ffmpeg / libavcodec / mpegaudioenc.c @ 5d6e4c16
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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 file is part of FFmpeg.

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*

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* FFmpeg 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 2.1 of the License, or (at your option) any later version.

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*

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* FFmpeg 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 FFmpeg; if not, write to the Free Software

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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 021101301 USA

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*/

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22 
/**

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* @file

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* The simplest mpeg audio layer 2 encoder.

25 
*/

26  
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#include "avcodec.h" 
28 
#include "put_bits.h" 
29  
30 
#undef CONFIG_MPEGAUDIO_HP

31 
#define CONFIG_MPEGAUDIO_HP 0 
32 
#include "mpegaudio.h" 
33  
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/* currently, cannot change these constants (need to modify

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quantization stage) */

36 
#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)

37  
38 
#define SAMPLES_BUF_SIZE 4096 
39  
40 
typedef struct MpegAudioContext { 
41 
PutBitContext pb; 
42 
int nb_channels;

43 
int lsf; /* 1 if mpeg2 low bitrate selected */ 
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int bitrate_index; /* bit rate */ 
45 
int freq_index;

46 
int frame_size; /* frame size, in bits, without padding */ 
47 
/* padding computation */

48 
int frame_frac, frame_frac_incr, do_padding;

49 
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]; 
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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; 
57 
} MpegAudioContext; 
58  
59 
/* define it to use floats in quantization (I don't like floats !) */

60 
#define USE_FLOATS

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

69 
int bitrate = avctx>bit_rate;

70 
int channels = avctx>channels;

71 
int i, v, table;

72 
float a;

73  
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if (channels <= 0  channels > 2){ 
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av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);

76 
return 1; 
77 
} 
78 
bitrate = bitrate / 1000;

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s>nb_channels = channels; 
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avctx>frame_size = MPA_FRAME_SIZE; 
81  
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/* encoding freq */

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s>lsf = 0;

84 
for(i=0;i<3;i++) { 
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if (ff_mpa_freq_tab[i] == freq)

86 
break;

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

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break;

90 
} 
91 
} 
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if (i == 3){ 
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av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);

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return 1; 
95 
} 
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s>freq_index = i; 
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/* encoding bitrate & frequency */

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

102 
} 
103 
if (i == 15){ 
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av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);

105 
return 1; 
106 
} 
107 
s>bitrate_index = i; 
108  
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/* compute total header size & pad bit */

110  
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a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); 
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s>frame_size = ((int)a) * 8; 
113  
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/* frame fractional size to compute padding */

115 
s>frame_frac = 0;

116 
s>frame_frac_incr = (int)((a  floor(a)) * 65536.0); 
117  
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/* select the right allocation table */

119 
table = ff_mpa_l2_select_table(bitrate, s>nb_channels, freq, s>lsf); 
120  
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/* number of used subbands */

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s>sblimit = ff_mpa_sblimit_table[table]; 
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s>alloc_table = ff_mpa_alloc_tables[table]; 
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dprintf(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",

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bitrate, freq, s>frame_size, table, s>frame_frac_incr); 
127  
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for(i=0;i<s>nb_channels;i++) 
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s>samples_offset[i] = 0;

130  
131 
for(i=0;i<257;i++) { 
132 
int v;

133 
v = ff_mpa_enwindow[i]; 
134 
#if WFRAC_BITS != 16 
135 
v = (v + (1 << (16  WFRAC_BITS  1))) >> (16  WFRAC_BITS); 
136 
#endif

137 
filter_bank[i] = v; 
138 
if ((i & 63) != 0) 
139 
v = v; 
140 
if (i != 0) 
141 
filter_bank[512  i] = v;

142 
} 
143  
144 
for(i=0;i<64;i++) { 
145 
v = (int)(pow(2.0, (3  i) / 3.0) * (1 << 20)); 
146 
if (v <= 0) 
147 
v = 1;

148 
scale_factor_table[i] = v; 
149 
#ifdef USE_FLOATS

150 
scale_factor_inv_table[i] = pow(2.0, (3  i) / 3.0) / (float)(1 << 20); 
151 
#else

