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avfilter/af_afir: add irnorm and irlink options

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Deprecate gtype option.
This commit is contained in:
Paul B Mahol 2023-11-18 00:36:18 +01:00
parent 07c303b708
commit 5452cbdc15
4 changed files with 90 additions and 118 deletions
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111
libavfilter/afir_template.c
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@ -29,6 +29,8 @@
#undef HYPOT
#undef SAMPLE_FORMAT
#undef TX_TYPE
#undef FABS
#undef POW
#if DEPTH == 32
#define SAMPLE_FORMAT float
#define SQRT sqrtf
@ -36,6 +38,8 @@
#define ctype AVComplexFloat
#define ftype float
#define TX_TYPE AV_TX_FLOAT_RDFT
#define FABS fabsf
#define POW powf
#else
#define SAMPLE_FORMAT double
#define SQRT sqrt
@ -43,6 +47,8 @@
#define ctype AVComplexDouble
#define ftype double
#define TX_TYPE AV_TX_DOUBLE_RDFT
#define FABS fabs
#define POW pow
#endif
#define fn3(a,b) a##_##b
@ -139,95 +145,32 @@ end:
av_free(mag);
}
static int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s,
int cur_nb_taps, int ch,
ftype *time)
static ftype fn(ir_gain)(AVFilterContext *ctx, AudioFIRContext *s,
int cur_nb_taps, const ftype *time)
{
ftype ch_gain = 1;
ftype ch_gain, sum = 0;
switch (s->gtype) {
case -1:
if (s->ir_norm < 0.f) {
ch_gain = 1;
break;
case 0:
{
ftype sum = 0;
} else if (s->ir_norm == 0.f) {
for (int i = 0; i < cur_nb_taps; i++)
sum += time[i];
ch_gain = 1. / sum;
} else {
ftype ir_norm = s->ir_norm;
for (int i = 0; i < cur_nb_taps; i++)
sum += FFABS(time[i]);
ch_gain = 1. / sum;
}
break;
case 1:
{
ftype sum = 0;
for (int i = 0; i < cur_nb_taps; i++)
sum += time[i];
ch_gain = 1. / sum;
}
break;
case 2:
{
ftype sum = 0;
for (int i = 0; i < cur_nb_taps; i++)
sum += time[i] * time[i];
ch_gain = 1. / SQRT(sum);
}
break;
case 3:
case 4:
{
ftype *inc, *outc, scale, power;
AVTXContext *tx;
av_tx_fn tx_fn;
int ret, size;
size = 1 << av_ceil_log2_c(cur_nb_taps);
inc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT));
outc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT));
if (!inc || !outc) {
av_free(outc);
av_free(inc);
break;
}
scale = 1.;
ret = av_tx_init(&tx, &tx_fn, TX_TYPE, 0, size, &scale, 0);
if (ret < 0) {
av_free(outc);
av_free(inc);
break;
}
{
memcpy(inc, time, cur_nb_taps * sizeof(SAMPLE_FORMAT));
tx_fn(tx, outc, inc, sizeof(SAMPLE_FORMAT));
power = 0;
if (s->gtype == 3) {
for (int i = 0; i < size / 2 + 1; i++)
power = FFMAX(power, HYPOT(outc[i * 2], outc[i * 2 + 1]));
} else {
ftype sum = 0;
for (int i = 0; i < size / 2 + 1; i++)
sum += HYPOT(outc[i * 2], outc[i * 2 + 1]);
power = SQRT(sum / (size / 2 + 1));
}
ch_gain = 1. / power;
}
av_tx_uninit(&tx);
av_free(outc);
av_free(inc);
}
break;
default:
return AVERROR_BUG;
for (int i = 0; i < cur_nb_taps; i++)
sum += POW(FABS(time[i]), ir_norm);
ch_gain = 1. / POW(sum, 1. / ir_norm);
}
return ch_gain;
}
static void fn(ir_scale)(AVFilterContext *ctx, AudioFIRContext *s,
int cur_nb_taps, int ch,
ftype *time, ftype ch_gain)
{
if (ch_gain != 1. || s->ir_gain != 1.) {
ftype gain = ch_gain * s->ir_gain;
@ -238,8 +181,6 @@ static int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s,
s->fdsp->vector_dmul_scalar(time, time, gain, FFALIGN(cur_nb_taps, 8));
#endif
}
return 0;
}
static void fn(convert_channel)(AVFilterContext *ctx, AudioFIRContext *s, int ch,