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); |