[SCM] libav/experimental: polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample filters
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 15:40:41 UTC 2013
The following commit has been merged in the experimental branch:
commit aaaf1635c058dd17bf977356f0deb10b009bc059
Author: Michael Niedermayer <michaelni at gmx.at>
Date: Thu Jun 17 15:43:23 2004 +0000
polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample filters
Originally committed as revision 3228 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index de6fd95..c71e4a9 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -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;
diff --git a/libavcodec/imgresample.c b/libavcodec/imgresample.c
index da57ad7..35aff28 100644
--- a/libavcodec/imgresample.c
+++ b/libavcodec/imgresample.c
@@ -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:
diff --git a/libavcodec/resample.c b/libavcodec/resample.c
index f6f0bf4..b43b4da 100644
--- a/libavcodec/resample.c
+++ b/libavcodec/resample.c
@@ -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);
}
diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c
new file mode 100644
index 0000000..7ea623e
--- /dev/null
+++ b/libavcodec/resample2.c
@@ -0,0 +1,214 @@
+/*
+ * audio resampling
+ * Copyright (c) 2004 Michael Niedermayer <michaelni at 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 at 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;
+}
--
Libav/FFmpeg packaging
More information about the pkg-multimedia-commits
mailing list