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/avassert.h"
33 #include "libavutil/channel_layout.h"
34 #include "libavutil/float_dsp.h"
38 #include "qcelpdata.h"
39 #include "celp_filters.h"
40 #include "acelp_filters.h"
41 #include "acelp_vectors.h"
45 I_F_Q = -1, /**< insufficient frame quality */
55 qcelp_packet_rate bitrate;
56 QCELPFrame frame; /**< unpacked data frame */
58 uint8_t erasure_count;
59 uint8_t octave_count; /**< count the consecutive RATE_OCTAVE frames */
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;
72 uint8_t warned_buf_mismatch_bitrate;
75 float postfilter_synth_mem[10];
76 float postfilter_agc_mem;
77 float postfilter_tilt_mem;
81 * Initialize the speech codec according to the specification.
83 * TIA/EIA/IS-733 2.4.9
85 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
87 QCELPContext *q = avctx->priv_data;
91 avctx->channel_layout = AV_CH_LAYOUT_MONO;
92 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
94 for (i = 0; i < 10; i++)
95 q->prev_lspf[i] = (i + 1) / 11.0;
101 * Decode the 10 quantized LSP frequencies from the LSPV/LSP
102 * transmission codes of any bitrate and check for badly received packets.
104 * @param q the context
105 * @param lspf line spectral pair frequencies
107 * @return 0 on success, -1 if the packet is badly received
109 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
111 static int decode_lspf(QCELPContext *q, float *lspf)
114 float tmp_lspf, smooth, erasure_coeff;
115 const float *predictors;
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
122 if (q->bitrate == RATE_OCTAVE) {
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);
132 smooth = q->octave_count < 10 ? .875 : 0.1;
134 erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
136 av_assert2(q->bitrate == I_F_Q);
138 if (q->erasure_count > 1)
139 erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;
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];
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);
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);
158 // Low-pass filter the LSP frequencies.
159 ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);
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;
169 // Check for badly received packets.
170 if (q->bitrate == RATE_QUARTER) {
171 if (lspf[9] <= .70 || lspf[9] >= .97)
173 for (i = 3; i < 10; i++)
174 if (fabs(lspf[i] - lspf[i - 2]) < .08)
177 if (lspf[9] <= .66 || lspf[9] >= .985)
179 for (i = 4; i < 10; i++)
180 if (fabs(lspf[i] - lspf[i - 4]) < .0931)
188 * Convert codebook transmission codes to GAIN and INDEX.
190 * @param q the context
191 * @param gain array holding the decoded gain
193 * TIA/EIA/IS-733 2.4.6.2
195 static void decode_gain_and_index(QCELPContext *q, float *gain)
197 int i, subframes_count, g1[16];
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;
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);
212 gain[i] = qcelp_g12ga[g1[i]];
214 if (q->frame.cbsign[i]) {
216 q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;
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]];
224 if (q->bitrate == RATE_QUARTER) {
225 // Provide smoothing of the unvoiced excitation energy.
227 gain[6] = 0.4 * gain[3] + 0.6 * gain[4];
229 gain[4] = 0.8 * gain[2] + 0.2 * gain[3];
230 gain[3] = 0.2 * gain[1] + 0.8 * gain[2];
232 gain[1] = 0.6 * gain[0] + 0.4 * gain[1];
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);
240 av_assert2(q->bitrate == I_F_Q);
242 g1[0] = q->prev_g1[1];
243 switch (q->erasure_count) {
245 case 2 : g1[0] -= 1; break;
246 case 3 : g1[0] -= 2; break;
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;
258 q->last_codebook_gain = gain[i - 2];
259 q->prev_g1[0] = q->prev_g1[1];
260 q->prev_g1[1] = g1[0];
265 * If the received packet is Rate 1/4 a further sanity check is made of the
268 * @param cbgain the unpacked cbgain array
269 * @return -1 if the sanity check fails, 0 otherwise
271 * TIA/EIA/IS-733 2.4.8.7.3
273 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
275 int i, diff, prev_diff = 0;
277 for (i = 1; i < 5; i++) {
278 diff = cbgain[i] - cbgain[i-1];
279 if (FFABS(diff) > 10)
281 else if (FFABS(diff - prev_diff) > 12)
289 * Compute the scaled codebook vector Cdn From INDEX and GAIN
292 * The specification lacks some information here.
