Commit aaaf1635 authored by Michael Niedermayer's avatar Michael Niedermayer
Browse files

polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample filters

Originally committed as revision 3228 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 4904d6c2
......@@ -1846,6 +1846,7 @@ extern AVCodec ac3_decoder;
/* resample.c */
struct ReSampleContext;
struct AVResampleContext;
typedef struct ReSampleContext ReSampleContext;
......@@ -1854,6 +1855,9 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
void audio_resample_close(ReSampleContext *s);
struct AVResampleContext *av_resample_init(int out_rate, int in_rate);
int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
/* YUV420 format is assumed ! */
struct ImgReSampleContext;
......
......@@ -55,6 +55,8 @@ struct ImgReSampleContext {
uint8_t *line_buf;
};
void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type);
static inline int get_phase(int pos)
{
return ((pos) >> (POS_FRAC_BITS - PHASE_BITS)) & ((1 << PHASE_BITS) - 1);
......@@ -540,48 +542,6 @@ static void component_resample(ImgReSampleContext *s,
}
}
/* XXX: the following filter is quite naive, but it seems to suffice
for 4 taps */
static void build_filter(int16_t *filter, float factor)
{
int ph, i, v;
float x, y, tab[NB_TAPS], norm, mult, target;
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
for(ph=0;ph<NB_PHASES;ph++) {
norm = 0;
for(i=0;i<NB_TAPS;i++) {
#if 1
const float d= -0.5; //first order derivative = -0.5
x = fabs(((float)(i - FCENTER) - (float)ph / NB_PHASES) * factor);
if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
else y= d*(-4 + 8*x - 5*x*x + x*x*x);
#else
x = M_PI * ((float)(i - FCENTER) - (float)ph / NB_PHASES) * factor;
if (x == 0)
y = 1.0;
else
y = sin(x) / x;
#endif
tab[i] = y;
norm += y;
}
/* normalize so that an uniform color remains the same */
target= 1 << FILTER_BITS;
for(i=0;i<NB_TAPS;i++) {
mult = target / norm;
v = lrintf(tab[i] * mult);
filter[ph * NB_TAPS + i] = v;
norm -= tab[i];
target -= v;
}
}
}
ImgReSampleContext *img_resample_init(int owidth, int oheight,
int iwidth, int iheight)
{
......@@ -626,10 +586,10 @@ ImgReSampleContext *img_resample_full_init(int owidth, int oheight,
s->h_incr = ((iwidth - leftBand - rightBand) * POS_FRAC) / s->pad_owidth;
s->v_incr = ((iheight - topBand - bottomBand) * POS_FRAC) / s->pad_oheight;
build_filter(&s->h_filters[0][0], (float) s->pad_owidth /
(float) (iwidth - leftBand - rightBand));
build_filter(&s->v_filters[0][0], (float) s->pad_oheight /
(float) (iheight - topBand - bottomBand));
av_build_filter(&s->h_filters[0][0], (float) s->pad_owidth /
(float) (iwidth - leftBand - rightBand), NB_TAPS, NB_PHASES, 1<<FILTER_BITS, 0);
av_build_filter(&s->v_filters[0][0], (float) s->pad_oheight /
(float) (iheight - topBand - bottomBand), NB_TAPS, NB_PHASES, 1<<FILTER_BITS, 0);
return s;
fail:
......
