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/*
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 * Copyright (c) 2012 Justin Ruggles 
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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 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 */