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https://github.com/FFmpeg/FFmpeg.git
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19af142d45
internal.h doesn't rely on it; instead include it directly in every user that needs it (a filter needing it is basically equivalent to it using FILTER_QUERY_FUNC, i.e. a majority of filters doesn't need it). Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
174 lines
5.1 KiB
C
174 lines
5.1 KiB
C
/*
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* Copyright (c) 2018 Paul B Mahol
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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 License
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* 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
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* GNU 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 License
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* along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/channel_layout.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 "formats.h"
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#include "internal.h"
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#include "filters.h"
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#include "window_func.h"
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typedef struct HilbertContext {
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const AVClass *class;
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int sample_rate;
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int nb_taps;
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int nb_samples;
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int win_func;
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float *taps;
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int64_t pts;
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} HilbertContext;
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#define OFFSET(x) offsetof(HilbertContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption hilbert_options[] = {
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{ "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
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{ "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
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{ "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=22051}, 11, UINT16_MAX, FLAGS },
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{ "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=22051}, 11, UINT16_MAX, FLAGS },
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{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
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{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
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WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
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WIN_FUNC_OPTION("w", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
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{NULL}
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};
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AVFILTER_DEFINE_CLASS(hilbert);
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static av_cold int init(AVFilterContext *ctx)
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{
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HilbertContext *s = ctx->priv;
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if (!(s->nb_taps & 1)) {
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av_log(s, AV_LOG_ERROR, "Number of taps %d must be odd length.\n", s->nb_taps);
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return AVERROR(EINVAL);
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}
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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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HilbertContext *s = ctx->priv;
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av_freep(&s->taps);
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}
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static av_cold int query_formats(AVFilterContext *ctx)
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{
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HilbertContext *s = ctx->priv;
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static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
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int sample_rates[] = { s->sample_rate, -1 };
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE
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};
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int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
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if (ret < 0)
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return ret;
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ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts);
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if (ret < 0)
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return ret;
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return ff_set_common_samplerates_from_list(ctx, sample_rates);
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}
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static av_cold int config_props(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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HilbertContext *s = ctx->priv;
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float overlap;
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int i;
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s->taps = av_malloc_array(s->nb_taps, sizeof(*s->taps));
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if (!s->taps)
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return AVERROR(ENOMEM);
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generate_window_func(s->taps, s->nb_taps, s->win_func, &overlap);
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for (i = 0; i < s->nb_taps; i++) {
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int k = -(s->nb_taps / 2) + i;
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if (k & 1) {
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float pk = M_PI * k;
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s->taps[i] *= (1.f - cosf(pk)) / pk;
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} else {
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s->taps[i] = 0.f;
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}
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}
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s->pts = 0;
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return 0;
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}
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static int activate(AVFilterContext *ctx)
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{
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AVFilterLink *outlink = ctx->outputs[0];
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HilbertContext *s = ctx->priv;
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AVFrame *frame;
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int nb_samples;
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if (!ff_outlink_frame_wanted(outlink))
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return FFERROR_NOT_READY;
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nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);
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if (nb_samples <= 0) {
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ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
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return 0;
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}
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if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
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return AVERROR(ENOMEM);
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memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float));
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frame->pts = s->pts;
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s->pts += nb_samples;
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return ff_filter_frame(outlink, frame);
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}
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static const AVFilterPad hilbert_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_props,
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},
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};
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const AVFilter ff_asrc_hilbert = {
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.name = "hilbert",
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.description = NULL_IF_CONFIG_SMALL("Generate a Hilbert transform FIR coefficients."),
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.init = init,
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.uninit = uninit,
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.activate = activate,
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.priv_size = sizeof(HilbertContext),
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.inputs = NULL,
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FILTER_OUTPUTS(hilbert_outputs),
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FILTER_QUERY_FUNC(query_formats),
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.priv_class = &hilbert_class,
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};
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