patch attached.
On Fri, Nov 4, 2016 at 3:29 PM, Michael Niedermayer <mich...@niedermayer.cc>
wrote:
> 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 <at...@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
>
>
> [...]
>
>
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> Breaking DRM is a little like attempting to break through a door even
> though the window is wide open and the only thing in the house is a bunch
> of things you dont want and which you would get tomorrow for free anyway
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
>
From 5b7efa32b1e3970d6accbb23fee02331278b118b Mon Sep 17 00:00:00 2001
From: Atana <at...@openmailbox.org>
Date: Fri, 4 Nov 2016 07:43:29 +0530
Subject: [PATCH] avfilter: added peakpoints filter
---
libavfilter/Makefile | 1 +
libavfilter/af_peakpoints.c | 267 ++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
4 files changed, 270 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..cb64b82
--- /dev/null
+++ b/libavfilter/af_peakpoints.c
@@ -0,0 +1,267 @@
+/*
+ * 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);
+ av_freep(&fft_res);
+ 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]);
+ }
+ // free peaks and set size to zero
+ av_freep(&p->peaks);
+ p->size = 0;
+ } 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
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