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

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Originally committed as revision 3228 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer
2004-06-17 15:43:23 +00:00
parent 4904d6c2d3
commit aaaf1635c0
4 changed files with 249 additions and 174 deletions
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153
libavcodec/resample.c
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@ -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);
}