2 * The simplest mpeg audio layer 2 encoder
3 * Copyright (c) 2000, 2001 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
23 * @file libavcodec/mpegaudio.c
24 * The simplest mpeg audio layer 2 encoder.
30 #undef CONFIG_MPEGAUDIO_HP
31 #define CONFIG_MPEGAUDIO_HP 0
32 #include "mpegaudio.h"
34 /* currently, cannot change these constants (need to modify
35 quantization stage) */
36 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
38 #define SAMPLES_BUF_SIZE 4096
40 typedef struct MpegAudioContext {
44 int lsf; /* 1 if mpeg2 low bitrate selected */
45 int bitrate_index; /* bit rate */
47 int frame_size; /* frame size, in bits, without padding */
48 int64_t nb_samples; /* total number of samples encoded */
49 /* padding computation */
50 int frame_frac, frame_frac_incr, do_padding;
51 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
52 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
53 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
54 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
55 /* code to group 3 scale factors */
56 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
57 int sblimit; /* number of used subbands */
58 const unsigned char *alloc_table;
61 /* define it to use floats in quantization (I don't like floats !) */
64 #include "mpegaudiodata.h"
65 #include "mpegaudiotab.h"
67 static av_cold int MPA_encode_init(AVCodecContext *avctx)
69 MpegAudioContext *s = avctx->priv_data;
70 int freq = avctx->sample_rate;
71 int bitrate = avctx->bit_rate;
72 int channels = avctx->channels;
76 if (channels <= 0 || channels > 2){
77 av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
80 bitrate = bitrate / 1000;
81 s->nb_channels = channels;
83 s->bit_rate = bitrate * 1000;
84 avctx->frame_size = MPA_FRAME_SIZE;
89 if (ff_mpa_freq_tab[i] == freq)
91 if ((ff_mpa_freq_tab[i] / 2) == freq) {
97 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
102 /* encoding bitrate & frequency */
104 if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
108 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
111 s->bitrate_index = i;
113 /* compute total header size & pad bit */
115 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
116 s->frame_size = ((int)a) * 8;
118 /* frame fractional size to compute padding */
120 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
122 /* select the right allocation table */
123 table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
125 /* number of used subbands */
126 s->sblimit = ff_mpa_sblimit_table[table];
127 s->alloc_table = ff_mpa_alloc_tables[table];
129 dprintf(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
130 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
132 for(i=0;i<s->nb_channels;i++)
133 s->samples_offset[i] = 0;
137 v = ff_mpa_enwindow[i];
139 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
145 filter_bank[512 - i] = v;
149 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
152 scale_factor_table[i] = v;
154 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
157 scale_factor_shift[i] = 21 - P - (i / 3);
158 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
173 scale_diff_table[i] = v;
177 v = ff_mpa_quant_bits[i];
182 total_quant_bits[i] = 12 * v;
185 avctx->coded_frame= avcodec_alloc_frame();
186 avctx->coded_frame->key_frame= 1;
191 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
192 static void idct32(int *out, int *tab)
196 const int *xp = costab32;
198 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
237 x3 = MUL(t[16], FIX(SQRT2*0.5));
241 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
242 x1 = MUL((t[8] - x2), xp[0]);
243 x2 = MUL((t[8] + x2), xp[1]);
256 xr = MUL(t[28],xp[0]);
260 xr = MUL(t[4],xp[1]);
261 t[ 4] = (t[24] - xr);
262 t[24] = (t[24] + xr);
264 xr = MUL(t[20],xp[2]);
268 xr = MUL(t[12],xp[3]);
269 t[12] = (t[16] - xr);
270 t[16] = (t[16] + xr);
275 for (i = 0; i < 4; i++) {
276 xr = MUL(tab[30-i*4],xp[0]);
277 tab[30-i*4] = (tab[i*4] - xr);
278 tab[ i*4] = (tab[i*4] + xr);
280 xr = MUL(tab[ 2+i*4],xp[1]);
281 tab[ 2+i*4] = (tab[28-i*4] - xr);
282 tab[28-i*4] = (tab[28-i*4] + xr);
284 xr = MUL(tab[31-i*4],xp[0]);
285 tab[31-i*4] = (tab[1+i*4] - xr);
286 tab[ 1+i*4] = (tab[1+i*4] + xr);
288 xr = MUL(tab[ 3+i*4],xp[1]);
289 tab[ 3+i*4] = (tab[29-i*4] - xr);
290 tab[29-i*4] = (tab[29-i*4] + xr);
298 xr = MUL(t1[0], *xp);
307 out[i] = tab[bitinv32[i]];
311 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
313 static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
316 int sum, offset, i, j;
321 // print_pow1(samples, 1152);
