Enable transport seq num extension on receive channel to suppress log warning.
TBR=pbos@webrtc.org
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1608563005
Cr-Commit-Position: refs/heads/master@{#11338}
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index eb008b3..b241bed 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -9,6 +9,7 @@
*/
#include <string>
+#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
@@ -90,6 +91,9 @@
EXPECT_CALL(*channel_proxy_,
SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
+ EXPECT_CALL(*channel_proxy_, EnableReceiveTransportSequenceNumber(
+ kTransportSequenceNumberId))
+ .Times(1);
EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(
nullptr, nullptr, &packet_router_))
.Times(1);
@@ -107,6 +111,8 @@
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+ stream_config_.rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
}
MockCongestionController* congestion_controller() {
@@ -261,8 +267,6 @@
ConfigHelper helper;
helper.config().combined_audio_video_bwe = true;
helper.config().rtp.transport_cc = true;
- helper.config().rtp.extensions.push_back(RtpExtension(
- RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
helper.SetupMockForBweFeedback(true);
internal::AudioReceiveStream recv_stream(
helper.congestion_controller(), helper.config(), helper.audio_state());