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webrtc / src / a6d26ec6a2a6eccb35b3456010956bf4f53a2659 / . / webrtc / api / peerconnectioninterface_unittest.cc
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/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include <string>
#include <utility>
#include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/api/audiotrack.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/mediastream.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnection.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/api/rtpsenderinterface.h"
#include "webrtc/api/streamcollection.h"
#ifdef WEBRTC_ANDROID
#include "webrtc/api/test/androidtestinitializer.h"
#endif
#include "webrtc/api/test/fakeconstraints.h"
#include "webrtc/api/test/fakertccertificategenerator.h"
#include "webrtc/api/test/fakevideotracksource.h"
#include "webrtc/api/test/mockpeerconnectionobservers.h"
#include "webrtc/api/test/testsdpstrings.h"
#include "webrtc/api/videocapturertracksource.h"
#include "webrtc/api/videotrack.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/thread.h"
#include "webrtc/media/base/fakevideocapturer.h"
#include "webrtc/media/sctp/sctpdataengine.h"
#include "webrtc/p2p/base/fakeportallocator.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/pc/mediasession.h"
static const char kStreamLabel1[] = "local_stream_1";
static const char kStreamLabel2[] = "local_stream_2";
static const char kStreamLabel3[] = "local_stream_3";
static const int kDefaultStunPort = 3478;
static const char kStunAddressOnly[] = "stun:address";
static const char kStunInvalidPort[] = "stun:address:-1";
static const char kStunAddressPortAndMore1[] = "stun:address:port:more";
static const char kStunAddressPortAndMore2[] = "stun:address:port more";
static const char kTurnIceServerUri[] = "turn:user@turn.example.org";
static const char kTurnUsername[] = "user";
static const char kTurnPassword[] = "password";
static const char kTurnHostname[] = "turn.example.org";
static const uint32_t kTimeout = 10000U;
static const char kStreams[][8] = {"stream1", "stream2"};
static const char kAudioTracks[][32] = {"audiotrack0", "audiotrack1"};
static const char kVideoTracks[][32] = {"videotrack0", "videotrack1"};
static const char kRecvonly[] = "recvonly";
static const char kSendrecv[] = "sendrecv";
// Reference SDP with a MediaStream with label "stream1" and audio track with
// id "audio_1" and a video track with id "video_1;
static const char kSdpStringWithStream1[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtpmap:120 VP8/90000\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 mslabel:stream1\r\n"
"a=ssrc:2 label:videotrack0\r\n";
// Reference SDP with two MediaStreams with label "stream1" and "stream2. Each
// MediaStreams have one audio track and one video track.
// This uses MSID.
static const char kSdpStringWithStream1And2[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=msid-semantic: WMS stream1 stream2\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n"
"a=ssrc:3 cname:stream2\r\n"
"a=ssrc:3 msid:stream2 audiotrack1\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtpmap:120 VP8/0\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n"
"a=ssrc:4 cname:stream2\r\n"
"a=ssrc:4 msid:stream2 videotrack1\r\n";
// Reference SDP without MediaStreams. Msid is not supported.
static const char kSdpStringWithoutStreams[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams. Msid is supported.
static const char kSdpStringWithMsidWithoutStreams[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=msid-semantic: WMS\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams and audio only.
static const char kSdpStringWithoutStreamsAudioOnly[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
// Reference SENDONLY SDP without MediaStreams. Msid is not supported.
static const char kSdpStringSendOnlyWithoutStreams[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtpmap:120 VP8/90000\r\n";
static const char kSdpStringInit[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=msid-semantic: WMS\r\n";
static const char kSdpStringAudio[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
static const char kSdpStringVideo[] =
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtpmap:120 VP8/90000\r\n";
static const char kSdpStringMs1Audio0[] =
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n";
static const char kSdpStringMs1Video0[] =
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n";
static const char kSdpStringMs1Audio1[] =
"a=ssrc:3 cname:stream1\r\n"
"a=ssrc:3 msid:stream1 audiotrack1\r\n";
static const char kSdpStringMs1Video1[] =
"a=ssrc:4 cname:stream1\r\n"
"a=ssrc:4 msid:stream1 videotrack1\r\n";
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
using ::testing::Exactly;
using cricket::StreamParams;
using webrtc::AudioSourceInterface;
using webrtc::AudioTrack;
using webrtc::AudioTrackInterface;
using webrtc::DataBuffer;
using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
using webrtc::IceCandidateInterface;
using webrtc::JsepSessionDescription;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStream;
using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
using webrtc::MockCreateSessionDescriptionObserver;
using webrtc::MockDataChannelObserver;
using webrtc::MockSetSessionDescriptionObserver;
using webrtc::MockStatsObserver;
using webrtc::NotifierInterface;
using webrtc::ObserverInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
using webrtc::RtpReceiverInterface;
using webrtc::RtpSenderInterface;
using webrtc::SdpParseError;
using webrtc::SessionDescriptionInterface;
using webrtc::StreamCollection;
using webrtc::StreamCollectionInterface;
using webrtc::VideoTrackSourceInterface;
using webrtc::VideoTrack;
using webrtc::VideoTrackInterface;
typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
namespace {
// Gets the first ssrc of given content type from the ContentInfo.
bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) {
if (!content_info || !ssrc) {
return false;
}
const cricket::MediaContentDescription* media_desc =
static_cast<const cricket::MediaContentDescription*>(
content_info->description);
if (!media_desc || media_desc->streams().empty()) {
return false;
}
*ssrc = media_desc->streams().begin()->first_ssrc();
return true;
}
void SetSsrcToZero(std::string* sdp) {
const char kSdpSsrcAtribute[] = "a=ssrc:";
const char kSdpSsrcAtributeZero[] = "a=ssrc:0";
size_t ssrc_pos = 0;
while ((ssrc_pos = sdp->find(kSdpSsrcAtribute, ssrc_pos)) !=
std::string::npos) {
size_t end_ssrc = sdp->find(" ", ssrc_pos);
sdp->replace(ssrc_pos, end_ssrc - ssrc_pos, kSdpSsrcAtributeZero);
ssrc_pos = end_ssrc;
}
}
// Check if |streams| contains the specified track.
bool ContainsTrack(const std::vector<cricket::StreamParams>& streams,
const std::string& stream_label,
const std::string& track_id) {
for (const cricket::StreamParams& params : streams) {
if (params.sync_label == stream_label && params.id == track_id) {
return true;
}
}
return false;
}
// Check if |senders| contains the specified sender, by id.
bool ContainsSender(
const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
const std::string& id) {
for (const auto& sender : senders) {
if (sender->id() == id) {
return true;
}
}
return false;
}
// Check if |senders| contains the specified sender, by id and stream id.
bool ContainsSender(
const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
const std::string& id,
const std::string& stream_id) {
for (const auto& sender : senders) {
if (sender->id() == id && sender->stream_ids()[0] == stream_id) {
return true;
}
}
return false;
}
// Create a collection of streams.
// CreateStreamCollection(1) creates a collection that
// correspond to kSdpStringWithStream1.
// CreateStreamCollection(2) correspond to kSdpStringWithStream1And2.
rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
int number_of_streams,
int tracks_per_stream) {
rtc::scoped_refptr<StreamCollection> local_collection(
StreamCollection::Create());
for (int i = 0; i < number_of_streams; ++i) {
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
webrtc::MediaStream::Create(kStreams[i]));
for (int j = 0; j < tracks_per_stream; ++j) {
// Add a local audio track.
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j],
nullptr));
stream->AddTrack(audio_track);
// Add a local video track.
