[FFmpeg-soc] [soc]: r645 - in eac3/ac3: . ac3.c ac3.h ac3dec.c
bwolowiec
subversion at mplayerhq.hu
Thu Aug 9 10:52:37 CEST 2007
Author: bwolowiec
Date: Thu Aug 9 10:52:37 2007
New Revision: 645
Log:
add current version of ac3.c ac3dec.c ac3.h
Added:
eac3/ac3/
eac3/ac3/ac3.c
eac3/ac3/ac3.h
eac3/ac3/ac3dec.c
Added: eac3/ac3/ac3.c
==============================================================================
--- (empty file)
+++ eac3/ac3/ac3.c Thu Aug 9 10:52:37 2007
@@ -0,0 +1,239 @@
+/*
+ * Common code between AC3 encoder and decoder
+ * Copyright (c) 2000 Fabrice Bellard.
+ *
+ * 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 ac3.c
+ * Common code between AC3 encoder and decoder.
+ */
+
+#include "avcodec.h"
+#include "ac3.h"
+#include "bitstream.h"
+
+static uint8_t bndtab[51];
+static uint8_t masktab[253];
+
+static inline int calc_lowcomp1(int a, int b0, int b1, int c)
+{
+ if ((b0 + 256) == b1) {
+ a = c;
+ } else if (b0 > b1) {
+ a = FFMAX(a - 64, 0);
+ }
+ return a;
+}
+
+static inline int calc_lowcomp(int a, int b0, int b1, int bin)
+{
+ if (bin < 7) {
+ return calc_lowcomp1(a, b0, b1, 384);
+ } else if (bin < 20) {
+ return calc_lowcomp1(a, b0, b1, 320);
+ } else {
+ return FFMAX(a - 128, 0);
+ }
+}
+
+void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
+ int16_t *bndpsd)
+{
+ int bin, i, j, k, end1, v;
+
+ /* exponent mapping to PSD */
+ for(bin=start;bin<end;bin++) {
+ psd[bin]=(3072 - (exp[bin] << 7));
+ }
+
+ /* PSD integration */
+ j=start;
+ k=masktab[start];
+ do {
+ v=psd[j];
+ j++;
+ end1 = FFMIN(bndtab[k+1], end);
+ for(i=j;i<end1;i++) {
+ /* logadd */
+ int adr = FFMIN(FFABS(v - psd[j]) >> 1, 255);
+ v = FFMAX(v, psd[j]) + ff_ac3_latab[adr];
+ j++;
+ }
+ bndpsd[k]=v;
+ k++;
+ } while (end > bndtab[k]);
+}
+
+void ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *bndpsd,
+ int start, int end, int fgain, int is_lfe,
+ int deltbae, int deltnseg, uint8_t *deltoffst,
+ uint8_t *deltlen, uint8_t *deltba,
+ int16_t *mask)
+{
+ int16_t excite[50]; /* excitation */
+ int bin, k;
+ int bndstrt, bndend, begin, end1, tmp;
+ int lowcomp, fastleak, slowleak;
+
+ /* excitation function */
+ bndstrt = masktab[start];
+ bndend = masktab[end-1] + 1;
+
+ if (bndstrt == 0) {
+ lowcomp = 0;
+ lowcomp = calc_lowcomp1(lowcomp, bndpsd[0], bndpsd[1], 384);
+ excite[0] = bndpsd[0] - fgain - lowcomp;
+ lowcomp = calc_lowcomp1(lowcomp, bndpsd[1], bndpsd[2], 384);
+ excite[1] = bndpsd[1] - fgain - lowcomp;
+ begin = 7;
+ for (bin = 2; bin < 7; bin++) {
+ if (!(is_lfe && bin == 6))
+ lowcomp = calc_lowcomp1(lowcomp, bndpsd[bin], bndpsd[bin+1], 384);
+ fastleak = bndpsd[bin] - fgain;
+ slowleak = bndpsd[bin] - s->sgain;
+ excite[bin] = fastleak - lowcomp;
+ if (!(is_lfe && bin == 6)) {
+ if (bndpsd[bin] <= bndpsd[bin+1]) {
+ begin = bin + 1;
+ break;
+ }
+ }
+ }
+
+ end1=bndend;
+ if (end1 > 22) end1=22;
+
+ for (bin = begin; bin < end1; bin++) {
+ if (!(is_lfe && bin == 6))
+ lowcomp = calc_lowcomp(lowcomp, bndpsd[bin], bndpsd[bin+1], bin);
+
+ fastleak = FFMAX(fastleak - s->fdecay, bndpsd[bin] - fgain);
+ slowleak = FFMAX(slowleak - s->sdecay, bndpsd[bin] - s->sgain);
+ excite[bin] = FFMAX(fastleak - lowcomp, slowleak);
+ }
+ begin = 22;
+ } else {
+ /* coupling channel */
+ begin = bndstrt;
+
+ fastleak = (s->cplfleak << 8) + 768;
+ slowleak = (s->cplsleak << 8) + 768;
+ }
+
+ for (bin = begin; bin < bndend; bin++) {
+ fastleak = FFMAX(fastleak - s->fdecay, bndpsd[bin] - fgain);
+ slowleak = FFMAX(slowleak - s->sdecay, bndpsd[bin] - s->sgain);
+ excite[bin] = FFMAX(fastleak, slowleak);
+ }
+
+ /* compute masking curve */
+
+ for (bin = bndstrt; bin < bndend; bin++) {
+ tmp = s->dbknee - bndpsd[bin];
+ if (tmp > 0) {
+ excite[bin] += tmp >> 2;
+ }
+ mask[bin] = FFMAX(ff_ac3_hth[bin >> s->halfratecod][s->fscod], excite[bin]);
+ }
+
+ /* delta bit allocation */
+
+ if (deltbae == DBA_REUSE || deltbae == DBA_NEW) {
+ int band, seg, delta;
+ band = 0;
+ for (seg = 0; seg < deltnseg; seg++) {
+ band += deltoffst[seg];
+ if (deltba[seg] >= 4) {
+ delta = (deltba[seg] - 3) << 7;
+ } else {
+ delta = (deltba[seg] - 4) << 7;
+ }
+ for (k = 0; k < deltlen[seg]; k++) {
+ mask[band] += delta;
+ band++;
+ }
+ }
+ }
+}
+
+void ff_ac3_bit_alloc_calc_bap(int16_t *mask, int16_t *psd, int start, int end,
+ int snroffset, int floor, uint8_t *bap)
+{
+ int i, j, k, end1, v, address;
+
+ /* special case, if snroffset is -960, set all bap's to zero */
+ if(snroffset == -960) {
+ memset(bap, 0, 256);
+ return;
+ }
+
+ i = start;
+ j = masktab[start];
+ do {
+ v = (FFMAX(mask[j] - snroffset - floor, 0) & 0x1FE0) + floor;
+ end1 = FFMIN(bndtab[j] + ff_ac3_bndsz[j], end);
+ for (k = i; k < end1; k++) {
+ address = av_clip((psd[i] - v) >> 5, 0, 63);
+ bap[i] = ff_ac3_baptab[address];
+ i++;
+ }
+ } while (end > bndtab[j++]);
+}
+
+/* AC3 bit allocation. The algorithm is the one described in the AC3
+ spec. */
+void ac3_parametric_bit_allocation(AC3BitAllocParameters *s, uint8_t *bap,
+ int8_t *exp, int start, int end,
+ int snroffset, int fgain, int is_lfe,
+ int deltbae,int deltnseg,
+ uint8_t *deltoffst, uint8_t *deltlen,
+ uint8_t *deltba)
+{
+ int16_t psd[256]; /* scaled exponents */
+ int16_t bndpsd[50]; /* interpolated exponents */
+ int16_t mask[50]; /* masking value */
+
+ ff_ac3_bit_alloc_calc_psd(exp, start, end, psd, bndpsd);
+
+ ff_ac3_bit_alloc_calc_mask(s, bndpsd, start, end, fgain, is_lfe,
+ deltbae, deltnseg, deltoffst, deltlen, deltba,
+ mask);
+
+ ff_ac3_bit_alloc_calc_bap(mask, psd, start, end, snroffset, s->floor, bap);
+}
+
+/**
+ * Initializes some tables.