323 offset = s->samples_offset[ch];
324 out = &s->sb_samples[ch][0][0][0];
326 /* 32 samples at once */
328 s->samples_buf[ch][offset + (31 - i)] = samples[0];
333 p = s->samples_buf[ch] + offset;
337 sum = p[0*64] * q[0*64];
338 sum += p[1*64] * q[1*64];
339 sum += p[2*64] * q[2*64];
340 sum += p[3*64] * q[3*64];
341 sum += p[4*64] * q[4*64];
342 sum += p[5*64] * q[5*64];
343 sum += p[6*64] * q[6*64];
344 sum += p[7*64] * q[7*64];
349 tmp1[0] = tmp[16] >> WSHIFT;
350 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
351 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
355 /* advance of 32 samples */
358 /* handle the wrap around */
360 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
361 s->samples_buf[ch], (512 - 32) * 2);
362 offset = SAMPLES_BUF_SIZE - 512;
365 s->samples_offset[ch] = offset;
367 // print_pow(s->sb_samples, 1152);
370 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
371 unsigned char scale_factors[SBLIMIT][3],
372 int sb_samples[3][12][SBLIMIT],
375 int *p, vmax, v, n, i, j, k, code;
377 unsigned char *sf = &scale_factors[0][0];
379 for(j=0;j<sblimit;j++) {
381 /* find the max absolute value */
382 p = &sb_samples[i][0][j];
390 /* compute the scale factor index using log 2 computations */
393 /* n is the position of the MSB of vmax. now
394 use at most 2 compares to find the index */
395 index = (21 - n) * 3 - 3;
397 while (vmax <= scale_factor_table[index+1])
400 index = 0; /* very unlikely case of overflow */
403 index = 62; /* value 63 is not allowed */
407 printf("%2d:%d in=%x %x %d\n",
408 j, i, vmax, scale_factor_table[index], index);
410 /* store the scale factor */
411 assert(index >=0 && index <= 63);
415 /* compute the transmission factor : look if the scale factors
416 are close enough to each other */
417 d1 = scale_diff_table[sf[0] - sf[1] + 64];
418 d2 = scale_diff_table[sf[1] - sf[2] + 64];
420 /* handle the 25 cases */
421 switch(d1 * 5 + d2) {
453 sf[1] = sf[2] = sf[0];
458 sf[0] = sf[1] = sf[2];
464 sf[0] = sf[2] = sf[1];
470 sf[1] = sf[2] = sf[0];
473 assert(0); //cannot happen
474 code = 0; /* kill warning */
478 printf("%d: %2d %2d %2d %d %d -> %d\n", j,
479 sf[0], sf[1], sf[2], d1, d2, code);
481 scale_code[j] = code;
486 /* The most important function : psycho acoustic module. In this
487 encoder there is basically none, so this is the worst you can do,
488 but also this is the simpler. */
489 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
493 for(i=0;i<s->sblimit;i++) {
494 smr[i] = (int)(fixed_smr[i] * 10);
499 #define SB_NOTALLOCATED 0
500 #define SB_ALLOCATED 1
503 /* Try to maximize the smr while using a number of bits inferior to
504 the frame size. I tried to make the code simpler, faster and
505 smaller than other encoders :-) */
506 static void compute_bit_allocation(MpegAudioContext *s,
507 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
508 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
511 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
513 short smr[MPA_MAX_CHANNELS][SBLIMIT];
514 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
515 const unsigned char *alloc;
517 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
518 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
519 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
521 /* compute frame size and padding */
522 max_frame_size = s->frame_size;
523 s->frame_frac += s->frame_frac_incr;
524 if (s->frame_frac >= 65536) {
525 s->frame_frac -= 65536;
532 /* compute the header + bit alloc size */
533 current_frame_size = 32;
534 alloc = s->alloc_table;
535 for(i=0;i<s->sblimit;i++) {
537 current_frame_size += incr * s->nb_channels;
541 /* look for the subband with the largest signal to mask ratio */
545 for(ch=0;ch<s->nb_channels;ch++) {
546 for(i=0;i<s->sblimit;i++) {
547 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
548 max_smr = smr[ch][i];
555 printf("current=%d max=%d max_sb=%d alloc=%d\n",
556 current_frame_size, max_frame_size, max_sb,
562 /* find alloc table entry (XXX: not optimal, should use
564 alloc = s->alloc_table;
565 for(i=0;i<max_sb;i++) {
566 alloc += 1 << alloc[0];
569 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
570 /* nothing was coded for this band: add the necessary bits */
571 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
572 incr += total_quant_bits[alloc[1]];
574 /* increments bit allocation */
575 b = bit_alloc[max_ch][max_sb];
576 incr = total_quant_bits[alloc[b + 1]] -
577 total_quant_bits[alloc[b]];
580 if (current_frame_size + incr <= max_frame_size) {
581 /* can increase size */
582 b = ++bit_alloc[max_ch][max_sb];
583 current_frame_size += incr;
584 /* decrease smr by the resolution we added */
585 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
586 /* max allocation size reached ? */
587 if (b == ((1 << alloc[0]) - 1))
588 subband_status[max_ch][max_sb] = SB_NOMORE;
590 subband_status[max_ch][max_sb] = SB_ALLOCATED;
592 /* cannot increase the size of this subband */
593 subband_status[max_ch][max_sb] = SB_NOMORE;
596 *padding = max_frame_size - current_frame_size;
597 assert(*padding >= 0);
600 for(i=0;i<s->sblimit;i++) {
601 printf("%d ", bit_alloc[i]);
608 * Output the mpeg audio layer 2 frame. Note how the code is small
609 * compared to other encoders :-)
611 static void encode_frame(MpegAudioContext *s,
612 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
615 int i, j, k, l, bit_alloc_bits, b, ch;
618 PutBitContext *p = &s->pb;
622 put_bits(p, 12, 0xfff);
623 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
624 put_bits(p, 2, 4-2); /* layer 2 */
625 put_bits(p, 1, 1); /* no error protection */
626 put_bits(p, 4, s->bitrate_index);
627 put_bits(p, 2, s->freq_index);
628 put_bits(p, 1, s->do_padding); /* use padding */
629 put_bits(p, 1, 0); /* private_bit */
630 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
631 put_bits(p, 2, 0); /* mode_ext */
632 put_bits(p, 1, 0); /* no copyright */
633 put_bits(p, 1, 1); /* original */
634 put_bits(p, 2, 0); /* no emphasis */
638 for(i=0;i<s->sblimit;i++) {
639 bit_alloc_bits = s->alloc_table[j];
640 for(ch=0;ch<s->nb_channels;ch++) {
641 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
643 j += 1 << bit_alloc_bits;
647 for(i=0;i<s->sblimit;i++) {
648 for(ch=0;ch<s->nb_channels;ch++) {
649 if (bit_alloc[ch][i])
650 put_bits(p, 2, s->scale_code[ch][i]);
655 for(i=0;i<s->sblimit;i++) {
656 for(ch=0;ch<s->nb_channels;ch++) {
657 if (bit_alloc[ch][i]) {
658 sf = &s->scale_factors[ch][i][0];
659 switch(s->scale_code[ch][i]) {
661 put_bits(p, 6, sf[0]);
662 put_bits(p, 6, sf[1]);
663 put_bits(p, 6, sf[2]);
667 put_bits(p, 6, sf[0]);
668 put_bits(p, 6, sf[2]);
671 put_bits(p, 6, sf[0]);
678 /* quantization & write sub band samples */
683 for(i=0;i<s->sblimit;i++) {
684 bit_alloc_bits = s->alloc_table[j];
685 for(ch=0;ch<s->nb_channels;ch++) {
686 b = bit_alloc[ch][i];
688 int qindex, steps, m, sample, bits;
689 /* we encode 3 sub band samples of the same sub band at a time */
690 qindex = s->alloc_table[j+b];
691 steps = ff_mpa_quant_steps[qindex];
693 sample = s->sb_samples[ch][k][l + m][i];
694 /* divide by scale factor */
698 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
699 q[m] = (int)((a + 1.0) * steps * 0.5);
703 int q1, e, shift, mult;
704 e = s->scale_factors[ch][i][k];
705 shift = scale_factor_shift[e];
706 mult = scale_factor_mult[e];
708 /* normalize to P bits */
710 q1 = sample << (-shift);
712 q1 = sample >> shift;
713 q1 = (q1 * mult) >> P;
714 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
719 assert(q[m] >= 0 && q[m] < steps);
721 bits = ff_mpa_quant_bits[qindex];
723 /* group the 3 values to save bits */
725 q[0] + steps * (q[1] + steps * q[2]));
727 printf("%d: gr1 %d\n",
728 i, q[0] + steps * (q[1] + steps * q[2]));
732 printf("%d: gr3 %d %d %d\n",
733 i, q[0], q[1], q[2]);
735 put_bits(p, bits, q[0]);
736 put_bits(p, bits, q[1]);
737 put_bits(p, bits, q[2]);
741 /* next subband in alloc table */
742 j += 1 << bit_alloc_bits;
748 for(i=0;i<padding;i++)
755 static int MPA_encode_frame(AVCodecContext *avctx,
756 unsigned char *frame, int buf_size, void *data)
758 MpegAudioContext *s = avctx->priv_data;
759 short *samples = data;
760 short smr[MPA_MAX_CHANNELS][SBLIMIT];
761 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
764 for(i=0;i<s->nb_channels;i++) {
765 filter(s, i, samples + i, s->nb_channels);
768 for(i=0;i<s->nb_channels;i++) {
769 compute_scale_factors(s->scale_code[i], s->scale_factors[i],
770 s->sb_samples[i], s->sblimit);
772 for(i=0;i<s->nb_channels;i++) {
773 psycho_acoustic_model(s, smr[i]);
775 compute_bit_allocation(s, smr, bit_alloc, &padding);
777 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
779 encode_frame(s, bit_alloc, padding);
781 s->nb_samples += MPA_FRAME_SIZE;
782 return put_bits_ptr(&s->pb) - s->pb.buf;
785 static av_cold int MPA_encode_close(AVCodecContext *avctx)
787 av_freep(&avctx->coded_frame);
791 AVCodec mp2_encoder = {
795 sizeof(MpegAudioContext),
800 .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
801 .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),