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* The simplest mpeg audio layer 2 encoder
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* Copyright (c) 2000, 2001 Fabrice Bellard.
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* This library 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 of the License, or (at your option) any later version.
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* This library 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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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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* The simplest mpeg audio layer 2 encoder.
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#include "mpegaudio.h"
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/* currently, cannot change these constants (need to modify
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quantization stage) */
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#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
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#define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
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#define SAMPLES_BUF_SIZE 4096
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typedef struct MpegAudioContext {
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int lsf; /* 1 if mpeg2 low bitrate selected */
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int bitrate_index; /* bit rate */
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int frame_size; /* frame size, in bits, without padding */
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int64_t nb_samples; /* total number of samples encoded */
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/* padding computation */
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int frame_frac, frame_frac_incr, do_padding;
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short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
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int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
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int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
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unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
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/* code to group 3 scale factors */
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unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
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int sblimit; /* number of used subbands */
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const unsigned char *alloc_table;
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/* define it to use floats in quantization (I don't like floats !) */
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#include "mpegaudiotab.h"
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static int MPA_encode_init(AVCodecContext *avctx)
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MpegAudioContext *s = avctx->priv_data;
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int freq = avctx->sample_rate;
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int bitrate = avctx->bit_rate;
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int channels = avctx->channels;
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bitrate = bitrate / 1000;
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s->nb_channels = channels;
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s->bit_rate = bitrate * 1000;
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avctx->frame_size = MPA_FRAME_SIZE;
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if (mpa_freq_tab[i] == freq)
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if ((mpa_freq_tab[i] / 2) == freq) {
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/* encoding bitrate & frequency */
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if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
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s->bitrate_index = i;
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/* compute total header size & pad bit */
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a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
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s->frame_size = ((int)a) * 8;
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/* frame fractional size to compute padding */
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s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
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/* select the right allocation table */
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table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
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/* number of used subbands */
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s->sblimit = sblimit_table[table];
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s->alloc_table = alloc_tables[table];
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printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
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bitrate, freq, s->frame_size, table, s->frame_frac_incr);
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for(i=0;i<s->nb_channels;i++)
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s->samples_offset[i] = 0;
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v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
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filter_bank[512 - i] = v;
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v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
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scale_factor_table[i] = v;
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scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
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scale_factor_shift[i] = 21 - P - (i / 3);
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scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
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scale_diff_table[i] = v;
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total_quant_bits[i] = 12 * v;
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avctx->coded_frame= avcodec_alloc_frame();
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avctx->coded_frame->key_frame= 1;
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/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
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static void idct32(int *out, int *tab)
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const int *xp = costab32;
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for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
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x3 = MUL(t[16], FIX(SQRT2*0.5));
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x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
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x1 = MUL((t[8] - x2), xp[0]);
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x2 = MUL((t[8] + x2), xp[1]);
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xr = MUL(t[28],xp[0]);
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xr = MUL(t[4],xp[1]);
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t[ 4] = (t[24] - xr);
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t[24] = (t[24] + xr);
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xr = MUL(t[20],xp[2]);
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xr = MUL(t[12],xp[3]);
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t[12] = (t[16] - xr);
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t[16] = (t[16] + xr);
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for (i = 0; i < 4; i++) {
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xr = MUL(tab[30-i*4],xp[0]);
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tab[30-i*4] = (tab[i*4] - xr);
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tab[ i*4] = (tab[i*4] + xr);
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xr = MUL(tab[ 2+i*4],xp[1]);
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tab[ 2+i*4] = (tab[28-i*4] - xr);
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tab[28-i*4] = (tab[28-i*4] + xr);
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xr = MUL(tab[31-i*4],xp[0]);
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tab[31-i*4] = (tab[1+i*4] - xr);
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tab[ 1+i*4] = (tab[1+i*4] + xr);
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xr = MUL(tab[ 3+i*4],xp[1]);
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tab[ 3+i*4] = (tab[29-i*4] - xr);
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tab[29-i*4] = (tab[29-i*4] + xr);
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xr = MUL(t1[0], *xp);
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out[i] = tab[bitinv32[i]];
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#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
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static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
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int sum, offset, i, j;
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// print_pow1(samples, 1152);
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offset = s->samples_offset[ch];
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out = &s->sb_samples[ch][0][0][0];
