| Index: talk/app/webrtc/peerconnection_unittest.cc
|
| diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
|
| deleted file mode 100644
|
| index cad13e24412c881cb1b3f256996acbae8d697e08..0000000000000000000000000000000000000000
|
| --- a/talk/app/webrtc/peerconnection_unittest.cc
|
| +++ /dev/null
|
| @@ -1,2029 +0,0 @@
|
| -/*
|
| - * libjingle
|
| - * Copyright 2012 Google Inc.
|
| - *
|
| - * Redistribution and use in source and binary forms, with or without
|
| - * modification, are permitted provided that the following conditions are met:
|
| - *
|
| - * 1. Redistributions of source code must retain the above copyright notice,
|
| - * this list of conditions and the following disclaimer.
|
| - * 2. Redistributions in binary form must reproduce the above copyright notice,
|
| - * this list of conditions and the following disclaimer in the documentation
|
| - * and/or other materials provided with the distribution.
|
| - * 3. The name of the author may not be used to endorse or promote products
|
| - * derived from this software without specific prior written permission.
|
| - *
|
| - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
| - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
| - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
| - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
| - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
| - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
| - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
| - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
| - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
| - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
| - */
|
| -
|
| -#include <stdio.h>
|
| -
|
| -#include <algorithm>
|
| -#include <list>
|
| -#include <map>
|
| -#include <utility>
|
| -#include <vector>
|
| -
|
| -#include "talk/app/webrtc/dtmfsender.h"
|
| -#include "talk/app/webrtc/fakemetricsobserver.h"
|
| -#include "talk/app/webrtc/localaudiosource.h"
|
| -#include "talk/app/webrtc/mediastreaminterface.h"
|
| -#include "talk/app/webrtc/peerconnection.h"
|
| -#include "talk/app/webrtc/peerconnectionfactory.h"
|
| -#include "talk/app/webrtc/peerconnectioninterface.h"
|
| -#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
|
| -#include "talk/app/webrtc/test/fakeconstraints.h"
|
| -#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
|
| -#include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
|
| -#include "talk/app/webrtc/test/fakevideotrackrenderer.h"
|
| -#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
|
| -#include "talk/app/webrtc/videosourceinterface.h"
|
| -#include "talk/session/media/mediasession.h"
|
| -#include "webrtc/base/gunit.h"
|
| -#include "webrtc/base/physicalsocketserver.h"
|
| -#include "webrtc/base/scoped_ptr.h"
|
| -#include "webrtc/base/ssladapter.h"
|
| -#include "webrtc/base/sslstreamadapter.h"
|
| -#include "webrtc/base/thread.h"
|
| -#include "webrtc/base/virtualsocketserver.h"
|
| -#include "webrtc/media/webrtc/fakewebrtcvideoengine.h"
|
| -#include "webrtc/p2p/base/constants.h"
|
| -#include "webrtc/p2p/base/sessiondescription.h"
|
| -#include "webrtc/p2p/client/fakeportallocator.h"
|
| -
|
| -#define MAYBE_SKIP_TEST(feature) \
|
| - if (!(feature())) { \
|
| - LOG(LS_INFO) << "Feature disabled... skipping"; \
|
| - return; \
|
| - }
|
| -
|
| -using cricket::ContentInfo;
|
| -using cricket::FakeWebRtcVideoDecoder;
|
| -using cricket::FakeWebRtcVideoDecoderFactory;
|
| -using cricket::FakeWebRtcVideoEncoder;
|
| -using cricket::FakeWebRtcVideoEncoderFactory;
|
| -using cricket::MediaContentDescription;
|
| -using webrtc::DataBuffer;
|
| -using webrtc::DataChannelInterface;
|
| -using webrtc::DtmfSender;
|
| -using webrtc::DtmfSenderInterface;
|
| -using webrtc::DtmfSenderObserverInterface;
|
| -using webrtc::FakeConstraints;
|
| -using webrtc::MediaConstraintsInterface;
|
| -using webrtc::MediaStreamInterface;
|
| -using webrtc::MediaStreamTrackInterface;
|
| -using webrtc::MockCreateSessionDescriptionObserver;
|
| -using webrtc::MockDataChannelObserver;
|
| -using webrtc::MockSetSessionDescriptionObserver;
|
| -using webrtc::MockStatsObserver;
|
| -using webrtc::ObserverInterface;
|
| -using webrtc::PeerConnectionInterface;
|
| -using webrtc::PeerConnectionFactory;
|
| -using webrtc::SessionDescriptionInterface;
|
| -using webrtc::StreamCollectionInterface;
|
| -
|
| -static const int kMaxWaitMs = 10000;
|
| -// Disable for TSan v2, see
|
| -// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
|
| -// This declaration is also #ifdef'd as it causes uninitialized-variable
|
| -// warnings.
|
| -#if !defined(THREAD_SANITIZER)
|
| -static const int kMaxWaitForStatsMs = 3000;
|
| -#endif
|
| -static const int kMaxWaitForActivationMs = 5000;
|
| -static const int kMaxWaitForFramesMs = 10000;
|
| -static const int kEndAudioFrameCount = 3;
|
| -static const int kEndVideoFrameCount = 3;
|
| -
|
| -static const char kStreamLabelBase[] = "stream_label";
|
| -static const char kVideoTrackLabelBase[] = "video_track";
|
| -static const char kAudioTrackLabelBase[] = "audio_track";
|
| -static const char kDataChannelLabel[] = "data_channel";
|
| -
|
| -// Disable for TSan v2, see
|
| -// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
|
| -// This declaration is also #ifdef'd as it causes unused-variable errors.
|
| -#if !defined(THREAD_SANITIZER)
|
| -// SRTP cipher name negotiated by the tests. This must be updated if the
|
| -// default changes.
|
| -static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32;
|
| -#endif
|
| -
|
| -static void RemoveLinesFromSdp(const std::string& line_start,
|
| - std::string* sdp) {
|
| - const char kSdpLineEnd[] = "\r\n";
|
| - size_t ssrc_pos = 0;
|
| - while ((ssrc_pos = sdp->find(line_start, ssrc_pos)) !=
|
| - std::string::npos) {
|
| - size_t end_ssrc = sdp->find(kSdpLineEnd, ssrc_pos);
|
| - sdp->erase(ssrc_pos, end_ssrc - ssrc_pos + strlen(kSdpLineEnd));
|
| - }
|
| -}
|
| -
|
| -class SignalingMessageReceiver {
|
| - public:
|
| - virtual void ReceiveSdpMessage(const std::string& type,
|
| - std::string& msg) = 0;
|
| - virtual void ReceiveIceMessage(const std::string& sdp_mid,
|
| - int sdp_mline_index,
|
| - const std::string& msg) = 0;
|
| -
|
| - protected:
|
| - SignalingMessageReceiver() {}
|
| - virtual ~SignalingMessageReceiver() {}
|
| -};
|
| -
|
| -class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
|
| - public SignalingMessageReceiver,
|
| - public ObserverInterface {
|
| - public:
|
| - static PeerConnectionTestClient* CreateClientWithDtlsIdentityStore(
|
| - const std::string& id,
|
| - const MediaConstraintsInterface* constraints,
|
| - const PeerConnectionFactory::Options* options,
|
| - rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
|
| - PeerConnectionTestClient* client(new PeerConnectionTestClient(id));
|
| - if (!client->Init(constraints, options, std::move(dtls_identity_store))) {
|
| - delete client;
|
| - return nullptr;
|
| - }
|
| - return client;
|
| - }
|
| -
|
| - static PeerConnectionTestClient* CreateClient(
|
| - const std::string& id,
|
| - const MediaConstraintsInterface* constraints,
|
| - const PeerConnectionFactory::Options* options) {
|
| - rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
|
| - rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
|
| - : nullptr);
|
| -
|
| - return CreateClientWithDtlsIdentityStore(id, constraints, options,
|
| - std::move(dtls_identity_store));
|
| - }
|
| -
|
| - ~PeerConnectionTestClient() {
|
| - }
|
| -
|
| - void Negotiate() { Negotiate(true, true); }
|
| -
|
| - void Negotiate(bool audio, bool video) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> offer;
|
| - ASSERT_TRUE(DoCreateOffer(offer.use()));
|
| -
|
| - if (offer->description()->GetContentByName("audio")) {
|
| - offer->description()->GetContentByName("audio")->rejected = !audio;
|
| - }
|
| - if (offer->description()->GetContentByName("video")) {
|
| - offer->description()->GetContentByName("video")->rejected = !video;
|
| - }
|
| -
|
| - std::string sdp;
|
| - EXPECT_TRUE(offer->ToString(&sdp));
|
| - EXPECT_TRUE(DoSetLocalDescription(offer.release()));
|
| - signaling_message_receiver_->ReceiveSdpMessage(
|
| - webrtc::SessionDescriptionInterface::kOffer, sdp);
|
| - }
|
| -
|
| - // SignalingMessageReceiver callback.
|
| - void ReceiveSdpMessage(const std::string& type, std::string& msg) override {
|
| - FilterIncomingSdpMessage(&msg);
|
| - if (type == webrtc::SessionDescriptionInterface::kOffer) {
|
| - HandleIncomingOffer(msg);
|
| - } else {
|
| - HandleIncomingAnswer(msg);
|
| - }
|
| - }
|
| -
|
| - // SignalingMessageReceiver callback.
|
| - void ReceiveIceMessage(const std::string& sdp_mid,
|
| - int sdp_mline_index,
|
| - const std::string& msg) override {
|
| - LOG(INFO) << id_ << "ReceiveIceMessage";
|
| - rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate(
|
| - webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr));
|
| - EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
|
| - }
|
| -
|
| - // PeerConnectionObserver callbacks.
