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authorFabrice Bellard <fabrice@bellard.org>2001-07-22 14:18:56 +0000
committerFabrice Bellard <fabrice@bellard.org>2001-07-22 14:18:56 +0000
commitde6d9b6404bfd1c589799142da5a95428f146edd (patch)
tree75ae0cbb74bdfafb6f1a40922db111a103db3bcf /libavcodec/mpegaudio.c
parent1b58d58ddaf8a8c766a0353885ff504babed0453 (diff)
downloadffmpeg-de6d9b6404bfd1c589799142da5a95428f146edd.tar.gz
Initial revision
Originally committed as revision 5 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/mpegaudio.c')
-rw-r--r--libavcodec/mpegaudio.c774
1 files changed, 774 insertions, 0 deletions
diff --git a/libavcodec/mpegaudio.c b/libavcodec/mpegaudio.c
new file mode 100644
index 0000000000..af05e29279
--- /dev/null
+++ b/libavcodec/mpegaudio.c
@@ -0,0 +1,774 @@
+/*
+ * The simplest mpeg audio layer 2 encoder
+ * Copyright (c) 2000 Gerard Lantau.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "avcodec.h"
+#include "mpegaudio.h"
+
+#define NDEBUG
+#include <assert.h>
+
+/* define it to use floats in quantization (I don't like floats !) */
+//#define USE_FLOATS
+
+#define MPA_STEREO 0
+#define MPA_JSTEREO 1
+#define MPA_DUAL 2
+#define MPA_MONO 3
+
+#include "mpegaudiotab.h"
+
+int MPA_encode_init(AVCodecContext *avctx)
+{
+ MpegAudioContext *s = avctx->priv_data;
+ int freq = avctx->sample_rate;
+ int bitrate = avctx->bit_rate;
+ int channels = avctx->channels;
+ int i, v, table, ch_bitrate;
+ float a;
+
+ if (channels > 2)
+ return -1;
+ bitrate = bitrate / 1000;
+ s->nb_channels = channels;
+ s->freq = freq;
+ s->bit_rate = bitrate * 1000;
+ avctx->frame_size = MPA_FRAME_SIZE;
+ avctx->key_frame = 1; /* always key frame */
+
+ /* encoding freq */
+ s->lsf = 0;
+ for(i=0;i<3;i++) {
+ if (freq_tab[i] == freq)
+ break;
+ if ((freq_tab[i] / 2) == freq) {
+ s->lsf = 1;
+ break;
+ }
+ }
+ if (i == 3)
+ return -1;
+ s->freq_index = i;
+
+ /* encoding bitrate & frequency */
+ for(i=0;i<15;i++) {
+ if (bitrate_tab[1-s->lsf][i] == bitrate)
+ break;
+ }
+ if (i == 15)
+ return -1;
+ s->bitrate_index = i;
+
+ /* compute total header size & pad bit */
+
+ a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
+ s->frame_size = ((int)a) * 8;
+
+ /* frame fractional size to compute padding */
+ s->frame_frac = 0;
+ s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
+
+ /* select the right allocation table */
+ ch_bitrate = bitrate / s->nb_channels;
+ if (!s->lsf) {
+ if ((freq == 48000 && ch_bitrate >= 56) ||
+ (ch_bitrate >= 56 && ch_bitrate <= 80))
+ table = 0;
+ else if (freq != 48000 && ch_bitrate >= 96)
+ table = 1;
+ else if (freq != 32000 && ch_bitrate <= 48)
+ table = 2;
+ else
+ table = 3;
+ } else {
+ table = 4;
+ }
+ /* number of used subbands */
+ s->sblimit = sblimit_table[table];
+ s->alloc_table = alloc_tables[table];
+
+#ifdef DEBUG
+ printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
