| Message ID | CAC7Kmr0ppU0zWMQaoe8yZPNuShJp5J_k7i=b-wy4tRxpsk_2YA@mail.gmail.com |
|---|---|
| State | Superseded |
| Headers | show |
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 > > > Regards, > Atana > > On Fri, Oct 28, 2016 at 4:38 AM, Michael Niedermayer <michael@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@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@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@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@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@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@ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > > > > af_peakpoints.c | 300 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > 1 file changed, 300 insertions(+) > 993ac68b0e42e9c176a8452eff9126ddcd223c4c 0001-avfilter-added-peakpoints-filter.patch > From 5ef0d52dbf2fa16ce3630f320c06eb72d83b6c2c Mon Sep 17 00:00:00 2001 > From: Atana <atana@openmailbox.org> > Date: Wed, 2 Nov 2016 04:49:15 +0530 > Subject: [PATCH] avfilter: added peakpoints filter > > --- > libavfilter/af_peakpoints.c | 300 ++++++++++++++++++++++++++++++++++++++++++++ > 1 file changed, 300 insertions(+) > create mode 100644 libavfilter/af_peakpoints.c > > diff --git a/libavfilter/af_peakpoints.c b/libavfilter/af_peakpoints.c > new file mode 100644 > index 0000000..e5c4494 > --- /dev/null > +++ b/libavfilter/af_peakpoints.c > @@ -0,0 +1,300 @@ > +/* > + * 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 "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) { > + max = getMax(res_arr, startIndex, i); > + peaks[peakIndex++] = max; > + startIndex = startIndex + chunkSize; > + } > + 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 = av_frame_alloc(); > + void *avc; > + ssize_t nSamples = 0; > + > + // open the file and autodetect the format > + if (avformat_open_input (&formatContext, ppc->inputFile, NULL, NULL) < 0) > + av_log(avc, AV_LOG_ERROR, "Can not open input file %s", ppc->inputFile); This still opens the file instead of using the filter input The way filters work is that they receive their data through the .filter_frame() callback not through calling some open_file() see for example a simple filter like libavfilter/af_volumedetect.c you need no AVPacket, no av_read_frame(), no avcodec_decode_audio4() no avformat_open_input() you get a already decoded AVFrame in your filter_frame() callback [...] > +static av_cold void uninit(AVFilterContext *ctx) > +{ > + PeakPointsContext *p = ctx->priv; > + int i; > + > + if (p->data) { > + av_free(p->data); > + p->data = NULL; > + } this should use av_freep() which does the "= NULL" automatically see existing uses of av_freep() on how its used [...]
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. [...]
From 5ef0d52dbf2fa16ce3630f320c06eb72d83b6c2c Mon Sep 17 00:00:00 2001 From: Atana <atana@openmailbox.org> Date: Wed, 2 Nov 2016 04:49:15 +0530 Subject: [PATCH] avfilter: added peakpoints filter --- libavfilter/af_peakpoints.c | 300 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 300 insertions(+) create mode 100644 libavfilter/af_peakpoints.c diff --git a/libavfilter/af_peakpoints.c b/libavfilter/af_peakpoints.c new file mode 100644 index 0000000..e5c4494 --- /dev/null +++ b/libavfilter/af_peakpoints.c @@ -0,0 +1,300 @@ +/* + * 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 "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) { + max = getMax(res_arr, startIndex, i); + peaks[peakIndex++] = max; + startIndex = startIndex + chunkSize; + } + 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 = av_frame_alloc(); + void *avc; + ssize_t nSamples = 0; + + // open the file and autodetect the format + if (avformat_open_input (&formatContext, ppc->inputFile, NULL, NULL) < 0) + av_log(avc, AV_LOG_ERROR, "Can not open input file %s", ppc->inputFile); + + if (avformat_find_stream_info (formatContext, NULL) < 0) + av_log(avc, AV_LOG_ERROR, "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) + av_log(avc, AV_LOG_ERROR, "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) + av_log(avc, AV_LOG_ERROR, "Can not open audio decoder"); + + // read all packets + int i = 0; + av_init_packet(&packet); + packet.data = NULL; + packet.size = 0; + while (av_read_frame (formatContext, &packet) == 0) { + 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){ + av_log(avc, AV_LOG_ERROR, "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) { + av_log(avc, AV_LOG_ERROR, "av_samples_get_buffer_size invalid value"); + } + + ppc->data[i] = (double)*frame->data[0]; + i += 1; + nSamples += frame->nb_samples; + } + } + av_free_packet (&packet); + } + ppc->nsamples = i; + + if (nSamples < 1) + av_log(avc, AV_LOG_WARNING, "Decoded audio data is empty"); + + int sampleSize = av_get_bytes_per_sample (codecContext->sample_fmt); + if (sampleSize < 1) + av_log(avc, AV_LOG_WARNING, "Invalid sample format"); + + // optional, return value with sample rate + if (sampleRate) *sampleRate = codecContext->sample_rate; + + if (codecContext) avcodec_close (codecContext); + + if (frame) { + av_free(frame); + frame = NULL; + } + avformat_close_input (&formatContext); +} + +/* Get peaks points from windowed frequency domain data*/ +static int getPeakPoints(PeakPointsContext *ppc) { + int i, k, size, chunkSize, chunkSampleSize, resSize, sample_rate; + double *fft_res; + int m; + void *avc; + + ppc->data = av_malloc(sizeof(double)*10000); + + if (!ppc->data) { + av_log(avc, AV_LOG_ERROR, "Cann't allocate memmory for audio data\n"); + return 0; + } + + readAudio (&sample_rate, ppc); + + size = ppc->nsamples; + 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; + } + + RDFTContext *rdftC = av_rdft_init(m, DFT_R2C); + FFTSample *data; + data = av_malloc(sizeof(FFTSample)*chunkSize); + + 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); + int 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[] = { + { "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\n"); + return AVERROR(EINVAL); + } + + // get peaks + int i; + if (getPeakPoints(p)) { + // 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 { + av_log(ctx, AV_LOG_INFO, "Peak points not retrieved\n"); + return AVERROR(EINVAL); + } + + return 0; +} + + +static av_cold void uninit(AVFilterContext *ctx) +{ + PeakPointsContext *p = ctx->priv; + int i; + + if (p->data) { + av_free(p->data); + p->data = NULL; + } + + // free memory allocated for peaks + if (p->peaks) { + av_free(p->peaks); + p->peaks = 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 = inputs, + //.outputs = outputs, + .priv_class = &peakpoints_class, +}; -- 1.9.1