[FFmpeg-devel] [PATCH] avfilter: add dcshift filter
Paul B Mahol
onemda at gmail.com
Mon Feb 9 11:30:34 CET 2015
On 1/30/15, Michael Niedermayer <michaelni at gmx.at> wrote:
> On Fri, Jan 30, 2015 at 10:17:59AM +0000, Paul B Mahol wrote:
>> Signed-off-by: Paul B Mahol <onemda at 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
>>
>> + 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 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
>
Will apply it as is if there is no more comments.
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