3 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
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
10 * version 2.1 of the License, or (at your option) any later version.
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 * @author Reynaldo H. Verdejo Pinochet
26 * @remark Libav merging spearheaded by Kenan Gillet
27 * @remark Development mentored by Benjamin Larson
32 #include "libavutil/channel_layout.h"
33 #include "libavutil/float_dsp.h"
37 #include "qcelpdata.h"
38 #include "celp_filters.h"
39 #include "acelp_filters.h"
40 #include "acelp_vectors.h"
47 I_F_Q
= -1, /**< insufficient frame quality */
57 qcelp_packet_rate bitrate
;
58 QCELPFrame frame
; /**< unpacked data frame */
60 uint8_t erasure_count
;
61 uint8_t octave_count
; /**< count the consecutive RATE_OCTAVE frames */
63 float predictor_lspf
[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
64 float pitch_synthesis_filter_mem
[303];
65 float pitch_pre_filter_mem
[303];
66 float rnd_fir_filter_mem
[180];
67 float formant_mem
[170];
68 float last_codebook_gain
;
74 uint8_t warned_buf_mismatch_bitrate
;
77 float postfilter_synth_mem
[10];
78 float postfilter_agc_mem
;
79 float postfilter_tilt_mem
;
83 * Initialize the speech codec according to the specification.
85 * TIA/EIA/IS-733 2.4.9
87 static av_cold
int qcelp_decode_init(AVCodecContext
*avctx
)
89 QCELPContext
*q
= avctx
->priv_data
;
93 avctx
->channel_layout
= AV_CH_LAYOUT_MONO
;
94 avctx
->sample_fmt
= AV_SAMPLE_FMT_FLT
;
96 for (i
= 0; i
< 10; i
++)
97 q
->prev_lspf
[i
] = (i
+ 1) / 11.;
103 * Decode the 10 quantized LSP frequencies from the LSPV/LSP
104 * transmission codes of any bitrate and check for badly received packets.
106 * @param q the context
107 * @param lspf line spectral pair frequencies
109 * @return 0 on success, -1 if the packet is badly received
111 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
113 static int decode_lspf(QCELPContext
*q
, float *lspf
)
116 float tmp_lspf
, smooth
, erasure_coeff
;
117 const float *predictors
;
119 if (q
->bitrate
== RATE_OCTAVE
|| q
->bitrate
== I_F_Q
) {
120 predictors
= q
->prev_bitrate
!= RATE_OCTAVE
&&
121 q
->prev_bitrate
!= I_F_Q
? q
->prev_lspf
124 if (q
->bitrate
== RATE_OCTAVE
) {
127 for (i
= 0; i
< 10; i
++) {
128 q
->predictor_lspf
[i
] =
129 lspf
[i
] = (q
->frame
.lspv
[i
] ? QCELP_LSP_SPREAD_FACTOR
130 : -QCELP_LSP_SPREAD_FACTOR
) +
131 predictors
[i
] * QCELP_LSP_OCTAVE_PREDICTOR
+
132 (i
+ 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR
) / 11);
134 smooth
= q
->octave_count
< 10 ? .875 : 0.1;
136 erasure_coeff
= QCELP_LSP_OCTAVE_PREDICTOR
;
138 assert(q
->bitrate
== I_F_Q
);
140 if (q
->erasure_count
> 1)
141 erasure_coeff
*= q
->erasure_count
< 4 ? 0.9 : 0.7;
143 for (i
= 0; i
< 10; i
++) {
144 q
->predictor_lspf
[i
] =
145 lspf
[i
] = (i
+ 1) * (1 - erasure_coeff
) / 11 +
146 erasure_coeff
* predictors
[i
];
151 // Check the stability of the LSP frequencies.
