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Diffstat (limited to 'third_party/resample/resample.c')
| -rw-r--r-- | third_party/resample/resample.c | 695 |
1 files changed, 695 insertions, 0 deletions
diff --git a/third_party/resample/resample.c b/third_party/resample/resample.c new file mode 100644 index 00000000..5e2a8aae --- /dev/null +++ b/third_party/resample/resample.c @@ -0,0 +1,695 @@ +/* $Id$ */ +/* + * 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); +} + + |