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"
58 /* the different Cook versions */
59 #define MONO 0x1000001
60 #define STEREO 0x1000002
61 #define JOINT_STEREO 0x1000003
62 #define MC_COOK 0x2000000 // multichannel Cook, not supported
64 #define SUBBAND_SIZE 20
65 #define MAX_SUBPACKETS 5
67 typedef struct cook_gains {
72 typedef struct COOKSubpacket {
80 int samples_per_channel;
81 int log2_numvector_size;
82 unsigned int channel_mask;
85 int bits_per_subpacket;
88 int numvector_size; // 1 << log2_numvector_size;
90 float mono_previous_buffer1[1024];
91 float mono_previous_buffer2[1024];
101 typedef struct cook {
103 * The following 5 functions provide the lowlevel arithmetic on
104 * the internal audio buffers.
106 void (*scalar_dequant)(struct cook *q, int index, int quant_index,
107 int *subband_coef_index, int *subband_coef_sign,
110 void (*decouple)(struct cook *q,
114 float *decode_buffer,
115 float *mlt_buffer1, float *mlt_buffer2);
117 void (*imlt_window)(struct cook *q, float *buffer1,
118 cook_gains *gains_ptr, float *previous_buffer);
120 void (*interpolate)(struct cook *q, float *buffer,
121 int gain_index, int gain_index_next);
123 void (*saturate_output)(struct cook *q, float *out);
125 AVCodecContext* avctx;
126 AudioDSPContext adsp;
130 int samples_per_channel;
133 int discarded_packets;
140 VLC envelope_quant_index[13];
141 VLC sqvh[7]; // scalar quantization
143 /* generate tables and related variables */
144 int gain_size_factor;
145 float gain_table[23];
149 uint8_t* decoded_bytes_buffer;
150 DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
151 float decode_buffer_1[1024];
152 float decode_buffer_2[1024];
153 float decode_buffer_0[1060]; /* static allocation for joint decode */
155 const float *cplscales[5];
157 COOKSubpacket subpacket[MAX_SUBPACKETS];
160 static float pow2tab[127];
161 static float rootpow2tab[127];
163 /*************** init functions ***************/
165 /* table generator */
166 static av_cold void init_pow2table(void)
169 for (i = -63; i < 64; i++) {
170 pow2tab[63 + i] = pow(2, i);
171 rootpow2tab[63 + i] = sqrt(pow(2, i));
175 /* table generator */
176 static av_cold void init_gain_table(COOKContext *q)
179 q->gain_size_factor = q->samples_per_channel / 8;
180 for (i = 0; i < 23; i++)
181 q->gain_table[i] = pow(pow2tab[i + 52],
182 (1.0 / (double) q->gain_size_factor));
186 static av_cold int init_cook_vlc_tables(COOKContext *q)
191 for (i = 0; i < 13; i++) {
192 result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
193 envelope_quant_index_huffbits[i], 1, 1,
194 envelope_quant_index_huffcodes[i], 2, 2, 0);
196 av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
197 for (i = 0; i < 7; i++) {
198 result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
199 cvh_huffbits[i], 1, 1,
200 cvh_huffcodes[i], 2, 2, 0);
203 for (i = 0; i < q->num_subpackets; i++) {
204 if (q->subpacket[i].joint_stereo == 1) {
205 result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
206 (1 << q->subpacket[i].js_vlc_bits) - 1,
207 ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
208 ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
209 av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
213 av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
217 static av_cold int init_cook_mlt(COOKContext *q)
220 int mlt_size = q->samples_per_channel;
222 if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)
223 return AVERROR(ENOMEM);
225 /* Initialize the MLT window: simple sine window. */
226 ff_sine_window_init(q->mlt_window, mlt_size);
227 for (j = 0; j < mlt_size; j++)
228 q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
230 /* Initialize the MDCT. */
231 if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
232 av_free(q->mlt_window);
235 av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
236 av_log2(mlt_size) + 1);
241 static av_cold void init_cplscales_table(COOKContext *q)
244 for (i = 0; i < 5; i++)
245 q->cplscales[i] = cplscales[i];
248 /*************** init functions end ***********/
250 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
251 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
254 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
255 * Why? No idea, some checksum/error detection method maybe.
257 * Out buffer size: extra bytes are needed to cope with
258 * padding/misalignment.
259 * Subpackets passed to the decoder can contain two, consecutive
260 * half-subpackets, of identical but arbitrary size.
