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internal.h
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1 /*
2  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
3  *
4  * This file is part of FFmpeg.
5  *
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.
10  *
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.
15  *
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
19  */
20 
21 #ifndef AVRESAMPLE_INTERNAL_H
22 #define AVRESAMPLE_INTERNAL_H
23 
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"
29 
30 typedef struct AudioData AudioData;
31 typedef struct AudioConvert AudioConvert;
32 typedef struct AudioMix AudioMix;
33 typedef struct ResampleContext ResampleContext;
34 
35 enum RemapPoint {
41 };
42 
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 */
52 
54  const AVClass *av_class; /**< AVClass for logging and AVOptions */
55 
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 */
76 
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 */
87 
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 */
92 
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 */
98 
99  /**
100  * mix matrix
101  * only used if avresample_set_matrix() is called before avresample_open()
102  */
103  double *mix_matrix;
104 
108 };
109 
110 
112  enum AVSampleFormat sample_fmt);
114  enum AVSampleFormat sample_fmt);
115 
116 #endif /* AVRESAMPLE_INTERNAL_H */