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76 lines
4.1 KiB
76 lines
4.1 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_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 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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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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#endif /* AVRESAMPLE_INTERNAL_H */
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