[FFmpeg-devel] [PATCH 07/12] libavcodec: Implementation of AAC_fixed_decoder (SBR-module) [3/3]
Nedeljko Babic
nedeljko.babic at imgtec.com
Tue Jun 30 11:53:09 CEST 2015
From: Djordje Pesut <djordje.pesut at imgtec.com>
Add fixed poind code.
Signed-off-by: Nedeljko Babic <nedeljko.babic at imgtec.com>
---
libavcodec/Makefile | 5 +-
libavcodec/aac.h | 52 +---
libavcodec/aac_defines.h | 78 ++++++
libavcodec/aacdec_template.c | 14 +-
libavcodec/aacsbr.c | 1 +
libavcodec/aacsbr.h | 12 +-
libavcodec/aacsbr_fixed.c | 586 +++++++++++++++++++++++++++++++++++++++++++
libavcodec/aacsbr_template.c | 211 ++++++++++++----
libavcodec/lpc.h | 15 +-
libavcodec/sbr.h | 78 +++---
libavcodec/sbrdsp.c | 3 +
libavcodec/sbrdsp.h | 36 +--
libavcodec/sbrdsp_fixed.c | 286 +++++++++++++++++++++
libavcodec/sbrdsp_template.c | 42 ++--
libavutil/softfloat.h | 8 +
15 files changed, 1237 insertions(+), 190 deletions(-)
create mode 100644 libavcodec/aac_defines.h
create mode 100644 libavcodec/aacsbr_fixed.c
create mode 100644 libavcodec/sbrdsp_fixed.c
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 642da3d..8bdf5a8 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -119,8 +119,9 @@ OBJS-$(CONFIG_A64MULTI5_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o \
aacadtsdec.o mpeg4audio.o kbdwin.o \
sbrdsp.o aacpsdsp.o
-OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o \
- aacadtsdec.o mpeg4audio.o kbdwin.o
+OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o aacsbr_fixed.o \
+ aacadtsdec.o mpeg4audio.o kbdwin.o \
+ sbrdsp_fixed.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
aacpsy.o aactab.o \
psymodel.o mpeg4audio.o kbdwin.o
diff --git a/libavcodec/aac.h b/libavcodec/aac.h
index f6fd446..d62455d 100644
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -30,58 +30,8 @@
#ifndef AVCODEC_AAC_H
#define AVCODEC_AAC_H
-#ifndef USE_FIXED
-#define USE_FIXED 0
-#endif
-
-#if USE_FIXED
-
-#include "libavutil/softfloat.h"
-
-#define FFT_FLOAT 0
-#define FFT_FIXED_32 1
-
-#define AAC_RENAME(x) x ## _fixed
-#define AAC_RENAME_32(x) x ## _fixed_32
-#define AAC_FLOAT SoftFloat
-#define INTFLOAT int
-#define SHORTFLOAT int16_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 AAC_FLOAT float
-#define INTFLOAT float
-#define SHORTFLOAT 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 "aac_defines.h"
#include "libavutil/float_dsp.h"
#include "libavutil/fixed_dsp.h"
#include "avcodec.h"
diff --git a/libavcodec/aac_defines.h b/libavcodec/aac_defines.h
new file mode 100644
index 0000000..0f3905f
--- /dev/null
+++ b/libavcodec/aac_defines.h
@@ -0,0 +1,78 @@
+/*
+ * AAC defines
+ *
+ * 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
+ */
+
+#ifndef AVCODEC_AAC_DEFINES_H
+#define AVCODEC_AAC_DEFINES_H
+
+#ifndef USE_FIXED
+#define USE_FIXED 0
+#endif
+
+#if USE_FIXED
+
+#include "libavutil/softfloat.h"
+
+#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 SoftFloat
+#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)
+#define AAC_SRA_R(x, y) (int)(((x) + (1 << ((y) - 1))) >> (y))
+
+#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))
+#define AAC_SRA_R(x, y) (x)
+
+#endif /* USE_FIXED */
+
+#endif /* AVCODEC_AAC_DEFINES_H */
diff --git a/libavcodec/aacdec_template.c b/libavcodec/aacdec_template.c
index d8eaca3..b4eee85 100644
--- a/libavcodec/aacdec_template.c
+++ b/libavcodec/aacdec_template.c
@@ -132,7 +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))) {
@@ -147,7 +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;
@@ -1126,7 +1126,7 @@ 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);
@@ -2315,7 +2315,7 @@ 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);
@@ -2357,7 +2357,7 @@ static void apply_tns(INTFLOAT 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)];
@@ -2738,7 +2738,7 @@ 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)
@@ -3153,7 +3153,7 @@ 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]);
}
}
diff --git a/libavcodec/aacsbr.c b/libavcodec/aacsbr.c
index 766c47b..81f1902 100644
--- a/libavcodec/aacsbr.c
+++ b/libavcodec/aacsbr.c
@@ -25,6 +25,7 @@
* AAC Spectral Band Replication decoding functions
* @author Robert Swain ( rob opendot cl )
*/
+#define USE_FIXED 0
#include "aac.h"
#include "sbr.h"
diff --git a/libavcodec/aacsbr.h b/libavcodec/aacsbr.h
index 476bc65..ed1a7f9 100644
--- a/libavcodec/aacsbr.h
+++ b/libavcodec/aacsbr.h
@@ -79,17 +79,17 @@ static const int8_t vlc_sbr_lav[10] =
{ name ## _codes, name ## _bits, sizeof(name ## _codes), sizeof(name ## _codes[0]) }
/** Initialize SBR. */
-void ff_aac_sbr_init(void);
+void AAC_RENAME(ff_aac_sbr_init)(void);
/** Initialize one SBR context. */
-void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr);
+void AAC_RENAME(ff_aac_sbr_ctx_init)(AACContext *ac, SpectralBandReplication *sbr);
/** Close one SBR context. */
-void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr);
+void AAC_RENAME(ff_aac_sbr_ctx_close)(SpectralBandReplication *sbr);
/** Decode one SBR element. */
-int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
+int AAC_RENAME(ff_decode_sbr_extension)(AACContext *ac, SpectralBandReplication *sbr,
GetBitContext *gb, int crc, int cnt, int id_aac);
/** Apply one SBR element to one AAC element. */
-void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
- float* L, float *R);
+void AAC_RENAME(ff_sbr_apply)(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
+ INTFLOAT* L, INTFLOAT *R);
void ff_aacsbr_func_ptr_init_mips(AACSBRContext *c);
diff --git a/libavcodec/aacsbr_fixed.c b/libavcodec/aacsbr_fixed.c
new file mode 100644
index 0000000..5a5c9cc
--- /dev/null
+++ b/libavcodec/aacsbr_fixed.c
@@ -0,0 +1,586 @@
+/*
+ * 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 Spectral Band Replication decoding functions (fixed-point)
+ * Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
+ * Copyright (c) 2009-2010 Alex Converse <alex.converse at 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 Spectral Band Replication decoding functions (fixed-point)
+ * Note: Rounding-to-nearest used unless otherwise stated
+ * @author Robert Swain ( rob opendot cl )
+ * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
+ */
+#define USE_FIXED 1
+
+#include "aac.h"
+#include "sbr.h"
+#include "aacsbr.h"
+#include "aacsbrdata.h"
+#include "aacsbr_fixed_tablegen.h"
+#include "fft.h"
+#include "aacps.h"
+#include "sbrdsp.h"
+#include "libavutil/internal.h"
+#include "libavutil/libm.h"
+#include "libavutil/avassert.h"
+
+#include <stdint.h>
+#include <float.h>
+#include <math.h>
+
+static VLC vlc_sbr[10];
+static void aacsbr_func_ptr_init(AACSBRContext *c);
+static const int CONST_LN2 = Q31(0.6931471806/256); // ln(2)/256
+static const int CONST_RECIP_LN2 = Q31(0.7213475204); // 0.5/ln(2)
+static const int CONST_SQRT2 = Q30(0.7071067812); // sqrt(2)/2
+static const int CONST_076923 = Q31(0.76923076923076923077f);
+
+int fixed_log_table[10] =
+{
+ Q31(1.0/2), Q31(1.0/3), Q31(1.0/4), Q31(1.0/5), Q31(1.0/6),
+ Q31(1.0/7), Q31(1.0/8), Q31(1.0/9), Q31(1.0/10), Q31(1.0/11)
+};
+
+static int fixed_log(int x)
+{
+ int i, ret, xpow, tmp;
+
+ ret = x;
+ xpow = x;
+ for (i=0; i<10; i+=2){
+ xpow = (int)(((int64_t)xpow * x + 0x40000000) >> 31);
+ tmp = (int)(((int64_t)xpow * fixed_log_table[i] + 0x40000000) >> 31);
+ ret -= tmp;
+
+ xpow = (int)(((int64_t)xpow * x + 0x40000000) >> 31);
+ tmp = (int)(((int64_t)xpow * fixed_log_table[i+1] + 0x40000000) >> 31);
+ ret += tmp;
+ }
+
+ return ret;
+}
+
+int fixed_exp_table[7] =
+{
+ Q31(1.0/2), Q31(1.0/6), Q31(1.0/24), Q31(1.0/120),
+ Q31(1.0/720), Q31(1.0/5040), Q31(1.0/40320)
+};
+
+static int fixed_exp(int x)
+{
+ int i, ret, xpow, tmp;
+
+ ret = 0x800000 + x;
+ xpow = x;
+ for (i=0; i<7; i++){
+ xpow = (int)(((int64_t)xpow * x + 0x400000) >> 23);
+ tmp = (int)(((int64_t)xpow * fixed_exp_table[i] + 0x40000000) >> 31);
+ ret += tmp;
+ }
+
+ return ret;
+}
+
+static void make_bands(int16_t* bands, int start, int stop, int num_bands)
+{
+ int k, previous, present;
+ int base, prod, nz = 0;
+
+ base = (stop << 23) / start;
+ while (base < 0x40000000){
+ base <<= 1;
+ nz++;
+ }
+ base = fixed_log(base - 0x80000000);
+ base = (((base + 0x80) >> 8) + (8-nz)*CONST_LN2) / num_bands;
+ base = fixed_exp(base);
+
+ previous = start;
+ prod = start << 23;
+
+ for (k = 0; k < num_bands-1; k++) {
+ prod = (int)(((int64_t)prod * base + 0x400000) >> 23);
+ present = (prod + 0x400000) >> 23;
+ bands[k] = present - previous;
+ previous = present;
+ }
+ bands[num_bands-1] = stop - previous;
+}
+
+/// Dequantization and stereo decoding (14496-3 sp04 p203)
+static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
+{
+ int k, e;
+ int ch;
+
+ if (id_aac == TYPE_CPE && sbr->bs_coupling) {
+ int alpha = sbr->data[0].bs_amp_res ? 2 : 1;
+ int pan_offset = sbr->data[0].bs_amp_res ? 12 : 24;
+ for (e = 1; e <= sbr->data[0].bs_num_env; e++) {
+ for (k = 0; k < sbr->n[sbr->data[0].bs_freq_res[e]]; k++) {
+ SoftFloat temp1, temp2, fac;
+
+ temp1.exp = sbr->data[0].env_facs[e][k].mant * alpha + 14;
+ if (temp1.exp & 1)
+ temp1.mant = 759250125;
+ else
+ temp1.mant = 0x20000000;
+ temp1.exp = (temp1.exp >> 1) + 1;
+
+ temp2.exp = (pan_offset - sbr->data[1].env_facs[e][k].mant) * alpha;
+ if (temp2.exp & 1)
+ temp2.mant = 759250125;
+ else
+ temp2.mant = 0x20000000;
+ temp2.exp = (temp2.exp >> 1) + 1;
+ fac = av_div_sf(temp1, av_add_sf(FLOAT_1, temp2));
+ sbr->data[0].env_facs[e][k] = fac;
+ sbr->data[1].env_facs[e][k] = av_mul_sf(fac, temp2);
+ }
+ }
+ for (e = 1; e <= sbr->data[0].bs_num_noise; e++) {
+ for (k = 0; k < sbr->n_q; k++) {
+ SoftFloat temp1, temp2, fac;
+
+ temp1.exp = NOISE_FLOOR_OFFSET - \
+ sbr->data[0].noise_facs[e][k].mant + 2;
+ temp1.mant = 0x20000000;
+ temp2.exp = 12 - sbr->data[1].noise_facs[e][k].mant + 1;
+ temp2.mant = 0x20000000;
+ fac = av_div_sf(temp1, av_add_sf(FLOAT_1, temp2));
+ sbr->data[0].noise_facs[e][k] = fac;
+ sbr->data[1].noise_facs[e][k] = av_mul_sf(fac, temp2);
+ }
+ }
+ } else { // SCE or one non-coupled CPE
+ for (ch = 0; ch < (id_aac == TYPE_CPE) + 1; ch++) {
+ int alpha = sbr->data[ch].bs_amp_res ? 2 : 1;
+ for (e = 1; e <= sbr->data[ch].bs_num_env; e++)
+ for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++){
+ SoftFloat temp1;
+
+ temp1.exp = alpha * sbr->data[ch].env_facs[e][k].mant + 12;
+ if (temp1.exp & 1)
+ temp1.mant = 759250125;
+ else
+ temp1.mant = 0x20000000;
+ temp1.exp = (temp1.exp >> 1) + 1;
+
+ sbr->data[ch].env_facs[e][k] = temp1;
+ }
+ for (e = 1; e <= sbr->data[ch].bs_num_noise; e++)
+ for (k = 0; k < sbr->n_q; k++){
+ sbr->data[ch].noise_facs[e][k].exp = NOISE_FLOOR_OFFSET - \
+ sbr->data[ch].noise_facs[e][k].mant + 1;
+ sbr->data[ch].noise_facs[e][k].mant = 0x20000000;
+ }
+ }
+ }
+}
+
+/** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
+ * (14496-3 sp04 p214)
+ * Warning: This routine does not seem numerically stable.
+ */
+static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
+ int (*alpha0)[2], int (*alpha1)[2],
+ const int X_low[32][40][2], int k0)
+{
+ int k;
+ int shift, round;
+
+ for (k = 0; k < k0; k++) {
+ SoftFloat phi[3][2][2];
+ SoftFloat a00, a01, a10, a11;
+ SoftFloat dk;
+
+ dsp->autocorrelate(X_low[k], phi);
+
+ dk = av_sub_sf(av_mul_sf(phi[2][1][0], phi[1][0][0]),
+ av_mul_sf(av_add_sf(av_mul_sf(phi[1][1][0], phi[1][1][0]),
+ av_mul_sf(phi[1][1][1], phi[1][1][1])), FLOAT_0999999));
+
+ if (!dk.mant) {
+ a10 = FLOAT_0;
+ a11 = FLOAT_0;
+ } else {
+ SoftFloat temp_real, temp_im;
+ temp_real = av_sub_sf(av_sub_sf(av_mul_sf(phi[0][0][0], phi[1][1][0]),
+ av_mul_sf(phi[0][0][1], phi[1][1][1])),
+ av_mul_sf(phi[0][1][0], phi[1][0][0]));
+ temp_im = av_sub_sf(av_add_sf(av_mul_sf(phi[0][0][0], phi[1][1][1]),
+ av_mul_sf(phi[0][0][1], phi[1][1][0])),
+ av_mul_sf(phi[0][1][1], phi[1][0][0]));
+
+ a10 = av_div_sf(temp_real, dk);
+ a11 = av_div_sf(temp_im, dk);
+ }
+
+ if (!phi[1][0][0].mant) {
+ a00 = FLOAT_0;
+ a01 = FLOAT_0;
+ } else {
+ SoftFloat temp_real, temp_im;
+ temp_real = av_add_sf(phi[0][0][0],
+ av_add_sf(av_mul_sf(a10, phi[1][1][0]),
+ av_mul_sf(a11, phi[1][1][1])));
+ temp_im = av_add_sf(phi[0][0][1],
+ av_sub_sf(av_mul_sf(a11, phi[1][1][0]),
+ av_mul_sf(a10, phi[1][1][1])));
+
+ temp_real.mant = -temp_real.mant;
+ temp_im.mant = -temp_im.mant;
+ a00 = av_div_sf(temp_real, phi[1][0][0]);
+ a01 = av_div_sf(temp_im, phi[1][0][0]);
+ }
+
+ shift = a00.exp;
+ if (shift >= 3)
+ alpha0[k][0] = 0x7fffffff;
+ else {
+ a00.mant <<= 1;
+ shift = 2-shift;
+ if (shift == 0)
+ alpha0[k][0] = a00.mant;
+ else {
+ round = 1 << (shift-1);
+ alpha0[k][0] = (a00.mant + round) >> shift;
+ }
+ }
+
+ shift = a01.exp;
+ if (shift >= 3)
+ alpha0[k][1] = 0x7fffffff;
+ else {
+ a01.mant <<= 1;
+ shift = 2-shift;
+ if (shift == 0)
+ alpha0[k][1] = a01.mant;
+ else {
+ round = 1 << (shift-1);
+ alpha0[k][1] = (a01.mant + round) >> shift;
+ }
+ }
+ shift = a10.exp;
+ if (shift >= 3)
+ alpha1[k][0] = 0x7fffffff;
+ else {
+ a10.mant <<= 1;
+ shift = 2-shift;
+ if (shift == 0)
+ alpha1[k][0] = a10.mant;
+ else {
+ round = 1 << (shift-1);
+ alpha1[k][0] = (a10.mant + round) >> shift;
+ }
+ }
+
+ shift = a11.exp;
+ if (shift >= 3)
+ alpha1[k][1] = 0x7fffffff;
+ else {
+ a11.mant <<= 1;
+ shift = 2-shift;
+ if (shift == 0)
+ alpha1[k][1] = a11.mant;
+ else {
+ round = 1 << (shift-1);
+ alpha1[k][1] = (a11.mant + round) >> shift;
+ }
+ }
+
+ shift = (int)(((int64_t)(alpha1[k][0]>>1) * (alpha1[k][0]>>1) + \
+ (int64_t)(alpha1[k][1]>>1) * (alpha1[k][1]>>1) + \
+ 0x40000000) >> 31);
+ if (shift >= 0x20000000){
+ alpha1[k][0] = 0;
+ alpha1[k][1] = 0;
+ alpha0[k][0] = 0;
+ alpha0[k][1] = 0;
+ }
+
+ shift = (int)(((int64_t)(alpha0[k][0]>>1) * (alpha0[k][0]>>1) + \
+ (int64_t)(alpha0[k][1]>>1) * (alpha0[k][1]>>1) + \
+ 0x40000000) >> 31);
+ if (shift >= 0x20000000){
+ alpha1[k][0] = 0;
+ alpha1[k][1] = 0;
+ alpha0[k][0] = 0;
+ alpha0[k][1] = 0;
+ }
+ }
+}
+
+/// Chirp Factors (14496-3 sp04 p214)
+static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
+{
+ int i;
+ int new_bw;
+ static const int bw_tab[] = { 0, 1610612736, 1932735283, 2104533975 };
+ int64_t accu;
+
+ for (i = 0; i < sbr->n_q; i++) {
+ if (ch_data->bs_invf_mode[0][i] + ch_data->bs_invf_mode[1][i] == 1)
+ new_bw = 1288490189;
+ else
+ new_bw = bw_tab[ch_data->bs_invf_mode[0][i]];
+
+ if (new_bw < ch_data->bw_array[i]){
+ accu = (int64_t)new_bw * 1610612736;
+ accu += (int64_t)ch_data->bw_array[i] * 0x20000000;
+ new_bw = (int)((accu + 0x40000000) >> 31);
+ } else {
+ accu = (int64_t)new_bw * 1946157056;
+ accu += (int64_t)ch_data->bw_array[i] * 201326592;
+ new_bw = (int)((accu + 0x40000000) >> 31);
+ }
+ ch_data->bw_array[i] = new_bw < 0x2000000 ? 0 : new_bw;
+ }
+}
+
+/**
+ * Calculation of levels of additional HF signal components (14496-3 sp04 p219)
+ * and Calculation of gain (14496-3 sp04 p219)
+ */
+static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
+ SBRData *ch_data, const int e_a[2])
+{
+ int e, k, m;
+ // max gain limits : -3dB, 0dB, 3dB, inf dB (limiter off)
+ static const SoftFloat limgain[4] = { { 760155524, 0 }, { 0x20000000, 1 },
+ { 758351638, 1 }, { 625000000, 34 } };
+
+ for (e = 0; e < ch_data->bs_num_env; e++) {
+ int delta = !((e == e_a[1]) || (e == e_a[0]));
+ for (k = 0; k < sbr->n_lim; k++) {
+ SoftFloat gain_boost, gain_max;
+ SoftFloat sum[2] = { { 0, 0}, { 0, 0 } };
+ for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+ const SoftFloat temp = av_div_sf(sbr->e_origmapped[e][m],
+ av_add_sf(FLOAT_1, sbr->q_mapped[e][m]));
+ sbr->q_m[e][m] = av_sqrt_sf(av_mul_sf(temp, sbr->q_mapped[e][m]));
+ sbr->s_m[e][m] = av_sqrt_sf(av_mul_sf(temp, av_int2sf(ch_data->s_indexmapped[e + 1][m], 0)));
+ if (!sbr->s_mapped[e][m]) {
+ if (delta) {
+ sbr->gain[e][m] = av_sqrt_sf(av_div_sf(sbr->e_origmapped[e][m],
+ av_mul_sf(av_add_sf(FLOAT_1, sbr->e_curr[e][m]),
+ av_add_sf(FLOAT_1, sbr->q_mapped[e][m]))));
+ } else {
+ sbr->gain[e][m] = av_sqrt_sf(av_div_sf(sbr->e_origmapped[e][m],
+ av_add_sf(FLOAT_1, sbr->e_curr[e][m])));
+ }
+ } else {
+ sbr->gain[e][m] = av_sqrt_sf(
+ av_div_sf(
+ av_mul_sf(sbr->e_origmapped[e][m], sbr->q_mapped[e][m]),
+ av_mul_sf(
+ av_add_sf(FLOAT_1, sbr->e_curr[e][m]),
+ av_add_sf(FLOAT_1, sbr->q_mapped[e][m]))));
+ }
+ }
+ for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+ sum[0] = av_add_sf(sum[0], sbr->e_origmapped[e][m]);
+ sum[1] = av_add_sf(sum[1], sbr->e_curr[e][m]);
+ }
+ gain_max = av_mul_sf(limgain[sbr->bs_limiter_gains],
+ av_sqrt_sf(
+ av_div_sf(
+ av_add_sf(FLOAT_EPSILON, sum[0]),
+ av_add_sf(FLOAT_EPSILON, sum[1]))));
+ if (av_gt_sf(gain_max, FLOAT_100000))
+ gain_max = FLOAT_100000;
+ for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+ SoftFloat q_m_max = av_div_sf(
+ av_mul_sf(sbr->q_m[e][m], gain_max),
+ sbr->gain[e][m]);
+ if (av_gt_sf(sbr->q_m[e][m], q_m_max))
+ sbr->q_m[e][m] = q_m_max;
+ if (av_gt_sf(sbr->gain[e][m], gain_max))
+ sbr->gain[e][m] = gain_max;
+ }
+ sum[0] = sum[1] = FLOAT_0;
+ for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+ sum[0] = av_add_sf(sum[0], sbr->e_origmapped[e][m]);
+ sum[1] = av_add_sf(sum[1],
+ av_mul_sf(
+ av_mul_sf(sbr->e_curr[e][m],
+ sbr->gain[e][m]),
+ sbr->gain[e][m]));
+ sum[1] = av_add_sf(sum[1],
+ av_mul_sf(sbr->s_m[e][m], sbr->s_m[e][m]));
+ if (delta && !sbr->s_m[e][m].mant)
+ sum[1] = av_add_sf(sum[1],
+ av_mul_sf(sbr->q_m[e][m], sbr->q_m[e][m]));
+ }
+ gain_boost = av_sqrt_sf(
+ av_div_sf(
+ av_add_sf(FLOAT_EPSILON, sum[0]),
+ av_add_sf(FLOAT_EPSILON, sum[1])));
+ if (av_gt_sf(gain_boost, FLOAT_1584893192))
+ gain_boost = FLOAT_1584893192;
+
+ for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
+ sbr->gain[e][m] = av_mul_sf(sbr->gain[e][m], gain_boost);
+ sbr->q_m[e][m] = av_mul_sf(sbr->q_m[e][m], gain_boost);
+ sbr->s_m[e][m] = av_mul_sf(sbr->s_m[e][m], gain_boost);
+ }
+ }
+ }
+}
+
+/// Assembling HF Signals (14496-3 sp04 p220)
+static void sbr_hf_assemble(int Y1[38][64][2],
+ const int X_high[64][40][2],
+ SpectralBandReplication *sbr, SBRData *ch_data,
+ const int e_a[2])
+{
+ int e, i, j, m;
+ const int h_SL = 4 * !sbr->bs_smoothing_mode;
+ const int kx = sbr->kx[1];
+ const int m_max = sbr->m[1];
+ static const SoftFloat h_smooth[5] = {
+ { 715827883, -1 },
+ { 647472402, -1 },
+ { 937030863, -2 },
+ { 989249804, -3 },
+ { 546843842, -4 },
+ };
+ SoftFloat (*g_temp)[48] = ch_data->g_temp, (*q_temp)[48] = ch_data->q_temp;
+ int indexnoise = ch_data->f_indexnoise;
+ int indexsine = ch_data->f_indexsine;
+
+ if (sbr->reset) {
+ for (i = 0; i < h_SL; i++) {
+ memcpy(g_temp[i + 2*ch_data->t_env[0]], sbr->gain[0], m_max * sizeof(sbr->gain[0][0]));
+ memcpy(q_temp[i + 2*ch_data->t_env[0]], sbr->q_m[0], m_max * sizeof(sbr->q_m[0][0]));
+ }
+ } else if (h_SL) {
+ for (i = 0; i < 4; i++) {
+ memcpy(g_temp[i + 2 * ch_data->t_env[0]],
+ g_temp[i + 2 * ch_data->t_env_num_env_old],
+ sizeof(g_temp[0]));
+ memcpy(q_temp[i + 2 * ch_data->t_env[0]],
+ q_temp[i + 2 * ch_data->t_env_num_env_old],
+ sizeof(q_temp[0]));
+ }
+ }
+
+ for (e = 0; e < ch_data->bs_num_env; e++) {
+ for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
+ memcpy(g_temp[h_SL + i], sbr->gain[e], m_max * sizeof(sbr->gain[0][0]));
+ memcpy(q_temp[h_SL + i], sbr->q_m[e], m_max * sizeof(sbr->q_m[0][0]));
+ }
+ }
+
+ for (e = 0; e < ch_data->bs_num_env; e++) {
+ for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
+ SoftFloat g_filt_tab[48];
+ SoftFloat q_filt_tab[48];
+ SoftFloat *g_filt, *q_filt;
+
+ if (h_SL && e != e_a[0] && e != e_a[1]) {
+ g_filt = g_filt_tab;
+ q_filt = q_filt_tab;
+ for (m = 0; m < m_max; m++) {
+ const int idx1 = i + h_SL;
+ g_filt[m].mant = g_filt[m].exp = 0;
+ q_filt[m].mant = q_filt[m].exp = 0;
+ for (j = 0; j <= h_SL; j++) {
+ g_filt[m] = av_add_sf(g_filt[m],
+ av_mul_sf(g_temp[idx1 - j][m],
+ h_smooth[j]));
+ q_filt[m] = av_add_sf(q_filt[m],
+ av_mul_sf(q_temp[idx1 - j][m],
+ h_smooth[j]));
+ }
+ }
+ } else {
+ g_filt = g_temp[i + h_SL];
+ q_filt = q_temp[i];
+ }
+
+ sbr->dsp.hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
+ i + ENVELOPE_ADJUSTMENT_OFFSET);
+
+ if (e != e_a[0] && e != e_a[1]) {
+ sbr->dsp.hf_apply_noise[indexsine](Y1[i] + kx, sbr->s_m[e],
+ q_filt, indexnoise,
+ kx, m_max);
+ } else {
+ int idx = indexsine&1;
+ int A = (1-((indexsine+(kx & 1))&2));
+ int B = (A^(-idx)) + idx;
+ int *out = &Y1[i][kx][idx];
+ int shift, round;
+
+ SoftFloat *in = sbr->s_m[e];
+ for (m = 0; m+1 < m_max; m+=2) {
+ shift = 22 - in[m ].exp;
+ round = 1 << (shift-1);
+ out[2*m ] += (in[m ].mant * A + round) >> shift;
+
+ shift = 22 - in[m+1].exp;
+ round = 1 << (shift-1);
+ out[2*m+2] += (in[m+1].mant * B + round) >> shift;
+ }
+ if(m_max&1)
+ {
+ shift = 22 - in[m ].exp;
+ round = 1 << (shift-1);
+
+ out[2*m ] += (in[m ].mant * A + round) >> shift;
+ }
+ }
+ indexnoise = (indexnoise + m_max) & 0x1ff;
+ indexsine = (indexsine + 1) & 3;
+ }
+ }
+ ch_data->f_indexnoise = indexnoise;
+ ch_data->f_indexsine = indexsine;
+}
+
+#include "aacsbr_template.c"
diff --git a/libavcodec/aacsbr_template.c b/libavcodec/aacsbr_template.c
index 8e81230..e40318f 100644
--- a/libavcodec/aacsbr_template.c
+++ b/libavcodec/aacsbr_template.c
@@ -3,6 +3,10 @@
* Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
* Copyright (c) 2009-2010 Alex Converse <alex.converse at gmail.com>
*
+ * Fixed point code
+ * Copyright (c) 2013
+ * MIPS Technologies, Inc., California.
