initial commit

pjproject-2.11.1/third_party/resample/src/libresample_dll.c

@@ -0,0 +1,12 @@
+// libresample_dll.cpp : Defines the entry point for the DLL application.
+//
+#include <windows.h>
+
+BOOL APIENTRY DllMain( HANDLE hModule, 
+                       DWORD  ul_reason_for_call, 
+                       LPVOID lpReserved
+					 )
+{
+    return TRUE;
+}
+

pjproject-2.11.1/third_party/resample/src/resample.h

@@ -0,0 +1,67 @@
+/*
+ * The configuration constants below govern
+ * the number of bits in the input sample and filter coefficients, the 
+ * number of bits to the right of the binary-point for fixed-point math, etc.
+ *
+ */
+
+/* Conversion constants */
+#define Nhc       8
+#define Na        7
+#define Np       (Nhc+Na)
+#define Npc      (1<<Nhc)
+#define Amask    ((1<<Na)-1)
+#define Pmask    ((1<<Np)-1)
+#define Nh       16
+#define Nb       16
+#define Nhxn     14
+#define Nhg      (Nh-Nhxn)
+#define NLpScl   13
+
+/* Description of constants:
+ *
+ * Npc - is the number of look-up values available for the lowpass filter
+ *    between the beginning of its impulse response and the "cutoff time"
+ *    of the filter.  The cutoff time is defined as the reciprocal of the
+ *    lowpass-filter cut off frequence in Hz.  For example, if the
+ *    lowpass filter were a sinc function, Npc would be the index of the
+ *    impulse-response lookup-table corresponding to the first zero-
+ *    crossing of the sinc function.  (The inverse first zero-crossing
+ *    time of a sinc function equals its nominal cutoff frequency in Hz.)
+ *    Npc must be a power of 2 due to the details of the current
+ *    implementation. The default value of 512 is sufficiently high that
+ *    using linear interpolation to fill in between the table entries
+ *    gives approximately 16-bit accuracy in filter coefficients.
+ *
+ * Nhc - is log base 2 of Npc.
+ *
+ * Na - is the number of bits devoted to linear interpolation of the
+ *    filter coefficients.
+ *
+ * Np - is Na + Nhc, the number of bits to the right of the binary point
+ *    in the integer "time" variable. To the left of the point, it indexes
+ *    the input array (X), and to the right, it is interpreted as a number
+ *    between 0 and 1 sample of the input X.  Np must be less than 16 in
+ *    this implementation.
+ *
+ * Nh - is the number of bits in the filter coefficients. The sum of Nh and
+ *    the number of bits in the input data (typically 16) cannot exceed 32.
+ *    Thus Nh should be 16.  The largest filter coefficient should nearly
+ *    fill 16 bits (32767).
+ *
+ * Nb - is the number of bits in the input data. The sum of Nb and Nh cannot
+ *    exceed 32.
+ *
+ * Nhxn - is the number of bits to right shift after multiplying each input
+ *    sample times a filter coefficient. It can be as great as Nh and as
+ *    small as 0. Nhxn = Nh-2 gives 2 guard bits in the multiply-add
+ *    accumulation.  If Nhxn=0, the accumulation will soon overflow 32 bits.
+ *
+ * Nhg - is the number of guard bits in mpy-add accumulation (equal to Nh-Nhxn)
+ *
+ * NLpScl - is the number of bits allocated to the unity-gain normalization
+ *    factor.  The output of the lowpass filter is multiplied by LpScl and
+ *    then right-shifted NLpScl bits. To avoid overflow, we must have 
+ *    Nb+Nhg+NLpScl < 32.
+ */
+

pjproject-2.11.1/third_party/resample/src/resamplesubs.c

@@ -0,0 +1,377 @@
+/* $Id$ */
+/*
+ * Digital Audio Resampling Home Page located at
+ * http://www-ccrma.stanford.edu/~jos/resample/.
+ *
+ * SOFTWARE FOR SAMPLING-RATE CONVERSION AND FIR DIGITAL FILTER DESIGN
+ *
+ * Snippet from the resample.1 man page:
+ *
+ * HISTORY
+ *
+ * The first version of this software was written by Julius O. Smith III
+ * <jos@ccrma.stanford.edu> at CCRMA <http://www-ccrma.stanford.edu> in
+ * 1981.  It was called SRCONV and was written in SAIL for PDP-10
+ * compatible machines.  The algorithm was first published in
+ *
+ * Smith, Julius O. and Phil Gossett. ``A Flexible Sampling-Rate
+ * Conversion Method,'' Proceedings (2): 19.4.1-19.4.4, IEEE Conference
+ * on Acoustics, Speech, and Signal Processing, San Diego, March 1984.
