FFmpeg/libavfilter/audio.h
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

87 lines
3.2 KiB
C

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