Merge branch 'mirror' into vdpau
[FFMpeg-mirror/ffmpeg-vdpau.git] / libavcodec / audioconvert.h
blob4b767101a45ebe0e7f5e4b4d15fc829fcc80427c
1 /*
2 * audio conversion
3 * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
4 * Copyright (c) 2008 Peter Ross
6 * This file is part of FFmpeg.
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.
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.
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
23 #ifndef AVCODEC_AUDIOCONVERT_H
24 #define AVCODEC_AUDIOCONVERT_H
26 /**
27 * @file audioconvert.h
28 * Audio format conversion routines
32 #include "avcodec.h"
35 /**
36 * Generate string corresponding to the sample format with
37 * number sample_fmt, or a header if sample_fmt is negative.
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.
45 void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
47 /**
48 * @return NULL on error
50 const char *avcodec_get_sample_fmt_name(int sample_fmt);
52 /**
53 * @return SAMPLE_FMT_NONE on error
55 enum SampleFormat avcodec_get_sample_fmt(const char* name);
57 /**
58 * @return NULL on error
60 const char *avcodec_get_channel_name(int channel_id);
62 /**
63 * Return description of channel layout
65 void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout);
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
74 int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name);
77 struct AVAudioConvert;
78 typedef struct AVAudioConvert AVAudioConvert;
80 /**
81 * Create an audio sample format converter context
82 * @param out_fmt Output sample format
83 * @param out_channels Number of output channels
84 * @param in_fmt Input sample format
85 * @param in_channels Number of input channels
86 * @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
87 * @param flags See FF_MM_xx
88 * @return NULL on error
90 AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
91 enum SampleFormat in_fmt, int in_channels,
92 const float *matrix, int flags);
94 /**
95 * Free audio sample format converter context
97 void av_audio_convert_free(AVAudioConvert *ctx);
99 /**
100 * Convert between audio sample formats
101 * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
102 * @param[in] out_stride distance between consecutive input samples (measured in bytes)
103 * @param[in] in array of input buffers for each channel
104 * @param[in] in_stride distance between consecutive output samples (measured in bytes)
105 * @param len length of audio frame size (measured in samples)
107 int av_audio_convert(AVAudioConvert *ctx,
108 void * const out[6], const int out_stride[6],
109 const void * const in[6], const int in_stride[6], int len);
111 #endif /* AVCODEC_AUDIOCONVERT_H */