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libavcodec/mpegaudioenc.c

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00001 /*
00002  * The simplest mpeg audio layer 2 encoder
00003  * Copyright (c) 2000, 2001 Fabrice Bellard
00004  *
00005  * This file is part of FFmpeg.
00006  *
00007  * FFmpeg is free software; you can redistribute it and/or
00008  * modify it under the terms of the GNU Lesser General Public
00009  * License as published by the Free Software Foundation; either
00010  * version 2.1 of the License, or (at your option) any later version.
00011  *
00012  * FFmpeg is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00015  * Lesser General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU Lesser General Public
00018  * License along with FFmpeg; if not, write to the Free Software
00019  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00020  */
00021 
00027 #include "avcodec.h"
00028 #include "put_bits.h"
00029 
00030 #define FRAC_BITS   15   /* fractional bits for sb_samples and dct */
00031 #define WFRAC_BITS  14   /* fractional bits for window */
00032 
00033 #include "mpegaudio.h"
00034 
00035 /* currently, cannot change these constants (need to modify
00036    quantization stage) */
00037 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
00038 
00039 #define SAMPLES_BUF_SIZE 4096
00040 
00041 typedef struct MpegAudioContext {
00042     PutBitContext pb;
00043     int nb_channels;
00044     int lsf;           /* 1 if mpeg2 low bitrate selected */
00045     int bitrate_index; /* bit rate */
00046     int freq_index;
00047     int frame_size; /* frame size, in bits, without padding */
00048     /* padding computation */
00049     int frame_frac, frame_frac_incr, do_padding;
00050     short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
00051     int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
00052     int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
00053     unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
00054     /* code to group 3 scale factors */
00055     unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
00056     int sblimit; /* number of used subbands */
00057     const unsigned char *alloc_table;
00058 } MpegAudioContext;
00059 
00060 /* define it to use floats in quantization (I don't like floats !) */
00061 #define USE_FLOATS
00062 
00063 #include "mpegaudiodata.h"
00064 #include "mpegaudiotab.h"
00065 
00066 static av_cold int MPA_encode_init(AVCodecContext *avctx)
00067 {
00068     MpegAudioContext *s = avctx->priv_data;
00069     int freq = avctx->sample_rate;
00070     int bitrate = avctx->bit_rate;
00071     int channels = avctx->channels;
00072     int i, v, table;
00073     float a;
00074 
00075     if (channels <= 0 || channels > 2){
00076         av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
00077         return -1;
00078     }
00079     bitrate = bitrate / 1000;
00080     s->nb_channels = channels;
00081     avctx->frame_size = MPA_FRAME_SIZE;
00082 
00083     /* encoding freq */
00084     s->lsf = 0;
00085     for(i=0;i<3;i++) {
00086         if (ff_mpa_freq_tab[i] == freq)
00087             break;
00088         if ((ff_mpa_freq_tab[i] / 2) == freq) {
00089             s->lsf = 1;
00090             break;
00091         }
00092     }
00093     if (i == 3){
00094         av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
00095         return -1;
00096     }
00097     s->freq_index = i;
00098 
00099     /* encoding bitrate & frequency */
00100     for(i=0;i<15;i++) {
00101         if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
00102             break;
00103     }
00104     if (i == 15){
00105         av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
00106         return -1;
00107     }
00108     s->bitrate_index = i;
00109 
00110     /* compute total header size & pad bit */
00111 
00112     a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
00113     s->frame_size = ((int)a) * 8;
00114 
00115     /* frame fractional size to compute padding */
00116     s->frame_frac = 0;
00117     s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
00118 
00119     /* select the right allocation table */
00120     table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
00121 
00122     /* number of used subbands */
00123     s->sblimit = ff_mpa_sblimit_table[table];
00124     s->alloc_table = ff_mpa_alloc_tables[table];
00125 
00126     av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
00127             bitrate, freq, s->frame_size, table, s->frame_frac_incr);
