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102 lines
4.0 KiB
102 lines
4.0 KiB
/* |
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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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