GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/g729dec.c Lines: 0 248 0.0 %
Date: 2019-11-18 18:00:01 Branches: 0 127 0.0 %

Line Branch Exec Source
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/*
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 * G.729, G729 Annex D decoders
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 * Copyright (c) 2008 Vladimir Voroshilov
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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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 */
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#include <inttypes.h>
23
#include <string.h>
24
25
#include "avcodec.h"
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#include "libavutil/avutil.h"
27
#include "get_bits.h"
28
#include "audiodsp.h"
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#include "internal.h"
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31
32
#include "g729.h"
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#include "lsp.h"
34
#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_pitch_delay.h"
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#include "acelp_vectors.h"
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#include "g729data.h"
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#include "g729postfilter.h"
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42
/**
43
 * minimum quantized LSF value (3.2.4)
44
 * 0.005 in Q13
45
 */
46
#define LSFQ_MIN                   40
47
48
/**
49
 * maximum quantized LSF value (3.2.4)
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 * 3.135 in Q13
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 */
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#define LSFQ_MAX                   25681
53
54
/**
55
 * minimum LSF distance (3.2.4)
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 * 0.0391 in Q13
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 */
58
#define LSFQ_DIFF_MIN              321
59
60
/// interpolation filter length
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#define INTERPOL_LEN              11
62
63
/**
64
 * minimum gain pitch value (3.8, Equation 47)
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 * 0.2 in (1.14)
66
 */
67
#define SHARP_MIN                  3277
68
69
/**
70
 * maximum gain pitch value (3.8, Equation 47)
71
 * (EE) This does not comply with the specification.
72
 * Specification says about 0.8, which should be
73
 * 13107 in (1.14), but reference C code uses
74
 * 13017 (equals to 0.7945) instead of it.
75
 */
76
#define SHARP_MAX                  13017
77
78
/**
79
 * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26  * subframe_size) in (7.13)
80
 */
81
#define MR_ENERGY 1018156
82
83
#define DECISION_NOISE        0
84
#define DECISION_INTERMEDIATE 1
85
#define DECISION_VOICE        2
86
87
typedef enum {
88
    FORMAT_G729_8K = 0,
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    FORMAT_G729D_6K4,
90
    FORMAT_COUNT,
91
} G729Formats;
92
93
typedef struct {
94
    uint8_t ac_index_bits[2];   ///< adaptive codebook index for second subframe (size in bits)
95
    uint8_t parity_bit;         ///< parity bit for pitch delay
96
    uint8_t gc_1st_index_bits;  ///< gain codebook (first stage) index (size in bits)
97
    uint8_t gc_2nd_index_bits;  ///< gain codebook (second stage) index (size in bits)
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    uint8_t fc_signs_bits;      ///< number of pulses in fixed-codebook vector
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    uint8_t fc_indexes_bits;    ///< size (in bits) of fixed-codebook index entry
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} G729FormatDescription;
101
102
typedef struct {
103
    /// past excitation signal buffer
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    int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
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106
    int16_t* exc;               ///< start of past excitation data in buffer
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    int pitch_delay_int_prev;   ///< integer part of previous subframe's pitch delay (4.1.3)
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    /// (2.13) LSP quantizer outputs
110
    int16_t  past_quantizer_output_buf[MA_NP + 1][10];
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    int16_t* past_quantizer_outputs[MA_NP + 1];
112
113
    int16_t lsfq[10];           ///< (2.13) quantized LSF coefficients from previous frame
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    int16_t lsp_buf[2][10];     ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
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    int16_t *lsp[2];            ///< pointers to lsp_buf
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    int16_t quant_energy[4];    ///< (5.10) past quantized energy
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    /// previous speech data for LP synthesis filter
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    int16_t syn_filter_data[10];
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    /// residual signal buffer (used in long-term postfilter)
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    int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
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    /// previous speech data for residual calculation filter
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    int16_t res_filter_data[SUBFRAME_SIZE+10];
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    /// previous speech data for short-term postfilter
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    int16_t pos_filter_data[SUBFRAME_SIZE+10];
