| Index: webrtc/api/peerconnectionendtoend_unittest.cc
|
| diff --git a/webrtc/api/peerconnectionendtoend_unittest.cc b/webrtc/api/peerconnectionendtoend_unittest.cc
|
| deleted file mode 100644
|
| index fedadd9e12a73880357caae0f4b658dc80e6e005..0000000000000000000000000000000000000000
|
| --- a/webrtc/api/peerconnectionendtoend_unittest.cc
|
| +++ /dev/null
|
| @@ -1,457 +0,0 @@
|
| -/*
|
| - * Copyright 2013 The WebRTC project authors. All Rights Reserved.
|
| - *
|
| - * Use of this source code is governed by a BSD-style license
|
| - * that can be found in the LICENSE file in the root of the source
|
| - * tree. An additional intellectual property rights grant can be found
|
| - * in the file PATENTS. All contributing project authors may
|
| - * be found in the AUTHORS file in the root of the source tree.
|
| - */
|
| -
|
| -#include <memory>
|
| -
|
| -#include "webrtc/api/test/peerconnectiontestwrapper.h"
|
| -// Notice that mockpeerconnectionobservers.h must be included after the above!
|
| -#include "webrtc/api/test/mockpeerconnectionobservers.h"
|
| -#ifdef WEBRTC_ANDROID
|
| -#include "webrtc/api/test/androidtestinitializer.h"
|
| -#endif
|
| -#include "webrtc/base/gunit.h"
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/base/ssladapter.h"
|
| -#include "webrtc/base/thread.h"
|
| -#include "webrtc/base/sslstreamadapter.h"
|
| -#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;
|
| -using webrtc::MediaStreamInterface;
|
| -using webrtc::PeerConnectionInterface;
|
| -
|
| -namespace {
|
| -
|
| -const int kMaxWait = 10000;
|
| -
|
| -} // namespace
|
| -
|
| -class PeerConnectionEndToEndTest
|
| - : public sigslot::has_slots<>,
|
| - public testing::Test {
|
| - public:
|
| - typedef std::vector<rtc::scoped_refptr<DataChannelInterface> >
|
| - DataChannelList;
|
| -
|
| - PeerConnectionEndToEndTest() {
|
| - RTC_CHECK(network_thread_.Start());
|
| - RTC_CHECK(worker_thread_.Start());
|
| - caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
|
| - "caller", &network_thread_, &worker_thread_);
|
| - callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
|
| - "callee", &network_thread_, &worker_thread_);
|
| - webrtc::PeerConnectionInterface::IceServer ice_server;
|
| - ice_server.uri = "stun:stun.l.google.com:19302";
|
| - config_.servers.push_back(ice_server);
|
| -
|
| -#ifdef WEBRTC_ANDROID
|
| - webrtc::InitializeAndroidObjects();
|
| -#endif
|
| - }
|
| -
|
| - void CreatePcs() {
|
| - CreatePcs(NULL);
|
| - }
|
| -
|
| - void CreatePcs(const MediaConstraintsInterface* pc_constraints) {
|
| - EXPECT_TRUE(caller_->CreatePc(pc_constraints, config_));
|
| - EXPECT_TRUE(callee_->CreatePc(pc_constraints, config_));
|
| - PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
|
| -
|
| - caller_->SignalOnDataChannel.connect(
|
| - this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel);
|
| - callee_->SignalOnDataChannel.connect(
|
| - this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel);
|
| - }
|
| -
|
| - void GetAndAddUserMedia() {
|
| - FakeConstraints audio_constraints;
|
| - FakeConstraints video_constraints;
|
| - GetAndAddUserMedia(true, audio_constraints, true, video_constraints);
|
| - }
|
| -
|
| - void GetAndAddUserMedia(bool audio, FakeConstraints audio_constraints,
|
| - bool video, FakeConstraints video_constraints) {
|
| - caller_->GetAndAddUserMedia(audio, audio_constraints,
|
| - video, video_constraints);
|
| - callee_->GetAndAddUserMedia(audio, audio_constraints,
|
| - video, video_constraints);
|
| - }
|
| -
|
| - void Negotiate() {
|
| - caller_->CreateOffer(NULL);
|
| - }
|
| -
|
| - void WaitForCallEstablished() {
|
| - caller_->WaitForCallEstablished();
|
| - callee_->WaitForCallEstablished();
|
| - }
|
| -
|
| - void WaitForConnection() {
|
| - caller_->WaitForConnection();
|
| - callee_->WaitForConnection();
|
| - }
|
| -
|
| - void OnCallerAddedDataChanel(DataChannelInterface* dc) {
|
| - caller_signaled_data_channels_.push_back(dc);
|
| - }
|
| -
|
| - void OnCalleeAddedDataChannel(DataChannelInterface* dc) {
|
| - callee_signaled_data_channels_.push_back(dc);
|
| - }
|
| -
|
| - // Tests that |dc1| and |dc2| can send to and receive from each other.
