[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 80967b2966947bc447051c3e837c6d9f3ce8ec4b..1ca7831ab2cb9c153cae70c4f2d204a5f759efdf 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -1281,9 +1281,9 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
   // For Telephone events, payload is not added to the registered payload list,
   // it will register only the payload used for audio stream.
   // Registering the payload again for audio stream with different payload name.
-  strcpy(payload_name, "payload_name");
+  const char kPayloadName[] = "payload_name";
   ASSERT_EQ(
-      0, rtp_sender_->RegisterPayload(payload_name, payload_type, 8000, 1, 0));
+      0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   // DTMF event key=9, duration=500 and attenuationdB=10
   rtp_sender_->SendTelephoneEvent(9, 500, 10);