3 * Copyright (c) 2010 Marcelo Galvao Povoa
5 * This file is part of Libav.
7 * Libav 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
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17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 * AMR wideband decoder
27 #include "libavutil/common.h"
28 #include "libavutil/lfg.h"
33 #include "celp_filters.h"
34 #include "acelp_filters.h"
35 #include "acelp_vectors.h"
36 #include "acelp_pitch_delay.h"
38 #define AMR_USE_16BIT_TABLES
41 #include "amrwbdata.h"
44 AVFrame avframe; ///< AVFrame for decoded samples
45 AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream
46 enum Mode fr_cur_mode; ///< mode index of current frame
47 uint8_t fr_quality; ///< frame quality index (FQI)
48 float isf_cur[LP_ORDER]; ///< working ISF vector from current frame
49 float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
50 float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
51 double isp[4][LP_ORDER]; ///< ISP vectors from current frame
52 double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
54 float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
56 uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
57 uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe
59 float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
60 float *excitation; ///< points to current excitation in excitation_buf[]
62 float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
63 float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
65 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
66 float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
67 float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
69 float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
71 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
72 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
73 float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
75 float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz
76 float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling
77 float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
79 float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
80 float demph_mem[1]; ///< previous value in the de-emphasis filter
81 float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
82 float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
84 AVLFG prng; ///< random number generator for white noise excitation
85 uint8_t first_frame; ///< flag active during decoding of the first frame
88 static av_cold int amrwb_decode_init(AVCodecContext *avctx)
90 AMRWBContext *ctx = avctx->priv_data;
93 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
95 av_lfg_init(&ctx->prng, 1);
97 ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
100 for (i = 0; i < LP_ORDER; i++)
101 ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
103 for (i = 0; i < 4; i++)
104 ctx->prediction_error[i] = MIN_ENERGY;
106 avcodec_get_frame_defaults(&ctx->avframe);
107 avctx->coded_frame = &ctx->avframe;
113 * Decode the frame header in the "MIME/storage" format. This format
114 * is simpler and does not carry the auxiliary frame information.
116 * @param[in] ctx The Context
117 * @param[in] buf Pointer to the input buffer
119 * @return The decoded header length in bytes
121 static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
123 /* Decode frame header (1st octet) */
124 ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
125 ctx->fr_quality = (buf[0] & 0x4) == 0x4;
131 * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
133 * @param[in] ind Array of 5 indexes
134 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
137 static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
141 for (i = 0; i < 9; i++)
142 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
144 for (i = 0; i < 7; i++)
145 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
147 for (i = 0; i < 5; i++)
148 isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
150 for (i = 0; i < 4; i++)
151 isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
153 for (i = 0; i < 7; i++)
154 isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
158 * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
160 * @param[in] ind Array of 7 indexes
161 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
164 static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
168 for (i = 0; i < 9; i++)
169 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
171 for (i = 0; i < 7; i++)
172 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
174 for (i = 0; i < 3; i++)
175 isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
177 for (i = 0; i < 3; i++)
178 isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
180 for (i = 0; i < 3; i++)
181 isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
183 for (i = 0; i < 3; i++)
184 isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
186 for (i = 0; i < 4; i++)
187 isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
191 * Apply mean and past ISF values using the prediction factor.
192 * Updates past ISF vector.
194 * @param[in,out] isf_q Current quantized ISF
195 * @param[in,out] isf_past Past quantized ISF
198 static void isf_add_mean_and_past(float *isf_q, float *isf_past)
203 for (i = 0; i < LP_ORDER; i++) {
205 isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
206 isf_q[i] += PRED_FACTOR * isf_past[i];
212 * Interpolate the fourth ISP vector from current and past frames
213 * to obtain an ISP vector for each subframe.
215 * @param[in,out] isp_q ISPs for each subframe
216 * @param[in] isp4_past Past ISP for subframe 4
218 static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
222 for (k = 0; k < 3; k++) {
223 float c = isfp_inter[k];
224 for (i = 0; i < LP_ORDER; i++)
225 isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
230 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
231 * Calculate integer lag and fractional lag always using 1/4 resolution.
232 * In 1st and 3rd subframes the index is relative to last subframe integer lag.
