diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/data_channel_unittest.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/data_channel_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/pc/data_channel_unittest.cc | 764 |
1 files changed, 764 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/data_channel_unittest.cc b/third_party/libwebrtc/pc/data_channel_unittest.cc new file mode 100644 index 0000000000..7b3e865d86 --- /dev/null +++ b/third_party/libwebrtc/pc/data_channel_unittest.cc @@ -0,0 +1,764 @@ +/* + * 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 <stdint.h> +#include <string.h> + +#include <memory> +#include <string> +#include <vector> + +#include "api/data_channel_interface.h" +#include "api/rtc_error.h" +#include "api/scoped_refptr.h" +#include "api/transport/data_channel_transport_interface.h" +#include "media/base/media_channel.h" +#include "media/sctp/sctp_transport_internal.h" +#include "pc/sctp_data_channel.h" +#include "pc/sctp_utils.h" +#include "pc/test/fake_data_channel_controller.h" +#include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/gunit.h" +#include "rtc_base/ssl_stream_adapter.h" +#include "rtc_base/third_party/sigslot/sigslot.h" +#include "rtc_base/thread.h" +#include "test/gtest.h" + +using webrtc::DataChannelInterface; +using webrtc::SctpDataChannel; +using webrtc::SctpSidAllocator; + +static constexpr int kDefaultTimeout = 10000; + +class FakeDataChannelObserver : public webrtc::DataChannelObserver { + public: + FakeDataChannelObserver() + : messages_received_(0), + on_state_change_count_(0), + on_buffered_amount_change_count_(0) {} + + void OnStateChange() { ++on_state_change_count_; } + + void OnBufferedAmountChange(uint64_t previous_amount) { + ++on_buffered_amount_change_count_; + } + + void OnMessage(const webrtc::DataBuffer& buffer) { ++messages_received_; } + + size_t messages_received() const { return messages_received_; } + + void ResetOnStateChangeCount() { on_state_change_count_ = 0; } + + void ResetOnBufferedAmountChangeCount() { + on_buffered_amount_change_count_ = 0; + } + + size_t on_state_change_count() const { return on_state_change_count_; } + + size_t on_buffered_amount_change_count() const { + return on_buffered_amount_change_count_; + } + + private: + size_t messages_received_; + size_t on_state_change_count_; + size_t on_buffered_amount_change_count_; +}; + +// TODO(deadbeef): The fact that these tests use a fake controller makes them +// not too valuable. Should rewrite using the +// peerconnection_datachannel_unittest.cc infrastructure. +// TODO(bugs.webrtc.org/11547): Incorporate a dedicated network thread. +class SctpDataChannelTest : public ::testing::Test { + protected: + SctpDataChannelTest() + : controller_(new FakeDataChannelController()), + webrtc_data_channel_(SctpDataChannel::Create(controller_.get(), + "test", + init_, + rtc::Thread::Current(), + rtc::Thread::Current())) {} + + void SetChannelReady() { + controller_->set_transport_available(true); + webrtc_data_channel_->OnTransportChannelCreated(); + if (webrtc_data_channel_->id() < 0) { + webrtc_data_channel_->SetSctpSid(0); + } + controller_->set_ready_to_send(true); + } + + void AddObserver() { + observer_.reset(new FakeDataChannelObserver()); + webrtc_data_channel_->RegisterObserver(observer_.get()); + } + + rtc::AutoThread main_thread_; + webrtc::InternalDataChannelInit init_; + std::unique_ptr<FakeDataChannelController> controller_; + std::unique_ptr<FakeDataChannelObserver> observer_; + rtc::scoped_refptr<SctpDataChannel> webrtc_data_channel_; +}; + +class