+ *
+ * An expanded tutorial based on this paper is available at the Digital
+ * Audio Resampling Home Page given above.
+ *
+ * Circa 1988, the SRCONV program was translated from SAIL to C by
+ * Christopher Lee Fraley working with Roger Dannenberg at CMU.
+ *
+ * Since then, the C version has been maintained by jos.
+ *
+ * Sndlib support was added 6/99 by John Gibson <jgg9c@virginia.edu>.
+ *
+ * The resample program is free software distributed in accordance
+ * with the Lesser GNU Public License (LGPL).  There is NO warranty; not
+ * even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ */
+
+/* PJMEDIA modification:
+ *  - remove resample(), just use SrcUp, SrcUD, and SrcLinear directly.
+ *  - move FilterUp() and FilterUD() from filterkit.c
+ *  - move stddefs.h and resample.h to this file.
+ *  - const correctness.
+ */
+
+#include <resamplesubs.h>
+#include "config.h"
+#include "stddefs.h"
+#include "resample.h"
+
+
+#ifdef _MSC_VER
+#   pragma warning(push, 3)
+//#   pragma warning(disable: 4245)   // Conversion from uint to ushort
+#   pragma warning(disable: 4244)   // Conversion from double to uint
+#   pragma warning(disable: 4146)   // unary minus operator applied to unsigned type, result still unsigned
+#   pragma warning(disable: 4761)   // integral size mismatch in argument; conversion supplied
+#endif
+
+#if defined(RESAMPLE_HAS_SMALL_FILTER) && RESAMPLE_HAS_SMALL_FILTER!=0
+#   include "smallfilter.h"
+#else
+#   define SMALL_FILTER_NMULT	0
+#   define SMALL_FILTER_SCALE	0
+#   define SMALL_FILTER_NWING	0
+#   define SMALL_FILTER_IMP	NULL
+#   define SMALL_FILTER_IMPD	NULL
+#endif
+
+#if defined(RESAMPLE_HAS_LARGE_FILTER) && RESAMPLE_HAS_LARGE_FILTER!=0
+#   include "largefilter.h"
+#else
+#   define LARGE_FILTER_NMULT	0
+#   define LARGE_FILTER_SCALE	0
+#   define LARGE_FILTER_NWING	0
+#   define LARGE_FILTER_IMP	NULL
+#   define LARGE_FILTER_IMPD	NULL
+#endif
+
+
+#undef INLINE
+#define INLINE
+#define HAVE_FILTER 0
+
+#ifndef NULL
+#   define NULL	0
+#endif
+
+
+static INLINE RES_HWORD WordToHword(RES_WORD v, int scl)
+{
+    RES_HWORD out;
+    RES_WORD llsb = (1<<(scl-1));
+    v += llsb;		/* round */
+    v >>= scl;
+    if (v>MAX_HWORD) {
+	v = MAX_HWORD;
+    } else if (v < MIN_HWORD) {
+	v = MIN_HWORD;
+    }
+    out = (RES_HWORD) v;
+    return out;
+}
+
+/* Sampling rate conversion using linear interpolation for maximum speed.
+ */
+static int
+  SrcLinear(const RES_HWORD X[], RES_HWORD Y[], double pFactor, RES_UHWORD nx)
+{
+    RES_HWORD iconst;
+    RES_UWORD time = 0;
+    const RES_HWORD *xp;
+    RES_HWORD *Ystart, *Yend;
+    RES_WORD v,x1,x2;
+
+    double dt;                  /* Step through input signal */
+    RES_UWORD dtb;                  /* Fixed-point version of Dt */
+    //RES_UWORD endTime;              /* When time reaches EndTime, return to user */
+
+    dt = 1.0/pFactor;            /* Output sampling period */
+    dtb = dt*(1<<Np) + 0.5;     /* Fixed-point representation */
+
+    Ystart = Y;
+    Yend = Ystart + (unsigned)(nx * pFactor + 0.5);
+    //endTime = time + (1<<Np)*(RES_WORD)nx;
+
+    // Integer round down in dtb calculation may cause (endTime % dtb > 0),
+    // so it may cause resample write pass the output buffer (Y >= Yend).