234 * @param[out] lag_int Decoded integer pitch lag
235 * @param[out] lag_frac Decoded fractional pitch lag
236 * @param[in] pitch_index Adaptive codebook pitch index
237 * @param[in,out] base_lag_int Base integer lag used in relative subframes
238 * @param[in] subframe Current subframe index (0 to 3)
240 static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
241 uint8_t *base_lag_int, int subframe)
243 if (subframe == 0 || subframe == 2) {
244 if (pitch_index < 376) {
245 *lag_int = (pitch_index + 137) >> 2;
246 *lag_frac = pitch_index - (*lag_int << 2) + 136;
247 } else if (pitch_index < 440) {
248 *lag_int = (pitch_index + 257 - 376) >> 1;
249 *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
250 /* the actual resolution is 1/2 but expressed as 1/4 */
252 *lag_int = pitch_index - 280;
255 /* minimum lag for next subframe */
256 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
257 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
258 // XXX: the spec states clearly that *base_lag_int should be
259 // the nearest integer to *lag_int (minus 8), but the ref code
260 // actually always uses its floor, I'm following the latter
262 *lag_int = (pitch_index + 1) >> 2;
263 *lag_frac = pitch_index - (*lag_int << 2);
264 *lag_int += *base_lag_int;
269 * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
270 * The description is analogous to decode_pitch_lag_high, but in 6k60 the
271 * relative index is used for all subframes except the first.
273 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
274 uint8_t *base_lag_int, int subframe, enum Mode mode)
276 if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
277 if (pitch_index < 116) {
278 *lag_int = (pitch_index + 69) >> 1;
279 *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;
281 *lag_int = pitch_index - 24;
284 // XXX: same problem as before
285 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
286 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
288 *lag_int = (pitch_index + 1) >> 1;
289 *lag_frac = (pitch_index - (*lag_int << 1)) << 1;
290 *lag_int += *base_lag_int;
295 * Find the pitch vector by interpolating the past excitation at the
296 * pitch delay, which is obtained in this function.
298 * @param[in,out] ctx The context
299 * @param[in] amr_subframe Current subframe data
300 * @param[in] subframe Current subframe index (0 to 3)
302 static void decode_pitch_vector(AMRWBContext *ctx,
303 const AMRWBSubFrame *amr_subframe,
306 int pitch_lag_int, pitch_lag_frac;
308 float *exc = ctx->excitation;
309 enum Mode mode = ctx->fr_cur_mode;
311 if (mode <= MODE_8k85) {
312 decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
313 &ctx->base_pitch_lag, subframe, mode);
315 decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
316 &ctx->base_pitch_lag, subframe);
318 ctx->pitch_lag_int = pitch_lag_int;
319 pitch_lag_int += pitch_lag_frac > 0;
321 /* Calculate the pitch vector by interpolating the past excitation at the
322 pitch lag using a hamming windowed sinc function */
323 ff_acelp_interpolatef(exc, exc + 1 - pitch_lag_int,
325 pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
326 LP_ORDER, AMRWB_SFR_SIZE + 1);
328 /* Check which pitch signal path should be used
329 * 6k60 and 8k85 modes have the ltp flag set to 0 */
330 if (amr_subframe->ltp) {
331 memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
333 for (i = 0; i < AMRWB_SFR_SIZE; i++)
334 ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
336 memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
340 /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
341 #define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))
343 /** Get the bit at specified position */
344 #define BIT_POS(x, p) (((x) >> (p)) & 1)
347 * The next six functions decode_[i]p_track decode exactly i pulses
348 * positions and amplitudes (-1 or 1) in a subframe track using
349 * an encoded pulse indexing (TS 26.190 section 5.8.2).
