Mercurial > libavcodec.hg
annotate atrac3.c @ 10184:9a577b684548 libavcodec
Remove useless code causing crash in Targa decoder
author | kostya |
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date | Thu, 17 Sep 2009 05:01:27 +0000 |
parents | b49a14edba84 |
children | 38ab367d4231 |
rev | line source |
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4856 | 1 /* |
2 * Atrac 3 compatible decoder | |
6844 | 3 * Copyright (c) 2006-2008 Maxim Poliakovski |
4 * Copyright (c) 2006-2008 Benjamin Larsson | |
4856 | 5 * |
6 * This file is part of FFmpeg. | |
7 * | |
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. | |
12 * | |
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. | |
17 * | |
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 | |
21 */ | |
22 | |
23 /** | |
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24 * @file libavcodec/atrac3.c |
4856 | 25 * Atrac 3 compatible decoder. |
6844 | 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). | |
4856 | 30 * |
31 * To use this decoder, a calling application must supply the extradata | |
6844 | 32 * bytes provided in the containers above. |
4856 | 33 */ |
34 | |
35 #include <math.h> | |
36 #include <stddef.h> | |
37 #include <stdio.h> | |
38 | |
39 #include "avcodec.h" | |
9428 | 40 #include "get_bits.h" |
4856 | 41 #include "dsputil.h" |
42 #include "bytestream.h" | |
43 | |
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44 #include "atrac.h" |
4856 | 45 #include "atrac3data.h" |
46 | |
47 #define JOINT_STEREO 0x12 | |
48 #define STEREO 0x2 | |
49 | |
50 | |
51 /* These structures are needed to store the parsed gain control data. */ | |
52 typedef struct { | |
53 int num_gain_data; | |
54 int levcode[8]; | |
55 int loccode[8]; | |
56 } gain_info; | |
57 | |
58 typedef struct { | |
59 gain_info gBlock[4]; | |
60 } gain_block; | |
61 | |
62 typedef struct { | |
63 int pos; | |
64 int numCoefs; | |
65 float coef[8]; | |
66 } tonal_component; | |
67 | |
68 typedef struct { | |
69 int bandsCoded; | |
70 int numComponents; | |
71 tonal_component components[64]; | |
72 float prevFrame[1024]; | |
73 int gcBlkSwitch; | |
74 gain_block gainBlock[2]; | |
75 | |
76 DECLARE_ALIGNED_16(float, spectrum[1024]); | |
77 DECLARE_ALIGNED_16(float, IMDCT_buf[1024]); | |
78 | |
79 float delayBuf1[46]; ///<qmf delay buffers | |
80 float delayBuf2[46]; | |
81 float delayBuf3[46]; | |
82 } channel_unit; | |
83 | |
84 typedef struct { | |
85 GetBitContext gb; | |
86 //@{ | |
87 /** stream data */ | |
88 int channels; | |
89 int codingMode; | |
90 int bit_rate; | |
91 int sample_rate; | |
92 int samples_per_channel; | |
93 int samples_per_frame; | |
94 | |
95 int bits_per_frame; | |
96 int bytes_per_frame; | |
97 int pBs; | |
98 channel_unit* pUnits; | |
99 //@} | |
100 //@{ | |
101 /** joint-stereo related variables */ | |
102 int matrix_coeff_index_prev[4]; | |
103 int matrix_coeff_index_now[4]; | |
104 int matrix_coeff_index_next[4]; | |
105 int weighting_delay[6]; | |
106 //@} | |
107 //@{ | |
108 /** data buffers */ | |
109 float outSamples[2048]; | |
110 uint8_t* decoded_bytes_buffer; | |
111 float tempBuf[1070]; | |
112 //@} | |
113 //@{ | |
114 /** extradata */ | |
115 int atrac3version; | |
116 int delay; | |
117 int scrambled_stream; | |
118 int frame_factor; | |
119 //@} | |
120 } ATRAC3Context; | |
121 | |
122 static DECLARE_ALIGNED_16(float,mdct_window[512]); | |
123 static VLC spectral_coeff_tab[7]; | |
124 static float gain_tab1[16]; | |
125 static float gain_tab2[31]; | |
126 static MDCTContext mdct_ctx; | |
127 static DSPContext dsp; | |
128 | |
129 | |
130 /** | |
131 * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands | |
132 * caused by the reverse spectra of the QMF. | |
133 * | |
134 * @param pInput float input | |
135 * @param pOutput float output | |
136 * @param odd_band 1 if the band is an odd band | |
137 */ | |
138 | |
7546 | 139 static void IMLT(float *pInput, float *pOutput, int odd_band) |
4856 | 140 { |
141 int i; | |
142 | |
143 if (odd_band) { | |
144 /** | |
145 * Reverse the odd bands before IMDCT, this is an effect of the QMF transform | |
146 * or it gives better compression to do it this way. | |
147 * FIXME: It should be possible to handle this in ff_imdct_calc | |
148 * for that to happen a modification of the prerotation step of | |
149 * all SIMD code and C code is needed. | |
150 * Or fix the functions before so they generate a pre reversed spectrum. | |