152 
#define P 15 
153 
scale_factor_shift[i] = 21  P  (i / 3); 
154 
scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0); 
155 
#endif

156 
} 
157 
for(i=0;i<128;i++) { 
158 
v = i  64;

159 
if (v <= 3) 
160 
v = 0;

161 
else if (v < 0) 
162 
v = 1;

163 
else if (v == 0) 
164 
v = 2;

165 
else if (v < 3) 
166 
v = 3;

167 
else

168 
v = 4;

169 
scale_diff_table[i] = v; 
170 
} 
171  
172 
for(i=0;i<17;i++) { 
173 
v = ff_mpa_quant_bits[i]; 
174 
if (v < 0) 
175 
v = v; 
176 
else

177 
v = v * 3;

178 
total_quant_bits[i] = 12 * v;

179 
} 
180  
181 
avctx>coded_frame= avcodec_alloc_frame(); 
182 
avctx>coded_frame>key_frame= 1;

183  
184 
return 0; 
185 
} 
186  
187 
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */

188 
static void idct32(int *out, int *tab) 
189 
{ 
190 
int i, j;

191 
int *t, *t1, xr;

192 
const int *xp = costab32; 
193  
194 
for(j=31;j>=3;j=2) tab[j] += tab[j  2]; 
195  
196 
t = tab + 30;

197 
t1 = tab + 2;

198 
do {

199 
t[0] += t[4]; 
200 
t[1] += t[1  4]; 
201 
t = 4;

202 
} while (t != t1);

203  
204 
t = tab + 28;

205 
t1 = tab + 4;

206 
do {

207 
t[0] += t[8]; 
208 
t[1] += t[18]; 
209 
t[2] += t[28]; 
210 
t[3] += t[38]; 
211 
t = 8;

212 
} while (t != t1);

213  
214 
t = tab; 
215 
t1 = tab + 32;

216 
do {

217 
t[ 3] = t[ 3]; 
218 
t[ 6] = t[ 6]; 
219  
220 
t[11] = t[11]; 
221 
t[12] = t[12]; 
222 
t[13] = t[13]; 
223 
t[15] = t[15]; 
224 
t += 16;

225 
} while (t != t1);

226  
227  
228 
t = tab; 
229 
t1 = tab + 8;

230 
do {

231 
int x1, x2, x3, x4;

232  
233 
x3 = MUL(t[16], FIX(SQRT2*0.5)); 
234 
x4 = t[0]  x3;

235 
x3 = t[0] + x3;

236  
237 
x2 = MUL((t[24] + t[8]), FIX(SQRT2*0.5)); 
238 
x1 = MUL((t[8]  x2), xp[0]); 
239 
x2 = MUL((t[8] + x2), xp[1]); 
240  
241 
t[ 0] = x3 + x1;

242 
t[ 8] = x4  x2;

243 
t[16] = x4 + x2;

244 
t[24] = x3  x1;

245 
t++; 
246 
} while (t != t1);

247  
248 
xp += 2;

249 
t = tab; 
250 
t1 = tab + 4;

251 
do {

252 
xr = MUL(t[28],xp[0]); 
253 
t[28] = (t[0]  xr); 
254 
t[0] = (t[0] + xr); 
255  
256 
xr = MUL(t[4],xp[1]); 
257 
t[ 4] = (t[24]  xr); 
258 
t[24] = (t[24] + xr); 
259  
260 
xr = MUL(t[20],xp[2]); 
261 
t[20] = (t[8]  xr); 
262 
t[ 8] = (t[8] + xr); 
263  
264 
xr = MUL(t[12],xp[3]); 
265 
t[12] = (t[16]  xr); 
266 
t[16] = (t[16] + xr); 
267 
t++; 
268 
} while (t != t1);

269 
xp += 4;

