2 * Atrac 3 compatible decoder
3 * Copyright (c) 2006-2008 Maxim Poliakovski
4 * Copyright (c) 2006-2008 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
24 * @file libavcodec/atrac3.c
25 * Atrac 3 compatible decoder.
26 * This decoder handles Sony's ATRAC3 data.
28 * Container formats used to store atrac 3 data:
29 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
31 * To use this decoder, a calling application must supply the extradata
32 * bytes provided in the containers above.
40 #include "atrac3data.h"
41 #include "atrac3data_fixed.h"
42 #include "fixp_math.h"
43 #include "../lib/mdct2.h"
45 #define JOINT_STEREO 0x12
53 /* FFMAX/MIN/SWAP and av_clip were taken from libavutil/common.h */
54 #define FFMAX(a,b) ((a) > (b) ? (a) : (b))
55 #define FFMIN(a,b) ((a) > (b) ? (b) : (a))
56 #define FFSWAP(type,a,b) do{type SWAP_tmp= b; b= a; a= SWAP_tmp;}while(0)
58 static int32_t qmf_window
[48] IBSS_ATTR
;
59 static VLC spectral_coeff_tab
[7];
60 static channel_unit channel_units
[2] IBSS_ATTR_LARGE_IRAM
;
63 * Matrixing within quadrature mirror synthesis filter.
65 * @param p3 output buffer
66 * @param inlo lower part of spectrum
67 * @param inhi higher part of spectrum
68 * @param nIn size of spectrum buffer
73 atrac3_iqmf_matrixing(int32_t *p3
,
79 atrac3_iqmf_matrixing(int32_t *p3
,
85 for(i
=0; i
<nIn
; i
+=2){
86 p3
[2*i
+0] = inlo
[i
] + inhi
[i
];
87 p3
[2*i
+1] = inlo
[i
] - inhi
[i
];
88 p3
[2*i
+2] = inlo
[i
+1] + inhi
[i
+1];
89 p3
[2*i
+3] = inlo
[i
+1] - inhi
[i
+1];
95 * Matrixing within quadrature mirror synthesis filter.
97 * @param out output buffer
98 * @param in input buffer
99 * @param win windowing coefficients
100 * @param nIn size of spectrum buffer
101 * Reference implementation:
103 * for (j = nIn; j != 0; j--) {
104 * s1 = fixmul32(in[0], win[0]);
105 * s2 = fixmul32(in[1], win[1]);
106 * for (i = 2; i < 48; i += 2) {
107 * s1 += fixmul31(in[i ], win[i ]);
108 * s2 += fixmul31(in[i+1], win[i+1]);
119 atrac3_iqmf_dewindowing(int32_t *out
,
125 atrac3_iqmf_dewindowing(int32_t *out
,
130 int32_t i
, j
, s1
, s2
;
132 for (j
= nIn
; j
!= 0; j
--) {
135 s1
= fixmul31(win
[i
], in
[i
]); i
++;
136 s2
= fixmul31(win
[i
], in
[i
]); i
++;
137 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
138 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
139 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
140 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
141 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
142 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
144 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
145 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
146 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
147 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
148 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
149 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
150 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
151 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
153 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
154 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
155 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
156 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
157 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
158 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
159 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
160 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
162 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
163 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
164 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
165 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
166 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
167 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
168 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
169 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
171 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
172 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
173 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
174 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
175 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
176 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
177 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
178 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
180 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
181 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
182 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
183 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
184 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
185 s2
+= fixmul31(win
[i
], in
[i
]); i
++;
186 s1
+= fixmul31(win
[i
], in
[i
]); i
++;
187 s2
+= fixmul31(win
[i
], in
[i
]);
201 * @param buffer sample buffer
202 * @param win window coefficients
206 atrac3_imdct_windowing(int32_t *buffer
,
210 /* win[0..127] = win[511..384], win[128..383] = 1 */
211 for(i
= 0; i
<128; i
++) {
212 buffer
[ i
] = fixmul31(win
[i
], buffer
[ i
]);
213 buffer
[511-i
] = fixmul31(win
[i
], buffer
[511-i
]);
218 * Quadrature mirror synthesis filter.
