pcm: define AVCodec instances only for enabled codecs
[libav.git] / libavcodec / atrac3.c
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1/*
2 * Atrac 3 compatible decoder
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3 * Copyright (c) 2006-2008 Maxim Poliakovski
4 * Copyright (c) 2006-2008 Benjamin Larsson
10e26bc7 5 *
2912e87a 6 * This file is part of Libav.
10e26bc7 7 *
2912e87a 8 * Libav is free software; you can redistribute it and/or
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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.
12 *
2912e87a 13 * Libav is distributed in the hope that it will be useful,
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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.
17 *
18 * You should have received a copy of the GNU Lesser General Public
2912e87a 19 * License along with Libav; if not, write to the Free Software
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23/**
ba87f080 24 * @file
10e26bc7 25 * Atrac 3 compatible decoder.
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26 * This decoder handles Sony's ATRAC3 data.
27 *
28 * Container formats used to store atrac 3 data:
29 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
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30 *
31 * To use this decoder, a calling application must supply the extradata
d311f8f3 32 * bytes provided in the containers above.
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33 */
34
35#include <math.h>
36#include <stddef.h>
37#include <stdio.h>
38
d5a7229b 39#include "libavutil/float_dsp.h"
10e26bc7 40#include "avcodec.h"
9106a698 41#include "get_bits.h"
10e26bc7 42#include "bytestream.h"
1429224b 43#include "fft.h"
5e76b8bb 44#include "fmtconvert.h"
10e26bc7 45
0e1baede 46#include "atrac.h"
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47#include "atrac3data.h"
48
49#define JOINT_STEREO 0x12
50#define STEREO 0x2
51
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52#define SAMPLES_PER_FRAME 1024
53#define MDCT_SIZE 512
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54
55/* These structures are needed to store the parsed gain control data. */
56typedef struct {
57 int num_gain_data;
58 int levcode[8];
59 int loccode[8];
60} gain_info;
61
62typedef struct {
63 gain_info gBlock[4];
64} gain_block;
65
66typedef struct {
67 int pos;
68 int numCoefs;
69 float coef[8];
70} tonal_component;
71
72typedef struct {
73 int bandsCoded;
74 int numComponents;
75 tonal_component components[64];
c9161385 76 float prevFrame[SAMPLES_PER_FRAME];
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77 int gcBlkSwitch;
78 gain_block gainBlock[2];
79
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80 DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
81 DECLARE_ALIGNED(32, float, IMDCT_buf)[SAMPLES_PER_FRAME];
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82
83 float delayBuf1[46]; ///<qmf delay buffers
84 float delayBuf2[46];
85 float delayBuf3[46];
86} channel_unit;
87
88typedef struct {
0eea2129 89 AVFrame frame;
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90 GetBitContext gb;
91 //@{
92 /** stream data */
93 int channels;
94 int codingMode;
95 int bit_rate;
96 int sample_rate;
97 int samples_per_channel;
98 int samples_per_frame;
99
100 int bits_per_frame;
101 int bytes_per_frame;
102 int pBs;
103 channel_unit* pUnits;
104 //@}
105 //@{
106 /** joint-stereo related variables */
107 int matrix_coeff_index_prev[4];
108 int matrix_coeff_index_now[4];
109 int matrix_coeff_index_next[4];
110 int weighting_delay[6];
111 //@}
112 //@{
113 /** data buffers */
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114 uint8_t* decoded_bytes_buffer;
115 float tempBuf[1070];
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116 //@}
117 //@{
118 /** extradata */
119 int atrac3version;
120 int delay;
121 int scrambled_stream;
122 int frame_factor;
123 //@}
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124
125 FFTContext mdct_ctx;
5e76b8bb 126 FmtConvertContext fmt_conv;
d5a7229b 127 AVFloatDSPContext fdsp;
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128} ATRAC3Context;
129
c9161385 130static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
10e26bc7 131static VLC spectral_coeff_tab[7];
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132static float gain_tab1[16];
133static float gain_tab2[31];
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134
135
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136/**
137 * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
138 * caused by the reverse spectra of the QMF.
139 *
140 * @param pInput float input
141 * @param pOutput float output
142 * @param odd_band 1 if the band is an odd band
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143 */
144
a28cccf6 145static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
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146{
147 int i;
148
149 if (odd_band) {
150 /**
151 * Reverse the odd bands before IMDCT, this is an effect of the QMF transform
152 * or it gives better compression to do it this way.