00128 
00129     for(i=0;i<s->nb_channels;i++)
00130         s->samples_offset[i] = 0;
00131 
00132     for(i=0;i<257;i++) {
00133         int v;
00134         v = ff_mpa_enwindow[i];
00135 #if WFRAC_BITS != 16
00136         v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
00137 #endif
00138         filter_bank[i] = v;
00139         if ((i & 63) != 0)
00140             v = -v;
00141         if (i != 0)
00142             filter_bank[512 - i] = v;
00143     }
00144 
00145     for(i=0;i<64;i++) {
00146         v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
00147         if (v <= 0)
00148             v = 1;
00149         scale_factor_table[i] = v;
00150 #ifdef USE_FLOATS
00151         scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
00152 #else
00153 #define P 15
00154         scale_factor_shift[i] = 21 - P - (i / 3);
00155         scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
00156 #endif
00157     }
00158     for(i=0;i<128;i++) {
00159         v = i - 64;
00160         if (v <= -3)
00161             v = 0;
00162         else if (v < 0)
00163             v = 1;
00164         else if (v == 0)
00165             v = 2;
00166         else if (v < 3)
00167             v = 3;
00168         else
00169             v = 4;
00170         scale_diff_table[i] = v;
00171     }
00172 
00173     for(i=0;i<17;i++) {
00174         v = ff_mpa_quant_bits[i];
00175         if (v < 0)
00176             v = -v;
00177         else
00178             v = v * 3;
00179         total_quant_bits[i] = 12 * v;
00180     }
00181 
00182     avctx->coded_frame= avcodec_alloc_frame();
00183     avctx->coded_frame->key_frame= 1;
00184 
00185     return 0;
00186 }
00187 
00188 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
00189 static void idct32(int *out, int *tab)
00190 {
00191     int i, j;
00192     int *t, *t1, xr;
00193     const int *xp = costab32;
00194 
00195     for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
00196 
00197     t = tab + 30;
00198     t1 = tab + 2;
00199     do {
00200         t[0] += t[-4];
00201         t[1] += t[1 - 4];
00202         t -= 4;
00203     } while (t != t1);
00204 
00205     t = tab + 28;
00206     t1 = tab + 4;
00207     do {
00208         t[0] += t[-8];
00209         t[1] += t[1-8];
00210         t[2] += t[2-8];
00211         t[3] += t[3-8];
00212         t -= 8;
00213     } while (t != t1);
00214 
00215     t = tab;
00216     t1 = tab + 32;
00217     do {
00218         t[ 3] = -t[ 3];
00219         t[ 6] = -t[ 6];
00220 
00221         t[11] = -t[11];
00222         t[12] = -t[12];
00223         t[13] = -t[13];
00224         t[15] = -t[15];
00225         t += 16;
00226     } while (t != t1);
00227 
00228 
00229     t = tab;
00230     t1 = tab + 8;
00231     do {
00232         int x1, x2, x3, x4;
00233 
00234         x3 = MUL(t[16], FIX(SQRT2*0.5));
00235         x4 = t[0] - x3;
00236         x3 = t[0] + x3;
00237 
00238         x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
00239         x1 = MUL((t[8] - x2), xp[0]);
00240         x2 = MUL((t[8] + x2), xp[1]);
00241 
00242         t[ 0] = x3 + x1;
00243         t[ 8] = x4 - x2;
00244         t[16] = x4 + x2;
00245         t[24] = x3 - x1;
00246         t++;
00247     } while (t != t1);
00248 
00249     xp += 2;
00250     t = tab;
00251     t1 = tab + 4;
00252     do {
00253         xr = MUL(t[28],xp[0]);
00254         t[28] = (t[0] - xr);
00255         t[0] = (t[0] + xr);
00256 
00257         xr = MUL(t[4],xp[1]);
00258         t[ 4] = (t[24] - xr);
00259         t[24] = (t[24] + xr);
00260 
00261         xr = MUL(t[20],xp[2]);
00262         t[20] = (t[8] - xr);
00263         t[ 8] = (t[8] + xr);
00264 
00265         xr = MUL(t[12],xp[3]);
00266         t[12] = (t[16] - xr);
00267         t[16] = (t[16] + xr);
00268         t++;
00269     } while (t != t1);
00270     xp += 4;
00271 
00272     for (i = 0; i < 4; i++) {
00273         xr = MUL(tab[30-i*4],xp[0]);
00274         tab[30-i*4] = (tab[i*4] - xr);
00275         tab[   i*4] = (tab[i*4] + xr);
00276 
00277         xr = MUL(tab[ 2+i*4],xp[1]);
00278         tab[ 2+i*4] = (tab[28-i*4] - xr);
00279         tab[28-i*4] = (tab[28-i*4] + xr);
00280 
00281         xr = MUL(tab[31-i*4],xp[0]);
00282         tab[31-i*4] = (tab[1+i*4] - xr);
00283         tab[ 1+i*4] = (tab[1+i*4] + xr);
00284 
00285         xr = MUL(tab[ 3+i*4],xp[1]);
00286         tab[ 3+i*4] = (tab[29-i*4] - xr);
00287         tab[29-i*4] = (tab[29-i*4] + xr);
00288 
00289         xp += 2;
00290     }
00291 
00292     t = tab + 30;
00293     t1 = tab + 1;
00294     do {