220 * @param inlo lower part of spectrum
221 * @param inhi higher part of spectrum
222 * @param nIn size of spectrum buffer
223 * @param pOut out buffer
224 * @param delayBuf delayBuf buffer
225 * @param temp temp buffer
228 static void iqmf (int32_t *inlo
, int32_t *inhi
, unsigned int nIn
, int32_t *pOut
, int32_t *delayBuf
, int32_t *temp
)
230 /* Restore the delay buffer */
231 memcpy(temp
, delayBuf
, 46*sizeof(int32_t));
233 /* loop1: matrixing */
234 atrac3_iqmf_matrixing(temp
+ 46, inlo
, inhi
, nIn
);
236 /* loop2: dewindowing */
237 atrac3_iqmf_dewindowing(pOut
, temp
, qmf_window
, nIn
);
239 /* Save the delay buffer */
240 memcpy(delayBuf
, temp
+ (nIn
<< 1), 46*sizeof(int32_t));
244 * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
245 * caused by the reverse spectra of the QMF.
247 * @param pInput float input
248 * @param pOutput float output
249 * @param odd_band 1 if the band is an odd band
252 static void IMLT(int32_t *pInput
, int32_t *pOutput
)
254 /* Apply the imdct. */
255 mdct_backward(512, pInput
, pOutput
);
258 atrac3_imdct_windowing(pOutput
, window_lookup
);
263 * Atrac 3 indata descrambling, only used for data coming from the rm container
265 * @param in pointer to 8 bit array of indata
266 * @param bits amount of bits
267 * @param out pointer to 8 bit array of outdata
270 static int decode_bytes(const uint8_t* inbuffer
, uint8_t* out
, int bytes
){
274 uint32_t* obuf
= (uint32_t*) out
;
276 #if ((defined(TEST) || defined(SIMULATOR)) && !defined(CPU_ARM))
277 off
= 0; //no check for memory alignment of inbuffer
279 off
= (intptr_t)inbuffer
& 3;
281 buf
= (const uint32_t*) (inbuffer
- off
);
283 c
= be2me_32((0x537F6103 >> (off
*8)) | (0x537F6103 << (32-(off
*8))));
285 for (i
= 0; i
< bytes
/4; i
++)
286 obuf
[i
] = c
^ buf
[i
];
292 static void init_atrac3_transforms(void) {
296 /* Generate the mdct window, for details see
297 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
299 /* mdct window had been generated and saved as a lookup table in atrac3data_fixed.h */
301 /* Generate the QMF window. */
302 for (i
=0 ; i
<24; i
++) {
303 s
= qmf_48tap_half_fix
[i
] << 1;
305 qmf_window
[47 - i
] = s
;
312 * @param gb the GetBit context
313 * @param selector what table is the output values coded with
314 * @param codingFlag constant length coding or variable length coding
315 * @param mantissas mantissa output table
316 * @param numCodes amount of values to get
319 static void readQuantSpectralCoeffs (GetBitContext
*gb
, int selector
, int codingFlag
, int* mantissas
, int numCodes
)
321 int numBits
, cnt
, code
, huffSymb
;
326 if (codingFlag
!= 0) {
327 /* constant length coding (CLC) */
328 numBits
= CLCLengthTab
[selector
];
331 for (cnt
= 0; cnt
< numCodes
; cnt
++) {
333 code
= get_sbits(gb
, numBits
);
336 mantissas
[cnt
] = code
;
339 for (cnt
= 0; cnt
< numCodes
; cnt
++) {
341 code
= get_bits(gb
, numBits
); //numBits is always 4 in this case
344 mantissas
[cnt
*2] = seTab_0
[code
>> 2];
345 mantissas
[cnt
*2+1] = seTab_0
[code
& 3];
349 /* variable length coding (VLC) */
351 for (cnt
= 0; cnt
< numCodes
; cnt
++) {
352 huffSymb
= get_vlc2(gb
, spectral_coeff_tab
[selector
-1].table
, spectral_coeff_tab
[selector
-1].bits
, 3);
354 code
= huffSymb
>> 1;
357 mantissas
[cnt
] = code
;
360 for (cnt
= 0; cnt
< numCodes
; cnt
++) {
361 huffSymb
= get_vlc2(gb
, spectral_coeff_tab
[selector
-1].table
, spectral_coeff_tab
[selector
-1].bits
, 3);
362 mantissas
[cnt
*2] = decTable1
[huffSymb
*2];
363 mantissas
[cnt
*2+1] = decTable1
[huffSymb
*2+1];