|
| - void OnSignalingChange(
|
| - webrtc::PeerConnectionInterface::SignalingState new_state) override {
|
| - EXPECT_EQ(pc()->signaling_state(), new_state);
|
| - }
|
| - void OnAddStream(MediaStreamInterface* media_stream) override {
|
| - media_stream->RegisterObserver(this);
|
| - for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) {
|
| - const std::string id = media_stream->GetVideoTracks()[i]->id();
|
| - ASSERT_TRUE(fake_video_renderers_.find(id) ==
|
| - fake_video_renderers_.end());
|
| - fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
|
| - media_stream->GetVideoTracks()[i]));
|
| - }
|
| - }
|
| - void OnRemoveStream(MediaStreamInterface* media_stream) override {}
|
| - void OnRenegotiationNeeded() override {}
|
| - void OnIceConnectionChange(
|
| - webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
|
| - EXPECT_EQ(pc()->ice_connection_state(), new_state);
|
| - }
|
| - void OnIceGatheringChange(
|
| - webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
|
| - EXPECT_EQ(pc()->ice_gathering_state(), new_state);
|
| - }
|
| - void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
|
| - LOG(INFO) << id_ << "OnIceCandidate";
|
| -
|
| - std::string ice_sdp;
|
| - EXPECT_TRUE(candidate->ToString(&ice_sdp));
|
| - if (signaling_message_receiver_ == nullptr) {
|
| - // Remote party may be deleted.
|
| - return;
|
| - }
|
| - signaling_message_receiver_->ReceiveIceMessage(
|
| - candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
|
| - }
|
| -
|
| - // MediaStreamInterface callback
|
| - void OnChanged() override {
|
| - // Track added or removed from MediaStream, so update our renderers.
|
| - rtc::scoped_refptr<StreamCollectionInterface> remote_streams =
|
| - pc()->remote_streams();
|
| - // Remove renderers for tracks that were removed.
|
| - for (auto it = fake_video_renderers_.begin();
|
| - it != fake_video_renderers_.end();) {
|
| - if (remote_streams->FindVideoTrack(it->first) == nullptr) {
|
| - auto to_remove = it++;
|
| - removed_fake_video_renderers_.push_back(std::move(to_remove->second));
|
| - fake_video_renderers_.erase(to_remove);
|
| - } else {
|
| - ++it;
|
| - }
|
| - }
|
| - // Create renderers for new video tracks.
|
| - for (size_t stream_index = 0; stream_index < remote_streams->count();
|
| - ++stream_index) {
|
| - MediaStreamInterface* remote_stream = remote_streams->at(stream_index);
|
| - for (size_t track_index = 0;
|
| - track_index < remote_stream->GetVideoTracks().size();
|
| - ++track_index) {
|
| - const std::string id =
|
| - remote_stream->GetVideoTracks()[track_index]->id();
|
| - if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) {
|
| - continue;
|
| - }
|
| - fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
|
| - remote_stream->GetVideoTracks()[track_index]));
|
| - }
|
| - }
|
| - }
|
| -
|
| - void SetVideoConstraints(const webrtc::FakeConstraints& video_constraint) {
|
| - video_constraints_ = video_constraint;
|
| - }
|
| -
|
| - void AddMediaStream(bool audio, bool video) {
|
| - std::string stream_label =
|
| - kStreamLabelBase +
|
| - rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count()));
|
| - rtc::scoped_refptr<MediaStreamInterface> stream =
|
| - peer_connection_factory_->CreateLocalMediaStream(stream_label);
|
| -
|
| - if (audio && can_receive_audio()) {
|
| - stream->AddTrack(CreateLocalAudioTrack(stream_label));
|
| - }
|
| - if (video && can_receive_video()) {
|
| - stream->AddTrack(CreateLocalVideoTrack(stream_label));
|
| - }
|
| -
|
| - EXPECT_TRUE(pc()->AddStream(stream));
|
| - }
|
| -
|
| - size_t NumberOfLocalMediaStreams() { return pc()->local_streams()->count(); }
|
| -
|
| - bool SessionActive() {
|
| - return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
|
| - }
|
| -
|
| - // Automatically add a stream when receiving an offer, if we don't have one.
|
| - // Defaults to true.
|
| - void set_auto_add_stream(bool auto_add_stream) {
|
| - auto_add_stream_ = auto_add_stream;
|
| - }
|
| -
|
| - void set_signaling_message_receiver(
|
| - SignalingMessageReceiver* signaling_message_receiver) {
|
| - signaling_message_receiver_ = signaling_message_receiver;
|
| - }
|
| -
|
| - void EnableVideoDecoderFactory() {
|
| - video_decoder_factory_enabled_ = true;
|
| - fake_video_decoder_factory_->AddSupportedVideoCodecType(
|
| - webrtc::kVideoCodecVP8);
|
| - }
|
| -
|
| - void IceRestart() {
|
| - session_description_constraints_.SetMandatoryIceRestart(true);
|
| - SetExpectIceRestart(true);
|
| - }
|
| -
|
| - void SetExpectIceRestart(bool expect_restart) {
|
| - expect_ice_restart_ = expect_restart;
|
| - }
|
| -
|
| - bool ExpectIceRestart() const { return expect_ice_restart_; }
|
| -
|
| - void SetReceiveAudioVideo(bool audio, bool video) {
|
| - SetReceiveAudio(audio);
|
| - SetReceiveVideo(video);
|
| - ASSERT_EQ(audio, can_receive_audio());
|
| - ASSERT_EQ(video, can_receive_video());
|
| - }
|
| -
|
| - void SetReceiveAudio(bool audio) {
|
| - if (audio && can_receive_audio())
|
| - return;
|
| - session_description_constraints_.SetMandatoryReceiveAudio(audio);
|
| - }
|
| -
|
| - void SetReceiveVideo(bool video) {
|
| - if (video && can_receive_video())
|
| - return;
|
| - session_description_constraints_.SetMandatoryReceiveVideo(video);
|
| - }
|
| -
|
| - void RemoveMsidFromReceivedSdp(bool remove) { remove_msid_ = remove; }
|
| -
|
| - void RemoveSdesCryptoFromReceivedSdp(bool remove) { remove_sdes_ = remove; }
|
| -
|
| - void RemoveBundleFromReceivedSdp(bool remove) { remove_bundle_ = remove; }
|
| -
|
| - bool can_receive_audio() {
|
| - bool value;
|
| - if (webrtc::FindConstraint(&session_description_constraints_,
|
| - MediaConstraintsInterface::kOfferToReceiveAudio,
|
| - &value, nullptr)) {
|
| - return value;
|
| - }
|
| - return true;
|
| - }
|
| -
|
| - bool can_receive_video() {
|
| - bool value;
|
| - if (webrtc::FindConstraint(&session_description_constraints_,
|
| - MediaConstraintsInterface::kOfferToReceiveVideo,
|
| - &value, nullptr)) {
|
| - return value;
|
| - }
|
| - return true;
|
| - }
|
| -
|
| - void OnDataChannel(DataChannelInterface* data_channel) override {
|
| - LOG(INFO) << id_ << "OnDataChannel";
|
| - data_channel_ = data_channel;
|
| - data_observer_.reset(new MockDataChannelObserver(data_channel));
|
| - }
|
| -
|
| - void CreateDataChannel() {
|
| - data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, nullptr);
|
| - ASSERT_TRUE(data_channel_.get() != nullptr);
|
| - data_observer_.reset(new MockDataChannelObserver(data_channel_));
|
| - }
|
| -
|
| - rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
|
| - const std::string& stream_label) {
|
| - FakeConstraints constraints;
|
| - // Disable highpass filter so that we can get all the test audio frames.
|
| - constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
|
| - rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
|
| - peer_connection_factory_->CreateAudioSource(&constraints);
|
| - // TODO(perkj): Test audio source when it is implemented. Currently audio
|
| - // always use the default input.
|
| - std::string label = stream_label + kAudioTrackLabelBase;
|
| - return peer_connection_factory_->CreateAudioTrack(label, source);
|
| - }
|
| -
|
| - rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
|
| - const std::string& stream_label) {
|
| - // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
|
| - FakeConstraints source_constraints = video_constraints_;
|
| - source_constraints.SetMandatoryMaxFrameRate(10);
|
| -
|
| - cricket::FakeVideoCapturer* fake_capturer =
|
| - new webrtc::FakePeriodicVideoCapturer();
|
| - video_capturers_.push_back(fake_capturer);
|
| - rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
|
| - peer_connection_factory_->CreateVideoSource(fake_capturer,
|
| - &source_constraints);
|
| - std::string label = stream_label + kVideoTrackLabelBase;
|
| - return peer_connection_factory_->CreateVideoTrack(label, source);
|
| - }
|
| -
|
| - DataChannelInterface* data_channel() { return data_channel_; }
|
| - const MockDataChannelObserver* data_observer() const {
|
| - return data_observer_.get();
|
| - }
|
| -
|
| - webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); }
|
| -
|
| - void StopVideoCapturers() {
|
| - for (std::vector<cricket::VideoCapturer*>::iterator it =
|
| - video_capturers_.begin();
|
| - it != video_capturers_.end(); ++it) {
|
| - (*it)->Stop();
|
| - }
|
| - }
|
| -
|
| - bool AudioFramesReceivedCheck(int number_of_frames) const {
|
| - return number_of_frames <= fake_audio_capture_module_->frames_received();
|
| - }
|
| -
|
| - int audio_frames_received() const {
|
| - return fake_audio_capture_module_->frames_received();
|
| - }
|
| -
|
| - bool VideoFramesReceivedCheck(int number_of_frames) {
|
| - if (video_decoder_factory_enabled_) {
|
| - const std::vector<FakeWebRtcVideoDecoder*>& decoders
|
| - = fake_video_decoder_factory_->decoders();
|
| - if (decoders.empty()) {
|
| - return number_of_frames <= 0;
|
| - }
|
| -
|
| - for (FakeWebRtcVideoDecoder* decoder : decoders) {
|
| - if (number_of_frames > decoder->GetNumFramesReceived()) {
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| - } else {
|
| - if (fake_video_renderers_.empty()) {
|
| - return number_of_frames <= 0;
|
| - }
|
| -
|
| - for (const auto& pair : fake_video_renderers_) {
|
| - if (number_of_frames > pair.second->num_rendered_frames()) {
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| - }
|
| - }
|
| -
|
| - int video_frames_received() const {
|
| - int total = 0;
|
| - if (video_decoder_factory_enabled_) {
|
| - const std::vector<FakeWebRtcVideoDecoder*>& decoders =
|
| - fake_video_decoder_factory_->decoders();
|
| - for (const FakeWebRtcVideoDecoder* decoder : decoders) {
|
| - total += decoder->GetNumFramesReceived();
|
| - }
|
| - } else {
|
| - for (const auto& pair : fake_video_renderers_) {
|
| - total += pair.second->num_rendered_frames();
|
| - }
|
| - for (const auto& renderer : removed_fake_video_renderers_) {
|
| - total += renderer->num_rendered_frames();
|
| - }
|
| - }
|
| - return total;
|
| - }
|
| -
|
| - // Verify the CreateDtmfSender interface
|
| - void VerifyDtmf() {
|
| - rtc::scoped_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
|
| - rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender;
|
| -
|
| - // We can't create a DTMF sender with an invalid audio track or a non local
|
| - // track.
