Imported Debian version 2.5.2~trusty
[deb_ffmpeg.git] / ffmpeg / libswresample / swresample_internal.h
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 enum SwrDitherType 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 struct SwrContext {
72 const AVClass *av_class; ///< AVClass used for AVOption and av_log()
73 int log_level_offset; ///< logging level offset
74 void *log_ctx; ///< parent logging context
75 enum AVSampleFormat in_sample_fmt; ///< input sample format
76 enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
77 enum AVSampleFormat out_sample_fmt; ///< output sample format
78 int64_t in_ch_layout; ///< input channel layout
79 int64_t out_ch_layout; ///< output channel layout
80 int in_sample_rate; ///< input sample rate
81 int out_sample_rate; ///< output sample rate
82 int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
83 float slev; ///< surround mixing level
84 float clev; ///< center mixing level
85 float lfe_mix_level; ///< LFE mixing level
86 float rematrix_volume; ///< rematrixing volume coefficient
87 float rematrix_maxval; ///< maximum value for rematrixing output
88 enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
89 const int *channel_map; ///< channel index (or -1 if muted channel) map
90 int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
91 enum SwrEngine engine;
92
93 struct DitherContext dither;
94
95 int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
96 int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
97 int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
98 double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
99 enum SwrFilterType filter_type; /**< swr resampling filter type */
100 int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
101 double precision; /**< soxr resampling precision (in bits) */
102 int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
103
104 float min_compensation; ///< swr minimum below which no compensation will happen
105 float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
106 float soft_compensation_duration; ///< swr duration over which soft compensation is applied
107 float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
108 float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
109 int64_t firstpts_in_samples; ///< swr first pts in samples
110
111 int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
112 int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
113 int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
114
115 AudioData in; ///< input audio data
116 AudioData postin; ///< post-input audio data: used for rematrix/resample
117 AudioData midbuf; ///< intermediate audio data (postin/preout)
118 AudioData preout; ///< pre-output audio data: used for rematrix/resample
119 AudioData out; ///< converted output audio data
120 AudioData in_buffer; ///< cached audio data (convert and resample purpose)
121 AudioData silence; ///< temporary with silence
122 AudioData drop_temp; ///< temporary used to discard output
123 int in_buffer_index; ///< cached buffer position
124 int in_buffer_count; ///< cached buffer length
125 int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
126 int flushed; ///< 1 if data is to be flushed and no further input is expected
127 int64_t outpts; ///< output PTS
128 int64_t firstpts; ///< first PTS
129 int drop_output; ///< number of output samples to drop
130
131 struct AudioConvert *in_convert; ///< input conversion context
132 struct AudioConvert *out_convert; ///< output conversion context
133 struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
134 struct ResampleContext *resample; ///< resampling context
135 struct Resampler const *resampler; ///< resampler virtual function table
136
137 float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
138 uint8_t *native_matrix;
139 uint8_t *native_one;
140 uint8_t *native_simd_one;
141 uint8_t *native_simd_matrix;
142 int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
143 uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
144 mix_1_1_func_type *mix_1_1_f;
145 mix_1_1_func_type *mix_1_1_simd;
146
147 mix_2_1_func_type *mix_2_1_f;
148 mix_2_1_func_type *mix_2_1_simd;
149
150 mix_any_func_type *mix_any_f;
151
152 /* TODO: callbacks for ASM optimizations */
153 };
154
155 typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
156 double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby);
157 typedef void (* resample_free_func)(struct ResampleContext **c);
158 typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
159 typedef int (* resample_flush_func)(struct SwrContext *c);
160 typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
161 typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
162 typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);
163
164 struct Resampler {
165 resample_init_func init;
166 resample_free_func free;
167 multiple_resample_func multiple_resample;
168 resample_flush_func flush;
169 set_compensation_func set_compensation;
170 get_delay_func get_delay;
171 invert_initial_buffer_func invert_initial_buffer;
172 };
173
174 extern struct Resampler const swri_resampler;
175
176 int swri_realloc_audio(AudioData *a, int count);
177
178 void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
179 void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
180 void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
181 void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
182
183 int swri_rematrix_init(SwrContext *s);
184 void swri_rematrix_free(SwrContext *s);
185 int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
186 void swri_rematrix_init_x86(struct SwrContext *s);
187
188 void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
189 int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
190
191 void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
192 enum AVSampleFormat out_fmt,
193 enum AVSampleFormat in_fmt,
194 int channels);
195 void swri_audio_convert_init_arm(struct AudioConvert *ac,
196 enum AVSampleFormat out_fmt,
197 enum AVSampleFormat in_fmt,
198 int channels);
199 void swri_audio_convert_init_x86(struct AudioConvert *ac,
200 enum AVSampleFormat out_fmt,
201 enum AVSampleFormat in_fmt,
202 int channels);
203 #endif