forked from KolibriOS/kolibrios
727 lines
26 KiB
C
727 lines
26 KiB
C
/*
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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>
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#include <string.h>
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#include "avcodec.h"
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#include "libavutil/avutil.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "internal.h"
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#include "g729.h"
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#include "lsp.h"
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#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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/**
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* minimum quantized LSF value (3.2.4)
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* 0.005 in Q13
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*/
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#define LSFQ_MIN 40
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/**
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* 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
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/**
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* minimum LSF distance (3.2.4)
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* 0.0391 in Q13
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*/
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#define LSFQ_DIFF_MIN 321
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/// interpolation filter length
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#define INTERPOL_LEN 11
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/**
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* minimum gain pitch value (3.8, Equation 47)
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* 0.2 in (1.14)
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*/
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#define SHARP_MIN 3277
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/**
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* maximum gain pitch value (3.8, Equation 47)
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* (EE) This does not comply with the specification.
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* Specification says about 0.8, which should be
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* 13107 in (1.14), but reference C code uses
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* 13017 (equals to 0.7945) instead of it.
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*/
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#define SHARP_MAX 13017
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/**
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* MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7.13)
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*/
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#define MR_ENERGY 1018156
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#define DECISION_NOISE 0
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#define DECISION_INTERMEDIATE 1
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#define DECISION_VOICE 2
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typedef enum {
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FORMAT_G729_8K = 0,
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FORMAT_G729D_6K4,
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FORMAT_COUNT,
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} G729Formats;
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typedef struct {
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uint8_t ac_index_bits[2]; ///< adaptive codebook index for second subframe (size in bits)
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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)
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uint8_t gc_2nd_index_bits; ///< gain codebook (second stage) index (size in bits)
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uint8_t fc_signs_bits; ///< number of pulses in fixed-codebook vector
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uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook index entry
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} G729FormatDescription;
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typedef struct {
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DSPContext dsp;
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/// 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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int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
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int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
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int16_t *lsp[2]; ///< pointers to lsp_buf
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int16_t quant_energy[4]; ///< (5.10) past quantized energy
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/// previous speech data for LP synthesis filter
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int16_t syn_filter_data[10];
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/// residual signal buffer (used in long-term postfilter)
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int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
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/// previous speech data for residual calculation filter
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int16_t res_filter_data[SUBFRAME_SIZE+10];
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/// previous speech data for short-term postfilter
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int16_t pos_filter_data[SUBFRAME_SIZE+10];
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/// (1.14) pitch gain of current and five previous subframes
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int16_t past_gain_pitch[6];
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/// (14.1) gain code from current and previous subframe
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int16_t past_gain_code[2];
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/// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
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int16_t voice_decision;
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int16_t onset; ///< detected onset level (0-2)
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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
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int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
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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
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/// (14.14) high-pass filter data (past input)
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int hpf_f[2];
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/// high-pass filter data (past output)
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int16_t hpf_z[2];
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} 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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};
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static const G729FormatDescription format_g729d_6k4 = {
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.ac_index_bits = {8,4},
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.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,
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};
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/**
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* @brief pseudo random number generator
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*/
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static inline uint16_t g729_prng(uint16_t value)
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{
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return 31821 * value + 13849;
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}
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/**
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* Get parity bit of bit 2..7
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*/
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static inline int get_parity(uint8_t value)
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{
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return (0x6996966996696996ULL >> (value >> 2)) & 1;
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}
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/**
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* Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
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* @param[out] lsfq (2.13) quantized LSF coefficients
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* @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
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* @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
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*/
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static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
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int16_t ma_predictor,
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int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
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{
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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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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];
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}
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for (j = 0; j < 2; j++) {
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for (i = 1; i < 10; i++) {
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int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
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if (diff > 0) {
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quantizer_output[i - 1] -= diff;
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quantizer_output[i ] += diff;
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}
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}
