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; |