ffmpeg / libavcodec / mpegaudio.c @ 029911d1
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/*


2 
* The simplest mpeg audio layer 2 encoder

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* Copyright (c) 2000, 2001 Fabrice Bellard.

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*

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* This library is free software; you can redistribute it and/or

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* modify it under the terms of the GNU Lesser General Public

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* License as published by the Free Software Foundation; either

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* version 2 of the License, or (at your option) any later version.

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*

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* This library is distributed in the hope that it will be useful,

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* but WITHOUT ANY WARRANTY; without even the implied warranty of

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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

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* Lesser General Public License for more details.

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*

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* You should have received a copy of the GNU Lesser General Public

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* License along with this library; if not, write to the Free Software

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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 021111307 USA

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*/

19 

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/**

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* @file mpegaudio.c

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* The simplest mpeg audio layer 2 encoder.

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*/

24 

25 
#include "avcodec.h" 
26 
#include "mpegaudio.h" 
27  
28 
/* currently, cannot change these constants (need to modify

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quantization stage) */

30 
#define FRAC_BITS 15 
31 
#define WFRAC_BITS 14 
32 
#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)

33 
#define FIX(a) ((int)((a) * (1 << FRAC_BITS))) 
34  
35 
#define SAMPLES_BUF_SIZE 4096 
36  
37 
typedef struct MpegAudioContext { 
38 
PutBitContext pb; 
39 
int nb_channels;

40 
int freq, bit_rate;

41 
int lsf; /* 1 if mpeg2 low bitrate selected */ 
42 
int bitrate_index; /* bit rate */ 
43 
int freq_index;

44 
int frame_size; /* frame size, in bits, without padding */ 
45 
int64_t nb_samples; /* total number of samples encoded */

46 
/* padding computation */

47 
int frame_frac, frame_frac_incr, do_padding;

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short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */ 
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int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */ 
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int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; 
51 
unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ 
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/* code to group 3 scale factors */

53 
unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; 
54 
int sblimit; /* number of used subbands */ 
55 
const unsigned char *alloc_table; 
56 
} MpegAudioContext; 
57  
58 
/* define it to use floats in quantization (I don't like floats !) */

59 
//#define USE_FLOATS

60  
61 
#include "mpegaudiotab.h" 
62  
63 
static int MPA_encode_init(AVCodecContext *avctx) 
64 
{ 
65 
MpegAudioContext *s = avctx>priv_data; 
66 
int freq = avctx>sample_rate;

67 
int bitrate = avctx>bit_rate;

68 
int channels = avctx>channels;

69 
int i, v, table;

70 
float a;

71  
72 
if (channels > 2) 
73 
return 1; 
74 
bitrate = bitrate / 1000;

75 
s>nb_channels = channels; 
76 
s>freq = freq; 
77 
s>bit_rate = bitrate * 1000;

78 
avctx>frame_size = MPA_FRAME_SIZE; 
79  
80 
/* encoding freq */

81 
s>lsf = 0;

82 
for(i=0;i<3;i++) { 
83 
if (mpa_freq_tab[i] == freq)

84 
break;

85 
if ((mpa_freq_tab[i] / 2) == freq) { 
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s>lsf = 1;

87 
break;

88 
} 
89 
} 
90 
if (i == 3) 
91 
return 1; 
92 
s>freq_index = i; 
93  
94 
/* encoding bitrate & frequency */

95 
for(i=0;i<15;i++) { 
96 
if (mpa_bitrate_tab[s>lsf][1][i] == bitrate) 
97 
break;

98 
} 
99 
if (i == 15) 
100 
return 1; 
101 
s>bitrate_index = i; 
102  
103 
/* compute total header size & pad bit */

104 

105 
a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); 
106 
s>frame_size = ((int)a) * 8; 
107  
108 
/* frame fractional size to compute padding */

109 
s>frame_frac = 0;

110 
s>frame_frac_incr = (int)((a  floor(a)) * 65536.0); 
111 

112 
/* select the right allocation table */

113 
table = l2_select_table(bitrate, s>nb_channels, freq, s>lsf); 
114  
115 
/* number of used subbands */

