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"
34 #include "celp_math.h"
35 #include "celp_filters.h"
36 #include "acelp_filters.h"
37 #include "acelp_pitch_delay.h"
38 #include "acelp_vectors.h"
40 #include "g729postfilter.h"
43 * minimum quantized LSF value (3.2.4)
49 * maximum quantized LSF value (3.2.4)
52 #define LSFQ_MAX 25681
55 * minimum LSF distance (3.2.4)
58 #define LSFQ_DIFF_MIN 321
60 /// interpolation filter length
61 #define INTERPOL_LEN 11
64 * minimum gain pitch value (3.8, Equation 47)
67 #define SHARP_MIN 3277
70 * maximum gain pitch value (3.8, Equation 47)
71 * (EE) This does not comply with the specification.
72 * Specification says about 0.8, which should be
73 * 13107 in (1.14), but reference C code uses
74 * 13017 (equals to 0.7945) instead of it.
76 #define SHARP_MAX 13017
79 * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7.13)
81 #define MR_ENERGY 1018156
83 #define DECISION_NOISE 0
84 #define DECISION_INTERMEDIATE 1
85 #define DECISION_VOICE 2
94 uint8_t ac_index_bits[2]; ///< adaptive codebook index for second subframe (size in bits)
95 uint8_t parity_bit; ///< parity bit for pitch delay
96 uint8_t gc_1st_index_bits; ///< gain codebook (first stage) index (size in bits)
97 uint8_t gc_2nd_index_bits; ///< gain codebook (second stage) index (size in bits)
98 uint8_t fc_signs_bits; ///< number of pulses in fixed-codebook vector
99 uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook index entry
100 } G729FormatDescription;
106 /// past excitation signal buffer
107 int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
109 int16_t* exc; ///< start of past excitation data in buffer
110 int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
112 /// (2.13) LSP quantizer outputs
113 int16_t past_quantizer_output_buf[MA_NP + 1][10];
114 int16_t* past_quantizer_outputs[MA_NP + 1];
116 int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
117 int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
118 int16_t *lsp[2]; ///< pointers to lsp_buf
120 int16_t quant_energy[4]; ///< (5.10) past quantized energy
122 /// previous speech data for LP synthesis filter
123 int16_t syn_filter_data[10];
126 /// residual signal buffer (used in long-term postfilter)
127 int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
129 /// previous speech data for residual calculation filter
130 int16_t res_filter_data[SUBFRAME_SIZE+10];
132 /// previous speech data for short-term postfilter
133 int16_t pos_filter_data[SUBFRAME_SIZE+10];
135 /// (1.14) pitch gain of current and five previous subframes
136 int16_t past_gain_pitch[6];
138 /// (14.1) gain code from current and previous subframe
139 int16_t past_gain_code[2];
141 /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
142 int16_t voice_decision;
144 int16_t onset; ///< detected onset level (0-2)
145 int16_t was_periodic; ///< whether previous frame was declared as periodic or not (4.4)
146 int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
147 int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
148 uint16_t rand_value; ///< random number generator value (4.4.4)
149 int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame
151 /// (14.14) high-pass filter data (past input)
154 /// high-pass filter data (past output)
158 static const G729FormatDescription format_g729_8k = {
159 .ac_index_bits = {8,5},
161 .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
162 .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
164 .fc_indexes_bits = 13,
167 static const G729FormatDescription format_g729d_6k4 = {
168 .ac_index_bits = {8,4},
170 .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
171 .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
173 .fc_indexes_bits = 9,
177 * @brief pseudo random number generator
179 static inline uint16_t g729_prng(uint16_t value)
181 return 31821 * value + 13849;
185 * Get parity bit of bit 2..7
187 static inline int get_parity(uint8_t value)
189 return (0x6996966996696996ULL >> (value >> 2)) & 1;
193 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
194 * @param[out] lsfq (2.13) quantized LSF coefficients
195 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
196 * @param ma_predictor switched MA predictor of LSP quantizer
197 * @param vq_1st first stage vector of quantizer
198 * @param vq_2nd_low second stage lower vector of LSP quantizer
199 * @param vq_2nd_high second stage higher vector of LSP quantizer
201 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
202 int16_t ma_predictor,
203 int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
206 static const uint8_t min_distance[2]={10, 5}; //(2.13)
207 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
