[FFmpeg-devel] [PATCH 03/14] libavcodec/libavutil: Implementation of AAC_fixed_decoder (LC-module) [3/4]
Nedeljko Babic
nedeljko.babic at imgtec.com
Wed Apr 15 18:29:18 CEST 2015
From: Djordje Pesut <djordje.pesut at imgtec.com>
Add fixed point implementation
Signed-off-by: Nedeljko Babic <nedeljko.babic at imgtec.com>
---
libavcodec/aac.h | 101 ++++++++--
libavcodec/aacdec.c | 5 +
libavcodec/aacdec_fixed.c | 448 +++++++++++++++++++++++++++++++++++++++++++
libavcodec/aacdec_template.c | 433 ++++++++++++++++++++++++++++-------------
libavcodec/lpc.h | 15 +-
libavcodec/mdct_template.c | 5 +
libavutil/fixed_dsp.c | 70 ++++++-
libavutil/fixed_dsp.h | 53 +++++
8 files changed, 967 insertions(+), 163 deletions(-)
create mode 100644 libavcodec/aacdec_fixed.c
diff --git a/libavcodec/aac.h b/libavcodec/aac.h
index 23ec085..571f8b9 100644
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -30,9 +30,62 @@
#ifndef AVCODEC_AAC_H
#define AVCODEC_AAC_H
+#ifndef USE_FIXED
+#define USE_FIXED 0
+#endif
+
+#if USE_FIXED
+
+#define FFT_FLOAT 0
+#define FFT_FIXED_32 1
+
+#define AAC_RENAME(x) x ## _fixed
+#define AAC_RENAME_32(x) x ## _fixed_32
+#define INTFLOAT int
+#define SHORTFLOAT int16_t
+#define AAC_FLOAT aac_float_t
+#define AAC_SIGNE int
+#define FIXR(a) ((int)((a) * 1 + 0.5))
+#define FIXR10(a) ((int)((a) * 1024.0 + 0.5))
+#define Q23(a) (int)((a) * 8388608.0 + 0.5)
+#define Q30(x) (int)((x)*1073741824.0 + 0.5)
+#define Q31(x) (int)((x)*2147483648.0 + 0.5)
+#define RANGE15(x) x
+#define GET_GAIN(x, y) (-(y) << (x)) + 1024
+#define AAC_MUL26(x, y) (int)(((int64_t)(x) * (y) + 0x2000000) >> 26)
+#define AAC_MUL30(x, y) (int)(((int64_t)(x) * (y) + 0x20000000) >> 30)
+#define AAC_MUL31(x, y) (int)(((int64_t)(x) * (y) + 0x40000000) >> 31)
+
+#else
+
+#define FFT_FLOAT 1
+#define FFT_FIXED_32 0
+
+#define AAC_RENAME(x) x
+#define AAC_RENAME_32(x) x
+#define INTFLOAT float
+#define SHORTFLOAT float
+#define AAC_FLOAT float
+#define AAC_SIGNE unsigned
+#define FIXR(x) ((float)(x))
+#define FIXR10(x) ((float)(x))
+#define Q23(x) x
+#define Q30(x) x
+#define Q31(x) x
+#define RANGE15(x) (32768.0 * (x))
+#define GET_GAIN(x, y) powf((x), -(y))
+#define AAC_MUL26(x, y) ((x) * (y))
+#define AAC_MUL30(x, y) ((x) * (y))
+#define AAC_MUL31(x, y) ((x) * (y))
+
+#endif /* USE_FIXED */
+
#include "libavutil/float_dsp.h"
+#include "libavutil/fixed_dsp.h"
#include "avcodec.h"
+#if !USE_FIXED
#include "imdct15.h"
+#endif
#include "fft.h"
#include "mpeg4audio.h"
#include "sbr.h"
@@ -125,12 +178,12 @@ typedef struct OutputConfiguration {
* Predictor State
*/
typedef struct PredictorState {
- float cor0;
- float cor1;
- float var0;
- float var1;
- float r0;
- float r1;
+ AAC_FLOAT cor0;
+ AAC_FLOAT cor1;
+ AAC_FLOAT var0;
+ AAC_FLOAT var1;
+ AAC_FLOAT r0;
+ AAC_FLOAT r1;
} PredictorState;
#define MAX_PREDICTORS 672
@@ -151,7 +204,7 @@ typedef struct PredictorState {
typedef struct LongTermPrediction {
int8_t present;
int16_t lag;
- float coef;
+ INTFLOAT coef;
int8_t used[MAX_LTP_LONG_SFB];
} LongTermPrediction;
@@ -185,7 +238,7 @@ typedef struct TemporalNoiseShaping {
int length[8][4];
int direction[8][4];
int order[8][4];
- float coef[8][4][TNS_MAX_ORDER];
+ INTFLOAT coef[8][4][TNS_MAX_ORDER];
} TemporalNoiseShaping;
/**
@@ -222,7 +275,7 @@ typedef struct ChannelCoupling {
int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
* [2] list of gains for left channel; [3] lists of gains for both channels
*/
- float gain[16][120];
+ INTFLOAT gain[16][120];
} ChannelCoupling;
/**
@@ -234,16 +287,16 @@ typedef struct SingleChannelElement {
Pulse pulse;
enum BandType band_type[128]; ///< band types
int band_type_run_end[120]; ///< band type run end points
- float sf[120]; ///< scalefactors
+ INTFLOAT sf[120]; ///< scalefactors
int sf_idx[128]; ///< scalefactor indices (used by encoder)
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
- DECLARE_ALIGNED(32, float, pcoeffs)[1024]; ///< coefficients for IMDCT, pristine
- DECLARE_ALIGNED(32, float, coeffs)[1024]; ///< coefficients for IMDCT, maybe processed
- DECLARE_ALIGNED(32, float, saved)[1536]; ///< overlap
- DECLARE_ALIGNED(32, float, ret_buf)[2048]; ///< PCM output buffer
- DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP
+ DECLARE_ALIGNED(32, INTFLOAT, pcoeffs)[1024]; ///< coefficients for IMDCT, pristine
+ DECLARE_ALIGNED(32, INTFLOAT, coeffs)[1024]; ///< coefficients for IMDCT, maybe processed
+ DECLARE_ALIGNED(32, INTFLOAT, saved)[1536]; ///< overlap
+ DECLARE_ALIGNED(32, INTFLOAT, ret_buf)[2048]; ///< PCM output buffer
+ DECLARE_ALIGNED(16, INTFLOAT, ltp_state)[3072];///< time signal for LTP
PredictorState predictor_state[MAX_PREDICTORS];
- float *ret; ///< PCM output
+ INTFLOAT *ret; ///< PCM output
} SingleChannelElement;
/**
@@ -288,7 +341,7 @@ struct AACContext {
* (We do not want to have these on the stack.)
