mirror of https://git.ffmpeg.org/ffmpeg.git
318 lines
9.7 KiB
C
318 lines
9.7 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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* @remark FFmpeg merging spearheaded by Kenan Gillet
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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, const float *in_a,
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const float *in_b, float weight_coeff_a,
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float weight_coeff_b, int length)
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{
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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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* Initialize the speech codec according to the specification.
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*
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* TIA/EIA/IS-733 2.4.9
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*/
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static av_cold int qcelp_decode_init(AVCodecContext *avctx) {
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QCELPContext *q = avctx->priv_data;
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int i;
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avctx->sample_fmt = SAMPLE_FMT_FLT;
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for (i = 0; i < 10; i++)
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q->prev_lspf[i] = (i + 1) / 11.;
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return 0;
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}
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/**
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* Computes the scaled codebook vector Cdn From INDEX and GAIN
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* for all rates.
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*
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* The specification lacks some information here.
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*
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* TIA/EIA/IS-733 has an omission on the codebook index determination
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* formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
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* you have to subtract the decoded index parameter from the given scaled
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* codebook vector index 'n' to get the desired circular codebook index, but
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* it does not mention that you have to clamp 'n' to [0-9] in order to get
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* RI-compliant results.
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*
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* The reason for this mistake seems to be the fact they forgot to mention you
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* have to do these calculations per codebook subframe and adjust given
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* equation values accordingly.
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*
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* @param q the context
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* @param gain array holding the 4 pitch subframe gain values
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* @param cdn_vector array for the generated scaled codebook vector
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*/
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static void compute_svector(const QCELPContext *q,
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const float *gain,
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float *cdn_vector) {
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int i, j, k;
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uint16_t cbseed, cindex;
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float *rnd, tmp_gain, fir_filter_value;
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switch (q->framerate) {
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case RATE_FULL:
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for (i = 0; i < 16; i++) {
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tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
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cindex = -q->cindex[i];
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for (j = 0; j < 10; j++)
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*cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
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}
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break;
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case RATE_HALF:
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for (i = 0; i < 4; i++) {
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tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
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cindex = -q->cindex[i];
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for (j = 0; j < 40; j++)
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*cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
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}
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break;
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case RATE_QUARTER:
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cbseed = (0x0003 & q->lspv[4])<<14 |
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(0x003F & q->lspv[3])<< 8 |
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(0x0060 & q->lspv[2])<< 1 |
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(0x0007 & q->lspv[1])<< 3 |
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(0x0038 & q->lspv[0])>> 3 ;
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rnd = q->rnd_fir_filter_mem + 20;
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for (i = 0; i < 8; i++) {
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tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
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for (k = 0; k < 20; k++) {
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cbseed = 521 * cbseed + 259;
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*rnd = (int16_t)cbseed;
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// FIR filter
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fir_filter_value = 0.0;
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for (j = 0; j < 10; j++)
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fir_filter_value += qcelp_rnd_fir_coefs[j ] * (rnd[-j ] + rnd[-20+j]);
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fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
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*cdn_vector++ = tmp_gain * fir_filter_value;
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rnd++;
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}
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}
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memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
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break;
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case RATE_OCTAVE:
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cbseed = q->first16bits;
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for (i = 0; i < 8; i++) {
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tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
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for (j = 0; j < 20; j++) {
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cbseed = 521 * cbseed + 259;
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*cdn_vector++ = tmp_gain * (int16_t)cbseed;
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}
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}
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break;
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case I_F_Q:
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cbseed = -44; // random codebook index
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for (i = 0; i < 4; i++) {
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tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
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for (j = 0; j < 40; j++)
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*cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
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}
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break;
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}
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}
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/**
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* Apply generic gain control.
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*
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* @param v_out output vector
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* @param v_in gain-controlled vector
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* @param v_ref vector to control gain of
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*
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* FIXME: If v_ref is a zero vector, it energy is zero
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* and the behavior of the gain control is
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* undefined in the specs.
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*
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* TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
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*/
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static void apply_gain_ctrl(float *v_out,
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const float *v_ref,
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const float *v_in) {
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int i, j, len;
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float scalefactor;
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for (i = 0, j = 0; i < 4; i++) {
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scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
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if (scalefactor)
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scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40) / scalefactor);
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else
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av_log_missing_feature(NULL, "Zero energy for gain control", 1);
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for (len = j + 40; j < len; j++)
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v_out[j] = scalefactor * v_in[j];
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}
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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], const float v_in[160],
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const float gain[4], const uint8_t *lag,
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const uint8_t pfrac[4])
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{
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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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{
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if(gain[i])
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{
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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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{
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if(pfrac[i]) // If it is a fractional lag...
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{
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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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{
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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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}
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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, const float *curr_lspf, float *lpc,
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const int subframe_num)
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{
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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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if(weight != 1.0)
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{
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weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
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weight, 1.0 - weight, 10);
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qcelp_lspf2lpc(interpolated_lspf, lpc);
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}else if(q->framerate >= RATE_QUARTER || (q->framerate == I_F_Q && !subframe_num))
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qcelp_lspf2lpc(curr_lspf, lpc);
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}
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static int buf_size2framerate(const int buf_size)
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{
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switch(buf_size)
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{
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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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{
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av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
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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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