lavfi: switch to AVFrame.
[FFMpeg-mirror/mplayer-patches.git] / libavcodec / qcelpdec.c
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1 /*
2 * QCELP decoder
3 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
5 * This file is part of Libav.
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.
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.
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
22 /**
23 * @file
24 * QCELP decoder
25 * @author Reynaldo H. Verdejo Pinochet
26 * @remark Libav merging spearheaded by Kenan Gillet
27 * @remark Development mentored by Benjamin Larson
30 #include <stddef.h>
32 #include "libavutil/channel_layout.h"
33 #include "libavutil/float_dsp.h"
34 #include "avcodec.h"
35 #include "internal.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 #undef NDEBUG
44 #include <assert.h>
46 typedef enum {
47 I_F_Q = -1, /**< insufficient frame quality */
48 SILENCE,
49 RATE_OCTAVE,
50 RATE_QUARTER,
51 RATE_HALF,
52 RATE_FULL
53 } qcelp_packet_rate;
55 typedef struct {
56 GetBitContext gb;
57 qcelp_packet_rate bitrate;
58 QCELPFrame frame; /**< unpacked data frame */
60 uint8_t erasure_count;
61 uint8_t octave_count; /**< count the consecutive RATE_OCTAVE frames */
62 float prev_lspf[10];
63 float predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
64 float pitch_synthesis_filter_mem[303];
65 float pitch_pre_filter_mem[303];
66 float rnd_fir_filter_mem[180];
67 float formant_mem[170];
68 float last_codebook_gain;
69 int prev_g1[2];
70 int prev_bitrate;
71 float pitch_gain[4];
72 uint8_t pitch_lag[4];
73 uint16_t first16bits;
74 uint8_t warned_buf_mismatch_bitrate;
76 /* postfilter */
77 float postfilter_synth_mem[10];
78 float postfilter_agc_mem;
79 float postfilter_tilt_mem;
80 } QCELPContext;
82 /**
83 * Initialize the speech codec according to the specification.
85 * TIA/EIA/IS-733 2.4.9
87 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
89 QCELPContext *q = avctx->priv_data;
90 int i;
92 avctx->channels = 1;
93 avctx->channel_layout = AV_CH_LAYOUT_MONO;
94 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
96 for (i = 0; i < 10; i++)
97 q->prev_lspf[i] = (i + 1) / 11.;
99 return 0;
103 * Decode the 10 quantized LSP frequencies from the LSPV/LSP
104 * transmission codes of any bitrate and check for badly received packets.
106 * @param q the context
107 * @param lspf line spectral pair frequencies
109 * @return 0 on success, -1 if the packet is badly received
111 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
113 static int decode_lspf(QCELPContext *q, float *lspf)
115 int i;
116 float tmp_lspf, smooth, erasure_coeff;
117 const float *predictors;
119 if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {
120 predictors = q->prev_bitrate != RATE_OCTAVE &&
121 q->prev_bitrate != I_F_Q ? q->prev_lspf
122 : q->predictor_lspf;
124 if (q->bitrate == RATE_OCTAVE) {
125 q->octave_count++;
127 for (i = 0; i < 10; i++) {
128 q->predictor_lspf[i] =
129 lspf[i] = (q->frame.lspv[i] ? QCELP_LSP_SPREAD_FACTOR
130 : -QCELP_LSP_SPREAD_FACTOR) +
131 predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR +
132 (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11);
134 smooth = q->octave_count < 10 ? .875 : 0.1;
135 } else {
136 erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
138 assert(q->bitrate == I_F_Q);
140 if (q->erasure_count > 1)
141 erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;
143 for (i = 0; i < 10; i++) {
144 q->predictor_lspf[i] =
145 lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 +
146 erasure_coeff * predictors[i];
148 smooth = 0.125;
151 // Check the stability of the LSP frequencies.
152 lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
153 for (i = 1; i < 10; i++)
154 lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR);
156 lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR);
157 for (i = 9; i > 0; i--)
158 lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR);
160 // Low-pass filter the LSP frequencies.
161 ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);
162 } else {
163 q->octave_count = 0;
165 tmp_lspf = 0.;
166 for (i = 0; i < 5; i++) {
167 lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
168 lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
171 // Check for badly received packets.
