FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavcodec/mpegaudioenc.c
Date: 2026-10-17 12:12:06
Exec Total Coverage
Lines: 404 412 98.1%
Functions: 10 10 100.0%
Branches: 171 181 94.5%
Line Branch Exec Source
1 /*
2 * The simplest mpeg audio layer 2 encoder
3 * Copyright (c) 2000, 2001 Fabrice Bellard
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file
24 * The simplest mpeg audio layer 2 encoder.
25 */
26
27 #include "config.h"
28 #include "config_components.h"
29
30 #include "libavutil/avassert.h"
31 #include "libavutil/channel_layout.h"
32
33 #include "avcodec.h"
34 #include "codec_internal.h"
35 #include "encode.h"
36 #include "put_bits.h"
37
38 #define FRAC_BITS 15 /* fractional bits for sb_samples and dct */
39 #define WFRAC_BITS 14 /* fractional bits for window */
40
41 #include "mpegaudio.h"
42 #include "mpegaudiodsp.h"
43 #include "mpegaudiodata.h"
44 #include "mpegaudiotab.h"
45
46 /* currently, cannot change these constants (need to modify
47 quantization stage) */
48 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
49
50 #define SAMPLES_BUF_SIZE 4096
51
52 typedef struct MpegAudioContext {
53 int nb_channels;
54 int lsf; /* 1 if mpeg2 low bitrate selected */
55 int bitrate_index; /* bit rate */
56 int freq_index;
57 int frame_size; /* frame size, in bits, without padding */
58 int is_fixed;
59 /* padding computation */
60 int frame_frac, frame_frac_incr, do_padding;
61 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
62 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
63 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
64 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
65 /* code to group 3 scale factors */
66 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
67 int sblimit; /* number of used subbands */
68 const unsigned char *alloc_table;
69 int16_t filter_bank[512];
70 int scale_factor_table[64];
71 unsigned char scale_diff_table[128];
72 union {
73 float scale_factor_inv_table[64];
74 struct {
75 int8_t scale_factor_shift[64];
76 unsigned short scale_factor_mult[64];
77 };
78 };
79 unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
80 } MpegAudioContext;
81
82 #define IS_FIXED(s) (CONFIG_MP2_ENCODER && CONFIG_MP2FIXED_ENCODER ? (s)->is_fixed : CONFIG_MP2FIXED_ENCODER)
83
84 129 static av_cold int mpa_encode_init(AVCodecContext *avctx)
85 {
86 129 MpegAudioContext *s = avctx->priv_data;
87 129 int freq = avctx->sample_rate;
88 129 int bitrate = avctx->bit_rate;
89 129 int channels = avctx->ch_layout.nb_channels;
90 int i, table;
91 float a;
92
93 129 bitrate = bitrate / 1000;
94 129 s->nb_channels = channels;
95 129 avctx->frame_size = MPA_FRAME_SIZE;
96 129 avctx->initial_padding = 512 - 32 + 1;
97
98 /* encoding freq */
99 129 s->lsf = 0;
100 129 for (i = 0;; i++) {
101 204 av_assert1(i < 3);
102
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 333 if (ff_mpa_freq_tab[i] == freq)
103 27 break;
104
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 306 if ((ff_mpa_freq_tab[i] / 2) == freq) {
105 102 s->lsf = 1;
106 102 break;
107 }
108 }
109 129 s->freq_index = i;
110
111 /* encoding bitrate & frequency */
112
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 417 for(i=1;i<15;i++) {
113
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 400 if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
114 112 break;
115 }
116
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 129 if (i == 15 && !avctx->bit_rate) {
117 17 i = 14;
118 17 bitrate = ff_mpa_bitrate_tab[s->lsf][1][i];
119 17 avctx->bit_rate = bitrate * 1000;
120 }
121
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 129 if (i == 15){
122 ✗ av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
123 ✗ return AVERROR(EINVAL);
124 }
125 129 s->bitrate_index = i;
126
127 /* compute total header size & pad bit */
128
129 129 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
130 129 s->frame_size = ((int)a) * 8;
131
132 /* frame fractional size to compute padding */
133 129 s->frame_frac = 0;
134 129 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
135
136 /* select the right allocation table */