+    // while (time < endTime)
+    while (Y < Yend)
+    {
+	iconst = (time) & Pmask;
+	xp = &X[(time)>>Np];      /* Ptr to current input sample */
+	x1 = *xp++;
+	x2 = *xp;
+	x1 *= ((1<<Np)-iconst);
+	x2 *= iconst;
+	v = x1 + x2;
+	*Y++ = WordToHword(v,Np);   /* Deposit output */
+	time += dtb;		    /* Move to next sample by time increment */
+    }
+    return (Y - Ystart);            /* Return number of output samples */
+}
+
+static RES_WORD FilterUp(const RES_HWORD Imp[], const RES_HWORD ImpD[],
+		     RES_UHWORD Nwing, RES_BOOL Interp,
+		     const RES_HWORD *Xp, RES_HWORD Ph, RES_HWORD Inc)
+{
+    const RES_HWORD *Hp;
+    const RES_HWORD *Hdp = NULL;
+    const RES_HWORD *End;
+    RES_HWORD a = 0;
+    RES_WORD v, t;
+
+    v=0;
+    Hp = &Imp[Ph>>Na];
+    End = &Imp[Nwing];
+    if (Interp) {
+	Hdp = &ImpD[Ph>>Na];
+	a = Ph & Amask;
+    }
+    if (Inc == 1)		/* If doing right wing...              */
+    {				/* ...drop extra coeff, so when Ph is  */
+	End--;			/*    0.5, we don't do too many mult's */
+	if (Ph == 0)		/* If the phase is zero...           */
+	{			/* ...then we've already skipped the */
+	    Hp += Npc;		/*    first sample, so we must also  */
+	    Hdp += Npc;		/*    skip ahead in Imp[] and ImpD[] */
+	}
+    }
+    if (Interp)
+      while (Hp < End) {
+	  t = *Hp;		/* Get filter coeff */
+	  t += (((RES_WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
+	  Hdp += Npc;		/* Filter coeff differences step */
+	  t *= *Xp;		/* Mult coeff by input sample */
+	  if (t & (1<<(Nhxn-1)))  /* Round, if needed */
+	    t += (1<<(Nhxn-1));
+	  t >>= Nhxn;		/* Leave some guard bits, but come back some */
+	  v += t;			/* The filter output */
+	  Hp += Npc;		/* Filter coeff step */
+
+	  Xp += Inc;		/* Input signal step. NO CHECK ON BOUNDS */
+      }
+    else
+      while (Hp < End) {
+	  t = *Hp;		/* Get filter coeff */
+	  t *= *Xp;		/* Mult coeff by input sample */
+	  if (t & (1<<(Nhxn-1)))  /* Round, if needed */
+	    t += (1<<(Nhxn-1));
+	  t >>= Nhxn;		/* Leave some guard bits, but come back some */
+	  v += t;			/* The filter output */
+	  Hp += Npc;		/* Filter coeff step */
+	  Xp += Inc;		/* Input signal step. NO CHECK ON BOUNDS */
+      }
+    return(v);
+}
+
+
+static RES_WORD FilterUD(const RES_HWORD Imp[], const RES_HWORD ImpD[],
+		     RES_UHWORD Nwing, RES_BOOL Interp,
+		     const RES_HWORD *Xp, RES_HWORD Ph, RES_HWORD Inc, RES_UHWORD dhb)
+{
+    RES_HWORD a;
+    const RES_HWORD *Hp, *Hdp, *End;
+    RES_WORD v, t;
+    RES_UWORD Ho;
+
+    v=0;
+    Ho = (Ph*(RES_UWORD)dhb)>>Np;
+    End = &Imp[Nwing];
+    if (Inc == 1)		/* If doing right wing...              */
+    {				/* ...drop extra coeff, so when Ph is  */
+	End--;			/*    0.5, we don't do too many mult's */
+	if (Ph == 0)		/* If the phase is zero...           */
+	  Ho += dhb;		/* ...then we've already skipped the */
+    }				/*    first sample, so we must also  */
+				/*    skip ahead in Imp[] and ImpD[] */
+    if (Interp)
+      while ((Hp = &Imp[Ho>>Na]) < End) {
+	  t = *Hp;		/* Get IR sample */
+	  Hdp = &ImpD[Ho>>Na];  /* get interp (lower Na) bits from diff table*/
+	  a = Ho & Amask;	/* a is logically between 0 and 1 */
+	  t += (((RES_WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
+	  t *= *Xp;		/* Mult coeff by input sample */
+	  if (t & 1<<(Nhxn-1))	/* Round, if needed */
+	    t += 1<<(Nhxn-1);
+	  t >>= Nhxn;		/* Leave some guard bits, but come back some */
