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sipr.c
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1/*
2 * SIPR / ACELP.NET decoder
3 *
4 * Copyright (c) 2008 Vladimir Voroshilov
5 * Copyright (c) 2009 Vitor Sessak
6 *
7 * This file is part of FFmpeg.
8 *
9 * FFmpeg is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Lesser General Public
11 * License as published by the Free Software Foundation; either
12 * version 2.1 of the License, or (at your option) any later version.
13 *
14 * FFmpeg is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Lesser General Public License for more details.
18 *
19 * You should have received a copy of the GNU Lesser General Public
20 * License along with FFmpeg; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 */
23
24#include <math.h>
25#include <stdint.h>
26#include <string.h>
27
29#include "libavutil/float_dsp.h"
31
32#define BITSTREAM_READER_LE
33#include "avcodec.h"
34#include "codec_internal.h"
35#include "decode.h"
36#include "get_bits.h"
37#include "lsp.h"
38#include "acelp_vectors.h"
39#include "acelp_pitch_delay.h"
40#include "acelp_filters.h"
41#include "celp_filters.h"
42
43#define MAX_SUBFRAME_COUNT 5
44
45#include "sipr.h"
46#include "siprdata.h"
47
48typedef struct SiprModeParam {
49 const char *mode_name;
54
55 /* bitstream parameters */
57 uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor
58
59 /** size in bits of the i-th stage vector of quantizer */
60 uint8_t vq_indexes_bits[5];
61
62 /** size in bits of the adaptive-codebook index for every subframe */
63 uint8_t pitch_delay_bits[5];
64
66 uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes
67 uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes
69
71 [MODE_16k] = {
72 .mode_name = "16k",
73 .bits_per_frame = 160,
74 .subframe_count = SUBFRAME_COUNT_16k,
75 .frames_per_packet = 1,
76 .pitch_sharp_factor = 0.00,
77
78 .number_of_fc_indexes = 10,
79 .ma_predictor_bits = 1,
80 .vq_indexes_bits = {7, 8, 7, 7, 7},
81 .pitch_delay_bits = {9, 6},
82 .gp_index_bits = 4,
83 .fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5},
84 .gc_index_bits = 5
85 },
86
87 [MODE_8k5] = {
88 .mode_name = "8k5",
89 .bits_per_frame = 152,
90 .subframe_count = 3,
91 .frames_per_packet = 1,
92 .pitch_sharp_factor = 0.8,
93
94 .number_of_fc_indexes = 3,
95 .ma_predictor_bits = 0,
96 .vq_indexes_bits = {6, 7, 7, 7, 5},
97 .pitch_delay_bits = {8, 5, 5},
98 .gp_index_bits = 0,
99 .fc_index_bits = {9, 9, 9},
100 .gc_index_bits = 7
101 },
102
103 [MODE_6k5] = {
104 .mode_name = "6k5",
105 .bits_per_frame = 232,
106 .subframe_count = 3,
107 .frames_per_packet = 2,
108 .pitch_sharp_factor = 0.8,
109
110 .number_of_fc_indexes = 3,
111 .ma_predictor_bits = 0,
112 .vq_indexes_bits = {6, 7, 7, 7, 5},
113 .pitch_delay_bits = {8, 5, 5},
114 .gp_index_bits = 0,
115 .fc_index_bits = {5, 5, 5},
116 .gc_index_bits = 7
117 },
118
119 [MODE_5k0] = {
120 .mode_name = "5k0",
121 .bits_per_frame = 296,
122 .subframe_count = 5,
123 .frames_per_packet = 2,
124 .pitch_sharp_factor = 0.85,
125
126 .number_of_fc_indexes = 1,
127 .ma_predictor_bits = 0,
128 .vq_indexes_bits = {6, 7, 7, 7, 5},
129 .pitch_delay_bits = {8, 5, 8, 5, 5},
130 .gp_index_bits = 0,
131 .fc_index_bits = {10},
132 .gc_index_bits = 7
133 }
134};
135
136const float ff_pow_0_5[] = {
137 1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4),
138 1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8),
139 1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12),
140 1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16)
141};
142
143static void dequant(float *out, const int *idx, const float * const cbs[])
144{
145 int i;
146 int stride = 2;
147 int num_vec = 5;
148
149 for (i = 0; i < num_vec; i++)
150 memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float));
151
152}
153
154static void lsf_decode_fp(float *lsfnew, float *lsf_history,
155 const SiprParameters *parm)
156{
157 int i;
158 float lsf_tmp[LP_FILTER_ORDER];
159
160 dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks);
161
162 for (i = 0; i < LP_FILTER_ORDER; i++)
163 lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i];
164
166
167 /* Note that a minimum distance is not enforced between the last value and
168 the previous one, contrary to what is done in ff_acelp_reorder_lsf() */
170 lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI);
171
172 memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history));
173
174 for (i = 0; i < LP_FILTER_ORDER - 1; i++)
175 lsfnew[i] = cos(lsfnew[i]);
176 lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI;
177}
178
179/** Apply pitch lag to the fixed vector (AMR section 6.1.2). */
180static void pitch_sharpening(int pitch_lag_int, float beta,
181 float *fixed_vector)
182{
183 int i;
184
185 for (i = pitch_lag_int; i < SUBFR_SIZE; i++)
186 fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int];
187}
188
189/**
190 * Extract decoding parameters from the input bitstream.
191 * @param parms parameters structure
192 * @param pgb pointer to initialized GetBitContext structure
193 */
195 const SiprModeParam *p)
196{
197 int i, j;
198
199 if (p->ma_predictor_bits)
200 parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits);
201
202 for (i = 0; i < 5; i++)
203 parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]);
204
205 for (i = 0; i < p->subframe_count; i++) {
206 parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]);
207 if (p->gp_index_bits)
208 parms->gp_index[i] = get_bits(pgb, p->gp_index_bits);
209
210 for (j = 0; j < p->number_of_fc_indexes; j++)
211 parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]);
212
213 parms->gc_index[i] = get_bits(pgb, p->gc_index_bits);
214 }
215}
216
217static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
218 int num_subfr)
219{
220 double lsfint[LP_FILTER_ORDER];
221 int i,j;
222 float t, t0 = 1.0 / num_subfr;
223
224 t = t0 * 0.5;
225 for (i = 0; i < num_subfr; i++) {
226 for (j = 0; j < LP_FILTER_ORDER; j++)
227 lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
228
230 Az += LP_FILTER_ORDER;
231 t += t0;
232 }
233}
234
235/**
236 * Evaluate the adaptive impulse response.
237 */
238static void eval_ir(const float *Az, int pitch_lag, float *freq,
239 float pitch_sharp_factor)
240{
241 float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1];
242 int i;
243
244 tmp1[0] = 1.0;
245 for (i = 0; i < LP_FILTER_ORDER; i++) {
246 tmp1[i+1] = Az[i] * ff_pow_0_55[i];
247 tmp2[i ] = Az[i] * ff_pow_0_7 [i];
248 }
249 memset(tmp1 + 11, 0, 37 * sizeof(float));
250
253
254 pitch_sharpening(pitch_lag, pitch_sharp_factor, freq);
255}
256
257/**
258 * Evaluate the convolution of a vector with a sparse vector.
259 */
260static void convolute_with_sparse(float *out, const AMRFixed *pulses,
261 const float *shape, int length)
262{
263 int i, j;
264
265 memset(out, 0, length*sizeof(float));
266 for (i = 0; i < pulses->n; i++)
267 for (j = pulses->x[i]; j < length; j++)
268 out[j] += pulses->y[i] * shape[j - pulses->x[i]];
269}
270
271/**
272 * Apply postfilter, very similar to AMR one.