+ * note: This function must remain thread safe because it is called by the
+ * AVParser init code.
+ */
+void ac3_common_init(void)
+{
+ int i, j, k, l, v;
+ /* compute bndtab and masktab from bandsz */
+ k = 0;
+ l = 0;
+ for(i=0;i<50;i++) {
+ bndtab[i] = l;
+ v = ff_ac3_bndsz[i];
+ for(j=0;j<v;j++) masktab[k++]=i;
+ l += v;
+ }
+ bndtab[50] = l;
+}
Added: eac3/ac3/ac3.h
==============================================================================
--- (empty file)
+++ eac3/ac3/ac3.h Thu Aug 9 10:52:37 2007
@@ -0,0 +1,175 @@
+/*
+ * Common code between AC3 encoder and decoder
+ * Copyright (c) 2000, 2001, 2002 Fabrice Bellard.
+ *
+ * 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 ac3.h
+ * Common code between AC3 encoder and decoder.
+ */
+
+#ifndef AC3_H
+#define AC3_H
+
+#include "ac3tab.h"
+
+#define AC3_MAX_CODED_FRAME_SIZE 3840 /* in bytes */
+#define AC3_MAX_CHANNELS 6 /* including LFE channel */
+
+#define NB_BLOCKS 6 /* number of PCM blocks inside an AC3 frame */
+#define AC3_FRAME_SIZE (NB_BLOCKS * 256)
+
+/* exponent encoding strategy */
+#define EXP_REUSE 0
+#define EXP_NEW 1
+
+#define EXP_D15 1
+#define EXP_D25 2
+#define EXP_D45 3
+
+/** Delta bit allocation strategy */
+typedef enum {
+ DBA_REUSE = 0,
+ DBA_NEW,
+ DBA_NONE,
+ DBA_RESERVED
+} AC3DeltaStrategy;
+
+/** Channel mode (audio coding mode) */
+typedef enum {
+ AC3_ACMOD_DUALMONO = 0,
+ AC3_ACMOD_MONO,
+ AC3_ACMOD_STEREO,
+ AC3_ACMOD_3F,
+ AC3_ACMOD_2F1R,
+ AC3_ACMOD_3F1R,
+ AC3_ACMOD_2F2R,
+ AC3_ACMOD_3F2R
+} AC3ChannelMode;
+
+typedef struct AC3BitAllocParameters {
+ int fscod; /* frequency */
+ int halfratecod;
+ int sgain, sdecay, fdecay, dbknee, floor;
+ int cplfleak, cplsleak;
+} AC3BitAllocParameters;
+
+/**
+ * @struct AC3HeaderInfo
+ * Coded AC-3 header values up to the lfeon element, plus derived values.
+ */
+typedef struct {
+ /** @defgroup coded Coded elements
+ * @{
+ */
+ uint16_t sync_word;
+ uint16_t crc1;
+ uint8_t fscod;
+ uint8_t frmsizecod;
+ uint8_t bsid;
+ uint8_t bsmod;
+ uint8_t acmod;
+ uint8_t cmixlev;
+ uint8_t surmixlev;
+ uint8_t dsurmod;
+ uint8_t lfeon;
+ /** @} */
+
+ /** @defgroup derived Derived values
+ * @{
+ */
+ uint8_t halfratecod;
+ uint16_t sample_rate;
+ uint32_t bit_rate;
+ uint8_t channels;
+ uint16_t frame_size;
+ /** @} */
+} AC3HeaderInfo;
+
+
+void ac3_common_init(void);
+
+/**
+ * Calculates the log power-spectral density of the input signal.
+ * This gives a rough estimate of signal power in the frequency domain by using
+ * the spectral envelope (exponents). The psd is also separately grouped
+ * into critical bands for use in the calculating the masking curve.
+ * 128 units in psd = -6 dB. The dbknee parameter in AC3BitAllocParameters
+ * determines the reference level.
+ *
+ * @param[in] exp frequency coefficient exponents
+ * @param[in] start starting bin location
+ * @param[in] end ending bin location
+ * @param[out] psd signal power for each frequency bin
+ * @param[out] bndpsd signal power for each critical band
+ */
+void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
+ int16_t *bndpsd);
+
+/**
+ * Calculates the masking curve.
+ * First, the excitation is calculated using parameters in \p s and the signal
+ * power in each critical band. The excitation is compared with a predefined
+ * hearing threshold table to produce the masking curve. If delta bit
+ * allocation information is provided, it is used for adjusting the masking
+ * curve, usually to give a closer match to a better psychoacoustic model.
+ *
+ * @param[in] s adjustable bit allocation parameters
+ * @param[in] bndpsd signal power for each critical band
+ * @param[in] start starting bin location
+ * @param[in] end ending bin location
+ * @param[in] fgain fast gain (estimated signal-to-mask ratio)
+ * @param[in] is_lfe whether or not the channel being processed is the LFE
+ * @param[in] deltbae delta bit allocation exists (none, reuse, or new)
+ * @param[in] deltnseg number of delta segments
+ * @param[in] deltoffst location offsets for each segment
+ * @param[in] deltlen length of each segment
+ * @param[in] deltba delta bit allocation for each segment
+ * @param[out] mask calculated masking curve
+ */
+void ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *bndpsd,
+ int start, int end, int fgain, int is_lfe,
+ int deltbae, int deltnseg, uint8_t *deltoffst,
+ uint8_t *deltlen, uint8_t *deltba,
+ int16_t *mask);
+
+/**
+ * Calculates bit allocation pointers.
+ * The SNR is the difference between the masking curve and the signal. AC-3
+ * uses this value for each frequency bin to allocate bits. The \p snroffset
+ * parameter is a global adjustment to the SNR for all bins.
+ *
+ * @param[in] mask masking curve
+ * @param[in] psd signal power for each frequency bin
+ * @param[in] start starting bin location
+ * @param[in] end ending bin location
+ * @param[in] snroffset SNR adjustment
+ * @param[in] floor noise floor
+ * @param[out] bap bit allocation pointers
+ */
+void ff_ac3_bit_alloc_calc_bap(int16_t *mask, int16_t *psd, int start, int end,
+ int snroffset, int floor, uint8_t *bap);
+
+void ac3_parametric_bit_allocation(AC3BitAllocParameters *s, uint8_t *bap,
+ int8_t *exp, int start, int end,
+ int snroffset, int fgain, int is_lfe,
+ int deltbae,int deltnseg,
+ uint8_t *deltoffst, uint8_t *deltlen, uint8_t *deltba);
+
+#endif /* AC3_H */
Added: eac3/ac3/ac3dec.c
==============================================================================
--- (empty file)
+++ eac3/ac3/ac3dec.c Thu Aug 9 10:52:37 2007
@@ -0,0 +1,1162 @@
+/*
+ * AC-3 Audio Decoder
+ * This code is developed as part of Google Summer of Code 2006 Program.
+ *
+ * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com).
