avformat/mpeg: demux ivtv captions
[ffmpeg.git] / libavcodec / qcelpdec.c
blob1435fecc2eb2172e43ba102da2d3bfc837785074
1 /*
2 * QCELP decoder
3 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
5 * This file is part of FFmpeg.
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.
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.
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
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
30 #include "libavutil/avassert.h"
31 #include "libavutil/channel_layout.h"
32 #include "libavutil/float_dsp.h"
33 #include "avcodec.h"
34 #include "codec_internal.h"
35 #include "decode.h"
36 #include "get_bits.h"
37 #include "qcelpdata.h"
38 #include "celp_filters.h"
39 #include "acelp_filters.h"
40 #include "acelp_vectors.h"
41 #include "lsp.h"
43 typedef enum {
44 I_F_Q = -1, /**< insufficient frame quality */
45 SILENCE,
46 RATE_OCTAVE,
47 RATE_QUARTER,
48 RATE_HALF,
49 RATE_FULL
50 } qcelp_packet_rate;
52 typedef struct QCELPContext {
53 GetBitContext gb;
54 qcelp_packet_rate bitrate;
55 QCELPFrame frame; /**< unpacked data frame */
57 uint8_t erasure_count;
58 uint8_t octave_count; /**< count the consecutive RATE_OCTAVE frames */
59 float prev_lspf[10];
60 float predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
61 float pitch_synthesis_filter_mem[303];
62 float pitch_pre_filter_mem[303];
63 float rnd_fir_filter_mem[180];
64 float formant_mem[170];
65 float last_codebook_gain;
66 int prev_g1[2];
67 int prev_bitrate;
68 float pitch_gain[4];
69 uint8_t pitch_lag[4];
70 uint16_t first16bits;
71 uint8_t warned_buf_mismatch_bitrate;
73 /* postfilter */
74 float postfilter_synth_mem[10];
75 float postfilter_agc_mem;
76 float postfilter_tilt_mem;
77 } QCELPContext;
79 /**
80 * Initialize the speech codec according to the specification.
82 * TIA/EIA/IS-733 2.4.9
84 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
86 QCELPContext *q = avctx->priv_data;
87 int i;
89 av_channel_layout_uninit(&avctx->ch_layout);
90 avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
91 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
93 for (i = 0; i < 10; i++)
94 q->prev_lspf[i] = (i + 1) / 11.0;
96 return 0;
99 /**
100 * Decode the 10 quantized LSP frequencies from the LSPV/LSP
101 * transmission codes of any bitrate and check for badly received packets.
103 * @param q the context
104 * @param lspf line spectral pair frequencies
106 * @return 0 on success, -1 if the packet is badly received
108 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
110 static int decode_lspf(QCELPContext *q, float *lspf)
112 int i;
113 float tmp_lspf, smooth, erasure_coeff;
114 const float *predictors;
116 if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {
117 predictors = q->prev_bitrate != RATE_OCTAVE &&
118 q->prev_bitrate != I_F_Q ? q->prev_lspf
119 : q->predictor_lspf;
121 if (q->bitrate == RATE_OCTAVE) {
122 q->octave_count++;
124 for (i = 0; i < 10; i++) {
125 q->predictor_lspf[i] =
126 lspf[i] = (q->frame.lspv[i] ? QCELP_LSP_SPREAD_FACTOR
127 : -QCELP_LSP_SPREAD_FACTOR) +
128 predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR +
129 (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11);
131 smooth = q->octave_count < 10 ? .875 : 0.1;
132 } else {
133 erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
135 av_assert2(q->bitrate == I_F_Q);
137 if (q->erasure_count > 1)
138 erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;
140 for (i = 0; i < 10; i++) {
141 q->predictor_lspf[i] =
142 lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 +
143 erasure_coeff * predictors[i];
145 smooth = 0.125;
148 // Check the stability of the LSP frequencies.