+ bitrate, freq, s->frame_size, table, s->frame_frac_incr);
+#endif
+
+ for(i=0;i<s->nb_channels;i++)
+ s->samples_offset[i] = 0;
+
+ for(i=0;i<512;i++) {
+ float a = enwindow[i] * 32768.0 * 16.0;
+ filter_bank[i] = (int)(a);
+ }
+ for(i=0;i<64;i++) {
+ v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
+ if (v <= 0)
+ v = 1;
+ scale_factor_table[i] = v;
+#ifdef USE_FLOATS
+ scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
+#else
+#define P 15
+ scale_factor_shift[i] = 21 - P - (i / 3);
+ scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
+#endif
+ }
+ for(i=0;i<128;i++) {
+ v = i - 64;
+ if (v <= -3)
+ v = 0;
+ else if (v < 0)
+ v = 1;
+ else if (v == 0)
+ v = 2;
+ else if (v < 3)
+ v = 3;
+ else
+ v = 4;
+ scale_diff_table[i] = v;
+ }
+
+ for(i=0;i<17;i++) {
+ v = quant_bits[i];
+ if (v < 0)
+ v = -v;
+ else
+ v = v * 3;
+ total_quant_bits[i] = 12 * v;
+ }
+
+ return 0;
+}
+
+/* 32 point floating point IDCT */
+static void idct32(int *out, int *tab, int sblimit, int left_shift)
+{
+ int i, j;
+ int *t, *t1, xr;
+ const int *xp = costab32;
+
+ for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
+
+ t = tab + 30;
+ t1 = tab + 2;
+ do {
+ t[0] += t[-4];
+ t[1] += t[1 - 4];
+ t -= 4;
+ } while (t != t1);
+
+ t = tab + 28;
+ t1 = tab + 4;
+ do {
+ t[0] += t[-8];
+ t[1] += t[1-8];
+ t[2] += t[2-8];
+ t[3] += t[3-8];
+ t -= 8;
+ } while (t != t1);
+
+ t = tab;
+ t1 = tab + 32;
+ do {
+ t[ 3] = -t[ 3];
+ t[ 6] = -t[ 6];
+
+ t[11] = -t[11];
+ t[12] = -t[12];
+ t[13] = -t[13];
+ t[15] = -t[15];
+ t += 16;
+ } while (t != t1);
+
+
+ t = tab;
+ t1 = tab + 8;
+ do {
+ int x1, x2, x3, x4;
+
+ x3 = MUL(t[16], FIX(SQRT2*0.5));
+ x4 = t[0] - x3;
+ x3 = t[0] + x3;
+
+ x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
+ x1 = MUL((t[8] - x2), xp[0]);
+ x2 = MUL((t[8] + x2), xp[1]);
+
+ t[ 0] = x3 + x1;
+ t[ 8] = x4 - x2;
+ t[16] = x4 + x2;
+ t[24] = x3 - x1;
+ t++;
+ } while (t != t1);
+
+ xp += 2;
+ t = tab;
+ t1 = tab + 4;
+ do {
+ xr = MUL(t[28],xp[0]);
+ t[28] = (t[0] - xr);
+ t[0] = (t[0] + xr);
+
+ xr = MUL(t[4],xp[1]);
+ t[ 4] = (t[24] - xr);
+ t[24] = (t[24] + xr);
+
+ xr = MUL(t[20],xp[2]);
+ t[20] = (t[8] - xr);
+ t[ 8] = (t[8] + xr);
+
+ xr = MUL(t[12],xp[3]);
+ t[12] = (t[16] - xr);
+ t[16] = (t[16] + xr);
+ t++;
+ } while (t != t1);
+ xp += 4;
+
+ for (i = 0; i < 4; i++) {
+ xr = MUL(tab[30-i*4],xp[0]);
+ tab[30-i*4] = (tab[i*4] - xr);
+ tab[ i*4] = (tab[i*4] + xr);
+
+ xr = MUL(tab[ 2+i*4],xp[1]);
+ tab[ 2+i*4] = (tab[28-i*4] - xr);
+ tab[28-i*4] = (tab[28-i*4] + xr);
+
+ xr = MUL(tab[31-i*4],xp[0]);
+ tab[31-i*4] = (tab[1+i*4] - xr);
+ tab[ 1+i*4] = (tab[1+i*4] + xr);
+
+ xr = MUL(tab[ 3+i*4],xp[1]);
+ tab[ 3+i*4] = (tab[29-i*4] - xr);
+ tab[29-i*4] = (tab[29-i*4] + xr);
+
+ xp += 2;
+ }
+
+ t = tab + 30;
+ t1 = tab + 1;
+ do {
+ xr = MUL(t1[0], *xp);
+ t1[0] = (t[0] - xr);
+ t[0] = (t[0] + xr);
