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Parameterize PeerConnection data channel tests for Unified Plan

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Bug: webrtc:8765
Change-Id: Ifac06b2f36230adb093169af0a88dda5463a1216
Reviewed-on: https://webrtc-review.googlesource.com/40503
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21704}
This commit is contained in:
Steve Anton
2018-01-19 15:23:40 -08:00
committed by Commit Bot
parent 71182f4ded
commit dbf9d03204
2 changed files with 62 additions and 30 deletions
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78
pc/peerconnection_datachannel_unittest.cc
View File

@ -16,6 +16,7 @@
#include "pc/peerconnection.h"
#include "pc/peerconnectionfactory.h"
#include "pc/peerconnectionwrapper.h"
#include "pc/sdputils.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#endif
@ -84,12 +85,14 @@ class PeerConnectionWrapperForDataChannelTest : public PeerConnectionWrapper {
FakeSctpTransportFactory* sctp_transport_factory_ = nullptr;
};
class PeerConnectionDataChannelTest : public ::testing::Test {
class PeerConnectionDataChannelBaseTest : public ::testing::Test {
protected:
typedef std::unique_ptr<PeerConnectionWrapperForDataChannelTest> WrapperPtr;
PeerConnectionDataChannelTest()
: vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
explicit PeerConnectionDataChannelBaseTest(SdpSemantics sdp_semantics)
: vss_(new rtc::VirtualSocketServer()),
main_(vss_.get()),
sdp_semantics_(sdp_semantics) {
#ifdef WEBRTC_ANDROID
InitializeAndroidObjects();
#endif
@ -112,8 +115,10 @@ class PeerConnectionDataChannelTest : public ::testing::Test {
pc_factory->SetOptions(factory_options);
RTC_CHECK(pc_factory->Initialize());
auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
auto pc = pc_factory->CreatePeerConnection(config, nullptr, nullptr,
observer.get());
RTCConfiguration modified_config = config;
modified_config.sdp_semantics = sdp_semantics_;
auto pc = pc_factory->CreatePeerConnection(modified_config, nullptr,
nullptr, observer.get());
if (!pc) {
return nullptr;
}
@ -153,9 +158,18 @@ class PeerConnectionDataChannelTest : public ::testing::Test {
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
const SdpSemantics sdp_semantics_;
};
TEST_F(PeerConnectionDataChannelTest,
class PeerConnectionDataChannelTest
: public PeerConnectionDataChannelBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionDataChannelTest()
: PeerConnectionDataChannelBaseTest(GetParam()) {}
};
TEST_P(PeerConnectionDataChannelTest,
NoSctpTransportCreatedIfRtpDataChannelEnabled) {
RTCConfiguration config;
config.enable_rtp_data_channel = true;
@ -165,7 +179,7 @@ TEST_F(PeerConnectionDataChannelTest,
EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
}
TEST_F(PeerConnectionDataChannelTest,
TEST_P(PeerConnectionDataChannelTest,
RtpDataChannelCreatedEvenIfSctpAvailable) {
RTCConfiguration config;
config.enable_rtp_data_channel = true;
@ -179,7 +193,7 @@ TEST_F(PeerConnectionDataChannelTest,
// Test that sctp_content_name/sctp_transport_name (used for stats) are correct
// before and after BUNDLE is negotiated.
TEST_F(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
TEST_P(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
@ -193,12 +207,24 @@ TEST_F(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
caller->AddAudioTrack("a");
caller->AddVideoTrack("v");
caller->pc()->CreateDataChannel("dc", nullptr);
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
auto offer = caller->CreateOffer();
const auto& offer_contents = offer->description()->contents();
ASSERT_EQ(cricket::MEDIA_TYPE_AUDIO,
offer_contents[0].media_description()->type());
std::string audio_mid = offer_contents[0].name;
ASSERT_EQ(cricket::MEDIA_TYPE_DATA,
offer_contents[2].media_description()->type());
std::string data_mid = offer_contents[2].name;
ASSERT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
ASSERT_TRUE(caller->sctp_content_name());
EXPECT_EQ(cricket::CN_DATA, *caller->sctp_content_name());
EXPECT_EQ(data_mid, *caller->sctp_content_name());
ASSERT_TRUE(caller->sctp_transport_name());
EXPECT_EQ(cricket::CN_DATA, *caller->sctp_transport_name());
EXPECT_EQ(data_mid, *caller->sctp_transport_name());
// Create answer that finishes BUNDLE negotiation, which means everything
// should be bundled on the first transport (audio).
