Added second pass processing (overlap, audio silence detection)

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 */