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Diffstat (limited to 'third_party/libwebrtc/video/end_to_end_tests/stats_tests.cc')
-rw-r--r--third_party/libwebrtc/video/end_to_end_tests/stats_tests.cc8
1 files changed, 4 insertions, 4 deletions
diff --git a/third_party/libwebrtc/video/end_to_end_tests/stats_tests.cc b/third_party/libwebrtc/video/end_to_end_tests/stats_tests.cc
index cc0b328b2b..d6820eeac2 100644
--- a/third_party/libwebrtc/video/end_to_end_tests/stats_tests.cc
+++ b/third_party/libwebrtc/video/end_to_end_tests/stats_tests.cc
@@ -518,9 +518,9 @@ TEST_F(StatsEndToEndTest, MAYBE_ContentTypeSwitches) {
metrics::Reset();
- CallConfig send_config(send_event_log_.get());
+ CallConfig send_config = SendCallConfig();
test.ModifySenderBitrateConfig(&send_config.bitrate_config);
- CallConfig recv_config(recv_event_log_.get());
+ CallConfig recv_config = RecvCallConfig();
test.ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
VideoEncoderConfig encoder_config_with_screenshare;
@@ -732,13 +732,13 @@ TEST_F(StatsEndToEndTest, CallReportsRttForSender) {
Start();
});
- int64_t start_time_ms = clock_->TimeInMilliseconds();
+ int64_t start_time_ms = env().clock().TimeInMilliseconds();
while (true) {
Call::Stats stats;
SendTask(task_queue(),
[this, &stats]() { stats = sender_call_->GetStats(); });
ASSERT_GE(start_time_ms + test::VideoTestConstants::kDefaultTimeout.ms(),
- clock_->TimeInMilliseconds())
+ env().clock().TimeInMilliseconds())
<< "No RTT stats before timeout!";
if (stats.rtt_ms != -1) {
// To avoid failures caused by rounding or minor ntp clock adjustments,