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pjproject-2.11.1/third_party/resample/COPYING
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GNU LESSER GENERAL PUBLIC LICENSE
Version 2.1, February 1999
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Everyone is permitted to copy and distribute verbatim copies
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pjproject-2.11.1/third_party/resample/README.resample
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README file for resample-1.x.tgz from the
Digital Audio Resampling Home Page located at
http://ccrma.stanford.edu/~jos/resample/.
SOFTWARE FOR SAMPLING-RATE CONVERSION AND FIR DIGITAL FILTER DESIGN
For installation instructions, read the INSTALL file in this directory.
The resample program "resamples" a soundfile to change its sampling
rate. For example, it can be used to convert the sampling rate from
48 kHz (used by DAT machines) to 44.1 kHz (the standard sampling rate
for Compact Discs). The command line for this operation would look
something like
resample -by 0.91875 dat.snd cd.snd
or, more simply,
resample -to 44100 dat.snd cd.snd
Any reasonable sampling rate can be converted to any other.
The windowfilter program designs Finite-Impulse-Response (FIR) digital
filters by the so-called "window method." In this method, the ideal
impulse response (a sinc function) is "windowed" by a Kaiser window (a
popular window used in spectrum analysis).
The resample program uses 32-bit fixed-point arithmetic: 16-bits data
and 16-bits coefficients. The input soundfile must be 16-bit mono or
stereo (interleaved) audio data.
SNDLIB
The program uses elements of Bill Schottstaedt's sndlib sound file
library. This means resample can read many different kinds of sound
file header (AIFF, WAV, NeXT, IRCAM, etc.).
The sndlib files used by resample are included in this directory to
ensure stability. The latest version of sndlib should be available as
ftp://ccrma-ftp.stanford.edu/pub/Lisp/sndlib.tar.gz
See sndlib.html in the sndlib distribution for documentation of SNDLIB.
CONTENTS of ./src directory
resample.c Sampling-rate conversion program.
resample.1 Manual page for resample. Try "nroff -man resample.1".
resamplesubs.c Subroutines used by resample.
resample.h Configuration constants for the sampling rate converter.
stdefs.h Machine-dependent definitions, useful constants and macros.
windowfilter.c Program for designing FIR digital filters used by resample.
windowfilter.1 Manual page for windowfilter.
filterkit.c Library for filter design, application, and file management.
filterkit.h Declarations (procedure prototypes) for the filterkit library.
README This file.
README.deemph A word about deemphasis filtering.
LGPL GNU Lesser General Public License (LGPL)
SNDLIB files:
io.c
audio.c
headers.c
sound.c
sndlib.h
sndlib-strings.h
COPYING
SNDLIB files are Copyright 2000 by Bill Schottstaedt <bil@ccrma.stanford.edu>.
The remaining files in this package, unless otherwise noted, are
Copyright 1994-2006 by Julius O. Smith III <jos@ccrma.stanford.edu>,
all rights reserved. Permission to use and copy is granted subject to
the terms of the "GNU Lesser General Public License" (LGPL) as
published by the Free Software Foundation; either version 2.1 of the
License, or any later version. In addition, we request that a copy of
any modified files be sent by email to jos@ccrma.stanford.edu so that
we may incorporate them into the CCRMA 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.
FILTERKIT CONTENTS
LpFilter() - Calculates the filter coeffs for a Kaiser-windowed
low-pass filter with a given roll-off frequency. These
coeffs are stored into a array of doubles.
writeFilter() - Writes a filter to a file.
makeFilter() - A section of the original SAIL program. Calls
LpFilter() to create a filter, then scales the double
coeffs into a array of half words.
readFilter() - Reads a filter from a file.
FilterUp() - Applies a filter to a given sample when up-converting.
FilterUD() - Applies a filter to a given sample when up- or down-
converting. Both are repoductions of the original SAIL
program.
initZerox() - Initialization routine for the zerox() function. Must
be called before zerox() is called. This routine loads
the correct filter so zerox() can use it.
zerox() - Given a pointer into a sample, finds a zero-crossing on the
interval [pointer-1:pointer+2] by iteration.
Query() - Ask the user for a yes/no question with prompt, default,
and optional help.
GetUShort() - Ask the user for a unsigned short with prompt, default,
and optional help.
GetDouble() - Ask the user for a double with prompt, default, and
optional help.
GetString() - Ask the user for a string with prompt, default, and
optional help.
FILTER FILE FORMAT
File Name: "F" Nmult "T" Nhc ".filter"
example: "F13T8.filter" and "F27T8.filter"
Structure of File:
"ScaleFactor" LpScl
"Length" Nwing
"Coeffs:"
Imp[0]
Imp[1]
:
Imp[Nwing-1]
"Differences:"
ImpD[0]
ImpD[1]
:
ImpD[Nwing-1]
EOF
where: Something enclosed in "" indicates specific characters in the file.
Nmult, Nwing, Imp[], and ImpD[] are variables (HWORD)
Npc is a conversion constant.
EOF is the end of the file.
See writeFilter() and readFilter() in "filterkit.c" for more details.
pjproject-2.11.1/third_party/resample/include/resamplesubs.h
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#ifndef __RESAMPLESUBS_H__
#define __RESAMPLESUBS_H__
typedef char RES_BOOL;
typedef short RES_HWORD;
typedef int RES_WORD;
typedef unsigned short RES_UHWORD;
typedef unsigned int RES_UWORD;
#ifdef _USRDLL
# define DECL(T) __declspec(dllexport) T
#else
# define DECL(T) T
#endif
#ifdef __cplusplus
extern "C"
{
#endif
DECL(int) res_SrcLinear(const RES_HWORD X[], RES_HWORD Y[],
double pFactor, RES_UHWORD nx);
DECL(int) res_Resample(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
RES_UHWORD nx, RES_BOOL LargeF, RES_BOOL Interp);
DECL(int) res_GetXOFF(double pFactor, RES_BOOL LargeF);
#ifdef __cplusplus
}
#endif
#endif
pjproject-2.11.1/third_party/resample/src/largefilter.h
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pjproject-2.11.1/third_party/resample/src/libresample_dll.c
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// 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
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/*
* 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
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/* $Id$ */
/*
* 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/smallfilter.h
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#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)