152 lspf
[0] = FFMAX(lspf
[0], QCELP_LSP_SPREAD_FACTOR
);
153 for (i
= 1; i
< 10; i
++)
154 lspf
[i
] = FFMAX(lspf
[i
], lspf
[i
- 1] + QCELP_LSP_SPREAD_FACTOR
);
156 lspf
[9] = FFMIN(lspf
[9], 1.0 - QCELP_LSP_SPREAD_FACTOR
);
157 for (i
= 9; i
> 0; i
--)
158 lspf
[i
- 1] = FFMIN(lspf
[i
- 1], lspf
[i
] - QCELP_LSP_SPREAD_FACTOR
);
160 // Low-pass filter the LSP frequencies.
161 ff_weighted_vector_sumf(lspf
, lspf
, q
->prev_lspf
, smooth
, 1.0 - smooth
, 10);
166 for (i
= 0; i
< 5; i
++) {
167 lspf
[2 * i
+ 0] = tmp_lspf
+= qcelp_lspvq
[i
][q
->frame
.lspv
[i
]][0] * 0.0001;
168 lspf
[2 * i
+ 1] = tmp_lspf
+= qcelp_lspvq
[i
][q
->frame
.lspv
[i
]][1] * 0.0001;
171 // Check for badly received packets.
172 if (q
->bitrate
== RATE_QUARTER
) {
173 if (lspf
[9] <= .70 || lspf
[9] >= .97)
175 for (i
= 3; i
< 10; i
++)
176 if (fabs(lspf
[i
] - lspf
[i
- 2]) < .08)
179 if (lspf
[9] <= .66 || lspf
[9] >= .985)
181 for (i
= 4; i
< 10; i
++)
182 if (fabs(lspf
[i
] - lspf
[i
- 4]) < .0931)
190 * Convert codebook transmission codes to GAIN and INDEX.
192 * @param q the context
193 * @param gain array holding the decoded gain
195 * TIA/EIA/IS-733 2.4.6.2
197 static void decode_gain_and_index(QCELPContext
*q
, float *gain
)
199 int i
, subframes_count
, g1
[16];
202 if (q
->bitrate
>= RATE_QUARTER
) {
203 switch (q
->bitrate
) {
204 case RATE_FULL
: subframes_count
= 16; break;
205 case RATE_HALF
: subframes_count
= 4; break;
206 default: subframes_count
= 5;
208 for (i
= 0; i
< subframes_count
; i
++) {
209 g1
[i
] = 4 * q
->frame
.cbgain
[i
];
210 if (q
->bitrate
== RATE_FULL
&& !((i
+ 1) & 3)) {
211 g1
[i
] += av_clip((g1
[i
- 1] + g1
[i
- 2] + g1
[i
- 3]) / 3 - 6, 0, 32);
214 gain
[i
] = qcelp_g12ga
[g1
[i
]];
216 if (q
->frame
.cbsign
[i
]) {
218 q
->frame
.cindex
[i
] = (q
->frame
.cindex
[i
] - 89) & 127;
222 q
->prev_g1
[0] = g1
[i
- 2];
223 q
->prev_g1
[1] = g1
[i
- 1];
224 q
->last_codebook_gain
= qcelp_g12ga
[g1
[i
- 1]];
226 if (q
->bitrate
== RATE_QUARTER
) {
227 // Provide smoothing of the unvoiced excitation energy.
229 gain
[6] = 0.4 * gain
[3] + 0.6 * gain
[4];
231 gain
[4] = 0.8 * gain
[2] + 0.2 * gain
[3];
232 gain
[3] = 0.2 * gain
[1] + 0.8 * gain
[2];
234 gain
[1] = 0.6 * gain
[0] + 0.4 * gain
[1];
236 } else if (q
->bitrate
!= SILENCE
) {
237 if (q
->bitrate
== RATE_OCTAVE
) {
238 g1
[0] = 2 * q
->frame
.cbgain
[0] +
239 av_clip((q
->prev_g1
[0] + q
->prev_g1
[1]) / 2 - 5, 0, 54);
242 assert(q
->bitrate
== I_F_Q
);
244 g1
[0] = q
->prev_g1
[1];
245 switch (q
->erasure_count
) {
247 case 2 : g1
[0] -= 1; break;
248 case 3 : g1
[0] -= 2; break;
255 // This interpolation is done to produce smoother background noise.
