libavcodec: Implementation of AC3 fixedpoint decoder

Signed-off-by: Nedeljko Babic <nbabic@mips.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Changelog

@@ -2,6 +2,7 @@ Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
 version <next>:
+- added AC3 fixed-point decoding
 - shuffleplanes filter
 - subfile protocol
 - Phantom Cine demuxer

configure

@@ -1951,6 +1951,7 @@ aac_decoder_select="mdct sinewin"
 aac_encoder_select="audio_frame_queue mdct sinewin"
 aac_latm_decoder_select="aac_decoder aac_latm_parser"
 ac3_decoder_select="mdct ac3dsp ac3_parser dsputil"
+ac3_fixed_decoder_select="mdct ac3dsp ac3_parser dsputil"
 ac3_encoder_select="mdct ac3dsp dsputil"
 ac3_fixed_encoder_select="mdct ac3dsp dsputil"
 aic_decoder_select="dsputil golomb"

doc/general.texi

@@ -820,7 +820,7 @@ following image formats are supported:
     @tab encoding supported through external library libaacplus
 @item AAC                    @tab  E  @tab  X
     @tab encoding supported through external library libfaac and libvo-aacenc
-@item AC-3                   @tab IX  @tab  X
+@item AC-3                   @tab IX  @tab  IX
 @item ADPCM 4X Movie         @tab     @tab  X
 @item ADPCM CDROM XA         @tab     @tab  X
 @item ADPCM Creative Technology @tab     @tab  X

libavcodec/Makefile

@@ -94,7 +94,8 @@ OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
                                           psymodel.o iirfilter.o \
                                           mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
-OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
+OBJS-$(CONFIG_AC3_DECODER)             += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o
+OBJS-$(CONFIG_AC3_FIXED_DECODER)       += ac3dec_fixed.o ac3dec_data.o ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3enc.o ac3tab.o \
                                           ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_FIXED_ENCODER)       += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o

libavcodec/ac3.h

@@ -51,6 +51,52 @@
 #define EXP_D25   2
 #define EXP_D45   3
 
+#ifndef CONFIG_AC3_FIXED
+#define CONFIG_AC3_FIXED 0
+#endif
+
+#if CONFIG_AC3_FIXED
+
+#define FFT_FLOAT 0
+
+#define FIXR(a)                 ((int)((a) * 0 + 0.5))
+#define FIXR12(a)               ((int)((a) * 4096 + 0.5))
+#define FIXR15(a)               ((int)((a) * 32768 + 0.5))
+#define ROUND15(x)              ((x) + 16384) >> 15
+
+#define AC3_RENAME(x)           x ## _fixed
+#define AC3_NORM(norm)          (1<<24)/(norm)
+#define AC3_MUL(a,b)            ((((int64_t) (a)) * (b))>>12)
+#define AC3_RANGE(x)            (x)
+#define AC3_DYNAMIC_RANGE(x)    (x)
+#define AC3_SPX_BLEND(x)        (x)
+#define AC3_DYNAMIC_RANGE1      0
+
+#define INTFLOAT                int
+#define SHORTFLOAT              int16_t
+
+#else /* CONFIG_AC3_FIXED */
+
+#define FIXR(x)                 ((float)(x))
+#define FIXR12(x)               ((float)(x))
+#define FIXR15(x)               ((float)(x))
+#define ROUND15(x)              (x)
+
+#define AC3_RENAME(x)           x
+#define AC3_NORM(norm)          (1.0f/(norm))
+#define AC3_MUL(a,b)            ((a) * (b))
+#define AC3_RANGE(x)            (dynamic_range_tab[(x)])
+#define AC3_DYNAMIC_RANGE(x)    (powf(x,  s->drc_scale))
+#define AC3_SPX_BLEND(x)        (x)* (1.0f/32)
+#define AC3_DYNAMIC_RANGE1      1.0f
+
+#define INTFLOAT                float
+#define SHORTFLOAT              float
+
+#endif /* CONFIG_AC3_FIXED */
+
+#define AC3_LEVEL(x)            ROUND15((x) * FIXR15(0.7071067811865476))
+
 /* pre-defined gain values */
 #define LEVEL_PLUS_3DB          1.4142135623730950
 #define LEVEL_PLUS_1POINT5DB    1.1892071150027209

libavcodec/ac3dec.h

@@ -51,6 +51,7 @@
 #define AVCODEC_AC3DEC_H
 
 #include "libavutil/float_dsp.h"
+#include "libavutil/fixed_dsp.h"
 #include "libavutil/lfg.h"
 #include "ac3.h"
 #include "ac3dsp.h"
@@ -138,8 +139,8 @@ typedef struct AC3DecodeContext {
     int num_spx_bands;                          ///< number of spx bands                    (nspxbnds)
     uint8_t spx_band_sizes[SPX_MAX_BANDS];      ///< number of bins in each spx band
     uint8_t first_spx_coords[AC3_MAX_CHANNELS]; ///< first spx coordinates states           (firstspxcos)
-    float spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor  (nblendfact)
-    float spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact)
+    INTFLOAT spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor  (nblendfact)
+    INTFLOAT spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact)
 ///@}
 
