[FFmpeg-devel] [OPW] OPW Project Proposal
Pallavi Kumari
pallavikumari.in at gmail.com
Wed Nov 2 01:30:09 EET 2016
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
Regards,
Atana
On Fri, Oct 28, 2016 at 4:38 AM, Michael Niedermayer <michael at niedermayer.cc
> wrote:
> On Thu, Oct 27, 2016 at 11:38:27PM +0530, Pallavi Kumari wrote:
> > Hi Michael,
> >
> > I have attached a patch with the mail.
> >
> > With `avcodec_get_frame_defaults (&frame)` in the function readAudio
> > program was working fine with old version in my system. But this function
> > is not in the latest git repo so I used `av_frame_unref(&frame)` instead
> > and the program segfaults. I am not sure why or how to fix this.
>
> theres "AVFrame frame;"
>
> thats bad, as it depends on sizef(AVFrame) to allocat frame on the
> stack.
> you should always allocate/clone/... one if one is not already
> available, each of these would be setup correctly and not need
> avcodec_get_frame_defaults()
>
>
> >
> > Also, I haven't properly understood the input and output flow of
> > libavfilters (how to write input and output links? and overall flow).
> Could
> > you explain it to me?
>
> see existing filters, af_*.c
> maybe libavfilter/af_astats.c has a similar structure to the filter
> here
>
> theres also
> doc/filter_design.txt
>
>
> >
> >
> >
> > On Thu, Oct 27, 2016 at 3:24 AM, Michael Niedermayer
> <michael at niedermayer.cc
> > > wrote:
> >
> > > On Thu, Oct 27, 2016 at 12:40:53AM +0530, Pallavi Kumari wrote:
> > > > I mean deciding a timeline for the opw project. Which is to be
> mentioned
> > > in
> > > > application
> > >
> > > its your application, you can choose whatever timeline you feel makes
> > > sense.
> > > As far as iam concered whats important is to finish the qualification
> > > task well before we have to choose the applicant(s) we accept this
> > > year.
> > >
> > > The timeline for the main project in dec-mar can be segmented as you
> > > like but leave plenty of time toward the end, problems always occur
> > > and things get delayed.
> > > Also both for qualification and the main project submit code early
> > > if you have questions ask early,
> > > if you dont get satisfactory reply from me on something ask again and
> > > louder/clearer ... i do sometimes also forget to reply until i then
> > > remember again days later ...
> > >
> > > also i think the architecture wasnt discussed much yet
> > > do you have some plans about it ?
> > >
> > > I mean for example there could be a avfilter that passes audio through
> > > and adds fingerint(s) as metadata into thf AVFrame
> > > (as with avpriv_frame_get_metadatap in vf_idet.c) and also print the
> > > fingerprints via av_log()
> > >
> > > a 2nd filter could then look these fingerprints from the metadata
> > > up in some file/database
> > >
> > > a 3rd one could add the fingerprints to the file/database
> > >
> > > or these also could all be in one filter of course
> > >
> > > this can be extended in the future to for example have a filter
> > > delay the audio until lookup succeeds and then once identified
> > > lookup lyrics online and add them or such (this of course would be a
> > > future thing after outreachy/opw)
> > >
> > > but maybe you have a different plan on how the components would
> > > interact ?
> > >
> > > thx
> > >
> > > >
> > > > On Wed, Oct 26, 2016 at 6:01 PM, Michael Niedermayer
> > > <michael at niedermayer.cc
> > > > > wrote:
> > > >
> > > > > Hi
> > > > >
> > > > > On Mon, Oct 17, 2016 at 02:06:26PM +0530, Pallavi Kumari wrote:
> > > > > > Hi Michael,
> > > > > >
> > > > > > I figured out the use of fft. Help me with the time line setting.
> > > Thanks
> > > > >
> > > > > I dont understand the question,
> > > > > if its about AVFILTER_FLAG_SUPPORT_TIMELINE*, please ignore this
> for
> > > > > now, its not needed
> > > > >
> > > > > [...]
> > > > > --
> > > > > Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC7
> > > 87040B0FAB
> > > > >
> > > > > Rewriting code that is poorly written but fully understood is good.
> > > > > Rewriting code that one doesnt understand is a sign that one is
> less
> > > smart
> > > > > then the original author, trying to rewrite it will not make it
> better.
