Dana 20. 9. 2015. 06:28 osoba "Kyle Swanson" <k...@ylo.ph> napisala je:
>
> Signed-off-by: Kyle Swanson <k...@ylo.ph>
> ---
> doc/filters.texi | 19 ++++++
> libavfilter/Makefile | 1 +
> libavfilter/af_tremolo.c | 173
+++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> libavfilter/version.h | 2 +-
> 5 files changed, 195 insertions(+), 1 deletion(-)
> create mode 100644 libavfilter/af_tremolo.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 88cb3ce..5bbbaf0 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -2568,6 +2568,25 @@ slope
> Determine how steep is the filter's shelf transition.
> @end table
>
> +@section tremolo
> +
> +Sinusoidal amplitude modulation.
> +
> +The filter accepts the following options:
> +
> +@table @option
> +@item f
> +Modulation frequency in Hertz. Modulation frequencies in the subharmonic
range
> +(20 Hz or lower) will result in a tremolo effect.
> +This filter may also be used as a ring modulator by specifying
> +a modulation frequency higher than 20 Hz.
> +Range is 0.1 - 20000.0. Default value is 5.0 Hz.
> +
> +@item d
> +Depth of modulation as a percentage. Range is 0.0 - 1.0.
> +Default value is 0.5.
> +@end table
> +
> @section volume
>
> Adjust the input audio volume.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 05effd6..45fca3b 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -82,6 +82,7 @@ OBJS-$(CONFIG_SILENCEDETECT_FILTER) +=
af_silencedetect.o
> OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
> OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o
> OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o
> +OBJS-$(CONFIG_TREMOLO_FILTER) += af_tremolo.o
generate_wave_table.o
nit: Should be after treble.
> OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o
> OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
> OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o
> diff --git a/libavfilter/af_tremolo.c b/libavfilter/af_tremolo.c
> new file mode 100644
> index 0000000..7896127
> --- /dev/null
> +++ b/libavfilter/af_tremolo.c
> @@ -0,0 +1,173 @@
> +/*
> + * Tremolo
> + * Copyright (c) 2015 Kyle Swanson <k...@ylo.ph>. Some rights reserved.
What this means? Better remove: Some rights reserved.
> + *
> + * 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
> + */
> +
> +/**
> + * @file
> + * Tremolo
> + */
> +
> +#include "libavutil/opt.h"
> +#include "avfilter.h"
> +#include "internal.h"
> +#include "audio.h"
> +#include "generate_wave_table.h"
> +
> +typedef struct TremoloContext {
> + const AVClass *class;
> + double freq;
> + double depth;
> + double *wave_table;
> + int wave_table_index;
> + int sample_rate;
> +} TremoloContext;
> +
> +#define OFFSET(x) offsetof(TremoloContext, x)
> +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption tremolo_options[] = {
> + { "f", "set frequency in hertz", OFFSET(freq),
AV_OPT_TYPE_DOUBLE, {.dbl = 5.0}, 0.1, 20000.0, FLAGS },
> + { "d", "set depth as percentage", OFFSET(depth),
AV_OPT_TYPE_DOUBLE, {.dbl = 0.5}, 0.0, 1.0, FLAGS },
> + { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(tremolo);
> +
> +static float trem_env(AVFilterContext *ctx)
> +{
> + TremoloContext *s = ctx->priv;
> + float env = s->wave_table[s->wave_table_index];
> + s->wave_table_index++;
> + if (s->wave_table_index >= s->sample_rate / s->freq)
> + s->wave_table_index = 0;
> + return 1.0 - (s->depth * env);
> +}
Shouldnt this use doubles instead of floats?
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + AVFilterLink *outlink = ctx->outputs[0];
> + 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);
> + }
> +
> + int channels = inlink->channels;
> + int nb_samples = in->nb_samples;
> + double *dst = (double *)out->data[0];
> + int n, c;
> +
> + for (n = 0; n < nb_samples; n++) {
> + float env = trem_env(ctx);
> + for (c = 0; c < channels; c++) {
> + dst[c] *= env;
Out can be zero if frame is not writable. So use: dst - src * env.
> + }
> + dst += channels;
> + }
> +
> + if (in != out)
> + av_frame_free(&in);
> +
> + return ff_filter_frame(outlink, out);
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterFormats *formats;
> + AVFilterChannelLayouts *layouts;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_DBL,
> + AV_SAMPLE_FMT_NONE
> + };
> + int 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_make_format_list(sample_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ret = ff_set_common_formats(ctx, formats);
> + if (ret < 0)
> + return ret;
> +
> + formats = ff_all_samplerates();
> + if (!formats)
> + return AVERROR(ENOMEM);
> + return ff_set_common_samplerates(ctx, formats);
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> + TremoloContext *s = ctx->priv;
> + av_free(s->wave_table);
> +}
> +
> +static int config_input(AVFilterLink *inlink)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + TremoloContext *s = ctx->priv;
> + s->sample_rate = inlink->sample_rate;
> + s->wave_table = av_malloc_array(sizeof(double), s->sample_rate /
s->freq);
> + ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_DBL, s->wave_table,
s->sample_rate / s->freq, 0.0, 1.0, 0.0);
> + s->wave_table_index = 0;
> + return 0;
> +}
> +
> +static const AVFilterPad avfilter_af_tremolo_inputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .config_props = config_input,
> + .filter_frame = filter_frame,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad avfilter_af_tremolo_outputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + },
> + { NULL }
> +};
> +
> +AVFilter ff_af_tremolo = {
> + .name = "tremolo",
> + .description = NULL_IF_CONFIG_SMALL("Audio Tremolo."),
> + .priv_size = sizeof(TremoloContext),
> + .priv_class = &tremolo_class,
> + .uninit = uninit,
> + .query_formats = query_formats,
> + .inputs = avfilter_af_tremolo_inputs,
> + .outputs = avfilter_af_tremolo_outputs,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index cab4564..59ba5f5 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -105,6 +105,7 @@ void avfilter_register_all(void)
> REGISTER_FILTER(STEREOTOOLS, stereotools, af);
> REGISTER_FILTER(STEREOWIDEN, stereowiden, af);
> REGISTER_FILTER(TREBLE, treble, af);
> + REGISTER_FILTER(TREMOLO, tremolo, af);
> REGISTER_FILTER(VOLUME, volume, af);
> REGISTER_FILTER(VOLUMEDETECT, volumedetect, af);
>
> diff --git a/libavfilter/version.h b/libavfilter/version.h
> index e918184..9d44fd0 100644
> --- a/libavfilter/version.h
> +++ b/libavfilter/version.h
> @@ -30,7 +30,7 @@
> #include "libavutil/version.h"
>
> #define LIBAVFILTER_VERSION_MAJOR 6
> -#define LIBAVFILTER_VERSION_MINOR 5
> +#define LIBAVFILTER_VERSION_MINOR 6
> #define LIBAVFILTER_VERSION_MICRO 100
>
> #define LIBAVFILTER_VERSION_INT
AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
> --
> 1.8.4
>
Will do some minor changes, and apply after are questions I raised resolved.
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> ffmpeg-devel@ffmpeg.org
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