00295         xr = MUL(t1[0], *xp);
00296         t1[0] = (t[0] - xr);
00297         t[0] = (t[0] + xr);
00298         t -= 2;
00299         t1 += 2;
00300         xp++;
00301     } while (t >= tab);
00302 
00303     for(i=0;i<32;i++) {
00304         out[i] = tab[bitinv32[i]];
00305     }
00306 }
00307 
00308 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
00309 
00310 static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
00311 {
00312     short *p, *q;
00313     int sum, offset, i, j;
00314     int tmp[64];
00315     int tmp1[32];
00316     int *out;
00317 
00318     //    print_pow1(samples, 1152);
00319 
00320     offset = s->samples_offset[ch];
00321     out = &s->sb_samples[ch][0][0][0];
00322     for(j=0;j<36;j++) {
00323         /* 32 samples at once */
00324         for(i=0;i<32;i++) {
00325             s->samples_buf[ch][offset + (31 - i)] = samples[0];
00326             samples += incr;
00327         }
00328 
00329         /* filter */
00330         p = s->samples_buf[ch] + offset;
00331         q = filter_bank;
00332         /* maxsum = 23169 */
00333         for(i=0;i<64;i++) {
00334             sum = p[0*64] * q[0*64];
00335             sum += p[1*64] * q[1*64];
00336             sum += p[2*64] * q[2*64];
00337             sum += p[3*64] * q[3*64];
00338             sum += p[4*64] * q[4*64];
00339             sum += p[5*64] * q[5*64];
00340             sum += p[6*64] * q[6*64];
00341             sum += p[7*64] * q[7*64];
00342             tmp[i] = sum;
00343             p++;
00344             q++;
00345         }
00346         tmp1[0] = tmp[16] >> WSHIFT;
00347         for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
00348         for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
00349 
00350         idct32(out, tmp1);
00351 
00352         /* advance of 32 samples */
00353         offset -= 32;
00354         out += 32;
00355         /* handle the wrap around */
00356         if (offset < 0) {
00357             memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
00358                     s->samples_buf[ch], (512 - 32) * 2);
00359             offset = SAMPLES_BUF_SIZE - 512;
00360         }
00361     }
00362     s->samples_offset[ch] = offset;
00363 
00364     //    print_pow(s->sb_samples, 1152);
00365 }
00366 
00367 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
00368                                   unsigned char scale_factors[SBLIMIT][3],
00369                                   int sb_samples[3][12][SBLIMIT],
00370                                   int sblimit)
00371 {
00372     int *p, vmax, v, n, i, j, k, code;
00373     int index, d1, d2;
00374     unsigned char *sf = &scale_factors[0][0];
00375 
00376     for(j=0;j<sblimit;j++) {
00377         for(i=0;i<3;i++) {
00378             /* find the max absolute value */
00379             p = &sb_samples[i][0][j];
00380             vmax = abs(*p);
00381             for(k=1;k<12;k++) {
00382                 p += SBLIMIT;
00383                 v = abs(*p);
00384                 if (v > vmax)
00385                     vmax = v;
00386             }
00387             /* compute the scale factor index using log 2 computations */
00388             if (vmax > 1) {
00389                 n = av_log2(vmax);
00390                 /* n is the position of the MSB of vmax. now
00391                    use at most 2 compares to find the index */
00392                 index = (21 - n) * 3 - 3;
00393                 if (index >= 0) {
00394                     while (vmax <= scale_factor_table[index+1])
00395                         index++;
00396                 } else {
00397                     index = 0; /* very unlikely case of overflow */
00398                 }
00399             } else {
00400                 index = 62; /* value 63 is not allowed */
00401             }
00402 
00403 #if 0
00404             printf("%2d:%d in=%x %x %d\n",
00405                    j, i, vmax, scale_factor_table[index], index);
00406 #endif
00407             /* store the scale factor */
00408             assert(index >=0 && index <= 63);
00409             sf[i] = index;
00410         }
00411 
00412         /* compute the transmission factor : look if the scale factors
00413            are close enough to each other */
00414         d1 = scale_diff_table[sf[0] - sf[1] + 64];
00415         d2 = scale_diff_table[sf[1] - sf[2] + 64];
00416 
00417         /* handle the 25 cases */
00418         switch(d1 * 5 + d2) {
00419         case 0*5+0:
00420         case 0*5+4:
00421         case 3*5+4:
00422         case 4*5+0:
00423         case 4*5+4:
00424             code = 0;
00425             break;
00426         case 0*5+1:
00427         case 0*5+2:
00428         case 4*5+1:
00429         case 4*5+2:
00430             code = 3;
00431             sf[2] = sf[1];
00432             break;
00433         case 0*5+3:
00434         case 4*5+3:
00435             code = 3;
00436             sf[1] = sf[2];
00437             break;
00438         case 1*5+0:
00439         case 1*5+4:
00440         case 2*5+4:
00441             code = 1;
00442             sf[1] = sf[0];
00443             break;
00444         case 1*5+1:
00445         case 1*5+2:
00446         case 2*5+0:
00447         case 2*5+1:
00448         case 2*5+2:
00449             code = 2;
00450             sf[1] = sf[2] = sf[0];
00451             break;
00452         case 2*5+3:
00453         case 3*5+3:
00454             code = 2;
00455             sf[0] = sf[1] = sf[2];
00456             break;
00457         case 3*5+0:
00458         case 3*5+1:
00459         case 3*5+2:
00460             code = 2;
00461             sf[0] = sf[2] = sf[1];
00462             break;
00463         case 1*5+3:
00464             code = 2;
00465             if (sf[0] > sf[2])
00466               sf[0] = sf[2];
00467             sf[1] = sf[2] = sf[0];
00468             break;
00469         default:
00470             assert(0); //cannot happen
00471             code = 0;           /* kill warning */
00472         }
00473 
00474 #if 0
00475         printf("%d: %2d %2d %2d %d %d -> %d\n", j,
00476                sf[0], sf[1], sf[2], d1, d2, code);
00477 #endif
00478         scale_code[j] = code;
00479         sf += 3;
00480     }
00481 }
00482 
00483 /* The most important function : psycho acoustic module. In this
00484    encoder there is basically none, so this is the worst you can do,
00485    but also this is the simpler. */
00486 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
00487 {
00488     int i;
00489 
00490     for(i=0;i<s->sblimit;i++) {
00491         smr[i] = (int)(fixed_smr[i] * 10);
00492     }
00493 }
00494 
00495 
00496 #define SB_NOTALLOCATED  0
00497 #define SB_ALLOCATED     1
00498 #define SB_NOMORE        2
00499 
00500 /* Try to maximize the smr while using a number of bits inferior to
00501    the frame size. I tried to make the code simpler, faster and
00502    smaller than other encoders :-) */
00503 static void compute_bit_allocation(MpegAudioContext *s,
00504                                    short smr1[MPA_MAX_CHANNELS][SBLIMIT],
00505                                    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
00506                                    int *padding)
00507 {
00508     int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
00509     int incr;
00510     short smr[MPA_MAX_CHANNELS][SBLIMIT];
00511     unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
00512     const unsigned char *alloc;
00513 
00514     memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
00515     memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
00516     memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
00517 
00518     /* compute frame size and padding */
00519     max_frame_size = s->frame_size;
00520     s->frame_frac += s->frame_frac_incr;
00521     if (s->frame_frac >= 65536) {
00522         s->frame_frac -= 65536;
00523         s->do_padding = 1;
00524         max_frame_size += 8;
00525     } else {
00526         s->do_padding = 0;
00527     }
00528 
00529     /* compute the header + bit alloc size */
00530     current_frame_size = 32;
00531     alloc = s->alloc_table;
00532     for(i=0;i<s->sblimit;i++) {
00533         incr = alloc[0];
00534         current_frame_size += incr * s->nb_channels;
00535         alloc += 1 << incr;
00536     }
00537     for(;;) {
00538         /* look for the subband with the largest signal to mask ratio */
00539         max_sb = -1;
00540         max_ch = -1;
00541         max_smr = INT_MIN;
00542         for(ch=0;ch<s->nb_channels;ch++) {
00543             for(i=0;i<s->sblimit;i++) {
00544                 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
00545                     max_smr = smr[ch][i];
00546                     max_sb = i;
00547                     max_ch = ch;
00548                 }
00549             }
00550         }
00551         if (max_sb < 0)
00552             break;
00553         av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
00554                 current_frame_size, max_frame_size, max_sb, max_ch,
00555                 bit_alloc[max_ch][max_sb]);
00556 
