FFmpeg  4.4.6
ra144enc.c
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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 
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 
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,
67  if (ret < 0)
68  return ret;
69 
70  ff_af_queue_init(avctx, &ractx->afq);
71 
72  return 0;
73 }
74 
75 
76 /**
77  * Quantize a value by searching a sorted table for the element with the
78  * nearest value
79  *
80  * @param value value to quantize
81  * @param table array containing the quantization table
82  * @param size size of the quantization table
83  * @return index of the quantization table corresponding to the element with the
84  * nearest value
85  */
86 static int quantize(int value, const int16_t *table, unsigned int size)
87 {
88  unsigned int low = 0, high = size - 1;
89 
90  while (1) {
91  int index = (low + high) >> 1;
92  int error = table[index] - value;
93 
94  if (index == low)
95  return table[high] + error > value ? low : high;
96  if (error > 0) {
97  high = index;
98  } else {
99  low = index;
100  }
101  }
102 }
103 
104 
105 /**
106  * Orthogonalize a vector to another vector
107  *
108  * @param v vector to orthogonalize
109  * @param u vector against which orthogonalization is performed
110  */
111 static void orthogonalize(float *v, const float *u)
112 {
113  int i;
114  float num = 0, den = 0;
115 
116  for (i = 0; i < BLOCKSIZE; i++) {
117  num += v[i] * u[i];
118  den += u[i] * u[i];
119  }
120  num /= den;
121  for (i = 0; i < BLOCKSIZE; i++)
122  v[i] -= num * u[i];
123 }
124 
125 
126 /**
127  * Calculate match score and gain of an LPC-filtered vector with respect to
128  * input data, possibly orthogonalizing it to up to two other vectors.
129  *
130  * @param work array used to calculate the filtered vector
131  * @param coefs coefficients of the LPC filter
132  * @param vect original vector
133  * @param ortho1 first vector against which orthogonalization is performed
134  * @param ortho2 second vector against which orthogonalization is performed
135  * @param data input data
136  * @param score pointer to variable where match score is returned
137  * @param gain pointer to variable where gain is returned
138  */
139 static void get_match_score(float *work, const float *coefs, float *vect,
140  const float *ortho1, const float *ortho2,
141  const float *data, float *score, float *gain)
142 {
143  float c, g;
144  int i;
145 
146  ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
147  if (ortho1)
148  orthogonalize(work, ortho1);
149  if (ortho2)
150  orthogonalize(work, ortho2);
151  c = g = 0;
152  for (i = 0; i < BLOCKSIZE; i++) {
153  g += work[i] * work[i];
154  c += data[i] * work[i];
155  }
156  if (c <= 0) {
157  *score = 0;
158  return;
159  }
160  *gain = c / g;
161  *score = *gain * c;
162 }
163 
164 
165 /**
166  * Create a vector from the adaptive codebook at a given lag value
167  *
168  * @param vect array where vector is stored
169  * @param cb adaptive codebook
170  * @param lag lag value
171  */
172 static void create_adapt_vect(float *vect, const int16_t *cb, int lag)
173 {
174  int i;
175 
176  cb += BUFFERSIZE - lag;
177  for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++)
178  vect[i] = cb[i];
179  if (lag < BLOCKSIZE)
180  for (i = 0; i < BLOCKSIZE - lag; i++)
181  vect[lag + i] = cb[i];
182 }
183 
184 
185 /**
186  * Search the adaptive codebook for the best entry and gain and remove its
187  * contribution from input data
188  *
189  * @param adapt_cb array from which the adaptive codebook is extracted
190  * @param work array used to calculate LPC-filtered vectors
191  * @param coefs coefficients of the LPC filter
192  * @param data input data
193  * @return index of the best entry of the adaptive codebook
194  */
195 static int adaptive_cb_search(const int16_t *adapt_cb, float *work,
196  const float *coefs, float *data)
197 {
198  int i, av_uninit(best_vect);
199  float score, gain, best_score, av_uninit(best_gain);
200  float exc[BLOCKSIZE];
201 
202  gain = best_score = 0;
203  for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) {
204  create_adapt_vect(exc, adapt_cb, i);
205  get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain);
206  if (score > best_score) {
207  best_score = score;
208  best_vect = i;
209  best_gain = gain;
210  }
211  }
212  if (!best_score)
213  return 0;
214 
215  /**
216  * Re-calculate the filtered vector from the vector with maximum match score
217  * and remove its contribution from input data.
