ffmpeg / libavcodec / mpegaudio.c @ ea937d01
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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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*/

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#include "avcodec.h" 
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#include "mpegaudio.h" 
21  
22 
/* currently, cannot change these constants (need to modify

23 
quantization stage) */

24 
#define FRAC_BITS 15 
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#define WFRAC_BITS 14 
26 
#define MUL(a,b) (((INT64)(a) * (INT64)(b)) >> FRAC_BITS)

27 
#define FIX(a) ((int)((a) * (1 << FRAC_BITS))) 
28  
29 
#define SAMPLES_BUF_SIZE 4096 
30  
31 
typedef struct MpegAudioContext { 
32 
PutBitContext pb; 
33 
int nb_channels;

34 
int freq, bit_rate;

35 
int lsf; /* 1 if mpeg2 low bitrate selected */ 
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int bitrate_index; /* bit rate */ 
37 
int freq_index;

38 
int frame_size; /* frame size, in bits, without padding */ 
39 
INT64 nb_samples; /* total number of samples encoded */

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/* padding computation */

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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]; 
45 
unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ 
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/* code to group 3 scale factors */

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unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; 
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int sblimit; /* number of used subbands */ 
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const unsigned char *alloc_table; 
50 
} MpegAudioContext; 
51  
52 
/* define it to use floats in quantization (I don't like floats !) */

53 
//#define USE_FLOATS

54  
55 
#include "mpegaudiotab.h" 
56  
57 
int MPA_encode_init(AVCodecContext *avctx)

58 
{ 
59 
MpegAudioContext *s = avctx>priv_data; 
60 
int freq = avctx>sample_rate;

61 
int bitrate = avctx>bit_rate;

62 
int channels = avctx>channels;

63 
int i, v, table;

64 
float a;

65  
66 
if (channels > 2) 
67 
return 1; 
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bitrate = bitrate / 1000;

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s>nb_channels = channels; 
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s>freq = freq; 
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s>bit_rate = bitrate * 1000;

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avctx>frame_size = MPA_FRAME_SIZE; 
73  
74 
/* encoding freq */

75 
s>lsf = 0;

76 
for(i=0;i<3;i++) { 
77 
if (mpa_freq_tab[i] == freq)

78 
break;

79 
if ((mpa_freq_tab[i] / 2) == freq) { 
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s>lsf = 1;

81 
break;

82 
} 
83 
} 
84 
if (i == 3) 
85 
return 1; 
86 
s>freq_index = i; 
87  
88 
/* encoding bitrate & frequency */

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for(i=0;i<15;i++) { 
90 
if (mpa_bitrate_tab[s>lsf][1][i] == bitrate) 
91 
break;

92 
} 
93 
if (i == 15) 
94 
return 1; 
95 
s>bitrate_index = i; 
96  
97 
/* compute total header size & pad bit */

98 

99 
a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); 
100 
s>frame_size = ((int)a) * 8; 
101  
102 
/* frame fractional size to compute padding */

103 
s>frame_frac = 0;

104 
s>frame_frac_incr = (int)((a  floor(a)) * 65536.0); 
105 

106 
/* select the right allocation table */

107 
table = l2_select_table(bitrate, s>nb_channels, freq, s>lsf); 
108  
109 
/* number of used subbands */

110 
s>sblimit = sblimit_table[table]; 
111 
s>alloc_table = alloc_tables[table]; 
112  
113 
#ifdef DEBUG

114 
printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",

115 
bitrate, freq, s>frame_size, table, s>frame_frac_incr); 
116 
#endif

117  
118 
for(i=0;i<s>nb_channels;i++) 
119 
s>samples_offset[i] = 0;

120  
121 
for(i=0;i<257;i++) { 
122 
int v;

123 
v = mpa_enwindow[i]; 
124 
#if WFRAC_BITS != 16 
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v = (v + (1 << (16  WFRAC_BITS  1))) >> (16  WFRAC_BITS); 
126 
#endif

127 
filter_bank[i] = v; 
128 
if ((i & 63) != 0) 
129 
v = v; 
130 
if (i != 0) 
131 
filter_bank[512  i] = v;