@ -208,12 +234,12 @@ TEST_F(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
ASSERT_TRUE(caller->sctp_content_name());
EXPECT_EQ(cricket::CN_DATA, *caller->sctp_content_name());
EXPECT_EQ(data_mid, *caller->sctp_content_name());
ASSERT_TRUE(caller->sctp_transport_name());
EXPECT_EQ(cricket::CN_AUDIO, *caller->sctp_transport_name());
EXPECT_EQ(audio_mid, *caller->sctp_transport_name());
}
TEST_F(PeerConnectionDataChannelTest,
TEST_P(PeerConnectionDataChannelTest,
CreateOfferWithNoDataChannelsGivesNoDataSection) {
auto caller = CreatePeerConnection();
auto offer = caller->CreateOffer();
@ -222,7 +248,7 @@ TEST_F(PeerConnectionDataChannelTest,
EXPECT_FALSE(offer->description()->GetTransportInfoByName(cricket::CN_DATA));
}
TEST_F(PeerConnectionDataChannelTest,
TEST_P(PeerConnectionDataChannelTest,
CreateAnswerWithRemoteSctpDataChannelIncludesDataSection) {
auto caller = CreatePeerConnectionWithDataChannel();
auto callee = CreatePeerConnection();
@ -231,14 +257,14 @@ TEST_F(PeerConnectionDataChannelTest,
auto answer = callee->CreateAnswer();
ASSERT_TRUE(answer);
auto* data_content =
answer->description()->GetContentByName(cricket::CN_DATA);
auto* data_content = cricket::GetFirstDataContent(answer->description());
ASSERT_TRUE(data_content);
EXPECT_FALSE(data_content->rejected);
EXPECT_TRUE(answer->description()->GetTransportInfoByName(cricket::CN_DATA));
EXPECT_TRUE(
answer->description()->GetTransportInfoByName(data_content->name));
}
TEST_F(PeerConnectionDataChannelTest,
TEST_P(PeerConnectionDataChannelTest,
CreateDataChannelWithDtlsDisabledSucceeds) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
@ -247,7 +273,7 @@ TEST_F(PeerConnectionDataChannelTest,
EXPECT_TRUE(caller->pc()->CreateDataChannel("dc", nullptr));
}
TEST_F(PeerConnectionDataChannelTest, CreateDataChannelWithSctpDisabledFails) {
TEST_P(PeerConnectionDataChannelTest, CreateDataChannelWithSctpDisabledFails) {
PeerConnectionFactoryInterface::Options options;
options.disable_sctp_data_channels = true;
auto caller = CreatePeerConnection(RTCConfiguration(), options);
@ -258,7 +284,7 @@ TEST_F(PeerConnectionDataChannelTest, CreateDataChannelWithSctpDisabledFails) {
// Test that if a callee has SCTP disabled and receives an offer with an SCTP
// data channel, the data section is rejected and no SCTP transport is created
// on the callee.
TEST_F(PeerConnectionDataChannelTest,
TEST_P(PeerConnectionDataChannelTest,
DataSectionRejectedIfCalleeHasSctpDisabled) {
auto caller = CreatePeerConnectionWithDataChannel();
PeerConnectionFactoryInterface::Options options;
@ -270,13 +296,12 @@ TEST_F(PeerConnectionDataChannelTest,
EXPECT_FALSE(callee->sctp_transport_factory()->last_fake_sctp_transport());
auto answer = callee->CreateAnswer();
auto* data_content =
answer->description()->GetContentByName(cricket::CN_DATA);
auto* data_content = cricket::GetFirstDataContent(answer->description());
ASSERT_TRUE(data_content);
EXPECT_TRUE(data_content->rejected);
}
TEST_F(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
TEST_P(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
constexpr int kNewSendPort = 9998;
constexpr int kNewRecvPort = 7775;
@ -298,4 +323,9 @@ TEST_F(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
EXPECT_EQ(kNewRecvPort, callee_transport->local_port());
}
INSTANTIATE_TEST_CASE_P(PeerConnectionDataChannelTest,
PeerConnectionDataChannelTest,
Values(SdpSemantics::kPlanB,
SdpSemantics::kUnifiedPlan));
} // namespace webrtc