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/*
* 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_INTERNAL_H
#define AVRESAMPLE_INTERNAL_H
#include "libavutil/audio_fifo.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "audio_convert.h"
#include "audio_data.h"
#include "audio_mix.h"
#include "resample.h"
struct AVAudioResampleContext {
const AVClass *av_class; /**< AVClass for logging and AVOptions */
uint64_t in_channel_layout; /**< input channel layout */
enum AVSampleFormat in_sample_fmt; /**< input sample format */
int in_sample_rate; /**< input sample rate */
uint64_t out_channel_layout; /**< output channel layout */
enum AVSampleFormat out_sample_fmt; /**< output sample format */
int out_sample_rate; /**< output sample rate */
enum AVSampleFormat internal_sample_fmt; /**< internal sample format */
enum AVMixCoeffType mix_coeff_type; /**< mixing coefficient type */
double center_mix_level; /**< center mix level */
double surround_mix_level; /**< surround mix level */
double lfe_mix_level; /**< lfe mix level */
int force_resampling; /**< force resampling */
int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
enum AVResampleFilterType filter_type; /**< resampling filter type */
int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
int in_channels; /**< number of input channels */
int out_channels; /**< number of output channels */
int resample_channels; /**< number of channels used for resampling */
int downmix_needed; /**< downmixing is needed */
int upmix_needed; /**< upmixing is needed */
int mixing_needed; /**< either upmixing or downmixing is needed */
int resample_needed; /**< resampling is needed */
int in_convert_needed; /**< input sample format conversion is needed */
int out_convert_needed; /**< output sample format conversion is needed */
AudioData *in_buffer; /**< buffer for converted input */
AudioData *resample_out_buffer; /**< buffer for output from resampler */
AudioData *out_buffer; /**< buffer for converted output */
AVAudioFifo *out_fifo; /**< FIFO for output samples */
AudioConvert *ac_in; /**< input sample format conversion context */
AudioConvert *ac_out; /**< output sample format conversion context */
ResampleContext *resample; /**< resampling context */
AudioMix *am; /**< channel mixing context */
enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
};
#endif /* AVRESAMPLE_INTERNAL_H */
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