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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 FFmpeg.
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 *
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 * FFmpeg 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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 * FFmpeg 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 FFmpeg; 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
/**
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 * @file qcelpdec.c
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 * QCELP decoder
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 * @author Reynaldo H. Verdejo Pinochet
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 * @remark FFmpeg merging spearheaded by Kenan Gillet
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 * @remark Development mentored by Benjamin Larson
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 */
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#include <stddef.h>
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#include "avcodec.h"
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#include "internal.h"
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#include "bitstream.h"
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#include "qcelpdata.h"
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#include "celp_math.h"
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#include "celp_filters.h"
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#undef NDEBUG
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#include <assert.h>
43

    
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typedef enum
45
{
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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;
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typedef struct
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{
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    GetBitContext     gb;
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    qcelp_packet_rate bitrate;
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    QCELPFrame        frame;    /*!< unpacked data frame */
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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;
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} QCELPContext;
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/**
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 * Reconstructs LPC coefficients from the line spectral pair frequencies.
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 *
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 * TIA/EIA/IS-733 2.4.3.3.5
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 */
82
void ff_qcelp_lspf2lpc(const float *lspf, float *lpc);
83

    
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static void weighted_vector_sumf(float *out, const float *in_a,
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                                 const float *in_b, float weight_coeff_a,
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                                 float weight_coeff_b, int length)
87
{
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    int i;
89

    
90
    for(i=0; i<length; i++)
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        out[i] = weight_coeff_a * in_a[i]
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               + weight_coeff_b * in_b[i];
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}
94

    
95
/**
96
 * Initialize the speech codec according to the specification.
97
 *
98
 * TIA/EIA/IS-733 2.4.9
99
 */
100
static av_cold int qcelp_decode_init(AVCodecContext *avctx)
101
{
102
    QCELPContext *q = avctx->priv_data;
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    int i;
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    avctx->sample_fmt = SAMPLE_FMT_FLT;
106

    
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    for(i=0; i<10; i++)
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        q->prev_lspf[i] = (i+1)/11.;
109

    
110
    return 0;
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}
112

    
113
/**
114
 * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
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 * transmission codes of any bitrate and checks for badly received packets.
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 *
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 * @param q the context
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 * @param lspf line spectral pair frequencies
119
 *
120
 * @return 0 on success, -1 if the packet is badly received
121
 *
122
 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
123
 */
124
static int decode_lspf(QCELPContext *q, float *lspf)
125
{
126
    int i;
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    float tmp_lspf, smooth, erasure_coeff;
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    const float *predictors;
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    if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
131
    {
132
        predictors = (q->prev_bitrate != RATE_OCTAVE &&
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                       q->prev_bitrate != I_F_Q ?
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                       q->prev_lspf : q->predictor_lspf);
135

    
136
        if(q->bitrate == RATE_OCTAVE)
137
        {
138
            q->octave_count++;
139

    
140
            for(i=0; i<10; i++)
141
            {
142
                q->predictor_lspf[i] =
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                             lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
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                                                         : -QCELP_LSP_SPREAD_FACTOR)
145
                                     + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
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                                     + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
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            }
148
            smooth = (q->octave_count < 10 ? .875 : 0.1);
149
        }else
150
        {
151
            erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
152

    
153
            assert(q->bitrate == I_F_Q);
154

    
155
            if(q->erasure_count > 1)
156
                erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
157

    
158
            for(i=0; i<10; i++)
159
            {
160
                q->predictor_lspf[i] =
161
                             lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
162
                                     + erasure_coeff * predictors[i];
163
            }
164
            smooth = 0.125;
165
        }
166

    
167
        // Check the stability of the LSP frequencies.
168
        lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
169
        for(i=1; i<10; i++)
170
            lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
171

    
172
        lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
173
        for(i=9; i>0; i--)
174
            lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
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176
        // Low-pass filter the LSP frequencies.
177
        weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
178
    }else
179
    {
180
        q->octave_count = 0;
181