StateSignalsListener : public sigslot::has_slots<> { + public: + int opened_count() const { return opened_count_; } + int closed_count() const { return closed_count_; } + + void OnSignalOpened(DataChannelInterface* data_channel) { ++opened_count_; } + + void OnSignalClosed(DataChannelInterface* data_channel) { ++closed_count_; } + + private: + int opened_count_ = 0; + int closed_count_ = 0; +}; + +// Verifies that the data channel is connected to the transport after creation. +TEST_F(SctpDataChannelTest, ConnectedToTransportOnCreated) { + controller_->set_transport_available(true); + rtc::scoped_refptr<SctpDataChannel> dc = + SctpDataChannel::Create(controller_.get(), "test1", init_, + rtc::Thread::Current(), rtc::Thread::Current()); + + EXPECT_TRUE(controller_->IsConnected(dc.get())); + // The sid is not set yet, so it should not have added the streams. + EXPECT_FALSE(controller_->IsSendStreamAdded(dc->id())); + EXPECT_FALSE(controller_->IsRecvStreamAdded(dc->id())); + + dc->SetSctpSid(0); + EXPECT_TRUE(controller_->IsSendStreamAdded(dc->id())); + EXPECT_TRUE(controller_->IsRecvStreamAdded(dc->id())); +} + +// Verifies that the data channel is connected to the transport if the transport +// is not available initially and becomes available later. +TEST_F(SctpDataChannelTest, ConnectedAfterTransportBecomesAvailable) { + EXPECT_FALSE(controller_->IsConnected(webrtc_data_channel_.get())); + + controller_->set_transport_available(true); + webrtc_data_channel_->OnTransportChannelCreated(); + EXPECT_TRUE(controller_->IsConnected(webrtc_data_channel_.get())); +} + +// Tests the state of the data channel. +TEST_F(SctpDataChannelTest, StateTransition) { + StateSignalsListener state_signals_listener; + webrtc_data_channel_->SignalOpened.connect( + &state_signals_listener, &StateSignalsListener::OnSignalOpened); + webrtc_data_channel_->SignalClosed.connect( + &state_signals_listener, &StateSignalsListener::OnSignalClosed); + EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, + webrtc_data_channel_->state()); + EXPECT_EQ(state_signals_listener.opened_count(), 0); + EXPECT_EQ(state_signals_listener.closed_count(), 0); + SetChannelReady(); + + EXPECT_EQ(webrtc::DataChannelInterface::kOpen, webrtc_data_channel_->state()); + EXPECT_EQ(state_signals_listener.opened_count(), 1); + EXPECT_EQ(state_signals_listener.closed_count(), 0); + webrtc_data_channel_->Close(); + EXPECT_EQ(webrtc::DataChannelInterface::kClosed, + webrtc_data_channel_->state()); + EXPECT_TRUE(webrtc_data_channel_->error().ok()); + EXPECT_EQ(state_signals_listener.opened_count(), 1); + EXPECT_EQ(state_signals_listener.closed_count(), 1); + // Verifies that it's disconnected from the transport. + EXPECT_FALSE(controller_->IsConnected(webrtc_data_channel_.get())); +} + +// Tests that DataChannel::buffered_amount() is correct after the channel is +// blocked. +TEST_F(SctpDataChannelTest, BufferedAmountWhenBlocked) { + AddObserver(); + SetChannelReady(); + webrtc::DataBuffer buffer("abcd"); + EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); + size_t successful_send_count = 1; + + EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); + EXPECT_EQ(successful_send_count, + observer_->on_buffered_amount_change_count()); + + controller_->set_send_blocked(true); + + const int number_of_packets = 3; + for (int i = 0; i < number_of_packets; ++i) { + EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); + } + EXPECT_EQ(buffer.data.size() * number_of_packets, + webrtc_data_channel_->buffered_amount()); + EXPECT_EQ(successful_send_count, + observer_->on_buffered_amount_change_count()); + + controller_->set_send_blocked(false); + successful_send_count += number_of_packets; + EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); + EXPECT_EQ(successful_send_count, + observer_->on_buffered_amount_change_count()); +} + +// Tests that the queued data are sent when the channel transitions from blocked +// to unblocked. +TEST_F(SctpDataChannelTest, QueuedDataSentWhenUnblocked) { + AddObserver(); + SetChannelReady(); + webrtc::DataBuffer buffer("abcd"); + controller_->set_send_blocked(true); + EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); + + EXPECT_EQ(0U, observer_->on_buffered_amount_change_count()); + + controller_->set_send_blocked(false); + SetChannelReady(); + EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); + EXPECT_EQ(1U, observer_->on_buffered_amount_change_count()); +} + +// Tests that no crash when the channel is blocked right away while trying to +// send queued data. +TEST_F(SctpDataChannelTest, BlockedWhenSendQueuedDataNoCrash) { + AddObserver(); + SetChannelReady(); + webrtc::DataBuffer buffer("abcd"); + controller_->set_send_blocked(true); + EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); + EXPECT_EQ(0U, observer_->on_buffered_amount_change_count()); + + // Set channel ready while it is still blocked. + SetChannelReady(); + EXPECT_EQ(buffer.size(), webrtc_data_channel_->buffered_amount()); + EXPECT_EQ(0U, observer_->on_buffered_amount_change_count()); + + // Unblock the channel to send queued data again, there should be no crash. + controller_->set_send_blocked(false); + SetChannelReady(); + EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); + EXPECT_EQ(1U, observer_->on_buffered_amount_change_count()); +} + +// Tests that DataChannel::messages_sent() and DataChannel::bytes_sent() are +// correct, sending data both while unblocked and while blocked. +TEST_F(SctpDataChannelTest, VerifyMessagesAndBytesSent) { + AddObserver(); + SetChannelReady(); + std::vector<webrtc::DataBuffer> buffers({ + webrtc::DataBuffer("message 1"), + webrtc::DataBuffer("msg 2"), + webrtc::DataBuffer("message three"), + webrtc::DataBuffer("quadra message"), + webrtc::DataBuffer("fifthmsg"), + webrtc::DataBuffer("message of the beast"), + }); + + // Default values. + EXPECT_EQ(0U, webrtc_data_channel_->messages_sent()); + EXPECT_EQ(0U, webrtc_data_channel_->bytes_sent()); + + // Send three buffers while not blocked. + controller_->set_send_blocked(false); + EXPECT_TRUE(webrtc_data_channel_->Send(buffers[0])); + EXPECT_TRUE(webrtc_data_channel_->Send(buffers[1])); + EXPECT_TRUE(webrtc_data_channel_->Send(buffers[2])); + size_t bytes_sent = buffers[0].size() + buffers[1].size() + buffers[2].size(); + EXPECT_EQ_WAIT(0U, webrtc_data_channel_->buffered_amount(), kDefaultTimeout); + EXPECT_EQ(3U, webrtc_data_channel_->messages_sent()); + EXPECT_EQ(bytes_sent, webrtc_data_channel_->bytes_sent()); + + // Send three buffers while blocked, queuing the buffers. + controller_->set_send_blocked(true); + EXPECT_TRUE(webrtc_data_channel_->Send(buffers[3])); + EXPECT_TRUE(webrtc_data_channel_->Send(buffers[4])); + EXPECT_TRUE(webrtc_data_channel_->Send(buffers[5])); + size_t bytes_queued = + buffers[3].size() + buffers[4].size() + buffers[5].size(); + EXPECT_EQ(bytes_queued, webrtc_data_channel_->buffered_amount()); + EXPECT_EQ(3U, webrtc_data_channel_->messages_sent()); + EXPECT_EQ(bytes_sent, webrtc_data_channel_->bytes_sent()); + + // Unblock