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j],
webrtc::FakeVideoTrackSource::Create()));
stream->AddTrack(video_track);
}
local_collection->AddStream(stream);
}
return local_collection;
}
// Check equality of StreamCollections.
bool CompareStreamCollections(StreamCollectionInterface* s1,
StreamCollectionInterface* s2) {
if (s1 == nullptr || s2 == nullptr || s1->count() != s2->count()) {
return false;
}
for (size_t i = 0; i != s1->count(); ++i) {
if (s1->at(i)->label() != s2->at(i)->label()) {
return false;
}
webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
if (audio_tracks1.size() != audio_tracks2.size()) {
return false;
}
for (size_t j = 0; j != audio_tracks1.size(); ++j) {
if (audio_tracks1[j]->id() != audio_tracks2[j]->id()) {
return false;
}
}
if (video_tracks1.size() != video_tracks2.size()) {
return false;
}
for (size_t j = 0; j != video_tracks1.size(); ++j) {
if (video_tracks1[j]->id() != video_tracks2[j]->id()) {
return false;
}
}
}
return true;
}
// Helper class to test Observer.
class MockTrackObserver : public ObserverInterface {
public:
explicit MockTrackObserver(NotifierInterface* notifier)
: notifier_(notifier) {
notifier_->RegisterObserver(this);
}
~MockTrackObserver() { Unregister(); }
void Unregister() {
if (notifier_) {
notifier_->UnregisterObserver(this);
notifier_ = nullptr;
}
}
MOCK_METHOD0(OnChanged, void());
private:
NotifierInterface* notifier_;
};
class MockPeerConnectionObserver : public PeerConnectionObserver {
public:
MockPeerConnectionObserver() : remote_streams_(StreamCollection::Create()) {}
virtual ~MockPeerConnectionObserver() {
}
void SetPeerConnectionInterface(PeerConnectionInterface* pc) {
pc_ = pc;
if (pc) {
state_ = pc_->signaling_state();
}
}
void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) override {
EXPECT_EQ(pc_->signaling_state(), new_state);
state_ = new_state;
}
// TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
virtual void OnStateChange(StateType state_changed) {
if (pc_.get() == NULL)
return;
switch (state_changed) {
case kSignalingState:
// OnSignalingChange and OnStateChange(kSignalingState) should always
// be called approximately simultaneously. To ease testing, we require
// that they always be called in that order. This check verifies
// that OnSignalingChange has just been called.
EXPECT_EQ(pc_->signaling_state(), state_);
break;
case kIceState:
ADD_FAILURE();
break;
default:
ADD_FAILURE();
break;
}
}
MediaStreamInterface* RemoteStream(const std::string& label) {
return remote_streams_->find(label);
}
StreamCollectionInterface* remote_streams() const { return remote_streams_; }
void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override {
last_added_stream_ = stream;
remote_streams_->AddStream(stream);
}
void OnRemoveStream(
rtc::scoped_refptr<MediaStreamInterface> stream) override {
last_removed_stream_ = stream;
remote_streams_->RemoveStream(stream);
}
void OnRenegotiationNeeded() override { renegotiation_needed_ = true; }
void OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) override {
last_datachannel_ = data_channel;
}
void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) override {
EXPECT_EQ(pc_->ice_connection_state(), new_state);
callback_triggered = true;
}
void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) override {
EXPECT_EQ(pc_->ice_gathering_state(), new_state);
ice_complete_ = new_state == PeerConnectionInterface::kIceGatheringComplete;
callback_triggered = true;
}
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
EXPECT_NE(PeerConnectionInterface::kIceGatheringNew,
pc_->ice_gathering_state());
std::string sdp;
EXPECT_TRUE(candidate->ToString(&sdp));
EXPECT_LT(0u, sdp.size());
last_candidate_.reset(webrtc::CreateIceCandidate(candidate->sdp_mid(),
candidate->sdp_mline_index(), sdp, NULL));
EXPECT_TRUE(last_candidate_.get() != NULL);
callback_triggered = true;
}
void OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) override {
callback_triggered = true;
}
void OnIceConnectionReceivingChange(bool receiving) override {
callback_triggered = true;
}
// Returns the label of the last added stream.
// Empty string if no stream have been added.
std::string GetLastAddedStreamLabel() {
if (last_added_stream_.get())
return last_added_stream_->label();
return "";
}
std::string GetLastRemovedStreamLabel() {
if (last_removed_stream_.get())
return last_removed_stream_->label();
return "";
}
rtc::scoped_refptr<PeerConnectionInterface> pc_;
PeerConnectionInterface::SignalingState state_;
std::unique_ptr<IceCandidateInterface> last_candidate_;
rtc::scoped_refptr<DataChannelInterface> last_datachannel_;
rtc::scoped_refptr<StreamCollection> remote_streams_;
bool renegotiation_needed_ = false;
bool ice_complete_ = false;
bool callback_triggered = false;
private:
rtc::scoped_refptr<MediaStreamInterface> last_added_stream_;
rtc::scoped_refptr<MediaStreamInterface> last_removed_stream_;
};
} // namespace
// The PeerConnectionMediaConfig tests below verify that configuration
// and constraints are propagated into the MediaConfig passed to
// CreateMediaController. These settings are intended for MediaChannel
// constructors, but that is not exercised by these unittest.
class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
public:
webrtc::MediaControllerInterface* CreateMediaController(
const cricket::MediaConfig& config) const override {
create_media_controller_called_ = true;
create_media_controller_config_ = config;
webrtc::MediaControllerInterface* mc =
PeerConnectionFactory::CreateMediaController(config);
EXPECT_TRUE(mc != nullptr);
return mc;
}
cricket::TransportController* CreateTransportController(
cricket::PortAllocator* port_allocator,
bool redetermine_role_on_ice_restart) override {
transport_controller = new cricket::TransportController(
rtc::Thread::Current(), rtc::Thread::Current(), port_allocator,
redetermine_role_on_ice_restart);
return transport_controller;
}
cricket::TransportController* transport_controller;
// Mutable, so they can be modified in the above const-declared method.
mutable bool create_media_controller_called_ = false;
mutable cricket::MediaConfig create_media_controller_config_;
};
class PeerConnectionInterfaceTest : public testing::Test {
protected:
PeerConnectionInterfaceTest() {
#ifdef WEBRTC_ANDROID
webrtc::InitializeAndroidObjects();
#endif
}
virtual void SetUp() {
pc_factory_ = webrtc::CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
nullptr, nullptr, nullptr);
ASSERT_TRUE(pc_factory_);
pc_factory_for_test_ =
new rtc::RefCountedObject<PeerConnectionFactoryForTest>();
pc_factory_for_test_->Initialize();
}
void CreatePeerConnection() {
CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(), nullptr);
}
void CreatePeerConnection(webrtc::MediaConstraintsInterface* constraints) {
CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
constraints);
}
void CreatePeerConnectionWithIceTransportsType(
PeerConnectionInterface::IceTransportsType type) {
PeerConnectionInterface::RTCConfiguration config;
config.type = type;
return CreatePeerConnection(config, nullptr);
}
void CreatePeerConnectionWithIceServer(const std::string& uri,
const std::string& password) {
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = uri;
server.password = password;
config.servers.push_back(server);
CreatePeerConnection(config, nullptr);
}
void CreatePeerConnection(PeerConnectionInterface::RTCConfiguration config,
webrtc::MediaConstraintsInterface* constraints) {
std::unique_ptr<cricket::FakePortAllocator> port_allocator(
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
port_allocator_ = port_allocator.get();
// DTLS does not work in a loopback call, so is disabled for most of the
// tests in this file. We only create a FakeIdentityService if the test
// explicitly sets the constraint.
