2 * G.729, G729 Annex D decoders
3 * Copyright (c) 2008 Vladimir Voroshilov
5 * This file is part of FFmpeg.
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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;
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)
155 static const G729FormatDescription format_g729_8k = {
156 .ac_index_bits = {8,5},
158 .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
159 .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
161 .fc_indexes_bits = 13,
164 static const G729FormatDescription format_g729d_6k4 = {
165 .ac_index_bits = {8,4},
167 .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
168 .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
170 .fc_indexes_bits = 9,
174 * @brief pseudo random number generator
176 static inline uint16_t g729_prng(uint16_t value)
178 return 31821 * value + 13849;
182 * Get parity bit of bit 2..7
184 static inline int get_parity(uint8_t value)
186 return (0x6996966996696996ULL >> (value >> 2)) & 1;
190 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
191 * @param lsfq [out] (2.13) quantized LSF coefficients
192 * @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
193 * @param ma_predictor switched MA predictor of LSP quantizer
194 * @param vq_1st first stage vector of quantizer
195 * @param vq_2nd_low second stage lower vector of LSP quantizer
196 * @param vq_2nd_high second stage higher vector of LSP quantizer
198 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
199 int16_t ma_predictor,
200 int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
203 static const uint8_t min_distance[2]={10, 5}; //(2.13)
204 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
206 for (i = 0; i < 5; i++) {
207 quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
208 quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
211 for (j = 0; j < 2; j++) {
212 for (i = 1; i < 10; i++) {
213 int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
215 quantizer_output[i - 1] -= diff;
216 quantizer_output[i ] += diff;
221 for (i = 0; i < 10; i++) {
222 int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
223 for (j = 0; j < MA_NP; j++)
224 sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
229 ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
233 * Restores past LSP quantizer output using LSF from previous frame
234 * @param lsfq [in/out] (2.13) quantized LSF coefficients
235 * @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
236 * @param ma_predictor_prev MA predictor from previous frame
237 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
239 static void lsf_restore_from_previous(int16_t* lsfq,
240 int16_t* past_quantizer_outputs[MA_NP + 1],
241 int ma_predictor_prev)
243 int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
246 for (i = 0; i < 10; i++) {
247 int tmp = lsfq[i] << 15;
249 for (k = 0; k < MA_NP; k++)
250 tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
252 quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
257 * Constructs new excitation signal and applies phase filter to it
258 * @param out[out] constructed speech signal
259 * @param in original excitation signal
260 * @param fc_cur (2.13) original fixed-codebook vector
261 * @param gain_code (14.1) gain code
262 * @param subframe_size length of the subframe
264 static void g729d_get_new_exc(
267 const int16_t* fc_cur,
273 int16_t fc_new[SUBFRAME_SIZE];
275 ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
277 for(i=0; i<subframe_size; i++)
280 out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
281 out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
286 * Makes decision about onset in current subframe
287 * @param past_onset decision result of previous subframe
288 * @param past_gain_code gain code of current and previous subframe
290 * @return onset decision result for current subframe
292 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
294 if((past_gain_code[0] >> 1) > past_gain_code[1])
297 return FFMAX(past_onset-1, 0);
301 * Makes decision about voice presence in current subframe
302 * @param onset onset level
303 * @param prev_voice_decision voice decision result from previous subframe
304 * @param past_gain_pitch pitch gain of current and previous subframes
306 * @return voice decision result for current subframe
308 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
310 int i, low_gain_pitch_cnt, voice_decision;
312 if(past_gain_pitch[0] >= 14745) // 0.9
313 voice_decision = DECISION_VOICE;
314 else if (past_gain_pitch[0] <= 9830) // 0.6
315 voice_decision = DECISION_NOISE;
317 voice_decision = DECISION_INTERMEDIATE;
319 for(i=0, low_gain_pitch_cnt=0; i<6; i++)
320 if(past_gain_pitch[i] < 9830)
321 low_gain_pitch_cnt++;
323 if(low_gain_pitch_cnt > 2 && !onset)
324 voice_decision = DECISION_NOISE;
326 if(!onset && voice_decision > prev_voice_decision + 1)
329 if(onset && voice_decision < DECISION_VOICE)
332 return voice_decision;
335 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order, int shift)
340 res += (*v1++ * *v2++) >> shift;
345 static av_cold int decoder_init(AVCodecContext * avctx)
347 G729Context* ctx = avctx->priv_data;
350 if (avctx->channels != 1) {
351 av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
352 return AVERROR(EINVAL);
354 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
