2 * The simplest AC-3 encoder
3 * Copyright (c) 2000 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 * The simplest AC-3 encoder.
29 #include "libavcore/audioconvert.h"
30 #include "libavutil/crc.h"
34 #include "audioconvert.h"
38 #define MDCT_SAMPLES (1 << MDCT_NBITS)
40 /** Scale a float value by 2^bits and convert to an integer. */
41 #define SCALE_FLOAT(a, bits) lrintf((a) * (float)(1 << (bits)))
43 /** Scale a float value by 2^15, convert to an integer, and clip to int16_t range. */
44 #define FIX15(a) av_clip_int16(SCALE_FLOAT(a, 15))
49 * Used in fixed-point MDCT calculation.
51 typedef struct IComplex {
56 * AC-3 encoder private context.
58 typedef struct AC3EncodeContext {
59 PutBitContext pb; ///< bitstream writer context
61 int bitstream_id; ///< bitstream id (bsid)
62 int bitstream_mode; ///< bitstream mode (bsmod)
64 int bit_rate; ///< target bit rate, in bits-per-second
65 int sample_rate; ///< sampling frequency, in Hz
67 int frame_size_min; ///< minimum frame size in case rounding is necessary
68 int frame_size; ///< current frame size in bytes
69 int frame_size_code; ///< frame size code (frmsizecod)
70 int bits_written; ///< bit count (used to avg. bitrate)
71 int samples_written; ///< sample count (used to avg. bitrate)
73 int fbw_channels; ///< number of full-bandwidth channels (nfchans)
74 int channels; ///< total number of channels (nchans)
75 int lfe_on; ///< indicates if there is an LFE channel (lfeon)
76 int lfe_channel; ///< channel index of the LFE channel
77 int channel_mode; ///< channel mode (acmod)
78 const uint8_t *channel_map; ///< channel map used to reorder channels
80 int bandwidth_code[AC3_MAX_CHANNELS]; ///< bandwidth code (0 to 60) (chbwcod)
81 int nb_coefs[AC3_MAX_CHANNELS];
83 /* bitrate allocation control */
84 int slow_gain_code; ///< slow gain code (sgaincod)
85 int slow_decay_code; ///< slow decay code (sdcycod)
86 int fast_decay_code; ///< fast decay code (fdcycod)
87 int db_per_bit_code; ///< dB/bit code (dbpbcod)
88 int floor_code; ///< floor code (floorcod)
89 AC3BitAllocParameters bit_alloc; ///< bit allocation parameters
90 int coarse_snr_offset; ///< coarse SNR offsets (csnroffst)
91 int fast_gain_code[AC3_MAX_CHANNELS]; ///< fast gain codes (signal-to-mask ratio) (fgaincod)
92 int fine_snr_offset[AC3_MAX_CHANNELS]; ///< fine SNR offsets (fsnroffst)
94 /* mantissa encoding */
95 int mant1_cnt, mant2_cnt, mant4_cnt; ///< mantissa counts for bap=1,2,4
97 int16_t last_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< last 256 samples from previous frame
101 /** MDCT and FFT tables */
102 static int16_t costab[64];
103 static int16_t sintab[64];
104 static int16_t xcos1[128];
105 static int16_t xsin1[128];
109 * Adjust the frame size to make the average bit rate match the target bit rate.
110 * This is only needed for 11025, 22050, and 44100 sample rates.
112 static void adjust_frame_size(AC3EncodeContext *s)
114 while (s->bits_written >= s->bit_rate && s->samples_written >= s->sample_rate) {
115 s->bits_written -= s->bit_rate;
116 s->samples_written -= s->sample_rate;
118 s->frame_size = s->frame_size_min + 2 * (s->bits_written * s->sample_rate < s->samples_written * s->bit_rate);
119 s->bits_written += s->frame_size * 8;
120 s->samples_written += AC3_FRAME_SIZE;
125 * Deinterleave input samples.
126 * Channels are reordered from FFmpeg's default order to AC-3 order.
128 static void deinterleave_input_samples(AC3EncodeContext *s,
129 const int16_t *samples,
130 int16_t planar_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE+AC3_FRAME_SIZE])
134 /* deinterleave and remap input samples */
135 for (ch = 0; ch < s->channels; ch++) {
139 /* copy last 256 samples of previous frame to the start of the current frame */
140 memcpy(&planar_samples[ch][0], s->last_samples[ch],
141 AC3_BLOCK_SIZE * sizeof(planar_samples[0][0]));
145 sptr = samples + s->channel_map[ch];
146 for (i = AC3_BLOCK_SIZE; i < AC3_FRAME_SIZE+AC3_BLOCK_SIZE; i++) {
147 planar_samples[ch][i] = *sptr;
151 /* save last 256 samples for next frame */
152 memcpy(s->last_samples[ch], &planar_samples[ch][6* AC3_BLOCK_SIZE],
153 AC3_BLOCK_SIZE * sizeof(planar_samples[0][0]));
159 * Initialize FFT tables.
160 * @param ln log2(FFT size)
162 static av_cold void fft_init(int ln)
170 for (i = 0; i < n2; i++) {
171 alpha = 2.0 * M_PI * i / n;
172 costab[i] = FIX15(cos(alpha));
173 sintab[i] = FIX15(sin(alpha));
179 * Initialize MDCT tables.
180 * @param nbits log2(MDCT size)
182 static av_cold void mdct_init(int nbits)
191 for (i = 0; i < n4; i++) {
192 float alpha = 2.0 * M_PI * (i + 1.0 / 8.0) / n;
193 xcos1[i] = FIX15(-cos(alpha));
194 xsin1[i] = FIX15(-sin(alpha));
200 #define BF(pre, pim, qre, qim, pre1, pim1, qre1, qim1) \
202 int ax, ay, bx, by; \
207 pre = (bx + ax) >> 1; \
208 pim = (by + ay) >> 1; \
209 qre = (bx - ax) >> 1; \
210 qim = (by - ay) >> 1; \
214 /** Complex multiply */
215 #define CMUL(pre, pim, are, aim, bre, bim) \
217 pre = (MUL16(are, bre) - MUL16(aim, bim)) >> 15; \
218 pim = (MUL16(are, bim) + MUL16(bre, aim)) >> 15; \
223 * Calculate a 2^n point complex FFT on 2^ln points.