|
| - EXPECT_TRUE(peer_connection_->CreateDtmfSender(nullptr) == nullptr);
|
| - rtc::scoped_refptr<webrtc::AudioTrackInterface> non_localtrack(
|
| - peer_connection_factory_->CreateAudioTrack("dummy_track", nullptr));
|
| - EXPECT_TRUE(peer_connection_->CreateDtmfSender(non_localtrack) == nullptr);
|
| -
|
| - // We should be able to create a DTMF sender from a local track.
|
| - webrtc::AudioTrackInterface* localtrack =
|
| - peer_connection_->local_streams()->at(0)->GetAudioTracks()[0];
|
| - dtmf_sender = peer_connection_->CreateDtmfSender(localtrack);
|
| - EXPECT_TRUE(dtmf_sender.get() != nullptr);
|
| - dtmf_sender->RegisterObserver(observer.get());
|
| -
|
| - // Test the DtmfSender object just created.
|
| - EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
|
| - EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50));
|
| -
|
| - // We don't need to verify that the DTMF tones are actually sent out because
|
| - // that is already covered by the tests of the lower level components.
|
| -
|
| - EXPECT_TRUE_WAIT(observer->completed(), kMaxWaitMs);
|
| - std::vector<std::string> tones;
|
| - tones.push_back("1");
|
| - tones.push_back("a");
|
| - tones.push_back("");
|
| - observer->Verify(tones);
|
| -
|
| - dtmf_sender->UnregisterObserver();
|
| - }
|
| -
|
| - // Verifies that the SessionDescription have rejected the appropriate media
|
| - // content.
|
| - void VerifyRejectedMediaInSessionDescription() {
|
| - ASSERT_TRUE(peer_connection_->remote_description() != nullptr);
|
| - ASSERT_TRUE(peer_connection_->local_description() != nullptr);
|
| - const cricket::SessionDescription* remote_desc =
|
| - peer_connection_->remote_description()->description();
|
| - const cricket::SessionDescription* local_desc =
|
| - peer_connection_->local_description()->description();
|
| -
|
| - const ContentInfo* remote_audio_content = GetFirstAudioContent(remote_desc);
|
| - if (remote_audio_content) {
|
| - const ContentInfo* audio_content =
|
| - GetFirstAudioContent(local_desc);
|
| - EXPECT_EQ(can_receive_audio(), !audio_content->rejected);
|
| - }
|
| -
|
| - const ContentInfo* remote_video_content = GetFirstVideoContent(remote_desc);
|
| - if (remote_video_content) {
|
| - const ContentInfo* video_content =
|
| - GetFirstVideoContent(local_desc);
|
| - EXPECT_EQ(can_receive_video(), !video_content->rejected);
|
| - }
|
| - }
|
| -
|
| - void VerifyLocalIceUfragAndPassword() {
|
| - ASSERT_TRUE(peer_connection_->local_description() != nullptr);
|
| - const cricket::SessionDescription* desc =
|
| - peer_connection_->local_description()->description();
|
| - const cricket::ContentInfos& contents = desc->contents();
|
| -
|
| - for (size_t index = 0; index < contents.size(); ++index) {
|
| - if (contents[index].rejected)
|
| - continue;
|
| - const cricket::TransportDescription* transport_desc =
|
| - desc->GetTransportDescriptionByName(contents[index].name);
|
| -
|
| - std::map<int, IceUfragPwdPair>::const_iterator ufragpair_it =
|
| - ice_ufrag_pwd_.find(static_cast<int>(index));
|
| - if (ufragpair_it == ice_ufrag_pwd_.end()) {
|
| - ASSERT_FALSE(ExpectIceRestart());
|
| - ice_ufrag_pwd_[static_cast<int>(index)] =
|
| - IceUfragPwdPair(transport_desc->ice_ufrag, transport_desc->ice_pwd);
|
| - } else if (ExpectIceRestart()) {
|
| - const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second;
|
| - EXPECT_NE(ufrag_pwd.first, transport_desc->ice_ufrag);
|
| - EXPECT_NE(ufrag_pwd.second, transport_desc->ice_pwd);
|
| - } else {
|
| - const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second;
|
| - EXPECT_EQ(ufrag_pwd.first, transport_desc->ice_ufrag);
|
| - EXPECT_EQ(ufrag_pwd.second, transport_desc->ice_pwd);
|
| - }
|
| - }
|
| - }
|
| -
|
| - int GetAudioOutputLevelStats(webrtc::MediaStreamTrackInterface* track) {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->AudioOutputLevel();
|
| - }
|
| -
|
| - int GetAudioInputLevelStats() {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->AudioInputLevel();
|
| - }
|
| -
|
| - int GetBytesReceivedStats(webrtc::MediaStreamTrackInterface* track) {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->BytesReceived();
|
| - }
|
| -
|
| - int GetBytesSentStats(webrtc::MediaStreamTrackInterface* track) {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->BytesSent();
|
| - }
|
| -
|
| - int GetAvailableReceivedBandwidthStats() {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - int bw = observer->AvailableReceiveBandwidth();
|
| - return bw;
|
| - }
|
| -
|
| - std::string GetDtlsCipherStats() {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->DtlsCipher();
|
| - }
|
| -
|
| - std::string GetSrtpCipherStats() {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->SrtpCipher();
|
| - }
|
| -
|
| - int rendered_width() {
|
| - EXPECT_FALSE(fake_video_renderers_.empty());
|
| - return fake_video_renderers_.empty() ? 1 :
|
| - fake_video_renderers_.begin()->second->width();
|
| - }
|
| -
|
| - int rendered_height() {
|
| - EXPECT_FALSE(fake_video_renderers_.empty());
|
| - return fake_video_renderers_.empty() ? 1 :
|
| - fake_video_renderers_.begin()->second->height();
|
| - }
|
| -
|
| - size_t number_of_remote_streams() {
|
| - if (!pc())
|
| - return 0;
|
| - return pc()->remote_streams()->count();
|
| - }
|
| -
|
| - StreamCollectionInterface* remote_streams() {
|
| - if (!pc()) {
|
| - ADD_FAILURE();
|
| - return nullptr;
|
| - }
|
| - return pc()->remote_streams();
|
| - }
|
| -
|
| - StreamCollectionInterface* local_streams() {
|
| - if (!pc()) {
|
| - ADD_FAILURE();
|
| - return nullptr;
|
| - }
|
| - return pc()->local_streams();
|
| - }
|
| -
|
| - webrtc::PeerConnectionInterface::SignalingState signaling_state() {
|
| - return pc()->signaling_state();
|
| - }
|
| -
|
| - webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
|
| - return pc()->ice_connection_state();
|
| - }
|
| -
|
| - webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
|
| - return pc()->ice_gathering_state();
|
| - }
|
| -
|
| - private:
|
| - class DummyDtmfObserver : public DtmfSenderObserverInterface {
|
| - public:
|
| - DummyDtmfObserver() : completed_(false) {}
|
| -
|
| - // Implements DtmfSenderObserverInterface.
|
| - void OnToneChange(const std::string& tone) override {
|
| - tones_.push_back(tone);
|
| - if (tone.empty()) {
|
| - completed_ = true;
|
| - }
|
| - }
|
| -
|
| - void Verify(const std::vector<std::string>& tones) const {
|
| - ASSERT_TRUE(tones_.size() == tones.size());
|
| - EXPECT_TRUE(std::equal(tones.begin(), tones.end(), tones_.begin()));
|
| - }
|
| -
|
| - bool completed() const { return completed_; }
|
| -
|
| - private:
|
| - bool completed_;
|
| - std::vector<std::string> tones_;
|
| - };
|
| -
|
| - explicit PeerConnectionTestClient(const std::string& id) : id_(id) {}
|
| -
|
| - bool Init(
|
| - const MediaConstraintsInterface* constraints,
|
| - const PeerConnectionFactory::Options* options,
|
| - rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
|
| - EXPECT_TRUE(!peer_connection_);
|
| - EXPECT_TRUE(!peer_connection_factory_);
|
| - rtc::scoped_ptr<cricket::PortAllocator> port_allocator(
|
| - new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
|
| - fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
|
| -
|
| - if (fake_audio_capture_module_ == nullptr) {
|
| - return false;
|
| - }
|
| - fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory();
|
| - fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory();
|
| - peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
|
| - rtc::Thread::Current(), rtc::Thread::Current(),
|
| - fake_audio_capture_module_, fake_video_encoder_factory_,
|
| - fake_video_decoder_factory_);
|
| - if (!peer_connection_factory_) {
|
| - return false;
|
| - }
|
| - if (options) {
|
| - peer_connection_factory_->SetOptions(*options);
|
| - }
|
| - peer_connection_ = CreatePeerConnection(
|
| - std::move(port_allocator), constraints, std::move(dtls_identity_store));
|
| - return peer_connection_.get() != nullptr;
|
| - }
|
| -
|
| - rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
|
| - rtc::scoped_ptr<cricket::PortAllocator> port_allocator,
|
| - const MediaConstraintsInterface* constraints,
|
| - rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
|
| - // CreatePeerConnection with RTCConfiguration.
