Index: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc |
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc |
index 291dded3b22b8f5b32149aaff0c5c4b23e22a7c2..bb93f57e93e1035c98cae90f5f2bf641a6a6552c 100644 |
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc |
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc |
@@ -162,9 +162,6 @@ TEST_F(RtpRtcpAudioTest, Basic) { |
module1->SetSSRC(test_ssrc); |
module1->SetStartTimestamp(test_timestamp); |
- // Test detection at the end of a DTMF tone. |
- // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true)); |
- |
EXPECT_EQ(0, module1->SetSendingStatus(true)); |
// Start basic RTP test. |