avfilter: add fractional delay IR source filter

This commit is contained in:
Paul B Mahol 2023-01-08 13:53:39 +01:00
parent 1ab0f83b0a
commit ac7d21284b
6 changed files with 204 additions and 1 deletions

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@ -28,6 +28,7 @@ version <next>:
- showcwt multimedia filter
- corr video filter
- adrc audio filter
- afdelaysrc audio filter
version 5.1:

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@ -7519,6 +7519,33 @@ 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.
@item channel_layout, c
Specifies the channel layout, and can be a string representing a channel layout.
The default value of @var{channel_layout} is "stereo".
@end table
@section afirsrc
Generate a FIR coefficients using frequency sampling method.

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@ -171,6 +171,7 @@ 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

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@ -159,6 +159,7 @@ 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;

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@ -0,0 +1,173 @@
/*
* 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;
AVChannelLayout chlayout;
char *chlayout_str;
int64_t pts;
} AFDelaySrcContext;
static av_cold int init(AVFilterContext *ctx)
{
AFDelaySrcContext *s = ctx->priv;
int ret;
ret = ff_parse_channel_layout(&s->chlayout, NULL, s->chlayout_str, ctx);
if (ret < 0)
return ret;
return 0;
}
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);
}
for (int ch = 1; ch < frame->ch_layout.nb_channels; ch++)
memcpy(frame->extended_data[ch], dst, sizeof(*dst) * nb_samples);
frame->pts = s->pts;
s->pts += nb_samples;
return ff_filter_frame(outlink, frame);
}
static int query_formats(AVFilterContext *ctx)
{
AFDelaySrcContext *s = ctx->priv;
AVChannelLayout chlayouts[] = { s->chlayout, { 0 } };
int sample_rates[] = { s->sample_rate, -1 };
static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP,
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,
},
};
static av_cold void uninit(AVFilterContext *ctx)
{
AFDelaySrcContext *s = ctx->priv;
av_channel_layout_uninit(&s->chlayout);
}
#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 },
{ "channel_layout", "set channel layout", OFFSET(chlayout_str),AV_OPT_TYPE_STRING,{.str="stereo"},0, 0, AF },
{ "c", "set channel layout", OFFSET(chlayout_str),AV_OPT_TYPE_STRING,{.str="stereo"},0, 0, 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,
.init = init,
.activate = activate,
.uninit = uninit,
.inputs = NULL,
FILTER_OUTPUTS(afdelaysrc_outputs),
FILTER_QUERY_FUNC(query_formats),
};

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@ -31,7 +31,7 @@
#include "version_major.h"
#define LIBAVFILTER_VERSION_MINOR 53
#define LIBAVFILTER_VERSION_MINOR 54
#define LIBAVFILTER_VERSION_MICRO 100