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29 * AAC decoder fixed-point implementation
31 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
32 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
34 * This file is part of FFmpeg.
36 * FFmpeg is free software; you can redistribute it and/or
37 * modify it under the terms of the GNU Lesser General Public
38 * License as published by the Free Software Foundation; either
39 * version 2.1 of the License, or (at your option) any later version.
41 * FFmpeg is distributed in the hope that it will be useful,
42 * but WITHOUT ANY WARRANTY; without even the implied warranty of
43 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
44 * Lesser General Public License for more details.
46 * You should have received a copy of the GNU Lesser General Public
47 * License along with FFmpeg; if not, write to the Free Software
48 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
54 * @author Oded Shimon ( ods15 ods15 dyndns org )
55 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
57 * Fixed point implementation
58 * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
64 #include "libavutil/fixed_dsp.h"
65 #include "libavutil/opt.h"
67 #include "codec_internal.h"
72 #include "sinewin_fixed_tablegen.h"
76 #include "aacdectab.h"
77 #include "adts_header.h"
78 #include "cbrt_data.h"
81 #include "mpeg4audio.h"
83 #include "libavutil/intfloat.h"
88 DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_long_1024))[1024];
89 DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_short_128))[128];
91 static av_always_inline void reset_predict_state(PredictorState *ps)
101 ps->var0.mant = 0x20000000;
103 ps->var1.mant = 0x20000000;
107 static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) }; // 2^0, 2^0.25, 2^0.5, 2^0.75
109 static inline int *DEC_SPAIR(int *dst, unsigned idx)
111 dst[0] = (idx & 15) - 4;
112 dst[1] = (idx >> 4 & 15) - 4;
117 static inline int *DEC_SQUAD(int *dst, unsigned idx)
119 dst[0] = (idx & 3) - 1;
120 dst[1] = (idx >> 2 & 3) - 1;
121 dst[2] = (idx >> 4 & 3) - 1;
122 dst[3] = (idx >> 6 & 3) - 1;
127 static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
129 dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
130 dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
135 static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
137 unsigned nz = idx >> 12;
139 dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
142 dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
145 dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
148 dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
153 static void vector_pow43(int *coefs, int len)
157 for (i=0; i<len; i++) {
160 coef = -(int)ff_cbrt_tab_fixed[(-coef) & 8191];
162 coef = (int)ff_cbrt_tab_fixed[ coef & 8191];
167 static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
169 int ssign = scale < 0 ? -1 : 1;
170 int s = FFABS(scale);
172 int i, out, c = exp2tab[s & 3];
174 s = offset - (s >> 2);
177 for (i=0; i<len; i++) {
182 for (i=0; i<len; i++) {
183 out = (int)(((int64_t)src[i] * c) >> 32);
184 dst[i] = ((int)(out+round) >> s) * ssign;
186 } else if (s > -32) {
189 for (i=0; i<len; i++) {
190 out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
191 dst[i] = out * (unsigned)ssign;
194 av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
198 static void noise_scale(int *coefs, int scale, int band_energy, int len)
202 int i, out, c = exp2tab[s & 3];
206 while (band_energy > 0x7fff) {
211 s = 21 + nlz - (s >> 2);
214 for (i=0; i<len; i++) {
218 round = s ? 1 << (s-1) : 0;
219 for (i=0; i<len; i++) {
220 out = (int)(((int64_t)coefs[i] * c) >> 32);
221 coefs[i] = -((int)(out+round) >> s);
228 for (i=0; i<len; i++) {
229 out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
233 for (i=0; i<len; i++)
234 coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
239 static av_always_inline SoftFloat flt16_round(SoftFloat pf)
246 tmp.mant = (pf.mant ^ s) - s;
247 tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
248 tmp.mant = (tmp.mant ^ s) - s;
253 static av_always_inline SoftFloat flt16_even(SoftFloat pf)
260 tmp.mant = (pf.mant ^ s) - s;
261 tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
262 tmp.mant = (tmp.mant ^ s) - s;
267 static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
274 pun.mant = (pf.mant ^ s) - s;
275 pun.mant = pun.mant & 0xFFC00000U;
276 pun.mant = (pun.mant ^ s) - s;
281 static av_always_inline void predict(PredictorState *ps, int *coef,
284 const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
285 const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
289 SoftFloat r0 = ps->r0, r1 = ps->r1;
290 SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
291 SoftFloat var0 = ps->var0, var1 = ps->var1;
294 if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
295 k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
302 if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
303 k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
310 tmp = av_mul_sf(k1, r0);
311 pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
313 int shift = 28 - pv.exp;
317 *coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
319 *coef += (unsigned)pv.mant << -shift;
323 e0 = av_int2sf(*coef, 2);
324 e1 = av_sub_sf(e0, tmp);
326 ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
327 tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
329 ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
330 ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
331 tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
333 ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
335 ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
336 ps->r0 = flt16_trunc(av_mul_sf(a, e0));
340 static const int cce_scale_fixed[8] = {
342 Q30(1.0905077327), //2^(1/8)
343 Q30(1.1892071150), //2^(2/8)
344 Q30(1.2968395547), //2^(3/8)
345 Q30(1.4142135624), //2^(4/8)
346 Q30(1.5422108254), //2^(5/8)
347 Q30(1.6817928305), //2^(6/8)
348 Q30(1.8340080864), //2^(7/8)
352 * Apply dependent channel coupling (applied before IMDCT).