132 
} 
133  
134 
for(i=0;i<64;i++) { 
135 
v = (int)(pow(2.0, (3  i) / 3.0) * (1 << 20)); 
136 
if (v <= 0) 
137 
v = 1;

138 
scale_factor_table[i] = v; 
139 
#ifdef USE_FLOATS

140 
scale_factor_inv_table[i] = pow(2.0, (3  i) / 3.0) / (float)(1 << 20); 
141 
#else

142 
#define P 15 
143 
scale_factor_shift[i] = 21  P  (i / 3); 
144 
scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0); 
145 
#endif

146 
} 
147 
for(i=0;i<128;i++) { 
148 
v = i  64;

149 
if (v <= 3) 
150 
v = 0;

151 
else if (v < 0) 
152 
v = 1;

153 
else if (v == 0) 
154 
v = 2;

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

157 
else

158 
v = 4;

159 
scale_diff_table[i] = v; 
160 
} 
161  
162 
for(i=0;i<17;i++) { 
163 
v = quant_bits[i]; 
164 
if (v < 0) 
165 
v = v; 
166 
else

167 
v = v * 3;

168 
total_quant_bits[i] = 12 * v;

169 
} 
170  
171 
avctx>coded_frame= avcodec_alloc_frame(); 
172 
avctx>coded_frame>key_frame= 1;

173  
174 
return 0; 
175 
} 
176  
177 
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */

178 
static void idct32(int *out, int *tab) 
179 
{ 
180 
int i, j;

181 
int *t, *t1, xr;

182 
const int *xp = costab32; 
183  
184 
for(j=31;j>=3;j=2) tab[j] += tab[j  2]; 
185 

186 
t = tab + 30;

187 
t1 = tab + 2;

188 
do {

189 
t[0] += t[4]; 
190 
t[1] += t[1  4]; 
191 
t = 4;

192 
} while (t != t1);

193  
194 
t = tab + 28;

195 
t1 = tab + 4;

196 
do {

197 
t[0] += t[8]; 
198 
t[1] += t[18]; 
199 
t[2] += t[28]; 
200 
t[3] += t[38]; 
201 
t = 8;

202 
} while (t != t1);

203 

204 
t = tab; 
205 
t1 = tab + 32;

206 
do {

207 
t[ 3] = t[ 3]; 
208 
t[ 6] = t[ 6]; 
209 

210 
t[11] = t[11]; 
211 
t[12] = t[12]; 
212 
t[13] = t[13]; 
213 
t[15] = t[15]; 
214 
t += 16;

215 
} while (t != t1);

216  
217 

218 
t = tab; 
219 
t1 = tab + 8;

220 
do {

221 
int x1, x2, x3, x4;

222 

223 
x3 = MUL(t[16], FIX(SQRT2*0.5)); 
224 
x4 = t[0]  x3;

225 
x3 = t[0] + x3;

226 

227 
x2 = MUL((t[24] + t[8]), FIX(SQRT2*0.5)); 
228 
x1 = MUL((t[8]  x2), xp[0]); 
229 
x2 = MUL((t[8] + x2), xp[1]); 
230  
231 
t[ 0] = x3 + x1;

232 
t[ 8] = x4  x2;

233 
t[16] = x4 + x2;

234 
t[24] = x3  x1;

235 
t++; 
236 
} while (t != t1);

237  
238 
xp += 2;

239 
t = tab; 
240 
t1 = tab + 4;

241 
do {

242 
xr = MUL(t[28],xp[0]); 
243 
t[28] = (t[0]  xr); 
244 
t[0] = (t[0] + xr); 
245  
246 
xr = MUL(t[4],xp[1]); 
247 
t[ 4] = (t[24]  xr); 
248 
t[24] = (t[24] + xr); 
249 

250 
xr = MUL(t[20],xp[2]); 
251 
t[20] = (t[8]  xr); 
252 
t[ 8] = (t[8] + xr); 
253 

254 
xr = MUL(t[12],xp[3]); 
255 
t[12] = (t[16]  xr); 
256 
t[16] = (t[16] + xr); 
257 
t++; 
258 
} while (t != t1);

259 
xp += 4;

260  
261 
for (i = 0; i < 4; i++) { 
262 
xr = MUL(tab[30i*4],xp[0]); 
263 
tab[30i*4] = (tab[i*4]  xr); 
264 
tab[ i*4] = (tab[i*4] + xr); 
265 

266 
xr = MUL(tab[ 2+i*4],xp[1]); 
267 
tab[ 2+i*4] = (tab[28i*4]  xr); 
268 
tab[28i*4] = (tab[28i*4] + xr); 
269 

270 
xr = MUL(tab[31i*4],xp[0]); 
271 
tab[31i*4] = (tab[1+i*4]  xr); 
272 
tab[ 1+i*4] = (tab[1+i*4] + xr); 
273 

274 
xr = MUL(tab[ 3+i*4],xp[1]); 
275 
tab[ 3+i*4] = (tab[29i*4]  xr); 
276 
tab[29i*4] = (tab[29i*4] + xr); 
277 

278 
xp += 2;

279 
} 
280  
281 
t = tab + 30;

282 
t1 = tab + 1;

283 
do {

284 
xr = MUL(t1[0], *xp);

285 
t1[0] = (t[0]  xr); 
286 
t[0] = (t[0] + xr); 
287 
t = 2;

288 
t1 += 2;

289 
xp++; 
290 
} while (t >= tab);

291  
292 
for(i=0;i<32;i++) { 
293 
out[i] = tab[bitinv32[i]]; 
294 
} 
295 
} 
296  
297 
#define WSHIFT (WFRAC_BITS + 15  FRAC_BITS) 
298  
299 
static void filter(MpegAudioContext *s, int ch, short *samples, int incr) 
300 
{ 
301 
short *p, *q;

302 
int sum, offset, i, j;

303 
int tmp[64]; 
304 
int tmp1[32]; 
305 
int *out;

306  
307 
// print_pow1(samples, 1152);

308  
309 
offset = s>samples_offset[ch]; 
310 
out = &s>sb_samples[ch][0][0][0]; 
311 
for(j=0;j<36;j++) { 
312 
/* 32 samples at once */

313 
for(i=0;i<32;i++) { 
314 
s>samples_buf[ch][offset + (31  i)] = samples[0]; 
315 
samples += incr; 
316 
} 
317  
318 
/* filter */

319 
p = s>samples_buf[ch] + offset; 
320 
q = filter_bank; 
321 
/* maxsum = 23169 */

322 
for(i=0;i<64;i++) { 
323 
sum = p[0*64] * q[0*64]; 
324 
sum += p[1*64] * q[1*64]; 
325 
sum += p[2*64] * q[2*64]; 
326 
sum += p[3*64] * q[3*64]; 
327 
sum += p[4*64] * q[4*64]; 
328 
sum += p[5*64] * q[5*64]; 
329 
sum += p[6*64] * q[6*64]; 
330 
sum += p[7*64] * q[7*64]; 
331 
tmp[i] = sum; 
332 
p++; 
333 
q++; 
334 
} 
335 
tmp1[0] = tmp[16] >> WSHIFT; 
336 
for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16i]) >> WSHIFT; 
337 
for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]tmp[80i]) >> WSHIFT; 
338  
339 
idct32(out, tmp1); 
340  
341 
/* advance of 32 samples */

342 
offset = 32;

343 
out += 32;

344 
/* handle the wrap around */

345 
if (offset < 0) { 
346 
memmove(s>samples_buf[ch] + SAMPLES_BUF_SIZE  (512  32), 
347 
s>samples_buf[ch], (512  32) * 2); 
348 
offset = SAMPLES_BUF_SIZE  512;

349 
} 
350 
} 
351 
s>samples_offset[ch] = offset; 
352  
353 
// print_pow(s>sb_samples, 1152);

354 
} 
355  
356 
static void compute_scale_factors(unsigned char scale_code[SBLIMIT], 
357 
unsigned char scale_factors[SBLIMIT][3], 
358 
int sb_samples[3][12][SBLIMIT], 
359 
int sblimit)

360 
{ 
361 
int *p, vmax, v, n, i, j, k, code;

362 
int index, d1, d2;

363 
unsigned char *sf = &scale_factors[0][0]; 
364 

365 
for(j=0;j<sblimit;j++) { 
366 
for(i=0;i<3;i++) { 
367 
/* find the max absolute value */

368 
p = &sb_samples[i][0][j];

369 
vmax = abs(*p); 
370 
for(k=1;k<12;k++) { 
371 
p += SBLIMIT; 
372 
v = abs(*p); 
373 
if (v > vmax)

374 
vmax = v; 
375 
} 
376 
/* compute the scale factor index using log 2 computations */

377 
if (vmax > 0) { 
378 
n = av_log2(vmax); 
379 
/* n is the position of the MSB of vmax. now

380 
use at most 2 compares to find the index */

381 
index = (21  n) * 3  3; 
382 
if (index >= 0) { 
383 
while (vmax <= scale_factor_table[index+1]) 
384 
index++; 
385 
} else {

386 
index = 0; /* very unlikely case of overflow */ 
387 
} 
388 
} else {

389 
index = 62; /* value 63 is not allowed */ 
390 
} 
391  
392 
#if 0

393 
printf("%2d:%d in=%x %x %d\n",

394 
j, i, vmax, scale_factor_table[index], index);

395 
#endif

396 
/* store the scale factor */

397 
assert(index >=0 && index <= 63); 
398 
sf[i] = index; 
399 
} 
400  
401 
/* compute the transmission factor : look if the scale factors

402 
are close enough to each other */

403 
d1 = scale_diff_table[sf[0]  sf[1] + 64]; 
404 
d2 = scale_diff_table[sf[1]  sf[2] + 64]; 
405 

406 
/* handle the 25 cases */

407 
switch(d1 * 5 + d2) { 
408 
case 0*5+0: 
409 
case 0*5+4: 
410 
case 3*5+4: 
411 
case 4*5+0: 
412 
case 4*5+4: 
413 
code = 0;

414 
break;

415 
case 0*5+1: 
416 
case 0*5+2: 
417 
case 4*5+1: 
418 
case 4*5+2: 
419 
code = 3;

420 
sf[2] = sf[1]; 
421 
break;

422 
case 0*5+3: 
423 
case 4*5+3: 
424 
code = 3;

425 
sf[1] = sf[2]; 
426 
break;

427 
case 1*5+0: 
428 
case 1*5+4: 
429 
case 2*5+4: 
430 
code = 1;

431 
sf[1] = sf[0]; 
432 
break;

433 
case 1*5+1: 
434 
case 1*5+2: 
435 
case 2*5+0: 
436 
case 2*5+1: 
437 
case 2*5+2: 
438 
code = 2;

439 
sf[1] = sf[2] = sf[0]; 
440 
break;

441 
case 2*5+3: 
442 
case 3*5+3: 
443 
code = 2;

444 
sf[0] = sf[1] = sf[2]; 
445 
break;

446 
case 3*5+0: 
447 
case 3*5+1: 
448 
case 3*5+2: 
449 
code = 2;

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

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

454 
if (sf[0] > sf[2]) 
455 
sf[0] = sf[2]; 
456 
sf[1] = sf[2] = sf[0]; 
457 
break;

458 
default:

459 
av_abort(); 
460 
} 
461 

462 
#if 0

463 
printf("%d: %2d %2d %2d %d %d > %d\n", j,

464 
sf[0], sf[1], sf[2], d1, d2, code);

465 
#endif

466 
scale_code[j] = code; 
467 
sf += 3;

468 
} 
469 
} 
470  
471 
/* The most important function : psycho acoustic module. In this

472 
encoder there is basically none, so this is the worst you can do,

473 
but also this is the simpler. */

474 
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) 
475 
{ 
476 
int i;

477  
478 
for(i=0;i<s>sblimit;i++) { 
479 
smr[i] = (int)(fixed_smr[i] * 10); 
480 
} 
481 
} 
482  
483  
484 
#define SB_NOTALLOCATED 0 
485 
#define SB_ALLOCATED 1 
486 
#define SB_NOMORE 2 
487  
488 
/* Try to maximize the smr while using a number of bits inferior to

489 
the frame size. I tried to make the code simpler, faster and

490 
smaller than other encoders :) */

491 
static void compute_bit_allocation(MpegAudioContext *s, 
492 
short smr1[MPA_MAX_CHANNELS][SBLIMIT],

493 
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 
494 
int *padding)

495 
{ 
496 
int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;

497 
int incr;

498 
short smr[MPA_MAX_CHANNELS][SBLIMIT];

499 
unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; 
500 
const unsigned char *alloc; 
501  
502 
memcpy(smr, smr1, s>nb_channels * sizeof(short) * SBLIMIT); 
503 
memset(subband_status, SB_NOTALLOCATED, s>nb_channels * SBLIMIT); 
504 
memset(bit_alloc, 0, s>nb_channels * SBLIMIT);

505 

506 
/* compute frame size and padding */

507 
max_frame_size = s>frame_size; 
508 
s>frame_frac += s>frame_frac_incr; 
509 
if (s>frame_frac >= 65536) { 
510 
s>frame_frac = 65536;

511 
s>do_padding = 1;

512 
max_frame_size += 8;

513 
} else {

514 
s>do_padding = 0;

515 
} 
516  
517 
/* compute the header + bit alloc size */

518 
current_frame_size = 32;

519 
alloc = s>alloc_table; 
520 
for(i=0;i<s>sblimit;i++) { 
521 
incr = alloc[0];

522 
current_frame_size += incr * s>nb_channels; 
523 
alloc += 1 << incr;

524 
} 
525 
for(;;) {

526 
/* look for the subband with the largest signal to mask ratio */

527 
max_sb = 1;

528 
max_ch = 1;

529 
max_smr = 0x80000000;

530 
for(ch=0;ch<s>nb_channels;ch++) { 
531 
for(i=0;i<s>sblimit;i++) { 
532 
if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {

533 
max_smr = smr[ch][i]; 
534 
max_sb = i; 
535 
max_ch = ch; 
536 
} 
537 
} 
538 
} 
539 
#if 0

540 
printf("current=%d max=%d max_sb=%d alloc=%d\n",

541 
current_frame_size, max_frame_size, max_sb,

542 
bit_alloc[max_sb]);

543 
#endif

544 
if (max_sb < 0) 
545 
break;

546 

547 
/* find alloc table entry (XXX: not optimal, should use

548 
pointer table) */

549 
alloc = s>alloc_table; 
550 
for(i=0;i<max_sb;i++) { 
551 
alloc += 1 << alloc[0]; 
552 
} 
553  
554 
if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {

555 
/* nothing was coded for this band: add the necessary bits */

556 
incr = 2 + nb_scale_factors[s>scale_code[max_ch][max_sb]] * 6; 
557 
incr += total_quant_bits[alloc[1]];

558 
} else {

559 
/* increments bit allocation */

560 
b = bit_alloc[max_ch][max_sb]; 
561 
incr = total_quant_bits[alloc[b + 1]] 

562 
total_quant_bits[alloc[b]]; 
563 
} 
564  
565 
if (current_frame_size + incr <= max_frame_size) {

566 
/* can increase size */

567 
b = ++bit_alloc[max_ch][max_sb]; 
568 
current_frame_size += incr; 
569 
/* decrease smr by the resolution we added */

570 
smr[max_ch][max_sb] = smr1[max_ch][max_sb]  quant_snr[alloc[b]]; 
571 
/* max allocation size reached ? */

572 
if (b == ((1 << alloc[0])  1)) 
573 
subband_status[max_ch][max_sb] = SB_NOMORE; 
574 
else

575 
subband_status[max_ch][max_sb] = SB_ALLOCATED; 
576 
} else {

577 
/* cannot increase the size of this subband */

578 
subband_status[max_ch][max_sb] = SB_NOMORE; 
579 
} 
580 
} 
581 
*padding = max_frame_size  current_frame_size; 
582 
assert(*padding >= 0);

583  
584 
#if 0

585 
for(i=0;i<s>sblimit;i++) {

586 
printf("%d ", bit_alloc[i]);

587 
}

588 
printf("\n");

589 
#endif

590 
} 
591  
592 
/*

593 
* Output the mpeg audio layer 2 frame. Note how the code is small

594 
* compared to other encoders :)

595 
*/

596 
static void encode_frame(MpegAudioContext *s, 
597 
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 
598 
int padding)

599 
{ 
600 
int i, j, k, l, bit_alloc_bits, b, ch;

601 
unsigned char *sf; 
602 
int q[3]; 
603 
PutBitContext *p = &s>pb; 
604  
605 
/* header */

606  
607 
put_bits(p, 12, 0xfff); 
608 
put_bits(p, 1, 1  s>lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ 
609 
put_bits(p, 2, 42); /* layer 2 */ 
610 
put_bits(p, 1, 1); /* no error protection */ 
611 
put_bits(p, 4, s>bitrate_index);

612 
put_bits(p, 2, s>freq_index);

613 
put_bits(p, 1, s>do_padding); /* use padding */ 
614 
put_bits(p, 1, 0); /* private_bit */ 
615 
put_bits(p, 2, s>nb_channels == 2 ? MPA_STEREO : MPA_MONO); 
616 
put_bits(p, 2, 0); /* mode_ext */ 
617 
put_bits(p, 1, 0); /* no copyright */ 
618 
put_bits(p, 1, 1); /* original */ 
619 
put_bits(p, 2, 0); /* no emphasis */ 
620  
621 
/* bit allocation */

622 
j = 0;

623 
for(i=0;i<s>sblimit;i++) { 
624 
bit_alloc_bits = s>alloc_table[j]; 
625 
for(ch=0;ch<s>nb_channels;ch++) { 
626 
put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); 
627 
} 
628 
j += 1 << bit_alloc_bits;

629 
} 
630 

631 
/* scale codes */

632 
for(i=0;i<s>sblimit;i++) { 
633 
for(ch=0;ch<s>nb_channels;ch++) { 
634 
if (bit_alloc[ch][i])

635 
put_bits(p, 2, s>scale_code[ch][i]);

636 
} 
637 
} 
638  
639 
/* scale factors */

640 
for(i=0;i<s>sblimit;i++) { 
641 
for(ch=0;ch<s>nb_channels;ch++) { 
642 
if (bit_alloc[ch][i]) {

643 
sf = &s>scale_factors[ch][i][0];

644 
switch(s>scale_code[ch][i]) {

645 
case 0: 
646 
put_bits(p, 6, sf[0]); 
647 
put_bits(p, 6, sf[1]); 
648 
put_bits(p, 6, sf[2]); 
649 
break;

650 
case 3: 
651 
case 1: 
652 
put_bits(p, 6, sf[0]); 
653 
put_bits(p, 6, sf[2]); 
654 
break;

655 
case 2: 
656 
put_bits(p, 6, sf[0]); 
657 
break;

658 
} 
659 
} 
660 
} 
661 
} 
662 

663 
/* quantization & write sub band samples */

664  
665 
for(k=0;k<3;k++) { 
666 
for(l=0;l<12;l+=3) { 
667 
j = 0;

668 
for(i=0;i<s>sblimit;i++) { 
669 
bit_alloc_bits = s>alloc_table[j]; 
670 
for(ch=0;ch<s>nb_channels;ch++) { 
671 
b = bit_alloc[ch][i]; 
672 
if (b) {

673 
int qindex, steps, m, sample, bits;

674 
/* we encode 3 sub band samples of the same sub band at a time */

675 
qindex = s>alloc_table[j+b]; 
676 
steps = quant_steps[qindex]; 
677 
for(m=0;m<3;m++) { 
678 
sample = s>sb_samples[ch][k][l + m][i]; 
679 
/* divide by scale factor */

