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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 */
-
- 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 */
- };
-
- #endif /* AVRESAMPLE_INTERNAL_H */
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