2 * G.729, G729 Annex D decoders
3 * Copyright (c) 2008 Vladimir Voroshilov
5 * This file is part of FFmpeg.
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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
30 #include "libavutil/avutil.h"
36 #include "celp_math.h"
37 #include "celp_filters.h"
38 #include "acelp_filters.h"
39 #include "acelp_pitch_delay.h"
40 #include "acelp_vectors.h"
42 #include "g729postfilter.h"
45 * minimum quantized LSF value (3.2.4)
51 * maximum quantized LSF value (3.2.4)
54 #define LSFQ_MAX 25681
57 * minimum LSF distance (3.2.4)
60 #define LSFQ_DIFF_MIN 321
62 /// interpolation filter length
63 #define INTERPOL_LEN 11
66 * minimum gain pitch value (3.8, Equation 47)
69 #define SHARP_MIN 3277
72 * maximum gain pitch value (3.8, Equation 47)
73 * (EE) This does not comply with the specification.
74 * Specification says about 0.8, which should be
75 * 13107 in (1.14), but reference C code uses
76 * 13017 (equals to 0.7945) instead of it.
78 #define SHARP_MAX 13017
81 * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7.13)
83 #define MR_ENERGY 1018156
85 #define DECISION_NOISE 0
86 #define DECISION_INTERMEDIATE 1
87 #define DECISION_VOICE 2
96 uint8_t ac_index_bits[2]; ///< adaptive codebook index for second subframe (size in bits)
97 uint8_t parity_bit; ///< parity bit for pitch delay
98 uint8_t gc_1st_index_bits; ///< gain codebook (first stage) index (size in bits)
99 uint8_t gc_2nd_index_bits; ///< gain codebook (second stage) index (size in bits)
100 uint8_t fc_signs_bits; ///< number of pulses in fixed-codebook vector
101 uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook index entry
102 } G729FormatDescription;
107 /// past excitation signal buffer
108 int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
110 int16_t* exc; ///< start of past excitation data in buffer
111 int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
113 /// (2.13) LSP quantizer outputs
114 int16_t past_quantizer_output_buf[MA_NP + 1][10];
115 int16_t* past_quantizer_outputs[MA_NP + 1];
117 int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
118 int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
119 int16_t *lsp[2]; ///< pointers to lsp_buf
121 int16_t quant_energy[4]; ///< (5.10) past quantized energy
123 /// previous speech data for LP synthesis filter
124 int16_t syn_filter_data[10];
127 /// residual signal buffer (used in long-term postfilter)
128 int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
130 /// previous speech data for residual calculation filter
131 int16_t res_filter_data[SUBFRAME_SIZE+10];
133 /// previous speech data for short-term postfilter
134 int16_t pos_filter_data[SUBFRAME_SIZE+10];
136 /// (1.14) pitch gain of current and five previous subframes
137 int16_t past_gain_pitch[6];
139 /// (14.1) gain code from current and previous subframe
140 int16_t past_gain_code[2];
142 /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
143 int16_t voice_decision;
145 int16_t onset; ///< detected onset level (0-2)
146 int16_t was_periodic; ///< whether previous frame was declared as periodic or not (4.4)
147 int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
148 int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
149 uint16_t rand_value; ///< random number generator value (4.4.4)
150 int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame
152 /// (14.14) high-pass filter data (past input)
155 /// high-pass filter data (past output)
159 static const G729FormatDescription format_g729_8k = {
160 .ac_index_bits = {8,5},
162 .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
163 .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
165 .fc_indexes_bits = 13,
168 static const G729FormatDescription format_g729d_6k4 = {
169 .ac_index_bits = {8,4},
171 .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
172 .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
174 .fc_indexes_bits = 9,
178 * @brief pseudo random number generator
180 static inline uint16_t g729_prng(uint16_t value)
182 return 31821 * value + 13849;
186 * Get parity bit of bit 2..7
188 static inline int get_parity(uint8_t value)
190 return (0x6996966996696996ULL >> (value >> 2)) & 1;
194 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
195 * @param lsfq [out] (2.13) quantized LSF coefficients
196 * @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
197 * @param ma_predictor switched MA predictor of LSP quantizer
198 * @param vq_1st first stage vector of quantizer
199 * @param vq_2nd_low second stage lower vector of LSP quantizer