270  
271 
for (i = 0; i < 4; i++) { 
272 
xr = MUL(tab[30i*4],xp[0]); 
273 
tab[30i*4] = (tab[i*4]  xr); 
274 
tab[ i*4] = (tab[i*4] + xr); 
275  
276 
xr = MUL(tab[ 2+i*4],xp[1]); 
277 
tab[ 2+i*4] = (tab[28i*4]  xr); 
278 
tab[28i*4] = (tab[28i*4] + xr); 
279  
280 
xr = MUL(tab[31i*4],xp[0]); 
281 
tab[31i*4] = (tab[1+i*4]  xr); 
282 
tab[ 1+i*4] = (tab[1+i*4] + xr); 
283  
284 
xr = MUL(tab[ 3+i*4],xp[1]); 
285 
tab[ 3+i*4] = (tab[29i*4]  xr); 
286 
tab[29i*4] = (tab[29i*4] + xr); 
287  
288 
xp += 2;

289 
} 
290  
291 
t = tab + 30;

292 
t1 = tab + 1;

293 
do {

294 
xr = MUL(t1[0], *xp);

295 
t1[0] = (t[0]  xr); 
296 
t[0] = (t[0] + xr); 
297 
t = 2;

298 
t1 += 2;

299 
xp++; 
300 
} while (t >= tab);

301  
302 
for(i=0;i<32;i++) { 
303 
out[i] = tab[bitinv32[i]]; 
304 
} 
305 
} 
306  
307 
#define WSHIFT (WFRAC_BITS + 15  FRAC_BITS) 
308  
309 
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr) 
310 
{ 
311 
short *p, *q;

312 
int sum, offset, i, j;

313 
int tmp[64]; 
314 
int tmp1[32]; 
315 
int *out;

316  
317 
// print_pow1(samples, 1152);

318  
319 
offset = s>samples_offset[ch]; 
320 
out = &s>sb_samples[ch][0][0][0]; 
321 
for(j=0;j<36;j++) { 
322 
/* 32 samples at once */

323 
for(i=0;i<32;i++) { 
324 
s>samples_buf[ch][offset + (31  i)] = samples[0]; 
325 
samples += incr; 
326 
} 
327  
328 
/* filter */

329 
p = s>samples_buf[ch] + offset; 
330 
q = filter_bank; 
331 
/* maxsum = 23169 */

332 
for(i=0;i<64;i++) { 
333 
sum = p[0*64] * q[0*64]; 
334 
sum += p[1*64] * q[1*64]; 
335 
sum += p[2*64] * q[2*64]; 
336 
sum += p[3*64] * q[3*64]; 
337 
sum += p[4*64] * q[4*64]; 
338 
sum += p[5*64] * q[5*64]; 
339 
sum += p[6*64] * q[6*64]; 
340 
sum += p[7*64] * q[7*64]; 
341 
tmp[i] = sum; 
342 
p++; 
343 
q++; 
344 
} 
345 
tmp1[0] = tmp[16] >> WSHIFT; 
346 
for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16i]) >> WSHIFT; 
347 
for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]tmp[80i]) >> WSHIFT; 
348  
349 
idct32(out, tmp1); 
350  
351 
/* advance of 32 samples */

352 
offset = 32;

353 
out += 32;

354 
/* handle the wrap around */

355 
if (offset < 0) { 
356 
memmove(s>samples_buf[ch] + SAMPLES_BUF_SIZE  (512  32), 
357 
s>samples_buf[ch], (512  32) * 2); 
358 
offset = SAMPLES_BUF_SIZE  512;

359 
} 
360 
} 
361 
s>samples_offset[ch] = offset; 
362  
363 
// print_pow(s>sb_samples, 1152);

364 
} 
365  
366 
static void compute_scale_factors(unsigned char scale_code[SBLIMIT], 
367 
unsigned char scale_factors[SBLIMIT][3], 
368 
int sb_samples[3][12][SBLIMIT], 
369 
int sblimit)

370 
{ 
371 
int *p, vmax, v, n, i, j, k, code;

372 
int index, d1, d2;