137 129 table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
138
139 /* number of used subbands */
140 129 s->sblimit = ff_mpa_sblimit_table[table];
141 129 s->alloc_table = ff_mpa_alloc_tables[table];
142
143 ff_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
144 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
145
146
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 261 for(i=0;i<s->nb_channels;i++)
147 132 s->samples_offset[i] = 0;
148
149
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 33282 for(i=0;i<257;i++) {
150 int v;
151 33153 v = ff_mpa_enwindow[i];
152 #if WFRAC_BITS != 16
153 33153 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
154 #endif
155 33153 s->filter_bank[i] = v;
156
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 33153 if ((i & 63) != 0)
157 32508 v = -v;
158
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 33153 if (i != 0)
159 33024 s->filter_bank[512 - i] = v;
160 }
161
162
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 8385 for(i=0;i<64;i++) {
163 8256 int v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
164
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 8256 if (v <= 0)
165 ✗ v = 1;
166 8256 s->scale_factor_table[i] = v;
167
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 8256 if (IS_FIXED(s)) {
168 #define P 15
169 7488 s->scale_factor_shift[i] = 21 - P - (i / 3);
170 7488 s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
171 } else {
172 768 s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
173 }
174 }
175
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 16641 for(i=0;i<128;i++) {
176 16512 int v = i - 64;
177
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 16512 if (v <= -3)
178 7998 v = 0;
179
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 8514 else if (v < 0)
180 258 v = 1;
181
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 8256 else if (v == 0)
182 129 v = 2;
183
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 8127 else if (v < 3)
184 258 v = 3;
185 else
186 7869 v = 4;
187 16512 s->scale_diff_table[i] = v;
188 }
189
190
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 2322 for(i=0;i<17;i++) {
191 2193 int v = ff_mpa_quant_bits[i];
192
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 2193 if (v < 0)
193 387 v = -v;
194 else
195 1806 v = v * 3;
196 2193 s->total_quant_bits[i] = 12 * v;
197 }
198
199 129 return 0;
200 }
201
202 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
203 467820 static void idct32(int *out, int *tab)
204 {
205 int i, j;
206 int *t, *t1, xr;
207 467820 const int *xp = costab32;
208
209
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 7485120 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
210
211 467820 t = tab + 30;
212 467820 t1 = tab + 2;
213 do {
214 3274740 t[0] += t[-4];
215 3274740 t[1] += t[1 - 4];
216 3274740 t -= 4;
217
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 3274740 } while (t != t1);
218
219 467820 t = tab + 28;
220 467820 t1 = tab + 4;
221 do {
222 1403460 t[0] += t[-8];
223 1403460 t[1] += t[1-8];
224 1403460 t[2] += t[2-8];
225 1403460 t[3] += t[3-8];
226 1403460 t -= 8;
227
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 1403460 } while (t != t1);
228
229 467820 t = tab;
230 467820 t1 = tab + 32;
231 do {
232 935640 t[ 3] = -t[ 3];
233 935640 t[ 6] = -t[ 6];
234
235 935640 t[11] = -t[11];
236 935640 t[12] = -t[12];
237 935640 t[13] = -t[13];
238 935640 t[15] = -t[15];
239 935640 t += 16;
240
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 935640 } while (t != t1);
241
242
243 467820 t = tab;
244 467820 t1 = tab + 8;
245 do {
246 int x1, x2, x3, x4;
247
248 3742560 x3 = MUL(t[16], FIX(M_SQRT2*0.5));
249 3742560 x4 = t[0] - x3;
250 3742560 x3 = t[0] + x3;
251
252 3742560 x2 = MUL(-(t[24] + t[8]), FIX(M_SQRT2*0.5));
253 3742560 x1 = MUL((t[8] - x2), xp[0]);
254 3742560 x2 = MUL((t[8] + x2), xp[1]);
255
256 3742560 t[ 0] = x3 + x1;
257 3742560 t[ 8] = x4 - x2;
258 3742560 t[16] = x4 + x2;
259 3742560 t[24] = x3 - x1;
260 3742560 t++;
261
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 3742560 } while (t != t1);
