lavfi: add channelsplit audio filter.

pull/59/head
Anton Khirnov 13 years ago
parent 4795362660
commit d625136877
  1. 1
      Changelog
  2. 25
      doc/filters.texi
  3. 1
      libavfilter/Makefile
  4. 146
      libavfilter/af_channelsplit.c
  5. 1
      libavfilter/allfilters.c
  6. 12
      libavfilter/formats.c
  7. 5
      libavfilter/formats.h
  8. 2
      libavfilter/version.h

@ -24,6 +24,7 @@ version <next>:
- avprobe output is now standard INI or JSON. The old format can still
be used with -of old.
- Indeo Audio decoder
- channelsplit audio filter
version 0.8:

@ -207,6 +207,31 @@ Maximum compensation in samples per second.
@end table
@section channelsplit
Split each channel in input audio stream into a separate output stream.
This filter accepts the following named parameters:
@table @option
@item channel_layout
Channel layout of the input stream. Default is "stereo".
@end table
For example, assuming a stereo input MP3 file
@example
avconv -i in.mp3 -filter_complex channelsplit out.mkv
@end example
will create an output Matroska file with two audio streams, one containing only
the left channel and the other the right channel.
To split a 5.1 WAV file into per-channel files
@example
avconv -i in.wav -filter_complex
'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
-map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
side_right.wav
@end example
@section resample
Convert the audio sample format, sample rate and channel layout. This filter is
not meant to be used directly, it is inserted automatically by libavfilter

@ -29,6 +29,7 @@ OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASYNCTS_FILTER) += af_asyncts.o
OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o

@ -0,0 +1,146 @@
/*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Channel split filter
*
* Split an audio stream into per-channel streams.
*/
#include "libavutil/audioconvert.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
typedef struct ChannelSplitContext {
const AVClass *class;
uint64_t channel_layout;
char *channel_layout_str;
} ChannelSplitContext;
#define OFFSET(x) offsetof(ChannelSplitContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
static const AVOption options[] = {
{ "channel_layout", "Input channel layout.", OFFSET(channel_layout_str), AV_OPT_TYPE_STRING, { .str = "stereo" }, .flags = A },
{ NULL },
};
static const AVClass channelsplit_class = {
.class_name = "channelsplit filter",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static int init(AVFilterContext *ctx, const char *arg, void *opaque)
{
ChannelSplitContext *s = ctx->priv;
int nb_channels;
int ret = 0, i;
s->class = &channelsplit_class;
av_opt_set_defaults(s);
if ((ret = av_set_options_string(s, arg, "=", ":")) < 0) {
av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", arg);
return ret;
}
if (!(s->channel_layout = av_get_channel_layout(s->channel_layout_str))) {
av_log(ctx, AV_LOG_ERROR, "Error parsing channel layout '%s'.\n",
s->channel_layout_str);
ret = AVERROR(EINVAL);
goto fail;
}
nb_channels = av_get_channel_layout_nb_channels(s->channel_layout);
for (i = 0; i < nb_channels; i++) {
uint64_t channel = av_channel_layout_extract_channel(s->channel_layout, i);
AVFilterPad pad = { 0 };
pad.type = AVMEDIA_TYPE_AUDIO;
pad.name = av_get_channel_name(channel);
ff_insert_outpad(ctx, i, &pad);
}
fail:
av_opt_free(s);
return ret;
}
static int query_formats(AVFilterContext *ctx)
{
ChannelSplitContext *s = ctx->priv;
AVFilterChannelLayouts *in_layouts = NULL;
int i;
ff_set_common_formats (ctx, ff_planar_sample_fmts());
ff_set_common_samplerates(ctx, ff_all_samplerates());
ff_add_channel_layout(&in_layouts, s->channel_layout);
ff_channel_layouts_ref(in_layouts, &ctx->inputs[0]->out_channel_layouts);
for (i = 0; i < ctx->output_count; i++) {
AVFilterChannelLayouts *out_layouts = NULL;
uint64_t channel = av_channel_layout_extract_channel(s->channel_layout, i);
ff_add_channel_layout(&out_layouts, channel);
ff_channel_layouts_ref(out_layouts, &ctx->outputs[i]->in_channel_layouts);
}
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
int i;
for (i = 0; i < ctx->output_count; i++) {
AVFilterBufferRef *buf_out = avfilter_ref_buffer(buf, ~AV_PERM_WRITE);
if (!buf_out)
return;
buf_out->data[0] = buf_out->extended_data[0] = buf_out->extended_data[i];
buf_out->audio->channel_layout =
av_channel_layout_extract_channel(buf->audio->channel_layout, i);
ff_filter_samples(ctx->outputs[i], buf_out);
}
avfilter_unref_buffer(buf);
}
AVFilter avfilter_af_channelsplit = {
.name = "channelsplit",
.description = NULL_IF_CONFIG_SMALL("Split audio into per-channel streams"),
.priv_size = sizeof(ChannelSplitContext),
.init = init,
.query_formats = query_formats,
.inputs = (const AVFilterPad[]){{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples, },
{ NULL }},
.outputs = (const AVFilterPad[]){{ NULL }},
};

@ -39,6 +39,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (ANULL, anull, af);
REGISTER_FILTER (ASPLIT, asplit, af);
REGISTER_FILTER (ASYNCTS, asyncts, af);
REGISTER_FILTER (CHANNELSPLIT,channelsplit,af);
REGISTER_FILTER (RESAMPLE, resample, af);
REGISTER_FILTER (ANULLSRC, anullsrc, asrc);

@ -220,6 +220,18 @@ AVFilterFormats *ff_all_formats(enum AVMediaType type)
return ret;
}
AVFilterFormats *ff_planar_sample_fmts(void)
{
AVFilterFormats *ret = NULL;
int fmt;
for (fmt = 0; fmt < AV_SAMPLE_FMT_NB; fmt++)
if (av_sample_fmt_is_planar(fmt))
ff_add_format(&ret, fmt);
return ret;
}
AVFilterFormats *ff_all_samplerates(void)
{
AVFilterFormats *ret = av_mallocz(sizeof(*ret));

@ -159,6 +159,11 @@ int ff_add_format(AVFilterFormats **avff, int fmt);
*/
AVFilterFormats *ff_all_formats(enum AVMediaType type);
/**
* Construct a formats list containing all planar sample formats.
*/
AVFilterFormats *ff_planar_sample_fmts(void);
/**
* Return a format list which contains the intersection of the formats of
* a and b. Also, all the references of a, all the references of b, and

@ -29,7 +29,7 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 2
#define LIBAVFILTER_VERSION_MINOR 20
#define LIBAVFILTER_VERSION_MINOR 21
#define LIBAVFILTER_VERSION_MICRO 0
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \

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