1 /*
2 * QCELP decoder
3 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
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 * QCELP decoder
25 * @author Reynaldo H. Verdejo Pinochet
26 * @remark FFmpeg merging spearheaded by Kenan Gillet
27 * @remark Development mentored by Benjamin Larson
28 */
29
30 #include <stddef.h>
31
32 #include "libavutil/avassert.h"
33 #include "libavutil/channel_layout.h"
34 #include "libavutil/float_dsp.h"
35 #include "avcodec.h"
36 #include "internal.h"
37 #include "get_bits.h"
38 #include "qcelpdata.h"
39 #include "celp_filters.h"
40 #include "acelp_filters.h"
41 #include "acelp_vectors.h"
42 #include "lsp.h"
43
44 typedef enum {
45 I_F_Q = -1, /**< insufficient frame quality */
46 SILENCE,
47 RATE_OCTAVE,
48 RATE_QUARTER,
49 RATE_HALF,
50 RATE_FULL
51 } qcelp_packet_rate;
52
53 typedef struct QCELPContext {
54 GetBitContext gb;
55 qcelp_packet_rate bitrate;
56 QCELPFrame frame; /**< unpacked data frame */
57
58 uint8_t erasure_count;
59 uint8_t octave_count; /**< count the consecutive RATE_OCTAVE frames */
60 float prev_lspf[10];
61 float predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
62 float pitch_synthesis_filter_mem[303];
63 float pitch_pre_filter_mem[303];
64 float rnd_fir_filter_mem[180];
65 float formant_mem[170];
66 float last_codebook_gain;
67 int prev_g1[2];
68 int prev_bitrate;
69 float pitch_gain[4];
70 uint8_t pitch_lag[4];
71 uint16_t first16bits;
72 uint8_t warned_buf_mismatch_bitrate;
73
74 /* postfilter */
75 float postfilter_synth_mem[10];
76 float postfilter_agc_mem;
77 float postfilter_tilt_mem;
78 } QCELPContext;
79
80 /**
81 * Initialize the speech codec according to the specification.
82 *
83 * TIA/EIA/IS-733 2.4.9
84 */
qcelp_decode_init(AVCodecContext * avctx)85 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
86 {
87 QCELPContext *q = avctx->priv_data;
88 int i;
89
90 avctx->channels = 1;
91 avctx->channel_layout = AV_CH_LAYOUT_MONO;
92 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
93
94 for (i = 0; i < 10; i++)
95 q->prev_lspf[i] = (i + 1) / 11.0;
96
97 return 0;
98 }
99
100 /**
101 * Decode the 10 quantized LSP frequencies from the LSPV/LSP
102 * transmission codes of any bitrate and check for badly received packets.
103 *
104 * @param q the context
105 * @param lspf line spectral pair frequencies
106 *
107 * @return 0 on success, -1 if the packet is badly received
108 *
109 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
110 */
decode_lspf(QCELPContext * q,float * lspf)111 static int decode_lspf(QCELPContext *q, float *lspf)
112 {
113 int i;
114 float tmp_lspf, smooth, erasure_coeff;
115 const float *predictors;
116
117 if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {
118 predictors = q->prev_bitrate != RATE_OCTAVE &&
119 q->prev_bitrate != I_F_Q ? q->prev_lspf
120 : q->predictor_lspf;
121
122 if (q->bitrate == RATE_OCTAVE) {
123 q->octave_count++;
124
125 for (i = 0; i < 10; i++) {
126 q->predictor_lspf[i] =
127 lspf[i] = (q->frame.lspv[i] ? QCELP_LSP_SPREAD_FACTOR
128 : -QCELP_LSP_SPREAD_FACTOR) +
129 predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR +
130 (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11);
131 }
132 smooth = q->octave_count < 10 ? .875 : 0.1;
133 } else {
134 erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
135
136 av_assert2(q->bitrate == I_F_Q);
137
138 if (q->erasure_count > 1)
139 erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;
140
141 for (i = 0; i < 10; i++) {
142 q->predictor_lspf[i] =
143 lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 +
144 erasure_coeff * predictors[i];
145 }
146 smooth = 0.125;
147 }
148
149 // Check the stability of the LSP frequencies.
150 lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
151 for (i = 1; i < 10; i++)
152 lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR);
153
154 lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR);
155 for (i = 9; i > 0; i--)
156 lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR);
157
158 // Low-pass filter the LSP frequencies.