256 slope
= 0.5 * (qcelp_g12ga
[g1
[0]] - q
->last_codebook_gain
) / subframes_count
;
257 for (i
= 1; i
<= subframes_count
; i
++)
258 gain
[i
- 1] = q
->last_codebook_gain
+ slope
* i
;
260 q
->last_codebook_gain
= gain
[i
- 2];
261 q
->prev_g1
[0] = q
->prev_g1
[1];
262 q
->prev_g1
[1] = g1
[0];
267 * If the received packet is Rate 1/4 a further sanity check is made of the
270 * @param cbgain the unpacked cbgain array
271 * @return -1 if the sanity check fails, 0 otherwise
273 * TIA/EIA/IS-733 2.4.8.7.3
275 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain
)
277 int i
, diff
, prev_diff
= 0;
279 for (i
= 1; i
< 5; i
++) {
280 diff
= cbgain
[i
] - cbgain
[i
-1];
281 if (FFABS(diff
) > 10)
283 else if (FFABS(diff
- prev_diff
) > 12)
291 * Compute the scaled codebook vector Cdn From INDEX and GAIN
294 * The specification lacks some information here.
296 * TIA/EIA/IS-733 has an omission on the codebook index determination
297 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
298 * you have to subtract the decoded index parameter from the given scaled
299 * codebook vector index 'n' to get the desired circular codebook index, but
300 * it does not mention that you have to clamp 'n' to [0-9] in order to get
301 * RI-compliant results.
303 * The reason for this mistake seems to be the fact they forgot to mention you
304 * have to do these calculations per codebook subframe and adjust given
305 * equation values accordingly.
307 * @param q the context
308 * @param gain array holding the 4 pitch subframe gain values
309 * @param cdn_vector array for the generated scaled codebook vector
311 static void compute_svector(QCELPContext
*q
, const float *gain
,
315 uint16_t cbseed
, cindex
;
316 float *rnd
, tmp_gain
, fir_filter_value
;
318 switch (q
->bitrate
) {
320 for (i
= 0; i
< 16; i
++) {
321 tmp_gain
= gain
[i
] * QCELP_RATE_FULL_CODEBOOK_RATIO
;
322 cindex
= -q
->frame
.cindex
[i
];
323 for (j
= 0; j
< 10; j
++)
324 *cdn_vector
++ = tmp_gain
* qcelp_rate_full_codebook
[cindex
++ & 127];
328 for (i
= 0; i
< 4; i
++) {
329 tmp_gain
= gain
[i
] * QCELP_RATE_HALF_CODEBOOK_RATIO
;
330 cindex
= -q
->frame
.cindex
[i
];
331 for (j
= 0; j
< 40; j
++)
332 *cdn_vector
++ = tmp_gain
* qcelp_rate_half_codebook
[cindex
++ & 127];
336 cbseed
= (0x0003 & q
->frame
.lspv
[4]) << 14 |
337 (0x003F & q
->frame
.lspv
[3]) << 8 |
338 (0x0060 & q
->frame
.lspv
[2]) << 1 |
339 (0x0007 & q
->frame
.lspv
[1]) << 3 |
340 (0x0038 & q
->frame
.lspv
[0]) >> 3;
341 rnd
= q
->rnd_fir_filter_mem
+ 20;
342 for (i
= 0; i
< 8; i
++) {
343 tmp_gain
= gain
[i
] * (QCELP_SQRT1887
/ 32768.0);
344 for (k
= 0; k
< 20; k
++) {
345 cbseed
= 521 * cbseed
+ 259;
346 *rnd
= (int16_t) cbseed
;
349 fir_filter_value
= 0.0;
350 for (j
= 0; j
< 10; j
++)
351 fir_filter_value
+= qcelp_rnd_fir_coefs
[j
] *
352 (rnd
[-j
] + rnd
[-20+j
]);
354 fir_filter_value
+= qcelp_rnd_fir_coefs
[10] * rnd
[-10];
355 *cdn_vector
++ = tmp_gain
* fir_filter_value
;
359 memcpy(q
->rnd_fir_filter_mem
, q
->rnd_fir_filter_mem
+ 160,
363 cbseed
= q
->first16bits
;
364 for (i
= 0; i
< 8; i
++) {
365 tmp_gain
= gain
[i
] * (QCELP_SQRT1887
/ 32768.0);
366 for (j
= 0; j
< 20; j
++) {
367 cbseed
= 521 * cbseed
+ 259;
368 *cdn_vector
++ = tmp_gain
* (int16_t) cbseed
;
373 cbseed
= -44; // random codebook index
374 for (i
= 0; i
< 4; i
++) {
375 tmp_gain
= gain
[i
] * QCELP_RATE_FULL_CODEBOOK_RATIO
;
376 for (j
= 0; j
< 40; j
++)
377 *cdn_vector
++ = tmp_gain
* qcelp_rate_full_codebook
[cbseed
++ & 127];
381 memset(cdn_vector
, 0, 160 * sizeof(float));
387 * Apply generic gain control.
