[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter

Paul B Mahol onemda at gmail.com
Tue May 2 21:47:33 EEST 2017


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.

>
>
>
>> +
>> +    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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