diff --git a/api/video/encoded_frame.cc b/api/video/encoded_frame.cc
index 42d6b06b84..86d1a698a7 100644
--- a/api/video/encoded_frame.cc
+++ b/api/video/encoded_frame.cc
@@ -13,7 +13,7 @@
 namespace webrtc {
 
 bool EncodedFrame::delayed_by_retransmission() const {
-  return 0;
+  return false;
 }
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index 6ffae0975f..461ee7fa4a 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -159,7 +159,7 @@ int Normal::Process(const int16_t* input,
 
     if (cng_decoder) {
       // Generate long enough for 48kHz.
-      if (!cng_decoder->Generate(cng_output, 0)) {
+      if (!cng_decoder->Generate(cng_output, false)) {
         // Error returned; set return vector to all zeros.
         memset(cng_output, 0, sizeof(cng_output));
       }
diff --git a/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
index a82f93bea2..6f27cdad4f 100644
--- a/modules/audio_coding/neteq/test/neteq_decoding_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -193,7 +193,7 @@ void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
   rtp_info->timestamp = timestamp;
   rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
   rtp_info->payloadType = 94;  // PCM16b WB codec.
-  rtp_info->markerBit = 0;
+  rtp_info->markerBit = false;
 }
 
 void NetEqDecodingTest::PopulateCng(int frame_index,
@@ -205,7 +205,7 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
   rtp_info->timestamp = timestamp;
   rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
   rtp_info->payloadType = 98;  // WB CNG.
-  rtp_info->markerBit = 0;
+  rtp_info->markerBit = false;
   payload[0] = 64;   // Noise level -64 dBov, quite arbitrarily chosen.
   *payload_len = 1;  // Only noise level, no spectral parameters.
 }
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 1b1222c077..599fafb602 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -508,7 +508,7 @@ void TestStereo::Run(TestPackStereo* channel,
   in_file_stereo_->FastForward(100);
   in_file_mono_->FastForward(100);
 
-  while (1) {
+  while (true) {
     // Simulate packet loss by setting `packet_loss_` to "true" in
     // `percent_loss` percent of the loops.
     if (percent_loss > 0) {
diff --git a/modules/video_coding/utility/vp9_uncompressed_header_parser.cc b/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
index 867967ddc0..bf9d51f692 100644
--- a/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
+++ b/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
@@ -158,7 +158,8 @@ void Vp9ReadQp(BitstreamReader& br, Vp9UncompressedHeader* frame_info) {
 void Vp9ReadSegmentationParams(BitstreamReader& br,
                                Vp9UncompressedHeader* frame_info) {
   constexpr int kSegmentationFeatureBits[kVp9SegLvlMax] = {8, 6, 2, 0};
-  constexpr bool kSegmentationFeatureSigned[kVp9SegLvlMax] = {1, 1, 0, 0};
+  constexpr bool kSegmentationFeatureSigned[kVp9SegLvlMax] = {true, true, false,
+                                                              false};
 
   frame_info->segmentation_enabled = br.Read<bool>();
   if (!frame_info->segmentation_enabled) {
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 045374d358..b1e593ba0c 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -694,7 +694,7 @@ absl::optional<Syncable::Info> VideoReceiveStream2::GetInfo() const {
 bool VideoReceiveStream2::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
                                                  int64_t* time_ms) const {
   RTC_DCHECK_NOTREACHED();
-  return 0;
+  return false;
 }
 
 void VideoReceiveStream2::SetEstimatedPlayoutNtpTimestampMs(