/*
  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
  *
  * This file is part of Libav.
  *
  * Libav is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * Libav is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #ifndef AVRESAMPLE_AUDIO_CONVERT_H
 #define AVRESAMPLE_AUDIO_CONVERT_H
 
 #include "libavutil/samplefmt.h"
 #include "avresample.h"
 #include "audio_data.h"
 
 typedef struct AudioConvert AudioConvert;
 
 /**
  * Set conversion function if the parameters match.
  *
  * This compares the parameters of the conversion function to the parameters
  * in the AudioConvert context. If the parameters do not match, no changes are
  * made to the active functions. If the parameters do match and the alignment
  * is not constrained, the function is set as the generic conversion function.
  * If the parameters match and the alignment is constrained, the function is
  * set as the optimized conversion function.
  *
  * @param ac             AudioConvert context
  * @param out_fmt        output sample format
  * @param in_fmt         input sample format
  * @param channels       number of channels, or 0 for any number of channels
  * @param ptr_align      buffer pointer alignment, in bytes

  * @param samples_align  buffer size alignment, in samples

  * @param descr          function type description (e.g. "C" or "SSE")
  * @param conv           conversion function pointer
  */
 void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
                                enum AVSampleFormat in_fmt, int channels,
                                int ptr_align, int samples_align,
                                const char *descr, void *conv);
 
 /**
  * Allocate and initialize AudioConvert context for sample format conversion.
  *

  * @param avr         AVAudioResampleContext
  * @param out_fmt     output sample format
  * @param in_fmt      input sample format
  * @param channels    number of channels
  * @param sample_rate sample rate (used for dithering)
  * @return            newly-allocated AudioConvert context

  */
 AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
                                      enum AVSampleFormat out_fmt,
                                      enum AVSampleFormat in_fmt,

                                      int channels, int sample_rate);
 
 /**
  * Free AudioConvert.
  *
  * The AudioConvert must have been previously allocated with ff_audio_convert_alloc().
  *
  * @param ac  AudioConvert struct
  */
 void ff_audio_convert_free(AudioConvert **ac);

 
 /**
  * Convert audio data from one sample format to another.
  *
  * For each call, the alignment of the input and output AudioData buffers are
  * examined to determine whether to use the generic or optimized conversion
  * function (when available).
  *

  * The number of samples to convert is determined by in->nb_samples. The output
  * buffer must be large enough to handle this many samples. out->nb_samples is
  * set by this function before a successful return.
  *

  * @param ac     AudioConvert context
  * @param out    output audio data
  * @param in     input audio data
  * @return       0 on success, negative AVERROR code on failure
  */

 int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in);

 
 /* arch-specific initialization functions */
 

 void ff_audio_convert_init_arm(AudioConvert *ac);

 void ff_audio_convert_init_x86(AudioConvert *ac);
 
 #endif /* AVRESAMPLE_AUDIO_CONVERT_H */