|
| - void TestDataChannelSendAndReceive(
|
| - DataChannelInterface* dc1, DataChannelInterface* dc2) {
|
| - std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer(
|
| - new webrtc::MockDataChannelObserver(dc1));
|
| -
|
| - std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer(
|
| - new webrtc::MockDataChannelObserver(dc2));
|
| -
|
| - static const std::string kDummyData = "abcdefg";
|
| - webrtc::DataBuffer buffer(kDummyData);
|
| - EXPECT_TRUE(dc1->Send(buffer));
|
| - EXPECT_EQ_WAIT(kDummyData, dc2_observer->last_message(), kMaxWait);
|
| -
|
| - EXPECT_TRUE(dc2->Send(buffer));
|
| - EXPECT_EQ_WAIT(kDummyData, dc1_observer->last_message(), kMaxWait);
|
| -
|
| - EXPECT_EQ(1U, dc1_observer->received_message_count());
|
| - EXPECT_EQ(1U, dc2_observer->received_message_count());
|
| - }
|
| -
|
| - void WaitForDataChannelsToOpen(DataChannelInterface* local_dc,
|
| - const DataChannelList& remote_dc_list,
|
| - size_t remote_dc_index) {
|
| - EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait);
|
| -
|
| - EXPECT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait);
|
| - EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
|
| - remote_dc_list[remote_dc_index]->state(),
|
| - kMaxWait);
|
| - EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id());
|
| - }
|
| -
|
| - void CloseDataChannels(DataChannelInterface* local_dc,
|
| - const DataChannelList& remote_dc_list,
|
| - size_t remote_dc_index) {
|
| - local_dc->Close();
|
| - EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait);
|
| - EXPECT_EQ_WAIT(DataChannelInterface::kClosed,
|
| - remote_dc_list[remote_dc_index]->state(),
|
| - kMaxWait);
|
| - }
|
| -
|
| - protected:
|
| - rtc::Thread network_thread_;
|
| - rtc::Thread worker_thread_;
|
| - rtc::scoped_refptr<PeerConnectionTestWrapper> caller_;
|
| - rtc::scoped_refptr<PeerConnectionTestWrapper> callee_;
|
| - DataChannelList caller_signaled_data_channels_;
|
| - DataChannelList callee_signaled_data_channels_;
|
| - webrtc::PeerConnectionInterface::RTCConfiguration config_;
|
| -};
|
| -
|
| -// Disabled for TSan v2, see
|
| -// https://bugs.chromium.org/p/webrtc/issues/detail?id=4719 for details.
|
| -// Disabled for Mac, see
|
| -// https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details.
|
| -#if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
|
| -TEST_F(PeerConnectionEndToEndTest, Call) {
|
| - CreatePcs();
|
| - GetAndAddUserMedia();
|
| - Negotiate();
|
| - WaitForCallEstablished();
|
| -}
|
| -#endif // if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
|
| -
|
| -#if !defined(ADDRESS_SANITIZER)
|
| -TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
|
| - FakeConstraints pc_constraints;
|
| - pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - false);
|
| - CreatePcs(&pc_constraints);
|
| - GetAndAddUserMedia();
|
| - Negotiate();
|
| - WaitForCallEstablished();
|
| -}
|
| -#endif // !defined(ADDRESS_SANITIZER)
|
| -
|
| -// 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;
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
| - caller_->CreateDataChannel("data", init));
|
| - rtc::scoped_refptr<DataChannelInterface> callee_dc(
|
| - callee_->CreateDataChannel("data", init));
|
| -
|
| - Negotiate();
|
| - WaitForConnection();
|
| -
|
| - WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
|
| - WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
|
| -
|
| - TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[0]);
|
| - TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
|
| -
|
| - CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
|
| - CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
|
| -}
|
| -
|
| -// 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;
|
| -
|
| - // This DataChannel is for creating the data content in the negotiation.