389 * @param v_out output vector
390 * @param v_in gain-controlled vector
391 * @param v_ref vector to control gain of
393 * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
395 static void apply_gain_ctrl(float *v_out
, const float *v_ref
, const float *v_in
)
399 for (i
= 0; i
< 160; i
+= 40) {
400 float res
= avpriv_scalarproduct_float_c(v_ref
+ i
, v_ref
+ i
, 40);
401 ff_scale_vector_to_given_sum_of_squares(v_out
+ i
, v_in
+ i
, res
, 40);
406 * Apply filter in pitch-subframe steps.
408 * @param memory buffer for the previous state of the filter
409 * - must be able to contain 303 elements
410 * - the 143 first elements are from the previous state
411 * - the next 160 are for output
412 * @param v_in input filter vector
413 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
414 * @param lag per-subframe lag array, each element is
415 * - between 16 and 143 if its corresponding pfrac is 0,
416 * - between 16 and 139 otherwise
417 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
420 * @return filter output vector
422 static const float *do_pitchfilter(float memory
[303], const float v_in
[160],
423 const float gain
[4], const uint8_t *lag
,
424 const uint8_t pfrac
[4])
427 float *v_lag
, *v_out
;
430 v_out
= memory
+ 143; // Output vector starts at memory[143].
432 for (i
= 0; i
< 4; i
++) {
434 v_lag
= memory
+ 143 + 40 * i
- lag
[i
];
435 for (v_len
= v_in
+ 40; v_in
< v_len
; v_in
++) {
436 if (pfrac
[i
]) { // If it is a fractional lag...
437 for (j
= 0, *v_out
= 0.; j
< 4; j
++)
438 *v_out
+= qcelp_hammsinc_table
[j
] * (v_lag
[j
- 4] + v_lag
[3 - j
]);
442 *v_out
= *v_in
+ gain
[i
] * *v_out
;
448 memcpy(v_out
, v_in
, 40 * sizeof(float));
454 memmove(memory
, memory
+ 160, 143 * sizeof(float));
459 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
460 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
462 * @param q the context
463 * @param cdn_vector the scaled codebook vector
465 static void apply_pitch_filters(QCELPContext
*q
, float *cdn_vector
)
468 const float *v_synthesis_filtered
, *v_pre_filtered
;
470 if (q
->bitrate
>= RATE_HALF
|| q
->bitrate
== SILENCE
||
471 (q
->bitrate
== I_F_Q
&& (q
->prev_bitrate
>= RATE_HALF
))) {
473 if (q
->bitrate
>= RATE_HALF
) {
474 // Compute gain & lag for the whole frame.