* @{
*/
- DECLARE_ALIGNED(32, float, buf_mdct)[1024];
+ DECLARE_ALIGNED(32, INTFLOAT, buf_mdct)[1024];
/** @} */
/**
@@ -299,8 +352,12 @@ struct AACContext {
FFTContext mdct_small;
FFTContext mdct_ld;
FFTContext mdct_ltp;
+#if USE_FIXED
+ AVFixedDSPContext *fdsp;
+#else
IMDCT15Context *mdct480;
AVFloatDSPContext *fdsp;
+#endif /* USE_FIXED */
int random_state;
/** @} */
@@ -320,7 +377,7 @@ struct AACContext {
int dmono_mode; ///< 0->not dmono, 1->use first channel, 2->use second channel
/** @} */
- DECLARE_ALIGNED(32, float, temp)[128];
+ DECLARE_ALIGNED(32, INTFLOAT, temp)[128];
OutputConfiguration oc[2];
int warned_num_aac_frames;
@@ -328,11 +385,13 @@ struct AACContext {
/* aacdec functions pointers */
void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
- void (*apply_tns)(float coef[1024], TemporalNoiseShaping *tns,
+ void (*apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
IndividualChannelStream *ics, int decode);
- void (*windowing_and_mdct_ltp)(AACContext *ac, float *out,
- float *in, IndividualChannelStream *ics);
+ void (*windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out,
+ INTFLOAT *in, IndividualChannelStream *ics);
void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
+ void (*vector_pow43)(int *coefs, int len);
+ void (*subband_scale)(int *dst, int *src, int scale, int offset, int len);
};
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 1d1abc9..5a9c57c 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -32,6 +32,11 @@
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
+#define FFT_FLOAT 1
+#define FFT_FIXED_32 0
+#define USE_FIXED 0
+#define CONFIG_FIXED 0
+
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avcodec.h"
diff --git a/libavcodec/aacdec_fixed.c b/libavcodec/aacdec_fixed.c
new file mode 100644
index 0000000..970f595
--- /dev/null
+++ b/libavcodec/aacdec_fixed.c
@@ -0,0 +1,448 @@
+/*
+ * 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
+#define CONFIG_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 "libavutil/intfloat.h"
+
+#include <assert.h>
+#include <errno.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;
+}
+
+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)) {
+ tmp = av_recip_sf(var0);
+ k1 = av_mul_sf(cor0, flt16_even(av_mul_sf(a, tmp)));
+ }
+ else {
+ k1.mant = 0;
+ k1.exp = 0;
+ }
+
+ if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
+ tmp = av_recip_sf(var1);
+ k2 = av_mul_sf(cor1, flt16_even(av_mul_sf(a, tmp)));
+ }
+ 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, 28);
+ 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 = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
+ .channel_layouts = aac_channel_layout,
+ .flush = flush,
+};
diff --git a/libavcodec/aacdec_template.c b/libavcodec/aacdec_template.c
index 1c02f77..45ce5e1 100644
--- a/libavcodec/aacdec_template.c
+++ b/libavcodec/aacdec_template.c
@@ -8,6 +8,10 @@
* Copyright (c) 2008-2010 Paul Kendall <paul at kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-libav at jannau.net>
*
+ * AAC decoder fixed-point implementation
+ * Copyright (c) 2013
+ * MIPS Technologies, Inc., California.
+ *
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
@@ -30,6 +34,10 @@
* AAC decoder
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * AAC decoder fixed-point implementation
+ * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
+ * @author Nedeljko Babic ( nedeljko.babic imgtec com )
*/
/*
@@ -124,7 +132,7 @@ static av_cold int che_configure(AACContext *ac,
if (!ac->che[type][id]) {
if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
return AVERROR(ENOMEM);
- ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
+ AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr);
}
if (type != TYPE_CCE) {
if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
@@ -139,7 +147,7 @@ static av_cold int che_configure(AACContext *ac,
}
} else {
if (ac->che[type][id])
- ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
+ AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][id]->sbr);
av_freep(&ac->che[type][id]);
}
return 0;
@@ -173,7 +181,7 @@ static int frame_configure_elements(AVCodecContext *avctx)
/* map output channel pointers to AVFrame data */
for (ch = 0; ch < avctx->channels; ch++) {
if (ac->output_element[ch])
- ac->output_element[ch]->ret = (float *)ac->frame->extended_data[ch];
+ ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
}
return 0;
@@ -845,8 +853,14 @@ static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
m4ac->ps = 0;
m4ac->sbr = 0;
-
+#if USE_FIXED
+ if (get_bits1(gb)) { // frameLengthFlag
+ avpriv_request_sample(avctx, "960/120 MDCT window");
+ return AVERROR_PATCHWELCOME;
+ }
+#else
m4ac->frame_length_short = get_bits1(gb);
+#endif
res_flags = get_bits(gb, 3);
if (res_flags) {
avpriv_report_missing_feature(avctx,
@@ -1032,8 +1046,11 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
aacdec_init(ac);
-
+#if USE_FIXED
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+#else
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+#endif /* USE_FIXED */
if (avctx->extradata_size > 0) {
if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
@@ -1088,9 +1105,13 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
AAC_INIT_VLC_STATIC( 9, 366);
AAC_INIT_VLC_STATIC(10, 462);
- ff_aac_sbr_init();
+ AAC_RENAME(ff_aac_sbr_init)();
+#if USE_FIXED
+ ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & CODEC_FLAG_BITEXACT);
+#else
ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
+#endif /* USE_FIXED */
if (!ac->fdsp) {
return AVERROR(ENOMEM);
}
@@ -1109,22 +1130,23 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
sizeof(ff_aac_scalefactor_code[0]),
352);
- ff_mdct_init(&ac->mdct, 11, 1, 1.0 / (32768.0 * 1024.0));
- ff_mdct_init(&ac->mdct_ld, 10, 1, 1.0 / (32768.0 * 512.0));
- ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0));
- ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0);
+ AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
+ AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
+ AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
+ AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
+#if !USE_FIXED
ret = ff_imdct15_init(&ac->mdct480, 5);
if (ret < 0)
return ret;
-
