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
25 * @author Reynaldo H. Verdejo Pinochet
26 * @remark FFmpeg merging spearheaded by Kenan Gillet
27 * @remark Development mentored by Benjamin Larson
32 #include "libavutil/channel_layout.h"
33 #include "libavutil/float_dsp.h"
37 #include "qcelpdata.h"
38 #include "celp_filters.h"
39 #include "acelp_filters.h"
40 #include "acelp_vectors.h"
47 I_F_Q = -1, /**< insufficient frame quality */
58 qcelp_packet_rate bitrate;
59 QCELPFrame frame; /**< unpacked data frame */
61 uint8_t erasure_count;
62 uint8_t octave_count; /**< count the consecutive RATE_OCTAVE frames */
64 float predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
65 float pitch_synthesis_filter_mem[303];
66 float pitch_pre_filter_mem[303];
67 float rnd_fir_filter_mem[180];
68 float formant_mem[170];
69 float last_codebook_gain;
75 uint8_t warned_buf_mismatch_bitrate;
78 float postfilter_synth_mem[10];
79 float postfilter_agc_mem;
80 float postfilter_tilt_mem;
84 * Initialize the speech codec according to the specification.
86 * TIA/EIA/IS-733 2.4.9
88 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
90 QCELPContext *q = avctx->priv_data;
94 avctx->channel_layout = AV_CH_LAYOUT_MONO;
95 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
97 for (i = 0; i < 10; i++)
98 q->prev_lspf[i] = (i + 1) / 11.;
100 avcodec_get_frame_defaults(&q->avframe);
101 avctx->coded_frame = &q->avframe;
107 * Decode the 10 quantized LSP frequencies from the LSPV/LSP
108 * transmission codes of any bitrate and check for badly received packets.
110 * @param q the context
111 * @param lspf line spectral pair frequencies
113 * @return 0 on success, -1 if the packet is badly received
115 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
117 static int decode_lspf(QCELPContext *q, float *lspf)
120 float tmp_lspf, smooth, erasure_coeff;
121 const float *predictors;
123 if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {
124 predictors = q->prev_bitrate != RATE_OCTAVE &&
125 q->prev_bitrate != I_F_Q ? q->prev_lspf
128 if (q->bitrate == RATE_OCTAVE) {
131 for (i = 0; i < 10; i++) {
132 q->predictor_lspf[i] =
133 lspf[i] = (q->frame.lspv[i] ? QCELP_LSP_SPREAD_FACTOR
134 : -QCELP_LSP_SPREAD_FACTOR) +
135 predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR +
136 (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11);
138 smooth = q->octave_count < 10 ? .875 : 0.1;
140 erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
142 assert(q->bitrate == I_F_Q);
144 if (q->erasure_count > 1)
145 erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;
147 for (i = 0; i < 10; i++) {
148 q->predictor_lspf[i] =
149 lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 +
150 erasure_coeff * predictors[i];
155 // Check the stability of the LSP frequencies.
156 lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
157 for (i = 1; i < 10; i++)
158 lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR);
160 lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR);
161 for (i = 9; i > 0; i--)
162 lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR);
164 // Low-pass filter the LSP frequencies.
165 ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);
170 for (i = 0; i < 5; i++) {
171 lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
172 lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
175 // Check for badly received packets.
176 if (q->bitrate == RATE_QUARTER) {
177 if (lspf[9] <= .70 || lspf[9] >= .97)
179 for (i = 3; i < 10; i++)
180 if (fabs(lspf[i] - lspf[i - 2]) < .08)
183 if (lspf[9] <= .66 || lspf[9] >= .985)
185 for (i = 4; i < 10; i++)
186 if (fabs(lspf[i] - lspf[i - 4]) < .0931)
194 * Convert codebook transmission codes to GAIN and INDEX.
196 * @param q the context
197 * @param gain array holding the decoded gain
199 * TIA/EIA/IS-733 2.4.6.2
201 static void decode_gain_and_index(QCELPContext *q, float *gain)
203 int i, subframes_count, g1[16];
206 if (q->bitrate >= RATE_QUARTER) {
207 switch (q->bitrate) {
208 case RATE_FULL: subframes_count = 16; break;
209 case RATE_HALF: subframes_count = 4; break;
210 default: subframes_count = 5;
212 for (i = 0; i < subframes_count; i++) {
213 g1[i] = 4 * q->frame.cbgain[i];
214 if (q->bitrate == RATE_FULL && !((i + 1) & 3)) {
215 g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32);
218 gain[i] = qcelp_g12ga[g1[i]];
220 if (q->frame.cbsign[i]) {
222 q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;
226 q->prev_g1[0] = g1[i - 2];
227 q->prev_g1[1] = g1[i - 1];
228 q->last_codebook_gain = qcelp_g12ga[g1[i - 1]];
230 if (q->bitrate == RATE_QUARTER) {
231 // Provide smoothing of the unvoiced excitation energy.