200 * @param vq_2nd_high second stage higher vector of LSP quantizer
202 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
203 int16_t ma_predictor,
204 int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
207 static const uint8_t min_distance[2]={10, 5}; //(2.13)
208 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
210 for (i = 0; i < 5; i++) {
211 quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
212 quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
215 for (j = 0; j < 2; j++) {
216 for (i = 1; i < 10; i++) {
217 int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
219 quantizer_output[i - 1] -= diff;
220 quantizer_output[i ] += diff;
225 for (i = 0; i < 10; i++) {
226 int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
227 for (j = 0; j < MA_NP; j++)
228 sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
233 ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
237 * Restores past LSP quantizer output using LSF from previous frame
238 * @param lsfq [in/out] (2.13) quantized LSF coefficients
239 * @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
240 * @param ma_predictor_prev MA predictor from previous frame
241 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
243 static void lsf_restore_from_previous(int16_t* lsfq,
244 int16_t* past_quantizer_outputs[MA_NP + 1],
245 int ma_predictor_prev)
247 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
250 for (i = 0; i < 10; i++) {
251 int tmp = lsfq[i] << 15;
253 for (k = 0; k < MA_NP; k++)
254 tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
256 quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
261 * Constructs new excitation signal and applies phase filter to it
262 * @param out[out] constructed speech signal
263 * @param in original excitation signal
264 * @param fc_cur (2.13) original fixed-codebook vector
265 * @param gain_code (14.1) gain code
266 * @param subframe_size length of the subframe
268 static void g729d_get_new_exc(
271 const int16_t* fc_cur,
277 int16_t fc_new[SUBFRAME_SIZE];
279 ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
281 for(i=0; i<subframe_size; i++)
284 out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
285 out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
290 * Makes decision about onset in current subframe
291 * @param past_onset decision result of previous subframe
292 * @param past_gain_code gain code of current and previous subframe
294 * @return onset decision result for current subframe
296 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
298 if((past_gain_code[0] >> 1) > past_gain_code[1])
301 return FFMAX(past_onset-1, 0);
305 * Makes decision about voice presence in current subframe
306 * @param onset onset level
307 * @param prev_voice_decision voice decision result from previous subframe
308 * @param past_gain_pitch pitch gain of current and previous subframes
310 * @return voice decision result for current subframe
312 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
314 int i, low_gain_pitch_cnt, voice_decision;
316 if(past_gain_pitch[0] >= 14745) // 0.9
317 voice_decision = DECISION_VOICE;
318 else if (past_gain_pitch[0] <= 9830) // 0.6
319 voice_decision = DECISION_NOISE;
321 voice_decision = DECISION_INTERMEDIATE;
323 for(i=0, low_gain_pitch_cnt=0; i<6; i++)
324 if(past_gain_pitch[i] < 9830)
325 low_gain_pitch_cnt++;
327 if(low_gain_pitch_cnt > 2 && !onset)
328 voice_decision = DECISION_NOISE;
330 if(!onset && voice_decision > prev_voice_decision + 1)
333 if(onset && voice_decision < DECISION_VOICE)
336 return voice_decision;
339 static av_cold int decoder_init(AVCodecContext * avctx)
341 G729Context* ctx = avctx->priv_data;
344 if (avctx->channels != 1) {
345 av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
346 return AVERROR(EINVAL);
348 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
350 /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
351 avctx->frame_size = SUBFRAME_SIZE << 1;
353 ctx->gain_coeff = 16384; // 1.0 in (1.14)
355 for (k = 0; k < MA_NP + 1; k++) {
356 ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
357 for (i = 1; i < 11; i++)
358 ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
361 ctx->lsp[0] = ctx->lsp_buf[0];
362 ctx->lsp[1] = ctx->lsp_buf[1];
363 memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
365 ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
367 /* random seed initialization */
368 ctx->rand_value = 21845;
370 /* quantized prediction error */
372 ctx->quant_energy[i] = -14336; // -14 in (5.10)
374 avctx->dsp_mask= ~AV_CPU_FLAG_FORCE;
375 dsputil_init(&ctx->dsp, avctx);