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    /// (1.14) pitch gain of current and five previous subframes
133
    int16_t past_gain_pitch[6];
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135
    /// (14.1) gain code from current and previous subframe
136
    int16_t past_gain_code[2];
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138
    /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
139
    int16_t voice_decision;
140
141
    int16_t onset;              ///< detected onset level (0-2)
142
    int16_t was_periodic;       ///< whether previous frame was declared as periodic or not (4.4)
143
    int16_t ht_prev_data;       ///< previous data for 4.2.3, equation 86
144
    int gain_coeff;             ///< (1.14) gain coefficient (4.2.4)
145
    uint16_t rand_value;        ///< random number generator value (4.4.4)
146
    int ma_predictor_prev;      ///< switched MA predictor of LSP quantizer from last good frame
147
148
    /// (14.14) high-pass filter data (past input)
149
    int hpf_f[2];
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151
    /// high-pass filter data (past output)
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    int16_t hpf_z[2];
153
}  G729ChannelContext;
154
155
typedef struct {
156
    AudioDSPContext adsp;
157
158
    G729ChannelContext *channel_context;
159
} G729Context;
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161
static const G729FormatDescription format_g729_8k = {
162
    .ac_index_bits     = {8,5},
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    .parity_bit        = 1,
164
    .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
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    .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
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    .fc_signs_bits     = 4,
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    .fc_indexes_bits   = 13,
168
};
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170
static const G729FormatDescription format_g729d_6k4 = {
171
    .ac_index_bits     = {8,4},
172
    .parity_bit        = 0,
173
    .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
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    .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
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    .fc_signs_bits     = 2,
176
    .fc_indexes_bits   = 9,
177
};
178
179
/**
180
 * @brief pseudo random number generator
181
 */
182
static inline uint16_t g729_prng(uint16_t value)
183
{
184
    return 31821 * value + 13849;
185
}
186
187
/**
188
 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
189
 * @param[out] lsfq (2.13) quantized LSF coefficients
190
 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
191
 * @param ma_predictor switched MA predictor of LSP quantizer
192
 * @param vq_1st first stage vector of quantizer
193
 * @param vq_2nd_low second stage lower vector of LSP quantizer
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 * @param vq_2nd_high second stage higher vector of LSP quantizer
195
 */
196
static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
197
                       int16_t ma_predictor,
198
                       int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
199
{
200
    int i,j;
201
    static const uint8_t min_distance[2]={10, 5}; //(2.13)
202
    int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
203
204
    for (i = 0; i < 5; i++) {
205
        quantizer_output[i]     = cb_lsp_1st[vq_1st][i    ] + cb_lsp_2nd[vq_2nd_low ][i    ];
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        quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
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    }
208
209
    for (j = 0; j < 2; j++) {
210
        for (i = 1; i < 10; i++) {
211
            int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
212
            if (diff > 0) {
213
                quantizer_output[i - 1] -= diff;
214
                quantizer_output[i    ] += diff;
215
            }
216
        }
217
    }
218
219
    for (i = 0; i < 10; i++) {
220
        int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
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        for (j = 0; j < MA_NP; j++)
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            sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
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        lsfq[i] = sum >> 15;
225
    }
226
227
    ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
228
}
229
230
/**
231
 * Restores past LSP quantizer output using LSF from previous frame
232
 * @param[in,out] lsfq (2.13) quantized LSF coefficients
233
 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
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 * @param ma_predictor_prev MA predictor from previous frame
235
 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
236
 */
237
static void lsf_restore_from_previous(int16_t* lsfq,
238
                                      int16_t* past_quantizer_outputs[MA_NP + 1],
239
                                      int ma_predictor_prev)
240
{
241
    int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
242
    int i,k;
243
244
    for (i = 0; i < 10; i++) {
245
        int tmp = lsfq[i] << 15;
246
247
        for (k = 0; k < MA_NP; k++)
248
            tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
249
250
        quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
251
    }
252
}
253
254
/**
255
 * Constructs new excitation signal and applies phase filter to it
256
 * @param[out] out constructed speech signal
257
 * @param in original excitation signal
258
 * @param fc_cur (2.13) original fixed-codebook vector
259
 * @param gain_code (14.1) gain code
260
 * @param subframe_size length of the subframe