|
| - webrtc::PeerConnectionInterface::RTCConfiguration config;
|
| - webrtc::PeerConnectionInterface::IceServer ice_server;
|
| - ice_server.uri = "stun:stun.l.google.com:19302";
|
| - config.servers.push_back(ice_server);
|
| -
|
| - return peer_connection_factory_->CreatePeerConnection(
|
| - config, constraints, std::move(port_allocator),
|
| - std::move(dtls_identity_store), this);
|
| - }
|
| -
|
| - void HandleIncomingOffer(const std::string& msg) {
|
| - LOG(INFO) << id_ << "HandleIncomingOffer ";
|
| - if (NumberOfLocalMediaStreams() == 0 && auto_add_stream_) {
|
| - // If we are not sending any streams ourselves it is time to add some.
|
| - AddMediaStream(true, true);
|
| - }
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription("offer", msg, nullptr));
|
| - EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
|
| - rtc::scoped_ptr<SessionDescriptionInterface> answer;
|
| - EXPECT_TRUE(DoCreateAnswer(answer.use()));
|
| - std::string sdp;
|
| - EXPECT_TRUE(answer->ToString(&sdp));
|
| - EXPECT_TRUE(DoSetLocalDescription(answer.release()));
|
| - if (signaling_message_receiver_) {
|
| - signaling_message_receiver_->ReceiveSdpMessage(
|
| - webrtc::SessionDescriptionInterface::kAnswer, sdp);
|
| - }
|
| - }
|
| -
|
| - void HandleIncomingAnswer(const std::string& msg) {
|
| - LOG(INFO) << id_ << "HandleIncomingAnswer";
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription("answer", msg, nullptr));
|
| - EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
|
| - }
|
| -
|
| - bool DoCreateOfferAnswer(SessionDescriptionInterface** desc,
|
| - bool offer) {
|
| - rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
|
| - observer(new rtc::RefCountedObject<
|
| - MockCreateSessionDescriptionObserver>());
|
| - if (offer) {
|
| - pc()->CreateOffer(observer, &session_description_constraints_);
|
| - } else {
|
| - pc()->CreateAnswer(observer, &session_description_constraints_);
|
| - }
|
| - EXPECT_EQ_WAIT(true, observer->called(), kMaxWaitMs);
|
| - *desc = observer->release_desc();
|
| - if (observer->result() && ExpectIceRestart()) {
|
| - EXPECT_EQ(0u, (*desc)->candidates(0)->count());
|
| - }
|
| - return observer->result();
|
| - }
|
| -
|
| - bool DoCreateOffer(SessionDescriptionInterface** desc) {
|
| - return DoCreateOfferAnswer(desc, true);
|
| - }
|
| -
|
| - bool DoCreateAnswer(SessionDescriptionInterface** desc) {
|
| - return DoCreateOfferAnswer(desc, false);
|
| - }
|
| -
|
| - bool DoSetLocalDescription(SessionDescriptionInterface* desc) {
|
| - rtc::scoped_refptr<MockSetSessionDescriptionObserver>
|
| - observer(new rtc::RefCountedObject<
|
| - MockSetSessionDescriptionObserver>());
|
| - LOG(INFO) << id_ << "SetLocalDescription ";
|
| - pc()->SetLocalDescription(observer, desc);
|
| - // Ignore the observer result. If we wait for the result with
|
| - // EXPECT_TRUE_WAIT, local ice candidates might be sent to the remote peer
|
| - // before the offer which is an error.
|
| - // The reason is that EXPECT_TRUE_WAIT uses
|
| - // rtc::Thread::Current()->ProcessMessages(1);
|
| - // ProcessMessages waits at least 1ms but processes all messages before
|
| - // returning. Since this test is synchronous and send messages to the remote
|
| - // peer whenever a callback is invoked, this can lead to messages being
|
| - // sent to the remote peer in the wrong order.
|
| - // TODO(perkj): Find a way to check the result without risking that the
|
| - // order of sent messages are changed. Ex- by posting all messages that are
|
| - // sent to the remote peer.
|
| - return true;
|
| - }
|
| -
|
| - bool DoSetRemoteDescription(SessionDescriptionInterface* desc) {
|
| - rtc::scoped_refptr<MockSetSessionDescriptionObserver>
|
| - observer(new rtc::RefCountedObject<
|
| - MockSetSessionDescriptionObserver>());
|
| - LOG(INFO) << id_ << "SetRemoteDescription ";
|
| - pc()->SetRemoteDescription(observer, desc);
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - return observer->result();
|
| - }
|
| -
|
| - // This modifies all received SDP messages before they are processed.
|
| - void FilterIncomingSdpMessage(std::string* sdp) {
|
| - if (remove_msid_) {
|
| - const char kSdpSsrcAttribute[] = "a=ssrc:";
|
| - RemoveLinesFromSdp(kSdpSsrcAttribute, sdp);
|
| - const char kSdpMsidSupportedAttribute[] = "a=msid-semantic:";
|
| - RemoveLinesFromSdp(kSdpMsidSupportedAttribute, sdp);
|
| - }
|
| - if (remove_bundle_) {
|
| - const char kSdpBundleAttribute[] = "a=group:BUNDLE";
|
| - RemoveLinesFromSdp(kSdpBundleAttribute, sdp);
|
| - }
|
| - if (remove_sdes_) {
|
| - const char kSdpSdesCryptoAttribute[] = "a=crypto";
|
| - RemoveLinesFromSdp(kSdpSdesCryptoAttribute, sdp);
|
| - }
|
| - }
|
| -
|
| - std::string id_;
|
| -
|
| - rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
|
| - rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
|
| - peer_connection_factory_;
|
| -
|
| - bool auto_add_stream_ = true;
|
| -
|
| - typedef std::pair<std::string, std::string> IceUfragPwdPair;
|
| - std::map<int, IceUfragPwdPair> ice_ufrag_pwd_;
|
| - bool expect_ice_restart_ = false;
|
| -
|
| - // Needed to keep track of number of frames sent.
|
| - rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
|
| - // Needed to keep track of number of frames received.
|
| - std::map<std::string, rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
|
| - fake_video_renderers_;
|
| - // Needed to ensure frames aren't received for removed tracks.
|
| - std::vector<rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
|
| - removed_fake_video_renderers_;
|
| - // Needed to keep track of number of frames received when external decoder
|
| - // used.
|
| - FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr;
|
| - FakeWebRtcVideoEncoderFactory* fake_video_encoder_factory_ = nullptr;
|
| - bool video_decoder_factory_enabled_ = false;
|
| - webrtc::FakeConstraints video_constraints_;
|
| -
|
| - // For remote peer communication.
|
| - SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
|
| -
|
| - // Store references to the video capturers we've created, so that we can stop
|
| - // them, if required.
|
| - std::vector<cricket::VideoCapturer*> video_capturers_;
|
| -
|
| - webrtc::FakeConstraints session_description_constraints_;
|
| - bool remove_msid_ = false; // True if MSID should be removed in received SDP.
|
| - bool remove_bundle_ =
|
| - false; // True if bundle should be removed in received SDP.
|
| - bool remove_sdes_ =
|
| - false; // True if a=crypto should be removed in received SDP.
|
| -
|
| - rtc::scoped_refptr<DataChannelInterface> data_channel_;
|
| - rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
|
| -};
|
| -
|
| -class P2PTestConductor : public testing::Test {
|
| - public:
|
| - P2PTestConductor()
|
| - : pss_(new rtc::PhysicalSocketServer),
|
| - ss_(new rtc::VirtualSocketServer(pss_.get())),
|
| - ss_scope_(ss_.get()) {}
|
| -
|
| - bool SessionActive() {
|
| - return initiating_client_->SessionActive() &&
|
| - receiving_client_->SessionActive();
|
| - }
|
| -
|
| - // Return true if the number of frames provided have been received or it is
|
| - // known that that will never occur (e.g. no frames will be sent or
|
| - // captured).
|
| - bool FramesNotPending(int audio_frames_to_receive,
|
| - int video_frames_to_receive) {
|
| - return VideoFramesReceivedCheck(video_frames_to_receive) &&
|
| - AudioFramesReceivedCheck(audio_frames_to_receive);
|
| - }
|
| - bool AudioFramesReceivedCheck(int frames_received) {
|
| - return initiating_client_->AudioFramesReceivedCheck(frames_received) &&
|
| - receiving_client_->AudioFramesReceivedCheck(frames_received);
|
| - }
|
| - bool VideoFramesReceivedCheck(int frames_received) {
|
| - return initiating_client_->VideoFramesReceivedCheck(frames_received) &&
|
| - receiving_client_->VideoFramesReceivedCheck(frames_received);
|
| - }
|
| - void VerifyDtmf() {
|
| - initiating_client_->VerifyDtmf();
|
| - receiving_client_->VerifyDtmf();
|
| - }
|
| -
|
| - void TestUpdateOfferWithRejectedContent() {
|
| - // Renegotiate, rejecting the video m-line.
