diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
index 23db2cdd0c..b585ce3ec0 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
@@ -55,7 +55,8 @@ void BitrateController::MakeDecision(AudioEncoderRuntimeConfig* config) {
     if (config->frame_length_ms)
       frame_length_ms_ = *config->frame_length_ms;
     int overhead_rate_bps =
-        *overhead_bytes_per_packet_ * 8 * 1000 / frame_length_ms_;
+        static_cast<int>(*overhead_bytes_per_packet_ * 8 * 1000 /
+            frame_length_ms_);
     bitrate_bps_ = std::max(0, *target_audio_bitrate_bps_ - overhead_rate_bps);
   }
   config->bitrate_bps = rtc::Optional<int>(bitrate_bps_);
diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
index d7d1644aae..e47f92fe97 100644
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -282,7 +282,8 @@ void StatisticsCalculator::GetNetworkStatistics(
       discarded_secondary_packets_ * samples_per_packet;
   stats->secondary_discarded_rate = CalculateQ14Ratio(
       discarded_secondary_samples,
-      discarded_secondary_samples + secondary_decoded_samples_);
+      static_cast<uint32_t>(discarded_secondary_samples +
+        secondary_decoded_samples_));
 
   if (waiting_times_.size() == 0) {
     stats->mean_waiting_time_ms = -1;