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FFmpeg/libavcodec/g729dec.c
Vladimir Voroshilov 50cad25625 Frame erasure support for pitch delay decoding
2011-09-24 21:11:00 +02:00

375 lines
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/*
* G.729 decoder
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#include <inttypes.h>
#include <limits.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <assert.h>
#include "avcodec.h"
#include "libavutil/avutil.h"
#include "get_bits.h"
#include "g729.h"
#include "lsp.h"
#include "celp_math.h"
#include "acelp_filters.h"
#include "acelp_pitch_delay.h"
#include "acelp_vectors.h"
#include "g729data.h"
/**
* minimum quantized LSF value (3.2.4)
* 0.005 in Q13
*/
#define LSFQ_MIN 40
/**
* maximum quantized LSF value (3.2.4)
* 3.135 in Q13
*/
#define LSFQ_MAX 25681
/**
* minimum LSF distance (3.2.4)
* 0.0391 in Q13
*/
#define LSFQ_DIFF_MIN 321
/**
* minimum gain pitch value (3.8, Equation 47)
* 0.2 in (1.14)
*/
#define SHARP_MIN 3277
/**
* maximum gain pitch value (3.8, Equation 47)
* (EE) This does not comply with the specification.
* Specification says about 0.8, which should be
* 13107 in (1.14), but reference C code uses
* 13017 (equals to 0.7945) instead of it.
*/
#define SHARP_MAX 13017
typedef struct {
uint8_t ac_index_bits[2]; ///< adaptive codebook index for second subframe (size in bits)
uint8_t parity_bit; ///< parity bit for pitch delay
uint8_t gc_1st_index_bits; ///< gain codebook (first stage) index (size in bits)
uint8_t gc_2nd_index_bits; ///< gain codebook (second stage) index (size in bits)
uint8_t fc_signs_bits; ///< number of pulses in fixed-codebook vector
uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook index entry
} G729FormatDescription;
typedef struct {
int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
/// (2.13) LSP quantizer outputs
int16_t past_quantizer_output_buf[MA_NP + 1][10];
int16_t* past_quantizer_outputs[MA_NP + 1];
int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
int16_t *lsp[2]; ///< pointers to lsp_buf
int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame
} G729Context;
static const G729FormatDescription format_g729_8k = {
.ac_index_bits = {8,5},
.parity_bit = 1,
.gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
.gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
.fc_signs_bits = 4,
.fc_indexes_bits = 13,
};
static const G729FormatDescription format_g729d_6k4 = {
.ac_index_bits = {8,4},
.parity_bit = 0,
.gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
.gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
.fc_signs_bits = 2,
.fc_indexes_bits = 9,
};
/**
* @brief pseudo random number generator
*/
static inline uint16_t g729_prng(uint16_t value)
{
return 31821 * value + 13849;
}
/**
* Get parity bit of bit 2..7
*/
static inline int get_parity(uint8_t value)
{
return (0x6996966996696996ULL >> (value >> 2)) & 1;
}
/*
* Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
* @param lsfq [out] (2.13) quantized LSF coefficients
* @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
* @param ma_predictor switched MA predictor of LSP quantizer
* @param vq_1st first stage vector of quantizer
* @param vq_2nd_low second stage lower vector of LSP quantizer
* @param vq_2nd_high second stage higher vector of LSP quantizer
*/
static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
int16_t ma_predictor,
int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
{
int i,j;
static const uint8_t min_distance[2]={10, 5}; //(2.13)
int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
for (i = 0; i < 5; i++) {
quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
}
for (j = 0; j < 2; j++) {
for (i = 1; i < 10; i++) {
int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
if (diff > 0) {
quantizer_output[i - 1] -= diff;
quantizer_output[i ] += diff;
}
}
}
for (i = 0; i < 10; i++) {
int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
for (j = 0; j < MA_NP; j++)
sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
lsfq[i] = sum >> 15;
}
ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
}
/**
* Restores past LSP quantizer output using LSF from previous frame
* @param lsfq [in/out] (2.13) quantized LSF coefficients
* @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
* @param ma_predictor_prev MA predictor from previous frame
* @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
*/
static void lsf_restore_from_previous(int16_t* lsfq,
int16_t* past_quantizer_outputs[MA_NP + 1],
int ma_predictor_prev)
{
int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
int i,k;
for (i = 0; i < 10; i++) {
int tmp = lsfq[i] << 15;
for (k = 0; k < MA_NP; k++)
tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
}
}
static av_cold int decoder_init(AVCodecContext * avctx)
{
G729Context* ctx = avctx->priv_data;
int i,k;
if (avctx->channels != 1) {
av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
return AVERROR(EINVAL);
}
/* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
avctx->frame_size = SUBFRAME_SIZE << 1;
for (k = 0; k < MA_NP + 1; k++) {
ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
for (i = 1; i < 11; i++)
ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
}
ctx->lsp[0] = ctx->lsp_buf[0];
ctx->lsp[1] = ctx->lsp_buf[1];
memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
return 0;
}
static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *out_frame = data;
GetBitContext gb;
G729FormatDescription format;
int frame_erasure = 0; ///< frame erasure detected during decoding
int bad_pitch = 0; ///< parity check failed
int i;
int16_t *tmp;
G729Context *ctx = avctx->priv_data;
int16_t lp[2][11]; // (3.12)
uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
uint8_t quantizer_1st; ///< first stage vector of quantizer
uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
int pitch_delay_int; // pitch delay, integer part
int pitch_delay_3x; // pitch delay, multiplied by 3
if (*data_size < SUBFRAME_SIZE << 2) {
av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer too small\n");
return AVERROR(EIO);
}
if (buf_size == 10) {
format = format_g729_8k;
av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
} else if (buf_size == 8) {
format = format_g729d_6k4;
av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
} else {
av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
return AVERROR_INVALIDDATA;
}
for (i=0; i < buf_size; i++)
frame_erasure |= buf[i];
frame_erasure = !frame_erasure;
init_get_bits(&gb, buf, buf_size);
ma_predictor = get_bits(&gb, 1);
quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
if(frame_erasure)
lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
ctx->ma_predictor_prev);
else {
lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
ma_predictor,
quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
ctx->ma_predictor_prev = ma_predictor;
}
tmp = ctx->past_quantizer_outputs[MA_NP];
memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
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++) {
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 = (pitch_delay_3x + 1) / 3;
ff_acelp_weighted_vector_sum(fc + pitch_delay_int,
fc + pitch_delay_int,
fc, 1 << 14,
av_clip(ctx->gain_pitch, SHARP_MIN, SHARP_MAX),
0, 14,
SUBFRAME_SIZE - pitch_delay_int);
if (frame_erasure) {
ctx->gain_pitch = (29491 * ctx->gain_pitch) >> 15; // 0.90 (0.15)
ctx->gain_code = ( 2007 * ctx->gain_code ) >> 11; // 0.98 (0.11)
gain_corr_factor = 0;
} else {
ctx->gain_pitch = 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];
ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
ctx->exc + i * SUBFRAME_SIZE, fc,
(!voicing && frame_erasure) ? 0 : ctx->gain_pitch,
( voicing && frame_erasure) ? 0 : ctx->gain_code,
1 << 13, 14, SUBFRAME_SIZE);
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;
}
*data_size = SUBFRAME_SIZE << 2;
return buf_size;
}
AVCodec ff_g729_decoder =
{
"g729",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_G729,
sizeof(G729Context),
decoder_init,
NULL,
NULL,
decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("G.729"),
};
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