262
263 467820 xp += 2;
264 467820 t = tab;
265 467820 t1 = tab + 4;
266 do {
267 1871280 xr = MUL(t[28],xp[0]);
268 1871280 t[28] = (t[0] - xr);
269 1871280 t[0] = (t[0] + xr);
270
271 1871280 xr = MUL(t[4],xp[1]);
272 1871280 t[ 4] = (t[24] - xr);
273 1871280 t[24] = (t[24] + xr);
274
275 1871280 xr = MUL(t[20],xp[2]);
276 1871280 t[20] = (t[8] - xr);
277 1871280 t[ 8] = (t[8] + xr);
278
279 1871280 xr = MUL(t[12],xp[3]);
280 1871280 t[12] = (t[16] - xr);
281 1871280 t[16] = (t[16] + xr);
282 1871280 t++;
283
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 1871280 } while (t != t1);
284 467820 xp += 4;
285
286
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 2339100 for (i = 0; i < 4; i++) {
287 1871280 xr = MUL(tab[30-i*4],xp[0]);
288 1871280 tab[30-i*4] = (tab[i*4] - xr);
289 1871280 tab[ i*4] = (tab[i*4] + xr);
290
291 1871280 xr = MUL(tab[ 2+i*4],xp[1]);
292 1871280 tab[ 2+i*4] = (tab[28-i*4] - xr);
293 1871280 tab[28-i*4] = (tab[28-i*4] + xr);
294
295 1871280 xr = MUL(tab[31-i*4],xp[0]);
296 1871280 tab[31-i*4] = (tab[1+i*4] - xr);
297 1871280 tab[ 1+i*4] = (tab[1+i*4] + xr);
298
299 1871280 xr = MUL(tab[ 3+i*4],xp[1]);
300 1871280 tab[ 3+i*4] = (tab[29-i*4] - xr);
301 1871280 tab[29-i*4] = (tab[29-i*4] + xr);
302
303 1871280 xp += 2;
304 }
305
306 467820 t = tab + 30;
307 467820 t1 = tab + 1;
308 do {
309 7485120 xr = MUL(t1[0], *xp);
310 7485120 t1[0] = (t[0] - xr);
311 7485120 t[0] = (t[0] + xr);
312 7485120 t -= 2;
313 7485120 t1 += 2;
314 7485120 xp++;
315
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 7485120 } while (t >= tab);
316
317
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 15438060 for(i=0;i<32;i++) {
318 14970240 out[i] = tab[bitinv32[i]];
319 }
320 467820 }
321
322 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
323
324 12995 static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
325 {
326 short *p, *q;
327 int sum, offset, i, j;
328 int tmp[64];
329 int tmp1[32];
330 int *out;
331
332 12995 offset = s->samples_offset[ch];
333 12995 out = &s->sb_samples[ch][0][0][0];
334
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 480815 for(j=0;j<36;j++) {
335 /* 32 samples at once */
336
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 15438060 for(i=0;i<32;i++) {
337 14970240 s->samples_buf[ch][offset + (31 - i)] = samples[0];
338 14970240 samples += incr;
339 }
340
341 /* filter */
342 467820 p = s->samples_buf[ch] + offset;
343 467820 q = s->filter_bank;
344 /* maxsum = 23169 */
345
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 30408300 for(i=0;i<64;i++) {
346 29940480 sum = p[0*64] * q[0*64];
347 29940480 sum += p[1*64] * q[1*64];
348 29940480 sum += p[2*64] * q[2*64];
349 29940480 sum += p[3*64] * q[3*64];
350 29940480 sum += p[4*64] * q[4*64];
351 29940480 sum += p[5*64] * q[5*64];
352 29940480 sum += p[6*64] * q[6*64];
353 29940480 sum += p[7*64] * q[7*64];
354 29940480 tmp[i] = sum;
355 29940480 p++;
356 29940480 q++;
357 }
358 467820 tmp1[0] = tmp[16] >> WSHIFT;
359
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 7952940 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
360
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 7485120 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
361
362 467820 idct32(out, tmp1);
363
364 /* advance of 32 samples */
365 467820 offset -= 32;
366 467820 out += 32;
367 /* handle the wrap around */
368
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 467820 if (offset < 0) {
369 4171 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
370 4171 s->samples_buf[ch], (512 - 32) * 2);
371 4171 offset = SAMPLES_BUF_SIZE - 512;
372 }
373 }
374 12995 s->samples_offset[ch] = offset;
375 12995 }
376
377 12995 static void compute_scale_factors(MpegAudioContext *s,
378 unsigned char scale_code[SBLIMIT],
379 unsigned char scale_factors[SBLIMIT][3],
380 int sb_samples[3][12][SBLIMIT],
381 int sblimit)
382 {
383 int *p, vmax, v, n, i, j, k, code;
384 int index, d1, d2;
385 12995 unsigned char *sf = &scale_factors[0][0];
386
387
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 400253 for(j=0;j<sblimit;j++) {
