FFmpeg: libswresample/swresample_internal.h Source File

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