and make sure everything was sent. + controller_->set_send_blocked(false); + EXPECT_EQ_WAIT(0U, webrtc_data_channel_->buffered_amount(), kDefaultTimeout); + bytes_sent += bytes_queued; + EXPECT_EQ(6U, webrtc_data_channel_->messages_sent()); + EXPECT_EQ(bytes_sent, webrtc_data_channel_->bytes_sent()); +} + +// Tests that the queued control message is sent when channel is ready. +TEST_F(SctpDataChannelTest, OpenMessageSent) { + // Initially the id is unassigned. + EXPECT_EQ(-1, webrtc_data_channel_->id()); + + SetChannelReady(); + EXPECT_GE(webrtc_data_channel_->id(), 0); + EXPECT_EQ(webrtc::DataMessageType::kControl, + controller_->last_send_data_params().type); + EXPECT_EQ(controller_->last_sid(), webrtc_data_channel_->id()); +} + +TEST_F(SctpDataChannelTest, QueuedOpenMessageSent) { + controller_->set_send_blocked(true); + SetChannelReady(); + controller_->set_send_blocked(false); + + EXPECT_EQ(webrtc::DataMessageType::kControl, + controller_->last_send_data_params().type); + EXPECT_EQ(controller_->last_sid(), webrtc_data_channel_->id()); +} + +// Tests that the DataChannel created after transport gets ready can enter OPEN +// state. +TEST_F(SctpDataChannelTest, LateCreatedChannelTransitionToOpen) { + SetChannelReady(); + webrtc::InternalDataChannelInit init; + init.id = 1; + rtc::scoped_refptr<SctpDataChannel> dc = + SctpDataChannel::Create(controller_.get(), "test1", init, + rtc::Thread::Current(), rtc::Thread::Current()); + EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, dc->state()); + EXPECT_TRUE_WAIT(webrtc::DataChannelInterface::kOpen == dc->state(), 1000); +} + +// Tests that an unordered DataChannel sends data as ordered until the OPEN_ACK +// message is received. +TEST_F(SctpDataChannelTest, SendUnorderedAfterReceivesOpenAck) { + SetChannelReady(); + webrtc::InternalDataChannelInit init; + init.id = 1; + init.ordered = false; + rtc::scoped_refptr<SctpDataChannel> dc = + SctpDataChannel::Create(controller_.get(), "test1", init, + rtc::Thread::Current(), rtc::Thread::Current()); + + EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000); + + // Sends a message and verifies it's ordered. + webrtc::DataBuffer buffer("some data"); + ASSERT_TRUE(dc->Send(buffer)); + EXPECT_TRUE(controller_->last_send_data_params().ordered); + + // Emulates receiving an OPEN_ACK message. + cricket::ReceiveDataParams params; + params.sid = init.id; + params.type = webrtc::DataMessageType::kControl; + rtc::CopyOnWriteBuffer payload; + webrtc::WriteDataChannelOpenAckMessage(&payload); + dc->OnDataReceived(params, payload); + + // Sends another message and verifies it's unordered. + ASSERT_TRUE(dc->Send(buffer)); + EXPECT_FALSE(controller_->last_send_data_params().ordered); +} + +// Tests that an unordered DataChannel sends unordered data after any DATA +// message is received. +TEST_F(SctpDataChannelTest, SendUnorderedAfterReceiveData) { + SetChannelReady(); + webrtc::InternalDataChannelInit init; + init.id = 1; + init.ordered = false; + rtc::scoped_refptr<SctpDataChannel> dc = + SctpDataChannel::Create(controller_.get(), "test1", init, + rtc::Thread::Current(), rtc::Thread::Current()); + + EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000); + + // Emulates receiving a DATA message. + cricket::ReceiveDataParams params; + params.sid = init.id; + params.type = webrtc::DataMessageType::kText; + webrtc::DataBuffer buffer("data"); + dc->OnDataReceived(params, buffer.data); + + // Sends a message and verifies it's