224 * @param z complex input/output samples
225 * @param ln log2(FFT size)
227 static void fft(IComplex *z, int ln)
231 register IComplex *p,*q;
237 for (j = 0; j < np; j++) {
238 int k = av_reverse[j] >> (8 - ln);
240 FFSWAP(IComplex, z[k], z[j]);
248 BF(p[0].re, p[0].im, p[1].re, p[1].im,
249 p[0].re, p[0].im, p[1].re, p[1].im);
258 BF(p[0].re, p[0].im, p[2].re, p[2].im,
259 p[0].re, p[0].im, p[2].re, p[2].im);
260 BF(p[1].re, p[1].im, p[3].re, p[3].im,
261 p[1].re, p[1].im, p[3].im, -p[3].re);
273 for (j = 0; j < nblocks; j++) {
274 BF(p->re, p->im, q->re, q->im,
275 p->re, p->im, q->re, q->im);
278 for(l = nblocks; l < np2; l += nblocks) {
279 CMUL(tmp_re, tmp_im, costab[l], -sintab[l], q->re, q->im);
280 BF(p->re, p->im, q->re, q->im,
281 p->re, p->im, tmp_re, tmp_im);
288 nblocks = nblocks >> 1;
289 nloops = nloops << 1;
295 * Calculate a 512-point MDCT
296 * @param out 256 output frequency coefficients
297 * @param in 512 windowed input audio samples
299 static void mdct512(int32_t *out, int16_t *in)
301 int i, re, im, re1, im1;
302 int16_t rot[MDCT_SAMPLES];
303 IComplex x[MDCT_SAMPLES/4];
305 /* shift to simplify computations */
306 for (i = 0; i < MDCT_SAMPLES/4; i++)
307 rot[i] = -in[i + 3*MDCT_SAMPLES/4];
308 for (;i < MDCT_SAMPLES; i++)
309 rot[i] = in[i - MDCT_SAMPLES/4];
312 for (i = 0; i < MDCT_SAMPLES/4; i++) {
313 re = ((int)rot[ 2*i] - (int)rot[MDCT_SAMPLES -1-2*i]) >> 1;
314 im = -((int)rot[MDCT_SAMPLES/2+2*i] - (int)rot[MDCT_SAMPLES/2-1-2*i]) >> 1;
315 CMUL(x[i].re, x[i].im, re, im, -xcos1[i], xsin1[i]);
318 fft(x, MDCT_NBITS - 2);
321 for (i = 0; i < MDCT_SAMPLES/4; i++) {
324 CMUL(re1, im1, re, im, xsin1[i], xcos1[i]);
326 out[MDCT_SAMPLES/2-1-2*i] = re1;
332 * Apply KBD window to input samples prior to MDCT.
334 static void apply_window(int16_t *output, const int16_t *input,
335 const int16_t *window, int n)
340 for (i = 0; i < n2; i++) {
341 output[i] = MUL16(input[i], window[i]) >> 15;
342 output[n-i-1] = MUL16(input[n-i-1], window[i]) >> 15;
348 * Calculate the log2() of the maximum absolute value in an array.
349 * @param tab input array
350 * @param n number of values in the array
351 * @return log2(max(abs(tab[])))
353 static int log2_tab(int16_t *tab, int n)
358 for (i = 0; i < n; i++)
366 * Left-shift each value in an array by a specified amount.
367 * @param tab input array
368 * @param n number of values in the array
369 * @param lshift left shift amount. a negative value means right shift.
371 static void lshift_tab(int16_t *tab, int n, int lshift)
376 for(i = 0; i < n; i++)
378 } else if (lshift < 0) {
380 for (i = 0; i < n; i++)
387 * Normalize the input samples to use the maximum available precision.
388 * This assumes signed 16-bit input samples. Exponents are reduced by 9 to
389 * match the 24-bit internal precision for MDCT coefficients.
391 * @return exponent shift
393 static int normalize_samples(AC3EncodeContext *s,
394 int16_t windowed_samples[AC3_WINDOW_SIZE])
396 int v = 14 - log2_tab(windowed_samples, AC3_WINDOW_SIZE);
398 lshift_tab(windowed_samples, AC3_WINDOW_SIZE, v);
404 * Apply the MDCT to input samples to generate frequency coefficients.
405 * This applies the KBD window and normalizes the input to reduce precision
406 * loss due to fixed-point calculations.
408 static void apply_mdct(AC3EncodeContext *s,
409 int16_t planar_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE+AC3_FRAME_SIZE],
410 int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
411 int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
414 int16_t windowed_samples[AC3_WINDOW_SIZE];
416 for (ch = 0; ch < s->channels; ch++) {
417 for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
418 const int16_t *input_samples = &planar_samples[ch][blk * AC3_BLOCK_SIZE];
420 apply_window(windowed_samples, input_samples, ff_ac3_window, AC3_WINDOW_SIZE);
422 exp_shift[blk][ch] = normalize_samples(s, windowed_samples);
424 mdct512(mdct_coef[blk][ch], windowed_samples);
431 * Extract exponents from the MDCT coefficients.
432 * This takes into account the normalization that was done to the input samples
433 * by adjusting the exponents by the exponent shift values.
