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/*
* 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 limiterthreshold;
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->limiterthreshold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out;
DCShiftContext *s = ctx->priv;
int i, j;
double dcshift = s->dcshift;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(outlink, in->nb_samples);
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->limiterthreshold && dcshift > 0) {
d = (d - s->limiterthreshold) * s->limitergain /
(INT32_MAX - s->limiterthreshold) +
s->limiterthreshold + dcshift;
} else if (d < -s->limiterthreshold && dcshift < 0) {
d = (d + s->limiterthreshold) * s->limitergain /
(INT32_MAX - s->limiterthreshold) -
s->limiterthreshold + 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);
}
}
}
if (out != in)
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,
},
};
static const AVFilterPad dcshift_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const 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,
FILTER_INPUTS(dcshift_inputs),
FILTER_OUTPUTS(dcshift_outputs),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
};
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