[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter
James Almer
jamrial at gmail.com
Tue May 9 21:21:18 EEST 2017
On 5/8/2017 7:00 PM, Paul B Mahol wrote:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> configure | 2 +
> doc/filters.texi | 23 ++
> libavfilter/Makefile | 1 +
> libavfilter/af_afir.c | 535 +++++++++++++++++++++++++++++++++++++++++
> libavfilter/af_afir.h | 82 +++++++
> libavfilter/allfilters.c | 1 +
> libavfilter/x86/Makefile | 2 +
> libavfilter/x86/af_afir.asm | 53 ++++
> libavfilter/x86/af_afir_init.c | 35 +++
> 9 files changed, 734 insertions(+)
> create mode 100644 libavfilter/af_afir.c
> create mode 100644 libavfilter/af_afir.h
> create mode 100644 libavfilter/x86/af_afir.asm
> create mode 100644 libavfilter/x86/af_afir_init.c
>
> diff --git a/configure b/configure
> index 2e1786a..a46c375 100755
> --- a/configure
> +++ b/configure
> @@ -3081,6 +3081,8 @@ unix_protocol_select="network"
> # filters
> afftfilt_filter_deps="avcodec"
> afftfilt_filter_select="fft"
> +afir_filter_deps="avcodec"
> +afir_filter_select="fft"
You also need to add avcodec to avfilter_deps in the corresponding
section (Near the end of configure) when this filter is enabled.
> amovie_filter_deps="avcodec avformat"
> aresample_filter_deps="swresample"
> ass_filter_deps="libass"
> diff --git a/doc/filters.texi b/doc/filters.texi
> index f431274..0efce9a 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -878,6 +878,29 @@ afftfilt="1-clip((b/nb)*b,0,1)"
> @end example
> @end itemize
>
> + at section afir
> +
> +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.
> +
> +It accepts the following parameters:
> +
> + at table @option
> + at item dry
> +Set dry gain. This sets input gain.
> +
> + at item wet
> +Set wet gain. This sets final output gain.
> +
> + at item length
> +Set Impulse Response filter length. Default is 1, which means whole IR is processed.
> + at end table
> +
> @anchor{aformat}
> @section aformat
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 0f99086..de5f992 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -37,6 +37,7 @@ 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-$(CONFIG_AFIR_FILTER) += af_afir.o
> OBJS-$(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_afir.c b/libavfilter/af_afir.c
> new file mode 100644
> index 0000000..eb59d53
> --- /dev/null
> +++ b/libavfilter/af_afir.c
> @@ -0,0 +1,535 @@
> +/*
> + * 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/common.h"
> +#include "libavutil/float_dsp.h"
> +#include "libavutil/opt.h"
> +#include "libavcodec/avfft.h"
> +
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +#include "internal.h"
> +#include "af_afir.h"
> +
> +static void fcmul_add_c(float *sum, const float *t, const float *c, int len)
ptrdiff_t for len.
> +{
> + int n;
> +
> + for (n = 0; n < len; n++) {
> + const float cre = c[2 * n ];
> + const float cim = c[2 * n + 1];
> + const float tre = t[2 * n ];
> + const float tim = t[2 * n + 1];
> +
> + sum[2 * n ] += tre * cre - tim * cim;
> + sum[2 * n + 1] += tre * cim + tim * cre;
> + }
> +
> + sum[2 * n] += t[2 * n] * c[2 * n];
> +}
[...]
> +static int convert_coeffs(AVFilterContext *ctx)
> +{
> + AudioFIRContext *s = ctx->priv;
> + int i, ch, n, N;
> + float power = 0;
> +
> + s->nb_taps = av_audio_fifo_size(s->fifo[1]);
> + if (s->nb_taps <= 0)
> + return AVERROR(EINVAL);
> +
> + for (n = 4; (1 << n) < s->nb_taps; n++);
> + N = FFMIN(n, 16);
> + s->ir_length = 1 << n;
> + s->fft_length = (1 << (N + 1)) + 1;
> + s->part_size = 1 << (N - 1);
> + s->block_size = FFALIGN(s->fft_length, 16);
> + s->coeff_size = FFALIGN(s->part_size + 1, 16);
> + s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
> + s->nb_coeffs = s->ir_length + s->nb_partitions;
> +
> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
> + s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
> + if (!s->sum[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
> + s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
Might be a good idea to add some padding since this is used in a SIMD
function. Same with block below.
[...]
> diff --git a/libavfilter/x86/af_afir.asm b/libavfilter/x86/af_afir.asm
> new file mode 100644
> index 0000000..b425055
> --- /dev/null
> +++ b/libavfilter/x86/af_afir.asm
> @@ -0,0 +1,53 @@
> +;*****************************************************************************
> +;* x86-optimized functions for afir filter
> +;* 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
> +;******************************************************************************
> +
> +%include "libavutil/x86/x86util.asm"
> +
> +SECTION_RODATA 32
Unused
> +
> +SECTION .text
> +
> +;------------------------------------------------------------------------------
> +; void ff_fcmul_add(float *sum, const float *t, const float *c, int len)
> +;------------------------------------------------------------------------------
> +
> +INIT_XMM sse3
You can add an YMM avx version as well.
> +cglobal fcmul_add, 4,4,3, sum, t, c, len
> + shl lend, 3
> + add lend, mmsize
> + add tq, lenq
> + add cq, lenq
> + add sumq, lenq
> + neg lenq
> +ALIGN 16
> +.loop:
> + movsldup m0, [tq + lenq]
> + movaps m1, [cq + lenq]
> + mulps m0, m1
> + shufps m1, m1, 0xb1
For AVX you'll have to change this into shufps m1, m1, m1, 0xb1, or yasm
will fail to assemble it. Probably a bug in yasm or x86inc.
> + movshdup m2, [tq + lenq]
> + mulps m2, m1
> + addsubps m0, m2;
Stray ;
> + addps m0, [sumq + lenq]
> + movaps [sumq + lenq], m0
> + add lenq, mmsize
> + jl .loop
> + REP_RET
Just use RET.
> diff --git a/libavfilter/x86/af_afir_init.c b/libavfilter/x86/af_afir_init.c
> new file mode 100644
> index 0000000..1cd5290
> --- /dev/null
> +++ b/libavfilter/x86/af_afir_init.c
> @@ -0,0 +1,35 @@
> +/*
> + * 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
> + */
> +
> +#include "config.h"
> +#include "libavutil/attributes.h"
> +#include "libavutil/cpu.h"
> +#include "libavutil/x86/cpu.h"
> +#include "libavfilter/af_afir.h"
> +
> +void ff_fcmul_add_sse3(float *sum, const float *t, const float *c,
> + int len);
> +
> +av_cold void ff_afir_init_x86(AudioFIRContext *s)
> +{
> + int cpu_flags = av_get_cpu_flags();
> +
> + if (EXTERNAL_SSE3(cpu_flags)) {
> + s->fcmul_add = ff_fcmul_add_sse3;
> + }
> +}
>
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