435 static void extract_exponents(AC3EncodeContext *s,
436 int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
437 int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
438 uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
442 /* extract exponents */
443 for (ch = 0; ch < s->channels; ch++) {
444 for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
445 /* compute "exponents". We take into account the normalization there */
446 for (i = 0; i < AC3_MAX_COEFS; i++) {
448 int v = abs(mdct_coef[blk][ch][i]);
452 e = 23 - av_log2(v) + exp_shift[blk][ch];
455 mdct_coef[blk][ch][i] = 0;
466 * Calculate the sum of absolute differences (SAD) between 2 sets of exponents.
468 static int calc_exp_diff(uint8_t *exp1, uint8_t *exp2, int n)
472 for (i = 0; i < n; i++)
473 sum += abs(exp1[i] - exp2[i]);
479 * Exponent Difference Threshold.
480 * New exponents are sent if their SAD exceed this number.
482 #define EXP_DIFF_THRESHOLD 1000
486 * Calculate exponent strategies for all blocks in a single channel.
488 static void compute_exp_strategy_ch(uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
489 uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
495 /* estimate if the exponent variation & decide if they should be
496 reused in the next frame */
497 exp_strategy[0][ch] = EXP_NEW;
498 for (blk = 1; blk < AC3_MAX_BLOCKS; blk++) {
499 exp_diff = calc_exp_diff(exp[blk][ch], exp[blk-1][ch], AC3_MAX_COEFS);
500 if (exp_diff > EXP_DIFF_THRESHOLD)
501 exp_strategy[blk][ch] = EXP_NEW;
503 exp_strategy[blk][ch] = EXP_REUSE;
508 /* now select the encoding strategy type : if exponents are often
509 recoded, we use a coarse encoding */
511 while (blk < AC3_MAX_BLOCKS) {
513 while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1][ch] == EXP_REUSE)
515 switch (blk1 - blk) {
516 case 1: exp_strategy[blk][ch] = EXP_D45; break;
518 case 3: exp_strategy[blk][ch] = EXP_D25; break;
519 default: exp_strategy[blk][ch] = EXP_D15; break;
527 * Calculate exponent strategies for all channels.
529 static void compute_exp_strategy(AC3EncodeContext *s,
530 uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
531 uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
535 for (ch = 0; ch < s->channels; ch++) {
536 compute_exp_strategy_ch(exp_strategy, exp, ch, ch == s->lfe_channel);
542 * Set each encoded exponent in a block to the minimum of itself and the
543 * exponent in the same frequency bin of a following block.
544 * exp[i] = min(exp[i], exp1[i]
546 static void exponent_min(uint8_t exp[AC3_MAX_COEFS], uint8_t exp1[AC3_MAX_COEFS], int n)
549 for (i = 0; i < n; i++) {
550 if (exp1[i] < exp[i])
557 * Update the exponents so that they are the ones the decoder will decode.
558 * @return the number of bits used to encode the exponents.
560 static int encode_exponents_blk_ch(uint8_t encoded_exp[AC3_MAX_COEFS],
561 uint8_t exp[AC3_MAX_COEFS],
562 int nb_exps, int exp_strategy)
564 int group_size, nb_groups, i, j, k, exp_min;
565 uint8_t exp1[AC3_MAX_COEFS];
567 group_size = exp_strategy + (exp_strategy == EXP_D45);
568 nb_groups = ((nb_exps + (group_size * 3) - 4) / (3 * group_size)) * 3;
570 /* for each group, compute the minimum exponent */
571 exp1[0] = exp[0]; /* DC exponent is handled separately */
573 for (i = 1; i <= nb_groups; i++) {
575 assert(exp_min >= 0 && exp_min <= 24);
576 for (j = 1; j < group_size; j++) {
577 if (exp[k+j] < exp_min)
584 /* constraint for DC exponent */
588 /* decrease the delta between each groups to within 2 so that they can be
589 differentially encoded */
590 for (i = 1; i <= nb_groups; i++)
591 exp1[i] = FFMIN(exp1[i], exp1[i-1] + 2);
592 for (i = nb_groups-1; i >= 0; i--)
593 exp1[i] = FFMIN(exp1[i], exp1[i+1] + 2);
595 /* now we have the exponent values the decoder will see */
596 encoded_exp[0] = exp1[0];
598 for (i = 1; i <= nb_groups; i++) {
599 for (j = 0; j < group_size; j++)
600 encoded_exp[k+j] = exp1[i];
604 return 4 + (nb_groups / 3) * 7;
609 * Encode exponents from original extracted form to what the decoder will see.
610 * This copies and groups exponents based on exponent strategy and reduces
611 * deltas between adjacent exponent groups so that they can be differentially
613 * @return bits needed to encode the exponents
615 static int encode_exponents(AC3EncodeContext *s,
616 uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
617 uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
618 uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
620 int blk, blk1, blk2, ch;
624 for (ch = 0; ch < s->channels; ch++) {
625 /* for the EXP_REUSE case we select the min of the exponents */
627 while (blk < AC3_MAX_BLOCKS) {
629 while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1][ch] == EXP_REUSE) {
630 exponent_min(exp[blk][ch], exp[blk1][ch], s->nb_coefs[ch]);
633 frame_bits += encode_exponents_blk_ch(encoded_exp[blk][ch],
634 exp[blk][ch], s->nb_coefs[ch],
635 exp_strategy[blk][ch]);
636 /* copy encoded exponents for reuse case */
637 for (blk2 = blk+1; blk2 < blk1; blk2++) {
638 memcpy(encoded_exp[blk2][ch], encoded_exp[blk][ch],
639 s->nb_coefs[ch] * sizeof(uint8_t));
650 * Calculate final exponents from the supplied MDCT coefficients and exponent shift.
651 * Extract exponents from MDCT coefficients, calculate exponent strategies,
652 * and encode final exponents.
653 * @return bits needed to encode the exponents
655 static int process_exponents(AC3EncodeContext *s,
656 int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
657 int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
658 uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
659 uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
660 uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
662 extract_exponents(s, mdct_coef, exp_shift, exp);
664 compute_exp_strategy(s, exp_strategy, exp);
666 return encode_exponents(s, exp, exp_strategy, encoded_exp);
671 * Calculate the number of bits needed to encode a set of mantissas.