159 ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);
160 } else {
161 q->octave_count = 0;
162
163 tmp_lspf = 0.0;
164 for (i = 0; i < 5; i++) {
165 lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
166 lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
167 }
168
169 // Check for badly received packets.
170 if (q->bitrate == RATE_QUARTER) {
171 if (lspf[9] <= .70 || lspf[9] >= .97)
172 return -1;
173 for (i = 3; i < 10; i++)
174 if (fabs(lspf[i] - lspf[i - 2]) < .08)
175 return -1;
176 } else {
177 if (lspf[9] <= .66 || lspf[9] >= .985)
178 return -1;
179 for (i = 4; i < 10; i++)
180 if (fabs(lspf[i] - lspf[i - 4]) < .0931)
181 return -1;
182 }
183 }
184 return 0;
185 }
186
187 /**
188 * Convert codebook transmission codes to GAIN and INDEX.
189 *
190 * @param q the context
191 * @param gain array holding the decoded gain
192 *
193 * TIA/EIA/IS-733 2.4.6.2
194 */
decode_gain_and_index(QCELPContext * q,float * gain)195 static void decode_gain_and_index(QCELPContext *q, float *gain)
196 {
197 int i, subframes_count, g1[16];
198 float slope;
199
200 if (q->bitrate >= RATE_QUARTER) {
201 switch (q->bitrate) {
202 case RATE_FULL: subframes_count = 16; break;
203 case RATE_HALF: subframes_count = 4; break;
204 default: subframes_count = 5;
205 }
206 for (i = 0; i < subframes_count; i++) {
207 g1[i] = 4 * q->frame.cbgain[i];
208 if (q->bitrate == RATE_FULL && !((i + 1) & 3)) {
209 g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32);
210 }
211
212 gain[i] = qcelp_g12ga[g1[i]];
213
214 if (q->frame.cbsign[i]) {
215 gain[i] = -gain[i];
216 q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;
217 }
218 }
219
220 q->prev_g1[0] = g1[i - 2];
221 q->prev_g1[1] = g1[i - 1];
222 q->last_codebook_gain = qcelp_g12ga[g1[i - 1]];
223
224 if (q->bitrate == RATE_QUARTER) {
225 // Provide smoothing of the unvoiced excitation energy.
226 gain[7] = gain[4];
227 gain[6] = 0.4 * gain[3] + 0.6 * gain[4];
228 gain[5] = gain[3];
229 gain[4] = 0.8 * gain[2] + 0.2 * gain[3];
230 gain[3] = 0.2 * gain[1] + 0.8 * gain[2];
231 gain[2] = gain[1];
232 gain[1] = 0.6 * gain[0] + 0.4 * gain[1];
233 }
234 } else if (q->bitrate != SILENCE) {
235 if (q->bitrate == RATE_OCTAVE) {
236 g1[0] = 2 * q->frame.cbgain[0] +
237 av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
238 subframes_count = 8;
239 } else {
240 av_assert2(q->bitrate == I_F_Q);
241
242 g1[0] = q->prev_g1[1];
243 switch (q->erasure_count) {
244 case 1 : break;
245 case 2 : g1[0] -= 1; break;
246 case 3 : g1[0] -= 2; break;
247 default: g1[0] -= 6;
248 }
249 if (g1[0] < 0)
250 g1[0] = 0;
251 subframes_count = 4;
252 }
253 // This interpolation is done to produce smoother background noise.
254 slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
255 for (i = 1; i <= subframes_count; i++)
256 gain[i - 1] = q->last_codebook_gain + slope * i;
257
258 q->last_codebook_gain = gain[i - 2];
259 q->prev_g1[0] = q->prev_g1[1];
260 q->prev_g1[1] = g1[0];
261 }
262 }
263
264 /**
265 * If the received packet is Rate 1/4 a further sanity check is made of the
266 * codebook gain.
267 *
268 * @param cbgain the unpacked cbgain array
269 * @return -1 if the sanity check fails, 0 otherwise
270 *
271 * TIA/EIA/IS-733 2.4.8.7.3
272 */
codebook_sanity_check_for_rate_quarter(const uint8_t * cbgain)273 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
274 {
275 int i, diff, prev_diff = 0;
276
277 for (i = 1; i < 5; i++) {
278 diff = cbgain[i] - cbgain[i-1];
279 if (FFABS(diff) > 10)
280 return -1;
281 else if (FFABS(diff - prev_diff) > 12)
282 return -1;
283 prev_diff = diff;
284 }
285 return 0;
286 }
287
288 /**
289 * Compute the scaled codebook vector Cdn From INDEX and GAIN
290 * for all rates.
291 *
292 * The specification lacks some information here.
293 *
294 * TIA/EIA/IS-733 has an omission on the codebook index determination
295 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
296 * you have to subtract the decoded index parameter from the given scaled
297 * codebook vector index 'n' to get the desired circular codebook index, but
298 * it does not mention that you have to clamp 'n' to [0-9] in order to get
299 * RI-compliant results.
300 *
301 * The reason for this mistake seems to be the fact they forgot to mention you
302 * have to do these calculations per codebook subframe and adjust given
303 * equation values accordingly.
304 *
305 * @param q the context
306 * @param gain array holding the 4 pitch subframe gain values
307 * @param cdn_vector array for the generated scaled codebook vector
308 */
compute_svector(QCELPContext * q,const float * gain,float * cdn_vector)309 static void compute_svector(QCELPContext *q, const float *gain,
310 float *cdn_vector)
311 {
312 int i, j, k;
313 uint16_t cbseed, cindex;
314 float *rnd, tmp_gain, fir_filter_value;
315
316 switch (q->bitrate) {
317 case RATE_FULL:
318 for (i = 0; i < 16; i++) {
319 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
320 cindex = -q->frame.cindex[i];
321 for (j = 0; j < 10; j++)
322 *cdn_vector++ = tmp_gain *
323 qcelp_rate_full_codebook[cindex++ & 127];
324 }
325 break;
326 case RATE_HALF:
327 for (i = 0; i < 4; i++) {
328 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
329 cindex = -q->frame.cindex[i];
330 for (j = 0; j < 40; j++)
331 *cdn_vector++ = tmp_gain *
332 qcelp_rate_half_codebook[cindex++ & 127];
333 }
334 break;
335 case RATE_QUARTER:
336 cbseed = (0x0003 & q->frame.lspv[4]) << 14 |
337 (0x003F & q->frame.lspv[3]) << 8 |
338 (0x0060 & q->frame.lspv[2]) << 1 |
339 (0x0007 & q->frame.lspv[1]) << 3 |
340 (0x0038 & q->frame.lspv[0]) >> 3;
341 rnd = q->rnd_fir_filter_mem + 20;
342 for (i = 0; i < 8; i++) {
343 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
344 for (k = 0; k < 20; k++) {
345 cbseed = 521 * cbseed + 259;
346 *rnd = (int16_t) cbseed;
347
348 // FIR filter
349 fir_filter_value = 0.0;
350 for (j = 0; j < 10; j++)
351 fir_filter_value += qcelp_rnd_fir_coefs[j] *
352 (rnd[-j] + rnd[-20+j]);
353
354 fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
355 *cdn_vector++ = tmp_gain * fir_filter_value;
356 rnd++;
357 }
358 }
359 memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160,
360 20 * sizeof(float));
361 break;
362 case RATE_OCTAVE:
363 cbseed = q->first16bits;
364 for (i = 0; i < 8; i++) {
365 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
366 for (j = 0; j < 20; j++) {
367 cbseed = 521 * cbseed + 259;
368 *cdn_vector++ = tmp_gain * (int16_t) cbseed;
369 }
370 }
371 break;
372 case I_F_Q:
373 cbseed = -44; // random codebook index
374 for (i = 0; i < 4; i++) {
375 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
376 for (j = 0; j < 40; j++)
377 *cdn_vector++ = tmp_gain *
378 qcelp_rate_full_codebook[cbseed++ & 127];
379 }
380 break;
381 case SILENCE:
382 memset(cdn_vector, 0, 160 * sizeof(float));
383 break;
384 }
385 }
386
387 /**
388 * Apply generic gain control.
389 *
390 * @param v_out output vector
391 * @param v_in gain-controlled vector
392 * @param v_ref vector to control gain of
393 *
394 * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
395 */
apply_gain_ctrl(float * v_out,const float * v_ref,const float * v_in)396 static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
397 {
398 int i;
399
400 for (i = 0; i < 160; i += 40) {
401 float res = avpriv_scalarproduct_float_c(v_ref + i, v_ref + i, 40);
402 ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40);
403 }
404 }
405
406 /**
407 * Apply filter in pitch-subframe steps.
408 *
409 * @param memory buffer for the previous state of the filter
410 * - must be able to contain 303 elements
411 * - the 143 first elements are from the previous state
412 * - the next 160 are for output
413 * @param v_in input filter vector
414 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
415 * @param lag per-subframe lag array, each element is
416 * - between 16 and 143 if its corresponding pfrac is 0,
417 * - between 16 and 139 otherwise
418 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
419 * otherwise
420 *
421 * @return filter output vector
422 */
do_pitchfilter(float memory[303],const float v_in[160],const float gain[4],const uint8_t * lag,const uint8_t pfrac[4])423 static const float *do_pitchfilter(float memory[303], const float v_in[160],
424 const float gain[4], const uint8_t *lag,
425 const uint8_t pfrac[4])
426 {
427 int i, j;
428 float *v_lag, *v_out;
429 const float *v_len;
430
431 v_out = memory + 143; // Output vector starts at memory[143].
432
433 for (i = 0; i < 4; i++) {
434 if (gain[i]) {
435 v_lag = memory + 143 + 40 * i - lag[i];
436 for (v_len = v_in + 40; v_in < v_len; v_in++) {
437 if (pfrac[i]) { // If it is a fractional lag...
438 for (j = 0, *v_out = 0.0; j < 4; j++)
439 *v_out += qcelp_hammsinc_table[j] *
440 (v_lag[j - 4] + v_lag[3 - j]);
441 } else
442 *v_out = *v_lag;
443
444 *v_out = *v_in + gain[i] * *v_out;
445
446 v_lag++;
447 v_out++;
448 }
449 } else {
450 memcpy(v_out, v_in, 40 * sizeof(float));
451 v_in += 40;
452 v_out += 40;
453 }
454 }
455
456 memmove(memory, memory + 160, 143 * sizeof(float));
457 return memory + 143;
458 }
459
460 /**
461 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
462 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
463 *
464 * @param q the context
465 * @param cdn_vector the scaled codebook vector
466 */
apply_pitch_filters(QCELPContext * q,float * cdn_vector)467 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
468 {
469 int i;
470 const float *v_synthesis_filtered, *v_pre_filtered;
471
472 if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||
473 (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
474
475 if (q->bitrate >= RATE_HALF) {
476 // Compute gain & lag for the whole frame.
477 for (i = 0; i < 4; i++) {
478 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
479
480 q->pitch_lag[i] = q->frame.plag[i] + 16;
481 }
482 } else {
483 float max_pitch_gain;
484
485 if (q->bitrate == I_F_Q) {
486 if (q->erasure_count < 3)
487 max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
488 else
489 max_pitch_gain = 0.0;
490 } else {
491 av_assert2(q->bitrate == SILENCE);
492 max_pitch_gain = 1.0;
493 }
494 for (i = 0; i < 4; i++)
495 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
496
497 memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
498 }
499
500 // pitch synthesis filter
501 v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
502 cdn_vector, q->pitch_gain,
503 q->pitch_lag, q->frame.pfrac);
504
505 // pitch prefilter update
506 for (i = 0; i < 4; i++)
507 q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
508
509 v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
510 v_synthesis_filtered,
511 q->pitch_gain, q->pitch_lag,
512 q->frame.pfrac);
513
514 apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
515 } else {
516 memcpy(q->pitch_synthesis_filter_mem,
517 cdn_vector + 17, 143 * sizeof(float));
518 memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
519 memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
520 memset(q->pitch_lag, 0, sizeof(q->pitch_lag));
521 }
522 }
523
524 /**
525 * Reconstruct LPC coefficients from the line spectral pair frequencies
526 * and perform bandwidth expansion.
