2 * G.729, G729 Annex D decoders
3 * Copyright (c) 2008 Vladimir Voroshilov
5 * This file is part of FFmpeg.
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8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
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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;
103 /// past excitation signal buffer
104 int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
106 int16_t* exc; ///< start of past excitation data in buffer
107 int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
109 /// (2.13) LSP quantizer outputs
110 int16_t past_quantizer_output_buf[MA_NP + 1][10];
111 int16_t* past_quantizer_outputs[MA_NP + 1];
113 int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
114 int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
115 int16_t *lsp[2]; ///< pointers to lsp_buf
117 int16_t quant_energy[4]; ///< (5.10) past quantized energy
119 /// previous speech data for LP synthesis filter
120 int16_t syn_filter_data[10];
123 /// residual signal buffer (used in long-term postfilter)
124 int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
126 /// previous speech data for residual calculation filter
127 int16_t res_filter_data[SUBFRAME_SIZE+10];
129 /// previous speech data for short-term postfilter
130 int16_t pos_filter_data[SUBFRAME_SIZE+10];
132 /// (1.14) pitch gain of current and five previous subframes
133 int16_t past_gain_pitch[6];
135 /// (14.1) gain code from current and previous subframe
136 int16_t past_gain_code[2];
138 /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
139 int16_t voice_decision;
141 int16_t onset; ///< detected onset level (0-2)
142 int16_t was_periodic; ///< whether previous frame was declared as periodic or not (4.4)
143 int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
144 int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
145 uint16_t rand_value; ///< random number generator value (4.4.4)
146 int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame
148 /// (14.14) high-pass filter data (past input)
151 /// high-pass filter data (past output)
153 } G729ChannelContext;
156 AudioDSPContext adsp;
158 G729ChannelContext *channel_context;
161 static const G729FormatDescription format_g729_8k = {
162 .ac_index_bits = {8,5},
164 .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
165 .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
167 .fc_indexes_bits = 13,
170 static const G729FormatDescription format_g729d_6k4 = {
171 .ac_index_bits = {8,4},
173 .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
174 .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
176 .fc_indexes_bits = 9,
180 * @brief pseudo random number generator
182 static inline uint16_t g729_prng(uint16_t value)
184 return 31821 * value + 13849;
188 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
189 * @param[out] lsfq (2.13) quantized LSF coefficients
190 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
191 * @param ma_predictor switched MA predictor of LSP quantizer
192 * @param vq_1st first stage vector of quantizer
193 * @param vq_2nd_low second stage lower vector of LSP quantizer
194 * @param vq_2nd_high second stage higher vector of LSP quantizer
196 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
197 int16_t ma_predictor,
198 int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
201 static const uint8_t min_distance[2]={10, 5}; //(2.13)
202 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
204 for (i = 0; i < 5; i++) {
205 quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
206 quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
209 for (j = 0; j < 2; j++) {
210 for (i = 1; i < 10; i++) {
211 int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
213 quantizer_output[i - 1] -= diff;
214 quantizer_output[i ] += diff;
219 for (i = 0; i < 10; i++) {
220 int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
221 for (j = 0; j < MA_NP; j++)
222 sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
227 ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
231 * Restores past LSP quantizer output using LSF from previous frame
232 * @param[in,out] lsfq (2.13) quantized LSF coefficients
233 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
234 * @param ma_predictor_prev MA predictor from previous frame
235 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
237 static void lsf_restore_from_previous(int16_t* lsfq,
238 int16_t* past_quantizer_outputs[MA_NP + 1],
239 int ma_predictor_prev)
241 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
244 for (i = 0; i < 10; i++) {
245 int tmp = lsfq[i] << 15;
247 for (k = 0; k < MA_NP; k++)
248 tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
250 quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
255 * Constructs new excitation signal and applies phase filter to it
256 * @param[out] out constructed speech signal