 ///@name Adaptive hybrid transform
@@ -151,15 +152,15 @@ typedef struct AC3DecodeContext {
     int fbw_channels;                           ///< number of full-bandwidth channels
     int channels;                               ///< number of total channels
     int lfe_ch;                                 ///< index of LFE channel
-    float downmix_coeffs[AC3_MAX_CHANNELS][2];  ///< stereo downmix coefficients
+    SHORTFLOAT downmix_coeffs[AC3_MAX_CHANNELS][2];  ///< stereo downmix coefficients
     int downmixed;                              ///< indicates if coeffs are currently downmixed
     int output_mode;                            ///< output channel configuration
     int out_channels;                           ///< number of output channels
 ///@}
 
 ///@name Dynamic range
-    float dynamic_range[2];                 ///< dynamic range
-    float drc_scale;                        ///< percentage of dynamic range compression to be applied
+    INTFLOAT dynamic_range[2];                 ///< dynamic range
+    INTFLOAT drc_scale;                        ///< percentage of dynamic range compression to be applied
 ///@}
 
 ///@name Bandwidth
@@ -207,22 +208,26 @@ typedef struct AC3DecodeContext {
 
 ///@name Optimization
     DSPContext dsp;                         ///< for optimization
+#if CONFIG_AC3_FIXED
+    AVFixedDSPContext *fdsp;
+#else
     AVFloatDSPContext fdsp;
+#endif
     AC3DSPContext ac3dsp;
     FmtConvertContext fmt_conv;             ///< optimized conversion functions
 ///@}
 
-    float *outptr[AC3_MAX_CHANNELS];
-    float *xcfptr[AC3_MAX_CHANNELS];
-    float *dlyptr[AC3_MAX_CHANNELS];
+    SHORTFLOAT *outptr[AC3_MAX_CHANNELS];
+    INTFLOAT *xcfptr[AC3_MAX_CHANNELS];
+    INTFLOAT *dlyptr[AC3_MAX_CHANNELS];
 
 ///@name Aligned arrays
-    DECLARE_ALIGNED(16, int32_t, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];     ///< fixed-point transform coefficients
-    DECLARE_ALIGNED(32, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< transform coefficients
-    DECLARE_ALIGNED(32, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];             ///< delay - added to the next block
-    DECLARE_ALIGNED(32, float, window)[AC3_BLOCK_SIZE];                              ///< window coefficients
-    DECLARE_ALIGNED(32, float, tmp_output)[AC3_BLOCK_SIZE];                          ///< temporary storage for output before windowing
-    DECLARE_ALIGNED(32, float, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];            ///< output after imdct transform and windowing
+    DECLARE_ALIGNED(16, int,   fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];       ///< fixed-point transform coefficients
+    DECLARE_ALIGNED(32, INTFLOAT, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< transform coefficients
+    DECLARE_ALIGNED(32, INTFLOAT, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];             ///< delay - added to the next block
+    DECLARE_ALIGNED(32, INTFLOAT, window)[AC3_BLOCK_SIZE];                              ///< window coefficients
+    DECLARE_ALIGNED(32, INTFLOAT, tmp_output)[AC3_BLOCK_SIZE];                          ///< temporary storage for output before windowing
+    DECLARE_ALIGNED(32, SHORTFLOAT, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];            ///< output after imdct transform and windowing
     DECLARE_ALIGNED(32, uint8_t, input_buffer)[AC3_FRAME_BUFFER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; ///< temp buffer to prevent overread
 ///@}
 } AC3DecodeContext;

