avfilter: add dcshift filter

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

1 files changed, 164 insertions, 0 deletions

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,
+};