218  */
219  create_adapt_vect(exc, adapt_cb, best_vect);
221  for (i = 0; i < BLOCKSIZE; i++)
222  data[i] -= best_gain * work[i];
223  return best_vect - BLOCKSIZE / 2 + 1;
224 }
225 
226 
227 /**
228  * Find the best vector of a fixed codebook by applying an LPC filter to
229  * codebook entries, possibly orthogonalizing them to up to two other vectors
230  * and matching the results with input data.
231  *
232  * @param work array used to calculate the filtered vectors
233  * @param coefs coefficients of the LPC filter
234  * @param cb fixed codebook
235  * @param ortho1 first vector against which orthogonalization is performed
236  * @param ortho2 second vector against which orthogonalization is performed
237  * @param data input data
238  * @param idx pointer to variable where the index of the best codebook entry is
239  * returned
240  * @param gain pointer to variable where the gain of the best codebook entry is
241  * returned
242  */
243 static void find_best_vect(float *work, const float *coefs,
244  const int8_t cb[][BLOCKSIZE], const float *ortho1,
245  const float *ortho2, float *data, int *idx,
246  float *gain)
247 {
248  int i, j;
249  float g, score, best_score;
250  float vect[BLOCKSIZE];
251 
252  *idx = *gain = best_score = 0;
253  for (i = 0; i < FIXED_CB_SIZE; i++) {
254  for (j = 0; j < BLOCKSIZE; j++)
255  vect[j] = cb[i][j];
256  get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g);
257  if (score > best_score) {
258  best_score = score;
259  *idx = i;
260  *gain = g;
261  }
262  }
263 }
264 
265 
266 /**
267  * Search the two fixed codebooks for the best entry and gain
268  *
269  * @param work array used to calculate LPC-filtered vectors
270  * @param coefs coefficients of the LPC filter
271  * @param data input data
272  * @param cba_idx index of the best entry of the adaptive codebook
273  * @param cb1_idx pointer to variable where the index of the best entry of the
274  * first fixed codebook is returned
275  * @param cb2_idx pointer to variable where the index of the best entry of the
276  * second fixed codebook is returned
277  */
278 static void fixed_cb_search(float *work, const float *coefs, float *data,
279  int cba_idx, int *cb1_idx, int *cb2_idx)
280 {
281  int i, ortho_cb1;
282  float gain;
283  float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE];
284  float vect[BLOCKSIZE];
285 
286  /**
287  * The filtered vector from the adaptive codebook can be retrieved from
288  * work, because this function is called just after adaptive_cb_search().
289  */
290  if (cba_idx)
291  memcpy(cba_vect, work, sizeof(cba_vect));
292 
293  find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL,
294  data, cb1_idx, &gain);
295 
296  /**
297  * Re-calculate the filtered vector from the vector with maximum match score
298  * and remove its contribution from input data.
