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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
FFmpeg/libavfilter/af_channelsplit.c
Michael Niedermayer f8911b987d Merge remote-tracking branch 'qatar/master'
* qatar/master:
  mss3: use standard zigzag table
  mss3: split DSP functions that are used in MTS2(MSS4) into separate file
  motion-test: do not use getopt()
  tcp: add initial timeout limit for incoming connections
  configure: Change the rdtsc check to a linker check
  avconv: propagate fatal errors from lavfi.
  lavfi: add error handling to filter_samples().
  fate-run: make avconv() properly deal with multiple inputs.
  asplit: don't leak the input buffer.
  af_resample: fix request_frame() behavior.
  af_asyncts: fix request_frame() behavior.
  libx264: support aspect ratio switching
  matroskadec: honor error_recognition when encountering unknown elements.
  lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
  lavr: resampling: add filter type and Kaiser window beta to AVOptions
  lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
  lavr: mix: validate internal sample format in ff_audio_mix_init()

Conflicts:
	ffmpeg.c
	ffplay.c
	libavcodec/libx264.c
	libavfilter/audio.c
	libavfilter/split.c
	libavformat/tcp.c
	tests/fate-run.sh

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-07-09 22:40:12 +02:00

147 lines
4.5 KiB
C

/*
* 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 channelsplit_options[] = {
{ "channel_layout", "Input channel layout.", OFFSET(channel_layout_str), AV_OPT_TYPE_STRING, { .str = "stereo" }, .flags = A },
{ NULL },
};
AVFILTER_DEFINE_CLASS(channelsplit);
static int init(AVFilterContext *ctx, const char *arg)
{
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->nb_outputs; 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 int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
int i, ret = 0;
for (i = 0; i < ctx->nb_outputs; i++) {
AVFilterBufferRef *buf_out = avfilter_ref_buffer(buf, ~AV_PERM_WRITE);
if (!buf_out) {
ret = AVERROR(ENOMEM);
break;
}
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);
ret = ff_filter_samples(ctx->outputs[i], buf_out);
if (ret < 0)
break;
}
avfilter_unref_buffer(buf);
return ret;
}
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 }},
};