+ *
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
@@ -24,9 +28,11 @@
* @file
* AAC Spectral Band Replication decoding functions
* @author Robert Swain ( rob opendot cl )
+ * @author Stanislav Ocovaj ( stanislav.ocovaj at imgtec.com )
+ * @author Zoran Basaric ( zoran.basaric at imgtec.com )
*/
-av_cold void ff_aac_sbr_init(void)
+av_cold void AAC_RENAME(ff_aac_sbr_init)(void)
{
static const struct {
const void *sbr_codes, *sbr_bits;
@@ -72,7 +78,7 @@ static void sbr_turnoff(SpectralBandReplication *sbr) {
memset(&sbr->spectrum_params, -1, sizeof(SpectrumParameters));
}
-av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
+av_cold void AAC_RENAME(ff_aac_sbr_ctx_init)(AACContext *ac, SpectralBandReplication *sbr)
{
if(sbr->mdct.mdct_bits)
return;
@@ -83,17 +89,17 @@ av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
/* SBR requires samples to be scaled to +/-32768.0 to work correctly.
* mdct scale factors are adjusted to scale up from +/-1.0 at analysis
* and scale back down at synthesis. */
- ff_mdct_init(&sbr->mdct, 7, 1, 1.0 / (64 * 32768.0));
- ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * 32768.0);
+ AAC_RENAME_32(ff_mdct_init)(&sbr->mdct, 7, 1, 1.0 / (64 * 32768.0));
+ AAC_RENAME_32(ff_mdct_init)(&sbr->mdct_ana, 7, 1, -2.0 * 32768.0);
ff_ps_ctx_init(&sbr->ps);
- ff_sbrdsp_init(&sbr->dsp);
+ AAC_RENAME(ff_sbrdsp_init)(&sbr->dsp);
aacsbr_func_ptr_init(&sbr->c);
}
-av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
+av_cold void AAC_RENAME(ff_aac_sbr_ctx_close)(SpectralBandReplication *sbr)
{
- ff_mdct_end(&sbr->mdct);
- ff_mdct_end(&sbr->mdct_ana);
+ AAC_RENAME_32(ff_mdct_end)(&sbr->mdct);
+ AAC_RENAME_32(ff_mdct_end)(&sbr->mdct_ana);
}
static int qsort_comparison_function_int16(const void *a, const void *b)
@@ -115,10 +121,10 @@ static void sbr_make_f_tablelim(SpectralBandReplication *sbr)
{
int k;
if (sbr->bs_limiter_bands > 0) {
- static const float bands_warped[3] = { 1.32715174233856803909f, //2^(0.49/1.2)
- 1.18509277094158210129f, //2^(0.49/2)
- 1.11987160404675912501f }; //2^(0.49/3)
- const float lim_bands_per_octave_warped = bands_warped[sbr->bs_limiter_bands - 1];
+ static const INTFLOAT bands_warped[3] = { Q23(1.32715174233856803909f), //2^(0.49/1.2)
+ Q23(1.18509277094158210129f), //2^(0.49/2)
+ Q23(1.11987160404675912501f) }; //2^(0.49/3)
+ const INTFLOAT lim_bands_per_octave_warped = bands_warped[sbr->bs_limiter_bands - 1];
int16_t patch_borders[7];
uint16_t *in = sbr->f_tablelim + 1, *out = sbr->f_tablelim;
@@ -138,7 +144,11 @@ static void sbr_make_f_tablelim(SpectralBandReplication *sbr)
sbr->n_lim = sbr->n[0] + sbr->num_patches - 1;
while (out < sbr->f_tablelim + sbr->n_lim) {
+#if USE_FIXED
+ if ((*in << 23) >= *out * lim_bands_per_octave_warped) {
+#else
if (*in >= *out * lim_bands_per_octave_warped) {
+#endif /* USE_FIXED */
*++out = *in++;
} else if (*in == *out ||
!in_table_int16(patch_borders, sbr->num_patches, *in)) {
@@ -344,6 +354,9 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
int two_regions, num_bands_0;
int vdk0_max, vdk1_min;
int16_t vk0[49];
+#if USE_FIXED
+ int tmp, nz = 0;
+#endif /* USE_FIXED */
if (49 * sbr->k[2] > 110 * sbr->k[0]) {
two_regions = 1;
@@ -353,7 +366,19 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
sbr->k[1] = sbr->k[2];
}
+#if USE_FIXED
+ tmp = (sbr->k[1] << 23) / sbr->k[0];
+ while (tmp < 0x40000000) {
+ tmp <<= 1;
+ nz++;
+ }
+ tmp = fixed_log(tmp - 0x80000000);
+ tmp = (int)(((int64_t)tmp * CONST_RECIP_LN2 + 0x20000000) >> 30);
+ tmp = (((tmp + 0x80) >> 8) + ((8 - nz) << 23)) * half_bands;
+ num_bands_0 = ((tmp + 0x400000) >> 23) * 2;
+#else
num_bands_0 = lrintf(half_bands * log2f(sbr->k[1] / (float)sbr->k[0])) * 2;
+#endif /* USE_FIXED */
if (num_bands_0 <= 0) { // Requirements (14496-3 sp04 p205)
av_log(ac->avctx, AV_LOG_ERROR, "Invalid num_bands_0: %d\n", num_bands_0);
@@ -378,11 +403,27 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
if (two_regions) {
int16_t vk1[49];
+#if USE_FIXED
+ int num_bands_1;
+
+ tmp = (sbr->k[2] << 23) / sbr->k[1];
+ nz = 0;
+ while (tmp < 0x40000000) {
+ tmp <<= 1;
+ nz++;
+ }
+ tmp = fixed_log(tmp - 0x80000000);
+ tmp = (int)(((int64_t)tmp * CONST_RECIP_LN2 + 0x20000000) >> 30);
+ tmp = (((tmp + 0x80) >> 8) + ((8 - nz) << 23)) * half_bands;
+ if (spectrum->bs_alter_scale)
+ tmp = (int)(((int64_t)tmp * CONST_076923 + 0x40000000) >> 31);
+ num_bands_1 = ((tmp + 0x400000) >> 23) * 2;
+#else
float invwarp = spectrum->bs_alter_scale ? 0.76923076923076923077f
: 1.0f; // bs_alter_scale = {0,1}
int num_bands_1 = lrintf(half_bands * invwarp *
log2f(sbr->k[2] / (float)sbr->k[1])) * 2;
-
+#endif /* USE_FIXED */
make_bands(vk1+1, sbr->k[1], sbr->k[2], num_bands_1);
vdk1_min = array_min_int16(vk1 + 1, num_bands_1);
@@ -487,6 +528,9 @@ static int sbr_hf_calc_npatches(AACContext *ac, SpectralBandReplication *sbr)
static int sbr_make_f_derived(AACContext *ac, SpectralBandReplication *sbr)
{
int k, temp;
+#if USE_FIXED
+ int nz = 0;
+#endif /* USE_FIXED */
sbr->n[1] = sbr->n_master - sbr->spectrum_params.bs_xover_band;
sbr->n[0] = (sbr->n[1] + 1) >> 1;
@@ -511,9 +555,24 @@ static int sbr_make_f_derived(AACContext *ac, SpectralBandReplication *sbr)
temp = sbr->n[1] & 1;
for (k = 1; k <= sbr->n[0]; k++)
sbr->f_tablelow[k] = sbr->f_tablehigh[2 * k - temp];
+#if USE_FIXED
+ temp = (sbr->k[2] << 23) / sbr->kx[1];
+ while (temp < 0x40000000) {
+ temp <<= 1;
+ nz++;
+ }
+ temp = fixed_log(temp - 0x80000000);
+ temp = (int)(((int64_t)temp * CONST_RECIP_LN2 + 0x20000000) >> 30);
+ temp = (((temp + 0x80) >> 8) + ((8 - nz) << 23)) * sbr->spectrum_params.bs_noise_bands;
+ sbr->n_q = (temp + 0x400000) >> 23;
+ if (sbr->n_q < 1)
+ sbr->n_q = 1;
+#else
sbr->n_q = FFMAX(1, lrintf(sbr->spectrum_params.bs_noise_bands *
log2f(sbr->k[2] / (float)sbr->kx[1]))); // 0 <= bs_noise_bands <= 3
+#endif /* USE_FIXED */
+
if (sbr->n_q > 5) {
av_log(ac->avctx, AV_LOG_ERROR, "Too many noise floor scale factors: %d\n", sbr->n_q);
return -1;
@@ -770,6 +829,31 @@ static void read_sbr_envelope(SpectralBandReplication *sbr, GetBitContext *gb,
}
}
+#if USE_FIXED
+ for (i = 0; i < ch_data->bs_num_env; i++) {
+ if (ch_data->bs_df_env[i]) {
+ // bs_freq_res[0] == bs_freq_res[bs_num_env] from prev frame
+ if (ch_data->bs_freq_res[i + 1] == ch_data->bs_freq_res[i]) {
+ for (j = 0; j < sbr->n[ch_data->bs_freq_res[i + 1]]; j++)
+ ch_data->env_facs[i + 1][j].mant = ch_data->env_facs[i][j].mant + delta * (get_vlc2(gb, t_huff, 9, 3) - t_lav);
+ } else if (ch_data->bs_freq_res[i + 1]) {
+ for (j = 0; j < sbr->n[ch_data->bs_freq_res[i + 1]]; j++) {
+ k = (j + odd) >> 1; // find k such that f_tablelow[k] <= f_tablehigh[j] < f_tablelow[k + 1]
+ ch_data->env_facs[i + 1][j].mant = ch_data->env_facs[i][k].mant + delta * (get_vlc2(gb, t_huff, 9, 3) - t_lav);
+ }
+ } else {
+ for (j = 0; j < sbr->n[ch_data->bs_freq_res[i + 1]]; j++) {
+ k = j ? 2*j - odd : 0; // find k such that f_tablehigh[k] == f_tablelow[j]
+ ch_data->env_facs[i + 1][j].mant = ch_data->env_facs[i][k].mant + delta * (get_vlc2(gb, t_huff, 9, 3) - t_lav);
+ }
+ }
+ } else {
+ ch_data->env_facs[i + 1][0].mant = delta * get_bits(gb, bits); // bs_env_start_value_balance
+ for (j = 1; j < sbr->n[ch_data->bs_freq_res[i + 1]]; j++)
+ ch_data->env_facs[i + 1][j].mant = ch_data->env_facs[i + 1][j - 1].mant + delta * (get_vlc2(gb, f_huff, 9, 3) - f_lav);
+ }
+ }
+#else
for (i = 0; i < ch_data->bs_num_env; i++) {
if (ch_data->bs_df_env[i]) {
// bs_freq_res[0] == bs_freq_res[bs_num_env] from prev frame
@@ -793,6 +877,7 @@ static void read_sbr_envelope(SpectralBandReplication *sbr, GetBitContext *gb,
ch_data->env_facs[i + 1][j] = ch_data->env_facs[i + 1][j - 1] + delta * (get_vlc2(gb, f_huff, 9, 3) - f_lav);
}
}
+#endif /* USE_FIXED */
//assign 0th elements of env_facs from last elements
memcpy(ch_data->env_facs[0], ch_data->env_facs[ch_data->bs_num_env],
@@ -819,6 +904,18 @@ static void read_sbr_noise(SpectralBandReplication *sbr, GetBitContext *gb,
f_lav = vlc_sbr_lav[F_HUFFMAN_ENV_3_0DB];
}
+#if USE_FIXED
+ for (i = 0; i < ch_data->bs_num_noise; i++) {
+ if (ch_data->bs_df_noise[i]) {
+ for (j = 0; j < sbr->n_q; j++)
+ ch_data->noise_facs[i + 1][j].mant = ch_data->noise_facs[i][j].mant + delta * (get_vlc2(gb, t_huff, 9, 2) - t_lav);
+ } else {
+ ch_data->noise_facs[i + 1][0].mant = delta * get_bits(gb, 5); // bs_noise_start_value_balance or bs_noise_start_value_level
+ for (j = 1; j < sbr->n_q; j++)
+ ch_data->noise_facs[i + 1][j].mant = ch_data->noise_facs[i + 1][j - 1].mant + delta * (get_vlc2(gb, f_huff, 9, 3) - f_lav);
+ }
+ }
+#else
for (i = 0; i < ch_data->bs_num_noise; i++) {
if (ch_data->bs_df_noise[i]) {
for (j = 0; j < sbr->n_q; j++)
@@ -829,6 +926,7 @@ static void read_sbr_noise(SpectralBandReplication *sbr, GetBitContext *gb,
ch_data->noise_facs[i + 1][j] = ch_data->noise_facs[i + 1][j - 1] + delta * (get_vlc2(gb, f_huff, 9, 3) - f_lav);
}
}
+#endif /* USE_FIXED */
//assign 0th elements of noise_facs from last elements
memcpy(ch_data->noise_facs[0], ch_data->noise_facs[ch_data->bs_num_noise],
@@ -990,7 +1088,7 @@ static void sbr_reset(AACContext *ac, SpectralBandReplication *sbr)
*
* @return Returns number of bytes consumed from the TYPE_FIL element.