373 
unsigned char *sf = &scale_factors[0][0]; 
374  
375 
for(j=0;j<sblimit;j++) { 
376 
for(i=0;i<3;i++) { 
377 
/* find the max absolute value */

378 
p = &sb_samples[i][0][j];

379 
vmax = abs(*p); 
380 
for(k=1;k<12;k++) { 
381 
p += SBLIMIT; 
382 
v = abs(*p); 
383 
if (v > vmax)

384 
vmax = v; 
385 
} 
386 
/* compute the scale factor index using log 2 computations */

387 
if (vmax > 1) { 
388 
n = av_log2(vmax); 
389 
/* n is the position of the MSB of vmax. now

390 
use at most 2 compares to find the index */

391 
index = (21  n) * 3  3; 
392 
if (index >= 0) { 
393 
while (vmax <= scale_factor_table[index+1]) 
394 
index++; 
395 
} else {

396 
index = 0; /* very unlikely case of overflow */ 
397 
} 
398 
} else {

399 
index = 62; /* value 63 is not allowed */ 
400 
} 
401  
402 
#if 0

403 
printf("%2d:%d in=%x %x %d\n",

404 
j, i, vmax, scale_factor_table[index], index);

405 
#endif

406 
/* store the scale factor */

407 
assert(index >=0 && index <= 63); 
408 
sf[i] = index; 
409 
} 
410  
411 
/* compute the transmission factor : look if the scale factors

412 
are close enough to each other */

413 
d1 = scale_diff_table[sf[0]  sf[1] + 64]; 
414 
d2 = scale_diff_table[sf[1]  sf[2] + 64]; 
415  
416 
/* handle the 25 cases */

417 
switch(d1 * 5 + d2) { 
418 
case 0*5+0: 
419 
case 0*5+4: 
420 
case 3*5+4: 
421 
case 4*5+0: 
422 
case 4*5+4: 
423 
code = 0;

424 
break;

425 
case 0*5+1: 
426 
case 0*5+2: 
427 
case 4*5+1: 
428 
case 4*5+2: 
429 
code = 3;

430 
sf[2] = sf[1]; 
431 
break;

432 
case 0*5+3: 
433 
case 4*5+3: 
434 
code = 3;

435 
sf[1] = sf[2]; 
436 
break;

437 
case 1*5+0: 
438 
case 1*5+4: 
439 
case 2*5+4: 
440 
code = 1;

441 
sf[1] = sf[0]; 
442 
break;

443 
case 1*5+1: 
444 
case 1*5+2: 
445 
case 2*5+0: 
446 
case 2*5+1: 
447 
case 2*5+2: 
448 
code = 2;

449 
sf[1] = sf[2] = sf[0]; 
450 
break;

451 
case 2*5+3: 
452 
case 3*5+3: 
453 
code = 2;

454 
sf[0] = sf[1] = sf[2]; 
455 
break;

456 
case 3*5+0: 
457 
case 3*5+1: 
458 
case 3*5+2: 
459 
code = 2;

460 
sf[0] = sf[2] = sf[1]; 
461 
break;

462 
case 1*5+3: 
463 
code = 2;

464 
if (sf[0] > sf[2]) 
465 
sf[0] = sf[2]; 
466 
sf[1] = sf[2] = sf[0]; 
467 
break;

468 
default:

469 
assert(0); //cannot happen 
470 
code = 0; /* kill warning */ 
471 
} 
472  
473 
#if 0

474 
printf("%d: %2d %2d %2d %d %d > %d\n", j,

475 
sf[0], sf[1], sf[2], d1, d2, code);

476 
#endif

477 
scale_code[j] = code; 
478 
sf += 3;

479 
} 
480 
} 
481  
482 
/* The most important function : psycho acoustic module. In this

483 
encoder there is basically none, so this is the worst you can do,

484 
but also this is the simpler. */

485 
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) 
486 
{ 
487 
int i;