116 
s>sblimit = sblimit_table[table]; 
117 
s>alloc_table = alloc_tables[table]; 
118  
119 
#ifdef DEBUG

120 
printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",

121 
bitrate, freq, s>frame_size, table, s>frame_frac_incr); 
122 
#endif

123  
124 
for(i=0;i<s>nb_channels;i++) 
125 
s>samples_offset[i] = 0;

126  
127 
for(i=0;i<257;i++) { 
128 
int v;

129 
v = mpa_enwindow[i]; 
130 
#if WFRAC_BITS != 16 
131 
v = (v + (1 << (16  WFRAC_BITS  1))) >> (16  WFRAC_BITS); 
132 
#endif

133 
filter_bank[i] = v; 
134 
if ((i & 63) != 0) 
135 
v = v; 
136 
if (i != 0) 
137 
filter_bank[512  i] = v;

138 
} 
139  
140 
for(i=0;i<64;i++) { 
141 
v = (int)(pow(2.0, (3  i) / 3.0) * (1 << 20)); 
142 
if (v <= 0) 
143 
v = 1;

144 
scale_factor_table[i] = v; 
145 
#ifdef USE_FLOATS

146 
scale_factor_inv_table[i] = pow(2.0, (3  i) / 3.0) / (float)(1 << 20); 
147 
#else

148 
#define P 15 
149 
scale_factor_shift[i] = 21  P  (i / 3); 
150 
scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0); 
151 
#endif

152 
} 
153 
for(i=0;i<128;i++) { 
154 
v = i  64;

155 
if (v <= 3) 
156 
v = 0;

157 
else if (v < 0) 
158 
v = 1;

159 
else if (v == 0) 
160 
v = 2;

161 
else if (v < 3) 
162 
v = 3;

163 
else

164 
v = 4;

165 
scale_diff_table[i] = v; 
166 
} 
167  
168 
for(i=0;i<17;i++) { 
169 
v = quant_bits[i]; 
170 
if (v < 0) 
171 
v = v; 
172 
else

173 
v = v * 3;

174 
total_quant_bits[i] = 12 * v;

175 
} 
176  
177 
avctx>coded_frame= avcodec_alloc_frame(); 
178 
avctx>coded_frame>key_frame= 1;

179  
180 
return 0; 
181 
} 
182  
183 
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */

184 
static void idct32(int *out, int *tab) 
185 
{ 
186 
int i, j;

187 
int *t, *t1, xr;

188 
const int *xp = costab32; 
189  
190 
for(j=31;j>=3;j=2) tab[j] += tab[j  2]; 
191 

192 
t = tab + 30;

193 
t1 = tab + 2;

194 
do {

195 
t[0] += t[4]; 
196 
t[1] += t[1  4]; 
197 
t = 4;

198 
} while (t != t1);

199  
200 
t = tab + 28;

201 
t1 = tab + 4;

202 
do {

203 
t[0] += t[8]; 
204 
t[1] += t[18]; 
205 
t[2] += t[28]; 
206 
t[3] += t[38]; 
207 
t = 8;

208 
} while (t != t1);

209 

210 
t = tab; 
211 
t1 = tab + 32;

212 
do {

213 
t[ 3] = t[ 3]; 
214 
t[ 6] = t[ 6]; 
215 

216 
t[11] = t[11]; 
217 
t[12] = t[12]; 
218 
t[13] = t[13]; 
219 
t[15] = t[15]; 
220 
t += 16;

221 
} while (t != t1);

222  
223 

224 
t = tab; 
225 
t1 = tab + 8;

226 
do {

227 
int x1, x2, x3, x4;

228 

229 
x3 = MUL(t[16], FIX(SQRT2*0.5)); 
230 
x4 = t[0]  x3;

231 
x3 = t[0] + x3;

232 

233 
x2 = MUL((t[24] + t[8]), FIX(SQRT2*0.5)); 
234 
x1 = MUL((t[8]  x2), xp[0]); 
235 
x2 = MUL((t[8] + x2), xp[1]); 
236  
237 
t[ 0] = x3 + x1;

238 
t[ 8] = x4  x2;

239 
t[16] = x4 + x2;

240 
t[24] = x3  x1;

241 
t++; 
242 
} while (t != t1);