209 for (i = 0; i < 5; i++) {
210 quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
211 quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
214 for (j = 0; j < 2; j++) {
215 for (i = 1; i < 10; i++) {
216 int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
218 quantizer_output[i - 1] -= diff;
219 quantizer_output[i ] += diff;
224 for (i = 0; i < 10; i++) {
225 int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
226 for (j = 0; j < MA_NP; j++)
227 sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
232 ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
236 * Restores past LSP quantizer output using LSF from previous frame
237 * @param[in,out] lsfq (2.13) quantized LSF coefficients
238 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
239 * @param ma_predictor_prev MA predictor from previous frame
240 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
242 static void lsf_restore_from_previous(int16_t* lsfq,
243 int16_t* past_quantizer_outputs[MA_NP + 1],
244 int ma_predictor_prev)
246 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
249 for (i = 0; i < 10; i++) {
250 int tmp = lsfq[i] << 15;
252 for (k = 0; k < MA_NP; k++)
253 tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
255 quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
260 * Constructs new excitation signal and applies phase filter to it
261 * @param[out] out constructed speech signal
262 * @param in original excitation signal
263 * @param fc_cur (2.13) original fixed-codebook vector
264 * @param gain_code (14.1) gain code
265 * @param subframe_size length of the subframe
267 static void g729d_get_new_exc(
270 const int16_t* fc_cur,
276 int16_t fc_new[SUBFRAME_SIZE];
278 ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
280 for(i=0; i<subframe_size; i++)
283 out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
284 out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
289 * Makes decision about onset in current subframe
290 * @param past_onset decision result of previous subframe
291 * @param past_gain_code gain code of current and previous subframe
293 * @return onset decision result for current subframe
295 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
297 if((past_gain_code[0] >> 1) > past_gain_code[1])
300 return FFMAX(past_onset-1, 0);
304 * Makes decision about voice presence in current subframe
305 * @param onset onset level
306 * @param prev_voice_decision voice decision result from previous subframe
307 * @param past_gain_pitch pitch gain of current and previous subframes
309 * @return voice decision result for current subframe
311 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
313 int i, low_gain_pitch_cnt, voice_decision;
315 if(past_gain_pitch[0] >= 14745) // 0.9
316 voice_decision = DECISION_VOICE;
317 else if (past_gain_pitch[0] <= 9830) // 0.6
318 voice_decision = DECISION_NOISE;
320 voice_decision = DECISION_INTERMEDIATE;
322 for(i=0, low_gain_pitch_cnt=0; i<6; i++)
323 if(past_gain_pitch[i] < 9830)
324 low_gain_pitch_cnt++;
326 if(low_gain_pitch_cnt > 2 && !onset)
327 voice_decision = DECISION_NOISE;
329 if(!onset && voice_decision > prev_voice_decision + 1)
332 if(onset && voice_decision < DECISION_VOICE)
335 return voice_decision;
338 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
343 res += *v1++ * *v2++;
348 static av_cold int decoder_init(AVCodecContext * avctx)
350 G729Context* ctx = avctx->priv_data;
353 if (avctx->channels != 1) {
354 av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
355 return AVERROR(EINVAL);
357 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
359 /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
360 avctx->frame_size = SUBFRAME_SIZE << 1;
362 ctx->gain_coeff = 16384; // 1.0 in (1.14)
364 for (k = 0; k < MA_NP + 1; k++) {
365 ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
366 for (i = 1; i < 11; i++)
367 ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
370 ctx->lsp[0] = ctx->lsp_buf[0];
371 ctx->lsp[1] = ctx->lsp_buf[1];
372 memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
374 ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
376 ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
378 /* random seed initialization */
379 ctx->rand_value = 21845;
381 /* quantized prediction error */
383 ctx->quant_energy[i] = -14336; // -14 in (5.10)
385 ff_dsputil_init(&ctx->dsp, avctx);
386 ctx->dsp.scalarproduct_int16 = scalarproduct_int16_c;
388 avcodec_get_frame_defaults(&ctx->frame);
389 avctx->coded_frame = &ctx->frame;
394 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
397 const uint8_t *buf = avpkt->data;
398 int buf_size = avpkt->size;
