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