261
 */
262
static void g729d_get_new_exc(
263
        int16_t* out,
264
        const int16_t* in,
265
        const int16_t* fc_cur,
266
        int dstate,
267
        int gain_code,
268
        int subframe_size)
269
{
270
    int i;
271
    int16_t fc_new[SUBFRAME_SIZE];
272
273
    ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
274
275
    for (i = 0; i < subframe_size; i++) {
276
        out[i]  = in[i];
277
        out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
278
        out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
279
    }
280
}
281
282
/**
283
 * Makes decision about onset in current subframe
284
 * @param past_onset decision result of previous subframe
285
 * @param past_gain_code gain code of current and previous subframe
286
 *
287
 * @return onset decision result for current subframe
288
 */
289
static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
290
{
291
    if ((past_gain_code[0] >> 1) > past_gain_code[1])
292
        return 2;
293
294
    return FFMAX(past_onset-1, 0);
295
}
296
297
/**
298
 * Makes decision about voice presence in current subframe
299
 * @param onset onset level
300
 * @param prev_voice_decision voice decision result from previous subframe
301
 * @param past_gain_pitch pitch gain of current and previous subframes
302
 *
303
 * @return voice decision result for current subframe
304
 */
305
static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
306
{
307
    int i, low_gain_pitch_cnt, voice_decision;
308
309
    if (past_gain_pitch[0] >= 14745) {       // 0.9
310
        voice_decision = DECISION_VOICE;
311
    } else if (past_gain_pitch[0] <= 9830) { // 0.6
312
        voice_decision = DECISION_NOISE;
313
    } else {
314
        voice_decision = DECISION_INTERMEDIATE;
315
    }
316
317
    for (i = 0, low_gain_pitch_cnt = 0; i < 6; i++)
318
        if (past_gain_pitch[i] < 9830)
319
            low_gain_pitch_cnt++;
320
321
    if (low_gain_pitch_cnt > 2 && !onset)
322
        voice_decision = DECISION_NOISE;
323
324
    if (!onset && voice_decision > prev_voice_decision + 1)
325
        voice_decision--;
326
327
    if (onset && voice_decision < DECISION_VOICE)
328
        voice_decision++;
329
330
    return voice_decision;
331
}
332
333
static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
334
{
335
    int res = 0;
336
337
    while (order--)
338
        res += *v1++ * *v2++;
339
340
    return res;
341
}
342
343
static av_cold int decoder_init(AVCodecContext * avctx)
344
{
345
    G729Context *s = avctx->priv_data;
346
    G729ChannelContext *ctx;
347
    int c,i,k;
348
349
    if (avctx->channels < 1 || avctx->channels > 2) {
350
        av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels);
351
        return AVERROR(EINVAL);
352
    }
353
    avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
354
355
    /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
356
    avctx->frame_size = SUBFRAME_SIZE << 1;
357
358
    ctx =
359
    s->channel_context = av_mallocz(sizeof(G729ChannelContext) * avctx->channels);
360
    if (!ctx)
361
        return AVERROR(ENOMEM);
362
363
    for (c = 0; c < avctx->channels; c++) {
364
        ctx->gain_coeff = 16384; // 1.0 in (1.14)
365
366
        for (k = 0; k < MA_NP + 1; k++) {
367
            ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
368
            for (i = 1; i < 11; i++)
369
                ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
370
        }
371
372
        ctx->lsp[0] = ctx->lsp_buf[0];
373
        ctx->lsp[1] = ctx->lsp_buf[1];
374
        memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
375
376
        ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
377
378
        ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
379
380
        /* random seed initialization */
381
        ctx->rand_value = 21845;
382
383
        /* quantized prediction error */
384
        for (i = 0; i < 4; i++)
385
            ctx->quant_energy[i] = -14336; // -14 in (5.10)
386
387
        ctx++;
388
    }
389
390
    ff_audiodsp_init(&s->adsp);
391
    s->adsp.scalarproduct_int16 = scalarproduct_int16_c;
392
393
    return 0;
394
}
395
396
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
397
                        AVPacket *avpkt)
398
{
399
    const uint8_t *buf = avpkt->data;
400
    int buf_size       = avpkt->size;
401
    int16_t *out_frame;
402
    GetBitContext gb;
403
    const G729FormatDescription *format;
404
    int c, i;
405
    int16_t *tmp;
406
    G729Formats packet_type;
407
    G729Context *s = avctx->priv_data;
408
    G729ChannelContext *ctx = s->channel_context;
409
    int16_t lp[2][11];           // (3.12)
410
    uint8_t ma_predictor;     ///< switched MA predictor of LSP quantizer
411
    uint8_t quantizer_1st;    ///< first stage vector of quantizer
412
    uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
413
    uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
414
415
    int pitch_delay_int[2];      // pitch delay, integer part
416
    int pitch_delay_3x;          // pitch delay, multiplied by 3
417
    int16_t fc[SUBFRAME_SIZE];   // fixed-codebook vector
418
    int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
419
    int j, ret;
420
    int gain_before, gain_after;
421
    AVFrame *frame = data;
422
423
    frame->nb_samples = SUBFRAME_SIZE<<1;
424
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
425
        return ret;
426
427
    if (buf_size % ((G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels) == 0) {
428
        packet_type = FORMAT_G729_8K;
429
        format = &format_g729_8k;
430
        //Reset voice decision
431
        ctx->onset = 0;
432
        ctx->voice_decision = DECISION_VOICE;
433
        av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