257 * @param in original excitation signal
258 * @param fc_cur (2.13) original fixed-codebook vector
259 * @param gain_code (14.1) gain code
260 * @param subframe_size length of the subframe
262 static void g729d_get_new_exc(
265 const int16_t* fc_cur,
271 int16_t fc_new[SUBFRAME_SIZE];
273 ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
275 for (i = 0; i < subframe_size; i++) {
277 out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
278 out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
283 * Makes decision about onset in current subframe
284 * @param past_onset decision result of previous subframe
285 * @param past_gain_code gain code of current and previous subframe
287 * @return onset decision result for current subframe
289 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
291 if ((past_gain_code[0] >> 1) > past_gain_code[1])
294 return FFMAX(past_onset-1, 0);
298 * Makes decision about voice presence in current subframe
299 * @param onset onset level
300 * @param prev_voice_decision voice decision result from previous subframe
301 * @param past_gain_pitch pitch gain of current and previous subframes
303 * @return voice decision result for current subframe
305 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
307 int i, low_gain_pitch_cnt, voice_decision;
309 if (past_gain_pitch[0] >= 14745) { // 0.9
310 voice_decision = DECISION_VOICE;
311 } else if (past_gain_pitch[0] <= 9830) { // 0.6
312 voice_decision = DECISION_NOISE;
314 voice_decision = DECISION_INTERMEDIATE;
317 for (i = 0, low_gain_pitch_cnt = 0; i < 6; i++)
318 if (past_gain_pitch[i] < 9830)
319 low_gain_pitch_cnt++;
321 if (low_gain_pitch_cnt > 2 && !onset)
322 voice_decision = DECISION_NOISE;
324 if (!onset && voice_decision > prev_voice_decision + 1)
327 if (onset && voice_decision < DECISION_VOICE)
330 return voice_decision;
333 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
338 res += *v1++ * *v2++;
343 static av_cold int decoder_init(AVCodecContext * avctx)
345 G729Context *s = avctx->priv_data;
346 G729ChannelContext *ctx;
349 if (avctx->channels < 1 || avctx->channels > 2) {
350 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels);
351 return AVERROR(EINVAL);
353 avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
355 /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
356 avctx->frame_size = SUBFRAME_SIZE << 1;
359 s->channel_context = av_mallocz(sizeof(G729ChannelContext) * avctx->channels);
361 return AVERROR(ENOMEM);
363 for (c = 0; c < avctx->channels; c++) {
364 ctx->gain_coeff = 16384; // 1.0 in (1.14)
366 for (k = 0; k < MA_NP + 1; k++) {
367 ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
368 for (i = 1; i < 11; i++)
369 ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
372 ctx->lsp[0] = ctx->lsp_buf[0];
373 ctx->lsp[1] = ctx->lsp_buf[1];
374 memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
376 ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
378 ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
380 /* random seed initialization */
381 ctx->rand_value = 21845;
383 /* quantized prediction error */
384 for (i = 0; i < 4; i++)
385 ctx->quant_energy[i] = -14336; // -14 in (5.10)
390 ff_audiodsp_init(&s->adsp);
391 s->adsp.scalarproduct_int16 = scalarproduct_int16_c;
396 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
399 const uint8_t *buf = avpkt->data;
400 int buf_size = avpkt->size;
403 const G729FormatDescription *format;
406 G729Formats packet_type;
407 G729Context *s = avctx->priv_data;
408 G729ChannelContext *ctx = s->channel_context;
409 int16_t lp[2][11]; // (3.12)
410 uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
411 uint8_t quantizer_1st; ///< first stage vector of quantizer
412 uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
413 uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
415 int pitch_delay_int[2]; // pitch delay, integer part
416 int pitch_delay_3x; // pitch delay, multiplied by 3
417 int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
418 int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
420 int gain_before, gain_after;
421 AVFrame *frame = data;
423 frame->nb_samples = SUBFRAME_SIZE<<1;
424 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
427 if (buf_size % ((G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels) == 0) {
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 == G729D_6K4_BLOCK_SIZE * avctx->channels) {
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 (c = 0; c < avctx->channels; c++) {
444 int frame_erasure = 0; ///< frame erasure detected during decoding
445 int bad_pitch = 0; ///< parity check failed
446 int is_periodic = 0; ///< whether one of the subframes is declared as periodic or not
447 out_frame = (int16_t*)frame->data[c];
448 if (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN) {