351 * The results are given in out[], in which a negative number means
352 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
354 * @param[out] out Output buffer (writes i elements)
355 * @param[in] code Pulse index (no. of bits varies, see below)
356 * @param[in] m (log2) Number of potential positions
357 * @param[in] off Offset for decoded positions
359 static inline void decode_1p_track(int *out, int code, int m, int off)
361 int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
363 out[0] = BIT_POS(code, m) ? -pos : pos;
366 static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
368 int pos0 = BIT_STR(code, m, m) + off;
369 int pos1 = BIT_STR(code, 0, m) + off;
371 out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
372 out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
373 out[1] = pos0 > pos1 ? -out[1] : out[1];
376 static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
378 int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
380 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
381 m - 1, off + half_2p);
382 decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
385 static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
387 int half_4p, subhalf_2p;
388 int b_offset = 1 << (m - 1);
390 switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
391 case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
392 half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
393 subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
395 decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
396 m - 2, off + half_4p + subhalf_2p);
397 decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
398 m - 1, off + half_4p);
400 case 1: /* 1 pulse in A, 3 pulses in B */
401 decode_1p_track(out, BIT_STR(code, 3*m - 2, m),
403 decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
404 m - 1, off + b_offset);
406 case 2: /* 2 pulses in each half */
407 decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
409 decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
410 m - 1, off + b_offset);
412 case 3: /* 3 pulses in A, 1 pulse in B */
413 decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
415 decode_1p_track(out + 3, BIT_STR(code, 0, m),
416 m - 1, off + b_offset);
421 static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
423 int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
425 decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
426 m - 1, off + half_3p);
428 decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
431 static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
433 int b_offset = 1 << (m - 1);
434 /* which half has more pulses in cases 0 to 2 */
435 int half_more = BIT_POS(code, 6*m - 5) << (m - 1);
436 int half_other = b_offset - half_more;
438 switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
439 case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
440 decode_1p_track(out, BIT_STR(code, 0, m),
441 m - 1, off + half_more);
442 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
443 m - 1, off + half_more);
445 case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
446 decode_1p_track(out, BIT_STR(code, 0, m),
447 m - 1, off + half_other);
448 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
449 m - 1, off + half_more);
451 case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
452 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
453 m - 1, off + half_other);
454 decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
455 m - 1, off + half_more);
457 case 3: /* 3 pulses in A, 3 pulses in B */
458 decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
460 decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
461 m - 1, off + b_offset);
467 * Decode the algebraic codebook index to pulse positions and signs,
468 * then construct the algebraic codebook vector.
470 * @param[out] fixed_vector Buffer for the fixed codebook excitation
471 * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)
472 * @param[in] pulse_lo LSBs part of the pulse index array
473 * @param[in] mode Mode of the current frame
475 static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
476 const uint16_t *pulse_lo, const enum Mode mode)
478 /* sig_pos stores for each track the decoded pulse position indexes
479 * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
481 int spacing = (mode == MODE_6k60) ? 2 : 4;
486 for (i = 0; i < 2; i++)
487 decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
490 for (i = 0; i < 4; i++)
491 decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
494 for (i = 0; i < 4; i++)
495 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
498 for (i = 0; i < 2; i++)
499 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
500 for (i = 2; i < 4; i++)
501 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
504 for (i = 0; i < 4; i++)
505 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
508 for (i = 0; i < 4; i++)
509 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
510 ((int) pulse_hi[i] << 14), 4, 1);
513 for (i = 0; i < 2; i++)
514 decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
515 ((int) pulse_hi[i] << 10), 4, 1);
516 for (i = 2; i < 4; i++)
517 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
518 ((int) pulse_hi[i] << 14), 4, 1);
522 for (i = 0; i < 4; i++)
523 decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
524 ((int) pulse_hi[i] << 11), 4, 1);
528 memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
530 for (i = 0; i < 4; i++)
531 for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
532 int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
534 fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
539 * Decode pitch gain and fixed gain correction factor.
541 * @param[in] vq_gain Vector-quantized index for gains
542 * @param[in] mode Mode of the current frame
543 * @param[out] fixed_gain_factor Decoded fixed gain correction factor
544 * @param[out] pitch_gain Decoded pitch gain
546 static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
547 float *fixed_gain_factor, float *pitch_gain)
549 const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
550 qua_gain_7b[vq_gain]);
552 *pitch_gain = gains[0] * (1.0f / (1 << 14));
553 *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
557 * Apply pitch sharpening filters to the fixed codebook vector.
559 * @param[in] ctx The context
560 * @param[in,out] fixed_vector Fixed codebook excitation
562 // XXX: Spec states this procedure should be applied when the pitch
563 // lag is less than 64, but this checking seems absent in reference and AMR-NB
564 static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
569 for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
570 fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
572 /* Periodicity enhancement part */
573 for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
574 fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
578 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
580 * @param[in] p_vector, f_vector Pitch and fixed excitation vectors
581 * @param[in] p_gain, f_gain Pitch and fixed gains
583 // XXX: There is something wrong with the precision here! The magnitudes
584 // of the energies are not correct. Please check the reference code carefully
585 static float voice_factor(float *p_vector, float p_gain,
586 float *f_vector, float f_gain)
588 double p_ener = (double) ff_scalarproduct_float_c(p_vector, p_vector,
591 double f_ener = (double) ff_scalarproduct_float_c(f_vector, f_vector,
595 return (p_ener - f_ener) / (p_ener + f_ener);
599 * Reduce fixed vector sparseness by smoothing with one of three IR filters,
600 * also known as "adaptive phase dispersion".
