3 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
5 * This file is part of FFmpeg.
7 * FFmpeg 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 * FFmpeg 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 FFmpeg; 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 FFmpeg merging spearheaded by Kenan Gillet
27 * @remark Development mentored by Benjamin Larson
30 #include "libavutil/avassert.h"
31 #include "libavutil/channel_layout.h"
32 #include "libavutil/float_dsp.h"
34 #include "codec_internal.h"
37 #include "qcelpdata.h"
38 #include "celp_filters.h"
39 #include "acelp_filters.h"
40 #include "acelp_vectors.h"
44 I_F_Q
= -1, /**< insufficient frame quality */
52 typedef struct QCELPContext
{
54 qcelp_packet_rate bitrate
;
55 QCELPFrame frame
; /**< unpacked data frame */
57 uint8_t erasure_count
;
58 uint8_t octave_count
; /**< count the consecutive RATE_OCTAVE frames */
60 float predictor_lspf
[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
61 float pitch_synthesis_filter_mem
[303];
62 float pitch_pre_filter_mem
[303];
63 float rnd_fir_filter_mem
[180];
64 float formant_mem
[170];
65 float last_codebook_gain
;
71 uint8_t warned_buf_mismatch_bitrate
;
74 float postfilter_synth_mem
[10];
75 float postfilter_agc_mem
;
76 float postfilter_tilt_mem
;
80 * Initialize the speech codec according to the specification.
82 * TIA/EIA/IS-733 2.4.9
84 static av_cold
int qcelp_decode_init(AVCodecContext
*avctx
)
86 QCELPContext
*q
= avctx
->priv_data
;
89 av_channel_layout_uninit(&avctx
->ch_layout
);
90 avctx
->ch_layout
= (AVChannelLayout
)AV_CHANNEL_LAYOUT_MONO
;
91 avctx
->sample_fmt
= AV_SAMPLE_FMT_FLT
;
93 for (i
= 0; i
< 10; i
++)
94 q
->prev_lspf
[i
] = (i
+ 1) / 11.0;
100 * Decode the 10 quantized LSP frequencies from the LSPV/LSP
101 * transmission codes of any bitrate and check for badly received packets.
103 * @param q the context
104 * @param lspf line spectral pair frequencies
106 * @return 0 on success, -1 if the packet is badly received
108 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
110 static int decode_lspf(QCELPContext
*q
, float *lspf
)
113 float tmp_lspf
, smooth
, erasure_coeff
;
114 const float *predictors
;
116 if (q
->bitrate
== RATE_OCTAVE
|| q
->bitrate
== I_F_Q
) {
117 predictors
= q
->prev_bitrate
!= RATE_OCTAVE
&&
118 q
->prev_bitrate
!= I_F_Q
? q
->prev_lspf
121 if (q
->bitrate
== RATE_OCTAVE
) {
124 for (i
= 0; i
< 10; i
++) {
125 q
->predictor_lspf
[i
] =
126 lspf
[i
] = (q
->frame
.lspv
[i
] ? QCELP_LSP_SPREAD_FACTOR
127 : -QCELP_LSP_SPREAD_FACTOR
) +
128 predictors
[i
] * QCELP_LSP_OCTAVE_PREDICTOR
+
129 (i
+ 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR
) / 11);
131 smooth
= q
->octave_count
< 10 ? .875 : 0.1;
133 erasure_coeff
= QCELP_LSP_OCTAVE_PREDICTOR
;
135 av_assert2(q
->bitrate
== I_F_Q
);
137 if (q
->erasure_count
> 1)
138 erasure_coeff
*= q
->erasure_count
< 4 ? 0.9 : 0.7;
140 for (i
= 0; i
< 10; i
++) {
141 q
->predictor_lspf
[i
] =
142 lspf
[i
] = (i
+ 1) * (1 - erasure_coeff
) / 11 +
143 erasure_coeff
* predictors
[i
];
148 // Check the stability of the LSP frequencies.