449 if (*buf != ((avctx->channels - 1 - c) * 0x80 | 2))
450 avpriv_request_sample(avctx, "First byte value %x for channel %d", *buf, c);
454 for (i = 0; i < buf_size; i++)
455 frame_erasure |= buf[i];
456 frame_erasure = !frame_erasure;
458 init_get_bits(&gb, buf, 8*buf_size);
460 ma_predictor = get_bits(&gb, 1);
461 quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
462 quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
463 quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
466 lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
467 ctx->ma_predictor_prev);
469 lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
471 quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
472 ctx->ma_predictor_prev = ma_predictor;
475 tmp = ctx->past_quantizer_outputs[MA_NP];
476 memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
477 MA_NP * sizeof(int16_t*));
478 ctx->past_quantizer_outputs[0] = tmp;
480 ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
482 ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
484 FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
486 for (i = 0; i < 2; i++) {
487 int gain_corr_factor;
489 uint8_t ac_index; ///< adaptive codebook index
490 uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
491 int fc_indexes; ///< fixed-codebook indexes
492 uint8_t gc_1st_index; ///< gain codebook (first stage) index
493 uint8_t gc_2nd_index; ///< gain codebook (second stage) index
495 ac_index = get_bits(&gb, format->ac_index_bits[i]);
496 if (!i && format->parity_bit)
497 bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
498 fc_indexes = get_bits(&gb, format->fc_indexes_bits);
499 pulses_signs = get_bits(&gb, format->fc_signs_bits);
500 gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
501 gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
504 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
507 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
509 pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
512 int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
513 PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
515 if (packet_type == FORMAT_G729D_6K4) {
516 pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
518 pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
522 /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
523 pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
524 if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
525 av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
526 pitch_delay_int[i] = PITCH_DELAY_MAX;
530 ctx->rand_value = g729_prng(ctx->rand_value);
531 fc_indexes = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
533 ctx->rand_value = g729_prng(ctx->rand_value);
534 pulses_signs = ctx->rand_value;
538 memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
539 switch (packet_type) {
541 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
542 ff_fc_4pulses_8bits_track_4,
543 fc_indexes, pulses_signs, 3, 3);
545 case FORMAT_G729D_6K4:
546 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
547 ff_fc_2pulses_9bits_track2_gray,
548 fc_indexes, pulses_signs, 1, 4);
553 This filter enhances harmonic components of the fixed-codebook vector to
554 improve the quality of the reconstructed speech.
556 / fc_v[i], i < pitch_delay
558 \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
560 if (SUBFRAME_SIZE > pitch_delay_int[i])
561 ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
562 fc + pitch_delay_int[i],
564 av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
566 SUBFRAME_SIZE - pitch_delay_int[i]);
568 memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
569 ctx->past_gain_code[1] = ctx->past_gain_code[0];
572 ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
573 ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
575 gain_corr_factor = 0;
577 if (packet_type == FORMAT_G729D_6K4) {
578 ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
579 cb_gain_2nd_6k4[gc_2nd_index][0];
580 gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
581 cb_gain_2nd_6k4[gc_2nd_index][1];
583 /* Without check below overflow can occur in ff_acelp_update_past_gain.
584 It is not issue for G.729, because gain_corr_factor in it's case is always
585 greater than 1024, while in G.729D it can be even zero. */
586 gain_corr_factor = FFMAX(gain_corr_factor, 1024);
587 #ifndef G729_BITEXACT
588 gain_corr_factor >>= 1;
591 ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
592 cb_gain_2nd_8k[gc_2nd_index][0];
593 gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
594 cb_gain_2nd_8k[gc_2nd_index][1];
597 /* Decode the fixed-codebook gain. */
598 ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&s->adsp, gain_corr_factor,
605 This correction required to get bit-exact result with
606 reference code, because gain_corr_factor in G.729D is
607 two times larger than in original G.729.
