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[FFmpeg-devel] avfilter: add asupcut filter

Message ID 20201123185223.14792-1-onemda@gmail.com
State New
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Series [FFmpeg-devel] avfilter: add asupcut filter
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Commit Message

Paul B Mahol Nov. 23, 2020, 6:52 p.m. UTC
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 doc/filters.texi         |  15 +++
 libavfilter/Makefile     |   1 +
 libavfilter/af_asupcut.c | 248 +++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |   1 +
 4 files changed, 265 insertions(+)
 create mode 100644 libavfilter/af_asupcut.c

Comments

Moritz Barsnick Nov. 24, 2020, 11:21 p.m. UTC | #1
On Mon, Nov 23, 2020 at 19:52:23 +0100, Paul B Mahol wrote:
> +@item cutoff
> +Set cutoff frequency in herz. Allowed range is 20000 to 192000.

Nit: Hertz (with "tz" and capitalized).

Moritz
Paul B Mahol Nov. 26, 2020, 12:21 p.m. UTC | #2
Will apply with minor fixes.

On Mon, Nov 23, 2020 at 7:52 PM Paul B Mahol <onemda@gmail.com> wrote:

> Signed-off-by: Paul B Mahol <onemda@gmail.com>
> ---
>  doc/filters.texi         |  15 +++
>  libavfilter/Makefile     |   1 +
>  libavfilter/af_asupcut.c | 248 +++++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c |   1 +
>  4 files changed, 265 insertions(+)
>  create mode 100644 libavfilter/af_asupcut.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 15acae9709..f8a8012b16 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -2586,6 +2586,21 @@ Default value is 20.
>
>  This filter supports the all above options as @ref{commands}.
>
> +@section asupcut
> +Cut super frequencies.
> +
> +The filter accepts the following options:
> +
> +@table @option
> +@item cutoff
> +Set cutoff frequency in herz. Allowed range is 20000 to 192000.
> +Default value is 20000.
> +@end table
> +
> +@subsection Commands
> +
> +This filter supports the all above options as @ref{commands}.
> +
>  @section atempo
>
>  Adjust audio tempo.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 36f3d2d0e4..47094b7157 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -90,6 +90,7 @@ OBJS-$(CONFIG_ASR_FILTER)                    += af_asr.o
>  OBJS-$(CONFIG_ASTATS_FILTER)                 += af_astats.o
>  OBJS-$(CONFIG_ASTREAMSELECT_FILTER)          += f_streamselect.o
> framesync.o
>  OBJS-$(CONFIG_ASUBBOOST_FILTER)              += af_asubboost.o
> +OBJS-$(CONFIG_ASUPCUT_FILTER)                += af_asupcut.o
>  OBJS-$(CONFIG_ATEMPO_FILTER)                 += af_atempo.o
>  OBJS-$(CONFIG_ATRIM_FILTER)                  += trim.o
>  OBJS-$(CONFIG_AXCORRELATE_FILTER)            += af_axcorrelate.o
> diff --git a/libavfilter/af_asupcut.c b/libavfilter/af_asupcut.c
> new file mode 100644
> index 0000000000..4a25a12844
> --- /dev/null
> +++ b/libavfilter/af_asupcut.c
> @@ -0,0 +1,248 @@
> +/*
> + * 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 BiquadCoeffs {
> +    double a1, a2;
> +    double b0, b1, b2;
> +} BiquadCoeffs;
> +
> +typedef struct ASupCutContext {
> +    const AVClass *class;
> +
> +    double cutoff;
> +
> +    int bypass;
> +
> +    BiquadCoeffs coeffs[5];
> +
> +    AVFrame *w;
> +} ASupCutContext;
> +
> +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 int get_coeffs(AVFilterContext *ctx)
> +{
> +    ASupCutContext *s = ctx->priv;
> +    AVFilterLink *inlink = ctx->inputs[0];
> +    double w0 = s->cutoff / inlink->sample_rate;
> +    double K = tan(M_PI * w0);
> +    double q[5];
> +
> +    if (w0 >= 0.5) {
> +        s->bypass = 1;
> +        return 0;
> +    }
> +
> +    q[0] = 0.50623256;
> +    q[1] = 0.56116312;
> +    q[2] = 0.70710678;
> +    q[3] = 1.10134463;
> +    q[4] = 3.19622661;
> +
> +    for (int b = 0; b < 5; b++) {
> +        BiquadCoeffs *coeffs = &s->coeffs[b];
> +        double norm = 1.0 / (1.0 + K / q[b] + K * K);
> +
> +        coeffs->b0 = K * K * norm;
> +        coeffs->b1 = 2.0 * coeffs->b0;
> +        coeffs->b2 = coeffs->b0;
> +        coeffs->a1 = -2.0 * (K * K - 1.0) * norm;
> +        coeffs->a2 = -(1.0 - K / q[b] + K * K) * norm;
> +    }
> +
> +    return 0;
> +}
> +
> +static int config_input(AVFilterLink *inlink)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    ASupCutContext *s = ctx->priv;
> +
> +    s->w = ff_get_audio_buffer(inlink, 2 * 5);
> +    if (!s->w)
> +        return AVERROR(ENOMEM);
> +
> +    return get_coeffs(ctx);
> +}
> +
> +typedef struct ThreadData {
> +    AVFrame *in, *out;
> +} ThreadData;
> +
> +static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr,
> int nb_jobs)
> +{
> +    ASupCutContext *s = ctx->priv;
> +    ThreadData *td = arg;
> +    AVFrame *out = td->out;
> +    AVFrame *in = td->in;
> +    const int start = (in->channels * jobnr) / nb_jobs;
> +    const int end = (in->channels * (jobnr+1)) / nb_jobs;
> +
> +    for (int ch = start; ch < end; ch++) {
> +        const double *src = (const double *)in->extended_data[ch];
> +        double *dst = (double *)out->extended_data[ch];
> +
> +        for (int b = 0; b < 5; b++) {
> +            BiquadCoeffs *coeffs = &s->coeffs[b];
> +            const double a1 = coeffs->a1;
> +            const double a2 = coeffs->a2;
> +            const double b0 = coeffs->b0;
> +            const double b1 = coeffs->b1;
> +            const double b2 = coeffs->b2;
> +            double *w = ((double *)s->w->extended_data[ch]) + b * 2;
> +
> +            for (int n = 0; n < in->nb_samples; n++) {
> +                double sin = b ? dst[n] : src[n];
> +                double sout = sin * b0 + w[0];
> +
> +                w[0] = b1 * sin + w[1] + a1 * sout;
> +                w[1] = b2 * sin + a2 * sout;
> +
> +                dst[n] = sout;
> +            }
> +        }
> +    }
> +
> +    return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    ASupCutContext *s = ctx->priv;
> +    AVFilterLink *outlink = ctx->outputs[0];
> +    ThreadData td;
> +    AVFrame *out;
> +
> +    if (s->bypass)
> +        return ff_filter_frame(outlink, in);
> +
> +    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);
> +    }
> +
> +    td.in = in; td.out = out;
> +    ctx->internal->execute(ctx, filter_channels, &td, NULL,
> FFMIN(inlink->channels,
> +
> ff_filter_get_nb_threads(ctx)));
> +
> +    if (out != in)
> +        av_frame_free(&in);
> +    return ff_filter_frame(outlink, out);
> +}
> +
> +static int process_command(AVFilterContext *ctx, const char *cmd, const
> char *args,
> +                           char *res, int res_len, int flags)
> +{
> +    int ret;
> +
> +    ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
> +    if (ret < 0)
> +        return ret;
> +
> +    return get_coeffs(ctx);
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> +    ASupCutContext *s = ctx->priv;
> +
> +    av_frame_free(&s->w);
> +}
> +
> +#define OFFSET(x) offsetof(ASupCutContext, x)
> +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption asupcut_options[] = {
> +    { "cutoff", "set cutoff frequency", OFFSET(cutoff),
> AV_OPT_TYPE_DOUBLE, {.dbl=20000}, 20000, 192000, FLAGS },
> +    { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(asupcut);
> +
> +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_asupcut = {
> +    .name            = "asupcut",
> +    .description     = NULL_IF_CONFIG_SMALL("Cut super frequencies."),
> +    .query_formats   = query_formats,
> +    .priv_size       = sizeof(ASupCutContext),
> +    .priv_class      = &asupcut_class,
> +    .uninit          = uninit,
> +    .inputs          = inputs,
> +    .outputs         = outputs,
> +    .process_command = process_command,
> +    .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
> +                       AVFILTER_FLAG_SLICE_THREADS,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index fde535d50c..bf7fe2ce49 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -83,6 +83,7 @@ extern AVFilter ff_af_asr;
>  extern AVFilter ff_af_astats;
>  extern AVFilter ff_af_astreamselect;
>  extern AVFilter ff_af_asubboost;
> +extern AVFilter ff_af_asupcut;
>  extern AVFilter ff_af_atempo;
>  extern AVFilter ff_af_atrim;
>  extern AVFilter ff_af_axcorrelate;
> --
> 2.17.1
>
>
diff mbox series