+ * Copyright (c) 2007 Justin Ruggles
+ *
+ * Portions of this code are derived from liba52
+ * http://liba52.sourceforge.net
+ * Copyright (C) 2000-2003 Michel Lespinasse <walken at zoy.org>
+ * Copyright (C) 1999-2000 Aaron Holtzman <aholtzma at ess.engr.uvic.ca>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 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
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU 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
+ */
+
+#include <stdio.h>
+#include <stddef.h>
+#include <math.h>
+#include <string.h>
+
+#include "avcodec.h"
+#include "ac3_parser.h"
+#include "bitstream.h"
+#include "dsputil.h"
+#include "random.h"
+
+/**
+ * Table of bin locations for rematrixing bands
+ * reference: Section 7.5.2 Rematrixing : Frequency Band Definitions
+ */
+static const uint8_t rematrix_band_tbl[5] = { 13, 25, 37, 61, 253 };
+
+/**
+ * table for exponent to scale_factor mapping
+ * scale_factors[i] = 2 ^ -i
+ */
+static float scale_factors[25];
+
+/** table for grouping exponents */
+static uint8_t exp_ungroup_tbl[128][3];
+
+
+/** tables for ungrouping mantissas */
+static float b1_mantissas[32][3];
+static float b2_mantissas[128][3];
+static float b3_mantissas[8];
+static float b4_mantissas[128][2];
+static float b5_mantissas[16];
+
+/**
+ * Quantization table: levels for symmetric. bits for asymmetric.
+ * reference: Table 7.18 Mapping of bap to Quantizer
+ */
+static const uint8_t qntztab[16] = {
+ 0, 3, 5, 7, 11, 15,
+ 5, 6, 7, 8, 9, 10, 11, 12, 14, 16
+};
+
+/** dynamic range table. converts codes to scale factors. */
+static float dynrng_tbl[256];
+
+/** dialogue normalization table */
+static float dialnorm_tbl[32];
+
+/** Adjustments in dB gain */
+#define LEVEL_MINUS_3DB 0.7071067811865476
+#define LEVEL_MINUS_4POINT5DB 0.5946035575013605
+#define LEVEL_MINUS_6DB 0.5000000000000000
+#define LEVEL_MINUS_9DB 0.3535533905932738
+#define LEVEL_ZERO 0.0000000000000000
+#define LEVEL_ONE 1.0000000000000000
+
+static const float gain_levels[6] = {
+ LEVEL_ZERO,
+ LEVEL_ONE,
+ LEVEL_MINUS_3DB,
+ LEVEL_MINUS_4POINT5DB,
+ LEVEL_MINUS_6DB,
+ LEVEL_MINUS_9DB
+};
+
+/**
+ * Table for center mix levels
+ * reference: Section 5.4.2.4 cmixlev
+ */
+static const uint8_t clevs[4] = { 2, 3, 4, 3 };
+
+/**
+ * Table for surround mix levels
+ * reference: Section 5.4.2.5 surmixlev
+ */
+static const uint8_t slevs[4] = { 2, 4, 0, 4 };
+
+/**
+ * Table for default stereo downmixing coefficients
+ * reference: Section 7.8.2 Downmixing Into Two Channels
+ */
+static const uint8_t ac3_default_coeffs[8][5][2] = {
+ { { 1, 0 }, { 0, 1 }, },
+ { { 2, 2 }, },
+ { { 1, 0 }, { 0, 1 }, },
+ { { 1, 0 }, { 3, 3 }, { 0, 1 }, },
+ { { 1, 0 }, { 0, 1 }, { 4, 4 }, },
+ { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 5, 5 }, },
+ { { 1, 0 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, },
+ { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, },
+};
+
+/* override ac3.h to include coupling channel */
+#undef AC3_MAX_CHANNELS
+#define AC3_MAX_CHANNELS 7
+#define CPL_CH 0
+
+#define AC3_OUTPUT_LFEON 8
+
+typedef struct {
+ int acmod; ///< audio coding mode
+ int dsurmod; ///< dolby surround mode
+ int blksw[AC3_MAX_CHANNELS]; ///< block switch flags
+ int dithflag[AC3_MAX_CHANNELS]; ///< dither flags
+ int dither_all; ///< true if all channels are dithered
+ int cplinu; ///< coupling in use
+ int chincpl[AC3_MAX_CHANNELS]; ///< channel in coupling
+ int phsflginu; ///< phase flags in use
+ int cplbndstrc[18]; ///< coupling band structure
+ int rematstr; ///< rematrixing strategy
+ int nrematbnd; ///< number of rematrixing bands
+ int rematflg[4]; ///< rematrixing flags
+ int expstr[AC3_MAX_CHANNELS]; ///< exponent strategies
+ int snroffst[AC3_MAX_CHANNELS]; ///< signal-to-noise ratio offsets
+ int fgain[AC3_MAX_CHANNELS]; ///< fast gain values (signal-to-mask ratio)
+ int deltbae[AC3_MAX_CHANNELS]; ///< delta bit allocation exists
+ int deltnseg[AC3_MAX_CHANNELS]; ///< number of delta segments
+ uint8_t deltoffst[AC3_MAX_CHANNELS][8]; ///< delta segment offsets
+ uint8_t deltlen[AC3_MAX_CHANNELS][8]; ///< delta segment lengths
+ uint8_t deltba[AC3_MAX_CHANNELS][8]; ///< delta values for each segment
+
+ int sampling_rate; ///< sample frequency, in Hz
+ int bit_rate; ///< stream bit rate, in bits-per-second
+ int frame_size; ///< current frame size, in bytes
+
+ int nchans; ///< number of total channels
+ int nfchans; ///< number of full-bandwidth channels
+ int lfeon; ///< lfe channel in use
+ int lfe_ch; ///< index of LFE channel
+ int output_mode; ///< output channel configuration
+ int out_channels; ///< number of output channels
+
+ float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
+ float dialnorm[2]; ///< dialogue normalization
+ float dynrng[2]; ///< dynamic range
+ float cplco[AC3_MAX_CHANNELS][18]; ///< coupling coordinates
+ int ncplbnd; ///< number of coupling bands
+ int ncplsubnd; ///< number of coupling sub bands
+ int startmant[AC3_MAX_CHANNELS]; ///< start frequency bin
+ int endmant[AC3_MAX_CHANNELS]; ///< end frequency bin
+ AC3BitAllocParameters bit_alloc_params; ///< bit allocation parameters
+
+ int8_t dexps[AC3_MAX_CHANNELS][256]; ///< decoded exponents
+ uint8_t bap[AC3_MAX_CHANNELS][256]; ///< bit allocation pointers
+ int16_t psd[AC3_MAX_CHANNELS][256]; ///< scaled exponents
+ int16_t bndpsd[AC3_MAX_CHANNELS][50]; ///< interpolated exponents
+ int16_t mask[AC3_MAX_CHANNELS][50]; ///< masking curve values
+
+ DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]); ///< transform coefficients
+
+ /* For IMDCT. */
+ MDCTContext imdct_512; ///< for 512 sample IMDCT
+ MDCTContext imdct_256; ///< for 256 sample IMDCT
+ DSPContext dsp; ///< for optimization
+ float add_bias; ///< offset for float_to_int16 conversion
+ float mul_bias; ///< scaling for float_to_int16 conversion
+
+ DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS-1][256]); ///< output after imdct transform and windowing
+ DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output
+ DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS-1][256]); ///< delay - added to the next block
+ DECLARE_ALIGNED_16(float, tmp_imdct[256]); ///< temporary storage for imdct transform
+ DECLARE_ALIGNED_16(float, tmp_output[512]); ///< temporary storage for output before windowing
+ DECLARE_ALIGNED_16(float, window[256]); ///< window coefficients
+
+ /* Miscellaneous. */
+ GetBitContext gb; ///< bitstream reader
+ AVRandomState dith_state; ///< for dither generation
+ AVCodecContext *avctx; ///< parent context
+} AC3DecodeContext;
+
+/**
+ * Generate a Kaiser-Bessel Derived Window.
