Libav
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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 #undef CONFIG_MPEGAUDIO_HP 00031 #define CONFIG_MPEGAUDIO_HP 0 00032 #include "mpegaudio.h" 00033 00034 /* currently, cannot change these constants (need to modify 00035 quantization stage) */ 00036 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) 00037 00038 #define SAMPLES_BUF_SIZE 4096 00039 00040 typedef struct MpegAudioContext { 00041 PutBitContext pb; 00042 int nb_channels; 00043 int freq, bit_rate; 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 int64_t nb_samples; /* total number of samples encoded */ 00049 /* padding computation */ 00050 int frame_frac, frame_frac_incr, do_padding; 00051 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */ 00052 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */ 00053 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; 00054 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ 00055 /* code to group 3 scale factors */ 00056 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; 00057 int sblimit; /* number of used subbands */ 00058 const unsigned char *alloc_table; 00059 } MpegAudioContext; 00060 00061 /* define it to use floats in quantization (I don't like floats !) */ 00062 #define USE_FLOATS 00063 00064 #include "mpegaudiodata.h" 00065 #include "mpegaudiotab.h" 00066 00067 static av_cold int MPA_encode_init(AVCodecContext *avctx) 00068 { 00069 MpegAudioContext *s = avctx->priv_data; 00070 int freq = avctx->sample_rate; 00071 int bitrate = avctx->bit_rate; 00072 int channels = avctx->channels; 00073 int i, v, table; 00074 float a; 00075 00076 if (channels <= 0 || channels > 2){ 00077 av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels); 00078 return -1; 00079 } 00080 bitrate = bitrate / 1000; 00081 s->nb_channels = channels; 00082 s->freq = freq; 00083 s->bit_rate = bitrate * 1000; 00084 avctx->frame_size = MPA_FRAME_SIZE; 00085 00086 /* encoding freq */ 00087 s->lsf = 0; 00088 for(i=0;i<3;i++) { 00089 if (ff_mpa_freq_tab[i] == freq) 00090 break; 00091 if ((ff_mpa_freq_tab[i] / 2) == freq) { 00092 s->lsf = 1; 00093 break; 00094 } 00095 } 00096 if (i == 3){ 00097 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq); 00098 return -1; 00099 } 00100 s->freq_index = i; 00101 00102 /* encoding bitrate & frequency */ 00103 for(i=0;i<15;i++) { 00104 if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate) 00105 break; 00106 } 00107 if (i == 15){ 00108 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate); 00109 return -1; 00110 } 00111 s->bitrate_index = i; 00112 00113 /* compute total header size & pad bit */ 00114 00115 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); 00116 s->frame_size = ((int)a) * 8; 00117 00118 /* frame fractional size to compute padding */ 00119 s->frame_frac = 0; 00120 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0); 00121 00122 /* select the right allocation table */ 00123 table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf); 00124 00125 /* number of used subbands */ 00126 s->sblimit = ff_mpa_sblimit_table[table]; 00127 s->alloc_table = ff_mpa_alloc_tables[table]; 00128 00129 dprintf(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 00130 bitrate, freq, s->frame_size, table, s->frame_frac_incr); 00131 00132 for(i=0;i<s->nb_channels;i++) 00133 s->samples_offset[i] = 0; 00134 00135 for(i=0;i<257;i++) { 00136 int v; 00137 v = ff_mpa_enwindow[i]; 00138 #if WFRAC_BITS != 16 00139 