|
| - initiating_client_->Negotiate(true, false);
|
| - ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
|
| -
|
| - int pc1_audio_received = initiating_client_->audio_frames_received();
|
| - int pc1_video_received = initiating_client_->video_frames_received();
|
| - int pc2_audio_received = receiving_client_->audio_frames_received();
|
| - int pc2_video_received = receiving_client_->video_frames_received();
|
| -
|
| - // Wait for some additional audio frames to be received.
|
| - EXPECT_TRUE_WAIT(initiating_client_->AudioFramesReceivedCheck(
|
| - pc1_audio_received + kEndAudioFrameCount) &&
|
| - receiving_client_->AudioFramesReceivedCheck(
|
| - pc2_audio_received + kEndAudioFrameCount),
|
| - kMaxWaitForFramesMs);
|
| -
|
| - // During this time, we shouldn't have received any additional video frames
|
| - // for the rejected video tracks.
|
| - EXPECT_EQ(pc1_video_received, initiating_client_->video_frames_received());
|
| - EXPECT_EQ(pc2_video_received, receiving_client_->video_frames_received());
|
| - }
|
| -
|
| - void VerifyRenderedSize(int width, int height) {
|
| - EXPECT_EQ(width, receiving_client()->rendered_width());
|
| - EXPECT_EQ(height, receiving_client()->rendered_height());
|
| - EXPECT_EQ(width, initializing_client()->rendered_width());
|
| - EXPECT_EQ(height, initializing_client()->rendered_height());
|
| - }
|
| -
|
| - void VerifySessionDescriptions() {
|
| - initiating_client_->VerifyRejectedMediaInSessionDescription();
|
| - receiving_client_->VerifyRejectedMediaInSessionDescription();
|
| - initiating_client_->VerifyLocalIceUfragAndPassword();
|
| - receiving_client_->VerifyLocalIceUfragAndPassword();
|
| - }
|
| -
|
| - ~P2PTestConductor() {
|
| - if (initiating_client_) {
|
| - initiating_client_->set_signaling_message_receiver(nullptr);
|
| - }
|
| - if (receiving_client_) {
|
| - receiving_client_->set_signaling_message_receiver(nullptr);
|
| - }
|
| - }
|
| -
|
| - bool CreateTestClients() { return CreateTestClients(nullptr, nullptr); }
|
| -
|
| - bool CreateTestClients(MediaConstraintsInterface* init_constraints,
|
| - MediaConstraintsInterface* recv_constraints) {
|
| - return CreateTestClients(init_constraints, nullptr, recv_constraints,
|
| - nullptr);
|
| - }
|
| -
|
| - void SetSignalingReceivers() {
|
| - initiating_client_->set_signaling_message_receiver(receiving_client_.get());
|
| - receiving_client_->set_signaling_message_receiver(initiating_client_.get());
|
| - }
|
| -
|
| - bool CreateTestClients(MediaConstraintsInterface* init_constraints,
|
| - PeerConnectionFactory::Options* init_options,
|
| - MediaConstraintsInterface* recv_constraints,
|
| - PeerConnectionFactory::Options* recv_options) {
|
| - initiating_client_.reset(PeerConnectionTestClient::CreateClient(
|
| - "Caller: ", init_constraints, init_options));
|
| - receiving_client_.reset(PeerConnectionTestClient::CreateClient(
|
| - "Callee: ", recv_constraints, recv_options));
|
| - if (!initiating_client_ || !receiving_client_) {
|
| - return false;
|
| - }
|
| - SetSignalingReceivers();
|
| - return true;
|
| - }
|
| -
|
| - void SetVideoConstraints(const webrtc::FakeConstraints& init_constraints,
|
| - const webrtc::FakeConstraints& recv_constraints) {
|
| - initiating_client_->SetVideoConstraints(init_constraints);
|
| - receiving_client_->SetVideoConstraints(recv_constraints);
|
| - }
|
| -
|
| - void EnableVideoDecoderFactory() {
|
| - initiating_client_->EnableVideoDecoderFactory();
|
| - receiving_client_->EnableVideoDecoderFactory();
|
| - }
|
| -
|
| - // This test sets up a call between two parties. Both parties send static
|
| - // frames to each other. Once the test is finished the number of sent frames
|
| - // is compared to the number of received frames.
|
| - void LocalP2PTest() {
|
| - if (initiating_client_->NumberOfLocalMediaStreams() == 0) {
|
| - initiating_client_->AddMediaStream(true, true);
|
| - }
|
| - initiating_client_->Negotiate();
|
| - // Assert true is used here since next tests are guaranteed to fail and
|
| - // would eat up 5 seconds.
|
| - ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
|
| - VerifySessionDescriptions();
|
| -
|
| - int audio_frame_count = kEndAudioFrameCount;
|
| - // TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
|
| - if (!initiating_client_->can_receive_audio() ||
|
| - !receiving_client_->can_receive_audio()) {
|
| - audio_frame_count = -1;
|
| - }
|
| - int video_frame_count = kEndVideoFrameCount;
|
| - if (!initiating_client_->can_receive_video() ||
|
| - !receiving_client_->can_receive_video()) {
|
| - video_frame_count = -1;
|
| - }
|
| -
|
| - if (audio_frame_count != -1 || video_frame_count != -1) {
|
| - // Audio or video is expected to flow, so both clients should reach the
|
| - // Connected state, and the offerer (ICE controller) should proceed to
|
| - // Completed.
|
| - // Note: These tests have been observed to fail under heavy load at
|
| - // shorter timeouts, so they may be flaky.
|
| - EXPECT_EQ_WAIT(
|
| - webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
| - initiating_client_->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| - EXPECT_EQ_WAIT(
|
| - webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
| - receiving_client_->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| - }
|
| -
|
| - if (initiating_client_->can_receive_audio() ||
|
| - initiating_client_->can_receive_video()) {
|
| - // The initiating client can receive media, so it must produce candidates
|
| - // that will serve as destinations for that media.
|
| - // TODO(bemasc): Understand why the state is not already Complete here, as
|
| - // seems to be the case for the receiving client. This may indicate a bug
|
| - // in the ICE gathering system.
|
| - EXPECT_NE(webrtc::PeerConnectionInterface::kIceGatheringNew,
|
| - initiating_client_->ice_gathering_state());
|
| - }
|
| - if (receiving_client_->can_receive_audio() ||
|
| - receiving_client_->can_receive_video()) {
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
|
| - receiving_client_->ice_gathering_state(),
|
| - kMaxWaitForFramesMs);
|
| - }
|
| -
|
| - EXPECT_TRUE_WAIT(FramesNotPending(audio_frame_count, video_frame_count),
|
| - kMaxWaitForFramesMs);
|
| - }
|
| -
|
| - void SetupAndVerifyDtlsCall() {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - LocalP2PTest();
|
| - VerifyRenderedSize(640, 480);
|
| - }
|
| -
|
| - PeerConnectionTestClient* CreateDtlsClientWithAlternateKey() {
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| -
|
| - rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
|
| - rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
|
| - : nullptr);
|
| - dtls_identity_store->use_alternate_key();
|
| -
|
| - // Make sure the new client is using a different certificate.
|
| - return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore(
|
| - "New Peer: ", &setup_constraints, nullptr,
|
| - std::move(dtls_identity_store));
|
| - }
|
| -
|
| - void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
|
| - // Messages may get lost on the unreliable DataChannel, so we send multiple
|
| - // times to avoid test flakiness.
|
| - static const size_t kSendAttempts = 5;
|
| -
|
| - for (size_t i = 0; i < kSendAttempts; ++i) {
|
| - dc->Send(DataBuffer(data));
|
| - }
|
| - }
|
| -
|
| - PeerConnectionTestClient* initializing_client() {
|
| - return initiating_client_.get();
|
| - }
|
| -
|
| - // Set the |initiating_client_| to the |client| passed in and return the
|
| - // original |initiating_client_|.
|
| - PeerConnectionTestClient* set_initializing_client(
|
| - PeerConnectionTestClient* client) {
|
| - PeerConnectionTestClient* old = initiating_client_.release();
|
| - initiating_client_.reset(client);
|
| - return old;
|
| - }
|
| -
|
| - PeerConnectionTestClient* receiving_client() {
|
| - return receiving_client_.get();
|
| - }
|
| -
|
| - // Set the |receiving_client_| to the |client| passed in and return the
|
| - // original |receiving_client_|.
|
| - PeerConnectionTestClient* set_receiving_client(
|
| - PeerConnectionTestClient* client) {
|
| - PeerConnectionTestClient* old = receiving_client_.release();
|
| - receiving_client_.reset(client);
|
| - return old;
|
| - }
|
| -
|
| - private:
|
| - rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
|
| - rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
|
| - rtc::SocketServerScope ss_scope_;
|
| - rtc::scoped_ptr<PeerConnectionTestClient> initiating_client_;
|
| - rtc::scoped_ptr<PeerConnectionTestClient> receiving_client_;
|
| -};
|
| -
|
| -// Disable for TSan v2, see
|
| -// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
|
| -#if !defined(THREAD_SANITIZER)
|
| -
|
| -// This test sets up a Jsep call between two parties and test Dtmf.
|
| -// TODO(holmer): Disabled due to sometimes crashing on buildbots.
|
| -// See issue webrtc/2378.
|
| -TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| - VerifyDtmf();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties and test that we can get a
|
| -// video aspect ratio of 16:9.
