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
24 * The simplest mpeg audio layer 2 encoder.
27 #include "libavutil/channel_layout.h"
33 #define FRAC_BITS 15 /* fractional bits for sb_samples and dct */
34 #define WFRAC_BITS 14 /* fractional bits for window */
36 /* define it to use floats in quantization (I don't like floats !) */
39 #include "mpegaudio.h"
40 #include "mpegaudiodsp.h"
41 #include "mpegaudiodata.h"
42 #include "mpegaudiotab.h"
44 /* currently, cannot change these constants (need to modify
45 quantization stage) */
46 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
48 #define SAMPLES_BUF_SIZE 4096
50 typedef struct MpegAudioContext {
53 int lsf; /* 1 if mpeg2 low bitrate selected */
54 int bitrate_index; /* bit rate */
56 int frame_size; /* frame size, in bits, without padding */
57 /* padding computation */
58 int frame_frac, frame_frac_incr, do_padding;
59 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
60 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
61 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
62 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
63 /* code to group 3 scale factors */
64 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
65 int sblimit; /* number of used subbands */
66 const unsigned char *alloc_table;
67 int16_t filter_bank[512];
68 int scale_factor_table[64];
69 unsigned char scale_diff_table[128];
71 float scale_factor_inv_table[64];
73 int8_t scale_factor_shift[64];
74 unsigned short scale_factor_mult[64];
76 unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
79 static av_cold int MPA_encode_init(AVCodecContext *avctx)
81 MpegAudioContext *s = avctx->priv_data;
82 int freq = avctx->sample_rate;
83 int bitrate = avctx->bit_rate;
84 int channels = avctx->channels;
88 if (channels <= 0 || channels > 2){
89 av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
90 return AVERROR(EINVAL);
92 bitrate = bitrate / 1000;
93 s->nb_channels = channels;
94 avctx->frame_size = MPA_FRAME_SIZE;
95 avctx->delay = 512 - 32 + 1;
100 if (avpriv_mpa_freq_tab[i] == freq)
102 if ((avpriv_mpa_freq_tab[i] / 2) == freq) {
108 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
109 return AVERROR(EINVAL);
113 /* encoding bitrate & frequency */
115 if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
119 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
120 return AVERROR(EINVAL);
122 s->bitrate_index = i;
124 /* compute total header size & pad bit */
126 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
127 s->frame_size = ((int)a) * 8;
129 /* frame fractional size to compute padding */
131 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
133 /* select the right allocation table */
134 table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
136 /* number of used subbands */
137 s->sblimit = ff_mpa_sblimit_table[table];
138 s->alloc_table = ff_mpa_alloc_tables[table];
140 av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
141 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
143 for(i=0;i<s->nb_channels;i++)
144 s->samples_offset[i] = 0;
148 v = ff_mpa_enwindow[i];
150 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
152 s->filter_bank[i] = v;
156 s->filter_bank[512 - i] = v;
160 v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
163 s->scale_factor_table[i] = v;
165 s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
168 s->scale_factor_shift[i] = 21 - P - (i / 3);
169 s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
184 s->scale_diff_table[i] = v;
188 v = ff_mpa_quant_bits[i];
193 s->total_quant_bits[i] = 12 * v;
199 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
200 static void idct32(int *out, int *tab)
204 const int *xp = costab32;
206 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
245 x3 = MUL(t[16], FIX(SQRT2*0.5));
249 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
250 x1 = MUL((t[8] - x2), xp[0]);
251 x2 = MUL((t[8] + x2), xp[1]);
264 xr = MUL(t[28],xp[0]);
268 xr = MUL(t[4],xp[1]);
269 t[ 4] = (t[24] - xr);
270 t[24] = (t[24] + xr);
272 xr = MUL(t[20],xp[2]);
276 xr = MUL(t[12],xp[3]);
277 t[12] = (t[16] - xr);
278 t[16] = (t[16] + xr);
283 for (i = 0; i < 4; i++) {
284 xr = MUL(tab[30-i*4],xp[0]);
285 tab[30-i*4] = (tab[i*4] - xr);
286 tab[ i*4] = (tab[i*4] + xr);
288 xr = MUL(tab[ 2+i*4],xp[1]);
289 tab[ 2+i*4] = (tab[28-i*4] - xr);
290 tab[28-i*4] = (tab[28-i*4] + xr);
292 xr = MUL(tab[31-i*4],xp[0]);
293 tab[31-i*4] = (tab[1+i*4] - xr);
294 tab[ 1+i*4] = (tab[1+i*4] + xr);
296 xr = MUL(tab[ 3+i*4],xp[1]);
297 tab[ 3+i*4] = (tab[29-i*4] - xr);
298 tab[29-i*4] = (tab[29-i*4] + xr);
306 xr = MUL(t1[0], *xp);
315 out[i] = tab[bitinv32[i]];
319 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
321 static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
324 int sum, offset, i, j;
329 offset = s->samples_offset[ch];
330 out = &s->sb_samples[ch][0][0][0];
332 /* 32 samples at once */
334 s->samples_buf[ch][offset + (31 - i)] = samples[0];
339 p = s->samples_buf[ch] + offset;
343 sum = p[0*64] * q[0*64];
344 sum += p[1*64] * q[1*64];
345 sum += p[2*64] * q[2*64];
346 sum += p[3*64] * q[3*64];
347 sum += p[4*64] * q[4*64];
348 sum += p[5*64] * q[5*64];
349 sum += p[6*64] * q[6*64];
350 sum += p[7*64] * q[7*64];
355 tmp1[0] = tmp[16] >> WSHIFT;
356 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
357 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
361 /* advance of 32 samples */
364 /* handle the wrap around */
366 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
367 s->samples_buf[ch], (512 - 32) * 2);
368 offset = SAMPLES_BUF_SIZE - 512;
371 s->samples_offset[ch] = offset;
374 static void compute_scale_factors(MpegAudioContext *s,
375 unsigned char scale_code[SBLIMIT],
376 unsigned char scale_factors[SBLIMIT][3],
377 int sb_samples[3][12][SBLIMIT],
380 int *p, vmax, v, n, i, j, k, code;
382 unsigned char *sf = &scale_factors[0][0];
384 for(j=0;j<sblimit;j++) {
386 /* find the max absolute value */
387 p = &sb_samples[i][0][j];
395 /* compute the scale factor index using log 2 computations */
398 /* n is the position of the MSB of vmax. now
399 use at most 2 compares to find the index */
400 index = (21 - n) * 3 - 3;
402 while (vmax <= s->scale_factor_table[index+1])
405 index = 0; /* very unlikely case of overflow */
408 index = 62; /* value 63 is not allowed */
411 av_dlog(NULL, "%2d:%d in=%x %x %d\n",
412 j, i, vmax, s->scale_factor_table[index], index);
413 /* store the scale factor */
414 av_assert2(index >=0 && index <= 63);
418 /* compute the transmission factor : look if the scale factors
419 are close enough to each other */
420 d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
421 d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
423 /* handle the 25 cases */
424 switch(d1 * 5 + d2) {
456 sf[1] = sf[2] = sf[0];
461 sf[0] = sf[1] = sf[2];
467 sf[0] = sf[2] = sf[1];
473 sf[1] = sf[2] = sf[0];
476 av_assert2(0); //cannot happen
477 code = 0; /* kill warning */
480 av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
481 sf[0], sf[1], sf[2], d1, d2, code);
482 scale_code[j] = code;
487 /* The most important function : psycho acoustic module. In this
488 encoder there is basically none, so this is the worst you can do,
489 but also this is the simpler. */
490 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
494 for(i=0;i<s->sblimit;i++) {
495 smr[i] = (int)(fixed_smr[i] * 10);
500 #define SB_NOTALLOCATED 0
501 #define SB_ALLOCATED 1
504 /* Try to maximize the smr while using a number of bits inferior to
505 the frame size. I tried to make the code simpler, faster and
506 smaller than other encoders :-) */
507 static void compute_bit_allocation(MpegAudioContext *s,
508 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
509 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
512 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
514 short smr[MPA_MAX_CHANNELS][SBLIMIT];
515 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
516 const unsigned char *alloc;
518 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
519 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
520 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
522 /* compute frame size and padding */
523 max_frame_size = s->frame_size;
524 s->frame_frac += s->frame_frac_incr;
525 if (s->frame_frac >= 65536) {
526 s->frame_frac -= 65536;
533 /* compute the header + bit alloc size */
534 current_frame_size = 32;
535 alloc = s->alloc_table;
536 for(i=0;i<s->sblimit;i++) {
538 current_frame_size += incr * s->nb_channels;
542 /* look for the subband with the largest signal to mask ratio */
546 for(ch=0;ch<s->nb_channels;ch++) {
547 for(i=0;i<s->sblimit;i++) {
548 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
549 max_smr = smr[ch][i];
557 av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
558 current_frame_size, max_frame_size, max_sb, max_ch,
559 bit_alloc[max_ch][max_sb]);
561 /* find alloc table entry (XXX: not optimal, should use
563 alloc = s->alloc_table;
564 for(i=0;i<max_sb;i++) {
565 alloc += 1 << alloc[0];
568 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
569 /* nothing was coded for this band: add the necessary bits */
570 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
571 incr += s->total_quant_bits[alloc[1]];
573 /* increments bit allocation */
574 b = bit_alloc[max_ch][max_sb];
575 incr = s->total_quant_bits[alloc[b + 1]] -
576 s->total_quant_bits[alloc[b]];
579 if (current_frame_size + incr <= max_frame_size) {
580 /* can increase size */
581 b = ++bit_alloc[max_ch][max_sb];
