Mercurial > libavcodec.hg
view acelp_vectors.c @ 10674:f04468776158 libavcodec
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author | michael |
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date | Fri, 11 Dec 2009 21:50:08 +0000 |
parents | 8e91a3efdbd2 |
children | 8ee37f5571dc |
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/* * adaptive and fixed codebook vector operations 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_vectors.h" #include "celp_math.h" const uint8_t ff_fc_2pulses_9bits_track1[16] = { 1, 3, 6, 8, 11, 13, 16, 18, 21, 23, 26, 28, 31, 33, 36, 38 }; const uint8_t ff_fc_2pulses_9bits_track1_gray[16] = { 1, 3, 8, 6, 18, 16, 11, 13, 38, 36, 31, 33, 21, 23, 28, 26, }; const uint8_t ff_fc_2pulses_9bits_track2_gray[32] = { 0, 2, 5, 4, 12, 10, 7, 9, 25, 24, 20, 22, 14, 15, 19, 17, 36, 31, 21, 26, 1, 6, 16, 11, 27, 29, 32, 30, 39, 37, 34, 35, }; const uint8_t ff_fc_4pulses_8bits_tracks_13[16] = { 0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75, }; const uint8_t ff_fc_4pulses_8bits_track_4[32] = { 3, 4, 8, 9, 13, 14, 18, 19, 23, 24, 28, 29, 33, 34, 38, 39, 43, 44, 48, 49, 53, 54, 58, 59, 63, 64, 68, 69, 73, 74, 78, 79, }; #if 0 static uint8_t gray_decode[32] = { 0, 1, 3, 2, 7, 6, 4, 5, 15, 14, 12, 13, 8, 9, 11, 10, 31, 30, 28, 29, 24, 25, 27, 26, 16, 17, 19, 18, 23, 22, 20, 21 }; #endif void ff_acelp_fc_pulse_per_track( int16_t* fc_v, const uint8_t *tab1, const uint8_t *tab2, int pulse_indexes, int pulse_signs, int pulse_count, int bits) { int mask = (1 << bits) - 1; int i; for(i=0; i<pulse_count; i++) { fc_v[i + tab1[pulse_indexes & mask]] += (pulse_signs & 1) ? 8191 : -8192; // +/-1 in (2.13) pulse_indexes >>= bits; pulse_signs >>= 1; } fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192; } void ff_acelp_weighted_vector_sum( int16_t* out, const int16_t *in_a, const int16_t *in_b, int16_t weight_coeff_a, int16_t weight_coeff_b, int16_t rounder, int shift, int length) { int i; // Clipping required here; breaks OVERFLOW test. for(i=0; i<length; i++) out[i] = av_clip_int16(( in_a[i] * weight_coeff_a + in_b[i] * weight_coeff_b + rounder) >> shift); } void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length) { int i; for(i=0; i<length; i++) out[i] = weight_coeff_a * in_a[i] + weight_coeff_b * in_b[i]; } void ff_adaptative_gain_control(float *buf_out, float speech_energ, int size, float alpha, float *gain_mem) { int i; float postfilter_energ = ff_dot_productf(buf_out, buf_out, size); float gain_scale_factor = 1.0; float mem = *gain_mem; if (postfilter_energ) gain_scale_factor = sqrt(speech_energ / postfilter_energ); gain_scale_factor *= 1.0 - alpha; for (i = 0; i < size; i++) { mem = alpha * mem + gain_scale_factor; buf_out[i] *= mem; } *gain_mem = mem; } void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in, float sum_of_squares, const int n) { int i; float scalefactor = ff_dot_productf(in, in, n); if (scalefactor) scalefactor = sqrt(sum_of_squares / scalefactor); for (i = 0; i < n; i++) out[i] = in[i] * scalefactor; }