mirror of
https://github.com/xenia-project/FFmpeg.git
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6aaac24d72
Many of the functions from avfilter/formats can return errors, usually AVERROR(ENOMEM). This propagates the return values. All of these were found by using av_warn_unused_result, demonstrating its utility. Tested with FATE. I am least sure of the changes to avfilter/filtergraph, since I don't know what/how reduce_format is intended to behave and how it should react to errors. Fixes: CID 1325680, 1325679, 1325678. Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Previous version Reviewed-by: Nicolas George <george@nsup.org> Previous version Reviewed-by: Clément Bœsch <u@pkh.me> Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
357 lines
12 KiB
C
357 lines
12 KiB
C
/*
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* Copyright (c) 2011 Nicolas George <nicolas.george@normalesup.org>
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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
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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
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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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/**
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* @file
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* Audio merging filter
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*/
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#include "libavutil/avstring.h"
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#include "libavutil/bprint.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "bufferqueue.h"
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#include "internal.h"
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#define SWR_CH_MAX 64
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typedef struct {
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const AVClass *class;
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int nb_inputs;
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int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
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int bps;
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struct amerge_input {
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struct FFBufQueue queue;
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int nb_ch; /**< number of channels for the input */
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int nb_samples;
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int pos;
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} *in;
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} AMergeContext;
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#define OFFSET(x) offsetof(AMergeContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption amerge_options[] = {
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{ "inputs", "specify the number of inputs", OFFSET(nb_inputs),
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AV_OPT_TYPE_INT, { .i64 = 2 }, 2, SWR_CH_MAX, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(amerge);
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AMergeContext *s = ctx->priv;
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int i;
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for (i = 0; i < s->nb_inputs; i++) {
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if (s->in)
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ff_bufqueue_discard_all(&s->in[i].queue);
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if (ctx->input_pads)
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av_freep(&ctx->input_pads[i].name);
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}
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av_freep(&s->in);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AMergeContext *s = ctx->priv;
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int64_t inlayout[SWR_CH_MAX], outlayout = 0;
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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int i, ret, overlap = 0, nb_ch = 0;
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for (i = 0; i < s->nb_inputs; i++) {
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if (!ctx->inputs[i]->in_channel_layouts ||
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!ctx->inputs[i]->in_channel_layouts->nb_channel_layouts) {
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av_log(ctx, AV_LOG_WARNING,
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"No channel layout for input %d\n", i + 1);
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return AVERROR(EAGAIN);
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}
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inlayout[i] = ctx->inputs[i]->in_channel_layouts->channel_layouts[0];
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if (ctx->inputs[i]->in_channel_layouts->nb_channel_layouts > 1) {
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char buf[256];
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av_get_channel_layout_string(buf, sizeof(buf), 0, inlayout[i]);
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av_log(ctx, AV_LOG_INFO, "Using \"%s\" for input %d\n", buf, i + 1);
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}
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s->in[i].nb_ch = av_get_channel_layout_nb_channels(inlayout[i]);
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if (outlayout & inlayout[i])
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overlap++;
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outlayout |= inlayout[i];
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nb_ch += s->in[i].nb_ch;
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}
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if (nb_ch > SWR_CH_MAX) {
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av_log(ctx, AV_LOG_ERROR, "Too many channels (max %d)\n", SWR_CH_MAX);
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return AVERROR(EINVAL);
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}
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if (overlap) {
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av_log(ctx, AV_LOG_WARNING,
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"Input channel layouts overlap: "
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"output layout will be determined by the number of distinct input channels\n");
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for (i = 0; i < nb_ch; i++)
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s->route[i] = i;
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outlayout = av_get_default_channel_layout(nb_ch);
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if (!outlayout && nb_ch)
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outlayout = 0xFFFFFFFFFFFFFFFFULL >> (64 - nb_ch);
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} else {
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int *route[SWR_CH_MAX];
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int c, out_ch_number = 0;
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route[0] = s->route;
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for (i = 1; i < s->nb_inputs; i++)
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route[i] = route[i - 1] + s->in[i - 1].nb_ch;
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for (c = 0; c < 64; c++)
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for (i = 0; i < s->nb_inputs; i++)
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if ((inlayout[i] >> c) & 1)
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*(route[i]++) = out_ch_number++;
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}
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formats = ff_make_format_list(ff_packed_sample_fmts_array);
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if ((ret = ff_set_common_formats(ctx, formats)) < 0)
