diff --git a/webrtc/video_engine/BUILD.gn b/webrtc/video_engine/BUILD.gn
index 8a805f7..b098227 100644
--- a/webrtc/video_engine/BUILD.gn
+++ b/webrtc/video_engine/BUILD.gn
@@ -54,16 +54,6 @@
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
-  if (is_win) {
-    cflags = [
-      # TODO(jschuh): Bug 1348: fix size_t to int truncations.
-      "/wd4267",  # size_t to int truncation.
-
-      # Bug 261.
-      "/wd4373",  # legacy warning for ignoring const / volatile in signatures.
-    ]
-  }
-
   deps = [
     "..:webrtc_common",
     "../common_video",
diff --git a/webrtc/video_engine/overuse_frame_detector.cc b/webrtc/video_engine/overuse_frame_detector.cc
index 88d5d05..23ee59f 100644
--- a/webrtc/video_engine/overuse_frame_detector.cc
+++ b/webrtc/video_engine/overuse_frame_detector.cc
@@ -250,7 +250,7 @@
   }
 
   void Reset() { frame_times_.clear(); }
-  int NumFrames() const { return frame_times_.size(); }
+  int NumFrames() const { return static_cast<int>(frame_times_.size()); }
   int last_processing_time_ms() const { return last_processing_time_ms_; }
 
  private:
diff --git a/webrtc/video_engine/report_block_stats.cc b/webrtc/video_engine/report_block_stats.cc
index ea18e83..6df6288 100644
--- a/webrtc/video_engine/report_block_stats.cc
+++ b/webrtc/video_engine/report_block_stats.cc
@@ -70,8 +70,8 @@
   // Fraction lost since previous report block.
   aggregate.fractionLost =
       FractionLost(num_lost_sequence_numbers, num_sequence_numbers);
-  aggregate.jitter =
-      (aggregate.jitter + report_blocks.size() / 2) / report_blocks.size();
+  aggregate.jitter = static_cast<uint32_t>(
+      (aggregate.jitter + report_blocks.size() / 2) / report_blocks.size());
   return aggregate;
 }
 
diff --git a/webrtc/video_engine/video_engine_core.gypi b/webrtc/video_engine/video_engine_core.gypi
index dc95671..4d295ae 100644
--- a/webrtc/video_engine/video_engine_core.gypi
+++ b/webrtc/video_engine/video_engine_core.gypi
@@ -66,8 +66,6 @@
         'vie_remb.cc',
         'vie_sync_module.cc',
       ], # source
-      # TODO(jschuh): Bug 1348: fix size_t to int truncations.
-      'msvs_disabled_warnings': [ 4267, ],
     },
   ], # targets
   'conditions': [
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 9c6b152..61096b3 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -271,23 +271,32 @@
     rtp_rtx.Add(rtx);
     elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(now) / 1000;
     if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.BitrateReceivedInKbps",
-          rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
-          rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PaddingBitrateReceivedInKbps",
-          rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / 1000);
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.BitrateReceivedInKbps",
+          static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                           1000));
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.MediaBitrateReceivedInKbps",
+          static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.PaddingBitrateReceivedInKbps",
+          static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
+                           1000));
       RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
-              rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+          static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
+                           elapsed_sec / 1000));
       uint32_t ssrc = 0;
       if (vie_receiver_.GetRtxSsrc(&ssrc)) {
-        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
-            rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+        RTC_HISTOGRAM_COUNTS_10000(
+            "WebRTC.Video.RtxBitrateReceivedInKbps",
+            static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                             1000));
       }
       if (vie_receiver_.IsFecEnabled()) {
         RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateReceivedInKbps",
-            rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000);
+                                   static_cast<int>(rtp_rtx.fec.TotalBytes() *
+                                                    8 / elapsed_sec / 1000));
       }
     }
   }
@@ -305,25 +314,34 @@
   if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
     return;
   }
-  RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.BitrateSentInKbps",
-      rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
-  RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateSentInKbps",
-      rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
-  RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PaddingBitrateSentInKbps",
-      rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / 1000);
-  RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RetransmittedBitrateSentInKbps",
-      rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+  RTC_HISTOGRAM_COUNTS_100000(
+      "WebRTC.Video.BitrateSentInKbps",
+      static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                       1000));
+  RTC_HISTOGRAM_COUNTS_10000(
+      "WebRTC.Video.MediaBitrateSentInKbps",
+      static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
+  RTC_HISTOGRAM_COUNTS_10000(
+      "WebRTC.Video.PaddingBitrateSentInKbps",
+      static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
+                       1000));
+  RTC_HISTOGRAM_COUNTS_10000(
+      "WebRTC.Video.RetransmittedBitrateSentInKbps",
+      static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec /
+                       1000));
   if (rtp_rtcp_->RtxSendStatus() != kRtxOff) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateSentInKbps",
-        rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+                               static_cast<int>(rtx.transmitted.TotalBytes() *
+                                                8 / elapsed_sec / 1000));
   }
   bool fec_enabled = false;
   uint8_t pltype_red;
   uint8_t pltype_fec;
   rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
   if (fec_enabled) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps",
-        rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000);
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.FecBitrateSentInKbps",
+        static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000));
   }
 }
 
