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 "get_bits.h"
28			#include "audiodsp.h"
29			#include "codec_internal.h"
30			#include "decode.h"
31
32
33			#include "g729.h"
34			#include "lsp.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			uint8_t block_size;
101			} G729FormatDescription;
102
103			typedef struct {
104			/// past excitation signal buffer
105			int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
106
107			int16_t* exc; ///< start of past excitation data in buffer
108			int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
109
110			/// (2.13) LSP quantizer outputs
111			int16_t past_quantizer_output_buf[MA_NP + 1][10];
112			int16_t* past_quantizer_outputs[MA_NP + 1];
113
114			int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
115			int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
116			int16_t *lsp[2]; ///< pointers to lsp_buf
117
118			int16_t quant_energy[4]; ///< (5.10) past quantized energy
119
120			/// previous speech data for LP synthesis filter
121			int16_t syn_filter_data[10];
122
123
124			/// residual signal buffer (used in long-term postfilter)
125			int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
126
127			/// previous speech data for residual calculation filter
128			int16_t res_filter_data[SUBFRAME_SIZE+10];
129
130			/// previous speech data for short-term postfilter
131			int16_t pos_filter_data[SUBFRAME_SIZE+10];
132
133			/// (1.14) pitch gain of current and five previous subframes
134			int16_t past_gain_pitch[6];
135
136			/// (14.1) gain code from current and previous subframe
137			int16_t past_gain_code[2];
138
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)
144			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)
147			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];
151
152			/// high-pass filter data (past output)
153			int16_t hpf_z[2];
154			} G729ChannelContext;
155
156			typedef struct {
157			AudioDSPContext adsp;
158
159			G729ChannelContext *channel_context;
160			} G729Context;
161
162			static const G729FormatDescription format_g729_8k = {
163			.ac_index_bits = {8,5},
164			.parity_bit = 1,
165			.gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
166			.gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
167			.fc_signs_bits = 4,
168			.fc_indexes_bits = 13,
169			.block_size = G729_8K_BLOCK_SIZE,
170			};
171
172			static const G729FormatDescription format_g729d_6k4 = {
173			.ac_index_bits = {8,4},
174			.parity_bit = 0,
175			.gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
176			.gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
177			.fc_signs_bits = 2,
178			.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
195			* @param vq_1st first stage vector of quantizer
196			* @param vq_2nd_low second stage lower vector of LSP quantizer
197			* @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,
201			int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
202			{
203			int i,j;
204			static const uint8_t min_distance[2]={10, 5}; //(2.13)
205		✗	int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
206
207		✗	for (i = 0; i < 5; i++) {
208		✗	quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
209		✗	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;
217		✗	quantizer_output[i ] += diff;
218			}
219			}
220			}
221
222		✗	for (i = 0; i < 10; i++) {
223		✗	int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
224		✗	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
238			* @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 channels = avctx->ch_layout.nb_channels;
354			int c,i,k;
355
356		✗	if (channels < 1 || channels > 2) {
357		✗	av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", channels);
358		✗	return AVERROR(EINVAL);
359			}
360		✗	avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
361
362			/* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
363		✗	avctx->frame_size = SUBFRAME_SIZE << 1;
364
365		✗	ctx =
366		✗	s->channel_context = av_mallocz(sizeof(G729ChannelContext) * channels);
367		✗	if (!ctx)
368		✗	return AVERROR(ENOMEM);
369
370		✗	for (c = 0; c < channels; c++) {
371		✗	ctx->gain_coeff = 16384; // 1.0 in (1.14)
372
373		✗	for (k = 0; k < MA_NP + 1; k++) {
374		✗	ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
375		✗	for (i = 1; i < 11; i++)
376		✗	ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
377			}
378
379		✗	ctx->lsp[0] = ctx->lsp_buf[0];
380		✗	ctx->lsp[1] = ctx->lsp_buf[1];
381		✗	memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
382
383		✗	ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
384
385		✗	ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
386
387			/* random seed initialization */
388		✗	ctx->rand_value = 21845;
389
390			/* quantized prediction error */
391		✗	for (i = 0; i < 4; i++)
392		✗	ctx->quant_energy[i] = -14336; // -14 in (5.10)
393
394		✗	ctx++;
395			}
396
397		✗	ff_audiodsp_init(&s->adsp);
398		✗	s->adsp.scalarproduct_int16 = scalarproduct_int16_c;
399
400		✗	return 0;
401			}
402
403		✗	static int decode_frame(AVCodecContext *avctx, AVFrame *frame,
404			int *got_frame_ptr, AVPacket *avpkt)
405			{
406		✗	const uint8_t *buf = avpkt->data;
407		✗	int buf_size = avpkt->size;
408			int16_t *out_frame;
409			GetBitContext gb;
410			const G729FormatDescription *format;
411			int c, i;
412			int16_t *tmp;
413			G729Formats packet_type;
414		✗	G729Context *s = avctx->priv_data;
415		✗	G729ChannelContext *ctx = s->channel_context;
416		✗	int channels = avctx->ch_layout.nb_channels;
417			int16_t lp[2][11]; // (3.12)
418			uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