    
182
        tmp_lspf = 0.;
183
        for(i=0; i<5 ; i++)
184
        {
185
            lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
186
            lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
187
        }
188

    
189
        // Check for badly received packets.
190
        if(q->bitrate == RATE_QUARTER)
191
        {
192
            if(lspf[9] <= .70 || lspf[9] >=  .97)
193
                return -1;
194
            for(i=3; i<10; i++)
195
                if(fabs(lspf[i] - lspf[i-2]) < .08)
196
                    return -1;
197
        }else
198
        {
199
            if(lspf[9] <= .66 || lspf[9] >= .985)
200
                return -1;
201
            for(i=4; i<10; i++)
202
                if (fabs(lspf[i] - lspf[i-4]) < .0931)
203
                    return -1;
204
        }
205
    }
206
    return 0;
207
}
208

    
209
/**
210
 * Converts codebook transmission codes to GAIN and INDEX.
211
 *
212
 * @param q the context
213
 * @param gain array holding the decoded gain
214
 *
215
 * TIA/EIA/IS-733 2.4.6.2
216
 */
217
static void decode_gain_and_index(QCELPContext  *q,
218
                                  float *gain) {
219
    int   i, subframes_count, g1[16];
220
    float slope;
221

    
222
    if(q->bitrate >= RATE_QUARTER)
223
    {
224
        switch(q->bitrate)
225
        {
226
            case RATE_FULL: subframes_count = 16; break;
227
            case RATE_HALF: subframes_count = 4;  break;
228
            default:        subframes_count = 5;
229
        }
230
        for(i=0; i<subframes_count; i++)
231
        {
232
            g1[i] = 4 * q->frame.cbgain[i];
233
            if(q->bitrate == RATE_FULL && !((i+1) & 3))
234
            {
235
                g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
236
            }
237

    
238
            gain[i] = qcelp_g12ga[g1[i]];
239

    
240
            if(q->frame.cbsign[i])
241
            {
242
                gain[i] = -gain[i];
243
                q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
244
            }
245
        }
246

    
247
        q->prev_g1[0] = g1[i-2];
248
        q->prev_g1[1] = g1[i-1];
249
        q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
250

    
251
        if(q->bitrate == RATE_QUARTER)
252
        {
253
            // Provide smoothing of the unvoiced excitation energy.
254
            gain[7] =     gain[4];
255
            gain[6] = 0.4*gain[3] + 0.6*gain[4];
256
            gain[5] =     gain[3];
257
            gain[4] = 0.8*gain[2] + 0.2*gain[3];
258
            gain[3] = 0.2*gain[1] + 0.8*gain[2];
259
            gain[2] =     gain[1];
260
            gain[1] = 0.6*gain[0] + 0.4*gain[1];
261
        }
262
    }else
263
    {
264
        if(q->bitrate == RATE_OCTAVE)
265
        {
266
            g1[0] = 2 * q->frame.cbgain[0]
267
                  + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
268
            subframes_count = 8;
269
        }else
270
        {
271
            assert(q->bitrate == I_F_Q);
272

    
273
            g1[0] = q->prev_g1[1];
274
            switch(q->erasure_count)
275
            {
276
                case 1 : break;
277
                case 2 : g1[0] -= 1; break;
278
                case 3 : g1[0] -= 2; break;
279
                default: g1[0] -= 6;
280
            }
281
            if(g1[0] < 0)
282
                g1[0] = 0;
283
            subframes_count = 4;
284
        }
285
        // This interpolation is done to produce smoother background noise.
286
        slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
287
        for(i=1; i<=subframes_count; i++)
288
            gain[i-1] = q->last_codebook_gain + slope * i;
289

    
290
        q->last_codebook_gain = gain[i-2];
291
        q->prev_g1[0] = q->prev_g1[1];
292
        q->prev_g1[1] = g1[0];
293
    }
294
}
295

    
296
/**
297
 * If the received packet is Rate 1/4 a further sanity check is made of the
298
 * codebook gain.
299
 *
300
 * @param cbgain the unpacked cbgain array
301
 * @return -1 if the sanity check fails, 0 otherwise
302
 *
303
 * TIA/EIA/IS-733 2.4.8.7.3
304
 */
305
static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
306
{
307
    int i, diff, prev_diff=0;
308