+#endif
// window initialization
- ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
- ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
- ff_init_ff_sine_windows(10);
- ff_init_ff_sine_windows( 9);
- ff_init_ff_sine_windows( 7);
+ AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
+ AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);
+ AAC_RENAME(ff_init_ff_sine_windows)(10);
+ AAC_RENAME(ff_init_ff_sine_windows)( 9);
+ AAC_RENAME(ff_init_ff_sine_windows)( 7);
- cbrt_tableinit();
+ AAC_RENAME(cbrt_tableinit)();
return 0;
}
@@ -1345,7 +1367,7 @@ static int decode_band_types(AACContext *ac, enum BandType band_type[120],
*
* @return Returns error status. 0 - OK, !0 - error
*/
-static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
+static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb,
unsigned int global_gain,
IndividualChannelStream *ics,
enum BandType band_type[120],
@@ -1360,7 +1382,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
int run_end = band_type_run_end[idx];
if (band_type[idx] == ZERO_BT) {
for (; i < run_end; i++, idx++)
- sf[idx] = 0.0;
+ sf[idx] = FIXR(0.);
} else if ((band_type[idx] == INTENSITY_BT) ||
(band_type[idx] == INTENSITY_BT2)) {
for (; i < run_end; i++, idx++) {
@@ -1372,7 +1394,11 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
"Clipped intensity stereo position (%d -> %d)",
offset[2], clipped_offset);
}
+#if USE_FIXED
+ sf[idx] = 100 - clipped_offset;
+#else
sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
+#endif /* USE_FIXED */
}
} else if (band_type[idx] == NOISE_BT) {
for (; i < run_end; i++, idx++) {
@@ -1387,7 +1413,11 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
"Clipped noise gain (%d -> %d)",
offset[1], clipped_offset);
}
+#if USE_FIXED
+ sf[idx] = -(100 + clipped_offset);
+#else
sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
+#endif /* USE_FIXED */
}
} else {
for (; i < run_end; i++, idx++) {
@@ -1397,7 +1427,11 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
"Scalefactor (%d) out of range.\n", offset[0]);
return AVERROR_INVALIDDATA;
}
+#if USE_FIXED
+ sf[idx] = -offset[0];
+#else
sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
+#endif /* USE_FIXED */
}
}
}
@@ -1503,8 +1537,8 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
*
* @return Returns error status. 0 - OK, !0 - error
*/
-static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
- GetBitContext *gb, const float sf[120],
+static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
+ GetBitContext *gb, const INTFLOAT sf[120],
int pulse_present, const Pulse *pulse,
const IndividualChannelStream *ics,
enum BandType band_type[120])
@@ -1512,49 +1546,63 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
int i, k, g, idx = 0;
const int c = 1024 / ics->num_windows;
const uint16_t *offsets = ics->swb_offset;
- float *coef_base = coef;
+ INTFLOAT *coef_base = coef;
for (g = 0; g < ics->num_windows; g++)
memset(coef + g * 128 + offsets[ics->max_sfb], 0,
- sizeof(float) * (c - offsets[ics->max_sfb]));
+ sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
for (g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (i = 0; i < ics->max_sfb; i++, idx++) {
const unsigned cbt_m1 = band_type[idx] - 1;
- float *cfo = coef + offsets[i];
+ INTFLOAT *cfo = coef + offsets[i];
int off_len = offsets[i + 1] - offsets[i];
int group;
if (cbt_m1 >= INTENSITY_BT2 - 1) {
- for (group = 0; group < g_len; group++, cfo+=128) {
- memset(cfo, 0, off_len * sizeof(float));
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ memset(cfo, 0, off_len * sizeof(INTFLOAT));
}
} else if (cbt_m1 == NOISE_BT - 1) {
- for (group = 0; group < g_len; group++, cfo+=128) {
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+#if !USE_FIXED
float scale;
- float band_energy;
+#endif /* !USE_FIXED */
+ INTFLOAT band_energy;
for (k = 0; k < off_len; k++) {
ac->random_state = lcg_random(ac->random_state);
+#if USE_FIXED
+ cfo[k] = ac->random_state >> 3;
+#else
cfo[k] = ac->random_state;
+#endif /* USE_FIXED */
}
+#if USE_FIXED
+ band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
+ band_energy = fixed_sqrt(band_energy, 31);
+ noise_scale(cfo, sf[idx], band_energy, off_len);
+#else
band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
scale = sf[idx] / sqrtf(band_energy);
ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
+#endif /* USE_FIXED */
}
} else {
+#if !USE_FIXED
const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
+#endif /* !USE_FIXED */
const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
OPEN_READER(re, gb);
switch (cbt_m1 >> 1) {
case 0:
- for (group = 0; group < g_len; group++, cfo+=128) {
- float *cf = cfo;
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ INTFLOAT *cf = cfo;
int len = off_len;
do {
@@ -1564,14 +1612,18 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
+#if USE_FIXED
+ cf = DEC_SQUAD(cf, cb_idx);
+#else
cf = VMUL4(cf, vq, cb_idx, sf + idx);
+#endif /* USE_FIXED */
} while (len -= 4);
}
break;
case 1:
- for (group = 0; group < g_len; group++, cfo+=128) {
- float *cf = cfo;
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ INTFLOAT *cf = cfo;
int len = off_len;
do {
@@ -1586,14 +1638,18 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
nnz = cb_idx >> 8 & 15;
bits = nnz ? GET_CACHE(re, gb) : 0;
LAST_SKIP_BITS(re, gb, nnz);
+#if USE_FIXED
+ cf = DEC_UQUAD(cf, cb_idx, bits);
+#else
cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
+#endif /* USE_FIXED */
} while (len -= 4);
}
break;
case 2:
- for (group = 0; group < g_len; group++, cfo+=128) {
- float *cf = cfo;
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ INTFLOAT *cf = cfo;
int len = off_len;
do {
@@ -1603,15 +1659,19 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
+#if USE_FIXED
+ cf = DEC_SPAIR(cf, cb_idx);
+#else
cf = VMUL2(cf, vq, cb_idx, sf + idx);
+#endif /* USE_FIXED */
} while (len -= 2);
}
break;
case 3:
case 4:
- for (group = 0; group < g_len; group++, cfo+=128) {
- float *cf = cfo;
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ INTFLOAT *cf = cfo;
int len = off_len;
do {
@@ -1626,15 +1686,24 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
nnz = cb_idx >> 8 & 15;
sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
