282255bbd2
Originally committed as revision 25984 to svn://svn.ffmpeg.org/ffmpeg/trunk
1490 lines
46 KiB
C
1490 lines
46 KiB
C
/*
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* The simplest AC-3 encoder
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* Copyright (c) 2000 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* The simplest AC-3 encoder.
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*/
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//#define DEBUG
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#include "libavcore/audioconvert.h"
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#include "libavutil/crc.h"
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#include "avcodec.h"
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#include "put_bits.h"
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#include "ac3.h"
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#include "audioconvert.h"
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#define MDCT_NBITS 9
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#define MDCT_SAMPLES (1 << MDCT_NBITS)
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/** Scale a float value by 2^bits and convert to an integer. */
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#define SCALE_FLOAT(a, bits) lrintf((a) * (float)(1 << (bits)))
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/** Scale a float value by 2^15, convert to an integer, and clip to int16_t range. */
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#define FIX15(a) av_clip_int16(SCALE_FLOAT(a, 15))
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/**
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* Compex number.
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* Used in fixed-point MDCT calculation.
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*/
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typedef struct IComplex {
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int16_t re,im;
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} IComplex;
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/**
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* AC-3 encoder private context.
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*/
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typedef struct AC3EncodeContext {
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PutBitContext pb; ///< bitstream writer context
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int bitstream_id; ///< bitstream id (bsid)
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int bitstream_mode; ///< bitstream mode (bsmod)
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int bit_rate; ///< target bit rate, in bits-per-second
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int sample_rate; ///< sampling frequency, in Hz
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int frame_size_min; ///< minimum frame size in case rounding is necessary
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int frame_size; ///< current frame size in bytes
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int frame_size_code; ///< frame size code (frmsizecod)
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int bits_written; ///< bit count (used to avg. bitrate)
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int samples_written; ///< sample count (used to avg. bitrate)
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int fbw_channels; ///< number of full-bandwidth channels (nfchans)
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int channels; ///< total number of channels (nchans)
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int lfe_on; ///< indicates if there is an LFE channel (lfeon)
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int lfe_channel; ///< channel index of the LFE channel
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int channel_mode; ///< channel mode (acmod)
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const uint8_t *channel_map; ///< channel map used to reorder channels
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int bandwidth_code[AC3_MAX_CHANNELS]; ///< bandwidth code (0 to 60) (chbwcod)
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int nb_coefs[AC3_MAX_CHANNELS];
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/* bitrate allocation control */
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int slow_gain_code; ///< slow gain code (sgaincod)
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int slow_decay_code; ///< slow decay code (sdcycod)
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int fast_decay_code; ///< fast decay code (fdcycod)
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int db_per_bit_code; ///< dB/bit code (dbpbcod)
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int floor_code; ///< floor code (floorcod)
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AC3BitAllocParameters bit_alloc; ///< bit allocation parameters
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int coarse_snr_offset; ///< coarse SNR offsets (csnroffst)
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int fast_gain_code[AC3_MAX_CHANNELS]; ///< fast gain codes (signal-to-mask ratio) (fgaincod)
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int fine_snr_offset[AC3_MAX_CHANNELS]; ///< fine SNR offsets (fsnroffst)
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/* mantissa encoding */
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int mant1_cnt, mant2_cnt, mant4_cnt; ///< mantissa counts for bap=1,2,4
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int16_t last_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< last 256 samples from previous frame
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} AC3EncodeContext;
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/** MDCT and FFT tables */
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static int16_t costab[64];
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static int16_t sintab[64];
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static int16_t xcos1[128];
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static int16_t xsin1[128];
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/**
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* Initialize FFT tables.
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* @param ln log2(FFT size)
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*/
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static av_cold void fft_init(int ln)
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{
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int i, n, n2;
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float alpha;
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n = 1 << ln;
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n2 = n >> 1;
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for (i = 0; i < n2; i++) {
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alpha = 2.0 * M_PI * i / n;
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costab[i] = FIX15(cos(alpha));
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sintab[i] = FIX15(sin(alpha));
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}
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}
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/**
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* Initialize MDCT tables.
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* @param nbits log2(MDCT size)
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*/
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static av_cold void mdct_init(int nbits)
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{
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int i, n, n4;
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n = 1 << nbits;
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n4 = n >> 2;
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fft_init(nbits - 2);
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for (i = 0; i < n4; i++) {
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float alpha = 2.0 * M_PI * (i + 1.0 / 8.0) / n;
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xcos1[i] = FIX15(-cos(alpha));
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xsin1[i] = FIX15(-sin(alpha));
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}
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}
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/** Butterfly op */
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#define BF(pre, pim, qre, qim, pre1, pim1, qre1, qim1) \
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{ \
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int ax, ay, bx, by; \
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bx = pre1; \
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by = pim1; \
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ax = qre1; \
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ay = qim1; \
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pre = (bx + ax) >> 1; \
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pim = (by + ay) >> 1; \
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qre = (bx - ax) >> 1; \
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qim = (by - ay) >> 1; \
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}
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/** Complex multiply */
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#define CMUL(pre, pim, are, aim, bre, bim) \
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{ \
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pre = (MUL16(are, bre) - MUL16(aim, bim)) >> 15; \
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pim = (MUL16(are, bim) + MUL16(bre, aim)) >> 15; \
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}
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/**
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* Calculate a 2^n point complex FFT on 2^ln points.
