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1 /*
2  * G.729, G729 Annex D decoders
3  * Copyright (c) 2008 Vladimir Voroshilov
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21
22 #include <inttypes.h>
23 #include <string.h>
24
25 #include "avcodec.h"
26 #include "libavutil/avutil.h"
27 #include "get_bits.h"
28 #include "audiodsp.h"
29 #include "internal.h"
30
31
32 #include "g729.h"
33 #include "lsp.h"
34 #include "celp_math.h"
35 #include "celp_filters.h"
36 #include "acelp_filters.h"
37 #include "acelp_pitch_delay.h"
38 #include "acelp_vectors.h"
39 #include "g729data.h"
40 #include "g729postfilter.h"
41
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)
50  * 3.135 in Q13
51  */
52 #define LSFQ_MAX                   25681
53
54 /**
55  * minimum LSF distance (3.2.4)
56  * 0.0391 in Q13
57  */
58 #define LSFQ_DIFF_MIN              321
59
60 /// interpolation filter length
61 #define INTERPOL_LEN              11
62
63 /**
64  * minimum gain pitch value (3.8, Equation 47)
65  * 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,
89     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)
98     uint8_t fc_signs_bits;      ///< number of pulses in fixed-codebook vector
99     uint8_t fc_indexes_bits;    ///< size (in bits) of fixed-codebook index entry
100 } G729FormatDescription;
101
102 typedef struct {
103     AudioDSPContext adsp;
104
105     /// past excitation signal buffer
106     int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
107
108     int16_t* exc;               ///< start of past excitation data in buffer
109     int pitch_delay_int_prev;   ///< integer part of previous subframe's pitch delay (4.1.3)
110
111     /// (2.13) LSP quantizer outputs
112     int16_t  past_quantizer_output_buf[MA_NP + 1][10];
113     int16_t* past_quantizer_outputs[MA_NP + 1];
114
115     int16_t lsfq[10];           ///< (2.13) quantized LSF coefficients from previous frame
116     int16_t lsp_buf[2][10];     ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
117     int16_t *lsp[2];            ///< pointers to lsp_buf
118
119     int16_t quant_energy[4];    ///< (5.10) past quantized energy
120
121     /// previous speech data for LP synthesis filter
122     int16_t syn_filter_data[10];
123
124
125     /// residual signal buffer (used in long-term postfilter)
126     int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
127
128     /// previous speech data for residual calculation filter
129     int16_t res_filter_data[SUBFRAME_SIZE+10];
130
131     /// previous speech data for short-term postfilter
132     int16_t pos_filter_data[SUBFRAME_SIZE+10];
133
134     /// (1.14) pitch gain of current and five previous subframes
135     int16_t past_gain_pitch[6];
136
137     /// (14.1) gain code from current and previous subframe
138     int16_t past_gain_code[2];
139
140     /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
141     int16_t voice_decision;
142
143     int16_t onset;              ///< detected onset level (0-2)
144     int16_t was_periodic;       ///< whether previous frame was declared as periodic or not (4.4)
145     int16_t ht_prev_data;       ///< previous data for 4.2.3, equation 86
146     int gain_coeff;             ///< (1.14) gain coefficient (4.2.4)
147     uint16_t rand_value;        ///< random number generator value (4.4.4)
148     int ma_predictor_prev;      ///< switched MA predictor of LSP quantizer from last good frame
149
150     /// (14.14) high-pass filter data (past input)
151     int hpf_f[2];
152
153     /// high-pass filter data (past output)
154     int16_t hpf_z[2];
155 }  G729Context;
156
157 static const G729FormatDescription format_g729_8k = {
158     .ac_index_bits     = {8,5},
159     .parity_bit        = 1,
160     .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
161     .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
162     .fc_signs_bits     = 4,
163     .fc_indexes_bits   = 13,
164 };
165
166 static const G729FormatDescription format_g729d_6k4 = {
167     .ac_index_bits     = {8,4},
168     .parity_bit        = 0,
169     .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
170     .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
171     .fc_signs_bits     = 2,
172     .fc_indexes_bits   = 9,
173 };
174
175 /**
176  * @brief pseudo random number generator
177  */
178 static inline uint16_t g729_prng(uint16_t value)
179 {
180     return 31821 * value + 13849;
181 }
182
183 /**
184  * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
185  * @param[out] lsfq (2.13) quantized LSF coefficients
186  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
187  * @param ma_predictor switched MA predictor of LSP quantizer
188  * @param vq_1st first stage vector of quantizer
189  * @param vq_2nd_low second stage lower vector of LSP quantizer
190  * @param vq_2nd_high second stage higher vector of LSP quantizer
191  */
192 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
193                        int16_t ma_predictor,
194                        int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
195 {
196     int i,j;
197     static const uint8_t min_distance[2]={10, 5}; //(2.13)
198     int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
199
200     for (i = 0; i < 5; i++) {
201         quantizer_output[i]     = cb_lsp_1st[vq_1st][i    ] + cb_lsp_2nd[vq_2nd_low ][i    ];
202         quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
203     }
204
205     for (j = 0; j < 2; j++) {
206         for (i = 1; i < 10; i++) {
207             int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
208             if (diff > 0) {
209                 quantizer_output[i - 1] -= diff;
210                 quantizer_output[i    ] += diff;
211             }
212         }
213     }
214
215     for (i = 0; i < 10; i++) {
216         int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
217         for (j = 0; j < MA_NP; j++)
218             sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
219
220         lsfq[i] = sum >> 15;
221     }
222
223     ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
224 }
225
226 /**
227  * Restores past LSP quantizer output using LSF from previous frame
228  * @param[in,out] lsfq (2.13) quantized LSF coefficients
229  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
230  * @param ma_predictor_prev MA predictor from previous frame
231  * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
232  */
233 static void lsf_restore_from_previous(int16_t* lsfq,
234                                       int16_t* past_quantizer_outputs[MA_NP + 1],
235                                       int ma_predictor_prev)
236 {
237     int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
238     int i,k;
239
240     for (i = 0; i < 10; i++) {
241         int tmp = lsfq[i] << 15;
242
243         for (k = 0; k < MA_NP; k++)
244             tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
245
246         quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
247     }
248 }
249
250 /**
251  * Constructs new excitation signal and applies phase filter to it
252  * @param[out] out constructed speech signal
253  * @param in original excitation signal
254  * @param fc_cur (2.13) original fixed-codebook vector
255  * @param gain_code (14.1) gain code
256  * @param subframe_size length of the subframe
257  */
258 static void g729d_get_new_exc(
259         int16_t* out,
260         const int16_t* in,
261         const int16_t* fc_cur,
262         int dstate,
263         int gain_code,
264         int subframe_size)
265 {
266     int i;
267     int16_t fc_new[SUBFRAME_SIZE];
268
269     ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
270
271     for(i=0; i<subframe_size; i++)
272     {
273         out[i]  = in[i];
274         out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
275         out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
276     }
277 }
278
279 /**
280  * Makes decision about onset in current subframe
281  * @param past_onset decision result of previous subframe
282  * @param past_gain_code gain code of current and previous subframe
283  *
284  * @return onset decision result for current subframe
285  */
286 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
287 {
288     if((past_gain_code[0] >> 1) > past_gain_code[1])
289         return 2;
290     else
291         return FFMAX(past_onset-1, 0);
292 }
293
294 /**
295  * Makes decision about voice presence in current subframe
296  * @param onset onset level
297  * @param prev_voice_decision voice decision result from previous subframe
298  * @param past_gain_pitch pitch gain of current and previous subframes
299  *
300  * @return voice decision result for current subframe
301  */
302 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
303 {
304     int i, low_gain_pitch_cnt, voice_decision;
305
306     if(past_gain_pitch[0] >= 14745)      // 0.9
307         voice_decision = DECISION_VOICE;
308     else if (past_gain_pitch[0] <= 9830) // 0.6
309         voice_decision = DECISION_NOISE;
310     else
311         voice_decision = DECISION_INTERMEDIATE;
312
313     for(i=0, low_gain_pitch_cnt=0; i<6; i++)
314         if(past_gain_pitch[i] < 9830)
315             low_gain_pitch_cnt++;
316
317     if(low_gain_pitch_cnt > 2 && !onset)
318         voice_decision = DECISION_NOISE;
319
320     if(!onset && voice_decision > prev_voice_decision + 1)
321         voice_decision--;
322
323     if(onset && voice_decision < DECISION_VOICE)
324         voice_decision++;
325
326     return voice_decision;
327 }
328
329 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
330 {
331     int res = 0;
332
333     while (order--)
334         res += *v1++ * *v2++;
335
336     return res;
337 }
338
339 static av_cold int decoder_init(AVCodecContext * avctx)
340 {
341     G729Context* ctx = avctx->priv_data;
342     int i,k;
343
344     if (avctx->channels != 1) {
345         av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
346         return AVERROR(EINVAL);
347     }
348     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
349
350     /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
351     avctx->frame_size = SUBFRAME_SIZE << 1;
352
353     ctx->gain_coeff = 16384; // 1.0 in (1.14)
354