434
    } else if (buf_size == G729D_6K4_BLOCK_SIZE * avctx->channels) {
435
        packet_type = FORMAT_G729D_6K4;
436
        format = &format_g729d_6k4;
437
        av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
438
    } else {
439
        av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
440
        return AVERROR_INVALIDDATA;
441
    }
442
443
    for (c = 0; c < avctx->channels; c++) {
444
        int frame_erasure = 0; ///< frame erasure detected during decoding
445
        int bad_pitch = 0;     ///< parity check failed
446
        int is_periodic = 0;   ///< whether one of the subframes is declared as periodic or not
447
        out_frame = (int16_t*)frame->data[c];
448
        if (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN) {
449
            if (*buf != ((avctx->channels - 1 - c) * 0x80 | 2))
450
                avpriv_request_sample(avctx, "First byte value %x for channel %d", *buf, c);
451
            buf++;
452
        }
453
454
        for (i = 0; i < buf_size; i++)
455
            frame_erasure |= buf[i];
456
        frame_erasure = !frame_erasure;
457
458
        init_get_bits(&gb, buf, 8*buf_size);
459
460
        ma_predictor     = get_bits(&gb, 1);
461
        quantizer_1st    = get_bits(&gb, VQ_1ST_BITS);
462
        quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
463
        quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
464
465
        if (frame_erasure) {
466
            lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
467
                                      ctx->ma_predictor_prev);
468
        } else {
469
            lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
470
                       ma_predictor,
471
                       quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
472
            ctx->ma_predictor_prev = ma_predictor;
473
        }
474
475
        tmp = ctx->past_quantizer_outputs[MA_NP];
476
        memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
477
                MA_NP * sizeof(int16_t*));
478
        ctx->past_quantizer_outputs[0] = tmp;
479
480
        ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
481
482
        ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
483
484
        FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
485
486
        for (i = 0; i < 2; i++) {
487
            int gain_corr_factor;
488
489
            uint8_t ac_index;      ///< adaptive codebook index
490
            uint8_t pulses_signs;  ///< fixed-codebook vector pulse signs
491
            int fc_indexes;        ///< fixed-codebook indexes
492
            uint8_t gc_1st_index;  ///< gain codebook (first stage) index
493
            uint8_t gc_2nd_index;  ///< gain codebook (second stage) index
494
495
            ac_index      = get_bits(&gb, format->ac_index_bits[i]);
496
            if (!i && format->parity_bit)
497
                bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
498
            fc_indexes    = get_bits(&gb, format->fc_indexes_bits);
499
            pulses_signs  = get_bits(&gb, format->fc_signs_bits);
500
            gc_1st_index  = get_bits(&gb, format->gc_1st_index_bits);
501
            gc_2nd_index  = get_bits(&gb, format->gc_2nd_index_bits);
502
503
            if (frame_erasure) {
504
                pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
505
            } else if (!i) {
506
                if (bad_pitch) {
507
                    pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
508
                } else {
509
                    pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
510
                }
511
            } else {
512
                int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
513
                                              PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
514
515
                if (packet_type == FORMAT_G729D_6K4) {
516
                    pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
517
                } else {
518
                    pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
519
                }
520
            }
521
522
            /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
523
            pitch_delay_int[i]  = (pitch_delay_3x + 1) / 3;
524
            if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
525
                av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
526
                pitch_delay_int[i] = PITCH_DELAY_MAX;
527
            }
528
529
            if (frame_erasure) {
530
                ctx->rand_value = g729_prng(ctx->rand_value);
531
                fc_indexes   = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
532
533
                ctx->rand_value = g729_prng(ctx->rand_value);
534
                pulses_signs = ctx->rand_value;
535
            }
536
537
538
            memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
539
            switch (packet_type) {
540
                case FORMAT_G729_8K:
541
                    ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
542
                                                ff_fc_4pulses_8bits_track_4,
543
                                                fc_indexes, pulses_signs, 3, 3);
544
                    break;
545
                case FORMAT_G729D_6K4:
546
                    ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
547
                                                ff_fc_2pulses_9bits_track2_gray,
548
                                                fc_indexes, pulses_signs, 1, 4);
549
                    break;
550
            }
551
552
            /*
553
              This filter enhances harmonic components of the fixed-codebook vector to
554
              improve the quality of the reconstructed speech.