299  */
300  if (gain) {
301  for (i = 0; i < BLOCKSIZE; i++)
302  vect[i] = ff_cb1_vects[*cb1_idx][i];
303  ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
304  if (cba_idx)
305  orthogonalize(work, cba_vect);
306  for (i = 0; i < BLOCKSIZE; i++)
307  data[i] -= gain * work[i];
308  memcpy(cb1_vect, work, sizeof(cb1_vect));
309  ortho_cb1 = 1;
310  } else
311  ortho_cb1 = 0;
312 
313  find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL,
314  ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);
315 }
316 
317 
318 /**
319  * Encode a subblock of the current frame
320  *
321  * @param ractx encoder context
322  * @param sblock_data input data of the subblock
323  * @param lpc_coefs coefficients of the LPC filter
324  * @param rms RMS of the reflection coefficients
325  * @param pb pointer to PutBitContext of the current frame
326  */
328  const int16_t *sblock_data,
329  const int16_t *lpc_coefs, unsigned int rms,
330  PutBitContext *pb)
331 {
332  float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE];
333  float coefs[LPC_ORDER];
334  float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
335  int cba_idx, cb1_idx, cb2_idx, gain;
336  int i, n;
337  unsigned m[3];
338  float g[3];
339  float error, best_error;
340 
341  for (i = 0; i < LPC_ORDER; i++) {
342  work[i] = ractx->curr_sblock[BLOCKSIZE + i];
343  coefs[i] = lpc_coefs[i] * (1/4096.0);
344  }
345 
346  /**
347  * Calculate the zero-input response of the LPC filter and subtract it from
348  * input data.
349  */
351  LPC_ORDER);
352  for (i = 0; i < BLOCKSIZE; i++) {
353  zero[i] = work[LPC_ORDER + i];
354  data[i] = sblock_data[i] - zero[i];
355  }
356 
357  /**
358  * Codebook search is performed without taking into account the contribution
359  * of the previous subblock, since it has been just subtracted from input
360  * data.
361  */
362  memset(work, 0, LPC_ORDER * sizeof(*work));
363 
364  cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs,
365  data);
366  if (cba_idx) {
367  /**
368  * The filtered vector from the adaptive codebook can be retrieved from
369  * work, see implementation of adaptive_cb_search().
370  */
371  memcpy(cba, work + LPC_ORDER, sizeof(cba));
372 
373  ff_copy_and_dup(ractx->buffer_a, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);
374  m[0] = (ff_irms(&ractx->adsp, ractx->buffer_a) * rms) >> 12;
375  }
376  fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx);
377  for (i = 0; i < BLOCKSIZE; i++) {
378  cb1[i] = ff_cb1_vects[cb1_idx][i];
379  cb2[i] = ff_cb2_vects[cb2_idx][i];
380  }
382  LPC_ORDER);
383  memcpy(cb1, work + LPC_ORDER, sizeof(cb1));
384  m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8;
386  LPC_ORDER);
387  memcpy(cb2, work + LPC_ORDER, sizeof(cb2));
388  m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8;
389  best_error = FLT_MAX;
390  gain = 0;
391  for (n = 0; n < 256; n++) {
392  g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) *
393  (1/4096.0);
394  g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) *
395  (1/4096.0);
396  error = 0;
397  if (cba_idx) {
398  g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) *
399  (1/4096.0);
400  for (i = 0; i < BLOCKSIZE; i++) {
401  data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] +
402  g[2] * cb2[i];
403  error += (data[i] - sblock_data[i]) *
404  (data[i] - sblock_data[i]);
405  }
406  } else {
407  for (i = 0; i < BLOCKSIZE; i++) {
408  data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i];
409  error += (data[i] - sblock_data[i]) *
410  (data[i] - sblock_data[i]);
411  }
412  }
413  if (error < best_error) {
414  best_error = error;
415  gain = n;
416  }
417  }
418  put_bits(pb, 7, cba_idx);
419  put_bits(pb, 8, gain);
420  put_bits(pb, 7, cb1_idx);
421  put_bits(pb, 7, cb2_idx);
422  ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms,
423  gain);
424 }
425 
426 
427 static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
428  const AVFrame *frame, int *got_packet_ptr)
429 {
430  static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
431  static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
432  RA144Context *ractx = avctx->priv_data;
433  PutBitContext pb;
434  int32_t lpc_data[NBLOCKS * BLOCKSIZE];
435  int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER];
436  int shift[LPC_ORDER];
437  int16_t block_coefs[NBLOCKS][LPC_ORDER];
438  int lpc_refl[LPC_ORDER]; /**< reflection coefficients of the frame */
439  unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */
440  const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
441  int energy = 0;
442  int i, idx, ret;
443 
444  if (ractx->last_frame)
445  return 0;
446 
447  if ((ret = ff_alloc_packet2(avctx, avpkt, FRAME_SIZE, 0)) < 0)
448  return ret;
449 
450  /**
451  * Since the LPC coefficients are calculated on a frame centered over the
452  * fourth subframe, to encode a given frame, data from the next frame is
453  * needed. In each call to this function, the previous frame (whose data are
454  * saved in the encoder context) is encoded, and data from the current frame
455  * are saved in the encoder context to be used in the next function call.