388
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 1549032 for(i=0;i<3;i++) {
389 /* find the max absolute value */
390 1161774 p = &sb_samples[i][0][j];
391 1161774 vmax = abs(*p);
392
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 13941288 for(k=1;k<12;k++) {
393 12779514 p += SBLIMIT;
394 12779514 v = abs(*p);
395
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 12779514 if (v > vmax)
396 2142780 vmax = v;
397 }
398 /* compute the scale factor index using log 2 computations */
399
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 1161774 if (vmax > 1) {
400 1138454 n = av_log2(vmax);
401 /* n is the position of the MSB of vmax. now
402 use at most 2 compares to find the index */
403 1138454 index = (21 - n) * 3 - 3;
404
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 1138454 if (index >= 0) {
405
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 2562300 while (vmax <= s->scale_factor_table[index+1])
406 1423846 index++;
407 } else {
408 ✗ index = 0; /* very unlikely case of overflow */
409 }
410 } else {
411 23320 index = 62; /* value 63 is not allowed */
412 }
413
414 ff_dlog(NULL, "%2d:%d in=%x %x %d\n",
415 j, i, vmax, s->scale_factor_table[index], index);
416 /* store the scale factor */
417 av_assert2(index >=0 && index <= 63);
418 1161774 sf[i] = index;
419 }
420
421 /* compute the transmission factor : look if the scale factors
422 are close enough to each other */
423 387258 d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
424 387258 d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
425
426 /* handle the 25 cases */
427
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 387258 switch(d1 * 5 + d2) {
428 8144 case 0*5+0:
429 case 0*5+4:
430 case 3*5+4:
431 case 4*5+0:
432 case 4*5+4:
433 8144 code = 0;
434 8144 break;
435 9121 case 0*5+1:
436 case 0*5+2:
437 case 4*5+1:
438 case 4*5+2:
439 9121 code = 3;
440 9121 sf[2] = sf[1];
441 9121 break;
442 6073 case 0*5+3:
443 case 4*5+3:
444 6073 code = 3;
445 6073 sf[1] = sf[2];
446 6073 break;
447 7664 case 1*5+0:
448 case 1*5+4:
449 case 2*5+4:
450 7664 code = 1;
451 7664 sf[1] = sf[0];
452 7664 break;
453 178408 case 1*5+1:
454 case 1*5+2:
455 case 2*5+0:
456 case 2*5+1:
457 case 2*5+2:
458 178408 code = 2;
459 178408 sf[1] = sf[2] = sf[0];
460 178408 break;
461 50674 case 2*5+3:
462 case 3*5+3:
463 50674 code = 2;
464 50674 sf[0] = sf[1] = sf[2];
465 50674 break;
466 82768 case 3*5+0:
467 case 3*5+1:
468 case 3*5+2:
469 82768 code = 2;
470 82768 sf[0] = sf[2] = sf[1];
471 82768 break;
472 44406 case 1*5+3:
473 44406 code = 2;
474
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 44406 if (sf[0] > sf[2])
475 7349 sf[0] = sf[2];
476 44406 sf[1] = sf[2] = sf[0];
477 44406 break;
478 ✗ default:
479 av_assert2(0); //cannot happen
480 ✗ code = 0; /* kill warning */
481 }
482
483 ff_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
484 sf[0], sf[1], sf[2], d1, d2, code);
485 387258 scale_code[j] = code;
486 387258 sf += 3;
487 }
488 12995 }
489
490 /* The most important function : psycho acoustic module. In this
491 encoder there is basically none, so this is the worst you can do,
492 but also this is the simpler. */
493 12995 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
494 {
495 int i;
496
497
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 400253 for(i=0;i<s->sblimit;i++) {
498 387258 smr[i] = (int)(fixed_smr[i] * 10);
499 }
500 12995 }
501
502
503 #define SB_NOTALLOCATED 0
504 #define SB_ALLOCATED 1
505 #define SB_NOMORE 2
506
507 /* Try to maximize the smr while using a number of bits inferior to
508 the frame size. I tried to make the code simpler, faster and
509 smaller than other encoders :-) */
510 12528 static unsigned compute_bit_allocation(MpegAudioContext *s,
511 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
512 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
513 int *padding)
514 {
515 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
516 int incr;
517 short smr[MPA_MAX_CHANNELS][SBLIMIT];
518 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