555
556
                         / fc_v[i],                                    i < pitch_delay
557
              fc_v[i] = <
558
                         \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
559
            */
560
            if (SUBFRAME_SIZE > pitch_delay_int[i])
561
                ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
562
                                             fc + pitch_delay_int[i],
563
                                             fc, 1 << 14,
564
                                             av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
565
                                             0, 14,
566
                                             SUBFRAME_SIZE - pitch_delay_int[i]);
567
568
            memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
569
            ctx->past_gain_code[1] = ctx->past_gain_code[0];
570
571
            if (frame_erasure) {
572
                ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
573
                ctx->past_gain_code[0]  = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
574
575
                gain_corr_factor = 0;
576
            } else {
577
                if (packet_type == FORMAT_G729D_6K4) {
578
                    ctx->past_gain_pitch[0]  = cb_gain_1st_6k4[gc_1st_index][0] +
579
                                               cb_gain_2nd_6k4[gc_2nd_index][0];
580
                    gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
581
                                       cb_gain_2nd_6k4[gc_2nd_index][1];
582
583
                    /* Without check below overflow can occur in ff_acelp_update_past_gain.
584
                       It is not issue for G.729, because gain_corr_factor in it's case is always
585
                       greater than 1024, while in G.729D it can be even zero. */
586
                    gain_corr_factor = FFMAX(gain_corr_factor, 1024);
587
    #ifndef G729_BITEXACT
588
                    gain_corr_factor >>= 1;
589
    #endif
590
                } else {
591
                    ctx->past_gain_pitch[0]  = cb_gain_1st_8k[gc_1st_index][0] +
592
                                               cb_gain_2nd_8k[gc_2nd_index][0];
593
                    gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
594
                                       cb_gain_2nd_8k[gc_2nd_index][1];
595
                }
596
597
                /* Decode the fixed-codebook gain. */
598
                ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&s->adsp, gain_corr_factor,
599
                                                                   fc, MR_ENERGY,
600
                                                                   ctx->quant_energy,
601
                                                                   ma_prediction_coeff,
602
                                                                   SUBFRAME_SIZE, 4);
603
    #ifdef G729_BITEXACT
604
                /*
605
                  This correction required to get bit-exact result with
606
                  reference code, because gain_corr_factor in G.729D is
607
                  two times larger than in original G.729.
608
609
                  If bit-exact result is not issue then gain_corr_factor
610
                  can be simpler divided by 2 before call to g729_get_gain_code
611
                  instead of using correction below.
