/* * 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 #include "libavutil/channel_layout.h" #include "libavutil/common.h" #include "libavutil/opt.h" #include "audio.h" #include "avfilter.h" #include "formats.h" #include "internal.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, "profile" }, { "default", "default profile", 0, AV_OPT_TYPE_CONST, { .i64 = BS2B_DEFAULT_CLEVEL }, 0, 0, A, "profile" }, { "cmoy", "Chu Moy circuit", 0, AV_OPT_TYPE_CONST, { .i64 = BS2B_CMOY_CLEVEL }, 0, 0, A, "profile" }, { "jmeier", "Jan Meier circuit", 0, AV_OPT_TYPE_CONST, { .i64 = BS2B_JMEIER_CLEVEL }, 0, 0, A, "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) { AVFilterFormats *formats = NULL; 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, AV_CH_LAYOUT_STEREO) != 0) return AVERROR(ENOMEM); ret = ff_set_common_channel_layouts(ctx, layouts); if (ret < 0) return ret; formats = ff_make_format_list(sample_fmts); if (!formats) return AVERROR(ENOMEM); ret = ff_set_common_formats(ctx, formats); if (ret < 0) return ret; formats = ff_all_samplerates(); if (!formats) return AVERROR(ENOMEM); return ff_set_common_samplerates(ctx, formats); } 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, }, { NULL } }; static const AVFilterPad bs2b_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, }, { NULL } }; AVFilter ff_af_bs2b = { .name = "bs2b", .description = NULL_IF_CONFIG_SMALL("Bauer stereo-to-binaural filter."), .query_formats = query_formats, .priv_size = sizeof(Bs2bContext), .priv_class = &bs2b_class, .init = init, .uninit = uninit, .inputs = bs2b_inputs, .outputs = bs2b_outputs, };