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/*
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 * QCELP decoder
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 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
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 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with Libav; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
21

    
22
/**
23
 * @file
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 * QCELP decoder
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 * @author Reynaldo H. Verdejo Pinochet
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 * @remark Libav merging spearheaded by Kenan Gillet
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 * @remark Development mentored by Benjamin Larson
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 */
29

    
30
#include <stddef.h>
31

    
32
#include "avcodec.h"
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#include "internal.h"
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#include "get_bits.h"
35

    
36
#include "qcelpdata.h"
37

    
38
#include "celp_math.h"
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#include "celp_filters.h"
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#include "acelp_filters.h"
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#include "acelp_vectors.h"
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#include "lsp.h"
43

    
44
#undef NDEBUG
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#include <assert.h>
46

    
47
typedef enum
48
{
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    I_F_Q = -1,    /*!< insufficient frame quality */
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    SILENCE,
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    RATE_OCTAVE,
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    RATE_QUARTER,
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    RATE_HALF,
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    RATE_FULL
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} qcelp_packet_rate;
56

    
57
typedef struct
58
{
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    GetBitContext     gb;
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    qcelp_packet_rate bitrate;
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    QCELPFrame        frame;    /*!< unpacked data frame */
62

    
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    uint8_t  erasure_count;
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    uint8_t  octave_count;      /*!< count the consecutive RATE_OCTAVE frames */
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    float    prev_lspf[10];
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    float    predictor_lspf[10];/*!< LSP predictor for RATE_OCTAVE and I_F_Q */
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    float    pitch_synthesis_filter_mem[303];
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    float    pitch_pre_filter_mem[303];
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    float    rnd_fir_filter_mem[180];
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    float    formant_mem[170];
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    float    last_codebook_gain;
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    int      prev_g1[2];
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    int      prev_bitrate;
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    float    pitch_gain[4];
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    uint8_t  pitch_lag[4];
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    uint16_t first16bits;
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    uint8_t  warned_buf_mismatch_bitrate;
78

    
79
    /* postfilter */
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    float    postfilter_synth_mem[10];
81
    float    postfilter_agc_mem;
82
    float    postfilter_tilt_mem;
83
} QCELPContext;
84

    
85
/**
86
 * Initialize the speech codec according to the specification.
87
 *
88
 * TIA/EIA/IS-733 2.4.9
89
 */
90
static av_cold int qcelp_decode_init(AVCodecContext *avctx)
91
{
92
    QCELPContext *q = avctx->priv_data;
93
    int i;
94

    
95
    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
96

    
97
    for(i=0; i<10; i++)
98
        q->prev_lspf[i] = (i+1)/11.;
99

    
100
    return 0;
101
}
102

    
103
/**
104
 * Decode the 10 quantized LSP frequencies from the LSPV/LSP
105
 * transmission codes of any bitrate and check for badly received packets.
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 *
107
 * @param q the context
108
 * @param lspf line spectral pair frequencies
109
 *
110
 * @return 0 on success, -1 if the packet is badly received
111
 *
112
 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
113
 */
114
static int decode_lspf(QCELPContext *q, float *lspf)
115
{
116
    int i;
117
    float tmp_lspf, smooth, erasure_coeff;
118
    const float *predictors;
119

    
120
    if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
121
    {
122
        predictors = (q->prev_bitrate != RATE_OCTAVE &&
123
                       q->prev_bitrate != I_F_Q ?
124
                       q->prev_lspf : q->predictor_lspf);
125

    
126
        if(q->bitrate == RATE_OCTAVE)
127
        {
128
            q->octave_count++;
129

    
130
            for(i=0; i<10; i++)
131
            {
132
                q->predictor_lspf[i] =
133
                             lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
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                                                         : -QCELP_LSP_SPREAD_FACTOR)
135
                                     + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
136
                                     + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
137
            }
138
            smooth = (q->octave_count < 10 ? .875 : 0.1);
139
        }else
140
        {
141
            erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
142

    
143
            assert(q->bitrate == I_F_Q);
144

    
145
            if(q->erasure_count > 1)
146
                erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
147

    
148
            for(i=0; i<10; i++)
149
            {
150
                q->predictor_lspf[i] =
151
                             lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
152
                                     + erasure_coeff * predictors[i];
153
            }
154
            smooth = 0.125;
155
        }
156

    
157
        // Check the stability of the LSP frequencies.
158
        lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
159
        for(i=1; i<10; i++)
160
            lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
161