FakeConstraints default_constraints;
if (!constraints) {
constraints = &default_constraints;
default_constraints.AddMandatory(
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
}
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
bool dtls;
if (FindConstraint(constraints,
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
&dtls,
nullptr) && dtls) {
cert_generator.reset(new FakeRTCCertificateGenerator());
}
pc_ = pc_factory_->CreatePeerConnection(
config, constraints, std::move(port_allocator),
std::move(cert_generator), &observer_);
ASSERT_TRUE(pc_.get() != NULL);
observer_.SetPeerConnectionInterface(pc_.get());
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePeerConnectionExpectFail(const std::string& uri) {
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = uri;
config.servers.push_back(server);
rtc::scoped_refptr<PeerConnectionInterface> pc;
pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr, nullptr,
&observer_);
EXPECT_EQ(nullptr, pc);
}
void CreatePeerConnectionWithDifferentConfigurations() {
CreatePeerConnectionWithIceServer(kStunAddressOnly, "");
EXPECT_EQ(1u, port_allocator_->stun_servers().size());
EXPECT_EQ(0u, port_allocator_->turn_servers().size());
EXPECT_EQ("address", port_allocator_->stun_servers().begin()->hostname());
EXPECT_EQ(kDefaultStunPort,
port_allocator_->stun_servers().begin()->port());
CreatePeerConnectionExpectFail(kStunInvalidPort);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore1);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore2);
CreatePeerConnectionWithIceServer(kTurnIceServerUri, kTurnPassword);
EXPECT_EQ(0u, port_allocator_->stun_servers().size());
EXPECT_EQ(1u, port_allocator_->turn_servers().size());
EXPECT_EQ(kTurnUsername,
port_allocator_->turn_servers()[0].credentials.username);
EXPECT_EQ(kTurnPassword,
port_allocator_->turn_servers()[0].credentials.password);
EXPECT_EQ(kTurnHostname,
port_allocator_->turn_servers()[0].ports[0].address.hostname());
}
void ReleasePeerConnection() {
pc_ = NULL;
observer_.SetPeerConnectionInterface(NULL);
}
void AddVideoStream(const std::string& label) {
// Create a local stream.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
rtc::scoped_refptr<VideoTrackSourceInterface> video_source(
pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer(), NULL));
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(label + "v0", video_source));
stream->AddTrack(video_track.get());
EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
void AddVoiceStream(const std::string& label) {
// Create a local stream.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack(label + "a0", NULL));
stream->AddTrack(audio_track.get());
EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
void AddAudioVideoStream(const std::string& stream_label,
const std::string& audio_track_label,
const std::string& video_track_label) {
// Create a local stream.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(stream_label));
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack(
audio_track_label, static_cast<AudioSourceInterface*>(NULL)));
stream->AddTrack(audio_track.get());
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
video_track_label,
pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer())));
stream->AddTrack(video_track.get());
EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
bool offer,
MediaConstraintsInterface* constraints) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockCreateSessionDescriptionObserver>());
if (offer) {
pc_->CreateOffer(observer, constraints);
} else {
pc_->CreateAnswer(observer, constraints);
}
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
desc->reset(observer->release_desc());
return observer->result();
}
bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc,
MediaConstraintsInterface* constraints) {
return DoCreateOfferAnswer(desc, true, constraints);
}
bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
MediaConstraintsInterface* constraints) {
return DoCreateOfferAnswer(desc, false, constraints);
}
bool DoSetSessionDescription(SessionDescriptionInterface* desc, bool local) {
rtc::scoped_refptr<MockSetSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockSetSessionDescriptionObserver>());
if (local) {
pc_->SetLocalDescription(observer, desc);
} else {
pc_->SetRemoteDescription(observer, desc);
}
if (pc_->signaling_state() != PeerConnectionInterface::kClosed) {
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
}
return observer->result();
}
bool DoSetLocalDescription(SessionDescriptionInterface* desc) {
return DoSetSessionDescription(desc, true);
}
bool DoSetRemoteDescription(SessionDescriptionInterface* desc) {
return DoSetSessionDescription(desc, false);
}
// Calls PeerConnection::GetStats and check the return value.
// It does not verify the values in the StatReports since a RTCP packet might
// be required.
bool DoGetStats(MediaStreamTrackInterface* track) {
rtc::scoped_refptr<MockStatsObserver> observer(
new rtc::RefCountedObject<MockStatsObserver>());
if (!pc_->GetStats(
observer, track, PeerConnectionInterface::kStatsOutputLevelStandard))
return false;
EXPECT_TRUE_WAIT(observer->called(), kTimeout);
return observer->called();
}
void InitiateCall() {
CreatePeerConnection();
// Create a local stream with audio&video tracks.
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
CreateOfferReceiveAnswer();
}
// Verify that RTP Header extensions has been negotiated for audio and video.
void VerifyRemoteRtpHeaderExtensions() {
const cricket::MediaContentDescription* desc =
cricket::GetFirstAudioContentDescription(
pc_->remote_description()->description());
ASSERT_TRUE(desc != NULL);
EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
desc = cricket::GetFirstVideoContentDescription(
pc_->remote_description()->description());
ASSERT_TRUE(desc != NULL);
EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
}
void CreateOfferAsRemoteDescription() {
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
SessionDescriptionInterface* remote_offer =
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
sdp, NULL);
EXPECT_TRUE(DoSetRemoteDescription(remote_offer));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
void CreateAndSetRemoteOffer(const std::string& sdp) {
SessionDescriptionInterface* remote_offer =
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
sdp, nullptr);
EXPECT_TRUE(DoSetRemoteDescription(remote_offer));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
void CreateAnswerAsLocalDescription() {
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
// TODO(perkj): Currently SetLocalDescription fails if any parameters in an
// audio codec change, even if the parameter has nothing to do with
// receiving. Not all parameters are serialized to SDP.
// Since CreatePrAnswerAsLocalDescription serialize/deserialize
// the SessionDescription, it is necessary to do that here to in order to
// get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass.
// https://code.google.com/p/webrtc/issues/detail?id=1356
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
SessionDescriptionInterface* new_answer =
webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
sdp, NULL);
EXPECT_TRUE(DoSetLocalDescription(new_answer));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePrAnswerAsLocalDescription() {
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
SessionDescriptionInterface* pr_answer =
webrtc::CreateSessionDescription(SessionDescriptionInterface::kPrAnswer,
sdp, NULL);
EXPECT_TRUE(DoSetLocalDescription(pr_answer));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalPrAnswer, observer_.state_);
}
void CreateOfferReceiveAnswer() {
CreateOfferAsLocalDescription();
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
CreateAnswerAsRemoteDescription(sdp);
}
void CreateOfferAsLocalDescription() {
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
// TODO(perkj): Currently SetLocalDescription fails if any parameters in an
// audio codec change, even if the parameter has nothing to do with
// receiving. Not all parameters are serialized to SDP.
// Since CreatePrAnswerAsLocalDescription serialize/deserialize
// the SessionDescription, it is necessary to do that here to in order to
// get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass.
// https://code.google.com/p/webrtc/issues/detail?id=1356
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
SessionDescriptionInterface* new_offer =
webrtc::CreateSessionDescription(
SessionDescriptionInterface::kOffer,
sdp, NULL);
EXPECT_TRUE(DoSetLocalDescription(new_offer));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
// Wait for the ice_complete message, so that SDP will have candidates.
