On Sat, Nov 28, 2015 at 5:26 PM, Paul B Mahol <one...@gmail.com> wrote:
> Signed-off-by: Paul B Mahol <one...@gmail.com>
> ---
> doc/filters.texi | 57 ++++++++++
> libavfilter/Makefile | 1 +
> libavfilter/af_apulsator.c | 270
> +++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 329 insertions(+)
> create mode 100644 libavfilter/af_apulsator.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index c8471e5..6d10a05 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -1027,6 +1027,63 @@ It accepts the following values:
> @end table
> @end table
>
> +@section apulsator
> +
> +Audio pulsator is something between an autopanner and a tremolo.
> +But it can produce funny stereo effects as well. Pulsator changes the volume
> +of left and right channel based on a LFO (low frequency oscillator) with
"left and right" -> "the left and right"
> +different waveforms and shifted phases.
> +This filter have ability to define an offset between left and right channel.
"have ability" -> "has the ability", "left and right" -> "the left and right"
> +An offset of 0 means that both LFO shapes match each other. Left and right
> +channel are altered equally - a conventional tremolo. An offset of 50% means
"Left and right channel" -> "The left and right channels"
> +that the shape of the right channel is exactly shifted in phase (or moved
> +backwards about half of the frequency) - Pulsator acts as an autopanner.
"Pulsator" -> "pulsator"
> +At 1 both curves match again. Every setting inbetween moves the phaseshift
"inbetween" -> "in between"
"phaseshift" -> "phase shift" or "phase-shift", prefer 1st
> +gapless between all stages and produces some "bypassing" sounds with sine and
> +triangle waveform. The more you set the offset near 1 (starting from the
"sine and triangle waveform" -> "sine and triangle waveforms", or
perhaps based on code "sine, triangle, square, sawup, or sawdown
waveforms". Up to you.
> +0.5) the faster the signal passes from left to right speaker.
"the 0.5" -> "0.5"
"left to right speaker" -> "the left to the right speaker".
> +
> +The filter accepts the following options:
> +
> +@table @option
> +@item level_in
> +Set input gain. By default it is 1. Range is between 0.015625 and 64.
> +
> +@item level_out
> +Set output gain. By default it is 1. Range is between 0.015625 and 64.
nit: 0.01625 and 64 inclusive or exclusive (i.e open or closed
interval) should be clarified since they are exactly representable
doubles.
> +
> +@item mode
> +Set waveform shape the LFO will use. Can be one of: sine, triangle, square,
> +sawup or sawdown. Default is sine.
> +
> +@item amount
> +Set modulation. Define how much of original signal is affected by the LFO.
> +
> +@item offset_l
> +Set left channel offset. Default is 0. Allowed range is from 0 to 1.
> +
> +@item offset_r
> +Set right channel offset. Default is 0.5. Allowed range is from 0 to 1.
Again, inclusive or exclusive.
> +
> +@item width
> +Set pulse width.
> +
> +@item timing
> +Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
> +
> +@item bpm
> +Set bpm. Default is 120. Allowed range is from 30 to 300. Only used if timing
> +is set to bpm.
> +
> +@item ms
> +Set ms. Default is 500. Allowed range is from 10 to 2000. Only used if timing
> +is set to ms
Same as above.
> +
> +@item hz
> +Set frequency in Hz. Default is 2. Allowed range is from 0.01 to 100. Only
> used
> +if timing is set to hz.
> +@end table
> +
> @anchor{aresample}
> @section aresample
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index e31bdaa..b6c0d7b 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -40,6 +40,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
> OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
> OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
> OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o
> generate_wave_table.o
> +OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
> OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
> OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
> OBJS-$(CONFIG_AREVERSE_FILTER) += f_reverse.o
> diff --git a/libavfilter/af_apulsator.c b/libavfilter/af_apulsator.c
> new file mode 100644
> index 0000000..c3579f4
> --- /dev/null
> +++ b/libavfilter/af_apulsator.c
> @@ -0,0 +1,270 @@
> +/*
> + * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald
> Johansen and others
> + *
> + * 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 "avfilter.h"
> +#include "internal.h"
> +#include "audio.h"
> +
> +enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES };
> +enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS };
> +
> +typedef struct SimpleLFO {
> + double phase;
> + double freq;
> + double offset;
> + double amount;
> + double pwidth;
> + int mode;
> + int srate;
> +} SimpleLFO;
I don't know the policy towards typedef'ed structures, kernel
explicitly forbids them. Seems like FFmpeg freely typedef's
structures, so feel free to ignore.