673 static int compute_mantissa_size(AC3EncodeContext *s, uint8_t *m, int nb_coefs)
678 for (i = 0; i < nb_coefs; i++) {
685 /* 3 mantissa in 5 bits */
686 if (s->mant1_cnt == 0)
688 if (++s->mant1_cnt == 3)
692 /* 3 mantissa in 7 bits */
693 if (s->mant2_cnt == 0)
695 if (++s->mant2_cnt == 3)
702 /* 2 mantissa in 7 bits */
703 if (s->mant4_cnt == 0)
705 if (++s->mant4_cnt == 2)
724 * Calculate masking curve based on the final exponents.
725 * Also calculate the power spectral densities to use in future calculations.
727 static void bit_alloc_masking(AC3EncodeContext *s,
728 uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
729 uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
730 int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
731 int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS])
734 int16_t band_psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS];
736 for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
737 for (ch = 0; ch < s->channels; ch++) {
738 if(exp_strategy[blk][ch] == EXP_REUSE) {
739 memcpy(psd[blk][ch], psd[blk-1][ch], AC3_MAX_COEFS*sizeof(psd[0][0][0]));
740 memcpy(mask[blk][ch], mask[blk-1][ch], AC3_CRITICAL_BANDS*sizeof(mask[0][0][0]));
742 ff_ac3_bit_alloc_calc_psd(encoded_exp[blk][ch], 0,
744 psd[blk][ch], band_psd[blk][ch]);
745 ff_ac3_bit_alloc_calc_mask(&s->bit_alloc, band_psd[blk][ch],
747 ff_ac3_fast_gain_tab[s->fast_gain_code[ch]],
748 ch == s->lfe_channel,
749 DBA_NONE, 0, NULL, NULL, NULL,
758 * Run the bit allocation with a given SNR offset.
759 * This calculates the bit allocation pointers that will be used to determine
760 * the quantization of each mantissa.
761 * @return the number of remaining bits (positive or negative) if the given
762 * SNR offset is used to quantize the mantissas.
764 static int bit_alloc(AC3EncodeContext *s,
765 int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS],
766 int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
767 uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
768 int frame_bits, int coarse_snr_offset, int fine_snr_offset)
773 snr_offset = (((coarse_snr_offset - 15) << 4) + fine_snr_offset) << 2;
775 for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
779 for (ch = 0; ch < s->channels; ch++) {
780 ff_ac3_bit_alloc_calc_bap(mask[blk][ch], psd[blk][ch], 0,
781 s->nb_coefs[ch], snr_offset,
782 s->bit_alloc.floor, ff_ac3_bap_tab,
784 frame_bits += compute_mantissa_size(s, bap[blk][ch], s->nb_coefs[ch]);
787 return 8 * s->frame_size - frame_bits;
794 * Perform bit allocation search.
795 * Finds the SNR offset value that maximizes quality and fits in the specified
796 * frame size. Output is the SNR offset and a set of bit allocation pointers
797 * used to quantize the mantissas.
799 static int compute_bit_allocation(AC3EncodeContext *s,
800 uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
801 uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
802 uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
806 int coarse_snr_offset, fine_snr_offset;
807 uint8_t bap1[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
808 int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
809 int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS];
810 static const int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 };
812 /* init default parameters */
813 s->slow_decay_code = 2;
814 s->fast_decay_code = 1;
815 s->slow_gain_code = 1;
816 s->db_per_bit_code = 2;
818 for (ch = 0; ch < s->channels; ch++)
819 s->fast_gain_code[ch] = 4;
821 /* compute real values */
822 s->bit_alloc.slow_decay = ff_ac3_slow_decay_tab[s->slow_decay_code] >> s->bit_alloc.sr_shift;
823 s->bit_alloc.fast_decay = ff_ac3_fast_decay_tab[s->fast_decay_code] >> s->bit_alloc.sr_shift;
824 s->bit_alloc.slow_gain = ff_ac3_slow_gain_tab[s->slow_gain_code];
825 s->bit_alloc.db_per_bit = ff_ac3_db_per_bit_tab[s->db_per_bit_code];
826 s->bit_alloc.floor = ff_ac3_floor_tab[s->floor_code];
830 // if (s->channel_mode == 2)
832 frame_bits += frame_bits_inc[s->channel_mode];
835 for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
836 frame_bits += s->fbw_channels * 2 + 2; /* blksw * c, dithflag * c, dynrnge, cplstre */
837 if (s->channel_mode == AC3_CHMODE_STEREO) {
838 frame_bits++; /* rematstr */
842 frame_bits += 2 * s->fbw_channels; /* chexpstr[2] * c */