475 for (i
= 0; i
< 4; i
++) {
476 q
->pitch_gain
[i
] = q
->frame
.plag
[i
] ? (q
->frame
.pgain
[i
] + 1) * 0.25 : 0.0;
478 q
->pitch_lag
[i
] = q
->frame
.plag
[i
] + 16;
481 float max_pitch_gain
;
483 if (q
->bitrate
== I_F_Q
) {
484 if (q
->erasure_count
< 3)
485 max_pitch_gain
= 0.9 - 0.3 * (q
->erasure_count
- 1);
487 max_pitch_gain
= 0.0;
489 assert(q
->bitrate
== SILENCE
);
490 max_pitch_gain
= 1.0;
492 for (i
= 0; i
< 4; i
++)
493 q
->pitch_gain
[i
] = FFMIN(q
->pitch_gain
[i
], max_pitch_gain
);
495 memset(q
->frame
.pfrac
, 0, sizeof(q
->frame
.pfrac
));
498 // pitch synthesis filter
499 v_synthesis_filtered
= do_pitchfilter(q
->pitch_synthesis_filter_mem
,
500 cdn_vector
, q
->pitch_gain
,
501 q
->pitch_lag
, q
->frame
.pfrac
);
503 // pitch prefilter update
504 for (i
= 0; i
< 4; i
++)
505 q
->pitch_gain
[i
] = 0.5 * FFMIN(q
->pitch_gain
[i
], 1.0);
507 v_pre_filtered
= do_pitchfilter(q
->pitch_pre_filter_mem
,
508 v_synthesis_filtered
,
509 q
->pitch_gain
, q
->pitch_lag
,
512 apply_gain_ctrl(cdn_vector
, v_synthesis_filtered
, v_pre_filtered
);
514 memcpy(q
->pitch_synthesis_filter_mem
, cdn_vector
+ 17, 143 * sizeof(float));
515 memcpy(q
->pitch_pre_filter_mem
, cdn_vector
+ 17, 143 * sizeof(float));
516 memset(q
->pitch_gain
, 0, sizeof(q
->pitch_gain
));
517 memset(q
->pitch_lag
, 0, sizeof(q
->pitch_lag
));
522 * Reconstruct LPC coefficients from the line spectral pair frequencies
523 * and perform bandwidth expansion.
525 * @param lspf line spectral pair frequencies
526 * @param lpc linear predictive coding coefficients
528 * @note: bandwidth_expansion_coeff could be precalculated into a table
529 * but it seems to be slower on x86
531 * TIA/EIA/IS-733 2.4.3.3.5
533 static void lspf2lpc(const float *lspf
, float *lpc
)
536 double bandwidth_expansion_coeff
= QCELP_BANDWIDTH_EXPANSION_COEFF
;
539 for (i
= 0; i
< 10; i
++)
540 lsp
[i
] = cos(M_PI
* lspf
[i
]);
542 ff_acelp_lspd2lpc(lsp
, lpc
, 5);
544 for (i
= 0; i
< 10; i
++) {
545 lpc
[i
] *= bandwidth_expansion_coeff
;
546 bandwidth_expansion_coeff
*= QCELP_BANDWIDTH_EXPANSION_COEFF
;
551 * Interpolate LSP frequencies and compute LPC coefficients
552 * for a given bitrate & pitch subframe.
554 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
556 * @param q the context
557 * @param curr_lspf LSP frequencies vector of the current frame
558 * @param lpc float vector for the resulting LPC
559 * @param subframe_num frame number in decoded stream
561 static void interpolate_lpc(QCELPContext
*q
, const float *curr_lspf
,
562 float *lpc
, const int subframe_num
)
564 float interpolated_lspf
[10];
567 if (q
->bitrate
>= RATE_QUARTER
)
568 weight
= 0.25 * (subframe_num
+ 1);
569 else if (q
->bitrate
== RATE_OCTAVE
&& !subframe_num
)
575 ff_weighted_vector_sumf(interpolated_lspf
, curr_lspf
, q
->prev_lspf
,
576 weight
, 1.0 - weight
, 10);
577 lspf2lpc(interpolated_lspf
, lpc
);
578 } else if (q
->bitrate
>= RATE_QUARTER
||
579 (q
->bitrate
== I_F_Q
&& !subframe_num
))
580 lspf2lpc(curr_lspf
, lpc
);
581 else if (q
->bitrate
== SILENCE
&& !subframe_num
)
582 lspf2lpc(q
->prev_lspf
, lpc
);
585 static qcelp_packet_rate
buf_size2bitrate(const int buf_size
)
588 case 35: return RATE_FULL
;
589 case 17: return RATE_HALF
;
590 case 8: return RATE_QUARTER
;
591 case 4: return RATE_OCTAVE
;
592 case 1: return SILENCE
;
599 * Determine the bitrate from the frame size and/or the first byte of the frame.
