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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 "cbrt_data.h"
81 #include "mpeg4audio.h"
82 #include "aacadtsdec.h"
84 #include "libavutil/intfloat.h"
89 static av_always_inline void reset_predict_state(PredictorState *ps)
99 ps->var0.mant = 0x20000000;
101 ps->var1.mant = 0x20000000;
105 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
107 static inline int *DEC_SPAIR(int *dst, unsigned idx)
109 dst[0] = (idx & 15) - 4;
110 dst[1] = (idx >> 4 & 15) - 4;
115 static inline int *DEC_SQUAD(int *dst, unsigned idx)
117 dst[0] = (idx & 3) - 1;
118 dst[1] = (idx >> 2 & 3) - 1;
119 dst[2] = (idx >> 4 & 3) - 1;
120 dst[3] = (idx >> 6 & 3) - 1;
125 static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
127 dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
128 dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
133 static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
135 unsigned nz = idx >> 12;
137 dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
140 dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
143 dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
146 dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
151 static void vector_pow43(int *coefs, int len)
155 for (i=0; i<len; i++) {
158 coef = -(int)ff_cbrt_tab_fixed[-coef];
160 coef = (int)ff_cbrt_tab_fixed[coef];
165 static void subband_scale(int *dst, int *src, int scale, int offset, int len)
167 int ssign = scale < 0 ? -1 : 1;
168 int s = FFABS(scale);
170 int i, out, c = exp2tab[s & 3];
172 s = offset - (s >> 2);
175 for (i=0; i<len; i++) {
180 for (i=0; i<len; i++) {
181 out = (int)(((int64_t)src[i] * c) >> 32);
182 dst[i] = ((int)(out+round) >> s) * ssign;
188 for (i=0; i<len; i++) {
189 out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
190 dst[i] = out * ssign;
195 static void noise_scale(int *coefs, int scale, int band_energy, int len)
197 int ssign = scale < 0 ? -1 : 1;
198 int s = FFABS(scale);
200 int i, out, c = exp2tab[s & 3];
203 while (band_energy > 0x7fff) {
208 s = 21 + nlz - (s >> 2);
211 for (i=0; i<len; i++) {
216 for (i=0; i<len; i++) {
217 out = (int)(((int64_t)coefs[i] * c) >> 32);
218 coefs[i] = ((int)(out+round) >> s) * ssign;
224 for (i=0; i<len; i++) {
225 out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
226 coefs[i] = out * ssign;
231 static av_always_inline SoftFloat flt16_round(SoftFloat pf)
238 tmp.mant = (pf.mant ^ s) - s;
239 tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
240 tmp.mant = (tmp.mant ^ s) - s;
245 static av_always_inline SoftFloat flt16_even(SoftFloat pf)
252 tmp.mant = (pf.mant ^ s) - s;
253 tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
254 tmp.mant = (tmp.mant ^ s) - s;
259 static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
266 pun.mant = (pf.mant ^ s) - s;
267 pun.mant = pun.mant & 0xFFC00000U;
268 pun.mant = (pun.mant ^ s) - s;
273 static av_always_inline void predict(PredictorState *ps, int *coef,
276 const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
277 const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
281 SoftFloat r0 = ps->r0, r1 = ps->r1;
282 SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
283 SoftFloat var0 = ps->var0, var1 = ps->var1;
286 if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
287 k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
294 if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
295 k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
302 tmp = av_mul_sf(k1, r0);
303 pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
305 int shift = 28 - pv.exp;
308 *coef += (pv.mant + (1 << (shift - 1))) >> shift;
311 e0 = av_int2sf(*coef, 2);
312 e1 = av_sub_sf(e0, tmp);
314 ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
315 tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
317 ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
318 ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
319 tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
321 ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
323 ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
324 ps->r0 = flt16_trunc(av_mul_sf(a, e0));
328 static const int cce_scale_fixed[8] = {
330 Q30(1.0905077327), //2^(1/8)
331 Q30(1.1892071150), //2^(2/8)
332 Q30(1.2968395547), //2^(3/8)
333 Q30(1.4142135624), //2^(4/8)
334 Q30(1.5422108254), //2^(5/8)
335 Q30(1.6817928305), //2^(6/8)
336 Q30(1.8340080864), //2^(7/8)
340 * Apply dependent channel coupling (applied before IMDCT).
342 * @param index index into coupling gain array
344 static void apply_dependent_coupling_fixed(AACContext *ac,
345 SingleChannelElement *target,
346 ChannelElement *cce, int index)
348 IndividualChannelStream *ics = &cce->ch[0].ics;
349 const uint16_t *offsets = ics->swb_offset;
350 int *dest = target->coeffs;
351 const int *src = cce->ch[0].coeffs;
352 int g, i, group, k, idx = 0;
353 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
354 av_log(ac->avctx, AV_LOG_ERROR,
355 "Dependent coupling is not supported together with LTP\n");
358 for (g = 0; g < ics->num_window_groups; g++) {
359 for (i = 0; i < ics->max_sfb; i++, idx++) {
360 if (cce->ch[0].band_type[idx] != ZERO_BT) {
361 const int gain = cce->coup.gain[index][idx];
362 int shift, round, c, tmp;
365 c = -cce_scale_fixed[-gain & 7];
366 shift = (-gain-1024) >> 3;
369 c = cce_scale_fixed[gain & 7];
370 shift = (gain-1024) >> 3;
375 round = 1 << (shift - 1);
377 for (group = 0; group < ics->group_len[g]; group++) {
378 for (k = offsets[i]; k < offsets[i + 1]; k++) {
379 tmp = (int)(((int64_t)src[group * 128 + k] * c + \
380 (int64_t)0x1000000000) >> 37);
381 dest[group * 128 + k] += (tmp + round) >> shift;
386 for (group = 0; group < ics->group_len[g]; group++) {
387 for (k = offsets[i]; k < offsets[i + 1]; k++) {
388 tmp = (int)(((int64_t)src[group * 128 + k] * c + \
389 (int64_t)0x1000000000) >> 37);
390 dest[group * 128 + k] += tmp << shift;
396 dest += ics->group_len[g] * 128;
397 src += ics->group_len[g] * 128;
402 * Apply independent channel coupling (applied after IMDCT).
404 * @param index index into coupling gain array
406 static void apply_independent_coupling_fixed(AACContext *ac,
407 SingleChannelElement *target,
408 ChannelElement *cce, int index)
410 int i, c, shift, round, tmp;
411 const int gain = cce->coup.gain[index][0];
412 const int *src = cce->ch[0].ret;
413 int *dest = target->ret;
414 const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
416 c = cce_scale_fixed[gain & 7];
417 shift = (gain-1024) >> 3;
420 round = 1 << (shift - 1);
422 for (i = 0; i < len; i++) {
423 tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
424 dest[i] += (tmp + round) >> shift;
428 for (i = 0; i < len; i++) {
429 tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
430 dest[i] += tmp << shift;
435 #include "aacdec_template.c"
437 AVCodec ff_aac_fixed_decoder = {
439 .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
440 .type = AVMEDIA_TYPE_AUDIO,
441 .id = AV_CODEC_ID_AAC,
442 .priv_data_size = sizeof(AACContext),
443 .init = aac_decode_init,
444 .close = aac_decode_close,
445 .decode = aac_decode_frame,
446 .sample_fmts = (const enum AVSampleFormat[]) {
447 AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
449 .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
450 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
451 .channel_layouts = aac_channel_layout,
452 .profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),