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.
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.
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
309 static void compute_svector(QCELPContext *q, const float *gain,
313 uint16_t cbseed, cindex;
314 float *rnd, tmp_gain, fir_filter_value;
316 switch (q->bitrate) {
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 * qcelp_rate_full_codebook[cindex++ & 127];
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 * qcelp_rate_half_codebook[cindex++ & 127];
334 cbseed = (0x0003 & q->frame.lspv[4]) << 14 |
335 (0x003F & q->frame.lspv[3]) << 8 |
336 (0x0060 & q->frame.lspv[2]) << 1 |
337 (0x0007 & q->frame.lspv[1]) << 3 |
338 (0x0038 & q->frame.lspv[0]) >> 3;
339 rnd = q->rnd_fir_filter_mem + 20;
340 for (i = 0; i < 8; i++) {
341 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
342 for (k = 0; k < 20; k++) {
343 cbseed = 521 * cbseed + 259;
344 *rnd = (int16_t) cbseed;
347 fir_filter_value = 0.0;
348 for (j = 0; j < 10; j++)
349 fir_filter_value += qcelp_rnd_fir_coefs[j] *
350 (rnd[-j] + rnd[-20+j]);
352 fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
353 *cdn_vector++ = tmp_gain * fir_filter_value;
357 memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160,
361 cbseed = q->first16bits;
362 for (i = 0; i < 8; i++) {
363 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
364 for (j = 0; j < 20; j++) {
365 cbseed = 521 * cbseed + 259;
366 *cdn_vector++ = tmp_gain * (int16_t) cbseed;
371 cbseed = -44; // random codebook index
372 for (i = 0; i < 4; i++) {
373 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
374 for (j = 0; j < 40; j++)
375 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
379 memset(cdn_vector, 0, 160 * sizeof(float));
385 * Apply generic gain control.
387 * @param v_out output vector
388 * @param v_in gain-controlled vector
389 * @param v_ref vector to control gain of
391 * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
393 static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
397 for (i = 0; i < 160; i += 40) {
398 float res = avpriv_scalarproduct_float_c(v_ref + i, v_ref + i, 40);
399 ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40);
404 * Apply filter in pitch-subframe steps.
406 * @param memory buffer for the previous state of the filter
407 * - must be able to contain 303 elements
408 * - the 143 first elements are from the previous state
409 * - the next 160 are for output
410 * @param v_in input filter vector
411 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
412 * @param lag per-subframe lag array, each element is
413 * - between 16 and 143 if its corresponding pfrac is 0,
414 * - between 16 and 139 otherwise
415 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
418 * @return filter output vector
420 static const float *do_pitchfilter(float memory[303], const float v_in[160],
421 const float gain[4], const uint8_t *lag,
422 const uint8_t pfrac[4])
425 float *v_lag, *v_out;
428 v_out = memory + 143; // Output vector starts at memory[143].
430 for (i = 0; i < 4; i++) {
432 v_lag = memory + 143 + 40 * i - lag[i];
433 for (v_len = v_in + 40; v_in < v_len; v_in++) {
434 if (pfrac[i]) { // If it is a fractional lag...
435 for (j = 0, *v_out = 0.0; j < 4; j++)
436 *v_out += qcelp_hammsinc_table[j] * (v_lag[j - 4] + v_lag[3 - j]);
440 *v_out = *v_in + gain[i] * *v_out;
446 memcpy(v_out, v_in, 40 * sizeof(float));
452 memmove(memory, memory + 160, 143 * sizeof(float));
457 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
458 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
460 * @param q the context
461 * @param cdn_vector the scaled codebook vector
463 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
466 const float *v_synthesis_filtered, *v_pre_filtered;
468 if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||
469 (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
471 if (q->bitrate >= RATE_HALF) {
472 // Compute gain & lag for the whole frame.