261 * 1234 1234 1234 1234 extraA extraB
262 * Case 1: AAAA BBBB 0 0
263 * Case 2: AAAA ABBB BB-- 3 3
264 * Case 3: AAAA AABB BBBB 2 2
265 * Case 4: AAAA AAAB BBBB BB-- 1 5
267 * Nice way to waste CPU cycles.
269 * @param inbuffer pointer to byte array of indata
270 * @param out pointer to byte array of outdata
271 * @param bytes number of bytes
273 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
275 static const uint32_t tab[4] = {
276 AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
277 AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
282 uint32_t *obuf = (uint32_t *) out;
283 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
284 * I'm too lazy though, should be something like
285 * for (i = 0; i < bitamount / 64; i++)
286 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
287 * Buffer alignment needs to be checked. */
289 off = (intptr_t) inbuffer & 3;
290 buf = (const uint32_t *) (inbuffer - off);
293 for (i = 0; i < bytes / 4; i++)
294 obuf[i] = c ^ buf[i];
299 static av_cold int cook_decode_close(AVCodecContext *avctx)
302 COOKContext *q = avctx->priv_data;
303 av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
305 /* Free allocated memory buffers. */
306 av_free(q->mlt_window);
307 av_free(q->decoded_bytes_buffer);
309 /* Free the transform. */
310 ff_mdct_end(&q->mdct_ctx);
312 /* Free the VLC tables. */
313 for (i = 0; i < 13; i++)
314 ff_free_vlc(&q->envelope_quant_index[i]);
315 for (i = 0; i < 7; i++)
316 ff_free_vlc(&q->sqvh[i]);
317 for (i = 0; i < q->num_subpackets; i++)
318 ff_free_vlc(&q->subpacket[i].channel_coupling);
320 av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
326 * Fill the gain array for the timedomain quantization.
328 * @param bc pointer to the BitstreamContext
329 * @param gaininfo array[9] of gain indexes
331 static void decode_gain_info(BitstreamContext *bc, int *gaininfo)
335 while (bitstream_read_bit(bc)) {
339 n = bitstream_tell(bc) - 1; // amount of elements * 2 to update
343 int index = bitstream_read(bc, 3);
344 int gain = bitstream_read_bit(bc) ? bitstream_read(bc, 4) - 7 : -1;
347 gaininfo[i++] = gain;
354 * Create the quant index table needed for the envelope.
356 * @param q pointer to the COOKContext
357 * @param quant_index_table pointer to the array
359 static int decode_envelope(COOKContext *q, COOKSubpacket *p,
360 int *quant_index_table)
364 quant_index_table[0] = bitstream_read(&q->bc, 6) - 6; // This is used later in categorize
366 for (i = 1; i < p->total_subbands; i++) {
368 if (i >= p->js_subband_start * 2) {
369 vlc_index -= p->js_subband_start;
376 vlc_index = 13; // the VLC tables >13 are identical to No. 13
378 j = bitstream_read_vlc(&q->bc, q->envelope_quant_index[vlc_index - 1].table,
379 q->envelope_quant_index[vlc_index - 1].bits, 2);
380 quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
381 if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
382 av_log(q->avctx, AV_LOG_ERROR,
383 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
384 quant_index_table[i], i);
385 return AVERROR_INVALIDDATA;
393 * Calculate the category and category_index vector.