680 
#ifdef USE_FLOATS

681 
{ 
682 
float a;

683 
a = (float)sample * scale_factor_inv_table[s>scale_factors[ch][i][k]];

684 
q[m] = (int)((a + 1.0) * steps * 0.5); 
685 
} 
686 
#else

687 
{ 
688 
int q1, e, shift, mult;

689 
e = s>scale_factors[ch][i][k]; 
690 
shift = scale_factor_shift[e]; 
691 
mult = scale_factor_mult[e]; 
692 

693 
/* normalize to P bits */

694 
if (shift < 0) 
695 
q1 = sample << (shift); 
696 
else

697 
q1 = sample >> shift; 
698 
q1 = (q1 * mult) >> P; 
699 
q[m] = ((q1 + (1 << P)) * steps) >> (P + 1); 
700 
} 
701 
#endif

702 
if (q[m] >= steps)

703 
q[m] = steps  1;

704 
assert(q[m] >= 0 && q[m] < steps);

705 
} 
706 
bits = quant_bits[qindex]; 
707 
if (bits < 0) { 
708 
/* group the 3 values to save bits */

709 
put_bits(p, bits, 
710 
q[0] + steps * (q[1] + steps * q[2])); 
711 
#if 0

712 
printf("%d: gr1 %d\n",

713 
i, q[0] + steps * (q[1] + steps * q[2]));

714 
#endif

715 
} else {

716 
#if 0

717 
printf("%d: gr3 %d %d %d\n",

718 
i, q[0], q[1], q[2]);

719 
#endif

720 
put_bits(p, bits, q[0]);

721 
put_bits(p, bits, q[1]);

722 
put_bits(p, bits, q[2]);

723 
} 
724 
} 
725 
} 
726 
/* next subband in alloc table */

727 
j += 1 << bit_alloc_bits;

728 
} 
729 
} 
730 
} 
731  
732 
/* padding */

733 
for(i=0;i<padding;i++) 
734 
put_bits(p, 1, 0); 
735  
736 
/* flush */

737 
flush_put_bits(p); 
738 
} 
739  
740 
int MPA_encode_frame(AVCodecContext *avctx,

741 
unsigned char *frame, int buf_size, void *data) 
742 
{ 
743 
MpegAudioContext *s = avctx>priv_data; 
744 
short *samples = data;

745 
short smr[MPA_MAX_CHANNELS][SBLIMIT];

746 
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; 
747 
int padding, i;

748  
749 
for(i=0;i<s>nb_channels;i++) { 
750 
filter(s, i, samples + i, s>nb_channels); 
751 
} 
752  
753 
for(i=0;i<s>nb_channels;i++) { 
754 
compute_scale_factors(s>scale_code[i], s>scale_factors[i], 
755 
s>sb_samples[i], s>sblimit); 
756 
} 
757 
for(i=0;i<s>nb_channels;i++) { 
758 
psycho_acoustic_model(s, smr[i]); 
759 
} 
760 
compute_bit_allocation(s, smr, bit_alloc, &padding); 
761  
762 
init_put_bits(&s>pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL); 
763  
764 
encode_frame(s, bit_alloc, padding); 
765 

766 
s>nb_samples += MPA_FRAME_SIZE; 
767 
return pbBufPtr(&s>pb)  s>pb.buf;

768 
} 
769  
770 
static int MPA_encode_close(AVCodecContext *avctx) 
771 
{ 
772 
av_freep(&avctx>coded_frame); 
773 
return 0; 
774 
} 
775  
776 
AVCodec mp2_encoder = { 
777 
"mp2",

778 
CODEC_TYPE_AUDIO, 
779 
CODEC_ID_MP2, 
780 
sizeof(MpegAudioContext),

781 
MPA_encode_init, 
782 
MPA_encode_frame, 
783 
MPA_encode_close, 
784 
NULL,

785 
}; 
786  
787 
#undef FIX
