Mercurial > libavcodec.hg
view audioconvert.h @ 7649:8c5d7ede9c96 libavcodec
rename pitch_delay_frac in ff_acelp_interpolate()
author | michael |
---|---|
date | Thu, 21 Aug 2008 22:36:32 +0000 |
parents | 283eeda62184 |
children | c4a4495715dd |
line wrap: on
line source
/* * audio conversion * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> * Copyright (c) 2008 Peter Ross * * 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 FFMPEG_AUDIOCONVERT_H #define FFMPEG_AUDIOCONVERT_H /** * @file audioconvert.h * Audio format conversion routines */ #include "avcodec.h" /** * Generate string corresponding to the sample format with * number sample_fmt, or a header if sample_fmt is negative. * * @param[in] buf the buffer where to write the string * @param[in] buf_size the size of buf * @param[in] sample_fmt the number of the sample format to print the corresponding info string, or * a negative value to print the corresponding header. * Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1. */ void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt); /** * @return NULL on error */ const char *avcodec_get_sample_fmt_name(int sample_fmt); /** * @return SAMPLE_FMT_NONE on error */ enum SampleFormat avcodec_get_sample_fmt(const char* name); struct AVAudioConvert; typedef struct AVAudioConvert AVAudioConvert; /** * Create an audio sample format converter context * @param out_fmt Output sample format * @param out_channels Number of output channels * @param in_fmt Input sample format * @param in_channels Number of input channels * @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore. * @param flags See FF_MM_xx * @return NULL on error */ AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels, enum SampleFormat in_fmt, int in_channels, const float *matrix, int flags); /** * Free audio sample format converter context */ void av_audio_convert_free(AVAudioConvert *ctx); /** * Convert between audio sample formats * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel. * @param[in] out_stride distance between consecutive input samples (measured in bytes) * @param[in] in array of input buffers for each channel * @param[in] in_stride distance between consecutive output samples (measured in bytes) * @param len length of audio frame size (measured in samples) */ int av_audio_convert(AVAudioConvert *ctx, void * const out[6], const int out_stride[6], const void * const in[6], const int in_stride[6], int len); #endif /* FFMPEG_AUDIOCONVERT_H */