|
| -TEST_F(P2PTestConductor, LocalP2PTest16To9) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - FakeConstraints constraint;
|
| - double requested_ratio = 640.0/360;
|
| - constraint.SetMandatoryMinAspectRatio(requested_ratio);
|
| - SetVideoConstraints(constraint, constraint);
|
| - LocalP2PTest();
|
| -
|
| - ASSERT_LE(0, initializing_client()->rendered_height());
|
| - double initiating_video_ratio =
|
| - static_cast<double>(initializing_client()->rendered_width()) /
|
| - initializing_client()->rendered_height();
|
| - EXPECT_LE(requested_ratio, initiating_video_ratio);
|
| -
|
| - ASSERT_LE(0, receiving_client()->rendered_height());
|
| - double receiving_video_ratio =
|
| - static_cast<double>(receiving_client()->rendered_width()) /
|
| - receiving_client()->rendered_height();
|
| - EXPECT_LE(requested_ratio, receiving_video_ratio);
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties and test that the
|
| -// received video has a resolution of 1280*720.
|
| -// TODO(mallinath): Enable when
|
| -// http://code.google.com/p/webrtc/issues/detail?id=981 is fixed.
|
| -TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - FakeConstraints constraint;
|
| - constraint.SetMandatoryMinWidth(1280);
|
| - constraint.SetMandatoryMinHeight(720);
|
| - SetVideoConstraints(constraint, constraint);
|
| - LocalP2PTest();
|
| - VerifyRenderedSize(1280, 720);
|
| -}
|
| -
|
| -// This test sets up a call between two endpoints that are configured to use
|
| -// DTLS key agreement. As a result, DTLS is negotiated and used for transport.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtls) {
|
| - SetupAndVerifyDtlsCall();
|
| -}
|
| -
|
| -// This test sets up a audio call initially and then upgrades to audio/video,
|
| -// using DTLS.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - receiving_client()->SetReceiveAudioVideo(true, false);
|
| - LocalP2PTest();
|
| - receiving_client()->SetReceiveAudioVideo(true, true);
|
| - receiving_client()->Negotiate();
|
| -}
|
| -
|
| -// This test sets up a call transfer to a new caller with a different DTLS
|
| -// fingerprint.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - SetupAndVerifyDtlsCall();
|
| -
|
| - // Keeping the original peer around which will still send packets to the
|
| - // receiving client. These SRTP packets will be dropped.
|
| - rtc::scoped_ptr<PeerConnectionTestClient> original_peer(
|
| - set_initializing_client(CreateDtlsClientWithAlternateKey()));
|
| - original_peer->pc()->Close();
|
| -
|
| - SetSignalingReceivers();
|
| - receiving_client()->SetExpectIceRestart(true);
|
| - LocalP2PTest();
|
| - VerifyRenderedSize(640, 480);
|
| -}
|
| -
|
| -// This test sets up a non-bundle call and apply bundle during ICE restart. When
|
| -// bundle is in effect in the restart, the channel can successfully reset its
|
| -// DTLS-SRTP context.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - receiving_client()->RemoveBundleFromReceivedSdp(true);
|
| - LocalP2PTest();
|
| - VerifyRenderedSize(640, 480);
|
| -
|
| - initializing_client()->IceRestart();
|
| - receiving_client()->SetExpectIceRestart(true);
|
| - receiving_client()->RemoveBundleFromReceivedSdp(false);
|
| - LocalP2PTest();
|
| - VerifyRenderedSize(640, 480);
|
| -}
|
| -
|
| -// This test sets up a call transfer to a new callee with a different DTLS
|
| -// fingerprint.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - SetupAndVerifyDtlsCall();
|
| -
|
| - // Keeping the original peer around which will still send packets to the
|
| - // receiving client. These SRTP packets will be dropped.
|
| - rtc::scoped_ptr<PeerConnectionTestClient> original_peer(
|
| - set_receiving_client(CreateDtlsClientWithAlternateKey()));
|
| - original_peer->pc()->Close();
|
| -
|
| - SetSignalingReceivers();
|
| - initializing_client()->IceRestart();
|
| - LocalP2PTest();
|
| - VerifyRenderedSize(640, 480);
|
| -}
|
| -
|
| -// This test sets up a call between two endpoints that are configured to use
|
| -// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
|
| -// negotiated and used for transport.
|
| -TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - receiving_client()->RemoveSdesCryptoFromReceivedSdp(true);
|
| - LocalP2PTest();
|
| - VerifyRenderedSize(640, 480);
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties, and the callee only
|
| -// accept to receive video.
|
| -TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - receiving_client()->SetReceiveAudioVideo(false, true);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties, and the callee only
|
| -// accept to receive audio.
|
| -TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - receiving_client()->SetReceiveAudioVideo(true, false);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties, and the callee reject both
|
| -// audio and video.
|
| -TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - receiving_client()->SetReceiveAudioVideo(false, false);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up an audio and video call between two parties. After the call
|
| -// runs for a while (10 frames), the caller sends an update offer with video
|
| -// being rejected. Once the re-negotiation is done, the video flow should stop
|
| -// and the audio flow should continue.
|
| -TEST_F(P2PTestConductor, UpdateOfferWithRejectedContent) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| - TestUpdateOfferWithRejectedContent();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties. The MSID is removed from
|
| -// the SDP strings from the caller.
|
| -TEST_F(P2PTestConductor, LocalP2PTestWithoutMsid) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - receiving_client()->RemoveMsidFromReceivedSdp(true);
|
| - // TODO(perkj): Currently there is a bug that cause audio to stop playing if
|
| - // audio and video is muxed when MSID is disabled. Remove
|
| - // SetRemoveBundleFromSdp once
|
| - // https://code.google.com/p/webrtc/issues/detail?id=1193 is fixed.
|
| - receiving_client()->RemoveBundleFromReceivedSdp(true);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties and the initiating peer
|
| -// sends two steams.
|
| -// TODO(perkj): Disabled due to
|
| -// https://code.google.com/p/webrtc/issues/detail?id=1454
|
| -TEST_F(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - // Set optional video constraint to max 320pixels to decrease CPU usage.
|
| - FakeConstraints constraint;
|
| - constraint.SetOptionalMaxWidth(320);
|
| - SetVideoConstraints(constraint, constraint);
|
| - initializing_client()->AddMediaStream(true, true);
|
| - initializing_client()->AddMediaStream(false, true);
|
| - ASSERT_EQ(2u, initializing_client()->NumberOfLocalMediaStreams());
|
| - LocalP2PTest();
|
| - EXPECT_EQ(2u, receiving_client()->number_of_remote_streams());
|
| -}
|
| -
|
| -// Test that we can receive the audio output level from a remote audio track.
|
| -TEST_F(P2PTestConductor, GetAudioOutputLevelStats) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| -
|
| - StreamCollectionInterface* remote_streams =
|
| - initializing_client()->remote_streams();
|
| - ASSERT_GT(remote_streams->count(), 0u);
|
| - ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u);
|
| - MediaStreamTrackInterface* remote_audio_track =
|
| - remote_streams->at(0)->GetAudioTracks()[0];
|
| -
|
| - // Get the audio output level stats. Note that the level is not available
|
| - // until a RTCP packet has been received.
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetAudioOutputLevelStats(remote_audio_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -}
|
| -
|
| -// Test that an audio input level is reported.
|
| -TEST_F(P2PTestConductor, GetAudioInputLevelStats) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| -
|
| - // Get the audio input level stats. The level should be available very
|
| - // soon after the test starts.
|
| - EXPECT_TRUE_WAIT(initializing_client()->GetAudioInputLevelStats() > 0,
|
| - kMaxWaitForStatsMs);
|
| -}
|
| -
|
| -// Test that we can get incoming byte counts from both audio and video tracks.
|
| -TEST_F(P2PTestConductor, GetBytesReceivedStats) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| -
|
| - StreamCollectionInterface* remote_streams =
|
| - initializing_client()->remote_streams();
|
| - ASSERT_GT(remote_streams->count(), 0u);
|
| - ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u);
|
| - MediaStreamTrackInterface* remote_audio_track =
|
| - remote_streams->at(0)->GetAudioTracks()[0];
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetBytesReceivedStats(remote_audio_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -
|
| - MediaStreamTrackInterface* remote_video_track =
|
| - remote_streams->at(0)->GetVideoTracks()[0];
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetBytesReceivedStats(remote_video_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -}
|
| -
|
| -// Test that we can get outgoing byte counts from both audio and video tracks.
|
| -TEST_F(P2PTestConductor, GetBytesSentStats) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| -
|
| - StreamCollectionInterface* local_streams =
|
| - initializing_client()->local_streams();
|
| - ASSERT_GT(local_streams->count(), 0u);
|
| - ASSERT_GT(local_streams->at(0)->GetAudioTracks().size(), 0u);
|
| - MediaStreamTrackInterface* local_audio_track =
|
| - local_streams->at(0)->GetAudioTracks()[0];
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetBytesSentStats(local_audio_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -
|
| - MediaStreamTrackInterface* local_video_track =
|
| - local_streams->at(0)->GetVideoTracks()[0];
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetBytesSentStats(local_video_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -}
|
| -
|
| -// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
|
| -TEST_F(P2PTestConductor, GetDtls12None) {
|
| - PeerConnectionFactory::Options init_options;
|
| - init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
| - PeerConnectionFactory::Options recv_options;
|
| - recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
| - ASSERT_TRUE(
|
| - CreateTestClients(nullptr, &init_options, nullptr, &recv_options));
|
| - rtc::scoped_refptr<webrtc::FakeMetricsObserver>
|
| - init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
|
| - initializing_client()->pc()->RegisterUMAObserver(init_observer);
|
| - LocalP2PTest();
|
| -
|
| - EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
|
| - rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
|
| - rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
|
| - initializing_client()->GetDtlsCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1, init_observer->GetEnumCounter(
|
| - webrtc::kEnumCounterAudioSslCipher,
|
| - rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
|
| - rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
|
| -
|
| - EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
| - initializing_client()->GetSrtpCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1,
|
| - init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
|
| - kDefaultSrtpCryptoSuite));
|
| -}
|
| -
|
| -// Test that DTLS 1.2 is used if both ends support it.