356 /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
357 avctx->frame_size = SUBFRAME_SIZE << 1;
359 ctx->gain_coeff = 16384; // 1.0 in (1.14)
361 for (k = 0; k < MA_NP + 1; k++) {
362 ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
363 for (i = 1; i < 11; i++)
364 ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
367 ctx->lsp[0] = ctx->lsp_buf[0];
368 ctx->lsp[1] = ctx->lsp_buf[1];
369 memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
371 ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
373 /* random seed initialization */
374 ctx->rand_value = 21845;
376 /* quantized prediction error */
378 ctx->quant_energy[i] = -14336; // -14 in (5.10)
380 dsputil_init(&ctx->dsp, avctx);
381 ctx->dsp.scalarproduct_int16 = scalarproduct_int16_c;
386 static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
389 const uint8_t *buf = avpkt->data;
390 int buf_size = avpkt->size;
391 int16_t *out_frame = data;
393 const G729FormatDescription *format;
394 int frame_erasure = 0; ///< frame erasure detected during decoding
395 int bad_pitch = 0; ///< parity check failed
398 G729Formats packet_type;
399 G729Context *ctx = avctx->priv_data;
400 int16_t lp[2][11]; // (3.12)
401 uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
402 uint8_t quantizer_1st; ///< first stage vector of quantizer
403 uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
404 uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
406 int pitch_delay_int[2]; // pitch delay, integer part
407 int pitch_delay_3x; // pitch delay, multiplied by 3
408 int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
409 int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
411 int gain_before, gain_after;
412 int is_periodic = 0; // whether one of the subframes is declared as periodic or not
414 if (*data_size < SUBFRAME_SIZE << 2) {
415 av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer too small\n");
419 if (buf_size == 10) {
420 packet_type = FORMAT_G729_8K;
421 format = &format_g729_8k;
422 //Reset voice decision
424 ctx->voice_decision = DECISION_VOICE;
425 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
426 } else if (buf_size == 8) {
427 packet_type = FORMAT_G729D_6K4;
428 format = &format_g729d_6k4;
429 av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
431 av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
432 return AVERROR_INVALIDDATA;
435 for (i=0; i < buf_size; i++)
436 frame_erasure |= buf[i];
437 frame_erasure = !frame_erasure;
439 init_get_bits(&gb, buf, 8*buf_size);
441 ma_predictor = get_bits(&gb, 1);
442 quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
443 quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
444 quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
447 lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
448 ctx->ma_predictor_prev);
450 lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
452 quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
453 ctx->ma_predictor_prev = ma_predictor;
456 tmp = ctx->past_quantizer_outputs[MA_NP];
457 memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
458 MA_NP * sizeof(int16_t*));
459 ctx->past_quantizer_outputs[0] = tmp;
461 ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
463 ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
465 FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
467 for (i = 0; i < 2; i++) {
468 int gain_corr_factor;
470 uint8_t ac_index; ///< adaptive codebook index
471 uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
472 int fc_indexes; ///< fixed-codebook indexes
473 uint8_t gc_1st_index; ///< gain codebook (first stage) index
474 uint8_t gc_2nd_index; ///< gain codebook (second stage) index
476 ac_index = get_bits(&gb, format->ac_index_bits[i]);
477 if(!i && format->parity_bit)
478 bad_pitch = get_parity(ac_index) == get_bits1(&gb);
479 fc_indexes = get_bits(&gb, format->fc_indexes_bits);
480 pulses_signs = get_bits(&gb, format->fc_signs_bits);
481 gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
482 gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
485 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
488 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
490 pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
492 int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
493 PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
495 if(packet_type == FORMAT_G729D_6K4)
496 pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
498 pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
501 /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
502 pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
505 ctx->rand_value = g729_prng(ctx->rand_value);
506 fc_indexes = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1);
508 ctx->rand_value = g729_prng(ctx->rand_value);
509 pulses_signs = ctx->rand_value;
513 memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
514 switch (packet_type) {
516 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
517 ff_fc_4pulses_8bits_track_4,
518 fc_indexes, pulses_signs, 3, 3);
520 case FORMAT_G729D_6K4:
521 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
522 ff_fc_2pulses_9bits_track2_gray,
523 fc_indexes, pulses_signs, 1, 4);
528 This filter enhances harmonic components of the fixed-codebook vector to
529 improve the quality of the reconstructed speech.
