb_samples
= q
->samples_per_channel
;
964 if ((ret
= ff_get_buffer(avctx
, frame
, 0)) < 0) {
965 av_log(avctx
, AV_LOG_ERROR
, "get_buffer() failed\n");
968 samples
= (float **)frame
->extended_data
;
971 /* estimate subpacket sizes */
972 q
->subpacket
[0].size
= avctx
->block_align
;
974 for (i
= 1; i
< q
->num_subpackets
; i
++) {
975 q
->subpacket
[i
].size
= 2 * buf
[avctx
->block_align
- q
->num_subpackets
+ i
];
976 q
->subpacket
[0].size
-= q
->subpacket
[i
].size
+ 1;
977 if (q
->subpacket
[0].size
< 0) {
978 av_log(avctx
, AV_LOG_DEBUG
,
979 "frame subpacket size total > avctx->block_align!\n");
980 return AVERROR_INVALIDDATA
;
984 /* decode supbackets */
985 for (i
= 0; i
< q
->num_subpackets
; i
++) {
986 q
->subpacket
[i
].bits_per_subpacket
= (q
->subpacket
[i
].size
* 8) >>
987 q
->subpacket
[i
].bits_per_subpdiv
;
988 q
->subpacket
[i
].ch_idx
= chidx
;
989 av_log(avctx
, AV_LOG_DEBUG
,
990 "subpacket[%i] size %i js %i %i block_align %i\n",
991 i
, q
->subpacket
[i
].size
, q
->subpacket
[i
].joint_stereo
, offset
,
994 if ((ret
= decode_subpacket(q
, &q
->subpacket
[i
], buf
+ offset
, samples
)) < 0)
996 offset
+= q
->subpacket
[i
].size
;
997 chidx
+= q
->subpacket
[i
].num_channels
;
998 av_log(avctx
, AV_LOG_DEBUG
, "subpacket[%i] %i %i\n",
999 i
, q
->subpacket
[i
].size
* 8, bitstream_tell(&q
->bc
));
1002 /* Discard the first two frames: no valid audio. */
1003 if (q
->discarded_packets
< 2) {
1004 q
->discarded_packets
++;
1006 return avctx
->block_align
;
1011 return avctx
->block_align
;
1015 static void dump_cook_context(COOKContext
*q
)
1018 #define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
1019 ff_dlog(q
->avctx
, "COOKextradata\n");
1020 ff_dlog(q
->avctx
, "cookversion=%x\n", q
->subpacket
[0].cookversion
);
1021 if (q
->subpacket
[0].cookversion
> STEREO
) {
1022 PRINT("js_subband_start", q
->subpacket
[0].js_subband_start
);
1023 PRINT("js_vlc_bits", q
->subpacket
[0].js_vlc_bits
);
1025 ff_dlog(q
->avctx
, "COOKContext\n");
1026 PRINT("nb_channels", q
->avctx
->channels
);
1027 PRINT("bit_rate", q
->avctx
->bit_rate
);
1028 PRINT("sample_rate", q
->avctx
->sample_rate
);
1029 PRINT("samples_per_channel", q
->subpacket
[0].samples_per_channel
);
1030 PRINT("subbands", q
->subpacket
[0].subbands
);
1031 PRINT("js_subband_start", q
->subpacket
[0].js_subband_start
);
1032 PRINT("log2_numvector_size", q
->subpacket
[0].log2_numvector_size
);
1033 PRINT("numvector_size", q
->subpacket
[0].numvector_size
);
1034 PRINT("total_subbands", q
->subpacket
[0].total_subbands
);
1039 * Cook initialization
1041 * @param avctx pointer to the AVCodecContext
1043 static av_cold
int cook_decode_init(AVCodecContext
*avctx
)
1045 COOKContext
*q
= avctx
->priv_data
;
1048 unsigned int channel_mask
= 0;
1049 int samples_per_frame
;
1053 /* Take care of the codec specific extradata. */
1054 if (avctx
->extradata_size
< 8) {
1055 av_log(avctx
, AV_LOG_ERROR
, "Necessary extradata missing!\n");
1056 return AVERROR_INVALIDDATA
;
1058 av_log(avctx
, AV_LOG_DEBUG
, "codecdata_length=%d\n", avctx
->extradata_size
);
1060 bytestream2_init(&gb
, avctx
->extradata
, avctx
->extradata_size
);
1062 /* Take data from the AVCodecContext (RM container). */
1063 if (!avctx
->channels
) {
1064 av_log(avctx
, AV_LOG_ERROR