|
| -TEST_F(P2PTestConductor, GetDtls12Both) {
|
| - PeerConnectionFactory::Options init_options;
|
| - init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
| - PeerConnectionFactory::Options recv_options;
|
| - recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
| - ASSERT_TRUE(
|
| - CreateTestClients(nullptr, &init_options, nullptr, &recv_options));
|
| - rtc::scoped_refptr<webrtc::FakeMetricsObserver>
|
| - init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
|
| - initializing_client()->pc()->RegisterUMAObserver(init_observer);
|
| - LocalP2PTest();
|
| -
|
| - EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
|
| - rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
|
| - rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)),
|
| - initializing_client()->GetDtlsCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1, init_observer->GetEnumCounter(
|
| - webrtc::kEnumCounterAudioSslCipher,
|
| - rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
|
| - rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)));
|
| -
|
| - EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
| - initializing_client()->GetSrtpCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1,
|
| - init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
|
| - kDefaultSrtpCryptoSuite));
|
| -}
|
| -
|
| -// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
|
| -// received supports 1.0.
|
| -TEST_F(P2PTestConductor, GetDtls12Init) {
|
| - PeerConnectionFactory::Options init_options;
|
| - init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
| - PeerConnectionFactory::Options recv_options;
|
| - recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
| - ASSERT_TRUE(
|
| - CreateTestClients(nullptr, &init_options, nullptr, &recv_options));
|
| - rtc::scoped_refptr<webrtc::FakeMetricsObserver>
|
| - init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
|
| - initializing_client()->pc()->RegisterUMAObserver(init_observer);
|
| - LocalP2PTest();
|
| -
|
| - EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
|
| - rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
|
| - rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
|
| - initializing_client()->GetDtlsCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1, init_observer->GetEnumCounter(
|
| - webrtc::kEnumCounterAudioSslCipher,
|
| - rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
|
| - rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
|
| -
|
| - EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
| - initializing_client()->GetSrtpCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1,
|
| - init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
|
| - kDefaultSrtpCryptoSuite));
|
| -}
|
| -
|
| -// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
|
| -// received supports 1.2.
|
| -TEST_F(P2PTestConductor, GetDtls12Recv) {
|
| - PeerConnectionFactory::Options init_options;
|
| - init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
| - PeerConnectionFactory::Options recv_options;
|
| - recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
| - ASSERT_TRUE(
|
| - CreateTestClients(nullptr, &init_options, nullptr, &recv_options));
|
| - rtc::scoped_refptr<webrtc::FakeMetricsObserver>
|
| - init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
|
| - initializing_client()->pc()->RegisterUMAObserver(init_observer);
|
| - LocalP2PTest();
|
| -
|
| - EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
|
| - rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
|
| - rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
|
| - initializing_client()->GetDtlsCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1, init_observer->GetEnumCounter(
|
| - webrtc::kEnumCounterAudioSslCipher,
|
| - rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
|
| - rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
|
| -
|
| - EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
| - initializing_client()->GetSrtpCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1,
|
| - init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
|
| - kDefaultSrtpCryptoSuite));
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, video and an RTP
|
| -// data channel.
|
| -TEST_F(P2PTestConductor, LocalP2PTestRtpDataChannel) {
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.SetAllowRtpDataChannels();
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - initializing_client()->CreateDataChannel();
|
| - LocalP2PTest();
|
| - ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| -
|
| - std::string data = "hello world";
|
| -
|
| - SendRtpData(initializing_client()->data_channel(), data);
|
| - EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -
|
| - SendRtpData(receiving_client()->data_channel(), data);
|
| - EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -
|
| - receiving_client()->data_channel()->Close();
|
| - // Send new offer and answer.
|
| - receiving_client()->Negotiate();
|
| - EXPECT_FALSE(initializing_client()->data_observer()->IsOpen());
|
| - EXPECT_FALSE(receiving_client()->data_observer()->IsOpen());
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, video and an SCTP
|
| -// data channel.
|
| -TEST_F(P2PTestConductor, LocalP2PTestSctpDataChannel) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - initializing_client()->CreateDataChannel();
|
| - LocalP2PTest();
|
| - ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
|
| -
|
| - std::string data = "hello world";
|
| -
|
| - initializing_client()->data_channel()->Send(DataBuffer(data));
|
| - EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -
|
| - receiving_client()->data_channel()->Send(DataBuffer(data));
|
| - EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -
|
| - receiving_client()->data_channel()->Close();
|
| - EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
|
| -}
|
| -
|
| -// This test sets up a call between two parties and creates a data channel.
|
| -// The test tests that received data is buffered unless an observer has been
|
| -// registered.
|
| -// Rtp data channels can receive data before the underlying
|
| -// transport has detected that a channel is writable and thus data can be
|
| -// received before the data channel state changes to open. That is hard to test
|
| -// but the same buffering is used in that case.
|
| -TEST_F(P2PTestConductor, RegisterDataChannelObserver) {
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.SetAllowRtpDataChannels();
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - initializing_client()->CreateDataChannel();
|
| - initializing_client()->Negotiate();
|
| -
|
| - ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
|
| - receiving_client()->data_channel()->state(), kMaxWaitMs);
|
| -
|
| - // Unregister the existing observer.
|
| - receiving_client()->data_channel()->UnregisterObserver();
|
| -
|
| - std::string data = "hello world";
|
| - SendRtpData(initializing_client()->data_channel(), data);
|
| -
|
| - // Wait a while to allow the sent data to arrive before an observer is
|
| - // registered..
|
| - rtc::Thread::Current()->ProcessMessages(100);
|
| -
|
| - MockDataChannelObserver new_observer(receiving_client()->data_channel());
|
| - EXPECT_EQ_WAIT(data, new_observer.last_message(), kMaxWaitMs);
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, video and but only
|
| -// the initiating client support data.
|
| -TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) {
|
| - FakeConstraints setup_constraints_1;
|
| - setup_constraints_1.SetAllowRtpDataChannels();
|
| - // Must disable DTLS to make negotiation succeed.
|
| - setup_constraints_1.SetMandatory(
|
| - MediaConstraintsInterface::kEnableDtlsSrtp, false);
|
| - FakeConstraints setup_constraints_2;
|
| - setup_constraints_2.SetMandatory(
|
| - MediaConstraintsInterface::kEnableDtlsSrtp, false);
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints_1, &setup_constraints_2));
|
| - initializing_client()->CreateDataChannel();
|
| - LocalP2PTest();
|
| - EXPECT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - EXPECT_FALSE(receiving_client()->data_channel());
|
| - EXPECT_FALSE(initializing_client()->data_observer()->IsOpen());
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, video. When audio
|
| -// and video is setup and flowing and data channel is negotiated.
|
| -TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) {
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.SetAllowRtpDataChannels();
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - LocalP2PTest();
|
| - initializing_client()->CreateDataChannel();
|
| - // Send new offer and answer.
|
| - initializing_client()->Negotiate();
|
| - ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| -}
|
| -
|
| -// This test sets up a Jsep call with SCTP DataChannel and verifies the
|
| -// negotiation is completed without error.
|
| -#ifdef HAVE_SCTP
|
| -TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints constraints;
|
| - constraints.SetMandatory(
|
| - MediaConstraintsInterface::kEnableDtlsSrtp, true);
|
| - ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
|
| - initializing_client()->CreateDataChannel();
|
| - initializing_client()->Negotiate(false, false);
|
| -}
|
| -#endif
|
| -
|
| -// This test sets up a call between two parties with audio, and video.
|
| -// During the call, the initializing side restart ice and the test verifies that
|
| -// new ice candidates are generated and audio and video still can flow.
|
| -TEST_F(P2PTestConductor, IceRestart) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| -
|
| - // Negotiate and wait for ice completion and make sure audio and video plays.
|
| - LocalP2PTest();
|
| -
|
| - // Create a SDP string of the first audio candidate for both clients.
|
| - const webrtc::IceCandidateCollection* audio_candidates_initiator =
|
| - initializing_client()->pc()->local_description()->candidates(0);
|
| - const webrtc::IceCandidateCollection* audio_candidates_receiver =
|
| - receiving_client()->pc()->local_description()->candidates(0);
|
| - ASSERT_GT(audio_candidates_initiator->count(), 0u);
|
| - ASSERT_GT(audio_candidates_receiver->count(), 0u);
|
| - std::string initiator_candidate;
|
| - EXPECT_TRUE(
|
| - audio_candidates_initiator->at(0)->ToString(&initiator_candidate));
|
| - std::string receiver_candidate;
|
| - EXPECT_TRUE(audio_candidates_receiver->at(0)->ToString(&receiver_candidate));
|
| -
|
| - // Restart ice on the initializing client.
|
| - receiving_client()->SetExpectIceRestart(true);
|
| - initializing_client()->IceRestart();
|
| -
|
| - // Negotiate and wait for ice completion again and make sure audio and video
|
| - // plays.
|
| - LocalP2PTest();
|
| -
|
| - // Create a SDP string of the first audio candidate for both clients again.
