On 18 November 2017 at 10:44, Paul B Mahol <one...@gmail.com> wrote:
> Signed-off-by: Paul B Mahol <one...@gmail.com>
> ---
> doc/filters.texi | 10 +++
> libavfilter/Makefile | 1 +
> libavfilter/af_acontrast.c | 219 ++++++++++++++++++++++++++++++
> +++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 231 insertions(+)
> create mode 100644 libavfilter/af_acontrast.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 5d99437871..e35952510b 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -429,6 +429,16 @@ How much to use compressed signal in output. Default
> is 1.
> Range is between 0 and 1.
> @end table
>
> +@section acontrast
> +Simple audio dynamic range commpression/expansion filter.
> +
> +The filter accepts the following options:
> +
> +@table @option
> +@item c
> +Set contrast. Default is 33. Allowed range is between 0 and 100.
> +@end table
> +
> @section acopy
>
> Copy the input audio source unchanged to the output. This is mainly
> useful for
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 9acae3ff5b..71c6333a52 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -31,6 +31,7 @@ OBJS-$(CONFIG_QSVVPP) += qsvvpp.o
> # audio filters
> OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o
> OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o
> +OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o
> OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o
> OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
> OBJS-$(CONFIG_ACRUSHER_FILTER) += af_acrusher.o
> diff --git a/libavfilter/af_acontrast.c b/libavfilter/af_acontrast.c
> new file mode 100644
> index 0000000000..38de08ffe5
> --- /dev/null
> +++ b/libavfilter/af_acontrast.c
> @@ -0,0 +1,219 @@
> +/*
> + * Copyright (c) 2008 Rob Sykes
> + * Copyright (c) 2017 Paul B Mahol
> + *
> + * 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/channel_layout.h"
> +#include "libavutil/opt.h"
> +#include "avfilter.h"
> +#include "audio.h"
> +#include "formats.h"
> +
> +typedef struct AudioContrastContext {
> + const AVClass *class;
> + float contrast;
> + void (*filter)(void **dst, const void **src,
> + int nb_samples, int channels, float contrast);
> +} AudioContrastContext;
> +
> +#define OFFSET(x) offsetof(AudioContrastContext, x)
> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption acontrast_options[] = {
> + { "c", "set contrast", OFFSET(contrast), AV_OPT_TYPE_FLOAT,
> {.dbl=33}, 0, 100, A },
>
"contrast" instead of "c"? Not sure if single letter options are a good
idea.
> + { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(acontrast);
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterFormats *formats = NULL;
> + AVFilterChannelLayouts *layouts = NULL;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
> + AV_SAMPLE_FMT_DBL, 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);
> +}
> +
> +static void filter_flt(void **d, const void **s,
> + int nb_samples, int channels,
> + float contrast)
> +{
> + const float *src = s[0];
> + float *dst = d[0];
> + int n, c;
> +
> + for (n = 0; n < nb_samples; n++) {
> + for (c = 0; c < channels; c++) {
> + double d = src[c] * M_PI_2;
> +
> + dst[c] = sin(d + contrast * sin(d * 4));
>
sinf() instead of sin()
> + }
> +
> + dst += c;
> + src += c;
> + }
> +}
> +
> +static void filter_dbl(void **d, const void **s,
> + int nb_samples, int channels,
> + float contrast)
> +{
> + const double *src = s[0];
> + double *dst = d[0];
> + int n, c;
> +
> + for (n = 0; n < nb_samples; n++) {
> + for (c = 0; c < channels; c++) {
> + double d = src[c] * M_PI_2;
> +
> + dst[c] = sin(d + contrast * sin(d * 4));
> + }
> +
> + dst += c;
> + src += c;
> + }
> +}
> +
> +static void filter_fltp(void **d, const void **s,
> + int nb_samples, int channels,
> + float contrast)
> +{
> + int n, c;
> +
> + for (c = 0; c < channels; c++) {
> + const float *src = s[c];
> + float *dst = d[c];
> +
> + for (n = 0; n < nb_samples; n++) {
> + double d = src[n] * M_PI_2;
> +
> + dst[n] = sin(d + contrast * sin(d * 4));
>
sinf() instead of sin()
> + }
> + }
> +}
> +
> +static void filter_dblp(void **d, const void **s,
> + int nb_samples, int channels,
> + float contrast)
> +{
> + int n, c;
> +
> + for (c = 0; c < channels; c++) {
> + const double *src = s[c];
> + double *dst = d[c];
> +
> + for (n = 0; n < nb_samples; n++) {
> + double d = src[n] * M_PI_2;
> +
> + dst[n] = sin(d + contrast * sin(d * 4));
> + }
> + }
> +}
>
Could you do the filtering in-place? Via av_frame_make_writeable?
> +
> +static int config_input(AVFilterLink *inlink)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + AudioContrastContext *s = ctx->priv;
> +
> + switch (inlink->format) {
> + case AV_SAMPLE_FMT_FLT: s->filter = filter_flt; break;
> + case AV_SAMPLE_FMT_DBL: s->filter = filter_dbl; break;
> + case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break;
> + case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break;
> + }
> +
> + return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + AVFilterLink *outlink = ctx->outputs[0];
> + AudioContrastContext *s = ctx->priv;
> + AVFrame *out;
> +
> + if (av_frame_is_writable(in)) {
> + out = in;
> + } else {
> + out = ff_get_audio_buffer(inlink, in->nb_samples);
> + if (!out) {
> + av_frame_free(&in);
> + return AVERROR(ENOMEM);
> + }
> + av_frame_copy_props(out, in);
> + }
> +
> + s->filter((void **)out->extended_data, (const void
> **)in->extended_data,
> + in->nb_samples, in->channels, s->contrast / 750);
>
Divide s->contrast by 750 during init?
> +
> + if (out != in)
> + av_frame_free(&in);
> +
> + return ff_filter_frame(outlink, out);
> +}
> +
> +static const AVFilterPad inputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = filter_frame,
> + .config_props = config_input,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad outputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + },
> + { NULL }
> +};
> +
> +AVFilter ff_af_acontrast = {
> + .name = "acontrast",
> + .description = NULL_IF_CONFIG_SMALL("Simple audio dynamic range
> compression/expansion filter."),
> + .query_formats = query_formats,
> + .priv_size = sizeof(AudioContrastContext),
> + .priv_class = &acontrast_class,
> + .inputs = inputs,
> + .outputs = outputs,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index a838309569..6d92b3ab5a 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -42,6 +42,7 @@ static void register_all(void)
> {
> REGISTER_FILTER(ABENCH, abench, af);
> REGISTER_FILTER(ACOMPRESSOR, acompressor, af);
> + REGISTER_FILTER(ACONTRAST, acontrast, af);
> REGISTER_FILTER(ACOPY, acopy, af);
> REGISTER_FILTER(ACROSSFADE, acrossfade, af);
> REGISTER_FILTER(ACRUSHER, acrusher, af);
> --
> 2.11.0
>
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> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
Apart from that lgtm
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