527 *
528 * @param lspf line spectral pair frequencies
529 * @param lpc linear predictive coding coefficients
530 *
531 * @note: bandwidth_expansion_coeff could be precalculated into a table
532 * but it seems to be slower on x86
533 *
534 * TIA/EIA/IS-733 2.4.3.3.5
535 */
lspf2lpc(const float * lspf,float * lpc)536 static void lspf2lpc(const float *lspf, float *lpc)
537 {
538 double lsp[10];
539 double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
540 int i;
541
542 for (i = 0; i < 10; i++)
543 lsp[i] = cos(M_PI * lspf[i]);
544
545 ff_acelp_lspd2lpc(lsp, lpc, 5);
546
547 for (i = 0; i < 10; i++) {
548 lpc[i] *= bandwidth_expansion_coeff;
549 bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
550 }
551 }
552
553 /**
554 * Interpolate LSP frequencies and compute LPC coefficients
555 * for a given bitrate & pitch subframe.
556 *
557 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
558 *
559 * @param q the context
560 * @param curr_lspf LSP frequencies vector of the current frame
561 * @param lpc float vector for the resulting LPC
562 * @param subframe_num frame number in decoded stream
563 */
interpolate_lpc(QCELPContext * q,const float * curr_lspf,float * lpc,const int subframe_num)564 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
565 float *lpc, const int subframe_num)
566 {
567 float interpolated_lspf[10];
568 float weight;
569
570 if (q->bitrate >= RATE_QUARTER)
571 weight = 0.25 * (subframe_num + 1);
572 else if (q->bitrate == RATE_OCTAVE && !subframe_num)
573 weight = 0.625;
574 else
575 weight = 1.0;
576
577 if (weight != 1.0) {
578 ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
579 weight, 1.0 - weight, 10);
580 lspf2lpc(interpolated_lspf, lpc);
581 } else if (q->bitrate >= RATE_QUARTER ||
582 (q->bitrate == I_F_Q && !subframe_num))
583 lspf2lpc(curr_lspf, lpc);
584 else if (q->bitrate == SILENCE && !subframe_num)
585 lspf2lpc(q->prev_lspf, lpc);
586 }
587
buf_size2bitrate(const int buf_size)588 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
589 {
590 switch (buf_size) {
591 case 35: return RATE_FULL;
592 case 17: return RATE_HALF;
593 case 8: return RATE_QUARTER;
594 case 4: return RATE_OCTAVE;
595 case 1: return SILENCE;
596 }
597
598 return I_F_Q;
599 }
600
601 /**
602 * Determine the bitrate from the frame size and/or the first byte of the frame.
603 *
604 * @param avctx the AV codec context
605 * @param buf_size length of the buffer
606 * @param buf the buffer
607 *
608 * @return the bitrate on success,
609 * I_F_Q if the bitrate cannot be satisfactorily determined
610 *
611 * TIA/EIA/IS-733 2.4.8.7.1
612 */
determine_bitrate(AVCodecContext * avctx,const int buf_size,const uint8_t ** buf)613 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx,
614 const int buf_size,
615 const uint8_t **buf)
616 {
617 qcelp_packet_rate bitrate;
618
619 if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
620 if (bitrate > **buf) {
621 QCELPContext *q = avctx->priv_data;
622 if (!q->warned_buf_mismatch_bitrate) {
623 av_log(avctx, AV_LOG_WARNING,
624 "Claimed bitrate and buffer size mismatch.\n");
625 q->warned_buf_mismatch_bitrate = 1;
626 }
627 bitrate = **buf;
628 } else if (bitrate < **buf) {
629 av_log(avctx, AV_LOG_ERROR,
630 "Buffer is too small for the claimed bitrate.\n");
631 return I_F_Q;
632 }
633 (*buf)++;
634 } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
635 av_log(avctx, AV_LOG_WARNING,
636 "Bitrate byte missing, guessing bitrate from packet size.\n");
637 } else
638 return I_F_Q;
639
640 if (bitrate == SILENCE) {
641 // FIXME: Remove this warning when tested with samples.