519 const unsigned char *alloc;
520
521 12528 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
522 12528 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
523 12528 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
524
525 /* compute frame size and padding */
526 12528 max_frame_size = s->frame_size;
527 12528 s->frame_frac += s->frame_frac_incr;
528
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 12528 if (s->frame_frac >= 65536) {
529 8461 s->frame_frac -= 65536;
530 8461 s->do_padding = 1;
531 8461 max_frame_size += 8;
532 } else {
533 4067 s->do_padding = 0;
534 }
535
536 /* compute the header + bit alloc size */
537 12528 current_frame_size = 32;
538 12528 alloc = s->alloc_table;
539
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 386487 for(i=0;i<s->sblimit;i++) {
540 373959 incr = alloc[0];
541 373959 current_frame_size += incr * s->nb_channels;
542 373959 alloc += 1 << incr;
543 }
544 for(;;) {
545 /* look for the subband with the largest signal to mask ratio */
546 2237020 max_sb = -1;
547 2237020 max_ch = -1;
548 2237020 max_smr = INT_MIN;
549
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 4569519 for(ch=0;ch<s->nb_channels;ch++) {
550
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551
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 69735936 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
552 5790128 max_smr = smr[ch][i];
553 5790128 max_sb = i;
554 5790128 max_ch = ch;
555 }
556 }
557 }
558
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 2237020 if (max_sb < 0)
559 12528 break;
560 ff_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
561 current_frame_size, max_frame_size, max_sb, max_ch,
562 bit_alloc[max_ch][max_sb]);
563
564 /* find alloc table entry (XXX: not optimal, should use
565 pointer table) */
566 2224492 alloc = s->alloc_table;
567
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 22443891 for(i=0;i<max_sb;i++) {
568 20219399 alloc += 1 << alloc[0];
569 }
570
571
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 2224492 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
572 /* nothing was coded for this band: add the necessary bits */
573 387258 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
574 387258 incr += s->total_quant_bits[alloc[1]];
575 } else {
576 /* increments bit allocation */
577 1837234 b = bit_alloc[max_ch][max_sb];
578 1837234 incr = s->total_quant_bits[alloc[b + 1]] -
579 1837234 s->total_quant_bits[alloc[b]];
580 }
581
582
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 2224492 if (current_frame_size + incr <= max_frame_size) {
583 /* can increase size */
584 1912384 b = ++bit_alloc[max_ch][max_sb];
585 1912384 current_frame_size += incr;
586 /* decrease smr by the resolution we added */
587 1912384 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
588 /* max allocation size reached ? */
589
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 1912384 if (b == ((1 << alloc[0]) - 1))
590 75150 subband_status[max_ch][max_sb] = SB_NOMORE;
591 else
592 1837234 subband_status[max_ch][max_sb] = SB_ALLOCATED;
593 } else {
594 /* cannot increase the size of this subband */
595 312108 subband_status[max_ch][max_sb] = SB_NOMORE;
596 }
597 }
598 12528 *padding = max_frame_size - current_frame_size;
599
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 12528 av_assert0(*padding >= 0);
600 12528 return max_frame_size / 8U;
601 }
602
603 /// Quantization & write sub band samples
604 12528 static av_always_inline void encode_subbands(MpegAudioContext *const s,
605 PutBitContext *const p,
606 const uint8_t bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
607 int is_fixed)
608 {
609
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 50112 for (int k = 0; k < 3; ++k) {
610
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 187920 for (int l = 0; l < 12; l += 3) {
611
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 4637844 for (int i = 0, j = 0; i < s->sblimit; ++i) {
612 4487508 const int bit_alloc_bits = s->alloc_table[j];
613
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 9134604 for (int ch = 0; ch < s->nb_channels; ++ch) {
