blob: cc4f381606ad35c011ef89e11c7cf22b2e5e780b
1 | /* |
2 | * Real Audio 1.0 (14.4K) encoder |
3 | * Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it> |
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 | * Real Audio 1.0 (14.4K) encoder |
25 | * @author Francesco Lavra <francescolavra@interfree.it> |
26 | */ |
27 | |
28 | #include <float.h> |
29 | |
30 | #include "avcodec.h" |
31 | #include "audio_frame_queue.h" |
32 | #include "celp_filters.h" |
33 | #include "internal.h" |
34 | #include "mathops.h" |
35 | #include "put_bits.h" |
36 | #include "ra144.h" |
37 | |
38 | static av_cold int ra144_encode_close(AVCodecContext *avctx) |
39 | { |
40 | RA144Context *ractx = avctx->priv_data; |
41 | ff_lpc_end(&ractx->lpc_ctx); |
42 | ff_af_queue_close(&ractx->afq); |
43 | return 0; |
44 | } |
45 | |
46 | |
47 | static av_cold int ra144_encode_init(AVCodecContext * avctx) |
48 | { |
49 | RA144Context *ractx; |
50 | int ret; |
51 | |
52 | if (avctx->channels != 1) { |
53 | av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", |
54 | avctx->channels); |
55 | return -1; |
56 | } |
57 | avctx->frame_size = NBLOCKS * BLOCKSIZE; |
58 | avctx->initial_padding = avctx->frame_size; |
59 | avctx->bit_rate = 8000; |
60 | ractx = avctx->priv_data; |
61 | ractx->lpc_coef[0] = ractx->lpc_tables[0]; |
62 | ractx->lpc_coef[1] = ractx->lpc_tables[1]; |
63 | ractx->avctx = avctx; |
64 | ff_audiodsp_init(&ractx->adsp); |
65 | ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER, |
66 | FF_LPC_TYPE_LEVINSON); |
67 | if (ret < 0) |
68 | goto error; |
69 | |
70 | ff_af_queue_init(avctx, &ractx->afq); |
71 | |
72 | return 0; |
73 | error: |
74 | ra144_encode_close(avctx); |
75 | return ret; |
76 | } |
77 | |
78 | |
79 | /** |
80 | * Quantize a value by searching a sorted table for the element with the |
81 | * nearest value |
82 | * |
83 | * @param value value to quantize |
84 | * @param table array containing the quantization table |
85 | * @param size size of the quantization table |
86 | * @return index of the quantization table corresponding to the element with the |
87 | * nearest value |
88 | */ |
89 | static int quantize(int value, const int16_t *table, unsigned int size) |
90 | { |
91 | unsigned int low = 0, high = size - 1; |
92 | |
93 | while (1) { |
94 | int index = (low + high) >> 1; |
95 | int error = table[index] - value; |
96 | |
97 | if (index == low) |
98 | return table[high] + error > value ? low : high; |
99 | if (error > 0) { |
100 | high = index; |
101 | } else { |
102 | low = index; |
103 | } |
104 | } |
105 | } |
106 | |
107 | |
108 | /** |
109 | * Orthogonalize a vector to another vector |
110 | * |
111 | * @param v vector to orthogonalize |
112 | * @param u vector against which orthogonalization is performed |
113 | */ |
114 | static void orthogonalize(float *v, const float *u) |
115 | { |
116 | int i; |
117 | float num = 0, den = 0; |
118 | |
119 | for (i = 0; i < BLOCKSIZE; i++) { |
120 | num += v[i] * u[i]; |
121 | den += u[i] * u[i]; |
122 | } |
123 | num /= den; |
124 | for (i = 0; i < BLOCKSIZE; i++) |
125 | v[i] -= num * u[i]; |
126 | } |
127 | |
128 | |
129 | /** |
130 | * Calculate match score and gain of an LPC-filtered vector with respect to |
131 | * input data, possibly orthogonalizing it to up to two other vectors. |
132 | * |
133 | * @param work array used to calculate the filtered vector |
134 | * @param coefs coefficients of the LPC filter |
135 | * @param vect original vector |
136 | * @param ortho1 first vector against which orthogonalization is performed |
137 | * @param ortho2 second vector against which orthogonalization is performed |
138 | * @param data input data |
139 | * @param score pointer to variable where match score is returned |
140 | * @param gain pointer to variable where gain is returned |
141 | */ |
142 | static void get_match_score(float *work, const float *coefs, float *vect, |
