diff --git a/src/audio/SDL_audiotypecvt.c b/src/audio/SDL_audiotypecvt.c
index fe8a89f7a..264ce9828 100644
--- a/src/audio/SDL_audiotypecvt.c
+++ b/src/audio/SDL_audiotypecvt.c
@@ -171,9 +171,9 @@ SDL_Convert_F32_to_S8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
 
     for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
         const float sample = *src;
-        if (sample > 1.0f) {
+        if (sample >= 1.0f) {
             *dst = 127;
-        } else if (sample < -1.0f) {
+        } else if (sample <= -1.0f) {
             *dst = -127;
         } else {
             *dst = (Sint8)(sample * 127.0f);
@@ -197,9 +197,9 @@ SDL_Convert_F32_to_U8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
 
     for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
         const float sample = *src;
-        if (sample > 1.0f) {
+        if (sample >= 1.0f) {
             *dst = 255;
-        } else if (sample < -1.0f) {
+        } else if (sample <= -1.0f) {
             *dst = 0;
         } else {
             *dst = (Uint8)((sample + 1.0f) * 127.0f);
@@ -223,9 +223,9 @@ SDL_Convert_F32_to_S16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
 
     for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
         const float sample = *src;
-        if (sample > 1.0f) {
+        if (sample >= 1.0f) {
             *dst = 32767;
-        } else if (sample < -1.0f) {
+        } else if (sample <= -1.0f) {
             *dst = -32767;
         } else {
             *dst = (Sint16)(sample * 32767.0f);
@@ -249,9 +249,9 @@ SDL_Convert_F32_to_U16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
 
     for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
         const float sample = *src;
-        if (sample > 1.0f) {
+        if (sample >= 1.0f) {
             *dst = 65534;
-        } else if (sample < -1.0f) {
+        } else if (sample <= -1.0f) {
             *dst = 0;
         } else {
             *dst = (Uint16)((sample + 1.0f) * 32767.0f);
@@ -275,9 +275,9 @@ SDL_Convert_F32_to_S32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
 
     for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
         const float sample = *src;
-        if (sample > 1.0f) {
+        if (sample >= 1.0f) {
             *dst = 2147483647;
-        } else if (sample < -1.0f) {
+        } else if (sample <= -1.0f) {
             *dst = -2147483647;
         } else {
             *dst = ((Sint32)(sample * 8388607.0f)) << 8;