| Index: webrtc/test/call_test.cc | 
| diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc | 
| index aaafdb3876d4e8e0d852a9701f76f9e0ae14a06d..aadcb8d5b89dcf320b00eb51ab2b41504aa23cc5 100644 | 
| --- a/webrtc/test/call_test.cc | 
| +++ b/webrtc/test/call_test.cc | 
| @@ -203,8 +203,8 @@ void CallTest::CreateSendConfig(size_t num_video_streams, | 
| size_t num_flexfec_streams, | 
| Transport* send_transport) { | 
| RTC_DCHECK(num_video_streams <= kNumSsrcs); | 
| -  RTC_DCHECK_LE(num_audio_streams, 1u); | 
| -  RTC_DCHECK_LE(num_flexfec_streams, 1u); | 
| +  RTC_DCHECK_LE(num_audio_streams, 1); | 
| +  RTC_DCHECK_LE(num_flexfec_streams, 1); | 
| RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); | 
| if (num_video_streams > 0) { | 
| video_send_config_ = VideoSendStream::Config(send_transport); | 
| @@ -261,7 +261,7 @@ void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) { | 
| } | 
| } | 
|  | 
| -  RTC_DCHECK_GE(1u, num_audio_streams_); | 
| +  RTC_DCHECK_GE(1, num_audio_streams_); | 
| if (num_audio_streams_ == 1) { | 
| RTC_DCHECK_LE(0, voe_send_.channel_id); | 
| AudioReceiveStream::Config audio_config; | 
|  |