401 const G729FormatDescription *format;
402 int frame_erasure = 0; ///< frame erasure detected during decoding
403 int bad_pitch = 0; ///< parity check failed
406 G729Formats packet_type;
407 G729Context *ctx = avctx->priv_data;
408 int16_t lp[2][11]; // (3.12)
409 uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
410 uint8_t quantizer_1st; ///< first stage vector of quantizer
411 uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
412 uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
414 int pitch_delay_int[2]; // pitch delay, integer part
415 int pitch_delay_3x; // pitch delay, multiplied by 3
416 int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
417 int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
419 int gain_before, gain_after;
420 int is_periodic = 0; // whether one of the subframes is declared as periodic or not
422 ctx->frame.nb_samples = SUBFRAME_SIZE<<1;
423 if ((ret = ff_get_buffer(avctx, &ctx->frame)) < 0) {
424 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
427 out_frame = (int16_t*) ctx->frame.data[0];
429 if (buf_size == 10) {
430 packet_type = FORMAT_G729_8K;
431 format = &format_g729_8k;
432 //Reset voice decision
434 ctx->voice_decision = DECISION_VOICE;
435 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
436 } else if (buf_size == 8) {
437 packet_type = FORMAT_G729D_6K4;
438 format = &format_g729d_6k4;
439 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
441 av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
442 return AVERROR_INVALIDDATA;
445 for (i=0; i < buf_size; i++)
446 frame_erasure |= buf[i];
447 frame_erasure = !frame_erasure;
449 init_get_bits(&gb, buf, 8*buf_size);
451 ma_predictor = get_bits(&gb, 1);
452 quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
453 quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
454 quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
457 lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
458 ctx->ma_predictor_prev);
460 lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
462 quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
463 ctx->ma_predictor_prev = ma_predictor;
466 tmp = ctx->past_quantizer_outputs[MA_NP];
467 memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
468 MA_NP * sizeof(int16_t*));
469 ctx->past_quantizer_outputs[0] = tmp;
471 ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
473 ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
475 FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
477 for (i = 0; i < 2; i++) {
478 int gain_corr_factor;
480 uint8_t ac_index; ///< adaptive codebook index
481 uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
482 int fc_indexes; ///< fixed-codebook indexes
483 uint8_t gc_1st_index; ///< gain codebook (first stage) index
484 uint8_t gc_2nd_index; ///< gain codebook (second stage) index
486 ac_index = get_bits(&gb, format->ac_index_bits[i]);
487 if(!i && format->parity_bit)
488 bad_pitch = get_parity(ac_index) == get_bits1(&gb);
489 fc_indexes = get_bits(&gb, format->fc_indexes_bits);
490 pulses_signs = get_bits(&gb, format->fc_signs_bits);
491 gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
492 gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
495 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
498 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
500 pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
502 int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
503 PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
505 if(packet_type == FORMAT_G729D_6K4)
506 pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
508 pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
511 /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
512 pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
513 if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
514 av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
515 pitch_delay_int[i] = PITCH_DELAY_MAX;
519 ctx->rand_value = g729_prng(ctx->rand_value);
520 fc_indexes = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1);
522 ctx->rand_value = g729_prng(ctx->rand_value);
523 pulses_signs = ctx->rand_value;
527 memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
528 switch (packet_type) {
530 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
531 ff_fc_4pulses_8bits_track_4,
532 fc_indexes, pulses_signs, 3, 3);
534 case FORMAT_G729D_6K4:
535 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
536 ff_fc_2pulses_9bits_track2_gray,
537 fc_indexes, pulses_signs, 1, 4);
542 This filter enhances harmonic components of the fixed-codebook vector to
543 improve the quality of the reconstructed speech.
