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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 64
  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. typedef int64_t (* get_out_samples_func)(struct SwrContext *s, int in_samples);
  80.  
  81. struct Resampler {
  82.   resample_init_func            init;
  83.   resample_free_func            free;
  84.   multiple_resample_func        multiple_resample;
  85.   resample_flush_func           flush;
  86.   set_compensation_func         set_compensation;
  87.   get_delay_func                get_delay;
  88.   invert_initial_buffer_func    invert_initial_buffer;
  89.   get_out_samples_func          get_out_samples;
  90. };
  91.  
  92. extern struct Resampler const swri_resampler;
  93. extern struct Resampler const swri_soxr_resampler;
  94.  
  95. struct SwrContext {
  96.     const AVClass *av_class;                        ///< AVClass used for AVOption and av_log()
  97.     int log_level_offset;                           ///< logging level offset
  98.     void *log_ctx;                                  ///< parent logging context
  99.     enum AVSampleFormat  in_sample_fmt;             ///< input sample format
  100.     enum AVSampleFormat int_sample_fmt;             ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
  101.     enum AVSampleFormat out_sample_fmt;             ///< output sample format
  102.     int64_t  in_ch_layout;                          ///< input channel layout
  103.     int64_t out_ch_layout;                          ///< output channel layout
  104.     int      in_sample_rate;                        ///< input sample rate
  105.     int     out_sample_rate;                        ///< output sample rate
  106.     int flags;                                      ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
  107.     float slev;                                     ///< surround mixing level
  108.     float clev;                                     ///< center mixing level
  109.     float lfe_mix_level;                            ///< LFE mixing level
  110.     float rematrix_volume;                          ///< rematrixing volume coefficient
  111.     float rematrix_maxval;                          ///< maximum value for rematrixing output
  112.     int matrix_encoding;                            /**< matrixed stereo encoding */
  113.     const int *channel_map;                         ///< channel index (or -1 if muted channel) map
  114.     int used_ch_count;                              ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
  115.     int engine;
  116.  
  117.     int user_in_ch_count;                           ///< User set input channel count
  118.     int user_out_ch_count;                          ///< User set output channel count
  119.     int user_used_ch_count;                         ///< User set used channel count
  120.     int64_t user_in_ch_layout;                      ///< User set input channel layout
  121.     int64_t user_out_ch_layout;                     ///< User set output channel layout
  122.     enum AVSampleFormat user_int_sample_fmt;        ///< User set internal sample format
  123.  
  124.     struct DitherContext dither;
  125.  
  126.     int filter_size;                                /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
  127.     int phase_shift;                                /**< log2 of the number of entries in the resampling polyphase filterbank */
  128.     int linear_interp;                              /**< if 1 then the resampling FIR filter will be linearly interpolated */
  129.     double cutoff;                                  /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
  130.     int filter_type;                                /**< swr resampling filter type */
  131.     int kaiser_beta;                                /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
  132.     double precision;                               /**< soxr resampling precision (in bits) */
  133.     int cheby;                                      /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
  134.  
  135.     float min_compensation;                         ///< swr minimum below which no compensation will happen
  136.     float min_hard_compensation;                    ///< swr minimum below which no silence inject / sample drop will happen
  137.     float soft_compensation_duration;               ///< swr duration over which soft compensation is applied
  138.     float max_soft_compensation;                    ///< swr maximum soft compensation in seconds over soft_compensation_duration
  139.     float async;                                    ///< swr simple 1 parameter async, similar to ffmpegs -async
  140.     int64_t firstpts_in_samples;                    ///< swr first pts in samples
  141.  
  142.     int resample_first;                             ///< 1 if resampling must come first, 0 if rematrixing
  143.     int rematrix;                                   ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
  144.     int rematrix_custom;                            ///< flag to indicate that a custom matrix has been defined
  145.  
  146.     AudioData in;                                   ///< input audio data
  147.     AudioData postin;                               ///< post-input audio data: used for rematrix/resample
  148.     AudioData midbuf;                               ///< intermediate audio data (postin/preout)
  149.     AudioData preout;                               ///< pre-output audio data: used for rematrix/resample
  150.     AudioData out;                                  ///< converted output audio data
  151.     AudioData in_buffer;                            ///< cached audio data (convert and resample purpose)
  152.     AudioData silence;                              ///< temporary with silence
  153.     AudioData drop_temp;                            ///< temporary used to discard output
  154.     int in_buffer_index;                            ///< cached buffer position
  155.     int in_buffer_count;                            ///< cached buffer length
  156.     int resample_in_constraint;                     ///< 1 if the input end was reach before the output end, 0 otherwise
  157.     int flushed;                                    ///< 1 if data is to be flushed and no further input is expected
  158.     int64_t outpts;                                 ///< output PTS
  159.     int64_t firstpts;                               ///< first PTS
  160.     int drop_output;                                ///< number of output samples to drop
  161.     double delayed_samples_fixup;                   ///< soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
  162.  
  163.     struct AudioConvert *in_convert;                ///< input conversion context
  164.     struct AudioConvert *out_convert;               ///< output conversion context
  165.     struct AudioConvert *full_convert;              ///< full conversion context (single conversion for input and output)
  166.     struct ResampleContext *resample;               ///< resampling context
  167.     struct Resampler const *resampler;              ///< resampler virtual function table
  168.  
  169.     float matrix[SWR_CH_MAX][SWR_CH_MAX];           ///< floating point rematrixing coefficients
  170.     uint8_t *native_matrix;
  171.     uint8_t *native_one;
  172.     uint8_t *native_simd_one;
  173.     uint8_t *native_simd_matrix;
  174.     int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX];       ///< 17.15 fixed point rematrixing coefficients
  175.     uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1];    ///< Lists of input channels per output channel that have non zero rematrixing coefficients
  176.     mix_1_1_func_type *mix_1_1_f;
  177.     mix_1_1_func_type *mix_1_1_simd;
  178.  
  179.     mix_2_1_func_type *mix_2_1_f;
  180.     mix_2_1_func_type *mix_2_1_simd;
  181.  
  182.     mix_any_func_type *mix_any_f;
  183.  
  184.     /* TODO: callbacks for ASM optimizations */
  185. };
  186.  
  187. int swri_realloc_audio(AudioData *a, int count);
  188.  
  189. void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
  190. void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
  191. void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
  192. void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
  193.  
  194. int swri_rematrix_init(SwrContext *s);
  195. void swri_rematrix_free(SwrContext *s);
  196. int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
  197. int swri_rematrix_init_x86(struct SwrContext *s);
  198.  
  199. int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
  200. int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
  201.  
  202. void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
  203.                                  enum AVSampleFormat out_fmt,
  204.                                  enum AVSampleFormat in_fmt,
  205.                                  int channels);
  206. void swri_audio_convert_init_arm(struct AudioConvert *ac,
  207.                                  enum AVSampleFormat out_fmt,
  208.                                  enum AVSampleFormat in_fmt,
  209.                                  int channels);
  210. void swri_audio_convert_init_x86(struct AudioConvert *ac,
  211.                                  enum AVSampleFormat out_fmt,
  212.                                  enum AVSampleFormat in_fmt,
  213.                                  int channels);
  214.  
  215. #endif
  216.