243  
244 
xp += 2;

245 
t = tab; 
246 
t1 = tab + 4;

247 
do {

248 
xr = MUL(t[28],xp[0]); 
249 
t[28] = (t[0]  xr); 
250 
t[0] = (t[0] + xr); 
251  
252 
xr = MUL(t[4],xp[1]); 
253 
t[ 4] = (t[24]  xr); 
254 
t[24] = (t[24] + xr); 
255 

256 
xr = MUL(t[20],xp[2]); 
257 
t[20] = (t[8]  xr); 
258 
t[ 8] = (t[8] + xr); 
259 

260 
xr = MUL(t[12],xp[3]); 
261 
t[12] = (t[16]  xr); 
262 
t[16] = (t[16] + xr); 
263 
t++; 
264 
} while (t != t1);

265 
xp += 4;

266  
267 
for (i = 0; i < 4; i++) { 
268 
xr = MUL(tab[30i*4],xp[0]); 
269 
tab[30i*4] = (tab[i*4]  xr); 
270 
tab[ i*4] = (tab[i*4] + xr); 
271 

272 
xr = MUL(tab[ 2+i*4],xp[1]); 
273 
tab[ 2+i*4] = (tab[28i*4]  xr); 
274 
tab[28i*4] = (tab[28i*4] + xr); 
275 

276 
xr = MUL(tab[31i*4],xp[0]); 
277 
tab[31i*4] = (tab[1+i*4]  xr); 
278 
tab[ 1+i*4] = (tab[1+i*4] + xr); 
279 

280 
xr = MUL(tab[ 3+i*4],xp[1]); 
281 
tab[ 3+i*4] = (tab[29i*4]  xr); 
282 
tab[29i*4] = (tab[29i*4] + xr); 
283 

284 
xp += 2;

285 
} 
286  
287 
t = tab + 30;

288 
t1 = tab + 1;

289 
do {

290 
xr = MUL(t1[0], *xp);

291 
t1[0] = (t[0]  xr); 
292 
t[0] = (t[0] + xr); 
293 
t = 2;

294 
t1 += 2;

295 
xp++; 
296 
} while (t >= tab);

297  
298 
for(i=0;i<32;i++) { 
299 
out[i] = tab[bitinv32[i]]; 
300 
} 
301 
} 
302  
303 
#define WSHIFT (WFRAC_BITS + 15  FRAC_BITS) 
304  
305 
static void filter(MpegAudioContext *s, int ch, short *samples, int incr) 
306 
{ 
307 
short *p, *q;

308 
int sum, offset, i, j;

309 
int tmp[64]; 
310 
int tmp1[32]; 
311 
int *out;

312  
313 
// print_pow1(samples, 1152);

314  
315 
offset = s>samples_offset[ch]; 
316 
out = &s>sb_samples[ch][0][0][0]; 
317 
for(j=0;j<36;j++) { 
318 
/* 32 samples at once */

319 
for(i=0;i<32;i++) { 
320 
s>samples_buf[ch][offset + (31  i)] = samples[0]; 
321 
samples += incr; 
322 
} 
323  
324 
/* filter */

325 
p = s>samples_buf[ch] + offset; 
326 
q = filter_bank; 
327 
/* maxsum = 23169 */

328 
for(i=0;i<64;i++) { 
329 
sum = p[0*64] * q[0*64]; 
330 
sum += p[1*64] * q[1*64]; 
331 
sum += p[2*64] * q[2*64]; 
332 
sum += p[3*64] * q[3*64]; 
333 
sum += p[4*64] * q[4*64]; 
334 
sum += p[5*64] * q[5*64]; 
335 
sum += p[6*64] * q[6*64]; 
336 
sum += p[7*64] * q[7*64]; 
337 
tmp[i] = sum; 
338 
p++; 
339 
q++; 
340 
} 
341 
tmp1[0] = tmp[16] >> WSHIFT; 
342 
for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16i]) >> WSHIFT; 
343 
for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]tmp[80i]) >> WSHIFT; 
344  
345 
idct32(out, tmp1); 
346  
347 
/* advance of 32 samples */

348 
offset = 32;

349 
out += 32;

