Mercurial > libavcodec.hg
view audioconvert.h @ 10184:9a577b684548 libavcodec
Remove useless code causing crash in Targa decoder
author | kostya |
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date | Thu, 17 Sep 2009 05:01:27 +0000 |
parents | c2dba7ed94dc |
children | 7dd2a45249a9 |
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/* * 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 AVCODEC_AUDIOCONVERT_H #define AVCODEC_AUDIOCONVERT_H /** * @file libavcodec/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); /** * @return NULL on error */ const char *avcodec_get_channel_name(int channel_id); /** * Return description of channel layout */ void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout); /** * Guess the channel layout * @param nb_channels * @param codec_id Codec identifier, or CODEC_ID_NONE if unknown * @param fmt_name Format name, or NULL if unknown * @return Channel layout mask */ int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name); /** * @return the number of channels in the channel layout. */ int avcodec_channel_layout_num_channels(int64_t channel_layout); 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 /* AVCODEC_AUDIOCONVERT_H */