Mercurial > libavcodec.hg
annotate audioconvert.h @ 7904:47717d5239fa libavcodec
AltiVec detection support for OpenBSD, patch by Brad, brad comstyle com.
author | diego |
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date | Mon, 22 Sep 2008 05:38:50 +0000 |
parents | c4a4495715dd |
children | c2ab7a8958ed |
rev | line source |
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1 /* |
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2 * audio conversion |
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3 * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> |
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4 * Copyright (c) 2008 Peter Ross |
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5 * |
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6 * This file is part of FFmpeg. |
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7 * |
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8 * FFmpeg is free software; you can redistribute it and/or |
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9 * modify it under the terms of the GNU Lesser General Public |
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10 * License as published by the Free Software Foundation; either |
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11 * version 2.1 of the License, or (at your option) any later version. |
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12 * |
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13 * FFmpeg is distributed in the hope that it will be useful, |
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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16 * Lesser General Public License for more details. |
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17 * |
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18 * You should have received a copy of the GNU Lesser General Public |
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19 * License along with FFmpeg; if not, write to the Free Software |
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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21 */ |
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22 |
7760 | 23 #ifndef AVCODEC_AUDIOCONVERT_H |
24 #define AVCODEC_AUDIOCONVERT_H | |
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25 |
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26 /** |
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27 * @file audioconvert.h |
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28 * Audio format conversion routines |
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29 */ |
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30 |
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31 |
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32 #include "avcodec.h" |
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33 |
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34 |
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35 /** |
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36 * Generate string corresponding to the sample format with |
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37 * number sample_fmt, or a header if sample_fmt is negative. |
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38 * |
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39 * @param[in] buf the buffer where to write the string |
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40 * @param[in] buf_size the size of buf |
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41 * @param[in] sample_fmt the number of the sample format to print the corresponding info string, or |
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42 * a negative value to print the corresponding header. |
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43 * Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1. |
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44 */ |
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45 void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt); |
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46 |
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47 /** |
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48 * @return NULL on error |
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49 */ |
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50 const char *avcodec_get_sample_fmt_name(int sample_fmt); |
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51 |
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52 /** |
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53 * @return SAMPLE_FMT_NONE on error |
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54 */ |
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55 enum SampleFormat avcodec_get_sample_fmt(const char* name); |
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56 |
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57 struct AVAudioConvert; |
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58 typedef struct AVAudioConvert AVAudioConvert; |
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59 |
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60 /** |
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61 * Create an audio sample format converter context |
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62 * @param out_fmt Output sample format |
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63 * @param out_channels Number of output channels |
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64 * @param in_fmt Input sample format |
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65 * @param in_channels Number of input channels |
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66 * @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore. |
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67 * @param flags See FF_MM_xx |
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68 * @return NULL on error |
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69 */ |
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70 AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels, |
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71 enum SampleFormat in_fmt, int in_channels, |
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72 const float *matrix, int flags); |
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73 |
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74 /** |
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75 * Free audio sample format converter context |
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76 */ |
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77 void av_audio_convert_free(AVAudioConvert *ctx); |
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78 |
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79 /** |
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80 * Convert between audio sample formats |
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81 * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel. |
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82 * @param[in] out_stride distance between consecutive input samples (measured in bytes) |
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83 * @param[in] in array of input buffers for each channel |
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84 * @param[in] in_stride distance between consecutive output samples (measured in bytes) |
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85 * @param len length of audio frame size (measured in samples) |
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86 */ |
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87 int av_audio_convert(AVAudioConvert *ctx, |
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88 void * const out[6], const int out_stride[6], |
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89 const void * const in[6], const int in_stride[6], int len); |
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90 |
7760 | 91 #endif /* AVCODEC_AUDIOCONVERT_H */ |