Fixed all VS6 and VS8 projects with new third party projects layout
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/split-3rd-party@1177 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/third_party/resample/COPYING b/third_party/resample/COPYING
new file mode 100644
index 0000000..4ccd6cc
--- /dev/null
+++ b/third_party/resample/COPYING
@@ -0,0 +1,463 @@
+
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+
diff --git a/third_party/resample/include/resamplesubs.h b/third_party/resample/include/resamplesubs.h
new file mode 100644
index 0000000..76563b7
--- /dev/null
+++ b/third_party/resample/include/resamplesubs.h
@@ -0,0 +1,18 @@
+#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;
+
+int res_SrcLinear(const RES_HWORD X[], RES_HWORD Y[],
+ double pFactor, RES_UHWORD nx);
+int res_Resample(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
+ RES_UHWORD nx, RES_BOOL LargeF, RES_BOOL Interp);
+int res_GetXOFF(double pFactor, RES_BOOL LargeF);
+
+
+#endif
+
diff --git a/third_party/resample/resample.c b/third_party/resample/resample.c
deleted file mode 100644
index 5e2a8aa..0000000
--- a/third_party/resample/resample.c
+++ /dev/null
@@ -1,695 +0,0 @@
-/* $Id$ */
-/*
- * Based on:
- * resample-1.8.tar.gz from the
- * 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 <pjmedia/resample.h>
-#include <pjmedia/errno.h>
-#include <pj/assert.h>
-#include <pj/log.h>
-#include <pj/pool.h>
-
-
-#define THIS_FILE "resample.c"
-
-
-/*
- * Taken from stddefs.h
- */
-#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
-
-typedef char RES_BOOL;
-typedef short RES_HWORD;
-typedef int RES_WORD;
-typedef unsigned short RES_UHWORD;
-typedef unsigned int RES_UWORD;
-
-#define MAX_HWORD (32767)
-#define MIN_HWORD (-32768)
-
-#ifdef DEBUG
-#define INLINE
-#else
-#define INLINE inline
-#endif
-
-/*
- * Taken from resample.h
- *
- * 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.
- */
-
-
-#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(PJMEDIA_HAS_SMALL_FILTER) && PJMEDIA_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(PJMEDIA_HAS_LARGE_FILTER) && PJMEDIA_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);
- endTime = time + (1<<Np)*(RES_WORD)nx;
- while (time < endTime)
- {
- 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);
- endTime = time + (1<<Np)*(RES_WORD)nx;
- while (time < endTime)
- {
- 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);
- endTime = time + (1<<Np)*(RES_WORD)nx;
- while (time < endTime)
- {
- 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 */
-}
-
-
-/* ***************************************************************************
- *
- * PJMEDIA RESAMPLE
- *
- * ***************************************************************************
- */
-
-struct pjmedia_resample
-{
- double factor; /* Conversion factor = rate_out / rate_in. */
- pj_bool_t large_filter; /* Large filter? */
- pj_bool_t high_quality; /* Not fast? */
- unsigned xoff; /* History and lookahead size, in samples */
- unsigned frame_size; /* Samples per frame. */
- pj_int16_t *buffer; /* Input buffer. */
-};
-
-
-PJ_DEF(pj_status_t) pjmedia_resample_create( pj_pool_t *pool,
- pj_bool_t high_quality,
- pj_bool_t large_filter,
- unsigned channel_count,
- unsigned rate_in,
- unsigned rate_out,
- unsigned samples_per_frame,
- pjmedia_resample **p_resample)
-{
- pjmedia_resample *resample;
-
- PJ_ASSERT_RETURN(pool && p_resample && rate_in &&
- rate_out && samples_per_frame, PJ_EINVAL);
-
- resample = pj_pool_alloc(pool, sizeof(pjmedia_resample));
- PJ_ASSERT_RETURN(resample, PJ_ENOMEM);
-
- PJ_UNUSED_ARG(channel_count);
-
- /*
- * If we're downsampling, always use the fast algorithm since it seems
- * to yield the same quality.
- */
- if (rate_out < rate_in) {
- //no this is not a good idea. It sounds pretty good with speech,
- //but very poor with background noise etc.
- //high_quality = 0;
- }
-
-#if !defined(PJMEDIA_HAS_LARGE_FILTER) || PJMEDIA_HAS_LARGE_FILTER==0
- /*
- * If large filter is excluded in the build, then prevent application
- * from using it.
