diff mbox

[FFmpeg-devel,OPW] OPW Project Proposal

Message ID CAC7Kmr1VbmSnf4VRCu+0jbAroetrMefPbmRNnY5P6ZpNoo8HzA@mail.gmail.com
State Superseded
Headers show

Commit Message

Pallavi Kumari Nov. 4, 2016, 2:16 a.m. UTC
Updated patch attached. PFA

On Thu, Nov 3, 2016 at 2:00 AM, Michael Niedermayer <michael@niedermayer.cc>
wrote:

> On Thu, Nov 03, 2016 at 01:10:26AM +0530, Pallavi Kumari wrote:
> > Necessary changes has been done. PFA.
> >
> > Usage:
> >
> > ./ffmpeg -i kpg.mp3 -filter_complex peakpoints=wsize=16 -f null -
> >
> > On Wed, Nov 2, 2016 at 6:14 AM, Michael Niedermayer
> <michael@niedermayer.cc>
> > wrote:
> >
> > > On Wed, Nov 02, 2016 at 05:00:09AM +0530, Pallavi Kumari wrote:
> > > > Hi Michael,
> > > >
> > > > I have attached a working patch with the mail. PFA.
> > > >
> > > > Usage:
> > > >
> > >
> > > > ./ffmpeg -i kpg.mp3 -filter_complex peakpoints=input=kpg.mp3:
> wsize=16
> > >
> > > I realize now, theres a mistake in this, you must provide a output
> > > as in
> > > ./ffmpeg -i kpg.mp3 -af peakpoints -f null -
> > >
> > > without some output like  "-f null -" it wont read the file fully and
> > > wont pass it through filter_frame()
> > >
> > > you could see this failure as in:
> > > ./ffmpeg -i ~/videos/matrixbench_mpeg2.mpg -af volumedetect  -f null -
> > > vs.
> > > ./ffmpeg -i ~/videos/matrixbench_mpeg2.mpg -af volumedetect
> > >
> > > you get the histogram from the volume detect filter in the first case
> > > but not the 2nd.
> > >
> > > [...]
> > > --
> > > Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC7
> 87040B0FAB
> > >
> > > Many things microsoft did are stupid, but not doing something just
> because
> > > microsoft did it is even more stupid. If everything ms did were stupid
> they
> > > would be bankrupt already.
> > >
>
> >  af_peakpoints.c |  226 ++++++++++++++++++++++++++++++
> ++++++++++++++++++++++++++
> >  1 file changed, 226 insertions(+)
> > 206d91b47bc6066dd01db1c3369d4674ac95f04c  0001-avfilter-added-
> peakpoints-filter.patch
> > From e10f73d363d0313774bcb132b3b1f2417fcfba11 Mon Sep 17 00:00:00 2001
> > From: Atana <atana@openmailbox.org>
> > Date: Thu, 3 Nov 2016 01:05:51 +0530
> > Subject: [PATCH] avfilter: added peakpoints filter
> >
> > ---
> >  libavfilter/af_peakpoints.c | 226 ++++++++++++++++++++++++++++++
> ++++++++++++++
> >  1 file changed, 226 insertions(+)
> >  create mode 100644 libavfilter/af_peakpoints.c
>
> This is missing changes to the Makefile and libavfilter/allfilters*
>
>
> >
> > diff --git a/libavfilter/af_peakpoints.c b/libavfilter/af_peakpoints.c
> > new file mode 100644
> > index 0000000..9265c47
> > --- /dev/null
> > +++ b/libavfilter/af_peakpoints.c
> > @@ -0,0 +1,226 @@
> > +/*
> > + * Copyright (c) 2016 Atana
> > + *
> > + * 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 "libavcodec/avcodec.h"
> > +#include "libavcodec/avfft.h"
> > +#include "libavformat/avformat.h"
> > +#include "libswscale/swscale.h"
> > +#include "avfilter.h"
> > +#include "audio.h"
> > +#include "libavutil/opt.h"
> > +
> > +
> > +/* Structure to contain peak points context */
> > +typedef struct {
> > +    const AVClass *class;
> > +    double *data;
> > +    int nsamples;
> > +    int index;
> > +    double *peaks;
> > +    int size; // number of peaks
> > +    int windowSize;
> > +    //char *inputFile;
> > +} PeakPointsContext;
> > +
> > +/* returns maximum value from an array conditioned on start and end
