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authorJochen Dolze <vdr@dolze.de>2011-01-29 15:58:36 +0100
committerJochen Dolze <vdr@dolze.de>2011-01-29 15:58:36 +0100
commit70d113055698c6b73c8ed13af8a9e2f3b38ab1f0 (patch)
treed5264ada5b3a1366ca2404468dd57c6ea62a6349 /command/audio_gain_analysis.h
parent9e964370ba635f57df44a96506fc4bf633004a86 (diff)
downloadvdr-plugin-markad-70d113055698c6b73c8ed13af8a9e2f3b38ab1f0.tar.gz
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Rewrite of demux/marks recognition (still incomplete)
Diffstat (limited to 'command/audio_gain_analysis.h')
-rw-r--r--command/audio_gain_analysis.h102
1 files changed, 0 insertions, 102 deletions
diff --git a/command/audio_gain_analysis.h b/command/audio_gain_analysis.h
deleted file mode 100644
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--- a/command/audio_gain_analysis.h
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * 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 */