Signed-off-by: Paul B Mahol <one...@gmail.com>
---
doc/filters.texi | 30 ++++
libavfilter/Makefile | 2 +
libavfilter/af_afreqshift.c | 274 ++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 2 +
4 files changed, 308 insertions(+)
create mode 100644 libavfilter/af_afreqshift.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 50ef692077..428bf4ce92 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1314,6 +1314,21 @@ Force the output to either unsigned 8-bit or signed
16-bit stereo
aformat=sample_fmts=u8|s16:channel_layouts=stereo
@end example
+@section afreqshift
+Apply frequency shift to input audio samples.
+
+The filter accepts the following options:
+
+@table @option
+@item shift
+Specify frequency shift. Allowed range is -INT_MAX to INT_MAX.
+Default value is 0.0.
+@end table
+
+@subsection Commands
+
+This filter supports the above option as @ref{commands}.
+
@section agate
A gate is mainly used to reduce lower parts of a signal. This kind of signal
@@ -2056,6 +2071,21 @@ It accepts the following values:
@end table
@end table
+@section aphaseshift
+Apply phase shift to input audio samples.
+
+The filter accepts the following options:
+
+@table @option
+@item shift
+Specify phase shift. Allowed range is -M_PI to M_PI.
+Default value is 0.0.
+@end table
+
+@subsection Commands
+
+This filter supports the above option as @ref{commands}.
+
@section apulsator
Audio pulsator is something between an autopanner and a tremolo.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 2691612179..480e191987 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -50,6 +50,7 @@ OBJS-$(CONFIG_AFFTDN_FILTER) += af_afftdn.o
OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o
OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
+OBJS-$(CONFIG_AFREQSHIFT_FILTER) += af_afreqshift.o
OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_AIIR_FILTER) += af_aiir.o
OBJS-$(CONFIG_AINTEGRAL_FILTER) += af_aderivative.o
@@ -69,6 +70,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o
generate_wave_table.o
+OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o
OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
diff --git a/libavfilter/af_afreqshift.c b/libavfilter/af_afreqshift.c
new file mode 100644
index 0000000000..6173be5a61
--- /dev/null
+++ b/libavfilter/af_afreqshift.c
@@ -0,0 +1,274 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/ffmath.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct AFreqShift {
+ const AVClass *class;
+
+ double shift;
+
+ double c[12];
+
+ int64_t in_samples;
+
+ AVFrame *i, *o;
+
+ void (*filter_channel)(AVFilterContext *ctx,
+ int nb_samples,
+ int sample_rate,
+ const double *src, double *dst,
+ double *i, double *o);
+} AFreqShift;
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static const double poles[12] =
+{
+ 0.3609, 2.7412, 11.1573, 44.7581, 179.6242, 798.4578,
+ 1.2524, 5.5671, 22.3423, 89.6271, 364.7914, 2770.1114,
+};
+
+static void pfilter_channel(AVFilterContext *ctx,
+ int nb_samples,
+ int sample_rate,
+ const double *src, double *dst,
+ double *i, double *o)
+{
+ AFreqShift *s = ctx->priv;
+ double *c = s->c;
+ double shift = s->shift;
+ double cos_theta = cos(shift);
+ double sin_theta = sin(shift);
+
+ for (int n = 0; n < nb_samples; n++) {
+ double xn1 = src[n], xn2 = src[n];
+ double I, Q;
+
+ for (int j = 0; j < 6; j++) {
+ I = c[j] * (xn1 - i[j]) + o[j];
+ o[j] = xn1;
+ i[j] = I;
+ xn1 = I;
+ }
+
+ for (int j = 6; j < 12; j++) {
+ Q = c[j] * (xn2 - i[j]) + o[j];
+ o[j] = xn2;
+ i[j] = Q;
+ xn2 = Q;
+ }
+
+ dst[n] = I * cos_theta - Q * sin_theta;
+ }
+}
+
+static void ffilter_channel(AVFilterContext *ctx,
+ int nb_samples,
+ int sample_rate,
+ const double *src, double *dst,
+ double *i, double *o)
+{
+ AFreqShift *s = ctx->priv;
+ double *c = s->c;
+ double ts = 1. / sample_rate;
+ double shift = s->shift;
+ int64_t N = s->in_samples;
+
+ for (int n = 0; n < nb_samples; n++) {
+ double xn1 = src[n], xn2 = src[n];
+ double I, Q, theta;
+
+ for (int j = 0; j < 6; j++) {
+ I = c[j] * (xn1 - i[j]) + o[j];
+ o[j] = xn1;
+ i[j] = I;
+ xn1 = I;
+ }
+
+ for (int j = 6; j < 12; j++) {
+ Q = c[j] * (xn2 - i[j]) + o[j];
+ o[j] = xn2;
+ i[j] = Q;
+ xn2 = Q;
+ }
+
+ theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.);
+ dst[n] = I * cos(theta) - Q * sin(theta);
+ }
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AFreqShift *s = ctx->priv;
+
+ for (int j = 0; j < 12; j++) {
+ double polefreq = poles[j] * 15.;
+ double rc = 1. / (2. * M_PI * polefreq);
+ double alpha = (1. / rc) * 0.5 * (1. / inlink->sample_rate);
+ double beta = (1. - alpha) / (1. + alpha);
+
+ s->c[j] = -beta;
+ }
+
+ s->i = ff_get_audio_buffer(inlink, 12);
+ s->o = ff_get_audio_buffer(inlink, 12);
+ if (!s->i || !s->o)
+ return AVERROR(ENOMEM);
+
+ if (!strcmp(ctx->filter->name, "afreqshift"))
+ s->filter_channel = ffilter_channel;
+ else
+ s->filter_channel = pfilter_channel;
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AFreqShift *s = ctx->priv;
+ AVFrame *out;
+
+ if (av_frame_is_writable(in)) {
+ out = in;
+ } else {
+ out = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+ }
+
+ for (int ch = 0; ch < in->channels; ch++) {
+ s->filter_channel(ctx, in->nb_samples,
+ in->sample_rate,
+ (const double *)in->extended_data[ch],
+ (double *)out->extended_data[ch],
+ (double *)s->i->extended_data[ch],
+ (double *)s->o->extended_data[ch]);
+ }
+
+ s->in_samples += in->nb_samples;
+
+ if (out != in)
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AFreqShift *s = ctx->priv;
+
+ av_frame_free(&s->i);
+ av_frame_free(&s->o);
+}
+
+#define OFFSET(x) offsetof(AFreqShift, x)
+#define FLAGS
AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption afreqshift_options[] = {
+ { "shift", "set frequency shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE,
{.dbl=0}, -INT_MAX, INT_MAX, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(afreqshift);
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_afreqshift = {
+ .name = "afreqshift",
+ .description = NULL_IF_CONFIG_SMALL("Apply frequency shifting to input
audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AFreqShift),
+ .priv_class = &afreqshift_class,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+ .process_command = ff_filter_process_command,
+};
+
+static const AVOption aphaseshift_options[] = {
+ { "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0},
-M_PI, M_PI, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aphaseshift);
+
+AVFilter ff_af_aphaseshift = {
+ .name = "aphaseshift",
+ .description = NULL_IF_CONFIG_SMALL("Apply phase shifting to input
audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AFreqShift),
+ .priv_class = &aphaseshift_class,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+ .process_command = ff_filter_process_command,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 26a8e87b0b..a5ec6bd4ca 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -43,6 +43,7 @@ extern AVFilter ff_af_afftdn;
extern AVFilter ff_af_afftfilt;
extern AVFilter ff_af_afir;
extern AVFilter ff_af_aformat;
+extern AVFilter ff_af_afreqshift;
extern AVFilter ff_af_agate;
extern AVFilter ff_af_aiir;
extern AVFilter ff_af_aintegral;
@@ -62,6 +63,7 @@ extern AVFilter ff_af_anull;
extern AVFilter ff_af_apad;
extern AVFilter ff_af_aperms;
extern AVFilter ff_af_aphaser;
+extern AVFilter ff_af_aphaseshift;
extern AVFilter ff_af_apulsator;
extern AVFilter ff_af_arealtime;
extern AVFilter ff_af_aresample;
--
2.17.1
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