26f548bb 153 * FIXME: It should be possible to handle this in imdct_calc
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154 * for that to happen a modification of the prerotation step of
155 * all SIMD code and C code is needed.
156 * Or fix the functions before so they generate a pre reversed spectrum.
157 */
158
159 for (i=0; i<128; i++)
160 FFSWAP(float, pInput[i], pInput[255-i]);
161 }
162
26f548bb 163 q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);
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164
165 /* Perform windowing on the output. */
d5a7229b 166 q->fdsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE);
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167
168}
169
170
171/**
172 * Atrac 3 indata descrambling, only used for data coming from the rm container
173 *
9a58234f 174 * @param inbuffer pointer to 8 bit array of indata
10e26bc7 175 * @param out pointer to 8 bit array of outdata
9a58234f 176 * @param bytes amount of bytes
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177 */
178
8687f767 179static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
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180 int i, off;
181 uint32_t c;
8687f767 182 const uint32_t* buf;
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183 uint32_t* obuf = (uint32_t*) out;
184
e05c8d06 185 off = (intptr_t)inbuffer & 3;
8687f767 186 buf = (const uint32_t*) (inbuffer - off);
8fc0162a 187 c = av_be2ne32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
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188 bytes += 3 + off;
189 for (i = 0; i < bytes/4; i++)
190 obuf[i] = c ^ buf[i];
191
192 if (off)
d9dee728 193 av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off);
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194
195 return off;
196}
197
198
9af4eaa8 199static av_cold int init_atrac3_transforms(ATRAC3Context *q) {
10e26bc7 200 float enc_window[256];
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201 int i;
202
203 /* Generate the mdct window, for details see
204 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
205 for (i=0 ; i<256; i++)
206 enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
207
208 if (!mdct_window[0])
209 for (i=0 ; i<256; i++) {
210 mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
211 mdct_window[511-i] = mdct_window[i];
212 }
213
10e26bc7 214 /* Initialize the MDCT transform. */
9af4eaa8 215 return ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768);
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216}
217
218/**
219 * Atrac3 uninit, free all allocated memory
220 */
221
5ef251e5 222static av_cold int atrac3_decode_close(AVCodecContext *avctx)
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223{
224 ATRAC3Context *q = avctx->priv_data;
225
226 av_free(q->pUnits);
227 av_free(q->decoded_bytes_buffer);
5e76b8bb 228
a28cccf6 229 ff_mdct_end(&q->mdct_ctx);
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230
231 return 0;
232}
233
234/**
235/ * Mantissa decoding
236 *
237 * @param gb the GetBit context
238 * @param selector what table is the output values coded with
239 * @param codingFlag constant length coding or variable length coding
240 * @param mantissas mantissa output table
241 * @param numCodes amount of values to get
242 */
243
244static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
245{
246 int numBits, cnt, code, huffSymb;
247
248 if (selector == 1)
249 numCodes /= 2;
250
251 if (codingFlag != 0) {
252 /* constant length coding (CLC) */
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253 numBits = CLCLengthTab[selector];
254
255 if (selector > 1) {
256 for (cnt = 0; cnt < numCodes; cnt++) {
257 if (numBits)
258 code = get_sbits(gb, numBits);
259 else
260 code = 0;
261 mantissas[cnt] = code;
262 }
263 } else {
264 for (cnt = 0; cnt < numCodes; cnt++) {
265 if (numBits)
266 code = get_bits(gb, numBits); //numBits is always 4 in this case
267 else
268 code = 0;
269 mantissas[cnt*2] = seTab_0[code >> 2];
270 mantissas[cnt*2+1] = seTab_0[code & 3];
271 }
272 }
273 } else {
274 /* variable length coding (VLC) */
275 if (selector != 1) {