+ t -= 2;
+ t1 += 2;
+ xp++;
+ } while (t >= tab);
+
+ for(i=0;i<32;i++) {
+ out[i] = tab[bitinv32[i]] << left_shift;
+ }
+}
+
+static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
+{
+ short *p, *q;
+ int sum, offset, i, j, norm, n;
+ short tmp[64];
+ int tmp1[32];
+ int *out;
+
+ // print_pow1(samples, 1152);
+
+ offset = s->samples_offset[ch];
+ out = &s->sb_samples[ch][0][0][0];
+ for(j=0;j<36;j++) {
+ /* 32 samples at once */
+ for(i=0;i<32;i++) {
+ s->samples_buf[ch][offset + (31 - i)] = samples[0];
+ samples += incr;
+ }
+
+ /* filter */
+ p = s->samples_buf[ch] + offset;
+ q = filter_bank;
+ /* maxsum = 23169 */
+ for(i=0;i<64;i++) {
+ sum = p[0*64] * q[0*64];
+ sum += p[1*64] * q[1*64];
+ sum += p[2*64] * q[2*64];
+ sum += p[3*64] * q[3*64];
+ sum += p[4*64] * q[4*64];
+ sum += p[5*64] * q[5*64];
+ sum += p[6*64] * q[6*64];
+ sum += p[7*64] * q[7*64];
+ tmp[i] = sum >> 14;
+ p++;
+ q++;
+ }
+ tmp1[0] = tmp[16];
+ for( i=1; i<=16; i++ ) tmp1[i] = tmp[i+16]+tmp[16-i];
+ for( i=17; i<=31; i++ ) tmp1[i] = tmp[i+16]-tmp[80-i];
+
+ /* integer IDCT 32 with normalization. XXX: There may be some
+ overflow left */
+ norm = 0;
+ for(i=0;i<32;i++) {
+ norm |= abs(tmp1[i]);
+ }
+ n = log2(norm) - 12;
+ if (n > 0) {
+ for(i=0;i<32;i++)
+ tmp1[i] >>= n;
+ } else {
+ n = 0;
+ }
+
+ idct32(out, tmp1, s->sblimit, n);
+
+ /* advance of 32 samples */
+ offset -= 32;
+ out += 32;
+ /* handle the wrap around */
+ if (offset < 0) {
+ memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
+ s->samples_buf[ch], (512 - 32) * 2);
+ offset = SAMPLES_BUF_SIZE - 512;
+ }
+ }
+ s->samples_offset[ch] = offset;
+
+ // print_pow(s->sb_samples, 1152);
+}
+
+static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
+ unsigned char scale_factors[SBLIMIT][3],
+ int sb_samples[3][12][SBLIMIT],
+ int sblimit)
+{
+ int *p, vmax, v, n, i, j, k, code;
+ int index, d1, d2;
+ unsigned char *sf = &scale_factors[0][0];
+
+ for(j=0;j<sblimit;j++) {
+ for(i=0;i<3;i++) {
+ /* find the max absolute value */
+ p = &sb_samples[i][0][j];
+ vmax = abs(*p);
+ for(k=1;k<12;k++) {
+ p += SBLIMIT;
+ v = abs(*p);
+ if (v > vmax)
+ vmax = v;
+ }
+ /* compute the scale factor index using log 2 computations */
+ if (vmax > 0) {
+ n = log2(vmax);
+ /* n is the position of the MSB of vmax. now
+ use at most 2 compares to find the index */
+ index = (21 - n) * 3 - 3;
+ if (index >= 0) {
+ while (vmax <= scale_factor_table[index+1])
+ index++;
+ } else {
+ index = 0; /* very unlikely case of overflow */
+ }
+ } else {
+ index = 63;
+ }
+
+#if 0
+ printf("%2d:%d in=%x %x %d\n",
+ j, i, vmax, scale_factor_table[index], index);
+#endif
+ /* store the scale factor */
+ assert(index >=0 && index <= 63);
+ sf[i] = index;
+ }
+
+ /* compute the transmission factor : look if the scale factors
+ are close enough to each other */
+ d1 = scale_diff_table[sf[0] - sf[1] + 64];