> index */
> > +static double getMax(double *res_arr, int startIndex, int endIndex) {
> > +    int i;
> > +    double max = res_arr[startIndex];
> > +    for (i = startIndex; i <= endIndex; i++) {
> > +         if (res_arr[i] > max) {
> > +             max = res_arr[i];
> > +         }
> > +    }
> > +    return max;
> > +}
> > +
> > +/* Stores peak frequency for each window(of chunkSize) in peaks array */
> > +static void getPeakPointInChunk(int chunkSize, double *res_arr, int
> size, double *peaks) {
> > +    int i = 0, peakIndex = 0;
> > +    int startIndex = 0;
> > +    double max;
> > +    // get a chunk and find max value in it
> > +    while (i < size) {
> > +         if (i % chunkSize-1 == 0) {
> > +            max = getMax(res_arr, startIndex, i);
> > +             peaks[peakIndex++] = max;
> > +             startIndex = startIndex + chunkSize;
> > +         }
> > +        i += 1;
> > +    }
> > +}
> > +
> > +/* Get peaks points from windowed frequency domain data*/
> > +static int getPeakPoints(PeakPointsContext *ppc) {
> > +    int i, m, k, size, chunkSize, pSize, chunkSampleSize, resSize;
> > +    double *fft_res;
> > +    void *avc;
> > +    RDFTContext *rdftC;
> > +    FFTSample *data;
> > +
> > +    size = ppc->index;
> > +    m = log2(ppc->windowSize);
> > +    chunkSize = ppc->windowSize;
> > +    chunkSampleSize = size/chunkSize;
> > +    resSize = chunkSize * chunkSampleSize;
> > +
> > +    fft_res = av_malloc(sizeof(double) * resSize);
> > +
> > +    if (!fft_res) {
> > +        av_log(avc, AV_LOG_ERROR, "Cann't allocate memmory for storing
> fft data\n");
> > +        return 0;
> > +    }
> > +
> > +
> > +    rdftC = av_rdft_init(m, DFT_R2C);
>
> > +    data = av_malloc(sizeof(FFTSample)*chunkSize);
>
> see av_malloc_array() (it avoids potential issues with the multiply
> overflowing)
>
>
> > +
> > +    if (!data) {
> > +        av_log(avc, AV_LOG_ERROR, "Cann't allocate memmory for chunk
> fft data\n");
> > +        return 0;
> > +    }
> > +    // FFT transform for windowed time domain data
> > +    // window is of size chunkSize
> > +    k = 0;
> > +    while (k < resSize) {
> > +        //copy data
> > +        for (i = 0; i < chunkSize; i++) {
> > +            data[i] = ppc->data[i+k];
> > +        }
> > +        //calculate FFT
> > +        av_rdft_calc(rdftC, data);
> > +        for (i = 0; i < chunkSize; i++) {
> > +         fft_res[i+k] = data[i];
> > +        }
> > +        k = k + chunkSize;
> > +    }
> > +
> > +    av_rdft_end(rdftC);
> > +    pSize = resSize/chunkSize;
> > +    ppc->size = pSize;
> > +    ppc->peaks = av_malloc(sizeof(double)*pSize);
> > +
> > +    if (!ppc->peaks) {
> > +        av_log(avc, AV_LOG_ERROR, "Cann't allocate memory for peak
> storage\n");
> > +        return 0;
> > +    }
> > +
> > +    getPeakPointInChunk(chunkSize, fft_res, resSize, ppc->peaks);
> > +    return 1;
> > +}
> > +
> > +
> > +#define OFFSET(x) offsetof(PeakPointsContext, x)
> > +
> > +static const AVOption peakpoints_options[] = {
> > +    { "wsize",  "set window size", OFFSET(windowSize),
> AV_OPT_TYPE_INT,    {.i64=16},    0, INT_MAX},
> > +    { NULL },
> > +};
> > +
> > +AVFILTER_DEFINE_CLASS(peakpoints);
> > +
>
> > +static av_cold int init(AVFilterContext *ctx)
> > +{
> > +    PeakPointsContext *p = ctx->priv;
> > +
> > +    if (p->windowSize < 16) {
> > +         av_log(ctx, AV_LOG_ERROR, "window size must be greater than or
> equal to 16\n");
> > +        return AVERROR(EINVAL);
> > +    }
> > +
> > +    p->index = 0;
> > +    p->size = 0;
>
> > +    p->data = av_malloc(sizeof(double)*10000);
>
> nothing gurantees that 10000 or any constant is large enough
> indeed it is not guranteed that all the decoded audio would fit in
> memory.
> What probably makes most sense is executing the
> fft from filter_frame() every time there is sufficient new data
> and overwriting the old data with new instead of trying to store
> all audio data
>
>
> > +
> > +    if (!p->data) {
> > +        av_log(ctx, AV_LOG_ERROR, "Cann't allocate memmory for audio
> data\n");
> > +        return AVERROR(EINVAL);
> > +    }
> > +
> > +    return 0;
> > +}
> > +
> > +static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
> > +{
> > +    AVFilterContext *ctx = inlink->dst;
> > +    PeakPointsContext *p = ctx->priv;
> > +
> > +    // store audio data
>
> > +    p->data[p->index] = (double)*samples->data[0];
>
> There are multiple channels and multiple samples
> this uses just the first sample of the first channel
>
> samples->nb_samples contains the number of samples (as in time)
> and av_frame_get_channels(samples) is the number of channels
>
> you can see in af_volumedetect.c how to access the samples of the
> channels.
> I guess its ok to use just the first channel for now but all samples
> of the frames (timewise) should be used
>
>
> > +    p->index = p->index + 1;
> > +
> > +    return ff_filter_frame(inlink->dst->outputs[0], samples);
> > +}
> > +
> > +static void ppointsStats(AVFilterContext *ctx, PeakPointsContext *p) {
> > +    int i, ret;
> > +    ret = getPeakPoints(p);
> > +
> > +    if (ret && p->size) {
> > +     // print peaks
> > +        av_log(ctx, AV_LOG_INFO, "######## Peak points are ########\n");
> > +        for (i = 0; i < p->size; i++) {
> > +             av_log(ctx, AV_LOG_INFO, "%f\n", p->peaks[i]);
> > +        }
> > +    } else if (p->size || !ret) {
> > +        av_log(ctx, AV_LOG_ERROR, "Peak points not retrieved\n");
> > +        return;
> > +    }
> > +}
> > +
> > +static av_cold void uninit(AVFilterContext *ctx)
> > +{
> > +    PeakPointsContext *p = ctx->priv;
> > +
> > +    ppointsStats(ctx, p);
> > +
> > +    // free allocated memories
> > +    av_freep(&p->data);
> > +    av_freep(&p->peaks);
> > +}
> > +
> > +static const AVFilterPad peakpoints_inputs[] = {
> > +    {
> > +        .name         = "default",
> > +        .type         = AVMEDIA_TYPE_AUDIO,
> > +        .filter_frame = filter_frame,
> > +    },
> > +    { NULL }
> > +};
> > +
> > +static const AVFilterPad peakpoints_outputs[] = {
> > +    {
> > +        .name = "default",
> > +        .type = AVMEDIA_TYPE_AUDIO,
> > +    },
> > +    { NULL }
> > +};
> > +
> > +AVFilter ff_af_peakpoints = {
> > +    .name          = "peakpoints",
> > +    .description   = NULL_IF_CONFIG_SMALL("peak points from frequency
> domain windowed data."),
> > +    .init          = init,
> > +    .uninit        = uninit,
>
> > +    //.query_formats = query_formats,
>
> without query_formats you cannot be sure that the data you get is
> if double type, see volumedetect for an example implementation, you
> only need to adapt it so it requests double and not S16 integers
>
>
> [...]
>
>
> --
> Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> The greatest way to live with honor in this world is to be what we pretend
> to be. -- Socrates
>

Comments

Michael Niedermayer Nov. 4, 2016, 9:59 a.m. UTC | #1
On Fri, Nov 04, 2016 at 07:46:34AM +0530, Pallavi Kumari wrote:
> Updated patch attached. PFA
[...]

>  Makefile        |    1 
>  af_peakpoints.c |  263 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++
>  allfilters.c    |    1 
>  version.h       |    2 
>  4 files changed, 266 insertions(+), 1 deletion(-)
> 7f6affa8dcb632cdea162553a18ab1489f0783ce  0001-avfilter-added-peakpoints-filter.patch
> From 46be941c87713f8afee686eed0262ca59a2896fd Mon Sep 17 00:00:00 2001
> From: Atana <atana@openmailbox.org>
> Date: Fri, 4 Nov 2016 07:43:29 +0530
> Subject: [PATCH] avfilter: added peakpoints filter
> 
> ---
>  libavfilter/Makefile        |   1 +
>  libavfilter/af_peakpoints.c | 263 ++++++++++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c    |   1 +
>  libavfilter/version.h       |   2 +-
>  4 files changed, 266 insertions(+), 1 deletion(-)
>  create mode 100644 libavfilter/af_peakpoints.c
> 
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 7ed4696..1a18902 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -96,6 +96,7 @@ OBJS-$(CONFIG_LADSPA_FILTER)                 += af_ladspa.o
>  OBJS-$(CONFIG_LOUDNORM_FILTER)               += af_loudnorm.o
>  OBJS-$(CONFIG_LOWPASS_FILTER)                += af_biquads.o
>  OBJS-$(CONFIG_PAN_FILTER)                    += af_pan.o
> +OBJS-$(CONFIG_PEAKPOINTS_FILTER)             += af_peakpoints.o
>  OBJS-$(CONFIG_REPLAYGAIN_FILTER)             += af_replaygain.o
>  OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o
>  OBJS-$(CONFIG_RUBBERBAND_FILTER)             += af_rubberband.o
> diff --git a/libavfilter/af_peakpoints.c b/libavfilter/af_peakpoints.c
> new file mode 100644
> index 0000000..da108ca
> --- /dev/null
> +++ b/libavfilter/af_peakpoints.c
> @@ -0,0 +1,263 @@
> +/*
> + * Copyright (c) 2016 Atana
> + *
> + * 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 "libavcodec/avcodec.h"
> +#include "libavcodec/avfft.h"
> +#include "libavformat/avformat.h"
> +#include "libswscale/swscale.h"
> +#include "avfilter.h"
> +#include "audio.h"
> +#include "libavutil/opt.h"
> +
> +#define SIZECHECK 4096
> +
> +/* Structure to contain peak points context */
> +typedef struct {
> +    const AVClass *class;
> +    double *data;
> +    int nsamples;
> +    int index;
> +    int isOnce;
> +    int buffFlag;
> +    double *peaks;
> +    int size; // number of peaks
> +    int windowSize;
> +    //char *inputFile;
> +} PeakPointsContext;
> +
> +/* returns maximum value from an array conditioned on start and end index */
> +static double getMax(double *res_arr, int startIndex, int endIndex) {
> +    int i;
> +    double max = res_arr[startIndex];
> +    for (i = startIndex; i <= endIndex; i++) {
> +	    if (res_arr[i] > max) {
> +	        max = res_arr[i];
> +	    }
> +    }