276 for (cnt = 0; cnt < numCodes; cnt++) {
277 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
278 huffSymb += 1;
279 code = huffSymb >> 1;
280 if (huffSymb & 1)
281 code = -code;
282 mantissas[cnt] = code;
283 }
284 } else {
285 for (cnt = 0; cnt < numCodes; cnt++) {
286 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
287 mantissas[cnt*2] = decTable1[huffSymb*2];
288 mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
289 }
290 }
291 }
292}
293
294/**
295 * Restore the quantized band spectrum coefficients
296 *
297 * @param gb the GetBit context
298 * @param pOut decoded band spectrum
299 * @return outSubbands subband counter, fix for broken specification/files
300 */
301
302static int decodeSpectrum (GetBitContext *gb, float *pOut)
303{
304 int numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
305 int subband_vlc_index[32], SF_idxs[32];
306 int mantissas[128];
307 float SF;
308
309 numSubbands = get_bits(gb, 5); // number of coded subbands
5fc32c27 310 codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
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311
312 /* Get the VLC selector table for the subbands, 0 means not coded. */
313 for (cnt = 0; cnt <= numSubbands; cnt++)
314 subband_vlc_index[cnt] = get_bits(gb, 3);
315
316 /* Read the scale factor indexes from the stream. */
317 for (cnt = 0; cnt <= numSubbands; cnt++) {
318 if (subband_vlc_index[cnt] != 0)
319 SF_idxs[cnt] = get_bits(gb, 6);
320 }
321
322 for (cnt = 0; cnt <= numSubbands; cnt++) {
323 first = subbandTab[cnt];
324 last = subbandTab[cnt+1];
325
326 subbWidth = last - first;
327
328 if (subband_vlc_index[cnt] != 0) {
329 /* Decode spectral coefficients for this subband. */
330 /* TODO: This can be done faster is several blocks share the
331 * same VLC selector (subband_vlc_index) */
332 readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
333
334 /* Decode the scale factor for this subband. */
82e1f217 335 SF = ff_atrac_sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
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336
337 /* Inverse quantize the coefficients. */
338 for (pIn=mantissas ; first<last; first++, pIn++)
339 pOut[first] = *pIn * SF;
340 } else {
341 /* This subband was not coded, so zero the entire subband. */
342 memset(pOut+first, 0, subbWidth*sizeof(float));
343 }
344 }
345
346 /* Clear the subbands that were not coded. */
347 first = subbandTab[cnt];
c9161385 348 memset(pOut+first, 0, (SAMPLES_PER_FRAME - first) * sizeof(float));
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349 return numSubbands;
350}
351
352/**
353 * Restore the quantized tonal components
354 *
355 * @param gb the GetBit context
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356 * @param pComponent tone component
357 * @param numBands amount of coded bands
358 */
359
b8c4a515 360static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
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361{
362 int i,j,k,cnt;
b8c4a515 363 int components, coding_mode_selector, coding_mode, coded_values_per_component;
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364 int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
365 int band_flags[4], mantissa[8];
366 float *pCoef;
367 float scalefactor;
b8c4a515 368 int component_count = 0;
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369
370 components = get_bits(gb,5);
371
372 /* no tonal components */
373 if (components == 0)
374 return 0;
375
376 coding_mode_selector = get_bits(gb,2);
377 if (coding_mode_selector == 2)
8f98577d 378 return AVERROR_INVALIDDATA;
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379
380 coding_mode = coding_mode_selector & 1;
381
382 for (i = 0; i < components; i++) {
383 for (cnt = 0; cnt <= numBands; cnt++)
384 band_flags[cnt] = get_bits1(gb);
385
386 coded_values_per_component = get_bits(gb,3);
387
388 quant_step_index = get_bits(gb,3);
389 if (quant_step_index <= 1)
8f98577d 390 return AVERROR_INVALIDDATA;
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391
392 if (coding_mode_selector == 3)
393 coding_mode = get_bits1(gb);
394
395 for (j = 0; j < (numBands + 1) * 4; j++) {
396 if (band_flags[j >> 2] == 0)