582 current_frame_size += incr;
583 /* decrease smr by the resolution we added */
584 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
585 /* max allocation size reached ? */
586 if (b == ((1 << alloc[0]) - 1))
587 subband_status[max_ch][max_sb] = SB_NOMORE;
589 subband_status[max_ch][max_sb] = SB_ALLOCATED;
591 /* cannot increase the size of this subband */
592 subband_status[max_ch][max_sb] = SB_NOMORE;
595 *padding = max_frame_size - current_frame_size;
596 av_assert0(*padding >= 0);
600 * Output the mpeg audio layer 2 frame. Note how the code is small
601 * compared to other encoders :-)
603 static void encode_frame(MpegAudioContext *s,
604 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
607 int i, j, k, l, bit_alloc_bits, b, ch;
610 PutBitContext *p = &s->pb;
614 put_bits(p, 12, 0xfff);
615 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
616 put_bits(p, 2, 4-2); /* layer 2 */
617 put_bits(p, 1, 1); /* no error protection */
618 put_bits(p, 4, s->bitrate_index);
619 put_bits(p, 2, s->freq_index);
620 put_bits(p, 1, s->do_padding); /* use padding */
621 put_bits(p, 1, 0); /* private_bit */
622 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
623 put_bits(p, 2, 0); /* mode_ext */
624 put_bits(p, 1, 0); /* no copyright */
625 put_bits(p, 1, 1); /* original */
626 put_bits(p, 2, 0); /* no emphasis */
630 for(i=0;i<s->sblimit;i++) {
631 bit_alloc_bits = s->alloc_table[j];
632 for(ch=0;ch<s->nb_channels;ch++) {
633 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
635 j += 1 << bit_alloc_bits;
639 for(i=0;i<s->sblimit;i++) {
640 for(ch=0;ch<s->nb_channels;ch++) {
641 if (bit_alloc[ch][i])
642 put_bits(p, 2, s->scale_code[ch][i]);
647 for(i=0;i<s->sblimit;i++) {
648 for(ch=0;ch<s->nb_channels;ch++) {
649 if (bit_alloc[ch][i]) {
650 sf = &s->scale_factors[ch][i][0];
651 switch(s->scale_code[ch][i]) {
653 put_bits(p, 6, sf[0]);
654 put_bits(p, 6, sf[1]);
655 put_bits(p, 6, sf[2]);
659 put_bits(p, 6, sf[0]);
660 put_bits(p, 6, sf[2]);
663 put_bits(p, 6, sf[0]);
670 /* quantization & write sub band samples */
675 for(i=0;i<s->sblimit;i++) {
676 bit_alloc_bits = s->alloc_table[j];
677 for(ch=0;ch<s->nb_channels;ch++) {
678 b = bit_alloc[ch][i];
680 int qindex, steps, m, sample, bits;
681 /* we encode 3 sub band samples of the same sub band at a time */
682 qindex = s->alloc_table[j+b];
683 steps = ff_mpa_quant_steps[qindex];
685 sample = s->sb_samples[ch][k][l + m][i];
686 /* divide by scale factor */
690 a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]];
691 q[m] = (int)((a + 1.0) * steps * 0.5);
695 int q1, e, shift, mult;
696 e = s->scale_factors[ch][i][k];
697 shift = s->scale_factor_shift[e];
698 mult = s->scale_factor_mult[e];
700 /* normalize to P bits */
702 q1 = sample << (-shift);
704 q1 = sample >> shift;
705 q1 = (q1 * mult) >> P;
706 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
713 av_assert2(q[m] >= 0 && q[m] < steps);
715 bits = ff_mpa_quant_bits[qindex];
717 /* group the 3 values to save bits */
719 q[0] + steps * (q[1] + steps * q[2]));
721 put_bits(p, bits, q[0]);
722 put_bits(p, bits, q[1]);
723 put_bits(p, bits, q[2]);
727 /* next subband in alloc table */
728 j += 1 << bit_alloc_bits;
734 for(i=0;i<padding;i++)
741 static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
742 const AVFrame *frame, int *got_packet_ptr)
744 MpegAudioContext *s = avctx->priv_data;
745 const int16_t *samples = (const int16_t *)frame->data[0];
746 short smr[MPA_MAX_CHANNELS][SBLIMIT];
747 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
750 for(i=0;i<s->nb_channels;i++) {
751 filter(s, i, samples + i, s->nb_channels);
754 for(i=0;i<s->nb_channels;i++) {
755 compute_scale_factors(s, s->scale_code[i], s->scale_factors[i],
756 s->sb_samples[i], s->sblimit);
758 for(i=0;i<s->nb_channels;i++) {
759 psycho_acoustic_model(s, smr[i]);
761 compute_bit_allocation(s, smr, bit_alloc, &padding);
763 if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
766 init_put_bits(&s->pb, avpkt->data, avpkt->size);
768 encode_frame(s, bit_alloc, padding);
770 if (frame->pts != AV_NOPTS_VALUE)
771 avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
773 avpkt->size = put_bits_count(&s->pb) / 8;
778 static const AVCodecDefault mp2_defaults[] = {
783 AVCodec ff_mp2_encoder = {
785 .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
786 .type = AVMEDIA_TYPE_AUDIO,
787 .id = AV_CODEC_ID_MP2,
788 .priv_data_size = sizeof(MpegAudioContext),
789 .init = MPA_encode_init,
790 .encode2 = MPA_encode_frame,
791 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
792 AV_SAMPLE_FMT_NONE },
793 .supported_samplerates = (const int[]){
794 44100, 48000, 32000, 22050, 24000, 16000, 0
796 .channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
799 .defaults = mp2_defaults,