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Diffstat (limited to 'libavcodec/acelp_filters.c')
-rw-r--r-- | libavcodec/acelp_filters.c | 125 |
1 files changed, 125 insertions, 0 deletions
diff --git a/libavcodec/acelp_filters.c b/libavcodec/acelp_filters.c new file mode 100644 index 0000000..b98c6f7 --- /dev/null +++ b/libavcodec/acelp_filters.c @@ -0,0 +1,125 @@ +/* + * 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 entire subframe (i.e. almost all + fc_in[i] are zero, in case of G.729D the buffer contains two non-zero + samples before the call to ff_acelp_enhance_harmonics, and (due to + pitch_delay bounded to [20; 143]) a maximum four non-zero samples + for a total of 40 after the call to it), it is faster to swap two loops + and process non-zero samples only. This will reduce the number of + multiplications from 40*40 to 4*40 for G.729D */ + 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; + } +} |