Mercurial > libavcodec.hg
view acelp_filters.c @ 6846:4346441afca6 libavcodec
adaptive and fixed codebook vector operations for ACELP-based codecs
author | voroshil |
---|---|
date | Fri, 23 May 2008 03:10:27 +0000 |
parents | 1f02f929b9ff |
children | 94465a2c3b34 |
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/* * various filters for ACELP-based codecs * * 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 <inttypes.h> #include "avcodec.h" #include "acelp_filters.h" #define FRAC_BITS 13 #include "mathops.h" void ff_acelp_convolve_circ( int16_t* fc_out, const int16_t* fc_in, const int16_t* filter, int subframe_size) { int i, k; memset(fc_out, 0, subframe_size * sizeof(int16_t)); /* Since there are few pulses over an entire subframe (i.e. almost all fc_in[i] are zero) it is faster to swap two loops and process non-zero samples only. In the case of G.729D the buffer contains two non-zero samples before the call to ff_acelp_enhance_harmonics and, due to pitch_delay being bounded by [20; 143], a maximum of four non-zero samples for a total of 40 after the call. */ for(i=0; i<subframe_size; i++) { if(fc_in[i]) { for(k=0; k<i; k++) fc_out[k] += (fc_in[i] * filter[subframe_size + k - i]) >> 15; for(k=i; k<subframe_size; k++) fc_out[k] += (fc_in[i] * filter[k - i]) >> 15; } } } int ff_acelp_lp_synthesis_filter( int16_t *out, const int16_t* filter_coeffs, const int16_t* in, int buffer_length, int filter_length, int stop_on_overflow) { int i,n; for(n=0; n<buffer_length; n++) { int sum = 0x800; for(i=1; i<filter_length; i++) sum -= filter_coeffs[i] * out[n-i]; sum = (sum >> 12) + in[n]; /* Check for overflow */ if(sum + 0x8000 > 0xFFFFU) { if(stop_on_overflow) return 1; sum = (sum >> 31) ^ 32767; } out[n] = sum; } return 0; } void ff_acelp_weighted_filter( int16_t *out, const int16_t* in, const int16_t *weight_pow, int filter_length) { int n; for(n=0; n<filter_length; n++) out[n] = (in[n] * weight_pow[n] + 0x4000) >> 15; /* (3.12) = (0.15) * (3.12) with rounding */ } void ff_acelp_high_pass_filter( int16_t* out, int hpf_f[2], const int16_t* in, int length) { int i; int tmp; for(i=0; i<length; i++) { tmp = MULL(hpf_f[0], 15836); /* (14.13) = (13.13) * (1.13) */ tmp += MULL(hpf_f[1], -7667); /* (13.13) = (13.13) * (0.13) */ tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]); /* (14.13) = (0.13) * (14.0) */ /* Multiplication by 2 with rounding can cause short type overflow, thus clipping is required. */ out[i] = av_clip_int16((tmp + 0x800) >> 12); /* (15.0) = 2 * (13.13) = (14.13) */ hpf_f[1] = hpf_f[0]; hpf_f[0] = tmp; } }