Index: webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc |
diff --git a/webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc b/webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc |
index d6dbf4b3d06cc5c34c41d38db1aa44a35dd73ba3..2e6c73311dca0701e32244734b20b3933617bec9 100644 |
--- a/webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc |
+++ b/webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc |
@@ -44,7 +44,7 @@ TEST_F(DtmfTest, ManualCanChangeDtmfPayloadType) { |
// Start by modifying the receiving side. |
for (int i = 0; i < voe_codec_->NumOfCodecs(); i++) { |
EXPECT_EQ(0, voe_codec_->GetCodec(i, codec_instance)); |
- if (!_stricmp("telephone-event", codec_instance.plname)) { |
+ if (!STR_CASE_CMP("telephone-event", codec_instance.plname)) { |
codec_instance.pltype = 88; // Use 88 instead of default 106. |
EXPECT_EQ(0, voe_base_->StopSend(channel_)); |
EXPECT_EQ(0, voe_base_->StopPlayout(channel_)); |