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/* 32 samples at once */
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s->samples_buf[ch][offset + (31 - i)] = samples[0];
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p = s->samples_buf[ch] + offset;
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sum = p[0*64] * q[0*64];
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sum += p[1*64] * q[1*64];
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sum += p[2*64] * q[2*64];
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sum += p[3*64] * q[3*64];
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sum += p[4*64] * q[4*64];
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sum += p[5*64] * q[5*64];
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sum += p[6*64] * q[6*64];
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sum += p[7*64] * q[7*64];
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tmp1[0] = tmp[16] >> WSHIFT;
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for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
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for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
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/* advance of 32 samples */
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/* handle the wrap around */
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memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
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s->samples_buf[ch], (512 - 32) * 2);
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offset = SAMPLES_BUF_SIZE - 512;
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s->samples_offset[ch] = offset;
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// print_pow(s->sb_samples, 1152);
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static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
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unsigned char scale_factors[SBLIMIT][3],
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int sb_samples[3][12][SBLIMIT],
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int *p, vmax, v, n, i, j, k, code;
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unsigned char *sf = &scale_factors[0][0];
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for(j=0;j<sblimit;j++) {
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/* find the max absolute value */
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p = &sb_samples[i][0][j];
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/* compute the scale factor index using log 2 computations */
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/* n is the position of the MSB of vmax. now
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use at most 2 compares to find the index */
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index = (21 - n) * 3 - 3;
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while (vmax <= scale_factor_table[index+1])
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index = 0; /* very unlikely case of overflow */
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index = 62; /* value 63 is not allowed */
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printf("%2d:%d in=%x %x %d\n",
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j, i, vmax, scale_factor_table[index], index);
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/* store the scale factor */
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assert(index >=0 && index <= 63);
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/* compute the transmission factor : look if the scale factors
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are close enough to each other */
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d1 = scale_diff_table[sf[0] - sf[1] + 64];
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d2 = scale_diff_table[sf[1] - sf[2] + 64];
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/* handle the 25 cases */
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switch(d1 * 5 + d2) {
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sf[1] = sf[2] = sf[0];
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sf[0] = sf[1] = sf[2];
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sf[0] = sf[2] = sf[1];
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sf[1] = sf[2] = sf[0];
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printf("%d: %2d %2d %2d %d %d -> %d\n", j,
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sf[0], sf[1], sf[2], d1, d2, code);
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scale_code[j] = code;
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/* The most important function : psycho acoustic module. In this
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encoder there is basically none, so this is the worst you can do,
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but also this is the simpler. */
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static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
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for(i=0;i<s->sblimit;i++) {
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smr[i] = (int)(fixed_smr[i] * 10);
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#define SB_NOTALLOCATED 0
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#define SB_ALLOCATED 1
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/* Try to maximize the smr while using a number of bits inferior to
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the frame size. I tried to make the code simpler, faster and
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smaller than other encoders :-) */
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static void compute_bit_allocation(MpegAudioContext *s,
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short smr1[MPA_MAX_CHANNELS][SBLIMIT],
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unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
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int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
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short smr[MPA_MAX_CHANNELS][SBLIMIT];
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unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
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const unsigned char *alloc;
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memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
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memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
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memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
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/* compute frame size and padding */
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max_frame_size = s->frame_size;
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s->frame_frac += s->frame_frac_incr;
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if (s->frame_frac >= 65536) {
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s->frame_frac -= 65536;
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/* compute the header + bit alloc size */
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current_frame_size = 32;
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alloc = s->alloc_table;
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for(i=0;i<s->sblimit;i++) {
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current_frame_size += incr * s->nb_channels;
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/* look for the subband with the largest signal to mask ratio */
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max_smr = 0x80000000;
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for(ch=0;ch<s->nb_channels;ch++) {
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for(i=0;i<s->sblimit;i++) {
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if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
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max_smr = smr[ch][i];
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printf("current=%d max=%d max_sb=%d alloc=%d\n",
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current_frame_size, max_frame_size, max_sb,
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/* find alloc table entry (XXX: not optimal, should use
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alloc = s->alloc_table;
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for(i=0;i<max_sb;i++) {
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alloc += 1 << alloc[0];
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if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
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/* nothing was coded for this band: add the necessary bits */
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incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
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incr += total_quant_bits[alloc[1]];
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/* increments bit allocation */
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b = bit_alloc[max_ch][max_sb];
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incr = total_quant_bits[alloc[b + 1]] -
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total_quant_bits[alloc[b]];
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if (current_frame_size + incr <= max_frame_size) {
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/* can increase size */
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b = ++bit_alloc[max_ch][max_sb];
