diff --git a/src/biquad.h b/src/biquad.h
index 7a7195c..88d47e4 100644
--- a/src/biquad.h
+++ b/src/biquad.h
@@ -8,6 +8,7 @@
 #ifndef SNDFILTER_BIQUAD__H
 #define SNDFILTER_BIQUAD__H
 
+#include "sf_common.h"
 #include "snd.h"
 
 // biquad filtering is a technique used to perform a variety of sound filters
@@ -41,6 +42,8 @@ typedef struct {
 	sf_sample_st yn2;
 } sf_biquad_state_st;
 
+SF_API_BEGIN
+
 // these functions will initialize an sf_biquad_state_st structure based on the desired filter
 void sf_lowpass  (sf_biquad_state_st *state, int rate, float cutoff, float resonance);
 void sf_highpass (sf_biquad_state_st *state, int rate, float cutoff, float resonance);
@@ -56,4 +59,6 @@ void sf_highshelf(sf_biquad_state_st *state, int rate, float freq, float Q, floa
 void sf_biquad_process(sf_biquad_state_st *state, int size, sf_sample_st *input,
 	sf_sample_st *output);
 
+SF_API_END
+
 #endif // SNDFILTER_BIQUAD__H
diff --git a/src/compressor.h b/src/compressor.h
index 13d7ef3..81f8ec7 100644
--- a/src/compressor.h
+++ b/src/compressor.h
@@ -8,6 +8,7 @@
 #ifndef SNDFILTER_COMPRESSOR__H
 #define SNDFILTER_COMPRESSOR__H
 
+#include "sf_common.h"
 #include "snd.h"
 
 // dynamic range compression is a complex topic with many different algorithms
@@ -75,6 +76,8 @@ typedef struct {
 	sf_sample_st delaybuf[SF_COMPRESSOR_MAXDELAY]; // predelay buffer
 } sf_compressor_state_st;
 
+SF_API_BEGIN
+
 // populate a compressor state with all default values
 void sf_defaultcomp(sf_compressor_state_st *state, int rate);
 
@@ -108,4 +111,6 @@ void sf_advancecomp(sf_compressor_state_st *state,
 void sf_compressor_process(sf_compressor_state_st *state, int size, sf_sample_st *input,
 	sf_sample_st *output);
 
+SF_API_END
+
 #endif // SNDFILTER_COMPRESSOR__H
diff --git a/src/reverb.h b/src/reverb.h
index 8b9e778..3fbf287 100644
--- a/src/reverb.h
+++ b/src/reverb.h
@@ -8,8 +8,11 @@
 #ifndef SNDFILTER_REVERB__H
 #define SNDFILTER_REVERB__H
 
+#include "sf_common.h"
 #include "snd.h"
 
+SF_API_BEGIN
+
 // this API works by first initializing an sf_reverb_state_st structure, then using it to process a
 // sample in chunks
 //
@@ -286,4 +289,6 @@ void sf_advancereverb(sf_reverb_state_st *rv,
 void sf_reverb_process(sf_reverb_state_st *state, int size, sf_sample_st *input,
 	sf_sample_st *output);
 
+SF_API_END
+
 #endif // SNDFILTER_REVERB__H
diff --git a/src/sf_common.h b/src/sf_common.h
new file mode 100644
index 0000000..ff26d4e
--- /dev/null
+++ b/src/sf_common.h
@@ -0,0 +1,20 @@
+// (c) Copyright 2021, Kazuki Tanaka (@tk-aria)
+// MIT License
+// Project Home: https://github.com/velipso/sndfilter
+
+#ifndef SNDFILTER_COMMON__H
+#define SNDFILTER_COMMON__H
+
+#ifdef __cplusplus
+
+#define SF_API_BEGIN extern "C" {
+#define SF_API_END }
+
+#else
+
+#define SF_API_BEGIN
+#define SF_API_END
+
+#endif
+
+#endif // SNDFILTER_COMMON__H
diff --git a/src/snd.h b/src/snd.h
index da86650..45f927f 100644
--- a/src/snd.h
+++ b/src/snd.h
@@ -8,6 +8,7 @@
 #define SNDFILTER_SND__H
 
 #include <stdbool.h>
+#include "sf_common.h"
 
 typedef struct {
 	float L; // left channel sample
@@ -20,7 +21,11 @@ typedef struct {
 	int rate; // samples per second
 } sf_snd_st, *sf_snd;
 
+SF_API_BEGIN
+
 sf_snd sf_snd_new(int size, int rate, bool clear);
 void   sf_snd_free(sf_snd snd);
 
+SF_API_END
+
 #endif // SNDFILTER_SND__H
diff --git a/src/wav.h b/src/wav.h
index 815cb08..5694b93 100644
--- a/src/wav.h
+++ b/src/wav.h
@@ -9,9 +9,14 @@
 #ifndef SNDFILTER_WAV__H
 #define SNDFILTER_WAV__H
 
+#include "sf_common.h"
 #include "snd.h"
 
+SF_API_BEGIN
+
 sf_snd sf_wavload(const char *file);
 bool   sf_wavsave(sf_snd snd, const char *file);
 
+SF_API_END
+
 #endif // SNDFILTER_WAV__H