avfilter: add dcshift filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

6 files changed, 188 insertions, 2 deletions

diff --git a/Changelog b/Changelog
index 49b4793f01..c663d5eb84 100644
--- a/Changelog
+++ b/Changelog
@@ -22,6 +22,7 @@ version <next>:
 - removed libmpcodecs
 - Changed default DNxHD colour range in QuickTime .mov derivatives to mpeg range
 - ported softpulldown filter from libmpcodecs as repeatfields filter
+- dcshift filter
 
 
 version 2.5:
diff --git a/doc/filters.texi b/doc/filters.texi
index 2f29c46f9c..8069554eb4 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -917,6 +917,7 @@ audio, the data is treated as if all the planes were concatenated.
 A list of Adler-32 checksums for each data plane.
 @end table
 
+@anchor{astats}
 @section astats
 
 Display time domain statistical information about the audio channels.
@@ -1394,6 +1395,24 @@ compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
 @end example
 @end itemize
 
+@section dcshift
+Apply a DC shift to the audio.
+
+This can be useful to remove a DC offset (caused perhaps by a hardware problem
+in the recording chain) from the audio. The effect of a DC offset is reduced
+headroom and hence volume. The @ref{astats} filter can be used to determine if
+a signal has a DC offset.
+
+@table @option
+@item shift
+Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift
+the audio.
+
+@item limitergain
+Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
+used to prevent clipping.
+@end table
+
 @section earwax
 
 Make audio easier to listen to on headphones.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 21a3fbe7ce..7d1ea9199b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -65,6 +65,7 @@ OBJS-$(CONFIG_BS2B_FILTER)                   += af_bs2b.o
 OBJS-$(CONFIG_CHANNELMAP_FILTER)             += af_channelmap.o
 OBJS-$(CONFIG_CHANNELSPLIT_FILTER)           += af_channelsplit.o
 OBJS-$(CONFIG_COMPAND_FILTER)                += af_compand.o
+OBJS-$(CONFIG_DCSHIFT_FILTER)                += af_dcshift.o
 OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o
 OBJS-$(CONFIG_EBUR128_FILTER)                += f_ebur128.o
 OBJS-$(CONFIG_EQUALIZER_FILTER)              += af_biquads.o
diff --git a/libavfilter/af_dcshift.c b/libavfilter/af_dcshift.c
new file mode 100644
index 0000000000..c1abb3c365
--- /dev/null
+++ b/libavfilter/af_dcshift.c
@@ -0,0 +1,164 @@
+/*
+ * Copyright (c) 2000 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
+ * Copyright (c) 2000 Fabien COELHO <fabien@coelho.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/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "internal.h"
+
+typedef struct DCShiftContext {
+    const AVClass *class;
+    double dcshift;
+    double limiterthreshhold;
+    double limitergain;
+} DCShiftContext;
+
+#define OFFSET(x) offsetof(DCShiftContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption dcshift_options[] = {
+    { "shift",       "set DC shift",     OFFSET(dcshift),       AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
+    { "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(dcshift);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    DCShiftContext *s = ctx->priv;
+
+    s->limiterthreshhold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));
+
+    return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterChannelLayouts *layouts;
+    AVFilterFormats *formats;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_S32P, 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 filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AVFrame *out = ff_get_audio_buffer(inlink, in->nb_samples);
+    DCShiftContext *s = ctx->priv;
+    int i, j;
+    double dcshift = s->dcshift;
+
+    if (!out) {
+        av_frame_free(&in);
+        return AVERROR(ENOMEM);
+    }
+    av_frame_copy_props(out, in);
+
+    if (s->limitergain > 0) {
+        for (i = 0; i < inlink->channels; i++) {
+            const int32_t *src = (int32_t *)in->extended_data[i];
+            int32_t *dst = (int32_t *)out->extended_data[i];
+
+            for (j = 0; j < in->nb_samples; j++) {
+                double d;
+
+                d = src[j];
+
+                if (d > s->limiterthreshhold && dcshift > 0) {
+                    d = (d - s->limiterthreshhold) * s->limitergain /
+                             (INT32_MAX - s->limiterthreshhold) +
+                             s->limiterthreshhold + dcshift;
+                } else if (d < -s->limiterthreshhold && dcshift < 0) {
+                    d = (d + s->limiterthreshhold) * s->limitergain /
+                             (INT32_MAX - s->limiterthreshhold) -
+                             s->limiterthreshhold + dcshift;
+                } else {
+                    d = dcshift * INT32_MAX + d;
+                }
+
+                dst[j] = av_clipl_int32(d);
+            }
+        }
+    } else {
+        for (i = 0; i < inlink->channels; i++) {
+            const int32_t *src = (int32_t *)in->extended_data[i];
+            int32_t *dst = (int32_t *)out->extended_data[i];
+
+            for (j = 0; j < in->nb_samples; j++) {
+                double d = dcshift * (INT32_MAX + 1.) + src[j];
+
+                dst[j] = av_clipl_int32(d);
+            }
+        }
+    }
+
+    av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+static const AVFilterPad dcshift_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad dcshift_outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_dcshift = {
+    .name           = "dcshift",
+    .description    = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(DCShiftContext),
+    .priv_class     = &dcshift_class,
+    .init           = init,
+    .inputs         = dcshift_inputs,
+    .outputs        = dcshift_outputs,
+    .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 9c6f2aeb0c..62d3eb33ce 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -81,6 +81,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(CHANNELMAP,     channelmap,     af);
     REGISTER_FILTER(CHANNELSPLIT,   channelsplit,   af);
     REGISTER_FILTER(COMPAND,        compand,        af);
+    REGISTER_FILTER(DCSHIFT,        dcshift,        af);
     REGISTER_FILTER(EARWAX,         earwax,         af);
     REGISTER_FILTER(EBUR128,        ebur128,        af);
     REGISTER_FILTER(EQUALIZER,      equalizer,      af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 49ea3a90da..4e506888e0 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,8 +30,8 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  5
-#define LIBAVFILTER_VERSION_MINOR  9
-#define LIBAVFILTER_VERSION_MICRO 104
+#define LIBAVFILTER_VERSION_MINOR  10
+#define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
                                                LIBAVFILTER_VERSION_MINOR, \