488  
489 
for(i=0;i<s>sblimit;i++) { 
490 
smr[i] = (int)(fixed_smr[i] * 10); 
491 
} 
492 
} 
493  
494  
495 
#define SB_NOTALLOCATED 0 
496 
#define SB_ALLOCATED 1 
497 
#define SB_NOMORE 2 
498  
499 
/* Try to maximize the smr while using a number of bits inferior to

500 
the frame size. I tried to make the code simpler, faster and

501 
smaller than other encoders :) */

502 
static void compute_bit_allocation(MpegAudioContext *s, 
503 
short smr1[MPA_MAX_CHANNELS][SBLIMIT],

504 
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 
505 
int *padding)

506 
{ 
507 
int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;

508 
int incr;

509 
short smr[MPA_MAX_CHANNELS][SBLIMIT];

510 
unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; 
511 
const unsigned char *alloc; 
512  
513 
memcpy(smr, smr1, s>nb_channels * sizeof(short) * SBLIMIT); 
514 
memset(subband_status, SB_NOTALLOCATED, s>nb_channels * SBLIMIT); 
515 
memset(bit_alloc, 0, s>nb_channels * SBLIMIT);

516  
517 
/* compute frame size and padding */

518 
max_frame_size = s>frame_size; 
519 
s>frame_frac += s>frame_frac_incr; 
520 
if (s>frame_frac >= 65536) { 
521 
s>frame_frac = 65536;

522 
s>do_padding = 1;

523 
max_frame_size += 8;

524 
} else {

525 
s>do_padding = 0;

526 
} 
527  
528 
/* compute the header + bit alloc size */

529 
current_frame_size = 32;

530 
alloc = s>alloc_table; 
531 
for(i=0;i<s>sblimit;i++) { 
532 
incr = alloc[0];

533 
current_frame_size += incr * s>nb_channels; 
534 
alloc += 1 << incr;

535 
} 
536 
for(;;) {

537 
/* look for the subband with the largest signal to mask ratio */

538 
max_sb = 1;

539 
max_ch = 1;

540 
max_smr = INT_MIN; 
541 
for(ch=0;ch<s>nb_channels;ch++) { 
542 
for(i=0;i<s>sblimit;i++) { 
543 
if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {

544 
max_smr = smr[ch][i]; 
545 
max_sb = i; 
546 
max_ch = ch; 
547 
} 
548 
} 
549 
} 
550 
#if 0

551 
printf("current=%d max=%d max_sb=%d alloc=%d\n",

552 
current_frame_size, max_frame_size, max_sb,

553 
bit_alloc[max_sb]);

554 
#endif

555 
if (max_sb < 0) 
556 
break;

557  
558 
/* find alloc table entry (XXX: not optimal, should use

559 
pointer table) */

560 
alloc = s>alloc_table; 
561 
for(i=0;i<max_sb;i++) { 
562 
alloc += 1 << alloc[0]; 
563 
} 
564  
565 
if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {

566 
/* nothing was coded for this band: add the necessary bits */

567 
incr = 2 + nb_scale_factors[s>scale_code[max_ch][max_sb]] * 6; 
568 
incr += total_quant_bits[alloc[1]];

569 
} else {

570 
/* increments bit allocation */

571 
b = bit_alloc[max_ch][max_sb]; 
572 
incr = total_quant_bits[alloc[b + 1]] 

573 
total_quant_bits[alloc[b]]; 
574 
} 
575  
576 
if (current_frame_size + incr <= max_frame_size) {

577 
/* can increase size */

578 
b = ++bit_alloc[max_ch][max_sb]; 
579 
current_frame_size += incr; 
580 
/* decrease smr by the resolution we added */

581 
smr[max_ch][max_sb] = smr1[max_ch][max_sb]  quant_snr[alloc[b]]; 
582 
/* max allocation size reached ? */

583 
if (b == ((1 << alloc[0])  1)) 
584 
subband_status[max_ch][max_sb] = SB_NOMORE; 
585 
else