LAST_SKIP_BITS(re, gb, nnz);
+#if USE_FIXED
+ cf = DEC_UPAIR(cf, cb_idx, sign);
+#else
cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
+#endif /* USE_FIXED */
} while (len -= 2);
}
break;
default:
- for (group = 0; group < g_len; group++, cfo+=128) {
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+#if USE_FIXED
+ int *icf = cfo;
+ int v;
+#else
float *cf = cfo;
uint32_t *icf = (uint32_t *) cf;
+#endif /* USE_FIXED */
int len = off_len;
do {
@@ -1678,18 +1747,33 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
b += 4;
n = (1 << b) + SHOW_UBITS(re, gb, b);
LAST_SKIP_BITS(re, gb, b);
+#if USE_FIXED
+ v = n;
+ if (bits & 1U<<31)
+ v = -v;
+ *icf++ = v;
+#else
*icf++ = cbrt_tab[n] | (bits & 1U<<31);
+#endif /* USE_FIXED */
bits <<= 1;
} else {
+#if USE_FIXED
+ v = cb_idx & 15;
+ if (bits & 1U<<31)
+ v = -v;
+ *icf++ = v;
+#else
unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
*icf++ = (bits & 1U<<31) | v;
+#endif /* USE_FIXED */
bits <<= !!v;
}
cb_idx >>= 4;
}
} while (len -= 2);
-
+#if !USE_FIXED
ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
+#endif /* !USE_FIXED */
}
}
@@ -1702,19 +1786,48 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
if (pulse_present) {
idx = 0;
for (i = 0; i < pulse->num_pulse; i++) {
- float co = coef_base[ pulse->pos[i] ];
+ INTFLOAT co = coef_base[ pulse->pos[i] ];
while (offsets[idx + 1] <= pulse->pos[i])
idx++;
if (band_type[idx] != NOISE_BT && sf[idx]) {
- float ico = -pulse->amp[i];
+ INTFLOAT ico = -pulse->amp[i];
+#if USE_FIXED
+ if (co) {
+ ico = co + (co > 0 ? -ico : ico);
+ }
+ coef_base[ pulse->pos[i] ] = ico;
+#else
if (co) {
co /= sf[idx];
ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
}
coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
+#endif /* USE_FIXED */
+ }
+ }
+ }
+#if USE_FIXED
+ coef = coef_base;
+ idx = 0;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ unsigned g_len = ics->group_len[g];
+
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ const unsigned cbt_m1 = band_type[idx] - 1;
+ int *cfo = coef + offsets[i];
+ int off_len = offsets[i + 1] - offsets[i];
+ int group;
+
+ if (cbt_m1 < NOISE_BT - 1) {
+ for (group = 0; group < (int)g_len; group++, cfo+=128) {
+ ac->vector_pow43(cfo, off_len);
+ ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
+ }
}
}
+ coef += g_len << 7;
}
+#endif /* USE_FIXED */
return 0;
}
@@ -1763,7 +1876,7 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
Pulse pulse;
TemporalNoiseShaping *tns = &sce->tns;
IndividualChannelStream *ics = &sce->ics;
- float *out = sce->coeffs;
+ INTFLOAT *out = sce->coeffs;
int global_gain, eld_syntax, er_syntax, pulse_present = 0;
int ret;
@@ -1837,8 +1950,8 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
{
const IndividualChannelStream *ics = &cpe->ch[0].ics;
- float *ch0 = cpe->ch[0].coeffs;
- float *ch1 = cpe->ch[1].coeffs;
+ INTFLOAT *ch0 = cpe->ch[0].coeffs;
+ INTFLOAT *ch1 = cpe->ch[1].coeffs;
int g, i, group, idx = 0;
const uint16_t *offsets = ics->swb_offset;
for (g = 0; g < ics->num_window_groups; g++) {
@@ -1846,10 +1959,17 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
if (cpe->ms_mask[idx] &&
cpe->ch[0].band_type[idx] < NOISE_BT &&
cpe->ch[1].band_type[idx] < NOISE_BT) {
+#if USE_FIXED
+ for (group = 0; group < ics->group_len[g]; group++) {
+ ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
+ ch1 + group * 128 + offsets[i],
+ offsets[i+1] - offsets[i]);
+#else
for (group = 0; group < ics->group_len[g]; group++) {
ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
ch1 + group * 128 + offsets[i],
offsets[i+1] - offsets[i]);
+#endif /* USE_FIXED */
}
}
}
@@ -1870,11 +1990,11 @@ static void apply_intensity_stereo(AACContext *ac,
{
const IndividualChannelStream *ics = &cpe->ch[1].ics;
SingleChannelElement *sce1 = &cpe->ch[1];
- float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
+ INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
const uint16_t *offsets = ics->swb_offset;
int g, group, i, idx = 0;
int c;
- float scale;
+ INTFLOAT scale;
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb;) {
if (sce1->band_type[idx] == INTENSITY_BT ||
@@ -1886,10 +2006,18 @@ static void apply_intensity_stereo(AACContext *ac,
c *= 1 - 2 * cpe->ms_mask[idx];
scale = c * sce1->sf[idx];
for (group = 0; group < ics->group_len[g]; group++)
+#if USE_FIXED
+ ac->subband_scale(coef1 + group * 128 + offsets[i],
+ coef0 + group * 128 + offsets[i],
+ scale,
+ 23,
+ offsets[i + 1] - offsets[i]);
+#else
ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
coef0 + group * 128 + offsets[i],
scale,
offsets[i + 1] - offsets[i]);
+#endif /* USE_FIXED */
}
} else {
int bt_run_end = sce1->band_type_run_end[idx];
@@ -1965,7 +2093,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
int num_gain = 0;
int c, g, sfb, ret;
int sign;
- float scale;
+ INTFLOAT scale;
SingleChannelElement *sce = &che->ch[0];
ChannelCoupling *coup = &che->coup;
@@ -1985,7 +2113,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
sign = get_bits(gb, 1);
- scale = cce_scale[get_bits(gb, 2)];
+ scale = AAC_RENAME(cce_scale)[get_bits(gb, 2)];
if ((ret = decode_ics(ac, sce, gb, 0, 0)))
return ret;
@@ -1994,11 +2122,11 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
int idx = 0;
int cge = 1;
int gain = 0;
- float gain_cache = 1.0;
+ INTFLOAT gain_cache = FIXR10(1.);
if (c) {
cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
- gain_cache = powf(scale, -gain);
+ gain_cache = GET_GAIN(scale, gain);
}
if (coup->coupling_point == AFTER_IMDCT) {
coup->gain[c][0] = gain_cache;
@@ -2015,7 +2143,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
s -= 2 * (t & 0x1);
t >>= 1;
}
- gain_cache = powf(scale, -t) * s;
+ gain_cache = GET_GAIN(scale, t) * s;
}
}
coup->gain[c][idx] = gain_cache;
@@ -2166,7 +2294,7 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
ac->oc[1].m4ac.sbr = 1;
ac->avctx->profile = FF_PROFILE_AAC_HE;
}
- res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
+ res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
break;
case EXT_DYNAMIC_RANGE:
res = decode_dynamic_range(&ac->che_drc, gb);
@@ -2189,14 +2317,14 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
* @param decode 1 if tool is used normally, 0 if tool is used in LTP.
