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17 * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
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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 )
62 #define FFT_FIXED_32 1
65 #include "libavutil/fixed_dsp.h"
66 #include "libavutil/opt.h"
77 #include "aacdectab.h"
78 #include "adts_header.h"
79 #include "cbrt_data.h"
82 #include "mpeg4audio.h"
84 #include "libavutil/intfloat.h"
89 DECLARE_ALIGNED(32, static int, AAC_KBD_RENAME(kbd_long_1024))[1024];
90 DECLARE_ALIGNED(32, static int, AAC_KBD_RENAME(kbd_short_128))[128];
92 static av_always_inline void reset_predict_state(PredictorState *ps)
102 ps->var0.mant = 0x20000000;
104 ps->var1.mant = 0x20000000;
108 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
110 static inline int *DEC_SPAIR(int *dst, unsigned idx)
112 dst[0] = (idx & 15) - 4;
113 dst[1] = (idx >> 4 & 15) - 4;
118 static inline int *DEC_SQUAD(int *dst, unsigned idx)
120 dst[0] = (idx & 3) - 1;
121 dst[1] = (idx >> 2 & 3) - 1;
122 dst[2] = (idx >> 4 & 3) - 1;
123 dst[3] = (idx >> 6 & 3) - 1;
128 static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
130 dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
131 dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
136 static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
138 unsigned nz = idx >> 12;
140 dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
143 dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
146 dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
149 dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
154 static void vector_pow43(int *coefs, int len)
158 for (i=0; i<len; i++) {
161 coef = -(int)ff_cbrt_tab_fixed[(-coef) & 8191];
163 coef = (int)ff_cbrt_tab_fixed[ coef & 8191];
168 static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
170 int ssign = scale < 0 ? -1 : 1;
171 int s = FFABS(scale);
173 int i, out, c = exp2tab[s & 3];
175 s = offset - (s >> 2);
178 for (i=0; i<len; i++) {
183 for (i=0; i<len; i++) {
184 out = (int)(((int64_t)src[i] * c) >> 32);
185 dst[i] = ((int)(out+round) >> s) * ssign;
187 } else if (s > -32) {
190 for (i=0; i<len; i++) {
191 out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
192 dst[i] = out * (unsigned)ssign;
195 av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
199 static void noise_scale(int *coefs, int scale, int band_energy, int len)
203 int i, out, c = exp2tab[s & 3];
207 while (band_energy > 0x7fff) {
212 s = 21 + nlz - (s >> 2);
215 for (i=0; i<len; i++) {
219 round = s ? 1 << (s-1) : 0;
220 for (i=0; i<len; i++) {
221 out = (int)(((int64_t)coefs[i] * c) >> 32);
222 coefs[i] = -((int)(out+round) >> s);
229 for (i=0; i<len; i++) {
230 out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
234 for (i=0; i<len; i++)
235 coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
240 static av_always_inline SoftFloat flt16_round(SoftFloat pf)
247 tmp.mant = (pf.mant ^ s) - s;
248 tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
249 tmp.mant = (tmp.mant ^ s) - s;
254 static av_always_inline SoftFloat flt16_even(SoftFloat pf)
261 tmp.mant = (pf.mant ^ s) - s;
262 tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
263 tmp.mant = (tmp.mant ^ s) - s;
268 static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
275 pun.mant = (pf.mant ^ s) - s;
276 pun.mant = pun.mant & 0xFFC00000U;
277 pun.mant = (pun.mant ^ s) - s;
282 static av_always_inline void predict(PredictorState *ps, int *coef,
285 const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
286 const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
290 SoftFloat r0 = ps->r0, r1 = ps->r1;
291 SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
292 SoftFloat var0 = ps->var0, var1 = ps->var1;
295 if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
296 k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
303 if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
304 k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
311 tmp = av_mul_sf(k1, r0);
312 pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
314 int shift = 28 - pv.exp;
318 *coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
320 *coef += (unsigned)pv.mant << -shift;
324 e0 = av_int2sf(*coef, 2);
325 e1 = av_sub_sf(e0, tmp);
327 ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
328 tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
330 ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
331 ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
332 tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
334 ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
336 ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
337 ps->r0 = flt16_trunc(av_mul_sf(a, e0));
341 static const int cce_scale_fixed[8] = {
343 Q30(1.0905077327), //2^(1/8)
344 Q30(1.1892071150), //2^(2/8)
345 Q30(1.2968395547), //2^(3/8)
346 Q30(1.4142135624), //2^(4/8)
347 Q30(1.5422108254), //2^(5/8)
348 Q30(1.6817928305), //2^(6/8)
349 Q30(1.8340080864), //2^(7/8)
353 * Apply dependent channel coupling (applied before IMDCT).
355 * @param index index into coupling gain array
357 static void apply_dependent_coupling_fixed(AACContext *ac,
358 SingleChannelElement *target,
359 ChannelElement *cce, int index)
361 IndividualChannelStream *ics = &cce->ch[0].ics;
362 const uint16_t *offsets = ics->swb_offset;
363 int *dest = target->coeffs;
364 const int *src = cce->ch[0].coeffs;
365 int g, i, group, k, idx = 0;
366 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
367 av_log(ac->avctx, AV_LOG_ERROR,
368 "Dependent coupling is not supported together with LTP\n");
371 for (g = 0; g < ics->num_window_groups; g++) {
372 for (i = 0; i < ics->max_sfb; i++, idx++) {
373 if (cce->ch[0].band_type[idx] != ZERO_BT) {
374 const int gain = cce->coup.gain[index][idx];
375 int shift, round, c, tmp;
378 c = -cce_scale_fixed[-gain & 7];
379 shift = (-gain-1024) >> 3;
382 c = cce_scale_fixed[gain & 7];
383 shift = (gain-1024) >> 3;
388 } else if (shift < 0) {
390 round = 1 << (shift - 1);
392 for (group = 0; group < ics->group_len[g]; group++) {
393 for (k = offsets[i]; k < offsets[i + 1]; k++) {
394 tmp = (int)(((int64_t)src[group * 128 + k] * c + \
395 (int64_t)0x1000000000) >> 37);
396 dest[group * 128 + k] += (tmp + (int64_t)round) >> shift;
401 for (group = 0; group < ics->group_len[g]; group++) {
402 for (k = offsets[i]; k < offsets[i + 1]; k++) {
403 tmp = (int)(((int64_t)src[group * 128 + k] * c + \
404 (int64_t)0x1000000000) >> 37);
405 dest[group * 128 + k] += tmp * (1U << shift);
411 dest += ics->group_len[g] * 128;
412 src += ics->group_len[g] * 128;
417 * Apply independent channel coupling (applied after IMDCT).
419 * @param index index into coupling gain array
421 static void apply_independent_coupling_fixed(AACContext *ac,
422 SingleChannelElement *target,
423 ChannelElement *cce, int index)
425 int i, c, shift, round, tmp;
426 const int gain = cce->coup.gain[index][0];
427 const int *src = cce->ch[0].ret;
428 unsigned int *dest = target->ret;
429 const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
431 c = cce_scale_fixed[gain & 7];
432 shift = (gain-1024) >> 3;
435 } else if (shift < 0) {
437 round = 1 << (shift - 1);
439 for (i = 0; i < len; i++) {
440 tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
441 dest[i] += (tmp + round) >> shift;
445 for (i = 0; i < len; i++) {
446 tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
447 dest[i] += tmp * (1U << shift);
452 #include "aacdec_template.c"
454 AVCodec ff_aac_fixed_decoder = {
456 .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
457 .type = AVMEDIA_TYPE_AUDIO,
458 .id = AV_CODEC_ID_AAC,
459 .priv_data_size = sizeof(AACContext),
460 .init = aac_decode_init,
461 .close = aac_decode_close,
462 .decode = aac_decode_frame,
463 .sample_fmts = (const enum AVSampleFormat[]) {
464 AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
466 .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
467 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
468 .channel_layouts = aac_channel_layout,
469 .profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),