00557         /* find alloc table entry (XXX: not optimal, should use
00558            pointer table) */
00559         alloc = s->alloc_table;
00560         for(i=0;i<max_sb;i++) {
00561             alloc += 1 << alloc[0];
00562         }
00563 
00564         if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
00565             /* nothing was coded for this band: add the necessary bits */
00566             incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
00567             incr += total_quant_bits[alloc[1]];
00568         } else {
00569             /* increments bit allocation */
00570             b = bit_alloc[max_ch][max_sb];
00571             incr = total_quant_bits[alloc[b + 1]] -
00572                 total_quant_bits[alloc[b]];
00573         }
00574 
00575         if (current_frame_size + incr <= max_frame_size) {
00576             /* can increase size */
00577             b = ++bit_alloc[max_ch][max_sb];
00578             current_frame_size += incr;
00579             /* decrease smr by the resolution we added */
00580             smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
00581             /* max allocation size reached ? */
00582             if (b == ((1 << alloc[0]) - 1))
00583                 subband_status[max_ch][max_sb] = SB_NOMORE;
00584             else
00585                 subband_status[max_ch][max_sb] = SB_ALLOCATED;
00586         } else {
00587             /* cannot increase the size of this subband */
00588             subband_status[max_ch][max_sb] = SB_NOMORE;
00589         }
00590     }
00591     *padding = max_frame_size - current_frame_size;
00592     assert(*padding >= 0);
00593 }
00594 
00595 /*
00596  * Output the mpeg audio layer 2 frame. Note how the code is small
00597  * compared to other encoders :-)
00598  */
00599 static void encode_frame(MpegAudioContext *s,
00600                          unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
00601                          int padding)
00602 {
00603     int i, j, k, l, bit_alloc_bits, b, ch;
00604     unsigned char *sf;
00605     int q[3];
00606     PutBitContext *p = &s->pb;
00607 
00608     /* header */
00609 
00610     put_bits(p, 12, 0xfff);
00611     put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
00612     put_bits(p, 2, 4-2);  /* layer 2 */
00613     put_bits(p, 1, 1); /* no error protection */
00614     put_bits(p, 4, s->bitrate_index);
00615     put_bits(p, 2, s->freq_index);
00616     put_bits(p, 1, s->do_padding); /* use padding */
00617     put_bits(p, 1, 0);             /* private_bit */
00618     put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
00619     put_bits(p, 2, 0); /* mode_ext */
00620     put_bits(p, 1, 0); /* no copyright */
00621     put_bits(p, 1, 1); /* original */
00622     put_bits(p, 2, 0); /* no emphasis */
00623 
00624     /* bit allocation */
00625     j = 0;
00626     for(i=0;i<s->sblimit;i++) {
00627         bit_alloc_bits = s->alloc_table[j];
00628         for(ch=0;ch<s->nb_channels;ch++) {
00629             put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
00630         }
00631         j += 1 << bit_alloc_bits;
00632     }
00633 
00634     /* scale codes */
00635     for(i=0;i<s->sblimit;i++) {
00636         for(ch=0;ch<s->nb_channels;ch++) {
00637             if (bit_alloc[ch][i])
00638                 put_bits(p, 2, s->scale_code[ch][i]);
00639         }
00640     }
00641 
00642     /* scale factors */
00643     for(i=0;i<s->sblimit;i++) {
00644         for(ch=0;ch<s->nb_channels;ch++) {
00645             if (bit_alloc[ch][i]) {
00646                 sf = &s->scale_factors[ch][i][0];
00647                 switch(s->scale_code[ch][i]) {
00648                 case 0:
00649                     put_bits(p, 6, sf[0]);
00650                     put_bits(p, 6, sf[1]);
00651                     put_bits(p, 6, sf[2]);
00652                     break;
00653                 case 3:
00654                 case 1:
00655                     put_bits(p, 6, sf[0]);
00656                     put_bits(p, 6, sf[2]);
00657                     break;
00658                 case 2:
00659                     put_bits(p, 6, sf[0]);
00660                     break;
00661                 }
00662             }
00663         }
00664     }
00665 
00666     /* quantization & write sub band samples */
00667 
00668     for(k=0;k<3;k++) {
00669         for(l=0;l<12;l+=3) {
00670             j = 0;
00671             for(i=0;i<s->sblimit;i++) {
00672                 bit_alloc_bits = s->alloc_table[j];
00673                 for(ch=0;ch<s->nb_channels;ch++) {