+	  v += t;			/* The filter output */
+	  Ho += dhb;		/* IR step */
+	  Xp += Inc;		/* Input signal step. NO CHECK ON BOUNDS */
+      }
+    else
+      while ((Hp = &Imp[Ho>>Na]) < End) {
+	  t = *Hp;		/* Get IR sample */
+	  t *= *Xp;		/* Mult coeff by input sample */
+	  if (t & 1<<(Nhxn-1))	/* Round, if needed */
+	    t += 1<<(Nhxn-1);
+	  t >>= Nhxn;		/* Leave some guard bits, but come back some */
+	  v += t;			/* The filter output */
+	  Ho += dhb;		/* IR step */
+	  Xp += Inc;		/* Input signal step. NO CHECK ON BOUNDS */
+      }
+    return(v);
+}
+
+/* Sampling rate up-conversion only subroutine;
+ * Slightly faster than down-conversion;
+ */
+static int SrcUp(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
+		 RES_UHWORD nx, RES_UHWORD pNwing, RES_UHWORD pLpScl,
+		 const RES_HWORD pImp[], const RES_HWORD pImpD[], RES_BOOL Interp)
+{
+    const RES_HWORD *xp;
+    RES_HWORD *Ystart, *Yend;
+    RES_WORD v;
+
+    double dt;                  /* Step through input signal */
+    RES_UWORD dtb;                  /* Fixed-point version of Dt */
+    RES_UWORD time = 0;
+    //RES_UWORD endTime;              /* When time reaches EndTime, return to user */
+
+    dt = 1.0/pFactor;            /* Output sampling period */
+    dtb = dt*(1<<Np) + 0.5;     /* Fixed-point representation */
+
+    Ystart = Y;
+    Yend = Ystart + (unsigned)(nx * pFactor + 0.5);
+    //endTime = time + (1<<Np)*(RES_WORD)nx;
+
+    // Integer round down in dtb calculation may cause (endTime % dtb > 0),
+    // so it may cause resample write pass the output buffer (Y >= Yend).
+    // while (time < endTime)
+    while (Y < Yend)
+    {
+	xp = &X[time>>Np];      /* Ptr to current input sample */
+	/* Perform left-wing inner product */
+	v = 0;
+	v = FilterUp(pImp, pImpD, pNwing, Interp, xp, (RES_HWORD)(time&Pmask),-1);
+
+	/* Perform right-wing inner product */
+	v += FilterUp(pImp, pImpD, pNwing, Interp, xp+1,  (RES_HWORD)((-time)&Pmask),1);
+
+	v >>= Nhg;		/* Make guard bits */
+	v *= pLpScl;		/* Normalize for unity filter gain */
+	*Y++ = WordToHword(v,NLpScl);   /* strip guard bits, deposit output */
+	time += dtb;		/* Move to next sample by time increment */
+    }
+    return (Y - Ystart);        /* Return the number of output samples */
+}
+
+
+/* Sampling rate conversion subroutine */
+
+static int SrcUD(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
+		 RES_UHWORD nx, RES_UHWORD pNwing, RES_UHWORD pLpScl,
+		 const RES_HWORD pImp[], const RES_HWORD pImpD[], RES_BOOL Interp)
+{
+    const RES_HWORD *xp;
+    RES_HWORD *Ystart, *Yend;
+    RES_WORD v;
+
+    double dh;                  /* Step through filter impulse response */
+    double dt;                  /* Step through input signal */
+    RES_UWORD time = 0;
+    //RES_UWORD endTime;          /* When time reaches EndTime, return to user */
+    RES_UWORD dhb, dtb;         /* Fixed-point versions of Dh,Dt */
+
+    dt = 1.0/pFactor;            /* Output sampling period */
+    dtb = dt*(1<<Np) + 0.5;     /* Fixed-point representation */
+
+    dh = MIN(Npc, pFactor*Npc);  /* Filter sampling period */
+    dhb = dh*(1<<Na) + 0.5;     /* Fixed-point representation */
+
+    Ystart = Y;
+    Yend = Ystart + (unsigned)(nx * pFactor + 0.5);
+    //endTime = time + (1<<Np)*(RES_WORD)nx;
+
+    // Integer round down in dtb calculation may cause (endTime % dtb > 0),
+    // so it may cause resample write pass the output buffer (Y >= Yend).