- */
- if (high_quality && large_filter) {
- large_filter = PJ_FALSE;
- PJ_LOG(5,(THIS_FILE,
- "Resample uses small filter because large filter is "
- "disabled"));
- }
-#endif
-
-#if !defined(PJMEDIA_HAS_SMALL_FILTER) || PJMEDIA_HAS_SMALL_FILTER==0
- /*
- * If small filter is excluded in the build and application wants to
- * use it, then drop to linear conversion.
- */
- if (high_quality && large_filter == 0) {
- high_quality = PJ_FALSE;
- PJ_LOG(4,(THIS_FILE,
- "Resample uses linear because small filter is disabled"));
- }
-#endif
-
- resample->factor = rate_out * 1.0 / rate_in;
- resample->large_filter = large_filter;
- resample->high_quality = high_quality;
- resample->frame_size = samples_per_frame;
-
- if (high_quality) {
- unsigned size;
-
- /* This is a bug in xoff calculation, thanks Stephane Lussier
- * of Macadamian dot com.
- * resample->xoff = large_filter ? 32 : 6;
- */
- if (large_filter)
- resample->xoff = (LARGE_FILTER_NMULT + 1) / 2.0 *
- MAX(1.0, 1.0/resample->factor);
- else
- resample->xoff = (SMALL_FILTER_NMULT + 1) / 2.0 *
- MAX(1.0, 1.0/resample->factor);
-
-
- size = (samples_per_frame + 2*resample->xoff) * sizeof(pj_int16_t);
- resample->buffer = pj_pool_alloc(pool, size);
- PJ_ASSERT_RETURN(resample->buffer, PJ_ENOMEM);
-
- pjmedia_zero_samples(resample->buffer, resample->xoff*2);
-
-
- } else {
- resample->xoff = 0;
- }
-
- *p_resample = resample;
-
- PJ_LOG(5,(THIS_FILE, "resample created: %s qualiy, %s filter, in/out "
- "rate=%d/%d",
- (high_quality?"high":"low"),
- (large_filter?"large":"small"),
- rate_in, rate_out));
- return PJ_SUCCESS;
-}
-
-
-
-PJ_DEF(void) pjmedia_resample_run( pjmedia_resample *resample,
- const pj_int16_t *input,
- pj_int16_t *output )
-{
- PJ_ASSERT_ON_FAIL(resample, return);
-
- if (resample->high_quality) {
- pj_int16_t *dst_buf;
- const pj_int16_t *src_buf;
-
- /* Okay chaps, here's how we do resampling.
- *
- * The original resample algorithm requires xoff samples *before* the
- * input buffer as history, and another xoff samples *after* the
- * end of the input buffer as lookahead. Since application can only
- * supply framesize buffer on each run, PJMEDIA needs to arrange the
- * buffer to meet these requirements.
- *
- * So here comes the trick.
- *
- * First of all, because of the history and lookahead requirement,
- * resample->buffer need to accomodate framesize+2*xoff samples in its
- * buffer. This is done when the buffer is created.
- *
- * On the first run, the input frame (supplied by application) is
- * copied to resample->buffer at 2*xoff position. The first 2*xoff
- * samples are initially zeroed (in the initialization). The resample
- * algorithm then invoked at resample->buffer+xoff ONLY, thus giving
- * it one xoff at the beginning as zero, and one xoff at the end
- * as the end of the original input. The resample algorithm will see
- * that the first xoff samples in the input as zero.
- *
- * So here's the layout of resample->buffer on the first run.
- *
- * run 0
- * +------+------+--------------+
- * | 0000 | 0000 | frame0... |
- * +------+------+--------------+
- * ^ ^ ^ ^
- * 0 xoff 2*xoff size+2*xoff
- *
- * (Note again: resample algorithm is called at resample->buffer+xoff)
- *
- * At the end of the run, 2*xoff samples from the end of
- * resample->buffer are copied to the beginning of resample->buffer.
- * The first xoff part of this will be used as history for the next
- * run, and the second xoff part of this is actually the start of
- * resampling for the next run.
- *
- * And the first run completes, the function returns.
- *
- *
- * On the next run, the input frame supplied by application is again
- * copied at 2*xoff position in the resample->buffer, and the
- * resample algorithm is again invoked at resample->buffer+xoff
- * position. So effectively, the resample algorithm will start its
- * operation on the last xoff from the previous frame, and gets the
- * history from the last 2*xoff of the previous frame, and the look-
- * ahead from the last xoff of current frame.