397 continue;
398
399 coded_components = get_bits(gb,3);
400
401 for (k=0; k<coded_components; k++) {
402 sfIndx = get_bits(gb,6);
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403 if (component_count >= 64)
404 return AVERROR_INVALIDDATA;
10e26bc7 405 pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
c9161385 406 max_coded_values = SAMPLES_PER_FRAME - pComponent[component_count].pos;
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407 coded_values = coded_values_per_component + 1;
408 coded_values = FFMIN(max_coded_values,coded_values);
409
82e1f217 410 scalefactor = ff_atrac_sf_table[sfIndx] * iMaxQuant[quant_step_index];
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411
412 readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
413
414 pComponent[component_count].numCoefs = coded_values;
415
416 /* inverse quant */
9d278d88 417 pCoef = pComponent[component_count].coef;
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418 for (cnt = 0; cnt < coded_values; cnt++)
419 pCoef[cnt] = mantissa[cnt] * scalefactor;
420
421 component_count++;
422 }
423 }
424 }
425
b8c4a515 426 return component_count;
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427}
428
429/**
430 * Decode gain parameters for the coded bands
431 *
432 * @param gb the GetBit context
433 * @param pGb the gainblock for the current band
434 * @param numBands amount of coded bands
435 */
436
437static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
438{
439 int i, cf, numData;
440 int *pLevel, *pLoc;
441
442 gain_info *pGain = pGb->gBlock;
443
444 for (i=0 ; i<=numBands; i++)
445 {
446 numData = get_bits(gb,3);
447 pGain[i].num_gain_data = numData;
448 pLevel = pGain[i].levcode;
449 pLoc = pGain[i].loccode;
450
451 for (cf = 0; cf < numData; cf++){
452 pLevel[cf]= get_bits(gb,4);
453 pLoc [cf]= get_bits(gb,5);
454 if(cf && pLoc[cf] <= pLoc[cf-1])
8f98577d 455 return AVERROR_INVALIDDATA;
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456 }
457 }
458
459 /* Clear the unused blocks. */
460 for (; i<4 ; i++)
461 pGain[i].num_gain_data = 0;
462
463 return 0;
464}
465
466/**
467 * Apply gain parameters and perform the MDCT overlapping part
468 *
469 * @param pIn input float buffer
470 * @param pPrev previous float buffer to perform overlap against
471 * @param pOut output float buffer
472 * @param pGain1 current band gain info
473 * @param pGain2 next band gain info
474 */
475
476static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
477{
478 /* gain compensation function */
479 float gain1, gain2, gain_inc;
480 int cnt, numdata, nsample, startLoc, endLoc;
481
482
483 if (pGain2->num_gain_data == 0)
484 gain1 = 1.0;
485 else
486 gain1 = gain_tab1[pGain2->levcode[0]];
487
488 if (pGain1->num_gain_data == 0) {
489 for (cnt = 0; cnt < 256; cnt++)
490 pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
491 } else {
492 numdata = pGain1->num_gain_data;
493 pGain1->loccode[numdata] = 32;
494 pGain1->levcode[numdata] = 4;
495
496 nsample = 0; // current sample = 0
497
498 for (cnt = 0; cnt < numdata; cnt++) {
499 startLoc = pGain1->loccode[cnt] * 8;
500 endLoc = startLoc + 8;
501
502 gain2 = gain_tab1[pGain1->levcode[cnt]];
503 gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
504
505 /* interpolate */
506 for (; nsample < startLoc; nsample++)
507 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
508
509 /* interpolation is done over eight samples */
510 for (; nsample < endLoc; nsample++) {
511 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
512 gain2 *= gain_inc;
513 }
514 }
515
516 for (; nsample < 256; nsample++)
517 pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
518 }
519
520 /* Delay for the overlapping part. */
521 memcpy(pPrev, &pIn[256], 256*sizeof(float));
522}
523
524/**
525 * Combine the tonal band spectrum and regular band spectrum
9d278d88 526 * Return position of the last tonal coefficient
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527 *
528 * @param pSpectrum output spectrum buffer
529 * @param numComponents amount of tonal components
530 * @param pComponent tonal components for this band
531 */
532