* @param coef spectral coefficients
*/
-static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
+static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
IndividualChannelStream *ics, int decode)
{
const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
int w, filt, m, i;
int bottom, top, order, start, end, size, inc;
- float lpc[TNS_MAX_ORDER];
- float tmp[TNS_MAX_ORDER+1];
+ INTFLOAT lpc[TNS_MAX_ORDER];
+ INTFLOAT tmp[TNS_MAX_ORDER+1];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
@@ -2208,7 +2336,7 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
continue;
// tns_decode_coef
- compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
+ AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
start = ics->swb_offset[FFMIN(bottom, mmm)];
end = ics->swb_offset[FFMIN( top, mmm)];
@@ -2226,13 +2354,13 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
// ar filter
for (m = 0; m < size; m++, start += inc)
for (i = 1; i <= FFMIN(m, order); i++)
- coef[start] -= coef[start - i * inc] * lpc[i - 1];
+ coef[start] -= AAC_MUL26(coef[start - i * inc], lpc[i - 1]);
} else {
// ma filter
for (m = 0; m < size; m++, start += inc) {
tmp[0] = coef[start];
for (i = 1; i <= FFMIN(m, order); i++)
- coef[start] += tmp[i] * lpc[i - 1];
+ coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
for (i = order; i > 0; i--)
tmp[i] = tmp[i - 1];
}
@@ -2245,25 +2373,25 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
* Apply windowing and MDCT to obtain the spectral
* coefficient from the predicted sample by LTP.
*/
-static void windowing_and_mdct_ltp(AACContext *ac, float *out,
- float *in, IndividualChannelStream *ics)
+static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out,
+ INTFLOAT *in, IndividualChannelStream *ics)
{
- const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
- const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
- const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
- const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+ const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
+ const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
+ const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
+ const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
} else {
- memset(in, 0, 448 * sizeof(float));
+ memset(in, 0, 448 * sizeof(INTFLOAT));
ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
}
if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
} else {
ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
- memset(in + 1024 + 576, 0, 448 * sizeof(float));
+ memset(in + 1024 + 576, 0, 448 * sizeof(INTFLOAT));
}
ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
}
@@ -2278,15 +2406,15 @@ static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
int i, sfb;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
- float *predTime = sce->ret;
- float *predFreq = ac->buf_mdct;
+ INTFLOAT *predTime = sce->ret;
+ INTFLOAT *predFreq = ac->buf_mdct;
int16_t num_samples = 2048;
if (ltp->lag < 1024)
num_samples = ltp->lag + 1024;
for (i = 0; i < num_samples; i++)
- predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
- memset(&predTime[i], 0, (2048 - i) * sizeof(float));
+ predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
+ memset(&predTime[i], 0, (2048 - i) * sizeof(INTFLOAT));
ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
@@ -2306,28 +2434,31 @@ static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
- float *saved = sce->saved;
- float *saved_ltp = sce->coeffs;
- const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
- const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+ INTFLOAT *saved = sce->saved;
+ INTFLOAT *saved_ltp = sce->coeffs;
+ const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
+ const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
int i;
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
- memcpy(saved_ltp, saved, 512 * sizeof(float));
- memset(saved_ltp + 576, 0, 448 * sizeof(float));
+ memcpy(saved_ltp, saved, 512 * sizeof(INTFLOAT));
+ memset(saved_ltp + 576, 0, 448 * sizeof(INTFLOAT));
ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+
for (i = 0; i < 64; i++)
- saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
+ saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
- memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
- memset(saved_ltp + 576, 0, 448 * sizeof(float));
+ memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(INTFLOAT));
+ memset(saved_ltp + 576, 0, 448 * sizeof(INTFLOAT));
ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+
for (i = 0; i < 64; i++)
- saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
+ saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
} else { // LONG_STOP or ONLY_LONG
ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
+
for (i = 0; i < 512; i++)
- saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
+ saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
}
memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
@@ -2341,22 +2472,27 @@ static void update_ltp(AACContext *ac, SingleChannelElement *sce)
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
- float *in = sce->coeffs;
- float *out = sce->ret;
- float *saved = sce->saved;
- const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
- const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
- const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
- float *buf = ac->buf_mdct;
- float *temp = ac->temp;
+ INTFLOAT *in = sce->coeffs;
+ INTFLOAT *out = sce->ret;
+ INTFLOAT *saved = sce->saved;
+ const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
+ const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
+ const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
+ INTFLOAT *buf = ac->buf_mdct;
+ INTFLOAT *temp = ac->temp;
int i;
// imdct
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
for (i = 0; i < 1024; i += 128)
ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
- } else
+ } else {
ac->mdct.imdct_half(&ac->mdct, buf, in);
+#if USE_FIXED
+ for (i=0; i<1024; i++)
+ buf[i] = (buf[i] + 4) >> 3;
+#endif /* USE_FIXED */
+ }
/* window overlapping
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
@@ -2368,7 +2504,7 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
} else {
- memcpy( out, saved, 448 * sizeof(float));
+ memcpy( out, saved, 448 * sizeof(INTFLOAT));
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
@@ -2376,65 +2512,71 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
- memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
+ memcpy( out + 448 + 4*128, temp, 64 * sizeof(INTFLOAT));
} else {
ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
- memcpy( out + 576, buf + 64, 448 * sizeof(float));
+ memcpy( out + 576, buf + 64, 448 * sizeof(INTFLOAT));
}
}
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
- memcpy( saved, temp + 64, 64 * sizeof(float));
+ memcpy( saved, temp + 64, 64 * sizeof(INTFLOAT));
ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
- memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
+ memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(INTFLOAT));
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
- memcpy( saved, buf + 512, 448 * sizeof(float));
- memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
+ memcpy( saved, buf + 512, 448 * sizeof(INTFLOAT));
+ memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(INTFLOAT));
} else { // LONG_STOP or ONLY_LONG
- memcpy( saved, buf + 512, 512 * sizeof(float));
+ memcpy( saved, buf + 512, 512 * sizeof(INTFLOAT));
}
}
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
- float *in = sce->coeffs;
- float *out = sce->ret;
- float *saved = sce->saved;
- float *buf = ac->buf_mdct;
+ INTFLOAT *in = sce->coeffs;
+ INTFLOAT *out = sce->ret;
+ INTFLOAT *saved = sce->saved;
+ INTFLOAT *buf = ac->buf_mdct;
// imdct
ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
+#if USE_FIXED
+ int i;
+ for (i = 0; i < 1024; i++)
+ buf[i] = (buf[i] + 2) >> 2;
+#endif /* USE_FIXED */
+
// window overlapping
if (ics->use_kb_window[1]) {
// AAC LD uses a low overlap sine window instead of a KBD window
- memcpy(out, saved, 192 * sizeof(float));
- ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
- memcpy( out + 320, buf + 64, 192 * sizeof(float));
+ memcpy(out, saved, 192 * sizeof(INTFLOAT));
+ ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
+ memcpy( out + 320, buf + 64, 192 * sizeof(INTFLOAT));
} else {
- ac->fdsp->vector_fmul_window(out, saved, buf, ff_sine_512, 256);
+ ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
}
// buffer update
- memcpy(saved, buf + 256, 256 * sizeof(float));
+ memcpy(saved, buf + 256, 256 * sizeof(INTFLOAT));
}
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
{
- float *in = sce->coeffs;
- float *out = sce->ret;
- float *saved = sce->saved;
- float *buf = ac->buf_mdct;
+ INTFLOAT *in = sce->coeffs;
+ INTFLOAT *out = sce->ret;
+ INTFLOAT *saved = sce->saved;
+ INTFLOAT *buf = ac->buf_mdct;
int i;
const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
const int n2 = n >> 1;
const int n4 = n >> 2;
- const float *const window = n == 480 ? ff_aac_eld_window_480 :
- ff_aac_eld_window_512;
+ const INTFLOAT *const window = n == 480 ? ff_aac_eld_window_480 :
+ AAC_RENAME(ff_aac_eld_window_512);
// Inverse transform, mapped to the conventional IMDCT by
// Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
@@ -2442,14 +2584,22 @@ static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
// International Conference on Audio, Language and Image Processing, ICALIP 2008.
// URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
for (i = 0; i < n2; i+=2) {
- float temp;
+ INTFLOAT temp;
temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
}
+#if !USE_FIXED
if (n == 480)
ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
else
+#endif
ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
+
+#if USE_FIXED
+ for (i = 0; i < 1024; i++)
+ buf[i] = (buf[i] + 1) >> 1;
+#endif /* USE_FIXED */
+
for (i = 0; i < n; i+=2) {
buf[i] = -buf[i];
}
@@ -2461,26 +2611,26 @@ static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
// The spec says to use samples [0..511] but the reference decoder uses
// samples [128..639].
for (i = n4; i < n2; i ++) {
- out[i - n4] = buf[n2 - 1 - i] * window[i - n4] +
- saved[ i + n2] * window[i + n - n4] +
- -saved[ n + n2 - 1 - i] * window[i + 2*n - n4] +
- -saved[2*n + n2 + i] * window[i + 3*n - n4];
+ out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
+ AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
+ AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
+ AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
}
for (i = 0; i < n2; i ++) {
- out[n4 + i] = buf[i] * window[i + n2 - n4] +
- -saved[ n - 1 - i] * window[i + n2 + n - n4] +
- -saved[ n + i] * window[i + n2 + 2*n - n4] +
- saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
+ out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
+ AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
+ AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
+ AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
}
for (i = 0; i < n4; i ++) {
- out[n2 + n4 + i] = buf[ i + n2] * window[i + n - n4] +
- -saved[ n2 - 1 - i] * window[i + 2*n - n4] +
- -saved[ n + n2 + i] * window[i + 3*n - n4];
+ out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
+ AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
+ AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
}
// buffer update
- memmove(saved + n, saved, 2 * n * sizeof(float));
- memcpy( saved, buf, n * sizeof(float));
+ memmove(saved + n, saved, 2 * n * sizeof(INTFLOAT));
+ memcpy( saved, buf, n * sizeof(INTFLOAT));
}
/**
@@ -2519,7 +2669,7 @@ static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
}
/**
- * Convert spectral data to float samples, applying all supported tools as appropriate.
+ * Convert spectral data to samples, applying all supported tools as appropriate.
