GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/g729dec.c Lines: 0 250 0.0 %
Date: 2020-04-04 00:26:16 Branches: 0 129 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>
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25
#include "avcodec.h"
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#include "libavutil/avutil.h"
27
#include "get_bits.h"
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#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)
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 * 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
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    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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    uint8_t block_size;
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} G729FormatDescription;
102
103
typedef struct {
104
    /// past excitation signal buffer
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    int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
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    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
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    int16_t  past_quantizer_output_buf[MA_NP + 1][10];
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    int16_t* past_quantizer_outputs[MA_NP + 1];
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114
    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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118
    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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123
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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
134
    int16_t past_gain_pitch[6];
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    /// (14.1) gain code from current and previous subframe
137
    int16_t past_gain_code[2];
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139
    /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
140
    int16_t voice_decision;
141
142
    int16_t onset;              ///< detected onset level (0-2)
143
    int16_t was_periodic;       ///< whether previous frame was declared as periodic or not (4.4)
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    int16_t ht_prev_data;       ///< previous data for 4.2.3, equation 86
145
    int gain_coeff;             ///< (1.14) gain coefficient (4.2.4)
146
    uint16_t rand_value;        ///< random number generator value (4.4.4)
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    int ma_predictor_prev;      ///< switched MA predictor of LSP quantizer from last good frame
148
149
    /// (14.14) high-pass filter data (past input)
150
    int hpf_f[2];
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    /// high-pass filter data (past output)
153
    int16_t hpf_z[2];
154
}  G729ChannelContext;
155
156
typedef struct {
157
    AudioDSPContext adsp;
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159
    G729ChannelContext *channel_context;
160
} G729Context;
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static const G729FormatDescription format_g729_8k = {
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    .ac_index_bits     = {8,5},
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    .parity_bit        = 1,
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    .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,
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    .block_size        = G729_8K_BLOCK_SIZE,
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};
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static const G729FormatDescription format_g729d_6k4 = {
173
    .ac_index_bits     = {8,4},
174
    .parity_bit        = 0,
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    .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,
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    .fc_indexes_bits   = 9,
179
    .block_size        = G729D_6K4_BLOCK_SIZE,
180
};
181
182
/**
183
 * @brief pseudo random number generator
184
 */
185
static inline uint16_t g729_prng(uint16_t value)
186
{
187
    return 31821 * value + 13849;
188
}
189
190
/**
191
 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
192
 * @param[out] lsfq (2.13) quantized LSF coefficients
193
 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
194
 * @param ma_predictor switched MA predictor of LSP quantizer
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 * @param vq_1st first stage vector of quantizer
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 * @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
198
 */
199
static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
200
                       int16_t ma_predictor,
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                       int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
202
{
203
    int i,j;
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    static const uint8_t min_distance[2]={10, 5}; //(2.13)
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    int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
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207
    for (i = 0; i < 5; i++) {
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        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];
210
    }
211
212
    for (j = 0; j < 2; j++) {
213
        for (i = 1; i < 10; i++) {
214
            int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
215
            if (diff > 0) {
216
                quantizer_output[i - 1] -= diff;
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                quantizer_output[i    ] += diff;
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            }
219
        }
220
    }
221
222
    for (i = 0; i < 10; i++) {
223
        int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
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        for (j = 0; j < MA_NP; j++)
225
            sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
226
227
        lsfq[i] = sum >> 15;
228
    }
229
230
    ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
231
}
232
233
/**
234
 * Restores past LSP quantizer output using LSF from previous frame
235
 * @param[in,out] lsfq (2.13) quantized LSF coefficients
236
 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
237
