mirror of https://github.com/FFmpeg/FFmpeg.git
Originally committed as revision 3228 to svn://svn.ffmpeg.org/ffmpeg/trunkpull/126/head
Michael Niedermayer
21 years ago
parent
4904d6c2d3
commit
aaaf1635c0
4 changed files with 249 additions and 174 deletions
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/*
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* audio resampling |
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> |
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* |
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* This library 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 of the License, or (at your option) any later version. |
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* |
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* This library 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 this library; if not, write to the Free Software |
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
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* |
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*/ |
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/**
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* @file resample2.c |
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* audio resampling |
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* @author Michael Niedermayer <michaelni@gmx.at> |
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*/ |
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#include "avcodec.h" |
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#include "common.h" |
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#define PHASE_SHIFT 10 |
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#define PHASE_COUNT (1<<PHASE_SHIFT) |
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#define PHASE_MASK (PHASE_COUNT-1) |
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#define FILTER_SHIFT 15 |
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typedef struct AVResampleContext{ |
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short *filter_bank; |
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int filter_length; |
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int ideal_dst_incr; |
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int dst_incr; |
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int index; |
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int frac; |
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int src_incr; |
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int compensation_distance; |
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}AVResampleContext; |
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/**
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* 0th order modified bessel function of the first kind. |
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*/ |
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double bessel(double x){ |
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double v=1; |
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double t=1; |
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int i; |
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for(i=1; i<50; i++){ |
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t *= i; |
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v += pow(x*x/4, i)/(t*t); |
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} |
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return v; |
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} |
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/**
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* builds a polyphase filterbank. |
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* @param factor resampling factor |
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* @param scale wanted sum of coefficients for each filter |
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* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16 |
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*/ |
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void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){ |
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int ph, i, v; |
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double x, y, w, tab[tap_count]; |
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const int center= (tap_count-1)/2; |
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/* if upsampling, only need to interpolate, no filter */ |
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if (factor > 1.0) |
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factor = 1.0; |
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for(ph=0;ph<phase_count;ph++) { |
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double norm = 0; |
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double e= 0; |
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for(i=0;i<tap_count;i++) { |
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x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; |
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if (x == 0) y = 1.0; |
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else y = sin(x) / x; |
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switch(type){ |
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case 0:{ |
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const float d= -0.5; //first order derivative = -0.5
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x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); |
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if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); |
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else y= d*(-4 + 8*x - 5*x*x + x*x*x); |
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break;} |
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case 1: |
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w = 2.0*x / (factor*tap_count) + M_PI; |
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y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); |
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break; |
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case 2: |
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w = 2.0*x / (factor*tap_count*M_PI); |
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y *= bessel(16*sqrt(FFMAX(1-w*w, 0))) / bessel(16); |
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break; |
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} |
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tab[i] = y; |
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norm += y; |
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} |
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/* normalize so that an uniform color remains the same */ |
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for(i=0;i<tap_count;i++) { |
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v = clip(lrintf(tab[i] * scale / norm) + e, -32768, 32767); |
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filter[ph * tap_count + i] = v; |
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e += tab[i] * scale / norm - v; |
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} |
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} |
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} |
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/**
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* initalizes a audio resampler. |
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* note, if either rate is not a integer then simply scale both rates up so they are |
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*/ |
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AVResampleContext *av_resample_init(int out_rate, int in_rate){ |
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AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); |
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double factor= FFMIN(out_rate / (double)in_rate, 1.0); |
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memset(c, 0, sizeof(AVResampleContext)); |
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c->filter_length= ceil(16.0/factor); |
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c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short)); |
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av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1); |
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c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 1]= (1<<FILTER_SHIFT)-1; |
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c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 2]= 1; |
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c->src_incr= out_rate; |
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c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT; |
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c->index= -PHASE_COUNT*((c->filter_length-1)/2); |
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return c; |
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} |
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void av_resample_close(AVResampleContext *c){ |
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av_freep(&c->filter_bank); |
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av_freep(&c); |
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} |
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void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ |
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assert(!c->compensation_distance); //FIXME
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c->compensation_distance= compensation_distance; |
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c->dst_incr-= c->ideal_dst_incr * sample_delta / compensation_distance; |
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} |
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/**
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* resamples. |
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* @param src an array of unconsumed samples |
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* @param consumed the number of samples of src which have been consumed are returned here |
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* @param src_size the number of unconsumed samples available |
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* @param dst_size the amount of space in samples available in dst |
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* @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context |
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* @return the number of samples written in dst or -1 if an error occured |
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*/ |
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int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ |
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int dst_index, i; |
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int index= c->index; |
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int frac= c->frac; |
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int dst_incr_frac= c->dst_incr % c->src_incr; |
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int dst_incr= c->dst_incr / c->src_incr; |
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if(c->compensation_distance && c->compensation_distance < dst_size) |
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dst_size= c->compensation_distance; |
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for(dst_index=0; dst_index < dst_size; dst_index++){ |
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short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK); |
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int sample_index= index >> PHASE_SHIFT; |
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int val=0; |
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if(sample_index < 0){ |
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for(i=0; i<c->filter_length; i++) |
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val += src[ABS(sample_index + i)] * filter[i]; |
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}else if(sample_index + c->filter_length > src_size){ |
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break; |
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}else{ |
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#if 0 |
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int64_t v=0; |
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int sub_phase= (frac<<12) / c->src_incr; |
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for(i=0; i<c->filter_length; i++){ |
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int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase; |
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v += src[sample_index + i] * coeff; |
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} |
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val= v>>12; |
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#else |
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for(i=0; i<c->filter_length; i++){ |
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val += src[sample_index + i] * filter[i]; |
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} |
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#endif |
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} |
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val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; |
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dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; |
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frac += dst_incr_frac; |
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index += dst_incr; |
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if(frac >= c->src_incr){ |
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frac -= c->src_incr; |
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index++; |
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} |
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} |
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if(update_ctx){ |
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if(c->compensation_distance){ |
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c->compensation_distance -= index; |
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if(!c->compensation_distance) |
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c->dst_incr= c->ideal_dst_incr; |
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} |
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c->frac= frac; |
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c->index=0; |
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} |
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*consumed= index >> PHASE_SHIFT; |
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return dst_index; |
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} |
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