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swresample_internal.h
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1 /*
2  * Copyright (C) 2011-2012 Michael Niedermayer (michaelni@gmx.at)
3  *
4  * This file is part of libswresample
5  *
6  * libswresample 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  * libswresample 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 libswresample; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #ifndef SWR_INTERNAL_H
22 #define SWR_INTERNAL_H
23 
24 #include "swresample.h"
25 #include "libavutil/channel_layout.h"
26 #include "config.h"
27 
28 #define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
29 
30 #if ARCH_X86_64
31 typedef int64_t integer;
32 #else
33 typedef int integer;
34 #endif
35 
36 typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
37 typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);
38 
39 typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
40 
41 typedef struct AudioData{
42  uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel
43  uint8_t *data; ///< samples buffer
44  int ch_count; ///< number of channels
45  int bps; ///< bytes per sample
46  int count; ///< number of samples
47  int planar; ///< 1 if planar audio, 0 otherwise
48  enum AVSampleFormat fmt; ///< sample format
49 } AudioData;
50 
51 struct SwrContext {
52  const AVClass *av_class; ///< AVClass used for AVOption and av_log()
53  int log_level_offset; ///< logging level offset
54  void *log_ctx; ///< parent logging context
55  enum AVSampleFormat in_sample_fmt; ///< input sample format
56  enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
57  enum AVSampleFormat out_sample_fmt; ///< output sample format
58  int64_t in_ch_layout; ///< input channel layout
59  int64_t out_ch_layout; ///< output channel layout
60  int in_sample_rate; ///< input sample rate
61  int out_sample_rate; ///< output sample rate
62  int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
63  float slev; ///< surround mixing level
64  float clev; ///< center mixing level
65  float lfe_mix_level; ///< LFE mixing level
66  float rematrix_volume; ///< rematrixing volume coefficient
67  enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
68  const int *channel_map; ///< channel index (or -1 if muted channel) map
69  int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
70  enum SwrEngine engine;
71  enum SwrDitherType dither_method;
72  int dither_pos;
73  float dither_scale;
74  int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
75  int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
76  int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
77  double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
78  enum SwrFilterType filter_type; /**< swr resampling filter type */
79  int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
80  double precision; /**< soxr resampling precision (in bits) */
81  int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
82 
83  float min_compensation; ///< swr minimum below which no compensation will happen
84  float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
85  float soft_compensation_duration; ///< swr duration over which soft compensation is applied
86  float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
87  float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
88 
89  int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
90  int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
91  int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
92 
93  AudioData in; ///< input audio data
94  AudioData postin; ///< post-input audio data: used for rematrix/resample
95  AudioData midbuf; ///< intermediate audio data (postin/preout)
96  AudioData preout; ///< pre-output audio data: used for rematrix/resample
97  AudioData out; ///< converted output audio data
98  AudioData in_buffer; ///< cached audio data (convert and resample purpose)
99  AudioData dither; ///< noise used for dithering
100  int in_buffer_index; ///< cached buffer position
101  int in_buffer_count; ///< cached buffer length
102  int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
103  int flushed; ///< 1 if data is to be flushed and no further input is expected
104  int64_t outpts; ///< output PTS
105  int drop_output; ///< number of output samples to drop
106 
107  struct AudioConvert *in_convert; ///< input conversion context
108  struct AudioConvert *out_convert; ///< output conversion context
109  struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
110  struct ResampleContext *resample; ///< resampling context
111  struct Resampler const *resampler; ///< resampler virtual function table
112 
113  float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
114  uint8_t *native_matrix;
115  uint8_t *native_one;
116  uint8_t *native_simd_matrix;
117  int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
118  uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
119  mix_1_1_func_type *mix_1_1_f;
120  mix_1_1_func_type *mix_1_1_simd;
121 
122  mix_2_1_func_type *mix_2_1_f;
123  mix_2_1_func_type *mix_2_1_simd;
124 
125  mix_any_func_type *mix_any_f;
126 
127  /* TODO: callbacks for ASM optimizations */
128 };
129 
130 typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
131  double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby);
132 typedef void (* resample_free_func)(struct ResampleContext **c);
133 typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
134 typedef int (* resample_flush_func)(struct SwrContext *c);
135 typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
136 typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
137 
138 struct Resampler {
139  resample_init_func init;
140  resample_free_func free;
141  multiple_resample_func multiple_resample;
142  resample_flush_func flush;
143  set_compensation_func set_compensation;
144  get_delay_func get_delay;
145 };
146 
147 extern struct Resampler const swri_resampler;
148 
149 int swri_realloc_audio(AudioData *a, int count);
150 int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
151 int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
152 int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx);
153 int swri_resample_double(struct ResampleContext *c,double *dst, const double *src, int *consumed, int src_size, int dst_size, int update_ctx);
154 
155 int swri_rematrix_init(SwrContext *s);
156 void swri_rematrix_free(SwrContext *s);
157 int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
158 void swri_rematrix_init_x86(struct SwrContext *s);
159 
160 void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
161 
162 void swri_audio_convert_init_arm(struct AudioConvert *ac,
163  enum AVSampleFormat out_fmt,
164  enum AVSampleFormat in_fmt,
165  int channels);
166 void swri_audio_convert_init_x86(struct AudioConvert *ac,
167  enum AVSampleFormat out_fmt,
168  enum AVSampleFormat in_fmt,
169  int channels);
170 #endif
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