151 */ | |
152 | |
153 for (i=0; i<128; i++) | |
154 FFSWAP(float, pInput[i], pInput[255-i]); | |
155 } | |
156 | |
7547 | 157 ff_imdct_calc(&mdct_ctx,pOutput,pInput); |
4856 | 158 |
159 /* Perform windowing on the output. */ | |
160 dsp.vector_fmul(pOutput,mdct_window,512); | |
161 | |
162 } | |
163 | |
164 | |
165 /** | |
166 * Atrac 3 indata descrambling, only used for data coming from the rm container | |
167 * | |
168 * @param in pointer to 8 bit array of indata | |
169 * @param bits amount of bits | |
170 * @param out pointer to 8 bit array of outdata | |
171 */ | |
172 | |
6228 | 173 static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){ |
4856 | 174 int i, off; |
175 uint32_t c; | |
6228 | 176 const uint32_t* buf; |
4856 | 177 uint32_t* obuf = (uint32_t*) out; |
178 | |
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179 off = (intptr_t)inbuffer & 3; |
6228 | 180 buf = (const uint32_t*) (inbuffer - off); |
4856 | 181 c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8)))); |
182 bytes += 3 + off; | |
183 for (i = 0; i < bytes/4; i++) | |
184 obuf[i] = c ^ buf[i]; | |
185 | |
186 if (off) | |
187 av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off); | |
188 | |
189 return off; | |
190 } | |
191 | |
192 | |
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193 static av_cold void init_atrac3_transforms(ATRAC3Context *q) { |
4856 | 194 float enc_window[256]; |
195 float s; | |
196 int i; | |
197 | |
198 /* Generate the mdct window, for details see | |
199 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */ | |
200 for (i=0 ; i<256; i++) | |
201 enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5; | |
202 | |
203 if (!mdct_window[0]) | |
204 for (i=0 ; i<256; i++) { | |
205 mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]); | |
206 mdct_window[511-i] = mdct_window[i]; | |
207 } | |
208 | |
209 /* Initialize the MDCT transform. */ | |
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210 ff_mdct_init(&mdct_ctx, 9, 1, 1.0); |
4856 | 211 } |
212 | |
213 /** | |
214 * Atrac3 uninit, free all allocated memory | |
215 */ | |
216 | |
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217 static av_cold int atrac3_decode_close(AVCodecContext *avctx) |
4856 | 218 { |
219 ATRAC3Context *q = avctx->priv_data; | |
220 | |
221 av_free(q->pUnits); | |
222 av_free(q->decoded_bytes_buffer); | |
223 | |
224 return 0; | |
225 } | |
226 | |
227 /** | |
228 / * Mantissa decoding | |
229 * | |
230 * @param gb the GetBit context | |
231 * @param selector what table is the output values coded with | |
232 * @param codingFlag constant length coding or variable length coding | |
233 * @param mantissas mantissa output table | |
234 * @param numCodes amount of values to get | |
235 */ | |
236 | |
237 static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes) | |
238 { | |
239 int numBits, cnt, code, huffSymb; | |
240 | |
241 if (selector == 1) | |
242 numCodes /= 2; | |
243 | |
244 if (codingFlag != 0) { | |
245 /* constant length coding (CLC) */ | |
246 numBits = CLCLengthTab[selector]; | |
247 | |
248 if (selector > 1) { | |
249 for (cnt = 0; cnt < numCodes; cnt++) { | |
250 if (numBits) | |
251 code = get_sbits(gb, numBits); | |
252 else | |
253 code = 0; | |
254 mantissas[cnt] = code; | |
255 } | |
256 } else { | |
257 for (cnt = 0; cnt < numCodes; cnt++) { | |
258 if (numBits) | |
259 code = get_bits(gb, numBits); //numBits is always 4 in this case | |
260 else | |
261 code = 0; | |
262 mantissas[cnt*2] = seTab_0[code >> 2]; | |
263 mantissas[cnt*2+1] = seTab_0[code & 3]; | |
264 } | |
265 } | |
266 } else { | |
267 /* variable length coding (VLC) */ | |
268 if (selector != 1) { | |
269 for (cnt = 0; cnt < numCodes; cnt++) { | |
270 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3); | |
271 huffSymb += 1; | |
272 code = huffSymb >> 1; | |
273 if (huffSymb & 1) | |
274 code = -code; | |
275 mantissas[cnt] = code; | |
276 } | |
277 } else { | |
278 for (cnt = 0; cnt < numCodes; cnt++) { | |
279 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3); | |
280 mantissas[cnt*2] = decTable1[huffSymb*2]; | |
281 mantissas[cnt*2+1] = decTable1[huffSymb*2+1]; | |
282 } | |
283 } | |
284 } | |
285 } | |
286 | |
287 /** | |
288 * Restore the quantized band spectrum coefficients | |
289 * | |
290 * @param gb the GetBit context | |
291 * @param pOut decoded band spectrum | |