 * @param ma_predictor_prev MA predictor from previous frame
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 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
239
 */
240
static void lsf_restore_from_previous(int16_t* lsfq,
241
                                      int16_t* past_quantizer_outputs[MA_NP + 1],
242
                                      int ma_predictor_prev)
243
{
244
    int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
245
    int i,k;
246
247
    for (i = 0; i < 10; i++) {
248
        int tmp = lsfq[i] << 15;
249
250
        for (k = 0; k < MA_NP; k++)
251
            tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
252
253
        quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
254
    }
255
}
256
257
/**
258
 * Constructs new excitation signal and applies phase filter to it
259
 * @param[out] out constructed speech signal
260
 * @param in original excitation signal
261
 * @param fc_cur (2.13) original fixed-codebook vector
262
 * @param gain_code (14.1) gain code
263
 * @param subframe_size length of the subframe
264
 */
265
static void g729d_get_new_exc(
266
        int16_t* out,
267
        const int16_t* in,
268
        const int16_t* fc_cur,
269
        int dstate,
270
        int gain_code,
271
        int subframe_size)
272
{
273
    int i;
274
    int16_t fc_new[SUBFRAME_SIZE];
275
276
    ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
277
278
    for (i = 0; i < subframe_size; i++) {
279
        out[i]  = in[i];
280
        out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
281
        out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
282
    }
283
}
284
285
/**
286
 * Makes decision about onset in current subframe
287
 * @param past_onset decision result of previous subframe
288
 * @param past_gain_code gain code of current and previous subframe
289
 *
290
 * @return onset decision result for current subframe
291
 */
292
static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
293
{
294
    if ((past_gain_code[0] >> 1) > past_gain_code[1])
295
        return 2;
296
297
    return FFMAX(past_onset-1, 0);
298
}
299
300
/**
301
 * Makes decision about voice presence in current subframe
302
 * @param onset onset level
303
 * @param prev_voice_decision voice decision result from previous subframe
304
 * @param past_gain_pitch pitch gain of current and previous subframes
305
 *
306
 * @return voice decision result for current subframe
307
 */
308
static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
309
{
310
    int i, low_gain_pitch_cnt, voice_decision;
311
312
    if (past_gain_pitch[0] >= 14745) {       // 0.9
313
        voice_decision = DECISION_VOICE;
314
    } else if (past_gain_pitch[0] <= 9830) { // 0.6
315
        voice_decision = DECISION_NOISE;
316
    } else {
317
        voice_decision = DECISION_INTERMEDIATE;
318
    }
319
320
    for (i = 0, low_gain_pitch_cnt = 0; i < 6; i++)
321
        if (past_gain_pitch[i] < 9830)
322
            low_gain_pitch_cnt++;
323
324
    if (low_gain_pitch_cnt > 2 && !onset)
325
        voice_decision = DECISION_NOISE;
326
327
    if (!onset && voice_decision > prev_voice_decision + 1)
328
        voice_decision--;
329
330
    if (onset && voice_decision < DECISION_VOICE)
331
        voice_decision++;
332
333
    return voice_decision;
334
}
335
336
static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
337
{
338
    int64_t res = 0;
339
340
    while (order--)
341
        res += *v1++ * *v2++;
342
343
    if      (res > INT32_MAX) return INT32_MAX;
344
    else if (res < INT32_MIN) return INT32_MIN;
345
346
    return res;
347
}
348
349
static av_cold int decoder_init(AVCodecContext * avctx)
350
{
351
    G729Context *s = avctx->priv_data;
352
    G729ChannelContext *ctx;
353
    int c,i,k;
354
355
    if (avctx->channels < 1 || avctx->channels > 2) {
356
        av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels);
357
        return AVERROR(EINVAL);
358
    }
359
    avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
360
361
    /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
362
    avctx->frame_size = SUBFRAME_SIZE << 1;
363
364
    ctx =
365
    s->channel_context = av_mallocz(sizeof(G729ChannelContext) * avctx->channels);
366
    if (!ctx)
367
        return AVERROR(ENOMEM);
368
369
    for (c = 0; c < avctx->channels; c++) {
370
        ctx->gain_coeff = 16384; // 1.0 in (1.14)
371
372
        for (k = 0; k < MA_NP + 1; k++) {
373
            ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
374
            for (i = 1; i < 11; i++)
375
                ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
376
        }
377
378
        ctx->lsp[0] = ctx->lsp_buf[0];
379
        ctx->lsp[1] = ctx->lsp_buf[1];
380
        memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
381
382
        ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
383
384
        ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
385
386
        /* random seed initialization */
387
        ctx->rand_value = 21845;
388
389
        /* quantized prediction error */
390
        for (i = 0; i < 4; i++)
391
            ctx->quant_energy[i] = -14336; // -14 in (5.10)
392
393
        ctx++;
394
    }
395
396
    ff_audiodsp_init(&s->adsp);
397
    s->adsp.scalarproduct_int16 = scalarproduct_int16_c;
398
399
    return 0;
400
}
401
402
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
403
                        AVPacket *avpkt)
404
{
405
    const uint8_t *buf = avpkt->data;
406
    int buf_size       = avpkt->size;
407
    int16_t *out_frame;
408
    GetBitContext gb;
409
    const G729FormatDescription *format;
410
    int c, i;
411
    int16_t *tmp;
412
    G729Formats packet_type;
413
    G729Context *s = avctx->priv_data;
414
    G729ChannelContext *ctx = s->channel_context;
415
    int16_t lp[2][11];           // (3.12)
416
    uint8_t ma_predictor;     ///< switched MA predictor of LSP quantizer
417
    uint8_t quantizer_1st;    ///< first stage vector of quantizer
418
    uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
419
    uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
420
421
    int pitch_delay_int[2];      // pitch delay, integer part
422
    int pitch_delay_3x;          // pitch delay, multiplied by 3
423
    int16_t fc[SUBFRAME_SIZE];   // fixed-codebook vector
424
    int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
425
    int j, ret;
