/* * 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 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->ch_layout.nb_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->ch_layout.nb_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."), .priv_size = sizeof(DCShiftContext), .priv_class = &dcshift_class, .init = init, FILTER_INPUTS(dcshift_inputs), FILTER_OUTPUTS(dcshift_outputs), FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_S32P), .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,};
/*
* 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 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->ch_layout.nb_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->ch_layout.nb_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."),
.priv_size = sizeof(DCShiftContext),
.priv_class = &dcshift_class,
.init = init,
FILTER_INPUTS(dcshift_inputs),
FILTER_OUTPUTS(dcshift_outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_S32P),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
};