602 * @param[in] ctx The context
603 * @param[in,out] fixed_vector Unfiltered fixed vector
604 * @param[out] buf Space for modified vector if necessary
606 * @return The potentially overwritten filtered fixed vector address
608 static float *anti_sparseness(AMRWBContext *ctx,
609 float *fixed_vector, float *buf)
613 if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
616 if (ctx->pitch_gain[0] < 0.6) {
617 ir_filter_nr = 0; // strong filtering
618 } else if (ctx->pitch_gain[0] < 0.9) {
619 ir_filter_nr = 1; // medium filtering
621 ir_filter_nr = 2; // no filtering
624 if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
625 if (ir_filter_nr < 2)
630 for (i = 0; i < 6; i++)
631 if (ctx->pitch_gain[i] < 0.6)
637 if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
641 /* update ir filter strength history */
642 ctx->prev_ir_filter_nr = ir_filter_nr;
644 ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
646 if (ir_filter_nr < 2) {
648 const float *coef = ir_filters_lookup[ir_filter_nr];
650 /* Circular convolution code in the reference
651 * decoder was modified to avoid using one
652 * extra array. The filtered vector is given by:
654 * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
657 memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
658 for (i = 0; i < AMRWB_SFR_SIZE; i++)
660 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
669 * Calculate a stability factor {teta} based on distance between
670 * current and past isf. A value of 1 shows maximum signal stability.
672 static float stability_factor(const float *isf, const float *isf_past)
677 for (i = 0; i < LP_ORDER - 1; i++)
678 acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
680 // XXX: This part is not so clear from the reference code
681 // the result is more accurate changing the "/ 256" to "* 512"
682 return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
686 * Apply a non-linear fixed gain smoothing in order to reduce
687 * fluctuation in the energy of excitation.
689 * @param[in] fixed_gain Unsmoothed fixed gain
690 * @param[in,out] prev_tr_gain Previous threshold gain (updated)
691 * @param[in] voice_fac Frame voicing factor
692 * @param[in] stab_fac Frame stability factor
694 * @return The smoothed gain
696 static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
697 float voice_fac, float stab_fac)
699 float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
702 // XXX: the following fixed-point constants used to in(de)crement
703 // gain by 1.5dB were taken from the reference code, maybe it could
705 if (fixed_gain < *prev_tr_gain) {
706 g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
707 (6226 * (1.0f / (1 << 15)))); // +1.5 dB
709 g0 = FFMAX(*prev_tr_gain, fixed_gain *
710 (27536 * (1.0f / (1 << 15)))); // -1.5 dB
712 *prev_tr_gain = g0; // update next frame threshold
714 return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
718 * Filter the fixed_vector to emphasize the higher frequencies.
720 * @param[in,out] fixed_vector Fixed codebook vector
721 * @param[in] voice_fac Frame voicing factor
723 static void pitch_enhancer(float *fixed_vector, float voice_fac)
726 float cpe = 0.125 * (1 + voice_fac);
727 float last = fixed_vector[0]; // holds c(i - 1)
729 fixed_vector[0] -= cpe * fixed_vector[1];
731 for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
732 float cur = fixed_vector[i];
734 fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
738 fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
742 * Conduct 16th order linear predictive coding synthesis from excitation.