    
309
    for(i=1; i<5; i++)
310
    {
311
        diff = cbgain[i] - cbgain[i-1];
312
        if(FFABS(diff) > 10)
313
            return -1;
314
        else if(FFABS(diff - prev_diff) > 12)
315
            return -1;
316
        prev_diff = diff;
317
    }
318
    return 0;
319
}
320

    
321
/**
322
 * Computes the scaled codebook vector Cdn From INDEX and GAIN
323
 * for all rates.
324
 *
325
 * The specification lacks some information here.
326
 *
327
 * TIA/EIA/IS-733 has an omission on the codebook index determination
328
 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
329
 * you have to subtract the decoded index parameter from the given scaled
330
 * codebook vector index 'n' to get the desired circular codebook index, but
331
 * it does not mention that you have to clamp 'n' to [0-9] in order to get
332
 * RI-compliant results.
333
 *
334
 * The reason for this mistake seems to be the fact they forgot to mention you
335
 * have to do these calculations per codebook subframe and adjust given
336
 * equation values accordingly.
337
 *
338
 * @param q the context
339
 * @param gain array holding the 4 pitch subframe gain values
340
 * @param cdn_vector array for the generated scaled codebook vector
341
 */
342
static void compute_svector(QCELPContext *q, const float *gain,
343
                            float *cdn_vector)
344
{
345
    int      i, j, k;
346
    uint16_t cbseed, cindex;
347
    float    *rnd, tmp_gain, fir_filter_value;
348

    
349
    switch(q->bitrate)
350
    {
351
        case RATE_FULL:
352
            for(i=0; i<16; i++)
353
            {
354
                tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
355
                cindex = -q->frame.cindex[i];
356
                for(j=0; j<10; j++)
357
                    *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
358
            }
359
        break;
360
        case RATE_HALF:
361
            for(i=0; i<4; i++)
362
            {
363
                tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
364
                cindex = -q->frame.cindex[i];
365
                for (j = 0; j < 40; j++)
366
                *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
367
            }
368
        break;
369
        case RATE_QUARTER:
370
            cbseed = (0x0003 & q->frame.lspv[4])<<14 |
371
                     (0x003F & q->frame.lspv[3])<< 8 |
372
                     (0x0060 & q->frame.lspv[2])<< 1 |
373
                     (0x0007 & q->frame.lspv[1])<< 3 |
374
                     (0x0038 & q->frame.lspv[0])>> 3 ;
375
            rnd = q->rnd_fir_filter_mem + 20;
376
            for(i=0; i<8; i++)
377
            {
378
                tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
379
                for(k=0; k<20; k++)
380
                {
381
                    cbseed = 521 * cbseed + 259;
382
                    *rnd = (int16_t)cbseed;
383

    
384
                    // FIR filter
385
                    fir_filter_value = 0.0;
386
                    for(j=0; j<10; j++)
387
                        fir_filter_value += qcelp_rnd_fir_coefs[j ]
388
                                          * (rnd[-j ] + rnd[-20+j]);
389

    
390
                    fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
391
                    *cdn_vector++ = tmp_gain * fir_filter_value;
392
                    rnd++;
393
                }
394
            }
395
            memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
396
        break;
397
        case RATE_OCTAVE:
398
            cbseed = q->first16bits;
399
            for(i=0; i<8; i++)
400
            {
401
                tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
402
                for(j=0; j<20; j++)
403
                {
404
                    cbseed = 521 * cbseed + 259;
405
                    *cdn_vector++ = tmp_gain * (int16_t)cbseed;
406
                }
407
            }
408
        break;
409
        case I_F_Q:
410
            cbseed = -44; // random codebook index
411
            for(i=0; i<4; i++)
412
            {
413
                tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
414
                for(j=0; j<40; j++)
415
                    *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
416
            }
417
        break;
418
    }
419
}
420