473 for (i = 0; i < 4; i++) {
474 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
476 q->pitch_lag[i] = q->frame.plag[i] + 16;
479 float max_pitch_gain;
481 if (q->bitrate == I_F_Q) {
482 if (q->erasure_count < 3)
483 max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
485 max_pitch_gain = 0.0;
487 av_assert2(q->bitrate == SILENCE);
488 max_pitch_gain = 1.0;
490 for (i = 0; i < 4; i++)
491 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
493 memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
496 // pitch synthesis filter
497 v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
498 cdn_vector, q->pitch_gain,
499 q->pitch_lag, q->frame.pfrac);
501 // pitch prefilter update
502 for (i = 0; i < 4; i++)
503 q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
505 v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
506 v_synthesis_filtered,
507 q->pitch_gain, q->pitch_lag,
510 apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
512 memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17, 143 * sizeof(float));
513 memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
514 memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
515 memset(q->pitch_lag, 0, sizeof(q->pitch_lag));
520 * Reconstruct LPC coefficients from the line spectral pair frequencies
521 * and perform bandwidth expansion.
523 * @param lspf line spectral pair frequencies
524 * @param lpc linear predictive coding coefficients
526 * @note: bandwidth_expansion_coeff could be precalculated into a table
527 * but it seems to be slower on x86
529 * TIA/EIA/IS-733 2.4.3.3.5
531 static void lspf2lpc(const float *lspf, float *lpc)
534 double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
537 for (i = 0; i < 10; i++)
538 lsp[i] = cos(M_PI * lspf[i]);
540 ff_acelp_lspd2lpc(lsp, lpc, 5);
542 for (i = 0; i < 10; i++) {
543 lpc[i] *= bandwidth_expansion_coeff;
544 bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
549 * Interpolate LSP frequencies and compute LPC coefficients
550 * for a given bitrate & pitch subframe.
552 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
554 * @param q the context
555 * @param curr_lspf LSP frequencies vector of the current frame
556 * @param lpc float vector for the resulting LPC
557 * @param subframe_num frame number in decoded stream
559 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
560 float *lpc, const int subframe_num)
562 float interpolated_lspf[10];
565 if (q->bitrate >= RATE_QUARTER)
566 weight = 0.25 * (subframe_num + 1);
567 else if (q->bitrate == RATE_OCTAVE && !subframe_num)
573 ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
574 weight, 1.0 - weight, 10);
575 lspf2lpc(interpolated_lspf, lpc);
576 } else if (q->bitrate >= RATE_QUARTER ||
577 (q->bitrate == I_F_Q && !subframe_num))
578 lspf2lpc(curr_lspf, lpc);
579 else if (q->bitrate == SILENCE && !subframe_num)
580 lspf2lpc(q->prev_lspf, lpc);
583 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
586 case 35: return RATE_FULL;
587 case 17: return RATE_HALF;
588 case 8: return RATE_QUARTER;
589 case 4: return RATE_OCTAVE;
590 case 1: return SILENCE;
597 * Determine the bitrate from the frame size and/or the first byte of the frame.
599 * @param avctx the AV codec context
600 * @param buf_size length of the buffer
601 * @param buf the bufffer
603 * @return the bitrate on success,
604 * I_F_Q if the bitrate cannot be satisfactorily determined
606 * TIA/EIA/IS-733 2.4.8.7.1
608 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx,
612 qcelp_packet_rate bitrate;
614 if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
615 if (bitrate > **buf) {
616 QCELPContext *q = avctx->priv_data;
617 if (!q->warned_buf_mismatch_bitrate) {
618 av_log(avctx, AV_LOG_WARNING,
619 "Claimed bitrate and buffer size mismatch.\n");
620 q->warned_buf_mismatch_bitrate = 1;
623 } else if (bitrate < **buf) {
624 av_log(avctx, AV_LOG_ERROR,
625 "Buffer is too small for the claimed bitrate.\n");
629 } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
630 av_log(avctx, AV_LOG_WARNING,
631 "Bitrate byte is missing, guessing the bitrate from packet size.\n");
635 if (bitrate == SILENCE) {
636 // FIXME: Remove this warning when tested with samples.
