[deb_ffmpeg.git] / ffmpeg / libswresample / swresample_internal.h

/*
 * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
 *
 * This file is part of libswresample
 *
 * libswresample is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * libswresample is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with libswresample; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#ifndef SWR_INTERNAL_H
#define SWR_INTERNAL_H

#include "swresample.h"
#include "libavutil/channel_layout.h"
#include "config.h"

#define SWR_CH_MAX 32

#define SQRT3_2      1.22474487139158904909  /* sqrt(3/2) */

#define NS_TAPS 20

#if ARCH_X86_64
typedef int64_t integer;
#else
typedef int integer;
#endif

typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);

typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);

typedef struct AudioData{
    uint8_t *ch[SWR_CH_MAX];    ///< samples buffer per channel
    uint8_t *data;              ///< samples buffer
    int ch_count;               ///< number of channels
    int bps;                    ///< bytes per sample
    int count;                  ///< number of samples
    int planar;                 ///< 1 if planar audio, 0 otherwise
    enum AVSampleFormat fmt;    ///< sample format
} AudioData;

struct DitherContext {
    enum SwrDitherType method;
    int noise_pos;
    float scale;
    float noise_scale;                              ///< Noise scale
    int ns_taps;                                    ///< Noise shaping dither taps
    float ns_scale;                                 ///< Noise shaping dither scale
    float ns_scale_1;                               ///< Noise shaping dither scale^-1
    int ns_pos;                                     ///< Noise shaping dither position
    float ns_coeffs[NS_TAPS];                       ///< Noise shaping filter coefficients
    float ns_errors[SWR_CH_MAX][2*NS_TAPS];
    AudioData noise;                                ///< noise used for dithering
    AudioData temp;                                 ///< temporary storage when writing into the input buffer isn't possible
    int output_sample_bits;                         ///< the number of used output bits, needed to scale dither correctly
};

struct SwrContext {
    const AVClass *av_class;                        ///< AVClass used for AVOption and av_log()
    int log_level_offset;                           ///< logging level offset
    void *log_ctx;                                  ///< parent logging context
    enum AVSampleFormat  in_sample_fmt;             ///< input sample format
    enum AVSampleFormat int_sample_fmt;             ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
    enum AVSampleFormat out_sample_fmt;             ///< output sample format
    int64_t  in_ch_layout;                          ///< input channel layout
    int64_t out_ch_layout;                          ///< output channel layout
    int      in_sample_rate;                        ///< input sample rate
    int     out_sample_rate;                        ///< output sample rate
    int flags;                                      ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
    float slev;                                     ///< surround mixing level
    float clev;                                     ///< center mixing level
    float lfe_mix_level;                            ///< LFE mixing level
    float rematrix_volume;                          ///< rematrixing volume coefficient
    float rematrix_maxval;                          ///< maximum value for rematrixing output
    enum AVMatrixEncoding matrix_encoding;          /**< matrixed stereo encoding */
    const int *channel_map;                         ///< channel index (or -1 if muted channel) map
    int used_ch_count;                              ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
    enum SwrEngine engine;

    struct DitherContext dither;

    int filter_size;                                /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
    int phase_shift;                                /**< log2 of the number of entries in the resampling polyphase filterbank */
    int linear_interp;                              /**< if 1 then the resampling FIR filter will be linearly interpolated */
    double cutoff;                                  /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
    enum SwrFilterType filter_type;                 /**< swr resampling filter type */
    int kaiser_beta;                                /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
    double precision;                               /**< soxr resampling precision (in bits) */
    int cheby;                                      /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */

    float min_compensation;                         ///< swr minimum below which no compensation will happen
    float min_hard_compensation;                    ///< swr minimum below which no silence inject / sample drop will happen
    float soft_compensation_duration;               ///< swr duration over which soft compensation is applied
    float max_soft_compensation;                    ///< swr maximum soft compensation in seconds over soft_compensation_duration
    float async;                                    ///< swr simple 1 parameter async, similar to ffmpegs -async
    int64_t firstpts_in_samples;                    ///< swr first pts in samples

    int resample_first;                             ///< 1 if resampling must come first, 0 if rematrixing
    int rematrix;                                   ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
    int rematrix_custom;                            ///< flag to indicate that a custom matrix has been defined

    AudioData in;                                   ///< input audio data
    AudioData postin;                               ///< post-input audio data: used for rematrix/resample
    AudioData midbuf;                               ///< intermediate audio data (postin/preout)
    AudioData preout;                               ///< pre-output audio data: used for rematrix/resample
    AudioData out;                                  ///< converted output audio data
    AudioData in_buffer;                            ///< cached audio data (convert and resample purpose)
    AudioData silence;                              ///< temporary with silence
    AudioData drop_temp;                            ///< temporary used to discard output
    int in_buffer_index;                            ///< cached buffer position
    int in_buffer_count;                            ///< cached buffer length
    int resample_in_constraint;                     ///< 1 if the input end was reach before the output end, 0 otherwise
    int flushed;                                    ///< 1 if data is to be flushed and no further input is expected
    int64_t outpts;                                 ///< output PTS
    int64_t firstpts;                               ///< first PTS
    int drop_output;                                ///< number of output samples to drop

    struct AudioConvert *in_convert;                ///< input conversion context
    struct AudioConvert *out_convert;               ///< output conversion context
    struct AudioConvert *full_convert;              ///< full conversion context (single conversion for input and output)
    struct ResampleContext *resample;               ///< resampling context
    struct Resampler const *resampler;              ///< resampler virtual function table

    float matrix[SWR_CH_MAX][SWR_CH_MAX];           ///< floating point rematrixing coefficients
    uint8_t *native_matrix;
    uint8_t *native_one;
    uint8_t *native_simd_one;
    uint8_t *native_simd_matrix;
    int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX];       ///< 17.15 fixed point rematrixing coefficients
    uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1];    ///< Lists of input channels per output channel that have non zero rematrixing coefficients
    mix_1_1_func_type *mix_1_1_f;
    mix_1_1_func_type *mix_1_1_simd;

    mix_2_1_func_type *mix_2_1_f;
    mix_2_1_func_type *mix_2_1_simd;

    mix_any_func_type *mix_any_f;

    /* TODO: callbacks for ASM optimizations */
};

typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
                                    double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby);
typedef void    (* resample_free_func)(struct ResampleContext **c);
typedef int     (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
typedef int     (* resample_flush_func)(struct SwrContext *c);
typedef int     (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
typedef int     (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);

struct Resampler {
  resample_init_func            init;
  resample_free_func            free;
  multiple_resample_func        multiple_resample;
  resample_flush_func           flush;
  set_compensation_func         set_compensation;
  get_delay_func                get_delay;
  invert_initial_buffer_func    invert_initial_buffer;
};

extern struct Resampler const swri_resampler;

int swri_realloc_audio(AudioData *a, int count);

void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);

int swri_rematrix_init(SwrContext *s);
void swri_rematrix_free(SwrContext *s);
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
void swri_rematrix_init_x86(struct SwrContext *s);

void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);

void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
                                 enum AVSampleFormat out_fmt,
                                 enum AVSampleFormat in_fmt,
                                 int channels);
void swri_audio_convert_init_arm(struct AudioConvert *ac,
                                 enum AVSampleFormat out_fmt,
                                 enum AVSampleFormat in_fmt,
                                 int channels);
void swri_audio_convert_init_x86(struct AudioConvert *ac,
                                 enum AVSampleFormat out_fmt,
                                 enum AVSampleFormat in_fmt,
                                 int channels);
#endif
Commit	Line	Data
2ba45a60 DM	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