350 
/* handle the wrap around */

351 
if (offset < 0) { 
352 
memmove(s>samples_buf[ch] + SAMPLES_BUF_SIZE  (512  32), 
353 
s>samples_buf[ch], (512  32) * 2); 
354 
offset = SAMPLES_BUF_SIZE  512;

355 
} 
356 
} 
357 
s>samples_offset[ch] = offset; 
358  
359 
// print_pow(s>sb_samples, 1152);

360 
} 
361  
362 
static void compute_scale_factors(unsigned char scale_code[SBLIMIT], 
363 
unsigned char scale_factors[SBLIMIT][3], 
364 
int sb_samples[3][12][SBLIMIT], 
365 
int sblimit)

366 
{ 
367 
int *p, vmax, v, n, i, j, k, code;

368 
int index, d1, d2;

369 
unsigned char *sf = &scale_factors[0][0]; 
370 

371 
for(j=0;j<sblimit;j++) { 
372 
for(i=0;i<3;i++) { 
373 
/* find the max absolute value */

374 
p = &sb_samples[i][0][j];

375 
vmax = abs(*p); 
376 
for(k=1;k<12;k++) { 
377 
p += SBLIMIT; 
378 
v = abs(*p); 
379 
if (v > vmax)

380 
vmax = v; 
381 
} 
382 
/* compute the scale factor index using log 2 computations */

383 
if (vmax > 0) { 
384 
n = av_log2(vmax); 
385 
/* n is the position of the MSB of vmax. now

386 
use at most 2 compares to find the index */

387 
index = (21  n) * 3  3; 
388 
if (index >= 0) { 
389 
while (vmax <= scale_factor_table[index+1]) 
390 
index++; 
391 
} else {

392 
index = 0; /* very unlikely case of overflow */ 
393 
} 
394 
} else {

395 
index = 62; /* value 63 is not allowed */ 
396 
} 
397  
398 
#if 0

399 
printf("%2d:%d in=%x %x %d\n",

400 
j, i, vmax, scale_factor_table[index], index);

401 
#endif

402 
/* store the scale factor */

403 
assert(index >=0 && index <= 63); 
404 
sf[i] = index; 
405 
} 
406  
407 
/* compute the transmission factor : look if the scale factors

408 
are close enough to each other */

409 
d1 = scale_diff_table[sf[0]  sf[1] + 64]; 
410 
d2 = scale_diff_table[sf[1]  sf[2] + 64]; 
411 

412 
/* handle the 25 cases */

413 
switch(d1 * 5 + d2) { 
414 
case 0*5+0: 
415 
case 0*5+4: 
416 
case 3*5+4: 
417 
case 4*5+0: 
418 
case 4*5+4: 
419 
code = 0;

420 
break;

421 
case 0*5+1: 
422 
case 0*5+2: 
423 
case 4*5+1: 
424 
case 4*5+2: 
425 
code = 3;

426 
sf[2] = sf[1]; 
427 
break;

428 
case 0*5+3: 
429 
case 4*5+3: 
430 
code = 3;

431 
sf[1] = sf[2]; 
432 
break;

433 
case 1*5+0: 
434 
case 1*5+4: 
435 
case 2*5+4: 
436 
code = 1;

437 
sf[1] = sf[0]; 
438 
break;

439 
case 1*5+1: 
440 
case 1*5+2: 
441 
case 2*5+0: 
442 
case 2*5+1: 
443 
case 2*5+2: 
444 
code = 2;

445 
sf[1] = sf[2] = sf[0]; 
446 
break;

447 
case 2*5+3: 
448 
case 3*5+3: 
449 
code = 2;

450 
sf[0] = sf[1] = sf[2]; 
451 
break;

452 
case 3*5+0: 
453 
case 3*5+1: 
454 
case 3*5+2: 
455 
code = 2;

456 
sf[0] = sf[2] = sf[1]; 
457 
break;

458 
case 1*5+3: 
459 
code = 2;

460 
if (sf[0] > sf[2]) 
461 
sf[0] = sf[2]; 
462 
sf[1] = sf[2] = sf[0]; 
463 
break;

464 
default:

465 
av_abort(); 
466 
} 
467 

468 
#if 0

469 
printf("%d: %2d %2d %2d %d %d > %d\n", j,

470 
sf[0], sf[1], sf[2], d1, d2, code);

471 
#endif

472 
scale_code[j] = code; 
473 
sf += 3;

474 
} 
475 
} 
476  
477 
/* The most important function : psycho acoustic module. In this

478 
encoder there is basically none, so this is the worst you can do,

479 
but also this is the simpler. */

480 
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) 
481 
{ 
482 
int i;

483  
484 
for(i=0;i<s>sblimit;i++) { 
485 
smr[i] = (int)(fixed_smr[i] * 10); 
486 
} 
487 
} 
488  
489  
490 
#define SB_NOTALLOCATED 0 
491 
#define SB_ALLOCATED 1 
492 
#define SB_NOMORE 2 
493  
494 
/* Try to maximize the smr while using a number of bits inferior to

495 
the frame size. I tried to make the code simpler, faster and

496 
smaller than other encoders :) */

497 
static void compute_bit_allocation(MpegAudioContext *s, 
498 
short smr1[MPA_MAX_CHANNELS][SBLIMIT],

499 
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 
500 
int *padding)

501 
{ 
502 
int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;

503 
int incr;

504 
short smr[MPA_MAX_CHANNELS][SBLIMIT];

505 
unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; 
506 
const unsigned char *alloc; 
507  
508 
memcpy(smr, smr1, s>nb_channels * sizeof(short) * SBLIMIT); 
509 
memset(subband_status, SB_NOTALLOCATED, s>nb_channels * SBLIMIT); 
510 
memset(bit_alloc, 0, s>nb_channels * SBLIMIT);

511 

512 
/* compute frame size and padding */

513 
max_frame_size = s>frame_size; 
514 
s>frame_frac += s>frame_frac_incr; 
515 
if (s>frame_frac >= 65536) { 
516 
s>frame_frac = 65536;

517 
s>do_padding = 1;

518 
max_frame_size += 8;

519 
} else {

520 
s>do_padding = 0;

521 
} 
522  
523 
/* compute the header + bit alloc size */

524 
current_frame_size = 32;

525 
alloc = s>alloc_table; 
526 
for(i=0;i<s>sblimit;i++) { 
527 
incr = alloc[0];

528 
current_frame_size += incr * s>nb_channels; 
529 
alloc += 1 << incr;

530 
} 
531 
for(;;) {

532 
/* look for the subband with the largest signal to mask ratio */

533 
max_sb = 1;

534 
max_ch = 1;

535 
max_smr = 0x80000000;

536 
for(ch=0;ch<s>nb_channels;ch++) { 
537 
for(i=0;i<s>sblimit;i++) { 
538 
if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {

539 
max_smr = smr[ch][i]; 
540 
max_sb = i; 
541 
max_ch = ch; 
542 
} 
543 
} 
544 
} 
545 
#if 0

546 
printf("current=%d max=%d max_sb=%d alloc=%d\n",

547 
current_frame_size, max_frame_size, max_sb,

548 
bit_alloc[max_sb]);

549 
#endif

550 
if (max_sb < 0) 
551 
break;

552 

553 
/* find alloc table entry (XXX: not optimal, should use

554 
pointer table) */

555 
alloc = s>alloc_table; 
556 
for(i=0;i<max_sb;i++) { 
557 
alloc += 1 << alloc[0]; 
558 
} 
559  
560 
if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {

561 
/* nothing was coded for this band: add the necessary bits */

562 
incr = 2 + nb_scale_factors[s>scale_code[max_ch][max_sb]] * 6; 
563 
incr += total_quant_bits[alloc[1]];

564 
} else {

565 
/* increments bit allocation */

566 
b = bit_alloc[max_ch][max_sb]; 
567 
incr = total_quant_bits[alloc[b + 1]] 

568 
total_quant_bits[alloc[b]]; 
569 
} 
570  
571 
if (current_frame_size + incr <= max_frame_size) {

572 
/* can increase size */

573 
b = ++bit_alloc[max_ch][max_sb]; 
574 
current_frame_size += incr; 
575 
/* decrease smr by the resolution we added */

