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https://github.com/FFmpeg/FFmpeg.git
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30e8ab8bc4
The floating point dsp code does not use MMX registers
since 2718a3be1f
.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
183 lines
5.3 KiB
C
183 lines
5.3 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 "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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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 (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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