diff --git a/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc b/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
index 9d86ba31be..cb8d0db5c7 100644
--- a/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
+++ b/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
@@ -36,7 +36,7 @@ class BweSimulation : public BweTest,
     VerboseLogging(true);
   }
 
-  Random random_;
+  test::Random random_;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(BweSimulation);
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index 56a309c59d..522489e921 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -64,6 +64,15 @@ std::vector<K> Keys(const std::map<K, V>& map) {
   return keys;
 }
 
+uint32_t ConvertMsTo24Bits(int64_t time_ms) {
+  uint32_t time_24_bits =
+      static_cast<uint32_t>(
+          ((static_cast<uint64_t>(time_ms) << kAbsSendTimeFraction) + 500) /
+          1000) &
+      0x00FFFFFF;
+  return time_24_bits;
+}
+
 bool RemoteBitrateEstimatorAbsSendTime::IsWithinClusterBounds(
     int send_delta_ms,
     const Cluster& cluster_aggregate) {
@@ -219,12 +228,8 @@ bool RemoteBitrateEstimatorAbsSendTime::IsBitrateImproving(
 void RemoteBitrateEstimatorAbsSendTime::IncomingPacketFeedbackVector(
     const std::vector<PacketInfo>& packet_feedback_vector) {
   for (const auto& packet_info : packet_feedback_vector) {
-    // TODO(holmer): We should get rid of this conversion if possible as we may
-    // lose precision.
-    uint32_t send_time_32bits = (packet_info.send_time_ms) / kTimestampToMs;
-    uint32_t send_time_24bits =
-        send_time_32bits >> kAbsSendTimeInterArrivalUpshift;
-    IncomingPacketInfo(packet_info.arrival_time_ms, send_time_24bits,
+    IncomingPacketInfo(packet_info.arrival_time_ms,
+                       ConvertMsTo24Bits(packet_info.send_time_ms),
                        packet_info.payload_size, 0, packet_info.was_paced);
   }
 }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 8dc8ca6e41..b21f115065 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -26,8 +26,9 @@
 namespace webrtc {
 
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
-const size_t kMaxPaddingLength = 224;
-const int kSendSideDelayWindowMs = 1000;
+static const size_t kMaxPaddingLength = 224;
+static const int kSendSideDelayWindowMs = 1000;
+static const uint32_t kAbsSendTimeFraction = 18;
 
 namespace {
 
@@ -45,6 +46,16 @@ const char* FrameTypeToString(FrameType frame_type) {
   return "";
 }
 
+// TODO(holmer): Merge this with the implementation in
+// remote_bitrate_estimator_abs_send_time.cc.
+uint32_t ConvertMsTo24Bits(int64_t time_ms) {
+  uint32_t time_24_bits =
+      static_cast<uint32_t>(
+          ((static_cast<uint64_t>(time_ms) << kAbsSendTimeFraction) + 500) /
+          1000) &
+      0x00FFFFFF;
+  return time_24_bits;
+}
 }  // namespace
 
 class BitrateAggregator {
@@ -1596,7 +1607,7 @@ void RTPSender::UpdateAbsoluteSendTime(uint8_t* rtp_packet,
   // Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
   // fractional part).
   ByteWriter<uint32_t, 3>::WriteBigEndian(rtp_packet + offset + 1,
-                                          ((now_ms << 18) / 1000) & 0x00ffffff);
+                                          ConvertMsTo24Bits(now_ms));
 }
 
 uint16_t RTPSender::UpdateTransportSequenceNumber(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index e4ace67a48..9be888c92a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -64,7 +64,7 @@ size_t GetPayloadDataLength(const RTPHeader& rtp_header,
 }
 
 uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
-  return 0x00fffffful & ((time_ms << 18) / 1000);
+  return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;
 }
 
 class LoopbackTransportTest : public webrtc::Transport {