380 static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
383 const uint8_t *buf = avpkt->data;
384 int buf_size = avpkt->size;
385 int16_t *out_frame = data;
387 G729FormatDescription format;
388 int frame_erasure = 0; ///< frame erasure detected during decoding
389 int bad_pitch = 0; ///< parity check failed
392 G729Formats packet_type;
393 G729Context *ctx = avctx->priv_data;
394 int16_t lp[2][11]; // (3.12)
395 uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
396 uint8_t quantizer_1st; ///< first stage vector of quantizer
397 uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
398 uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
400 int pitch_delay_int[2]; // pitch delay, integer part
401 int pitch_delay_3x; // pitch delay, multiplied by 3
402 int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
403 int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
405 int gain_before, gain_after;
406 int is_periodic = 0; // whether one of the subframes is declared as periodic or not
408 if (*data_size < SUBFRAME_SIZE << 2) {
409 av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer too small\n");
413 if (buf_size == 10) {
414 packet_type = FORMAT_G729_8K;
415 format = format_g729_8k;
416 //Reset voice decision
418 ctx->voice_decision = DECISION_VOICE;
419 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
420 } else if (buf_size == 8) {
421 packet_type = FORMAT_G729D_6K4;
422 format = format_g729d_6k4;
423 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
425 av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
426 return AVERROR_INVALIDDATA;
429 for (i=0; i < buf_size; i++)
430 frame_erasure |= buf[i];
431 frame_erasure = !frame_erasure;
433 init_get_bits(&gb, buf, buf_size);
435 ma_predictor = get_bits(&gb, 1);
436 quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
437 quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
438 quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
441 lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
442 ctx->ma_predictor_prev);
444 lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
446 quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
447 ctx->ma_predictor_prev = ma_predictor;
450 tmp = ctx->past_quantizer_outputs[MA_NP];
451 memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
452 MA_NP * sizeof(int16_t*));
453 ctx->past_quantizer_outputs[0] = tmp;
455 ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
457 ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
459 FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
461 for (i = 0; i < 2; i++) {
462 int gain_corr_factor;
464 uint8_t ac_index; ///< adaptive codebook index
465 uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
466 int fc_indexes; ///< fixed-codebook indexes
467 uint8_t gc_1st_index; ///< gain codebook (first stage) index
468 uint8_t gc_2nd_index; ///< gain codebook (second stage) index
470 ac_index = get_bits(&gb, format.ac_index_bits[i]);
471 if(!i && format.parity_bit)
472 bad_pitch = get_parity(ac_index) == get_bits1(&gb);
473 fc_indexes = get_bits(&gb, format.fc_indexes_bits);
474 pulses_signs = get_bits(&gb, format.fc_signs_bits);
475 gc_1st_index = get_bits(&gb, format.gc_1st_index_bits);
476 gc_2nd_index = get_bits(&gb, format.gc_2nd_index_bits);
479 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
482 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
484 pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
486 int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
487 PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
489 if(packet_type == FORMAT_G729D_6K4)
490 pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
492 pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
495 /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
496 pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
499 ctx->rand_value = g729_prng(ctx->rand_value);
500 fc_indexes = ctx->rand_value & ((1 << format.fc_indexes_bits) - 1);
502 ctx->rand_value = g729_prng(ctx->rand_value);
503 pulses_signs = ctx->rand_value;
507 memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
508 switch (packet_type) {
510 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
511 ff_fc_4pulses_8bits_track_4,
512 fc_indexes, pulses_signs, 3, 3);
514 case FORMAT_G729D_6K4:
515 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
516 ff_fc_2pulses_9bits_track2_gray,
517 fc_indexes, pulses_signs, 1, 4);
522 This filter enhances harmonic components of the fixed-codebook vector to
523 improve the quality of the reconstructed speech.