292 * @return outSubbands subband counter, fix for broken specification/files | |
293 */ | |
294 | |
295 static int decodeSpectrum (GetBitContext *gb, float *pOut) | |
296 { | |
297 int numSubbands, codingMode, cnt, first, last, subbWidth, *pIn; | |
298 int subband_vlc_index[32], SF_idxs[32]; | |
299 int mantissas[128]; | |
300 float SF; | |
301 | |
302 numSubbands = get_bits(gb, 5); // number of coded subbands | |
5513 | 303 codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC |
4856 | 304 |
305 /* Get the VLC selector table for the subbands, 0 means not coded. */ | |
306 for (cnt = 0; cnt <= numSubbands; cnt++) | |
307 subband_vlc_index[cnt] = get_bits(gb, 3); | |
308 | |
309 /* Read the scale factor indexes from the stream. */ | |
310 for (cnt = 0; cnt <= numSubbands; cnt++) { | |
311 if (subband_vlc_index[cnt] != 0) | |
312 SF_idxs[cnt] = get_bits(gb, 6); | |
313 } | |
314 | |
315 for (cnt = 0; cnt <= numSubbands; cnt++) { | |
316 first = subbandTab[cnt]; | |
317 last = subbandTab[cnt+1]; | |
318 | |
319 subbWidth = last - first; | |
320 | |
321 if (subband_vlc_index[cnt] != 0) { | |
322 /* Decode spectral coefficients for this subband. */ | |
323 /* TODO: This can be done faster is several blocks share the | |
324 * same VLC selector (subband_vlc_index) */ | |
325 readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth); | |
326 | |
327 /* Decode the scale factor for this subband. */ | |
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328 SF = sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]]; |
4856 | 329 |
330 /* Inverse quantize the coefficients. */ | |
331 for (pIn=mantissas ; first<last; first++, pIn++) | |
332 pOut[first] = *pIn * SF; | |
333 } else { | |
334 /* This subband was not coded, so zero the entire subband. */ | |
335 memset(pOut+first, 0, subbWidth*sizeof(float)); | |
336 } | |
337 } | |
338 | |
339 /* Clear the subbands that were not coded. */ | |
340 first = subbandTab[cnt]; | |
341 memset(pOut+first, 0, (1024 - first) * sizeof(float)); | |
342 return numSubbands; | |
343 } | |
344 | |
345 /** | |
346 * Restore the quantized tonal components | |
347 * | |
348 * @param gb the GetBit context | |
349 * @param pComponent tone component | |
350 * @param numBands amount of coded bands | |
351 */ | |
352 | |
4865 | 353 static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands) |
4856 | 354 { |
355 int i,j,k,cnt; | |
4865 | 356 int components, coding_mode_selector, coding_mode, coded_values_per_component; |
4856 | 357 int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components; |
358 int band_flags[4], mantissa[8]; | |
359 float *pCoef; | |
360 float scalefactor; | |
4865 | 361 int component_count = 0; |
4856 | 362 |
363 components = get_bits(gb,5); | |
364 | |
365 /* no tonal components */ | |
366 if (components == 0) | |
367 return 0; | |
368 | |
369 coding_mode_selector = get_bits(gb,2); | |
370 if (coding_mode_selector == 2) | |
371 return -1; | |
372 | |
373 coding_mode = coding_mode_selector & 1; | |
374 | |
375 for (i = 0; i < components; i++) { | |
376 for (cnt = 0; cnt <= numBands; cnt++) | |
377 band_flags[cnt] = get_bits1(gb); | |
378 | |
379 coded_values_per_component = get_bits(gb,3); | |
380 | |
381 quant_step_index = get_bits(gb,3); | |
382 if (quant_step_index <= 1) | |
383 return -1; | |
384 | |
385 if (coding_mode_selector == 3) | |
386 coding_mode = get_bits1(gb); | |
387 | |
388 for (j = 0; j < (numBands + 1) * 4; j++) { | |
389 if (band_flags[j >> 2] == 0) | |
390 continue; | |
391 | |
392 coded_components = get_bits(gb,3); | |
393 | |
394 for (k=0; k<coded_components; k++) { | |
395 sfIndx = get_bits(gb,6); | |
396 pComponent[component_count].pos = j * 64 + (get_bits(gb,6)); | |
397 max_coded_values = 1024 - pComponent[component_count].pos; | |
398 coded_values = coded_values_per_component + 1; | |
399 coded_values = FFMIN(max_coded_values,coded_values); | |
400 | |
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401 scalefactor = sf_table[sfIndx] * iMaxQuant[quant_step_index]; |
4856 | 402 |
403 readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values); | |
404 | |
405 pComponent[component_count].numCoefs = coded_values; | |
406 | |
407 /* inverse quant */ | |
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408 pCoef = pComponent[component_count].coef; |
4856 | 409 for (cnt = 0; cnt < coded_values; cnt++) |
410 pCoef[cnt] = mantissa[cnt] * scalefactor; | |