395 * @param q pointer to the COOKContext
396 * @param quant_index_table pointer to the array
397 * @param category pointer to the category array
398 * @param category_index pointer to the category_index array
400 static void categorize(COOKContext *q, COOKSubpacket *p, int *quant_index_table,
401 int *category, int *category_index)
403 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
404 int exp_index2[102] = { 0 };
405 int exp_index1[102] = { 0 };
407 int tmp_categorize_array[128 * 2] = { 0 };
408 int tmp_categorize_array1_idx = p->numvector_size;
409 int tmp_categorize_array2_idx = p->numvector_size;
411 bits_left = p->bits_per_subpacket - bitstream_tell(&q->bc);
413 if (bits_left > q->samples_per_channel)
414 bits_left = q->samples_per_channel +
415 ((bits_left - q->samples_per_channel) * 5) / 8;
420 for (i = 32; i > 0; i = i / 2) {
423 for (j = p->total_subbands; j > 0; j--) {
424 exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
426 num_bits += expbits_tab[exp_idx];
428 if (num_bits >= bits_left - 32)
432 /* Calculate total number of bits. */
434 for (i = 0; i < p->total_subbands; i++) {
435 exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
436 num_bits += expbits_tab[exp_idx];
437 exp_index1[i] = exp_idx;
438 exp_index2[i] = exp_idx;
440 tmpbias1 = tmpbias2 = num_bits;
442 for (j = 1; j < p->numvector_size; j++) {
443 if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
446 for (i = 0; i < p->total_subbands; i++) {
447 if (exp_index1[i] < 7) {
448 v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
457 tmp_categorize_array[tmp_categorize_array1_idx++] = index;
458 tmpbias1 -= expbits_tab[exp_index1[index]] -
459 expbits_tab[exp_index1[index] + 1];
464 for (i = 0; i < p->total_subbands; i++) {
465 if (exp_index2[i] > 0) {
466 v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
475 tmp_categorize_array[--tmp_categorize_array2_idx] = index;
476 tmpbias2 -= expbits_tab[exp_index2[index]] -
477 expbits_tab[exp_index2[index] - 1];
482 for (i = 0; i < p->total_subbands; i++)
483 category[i] = exp_index2[i];
485 for (i = 0; i < p->numvector_size - 1; i++)
486 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
491 * Expand the category vector.
493 * @param q pointer to the COOKContext
494 * @param category pointer to the category array
495 * @param category_index pointer to the category_index array
497 static inline void expand_category(COOKContext *q, int *category,
501 for (i = 0; i < q->num_vectors; i++)
503 int idx = category_index[i];
504 if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
510 * The real requantization of the mltcoefs
512 * @param q pointer to the COOKContext
514 * @param quant_index quantisation index
515 * @param subband_coef_index array of indexes to quant_centroid_tab
516 * @param subband_coef_sign signs of coefficients
517 * @param mlt_p pointer into the mlt buffer
519 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
520 int *subband_coef_index, int *subband_coef_sign,
526 for (i = 0; i < SUBBAND_SIZE; i++) {
527 if (subband_coef_index[i]) {
528 f1 = quant_centroid_tab[index][subband_coef_index[i]];
529 if (subband_coef_sign[i])
532 /* noise coding if subband_coef_index[i] == 0 */
533 f1 = dither_tab[index];
534 if (av_lfg_get(&q->random_state) < 0x80000000)
537 mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
541 * Unpack the subband_coef_index and subband_coef_sign vectors.
543 * @param q pointer to the COOKContext
544 * @param category pointer to the category array
545 * @param subband_coef_index array of indexes to quant_centroid_tab
546 * @param subband_coef_sign signs of coefficients
548 static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
549 int *subband_coef_index, int *subband_coef_sign)
552 int vlc, vd, tmp, result;
554 vd = vd_tab[category];
556 for (i = 0; i < vpr_tab[category]; i++) {
557 vlc = bitstream_read_vlc(&q->bc, q->sqvh[category].table, q->sqvh[category].bits, 3);
558 if (p->bits_per_subpacket < bitstream_tell(&q->bc)) {
562 for (j = vd - 1; j >= 0; j--) {
563 tmp = (vlc * invradix_tab[category]) / 0x100000;
564 subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
567 for (j = 0; j < vd; j++) {
568 if (subband_coef_index[i * vd + j]) {
569 if (bitstream_tell(&q->bc) < p->bits_per_subpacket) {
570 subband_coef_sign[i * vd + j] = bitstream_read_bit(&q->bc);
573 subband_coef_sign[i * vd + j] = 0;
576 subband_coef_sign[i * vd + j] = 0;
585 * Fill the mlt_buffer with mlt coefficients.