149 lspf
[0] = FFMAX(lspf
[0], QCELP_LSP_SPREAD_FACTOR
);
150 for (i
= 1; i
< 10; i
++)
151 lspf
[i
] = FFMAX(lspf
[i
], lspf
[i
- 1] + QCELP_LSP_SPREAD_FACTOR
);
153 lspf
[9] = FFMIN(lspf
[9], 1.0 - QCELP_LSP_SPREAD_FACTOR
);
154 for (i
= 9; i
> 0; i
--)
155 lspf
[i
- 1] = FFMIN(lspf
[i
- 1], lspf
[i
] - QCELP_LSP_SPREAD_FACTOR
);
157 // Low-pass filter the LSP frequencies.
158 ff_weighted_vector_sumf(lspf
, lspf
, q
->prev_lspf
, smooth
, 1.0 - smooth
, 10);
163 for (i
= 0; i
< 5; i
++) {
164 lspf
[2 * i
+ 0] = tmp_lspf
+= qcelp_lspvq
[i
][q
->frame
.lspv
[i
]][0] * 0.0001;
165 lspf
[2 * i
+ 1] = tmp_lspf
+= qcelp_lspvq
[i
][q
->frame
.lspv
[i
]][1] * 0.0001;
168 // Check for badly received packets.
169 if (q
->bitrate
== RATE_QUARTER
) {
170 if (lspf
[9] <= .70 || lspf
[9] >= .97)
172 for (i
= 3; i
< 10; i
++)
173 if (fabs(lspf
[i
] - lspf
[i
- 2]) < .08)
176 if (lspf
[9] <= .66 || lspf
[9] >= .985)
178 for (i
= 4; i
< 10; i
++)
179 if (fabs(lspf
[i
] - lspf
[i
- 4]) < .0931)
187 * Convert codebook transmission codes to GAIN and INDEX.
189 * @param q the context
190 * @param gain array holding the decoded gain
192 * TIA/EIA/IS-733 2.4.6.2
194 static void decode_gain_and_index(QCELPContext
*q
, float *gain
)
196 int i
, subframes_count
, g1
[16];
199 if (q
->bitrate
>= RATE_QUARTER
) {
200 switch (q
->bitrate
) {
201 case RATE_FULL
: subframes_count
= 16; break;
202 case RATE_HALF
: subframes_count
= 4; break;
203 default: subframes_count
= 5;
205 for (i
= 0; i
< subframes_count
; i
++) {
206 g1
[i
] = 4 * q
->frame
.cbgain
[i
];
207 if (q
->bitrate
== RATE_FULL
&& !((i
+ 1) & 3)) {
208 g1
[i
] += av_clip((g1
[i
- 1] + g1
[i
- 2] + g1
[i
- 3]) / 3 - 6, 0, 32);
211 gain
[i
] = qcelp_g12ga
[g1
[i
]];
213 if (q
->frame
.cbsign
[i
]) {
215 q
->frame
.cindex
[i
] = (q
->frame
.cindex
[i
] - 89) & 127;
219 q
->prev_g1
[0] = g1
[i
- 2];
220 q
->prev_g1
[1] = g1
[i
- 1];
221 q
->last_codebook_gain
= qcelp_g12ga
[g1
[i
- 1]];
223 if (q
->bitrate
== RATE_QUARTER
) {
224 // Provide smoothing of the unvoiced excitation energy.
226 gain
[6] = 0.4 * gain
[3] + 0.6 * gain
[4];
228 gain
[4] = 0.8 * gain
[2] + 0.2 * gain
[3];
229 gain
[3] = 0.2 * gain
[1] + 0.8 * gain
[2];
231 gain
[1] = 0.6 * gain
[0] + 0.4 * gain
[1];
233 } else if (q
->bitrate
!= SILENCE
) {
234 if (q
->bitrate
== RATE_OCTAVE
) {
235 g1
[0] = 2 * q
->frame
.cbgain
[0] +
236 av_clip((q
->prev_g1
[0] + q
->prev_g1
[1]) / 2 - 5, 0, 54);
239 av_assert2(q
->bitrate
== I_F_Q
);
241 g1
[0] = q
->prev_g1
[1];
242 switch (q
->erasure_count
) {
244 case 2 : g1
[0] -= 1; break;
245 case 3 : g1
[0] -= 2; break;
252 // This interpolation is done to produce smoother background noise.
