diff --git a/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc b/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
index bd243897a1..a0acfe7664 100644
--- a/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
+++ b/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
@@ -14,6 +14,7 @@
 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
 #include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/safe_conversions.h"
 
 namespace webrtc {
 
@@ -21,7 +22,7 @@ rtc::Optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
     const SdpAudioFormat& format) {
   Config config;
   config.sample_rate_hz = format.clockrate_hz;
-  config.num_channels = format.num_channels;
+  config.num_channels = rtc::checked_cast<int>(format.num_channels);
   return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
              ? rtc::Optional<Config>(config)
              : rtc::Optional<Config>();
diff --git a/webrtc/call/rtp_demuxer_unittest.cc b/webrtc/call/rtp_demuxer_unittest.cc
index 39dc67b0a1..6fe5fc8d42 100644
--- a/webrtc/call/rtp_demuxer_unittest.cc
+++ b/webrtc/call/rtp_demuxer_unittest.cc
@@ -24,6 +24,7 @@
 #include "webrtc/rtc_base/basictypes.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/safe_conversions.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 
@@ -346,7 +347,8 @@ TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkByRsid) {
   }
 
   for (size_t i = 0; i < arraysize(rsids); i++) {
-    auto packet = CreatePacketWithSsrcRsid(i, rsids[i]);
+    auto packet = CreatePacketWithSsrcRsid(rtc::checked_cast<uint32_t>(i),
+                                           rsids[i]);
     EXPECT_CALL(sinks[i], OnRtpPacket(SamePacketAs(*packet))).Times(1);
     EXPECT_TRUE(demuxer_.OnRtpPacket(*packet));
   }
@@ -360,7 +362,8 @@ TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkByMid) {
   }
 
   for (size_t i = 0; i < arraysize(mids); i++) {
-    auto packet = CreatePacketWithSsrcMid(i, mids[i]);
+    auto packet = CreatePacketWithSsrcMid(rtc::checked_cast<uint32_t>(i),
+                                          mids[i]);
     EXPECT_CALL(sinks[i], OnRtpPacket(SamePacketAs(*packet))).Times(1);
     EXPECT_TRUE(demuxer_.OnRtpPacket(*packet));
   }
@@ -414,7 +417,7 @@ TEST_F(RtpDemuxerTest, PacketsDeliveredInRightOrder) {
   std::unique_ptr<RtpPacketReceived> packets[5];
   for (size_t i = 0; i < arraysize(packets); i++) {
     packets[i] = CreatePacketWithSsrc(ssrc);
-    packets[i]->SetSequenceNumber(i);
+    packets[i]->SetSequenceNumber(rtc::checked_cast<uint16_t>(i));
   }
 
   InSequence sequence;