2 * G.729, G729 Annex D decoders
3 * Copyright (c) 2008 Vladimir Voroshilov
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
26 #include "libavutil/avutil.h"
32 #include "celp_math.h"
33 #include "celp_filters.h"
34 #include "acelp_filters.h"
35 #include "acelp_pitch_delay.h"
36 #include "acelp_vectors.h"
38 #include "g729postfilter.h"
41 * minimum quantized LSF value (3.2.4)
47 * maximum quantized LSF value (3.2.4)
50 #define LSFQ_MAX 25681
53 * minimum LSF distance (3.2.4)
56 #define LSFQ_DIFF_MIN 321
58 /// interpolation filter length
59 #define INTERPOL_LEN 11
62 * minimum gain pitch value (3.8, Equation 47)
65 #define SHARP_MIN 3277
68 * maximum gain pitch value (3.8, Equation 47)
69 * (EE) This does not comply with the specification.
70 * Specification says about 0.8, which should be
71 * 13107 in (1.14), but reference C code uses
72 * 13017 (equals to 0.7945) instead of it.
74 #define SHARP_MAX 13017
77 * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7.13)
79 #define MR_ENERGY 1018156
81 #define DECISION_NOISE 0
82 #define DECISION_INTERMEDIATE 1
83 #define DECISION_VOICE 2
92 uint8_t ac_index_bits[2]; ///< adaptive codebook index for second subframe (size in bits)
93 uint8_t parity_bit; ///< parity bit for pitch delay
94 uint8_t gc_1st_index_bits; ///< gain codebook (first stage) index (size in bits)
95 uint8_t gc_2nd_index_bits; ///< gain codebook (second stage) index (size in bits)
96 uint8_t fc_signs_bits; ///< number of pulses in fixed-codebook vector
97 uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook index entry
98 } G729FormatDescription;
104 /// past excitation signal buffer
105 int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
107 int16_t* exc; ///< start of past excitation data in buffer
108 int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
110 /// (2.13) LSP quantizer outputs
111 int16_t past_quantizer_output_buf[MA_NP + 1][10];
112 int16_t* past_quantizer_outputs[MA_NP + 1];
114 int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
115 int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
116 int16_t *lsp[2]; ///< pointers to lsp_buf
118 int16_t quant_energy[4]; ///< (5.10) past quantized energy
120 /// previous speech data for LP synthesis filter
121 int16_t syn_filter_data[10];
124 /// residual signal buffer (used in long-term postfilter)
125 int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
127 /// previous speech data for residual calculation filter
128 int16_t res_filter_data[SUBFRAME_SIZE+10];
130 /// previous speech data for short-term postfilter
131 int16_t pos_filter_data[SUBFRAME_SIZE+10];
133 /// (1.14) pitch gain of current and five previous subframes
134 int16_t past_gain_pitch[6];
136 /// (14.1) gain code from current and previous subframe
137 int16_t past_gain_code[2];
139 /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
140 int16_t voice_decision;
142 int16_t onset; ///< detected onset level (0-2)
143 int16_t was_periodic; ///< whether previous frame was declared as periodic or not (4.4)
144 int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
145 int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
146 uint16_t rand_value; ///< random number generator value (4.4.4)
147 int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame
149 /// (14.14) high-pass filter data (past input)
152 /// high-pass filter data (past output)
156 static const G729FormatDescription format_g729_8k = {
157 .ac_index_bits = {8,5},
159 .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
160 .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
162 .fc_indexes_bits = 13,
165 static const G729FormatDescription format_g729d_6k4 = {
166 .ac_index_bits = {8,4},
168 .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
169 .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
171 .fc_indexes_bits = 9,
175 * @brief pseudo random number generator
177 static inline uint16_t g729_prng(uint16_t value)
179 return 31821 * value + 13849;
183 * Get parity bit of bit 2..7
185 static inline int get_parity(uint8_t value)
187 return (0x6996966996696996ULL >> (value >> 2)) & 1;
191 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
192 * @param lsfq [out] (2.13) quantized LSF coefficients
193 * @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
194 * @param ma_predictor switched MA predictor of LSP quantizer
195 * @param vq_1st first stage vector of quantizer
196 * @param vq_2nd_low second stage lower vector of LSP quantizer
197 * @param vq_2nd_high second stage higher vector of LSP quantizer
199 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
200 int16_t ma_predictor,
201 int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
204 static const uint8_t min_distance[2]={10, 5}; //(2.13)
205 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
207 for (i = 0; i < 5; i++) {
208 quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
209 quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
212 for (j = 0; j < 2; j++) {
213 for (i = 1; i < 10; i++) {
214 int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
216 quantizer_output[i - 1] -= diff;
217 quantizer_output[i ] += diff;
222 for (i = 0; i < 10; i++) {
223 int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
224 for (j = 0; j < MA_NP; j++)
225 sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
230 ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
234 * Restores past LSP quantizer output using LSF from previous frame
235 * @param lsfq [in/out] (2.13) quantized LSF coefficients
236 * @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
237 * @param ma_predictor_prev MA predictor from previous frame
238 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
240 static void lsf_restore_from_previous(int16_t* lsfq,
241 int16_t* past_quantizer_outputs[MA_NP + 1],
242 int ma_predictor_prev)
244 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
247 for (i = 0; i < 10; i++) {
248 int tmp = lsfq[i] << 15;
250 for (k = 0; k < MA_NP; k++)
251 tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
253 quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
258 * Constructs new excitation signal and applies phase filter to it
259 * @param out[out] constructed speech signal
260 * @param in original excitation signal
261 * @param fc_cur (2.13) original fixed-codebook vector
262 * @param gain_code (14.1) gain code
263 * @param subframe_size length of the subframe
265 static void g729d_get_new_exc(
268 const int16_t* fc_cur,
274 int16_t fc_new[SUBFRAME_SIZE];
276 ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
278 for(i=0; i<subframe_size; i++)
281 out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
282 out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
287 * Makes decision about onset in current subframe
288 * @param past_onset decision result of previous subframe
289 * @param past_gain_code gain code of current and previous subframe
291 * @return onset decision result for current subframe
293 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
295 if((past_gain_code[0] >> 1) > past_gain_code[1])
298 return FFMAX(past_onset-1, 0);
302 * Makes decision about voice presence in current subframe
303 * @param onset onset level
304 * @param prev_voice_decision voice decision result from previous subframe
305 * @param past_gain_pitch pitch gain of current and previous subframes
307 * @return voice decision result for current subframe
309 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
311 int i, low_gain_pitch_cnt, voice_decision;
313 if(past_gain_pitch[0] >= 14745) // 0.9
314 voice_decision = DECISION_VOICE;
315 else if (past_gain_pitch[0] <= 9830) // 0.6
316 voice_decision = DECISION_NOISE;
318 voice_decision = DECISION_INTERMEDIATE;
320 for(i=0, low_gain_pitch_cnt=0; i<6; i++)
321 if(past_gain_pitch[i] < 9830)
322 low_gain_pitch_cnt++;
324 if(low_gain_pitch_cnt > 2 && !onset)
325 voice_decision = DECISION_NOISE;
327 if(!onset && voice_decision > prev_voice_decision + 1)
330 if(onset && voice_decision < DECISION_VOICE)
333 return voice_decision;
336 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
341 res += *v1++ * *v2++;
346 static av_cold int decoder_init(AVCodecContext * avctx)
348 G729Context* ctx = avctx->priv_data;
351 if (avctx->channels != 1) {
352 av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
353 return AVERROR(EINVAL);
355 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
357 /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
358 avctx->frame_size = SUBFRAME_SIZE << 1;
360 ctx->gain_coeff = 16384; // 1.0 in (1.14)
362 for (k = 0; k < MA_NP + 1; k++) {
363 ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
364 for (i = 1; i < 11; i++)
365 ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
368 ctx->lsp[0] = ctx->lsp_buf[0];
369 ctx->lsp[1] = ctx->lsp_buf[1];
370 memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
372 ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
374 ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
376 /* random seed initialization */
377 ctx->rand_value = 21845;
379 /* quantized prediction error */
381 ctx->quant_energy[i] = -14336; // -14 in (5.10)
383 ff_dsputil_init(&ctx->dsp, avctx);
384 ctx->dsp.scalarproduct_int16 = scalarproduct_int16_c;
386 avcodec_get_frame_defaults(&ctx->frame);
387 avctx->coded_frame = &ctx->frame;