612
                */
613
                if (packet_type == FORMAT_G729D_6K4) {
614
                    gain_corr_factor >>= 1;
615
                    ctx->past_gain_code[0] >>= 1;
616
                }
617
    #endif
618
            }
619
            ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
620
621
            /* Routine requires rounding to lowest. */
622
            ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
623
                                 ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
624
                                 ff_acelp_interp_filter, 6,
625
                                 (pitch_delay_3x % 3) << 1,
626
                                 10, SUBFRAME_SIZE);
627
628
            ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
629
                                         ctx->exc + i * SUBFRAME_SIZE, fc,
630
                                         (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
631
                                         ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
632
                                         1 << 13, 14, SUBFRAME_SIZE);
633
634
            memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
635
636
            if (ff_celp_lp_synthesis_filter(
637
                synth+10,
638
                &lp[i][1],
639
                ctx->exc  + i * SUBFRAME_SIZE,
640
                SUBFRAME_SIZE,
641
                10,
642
                1,
643
                0,
644
                0x800))
645
                /* Overflow occurred, downscale excitation signal... */
646
                for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
647
                    ctx->exc_base[j] >>= 2;
648
649
            /* ... and make synthesis again. */
650
            if (packet_type == FORMAT_G729D_6K4) {
651
                int16_t exc_new[SUBFRAME_SIZE];
652
653
                ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
654
                ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
655
656
                g729d_get_new_exc(exc_new, ctx->exc  + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
657
658
                ff_celp_lp_synthesis_filter(
659
                        synth+10,
660
                        &lp[i][1],
661
                        exc_new,
662
                        SUBFRAME_SIZE,
663
                        10,
664
                        0,
665
                        0,
666
                        0x800);
667
            } else {
668
                ff_celp_lp_synthesis_filter(
669
                        synth+10,
670
                        &lp[i][1],
671
                        ctx->exc  + i * SUBFRAME_SIZE,
672
                        SUBFRAME_SIZE,
673
                        10,
674
                        0,
675
                        0,
676
                        0x800);
677
            }
678
            /* Save data (without postfilter) for use in next subframe. */
679
            memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
680
681
            /* Calculate gain of unfiltered signal for use in AGC. */
682
            gain_before = 0;
683
            for (j = 0; j < SUBFRAME_SIZE; j++)
684
                gain_before += FFABS(synth[j+10]);
685
686
            /* Call postfilter and also update voicing decision for use in next frame. */
687
            ff_g729_postfilter(
688
                    &s->adsp,
689
                    &ctx->ht_prev_data,
690
                    &is_periodic,
691
                    &lp[i][0],
692
                    pitch_delay_int[0],
693
                    ctx->residual,
694
                    ctx->res_filter_data,
695
                    ctx->pos_filter_data,
696
                    synth+10,
697
                    SUBFRAME_SIZE);
698
699
            /* Calculate gain of filtered signal for use in AGC. */
700
            gain_after = 0;
701
            for (j = 0; j < SUBFRAME_SIZE; j++)
702
                gain_after += FFABS(synth[j+10]);
703
704
            ctx->gain_coeff = ff_g729_adaptive_gain_control(
705
                    gain_before,
706
                    gain_after,
707
                    synth+10,
708
                    SUBFRAME_SIZE,
709
                    ctx->gain_coeff);
710
711
            if (frame_erasure) {
712
                ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
713
            } else {
714
                ctx->pitch_delay_int_prev = pitch_delay_int[i];
715
            }
716
717
            memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
718
            ff_acelp_high_pass_filter(
719
                    out_frame + i*SUBFRAME_SIZE,
720
                    ctx->hpf_f,
721
                    synth+10,
722
                    SUBFRAME_SIZE);
723
            memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
724
        }
725
726
        ctx->was_periodic = is_periodic;
727
728
        /* Save signal for use in next frame. */
729
        memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
730
731
        buf += packet_type == FORMAT_G729_8K ? G729_8K_BLOCK_SIZE : G729D_6K4_BLOCK_SIZE;
732
        ctx++;
733
    }
734
735
    *got_frame_ptr = 1;
736
    return packet_type == FORMAT_G729_8K ? (G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels : G729D_6K4_BLOCK_SIZE * avctx->channels;
737
}
738
739
static av_cold int decode_close(AVCodecContext *avctx)
740
{
741
    G729Context *s = avctx->priv_data;
742
    av_freep(&s->channel_context);
743
744
    return 0;
745
}
746
747
AVCodec ff_g729_decoder = {
748
    .name           = "g729",
749
    .long_name      = NULL_IF_CONFIG_SMALL("G.729"),
750
    .type           = AVMEDIA_TYPE_AUDIO,
751
    .id             = AV_CODEC_ID_G729,
752
    .priv_data_size = sizeof(G729Context),
753
    .init           = decoder_init,
754
    .decode         = decode_frame,
755
    .close          = decode_close,
756
    .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
757
};
758
759
AVCodec ff_acelp_kelvin_decoder = {
760
    .name           = "acelp.kelvin",
761
    .long_name      = NULL_IF_CONFIG_SMALL("Sipro ACELP.KELVIN"),
762
    .type           = AVMEDIA_TYPE_AUDIO,
763
    .id             = AV_CODEC_ID_ACELP_KELVIN,
764
    .priv_data_size = sizeof(G729Context),
765
    .init           = decoder_init,
766
    .decode         = decode_frame,
767
    .close          = decode_close,
768
    .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
769
};