4cf4028331
Provides an option to disable mix level normalization Signed-off-by: Diego Biurrun <diego@biurrun.de>
80 lines
4.4 KiB
C
80 lines
4.4 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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#endif /* AVRESAMPLE_INTERNAL_H */
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