+ d2 = scale_diff_table[sf[1] - sf[2] + 64];
+
+ /* handle the 25 cases */
+ switch(d1 * 5 + d2) {
+ case 0*5+0:
+ case 0*5+4:
+ case 3*5+4:
+ case 4*5+0:
+ case 4*5+4:
+ code = 0;
+ break;
+ case 0*5+1:
+ case 0*5+2:
+ case 4*5+1:
+ case 4*5+2:
+ code = 3;
+ sf[2] = sf[1];
+ break;
+ case 0*5+3:
+ case 4*5+3:
+ code = 3;
+ sf[1] = sf[2];
+ break;
+ case 1*5+0:
+ case 1*5+4:
+ case 2*5+4:
+ code = 1;
+ sf[1] = sf[0];
+ break;
+ case 1*5+1:
+ case 1*5+2:
+ case 2*5+0:
+ case 2*5+1:
+ case 2*5+2:
+ code = 2;
+ sf[1] = sf[2] = sf[0];
+ break;
+ case 2*5+3:
+ case 3*5+3:
+ code = 2;
+ sf[0] = sf[1] = sf[2];
+ break;
+ case 3*5+0:
+ case 3*5+1:
+ case 3*5+2:
+ code = 2;
+ sf[0] = sf[2] = sf[1];
+ break;
+ case 1*5+3:
+ code = 2;
+ if (sf[0] > sf[2])
+ sf[0] = sf[2];
+ sf[1] = sf[2] = sf[0];
+ break;
+ default:
+ abort();
+ }
+
+#if 0
+ printf("%d: %2d %2d %2d %d %d -> %d\n", j,
+ sf[0], sf[1], sf[2], d1, d2, code);
+#endif
+ scale_code[j] = code;
+ sf += 3;
+ }
+}
+
+/* The most important function : psycho acoustic module. In this
+ encoder there is basically none, so this is the worst you can do,
+ but also this is the simpler. */
+static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
+{
+ int i;
+
+ for(i=0;i<s->sblimit;i++) {
+ smr[i] = (int)(fixed_smr[i] * 10);
+ }
+}
+
+
+#define SB_NOTALLOCATED 0
+#define SB_ALLOCATED 1
+#define SB_NOMORE 2
+
+/* Try to maximize the smr while using a number of bits inferior to
+ the frame size. I tried to make the code simpler, faster and
+ smaller than other encoders :-) */
+static void compute_bit_allocation(MpegAudioContext *s,
+ short smr1[MPA_MAX_CHANNELS][SBLIMIT],
+ unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
+ int *padding)
+{
+ int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
+ int incr;
+ short smr[MPA_MAX_CHANNELS][SBLIMIT];
+ unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
+ const unsigned char *alloc;
+
+ memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
+ memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
+ memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
+
+ /* compute frame size and padding */
+ max_frame_size = s->frame_size;
+ s->frame_frac += s->frame_frac_incr;
+ if (s->frame_frac >= 65536) {
+ s->frame_frac -= 65536;
+ s->do_padding = 1;
+ max_frame_size += 8;
+ } else {
+ s->do_padding = 0;
+ }
+
+ /* compute the header + bit alloc size */
+ current_frame_size = 32;
+ alloc = s->alloc_table;
+ for(i=0;i<s->sblimit;i++) {
+ incr = alloc[0];
+ current_frame_size += incr * s->nb_channels;
+ alloc += 1 << incr;
+ }
+ for(;;) {
+ /* look for the subband with the largest signal to mask ratio */
+ max_sb = -1;
+ max_ch = -1;
+ max_smr = 0x80000000;
+ for(ch=0;ch<s->nb_channels;ch++) {
+ for(i=0;i<s->sblimit;i++) {
+ if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
+ max_smr = smr[ch][i];
+ max_sb = i;
+ max_ch = ch;
+ }
+ }
