FFmpeg coverage

Line	Branch	Exec	Source
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 "libavutil/mem.h"
28			#include "get_bits.h"
29			#include "audiodsp.h"
30			#include "codec_internal.h"
31			#include "decode.h"
32
33
34			#include "g729.h"
35			#include "lsp.h"
36			#include "celp_filters.h"
37			#include "acelp_filters.h"
38			#include "acelp_pitch_delay.h"
39			#include "acelp_vectors.h"
40			#include "g729data.h"
41			#include "g729postfilter.h"
42
43			/**
44			* minimum quantized LSF value (3.2.4)
45			* 0.005 in Q13
46			*/
47			#define LSFQ_MIN 40
48
49			/**
50			* maximum quantized LSF value (3.2.4)
51			* 3.135 in Q13
52			*/
53			#define LSFQ_MAX 25681
54
55			/**
56			* minimum LSF distance (3.2.4)
57			* 0.0391 in Q13
58			*/
59			#define LSFQ_DIFF_MIN 321
60
61			/// interpolation filter length
62			#define INTERPOL_LEN 11
63
64			/**
65			* minimum gain pitch value (3.8, Equation 47)
66			* 0.2 in (1.14)
67			*/
68			#define SHARP_MIN 3277
69
70			/**
71			* maximum gain pitch value (3.8, Equation 47)
72			* (EE) This does not comply with the specification.
73			* Specification says about 0.8, which should be
74			* 13107 in (1.14), but reference C code uses
75			* 13017 (equals to 0.7945) instead of it.
76			*/
77			#define SHARP_MAX 13017
78
79			/**
80			* MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7.13)
81			*/
82			#define MR_ENERGY 1018156
83
84			#define DECISION_NOISE 0
85			#define DECISION_INTERMEDIATE 1
86			#define DECISION_VOICE 2
87
88			typedef enum {
89			FORMAT_G729_8K = 0,
90			FORMAT_G729D_6K4,
91			FORMAT_COUNT,
92			} G729Formats;
93
94			typedef struct {
95			uint8_t ac_index_bits[2]; ///< adaptive codebook index for second subframe (size in bits)
96			uint8_t parity_bit; ///< parity bit for pitch delay
97			uint8_t gc_1st_index_bits; ///< gain codebook (first stage) index (size in bits)
98			uint8_t gc_2nd_index_bits; ///< gain codebook (second stage) index (size in bits)
99			uint8_t fc_signs_bits; ///< number of pulses in fixed-codebook vector
100			uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook index entry
101			uint8_t block_size;
102			} G729FormatDescription;
103
104			typedef struct {
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			} G729ChannelContext;
156
157			typedef struct {
158			AudioDSPContext adsp;
159
160			G729ChannelContext *channel_context;
161			} G729Context;
162
163			static const G729FormatDescription format_g729_8k = {
164			.ac_index_bits = {8,5},
165			.parity_bit = 1,
166			.gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
167			.gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
168			.fc_signs_bits = 4,
169			.fc_indexes_bits = 13,
170			.block_size = G729_8K_BLOCK_SIZE,
171			};
172
173			static const G729FormatDescription format_g729d_6k4 = {
174			.ac_index_bits = {8,4},
175			.parity_bit = 0,
176			.gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
177			.gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
178			.fc_signs_bits = 2,
179			.fc_indexes_bits = 9,
180			.block_size = G729D_6K4_BLOCK_SIZE,
181			};
182
183			/**
184			* @brief pseudo random number generator
185			*/
186		✗	static inline uint16_t g729_prng(uint16_t value)
187			{
188		✗	return 31821 * value + 13849;
189			}
190
191			/**
192			* Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
193			* @param[out] lsfq (2.13) quantized LSF coefficients
194			* @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
195			* @param ma_predictor switched MA predictor of LSP quantizer
196			* @param vq_1st first stage vector of quantizer
197			* @param vq_2nd_low second stage lower vector of LSP quantizer
198			* @param vq_2nd_high second stage higher vector of LSP quantizer
199			*/
200		✗	static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
201			int16_t ma_predictor,
202			int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
203			{
204			int i,j;
205			static const uint8_t min_distance[2]={10, 5}; //(2.13)
206		✗	int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
207
208		✗	for (i = 0; i < 5; i++) {
209		✗	quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
210		✗	quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
211			}
212
213		✗	for (j = 0; j < 2; j++) {
214		✗	for (i = 1; i < 10; i++) {
215		✗	int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
216		✗	if (diff > 0) {
217		✗	quantizer_output[i - 1] -= diff;
218		✗	quantizer_output[i ] += diff;
219			}
220			}
221			}
222
223		✗	for (i = 0; i < 10; i++) {
224		✗	int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
225		✗	for (j = 0; j < MA_NP; j++)
226		✗	sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
227
228		✗	lsfq[i] = sum >> 15;
229			}
230
231		✗	ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
232		✗	}
233
234			/**
235			* Restores past LSP quantizer output using LSF from previous frame
236			* @param[in,out] lsfq (2.13) quantized LSF coefficients
237			* @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
238			* @param ma_predictor_prev MA predictor from previous frame
239			* @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
240			*/
241		✗	static void lsf_restore_from_previous(int16_t* lsfq,
242			int16_t* past_quantizer_outputs[MA_NP + 1],
243			int ma_predictor_prev)
244			{
245		✗	int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
246			int i,k;
247
248		✗	for (i = 0; i < 10; i++) {
249		✗	int tmp = lsfq[i] << 15;
250
251		✗	for (k = 0; k < MA_NP; k++)
252		✗	tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
253
254		✗	quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
255			}
256		✗	}
257
258			/**
259			* Constructs new excitation signal and applies phase filter to it
260			* @param[out] out constructed speech signal
261			* @param in original excitation signal
262			* @param fc_cur (2.13) original fixed-codebook vector
263			* @param gain_code (14.1) gain code
264			* @param subframe_size length of the subframe
265			*/
266		✗	static void g729d_get_new_exc(
267			int16_t* out,
268			const int16_t* in,
269			const int16_t* fc_cur,
270			int dstate,
271			int gain_code,
272			int subframe_size)
273			{
274			int i;
275			int16_t fc_new[SUBFRAME_SIZE];
276
277		✗	ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
278
279		✗	for (i = 0; i < subframe_size; i++) {
280		✗	out[i] = in[i];
281		✗	out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
282		✗	out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
283			}
284		✗	}
285
286			/**
287			* Makes decision about onset in current subframe
288			* @param past_onset decision result of previous subframe
289			* @param past_gain_code gain code of current and previous subframe
290			*
291			* @return onset decision result for current subframe
292			*/
293		✗	static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
294			{
295		✗	if ((past_gain_code[0] >> 1) > past_gain_code[1])
296		✗	return 2;
297
298		✗	return FFMAX(past_onset-1, 0);
299			}
300
301			/**
302			* Makes decision about voice presence in current subframe
303			* @param onset onset level
304			* @param prev_voice_decision voice decision result from previous subframe
305			* @param past_gain_pitch pitch gain of current and previous subframes
306			*
307			* @return voice decision result for current subframe
308			*/
309		✗	static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
310			{
311			int i, low_gain_pitch_cnt, voice_decision;
312
313		✗	if (past_gain_pitch[0] >= 14745) { // 0.9
314		✗	voice_decision = DECISION_VOICE;
315		✗	} else if (past_gain_pitch[0] <= 9830) { // 0.6
316		✗	voice_decision = DECISION_NOISE;
317			} else {
318		✗	voice_decision = DECISION_INTERMEDIATE;
319			}
320
321		✗	for (i = 0, low_gain_pitch_cnt = 0; i < 6; i++)
322		✗	if (past_gain_pitch[i] < 9830)
323		✗	low_gain_pitch_cnt++;
324
325		✗	if (low_gain_pitch_cnt > 2 && !onset)
326		✗	voice_decision = DECISION_NOISE;
327
328		✗	if (!onset && voice_decision > prev_voice_decision + 1)
329		✗	voice_decision--;
330
331		✗	if (onset && voice_decision < DECISION_VOICE)
332		✗	voice_decision++;
333
334		✗	return voice_decision;
335			}
336
337		✗	static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
338			{
339		✗	int64_t res = 0;
340
341		✗	while (order--)
342		✗	res += *v1++ * *v2++;
343
344		✗	if (res > INT32_MAX) return INT32_MAX;
345		✗	else if (res < INT32_MIN) return INT32_MIN;
346
347		✗	return res;
348			}
349
350		✗	static av_cold int decoder_init(AVCodecContext * avctx)
351			{
352		✗	G729Context *s = avctx->priv_data;
353			G729ChannelContext *ctx;
354		✗	int channels = avctx->ch_layout.nb_channels;
355			int c,i,k;
356
357		✗	if (channels < 1 || channels > 2) {
358		✗	av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", channels);
359		✗	return AVERROR(EINVAL);
360			}
361		✗	avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
362
363			/* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
364		✗	avctx->frame_size = SUBFRAME_SIZE << 1;
365
366		✗	ctx =
367		✗	s->channel_context = av_mallocz(sizeof(G729ChannelContext) * channels);
368		✗	if (!ctx)
369		✗	return AVERROR(ENOMEM);
370
371		✗	for (c = 0; c < channels; c++) {
372		✗	ctx->gain_coeff = 16384; // 1.0 in (1.14)
373
374		✗	for (k = 0; k < MA_NP + 1; k++) {
375		✗	ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
376		✗	for (i = 1; i < 11; i++)
377		✗	ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
378			}
379
380		✗	ctx->lsp[0] = ctx->lsp_buf[0];
381		✗	ctx->lsp[1] = ctx->lsp_buf[1];
382		✗	memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
383
384		✗	ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
385
386		✗	ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
387
388			/* random seed initialization */
389		✗	ctx->rand_value = 21845;
390
391			/* quantized prediction error */
392		✗	for (i = 0; i < 4; i++)
393		✗	ctx->quant_energy[i] = -14336; // -14 in (5.10)
394
395		✗	ctx++;
396			}
397
398		✗	ff_audiodsp_init(&s->adsp);
399		✗	s->adsp.scalarproduct_int16 = scalarproduct_int16_c;
400
401		✗	return 0;
402			}
403
404		✗	static int decode_frame(AVCodecContext *avctx, AVFrame *frame,
405			int *got_frame_ptr, AVPacket *avpkt)
406			{
407		✗	const uint8_t *buf = avpkt->data;
408		✗	int buf_size = avpkt->size;
409			int16_t *out_frame;
410			GetBitContext gb;
411			const G729FormatDescription *format;
412			int c, i;
413			int16_t *tmp;
414			G729Formats packet_type;
415		✗	G729Context *s = avctx->priv_data;
416		✗	G729ChannelContext *ctx = s->channel_context;
417		✗	int channels = avctx->ch_layout.nb_channels;
418			int16_t lp[2][11]; // (3.12)
419			uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
420			uint8_t quantizer_1st; ///< first stage vector of quantizer
421			uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
422			uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
423
424			int pitch_delay_int[2]; // pitch delay, integer part
425			int pitch_delay_3x; // pitch delay, multiplied by 3
426			int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
427			int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
428			int j, ret;
429			int gain_before, gain_after;
430
431		✗	frame->nb_samples = SUBFRAME_SIZE<<1;
432		✗	if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
433		✗	return ret;
434
435		✗	if (buf_size && buf_size % ((G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * channels) == 0) {
436		✗	packet_type = FORMAT_G729_8K;
437		✗	format = &format_g729_8k;
438			//Reset voice decision
439		✗	ctx->onset = 0;
440		✗	ctx->voice_decision = DECISION_VOICE;
441		✗	av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
442		✗	} else if (buf_size == G729D_6K4_BLOCK_SIZE * channels && avctx->codec_id != AV_CODEC_ID_ACELP_KELVIN) {
443		✗	packet_type = FORMAT_G729D_6K4;
444		✗	format = &format_g729d_6k4;
445		✗	av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
446			} else {
447		✗	av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
448		✗	return AVERROR_INVALIDDATA;
449			}
450
451		✗	for (c = 0; c < channels; c++) {
452		✗	int frame_erasure = 0; ///< frame erasure detected during decoding
453		✗	int bad_pitch = 0; ///< parity check failed
454		✗	int is_periodic = 0; ///< whether one of the subframes is declared as periodic or not
455		✗	out_frame = (int16_t*)frame->data[c];
456		✗	if (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN) {
457		✗	if (*buf != ((avctx->ch_layout.nb_channels - 1 - c) * 0x80 | 2))
458		✗	avpriv_request_sample(avctx, "First byte value %x for channel %d", *buf, c);
459		✗	buf++;
460			}
461
462		✗	for (i = 0; i < format->block_size; i++)
463		✗	frame_erasure |= buf[i];
464		✗	frame_erasure = !frame_erasure;
465
466		✗	init_get_bits8(&gb, buf, format->block_size);
467
468		✗	ma_predictor = get_bits(&gb, 1);
469		✗	quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
470		✗	quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
471		✗	quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
472
473		✗	if (frame_erasure) {
474		✗	lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
475			ctx->ma_predictor_prev);
476			} else {
477		✗	lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
478			ma_predictor,
479			quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
480		✗	ctx->ma_predictor_prev = ma_predictor;
481			}
482
483		✗	tmp = ctx->past_quantizer_outputs[MA_NP];
484		✗	memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
485			MA_NP * sizeof(int16_t*));
486		✗	ctx->past_quantizer_outputs[0] = tmp;
487
488		✗	ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
489
490		✗	ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
491
492		✗	FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
493
494		✗	for (i = 0; i < 2; i++) {
495			int gain_corr_factor;
496
497			uint8_t ac_index; ///< adaptive codebook index
498			uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
499			int fc_indexes; ///< fixed-codebook indexes
500			uint8_t gc_1st_index; ///< gain codebook (first stage) index
501			uint8_t gc_2nd_index; ///< gain codebook (second stage) index
502
503		✗	ac_index = get_bits(&gb, format->ac_index_bits[i]);
504		✗	if (!i && format->parity_bit)
505		✗	bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
506		✗	fc_indexes = get_bits(&gb, format->fc_indexes_bits);
507		✗	pulses_signs = get_bits(&gb, format->fc_signs_bits);
508		✗	gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
509		✗	gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
510
511		✗	if (frame_erasure) {
512		✗	pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
513		✗	} else if (!i) {
514		✗	if (bad_pitch) {
515		✗	pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
516			} else {
517		✗	pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
518			}
519			} else {
520		✗	int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
521			PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
522
523		✗	if (packet_type == FORMAT_G729D_6K4) {
524		✗	pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
525			} else {
526		✗	pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
527			}
528			}
529
530			/* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
531		✗	pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
532		✗	if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
533		✗	av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
534		✗	pitch_delay_int[i] = PITCH_DELAY_MAX;
535			}
536
537		✗	if (frame_erasure) {
538		✗	ctx->rand_value = g729_prng(ctx->rand_value);
539		✗	fc_indexes = av_zero_extend(ctx->rand_value, format->fc_indexes_bits);
540
541		✗	ctx->rand_value = g729_prng(ctx->rand_value);
542		✗	pulses_signs = ctx->rand_value;
543			}
544
545
546		✗	memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
547		✗	switch (packet_type) {
548		✗	case FORMAT_G729_8K:
549		✗	ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
550			ff_fc_4pulses_8bits_track_4,
551			fc_indexes, pulses_signs, 3, 3);
552		✗	break;
553		✗	case FORMAT_G729D_6K4:
554		✗	ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
555			ff_fc_2pulses_9bits_track2_gray,
556			fc_indexes, pulses_signs, 1, 4);
557		✗	break;
558			}
559
560			/*
561			This filter enhances harmonic components of the fixed-codebook vector to
562			improve the quality of the reconstructed speech.
563
564			/ fc_v[i], i < pitch_delay
565			fc_v[i] = <
566			\ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
567			*/
568		✗	if (SUBFRAME_SIZE > pitch_delay_int[i])
569		✗	ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
570		✗	fc + pitch_delay_int[i],
571			fc, 1 << 14,
572		✗	av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
573			0, 14,
574		✗	SUBFRAME_SIZE - pitch_delay_int[i]);
575
576		✗	memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
577		✗	ctx->past_gain_code[1] = ctx->past_gain_code[0];
578
579		✗	if (frame_erasure) {
580		✗	ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
581		✗	ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
582
583		✗	gain_corr_factor = 0;
584			} else {
585		✗	if (packet_type == FORMAT_G729D_6K4) {
586		✗	ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
587		✗	cb_gain_2nd_6k4[gc_2nd_index][0];
588		✗	gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
589		✗	cb_gain_2nd_6k4[gc_2nd_index][1];
590
591			/* Without check below overflow can occur in ff_acelp_update_past_gain.
592			It is not issue for G.729, because gain_corr_factor in it's case is always
593			greater than 1024, while in G.729D it can be even zero. */
594		✗	gain_corr_factor = FFMAX(gain_corr_factor, 1024);
595			#ifndef G729_BITEXACT
596		✗	gain_corr_factor >>= 1;
597			#endif
598			} else {
599		✗	ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
600		✗	cb_gain_2nd_8k[gc_2nd_index][0];
601		✗	gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
602		✗	cb_gain_2nd_8k[gc_2nd_index][1];
603			}
604
605			/* Decode the fixed-codebook gain. */
606		✗	ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&s->adsp, gain_corr_factor,
607			fc, MR_ENERGY,
608		✗	ctx->quant_energy,
609			ma_prediction_coeff,
610			SUBFRAME_SIZE, 4);
611			#ifdef G729_BITEXACT
612			/*
613			This correction required to get bit-exact result with
614			reference code, because gain_corr_factor in G.729D is
615			two times larger than in original G.729.
616
617			If bit-exact result is not issue then gain_corr_factor
618			can be simpler divided by 2 before call to g729_get_gain_code
619			instead of using correction below.
620			*/
621			if (packet_type == FORMAT_G729D_6K4) {
622			gain_corr_factor >>= 1;
623			ctx->past_gain_code[0] >>= 1;
624			}
625			#endif
626			}
627		✗	ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
628
629			/* Routine requires rounding to lowest. */
630		✗	ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
631		✗	ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
632			ff_acelp_interp_filter, 6,
633		✗	(pitch_delay_3x % 3) << 1,
634			10, SUBFRAME_SIZE);
635
636		✗	ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
637		✗	ctx->exc + i * SUBFRAME_SIZE, fc,
638		✗	(!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
639		✗	( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
640			1 << 13, 14, SUBFRAME_SIZE);
641
642		✗	memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
643
644		✗	if (ff_celp_lp_synthesis_filter(
645			synth+10,
646		✗	&lp[i][1],
647		✗	ctx->exc + i * SUBFRAME_SIZE,
648			SUBFRAME_SIZE,
649			10,
650			1,
651			0,
652			0x800))
653			/* Overflow occurred, downscale excitation signal... */
654		✗	for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
655		✗	ctx->exc_base[j] >>= 2;
656
657			/* ... and make synthesis again. */
658		✗	if (packet_type == FORMAT_G729D_6K4) {
659			int16_t exc_new[SUBFRAME_SIZE];
660
661		✗	ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
662		✗	ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
663
664		✗	g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
665
666		✗	ff_celp_lp_synthesis_filter(
667			synth+10,
668		✗	&lp[i][1],
669			exc_new,
670			SUBFRAME_SIZE,
671			10,
672			0,
673			0,
674			0x800);
675			} else {
676		✗	ff_celp_lp_synthesis_filter(
677			synth+10,
678		✗	&lp[i][1],
679		✗	ctx->exc + i * SUBFRAME_SIZE,
680			SUBFRAME_SIZE,
681			10,
682			0,
683			0,
684			0x800);
685			}
686			/* Save data (without postfilter) for use in next subframe. */
687		✗	memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
688
689			/* Calculate gain of unfiltered signal for use in AGC. */
690		✗	gain_before = 0;
691		✗	for (j = 0; j < SUBFRAME_SIZE; j++)
692		✗	gain_before += FFABS(synth[j+10]);
693
694			/* Call postfilter and also update voicing decision for use in next frame. */
695		✗	ff_g729_postfilter(
696			&s->adsp,
697			&ctx->ht_prev_data,
698			&is_periodic,
699		✗	&lp[i][0],
700			pitch_delay_int[0],
701		✗	ctx->residual,
702		✗	ctx->res_filter_data,
703		✗	ctx->pos_filter_data,
704			synth+10,
705			SUBFRAME_SIZE);
706
707			/* Calculate gain of filtered signal for use in AGC. */
708		✗	gain_after = 0;
709		✗	for (j = 0; j < SUBFRAME_SIZE; j++)
710		✗	gain_after += FFABS(synth[j+10]);
711
712		✗	ctx->gain_coeff = ff_g729_adaptive_gain_control(
713			gain_before,
714			gain_after,
715			synth+10,
716			SUBFRAME_SIZE,
717		✗	ctx->gain_coeff);
718
719		✗	if (frame_erasure) {
720		✗	ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
721			} else {
722		✗	ctx->pitch_delay_int_prev = pitch_delay_int[i];
723			}
724
725		✗	memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
726		✗	ff_acelp_high_pass_filter(
727		✗	out_frame + i*SUBFRAME_SIZE,
728		✗	ctx->hpf_f,
729			synth+10,
730			SUBFRAME_SIZE);
731		✗	memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
732			}
733
734		✗	ctx->was_periodic = is_periodic;
735
736			/* Save signal for use in next frame. */
737		✗	memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
738
739		✗	buf += format->block_size;
740		✗	ctx++;
741			}
742
743		✗	*got_frame_ptr = 1;
744		✗	return (format->block_size + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * channels;
745			}
746
747		✗	static av_cold int decode_close(AVCodecContext *avctx)
748			{
749		✗	G729Context *s = avctx->priv_data;
750		✗	av_freep(&s->channel_context);
751
752		✗	return 0;
753			}
754
755			const FFCodec ff_g729_decoder = {
756			.p.name = "g729",
757			CODEC_LONG_NAME("G.729"),
758			.p.type = AVMEDIA_TYPE_AUDIO,
759			.p.id = AV_CODEC_ID_G729,
760			.priv_data_size = sizeof(G729Context),
761			.init = decoder_init,
762			FF_CODEC_DECODE_CB(decode_frame),
763			.close = decode_close,
764			.p.capabilities =
765			#if FF_API_SUBFRAMES
766			AV_CODEC_CAP_SUBFRAMES |
767			#endif
768			AV_CODEC_CAP_DR1,
769			};
770
771			const FFCodec ff_acelp_kelvin_decoder = {
772			.p.name = "acelp.kelvin",
773			CODEC_LONG_NAME("Sipro ACELP.KELVIN"),
774			.p.type = AVMEDIA_TYPE_AUDIO,
775			.p.id = AV_CODEC_ID_ACELP_KELVIN,
776			.priv_data_size = sizeof(G729Context),
777			.init = decoder_init,
778			FF_CODEC_DECODE_CB(decode_frame),
779			.close = decode_close,
780			.p.capabilities =
781			#if FF_API_SUBFRAMES
782			AV_CODEC_CAP_SUBFRAMES |
783			#endif
784			AV_CODEC_CAP_DR1,
785			};
786