253 slope
= 0.5 * (qcelp_g12ga
[g1
[0]] - q
->last_codebook_gain
) / subframes_count
;
254 for (i
= 1; i
<= subframes_count
; i
++)
255 gain
[i
- 1] = q
->last_codebook_gain
+ slope
* i
;
257 q
->last_codebook_gain
= gain
[i
- 2];
258 q
->prev_g1
[0] = q
->prev_g1
[1];
259 q
->prev_g1
[1] = g1
[0];
264 * If the received packet is Rate 1/4 a further sanity check is made of the
267 * @param cbgain the unpacked cbgain array
268 * @return -1 if the sanity check fails, 0 otherwise
270 * TIA/EIA/IS-733 2.4.8.7.3
272 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain
)
274 int i
, diff
, prev_diff
= 0;
276 for (i
= 1; i
< 5; i
++) {
277 diff
= cbgain
[i
] - cbgain
[i
-1];
278 if (FFABS(diff
) > 10)
280 else if (FFABS(diff
- prev_diff
) > 12)
288 * Compute the scaled codebook vector Cdn From INDEX and GAIN
291 * The specification lacks some information here.
293 * TIA/EIA/IS-733 has an omission on the codebook index determination
294 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
295 * you have to subtract the decoded index parameter from the given scaled
296 * codebook vector index 'n' to get the desired circular codebook index, but
297 * it does not mention that you have to clamp 'n' to [0-9] in order to get
298 * RI-compliant results.
300 * The reason for this mistake seems to be the fact they forgot to mention you
301 * have to do these calculations per codebook subframe and adjust given
302 * equation values accordingly.
304 * @param q the context
305 * @param gain array holding the 4 pitch subframe gain values
306 * @param cdn_vector array for the generated scaled codebook vector
308 static void compute_svector(QCELPContext
*q
, const float *gain
,
312 uint16_t cbseed
, cindex
;
313 float *rnd
, tmp_gain
, fir_filter_value
;
315 switch (q
->bitrate
) {
317 for (i
= 0; i
< 16; i
++) {
318 tmp_gain
= gain
[i
] * QCELP_RATE_FULL_CODEBOOK_RATIO
;
319 cindex
= -q
->frame
.cindex
[i
];
320 for (j
= 0; j
< 10; j
++)
321 *cdn_vector
++ = tmp_gain
*
322 qcelp_rate_full_codebook
[cindex
++ & 127];
326 for (i
= 0; i
< 4; i
++) {
327 tmp_gain
= gain
[i
] * QCELP_RATE_HALF_CODEBOOK_RATIO
;
328 cindex
= -q
->frame
.cindex
[i
];
329 for (j
= 0; j
< 40; j
++)
330 *cdn_vector
++ = tmp_gain
*
331 qcelp_rate_half_codebook
[cindex
++ & 127];
335 cbseed
= (0x0003 & q
->frame
.lspv
[4]) << 14 |
336 (0x003F & q
->frame
.lspv
[3]) << 8 |
337 (0x0060 & q
->frame
.lspv
[2]) << 1 |
338 (0x0007 & q
->frame
.lspv
[1]) << 3 |
339 (0x0038 & q
->frame
.lspv
[0]) >> 3;
340 rnd
= q
->rnd_fir_filter_mem
+ 20;
341 for (i
= 0; i
< 8; i
++) {
342 tmp_gain
= gain
[i
] * (QCELP_SQRT1887
/ 32768.0);
343 for (k
= 0; k
< 20; k
++) {
344 cbseed
= 521 * cbseed
+ 259;
345 *rnd
= (int16_t) cbseed
;
348 fir_filter_value
= 0.0;
349 for (j
= 0; j
< 10; j
++)
350 fir_filter_value
+= qcelp_rnd_fir_coefs
[j
] *
351 (rnd
[-j
] + rnd
[-20+j
]);
353 fir_filter_value
+= qcelp_rnd_fir_coefs