456  */
457  for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) {
458  lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
459  energy += (lpc_data[i] * lpc_data[i]) >> 4;
460  }
461  if (frame) {
462  int j;
463  for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) {
464  lpc_data[i] = samples[j] >> 2;
465  energy += (lpc_data[i] * lpc_data[i]) >> 4;
466  }
467  }
468  if (i < NBLOCKS * BLOCKSIZE)
469  memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data));
470  energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,
471  32)];
472 
473  ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
474  LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON,
475  0, ORDER_METHOD_EST, 0, 12, 0);
476  for (i = 0; i < LPC_ORDER; i++)
477  block_coefs[NBLOCKS - 1][i] = -lpc_coefs[LPC_ORDER - 1][i]
478  * (1 << (12 - shift[LPC_ORDER - 1]));
479 
480  /**
481  * TODO: apply perceptual weighting of the input speech through bandwidth
482  * expansion of the LPC filter.
483  */
484 
485  if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
486  /**
487  * The filter is unstable: use the coefficients of the previous frame.
488  */
489  ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]);
490  if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
491  /* the filter is still unstable. set reflection coeffs to zero. */
492  memset(lpc_refl, 0, sizeof(lpc_refl));
493  }
494  }
495  init_put_bits(&pb, avpkt->data, avpkt->size);
496  for (i = 0; i < LPC_ORDER; i++) {
497  idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);
498  put_bits(&pb, bit_sizes[i], idx);
499  lpc_refl[i] = ff_lpc_refl_cb[i][idx];
500  }
501  ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
502  ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
503  refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
504  refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
505  energy <= ractx->old_energy,
506  ff_t_sqrt(energy * ractx->old_energy) >> 12);
507  refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
508  refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
509  ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]);
510  put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32));
511  for (i = 0; i < NBLOCKS; i++)
512  ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE,
513  block_coefs[i], refl_rms[i], &pb);
514  flush_put_bits(&pb);
515  ractx->old_energy = energy;
516  ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
517  FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
518 
519  /* copy input samples to current block for processing in next call */
520  i = 0;
521  if (frame) {
522  for (; i < frame->nb_samples; i++)
523  ractx->curr_block[i] = samples[i] >> 2;
524 
525  if ((ret = ff_af_queue_add(&ractx->afq, frame)) < 0)
526  return ret;
527  } else
528  ractx->last_frame = 1;
529  memset(&ractx->curr_block[i], 0,
530  (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block));
531 
532  /* Get the next frame pts/duration */
533  ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts,
534  &avpkt->duration);
535 
536  avpkt->size = FRAME_SIZE;
537  *got_packet_ptr = 1;
538  return 0;
539 }
540 
541 
543  .name = "real_144",
544  .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
545  .type = AVMEDIA_TYPE_AUDIO,
546  .id = AV_CODEC_ID_RA_144,
547  .priv_data_size = sizeof(RA144Context),
549  .encode2 = ra144_encode_frame,
550  .close = ra144_encode_close,
552  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
554  .supported_samplerates = (const int[]){ 8000, 0 },
555  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO, 0 },
556 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
#define av_uninit(x)
Definition: attributes.h:154
#define av_cold
Definition: attributes.h:88
uint8_t
int32_t
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
#define u(width, name, range_min, range_max)
Definition: cbs_h2645.c:264
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
Definition: celp_filters.c:84
#define FFSWAP(type, a, b)
Definition: common.h:108
#define FFMIN(a, b)
Definition: common.h:105
#define NULL
Definition: coverity.c:32
static AVFrame * frame
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:33
double value
Definition: eval.c:98
#define FRAME_SIZE
#define LPC_ORDER
Definition: g723_1.h:40
#define AV_CH_LAYOUT_MONO
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: codec.h:77
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: codec.h:82
@ AV_CODEC_ID_RA_144
Definition: codec_id.h:410
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
int index
Definition: gxfenc.c:89
static const int sizes[][2]
Definition: img2dec.c:54
int i
Definition: input.c:407
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:218
av_cold void ff_audiodsp_init(AudioDSPContext *c)
Definition: audiodsp.c:106
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
Definition: lpc.c:325
int ff_lpc_calc_coefs(LPCContext *s, const int32_t *samples, int blocksize, int min_order, int max_order, int precision, int32_t coefs[][MAX_LPC_ORDER], int *shift, enum FFLPCType lpc_type, int lpc_passes, int omethod, int min_shift, int max_shift, int zero_shift)
Calculate LPC coefficients for multiple orders.
Definition: lpc.c:201
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
Definition: lpc.c:303
#define ORDER_METHOD_EST
Definition: lpc.h:30
#define MAX_LPC_ORDER
Definition: lpc.h:38
@ FF_LPC_TYPE_LEVINSON
Levinson-Durbin recursion.
Definition: lpc.h:47
const char data[16]
Definition: mxf.c:142
static const uint16_t table[]
Definition: prosumer.c:206
bitstream writer API
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:57
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:110
unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy)
Definition: ra144.c:1678
int ff_irms(AudioDSPContext *adsp, const int16_t *data)
inverse root mean square
Definition: ra144.c:1684
const uint16_t ff_cb1_base[128]
Definition: ra144.c:1402
void ff_eval_coefs(int *coefs, const int *refl)
Evaluate the LPC filter coefficients from the reflection coefficients.
Definition: ra144.c:1593
const int8_t ff_cb1_vects[128][40]
Definition: ra144.c:114
void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain)
Definition: ra144.c:1694
const uint16_t ff_cb2_base[128]
Definition: ra144.c:1421
const int16_t ff_energy_tab[32]
Definition: ra144.c:1440
const int16_t ff_gain_val_tab[256][3]
Definition: ra144.c:28
void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset)
Copy the last offset values of *source to *target.
Definition: ra144.c:1530
int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx)
Evaluate the reflection coefficients from the filter coefficients.
Definition: ra144.c:1545
const int8_t ff_cb2_vects[128][40]
Definition: ra144.c:758
void ff_int_to_int16(int16_t *out, const int *inp)
Definition: ra144.c:1613
unsigned int ff_rms(const int *data)
Definition: ra144.c:1636
int ff_t_sqrt(unsigned int x)
Evaluate sqrt(x << 24).
Definition: ra144.c:1625
const int16_t *const ff_lpc_refl_cb[10]
Definition: ra144.c:1502
const uint8_t ff_gain_exp_tab[256]
Definition: ra144.c:95
int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy)
Definition: ra144.c:1657
#define NBLOCKS
number of subblocks within a block
Definition: ra144.h:33
#define BUFFERSIZE
the size of the adaptive codebook
Definition: ra144.h:35
#define FIXED_CB_SIZE
size of fixed codebooks
Definition: ra144.h:36
#define BLOCKSIZE
subblock size in 16-bit words
Definition: ra144.h:34
static void create_adapt_vect(float *vect, const int16_t *cb, int lag)
Create a vector from the adaptive codebook at a given lag value.
