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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00

avfilter/af_aiir: add polar zeros/poles format variant

Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2018-01-10 20:25:50 +01:00
parent de5b12c93f
commit de8a1d8d4d
2 changed files with 83 additions and 10 deletions

View File

@ -1084,8 +1084,17 @@ Set output gain.
@item f
Set coefficients format.
Can be @code{tf} - transfer function or @code{zp} - Z-plane zeros/poles.
Default is @code{zp}.
@table @samp
@item tf
transfer function
@item zp
Z-plane zeros/poles, cartesian (default)
@item zpr
Z-plane zeros/poles, polar radians
@item zpd
Z-plane zeros/poles, polar degrees
@end table
@item r
Set kind of processing.
@ -1094,7 +1103,7 @@ Can be @code{d} - direct or @code{s} - serial cascading. Defauls is @code{s}.
@item e
Set filtering precision.
@table @option
@table @samp
@item dbl
double-precision floating-point (default)
@item flt

View File

@ -283,7 +283,7 @@ static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
return 0;
}
static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, const char *format)
{
char *p, *arg, *old_str, *saveptr = NULL;
int i;
@ -296,7 +296,7 @@ static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
break;
p = NULL;
if (sscanf(arg, "%lf %lfi", &dst[i*2], &dst[i*2+1]) != 2) {
if (sscanf(arg, format, &dst[i*2], &dst[i*2+1]) != 2) {
av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
av_freep(&old_str);
return AVERROR(EINVAL);
@ -308,6 +308,8 @@ static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
return 0;
}
static const char *format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd" };
static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int ab)
{
AudioIIRContext *s = ctx->priv;
@ -332,14 +334,14 @@ static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str,
p = NULL;
iir->cache[ab] = av_calloc(iir->nb_ab[ab] + 1, sizeof(double));
iir->ab[ab] = av_calloc(iir->nb_ab[ab] * (s->format + 1), sizeof(double));
iir->ab[ab] = av_calloc(iir->nb_ab[ab] * (!!s->format + 1), sizeof(double));
if (!iir->ab[ab] || !iir->cache[ab]) {
av_freep(&old_str);
return AVERROR(ENOMEM);
}
if (s->format) {
ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]);
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]);
}
@ -586,6 +588,60 @@ static int decompose_zp2biquads(AVFilterContext *ctx, int channels)
return 0;
}
static void convert_pr2zp(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
int ch;
for (ch = 0; ch < channels; ch++) {
IIRChannel *iir = &s->iir[ch];
int n;
for (n = 0; n < iir->nb_ab[0]; n++) {
double r = iir->ab[0][2*n];
double angle = iir->ab[0][2*n+1];
iir->ab[0][2*n] = r * cos(angle);
iir->ab[0][2*n+1] = r * sin(angle);
}
for (n = 0; n < iir->nb_ab[1]; n++) {
double r = iir->ab[1][2*n];
double angle = iir->ab[1][2*n+1];
iir->ab[1][2*n] = r * cos(angle);
iir->ab[1][2*n+1] = r * sin(angle);
}
}
}
static void convert_pd2zp(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
int ch;
for (ch = 0; ch < channels; ch++) {
IIRChannel *iir = &s->iir[ch];
int n;
for (n = 0; n < iir->nb_ab[0]; n++) {
double r = iir->ab[0][2*n];
double angle = M_PI*iir->ab[0][2*n+1]/180.;
iir->ab[0][2*n] = r * cos(angle);
iir->ab[0][2*n+1] = r * sin(angle);
}
for (n = 0; n < iir->nb_ab[1]; n++) {
double r = iir->ab[1][2*n];
double angle = M_PI*iir->ab[1][2*n+1]/180.;
iir->ab[1][2*n] = r * cos(angle);
iir->ab[1][2*n+1] = r * sin(angle);
}
}
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
@ -610,10 +666,16 @@ static int config_output(AVFilterLink *outlink)
if (ret < 0)
return ret;
if (s->format == 2) {
convert_pr2zp(ctx, inlink->channels);
} else if (s->format == 3) {
convert_pd2zp(ctx, inlink->channels);
}
if (s->format == 0)
av_log(ctx, AV_LOG_WARNING, "tf coefficients format is not recommended for too high number of zeros/poles.\n");
if (s->format == 1 && s->process == 0) {
if (s->format > 0 && s->process == 0) {
av_log(ctx, AV_LOG_WARNING, "Direct processsing is not recommended for zp coefficients format.\n");
ret = convert_zp2tf(ctx, inlink->channels);
@ -622,7 +684,7 @@ static int config_output(AVFilterLink *outlink)
} else if (s->format == 0 && s->process == 1) {
av_log(ctx, AV_LOG_ERROR, "Serial cascading is not implemented for transfer function.\n");
return AVERROR_PATCHWELCOME;
} else if (s->format == 1 && s->process == 1) {
} else if (s->format > 0 && s->process == 1) {
if (inlink->format == AV_SAMPLE_FMT_S16P)
av_log(ctx, AV_LOG_WARNING, "Serial cascading is not recommended for i16 precision.\n");
@ -755,9 +817,11 @@ 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 },
{ "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "format" },
{ "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 3, AF, "format" },
{ "tf", "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" },
{ "pd", "Z-plane zeros/poles (polar degrees)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "format" },
{ "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "process" },
{ "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "process" },
{ "s", "serial cascading", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "process" },