587 * @param q pointer to the COOKContext
588 * @param category pointer to the category array
589 * @param quant_index_table pointer to the array
590 * @param mlt_buffer pointer to mlt coefficients
592 static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
593 int *quant_index_table, float *mlt_buffer)
595 /* A zero in this table means that the subband coefficient is
596 random noise coded. */
597 int subband_coef_index[SUBBAND_SIZE];
598 /* A zero in this table means that the subband coefficient is a
599 positive multiplicator. */
600 int subband_coef_sign[SUBBAND_SIZE];
604 for (band = 0; band < p->total_subbands; band++) {
605 index = category[band];
606 if (category[band] < 7) {
607 if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
609 for (j = 0; j < p->total_subbands; j++)
610 category[band + j] = 7;
614 memset(subband_coef_index, 0, sizeof(subband_coef_index));
615 memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
617 q->scalar_dequant(q, index, quant_index_table[band],
618 subband_coef_index, subband_coef_sign,
619 &mlt_buffer[band * SUBBAND_SIZE]);
622 /* FIXME: should this be removed, or moved into loop above? */
623 if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
628 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
630 int category_index[128] = { 0 };
631 int category[128] = { 0 };
632 int quant_index_table[102];
635 if ((res = decode_envelope(q, p, quant_index_table)) < 0)
637 q->num_vectors = bitstream_read(&q->bc, p->log2_numvector_size);
638 categorize(q, p, quant_index_table, category, category_index);
639 expand_category(q, category, category_index);
640 decode_vectors(q, p, category, quant_index_table, mlt_buffer);
647 * the actual requantization of the timedomain samples
649 * @param q pointer to the COOKContext
650 * @param buffer pointer to the timedomain buffer
651 * @param gain_index index for the block multiplier
652 * @param gain_index_next index for the next block multiplier
654 static void interpolate_float(COOKContext *q, float *buffer,
655 int gain_index, int gain_index_next)
659 fc1 = pow2tab[gain_index + 63];
661 if (gain_index == gain_index_next) { // static gain
662 for (i = 0; i < q->gain_size_factor; i++)
664 } else { // smooth gain
665 fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
666 for (i = 0; i < q->gain_size_factor; i++) {
674 * Apply transform window, overlap buffers.
676 * @param q pointer to the COOKContext
677 * @param inbuffer pointer to the mltcoefficients
678 * @param gains_ptr current and previous gains
679 * @param previous_buffer pointer to the previous buffer to be used for overlapping
681 static void imlt_window_float(COOKContext *q, float *inbuffer,
682 cook_gains *gains_ptr, float *previous_buffer)
684 const float fc = pow2tab[gains_ptr->previous[0] + 63];
686 /* The weird thing here, is that the two halves of the time domain
687 * buffer are swapped. Also, the newest data, that we save away for
688 * next frame, has the wrong sign. Hence the subtraction below.
689 * Almost sounds like a complex conjugate/reverse data/FFT effect.
692 /* Apply window and overlap */
693 for (i = 0; i < q->samples_per_channel; i++)
694 inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
695 previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
699 * The modulated lapped transform, this takes transform coefficients
700 * and transforms them into timedomain samples.
701 * Apply transform window, overlap buffers, apply gain profile
702 * and buffer management.
704 * @param q pointer to the COOKContext
705 * @param inbuffer pointer to the mltcoefficients
706 * @param gains_ptr current and previous gains
707 * @param previous_buffer pointer to the previous buffer to be used for overlapping
709 static void imlt_gain(COOKContext *q, float *inbuffer,
710 cook_gains *gains_ptr, float *previous_buffer)
712 float *buffer0 = q->mono_mdct_output;
713 float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
716 /* Inverse modified discrete cosine transform */
717 q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
719 q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
721 /* Apply gain profile */
722 for (i = 0; i < 8; i++)
723 if (gains_ptr->now[i] || gains_ptr->now[i + 1])
724 q->interpolate(q, &buffer1[q->gain_size_factor * i],
725 gains_ptr->now[i], gains_ptr->now[i + 1]);
727 /* Save away the current to be previous block. */
728 memcpy(previous_buffer, buffer0,
729 q->samples_per_channel * sizeof(*previous_buffer));
734 * function for getting the jointstereo coupling information
736 * @param q pointer to the COOKContext
737 * @param decouple_tab decoupling array
739 static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
742 int vlc = bitstream_read_bit(&q->bc);
743 int start = cplband[p->js_subband_start];
744 int end = cplband[p->subbands - 1];
745 int length = end - start + 1;
751 for (i = 0; i < length; i++)
752 decouple_tab[start + i] =
753 bitstream_read_vlc(&q->bc,
754 p->channel_coupling.table,
755 p->channel_coupling.bits, 2);
757 for (i = 0; i < length; i++)
758 decouple_tab[start + i] = bitstream_read(&q->bc, p->js_vlc_bits);
762 * function decouples a pair of signals from a single signal via multiplication.
