[FFmpeg-devel] [PATCH] avfilter: add acomb filter

Paul B Mahol onemda at gmail.com
Wed Oct 2 18:57:59 EEST 2019


On 10/2/19, Paul B Mahol <onemda at gmail.com> wrote:
> 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.

Fixed locally, but note that bunch of other filters may need to be changed too.

>
>>
>>> +    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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