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14758e3211
This allows AudioMix to be treated the same way as other conversion contexts and removes the requirement to allocate it at the same time as the AVAudioResampleContext. The current matrix get/set functions are split between the public interface and AudioMix private functions.
86 lines
4.5 KiB
C
86 lines
4.5 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_INTERNAL_H
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#define AVRESAMPLE_INTERNAL_H
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#include "libavutil/audio_fifo.h"
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#include "libavutil/log.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avresample.h"
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#include "audio_convert.h"
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#include "audio_data.h"
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#include "audio_mix.h"
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#include "resample.h"
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struct AVAudioResampleContext {
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const AVClass *av_class; /**< AVClass for logging and AVOptions */
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uint64_t in_channel_layout; /**< input channel layout */
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enum AVSampleFormat in_sample_fmt; /**< input sample format */
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int in_sample_rate; /**< input sample rate */
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uint64_t out_channel_layout; /**< output channel layout */
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enum AVSampleFormat out_sample_fmt; /**< output sample format */
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int out_sample_rate; /**< output sample rate */
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enum AVSampleFormat internal_sample_fmt; /**< internal sample format */
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enum AVMixCoeffType mix_coeff_type; /**< mixing coefficient type */
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double center_mix_level; /**< center mix level */
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double surround_mix_level; /**< surround mix level */
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double lfe_mix_level; /**< lfe mix level */
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int normalize_mix_level; /**< enable mix level normalization */
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int force_resampling; /**< force resampling */
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int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
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int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
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int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
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double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
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enum AVResampleFilterType filter_type; /**< resampling filter type */
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int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
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int in_channels; /**< number of input channels */
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int out_channels; /**< number of output channels */
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int resample_channels; /**< number of channels used for resampling */
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int downmix_needed; /**< downmixing is needed */
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int upmix_needed; /**< upmixing is needed */
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int mixing_needed; /**< either upmixing or downmixing is needed */
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int resample_needed; /**< resampling is needed */
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int in_convert_needed; /**< input sample format conversion is needed */
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int out_convert_needed; /**< output sample format conversion is needed */
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AudioData *in_buffer; /**< buffer for converted input */
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AudioData *resample_out_buffer; /**< buffer for output from resampler */
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AudioData *out_buffer; /**< buffer for converted output */
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AVAudioFifo *out_fifo; /**< FIFO for output samples */
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AudioConvert *ac_in; /**< input sample format conversion context */
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AudioConvert *ac_out; /**< output sample format conversion context */
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ResampleContext *resample; /**< resampling context */
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AudioMix *am; /**< channel mixing context */
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enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
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/**
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* mix matrix
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* only used if avresample_set_matrix() is called before avresample_open()
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*/
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double *mix_matrix;
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};
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#endif /* AVRESAMPLE_INTERNAL_H */
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