233 gain[6] = 0.4 * gain[3] + 0.6 * gain[4];
235 gain[4] = 0.8 * gain[2] + 0.2 * gain[3];
236 gain[3] = 0.2 * gain[1] + 0.8 * gain[2];
238 gain[1] = 0.6 * gain[0] + 0.4 * gain[1];
240 } else if (q->bitrate != SILENCE) {
241 if (q->bitrate == RATE_OCTAVE) {
242 g1[0] = 2 * q->frame.cbgain[0] +
243 av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
246 assert(q->bitrate == I_F_Q);
248 g1[0] = q->prev_g1[1];
249 switch (q->erasure_count) {
251 case 2 : g1[0] -= 1; break;
252 case 3 : g1[0] -= 2; break;
259 // This interpolation is done to produce smoother background noise.
260 slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
261 for (i = 1; i <= subframes_count; i++)
262 gain[i - 1] = q->last_codebook_gain + slope * i;
264 q->last_codebook_gain = gain[i - 2];
265 q->prev_g1[0] = q->prev_g1[1];
266 q->prev_g1[1] = g1[0];
271 * If the received packet is Rate 1/4 a further sanity check is made of the
274 * @param cbgain the unpacked cbgain array
275 * @return -1 if the sanity check fails, 0 otherwise
277 * TIA/EIA/IS-733 2.4.8.7.3
279 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
281 int i, diff, prev_diff = 0;
283 for (i = 1; i < 5; i++) {
284 diff = cbgain[i] - cbgain[i-1];
285 if (FFABS(diff) > 10)
287 else if (FFABS(diff - prev_diff) > 12)
295 * Compute the scaled codebook vector Cdn From INDEX and GAIN
298 * The specification lacks some information here.
300 * TIA/EIA/IS-733 has an omission on the codebook index determination
301 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
302 * you have to subtract the decoded index parameter from the given scaled
303 * codebook vector index 'n' to get the desired circular codebook index, but
304 * it does not mention that you have to clamp 'n' to [0-9] in order to get
305 * RI-compliant results.
307 * The reason for this mistake seems to be the fact they forgot to mention you
308 * have to do these calculations per codebook subframe and adjust given
309 * equation values accordingly.
311 * @param q the context
312 * @param gain array holding the 4 pitch subframe gain values
313 * @param cdn_vector array for the generated scaled codebook vector
315 static void compute_svector(QCELPContext *q, const float *gain,
319 uint16_t cbseed, cindex;
320 float *rnd, tmp_gain, fir_filter_value;
322 switch (q->bitrate) {
324 for (i = 0; i < 16; i++) {
325 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
326 cindex = -q->frame.cindex[i];
327 for (j = 0; j < 10; j++)
328 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
332 for (i = 0; i < 4; i++) {
333 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
334 cindex = -q->frame.cindex[i];
335 for (j = 0; j < 40; j++)
336 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
340 cbseed = (0x0003 & q->frame.lspv[4]) << 14 |
341 (0x003F & q->frame.lspv[3]) << 8 |
342 (0x0060 & q->frame.lspv[2]) << 1 |
343 (0x0007 & q->frame.lspv[1]) << 3 |
344 (0x0038 & q->frame.lspv[0]) >> 3;
345 rnd = q->rnd_fir_filter_mem + 20;
346 for (i = 0; i < 8; i++) {
347 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
348 for (k = 0; k < 20; k++) {
349 cbseed = 521 * cbseed + 259;
350 *rnd = (int16_t) cbseed;
353 fir_filter_value = 0.0;
354 for (j = 0; j < 10; j++)
355 fir_filter_value += qcelp_rnd_fir_coefs[j] *
356 (rnd[-j] + rnd[-20+j]);
358 fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
359 *cdn_vector++ = tmp_gain * fir_filter_value;
363 memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160,
367 cbseed = q->first16bits;
368 for (i = 0; i < 8; i++) {
369 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
370 for (j = 0; j < 20; j++) {
371 cbseed = 521 * cbseed + 259;
372 *cdn_vector++ = tmp_gain * (int16_t) cbseed;
377 cbseed = -44; // random codebook index
378 for (i = 0; i < 4; i++) {
379 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
380 for (j = 0; j < 40; j++)
381 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
385 memset(cdn_vector, 0, 160 * sizeof(float));
391 * Apply generic gain control.
393 * @param v_out output vector
394 * @param v_in gain-controlled vector
395 * @param v_ref vector to control gain of
397 * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
399 static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
403 for (i = 0; i < 160; i += 40) {
404 float res = avpriv_scalarproduct_float_c(v_ref + i, v_ref + i, 40);
405 ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40);
410 * Apply filter in pitch-subframe steps.