Definition: ra144enc.c:172
static void orthogonalize(float *v, const float *u)
Orthogonalize a vector to another vector.
Definition: ra144enc.c:111
static av_cold int ra144_encode_init(AVCodecContext *avctx)
Definition: ra144enc.c:47
static void fixed_cb_search(float *work, const float *coefs, float *data, int cba_idx, int *cb1_idx, int *cb2_idx)
Search the two fixed codebooks for the best entry and gain.
Definition: ra144enc.c:278
static void ra144_encode_subblock(RA144Context *ractx, const int16_t *sblock_data, const int16_t *lpc_coefs, unsigned int rms, PutBitContext *pb)
Encode a subblock of the current frame.
Definition: ra144enc.c:327
static av_cold int ra144_encode_close(AVCodecContext *avctx)
Definition: ra144enc.c:38
static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: ra144enc.c:427
static int adaptive_cb_search(const int16_t *adapt_cb, float *work, const float *coefs, float *data)
Search the adaptive codebook for the best entry and gain and remove its contribution from input data.
Definition: ra144enc.c:195
AVCodec ff_ra_144_encoder
Definition: ra144enc.c:542
static int quantize(int value, const int16_t *table, unsigned int size)
Quantize a value by searching a sorted table for the element with the nearest value.
Definition: ra144enc.c:86
static void find_best_vect(float *work, const float *coefs, const int8_t cb[][BLOCKSIZE], const float *ortho1, const float *ortho2, float *data, int *idx, float *gain)
Find the best vector of a fixed codebook by applying an LPC filter to codebook entries,...
Definition: ra144enc.c:243
static void get_match_score(float *work, const float *coefs, float *vect, const float *ortho1, const float *ortho2, const float *data, float *score, float *gain)
Calculate match score and gain of an LPC-filtered vector with respect to input data,...
Definition: ra144enc.c:139
#define zero
Definition: regdef.h:64
static int shift(int a, int b)
Definition: sonic.c:82
main external API structure.
Definition: avcodec.h:536
int64_t bit_rate
the average bitrate
Definition: avcodec.h:586
int initial_padding
Audio only.
Definition: avcodec.h:2066
int channels
number of audio channels
Definition: avcodec.h:1197
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1216
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:332
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:387
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:362
uint8_t * data
Definition: packet.h:369
unsigned int lpc_tables[2][10]
Definition: ra144.h:49
int16_t adapt_cb[146+2]
Adaptive codebook, its size is two units bigger to avoid a buffer overflow.
Definition: ra144.h:64
AudioDSPContext adsp
Definition: ra144.h:42
AVCodecContext * avctx
Definition: ra144.h:41
int16_t buffer_a[FFALIGN(BLOCKSIZE, 16)]
Definition: ra144.h:66
unsigned int old_energy
previous frame energy
Definition: ra144.h:47
int last_frame
Definition: ra144.h:45
int16_t curr_block[NBLOCKS *BLOCKSIZE]
Definition: ra144.h:57
unsigned int * lpc_coef[2]
LPC coefficients: lpc_coef[0] is the coefficients of the current frame and lpc_coef[1] of the previou...
Definition: ra144.h:53
AudioFrameQueue afq
Definition: ra144.h:44
LPCContext lpc_ctx
Definition: ra144.h:43
int16_t curr_sblock[50]
The current subblock padded by the last 10 values of the previous one.
Definition: ra144.h:60
unsigned int lpc_refl_rms[2]
Definition: ra144.h:55
#define av_log(a,...)
static void error(const char *err)
int size
const char * g
Definition: vf_curves.c:117
static double cb(void *priv, double x, double y)
Definition: vf_geq.c:215
if(ret< 0)
Definition: vf_mcdeint.c:282
static double c[64]