On 11/18/17, Rostislav Pehlivanov <atomnu...@gmail.com> wrote:
> 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()

ok

>
>
>
>> +        }
>> +
>> +        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()
>

ok

>
>
>> +        }
>> +    }
>> +}
>> +
>> +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?
>

Both are supported.

>
>
>> +
>> +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?

Doesn't cost much, and also it is less lines this way.
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