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111 lines
3.7 KiB
111 lines
3.7 KiB
/* |
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* audio conversion |
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* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> |
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* Copyright (c) 2008 Peter Ross |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#ifndef AVCODEC_AUDIOCONVERT_H |
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#define AVCODEC_AUDIOCONVERT_H |
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/** |
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* @file libavcodec/audioconvert.h |
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* Audio format conversion routines |
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*/ |
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#include "avcodec.h" |
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/** |
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* Generate string corresponding to the sample format with |
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* number sample_fmt, or a header if sample_fmt is negative. |
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* |
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* @param[in] buf the buffer where to write the string |
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* @param[in] buf_size the size of buf |
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* @param[in] sample_fmt the number of the sample format to print the corresponding info string, or |
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* a negative value to print the corresponding header. |
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* Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1. |
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*/ |
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void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt); |
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/** |
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* @return NULL on error |
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*/ |
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const char *avcodec_get_sample_fmt_name(int sample_fmt); |
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/** |
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* @return SAMPLE_FMT_NONE on error |
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*/ |
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enum SampleFormat avcodec_get_sample_fmt(const char* name); |
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/** |
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* @return NULL on error |
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*/ |
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const char *avcodec_get_channel_name(int channel_id); |
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/** |
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* Return description of channel layout |
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*/ |
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void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout); |
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/** |
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* Guess the channel layout |
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* @param nb_channels |
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* @param codec_id Codec identifier, or CODEC_ID_NONE if unknown |
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* @param fmt_name Format name, or NULL if unknown |
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* @return Channel layout mask |
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*/ |
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int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name); |
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struct AVAudioConvert; |
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typedef struct AVAudioConvert AVAudioConvert; |
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/** |
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* Create an audio sample format converter context |
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* @param out_fmt Output sample format |
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* @param out_channels Number of output channels |
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* @param in_fmt Input sample format |
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* @param in_channels Number of input channels |
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* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore. |
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* @param flags See FF_MM_xx |
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* @return NULL on error |
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*/ |
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AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels, |
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enum SampleFormat in_fmt, int in_channels, |
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const float *matrix, int flags); |
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/** |
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* Free audio sample format converter context |
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*/ |
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void av_audio_convert_free(AVAudioConvert *ctx); |
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/** |
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* Convert between audio sample formats |
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* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel. |
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* @param[in] out_stride distance between consecutive input samples (measured in bytes) |
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* @param[in] in array of input buffers for each channel |
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* @param[in] in_stride distance between consecutive output samples (measured in bytes) |
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* @param len length of audio frame size (measured in samples) |
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*/ |
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int av_audio_convert(AVAudioConvert *ctx, |
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void * const out[6], const int out_stride[6], |
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const void * const in[6], const int in_stride[6], int len); |
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#endif /* AVCODEC_AUDIOCONVERT_H */
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