[FFmpeg-devel] [PATCH] avfilter: add acomb filter
James Almer
jamrial at gmail.com
Wed Oct 2 18:29:43 EEST 2019
On 10/2/2019 12:11 PM, Paul B Mahol wrote:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> doc/filters.texi | 28 ++++++
> libavfilter/Makefile | 1 +
> libavfilter/af_acomb.c | 188 +++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 218 insertions(+)
> create mode 100644 libavfilter/af_acomb.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index e46839bfec..9c50b2e4b2 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -355,6 +355,34 @@ build.
>
> Below is a description of the currently available audio filters.
>
> + at section acomb
> +Apply comb audio filtering.
> +
> +Amplifies or attenuates certain frequencies by the superposition of a
> +delayed version of the original audio signal onto itself.
> +
> + at table @option
> + at item t
> +Set comb filtering type.
> +
> +It accepts the following values:
> + at table @option
> + at item f
> +set feedforward type
> + at item b
> +set feedback type
> + at end table
> +
> + at item b0
> +Set direct signal gain. Default is 1. Allowed range is from 0 to 1.
> +
> + at item xM
> +Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1.
> +
> + at item M
> +Set delay in number of samples. Default is 10. Allowed range is from 1 to 327680.
> + at end table
> +
> @section acompressor
>
> A compressor is mainly used to reduce the dynamic range of a signal.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 182fe9df4b..d8a16d6e15 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -31,6 +31,7 @@ include $(SRC_PATH)/libavfilter/dnn/Makefile
>
> # audio filters
> OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o
> +OBJS-$(CONFIG_ACOMB_FILTER) += af_acomb.o
> OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o
> OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o
> OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o
> diff --git a/libavfilter/af_acomb.c b/libavfilter/af_acomb.c
> new file mode 100644
> index 0000000000..3b0730c363
> --- /dev/null
> +++ b/libavfilter/af_acomb.c
> @@ -0,0 +1,188 @@
> +/*
> + * 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/opt.h"
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "internal.h"
> +
> +typedef struct AudioCombContext {
> + const AVClass *class;
> +
> + double b0, xM;
> + int t, M;
> +
> + int head;
> + int tail;
> +
> + AVFrame *delayframe;
> +
> + void (*filter)(struct AudioCombContext *s, AVFrame *in, AVFrame *out);
> +} AudioCombContext;
> +
> +#define OFFSET(x) offsetof(AudioCombContext, x)
> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption acomb_options[] = {
> + { "t", "set comb filter type", OFFSET(t), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "t" },
> + { "f", "feedforward", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "t" },
> + { "b", "feedback", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "t" },
> + { "b0", "set direct signal gain", OFFSET(b0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A },
> + { "xM", "set delayed line gain", OFFSET(xM), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A },
> + { "M", "set delay in number of samples", OFFSET(M), AV_OPT_TYPE_INT, {.i64=10}, 1, 327680, A },
> + { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(acomb);
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterFormats *formats = NULL;
> + AVFilterChannelLayouts *layouts = NULL;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_FLTP,
> + AV_SAMPLE_FMT_DBLP,
> + AV_SAMPLE_FMT_NONE
> + };
> + int ret;
> +
> + formats = ff_make_format_list(sample_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ret = ff_set_common_formats(ctx, formats);
> + if (ret < 0)
> + return ret;
> +
> + layouts = ff_all_channel_counts();
> + if (!layouts)
> + return AVERROR(ENOMEM);
> +
> + ret = ff_set_common_channel_layouts(ctx, layouts);
> + if (ret < 0)
> + return ret;
> +
> + formats = ff_all_samplerates();
> + return ff_set_common_samplerates(ctx, formats);
> +}
> +
> +#define COMB(name, type, dir, t) \
> +static void acomb_## name ## _ ##dir(AudioCombContext *s, \
> + AVFrame *in, AVFrame *out) \
> +{ \
> + const type b0 = s->b0; \
> + const type xM = s->xM; \
> + const int M = s->M; \
> + int head; \
> + \
> + for (int c = 0; c < in->channels; c++) { \
> + const type *src = (const type *)in->extended_data[c]; \
> + type *delay = (type *)s->delayframe->extended_data[c]; \
> + type *dst = (type *)out->extended_data[c]; \
> + \
> + head = s->head; \
> + for (int n = 0; n < in->nb_samples; n++) { \
> + dst[n] = b0 * src[n] + t * xM * delay[head]; \
> + if (t == 1) \
> + delay[head] = src[n]; \
> + else \
> + delay[head] = dst[n]; \
> + head++; \
> + if (head >= M) \
> + head = 0; \
> + } \
> + } \
> + \
> + s->head = head; \
> +}
> +
> +COMB(fltp, float, f, 1)
> +COMB(dblp, double, f, 1)
> +COMB(fltp, float, b, -1)
> +COMB(dblp, double, b, -1)
> +
> +static int config_input(AVFilterLink *inlink)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + AudioCombContext *s = ctx->priv;
> +
> + s->delayframe = ff_get_audio_buffer(inlink, s->M);
You're leaking s->delayframe every time config_input() is called after
the first time.
> + if (!s->delayframe)
> + return AVERROR(ENOMEM);
> +
> + switch (inlink->format) {
> + case AV_SAMPLE_FMT_FLTP: s->filter = s->t ? acomb_fltp_b : acomb_fltp_f; break;
> + case AV_SAMPLE_FMT_DBLP: s->filter = s->t ? acomb_dblp_b : acomb_dblp_f; break;
> + }
> +
> + return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + AudioCombContext *s = ctx->priv;
> + AVFilterLink *outlink = ctx->outputs[0];
> + AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
> +
> + if (!out) {
> + av_frame_free(&in);
> + return AVERROR(ENOMEM);
> + }
> + av_frame_copy_props(out, in);
> +
> + s->filter(s, in, out);
> +
> + av_frame_free(&in);
> + return ff_filter_frame(outlink, out);
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> + AudioCombContext *s = ctx->priv;
> +
> + av_frame_free(&s->delayframe);
> +}
> +
> +static const AVFilterPad acomb_inputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = filter_frame,
> + .config_props = config_input,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad acomb_outputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + },
> + { NULL }
> +};
> +
> +AVFilter ff_af_acomb = {
> + .name = "acomb",
> + .description = NULL_IF_CONFIG_SMALL("Apply comb audio filter."),
> + .query_formats = query_formats,
> + .priv_size = sizeof(AudioCombContext),
> + .priv_class = &acomb_class,
> + .uninit = uninit,
> + .inputs = acomb_inputs,
> + .outputs = acomb_outputs,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 1a26129069..7417f9656d 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -24,6 +24,7 @@
> #include "config.h"
>
> extern AVFilter ff_af_abench;
> +extern AVFilter ff_af_acomb;
> extern AVFilter ff_af_acompressor;
> extern AVFilter ff_af_acontrast;
> extern AVFilter ff_af_acopy;
>
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