+ */
+static void ac3_window_init(float *window)
+{
+ int i, j;
+ double sum = 0.0, bessel, tmp;
+ double local_window[256];
+ double alpha2 = (5.0 * M_PI / 256.0) * (5.0 * M_PI / 256.0);
+
+ for (i = 0; i < 256; i++) {
+ tmp = i * (256 - i) * alpha2;
+ bessel = 1.0;
+ for (j = 100; j > 0; j--) /* default to 100 iterations */
+ bessel = bessel * tmp / (j * j) + 1;
+ sum += bessel;
+ local_window[i] = sum;
+ }
+
+ sum++;
+ for (i = 0; i < 256; i++)
+ window[i] = sqrt(local_window[i] / sum);
+}
+
+/**
+ * Symmetrical Dequantization
+ * reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization
+ * Tables 7.19 to 7.23
+ */
+static inline float
+symmetric_dequant(int code, int levels)
+{
+ return (code - (levels >> 1)) * (2.0f / levels);
+}
+
+/*
+ * Initialize tables at runtime.
+ */
+static void ac3_tables_init(void)
+{
+ int i;
+
+ /* generate grouped mantissa tables
+ reference: Section 7.3.5 Ungrouping of Mantissas */
+ for(i=0; i<32; i++) {
+ /* bap=1 mantissas */
+ b1_mantissas[i][0] = symmetric_dequant( i / 9 , 3);
+ b1_mantissas[i][1] = symmetric_dequant((i % 9) / 3, 3);
+ b1_mantissas[i][2] = symmetric_dequant((i % 9) % 3, 3);
+ }
+ for(i=0; i<128; i++) {
+ /* bap=2 mantissas */
+ b2_mantissas[i][0] = symmetric_dequant( i / 25 , 5);
+ b2_mantissas[i][1] = symmetric_dequant((i % 25) / 5, 5);
+ b2_mantissas[i][2] = symmetric_dequant((i % 25) % 5, 5);
+
+ /* bap=4 mantissas */
+ b4_mantissas[i][0] = symmetric_dequant(i / 11, 11);
+ b4_mantissas[i][1] = symmetric_dequant(i % 11, 11);
+ }
+ /* generate ungrouped mantissa tables
+ reference: Tables 7.21 and 7.23 */
+ for(i=0; i<7; i++) {
+ /* bap=3 mantissas */
+ b3_mantissas[i] = symmetric_dequant(i, 7);
+ }
+ for(i=0; i<15; i++) {
+ /* bap=5 mantissas */
+ b5_mantissas[i] = symmetric_dequant(i, 15);
+ }
+
+ /* generate dynamic range table
+ reference: Section 7.7.1 Dynamic Range Control */
+ for(i=0; i<256; i++) {
+ int v = (i >> 5) - ((i >> 7) << 3) - 5;
+ dynrng_tbl[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20);
+ }
+
+ /* generate dialogue normalization table
+ references: Section 5.4.2.8 dialnorm
+ Section 7.6 Dialogue Normalization */
+ for(i=1; i<32; i++) {
+ dialnorm_tbl[i] = expf((i-31) * M_LN10 / 20.0f);
+ }
+ dialnorm_tbl[0] = dialnorm_tbl[31];
+
+ /* generate scale factors for exponents and asymmetrical dequantization
+ reference: Section 7.3.2 Expansion of Mantissas for Asymmetric Quantization */
+ for (i = 0; i < 25; i++)
+ scale_factors[i] = pow(2.0, -i);
+
+ /* generate exponent tables
+ reference: Section 7.1.3 Exponent Decoding */
+ for(i=0; i<128; i++) {
+ exp_ungroup_tbl[i][0] = i / 25;
+ exp_ungroup_tbl[i][1] = (i % 25) / 5;
+ exp_ungroup_tbl[i][2] = (i % 25) % 5;
+ }
+}
+
+
+/**
+ * AVCodec initialization
+ */
+static int ac3_decode_init(AVCodecContext *avctx)
+{
+ AC3DecodeContext *ctx = avctx->priv_data;
+ ctx->avctx = avctx;
+
+ ac3_common_init();
+ ac3_tables_init();
+ ff_mdct_init(&ctx->imdct_256, 8, 1);
+ ff_mdct_init(&ctx->imdct_512, 9, 1);
+ ac3_window_init(ctx->window);
+ dsputil_init(&ctx->dsp, avctx);
+ av_init_random(0, &ctx->dith_state);
+
+ /* set bias values for float to int16 conversion */
+ if(ctx->dsp.float_to_int16 == ff_float_to_int16_c) {
+ ctx->add_bias = 385.0f;
+ ctx->mul_bias = 1.0f;
+ } else {
+ ctx->add_bias = 0.0f;
+ ctx->mul_bias = 32767.0f;
+ }
+
+ return 0;
+}
+
+/**
+ * Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream.
+ * GetBitContext within AC3DecodeContext must point to
+ * start of the synchronized ac3 bitstream.
+ */
+static int ac3_parse_header(AC3DecodeContext *ctx)
+{
+ AC3HeaderInfo hdr;
+ GetBitContext *gb = &ctx->gb;
+ float cmixlev, surmixlev;
+ int err, i;
+
+ err = ff_ac3_parse_header(gb->buffer, &hdr);
+ if(err)
+ return err;
+
+ /* get decoding parameters from header info */
+ ctx->bit_alloc_params.fscod = hdr.fscod;
+ ctx->acmod = hdr.acmod;
+ cmixlev = gain_levels[clevs[hdr.cmixlev]];
+ surmixlev = gain_levels[slevs[hdr.surmixlev]];
+ ctx->dsurmod = hdr.dsurmod;
+ ctx->lfeon = hdr.lfeon;
+ ctx->bit_alloc_params.halfratecod = hdr.halfratecod;
+ ctx->sampling_rate = hdr.sample_rate;
+ ctx->bit_rate = hdr.bit_rate;
+ ctx->nchans = hdr.channels;
+ ctx->nfchans = ctx->nchans - ctx->lfeon;
+ ctx->lfe_ch = ctx->nfchans + 1;
+ ctx->frame_size = hdr.frame_size;
+
+ /* set default output to all source channels */
+ ctx->out_channels = ctx->nchans;
+ ctx->output_mode = ctx->acmod;
+ if(ctx->lfeon)
+ ctx->output_mode |= AC3_OUTPUT_LFEON;
+
+ /* skip over portion of header which has already been read */
+ skip_bits(gb, 16); // skip the sync_word
+ skip_bits(gb, 16); // skip crc1
+ skip_bits(gb, 8); // skip fscod and frmsizecod
+ skip_bits(gb, 11); // skip bsid, bsmod, and acmod
+ if(ctx->acmod == AC3_ACMOD_STEREO) {
+ skip_bits(gb, 2); // skip dsurmod
+ } else {
+ if((ctx->acmod & 1) && ctx->acmod != AC3_ACMOD_MONO)
+ skip_bits(gb, 2); // skip cmixlev
+ if(ctx->acmod & 4)
+ skip_bits(gb, 2); // skip surmixlev
+ }
+ skip_bits1(gb); // skip lfeon
+
+ /* read the rest of the bsi. read twice for dual mono mode. */
+ i = !(ctx->acmod);
+ do {
+ ctx->dialnorm[i] = dialnorm_tbl[get_bits(gb, 5)]; // dialogue normalization
+ if (get_bits1(gb))
+ skip_bits(gb, 8); //skip compression
+ if (get_bits1(gb))
+ skip_bits(gb, 8); //skip language code
+ if (get_bits1(gb))
+ skip_bits(gb, 7); //skip audio production information
+ } while (i--);
+
+ skip_bits(gb, 2); //skip copyright bit and original bitstream bit
+
+ /* skip the timecodes (or extra bitstream information for Alternate Syntax)
+ TODO: read & use the xbsi1 downmix levels */
+ if (get_bits1(gb))
+ skip_bits(gb, 14); //skip timecode1 / xbsi1
+ if (get_bits1(gb))
+ skip_bits(gb, 14); //skip timecode2 / xbsi2
+
+ /* skip additional bitstream info */
+ if (get_bits1(gb)) {
+ i = get_bits(gb, 6);
+ do {
+ skip_bits(gb, 8);
+ } while(i--);
+ }
+
+ /* set stereo downmixing coefficients
+ reference: Section 7.8.2 Downmixing Into Two Channels */
+ for(i=0; i<ctx->nfchans; i++) {
+ ctx->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[ctx->acmod][i][0]];
+ ctx->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[ctx->acmod][i][1]];
+ }
+ if(ctx->acmod > 1 && ctx->acmod & 1) {
+ ctx->downmix_coeffs[1][0] = ctx->downmix_coeffs[1][1] = cmixlev;
+ }
+ if(ctx->acmod == AC3_ACMOD_2F1R || ctx->acmod == AC3_ACMOD_3F1R) {
+ int nf = ctx->acmod - 2;
+ ctx->downmix_coeffs[nf][0] = ctx->downmix_coeffs[nf][1] = surmixlev * LEVEL_MINUS_3DB;
+ }
+ if(ctx->acmod == AC3_ACMOD_2F2R || ctx->acmod == AC3_ACMOD_3F2R) {
+ int nf = ctx->acmod - 4;
+ ctx->downmix_coeffs[nf][0] = ctx->downmix_coeffs[nf+1][1] = surmixlev;
+ }
+
+ return 0;
+}
+
+/**
+ * Decode the grouped exponents according to exponent strategy.
