| 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));
|
|
|