unordered. + ASSERT_TRUE(dc->Send(buffer)); + EXPECT_FALSE(controller_->last_send_data_params().ordered); +} + +// Tests that the channel can't open until it's successfully sent the OPEN +// message. +TEST_F(SctpDataChannelTest, OpenWaitsForOpenMesssage) { + webrtc::DataBuffer buffer("foo"); + + controller_->set_send_blocked(true); + SetChannelReady(); + EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, + webrtc_data_channel_->state()); + controller_->set_send_blocked(false); + EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, + webrtc_data_channel_->state(), 1000); + EXPECT_EQ(webrtc::DataMessageType::kControl, + controller_->last_send_data_params().type); +} + +// Tests that close first makes sure all queued data gets sent. +TEST_F(SctpDataChannelTest, QueuedCloseFlushes) { + webrtc::DataBuffer buffer("foo"); + + controller_->set_send_blocked(true); + SetChannelReady(); + EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, + webrtc_data_channel_->state()); + controller_->set_send_blocked(false); + EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, + webrtc_data_channel_->state(), 1000); + controller_->set_send_blocked(true); + webrtc_data_channel_->Send(buffer); + webrtc_data_channel_->Close(); + controller_->set_send_blocked(false); + EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, + webrtc_data_channel_->state(), 1000); + EXPECT_TRUE(webrtc_data_channel_->error().ok()); + EXPECT_EQ(webrtc::DataMessageType::kText, + controller_->last_send_data_params().type); +} + +// Tests that messages are sent with the right id. +TEST_F(SctpDataChannelTest, SendDataId) { + webrtc_data_channel_->SetSctpSid(1); + SetChannelReady(); + webrtc::DataBuffer buffer("data"); + EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); + EXPECT_EQ(1, controller_->last_sid()); +} + +// Tests that the incoming messages with wrong ids are rejected. +TEST_F(SctpDataChannelTest, ReceiveDataWithInvalidId) { + webrtc_data_channel_->SetSctpSid(1); + SetChannelReady(); + + AddObserver(); + + cricket::ReceiveDataParams params; + params.sid = 0; + webrtc::DataBuffer buffer("abcd"); + webrtc_data_channel_->OnDataReceived(params, buffer.data); + + EXPECT_EQ(0U, observer_->messages_received()); +} + +// Tests that the incoming messages with right ids are accepted. +TEST_F(SctpDataChannelTest, ReceiveDataWithValidId) { + webrtc_data_channel_->SetSctpSid(1); + SetChannelReady(); + + AddObserver(); + + cricket::ReceiveDataParams params; + params.sid = 1; + webrtc::DataBuffer buffer("abcd"); + + webrtc_data_channel_->OnDataReceived(params, buffer.data); + EXPECT_EQ(1U, observer_->messages_received()); +} + +// Tests that no CONTROL message is sent if the datachannel is negotiated and +// not created from an OPEN message. +TEST_F(SctpDataChannelTest, NoMsgSentIfNegotiatedAndNotFromOpenMsg) { + webrtc::InternalDataChannelInit config; + config.id = 1; + config.negotiated = true; + config.open_handshake_role = webrtc::InternalDataChannelInit::kNone; + + SetChannelReady(); + rtc::scoped_refptr<SctpDataChannel> dc = + SctpDataChannel::Create(controller_.get(), "test1", config, + rtc::Thread::Current(), rtc::Thread::Current()); + + EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000); + EXPECT_EQ(0, controller_->last_sid()); +} + +// Tests that DataChannel::messages_received() and DataChannel::bytes_received() +// are correct, receiving data both while not open and while open. +TEST_F(SctpDataChannelTest, VerifyMessagesAndBytesReceived) { + AddObserver(); + std::vector<webrtc::DataBuffer> buffers({ + webrtc::DataBuffer("message 1"), + webrtc::DataBuffer("msg 2"), + webrtc::DataBuffer("message three"), + webrtc::DataBuffer("quadra message"), + webrtc::DataBuffer("fifthmsg"), + webrtc::DataBuffer("message of the beast"), + }); + + webrtc_data_channel_->SetSctpSid(1); + cricket::ReceiveDataParams params; + params.sid = 1; + + // Default values. + EXPECT_EQ(0U, webrtc_data_channel_->messages_received()); + EXPECT_EQ(0U, webrtc_data_channel_->bytes_received()); + + // Receive three buffers while data channel isn't open. + webrtc_data_channel_->OnDataReceived(params, buffers[0].data); + webrtc_data_channel_->OnDataReceived(params, buffers[1].data); + webrtc_data_channel_->OnDataReceived(params, buffers[2].data); + EXPECT_EQ(0U, observer_->messages_received()); + EXPECT_EQ(0U, webrtc_data_channel_->messages_received()); + EXPECT_EQ(0U, webrtc_data_channel_->bytes_received()); + + // Open channel and make sure everything was received. + SetChannelReady(); + size_t bytes_received = + buffers[0].size() + buffers[1].size() + buffers[2].size(); + EXPECT_EQ(3U, observer_->messages_received()); + EXPECT_EQ(3U, webrtc_data_channel_->messages_received()); + EXPECT_EQ(bytes_received, webrtc_data_channel_->bytes_received()); + + // Receive three buffers while open. + webrtc_data_channel_->OnDataReceived(params, buffers[3].data); + webrtc_data_channel_->OnDataReceived(params, buffers[4].data); + webrtc_data_channel_->OnDataReceived(params, buffers[5].data); + bytes_received += buffers[3].size() + buffers[4].size() + buffers[5].size(); + EXPECT_EQ(6U, observer_->messages_received()); + EXPECT_EQ(6U, webrtc_data_channel_->messages_received()); + EXPECT_EQ(bytes_received, webrtc_data_channel_->bytes_received()); +} + +// Tests that OPEN_ACK message is sent if the datachannel is created from an +// OPEN message. +TEST_F(SctpDataChannelTest, OpenAckSentIfCreatedFromOpenMessage) { + webrtc::InternalDataChannelInit config; + config.id = 1; + config.negotiated = true; + config.open_handshake_role = webrtc::InternalDataChannelInit::kAcker; + + SetChannelReady(); + rtc::scoped_refptr<SctpDataChannel> dc = + SctpDataChannel::Create(controller_.get(), "test1", config, + rtc::Thread::Current(), rtc::Thread::Current()); + + EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000); + + EXPECT_EQ(config.id, controller_->last_sid()); + EXPECT_EQ(webrtc::DataMessageType::kControl, + controller_->last_send_data_params().type); +} + +// Tests the OPEN_ACK role assigned by InternalDataChannelInit. +TEST_F(SctpDataChannelTest, OpenAckRoleInitialization) { + webrtc::InternalDataChannelInit init; + EXPECT_EQ(webrtc::InternalDataChannelInit::kOpener, init.open_handshake_role); + EXPECT_FALSE(init.negotiated); + + webrtc::DataChannelInit base; + base.negotiated = true; + webrtc::InternalDataChannelInit init2(base); + EXPECT_EQ(webrtc::InternalDataChannelInit::kNone, init2.open_handshake_role); +} + +// Tests that that Send() returns false if the sending buffer is full +// and the channel stays open. +TEST_F(SctpDataChannelTest, OpenWhenSendBufferFull) { + SetChannelReady(); + + const size_t packetSize = 1024; + + rtc::CopyOnWriteBuffer buffer(packetSize); + memset(buffer.MutableData(), 0, buffer.size()); + + webrtc::DataBuffer packet(buffer, true); + controller_->set_send_blocked(true); + + for (size_t i = 0; + i < webrtc::DataChannelInterface::MaxSendQueueSize() / packetSize; ++i) { + EXPECT_TRUE(webrtc_data_channel_->Send(packet)); + } + + // The sending buffer shoul be full, send returns