273 */
274static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
275{
276 float buf[SUBFR_SIZE + LP_FILTER_ORDER];
277 float *pole_out = buf + LP_FILTER_ORDER;
278 float lpc_n[LP_FILTER_ORDER];
279 float lpc_d[LP_FILTER_ORDER];
280 int i;
281
282 for (i = 0; i < LP_FILTER_ORDER; i++) {
283 lpc_d[i] = lpc[i] * ff_pow_0_75[i];
284 lpc_n[i] = lpc[i] * ff_pow_0_5 [i];
285 };
286
287 memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem,
288 LP_FILTER_ORDER*sizeof(float));
289
290 ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE,
292
293 memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
294 LP_FILTER_ORDER*sizeof(float));
295
296 ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE);
297
298 memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0,
299 LP_FILTER_ORDER*sizeof(*pole_out));
300
301 memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
302 LP_FILTER_ORDER*sizeof(*pole_out));
303
304 ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE,
306
307}
308
309static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses,
310 SiprMode mode, int low_gain)
311{
312 int i;
313
314 switch (mode) {
315 case MODE_6k5:
316 for (i = 0; i < 3; i++) {
317 fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i;
318 fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1;
319 }
320 fixed_sparse->n = 3;
321 break;
322 case MODE_8k5:
323 for (i = 0; i < 3; i++) {
324 fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i;
325 fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i;
326
327 fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0;
328
329 fixed_sparse->y[2*i + 1] =
330 (fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ?
331 -fixed_sparse->y[2*i ] : fixed_sparse->y[2*i];
332 }
333
334 fixed_sparse->n = 6;
335 break;
336 case MODE_5k0:
337 default:
338 if (low_gain) {
339 int offset = (pulses[0] & 0x200) ? 2 : 0;
340 int val = pulses[0];
341
342 for (i = 0; i < 3; i++) {
343 int index = (val & 0x7) * 6 + 4 - i*2;
344
345 fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1;
346 fixed_sparse->x[i] = index;
347
348 val >>= 3;
349 }
350 fixed_sparse->n = 3;
351 } else {
352 int pulse_subset = (pulses[0] >> 8) & 1;
353
354 fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset;
355 fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1;
356
357 fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1;
358 fixed_sparse->y[1] = -fixed_sparse->y[0];
359 fixed_sparse->n = 2;
360 }
361 break;
362 }
363}
364
366 float *out_data)
367{
368 int i, j;
369 int subframe_count = modes[ctx->mode].subframe_count;
370 int frame_size = subframe_count * SUBFR_SIZE;
372 float *excitation;
373 float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER];
374 float lsf_new[LP_FILTER_ORDER];
375 float *impulse_response = ir_buf + LP_FILTER_ORDER;
376 float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for
377 // memory alignment
378 int t0_first = 0;
379 AMRFixed fixed_cb;
380
381 memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float));
382 lsf_decode_fp(lsf_new, ctx->lsf_history, params);
383
384 sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count);
385
386 memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float));
387
388 excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL;
389
390 for (i = 0; i < subframe_count; i++) {
391 float *pAz = Az + i*LP_FILTER_ORDER;
392 float fixed_vector[SUBFR_SIZE];
393 int T0,T0_frac;
394 float pitch_gain, gain_code, avg_energy;
395
396 ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i,
397 ctx->mode == MODE_5k0, 6);
398
399 if (i == 0 || (i == 2 && ctx->mode == MODE_5k0))
400 t0_first = T0;
401
402 ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0),
403 ff_b60_sinc, 6,
404 2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER,
405 SUBFR_SIZE);
406
407 decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode,
408 ctx->past_pitch_gain < 0.8);
409
410 eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor);
411
412 convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
413 SUBFR_SIZE);
414
415 avg_energy = (0.01 + ff_scalarproduct_float_c(fixed_vector,
416 fixed_vector,
417 SUBFR_SIZE)) /
419
420 ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
421
422 gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1],
423 avg_energy, ctx->energy_history,
424 34 - 15.0/(0.05*M_LN10/M_LN2),
425 pred);
426
427 ff_weighted_vector_sumf(excitation, excitation, fixed_vector,
428 pitch_gain, gain_code, SUBFR_SIZE);
429
430 pitch_gain *= 0.5 * pitch_gain;