    
421
/**
422
 * Apply generic gain control.
423
 *
424
 * @param v_out output vector
425
 * @param v_in gain-controlled vector
426
 * @param v_ref vector to control gain of
427
 *
428
 * FIXME: If v_ref is a zero vector, it energy is zero
429
 *        and the behavior of the gain control is
430
 *        undefined in the specs.
431
 *
432
 * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
433
 */
434
static void apply_gain_ctrl(float *v_out, const float *v_ref,
435
                            const float *v_in)
436
{
437
    int   i, j, len;
438
    float scalefactor;
439

    
440
    for(i=0, j=0; i<4; i++)
441
    {
442
        scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
443
        if(scalefactor)
444
            scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40)
445
                        / scalefactor);
446
        else
447
            ff_log_missing_feature(NULL, "Zero energy for gain control", 1);
448
        for(len=j+40; j<len; j++)
449
            v_out[j] = scalefactor * v_in[j];
450
    }
451
}
452

    
453
/**
454
 * Apply filter in pitch-subframe steps.
455
 *
456
 * @param memory buffer for the previous state of the filter
457
 *        - must be able to contain 303 elements
458
 *        - the 143 first elements are from the previous state
459
 *        - the next 160 are for output
460
 * @param v_in input filter vector
461
 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
462
 * @param lag per-subframe lag array, each element is
463
 *        - between 16 and 143 if its corresponding pfrac is 0,
464
 *        - between 16 and 139 otherwise
465
 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
466
 *        otherwise
467
 *
468
 * @return filter output vector
469
 */
470
static const float *do_pitchfilter(float memory[303], const float v_in[160],
471
                                   const float gain[4], const uint8_t *lag,
472
                                   const uint8_t pfrac[4])
473
{
474
    int         i, j;
475
    float       *v_lag, *v_out;
476
    const float *v_len;
477

    
478
    v_out = memory + 143; // Output vector starts at memory[143].
479

    
480
    for(i=0; i<4; i++)
481
    {
482
        if(gain[i])
483
        {
484
            v_lag = memory + 143 + 40 * i - lag[i];
485
            for(v_len=v_in+40; v_in<v_len; v_in++)
486
            {
487
                if(pfrac[i]) // If it is a fractional lag...
488
                {
489
                    for(j=0, *v_out=0.; j<4; j++)
490
                        *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
491
                }else
492
                    *v_out = *v_lag;
493

    
494
                *v_out = *v_in + gain[i] * *v_out;
495

    
496
                v_lag++;
497
                v_out++;
498
            }
499
        }else
500
        {
501
            memcpy(v_out, v_in, 40 * sizeof(float));
502
            v_in  += 40;
503
            v_out += 40;
504
        }
505
    }
506

    
507
    memmove(memory, memory + 160, 143 * sizeof(float));
508
    return memory + 143;
509
}
510

    
511
/**
512
 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
513
 * TIA/EIA/IS-733 2.4.5.2
514
 *
515
 * @param q the context
516
 * @param cdn_vector the scaled codebook vector
517
 */
518
static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
519
{
520
    int         i;
521
    const float *v_synthesis_filtered, *v_pre_filtered;
522

    
523
    if(q->bitrate >= RATE_HALF ||
524
       (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
525
    {
526

    
527
        if(q->bitrate >= RATE_HALF)
528
        {
529

    
530
            // Compute gain & lag for the whole frame.
531
            for(i=0; i<4; i++)
532
            {
533
                q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
534

    
535
                q->pitch_lag[i] = q->frame.plag[i] + 16;
536
            }
537
        }else
538
        {
539
            float max_pitch_gain;
540

    
541
            if (q->erasure_count < 3)
542
                max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
543
             else
544
                max_pitch_gain = 0.0;
545
            for(i=0; i<4; i++)
546
                q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
547

    
548
            memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
549
        }
550

    
551
        // pitch synthesis filter
552
        v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
553
                                              cdn_vector, q->pitch_gain,
554
                                              q->pitch_lag, q->frame.pfrac);
555