172 if (q->bitrate == RATE_QUARTER) {
173 if (lspf[9] <= .70 || lspf[9] >= .97)
174 return -1;
175 for (i = 3; i < 10; i++)
176 if (fabs(lspf[i] - lspf[i - 2]) < .08)
177 return -1;
178 } else {
179 if (lspf[9] <= .66 || lspf[9] >= .985)
180 return -1;
181 for (i = 4; i < 10; i++)
182 if (fabs(lspf[i] - lspf[i - 4]) < .0931)
183 return -1;
186 return 0;
190 * Convert codebook transmission codes to GAIN and INDEX.
192 * @param q the context
193 * @param gain array holding the decoded gain
195 * TIA/EIA/IS-733 2.4.6.2
197 static void decode_gain_and_index(QCELPContext *q, float *gain)
199 int i, subframes_count, g1[16];
200 float slope;
202 if (q->bitrate >= RATE_QUARTER) {
203 switch (q->bitrate) {
204 case RATE_FULL: subframes_count = 16; break;
205 case RATE_HALF: subframes_count = 4; break;
206 default: subframes_count = 5;
208 for (i = 0; i < subframes_count; i++) {
209 g1[i] = 4 * q->frame.cbgain[i];
210 if (q->bitrate == RATE_FULL && !((i + 1) & 3)) {
211 g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32);
214 gain[i] = qcelp_g12ga[g1[i]];
216 if (q->frame.cbsign[i]) {
217 gain[i] = -gain[i];
218 q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;
222 q->prev_g1[0] = g1[i - 2];
223 q->prev_g1[1] = g1[i - 1];
224 q->last_codebook_gain = qcelp_g12ga[g1[i - 1]];
226 if (q->bitrate == RATE_QUARTER) {
227 // Provide smoothing of the unvoiced excitation energy.
228 gain[7] = gain[4];
229 gain[6] = 0.4 * gain[3] + 0.6 * gain[4];
230 gain[5] = gain[3];
231 gain[4] = 0.8 * gain[2] + 0.2 * gain[3];
232 gain[3] = 0.2 * gain[1] + 0.8 * gain[2];
233 gain[2] = gain[1];
234 gain[1] = 0.6 * gain[0] + 0.4 * gain[1];
236 } else if (q->bitrate != SILENCE) {
237 if (q->bitrate == RATE_OCTAVE) {
238 g1[0] = 2 * q->frame.cbgain[0] +
239 av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
240 subframes_count = 8;
241 } else {
242 assert(q->bitrate == I_F_Q);
244 g1[0] = q->prev_g1[1];
245 switch (q->erasure_count) {
246 case 1 : break;
247 case 2 : g1[0] -= 1; break;
248 case 3 : g1[0] -= 2; break;
249 default: g1[0] -= 6;
251 if (g1[0] < 0)
252 g1[0] = 0;
253 subframes_count = 4;
255 // This interpolation is done to produce smoother background noise.
256 slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
257 for (i = 1; i <= subframes_count; i++)
258 gain[i - 1] = q->last_codebook_gain + slope * i;
260 q->last_codebook_gain = gain[i - 2];
261 q->prev_g1[0] = q->prev_g1[1];
262 q->prev_g1[1] = g1[0];
267 * If the received packet is Rate 1/4 a further sanity check is made of the
268 * codebook gain.
270 * @param cbgain the unpacked cbgain array
271 * @return -1 if the sanity check fails, 0 otherwise
273 * TIA/EIA/IS-733 2.4.8.7.3
275 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
277 int i, diff, prev_diff = 0;
279 for (i = 1; i < 5; i++) {
280 diff = cbgain[i] - cbgain[i-1];
281 if (FFABS(diff) > 10)
282 return -1;
283 else if (FFABS(diff - prev_diff) > 12)
284 return -1;
285 prev_diff = diff;
287 return 0;
291 * Compute the scaled codebook vector Cdn From INDEX and GAIN
292 * for all rates.
294 * The specification lacks some information here.
296 * TIA/EIA/IS-733 has an omission on the codebook index determination
297 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
298 * you have to subtract the decoded index parameter from the given scaled
299 * codebook vector index 'n' to get the desired circular codebook index, but
300 * it does not mention that you have to clamp 'n' to [0-9] in order to get
301 * RI-compliant results.
303 * The reason for this mistake seems to be the fact they forgot to mention you
304 * have to do these calculations per codebook subframe and adjust given
305 * equation values accordingly.