*/
static void spectral_to_sample(AACContext *ac)
{
@@ -2540,7 +2690,7 @@ static void spectral_to_sample(AACContext *ac)
ChannelElement *che = ac->che[type][i];
if (che && che->present) {
if (type <= TYPE_CPE)
- apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
+ apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(apply_dependent_coupling));
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
if (che->ch[0].ics.predictor_present) {
if (che->ch[0].ics.ltp.present)
@@ -2554,7 +2704,7 @@ static void spectral_to_sample(AACContext *ac)
if (che->ch[1].tns.present)
ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
if (type <= TYPE_CPE)
- apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
+ apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, AAC_RENAME(apply_dependent_coupling));
if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
imdct_and_window(ac, &che->ch[0]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
@@ -2565,11 +2715,22 @@ static void spectral_to_sample(AACContext *ac)
ac->update_ltp(ac, &che->ch[1]);
}
if (ac->oc[1].m4ac.sbr > 0) {
- ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
+ AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
}
}
if (type <= TYPE_CCE)
- apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
+ apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling));
+
+#if USE_FIXED
+ {
+ int j;
+ /* preparation for resampler */
+ for(j = 0; j<2048; j++){
+ che->ch[0].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[0].ret[j]<<7)+0x8000;
+ che->ch[1].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[1].ret[j]<<7)+0x8000;
+ }
+ }
+#endif /* USE_FIXED */
che->present = 0;
} else if (che) {
av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
@@ -2954,7 +3115,7 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
for (i = 0; i < MAX_ELEM_ID; i++) {
for (type = 0; type < 4; type++) {
if (ac->che[type][i])
- ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
+ AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr);
av_freep(&ac->che[type][i]);
}
}
@@ -2963,7 +3124,9 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
ff_mdct_end(&ac->mdct_small);
ff_mdct_end(&ac->mdct_ld);
ff_mdct_end(&ac->mdct_ltp);
+#if !USE_FIXED
ff_imdct15_uninit(&ac->mdct480);
+#endif
av_freep(&ac->fdsp);
return 0;
}
@@ -2975,9 +3138,15 @@ static void aacdec_init(AACContext *c)
c->apply_tns = apply_tns;
c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
c->update_ltp = update_ltp;
+#if USE_FIXED
+ c->vector_pow43 = vector_pow43;
+ c->subband_scale = subband_scale;
+#endif
+#if !USE_FIXED
if(ARCH_MIPS)
ff_aacdec_init_mips(c);
+#endif /* !USE_FIXED */
}
/**
* AVOptions for Japanese DTV specific extensions (ADTS only)
diff --git a/libavcodec/lpc.h b/libavcodec/lpc.h
index 96acb37..9b76e2f 100644
--- a/libavcodec/lpc.h
+++ b/libavcodec/lpc.h
@@ -25,6 +25,7 @@
#include <stdint.h>
#include "libavutil/avassert.h"
#include "libavutil/lls.h"
+#include "aac_defines.h"
#define ORDER_METHOD_EST 0
#define ORDER_METHOD_2LEVEL 1
@@ -111,11 +112,15 @@ void ff_lpc_init_x86(LPCContext *s);
*/
void ff_lpc_end(LPCContext *s);
+#if USE_FIXED
+#define LPC_TYPE int
+#else
#ifdef LPC_USE_DOUBLE
#define LPC_TYPE double
#else
#define LPC_TYPE float
#endif
+#endif // USE_FIXED
/**
* Schur recursion.
@@ -152,7 +157,7 @@ static inline void compute_ref_coefs(const LPC_TYPE *autoc, int max_order,
* Levinson-Durbin recursion.
* Produce LPC coefficients from autocorrelation data.
*/
-static inline int compute_lpc_coefs(const LPC_TYPE *autoc, int max_order,
+static inline int AAC_RENAME(compute_lpc_coefs)(const LPC_TYPE *autoc, int max_order,
LPC_TYPE *lpc, int lpc_stride, int fail,
int normalize)
{
@@ -169,14 +174,14 @@ static inline int compute_lpc_coefs(const LPC_TYPE *autoc, int max_order,
return -1;
for(i=0; i<max_order; i++) {
- LPC_TYPE r = -autoc[i];
+ LPC_TYPE r = AAC_SRA_R(-autoc[i], 5);
if (normalize) {
for(j=0; j<i; j++)
r -= lpc_last[j] * autoc[i-j-1];
r /= err;
- err *= 1.0 - (r * r);
+ err *= FIXR(1.0) - (r * r);
}
lpc[i] = r;
@@ -184,8 +189,8 @@ static inline int compute_lpc_coefs(const LPC_TYPE *autoc, int max_order,
for(j=0; j < (i+1)>>1; j++) {
LPC_TYPE f = lpc_last[ j];
LPC_TYPE b = lpc_last[i-1-j];
- lpc[ j] = f + r * b;
- lpc[i-1-j] = b + r * f;
+ lpc[ j] = f + AAC_MUL26(r, b);
+ lpc[i-1-j] = b + AAC_MUL26(r, f);
}
if (fail && err < 0)
diff --git a/libavcodec/mdct_template.c b/libavcodec/mdct_template.c
index 7fa8bcc..e7e5f62 100644
--- a/libavcodec/mdct_template.c
+++ b/libavcodec/mdct_template.c
@@ -81,8 +81,13 @@ av_cold int ff_mdct_init(FFTContext *s, int nbits, int inverse, double scale)
scale = sqrt(fabs(scale));
for(i=0;i<n4;i++) {
alpha = 2 * M_PI * (i + theta) / n;
+#if FFT_FIXED_32
+ s->tcos[i*tstep] = (FFTSample)floor(-cos(alpha) * 2147483648.0 + 0.5);
+ s->tsin[i*tstep] = (FFTSample)floor(-sin(alpha) * 2147483648.0 + 0.5);
+#else
s->tcos[i*tstep] = FIX15(-cos(alpha) * scale);
s->tsin[i*tstep] = FIX15(-sin(alpha) * scale);
+#endif
}
return 0;
fail:
diff --git a/libavutil/fixed_dsp.c b/libavutil/fixed_dsp.c
index 9f2e841..8ddcdef 100644
--- a/libavutil/fixed_dsp.c
+++ b/libavutil/fixed_dsp.c
@@ -26,7 +26,7 @@
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*
- * Author: Nedeljko Babic (nbabic at mips.com)
+ * Author: Nedeljko Babic (nedeljko.babic imgtec com)
*
* This file is part of FFmpeg.
