[FFmpeg-devel] [PATCH] avfilter: add acomb filter
Paul B Mahol
onemda at gmail.com
Wed Oct 2 18:37:09 EEST 2019
On 10/2/19, James Almer <jamrial at gmail.com> wrote:
> 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.
Sorry, but since when its ok to call config_input() multiple times?
It was never ok, only filter is allowed to call it by itself.
>
>> + 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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