/* * Copyright (c) 2012 Justin Ruggles * * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; 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" typedef struct AudioData AudioData; typedef struct AudioConvert AudioConvert; typedef struct AudioMix AudioMix; typedef struct ResampleContext ResampleContext; enum RemapPoint { REMAP_NONE, REMAP_IN_COPY, REMAP_IN_CONVERT, REMAP_OUT_COPY, REMAP_OUT_CONVERT, }; typedef struct ChannelMapInfo { int channel_map[AVRESAMPLE_MAX_CHANNELS]; /**< source index of each output channel, -1 if not remapped */ int do_remap; /**< remap needed */ int channel_copy[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to copy from */ int do_copy; /**< copy needed */ int channel_zero[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to zero */ int do_zero; /**< zeroing needed */ int input_map[AVRESAMPLE_MAX_CHANNELS]; /**< dest index of each input channel */ } ChannelMapInfo; 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 normalize_mix_level; /**< enable mix level normalization */ 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) */ enum AVResampleDitherMethod dither_method; /**< dither method */ 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 */ int in_copy_needed; /**< input data copy 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 */ /** * mix matrix * only used if avresample_set_matrix() is called before avresample_open() */ double *mix_matrix; int use_channel_map; enum RemapPoint remap_point; ChannelMapInfo ch_map_info; }; void ff_audio_resample_init_aarch64(ResampleContext *c, enum AVSampleFormat sample_fmt); void ff_audio_resample_init_arm(ResampleContext *c, enum AVSampleFormat sample_fmt); #endif /* AVRESAMPLE_INTERNAL_H */