EXPECT_TRUE_WAIT(observer_.ice_complete_, kTimeout);
}
void CreateAnswerAsRemoteDescription(const std::string& sdp) {
webrtc::JsepSessionDescription* answer = new webrtc::JsepSessionDescription(
SessionDescriptionInterface::kAnswer);
EXPECT_TRUE(answer->Initialize(sdp, NULL));
EXPECT_TRUE(DoSetRemoteDescription(answer));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) {
webrtc::JsepSessionDescription* pr_answer =
new webrtc::JsepSessionDescription(
SessionDescriptionInterface::kPrAnswer);
EXPECT_TRUE(pr_answer->Initialize(sdp, NULL));
EXPECT_TRUE(DoSetRemoteDescription(pr_answer));
EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_);
webrtc::JsepSessionDescription* answer =
new webrtc::JsepSessionDescription(
SessionDescriptionInterface::kAnswer);
EXPECT_TRUE(answer->Initialize(sdp, NULL));
EXPECT_TRUE(DoSetRemoteDescription(answer));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
// Help function used for waiting until a the last signaled remote stream has
// the same label as |stream_label|. In a few of the tests in this file we
// answer with the same session description as we offer and thus we can
// check if OnAddStream have been called with the same stream as we offer to
// send.
void WaitAndVerifyOnAddStream(const std::string& stream_label) {
EXPECT_EQ_WAIT(stream_label, observer_.GetLastAddedStreamLabel(), kTimeout);
}
// Creates an offer and applies it as a local session description.
// Creates an answer with the same SDP an the offer but removes all lines
// that start with a:ssrc"
void CreateOfferReceiveAnswerWithoutSsrc() {
CreateOfferAsLocalDescription();
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
SetSsrcToZero(&sdp);
CreateAnswerAsRemoteDescription(sdp);
}
// This function creates a MediaStream with label kStreams[0] and
// |number_of_audio_tracks| and |number_of_video_tracks| tracks and the
// corresponding SessionDescriptionInterface. The SessionDescriptionInterface
// is returned and the MediaStream is stored in
// |reference_collection_|
std::unique_ptr<SessionDescriptionInterface>
CreateSessionDescriptionAndReference(size_t number_of_audio_tracks,
size_t number_of_video_tracks) {
EXPECT_LE(number_of_audio_tracks, 2u);
EXPECT_LE(number_of_video_tracks, 2u);
reference_collection_ = StreamCollection::Create();
std::string sdp_ms1 = std::string(kSdpStringInit);
std::string mediastream_label = kStreams[0];
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
webrtc::MediaStream::Create(mediastream_label));
reference_collection_->AddStream(stream);
if (number_of_audio_tracks > 0) {
sdp_ms1 += std::string(kSdpStringAudio);
sdp_ms1 += std::string(kSdpStringMs1Audio0);
AddAudioTrack(kAudioTracks[0], stream);
}
if (number_of_audio_tracks > 1) {
sdp_ms1 += kSdpStringMs1Audio1;
AddAudioTrack(kAudioTracks[1], stream);
}
if (number_of_video_tracks > 0) {
sdp_ms1 += std::string(kSdpStringVideo);
sdp_ms1 += std::string(kSdpStringMs1Video0);
AddVideoTrack(kVideoTracks[0], stream);
}
if (number_of_video_tracks > 1) {
sdp_ms1 += kSdpStringMs1Video1;
AddVideoTrack(kVideoTracks[1], stream);
}
return std::unique_ptr<SessionDescriptionInterface>(
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
sdp_ms1, nullptr));
}
void AddAudioTrack(const std::string& track_id,
MediaStreamInterface* stream) {
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(track_id, nullptr));
ASSERT_TRUE(stream->AddTrack(audio_track));
}
void AddVideoTrack(const std::string& track_id,
MediaStreamInterface* stream) {
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
webrtc::VideoTrack::Create(track_id,
webrtc::FakeVideoTrackSource::Create()));
ASSERT_TRUE(stream->AddTrack(video_track));
}
std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
CreatePeerConnection();
AddVoiceStream(kStreamLabel1);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
return offer;
}
std::unique_ptr<SessionDescriptionInterface>
CreateAnswerWithOneAudioStream() {
std::unique_ptr<SessionDescriptionInterface> offer =
CreateOfferWithOneAudioStream();
EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
return answer;
}
const std::string& GetFirstAudioStreamCname(
const SessionDescriptionInterface* desc) {
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(desc->description());
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
return audio_desc->streams()[0].cname;
}
cricket::FakePortAllocator* port_allocator_ = nullptr;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
rtc::scoped_refptr<PeerConnectionFactoryForTest> pc_factory_for_test_;
rtc::scoped_refptr<PeerConnectionInterface> pc_;
MockPeerConnectionObserver observer_;
rtc::scoped_refptr<StreamCollection> reference_collection_;
};
// Test that no callbacks on the PeerConnectionObserver are called after the
// PeerConnection is closed.
TEST_F(PeerConnectionInterfaceTest, CloseAndTestCallbackFunctions) {
rtc::scoped_refptr<PeerConnectionInterface> pc(
pc_factory_for_test_->CreatePeerConnection(
PeerConnectionInterface::RTCConfiguration(), nullptr, nullptr,
nullptr, &observer_));
observer_.SetPeerConnectionInterface(pc.get());
pc->Close();
// No callbacks is expected to be called.
observer_.callback_triggered = false;
std::vector<cricket::Candidate> candidates;
pc_factory_for_test_->transport_controller->SignalGatheringState(
cricket::IceGatheringState{});
pc_factory_for_test_->transport_controller->SignalCandidatesGathered(
"", candidates);
pc_factory_for_test_->transport_controller->SignalConnectionState(
cricket::IceConnectionState{});
pc_factory_for_test_->transport_controller->SignalCandidatesRemoved(
candidates);
pc_factory_for_test_->transport_controller->SignalReceiving(false);
EXPECT_FALSE(observer_.callback_triggered);
}
// Generate different CNAMEs when PeerConnections are created.
// The CNAMEs are expected to be generated randomly. It is possible
// that the test fails, though the possibility is very low.
TEST_F(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
std::unique_ptr<SessionDescriptionInterface> offer1 =
CreateOfferWithOneAudioStream();
std::unique_ptr<SessionDescriptionInterface> offer2 =
CreateOfferWithOneAudioStream();
EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
GetFirstAudioStreamCname(offer2.get()));
}
TEST_F(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
std::unique_ptr<SessionDescriptionInterface> answer1 =
CreateAnswerWithOneAudioStream();
std::unique_ptr<SessionDescriptionInterface> answer2 =
CreateAnswerWithOneAudioStream();
EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
GetFirstAudioStreamCname(answer2.get()));
}
TEST_F(PeerConnectionInterfaceTest,
CreatePeerConnectionWithDifferentConfigurations) {
CreatePeerConnectionWithDifferentConfigurations();
}
TEST_F(PeerConnectionInterfaceTest,
CreatePeerConnectionWithDifferentIceTransportsTypes) {
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNone);
EXPECT_EQ(cricket::CF_NONE, port_allocator_->candidate_filter());
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kRelay);
EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNoHost);
EXPECT_EQ(cricket::CF_ALL & ~cricket::CF_HOST,
port_allocator_->candidate_filter());
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kAll);
EXPECT_EQ(cricket::CF_ALL, port_allocator_->candidate_filter());
}
// Test that when a PeerConnection is created with a nonzero candidate pool
// size, the pooled PortAllocatorSession is created with all the attributes
// in the RTCConfiguration.