    
162
        lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
163
        for(i=9; i>0; i--)
164
            lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
165

    
166
        // Low-pass filter the LSP frequencies.
167
        ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
168
    }else
169
    {
170
        q->octave_count = 0;
171

    
172
        tmp_lspf = 0.;
173
        for(i=0; i<5 ; i++)
174
        {
175
            lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
176
            lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
177
        }
178

    
179
        // Check for badly received packets.
180
        if(q->bitrate == RATE_QUARTER)
181
        {
182
            if(lspf[9] <= .70 || lspf[9] >=  .97)
183
                return -1;
184
            for(i=3; i<10; i++)
185
                if(fabs(lspf[i] - lspf[i-2]) < .08)
186
                    return -1;
187
        }else
188
        {
189
            if(lspf[9] <= .66 || lspf[9] >= .985)
190
                return -1;
191
            for(i=4; i<10; i++)
192
                if (fabs(lspf[i] - lspf[i-4]) < .0931)
193
                    return -1;
194
        }
195
    }
196
    return 0;
197
}
198

    
199
/**
200
 * Convert codebook transmission codes to GAIN and INDEX.
201
 *
202
 * @param q the context
203
 * @param gain array holding the decoded gain
204
 *
205
 * TIA/EIA/IS-733 2.4.6.2
206
 */
207
static void decode_gain_and_index(QCELPContext  *q,
208
                                  float *gain) {
209
    int   i, subframes_count, g1[16];
210
    float slope;
211

    
212
    if(q->bitrate >= RATE_QUARTER)
213
    {
214
        switch(q->bitrate)
215
        {
216
            case RATE_FULL: subframes_count = 16; break;
217
            case RATE_HALF: subframes_count = 4;  break;
218
            default:        subframes_count = 5;
219
        }
220
        for(i=0; i<subframes_count; i++)
221
        {
222
            g1[i] = 4 * q->frame.cbgain[i];
223
            if(q->bitrate == RATE_FULL && !((i+1) & 3))
224
            {
225
                g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
226
            }
227

    
228
            gain[i] = qcelp_g12ga[g1[i]];
229

    
230
            if(q->frame.cbsign[i])
231
            {
232
                gain[i] = -gain[i];
233
                q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
234
            }
235
        }
236

    
237
        q->prev_g1[0] = g1[i-2];
238
        q->prev_g1[1] = g1[i-1];
239
        q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
240

    
241
        if(q->bitrate == RATE_QUARTER)
242
        {
243
            // Provide smoothing of the unvoiced excitation energy.
244
            gain[7] =     gain[4];
245
            gain[6] = 0.4*gain[3] + 0.6*gain[4];
246
            gain[5] =     gain[3];
247
            gain[4] = 0.8*gain[2] + 0.2*gain[3];
248
            gain[3] = 0.2*gain[1] + 0.8*gain[2];
249
            gain[2] =     gain[1];
250
            gain[1] = 0.6*gain[0] + 0.4*gain[1];
251
        }
252
    }else if (q->bitrate != SILENCE)
253
    {
254
        if(q->bitrate == RATE_OCTAVE)
255
        {
256
            g1[0] = 2 * q->frame.cbgain[0]
257
                  + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
258
            subframes_count = 8;
259
        }else
260
        {
261
            assert(q->bitrate == I_F_Q);
262

    
263
            g1[0] = q->prev_g1[1];
264
            switch(q->erasure_count)
265
            {
266
                case 1 : break;
267
                case 2 : g1[0] -= 1; break;
268
                case 3 : g1[0] -= 2; break;
269
                default: g1[0] -= 6;
270
            }
271
            if(g1[0] < 0)
272
                g1[0] = 0;
273
            subframes_count = 4;
274
        }
275
        // This interpolation is done to produce smoother background noise.
276
        slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
277
        for(i=1; i<=subframes_count; i++)
278
            gain[i-1] = q->last_codebook_gain + slope * i;
279

    
280
        q->last_codebook_gain = gain[i-2];
281
        q->prev_g1[0] = q->prev_g1[1];
282
        q->prev_g1[1] = g1[0];
283
    }
284
}
285

    
286
/**
287
 * If the received packet is Rate 1/4 a further sanity check is made of the
288
 * codebook gain.
289
 *
290
 * @param cbgain the unpacked cbgain array
291
 * @return -1 if the sanity check fails, 0 otherwise
292
 *
293
 * TIA/EIA/IS-733 2.4.8.7.3
294
 */
295
static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
296
{
297
    int i, diff, prev_diff=0;
298

    
299
    for(i=1; i<5; i++)
300
    {
301
        diff = cbgain[i] - cbgain[i-1];
302
        if(FFABS(diff) > 10)
303
            return -1;
304
        else if(FFABS(diff - prev_diff) > 12)
305
            return -1;
306
        prev_diff = diff;
307
    }
308
    return 0;
309
}
310

    
311
/**
312
 * Compute the scaled codebook vector Cdn From INDEX and GAIN
313
 * for all rates.
314
 *
315
 * The specification lacks some information here.
316
 *
317
 * TIA/EIA/IS-733 has an omission on the codebook index determination
318
 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
319
 * you have to subtract the decoded index parameter from the given scaled
320
 * codebook vector index 'n' to get the desired circular codebook index, but
321
 * it does not mention that you have to clamp 'n' to [0-9] in order to get
322
 * RI-compliant results.
323
 *
324
 * The reason for this mistake seems to be the fact they forgot to mention you
325
 * have to do these calculations per codebook subframe and adjust given
326
 * equation values accordingly.
327
 *
328
 * @param q the context
329
 * @param gain array holding the 4 pitch subframe gain values
330
 * @param cdn_vector array for the generated scaled codebook vector
331
 */
332
static void compute_svector(QCELPContext *q, const float *gain,
333
                            float *cdn_vector)
334
{
335
    int      i, j, k;
336
    uint16_t cbseed, cindex;
337
    float    *rnd, tmp_gain, fir_filter_value;
338

    
339
    switch(q->bitrate)
340
    {
341
        case RATE_FULL:
342
            for(i=0; i<16; i++)
343
            {
344
                tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
345
                cindex = -q->frame.cindex[i];
346
                for(j=0; j<10; j++)
347
                    *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
348
            }
349
        break;
350
        case RATE_HALF:
351
            for(i=0; i<4; i++)
352
            {
353
                tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
354
                cindex = -q->frame.cindex[i];
355
                for (j = 0; j < 40; j++)
356
                *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
357
            }
358
        break;
359
        case RATE_QUARTER:
360
            cbseed = (0x0003 & q->frame.lspv[4])<<14 |
361
                     (0x003F & q->frame.lspv[3])<< 8 |
362
                     (0x0060 & q->frame.lspv[2])<< 1 |
363
                     (0x0007 & q->frame.lspv[1])<< 3 |
364
                     (0x0038 & q->frame.lspv[0])>> 3 ;
365
            rnd = q->rnd_fir_filter_mem + 20;
366
            for(i=0; i<8; i++)
367
            {
368
                tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
369
                for(k=0; k<20; k++)
370
                {
371
                    cbseed = 521 * cbseed + 259;
372
                    *rnd = (int16_t)cbseed;
373

    
374
                    // FIR filter
375
                    fir_filter_value = 0.0;
376
                    for(j=0; j<10; j++)
377
                        fir_filter_value += qcelp_rnd_fir_coefs[j ]
378
                                          * (rnd[-j ] + rnd[-20+j]);
379

    
380
                    fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
381
                    *cdn_vector++ = tmp_gain * fir_filter_value;
382
                    rnd++;
383
                }
384
            }
385
            memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
386
        break;
387
        case RATE_OCTAVE:
388
            cbseed = q->first16bits;
389
            for(i=0; i<8; i++)
390
            {
391
                tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
392
                for(j=0; j<20; j++)
393
                {
394
                    cbseed = 521 * cbseed + 259;
395
                    *cdn_vector++ = tmp_gain * (int16_t)cbseed;
396
                }
397
            }
398
        break;
399
        case I_F_Q:
400
            cbseed = -44; // random codebook index
401
            for(i=0; i<4; i++)
402
            {
403
                tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
404
                for(j=0; j<40; j++)
405
                    *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
406
            }
407
        break;
408
        case SILENCE:
409
            memset(cdn_vector, 0, 160 * sizeof(float));
410
        break;
411
    }
412
}
413

    
414
/**
415
 * Apply generic gain control.
416
 *
417
 * @param v_out output vector
418
 * @param v_in gain-controlled vector
419
 * @param v_ref vector to control gain of
420
 *
421
 * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
422
 */
423
static void apply_gain_ctrl(float *v_out, const float *v_ref,
424
                            const float *v_in)
425
{
426
    int i;
427

    
428
    for (i = 0; i < 160; i += 40)
429
        ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
430
                                                ff_dot_productf(v_ref + i,
431
                                                                v_ref + i, 40),
432
                                                40);
433
}
434