307 * @param q the context
308 * @param gain array holding the 4 pitch subframe gain values
309 * @param cdn_vector array for the generated scaled codebook vector
311 static void compute_svector(QCELPContext *q, const float *gain,
312 float *cdn_vector)
314 int i, j, k;
315 uint16_t cbseed, cindex;
316 float *rnd, tmp_gain, fir_filter_value;
318 switch (q->bitrate) {
319 case RATE_FULL:
320 for (i = 0; i < 16; i++) {
321 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
322 cindex = -q->frame.cindex[i];
323 for (j = 0; j < 10; j++)
324 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
326 break;
327 case RATE_HALF:
328 for (i = 0; i < 4; i++) {
329 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
330 cindex = -q->frame.cindex[i];
331 for (j = 0; j < 40; j++)
332 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
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;
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]);
354 fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
355 *cdn_vector++ = tmp_gain * fir_filter_value;
356 rnd++;
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;
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 * 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.; j < 4; j++)
438 *v_out += qcelp_hammsinc_table[j] * (v_lag[j - 4] + v_lag[3 - j]);
439 } else
440 *v_out = *v_lag;
442 *v_out = *v_in + gain[i] * *v_out;
444 v_lag++;
445 v_out++;
447 } else {
448 memcpy(v_out, v_in, 40 * sizeof(float));
449 v_in += 40;
450 v_out += 40;
454 memmove(memory, memory + 160, 143 * sizeof(float));
455 return memory + 143;
459 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
460 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
462 * @param q the context
463 * @param cdn_vector the scaled codebook vector
465 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
467 int i;
468 const float *v_synthesis_filtered, *v_pre_filtered;
470 if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||
471 (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
473 if (q->bitrate >= RATE_HALF) {
474 // Compute gain & lag for the whole frame.
475 for (i = 0; i < 4; i++) {
476 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
478 q->pitch_lag[i] = q->frame.plag[i] + 16;
480 } else {
481 float max_pitch_gain;
483 if (q->bitrate == I_F_Q) {
484 if (q->erasure_count < 3)
485 max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
486 else
487 max_pitch_gain = 0.0;
488 } else {
489 assert(q->bitrate == SILENCE);
490 max_pitch_gain = 1.0;
492 for (i = 0; i < 4; i++)
493 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
495 memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
498 // pitch synthesis filter
499 v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
500 cdn_vector, q->pitch_gain,
501 q->pitch_lag, q->frame.pfrac);
503 // pitch prefilter update
504 for (i = 0; i < 4; i++)
505 q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
507 v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
508 v_synthesis_filtered,
509 q->pitch_gain, q->pitch_lag,
510 q->frame.pfrac);
512 apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
513 } else {
514 memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17, 143 * sizeof(float));
515 memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
516 memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
517 memset(q->pitch_lag, 0, sizeof(q->pitch_lag));
522 * Reconstruct LPC coefficients from the line spectral pair frequencies
523 * and perform bandwidth expansion.
525 * @param lspf line spectral pair frequencies
526 * @param lpc linear predictive coding coefficients
528 * @note: bandwidth_expansion_coeff could be precalculated into a table
529 * but it seems to be slower on x86
531 * TIA/EIA/IS-733 2.4.3.3.5
533 static void lspf2lpc(const float *lspf, float *lpc)
535 double lsp[10];
536 double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
537 int i;
539 for (i = 0; i < 10; i++)
540 lsp[i] = cos(M_PI * lspf[i]);
542 ff_acelp_lspd2lpc(lsp, lpc, 5);
544 for (i = 0; i < 10; i++) {
545 lpc[i] *= bandwidth_expansion_coeff;
546 bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
551 * Interpolate LSP frequencies and compute LPC coefficients
552 * for a given bitrate & pitch subframe.
554 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
556 * @param q the context
557 * @param curr_lspf LSP frequencies vector of the current frame
558 * @param lpc float vector for the resulting LPC
559 * @param subframe_num frame number in decoded stream
561 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
562 float *lpc, const int subframe_num)
564 float interpolated_lspf[10];
565 float weight;
567 if (q->bitrate >= RATE_QUARTER)
568 weight = 0.25 * (subframe_num + 1);
569 else if (q->bitrate == RATE_OCTAVE && !subframe_num)
570 weight = 0.625;
571 else
572 weight = 1.0;
574 if (weight != 1.0) {
575 ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
576 weight, 1.0 - weight, 10);
577 lspf2lpc(interpolated_lspf, lpc);
578 } else if (q->bitrate >= RATE_QUARTER ||
579 (q->bitrate == I_F_Q && !subframe_num))
580 lspf2lpc(curr_lspf, lpc);
581 else if (q->bitrate == SILENCE && !subframe_num)
582 lspf2lpc(q->prev_lspf, lpc);
585 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
587 switch (buf_size) {
588 case 35: return RATE_FULL;
589 case 17: return RATE_HALF;
590 case 8: return RATE_QUARTER;
591 case 4: return RATE_OCTAVE;
592 case 1: return SILENCE;
595 return I_F_Q;
599 * Determine the bitrate from the frame size and/or the first byte of the frame.