431 pitch_gain = FFMIN(pitch_gain, 0.4);
432
433 ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain;
434 ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain);
435 gain_code *= ctx->gain_mem;
436
437 for (j = 0; j < SUBFR_SIZE; j++)
438 fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j];
439
440 if (ctx->mode == MODE_5k0) {
441 postfilter_5k0(ctx, pAz, fixed_vector);
442
444 pAz, excitation, SUBFR_SIZE,
446 }
447
448 ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector,
450
451 excitation += SUBFR_SIZE;
452 }
453
454 memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER,
455 LP_FILTER_ORDER * sizeof(float));
456
457 if (ctx->mode == MODE_5k0) {
458 for (i = 0; i < subframe_count; i++) {
459 float energy = ff_scalarproduct_float_c(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
460 ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
461 SUBFR_SIZE);
463 &synth[i * SUBFR_SIZE], energy,
464 SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
465 }
466
467 memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
468 LP_FILTER_ORDER*sizeof(float));
469 }
470 memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
471 (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
472
474 (const float[2]) {-1.99997 , 1.000000000},
475 (const float[2]) {-1.93307352, 0.935891986},
476 0.939805806,
477 ctx->highpass_filt_mem,
478 frame_size);
479}
480
482{
483 SiprContext *ctx = avctx->priv_data;
484 int i;
485
486 switch (avctx->block_align) {
487 case 20: ctx->mode = MODE_16k; break;
488 case 19: ctx->mode = MODE_8k5; break;
489 case 29: ctx->mode = MODE_6k5; break;
490 case 37: ctx->mode = MODE_5k0; break;
491 default:
492 if (avctx->bit_rate > 12200) ctx->mode = MODE_16k;
493 else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5;
494 else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5;
495 else ctx->mode = MODE_5k0;
496 av_log(avctx, AV_LOG_WARNING,
497 "Invalid block_align: %d. Mode %s guessed based on bitrate: %"PRId64"\n",
498 avctx->block_align, modes[ctx->mode].mode_name, avctx->bit_rate);
499 }
500
501 av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name);
502
503 if (ctx->mode == MODE_16k) {
505 ctx->decode_frame = ff_sipr_decode_frame_16k;
506 } else {
507 ctx->decode_frame = decode_frame;
508 }
509
510 for (i = 0; i < LP_FILTER_ORDER; i++)
511 ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1));
512
513 for (i = 0; i < 4; i++)
514 ctx->energy_history[i] = -14;
515
519
520 return 0;
521}
522
524 int *got_frame_ptr, AVPacket *avpkt)
525{
526 SiprContext *ctx = avctx->priv_data;
527 const uint8_t *buf=avpkt->data;
528 SiprParameters parm;
529 const SiprModeParam *mode_par = &modes[ctx->mode];
530 GetBitContext gb;
531 float *samples;
532 int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE;
533 int i, ret;
534
535 if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
536 av_log(avctx, AV_LOG_ERROR,
537 "Error processing packet: packet size (%d) too small\n",
538 avpkt->size);
539 return AVERROR_INVALIDDATA;
540 }
541
542 /* get output buffer */
543 frame->nb_samples = mode_par->frames_per_packet * subframe_size *
544 mode_par->subframe_count;
545 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
546 return ret;
547 samples = (float *)frame->data[0];
548
549 init_get_bits(&gb, buf, mode_par->bits_per_frame);
550
551 for (i = 0; i < mode_par->frames_per_packet; i++) {
552 decode_parameters(&parm, &gb, mode_par);
553
554 ctx->decode_frame(ctx, &parm, samples);
555
556 samples += subframe_size * mode_par->subframe_count;
557 }
558
559 *got_frame_ptr = 1;
560
561 return mode_par->bits_per_frame >> 3;
562}
563
565 .p.name = "sipr",
566 CODEC_LONG_NAME("RealAudio SIPR / ACELP.NET"),
567 .p.type = AVMEDIA_TYPE_AUDIO,
568 .p.id = AV_CODEC_ID_SIPR,
569 .priv_data_size = sizeof(SiprContext),
573};
SwsAArch64OpImplParams params
Definition ops.c:51
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n)
Apply an order 2 rational transfer function in-place.
void ff_acelp_interpolatef(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Floating point version of ff_acelp_interpolate()
float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy, float *prediction_error, float energy_mean, const float *pred_table)
Calculate fixed gain (part of section 6.1.3 of AMR spec)
void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index, const int prev_lag_int, const int subframe, int third_as_first, int resolution)
Decode the adaptive codebook index to the integer and fractional parts of the pitch lag for one subfr...