9d278d88 533static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
10e26bc7 534{
9d278d88 535 int cnt, i, lastPos = -1;
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536 float *pIn, *pOut;
537
538 for (cnt = 0; cnt < numComponents; cnt++){
9d278d88 539 lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);
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540 pIn = pComponent[cnt].coef;
541 pOut = &(pSpectrum[pComponent[cnt].pos]);
542
543 for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
544 pOut[i] += pIn[i];
545 }
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546
547 return lastPos;
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548}
549
550
551#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
552
553static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
554{
555 int i, band, nsample, s1, s2;
556 float c1, c2;
557 float mc1_l, mc1_r, mc2_l, mc2_r;
558
559 for (i=0,band = 0; band < 4*256; band+=256,i++) {
560 s1 = pPrevCode[i];
561 s2 = pCurrCode[i];
562 nsample = 0;
563
564 if (s1 != s2) {
565 /* Selector value changed, interpolation needed. */
566 mc1_l = matrixCoeffs[s1*2];
567 mc1_r = matrixCoeffs[s1*2+1];
568 mc2_l = matrixCoeffs[s2*2];
569 mc2_r = matrixCoeffs[s2*2+1];
570
571 /* Interpolation is done over the first eight samples. */
572 for(; nsample < 8; nsample++) {
573 c1 = su1[band+nsample];
574 c2 = su2[band+nsample];
575 c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
576 su1[band+nsample] = c2;
577 su2[band+nsample] = c1 * 2.0 - c2;
578 }
579 }
580
581 /* Apply the matrix without interpolation. */
582 switch (s2) {
583 case 0: /* M/S decoding */
584 for (; nsample < 256; nsample++) {
585 c1 = su1[band+nsample];
586 c2 = su2[band+nsample];
587 su1[band+nsample] = c2 * 2.0;
588 su2[band+nsample] = (c1 - c2) * 2.0;
589 }
590 break;
591
592 case 1:
593 for (; nsample < 256; nsample++) {
594 c1 = su1[band+nsample];
595 c2 = su2[band+nsample];
596 su1[band+nsample] = (c1 + c2) * 2.0;
597 su2[band+nsample] = c2 * -2.0;
598 }
599 break;
600 case 2:
601 case 3:
602 for (; nsample < 256; nsample++) {
603 c1 = su1[band+nsample];
604 c2 = su2[band+nsample];
605 su1[band+nsample] = c1 + c2;
606 su2[band+nsample] = c1 - c2;
607 }
608 break;
609 default:
610 assert(0);
611 }
612 }
613}
614
615static void getChannelWeights (int indx, int flag, float ch[2]){
616
617 if (indx == 7) {
618 ch[0] = 1.0;
619 ch[1] = 1.0;
620 } else {
621 ch[0] = (float)(indx & 7) / 7.0;
622 ch[1] = sqrt(2 - ch[0]*ch[0]);
623 if(flag)
624 FFSWAP(float, ch[0], ch[1]);
625 }
626}
627
628static void channelWeighting (float *su1, float *su2, int *p3)
629{
630 int band, nsample;
631 /* w[x][y] y=0 is left y=1 is right */
632 float w[2][2];
633
634 if (p3[1] != 7 || p3[3] != 7){
635 getChannelWeights(p3[1], p3[0], w[0]);
636 getChannelWeights(p3[3], p3[2], w[1]);
637
638 for(band = 1; band < 4; band++) {
639 /* scale the channels by the weights */
640 for(nsample = 0; nsample < 8; nsample++) {
641 su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
642 su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
643 }
644
645 for(; nsample < 256; nsample++) {
646 su1[band*256+nsample] *= w[1][0];
647 su2[band*256+nsample] *= w[1][1];
648 }
649 }
650 }
651}
652
653
654/**
655 * Decode a Sound Unit
656 *
657 * @param gb the GetBit context
658 * @param pSnd the channel unit to be used
659 * @param pOut the decoded samples before IQMF in float representation
660 * @param channelNum channel number
661 * @param codingMode the coding mode (JOINT_STEREO or regular stereo/mono)
662 */
663
664
665static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
666{
9d278d88 667 int band, result=0, numSubbands, lastTonal, numBands;
10e26bc7
BL
668
669 if (codingMode == JOINT_STEREO && channelNum == 1) {
670 if (get_bits(gb,2) != 3) {
671 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
8f98577d 672 return AVERROR_INVALIDDATA;
10e26bc7
BL
673 }
674 } else {
675 if (get_bits(gb,6) != 0x28) {
676 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
8f98577d 677 return AVERROR_INVALIDDATA;
10e26bc7
BL
678 }
679 }
680
681 /* number of coded QMF bands */
682 pSnd->bandsCoded = get_bits(gb,2);
683