|
| - rtc::scoped_refptr<DataChannelInterface> dummy(
|
| - caller_->CreateDataChannel("data", init));
|
| - Negotiate();
|
| - WaitForConnection();
|
| -
|
| - // Wait for the data channel created pre-negotiation to be opened.
|
| - WaitForDataChannelsToOpen(dummy, callee_signaled_data_channels_, 0);
|
| -
|
| - // Create new DataChannels after the negotiation and verify their states.
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
| - caller_->CreateDataChannel("hello", init));
|
| - rtc::scoped_refptr<DataChannelInterface> callee_dc(
|
| - callee_->CreateDataChannel("hello", init));
|
| -
|
| - WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
|
| - WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
|
| -
|
| - TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
|
| - TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
|
| -
|
| - CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
|
| - CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
|
| -}
|
| -
|
| -// 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;
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
|
| - caller_->CreateDataChannel("data", init));
|
| - rtc::scoped_refptr<DataChannelInterface> callee_dc_1(
|
| - callee_->CreateDataChannel("data", init));
|
| -
|
| - Negotiate();
|
| - WaitForConnection();
|
| -
|
| - EXPECT_EQ(1U, caller_dc_1->id() % 2);
|
| - EXPECT_EQ(0U, callee_dc_1->id() % 2);
|
| -
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
|
| - caller_->CreateDataChannel("data", init));
|
| - rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
|
| - callee_->CreateDataChannel("data", init));
|
| -
|
| - EXPECT_EQ(1U, caller_dc_2->id() % 2);
|
| - EXPECT_EQ(0U, callee_dc_2->id() % 2);
|
| -}
|
| -
|
| -// Verifies that the message is received by the right remote DataChannel when
|
| -// there are multiple DataChannels.
|
| -TEST_F(PeerConnectionEndToEndTest,
|
| - MessageTransferBetweenTwoPairsOfDataChannels) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| -
|
| - CreatePcs();
|
| -
|
| - webrtc::DataChannelInit init;
|
| -
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
|
| - caller_->CreateDataChannel("data", init));
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
|
| - caller_->CreateDataChannel("data", init));
|
| -
|
| - Negotiate();
|
| - WaitForConnection();
|
| - WaitForDataChannelsToOpen(caller_dc_1, callee_signaled_data_channels_, 0);
|
| - WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1);
|
| -
|
| - std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
|
| - new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0]));
|
| -
|
| - std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
|
| - new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1]));
|
| -
|
| - const std::string message_1 = "hello 1";
|
| - const std::string message_2 = "hello 2";
|
| -
|
| - caller_dc_1->Send(webrtc::DataBuffer(message_1));
|
| - EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
|
| -
|
| - caller_dc_2->Send(webrtc::DataBuffer(message_2));
|
| - EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
|
| -
|
| - EXPECT_EQ(1U, dc_1_observer->received_message_count());
|
| - EXPECT_EQ(1U, dc_2_observer->received_message_count());
|
| -}
|
| -
|
| -#ifdef HAVE_QUIC
|
| -// Test that QUIC data channels can be used and that messages go to the correct
|
| -// remote data channel when both peers want to use QUIC. It is assumed that the
|
| -// application has externally negotiated the data channel parameters.