+ * reference: Section 7.1.3 Exponent Decoding
+ */
+static void decode_exponents(GetBitContext *gb, int expstr, int ngrps,
+ uint8_t absexp, int8_t *dexps)
+{
+ int i, j, grp, grpsize;
+ int dexp[256];
+ int expacc, prevexp;
+
+ /* unpack groups */
+ grpsize = expstr + (expstr == EXP_D45);
+ for(grp=0,i=0; grp<ngrps; grp++) {
+ expacc = get_bits(gb, 7);
+ dexp[i++] = exp_ungroup_tbl[expacc][0];
+ dexp[i++] = exp_ungroup_tbl[expacc][1];
+ dexp[i++] = exp_ungroup_tbl[expacc][2];
+ }
+
+ /* convert to absolute exps and expand groups */
+ prevexp = absexp;
+ for(i=0; i<ngrps*3; i++) {
+ prevexp = av_clip(prevexp + dexp[i]-2, 0, 24);
+ for(j=0; j<grpsize; j++) {
+ dexps[(i*grpsize)+j] = prevexp;
+ }
+ }
+}
+
+/**
+ * Generate transform coefficients for each coupled channel in the coupling
+ * range using the coupling coefficients and coupling coordinates.
+ * reference: Section 7.4.3 Coupling Coordinate Format
+ */
+static void uncouple_channels(AC3DecodeContext *ctx)
+{
+ int i, j, ch, bnd, subbnd;
+
+ subbnd = -1;
+ i = ctx->startmant[CPL_CH];
+ for(bnd=0; bnd<ctx->ncplbnd; bnd++) {
+ do {
+ subbnd++;
+ for(j=0; j<12; j++) {
+ for(ch=1; ch<=ctx->nfchans; ch++) {
+ if(ctx->chincpl[ch])
+ ctx->transform_coeffs[ch][i] = ctx->transform_coeffs[CPL_CH][i] * ctx->cplco[ch][bnd] * 8.0f;
+ }
+ i++;
+ }
+ } while(ctx->cplbndstrc[subbnd]);
+ }
+}
+
+/**
+ * Grouped mantissas for 3-level 5-level and 11-level quantization
+ */
+typedef struct {
+ float b1_mant[3];
+ float b2_mant[3];
+ float b4_mant[2];
+ int b1ptr;
+ int b2ptr;
+ int b4ptr;
+} mant_groups;
+
+/**
+ * Get the transform coefficients for a particular channel
+ * reference: Section 7.3 Quantization and Decoding of Mantissas
+ */
+static int get_transform_coeffs_ch(AC3DecodeContext *ctx, int ch_index, mant_groups *m)
+{
+ GetBitContext *gb = &ctx->gb;
+ int i, gcode, tbap, start, end;
+ uint8_t *exps;
+ uint8_t *bap;
+ float *coeffs;
+
+ exps = ctx->dexps[ch_index];
+ bap = ctx->bap[ch_index];
+ coeffs = ctx->transform_coeffs[ch_index];
+ start = ctx->startmant[ch_index];
+ end = ctx->endmant[ch_index];
+
+ for (i = start; i < end; i++) {
+ tbap = bap[i];
+ switch (tbap) {
+ case 0:
+ coeffs[i] = ((av_random(&ctx->dith_state) & 0xFFFF) * LEVEL_MINUS_3DB) / 32768.0f;
+ break;
+
+ case 1:
+ if(m->b1ptr > 2) {
+ gcode = get_bits(gb, 5);
+ m->b1_mant[0] = b1_mantissas[gcode][0];
+ m->b1_mant[1] = b1_mantissas[gcode][1];
+ m->b1_mant[2] = b1_mantissas[gcode][2];
+ m->b1ptr = 0;
+ }
+ coeffs[i] = m->b1_mant[m->b1ptr++];
+ break;
+
+ case 2:
+ if(m->b2ptr > 2) {
+ gcode = get_bits(gb, 7);
+ m->b2_mant[0] = b2_mantissas[gcode][0];
+ m->b2_mant[1] = b2_mantissas[gcode][1];
+ m->b2_mant[2] = b2_mantissas[gcode][2];
+ m->b2ptr = 0;
+ }
+ coeffs[i] = m->b2_mant[m->b2ptr++];
+ break;
+
+ case 3:
+ coeffs[i] = b3_mantissas[get_bits(gb, 3)];
+ break;
+
+ case 4:
+ if(m->b4ptr > 1) {
+ gcode = get_bits(gb, 7);
+ m->b4_mant[0] = b4_mantissas[gcode][0];
+ m->b4_mant[1] = b4_mantissas[gcode][1];
+ m->b4ptr = 0;
+ }
+ coeffs[i] = m->b4_mant[m->b4ptr++];
+ break;
+
+ case 5:
+ coeffs[i] = b5_mantissas[get_bits(gb, 4)];
+ break;
+
+ default:
+ /* asymmetric dequantization */
+ coeffs[i] = get_sbits(gb, qntztab[tbap]) * scale_factors[qntztab[tbap]-1];
+ break;
+ }
+ coeffs[i] *= scale_factors[exps[i]];
+ }
+
+ return 0;
+}
+
+/**
+ * Remove random dithering from coefficients with zero-bit mantissas
+ * reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0)
+ */
+static void remove_dithering(AC3DecodeContext *ctx) {
+ int ch, i;
+ int end=0;
+ float *coeffs;
+ uint8_t *bap;
+
+ for(ch=1; ch<=ctx->nfchans; ch++) {
+ if(!ctx->dithflag[ch]) {
+ coeffs = ctx->transform_coeffs[ch];
+ bap = ctx->bap[ch];
+ if(ctx->chincpl[ch])
+ end = ctx->startmant[CPL_CH];
+ else
+ end = ctx->endmant[ch];
+ for(i=0; i<end; i++) {
+ if(bap[i] == 0)
+ coeffs[i] = 0.0f;
+ }
+ if(ctx->chincpl[ch]) {
+ bap = ctx->bap[CPL_CH];
+ for(; i<ctx->endmant[CPL_CH]; i++) {
+ if(bap[i] == 0)
+ coeffs[i] = 0.0f;
+ }
+ }
+ }
+ }
+}
+
+/**
+ * Get the transform coefficients.