614 4647096 const int b = bit_alloc[ch][i];
615
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 4647096 if (b) {
616 /* we encode 3 sub band samples of the same sub band at a time */
617 2794248 const int qindex = s->alloc_table[j + b];
618 2794248 const int steps = ff_mpa_quant_steps[qindex];
619 int q[3];
620
621
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 11176992 for (int m = 0; m < 3; ++m) {
622 8382744 const int sample = s->sb_samples[ch][k][l + m][i];
623 /* divide by scale factor */
624
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 8382744 if (!is_fixed) {
625 606960 float a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]];
626 606960 q[m] = (int)((a + 1.0) * steps * 0.5);
627 } else {
628 7775784 const int e = s->scale_factors[ch][i][k];
629 7775784 const int shift = s->scale_factor_shift[e];
630 7775784 const int mult = s->scale_factor_mult[e];
631 int q1;
632
633 /* normalize to P bits */
634
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 7775784 if (shift < 0)
635 6954108 q1 = sample * (1 << -shift);
636 else
637 821676 q1 = sample >> shift;
638 7775784 q1 = (q1 * mult) >> P;
639 7775784 q1 += 1 << P;
640
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 7775784 if (q1 < 0)
641 ✗ q1 = 0;
642 7775784 q[m] = (q1 * (unsigned)steps) >> (P + 1);
643 }
644
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 8382744 if (q[m] >= steps)
645 13395 q[m] = steps - 1;
646 av_assert2(q[m] >= 0 && q[m] < steps);
647 }
648 2794248 const int bits = ff_mpa_quant_bits[qindex];
649
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 2794248 if (bits < 0) {
650 /* group the 3 values to save bits */
651 506916 put_bits(p, -bits,
652 506916 q[0] + steps * (q[1] + steps * q[2]));
653 } else {
654 2287332 put_bits(p, bits, q[0]);
655 2287332 put_bits(p, bits, q[1]);
656 2287332 put_bits(p, bits, q[2]);
657 }
658 }
659 }
660 /* next subband in alloc table */
661 4487508 j += 1 << bit_alloc_bits;
662 }
663 }
664 }
665 12528 }
666
667 /*
668 * Output the MPEG audio layer 2 frame. Note how the code is small
669 * compared to other encoders :-)
670 */
671 12528 static void encode_frame(MpegAudioContext *s, uint8_t *buf, unsigned buf_size,
672 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
673 int padding)
674 {
675 int i, j, bit_alloc_bits, ch;
676 unsigned char *sf;
677 12528 PutBitContext p0, *p = &p0;
678
679 12528 init_put_bits(p, buf, buf_size);
680
681 /* header */
682
683 12528 put_bits(p, 12, 0xfff);
684 12528 put_bits(p, 1, 1 - s->lsf); /* 1 = MPEG-1 ID, 0 = MPEG-2 lsf ID */
685 12528 put_bits(p, 2, 4-2); /* layer 2 */
686 12528 put_bits(p, 1, 1); /* no error protection */
687 12528 put_bits(p, 4, s->bitrate_index);
688 12528 put_bits(p, 2, s->freq_index);
689 12528 put_bits(p, 1, s->do_padding); /* use padding */
690 12528 put_bits(p, 1, 0); /* private_bit */
691
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 12528 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
692 12528 put_bits(p, 2, 0); /* mode_ext */
693 12528 put_bits(p, 1, 0); /* no copyright */
694 12528 put_bits(p, 1, 1); /* original */
695 12528 put_bits(p, 2, 0); /* no emphasis */
696
697 /* bit allocation */
698 12528 j = 0;
699
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 386487 for(i=0;i<s->sblimit;i++) {
700 373959 bit_alloc_bits = s->alloc_table[j];
701
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 761217 for(ch=0;ch<s->nb_channels;ch++) {
702 387258 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
703 }
704 373959 j += 1 << bit_alloc_bits;
705 }
706
707 /* scale codes */
708
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709
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710
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 387258 if (bit_alloc[ch][i])
711 232854 put_bits(p, 2, s->scale_code[ch][i]);
712 }
713 }
714
715 /* scale factors */
716
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 386487 for(i=0;i<s->sblimit;i++) {
717
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 761217 for(ch=0;ch<s->nb_channels;ch++) {
718