576 
smr[max_ch][max_sb] = smr1[max_ch][max_sb]  quant_snr[alloc[b]]; 
577 
/* max allocation size reached ? */

578 
if (b == ((1 << alloc[0])  1)) 
579 
subband_status[max_ch][max_sb] = SB_NOMORE; 
580 
else

581 
subband_status[max_ch][max_sb] = SB_ALLOCATED; 
582 
} else {

583 
/* cannot increase the size of this subband */

584 
subband_status[max_ch][max_sb] = SB_NOMORE; 
585 
} 
586 
} 
587 
*padding = max_frame_size  current_frame_size; 
588 
assert(*padding >= 0);

589  
590 
#if 0

591 
for(i=0;i<s>sblimit;i++) {

592 
printf("%d ", bit_alloc[i]);

593 
}

594 
printf("\n");

595 
#endif

596 
} 
597  
598 
/*

599 
* Output the mpeg audio layer 2 frame. Note how the code is small

600 
* compared to other encoders :)

601 
*/

602 
static void encode_frame(MpegAudioContext *s, 
603 
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 
604 
int padding)

605 
{ 
606 
int i, j, k, l, bit_alloc_bits, b, ch;

607 
unsigned char *sf; 
608 
int q[3]; 
609 
PutBitContext *p = &s>pb; 
610  
611 
/* header */

612  
613 
put_bits(p, 12, 0xfff); 
614 
put_bits(p, 1, 1  s>lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ 
615 
put_bits(p, 2, 42); /* layer 2 */ 
616 
put_bits(p, 1, 1); /* no error protection */ 
617 
put_bits(p, 4, s>bitrate_index);

618 
put_bits(p, 2, s>freq_index);

619 
put_bits(p, 1, s>do_padding); /* use padding */ 
620 
put_bits(p, 1, 0); /* private_bit */ 
621 
put_bits(p, 2, s>nb_channels == 2 ? MPA_STEREO : MPA_MONO); 
622 
put_bits(p, 2, 0); /* mode_ext */ 
623 
put_bits(p, 1, 0); /* no copyright */ 
624 
put_bits(p, 1, 1); /* original */ 
625 
put_bits(p, 2, 0); /* no emphasis */ 
626  
627 
/* bit allocation */

628 
j = 0;

629 
for(i=0;i<s>sblimit;i++) { 
630 
bit_alloc_bits = s>alloc_table[j]; 
631 
for(ch=0;ch<s>nb_channels;ch++) { 
632 
put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); 
633 
} 
634 
j += 1 << bit_alloc_bits;

635 
} 
636 

637 
/* scale codes */

638 
for(i=0;i<s>sblimit;i++) { 
639 
for(ch=0;ch<s>nb_channels;ch++) { 
640 
if (bit_alloc[ch][i])

641 
put_bits(p, 2, s>scale_code[ch][i]);

642 
} 
643 
} 
644  
645 
/* scale factors */

646 
for(i=0;i<s>sblimit;i++) { 
647 
for(ch=0;ch<s>nb_channels;ch++) { 
648 
if (bit_alloc[ch][i]) {

649 
sf = &s>scale_factors[ch][i][0];

650 
switch(s>scale_code[ch][i]) {

651 
case 0: 
652 
put_bits(p, 6, sf[0]); 
653 
put_bits(p, 6, sf[1]); 
654 
put_bits(p, 6, sf[2]); 
655 
break;

656 
case 3: 
657 
case 1: 
658 
put_bits(p, 6, sf[0]); 
659 
put_bits(p, 6, sf[2]); 
660 
break;

661 
case 2: 
662 
put_bits(p, 6, sf[0]); 
663 
break;

664 
} 
665 
} 
666 
} 
667 
} 
668 

669 
/* quantization & write sub band samples */

670  
671 
for(k=0;k<3;k++) { 
672 
for(l=0;l<12;l+=3) { 
673 
j = 0;

674 
for(i=0;i<s>sblimit;i++) { 
675 
bit_alloc_bits = s>alloc_table[j]; 
676 
for(ch=0;ch<s>nb_channels;ch++) { 
677 
b = bit_alloc[ch][i]; 
678 
if (b) {

679 
int qindex, steps, m, sample, bits;