744 * @param[in] ctx Pointer to the AMRWBContext
745 * @param[in] lpc Pointer to the LPC coefficients
746 * @param[out] excitation Buffer for synthesis final excitation
747 * @param[in] fixed_gain Fixed codebook gain for synthesis
748 * @param[in] fixed_vector Algebraic codebook vector
749 * @param[in,out] samples Pointer to the output samples and memory
751 static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
752 float fixed_gain, const float *fixed_vector,
755 ff_weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
756 ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
758 /* emphasize pitch vector contribution in low bitrate modes */
759 if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
761 float energy = ff_scalarproduct_float_c(excitation, excitation,
764 // XXX: Weird part in both ref code and spec. A unknown parameter
765 // {beta} seems to be identical to the current pitch gain
766 float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
768 for (i = 0; i < AMRWB_SFR_SIZE; i++)
769 excitation[i] += pitch_factor * ctx->pitch_vector[i];
771 ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
772 energy, AMRWB_SFR_SIZE);
775 ff_celp_lp_synthesis_filterf(samples, lpc, excitation,
776 AMRWB_SFR_SIZE, LP_ORDER);
780 * Apply to synthesis a de-emphasis filter of the form:
781 * H(z) = 1 / (1 - m * z^-1)
783 * @param[out] out Output buffer
784 * @param[in] in Input samples array with in[-1]
785 * @param[in] m Filter coefficient
786 * @param[in,out] mem State from last filtering
788 static void de_emphasis(float *out, float *in, float m, float mem[1])
792 out[0] = in[0] + m * mem[0];
794 for (i = 1; i < AMRWB_SFR_SIZE; i++)
795 out[i] = in[i] + out[i - 1] * m;
797 mem[0] = out[AMRWB_SFR_SIZE - 1];
801 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
802 * a FIR interpolation filter. Uses past data from before *in address.
804 * @param[out] out Buffer for interpolated signal
805 * @param[in] in Current signal data (length 0.8*o_size)
806 * @param[in] o_size Output signal length
808 static void upsample_5_4(float *out, const float *in, int o_size)
810 const float *in0 = in - UPS_FIR_SIZE + 1;
812 int int_part = 0, frac_part;
815 for (j = 0; j < o_size / 5; j++) {
816 out[i] = in[int_part];
820 for (k = 1; k < 5; k++) {
821 out[i] = ff_scalarproduct_float_c(in0 + int_part,
822 upsample_fir[4 - frac_part],
832 * Calculate the high-band gain based on encoded index (23k85 mode) or
833 * on the low-band speech signal and the Voice Activity Detection flag.
835 * @param[in] ctx The context
836 * @param[in] synth LB speech synthesis at 12.8k
837 * @param[in] hb_idx Gain index for mode 23k85 only
838 * @param[in] vad VAD flag for the frame
840 static float find_hb_gain(AMRWBContext *ctx, const float *synth,
841 uint16_t hb_idx, uint8_t vad)
846 if (ctx->fr_cur_mode == MODE_23k85)
847 return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
849 tilt = ff_scalarproduct_float_c(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
850 ff_scalarproduct_float_c(synth, synth, AMRWB_SFR_SIZE);
852 /* return gain bounded by [0.1, 1.0] */
853 return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
857 * Generate the high-band excitation with the same energy from the lower
858 * one and scaled by the given gain.
860 * @param[in] ctx The context
861 * @param[out] hb_exc Buffer for the excitation
862 * @param[in] synth_exc Low-band excitation used for synthesis
863 * @param[in] hb_gain Wanted excitation gain
865 static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
866 const float *synth_exc, float hb_gain)
869 float energy = ff_scalarproduct_float_c(synth_exc, synth_exc, AMRWB_SFR_SIZE);
871 /* Generate a white-noise excitation */
872 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
873 hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
875 ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
876 energy * hb_gain * hb_gain,
881 * Calculate the auto-correlation for the ISF difference vector.
883 static float auto_correlation(float *diff_isf, float mean, int lag)
888 for (i = 7; i < LP_ORDER - 2; i++) {
889 float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
896 * Extrapolate a ISF vector to the 16kHz range (20th order LP)
897 * used at mode 6k60 LP filter for the high frequency band.