642 avpriv_request_sample(avctx, "Blank frame handling");
643 }
644 return bitrate;
645 }
646
warn_insufficient_frame_quality(AVCodecContext * avctx,const char * message)647 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
648 const char *message)
649 {
650 av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n",
651 avctx->frame_number, message);
652 }
653
postfilter(QCELPContext * q,float * samples,float * lpc)654 static void postfilter(QCELPContext *q, float *samples, float *lpc)
655 {
656 static const float pow_0_775[10] = {
657 0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
658 0.216676, 0.167924, 0.130141, 0.100859, 0.078166
659 }, pow_0_625[10] = {
660 0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
661 0.059605, 0.037253, 0.023283, 0.014552, 0.009095
662 };
663 float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
664 int n;
665
666 for (n = 0; n < 10; n++) {
667 lpc_s[n] = lpc[n] * pow_0_625[n];
668 lpc_p[n] = lpc[n] * pow_0_775[n];
669 }
670
671 ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
672 q->formant_mem + 10, 160, 10);
673 memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
674 ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
675 memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
676
677 ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
678
679 ff_adaptive_gain_control(samples, pole_out + 10,
680 avpriv_scalarproduct_float_c(q->formant_mem + 10,
681 q->formant_mem + 10,
682 160),
683 160, 0.9375, &q->postfilter_agc_mem);
684 }
685
qcelp_decode_frame(AVCodecContext * avctx,void * data,int * got_frame_ptr,AVPacket * avpkt)686 static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
687 int *got_frame_ptr, AVPacket *avpkt)
688 {
689 const uint8_t *buf = avpkt->data;
690 int buf_size = avpkt->size;
691 QCELPContext *q = avctx->priv_data;
692 AVFrame *frame = data;
693 float *outbuffer;
694 int i, ret;
695 float quantized_lspf[10], lpc[10];
696 float gain[16];
697 float *formant_mem;
698
699 /* get output buffer */
700 frame->nb_samples = 160;
701 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
702 return ret;
703 outbuffer = (float *)frame->data[0];
704
705 if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
706 warn_insufficient_frame_quality(avctx, "Bitrate cannot be determined.");
707 goto erasure;
708 }
709
710 if (q->bitrate == RATE_OCTAVE &&
711 (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
712 warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
713 goto erasure;
714 }
715
716 if (q->bitrate > SILENCE) {
717 const QCELPBitmap *bitmaps = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
718 const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +
719 qcelp_unpacking_bitmaps_lengths[q->bitrate];
720 uint8_t *unpacked_data = (uint8_t *)&q->frame;
721
722 if ((ret = init_get_bits8(&q->gb, buf, buf_size)) < 0)
723 return ret;
724
725 memset(&q->frame, 0, sizeof(QCELPFrame));
726
727 for (; bitmaps < bitmaps_end; bitmaps++)
728 unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
729
730 // Check for erasures/blanks on rates 1, 1/4 and 1/8.
731 if (q->frame.reserved) {
732 warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
733 goto erasure;
734 }
735 if (q->bitrate == RATE_QUARTER &&
736 codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) {
737 warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
738 goto erasure;
739 }
740
741 if (q->bitrate >= RATE_HALF) {
742 for (i = 0; i < 4; i++) {
743 if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
744 warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
745 goto erasure;
746 }
747 }
748 }
749 }
750
751 decode_gain_and_index(q, gain);
752 compute_svector(q, gain, outbuffer);
753
754 if (decode_lspf(q, quantized_lspf) < 0) {
755 warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
756 goto erasure;
757 }
758
759 apply_pitch_filters(q, outbuffer);
760
761 if (q->bitrate == I_F_Q) {
762 erasure:
763 q->bitrate = I_F_Q;
764 q->erasure_count++;
765 decode_gain_and_index(q, gain);
766 compute_svector(q, gain, outbuffer);
767 decode_lspf(q, quantized_lspf);
768 apply_pitch_filters(q, outbuffer);
769 } else
770 q->erasure_count = 0;
771
772 formant_mem = q->formant_mem + 10;
773 for (i = 0; i < 4; i++) {
774 interpolate_lpc(q, quantized_lspf, lpc, i);
775 ff_celp_lp_synthesis_filterf(formant_mem, lpc,
776 outbuffer + i * 40, 40, 10);
777 formant_mem += 40;
778 }
779
780 // postfilter, as per TIA/EIA/IS-733 2.4.8.6
781 postfilter(q, outbuffer, lpc);
782
783 memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
784
785 memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
786 q->prev_bitrate = q->bitrate;
787
788 *got_frame_ptr = 1;
789
790 return buf_size;
791 }
792
793 AVCodec ff_qcelp_decoder = {
794 .name = "qcelp",
795 .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
796 .type = AVMEDIA_TYPE_AUDIO,
797 .id = AV_CODEC_ID_QCELP,
798 .init = qcelp_decode_init,
799 .decode = qcelp_decode_frame,
800 .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
801 .priv_data_size = sizeof(QCELPContext),
802 };
803