844 frame_bits++; /* lfeexpstr */
845 for (ch = 0; ch < s->fbw_channels; ch++) {
846 if (exp_strategy[blk][ch] != EXP_REUSE)
847 frame_bits += 6 + 2; /* chbwcod[6], gainrng[2] */
849 frame_bits++; /* baie */
850 frame_bits++; /* snr */
851 frame_bits += 2; /* delta / skip */
853 frame_bits++; /* cplinu for block 0 */
855 /* sdcycod[2], fdcycod[2], sgaincod[2], dbpbcod[2], floorcod[3] */
857 /* (fsnoffset[4] + fgaincod[4]) * c */
858 frame_bits += 2*4 + 3 + 6 + s->channels * (4 + 3);
860 /* auxdatae, crcrsv */
866 /* calculate psd and masking curve before doing bit allocation */
867 bit_alloc_masking(s, encoded_exp, exp_strategy, psd, mask);
869 /* now the big work begins : do the bit allocation. Modify the snr
870 offset until we can pack everything in the requested frame size */
872 coarse_snr_offset = s->coarse_snr_offset;
873 while (coarse_snr_offset >= 0 &&
874 bit_alloc(s, mask, psd, bap, frame_bits, coarse_snr_offset, 0) < 0)
875 coarse_snr_offset -= SNR_INC1;
876 if (coarse_snr_offset < 0) {
877 av_log(NULL, AV_LOG_ERROR, "Bit allocation failed. Try increasing the bitrate.\n");
880 while (coarse_snr_offset + SNR_INC1 <= 63 &&
881 bit_alloc(s, mask, psd, bap1, frame_bits,
882 coarse_snr_offset + SNR_INC1, 0) >= 0) {
883 coarse_snr_offset += SNR_INC1;
884 memcpy(bap, bap1, sizeof(bap1));
886 while (coarse_snr_offset + 1 <= 63 &&
887 bit_alloc(s, mask, psd, bap1, frame_bits, coarse_snr_offset + 1, 0) >= 0) {
889 memcpy(bap, bap1, sizeof(bap1));
893 while (fine_snr_offset + SNR_INC1 <= 15 &&
894 bit_alloc(s, mask, psd, bap1, frame_bits,
895 coarse_snr_offset, fine_snr_offset + SNR_INC1) >= 0) {
896 fine_snr_offset += SNR_INC1;
897 memcpy(bap, bap1, sizeof(bap1));
899 while (fine_snr_offset + 1 <= 15 &&
900 bit_alloc(s, mask, psd, bap1, frame_bits,
901 coarse_snr_offset, fine_snr_offset + 1) >= 0) {
903 memcpy(bap, bap1, sizeof(bap1));
906 s->coarse_snr_offset = coarse_snr_offset;
907 for (ch = 0; ch < s->channels; ch++)
908 s->fine_snr_offset[ch] = fine_snr_offset;
915 * Write the AC-3 frame header to the output bitstream.
917 static void output_frame_header(AC3EncodeContext *s)
919 put_bits(&s->pb, 16, 0x0b77); /* frame header */
920 put_bits(&s->pb, 16, 0); /* crc1: will be filled later */
921 put_bits(&s->pb, 2, s->bit_alloc.sr_code);
922 put_bits(&s->pb, 6, s->frame_size_code + (s->frame_size - s->frame_size_min) / 2);
923 put_bits(&s->pb, 5, s->bitstream_id);
924 put_bits(&s->pb, 3, s->bitstream_mode);
925 put_bits(&s->pb, 3, s->channel_mode);
926 if ((s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO)
927 put_bits(&s->pb, 2, 1); /* XXX -4.5 dB */
928 if (s->channel_mode & 0x04)
929 put_bits(&s->pb, 2, 1); /* XXX -6 dB */
930 if (s->channel_mode == AC3_CHMODE_STEREO)
931 put_bits(&s->pb, 2, 0); /* surround not indicated */
932 put_bits(&s->pb, 1, s->lfe_on); /* LFE */
933 put_bits(&s->pb, 5, 31); /* dialog norm: -31 db */
934 put_bits(&s->pb, 1, 0); /* no compression control word */
935 put_bits(&s->pb, 1, 0); /* no lang code */
936 put_bits(&s->pb, 1, 0); /* no audio production info */
937 put_bits(&s->pb, 1, 0); /* no copyright */
938 put_bits(&s->pb, 1, 1); /* original bitstream */
939 put_bits(&s->pb, 1, 0); /* no time code 1 */
940 put_bits(&s->pb, 1, 0); /* no time code 2 */
941 put_bits(&s->pb, 1, 0); /* no additional bit stream info */
946 * Symmetric quantization on 'levels' levels.
948 static inline int sym_quant(int c, int e, int levels)
953 v = (levels * (c << e)) >> 24;
955 v = (levels >> 1) + v;
957 v = (levels * ((-c) << e)) >> 24;
959 v = (levels >> 1) - v;
961 assert (v >= 0 && v < levels);
967 * Asymmetric quantization on 2^qbits levels.
969 static inline int asym_quant(int c, int e, int qbits)
973 lshift = e + qbits - 24;
980 m = (1 << (qbits-1));
984 return v & ((1 << qbits)-1);
989 * Write one audio block to the output bitstream.
991 static void output_audio_block(AC3EncodeContext *s,
992 uint8_t exp_strategy[AC3_MAX_CHANNELS],
993 uint8_t encoded_exp[AC3_MAX_CHANNELS][AC3_MAX_COEFS],
994 uint8_t bap[AC3_MAX_CHANNELS][AC3_MAX_COEFS],
995 int32_t mdct_coef[AC3_MAX_CHANNELS][AC3_MAX_COEFS],
996 int8_t exp_shift[AC3_MAX_CHANNELS],
999 int ch, nb_groups, group_size, i, baie, rbnd;
1001 uint16_t qmant[AC3_MAX_CHANNELS][AC3_MAX_COEFS];
1003 int mant1_cnt, mant2_cnt, mant4_cnt;