+ */
+static int get_transform_coeffs(AC3DecodeContext * ctx)
+{
+ int ch, end;
+ int got_cplchan = 0;
+ mant_groups m;
+
+ m.b1ptr = m.b2ptr = m.b4ptr = 3;
+
+ for (ch = 1; ch <= ctx->nchans; ch++) {
+ /* transform coefficients for full-bandwidth channel */
+ if (get_transform_coeffs_ch(ctx, ch, &m))
+ return -1;
+ /* tranform coefficients for coupling channel come right after the
+ coefficients for the first coupled channel*/
+ if (ctx->chincpl[ch]) {
+ if (!got_cplchan) {
+ if (get_transform_coeffs_ch(ctx, CPL_CH, &m)) {
+ av_log(ctx->avctx, AV_LOG_ERROR, "error in decoupling channels\n");
+ return -1;
+ }
+ uncouple_channels(ctx);
+ got_cplchan = 1;
+ }
+ end = ctx->endmant[CPL_CH];
+ } else {
+ end = ctx->endmant[ch];
+ }
+ do
+ ctx->transform_coeffs[ch][end] = 0;
+ while(++end < 256);
+ }
+
+ /* if any channel doesn't use dithering, zero appropriate coefficients */
+ if(!ctx->dither_all)
+ remove_dithering(ctx);
+
+ return 0;
+}
+
+/**
+ * Stereo rematrixing.
+ * reference: Section 7.5.4 Rematrixing : Decoding Technique
+ */
+static void do_rematrixing(AC3DecodeContext *ctx)
+{
+ int bnd, i;
+ int end, bndend;
+ float tmp0, tmp1;
+
+ end = FFMIN(ctx->endmant[1], ctx->endmant[2]);
+
+ for(bnd=0; bnd<ctx->nrematbnd; bnd++) {
+ if(ctx->rematflg[bnd]) {
+ bndend = FFMIN(end, rematrix_band_tbl[bnd+1]);
+ for(i=rematrix_band_tbl[bnd]; i<bndend; i++) {
+ tmp0 = ctx->transform_coeffs[1][i];
+ tmp1 = ctx->transform_coeffs[2][i];
+ ctx->transform_coeffs[1][i] = tmp0 + tmp1;
+ ctx->transform_coeffs[2][i] = tmp0 - tmp1;
+ }
+ }
+ }
+}
+
+/**
+ * Perform the 256-point IMDCT
+ */
+static void do_imdct_256(AC3DecodeContext *ctx, int chindex)
+{
+ int i, k;
+ DECLARE_ALIGNED_16(float, x[128]);
+ FFTComplex z[2][64];
+ float *o_ptr = ctx->tmp_output;
+
+ for(i=0; i<2; i++) {
+ /* de-interleave coefficients */
+ for(k=0; k<128; k++) {
+ x[k] = ctx->transform_coeffs[chindex][2*k+i];
+ }
+
+ /* run standard IMDCT */
+ ctx->imdct_256.fft.imdct_calc(&ctx->imdct_256, o_ptr, x, ctx->tmp_imdct);
+
+ /* reverse the post-rotation & reordering from standard IMDCT */
+ for(k=0; k<32; k++) {
+ z[i][32+k].re = -o_ptr[128+2*k];
+ z[i][32+k].im = -o_ptr[2*k];
+ z[i][31-k].re = o_ptr[2*k+1];
+ z[i][31-k].im = o_ptr[128+2*k+1];
+ }
+ }
+
+ /* apply AC-3 post-rotation & reordering */
+ for(k=0; k<64; k++) {
+ o_ptr[ 2*k ] = -z[0][ k].im;
+ o_ptr[ 2*k+1] = z[0][63-k].re;
+ o_ptr[128+2*k ] = -z[0][ k].re;
+ o_ptr[128+2*k+1] = z[0][63-k].im;
+ o_ptr[256+2*k ] = -z[1][ k].re;
+ o_ptr[256+2*k+1] = z[1][63-k].im;
+ o_ptr[384+2*k ] = z[1][ k].im;
+ o_ptr[384+2*k+1] = -z[1][63-k].re;
+ }
+}
+
+/**
+ * Inverse MDCT Transform.
+ * Convert frequency domain coefficients to time-domain audio samples.
+ * reference: Section 7.9.4 Transformation Equations
+ */
+static inline void do_imdct(AC3DecodeContext *ctx)
+{
+ int ch;
+ int nchans;
+
+ /* Don't perform the IMDCT on the LFE channel unless it's used in the output */
+ nchans = ctx->nfchans;
+ if(ctx->output_mode & AC3_OUTPUT_LFEON)
+ nchans++;
+
+ for (ch=1; ch<=nchans; ch++) {
+ if (ctx->blksw[ch]) {
+ do_imdct_256(ctx, ch);
+ } else {
+ ctx->imdct_512.fft.imdct_calc(&ctx->imdct_512, ctx->tmp_output,
+ ctx->transform_coeffs[ch],
+ ctx->tmp_imdct);
+ }
+ /* For the first half of the block, apply the window, add the delay
+ from the previous block, and send to output */
+ ctx->dsp.vector_fmul_add_add(ctx->output[ch-1], ctx->tmp_output,
+ ctx->window, ctx->delay[ch-1], 0, 256, 1);
+ /* For the second half of the block, apply the window and store the
+ samples to delay, to be combined with the next block */
+ ctx->dsp.vector_fmul_reverse(ctx->delay[ch-1], ctx->tmp_output+256,
+ ctx->window, 256);
+ }
+}
+
+/**
+ * Downmix the output to mono or stereo.
+ */
+static void ac3_downmix(float samples[AC3_MAX_CHANNELS][256], int nfchans,
+ int output_mode, float coef[AC3_MAX_CHANNELS][2])
+{
+ int i, j;
+ float v0, v1, s0, s1;
+
+ for(i=0; i<256; i++) {
+ v0 = v1 = s0 = s1 = 0.0f;
+ for(j=0; j<nfchans; j++) {
+ v0 += samples[j][i] * coef[j][0];
+ v1 += samples[j][i] * coef[j][1];
+ s0 += coef[j][0];
+ s1 += coef[j][1];
+ }
+ v0 /= s0;
+ v1 /= s1;
+ if(output_mode == AC3_ACMOD_MONO) {
+ samples[0][i] = (v0 + v1) * LEVEL_MINUS_3DB;
+ } else if(output_mode == AC3_ACMOD_STEREO) {
+ samples[0][i] = v0;
+ samples[1][i] = v1;
+ }
+ }
+}
+
+/**
+ * Parse an audio block from AC-3 bitstream.
