Libav
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 Libav.
6  *
7  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
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 
40 {
41  RA144Context *ractx = avctx->priv_data;
42  ff_lpc_end(&ractx->lpc_ctx);
43  ff_af_queue_close(&ractx->afq);
44  return 0;
45 }
46 
47 
49 {
50  RA144Context *ractx;
51  int ret;
52 
53  if (avctx->channels != 1) {
54  av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n",
55  avctx->channels);
56  return -1;
57  }
58  avctx->frame_size = NBLOCKS * BLOCKSIZE;
59  avctx->delay = avctx->frame_size;
60  avctx->bit_rate = 8000;
61  ractx = avctx->priv_data;
62  ractx->lpc_coef[0] = ractx->lpc_tables[0];
63  ractx->lpc_coef[1] = ractx->lpc_tables[1];
64  ractx->avctx = avctx;
65  ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER,
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 
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 
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 
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 
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 
198 static int adaptive_cb_search(const int16_t *adapt_cb, float *work,
199  const float *coefs, float *data)
200 {
201  int i, best_vect;
202  float score, gain, best_score, 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 
222  create_adapt_vect(exc, adapt_cb, best_vect);
224  for (i = 0; i < BLOCKSIZE; i++)
225  data[i] -= best_gain * work[i];
226  return best_vect - BLOCKSIZE / 2 + 1;
227 }
228 
229 
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 
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 
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 
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 
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  int16_t cba_vect[BLOCKSIZE];
339  int cba_idx, cb1_idx, cb2_idx, gain;
340  int i, n, 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 
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 
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) {
374  memcpy(cba, work + LPC_ORDER, sizeof(cba));
375 
376  ff_copy_and_dup(cba_vect, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);
377  m[0] = (ff_irms(cba_vect) * 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];
442  unsigned int refl_rms[NBLOCKS];
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_packet(avpkt, FRAMESIZE))) {
451  av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
452  return ret;
453  }
454 
462  for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) {
463  lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
464  energy += (lpc_data[i] * lpc_data[i]) >> 4;
465  }
466  if (frame) {
467  int j;
468  for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) {
469  lpc_data[i] = samples[j] >> 2;
470  energy += (lpc_data[i] * lpc_data[i]) >> 4;
471  }
472  }
473  if (i < NBLOCKS * BLOCKSIZE)
474  memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data));
475  energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,
476  32)];
477 
478  ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
479  LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON,
480  0, ORDER_METHOD_EST, 12, 0);
481  for (i = 0; i < LPC_ORDER; i++)
482  block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] <<
483  (12 - shift[LPC_ORDER - 1]));
484 
490  if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
494  ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]);
495  if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
496  /* the filter is still unstable. set reflection coeffs to zero. */
497  memset(lpc_refl, 0, sizeof(lpc_refl));
498  }
499  }
500  init_put_bits(&pb, avpkt->data, avpkt->size);
501  for (i = 0; i < LPC_ORDER; i++) {
502  idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);
503  put_bits(&pb, bit_sizes[i], idx);
504  lpc_refl[i] = ff_lpc_refl_cb[i][idx];
505  }
506  ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
507  ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
508  refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
509  refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
510  energy <= ractx->old_energy,
511  ff_t_sqrt(energy * ractx->old_energy) >> 12);
512  refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
513  refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
514  ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]);
515  put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32));
516  for (i = 0; i < NBLOCKS; i++)
517  ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE,
518  block_coefs[i], refl_rms[i], &pb);
519  flush_put_bits(&pb);
520  ractx->old_energy = energy;
521  ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
522  FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
523 
524  /* copy input samples to current block for processing in next call */
525  i = 0;
526  if (frame) {
527  for (; i < frame->nb_samples; i++)
528  ractx->curr_block[i] = samples[i] >> 2;
529 
530  if ((ret = ff_af_queue_add(&ractx->afq, frame)) < 0)
531  return ret;
532  } else
533  ractx->last_frame = 1;
534  memset(&ractx->curr_block[i], 0,
535  (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block));
536 
537  /* Get the next frame pts/duration */
538  ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts,
539  &avpkt->duration);
540 
541  avpkt->size = FRAMESIZE;
542  *got_packet_ptr = 1;
543  return 0;
544 }
545 
546 
548  .name = "real_144",
549  .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
550  .type = AVMEDIA_TYPE_AUDIO,
551  .id = AV_CODEC_ID_RA_144,
552  .priv_data_size = sizeof(RA144Context),
554  .encode2 = ra144_encode_frame,
557  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
559 };
unsigned int lpc_tables[2][10]
Definition: ra144.h:44
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:83
int size
int ff_t_sqrt(unsigned int x)
Evaluate sqrt(x << 24).
Definition: ra144.c:1618
This structure describes decoded (raw) audio or video data.
Definition: frame.h:135
const int16_t *const ff_lpc_refl_cb[10]
Definition: ra144.c:1502
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:198
#define MAX_LPC_ORDER
Definition: lpc.h:35
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 max_shift, int zero_shift)
Calculate LPC coefficients for multiple orders.