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return ret;
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for (i = 0; i < s->nb_inputs; i++) {
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layouts = NULL;
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if ((ret = ff_add_channel_layout(&layouts, inlayout[i])) < 0)
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return ret;
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if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
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return ret;
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}
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layouts = NULL;
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if ((ret = ff_add_channel_layout(&layouts, outlayout)) < 0)
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return ret;
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if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
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return ret;
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return ff_set_common_samplerates(ctx, ff_all_samplerates());
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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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AMergeContext *s = ctx->priv;
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AVBPrint bp;
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int i;
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for (i = 1; i < s->nb_inputs; i++) {
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if (ctx->inputs[i]->sample_rate != ctx->inputs[0]->sample_rate) {
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av_log(ctx, AV_LOG_ERROR,
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"Inputs must have the same sample rate "
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"%d for in%d vs %d\n",
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ctx->inputs[i]->sample_rate, i, ctx->inputs[0]->sample_rate);
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return AVERROR(EINVAL);
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}
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}
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s->bps = av_get_bytes_per_sample(ctx->outputs[0]->format);
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outlink->sample_rate = ctx->inputs[0]->sample_rate;
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outlink->time_base = ctx->inputs[0]->time_base;
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av_bprint_init(&bp, 0, 1);
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for (i = 0; i < s->nb_inputs; i++) {
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av_bprintf(&bp, "%sin%d:", i ? " + " : "", i);
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av_bprint_channel_layout(&bp, -1, ctx->inputs[i]->channel_layout);
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}
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av_bprintf(&bp, " -> out:");
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av_bprint_channel_layout(&bp, -1, ctx->outputs[0]->channel_layout);
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av_log(ctx, AV_LOG_VERBOSE, "%s\n", bp.str);
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return 0;
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}
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static int request_frame(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AMergeContext *s = ctx->priv;
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int i, ret;
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for (i = 0; i < s->nb_inputs; i++)
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if (!s->in[i].nb_samples)
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if ((ret = ff_request_frame(ctx->inputs[i])) < 0)
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return ret;
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return 0;
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}
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/**
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* Copy samples from several input streams to one output stream.
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* @param nb_inputs number of inputs
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* @param in inputs; used only for the nb_ch field;
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* @param route routing values;
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* input channel i goes to output channel route[i];
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* i < in[0].nb_ch are the channels from the first output;
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* i >= in[0].nb_ch are the channels from the second output
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* @param ins pointer to the samples of each inputs, in packed format;
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* will be left at the end of the copied samples
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* @param outs pointer to the samples of the output, in packet format;
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* must point to a buffer big enough;
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* will be left at the end of the copied samples
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* @param ns number of samples to copy
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* @param bps bytes per sample
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*/
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static inline void copy_samples(int nb_inputs, struct amerge_input in[],
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int *route, uint8_t *ins[],
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uint8_t **outs, int ns, int bps)
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{
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int *route_cur;
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int i, c, nb_ch = 0;
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for (i = 0; i < nb_inputs; i++)
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nb_ch += in[i].nb_ch;
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while (ns--) {
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route_cur = route;
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for (i = 0; i < nb_inputs; i++) {
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for (c = 0; c < in[i].nb_ch; c++) {
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memcpy((*outs) + bps * *(route_cur++), ins[i], bps);
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ins[i] += bps;
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}
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}
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*outs += nb_ch * bps;
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}
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
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{
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AVFilterContext *ctx = inlink->dst;
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AMergeContext *s = ctx->priv;
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AVFilterLink *const outlink = ctx->outputs[0];
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int input_number;
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int nb_samples, ns, i;
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AVFrame *outbuf, *inbuf[SWR_CH_MAX];
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uint8_t *ins[SWR_CH_MAX], *outs;
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for (input_number = 0; input_number < s->nb_inputs; input_number++)
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if (inlink == ctx->inputs[input_number])
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break;
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av_assert1(input_number < s->nb_inputs);
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if (ff_bufqueue_is_full(&s->in[input_number].queue)) {