370 * Restore the quantized band spectrum coefficients
372 * @param gb the GetBit context
373 * @param pOut decoded band spectrum
374 * @return outSubbands subband counter, fix for broken specification/files
377 static int decodeSpectrum (GetBitContext
*gb
, int32_t *pOut
)
379 int numSubbands
, codingMode
, cnt
, first
, last
, subbWidth
, *pIn
;
380 int subband_vlc_index
[32], SF_idxs
[32];
384 numSubbands
= get_bits(gb
, 5); // number of coded subbands
385 codingMode
= get_bits1(gb
); // coding Mode: 0 - VLC/ 1-CLC
387 /* Get the VLC selector table for the subbands, 0 means not coded. */
388 for (cnt
= 0; cnt
<= numSubbands
; cnt
++)
389 subband_vlc_index
[cnt
] = get_bits(gb
, 3);
391 /* Read the scale factor indexes from the stream. */
392 for (cnt
= 0; cnt
<= numSubbands
; cnt
++) {
393 if (subband_vlc_index
[cnt
] != 0)
394 SF_idxs
[cnt
] = get_bits(gb
, 6);
397 for (cnt
= 0; cnt
<= numSubbands
; cnt
++) {
398 first
= subbandTab
[cnt
];
399 last
= subbandTab
[cnt
+1];
401 subbWidth
= last
- first
;
403 if (subband_vlc_index
[cnt
] != 0) {
404 /* Decode spectral coefficients for this subband. */
405 /* TODO: This can be done faster is several blocks share the
406 * same VLC selector (subband_vlc_index) */
407 readQuantSpectralCoeffs (gb
, subband_vlc_index
[cnt
], codingMode
, mantissas
, subbWidth
);
409 /* Decode the scale factor for this subband. */
410 SF
= fixmul31(SFTable_fixed
[SF_idxs
[cnt
]], iMaxQuant_fix
[subband_vlc_index
[cnt
]]);
412 /* Inverse quantize the coefficients. */
414 /* Odd band - Reverse coefficients */
415 for (pIn
=mantissas
; last
>first
; last
--, pIn
++)
416 pOut
[last
] = fixmul16(*pIn
, SF
);
418 for (pIn
=mantissas
; first
<last
; first
++, pIn
++)
419 pOut
[first
] = fixmul16(*pIn
, SF
);
423 /* This subband was not coded, so zero the entire subband. */
424 memset(pOut
+first
, 0, subbWidth
*sizeof(int32_t));
428 /* Clear the subbands that were not coded. */
429 first
= subbandTab
[cnt
];
430 memset(pOut
+first
, 0, (1024 - first
) * sizeof(int32_t));
435 * Restore the quantized tonal components
437 * @param gb the GetBit context
438 * @param pComponent tone component
439 * @param numBands amount of coded bands
442 static int decodeTonalComponents (GetBitContext
*gb
, tonal_component
*pComponent
, int numBands
)
445 int components
, coding_mode_selector
, coding_mode
, coded_values_per_component
;
446 int sfIndx
, coded_values
, max_coded_values
, quant_step_index
, coded_components
;
447 int band_flags
[4], mantissa
[8];
450 int component_count
= 0;
452 components
= get_bits(gb
,5);
454 /* no tonal components */
458 coding_mode_selector
= get_bits(gb
,2);
459 if (coding_mode_selector
== 2)
462 coding_mode
= coding_mode_selector
& 1;
464 for (i
= 0; i
< components
; i
++) {
465 for (cnt
= 0; cnt
<= numBands
; cnt
++)
466 band_flags
[cnt
] = get_bits1(gb
);
468 coded_values_per_component
= get_bits(gb
,3);
470 quant_step_index
= get_bits(gb
,3);
471 if (quant_step_index
<= 1)
474 if (coding_mode_selector
== 3)
475 coding_mode
= get_bits1(gb
);
477 for (j
= 0; j
< (numBands
+ 1) * 4; j
++) {
478 if (band_flags
[j
>> 2] == 0)
481 coded_components
= get_bits(gb
,3);
483 for (k
=0; k
<coded_components
; k
++) {
484 sfIndx
= get_bits(gb
,6);
485 pComponent
[component_count
].pos
= j
* 64 + (get_bits(gb
,6));
486 max_coded_values
= 1024 - pComponent
[component_count
].pos
;
487 coded_values
= coded_values_per_component
+ 1;
488 coded_values
= FFMIN(max_coded_values
,coded_values
);
490 scalefactor
= fixmul31(SFTable_fixed
[sfIndx
], iMaxQuant_fix
[quant_step_index
]);
492 readQuantSpectralCoeffs(gb