+    // while (time < endTime)
+    while (Y < Yend)
+    {
+	xp = &X[time>>Np];	/* Ptr to current input sample */
+	v = FilterUD(pImp, pImpD, pNwing, Interp, xp, (RES_HWORD)(time&Pmask),
+		     -1, dhb);	/* Perform left-wing inner product */
+	v += FilterUD(pImp, pImpD, pNwing, Interp, xp+1, (RES_HWORD)((-time)&Pmask),
+		      1, dhb);	/* Perform right-wing inner product */
+	v >>= Nhg;		/* Make guard bits */
+	v *= pLpScl;		/* Normalize for unity filter gain */
+	*Y++ = WordToHword(v,NLpScl);   /* strip guard bits, deposit output */
+	time += dtb;		/* Move to next sample by time increment */
+    }
+    return (Y - Ystart);        /* Return the number of output samples */
+}
+
+
+DECL(int) res_SrcLinear(const RES_HWORD X[], RES_HWORD Y[],
+		        double pFactor, RES_UHWORD nx)
+{
+    return SrcLinear(X, Y, pFactor, nx);
+}
+
+DECL(int) res_Resample(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
+		       RES_UHWORD nx, RES_BOOL LargeF, RES_BOOL Interp)
+{
+    if (pFactor >= 1) {
+
+	if (LargeF)
+	    return SrcUp(X, Y, pFactor, nx,
+			 LARGE_FILTER_NWING, LARGE_FILTER_SCALE,
+			 LARGE_FILTER_IMP, LARGE_FILTER_IMPD, Interp);
+	else
+	    return SrcUp(X, Y, pFactor, nx,
+			 SMALL_FILTER_NWING, SMALL_FILTER_SCALE,
+			 SMALL_FILTER_IMP, SMALL_FILTER_IMPD, Interp);
+
+    } else {
+
+	if (LargeF)
+	    return SrcUD(X, Y, pFactor, nx,
+			 LARGE_FILTER_NWING, LARGE_FILTER_SCALE * pFactor + 0.5,
+			 LARGE_FILTER_IMP, LARGE_FILTER_IMPD, Interp);
+	else
+	    return SrcUD(X, Y, pFactor, nx,
+			 SMALL_FILTER_NWING, SMALL_FILTER_SCALE * pFactor + 0.5,
+			 SMALL_FILTER_IMP, SMALL_FILTER_IMPD, Interp);
+
+    }
+}
+
+DECL(int) res_GetXOFF(double pFactor, RES_BOOL LargeF)
+{
+    if (LargeF)
+	return (LARGE_FILTER_NMULT + 1) / 2.0  *
+	        MAX(1.0, 1.0/pFactor);
+    else
+	return (SMALL_FILTER_NMULT + 1) / 2.0  *
+		MAX(1.0, 1.0/pFactor);
+}
+

pjproject-2.11.1/third_party/resample/src/stddefs.h

@@ -0,0 +1,29 @@
+#ifndef PI
+#define PI (3.14159265358979232846)
+#endif
+
+#ifndef PI2
+#define PI2 (6.28318530717958465692)
+#endif
+
+#define D2R (0.01745329348)          /* (2*pi)/360 */
+#define R2D (57.29577951)            /* 360/(2*pi) */
+
+#ifndef MAX
+#define MAX(x,y) ((x)>(y) ?(x):(y))
+#endif
+#ifndef MIN
+#define MIN(x,y) ((x)<(y) ?(x):(y))
+#endif
+
+#ifndef ABS
+#define ABS(x)   ((x)<0   ?(-(x)):(x))
+#endif
+
+#ifndef SGN
+#define SGN(x)   ((x)<0   ?(-1):((x)==0?(0):(1)))
+#endif
+
+#define MAX_HWORD (32767)
+#define MIN_HWORD (-32768)
+