684 result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
685 if (result) return result;
686
b8c4a515
BL
687 pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
688 if (pSnd->numComponents == -1) return -1;
10e26bc7
BL
689
690 numSubbands = decodeSpectrum (gb, pSnd->spectrum);
691
692 /* Merge the decoded spectrum and tonal components. */
9d278d88 693 lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
10e26bc7
BL
694
695
9d278d88 696 /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
10e26bc7 697 numBands = (subbandTab[numSubbands] - 1) >> 8;
9d278d88
MP
698 if (lastTonal >= 0)
699 numBands = FFMAX((lastTonal + 256) >> 8, numBands);
10e26bc7
BL
700
701
702 /* Reconstruct time domain samples. */
703 for (band=0; band<4; band++) {
704 /* Perform the IMDCT step without overlapping. */
705 if (band <= numBands) {
a28cccf6 706 IMLT(q, &(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
10e26bc7
BL
707 } else
708 memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
709
710 /* gain compensation and overlapping */
ee41963f
DB
711 gainCompensateAndOverlap(pSnd->IMDCT_buf, &pSnd->prevFrame[band * 256],
712 &pOut[band * 256],
713 &pSnd->gainBlock[1 - pSnd->gcBlkSwitch].gBlock[band],
714 &pSnd->gainBlock[ pSnd->gcBlkSwitch].gBlock[band]);
10e26bc7
BL
715 }
716
717 /* Swap the gain control buffers for the next frame. */
718 pSnd->gcBlkSwitch ^= 1;
719
720 return 0;
721}
722
723/**
724 * Frame handling
725 *
726 * @param q Atrac3 private context
727 * @param databuf the input data
728 */
729
8af33cb3 730static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
6ba7f78b 731 float **out_samples)
10e26bc7
BL
732{
733 int result, i;
734 float *p1, *p2, *p3, *p4;
15ae1959 735 uint8_t *ptr1;
10e26bc7
BL
736
737 if (q->codingMode == JOINT_STEREO) {
738
739 /* channel coupling mode */
740 /* decode Sound Unit 1 */
741 init_get_bits(&q->gb,databuf,q->bits_per_frame);
742
6ba7f78b 743 result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, out_samples[0], 0, JOINT_STEREO);
10e26bc7 744 if (result != 0)
d4b63054 745 return result;
10e26bc7
BL
746
747 /* Framedata of the su2 in the joint-stereo mode is encoded in
748 * reverse byte order so we need to swap it first. */
15ae1959
AJ
749 if (databuf == q->decoded_bytes_buffer) {
750 uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1;
751 ptr1 = q->decoded_bytes_buffer;
b37b1306
AJ
752 for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
753 FFSWAP(uint8_t,*ptr1,*ptr2);
754 }
15ae1959
AJ
755 } else {
756 const uint8_t *ptr2 = databuf+q->bytes_per_frame-1;
757 for (i = 0; i < q->bytes_per_frame; i++)
758 q->decoded_bytes_buffer[i] = *ptr2--;
759 }
10e26bc7
BL
760
761 /* Skip the sync codes (0xF8). */
15ae1959 762 ptr1 = q->decoded_bytes_buffer;
10e26bc7
BL
763 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
764 if (i >= q->bytes_per_frame)
8f98577d 765 return AVERROR_INVALIDDATA;
10e26bc7
BL
766 }
767
768
769 /* set the bitstream reader at the start of the second Sound Unit*/
770 init_get_bits(&q->gb,ptr1,q->bits_per_frame);
771
772 /* Fill the Weighting coeffs delay buffer */
773 memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
5fc32c27 774 q->weighting_delay[4] = get_bits1(&q->gb);
10e26bc7
BL
775 q->weighting_delay[5] = get_bits(&q->gb,3);
776
777 for (i = 0; i < 4; i++) {
778 q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
779 q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
780 q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
781 }
782
783 /* Decode Sound Unit 2. */
6ba7f78b 784 result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], out_samples[1], 1, JOINT_STEREO);
10e26bc7 785 if (result != 0)
d4b63054 786 return result;
10e26bc7
BL
787
788 /* Reconstruct the channel coefficients. */
6ba7f78b 789 reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
10e26bc7 790
6ba7f78b 791 channelWeighting(out_samples[0], out_samples[1], q->weighting_delay);
10e26bc7
BL
792
793 } else {
794 /* normal stereo mode or mono */
795 /* Decode the channel sound units. */
796 for (i=0 ; i<q->channels ; i++) {
797
798 /* Set the bitstream reader at the start of a channel sound unit. */
ee41963f
DB
