[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter
Muhammad Faiz
mfcc64 at gmail.com
Fri May 5 16:02:00 EEST 2017
On Wed, May 3, 2017 at 4:12 PM, Muhammad Faiz <mfcc64 at gmail.com> wrote:
> On Wed, May 3, 2017 at 1:47 AM, Paul B Mahol <onemda at gmail.com> wrote:
>> On 5/2/17, Muhammad Faiz <mfcc64 at gmail.com> wrote:
>>> On Mon, May 1, 2017 at 3:30 PM, Paul B Mahol <onemda at gmail.com> wrote:
>>>> Signed-off-by: Paul B Mahol <onemda at 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
>>>>
>>>> + at 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.
Probably, like in asetnsamples filter.
Thank's.
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