libavcodec/audioconvert.h

   1 /*
   2  * audio conversion
   3  * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
   4  * Copyright (c) 2008 Peter Ross
   5  *
   6  * This file is part of FFmpeg.
   7  *
   8  * FFmpeg is free software; you can redistribute it and/or
   9  * modify it under the terms of the GNU Lesser General Public
  10  * License as published by the Free Software Foundation; either
  11  * version 2.1 of the License, or (at your option) any later version.
  12  *
  13  * FFmpeg is distributed in the hope that it will be useful,
  14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16  * Lesser General Public License for more details.
  17  *
  18  * You should have received a copy of the GNU Lesser General Public
  19  * License along with FFmpeg; if not, write to the Free Software
  20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21  */
  22
  23 #ifndef AVCODEC_AUDIOCONVERT_H
  24 #define AVCODEC_AUDIOCONVERT_H
  25
  26 /**
  27  * @file libavcodec/audioconvert.h
  28  * Audio format conversion routines
  29  */
  30
  31
  32 #include "avcodec.h"
  33
  34
  35 /**
  36  * Generate string corresponding to the sample format with
  37  * number sample_fmt, or a header if sample_fmt is negative.
  38  *
  39  * @param[in] buf the buffer where to write the string
  40  * @param[in] buf_size the size of buf
  41  * @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
  42  * a negative value to print the corresponding header.
  43  * Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
  44  */
  45 void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
  46
  47 /**
  48  * @return NULL on error
  49  */
  50 const char *avcodec_get_sample_fmt_name(int sample_fmt);
  51
  52 /**
  53  * @return SAMPLE_FMT_NONE on error
  54  */
  55 enum SampleFormat avcodec_get_sample_fmt(const char* name);
  56
  57 /**
  58  * @return NULL on error
  59  */
  60 const char *avcodec_get_channel_name(int channel_id);
  61
  62 /**
  63  * Return description of channel layout
  64  */
  65 void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout);
  66
  67 /**
  68  * Guess the channel layout
  69  * @param nb_channels
  70  * @param codec_id Codec identifier, or CODEC_ID_NONE if unknown
  71  * @param fmt_name Format name, or NULL if unknown
  72  * @return Channel layout mask
  73  */
  74 int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name);
  75
  76 /**
  77  * @return the number of channels in the channel layout.
  78  */
  79 int avcodec_channel_layout_num_channels(int64_t channel_layout);
  80
  81 struct AVAudioConvert;
  82 typedef struct AVAudioConvert AVAudioConvert;
  83
  84 /**
  85  * Create an audio sample format converter context
  86  * @param out_fmt Output sample format
  87  * @param out_channels Number of output channels
  88  * @param in_fmt Input sample format
  89  * @param in_channels Number of input channels
  90  * @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
  91  * @param flags See FF_MM_xx
  92  * @return NULL on error
  93  */
  94 AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
  95                                        enum SampleFormat in_fmt, int in_channels,
  96                                        const float *matrix, int flags);
  97
  98 /**
  99  * Free audio sample format converter context
 100  */
 101 void av_audio_convert_free(AVAudioConvert *ctx);
 102
 103 /**
 104  * Convert between audio sample formats
 105  * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
 106  * @param[in] out_stride distance between consecutive input samples (measured in bytes)
 107  * @param[in] in array of input buffers for each channel
 108  * @param[in] in_stride distance between consecutive output samples (measured in bytes)
 109  * @param len length of audio frame size (measured in samples)
 110  */
 111 int av_audio_convert(AVAudioConvert *ctx,
 112                            void * const out[6], const int out_stride[6],
 113                      const void * const  in[6], const int  in_stride[6], int len);
 114
 115 #endif /* AVCODEC_AUDIOCONVERT_H */