149 lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
150 for (i = 1; i < 10; i++)
151 lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR);
153 lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR);
154 for (i = 9; i > 0; i--)
155 lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR);
157 // Low-pass filter the LSP frequencies.
158 ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);
159 } else {
160 q->octave_count = 0;
162 tmp_lspf = 0.0;
163 for (i = 0; i < 5; i++) {
164 lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
165 lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
168 // Check for badly received packets.
169 if (q->bitrate == RATE_QUARTER) {
170 if (lspf[9] <= .70 || lspf[9] >= .97)
171 return -1;
172 for (i = 3; i < 10; i++)
173 if (fabs(lspf[i] - lspf[i - 2]) < .08)
174 return -1;
175 } else {
176 if (lspf[9] <= .66 || lspf[9] >= .985)
177 return -1;
178 for (i = 4; i < 10; i++)
179 if (fabs(lspf[i] - lspf[i - 4]) < .0931)
180 return -1;
183 return 0;
187 * Convert codebook transmission codes to GAIN and INDEX.
189 * @param q the context
190 * @param gain array holding the decoded gain
192 * TIA/EIA/IS-733 2.4.6.2
194 static void decode_gain_and_index(QCELPContext *q, float *gain)
196 int i, subframes_count, g1[16];
197 float slope;
199 if (q->bitrate >= RATE_QUARTER) {
200 switch (q->bitrate) {
201 case RATE_FULL: subframes_count = 16; break;
202 case RATE_HALF: subframes_count = 4; break;
203 default: subframes_count = 5;
205 for (i = 0; i < subframes_count; i++) {
206 g1[i] = 4 * q->frame.cbgain[i];
207 if (q->bitrate == RATE_FULL && !((i + 1) & 3)) {
208 g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32);
211 gain[i] = qcelp_g12ga[g1[i]];
213 if (q->frame.cbsign[i]) {
214 gain[i] = -gain[i];
215 q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;
219 q->prev_g1[0] = g1[i - 2];
220 q->prev_g1[1] = g1[i - 1];
221 q->last_codebook_gain = qcelp_g12ga[g1[i - 1]];
223 if (q->bitrate == RATE_QUARTER) {
224 // Provide smoothing of the unvoiced excitation energy.
225 gain[7] = gain[4];
226 gain[6] = 0.4 * gain[3] + 0.6 * gain[4];
227 gain[5] = gain[3];
228 gain[4] = 0.8 * gain[2] + 0.2 * gain[3];
229 gain[3] = 0.2 * gain[1] + 0.8 * gain[2];
230 gain[2] = gain[1];
231 gain[1] = 0.6 * gain[0] + 0.4 * gain[1];
233 } else if (q->bitrate != SILENCE) {
234 if (q->bitrate == RATE_OCTAVE) {
235 g1[0] = 2 * q->frame.cbgain[0] +
236 av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
237 subframes_count = 8;
238 } else {
239 av_assert2(q->bitrate == I_F_Q);
241 g1[0] = q->prev_g1[1];
242 switch (q->erasure_count) {
243 case 1 : break;
244 case 2 : g1[0] -= 1; break;
245 case 3 : g1[0] -= 2; break;
246 default: g1[0] -= 6;
248 if (g1[0] < 0)
249 g1[0] = 0;
250 subframes_count = 4;
252 // This interpolation is done to produce smoother background noise.
253 slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
254 for (i = 1; i <= subframes_count; i++)
255 gain[i - 1] = q->last_codebook_gain + slope * i;
257 q->last_codebook_gain = gain[i - 2];
258 q->prev_g1[0] = q->prev_g1[1];
259 q->prev_g1[1] = g1[0];
264 * If the received packet is Rate 1/4 a further sanity check is made of the
265 * codebook gain.
267 * @param cbgain the unpacked cbgain array
268 * @return -1 if the sanity check fails, 0 otherwise
270 * TIA/EIA/IS-733 2.4.8.7.3
272 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
274 int i, diff, prev_diff = 0;
276 for (i = 1; i < 5; i++) {
277 diff = cbgain[i] - cbgain[i-1];
278 if (FFABS(diff) > 10)
279 return -1;
280 else if (FFABS(diff - prev_diff) > 12)
281 return -1;
282 prev_diff = diff;
284 return 0;
288 * Compute the scaled codebook vector Cdn From INDEX and GAIN
289 * for all rates.
291 * The specification lacks some information here.
293 * TIA/EIA/IS-733 has an omission on the codebook index determination
294 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
295 * you have to subtract the decoded index parameter from the given scaled
296 * codebook vector index 'n' to get the desired circular codebook index, but
297 * it does not mention that you have to clamp 'n' to [0-9] in order to get
298 * RI-compliant results.
300 * The reason for this mistake seems to be the fact they forgot to mention you
301 * have to do these calculations per codebook subframe and adjust given
302 * equation values accordingly.
304 * @param q the context
305 * @param gain array holding the 4 pitch subframe gain values
306 * @param cdn_vector array for the generated scaled codebook vector
308 static void compute_svector(QCELPContext *q, const float *gain,
309 float *cdn_vector)
311 int i, j, k;
312 uint16_t cbseed, cindex;
313 float *rnd, tmp_gain, fir_filter_value;
315 switch (q->bitrate) {
316 case RATE_FULL:
317 for (i = 0; i < 16; i++) {
318 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
319 cindex = -q->frame.cindex[i];
320 for (j = 0; j < 10; j++)
321 *cdn_vector++ = tmp_gain *
322 qcelp_rate_full_codebook[cindex++ & 127];
324 break;
325 case RATE_HALF:
326 for (i = 0; i < 4; i++) {
327 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
328 cindex = -q->frame.cindex[i];
329 for (j = 0; j < 40; j++)
330 *cdn_vector++ = tmp_gain *
331 qcelp_rate_half_codebook[cindex++ & 127];
333 break;
334 case RATE_QUARTER:
335 cbseed = (0x0003 & q->frame.lspv[4]) << 14 |
336 (0x003F & q->frame.lspv[3]) << 8 |
337 (0x0060 & q->frame.lspv[2]) << 1 |
338 (0x0007 & q->frame.lspv[1]) << 3 |
339 (0x0038 & q->frame.lspv[0]) >> 3;
340 rnd = q->rnd_fir_filter_mem + 20;
341 for (i = 0; i < 8; i++) {
342 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
343 for (k = 0; k < 20; k++) {
344 cbseed = 521 * cbseed + 259;
345 *rnd = (int16_t) cbseed;
347 // FIR filter
348 fir_filter_value = 0.0;
349 for (j = 0; j < 10; j++)
350 fir_filter_value += qcelp_rnd_fir_coefs[j] *
351 (rnd[-j] + rnd[-20+j]);
353 fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
354 *cdn_vector++ = tmp_gain * fir_filter_value;
355 rnd++;
358 memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160,
359 20 * sizeof(float));
360 break;
361 case RATE_OCTAVE:
362 cbseed = q->first16bits;
363 for (i = 0; i < 8; i++) {
364 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
365 for (j = 0; j < 20; j++) {
366 cbseed = 521 * cbseed + 259;
367 *cdn_vector++ = tmp_gain * (int16_t) cbseed;
370 break;
371 case I_F_Q:
372 cbseed = -44; // random codebook index
373 for (i = 0; i < 4; i++) {
374 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
375 for (j = 0; j < 40; j++)
376 *cdn_vector++ = tmp_gain *
377 qcelp_rate_full_codebook[cbseed++ & 127];
379 break;
380 case SILENCE:
381 memset(cdn_vector, 0, 160 * sizeof(float));
382 break;
387 * Apply generic gain control.