531 / fc_v[i], i < pitch_delay
533 \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
535 ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
536 fc + pitch_delay_int[i],
538 av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
540 SUBFRAME_SIZE - pitch_delay_int[i]);
542 memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
543 ctx->past_gain_code[1] = ctx->past_gain_code[0];
546 ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
547 ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
549 gain_corr_factor = 0;
551 if (packet_type == FORMAT_G729D_6K4) {
552 ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
553 cb_gain_2nd_6k4[gc_2nd_index][0];
554 gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
555 cb_gain_2nd_6k4[gc_2nd_index][1];
557 /* Without check below overflow can occure in ff_acelp_update_past_gain.
558 It is not issue for G.729, because gain_corr_factor in it's case is always
559 greater than 1024, while in G.729D it can be even zero. */
560 gain_corr_factor = FFMAX(gain_corr_factor, 1024);
561 #ifndef G729_BITEXACT
562 gain_corr_factor >>= 1;
565 ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
566 cb_gain_2nd_8k[gc_2nd_index][0];
567 gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
568 cb_gain_2nd_8k[gc_2nd_index][1];
571 /* Decode the fixed-codebook gain. */
572 ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
579 This correction required to get bit-exact result with
580 reference code, because gain_corr_factor in G.729D is
581 two times larger than in original G.729.
583 If bit-exact result is not issue then gain_corr_factor
584 can be simpler devided by 2 before call to g729_get_gain_code
585 instead of using correction below.
587 if (packet_type == FORMAT_G729D_6K4) {
588 gain_corr_factor >>= 1;
589 ctx->past_gain_code[0] >>= 1;
593 ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
595 /* Routine requires rounding to lowest. */
596 ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
597 ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
598 ff_acelp_interp_filter, 6,
599 (pitch_delay_3x % 3) << 1,
602 ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
603 ctx->exc + i * SUBFRAME_SIZE, fc,
604 (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
605 ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
606 1 << 13, 14, SUBFRAME_SIZE);
608 memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
610 if (ff_celp_lp_synthesis_filter(
613 ctx->exc + i * SUBFRAME_SIZE,
619 /* Overflow occured, downscale excitation signal... */
620 for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
621 ctx->exc_base[j] >>= 2;
623 /* ... and make synthesis again. */
624 if (packet_type == FORMAT_G729D_6K4) {
625 int16_t exc_new[SUBFRAME_SIZE];
627 ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
628 ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
630 g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
632 ff_celp_lp_synthesis_filter(
642 ff_celp_lp_synthesis_filter(
645 ctx->exc + i * SUBFRAME_SIZE,
652 /* Save data (without postfilter) for use in next subframe. */
653 memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
655 /* Calculate gain of unfiltered signal for use in AGC. */
657 for (j = 0; j < SUBFRAME_SIZE; j++)
658 gain_before += FFABS(synth[j+10]);
660 /* Call postfilter and also update voicing decision for use in next frame. */
668 ctx->res_filter_data,
669 ctx->pos_filter_data,
673 /* Calculate gain of filtered signal for use in AGC. */
675 for(j=0; j<SUBFRAME_SIZE; j++)
676 gain_after += FFABS(synth[j+10]);
678 ctx->gain_coeff = ff_g729_adaptive_gain_control(
686 ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
688 ctx->pitch_delay_int_prev = pitch_delay_int[i];
690 memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
691 ff_acelp_high_pass_filter(
692 out_frame + i*SUBFRAME_SIZE,
696 memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
699 ctx->was_periodic = is_periodic;
701 /* Save signal for use in next frame. */
702 memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
704 *data_size = SUBFRAME_SIZE << 2;
708 AVCodec ff_g729_decoder =
718 .long_name = NULL_IF_CONFIG_SMALL("G.729"),