/*
  * 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 "adts_header.h"

 #include "cbrt_data.h"

 #include "sbr.h"
 #include "aacsbr.h"
 #include "mpeg4audio.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) {
             if (shift > 0) {

                 *coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);

             } else

                 *coef += (unsigned)pv.mant << -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 * (1U << 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 * (1 << 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,
 };