Updated patch attached. PFA On Thu, Nov 3, 2016 at 2:00 AM, Michael Niedermayer <mich...@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 > <mich...@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 <at...@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 >
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]; + } + } + 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
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