411 | |
412 component_count++; | |
413 } | |
414 } | |
415 } | |
416 | |
4865 | 417 return component_count; |
4856 | 418 } |
419 | |
420 /** | |
421 * Decode gain parameters for the coded bands | |
422 * | |
423 * @param gb the GetBit context | |
424 * @param pGb the gainblock for the current band | |
425 * @param numBands amount of coded bands | |
426 */ | |
427 | |
428 static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands) | |
429 { | |
430 int i, cf, numData; | |
431 int *pLevel, *pLoc; | |
432 | |
433 gain_info *pGain = pGb->gBlock; | |
434 | |
435 for (i=0 ; i<=numBands; i++) | |
436 { | |
437 numData = get_bits(gb,3); | |
438 pGain[i].num_gain_data = numData; | |
439 pLevel = pGain[i].levcode; | |
440 pLoc = pGain[i].loccode; | |
441 | |
442 for (cf = 0; cf < numData; cf++){ | |
443 pLevel[cf]= get_bits(gb,4); | |
444 pLoc [cf]= get_bits(gb,5); | |
445 if(cf && pLoc[cf] <= pLoc[cf-1]) | |
446 return -1; | |
447 } | |
448 } | |
449 | |
450 /* Clear the unused blocks. */ | |
451 for (; i<4 ; i++) | |
452 pGain[i].num_gain_data = 0; | |
453 | |
454 return 0; | |
455 } | |
456 | |
457 /** | |
458 * Apply gain parameters and perform the MDCT overlapping part | |
459 * | |
460 * @param pIn input float buffer | |
461 * @param pPrev previous float buffer to perform overlap against | |
462 * @param pOut output float buffer | |
463 * @param pGain1 current band gain info | |
464 * @param pGain2 next band gain info | |
465 */ | |
466 | |
467 static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2) | |
468 { | |
469 /* gain compensation function */ | |
470 float gain1, gain2, gain_inc; | |
471 int cnt, numdata, nsample, startLoc, endLoc; | |
472 | |
473 | |
474 if (pGain2->num_gain_data == 0) | |
475 gain1 = 1.0; | |
476 else | |
477 gain1 = gain_tab1[pGain2->levcode[0]]; | |
478 | |
479 if (pGain1->num_gain_data == 0) { | |
480 for (cnt = 0; cnt < 256; cnt++) | |
481 pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt]; | |
482 } else { | |
483 numdata = pGain1->num_gain_data; | |
484 pGain1->loccode[numdata] = 32; | |
485 pGain1->levcode[numdata] = 4; | |
486 | |
487 nsample = 0; // current sample = 0 | |
488 | |
489 for (cnt = 0; cnt < numdata; cnt++) { | |
490 startLoc = pGain1->loccode[cnt] * 8; | |
491 endLoc = startLoc + 8; | |
492 | |
493 gain2 = gain_tab1[pGain1->levcode[cnt]]; | |
494 gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15]; | |
495 | |
496 /* interpolate */ | |
497 for (; nsample < startLoc; nsample++) | |
498 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; | |
499 | |
500 /* interpolation is done over eight samples */ | |
501 for (; nsample < endLoc; nsample++) { | |
502 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; | |
503 gain2 *= gain_inc; | |
504 } | |
505 } | |
506 | |
507 for (; nsample < 256; nsample++) | |
508 pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample]; | |
509 } | |
510 | |
511 /* Delay for the overlapping part. */ | |
512 memcpy(pPrev, &pIn[256], 256*sizeof(float)); | |
513 } | |
514 | |
515 /** | |
516 * Combine the tonal band spectrum and regular band spectrum | |
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517 * Return position of the last tonal coefficient |
4856 | 518 * |
519 * @param pSpectrum output spectrum buffer | |
520 * @param numComponents amount of tonal components | |
521 * @param pComponent tonal components for this band | |
522 */ | |
523 | |
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524 static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent) |
4856 | 525 { |
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526 int cnt, i, lastPos = -1; |
4856 | 527 float *pIn, *pOut; |
528 | |
529 for (cnt = 0; cnt < numComponents; cnt++){ | |
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530 lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos); |
4856 | 531 pIn = pComponent[cnt].coef; |
532 pOut = &(pSpectrum[pComponent[cnt].pos]); | |
533 | |
534 for (i=0 ; i<pComponent[cnt].numCoefs ; i++) | |
535 pOut[i] += pIn[i]; | |
536 } | |
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537 |
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538 return lastPos; |
4856 | 539 } |
540 | |
541 | |
542 #define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old))) | |
543 | |
544 static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode) | |
545 { | |
546 int i, band, nsample, s1, s2; | |
547 float c1, c2; | |
548 float mc1_l, mc1_r, mc2_l, mc2_r; | |
549 | |
550 for (i=0,band = 0; band < 4*256; band+=256,i++) { | |
551 s1 = pPrevCode[i]; | |
552 s2 = pCurrCode[i]; | |
553 nsample = 0; | |
554 | |
555 if (s1 != s2) { | |
556 /* Selector value changed, interpolation needed. */ | |
557 mc1_l = matrixCoeffs[s1*2]; | |
558 mc1_r = matrixCoeffs[s1*2+1]; | |
559 mc2_l = matrixCoeffs[s2*2]; | |
560 mc2_r = matrixCoeffs[s2*2+1]; | |
561 | |
562 /* Interpolation is done over the first eight samples. */ | |
563 for(; nsample < 8; nsample++) { | |
564 c1 = su1[band+nsample]; | |
565 c2 = su2[band+nsample]; | |
566 c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample); | |
567 su1[band+nsample] = c2; | |
568 su2[band+nsample] = c1 * 2.0 - c2; | |
569 } | |
570 } | |
571 | |
572 /* Apply the matrix without interpolation. */ | |
573 switch (s2) { | |
574 case 0: /* M/S decoding */ | |
575 for (; nsample < 256; nsample++) { | |
576 c1 = su1[band+nsample]; | |
577 c2 = su2[band+nsample]; | |
578 su1[band+nsample] = c2 * 2.0; | |
579 su2[band+nsample] = (c1 - c2) * 2.0; | |
580 } | |
581 break; | |
582 | |
583 case 1: | |
584 for (; nsample < 256; nsample++) { | |
585 c1 = su1[band+nsample]; | |
586 c2 = su2[band+nsample]; | |
587 su1[band+nsample] = (c1 + c2) * 2.0; | |
588 su2[band+nsample] = c2 * -2.0; | |
589 } | |
590 break; | |
591 case 2: | |
592 case 3: | |
593 for (; nsample < 256; nsample++) { | |
594 c1 = su1[band+nsample]; | |
595 c2 = su2[band+nsample]; | |
596 su1[band+nsample] = c1 + c2; | |
597 su2[band+nsample] = c1 - c2; | |
598 } | |
599 break; | |
600 default: | |
601 assert(0); | |
602 } | |
603 } | |
604 } | |
605 | |
606 static void getChannelWeights (int indx, int flag, float ch[2]){ | |
607 | |
608 if (indx == 7) { | |
609 ch[0] = 1.0; | |
610 ch[1] = 1.0; | |
611 } else { | |
612 ch[0] = (float)(indx & 7) / 7.0; | |
613 ch[1] = sqrt(2 - ch[0]*ch[0]); | |
614 if(flag) | |
615 FFSWAP(float, ch[0], ch[1]); | |
616 } | |
617 } | |
618 | |
619 static void channelWeighting (float *su1, float *su2, int *p3) | |
620 { | |
621 int band, nsample; | |
622 /* w[x][y] y=0 is left y=1 is right */ | |
623 float w[2][2]; | |
624 | |
625 if (p3[1] != 7 || p3[3] != 7){ | |
626 getChannelWeights(p3[1], p3[0], w[0]); | |
627 getChannelWeights(p3[3], p3[2], w[1]); | |
628 | |
629 for(band = 1; band < 4; band++) { | |
630 /* scale the channels by the weights */ | |
631 for(nsample = 0; nsample < 8; nsample++) { | |
632 su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample); | |
633 su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample); | |
634 } | |
635 | |
636 for(; nsample < 256; nsample++) { | |
637 su1[band*256+nsample] *= w[1][0]; | |
638 su2[band*256+nsample] *= w[1][1]; | |
639 } | |
640 } | |
641 } | |
642 } | |
643 | |
644 | |
645 /** | |
646 * Decode a Sound Unit | |
647 * | |
648 * @param gb the GetBit context | |
649 * @param pSnd the channel unit to be used | |
650 * @param pOut the decoded samples before IQMF in float representation | |
651 * @param channelNum channel number | |
652 * @param codingMode the coding mode (JOINT_STEREO or regular stereo/mono) | |
653 */ | |
654 | |
655 | |
656 static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode) | |
657 { | |
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658 int band, result=0, numSubbands, lastTonal, numBands; |
4856 | 659 |
660 if (codingMode == JOINT_STEREO && channelNum == 1) { | |
661 if (get_bits(gb,2) != 3) { | |
662 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n"); | |
663 return -1; | |
664 } | |
665 } else { | |
666 if (get_bits(gb,6) != 0x28) { | |
667 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n"); | |
668 return -1; | |
669 } | |
670 } | |
671 | |
672 /* number of coded QMF bands */ | |
673 pSnd->bandsCoded = get_bits(gb,2); | |
674 | |
675 result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded); | |
676 if (result) return result; | |
677 | |
4865 | 678 pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded); |
679 if (pSnd->numComponents == -1) return -1; | |
4856 | 680 |
681 numSubbands = decodeSpectrum (gb, pSnd->spectrum); | |
682 | |
683 /* Merge the decoded spectrum and tonal components. */ | |
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684 lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components); |
4856 | 685 |
686 | |