+ */
+static int ac3_parse_audio_block(AC3DecodeContext *ctx, int blk)
+{
+ int nfchans = ctx->nfchans;
+ int acmod = ctx->acmod;
+ int i, bnd, seg, ch;
+ GetBitContext *gb = &ctx->gb;
+ uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
+
+ memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
+
+ /* block switch flags */
+ for (ch = 1; ch <= nfchans; ch++)
+ ctx->blksw[ch] = get_bits1(gb);
+
+ /* dithering flags */
+ ctx->dither_all = 1;
+ for (ch = 1; ch <= nfchans; ch++) {
+ ctx->dithflag[ch] = get_bits1(gb);
+ if(!ctx->dithflag[ch])
+ ctx->dither_all = 0;
+ }
+
+ /* dynamic range */
+ i = !(ctx->acmod);
+ do {
+ if(get_bits1(gb)) {
+ ctx->dynrng[i] = dynrng_tbl[get_bits(gb, 8)];
+ } else if(blk == 0) {
+ ctx->dynrng[i] = 1.0f;
+ }
+ } while(i--);
+
+ /* coupling strategy */
+ if (get_bits1(gb)) {
+ memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
+ ctx->cplinu = get_bits1(gb);
+ if (ctx->cplinu) {
+ /* coupling in use */
+ int cplbegf, cplendf;
+
+ /* determine which channels are coupled */
+ for (ch = 1; ch <= nfchans; ch++)
+ ctx->chincpl[ch] = get_bits1(gb);
+
+ /* phase flags in use */
+ if (acmod == AC3_ACMOD_STEREO)
+ ctx->phsflginu = get_bits1(gb);
+
+ /* coupling frequency range and band structure */
+ cplbegf = get_bits(gb, 4);
+ cplendf = get_bits(gb, 4);
+ if (3 + cplendf - cplbegf < 0) {
+ av_log(ctx->avctx, AV_LOG_ERROR, "cplendf = %d < cplbegf = %d\n", cplendf, cplbegf);
+ return -1;
+ }
+ ctx->ncplbnd = ctx->ncplsubnd = 3 + cplendf - cplbegf;
+ ctx->startmant[CPL_CH] = cplbegf * 12 + 37;
+ ctx->endmant[CPL_CH] = cplendf * 12 + 73;
+ for (bnd = 0; bnd < ctx->ncplsubnd - 1; bnd++) {
+ if (get_bits1(gb)) {
+ ctx->cplbndstrc[bnd] = 1;
+ ctx->ncplbnd--;
+ }
+ }
+ } else {
+ /* coupling not in use */
+ for (ch = 1; ch <= nfchans; ch++)
+ ctx->chincpl[ch] = 0;
+ }
+ }
+
+ /* coupling coordinates */
+ if (ctx->cplinu) {
+ int cplcoe = 0;
+
+ for (ch = 1; ch <= nfchans; ch++) {
+ if (ctx->chincpl[ch]) {
+ if (get_bits1(gb)) {
+ int mstrcplco, cplcoexp, cplcomant;
+ cplcoe = 1;
+ mstrcplco = 3 * get_bits(gb, 2);
+ for (bnd = 0; bnd < ctx->ncplbnd; bnd++) {
+ cplcoexp = get_bits(gb, 4);
+ cplcomant = get_bits(gb, 4);
+ if (cplcoexp == 15)
+ ctx->cplco[ch][bnd] = cplcomant / 16.0f;
+ else
+ ctx->cplco[ch][bnd] = (cplcomant + 16.0f) / 32.0f;
+ ctx->cplco[ch][bnd] *= scale_factors[cplcoexp + mstrcplco];
+ }
+ }
+ }
+ }
+ /* phase flags */
+ if (acmod == AC3_ACMOD_STEREO && ctx->phsflginu && cplcoe) {
+ for (bnd = 0; bnd < ctx->ncplbnd; bnd++) {
+ if (get_bits1(gb))
+ ctx->cplco[2][bnd] = -ctx->cplco[2][bnd];
+ }
+ }
+ }
+
+ /* stereo rematrixing strategy and band structure */
+ if (acmod == AC3_ACMOD_STEREO) {
+ ctx->rematstr = get_bits1(gb);
+ if (ctx->rematstr) {
+ ctx->nrematbnd = 4;
+ if(ctx->cplinu && ctx->startmant[CPL_CH] <= 61)
+ ctx->nrematbnd -= 1 + (ctx->startmant[CPL_CH] == 37);
+ for(bnd=0; bnd<ctx->nrematbnd; bnd++)
+ ctx->rematflg[bnd] = get_bits1(gb);
+ }
+ }
+
+ /* exponent strategies for each channel */
+ ctx->expstr[CPL_CH] = EXP_REUSE;
+ ctx->expstr[ctx->lfe_ch] = EXP_REUSE;
+ for (ch = !ctx->cplinu; ch <= ctx->nchans; ch++) {
+ if(ch == ctx->lfe_ch)
+ ctx->expstr[ch] = get_bits(gb, 1);
+ else
+ ctx->expstr[ch] = get_bits(gb, 2);
+ if(ctx->expstr[ch] != EXP_REUSE)
+ bit_alloc_stages[ch] = 3;
+ }
+
+ /* channel bandwidth */
+ for (ch = 1; ch <= nfchans; ch++) {
+ ctx->startmant[ch] = 0;
+ if (ctx->expstr[ch] != EXP_REUSE) {
+ int prev = ctx->endmant[ch];
+ if (ctx->chincpl[ch])
+ ctx->endmant[ch] = ctx->startmant[CPL_CH];
+ else {
+ int chbwcod = get_bits(gb, 6);
+ if (chbwcod > 60) {
+ av_log(ctx->avctx, AV_LOG_ERROR, "chbwcod = %d > 60", chbwcod);
+ return -1;
+ }
+ ctx->endmant[ch] = chbwcod * 3 + 73;
+ }
+ if(blk > 0 && ctx->endmant[ch] != prev)
+ memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
+ }
+ }
+ ctx->startmant[ctx->lfe_ch] = 0;
+ ctx->endmant[ctx->lfe_ch] = 7;
+
+ /* decode exponents for each channel */
+ for (ch = !ctx->cplinu; ch <= ctx->nchans; ch++) {
+ if (ctx->expstr[ch] != EXP_REUSE) {
+ int grpsize, ngrps;
+ grpsize = 3 << (ctx->expstr[ch] - 1);
+ if(ch == CPL_CH)
+ ngrps = (ctx->endmant[ch] - ctx->startmant[ch]) / grpsize;
+ else if(ch == ctx->lfe_ch)
+ ngrps = 2;
+ else
+ ngrps = (ctx->endmant[ch] + grpsize - 4) / grpsize;
+ ctx->dexps[ch][0] = get_bits(gb, 4) << !ch;
+ decode_exponents(gb, ctx->expstr[ch], ngrps, ctx->dexps[ch][0],
+ &ctx->dexps[ch][ctx->startmant[ch]+!!ch]);
+ if(ch != CPL_CH && ch != ctx->lfe_ch)
+ skip_bits(gb, 2); /* skip gainrng */
+ }
+ }
+
+ /* bit allocation information */
+ if (get_bits1(gb)) {
+ ctx->bit_alloc_params.sdecay = ff_sdecaytab[get_bits(gb, 2)];
+ ctx->bit_alloc_params.fdecay = ff_fdecaytab[get_bits(gb, 2)];
+ ctx->bit_alloc_params.sgain = ff_sgaintab[get_bits(gb, 2)];
+ ctx->bit_alloc_params.dbknee = ff_dbkneetab[get_bits(gb, 2)];
+ ctx->bit_alloc_params.floor = ff_floortab[get_bits(gb, 3)];
+ for(ch=!ctx->cplinu; ch<=ctx->nchans; ch++) {
+ bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
+ }
+ }
+
+ /* signal-to-noise ratio offsets and fast gains (signal-to-mask ratios) */
+ if (get_bits1(gb)) {
+ int csnr;
+ csnr = (get_bits(gb, 6) - 15) << 4;
+ for (ch = !ctx->cplinu; ch <= ctx->nchans; ch++) { /* snr offset and fast gain */
+ ctx->snroffst[ch] = (csnr + get_bits(gb, 4)) << 2;
+ ctx->fgain[ch] = ff_fgaintab[get_bits(gb, 3)];
+ }
+ memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
+ }
+
+ /* coupling leak information */
+ if (ctx->cplinu && get_bits1(gb)) {
+ ctx->bit_alloc_params.cplfleak = get_bits(gb, 3);
