Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(301)

Unified Diff: webrtc/video/end_to_end_tests.cc

Issue 2348623003: Unify rtcp packet setters (Closed)
Patch Set: Comments Created 4 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/video/end_to_end_tests.cc
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 55d5cc073d045d4499c0d1f2681c499bf9ed620f..ced9ddfd40b4d3712cf135564ed9b60898a3f849 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -528,10 +528,10 @@ TEST_F(EndToEndTest, ReceivesNackAndRetransmitsAudio) {
} else {
// Send a NACK as often as necessary until retransmission is received.
rtcp::Nack nack;
- nack.From(local_ssrc_);
- nack.To(remote_ssrc_);
+ nack.SetSenderSsrc(local_ssrc_);
+ nack.SetMediaSsrc(remote_ssrc_);
uint16_t nack_list[] = {*sequence_number_to_retransmit_};
- nack.WithList(nack_list, 1);
+ nack.SetPacketIds(nack_list, 1);
rtc::Buffer buffer = nack.Build();
EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));

Powered by Google App Engine
This is Rietveld 408576698