764 * @param q pointer to the COOKContext
765 * @param subband index of the current subband
766 * @param f1 multiplier for channel 1 extraction
767 * @param f2 multiplier for channel 2 extraction
768 * @param decode_buffer input buffer
769 * @param mlt_buffer1 pointer to left channel mlt coefficients
770 * @param mlt_buffer2 pointer to right channel mlt coefficients
772 static void decouple_float(COOKContext *q,
776 float *decode_buffer,
777 float *mlt_buffer1, float *mlt_buffer2)
780 for (j = 0; j < SUBBAND_SIZE; j++) {
781 tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
782 mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
783 mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
788 * function for decoding joint stereo data
790 * @param q pointer to the COOKContext
791 * @param mlt_buffer1 pointer to left channel mlt coefficients
792 * @param mlt_buffer2 pointer to right channel mlt coefficients
794 static int joint_decode(COOKContext *q, COOKSubpacket *p,
795 float *mlt_buffer_left, float *mlt_buffer_right)
798 int decouple_tab[SUBBAND_SIZE] = { 0 };
799 float *decode_buffer = q->decode_buffer_0;
802 const float *cplscale;
804 memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
806 /* Make sure the buffers are zeroed out. */
807 memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
808 memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
809 decouple_info(q, p, decouple_tab);
810 if ((res = mono_decode(q, p, decode_buffer)) < 0)
813 /* The two channels are stored interleaved in decode_buffer. */
814 for (i = 0; i < p->js_subband_start; i++) {
815 for (j = 0; j < SUBBAND_SIZE; j++) {
816 mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
817 mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
821 /* When we reach js_subband_start (the higher frequencies)
822 the coefficients are stored in a coupling scheme. */
823 idx = (1 << p->js_vlc_bits) - 1;
824 for (i = p->js_subband_start; i < p->subbands; i++) {
825 cpl_tmp = cplband[i];
826 idx -= decouple_tab[cpl_tmp];
827 cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
828 f1 = cplscale[decouple_tab[cpl_tmp] + 1];
830 q->decouple(q, p, i, f1, f2, decode_buffer,
831 mlt_buffer_left, mlt_buffer_right);
832 idx = (1 << p->js_vlc_bits) - 1;
839 * First part of subpacket decoding:
840 * decode raw stream bytes and read gain info.
842 * @param q pointer to the COOKContext
843 * @param inbuffer pointer to raw stream data
844 * @param gains_ptr array of current/prev gain pointers
846 static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
847 const uint8_t *inbuffer,
848 cook_gains *gains_ptr)
852 offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
853 p->bits_per_subpacket / 8);
854 bitstream_init(&q->bc, q->decoded_bytes_buffer + offset,
855 p->bits_per_subpacket);
856 decode_gain_info(&q->bc, gains_ptr->now);
858 /* Swap current and previous gains */
859 FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
863 * Saturate the output signal and interleave.
865 * @param q pointer to the COOKContext
866 * @param out pointer to the output vector
868 static void saturate_output_float(COOKContext *q, float *out)
870 q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
871 FFALIGN(q->samples_per_channel, 8), -1.0f, 1.0f);
876 * Final part of subpacket decoding:
877 * Apply modulated lapped transform, gain compensation,
878 * clip and convert to integer.
880 * @param q pointer to the COOKContext
881 * @param decode_buffer pointer to the mlt coefficients
882 * @param gains_ptr array of current/prev gain pointers
883 * @param previous_buffer pointer to the previous buffer to be used for overlapping
884 * @param out pointer to the output buffer
886 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
887 cook_gains *gains_ptr, float *previous_buffer,
890 imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
892 q->saturate_output(q, out);
897 * Cook subpacket decoding. This function returns one decoded subpacket,
898 * usually 1024 samples per channel.