> > > > >
> > > > _______________________________________________
> > > > ffmpeg-devel mailing list
> > > > ffmpeg-devel at ffmpeg.org
> > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> > >
> > > --
> > > Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC7
> 87040B0FAB
> > >
> > > He who knows, does not speak. He who speaks, does not know. -- Lao Tsu
> > >
> > > _______________________________________________
> > > ffmpeg-devel mailing list
> > > ffmpeg-devel at ffmpeg.org
> > > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> > >
> > >
>
> > Makefile | 1
> > af_peakpoints.c | 263 ++++++++++++++++++++++++++++++
> ++++++++++++++++++++++++++
> > allfilters.c | 1
> > version.h | 2
> > 4 files changed, 266 insertions(+), 1 deletion(-)
> > 5cc989fc1ba9466043f833059ee82efcaa3cf898 0001-avfilter-add-peakpoints-
> filter.patch
> > From 60429fb7d83d11d6dd733c3477950a940bd0d84a Mon Sep 17 00:00:00 2001
> > From: Atana <atana at openmailbox.org>
> > Date: Mon, 24 Oct 2016 17:16:09 +0530
> > Subject: [PATCH] avfilter: add 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..84b51da
> > --- /dev/null
> > +++ b/libavfilter/af_peakpoints.c
> > @@ -0,0 +1,263 @@
> > +/*
> > + * Copyright (c) 2016 Pallavi Kumari
> > + *
> > + * 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 "libavutil/opt.h"
> > +
> > +
> > +/* Structure to contain peak points context */
> > +typedef struct {
> > + const AVClass *class;
> > + double *data;
> > + int nsamples;
> > + 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 endIndex, 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) {
>
> % is slow (compared to &)
> also doing a i++ loop instead of i+= chunkSize-1 is slow too
> does lots of unneeed operations, which the compiler might or might
> not optimize out
>
>
> > + max = getMax(res_arr, startIndex, i);
> > + peaks[peakIndex++] = max;
> > + startIndex = startIndex + chunkSize;
> > + }
>
> tabs are also forbidden in ffmpeg git
>
>
> > + i += 1;
> > + }
> > +}
> > +
>
> > +/* read audio stream from input file */
> > +static void readAudio (int *sampleRate, PeakPointsContext *ppc) {
> > + // Initialize FFmpeg
> > + av_register_all ();
> > +
> > + AVFormatContext *formatContext = NULL;
> > + AVCodecContext *codecContext = NULL;
> > + AVCodec *codec = NULL;
> > + AVPacket packet;
> > + AVFrame frame;
> > +
> > + ssize_t nSamples = 0;
> > +
> > + // open the file and autodetect the format
> > + if (avformat_open_input (&formatContext, ppc->inputFile, NULL,
> NULL) < 0)
> > + printf("Can not open input file %s", ppc->inputFile);
> > +
> > + if (avformat_find_stream_info (formatContext, NULL) < 0)
> > + printf("Can not find stream information");
> > +
> > + // find audio stream (between video/audio/subtitles/.. streams)
> > + int audioStreamId = av_find_best_stream (formatContext,
> AVMEDIA_TYPE_AUDIO,
> > + -1, -1, &codec, 0);
> > + if (audioStreamId < 0)
> > + printf("Can not find audio stream in the input file");
> > +
> > + codecContext = formatContext->streams[audioStreamId]->codec;
> > +
> > + // init the audio decoder
> > + if (avcodec_open2 (codecContext, codec, NULL) < 0)
> > + printf("Can not open audio decoder");
> > +
> > + // read all packets
> > + int i = 0;
> > + while (av_read_frame (formatContext, &packet) == 0) {
>
> A libavfilter would generally analyze its audio input stream instead
> of opening a audio file on its own
>
>
>
> > + if (packet.stream_index == audioStreamId) {
> > + //avcodec_get_frame_defaults (&frame);
> > + av_frame_unref (&frame);
> > + int gotFrame = 0;
> > + if (avcodec_decode_audio4 (codecContext,&frame,&gotFrame,&packet)
> < 0){
> > + printf("Error decoding audio");
> > + }
> > + if (gotFrame) {
> > + // audio frame has been decoded
> > + int size = av_samples_get_buffer_size (NULL,
> > +
> codecContext->channels,
> > + frame.nb_samples,
> > +
> codecContext->sample_fmt,
> > + 1);
> > + if (size < 0) {
