Fix for sanitizer bot failure in AudioUsesAbsSendTimeExtension

Bug: webrtc:10904
Change-Id: Id37a88afd85c522a7973f6dc9e8dd331a04d3fae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150325
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28981}
This commit is contained in:
Sebastian Jansson 2019-08-23 14:52:14 +02:00 committed by Commit Bot
parent 85ba9972c4
commit 7f65932073

View File

@ -72,6 +72,8 @@ TEST(RemoteEstimateEndToEnd, OfferedCapabilityIsInAnswer) {
TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
ScopedFieldTrials trials("WebRTC-KeepAbsSendTimeExtension/Enabled/");
// Defined before PeerScenario so it gets destructed after, to avoid use after free.
rtc::Event received_abs_send_time;
PeerScenario s;
auto* caller = s.CreateClient(PeerScenarioClient::Config());
@ -99,7 +101,6 @@ TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
offer_exchange_done.Set();
});
EXPECT_TRUE(s.WaitAndProcess(&offer_exchange_done));
rtc::Event received_abs_send_time;
send_node->router()->SetWatcher(
[extension_map, &received_abs_send_time](const EmulatedIpPacket& packet) {
auto extensions = GetRtpPacketExtensions(packet.data, extension_map);