[10] * rnd
[-10];
354 *cdn_vector
++ = tmp_gain
* fir_filter_value
;
358 memcpy(q
->rnd_fir_filter_mem
, q
->rnd_fir_filter_mem
+ 160,
362 cbseed
= q
->first16bits
;
363 for (i
= 0; i
< 8; i
++) {
364 tmp_gain
= gain
[i
] * (QCELP_SQRT1887
/ 32768.0);
365 for (j
= 0; j
< 20; j
++) {
366 cbseed
= 521 * cbseed
+ 259;
367 *cdn_vector
++ = tmp_gain
* (int16_t) cbseed
;
372 cbseed
= -44; // random codebook index
373 for (i
= 0; i
< 4; i
++) {
374 tmp_gain
= gain
[i
] * QCELP_RATE_FULL_CODEBOOK_RATIO
;
375 for (j
= 0; j
< 40; j
++)
376 *cdn_vector
++ = tmp_gain
*
377 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
= 0.0; j
< 4; j
++)
438 *v_out
+= qcelp_hammsinc_table
[j
] *
439 (v_lag
[j
- 4] + v_lag
[3 - j
]);
443 *v_out
= *v_in
+ gain
[i
] * *v_out
;
449 memcpy(v_out
, v_in
, 40 * sizeof(float));
455 memmove(memory
, memory
+ 160, 143 * sizeof(float));
460 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
461 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
463 * @param q the context
464 * @param cdn_vector the scaled codebook vector
466 static void apply_pitch_filters(QCELPContext
*q
, float *cdn_vector
)
469 const float *v_synthesis_filtered
, *v_pre_filtered
;
471 if (q
->bitrate
>= RATE_HALF
|| q
->bitrate
== SILENCE
||
472 (q
->bitrate
== I_F_Q
&& (q
->prev_bitrate
>= RATE_HALF
))) {
474 if (q
->bitrate
>= RATE_HALF
) {
475 // Compute gain & lag for the whole frame.
476 for (i
= 0; i
< 4; i
++) {
477 q
->pitch_gain
[i
] = q
->frame
.plag
[i
] ? (q
->frame
.pgain
[i
] + 1) * 0.25 : 0.0;
479 q
->pitch_lag
[i
] = q
->frame
.plag
[i
] + 16;
482 float max_pitch_gain
;
484 if (q
->bitrate
== I_F_Q
) {
485 if (q
->erasure_count
< 3)
486 max_pitch_gain
= 0.9 - 0.3 * (q
->erasure_count
- 1);
488 max_pitch_gain
= 0.0;
490 av_assert2(q
->bitrate
== SILENCE
);
491 max_pitch_gain
= 1.0;
493 for (i
= 0; i
< 4; i
++)
494 q
->pitch_gain
[i
] = FFMIN(q
->pitch_gain
[i
], max_pitch_gain
);
496 memset(q
->frame
.pfrac
, 0, sizeof(q
->frame
.pfrac
));
499 // pitch synthesis filter
500 v_synthesis_filtered
= do_pitchfilter(q
->pitch_synthesis_filter_mem
,
501 cdn_vector
, q
->pitch_gain
,
502 q
->pitch_lag
, q
->frame
.pfrac
);
504 // pitch prefilter update
505 for (i
= 0; i
< 4; i
++)
506 q
->pitch_gain
[i
] = 0.5 * FFMIN(q
->pitch_gain
[i
], 1.0);
508 v_pre_filtered
= do_pitchfilter(q
->pitch_pre_filter_mem
,
509 v_synthesis_filtered
,
510 q
->pitch_gain
, q
->pitch_lag
,
513 apply_gain_ctrl(cdn_vector
, v_synthesis_filtered
, v_pre_filtered
);
515 memcpy(q
->pitch_synthesis_filter_mem
,
516 cdn_vector
+ 17, 143 * sizeof(float));
517 memcpy(q
->pitch_pre_filter_mem
, cdn_vector
+ 17, 143 * sizeof(float));
518 memset(q
->pitch_gain
, 0, sizeof(q
->pitch_gain
));
519 memset(q
->pitch_lag
, 0, sizeof(q
->pitch_lag
));
524 * Reconstruct LPC coefficients from the line spectral pair frequencies
525 * and perform bandwidth expansion.