355     for (k = 0; k < MA_NP + 1; k++) {
356         ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
357         for (i = 1; i < 11; i++)
358             ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
359     }
360
361     ctx->lsp[0] = ctx->lsp_buf[0];
362     ctx->lsp[1] = ctx->lsp_buf[1];
363     memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
364
365     ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
366
367     ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
368
369     /* random seed initialization */
370     ctx->rand_value = 21845;
371
372     /* quantized prediction error */
373     for(i=0; i<4; i++)
374         ctx->quant_energy[i] = -14336; // -14 in (5.10)
375
376     ff_audiodsp_init(&ctx->adsp);
377     ctx->adsp.scalarproduct_int16 = scalarproduct_int16_c;
378
379     return 0;
380 }
381
382 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
383                         AVPacket *avpkt)
384 {
385     const uint8_t *buf = avpkt->data;
386     int buf_size       = avpkt->size;
387     int16_t *out_frame;
388     GetBitContext gb;
389     const G729FormatDescription *format;
390     int frame_erasure = 0;    ///< frame erasure detected during decoding
391     int bad_pitch = 0;        ///< parity check failed
392     int i;
393     int16_t *tmp;
394     G729Formats packet_type;
395     G729Context *ctx = avctx->priv_data;
396     int16_t lp[2][11];           // (3.12)
397     uint8_t ma_predictor;     ///< switched MA predictor of LSP quantizer
398     uint8_t quantizer_1st;    ///< first stage vector of quantizer
399     uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
400     uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
401
402     int pitch_delay_int[2];      // pitch delay, integer part
403     int pitch_delay_3x;          // pitch delay, multiplied by 3
404     int16_t fc[SUBFRAME_SIZE];   // fixed-codebook vector
405     int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
406     int j, ret;
407     int gain_before, gain_after;
408     int is_periodic = 0;         // whether one of the subframes is declared as periodic or not
409     AVFrame *frame = data;
410
411     frame->nb_samples = SUBFRAME_SIZE<<1;
412     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
413         return ret;
414     out_frame = (int16_t*) frame->data[0];
415
416     if (buf_size % 10 == 0) {
417         packet_type = FORMAT_G729_8K;
418         format = &format_g729_8k;
419         //Reset voice decision
420         ctx->onset = 0;
421         ctx->voice_decision = DECISION_VOICE;
422         av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
423     } else if (buf_size == 8) {
424         packet_type = FORMAT_G729D_6K4;
425         format = &format_g729d_6k4;
426         av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
427     } else {
428         av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
429         return AVERROR_INVALIDDATA;
430     }
431
432     for (i=0; i < buf_size; i++)
433         frame_erasure |= buf[i];
434     frame_erasure = !frame_erasure;
435
436     init_get_bits(&gb, buf, 8*buf_size);
437
438     ma_predictor     = get_bits(&gb, 1);
439     quantizer_1st    = get_bits(&gb, VQ_1ST_BITS);
440     quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
441     quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
442
443     if(frame_erasure)
444         lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
445                                   ctx->ma_predictor_prev);
446     else {
447         lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
448                    ma_predictor,
449                    quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
450         ctx->ma_predictor_prev = ma_predictor;
451     }
452
453     tmp = ctx->past_quantizer_outputs[MA_NP];
454     memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
455             MA_NP * sizeof(int16_t*));
456     ctx->past_quantizer_outputs[0] = tmp;
457
458     ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
459
460     ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
461
462     FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
463
464     for (i = 0; i < 2; i++) {
465         int gain_corr_factor;
466
467         uint8_t ac_index;      ///< adaptive codebook index
468         uint8_t pulses_signs;  ///< fixed-codebook vector pulse signs
469         int fc_indexes;        ///< fixed-codebook indexes
470         uint8_t gc_1st_index;  ///< gain codebook (first stage) index
471         uint8_t gc_2nd_index;  ///< gain codebook (second stage) index
472
473         ac_index      = get_bits(&gb, format->ac_index_bits[i]);
474         if(!i && format->parity_bit)
475             bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
476         fc_indexes    = get_bits(&gb, format->fc_indexes_bits);
477         pulses_signs  = get_bits(&gb, format->fc_signs_bits);
478         gc_1st_index  = get_bits(&gb, format->gc_1st_index_bits);
479         gc_2nd_index  = get_bits(&gb, format->gc_2nd_index_bits);
480
481         if (frame_erasure)
482             pitch_delay_3x   = 3 * ctx->pitch_delay_int_prev;
483         else if(!i) {
484             if (bad_pitch)
485                 pitch_delay_3x   = 3 * ctx->pitch_delay_int_prev;
486             else
487                 pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
488         } else {
489             int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
490                                           PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
491
492             if(packet_type == FORMAT_G729D_6K4)
493                 pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
494             else
495                 pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
496         }
497
498         /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
499         pitch_delay_int[i]  = (pitch_delay_3x + 1) / 3;
500         if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
501             av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
502             pitch_delay_int[i] = PITCH_DELAY_MAX;
503         }
504
505         if (frame_erasure) {
506             ctx->rand_value = g729_prng(ctx->rand_value);
507             fc_indexes   = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
508
509             ctx->rand_value = g729_prng(ctx->rand_value);
510             pulses_signs = ctx->rand_value;
511         }
512
513
514         memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
515         switch (packet_type) {
516             case FORMAT_G729_8K:
517                 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
518                                             ff_fc_4pulses_8bits_track_4,
519                                             fc_indexes, pulses_signs, 3, 3);
520                 break;
521             case FORMAT_G729D_6K4:
522                 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
523                                             ff_fc_2pulses_9bits_track2_gray,
524                                             fc_indexes, pulses_signs, 1, 4);
525                 break;
526         }
527
528         /*
529           This filter enhances harmonic components of the fixed-codebook vector to
530           improve the quality of the reconstructed speech.
531
532                      / fc_v[i],                                    i < pitch_delay
533           fc_v[i] = <
534                      \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
535         */
536         ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
537                                      fc + pitch_delay_int[i],
538                                      fc, 1 << 14,
539                                      av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
540                                      0, 14,
541                                      SUBFRAME_SIZE - pitch_delay_int[i]);
542
543         memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
544         ctx->past_gain_code[1] = ctx->past_gain_code[0];
545
546         if (frame_erasure) {
547             ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
548             ctx->past_gain_code[0]  = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
549
550             gain_corr_factor = 0;
551         } else {
552             if (packet_type == FORMAT_G729D_6K4) {
553                 ctx->past_gain_pitch[0]  = cb_gain_1st_6k4[gc_1st_index][0] +
554                                            cb_gain_2nd_6k4[gc_2nd_index][0];
555                 gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
556                                    cb_gain_2nd_6k4[gc_2nd_index][1];
557
558                 /* Without check below overflow can occur in ff_acelp_update_past_gain.
559                    It is not issue for G.729, because gain_corr_factor in it's case is always
560                    greater than 1024, while in G.729D it can be even zero. */
561                 gain_corr_factor = FFMAX(gain_corr_factor, 1024);
562 #ifndef G729_BITEXACT
563                 gain_corr_factor >>= 1;
564 #endif
565             } else {
566                 ctx->past_gain_pitch[0]  = cb_gain_1st_8k[gc_1st_index][0] +
567                                            cb_gain_2nd_8k[gc_2nd_index][0];
568                 gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
569                                    cb_gain_2nd_8k[gc_2nd_index][1];
570             }
571
572             /* Decode the fixed-codebook gain. */
573             ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->adsp, gain_corr_factor,
574                                                                fc, MR_ENERGY,
575                                                                ctx->quant_energy,
576                                                                ma_prediction_coeff,
577                                                                SUBFRAME_SIZE, 4);
578 #ifdef G729_BITEXACT
579             /*
580               This correction required to get bit-exact result with
581               reference code, because gain_corr_factor in G.729D is
582               two times larger than in original G.729.
583
584               If bit-exact result is not issue then gain_corr_factor
585               can be simpler divided by 2 before call to g729_get_gain_code
586               instead of using correction below.
587             */
588             if (packet_type == FORMAT_G729D_6K4) {
589                 gain_corr_factor >>= 1;
590                 ctx->past_gain_code[0] >>= 1;
591             }
592 #endif
593         }
594         ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
595
596         /* Routine requires rounding to lowest. */
597         ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
598                              ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
599                              ff_acelp_interp_filter, 6,
600                              (pitch_delay_3x % 3) << 1,
601                              10, SUBFRAME_SIZE);
602
603         ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
604                                      ctx->exc + i * SUBFRAME_SIZE, fc,
605                                      (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
606                                      ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
607                                      1 << 13, 14, SUBFRAME_SIZE);
608
609         memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
610
611         if (ff_celp_lp_synthesis_filter(
612             synth+10,
613             &lp[i][1],
614             ctx->exc  + i * SUBFRAME_SIZE,
615             SUBFRAME_SIZE,
616             10,
617             1,
618             0,
619             0x800))
620             /* Overflow occurred, downscale excitation signal... */
621             for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
622                 ctx->exc_base[j] >>= 2;
623
624         /* ... and make synthesis again. */
625         if (packet_type == FORMAT_G729D_6K4) {
626             int16_t exc_new[SUBFRAME_SIZE];
627
628             ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
629             ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
630
631             g729d_get_new_exc(exc_new, ctx->exc  + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
632
633             ff_celp_lp_synthesis_filter(
634                     synth+10,
635                     &lp[i][1],
636                     exc_new,
637                     SUBFRAME_SIZE,
638                     10,
639                     0,
640                     0,
641                     0x800);
642         } else {
643             ff_celp_lp_synthesis_filter(
644                     synth+10,
645                     &lp[i][1],
646                     ctx->exc  + i * SUBFRAME_SIZE,
647                     SUBFRAME_SIZE,
648                     10,
649                     0,
650                     0,
651                     0x800);
652         }
653         /* Save data (without postfilter) for use in next subframe. */
654         memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
655
656         /* Calculate gain of unfiltered signal for use in AGC. */
657         gain_before = 0;
658         for (j = 0; j < SUBFRAME_SIZE; j++)
659             gain_before += FFABS(synth[j+10]);
660
661         /* Call postfilter and also update voicing decision for use in next frame. */
662         ff_g729_postfilter(
663                 &ctx->adsp,
664                 &ctx->ht_prev_data,
665                 &is_periodic,
666                 &lp[i][0],
667                 pitch_delay_int[0],
668                 ctx->residual,
669                 ctx->res_filter_data,
670                 ctx->pos_filter_data,
671                 synth+10,
672                 SUBFRAME_SIZE);
673
674         /* Calculate gain of filtered signal for use in AGC. */
675         gain_after = 0;
676         for(j=0; j<SUBFRAME_SIZE; j++)
677             gain_after += FFABS(synth[j+10]);
678
679         ctx->gain_coeff = ff_g729_adaptive_gain_control(
680                 gain_before,
681                 gain_after,
682                 synth+10,
683                 SUBFRAME_SIZE,
684                 ctx->gain_coeff);
685
686         if (frame_erasure)
687             ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
688         else
689             ctx->pitch_delay_int_prev = pitch_delay_int[i];
690
691         memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
692         ff_acelp_high_pass_filter(
693                 out_frame + i*SUBFRAME_SIZE,
694                 ctx->hpf_f,
695                 synth+10,
696                 SUBFRAME_SIZE);
697         memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
698     }
699
700     ctx->was_periodic = is_periodic;
701
702     /* Save signal for use in next frame. */
703     memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
704
705     *got_frame_ptr = 1;
706     return packet_type == FORMAT_G729_8K ? 10 : 8;
707 }
708
709 AVCodec ff_g729_decoder = {
710     .name           = "g729",
711     .long_name      = NULL_IF_CONFIG_SMALL("G.729"),
712     .type           = AVMEDIA_TYPE_AUDIO,
713     .id             = AV_CODEC_ID_G729,
714     .priv_data_size = sizeof(G729Context),
715     .init           = decoder_init,
716     .decode         = decode_frame,
717     .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
718 };