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av_frame_free(&insamples);
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return AVERROR(ENOMEM);
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}
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ff_bufqueue_add(ctx, &s->in[input_number].queue, av_frame_clone(insamples));
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s->in[input_number].nb_samples += insamples->nb_samples;
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av_frame_free(&insamples);
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nb_samples = s->in[0].nb_samples;
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for (i = 1; i < s->nb_inputs; i++)
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nb_samples = FFMIN(nb_samples, s->in[i].nb_samples);
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if (!nb_samples)
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return 0;
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outbuf = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
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if (!outbuf)
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return AVERROR(ENOMEM);
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outs = outbuf->data[0];
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for (i = 0; i < s->nb_inputs; i++) {
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inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0);
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ins[i] = inbuf[i]->data[0] +
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s->in[i].pos * s->in[i].nb_ch * s->bps;
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}
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av_frame_copy_props(outbuf, inbuf[0]);
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outbuf->pts = inbuf[0]->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
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inbuf[0]->pts +
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av_rescale_q(s->in[0].pos,
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av_make_q(1, ctx->inputs[0]->sample_rate),
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ctx->outputs[0]->time_base);
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outbuf->nb_samples = nb_samples;
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outbuf->channel_layout = outlink->channel_layout;
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av_frame_set_channels(outbuf, outlink->channels);
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while (nb_samples) {
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ns = nb_samples;
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for (i = 0; i < s->nb_inputs; i++)
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ns = FFMIN(ns, inbuf[i]->nb_samples - s->in[i].pos);
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/* Unroll the most common sample formats: speed +~350% for the loop,
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+~13% overall (including two common decoders) */
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switch (s->bps) {
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case 1:
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copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 1);
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break;
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case 2:
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copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 2);
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break;
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case 4:
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copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 4);
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break;
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default:
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copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, s->bps);
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break;
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}
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nb_samples -= ns;
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for (i = 0; i < s->nb_inputs; i++) {
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s->in[i].nb_samples -= ns;
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s->in[i].pos += ns;
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if (s->in[i].pos == inbuf[i]->nb_samples) {
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s->in[i].pos = 0;
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av_frame_free(&inbuf[i]);
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ff_bufqueue_get(&s->in[i].queue);
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inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0);
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ins[i] = inbuf[i] ? inbuf[i]->data[0] : NULL;
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}
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}
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}
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return ff_filter_frame(ctx->outputs[0], outbuf);
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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AMergeContext *s = ctx->priv;
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int i;
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s->in = av_calloc(s->nb_inputs, sizeof(*s->in));
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if (!s->in)
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return AVERROR(ENOMEM);
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for (i = 0; i < s->nb_inputs; i++) {
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char *name = av_asprintf("in%d", i);
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AVFilterPad pad = {
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.name = name,
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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};
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if (!name)
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return AVERROR(ENOMEM);
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ff_insert_inpad(ctx, i, &pad);
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}
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return 0;
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}
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static const AVFilterPad amerge_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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.request_frame = request_frame,
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},
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{ NULL }
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};
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AVFilter ff_af_amerge = {
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.name = "amerge",
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.description = NULL_IF_CONFIG_SMALL("Merge two or more audio streams into "
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"a single multi-channel stream."),
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.priv_size = sizeof(AMergeContext),
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.init = init,
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.uninit = uninit,
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.query_formats = query_formats,
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.inputs = NULL,
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.outputs = amerge_outputs,
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.priv_class = &amerge_class,
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.flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
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};
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