......@@ -24,103 +24,17 @@
#include "avcodec.h"
typedef struct {
/* fractional resampling */
uint32_t incr; /* fractional increment */
uint32_t frac;
int last_sample;
/* integer down sample */
int iratio; /* integer divison ratio */
int icount, isum;
int inv;
} ReSampleChannelContext;
struct AVResampleContext;
struct ReSampleContext {
ReSampleChannelContext channel_ctx[2];
struct AVResampleContext *resample_context;
short *temp[2];
int temp_len;
float ratio;
/* channel convert */
int input_channels, output_channels, filter_channels;
};
#define FRAC_BITS 16
#define FRAC (1 << FRAC_BITS)
static void init_mono_resample(ReSampleChannelContext *s, float ratio)
{
ratio = 1.0 / ratio;
s->iratio = (int)floorf(ratio);
if (s->iratio == 0)
s->iratio = 1;
s->incr = (int)((ratio / s->iratio) * FRAC);
s->frac = FRAC;
s->last_sample = 0;
s->icount = s->iratio;
s->isum = 0;
s->inv = (FRAC / s->iratio);
}
/* fractional audio resampling */
static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
unsigned int frac, incr;
int l0, l1;
short *q, *p, *pend;
l0 = s->last_sample;
incr = s->incr;
frac = s->frac;
p = input;
pend = input + nb_samples;
q = output;
l1 = *p++;
for(;;) {
/* interpolate */
*q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
frac = frac + s->incr;
while (frac >= FRAC) {
frac -= FRAC;
if (p >= pend)
goto the_end;
l0 = l1;
l1 = *p++;
}
}
the_end:
s->last_sample = l1;
s->frac = frac;
return q - output;
}
static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
short *q, *p, *pend;
int c, sum;
p = input;
pend = input + nb_samples;
q = output;
c = s->icount;
sum = s->isum;
for(;;) {
sum += *p++;
if (--c == 0) {
*q++ = (sum * s->inv) >> FRAC_BITS;
c = s->iratio;
sum = 0;
}
if (p >= pend)
break;
}
s->isum = sum;
s->icount = c;
return q - output;
}
/* n1: number of samples */
static void stereo_to_mono(short *output, short *input, int n1)
{
......@@ -210,31 +124,6 @@ static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
}
}
static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
short *buf1;
short *buftmp;
buf1= (short*)av_malloc( nb_samples * sizeof(short) );
/* first downsample by an integer factor with averaging filter */
if (s->iratio > 1) {
buftmp = buf1;
nb_samples = integer_downsample(s, buftmp, input, nb_samples);
} else {
buftmp = input;
}
/* then do a fractional resampling with linear interpolation */
if (s->incr != FRAC) {
nb_samples = fractional_resample(s, output, buftmp, nb_samples);
} else {
memcpy(output, buftmp, nb_samples * sizeof(short));
}
av_free(buf1);
return nb_samples;
}
ReSampleContext *audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate)
{
......@@ -271,16 +160,13 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
if(s->filter_channels>2)
s->filter_channels = 2;
for(i=0;i<s->filter_channels;i++) {
init_mono_resample(&s->channel_ctx[i], s->ratio);
}
s->resample_context= av_resample_init(output_rate, input_rate);
return s;
}
/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
/* XXX: do it with polyphase filters, since the quality here is
HORRIBLE. Return the number of samples available in output */
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
{
int i, nb_samples1;
......@@ -296,8 +182,11 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
}
/* XXX: move those malloc to resample init code */
bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
for(i=0; i<s->filter_channels; i++){
bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
buftmp2[i] = bufin[i] + s->temp_len;
}
/* make some zoom to avoid round pb */
lenout= (int)(nb_samples * s->ratio) + 16;
......@@ -306,27 +195,32 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
if (s->input_channels == 2 &&
s->output_channels == 1) {
buftmp2[0] = bufin[0];
buftmp3[0] = output;
stereo_to_mono(buftmp2[0], input, nb_samples);
} else if (s->output_channels >= 2 && s->input_channels == 1) {
buftmp2[0] = input;
buftmp3[0] = bufout[0];
memcpy(buftmp2[0], input, nb_samples*sizeof(short));
} else if (s->output_channels >= 2) {
buftmp2[0] = bufin[0];
buftmp2[1] = bufin[1];
buftmp3[0] = bufout[0];
buftmp3[1] = bufout[1];
stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
} else {
buftmp2[0] = input;
buftmp3[0] = output;
memcpy(buftmp2[0], input, nb_samples*sizeof(short));
}
nb_samples += s->temp_len;
/* resample each channel */
nb_samples1 = 0; /* avoid warning */
for(i=0;i<s->filter_channels;i++) {
nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
int consumed;
int is_last= i+1 == s->filter_channels;
nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
s->temp_len= nb_samples - consumed;
s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
}
if (s->output_channels == 2 && s->input_channels == 1) {
......@@ -347,5 +241,8 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
void audio_resample_close(ReSampleContext *s)
{
av_resample_close(s->resample_context);
av_freep(&s->temp[0]);
av_freep(&s->temp[1]);
av_free(s);
}
/*
* audio resampling
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* This library 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 of the License, or (at your option) any later version.