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687 /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */ |
4856 | 688 numBands = (subbandTab[numSubbands] - 1) >> 8; |
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689 if (lastTonal >= 0) |
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690 numBands = FFMAX((lastTonal + 256) >> 8, numBands); |
4856 | 691 |
692 | |
693 /* Reconstruct time domain samples. */ | |
694 for (band=0; band<4; band++) { | |
695 /* Perform the IMDCT step without overlapping. */ | |
696 if (band <= numBands) { | |
7546 | 697 IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1); |
4856 | 698 } else |
699 memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float)); | |
700 | |
701 /* gain compensation and overlapping */ | |
702 gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]), | |
703 &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]), | |
704 &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band])); | |
705 } | |
706 | |
707 /* Swap the gain control buffers for the next frame. */ | |
708 pSnd->gcBlkSwitch ^= 1; | |
709 | |
710 return 0; | |
711 } | |
712 | |
713 /** | |
714 * Frame handling | |
715 * | |
716 * @param q Atrac3 private context | |
717 * @param databuf the input data | |
718 */ | |
719 | |
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720 static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf) |
4856 | 721 { |
722 int result, i; | |
723 float *p1, *p2, *p3, *p4; | |
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724 uint8_t *ptr1; |
4856 | 725 |
726 if (q->codingMode == JOINT_STEREO) { | |
727 | |
728 /* channel coupling mode */ | |
729 /* decode Sound Unit 1 */ | |
730 init_get_bits(&q->gb,databuf,q->bits_per_frame); | |
731 | |
732 result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO); | |
733 if (result != 0) | |
734 return (result); | |
735 | |
736 /* Framedata of the su2 in the joint-stereo mode is encoded in | |
737 * reverse byte order so we need to swap it first. */ | |
7939
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738 if (databuf == q->decoded_bytes_buffer) { |
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739 uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1; |
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740 ptr1 = q->decoded_bytes_buffer; |
7987 | 741 for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) { |
742 FFSWAP(uint8_t,*ptr1,*ptr2); | |
743 } | |
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744 } else { |
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745 const uint8_t *ptr2 = databuf+q->bytes_per_frame-1; |
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746 for (i = 0; i < q->bytes_per_frame; i++) |
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747 q->decoded_bytes_buffer[i] = *ptr2--; |
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748 } |
4856 | 749 |
750 /* Skip the sync codes (0xF8). */ | |
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751 ptr1 = q->decoded_bytes_buffer; |
4856 | 752 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) { |
753 if (i >= q->bytes_per_frame) | |
754 return -1; | |
755 } | |
756 | |
757 | |
758 /* set the bitstream reader at the start of the second Sound Unit*/ | |
759 init_get_bits(&q->gb,ptr1,q->bits_per_frame); | |
760 | |
761 /* Fill the Weighting coeffs delay buffer */ | |
762 memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int)); | |
5513 | 763 q->weighting_delay[4] = get_bits1(&q->gb); |
4856 | 764 q->weighting_delay[5] = get_bits(&q->gb,3); |
765 | |
766 for (i = 0; i < 4; i++) { | |
767 q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i]; | |
768 q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i]; | |
769 q->matrix_coeff_index_next[i] = get_bits(&q->gb,2); | |
770 } | |
771 | |
772 /* Decode Sound Unit 2. */ | |
773 result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO); | |
774 if (result != 0) | |
775 return (result); | |
776 | |
777 /* Reconstruct the channel coefficients. */ | |
778 reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now); | |
779 | |
780 channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay); | |
781 | |
782 } else { | |
783 /* normal stereo mode or mono */ | |
784 /* Decode the channel sound units. */ | |
785 for (i=0 ; i<q->channels ; i++) { | |
786 | |
787 /* Set the bitstream reader at the start of a channel sound unit. */ | |
788 init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels); | |
