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72415b2adb
is deprecated and will be dropped at the next major bump. Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk
327 lines
11 KiB
C
327 lines
11 KiB
C
/*
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* G.729 decoder
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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 <stdlib.h>
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#include <inttypes.h>
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#include <limits.h>
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#include <stdio.h>
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#include <string.h>
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#include <math.h>
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#include <assert.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 "g729.h"
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#include "lsp.h"
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#include "celp_math.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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/**
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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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/**
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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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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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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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} 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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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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/* Rotate past_quantizer_outputs. */
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memmove(past_quantizer_outputs + 1, past_quantizer_outputs, MA_NP * sizeof(int16_t*));
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past_quantizer_outputs[0] = quantizer_output;
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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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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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/* 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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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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return 0;
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}
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static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
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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 = data;
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GetBitContext gb;
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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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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; // pitch delay, integer part
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int pitch_delay_3x; // pitch delay, multiplied by 3
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if (*data_size < SUBFRAME_SIZE << 2) {
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av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer too small\n");
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return AVERROR(EIO);
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}
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if (buf_size == 10) {
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format = format_g729_8k;
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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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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, 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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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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ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
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ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
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FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
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for (i = 0; i < 2; i++) {
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uint8_t ac_index; ///< adaptive codebook index
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uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
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int fc_indexes; ///< fixed-codebook indexes
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uint8_t gc_1st_index; ///< gain codebook (first stage) index
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uint8_t gc_2nd_index; ///< gain codebook (second stage) index
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ac_index = get_bits(&gb, format.ac_index_bits[i]);
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if(!i && format.parity_bit)
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bad_pitch = get_parity(ac_index) == get_bits1(&gb);
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fc_indexes = get_bits(&gb, format.fc_indexes_bits);
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pulses_signs = get_bits(&gb, format.fc_signs_bits);
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gc_1st_index = get_bits(&gb, format.gc_1st_index_bits);
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gc_2nd_index = get_bits(&gb, format.gc_2nd_index_bits);
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if(!i) {
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if (bad_pitch)
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pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
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else
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pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
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} else {
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int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
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PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
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if(packet_type == FORMAT_G729D_6K4)
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pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
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else
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pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
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}
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/* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
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pitch_delay_int = (pitch_delay_3x + 1) / 3;
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ff_acelp_weighted_vector_sum(fc + pitch_delay_int,
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fc + pitch_delay_int,
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fc, 1 << 14,
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av_clip(ctx->gain_pitch, SHARP_MIN, SHARP_MAX),
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0, 14,
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SUBFRAME_SIZE - pitch_delay_int);
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if (frame_erasure) {
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ctx->gain_pitch = (29491 * ctx->gain_pitch) >> 15; // 0.90 (0.15)
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ctx->gain_code = ( 2007 * ctx->gain_code ) >> 11; // 0.98 (0.11)
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gain_corr_factor = 0;
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} else {
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ctx->gain_pitch = cb_gain_1st_8k[gc_1st_index][0] +
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cb_gain_2nd_8k[gc_2nd_index][0];
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gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
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cb_gain_2nd_8k[gc_2nd_index][1];
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ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
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ctx->exc + i * SUBFRAME_SIZE, fc,
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(!voicing && frame_erasure) ? 0 : ctx->gain_pitch,
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( voicing && frame_erasure) ? 0 : ctx->gain_code,
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1 << 13, 14, SUBFRAME_SIZE);
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ctx->pitch_delay_int_prev = pitch_delay_int;
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}
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*data_size = SUBFRAME_SIZE << 2;
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return buf_size;
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}
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AVCodec g729_decoder =
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{
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"g729",
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AVMEDIA_TYPE_AUDIO,
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CODEC_ID_G729,
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sizeof(G729Context),
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decoder_init,
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NULL,
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NULL,
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decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("G.729"),
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};
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