Commit | Line | Data |
---|---|---|
2ba45a60 DM |
1 | /* |
2 | * Copyright (C) 2011-2013 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 SWR_CH_MAX 32 | |
29 | ||
30 | #define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */ | |
31 | ||
32 | #define NS_TAPS 20 | |
33 | ||
34 | #if ARCH_X86_64 | |
35 | typedef int64_t integer; | |
36 | #else | |
37 | typedef int integer; | |
38 | #endif | |
39 | ||
40 | typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len); | |
41 | typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len); | |
42 | ||
43 | typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len); | |
44 | ||
45 | typedef struct AudioData{ | |
46 | uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel | |
47 | uint8_t *data; ///< samples buffer | |
48 | int ch_count; ///< number of channels | |
49 | int bps; ///< bytes per sample | |
50 | int count; ///< number of samples | |
51 | int planar; ///< 1 if planar audio, 0 otherwise | |
52 | enum AVSampleFormat fmt; ///< sample format | |
53 | } AudioData; | |
54 | ||
55 | struct DitherContext { | |
56 | enum SwrDitherType method; | |
57 | int noise_pos; | |
58 | float scale; | |
59 | float noise_scale; ///< Noise scale | |
60 | int ns_taps; ///< Noise shaping dither taps | |
61 | float ns_scale; ///< Noise shaping dither scale | |
62 | float ns_scale_1; ///< Noise shaping dither scale^-1 | |
63 | int ns_pos; ///< Noise shaping dither position | |
64 | float ns_coeffs[NS_TAPS]; ///< Noise shaping filter coefficients | |
65 | float ns_errors[SWR_CH_MAX][2*NS_TAPS]; | |
66 | AudioData noise; ///< noise used for dithering | |
67 | AudioData temp; ///< temporary storage when writing into the input buffer isn't possible | |
68 | int output_sample_bits; ///< the number of used output bits, needed to scale dither correctly | |
69 | }; | |
70 | ||
71 | struct SwrContext { | |
72 | const AVClass *av_class; ///< AVClass used for AVOption and av_log() | |
73 | int log_level_offset; ///< logging level offset | |
74 | void *log_ctx; ///< parent logging context | |
75 | enum AVSampleFormat in_sample_fmt; ///< input sample format | |
76 | enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P) | |
77 | enum AVSampleFormat out_sample_fmt; ///< output sample format | |
78 | int64_t in_ch_layout; ///< input channel layout | |
79 | int64_t out_ch_layout; ///< output channel layout | |
80 | int in_sample_rate; ///< input sample rate | |
81 | int out_sample_rate; ///< output sample rate | |
82 | int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE | |
83 | float slev; ///< surround mixing level | |
84 | float clev; ///< center mixing level | |
85 | float lfe_mix_level; ///< LFE mixing level | |
86 | float rematrix_volume; ///< rematrixing volume coefficient | |
87 | float rematrix_maxval; ///< maximum value for rematrixing output | |
88 | enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */ | |
89 | const int *channel_map; ///< channel index (or -1 if muted channel) map | |
90 | int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count) | |
91 | enum SwrEngine engine; | |
92 | ||
93 | struct DitherContext dither; | |
94 | ||
95 | int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */ | |
96 | int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */ | |
97 | int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */ | |
98 | double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */ | |
99 | enum SwrFilterType filter_type; /**< swr resampling filter type */ | |
100 | int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */ | |
101 | double precision; /**< soxr resampling precision (in bits) */ | |
102 | int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */ | |
103 | ||
104 | float min_compensation; ///< swr minimum below which no compensation will happen | |
105 | float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen | |
106 | float soft_compensation_duration; ///< swr duration over which soft compensation is applied | |
107 | float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration | |
108 | float async; ///< swr simple 1 parameter async, similar to ffmpegs -async | |
109 | int64_t firstpts_in_samples; ///< swr first pts in samples | |
110 | ||
111 | int resample_first; ///< 1 if resampling must come first, 0 if rematrixing | |
112 | int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch) | |