900 * @param q pointer to the COOKContext
901 * @param inbuffer pointer to the inbuffer
902 * @param outbuffer pointer to the outbuffer
904 static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
905 const uint8_t *inbuffer, float **outbuffer)
907 int sub_packet_size = p->size;
910 memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
911 decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
913 if (p->joint_stereo) {
914 if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
917 if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
920 if (p->num_channels == 2) {
921 decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
922 if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
927 mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
928 p->mono_previous_buffer1,
929 outbuffer ? outbuffer[p->ch_idx] : NULL);
931 if (p->num_channels == 2)
933 mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
934 p->mono_previous_buffer2,
935 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
937 mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
938 p->mono_previous_buffer2,
939 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
945 static int cook_decode_frame(AVCodecContext *avctx, void *data,
946 int *got_frame_ptr, AVPacket *avpkt)
948 AVFrame *frame = data;
949 const uint8_t *buf = avpkt->data;
950 int buf_size = avpkt->size;
951 COOKContext *q = avctx->priv_data;
952 float **samples = NULL;
957 if (buf_size < avctx->block_align)
960 /* get output buffer */
961 if (q->discarded_packets >= 2) {
962 frame->nb_samples = q->samples_per_channel;
963 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
964 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
967 samples = (float **)frame->extended_data;
970 /* estimate subpacket sizes */
971 q->subpacket[0].size = avctx->block_align;
973 for (i = 1; i < q->num_subpackets; i++) {
974 q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
975 q->subpacket[0].size -= q->subpacket[i].size + 1;
976 if (q->subpacket[0].size < 0) {
977 av_log(avctx, AV_LOG_DEBUG,
978 "frame subpacket size total > avctx->block_align!\n");
979 return AVERROR_INVALIDDATA;
983 /* decode supbackets */
984 for (i = 0; i < q->num_subpackets; i++) {
985 q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
986 q->subpacket[i].bits_per_subpdiv;
987 q->subpacket[i].ch_idx = chidx;
988 av_log(avctx, AV_LOG_DEBUG,
989 "subpacket[%i] size %i js %i %i block_align %i\n",
990 i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
993 if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
995 offset += q->subpacket[i].size;
996 chidx += q->subpacket[i].num_channels;
997 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
998 i, q->subpacket[i].size * 8, bitstream_tell(&q->bc));
1001 /* Discard the first two frames: no valid audio. */
1002 if (q->discarded_packets < 2) {
1003 q->discarded_packets++;
1005 return avctx->block_align;
1010 return avctx->block_align;
1014 static void dump_cook_context(COOKContext *q)
1017 #define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
1018 ff_dlog(q->avctx, "COOKextradata\n");
1019 ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1020 if (q->subpacket[0].cookversion > STEREO) {
1021 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1022 PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1024 ff_dlog(q->avctx, "COOKContext\n");
1025 PRINT("nb_channels", q->avctx->channels);
1026 PRINT("bit_rate", q->avctx->bit_rate);
1027 PRINT("sample_rate", q->avctx->sample_rate);
1028 PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1029 PRINT("subbands", q->subpacket[0].subbands);
1030 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1031 PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1032 PRINT("numvector_size", q->subpacket[0].numvector_size);
1033 PRINT("total_subbands", q->subpacket[0].total_subbands);
1038 * Cook initialization
1040 * @param avctx pointer to the AVCodecContext
1042 static av_cold int cook_decode_init(AVCodecContext *avctx)
1044 COOKContext *q = avctx->priv_data;
1047 unsigned int channel_mask = 0;
1048 int samples_per_frame;
1052 /* Take care of the codec specific extradata. */
1053 if (avctx->extradata_size < 8) {
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 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1061 /* Take data from the AVCodecContext (RM container). */
1062 if (!avctx->channels) {
1063 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1064 return AVERROR_INVALIDDATA;
1067 /* Initialize RNG. */
1068 av_lfg_init(&q->random_state, 0);
1070 ff_audiodsp_init(&q->adsp);
1072 while (bytestream2_get_bytes_left(&gb)) {
1073 /* 8 for mono, 16 for stereo, ? for multichannel
1074 Swap to right endianness so we don't need to care later on. */
1075 q->subpacket[s].cookversion = bytestream2_get_be32(&gb);
1076 samples_per_frame = bytestream2_get_be16(&gb);
1077 q->subpacket[s].subbands = bytestream2_get_be16(&gb);
1078 bytestream2_get_be32(&gb); // Unknown unused
1079 q->subpacket[s].js_subband_start = bytestream2_get_be16(&gb);
1080 q->subpacket[s].js_vlc_bits = bytestream2_get_be16(&gb);
1082 /* Initialize extradata related variables. */
