libavcodec/aacdec_fixed.c - nest-android-app/ffmpeg - Git at Google

 /*
  * 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_tablegen.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) << 1));

     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) << 1));
     sign <<= nz & 1;
     nz >>= 1;
     dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) << 1));
     sign <<= nz & 1;
     nz >>= 1;
     dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) << 1));
     sign <<= nz & 1;
     nz >>= 1;
     dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) << 1));

     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)cbrt_tab[-coef];
         else
             coef = (int)cbrt_tab[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 > 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 {
         s = s + 32;
         round = 1 << (s-1);
         for (i=0; i<len; i++) {
             out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
             dst[i] = out * ssign;
         }
     }
 }

 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 > 0) {
         round = 1 << (s-1);
         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 < 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 << 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 < 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,
 };
	/*
	* 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_tablegen.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) << 1));

	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) << 1));
	sign <<= nz & 1;
	nz >>= 1;
	dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) << 1));
	sign <<= nz & 1;
	nz >>= 1;
	dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) << 1));
	sign <<= nz & 1;
	nz >>= 1;
	dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) << 1));

	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)cbrt_tab[-coef];
	else
	coef = (int)cbrt_tab[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 > 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 {
	s = s + 32;
	round = 1 << (s-1);
	for (i=0; i<len; i++) {
	out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
	dst[i] = out * ssign;
	}
	}
	}

	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 > 0) {
	round = 1 << (s-1);
	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 < 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 << 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 < 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,
	};