419			uint8_t quantizer_1st; ///< first stage vector of quantizer
420			uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
421			uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
422
423			int pitch_delay_int[2]; // pitch delay, integer part
424			int pitch_delay_3x; // pitch delay, multiplied by 3
425			int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
426			int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
427			int j, ret;
428			int gain_before, gain_after;
429
430		✗	frame->nb_samples = SUBFRAME_SIZE<<1;
431		✗	if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
432		✗	return ret;
433
434		✗	if (buf_size && buf_size % ((G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * channels) == 0) {
435		✗	packet_type = FORMAT_G729_8K;
436		✗	format = &format_g729_8k;
437			//Reset voice decision
438		✗	ctx->onset = 0;
439		✗	ctx->voice_decision = DECISION_VOICE;
440		✗	av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
441		✗	} else if (buf_size == G729D_6K4_BLOCK_SIZE * channels && avctx->codec_id != AV_CODEC_ID_ACELP_KELVIN) {
442		✗	packet_type = FORMAT_G729D_6K4;
443		✗	format = &format_g729d_6k4;
444		✗	av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
445			} else {
446		✗	av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
447		✗	return AVERROR_INVALIDDATA;
448			}
449
450		✗	for (c = 0; c < channels; c++) {
451		✗	int frame_erasure = 0; ///< frame erasure detected during decoding
452		✗	int bad_pitch = 0; ///< parity check failed
453		✗	int is_periodic = 0; ///< whether one of the subframes is declared as periodic or not
454		✗	out_frame = (int16_t*)frame->data[c];
455		✗	if (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN) {
456		✗	if (*buf != ((avctx->ch_layout.nb_channels - 1 - c) * 0x80 | 2))
457		✗	avpriv_request_sample(avctx, "First byte value %x for channel %d", *buf, c);
458		✗	buf++;
459			}
460
461		✗	for (i = 0; i < format->block_size; i++)
462		✗	frame_erasure |= buf[i];
463		✗	frame_erasure = !frame_erasure;
464
465		✗	init_get_bits8(&gb, buf, format->block_size);
466
467		✗	ma_predictor = get_bits(&gb, 1);
468		✗	quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
469		✗	quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
470		✗	quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
471
472		✗	if (frame_erasure) {
473		✗	lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
474			ctx->ma_predictor_prev);
475			} else {
476		✗	lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
477			ma_predictor,
478			quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
479		✗	ctx->ma_predictor_prev = ma_predictor;
480			}
481
482		✗	tmp = ctx->past_quantizer_outputs[MA_NP];
483		✗	memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
484			MA_NP * sizeof(int16_t*));
485		✗	ctx->past_quantizer_outputs[0] = tmp;
486
487		✗	ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
488
489		✗	ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
490
491		✗	FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
492
493		✗	for (i = 0; i < 2; i++) {
494			int gain_corr_factor;
495
496			uint8_t ac_index; ///< adaptive codebook index
497			uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
498			int fc_indexes; ///< fixed-codebook indexes
499			uint8_t gc_1st_index; ///< gain codebook (first stage) index
500			uint8_t gc_2nd_index; ///< gain codebook (second stage) index
501
502		✗	ac_index = get_bits(&gb, format->ac_index_bits[i]);
503		✗	if (!i && format->parity_bit)
504		✗	bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
505		✗	fc_indexes = get_bits(&gb, format->fc_indexes_bits);
506		✗	pulses_signs = get_bits(&gb, format->fc_signs_bits);
507		✗	gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
508		✗	gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
509
510		✗	if (frame_erasure) {
511		✗	pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
512		✗	} else if (!i) {
513		✗	if (bad_pitch) {
514		✗	pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
515			} else {
516		✗	pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
517			}
518			} else {
519		✗	int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
520			PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
521
522		✗	if (packet_type == FORMAT_G729D_6K4) {
523		✗	pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
524			} else {
525		✗	pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
526			}
527			}
528
529			/* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
530		✗	pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
531		✗	if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
532		✗	av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
533		✗	pitch_delay_int[i] = PITCH_DELAY_MAX;
534			}
535
536		✗	if (frame_erasure) {
537		✗	ctx->rand_value = g729_prng(ctx->rand_value);
538		✗	fc_indexes = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
539
540		✗	ctx->rand_value = g729_prng(ctx->rand_value);
541		✗	pulses_signs = ctx->rand_value;
542			}
543
544
545		✗	memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
546		✗	switch (packet_type) {
547		✗	case FORMAT_G729_8K:
548		✗	ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
549			ff_fc_4pulses_8bits_track_4,
550			fc_indexes, pulses_signs, 3, 3);
551		✗	break;
552		✗	case FORMAT_G729D_6K4:
553		✗	ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
554			ff_fc_2pulses_9bits_track2_gray,
555			fc_indexes, pulses_signs, 1, 4);
556		✗	break;
557			}
558
559			/*
560			This filter enhances harmonic components of the fixed-codebook vector to
561			improve the quality of the reconstructed speech.
562
563			/ fc_v[i], i < pitch_delay
564			fc_v[i] = <
565			\ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
566			*/
567		✗	if (SUBFRAME_SIZE > pitch_delay_int[i])
568		✗	ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
569		✗	fc + pitch_delay_int[i],
570			fc, 1 << 14,
571		✗	av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
572			0, 14,
573		✗	SUBFRAME_SIZE - pitch_delay_int[i]);
574
575		✗	memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
576		✗	ctx->past_gain_code[1] = ctx->past_gain_code[0];
577
578		✗	if (frame_erasure) {
579		✗	ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
580		✗	ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
581
582		✗	gain_corr_factor = 0;
583			} else {
584		✗	if (packet_type == FORMAT_G729D_6K4) {
585		✗	ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
586		✗	cb_gain_2nd_6k4[gc_2nd_index][0];
587		✗	gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
588		✗	cb_gain_2nd_6k4[gc_2nd_index][1];
589
590			/* Without check below overflow can occur in ff_acelp_update_past_gain.
591			It is not issue for G.729, because gain_corr_factor in it's case is always
592			greater than 1024, while in G.729D it can be even zero. */
593		✗	gain_corr_factor = FFMAX(gain_corr_factor, 1024);
594			#ifndef G729_BITEXACT
595		✗	gain_corr_factor >>= 1;
596			#endif
597			} else {
598		✗	ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
599		✗	cb_gain_2nd_8k[gc_2nd_index][0];
600		✗	gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
601		✗	cb_gain_2nd_8k[gc_2nd_index][1];
602			}
603
604			/* Decode the fixed-codebook gain. */
605		✗	ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&s->adsp, gain_corr_factor,
606			fc, MR_ENERGY,
607		✗	ctx->quant_energy,
608			ma_prediction_coeff,
609			SUBFRAME_SIZE, 4);
610			#ifdef G729_BITEXACT
611			/*
612			This correction required to get bit-exact result with
613			reference code, because gain_corr_factor in G.729D is
614			two times larger than in original G.729.
615
616			If bit-exact result is not issue then gain_corr_factor
617			can be simpler divided by 2 before call to g729_get_gain_code
618			instead of using correction below.
