[FFmpeg-cvslog] avfilter: add flanger filter
Paul B Mahol
git at videolan.org
Thu Jul 3 10:53:28 CEST 2014
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Fri Jun 27 08:42:35 2014 +0000| [b52c26c66f65e0f9242e7effbf06ae2fd3e304f0] | committer: Paul B Mahol
avfilter: add flanger filter
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b52c26c66f65e0f9242e7effbf06ae2fd3e304f0
---
Changelog | 1 +
doc/filters.texi | 36 +++++++
libavfilter/Makefile | 1 +
libavfilter/af_flanger.c | 241 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 281 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 0346877..69f928d 100644
--- a/Changelog
+++ b/Changelog
@@ -30,6 +30,7 @@ version <next>:
- zoompan filter
- signalstats filter
- hqx filter (hq2x, hq3x, hq4x)
+- flanger filter
version 2.2:
diff --git a/doc/filters.texi b/doc/filters.texi
index f119a3a..ada33a7 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1439,6 +1439,42 @@ equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g
@end example
@end itemize
+ at section flanger
+Apply a flanging effect to the audio.
+
+The filter accepts the following options:
+
+ at table @option
+ at item delay
+Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
+
+ at item depth
+Set added swep delay in milliseconds. Range from 0 to 10. Default value is 2.
+
+ at item regen
+Set percentage regeneneration (delayed signal feedback). Range from -95 to 95.
+Default value is 0.
+
+ at item width
+Set percentage of delayed signal mixed with original. Range from 0 to 100.
+Default valu is 71.
+
+ at item speed
+Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
+
+ at item shape
+Set swept wave shape, can be @var{triangular} or @var{sinusoidal}.
+Default value is @var{sinusoidal}.
+
+ at item phase
+Set swept wave percentage-shift for multi channel. Range from 0 to 100.
+Default value is 25.
+
+ at item interp
+Set delay-line interpolation, @var{linear} or @var{quadratic}.
+Default is @var{linear}.
+ at end table
+
@section highpass
Apply a high-pass filter with 3dB point frequency.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 6acd43f..0f54381 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -69,6 +69,7 @@ OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o
OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o
+OBJS-$(CONFIG_FLANGER_FILTER) += af_flanger.o generate_wave_table.o
OBJS-$(CONFIG_HIGHPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
OBJS-$(CONFIG_LADSPA_FILTER) += af_ladspa.o
diff --git a/libavfilter/af_flanger.c b/libavfilter/af_flanger.c
new file mode 100644
index 0000000..5ff3786
--- /dev/null
+++ b/libavfilter/af_flanger.c
@@ -0,0 +1,241 @@
+/*
+ * Copyright (c) 2006 Rob Sykes <robs at users.sourceforge.net>
+ *
+ * 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/avstring.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "internal.h"
+#include "generate_wave_table.h"
+
+#define INTERPOLATION_LINEAR 0
+#define INTERPOLATION_QUADRATIC 1
+
+typedef struct FlangerContext {
+ const AVClass *class;
+ double delay_min;
+ double delay_depth;
+ double feedback_gain;
+ double delay_gain;
+ double speed;
+ int wave_shape;
+ double channel_phase;
+ int interpolation;
+ double in_gain;
+ int max_samples;
+ uint8_t **delay_buffer;
+ int delay_buf_pos;
+ double *delay_last;
+ float *lfo;
+ int lfo_length;
+ int lfo_pos;
+} FlangerContext;
+
+#define OFFSET(x) offsetof(FlangerContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption flanger_options[] = {
+ { "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
+ { "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
+ { "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
+ { "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
+ { "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
+ { "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
+ { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
+ { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
+ { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
+ { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
+ { "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
+ { "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
+ { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, "itype" },
+ { "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(flanger);
+
+static int init(AVFilterContext *ctx)