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); 00140 #endif 00141 filter_bank[i] = v; 00142 if ((i & 63) != 0) 00143 v = -v; 00144 if (i != 0) 00145 filter_bank[512 - i] = v; 00146 } 00147 00148 for(i=0;i<64;i++) { 00149 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20)); 00150 if (v <= 0) 00151 v = 1; 00152 scale_factor_table[i] = v; 00153 #ifdef USE_FLOATS 00154 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20); 00155 #else 00156 #define P 15 00157 scale_factor_shift[i] = 21 - P - (i / 3); 00158 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0); 00159 #endif 00160 } 00161 for(i=0;i<128;i++) { 00162 v = i - 64; 00163 if (v <= -3) 00164 v = 0; 00165 else if (v < 0) 00166 v = 1; 00167 else if (v == 0) 00168 v = 2; 00169 else if (v < 3) 00170 v = 3; 00171 else 00172 v = 4; 00173 scale_diff_table[i] = v; 00174 } 00175 00176 for(i=0;i<17;i++) { 00177 v = ff_mpa_quant_bits[i]; 00178 if (v < 0) 00179 v = -v; 00180 else 00181 v = v * 3; 00182 total_quant_bits[i] = 12 * v; 00183 } 00184 00185 avctx->coded_frame= avcodec_alloc_frame(); 00186 avctx->coded_frame->key_frame= 1; 00187 00188 return 0; 00189 } 00190 00191 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ 00192 static void idct32(int *out, int *tab) 00193 { 00194 int i, j; 00195 int *t, *t1, xr; 00196 const int *xp = costab32; 00197 00198 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2]; 00199 00200 t = tab + 30; 00201 t1 = tab + 2; 00202 do { 00203 t[0] += t[-4]; 00204 t[1] += t[1 - 4]; 00205 t -= 4; 00206 } while (t != t1); 00207 00208 t = tab + 28; 00209 t1 = tab + 4; 00210 do { 00211 t[0] += t[-8]; 00212 t[1] += t[1-8]; 00213 t[2] += t[2-8]; 00214 t[3] += t[3-8]; 00215 t -= 8; 00216 } while (t != t1); 00217 00218 t = tab; 00219 t1 = tab + 32; 00220 do { 00221 t[ 3] = -t[ 3]; 00222 t[ 6] = -t[ 6]; 00223 00224 t[11] = -t[11]; 00225 t[12] = -t[12]; 00226 t[13] = -t[13]; 00227 t[15] = -t[15]; 00228 t += 16; 00229 } while (t != t1); 00230 00231 00232 t = tab; 00233 t1 = tab + 8; 00234 do { 00235 int x1, x2, x3, x4; 00236 00237 x3 = MUL(t[16], FIX(SQRT2*0.5)); 00238 x4 = t[0] - x3; 00239 x3 = t[0] + x3; 00240 00241 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5)); 00242 x1 = MUL((t[8] - x2), xp[0]); 00243 x2 = MUL((t[8] + x2), xp[1]); 00244 00245 t[ 0] = x3 + x1; 00246 t[ 8] = x4 - x2; 00247 t[16] = x4 + x2; 00248 t[24] = x3 - x1; 00249 t++; 00250 } while (t != t1); 00251 00252 xp += 2; 00253 t = tab; 00254 t1 = tab + 4; 00255 do { 00256 xr = MUL(t[28],xp[0]); 00257 t[28] = (t[0] - xr); 00258 t[0] = (t[0] + xr); 00259 00260 xr = MUL(t[4],xp[1]); 00261 t[ 4] = (t[24] - xr); 00262 t[24] = (t[24] + xr); 00263 00264 xr = MUL(t[20],xp[2]); 00265 t[20] = (t[8] - xr); 00266 t[ 8] = (t[8] + xr); 00267 00268 xr = MUL(t[12],xp[3]); 00269 t[12] = (t[16] - xr); 00270 t[16] = (t[16] + xr); 00271 t++; 00272 } while (t != t1); 00273 xp += 4; 00274 00275 for (i = 0; i < 4; i++) { 00276 xr = MUL(tab[30-i*4],xp[0]); 00277 tab[30-i*4] = (tab[i*4] - xr); 00278 tab[ i*4] = (tab[i*4] + xr); 00279 00280 xr = MUL(tab[ 2+i*4],xp[1]); 00281 tab[ 2+i*4] = (tab[28-i*4] - xr); 00282 tab[28-i*4] = (tab[28-i*4] + xr); 00283 00284 xr = MUL(tab[31-i*4],xp[0]); 00285 tab[31-i*4] = (tab[1+i*4] - xr); 00286 tab[ 1+i*4] = (tab[1+i*4] + xr); 00287 00288 xr = MUL(tab[ 3+i*4],xp[1]); 00289 tab[ 3+i*4] = (tab[29-i*4] - xr); 00290 tab[29-i*4] = (tab[29-i*4] + xr); 00291 00292 xp += 2; 00293 } 00294 00295 t = tab + 30; 00296 t1 = tab + 1; 00297 do { 00298 xr = MUL(t1[0], *xp); 00299 t1[0] = (t[0] - xr); 00300 t[0] = (t[0] + xr); 00301 t -= 2; 00302 t1 += 2; 00303 xp++; 00304 } while (t >= tab); 00305 00306 for(i=0;i<32;i++) { 00307 out[i] = tab[bitinv32[i]]; 00308 } 00309 } 00310 00311 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS) 00312 00313 static void filter(MpegAudioContext *s, int ch, short *samples, int incr) 00314 { 00315 short *p, *q; 00316 int sum, offset, i, j; 00317 int tmp[64]; 00318 int tmp1[32]; 00319 int *out; 00320 00321 // print_pow1(samples, 1152); 00322 00323 offset = s->samples_offset[ch]; 00324 out = &s->sb_samples[ch][0][0][0]; 00325 for(j=0;j<36;j++) { 00326 /* 32 samples at once */ 00327 for(i=0;i<32;i++) { 00328 s->samples_buf[ch][offset + (31 - i)] = samples[0]; 00329 samples += incr; 00330 } 00331 00332 /* filter */ 00333 p = s->samples_buf[ch] + offset; 00334 q = filter_bank; 00335 /* maxsum = 23169 */ 00336 for(i=0;i<64;i++) { 00337 sum = p[0*64] * q[0*64]; 00338 sum += p[1*64] * q[1*64]; 00339 sum += p[2*64] * q[2*64]; 00340 sum += p[3*64] * q[3*64]; 00341 sum += p[4*64] * q[4*64]; 00342 sum += p[5*64] * q[5*64]; 00343 sum += p[6*64] * q[6*64]; 00344 sum += p[7*64] * q[7*64]; 00345 tmp[i] = sum; 00346 p++; 00347 q++; 00348 } 00349 tmp1[0] = tmp[16] >> WSHIFT; 00350 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT; 00351 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT; 00352 00353 idct32(out, tmp1); 00354 00355 /* advance of 32 samples */ 00356 offset -= 32; 00357 out += 32; 00358 /* handle the wrap around */ 00359 if (offset < 0) { 00360 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), 00361 s->samples_buf[ch], (512 - 32) * 2); 00362 offset = SAMPLES_BUF_SIZE - 512; 00363 } 00364 } 00365 s->samples_offset[ch] = offset; 00366 00367 // print_pow(s->sb_samples, 1152); 00368 } 00369 00370 static void compute_scale_factors(unsigned char scale_code[SBLIMIT], 00371 unsigned char scale_factors[SBLIMIT][3], 00372 int sb_samples[3][12][SBLIMIT], 00373 int sblimit) 00374 { 00375 int *p, vmax, v, n, i, j, k, code; 00376 int index, d1, d2; 00377 unsigned char *sf = &scale_factors[0][0]; 00378 00379 for(j=0;j<sblimit;j++) { 00380 for(i=0;i<3;i++) { 00381 /* find the max absolute value */ 00382 p = &sb_samples[i][0][j]; 00383 vmax = abs(*p); 00384 for(k=1;k<12;k++) { 00385 p += SBLIMIT; 00386 v = abs(*p); 00387 if (v > vmax) 00388 vmax = v; 00389 } 00390 /* compute the scale factor index using log 2 computations */ 00391 if (vmax > 1) { 00392 n = av_log2(vmax); 00393 /* n is the position of the MSB of vmax. now 00394 use at most 2 compares to find the index */ 00395 index = (21 - n) * 3 - 3; 00396 if (index >= 0) { 00397 while (vmax <= scale_factor_table[index+1]) 00398 index++; 00399 } else { 00400 index = 0; /* very unlikely case of overflow */ 00401 } 00402 } else { 00403 index = 62; /* value 63 is not allowed */ 00404 } 00405 00406 #if 0 00407 printf("%2d:%d in=%x %x %d\n", 00408 j, i, vmax, scale_factor_table[index], index); 00409 #endif 00410 /* store the scale factor */ 00411 assert(index >=0 && index <= 63); 00412 sf[i] = index; 00413 } 00414 00415 /* compute the transmission factor : look if the scale factors 00416 are close enough to each other */ 00417 d1 = scale_diff_table[sf[0] - sf[1] + 64]; 00418 d2 = scale_diff_table[sf[1] - sf[2] + 64]; 00419 00420 /* handle the 25 cases */ 00421 switch(d1 * 5 + d2) { 00422 case 0*5+0: 00423 case 0*5+4: 00424 case 3*5+4: 00425 case 4*5+0: 00426 case 4*5+4: 00427 code = 0; 