392 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
395 const uint8_t *buf = avpkt->data;
396 int buf_size = avpkt->size;
399 const G729FormatDescription *format;
400 int frame_erasure = 0; ///< frame erasure detected during decoding
401 int bad_pitch = 0; ///< parity check failed
404 G729Formats packet_type;
405 G729Context *ctx = avctx->priv_data;
406 int16_t lp[2][11]; // (3.12)
407 uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
408 uint8_t quantizer_1st; ///< first stage vector of quantizer
409 uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
410 uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
412 int pitch_delay_int[2]; // pitch delay, integer part
413 int pitch_delay_3x; // pitch delay, multiplied by 3
414 int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
415 int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
417 int gain_before, gain_after;
418 int is_periodic = 0; // whether one of the subframes is declared as periodic or not
420 ctx->frame.nb_samples = SUBFRAME_SIZE<<1;
421 if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
422 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
425 out_frame = (int16_t*) ctx->frame.data[0];
427 if (buf_size == 10) {
428 packet_type = FORMAT_G729_8K;
429 format = &format_g729_8k;
430 //Reset voice decision
432 ctx->voice_decision = DECISION_VOICE;
433 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
434 } else if (buf_size == 8) {
435 packet_type = FORMAT_G729D_6K4;
436 format = &format_g729d_6k4;
437 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
439 av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
440 return AVERROR_INVALIDDATA;
443 for (i=0; i < buf_size; i++)
444 frame_erasure |= buf[i];
445 frame_erasure = !frame_erasure;
447 init_get_bits(&gb, buf, 8*buf_size);
449 ma_predictor = get_bits(&gb, 1);
450 quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
451 quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
452 quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
455 lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
456 ctx->ma_predictor_prev);
458 lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
460 quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
461 ctx->ma_predictor_prev = ma_predictor;
464 tmp = ctx->past_quantizer_outputs[MA_NP];
465 memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
466 MA_NP * sizeof(int16_t*));
467 ctx->past_quantizer_outputs[0] = tmp;
469 ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
471 ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
473 FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
475 for (i = 0; i < 2; i++) {
476 int gain_corr_factor;
478 uint8_t ac_index; ///< adaptive codebook index
479 uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
480 int fc_indexes; ///< fixed-codebook indexes
481 uint8_t gc_1st_index; ///< gain codebook (first stage) index
482 uint8_t gc_2nd_index; ///< gain codebook (second stage) index
484 ac_index = get_bits(&gb, format->ac_index_bits[i]);
485 if(!i && format->parity_bit)
486 bad_pitch = get_parity(ac_index) == get_bits1(&gb);
487 fc_indexes = get_bits(&gb, format->fc_indexes_bits);
488 pulses_signs = get_bits(&gb, format->fc_signs_bits);
489 gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
490 gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
493 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
496 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
498 pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
500 int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
501 PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
503 if(packet_type == FORMAT_G729D_6K4)
504 pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
506 pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
509 /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
510 pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
513 ctx->rand_value = g729_prng(ctx->rand_value);
514 fc_indexes = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1);
516 ctx->rand_value = g729_prng(ctx->rand_value);
517 pulses_signs = ctx->rand_value;
521 memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
522 switch (packet_type) {
524 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
525 ff_fc_4pulses_8bits_track_4,
526 fc_indexes, pulses_signs, 3, 3);
528 case FORMAT_G729D_6K4:
529 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
530 ff_fc_2pulses_9bits_track2_gray,
531 fc_indexes, pulses_signs, 1, 4);
536 This filter enhances harmonic components of the fixed-codebook vector to
537 improve the quality of the reconstructed speech.
