[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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