    
556
        // pitch prefilter update
557
        for(i=0; i<4; i++)
558
            q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
559

    
560
        v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
561
                                        v_synthesis_filtered,
562
                                        q->pitch_gain, q->pitch_lag,
563
                                        q->frame.pfrac);
564

    
565
        apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
566
    }else
567
    {
568
        memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
569
               143 * sizeof(float));
570
        memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
571
        memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
572
        memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
573
    }
574
}
575

    
576
/**
577
 * Interpolates LSP frequencies and computes LPC coefficients
578
 * for a given bitrate & pitch subframe.
579
 *
580
 * TIA/EIA/IS-733 2.4.3.3.4
581
 *
582
 * @param q the context
583
 * @param curr_lspf LSP frequencies vector of the current frame
584
 * @param lpc float vector for the resulting LPC
585
 * @param subframe_num frame number in decoded stream
586
 */
587
void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
588
                     const int subframe_num)
589
{
590
    float interpolated_lspf[10];
591
    float weight;
592

    
593
    if(q->bitrate >= RATE_QUARTER)
594
        weight = 0.25 * (subframe_num + 1);
595
    else if(q->bitrate == RATE_OCTAVE && !subframe_num)
596
        weight = 0.625;
597
    else
598
        weight = 1.0;
599

    
600
    if(weight != 1.0)
601
    {
602
        weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
603
                             weight, 1.0 - weight, 10);
604
        ff_qcelp_lspf2lpc(interpolated_lspf, lpc);
605
    }else if(q->bitrate >= RATE_QUARTER ||
606
             (q->bitrate == I_F_Q && !subframe_num))
607
        ff_qcelp_lspf2lpc(curr_lspf, lpc);
608
}
609

    
610
static qcelp_packet_rate buf_size2bitrate(const int buf_size)
611
{
612
    switch(buf_size)
613
    {
614
        case 35: return RATE_FULL;
615
        case 17: return RATE_HALF;
616
        case  8: return RATE_QUARTER;
617
        case  4: return RATE_OCTAVE;
618
        case  1: return SILENCE;
619
    }
620

    
621
    return I_F_Q;
622
}
623

    
624
/**
625
 * Determine the bitrate from the frame size and/or the first byte of the frame.
626
 *
627
 * @param avctx the AV codec context
628
 * @param buf_size length of the buffer
629
 * @param buf the bufffer
630
 *
631
 * @return the bitrate on success,
632
 *         I_F_Q  if the bitrate cannot be satisfactorily determined
633
 *
634
 * TIA/EIA/IS-733 2.4.8.7.1
635
 */
636
static int determine_bitrate(AVCodecContext *avctx, const int buf_size,
637
                             const uint8_t **buf)
638
{
639
    qcelp_packet_rate bitrate;
640

    
641
    if((bitrate = buf_size2bitrate(buf_size)) >= 0)
642
    {
643
        if(bitrate > **buf)
644
        {
645
            QCELPContext *q = avctx->priv_data;
646
            if (!q->warned_buf_mismatch_bitrate)
647
            {
648
            av_log(avctx, AV_LOG_WARNING,
649
                   "Claimed bitrate and buffer size mismatch.\n");
650
                q->warned_buf_mismatch_bitrate = 1;
651
            }
652
            bitrate = **buf;
653
        }else if(bitrate < **buf)
654
        {
655
            av_log(avctx, AV_LOG_ERROR,
656
                   "Buffer is too small for the claimed bitrate.\n");
657
            return I_F_Q;
658
        }
659
        (*buf)++;
660
    }else if((bitrate = buf_size2bitrate(buf_size + 1)) >= 0)
661
    {
662
        av_log(avctx, AV_LOG_WARNING,
663
               "Bitrate byte is missing, guessing the bitrate from packet size.\n");
664
    }else
665
        return I_F_Q;
666

    
667
    if(bitrate == SILENCE)
668
    {
669
        // FIXME: the decoder should not handle SILENCE frames as I_F_Q frames
670
        ff_log_missing_feature(avctx, "Blank frame", 1);
671
        bitrate = I_F_Q;
672
    }
673
    return bitrate;
674
}
675