354 * @param index index into coupling gain array
356 static void apply_dependent_coupling_fixed(AACContext *ac,
357 SingleChannelElement *target,
358 ChannelElement *cce, int index)
360 IndividualChannelStream *ics = &cce->ch[0].ics;
361 const uint16_t *offsets = ics->swb_offset;
362 int *dest = target->coeffs;
363 const int *src = cce->ch[0].coeffs;
364 int g, i, group, k, idx = 0;
365 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
366 av_log(ac->avctx, AV_LOG_ERROR,
367 "Dependent coupling is not supported together with LTP\n");
370 for (g = 0; g < ics->num_window_groups; g++) {
371 for (i = 0; i < ics->max_sfb; i++, idx++) {
372 if (cce->ch[0].band_type[idx] != ZERO_BT) {
373 const int gain = cce->coup.gain[index][idx];
374 int shift, round, c, tmp;
377 c = -cce_scale_fixed[-gain & 7];
378 shift = (-gain-1024) >> 3;
381 c = cce_scale_fixed[gain & 7];
382 shift = (gain-1024) >> 3;
387 } else if (shift < 0) {
389 round = 1 << (shift - 1);
391 for (group = 0; group < ics->group_len[g]; group++) {
392 for (k = offsets[i]; k < offsets[i + 1]; k++) {
393 tmp = (int)(((int64_t)src[group * 128 + k] * c + \
394 (int64_t)0x1000000000) >> 37);
395 dest[group * 128 + k] += (tmp + (int64_t)round) >> shift;
400 for (group = 0; group < ics->group_len[g]; group++) {
401 for (k = offsets[i]; k < offsets[i + 1]; k++) {
402 tmp = (int)(((int64_t)src[group * 128 + k] * c + \
403 (int64_t)0x1000000000) >> 37);
404 dest[group * 128 + k] += tmp * (1U << shift);
410 dest += ics->group_len[g] * 128;
411 src += ics->group_len[g] * 128;
416 * Apply independent channel coupling (applied after IMDCT).
418 * @param index index into coupling gain array
420 static void apply_independent_coupling_fixed(AACContext *ac,
421 SingleChannelElement *target,
422 ChannelElement *cce, int index)
424 int i, c, shift, round, tmp;
425 const int gain = cce->coup.gain[index][0];
426 const int *src = cce->ch[0].ret;
427 unsigned int *dest = target->ret;
428 const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
430 c = cce_scale_fixed[gain & 7];
431 shift = (gain-1024) >> 3;
434 } else if (shift < 0) {
436 round = 1 << (shift - 1);
438 for (i = 0; i < len; i++) {
439 tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
440 dest[i] += (tmp + round) >> shift;
444 for (i = 0; i < len; i++) {
445 tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
446 dest[i] += tmp * (1U << shift);
451 #include "aacdec_template.c"
453 const FFCodec ff_aac_fixed_decoder = {
454 .p.name = "aac_fixed",
455 .p.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
456 .p.type = AVMEDIA_TYPE_AUDIO,
457 .p.id = AV_CODEC_ID_AAC,
458 .priv_data_size = sizeof(AACContext),
459 .init = aac_decode_init,
460 .close = aac_decode_close,
461 FF_CODEC_DECODE_CB(aac_decode_frame),
462 .p.sample_fmts = (const enum AVSampleFormat[]) {
463 AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
465 .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
466 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
467 #if FF_API_OLD_CHANNEL_LAYOUT
468 .p.channel_layouts = aac_channel_layout,
470 .p.ch_layouts = aac_ch_layout,
471 .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),