545 / fc_v[i], i < pitch_delay
547 \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
549 ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
550 fc + pitch_delay_int[i],
552 av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
554 SUBFRAME_SIZE - pitch_delay_int[i]);
556 memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
557 ctx->past_gain_code[1] = ctx->past_gain_code[0];
560 ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
561 ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
563 gain_corr_factor = 0;
565 if (packet_type == FORMAT_G729D_6K4) {
566 ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
567 cb_gain_2nd_6k4[gc_2nd_index][0];
568 gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
569 cb_gain_2nd_6k4[gc_2nd_index][1];
571 /* Without check below overflow can occur in ff_acelp_update_past_gain.
572 It is not issue for G.729, because gain_corr_factor in it's case is always
573 greater than 1024, while in G.729D it can be even zero. */
574 gain_corr_factor = FFMAX(gain_corr_factor, 1024);
575 #ifndef G729_BITEXACT
576 gain_corr_factor >>= 1;
579 ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
580 cb_gain_2nd_8k[gc_2nd_index][0];
581 gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
582 cb_gain_2nd_8k[gc_2nd_index][1];
585 /* Decode the fixed-codebook gain. */
586 ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
593 This correction required to get bit-exact result with
594 reference code, because gain_corr_factor in G.729D is
595 two times larger than in original G.729.
597 If bit-exact result is not issue then gain_corr_factor
598 can be simpler divided by 2 before call to g729_get_gain_code
599 instead of using correction below.
601 if (packet_type == FORMAT_G729D_6K4) {
602 gain_corr_factor >>= 1;
603 ctx->past_gain_code[0] >>= 1;
607 ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
609 /* Routine requires rounding to lowest. */
610 ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
611 ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
612 ff_acelp_interp_filter, 6,
613 (pitch_delay_3x % 3) << 1,
616 ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
617 ctx->exc + i * SUBFRAME_SIZE, fc,
618 (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
619 ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
620 1 << 13, 14, SUBFRAME_SIZE);
622 memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
624 if (ff_celp_lp_synthesis_filter(
627 ctx->exc + i * SUBFRAME_SIZE,
633 /* Overflow occurred, downscale excitation signal... */
634 for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
635 ctx->exc_base[j] >>= 2;
637 /* ... and make synthesis again. */
638 if (packet_type == FORMAT_G729D_6K4) {
639 int16_t exc_new[SUBFRAME_SIZE];
641 ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
642 ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
644 g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
646 ff_celp_lp_synthesis_filter(
656 ff_celp_lp_synthesis_filter(
659 ctx->exc + i * SUBFRAME_SIZE,
666 /* Save data (without postfilter) for use in next subframe. */
667 memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
669 /* Calculate gain of unfiltered signal for use in AGC. */
671 for (j = 0; j < SUBFRAME_SIZE; j++)
672 gain_before += FFABS(synth[j+10]);
674 /* Call postfilter and also update voicing decision for use in next frame. */
682 ctx->res_filter_data,
683 ctx->pos_filter_data,
687 /* Calculate gain of filtered signal for use in AGC. */
689 for(j=0; j<SUBFRAME_SIZE; j++)
690 gain_after += FFABS(synth[j+10]);
692 ctx->gain_coeff = ff_g729_adaptive_gain_control(
700 ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
702 ctx->pitch_delay_int_prev = pitch_delay_int[i];
704 memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
705 ff_acelp_high_pass_filter(
706 out_frame + i*SUBFRAME_SIZE,
710 memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
713 ctx->was_periodic = is_periodic;
715 /* Save signal for use in next frame. */
716 memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
719 *(AVFrame*)data = ctx->frame;
723 AVCodec ff_g729_decoder = {
725 .type = AVMEDIA_TYPE_AUDIO,
726 .id = AV_CODEC_ID_G729,
727 .priv_data_size = sizeof(G729Context),
728 .init = decoder_init,
729 .decode = decode_frame,
730 .capabilities = CODEC_CAP_DR1,
731 .long_name = NULL_IF_CONFIG_SMALL("G.729"),