389 * @param v_out output vector
390 * @param v_in gain-controlled vector
391 * @param v_ref vector to control gain of
393 * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
395 static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
397 int i;
399 for (i = 0; i < 160; i += 40) {
400 float res = avpriv_scalarproduct_float_c(v_ref + i, v_ref + i, 40);
401 ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40);
406 * Apply filter in pitch-subframe steps.
408 * @param memory buffer for the previous state of the filter
409 * - must be able to contain 303 elements
410 * - the 143 first elements are from the previous state
411 * - the next 160 are for output
412 * @param v_in input filter vector
413 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
414 * @param lag per-subframe lag array, each element is
415 * - between 16 and 143 if its corresponding pfrac is 0,
416 * - between 16 and 139 otherwise
417 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
418 * otherwise
420 * @return filter output vector
422 static const float *do_pitchfilter(float memory[303], const float v_in[160],
423 const float gain[4], const uint8_t *lag,
424 const uint8_t pfrac[4])
426 int i, j;
427 float *v_lag, *v_out;
428 const float *v_len;
430 v_out = memory + 143; // Output vector starts at memory[143].
432 for (i = 0; i < 4; i++) {
433 if (gain[i]) {
434 v_lag = memory + 143 + 40 * i - lag[i];
435 for (v_len = v_in + 40; v_in < v_len; v_in++) {
436 if (pfrac[i]) { // If it is a fractional lag...
437 for (j = 0, *v_out = 0.0; j < 4; j++)
438 *v_out += qcelp_hammsinc_table[j] *
439 (v_lag[j - 4] + v_lag[3 - j]);
440 } else
441 *v_out = *v_lag;
443 *v_out = *v_in + gain[i] * *v_out;
445 v_lag++;
446 v_out++;
448 } else {
449 memcpy(v_out, v_in, 40 * sizeof(float));
450 v_in += 40;
451 v_out += 40;
455 memmove(memory, memory + 160, 143 * sizeof(float));
456 return memory + 143;
460 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
461 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
463 * @param q the context
464 * @param cdn_vector the scaled codebook vector
466 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
468 int i;
469 const float *v_synthesis_filtered, *v_pre_filtered;
471 if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||
472 (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
474 if (q->bitrate >= RATE_HALF) {
475 // Compute gain & lag for the whole frame.
476 for (i = 0; i < 4; i++) {
477 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
479 q->pitch_lag[i] = q->frame.plag[i] + 16;
481 } else {
482 float max_pitch_gain;
484 if (q->bitrate == I_F_Q) {
485 if (q->erasure_count < 3)
486 max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
487 else
488 max_pitch_gain = 0.0;
489 } else {
490 av_assert2(q->bitrate == SILENCE);
491 max_pitch_gain = 1.0;
493 for (i = 0; i < 4; i++)
494 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
496 memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
499 // pitch synthesis filter
500 v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
501 cdn_vector, q->pitch_gain,
502 q->pitch_lag, q->frame.pfrac);
504 // pitch prefilter update
505 for (i = 0; i < 4; i++)
506 q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
508 v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
509 v_synthesis_filtered,
510 q->pitch_gain, q->pitch_lag,
511 q->frame.pfrac);
513 apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
514 } else {
515 memcpy(q->pitch_synthesis_filter_mem,
516 cdn_vector + 17, 143 * sizeof(float));
517 memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
518 memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
519 memset(q->pitch_lag, 0, sizeof(q->pitch_lag));
524 * Reconstruct LPC coefficients from the line spectral pair frequencies
525 * and perform bandwidth expansion.