+ ctx->bit_alloc_params.cplsleak = get_bits(gb, 3);
+ bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2);
+ }
+
+ /* delta bit allocation information */
+ if (get_bits1(gb)) {
+ /* delta bit allocation exists (strategy) */
+ for (ch = !ctx->cplinu; ch <= nfchans; ch++) {
+ ctx->deltbae[ch] = get_bits(gb, 2);
+ if (ctx->deltbae[ch] == DBA_RESERVED) {
+ av_log(ctx->avctx, AV_LOG_ERROR, "delta bit allocation strategy reserved\n");
+ return -1;
+ }
+ bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
+ }
+ /* channel delta offset, len and bit allocation */
+ for (ch = !ctx->cplinu; ch <= nfchans; ch++) {
+ if (ctx->deltbae[ch] == DBA_NEW) {
+ ctx->deltnseg[ch] = get_bits(gb, 3);
+ for (seg = 0; seg <= ctx->deltnseg[ch]; seg++) {
+ ctx->deltoffst[ch][seg] = get_bits(gb, 5);
+ ctx->deltlen[ch][seg] = get_bits(gb, 4);
+ ctx->deltba[ch][seg] = get_bits(gb, 3);
+ }
+ }
+ }
+ } else if(blk == 0) {
+ for(ch=0; ch<=ctx->nchans; ch++) {
+ ctx->deltbae[ch] = DBA_NONE;
+ }
+ }
+
+ /* Bit allocation */
+ for(ch=!ctx->cplinu; ch<=ctx->nchans; ch++) {
+ if(bit_alloc_stages[ch] > 2) {
+ /* Exponent mapping into PSD and PSD integration */
+ ff_ac3_bit_alloc_calc_psd(ctx->dexps[ch],
+ ctx->startmant[ch], ctx->endmant[ch],
+ ctx->psd[ch], ctx->bndpsd[ch]);
+ }
+ if(bit_alloc_stages[ch] > 1) {
+ /* Compute excitation function, Compute masking curve, and
+ Apply delta bit allocation */
+ ff_ac3_bit_alloc_calc_mask(&ctx->bit_alloc_params, ctx->bndpsd[ch],
+ ctx->startmant[ch], ctx->endmant[ch],
+ ctx->fgain[ch], (ch == ctx->lfe_ch),
+ ctx->deltbae[ch], ctx->deltnseg[ch],
+ ctx->deltoffst[ch], ctx->deltlen[ch],
+ ctx->deltba[ch], ctx->mask[ch]);
+ }
+ if(bit_alloc_stages[ch] > 0) {
+ /* Compute bit allocation */
+ ff_ac3_bit_alloc_calc_bap(ctx->mask[ch], ctx->psd[ch],
+ ctx->startmant[ch], ctx->endmant[ch],
+ ctx->snroffst[ch],
+ ctx->bit_alloc_params.floor,
+ ctx->bap[ch]);
+ }
+ }
+
+ /* unused dummy data */
+ if (get_bits1(gb)) {
+ int skipl = get_bits(gb, 9);
+ while(skipl--)
+ skip_bits(gb, 8);
+ }
+
+ /* unpack the transform coefficients
+ this also uncouples channels if coupling is in use. */
+ if (get_transform_coeffs(ctx)) {
+ av_log(ctx->avctx, AV_LOG_ERROR, "Error in routine get_transform_coeffs\n");
+ return -1;
+ }
+
+ /* recover coefficients if rematrixing is in use */
+ if(ctx->acmod == AC3_ACMOD_STEREO)
+ do_rematrixing(ctx);
+
+ /* apply scaling to coefficients (headroom, dialnorm, dynrng) */
+ for(ch=1; ch<=ctx->nchans; ch++) {
+ float gain = 2.0f * ctx->mul_bias;
+ if(ctx->acmod == AC3_ACMOD_DUALMONO) {
+ gain *= ctx->dialnorm[ch-1] * ctx->dynrng[ch-1];
+ } else {
+ gain *= ctx->dialnorm[0] * ctx->dynrng[0];
+ }
+ for(i=0; i<ctx->endmant[ch]; i++) {
+ ctx->transform_coeffs[ch][i] *= gain;
+ }
+ }
+
+ do_imdct(ctx);
+
+ /* downmix output if needed */
+ if(ctx->nchans != ctx->out_channels && !((ctx->output_mode & AC3_OUTPUT_LFEON) &&
+ ctx->nfchans == ctx->out_channels)) {
+ ac3_downmix(ctx->output, ctx->nfchans, ctx->output_mode,
+ ctx->downmix_coeffs);
+ }
+
+ /* convert float to 16-bit integer */
+ for(ch=0; ch<ctx->out_channels; ch++) {
+ for(i=0; i<256; i++) {
+ ctx->output[ch][i] += ctx->add_bias;
+ }
+ ctx->dsp.float_to_int16(ctx->int_output[ch], ctx->output[ch], 256);
+ }
+
+ return 0;
+}
+
+/**
+ * Decode a single AC-3 frame.
+ */
+static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, uint8_t *buf, int buf_size)
+{
+ AC3DecodeContext *ctx = (AC3DecodeContext *)avctx->priv_data;
+ int16_t *out_samples = (int16_t *)data;
+ int i, blk, ch;
+
+ /* initialize the GetBitContext with the start of valid AC-3 Frame */
+ init_get_bits(&ctx->gb, buf, buf_size * 8);
+
+ /* parse the syncinfo */
+ if (ac3_parse_header(ctx)) {
+ av_log(avctx, AV_LOG_ERROR, "\n");
+ *data_size = 0;
+ return buf_size;
+ }
+
+ avctx->sample_rate = ctx->sampling_rate;
+ avctx->bit_rate = ctx->bit_rate;
+
+ /* channel config */
+ ctx->out_channels = ctx->nchans;
+ if (avctx->channels == 0) {
+ avctx->channels = ctx->out_channels;
+ } else if(ctx->out_channels < avctx->channels) {
+ av_log(avctx, AV_LOG_ERROR, "Cannot upmix AC3 from %d to %d channels.\n",
+ ctx->out_channels, avctx->channels);
+ return -1;
+ }
+ if(avctx->channels == 2) {
+ ctx->output_mode = AC3_ACMOD_STEREO;
+ } else if(avctx->channels == 1) {
+ ctx->output_mode = AC3_ACMOD_MONO;
+ } else if(avctx->channels != ctx->out_channels) {
+ av_log(avctx, AV_LOG_ERROR, "Cannot downmix AC3 from %d to %d channels.\n",
+ ctx->out_channels, avctx->channels);
+ return -1;
+ }
+ ctx->out_channels = avctx->channels;
+
+ /* parse the audio blocks */
+ for (blk = 0; blk < NB_BLOCKS; blk++) {
+ if (ac3_parse_audio_block(ctx, blk)) {
+ av_log(avctx, AV_LOG_ERROR, "error parsing the audio block\n");
+ *data_size = 0;
+ return ctx->frame_size;
+ }
+ for (i = 0; i < 256; i++)
+ for (ch = 0; ch < ctx->out_channels; ch++)
+ *(out_samples++) = ctx->int_output[ch][i];
+ }
+ *data_size = NB_BLOCKS * 256 * avctx->channels * sizeof (int16_t);
+ return ctx->frame_size;
+}
+
+/**
+ * Uninitialize the AC-3 decoder.
+ */
+static int ac3_decode_end(AVCodecContext *avctx)
+{
+ AC3DecodeContext *ctx = (AC3DecodeContext *)avctx->priv_data;
+ ff_mdct_end(&ctx->imdct_512);
+ ff_mdct_end(&ctx->imdct_256);
+
+ return 0;
+}
+
+AVCodec ac3_decoder = {
+ .name = "ac3",
+ .type = CODEC_TYPE_AUDIO,
+ .id = CODEC_ID_AC3,
+ .priv_data_size = sizeof (AC3DecodeContext),
+ .init = ac3_decode_init,
+ .close = ac3_decode_end,
+ .decode = ac3_decode_frame,
+};
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