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.
>
>>
>>> + 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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