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}
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for (i = 0; i < 10; i++) {
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int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
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for (j = 0; j < MA_NP; j++)
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sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
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lsfq[i] = sum >> 15;
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}
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ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
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}
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/**
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* Restores past LSP quantizer output using LSF from previous frame
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* @param[in,out] lsfq (2.13) quantized LSF coefficients
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* @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
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* @param ma_predictor_prev MA predictor from previous frame
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* @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
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*/
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static void lsf_restore_from_previous(int16_t* lsfq,
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int16_t* past_quantizer_outputs[MA_NP + 1],
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int ma_predictor_prev)
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{
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int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
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int i,k;
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for (i = 0; i < 10; i++) {
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int tmp = lsfq[i] << 15;
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for (k = 0; k < MA_NP; k++)
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tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
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quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
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}
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}
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/**
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* Constructs new excitation signal and applies phase filter to it
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* @param[out] out constructed speech signal
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* @param in original excitation signal
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* @param fc_cur (2.13) original fixed-codebook vector
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* @param gain_code (14.1) gain code
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* @param subframe_size length of the subframe
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*/
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static void g729d_get_new_exc(
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int16_t* out,
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const int16_t* in,
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const int16_t* fc_cur,
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int dstate,
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int gain_code,
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int subframe_size)
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{
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int i;
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int16_t fc_new[SUBFRAME_SIZE];
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ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
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for(i=0; i<subframe_size; i++)
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{
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out[i] = in[i];
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out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
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out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
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}
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}
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/**
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* Makes decision about onset in current subframe
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* @param past_onset decision result of previous subframe
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* @param past_gain_code gain code of current and previous subframe
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*
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* @return onset decision result for current subframe
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*/
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static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
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{
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if((past_gain_code[0] >> 1) > past_gain_code[1])
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return 2;
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else
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return FFMAX(past_onset-1, 0);
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}
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/**
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* Makes decision about voice presence in current subframe
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* @param onset onset level
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* @param prev_voice_decision voice decision result from previous subframe
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* @param past_gain_pitch pitch gain of current and previous subframes
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*
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* @return voice decision result for current subframe
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*/
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static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
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{
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int i, low_gain_pitch_cnt, voice_decision;
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if(past_gain_pitch[0] >= 14745) // 0.9
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voice_decision = DECISION_VOICE;
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else if (past_gain_pitch[0] <= 9830) // 0.6
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voice_decision = DECISION_NOISE;
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else
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voice_decision = DECISION_INTERMEDIATE;
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for(i=0, low_gain_pitch_cnt=0; i<6; i++)
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if(past_gain_pitch[i] < 9830)
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low_gain_pitch_cnt++;
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if(low_gain_pitch_cnt > 2 && !onset)
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voice_decision = DECISION_NOISE;
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if(!onset && voice_decision > prev_voice_decision + 1)
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voice_decision--;
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if(onset && voice_decision < DECISION_VOICE)
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voice_decision++;
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return voice_decision;
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}
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static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
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{
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int res = 0;
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while (order--)
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res += *v1++ * *v2++;
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return res;
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}
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static av_cold int decoder_init(AVCodecContext * avctx)
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{
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G729Context* ctx = avctx->priv_data;
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int i,k;
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if (avctx->channels != 1) {
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av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
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return AVERROR(EINVAL);
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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/* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
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avctx->frame_size = SUBFRAME_SIZE << 1;
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ctx->gain_coeff = 16384; // 1.0 in (1.14)
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for (k = 0; k < MA_NP + 1; k++) {
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ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
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for (i = 1; i < 11; i++)
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ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
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}
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ctx->lsp[0] = ctx->lsp_buf[0];
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ctx->lsp[1] = ctx->lsp_buf[1];
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memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
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ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
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ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
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/* random seed initialization */
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ctx->rand_value = 21845;
