On Fri, Jan 30, 2015 at 10:17:59AM +0000, Paul B Mahol wrote:
> Signed-off-by: Paul B Mahol <one...@gmail.com>
> ---
> doc/filters.texi | 19 ++++++
> libavfilter/Makefile | 1 +
> libavfilter/af_dcshift.c | 161
> +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 182 insertions(+)
> create mode 100644 libavfilter/af_dcshift.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index cba2697..27a745f 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 7e0d456..60072f9 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..25fc66a
> --- /dev/null
> +++ b/libavfilter/af_dcshift.c
> @@ -0,0 +1,161 @@
> +/*
> + * 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_c(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_c(d);i think this should use some rounding function like llrint() ? though with 32bit precission it probably doesnt really matter much [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB No human being will ever know the Truth, for even if they happen to say it by chance, they would not even known they had done so. -- Xenophanes
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