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FFmpeg/libavfilter/af_amerge.c
James Almer 1f96db959c avfilter: convert to new channel layout API
Signed-off-by: James Almer <jamrial@gmail.com>
2022-03-15 09:42:46 -03:00

359 lines
12 KiB
C

/*
* Copyright (c) 2011 Nicolas George <nicolas.george@normalesup.org>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Audio merging filter
*/
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "filters.h"
#include "audio.h"
#include "internal.h"
#define SWR_CH_MAX 64
typedef struct AMergeContext {
const AVClass *class;
int nb_inputs;
int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
int bps;
struct amerge_input {
int nb_ch; /**< number of channels for the input */
} *in;
} AMergeContext;
#define OFFSET(x) offsetof(AMergeContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption amerge_options[] = {
{ "inputs", "specify the number of inputs", OFFSET(nb_inputs),
AV_OPT_TYPE_INT, { .i64 = 2 }, 1, SWR_CH_MAX, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(amerge);
static av_cold void uninit(AVFilterContext *ctx)
{
AMergeContext *s = ctx->priv;
av_freep(&s->in);
}
static int query_formats(AVFilterContext *ctx)
{
static const enum AVSampleFormat packed_sample_fmts[] = {
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE
};
AMergeContext *s = ctx->priv;
AVChannelLayout *inlayout[SWR_CH_MAX] = { NULL }, outlayout = { 0 };
uint64_t outmask = 0;
AVFilterChannelLayouts *layouts;
int i, ret, overlap = 0, nb_ch = 0;
for (i = 0; i < s->nb_inputs; i++) {
if (!ctx->inputs[i]->incfg.channel_layouts ||
!ctx->inputs[i]->incfg.channel_layouts->nb_channel_layouts) {
av_log(ctx, AV_LOG_WARNING,
"No channel layout for input %d\n", i + 1);
return AVERROR(EAGAIN);
}
inlayout[i] = &ctx->inputs[i]->incfg.channel_layouts->channel_layouts[0];
if (ctx->inputs[i]->incfg.channel_layouts->nb_channel_layouts > 1) {
char buf[256];
av_channel_layout_describe(inlayout[i], buf, sizeof(buf));
av_log(ctx, AV_LOG_INFO, "Using \"%s\" for input %d\n", buf, i + 1);
}
s->in[i].nb_ch = FF_LAYOUT2COUNT(inlayout[i]);
if (s->in[i].nb_ch) {
overlap++;
} else {
s->in[i].nb_ch = inlayout[i]->nb_channels;
if (av_channel_layout_subset(inlayout[i], outmask))
overlap++;
outmask |= inlayout[i]->order == AV_CHANNEL_ORDER_NATIVE ?
inlayout[i]->u.mask : 0;
}
nb_ch += s->in[i].nb_ch;
}
if (nb_ch > SWR_CH_MAX) {
av_log(ctx, AV_LOG_ERROR, "Too many channels (max %d)\n", SWR_CH_MAX);
return AVERROR(EINVAL);
}
if (overlap) {
av_log(ctx, AV_LOG_WARNING,
"Input channel layouts overlap: "
"output layout will be determined by the number of distinct input channels\n");
for (i = 0; i < nb_ch; i++)
s->route[i] = i;
av_channel_layout_default(&outlayout, nb_ch);
if (!KNOWN(&outlayout) && nb_ch)
av_channel_layout_from_mask(&outlayout, 0xFFFFFFFFFFFFFFFFULL >> (64 - nb_ch));
} else {
int *route[SWR_CH_MAX];
int c, out_ch_number = 0;
av_channel_layout_from_mask(&outlayout, outmask);
route[0] = s->route;
for (i = 1; i < s->nb_inputs; i++)
route[i] = route[i - 1] + s->in[i - 1].nb_ch;
for (c = 0; c < 64; c++)
for (i = 0; i < s->nb_inputs; i++)
if (av_channel_layout_index_from_channel(inlayout[i], c) >= 0)
*(route[i]++) = out_ch_number++;
}
if ((ret = ff_set_common_formats_from_list(ctx, packed_sample_fmts)) < 0)
return ret;
for (i = 0; i < s->nb_inputs; i++) {
layouts = NULL;
if ((ret = ff_add_channel_layout(&layouts, inlayout[i])) < 0)
return ret;
if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->outcfg.channel_layouts)) < 0)
return ret;
}
layouts = NULL;
if ((ret = ff_add_channel_layout(&layouts, &outlayout)) < 0)
return ret;
if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AMergeContext *s = ctx->priv;
AVBPrint bp;
char buf[128];
int i;
s->bps = av_get_bytes_per_sample(ctx->outputs[0]->format);
outlink->time_base = ctx->inputs[0]->time_base;
av_bprint_init(&bp, 0, AV_BPRINT_SIZE_AUTOMATIC);
for (i = 0; i < s->nb_inputs; i++) {
av_bprintf(&bp, "%sin%d:", i ? " + " : "", i);
av_channel_layout_describe(&ctx->inputs[i]->ch_layout, buf, sizeof(buf));
av_bprintf(&bp, "%s", buf);
}
av_bprintf(&bp, " -> out:");
av_channel_layout_describe(&ctx->outputs[0]->ch_layout, buf, sizeof(buf));
av_bprintf(&bp, "%s", buf);
av_log(ctx, AV_LOG_VERBOSE, "%s\n", bp.str);
return 0;
}
/**
* Copy samples from several input streams to one output stream.
