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[FFmpeg-devel] avfilter: add asubboost filter

Message ID 20200427180532.2044-1-onemda@gmail.com
State Accepted
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Series [FFmpeg-devel] avfilter: add asubboost filter
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Commit Message

Paul B Mahol April 27, 2020, 6:05 p.m. UTC
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 doc/filters.texi           |  31 ++++++
 libavfilter/Makefile       |   1 +
 libavfilter/af_asubboost.c | 210 +++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c   |   1 +
 4 files changed, 243 insertions(+)
 create mode 100644 libavfilter/af_asubboost.c

Comments

Paul B Mahol April 30, 2020, 7:41 a.m. UTC | #1
On 4/27/20, Paul B Mahol <onemda@gmail.com> wrote:
> Signed-off-by: Paul B Mahol <onemda@gmail.com>
> ---
>  doc/filters.texi           |  31 ++++++
>  libavfilter/Makefile       |   1 +
>  libavfilter/af_asubboost.c | 210 +++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c   |   1 +
>  4 files changed, 243 insertions(+)
>  create mode 100644 libavfilter/af_asubboost.c
>

Will apply shortly.
diff mbox series

Patch

diff --git a/doc/filters.texi b/doc/filters.texi
index 71a6787289..18c8528da7 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2446,6 +2446,37 @@  Number of points where the waveform crosses the zero level axis.
 Rate of Zero crossings and number of audio samples.
 @end table
 
+@section asubboost
+Boost subwoofer frequencies.
+
+The filter accepts the following options:
+
+@table @option
+@item dry
+Set dry gain, how much of original signal is kept. Allowed range is from 0 to 1.
+Default value is 0.5.
+
+@item wet
+Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to 1.
+Default value is 0.8.
+
+@item decay
+Set delay line decay gain value. Allowed range is from 0 to 1.
+Default value is 0.7.
+
+@item feedback
+Set delay line feedback gain value. Allowed range is from 0 to 1.
+Default value is 0.5.
+
+@item cutoff
+Set cutoff frequenciy in herz. Allowed range is 50 to 250.
+Default value is 100.
+
+@item delay
+Set delay. Allowed range is from 1 to 100.
+Default value is 20.
+@end table
+
 @section atempo
 
 Adjust audio tempo.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index e1205eb063..f982afe15f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -86,6 +86,7 @@  OBJS-$(CONFIG_ASPLIT_FILTER)                 += split.o
 OBJS-$(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_ATEMPO_FILTER)                 += af_atempo.o
 OBJS-$(CONFIG_ATRIM_FILTER)                  += trim.o
 OBJS-$(CONFIG_AXCORRELATE_FILTER)            += af_axcorrelate.o
diff --git a/libavfilter/af_asubboost.c b/libavfilter/af_asubboost.c
new file mode 100644
index 0000000000..37ed853687
--- /dev/null
+++ b/libavfilter/af_asubboost.c
@@ -0,0 +1,210 @@ 
+/*
+ * 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 ASubBoostContext {
+    const AVClass *class;
+
+    double dry_gain;
+    double wet_gain;
+    double feedback;
+    double decay;
+    double delay;
+    double cutoff;
+
+    double a0, a1, a2;
+    double b0, b1, b2;
+
+    int write_pos;
+    int buffer_samples;
+
+    AVFrame *i, *o;
+    AVFrame *buffer;
+} ASubBoostContext;
+
+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 config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ASubBoostContext *s = ctx->priv;
+
+    double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
+    double alpha = sin(w0) / 2 * sqrt(2 * (1 / 0.5 - 1) + 2);
+
+    s->a0 =  1 + alpha;
+    s->a1 = -2 * cos(w0);
+    s->a2 =  1 - alpha;
+    s->b0 = (1 - cos(w0)) / 2;
+    s->b1 =  1 - cos(w0);
+    s->b2 = (1 - cos(w0)) / 2;
+
+    s->a1 /= s->a0;
+    s->a2 /= s->a0;
+    s->b0 /= s->a0;
+    s->b1 /= s->a0;
+    s->b2 /= s->a0;
+
+    s->buffer_samples = inlink->sample_rate * s->delay / 1000;
+
+    s->buffer = ff_get_audio_buffer(inlink, s->buffer_samples);
+    s->i = ff_get_audio_buffer(inlink, 2);
+    s->o = ff_get_audio_buffer(inlink, 2);
+    if (!s->buffer || !s->i || !s->o)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    ASubBoostContext *s = ctx->priv;
+    const float wet = s->wet_gain, dry = s->dry_gain, feedback = s->feedback, decay = s->decay;
+    int write_pos;
+    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++) {
+        const double *src = (const double *)in->extended_data[ch];
+        double *dst = (double *)out->extended_data[ch];
+        double *buffer = (double *)s->buffer->extended_data[ch];
+        double *ix = (double *)s->i->extended_data[ch];
+        double *ox = (double *)s->o->extended_data[ch];
+
+        write_pos = s->write_pos;
+        for (int n = 0; n < in->nb_samples; n++) {
+            double out_sample;
+
+            out_sample = src[n] * s->b0 + ix[0] * s->b1 + ix[1] * s->b2 - ox[0] * s->a1 - ox[1] * s->a2;
+            ix[1] = ix[0];
+            ix[0] = src[n];
+            ox[1] = ox[0];
+            ox[0] = out_sample;
+
+            buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
+            dst[n] = src[n] * dry + buffer[write_pos] * wet;
+
+            if (++write_pos >= s->buffer_samples)
+                write_pos = 0;
+        }
+    }
+
+    s->write_pos = write_pos;
+
+    if (out != in)
+        av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    ASubBoostContext *s = ctx->priv;
+
+    av_frame_free(&s->buffer);
+    av_frame_free(&s->i);
+    av_frame_free(&s->o);
+}
+
+#define OFFSET(x) offsetof(ASubBoostContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption asubboost_options[] = {
+    { "dry",      "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.5},  0,   1, FLAGS },
+    { "wet",      "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.8},  0,   1, FLAGS },
+    { "decay",    "set decay",    OFFSET(decay),    AV_OPT_TYPE_DOUBLE, {.dbl=0.7},  0,   1, FLAGS },
+    { "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.5},  0,   1, FLAGS },
+    { "cutoff",   "set cutoff",   OFFSET(cutoff),   AV_OPT_TYPE_DOUBLE, {.dbl=100}, 50, 250, FLAGS },
+    { "delay",    "set delay",    OFFSET(delay),    AV_OPT_TYPE_DOUBLE, {.dbl=20},   1, 100, FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(asubboost);
+
+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_asubboost = {
+    .name           = "asubboost",
+    .description    = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(ASubBoostContext),
+    .priv_class     = &asubboost_class,
+    .uninit         = uninit,
+    .inputs         = inputs,
+    .outputs        = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 6354375e91..1b94501da0 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -79,6 +79,7 @@  extern AVFilter ff_af_asplit;
 extern AVFilter ff_af_asr;
 extern AVFilter ff_af_astats;
 extern AVFilter ff_af_astreamselect;
+extern AVFilter ff_af_asubboost;
 extern AVFilter ff_af_atempo;
 extern AVFilter ff_af_atrim;
 extern AVFilter ff_af_axcorrelate;