/* * Copyright (c) 2013 * MIPS Technologies, Inc., California. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer in the * documentation and/or other materials provided with the distribution. * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its * contributors may be used to endorse or promote products derived from * this software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE * ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF * SUCH DAMAGE. * * AAC decoder fixed-point implementation * * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * AAC decoder * @author Oded Shimon ( ods15 ods15 dyndns org ) * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) * * Fixed point implementation * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com ) */ #define FFT_FLOAT 0 #define FFT_FIXED_32 1 #define USE_FIXED 1 #include "libavutil/fixed_dsp.h" #include "libavutil/opt.h" #include "avcodec.h" #include "internal.h" #include "get_bits.h" #include "fft.h" #include "lpc.h" #include "kbdwin.h" #include "sinewin.h" #include "aac.h" #include "aactab.h" #include "aacdectab.h" #include "cbrt_data.h" #include "sbr.h" #include "aacsbr.h" #include "mpeg4audio.h" #include "aacadtsdec.h" #include "profiles.h" #include "libavutil/intfloat.h" #include <math.h> #include <string.h> static av_always_inline void reset_predict_state(PredictorState *ps) { ps->r0.mant = 0; ps->r0.exp = 0; ps->r1.mant = 0; ps->r1.exp = 0; ps->cor0.mant = 0; ps->cor0.exp = 0; ps->cor1.mant = 0; ps->cor1.exp = 0; ps->var0.mant = 0x20000000; ps->var0.exp = 1; ps->var1.mant = 0x20000000; ps->var1.exp = 1; } 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 static inline int *DEC_SPAIR(int *dst, unsigned idx) { dst[0] = (idx & 15) - 4; dst[1] = (idx >> 4 & 15) - 4; return dst + 2; } static inline int *DEC_SQUAD(int *dst, unsigned idx) { dst[0] = (idx & 3) - 1; dst[1] = (idx >> 2 & 3) - 1; dst[2] = (idx >> 4 & 3) - 1; dst[3] = (idx >> 6 & 3) - 1; return dst + 4; } static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign) { dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE)); dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2)); return dst + 2; } static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign) { unsigned nz = idx >> 12; dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2)); sign <<= nz & 1; nz >>= 1; dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2)); sign <<= nz & 1; nz >>= 1; dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2)); sign <<= nz & 1; nz >>= 1; dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2)); return dst + 4; } static void vector_pow43(int *coefs, int len) { int i, coef; for (i=0; i<len; i++) { coef = coefs[i]; if (coef < 0) coef = -(int)ff_cbrt_tab_fixed[-coef]; else coef = (int)ff_cbrt_tab_fixed[coef]; coefs[i] = coef; } } static void subband_scale(int *dst, int *src, int scale, int offset, int len) { int ssign = scale < 0 ? -1 : 1; int s = FFABS(scale); unsigned int round; int i, out, c = exp2tab[s & 3]; s = offset - (s >> 2); if (s > 31) { for (i=0; i<len; i++) { dst[i] = 0; } } else if (s > 0) { round = 1 << (s-1); for (i=0; i<len; i++) { out = (int)(((int64_t)src[i] * c) >> 32); dst[i] = ((int)(out+round) >> s) * ssign; } } else if (s > -32) { s = s + 32; round = 1U << (s-1); for (i=0; i<len; i++) { out = (int)((int64_t)((int64_t)src[i] * c + round) >> s); dst[i] = out * (unsigned)ssign; } } else { av_log(NULL, AV_LOG_ERROR, "Overflow in subband_scale()\n"); } } static void noise_scale(int *coefs, int scale, int band_energy, int len) { int ssign = scale < 0 ? -1 : 1; int s = FFABS(scale); unsigned int round; int i, out, c = exp2tab[s & 3]; int nlz = 0; while (band_energy > 0x7fff) { band_energy >>= 1; nlz++; } c /= band_energy; s = 21 + nlz - (s >> 2); if (s > 31) { for (i=0; i<len; i++) { coefs[i] = 0; } } else if (s >= 0) { round = s ? 1 << (s-1) : 0; for (i=0; i<len; i++) { out = (int)(((int64_t)coefs[i] * c) >> 32); coefs[i] = ((int)(out+round) >> s) * ssign; } } else { s = s + 32; round = 1 << (s-1); for (i=0; i<len; i++) { out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s); coefs[i] = out * ssign; } } } static av_always_inline SoftFloat flt16_round(SoftFloat pf) { SoftFloat tmp; int s; tmp.exp = pf.exp; s = pf.mant >> 31; tmp.mant = (pf.mant ^ s) - s; tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U; tmp.mant = (tmp.mant ^ s) - s; return tmp; } static av_always_inline SoftFloat flt16_even(SoftFloat pf) { SoftFloat tmp; int s; tmp.exp = pf.exp; s = pf.mant >> 31; tmp.mant = (pf.mant ^ s) - s; tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U; tmp.mant = (tmp.mant ^ s) - s; return tmp; } static av_always_inline SoftFloat flt16_trunc(SoftFloat pf) { SoftFloat