/* * Copyright (c) 2018 Paul B Mahol * * 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/channel_layout.h" #include "libavutil/mem.h" #include "libavutil/opt.h" #include "audio.h" #include "avfilter.h" #include "formats.h" #include "filters.h" #include "window_func.h" typedef struct HilbertContext { const AVClass *class; int sample_rate; int nb_taps; int nb_samples; int win_func; float *taps; int64_t pts; } HilbertContext; #define OFFSET(x) offsetof(HilbertContext, x) #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM static const AVOption hilbert_options[] = { { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS }, { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS }, { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=22051}, 11, UINT16_MAX, FLAGS }, { "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=22051}, 11, UINT16_MAX, FLAGS }, { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS }, { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS }, WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN), WIN_FUNC_OPTION("w", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN), {NULL} }; AVFILTER_DEFINE_CLASS(hilbert); static av_cold int init(AVFilterContext *ctx) { HilbertContext *s = ctx->priv; if (!(s->nb_taps & 1)) { av_log(s, AV_LOG_ERROR, "Number of taps %d must be odd length.\n", s->nb_taps); return AVERROR(EINVAL); } return 0; } static av_cold void uninit(AVFilterContext *ctx) { HilbertContext *s = ctx->priv; av_freep(&s->taps); } static av_cold int query_formats(const AVFilterContext *ctx, AVFilterFormatsConfig **cfg_in, AVFilterFormatsConfig **cfg_out) { HilbertContext *s = ctx->priv; static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } }; int sample_rates[] = { s->sample_rate, -1 }; static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_NONE }; int ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts); if (ret < 0) return ret; ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, chlayouts); if (ret < 0) return ret; return ff_set_common_samplerates_from_list2(ctx, cfg_in, cfg_out, sample_rates); } static av_cold int config_props(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; HilbertContext *s = ctx->priv; float overlap; int i; s->taps = av_malloc_array(s->nb_taps, sizeof(*s->taps)); if (!s->taps) return AVERROR(ENOMEM); generate_window_func(s->taps, s->nb_taps, s->win_func, &overlap); for (i = 0; i < s->nb_taps; i++) { int k = -(s->nb_taps / 2) + i; if (k & 1) { float pk = M_PI * k; s->taps[i] *= (1.f - cosf(pk)) / pk; } else { s->taps[i] = 0.f; } } s->pts = 0; return 0; } static int activate(AVFilterContext *ctx) { AVFilterLink *outlink = ctx->outputs[0]; HilbertContext *s = ctx->priv; AVFrame *frame; int nb_samples; if (!ff_outlink_frame_wanted(outlink)) return FFERROR_NOT_READY; nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts); if (nb_samples <= 0) { ff_outlink_set_status(outlink, AVERROR_EOF, s->pts); return 0; } if (!(frame = ff_get_audio_buffer(outlink, nb_samples))) return AVERROR(ENOMEM); memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float)); frame->pts = s->pts; s->pts += nb_samples; return ff_filter_frame(outlink, frame); } static const AVFilterPad hilbert_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_props, }, }; const AVFilter ff_asrc_hilbert = { .name = "hilbert", .description = NULL_IF_CONFIG_SMALL("Generate a Hilbert transform FIR coefficients."), .init = init, .uninit = uninit, .activate = activate, .priv_size = sizeof(HilbertContext), .inputs = NULL, FILTER_OUTPUTS(hilbert_outputs), FILTER_QUERY_FUNC2(query_formats), .priv_class = &hilbert_class, };