1083 q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1084 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1086 /* Initialize default data states. */
1087 q->subpacket[s].log2_numvector_size = 5;
1088 q->subpacket[s].total_subbands = q->subpacket[s].subbands;
1089 q->subpacket[s].num_channels = 1;
1091 /* Initialize version-dependent variables */
1093 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1094 q->subpacket[s].cookversion);
1095 q->subpacket[s].joint_stereo = 0;
1096 switch (q->subpacket[s].cookversion) {
1098 if (avctx->channels != 1) {
1099 avpriv_request_sample(avctx, "Container channels != 1");
1100 return AVERROR_PATCHWELCOME;
1102 av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1105 if (avctx->channels != 1) {
1106 q->subpacket[s].bits_per_subpdiv = 1;
1107 q->subpacket[s].num_channels = 2;
1109 av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1112 if (avctx->channels != 2) {
1113 avpriv_request_sample(avctx, "Container channels != 2");
1114 return AVERROR_PATCHWELCOME;
1116 av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1117 if (avctx->extradata_size >= 16) {
1118 q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1119 q->subpacket[s].js_subband_start;
1120 q->subpacket[s].joint_stereo = 1;
1121 q->subpacket[s].num_channels = 2;
1123 if (q->subpacket[s].samples_per_channel > 256) {
1124 q->subpacket[s].log2_numvector_size = 6;
1126 if (q->subpacket[s].samples_per_channel > 512) {
1127 q->subpacket[s].log2_numvector_size = 7;
1131 av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1132 channel_mask |= q->subpacket[s].channel_mask = bytestream2_get_be32(&gb);
1134 if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {
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;
1139 q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1141 if (q->subpacket[s].samples_per_channel > 256) {
1142 q->subpacket[s].log2_numvector_size = 6;
1144 if (q->subpacket[s].samples_per_channel > 512) {
1145 q->subpacket[s].log2_numvector_size = 7;
1148 q->subpacket[s].samples_per_channel = samples_per_frame;
1152 avpriv_request_sample(avctx, "Cook version %d",
1153 q->subpacket[s].cookversion);
1154 return AVERROR_PATCHWELCOME;
1157 if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1158 av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1159 return AVERROR_INVALIDDATA;
1161 q->samples_per_channel = q->subpacket[0].samples_per_channel;
1164 /* Initialize variable relations */
1165 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
1167 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1168 if (q->subpacket[s].total_subbands > 53) {
1169 avpriv_request_sample(avctx, "total_subbands > 53");
1170 return AVERROR_PATCHWELCOME;
1173 if ((q->subpacket[s].js_vlc_bits > 6) ||
1174 (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1175 av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1176 q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1177 return AVERROR_INVALIDDATA;
1180 if (q->subpacket[s].subbands > 50) {
1181 avpriv_request_sample(avctx, "subbands > 50");
1182 return AVERROR_PATCHWELCOME;
1184 q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1185 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1186 q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1187 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1189 q->num_subpackets++;
1191 if (s > MAX_SUBPACKETS) {
1192 avpriv_request_sample(avctx, "subpackets > %d", MAX_SUBPACKETS);
1193 return AVERROR_PATCHWELCOME;
1196 /* Generate tables */
1199 init_cplscales_table(q);
1201 if ((ret = init_cook_vlc_tables(q)))
1205 if (avctx->block_align >= UINT_MAX / 2)
1206 return AVERROR(EINVAL);
1208 /* Pad the databuffer with:
1209 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1210 AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1211 q->decoded_bytes_buffer =
1212 av_mallocz(avctx->block_align
1213 + DECODE_BYTES_PAD1(avctx->block_align)
1214 + AV_INPUT_BUFFER_PADDING_SIZE);
1215 if (!q->decoded_bytes_buffer)
1216 return AVERROR(ENOMEM);
1218 /* Initialize transform. */
1219 if ((ret = init_cook_mlt(q)))
1222 /* Initialize COOK signal arithmetic handling */
1224 q->scalar_dequant = scalar_dequant_float;
1225 q->decouple = decouple_float;
1226 q->imlt_window = imlt_window_float;
1227 q->interpolate = interpolate_float;
1228 q->saturate_output = saturate_output_float;
1231 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1232 if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1233 q->samples_per_channel != 1024) {
1234 avpriv_request_sample(avctx, "samples_per_channel = %d",
1235 q->samples_per_channel);
1236 return AVERROR_PATCHWELCOME;
1239 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1241 avctx->channel_layout = channel_mask;
1243 avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1246 dump_cook_context(q);
1251 AVCodec ff_cook_decoder = {
1253 .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1254 .type = AVMEDIA_TYPE_AUDIO,
1255 .id = AV_CODEC_ID_COOK,
1256 .priv_data_size = sizeof(COOKContext),
1257 .init = cook_decode_init,
1258 .close = cook_decode_close,
1259 .decode = cook_decode_frame,
1260 .capabilities = AV_CODEC_CAP_DR1,
1261 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1262 AV_SAMPLE_FMT_NONE },