mirror of
https://github.com/FFmpeg/FFmpeg.git
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120 lines
3.7 KiB
C
120 lines
3.7 KiB
C
/*
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* Copyright (c) 2013 Nicolas George
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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/opt.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct {
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const AVClass *class;
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int sample_rate;
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int rescale_pts;
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} ASetRateContext;
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#define CONTEXT ASetRateContext
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \
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{ name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type, \
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{ .deffield = def }, min, max, FLAGS, __VA_ARGS__ }
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#define OPT_INT(name, field, def, min, max, descr, ...) \
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OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)
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static const AVOption asetrate_options[] = {
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OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate"),
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OPT_INT("r", sample_rate, 44100, 1, INT_MAX, "set the sample rate"),
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{NULL},
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};
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AVFILTER_DEFINE_CLASS(asetrate);
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static av_cold int query_formats(AVFilterContext *ctx)
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{
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ASetRateContext *sr = ctx->priv;
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int sample_rates[] = { sr->sample_rate, -1 };
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ff_formats_ref(ff_make_format_list(sample_rates),
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&ctx->outputs[0]->in_samplerates);
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return 0;
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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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ASetRateContext *sr = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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AVRational intb = ctx->inputs[0]->time_base;
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int inrate = inlink->sample_rate;
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if (intb.num == 1 && intb.den == inrate) {
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outlink->time_base.num = 1;
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outlink->time_base.den = outlink->sample_rate;
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} else {
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outlink->time_base = intb;
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sr->rescale_pts = 1;
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if (av_q2d(intb) > 1.0 / FFMAX(inrate, outlink->sample_rate))
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av_log(ctx, AV_LOG_WARNING, "Time base is inaccurate\n");
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}
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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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AVFilterContext *ctx = inlink->dst;
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ASetRateContext *sr = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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frame->sample_rate = outlink->sample_rate;
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if (sr->rescale_pts)
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frame->pts = av_rescale(frame->pts, inlink->sample_rate,
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outlink->sample_rate);
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return ff_filter_frame(outlink, frame);
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}
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static const AVFilterPad asetrate_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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{ NULL }
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};
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static const AVFilterPad asetrate_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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{ NULL }
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};
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AVFilter avfilter_af_asetrate = {
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.name = "asetrate",
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.description = NULL_IF_CONFIG_SMALL("Change the sample rate without "
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"altering the data."),
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.query_formats = query_formats,
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.priv_size = sizeof(ASetRateContext),
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.inputs = asetrate_inputs,
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.outputs = asetrate_outputs,
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.priv_class = &asetrate_class,
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};
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