mirror of https://git.ffmpeg.org/ffmpeg.git
170 lines
5.2 KiB
C
170 lines
5.2 KiB
C
/*
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* QCELP decoder
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* Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
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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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/**
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* @file qcelpdec.c
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* QCELP decoder
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* @author Reynaldo H. Verdejo Pinochet
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*/
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#include <stddef.h>
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#include "avcodec.h"
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#include "bitstream.h"
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#include "qcelp.h"
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#include "qcelpdata.h"
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#include "celp_math.h"
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#include "celp_filters.h"
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#undef NDEBUG
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#include <assert.h>
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static void weighted_vector_sumf(float *out,
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const float *in_a,
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const float *in_b,
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float weight_coeff_a,
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float weight_coeff_b,
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int length) {
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int i;
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for (i = 0; i < length; i++)
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out[i] = weight_coeff_a * in_a[i]
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+ weight_coeff_b * in_b[i];
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}
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/**
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* Apply filter in pitch-subframe steps.
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*
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* @param memory buffer for the previous state of the filter
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* - must be able to contain 303 elements
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* - the 143 first elements are from the previous state
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* - the next 160 are for output
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* @param v_in input filter vector
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* @param gain per-subframe gain array, each element is between 0.0 and 2.0
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* @param lag per-subframe lag array, each element is
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* - between 16 and 143 if its corresponding pfrac is 0,
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* - between 16 and 139 otherwise
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* @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0 otherwise
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*
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* @return filter output vector
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*/
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static const float *do_pitchfilter(float memory[303],
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const float v_in[160],
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const float gain[4],
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const uint8_t *lag,
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const uint8_t pfrac[4]) {
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int i, j;
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float *v_lag, *v_out;
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const float *v_len;
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v_out = memory + 143; // Output vector starts at memory[143].
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for (i = 0; i < 4; i++)
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if (gain[i]) {
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v_lag = memory + 143 + 40 * i - lag[i];
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for (v_len = v_in + 40; v_in < v_len; v_in++) {
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if (pfrac[i]) { // If it is a fractional lag...
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for (j = 0, *v_out = 0.; j < 4; j++)
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*v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
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} else
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*v_out = *v_lag;
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*v_out = *v_in + gain[i] * *v_out;
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v_lag++;
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v_out++;
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}
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} else {
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memcpy(v_out, v_in, 40 * sizeof(float));
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v_in += 40;
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v_out += 40;
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}
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memmove(memory, memory + 160, 143 * sizeof(float));
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return memory + 143;
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}
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/**
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* Interpolates LSP frequencies and computes LPC coefficients
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* for a given framerate & pitch subframe.
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*
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* TIA/EIA/IS-733 2.4.3.3.4
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*
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* @param q the context
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* @param curr_lspf LSP frequencies vector of the current frame
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* @param lpc float vector for the resulting LPC
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* @param subframe_num frame number in decoded stream
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*/
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void interpolate_lpc(QCELPContext *q,
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const float *curr_lspf,
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float *lpc,
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const int subframe_num) {
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float interpolated_lspf[10];
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float weight;
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if (q->framerate >= RATE_QUARTER) {
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weight = 0.25 * (subframe_num + 1);
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} else if (q->framerate == RATE_OCTAVE && !subframe_num) {
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weight = 0.625;
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} else {
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weight = 1.0;
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}
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if (weight != 1.0) {
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weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf, weight, 1.0 - weight, 10);
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lspf2lpc(q, interpolated_lspf, lpc);
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} else if (q->framerate >= RATE_QUARTER || (q->framerate == I_F_Q && !subframe_num))
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lspf2lpc(q, curr_lspf, lpc);
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}
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static int buf_size2framerate(const int buf_size) {
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switch (buf_size) {
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case 35:
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return RATE_FULL;
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case 17:
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return RATE_HALF;
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case 8:
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return RATE_QUARTER;
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case 4:
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return RATE_OCTAVE;
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case 1:
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return SILENCE;
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}
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return -1;
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}
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static void warn_insufficient_frame_quality(AVCodecContext *avctx,
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const char *message) {
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av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number, message);
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}
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AVCodec qcelp_decoder =
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{
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.name = "qcelp",
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.type = CODEC_TYPE_AUDIO,
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.id = CODEC_ID_QCELP,
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.init = qcelp_decode_init,
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.decode = qcelp_decode_frame,
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.priv_data_size = sizeof(QCELPContext),
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.long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
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};
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