899 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
902 static void extrapolate_isf(float isf[LP_ORDER_16k])
904 float diff_isf[LP_ORDER - 2], diff_mean;
905 float *diff_hi = diff_isf - LP_ORDER + 1; // diff array for extrapolated indexes
910 isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
912 /* Calculate the difference vector */
913 for (i = 0; i < LP_ORDER - 2; i++)
914 diff_isf[i] = isf[i + 1] - isf[i];
917 for (i = 2; i < LP_ORDER - 2; i++)
918 diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
920 /* Find which is the maximum autocorrelation */
922 for (i = 0; i < 3; i++) {
923 corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
925 if (corr_lag[i] > corr_lag[i_max_corr])
930 for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
931 isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
932 - isf[i - 2 - i_max_corr];
934 /* Calculate an estimate for ISF(18) and scale ISF based on the error */
935 est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
936 scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
937 (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
939 for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
940 diff_hi[i] = scale * (isf[i] - isf[i - 1]);
942 /* Stability insurance */
943 for (i = LP_ORDER; i < LP_ORDER_16k - 1; i++)
944 if (diff_hi[i] + diff_hi[i - 1] < 5.0) {
945 if (diff_hi[i] > diff_hi[i - 1]) {
946 diff_hi[i - 1] = 5.0 - diff_hi[i];
948 diff_hi[i] = 5.0 - diff_hi[i - 1];
951 for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
952 isf[i] = isf[i - 1] + diff_hi[i] * (1.0f / (1 << 15));
954 /* Scale the ISF vector for 16000 Hz */
955 for (i = 0; i < LP_ORDER_16k - 1; i++)
960 * Spectral expand the LP coefficients using the equation:
961 * y[i] = x[i] * (gamma ** i)
963 * @param[out] out Output buffer (may use input array)
964 * @param[in] lpc LP coefficients array
965 * @param[in] gamma Weighting factor
966 * @param[in] size LP array size
968 static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
973 for (i = 0; i < size; i++) {
974 out[i] = lpc[i] * fac;
980 * Conduct 20th order linear predictive coding synthesis for the high
981 * frequency band excitation at 16kHz.
983 * @param[in] ctx The context
984 * @param[in] subframe Current subframe index (0 to 3)
985 * @param[in,out] samples Pointer to the output speech samples
986 * @param[in] exc Generated white-noise scaled excitation
987 * @param[in] isf Current frame isf vector
988 * @param[in] isf_past Past frame final isf vector
990 static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
991 const float *exc, const float *isf, const float *isf_past)
993 float hb_lpc[LP_ORDER_16k];
994 enum Mode mode = ctx->fr_cur_mode;
996 if (mode == MODE_6k60) {
997 float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
998 double e_isp[LP_ORDER_16k];
1000 ff_weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1001 1.0 - isfp_inter[subframe], LP_ORDER);
1003 extrapolate_isf(e_isf);
1005 e_isf[LP_ORDER_16k - 1] *= 2.0;
1006 ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
1007 ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
1009 lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
1011 lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
1014 ff_celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
1015 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
1019 * Apply a 15th order filter to high-band samples.
1020 * The filter characteristic depends on the given coefficients.
1022 * @param[out] out Buffer for filtered output
1023 * @param[in] fir_coef Filter coefficients
1024 * @param[in,out] mem State from last filtering (updated)
1025 * @param[in] in Input speech data (high-band)
1027 * @remark It is safe to pass the same array in in and out parameters
1029 static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
1030 float mem[HB_FIR_SIZE], const float *in)
1033 float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
1035 memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
1036 memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
1038 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
1040 for (j = 0; j <= HB_FIR_SIZE; j++)
1041 out[i] += data[i + j] * fir_coef[j];
1044 memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
1048 * Update context state before the next subframe.
1050 static void update_sub_state(AMRWBContext *ctx)
1052 memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
1053 (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
1055 memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
1056 memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
1058 memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
1059 LP_ORDER * sizeof(float));
1060 memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
1061 UPS_MEM_SIZE * sizeof(float));
1062 memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
1063 LP_ORDER_16k * sizeof(float));
1066 static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
1067 int *got_frame_ptr, AVPacket *avpkt)
1069 AMRWBContext *ctx = avctx->priv_data;
1070 AMRWBFrame *cf = &ctx->frame;
1071 const uint8_t *buf = avpkt->data;
1072 int buf_size = avpkt->size;
1073 int expected_fr_size, header_size;
1075 float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
1076 float fixed_gain_factor; // fixed gain correction factor (gamma)
1077 float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
1078 float synth_fixed_gain; // the fixed gain that synthesis should use
1079 float voice_fac, stab_fac; // parameters used for gain smoothing
1080 float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis
1081 float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
1082 float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
1086 /* get output buffer */
1087 ctx->avframe.nb_samples = 4 * AMRWB_SFR_SIZE_16k;
1088 if ((ret = avctx->get_buffer(avctx, &ctx->avframe)) < 0) {
1089 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1092 buf_out = (float *)ctx->avframe.data[0];
1094 header_size = decode_mime_header(ctx, buf);
1095 if (ctx->fr_cur_mode > MODE_SID) {
1096 av_log(avctx, AV_LOG_ERROR,
1097 "Invalid mode %d\n", ctx->fr_cur_mode);
1098 return AVERROR_INVALIDDATA;
1100 expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
1102 if (buf_size < expected_fr_size) {
1103 av_log(avctx, AV_LOG_ERROR,
1104 "Frame too small (%d bytes). Truncated file?\n", buf_size);
1106 return AVERROR_INVALIDDATA;
1109 if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
1110 av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
1112 if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
1113 av_log_missing_feature(avctx, "SID mode", 1);
1117 ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
1118 buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
1120 /* Decode the quantized ISF vector */
1121 if (ctx->fr_cur_mode == MODE_6k60) {
1122 decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
1124 decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
1127 isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
1128 ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
1130 stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
1132 ctx->isf_cur[LP_ORDER - 1] *= 2.0;
1133 ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
1135 /* Generate a ISP vector for each subframe */
1136 if (ctx->first_frame) {
1137 ctx->first_frame = 0;
1138 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
1140 interpolate_isp(ctx->isp, ctx->isp_sub4_past);
1142 for (sub = 0; sub < 4; sub++)
1143 ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
1145 for (sub = 0; sub < 4; sub++) {
1146 const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
1147 float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
1149 /* Decode adaptive codebook (pitch vector) */
1150 decode_pitch_vector(ctx, cur_subframe, sub);
1151 /* Decode innovative codebook (fixed vector) */
1152 decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
1153 cur_subframe->pul_il, ctx->fr_cur_mode);
1155 pitch_sharpening(ctx, ctx->fixed_vector);
1157 decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
1158 &fixed_gain_factor, &ctx->pitch_gain[0]);
1160 ctx->fixed_gain[0] =
1161 ff_amr_set_fixed_gain(fixed_gain_factor,
1162 ff_scalarproduct_float_c(ctx->fixed_vector,
1166 ctx->prediction_error,
1167 ENERGY_MEAN, energy_pred_fac);
1169 /* Calculate voice factor and store tilt for next subframe */
1170 voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
1171 ctx->fixed_vector, ctx->fixed_gain[0]);
1172 ctx->tilt_coef = voice_fac * 0.25 + 0.25;
1174 /* Construct current excitation */
1175 for (i = 0; i < AMRWB_SFR_SIZE; i++) {
1176 ctx->excitation[i] *= ctx->pitch_gain[0];
1177 ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
1178 ctx->excitation[i] = truncf(ctx->excitation[i]);
1181 /* Post-processing of excitation elements */
1182 synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
1183 voice_fac, stab_fac);
1185 synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
1188 pitch_enhancer(synth_fixed_vector, voice_fac);
1190 synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
1191 synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
1193 /* Synthesis speech post-processing */
1194 de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
1195 &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
1197 ff_acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
1198 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
1199 hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
1201 upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
1202 AMRWB_SFR_SIZE_16k);
1204 /* High frequency band (6.4 - 7.0 kHz) generation part */
1205 ff_acelp_apply_order_2_transfer_function(hb_samples,
1206 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
1207 hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
1209 hb_gain = find_hb_gain(ctx, hb_samples,
1210 cur_subframe->hb_gain, cf->vad);
1212 scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
1214 hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
1215 hb_exc, ctx->isf_cur, ctx->isf_past_final);
1217 /* High-band post-processing filters */
1218 hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
1219 &ctx->samples_hb[LP_ORDER_16k]);
1221 if (ctx->fr_cur_mode == MODE_23k85)
1222 hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
1225 /* Add the low and high frequency bands */
1226 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
1227 sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
1229 /* Update buffers and history */
1230 update_sub_state(ctx);
1233 /* update state for next frame */
1234 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
1235 memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
1238 *(AVFrame *)data = ctx->avframe;
1240 return expected_fr_size;
1243 AVCodec ff_amrwb_decoder = {
1245 .type = AVMEDIA_TYPE_AUDIO,
1246 .id = AV_CODEC_ID_AMR_WB,
1247 .priv_data_size = sizeof(AMRWBContext),
1248 .init = amrwb_decode_init,
1249 .decode = amrwb_decode_frame,
1250 .capabilities = CODEC_CAP_DR1,
1251 .long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
1252 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1253 AV_SAMPLE_FMT_NONE },