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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 SQRT3_2      1.22474487139158904909  /* sqrt(3/2) */
  29.  
  30. #define NS_TAPS 20
  31.  
  32. #if ARCH_X86_64
  33. typedef int64_t integer;
  34. #else
  35. typedef int integer;
  36. #endif
  37.  
  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);
  40.  
  41. typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
  42.  
  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
  51. } AudioData;
  52.  
  53. struct DitherContext {
  54.     enum SwrDitherType method;
  55.     int noise_pos;
  56.     float scale;
  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
  65.     AudioData temp;                                 ///< temporary storage when writing into the input buffer isnt possible
  66.     int output_sample_bits;                         ///< the number of used output bits, needed to scale dither correctly
  67. };
  68.  
  69. struct SwrContext {
  70.     const AVClass *av_class;                        ///< AVClass used for AVOption and av_log()
  71.     int log_level_offset;                           ///< logging level offset
  72.     void *log_ctx;                                  ///< parent logging context
  73.     enum AVSampleFormat  in_sample_fmt;             ///< input sample format
  74.     enum AVSampleFormat int_sample_fmt;             ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
  75.     enum AVSampleFormat out_sample_fmt;             ///< output sample format
  76.     int64_t  in_ch_layout;                          ///< input channel layout
  77.     int64_t out_ch_layout;                          ///< output channel layout
  78.     int      in_sample_rate;                        ///< input sample rate
  79.     int     out_sample_rate;                        ///< output sample rate
  80.     int flags;                                      ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
  81.     float slev;                                     ///< surround mixing level
  82.     float clev;                                     ///< center mixing level
  83.     float lfe_mix_level;                            ///< LFE mixing level
  84.     float rematrix_volume;                          ///< rematrixing volume coefficient
  85.     float rematrix_maxval;                          ///< maximum value for rematrixing output
  86.     enum AVMatrixEncoding matrix_encoding;          /**< matrixed stereo encoding */
  87.     const int *channel_map;                         ///< channel index (or -1 if muted channel) map
  88.     int used_ch_count;                              ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
  89.     enum SwrEngine engine;
  90.  
  91.     struct DitherContext dither;
  92.  
  93.     int filter_size;                                /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
  94.     int phase_shift;                                /**< log2 of the number of entries in the resampling polyphase filterbank */
  95.     int linear_interp;                              /**< if 1 then the resampling FIR filter will be linearly interpolated */
  96.     double cutoff;                                  /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
  97.     enum SwrFilterType filter_type;                 /**< swr resampling filter type */
  98.     int kaiser_beta;                                /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
  99.     double precision;                               /**< soxr resampling precision (in bits) */
  100.     int cheby;                                      /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
  101.  
  102.     float min_compensation;                         ///< swr minimum below which no compensation will happen
  103.     float min_hard_compensation;                    ///< swr minimum below which no silence inject / sample drop will happen
  104.     float soft_compensation_duration;               ///< swr duration over which soft compensation is applied
  105.     float max_soft_compensation;                    ///< swr maximum soft compensation in seconds over soft_compensation_duration
  106.     float async;                                    ///< swr simple 1 parameter async, similar to ffmpegs -async
  107.     int64_t firstpts_in_samples;                    ///< swr first pts in samples
  108.  
  109.     int resample_first;                             ///< 1 if resampling must come first, 0 if rematrixing
  110.     int rematrix;                                   ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
  111.     int rematrix_custom;                            ///< flag to indicate that a custom matrix has been defined
  112.  
  113.     AudioData in;                                   ///< input audio data
  114.     AudioData postin;                               ///< post-input audio data: used for rematrix/resample
  115.     AudioData midbuf;                               ///< intermediate audio data (postin/preout)
  116.     AudioData preout;                               ///< pre-output audio data: used for rematrix/resample
  117.     AudioData out;                                  ///< converted output audio data
  118.     AudioData in_buffer;                            ///< cached audio data (convert and resample purpose)
  119.     AudioData silence;                              ///< temporary with silence
  120.     AudioData drop_temp;                            ///< temporary used to discard output
  121.     int in_buffer_index;                            ///< cached buffer position
  122.     int in_buffer_count;                            ///< cached buffer length
  123.     int resample_in_constraint;                     ///< 1 if the input end was reach before the output end, 0 otherwise
  124.     int flushed;                                    ///< 1 if data is to be flushed and no further input is expected
  125.     int64_t outpts;                                 ///< output PTS
  126.     int64_t firstpts;                               ///< first PTS
  127.     int drop_output;                                ///< number of output samples to drop
  128.  
  129.     struct AudioConvert *in_convert;                ///< input conversion context
  130.     struct AudioConvert *out_convert;               ///< output conversion context
  131.     struct AudioConvert *full_convert;              ///< full conversion context (single conversion for input and output)
  132.     struct ResampleContext *resample;               ///< resampling context
  133.     struct Resampler const *resampler;              ///< resampler virtual function table
  134.  
  135.     float matrix[SWR_CH_MAX][SWR_CH_MAX];           ///< floating point rematrixing coefficients
  136.     uint8_t *native_matrix;
  137.     uint8_t *native_one;
  138.     uint8_t *native_simd_one;
  139.     uint8_t *native_simd_matrix;
  140.     int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX];       ///< 17.15 fixed point rematrixing coefficients
  141.     uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1];    ///< Lists of input channels per output channel that have non zero rematrixing coefficients
  142.     mix_1_1_func_type *mix_1_1_f;
  143.     mix_1_1_func_type *mix_1_1_simd;
  144.  
  145.     mix_2_1_func_type *mix_2_1_f;
  146.     mix_2_1_func_type *mix_2_1_simd;
  147.  
  148.     mix_any_func_type *mix_any_f;
  149.  
  150.     /* TODO: callbacks for ASM optimizations */
  151. };
  152.  
  153. typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
  154.                                     double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby);
  155. typedef void    (* resample_free_func)(struct ResampleContext **c);
  156. typedef int     (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
  157. typedef int     (* resample_flush_func)(struct SwrContext *c);
  158. typedef int     (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
  159. typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
  160.  
  161. struct Resampler {
  162.   resample_init_func            init;
  163.   resample_free_func            free;
  164.   multiple_resample_func        multiple_resample;
  165.   resample_flush_func           flush;
  166.   set_compensation_func         set_compensation;
  167.   get_delay_func                get_delay;
  168. };
  169.  
  170. extern struct Resampler const swri_resampler;
  171.  
  172. int swri_realloc_audio(AudioData *a, int count);
  173. 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);
  174. 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);
  175. int swri_resample_float(struct ResampleContext *c, float   *dst, const float   *src, int *consumed, int src_size, int dst_size, int update_ctx);
  176. int swri_resample_double(struct ResampleContext *c,double  *dst, const double  *src, int *consumed, int src_size, int dst_size, int update_ctx);
  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_arm(struct AudioConvert *ac,
  192.                                  enum AVSampleFormat out_fmt,
  193.                                  enum AVSampleFormat in_fmt,
  194.                                  int channels);
  195. void swri_audio_convert_init_x86(struct AudioConvert *ac,
  196.                                  enum AVSampleFormat out_fmt,
  197.                                  enum AVSampleFormat in_fmt,
  198.                                  int channels);
  199. #endif
  200.