@end example
@end itemize
+@section compensationdelay
+
+Compensation Delay Line is a metric based delay to compensate differing
+positions of microphones or speakers.
+
+For example, you have recorded guitar with two microphones placed in
+different location. Because the front of sound wave has fixed speed in
+normal conditions, the phasing of microphones can vary and depends on
+their location and interposition. The best sound mix can be achieved when
+these microphones are in phase (synchronized). Note that distance of
+~30 cm between microphones makes one microphone to capture signal in
+antiphase to another microphone. That makes the final mix sounding moody.
+This filter helps to solve phasing problems by adding different delays
+to each microphone track and make them synchronized.
+
+The best result can be reached when you take one track as base and
+synchronize other tracks one by one with it.
+Remember that synchronization/delay tolerance depends on sample rate, too.
+Higher sample rates will give more tolerance.
+
+It accepts the following parameters:
+
+@table @option
+@item mm
+Set millimeters distance. This is compensation distance for fine tuning.
+Default is 0.
+
+@item cm
+Set cm distance. This is compensation distance for tightening distance setup.
+Default is 0.
+
+@item m
+Set meters distance. This is compensation distance for hard distance setup.
+Default is 0.
+
+@item dry
+Set dry amount. Amount of unprocessed (dry) signal.
+Default is 0.
+
+@item wet
+Set wet amount. Amount of processed (wet) signal.
+Default is 1.
+
+@item temp
+Set temperature degree in Celsius. This is the temperature of the environment.
+Default is 20.
+@end table
+
@section dcshift
Apply a DC shift to the audio.
--- /dev/null
+/*
+ * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Vladimir Sadovnikov and others
+ * Copyright (c) 2015 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
+ */
+
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "internal.h"
+
+typedef struct CompensationDelayContext {
+ const AVClass *class;
+ int distance_mm;
+ int distance_cm;
+ int distance_m;
+ double dry, wet;
+ int temp;
+
+ unsigned delay;
+ unsigned w_ptr;
+ unsigned buf_size;
+ AVFrame *delay_frame;
+} CompensationDelayContext;
+
+#define OFFSET(x) offsetof(CompensationDelayContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption compensationdelay_options[] = {
+ { "mm", "set mm distance", OFFSET(distance_mm), AV_OPT_TYPE_INT, {.i64=0}, 0, 10, A },
+ { "cm", "set cm distance", OFFSET(distance_cm), AV_OPT_TYPE_INT, {.i64=0}, 0, 100, A },
+ { "m", "set meter distance", OFFSET(distance_m), AV_OPT_TYPE_INT, {.i64=0}, 0, 100, A },
+ { "dry", "set dry amount", OFFSET(dry), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
+ { "wet", "set wet amount", OFFSET(wet), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A },
+ { "temp", "set temperature °C", OFFSET(temp), AV_OPT_TYPE_INT, {.i64=20}, -50, 50, A },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(compensationdelay);
+
+// The maximum distance for options
+#define COMP_DELAY_MAX_DISTANCE (100.0 * 100.0 + 100.0 * 1.0 + 1.0)
+// The actual speed of sound in normal conditions
+#define COMP_DELAY_SOUND_SPEED_KM_H(temp) 1.85325 * (643.95 * pow(((temp + 273.15) / 273.15), 0.5))
+#define COMP_DELAY_SOUND_SPEED_CM_S(temp) (COMP_DELAY_SOUND_SPEED_KM_H(temp) * (1000.0 * 100.0) /* cm/km */ / (60.0 * 60.0) /* s/h */)
+#define COMP_DELAY_SOUND_FRONT_DELAY(temp) (1.0 / COMP_DELAY_SOUND_SPEED_CM_S(temp))
+// The maximum delay may be reached by this filter
+#define COMP_DELAY_MAX_DELAY (COMP_DELAY_MAX_DISTANCE * COMP_DELAY_SOUND_FRONT_DELAY(50))
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterChannelLayouts *layouts;
+ AVFilterFormats *formats;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBLP,
+ 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 int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ CompensationDelayContext *s = ctx->priv;
+ unsigned min_size, new_size = 1;
+
+ s->delay = (s->distance_m * 100. + s->distance_cm * 1. + s->distance_mm * .1) *
+ COMP_DELAY_SOUND_FRONT_DELAY(s->temp) * inlink->sample_rate;
+ min_size = inlink->sample_rate * COMP_DELAY_MAX_DELAY;
+
+ while (new_size < min_size)
+ new_size <<= 1;
+
+ s->delay_frame = av_frame_alloc();
+ if (!s->delay_frame)
+ return AVERROR(ENOMEM);
+
+ s->buf_size = new_size;
+ s->delay_frame->format = inlink->format;
+ s->delay_frame->nb_samples = new_size;
+ s->delay_frame->channel_layout = inlink->channel_layout;
+
+ return av_frame_get_buffer(s->delay_frame, 32);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ CompensationDelayContext *s = ctx->priv;
+ const unsigned b_mask = s->buf_size - 1;
+ const unsigned buf_size = s->buf_size;
+ const unsigned delay = s->delay;
+ const double dry = s->dry;
+ const double wet = s->wet;
+ unsigned r_ptr, w_ptr;
+ AVFrame *out;
+ int n, ch;
+
+ out = ff_get_audio_buffer(inlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+
+ for (ch = 0; ch < inlink->channels; ch++) {
+ const double *src = (const double *)in->extended_data[ch];
+ double *dst = (double *)out->extended_data[ch];
+ double *buffer = (double *)s->delay_frame->extended_data[ch];
+
+ w_ptr = s->w_ptr;
+ r_ptr = (w_ptr + buf_size - delay) & b_mask;
+
+ for (n = 0; n < in->nb_samples; n++) {
+ const double sample = src[n];
+
+ buffer[w_ptr] = sample;
+ dst[n] = dry * sample + wet * buffer[r_ptr];
+ w_ptr = (w_ptr + 1) & b_mask;
+ r_ptr = (r_ptr + 1) & b_mask;
+ }
+ }
+ s->w_ptr = w_ptr;
+
+ av_frame_free(&in);
+ return ff_filter_frame(ctx->outputs[0], out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ CompensationDelayContext *s = ctx->priv;
+
+ av_frame_free(&s->delay_frame);
+}
+
+static const AVFilterPad compensationdelay_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad compensationdelay_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_compensationdelay = {
+ .name = "compensationdelay",
+ .description = NULL_IF_CONFIG_SMALL("Audio Compensation Delay Line."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(CompensationDelayContext),
+ .priv_class = &compensationdelay_class,
+ .uninit = uninit,
+ .inputs = compensationdelay_inputs,
+ .outputs = compensationdelay_outputs,
+};
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