143 | const float *ortho1, const float *ortho2, |
144 | const float *data, float *score, float *gain) |
145 | { |
146 | float c, g; |
147 | int i; |
148 | |
149 | ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER); |
150 | if (ortho1) |
151 | orthogonalize(work, ortho1); |
152 | if (ortho2) |
153 | orthogonalize(work, ortho2); |
154 | c = g = 0; |
155 | for (i = 0; i < BLOCKSIZE; i++) { |
156 | g += work[i] * work[i]; |
157 | c += data[i] * work[i]; |
158 | } |
159 | if (c <= 0) { |
160 | *score = 0; |
161 | return; |
162 | } |
163 | *gain = c / g; |
164 | *score = *gain * c; |
165 | } |
166 | |
167 | |
168 | /** |
169 | * Create a vector from the adaptive codebook at a given lag value |
170 | * |
171 | * @param vect array where vector is stored |
172 | * @param cb adaptive codebook |
173 | * @param lag lag value |
174 | */ |
175 | static void create_adapt_vect(float *vect, const int16_t *cb, int lag) |
176 | { |
177 | int i; |
178 | |
179 | cb += BUFFERSIZE - lag; |
180 | for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++) |
181 | vect[i] = cb[i]; |
182 | if (lag < BLOCKSIZE) |
183 | for (i = 0; i < BLOCKSIZE - lag; i++) |
184 | vect[lag + i] = cb[i]; |
185 | } |
186 | |
187 | |
188 | /** |
189 | * Search the adaptive codebook for the best entry and gain and remove its |
190 | * contribution from input data |
191 | * |
192 | * @param adapt_cb array from which the adaptive codebook is extracted |
193 | * @param work array used to calculate LPC-filtered vectors |
194 | * @param coefs coefficients of the LPC filter |
195 | * @param data input data |
196 | * @return index of the best entry of the adaptive codebook |
197 | */ |
198 | static int adaptive_cb_search(const int16_t *adapt_cb, float *work, |
199 | const float *coefs, float *data) |
200 | { |
201 | int i, av_uninit(best_vect); |
202 | float score, gain, best_score, av_uninit(best_gain); |
203 | float exc[BLOCKSIZE]; |
204 | |
205 | gain = best_score = 0; |
206 | for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) { |
207 | create_adapt_vect(exc, adapt_cb, i); |
208 | get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain); |
209 | if (score > best_score) { |
210 | best_score = score; |
211 | best_vect = i; |
212 | best_gain = gain; |
213 | } |
214 | } |
215 | if (!best_score) |
216 | return 0; |
217 | |
218 | /** |
219 | * Re-calculate the filtered vector from the vector with maximum match score |
220 | * and remove its contribution from input data. |
221 | */ |
222 | create_adapt_vect(exc, adapt_cb, best_vect); |
223 | ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER); |
224 | for (i = 0; i < BLOCKSIZE; i++) |
225 | data[i] -= best_gain * work[i]; |
226 | return best_vect - BLOCKSIZE / 2 + 1; |
227 | } |
228 | |
229 | |
230 | /** |
231 | * Find the best vector of a fixed codebook by applying an LPC filter to |
232 | * codebook entries, possibly orthogonalizing them to up to two other vectors |
233 | * and matching the results with input data. |
234 | * |
235 | * @param work array used to calculate the filtered vectors |
236 | * @param coefs coefficients of the LPC filter |
237 | * @param cb fixed codebook |
238 | * @param ortho1 first vector against which orthogonalization is performed |
239 | * @param ortho2 second vector against which orthogonalization is performed |
240 | * @param data input data |
241 | * @param idx pointer to variable where the index of the best codebook entry is |
242 | * returned |
243 | * @param gain pointer to variable where the gain of the best codebook entry is |
244 | * returned |
245 | */ |
246 | static void find_best_vect(float *work, const float *coefs, |
247 | const int8_t cb[][BLOCKSIZE], const float *ortho1, |
248 | const float *ortho2, float *data, int *idx, |
249 | float *gain) |
250 | { |
251 | int i, j; |
252 | float g, score, best_score; |
253 | float vect[BLOCKSIZE]; |
254 | |
255 | *idx = *gain = best_score = 0; |
256 | for (i = 0; i < FIXED_CB_SIZE; i++) { |