|
| - const webrtc::IceCandidateCollection* audio_candidates_initiator_restart =
|
| - initializing_client()->pc()->local_description()->candidates(0);
|
| - const webrtc::IceCandidateCollection* audio_candidates_reciever_restart =
|
| - receiving_client()->pc()->local_description()->candidates(0);
|
| - ASSERT_GT(audio_candidates_initiator_restart->count(), 0u);
|
| - ASSERT_GT(audio_candidates_reciever_restart->count(), 0u);
|
| - std::string initiator_candidate_restart;
|
| - EXPECT_TRUE(audio_candidates_initiator_restart->at(0)->ToString(
|
| - &initiator_candidate_restart));
|
| - std::string receiver_candidate_restart;
|
| - EXPECT_TRUE(audio_candidates_reciever_restart->at(0)->ToString(
|
| - &receiver_candidate_restart));
|
| -
|
| - // Verify that the first candidates in the local session descriptions has
|
| - // changed.
|
| - EXPECT_NE(initiator_candidate, initiator_candidate_restart);
|
| - EXPECT_NE(receiver_candidate, receiver_candidate_restart);
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, and video.
|
| -// It then renegotiates setting the video m-line to "port 0", then later
|
| -// renegotiates again, enabling video.
|
| -TEST_F(P2PTestConductor, LocalP2PTestVideoDisableEnable) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| -
|
| - // Do initial negotiation. Will result in video and audio sendonly m-lines.
|
| - receiving_client()->set_auto_add_stream(false);
|
| - initializing_client()->AddMediaStream(true, true);
|
| - initializing_client()->Negotiate();
|
| -
|
| - // Negotiate again, disabling the video m-line (receiving client will
|
| - // set port to 0 due to mandatory "OfferToReceiveVideo: false" constraint).
|
| - receiving_client()->SetReceiveVideo(false);
|
| - initializing_client()->Negotiate();
|
| -
|
| - // Enable video and do negotiation again, making sure video is received
|
| - // end-to-end.
|
| - receiving_client()->SetReceiveVideo(true);
|
| - receiving_client()->AddMediaStream(true, true);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties with external
|
| -// VideoDecoderFactory.
|
| -// TODO(holmer): Disabled due to sometimes crashing on buildbots.
|
| -// See issue webrtc/2378.
|
| -TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - EnableVideoDecoderFactory();
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This tests that if we negotiate after calling CreateSender but before we
|
| -// have a track, then set a track later, frames from the newly-set track are
|
| -// received end-to-end.
|
| -TEST_F(P2PTestConductor, EarlyWarmupTest) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - auto audio_sender =
|
| - initializing_client()->pc()->CreateSender("audio", "stream_id");
|
| - auto video_sender =
|
| - initializing_client()->pc()->CreateSender("video", "stream_id");
|
| - initializing_client()->Negotiate();
|
| - // Wait for ICE connection to complete, without any tracks.
|
| - // Note that the receiving client WILL (in HandleIncomingOffer) create
|
| - // tracks, so it's only the initiator here that's doing early warmup.
|
| - ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
|
| - VerifySessionDescriptions();
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
| - initializing_client()->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
| - receiving_client()->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| - // Now set the tracks, and expect frames to immediately start flowing.
|
| - EXPECT_TRUE(
|
| - audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack("")));
|
| - EXPECT_TRUE(
|
| - video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack("")));
|
| - EXPECT_TRUE_WAIT(FramesNotPending(kEndAudioFrameCount, kEndVideoFrameCount),
|
| - kMaxWaitForFramesMs);
|
| -}
|
| -
|
| -class IceServerParsingTest : public testing::Test {
|
| - public:
|
| - // Convenience for parsing a single URL.
|
| - bool ParseUrl(const std::string& url) {
|
| - return ParseUrl(url, std::string(), std::string());
|
| - }
|
| -
|
| - bool ParseUrl(const std::string& url,
|
| - const std::string& username,
|
| - const std::string& password) {
|
| - PeerConnectionInterface::IceServers servers;
|
| - PeerConnectionInterface::IceServer server;
|
| - server.urls.push_back(url);
|
| - server.username = username;
|
| - server.password = password;
|
| - servers.push_back(server);
|
| - return webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_);
|
| - }
|
| -
|
| - protected:
|
| - cricket::ServerAddresses stun_servers_;
|
| - std::vector<cricket::RelayServerConfig> turn_servers_;
|
| -};
|
| -
|
| -// Make sure all STUN/TURN prefixes are parsed correctly.
|
| -TEST_F(IceServerParsingTest, ParseStunPrefixes) {
|
| - EXPECT_TRUE(ParseUrl("stun:hostname"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ(0U, turn_servers_.size());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stuns:hostname"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ(0U, turn_servers_.size());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("turn:hostname"));
|
| - EXPECT_EQ(0U, stun_servers_.size());
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_FALSE(turn_servers_[0].ports[0].secure);
|
| - turn_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("turns:hostname"));
|
| - EXPECT_EQ(0U, stun_servers_.size());
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_TRUE(turn_servers_[0].ports[0].secure);
|
| - turn_servers_.clear();
|
| -
|
| - // invalid prefixes
|
| - EXPECT_FALSE(ParseUrl("stunn:hostname"));
|
| - EXPECT_FALSE(ParseUrl(":hostname"));
|
| - EXPECT_FALSE(ParseUrl(":"));
|
| - EXPECT_FALSE(ParseUrl(""));
|
| -}
|
| -
|
| -TEST_F(IceServerParsingTest, VerifyDefaults) {
|
| - // TURNS defaults
|
| - EXPECT_TRUE(ParseUrl("turns:hostname"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ(5349, turn_servers_[0].ports[0].address.port());
|
| - EXPECT_EQ(cricket::PROTO_TCP, turn_servers_[0].ports[0].proto);
|
| - turn_servers_.clear();
|
| -
|
| - // TURN defaults
|
| - EXPECT_TRUE(ParseUrl("turn:hostname"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ(3478, turn_servers_[0].ports[0].address.port());
|
| - EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
|
| - turn_servers_.clear();
|
| -
|
| - // STUN defaults
|
| - EXPECT_TRUE(ParseUrl("stun:hostname"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ(3478, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -}
|
| -
|
| -// Check that the 6 combinations of IPv4/IPv6/hostname and with/without port
|
| -// can be parsed correctly.
|
| -TEST_F(IceServerParsingTest, ParseHostnameAndPort) {
|
| - EXPECT_TRUE(ParseUrl("stun:1.2.3.4:1234"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(1234, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]:4321"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(4321, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:hostname:9999"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("hostname", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(9999, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:1.2.3.4"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(3478, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(3478, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:hostname"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("hostname", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(3478, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - // Try some invalid hostname:port strings.
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:99a99"));
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:-1"));
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:port:more"));
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:port more"));
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:"));
|
| - EXPECT_FALSE(ParseUrl("stun:[1:2:3:4:5:6:7:8]junk:1000"));
|
| - EXPECT_FALSE(ParseUrl("stun::5555"));
|
| - EXPECT_FALSE(ParseUrl("stun:"));
|
| -}
|
| -
|
| -// Test parsing the "?transport=xxx" part of the URL.
|
| -TEST_F(IceServerParsingTest, ParseTransport) {
|
| - EXPECT_TRUE(ParseUrl("turn:hostname:1234?transport=tcp"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ(cricket::PROTO_TCP, turn_servers_[0].ports[0].proto);
|
| - turn_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("turn:hostname?transport=udp"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
|
| - turn_servers_.clear();
|
| -
|
| - EXPECT_FALSE(ParseUrl("turn:hostname?transport=invalid"));
|
| -}
|
| -
|
| -// Test parsing ICE username contained in URL.
|
| -TEST_F(IceServerParsingTest, ParseUsername) {
|
| - EXPECT_TRUE(ParseUrl("turn:user@hostname"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ("user", turn_servers_[0].credentials.username);
|
| - turn_servers_.clear();
|
| -
|
| - EXPECT_FALSE(ParseUrl("turn:@hostname"));
|
| - EXPECT_FALSE(ParseUrl("turn:username@"));
|
| - EXPECT_FALSE(ParseUrl("turn:@"));
|
| - EXPECT_FALSE(ParseUrl("turn:user@name@hostname"));
|
| -}
|
| -
|
| -// Test that username and password from IceServer is copied into the resulting
|
| -// RelayServerConfig.
|
| -TEST_F(IceServerParsingTest, CopyUsernameAndPasswordFromIceServer) {
|
| - EXPECT_TRUE(ParseUrl("turn:hostname", "username", "password"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ("username", turn_servers_[0].credentials.username);
|
| - EXPECT_EQ("password", turn_servers_[0].credentials.password);
|
| -}
|
| -
|
| -// Ensure that if a server has multiple URLs, each one is parsed.
|
| -TEST_F(IceServerParsingTest, ParseMultipleUrls) {
|
| - PeerConnectionInterface::IceServers servers;
|
| - PeerConnectionInterface::IceServer server;
|
| - server.urls.push_back("stun:hostname");
|
| - server.urls.push_back("turn:hostname");
|
| - servers.push_back(server);
|
| - EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| -}
|
| -
|
| -// Ensure that TURN servers are given unique priorities,
|
| -// so that their resulting candidates have unique priorities.
|
| -TEST_F(IceServerParsingTest, TurnServerPrioritiesUnique) {
|
| - PeerConnectionInterface::IceServers servers;
|
| - PeerConnectionInterface::IceServer server;
|
| - server.urls.push_back("turn:hostname");
|
| - server.urls.push_back("turn:hostname2");
|
| - servers.push_back(server);
|
| - EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
|
| - EXPECT_EQ(2U, turn_servers_.size());
|
| - EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
|
| -}
|
| -
|
| -#endif // if !defined(THREAD_SANITIZER)
|
|
|
Chromium Code Reviews
Issue 1610243002:
Move talk/app/webrtc to webrtc/api (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master