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Diffstat (limited to 'command/audio_gain_analysis.h')
| -rw-r--r-- | command/audio_gain_analysis.h | 102 |
1 files changed, 102 insertions, 0 deletions
diff --git a/command/audio_gain_analysis.h b/command/audio_gain_analysis.h new file mode 100644 index 0000000..abed123 --- /dev/null +++ b/command/audio_gain_analysis.h @@ -0,0 +1,102 @@ +/* + * ReplayGainAnalysis - analyzes input samples and give the recommended dB change + * Copyright (C) 2001-2009 David Robinson and Glen Sawyer + * + * This library 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. + * + * This library 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 this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * concept and filter values by David Robinson (David@Robinson.org) + * -- blame him if you think the idea is flawed + * coding by Glen Sawyer (mp3gain@hotmail.com) 735 W 255 N, Orem, UT 84057-4505 USA + * -- blame him if you think this runs too slowly, or the coding is otherwise flawed + * + * For an explanation of the concepts and the basic algorithms involved, go to: + * http://www.replaygain.org/ + */ + +#ifndef GAIN_ANALYSIS_H +#define GAIN_ANALYSIS_H + +#include <stddef.h> + +#define GAIN_NOT_ENOUGH_SAMPLES -24601 +#define GAIN_ANALYSIS_ERROR 0 +#define GAIN_ANALYSIS_OK 1 + +#define INIT_GAIN_ANALYSIS_ERROR 0 +#define INIT_GAIN_ANALYSIS_OK 1 + +typedef double Float_t; // Type used for filtering + +class cMarkAdAudioGainAnalysis +{ +private: + typedef unsigned int Uint32_t; + typedef signed int Int32_t; + +#define YULE_ORDER 10 +#define BUTTER_ORDER 2 +#define YULE_FILTER filterYule +#define BUTTER_FILTER filterButter +#define RMS_PERCENTILE 0.95 // percentile which is louder than the proposed level +#define MAX_SAMP_FREQ 96000. // maximum allowed sample frequency [Hz] +#define RMS_WINDOW_TIME 0.050 // Time slice size [s] +#define STEPS_per_dB 100. // Table entries per dB +#define MAX_dB 120. // Table entries for 0...MAX_dB (normal max. values are 70...80 dB) + +#define MAX_ORDER (BUTTER_ORDER > YULE_ORDER ? BUTTER_ORDER : YULE_ORDER) +#define MAX_SAMPLES_PER_WINDOW (size_t) (MAX_SAMP_FREQ * RMS_WINDOW_TIME + 1) // max. Samples per Time slice +#define PINK_REF 64.82 //298640883795 // calibration value + + Float_t linprebuf [MAX_ORDER * 2]; + Float_t* linpre; // left input samples, with pre-buffer + Float_t lstepbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER]; + Float_t* lstep; // left "first step" (i.e. post first filter) samples + Float_t loutbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER]; + Float_t* lout; // left "out" (i.e. post second filter) samples + Float_t rinprebuf [MAX_ORDER * 2]; + Float_t* rinpre; // right input samples ... + Float_t rstepbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER]; + Float_t* rstep; + Float_t routbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER]; + Float_t* rout; + long sampleWindow; // number of samples required to reach number of milliseconds required for RMS window + long totsamp; + int gnum_samples; + double lsum; + double rsum; + int freqindex; + int first; + Uint32_t A [(size_t)(STEPS_per_dB * MAX_dB)]; + Uint32_t B [(size_t)(STEPS_per_dB * MAX_dB)]; + + static const Float_t ABButter[12][2*BUTTER_ORDER + 1]; + static const Float_t ABYule[12][2*YULE_ORDER + 1]; + + void filterButter (const Float_t* input, Float_t* output, size_t nSamples, const Float_t* kernel); + void filterYule (const Float_t* input, Float_t* output, size_t nSamples, const Float_t* kernel); + + int ResetSampleFrequency ( long samplefreq ); + Float_t analyzeResult ( Uint32_t* Array, size_t len ); +public: + int Init( long samplefreq ); + int AnalyzeSamples ( const Float_t* left_samples, const Float_t* right_samples, size_t num_samples, int num_channels ); + int AnalyzedSamples() + { + return (int) gnum_samples; + }; + Float_t GetGain(void); +}; + +#endif /* GAIN_ANALYSIS_H */ |