257 | for (j = 0; j < BLOCKSIZE; j++) |
258 | vect[j] = cb[i][j]; |
259 | get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g); |
260 | if (score > best_score) { |
261 | best_score = score; |
262 | *idx = i; |
263 | *gain = g; |
264 | } |
265 | } |
266 | } |
267 | |
268 | |
269 | /** |
270 | * Search the two fixed codebooks for the best entry and gain |
271 | * |
272 | * @param work array used to calculate LPC-filtered vectors |
273 | * @param coefs coefficients of the LPC filter |
274 | * @param data input data |
275 | * @param cba_idx index of the best entry of the adaptive codebook |
276 | * @param cb1_idx pointer to variable where the index of the best entry of the |
277 | * first fixed codebook is returned |
278 | * @param cb2_idx pointer to variable where the index of the best entry of the |
279 | * second fixed codebook is returned |
280 | */ |
281 | static void fixed_cb_search(float *work, const float *coefs, float *data, |
282 | int cba_idx, int *cb1_idx, int *cb2_idx) |
283 | { |
284 | int i, ortho_cb1; |
285 | float gain; |
286 | float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE]; |
287 | float vect[BLOCKSIZE]; |
288 | |
289 | /** |
290 | * The filtered vector from the adaptive codebook can be retrieved from |
291 | * work, because this function is called just after adaptive_cb_search(). |
292 | */ |
293 | if (cba_idx) |
294 | memcpy(cba_vect, work, sizeof(cba_vect)); |
295 | |
296 | find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL, |
297 | data, cb1_idx, &gain); |
298 | |
299 | /** |
300 | * Re-calculate the filtered vector from the vector with maximum match score |
301 | * and remove its contribution from input data. |
302 | */ |
303 | if (gain) { |
304 | for (i = 0; i < BLOCKSIZE; i++) |
305 | vect[i] = ff_cb1_vects[*cb1_idx][i]; |
306 | ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER); |
307 | if (cba_idx) |
308 | orthogonalize(work, cba_vect); |
309 | for (i = 0; i < BLOCKSIZE; i++) |
310 | data[i] -= gain * work[i]; |
311 | memcpy(cb1_vect, work, sizeof(cb1_vect)); |
312 | ortho_cb1 = 1; |
313 | } else |
314 | ortho_cb1 = 0; |
315 | |
316 | find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL, |
317 | ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain); |
318 | } |
319 | |
320 | |
321 | /** |
322 | * Encode a subblock of the current frame |
323 | * |
324 | * @param ractx encoder context |
325 | * @param sblock_data input data of the subblock |
326 | * @param lpc_coefs coefficients of the LPC filter |
327 | * @param rms RMS of the reflection coefficients |
328 | * @param pb pointer to PutBitContext of the current frame |
329 | */ |
330 | static void ra144_encode_subblock(RA144Context *ractx, |
331 | const int16_t *sblock_data, |
332 | const int16_t *lpc_coefs, unsigned int rms, |
333 | PutBitContext *pb) |
334 | { |
335 | float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE]; |
336 | float coefs[LPC_ORDER]; |
337 | float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE]; |
338 | int cba_idx, cb1_idx, cb2_idx, gain; |
339 | int i, n; |
340 | unsigned m[3]; |
341 | float g[3]; |
342 | float error, best_error; |
343 | |
344 | for (i = 0; i < LPC_ORDER; i++) { |
345 | work[i] = ractx->curr_sblock[BLOCKSIZE + i]; |
346 | coefs[i] = lpc_coefs[i] * (1/4096.0); |
347 | } |
348 | |
349 | /** |
350 | * Calculate the zero-input response of the LPC filter and subtract it from |
351 | * input data. |
352 | */ |
353 | ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE, |
354 | LPC_ORDER); |
355 | for (i = 0; i < BLOCKSIZE; i++) { |
356 | zero[i] = work[LPC_ORDER + i]; |
357 | data[i] = sblock_data[i] - zero[i]; |
358 | } |
359 | |
360 | /** |
361 | * Codebook search is performed without taking into account the contribution |
362 | * of the previous subblock, since it has been just subtracted from input |
363 | * data. |
364 | */ |
365 | memset(work, 0, LPC_ORDER * sizeof(*work)); |