601 * @param avctx the AV codec context
602 * @param buf_size length of the buffer
603 * @param buf the bufffer
605 * @return the bitrate on success,
606 * I_F_Q if the bitrate cannot be satisfactorily determined
608 * TIA/EIA/IS-733 2.4.8.7.1
610 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx,
611 const int buf_size,
612 const uint8_t **buf)
614 qcelp_packet_rate bitrate;
616 if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
617 if (bitrate > **buf) {
618 QCELPContext *q = avctx->priv_data;
619 if (!q->warned_buf_mismatch_bitrate) {
620 av_log(avctx, AV_LOG_WARNING,
621 "Claimed bitrate and buffer size mismatch.\n");
622 q->warned_buf_mismatch_bitrate = 1;
624 bitrate = **buf;
625 } else if (bitrate < **buf) {
626 av_log(avctx, AV_LOG_ERROR,
627 "Buffer is too small for the claimed bitrate.\n");
628 return I_F_Q;
630 (*buf)++;
631 } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
632 av_log(avctx, AV_LOG_WARNING,
633 "Bitrate byte is missing, guessing the bitrate from packet size.\n");
634 } else
635 return I_F_Q;
637 if (bitrate == SILENCE) {
638 //FIXME: Remove experimental warning when tested with samples.
639 av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
641 return bitrate;
644 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
645 const char *message)
647 av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n",
648 avctx->frame_number, message);
651 static void postfilter(QCELPContext *q, float *samples, float *lpc)
653 static const float pow_0_775[10] = {
654 0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
655 0.216676, 0.167924, 0.130141, 0.100859, 0.078166
656 }, pow_0_625[10] = {
657 0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
658 0.059605, 0.037253, 0.023283, 0.014552, 0.009095
660 float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
661 int n;
663 for (n = 0; n < 10; n++) {
664 lpc_s[n] = lpc[n] * pow_0_625[n];
665 lpc_p[n] = lpc[n] * pow_0_775[n];
668 ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
669 q->formant_mem + 10, 160, 10);
670 memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
671 ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
672 memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
674 ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
676 ff_adaptive_gain_control(samples, pole_out + 10,
677 avpriv_scalarproduct_float_c(q->formant_mem + 10,
678 q->formant_mem + 10,
679 160),
680 160, 0.9375, &q->postfilter_agc_mem);
683 static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
684 int *got_frame_ptr, AVPacket *avpkt)
686 const uint8_t *buf = avpkt->data;
687 int buf_size = avpkt->size;
688 QCELPContext *q = avctx->priv_data;
689 AVFrame *frame = data;
690 float *outbuffer;
691 int i, ret;
692 float quantized_lspf[10], lpc[10];
693 float gain[16];
694 float *formant_mem;
696 /* get output buffer */
697 frame->nb_samples = 160;
698 if ((ret = ff_get_buffer(avctx, frame)) < 0) {
699 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
700 return ret;
702 outbuffer = (float *)frame->data[0];
704 if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
705 warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
706 goto erasure;
709 if (q->bitrate == RATE_OCTAVE &&
710 (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
711 warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
712 goto erasure;
715 if (q->bitrate > SILENCE) {
716 const QCELPBitmap *bitmaps = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
717 const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +
718 qcelp_unpacking_bitmaps_lengths[q->bitrate];
719 uint8_t *unpacked_data = (uint8_t *)&q->frame;
721 init_get_bits(&q->gb, buf, 8 * buf_size);
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, outbuffer + i * 40, 40, 10);
774 formant_mem += 40;
777 // postfilter, as per TIA/EIA/IS-733 2.4.8.6
778 postfilter(q, outbuffer, lpc);
780 memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
782 memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
783 q->prev_bitrate = q->bitrate;
785 *got_frame_ptr = 1;
787 return buf_size;
790 AVCodec ff_qcelp_decoder = {
791 .name = "qcelp",
792 .type = AVMEDIA_TYPE_AUDIO,
793 .id = AV_CODEC_ID_QCELP,
794 .init = qcelp_decode_init,
795 .decode = qcelp_decode_frame,
796 .capabilities = CODEC_CAP_DR1,
797 .priv_data_size = sizeof(QCELPContext),
798 .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),