libavcodec/ac3dec_fixed.c

+/*
+ * Copyright (c) 2012
+ *      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.
+ *
+ * Author:  Stanislav Ocovaj (socovaj@mips.com)
+ *
+ * AC3 fixed-point decoder for MIPS platforms
+ *
+ * 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
+ */
+
+#define FFT_FLOAT 0
+#define CONFIG_AC3_FIXED 1
+#define FFT_FIXED_32 1
+#include "ac3dec.h"
+
+
+/**
+ * Table for center mix levels
+ * reference: Section 5.4.2.4 cmixlev
+ */
+static const uint8_t center_levels[4] = { 4, 5, 6, 5 };
+
+/**
+ * Table for surround mix levels
+ * reference: Section 5.4.2.5 surmixlev
+ */
+static const uint8_t surround_levels[4] = { 4, 6, 7, 6 };
+
+int end_freq_inv_tab[8] =
+{
+    50529027, 44278013, 39403370, 32292987, 27356480, 23729101, 20951060, 18755316
+};
+
+static void scale_coefs (
+    int32_t *dst,
+    const int32_t *src,
+    int dynrng,
+    int len)
+{
+    int i, shift, round;
+    int16_t mul;
+    int temp, temp1, temp2, temp3, temp4, temp5, temp6, temp7;
+
+    mul = (dynrng & 0x1f) + 0x20;
+    shift = 4 - ((dynrng << 24) >> 29);
+    round = 1 << (shift-1);
+    for (i=0; i<len; i+=8) {
+
+        temp = src[i] * mul;
+        temp1 = src[i+1] * mul;
+        temp = temp + round;
+        temp2 = src[i+2] * mul;
+
+        temp1 = temp1 + round;
+        dst[i] = temp >> shift;
+        temp3 = src[i+3] * mul;
+        temp2 = temp2 + round;
+
+        dst[i+1] = temp1 >> shift;
+        temp4 = src[i + 4] * mul;
+        temp3 = temp3 + round;
+        dst[i+2] = temp2 >> shift;
+
+        temp5 = src[i+5] * mul;
+        temp4 = temp4 + round;
+        dst[i+3] = temp3 >> shift;
+        temp6 = src[i+6] * mul;
+
+        dst[i+4] = temp4 >> shift;
+        temp5 = temp5 + round;
+        temp7 = src[i+7] * mul;
+        temp6 = temp6 + round;
+
+        dst[i+5] = temp5 >> shift;
+        temp7 = temp7 + round;
+        dst[i+6] = temp6 >> shift;
+        dst[i+7] = temp7 >> shift;
+
+    }
+}
+
+/**
+ * Downmix samples from original signal to stereo or mono (this is for 16-bit samples
+ * and fixed point decoder - original (for 32-bit samples) is in ac3dsp.c).
+ */
+static void ac3_downmix_c_fixed16(int16_t **samples, int16_t (*matrix)[2],
+                                  int out_ch, int in_ch, int len)
+{
+    int i, j;
+    int v0, v1;
+    if (out_ch == 2) {
+        for (i = 0; i < len; i++) {
+            v0 = v1 = 0;
+            for (j = 0; j < in_ch; j++) {
+                v0 += samples[j][i] * matrix[j][0];
+                v1 += samples[j][i] * matrix[j][1];
+            }
+            samples[0][i] = (v0+2048)>>12;
+            samples[1][i] = (v1+2048)>>12;
+        }
+    } else if (out_ch == 1) {
+        for (i = 0; i < len; i++) {
+            v0 = 0;
+            for (j = 0; j < in_ch; j++)
+                v0 += samples[j][i] * matrix[j][0];
+            samples[0][i] = (v0+2048)>>12;
+        }
+    }
+}
+
+#include "ac3dec.c"
+
+static const AVOption options[] = {
+    { NULL},
+};
+
+static const AVClass ac3_decoder_class = {
+    .class_name = "Fixed-Point AC-3 Decoder",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_ac3_fixed_decoder = {
+    .name           = "ac3_fixed",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_AC3,
+    .priv_data_size = sizeof (AC3DecodeContext),
+    .init           = ac3_decode_init,
+    .close          = ac3_decode_end,
+    .decode         = ac3_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+                                                      AV_SAMPLE_FMT_NONE },
+    .priv_class     = &ac3_decoder_class,
+};

libavcodec/ac3dec_float.c

+/*
+ * AC-3 Audio Decoder
+ * This code was developed as part of Google Summer of Code 2006.
+ * E-AC-3 support was added as part of Google Summer of Code 2007.
+ *
+ * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com)
+ * Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
+ * Copyright (c) 2007 Justin Ruggles <justin.ruggles@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
+ */
+
+/**
+ * Upmix delay samples from stereo to original channel layout.
+ */
+#include "ac3dec.h"
+#include "ac3dec.c"
+
+static const AVOption options[] = {
+    { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 6.0, PAR },
+
+{"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 2, 0, "dmix_mode"},
+{"ltrt_cmixlev",   "Lt/Rt Center Mix Level",   OFFSET(ltrt_center_mix_level),    AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+{"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level),  AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+{"loro_cmixlev",   "Lo/Ro Center Mix Level",   OFFSET(loro_center_mix_level),    AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+{"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level),  AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+
+    { NULL},
+};
+
+static const AVClass ac3_decoder_class = {
+    .class_name = "AC3 decoder",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_ac3_decoder = {
+    .name           = "ac3",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_AC3,
+    .priv_data_size = sizeof (AC3DecodeContext),
+    .init           = ac3_decode_init,
+    .close          = ac3_decode_end,
+    .decode         = ac3_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+                                                      AV_SAMPLE_FMT_NONE },
+    .priv_class     = &ac3_decoder_class,
+};
+
+#if CONFIG_EAC3_DECODER
+static const AVClass eac3_decoder_class = {
+    .class_name = "E-AC3 decoder",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_eac3_decoder = {
+    .name           = "eac3",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_EAC3,
+    .priv_data_size = sizeof (AC3DecodeContext),
+    .init           = ac3_decode_init,
+    .close          = ac3_decode_end,
+    .decode         = ac3_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+                                                      AV_SAMPLE_FMT_NONE },
+    .priv_class     = &eac3_decoder_class,
+};
+#endif

