[FFmpeg-cvslog] swr: resampling: add filter type and Kaiser window beta to AVOptions
Justin Ruggles
git at videolan.org
Mon Jul 23 19:27:09 CEST 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Sat May 26 14:50:02 2012 -0400| [7e15df7551cf45ad1d3e39d20fdc8d6c651d4705] | committer: Michael Niedermayer
swr: resampling: add filter type and Kaiser window beta to AVOptions
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7e15df7551cf45ad1d3e39d20fdc8d6c651d4705
---
libswresample/resample.c | 33 ++++++++++++++++++++-------------
libswresample/swresample.c | 7 ++++++-
libswresample/swresample.h | 7 +++++++
libswresample/swresample_internal.h | 4 +++-
4 files changed, 36 insertions(+), 15 deletions(-)
diff --git a/libswresample/resample.c b/libswresample/resample.c
index 4aa53ee..cef2a81 100644
--- a/libswresample/resample.c
+++ b/libswresample/resample.c
@@ -29,9 +29,6 @@
#include "libavutil/avassert.h"
#include "swresample_internal.h"
-#define WINDOW_TYPE 9
-
-
typedef struct ResampleContext {
const AVClass *av_class;
@@ -47,6 +44,8 @@ typedef struct ResampleContext {
int phase_shift;
int phase_mask;
int linear;
+ enum SwrFilterType filter_type;
+ int kaiser_beta;
double factor;
enum AVSampleFormat format;
int felem_size;
@@ -87,10 +86,12 @@ static double bessel(double x){
* 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..16->kaiser windowed sinc beta=2..16
+ * @param filter_type filter type
+ * @param kaiser_beta kaiser window beta
* @return 0 on success, negative on error
*/
-static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale, int type){
+static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,
+ int filter_type, int kaiser_beta){
int ph, i;
double x, y, w;
double *tab = av_malloc(tap_count * sizeof(*tab));
@@ -109,21 +110,23 @@ static int build_filter(ResampleContext *c, void *filter, double factor, int tap
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:{
+ switch(filter_type){
+ case SWR_FILTER_TYPE_CUBIC:{
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:
+ case SWR_FILTER_TYPE_BLACKMAN_NUTTALL:
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;
- default:
+ case SWR_FILTER_TYPE_KAISER:
w = 2.0*x / (factor*tap_count*M_PI);
- y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
+ y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));
break;
+ default:
+ av_assert0(0);
}
tab[i] = y;
@@ -191,12 +194,14 @@ static int build_filter(ResampleContext *c, void *filter, double factor, int tap
return 0;
}
-ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat format){
+ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
+ double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta){
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
- || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format) {
+ || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format
+ || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {
c = av_mallocz(sizeof(*c));
if (!c)
return NULL;
@@ -228,9 +233,11 @@ ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rat
c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
c->filter_alloc = FFALIGN(c->filter_length, 8);
c->filter_bank = av_mallocz(c->filter_alloc*(phase_count+1)*c->felem_size);
+ c->filter_type = filter_type;
+ c->kaiser_beta = kaiser_beta;
if (!c->filter_bank)
goto error;
- if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, WINDOW_TYPE))
+ if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, filter_type, kaiser_beta))
goto error;
memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size);
memcpy(c->filter_bank + (c->filter_alloc*phase_count )*c->felem_size, c->filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);
diff --git a/libswresample/swresample.c b/libswresample/swresample.c
index 0c869f9..89fe0f7 100644
--- a/libswresample/swresample.c
+++ b/libswresample/swresample.c
@@ -88,6 +88,11 @@ static const AVOption options[]={
, OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM },
{"max_soft_comp" , "Maximum factor by which data is stretched/squeezed to make it match the timestamps."
, OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
+{ "filter_type" , "Filter Type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
+ { "cubic" , "Cubic" , 0 , AV_OPT_TYPE_CONST, { SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+ { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+ { "kaiser" , "Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+{ "kaiser_beta" , "Kaiser Window Beta" ,OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.dbl=9 }, 2 , 16 , PARAM },
{0}
};
@@ -244,7 +249,7 @@ int swr_init(struct SwrContext *s){
set_audiodata_fmt(&s->out, s->out_sample_fmt);
if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
- s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt);
+ s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta);
}else
swri_resample_free(&s->resample);
if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
diff --git a/libswresample/swresample.h b/libswresample/swresample.h
index 85a337a..73dcae3 100644
--- a/libswresample/swresample.h
+++ b/libswresample/swresample.h
@@ -53,6 +53,13 @@ enum SwrDitherType {
SWR_DITHER_NB, ///< not part of API/ABI
};
+/** Resampling Filter Types */
+enum SwrFilterType {
+ SWR_FILTER_TYPE_CUBIC, /**< Cubic */
+ SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
+ SWR_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
+};
+
typedef struct SwrContext SwrContext;
/**
diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h
index 52b60ce..e569a44 100644
--- a/libswresample/swresample_internal.h
+++ b/libswresample/swresample_internal.h
@@ -63,6 +63,8 @@ struct SwrContext {
int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
+ enum SwrFilterType filter_type; /**< resampling filter type */
+ int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
float min_compensation; ///< minimum below which no compensation will happen
float min_hard_compensation; ///< minimum below which no silence inject / sample drop will happen
@@ -109,7 +111,7 @@ struct SwrContext {
/* TODO: callbacks for ASM optimizations */
};
-struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat);
+struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat, enum SwrFilterType, int kaiser_beta);
void swri_resample_free(struct ResampleContext **c);
int swri_multiple_resample(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
void swri_resample_compensate(struct ResampleContext *c, int sample_delta, int compensation_distance);
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