619			*/
620			if (packet_type == FORMAT_G729D_6K4) {
621			gain_corr_factor >>= 1;
622			ctx->past_gain_code[0] >>= 1;
623			}
624			#endif
625			}
626		✗	ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
627
628			/* Routine requires rounding to lowest. */
629		✗	ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
630		✗	ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
631			ff_acelp_interp_filter, 6,
632		✗	(pitch_delay_3x % 3) << 1,
633			10, SUBFRAME_SIZE);
634
635		✗	ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
636		✗	ctx->exc + i * SUBFRAME_SIZE, fc,
637		✗	(!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
638		✗	( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
639			1 << 13, 14, SUBFRAME_SIZE);
640
641		✗	memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
642
643		✗	if (ff_celp_lp_synthesis_filter(
644			synth+10,
645		✗	&lp[i][1],
646		✗	ctx->exc + i * SUBFRAME_SIZE,
647			SUBFRAME_SIZE,
648			10,
649			1,
650			0,
651			0x800))
652			/* Overflow occurred, downscale excitation signal... */
653		✗	for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
654		✗	ctx->exc_base[j] >>= 2;
655
656			/* ... and make synthesis again. */
657		✗	if (packet_type == FORMAT_G729D_6K4) {
658			int16_t exc_new[SUBFRAME_SIZE];
659
660		✗	ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
661		✗	ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
662
663		✗	g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
664
665		✗	ff_celp_lp_synthesis_filter(
666			synth+10,
667		✗	&lp[i][1],
668			exc_new,
669			SUBFRAME_SIZE,
670			10,
671			0,
672			0,
673			0x800);
674			} else {
675		✗	ff_celp_lp_synthesis_filter(
676			synth+10,
677		✗	&lp[i][1],
678		✗	ctx->exc + i * SUBFRAME_SIZE,
679			SUBFRAME_SIZE,
680			10,
681			0,
682			0,
683			0x800);
684			}
685			/* Save data (without postfilter) for use in next subframe. */
686		✗	memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
687
688			/* Calculate gain of unfiltered signal for use in AGC. */
689		✗	gain_before = 0;
690		✗	for (j = 0; j < SUBFRAME_SIZE; j++)
691		✗	gain_before += FFABS(synth[j+10]);
692
693			/* Call postfilter and also update voicing decision for use in next frame. */
694		✗	ff_g729_postfilter(
695			&s->adsp,
696			&ctx->ht_prev_data,
697			&is_periodic,
698		✗	&lp[i][0],
699			pitch_delay_int[0],
700		✗	ctx->residual,
701		✗	ctx->res_filter_data,
702		✗	ctx->pos_filter_data,
703			synth+10,
704			SUBFRAME_SIZE);
705
706			/* Calculate gain of filtered signal for use in AGC. */
707		✗	gain_after = 0;
708		✗	for (j = 0; j < SUBFRAME_SIZE; j++)
709		✗	gain_after += FFABS(synth[j+10]);
710
711		✗	ctx->gain_coeff = ff_g729_adaptive_gain_control(
712			gain_before,
713			gain_after,
714			synth+10,
715			SUBFRAME_SIZE,
716		✗	ctx->gain_coeff);
717
718		✗	if (frame_erasure) {
719		✗	ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
720			} else {
721		✗	ctx->pitch_delay_int_prev = pitch_delay_int[i];
722			}
723
724		✗	memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
725		✗	ff_acelp_high_pass_filter(
726		✗	out_frame + i*SUBFRAME_SIZE,
727		✗	ctx->hpf_f,
728			synth+10,
729			SUBFRAME_SIZE);
730		✗	memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
731			}
732
733		✗	ctx->was_periodic = is_periodic;
734
735			/* Save signal for use in next frame. */
736		✗	memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
737
738		✗	buf += format->block_size;
739		✗	ctx++;
740			}
741
742		✗	*got_frame_ptr = 1;
743		✗	return (format->block_size + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * channels;
744			}
745
746		✗	static av_cold int decode_close(AVCodecContext *avctx)
747			{
748		✗	G729Context *s = avctx->priv_data;
749		✗	av_freep(&s->channel_context);
750
751		✗	return 0;
752			}
753
754			const FFCodec ff_g729_decoder = {
755			.p.name = "g729",
756			CODEC_LONG_NAME("G.729"),
757			.p.type = AVMEDIA_TYPE_AUDIO,
758			.p.id = AV_CODEC_ID_G729,
759			.priv_data_size = sizeof(G729Context),
760			.init = decoder_init,
761			FF_CODEC_DECODE_CB(decode_frame),
762			.close = decode_close,
763			.p.capabilities =
764			#if FF_API_SUBFRAMES
765			AV_CODEC_CAP_SUBFRAMES |
766			#endif
767			AV_CODEC_CAP_DR1,
768			};
769
770			const FFCodec ff_acelp_kelvin_decoder = {
771			.p.name = "acelp.kelvin",
772			CODEC_LONG_NAME("Sipro ACELP.KELVIN"),
773			.p.type = AVMEDIA_TYPE_AUDIO,
774			.p.id = AV_CODEC_ID_ACELP_KELVIN,
775			.priv_data_size = sizeof(G729Context),
776			.init = decoder_init,
777			FF_CODEC_DECODE_CB(decode_frame),
778			.close = decode_close,
779			.p.capabilities =
780			#if FF_API_SUBFRAMES
781			AV_CODEC_CAP_SUBFRAMES |
782			#endif
783			AV_CODEC_CAP_DR1,
784			};
785