*
* This library 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 this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
/**
* @file resample2.c
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "avcodec.h"
#include "common.h"
#define PHASE_SHIFT 10
#define PHASE_COUNT (1<<PHASE_SHIFT)
#define PHASE_MASK (PHASE_COUNT-1)
#define FILTER_SHIFT 15
typedef struct AVResampleContext{
short *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
}AVResampleContext;
/**
* 0th order modified bessel function of the first kind.
*/
double bessel(double x){
double v=1;
double t=1;
int i;
for(i=1; i<50; i++){
t *= i;
v += pow(x*x/4, i)/(t*t);
}
return v;
}
/**
* builds a polyphase filterbank.
* @param factor resampling factor
* @param scale wanted sum of coefficients for each filter
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16
*/
void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
int ph, i, v;
double x, y, w, tab[tap_count];
const int center= (tap_count-1)/2;
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
for(ph=0;ph<phase_count;ph++) {
double norm = 0;
double e= 0;
for(i=0;i<tap_count;i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else y = sin(x) / x;
switch(type){
case 0:{
const float d= -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
else y= d*(-4 + 8*x - 5*x*x + x*x*x);
break;}
case 1:
w = 2.0*x / (factor*tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
break;
case 2:
w = 2.0*x / (factor*tap_count*M_PI);
y *= bessel(16*sqrt(FFMAX(1-w*w, 0))) / bessel(16);
break;
}
tab[i] = y;
norm += y;
}
/* normalize so that an uniform color remains the same */
for(i=0;i<tap_count;i++) {
v = clip(lrintf(tab[i] * scale / norm) + e, -32768, 32767);
filter[ph * tap_count + i] = v;
e += tab[i] * scale / norm - v;
}
}
}
/**
* initalizes a audio resampler.
* note, if either rate is not a integer then simply scale both rates up so they are
*/
AVResampleContext *av_resample_init(int out_rate, int in_rate){
AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
double factor= FFMIN(out_rate / (double)in_rate, 1.0);
memset(c, 0, sizeof(AVResampleContext));
c->filter_length= ceil(16.0/factor);
c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 1]= (1<<FILTER_SHIFT)-1;
c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 2]= 1;
c->src_incr= out_rate;
c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
c->index= -PHASE_COUNT*((c->filter_length-1)/2);
return c;
}
void av_resample_close(AVResampleContext *c){
av_freep(&c->filter_bank);
av_freep(&c);
}
void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
assert(!c->compensation_distance); //FIXME
c->compensation_distance= compensation_distance;
c->dst_incr-= c->ideal_dst_incr * sample_delta / compensation_distance;
}
/**
* resamples.
* @param src an array of unconsumed samples
* @param consumed the number of samples of src which have been consumed are returned here
* @param src_size the number of unconsumed samples available
* @param dst_size the amount of space in samples available in dst
* @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context
* @return the number of samples written in dst or -1 if an error occured
*/
int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
int dst_index, i;
int index= c->index;
int frac= c->frac;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
if(c->compensation_distance && c->compensation_distance < dst_size)
dst_size= c->compensation_distance;
for(dst_index=0; dst_index < dst_size; dst_index++){
short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
int sample_index= index >> PHASE_SHIFT;
int val=0;
if(sample_index < 0){
for(i=0; i<c->filter_length; i++)
val += src[ABS(sample_index + i)] * filter[i];
}else if(sample_index + c->filter_length > src_size){
break;
}else{
#if 0
int64_t v=0;
int sub_phase= (frac<<12) / c->src_incr;
for(i=0; i<c->filter_length; i++){
int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase;
v += src[sample_index + i] * coeff;
}
val= v>>12;
#else
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * filter[i];
}
#endif
}
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
frac += dst_incr_frac;
index += dst_incr;
if(frac >= c->src_incr){
frac -= c->src_incr;
index++;
}
}
if(update_ctx){
if(c->compensation_distance){
c->compensation_distance -= index;
if(!c->compensation_distance)
c->dst_incr= c->ideal_dst_incr;
}
c->frac= frac;
c->index=0;
}
*consumed= index >> PHASE_SHIFT;
return dst_index;
}
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