00674                     b = bit_alloc[ch][i];
00675                     if (b) {
00676                         int qindex, steps, m, sample, bits;
00677                         /* we encode 3 sub band samples of the same sub band at a time */
00678                         qindex = s->alloc_table[j+b];
00679                         steps = ff_mpa_quant_steps[qindex];
00680                         for(m=0;m<3;m++) {
00681                             sample = s->sb_samples[ch][k][l + m][i];
00682                             /* divide by scale factor */
00683 #ifdef USE_FLOATS
00684                             {
00685                                 float a;
00686                                 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
00687                                 q[m] = (int)((a + 1.0) * steps * 0.5);
00688                             }
00689 #else
00690                             {
00691                                 int q1, e, shift, mult;
00692                                 e = s->scale_factors[ch][i][k];
00693                                 shift = scale_factor_shift[e];
00694                                 mult = scale_factor_mult[e];
00695 
00696                                 /* normalize to P bits */
00697                                 if (shift < 0)
00698                                     q1 = sample << (-shift);
00699                                 else
00700                                     q1 = sample >> shift;
00701                                 q1 = (q1 * mult) >> P;
00702                                 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
00703                             }
00704 #endif
00705                             if (q[m] >= steps)
00706                                 q[m] = steps - 1;
00707                             assert(q[m] >= 0 && q[m] < steps);
00708                         }
00709                         bits = ff_mpa_quant_bits[qindex];
00710                         if (bits < 0) {
00711                             /* group the 3 values to save bits */
00712                             put_bits(p, -bits,
00713                                      q[0] + steps * (q[1] + steps * q[2]));
00714                         } else {
00715                             put_bits(p, bits, q[0]);
00716                             put_bits(p, bits, q[1]);
00717                             put_bits(p, bits, q[2]);
00718                         }
00719                     }
00720                 }
00721                 /* next subband in alloc table */
00722                 j += 1 << bit_alloc_bits;
00723             }
00724         }
00725     }
00726 
00727     /* padding */
00728     for(i=0;i<padding;i++)
00729         put_bits(p, 1, 0);
00730 
00731     /* flush */
00732     flush_put_bits(p);
00733 }
00734 
00735 static int MPA_encode_frame(AVCodecContext *avctx,
00736                             unsigned char *frame, int buf_size, void *data)
00737 {
00738     MpegAudioContext *s = avctx->priv_data;
00739     const short *samples = data;
00740     short smr[MPA_MAX_CHANNELS][SBLIMIT];
00741     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
00742     int padding, i;
00743 
00744     for(i=0;i<s->nb_channels;i++) {
00745         filter(s, i, samples + i, s->nb_channels);
00746     }
00747 
00748     for(i=0;i<s->nb_channels;i++) {
00749         compute_scale_factors(s->scale_code[i], s->scale_factors[i],
00750                               s->sb_samples[i], s->sblimit);
00751     }
00752     for(i=0;i<s->nb_channels;i++) {
00753         psycho_acoustic_model(s, smr[i]);
00754     }
00755     compute_bit_allocation(s, smr, bit_alloc, &padding);
00756 
00757     init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
00758 
00759     encode_frame(s, bit_alloc, padding);
00760 
00761     return put_bits_ptr(&s->pb) - s->pb.buf;
00762 }
00763 
00764 static av_cold int MPA_encode_close(AVCodecContext *avctx)
00765 {
00766     av_freep(&avctx->coded_frame);
00767     return 0;
00768 }
00769 
00770 AVCodec ff_mp2_encoder = {
00771     "mp2",
00772     AVMEDIA_TYPE_AUDIO,
00773     CODEC_ID_MP2,
00774     sizeof(MpegAudioContext),
00775     MPA_encode_init,
00776     MPA_encode_frame,
00777     MPA_encode_close,
00778     NULL,
00779     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
00780     .supported_samplerates= (const int[]){44100, 48000,  32000, 22050, 24000, 16000, 0},
00781     .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
00782 };

Generated on Fri Feb 22 2013 07:24:27 for FFmpeg by  doxygen 1.7.1