|
| -TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) {
|
| - config_.enable_quic = true;
|
| - CreatePcs();
|
| -
|
| - webrtc::DataChannelInit init_1;
|
| - init_1.id = 0;
|
| - init_1.ordered = false;
|
| - init_1.reliable = true;
|
| -
|
| - webrtc::DataChannelInit init_2;
|
| - init_2.id = 1;
|
| - init_2.ordered = false;
|
| - init_2.reliable = true;
|
| -
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
|
| - caller_->CreateDataChannel("data", init_1));
|
| - ASSERT_NE(nullptr, caller_dc_1);
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
|
| - caller_->CreateDataChannel("data", init_2));
|
| - ASSERT_NE(nullptr, caller_dc_2);
|
| - rtc::scoped_refptr<DataChannelInterface> callee_dc_1(
|
| - callee_->CreateDataChannel("data", init_1));
|
| - ASSERT_NE(nullptr, callee_dc_1);
|
| - rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
|
| - callee_->CreateDataChannel("data", init_2));
|
| - ASSERT_NE(nullptr, callee_dc_2);
|
| -
|
| - Negotiate();
|
| - WaitForConnection();
|
| - EXPECT_TRUE_WAIT(caller_dc_1->state() == webrtc::DataChannelInterface::kOpen,
|
| - kMaxWait);
|
| - EXPECT_TRUE_WAIT(callee_dc_1->state() == webrtc::DataChannelInterface::kOpen,
|
| - kMaxWait);
|
| - EXPECT_TRUE_WAIT(caller_dc_2->state() == webrtc::DataChannelInterface::kOpen,
|
| - kMaxWait);
|
| - EXPECT_TRUE_WAIT(callee_dc_2->state() == webrtc::DataChannelInterface::kOpen,
|
| - kMaxWait);
|
| -
|
| - std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
|
| - new webrtc::MockDataChannelObserver(callee_dc_1.get()));
|
| -
|
| - std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
|
| - new webrtc::MockDataChannelObserver(callee_dc_2.get()));
|
| -
|
| - const std::string message_1 = "hello 1";
|
| - const std::string message_2 = "hello 2";
|
| -
|
| - // Send data from caller to callee.
|
| - caller_dc_1->Send(webrtc::DataBuffer(message_1));
|
| - EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
|
| -
|
| - caller_dc_2->Send(webrtc::DataBuffer(message_2));
|
| - EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
|
| -
|
| - EXPECT_EQ(1U, dc_1_observer->received_message_count());
|
| - EXPECT_EQ(1U, dc_2_observer->received_message_count());
|
| -
|
| - // Send data from callee to caller.
|
| - dc_1_observer.reset(new webrtc::MockDataChannelObserver(caller_dc_1.get()));
|
| - dc_2_observer.reset(new webrtc::MockDataChannelObserver(caller_dc_2.get()));
|
| -
|
| - callee_dc_1->Send(webrtc::DataBuffer(message_1));
|
| - EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
|
| -
|
| - callee_dc_2->Send(webrtc::DataBuffer(message_2));
|
| - EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
|
| -
|
| - EXPECT_EQ(1U, dc_1_observer->received_message_count());
|
| - EXPECT_EQ(1U, dc_2_observer->received_message_count());
|
| -}
|
| -#endif // HAVE_QUIC
|
| -
|
| -// Verifies that a DataChannel added from an OPEN message functions after
|
| -// a channel has been previously closed (webrtc issue 3778).
|
| -// This previously failed because the new channel re-uses the ID of the closed
|
| -// channel, and the closed channel was incorrectly still assigned to the id.
|
| -// TODO(deadbeef): This is disabled because there's currently a race condition
|
| -// caused by the fact that a data channel signals that it's closed before it
|
| -// really is. Re-enable this test once that's fixed.
|
| -// 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;
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
| - caller_->CreateDataChannel("data", init));
|
| -
|
| - Negotiate();
|
| - WaitForConnection();
|
| -
|
| - WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
|
| - CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
|
| -
|
| - // Create a new channel and ensure it works after closing the previous one.
|
| - caller_dc = caller_->CreateDataChannel("data2", init);
|
| -
|
| - WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
|
| - TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
|
| -
|
| - CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
|
| -}
|
| -
|
| -// This tests that if a data channel is closed remotely while not referenced
|
| -// by the application (meaning only the PeerConnection contributes to its
|
| -// 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;
|
| - rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
| - caller_->CreateDataChannel("data", init));
|
| -
|
| - Negotiate();
|
| - WaitForConnection();
|
| -
|
| - WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
|
| - // This removes the reference to the remote data channel that we hold.
|
| - callee_signaled_data_channels_.clear();
|
| - caller_dc->Close();
|
| - EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
|
| -
|
| - // Wait for a bit longer so the remote data channel will receive the
|
| - // close message and be destroyed.
|
| - rtc::Thread::Current()->ProcessMessages(100);
|
| -}
|
|
|