525 / fc_v[i], i < pitch_delay
527 \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
529 ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
530 fc + pitch_delay_int[i],
532 av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
534 SUBFRAME_SIZE - pitch_delay_int[i]);
536 memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
537 ctx->past_gain_code[1] = ctx->past_gain_code[0];
540 ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
541 ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
543 gain_corr_factor = 0;
545 if (packet_type == FORMAT_G729D_6K4) {
546 ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
547 cb_gain_2nd_6k4[gc_2nd_index][0];
548 gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
549 cb_gain_2nd_6k4[gc_2nd_index][1];
551 /* Without check below overflow can occure in ff_acelp_update_past_gain.
552 It is not issue for G.729, because gain_corr_factor in it's case is always
553 greater than 1024, while in G.729D it can be even zero. */
554 gain_corr_factor = FFMAX(gain_corr_factor, 1024);
555 #ifndef G729_BITEXACT
556 gain_corr_factor >>= 1;
559 ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
560 cb_gain_2nd_8k[gc_2nd_index][0];
561 gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
562 cb_gain_2nd_8k[gc_2nd_index][1];
565 /* Decode the fixed-codebook gain. */
566 ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
573 This correction required to get bit-exact result with
574 reference code, because gain_corr_factor in G.729D is
575 two times larger than in original G.729.
577 If bit-exact result is not issue then gain_corr_factor
578 can be simpler devided by 2 before call to g729_get_gain_code
579 instead of using correction below.
581 if (packet_type == FORMAT_G729D_6K4) {
582 gain_corr_factor >>= 1;
583 ctx->past_gain_code[0] >>= 1;
587 ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
589 /* Routine requires rounding to lowest. */
590 ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
591 ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
592 ff_acelp_interp_filter, 6,
593 (pitch_delay_3x % 3) << 1,
596 ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
597 ctx->exc + i * SUBFRAME_SIZE, fc,
598 (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
599 ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
600 1 << 13, 14, SUBFRAME_SIZE);
602 memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
604 if (ff_celp_lp_synthesis_filter(
607 ctx->exc + i * SUBFRAME_SIZE,
612 /* Overflow occured, downscale excitation signal... */
613 for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
614 ctx->exc_base[j] >>= 2;
616 /* ... and make synthesis again. */
617 if (packet_type == FORMAT_G729D_6K4) {
618 int16_t exc_new[SUBFRAME_SIZE];
620 ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
621 ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
623 g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
625 ff_celp_lp_synthesis_filter(
634 ff_celp_lp_synthesis_filter(
637 ctx->exc + i * SUBFRAME_SIZE,
643 /* Save data (without postfilter) for use in next subframe. */
644 memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
646 /* Calculate gain of unfiltered signal for use in AGC. */
648 for (j = 0; j < SUBFRAME_SIZE; j++)
649 gain_before += FFABS(synth[j+10]);
651 /* Call postfilter and also update voicing decision for use in next frame. */
659 ctx->res_filter_data,
660 ctx->pos_filter_data,
664 /* Calculate gain of filtered signal for use in AGC. */
666 for(j=0; j<SUBFRAME_SIZE; j++)
667 gain_after += FFABS(synth[j+10]);
669 ctx->gain_coeff = g729_adaptive_gain_control(
677 ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
679 ctx->pitch_delay_int_prev = pitch_delay_int[i];
681 memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
682 ff_acelp_high_pass_filter(
683 out_frame + i*SUBFRAME_SIZE,
687 memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
690 ctx->was_periodic = is_periodic;
692 /* Save signal for use in next frame. */
693 memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
695 *data_size = SUBFRAME_SIZE << 2;
699 AVCodec ff_g729_decoder =
709 .long_name = NULL_IF_CONFIG_SMALL("G.729"),