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* @param z complex input/output samples
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* @param ln log2(FFT size)
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*/
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static void fft(IComplex *z, int ln)
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{
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int j, l, np, np2;
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int nblocks, nloops;
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register IComplex *p,*q;
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int tmp_re, tmp_im;
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np = 1 << ln;
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/* reverse */
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for (j = 0; j < np; j++) {
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int k = av_reverse[j] >> (8 - ln);
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if (k < j)
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FFSWAP(IComplex, z[k], z[j]);
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}
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/* pass 0 */
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p = &z[0];
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j = np >> 1;
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do {
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BF(p[0].re, p[0].im, p[1].re, p[1].im,
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p[0].re, p[0].im, p[1].re, p[1].im);
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p += 2;
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} while (--j);
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/* pass 1 */
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p = &z[0];
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j = np >> 2;
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do {
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BF(p[0].re, p[0].im, p[2].re, p[2].im,
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p[0].re, p[0].im, p[2].re, p[2].im);
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BF(p[1].re, p[1].im, p[3].re, p[3].im,
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p[1].re, p[1].im, p[3].im, -p[3].re);
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p+=4;
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} while (--j);
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/* pass 2 .. ln-1 */
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nblocks = np >> 3;
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nloops = 1 << 2;
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np2 = np >> 1;
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do {
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p = z;
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q = z + nloops;
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for (j = 0; j < nblocks; j++) {
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BF(p->re, p->im, q->re, q->im,
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p->re, p->im, q->re, q->im);
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p++;
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q++;
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for(l = nblocks; l < np2; l += nblocks) {
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CMUL(tmp_re, tmp_im, costab[l], -sintab[l], q->re, q->im);
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BF(p->re, p->im, q->re, q->im,
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p->re, p->im, tmp_re, tmp_im);
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p++;
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q++;
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}
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p += nloops;
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q += nloops;
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}
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nblocks = nblocks >> 1;
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nloops = nloops << 1;
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} while (nblocks);
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}
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/**
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* Calculate a 512-point MDCT
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* @param out 256 output frequency coefficients
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* @param in 512 windowed input audio samples
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*/
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static void mdct512(int32_t *out, int16_t *in)
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{
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int i, re, im, re1, im1;
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int16_t rot[MDCT_SAMPLES];
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IComplex x[MDCT_SAMPLES/4];
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/* shift to simplify computations */
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for (i = 0; i < MDCT_SAMPLES/4; i++)
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rot[i] = -in[i + 3*MDCT_SAMPLES/4];
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for (;i < MDCT_SAMPLES; i++)
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rot[i] = in[i - MDCT_SAMPLES/4];
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/* pre rotation */
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for (i = 0; i < MDCT_SAMPLES/4; i++) {
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re = ((int)rot[ 2*i] - (int)rot[MDCT_SAMPLES -1-2*i]) >> 1;
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im = -((int)rot[MDCT_SAMPLES/2+2*i] - (int)rot[MDCT_SAMPLES/2-1-2*i]) >> 1;
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CMUL(x[i].re, x[i].im, re, im, -xcos1[i], xsin1[i]);
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}
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fft(x, MDCT_NBITS - 2);
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/* post rotation */
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for (i = 0; i < MDCT_SAMPLES/4; i++) {
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re = x[i].re;
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im = x[i].im;
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CMUL(re1, im1, re, im, xsin1[i], xcos1[i]);
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out[ 2*i] = im1;
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out[MDCT_SAMPLES/2-1-2*i] = re1;
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}
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}
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/**
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* Calculate the log2() of the maximum absolute value in an array.
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* @param tab input array
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* @param n number of values in the array
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* @return log2(max(abs(tab[])))
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*/
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static int log2_tab(int16_t *tab, int n)
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{
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int i, v;
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v = 0;
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for (i = 0; i < n; i++)
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v |= abs(tab[i]);
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return av_log2(v);
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}
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/**
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* Left-shift each value in an array by a specified amount.
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* @param tab input array
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* @param n number of values in the array
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* @param lshift left shift amount. a negative value means right shift.
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*/
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static void lshift_tab(int16_t *tab, int n, int lshift)
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{
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int i;
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if (lshift > 0) {
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for(i = 0; i < n; i++)
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tab[i] <<= lshift;
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} else if (lshift < 0) {
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lshift = -lshift;
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for (i = 0; i < n; i++)
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tab[i] >>= lshift;
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}
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}
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/**
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* Calculate the sum of absolute differences (SAD) between 2 sets of exponents.
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*/
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static int calc_exp_diff(uint8_t *exp1, uint8_t *exp2, int n)
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{
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int sum, i;
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sum = 0;
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for (i = 0; i < n; i++)
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sum += abs(exp1[i] - exp2[i]);
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return sum;
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}
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/**
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* Exponent Difference Threshold.
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* New exponents are sent if their SAD exceed this number.
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*/
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#define EXP_DIFF_THRESHOLD 1000
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/**
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* Calculate exponent strategies for all blocks in a single channel.
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*/
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static void compute_exp_strategy_ch(uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
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uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
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int ch, int is_lfe)
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{
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int blk, blk1;
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int exp_diff;
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/* estimate if the exponent variation & decide if they should be
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reused in the next frame */
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exp_strategy[0][ch] = EXP_NEW;
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for (blk = 1; blk < AC3_MAX_BLOCKS; blk++) {
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exp_diff = calc_exp_diff(exp[blk][ch], exp[blk-1][ch], AC3_MAX_COEFS);
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if (exp_diff > EXP_DIFF_THRESHOLD)
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exp_strategy[blk][ch] = EXP_NEW;
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else
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exp_strategy[blk][ch] = EXP_REUSE;
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}
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if (is_lfe)
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return;
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/* now select the encoding strategy type : if exponents are often
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recoded, we use a coarse encoding */
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blk = 0;
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while (blk < AC3_MAX_BLOCKS) {
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blk1 = blk + 1;
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while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1][ch] == EXP_REUSE)
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blk1++;
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switch (blk1 - blk) {
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case 1: exp_strategy[blk][ch] = EXP_D45; break;
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case 2:
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case 3: exp_strategy[blk][ch] = EXP_D25; break;
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default: exp_strategy[blk][ch] = EXP_D15; break;
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}
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blk = blk1;
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}
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}
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/**
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* Set each encoded exponent in a block to the minimum of itself and the
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* exponent in the same frequency bin of a following block.
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* exp[i] = min(exp[i], exp1[i]
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*/
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static void exponent_min(uint8_t exp[AC3_MAX_COEFS], uint8_t exp1[AC3_MAX_COEFS], int n)
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{
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int i;
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for (i = 0; i < n; i++) {
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if (exp1[i] < exp[i])
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exp[i] = exp1[i];
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}
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}
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/**
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* Update the exponents so that they are the ones the decoder will decode.
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* @return the number of bits used to encode the exponents.
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*/
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static int encode_exponents_blk_ch(uint8_t encoded_exp[AC3_MAX_COEFS],
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uint8_t exp[AC3_MAX_COEFS],
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int nb_exps, int exp_strategy)
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{
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int group_size, nb_groups, i, j, k, exp_min;
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uint8_t exp1[AC3_MAX_COEFS];
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group_size = exp_strategy + (exp_strategy == EXP_D45);
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nb_groups = ((nb_exps + (group_size * 3) - 4) / (3 * group_size)) * 3;
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/* for each group, compute the minimum exponent */
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exp1[0] = exp[0]; /* DC exponent is handled separately */
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k = 1;
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for (i = 1; i <= nb_groups; i++) {
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exp_min = exp[k];
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assert(exp_min >= 0 && exp_min <= 24);
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for (j = 1; j < group_size; j++) {
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if (exp[k+j] < exp_min)
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exp_min = exp[k+j];
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}
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exp1[i] = exp_min;
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k += group_size;
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}
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/* constraint for DC exponent */
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if (exp1[0] > 15)
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exp1[0] = 15;
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/* decrease the delta between each groups to within 2 so that they can be
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differentially encoded */
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for (i = 1; i <= nb_groups; i++)
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exp1[i] = FFMIN(exp1[i], exp1[i-1] + 2);
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for (i = nb_groups-1; i >= 0; i--)
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exp1[i] = FFMIN(exp1[i], exp1[i+1] + 2);
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/* now we have the exponent values the decoder will see */
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encoded_exp[0] = exp1[0];
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k = 1;
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for (i = 1; i <= nb_groups; i++) {
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for (j = 0; j < group_size; j++)
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encoded_exp[k+j] = exp1[i];
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k += group_size;
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}
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return 4 + (nb_groups / 3) * 7;
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}
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/**
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* Calculate the number of bits needed to encode a set of mantissas.