1004 uint16_t *qmant1_ptr, *qmant2_ptr, *qmant4_ptr;
1005 int delta0, delta1, delta2;
1007 for (ch = 0; ch < s->fbw_channels; ch++)
1008 put_bits(&s->pb, 1, 0); /* no block switching */
1009 for (ch = 0; ch < s->fbw_channels; ch++)
1010 put_bits(&s->pb, 1, 1); /* no dither */
1011 put_bits(&s->pb, 1, 0); /* no dynamic range */
1013 put_bits(&s->pb, 1, 1); /* coupling strategy present */
1014 put_bits(&s->pb, 1, 0); /* no coupling strategy */
1016 put_bits(&s->pb, 1, 0); /* no new coupling strategy */
1019 if (s->channel_mode == AC3_CHMODE_STEREO) {
1021 /* first block must define rematrixing (rematstr) */
1022 put_bits(&s->pb, 1, 1);
1024 /* dummy rematrixing rematflg(1:4)=0 */
1025 for (rbnd = 0; rbnd < 4; rbnd++)
1026 put_bits(&s->pb, 1, 0);
1028 /* no matrixing (but should be used in the future) */
1029 put_bits(&s->pb, 1, 0);
1033 /* exponent strategy */
1034 for (ch = 0; ch < s->fbw_channels; ch++)
1035 put_bits(&s->pb, 2, exp_strategy[ch]);
1038 put_bits(&s->pb, 1, exp_strategy[s->lfe_channel]);
1041 for (ch = 0; ch < s->fbw_channels; ch++) {
1042 if (exp_strategy[ch] != EXP_REUSE)
1043 put_bits(&s->pb, 6, s->bandwidth_code[ch]);
1047 for (ch = 0; ch < s->channels; ch++) {
1048 if (exp_strategy[ch] == EXP_REUSE)
1050 group_size = exp_strategy[ch] + (exp_strategy[ch] == EXP_D45);
1051 nb_groups = (s->nb_coefs[ch] + (group_size * 3) - 4) / (3 * group_size);
1052 p = encoded_exp[ch];
1054 /* first exponent */
1056 put_bits(&s->pb, 4, exp1);
1058 /* next ones are delta encoded */
1059 for (i = 0; i < nb_groups; i++) {
1060 /* merge three delta in one code */
1064 delta0 = exp1 - exp0 + 2;
1069 delta1 = exp1 - exp0 + 2;
1074 delta2 = exp1 - exp0 + 2;
1076 put_bits(&s->pb, 7, ((delta0 * 5 + delta1) * 5) + delta2);
1079 if (ch != s->lfe_channel)
1080 put_bits(&s->pb, 2, 0); /* no gain range info */
1083 /* bit allocation info */
1084 baie = (block_num == 0);
1085 put_bits(&s->pb, 1, baie);
1087 put_bits(&s->pb, 2, s->slow_decay_code);
1088 put_bits(&s->pb, 2, s->fast_decay_code);
1089 put_bits(&s->pb, 2, s->slow_gain_code);
1090 put_bits(&s->pb, 2, s->db_per_bit_code);
1091 put_bits(&s->pb, 3, s->floor_code);
1095 put_bits(&s->pb, 1, baie);
1097 put_bits(&s->pb, 6, s->coarse_snr_offset);
1098 for (ch = 0; ch < s->channels; ch++) {
1099 put_bits(&s->pb, 4, s->fine_snr_offset[ch]);
1100 put_bits(&s->pb, 3, s->fast_gain_code[ch]);
1104 put_bits(&s->pb, 1, 0); /* no delta bit allocation */
1105 put_bits(&s->pb, 1, 0); /* no data to skip */
1107 /* mantissa encoding : we use two passes to handle the grouping. A
1108 one pass method may be faster, but it would necessitate to
1109 modify the output stream. */
1111 /* first pass: quantize */
1112 mant1_cnt = mant2_cnt = mant4_cnt = 0;
1113 qmant1_ptr = qmant2_ptr = qmant4_ptr = NULL;
1115 for (ch = 0; ch < s->channels; ch++) {
1118 for (i = 0; i < s->nb_coefs[ch]; i++) {
1119 c = mdct_coef[ch][i];
1120 e = encoded_exp[ch][i] - exp_shift[ch];
1127 v = sym_quant(c, e, 3);
1128 switch (mant1_cnt) {
1130 qmant1_ptr = &qmant[ch][i];
1135 *qmant1_ptr += 3 * v;
1147 v = sym_quant(c, e, 5);
1148 switch (mant2_cnt) {
1150 qmant2_ptr = &qmant[ch][i];
1155 *qmant2_ptr += 5 * v;
1167 v = sym_quant(c, e, 7);
1170 v = sym_quant(c, e, 11);
1171 switch (mant4_cnt) {
1173 qmant4_ptr = &qmant[ch][i];
1185 v = sym_quant(c, e, 15);
1188 v = asym_quant(c, e, 14);
1191 v = asym_quant(c, e, 16);
1194 v = asym_quant(c, e, b - 1);
1201 /* second pass : output the values */
1202 for (ch = 0; ch < s->channels; ch++) {
1205 for (i = 0; i < s->nb_coefs[ch]; i++) {
1210 case 1: if (q != 128) put_bits(&s->pb, 5, q); break;
1211 case 2: if (q != 128) put_bits(&s->pb, 7, q); break;
1212 case 3: put_bits(&s->pb, 3, q); break;
1213 case 4: if (q != 128) put_bits(&s->pb, 7, q); break;
1214 case 14: put_bits(&s->pb, 14, q); break;
1215 case 15: put_bits(&s->pb, 16, q); break;
1216 default: put_bits(&s->pb, b-1, q); break;
1223 /** CRC-16 Polynomial */
1224 #define CRC16_POLY ((1 << 0) | (1 << 2) | (1 << 15) | (1 << 16))
1227 static unsigned int mul_poly(unsigned int a, unsigned int b, unsigned int poly)
1244 static unsigned int pow_poly(unsigned int a, unsigned int n, unsigned int poly)
1250 r = mul_poly(r, a, poly);
1251 a = mul_poly(a, a, poly);
1259 * Fill the end of the frame with 0's and compute the two CRCs.