+ }
+#if 0
+ printf("current=%d max=%d max_sb=%d alloc=%d\n",
+ current_frame_size, max_frame_size, max_sb,
+ bit_alloc[max_sb]);
+#endif
+ if (max_sb < 0)
+ break;
+
+ /* find alloc table entry (XXX: not optimal, should use
+ pointer table) */
+ alloc = s->alloc_table;
+ for(i=0;i<max_sb;i++) {
+ alloc += 1 << alloc[0];
+ }
+
+ if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
+ /* nothing was coded for this band: add the necessary bits */
+ incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
+ incr += total_quant_bits[alloc[1]];
+ } else {
+ /* increments bit allocation */
+ b = bit_alloc[max_ch][max_sb];
+ incr = total_quant_bits[alloc[b + 1]] -
+ total_quant_bits[alloc[b]];
+ }
+
+ if (current_frame_size + incr <= max_frame_size) {
+ /* can increase size */
+ b = ++bit_alloc[max_ch][max_sb];
+ current_frame_size += incr;
+ /* decrease smr by the resolution we added */
+ smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
+ /* max allocation size reached ? */
+ if (b == ((1 << alloc[0]) - 1))
+ subband_status[max_ch][max_sb] = SB_NOMORE;
+ else
+ subband_status[max_ch][max_sb] = SB_ALLOCATED;
+ } else {
+ /* cannot increase the size of this subband */
+ subband_status[max_ch][max_sb] = SB_NOMORE;
+ }
+ }
+ *padding = max_frame_size - current_frame_size;
+ assert(*padding >= 0);
+
+#if 0
+ for(i=0;i<s->sblimit;i++) {
+ printf("%d ", bit_alloc[i]);
+ }
+ printf("\n");
+#endif
+}
+
+/*
+ * Output the mpeg audio layer 2 frame. Note how the code is small
+ * compared to other encoders :-)
+ */
+static void encode_frame(MpegAudioContext *s,
+ unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
+ int padding)
+{
+ int i, j, k, l, bit_alloc_bits, b, ch;
+ unsigned char *sf;
+ int q[3];
+ PutBitContext *p = &s->pb;
+
+ /* header */
+
+ put_bits(p, 12, 0xfff);
+ put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
+ put_bits(p, 2, 4-2); /* layer 2 */
+ put_bits(p, 1, 1); /* no error protection */
+ put_bits(p, 4, s->bitrate_index);
+ put_bits(p, 2, s->freq_index);
+ put_bits(p, 1, s->do_padding); /* use padding */
+ put_bits(p, 1, 0); /* private_bit */
+ put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
+ put_bits(p, 2, 0); /* mode_ext */
+ put_bits(p, 1, 0); /* no copyright */
+ put_bits(p, 1, 1); /* original */
+ put_bits(p, 2, 0); /* no emphasis */
+
+ /* bit allocation */
+ j = 0;
+ for(i=0;i<s->sblimit;i++) {
+ bit_alloc_bits = s->alloc_table[j];
+ for(ch=0;ch<s->nb_channels;ch++) {
+ put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
+ }
+ j += 1 << bit_alloc_bits;
+ }
+
+ /* scale codes */
+ for(i=0;i<s->sblimit;i++) {
+ for(ch=0;ch<s->nb_channels;ch++) {
+ if (bit_alloc[ch][i])
+ put_bits(p, 2, s->scale_code[ch][i]);
+ }
+ }
+
+ /* scale factors */
+ for(i=0;i<s->sblimit;i++) {
+ for(ch=0;ch<s->nb_channels;ch++) {
+ if (bit_alloc[ch][i]) {
+ sf = &s->scale_factors[ch][i][0];
+ switch(s->scale_code[ch][i]) {