366 | |
367 | cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs, |
368 | data); |
369 | if (cba_idx) { |
370 | /** |
371 | * The filtered vector from the adaptive codebook can be retrieved from |
372 | * work, see implementation of adaptive_cb_search(). |
373 | */ |
374 | memcpy(cba, work + LPC_ORDER, sizeof(cba)); |
375 | |
376 | ff_copy_and_dup(ractx->buffer_a, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1); |
377 | m[0] = (ff_irms(&ractx->adsp, ractx->buffer_a) * rms) >> 12; |
378 | } |
379 | fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx); |
380 | for (i = 0; i < BLOCKSIZE; i++) { |
381 | cb1[i] = ff_cb1_vects[cb1_idx][i]; |
382 | cb2[i] = ff_cb2_vects[cb2_idx][i]; |
383 | } |
384 | ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb1, BLOCKSIZE, |
385 | LPC_ORDER); |
386 | memcpy(cb1, work + LPC_ORDER, sizeof(cb1)); |
387 | m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8; |
388 | ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb2, BLOCKSIZE, |
389 | LPC_ORDER); |
390 | memcpy(cb2, work + LPC_ORDER, sizeof(cb2)); |
391 | m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8; |
392 | best_error = FLT_MAX; |
393 | gain = 0; |
394 | for (n = 0; n < 256; n++) { |
395 | g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) * |
396 | (1/4096.0); |
397 | g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) * |
398 | (1/4096.0); |
399 | error = 0; |
400 | if (cba_idx) { |
401 | g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) * |
402 | (1/4096.0); |
403 | for (i = 0; i < BLOCKSIZE; i++) { |
404 | data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] + |
405 | g[2] * cb2[i]; |
406 | error += (data[i] - sblock_data[i]) * |
407 | (data[i] - sblock_data[i]); |
408 | } |
409 | } else { |
410 | for (i = 0; i < BLOCKSIZE; i++) { |
411 | data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i]; |
412 | error += (data[i] - sblock_data[i]) * |
413 | (data[i] - sblock_data[i]); |
414 | } |
415 | } |
416 | if (error < best_error) { |
417 | best_error = error; |
418 | gain = n; |
419 | } |
420 | } |
421 | put_bits(pb, 7, cba_idx); |
422 | put_bits(pb, 8, gain); |
423 | put_bits(pb, 7, cb1_idx); |
424 | put_bits(pb, 7, cb2_idx); |
425 | ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms, |
426 | gain); |
427 | } |
428 | |
429 | |
430 | static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, |
431 | const AVFrame *frame, int *got_packet_ptr) |
432 | { |
433 | static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4}; |
434 | static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; |
435 | RA144Context *ractx = avctx->priv_data; |
436 | PutBitContext pb; |
437 | int32_t lpc_data[NBLOCKS * BLOCKSIZE]; |
438 | int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER]; |
439 | int shift[LPC_ORDER]; |
440 | int16_t block_coefs[NBLOCKS][LPC_ORDER]; |
441 | int lpc_refl[LPC_ORDER]; /**< reflection coefficients of the frame */ |
442 | unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */ |
443 | const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL; |
444 | int energy = 0; |
445 | int i, idx, ret; |
446 | |
447 | if (ractx->last_frame) |
448 | return 0; |
449 | |
450 | if ((ret = ff_alloc_packet2(avctx, avpkt, FRAME_SIZE, 0)) < 0) |
451 | return ret; |
452 | |
453 | /** |
454 | * Since the LPC coefficients are calculated on a frame centered over the |
455 | * fourth subframe, to encode a given frame, data from the next frame is |
456 | * needed. In each call to this function, the previous frame (whose data are |
457 | * saved in the encoder context) is encoded, and data from the current frame |
458 | * are saved in the encoder context to be used in the next function call. |
459 | */ |
460 | for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) { |
461 | lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i]; |
462 | energy += (lpc_data[i] * lpc_data[i]) >> 4; |
463 | } |