TEST_F(PeerConnectionInterfaceTest, CreatePeerConnectionWithPooledCandidates) {
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = kStunAddressOnly;
config.servers.push_back(server);
config.type = PeerConnectionInterface::kRelay;
config.disable_ipv6 = true;
config.tcp_candidate_policy =
PeerConnectionInterface::kTcpCandidatePolicyDisabled;
config.candidate_network_policy =
PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
config.ice_candidate_pool_size = 1;
CreatePeerConnection(config, nullptr);
const cricket::FakePortAllocatorSession* session =
static_cast<const cricket::FakePortAllocatorSession*>(
port_allocator_->GetPooledSession());
ASSERT_NE(nullptr, session);
EXPECT_EQ(1UL, session->stun_servers().size());
EXPECT_EQ(0U, session->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6);
EXPECT_LT(0U, session->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
EXPECT_LT(0U,
session->flags() & cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
}
// Test that the PeerConnection initializes the port allocator passed into it,
// and on the correct thread.
TEST_F(PeerConnectionInterfaceTest,
CreatePeerConnectionInitializesPortAllocator) {
rtc::Thread network_thread;
network_thread.Start();
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory(
webrtc::CreatePeerConnectionFactory(
&network_thread, rtc::Thread::Current(), rtc::Thread::Current(),
nullptr, nullptr, nullptr));
std::unique_ptr<cricket::FakePortAllocator> port_allocator(
new cricket::FakePortAllocator(&network_thread, nullptr));
cricket::FakePortAllocator* raw_port_allocator = port_allocator.get();
PeerConnectionInterface::RTCConfiguration config;
rtc::scoped_refptr<PeerConnectionInterface> pc(
pc_factory->CreatePeerConnection(
config, nullptr, std::move(port_allocator), nullptr, &observer_));
// FakePortAllocator RTC_CHECKs that it's initialized on the right thread,
// so all we have to do here is check that it's initialized.
EXPECT_TRUE(raw_port_allocator->initialized());
}
TEST_F(PeerConnectionInterfaceTest, AddStreams) {
CreatePeerConnection();
AddVideoStream(kStreamLabel1);
AddVoiceStream(kStreamLabel2);
ASSERT_EQ(2u, pc_->local_streams()->count());
// Test we can add multiple local streams to one peerconnection.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamLabel3));
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack(kStreamLabel3,
static_cast<AudioSourceInterface*>(NULL)));
stream->AddTrack(audio_track.get());
EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_EQ(3u, pc_->local_streams()->count());
// Remove the third stream.
pc_->RemoveStream(pc_->local_streams()->at(2));
EXPECT_EQ(2u, pc_->local_streams()->count());
// Remove the second stream.
pc_->RemoveStream(pc_->local_streams()->at(1));
EXPECT_EQ(1u, pc_->local_streams()->count());
// Remove the first stream.
pc_->RemoveStream(pc_->local_streams()->at(0));
EXPECT_EQ(0u, pc_->local_streams()->count());
}
// Test that the created offer includes streams we added.
TEST_F(PeerConnectionInterfaceTest, AddedStreamsPresentInOffer) {
CreatePeerConnection();
AddAudioVideoStream(kStreamLabel1, "audio_track", "video_track");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
EXPECT_TRUE(
ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
const cricket::VideoContentDescription* video_desc =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
EXPECT_TRUE(
ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
// Add another stream and ensure the offer includes both the old and new
// streams.
AddAudioVideoStream(kStreamLabel2, "audio_track2", "video_track2");
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
audio_content = cricket::GetFirstAudioContent(offer->description());
audio_desc = static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
EXPECT_TRUE(
ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
EXPECT_TRUE(
ContainsTrack(audio_desc->streams(), kStreamLabel2, "audio_track2"));
video_content = cricket::GetFirstVideoContent(offer->description());
video_desc = static_cast<const cricket::VideoContentDescription*>(
video_content->description);
EXPECT_TRUE(
ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
EXPECT_TRUE(
ContainsTrack(video_desc->streams(), kStreamLabel2, "video_track2"));
}
TEST_F(PeerConnectionInterfaceTest, RemoveStream) {
CreatePeerConnection();
AddVideoStream(kStreamLabel1);
ASSERT_EQ(1u, pc_->local_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
EXPECT_EQ(0u, pc_->local_streams()->count());
}
// Test for AddTrack and RemoveTrack methods.
// Tests that the created offer includes tracks we added,
// and that the RtpSenders are created correctly.
// Also tests that RemoveTrack removes the tracks from subsequent offers.
TEST_F(PeerConnectionInterfaceTest, AddTrackRemoveTrack) {
CreatePeerConnection();
// Create a dummy stream, so tracks share a stream label.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamLabel1));
std::vector<MediaStreamInterface*> stream_list;
stream_list.push_back(stream.get());
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
"video_track",
pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer())));
auto audio_sender = pc_->AddTrack(audio_track, stream_list);
auto video_sender = pc_->AddTrack(video_track, stream_list);
EXPECT_EQ(1UL, audio_sender->stream_ids().size());
EXPECT_EQ(kStreamLabel1, audio_sender->stream_ids()[0]);
EXPECT_EQ("audio_track", audio_sender->id());
EXPECT_EQ(audio_track, audio_sender->track());
EXPECT_EQ(1UL, video_sender->stream_ids().size());
EXPECT_EQ(kStreamLabel1, video_sender->stream_ids()[0]);
EXPECT_EQ("video_track", video_sender->id());
EXPECT_EQ(video_track, video_sender->track());
// Now create an offer and check for the senders.
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
EXPECT_TRUE(
ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
const cricket::VideoContentDescription* video_desc =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
EXPECT_TRUE(
ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
EXPECT_TRUE(DoSetLocalDescription(offer.release()));
// Now try removing the tracks.
EXPECT_TRUE(pc_->RemoveTrack(audio_sender));
EXPECT_TRUE(pc_->RemoveTrack(video_sender));
// Create a new offer and ensure it doesn't contain the removed senders.
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
audio_content = cricket::GetFirstAudioContent(offer->description());
audio_desc = static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
EXPECT_FALSE(
ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
video_content = cricket::GetFirstVideoContent(offer->description());
video_desc = static_cast<const cricket::VideoContentDescription*>(
video_content->description);
EXPECT_FALSE(
ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
EXPECT_TRUE(DoSetLocalDescription(offer.release()));
// Calling RemoveTrack on a sender no longer attached to a PeerConnection
// should return false.
EXPECT_FALSE(pc_->RemoveTrack(audio_sender));
EXPECT_FALSE(pc_->RemoveTrack(video_sender));
}
// Test creating senders without a stream specified,
// expecting a random stream ID to be generated.
TEST_F(PeerConnectionInterfaceTest, AddTrackWithoutStream) {
CreatePeerConnection();
// Create a dummy stream, so tracks share a stream label.
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
"video_track",
pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer())));
auto audio_sender =
pc_->AddTrack(audio_track, std::vector<MediaStreamInterface*>());
auto video_sender =
pc_->AddTrack(video_track, std::vector<MediaStreamInterface*>());
EXPECT_EQ("audio_track", audio_sender->id());
EXPECT_EQ(audio_track, audio_sender->track());
EXPECT_EQ("video_track", video_sender->id());
EXPECT_EQ(video_track, video_sender->track());
// If the ID is truly a random GUID, it should be infinitely unlikely they
// will be the same.