    
676
static void warn_insufficient_frame_quality(AVCodecContext *avctx,
677
                                            const char *message)
678
{
679
    av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
680
           message);
681
}
682

    
683
static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
684
                              const uint8_t *buf, int buf_size)
685
{
686
    QCELPContext *q = avctx->priv_data;
687
    float *outbuffer = data;
688
    int   i;
689
    float quantized_lspf[10], lpc[10];
690
    float gain[16];
691
    float *formant_mem;
692

    
693
    if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
694
    {
695
        warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
696
        goto erasure;
697
    }
698

    
699
    if(q->bitrate == RATE_OCTAVE &&
700
       (q->first16bits = AV_RB16(buf)) == 0xFFFF)
701
    {
702
        warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
703
        goto erasure;
704
    }
705

    
706
    if(q->bitrate > SILENCE)
707
    {
708
        const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
709
        const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
710
                                       + qcelp_unpacking_bitmaps_lengths[q->bitrate];
711
        uint8_t           *unpacked_data = (uint8_t *)&q->frame;
712

    
713
        init_get_bits(&q->gb, buf, 8*buf_size);
714

    
715
        memset(&q->frame, 0, sizeof(QCELPFrame));
716

    
717
        for(; bitmaps < bitmaps_end; bitmaps++)
718
            unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
719

    
720
        // Check for erasures/blanks on rates 1, 1/4 and 1/8.
721
        if(q->frame.reserved)
722
        {
723
            warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
724
            goto erasure;
725
        }
726
        if(q->bitrate == RATE_QUARTER &&
727
           codebook_sanity_check_for_rate_quarter(q->frame.cbgain))
728
        {
729
            warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
730
            goto erasure;
731
        }
732

    
733
        if(q->bitrate >= RATE_HALF)
734
        {
735
            for(i=0; i<4; i++)
736
            {
737
                if(q->frame.pfrac[i] && q->frame.plag[i] >= 124)
738
                {
739
                    warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
740
                    goto erasure;
741
                }
742
            }
743
        }
744
    }
745

    
746
    decode_gain_and_index(q, gain);
747
    compute_svector(q, gain, outbuffer);
748

    
749
    if(decode_lspf(q, quantized_lspf) < 0)
750
    {
751
        warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
752
        goto erasure;
753
    }
754

    
755

    
756
    apply_pitch_filters(q, outbuffer);
757

    
758
    if(q->bitrate == I_F_Q)
759
    {
760
erasure:
761
        q->bitrate = I_F_Q;
762
        q->erasure_count++;
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);
767
    }else
768
        q->erasure_count = 0;
769

    
770
    formant_mem = q->formant_mem + 10;
771
    for(i=0; i<4; i++)
772
    {
773
        interpolate_lpc(q, quantized_lspf, lpc, i);
774
        ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
775
                                     10);
776
        formant_mem += 40;
777
    }
778
    memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
779

    
780
    // FIXME: postfilter and final gain control should be here.
781
    // TIA/EIA/IS-733 2.4.8.6
782

    
783
    formant_mem = q->formant_mem + 10;
784
    for(i=0; i<160; i++)
785
        *outbuffer++ = av_clipf(*formant_mem++, QCELP_CLIP_LOWER_BOUND,
786
                                QCELP_CLIP_UPPER_BOUND);
787

    
788
    memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
789
    q->prev_bitrate = q->bitrate;
790

    
791
    *data_size = 160 * sizeof(*outbuffer);
792

    
793
    return *data_size;
794
}
795

    
796
AVCodec qcelp_decoder =
797
{
798
    .name   = "qcelp",
799
    .type   = CODEC_TYPE_AUDIO,
800
    .id     = CODEC_ID_QCELP,
801
    .init   = qcelp_decode_init,
802
    .decode = qcelp_decode_frame,
803
    .priv_data_size = sizeof(QCELPContext),
804
    .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
805
};