113 | int rematrix_custom; ///< flag to indicate that a custom matrix has been defined | |
114 | ||
115 | AudioData in; ///< input audio data | |
116 | AudioData postin; ///< post-input audio data: used for rematrix/resample | |
117 | AudioData midbuf; ///< intermediate audio data (postin/preout) | |
118 | AudioData preout; ///< pre-output audio data: used for rematrix/resample | |
119 | AudioData out; ///< converted output audio data | |
120 | AudioData in_buffer; ///< cached audio data (convert and resample purpose) | |
121 | AudioData silence; ///< temporary with silence | |
122 | AudioData drop_temp; ///< temporary used to discard output | |
123 | int in_buffer_index; ///< cached buffer position | |
124 | int in_buffer_count; ///< cached buffer length | |
125 | int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise | |
126 | int flushed; ///< 1 if data is to be flushed and no further input is expected | |
127 | int64_t outpts; ///< output PTS | |
128 | int64_t firstpts; ///< first PTS | |
129 | int drop_output; ///< number of output samples to drop | |
130 | ||
131 | struct AudioConvert *in_convert; ///< input conversion context | |
132 | struct AudioConvert *out_convert; ///< output conversion context | |
133 | struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output) | |
134 | struct ResampleContext *resample; ///< resampling context | |
135 | struct Resampler const *resampler; ///< resampler virtual function table | |
136 | ||
137 | float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients | |
138 | uint8_t *native_matrix; | |
139 | uint8_t *native_one; | |
140 | uint8_t *native_simd_one; | |
141 | uint8_t *native_simd_matrix; | |
142 | int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients | |
143 | uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients | |
144 | mix_1_1_func_type *mix_1_1_f; | |
145 | mix_1_1_func_type *mix_1_1_simd; | |
146 | ||
147 | mix_2_1_func_type *mix_2_1_f; | |
148 | mix_2_1_func_type *mix_2_1_simd; | |
149 | ||
150 | mix_any_func_type *mix_any_f; | |
151 | ||
152 | /* TODO: callbacks for ASM optimizations */ | |
153 | }; | |
154 | ||
155 | typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, | |
156 | double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby); | |
157 | typedef void (* resample_free_func)(struct ResampleContext **c); | |
158 | typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed); | |
159 | typedef int (* resample_flush_func)(struct SwrContext *c); | |
160 | typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance); | |
161 | typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base); | |
162 | typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count); | |
163 | ||
164 | struct Resampler { | |
165 | resample_init_func init; | |
166 | resample_free_func free; | |
167 | multiple_resample_func multiple_resample; | |
168 | resample_flush_func flush; | |
169 | set_compensation_func set_compensation; | |
170 | get_delay_func get_delay; | |
171 | invert_initial_buffer_func invert_initial_buffer; | |
172 | }; | |
173 | ||
174 | extern struct Resampler const swri_resampler; | |
175 | ||
176 | int swri_realloc_audio(AudioData *a, int count); | |
177 | ||
178 | void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); | |
179 | void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); | |
180 | void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); | |
181 | void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); | |
182 | ||
183 | int swri_rematrix_init(SwrContext *s); | |
184 | void swri_rematrix_free(SwrContext *s); | |
185 | int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy); | |
186 | void swri_rematrix_init_x86(struct SwrContext *s); | |
187 | ||
188 | void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt); | |
189 | int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt); | |
190 | ||
191 | void swri_audio_convert_init_aarch64(struct AudioConvert *ac, | |
192 | enum AVSampleFormat out_fmt, | |
193 | enum AVSampleFormat in_fmt, | |
194 | int channels); | |
195 | void swri_audio_convert_init_arm(struct AudioConvert *ac, | |
196 | enum AVSampleFormat out_fmt, | |
197 | enum AVSampleFormat in_fmt, | |
198 | int channels); | |
199 | void swri_audio_convert_init_x86(struct AudioConvert *ac, | |
200 | enum AVSampleFormat out_fmt, | |
201 | enum AVSampleFormat in_fmt, | |
202 | int channels); | |
203 | #endif |