*/
-int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
+int AAC_RENAME(ff_decode_sbr_extension)(AACContext *ac, SpectralBandReplication *sbr,
GetBitContext *gb_host, int crc, int cnt, int id_aac)
{
unsigned int num_sbr_bits = 0, num_align_bits;
@@ -1042,9 +1140,13 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
* @param W array of complex-valued samples split into subbands
*/
#ifndef sbr_qmf_analysis
+#if USE_FIXED
+static void sbr_qmf_analysis(AVFixedDSPContext *dsp, FFTContext *mdct,
+#else
static void sbr_qmf_analysis(AVFloatDSPContext *dsp, FFTContext *mdct,
- SBRDSPContext *sbrdsp, const float *in, float *x,
- float z[320], float W[2][32][32][2], int buf_idx)
+#endif /* USE_FIXED */
+ SBRDSPContext *sbrdsp, const INTFLOAT *in, INTFLOAT *x,
+ INTFLOAT z[320], INTFLOAT W[2][32][32][2], int buf_idx)
{
int i;
memcpy(x , x+1024, (320-32)*sizeof(x[0]));
@@ -1067,19 +1169,23 @@ static void sbr_qmf_analysis(AVFloatDSPContext *dsp, FFTContext *mdct,
*/
#ifndef sbr_qmf_synthesis
static void sbr_qmf_synthesis(FFTContext *mdct,
+#if USE_FIXED
+ SBRDSPContext *sbrdsp, AVFixedDSPContext *dsp,
+#else
SBRDSPContext *sbrdsp, AVFloatDSPContext *dsp,
- float *out, float X[2][38][64],
- float mdct_buf[2][64],
- float *v0, int *v_off, const unsigned int div)
+#endif /* USE_FIXED */
+ INTFLOAT *out, INTFLOAT X[2][38][64],
+ INTFLOAT mdct_buf[2][64],
+ INTFLOAT *v0, int *v_off, const unsigned int div)
{
int i, n;
- const float *sbr_qmf_window = div ? sbr_qmf_window_ds : sbr_qmf_window_us;
+ const INTFLOAT *sbr_qmf_window = div ? sbr_qmf_window_ds : sbr_qmf_window_us;
const int step = 128 >> div;
- float *v;
+ INTFLOAT *v;
for (i = 0; i < 32; i++) {
if (*v_off < step) {
int saved_samples = (1280 - 128) >> div;
- memcpy(&v0[SBR_SYNTHESIS_BUF_SIZE - saved_samples], v0, saved_samples * sizeof(float));
+ memcpy(&v0[SBR_SYNTHESIS_BUF_SIZE - saved_samples], v0, saved_samples * sizeof(INTFLOAT));
*v_off = SBR_SYNTHESIS_BUF_SIZE - saved_samples - step;
} else {
*v_off -= step;
@@ -1115,7 +1221,7 @@ static void sbr_qmf_synthesis(FFTContext *mdct,
/// Generate the subband filtered lowband
static int sbr_lf_gen(AACContext *ac, SpectralBandReplication *sbr,
- float X_low[32][40][2], const float W[2][32][32][2],
+ INTFLOAT X_low[32][40][2], const INTFLOAT W[2][32][32][2],
int buf_idx)
{
int i, k;
@@ -1140,9 +1246,9 @@ static int sbr_lf_gen(AACContext *ac, SpectralBandReplication *sbr,
/// High Frequency Generator (14496-3 sp04 p215)
static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
- float X_high[64][40][2], const float X_low[32][40][2],
- const float (*alpha0)[2], const float (*alpha1)[2],
- const float bw_array[5], const uint8_t *t_env,
+ INTFLOAT X_high[64][40][2], const INTFLOAT X_low[32][40][2],
+ const INTFLOAT (*alpha0)[2], const INTFLOAT (*alpha1)[2],
+ const INTFLOAT bw_array[5], const uint8_t *t_env,
int bs_num_env)
{
int j, x;
@@ -1174,9 +1280,9 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
}
/// Generate the subband filtered lowband
-static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
- const float Y0[38][64][2], const float Y1[38][64][2],
- const float X_low[32][40][2], int ch)
+static int sbr_x_gen(SpectralBandReplication *sbr, INTFLOAT X[2][38][64],
+ const INTFLOAT Y0[38][64][2], const INTFLOAT Y1[38][64][2],
+ const INTFLOAT X_low[32][40][2], int ch)
{
int k, i;
const int i_f = 32;
@@ -1268,7 +1374,7 @@ static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
}
/// Estimation of current envelope (14496-3 sp04 p218)
-static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
+static void sbr_env_estimate(AAC_FLOAT (*e_curr)[48], INTFLOAT X_high[64][40][2],
SpectralBandReplication *sbr, SBRData *ch_data)
{
int e, m;
@@ -1276,13 +1382,21 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
if (sbr->bs_interpol_freq) {
for (e = 0; e < ch_data->bs_num_env; e++) {
+#if USE_FIXED
+ const SoftFloat recip_env_size = av_int2sf(0x20000000 / (ch_data->t_env[e + 1] - ch_data->t_env[e]), 30);
+#else
const float recip_env_size = 0.5f / (ch_data->t_env[e + 1] - ch_data->t_env[e]);
+#endif /* USE_FIXED */
int ilb = ch_data->t_env[e] * 2 + ENVELOPE_ADJUSTMENT_OFFSET;
int iub = ch_data->t_env[e + 1] * 2 + ENVELOPE_ADJUSTMENT_OFFSET;
for (m = 0; m < sbr->m[1]; m++) {
- float sum = sbr->dsp.sum_square(X_high[m+kx1] + ilb, iub - ilb);
+ AAC_FLOAT sum = sbr->dsp.sum_square(X_high[m+kx1] + ilb, iub - ilb);
+#if USE_FIXED
+ e_curr[e][m] = av_mul_sf(sum, recip_env_size);
+#else
e_curr[e][m] = sum * recip_env_size;
+#endif /* USE_FIXED */
}
}
} else {
@@ -1295,6 +1409,14 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
const uint16_t *table = ch_data->bs_freq_res[e + 1] ? sbr->f_tablehigh : sbr->f_tablelow;
for (p = 0; p < sbr->n[ch_data->bs_freq_res[e + 1]]; p++) {
+#if USE_FIXED
+ SoftFloat sum = { 0, 0 };
+ const SoftFloat den = av_int2sf(0x20000000 / (env_size * (table[p + 1] - table[p])), 29);
+ for (k = table[p]; k < table[p + 1]; k++) {
+ sum = av_add_sf(sum, sbr->dsp.sum_square(X_high[k] + ilb, iub - ilb));
+ }
+ sum = av_mul_sf(sum, den);
+#else
float sum = 0.0f;
const int den = env_size * (table[p + 1] - table[p]);
@@ -1302,6 +1424,7 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
sum += sbr->dsp.sum_square(X_high[k] + ilb, iub - ilb);
}
sum /= den;
+#endif /* USE_FIXED */
for (k = table[p]; k < table[p + 1]; k++) {
e_curr[e][k - kx1] = sum;
}
@@ -1310,8 +1433,8 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
}
}
-void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
- float* L, float* R)