426
    int gain_before, gain_after;
427
    AVFrame *frame = data;
428
429
    frame->nb_samples = SUBFRAME_SIZE<<1;
430
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
431
        return ret;
432
433
    if (buf_size && buf_size % ((G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels) == 0) {
434
        packet_type = FORMAT_G729_8K;
435
        format = &format_g729_8k;
436
        //Reset voice decision
437
        ctx->onset = 0;
438
        ctx->voice_decision = DECISION_VOICE;
439
        av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
440
    } else if (buf_size == G729D_6K4_BLOCK_SIZE * avctx->channels && avctx->codec_id != AV_CODEC_ID_ACELP_KELVIN) {
441
        packet_type = FORMAT_G729D_6K4;
442
        format = &format_g729d_6k4;
443
        av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
444
    } else {
445
        av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
446
        return AVERROR_INVALIDDATA;
447
    }
448
449
    for (c = 0; c < avctx->channels; c++) {
450
        int frame_erasure = 0; ///< frame erasure detected during decoding
451
        int bad_pitch = 0;     ///< parity check failed
452
        int is_periodic = 0;   ///< whether one of the subframes is declared as periodic or not
453
        out_frame = (int16_t*)frame->data[c];
454
        if (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN) {
455
            if (*buf != ((avctx->channels - 1 - c) * 0x80 | 2))
456
                avpriv_request_sample(avctx, "First byte value %x for channel %d", *buf, c);
457
            buf++;
458
        }
459
460
        for (i = 0; i < format->block_size; i++)
461
            frame_erasure |= buf[i];
462
        frame_erasure = !frame_erasure;
463
464
        init_get_bits8(&gb, buf, format->block_size);
465
466
        ma_predictor     = get_bits(&gb, 1);
467
        quantizer_1st    = get_bits(&gb, VQ_1ST_BITS);
468
        quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
469
        quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
470
471
        if (frame_erasure) {
472
            lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
473
                                      ctx->ma_predictor_prev);
474
        } else {
475
            lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
476
                       ma_predictor,
477
                       quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
478
            ctx->ma_predictor_prev = ma_predictor;
479
        }
480
481
        tmp = ctx->past_quantizer_outputs[MA_NP];
482
        memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
483
                MA_NP * sizeof(int16_t*));
484
        ctx->past_quantizer_outputs[0] = tmp;
485
486
        ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
487
488
        ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
489
490
        FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
491
492
        for (i = 0; i < 2; i++) {
493
            int gain_corr_factor;
494
495
            uint8_t ac_index;      ///< adaptive codebook index
496
            uint8_t pulses_signs;  ///< fixed-codebook vector pulse signs
497
            int fc_indexes;        ///< fixed-codebook indexes
498
            uint8_t gc_1st_index;  ///< gain codebook (first stage) index
499
            uint8_t gc_2nd_index;  ///< gain codebook (second stage) index
500
501
            ac_index      = get_bits(&gb, format->ac_index_bits[i]);
502
            if (!i && format->parity_bit)
503
                bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
504
            fc_indexes    = get_bits(&gb, format->fc_indexes_bits);
505
            pulses_signs  = get_bits(&gb, format->fc_signs_bits);
506
            gc_1st_index  = get_bits(&gb, format->gc_1st_index_bits);
507
            gc_2nd_index  = get_bits(&gb, format->gc_2nd_index_bits);
508
509
            if (frame_erasure) {
510
                pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
511
            } else if (!i) {
512
                if (bad_pitch) {
513
                    pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
514
                } else {
515
                    pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
516
                }
517
            } else {
518
                int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
519
                                              PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
520
521
                if (packet_type == FORMAT_G729D_6K4) {
522
                    pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
523
                } else {
524
                    pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
525
                }
526
            }
527
528
            /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
529
            pitch_delay_int[i]  = (pitch_delay_3x + 1) / 3;
530
            if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
531
                av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
532
                pitch_delay_int[i] = PITCH_DELAY_MAX;
533
            }
534
535
            if (frame_erasure) {
536
                ctx->rand_value = g729_prng(ctx->rand_value);
537
                fc_indexes   = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
538
539
                ctx->rand_value = g729_prng(ctx->rand_value);
540
                pulses_signs = ctx->rand_value;
541
            }
542
543
544
            memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
545
            switch (packet_type) {
546
                case FORMAT_G729_8K:
547
                    ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
548
                                                ff_fc_4pulses_8bits_track_4,
549
                                                fc_indexes, pulses_signs, 3, 3);
550
                    break;
551
                case FORMAT_G729D_6K4:
552
                    ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
553
                                                ff_fc_2pulses_9bits_track2_gray,
554
                                                fc_indexes, pulses_signs, 1, 4);
555
                    break;
556
            }
557
558
            /*
559
              This filter enhances harmonic components of the fixed-codebook vector to
560
              improve the quality of the reconstructed speech.
