avfilter: add flanger filter
authorPaul B Mahol <onemda@gmail.com>
Fri, 27 Jun 2014 08:42:35 +0000 (08:42 +0000)
committerPaul B Mahol <onemda@gmail.com>
Thu, 3 Jul 2014 08:07:42 +0000 (08:07 +0000)
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Changelog
doc/filters.texi
libavfilter/Makefile
libavfilter/af_flanger.c [new file with mode: 0644]
libavfilter/allfilters.c
libavfilter/version.h

index 0346877..69f928d 100644 (file)
--- a/Changelog
+++ b/Changelog
@@ -30,6 +30,7 @@ version <next>:
 - zoompan filter
 - signalstats filter
 - hqx filter (hq2x, hq3x, hq4x)
+- flanger filter
 
 
 version 2.2:
index f119a3a..ada33a7 100644 (file)
@@ -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
 
+@section flanger
+Apply a flanging effect to the audio.
+
+The filter accepts the following options:
+
+@table @option
+@item delay
+Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
+
+@item depth
+Set added swep delay in milliseconds. Range from 0 to 10. Default value is 2.
+
+@item regen
+Set percentage regeneneration (delayed signal feedback). Range from -95 to 95.
+Default value is 0.
+
+@item width
+Set percentage of delayed signal mixed with original. Range from 0 to 100.
+Default valu is 71.
+
+@item speed
+Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
+
+@item shape
+Set swept wave shape, can be @var{triangular} or @var{sinusoidal}.
+Default value is @var{sinusoidal}.
+
+@item phase
+Set swept wave percentage-shift for multi channel. Range from 0 to 100.
+Default value is 25.
+
+@item interp
+Set delay-line interpolation, @var{linear} or @var{quadratic}.
+Default is @var{linear}.
+@end table
+
 @section highpass
 
 Apply a high-pass filter with 3dB point frequency.
index 6acd43f..0f54381 100644 (file)
@@ -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 (file)
index 0000000..5ff3786
--- /dev/null
@@ -0,0 +1,241 @@
+/*
+ * Copyright (c) 2006 Rob Sykes <robs@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,
+};
index e4ac983..1877557 100644 (file)
@@ -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);
index f125032..bf9191e 100644 (file)
@@ -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, \