1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
|
/*
* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample 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.
*
* libswresample 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 libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWRESAMPLE_SWRESAMPLE_INTERNAL_H
#define SWRESAMPLE_SWRESAMPLE_INTERNAL_H
#include "swresample.h"
#include "libavutil/channel_layout.h"
#include "config.h"
#define SWR_CH_MAX 64
#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
#define NS_TAPS 20
#if ARCH_X86_64
typedef int64_t integer;
#else
typedef int integer;
#endif
typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);
typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
typedef struct AudioData{
uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel
uint8_t *data; ///< samples buffer
int ch_count; ///< number of channels
int bps; ///< bytes per sample
int count; ///< number of samples
int planar; ///< 1 if planar audio, 0 otherwise
enum AVSampleFormat fmt; ///< sample format
} AudioData;
struct DitherContext {
int method;
int noise_pos;
float scale;
float noise_scale; ///< Noise scale
int ns_taps; ///< Noise shaping dither taps
float ns_scale; ///< Noise shaping dither scale
float ns_scale_1; ///< Noise shaping dither scale^-1
int ns_pos; ///< Noise shaping dither position
float ns_coeffs[NS_TAPS]; ///< Noise shaping filter coefficients
float ns_errors[SWR_CH_MAX][2*NS_TAPS];
AudioData noise; ///< noise used for dithering
AudioData temp; ///< temporary storage when writing into the input buffer isn't possible
int output_sample_bits; ///< the number of used output bits, needed to scale dither correctly
};
typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby, int exact_rational);
typedef void (* resample_free_func)(struct ResampleContext **c);
typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
typedef int (* resample_flush_func)(struct SwrContext *c);
typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);
typedef int64_t (* get_out_samples_func)(struct SwrContext *s, int in_samples);
struct Resampler {
resample_init_func init;
resample_free_func free;
multiple_resample_func multiple_resample;
resample_flush_func flush;
set_compensation_func set_compensation;
get_delay_func get_delay;
invert_initial_buffer_func invert_initial_buffer;
get_out_samples_func get_out_samples;
};
extern struct Resampler const swri_resampler;
extern struct Resampler const swri_soxr_resampler;
struct SwrContext {
const AVClass *av_class; ///< AVClass used for AVOption and av_log()
int log_level_offset; ///< logging level offset
void *log_ctx; ///< parent logging context
enum AVSampleFormat in_sample_fmt; ///< input sample format
enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
enum AVSampleFormat out_sample_fmt; ///< output sample format
AVChannelLayout used_ch_layout; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
AVChannelLayout in_ch_layout; ///< input channel layout
AVChannelLayout out_ch_layout; ///< output channel layout
int in_sample_rate; ///< input sample rate
int out_sample_rate; ///< output sample rate
int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
float slev; ///< surround mixing level
float clev; ///< center mixing level
float lfe_mix_level; ///< LFE mixing level
float rematrix_volume; ///< rematrixing volume coefficient
float rematrix_maxval; ///< maximum value for rematrixing output
int matrix_encoding; /**< matrixed stereo encoding */
const int *channel_map; ///< channel index (or -1 if muted channel) map
int engine;
#if FF_API_OLD_CHANNEL_LAYOUT
int user_used_ch_count; ///< User set used channel count
int user_in_ch_count; ///< User set input channel count
int user_out_ch_count; ///< User set output channel count
int64_t user_in_ch_layout; ///< User set input channel layout
int64_t user_out_ch_layout; ///< User set output channel layout
#endif
AVChannelLayout user_used_chlayout; ///< User set used channel layout
AVChannelLayout user_in_chlayout; ///< User set input channel layout
AVChannelLayout user_out_chlayout; ///< User set output channel layout
enum AVSampleFormat user_int_sample_fmt; ///< User set internal sample format
int user_dither_method; ///< User set dither method
struct DitherContext dither;
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 */
int exact_rational; /**< if 1 then enable non power of 2 phase_count */
double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
int filter_type; /**< swr resampling filter type */
double kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
double precision; /**< soxr resampling precision (in bits) */
int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
float min_compensation; ///< swr minimum below which no compensation will happen
float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
float soft_compensation_duration; ///< swr duration over which soft compensation is applied
float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
int64_t firstpts_in_samples; ///< swr first pts in samples
int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
AudioData in; ///< input audio data
AudioData postin; ///< post-input audio data: used for rematrix/resample
AudioData midbuf; ///< intermediate audio data (postin/preout)
AudioData preout; ///< pre-output audio data: used for rematrix/resample
AudioData out; ///< converted output audio data
AudioData in_buffer; ///< cached audio data (convert and resample purpose)
AudioData silence; ///< temporary with silence
AudioData drop_temp; ///< temporary used to discard output
int in_buffer_index; ///< cached buffer position
int in_buffer_count; ///< cached buffer length
int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
int flushed; ///< 1 if data is to be flushed and no further input is expected
int64_t outpts; ///< output PTS
int64_t firstpts; ///< first PTS
int drop_output; ///< number of output samples to drop
double delayed_samples_fixup; ///< soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
struct AudioConvert *in_convert; ///< input conversion context
struct AudioConvert *out_convert; ///< output conversion context
struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
struct ResampleContext *resample; ///< resampling context
struct Resampler const *resampler; ///< resampler virtual function table
double matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
float matrix_flt[SWR_CH_MAX][SWR_CH_MAX]; ///< single precision floating point rematrixing coefficients
uint8_t *native_matrix;
uint8_t *native_one;
uint8_t *native_simd_one;
uint8_t *native_simd_matrix;
int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
mix_1_1_func_type *mix_1_1_f;
mix_1_1_func_type *mix_1_1_simd;
mix_2_1_func_type *mix_2_1_f;
mix_2_1_func_type *mix_2_1_simd;
mix_any_func_type *mix_any_f;
/* TODO: callbacks for ASM optimizations */
};
av_warn_unused_result
int swri_realloc_audio(AudioData *a, int count);
void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
av_warn_unused_result
int swri_rematrix_init(SwrContext *s);
void swri_rematrix_free(SwrContext *s);
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
int swri_rematrix_init_x86(struct SwrContext *s);
av_warn_unused_result
int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
av_warn_unused_result
int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels);
void swri_audio_convert_init_arm(struct AudioConvert *ac,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels);
void swri_audio_convert_init_x86(struct AudioConvert *ac,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels);
#endif
|
