[FFmpeg-cvslog] avfilter/af_afreqshift: allow to change order of filters
Paul B Mahol
git at videolan.org
Mon Aug 23 20:45:21 EEST 2021
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon Aug 23 19:30:48 2021 +0200| [7f194c7d965ff9791e498074ca803ab0a097fa01] | committer: Paul B Mahol
avfilter/af_afreqshift: allow to change order of filters
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7f194c7d965ff9791e498074ca803ab0a097fa01
---
doc/filters.texi | 8 ++++++++
libavfilter/af_afreqshift.c | 34 ++++++++++++++++++++--------------
2 files changed, 28 insertions(+), 14 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index c84202cf85..b902aca12d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1487,6 +1487,10 @@ Default value is 0.0.
@item level
Set output gain applied to final output. Allowed range is from 0.0 to 1.0.
Default value is 1.0.
+
+ at item order
+Set filter order used for filtering. Allowed range is from 1 to 16.
+Default value is 8.
@end table
@subsection Commands
@@ -2335,6 +2339,10 @@ Default value is 0.0.
@item level
Set output gain applied to final output. Allowed range is from 0.0 to 1.0.
Default value is 1.0.
+
+ at item order
+Set filter order used for filtering. Allowed range is from 1 to 16.
+Default value is 8.
@end table
@subsection Commands
diff --git a/libavfilter/af_afreqshift.c b/libavfilter/af_afreqshift.c
index 3be393d9f7..90f0c3014e 100644
--- a/libavfilter/af_afreqshift.c
+++ b/libavfilter/af_afreqshift.c
@@ -26,16 +26,18 @@
#include "audio.h"
#include "formats.h"
-#define NB_COEFS 16
+#define MAX_NB_COEFFS 16
typedef struct AFreqShift {
const AVClass *class;
double shift;
double level;
+ int nb_coeffs;
+ int old_nb_coeffs;
- double cd[NB_COEFS];
- float cf[NB_COEFS];
+ double cd[MAX_NB_COEFFS * 2];
+ float cf[MAX_NB_COEFFS * 2];
int64_t in_samples;
@@ -88,7 +90,7 @@ static void pfilter_channel_## name(AVFilterContext *ctx, \
type xn1 = src[n], xn2 = src[n]; \
type I, Q; \
\
- for (int j = 0; j < NB_COEFS / 2; j++) { \
+ for (int j = 0; j < s->nb_coeffs; j++) { \
I = c[j] * (xn1 + o2[j]) - i2[j]; \
i2[j] = i1[j]; \
i1[j] = xn1; \
@@ -97,7 +99,7 @@ static void pfilter_channel_## name(AVFilterContext *ctx, \
xn1 = I; \
} \
\
- for (int j = NB_COEFS / 2; j < NB_COEFS; j++) { \
+ for (int j = s->nb_coeffs; j < s->nb_coeffs*2; j++) { \
Q = c[j] * (xn2 + o2[j]) - i2[j]; \
i2[j] = i1[j]; \
i1[j] = xn2; \
@@ -105,7 +107,7 @@ static void pfilter_channel_## name(AVFilterContext *ctx, \
o1[j] = Q; \
xn2 = Q; \
} \
- Q = o2[NB_COEFS - 1]; \
+ Q = o2[s->nb_coeffs * 2 - 1]; \
\
dst[n] = (I * cos_theta - Q * sin_theta) * level; \
} \
@@ -137,7 +139,7 @@ static void ffilter_channel_## name(AVFilterContext *ctx, \
type xn1 = src[n], xn2 = src[n]; \
type I, Q, theta; \
\
- for (int j = 0; j < NB_COEFS / 2; j++) { \
+ for (int j = 0; j < s->nb_coeffs; j++) { \
I = c[j] * (xn1 + o2[j]) - i2[j]; \
i2[j] = i1[j]; \
i1[j] = xn1; \
@@ -146,7 +148,7 @@ static void ffilter_channel_## name(AVFilterContext *ctx, \
xn1 = I; \
} \
\
- for (int j = NB_COEFS / 2; j < NB_COEFS; j++) { \
+ for (int j = s->nb_coeffs; j < s->nb_coeffs*2; j++) { \
Q = c[j] * (xn2 + o2[j]) - i2[j]; \
i2[j] = i1[j]; \
i1[j] = xn2; \
@@ -154,7 +156,7 @@ static void ffilter_channel_## name(AVFilterContext *ctx, \
o1[j] = Q; \
xn2 = Q; \
} \
- Q = o2[NB_COEFS - 1]; \
+ Q = o2[s->nb_coeffs * 2 - 1]; \
\
theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.); \
dst[n] = (I * cos(theta) - Q * sin(theta)) * level; \
@@ -266,12 +268,14 @@ static int config_input(AVFilterLink *inlink)
AVFilterContext *ctx = inlink->dst;
AFreqShift *s = ctx->priv;
- compute_coefs(s->cd, s->cf, NB_COEFS, 2. * 20. / inlink->sample_rate);
+ if (s->old_nb_coeffs != s->nb_coeffs)
+ compute_coefs(s->cd, s->cf, s->nb_coeffs * 2, 2. * 20. / inlink->sample_rate);
+ s->old_nb_coeffs = s->nb_coeffs;
- s->i1 = ff_get_audio_buffer(inlink, NB_COEFS);
- s->o1 = ff_get_audio_buffer(inlink, NB_COEFS);
- s->i2 = ff_get_audio_buffer(inlink, NB_COEFS);
- s->o2 = ff_get_audio_buffer(inlink, NB_COEFS);
+ s->i1 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
+ s->o1 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
+ s->i2 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
+ s->o2 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
if (!s->i1 || !s->o1 || !s->i2 || !s->o2)
return AVERROR(ENOMEM);
@@ -355,6 +359,7 @@ static av_cold void uninit(AVFilterContext *ctx)
static const AVOption afreqshift_options[] = {
{ "shift", "set frequency shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -INT_MAX, INT_MAX, FLAGS },
{ "level", "set output level", OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0, 1.0, FLAGS },
+ { "order", "set filter order", OFFSET(nb_coeffs),AV_OPT_TYPE_INT, {.i64=8}, 1, MAX_NB_COEFFS, FLAGS },
{ NULL }
};
@@ -393,6 +398,7 @@ const AVFilter ff_af_afreqshift = {
static const AVOption aphaseshift_options[] = {
{ "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1.0, 1.0, FLAGS },
{ "level", "set output level",OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0, 1.0, FLAGS },
+ { "order", "set filter order",OFFSET(nb_coeffs), AV_OPT_TYPE_INT,{.i64=8}, 1, MAX_NB_COEFFS, FLAGS },
{ NULL }
};
More information about the ffmpeg-cvslog
mailing list