527 * @param lspf line spectral pair frequencies
528 * @param lpc linear predictive coding coefficients
530 * @note: bandwidth_expansion_coeff could be precalculated into a table
531 * but it seems to be slower on x86
533 * TIA/EIA/IS-733 2.4.3.3.5
535 static void lspf2lpc(const float *lspf
, float *lpc
)
538 double bandwidth_expansion_coeff
= QCELP_BANDWIDTH_EXPANSION_COEFF
;
541 for (i
= 0; i
< 10; i
++)
542 lsp
[i
] = cos(M_PI
* lspf
[i
]);
544 ff_acelp_lspd2lpc(lsp
, lpc
, 5);
546 for (i
= 0; i
< 10; i
++) {
547 lpc
[i
] *= bandwidth_expansion_coeff
;
548 bandwidth_expansion_coeff
*= QCELP_BANDWIDTH_EXPANSION_COEFF
;
553 * Interpolate LSP frequencies and compute LPC coefficients
554 * for a given bitrate & pitch subframe.
556 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
558 * @param q the context
559 * @param curr_lspf LSP frequencies vector of the current frame
560 * @param lpc float vector for the resulting LPC
561 * @param subframe_num frame number in decoded stream
563 static void interpolate_lpc(QCELPContext
*q
, const float *curr_lspf
,
564 float *lpc
, const int subframe_num
)
566 float interpolated_lspf
[10];
569 if (q
->bitrate
>= RATE_QUARTER
)
570 weight
= 0.25 * (subframe_num
+ 1);
571 else if (q
->bitrate
== RATE_OCTAVE
&& !subframe_num
)
577 ff_weighted_vector_sumf(interpolated_lspf
, curr_lspf
, q
->prev_lspf
,
578 weight
, 1.0 - weight
, 10);
579 lspf2lpc(interpolated_lspf
, lpc
);
580 } else if (q
->bitrate
>= RATE_QUARTER
||
581 (q
->bitrate
== I_F_Q
&& !subframe_num
))
582 lspf2lpc(curr_lspf
, lpc
);
583 else if (q
->bitrate
== SILENCE
&& !subframe_num
)
584 lspf2lpc(q
->prev_lspf
, lpc
);
587 static qcelp_packet_rate
buf_size2bitrate(const int buf_size
)
590 case 35: return RATE_FULL
;
591 case 17: return RATE_HALF
;
592 case 8: return RATE_QUARTER
;
593 case 4: return RATE_OCTAVE
;
594 case 1: return SILENCE
;
601 * Determine the bitrate from the frame size and/or the first byte of the frame.
603 * @param avctx the AV codec context
604 * @param buf_size length of the buffer
605 * @param buf the buffer
607 * @return the bitrate on success,
608 * I_F_Q if the bitrate cannot be satisfactorily determined
610 * TIA/EIA/IS-733 2.4.8.7.1
612 static qcelp_packet_rate
determine_bitrate(AVCodecContext
*avctx
,
616 qcelp_packet_rate bitrate
;
618 if ((bitrate
= buf_size2bitrate(buf_size
)) >= 0) {
619 if (bitrate
> **buf
) {
620 QCELPContext
*q
= avctx
->priv_data
;
621 if (!q
->warned_buf_mismatch_bitrate
) {
622 av_log(avctx
, AV_LOG_WARNING
,
623 "Claimed bitrate and buffer size mismatch.\n");
624 q
->warned_buf_mismatch_bitrate
= 1;
627 } else if (bitrate
< **buf
) {
628 av_log(avctx
, AV_LOG_ERROR
,
629 "Buffer is too small for the claimed bitrate.\n");
633 } else if ((bitrate
= buf_size2bitrate(buf_size
+ 1)) >= 0) {
634 av_log(avctx
, AV_LOG_WARNING
,
635 "Bitrate byte missing, guessing bitrate from packet size.\n");
639 if (bitrate
== SILENCE
) {
640 // FIXME: Remove this warning when tested with samples.
641 avpriv_request_sample(avctx
, "Blank frame handling");
646 static void warn_insufficient_frame_quality(AVCodecContext
*avctx
,
649 av_log(avctx
, AV_LOG_WARNING
, "Frame #%"PRId64
", IFQ: %s\n",
650 avctx
->frame_num
, message
);
653 static void postfilter(QCELPContext
*q
, float *samples
, float *lpc
)
655 static const float pow_0_775
[10] = {
656 0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
657 0.216676, 0.167924, 0.130141, 0.100859, 0.078166
659 0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
660 0.059605, 0.037253, 0.023283, 0.014552, 0.009095
662 float lpc_s
[10], lpc_p
[10], pole_out
[170], zero_out
[160];
665 for (n
= 0; n
< 10; n
++) {
666 lpc_s
[n
] = lpc
[n
] * pow_0_625
[n
];
667 lpc_p
[n
] = lpc
[n
] * pow_0_775
[n
];
670 ff_celp_lp_zero_synthesis_filterf(zero_out
, lpc_s
,
671 q
->formant_mem
+ 10, 160, 10);
672 memcpy(pole_out
, q
->postfilter_synth_mem
, sizeof(float) * 10);
673 ff_celp_lp_synthesis_filterf(pole_out
+ 10, lpc_p
, zero_out
, 160, 10);
674 memcpy(q
->postfilter_synth_mem
, pole_out
+ 160, sizeof(float) * 10);
676 ff_tilt_compensation(&q
->postfilter_tilt_mem
, 0.3, pole_out
+ 10, 160);
678 ff_adaptive_gain_control(samples
, pole_out
+ 10,
679 avpriv_scalarproduct_float_c(q
->formant_mem
+ 10,
682 160, 0.9375, &q
->postfilter_agc_mem
);
685 static int qcelp_decode_frame(AVCodecContext
*avctx
, AVFrame
*frame
,
686 int *got_frame_ptr
, AVPacket
*avpkt
)
688 const uint8_t *buf
= avpkt
->data
;
689 int buf_size
= avpkt
->size
;
690 QCELPContext
*q
= avctx
->priv_data
;
693 float quantized_lspf
[10], lpc
[10];
697 /* get output buffer */
698 frame
->nb_samples
= 160;
699 if ((ret
= ff_get_buffer(avctx
, frame
, 0)) < 0)
701 outbuffer
= (float *)frame
->data
[0];
703 if ((q
->bitrate
= determine_bitrate(avctx
, buf_size
, &buf
)) == I_F_Q
) {
704 warn_insufficient_frame_quality(avctx
, "Bitrate cannot be determined.");
708 if (q
->bitrate
== RATE_OCTAVE
&&
709 (q
->first16bits
= AV_RB16(buf
)) == 0xFFFF) {
710 warn_insufficient_frame_quality(avctx
, "Bitrate is 1/8 and first 16 bits are on.");
714 if (q
->bitrate
> SILENCE
) {
715 const QCELPBitmap
*bitmaps
= qcelp_unpacking_bitmaps_per_rate
[q
->bitrate
];
716 const QCELPBitmap
*bitmaps_end
= qcelp_unpacking_bitmaps_per_rate
[q
->bitrate
] +
717 qcelp_unpacking_bitmaps_lengths
[q
->bitrate
];
718 uint8_t *unpacked_data
= (uint8_t *)&q
->frame
;
720 if ((ret
= init_get_bits8(&q
->gb
, buf
, buf_size
)) < 0)
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
,
774 outbuffer
+ i
* 40, 40, 10);
778 // postfilter, as per TIA/EIA/IS-733 2.4.8.6
779 postfilter(q
, outbuffer
, lpc
);
781 memcpy(q
->formant_mem
, q
->formant_mem
+ 160, 10 * sizeof(float));
783 memcpy(q
->prev_lspf
, quantized_lspf
, sizeof(q
->prev_lspf
));
784 q
->prev_bitrate
= q
->bitrate
;
791 const FFCodec ff_qcelp_decoder
= {
793 CODEC_LONG_NAME("QCELP / PureVoice"),
794 .p
.type
= AVMEDIA_TYPE_AUDIO
,
795 .p
.id
= AV_CODEC_ID_QCELP
,
796 .init
= qcelp_decode_init
,
797 FF_CODEC_DECODE_CB(qcelp_decode_frame
),
798 .p
.capabilities
= AV_CODEC_CAP_DR1
| AV_CODEC_CAP_CHANNEL_CONF
,
799 .priv_data_size
= sizeof(QCELPContext
),