mirror of
https://github.com/librempeg/librempeg
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103 lines
4.0 KiB
C
103 lines
4.0 KiB
C
/*
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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 AVRESAMPLE_AUDIO_CONVERT_H
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#define AVRESAMPLE_AUDIO_CONVERT_H
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#include "libavutil/samplefmt.h"
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#include "avresample.h"
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#include "internal.h"
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#include "audio_data.h"
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/**
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* Set conversion function if the parameters match.
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*
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* This compares the parameters of the conversion function to the parameters
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* in the AudioConvert context. If the parameters do not match, no changes are
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* made to the active functions. If the parameters do match and the alignment
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* is not constrained, the function is set as the generic conversion function.
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* If the parameters match and the alignment is constrained, the function is
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* set as the optimized conversion function.
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*
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* @param ac AudioConvert context
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* @param out_fmt output sample format
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* @param in_fmt input sample format
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* @param channels number of channels, or 0 for any number of channels
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* @param ptr_align buffer pointer alignment, in bytes
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* @param samples_align buffer size alignment, in samples
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* @param descr function type description (e.g. "C" or "SSE")
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* @param conv conversion function pointer
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*/
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void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
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enum AVSampleFormat in_fmt, int channels,
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int ptr_align, int samples_align,
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const char *descr, void *conv);
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/**
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* Allocate and initialize AudioConvert context for sample format conversion.
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*
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* @param avr AVAudioResampleContext
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* @param out_fmt output sample format
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* @param in_fmt input sample format
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* @param channels number of channels
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* @param sample_rate sample rate (used for dithering)
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* @param apply_map apply channel map during conversion
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* @return newly-allocated AudioConvert context
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*/
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AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
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enum AVSampleFormat out_fmt,
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enum AVSampleFormat in_fmt,
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int channels, int sample_rate,
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int apply_map);
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/**
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* Free AudioConvert.
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*
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* The AudioConvert must have been previously allocated with ff_audio_convert_alloc().
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*
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* @param ac AudioConvert struct
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*/
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void ff_audio_convert_free(AudioConvert **ac);
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/**
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* Convert audio data from one sample format to another.
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*
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* For each call, the alignment of the input and output AudioData buffers are
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* examined to determine whether to use the generic or optimized conversion
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* function (when available).
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*
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* The number of samples to convert is determined by in->nb_samples. The output
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* buffer must be large enough to handle this many samples. out->nb_samples is
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* set by this function before a successful return.
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*
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* @param ac AudioConvert context
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* @param out output audio data
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* @param in input audio data
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* @return 0 on success, negative AVERROR code on failure
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*/
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int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in);
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/* arch-specific initialization functions */
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void ff_audio_convert_init_arm(AudioConvert *ac);
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void ff_audio_convert_init_x86(AudioConvert *ac);
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#endif /* AVRESAMPLE_AUDIO_CONVERT_H */
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