* [FFmpeg-devel] [PATCH] avfilter: add fractional delay IR source filter
@ 2023-01-08 14:03 Paul B Mahol
2023-01-15 9:45 ` Paul B Mahol
0 siblings, 1 reply; 2+ messages in thread
From: Paul B Mahol @ 2023-01-08 14:03 UTC (permalink / raw)
To: FFmpeg development discussions and patches
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Patch attached.
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From 7492dad5bc3a7826e5abd67f26503c74bd5745d2 Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Sun, 8 Jan 2023 13:53:39 +0100
Subject: [PATCH] avfilter: add fractional delay IR source filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
doc/filters.texi | 23 ++++++
libavfilter/Makefile | 1 +
libavfilter/allfilters.c | 1 +
libavfilter/asrc_afdelaysrc.c | 145 ++++++++++++++++++++++++++++++++++
4 files changed, 170 insertions(+)
create mode 100644 libavfilter/asrc_afdelaysrc.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 9c32339141..e8febc2bfd 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -7519,6 +7519,29 @@ aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"
@end itemize
+@section afdelaysrc
+
+Generate a fractional delay FIR coefficients.
+
+The resulting stream can be used with @ref{afir} filter for filtering the audio signal.
+
+The filter accepts the following options:
+
+@table @option
+@item delay, d
+Set the fractional delay. Default is 0.
+
+@item sample_rate, r
+Set the sample rate, default is 44100.
+
+@item nb_samples, n
+Set the number of samples per each frame. Default is 1024.
+
+@item taps, t
+Set the number of filter coefficents in output audio stream.
+Default value is 0.
+@end table
+
@section afirsrc
Generate a FIR coefficients using frequency sampling method.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index f235ac047e..65cf3e7d7e 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -172,6 +172,7 @@ OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o
OBJS-$(CONFIG_AEVALSRC_FILTER) += aeval.o
+OBJS-$(CONFIG_AFDELAYSRC_FILTER) += asrc_afdelaysrc.o
OBJS-$(CONFIG_AFIRSRC_FILTER) += asrc_afirsrc.o
OBJS-$(CONFIG_ANOISESRC_FILTER) += asrc_anoisesrc.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 2ece5c15c8..c01cae3e84 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -160,6 +160,7 @@ extern const AVFilter ff_af_volume;
extern const AVFilter ff_af_volumedetect;
extern const AVFilter ff_asrc_aevalsrc;
+extern const AVFilter ff_asrc_afdelaysrc;
extern const AVFilter ff_asrc_afirsrc;
extern const AVFilter ff_asrc_anoisesrc;
extern const AVFilter ff_asrc_anullsrc;
diff --git a/libavfilter/asrc_afdelaysrc.c b/libavfilter/asrc_afdelaysrc.c
new file mode 100644
index 0000000000..784771f84b
--- /dev/null
+++ b/libavfilter/asrc_afdelaysrc.c
@@ -0,0 +1,145 @@
+/*
+ * Copyright (c) 2023 Paul B Mahol
+ *
+ * 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/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "filters.h"
+#include "internal.h"
+
+typedef struct AFDelaySrcContext {
+ const AVClass *class;
+
+ double delay;
+ int sample_rate;
+ int nb_samples;
+ int nb_taps;
+
+ int64_t pts;
+} AFDelaySrcContext;
+
+static float sincf(float x)
+{
+ if (x == 0.f)
+ return 1.f;
+ return sinf(M_PI * x) / (M_PI * x);
+}
+
+static int activate(AVFilterContext *ctx)
+{
+ AVFilterLink *outlink = ctx->outputs[0];
+ AFDelaySrcContext *s = ctx->priv;
+ AVFrame *frame = NULL;
+ int nb_samples;
+ float *dst;
+
+ if (!ff_outlink_frame_wanted(outlink))
+ return FFERROR_NOT_READY;
+
+ nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);
+ if (nb_samples <= 0) {
+ ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
+ return 0;
+ }
+
+ if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
+ return AVERROR(ENOMEM);
+
+ dst = (float *)frame->extended_data[0];
+ for (int n = 0; n < nb_samples; n++) {
+ float x = s->pts + n;
+ dst[n] = sincf(x - s->delay) * cosf(M_PI * (x - s->delay) / s->nb_taps) / sincf((x - s->delay) / s->nb_taps);
+ }
+
+ frame->pts = s->pts;
+ s->pts += nb_samples;
+
+ return ff_filter_frame(outlink, frame);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AFDelaySrcContext *s = ctx->priv;
+ static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
+ int sample_rates[] = { s->sample_rate, -1 };
+ static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_NONE };
+ int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
+ if (ret < 0)
+ return ret;
+
+ ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts);
+ if (ret < 0)
+ return ret;
+
+ return ff_set_common_samplerates_from_list(ctx, sample_rates);
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AFDelaySrcContext *s = ctx->priv;
+
+ outlink->sample_rate = s->sample_rate;
+ s->pts = 0;
+ if (s->nb_taps <= 0)
+ s->nb_taps = s->delay * 8 + 1;
+
+ return 0;
+}
+
+static const AVFilterPad afdelaysrc_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+};
+
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define OFFSET(x) offsetof(AFDelaySrcContext, x)
+
+static const AVOption afdelaysrc_options[] = {
+ { "delay", "set fractional delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE,{.dbl=0}, 0, INT16_MAX, AF },
+ { "d", "set fractional delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE,{.dbl=0}, 0, INT16_MAX, AF },
+ { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
+ { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
+ { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
+ { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
+ { "taps", "set number of taps for delay filter", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
+ { "t", "set number of taps for delay filter", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(afdelaysrc);
+
+const AVFilter ff_asrc_afdelaysrc = {
+ .name = "afdelaysrc",
+ .description = NULL_IF_CONFIG_SMALL("Generate a Fractional delay FIR coefficients."),
+ .priv_size = sizeof(AFDelaySrcContext),
+ .priv_class = &afdelaysrc_class,
+ .activate = activate,
+ .inputs = NULL,
+ FILTER_OUTPUTS(afdelaysrc_outputs),
+ FILTER_QUERY_FUNC(query_formats),
+};
--
2.37.2
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^ permalink raw reply [flat|nested] 2+ messages in thread
* Re: [FFmpeg-devel] [PATCH] avfilter: add fractional delay IR source filter
2023-01-08 14:03 [FFmpeg-devel] [PATCH] avfilter: add fractional delay IR source filter Paul B Mahol
@ 2023-01-15 9:45 ` Paul B Mahol
0 siblings, 0 replies; 2+ messages in thread
From: Paul B Mahol @ 2023-01-15 9:45 UTC (permalink / raw)
To: FFmpeg development discussions and patches
On 1/8/23, Paul B Mahol <onemda@gmail.com> wrote:
> Patch attached.
>
will apply soon
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