1261 static void output_frame_end(AC3EncodeContext *s)
1263 int frame_size, frame_size_58, pad_bytes, crc1, crc2, crc_inv;
1266 frame_size = s->frame_size; /* frame size in words */
1267 /* align to 8 bits */
1268 flush_put_bits(&s->pb);
1269 /* add zero bytes to reach the frame size */
1271 pad_bytes = s->frame_size - (put_bits_ptr(&s->pb) - frame) - 2;
1272 assert(pad_bytes >= 0);
1274 memset(put_bits_ptr(&s->pb), 0, pad_bytes);
1276 /* Now we must compute both crcs : this is not so easy for crc1
1277 because it is at the beginning of the data... */
1278 frame_size_58 = ((frame_size >> 2) + (frame_size >> 4)) << 1;
1280 crc1 = av_bswap16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
1281 frame + 4, frame_size_58 - 4));
1283 /* XXX: could precompute crc_inv */
1284 crc_inv = pow_poly((CRC16_POLY >> 1), (8 * frame_size_58) - 16, CRC16_POLY);
1285 crc1 = mul_poly(crc_inv, crc1, CRC16_POLY);
1286 AV_WB16(frame + 2, crc1);
1288 crc2 = av_bswap16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
1289 frame + frame_size_58,
1290 frame_size - frame_size_58 - 2));
1291 AV_WB16(frame + frame_size - 2, crc2);
1296 * Write the frame to the output bitstream.
1298 static void output_frame(AC3EncodeContext *s,
1299 unsigned char *frame,
1300 uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
1301 uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
1302 uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
1303 int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
1304 int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS])
1308 init_put_bits(&s->pb, frame, AC3_MAX_CODED_FRAME_SIZE);
1310 output_frame_header(s);
1312 for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
1313 output_audio_block(s, exp_strategy[blk], encoded_exp[blk],
1314 bap[blk], mdct_coef[blk], exp_shift[blk], blk);
1317 output_frame_end(s);
1322 * Encode a single AC-3 frame.
1324 static int ac3_encode_frame(AVCodecContext *avctx,
1325 unsigned char *frame, int buf_size, void *data)
1327 AC3EncodeContext *s = avctx->priv_data;
1328 const int16_t *samples = data;
1329 int16_t planar_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE+AC3_FRAME_SIZE];
1330 int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
1331 uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
1332 uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS];
1333 uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
1334 uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
1335 int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS];
1338 if (s->bit_alloc.sr_code == 1)
1339 adjust_frame_size(s);
1341 deinterleave_input_samples(s, samples, planar_samples);
1343 apply_mdct(s, planar_samples, exp_shift, mdct_coef);
1345 frame_bits = process_exponents(s, mdct_coef, exp_shift, exp, exp_strategy, encoded_exp);
1347 compute_bit_allocation(s, bap, encoded_exp, exp_strategy, frame_bits);
1349 output_frame(s, frame, exp_strategy, encoded_exp, bap, mdct_coef, exp_shift);
1351 return s->frame_size;
1356 * Finalize encoding and free any memory allocated by the encoder.
1358 static av_cold int ac3_encode_close(AVCodecContext *avctx)
1360 av_freep(&avctx->coded_frame);
1366 * Set channel information during initialization.
1368 static av_cold int set_channel_info(AC3EncodeContext *s, int channels,
1369 int64_t *channel_layout)
1373 if (channels < 1 || channels > AC3_MAX_CHANNELS)
1374 return AVERROR(EINVAL);
1375 if ((uint64_t)*channel_layout > 0x7FF)
1376 return AVERROR(EINVAL);
1377 ch_layout = *channel_layout;
1379 ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL);
1380 if (av_get_channel_layout_nb_channels(ch_layout) != channels)
1381 return AVERROR(EINVAL);
1383 s->lfe_on = !!(ch_layout & AV_CH_LOW_FREQUENCY);
1384 s->channels = channels;
1385 s->fbw_channels = channels - s->lfe_on;
1386 s->lfe_channel = s->lfe_on ? s->fbw_channels : -1;
1388 ch_layout -= AV_CH_LOW_FREQUENCY;
1390 switch (ch_layout) {
1391 case AV_CH_LAYOUT_MONO: s->channel_mode = AC3_CHMODE_MONO; break;
1392 case AV_CH_LAYOUT_STEREO: s->channel_mode = AC3_CHMODE_STEREO; break;
1393 case AV_CH_LAYOUT_SURROUND: s->channel_mode = AC3_CHMODE_3F; break;
1394 case AV_CH_LAYOUT_2_1: s->channel_mode = AC3_CHMODE_2F1R; break;
1395 case AV_CH_LAYOUT_4POINT0: s->channel_mode = AC3_CHMODE_3F1R; break;
1396 case AV_CH_LAYOUT_QUAD:
1397 case AV_CH_LAYOUT_2_2: s->channel_mode = AC3_CHMODE_2F2R; break;
1398 case AV_CH_LAYOUT_5POINT0:
1399 case AV_CH_LAYOUT_5POINT0_BACK: s->channel_mode = AC3_CHMODE_3F2R; break;
1401 return AVERROR(EINVAL);
1404 s->channel_map = ff_ac3_enc_channel_map[s->channel_mode][s->lfe_on];
1405 *channel_layout = ch_layout;
1407 *channel_layout |= AV_CH_LOW_FREQUENCY;
1413 static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s)
1417 /* validate channel layout */
1418 if (!avctx->channel_layout) {
1419 av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
1420 "encoder will guess the layout, but it "
1421 "might be incorrect.\n");
1423 ret = set_channel_info(s, avctx->channels, &avctx->channel_layout);
1425 av_log(avctx, AV_LOG_ERROR, "invalid channel layout\n");
1429 /* validate sample rate */
1430 for (i = 0; i < 9; i++) {
1431 if ((ff_ac3_sample_rate_tab[i / 3] >> (i % 3)) == avctx->sample_rate)
1435 av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n");
1436 return AVERROR(EINVAL);
1438 s->sample_rate = avctx->sample_rate;
1439 s->bit_alloc.sr_shift = i % 3;
1440 s->bit_alloc.sr_code = i / 3;
1442 /* validate bit rate */
1443 for (i = 0; i < 19; i++) {
1444 if ((ff_ac3_bitrate_tab[i] >> s->bit_alloc.sr_shift)*1000 == avctx->bit_rate)
1448 av_log(avctx, AV_LOG_ERROR, "invalid bit rate\n");
1449 return AVERROR(EINVAL);
1451 s->bit_rate = avctx->bit_rate;
1452 s->frame_size_code = i << 1;
1459 * Set bandwidth for all channels.
