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Unified Diff: webrtc/video/bitrate_estimator_tests.cc

Issue 1335923002: Add RTC_ prefix to (D)CHECKs and related macros. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 5 years, 3 months ago
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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));
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