@@ -375,8 +393,8 @@
     if (video_codec.numberOfSimulcastStreams > 0) {
       // Set correct bitrate to base layer.
       // Create our simulcast RTP modules.
-      int num_modules_to_add =
-          video_codec.numberOfSimulcastStreams - simulcast_rtp_rtcp_.size() - 1;
+      int num_modules_to_add = video_codec.numberOfSimulcastStreams -
+                               static_cast<int>(simulcast_rtp_rtcp_.size()) - 1;
       if (num_modules_to_add < 0) {
         num_modules_to_add = 0;
       }
@@ -425,8 +443,9 @@
       }
 
       // Remove last in list if we have too many.
-      for (int j = simulcast_rtp_rtcp_.size();
-           j > (video_codec.numberOfSimulcastStreams - 1); j--) {
+      for (size_t j = simulcast_rtp_rtcp_.size();
+           j > static_cast<size_t>(video_codec.numberOfSimulcastStreams - 1);
+           j--) {
         RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
         deregistered_modules.push_back(rtp_rtcp);
         rtp_rtcp->SetSendingStatus(false);
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 3ee5bae..7ed207f 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -163,8 +163,9 @@
       CriticalSectionScoped cs(data_cs_.get());
       send_padding_ = video_codec.numberOfSimulcastStreams > 1;
     }
-    if (vcm_->RegisterSendCodec(&video_codec, number_of_cores_,
-                                PayloadRouter::DefaultMaxPayloadLength()) !=
+    if (vcm_->RegisterSendCodec(
+            &video_codec, number_of_cores_,
+            static_cast<uint32_t>(PayloadRouter::DefaultMaxPayloadLength())) !=
         0) {
       return false;
     }
@@ -298,8 +299,9 @@
     // for realz.  https://code.google.com/p/chromium/issues/detail?id=348222
     current_send_codec.extra_options = NULL;
     size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
-    if (vcm_->RegisterSendCodec(&current_send_codec, number_of_cores_,
-                                max_data_payload_length) != VCM_OK) {
+    if (vcm_->RegisterSendCodec(
+            &current_send_codec, number_of_cores_,
+            static_cast<uint32_t>(max_data_payload_length)) != VCM_OK) {
       LOG(LS_INFO) << "De-registered the currently used external encoder ("
                    << static_cast<int>(pl_type) << ") and therefore tried to "
                    << "register the corresponding internal encoder, but none "
@@ -354,7 +356,8 @@
 
   size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
   if (vcm_->RegisterSendCodec(&modified_video_codec, number_of_cores_,
-                              max_data_payload_length) != VCM_OK) {
+                              static_cast<uint32_t>(max_data_payload_length)) !=
+      VCM_OK) {
     return -1;
   }
   return 0;
@@ -627,7 +630,8 @@
       codec.startBitrate = (current_bitrate_bps + 500) / 1000;
       size_t max_payload_length = send_payload_router_->MaxPayloadLength();
       if (vcm_->RegisterSendCodec(&codec, number_of_cores_,
-                                  max_payload_length) != 0) {
+                                  static_cast<uint32_t>(max_payload_length)) !=
+          0) {
         return -1;
       }
     }
diff --git a/webrtc/video_engine/vie_receiver.cc b/webrtc/video_engine/vie_receiver.cc
index a1baf29..18b6e34 100644
--- a/webrtc/video_engine/vie_receiver.cc
+++ b/webrtc/video_engine/vie_receiver.cc
@@ -69,13 +69,14 @@
 void ViEReceiver::UpdateHistograms() {
   FecPacketCounter counter = fec_receiver_->GetPacketCounter();
   if (counter.num_packets > 0) {
-    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedFecPacketsInPercent",
-        counter.num_fec_packets * 100 / counter.num_packets);
+    RTC_HISTOGRAM_PERCENTAGE(
+        "WebRTC.Video.ReceivedFecPacketsInPercent",
+        static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
   }
   if (counter.num_fec_packets > 0) {
-    RTC_HISTOGRAM_PERCENTAGE(
-        "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
-            counter.num_recovered_packets * 100 / counter.num_fec_packets);
+    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
+                             static_cast<int>(counter.num_recovered_packets *
+                                              100 / counter.num_fec_packets));
   }
 }