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avfilter/af_aiir: add analog transfer function format

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This commit is contained in:
Paul B Mahol 2020-10-18 18:25:51 +02:00
parent 6ef55f54fe
commit 847dc03787
2 changed files with 74 additions and 8 deletions
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10
doc/filters.texi
View File

@ -1408,6 +1408,8 @@ Set output gain.
Set coefficients format.
@table @samp
@item sf
analog transfer function
@item tf
digital transfer function
@item zp
@ -1466,7 +1468,7 @@ displayed. This option is used only when @var{response} is enabled.
Set video stream size. This option is used only when @var{response} is enabled.
@end table
Coefficients in @code{tf} format are separated by spaces and are in ascending
Coefficients in @code{tf} and @code{sf} format are separated by spaces and are in ascending
order.
Coefficients in @code{zp} format are separated by spaces and order of coefficients
@ -1491,6 +1493,12 @@ Same as above but in @code{zp} format:
@example
aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s
@end example
@item
Apply 3-rd order analog normalized Butterworth low-pass filter, using analog transfer function format:
@example
aiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d
@end example
@end itemize
@section alimiter

72
libavfilter/af_aiir.c
View File

@ -428,7 +428,7 @@ static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str,
return AVERROR(ENOMEM);
}
if (s->format) {
if (s->format > 0) {
ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab], format[s->format]);
} else {
ret = read_tf_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]);
@ -898,6 +898,60 @@ static void convert_sp2zp(AVFilterContext *ctx, int channels)
}
}
static double fact(double i)
{
if (i <= 0.)
return 1.;
return i * fact(i - 1.);
}
static double coef_sf2zf(double *a, int N, int n, double fs)
{
double z = 0.;
for (int i = 0; i <= N; i++) {
double acc = 0.;
for (int k = FFMAX(n - N + i, 0); k <= FFMIN(i, n); k++) {
acc += ((fact(i) * fact(N - i)) /
(fact(k) * fact(i - k) * fact(n - k) * fact(N - i - n + k))) *
((k & 1) ? -1. : 1.);;
}
z += a[i] * pow(2., i) * acc;
}
return z;
}
static void convert_sf2tf(AVFilterContext *ctx, int channels, int sample_rate)
{
AudioIIRContext *s = ctx->priv;
int ch;
for (ch = 0; ch < channels; ch++) {
IIRChannel *iir = &s->iir[ch];
double *temp0 = av_calloc(iir->nb_ab[0], sizeof(*temp0));
double *temp1 = av_calloc(iir->nb_ab[1], sizeof(*temp1));
if (!temp0 || !temp1)
goto next;
memcpy(temp0, iir->ab[0], iir->nb_ab[0] * sizeof(*temp0));
memcpy(temp1, iir->ab[1], iir->nb_ab[1] * sizeof(*temp1));
for (int n = 0; n < iir->nb_ab[0]; n++)
iir->ab[0][n] = coef_sf2zf(temp0, iir->nb_ab[0] - 1, n, sample_rate);
for (int n = 0; n < iir->nb_ab[1]; n++)
iir->ab[1][n] = coef_sf2zf(temp1, iir->nb_ab[1] - 1, n, sample_rate);
next:
av_free(temp0);
av_free(temp1);
}
}
static void convert_pd2zp(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
@ -1186,7 +1240,10 @@ static int config_output(AVFilterLink *outlink)
if (ret < 0)
return ret;
if (s->format == 2) {
if (s->format == -1) {
convert_sf2tf(ctx, inlink->channels, inlink->sample_rate);
s->format = 0;
} else if (s->format == 2) {
convert_pr2zp(ctx, inlink->channels);
} else if (s->format == 3) {
convert_pd2zp(ctx, inlink->channels);
@ -1207,7 +1264,7 @@ static int config_output(AVFilterLink *outlink)
}
if (s->format == 0)
av_log(ctx, AV_LOG_WARNING, "tf coefficients format is not recommended for too high number of zeros/poles.\n");
av_log(ctx, AV_LOG_WARNING, "transfer function coefficients format is not recommended for too high number of zeros/poles.\n");
if (s->format > 0 && s->process == 0) {
av_log(ctx, AV_LOG_WARNING, "Direct processsing is not recommended for zp coefficients format.\n");
@ -1215,10 +1272,10 @@ static int config_output(AVFilterLink *outlink)
ret = convert_zp2tf(ctx, inlink->channels);
if (ret < 0)
return ret;
} else if (s->format == 0 && s->process == 1) {
} else if (s->format <= 0 && s->process == 1) {
av_log(ctx, AV_LOG_ERROR, "Serial processing is not implemented for transfer function.\n");
return AVERROR_PATCHWELCOME;
} else if (s->format == 0 && s->process == 2) {
} else if (s->format <= 0 && s->process == 2) {
av_log(ctx, AV_LOG_ERROR, "Parallel processing is not implemented for transfer function.\n");
return AVERROR_PATCHWELCOME;
} else if (s->format > 0 && s->process == 1) {
@ -1416,8 +1473,9 @@ static const AVOption aiir_options[] = {
{ "k", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
{ "format", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 4, AF, "format" },
{ "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 4, AF, "format" },
{ "format", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -1, 4, AF, "format" },
{ "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -1, 4, AF, "format" },
{ "sf", "analog transfer function", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "format" },
{ "tf", "digital transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "format" },
{ "zp", "Z-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "format" },
{ "pr", "Z-plane zeros/poles (polar radians)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "format" },