1460 * The user can optionally supply a cutoff frequency. Otherwise an appropriate
1461 * default value will be used.
1463 static av_cold void set_bandwidth(AC3EncodeContext *s, int cutoff)
1468 /* calculate bandwidth based on user-specified cutoff frequency */
1470 cutoff = av_clip(cutoff, 1, s->sample_rate >> 1);
1471 fbw_coeffs = cutoff * 2 * AC3_MAX_COEFS / s->sample_rate;
1472 bw_code = av_clip((fbw_coeffs - 73) / 3, 0, 60);
1474 /* use default bandwidth setting */
1475 /* XXX: should compute the bandwidth according to the frame
1476 size, so that we avoid annoying high frequency artifacts */
1480 /* set number of coefficients for each channel */
1481 for (ch = 0; ch < s->fbw_channels; ch++) {
1482 s->bandwidth_code[ch] = bw_code;
1483 s->nb_coefs[ch] = bw_code * 3 + 73;
1486 s->nb_coefs[s->lfe_channel] = 7; /* LFE channel always has 7 coefs */
1491 * Initialize the encoder.
1493 static av_cold int ac3_encode_init(AVCodecContext *avctx)
1495 AC3EncodeContext *s = avctx->priv_data;
1498 avctx->frame_size = AC3_FRAME_SIZE;
1502 ret = validate_options(avctx, s);
1506 s->bitstream_id = 8 + s->bit_alloc.sr_shift;
1507 s->bitstream_mode = 0; /* complete main audio service */
1509 s->frame_size_min = 2 * ff_ac3_frame_size_tab[s->frame_size_code][s->bit_alloc.sr_code];
1510 s->bits_written = 0;
1511 s->samples_written = 0;
1512 s->frame_size = s->frame_size_min;
1514 set_bandwidth(s, avctx->cutoff);
1516 /* initial snr offset */
1517 s->coarse_snr_offset = 40;
1521 avctx->coded_frame= avcodec_alloc_frame();
1522 avctx->coded_frame->key_frame= 1;
1529 /*************************************************************************/
1532 #include "libavutil/lfg.h"
1534 #define FN (MDCT_SAMPLES/4)
1537 static void fft_test(AVLFG *lfg)
1539 IComplex in[FN], in1[FN];
1541 float sum_re, sum_im, a;
1543 for (i = 0; i < FN; i++) {
1544 in[i].re = av_lfg_get(lfg) % 65535 - 32767;
1545 in[i].im = av_lfg_get(lfg) % 65535 - 32767;
1551 for (k = 0; k < FN; k++) {
1554 for (n = 0; n < FN; n++) {
1555 a = -2 * M_PI * (n * k) / FN;
1556 sum_re += in1[n].re * cos(a) - in1[n].im * sin(a);
1557 sum_im += in1[n].re * sin(a) + in1[n].im * cos(a);
1559 av_log(NULL, AV_LOG_DEBUG, "%3d: %6d,%6d %6.0f,%6.0f\n",
1560 k, in[k].re, in[k].im, sum_re / FN, sum_im / FN);
1565 static void mdct_test(AVLFG *lfg)
1567 int16_t input[MDCT_SAMPLES];
1568 int32_t output[AC3_MAX_COEFS];
1569 float input1[MDCT_SAMPLES];
1570 float output1[AC3_MAX_COEFS];
1571 float s, a, err, e, emax;
1574 for (i = 0; i < MDCT_SAMPLES; i++) {
1575 input[i] = (av_lfg_get(lfg) % 65535 - 32767) * 9 / 10;
1576 input1[i] = input[i];
1579 mdct512(output, input);
1582 for (k = 0; k < AC3_MAX_COEFS; k++) {
1584 for (n = 0; n < MDCT_SAMPLES; n++) {
1585 a = (2*M_PI*(2*n+1+MDCT_SAMPLES/2)*(2*k+1) / (4 * MDCT_SAMPLES));
1586 s += input1[n] * cos(a);
1588 output1[k] = -2 * s / MDCT_SAMPLES;
1593 for (i = 0; i < AC3_MAX_COEFS; i++) {
1594 av_log(NULL, AV_LOG_DEBUG, "%3d: %7d %7.0f\n", i, output[i], output1[i]);
1595 e = output[i] - output1[i];
1600 av_log(NULL, AV_LOG_DEBUG, "err2=%f emax=%f\n", err / AC3_MAX_COEFS, emax);
1608 av_log_set_level(AV_LOG_DEBUG);
1619 AVCodec ac3_encoder = {
1623 sizeof(AC3EncodeContext),
1628 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
1629 .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
1630 .channel_layouts = (const int64_t[]){
1632 AV_CH_LAYOUT_STEREO,
1634 AV_CH_LAYOUT_SURROUND,
1637 AV_CH_LAYOUT_4POINT0,
1638 AV_CH_LAYOUT_5POINT0,
1639 AV_CH_LAYOUT_5POINT0_BACK,
1640 (AV_CH_LAYOUT_MONO | AV_CH_LOW_FREQUENCY),
1641 (AV_CH_LAYOUT_STEREO | AV_CH_LOW_FREQUENCY),
1642 (AV_CH_LAYOUT_2_1 | AV_CH_LOW_FREQUENCY),
1643 (AV_CH_LAYOUT_SURROUND | AV_CH_LOW_FREQUENCY),
1644 (AV_CH_LAYOUT_2_2 | AV_CH_LOW_FREQUENCY),
1645 (AV_CH_LAYOUT_QUAD | AV_CH_LOW_FREQUENCY),
1646 (AV_CH_LAYOUT_4POINT0 | AV_CH_LOW_FREQUENCY),
1647 AV_CH_LAYOUT_5POINT1,
1648 AV_CH_LAYOUT_5POINT1_BACK,