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*/
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static int compute_mantissa_size(AC3EncodeContext *s, uint8_t *m, int nb_coefs)
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{
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int bits, mant, i;
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bits = 0;
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for (i = 0; i < nb_coefs; i++) {
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mant = m[i];
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switch (mant) {
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case 0:
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/* nothing */
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break;
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case 1:
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/* 3 mantissa in 5 bits */
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if (s->mant1_cnt == 0)
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bits += 5;
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if (++s->mant1_cnt == 3)
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s->mant1_cnt = 0;
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break;
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case 2:
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/* 3 mantissa in 7 bits */
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if (s->mant2_cnt == 0)
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bits += 7;
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if (++s->mant2_cnt == 3)
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s->mant2_cnt = 0;
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break;
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case 3:
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bits += 3;
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break;
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case 4:
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/* 2 mantissa in 7 bits */
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if (s->mant4_cnt == 0)
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bits += 7;
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if (++s->mant4_cnt == 2)
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s->mant4_cnt = 0;
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break;
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case 14:
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bits += 14;
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break;
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case 15:
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bits += 16;
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break;
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default:
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bits += mant - 1;
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break;
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}
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}
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return bits;
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}
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/**
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* Calculate masking curve based on the final exponents.
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* Also calculate the power spectral densities to use in future calculations.
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*/
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static void bit_alloc_masking(AC3EncodeContext *s,
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uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
|
|
uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
|
|
int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
|
|
int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS])
|
|
{
|
|
int blk, ch;
|
|
int16_t band_psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS];
|
|
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
if(exp_strategy[blk][ch] == EXP_REUSE) {
|
|
memcpy(psd[blk][ch], psd[blk-1][ch], AC3_MAX_COEFS*sizeof(psd[0][0][0]));
|
|
memcpy(mask[blk][ch], mask[blk-1][ch], AC3_CRITICAL_BANDS*sizeof(mask[0][0][0]));
|
|
} else {
|
|
ff_ac3_bit_alloc_calc_psd(encoded_exp[blk][ch], 0,
|
|
s->nb_coefs[ch],
|
|
psd[blk][ch], band_psd[blk][ch]);
|
|
ff_ac3_bit_alloc_calc_mask(&s->bit_alloc, band_psd[blk][ch],
|
|
0, s->nb_coefs[ch],
|
|
ff_ac3_fast_gain_tab[s->fast_gain_code[ch]],
|
|
ch == s->lfe_channel,
|
|
DBA_NONE, 0, NULL, NULL, NULL,
|
|
mask[blk][ch]);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Run the bit allocation with a given SNR offset.
|
|
* This calculates the bit allocation pointers that will be used to determine
|
|
* the quantization of each mantissa.
|
|
* @return the number of remaining bits (positive or negative) if the given
|
|
* SNR offset is used to quantize the mantissas.
|
|
*/
|
|
static int bit_alloc(AC3EncodeContext *s,
|
|
int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS],
|
|
int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
|
|
uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
|
|
int frame_bits, int coarse_snr_offset, int fine_snr_offset)
|
|
{
|
|
int blk, ch;
|
|
int snr_offset;
|
|
|
|
snr_offset = (((coarse_snr_offset - 15) << 4) + fine_snr_offset) << 2;
|
|
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
s->mant1_cnt = 0;
|
|
s->mant2_cnt = 0;
|
|
s->mant4_cnt = 0;
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
ff_ac3_bit_alloc_calc_bap(mask[blk][ch], psd[blk][ch], 0,
|
|
s->nb_coefs[ch], snr_offset,
|
|
s->bit_alloc.floor, ff_ac3_bap_tab,
|
|
bap[blk][ch]);
|
|
frame_bits += compute_mantissa_size(s, bap[blk][ch], s->nb_coefs[ch]);
|
|
}
|
|
}
|
|
return 8 * s->frame_size - frame_bits;
|
|
}
|
|
|
|
|
|
#define SNR_INC1 4
|
|
|
|
/**
|
|
* Perform bit allocation search.
|
|
* Finds the SNR offset value that maximizes quality and fits in the specified
|
|
* frame size. Output is the SNR offset and a set of bit allocation pointers
|
|
* used to quantize the mantissas.
|
|
*/
|
|
static int compute_bit_allocation(AC3EncodeContext *s,
|
|
uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
|
|
uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
|
|
uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
|
|
int frame_bits)
|
|
{
|
|
int blk, ch;
|
|
int coarse_snr_offset, fine_snr_offset;
|
|
uint8_t bap1[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
|
|
int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
|
|
int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS];
|
|
static const int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 };
|
|
|
|
/* init default parameters */
|
|
s->slow_decay_code = 2;
|
|
s->fast_decay_code = 1;
|
|
s->slow_gain_code = 1;
|
|
s->db_per_bit_code = 2;
|
|
s->floor_code = 4;
|
|
for (ch = 0; ch < s->channels; ch++)
|
|
s->fast_gain_code[ch] = 4;
|
|
|
|
/* compute real values */
|
|
s->bit_alloc.slow_decay = ff_ac3_slow_decay_tab[s->slow_decay_code] >> s->bit_alloc.sr_shift;
|
|
s->bit_alloc.fast_decay = ff_ac3_fast_decay_tab[s->fast_decay_code] >> s->bit_alloc.sr_shift;
|
|
s->bit_alloc.slow_gain = ff_ac3_slow_gain_tab[s->slow_gain_code];
|
|
s->bit_alloc.db_per_bit = ff_ac3_db_per_bit_tab[s->db_per_bit_code];
|
|
s->bit_alloc.floor = ff_ac3_floor_tab[s->floor_code];
|
|
|
|
/* header size */
|
|
frame_bits += 65;
|
|
// if (s->channel_mode == 2)
|
|
// frame_bits += 2;
|
|
frame_bits += frame_bits_inc[s->channel_mode];
|
|
|
|
/* audio blocks */
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
frame_bits += s->fbw_channels * 2 + 2; /* blksw * c, dithflag * c, dynrnge, cplstre */
|
|
if (s->channel_mode == AC3_CHMODE_STEREO) {
|
|
frame_bits++; /* rematstr */
|
|
if (!blk)
|
|
frame_bits += 4;
|
|
}
|
|
frame_bits += 2 * s->fbw_channels; /* chexpstr[2] * c */
|
|
if (s->lfe_on)
|
|
frame_bits++; /* lfeexpstr */
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
if (exp_strategy[blk][ch] != EXP_REUSE)
|
|
frame_bits += 6 + 2; /* chbwcod[6], gainrng[2] */
|
|
}
|
|
frame_bits++; /* baie */
|
|
frame_bits++; /* snr */
|
|
frame_bits += 2; /* delta / skip */
|
|
}
|
|
frame_bits++; /* cplinu for block 0 */
|
|
/* bit alloc info */
|
|
/* sdcycod[2], fdcycod[2], sgaincod[2], dbpbcod[2], floorcod[3] */
|
|
/* csnroffset[6] */
|
|
/* (fsnoffset[4] + fgaincod[4]) * c */
|
|
frame_bits += 2*4 + 3 + 6 + s->channels * (4 + 3);
|
|
|
|
/* auxdatae, crcrsv */
|
|
frame_bits += 2;
|
|
|
|
/* CRC */
|
|
frame_bits += 16;
|
|
|
|
/* calculate psd and masking curve before doing bit allocation */
|
|
bit_alloc_masking(s, encoded_exp, exp_strategy, psd, mask);
|
|
|
|
/* now the big work begins : do the bit allocation. Modify the snr
|
|
offset until we can pack everything in the requested frame size */
|
|
|
|
coarse_snr_offset = s->coarse_snr_offset;
|
|
while (coarse_snr_offset >= 0 &&
|
|
bit_alloc(s, mask, psd, bap, frame_bits, coarse_snr_offset, 0) < 0)
|
|
coarse_snr_offset -= SNR_INC1;
|
|
if (coarse_snr_offset < 0) {
|
|
av_log(NULL, AV_LOG_ERROR, "Bit allocation failed. Try increasing the bitrate.\n");
|
|
return -1;
|
|
}
|
|
while (coarse_snr_offset + SNR_INC1 <= 63 &&
|
|
bit_alloc(s, mask, psd, bap1, frame_bits,
|
|
coarse_snr_offset + SNR_INC1, 0) >= 0) {
|
|
coarse_snr_offset += SNR_INC1;
|
|
memcpy(bap, bap1, sizeof(bap1));
|
|
}
|
|
while (coarse_snr_offset + 1 <= 63 &&
|
|
bit_alloc(s, mask, psd, bap1, frame_bits, coarse_snr_offset + 1, 0) >= 0) {
|
|
coarse_snr_offset++;
|
|
memcpy(bap, bap1, sizeof(bap1));
|
|
}
|
|
|
|
fine_snr_offset = 0;
|
|
while (fine_snr_offset + SNR_INC1 <= 15 &&
|
|
bit_alloc(s, mask, psd, bap1, frame_bits,
|
|
coarse_snr_offset, fine_snr_offset + SNR_INC1) >= 0) {
|
|
fine_snr_offset += SNR_INC1;
|
|
memcpy(bap, bap1, sizeof(bap1));
|
|
}
|
|
while (fine_snr_offset + 1 <= 15 &&
|
|
bit_alloc(s, mask, psd, bap1, frame_bits,
|
|
coarse_snr_offset, fine_snr_offset + 1) >= 0) {
|
|
fine_snr_offset++;
|
|
memcpy(bap, bap1, sizeof(bap1));
|
|
}
|
|
|
|
s->coarse_snr_offset = coarse_snr_offset;
|
|
for (ch = 0; ch < s->channels; ch++)
|
|
s->fine_snr_offset[ch] = fine_snr_offset;
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
/**
|
|
* Write the AC-3 frame header to the output bitstream.
|
|
*/
|
|
static void output_frame_header(AC3EncodeContext *s, unsigned char *frame)
|
|
{
|
|
init_put_bits(&s->pb, frame, AC3_MAX_CODED_FRAME_SIZE);
|
|
|
|
put_bits(&s->pb, 16, 0x0b77); /* frame header */
|
|
put_bits(&s->pb, 16, 0); /* crc1: will be filled later */
|
|
put_bits(&s->pb, 2, s->bit_alloc.sr_code);
|
|
put_bits(&s->pb, 6, s->frame_size_code + (s->frame_size - s->frame_size_min) / 2);
|
|
put_bits(&s->pb, 5, s->bitstream_id);
|
|
put_bits(&s->pb, 3, s->bitstream_mode);
|
|
put_bits(&s->pb, 3, s->channel_mode);
|
|
if ((s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO)
|
|
put_bits(&s->pb, 2, 1); /* XXX -4.5 dB */
|
|
if (s->channel_mode & 0x04)
|
|
put_bits(&s->pb, 2, 1); /* XXX -6 dB */
|
|
if (s->channel_mode == AC3_CHMODE_STEREO)
|
|
put_bits(&s->pb, 2, 0); /* surround not indicated */
|
|
put_bits(&s->pb, 1, s->lfe_on); /* LFE */
|
|
put_bits(&s->pb, 5, 31); /* dialog norm: -31 db */
|
|
put_bits(&s->pb, 1, 0); /* no compression control word */
|
|
put_bits(&s->pb, 1, 0); /* no lang code */
|
|
put_bits(&s->pb, 1, 0); /* no audio production info */
|
|
put_bits(&s->pb, 1, 0); /* no copyright */
|
|
put_bits(&s->pb, 1, 1); /* original bitstream */
|
|
put_bits(&s->pb, 1, 0); /* no time code 1 */
|
|
put_bits(&s->pb, 1, 0); /* no time code 2 */
|
|
put_bits(&s->pb, 1, 0); /* no additional bit stream info */
|
|
}
|
|
|
|
|
|
/**
|
|
* Symmetric quantization on 'levels' levels.
|
|
*/
|
|
static inline int sym_quant(int c, int e, int levels)
|
|
{
|
|
int v;
|
|
|
|
if (c >= 0) {
|
|
v = (levels * (c << e)) >> 24;
|
|
v = (v + 1) >> 1;
|
|
v = (levels >> 1) + v;
|
|
} else {
|
|
v = (levels * ((-c) << e)) >> 24;
|
|
v = (v + 1) >> 1;
|
|
v = (levels >> 1) - v;
|
|
}
|
|
assert (v >= 0 && v < levels);
|
|
return v;
|
|
}
|
|
|
|
|
|
/**
|
|
* Asymmetric quantization on 2^qbits levels.
|
|
*/
|
|
static inline int asym_quant(int c, int e, int qbits)
|
|
{
|
|
int lshift, m, v;
|
|
|
|
lshift = e + qbits - 24;
|
|
if (lshift >= 0)
|
|
v = c << lshift;
|
|
else
|
|
v = c >> (-lshift);
|
|
/* rounding */
|
|
v = (v + 1) >> 1;
|
|
m = (1 << (qbits-1));
|
|
if (v >= m)
|
|
v = m - 1;
|
|
assert(v >= -m);
|
|
return v & ((1 << qbits)-1);
|
|
}
|
|
|
|
|
|
/**
|
|
* Write one audio block to the output bitstream.
|
|
*/
|
|
static void output_audio_block(AC3EncodeContext *s,
|
|
uint8_t exp_strategy[AC3_MAX_CHANNELS],
|
|
uint8_t encoded_exp[AC3_MAX_CHANNELS][AC3_MAX_COEFS],
|
|
uint8_t bap[AC3_MAX_CHANNELS][AC3_MAX_COEFS],
|
|
int32_t mdct_coef[AC3_MAX_CHANNELS][AC3_MAX_COEFS],
|
|
int8_t exp_shift[AC3_MAX_CHANNELS],
|
|
int block_num)
|
|
{
|
|
int ch, nb_groups, group_size, i, baie, rbnd;
|
|
uint8_t *p;
|
|
uint16_t qmant[AC3_MAX_CHANNELS][AC3_MAX_COEFS];
|
|
int exp0, exp1;
|
|
int mant1_cnt, mant2_cnt, mant4_cnt;
|
|
uint16_t *qmant1_ptr, *qmant2_ptr, *qmant4_ptr;
|
|
int delta0, delta1, delta2;
|
|
|
|
for (ch = 0; ch < s->fbw_channels; ch++)
|
|
put_bits(&s->pb, 1, 0); /* no block switching */
|
|
for (ch = 0; ch < s->fbw_channels; ch++)
|
|
put_bits(&s->pb, 1, 1); /* no dither */
|
|
put_bits(&s->pb, 1, 0); /* no dynamic range */
|
|
if (!block_num) {
|
|
put_bits(&s->pb, 1, 1); /* coupling strategy present */
|
|
put_bits(&s->pb, 1, 0); /* no coupling strategy */
|
|
} else {
|
|
put_bits(&s->pb, 1, 0); /* no new coupling strategy */
|
|
}
|
|
|
|
if (s->channel_mode == AC3_CHMODE_STEREO) {
|
|
if (!block_num) {
|
|
/* first block must define rematrixing (rematstr) */
|
|
put_bits(&s->pb, 1, 1);
|
|
|
|
/* dummy rematrixing rematflg(1:4)=0 */
|
|
for (rbnd = 0; rbnd < 4; rbnd++)
|
|
put_bits(&s->pb, 1, 0);
|
|
} else {
|
|
/* no matrixing (but should be used in the future) */
|
|
put_bits(&s->pb, 1, 0);
|
|
}
|
|
}
|
|
|
|
/* exponent strategy */
|
|
for (ch = 0; ch < s->fbw_channels; ch++)
|
|
put_bits(&s->pb, 2, exp_strategy[ch]);
|
|
|
|
if (s->lfe_on)
|
|
put_bits(&s->pb, 1, exp_strategy[s->lfe_channel]);
|
|
|
|
/* bandwidth */
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
if (exp_strategy[ch] != EXP_REUSE)
|
|
put_bits(&s->pb, 6, s->bandwidth_code[ch]);
|
|
}
|
|
|
|
/* exponents */
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
if (exp_strategy[ch] == EXP_REUSE)
|
|
continue;
|
|
group_size = exp_strategy[ch] + (exp_strategy[ch] == EXP_D45);
|
|
nb_groups = (s->nb_coefs[ch] + (group_size * 3) - 4) / (3 * group_size);
|
|
p = encoded_exp[ch];
|
|
|
|
/* first exponent */
|
|
exp1 = *p++;
|
|
put_bits(&s->pb, 4, exp1);
|
|
|
|
/* next ones are delta encoded */
|
|
for (i = 0; i < nb_groups; i++) {
|
|
/* merge three delta in one code */
|
|
exp0 = exp1;
|
|
exp1 = p[0];
|
|
p += group_size;
|
|
delta0 = exp1 - exp0 + 2;
|
|
|
|
exp0 = exp1;
|
|
exp1 = p[0];
|
|
p += group_size;
|
|
delta1 = exp1 - exp0 + 2;
|
|
|
|
exp0 = exp1;
|
|
exp1 = p[0];
|
|
p += group_size;
|
|
delta2 = exp1 - exp0 + 2;
|
|
|
|
put_bits(&s->pb, 7, ((delta0 * 5 + delta1) * 5) + delta2);
|
|
}
|
|
|
|
if (ch != s->lfe_channel)
|
|
put_bits(&s->pb, 2, 0); /* no gain range info */
|
|
}
|
|
|
|
/* bit allocation info */
|
|
baie = (block_num == 0);
|
|
put_bits(&s->pb, 1, baie);
|
|
if (baie) {
|
|
put_bits(&s->pb, 2, s->slow_decay_code);
|
|
put_bits(&s->pb, 2, s->fast_decay_code);
|
|
put_bits(&s->pb, 2, s->slow_gain_code);
|
|
put_bits(&s->pb, 2, s->db_per_bit_code);
|
|
put_bits(&s->pb, 3, s->floor_code);
|
|
}
|
|
|
|
/* snr offset */
|
|
put_bits(&s->pb, 1, baie);
|
|
if (baie) {
|
|
put_bits(&s->pb, 6, s->coarse_snr_offset);
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
put_bits(&s->pb, 4, s->fine_snr_offset[ch]);
|
|
put_bits(&s->pb, 3, s->fast_gain_code[ch]);
|
|
}
|
|
}
|
|
|
|
put_bits(&s->pb, 1, 0); /* no delta bit allocation */
|
|
put_bits(&s->pb, 1, 0); /* no data to skip */
|
|
|
|
/* mantissa encoding : we use two passes to handle the grouping. A
|
|
one pass method may be faster, but it would necessitate to
|
|
modify the output stream. */
|
|
|
|
/* first pass: quantize */
|
|
mant1_cnt = mant2_cnt = mant4_cnt = 0;
|
|
qmant1_ptr = qmant2_ptr = qmant4_ptr = NULL;
|
|
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
int b, c, e, v;
|
|
|
|
for (i = 0; i < s->nb_coefs[ch]; i++) {
|
|
c = mdct_coef[ch][i];
|
|
e = encoded_exp[ch][i] - exp_shift[ch];
|
|
b = bap[ch][i];
|
|
switch (b) {
|
|
case 0:
|
|
v = 0;
|
|
break;
|
|
case 1:
|
|
v = sym_quant(c, e, 3);
|
|
switch (mant1_cnt) {
|
|
case 0:
|
|
qmant1_ptr = &qmant[ch][i];
|
|
v = 9 * v;
|
|
mant1_cnt = 1;
|
|
break;
|
|
case 1:
|
|
*qmant1_ptr += 3 * v;
|
|
mant1_cnt = 2;
|
|
v = 128;
|
|
break;
|
|
default:
|
|
*qmant1_ptr += v;
|
|
mant1_cnt = 0;
|
|
v = 128;
|
|
break;
|
|
}
|
|
break;
|
|
case 2:
|
|
v = sym_quant(c, e, 5);
|
|
switch (mant2_cnt) {
|
|
case 0:
|
|
qmant2_ptr = &qmant[ch][i];
|
|
v = 25 * v;
|
|
mant2_cnt = 1;
|
|
break;
|
|
case 1:
|
|
*qmant2_ptr += 5 * v;
|
|
mant2_cnt = 2;
|
|
v = 128;
|
|
break;
|
|
default:
|
|
*qmant2_ptr += v;
|
|
mant2_cnt = 0;
|
|
v = 128;
|
|
break;
|
|
}
|
|
break;
|
|
case 3:
|
|
v = sym_quant(c, e, 7);
|
|
break;
|
|
case 4:
|
|
v = sym_quant(c, e, 11);
|
|
switch (mant4_cnt) {
|
|
case 0:
|
|
qmant4_ptr = &qmant[ch][i];
|
|
v = 11 * v;
|
|
mant4_cnt = 1;
|
|
break;
|
|
default:
|
|
*qmant4_ptr += v;
|
|
mant4_cnt = 0;
|
|
v = 128;
|
|
break;
|
|
}
|
|
break;
|
|
case 5:
|
|
v = sym_quant(c, e, 15);
|
|
break;
|
|
case 14:
|
|
v = asym_quant(c, e, 14);
|
|
break;
|
|
case 15:
|
|
v = asym_quant(c, e, 16);
|
|
break;
|
|
default:
|
|
v = asym_quant(c, e, b - 1);
|
|
break;
|
|
}
|
|
qmant[ch][i] = v;
|
|
}
|
|
}
|
|
|
|
/* second pass : output the values */
