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libavresample
internal.h
Go to the documentation of this file.
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/*
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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 FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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typedef
struct
AudioData
AudioData
;
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typedef
struct
AudioConvert
AudioConvert
;
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typedef
struct
AudioMix
AudioMix
;
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typedef
struct
ResampleContext
ResampleContext
;
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enum
RemapPoint
{
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REMAP_NONE
,
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REMAP_IN_COPY
,
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REMAP_IN_CONVERT
,
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REMAP_OUT_COPY
,
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REMAP_OUT_CONVERT
,
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};
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typedef
struct
ChannelMapInfo
{
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int
channel_map
[
AVRESAMPLE_MAX_CHANNELS
];
/**< source index of each output channel, -1 if not remapped */
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int
do_remap
;
/**< remap needed */
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int
channel_copy
[
AVRESAMPLE_MAX_CHANNELS
];
/**< dest index to copy from */
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int
do_copy
;
/**< copy needed */
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int
channel_zero
[
AVRESAMPLE_MAX_CHANNELS
];
/**< dest index to zero */
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int
do_zero
;
/**< zeroing needed */
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int
input_map
[
AVRESAMPLE_MAX_CHANNELS
];
/**< dest index of each input channel */
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}
ChannelMapInfo
;
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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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enum
AVResampleDitherMethod
dither_method
;
/**< dither method */
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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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int
in_copy_needed
;
/**< input data copy 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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int
use_channel_map
;
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enum
RemapPoint
remap_point
;
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ChannelMapInfo
ch_map_info
;
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};
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void
ff_audio_resample_init_aarch64
(
ResampleContext
*
c
,
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enum
AVSampleFormat
sample_fmt);
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void
ff_audio_resample_init_arm
(
ResampleContext
*
c
,
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enum
AVSampleFormat
sample_fmt);
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#endif
/* AVRESAMPLE_INTERNAL_H */
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