> +
> +typedef struct AudioPulsatorContext {
> + const AVClass *class;
> + int mode;
> + double level_in;
> + double level_out;
> + double amount;
> + double offset_l;
> + double offset_r;
> + double pwidth;
> + double bpm;
> + double hz;
> + int ms;
> + int timing;
> +
> + SimpleLFO lfoL, lfoR;
> +} AudioPulsatorContext;
> +
> +#define OFFSET(x) offsetof(AudioPulsatorContext, x)
> +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption apulsator_options[] = {
> + { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE,
> {.dbl=1}, 0.015625, 64, FLAGS, },
> + { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE,
> {.dbl=1}, 0.015625, 64, FLAGS, },
> + { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT,
> {.i64=SINE}, SINE, NB_MODES-1, FLAGS, "mode" },
> + { "sine", NULL, 0, AV_OPT_TYPE_CONST,
> {.i64=SINE}, 0, 0, FLAGS, "mode" },
> + { "triangle", NULL, 0, AV_OPT_TYPE_CONST,
> {.i64=TRIANGLE},0, 0, FLAGS, "mode" },
> + { "square", NULL, 0, AV_OPT_TYPE_CONST,
> {.i64=SQUARE}, 0, 0, FLAGS, "mode" },
> + { "sawup", NULL, 0, AV_OPT_TYPE_CONST,
> {.i64=SAWUP}, 0, 0, FLAGS, "mode" },
> + { "sawdown", NULL, 0, AV_OPT_TYPE_CONST,
> {.i64=SAWDOWN}, 0, 0, FLAGS, "mode" },
> + { "amount", "set modulation", OFFSET(amount), AV_OPT_TYPE_DOUBLE,
> {.dbl=1}, 0, 1, FLAGS },
> + { "offset_l", "set offset L", OFFSET(offset_l), AV_OPT_TYPE_DOUBLE,
> {.dbl=0}, 0, 1, FLAGS },
> + { "offset_r", "set offset R", OFFSET(offset_r), AV_OPT_TYPE_DOUBLE,
> {.dbl=.5}, 0, 1, FLAGS },
> + { "width", "set pulse width", OFFSET(pwidth), AV_OPT_TYPE_DOUBLE,
> {.dbl=1}, 0, 2, FLAGS },
> + { "timing", "set timing", OFFSET(timing), AV_OPT_TYPE_INT,
> {.i64=2}, 0, NB_TIMINGS-1, FLAGS, "timing" },
> + { "bpm", NULL, 0, AV_OPT_TYPE_CONST,
> {.i64=UNIT_BPM}, 0, 0, FLAGS, "timing" },
> + { "ms", NULL, 0, AV_OPT_TYPE_CONST,
> {.i64=UNIT_MS}, 0, 0, FLAGS, "timing" },
> + { "hz", NULL, 0, AV_OPT_TYPE_CONST,
> {.i64=UNIT_HZ}, 0, 0, FLAGS, "timing" },
> + { "bpm", "set BPM", OFFSET(bpm), AV_OPT_TYPE_DOUBLE,
> {.dbl=120}, 30, 300, FLAGS },
> + { "ms", "set ms", OFFSET(ms), AV_OPT_TYPE_INT,
> {.i64=500}, 10, 2000, FLAGS },
> + { "hz", "set frequency", OFFSET(hz), AV_OPT_TYPE_DOUBLE,
> {.dbl=2}, 0.01, 100, FLAGS },
> + { NULL }
> +};
Please do check a build with "-Wgnu-zero-variadic-macro-arguments" (on
clang) if easily available:
https://lists.ffmpeg.org/pipermail/ffmpeg-devel/2015-October/181970.html,
or verify otherwise that it is not an issue.
> +
> +AVFILTER_DEFINE_CLASS(apulsator);
> +
> +static void lfo_advance(SimpleLFO *lfo, unsigned count)
> +{
> + lfo->phase = fabs((lfo->phase + count * lfo->freq * (1.0 / lfo->srate)));
> + if (lfo->phase >= 1.)
> + lfo->phase = fmod(lfo->phase, 1.);
Minor nit: change all 1. to 1, they are exactly representable.
More useful one: remove 1.0 / , simply do a count * lfo->freq /
lfo->srate, the 1.0 is redundant.
> +}
> +
> +static double lfo_get_value(SimpleLFO *lfo)
> +{
> + double val;
> + double phs = FFMIN(100., lfo->phase / FFMIN(1.99, FFMAX(0.01,
> lfo->pwidth)) + lfo->offset);
why "magic" 1.99? Usually when I see such things, they are a bad form
of 2 - epsilon, and epsilon is picked out of a hat with no rationale.
If 1.99, 0.01 is genuinely needed for some reason, be it compatibility
or something technical, please add a relevant comment.
minor nit: change 100. to 100, this applies in various places
throughout this code. I won't point them out below for brevity.