464 | if (frame) { |
465 | int j; |
466 | for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) { |
467 | lpc_data[i] = samples[j] >> 2; |
468 | energy += (lpc_data[i] * lpc_data[i]) >> 4; |
469 | } |
470 | } |
471 | if (i < NBLOCKS * BLOCKSIZE) |
472 | memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data)); |
473 | energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab, |
474 | 32)]; |
475 | |
476 | ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER, |
477 | LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON, |
478 | 0, ORDER_METHOD_EST, 0, 12, 0); |
479 | for (i = 0; i < LPC_ORDER; i++) |
480 | block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] << |
481 | (12 - shift[LPC_ORDER - 1])); |
482 | |
483 | /** |
484 | * TODO: apply perceptual weighting of the input speech through bandwidth |
485 | * expansion of the LPC filter. |
486 | */ |
487 | |
488 | if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) { |
489 | /** |
490 | * The filter is unstable: use the coefficients of the previous frame. |
491 | */ |
492 | ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]); |
493 | if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) { |
494 | /* the filter is still unstable. set reflection coeffs to zero. */ |
495 | memset(lpc_refl, 0, sizeof(lpc_refl)); |
496 | } |
497 | } |
498 | init_put_bits(&pb, avpkt->data, avpkt->size); |
499 | for (i = 0; i < LPC_ORDER; i++) { |
500 | idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]); |
501 | put_bits(&pb, bit_sizes[i], idx); |
502 | lpc_refl[i] = ff_lpc_refl_cb[i][idx]; |
503 | } |
504 | ractx->lpc_refl_rms[0] = ff_rms(lpc_refl); |
505 | ff_eval_coefs(ractx->lpc_coef[0], lpc_refl); |
506 | refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy); |
507 | refl_rms[1] = ff_interp(ractx, block_coefs[1], 2, |
508 | energy <= ractx->old_energy, |
509 | ff_t_sqrt(energy * ractx->old_energy) >> 12); |
510 | refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy); |
511 | refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy); |
512 | ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]); |
513 | put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32)); |
514 | for (i = 0; i < NBLOCKS; i++) |
515 | ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE, |
516 | block_coefs[i], refl_rms[i], &pb); |
517 | flush_put_bits(&pb); |
518 | ractx->old_energy = energy; |
519 | ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0]; |
520 | FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]); |
521 | |
522 | /* copy input samples to current block for processing in next call */ |
523 | i = 0; |
524 | if (frame) { |
525 | for (; i < frame->nb_samples; i++) |
526 | ractx->curr_block[i] = samples[i] >> 2; |
527 | |
528 | if ((ret = ff_af_queue_add(&ractx->afq, frame)) < 0) |
529 | return ret; |
530 | } else |
531 | ractx->last_frame = 1; |
532 | memset(&ractx->curr_block[i], 0, |
533 | (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block)); |
534 | |
535 | /* Get the next frame pts/duration */ |
536 | ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts, |
537 | &avpkt->duration); |
538 | |
539 | avpkt->size = FRAME_SIZE; |
540 | *got_packet_ptr = 1; |
541 | return 0; |
542 | } |
543 | |
544 | |
545 | AVCodec ff_ra_144_encoder = { |
546 | .name = "real_144", |
547 | .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"), |
548 | .type = AVMEDIA_TYPE_AUDIO, |
549 | .id = AV_CODEC_ID_RA_144, |
550 | .priv_data_size = sizeof(RA144Context), |
551 | .init = ra144_encode_init, |
552 | .encode2 = ra144_encode_frame, |
553 | .close = ra144_encode_close, |
554 | .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME, |
555 | .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, |
556 | AV_SAMPLE_FMT_NONE }, |
557 | .supported_samplerates = (const int[]){ 8000, 0 }, |
558 | .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO, 0 }, |
559 | }; |
560 |