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
int b, q;
|
|
|
|
for (i = 0; i < s->nb_coefs[ch]; i++) {
|
|
q = qmant[ch][i];
|
|
b = bap[ch][i];
|
|
switch (b) {
|
|
case 0: break;
|
|
case 1: if (q != 128) put_bits(&s->pb, 5, q); break;
|
|
case 2: if (q != 128) put_bits(&s->pb, 7, q); break;
|
|
case 3: put_bits(&s->pb, 3, q); break;
|
|
case 4: if (q != 128) put_bits(&s->pb, 7, q); break;
|
|
case 14: put_bits(&s->pb, 14, q); break;
|
|
case 15: put_bits(&s->pb, 16, q); break;
|
|
default: put_bits(&s->pb, b-1, q); break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/** CRC-16 Polynomial */
|
|
#define CRC16_POLY ((1 << 0) | (1 << 2) | (1 << 15) | (1 << 16))
|
|
|
|
|
|
static unsigned int mul_poly(unsigned int a, unsigned int b, unsigned int poly)
|
|
{
|
|
unsigned int c;
|
|
|
|
c = 0;
|
|
while (a) {
|
|
if (a & 1)
|
|
c ^= b;
|
|
a = a >> 1;
|
|
b = b << 1;
|
|
if (b & (1 << 16))
|
|
b ^= poly;
|
|
}
|
|
return c;
|
|
}
|
|
|
|
|
|
static unsigned int pow_poly(unsigned int a, unsigned int n, unsigned int poly)
|
|
{
|
|
unsigned int r;
|
|
r = 1;
|
|
while (n) {
|
|
if (n & 1)
|
|
r = mul_poly(r, a, poly);
|
|
a = mul_poly(a, a, poly);
|
|
n >>= 1;
|
|
}
|
|
return r;
|
|
}
|
|
|
|
|
|
/**
|
|
* Fill the end of the frame with 0's and compute the two CRCs.
|
|
*/
|
|
static void output_frame_end(AC3EncodeContext *s)
|
|
{
|
|
int frame_size, frame_size_58, pad_bytes, crc1, crc2, crc_inv;
|
|
uint8_t *frame;
|
|
|
|
frame_size = s->frame_size; /* frame size in words */
|
|
/* align to 8 bits */
|
|
flush_put_bits(&s->pb);
|
|
/* add zero bytes to reach the frame size */
|
|
frame = s->pb.buf;
|
|
pad_bytes = s->frame_size - (put_bits_ptr(&s->pb) - frame) - 2;
|
|
assert(pad_bytes >= 0);
|
|
if (pad_bytes > 0)
|
|
memset(put_bits_ptr(&s->pb), 0, pad_bytes);
|
|
|
|
/* Now we must compute both crcs : this is not so easy for crc1
|
|
because it is at the beginning of the data... */
|
|
frame_size_58 = ((frame_size >> 2) + (frame_size >> 4)) << 1;
|
|
|
|
crc1 = av_bswap16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
|
|
frame + 4, frame_size_58 - 4));
|
|
|
|
/* XXX: could precompute crc_inv */
|
|
crc_inv = pow_poly((CRC16_POLY >> 1), (8 * frame_size_58) - 16, CRC16_POLY);
|
|
crc1 = mul_poly(crc_inv, crc1, CRC16_POLY);
|
|
AV_WB16(frame + 2, crc1);
|
|
|
|
crc2 = av_bswap16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
|
|
frame + frame_size_58,
|
|
frame_size - frame_size_58 - 2));
|
|
AV_WB16(frame + frame_size - 2, crc2);
|
|
}
|
|
|
|
|
|
/**
|
|
* Encode a single AC-3 frame.
|
|
*/
|
|
static int ac3_encode_frame(AVCodecContext *avctx,
|
|
unsigned char *frame, int buf_size, void *data)
|
|
{
|
|
AC3EncodeContext *s = avctx->priv_data;
|
|
const int16_t *samples = data;
|
|
int v;
|
|
int blk, blk1, blk2, ch, i;
|
|
int16_t input_samples[AC3_WINDOW_SIZE];
|
|
int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
|
|
uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
|
|
uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS];
|
|
uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
|
|
uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
|
|
int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS];
|
|
int frame_bits;
|
|
|
|
frame_bits = 0;
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
int ich = s->channel_map[ch];
|
|
/* fixed mdct to the six sub blocks & exponent computation */
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
const int16_t *sptr;
|
|
int sinc;
|
|
|
|
/* compute input samples */
|
|
memcpy(input_samples, s->last_samples[ich], AC3_BLOCK_SIZE * sizeof(int16_t));
|
|
sinc = s->channels;
|
|
sptr = samples + (sinc * AC3_BLOCK_SIZE * blk) + ich;
|
|
for (i = 0; i < AC3_BLOCK_SIZE; i++) {
|
|
v = *sptr;
|
|
input_samples[i + AC3_BLOCK_SIZE] = v;
|
|
s->last_samples[ich][i] = v;
|
|
sptr += sinc;
|
|
}
|
|
|
|
/* apply the MDCT window */
|
|
for (i = 0; i < AC3_BLOCK_SIZE; i++) {
|
|
input_samples[i] = MUL16(input_samples[i],
|
|
ff_ac3_window[i]) >> 15;
|
|
input_samples[AC3_WINDOW_SIZE-i-1] = MUL16(input_samples[AC3_WINDOW_SIZE-i-1],
|
|
ff_ac3_window[i]) >> 15;
|
|
}
|
|
|
|
/* Normalize the samples to use the maximum available precision */
|
|
v = 14 - log2_tab(input_samples, AC3_WINDOW_SIZE);
|
|
if (v < 0)
|
|
v = 0;
|
|
exp_shift[blk][ch] = v - 9;
|
|
lshift_tab(input_samples, AC3_WINDOW_SIZE, v);
|
|
|
|
/* do the MDCT */
|
|
mdct512(mdct_coef[blk][ch], input_samples);
|
|
|
|
/* compute "exponents". We take into account the normalization there */
|
|
for (i = 0; i < AC3_MAX_COEFS; i++) {
|
|
int e;
|
|
v = abs(mdct_coef[blk][ch][i]);
|
|
if (v == 0)
|
|
e = 24;
|
|
else {
|
|
e = 23 - av_log2(v) + exp_shift[blk][ch];
|
|
if (e >= 24) {
|
|
e = 24;
|
|
mdct_coef[blk][ch][i] = 0;
|
|
}
|
|
}
|
|
exp[blk][ch][i] = e;
|
|
}
|
|
}
|
|
|
|
compute_exp_strategy_ch(exp_strategy, exp, ch, ch == s->lfe_channel);
|
|
|
|
/* compute the exponents as the decoder will see them. The
|
|
EXP_REUSE case must be handled carefully : we select the
|
|
min of the exponents */
|
|
blk = 0;
|
|
while (blk < AC3_MAX_BLOCKS) {
|
|
blk1 = blk + 1;
|
|
while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1][ch] == EXP_REUSE) {
|
|
exponent_min(exp[blk][ch], exp[blk1][ch], s->nb_coefs[ch]);
|
|
blk1++;
|
|
}
|
|