412 * @param memory buffer for the previous state of the filter
413 * - must be able to contain 303 elements
414 * - the 143 first elements are from the previous state
415 * - the next 160 are for output
416 * @param v_in input filter vector
417 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
418 * @param lag per-subframe lag array, each element is
419 * - between 16 and 143 if its corresponding pfrac is 0,
420 * - between 16 and 139 otherwise
421 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
424 * @return filter output vector
426 static const float *do_pitchfilter(float memory[303], const float v_in[160],
427 const float gain[4], const uint8_t *lag,
428 const uint8_t pfrac[4])
431 float *v_lag, *v_out;
434 v_out = memory + 143; // Output vector starts at memory[143].
436 for (i = 0; i < 4; i++) {
438 v_lag = memory + 143 + 40 * i - lag[i];
439 for (v_len = v_in + 40; v_in < v_len; v_in++) {
440 if (pfrac[i]) { // If it is a fractional lag...
441 for (j = 0, *v_out = 0.; j < 4; j++)
442 *v_out += qcelp_hammsinc_table[j] * (v_lag[j - 4] + v_lag[3 - j]);
446 *v_out = *v_in + gain[i] * *v_out;
452 memcpy(v_out, v_in, 40 * sizeof(float));
458 memmove(memory, memory + 160, 143 * sizeof(float));
463 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
464 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
466 * @param q the context
467 * @param cdn_vector the scaled codebook vector
469 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
472 const float *v_synthesis_filtered, *v_pre_filtered;
474 if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||
475 (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
477 if (q->bitrate >= RATE_HALF) {
478 // Compute gain & lag for the whole frame.
479 for (i = 0; i < 4; i++) {
480 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
482 q->pitch_lag[i] = q->frame.plag[i] + 16;
485 float max_pitch_gain;
487 if (q->bitrate == I_F_Q) {
488 if (q->erasure_count < 3)
489 max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
491 max_pitch_gain = 0.0;
493 assert(q->bitrate == SILENCE);
494 max_pitch_gain = 1.0;
496 for (i = 0; i < 4; i++)
497 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
499 memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
502 // pitch synthesis filter
503 v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
504 cdn_vector, q->pitch_gain,
505 q->pitch_lag, q->frame.pfrac);
507 // pitch prefilter update
508 for (i = 0; i < 4; i++)
509 q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
511 v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
512 v_synthesis_filtered,
513 q->pitch_gain, q->pitch_lag,
516 apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
518 memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17, 143 * sizeof(float));
519 memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
520 memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
521 memset(q->pitch_lag, 0, sizeof(q->pitch_lag));
526 * Reconstruct LPC coefficients from the line spectral pair frequencies
527 * and perform bandwidth expansion.
529 * @param lspf line spectral pair frequencies
530 * @param lpc linear predictive coding coefficients
532 * @note: bandwidth_expansion_coeff could be precalculated into a table
533 * but it seems to be slower on x86
535 * TIA/EIA/IS-733 2.4.3.3.5
537 static void lspf2lpc(const float *lspf, float *lpc)
540 double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
543 for (i = 0; i < 10; i++)
544 lsp[i] = cos(M_PI * lspf[i]);
546 ff_acelp_lspd2lpc(lsp, lpc, 5);
548 for (i = 0; i < 10; i++) {
549 lpc[i] *= bandwidth_expansion_coeff;
550 bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
555 * Interpolate LSP frequencies and compute LPC coefficients
556 * for a given bitrate & pitch subframe.
558 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
560 * @param q the context
561 * @param curr_lspf LSP frequencies vector of the current frame
562 * @param lpc float vector for the resulting LPC
563 * @param subframe_num frame number in decoded stream
565 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
566 float *lpc, const int subframe_num)
568 float interpolated_lspf[10];
571 if (q->bitrate >= RATE_QUARTER)
572 weight = 0.25 * (subframe_num + 1);
573 else if (q->bitrate == RATE_OCTAVE && !subframe_num)
579 ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
580 weight, 1.0 - weight, 10);
581 lspf2lpc(interpolated_lspf, lpc);
582 } else if (q->bitrate >= RATE_QUARTER ||
583 (q->bitrate == I_F_Q && !subframe_num))
584 lspf2lpc(curr_lspf, lpc);
585 else if (q->bitrate == SILENCE && !subframe_num)
586 lspf2lpc(q->prev_lspf, lpc);
589 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
592 case 35: return RATE_FULL;
593 case 17: return RATE_HALF;
594 case 8: return RATE_QUARTER;
595 case 4: return RATE_OCTAVE;
596 case 1: return SILENCE;
603 * Determine the bitrate from the frame size and/or the first byte of the frame.