527 * @param lspf line spectral pair frequencies
528 * @param lpc linear predictive coding coefficients
530 * @note: bandwidth_expansion_coeff could be precalculated into a table
531 * but it seems to be slower on x86
533 * TIA/EIA/IS-733 2.4.3.3.5
535 static void lspf2lpc(const float *lspf, float *lpc)
537 double lsp[10];
538 double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
539 int i;
541 for (i = 0; i < 10; i++)
542 lsp[i] = cos(M_PI * lspf[i]);
544 ff_acelp_lspd2lpc(lsp, lpc, 5);
546 for (i = 0; i < 10; i++) {
547 lpc[i] *= bandwidth_expansion_coeff;
548 bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
553 * Interpolate LSP frequencies and compute LPC coefficients
554 * for a given bitrate & pitch subframe.
556 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
558 * @param q the context
559 * @param curr_lspf LSP frequencies vector of the current frame
560 * @param lpc float vector for the resulting LPC
561 * @param subframe_num frame number in decoded stream
563 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
564 float *lpc, const int subframe_num)
566 float interpolated_lspf[10];
567 float weight;
569 if (q->bitrate >= RATE_QUARTER)
570 weight = 0.25 * (subframe_num + 1);
571 else if (q->bitrate == RATE_OCTAVE && !subframe_num)
572 weight = 0.625;
573 else
574 weight = 1.0;
576 if (weight != 1.0) {
577 ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
578 weight, 1.0 - weight, 10);
579 lspf2lpc(interpolated_lspf, lpc);
580 } else if (q->bitrate >= RATE_QUARTER ||
581 (q->bitrate == I_F_Q && !subframe_num))
582 lspf2lpc(curr_lspf, lpc);
583 else if (q->bitrate == SILENCE && !subframe_num)
584 lspf2lpc(q->prev_lspf, lpc);
587 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
589 switch (buf_size) {
590 case 35: return RATE_FULL;
591 case 17: return RATE_HALF;
592 case 8: return RATE_QUARTER;
593 case 4: return RATE_OCTAVE;
594 case 1: return SILENCE;
597 return I_F_Q;
601 * Determine the bitrate from the frame size and/or the first byte of the frame.
603 * @param avctx the AV codec context
604 * @param buf_size length of the buffer
605 * @param buf the buffer
607 * @return the bitrate on success,
608 * I_F_Q if the bitrate cannot be satisfactorily determined
610 * TIA/EIA/IS-733 2.4.8.7.1
612 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx,
613 const int buf_size,
614 const uint8_t **buf)
616 qcelp_packet_rate bitrate;
618 if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
619 if (bitrate > **buf) {
620 QCELPContext *q = avctx->priv_data;
621 if (!q->warned_buf_mismatch_bitrate) {
622 av_log(avctx, AV_LOG_WARNING,
623 "Claimed bitrate and buffer size mismatch.\n");
624 q->warned_buf_mismatch_bitrate = 1;
626 bitrate = **buf;
627 } else if (bitrate < **buf) {
628 av_log(avctx, AV_LOG_ERROR,
629 "Buffer is too small for the claimed bitrate.\n");
630 return I_F_Q;
632 (*buf)++;
633 } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
634 av_log(avctx, AV_LOG_WARNING,
635 "Bitrate byte missing, guessing bitrate from packet size.\n");
636 } else
637 return I_F_Q;
639 if (bitrate == SILENCE) {
640 // FIXME: Remove this warning when tested with samples.
641 avpriv_request_sample(avctx, "Blank frame handling");
643 return bitrate;
646 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
647 const char *message)
649 av_log(avctx, AV_LOG_WARNING, "Frame #%"PRId64", IFQ: %s\n",
650 avctx->frame_num, message);
653 static void postfilter(QCELPContext *q, float *samples, float *lpc)
655 static const float pow_0_775[10] = {
656 0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
657 0.216676, 0.167924, 0.130141, 0.100859, 0.078166
658 }, pow_0_625[10] = {
659 0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
660 0.059605, 0.037253, 0.023283, 0.014552, 0.009095
662 float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
663 int n;
665 for (n = 0; n < 10; n++) {
666 lpc_s[n] = lpc[n] * pow_0_625[n];
667 lpc_p[n] = lpc[n] * pow_0_775[n];
670 ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
671 q->formant_mem + 10, 160, 10);
672 memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
673 ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
674 memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
676 ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
678 ff_adaptive_gain_control(samples, pole_out + 10,
679 avpriv_scalarproduct_float_c(q->formant_mem + 10,
680 q->formant_mem + 10,
681 160),
682 160, 0.9375, &q->postfilter_agc_mem);
685 static int qcelp_decode_frame(AVCodecContext *avctx, AVFrame *frame,
686 int *got_frame_ptr, AVPacket *avpkt)
688 const uint8_t *buf = avpkt->data;
689 int buf_size = avpkt->size;
690 QCELPContext *q = avctx->priv_data;
691 float *outbuffer;
692 int i, ret;
693 float quantized_lspf[10], lpc[10];
694 float gain[16];
695 float *formant_mem;
697 /* get output buffer */
698 frame->nb_samples = 160;
699 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
700 return ret;
701 outbuffer = (float *)frame->data[0];
703 if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
704 warn_insufficient_frame_quality(avctx, "Bitrate cannot be determined.");
705 goto erasure;
708 if (q->bitrate == RATE_OCTAVE &&
709 (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
710 warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
711 goto erasure;
714 if (q->bitrate > SILENCE) {
715 const QCELPBitmap *bitmaps = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
716 const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +
717 qcelp_unpacking_bitmaps_lengths[q->bitrate];
718 uint8_t *unpacked_data = (uint8_t *)&q->frame;
720 if ((ret = init_get_bits8(&q->gb, buf, buf_size)) < 0)
721 return ret;
723 memset(&q->frame, 0, sizeof(QCELPFrame));
725 for (; bitmaps < bitmaps_end; bitmaps++)
726 unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
728 // Check for erasures/blanks on rates 1, 1/4 and 1/8.
729 if (q->frame.reserved) {
730 warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
731 goto erasure;
733 if (q->bitrate == RATE_QUARTER &&
734 codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) {
735 warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
736 goto erasure;
739 if (q->bitrate >= RATE_HALF) {
740 for (i = 0; i < 4; i++) {
741 if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
742 warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
743 goto erasure;
749 decode_gain_and_index(q, gain);
750 compute_svector(q, gain, outbuffer);
752 if (decode_lspf(q, quantized_lspf) < 0) {
753 warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
754 goto erasure;
757 apply_pitch_filters(q, outbuffer);
759 if (q->bitrate == I_F_Q) {
760 erasure:
761 q->bitrate = I_F_Q;
762 q->erasure_count++;
763 decode_gain_and_index(q, gain);
764 compute_svector(q, gain, outbuffer);
765 decode_lspf(q, quantized_lspf);
766 apply_pitch_filters(q, outbuffer);
767 } else
768 q->erasure_count = 0;
770 formant_mem = q->formant_mem + 10;
771 for (i = 0; i < 4; i++) {
772 interpolate_lpc(q, quantized_lspf, lpc, i);
773 ff_celp_lp_synthesis_filterf(formant_mem, lpc,
774 outbuffer + i * 40, 40, 10);
775 formant_mem += 40;
778 // postfilter, as per TIA/EIA/IS-733 2.4.8.6
779 postfilter(q, outbuffer, lpc);
781 memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
783 memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
784 q->prev_bitrate = q->bitrate;
786 *got_frame_ptr = 1;
788 return buf_size;
791 const FFCodec ff_qcelp_decoder = {
792 .p.name = "qcelp",
793 CODEC_LONG_NAME("QCELP / PureVoice"),
794 .p.type = AVMEDIA_TYPE_AUDIO,
795 .p.id = AV_CODEC_ID_QCELP,
796 .init = qcelp_decode_init,
797 FF_CODEC_DECODE_CB(qcelp_decode_frame),
798 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
799 .priv_data_size = sizeof(QCELPContext),