, quant_step_index
, coding_mode
, mantissa
, coded_values
);
494 pComponent
[component_count
].numCoefs
= coded_values
;
497 pCoef
= pComponent
[component_count
].coef
;
498 for (cnt
= 0; cnt
< coded_values
; cnt
++)
499 pCoef
[cnt
] = fixmul16(mantissa
[cnt
], scalefactor
);
506 return component_count
;
510 * Decode gain parameters for the coded bands
512 * @param gb the GetBit context
513 * @param pGb the gainblock for the current band
514 * @param numBands amount of coded bands
517 static int decodeGainControl (GetBitContext
*gb
, gain_block
*pGb
, int numBands
)
522 gain_info
*pGain
= pGb
->gBlock
;
524 for (i
=0 ; i
<=numBands
; i
++)
526 numData
= get_bits(gb
,3);
527 pGain
[i
].num_gain_data
= numData
;
528 pLevel
= pGain
[i
].levcode
;
529 pLoc
= pGain
[i
].loccode
;
531 for (cf
= 0; cf
< numData
; cf
++){
532 pLevel
[cf
]= get_bits(gb
,4);
533 pLoc
[cf
]= get_bits(gb
,5);
534 if(cf
&& pLoc
[cf
] <= pLoc
[cf
-1])
539 /* Clear the unused blocks. */
541 pGain
[i
].num_gain_data
= 0;
547 * Apply gain parameters and perform the MDCT overlapping part
549 * @param pIn input float buffer
550 * @param pPrev previous float buffer to perform overlap against
551 * @param pOut output float buffer
552 * @param pGain1 current band gain info
553 * @param pGain2 next band gain info
556 static void gainCompensateAndOverlap (int32_t *pIn
, int32_t *pPrev
, int32_t *pOut
, gain_info
*pGain1
, gain_info
*pGain2
)
558 /* gain compensation function */
559 int32_t gain1
, gain2
, gain_inc
;
560 int cnt
, numdata
, nsample
, startLoc
, endLoc
;
562 if (pGain2
->num_gain_data
== 0)
565 gain1
= gain_tab1
[pGain2
->levcode
[0]];
567 if (pGain1
->num_gain_data
== 0) {
568 for (cnt
= 0; cnt
< 256; cnt
++)
569 pOut
[cnt
] = fixmul16(pIn
[cnt
], gain1
) + pPrev
[cnt
];
571 numdata
= pGain1
->num_gain_data
;
572 pGain1
->loccode
[numdata
] = 32;
573 pGain1
->levcode
[numdata
] = 4;
575 nsample
= 0; // current sample = 0
577 for (cnt
= 0; cnt
< numdata
; cnt
++) {
578 startLoc
= pGain1
->loccode
[cnt
] * 8;
579 endLoc
= startLoc
+ 8;
581 gain2
= gain_tab1
[pGain1
->levcode
[cnt
]];
582 gain_inc
= gain_tab2
[(pGain1
->levcode
[cnt
+1] - pGain1
->levcode
[cnt
])+15];
585 for (; nsample
< startLoc
; nsample
++)
586 pOut
[nsample
] = fixmul16((fixmul16(pIn
[nsample
], gain1
) + pPrev
[nsample
]), gain2
);
588 /* interpolation is done over eight samples */
589 for (; nsample
< endLoc
; nsample
++) {
590 pOut
[nsample
] = fixmul16((fixmul16(pIn
[nsample
], gain1
) + pPrev
[nsample
]),gain2
);
591 gain2
= fixmul16(gain2
, gain_inc
);
595 for (; nsample
< 256; nsample
++)
596 pOut
[nsample
] = fixmul16(pIn
[nsample
], gain1
) + pPrev
[nsample
];
599 /* Delay for the overlapping part. */
600 memcpy(pPrev
, &pIn
[256], 256*sizeof(int32_t));
604 * Combine the tonal band spectrum and regular band spectrum
605 * Return position of the last tonal coefficient
608 * @param pSpectrum output spectrum buffer
609 * @param numComponents amount of tonal components
610 * @param pComponent tonal components for this band
613 static int addTonalComponents (int32_t *pSpectrum
, int numComponents
, tonal_component
*pComponent
)
615 int cnt
, i
, lastPos
= -1;
619 for (cnt
= 0; cnt
< numComponents
; cnt
++){
620 lastPos
= FFMAX(pComponent
[cnt
].pos
+ pComponent
[cnt
].numCoefs
, lastPos
);
621 pIn
= pComponent
[cnt
].coef
;
622 pOut
= &(pSpectrum
[pComponent
[cnt
].pos
]);
624 for (i
=0 ; i
<pComponent
[cnt
].numCoefs
; i
++)
632 * Linear equidistant interpolation between two points x and y. 7 interpolation
633 * points can be calculated. Result is scaled by <<16.
634 * Result for s=0 is x*ONE_16
635 * Result for s=8 is y*ONE_16
637 * @param x first input point
638 * @param y second input point
639 * @param s index of interpolation point (0..7)
643 #define INTERPOLATE(x, y, s) ((x*ONE_16) + fixmul16(((s*ONE_16)>>3), (((x) - (y))*ONE_16)))
645 #define INTERPOLATE(x, y, s) ((((x)<<3) + s*((y)-(x)))<<13)
647 static void reverseMatrixing(int32_t *su1
, int32_t *su2
, int *pPrevCode
, int *pCurrCode
)
649 int i
, band
, nsample
, s1
, s2
;
651 int32_t mc1_l
, mc1_r
, mc2_l
, mc2_r
;
653 for (i
=0,band
= 0; band
< 4*256; band
+=256,i
++) {
659 /* Selector value changed, interpolation needed. */
660 mc1_l
= matrixCoeffs_fix
[s1
<<1];
661 mc1_r
= matrixCoeffs_fix
[(s1
<<1)+1];
662 mc2_l
= matrixCoeffs_fix
[s2
<<1];
663 mc2_r
= matrixCoeffs_fix
[(s2
<<1)+1];
665 /* Interpolation is done over the first eight samples. */
666 for(; nsample
< 8; nsample
++) {
667 c1
= su1
[band
+nsample
];
668 c2
= su2
[band
+nsample
];
669 c2
= fixmul16(c1
, INTERPOLATE(mc1_l
, mc2_l
, nsample
)) + fixmul16(c2
, INTERPOLATE(mc1_r
, mc2_r
, nsample
));
670 su1
[band
+nsample
] = c2
;
671 su2
[band
+nsample
] = (c1
<< 1) - c2
;
675 /* Apply the matrix without interpolation. */
677 case 0: /* M/S decoding */
678 for (; nsample
< 256; nsample
++) {
679 c1
= su1
[band
+nsample
];
680 c2
= su2
[band
+nsample
];
681 su1
[band
+nsample
] = c2
<< 1;
682 su2
[band
+nsample
] = (c1
- c2
) << 1;
687 for (; nsample
< 256; nsample
++) {
688 c1
= su1
[band
+nsample
];
689 c2
= su2
[band
+nsample
];
690 su1
[band
+nsample
] = (c1
+ c2
) << 1;
691 su2
[band
+nsample
] = -1*(c2
<< 1);
696 for (; nsample
< 256; nsample
++) {
697 c1
= su1
[band
+nsample
];
698 c2
= su2
[band
+nsample
];
699 su1
[band
+nsample
] = c1
+ c2
;
700 su2
[band
+nsample
] = c1
- c2
;
710 static void getChannelWeights (int indx
, int flag
, int32_t ch
[2]){
711 /* Read channel weights from table */
713 /* Swap channel weights */
714 ch
[1] = channelWeights0
[indx
&7];
715 ch
[0] = channelWeights1
[indx
&7];
717 ch
[0] = channelWeights0
[indx
&7];
718 ch
[1] = channelWeights1
[indx
&7];
722 static void channelWeighting (int32_t *su1
, int32_t *su2
, int *p3
)
725 /* w[x][y] y=0 is left y=1 is right */
728 if (p3
[1] != 7 || p3
[3] != 7){
729 getChannelWeights(p3
[1], p3
[0], w
[0]);
730 getChannelWeights(p3
[3], p3
[2], w
[1]);
732 for(band
= 1; band
< 4; band
++) {
733 /* scale the channels by the weights */
734 for(nsample
= 0; nsample
< 8; nsample
++) {
735 su1
[band
*256+nsample
] = fixmul16(su1
[band
*256+nsample
], INTERPOLATE(w
[0][0], w
[0][1], nsample
));
736 su2
[band
*256+nsample
] = fixmul16(su2
[band
*256+nsample
], INTERPOLATE(w
[1][0], w
[1][1], nsample
));
739 for(; nsample
< 256; nsample
++) {
740 su1
[band
*256+nsample
] = fixmul16(su1
[band
*256+nsample
], w
[1][0]);
741 su2
[band
*256+nsample
] = fixmul16(su2
[band
*256+nsample
], w
[1][1]);
749 * Decode a Sound Unit
751 * @param gb the GetBit context
752 * @param pSnd the channel unit to be used
753 * @param pOut the decoded samples before IQMF in float representation
754 * @param channelNum channel number
755 * @param codingMode the coding mode (JOINT_STEREO or regular stereo/mono)
759 static int decodeChannelSoundUnit (GetBitContext
*gb
, channel_unit
*pSnd
, int32_t *pOut
, int channelNum
, int codingMode
)
761 int band
, result
=0, numSubbands
, lastTonal
, numBands
;
762 if (codingMode
== JOINT_STEREO
&& channelNum
== 1) {
763 if (get_bits(gb
,2) != 3) {
764 DEBUGF("JS mono Sound Unit id != 3.\n");
768 if (get_bits(gb
,6) != 0x28) {
769 DEBUGF("Sound Unit id != 0x28.\n");
774 /* number of coded QMF bands */
775 pSnd
->bandsCoded
= get_bits(gb
,2);
777 result
= decodeGainControl (gb
, &(pSnd
->gainBlock
[pSnd
->gcBlkSwitch
]), pSnd
->bandsCoded
);
778 if (result
) return result
;
780 pSnd
->numComponents
= decodeTonalComponents (gb
, pSnd
->components
, pSnd
->bandsCoded
);
781 if (pSnd
->numComponents
== -1) return -1;
783 numSubbands
= decodeSpectrum (gb
, pSnd
->spectrum
);
785 /* Merge the decoded spectrum and tonal components. */
786 lastTonal
= addTonalComponents (pSnd
->spectrum
, pSnd
->numComponents
, pSnd
->components
);
789 /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
790 numBands
= (subbandTab
[numSubbands
] - 1) >> 8;
792 numBands
= FFMAX((lastTonal
+ 256) >> 8, numBands
);
794 /* Remark: Hardcoded hack to add 2 bits (empty) fract part to internal sample
795 * representation. Needed for higher accuracy in internal calculations as
796 * well as for DSP configuration. See also: ../atrac3_rm.c, DSP_SET_SAMPLE_DEPTH
797 * Todo: Check spectral requantisation for using and outputting samples with
800 for (i
=0; i
<1024; ++i
) {
801 pSnd
->spectrum
[i
] <<= 2;
804 /* Reconstruct time domain samples. */
805 for (band
=0; band
<4; band
++) {
806 /* Perform the IMDCT step without overlapping. */
807 if (band
<= numBands
) {
808 IMLT(&(pSnd
->spectrum
[band
*256]), pSnd
->IMDCT_buf
);
810 memset(pSnd
->IMDCT_buf
, 0, 512 * sizeof(int32_t));
812 /* gain compensation and overlapping */
813 gainCompensateAndOverlap (pSnd
->IMDCT_buf
, &(pSnd
->prevFrame
[band
*256]), &(pOut
[band
*256]),
814 &((pSnd
->gainBlock
[1 - (pSnd
->gcBlkSwitch
)]).gBlock
[band
]),
815 &((pSnd
->gainBlock
[pSnd
->gcBlkSwitch
]).gBlock
[band
]));
818 /* Swap the gain control buffers for the next frame. */
819 pSnd
->gcBlkSwitch
^= 1;
827 * @param q Atrac3 private context
828 * @param databuf the input data
831 static int decodeFrame(ATRAC3Context
*q
, const uint8_t* databuf
, int off
)
834 int32_t *p1
, *p2
, *p3
, *p4
;
837 if (q
->codingMode
== JOINT_STEREO
) {
839 /* channel coupling mode */
840 /* decode Sound Unit 1 */
841 init_get_bits(&q
->gb
,databuf
,q
->bits_per_frame
);
843 result
= decodeChannelSoundUnit(&q
->gb
, q
->pUnits
, q
->outSamples
, 0, JOINT_STEREO
);
847 /* Framedata of the su2 in the joint-stereo mode is encoded in
848 * reverse byte order so we need to swap it first. */
849 if (databuf
== q
->decoded_bytes_buffer
) {
850 uint8_t *ptr2
= q
->decoded_bytes_buffer
+q
->bytes_per_frame
-1;
851 ptr1
= q
->decoded_bytes_buffer
;
852 for (i
= 0; i
< (q
->bytes_per_frame
/2); i
++, ptr1
++, ptr2
--) {
853 FFSWAP(uint8_t,*ptr1
,*ptr2
);
856 const uint8_t *ptr2
= databuf
+q
->bytes_per_frame
-1;
857 for (i
= 0; i
< q
->bytes_per_frame
; i
++)
858 q
->decoded_bytes_buffer
[i
] = *ptr2
--;
861 /* Skip the sync codes (0xF8). */
862 ptr1
= q
