mirror of
https://github.com/FFmpeg/FFmpeg.git
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a49188297f
Fixes compilation with msvc. Signed-off-by: James Almer <jamrial@gmail.com>
219 lines
6.0 KiB
C
219 lines
6.0 KiB
C
/*
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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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/**
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* @file
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* Bauer stereo-to-binaural filter
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*/
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#include <bs2b.h>
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "filters.h"
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#include "formats.h"
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typedef void (*filter_func)(t_bs2bdp bs2bdp, uint8_t *sample, int n);
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typedef struct Bs2bContext {
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const AVClass *class;
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int profile;
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int fcut;
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int feed;
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t_bs2bdp bs2bp;
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filter_func filter;
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} Bs2bContext;
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#define OFFSET(x) offsetof(Bs2bContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption bs2b_options[] = {
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{ "profile", "Apply a pre-defined crossfeed level",
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OFFSET(profile), AV_OPT_TYPE_INT, { .i64 = BS2B_DEFAULT_CLEVEL }, 0, INT_MAX, A, .unit = "profile" },
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{ "default", "default profile", 0, AV_OPT_TYPE_CONST, { .i64 = BS2B_DEFAULT_CLEVEL }, 0, 0, A, .unit = "profile" },
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{ "cmoy", "Chu Moy circuit", 0, AV_OPT_TYPE_CONST, { .i64 = BS2B_CMOY_CLEVEL }, 0, 0, A, .unit = "profile" },
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{ "jmeier", "Jan Meier circuit", 0, AV_OPT_TYPE_CONST, { .i64 = BS2B_JMEIER_CLEVEL }, 0, 0, A, .unit = "profile" },
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{ "fcut", "Set cut frequency (in Hz)",
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OFFSET(fcut), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, BS2B_MAXFCUT, A },
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{ "feed", "Set feed level (in Hz)",
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OFFSET(feed), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, BS2B_MAXFEED, A },
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{ NULL },
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};
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AVFILTER_DEFINE_CLASS(bs2b);
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static av_cold int init(AVFilterContext *ctx)
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{
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Bs2bContext *bs2b = ctx->priv;
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if (!(bs2b->bs2bp = bs2b_open()))
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return AVERROR(ENOMEM);
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bs2b_set_level(bs2b->bs2bp, bs2b->profile);
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if (bs2b->fcut)
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bs2b_set_level_fcut(bs2b->bs2bp, bs2b->fcut);
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if (bs2b->feed)
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bs2b_set_level_feed(bs2b->bs2bp, bs2b->feed);
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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Bs2bContext *bs2b = ctx->priv;
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if (bs2b->bs2bp)
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bs2b_close(bs2b->bs2bp);
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}
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static int query_formats(const AVFilterContext *ctx,
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AVFilterFormatsConfig **cfg_in,
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AVFilterFormatsConfig **cfg_out)
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{
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static const AVChannelLayout layouts[] = {
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AV_CHANNEL_LAYOUT_STEREO,
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{ .nb_channels = 0 },
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};
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_U8,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_S32,
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_DBL,
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AV_SAMPLE_FMT_NONE,
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};
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int ret;
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ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, layouts);
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if (ret < 0)
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return ret;
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ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts);
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if (ret < 0)
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return ret;
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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{
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int ret;
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AVFrame *out_frame;
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Bs2bContext *bs2b = inlink->dst->priv;
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AVFilterLink *outlink = inlink->dst->outputs[0];
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if (av_frame_is_writable(frame)) {
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out_frame = frame;
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} else {
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out_frame = ff_get_audio_buffer(outlink, frame->nb_samples);
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if (!out_frame) {
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av_frame_free(&frame);
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return AVERROR(ENOMEM);
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}
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av_frame_copy(out_frame, frame);
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ret = av_frame_copy_props(out_frame, frame);
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if (ret < 0) {
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av_frame_free(&out_frame);
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av_frame_free(&frame);
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return ret;
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}
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}
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bs2b->filter(bs2b->bs2bp, out_frame->extended_data[0], out_frame->nb_samples);
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if (frame != out_frame)
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av_frame_free(&frame);
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return ff_filter_frame(outlink, out_frame);
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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Bs2bContext *bs2b = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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int srate = inlink->sample_rate;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_U8:
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bs2b->filter = (filter_func) bs2b_cross_feed_u8;
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break;
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case AV_SAMPLE_FMT_S16:
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bs2b->filter = (filter_func) bs2b_cross_feed_s16;
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break;
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case AV_SAMPLE_FMT_S32:
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bs2b->filter = (filter_func) bs2b_cross_feed_s32;
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break;
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case AV_SAMPLE_FMT_FLT:
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bs2b->filter = (filter_func) bs2b_cross_feed_f;
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break;
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case AV_SAMPLE_FMT_DBL:
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bs2b->filter = (filter_func) bs2b_cross_feed_d;
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break;
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default:
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return AVERROR_BUG;
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}
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if ((srate < BS2B_MINSRATE) || (srate > BS2B_MAXSRATE))
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return AVERROR(ENOSYS);
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bs2b_set_srate(bs2b->bs2bp, srate);
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return 0;
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}
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static const AVFilterPad bs2b_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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static const AVFilterPad bs2b_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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};
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const AVFilter ff_af_bs2b = {
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.name = "bs2b",
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.description = NULL_IF_CONFIG_SMALL("Bauer stereo-to-binaural filter."),
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.priv_size = sizeof(Bs2bContext),
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.priv_class = &bs2b_class,
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.init = init,
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.uninit = uninit,
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FILTER_INPUTS(bs2b_inputs),
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FILTER_OUTPUTS(bs2b_outputs),
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FILTER_QUERY_FUNC2(query_formats),
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};
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