+{
+ FlangerContext *s = ctx->priv;
+
+ s->feedback_gain /= 100;
+ s->delay_gain /= 100;
+ s->channel_phase /= 100;
+ s->delay_min /= 1000;
+ s->delay_depth /= 1000;
+ s->in_gain = 1 / (1 + s->delay_gain);
+ s->delay_gain /= 1 + s->delay_gain;
+ s->delay_gain *= 1 - fabs(s->feedback_gain);
+
+ return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterChannelLayouts *layouts;
+ AVFilterFormats *formats;
+ 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 int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ FlangerContext *s = ctx->priv;
+
+ s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
+ s->lfo_length = inlink->sample_rate / s->speed;
+ s->delay_last = av_calloc(inlink->channels, sizeof(*s->delay_last));
+ s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
+ if (!s->lfo || !s->delay_last)
+ return AVERROR(ENOMEM);
+
+ ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
+ floor(s->delay_min * inlink->sample_rate + 0.5),
+ s->max_samples - 2., 3 * M_PI_2);
+
+ return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
+ inlink->channels, s->max_samples,
+ inlink->format, 0);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+ AVFilterContext *ctx = inlink->dst;
+ FlangerContext *s = ctx->priv;
+ AVFrame *out_frame;
+ int chan, i;
+
+ if (av_frame_is_writable(frame)) {
+ out_frame = frame;
+ } else {
+ out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
+ if (!out_frame)
+ return AVERROR(ENOMEM);
+ av_frame_copy_props(out_frame, frame);
+ }
+
+ for (i = 0; i < frame->nb_samples; i++) {
+
+ s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
+
+ for (chan = 0; chan < inlink->channels; chan++) {
+ double *src = (double *)frame->extended_data[chan];
+ double *dst = (double *)out_frame->extended_data[chan];
+ double delayed_0, delayed_1;
+ double delayed;
+ double in, out;
+ int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
+ double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
+ int int_delay = (int)delay;
+ double frac_delay = modf(delay, &delay);
+ double *delay_buffer = (double *)s->delay_buffer[chan];
+
+ in = src[i];
+ delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
+ s->feedback_gain;
+ delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
+ delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
+
+ if (s->interpolation == INTERPOLATION_LINEAR) {
+ delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
+ } else {
+ double a, b;
+ double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
+ delayed_2 -= delayed_0;
+ delayed_1 -= delayed_0;
+ a = delayed_2 * .5 - delayed_1;
+ b = delayed_1 * 2 - delayed_2 *.5;
+ delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
+ }
+
+ s->delay_last[chan] = delayed;
+ out = in * s->in_gain + delayed * s->delay_gain;
+ dst[i] = out;
+ }
+ s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
+ }
+
+ if (frame != out_frame)
+ av_frame_free(&frame);
+
+ return ff_filter_frame(ctx->outputs[0], out_frame);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ FlangerContext *s = ctx->priv;
+
+ av_freep(&s->lfo);
+ av_freep(&s->delay_last);
+
+ if (s->delay_buffer)
+ av_freep(&s->delay_buffer[0]);
+ av_freep(&s->delay_buffer);
+}
+
+static const AVFilterPad flanger_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad flanger_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_flanger = {
+ .name = "flanger",
+ .description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(FlangerContext),
+ .priv_class = &flanger_class,
+ .init = init,
+ .uninit = uninit,
+ .inputs = flanger_inputs,
+ .outputs = flanger_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index e4ac983..1877557 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -87,6 +87,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(EARWAX, earwax, af);
REGISTER_FILTER(EBUR128, ebur128, af);
REGISTER_FILTER(EQUALIZER, equalizer, af);
+ REGISTER_FILTER(FLANGER, flanger, af);
REGISTER_FILTER(HIGHPASS, highpass, af);
REGISTER_FILTER(JOIN, join, af);
REGISTER_FILTER(LADSPA, ladspa, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index f125032..bf9191e 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 4
-#define LIBAVFILTER_VERSION_MINOR 9
+#define LIBAVFILTER_VERSION_MINOR 10
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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