[FFmpeg-cvslog] avfilter: add dcshift filter
Paul B Mahol
git at videolan.org
Wed Feb 11 17:21:26 CET 2015
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed Jan 28 15:46:58 2015 +0000| [edf217ebb7d518be3030184d03b5534033e82d0f] | committer: Paul B Mahol
avfilter: add dcshift filter
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=edf217ebb7d518be3030184d03b5534033e82d0f
---
Changelog | 1 +
doc/filters.texi | 19 ++++++
libavfilter/Makefile | 1 +
libavfilter/af_dcshift.c | 164 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 4 +-
6 files changed, 188 insertions(+), 2 deletions(-)
diff --git a/Changelog b/Changelog
index 49b4793..c663d5e 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 2f29c46..8069554 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
+ at 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
+ at 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.
+
+ at table @option
+ at item shift
+Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift
+the audio.
+
+ at 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.
+ at end table
+
@section earwax
Make audio easier to listen to on headphones.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 21a3fbe..7d1ea91 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 0000000..c1abb3c
--- /dev/null
+++ b/libavfilter/af_dcshift.c
@@ -0,0 +1,164 @@
+/*
+ * Copyright (c) 2000 Chris Ausbrooks <weed at bucket.pp.ualr.edu>
+ * Copyright (c) 2000 Fabien COELHO <fabien at 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 9c6f2ae..62d3eb3 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 49ea3a9..4e50688 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, \
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