680 
/* we encode 3 sub band samples of the same sub band at a time */

681 
qindex = s>alloc_table[j+b]; 
682 
steps = quant_steps[qindex]; 
683 
for(m=0;m<3;m++) { 
684 
sample = s>sb_samples[ch][k][l + m][i]; 
685 
/* divide by scale factor */

686 
#ifdef USE_FLOATS

687 
{ 
688 
float a;

689 
a = (float)sample * scale_factor_inv_table[s>scale_factors[ch][i][k]];

690 
q[m] = (int)((a + 1.0) * steps * 0.5); 
691 
} 
692 
#else

693 
{ 
694 
int q1, e, shift, mult;

695 
e = s>scale_factors[ch][i][k]; 
696 
shift = scale_factor_shift[e]; 
697 
mult = scale_factor_mult[e]; 
698 

699 
/* normalize to P bits */

700 
if (shift < 0) 
701 
q1 = sample << (shift); 
702 
else

703 
q1 = sample >> shift; 
704 
q1 = (q1 * mult) >> P; 
705 
q[m] = ((q1 + (1 << P)) * steps) >> (P + 1); 
706 
} 
707 
#endif

708 
if (q[m] >= steps)

709 
q[m] = steps  1;

710 
assert(q[m] >= 0 && q[m] < steps);

711 
} 
712 
bits = quant_bits[qindex]; 
713 
if (bits < 0) { 
714 
/* group the 3 values to save bits */

715 
put_bits(p, bits, 
716 
q[0] + steps * (q[1] + steps * q[2])); 
717 
#if 0

718 
printf("%d: gr1 %d\n",

719 
i, q[0] + steps * (q[1] + steps * q[2]));

720 
#endif

721 
} else {

722 
#if 0

723 
printf("%d: gr3 %d %d %d\n",

724 
i, q[0], q[1], q[2]);

725 
#endif

726 
put_bits(p, bits, q[0]);

727 
put_bits(p, bits, q[1]);

728 
put_bits(p, bits, q[2]);

729 
} 
730 
} 
731 
} 
732 
/* next subband in alloc table */

733 
j += 1 << bit_alloc_bits;

734 
} 
735 
} 
736 
} 
737  
738 
/* padding */

739 
for(i=0;i<padding;i++) 
740 
put_bits(p, 1, 0); 
741  
742 
/* flush */

743 
flush_put_bits(p); 
744 
} 
745  
746 
static int MPA_encode_frame(AVCodecContext *avctx, 
747 
unsigned char *frame, int buf_size, void *data) 
748 
{ 
749 
MpegAudioContext *s = avctx>priv_data; 
750 
short *samples = data;

751 
short smr[MPA_MAX_CHANNELS][SBLIMIT];

752 
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; 
753 
int padding, i;

754  
755 
for(i=0;i<s>nb_channels;i++) { 
756 
filter(s, i, samples + i, s>nb_channels); 
757 
} 
758  
759 
for(i=0;i<s>nb_channels;i++) { 
760 
compute_scale_factors(s>scale_code[i], s>scale_factors[i], 
761 
s>sb_samples[i], s>sblimit); 
762 
} 
763 
for(i=0;i<s>nb_channels;i++) { 
764 
psycho_acoustic_model(s, smr[i]); 
765 
} 
766 
compute_bit_allocation(s, smr, bit_alloc, &padding); 
767  
768 
init_put_bits(&s>pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL); 
769  
770 
encode_frame(s, bit_alloc, padding); 
771 

772 
s>nb_samples += MPA_FRAME_SIZE; 
773 
return pbBufPtr(&s>pb)  s>pb.buf;

774 
} 
775  
776 
static int MPA_encode_close(AVCodecContext *avctx) 
777 
{ 
778 
av_freep(&avctx>coded_frame); 
779 
return 0; 
780 
} 
781  
782 
AVCodec mp2_encoder = { 
783 
"mp2",

784 
CODEC_TYPE_AUDIO, 
785 
CODEC_ID_MP2, 
786 
sizeof(MpegAudioContext),

787 
MPA_encode_init, 
788 
MPA_encode_frame, 
789 
MPA_encode_close, 
790 
NULL,

791 
}; 
792  
793 
#undef FIX
