Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(305)

Unified Diff: webrtc/audio/audio_receive_stream_unittest.cc

Issue 1608563005: Enable transport seq num extension on receive channel to suppress log warning. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add test. Created 4 years, 11 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/audio/audio_receive_stream_unittest.cc
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index eb008b3045408c32cedd5a1d9b9d2cec5208a32f..b241bed6b51e259c8e095e449bb5d150485a0433 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>
the sun 2016/01/21 12:24:52 lint warning?
stefan-webrtc 2016/01/21 13:17:43 Yes
#include "testing/gtest/include/gtest/gtest.h"
@@ -90,6 +91,9 @@ struct ConfigHelper {
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 @@ struct ConfigHelper {
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 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
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());

Powered by Google App Engine
This is Rietveld 408576698