799 init_get_bits(&q->gb,
800 databuf + i * q->bytes_per_frame / q->channels,
801 q->bits_per_frame / q->channels);
10e26bc7 802
6ba7f78b 803 result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], out_samples[i], i, q->codingMode);
10e26bc7 804 if (result != 0)
d4b63054 805 return result;
10e26bc7
BL
806 }
807 }
808
809 /* Apply the iQMF synthesis filter. */
10e26bc7 810 for (i=0 ; i<q->channels ; i++) {
6ba7f78b 811 p1 = out_samples[i];
10e26bc7
BL
812 p2= p1+256;
813 p3= p2+256;
814 p4= p3+256;
99560a4c
MS
815 ff_atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
816 ff_atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
817 ff_atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
10e26bc7
BL
818 }
819
820 return 0;
821}
822
823
824/**
825 * Atrac frame decoding
826 *
827 * @param avctx pointer to the AVCodecContext
828 */
829
0eea2129
JR
830static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
831 int *got_frame_ptr, AVPacket *avpkt)
832{
7a00bbad
TB
833 const uint8_t *buf = avpkt->data;
834 int buf_size = avpkt->size;
10e26bc7 835 ATRAC3Context *q = avctx->priv_data;
0eea2129 836 int result;
15ae1959 837 const uint8_t* databuf;
10e26bc7 838
46a76dec
VS
839 if (buf_size < avctx->block_align) {
840 av_log(avctx, AV_LOG_ERROR,
841 "Frame too small (%d bytes). Truncated file?\n", buf_size);
1fead73d 842 return AVERROR_INVALIDDATA;
46a76dec 843 }
10e26bc7 844
0eea2129
JR
845 /* get output buffer */
846 q->frame.nb_samples = SAMPLES_PER_FRAME;
847 if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) {
848 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
849 return result;
7e4881a2
JR
850 }
851
10e26bc7
BL
852 /* Check if we need to descramble and what buffer to pass on. */
853 if (q->scrambled_stream) {
854 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
855 databuf = q->decoded_bytes_buffer;
856 } else {
857 databuf = buf;
858 }
859
9af4eaa8 860 result = decodeFrame(q, databuf, (float **)q->frame.extended_data);
10e26bc7
BL
861
862 if (result != 0) {
863 av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
8f98577d 864 return result;
10e26bc7
BL
865 }
866
0eea2129
JR
867 *got_frame_ptr = 1;
868 *(AVFrame *)data = q->frame;
10e26bc7
BL
869
870 return avctx->block_align;
871}
872
873
874/**
875 * Atrac3 initialization
876 *
877 * @param avctx pointer to the AVCodecContext
878 */
879
5ef251e5 880static av_cold int atrac3_decode_init(AVCodecContext *avctx)
10e26bc7 881{
47b61702 882 int i, ret;
8687f767 883 const uint8_t *edata_ptr = avctx->extradata;
10e26bc7 884 ATRAC3Context *q = avctx->priv_data;
031b1cbe
BL
885 static VLC_TYPE atrac3_vlc_table[4096][2];
886 static int vlcs_initialized = 0;
10e26bc7
BL
887
888 /* Take data from the AVCodecContext (RM container). */
889 q->sample_rate = avctx->sample_rate;
890 q->channels = avctx->channels;
891 q->bit_rate = avctx->bit_rate;
892 q->bits_per_frame = avctx->block_align * 8;
893 q->bytes_per_frame = avctx->block_align;
894
895 /* Take care of the codec-specific extradata. */
896 if (avctx->extradata_size == 14) {
897 /* Parse the extradata, WAV format */
898 av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1
899 q->samples_per_channel = bytestream_get_le32(&edata_ptr);
900 q->codingMode = bytestream_get_le16(&edata_ptr);
901 av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
902 q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1
903 av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0
904
905 /* setup */
c9161385 906 q->samples_per_frame = SAMPLES_PER_FRAME * q->channels;
10e26bc7
BL
907 q->atrac3version = 4;
908 q->delay = 0x88E;
909 if (q->codingMode)
910 q->codingMode = JOINT_STEREO;
911 else
912 q->codingMode = STEREO;
913
914 q->scrambled_stream = 0;
915
916 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)) {
917 } else {
918 av_log(avctx,AV_LOG_ERROR,"Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor);
8f98577d 919 return AVERROR_INVALIDDATA;
10e26bc7
BL
920 }
921
922 } else if (avctx->extradata_size == 10) {
923 /* Parse the extradata, RM format. */
924 q->atrac3version = bytestream_get_be32(&edata_ptr);
925 q->samples_per_frame = bytestream_get_be16(&edata_ptr);