00428 break; 00429 case 0*5+1: 00430 case 0*5+2: 00431 case 4*5+1: 00432 case 4*5+2: 00433 code = 3; 00434 sf[2] = sf[1]; 00435 break; 00436 case 0*5+3: 00437 case 4*5+3: 00438 code = 3; 00439 sf[1] = sf[2]; 00440 break; 00441 case 1*5+0: 00442 case 1*5+4: 00443 case 2*5+4: 00444 code = 1; 00445 sf[1] = sf[0]; 00446 break; 00447 case 1*5+1: 00448 case 1*5+2: 00449 case 2*5+0: 00450 case 2*5+1: 00451 case 2*5+2: 00452 code = 2; 00453 sf[1] = sf[2] = sf[0]; 00454 break; 00455 case 2*5+3: 00456 case 3*5+3: 00457 code = 2; 00458 sf[0] = sf[1] = sf[2]; 00459 break; 00460 case 3*5+0: 00461 case 3*5+1: 00462 case 3*5+2: 00463 code = 2; 00464 sf[0] = sf[2] = sf[1]; 00465 break; 00466 case 1*5+3: 00467 code = 2; 00468 if (sf[0] > sf[2]) 00469 sf[0] = sf[2]; 00470 sf[1] = sf[2] = sf[0]; 00471 break; 00472 default: 00473 assert(0); //cannot happen 00474 code = 0; /* kill warning */ 00475 } 00476 00477 #if 0 00478 printf("%d: %2d %2d %2d %d %d -> %d\n", j, 00479 sf[0], sf[1], sf[2], d1, d2, code); 00480 #endif 00481 scale_code[j] = code; 00482 sf += 3; 00483 } 00484 } 00485 00486 /* The most important function : psycho acoustic module. In this 00487 encoder there is basically none, so this is the worst you can do, 00488 but also this is the simpler. */ 00489 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) 00490 { 00491 int i; 00492 00493 for(i=0;i<s->sblimit;i++) { 00494 smr[i] = (int)(fixed_smr[i] * 10); 00495 } 00496 } 00497 00498 00499 #define SB_NOTALLOCATED 0 00500 #define SB_ALLOCATED 1 00501 #define SB_NOMORE 2 00502 00503 /* Try to maximize the smr while using a number of bits inferior to 00504 the frame size. I tried to make the code simpler, faster and 00505 smaller than other encoders :-) */ 00506 static void compute_bit_allocation(MpegAudioContext *s, 00507 short smr1[MPA_MAX_CHANNELS][SBLIMIT], 00508 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 00509 int *padding) 00510 { 00511 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size; 00512 int incr; 00513 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 00514 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; 00515 const unsigned char *alloc; 00516 00517 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT); 00518 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT); 00519 memset(bit_alloc, 0, s->nb_channels * SBLIMIT); 00520 00521 /* compute frame size and padding */ 00522 max_frame_size = s->frame_size; 00523 s->frame_frac += s->frame_frac_incr; 00524 if (s->frame_frac >= 65536) { 00525 s->frame_frac -= 65536; 00526 s->do_padding = 1; 00527 max_frame_size += 8; 00528 } else { 00529 s->do_padding = 0; 00530 } 00531 00532 /* compute the header + bit alloc size */ 00533 current_frame_size = 32; 00534 alloc = s->alloc_table; 00535 for(i=0;i<s->sblimit;i++) { 00536 incr = alloc[0]; 00537 current_frame_size += incr * s->nb_channels; 00538 alloc += 1 << incr; 00539 } 00540 for(;;) { 00541 /* look for the subband with the largest signal to mask ratio */ 00542 max_sb = -1; 00543 max_ch = -1; 00544 max_smr = INT_MIN; 00545 for(ch=0;ch<s->nb_channels;ch++) { 00546 for(i=0;i<s->sblimit;i++) { 00547 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) { 00548 max_smr = smr[ch][i]; 00549 max_sb = i; 00550 max_ch = ch; 00551 } 00552 } 00553 } 00554 #if 0 00555 printf("current=%d max=%d max_sb=%d alloc=%d\n", 00556 current_frame_size, max_frame_size, max_sb, 00557 bit_alloc[max_sb]); 00558 #endif 00559 if (max_sb < 0) 00560 break; 00561 00562 /* find alloc table entry (XXX: not optimal, should use 00563 pointer table) */ 00564 alloc = s->alloc_table; 00565 for(i=0;i<max_sb;i++) { 00566 alloc += 1 << alloc[0]; 00567 } 00568 00569 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) { 00570 /* nothing was coded for this band: add the necessary bits */ 00571 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6; 00572 incr += total_quant_bits[alloc[1]]; 00573 } else { 00574 /* increments bit allocation */ 00575 b = bit_alloc[max_ch][max_sb]; 00576 incr = total_quant_bits[alloc[b + 1]] - 00577 total_quant_bits[alloc[b]]; 00578 } 00579 00580 if (current_frame_size + incr <= max_frame_size) { 00581 /* can increase size */ 00582 b = ++bit_alloc[max_ch][max_sb]; 00583 current_frame_size += incr; 00584 /* decrease smr by the resolution we added */ 00585 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]]; 00586 /* max allocation size reached ? */ 00587 if (b == ((1 << alloc[0]) - 1)) 00588 subband_status[max_ch][max_sb] = SB_NOMORE; 00589 else 00590 subband_status[max_ch][max_sb] = SB_ALLOCATED; 00591 } else { 00592 /* cannot increase the size of this subband */ 00593 subband_status[max_ch][max_sb] = SB_NOMORE; 00594 } 00595 } 00596 *padding = max_frame_size - current_frame_size; 00597 assert(*padding >= 0); 00598 00599 #if 0 00600 for(i=0;i<s->sblimit;i++) { 00601 printf("%d ", bit_alloc[i]); 00602 } 00603 printf("\n"); 00604 #endif 00605 } 00606 00607 /* 00608 * Output the mpeg audio layer 2 frame. Note how the code is small 00609 * compared to other encoders :-) 00610 */ 00611 static void encode_frame(MpegAudioContext *s, 00612 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 00613 int padding) 00614 { 00615 int i, j, k, l, bit_alloc_bits, b, ch; 00616 unsigned char *sf; 00617 int q[3]; 00618 PutBitContext *p = &s->pb; 00619 00620 /* header */ 00621 00622 put_bits(p, 12, 0xfff); 00623 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ 00624 put_bits(p, 2, 4-2); /* layer 2 */ 00625 put_bits(p, 1, 1); /* no error protection */ 00626 put_bits(p, 4, s->bitrate_index); 00627 put_bits(p, 2, s->freq_index); 00628 put_bits(p, 1, s->do_padding); /* use padding */ 00629 put_bits(p, 1, 0); /* private_bit */ 00630 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO); 00631 put_bits(p, 2, 0); /* mode_ext */ 00632 put_bits(p, 1, 0); /* no copyright */ 00633 put_bits(p, 1, 1); /* original */ 00634 put_bits(p, 2, 0); /* no emphasis */ 00635 00636 /* bit allocation */ 00637 j = 0; 00638 for(i=0;i<s->sblimit;i++) { 00639 bit_alloc_bits = s->alloc_table[j]; 00640 for(ch=0;ch<s->nb_channels;ch++) { 00641 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); 00642 } 00643 j += 1 << bit_alloc_bits; 00644 } 00645 00646 /* scale codes */ 00647 for(i=0;i<s->sblimit;i++) { 00648 for(ch=0;ch<s->nb_channels;ch++) { 00649 if (bit_alloc[ch][i]) 00650 put_bits(p, 2, s->scale_code[ch][i]); 00651 } 00652 } 00653 00654 /* scale factors */ 00655 for(i=0;i<s->sblimit;i++) { 00656 for(ch=0;ch<s->nb_channels;ch++) { 00657 if (bit_alloc[ch][i]) { 00658 sf = &s->scale_factors[ch][i][0]; 00659 switch(s->scale_code[ch][i]) { 00660 case 0: 00661 put_bits(p, 6, sf[0]); 00662 put_bits(p, 6, sf[1]); 00663 put_bits(p, 6, sf[2]); 00664 break; 00665 case 3: 00666 case 1: 00667 put_bits(p, 6, sf[0]); 00668 put_bits(p, 6, sf[2]); 00669 break; 00670 case 2: 00671 put_bits(p, 6, sf[0]); 00672 break; 00673 } 00674 } 00675 } 00676 } 00677 00678 /* quantization & write sub band samples */ 00679 