Definition: lpc.c:169
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:89
int size
Definition: avcodec.h:974
static av_cold int ra144_encode_close(AVCodecContext *avctx)
Definition: ra144enc.c:39
static void orthogonalize(float *v, const float *u)
Orthogonalize a vector to another vector.
Definition: ra144enc.c:114
uint16_t adapt_cb[146+2]
Adaptive codebook, its size is two units bigger to avoid a buffer overflow.
Definition: ra144.h:59
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, possibly othogonalizing it to up to 2 other vectors.
Definition: ra144enc.c:142
AVCodec.
Definition: avcodec.h:2812
AVCodec ff_ra_144_encoder
Definition: ra144enc.c:547
static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: ra144enc.c:430
#define NBLOCKS
number of subblocks within a block
Definition: ra144.h:29
uint8_t
#define av_cold
Definition: attributes.h:66
unsigned int lpc_refl_rms[2]
Definition: ra144.h:50
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
unsigned int ff_rms(const int *data)
Definition: ra144.c:1629
#define FIXED_CB_SIZE
size of fixed codebooks
Definition: ra144.h:32
const char data[16]
Definition: mxf.c:70
uint8_t * data
Definition: avcodec.h:973
const uint16_t ff_cb2_base[128]
Definition: ra144.c:1421
#define FRAMESIZE
size of encoded frame
Definition: ra144.h:33
void ff_subblock_synthesis(RA144Context *ractx, const uint16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain)
Definition: ra144.c:1690
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:991
AVCodecContext * avctx
Definition: ra144.h:37
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
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:48
#define CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: avcodec.h:713
static const int sizes[][2]
Definition: img2dec.c:46
const int8_t ff_cb1_vects[128][40]
Definition: ra144.c:114
sample_fmts
Definition: avconv_filter.c:68
#define CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:718
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:145
g
Definition: yuv2rgb.c:535
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:169
const char * name
Name of the codec implementation.
Definition: avcodec.h:2819
static void put_bits(PutBitContext *s, int n, unsigned int value)
Write up to 31 bits into a bitstream.
Definition: put_bits.h:134
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
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:281
AudioFrameQueue afq
Definition: ra144.h:39
int bit_rate
the average bitrate
Definition: avcodec.h:1114
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
Definition: lpc.c:284
#define FFMIN(a, b)
Definition: common.h:57
int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy)
Definition: ra144.c:1650
int32_t
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_alloc_packet(AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1266
int last_frame
Definition: ra144.h:40
void ff_int_to_int16(int16_t *out, const int *inp)
Definition: ra144.c:1606
const int16_t ff_gain_val_tab[256][3]
Definition: ra144.c:28
if(ac->has_optimized_func)
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1827
NULL
Definition: eval.c:55
#define BLOCKSIZE
subblock size in 16-bit words
Definition: ra144.h:30
void ff_eval_coefs(int *coefs, const int *refl)
Evaluate the LPC filter coefficients from the reflection coefficients.
Definition: ra144.c:1586
Libavcodec external API header.
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:61
AV_SAMPLE_FMT_NONE
Definition: avconv_filter.c:68
main external API structure.
Definition: avcodec.h:1050
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:490
int ff_irms(const int16_t *data)
inverse root mean square
Definition: ra144.c:1677
static av_cold int ra144_encode_init(AVCodecContext *avctx)
Definition: ra144enc.c:48
Levinson-Durbin recursion.
Definition: lpc.h:44
#define ORDER_METHOD_EST
Definition: lpc.h:27
LPCContext lpc_ctx
Definition: ra144.h:38
int index
Definition: gxfenc.c:72
int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx)
Evaluate the reflection coefficients from the filter coefficients.
Definition: ra144.c:1545
#define BUFFERSIZE
the size of the adaptive codebook
Definition: ra144.h:31
unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy)
Definition: ra144.c:1671
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
Definition: lpc.c:262
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:141
#define LPC_ORDER
Definition: g723_1_data.h:36
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, possibly othogonalizing them to up to 2 other vectors and matching the results with input data.
Definition: ra144enc.c:246
const int8_t ff_cb2_vects[128][40]
Definition: ra144.c:758
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:83
unsigned int old_energy
previous frame energy
Definition: ra144.h:42
const int16_t ff_energy_tab[32]
Definition: ra144.c:1440
signed 16 bits
Definition: samplefmt.h:64
const uint8_t ff_gain_exp_tab[256]
Definition: ra144.c:95
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
static av_cold int init(AVCodecParserContext *s)
Definition: h264_parser.c:499
int16_t curr_sblock[50]
The current subblock padded by the last 10 values of the previous one.
Definition: ra144.h:55
void * priv_data
Definition: avcodec.h:1092
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int *duration)
Remove frame(s) from the queue.
int channels
number of audio channels
Definition: avcodec.h:1808
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
const uint16_t ff_cb1_base[128]
Definition: ra144.c:1402
int16_t curr_block[NBLOCKS *BLOCKSIZE]
Definition: ra144.h:52
#define FFSWAP(type, a, b)
Definition: common.h:60
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:175
This structure stores compressed data.
Definition: avcodec.h:950
int delay
Codec delay.
Definition: avcodec.h:1212
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:179
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:966
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:330
bitstream writer API