637 avpriv_request_sample(avctx, "Blank frame handling");
642 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
645 av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n",
646 avctx->frame_number, message);
649 static void postfilter(QCELPContext *q, float *samples, float *lpc)
651 static const float pow_0_775[10] = {
652 0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
653 0.216676, 0.167924, 0.130141, 0.100859, 0.078166
655 0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
656 0.059605, 0.037253, 0.023283, 0.014552, 0.009095
658 float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
661 for (n = 0; n < 10; n++) {
662 lpc_s[n] = lpc[n] * pow_0_625[n];
663 lpc_p[n] = lpc[n] * pow_0_775[n];
666 ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
667 q->formant_mem + 10, 160, 10);
668 memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
669 ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
670 memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
672 ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
674 ff_adaptive_gain_control(samples, pole_out + 10,
675 avpriv_scalarproduct_float_c(q->formant_mem + 10,
678 160, 0.9375, &q->postfilter_agc_mem);
681 static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
682 int *got_frame_ptr, AVPacket *avpkt)
684 const uint8_t *buf = avpkt->data;
685 int buf_size = avpkt->size;
686 QCELPContext *q = avctx->priv_data;
687 AVFrame *frame = data;
690 float quantized_lspf[10], lpc[10];
694 /* get output buffer */
695 frame->nb_samples = 160;
696 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
698 outbuffer = (float *)frame->data[0];
700 if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
701 warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
705 if (q->bitrate == RATE_OCTAVE &&
706 (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
707 warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
711 if (q->bitrate > SILENCE) {
712 const QCELPBitmap *bitmaps = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
713 const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +
714 qcelp_unpacking_bitmaps_lengths[q->bitrate];
715 uint8_t *unpacked_data = (uint8_t *)&q->frame;
717 init_get_bits(&q->gb, buf, 8 * buf_size);
719 memset(&q->frame, 0, sizeof(QCELPFrame));
721 for (; bitmaps < bitmaps_end; bitmaps++)
722 unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
724 // Check for erasures/blanks on rates 1, 1/4 and 1/8.
725 if (q->frame.reserved) {
726 warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
729 if (q->bitrate == RATE_QUARTER &&
730 codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) {
731 warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
735 if (q->bitrate >= RATE_HALF) {
736 for (i = 0; i < 4; i++) {
737 if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
738 warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
745 decode_gain_and_index(q, gain);
746 compute_svector(q, gain, outbuffer);
748 if (decode_lspf(q, quantized_lspf) < 0) {
749 warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
753 apply_pitch_filters(q, outbuffer);
755 if (q->bitrate == I_F_Q) {
759 decode_gain_and_index(q, gain);
760 compute_svector(q, gain, outbuffer);
761 decode_lspf(q, quantized_lspf);
762 apply_pitch_filters(q, outbuffer);
764 q->erasure_count = 0;
766 formant_mem = q->formant_mem + 10;
767 for (i = 0; i < 4; i++) {
768 interpolate_lpc(q, quantized_lspf, lpc, i);
769 ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40, 10);
773 // postfilter, as per TIA/EIA/IS-733 2.4.8.6
774 postfilter(q, outbuffer, lpc);
776 memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
778 memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
779 q->prev_bitrate = q->bitrate;
786 AVCodec ff_qcelp_decoder = {
788 .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
789 .type = AVMEDIA_TYPE_AUDIO,
790 .id = AV_CODEC_ID_QCELP,
791 .init = qcelp_decode_init,
792 .decode = qcelp_decode_frame,
793 .capabilities = CODEC_CAP_DR1,
794 .priv_data_size = sizeof(QCELPContext),