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/* quantized prediction error */
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for(i=0; i<4; i++)
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ctx->quant_energy[i] = -14336; // -14 in (5.10)
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ff_dsputil_init(&ctx->dsp, avctx);
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ctx->dsp.scalarproduct_int16 = scalarproduct_int16_c;
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return 0;
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}
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static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
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AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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int16_t *out_frame;
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GetBitContext gb;
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const G729FormatDescription *format;
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int frame_erasure = 0; ///< frame erasure detected during decoding
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int bad_pitch = 0; ///< parity check failed
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int i;
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int16_t *tmp;
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G729Formats packet_type;
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G729Context *ctx = avctx->priv_data;
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int16_t lp[2][11]; // (3.12)
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uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
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uint8_t quantizer_1st; ///< first stage vector of quantizer
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uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
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uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
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int pitch_delay_int[2]; // pitch delay, integer part
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int pitch_delay_3x; // pitch delay, multiplied by 3
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int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
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int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
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int j, ret;
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int gain_before, gain_after;
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int is_periodic = 0; // whether one of the subframes is declared as periodic or not
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AVFrame *frame = data;
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frame->nb_samples = SUBFRAME_SIZE<<1;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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out_frame = (int16_t*) frame->data[0];
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if (buf_size == 10) {
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packet_type = FORMAT_G729_8K;
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format = &format_g729_8k;
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//Reset voice decision
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ctx->onset = 0;
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ctx->voice_decision = DECISION_VOICE;
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av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
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} else if (buf_size == 8) {
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packet_type = FORMAT_G729D_6K4;
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format = &format_g729d_6k4;
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av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
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} else {
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av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
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return AVERROR_INVALIDDATA;
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}
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for (i=0; i < buf_size; i++)
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frame_erasure |= buf[i];
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frame_erasure = !frame_erasure;
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init_get_bits(&gb, buf, 8*buf_size);
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ma_predictor = get_bits(&gb, 1);
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quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
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quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
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quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
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if(frame_erasure)
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lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
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ctx->ma_predictor_prev);
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else {
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lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
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ma_predictor,
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quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
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ctx->ma_predictor_prev = ma_predictor;
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}
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tmp = ctx->past_quantizer_outputs[MA_NP];
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memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
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MA_NP * sizeof(int16_t*));
|
|
ctx->past_quantizer_outputs[0] = tmp;
|
|
|
|
ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
|
|
|
|
ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
|
|
|
|
FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
|
|
|
|
for (i = 0; i < 2; i++) {
|
|
int gain_corr_factor;
|
|
|
|
uint8_t ac_index; ///< adaptive codebook index
|
|
uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
|
|
int fc_indexes; ///< fixed-codebook indexes
|
|
uint8_t gc_1st_index; ///< gain codebook (first stage) index
|
|
uint8_t gc_2nd_index; ///< gain codebook (second stage) index
|
|
|
|
ac_index = get_bits(&gb, format->ac_index_bits[i]);
|
|
if(!i && format->parity_bit)
|
|
bad_pitch = get_parity(ac_index) == get_bits1(&gb);
|
|
fc_indexes = get_bits(&gb, format->fc_indexes_bits);
|
|
pulses_signs = get_bits(&gb, format->fc_signs_bits);
|
|
gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
|
|
gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
|
|
|
|
if (frame_erasure)
|
|
pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
|
|
else if(!i) {
|
|
if (bad_pitch)
|
|
pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
|
|
else
|
|
pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
|
|
} else {
|
|
int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
|
|
PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
|
|
|
|
if(packet_type == FORMAT_G729D_6K4)
|
|
pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
|
|
else
|
|
pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
|
|
}
|
|
|
|
/* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
|
|
pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
|
|
if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
|
|
av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
|
|
pitch_delay_int[i] = PITCH_DELAY_MAX;
|
|
}
|
|
|
|
if (frame_erasure) {
|
|
ctx->rand_value = g729_prng(ctx->rand_value);
|
|
fc_indexes = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1);
|
|
|
|
ctx->rand_value = g729_prng(ctx->rand_value);
|
|
pulses_signs = ctx->rand_value;
|
|
}
|
|
|
|
|
|
memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
|
|
switch (packet_type) {
|
|
case FORMAT_G729_8K:
|
|
ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
|
|
ff_fc_4pulses_8bits_track_4,
|
|
fc_indexes, pulses_signs, 3, 3);
|
|
break;
|
|
case FORMAT_G729D_6K4:
|
|
ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
|
|
ff_fc_2pulses_9bits_track2_gray,
|
|
fc_indexes, pulses_signs, 1, 4);
|
|
break;
|
|
}
|
|
|
|
/*
|
|
This filter enhances harmonic components of the fixed-codebook vector to
|
|
improve the quality of the reconstructed speech.
|
|
|
|
/ fc_v[i], i < pitch_delay
|
|
fc_v[i] = <
|
|
\ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
|
|
*/
|
|
ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
|
|
fc + pitch_delay_int[i],
|
|
fc, 1 << 14,
|
|
av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
|
|
0, 14,
|
|
SUBFRAME_SIZE - pitch_delay_int[i]);
|
|
|
|
memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
|
|
ctx->past_gain_code[1] = ctx->past_gain_code[0];
|
|
|
|
if (frame_erasure) {
|
|
ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
|
|
ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
|
|
|
|
gain_corr_factor = 0;
|
|
} else {
|
|
if (packet_type == FORMAT_G729D_6K4) {
|
|
ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
|
|
cb_gain_2nd_6k4[gc_2nd_index][0];
|
|
gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
|
|
cb_gain_2nd_6k4[gc_2nd_index][1];
|
|
|
|
/* Without check below overflow can occur in ff_acelp_update_past_gain.
|
|
It is not issue for G.729, because gain_corr_factor in it's case is always
|
|
greater than 1024, while in G.729D it can be even zero. */
|
|
gain_corr_factor = FFMAX(gain_corr_factor, 1024);
|
|
#ifndef G729_BITEXACT
|
|
gain_corr_factor >>= 1;
|
|
#endif
|
|
} else {
|
|
ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
|
|
cb_gain_2nd_8k[gc_2nd_index][0];
|
|
gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
|
|
cb_gain_2nd_8k[gc_2nd_index][1];
|
|
}
|
|
|
|
/* Decode the fixed-codebook gain. */
|
|
ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
|
|
fc, MR_ENERGY,
|
|
ctx->quant_energy,
|
|
ma_prediction_coeff,
|
|
SUBFRAME_SIZE, 4);
|
|
#ifdef G729_BITEXACT
|
|
/*
|
|
This correction required to get bit-exact result with
|
|
reference code, because gain_corr_factor in G.729D is
|
|
two times larger than in original G.729.
