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