#define PITCH_DELAY_MAX
const float ff_pow_0_7[10]
Table of pow(0.7,n)
void ff_adaptive_gain_control(float *out, const float *in, float speech_energ, int size, float alpha, float *gain_mem)
Adaptive gain control (as used in AMR postfiltering)
const float ff_b60_sinc[61]
b60 hamming windowed sinc function coefficients
const float ff_pow_0_55[10]
Table of pow(0.55,n)
const float ff_pow_0_75[10]
Table of pow(0.75,n)
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length)
float implementation of weighted sum of two vectors.
static double val(void *priv, double ch)
Definition aeval.c:77
const FFCodec ff_sipr_decoder
Definition sipr.c:564
#define LP_FILTER_ORDER
linear predictive coding filter order
Definition amrnbdata.h:53
Libavcodec external API header.
#define i(width, name, range_min, range_max)
Definition cbs_h264.c:63
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP zero synthesis filter.
Public libavutil channel layout APIs header.
@ MODE_COUNT
Definition cinepakenc.c:76
#define FF_CODEC_DECODE_CB(func)
#define CODEC_LONG_NAME(str)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition decode.c:1777
static AVFrame * frame
int(* init)(AVBSFContext *ctx)
Definition dts2pts.c:608
float ff_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors of floats.
static const uint8_t frame_size[4]
Definition g723_1.h:222
static const int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
Definition g723_1.h:260
bitstream reader API header.
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition get_bits.h:337
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition get_bits.h:517
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition codec.h:49
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition codec.h:94
@ AV_CODEC_ID_SIPR
Definition codec_id.h:494
#define AV_CHANNEL_LAYOUT_MONO
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition error.h:61
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition log.h:231
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition log.h:216
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition log.h:210
@ AVMEDIA_TYPE_AUDIO
Definition avutil.h:201
@ AV_SAMPLE_FMT_FLT
float
Definition samplefmt.h:60
int index
Definition gxfenc.c:90
unsigned offset
Definition libaomenc.c:763
static int decode_frame(AVCodecContext *avctx, AVFrame *picture, int *got_frame, AVPacket *avpkt)
Definition 4xm.c:837
#define LSFQ_DIFF_MIN
minimum LSF distance (3.2.4) 0.0391 in Q13
Definition g729dec.c:59
#define av_cold
Definition attributes.h:117
void ff_set_min_dist_lsf(float *lsf, double min_spacing, int size)
Adjust the quantized LSFs so they are increasing and not too close.
Definition lsp.c:55
void ff_amrwb_lsp2lpc(const double *lsp, float *lp, int lp_order)
LSP to LP conversion (5.2.4 of AMR-WB)
Definition lsp.c:181
void ff_sort_nearly_sorted_floats(float *vals, int len)
Sort values in ascending order.
Definition lsp.c:245
#define FFMIN(a, b)
Definition macros.h:49
#define M_LN10
Definition mathematics.h:49
#define M_LN2
Definition mathematics.h:43
#define M_PI
Definition mathematics.h:67
static void lsf_decode_fp(float *lsfnew, float *lsf_history, const SiprParameters *parm)
Definition sipr.c:154
static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
Apply postfilter, very similar to AMR one.
Definition sipr.c:274
const float ff_pow_0_5[]
Definition sipr.c:136
static void pitch_sharpening(int pitch_lag_int, float beta, float *fixed_vector)
Apply pitch lag to the fixed vector (AMR section 6.1.2).
Definition sipr.c:180
static void eval_ir(const float *Az, int pitch_lag, float *freq, float pitch_sharp_factor)
Evaluate the adaptive impulse response.
Definition sipr.c:238
static void decode_frame(SiprContext *ctx, SiprParameters *params, float *out_data)
Definition sipr.c:365
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses, SiprMode mode, int low_gain)
Definition sipr.c:309
static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az, int num_subfr)
Definition sipr.c:217
static const SiprModeParam modes[MODE_COUNT]
Definition sipr.c:70
static int sipr_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition sipr.c:523
static void convolute_with_sparse(float *out, const AMRFixed *pulses, const float *shape, int length)
Evaluate the convolution of a vector with a sparse vector.