586 
subband_status[max_ch][max_sb] = SB_ALLOCATED; 
587 
} else {

588 
/* cannot increase the size of this subband */

589 
subband_status[max_ch][max_sb] = SB_NOMORE; 
590 
} 
591 
} 
592 
*padding = max_frame_size  current_frame_size; 
593 
assert(*padding >= 0);

594  
595 
#if 0

596 
for(i=0;i<s>sblimit;i++) {

597 
printf("%d ", bit_alloc[i]);

598 
}

599 
printf("\n");

600 
#endif

601 
} 
602  
603 
/*

604 
* Output the mpeg audio layer 2 frame. Note how the code is small

605 
* compared to other encoders :)

606 
*/

607 
static void encode_frame(MpegAudioContext *s, 
608 
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 
609 
int padding)

610 
{ 
611 
int i, j, k, l, bit_alloc_bits, b, ch;

612 
unsigned char *sf; 
613 
int q[3]; 
614 
PutBitContext *p = &s>pb; 
615  
616 
/* header */

617  
618 
put_bits(p, 12, 0xfff); 
619 
put_bits(p, 1, 1  s>lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ 
620 
put_bits(p, 2, 42); /* layer 2 */ 
621 
put_bits(p, 1, 1); /* no error protection */ 
622 
put_bits(p, 4, s>bitrate_index);

623 
put_bits(p, 2, s>freq_index);

624 
put_bits(p, 1, s>do_padding); /* use padding */ 
625 
put_bits(p, 1, 0); /* private_bit */ 
626 
put_bits(p, 2, s>nb_channels == 2 ? MPA_STEREO : MPA_MONO); 
627 
put_bits(p, 2, 0); /* mode_ext */ 
628 
put_bits(p, 1, 0); /* no copyright */ 
629 
put_bits(p, 1, 1); /* original */ 
630 
put_bits(p, 2, 0); /* no emphasis */ 
631  
632 
/* bit allocation */

633 
j = 0;

634 
for(i=0;i<s>sblimit;i++) { 
635 
bit_alloc_bits = s>alloc_table[j]; 
636 
for(ch=0;ch<s>nb_channels;ch++) { 
637 
put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); 
638 
} 
639 
j += 1 << bit_alloc_bits;

640 
} 
641  
642 
/* scale codes */

643 
for(i=0;i<s>sblimit;i++) { 
644 
for(ch=0;ch<s>nb_channels;ch++) { 
645 
if (bit_alloc[ch][i])

646 
put_bits(p, 2, s>scale_code[ch][i]);

647 
} 
648 
} 
649  
650 
/* scale factors */

651 
for(i=0;i<s>sblimit;i++) { 
652 
for(ch=0;ch<s>nb_channels;ch++) { 
653 
if (bit_alloc[ch][i]) {

654 
sf = &s>scale_factors[ch][i][0];

655 
switch(s>scale_code[ch][i]) {

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

661 
case 3: 
662 
case 1: 
663 
put_bits(p, 6, sf[0]); 
664 
put_bits(p, 6, sf[2]); 
665 
break;

666 
case 2: 
667 
put_bits(p, 6, sf[0]); 
668 
break;

669 
} 
670 
} 
671 
} 
672 
} 
673  
674 
/* quantization & write sub band samples */

675  
676 
for(k=0;k<3;k++) { 
677 
for(l=0;l<12;l+=3) { 
678 
j = 0;

679 
for(i=0;i<s>sblimit;i++) { 
680 
bit_alloc_bits = s>alloc_table[j]; 
681 
for(ch=0;ch<s>nb_channels;ch++) { 
682 
b = bit_alloc[ch][i]; 
683 
if (b) {

684 
int qindex, steps, m, sample, bits;

685 
/* we encode 3 sub band samples of the same sub band at a time */

686 
qindex = s>alloc_table[j+b]; 
687 
steps = ff_mpa_quant_steps[qindex]; 
688 
for(m=0;m<3;m++) { 
689 
sample = s>sb_samples[ch][k][l + m][i]; 
690 
/* divide by scale factor */

691 
#ifdef USE_FLOATS

692 
{ 
693 
float a;

