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50ea7389ec
Lots of audio filters use very simple inputs or outputs: An array with a single AVFilterPad whose name is "default" and whose type is AVMEDIA_TYPE_AUDIO; everything else is unset. Given that we never use pointer equality for inputs or outputs*, we can simply use a single AVFilterPad instead of dozens; this even saves .data.rel.ro (4784B here) as well as relocations. *: In fact, several filters (like the filters in af_biquads.c) already use the same inputs; furthermore, ff_filter_alloc() duplicates the input and output pads so that we do not even work with the pads directly. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
136 lines
4.3 KiB
C
136 lines
4.3 KiB
C
/*
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* Copyright (c) 2000 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
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* Copyright (c) 2000 Fabien COELHO <fabien@coelho.net>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "internal.h"
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typedef struct DCShiftContext {
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const AVClass *class;
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double dcshift;
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double limiterthreshold;
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double limitergain;
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} DCShiftContext;
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#define OFFSET(x) offsetof(DCShiftContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption dcshift_options[] = {
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{ "shift", "set DC shift", OFFSET(dcshift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
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{ "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(dcshift);
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static av_cold int init(AVFilterContext *ctx)
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{
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DCShiftContext *s = ctx->priv;
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s->limiterthreshold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AVFrame *out;
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DCShiftContext *s = ctx->priv;
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int i, j;
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double dcshift = s->dcshift;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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if (s->limitergain > 0) {
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for (i = 0; i < inlink->ch_layout.nb_channels; i++) {
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const int32_t *src = (int32_t *)in->extended_data[i];
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int32_t *dst = (int32_t *)out->extended_data[i];
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for (j = 0; j < in->nb_samples; j++) {
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double d;
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d = src[j];
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if (d > s->limiterthreshold && dcshift > 0) {
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d = (d - s->limiterthreshold) * s->limitergain /
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(INT32_MAX - s->limiterthreshold) +
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s->limiterthreshold + dcshift;
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} else if (d < -s->limiterthreshold && dcshift < 0) {
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d = (d + s->limiterthreshold) * s->limitergain /
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(INT32_MAX - s->limiterthreshold) -
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s->limiterthreshold + dcshift;
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} else {
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d = dcshift * INT32_MAX + d;
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}
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dst[j] = av_clipl_int32(d);
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}
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}
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} else {
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for (i = 0; i < inlink->ch_layout.nb_channels; i++) {
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const int32_t *src = (int32_t *)in->extended_data[i];
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int32_t *dst = (int32_t *)out->extended_data[i];
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for (j = 0; j < in->nb_samples; j++) {
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double d = dcshift * (INT32_MAX + 1.) + src[j];
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dst[j] = av_clipl_int32(d);
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}
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}
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}
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if (out != in)
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static const AVFilterPad dcshift_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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},
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};
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const AVFilter ff_af_dcshift = {
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.name = "dcshift",
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.description = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
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.priv_size = sizeof(DCShiftContext),
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.priv_class = &dcshift_class,
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.init = init,
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FILTER_INPUTS(dcshift_inputs),
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FILTER_OUTPUTS(ff_audio_default_filterpad),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_S32P),
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
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};
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