|
|
|
|
If bit-exact result is not issue then gain_corr_factor
|
|
can be simpler divided by 2 before call to g729_get_gain_code
|
|
instead of using correction below.
|
|
*/
|
|
if (packet_type == FORMAT_G729D_6K4) {
|
|
gain_corr_factor >>= 1;
|
|
ctx->past_gain_code[0] >>= 1;
|
|
}
|
|
#endif
|
|
}
|
|
ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
|
|
|
|
/* Routine requires rounding to lowest. */
|
|
ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
|
|
ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
|
|
ff_acelp_interp_filter, 6,
|
|
(pitch_delay_3x % 3) << 1,
|
|
10, SUBFRAME_SIZE);
|
|
|
|
ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
|
|
ctx->exc + i * SUBFRAME_SIZE, fc,
|
|
(!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
|
|
( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
|
|
1 << 13, 14, SUBFRAME_SIZE);
|
|
|
|
memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
|
|
|
|
if (ff_celp_lp_synthesis_filter(
|
|
synth+10,
|
|
&lp[i][1],
|
|
ctx->exc + i * SUBFRAME_SIZE,
|
|
SUBFRAME_SIZE,
|
|
10,
|
|
1,
|
|
0,
|
|
0x800))
|
|
/* Overflow occurred, downscale excitation signal... */
|
|
for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
|
|
ctx->exc_base[j] >>= 2;
|
|
|
|
/* ... and make synthesis again. */
|
|
if (packet_type == FORMAT_G729D_6K4) {
|
|
int16_t exc_new[SUBFRAME_SIZE];
|
|
|
|
ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
|
|
ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
|
|
|
|
g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
|
|
|
|
ff_celp_lp_synthesis_filter(
|
|
synth+10,
|
|
&lp[i][1],
|
|
exc_new,
|
|
SUBFRAME_SIZE,
|
|
10,
|
|
0,
|
|
0,
|
|
0x800);
|
|
} else {
|
|
ff_celp_lp_synthesis_filter(
|
|
synth+10,
|
|
&lp[i][1],
|
|
ctx->exc + i * SUBFRAME_SIZE,
|
|
SUBFRAME_SIZE,
|
|
10,
|
|
0,
|
|
0,
|
|
0x800);
|
|
}
|
|
/* Save data (without postfilter) for use in next subframe. */
|
|
memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
|
|
|
|
/* Calculate gain of unfiltered signal for use in AGC. */
|
|
gain_before = 0;
|
|
for (j = 0; j < SUBFRAME_SIZE; j++)
|
|
gain_before += FFABS(synth[j+10]);
|
|
|
|
/* Call postfilter and also update voicing decision for use in next frame. */
|
|
ff_g729_postfilter(
|
|
&ctx->dsp,
|
|
&ctx->ht_prev_data,
|
|
&is_periodic,
|
|
&lp[i][0],
|
|
pitch_delay_int[0],
|
|
ctx->residual,
|
|
ctx->res_filter_data,
|
|
ctx->pos_filter_data,
|
|
synth+10,
|
|
SUBFRAME_SIZE);
|
|
|
|
/* Calculate gain of filtered signal for use in AGC. */
|
|
gain_after = 0;
|
|
for(j=0; j<SUBFRAME_SIZE; j++)
|
|
gain_after += FFABS(synth[j+10]);
|
|
|
|
ctx->gain_coeff = ff_g729_adaptive_gain_control(
|
|
gain_before,
|
|
gain_after,
|
|
synth+10,
|
|
SUBFRAME_SIZE,
|
|
ctx->gain_coeff);
|
|
|
|
if (frame_erasure)
|
|
ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
|
|
else
|
|
ctx->pitch_delay_int_prev = pitch_delay_int[i];
|
|
|
|
memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
|
|
ff_acelp_high_pass_filter(
|
|
out_frame + i*SUBFRAME_SIZE,
|
|
ctx->hpf_f,
|
|
synth+10,
|
|
SUBFRAME_SIZE);
|
|
memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
|
|
}
|
|
|
|
ctx->was_periodic = is_periodic;
|
|
|
|
/* Save signal for use in next frame. */
|
|
memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
|
|
|
|
*got_frame_ptr = 1;
|
|
return buf_size;
|
|
}
|
|
|
|
AVCodec ff_g729_decoder = {
|
|
.name = "g729",
|
|
.long_name = NULL_IF_CONFIG_SMALL("G.729"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_G729,
|
|
.priv_data_size = sizeof(G729Context),
|
|
.init = decoder_init,
|
|
.decode = decode_frame,
|
|
.capabilities = CODEC_CAP_DR1,
|
|
};
|