diff --git a/src/frontend/sndprint.c b/src/frontend/sndprint.c
index 63a4dc64d..a0b389c47 100644
--- a/src/frontend/sndprint.c
+++ b/src/frontend/sndprint.c
@@ -10,16 +10,16 @@
 #include "ngspice/ngspice.h"
 
 
-int o_samplerate = 48000;
-int o_sndfmt = (SF_FORMAT_WAV | SF_FORMAT_PCM_24);
-float o_mult = 1.0;
-float o_off = 0.0;
+static int o_samplerate = 48000;
+static int o_sndfmt = (SF_FORMAT_WAV | SF_FORMAT_PCM_24);
+static float o_mult = 1.0;
+static float o_off = 0.0;
 
 //////////////////////////////////   aliki  //////////////////////////////////
 
 #define HDRSIZE 256
 
-void* my_open_aliki(char* fn, int nchannel) {
+static void* my_open_aliki(char* fn, int nchannel) {
 	char p[HDRSIZE];
 	FILE* aldfile;
 	if ((aldfile = fopen(fn, "w")) == 0) {
@@ -49,11 +49,11 @@ void* my_open_aliki(char* fn, int nchannel) {
 	return ((void*)aldfile);
 }
 
-size_t my_write_aliki(void* d, float val) {
+static size_t my_write_aliki(void* d, float val) {
 	return(fwrite(&val, sizeof(float), 1, (FILE*)d));
 }
 
-void my_close_aliki(void* d) {
+static void my_close_aliki(void* d) {
 	fclose((FILE*)d);
 }
 
@@ -67,8 +67,7 @@ typedef struct {
 	float* sf_buf;
 } SSFILE;
 
-void* my_open_sf(char* fn, int nchannel) {
-
+static void* my_open_sf(char* fn, int nchannel) {
 	SSFILE* d = calloc(1, sizeof(SSFILE));
 	SF_INFO sfinfo;
 
@@ -94,7 +93,7 @@ void* my_open_sf(char* fn, int nchannel) {
 	return ((void*)d);
 }
 
-int my_write_sf(void* d, float val) {
+static int my_write_sf(void* d, float val) {
 	SSFILE* p = (SSFILE*)d;
 	p->sf_buf[p->sf_bptr++] = val;
 	if (p->sf_bptr >= p->sf_channels) {
@@ -104,7 +103,7 @@ int my_write_sf(void* d, float val) {
 	return (1);
 }
 
-void my_close_sf(void* d) {
+static void my_close_sf(void* d) {
 	sf_close(((SSFILE*)d)->outfile);
 	free(((SSFILE*)d)->sf_buf);
 	free((SSFILE*)d);
@@ -121,15 +120,15 @@ typedef struct SP_BUF {
 	double* val;
 } SP_BUF;
 
-void (*p_close)(void*);
-void* (*p_open)(char*, int);
-int  (*p_write)(void*, float);
-void* outfile;
-uint32_t sample;
-int sp_nchannel;
+static void (*p_close)(void*);
+static void* (*p_open)(char*, int);
+static int  (*p_write)(void*, float);
+static void* outfile;
+static uint32_t sample;
+static int sp_nchannel;
 #define SP_MAX (2)
 SP_BUF* sp_buf;
-char* filename = NULL;
+static char* filename = NULL;
 
 #define HAVE_SRC
 
@@ -138,15 +137,15 @@ char* filename = NULL;
 #else
 #include <samplerate.h>
 #define OBUFSIZE 256
-int oversampling = 64;
+static int oversampling = 64;
 #define OVERSAMPLING ((double) oversampling)
-SRC_STATE* rabbit;
-int rabbit_err;
-float* interleaved;
-float* resampled;
-int iptr = 0;
+static SRC_STATE* rabbit;
+static int rabbit_err;
+static float* interleaved;
+static float* resampled;
+static int iptr = 0;
 
-int resample_wrapper(void* d, float val) {
+static int resample_wrapper(void* d, float val) {
 	interleaved[iptr++] = val;
 	size_t ibufsize = sp_nchannel * OBUFSIZE * oversampling;
 	size_t obufsize = sp_nchannel * OBUFSIZE;
diff --git a/src/spicelib/devices/vsrc/vsjack.c b/src/spicelib/devices/vsrc/vsjack.c
index a1197fe2c..2c5e30c1f 100644
--- a/src/spicelib/devices/vsrc/vsjack.c
+++ b/src/spicelib/devices/vsrc/vsjack.c
@@ -9,28 +9,28 @@
 #include <inttypes.h>
 #define VS_BUFSIZ 1024
 
-#include "ngspice/ngspice.h""
+#include "ngspice/ngspice.h"
 
 #define MAX_D 6 
-char* (sources[MAX_D]);
+static char* (sources[MAX_D]);
 
-SNDFILE* m_sndfile[MAX_D];
-int m_channels[MAX_D]; //< number of channles in src-file
-uint32_t m_samplerate[MAX_D]; //< samplerate of source
-uint32_t m_frames[MAX_D]; //< duration of source in frames
-float* (interleaved[MAX_D]); //< internal soundfile buffer
-uint32_t ilb_start[MAX_D]; //< first sample in buffer 
-uint32_t ilb_end[MAX_D]; //< last sample in buffer
+static SNDFILE* m_sndfile[MAX_D];
+static int m_channels[MAX_D]; //< number of channles in src-file
+static uint32_t m_samplerate[MAX_D]; //< samplerate of source
+static uint32_t m_frames[MAX_D]; //< duration of source in frames
+static float* (interleaved[MAX_D]); //< internal soundfile buffer
+static uint32_t ilb_start[MAX_D]; //< first sample in buffer 
+static uint32_t ilb_end[MAX_D]; //< last sample in buffer
 
 #define HAVE_SRC
 
 #ifdef HAVE_SRC
 #include <samplerate.h>
-double src_ratio = 64.0;
-#define SRC_RATIO src_ratio
-SRC_STATE* rabbit[MAX_D];
-int rabbit_err[MAX_D];
-float* (resampled[MAX_D]); //< internal soundfile buffer
+static double src_ratio = 64.0;
+#define SRC_RATIO 64
+static SRC_STATE* rabbit[MAX_D];
+static int rabbit_err[MAX_D];
+static float* (resampled[MAX_D]); //< internal soundfile buffer
 #endif
 
 void vsjack_initialize(void) {
@@ -102,7 +102,7 @@ void load_buffer(int d, uint32_t sample) {
 	sf_seek(m_sndfile[d], sample, SEEK_SET);
 	ilb_start[d] = sample;
 	uint32_t nframes;
-	if ((nframes = sf_readf_float(m_sndfile[d], (interleaved[d]), VS_BUFSIZ)) > 0) {
+	if ((nframes = (uint32_t)sf_readf_float(m_sndfile[d], (interleaved[d]), VS_BUFSIZ)) > 0) {
 		ilb_end[d] = ilb_start[d] + nframes;
 	}
 	else {
@@ -139,7 +139,7 @@ double get_value(int d, double time, int channel) {
 	}
 
 #ifdef HAVE_SRC
-	int offset = floor((sample - ilb_start[d]) * SRC_RATIO);
+	int offset = (int)floor((sample - ilb_start[d]) * SRC_RATIO);
 	if (offset > VS_BUFSIZ * SRC_RATIO || offset < 0) {
 		printf("value not in buffer:%i.\n", d);
 		return (0.0); // nan ?