On 5/2/17, Muhammad Faiz <mfc...@gmail.com> wrote:
> On Mon, May 1, 2017 at 3:30 PM, Paul B Mahol <one...@gmail.com> wrote:
>> Signed-off-by: Paul B Mahol <one...@gmail.com>
>> ---
>> configure | 2 +
>> doc/filters.texi | 10 ++
>> libavfilter/Makefile | 1 +
>> libavfilter/af_afirfilter.c | 409
>> ++++++++++++++++++++++++++++++++++++++++++++
>> libavfilter/allfilters.c | 1 +
>> 5 files changed, 423 insertions(+)
>> create mode 100644 libavfilter/af_afirfilter.c
>>
>> diff --git a/configure b/configure
>> index b3cb5b0..7fc7af4 100755
>> --- a/configure
>> +++ b/configure
>> @@ -3078,6 +3078,8 @@ unix_protocol_select="network"
>> # filters
>> afftfilt_filter_deps="avcodec"
>> afftfilt_filter_select="fft"
>> +afirfilter_filter_deps="avcodec"
>> +afirfilter_filter_select="fft"
>> amovie_filter_deps="avcodec avformat"
>> aresample_filter_deps="swresample"
>> ass_filter_deps="libass"
>> diff --git a/doc/filters.texi b/doc/filters.texi
>> index 119e747..ea343d1 100644
>> --- a/doc/filters.texi
>> +++ b/doc/filters.texi
>> @@ -878,6 +878,16 @@ afftfilt="1-clip((b/nb)*b,0,1)"
>> @end example
>> @end itemize
>>
>> +@section afirfilter
>> +
>> +Apply an Arbitary Frequency Impulse Response filter.
>> +
>> +This filter uses second stream as FIR coefficients.
>> +If second stream holds single channel, it will be used
>> +for all input channels in first stream, otherwise
>> +number of channels in second stream must be same as
>> +number of channels in first stream.
>> +
>> @anchor{aformat}
>> @section aformat
>>
>> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
>> index 66c36e4..1a0f24b 100644
>> --- a/libavfilter/Makefile
>> +++ b/libavfilter/Makefile
>> @@ -38,6 +38,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER) +=
>> af_aemphasis.o
>> OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
>> OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
>> OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o
>> window_func.o
>> +OBJS-$(CONFIG_AFIRFILTER_FILTER) += af_afirfilter.o
>> OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
>> OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
>> OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
>> diff --git a/libavfilter/af_afirfilter.c b/libavfilter/af_afirfilter.c
>> new file mode 100644
>> index 0000000..ef2488a
>> --- /dev/null
>> +++ b/libavfilter/af_afirfilter.c
>> @@ -0,0 +1,409 @@
>> +/*
>> + * Copyright (c) 2017 Paul B Mahol
>> + *
>> + * 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
>> + */
>> +
>> +/**
>> + * @file
>> + * An arbitrary audio FIR filter
>> + */
>> +
>> +#include "libavutil/audio_fifo.h"
>> +#include "libavutil/avassert.h"
>> +#include "libavutil/channel_layout.h"
>> +#include "libavutil/common.h"
>> +#include "libavutil/opt.h"
>> +#include "libavcodec/avfft.h"
>> +
>> +#include "audio.h"
>> +#include "avfilter.h"
>> +#include "formats.h"
>> +#include "internal.h"
>> +
>> +typedef struct FIRContext {
>> + const AVClass *class;
>> +
>> + int n;
>> + int eof_coeffs;
>> + int have_coeffs;
>> + int nb_taps;
>> + int fft_length;
>> + int nb_channels;
>> + int one2many;
>> +
>> + FFTContext *fft, *ifft;
>> + FFTComplex **fft_data;
>> + FFTComplex **fft_coef;
>
> Probably you may use rdft for performance reason.
I will concentrate on correctness of output first.
>
>
>
>> +
>> + AVAudioFifo *fifo[2];
>> + AVFrame *in[2];
>> + AVFrame *buffer;
>> + int64_t pts;
>> + int hop_size;
>> + int start, end;
>> +} FIRContext;
>> +
>> +static int fir_filter(FIRContext *s, AVFilterLink *outlink)
>> +{
>> + AVFilterContext *ctx = outlink->src;
>> + int start = s->start, end = s->end;
>> + int ret = 0, n, ch, j, k;
>> + int nb_samples;
>> + AVFrame *out;
>> +
>> + nb_samples = FFMIN(s->fft_length, av_audio_fifo_size(s->fifo[0]));
>> +
>> + s->in[0] = ff_get_audio_buffer(ctx->inputs[0], nb_samples);
>> + if (!s->in[0])
>> + return AVERROR(ENOMEM);
>> +
>> + av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data,
>> nb_samples);
>> +
>> + for (ch = 0; ch < outlink->channels; ch++) {
>> + const float *src = (float *)s->in[0]->extended_data[ch];
>> + float *buf = (float *)s->buffer->extended_data[ch];
>> + FFTComplex *fft_data = s->fft_data[ch];
>> + FFTComplex *fft_coef = s->fft_coef[ch];
>> +
>> + memset(fft_data, 0, sizeof(*fft_data) * s->fft_length);
>> + for (n = 0; n < nb_samples; n++) {
>> + fft_data[n].re = src[n];
>> + fft_data[n].im = 0;
>> + }
>> +
>> + av_fft_permute(s->fft, fft_data);
>> + av_fft_calc(s->fft, fft_data);
>> +
>> + fft_data[0].re *= fft_coef[0].re;
>> + fft_data[0].im *= fft_coef[0].im;
>> + for (n = 1; n < s->fft_length; n++) {
>> + const float re = fft_data[n].re;
>> + const float im = fft_data[n].im;
>> +
>> + fft_data[n].re = re * fft_coef[n].re - im * fft_coef[n].im;
>> + fft_data[n].im = re * fft_coef[n].im + im * fft_coef[n].re;
>> + }
>> +
>> + av_fft_permute(s->ifft, fft_data);
>> + av_fft_calc(s->ifft, fft_data);
>> +
>> + start = s->start;
>> + end = s->end;
>> + k = end;
>> +
>> + for (n = 0, j = start; j < k && n < s->fft_length; n++, j++) {
>> + buf[j] = fft_data[n].re;
>> + }
>> +
>> + for (; n < s->fft_length; n++, j++) {
>> + buf[j] = fft_data[n].re;
>> + }
>> +
>> + start += s->hop_size;
>> + end = j;
>> + }
>> +
>> + s->start = start;
>> + s->end = end;
>> +
>> + if (start >= nb_samples) {
>> + float *dst, *buf;
>> +
>> + start -= nb_samples;
>> + end -= nb_samples;
>> +
>> + s->start = start;
>> + s->end = end;
>> +
>> + out = ff_get_audio_buffer(outlink, nb_samples);
>> + if (!out)
>> + return AVERROR(ENOMEM);
>> +
>> + out->pts = s->pts;
>> + s->pts += nb_samples;
>
> Is pts handled correctly here? Seem it is not derived from input pts.
>
It can not be derived in any other way.
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