2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4 * This file is part of FFmpeg.
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 #ifndef AVRESAMPLE_INTERNAL_H
22 #define AVRESAMPLE_INTERNAL_H
24 #include "libavutil/audio_fifo.h"
25 #include "libavutil/log.h"
26 #include "libavutil/opt.h"
27 #include "libavutil/samplefmt.h"
28 #include "avresample.h"
30 typedef struct AudioData AudioData
;
31 typedef struct AudioConvert AudioConvert
;
32 typedef struct AudioMix AudioMix
;
33 typedef struct ResampleContext ResampleContext
;
43 typedef struct ChannelMapInfo
{
44 int channel_map
[AVRESAMPLE_MAX_CHANNELS
]; /**< source index of each output channel, -1 if not remapped */
45 int do_remap
; /**< remap needed */
46 int channel_copy
[AVRESAMPLE_MAX_CHANNELS
]; /**< dest index to copy from */
47 int do_copy
; /**< copy needed */
48 int channel_zero
[AVRESAMPLE_MAX_CHANNELS
]; /**< dest index to zero */
49 int do_zero
; /**< zeroing needed */
50 int input_map
[AVRESAMPLE_MAX_CHANNELS
]; /**< dest index of each input channel */
53 struct AVAudioResampleContext
{
54 const AVClass
*av_class
; /**< AVClass for logging and AVOptions */
56 uint64_t in_channel_layout
; /**< input channel layout */
57 enum AVSampleFormat in_sample_fmt
; /**< input sample format */
58 int in_sample_rate
; /**< input sample rate */
59 uint64_t out_channel_layout
; /**< output channel layout */
60 enum AVSampleFormat out_sample_fmt
; /**< output sample format */
61 int out_sample_rate
; /**< output sample rate */
62 enum AVSampleFormat internal_sample_fmt
; /**< internal sample format */
63 enum AVMixCoeffType mix_coeff_type
; /**< mixing coefficient type */
64 double center_mix_level
; /**< center mix level */
65 double surround_mix_level
; /**< surround mix level */
66 double lfe_mix_level
; /**< lfe mix level */
67 int normalize_mix_level
; /**< enable mix level normalization */
68 int force_resampling
; /**< force resampling */
69 int filter_size
; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
70 int phase_shift
; /**< log2 of the number of entries in the resampling polyphase filterbank */
71 int linear_interp
; /**< if 1 then the resampling FIR filter will be linearly interpolated */
72 double cutoff
; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
73 enum AVResampleFilterType filter_type
; /**< resampling filter type */
74 int kaiser_beta
; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
75 enum AVResampleDitherMethod dither_method
; /**< dither method */
77 int in_channels
; /**< number of input channels */
78 int out_channels
; /**< number of output channels */
79 int resample_channels
; /**< number of channels used for resampling */
80 int downmix_needed
; /**< downmixing is needed */
81 int upmix_needed
; /**< upmixing is needed */
82 int mixing_needed
; /**< either upmixing or downmixing is needed */
83 int resample_needed
; /**< resampling is needed */
84 int in_convert_needed
; /**< input sample format conversion is needed */
85 int out_convert_needed
; /**< output sample format conversion is needed */
86 int in_copy_needed
; /**< input data copy is needed */
88 AudioData
*in_buffer
; /**< buffer for converted input */
89 AudioData
*resample_out_buffer
; /**< buffer for output from resampler */
90 AudioData
*out_buffer
; /**< buffer for converted output */
91 AVAudioFifo
*out_fifo
; /**< FIFO for output samples */
93 AudioConvert
*ac_in
; /**< input sample format conversion context */
94 AudioConvert
*ac_out
; /**< output sample format conversion context */
95 ResampleContext
*resample
; /**< resampling context */
96 AudioMix
*am
; /**< channel mixing context */
97 enum AVMatrixEncoding matrix_encoding
; /**< matrixed stereo encoding */
101 * only used if avresample_set_matrix() is called before avresample_open()
106 enum RemapPoint remap_point
;
107 ChannelMapInfo ch_map_info
;
111 void ff_audio_resample_init_aarch64(ResampleContext
*c
,
112 enum AVSampleFormat sample_fmt
);
113 #endif /* AVRESAMPLE_INTERNAL_H */