frame_bits += encode_exponents_blk_ch(encoded_exp[blk][ch],
|
|
exp[blk][ch], s->nb_coefs[ch],
|
|
exp_strategy[blk][ch]);
|
|
/* copy encoded exponents for reuse case */
|
|
for (blk2 = blk+1; blk2 < blk1; blk2++) {
|
|
memcpy(encoded_exp[blk2][ch], encoded_exp[blk][ch],
|
|
s->nb_coefs[ch] * sizeof(uint8_t));
|
|
}
|
|
blk = blk1;
|
|
}
|
|
}
|
|
|
|
/* adjust for fractional frame sizes */
|
|
while (s->bits_written >= s->bit_rate && s->samples_written >= s->sample_rate) {
|
|
s->bits_written -= s->bit_rate;
|
|
s->samples_written -= s->sample_rate;
|
|
}
|
|
s->frame_size = s->frame_size_min + 2 * (s->bits_written * s->sample_rate < s->samples_written * s->bit_rate);
|
|
s->bits_written += s->frame_size * 8;
|
|
s->samples_written += AC3_FRAME_SIZE;
|
|
|
|
compute_bit_allocation(s, bap, encoded_exp, exp_strategy, frame_bits);
|
|
/* everything is known... let's output the frame */
|
|
output_frame_header(s, frame);
|
|
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
output_audio_block(s, exp_strategy[blk], encoded_exp[blk],
|
|
bap[blk], mdct_coef[blk], exp_shift[blk], blk);
|
|
}
|
|
output_frame_end(s);
|
|
|
|
return s->frame_size;
|
|
}
|
|
|
|
|
|
/**
|
|
* Finalize encoding and free any memory allocated by the encoder.
|
|
*/
|
|
static av_cold int ac3_encode_close(AVCodecContext *avctx)
|
|
{
|
|
av_freep(&avctx->coded_frame);
|
|
return 0;
|
|
}
|
|
|
|
|
|
/**
|
|
* Set channel information during initialization.
|
|
*/
|
|
static av_cold int set_channel_info(AC3EncodeContext *s, int channels,
|
|
int64_t *channel_layout)
|
|
{
|
|
int ch_layout;
|
|
|
|
if (channels < 1 || channels > AC3_MAX_CHANNELS)
|
|
return AVERROR(EINVAL);
|
|
if ((uint64_t)*channel_layout > 0x7FF)
|
|
return AVERROR(EINVAL);
|
|
ch_layout = *channel_layout;
|
|
if (!ch_layout)
|
|
ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL);
|
|
if (av_get_channel_layout_nb_channels(ch_layout) != channels)
|
|
return AVERROR(EINVAL);
|
|
|
|
s->lfe_on = !!(ch_layout & AV_CH_LOW_FREQUENCY);
|
|
s->channels = channels;
|
|
s->fbw_channels = channels - s->lfe_on;
|
|
s->lfe_channel = s->lfe_on ? s->fbw_channels : -1;
|
|
if (s->lfe_on)
|
|
ch_layout -= AV_CH_LOW_FREQUENCY;
|
|
|
|
switch (ch_layout) {
|
|
case AV_CH_LAYOUT_MONO: s->channel_mode = AC3_CHMODE_MONO; break;
|
|
case AV_CH_LAYOUT_STEREO: s->channel_mode = AC3_CHMODE_STEREO; break;
|
|
case AV_CH_LAYOUT_SURROUND: s->channel_mode = AC3_CHMODE_3F; break;
|
|
case AV_CH_LAYOUT_2_1: s->channel_mode = AC3_CHMODE_2F1R; break;
|
|
case AV_CH_LAYOUT_4POINT0: s->channel_mode = AC3_CHMODE_3F1R; break;
|
|
case AV_CH_LAYOUT_QUAD:
|
|
case AV_CH_LAYOUT_2_2: s->channel_mode = AC3_CHMODE_2F2R; break;
|
|
case AV_CH_LAYOUT_5POINT0:
|
|
case AV_CH_LAYOUT_5POINT0_BACK: s->channel_mode = AC3_CHMODE_3F2R; break;
|
|
default:
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
s->channel_map = ff_ac3_enc_channel_map[s->channel_mode][s->lfe_on];
|
|
*channel_layout = ch_layout;
|
|
if (s->lfe_on)
|
|
*channel_layout |= AV_CH_LOW_FREQUENCY;
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s)
|
|
{
|
|
int i, ret;
|
|
|
|
if (!avctx->channel_layout) {
|
|
av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
|
|
"encoder will guess the layout, but it "
|
|
"might be incorrect.\n");
|
|
}
|
|
ret = set_channel_info(s, avctx->channels, &avctx->channel_layout);
|
|
if (ret) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid channel layout\n");
|
|
return ret;
|
|
}
|
|
|
|
/* frequency */
|
|
for (i = 0; i < 9; i++) {
|
|
if ((ff_ac3_sample_rate_tab[i / 3] >> (i % 3)) == avctx->sample_rate)
|
|
break;
|
|
}
|
|
if (i == 9) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
s->sample_rate = avctx->sample_rate;
|
|
s->bit_alloc.sr_shift = i % 3;
|
|
s->bit_alloc.sr_code = i / 3;
|
|
|
|
/* bitrate & frame size */
|
|
for (i = 0; i < 19; i++) {
|
|
if ((ff_ac3_bitrate_tab[i] >> s->bit_alloc.sr_shift)*1000 == avctx->bit_rate)
|
|
break;
|
|
}
|
|
if (i == 19) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid bit rate\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
s->bit_rate = avctx->bit_rate;
|
|
s->frame_size_code = i << 1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
/**
|
|
* Set bandwidth for all channels.
|
|
* The user can optionally supply a cutoff frequency. Otherwise an appropriate
|
|
* default value will be used.
|
|
*/
|
|
static av_cold void set_bandwidth(AC3EncodeContext *s, int cutoff)
|
|
{
|
|
int ch, bw_code;
|
|
|
|
if (cutoff) {
|
|
/* calculate bandwidth based on user-specified cutoff frequency */
|
|
int fbw_coeffs;
|
|
cutoff = av_clip(cutoff, 1, s->sample_rate >> 1);
|
|
fbw_coeffs = cutoff * 2 * AC3_MAX_COEFS / s->sample_rate;
|
|
bw_code = av_clip((fbw_coeffs - 73) / 3, 0, 60);
|
|
} else {
|
|
/* use default bandwidth setting */
|
|
/* XXX: should compute the bandwidth according to the frame
|
|
size, so that we avoid annoying high frequency artifacts */
|
|
bw_code = 50;
|
|
}
|
|
|
|
/* set number of coefficients for each channel */
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
s->bandwidth_code[ch] = bw_code;
|
|
s->nb_coefs[ch] = bw_code * 3 + 73;
|
|
}
|
|
if (s->lfe_on)
|
|
s->nb_coefs[s->lfe_channel] = 7; /* LFE channel always has 7 coefs */
|
|
}
|
|
|
|
|
|
/**
|
|
* Initialize the encoder.
|
|
*/
|
|
static av_cold int ac3_encode_init(AVCodecContext *avctx)
|
|
{
|
|
AC3EncodeContext *s = avctx->priv_data;
|
|
int ret;
|
|
|
|
avctx->frame_size = AC3_FRAME_SIZE;
|
|
|
|
ac3_common_init();
|
|
|
|
ret = validate_options(avctx, s);
|
|
if (ret)
|
|
return ret;
|
|
|
|
s->bitstream_id = 8 + s->bit_alloc.sr_shift;
|
|
s->bitstream_mode = 0; /* complete main audio service */
|
|
|
|
s->frame_size_min = 2 * ff_ac3_frame_size_tab[s->frame_size_code][s->bit_alloc.sr_code];
|
|
s->bits_written = 0;
|
|
s->samples_written = 0;
|
|
s->frame_size = s->frame_size_min;
|
|
|
|
set_bandwidth(s, avctx->cutoff);
|
|
|
|
/* initial snr offset */
|
|
s->coarse_snr_offset = 40;
|
|
|
|
mdct_init(9);
|
|
|
|
avctx->coded_frame= avcodec_alloc_frame();
|
|
avctx->coded_frame->key_frame= 1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
#ifdef TEST
|
|
/*************************************************************************/
|
|
/* TEST */
|
|
|
|
#include "libavutil/lfg.h"
|
|
|
|
#define FN (MDCT_SAMPLES/4)
|
|
|
|
|
|
static void fft_test(AVLFG *lfg)
|
|
{
|
|
IComplex in[FN], in1[FN];
|
|
int k, n, i;
|
|
float sum_re, sum_im, a;
|
|
|
|
for (i = 0; i < FN; i++) {
|
|
in[i].re = av_lfg_get(lfg) % 65535 - 32767;
|
|
in[i].im = av_lfg_get(lfg) % 65535 - 32767;
|
|
in1[i] = in[i];
|
|
}
|
|
fft(in, 7);
|
|
|
|
/* do it by hand */
|
|
for (k = 0; k < FN; k++) {
|
|
sum_re = 0;
|
|
sum_im = 0;
|
|
for (n = 0; n < FN; n++) {
|
|
a = -2 * M_PI * (n * k) / FN;
|
|
sum_re += in1[n].re * cos(a) - in1[n].im * sin(a);
|
|
sum_im += in1[n].re * sin(a) + in1[n].im * cos(a);
|
|
}
|
|
av_log(NULL, AV_LOG_DEBUG, "%3d: %6d,%6d %6.0f,%6.0f\n",
|
|
k, in[k].re, in[k].im, sum_re / FN, sum_im / FN);
|
|
}
|
|
}
|
|
|
|
|
|
static void mdct_test(AVLFG *lfg)
|
|
{
|
|
int16_t input[MDCT_SAMPLES];
|
|
int32_t output[AC3_MAX_COEFS];
|
|
float input1[MDCT_SAMPLES];
|
|
float output1[AC3_MAX_COEFS];
|
|
float s, a, err, e, emax;
|
|
int i, k, n;
|
|
|
|
for (i = 0; i < MDCT_SAMPLES; i++) {
|
|
input[i] = (av_lfg_get(lfg) % 65535 - 32767) * 9 / 10;
|
|
input1[i] = input[i];
|
|
}
|
|
|
|
mdct512(output, input);
|
|
|
|
/* do it by hand */
|
|
for (k = 0; k < AC3_MAX_COEFS; k++) {
|
|
s = 0;
|
|
for (n = 0; n < MDCT_SAMPLES; n++) {
|
|
a = (2*M_PI*(2*n+1+MDCT_SAMPLES/2)*(2*k+1) / (4 * MDCT_SAMPLES));
|
|
s += input1[n] * cos(a);
|
|
}
|
|
output1[k] = -2 * s / MDCT_SAMPLES;
|
|
}
|
|
|
|
err = 0;
|
|
emax = 0;
|
|
for (i = 0; i < AC3_MAX_COEFS; i++) {
|
|
av_log(NULL, AV_LOG_DEBUG, "%3d: %7d %7.0f\n", i, output[i], output1[i]);
|
|
e = output[i] - output1[i];
|
|
if (e > emax)
|
|
emax = e;
|
|
err += e * e;
|
|
}
|
|
av_log(NULL, AV_LOG_DEBUG, "err2=%f emax=%f\n", err / AC3_MAX_COEFS, emax);
|
|
}
|
|
|
|
|
|
int main(void)
|
|
{
|
|
AVLFG lfg;
|
|
|
|
av_log_set_level(AV_LOG_DEBUG);
|
|
mdct_init(9);
|
|
|
|
fft_test(&lfg);
|
|
mdct_test(&lfg);
|
|
|
|
return 0;
|
|
}
|
|
#endif /* TEST */
|
|
|
|
|
|
AVCodec ac3_encoder = {
|
|
"ac3",
|
|
AVMEDIA_TYPE_AUDIO,
|
|
CODEC_ID_AC3,
|
|
sizeof(AC3EncodeContext),
|
|
ac3_encode_init,
|
|
ac3_encode_frame,
|
|
ac3_encode_close,
|
|
NULL,
|
|
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
|
|
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
|
|
.channel_layouts = (const int64_t[]){
|
|
AV_CH_LAYOUT_MONO,
|
|
AV_CH_LAYOUT_STEREO,
|
|
AV_CH_LAYOUT_2_1,
|
|
AV_CH_LAYOUT_SURROUND,
|
|
AV_CH_LAYOUT_2_2,
|
|
AV_CH_LAYOUT_QUAD,
|
|
AV_CH_LAYOUT_4POINT0,
|
|
AV_CH_LAYOUT_5POINT0,
|
|
AV_CH_LAYOUT_5POINT0_BACK,
|
|
(AV_CH_LAYOUT_MONO | AV_CH_LOW_FREQUENCY),
|
|
(AV_CH_LAYOUT_STEREO | AV_CH_LOW_FREQUENCY),
|
|
(AV_CH_LAYOUT_2_1 | AV_CH_LOW_FREQUENCY),
|
|
(AV_CH_LAYOUT_SURROUND | AV_CH_LOW_FREQUENCY),
|
|
(AV_CH_LAYOUT_2_2 | AV_CH_LOW_FREQUENCY),
|
|
(AV_CH_LAYOUT_QUAD | AV_CH_LOW_FREQUENCY),
|
|
(AV_CH_LAYOUT_4POINT0 | AV_CH_LOW_FREQUENCY),
|
|
AV_CH_LAYOUT_5POINT1,
|
|
AV_CH_LAYOUT_5POINT1_BACK,
|
|
0 },
|
|
};
|