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lavfi: add amerge audio filter.

This commit is contained in:
Nicolas George 2011-11-06 21:28:05 +01:00
parent e90a69e955
commit 4962edf889
6 changed files with 325 additions and 1 deletions
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1
Changelog
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@ -13,6 +13,7 @@ version next:
- asplit audio filter
- tinterlace video filter
- astreamsync audio filter
- amerge audio filter
version 0.9:

33
doc/filters.texi
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@ -156,6 +156,39 @@ aformat=u8\\,s16:mono:packed
aformat=s16:mono\\,stereo:all
@end example
@section amerge
Merge two audio streams into a single multi-channel stream.
This filter does not need any argument.
If the channel layouts of the inputs are disjoint, and therefore compatible,
the channel layout of the output will be set accordingly and the channels
will be reordered as necessary. If the channel layouts of the inputs are not
disjoint, the output will have all the channels of the first input then all
the channels of the second input, in that order, and the channel layout of
the output will be the default value corresponding to the total number of
channels.
For example, if the first input is in 2.1 (FL+FR+LF) and the second input
is FC+BL+BR, then the output will be in 5.1, with the channels in the
following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the
first input, b1 is the first channel of the second input).
On the other hand, if both input are in stereo, the output channels will be
in the default order: a1, a2, b1, b2, and the channel layout will be
arbitrarily set to 4.0, which may or may not be the expected value.
Both inputs must have the same sample rate, format and packing.
If inputs do not have the same duration, the output will stop with the
shortest.
Example: merge two mono files into a stereo stream:
@example
amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
@end example
@section anull
Pass the audio source unchanged to the output.

1
libavfilter/Makefile
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@ -26,6 +26,7 @@ OBJS-$(CONFIG_AVCODEC) += avcodec.o
OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o

