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

[Muhammad Faiz]

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.