605 * @param avctx the AV codec context
606 * @param buf_size length of the buffer
607 * @param buf the bufffer
609 * @return the bitrate on success,
610 * I_F_Q if the bitrate cannot be satisfactorily determined
612 * TIA/EIA/IS-733 2.4.8.7.1
614 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx,
618 qcelp_packet_rate bitrate;
620 if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
621 if (bitrate > **buf) {
622 QCELPContext *q = avctx->priv_data;
623 if (!q->warned_buf_mismatch_bitrate) {
624 av_log(avctx, AV_LOG_WARNING,
625 "Claimed bitrate and buffer size mismatch.\n");
626 q->warned_buf_mismatch_bitrate = 1;
629 } else if (bitrate < **buf) {
630 av_log(avctx, AV_LOG_ERROR,
631 "Buffer is too small for the claimed bitrate.\n");
635 } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
636 av_log(avctx, AV_LOG_WARNING,
637 "Bitrate byte is missing, guessing the bitrate from packet size.\n");
641 if (bitrate == SILENCE) {
642 //FIXME: Remove experimental warning when tested with samples.
643 av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
648 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
651 av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n",
652 avctx->frame_number, message);
655 static void postfilter(QCELPContext *q, float *samples, float *lpc)
657 static const float pow_0_775[10] = {
658 0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
659 0.216676, 0.167924, 0.130141, 0.100859, 0.078166
661 0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
662 0.059605, 0.037253, 0.023283, 0.014552, 0.009095
664 float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
667 for (n = 0; n < 10; n++) {
668 lpc_s[n] = lpc[n] * pow_0_625[n];
669 lpc_p[n] = lpc[n] * pow_0_775[n];
672 ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
673 q->formant_mem + 10, 160, 10);
674 memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
675 ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
676 memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
678 ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
680 ff_adaptive_gain_control(samples, pole_out + 10,
681 avpriv_scalarproduct_float_c(q->formant_mem + 10,
684 160, 0.9375, &q->postfilter_agc_mem);
687 static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
688 int *got_frame_ptr, AVPacket *avpkt)
690 const uint8_t *buf = avpkt->data;
691 int buf_size = avpkt->size;
692 QCELPContext *q = avctx->priv_data;
695 float quantized_lspf[10], lpc[10];
699 /* get output buffer */
700 q->avframe.nb_samples = 160;
701 if ((ret = ff_get_buffer(avctx, &q->avframe)) < 0) {
702 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
705 outbuffer = (float *)q->avframe.data[0];
707 if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
708 warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
712 if (q->bitrate == RATE_OCTAVE &&
713 (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
714 warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
718 if (q->bitrate > SILENCE) {
719 const QCELPBitmap *bitmaps = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
720 const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +
721 qcelp_unpacking_bitmaps_lengths[q->bitrate];
722 uint8_t *unpacked_data = (uint8_t *)&q->frame;
724 init_get_bits(&q->gb, buf, 8 * buf_size);
726 memset(&q->frame, 0, sizeof(QCELPFrame));
728 for (; bitmaps < bitmaps_end; bitmaps++)
729 unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
731 // Check for erasures/blanks on rates 1, 1/4 and 1/8.
732 if (q->frame.reserved) {
733 warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
736 if (q->bitrate == RATE_QUARTER &&
737 codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) {
738 warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
742 if (q->bitrate >= RATE_HALF) {
743 for (i = 0; i < 4; i++) {
744 if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
745 warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
752 decode_gain_and_index(q, gain);
753 compute_svector(q, gain, outbuffer);
755 if (decode_lspf(q, quantized_lspf) < 0) {
756 warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
760 apply_pitch_filters(q, outbuffer);
762 if (q->bitrate == I_F_Q) {
766 decode_gain_and_index(q, gain);
767 compute_svector(q, gain, outbuffer);
768 decode_lspf(q, quantized_lspf);
769 apply_pitch_filters(q, outbuffer);
771 q->erasure_count = 0;
773 formant_mem = q->formant_mem + 10;
774 for (i = 0; i < 4; i++) {
775 interpolate_lpc(q, quantized_lspf, lpc, i);
776 ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40, 10);
780 // postfilter, as per TIA/EIA/IS-733 2.4.8.6
781 postfilter(q, outbuffer, lpc);
783 memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
785 memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
786 q->prev_bitrate = q->bitrate;
789 *(AVFrame *)data = q->avframe;
794 AVCodec ff_qcelp_decoder = {
796 .type = AVMEDIA_TYPE_AUDIO,
797 .id = AV_CODEC_ID_QCELP,
798 .init = qcelp_decode_init,
799 .decode = qcelp_decode_frame,
800 .capabilities = CODEC_CAP_DR1,
801 .priv_data_size = sizeof(QCELPContext),
802 .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),