pun; int s; pun.exp = pf.exp; s = pf.mant >> 31; pun.mant = (pf.mant ^ s) - s; pun.mant = pun.mant & 0xFFC00000U; pun.mant = (pun.mant ^ s) - s; return pun; } static av_always_inline void predict(PredictorState *ps, int *coef, int output_enable) { const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64 const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32 SoftFloat e0, e1; SoftFloat pv; SoftFloat k1, k2; SoftFloat r0 = ps->r0, r1 = ps->r1; SoftFloat cor0 = ps->cor0, cor1 = ps->cor1; SoftFloat var0 = ps->var0, var1 = ps->var1; SoftFloat tmp; if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) { k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0))); } else { k1.mant = 0; k1.exp = 0; } if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) { k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1))); } else { k2.mant = 0; k2.exp = 0; } tmp = av_mul_sf(k1, r0); pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1))); if (output_enable) { int shift = 28 - pv.exp; if (shift < 31) *coef += (pv.mant + (1 << (shift - 1))) >> shift; } e0 = av_int2sf(*coef, 2); e1 = av_sub_sf(e0, tmp); ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1))); tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1)); tmp.exp--; ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp)); ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0))); tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0)); tmp.exp--; ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp)); ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0)))); ps->r0 = flt16_trunc(av_mul_sf(a, e0)); } static const int cce_scale_fixed[8] = { Q30(1.0), //2^(0/8) Q30(1.0905077327), //2^(1/8) Q30(1.1892071150), //2^(2/8) Q30(1.2968395547), //2^(3/8) Q30(1.4142135624), //2^(4/8) Q30(1.5422108254), //2^(5/8) Q30(1.6817928305), //2^(6/8) Q30(1.8340080864), //2^(7/8) }; /** * Apply dependent channel coupling (applied before IMDCT). * * @param index index into coupling gain array */ static void apply_dependent_coupling_fixed(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index) { IndividualChannelStream *ics = &cce->ch[0].ics; const uint16_t *offsets = ics->swb_offset; int *dest = target->coeffs; const int *src = cce->ch[0].coeffs; int g, i, group, k, idx = 0; if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) { av_log(ac->avctx, AV_LOG_ERROR, "Dependent coupling is not supported together with LTP\n"); return; } for (g = 0; g < ics->num_window_groups; g++) { for (i = 0; i < ics->max_sfb; i++, idx++) { if (cce->ch[0].band_type[idx] != ZERO_BT) { const int gain = cce->coup.gain[index][idx]; int shift, round, c, tmp; if (gain < 0) { c = -cce_scale_fixed[-gain & 7]; shift = (-gain-1024) >> 3; } else { c = cce_scale_fixed[gain & 7]; shift = (gain-1024) >> 3; } if (shift < -31) { // Nothing to do } else if (shift < 0) { shift = -shift; round = 1 << (shift - 1); for (group = 0; group < ics->group_len[g]; group++) { for (k = offsets[i]; k < offsets[i + 1]; k++) { tmp = (int)(((int64_t)src[group * 128 + k] * c + \ (int64_t)0x1000000000) >> 37); dest[group * 128 + k] += (tmp + round) >> shift; } } } else { for (group = 0; group < ics->group_len[g]; group++) { for (k = offsets[i]; k < offsets[i + 1]; k++) { tmp = (int)(((int64_t)src[group * 128 + k] * c + \ (int64_t)0x1000000000) >> 37); dest[group * 128 + k] += tmp * (1 << shift); } } } } } dest += ics->group_len[g] * 128; src += ics->group_len[g] * 128; } } /** * Apply independent channel coupling (applied after IMDCT). * * @param index index into coupling gain array */ static void apply_independent_coupling_fixed(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index) { int i, c, shift, round, tmp; const int gain = cce->coup.gain[index][0]; const int *src = cce->ch[0].ret; int *dest = target->ret; const int len = 1024 << (ac->oc[1].m4ac.sbr == 1); c = cce_scale_fixed[gain & 7]; shift = (gain-1024) >> 3; if (shift < -31) { return; } else if (shift < 0) { shift = -shift; round = 1 << (shift - 1); for (i = 0; i < len; i++) { tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37); dest[i] += (tmp + round) >> shift; } } else { for (i = 0; i < len; i++) { tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37); dest[i] += tmp << shift; } } } #include "aacdec_template.c" AVCodec ff_aac_fixed_decoder = { .name = "aac_fixed", .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_AAC, .priv_data_size = sizeof(AACContext), .init = aac_decode_init, .close = aac_decode_close, .decode = aac_decode_frame, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE }, .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, .channel_layouts = aac_channel_layout, .profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), .flush = flush, };