> +
> + if (phs > 1)
> + phs = fmod(phs, 1.);
> +
> + switch (lfo->mode) {
> + case SINE:
> + val = sin((phs * 360.) * M_PI / 180);
Simplify to sin(phs * 2 * M_PI)
> + break;
> + case TRIANGLE:
> + if (phs > 0.75)
> + val = (phs - 0.75) * 4 - 1;
> + else if (phs > 0.5)
> + val = (phs - 0.5) * 4 * -1;
> + else if (phs > 0.25)
> + val = 1 - (phs - 0.25) * 4;
Squash the two cases for > 0.25, 0.5 into one; it is a straight line
segment and does not need an additional branch.
> + else
> + val = phs * 4;
> + break;
> + case SQUARE:
> + val = (phs < 0.5) ? -1 : +1;
> + break;
> + case SAWUP:
> + val = phs * 2. - 1;
> + break;
> + case SAWDOWN:
> + val = 1 - phs * 2.;
> + break;
> + }
> +
> + return val * lfo->amount;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + AVFilterLink *outlink = ctx->outputs[0];
> + AudioPulsatorContext *s = ctx->priv;
> + const double *src = (const double *)in->data[0];
> + const int nb_samples = in->nb_samples;
> + const double level_out = s->level_out;
> + const double level_in = s->level_in;
> + const double amount = s->amount;
> + AVFrame *out;
> + double *dst;
> + int n;
> +
> + 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);
> + }
> + dst = (double *)out->data[0];
> +
> + for (n = 0; n < nb_samples; n++) {
> + double outL;
> + double outR;
> + double inL = src[0] * level_in;
> + double inR = src[1] * level_in;
> + double procL = inL;
> + double procR = inR;
> +
> + procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2;
> + procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2;
> +
> + outL = procL + inL * (1. - amount);
> + outR = procR + inR * (1. - amount);
> +
> + outL *= level_out;
> + outR *= level_out;
> +
> + dst[0] = outL;
> + dst[1] = outR;
> +
> + lfo_advance(&s->lfoL, 1);
> + lfo_advance(&s->lfoR, 1);
> +
> + dst += 2;
> + src += 2;
> + }
> +
> + if (in != out)
> + av_frame_free(&in);
> +
> + return ff_filter_frame(outlink, out);
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterChannelLayouts *layout = NULL;
> + AVFilterFormats *formats;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_DBL,
> + AV_SAMPLE_FMT_NONE
> + };
> + int ret;
> +
> + ret = ff_add_channel_layout(&layout, AV_CH_LAYOUT_STEREO);
> + if (ret < 0)
> + return ret;
> + ret = ff_set_common_channel_layouts(ctx, layout);
> + 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);
A potential memleak issue: what happens if e.g channel layout stuff
succeeds and formats stuff fails? goto fail may be useful.
> +}
> +
> +static int config_input(AVFilterLink *inlink)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + AudioPulsatorContext *s = ctx->priv;
> + double freq;
> +
> + switch (s->timing) {
> + case UNIT_BPM: freq = s->bpm / 60.; break;
> + case UNIT_MS: freq = 1. / (s->ms / 1000); break;
> + case UNIT_HZ: freq = s->hz; break;
> + }
> +
> + s->lfoL.freq = freq;
> + s->lfoR.freq = freq;
> + s->lfoL.mode = s->mode;
> + s->lfoR.mode = s->mode;
> + s->lfoL.offset = s->offset_l;
> + s->lfoR.offset = s->offset_r;
> + s->lfoL.srate = inlink->sample_rate;
> + s->lfoR.srate = inlink->sample_rate;
> + s->lfoL.amount = s->amount;
> + s->lfoR.amount = s->amount;
> + s->lfoL.pwidth = s->pwidth;
> + s->lfoR.pwidth = s->pwidth;
> +
> + return 0;
> +}
> +
> +static const AVFilterPad inputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .config_props = config_input,
> + .filter_frame = filter_frame,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad outputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + },
> + { NULL }
> +};
> +
> +AVFilter ff_af_apulsator = {
> + .name = "apulsator",
> + .description = NULL_IF_CONFIG_SMALL("Audio pulsator."),
> + .priv_size = sizeof(AudioPulsatorContext),
> + .priv_class = &apulsator_class,
> + .query_formats = query_formats,
> + .inputs = inputs,
> + .outputs = outputs,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index ccd3f35..9502ebf 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -62,6 +62,7 @@ void avfilter_register_all(void)
> REGISTER_FILTER(APAD, apad, af);
> REGISTER_FILTER(APERMS, aperms, af);
> REGISTER_FILTER(APHASER, aphaser, af);
> + REGISTER_FILTER(APULSATOR, apulsator, af);
> REGISTER_FILTER(AREALTIME, arealtime, af);
> REGISTER_FILTER(ARESAMPLE, aresample, af);
> REGISTER_FILTER(AREVERSE, areverse, af);
> --
> 1.9.1
Note: I have not tested the filter; all the above is purely based on
examination of the code.
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
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