2 * Copyright (C) 2011-2012 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) */
31 typedef int64_t integer;
36 typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
37 typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);
39 typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
41 typedef struct AudioData{
42 uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel
43 uint8_t *data; ///< samples buffer
44 int ch_count; ///< number of channels
45 int bps; ///< bytes per sample
46 int count; ///< number of samples
47 int planar; ///< 1 if planar audio, 0 otherwise
48 enum AVSampleFormat fmt; ///< sample format
52 const AVClass *av_class; ///< AVClass used for AVOption and av_log()
53 int log_level_offset; ///< logging level offset
54 void *log_ctx; ///< parent logging context
55 enum AVSampleFormat in_sample_fmt; ///< input sample format
56 enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
57 enum AVSampleFormat out_sample_fmt; ///< output sample format
58 int64_t in_ch_layout; ///< input channel layout
59 int64_t out_ch_layout; ///< output channel layout
60 int in_sample_rate; ///< input sample rate
61 int out_sample_rate; ///< output sample rate
62 int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
63 float slev; ///< surround mixing level
64 float clev; ///< center mixing level
65 float lfe_mix_level; ///< LFE mixing level
66 float rematrix_volume; ///< rematrixing volume coefficient
67 enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
68 const int *channel_map; ///< channel index (or -1 if muted channel) map
69 int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
70 enum SwrDitherType dither_method;
73 int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
74 int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
75 int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
76 double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
77 enum SwrFilterType filter_type; /**< resampling filter type */
78 int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
80 float min_compensation; ///< minimum below which no compensation will happen
81 float min_hard_compensation; ///< minimum below which no silence inject / sample drop will happen
82 float soft_compensation_duration; ///< duration over which soft compensation is applied
83 float max_soft_compensation; ///< maximum soft compensation in seconds over soft_compensation_duration
85 int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
86 int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
87 int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
89 AudioData in; ///< input audio data
90 AudioData postin; ///< post-input audio data: used for rematrix/resample
91 AudioData midbuf; ///< intermediate audio data (postin/preout)
92 AudioData preout; ///< pre-output audio data: used for rematrix/resample
93 AudioData out; ///< converted output audio data
94 AudioData in_buffer; ///< cached audio data (convert and resample purpose)
95 AudioData dither; ///< noise used for dithering
96 int in_buffer_index; ///< cached buffer position
97 int in_buffer_count; ///< cached buffer length
98 int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
99 int flushed; ///< 1 if data is to be flushed and no further input is expected
100 int64_t outpts; ///< output PTS
101 int drop_output; ///< number of output samples to drop
103 struct AudioConvert *in_convert; ///< input conversion context
104 struct AudioConvert *out_convert; ///< output conversion context
105 struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
106 struct ResampleContext *resample; ///< resampling context
108 float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
109 uint8_t *native_matrix;
111 uint8_t *native_simd_matrix;
112 int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
113 uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
114 mix_1_1_func_type *mix_1_1_f;
115 mix_1_1_func_type *mix_1_1_simd;
117 mix_2_1_func_type *mix_2_1_f;
118 mix_2_1_func_type *mix_2_1_simd;
120 mix_any_func_type *mix_any_f;
122 /* TODO: callbacks for ASM optimizations */
125 struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat, enum SwrFilterType, int kaiser_beta);
126 void swri_resample_free(struct ResampleContext **c);
127 int swri_multiple_resample(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
128 void swri_resample_compensate(struct ResampleContext *c, int sample_delta, int compensation_distance);
129 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);
130 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);
131 int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx);
132 int swri_resample_double(struct ResampleContext *c,double *dst, const double *src, int *consumed, int src_size, int dst_size, int update_ctx);
134 int swri_rematrix_init(SwrContext *s);
135 void swri_rematrix_free(SwrContext *s);
136 int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
137 void swri_rematrix_init_x86(struct SwrContext *s);
139 void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
141 void swri_audio_convert_init_arm(struct AudioConvert *ac,
142 enum AVSampleFormat out_fmt,
143 enum AVSampleFormat in_fmt,
145 void swri_audio_convert_init_x86(struct AudioConvert *ac,
146 enum AVSampleFormat out_fmt,
147 enum AVSampleFormat in_fmt,