+void AAC_RENAME(ff_sbr_apply)(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
+ INTFLOAT* L, INTFLOAT* R)
{
int downsampled = ac->oc[1].m4ac.ext_sample_rate < sbr->sample_rate;
int ch;
@@ -1331,20 +1454,20 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
for (ch = 0; ch < nch; ch++) {
/* decode channel */
sbr_qmf_analysis(ac->fdsp, &sbr->mdct_ana, &sbr->dsp, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
- (float*)sbr->qmf_filter_scratch,
+ (INTFLOAT*)sbr->qmf_filter_scratch,
sbr->data[ch].W, sbr->data[ch].Ypos);
sbr->c.sbr_lf_gen(ac, sbr, sbr->X_low,
- (const float (*)[32][32][2]) sbr->data[ch].W,
+ (const INTFLOAT (*)[32][32][2]) sbr->data[ch].W,
sbr->data[ch].Ypos);
sbr->data[ch].Ypos ^= 1;
if (sbr->start) {
sbr->c.sbr_hf_inverse_filter(&sbr->dsp, sbr->alpha0, sbr->alpha1,
- (const float (*)[40][2]) sbr->X_low, sbr->k[0]);
+ (const INTFLOAT (*)[40][2]) sbr->X_low, sbr->k[0]);
sbr_chirp(sbr, &sbr->data[ch]);
sbr_hf_gen(ac, sbr, sbr->X_high,
- (const float (*)[40][2]) sbr->X_low,
- (const float (*)[2]) sbr->alpha0,
- (const float (*)[2]) sbr->alpha1,
+ (const INTFLOAT (*)[40][2]) sbr->X_low,
+ (const INTFLOAT (*)[2]) sbr->alpha0,
+ (const INTFLOAT (*)[2]) sbr->alpha1,
sbr->data[ch].bw_array, sbr->data[ch].t_env,
sbr->data[ch].bs_num_env);
@@ -1354,7 +1477,7 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
sbr_env_estimate(sbr->e_curr, sbr->X_high, sbr, &sbr->data[ch]);
sbr_gain_calc(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
sbr->c.sbr_hf_assemble(sbr->data[ch].Y[sbr->data[ch].Ypos],
- (const float (*)[40][2]) sbr->X_high,
+ (const INTFLOAT (*)[40][2]) sbr->X_high,
sbr, &sbr->data[ch],
sbr->data[ch].e_a);
}
@@ -1362,9 +1485,9 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
/* synthesis */
sbr->c.sbr_x_gen(sbr, sbr->X[ch],
- (const float (*)[64][2]) sbr->data[ch].Y[1-sbr->data[ch].Ypos],
- (const float (*)[64][2]) sbr->data[ch].Y[ sbr->data[ch].Ypos],
- (const float (*)[40][2]) sbr->X_low, ch);
+ (const INTFLOAT (*)[64][2]) sbr->data[ch].Y[1-sbr->data[ch].Ypos],
+ (const INTFLOAT (*)[64][2]) sbr->data[ch].Y[ sbr->data[ch].Ypos],
+ (const INTFLOAT (*)[40][2]) sbr->X_low, ch);
}
if (ac->oc[1].m4ac.ps == 1) {
@@ -1396,6 +1519,8 @@ static void aacsbr_func_ptr_init(AACSBRContext *c)
c->sbr_x_gen = sbr_x_gen;
c->sbr_hf_inverse_filter = sbr_hf_inverse_filter;
+#if !USE_FIXED
if(ARCH_MIPS)
ff_aacsbr_func_ptr_init_mips(c);
+#endif
}
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/sbr.h b/libavcodec/sbr.h
index e28fccd..f78a37d 100644
--- a/libavcodec/sbr.h
+++ b/libavcodec/sbr.h
@@ -66,9 +66,9 @@ typedef struct SBRData {
*/
unsigned bs_frame_class;
unsigned bs_add_harmonic_flag;
- unsigned bs_num_env;
+ AAC_SIGNE bs_num_env;
uint8_t bs_freq_res[7];
- unsigned bs_num_noise;
+ AAC_SIGNE bs_num_noise;
uint8_t bs_df_env[5];
uint8_t bs_df_noise[2];
uint8_t bs_invf_mode[2][5];
@@ -80,25 +80,25 @@ typedef struct SBRData {
* @name State variables
* @{
*/
- DECLARE_ALIGNED(32, float, synthesis_filterbank_samples)[SBR_SYNTHESIS_BUF_SIZE];
- DECLARE_ALIGNED(32, float, analysis_filterbank_samples) [1312];
+ DECLARE_ALIGNED(32, INTFLOAT, synthesis_filterbank_samples)[SBR_SYNTHESIS_BUF_SIZE];
+ DECLARE_ALIGNED(32, INTFLOAT, analysis_filterbank_samples) [1312];
int synthesis_filterbank_samples_offset;
///l_APrev and l_A
int e_a[2];
///Chirp factors
- float bw_array[5];
+ INTFLOAT bw_array[5];
///QMF values of the original signal
- float W[2][32][32][2];
+ INTFLOAT W[2][32][32][2];
///QMF output of the HF adjustor
int Ypos;
- DECLARE_ALIGNED(16, float, Y)[2][38][64][2];
- DECLARE_ALIGNED(16, float, g_temp)[42][48];
- float q_temp[42][48];
+ DECLARE_ALIGNED(16, INTFLOAT, Y)[2][38][64][2];
+ DECLARE_ALIGNED(16, AAC_FLOAT, g_temp)[42][48];
+ AAC_FLOAT q_temp[42][48];
uint8_t s_indexmapped[8][48];
///Envelope scalefactors
- float env_facs[6][48];
+ AAC_FLOAT env_facs[6][48];
///Noise scalefactors
- float noise_facs[3][5];
+ AAC_FLOAT noise_facs[3][5];
///Envelope time borders
uint8_t t_env[8];
///Envelope time border of the last envelope of the previous frame
@@ -117,18 +117,18 @@ typedef struct SpectralBandReplication SpectralBandReplication;
*/
typedef struct AACSBRContext {
int (*sbr_lf_gen)(AACContext *ac, SpectralBandReplication *sbr,
- float X_low[32][40][2], const float W[2][32][32][2],
+ INTFLOAT X_low[32][40][2], const INTFLOAT W[2][32][32][2],
int buf_idx);
- void (*sbr_hf_assemble)(float Y1[38][64][2],
- const float X_high[64][40][2],
+ void (*sbr_hf_assemble)(INTFLOAT Y1[38][64][2],
+ const INTFLOAT X_high[64][40][2],
SpectralBandReplication *sbr, SBRData *ch_data,
const int e_a[2]);
- int (*sbr_x_gen)(SpectralBandReplication *sbr, float X[2][38][64],
- const float Y0[38][64][2], const float Y1[38][64][2],
- const float X_low[32][40][2], int ch);
+ int (*sbr_x_gen)(SpectralBandReplication *sbr, INTFLOAT X[2][38][64],
+ const INTFLOAT Y0[38][64][2], const INTFLOAT Y1[38][64][2],
+ const INTFLOAT X_low[32][40][2], int ch);
void (*sbr_hf_inverse_filter)(SBRDSPContext *dsp,
- float (*alpha0)[2], float (*alpha1)[2],
- const float X_low[32][40][2], int k0);
+ INTFLOAT (*alpha0)[2], INTFLOAT (*alpha1)[2],
+ const INTFLOAT X_low[32][40][2], int k0);
} AACSBRContext;
/**
@@ -150,23 +150,23 @@ struct SpectralBandReplication {
unsigned bs_smoothing_mode;
/** @} */
unsigned bs_coupling;
- unsigned k[5]; ///< k0, k1, k2
+ AAC_SIGNE k[5]; ///< k0, k1, k2
///kx', and kx respectively, kx is the first QMF subband where SBR is used.
///kx' is its value from the previous frame
- unsigned kx[2];
+ AAC_SIGNE kx[2];
///M' and M respectively, M is the number of QMF subbands that use SBR.