EXPECT_NE(video_sender->stream_ids(), audio_sender->stream_ids());
}
TEST_F(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) {
InitiateCall();
WaitAndVerifyOnAddStream(kStreamLabel1);
VerifyRemoteRtpHeaderExtensions();
}
TEST_F(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) {
CreatePeerConnection();
AddVideoStream(kStreamLabel1);
CreateOfferAsLocalDescription();
std::string offer;
EXPECT_TRUE(pc_->local_description()->ToString(&offer));
CreatePrAnswerAndAnswerAsRemoteDescription(offer);
WaitAndVerifyOnAddStream(kStreamLabel1);
}
TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) {
CreatePeerConnection();
AddVideoStream(kStreamLabel1);
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
WaitAndVerifyOnAddStream(kStreamLabel1);
}
TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
CreatePeerConnection();
AddVideoStream(kStreamLabel1);
CreateOfferAsRemoteDescription();
CreatePrAnswerAsLocalDescription();
CreateAnswerAsLocalDescription();
WaitAndVerifyOnAddStream(kStreamLabel1);
}
TEST_F(PeerConnectionInterfaceTest, Renegotiate) {
InitiateCall();
ASSERT_EQ(1u, pc_->remote_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
CreateOfferReceiveAnswer();
EXPECT_EQ(0u, pc_->remote_streams()->count());
AddVideoStream(kStreamLabel1);
CreateOfferReceiveAnswer();
}
// Tests that after negotiating an audio only call, the respondent can perform a
// renegotiation that removes the audio stream.
TEST_F(PeerConnectionInterfaceTest, RenegotiateAudioOnly) {
CreatePeerConnection();
AddVoiceStream(kStreamLabel1);
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
ASSERT_EQ(1u, pc_->remote_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
CreateOfferReceiveAnswer();
EXPECT_EQ(0u, pc_->remote_streams()->count());
}
// Test that candidates are generated and that we can parse our own candidates.
TEST_F(PeerConnectionInterfaceTest, IceCandidates) {
CreatePeerConnection();
EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate_.get()));
// SetRemoteDescription takes ownership of offer.
std::unique_ptr<SessionDescriptionInterface> offer;
AddVideoStream(kStreamLabel1);
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
// SetLocalDescription takes ownership of answer.
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(answer.release()));
EXPECT_TRUE_WAIT(observer_.last_candidate_.get() != NULL, kTimeout);
EXPECT_TRUE_WAIT(observer_.ice_complete_, kTimeout);
EXPECT_TRUE(pc_->AddIceCandidate(observer_.last_candidate_.get()));
}
// Test that CreateOffer and CreateAnswer will fail if the track labels are
// not unique.
TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
CreatePeerConnection();
// Create a regular offer for the CreateAnswer test later.
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(offer);
offer.reset();
// Create a local stream with audio&video tracks having same label.
AddAudioVideoStream(kStreamLabel1, "track_label", "track_label");
// Test CreateOffer
EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
// Test CreateAnswer
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
}
// Test that we will get different SSRCs for each tracks in the offer and answer
// we created.
TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
CreatePeerConnection();
// Create a local stream with audio&video tracks having different labels.
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
// Test CreateOffer
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
int audio_ssrc = 0;
int video_ssrc = 0;
EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(offer->description()),
&audio_ssrc));
EXPECT_TRUE(GetFirstSsrc(GetFirstVideoContent(offer->description()),
&video_ssrc));
EXPECT_NE(audio_ssrc, video_ssrc);
// Test CreateAnswer
EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
audio_ssrc = 0;
video_ssrc = 0;
EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(answer->description()),
&audio_ssrc));
EXPECT_TRUE(GetFirstSsrc(GetFirstVideoContent(answer->description()),
&video_ssrc));
EXPECT_NE(audio_ssrc, video_ssrc);
}
// Test that it's possible to call AddTrack on a MediaStream after adding
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
CreatePeerConnection();
// Create audio stream and add to PeerConnection.
AddVoiceStream(kStreamLabel1);
MediaStreamInterface* stream = pc_->local_streams()->at(0);
// Add video track to the audio-only stream.
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
"video_label",
pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer())));
stream->AddTrack(video_track.get());
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(video_desc != nullptr);
}
// Test that it's possible to call RemoveTrack on a MediaStream after adding
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
CreatePeerConnection();
// Create audio/video stream and add to PeerConnection.
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
MediaStreamInterface* stream = pc_->local_streams()->at(0);
// Remove the video track.
stream->RemoveTrack(stream->GetVideoTracks()[0]);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(video_desc == nullptr);
}
// Test creating a sender with a stream ID, and ensure the ID is populated
// in the offer.
TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
CreatePeerConnection();
pc_->CreateSender("video", kStreamLabel1);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
ASSERT_TRUE(video_desc != nullptr);
ASSERT_EQ(1u, video_desc->streams().size());
EXPECT_EQ(kStreamLabel1, video_desc->streams()[0].sync_label);
}
// Test that we can specify a certain track that we want statistics about.
TEST_F(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
InitiateCall();
ASSERT_LT(0u, pc_->remote_streams()->count());
ASSERT_LT(0u, pc_->remote_streams()->at(0)->GetAudioTracks().size());
rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
pc_->remote_streams()->at(0)->GetAudioTracks()[0];
EXPECT_TRUE(DoGetStats(remote_audio));
// Remove the stream. Since we are sending to our selves the local
// and the remote stream is the same.
pc_->RemoveStream(pc_->local_streams()->at(0));
// Do a re-negotiation.
CreateOfferReceiveAnswer();
ASSERT_EQ(0u, pc_->remote_streams()->count());
// Test that we still can get statistics for the old track. Even if it is not
// sent any longer.
EXPECT_TRUE(DoGetStats(remote_audio));
}
// Test that we can get stats on a video track.
TEST_F(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
InitiateCall();
ASSERT_LT(0u, pc_->remote_streams()->count());
ASSERT_LT(0u, pc_->remote_streams()->at(0)->GetVideoTracks().size());
rtc::scoped_refptr<MediaStreamTrackInterface> remote_video =
pc_->remote_streams()->at(0)->GetVideoTracks()[0];
EXPECT_TRUE(DoGetStats(remote_video));
}
// Test that we don't get statistics for an invalid track.
// TODO(tommi): Fix this test. DoGetStats will return true
// for the unknown track (since GetStats is async), but no
// data is returned for the track.
TEST_F(PeerConnectionInterfaceTest, DISABLED_GetStatsForInvalidTrack) {
InitiateCall();
rtc::scoped_refptr<AudioTrackInterface> unknown_audio_track(
pc_factory_->CreateAudioTrack("unknown track", NULL));
EXPECT_FALSE(DoGetStats(unknown_audio_track));
}
// This test setup two RTP data channels in loop back.
TEST_F(PeerConnectionInterfaceTest, TestDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
std::unique_ptr<MockDataChannelObserver> observer2(
new MockDataChannelObserver(data2));
EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
std::string data_to_send1 = "testing testing";
std::string data_to_send2 = "testing something else";
EXPECT_FALSE(data1->Send(DataBuffer(data_to_send1)));
CreateOfferReceiveAnswer();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
EXPECT_TRUE(data1->Send(DataBuffer(data_to_send1)));
EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
EXPECT_EQ_WAIT(data_to_send1, observer1->last_message(), kTimeout);
EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
data1->Close();
EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
CreateOfferReceiveAnswer();
EXPECT_FALSE(observer1->IsOpen());
EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
EXPECT_TRUE(observer2->IsOpen());
data_to_send2 = "testing something else again";
EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
}
// This test verifies that sendnig binary data over RTP data channels should
// fail.
TEST_F(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
std::unique_ptr<MockDataChannelObserver> observer2(
new MockDataChannelObserver(data2));
EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
CreateOfferReceiveAnswer();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
rtc::CopyOnWriteBuffer buffer("test", 4);
EXPECT_FALSE(data1->Send(DataBuffer(buffer, true)));
}
// This test setup a RTP data channels in loop back and test that a channel is
// opened even if the remote end answer with a zero SSRC.