926 q->delay = bytestream_get_be16(&edata_ptr);
927 q->codingMode = bytestream_get_be16(&edata_ptr);
928
929 q->samples_per_channel = q->samples_per_frame / q->channels;
930 q->scrambled_stream = 1;
931
932 } else {
933 av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
934 }
935 /* Check the extradata. */
936
937 if (q->atrac3version != 4) {
938 av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
8f98577d 939 return AVERROR_INVALIDDATA;
10e26bc7
BL
940 }
941
c9161385 942 if (q->samples_per_frame != SAMPLES_PER_FRAME && q->samples_per_frame != SAMPLES_PER_FRAME*2) {
10e26bc7 943 av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
8f98577d 944 return AVERROR_INVALIDDATA;
10e26bc7
BL
945 }
946
947 if (q->delay != 0x88E) {
948 av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
8f98577d 949 return AVERROR_INVALIDDATA;
10e26bc7
BL
950 }
951
952 if (q->codingMode == STEREO) {
953 av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
954 } else if (q->codingMode == JOINT_STEREO) {
955 av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
956 } else {
957 av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
8f98577d 958 return AVERROR_INVALIDDATA;
10e26bc7
BL
959 }
960
961 if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
962 av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
8f98577d 963 return AVERROR(EINVAL);
10e26bc7
BL
964 }
965
966
967 if(avctx->block_align >= UINT_MAX/2)
8f98577d 968 return AVERROR(EINVAL);
10e26bc7
BL
969
970 /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
971 * this is for the bitstream reader. */
972 if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE))) == NULL)
6611c0b4 973 return AVERROR(ENOMEM);
10e26bc7
BL
974
975
976 /* Initialize the VLC tables. */
031b1cbe 977 if (!vlcs_initialized) {
7dd55689
BL
978 for (i=0 ; i<7 ; i++) {
979 spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
980 spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - atrac3_vlc_offs[i];
981 init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
982 huff_bits[i], 1, 1,
983 huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
984 }
031b1cbe 985 vlcs_initialized = 1;
10e26bc7
BL
986 }
987
9af4eaa8 988 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
20732246 989
9af4eaa8 990 if ((ret = init_atrac3_transforms(q))) {
47b61702
JR
991 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
992 av_freep(&q->decoded_bytes_buffer);
993 return ret;
994 }
10e26bc7 995
99560a4c 996 ff_atrac_generate_tables();
10e26bc7
BL
997
998 /* Generate gain tables. */
999 for (i=0 ; i<16 ; i++)
1000 gain_tab1[i] = powf (2.0, (4 - i));
1001
1002 for (i=-15 ; i<16 ; i++)
1003 gain_tab2[i+15] = powf (2.0, i * -0.125);
1004
1005 /* init the joint-stereo decoding data */
1006 q->weighting_delay[0] = 0;
1007 q->weighting_delay[1] = 7;
1008 q->weighting_delay[2] = 0;
1009 q->weighting_delay[3] = 7;
1010 q->weighting_delay[4] = 0;
1011 q->weighting_delay[5] = 7;
1012
1013 for (i=0; i<4; i++) {
1014 q->matrix_coeff_index_prev[i] = 3;
1015 q->matrix_coeff_index_now[i] = 3;
1016 q->matrix_coeff_index_next[i] = 3;
1017 }
1018
d5a7229b 1019 avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
5e76b8bb 1020 ff_fmt_convert_init(&q->fmt_conv, avctx);
10e26bc7
BL
1021
1022 q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
6654296c 1023 if (!q->pUnits) {
47b61702 1024 atrac3_decode_close(avctx);
6654296c
PI
1025 return AVERROR(ENOMEM);
1026 }
10e26bc7 1027
0eea2129
JR
1028 avcodec_get_frame_defaults(&q->frame);
1029 avctx->coded_frame = &q->frame;
1030
10e26bc7
BL
1031 return 0;
1032}
1033
1034
d36beb3f 1035AVCodec ff_atrac3_decoder =
10e26bc7 1036{
00c3b67b
MS
1037 .name = "atrac3",
1038 .type = AVMEDIA_TYPE_AUDIO,
36ef5369 1039 .id = AV_CODEC_ID_ATRAC3,
10e26bc7 1040 .priv_data_size = sizeof(ATRAC3Context),
00c3b67b
MS
1041 .init = atrac3_decode_init,
1042 .close = atrac3_decode_close,
1043 .decode = atrac3_decode_frame,
1044 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
1045 .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
9af4eaa8
JR
1046 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1047 AV_SAMPLE_FMT_NONE },
10e26bc7 1048};