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f8911b987d
* 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>
45 lines
1.4 KiB
C
45 lines
1.4 KiB
C
/*
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* Copyright (c) 2010 S.N. Hemanth Meenakshisundaram <smeenaks@ucsd.edu>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avfilter.h"
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#include "internal.h"
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static int null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
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{
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return 0;
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}
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AVFilter avfilter_asink_anullsink = {
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.name = "anullsink",
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.description = NULL_IF_CONFIG_SMALL("Do absolutely nothing with the input audio."),
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.priv_size = 0,
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.inputs = (const AVFilterPad[]) {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = null_filter_samples,
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},
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{ .name = NULL},
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},
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.outputs = (const AVFilterPad[]) {{ .name = NULL }},
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};
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