diff --git a/webrtc/common_audio/signal_processing/include/spl_inl.h b/webrtc/common_audio/signal_processing/include/spl_inl.h
index 370834e694..812ab2d587 100644
--- a/webrtc/common_audio/signal_processing/include/spl_inl.h
+++ b/webrtc/common_audio/signal_processing/include/spl_inl.h
@@ -63,6 +63,11 @@ static __inline int WebRtcSpl_CountLeadingZeros64(uint64_t n) {
 #endif
 }
 
+static __inline int32_t WebRtcSpl_SatW64ToW32(int64_t x) {
+  int32_t x32 = (int32_t)x;
+  return x32 == x ? x32 : x < 0 ? INT32_MIN : INT32_MAX;
+}
+
 #ifdef WEBRTC_ARCH_ARM_V7
 #include "webrtc/common_audio/signal_processing/include/spl_inl_armv7.h"
 #else
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index 10e05bc31a..e4d399b61f 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -234,8 +234,7 @@ static void CalcInvArSpec(const int16_t *ARCoefQ12,
     for (n = 0; n < FRAMESAMPLES/8; n++) {
       const int64_t p =
           (WebRtcIsacfix_kCos[k][n] * (int64_t)CorrQ11[k + 1] + 2) >> 2;
-      RTC_DCHECK_EQ(p, (int32_t)p);  // p fits in 32 bits
-      CurveQ16[n] += (int32_t)p;
+      CurveQ16[n] += WebRtcSpl_SatW64ToW32(p);
     }
   }