| 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_));
|
|
|