[FFmpeg-devel] [OPW] OPW Project Proposal
Pallavi Kumari
pallavikumari.in at gmail.com
Fri Nov 4 13:57:34 EET 2016
patch attached.
On Fri, Nov 4, 2016 at 3:29 PM, Michael Niedermayer <michael at 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 <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 at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
>
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