    
435
/**
436
 * Apply filter in pitch-subframe steps.
437
 *
438
 * @param memory buffer for the previous state of the filter
439
 *        - must be able to contain 303 elements
440
 *        - the 143 first elements are from the previous state
441
 *        - the next 160 are for output
442
 * @param v_in input filter vector
443
 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
444
 * @param lag per-subframe lag array, each element is
445
 *        - between 16 and 143 if its corresponding pfrac is 0,
446
 *        - between 16 and 139 otherwise
447
 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
448
 *        otherwise
449
 *
450
 * @return filter output vector
451
 */
452
static const float *do_pitchfilter(float memory[303], const float v_in[160],
453
                                   const float gain[4], const uint8_t *lag,
454
                                   const uint8_t pfrac[4])
455
{
456
    int         i, j;
457
    float       *v_lag, *v_out;
458
    const float *v_len;
459

    
460
    v_out = memory + 143; // Output vector starts at memory[143].
461

    
462
    for(i=0; i<4; i++)
463
    {
464
        if(gain[i])
465
        {
466
            v_lag = memory + 143 + 40 * i - lag[i];
467
            for(v_len=v_in+40; v_in<v_len; v_in++)
468
            {
469
                if(pfrac[i]) // If it is a fractional lag...
470
                {
471
                    for(j=0, *v_out=0.; j<4; j++)
472
                        *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
473
                }else
474
                    *v_out = *v_lag;
475

    
476
                *v_out = *v_in + gain[i] * *v_out;
477

    
478
                v_lag++;
479
                v_out++;
480
            }
481
        }else
482
        {
483
            memcpy(v_out, v_in, 40 * sizeof(float));
484
            v_in  += 40;
485
            v_out += 40;
486
        }
487
    }
488

    
489
    memmove(memory, memory + 160, 143 * sizeof(float));
490
    return memory + 143;
491
}
492

    
493
/**
494
 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
495
 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
496
 *
497
 * @param q the context
498
 * @param cdn_vector the scaled codebook vector
499
 */
500
static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
501
{
502
    int         i;
503
    const float *v_synthesis_filtered, *v_pre_filtered;
504

    
505
    if(q->bitrate >= RATE_HALF ||
506
       q->bitrate == SILENCE ||
507
       (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
508
    {
509

    
510
        if(q->bitrate >= RATE_HALF)
511
        {
512

    
513
            // Compute gain & lag for the whole frame.
514
            for(i=0; i<4; i++)
515
            {
516
                q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
517

    
518
                q->pitch_lag[i] = q->frame.plag[i] + 16;
519
            }
520
        }else
521
        {
522
            float max_pitch_gain;
523

    
524
            if (q->bitrate == I_F_Q)
525
            {
526
                  if (q->erasure_count < 3)
527
                      max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
528
                  else
529
                      max_pitch_gain = 0.0;
530
            }else
531
            {
532
                assert(q->bitrate == SILENCE);
533
                max_pitch_gain = 1.0;
534
            }
535
            for(i=0; i<4; i++)
536
                q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
537

    
538
            memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
539
        }
540

    
541
        // pitch synthesis filter
542
        v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
543
                                              cdn_vector, q->pitch_gain,
544
                                              q->pitch_lag, q->frame.pfrac);
545

    
546
        // pitch prefilter update
547
        for(i=0; i<4; i++)
548
            q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
549

    
550
        v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
551
                                        v_synthesis_filtered,
552
                                        q->pitch_gain, q->pitch_lag,
553
                                        q->frame.pfrac);
554

    
555
        apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
556
    }else
557
    {
558
        memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
559
               143 * sizeof(float));
560
        memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
561
        memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
562
        memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
563
    }
564
}
565

    
566
/**
567
 * Reconstruct LPC coefficients from the line spectral pair frequencies
568
 * and perform bandwidth expansion.
569
 *
570
 * @param lspf line spectral pair frequencies
571
 * @param lpc linear predictive coding coefficients
572
 *
573
 * @note: bandwidth_expansion_coeff could be precalculated into a table
574
 *        but it seems to be slower on x86
575
 *
576
 * TIA/EIA/IS-733 2.4.3.3.5
577
 */
578
static void lspf2lpc(const float *lspf, float *lpc)
579
{
580
    double lsp[10];
581
    double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
582
    int   i;
583

    
584
    for (i=0; i<10; i++)
585
        lsp[i] = cos(M_PI * lspf[i]);
586

    
587
    ff_acelp_lspd2lpc(lsp, lpc, 5);
588

    
589
    for (i=0; i<10; i++)
590
    {
591
        lpc[i] *= bandwidth_expansion_coeff;
592
        bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
593
    }
594
}
595

    
596
/**
597
 * Interpolate LSP frequencies and compute LPC coefficients
598
 * for a given bitrate & pitch subframe.
599
 *
600
 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
601
 *
602
 * @param q the context
603
 * @param curr_lspf LSP frequencies vector of the current frame
604
 * @param lpc float vector for the resulting LPC
605
 * @param subframe_num frame number in decoded stream
606
 */
607
static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
608
                            float *lpc, const int subframe_num)
609
{
610
    float interpolated_lspf[10];
611
    float weight;
612

    
613
    if(q->bitrate >= RATE_QUARTER)
614
        weight = 0.25 * (subframe_num + 1);
615
    else if(q->bitrate == RATE_OCTAVE && !subframe_num)
616
        weight = 0.625;
617
    else
618
        weight = 1.0;
619

    
620
    if(weight != 1.0)
621
    {
622
        ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
623
                                weight, 1.0 - weight, 10);
624
        lspf2lpc(interpolated_lspf, lpc);
625
    }else if(q->bitrate >= RATE_QUARTER ||
626
             (q->bitrate == I_F_Q && !subframe_num))
627
        lspf2lpc(curr_lspf, lpc);
628
    else if(q->bitrate == SILENCE && !subframe_num)
629
        lspf2lpc(q->prev_lspf, lpc);
630
}
631

    
632
static qcelp_packet_rate buf_size2bitrate(const int buf_size)
633
{
634
    switch(buf_size)
635
    {
636
        case 35: return RATE_FULL;
637
        case 17: return RATE_HALF;
638
        case  8: return RATE_QUARTER;
639
        case  4: return RATE_OCTAVE;
640
        case  1: return SILENCE;
641
    }
642

    
643
    return I_F_Q;
644
}
645

    
646
/**
647
 * Determine the bitrate from the frame size and/or the first byte of the frame.
648
 *
649
 * @param avctx the AV codec context
650
 * @param buf_size length of the buffer
651
 * @param buf the bufffer
652
 *
653
 * @return the bitrate on success,
654
 *         I_F_Q  if the bitrate cannot be satisfactorily determined
655
 *
656
 * TIA/EIA/IS-733 2.4.8.7.1
657
 */
658
static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size,
659
                             const uint8_t **buf)
660
{
661
    qcelp_packet_rate bitrate;
662

    
663
    if((bitrate = buf_size2bitrate(buf_size)) >= 0)
664
    {
665
        if(bitrate > **buf)
666
        {
667
            QCELPContext *q = avctx->priv_data;
668
            if (!q->warned_buf_mismatch_bitrate)
669
            {
670
            av_log(avctx, AV_LOG_WARNING,
671
                   "Claimed bitrate and buffer size mismatch.\n");
672
                q->warned_buf_mismatch_bitrate = 1;
673
            }
674
            bitrate = **buf;
675
        }else if(bitrate < **buf)
676
        {
677
            av_log(avctx, AV_LOG_ERROR,
678
                   "Buffer is too small for the claimed bitrate.\n");
679
            return I_F_Q;
680
        }
681
        (*buf)++;
682
    }else if((bitrate = buf_size2bitrate(buf_size + 1)) >= 0)
683
    {
684
        av_log(avctx, AV_LOG_WARNING,
685
               "Bitrate byte is missing, guessing the bitrate from packet size.\n");
686
    }else
687
        return I_F_Q;
688

    
689
    if(bitrate == SILENCE)
690
    {
691
        //FIXME: Remove experimental warning when tested with samples.
692
        av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
693
    }
694
    return bitrate;
695
}
696

    
697
static void warn_insufficient_frame_quality(AVCodecContext *avctx,
698
                                            const char *message)
699
{
700
    av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
701
           message);
702
}
703

    
704
static void postfilter(QCELPContext *q, float *samples, float *lpc)
705
{
706
    static const float pow_0_775[10] = {
707
        0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
708
        0.216676, 0.167924, 0.130141, 0.100859, 0.078166
709
    }, pow_0_625[10] = {
710
        0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
711
        0.059605, 0.037253, 0.023283, 0.014552, 0.009095
712
    };
713
    float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
714
    int n;
715

    
716
    for (n = 0; n < 10; n++) {
717
        lpc_s[n] = lpc[n] * pow_0_625[n];
718
        lpc_p[n] = lpc[n] * pow_0_775[n];
719
    }
720

    
721
    ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
722
                                      q->formant_mem + 10, 160, 10);
723
    memcpy(pole_out, q->postfilter_synth_mem,       sizeof(float) * 10);
724
    ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
725
    memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
726

    
727
    ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
728

    
729
    ff_adaptive_gain_control(samples, pole_out + 10,
730
        ff_dot_productf(q->formant_mem + 10, q->formant_mem + 10, 160),
731
        160, 0.9375, &q->postfilter_agc_mem);
732
}
733

    
734
static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
735
                              AVPacket *avpkt)
736
{
737
    const uint8_t *buf = avpkt->data;
738
    int buf_size = avpkt->size;
739
    QCELPContext *q = avctx->priv_data;
740
    float *outbuffer = data;
741
    int   i;
742
    float quantized_lspf[10], lpc[10];
743
    float gain[16];
744
    float *formant_mem;
745

    
746
    if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
747
    {
748
        warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
749
        goto erasure;
750
    }
751

    
752
    if(q->bitrate == RATE_OCTAVE &&
753
       (q->first16bits = AV_RB16(buf)) == 0xFFFF)
754
    {
755
        warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
756
        goto erasure;
757
    }
758

    
759
    if(q->bitrate > SILENCE)
760
    {
761
        const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
762
        const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
763
                                       + qcelp_unpacking_bitmaps_lengths[q->bitrate];
764
        uint8_t           *unpacked_data = (uint8_t *)&q->frame;
765

    
766
        init_get_bits(&q->gb, buf, 8*buf_size);
767

    
768
        memset(&q->frame, 0, sizeof(QCELPFrame));
769

    
770
        for(; bitmaps < bitmaps_end; bitmaps++)
771
            unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
772

    
773
        // Check for erasures/blanks on rates 1, 1/4 and 1/8.
774
        if(q->frame.reserved)
775
        {
776
            warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
777
            goto erasure;
778
        }
779
        if(q->bitrate == RATE_QUARTER &&
780
           codebook_sanity_check_for_rate_quarter(q->frame.cbgain))
781
        {
782
            warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
783
            goto erasure;
784
        }
785

    
786
        if(q->bitrate >= RATE_HALF)
787
        {
788
            for(i=0; i<4; i++)
789
            {
790
                if(q->frame.pfrac[i] && q->frame.plag[i] >= 124)
791
                {
792
                    warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
793
                    goto erasure;
794
                }
795
            }
796
        }
797
    }
798

    
799
    decode_gain_and_index(q, gain);
800
    compute_svector(q, gain, outbuffer);
801

    
802
    if(decode_lspf(q, quantized_lspf) < 0)
803
    {
804
        warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
805
        goto erasure;
806
    }
807

    
808

    
809
    apply_pitch_filters(q, outbuffer);
810

    
811
    if(q->bitrate == I_F_Q)
812
    {
813
erasure:
814
        q->bitrate = I_F_Q;
815
        q->erasure_count++;
816
        decode_gain_and_index(q, gain);
817
        compute_svector(q, gain, outbuffer);
818
        decode_lspf(q, quantized_lspf);
819
        apply_pitch_filters(q, outbuffer);
820
    }else
821
        q->erasure_count = 0;
822

    
823
    formant_mem = q->formant_mem + 10;
824
    for(i=0; i<4; i++)
825
    {
826
        interpolate_lpc(q, quantized_lspf, lpc, i);
827
        ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
828
                                     10);
829
        formant_mem += 40;
830
    }
831

    
832
    // postfilter, as per TIA/EIA/IS-733 2.4.8.6
833
    postfilter(q, outbuffer, lpc);
834

    
835
    memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
836

    
837
    memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
838
    q->prev_bitrate = q->bitrate;
839

    
840
    *data_size = 160 * sizeof(*outbuffer);
841

    
842
    return *data_size;
843
}
844

    
845
AVCodec ff_qcelp_decoder =
846
{
847
    .name   = "qcelp",
848
    .type   = AVMEDIA_TYPE_AUDIO,
849
    .id     = CODEC_ID_QCELP,
850
    .init   = qcelp_decode_init,
851
    .decode = qcelp_decode_frame,
852
    .priv_data_size = sizeof(QCELPContext),
853
    .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
854
};