On 10/2/19, James Almer <jamr...@gmail.com> wrote:
> On 10/2/2019 12:37 PM, Paul B Mahol wrote:
>> On 10/2/19, James Almer <jamr...@gmail.com> wrote:
>>> On 10/2/2019 12:11 PM, Paul B Mahol wrote:
>>>> Signed-off-by: Paul B Mahol <one...@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.
>>>>
>>>> +@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.
>>>> +
>>>> +@table @option
>>>> +@item t
>>>> +Set comb filtering type.
>>>> +
>>>> +It accepts the following values:
>>>> +@table @option
>>>> +@item f
>>>> +set feedforward type
>>>> +@item b
>>>> +set feedback type
>>>> +@end table
>>>> +
>>>> +@item b0
>>>> +Set direct signal gain. Default is 1. Allowed range is from 0 to 1.
>>>> +
>>>> +@item xM
>>>> +Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1.
>>>> +
>>>> +@item M
>>>> +Set delay in number of samples. Default is 10. Allowed range is from 1
>>>> to
>>>> 327680.
>>>> +@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.
>
> I see, so it's an init function and not something called per frame.
> Disregard what i said, then. I'm not familiar with libavfilter internal
> workings, which is why i assumed it could happen.
It actually happens with astreamselect filter. But that filter is not used much.
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