diff --git a/webrtc/base/checks.h b/webrtc/base/checks.h
index 5438d5989d..f64d2730e0 100644
--- a/webrtc/base/checks.h
+++ b/webrtc/base/checks.h
@@ -217,6 +217,14 @@ class FatalMessage {
   std::ostringstream stream_;
 };
 
+// Performs the integer division a/b and returns the result. CHECKs that the
+// remainder is zero.
+template <typename T>
+inline T CheckedDivExact(T a, T b) {
+  CHECK_EQ(a % b, static_cast<T>(0));
+  return a / b;
+}
+
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_CHECKS_H_
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 9d36663320..6fc4dc6aa4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -20,11 +20,6 @@ namespace webrtc {
 
 const int kIsacPayloadType = 103;
 
-inline int DivExact(int a, int b) {
-  CHECK_EQ(a % b, 0);
-  return a / b;
-}
-
 template <typename T>
 AudioEncoderDecoderIsacT<T>::Config::Config()
     : payload_type(kIsacPayloadType),
@@ -115,7 +110,8 @@ template <typename T>
 int AudioEncoderDecoderIsacT<T>::Num10MsFramesInNextPacket() const {
   CriticalSectionScoped cs(lock_.get());
   const int samples_in_next_packet = T::GetNewFrameLen(isac_state_);
-  return DivExact(samples_in_next_packet, DivExact(sample_rate_hz(), 100));
+  return rtc::CheckedDivExact(samples_in_next_packet,
+                              rtc::CheckedDivExact(sample_rate_hz(), 100));
 }
 
 template <typename T>
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 358647f539..fce7721dd6 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -19,10 +19,6 @@ namespace {
 // We always encode at 48 kHz.
 const int kSampleRateHz = 48000;
 
-int DivExact(int a, int b) {
-  CHECK_EQ(a % b, 0);
-  return a / b;
-}
 
 int16_t ClampInt16(size_t x) {
   return static_cast<int16_t>(
@@ -52,11 +48,13 @@ bool AudioEncoderOpus::Config::IsOk() const {
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const Config& config)
-    : num_10ms_frames_per_packet_(DivExact(config.frame_size_ms, 10)),
+    : num_10ms_frames_per_packet_(
+          rtc::CheckedDivExact(config.frame_size_ms, 10)),
       num_channels_(config.num_channels),
       payload_type_(config.payload_type),
       application_(config.application),
-      samples_per_10ms_frame_(DivExact(kSampleRateHz, 100) * num_channels_) {
+      samples_per_10ms_frame_(rtc::CheckedDivExact(kSampleRateHz, 100) *
+                              num_channels_) {
   CHECK(config.IsOk());
   input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
   CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_, application_));
@@ -101,7 +99,8 @@ bool AudioEncoderOpus::EncodeInternal(uint32_t timestamp,
            samples_per_10ms_frame_);
   int16_t r = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
-      DivExact(CastInt16(input_buffer_.size()), num_channels_),
+      rtc::CheckedDivExact(CastInt16(input_buffer_.size()),
+                           static_cast<int16_t>(num_channels_)),
       ClampInt16(max_encoded_bytes), encoded);
   input_buffer_.clear();
   if (r < 0)