[FFmpeg-cvslog] avfilter/af_aiir: add support for alternative coefficients format
Paul B Mahol
git at videolan.org
Sun Jan 7 18:06:02 EET 2018
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sun Jan 7 09:43:50 2018 +0100| [6c65de3db06c5379f2ca9173175bfb5f1553518b] | committer: Paul B Mahol
avfilter/af_aiir: add support for alternative coefficients format
Support for zeros/poles syntax on Z-plane.
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=6c65de3db06c5379f2ca9173175bfb5f1553518b
---
doc/filters.texi | 23 +++++++-
libavfilter/af_aiir.c | 153 ++++++++++++++++++++++++++++++++++++++++++++++++--
2 files changed, 167 insertions(+), 9 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 1a2a93b97a..69c59d74a6 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1071,16 +1071,27 @@ It accepts the following parameters:
Set denominator/poles coefficients.
@item b
-Set nominator/zeros coefficients.
+Set numerator/zeros coefficients.
@item dry_gain
Set input gain.
@item wet_gain
Set output gain.
+
+ at item f
+Set coefficients format.
+Can be @code{tf} - transfer function or @code{zp} - Z-plane zeros/poles.
@end table
-Coefficients are separated by spaces and are in ascending order.
+Coefficients in @code{tf} format are separated by spaces and are in ascending
+order.
+
+Coefficients in @code{zp} format are separated by spaces and order of coefficients
+doesn't matter. Coefficients in @code{zp} format are complex numbers with @var{i}
+imaginary unit, also first number in numerator, option @var{b}, is not complex but
+real number and sets overall gain for channel.
+
Different coefficients can be provided for every channel, in such case
use '|' to separate coefficients. Last provided coefficients will be
used for all remaining channels.
@@ -1091,7 +1102,13 @@ used for all remaining channels.
@item
Apply 2 pole elliptic notch at arround 5000Hz for 48000 Hz sample rate:
@example
-aiir=b=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:a=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1
+aiir=b=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:a=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf
+ at end example
+
+ at item
+Same as above but in @code{zp} format:
+ at example
+aiir=b=0.79575848078096756 0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:a=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp
@end example
@end itemize
diff --git a/libavfilter/af_aiir.c b/libavfilter/af_aiir.c
index e14e464211..62ad7ebfa1 100644
--- a/libavfilter/af_aiir.c
+++ b/libavfilter/af_aiir.c
@@ -18,6 +18,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include <float.h>
+
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
@@ -29,6 +31,7 @@ typedef struct AudioIIRContext {
const AVClass *class;
char *a_str, *b_str;
double dry_gain, wet_gain;
+ int format;
int *nb_a, *nb_b;
double **a, **b;
@@ -137,7 +140,7 @@ static void count_coefficients(char *item_str, int *nb_items)
}
}
-static int read_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
+static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
{
char *p, *arg, *old_str, *saveptr = NULL;
int i;
@@ -161,8 +164,40 @@ static int read_coefficients(AVFilterContext *ctx, char *item_str, int nb_items,
return 0;
}
-static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int *nb, double **c, double **cache)
+static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, int is_zeros)
{
+ char *p, *arg, *old_str, *saveptr = NULL;
+ int i;
+
+ p = old_str = av_strdup(item_str);
+ if (!p)
+ return AVERROR(ENOMEM);
+ for (i = 0; i < nb_items; i++) {
+ if (!(arg = av_strtok(p, " ", &saveptr)))
+ break;
+
+ p = NULL;
+ if (i == 0 && is_zeros) {
+ if (sscanf(arg, "%lf", &dst[i]) != 1) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid gain supplied: %s\n", arg);
+ return AVERROR(EINVAL);
+ }
+ } else {
+ if (sscanf(arg, "%lf %lfi", &dst[i*2], &dst[i*2+1]) != 2) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
+ return AVERROR(EINVAL);
+ }
+ }
+ }
+
+ av_freep(&old_str);
+
+ return 0;
+}
+
+static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int *nb, double **c, double **cache, int is_zeros)
+{
+ AudioIIRContext *s = ctx->priv;
char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
int i, ret;
@@ -180,11 +215,17 @@ static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str,
p = NULL;
cache[i] = av_calloc(nb[i] + 1, sizeof(double));
- c[i] = av_calloc(nb[i], sizeof(double));
+ c[i] = av_calloc(nb[i] * (s->format + 1), sizeof(double));
if (!c[i] || !cache[i])
return AVERROR(ENOMEM);
- ret = read_coefficients(ctx, arg, nb[i], c[i]);
+ if (s->format) {
+ ret = read_zp_coefficients(ctx, arg, nb[i], c[i], is_zeros);
+ if (is_zeros)
+ nb[i]--;
+ } else {
+ ret = read_tf_coefficients(ctx, arg, nb[i], c[i]);
+ }
if (ret < 0)
return ret;
prev_arg = arg;
@@ -195,6 +236,97 @@ static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str,
return 0;
}
+static void multiply(double wre, double wim, int npz, double *coeffs)
+{
+ double nwre = -wre, nwim = -wim;
+ double cre, cim;
+ int i;
+
+ for (i = npz; i >= 1; i--) {
+ cre = coeffs[2 * i + 0];
+ cim = coeffs[2 * i + 1];
+
+ coeffs[2 * i + 0] = (nwre * cre - nwim * cim) + coeffs[2 * (i - 1) + 0];
+ coeffs[2 * i + 1] = (nwre * cim + nwim * cre) + coeffs[2 * (i - 1) + 1];
+ }
+
+ cre = coeffs[0];
+ cim = coeffs[1];
+ coeffs[0] = nwre * cre - nwim * cim;
+ coeffs[1] = nwre * cim + nwim * cre;
+}
+
+static int expand(AVFilterContext *ctx, double *pz, int nb, double *coeffs)
+{
+ int i;
+
+ coeffs[0] = 1.0;
+ coeffs[1] = 0.0;
+
+ for (i = 0; i < nb; i++) {
+ coeffs[2 * (i + 1) ] = 0.0;
+ coeffs[2 * (i + 1) + 1] = 0.0;
+ }
+
+ for (i = 0; i < nb; i++)
+ multiply(pz[2 * i], pz[2 * i + 1], nb, coeffs);
+
+ for (i = 0; i < nb + 1; i++) {
+ if (fabs(coeffs[2 * i + 1]) > DBL_EPSILON) {
+ av_log(ctx, AV_LOG_ERROR, "coeff: %lf of z^%d is not real; poles/zeros are not complex conjugates.\n",
+ coeffs[2 * i + i], i);
+ return AVERROR(EINVAL);
+ }
+ }
+
+ return 0;
+}
+
+static int convert_zp2tf(AVFilterContext *ctx, int channels)
+{
+ AudioIIRContext *s = ctx->priv;
+ int ch, i, j, ret;
+
+ for (ch = 0; ch < channels; ch++) {
+ double *topc, *botc, gain;
+
+ topc = av_calloc((s->nb_b[ch] + 1) * 2, sizeof(*topc));
+ botc = av_calloc((s->nb_a[ch] + 1) * 2, sizeof(*botc));
+ if (!topc || !botc)
+ return AVERROR(ENOMEM);
+
+ ret = expand(ctx, s->a[ch], s->nb_a[ch], botc);
+ if (ret < 0) {
+ av_free(topc);
+ av_free(botc);
+ return ret;
+ }
+
+ ret = expand(ctx, &s->b[ch][2], s->nb_b[ch], topc);
+ if (ret < 0) {
+ av_free(topc);
+ av_free(botc);
+ return ret;
+ }
+
+ gain = s->b[ch][0];
+ for (j = 0, i = s->nb_b[ch]; i >= 0; j++, i--) {
+ s->b[ch][j] = topc[2 * i] * gain;
+ }
+ s->nb_b[ch]++;
+
+ for (j = 0, i = s->nb_a[ch]; i >= 0; j++, i--) {
+ s->a[ch][j] = botc[2 * i];
+ }
+ s->nb_a[ch]++;
+
+ av_free(topc);
+ av_free(botc);
+ }
+
+ return 0;
+}
+
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
@@ -212,14 +344,20 @@ static int config_output(AVFilterLink *outlink)
if (!s->a || !s->b || !s->nb_a || !s->nb_b || !s->input || !s->output)
return AVERROR(ENOMEM);
- ret = read_channels(ctx, inlink->channels, s->a_str, s->nb_a, s->a, s->output);
+ ret = read_channels(ctx, inlink->channels, s->a_str, s->nb_a, s->a, s->output, 0);
if (ret < 0)
return ret;
- ret = read_channels(ctx, inlink->channels, s->b_str, s->nb_b, s->b, s->input);
+ ret = read_channels(ctx, inlink->channels, s->b_str, s->nb_b, s->b, s->input, 1);
if (ret < 0)
return ret;
+ if (s->format) {
+ ret = convert_zp2tf(ctx, inlink->channels);
+ if (ret < 0)
+ return ret;
+ }
+
for (ch = 0; ch < inlink->channels; ch++) {
for (i = 1; i < s->nb_a[ch]; i++) {
s->a[ch][i] /= s->a[ch][0];
@@ -336,6 +474,9 @@ static const AVOption aiir_options[] = {
{ "b", "set B/numerator/zeros coefficients", OFFSET(b_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=0}, 0, 1, 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" },
{ NULL },
};
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