1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-02 03:06:28 +02:00
FFmpeg/libavfilter/af_bs2b.c
Anton Khirnov 6d75d44d90 lavfi: drop internal.h
All that remains in it are things that belong in avfilter_internal.h.

Move them there and remove internal.h
2024-08-19 21:48:04 +02:00

216 lines
5.9 KiB
C

/*
* 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
*/
/**
* @file
* Bauer stereo-to-binaural filter
*/
#include <bs2b.h>
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
typedef void (*filter_func)(t_bs2bdp bs2bdp, uint8_t *sample, int n);
typedef struct Bs2bContext {
const AVClass *class;
int profile;
int fcut;
int feed;
t_bs2bdp bs2bp;
filter_func filter;
} Bs2bContext;
#define OFFSET(x) offsetof(Bs2bContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
static const AVOption bs2b_options[] = {
{ "profile", "Apply a pre-defined crossfeed level",
OFFSET(profile), AV_OPT_TYPE_INT, { .i64 = BS2B_DEFAULT_CLEVEL }, 0, INT_MAX, A, .unit = "profile" },
{ "default", "default profile", 0, AV_OPT_TYPE_CONST, { .i64 = BS2B_DEFAULT_CLEVEL }, 0, 0, A, .unit = "profile" },
{ "cmoy", "Chu Moy circuit", 0, AV_OPT_TYPE_CONST, { .i64 = BS2B_CMOY_CLEVEL }, 0, 0, A, .unit = "profile" },
{ "jmeier", "Jan Meier circuit", 0, AV_OPT_TYPE_CONST, { .i64 = BS2B_JMEIER_CLEVEL }, 0, 0, A, .unit = "profile" },
{ "fcut", "Set cut frequency (in Hz)",
OFFSET(fcut), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, BS2B_MAXFCUT, A },
{ "feed", "Set feed level (in Hz)",
OFFSET(feed), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, BS2B_MAXFEED, A },
{ NULL },
};
AVFILTER_DEFINE_CLASS(bs2b);
static av_cold int init(AVFilterContext *ctx)
{
Bs2bContext *bs2b = ctx->priv;
if (!(bs2b->bs2bp = bs2b_open()))
return AVERROR(ENOMEM);
bs2b_set_level(bs2b->bs2bp, bs2b->profile);
if (bs2b->fcut)
bs2b_set_level_fcut(bs2b->bs2bp, bs2b->fcut);
if (bs2b->feed)
bs2b_set_level_feed(bs2b->bs2bp, bs2b->feed);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
Bs2bContext *bs2b = ctx->priv;
if (bs2b->bs2bp)
bs2b_close(bs2b->bs2bp);
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterChannelLayouts *layouts = NULL;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE,
};
int ret;
if (ff_add_channel_layout(&layouts, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO) != 0)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
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 *frame)
{
int ret;
AVFrame *out_frame;
Bs2bContext *bs2b = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
if (av_frame_is_writable(frame)) {
out_frame = frame;
} else {
out_frame = ff_get_audio_buffer(outlink, frame->nb_samples);
if (!out_frame) {
av_frame_free(&frame);
return AVERROR(ENOMEM);
}
av_frame_copy(out_frame, frame);
ret = av_frame_copy_props(out_frame, frame);
if (ret < 0) {
av_frame_free(&out_frame);
av_frame_free(&frame);
return ret;
}
}
bs2b->filter(bs2b->bs2bp, out_frame->extended_data[0], out_frame->nb_samples);
if (frame != out_frame)
av_frame_free(&frame);
return ff_filter_frame(outlink, out_frame);
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
Bs2bContext *bs2b = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
int srate = inlink->sample_rate;
switch (inlink->format) {
case AV_SAMPLE_FMT_U8:
bs2b->filter = (filter_func) bs2b_cross_feed_u8;
break;
case AV_SAMPLE_FMT_S16:
bs2b->filter = (filter_func) bs2b_cross_feed_s16;
break;
case AV_SAMPLE_FMT_S32:
bs2b->filter = (filter_func) bs2b_cross_feed_s32;
break;
case AV_SAMPLE_FMT_FLT:
bs2b->filter = (filter_func) bs2b_cross_feed_f;
break;
case AV_SAMPLE_FMT_DBL:
bs2b->filter = (filter_func) bs2b_cross_feed_d;
break;
default:
return AVERROR_BUG;
}
if ((srate < BS2B_MINSRATE) || (srate > BS2B_MAXSRATE))
return AVERROR(ENOSYS);
bs2b_set_srate(bs2b->bs2bp, srate);
return 0;
}
static const AVFilterPad bs2b_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
};
static const AVFilterPad bs2b_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_bs2b = {
.name = "bs2b",
.description = NULL_IF_CONFIG_SMALL("Bauer stereo-to-binaural filter."),
.priv_size = sizeof(Bs2bContext),
.priv_class = &bs2b_class,
.init = init,
.uninit = uninit,
FILTER_INPUTS(bs2b_inputs),
FILTER_OUTPUTS(bs2b_outputs),
FILTER_QUERY_FUNC(query_formats),
};