Index: webrtc/video/bitrate_estimator_tests.cc |
diff --git a/webrtc/video/bitrate_estimator_tests.cc b/webrtc/video/bitrate_estimator_tests.cc |
index 059de351d3c1b2b6a093fecc9d8b251ca8b312ff..f7044ae33e788712eb2a1629905d4125ea1570ec 100644 |
--- a/webrtc/video/bitrate_estimator_tests.cc |
+++ b/webrtc/video/bitrate_estimator_tests.cc |
@@ -188,7 +188,7 @@ class BitrateEstimatorTest : public test::CallTest { |
test_->send_config_.encoder_settings.encoder = &fake_encoder_; |
send_stream_ = test_->sender_call_->CreateVideoSendStream( |
test_->send_config_, test_->encoder_config_); |
- DCHECK_EQ(1u, test_->encoder_config_.streams.size()); |
+ RTC_DCHECK_EQ(1u, test_->encoder_config_.streams.size()); |
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create( |
send_stream_->Input(), |
test_->encoder_config_.streams[0].width, |
@@ -201,9 +201,9 @@ class BitrateEstimatorTest : public test::CallTest { |
if (receive_audio) { |
AudioReceiveStream::Config receive_config; |
receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0]; |
- // Bogus non-default id to prevent hitting a DCHECK when creating the |
- // AudioReceiveStream. Every receive stream has to correspond to an |
- // underlying channel id. |
+ // Bogus non-default id to prevent hitting a RTC_DCHECK when creating |
+ // the AudioReceiveStream. Every receive stream has to correspond to |
+ // an underlying channel id. |
receive_config.voe_channel_id = 0; |
receive_config.rtp.extensions.push_back( |
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId)); |