libavcodec/ac3dsp.c

@@ -239,6 +239,31 @@ static void ac3_downmix_c(float **samples, float (*matrix)[2],
     }
 }
 
+static void ac3_downmix_c_fixed(int32_t **samples, int16_t (*matrix)[2],
+                                int out_ch, int in_ch, int len)
+{
+    int i, j;
+    int64_t v0, v1;
+    if (out_ch == 2) {
+        for (i = 0; i < len; i++) {
+            v0 = v1 = 0;
+            for (j = 0; j < in_ch; j++) {
+                v0 += (int64_t)samples[j][i] * matrix[j][0];
+                v1 += (int64_t)samples[j][i] * matrix[j][1];
+            }
+            samples[0][i] = (v0+2048)>>12;
+            samples[1][i] = (v1+2048)>>12;
+        }
+    } else if (out_ch == 1) {
+        for (i = 0; i < len; i++) {
+            v0 = 0;
+            for (j = 0; j < in_ch; j++)
+                v0 += (int64_t)samples[j][i] * matrix[j][0];
+            samples[0][i] = (v0+2048)>>12;
+        }
+    }
+}
+
 static void apply_window_int16_c(int16_t *output, const int16_t *input,
                                  const int16_t *window, unsigned int len)
 {
@@ -266,6 +291,7 @@ av_cold void ff_ac3dsp_init(AC3DSPContext *c, int bit_exact)
     c->sum_square_butterfly_int32 = ac3_sum_square_butterfly_int32_c;
     c->sum_square_butterfly_float = ac3_sum_square_butterfly_float_c;
     c->downmix = ac3_downmix_c;
+    c->downmix_fixed = ac3_downmix_c_fixed;
     c->apply_window_int16 = apply_window_int16_c;
 
     if (ARCH_ARM)

libavcodec/ac3dsp.h

@@ -135,6 +135,9 @@ typedef struct AC3DSPContext {
     void (*downmix)(float **samples, float (*matrix)[2], int out_ch,
                     int in_ch, int len);
 
+    void (*downmix_fixed)(int32_t **samples, int16_t (*matrix)[2], int out_ch,
+                          int in_ch, int len);
+
     /**
      * Apply symmetric window in 16-bit fixed-point.
      * @param output destination array

libavcodec/allcodecs.c

@@ -323,7 +323,7 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC (AAC,               aac);
     REGISTER_DECODER(AAC_LATM,          aac_latm);
     REGISTER_ENCDEC (AC3,               ac3);
-    REGISTER_ENCODER(AC3_FIXED,         ac3_fixed);
+    REGISTER_ENCDEC (AC3_FIXED,         ac3_fixed);
     REGISTER_ENCDEC (ALAC,              alac);
     REGISTER_DECODER(ALS,               als);
     REGISTER_DECODER(AMRNB,             amrnb);

libavcodec/kbdwin.c

@@ -45,3 +45,13 @@ av_cold void ff_kbd_window_init(float *window, float alpha, int n)
    for (i = 0; i < n; i++)
        window[i] = sqrt(local_window[i] / sum);
 }
+
+av_cold void ff_kbd_window_init_fixed(int32_t *window, float alpha, int n)
+{
+    int i;
+    float local_window[FF_KBD_WINDOW_MAX];
+
+    ff_kbd_window_init(local_window, alpha, n);
+    for (i = 0; i < n; i++)
+        window[i] = (int)floor(2147483647.0 * local_window[i] + 0.5);
+}

libavcodec/kbdwin.h

@@ -31,5 +31,6 @@
  * @param   n       size of half window, max FF_KBD_WINDOW_MAX
  */
 void ff_kbd_window_init(float *window, float alpha, int n);
+void ff_kbd_window_init_fixed(int32_t *window, float alpha, int n);
 
 #endif /* AVCODEC_KBDWIN_H */

libavcodec/version.h

@@ -29,7 +29,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVCODEC_VERSION_MAJOR 55
-#define LIBAVCODEC_VERSION_MINOR  55
+#define LIBAVCODEC_VERSION_MINOR  56
 #define LIBAVCODEC_VERSION_MICRO 107
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \