Re: [FFmpeg-devel] [PATCH] avfilter: add aderivative and aintegral filter

[Paul B Mahol]

On 5/14/18, Paul B Mahol  wrote:
> Signed-off-by: Paul B Mahol 
> ---
>  doc/filters.texi |   6 ++
>  libavfilter/Makefile |   2 +
>  libavfilter/af_aderivative.c | 207
> +++
>  libavfilter/allfilters.c |   2 +
>  4 files changed, 217 insertions(+)
>  create mode 100644 libavfilter/af_aderivative.c
>

Will apply.
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[FFmpeg-devel] [PATCH] avfilter: add aderivative and aintegral filter

[Paul B Mahol]

Signed-off-by: Paul B Mahol 
---
 doc/filters.texi |   6 ++
 libavfilter/Makefile |   2 +
 libavfilter/af_aderivative.c | 207 +++
 libavfilter/allfilters.c |   2 +
 4 files changed, 217 insertions(+)
 create mode 100644 libavfilter/af_aderivative.c

diff --git a/doc/filters.texi b/doc/filters.texi
index 30982cb6ab..ba31ed1316 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -585,6 +585,12 @@ adelay=0|500S|700S
 @end example
 @end itemize
 
+@section aderivative, aintegral
+
+Compute derivative/integral of audio stream.
+
+Applying both filters one after another produces original audio.
+
 @section aecho
 
 Apply echoing to the input audio.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index b2d6756e79..717aa83359 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -35,6 +35,8 @@ OBJS-$(CONFIG_ACOPY_FILTER)  += af_acopy.o
 OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
 OBJS-$(CONFIG_ACRUSHER_FILTER)   += af_acrusher.o
 OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
+OBJS-$(CONFIG_ADERIVATIVE_FILTER)+= af_aderivative.o
+OBJS-$(CONFIG_AINTEGRAL_FILTER)  += af_aderivative.o
 OBJS-$(CONFIG_AECHO_FILTER)  += af_aecho.o
 OBJS-$(CONFIG_AEMPHASIS_FILTER)  += af_aemphasis.o
 OBJS-$(CONFIG_AEVAL_FILTER)  += aeval.o
diff --git a/libavfilter/af_aderivative.c b/libavfilter/af_aderivative.c
new file mode 100644
index 00..a591515cbf
--- /dev/null
+++ b/libavfilter/af_aderivative.c
@@ -0,0 +1,207 @@
+/*
+ * 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 "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct ADerivativeContext {
+const AVClass *class;
+AVFrame *prev;
+void (*filter)(void **dst, void **prv, const void **src,
+   int nb_samples, int channels);
+} ADerivativeContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+AVFilterFormats *formats = NULL;
+AVFilterChannelLayouts *layouts = NULL;
+static const enum AVSampleFormat derivative_sample_fmts[] = {
+AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLTP,
+AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_DBLP,
+AV_SAMPLE_FMT_NONE
+};
+static const enum AVSampleFormat integral_sample_fmts[] = {
+AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
+AV_SAMPLE_FMT_NONE
+};
+int ret;
+
+formats = ff_make_format_list(strcmp(ctx->filter->name, "aintegral") ?
+  derivative_sample_fmts : 
integral_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);
+}
+
+#define DERIVATIVE(name, type)  \
+static void aderivative_## name ##p(void **d, void **p, const void **s, \
+int nb_samples, int channels)   \
+{   \
+int n, c;   \
+\
+for (c = 0; c < channels; c++) {\
+const type *src = s[c]; \
+type *dst = d[c];   \
+type *prv = p[c];   \
+\
+for (n = 0; n < nb_samples; n++) {  \
+const type current = src[n];\
+\
+dst[n] = current - prv[0];  \
+prv[0] = current;