- *
- * So on this run, the buffer layout is:
- *
- * run 1
- * +------+------+--------------+
- * | frm0 | frm0 | frame1... |
- * +------+------+--------------+
- * ^ ^ ^ ^
- * 0 xoff 2*xoff size+2*xoff
- *
- * As you can see from above diagram, the resampling algorithm is
- * actually called from the last xoff part of previous frame (frm0).
- *
- * And so on the process continues for the next frame, and the next,
- * and the next, ...
- *
- */
- dst_buf = resample->buffer + resample->xoff*2;
- pjmedia_copy_samples(dst_buf, input, resample->frame_size);
-
- if (resample->factor >= 1) {
-
- if (resample->large_filter) {
- SrcUp(resample->buffer + resample->xoff, output,
- resample->factor, resample->frame_size,
- LARGE_FILTER_NWING, LARGE_FILTER_SCALE,
- LARGE_FILTER_IMP, LARGE_FILTER_IMPD,
- PJ_TRUE);
- } else {
- SrcUp(resample->buffer + resample->xoff, output,
- resample->factor, resample->frame_size,
- SMALL_FILTER_NWING, SMALL_FILTER_SCALE,
- SMALL_FILTER_IMP, SMALL_FILTER_IMPD,
- PJ_TRUE);
- }
-
- } else {
-
- if (resample->large_filter) {
-
- SrcUD( resample->buffer + resample->xoff, output,
- resample->factor, resample->frame_size,
- LARGE_FILTER_NWING,
- LARGE_FILTER_SCALE * resample->factor + 0.5,
- LARGE_FILTER_IMP, LARGE_FILTER_IMPD,
- PJ_TRUE);
-
- } else {
-
- SrcUD( resample->buffer + resample->xoff, output,
- resample->factor, resample->frame_size,
- SMALL_FILTER_NWING,
- SMALL_FILTER_SCALE * resample->factor + 0.5,
- SMALL_FILTER_IMP, SMALL_FILTER_IMPD,
- PJ_TRUE);
-
- }
-
- }
-
- dst_buf = resample->buffer;
- src_buf = input + resample->frame_size - resample->xoff*2;
- pjmedia_copy_samples(dst_buf, src_buf, resample->xoff * 2);
-
- } else {
- SrcLinear( input, output, resample->factor, resample->frame_size);
- }
-}
-
-PJ_DEF(unsigned) pjmedia_resample_get_input_size(pjmedia_resample *resample)
-{
- PJ_ASSERT_RETURN(resample != NULL, 0);
- return resample->frame_size;
-}
-
-PJ_DEF(void) pjmedia_resample_destroy(pjmedia_resample *resample)
-{
- PJ_UNUSED_ARG(resample);
-}
-
-
diff --git a/third_party/resample/largefilter.h b/third_party/resample/src/largefilter.h
similarity index 100%
rename from third_party/resample/largefilter.h
rename to third_party/resample/src/largefilter.h
diff --git a/third_party/resample/src/resample.h b/third_party/resample/src/resample.h
new file mode 100644
index 0000000..3521251
--- /dev/null
+++ b/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.
+ */
+
diff --git a/third_party/resample/src/resamplesubs.c b/third_party/resample/src/resamplesubs.c
new file mode 100644
index 0000000..ae78add
--- /dev/null
+++ b/third_party/resample/src/resamplesubs.c
@@ -0,0 +1,365 @@
+/* $Id$ */
+/*
+ * 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);
+ endTime = time + (1<<Np)*(RES_WORD)nx;
+ while (time < endTime)
+ {
+ 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);
+ endTime = time + (1<<Np)*(RES_WORD)nx;
+ while (time < endTime)
+ {
+ 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);
+ endTime = time + (1<<Np)*(RES_WORD)nx;
+ while (time < endTime)
+ {
+ 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 */
+}
+
+
+int res_SrcLinear(const RES_HWORD X[], RES_HWORD Y[],
+ double pFactor, RES_UHWORD nx)
+{
+ return SrcLinear(X, Y, pFactor, nx);
+}
+
+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);
+
+ }
+}
+
+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);
+}
+
diff --git a/third_party/resample/smallfilter.h b/third_party/resample/src/smallfilter.h
similarity index 100%
rename from third_party/resample/smallfilter.h
rename to third_party/resample/src/smallfilter.h
diff --git a/third_party/resample/src/stddefs.h b/third_party/resample/src/stddefs.h
new file mode 100644
index 0000000..8349770
--- /dev/null
+++ b/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)
+