539 / fc_v[i], i < pitch_delay
541 \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
543 ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
544 fc + pitch_delay_int[i],
546 av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
548 SUBFRAME_SIZE - pitch_delay_int[i]);
550 memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
551 ctx->past_gain_code[1] = ctx->past_gain_code[0];
554 ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
555 ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
557 gain_corr_factor = 0;
559 if (packet_type == FORMAT_G729D_6K4) {
560 ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
561 cb_gain_2nd_6k4[gc_2nd_index][0];
562 gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
563 cb_gain_2nd_6k4[gc_2nd_index][1];
565 /* Without check below overflow can occur in ff_acelp_update_past_gain.
566 It is not issue for G.729, because gain_corr_factor in it's case is always
567 greater than 1024, while in G.729D it can be even zero. */
568 gain_corr_factor = FFMAX(gain_corr_factor, 1024);
569 #ifndef G729_BITEXACT
570 gain_corr_factor >>= 1;
573 ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
574 cb_gain_2nd_8k[gc_2nd_index][0];
575 gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
576 cb_gain_2nd_8k[gc_2nd_index][1];
579 /* Decode the fixed-codebook gain. */
580 ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
587 This correction required to get bit-exact result with
588 reference code, because gain_corr_factor in G.729D is
589 two times larger than in original G.729.
591 If bit-exact result is not issue then gain_corr_factor
592 can be simpler divided by 2 before call to g729_get_gain_code
593 instead of using correction below.
595 if (packet_type == FORMAT_G729D_6K4) {
596 gain_corr_factor >>= 1;
597 ctx->past_gain_code[0] >>= 1;
601 ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
603 /* Routine requires rounding to lowest. */
604 ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
605 ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
606 ff_acelp_interp_filter, 6,
607 (pitch_delay_3x % 3) << 1,
610 ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
611 ctx->exc + i * SUBFRAME_SIZE, fc,
612 (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
613 ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
614 1 << 13, 14, SUBFRAME_SIZE);
616 memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
618 if (ff_celp_lp_synthesis_filter(
621 ctx->exc + i * SUBFRAME_SIZE,
627 /* Overflow occurred, downscale excitation signal... */
628 for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
629 ctx->exc_base[j] >>= 2;
631 /* ... and make synthesis again. */
632 if (packet_type == FORMAT_G729D_6K4) {
633 int16_t exc_new[SUBFRAME_SIZE];
635 ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
636 ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
638 g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
640 ff_celp_lp_synthesis_filter(
650 ff_celp_lp_synthesis_filter(
653 ctx->exc + i * SUBFRAME_SIZE,
660 /* Save data (without postfilter) for use in next subframe. */
661 memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
663 /* Calculate gain of unfiltered signal for use in AGC. */
665 for (j = 0; j < SUBFRAME_SIZE; j++)
666 gain_before += FFABS(synth[j+10]);
668 /* Call postfilter and also update voicing decision for use in next frame. */
676 ctx->res_filter_data,
677 ctx->pos_filter_data,
681 /* Calculate gain of filtered signal for use in AGC. */
683 for(j=0; j<SUBFRAME_SIZE; j++)
684 gain_after += FFABS(synth[j+10]);
686 ctx->gain_coeff = ff_g729_adaptive_gain_control(
694 ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
696 ctx->pitch_delay_int_prev = pitch_delay_int[i];
698 memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
699 ff_acelp_high_pass_filter(
700 out_frame + i*SUBFRAME_SIZE,
704 memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
707 ctx->was_periodic = is_periodic;
709 /* Save signal for use in next frame. */
710 memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
713 *(AVFrame*)data = ctx->frame;
717 AVCodec ff_g729_decoder = {
719 .type = AVMEDIA_TYPE_AUDIO,
720 .id = AV_CODEC_ID_G729,
721 .priv_data_size = sizeof(G729Context),
722 .init = decoder_init,
723 .decode = decode_frame,
724 .capabilities = CODEC_CAP_DR1,
725 .long_name = NULL_IF_CONFIG_SMALL("G.729"),