[FFmpeg-devel] [PATCH] aphaser filter
Stefano Sabatini
stefasab at gmail.com
Sun Mar 31 23:00:28 CEST 2013
On date Saturday 2013-03-30 21:55:29 +0000, Paul B Mahol encoded:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> doc/filters.texi | 25 +++++
> libavfilter/Makefile | 1 +
> libavfilter/af_aphaser.c | 271 +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 298 insertions(+)
> create mode 100644 libavfilter/af_aphaser.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 4190cca..2b8e58b 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -6262,6 +6262,31 @@ following one, the permission might not be received as expected in that
> following filter. Inserting a @ref{format} or @ref{aformat} filter before the
> perms/aperms filter can avoid this problem.
>
> + at section aphaser
> +Add a phasing effect to the input audio.
> +
Maybe add a note about what a phaser is and what is used for.
> +The filter accepts parameters as a list of @var{key}=@var{value}
> +pairs, separated by ":".
> +
> +A description of the accepted parameters follows.
> +
> + at table @option
> + at item in_gain
> +Set input gain. Default is 0.4.
> +
> + at item out_gain
> +Set output gain. Default is 0.74
> +
> + at item delay
> +Set delay in miliseconds. Default is 3.0.
typo
> +
> + at item decay
> +Set decay. Default is 0.4.
> +
> + at item speed
> +Set modulation speed in Hz. Default is 0.5.
You could mention the ranges for the various options.
> + at end table
> +
> @section aselect, select
> Select frames to pass in output.
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 690b1cb..a4bdf2e 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -57,6 +57,7 @@ OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
> OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
> OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
> OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
> +OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o
> OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
> OBJS-$(CONFIG_ASELECT_FILTER) += f_select.o
> OBJS-$(CONFIG_ASENDCMD_FILTER) += f_sendcmd.o
> diff --git a/libavfilter/af_aphaser.c b/libavfilter/af_aphaser.c
> new file mode 100644
> index 0000000..dd5fd4c
> --- /dev/null
> +++ b/libavfilter/af_aphaser.c
> @@ -0,0 +1,271 @@
> +/*
> + * Copyright (c) 2013 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
> + */
> +
> +/**
> + * @file
> + * phaser audio filter
> + */
> +
> +#include "libavutil/avassert.h"
> +#include "libavutil/opt.h"
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "internal.h"
> +
> +enum WaveType {
> + WAVE_SINE,
> + WAVE_TRIANGLE,
> +};
> +
> +typedef struct {
> + const AVClass *class;
> + double in_gain, out_gain;
> + double delay;
> + double decay;
> + double speed;
> +
> + int delay_buffer_length;
> + double *delay_buffer;
> +
> + int modulation_buffer_length;
> + int32_t *modulation_buffer;
> +
> + int delay_pos, modulation_pos;
> +} AudioPhaserContext;
> +
> +#define OFFSET(x) offsetof(AudioPhaserContext, x)
> +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption aphaser_options[] = {
> + { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
> + { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
> + { "delay", "set delay in miliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
mili -> milli typo
> + { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
> + { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
> + { NULL },
> +};
> +
> +AVFILTER_DEFINE_CLASS(aphaser);
> +
> +static av_cold int init(AVFilterContext *ctx, const char *args)
> +{
> + AudioPhaserContext *p = ctx->priv;
> +
> + if (p->in_gain > (1 - p->decay * p->decay))
> + av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
> + if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
> + av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
> +
> + return 0;
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterFormats *formats;
> + AVFilterChannelLayouts *layouts;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_DBLP,
> + AV_SAMPLE_FMT_NONE
> + };
> +
> + layouts = ff_all_channel_layouts();
> + if (!layouts)
> + return AVERROR(ENOMEM);
> + ff_set_common_channel_layouts(ctx, layouts);
> +
> + formats = ff_make_format_list(sample_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_set_common_formats(ctx, formats);
> +
> + formats = ff_all_samplerates();
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_set_common_samplerates(ctx, formats);
> +
> + return 0;
> +}
> +
> +static void generate_wave_table(int wave_type, enum AVSampleFormat sample_fmt,
enum WaveType wave_type
> + void *table, int table_size,
> + double min, double max, double phase)
> +{
> + uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5;
> +
> + for (i = 0; i < table_size; i++) {