->decoded_bytes_buffer
;
863 for (i
= 4; *ptr1
== 0xF8; i
++, ptr1
++) {
864 if (i
>= q
->bytes_per_frame
)
869 /* set the bitstream reader at the start of the second Sound Unit*/
870 init_get_bits(&q
->gb
,ptr1
,q
->bits_per_frame
);
872 /* Fill the Weighting coeffs delay buffer */
873 memmove(q
->weighting_delay
,&(q
->weighting_delay
[2]),4*sizeof(int));
874 q
->weighting_delay
[4] = get_bits1(&q
->gb
);
875 q
->weighting_delay
[5] = get_bits(&q
->gb
,3);
877 for (i
= 0; i
< 4; i
++) {
878 q
->matrix_coeff_index_prev
[i
] = q
->matrix_coeff_index_now
[i
];
879 q
->matrix_coeff_index_now
[i
] = q
->matrix_coeff_index_next
[i
];
880 q
->matrix_coeff_index_next
[i
] = get_bits(&q
->gb
,2);
883 /* Decode Sound Unit 2. */
884 result
= decodeChannelSoundUnit(&q
->gb
, &q
->pUnits
[1], &q
->outSamples
[1024], 1, JOINT_STEREO
);
888 /* Reconstruct the channel coefficients. */
889 reverseMatrixing(q
->outSamples
, &q
->outSamples
[1024], q
->matrix_coeff_index_prev
, q
->matrix_coeff_index_now
);
891 channelWeighting(q
->outSamples
, &q
->outSamples
[1024], q
->weighting_delay
);
894 /* normal stereo mode or mono */
895 /* Decode the channel sound units. */
896 for (i
=0 ; i
<q
->channels
; i
++) {
898 /* Set the bitstream reader at the start of a channel sound unit. */
899 init_get_bits(&q
->gb
, databuf
+((i
*q
->bytes_per_frame
)/q
->channels
)+off
, (q
->bits_per_frame
)/q
->channels
);
901 result
= decodeChannelSoundUnit(&q
->gb
, &q
->pUnits
[i
], &q
->outSamples
[i
*1024], i
, q
->codingMode
);
907 /* Apply the iQMF synthesis filter. */
909 for (i
=0 ; i
<q
->channels
; i
++) {
913 iqmf (p1
, p2
, 256, p1
, q
->pUnits
[i
].delayBuf1
, q
->tempBuf
);
914 iqmf (p4
, p3
, 256, p3
, q
->pUnits
[i
].delayBuf2
, q
->tempBuf
);
915 iqmf (p1
, p3
, 512, p1
, q
->pUnits
[i
].delayBuf3
, q
->tempBuf
);
924 * Atrac frame decoding
926 * @param rmctx pointer to the AVCodecContext
929 int atrac3_decode_frame(RMContext
*rmctx
, ATRAC3Context
*q
,
930 int *data_size
, const uint8_t *buf
, int buf_size
) {
931 int result
= 0, off
= 0;
932 const uint8_t* databuf
;
934 if (buf_size
< rmctx
->block_align
)
937 /* Check if we need to descramble and what buffer to pass on. */
938 if (q
->scrambled_stream
) {
939 off
= decode_bytes(buf
, q
->decoded_bytes_buffer
, rmctx
->block_align
);
940 databuf
= q
->decoded_bytes_buffer
;
945 result
= decodeFrame(q
, databuf
, off
);
948 DEBUGF("Frame decoding error!\n");
952 if (q
->channels
== 1)
953 *data_size
= 1024 * sizeof(int32_t);
955 *data_size
= 2048 * sizeof(int32_t);
957 return rmctx
->block_align
;
962 * Atrac3 initialization
964 * @param rmctx pointer to the RMContext
967 int atrac3_decode_init(ATRAC3Context
*q
, RMContext
*rmctx
)
970 uint8_t *edata_ptr
= rmctx
->codec_extradata
;
971 static VLC_TYPE atrac3_vlc_table
[4096][2];
972 static int vlcs_initialized
= 0;
974 /* Take data from the AVCodecContext (RM container). */
975 q
->sample_rate
= rmctx
->sample_rate
;
976 q
->channels
= rmctx
->nb_channels
;
977 q
->bit_rate
= rmctx
->bit_rate
;
978 q
->bits_per_frame
= rmctx
->block_align
* 8;
979 q
->bytes_per_frame
= rmctx
->block_align
;
981 /* Take care of the codec-specific extradata. */
982 if (rmctx
->extradata_size
== 14) {
983 /* Parse the extradata, WAV format */
984 DEBUGF("[0-1] %d\n",rm_get_uint16le(&edata_ptr
[0])); //Unknown value always 1
985 q
->samples_per_channel
= rm_get_uint32le(&edata_ptr
[2]);
986 q
->codingMode
= rm_get_uint16le(&edata_ptr
[6]);
987 DEBUGF("[8-9] %d\n",rm_get_uint16le(&edata_ptr
[8])); //Dupe of coding mode
988 q
->frame_factor
= rm_get_uint16le(&edata_ptr
[10]); //Unknown always 1
989 DEBUGF("[12-13] %d\n",rm_get_uint16le(&edata_ptr
[12])); //Unknown always 0
992 q
->samples_per_frame
= 1024 * q
->channels
;
993 q
->atrac3version
= 4;
996 q
->codingMode
= JOINT_STEREO
;
998 q
->codingMode
= STEREO
;
999 q
->scrambled_stream
= 0;
1001 if ((q
->bytes_per_frame
== 96*q
->channels
*q
->frame_factor
) || (q
->bytes_per_frame
== 152*q
->channels
*q
->frame_factor
) || (q
->bytes_per_frame
== 192*q
->channels
*q
->frame_factor
)) {
1003 DEBUGF("Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q
->bytes_per_frame
, q
->channels
, q
->frame_factor
);
1007 } else if (rmctx
->extradata_size
== 10) {
1008 /* Parse the extradata, RM format. */
1009 q
->atrac3version
= rm_get_uint32be(&edata_ptr
[0]);
1010 q
->samples_per_frame
= rm_get_uint16be(&edata_ptr
[4]);
1011 q
->delay
= rm_get_uint16be(&edata_ptr
[6]);
1012 q
->codingMode
= rm_get_uint16be(&edata_ptr
[8]);
1014 q
->samples_per_channel
= q
->samples_per_frame
/ q
->channels
;
1015 q
->scrambled_stream
= 1;
1018 DEBUGF("Unknown extradata size %d.\n",rmctx
->extradata_size
);
1020 /* Check the extradata. */
1022 if (q
->atrac3version
!= 4) {
1023 DEBUGF("Version %d != 4.\n",q
->atrac3version
);
1027 if (q
->samples_per_frame
!= 1024 && q
->samples_per_frame
!= 2048) {
1028 DEBUGF("Unknown amount of samples per frame %d.\n",q
->samples_per_frame
);
1032 if (q
->delay
!= 0x88E) {
1033 DEBUGF("Unknown amount of delay %x != 0x88E.\n",q
->delay
);
1037 if (q
->codingMode
== STEREO
) {
1038 DEBUGF("Normal stereo detected.\n");
1039 } else if (q
->codingMode
== JOINT_STEREO
) {
1040 DEBUGF("Joint stereo detected.\n");
1042 DEBUGF("Unknown channel coding mode %x!\n",q
->codingMode
);
1046 if (rmctx
->nb_channels
<= 0 || rmctx
->nb_channels
> 2 /*|| ((rmctx->channels * 1024) != q->samples_per_frame)*/) {
1047 DEBUGF("Channel configuration error!\n");
1052 if(rmctx
->block_align
>= UINT16_MAX
/2)
1056 /* Initialize the VLC tables. */
1057 if (!vlcs_initialized
) {
1058 for (i
=0 ; i
<7 ; i
++) {
1059 spectral_coeff_tab
[i
].table
= &atrac3_vlc_table
[atrac3_vlc_offs
[i
]];
1060 spectral_coeff_tab
[i
].table_allocated
= atrac3_vlc_offs
[i
+ 1] - atrac3_vlc_offs
[i
];
1061 init_vlc (&spectral_coeff_tab
[i
], 9, huff_tab_sizes
[i
],
1063 huff_codes
[i
], 1, 1, INIT_VLC_USE_NEW_STATIC
);
1066 vlcs_initialized
= 1;
1070 init_atrac3_transforms();
1072 /* init the joint-stereo decoding data */
1073 q
->weighting_delay
[0] = 0;
1074 q
->weighting_delay
[1] = 7;
1075 q
->weighting_delay
[2] = 0;
1076 q
->weighting_delay
[3] = 7;
1077 q
->weighting_delay
[4] = 0;
1078 q
->weighting_delay
[5] = 7;
1080 for (i
=0; i
<4; i
++) {
1081 q
->matrix_coeff_index_prev
[i
] = 3;
1082 q
->matrix_coeff_index_now
[i
] = 3;
1083 q
->matrix_coeff_index_next
[i
] = 3;
1086 q
->pUnits
= channel_units
;