false. + EXPECT_FALSE(webrtc_data_channel_->Send(packet)); + + EXPECT_TRUE(webrtc::DataChannelInterface::kOpen == + webrtc_data_channel_->state()); +} + +// Tests that the DataChannel is closed on transport errors. +TEST_F(SctpDataChannelTest, ClosedOnTransportError) { + SetChannelReady(); + webrtc::DataBuffer buffer("abcd"); + controller_->set_transport_error(); + + EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); + + EXPECT_EQ(webrtc::DataChannelInterface::kClosed, + webrtc_data_channel_->state()); + EXPECT_FALSE(webrtc_data_channel_->error().ok()); + EXPECT_EQ(webrtc::RTCErrorType::NETWORK_ERROR, + webrtc_data_channel_->error().type()); + EXPECT_EQ(webrtc::RTCErrorDetailType::NONE, + webrtc_data_channel_->error().error_detail()); +} + +// Tests that the DataChannel is closed if the received buffer is full. +TEST_F(SctpDataChannelTest, ClosedWhenReceivedBufferFull) { + SetChannelReady(); + rtc::CopyOnWriteBuffer buffer(1024); + memset(buffer.MutableData(), 0, buffer.size()); + + cricket::ReceiveDataParams params; + params.sid = 0; + + // Receiving data without having an observer will overflow the buffer. + for (size_t i = 0; i < 16 * 1024 + 1; ++i) { + webrtc_data_channel_->OnDataReceived(params, buffer); + } + EXPECT_EQ(webrtc::DataChannelInterface::kClosed, + webrtc_data_channel_->state()); + EXPECT_FALSE(webrtc_data_channel_->error().ok()); + EXPECT_EQ(webrtc::RTCErrorType::RESOURCE_EXHAUSTED, + webrtc_data_channel_->error().type()); + EXPECT_EQ(webrtc::RTCErrorDetailType::NONE, + webrtc_data_channel_->error().error_detail()); +} + +// Tests that sending empty data returns no error and keeps the channel open. +TEST_F(SctpDataChannelTest, SendEmptyData) { + webrtc_data_channel_->SetSctpSid(1); + SetChannelReady(); + EXPECT_EQ(webrtc::DataChannelInterface::kOpen, webrtc_data_channel_->state()); + + webrtc::DataBuffer buffer(""); + EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); + EXPECT_EQ(webrtc::DataChannelInterface::kOpen, webrtc_data_channel_->state()); +} + +// Tests that a channel can be closed without being opened or assigned an sid. +TEST_F(SctpDataChannelTest, NeverOpened) { + controller_->set_transport_available(true); + webrtc_data_channel_->OnTransportChannelCreated(); + webrtc_data_channel_->Close(); +} + +// Test that the data channel goes to the "closed" state (and doesn't crash) +// when its transport goes away, even while data is buffered. +TEST_F(SctpDataChannelTest, TransportDestroyedWhileDataBuffered) { + SetChannelReady(); + + rtc::CopyOnWriteBuffer buffer(1024); + memset(buffer.MutableData(), 0, buffer.size()); + webrtc::DataBuffer packet(buffer, true); + + // Send a packet while sending is blocked so it ends up buffered. + controller_->set_send_blocked(true); + EXPECT_TRUE(webrtc_data_channel_->Send(packet)); + + // Tell the data channel that its transport is being destroyed. + // It should then stop using the transport (allowing us to delete it) and + // transition to the "closed" state. + webrtc::RTCError error(webrtc::RTCErrorType::OPERATION_ERROR_WITH_DATA, ""); + error.set_error_detail(webrtc::RTCErrorDetailType::SCTP_FAILURE); + webrtc_data_channel_->OnTransportChannelClosed(error); + controller_.reset(nullptr); + EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, + webrtc_data_channel_->state(), kDefaultTimeout); + EXPECT_FALSE(webrtc_data_channel_->error().ok()); + EXPECT_EQ(webrtc::RTCErrorType::OPERATION_ERROR_WITH_DATA, + webrtc_data_channel_->error().type()); + EXPECT_EQ(webrtc::RTCErrorDetailType::SCTP_FAILURE, + webrtc_data_channel_->error().error_detail()); +} + +TEST_F(SctpDataChannelTest, TransportGotErrorCode) { + SetChannelReady(); + + // Tell