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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 SWRESAMPLE_SWRESAMPLE_INTERNAL_H
22 #define SWRESAMPLE_SWRESAMPLE_INTERNAL_H
23 
24 #include "swresample.h"
25 #include "libavutil/channel_layout.h"
26 #include "config.h"
27 
28 #define SWR_CH_MAX 64
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     int 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 typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
72                                     double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby, int exact_rational);
73 typedef void    (* resample_free_func)(struct ResampleContext **c);
74 typedef int     (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
75 typedef int     (* resample_flush_func)(struct SwrContext *c);
76 typedef int     (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
77 typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
78 typedef int     (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);
79 typedef int64_t (* get_out_samples_func)(struct SwrContext *s, int in_samples);
80 
81 struct Resampler {
82   resample_init_func            init;
83   resample_free_func            free;
84   multiple_resample_func        multiple_resample;
85   resample_flush_func           flush;
86   set_compensation_func         set_compensation;
87   get_delay_func                get_delay;
88   invert_initial_buffer_func    invert_initial_buffer;
89   get_out_samples_func          get_out_samples;
90 };
91 
92 extern struct Resampler const swri_resampler;
93 extern struct Resampler const swri_soxr_resampler;
94 
95 struct SwrContext {
96     const AVClass *av_class;                        ///< AVClass used for AVOption and av_log()
97     int log_level_offset;                           ///< logging level offset
98     void *log_ctx;                                  ///< parent logging context
99     enum AVSampleFormat  in_sample_fmt;             ///< input sample format
100     enum AVSampleFormat int_sample_fmt;             ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
101     enum AVSampleFormat out_sample_fmt;             ///< output sample format
102     int64_t  in_ch_layout;                          ///< input channel layout
103     int64_t out_ch_layout;                          ///< output channel layout
104     int      in_sample_rate;                        ///< input sample rate
105     int     out_sample_rate;                        ///< output sample rate
106     int flags;                                      ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
107     float slev;                                     ///< surround mixing level
108     float clev;                                     ///< center mixing level
109     float lfe_mix_level;                            ///< LFE mixing level
110     float rematrix_volume;                          ///< rematrixing volume coefficient
111     float rematrix_maxval;                          ///< maximum value for rematrixing output
112     int matrix_encoding;                            /**< matrixed stereo encoding */
113     const int *channel_map;                         ///< channel index (or -1 if muted channel) map
114     int used_ch_count;                              ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
115     int engine;
116 
117     int user_in_ch_count;                           ///< User set input channel count
118     int user_out_ch_count;                          ///< User set output channel count
119     int user_used_ch_count;                         ///< User set used channel count
120     int64_t user_in_ch_layout;                      ///< User set input channel layout
121     int64_t user_out_ch_layout;                     ///< User set output channel layout
122     enum AVSampleFormat user_int_sample_fmt;        ///< User set internal sample format
123     int user_dither_method;                         ///< User set dither method
124 
125     struct DitherContext dither;
126 
127     int filter_size;                                /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
128     int phase_shift;                                /**< log2 of the number of entries in the resampling polyphase filterbank */
129     int linear_interp;                              /**< if 1 then the resampling FIR filter will be linearly interpolated */
130     int exact_rational;                             /**< if 1 then enable non power of 2 phase_count */
131     double cutoff;                                  /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
132     int filter_type;                                /**< swr resampling filter type */
133     double kaiser_beta;                                /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
134     double precision;                               /**< soxr resampling precision (in bits) */
135     int cheby;                                      /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
136 
137     float min_compensation;                         ///< swr minimum below which no compensation will happen
138     float min_hard_compensation;                    ///< swr minimum below which no silence inject / sample drop will happen
139     float soft_compensation_duration;               ///< swr duration over which soft compensation is applied
140     float max_soft_compensation;                    ///< swr maximum soft compensation in seconds over soft_compensation_duration
141     float async;                                    ///< swr simple 1 parameter async, similar to ffmpegs -async
142     int64_t firstpts_in_samples;                    ///< swr first pts in samples
143 
144     int resample_first;                             ///< 1 if resampling must come first, 0 if rematrixing
145     int rematrix;                                   ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
146     int rematrix_custom;                            ///< flag to indicate that a custom matrix has been defined
147 
148     AudioData in;                                   ///< input audio data
149     AudioData postin;                               ///< post-input audio data: used for rematrix/resample
150     AudioData midbuf;                               ///< intermediate audio data (postin/preout)
151     AudioData preout;                               ///< pre-output audio data: used for rematrix/resample
152     AudioData out;                                  ///< converted output audio data
153     AudioData in_buffer;                            ///< cached audio data (convert and resample purpose)
154     AudioData silence;                              ///< temporary with silence
155     AudioData drop_temp;                            ///< temporary used to discard output
156     int in_buffer_index;                            ///< cached buffer position
157     int in_buffer_count;                            ///< cached buffer length
158     int resample_in_constraint;                     ///< 1 if the input end was reach before the output end, 0 otherwise
159     int flushed;                                    ///< 1 if data is to be flushed and no further input is expected
160     int64_t outpts;                                 ///< output PTS
161     int64_t firstpts;                               ///< first PTS
162     int drop_output;                                ///< number of output samples to drop
163     double delayed_samples_fixup;                   ///< soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
164 
165     struct AudioConvert *in_convert;                ///< input conversion context
166     struct AudioConvert *out_convert;               ///< output conversion context
167     struct AudioConvert *full_convert;              ///< full conversion context (single conversion for input and output)
168     struct ResampleContext *resample;               ///< resampling context
169     struct Resampler const *resampler;              ///< resampler virtual function table
170 
171     double matrix[SWR_CH_MAX][SWR_CH_MAX];          ///< floating point rematrixing coefficients
172     float matrix_flt[SWR_CH_MAX][SWR_CH_MAX];       ///< single precision floating point rematrixing coefficients
173     uint8_t *native_matrix;
174     uint8_t *native_one;
175     uint8_t *native_simd_one;
176     uint8_t *native_simd_matrix;
177     int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX];       ///< 17.15 fixed point rematrixing coefficients
178     uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1];    ///< Lists of input channels per output channel that have non zero rematrixing coefficients
179     mix_1_1_func_type *mix_1_1_f;
180     mix_1_1_func_type *mix_1_1_simd;
181 
182     mix_2_1_func_type *mix_2_1_f;
183     mix_2_1_func_type *mix_2_1_simd;
184 
185     mix_any_func_type *mix_any_f;
186 
187     /* TODO: callbacks for ASM optimizations */
188 };
189 
190 av_warn_unused_result
191 int swri_realloc_audio(AudioData *a, int count);
192 
193 void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
194 void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
195 void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
196 void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
197 
198 av_warn_unused_result
199 int swri_rematrix_init(SwrContext *s);
200 void swri_rematrix_free(SwrContext *s);
201 int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
202 int swri_rematrix_init_x86(struct SwrContext *s);
203 
204 av_warn_unused_result
205 int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
206 av_warn_unused_result
207 int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
208 
209 void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
210                                  enum AVSampleFormat out_fmt,
211                                  enum AVSampleFormat in_fmt,
212                                  int channels);
213 void swri_audio_convert_init_arm(struct AudioConvert *ac,
214                                  enum AVSampleFormat out_fmt,
215                                  enum AVSampleFormat in_fmt,
216                                  int channels);
217 void swri_audio_convert_init_x86(struct AudioConvert *ac,
218                                  enum AVSampleFormat out_fmt,
219                                  enum AVSampleFormat in_fmt,
220                                  int channels);
221 
222 #endif
223