- unsigned m[2];
+ AAC_SIGNE m[2];
unsigned kx_and_m_pushed;
///The number of frequency bands in f_master
- unsigned n_master;
+ AAC_SIGNE n_master;
SBRData data[2];
PSContext ps;
///N_Low and N_High respectively, the number of frequency bands for low and high resolution
- unsigned n[2];
+ AAC_SIGNE n[2];
///Number of noise floor bands
- unsigned n_q;
+ AAC_SIGNE n_q;
///Number of limiter bands
- unsigned n_lim;
+ AAC_SIGNE n_lim;
///The master QMF frequency grouping
uint16_t f_master[49];
///Frequency borders for low resolution SBR
@@ -177,33 +177,33 @@ struct SpectralBandReplication {
uint16_t f_tablenoise[6];
///Frequency borders for the limiter
uint16_t f_tablelim[30];
- unsigned num_patches;
+ AAC_SIGNE num_patches;
uint8_t patch_num_subbands[6];
uint8_t patch_start_subband[6];
///QMF low frequency input to the HF generator
- DECLARE_ALIGNED(16, float, X_low)[32][40][2];
+ DECLARE_ALIGNED(16, INTFLOAT, X_low)[32][40][2];
///QMF output of the HF generator
- DECLARE_ALIGNED(16, float, X_high)[64][40][2];
+ DECLARE_ALIGNED(16, INTFLOAT, X_high)[64][40][2];
///QMF values of the reconstructed signal
- DECLARE_ALIGNED(16, float, X)[2][2][38][64];
+ DECLARE_ALIGNED(16, INTFLOAT, X)[2][2][38][64];
///Zeroth coefficient used to filter the subband signals
- DECLARE_ALIGNED(16, float, alpha0)[64][2];
+ DECLARE_ALIGNED(16, INTFLOAT, alpha0)[64][2];
///First coefficient used to filter the subband signals
- DECLARE_ALIGNED(16, float, alpha1)[64][2];
+ DECLARE_ALIGNED(16, INTFLOAT, alpha1)[64][2];
///Dequantized envelope scalefactors, remapped
- float e_origmapped[7][48];
+ AAC_FLOAT e_origmapped[7][48];
///Dequantized noise scalefactors, remapped
- float q_mapped[7][48];
+ AAC_FLOAT q_mapped[7][48];
///Sinusoidal presence, remapped
uint8_t s_mapped[7][48];
///Estimated envelope
- float e_curr[7][48];
+ AAC_FLOAT e_curr[7][48];
///Amplitude adjusted noise scalefactors
- float q_m[7][48];
+ AAC_FLOAT q_m[7][48];
///Sinusoidal levels
- float s_m[7][48];
- float gain[7][48];
- DECLARE_ALIGNED(32, float, qmf_filter_scratch)[5][64];
+ AAC_FLOAT s_m[7][48];
+ AAC_FLOAT gain[7][48];
+ DECLARE_ALIGNED(32, INTFLOAT, qmf_filter_scratch)[5][64];
FFTContext mdct_ana;
FFTContext mdct;
SBRDSPContext dsp;
diff --git a/libavcodec/sbrdsp.c b/libavcodec/sbrdsp.c
index aa60de4..cc432b6 100644
--- a/libavcodec/sbrdsp.c
+++ b/libavcodec/sbrdsp.c
@@ -20,6 +20,9 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#define USE_FIXED 0
+
+#include "aac.h"
#include "config.h"
#include "libavutil/attributes.h"
#include "libavutil/intfloat.h"
diff --git a/libavcodec/sbrdsp.h b/libavcodec/sbrdsp.h
index 1c1bcdf..66852de 100644
--- a/libavcodec/sbrdsp.h
+++ b/libavcodec/sbrdsp.h
@@ -22,29 +22,31 @@
#define AVCODEC_SBRDSP_H
#include <stdint.h>
+#include "aac_defines.h"
+#include "libavutil/softfloat.h"
typedef struct SBRDSPContext {
- void (*sum64x5)(float *z);
- float (*sum_square)(float (*x)[2], int n);
- void (*neg_odd_64)(float *x);
- void (*qmf_pre_shuffle)(float *z);
- void (*qmf_post_shuffle)(float W[32][2], const float *z);
- void (*qmf_deint_neg)(float *v, const float *src);
- void (*qmf_deint_bfly)(float *v, const float *src0, const float *src1);
- void (*autocorrelate)(const float x[40][2], float phi[3][2][2]);
- void (*hf_gen)(float (*X_high)[2], const float (*X_low)[2],
- const float alpha0[2], const float alpha1[2],
- float bw, int start, int end);
- void (*hf_g_filt)(float (*Y)[2], const float (*X_high)[40][2],
- const float *g_filt, int m_max, intptr_t ixh);
- void (*hf_apply_noise[4])(float (*Y)[2], const float *s_m,
- const float *q_filt, int noise,
+ void (*sum64x5)(INTFLOAT *z);
+ AAC_FLOAT (*sum_square)(INTFLOAT (*x)[2], int n);
+ void (*neg_odd_64)(INTFLOAT *x);
+ void (*qmf_pre_shuffle)(INTFLOAT *z);
+ void (*qmf_post_shuffle)(INTFLOAT W[32][2], const INTFLOAT *z);
+ void (*qmf_deint_neg)(INTFLOAT *v, const INTFLOAT *src);
+ void (*qmf_deint_bfly)(INTFLOAT *v, const INTFLOAT *src0, const INTFLOAT *src1);
+ void (*autocorrelate)(const INTFLOAT x[40][2], AAC_FLOAT phi[3][2][2]);
+ void (*hf_gen)(INTFLOAT (*X_high)[2], const INTFLOAT (*X_low)[2],
+ const INTFLOAT alpha0[2], const INTFLOAT alpha1[2],
+ INTFLOAT bw, int start, int end);
+ void (*hf_g_filt)(INTFLOAT (*Y)[2], const INTFLOAT (*X_high)[40][2],
+ const AAC_FLOAT *g_filt, int m_max, intptr_t ixh);
+ void (*hf_apply_noise[4])(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
+ const AAC_FLOAT *q_filt, int noise,
int kx, int m_max);
} SBRDSPContext;
-extern const float ff_sbr_noise_table[][2];
+extern const INTFLOAT AAC_RENAME(ff_sbr_noise_table)[][2];
-void ff_sbrdsp_init(SBRDSPContext *s);
+void AAC_RENAME(ff_sbrdsp_init)(SBRDSPContext *s);
void ff_sbrdsp_init_arm(SBRDSPContext *s);
void ff_sbrdsp_init_x86(SBRDSPContext *s);
void ff_sbrdsp_init_mips(SBRDSPContext *s);
diff --git a/libavcodec/sbrdsp_fixed.c b/libavcodec/sbrdsp_fixed.c
new file mode 100644
index 0000000..5b7b7a6
--- /dev/null
+++ b/libavcodec/sbrdsp_fixed.c
@@ -0,0 +1,286 @@
+/*
+ * AAC Spectral Band Replication decoding functions
+ * Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
+ * Copyright (c) 2009-2010 Alex Converse <alex.converse at 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
+ *
+ * Note: Rounding-to-nearest used unless otherwise stated
+ *
+ */
+
+#define USE_FIXED 1
+
+#include "aac.h"
+#include "config.h"
+#include "libavutil/attributes.h"
+#include "libavutil/intfloat.h"
+#include "sbrdsp.h"
+
+static SoftFloat sbr_sum_square_c(int (*x)[2], int n)
+{
+ SoftFloat ret;
+ int64_t accu = 0;
+ int i, nz, round;
+
+ for (i = 0; i < n; i += 2) {
+ accu += (int64_t)x[i + 0][0] * x[i + 0][0];
+ accu += (int64_t)x[i + 0][1] * x[i + 0][1];
+ accu += (int64_t)x[i + 1][0] * x[i + 1][0];
+ accu += (int64_t)x[i + 1][1] * x[i + 1][1];
+ }
+
+ i = (int)(accu >> 32);
+ if (i == 0) {
+ nz = 1;
+ } else {
+ nz = 0;
+ while (FFABS(i) < 0x40000000) {
+ i <<= 1;
+ nz++;
+ }
+ nz = 32 - nz;
+ }
+
+ round = 1 << (nz-1);
+ i = (int)((accu + round) >> nz);
+ i >>= 1;
+ ret = av_int2sf(i, 15 - nz);
+
+ return ret;
+}
+
+static void sbr_neg_odd_64_c(int *x)
+{
+ int i;
+ for (i = 1; i < 64; i += 2)
+ x[i] = -x[i];
+}
+
+static void sbr_qmf_pre_shuffle_c(int *z)
+{
+ int k;
+ z[64] = z[0];
+ z[65] = z[1];
+ for (k = 1; k < 32; k++) {
+ z[64+2*k ] = -z[64 - k];
+ z[64+2*k+1] = z[ k + 1];
+ }
+}
+
+static void sbr_qmf_post_shuffle_c(int W[32][2], const int *z)
+{
+ int k;
+ for (k = 0; k < 32; k++) {
+ W[k][0] = -z[63-k];
+ W[k][1] = z[k];
+ }
+}
+
+static void sbr_qmf_deint_neg_c(int *v, const int *src)
+{
+ int i;
+ for (i = 0; i < 32; i++) {
+ v[ i] = ( src[63 - 2*i ] + 0x10) >> 5;
+ v[63 - i] = (-src[63 - 2*i - 1] + 0x10) >> 5;
+ }
+}
+
+static av_always_inline SoftFloat autocorr_calc(int64_t accu)
+{
+ int nz, mant, expo, round;
+ int i = (int)(accu >> 32);
+ if (i == 0) {
+ nz = 1;
+ } else {
+ nz = 0;
+ while (FFABS(i) < 0x40000000) {
+ i <<= 1;
+ nz++;
+ }
+ nz = 32-nz;
+ }
+
+ round = 1 << (nz-1);
+ mant = (int)((accu + round) >> nz);
+ mant = (mant + 0x40)>>7;
+ mant <<= 6;
+ expo = nz + 15;
+ return av_int2sf(mant, 30 - expo);
+}
+
+static av_always_inline void autocorrelate(const int x[40][2], SoftFloat phi[3][2][2], int lag)
+{
+ int i;
+ int64_t real_sum, imag_sum;
+ int64_t accu_re = 0, accu_im = 0;
+
+ if (lag) {
+ for (i = 1; i < 38; i++) {
+ accu_re += (int64_t)x[i][0] * x[i+lag][0];
+ accu_re += (int64_t)x[i][1] * x[i+lag][1];
+ accu_im += (int64_t)x[i][0] * x[i+lag][1];
+ accu_im -= (int64_t)x[i][1] * x[i+lag][0];
+ }
+
+ real_sum = accu_re;
+ imag_sum = accu_im;
+
+ accu_re += (int64_t)x[ 0][0] * x[lag][0];
+ accu_re += (int64_t)x[ 0][1] * x[lag][1];
+ accu_im += (int64_t)x[ 0][0] * x[lag][1];
+ accu_im -= (int64_t)x[ 0][1] * x[lag][0];
+
+ phi[2-lag][1][0] = autocorr_calc(accu_re);
+ phi[2-lag][1][1] = autocorr_calc(accu_im);
+
+ if (lag == 1) {
+ accu_re = real_sum;
+ accu_im = imag_sum;
+ accu_re += (int64_t)x[38][0] * x[39][0];
+ accu_re += (int64_t)x[38][1] * x[39][1];
+ accu_im += (int64_t)x[38][0] * x[39][1];
+ accu_im -= (int64_t)x[38][1] * x[39][0];
+
+ phi[0][0][0] = autocorr_calc(accu_re);
+ phi[0][0][1] = autocorr_calc(accu_im);
+ }
+ } else {
+ for (i = 1; i < 38; i++) {
+ accu_re += (int64_t)x[i][0] * x[i][0];
+ accu_re += (int64_t)x[i][1] * x[i][1];
+ }
+ real_sum = accu_re;
+ accu_re += (int64_t)x[ 0][0] * x[ 0][0];
+ accu_re += (int64_t)x[ 0][1] * x[ 0][1];
+
+ phi[2][1][0] = autocorr_calc(accu_re);
+
+ accu_re = real_sum;
+ accu_re += (int64_t)x[38][0] * x[38][0];
+ accu_re += (int64_t)x[38][1] * x[38][1];
+
+ phi[1][0][0] = autocorr_calc(accu_re);
+ }
+}
+
+static void sbr_autocorrelate_c(const int x[40][2], SoftFloat phi[3][2][2])
+{
+ autocorrelate(x, phi, 0);
+ autocorrelate(x, phi, 1);
+ autocorrelate(x, phi, 2);
+}
+
+static void sbr_hf_gen_c(int (*X_high)[2], const int (*X_low)[2],
+ const int alpha0[2], const int alpha1[2],
+ int bw, int start, int end)
+{
+ int alpha[4];
+ int i;
+ int64_t accu;
+
+ accu = (int64_t)alpha0[0] * bw;
+ alpha[2] = (int)((accu + 0x40000000) >> 31);
+ accu = (int64_t)alpha0[1] * bw;
+ alpha[3] = (int)((accu + 0x40000000) >> 31);
+ accu = (int64_t)bw * bw;
+ bw = (int)((accu + 0x40000000) >> 31);
+ accu = (int64_t)alpha1[0] * bw;
+ alpha[0] = (int)((accu + 0x40000000) >> 31);
+ accu = (int64_t)alpha1[1] * bw;
+ alpha[1] = (int)((accu + 0x40000000) >> 31);
+
+ for (i = start; i < end; i++) {
+ accu = (int64_t)X_low[i][0] * 0x20000000;
+ accu += (int64_t)X_low[i - 2][0] * alpha[0];
+ accu -= (int64_t)X_low[i - 2][1] * alpha[1];
+ accu += (int64_t)X_low[i - 1][0] * alpha[2];
+ accu -= (int64_t)X_low[i - 1][1] * alpha[3];
+ X_high[i][0] = (int)((accu + 0x10000000) >> 29);
+
+ accu = (int64_t)X_low[i][1] * 0x20000000;
+ accu += (int64_t)X_low[i - 2][1] * alpha[0];
+ accu += (int64_t)X_low[i - 2][0] * alpha[1];
+ accu += (int64_t)X_low[i - 1][1] * alpha[2];
+ accu += (int64_t)X_low[i - 1][0] * alpha[3];
+ X_high[i][1] = (int)((accu + 0x10000000) >> 29);
+ }
+}
+
+static void sbr_hf_g_filt_c(int (*Y)[2], const int (*X_high)[40][2],
+ const SoftFloat *g_filt, int m_max, intptr_t ixh)
+{
+ int m, r;
+ int64_t accu;
+
+ for (m = 0; m < m_max; m++) {
+ r = 1 << (22-g_filt[m].exp);
+ accu = (int64_t)X_high[m][ixh][0] * ((g_filt[m].mant + 0x40)>>7);
+ Y[m][0] = (int)((accu + r) >> (23-g_filt[m].exp));
+
+ accu = (int64_t)X_high[m][ixh][1] * ((g_filt[m].mant + 0x40)>>7);
+ Y[m][1] = (int)((accu + r) >> (23-g_filt[m].exp));
+ }
+}
+
+static av_always_inline void sbr_hf_apply_noise(int (*Y)[2],
+ const SoftFloat *s_m,
+ const SoftFloat *q_filt,
+ int noise,
+ int phi_sign0,
+ int phi_sign1,
+ int m_max)
+{
+ int m;
+
+ for (m = 0; m < m_max; m++) {
+ int y0 = Y[m][0];
+ int y1 = Y[m][1];
+ noise = (noise + 1) & 0x1ff;
+ if (s_m[m].mant) {
+ int shift, round;
+
+ shift = 22 - s_m[m].exp;
+ if (shift < 30) {
+ round = 1 << (shift-1);
+ y0 += (s_m[m].mant * phi_sign0 + round) >> shift;
+ y1 += (s_m[m].mant * phi_sign1 + round) >> shift;
+ }
+ } else {
+ int shift, round, tmp;
+ int64_t accu;
+
+ shift = 22 - q_filt[m].exp;
+ if (shift < 30) {
+ round = 1 << (shift-1);
+
+ accu = (int64_t)q_filt[m].mant * ff_sbr_noise_table_fixed[noise][0];
+ tmp = (int)((accu + 0x40000000) >> 31);
+ y0 += (tmp + round) >> shift;
+
+ accu = (int64_t)q_filt[m].mant * ff_sbr_noise_table_fixed[noise][1];
+ tmp = (int)((accu + 0x40000000) >> 31);
+ y1 += (tmp + round) >> shift;
+ }
+ }
+ Y[m][0] = y0;
+ Y[m][1] = y1;
+ phi_sign1 = -phi_sign1;
+ }
+}
+
+#include "sbrdsp_template.c"
diff --git a/libavcodec/sbrdsp_template.c b/libavcodec/sbrdsp_template.c
index 410671e..b649dfd 100644
--- a/libavcodec/sbrdsp_template.c
+++ b/libavcodec/sbrdsp_template.c
@@ -20,55 +20,55 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-static void sbr_sum64x5_c(float *z)
+static void sbr_sum64x5_c(INTFLOAT *z)
{
int k;
for (k = 0; k < 64; k++) {
- float f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
+ INTFLOAT f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
z[k] = f;
}
}
-static void sbr_qmf_deint_bfly_c(float *v, const float *src0, const float *src1)
+static void sbr_qmf_deint_bfly_c(INTFLOAT *v, const INTFLOAT *src0, const INTFLOAT *src1)
{
int i;
for (i = 0; i < 64; i++) {
- v[ i] = src0[i] - src1[63 - i];
- v[127 - i] = src0[i] + src1[63 - i];
+ v[ i] = AAC_SRA_R((src0[i] - src1[63 - i]), 5);
+ v[127 - i] = AAC_SRA_R((src0[i] + src1[63 - i]), 5);
}
}
-static void sbr_hf_apply_noise_0(float (*Y)[2], const float *s_m,
- const float *q_filt, int noise,
+static void sbr_hf_apply_noise_0(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
+ const AAC_FLOAT *q_filt, int noise,
int kx, int m_max)
{
- sbr_hf_apply_noise(Y, s_m, q_filt, noise, 1.0, 0.0, m_max);
+ sbr_hf_apply_noise(Y, s_m, q_filt, noise, (INTFLOAT)1.0, (INTFLOAT)0.0, m_max);
}
-static void sbr_hf_apply_noise_1(float (*Y)[2], const float *s_m,
- const float *q_filt, int noise,
+static void sbr_hf_apply_noise_1(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
+ const AAC_FLOAT *q_filt, int noise,
int kx, int m_max)
{
- float phi_sign = 1 - 2 * (kx & 1);
- sbr_hf_apply_noise(Y, s_m, q_filt, noise, 0.0, phi_sign, m_max);
+ INTFLOAT phi_sign = 1 - 2 * (kx & 1);
+ sbr_hf_apply_noise(Y, s_m, q_filt, noise, (INTFLOAT)0.0, phi_sign, m_max);
}
-static void sbr_hf_apply_noise_2(float (*Y)[2], const float *s_m,
- const float *q_filt, int noise,
+static void sbr_hf_apply_noise_2(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
+ const AAC_FLOAT *q_filt, int noise,
int kx, int m_max)
{
- sbr_hf_apply_noise(Y, s_m, q_filt, noise, -1.0, 0.0, m_max);
+ sbr_hf_apply_noise(Y, s_m, q_filt, noise, (INTFLOAT)-1.0, (INTFLOAT)0.0, m_max);
}
-static void sbr_hf_apply_noise_3(float (*Y)[2], const float *s_m,
- const float *q_filt, int noise,
+static void sbr_hf_apply_noise_3(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
+ const AAC_FLOAT *q_filt, int noise,
int kx, int m_max)
{
- float phi_sign = 1 - 2 * (kx & 1);
- sbr_hf_apply_noise(Y, s_m, q_filt, noise, 0.0, -phi_sign, m_max);
+ INTFLOAT phi_sign = 1 - 2 * (kx & 1);
+ sbr_hf_apply_noise(Y, s_m, q_filt, noise, (INTFLOAT)0.0, -phi_sign, m_max);
}
-av_cold void ff_sbrdsp_init(SBRDSPContext *s)
+av_cold void AAC_RENAME(ff_sbrdsp_init)(SBRDSPContext *s)
{
s->sum64x5 = sbr_sum64x5_c;
s->sum_square = sbr_sum_square_c;
@@ -86,10 +86,12 @@ av_cold void ff_sbrdsp_init(SBRDSPContext *s)
s->hf_apply_noise[2] = sbr_hf_apply_noise_2;
s->hf_apply_noise[3] = sbr_hf_apply_noise_3;
+#if !USE_FIXED
if (ARCH_ARM)
ff_sbrdsp_init_arm(s);
if (ARCH_X86)
ff_sbrdsp_init_x86(s);
if (ARCH_MIPS)
ff_sbrdsp_init_mips(s);
+#endif /* !USE_FIXED */
}
diff --git a/libavutil/softfloat.h b/libavutil/softfloat.h
index 4272363..642a675 100644
--- a/libavutil/softfloat.h
+++ b/libavutil/softfloat.h
@@ -36,6 +36,14 @@ typedef struct SoftFloat{
int32_t exp;
}SoftFloat;
+static const SoftFloat FLOAT_0 = { 0, 0};
+static const SoftFloat FLOAT_05 = { 0x20000000, 0};
+static const SoftFloat FLOAT_1 = { 0x20000000, 1};
+static const SoftFloat FLOAT_EPSILON = { 0x29F16B12, -16};
+static const SoftFloat FLOAT_1584893192 = { 0x32B771ED, 1};
+static const SoftFloat FLOAT_100000 = { 0x30D40000, 17};
+static const SoftFloat FLOAT_0999999 = { 0x3FFFFBCE, 0};
+
static inline av_const double av_sf2double(SoftFloat v) {
v.exp -= ONE_BITS +1;
if(v.exp > 0) return (double)v.mant * (double)(1 << v.exp);
--
1.8.2.1
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