> + uint32_t point = (i + phase_offset) % table_size;
> + double d;
> +
> + switch (wave_type) {
> + case WAVE_SINE:
> + d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;
> + break;
> + case WAVE_TRIANGLE:
> + d = (double)point * 2 / table_size;
> + switch (4 * point / table_size) {
> + case 0: d = d + 0.5; break;
> + case 1:
> + case 2: d = 1.5 - d; break;
> + case 3: d = d - 1.5; break;
> + }
> + break;
> + default:
> + av_assert0(0);
> + }
> +
> + d = d * (max - min) + min;
> + switch (sample_fmt) {
> + case AV_SAMPLE_FMT_FLT: {
> + float *fp = (float *)table;
> + *fp++ = (float)d;
> + table = fp;
> + continue; }
> + case AV_SAMPLE_FMT_DBL: {
> + double *dp = (double *)table;
> + *dp++ = d;
> + table = dp;
> + continue; }
> + }
> +
> + d += d < 0 ? -0.5 : +0.5;
> + switch (sample_fmt) {
> + case AV_SAMPLE_FMT_S16: {
> + int16_t *sp = table;
> + *sp++ = (int16_t)d;
> + table = sp;
> + continue; }
> + case AV_SAMPLE_FMT_S32: {
> + int32_t *ip = table;
> + *ip++ = (int32_t)d;
> + table = ip;
> + continue; }
> + default:
> + av_assert0(0);
> + }
> + }
> +}
> +
> +static int config_input(AVFilterLink *inlink)
> +{
> + AudioPhaserContext *p = inlink->dst->priv;
> +
> + p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
> + p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
> + p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
> + p->modulation_buffer = av_malloc(p->modulation_buffer_length * sizeof(*p->modulation_buffer));
> +
> + if (!p->modulation_buffer || !p->delay_buffer)
> + return AVERROR(ENOMEM);
> +
> + generate_wave_table(WAVE_TRIANGLE, AV_SAMPLE_FMT_S32,
> + p->modulation_buffer, p->modulation_buffer_length,
> + 1., (double)p->delay_buffer_length, M_PI / 2.0);
> +
> + p->delay_pos = p->modulation_pos = 0;
> +
> + return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
> +{
> + AudioPhaserContext *p = inlink->dst->priv;
> + AVFilterLink *outlink = inlink->dst->outputs[0];
> + AVFrame *out_buf;
> + int i, c, delay_pos, modulation_pos;
> +
> + if (av_frame_is_writable(buf)) {
> + out_buf = buf;
> + } else {
> + out_buf = ff_get_audio_buffer(inlink, buf->nb_samples);
> + if (!out_buf)
> + return AVERROR(ENOMEM);
> + out_buf->pts = buf->pts;
copy props
> + }
> +
> + for (c = 0; c < av_frame_get_channels(buf); c++) {
> + double *src = (double *)buf->extended_data[c];
> + double *dst = (double *)out_buf->extended_data[c];
> + double *buffer = p->delay_buffer + c * p->delay_buffer_length;
> +
> + delay_pos = p->delay_pos;
> + modulation_pos = p->modulation_pos;
> +
> + for (i = 0; i < buf->nb_samples; i++, src++, dst++) {
> + double d = *src * p->in_gain + buffer[
> + (delay_pos + p->modulation_buffer[modulation_pos]) %
> + p->delay_buffer_length] * p->decay;
> +
> + modulation_pos = (modulation_pos + 1) % p->modulation_buffer_length;
> + delay_pos = (delay_pos + 1) % p->delay_buffer_length;
> + buffer[delay_pos] = d;
> +
> + *dst = d * p->out_gain;
> + }
> + }
> +
> + p->delay_pos = delay_pos;
> + p->modulation_pos = modulation_pos;
> +
> + if (buf != out_buf)
> + av_frame_free(&buf);
> +
> + return ff_filter_frame(outlink, out_buf);
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> + AudioPhaserContext *p = ctx->priv;
> +
> + av_freep(&p->delay_buffer);
> + av_freep(&p->modulation_buffer);
> +}
> +
> +static const AVFilterPad aphaser_inputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = filter_frame,
> + .config_props = config_input,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad aphaser_outputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + },
> + { NULL }
> +};
> +
> +static const char *const shorthand[] = { "in_gain", "out_gain", "delay", "decay", "speed", NULL };
> +
> +AVFilter avfilter_af_aphaser = {
> + .name = "aphaser",
> + .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
> + .query_formats = query_formats,
> + .priv_size = sizeof(AudioPhaserContext),
> + .init = init,
> + .uninit = uninit,
> + .inputs = aphaser_inputs,
> + .outputs = aphaser_outputs,
> + .priv_class = &aphaser_class,
> + .shorthand = shorthand,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 45a67e5..287d459 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -53,6 +53,7 @@ void avfilter_register_all(void)
> REGISTER_FILTER(ANULL, anull, af);
> REGISTER_FILTER(APAD, apad, af);
> REGISTER_FILTER(APERMS, aperms, af);
> + REGISTER_FILTER(APHASER, aphaser, af);
> REGISTER_FILTER(ARESAMPLE, aresample, af);
> REGISTER_FILTER(ASELECT, aselect, af);
> REGISTER_FILTER(ASENDCMD, asendcmd, af);
No more comments from me, can't comment on the algo itself but I can't
spot evident mistakes.
It could be useful to compare the output with the one from the sox
wrapper.
Thanks.
--
FFmpeg = Fascinating Fascinating Multimedia Plastic Earthshaking God
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