Patch

diff --git a/doc/filters.texi b/doc/filters.texi
index 15acae9709..f8a8012b16 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2586,6 +2586,21 @@  Default value is 20.
 
 This filter supports the all above options as @ref{commands}.
 
+@section asupcut
+Cut super frequencies.
+
+The filter accepts the following options:
+
+@table @option
+@item cutoff
+Set cutoff frequency in herz. Allowed range is 20000 to 192000.
+Default value is 20000.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @section atempo
 
 Adjust audio tempo.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 36f3d2d0e4..47094b7157 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -90,6 +90,7 @@  OBJS-$(CONFIG_ASR_FILTER)                    += af_asr.o
 OBJS-$(CONFIG_ASTATS_FILTER)                 += af_astats.o
 OBJS-$(CONFIG_ASTREAMSELECT_FILTER)          += f_streamselect.o framesync.o
 OBJS-$(CONFIG_ASUBBOOST_FILTER)              += af_asubboost.o
+OBJS-$(CONFIG_ASUPCUT_FILTER)                += af_asupcut.o
 OBJS-$(CONFIG_ATEMPO_FILTER)                 += af_atempo.o
 OBJS-$(CONFIG_ATRIM_FILTER)                  += trim.o
 OBJS-$(CONFIG_AXCORRELATE_FILTER)            += af_axcorrelate.o
diff --git a/libavfilter/af_asupcut.c b/libavfilter/af_asupcut.c
new file mode 100644
index 0000000000..4a25a12844
--- /dev/null
+++ b/libavfilter/af_asupcut.c
@@ -0,0 +1,248 @@ 
+/*
+ * 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 BiquadCoeffs {
+    double a1, a2;
+    double b0, b1, b2;
+} BiquadCoeffs;
+
+typedef struct ASupCutContext {
+    const AVClass *class;
+
+    double cutoff;
+
+    int bypass;
+
+    BiquadCoeffs coeffs[5];
+
+    AVFrame *w;
+} ASupCutContext;
+
+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 int get_coeffs(AVFilterContext *ctx)
+{
+    ASupCutContext *s = ctx->priv;
+    AVFilterLink *inlink = ctx->inputs[0];
+    double w0 = s->cutoff / inlink->sample_rate;
+    double K = tan(M_PI * w0);
+    double q[5];
+
+    if (w0 >= 0.5) {
+        s->bypass = 1;
+        return 0;
+    }
+
+    q[0] = 0.50623256;
+    q[1] = 0.56116312;
+    q[2] = 0.70710678;
+    q[3] = 1.10134463;
+    q[4] = 3.19622661;
+
+    for (int b = 0; b < 5; b++) {
+        BiquadCoeffs *coeffs = &s->coeffs[b];
+        double norm = 1.0 / (1.0 + K / q[b] + K * K);
+
+        coeffs->b0 = K * K * norm;
+        coeffs->b1 = 2.0 * coeffs->b0;
+        coeffs->b2 = coeffs->b0;
+        coeffs->a1 = -2.0 * (K * K - 1.0) * norm;
+        coeffs->a2 = -(1.0 - K / q[b] + K * K) * norm;
+    }
+
+    return 0;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ASupCutContext *s = ctx->priv;
+
+    s->w = ff_get_audio_buffer(inlink, 2 * 5);
+    if (!s->w)
+        return AVERROR(ENOMEM);
+
+    return get_coeffs(ctx);
+}
+
+typedef struct ThreadData {
+    AVFrame *in, *out;
+} ThreadData;
+
+static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+    ASupCutContext *s = ctx->priv;
+    ThreadData *td = arg;
+    AVFrame *out = td->out;
+    AVFrame *in = td->in;
+    const int start = (in->channels * jobnr) / nb_jobs;
+    const int end = (in->channels * (jobnr+1)) / nb_jobs;
+
+    for (int ch = start; ch < end; ch++) {
+        const double *src = (const double *)in->extended_data[ch];
+        double *dst = (double *)out->extended_data[ch];
+
+        for (int b = 0; b < 5; b++) {
+            BiquadCoeffs *coeffs = &s->coeffs[b];
+            const double a1 = coeffs->a1;
+            const double a2 = coeffs->a2;
+            const double b0 = coeffs->b0;
+            const double b1 = coeffs->b1;
+            const double b2 = coeffs->b2;
+            double *w = ((double *)s->w->extended_data[ch]) + b * 2;
+
+            for (int n = 0; n < in->nb_samples; n++) {
+                double sin = b ? dst[n] : src[n];
+                double sout = sin * b0 + w[0];
+
+                w[0] = b1 * sin + w[1] + a1 * sout;
+                w[1] = b2 * sin + a2 * sout;
+
+                dst[n] = sout;
+            }
+        }
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ASupCutContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    ThreadData td;
+    AVFrame *out;
+
+    if (s->bypass)
+        return ff_filter_frame(outlink, in);
+
+    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);
+    }
+
+    td.in = in; td.out = out;
+    ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
+                                                            ff_filter_get_nb_threads(ctx)));
+
+    if (out != in)
+        av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+
+static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
+                           char *res, int res_len, int flags)
+{
+    int ret;
+
+    ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
+    if (ret < 0)
+        return ret;
+
+    return get_coeffs(ctx);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    ASupCutContext *s = ctx->priv;
+
+    av_frame_free(&s->w);
+}
+
+#define OFFSET(x) offsetof(ASupCutContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption asupcut_options[] = {
+    { "cutoff", "set cutoff frequency", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=20000}, 20000, 192000, FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(asupcut);
+
+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_asupcut = {
+    .name            = "asupcut",
+    .description     = NULL_IF_CONFIG_SMALL("Cut super frequencies."),
+    .query_formats   = query_formats,
+    .priv_size       = sizeof(ASupCutContext),
+    .priv_class      = &asupcut_class,
+    .uninit          = uninit,
+    .inputs          = inputs,
+    .outputs         = outputs,
+    .process_command = process_command,
+    .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
+                       AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index fde535d50c..bf7fe2ce49 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -83,6 +83,7 @@  extern AVFilter ff_af_asr;
 extern AVFilter ff_af_astats;
 extern AVFilter ff_af_astreamselect;
 extern AVFilter ff_af_asubboost;
+extern AVFilter ff_af_asupcut;
 extern AVFilter ff_af_atempo;
 extern AVFilter ff_af_atrim;
 extern AVFilter ff_af_axcorrelate;