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current_frame_size += incr;
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/* decrease smr by the resolution we added */
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smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
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/* max allocation size reached ? */
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if (b == ((1 << alloc[0]) - 1))
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subband_status[max_ch][max_sb] = SB_NOMORE;
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subband_status[max_ch][max_sb] = SB_ALLOCATED;
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/* cannot increase the size of this subband */
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subband_status[max_ch][max_sb] = SB_NOMORE;
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*padding = max_frame_size - current_frame_size;
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assert(*padding >= 0);
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for(i=0;i<s->sblimit;i++) {
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printf("%d ", bit_alloc[i]);
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* Output the mpeg audio layer 2 frame. Note how the code is small
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* compared to other encoders :-)
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static void encode_frame(MpegAudioContext *s,
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unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
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int i, j, k, l, bit_alloc_bits, b, ch;
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PutBitContext *p = &s->pb;
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put_bits(p, 12, 0xfff);
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put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
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put_bits(p, 2, 4-2); /* layer 2 */
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put_bits(p, 1, 1); /* no error protection */
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put_bits(p, 4, s->bitrate_index);
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put_bits(p, 2, s->freq_index);
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put_bits(p, 1, s->do_padding); /* use padding */
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put_bits(p, 1, 0); /* private_bit */
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put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
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put_bits(p, 2, 0); /* mode_ext */
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put_bits(p, 1, 0); /* no copyright */
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put_bits(p, 1, 1); /* original */
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put_bits(p, 2, 0); /* no emphasis */
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for(i=0;i<s->sblimit;i++) {
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bit_alloc_bits = s->alloc_table[j];
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for(ch=0;ch<s->nb_channels;ch++) {
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put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
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j += 1 << bit_alloc_bits;
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for(i=0;i<s->sblimit;i++) {
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for(ch=0;ch<s->nb_channels;ch++) {
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if (bit_alloc[ch][i])
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put_bits(p, 2, s->scale_code[ch][i]);
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for(i=0;i<s->sblimit;i++) {
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for(ch=0;ch<s->nb_channels;ch++) {
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if (bit_alloc[ch][i]) {
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sf = &s->scale_factors[ch][i][0];
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switch(s->scale_code[ch][i]) {
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put_bits(p, 6, sf[0]);
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put_bits(p, 6, sf[1]);
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put_bits(p, 6, sf[2]);
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put_bits(p, 6, sf[0]);
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put_bits(p, 6, sf[2]);
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put_bits(p, 6, sf[0]);
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/* quantization & write sub band samples */
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for(i=0;i<s->sblimit;i++) {
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bit_alloc_bits = s->alloc_table[j];
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for(ch=0;ch<s->nb_channels;ch++) {
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b = bit_alloc[ch][i];
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int qindex, steps, m, sample, bits;
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/* we encode 3 sub band samples of the same sub band at a time */
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qindex = s->alloc_table[j+b];
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steps = quant_steps[qindex];
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sample = s->sb_samples[ch][k][l + m][i];
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/* divide by scale factor */
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a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
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q[m] = (int)((a + 1.0) * steps * 0.5);
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int q1, e, shift, mult;
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e = s->scale_factors[ch][i][k];
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shift = scale_factor_shift[e];
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mult = scale_factor_mult[e];
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/* normalize to P bits */
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q1 = sample << (-shift);
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q1 = sample >> shift;
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q1 = (q1 * mult) >> P;
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q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
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assert(q[m] >= 0 && q[m] < steps);
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bits = quant_bits[qindex];
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/* group the 3 values to save bits */
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q[0] + steps * (q[1] + steps * q[2]));
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printf("%d: gr1 %d\n",
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i, q[0] + steps * (q[1] + steps * q[2]));
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printf("%d: gr3 %d %d %d\n",
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i, q[0], q[1], q[2]);
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put_bits(p, bits, q[0]);
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put_bits(p, bits, q[1]);
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put_bits(p, bits, q[2]);
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/* next subband in alloc table */
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j += 1 << bit_alloc_bits;
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for(i=0;i<padding;i++)
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static int MPA_encode_frame(AVCodecContext *avctx,
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unsigned char *frame, int buf_size, void *data)
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MpegAudioContext *s = avctx->priv_data;
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short *samples = data;
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short smr[MPA_MAX_CHANNELS][SBLIMIT];
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unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
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for(i=0;i<s->nb_channels;i++) {
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filter(s, i, samples + i, s->nb_channels);
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for(i=0;i<s->nb_channels;i++) {
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compute_scale_factors(s->scale_code[i], s->scale_factors[i],
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s->sb_samples[i], s->sblimit);
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for(i=0;i<s->nb_channels;i++) {
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psycho_acoustic_model(s, smr[i]);
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compute_bit_allocation(s, smr, bit_alloc, &padding);
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init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
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encode_frame(s, bit_alloc, padding);
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s->nb_samples += MPA_FRAME_SIZE;
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return pbBufPtr(&s->pb) - s->pb.buf;
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static int MPA_encode_close(AVCodecContext *avctx)
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av_freep(&avctx->coded_frame);
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AVCodec mp2_encoder = {
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sizeof(MpegAudioContext),