tabs are forbidden in ffmpeg git


> +    return max;
> +}
> +
> +/* Stores peak frequency for each window(of chunkSize) in peaks array */
> +static void getPeakPointInChunk(int chunkSize, double *res_arr, int size, double *peaks) {
> +    int i = 0, peakIndex = 0;
> +    int startIndex = 0;
> +    double max;
> +    // get a chunk and find max value in it
> +    while (i < size) {
> +	    if (i % chunkSize-1 == 0) {
> +            max = getMax(res_arr, startIndex, i);
> +	        peaks[peakIndex++] = max;
> +	        startIndex = startIndex + chunkSize;
> +	    }
> +        i += 1;
> +    }
> +}
> +
> +/* Get peaks points from windowed frequency domain data*/
> +static int getPeakPoints(PeakPointsContext *ppc) {
> +    int i, m, k, size, chunkSize, pSize, chunkSampleSize, resSize;
> +    double *fft_res;
> +    void *avc;
> +    RDFTContext *rdftC;
> +    FFTSample *data;
> +
> +    size = ppc->index;
> +    m = log2(ppc->windowSize);
> +    chunkSize = ppc->windowSize;
> +    chunkSampleSize = size/chunkSize;
> +    resSize = chunkSize * chunkSampleSize;
> +

> +    fft_res = av_malloc_array(resSize, sizeof(double));

This seems never freed


> +
> +    if (!fft_res) {
> +        av_log(avc, AV_LOG_ERROR, "Cann't allocate memmory for storing fft data\n");
> +        return 0;
> +    }
> +
> +
> +    rdftC = av_rdft_init(m, DFT_R2C);
> +    data = av_malloc_array(chunkSize, sizeof(FFTSample));
> +
> +    if (!data) {
> +        av_log(avc, AV_LOG_ERROR, "Cann't allocate memmory for chunk fft data\n");
> +        return 0;
> +    }
> +    // FFT transform for windowed time domain data
> +    // window is of size chunkSize
> +    k = 0;
> +    while (k < resSize) {
> +        //copy data
> +        for (i = 0; i < chunkSize; i++) {
> +            data[i] = ppc->data[i+k];
> +        }
> +        //calculate FFT
> +        av_rdft_calc(rdftC, data);
> +        for (i = 0; i < chunkSize; i++) {
> +	    fft_res[i+k] = data[i];
> +        }
> +        k = k + chunkSize;
> +    }
> +
> +    av_rdft_end(rdftC);
> +    pSize = resSize/chunkSize;
> +    ppc->size = pSize;

> +    ppc->peaks = av_malloc_array(pSize, sizeof(double));

This is allocated potentially durng each filter_frame
but only deallocated once at the end, all except the last thus leak


[...]
> +static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    PeakPointsContext *p = ctx->priv;
> +    int i, nb_samples = samples->nb_samples;
> +
> +    // store audio data
> +    for (i = 0; i < nb_samples; i++) {

> +        p->data[p->index] = (double)samples->data[0][i];

This reads the double values as 8bit integers, you have to cast to
a pointer to double and dereference it to read the double values


[...]
diff mbox

Patch

From 46be941c87713f8afee686eed0262ca59a2896fd Mon Sep 17 00:00:00 2001
From: Atana <atana@openmailbox.org>
Date: Fri, 4 Nov 2016 07:43:29 +0530
Subject: [PATCH] avfilter: added peakpoints filter

---
 libavfilter/Makefile        |   1 +
 libavfilter/af_peakpoints.c | 263 ++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c    |   1 +
 libavfilter/version.h       |   2 +-
 4 files changed, 266 insertions(+), 1 deletion(-)
 create mode 100644 libavfilter/af_peakpoints.c

diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 7ed4696..1a18902 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -96,6 +96,7 @@  OBJS-$(CONFIG_LADSPA_FILTER)                 += af_ladspa.o
 OBJS-$(CONFIG_LOUDNORM_FILTER)               += af_loudnorm.o
 OBJS-$(CONFIG_LOWPASS_FILTER)                += af_biquads.o
 OBJS-$(CONFIG_PAN_FILTER)                    += af_pan.o
+OBJS-$(CONFIG_PEAKPOINTS_FILTER)             += af_peakpoints.o
 OBJS-$(CONFIG_REPLAYGAIN_FILTER)             += af_replaygain.o
 OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o
 OBJS-$(CONFIG_RUBBERBAND_FILTER)             += af_rubberband.o
diff --git a/libavfilter/af_peakpoints.c b/libavfilter/af_peakpoints.c
new file mode 100644
index 0000000..da108ca
--- /dev/null
+++ b/libavfilter/af_peakpoints.c
@@ -0,0 +1,263 @@ 
+/*
+ * Copyright (c) 2016 Atana
+ *
+ * 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 "libavcodec/avcodec.h"
+#include "libavcodec/avfft.h"
+#include "libavformat/avformat.h"
+#include "libswscale/swscale.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "libavutil/opt.h"
+
+#define SIZECHECK 4096
+
+/* Structure to contain peak points context */
+typedef struct {
+    const AVClass *class;
+    double *data;
+    int nsamples;
+    int index;
+    int isOnce;
+    int buffFlag;
+    double *peaks;
+    int size; // number of peaks
+    int windowSize;
+    //char *inputFile;
+} PeakPointsContext;
+
+/* returns maximum value from an array conditioned on start and end index */
+static double getMax(double *res_arr, int startIndex, int endIndex) {
+    int i;
+    double max = res_arr[startIndex];
+    for (i = startIndex; i <= endIndex; i++) {
+	    if (res_arr[i] > max) {
+	        max = res_arr[i];
+	    }
+    }
+    return max;
+}
+
+/* Stores peak frequency for each window(of chunkSize) in peaks array */
+static void getPeakPointInChunk(int chunkSize, double *res_arr, int size, double *peaks) {
+    int i = 0, peakIndex = 0;
+    int startIndex = 0;
+    double max;
+    // get a chunk and find max value in it
+    while (i < size) {
+	    if (i % chunkSize-1 == 0) {
+            max = getMax(res_arr, startIndex, i);
+	        peaks[peakIndex++] = max;
+	        startIndex = startIndex + chunkSize;
+	    }
+        i += 1;
+    }
+}
+
+/* Get peaks points from windowed frequency domain data*/
+static int getPeakPoints(PeakPointsContext *ppc) {
+    int i, m, k, size, chunkSize, pSize, chunkSampleSize, resSize;
+    double *fft_res;
+    void *avc;
+    RDFTContext *rdftC;
+    FFTSample *data;
+
+    size = ppc->index;
+    m = log2(ppc->windowSize);
+    chunkSize = ppc->windowSize;
+    chunkSampleSize = size/chunkSize;
+    resSize = chunkSize * chunkSampleSize;
+
+    fft_res = av_malloc_array(resSize, sizeof(double));
+
+    if (!fft_res) {
+        av_log(avc, AV_LOG_ERROR, "Cann't allocate memmory for storing fft data\n");
+        return 0;
+    }
+
+
+    rdftC = av_rdft_init(m, DFT_R2C);
+    data = av_malloc_array(chunkSize, sizeof(FFTSample));
+
+    if (!data) {
+        av_log(avc, AV_LOG_ERROR, "Cann't allocate memmory for chunk fft data\n");
+        return 0;
+    }
+    // FFT transform for windowed time domain data
+    // window is of size chunkSize
+    k = 0;
+    while (k < resSize) {
+        //copy data
+        for (i = 0; i < chunkSize; i++) {
+            data[i] = ppc->data[i+k];
+        }
+        //calculate FFT
+        av_rdft_calc(rdftC, data);
+        for (i = 0; i < chunkSize; i++) {
+	    fft_res[i+k] = data[i];
+        }
+        k = k + chunkSize;
+    }
+
+    av_rdft_end(rdftC);
+    pSize = resSize/chunkSize;
+    ppc->size = pSize;
+    ppc->peaks = av_malloc_array(pSize, sizeof(double));
+
+    if (!ppc->peaks) {
+        av_log(avc, AV_LOG_ERROR, "Cann't allocate memory for peak storage\n");
+        return 0;
+    }
+
+    getPeakPointInChunk(chunkSize, fft_res, resSize, ppc->peaks);
+    av_freep(&data);
+    return 1;
+}
+
+
+#define OFFSET(x) offsetof(PeakPointsContext, x)