561
562
                         / fc_v[i],                                    i < pitch_delay
563
              fc_v[i] = <
564
                         \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
565
            */
566
            if (SUBFRAME_SIZE > pitch_delay_int[i])
567
                ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
568
                                             fc + pitch_delay_int[i],
569
                                             fc, 1 << 14,
570
                                             av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
571
                                             0, 14,
572
                                             SUBFRAME_SIZE - pitch_delay_int[i]);
573
574
            memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
575
            ctx->past_gain_code[1] = ctx->past_gain_code[0];
576
577
            if (frame_erasure) {
578
                ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
579
                ctx->past_gain_code[0]  = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
580
581
                gain_corr_factor = 0;
582
            } else {
583
                if (packet_type == FORMAT_G729D_6K4) {
584
                    ctx->past_gain_pitch[0]  = cb_gain_1st_6k4[gc_1st_index][0] +
585
                                               cb_gain_2nd_6k4[gc_2nd_index][0];
586
                    gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
587
                                       cb_gain_2nd_6k4[gc_2nd_index][1];
588
589
                    /* Without check below overflow can occur in ff_acelp_update_past_gain.
590
                       It is not issue for G.729, because gain_corr_factor in it's case is always
591
                       greater than 1024, while in G.729D it can be even zero. */
592
                    gain_corr_factor = FFMAX(gain_corr_factor, 1024);
593
    #ifndef G729_BITEXACT
594
                    gain_corr_factor >>= 1;
595
    #endif
596
                } else {
597
                    ctx->past_gain_pitch[0]  = cb_gain_1st_8k[gc_1st_index][0] +
598
                                               cb_gain_2nd_8k[gc_2nd_index][0];
599
                    gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
600
                                       cb_gain_2nd_8k[gc_2nd_index][1];
601
                }
602
603
                /* Decode the fixed-codebook gain. */
604
                ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&s->adsp, gain_corr_factor,
605
                                                                   fc, MR_ENERGY,
606
                                                                   ctx->quant_energy,
607
                                                                   ma_prediction_coeff,
608
                                                                   SUBFRAME_SIZE, 4);
609
    #ifdef G729_BITEXACT
610
                /*
611
                  This correction required to get bit-exact result with
612
                  reference code, because gain_corr_factor in G.729D is
613
                  two times larger than in original G.729.
614
615
                  If bit-exact result is not issue then gain_corr_factor
616
                  can be simpler divided by 2 before call to g729_get_gain_code
617
                  instead of using correction below.
618
                */
619
                if (packet_type == FORMAT_G729D_6K4) {
620
                    gain_corr_factor >>= 1;
621
                    ctx->past_gain_code[0] >>= 1;
622
                }
623
    #endif
624
            }
625
            ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
626
627
            /* Routine requires rounding to lowest. */
628
            ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
629
                                 ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
630
                                 ff_acelp_interp_filter, 6,
631
                                 (pitch_delay_3x % 3) << 1,
632
                                 10, SUBFRAME_SIZE);
633
634
            ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
635
                                         ctx->exc + i * SUBFRAME_SIZE, fc,
636
                                         (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
637
                                         ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
638
                                         1 << 13, 14, SUBFRAME_SIZE);
639
640
            memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
641
642
            if (ff_celp_lp_synthesis_filter(
643
                synth+10,
644
                &lp[i][1],
645
                ctx->exc  + i * SUBFRAME_SIZE,
646
                SUBFRAME_SIZE,
647
                10,
648
                1,
649
                0,
650
                0x800))
651
                /* Overflow occurred, downscale excitation signal... */
652
                for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
653
                    ctx->exc_base[j] >>= 2;
654
655
            /* ... and make synthesis again. */
656
            if (packet_type == FORMAT_G729D_6K4) {
657
                int16_t exc_new[SUBFRAME_SIZE];
658
659
                ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
660
                ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
661
662
                g729d_get_new_exc(exc_new, ctx->exc  + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
663
664
                ff_celp_lp_synthesis_filter(
665
                        synth+10,
666
                        &lp[i][1],
667
                        exc_new,
668
                        SUBFRAME_SIZE,
669
                        10,
670
                        0,
671
                        0,
672
                        0x800);
673
            } else {
674
                ff_celp_lp_synthesis_filter(
675
                        synth+10,
676
                        &lp[i][1],
677
                        ctx->exc  + i * SUBFRAME_SIZE,
678
                        SUBFRAME_SIZE,
679
                        10,
680
                        0,
681
                        0,
682
                        0x800);
683
            }
684
            /* Save data (without postfilter) for use in next subframe. */
685
            memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
686
687
            /* Calculate gain of unfiltered signal for use in AGC. */
688
            gain_before = 0;
689
            for (j = 0; j < SUBFRAME_SIZE; j++)
690
                gain_before += FFABS(synth[j+10]);
691
692
            /* Call postfilter and also update voicing decision for use in next frame. */
693
            ff_g729_postfilter(
694
                    &s->adsp,
695
                    &ctx->ht_prev_data,
696
                    &is_periodic,
697
                    &lp[i][0],
698
                    pitch_delay_int[0],
699
                    ctx->residual,
700
                    ctx->res_filter_data,
701
                    ctx->pos_filter_data,
702
                    synth+10,
703
                    SUBFRAME_SIZE);
704
705
            /* Calculate gain of filtered signal for use in AGC. */
706
            gain_after = 0;
707
            for (j = 0; j < SUBFRAME_SIZE; j++)
708
                gain_after += FFABS(synth[j+10]);
709
710
            ctx->gain_coeff = ff_g729_adaptive_gain_control(
711
                    gain_before,
712
                    gain_after,
713
                    synth+10,
714
                    SUBFRAME_SIZE,
715
                    ctx->gain_coeff);
716
717
            if (frame_erasure) {
718
                ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
719
            } else {
720
                ctx->pitch_delay_int_prev = pitch_delay_int[i];
721
            }
722
723
            memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
724
            ff_acelp_high_pass_filter(
725
                    out_frame + i*SUBFRAME_SIZE,
726
                    ctx->hpf_f,
727
                    synth+10,
728
                    SUBFRAME_SIZE);
729
            memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
730
        }
731
732
        ctx->was_periodic = is_periodic;
733
734
        /* Save signal for use in next frame. */
735
        memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
736
737
        buf += format->block_size;
738
        ctx++;
739
    }
740
741
    *got_frame_ptr = 1;
742
    return (format->block_size + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels;
743
}
744
745
static av_cold int decode_close(AVCodecContext *avctx)
746
{
747
    G729Context *s = avctx->priv_data;
748
    av_freep(&s->channel_context);
749
750
    return 0;
751
}
752
753
AVCodec ff_g729_decoder = {
754
    .name           = "g729",
755
    .long_name      = NULL_IF_CONFIG_SMALL("G.729"),
756
    .type           = AVMEDIA_TYPE_AUDIO,
757
    .id             = AV_CODEC_ID_G729,
758
    .priv_data_size = sizeof(G729Context),
759
    .init           = decoder_init,
760
    .decode         = decode_frame,
761
    .close          = decode_close,
762
    .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
763
};
764
765
AVCodec ff_acelp_kelvin_decoder = {
766
    .name           = "acelp.kelvin",
767
    .long_name      = NULL_IF_CONFIG_SMALL("Sipro ACELP.KELVIN"),
768
    .type           = AVMEDIA_TYPE_AUDIO,
769
    .id             = AV_CODEC_ID_ACELP_KELVIN,
770
    .priv_data_size = sizeof(G729Context),
771
    .init           = decoder_init,
772
    .decode         = decode_frame,
773
    .close          = decode_close,
774
    .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
775
};