/*
* 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 "audio.h"
#include "bufferqueue.h"
#include "internal.h"
#define SWR_CH_MAX 64
typedef struct {
const AVClass *class;
int nb_inputs;
int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
int bps;
struct amerge_input {
struct FFBufQueue queue;
int nb_ch; /**< number of channels for the input */
int nb_samples;
int pos;
} *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 }, 2, SWR_CH_MAX, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(amerge);
static av_cold void uninit(AVFilterContext *ctx)
{
AMergeContext *s = ctx->priv;
int i;
for (i = 0; i < s->nb_inputs; i++) {
if (s->in)
ff_bufqueue_discard_all(&s->in[i].queue);
if (ctx->input_pads)
av_freep(&ctx->input_pads[i].name);
}
av_freep(&s->in);
}
static int query_formats(AVFilterContext *ctx)
{
AMergeContext *s = ctx->priv;
int64_t inlayout[SWR_CH_MAX], outlayout = 0;
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
int i, overlap = 0, nb_ch = 0;
for (i = 0; i < s->nb_inputs; i++) {
if (!ctx->inputs[i]->in_channel_layouts ||
!ctx->inputs[i]->in_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]->in_channel_layouts->channel_layouts[0];
if (ctx->inputs[i]->in_channel_layouts->nb_channel_layouts > 1) {
char buf[256];
av_get_channel_layout_string(buf, sizeof(buf), 0, inlayout[i]);
av_log(ctx, AV_LOG_INFO, "Using \"%s\" for input %d\n", buf, i + 1);
}
s->in[i].nb_ch = av_get_channel_layout_nb_channels(inlayout[i]);
if (outlayout & inlayout[i])
overlap++;
outlayout |= inlayout[i];
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;
outlayout = av_get_default_channel_layout(nb_ch);
if (!outlayout && nb_ch)
outlayout = 0xFFFFFFFFFFFFFFFFULL >> (64 - nb_ch);
} else {
int *route[SWR_CH_MAX];
int c, out_ch_number = 0;
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 ((inlayout[i] >> c) & 1)
*(route[i]++) = out_ch_number++;
}
formats = ff_make_format_list(ff_packed_sample_fmts_array);
ff_set_common_formats(ctx, formats);
for (i = 0; i < s->nb_inputs; i++) {
layouts = NULL;
ff_add_channel_layout(&layouts, inlayout[i]);
ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts);
}
layouts = NULL;
ff_add_channel_layout(&layouts, outlayout);
ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);
ff_set_common_samplerates(ctx, ff_all_samplerates());
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AMergeContext *s = ctx->priv;
AVBPrint bp;
int i;
for (i = 1; i < s->nb_inputs; i++) {
if (ctx->inputs[i]->sample_rate != ctx->inputs[0]->sample_rate) {
av_log(ctx, AV_LOG_ERROR,
"Inputs must have the same sample rate "
"%d for in%d vs %d\n",
ctx->inputs[i]->sample_rate, i, ctx->inputs[0]->sample_rate);
return AVERROR(EINVAL);
}
}
s->bps = av_get_bytes_per_sample(ctx->outputs[0]->format);
outlink->sample_rate = ctx->inputs[0]->sample_rate;
outlink->time_base = ctx->inputs[0]->time_base;
av_bprint_init(&bp, 0, 1);
for (i = 0; i < s->nb_inputs; i++) {
av_bprintf(&bp, "%sin%d:", i ? " + " : "", i);
av_bprint_channel_layout(&bp, -1, ctx->inputs[i]->channel_layout);
}
av_bprintf(&bp, " -> out:");
av_bprint_channel_layout(&bp, -1, ctx->outputs[0]->channel_layout);
av_log(ctx, AV_LOG_VERBOSE, "%s\n", bp.str);
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AMergeContext *s = ctx->priv;
int i, ret;
for (i = 0; i < s->nb_inputs; i++)
if (!s->in[i].nb_samples)
if ((ret = ff_request_frame(ctx->inputs[i])) < 0)
return ret;
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 int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
{
AVFilterContext *ctx = inlink->dst;
AMergeContext *s = ctx->priv;
AVFilterLink *const outlink = ctx->outputs[0];
int input_number;
int nb_samples, ns, i;
AVFrame *outbuf, *inbuf[SWR_CH_MAX];
uint8_t *ins[SWR_CH_MAX], *outs;
for (input_number = 0; input_number < s->nb_inputs; input_number++)
if (inlink == ctx->inputs[input_number])
break;
av_assert1(input_number < s->nb_inputs);
if (ff_bufqueue_is_full(&s->in[input_number].queue)) {
av_frame_free(&insamples);
return AVERROR(ENOMEM);
}
ff_bufqueue_add(ctx, &s->in[input_number].queue, av_frame_clone(insamples));
s->in[input_number].nb_samples += insamples->nb_samples;
av_frame_free(&insamples);
nb_samples = s->in[0].nb_samples;
for (i = 1; i < s->nb_inputs; i++)
nb_samples = FFMIN(nb_samples, s->in[i].nb_samples);
if (!nb_samples)
return 0;
outbuf = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
if (!outbuf)
return AVERROR(ENOMEM);
outs = outbuf->data[0];
for (i = 0; i < s->nb_inputs; i++) {
inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0);
ins[i] = inbuf[i]->data[0] +
s->in[i].pos * s->in[i].nb_ch * s->bps;
}
av_frame_copy_props(outbuf, inbuf[0]);
outbuf->pts = inbuf[0]->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
inbuf[0]->pts +
av_rescale_q(s->in[0].pos,
av_make_q(1, ctx->inputs[0]->sample_rate),
ctx->outputs[0]->time_base);
outbuf->nb_samples = nb_samples;
outbuf->channel_layout = outlink->channel_layout;
av_frame_set_channels(outbuf, outlink->channels);
while (nb_samples) {
ns = nb_samples;
for (i = 0; i < s->nb_inputs; i++)
ns = FFMIN(ns, inbuf[i]->nb_samples - s->in[i].pos);
/* 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, ns, 1);
break;
case 2:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 2);
break;
case 4:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 4);
break;
default:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, s->bps);
break;
}
nb_samples -= ns;
for (i = 0; i < s->nb_inputs; i++) {
s->in[i].nb_samples -= ns;
s->in[i].pos += ns;
if (s->in[i].pos == inbuf[i]->nb_samples) {
s->in[i].pos = 0;
av_frame_free(&inbuf[i]);
ff_bufqueue_get(&s->in[i].queue);
inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0);
ins[i] = inbuf[i] ? inbuf[i]->data[0] : NULL;
}
}
}
return ff_filter_frame(ctx->outputs[0], outbuf);
}
static av_cold int init(AVFilterContext *ctx)
{
AMergeContext *s = ctx->priv;
int i;
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,
.filter_frame = filter_frame,
};
if (!name)
return AVERROR(ENOMEM);
ff_insert_inpad(ctx, i, &pad);
}
return 0;
}
static const AVFilterPad amerge_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame,
},
{ NULL }
};
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,
.query_formats = query_formats,
.inputs = NULL,
.outputs = amerge_outputs,
.priv_class = &amerge_class,
.flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
};