+
+static const AVOption peakpoints_options[] = {
+    { "wsize",  "set window size", OFFSET(windowSize),  AV_OPT_TYPE_INT,    {.i64=16},    0, INT_MAX},
+    { NULL },
+};
+
+AVFILTER_DEFINE_CLASS(peakpoints);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    PeakPointsContext *p = ctx->priv;
+
+    if (p->windowSize < 16 || p->windowSize > SIZECHECK) {
+	    av_log(ctx, AV_LOG_ERROR, "window size must be in range 16 to %d\n", SIZECHECK);
+        return AVERROR(EINVAL);
+    }
+
+    p->index = 0;
+    p->size = 0;
+    p->isOnce = 1;
+    p->data = av_malloc_array(SIZECHECK, sizeof(double));
+
+    if (!p->data) {
+        av_log(ctx, AV_LOG_ERROR, "Cann't allocate memmory for audio data\n");
+        return AVERROR(EINVAL);
+    }
+
+    return 0;
+}
+
+static void ppointsStats(AVFilterContext *ctx, PeakPointsContext *p) {
+    int i, ret;
+    ret = getPeakPoints(p);
+
+    if (ret && p->size) {
+    	// print peaks
+        if (p->isOnce) {
+            av_log(ctx, AV_LOG_INFO, "######## Peak points are ########\n");
+            p->isOnce = 0;
+        }
+        for (i = 0; i < p->size; i++) {
+	        av_log(ctx, AV_LOG_INFO, "%f\n", p->peaks[i]);
+        }
+    } else if (p->size || !ret) {
+        av_log(ctx, AV_LOG_ERROR, "Peak points not retrieved\n");
+        return;
+    }
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_DBL,
+        AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+    AVFilterFormats *formats;
+
+    if (!(formats = ff_make_format_list(sample_fmts)))
+        return AVERROR(ENOMEM);
+    return ff_set_common_formats(ctx, formats);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
+{
+    AVFilterContext *ctx = inlink->dst;
+    PeakPointsContext *p = ctx->priv;
+    int i, nb_samples = samples->nb_samples;
+
+    // store audio data
+    for (i = 0; i < nb_samples; i++) {
+        p->data[p->index] = (double)samples->data[0][i];
+        p->buffFlag = 1;
+        p->index = p->index + 1;
+
+        // size check
+        if (p->index == SIZECHECK) {
+            // get peak points stats
+            ppointsStats(ctx, p);
+            p->index = 0;
+            p->buffFlag = 1;
+        }
+    }
+
+    return ff_filter_frame(inlink->dst->outputs[0], samples);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    PeakPointsContext *p = ctx->priv;
+
+    // if audio data in buffer get peak points
+    if (p->buffFlag) {
+        ppointsStats(ctx, p);
+    }
+
+    // free allocated memories
+    av_freep(&p->data);
+    av_freep(&p->peaks);
+}
+
+static const AVFilterPad peakpoints_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad peakpoints_outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_peakpoints = {
+    .name          = "peakpoints",
+    .description   = NULL_IF_CONFIG_SMALL("peak points from frequency domain windowed data."),
+    .init          = init,
+    .uninit        = uninit,
+    .query_formats = query_formats,
+    .priv_size     = sizeof(PeakPointsContext),
+    .inputs        = peakpoints_inputs,
+    .outputs       = peakpoints_outputs,
+    .priv_class    = &peakpoints_class,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 82a65ee..b3b0330 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -114,6 +114,7 @@  void avfilter_register_all(void)
     REGISTER_FILTER(LOUDNORM,       loudnorm,       af);
     REGISTER_FILTER(LOWPASS,        lowpass,        af);
     REGISTER_FILTER(PAN,            pan,            af);
+    REGISTER_FILTER(PEAKPOINTS,     peakpoints,     af);
     REGISTER_FILTER(REPLAYGAIN,     replaygain,     af);
     REGISTER_FILTER(RESAMPLE,       resample,       af);
     REGISTER_FILTER(RUBBERBAND,     rubberband,     af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 82a5f63..b8c9b81 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@ 
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  66
+#define LIBAVFILTER_VERSION_MINOR  67
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
-- 
1.9.1