601 * @param avctx the AV codec context
602 * @param buf_size length of the buffer
603 * @param buf the bufffer
605 * @return the bitrate on success,
606 * I_F_Q if the bitrate cannot be satisfactorily determined
608 * TIA/EIA/IS-733 2.4.8.7.1
610 static qcelp_packet_rate
determine_bitrate(AVCodecContext
*avctx
,
614 qcelp_packet_rate bitrate
;
616 if ((bitrate
= buf_size2bitrate(buf_size
)) >= 0) {
617 if (bitrate
> **buf
) {
618 QCELPContext
*q
= avctx
->priv_data
;
619 if (!q
->warned_buf_mismatch_bitrate
) {
620 av_log(avctx
, AV_LOG_WARNING
,
621 "Claimed bitrate and buffer size mismatch.\n");
622 q
->warned_buf_mismatch_bitrate
= 1;
625 } else if (bitrate
< **buf
) {
626 av_log(avctx
, AV_LOG_ERROR
,
627 "Buffer is too small for the claimed bitrate.\n");
631 } else if ((bitrate
= buf_size2bitrate(buf_size
+ 1)) >= 0) {
632 av_log(avctx
, AV_LOG_WARNING
,
633 "Bitrate byte is missing, guessing the bitrate from packet size.\n");
637 if (bitrate
== SILENCE
) {
638 //FIXME: Remove experimental warning when tested with samples.
639 av_log_ask_for_sample(avctx
, "'Blank frame handling is experimental.");
644 static void warn_insufficient_frame_quality(AVCodecContext
*avctx
,
647 av_log(avctx
, AV_LOG_WARNING
, "Frame #%d, IFQ: %s\n",
648 avctx
->frame_number
, message
);
651 static void postfilter(QCELPContext
*q
, float *samples
, float *lpc
)
653 static const float pow_0_775
[10] = {
654 0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
655 0.216676, 0.167924, 0.130141, 0.100859, 0.078166
657 0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
658 0.059605, 0.037253, 0.023283, 0.014552, 0.009095
660 float lpc_s
[10], lpc_p
[10], pole_out
[170], zero_out
[160];
663 for (n
= 0; n
< 10; n
++) {
664 lpc_s
[n
] = lpc
[n
] * pow_0_625
[n
];
665 lpc_p
[n
] = lpc
[n
] * pow_0_775
[n
];
668 ff_celp_lp_zero_synthesis_filterf(zero_out
, lpc_s
,
669 q
->formant_mem
+ 10, 160, 10);
670 memcpy(pole_out
, q
->postfilter_synth_mem
, sizeof(float) * 10);
671 ff_celp_lp_synthesis_filterf(pole_out
+ 10, lpc_p
, zero_out
, 160, 10);
672 memcpy(q
->postfilter_synth_mem
, pole_out
+ 160, sizeof(float) * 10);
674 ff_tilt_compensation(&q
->postfilter_tilt_mem
, 0.3, pole_out
+ 10, 160);
676 ff_adaptive_gain_control(samples
, pole_out
+ 10,
677 avpriv_scalarproduct_float_c(q
->formant_mem
+ 10,
680 160, 0.9375, &q
->postfilter_agc_mem
);
683 static int qcelp_decode_frame(AVCodecContext
*avctx
, void *data
,
684 int *got_frame_ptr
, AVPacket
*avpkt
)
686 const uint8_t *buf
= avpkt
->data
;
687 int buf_size
= avpkt
->size
;
688 QCELPContext
*q
= avctx
->priv_data
;
689 AVFrame
*frame
= data
;
692 float quantized_lspf
[10], lpc
[10];
696 /* get output buffer */
697 frame
->nb_samples
= 160;
698 if ((ret
= ff_get_buffer(avctx
, frame
)) < 0) {
699 av_log(avctx
, AV_LOG_ERROR
, "get_buffer() failed\n");
702 outbuffer
= (float *)frame
->data
[0];
704 if ((q
->bitrate
= determine_bitrate(avctx
, buf_size
, &buf
)) == I_F_Q
) {
705 warn_insufficient_frame_quality(avctx
, "bitrate cannot be determined.");
709 if (q
->bitrate
== RATE_OCTAVE
&&
710 (q
->first16bits
= AV_RB16(buf
)) == 0xFFFF) {
711 warn_insufficient_frame_quality(avctx
, "Bitrate is 1/8 and first 16 bits are on.");
715 if (q
->bitrate
> SILENCE
) {
716 const QCELPBitmap
*bitmaps
= qcelp_unpacking_bitmaps_per_rate
[q
->bitrate
];
717 const QCELPBitmap
*bitmaps_end
= qcelp_unpacking_bitmaps_per_rate
[q
->bitrate
] +
718 qcelp_unpacking_bitmaps_lengths
[q
->bitrate
];
719 uint8_t *unpacked_data
= (uint8_t *)&q
->frame
;
721 init_get_bits(&q
->gb
, buf
, 8 * buf_size
);
723 memset(&q
->frame
, 0, sizeof(QCELPFrame
));
725 for (; bitmaps
< bitmaps_end
; bitmaps
++)
726 unpacked_data
[bitmaps
->index
] |= get_bits(&q
->gb
, bitmaps
->bitlen
) << bitmaps
->bitpos
;
728 // Check for erasures/blanks on rates 1, 1/4 and 1/8.
729 if (q
->frame
.reserved
) {
730 warn_insufficient_frame_quality(avctx
, "Wrong data in reserved frame area.");
733 if (q
->bitrate
== RATE_QUARTER
&&
734 codebook_sanity_check_for_rate_quarter(q
->frame
.cbgain
)) {
735 warn_insufficient_frame_quality(avctx
, "Codebook gain sanity check failed.");
739 if (q
->bitrate
>= RATE_HALF
) {
740 for (i
= 0; i
< 4; i
++) {
741 if (q
->frame
.pfrac
[i
] && q
->frame
.plag
[i
] >= 124) {
742 warn_insufficient_frame_quality(avctx
, "Cannot initialize pitch filter.");
749 decode_gain_and_index(q
, gain
);
750 compute_svector(q
, gain
, outbuffer
);
752 if (decode_lspf(q
, quantized_lspf
) < 0) {
753 warn_insufficient_frame_quality(avctx
, "Badly received packets in frame.");
757 apply_pitch_filters(q
, outbuffer
);
759 if (q
->bitrate
== I_F_Q
) {
763 decode_gain_and_index(q
, gain
);
764 compute_svector(q
, gain
, outbuffer
);
765 decode_lspf(q
, quantized_lspf
);
766 apply_pitch_filters(q
, outbuffer
);
768 q
->erasure_count
= 0;
770 formant_mem
= q
->formant_mem
+ 10;
771 for (i
= 0; i
< 4; i
++) {
772 interpolate_lpc(q
, quantized_lspf
, lpc
, i
);
773 ff_celp_lp_synthesis_filterf(formant_mem
, lpc
, outbuffer
+ i
* 40, 40, 10);
777 // postfilter, as per TIA/EIA/IS-733 2.4.8.6
778 postfilter(q
, outbuffer
, lpc
);
780 memcpy(q
->formant_mem
, q
->formant_mem
+ 160, 10 * sizeof(float));
782 memcpy(q
->prev_lspf
, quantized_lspf
, sizeof(q
->prev_lspf
));
783 q
->prev_bitrate
= q
->bitrate
;
790 AVCodec ff_qcelp_decoder
= {
792 .type
= AVMEDIA_TYPE_AUDIO
,
793 .id
= AV_CODEC_ID_QCELP
,
794 .init
= qcelp_decode_init
,
795 .decode
= qcelp_decode_frame
,
796 .capabilities
= CODEC_CAP_DR1
,
797 .priv_data_size
= sizeof(QCELPContext
),
798 .long_name
= NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),