Definition sipr.c:260
static av_cold int sipr_decoder_init(AVCodecContext *avctx)
Definition sipr.c:481
static void dequant(float *out, const int *idx, const float *const cbs[])
Definition sipr.c:143
#define MAX_SUBFRAME_COUNT
Definition sipr.c:43
static void decode_parameters(SiprParameters *parms, GetBitContext *pgb, const SiprModeParam *p)
Extract decoding parameters from the input bitstream.
Definition sipr.c:194
SiprMode
Definition sipr.h:47
@ MODE_16k
Definition sipr.h:48
@ MODE_6k5
Definition sipr.h:50
@ MODE_8k5
Definition sipr.h:49
@ MODE_5k0
Definition sipr.h:51
#define L_SUBFR_16k
Definition sipr.h:31
#define L_INTERPOL
Number of past samples needed for excitation interpolation.
Definition sipr.h:40
#define SUBFR_SIZE
Subframe size for all modes except 16k.
Definition sipr.h:43
#define SUBFRAME_COUNT_16k
Definition sipr.h:45
void ff_sipr_init_16k(SiprContext *ctx)
Definition sipr16k.c:271
void ff_sipr_decode_frame_16k(SiprContext *ctx, SiprParameters *params, float *out_data)
Definition sipr16k.c:176
static const float pred[4]
Definition siprdata.h:259
static const float gain_cb[128][2]
Definition siprdata.h:213
static const float *const lsf_codebooks[]
Definition siprdata.h:209
static const float mean_lsf[10]
Definition siprdata.h:27
Sparse representation for the algebraic codebook (fixed) vector.
int x[10]
float y[10]
An AVChannelLayout holds information about the channel layout of audio data.
main external API structure.
Definition avcodec.h:443
AVChannelLayout ch_layout
Audio channel layout.
Definition avcodec.h:1055
enum AVSampleFormat sample_fmt
audio sample format
Definition avcodec.h:1047
int64_t bit_rate
the average bitrate
Definition avcodec.h:493
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition avcodec.h:1075
void * priv_data
Definition avcodec.h:470
This structure describes decoded (raw) audio or video data.
Definition frame.h:472
This structure stores compressed data.
Definition packet.h:580
int size
Definition packet.h:604
uint8_t * data
Definition packet.h:603
uint8_t number_of_fc_indexes
Definition sipr.c:56
uint8_t fc_index_bits[10]
size in bits of the fixed codebook indexes
Definition sipr.c:66
uint8_t gc_index_bits
size in bits of the gain codebook indexes
Definition sipr.c:67
uint8_t pitch_delay_bits[5]
size in bits of the adaptive-codebook index for every subframe
Definition sipr.c:63
uint8_t frames_per_packet
Definition sipr.c:52
uint8_t ma_predictor_bits
size in bits of the switched MA predictor
Definition sipr.c:57
float pitch_sharp_factor
Definition sipr.c:53
uint16_t bits_per_frame
Definition sipr.c:50
uint8_t gp_index_bits
Definition sipr.c:65
uint8_t vq_indexes_bits[5]
size in bits of the i-th stage vector of quantizer
Definition sipr.c:60
const char * mode_name
Definition sipr.c:49
uint8_t subframe_count
Definition sipr.c:51
int vq_indexes[5]
Definition sipr.h:57
int pitch_delay[5]
pitch delay
Definition sipr.h:58
int gp_index[5]
adaptive-codebook gain indexes
Definition sipr.h:59
int gc_index[5]
fixed-codebook gain indexes
Definition sipr.h:61
int ma_pred_switch
switched moving average predictor
Definition sipr.h:56
int16_t fc_indexes[5][10]
fixed-codebook indexes
Definition sipr.h:60
Definition swscale.c:71
#define stride
#define av_log(a,...)
static FILE * out
Definition movenc.c:55
static AVFormatContext * ctx
Definition movenc.c:49