TEST_F(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
CreateOfferReceiveAnswerWithoutSsrc();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
data1->Close();
EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
CreateOfferReceiveAnswerWithoutSsrc();
EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
EXPECT_FALSE(observer1->IsOpen());
}
// This test that if a data channel is added in an answer a receive only channel
// channel is created.
TEST_F(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
std::string offer_label = "offer_channel";
rtc::scoped_refptr<DataChannelInterface> offer_channel =
pc_->CreateDataChannel(offer_label, NULL);
CreateOfferAsLocalDescription();
// Replace the data channel label in the offer and apply it as an answer.
std::string receive_label = "answer_channel";
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
rtc::replace_substrs(offer_label.c_str(), offer_label.length(),
receive_label.c_str(), receive_label.length(),
&sdp);
CreateAnswerAsRemoteDescription(sdp);
// Verify that a new incoming data channel has been created and that
// it is open but can't we written to.
ASSERT_TRUE(observer_.last_datachannel_ != NULL);
DataChannelInterface* received_channel = observer_.last_datachannel_;
EXPECT_EQ(DataChannelInterface::kConnecting, received_channel->state());
EXPECT_EQ(receive_label, received_channel->label());
EXPECT_FALSE(received_channel->Send(DataBuffer("something")));
// Verify that the channel we initially offered has been rejected.
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
// Do another offer / answer exchange and verify that the data channel is
// opened.
CreateOfferReceiveAnswer();
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, received_channel->state(),
kTimeout);
}
// This test that no data channel is returned if a reliable channel is
// requested.
// TODO(perkj): Remove this test once reliable channels are implemented.
TEST_F(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
std::string label = "test";
webrtc::DataChannelInit config;
config.reliable = true;
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, &config);
EXPECT_TRUE(channel == NULL);
}
// Verifies that duplicated label is not allowed for RTP data channel.
TEST_F(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
std::string label = "test";
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(channel, nullptr);
rtc::scoped_refptr<DataChannelInterface> dup_channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_EQ(dup_channel, nullptr);
}
// This tests that a SCTP data channel is returned using different
// DataChannelInit configurations.
TEST_F(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
FakeConstraints constraints;
constraints.SetAllowDtlsSctpDataChannels();
CreatePeerConnection(&constraints);
webrtc::DataChannelInit config;
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel("1", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_TRUE(channel->reliable());
EXPECT_TRUE(observer_.renegotiation_needed_);
observer_.renegotiation_needed_ = false;
config.ordered = false;
channel = pc_->CreateDataChannel("2", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_TRUE(channel->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
config.ordered = true;
config.maxRetransmits = 0;
channel = pc_->CreateDataChannel("3", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_FALSE(channel->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
config.maxRetransmits = -1;
config.maxRetransmitTime = 0;
channel = pc_->CreateDataChannel("4", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_FALSE(channel->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
}
// This tests that no data channel is returned if both maxRetransmits and
// maxRetransmitTime are set for SCTP data channels.
TEST_F(PeerConnectionInterfaceTest,
CreateSctpDataChannelShouldFailForInvalidConfig) {
FakeConstraints constraints;
constraints.SetAllowDtlsSctpDataChannels();
CreatePeerConnection(&constraints);
std::string label = "test";
webrtc::DataChannelInit config;
config.maxRetransmits = 0;
config.maxRetransmitTime = 0;
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, &config);
EXPECT_TRUE(channel == NULL);
}
// The test verifies that creating a SCTP data channel with an id already in use
// or out of range should fail.
TEST_F(PeerConnectionInterfaceTest,
CreateSctpDataChannelWithInvalidIdShouldFail) {
FakeConstraints constraints;
constraints.SetAllowDtlsSctpDataChannels();
CreatePeerConnection(&constraints);
webrtc::DataChannelInit config;
rtc::scoped_refptr<DataChannelInterface> channel;
config.id = 1;
channel = pc_->CreateDataChannel("1", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_EQ(1, channel->id());
channel = pc_->CreateDataChannel("x", &config);
EXPECT_TRUE(channel == NULL);
config.id = cricket::kMaxSctpSid;
channel = pc_->CreateDataChannel("max", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_EQ(config.id, channel->id());
config.id = cricket::kMaxSctpSid + 1;
channel = pc_->CreateDataChannel("x", &config);
EXPECT_TRUE(channel == NULL);
}
// Verifies that duplicated label is allowed for SCTP data channel.
TEST_F(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
std::string label = "test";
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(channel, nullptr);
rtc::scoped_refptr<DataChannelInterface> dup_channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(dup_channel, nullptr);
}
// This test verifies that OnRenegotiationNeeded is fired for every new RTP
// DataChannel.
TEST_F(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> dc1 =
pc_->CreateDataChannel("test1", NULL);
EXPECT_TRUE(observer_.renegotiation_needed_);
observer_.renegotiation_needed_ = false;
rtc::scoped_refptr<DataChannelInterface> dc2 =
pc_->CreateDataChannel("test2", NULL);
EXPECT_TRUE(observer_.renegotiation_needed_);
}
// This test that a data channel closes when a PeerConnection is deleted/closed.
TEST_F(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
std::unique_ptr<MockDataChannelObserver> observer2(
new MockDataChannelObserver(data2));
CreateOfferReceiveAnswer();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
ReleasePeerConnection();
EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
EXPECT_EQ(DataChannelInterface::kClosed, data2->state());
}
// This test that data channels can be rejected in an answer.
TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> offer_channel(
pc_->CreateDataChannel("offer_channel", NULL));
CreateOfferAsLocalDescription();
// Create an answer where the m-line for data channels are rejected.
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
webrtc::JsepSessionDescription* answer = new webrtc::JsepSessionDescription(
SessionDescriptionInterface::kAnswer);
EXPECT_TRUE(answer->Initialize(sdp, NULL));
cricket::ContentInfo* data_info =
answer->description()->GetContentByName("data");
data_info->rejected = true;
DoSetRemoteDescription(answer);
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
}
// Test that we can create a session description from an SDP string from
// FireFox, use it as a remote session description, generate an answer and use
// the answer as a local description.
TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
SessionDescriptionInterface* desc =
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
webrtc::kFireFoxSdpOffer, nullptr);
EXPECT_TRUE(DoSetSessionDescription(desc, false));
CreateAnswerAsLocalDescription();
ASSERT_TRUE(pc_->local_description() != NULL);
ASSERT_TRUE(pc_->remote_description() != NULL);
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
content =
cricket::GetFirstVideoContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
#ifdef HAVE_SCTP
content =
cricket::GetFirstDataContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_TRUE(content->rejected);
#endif
}
// Test that we can create an audio only offer and receive an answer with a
// limited set of audio codecs and receive an updated offer with more audio
// codecs, where the added codecs are not supported.
TEST_F(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
CreatePeerConnection();
AddVoiceStream("audio_label");
CreateOfferAsLocalDescription();
SessionDescriptionInterface* answer =
webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
webrtc::kAudioSdp, nullptr);
EXPECT_TRUE(DoSetSessionDescription(answer, false));
SessionDescriptionInterface* updated_offer =
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
webrtc::kAudioSdpWithUnsupportedCodecs,
nullptr);
EXPECT_TRUE(DoSetSessionDescription(updated_offer, false));
CreateAnswerAsLocalDescription();
}
// Test that if we're receiving (but not sending) a track, subsequent offers
// will have m-lines with a=recvonly.
TEST_F(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
CreateAnswerAsLocalDescription();
// At this point we should be receiving stream 1, but not sending anything.