00680 for(k=0;k<3;k++) { 00681 for(l=0;l<12;l+=3) { 00682 j = 0; 00683 for(i=0;i<s->sblimit;i++) { 00684 bit_alloc_bits = s->alloc_table[j]; 00685 for(ch=0;ch<s->nb_channels;ch++) { 00686 b = bit_alloc[ch][i]; 00687 if (b) { 00688 int qindex, steps, m, sample, bits; 00689 /* we encode 3 sub band samples of the same sub band at a time */ 00690 qindex = s->alloc_table[j+b]; 00691 steps = ff_mpa_quant_steps[qindex]; 00692 for(m=0;m<3;m++) { 00693 sample = s->sb_samples[ch][k][l + m][i]; 00694 /* divide by scale factor */ 00695 #ifdef USE_FLOATS 00696 { 00697 float a; 00698 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]]; 00699 q[m] = (int)((a + 1.0) * steps * 0.5); 00700 } 00701 #else 00702 { 00703 int q1, e, shift, mult; 00704 e = s->scale_factors[ch][i][k]; 00705 shift = scale_factor_shift[e]; 00706 mult = scale_factor_mult[e]; 00707 00708 /* normalize to P bits */ 00709 if (shift < 0) 00710 q1 = sample << (-shift); 00711 else 00712 q1 = sample >> shift; 00713 q1 = (q1 * mult) >> P; 00714 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1); 00715 } 00716 #endif 00717 if (q[m] >= steps) 00718 q[m] = steps - 1; 00719 assert(q[m] >= 0 && q[m] < steps); 00720 } 00721 bits = ff_mpa_quant_bits[qindex]; 00722 if (bits < 0) { 00723 /* group the 3 values to save bits */ 00724 put_bits(p, -bits, 00725 q[0] + steps * (q[1] + steps * q[2])); 00726 #if 0 00727 printf("%d: gr1 %d\n", 00728 i, q[0] + steps * (q[1] + steps * q[2])); 00729 #endif 00730 } else { 00731 #if 0 00732 printf("%d: gr3 %d %d %d\n", 00733 i, q[0], q[1], q[2]); 00734 #endif 00735 put_bits(p, bits, q[0]); 00736 put_bits(p, bits, q[1]); 00737 put_bits(p, bits, q[2]); 00738 } 00739 } 00740 } 00741 /* next subband in alloc table */ 00742 j += 1 << bit_alloc_bits; 00743 } 00744 } 00745 } 00746 00747 /* padding */ 00748 for(i=0;i<padding;i++) 00749 put_bits(p, 1, 0); 00750 00751 /* flush */ 00752 flush_put_bits(p); 00753 } 00754 00755 static int MPA_encode_frame(AVCodecContext *avctx, 00756 unsigned char *frame, int buf_size, void *data) 00757 { 00758 MpegAudioContext *s = avctx->priv_data; 00759 short *samples = data; 00760 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 00761 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; 00762 int padding, i; 00763 00764 for(i=0;i<s->nb_channels;i++) { 00765 filter(s, i, samples + i, s->nb_channels); 00766 } 00767 00768 for(i=0;i<s->nb_channels;i++) { 00769 compute_scale_factors(s->scale_code[i], s->scale_factors[i], 00770 s->sb_samples[i], s->sblimit); 00771 } 00772 for(i=0;i<s->nb_channels;i++) { 00773 psycho_acoustic_model(s, smr[i]); 00774 } 00775 compute_bit_allocation(s, smr, bit_alloc, &padding); 00776 00777 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE); 00778 00779 encode_frame(s, bit_alloc, padding); 00780 00781 s->nb_samples += MPA_FRAME_SIZE; 00782 return put_bits_ptr(&s->pb) - s->pb.buf; 00783 } 00784 00785 static av_cold int MPA_encode_close(AVCodecContext *avctx) 00786 { 00787 av_freep(&avctx->coded_frame); 00788 return 0; 00789 } 00790 00791 AVCodec mp2_encoder = { 00792 "mp2", 00793 AVMEDIA_TYPE_AUDIO, 00794 CODEC_ID_MP2, 00795 sizeof(MpegAudioContext), 00796 MPA_encode_init, 00797 MPA_encode_frame, 00798 MPA_encode_close, 00799 NULL, 00800 .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, 00801 .supported_samplerates= (const int[]){44100, 48000, 32000, 22050, 24000, 16000, 0}, 00802 .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), 00803 }; 00804 00805 #undef FIX