Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ )
Reason for revert:
Broke chromium build, due to a config being removed. Will add it back and remove the dependency in a chromium CL.
Original issue's description:
> Removing #defines previously used for building without BoringSSL/OpenSSL.
>
> These defines don't work any more, so they only cause confusion:
>
> FEATURE_ENABLE_SSL
> HAVE_OPENSSL_SSL_H
> SSL_USE_OPENSSL
>
> BUG=webrtc:7025
>
> Review-Url: https://codereview.webrtc.org/2640513002
> Cr-Commit-Position: refs/heads/master@{#16196}
> Committed: eaa826c2ee
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2648003003
Cr-Commit-Position: refs/heads/master@{#16197}
This commit is contained in:
@ -24,6 +24,12 @@
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#include "webrtc/base/stringencode.h"
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#include "webrtc/base/stringutils.h"
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#define MAYBE_SKIP_TEST(feature) \
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if (!(feature())) { \
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LOG(LS_INFO) << "Feature disabled... skipping"; \
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return; \
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}
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using webrtc::DataChannelInterface;
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using webrtc::FakeConstraints;
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using webrtc::MediaConstraintsInterface;
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@ -192,6 +198,8 @@ TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
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// Verifies that a DataChannel created before the negotiation can transition to
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// "OPEN" and transfer data.
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TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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CreatePcs();
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webrtc::DataChannelInit init;
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@ -216,6 +224,8 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
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// Verifies that a DataChannel created after the negotiation can transition to
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// "OPEN" and transfer data.
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TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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CreatePcs();
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webrtc::DataChannelInit init;
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@ -247,6 +257,8 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
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// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
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TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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CreatePcs();
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webrtc::DataChannelInit init;
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@ -274,6 +286,8 @@ TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
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// there are multiple DataChannels.
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TEST_F(PeerConnectionEndToEndTest,
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MessageTransferBetweenTwoPairsOfDataChannels) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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CreatePcs();
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webrtc::DataChannelInit init;
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@ -395,6 +409,8 @@ TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) {
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// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
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TEST_F(PeerConnectionEndToEndTest,
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DISABLED_DataChannelFromOpenWorksAfterClose) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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CreatePcs();
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webrtc::DataChannelInit init;
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@ -421,6 +437,8 @@ TEST_F(PeerConnectionEndToEndTest,
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// reference count), no memory access violation will occur.
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// See: https://code.google.com/p/chromium/issues/detail?id=565048
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TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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CreatePcs();
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webrtc::DataChannelInit init;
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