@@ -6,6 +6,7 @@ version <next>:

 - tmix filter

 - amplify filter

 - fftdnoiz filter

+- aderivative and aintegral audio filters

 version 4.0:

@@ -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.

@@ -35,6 +35,7 @@ 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_AECHO_FILTER)                  += af_aecho.o

 OBJS-$(CONFIG_AEMPHASIS_FILTER)              += af_aemphasis.o

 OBJS-$(CONFIG_AEVAL_FILTER)                  += aeval.o

@@ -44,6 +45,7 @@ OBJS-$(CONFIG_AFIR_FILTER)                   += af_afir.o

 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o

 OBJS-$(CONFIG_AGATE_FILTER)                  += af_agate.o

 OBJS-$(CONFIG_AIIR_FILTER)                   += af_aiir.o

+OBJS-$(CONFIG_AINTEGRAL_FILTER)              += af_aderivative.o

 OBJS-$(CONFIG_AINTERLEAVE_FILTER)            += f_interleave.o

 OBJS-$(CONFIG_ALIMITER_FILTER)               += af_alimiter.o

 OBJS-$(CONFIG_ALLPASS_FILTER)                += af_biquads.o

new file mode 100644

@@ -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,

+};

@@ -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;

@@ -30,7 +30,7 @@

 #include "libavutil/version.h"

 #define LIBAVFILTER_VERSION_MAJOR   7

-#define LIBAVFILTER_VERSION_MINOR  22

+#define LIBAVFILTER_VERSION_MINOR  23

 #define LIBAVFILTER_VERSION_MICRO 100

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \