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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>
87 lines
3.2 KiB
C
87 lines
3.2 KiB
C
/*
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* Copyright (c) Stefano Sabatini | stefasab at gmail.com
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* Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at 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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#ifndef AVFILTER_AUDIO_H
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#define AVFILTER_AUDIO_H
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#include "avfilter.h"
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static const enum AVSampleFormat ff_packed_sample_fmts_array[] = {
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AV_SAMPLE_FMT_U8,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_S32,
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_DBL,
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AV_SAMPLE_FMT_NONE
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};
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static const enum AVSampleFormat ff_planar_sample_fmts_array[] = {
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AV_SAMPLE_FMT_U8P,
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AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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/** default handler for get_audio_buffer() for audio inputs */
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AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms,
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int nb_samples);
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/** get_audio_buffer() handler for filters which simply pass audio along */
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AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
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int nb_samples);
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/**
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* Request an audio samples buffer with a specific set of permissions.
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*
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* @param link the output link to the filter from which the buffer will
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* be requested
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* @param perms the required access permissions
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* @param nb_samples the number of samples per channel
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* @return A reference to the samples. This must be unreferenced with
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* avfilter_unref_buffer when you are finished with it.
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*/
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AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
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int nb_samples);
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/**
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* Send a buffer of audio samples to the next filter.
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*
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* @param link the output link over which the audio samples are being sent
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* @param samplesref a reference to the buffer of audio samples being sent. The
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* receiving filter will free this reference when it no longer
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* needs it or pass it on to the next filter.
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*
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* @return >= 0 on success, a negative AVERROR on error. The receiving filter
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* is responsible for unreferencing samplesref in case of error.
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*/
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int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
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/**
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* Send a buffer of audio samples to the next link, without checking
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* min_samples.
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*/
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int ff_filter_samples_framed(AVFilterLink *link,
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AVFilterBufferRef *samplesref);
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#endif /* AVFILTER_AUDIO_H */
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