Index: webrtc/api/peerconnectionendtoend_unittest.cc |
diff --git a/webrtc/api/peerconnectionendtoend_unittest.cc b/webrtc/api/peerconnectionendtoend_unittest.cc |
index 4110db07c92e18aaa1ff31e3dc67c70f38b4d952..436e7bf02b45cd804442820dec1a7463aaa7d4db 100644 |
--- a/webrtc/api/peerconnectionendtoend_unittest.cc |
+++ b/webrtc/api/peerconnectionendtoend_unittest.cc |
@@ -24,12 +24,6 @@ |
#include "webrtc/base/stringencode.h" |
#include "webrtc/base/stringutils.h" |
-#define MAYBE_SKIP_TEST(feature) \ |
- if (!(feature())) { \ |
- LOG(LS_INFO) << "Feature disabled... skipping"; \ |
- return; \ |
- } |
- |
using webrtc::DataChannelInterface; |
using webrtc::FakeConstraints; |
using webrtc::MediaConstraintsInterface; |
@@ -198,8 +192,6 @@ TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) { |
// Verifies that a DataChannel created before the negotiation can transition to |
// "OPEN" and transfer data. |
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
CreatePcs(); |
webrtc::DataChannelInit init; |
@@ -224,8 +216,6 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { |
// Verifies that a DataChannel created after the negotiation can transition to |
// "OPEN" and transfer data. |
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
CreatePcs(); |
webrtc::DataChannelInit init; |
@@ -257,8 +247,6 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) { |
// Verifies that DataChannel IDs are even/odd based on the DTLS roles. |
TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
CreatePcs(); |
webrtc::DataChannelInit init; |
@@ -286,8 +274,6 @@ TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) { |
// there are multiple DataChannels. |
TEST_F(PeerConnectionEndToEndTest, |
MessageTransferBetweenTwoPairsOfDataChannels) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
CreatePcs(); |
webrtc::DataChannelInit init; |
@@ -409,8 +395,6 @@ TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) { |
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453 |
TEST_F(PeerConnectionEndToEndTest, |
DISABLED_DataChannelFromOpenWorksAfterClose) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
CreatePcs(); |
webrtc::DataChannelInit init; |
@@ -437,8 +421,6 @@ TEST_F(PeerConnectionEndToEndTest, |
// reference count), no memory access violation will occur. |
// See: https://code.google.com/p/chromium/issues/detail?id=565048 |
TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) { |
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
- |
CreatePcs(); |
webrtc::DataChannelInit init; |