, "Invalid number of channels\n");
1065 return AVERROR_INVALIDDATA
;
1068 /* Initialize RNG. */
1069 av_lfg_init(&q
->random_state
, 0);
1071 ff_audiodsp_init(&q
->adsp
);
1073 while (bytestream2_get_bytes_left(&gb
)) {
1074 /* 8 for mono, 16 for stereo, ? for multichannel
1075 Swap to right endianness so we don't need to care later on. */
1076 q
->subpacket
[s
].cookversion
= bytestream2_get_be32(&gb
);
1077 samples_per_frame
= bytestream2_get_be16(&gb
);
1078 q
->subpacket
[s
].subbands
= bytestream2_get_be16(&gb
);
1079 bytestream2_get_be32(&gb
); // Unknown unused
1080 q
->subpacket
[s
].js_subband_start
= bytestream2_get_be16(&gb
);
1081 q
->subpacket
[s
].js_vlc_bits
= bytestream2_get_be16(&gb
);
1083 /* Initialize extradata related variables. */
1084 q
->subpacket
[s
].samples_per_channel
= samples_per_frame
/ avctx
->channels
;
1085 q
->subpacket
[s
].bits_per_subpacket
= avctx
->block_align
* 8;
1087 /* Initialize default data states. */
1088 q
->subpacket
[s
].log2_numvector_size
= 5;
1089 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
;
1090 q
->subpacket
[s
].num_channels
= 1;
1092 /* Initialize version-dependent variables */
1094 av_log(avctx
, AV_LOG_DEBUG
, "subpacket[%i].cookversion=%x\n", s
,
1095 q
->subpacket
[s
].cookversion
);
1096 q
->subpacket
[s
].joint_stereo
= 0;
1097 switch (q
->subpacket
[s
].cookversion
) {
1099 if (avctx
->channels
!= 1) {
1100 avpriv_request_sample(avctx
, "Container channels != 1");
1101 return AVERROR_PATCHWELCOME
;
1103 av_log(avctx
, AV_LOG_DEBUG
, "MONO\n");
1106 if (avctx
->channels
!= 1) {
1107 q
->subpacket
[s
].bits_per_subpdiv
= 1;
1108 q
->subpacket
[s
].num_channels
= 2;
1110 av_log(avctx
, AV_LOG_DEBUG
, "STEREO\n");
1113 if (avctx
->channels
!= 2) {
1114 avpriv_request_sample(avctx
, "Container channels != 2");
1115 return AVERROR_PATCHWELCOME
;
1117 av_log(avctx
, AV_LOG_DEBUG
, "JOINT_STEREO\n");
1118 if (avctx
->extradata_size
>= 16) {
1119 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
+
1120 q
->subpacket
[s
].js_subband_start
;
1121 q
->subpacket
[s
].joint_stereo
= 1;
1122 q
->subpacket
[s
].num_channels
= 2;
1124 if (q
->subpacket
[s
].samples_per_channel
> 256) {
1125 q
->subpacket
[s
].log2_numvector_size
= 6;
1127 if (q
->subpacket
[s
].samples_per_channel
> 512) {
1128 q
->subpacket
[s
].log2_numvector_size
= 7;
1132 av_log(avctx
, AV_LOG_DEBUG
, "MULTI_CHANNEL\n");
1133 channel_mask
|= q
->subpacket
[s
].channel_mask
= bytestream2_get_be32(&gb
);
1135 if (av_get_channel_layout_nb_channels(q
->subpacket
[s
].channel_mask
) > 1) {
1136 q
->subpacket
[s
].total_subbands
= q
->subpacket
[s
].subbands
+
1137 q
->subpacket
[s
].js_subband_start
;
1138 q
->subpacket
[s
].joint_stereo
= 1;
1139 q
->subpacket
[s
].num_channels
= 2;
1140 q
->subpacket
[s
].samples_per_channel
= samples_per_frame
>> 1;
1142 if (q
->subpacket
[s
].samples_per_channel
> 256) {
1143 q
->subpacket
[s
].log2_numvector_size
= 6;
1145 if (q
->subpacket
[s
].samples_per_channel
> 512) {
1146 q
->subpacket
[s
].log2_numvector_size
= 7;
1149 q
->subpacket
[s
].samples_per_channel
= samples_per_frame
;
1153 avpriv_request_sample(avctx
, "Cook version %d",
1154 q
->subpacket
[s
].cookversion
);
1155 return AVERROR_PATCHWELCOME
;
1158 if (s
> 1 && q
->subpacket
[s
].samples_per_channel
!= q
->samples_per_channel