789 | |
790 result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode); | |
791 if (result != 0) | |
792 return (result); | |
793 } | |
794 } | |
795 | |
796 /* Apply the iQMF synthesis filter. */ | |
797 p1= q->outSamples; | |
798 for (i=0 ; i<q->channels ; i++) { | |
799 p2= p1+256; | |
800 p3= p2+256; | |
801 p4= p3+256; | |
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802 atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf); |
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803 atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf); |
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804 atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf); |
4856 | 805 p1 +=1024; |
806 } | |
807 | |
808 return 0; | |
809 } | |
810 | |
811 | |
812 /** | |
813 * Atrac frame decoding | |
814 * | |
815 * @param avctx pointer to the AVCodecContext | |
816 */ | |
817 | |
818 static int atrac3_decode_frame(AVCodecContext *avctx, | |
819 void *data, int *data_size, | |
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820 AVPacket *avpkt) { |
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821 const uint8_t *buf = avpkt->data; |
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822 int buf_size = avpkt->size; |
4856 | 823 ATRAC3Context *q = avctx->priv_data; |
824 int result = 0, i; | |
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825 const uint8_t* databuf; |
4856 | 826 int16_t* samples = data; |
827 | |
828 if (buf_size < avctx->block_align) | |
829 return buf_size; | |
830 | |
831 /* Check if we need to descramble and what buffer to pass on. */ | |
832 if (q->scrambled_stream) { | |
833 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align); | |
834 databuf = q->decoded_bytes_buffer; | |
835 } else { | |
836 databuf = buf; | |
837 } | |
838 | |
839 result = decodeFrame(q, databuf); | |
840 | |
841 if (result != 0) { | |
842 av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n"); | |
843 return -1; | |
844 } | |
845 | |
846 if (q->channels == 1) { | |
847 /* mono */ | |
848 for (i = 0; i<1024; i++) | |
5523 | 849 samples[i] = av_clip_int16(round(q->outSamples[i])); |
4856 | 850 *data_size = 1024 * sizeof(int16_t); |
851 } else { | |
852 /* stereo */ | |
853 for (i = 0; i < 1024; i++) { | |
5523 | 854 samples[i*2] = av_clip_int16(round(q->outSamples[i])); |
855 samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i])); | |
4856 | 856 } |
857 *data_size = 2048 * sizeof(int16_t); | |
858 } | |
859 | |
860 return avctx->block_align; | |
861 } | |
862 | |
863 | |
864 /** | |
865 * Atrac3 initialization | |
866 * | |
867 * @param avctx pointer to the AVCodecContext | |
868 */ | |
869 | |
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870 static av_cold int atrac3_decode_init(AVCodecContext *avctx) |
4856 | 871 { |
872 int i; | |
6228 | 873 const uint8_t *edata_ptr = avctx->extradata; |
4856 | 874 ATRAC3Context *q = avctx->priv_data; |
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875 static VLC_TYPE atrac3_vlc_table[4096][2]; |
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876 static int vlcs_initialized = 0; |
4856 | 877 |
878 /* Take data from the AVCodecContext (RM container). */ | |
879 q->sample_rate = avctx->sample_rate; | |
880 q->channels = avctx->channels; | |
881 q->bit_rate = avctx->bit_rate; | |
882 q->bits_per_frame = avctx->block_align * 8; | |
883 q->bytes_per_frame = avctx->block_align; | |
884 | |
885 /* Take care of the codec-specific extradata. */ | |
886 if (avctx->extradata_size == 14) { | |
887 /* Parse the extradata, WAV format */ | |
888 av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1 | |
889 q->samples_per_channel = bytestream_get_le32(&edata_ptr); | |
890 q->codingMode = bytestream_get_le16(&edata_ptr); | |
891 av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode | |
892 q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1 | |
893 av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0 | |
894 | |
895 /* setup */ | |
896 q->samples_per_frame = 1024 * q->channels; | |
897 q->atrac3version = 4; | |
898 q->delay = 0x88E; | |
899 if (q->codingMode) | |
900 q->codingMode = JOINT_STEREO; | |
901 else | |
902 q->codingMode = STEREO; | |
903 | |
904 q->scrambled_stream = 0; | |
905 | |
906 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)) { | |
907 } else { | |
908 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); | |