the data channel that its transport is being destroyed with an + // error code. + // It should then report that error code. + webrtc::RTCError error(webrtc::RTCErrorType::OPERATION_ERROR_WITH_DATA, + "Transport channel closed"); + error.set_error_detail(webrtc::RTCErrorDetailType::SCTP_FAILURE); + error.set_sctp_cause_code( + static_cast<uint16_t>(cricket::SctpErrorCauseCode::kProtocolViolation)); + webrtc_data_channel_->OnTransportChannelClosed(error); + controller_.reset(nullptr); + EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, + webrtc_data_channel_->state(), kDefaultTimeout); + EXPECT_FALSE(webrtc_data_channel_->error().ok()); + EXPECT_EQ(webrtc::RTCErrorType::OPERATION_ERROR_WITH_DATA, + webrtc_data_channel_->error().type()); + EXPECT_EQ(webrtc::RTCErrorDetailType::SCTP_FAILURE, + webrtc_data_channel_->error().error_detail()); + EXPECT_EQ( + static_cast<uint16_t>(cricket::SctpErrorCauseCode::kProtocolViolation), + webrtc_data_channel_->error().sctp_cause_code()); +} + +class SctpSidAllocatorTest : public ::testing::Test { + protected: + SctpSidAllocator allocator_; +}; + +// Verifies that an even SCTP id is allocated for SSL_CLIENT and an odd id for +// SSL_SERVER. +TEST_F(SctpSidAllocatorTest, SctpIdAllocationBasedOnRole) { + int id; + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &id)); + EXPECT_EQ(1, id); + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &id)); + EXPECT_EQ(0, id); + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &id)); + EXPECT_EQ(3, id); + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &id)); + EXPECT_EQ(2, id); +} + +// Verifies that SCTP ids of existing DataChannels are not reused. +TEST_F(SctpSidAllocatorTest, SctpIdAllocationNoReuse) { + int old_id = 1; + EXPECT_TRUE(allocator_.ReserveSid(old_id)); + + int new_id; + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &new_id)); + EXPECT_NE(old_id, new_id); + + old_id = 0; + EXPECT_TRUE(allocator_.ReserveSid(old_id)); + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &new_id)); + EXPECT_NE(old_id, new_id); +} + +// Verifies that SCTP ids of removed DataChannels can be reused. +TEST_F(SctpSidAllocatorTest, SctpIdReusedForRemovedDataChannel) { + int odd_id = 1; + int even_id = 0; + EXPECT_TRUE(allocator_.ReserveSid(odd_id)); + EXPECT_TRUE(allocator_.ReserveSid(even_id)); + + int allocated_id = -1; + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &allocated_id)); + EXPECT_EQ(odd_id + 2, allocated_id); + + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &allocated_id)); + EXPECT_EQ(even_id + 2, allocated_id); + + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &allocated_id)); + EXPECT_EQ(odd_id + 4, allocated_id); + + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &allocated_id)); + EXPECT_EQ(even_id + 4, allocated_id); + + allocator_.ReleaseSid(odd_id); + allocator_.ReleaseSid(even_id); + + // Verifies that removed ids are reused. + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &allocated_id)); + EXPECT_EQ(odd_id, allocated_id); + + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &allocated_id)); + EXPECT_EQ(even_id, allocated_id); + + // Verifies that used higher ids are not reused. + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &allocated_id)); + EXPECT_EQ(odd_id + 6, allocated_id); + + EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &allocated_id)); + EXPECT_EQ(even_id + 6, allocated_id); +} |