*
@@ -47,7 +47,29 @@
#include "fixed_dsp.h"
-static void vector_fmul_window_fixed_scaled_c(int16_t *dst, const int32_t *src0,
+static void vector_fmul_add_c(int *dst, const int *src0, const int *src1, const int *src2, int len){
+ int i;
+ int64_t accu;
+
+ for (i=0; i<len; i++) {
+ accu = (int64_t)src0[i] * src1[i];
+ dst[i] = src2[i] + (int)((accu + 0x40000000) >> 31);
+ }
+}
+
+static void vector_fmul_reverse_c(int *dst, const int *src0, const int *src1, int len)
+{
+ int i;
+ int64_t accu;
+
+ src1 += len-1;
+ for (i=0; i<len; i++) {
+ accu = (int64_t)src0[i] * src1[-i];
+ dst[i] = (int)((accu+0x40000000) >> 31);
+ }
+}
+
+static void vector_fmul_window_scaled_c(int16_t *dst, const int32_t *src0,
const int32_t *src1, const int32_t *win,
int len, uint8_t bits)
{
@@ -68,7 +90,7 @@ static void vector_fmul_window_fixed_scaled_c(int16_t *dst, const int32_t *src0,
}
}
-static void vector_fmul_window_fixed_c(int32_t *dst, const int32_t *src0,
+static void vector_fmul_window_c(int32_t *dst, const int32_t *src0,
const int32_t *src1, const int32_t *win,
int len)
{
@@ -88,6 +110,39 @@ static void vector_fmul_window_fixed_c(int32_t *dst, const int32_t *src0,
}
}
+static void vector_fmul_c(int *dst, const int *src0, const int *src1, int len)
+{
+ int i;
+ int64_t accu;
+
+ for (i = 0; i < len; i++){
+ accu = (int64_t)src0[i] * src1[i];
+ dst[i] = (int)((accu+0x40000000) >> 31);
+ }
+}
+
+static int ff_scalarproduct_fixed_c(const int *v1, const int *v2, int len)
+{
+ int64_t p = 0;
+ int i;
+
+ for (i = 0; i < len; i++)
+ p += (int64_t)v1[i] * v2[i];
+
+ return (int)((p + 0x40000000) >> 31);
+}
+
+static void butterflies_fixed_c(int *v1, int *v2, int len)
+{
+ int i;
+
+ for (i = 0; i < len; i++){
+ int t = v1[i] - v2[i];
+ v1[i] += v2[i];
+ v2[i] = t;
+ }
+}
+
AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
{
AVFixedDSPContext * fdsp = av_malloc(sizeof(AVFixedDSPContext));
@@ -95,8 +150,13 @@ AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
if (!fdsp)
return NULL;
- fdsp->vector_fmul_window_scaled = vector_fmul_window_fixed_scaled_c;
- fdsp->vector_fmul_window = vector_fmul_window_fixed_c;
+ fdsp->vector_fmul_window_scaled = vector_fmul_window_scaled_c;
+ fdsp->vector_fmul_window = vector_fmul_window_c;
+ fdsp->vector_fmul = vector_fmul_c;
+ fdsp->vector_fmul_add = vector_fmul_add_c;
+ fdsp->vector_fmul_reverse = vector_fmul_reverse_c;
+ fdsp->butterflies_fixed = butterflies_fixed_c;
+ fdsp->scalarproduct_fixed = ff_scalarproduct_fixed_c;
return fdsp;
}
diff --git a/libavutil/fixed_dsp.h b/libavutil/fixed_dsp.h
index ff6f365..73859c0 100644
--- a/libavutil/fixed_dsp.h
+++ b/libavutil/fixed_dsp.h
@@ -54,6 +54,8 @@
#include "libavcodec/mathops.h"
typedef struct AVFixedDSPContext {
+ /* assume len is a multiple of 16, and arrays are 32-byte aligned */
+
/**
* Overlap/add with window function.
* Used primarily by MDCT-based audio codecs.
@@ -92,6 +94,57 @@ typedef struct AVFixedDSPContext {
*/
void (*vector_fmul_window)(int32_t *dst, const int32_t *src0, const int32_t *src1, const int32_t *win, int len);
+ /**
+ * Fixed-point multiplication that calculates the product of two vectors of
+ * integers and stores the result in a vector of integers.
+ *
+ * @param dst output vector
+ * constraints: 32-byte aligned
+ * @param src0 first input vector
+ * constraints: 32-byte aligned
+ * @param src1 second input vector
+ * constraints: 32-byte aligned
+ * @param len number of elements in the input
+ * constraints: multiple of 16
+ */
+ void (*vector_fmul)(int *dst, const int *src0, const int *src1,
+ int len);
+
+ void (*vector_fmul_reverse)(int *dst, const int *src0, const int *src1, int len);
+ /**
+ * Calculate the product of two vectors of integers, add a third vector of
+ * integers and store the result in a vector of integers.
+ *
+ * @param dst output vector
+ * constraints: 32-byte aligned
+ * @param src0 first input vector
+ * constraints: 32-byte aligned
+ * @param src1 second input vector
+ * constraints: 32-byte aligned
+ * @param src1 third input vector
+ * constraints: 32-byte aligned
+ * @param len number of elements in the input
+ * constraints: multiple of 16
+ */
+ void (*vector_fmul_add)(int *dst, const int *src0, const int *src1,
+ const int *src2, int len);
+
+ /**
+ * Calculate the scalar product of two vectors of integers.
+ * @param v1 first vector, 16-byte aligned
+ * @param v2 second vector, 16-byte aligned
+ * @param len length of vectors, multiple of 4
+ */
+ int (*scalarproduct_fixed)(const int *v1, const int *v2, int len);
+
+ /**
+ * Calculate the sum and difference of two vectors of integers.
+ *
+ * @param v1 first input vector, sum output, 16-byte aligned
+ * @param v2 second input vector, difference output, 16-byte aligned
+ * @param len length of vectors, multiple of 4
+ */
+ void (*butterflies_fixed)(int *av_restrict v1, int *av_restrict v2, int len);
} AVFixedDSPContext;
/**
--
1.8.2.1
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