288
libavfilter/af_amerge.c Normal file
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@ -0,0 +1,288 @@
/*
* 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 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 "libswresample/swresample.h" // only for SWR_CH_MAX
#include "avfilter.h"
#include "internal.h"
#define QUEUE_SIZE 16
typedef struct {
int nb_in_ch[2]; /**< number of channels for each input */
int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
int bps;
struct amerge_queue {
AVFilterBufferRef *buf[QUEUE_SIZE];
int nb_buf, nb_samples, pos;
} queue[2];
} AMergeContext;
static av_cold void uninit(AVFilterContext *ctx)
{
AMergeContext *am = ctx->priv;
int i, j;
for (i = 0; i < 2; i++)
for (j = 0; j < am->queue[i].nb_buf; j++)
avfilter_unref_buffer(am->queue[i].buf[j]);
}
static int query_formats(AVFilterContext *ctx)
{
AMergeContext *am = ctx->priv;
int64_t inlayout[2], outlayout;
const int packing_fmts[] = { AVFILTER_PACKED, -1 };
AVFilterFormats *formats;
int i;
for (i = 0; i < 2; i++) {
if (!ctx->inputs[i]->in_chlayouts ||
!ctx->inputs[i]->in_chlayouts->format_count) {
av_log(ctx, AV_LOG_ERROR,
"No channel layout for input %d\n", i + 1);
return AVERROR(EINVAL);
}
inlayout[i] = ctx->inputs[i]->in_chlayouts->formats[0];
if (ctx->inputs[i]->in_chlayouts->format_count > 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);
}
am->nb_in_ch[i] = av_get_channel_layout_nb_channels(inlayout[i]);
}
if (am->nb_in_ch[0] + am->nb_in_ch[1] > SWR_CH_MAX) {
av_log(ctx, AV_LOG_ERROR, "Too many channels (max %d)\n", SWR_CH_MAX);
return AVERROR(EINVAL);
}
if (inlayout[0] & inlayout[1]) {
av_log(ctx, AV_LOG_WARNING,
"Inputs overlap: output layout will be meaningless\n");
for (i = 0; i < am->nb_in_ch[0] + am->nb_in_ch[1]; i++)
am->route[i] = i;
outlayout = av_get_default_channel_layout(am->nb_in_ch[0] +
am->nb_in_ch[1]);
if (!outlayout)
outlayout = ((int64_t)1 << (am->nb_in_ch[0] + am->nb_in_ch[1])) - 1;
} else {
int *route[2] = { am->route, am->route + am->nb_in_ch[0] };
int c, out_ch_number = 0;
outlayout = inlayout[0] | inlayout[1];
for (c = 0; c < 64; c++)
for (i = 0; i < 2; i++)
if ((inlayout[i] >> c) & 1)
*(route[i]++) = out_ch_number++;
}
formats = avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO);
avfilter_set_common_sample_formats(ctx, formats);
formats = avfilter_make_format_list(packing_fmts);
avfilter_set_common_packing_formats(ctx, formats);
for (i = 0; i < 2; i++) {
formats = NULL;
avfilter_add_format(&formats, inlayout[i]);
avfilter_formats_ref(formats, &ctx->inputs[i]->out_chlayouts);
}
formats = NULL;
avfilter_add_format(&formats, outlayout);
avfilter_formats_ref(formats, &ctx->outputs[0]->in_chlayouts);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AMergeContext *am = ctx->priv;
int64_t layout;
char name[3][256];
int i;
if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
av_log(ctx, AV_LOG_ERROR,
"Inputs must have the same sample rate "
"(%"PRIi64" vs %"PRIi64")\n",
ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
return AVERROR(EINVAL);
}
am->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;
for (i = 0; i < 3; i++) {
layout = (i < 2 ? ctx->inputs[i] : ctx->outputs[0])->channel_layout;
av_get_channel_layout_string(name[i], sizeof(name[i]), -1, layout);
}
av_log(ctx, AV_LOG_INFO,
"in1:%s + in2:%s -> out:%s\n", name[0], name[1], name[2]);
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AMergeContext *am = ctx->priv;
int i;
for (i = 0; i < 2; i++)
if (!am->queue[i].nb_samples)
avfilter_request_frame(ctx->inputs[i]);
return 0;
}
/**
* Copy samples from two input streams to one output stream.
* @param nb_in_ch number of channels in each input stream
* @param route routing values;
* input channel i goes to output channel route[i];
* i < nb_in_ch[0] are the channels from the first output;
* i >= nb_in_ch[0] 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_in_ch[2], int *route, uint8_t *ins[2],
uint8_t **outs, int ns, int bps)
{
int *route_cur;
int i, c;
while (ns--) {
route_cur = route;
for (i = 0; i < 2; i++) {
for (c = 0; c < nb_in_ch[i]; c++) {
memcpy((*outs) + bps * *(route_cur++), ins[i], bps);
ins[i] += bps;
}
}
*outs += (nb_in_ch[0] + nb_in_ch[1]) * bps;
}
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AMergeContext *am = ctx->priv;
int input_number = inlink == ctx->inputs[1];
struct amerge_queue *inq = &am->queue[input_number];
int nb_samples, ns, i;
AVFilterBufferRef *outbuf, **inbuf[2];
uint8_t *ins[2], *outs;
if (inq->nb_buf == QUEUE_SIZE) {
av_log(ctx, AV_LOG_ERROR, "Packet queue overflow; dropped\n");
avfilter_unref_buffer(insamples);
return;
}
inq->buf[inq->nb_buf++] = avfilter_ref_buffer(insamples, AV_PERM_READ |
AV_PERM_PRESERVE);
inq->nb_samples += insamples->audio->nb_samples;
avfilter_unref_buffer(insamples);
if (!am->queue[!input_number].nb_samples)
return;
nb_samples = FFMIN(am->queue[0].nb_samples,
am->queue[1].nb_samples);
outbuf = avfilter_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE,
nb_samples);
outs = outbuf->data[0];
for (i = 0; i < 2; i++) {
inbuf[i] = am->queue[i].buf;
ins[i] = (*inbuf[i])->data[0] +
am->queue[i].pos * am->nb_in_ch[i] * am->bps;
}
while (nb_samples) {
ns = nb_samples;
for (i = 0; i < 2; i++)
ns = FFMIN(ns, (*inbuf[i])->audio->nb_samples - am->queue[i].pos);
/* Unroll the most common sample formats: speed +~350% for the loop,
+~13% overall (including two common decoders) */
switch (am->bps) {
case 1:
copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, 1);
break;
case 2:
copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, 2);
break;
case 4:
copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, 4);
break;
default:
copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, am->bps);
break;
}
nb_samples -= ns;
for (i = 0; i < 2; i++) {
am->queue[i].nb_samples -= ns;
am->queue[i].pos += ns;
if (am->queue[i].pos == (*inbuf[i])->audio->nb_samples) {
am->queue[i].pos = 0;
avfilter_unref_buffer(*inbuf[i]);
*inbuf[i] = NULL;
inbuf[i]++;
ins[i] = *inbuf[i] ? (*inbuf[i])->data[0] : NULL;
}
}
}
for (i = 0; i < 2; i++) {
int nbufused = inbuf[i] - am->queue[i].buf;
if (nbufused) {
am->queue[i].nb_buf -= nbufused;
memmove(am->queue[i].buf, inbuf[i],
am->queue[i].nb_buf * sizeof(**inbuf));
}
}
avfilter_filter_samples(ctx->outputs[0], outbuf);
}
AVFilter avfilter_af_amerge = {
.name = "amerge",
.description = NULL_IF_CONFIG_SMALL("Merge two audio streams into "
"a single multi-channel stream."),
.priv_size = sizeof(AMergeContext),
.uninit = uninit,
.query_formats = query_formats,
.inputs = (const AVFilterPad[]) {
{ .name = "in1",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = "in2",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = NULL }
},
.outputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame, },
{ .name = NULL }
},
};

1
libavfilter/allfilters.c
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@ -36,6 +36,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (ACONVERT, aconvert, af);
REGISTER_FILTER (AFORMAT, aformat, af);
REGISTER_FILTER (AMERGE, amerge, af);
REGISTER_FILTER (ANULL, anull, af);
REGISTER_FILTER (ARESAMPLE, aresample, af);
REGISTER_FILTER (ASHOWINFO, ashowinfo, af);

2
libavfilter/avfilter.h
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@ -30,7 +30,7 @@
#include "libavcodec/avcodec.h"
#define LIBAVFILTER_VERSION_MAJOR 2
#define LIBAVFILTER_VERSION_MINOR 56
#define LIBAVFILTER_VERSION_MINOR 57
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \