Signed-off-by: Paul B Mahol <one...@gmail.com>
---
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 0000000000..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; \
+ } \
+ } \
+}
+
+DERIVATIVE(flt, float)
+DERIVATIVE(dbl, double)
+DERIVATIVE(s16, int16_t)
+DERIVATIVE(s32, int32_t)
+
+#define INTEGRAL(name, type) \
+static void aintegral_## 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] = dst[n]; \
+ } \
+ } \
+}
+
+INTEGRAL(flt, float)
+INTEGRAL(dbl, double)
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ADerivativeContext *s = ctx->priv;
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_FLTP: s->filter = aderivative_fltp; break;
+ case AV_SAMPLE_FMT_DBLP: s->filter = aderivative_dblp; break;
+ case AV_SAMPLE_FMT_S32P: s->filter = aderivative_s32p; break;
+ case AV_SAMPLE_FMT_S16P: s->filter = aderivative_s16p; break;
+ }
+
+ if (strcmp(ctx->filter->name, "aintegral"))
+ return 0;
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_FLTP: s->filter = aintegral_fltp; break;
+ case AV_SAMPLE_FMT_DBLP: s->filter = aintegral_dblp; break;
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ADerivativeContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
+
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+
+ if (!s->prev) {
+ s->prev = ff_get_audio_buffer(inlink, 1);
+ if (!s->prev) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ }
+
+ s->filter((void **)out->extended_data, (void **)s->prev->extended_data,
(const void **)in->extended_data,
+ in->nb_samples, in->channels);
+
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ADerivativeContext *s = ctx->priv;
+
+ av_frame_free(&s->prev);
+}
+
+static const AVFilterPad aderivative_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad aderivative_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_aderivative = {
+ .name = "aderivative",
+ .description = NULL_IF_CONFIG_SMALL("Compute derivative of input
audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ADerivativeContext),
+ .uninit = uninit,
+ .inputs = aderivative_inputs,
+ .outputs = aderivative_outputs,
+};
+
+AVFilter ff_af_aintegral = {
+ .name = "aintegral",
+ .description = NULL_IF_CONFIG_SMALL("Compute integral of input audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ADerivativeContext),
+ .uninit = uninit,
+ .inputs = aderivative_inputs,
+ .outputs = aderivative_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index f28f6e47ee..5d3ed0a8a2 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -30,6 +30,7 @@ extern AVFilter ff_af_acopy;
extern AVFilter ff_af_acrossfade;
extern AVFilter ff_af_acrusher;
extern AVFilter ff_af_adelay;
+extern AVFilter ff_af_aderivative;
extern AVFilter ff_af_aecho;
extern AVFilter ff_af_aemphasis;
extern AVFilter ff_af_aeval;
@@ -39,6 +40,7 @@ extern AVFilter ff_af_afir;
extern AVFilter ff_af_aformat;
extern AVFilter ff_af_agate;
extern AVFilter ff_af_aiir;
+extern AVFilter ff_af_aintegral;
extern AVFilter ff_af_ainterleave;
extern AVFilter ff_af_alimiter;
extern AVFilter ff_af_allpass;
--
2.11.0
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