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https://github.com/FFmpeg/FFmpeg.git
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790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
185 lines
5.4 KiB
C
185 lines
5.4 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
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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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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/mem.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 "internal.h"
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typedef struct AudioMultiplyContext {
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const AVClass *class;
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AVFrame *frames[2];
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int planes;
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int channels;
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int samples_align;
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AVFloatDSPContext *fdsp;
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} AudioMultiplyContext;
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static int activate(AVFilterContext *ctx)
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{
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AudioMultiplyContext *s = ctx->priv;
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int i, ret, status;
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int nb_samples;
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int64_t pts;
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FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
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nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
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ff_inlink_queued_samples(ctx->inputs[1]));
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for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
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if (s->frames[i])
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continue;
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if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
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ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frames[i]);
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if (ret < 0)
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return ret;
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}
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}
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if (s->frames[0] && s->frames[1]) {
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AVFrame *out;
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int plane_samples;
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if (av_sample_fmt_is_planar(ctx->inputs[0]->format))
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plane_samples = FFALIGN(s->frames[0]->nb_samples, s->samples_align);
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else
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plane_samples = FFALIGN(s->frames[0]->nb_samples * s->channels, s->samples_align);
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out = ff_get_audio_buffer(ctx->outputs[0], s->frames[0]->nb_samples);
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if (!out)
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return AVERROR(ENOMEM);
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out->pts = s->frames[0]->pts;
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out->duration = s->frames[0]->duration;
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if (av_get_packed_sample_fmt(ctx->inputs[0]->format) == AV_SAMPLE_FMT_FLT) {
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for (i = 0; i < s->planes; i++) {
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s->fdsp->vector_fmul((float *)out->extended_data[i],
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(const float *)s->frames[0]->extended_data[i],
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(const float *)s->frames[1]->extended_data[i],
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plane_samples);
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}
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} else {
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for (i = 0; i < s->planes; i++) {
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s->fdsp->vector_dmul((double *)out->extended_data[i],
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(const double *)s->frames[0]->extended_data[i],
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(const double *)s->frames[1]->extended_data[i],
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plane_samples);
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}
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}
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av_frame_free(&s->frames[0]);
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av_frame_free(&s->frames[1]);
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ret = ff_filter_frame(ctx->outputs[0], out);
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if (ret < 0)
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return ret;
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}
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if (!nb_samples) {
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for (i = 0; i < 2; i++) {
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
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ff_outlink_set_status(ctx->outputs[0], status, pts);
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return 0;
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}
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}
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}
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if (ff_outlink_frame_wanted(ctx->outputs[0])) {
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for (i = 0; i < 2; i++) {
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if (s->frames[i] || ff_inlink_queued_samples(ctx->inputs[i]) > 0)
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continue;
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ff_inlink_request_frame(ctx->inputs[i]);
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return 0;
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}
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}
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return 0;
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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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AudioMultiplyContext *s = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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s->channels = inlink->ch_layout.nb_channels;
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s->planes = av_sample_fmt_is_planar(inlink->format) ? inlink->ch_layout.nb_channels : 1;
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s->samples_align = 16;
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return 0;
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioMultiplyContext *s = ctx->priv;
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s->fdsp = avpriv_float_dsp_alloc(0);
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if (!s->fdsp)
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return AVERROR(ENOMEM);
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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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AudioMultiplyContext *s = ctx->priv;
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av_freep(&s->fdsp);
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "multiply0",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{
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.name = "multiply1",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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static const AVFilterPad 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_amultiply = {
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.name = "amultiply",
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.description = NULL_IF_CONFIG_SMALL("Multiply two audio streams."),
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.priv_size = sizeof(AudioMultiplyContext),
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.init = init,
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.uninit = uninit,
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.activate = activate,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP),
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};
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