609 If bit-exact result is not issue then gain_corr_factor
610 can be simpler divided by 2 before call to g729_get_gain_code
611 instead of using correction below.
613 if (packet_type == FORMAT_G729D_6K4) {
614 gain_corr_factor >>= 1;
615 ctx->past_gain_code[0] >>= 1;
619 ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
621 /* Routine requires rounding to lowest. */
622 ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
623 ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
624 ff_acelp_interp_filter, 6,
625 (pitch_delay_3x % 3) << 1,
628 ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
629 ctx->exc + i * SUBFRAME_SIZE, fc,
630 (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
631 ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
632 1 << 13, 14, SUBFRAME_SIZE);
634 memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
636 if (ff_celp_lp_synthesis_filter(
639 ctx->exc + i * SUBFRAME_SIZE,
645 /* Overflow occurred, downscale excitation signal... */
646 for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
647 ctx->exc_base[j] >>= 2;
649 /* ... and make synthesis again. */
650 if (packet_type == FORMAT_G729D_6K4) {
651 int16_t exc_new[SUBFRAME_SIZE];
653 ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
654 ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
656 g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
658 ff_celp_lp_synthesis_filter(
668 ff_celp_lp_synthesis_filter(
671 ctx->exc + i * SUBFRAME_SIZE,
678 /* Save data (without postfilter) for use in next subframe. */
679 memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
681 /* Calculate gain of unfiltered signal for use in AGC. */
683 for (j = 0; j < SUBFRAME_SIZE; j++)
684 gain_before += FFABS(synth[j+10]);
686 /* Call postfilter and also update voicing decision for use in next frame. */
694 ctx->res_filter_data,
695 ctx->pos_filter_data,
699 /* Calculate gain of filtered signal for use in AGC. */
701 for (j = 0; j < SUBFRAME_SIZE; j++)
702 gain_after += FFABS(synth[j+10]);
704 ctx->gain_coeff = ff_g729_adaptive_gain_control(
712 ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
714 ctx->pitch_delay_int_prev = pitch_delay_int[i];
717 memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
718 ff_acelp_high_pass_filter(
719 out_frame + i*SUBFRAME_SIZE,
723 memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
726 ctx->was_periodic = is_periodic;
728 /* Save signal for use in next frame. */
729 memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
731 buf += packet_type == FORMAT_G729_8K ? G729_8K_BLOCK_SIZE : G729D_6K4_BLOCK_SIZE;
736 return packet_type == FORMAT_G729_8K ? (G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels : G729D_6K4_BLOCK_SIZE * avctx->channels;
739 static av_cold int decode_close(AVCodecContext *avctx)
741 G729Context *s = avctx->priv_data;
742 av_freep(&s->channel_context);
747 AVCodec ff_g729_decoder = {
749 .long_name = NULL_IF_CONFIG_SMALL("G.729"),
750 .type = AVMEDIA_TYPE_AUDIO,
751 .id = AV_CODEC_ID_G729,
752 .priv_data_size = sizeof(G729Context),
753 .init = decoder_init,
754 .decode = decode_frame,
755 .close = decode_close,
756 .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
759 AVCodec ff_acelp_kelvin_decoder = {
760 .name = "acelp.kelvin",
761 .long_name = NULL_IF_CONFIG_SMALL("Sipro ACELP.KELVIN"),
762 .type = AVMEDIA_TYPE_AUDIO,
763 .id = AV_CODEC_ID_ACELP_KELVIN,
764 .priv_data_size = sizeof(G729Context),
765 .init = decoder_init,
766 .decode = decode_frame,
767 .close = decode_close,
768 .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,