avfilter/af_acrossover: add option to adjust input gain

pull/358/head
Paul B Mahol 4 years ago
parent 68adb68e96
commit 66d89a8070
  1. 3
      doc/filters.texi
  2. 27
      libavfilter/af_acrossover.c

@ -539,6 +539,9 @@ Set filter order. Available values are:
@end table @end table
Default is @var{4th}. Default is @var{4th}.
@item level
Set input gain level. Allowed range is from 0 to 1. Default value is 1.
@end table @end table
@subsection Examples @subsection Examples

@ -27,6 +27,7 @@
#include "libavutil/avstring.h" #include "libavutil/avstring.h"
#include "libavutil/channel_layout.h" #include "libavutil/channel_layout.h"
#include "libavutil/eval.h" #include "libavutil/eval.h"
#include "libavutil/float_dsp.h"
#include "libavutil/internal.h" #include "libavutil/internal.h"
#include "libavutil/opt.h" #include "libavutil/opt.h"
@ -55,6 +56,7 @@ typedef struct AudioCrossoverContext {
char *splits_str; char *splits_str;
int order_opt; int order_opt;
float level_in;
int order; int order;
int filter_count; int filter_count;
@ -69,6 +71,8 @@ typedef struct AudioCrossoverContext {
AVFrame *frames[MAX_BANDS]; AVFrame *frames[MAX_BANDS];
int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs); int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
AVFloatDSPContext *fdsp;
} AudioCrossoverContext; } AudioCrossoverContext;
#define OFFSET(x) offsetof(AudioCrossoverContext, x) #define OFFSET(x) offsetof(AudioCrossoverContext, x)
@ -87,6 +91,7 @@ static const AVOption acrossover_options[] = {
{ "16th", "16th order", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, "m" }, { "16th", "16th order", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, "m" },
{ "18th", "18th order", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, "m" }, { "18th", "18th order", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, "m" },
{ "20th", "20th order", 0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, "m" }, { "20th", "20th order", 0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, "m" },
{ "level", "set input gain", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
{ NULL } { NULL }
}; };
@ -98,6 +103,10 @@ static av_cold int init(AVFilterContext *ctx)
char *p, *arg, *saveptr = NULL; char *p, *arg, *saveptr = NULL;
int i, ret = 0; int i, ret = 0;
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits)); s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
if (!s->splits) if (!s->splits)
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
@ -288,7 +297,7 @@ static void biquad_process_## name(BiquadContext *b, \
BIQUAD_PROCESS(fltp, float) BIQUAD_PROCESS(fltp, float)
BIQUAD_PROCESS(dblp, double) BIQUAD_PROCESS(dblp, double)
#define XOVER_PROCESS(name, type, one) \ #define XOVER_PROCESS(name, type, one, ff) \
static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \ static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \
{ \ { \
AudioCrossoverContext *s = ctx->priv; \ AudioCrossoverContext *s = ctx->priv; \
@ -299,23 +308,26 @@ static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, i
const int nb_samples = in->nb_samples; \ const int nb_samples = in->nb_samples; \
\ \
for (int ch = start; ch < end; ch++) { \ for (int ch = start; ch < end; ch++) { \
const type *src = (const type *)in->extended_data[ch]; \
CrossoverChannel *xover = &s->xover[ch]; \ CrossoverChannel *xover = &s->xover[ch]; \
\ \
s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src, \
s->level_in, nb_samples); \
emms_c(); \
\
for (int band = 0; band < ctx->nb_outputs; band++) { \ for (int band = 0; band < ctx->nb_outputs; band++) { \
for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \ for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \
const type *src = (const type *)in->extended_data[ch]; \
const type *prv = (const type *)frames[band]->extended_data[ch]; \ const type *prv = (const type *)frames[band]->extended_data[ch]; \
type *dst = (type *)frames[band + 1]->extended_data[ch]; \ type *dst = (type *)frames[band + 1]->extended_data[ch]; \
const type *hsrc = (band == 0 && f == 0) ? src : f == 0 ? prv : dst; \ const type *hsrc = f == 0 ? prv : dst; \
BiquadContext *hp = &xover->hp[band][f]; \ BiquadContext *hp = &xover->hp[band][f]; \
\ \
biquad_process_## name(hp, dst, hsrc, nb_samples); \ biquad_process_## name(hp, dst, hsrc, nb_samples); \
} \ } \
\ \
for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \ for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \
const type *src = (const type *)in->extended_data[ch]; \
type *dst = (type *)frames[band]->extended_data[ch]; \ type *dst = (type *)frames[band]->extended_data[ch]; \
const type *lsrc = (band == 0 && f == 0) ? src : dst; \ const type *lsrc = dst; \
BiquadContext *lp = &xover->lp[band][f]; \ BiquadContext *lp = &xover->lp[band][f]; \
\ \
biquad_process_## name(lp, dst, lsrc, nb_samples); \ biquad_process_## name(lp, dst, lsrc, nb_samples); \
@ -353,8 +365,8 @@ static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, i
return 0; \ return 0; \
} }
XOVER_PROCESS(fltp, float, 1.f) XOVER_PROCESS(fltp, float, 1.f, f)
XOVER_PROCESS(dblp, double, 1.0) XOVER_PROCESS(dblp, double, 1.0, d)
static int config_input(AVFilterLink *inlink) static int config_input(AVFilterLink *inlink)
{ {
@ -453,6 +465,7 @@ static av_cold void uninit(AVFilterContext *ctx)
AudioCrossoverContext *s = ctx->priv; AudioCrossoverContext *s = ctx->priv;
int i; int i;
av_freep(&s->fdsp);
av_freep(&s->splits); av_freep(&s->splits);
av_freep(&s->xover); av_freep(&s->xover);

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