909 return -1; | |
910 } | |
911 | |
912 } else if (avctx->extradata_size == 10) { | |
913 /* Parse the extradata, RM format. */ | |
914 q->atrac3version = bytestream_get_be32(&edata_ptr); | |
915 q->samples_per_frame = bytestream_get_be16(&edata_ptr); | |
916 q->delay = bytestream_get_be16(&edata_ptr); | |
917 q->codingMode = bytestream_get_be16(&edata_ptr); | |
918 | |
919 q->samples_per_channel = q->samples_per_frame / q->channels; | |
920 q->scrambled_stream = 1; | |
921 | |
922 } else { | |
923 av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size); | |
924 } | |
925 /* Check the extradata. */ | |
926 | |
927 if (q->atrac3version != 4) { | |
928 av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version); | |
929 return -1; | |
930 } | |
931 | |
932 if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) { | |
933 av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame); | |
934 return -1; | |
935 } | |
936 | |
937 if (q->delay != 0x88E) { | |
938 av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay); | |
939 return -1; | |
940 } | |
941 | |
942 if (q->codingMode == STEREO) { | |
943 av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n"); | |
944 } else if (q->codingMode == JOINT_STEREO) { | |
945 av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n"); | |
946 } else { | |
947 av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode); | |
948 return -1; | |
949 } | |
950 | |
951 if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) { | |
952 av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n"); | |
953 return -1; | |
954 } | |
955 | |
956 | |
957 if(avctx->block_align >= UINT_MAX/2) | |
958 return -1; | |
959 | |
960 /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE, | |
961 * this is for the bitstream reader. */ | |
962 if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE))) == NULL) | |
5407 | 963 return AVERROR(ENOMEM); |
4856 | 964 |
965 | |
966 /* Initialize the VLC tables. */ | |
9666
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967 if (!vlcs_initialized) { |
9667 | 968 for (i=0 ; i<7 ; i++) { |
969 spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]]; | |
970 spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - atrac3_vlc_offs[i]; | |
971 init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i], | |
972 huff_bits[i], 1, 1, | |
973 huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC); | |
974 } | |
9666
c80df3181479
Change from INIT_VLC_USE_STATIC to INIT_VLC_USE_NEW_STATIC in atrac3
banan
parents:
9658
diff
changeset
|
975 vlcs_initialized = 1; |
4856 | 976 } |
977 | |
978 init_atrac3_transforms(q); | |
979 | |
10150
29cedcc646fe
Split out common routines needed in the atrac1 decoder from atrac3.c to atrac.c.
banan
parents:
9667
diff
changeset
|
980 atrac_generate_tables(); |
4856 | 981 |
982 /* Generate gain tables. */ | |
983 for (i=0 ; i<16 ; i++) | |
984 gain_tab1[i] = powf (2.0, (4 - i)); | |
985 | |
986 for (i=-15 ; i<16 ; i++) | |
987 gain_tab2[i+15] = powf (2.0, i * -0.125); | |
988 | |
989 /* init the joint-stereo decoding data */ | |
990 q->weighting_delay[0] = 0; | |
991 q->weighting_delay[1] = 7; | |
992 q->weighting_delay[2] = 0; | |
993 q->weighting_delay[3] = 7; | |
994 q->weighting_delay[4] = 0; | |
995 q->weighting_delay[5] = 7; | |
996 | |
997 for (i=0; i<4; i++) { | |
998 q->matrix_coeff_index_prev[i] = 3; | |
999 q->matrix_coeff_index_now[i] = 3; | |
1000 q->matrix_coeff_index_next[i] = 3; | |
1001 } | |
1002 | |
1003 dsputil_init(&dsp, avctx); | |
1004 | |
1005 q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels); | |
5423 | 1006 if (!q->pUnits) { |
1007 av_free(q->decoded_bytes_buffer); | |
1008 return AVERROR(ENOMEM); | |
1009 } | |
4856 | 1010 |
7451
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
pross
parents:
7040
diff
changeset
|
1011 avctx->sample_fmt = SAMPLE_FMT_S16; |
4856 | 1012 return 0; |
1013 } | |
1014 | |
1015 | |
1016 AVCodec atrac3_decoder = | |
1017 { | |
6716 | 1018 .name = "atrac3", |
4856 | 1019 .type = CODEC_TYPE_AUDIO, |
1020 .id = CODEC_ID_ATRAC3, | |
1021 .priv_data_size = sizeof(ATRAC3Context), | |
1022 .init = atrac3_decode_init, | |
1023 .close = atrac3_decode_close, | |
1024 .decode = atrac3_decode_frame, | |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
6997
diff
changeset
|
1025 .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"), |
4856 | 1026 }; |