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 387258 if (bit_alloc[ch][i]) {
719 232854 sf = &s->scale_factors[ch][i][0];
720
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 232854 switch(s->scale_code[ch][i]) {
721 3832 case 0:
722 3832 put_bits(p, 18, sf[0] << 12 | sf[1] << 6 | sf[2]);
723 3832 break;
724 10191 case 3:
725 case 1:
726 10191 put_bits(p, 12, sf[0] << 6 | sf[2]);
727 10191 break;
728 218831 case 2:
729 218831 put_bits(p, 6, sf[0]);
730 218831 break;
731 }
732 }
733 }
734 }
735
736 #if CONFIG_SMALL
737 encode_subbands(s, p, bit_alloc, IS_FIXED(s));
738 #else
739
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 12528 if (IS_FIXED(s))
740 11799 encode_subbands(s, p, bit_alloc, 1);
741 else
742 729 encode_subbands(s, p, bit_alloc, 0);
743 #endif
744
745 av_assert1(put_bits_left(p) == padding);
746
747 /* flush */
748 12528 flush_put_bits(p);
749
750 /* padding */
751
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 12528 if (put_bytes_left(p, 0))
752 8870 memset(put_bits_ptr(p), 0, put_bytes_left(p, 0));
753 12528 }
754
755 12528 static int mpa_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
756 const AVFrame *frame, int *got_packet_ptr)
757 {
758 12528 MpegAudioContext *s = avctx->priv_data;
759 12528 const int16_t *samples = (const int16_t *)frame->data[0];
760 short smr[MPA_MAX_CHANNELS][SBLIMIT];
761 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
762 int padding, i, ret;
763
764
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 25523 for(i=0;i<s->nb_channels;i++) {
765 12995 filter(s, i, samples + i, s->nb_channels);
766 }
767
768
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 25523 for(i=0;i<s->nb_channels;i++) {
769 12995 compute_scale_factors(s, s->scale_code[i], s->scale_factors[i],
770 12995 s->sb_samples[i], s->sblimit);
771 }
772
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 25523 for(i=0;i<s->nb_channels;i++) {
773 12995 psycho_acoustic_model(s, smr[i]);
774 }
775 12528 unsigned frame_size = compute_bit_allocation(s, smr, bit_alloc, &padding);
776
777 12528 ret = ff_get_encode_buffer(avctx, avpkt, frame_size, 0);
778
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 12528 if (ret < 0)
779 ✗ return ret;
780
781 12528 encode_frame(s, avpkt->data, frame_size, bit_alloc, padding);
782
783
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 12528 if (frame->pts != AV_NOPTS_VALUE)
784 12528 avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
785
786 12528 *got_packet_ptr = 1;
787 12528 return 0;
788 }
789
790 static const FFCodecDefault mp2_defaults[] = {
791 { "b", "0" },
792 { NULL },
793 };
794
795 #if CONFIG_MP2_ENCODER
796 const FFCodec ff_mp2_encoder = {
797 .p.name = "mp2",
798 CODEC_LONG_NAME("MP2 (MPEG audio layer 2)"),
799 .p.type = AVMEDIA_TYPE_AUDIO,
800 .p.id = AV_CODEC_ID_MP2,
801 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE,
802 .priv_data_size = sizeof(MpegAudioContext),
803 .init = mpa_encode_init,
804 FF_CODEC_ENCODE_CB(mpa_encode_frame),
805 CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_S16),
806 CODEC_SAMPLERATES(44100, 48000, 32000, 22050, 24000, 16000),
807 CODEC_CH_LAYOUTS(AV_CHANNEL_LAYOUT_MONO, AV_CHANNEL_LAYOUT_STEREO),
808 .defaults = mp2_defaults,
809 };
810 #endif
811
812 #if CONFIG_MP2FIXED_ENCODER
813 117 static av_cold int mpa_fixed_encode_init(AVCodecContext *avctx)
814 {
815 117 MpegAudioContext *s = avctx->priv_data;
816
817 117 s->is_fixed = 1;
818 117 return mpa_encode_init(avctx);
819 }
820
821 const FFCodec ff_mp2fixed_encoder = {
822 .p.name = "mp2fixed",
823 CODEC_LONG_NAME("MP2 fixed point (MPEG audio layer 2)"),
824 .p.type = AVMEDIA_TYPE_AUDIO,
825 .p.id = AV_CODEC_ID_MP2,
826 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE,
827 .priv_data_size = sizeof(MpegAudioContext),
828 .init = mpa_fixed_encode_init,
829 FF_CODEC_ENCODE_CB(mpa_encode_frame),
830 CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_S16),
831 CODEC_SAMPLERATES(44100, 48000, 32000, 22050, 24000, 16000),
832 CODEC_CH_LAYOUTS(AV_CHANNEL_LAYOUT_MONO, AV_CHANNEL_LAYOUT_STEREO),
833 .defaults = mp2_defaults,
834 };
835 #endif
836