On Wed, May 3, 2017 at 1:47 AM, Paul B Mahol <one...@gmail.com> wrote:
> 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.
OK.
>
>>
>>
>>
>>> +
>>> + 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.
Probably, at least, first pts should be derived from input pts.
Also, is time_base always 1/sample_rate?
Thank's.
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