> > + printf("av_samples_get_buffer_size invalid value");
> > + }
> > +
> > + // printf("%d\n", *frame.data[0]);
> > + ppc->data[i] = (double)*frame.data[0];
> > + i += 1;
> > + nSamples += frame.nb_samples;
> > + }
> > + }
> > + av_free_packet (&packet);
> > + }
> > + ppc->nsamples = i;
> > +
> > + if (nSamples < 1)
> > + printf("Decoded audio data is empty");
> > +
> > + int sampleSize = av_get_bytes_per_sample (codecContext->sample_fmt);
> > + if (sampleSize < 1)
> > + printf("Invalid sample format");
> > +
> > + // optional, return value with sample rate
> > + if (sampleRate) *sampleRate = codecContext->sample_rate;
> > +
> > + if (codecContext) avcodec_close (codecContext);
> > +
> > + avformat_close_input (&formatContext);
> > +}
> > +
> > +/* Get peaks points from windowed frequency domain data*/
> > +static void getPeakPoints(PeakPointsContext *ppc) {
> > + int i, k, size, chunkSize, chunkSampleSize, resSize, sample_rate;
> > + double *fft_res;
> > + int m;
> > +
> > + ppc->data = malloc(sizeof(double)*10000);
> > + readAudio (&sample_rate, ppc);
> > +
> > + size = ppc->nsamples;
> > + m = log2(ppc->windowSize);
> > + chunkSize = ppc->windowSize;
> > + chunkSampleSize = size/chunkSize;
> > + resSize = chunkSize * chunkSampleSize;
> > + fft_res = malloc(sizeof(double) * resSize);
> > +
> > + RDFTContext *rdftC = av_rdft_init(m, DFT_R2C);
> > + FFTSample *data;
>
> > + data = malloc(sizeof(FFTSample)*chunkSize);
> > + // 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];
>
> missing malloc failure check
>
>
> > + }
> > + //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);
> > +
> > + int pSize = resSize/chunkSize;
> > + ppc->size = pSize;
> > + ppc->peaks = malloc(sizeof(double)*pSize);
> > + getPeakPointInChunk(chunkSize, fft_res, resSize, ppc->peaks);
> > +}
> > +
> > +
> > +#define OFFSET(x) offsetof(PeakPointsContext, x)
> > +
> > +static const AVOption peakpoints_options[] = {
> > + { "input", "set input audio file", OFFSET(inputFile),
> AV_OPT_TYPE_STRING, {.str="sample.mp3"}, CHAR_MIN, CHAR_MAX },
> > + { "wsize", "set window size", OFFSET(windowSize),
> AV_OPT_TYPE_INT, {.i64=16}, 0, INT_MAX},
> > + { NULL },
> > +};
> > +
> > +static const char *peakpoints_get_name(void *ctx) {
> > + return "peakpoints";
> > +}
> > +
> > +static const AVClass peakpoints_class = {
> > + .class_name = "PeakPointsContext",
> > + .item_name = peakpoints_get_name,
> > + .option = peakpoints_options,
> > +};
> > +
> > +
> > +AVFILTER_DEFINE_CLASS(peakpoints);
> > +
> > +static av_cold int init(AVFilterContext *ctx)
> > +{
> > + PeakPointsContext *p = ctx->priv;
> > +
> > + if (!(p->inputFile)) {
> > + av_log(ctx, AV_LOG_ERROR, "Input audio file must be passed\n");
> > + return AVERROR(EINVAL);
> > + }
> > +
> > + if (p->windowSize < 16) {
> > + av_log(ctx, AV_LOG_ERROR, "window size must be greater than
> or equal to 16");
> > + return AVERROR(EINVAL);
> > + }
> > + //printf("Getting peak points..");
> > + // get peaks
> > + getPeakPoints(p);
> > + //printf("Peak points collected.");
> > + // print peaks
> > + int i;
> > + for (i = 0; i < p->size; i++) {
> > + av_log(ctx, AV_LOG_DEBUG, "%f", p->peaks[i]);
> > + }
> > + return 0;
> > +}
> > +
>
> > +static av_cold void uninit(AVFilterContext *ctx)
> > +{
> > + PeakPointsContext *p = ctx->priv;
> > + int i;
> > +
>
> > + // free memory allocated for audio data
> > + for (i = 0; p->nsamples; i++) {
> > + free(&p->data[i]);
> > + }
> > +
> > + // free memory allocated for peaks
> > + for (i = 0; p->size; i++) {
> > + free(&p->data[i]);
> > + }
>
> looks like a double free
>
> also please use av_freep() and av_malloc*()
>
> [...]
>
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> No human being will ever know the Truth, for even if they happen to say it
> by chance, they would not even known they had done so. -- Xenophanes
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
>
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