* @param nb_inputs number of inputs
* @param in inputs; used only for the nb_ch field;
* @param route routing values;
* input channel i goes to output channel route[i];
* i < in[0].nb_ch are the channels from the first output;
* i >= in[0].nb_ch are the channels from the second output
* @param ins pointer to the samples of each inputs, in packed format;
* will be left at the end of the copied samples
* @param outs pointer to the samples of the output, in packet format;
* must point to a buffer big enough;
* will be left at the end of the copied samples
* @param ns number of samples to copy
* @param bps bytes per sample
*/
static inline void copy_samples(int nb_inputs, struct amerge_input in[],
int *route, uint8_t *ins[],
uint8_t **outs, int ns, int bps)
{
int *route_cur;
int i, c, nb_ch = 0;
for (i = 0; i < nb_inputs; i++)
nb_ch += in[i].nb_ch;
while (ns--) {
route_cur = route;
for (i = 0; i < nb_inputs; i++) {
for (c = 0; c < in[i].nb_ch; c++) {
memcpy((*outs) + bps * *(route_cur++), ins[i], bps);
ins[i] += bps;
}
}
*outs += nb_ch * bps;
}
}
static void free_frames(int nb_inputs, AVFrame **input_frames)
{
int i;
for (i = 0; i < nb_inputs; i++)
av_frame_free(&input_frames[i]);
}
static int try_push_frame(AVFilterContext *ctx, int nb_samples)
{
AMergeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int i, ret;
AVFrame *outbuf, *inbuf[SWR_CH_MAX] = { NULL };
uint8_t *outs, *ins[SWR_CH_MAX];
for (i = 0; i < ctx->nb_inputs; i++) {
ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &inbuf[i]);
if (ret < 0) {
free_frames(i, inbuf);
return ret;
}
ins[i] = inbuf[i]->data[0];
}
outbuf = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
if (!outbuf) {
free_frames(s->nb_inputs, inbuf);
return AVERROR(ENOMEM);
}
outs = outbuf->data[0];
outbuf->pts = inbuf[0]->pts;
outbuf->nb_samples = nb_samples;
if ((ret = av_channel_layout_copy(&outbuf->ch_layout, &outlink->ch_layout)) < 0)
return ret;
#if FF_API_OLD_CHANNEL_LAYOUT
FF_DISABLE_DEPRECATION_WARNINGS
outbuf->channel_layout = outlink->channel_layout;
outbuf->channels = outlink->ch_layout.nb_channels;
FF_ENABLE_DEPRECATION_WARNINGS
#endif
while (nb_samples) {
/* Unroll the most common sample formats: speed +~350% for the loop,
+~13% overall (including two common decoders) */
switch (s->bps) {
case 1:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, 1);
break;
case 2:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, 2);
break;
case 4:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, 4);
break;
default:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, s->bps);
break;
}
nb_samples = 0;
}
free_frames(s->nb_inputs, inbuf);
return ff_filter_frame(ctx->outputs[0], outbuf);
}
static int activate(AVFilterContext *ctx)
{
int i, status;
int ret, nb_samples;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
nb_samples = ff_inlink_queued_samples(ctx->inputs[0]);
for (i = 1; i < ctx->nb_inputs && nb_samples > 0; i++) {
nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[i]), nb_samples);
}
if (nb_samples) {
ret = try_push_frame(ctx, nb_samples);
if (ret < 0)
return ret;
}
for (i = 0; i < ctx->nb_inputs; i++) {
if (ff_inlink_queued_samples(ctx->inputs[i]))
continue;
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
ff_outlink_set_status(ctx->outputs[0], status, pts);
return 0;
} else if (ff_outlink_frame_wanted(ctx->outputs[0])) {
ff_inlink_request_frame(ctx->inputs[i]);
return 0;
}
}
return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
AMergeContext *s = ctx->priv;
int i, ret;
s->in = av_calloc(s->nb_inputs, sizeof(*s->in));
if (!s->in)
return AVERROR(ENOMEM);
for (i = 0; i < s->nb_inputs; i++) {
char *name = av_asprintf("in%d", i);
AVFilterPad pad = {
.name = name,
.type = AVMEDIA_TYPE_AUDIO,
};
if (!name)
return AVERROR(ENOMEM);
if ((ret = ff_append_inpad_free_name(ctx, &pad)) < 0)
return ret;
}
return 0;
}
static const AVFilterPad amerge_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_amerge = {
.name = "amerge",
.description = NULL_IF_CONFIG_SMALL("Merge two or more audio streams into "
"a single multi-channel stream."),
.priv_size = sizeof(AMergeContext),
.init = init,
.uninit = uninit,
.activate = activate,
.inputs = NULL,
FILTER_OUTPUTS(amerge_outputs),
FILTER_QUERY_FUNC(query_formats),
.priv_class = &amerge_class,
.flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
};