[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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