694 
a = (float)sample * scale_factor_inv_table[s>scale_factors[ch][i][k]];

695 
q[m] = (int)((a + 1.0) * steps * 0.5); 
696 
} 
697 
#else

698 
{ 
699 
int q1, e, shift, mult;

700 
e = s>scale_factors[ch][i][k]; 
701 
shift = scale_factor_shift[e]; 
702 
mult = scale_factor_mult[e]; 
703  
704 
/* normalize to P bits */

705 
if (shift < 0) 
706 
q1 = sample << (shift); 
707 
else

708 
q1 = sample >> shift; 
709 
q1 = (q1 * mult) >> P; 
710 
q[m] = ((q1 + (1 << P)) * steps) >> (P + 1); 
711 
} 
712 
#endif

713 
if (q[m] >= steps)

714 
q[m] = steps  1;

715 
assert(q[m] >= 0 && q[m] < steps);

716 
} 
717 
bits = ff_mpa_quant_bits[qindex]; 
718 
if (bits < 0) { 
719 
/* group the 3 values to save bits */

720 
put_bits(p, bits, 
721 
q[0] + steps * (q[1] + steps * q[2])); 
722 
#if 0

723 
printf("%d: gr1 %d\n",

724 
i, q[0] + steps * (q[1] + steps * q[2]));

725 
#endif

726 
} else {

727 
#if 0

728 
printf("%d: gr3 %d %d %d\n",

729 
i, q[0], q[1], q[2]);

730 
#endif

731 
put_bits(p, bits, q[0]);

732 
put_bits(p, bits, q[1]);

733 
put_bits(p, bits, q[2]);

734 
} 
735 
} 
736 
} 
737 
/* next subband in alloc table */

738 
j += 1 << bit_alloc_bits;

739 
} 
740 
} 
741 
} 
742  
743 
/* padding */

744 
for(i=0;i<padding;i++) 
745 
put_bits(p, 1, 0); 
746  
747 
/* flush */

748 
flush_put_bits(p); 
749 
} 
750  
751 
static int MPA_encode_frame(AVCodecContext *avctx, 
752 
unsigned char *frame, int buf_size, void *data) 
753 
{ 
754 
MpegAudioContext *s = avctx>priv_data; 
755 
const short *samples = data; 
756 
short smr[MPA_MAX_CHANNELS][SBLIMIT];

757 
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; 
758 
int padding, i;

759  
760 
for(i=0;i<s>nb_channels;i++) { 
761 
filter(s, i, samples + i, s>nb_channels); 
762 
} 
763  
764 
for(i=0;i<s>nb_channels;i++) { 
765 
compute_scale_factors(s>scale_code[i], s>scale_factors[i], 
766 
s>sb_samples[i], s>sblimit); 
767 
} 
768 
for(i=0;i<s>nb_channels;i++) { 
769 
psycho_acoustic_model(s, smr[i]); 
770 
} 
771 
compute_bit_allocation(s, smr, bit_alloc, &padding); 
772  
773 
init_put_bits(&s>pb, frame, MPA_MAX_CODED_FRAME_SIZE); 
774  
775 
encode_frame(s, bit_alloc, padding); 
776  
777 
return put_bits_ptr(&s>pb)  s>pb.buf;

778 
} 
779  
780 
static av_cold int MPA_encode_close(AVCodecContext *avctx) 
781 
{ 
782 
av_freep(&avctx>coded_frame); 
783 
return 0; 
784 
} 
785  
786 
AVCodec mp2_encoder = { 
787 
"mp2",

788 
AVMEDIA_TYPE_AUDIO, 
789 
CODEC_ID_MP2, 
790 
sizeof(MpegAudioContext),

791 
MPA_encode_init, 
792 
MPA_encode_frame, 
793 
MPA_encode_close, 
794 
NULL,

795 
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, 
796 
.supported_samplerates= (const int[]){44100, 48000, 32000, 22050, 24000, 16000, 0}, 
797 
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),

798 
}; 
799  
800 
#undef FIX
