| 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;
|
|
|