2 * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
4 * This file is part of libswresample
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.
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.
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
21 #ifndef SWR_INTERNAL_H
22 #define SWR_INTERNAL_H
24 #include "swresample.h"
25 #include "libavutil/channel_layout.h"
28 #define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
33 typedef int64_t integer;
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);
41 typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
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
53 struct DitherContext {
54 enum SwrDitherType method;
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
68 const AVClass *av_class; ///< AVClass used for AVOption and av_log()
69 int log_level_offset; ///< logging level offset
70 void *log_ctx; ///< parent logging context
71 enum AVSampleFormat in_sample_fmt; ///< input sample format
72 enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
73 enum AVSampleFormat out_sample_fmt; ///< output sample format
74 int64_t in_ch_layout; ///< input channel layout
75 int64_t out_ch_layout; ///< output channel layout
76 int in_sample_rate; ///< input sample rate
77 int out_sample_rate; ///< output sample rate
78 int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
79 float slev; ///< surround mixing level
80 float clev; ///< center mixing level
81 float lfe_mix_level; ///< LFE mixing level
82 float rematrix_volume; ///< rematrixing volume coefficient
83 enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
84 const int *channel_map; ///< channel index (or -1 if muted channel) map
85 int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
86 enum SwrEngine engine;
88 struct DitherContext dither;
90 int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
91 int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
92 int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
93 double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
94 enum SwrFilterType filter_type; /**< swr resampling filter type */
95 int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
96 double precision; /**< soxr resampling precision (in bits) */
97 int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
99 float min_compensation; ///< swr minimum below which no compensation will happen
100 float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
101 float soft_compensation_duration; ///< swr duration over which soft compensation is applied
102 float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
103 float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
105 int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
106 int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
107 int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
109 AudioData in; ///< input audio data
110 AudioData postin; ///< post-input audio data: used for rematrix/resample
111 AudioData midbuf; ///< intermediate audio data (postin/preout)
112 AudioData preout; ///< pre-output audio data: used for rematrix/resample
113 AudioData out; ///< converted output audio data
114 AudioData in_buffer; ///< cached audio data (convert and resample purpose)
115 int in_buffer_index; ///< cached buffer position
116 int in_buffer_count; ///< cached buffer length
117 int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
118 int flushed; ///< 1 if data is to be flushed and no further input is expected
119 int64_t outpts; ///< output PTS
120 int drop_output; ///< number of output samples to drop
122 struct AudioConvert *in_convert; ///< input conversion context
123 struct AudioConvert *out_convert; ///< output conversion context
124 struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
125 struct ResampleContext *resample; ///< resampling context
126 struct Resampler const *resampler; ///< resampler virtual function table
128 float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
129 uint8_t *native_matrix;
131 uint8_t *native_simd_matrix;
132 int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
133 uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
134 mix_1_1_func_type *mix_1_1_f;
135 mix_1_1_func_type *mix_1_1_simd;
137 mix_2_1_func_type *mix_2_1_f;
138 mix_2_1_func_type *mix_2_1_simd;
140 mix_any_func_type *mix_any_f;
142 /* TODO: callbacks for ASM optimizations */
145 typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
146 double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby);
147 typedef void (* resample_free_func)(struct ResampleContext **c);
148 typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
149 typedef int (* resample_flush_func)(struct SwrContext *c);
150 typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
151 typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
154 resample_init_func init;
155 resample_free_func free;
156 multiple_resample_func multiple_resample;
157 resample_flush_func flush;
158 set_compensation_func set_compensation;
159 get_delay_func get_delay;
162 extern struct Resampler const swri_resampler;
164 int swri_realloc_audio(AudioData *a, int count);
165 int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
166 int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
167 int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx);
168 int swri_resample_double(struct ResampleContext *c,double *dst, const double *src, int *consumed, int src_size, int dst_size, int update_ctx);
170 void swri_noise_shaping_int16 (SwrContext *s, AudioData *srcs, AudioData *noises, int count);
171 void swri_noise_shaping_int32 (SwrContext *s, AudioData *srcs, AudioData *noises, int count);
172 void swri_noise_shaping_float (SwrContext *s, AudioData *srcs, AudioData *noises, int count);
173 void swri_noise_shaping_double(SwrContext *s, AudioData *srcs, AudioData *noises, int count);
175 int swri_rematrix_init(SwrContext *s);
176 void swri_rematrix_free(SwrContext *s);
177 int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
178 void swri_rematrix_init_x86(struct SwrContext *s);
180 void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
181 int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
183 void swri_audio_convert_init_arm(struct AudioConvert *ac,
184 enum AVSampleFormat out_fmt,
185 enum AVSampleFormat in_fmt,
187 void swri_audio_convert_init_x86(struct AudioConvert *ac,
188 enum AVSampleFormat out_fmt,
189 enum AVSampleFormat in_fmt,