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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) << 1));
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) << 1));
140 dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) << 1));
143 dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) << 1));
146 dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) << 1));
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);
176 for (i=0; i<len; i++) {
177 out = (int)(((int64_t)src[i] * c) >> 32);
178 dst[i] = ((int)(out+round) >> s) * ssign;
184 for (i=0; i<len; i++) {
185 out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
186 dst[i] = out * ssign;
191 static void noise_scale(int *coefs, int scale, int band_energy, int len)
193 int ssign = scale < 0 ? -1 : 1;
194 int s = FFABS(scale);
196 int i, out, c = exp2tab[s & 3];
199 while (band_energy > 0x7fff) {
204 s = 21 + nlz - (s >> 2);
208 for (i=0; i<len; i++) {
209 out = (int)(((int64_t)coefs[i] * c) >> 32);
210 coefs[i] = ((int)(out+round) >> s) * ssign;
216 for (i=0; i<len; i++) {
217 out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
218 coefs[i] = out * ssign;
223 static av_always_inline SoftFloat flt16_round(SoftFloat pf)
230 tmp.mant = (pf.mant ^ s) - s;
231 tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
232 tmp.mant = (tmp.mant ^ s) - s;
237 static av_always_inline SoftFloat flt16_even(SoftFloat pf)
244 tmp.mant = (pf.mant ^ s) - s;
245 tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
246 tmp.mant = (tmp.mant ^ s) - s;
251 static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
258 pun.mant = (pf.mant ^ s) - s;
259 pun.mant = pun.mant & 0xFFC00000U;
260 pun.mant = (pun.mant ^ s) - s;
265 static av_always_inline void predict(PredictorState *ps, int *coef,
268 const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
269 const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
273 SoftFloat r0 = ps->r0, r1 = ps->r1;
274 SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
275 SoftFloat var0 = ps->var0, var1 = ps->var1;
278 if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
279 k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
286 if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
287 k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
294 tmp = av_mul_sf(k1, r0);
295 pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
297 int shift = 28 - pv.exp;
300 *coef += (pv.mant + (1 << (shift - 1))) >> shift;
303 e0 = av_int2sf(*coef, 2);
304 e1 = av_sub_sf(e0, tmp);
306 ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
307 tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
309 ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
310 ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
311 tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
313 ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
315 ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
316 ps->r0 = flt16_trunc(av_mul_sf(a, e0));
320 static const int cce_scale_fixed[8] = {
322 Q30(1.0905077327), //2^(1/8)
323 Q30(1.1892071150), //2^(2/8)
324 Q30(1.2968395547), //2^(3/8)
325 Q30(1.4142135624), //2^(4/8)
326 Q30(1.5422108254), //2^(5/8)
327 Q30(1.6817928305), //2^(6/8)
328 Q30(1.8340080864), //2^(7/8)
332 * Apply dependent channel coupling (applied before IMDCT).
334 * @param index index into coupling gain array
336 static void apply_dependent_coupling_fixed(AACContext *ac,
337 SingleChannelElement *target,
338 ChannelElement *cce, int index)
340 IndividualChannelStream *ics = &cce->ch[0].ics;
341 const uint16_t *offsets = ics->swb_offset;
342 int *dest = target->coeffs;
343 const int *src = cce->ch[0].coeffs;
344 int g, i, group, k, idx = 0;
345 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
346 av_log(ac->avctx, AV_LOG_ERROR,
347 "Dependent coupling is not supported together with LTP\n");
350 for (g = 0; g < ics->num_window_groups; g++) {
351 for (i = 0; i < ics->max_sfb; i++, idx++) {
352 if (cce->ch[0].band_type[idx] != ZERO_BT) {
353 const int gain = cce->coup.gain[index][idx];
354 int shift, round, c, tmp;
357 c = -cce_scale_fixed[-gain & 7];
358 shift = (-gain-1024) >> 3;
361 c = cce_scale_fixed[gain & 7];
362 shift = (gain-1024) >> 3;
367 round = 1 << (shift - 1);
369 for (group = 0; group < ics->group_len[g]; group++) {
370 for (k = offsets[i]; k < offsets[i + 1]; k++) {
371 tmp = (int)(((int64_t)src[group * 128 + k] * c + \
372 (int64_t)0x1000000000) >> 37);
373 dest[group * 128 + k] += (tmp + round) >> shift;
378 for (group = 0; group < ics->group_len[g]; group++) {
379 for (k = offsets[i]; k < offsets[i + 1]; k++) {
380 tmp = (int)(((int64_t)src[group * 128 + k] * c + \
381 (int64_t)0x1000000000) >> 37);
382 dest[group * 128 + k] += tmp << shift;
388 dest += ics->group_len[g] * 128;
389 src += ics->group_len[g] * 128;
394 * Apply independent channel coupling (applied after IMDCT).
396 * @param index index into coupling gain array
398 static void apply_independent_coupling_fixed(AACContext *ac,
399 SingleChannelElement *target,
400 ChannelElement *cce, int index)
402 int i, c, shift, round, tmp;
403 const int gain = cce->coup.gain[index][0];
404 const int *src = cce->ch[0].ret;
405 int *dest = target->ret;
406 const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
408 c = cce_scale_fixed[gain & 7];
409 shift = (gain-1024) >> 3;
412 round = 1 << (shift - 1);
414 for (i = 0; i < len; i++) {
415 tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
416 dest[i] += (tmp + round) >> shift;
420 for (i = 0; i < len; i++) {
421 tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
422 dest[i] += tmp << shift;
427 #include "aacdec_template.c"
429 AVCodec ff_aac_fixed_decoder = {
431 .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
432 .type = AVMEDIA_TYPE_AUDIO,
433 .id = AV_CODEC_ID_AAC,
434 .priv_data_size = sizeof(AACContext),
435 .init = aac_decode_init,
436 .close = aac_decode_close,
437 .decode = aac_decode_frame,
438 .sample_fmts = (const enum AVSampleFormat[]) {
439 AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
441 .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
442 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
443 .channel_layouts = aac_channel_layout,
444 .profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),