+ case 0:
+ put_bits(p, 6, sf[0]);
+ put_bits(p, 6, sf[1]);
+ put_bits(p, 6, sf[2]);
+ break;
+ case 3:
+ case 1:
+ put_bits(p, 6, sf[0]);
+ put_bits(p, 6, sf[2]);
+ break;
+ case 2:
+ put_bits(p, 6, sf[0]);
+ break;
+ }
+ }
+ }
+ }
+
+ /* quantization & write sub band samples */
+
+ for(k=0;k<3;k++) {
+ for(l=0;l<12;l+=3) {
+ j = 0;
+ for(i=0;i<s->sblimit;i++) {
+ bit_alloc_bits = s->alloc_table[j];
+ for(ch=0;ch<s->nb_channels;ch++) {
+ b = bit_alloc[ch][i];
+ if (b) {
+ int qindex, steps, m, sample, bits;
+ /* we encode 3 sub band samples of the same sub band at a time */
+ qindex = s->alloc_table[j+b];
+ steps = quant_steps[qindex];
+ for(m=0;m<3;m++) {
+ sample = s->sb_samples[ch][k][l + m][i];
+ /* divide by scale factor */
+#ifdef USE_FLOATS
+ {
+ float a;
+ a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
+ q[m] = (int)((a + 1.0) * steps * 0.5);
+ }
+#else
+ {
+ int q1, e, shift, mult;
+ e = s->scale_factors[ch][i][k];
+ shift = scale_factor_shift[e];
+ mult = scale_factor_mult[e];
+
+ /* normalize to P bits */
+ if (shift < 0)
+ q1 = sample << (-shift);
+ else
+ q1 = sample >> shift;
+ q1 = (q1 * mult) >> P;
+ q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
+ }
+#endif
+ if (q[m] >= steps)
+ q[m] = steps - 1;
+ assert(q[m] >= 0 && q[m] < steps);
+ }
+ bits = quant_bits[qindex];
+ if (bits < 0) {
+ /* group the 3 values to save bits */
+ put_bits(p, -bits,
+ q[0] + steps * (q[1] + steps * q[2]));
+#if 0
+ printf("%d: gr1 %d\n",
+ i, q[0] + steps * (q[1] + steps * q[2]));
+#endif
+ } else {
+#if 0
+ printf("%d: gr3 %d %d %d\n",
+ i, q[0], q[1], q[2]);
+#endif
+ put_bits(p, bits, q[0]);
+ put_bits(p, bits, q[1]);
+ put_bits(p, bits, q[2]);
+ }
+ }
+ }
+ /* next subband in alloc table */
+ j += 1 << bit_alloc_bits;
+ }
+ }
+ }
+
+ /* padding */
+ for(i=0;i<padding;i++)
+ put_bits(p, 1, 0);
+
+ /* flush */
+ flush_put_bits(p);
+}
+
+int MPA_encode_frame(AVCodecContext *avctx,
+ unsigned char *frame, int buf_size, void *data)
+{
+ MpegAudioContext *s = avctx->priv_data;
+ short *samples = data;
+ short smr[MPA_MAX_CHANNELS][SBLIMIT];
+ unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
+ int padding, i;
+
+ for(i=0;i<s->nb_channels;i++) {
+ filter(s, i, samples + i, s->nb_channels);
+ }
+
+ for(i=0;i<s->nb_channels;i++) {
+ compute_scale_factors(s->scale_code[i], s->scale_factors[i],
+ s->sb_samples[i], s->sblimit);
+ }
+ for(i=0;i<s->nb_channels;i++) {
+ psycho_acoustic_model(s, smr[i]);
+ }
+ compute_bit_allocation(s, smr, bit_alloc, &padding);
+
+ init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
+
+ encode_frame(s, bit_alloc, padding);
+
+ s->nb_samples += MPA_FRAME_SIZE;
+ return s->pb.buf_ptr - s->pb.buf;
+}
+
+
+AVCodec mp2_encoder = {
+ "mp2",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_MP2,
+ sizeof(MpegAudioContext),
+ MPA_encode_init,
+ MPA_encode_frame,
+ NULL,
+};