) {
1159 av_log(avctx
, AV_LOG_ERROR
, "different number of samples per channel!\n");
1160 return AVERROR_INVALIDDATA
;
1162 q
->samples_per_channel
= q
->subpacket
[0].samples_per_channel
;
1165 /* Initialize variable relations */
1166 q
->subpacket
[s
].numvector_size
= (1 << q
->subpacket
[s
].log2_numvector_size
);
1168 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1169 if (q
->subpacket
[s
].total_subbands
> 53) {
1170 avpriv_request_sample(avctx
, "total_subbands > 53");
1171 return AVERROR_PATCHWELCOME
;
1174 if ((q
->subpacket
[s
].js_vlc_bits
> 6) ||
1175 (q
->subpacket
[s
].js_vlc_bits
< 2 * q
->subpacket
[s
].joint_stereo
)) {
1176 av_log(avctx
, AV_LOG_ERROR
, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1177 q
->subpacket
[s
].js_vlc_bits
, 2 * q
->subpacket
[s
].joint_stereo
);
1178 return AVERROR_INVALIDDATA
;
1181 if (q
->subpacket
[s
].subbands
> 50) {
1182 avpriv_request_sample(avctx
, "subbands > 50");
1183 return AVERROR_PATCHWELCOME
;
1185 q
->subpacket
[s
].gains1
.now
= q
->subpacket
[s
].gain_1
;
1186 q
->subpacket
[s
].gains1
.previous
= q
->subpacket
[s
].gain_2
;
1187 q
->subpacket
[s
].gains2
.now
= q
->subpacket
[s
].gain_3
;
1188 q
->subpacket
[s
].gains2
.previous
= q
->subpacket
[s
].gain_4
;
1190 q
->num_subpackets
++;
1192 if (s
> MAX_SUBPACKETS
) {
1193 avpriv_request_sample(avctx
, "subpackets > %d", MAX_SUBPACKETS
);
1194 return AVERROR_PATCHWELCOME
;
1197 /* Generate tables */
1200 init_cplscales_table(q
);
1202 if ((ret
= init_cook_vlc_tables(q
)))
1206 if (avctx
->block_align
>= UINT_MAX
/ 2)
1207 return AVERROR(EINVAL
);
1209 /* Pad the databuffer with:
1210 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1211 AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1212 q
->decoded_bytes_buffer
=
1213 av_mallocz(avctx
->block_align
1214 + DECODE_BYTES_PAD1(avctx
->block_align
)
1215 + AV_INPUT_BUFFER_PADDING_SIZE
);
1216 if (!q
->decoded_bytes_buffer
)
1217 return AVERROR(ENOMEM
);
1219 /* Initialize transform. */
1220 if ((ret
= init_cook_mlt(q
)))
1223 /* Initialize COOK signal arithmetic handling */
1225 q
->scalar_dequant
= scalar_dequant_float
;
1226 q
->decouple
= decouple_float
;
1227 q
->imlt_window
= imlt_window_float
;
1228 q
->interpolate
= interpolate_float
;
1229 q
->saturate_output
= saturate_output_float
;
1232 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1233 if (q
->samples_per_channel
!= 256 && q
->samples_per_channel
!= 512 &&
1234 q
->samples_per_channel
!= 1024) {
1235 avpriv_request_sample(avctx
, "samples_per_channel = %d",
1236 q
->samples_per_channel
);
1237 return AVERROR_PATCHWELCOME
;
1240 avctx
->sample_fmt
= AV_SAMPLE_FMT_FLTP
;
1242 avctx
->channel_layout
= channel_mask
;
1244 avctx
->channel_layout
= (avctx
->channels
== 2) ? AV_CH_LAYOUT_STEREO
: AV_CH_LAYOUT_MONO
;
1247 dump_cook_context(q
);
1252 AVCodec ff_cook_decoder
= {
1254 .long_name
= NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1255 .type
= AVMEDIA_TYPE_AUDIO
,
1256 .id
= AV_CODEC_ID_COOK
,
1257 .priv_data_size
= sizeof(COOKContext
),
1258 .init
= cook_decode_init
,
1259 .close
= cook_decode_close
,
1260 .decode
= cook_decode_frame
,
1261 .capabilities
= AV_CODEC_CAP_DR1
,
1262 .sample_fmts
= (const enum AVSampleFormat
[]) { AV_SAMPLE_FMT_FLTP
,
1263 AV_SAMPLE_FMT_NONE
},