// A new offer should be recvonly.
std::unique_ptr<SessionDescriptionInterface> offer;
DoCreateOffer(&offer, nullptr);
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
const cricket::VideoContentDescription* video_desc =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
ASSERT_EQ(cricket::MD_RECVONLY, video_desc->direction());
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
ASSERT_EQ(cricket::MD_RECVONLY, audio_desc->direction());
}
// Test that if we're receiving (but not sending) a track, and the
// offerToReceiveVideo/offerToReceiveAudio constraints are explicitly set to
// false, the generated m-lines will be a=inactive.
TEST_F(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
CreateAnswerAsLocalDescription();
// At this point we should be receiving stream 1, but not sending anything.
// A new offer would be recvonly, but we'll set the "no receive" constraints
// to make it inactive.
std::unique_ptr<SessionDescriptionInterface> offer;
FakeConstraints offer_constraints;
offer_constraints.AddMandatory(
webrtc::MediaConstraintsInterface::kOfferToReceiveVideo, false);
offer_constraints.AddMandatory(
webrtc::MediaConstraintsInterface::kOfferToReceiveAudio, false);
DoCreateOffer(&offer, &offer_constraints);
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
const cricket::VideoContentDescription* video_desc =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
ASSERT_EQ(cricket::MD_INACTIVE, video_desc->direction());
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
ASSERT_EQ(cricket::MD_INACTIVE, audio_desc->direction());
}
// Test that we can use SetConfiguration to change the ICE servers of the
// PortAllocator.
TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = "stun:test_hostname";
config.servers.push_back(server);
EXPECT_TRUE(pc_->SetConfiguration(config));
EXPECT_EQ(1u, port_allocator_->stun_servers().size());
EXPECT_EQ("test_hostname",
port_allocator_->stun_servers().begin()->hostname());
}
TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesCandidateFilter) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
EXPECT_TRUE(pc_->SetConfiguration(config));
EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
}
// Test that when SetConfiguration changes both the pool size and other
// attributes, the pooled session is created with the updated attributes.
TEST_F(PeerConnectionInterfaceTest,
SetConfigurationCreatesPooledSessionCorrectly) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config;
config.ice_candidate_pool_size = 1;
PeerConnectionInterface::IceServer server;
server.uri = kStunAddressOnly;
config.servers.push_back(server);
config.type = PeerConnectionInterface::kRelay;
EXPECT_TRUE(pc_->SetConfiguration(config));
const cricket::FakePortAllocatorSession* session =
static_cast<const cricket::FakePortAllocatorSession*>(
port_allocator_->GetPooledSession());
ASSERT_NE(nullptr, session);
EXPECT_EQ(1UL, session->stun_servers().size());
}
// Test that PeerConnection::Close changes the states to closed and all remote
// tracks change state to ended.
TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
// Initialize a PeerConnection and negotiate local and remote session
// description.
InitiateCall();
ASSERT_EQ(1u, pc_->local_streams()->count());
ASSERT_EQ(1u, pc_->remote_streams()->count());
pc_->Close();
EXPECT_EQ(PeerConnectionInterface::kClosed, pc_->signaling_state());
EXPECT_EQ(PeerConnectionInterface::kIceConnectionClosed,
pc_->ice_connection_state());
EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete,
pc_->ice_gathering_state());
EXPECT_EQ(1u, pc_->local_streams()->count());
EXPECT_EQ(1u, pc_->remote_streams()->count());
rtc::scoped_refptr<MediaStreamInterface> remote_stream =
pc_->remote_streams()->at(0);
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
remote_stream->GetAudioTracks()[0]->state(), kTimeout);
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
remote_stream->GetVideoTracks()[0]->state(), kTimeout);
}
// Test that PeerConnection methods fails gracefully after
// PeerConnection::Close has been called.
TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
CreatePeerConnection();
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
ASSERT_EQ(1u, pc_->local_streams()->count());
rtc::scoped_refptr<MediaStreamInterface> local_stream =
pc_->local_streams()->at(0);
pc_->Close();
pc_->RemoveStream(local_stream);
EXPECT_FALSE(pc_->AddStream(local_stream));
ASSERT_FALSE(local_stream->GetAudioTracks().empty());
rtc::scoped_refptr<webrtc::DtmfSenderInterface> dtmf_sender(
pc_->CreateDtmfSender(local_stream->GetAudioTracks()[0]));
EXPECT_TRUE(NULL == dtmf_sender); // local stream has been removed.
EXPECT_TRUE(pc_->CreateDataChannel("test", NULL) == NULL);
EXPECT_TRUE(pc_->local_description() != NULL);
EXPECT_TRUE(pc_->remote_description() != NULL);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
std::string sdp;
ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
SessionDescriptionInterface* remote_offer =
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
sdp, NULL);
EXPECT_FALSE(DoSetRemoteDescription(remote_offer));
ASSERT_TRUE(pc_->local_description()->ToString(&sdp));
SessionDescriptionInterface* local_offer =
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
sdp, NULL);
EXPECT_FALSE(DoSetLocalDescription(local_offer));
}
// Test that GetStats can still be called after PeerConnection::Close.
TEST_F(PeerConnectionInterfaceTest, CloseAndGetStats) {
InitiateCall();
pc_->Close();
DoGetStats(NULL);
}
// NOTE: The series of tests below come from what used to be
// mediastreamsignaling_unittest.cc, and are mostly aimed at testing that
// setting a remote or local description has the expected effects.
// This test verifies that the remote MediaStreams corresponding to a received
// SDP string is created. In this test the two separate MediaStreams are
// signaled.
TEST_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference.get()));
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_TRUE(remote_stream->GetVideoTracks()[0]->GetSource() != nullptr);
// Create a session description based on another SDP with another
// MediaStream.
CreateAndSetRemoteOffer(kSdpStringWithStream1And2);
rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference2.get()));
}
// This test verifies that when remote tracks are added/removed from SDP, the
// created remote streams are updated appropriately.
TEST_F(PeerConnectionInterfaceTest,
AddRemoveTrackFromExistingRemoteMediaStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
std::unique_ptr<SessionDescriptionInterface> desc_ms1 =
CreateSessionDescriptionAndReference(1, 1);
EXPECT_TRUE(DoSetRemoteDescription(desc_ms1.release()));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
// Add extra audio and video tracks to the same MediaStream.
std::unique_ptr<SessionDescriptionInterface> desc_ms1_two_tracks =
CreateSessionDescriptionAndReference(2, 2);
EXPECT_TRUE(DoSetRemoteDescription(desc_ms1_two_tracks.release()));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
rtc::scoped_refptr<AudioTrackInterface> audio_track2 =
observer_.remote_streams()->at(0)->GetAudioTracks()[1];
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state());
rtc::scoped_refptr<VideoTrackInterface> video_track2 =
observer_.remote_streams()->at(0)->GetVideoTracks()[1];
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state());
// Remove the extra audio and video tracks.
std::unique_ptr<SessionDescriptionInterface> desc_ms2 =
CreateSessionDescriptionAndReference(1, 1);
MockTrackObserver audio_track_observer(audio_track2);
MockTrackObserver video_track_observer(video_track2);
EXPECT_CALL(audio_track_observer, OnChanged()).Times(Exactly(1));
EXPECT_CALL(video_track_observer, OnChanged()).Times(Exactly(1));
EXPECT_TRUE(DoSetRemoteDescription(desc_ms2.release()));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
audio_track2->state(), kTimeout);
EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
video_track2->state(), kTimeout);
}
// This tests that remote tracks are ended if a local session description is set
// that rejects the media content type.
TEST_F(PeerConnectionInterfaceTest, RejectMediaContent) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
// First create and set a remote offer, then reject its video content in our
// answer.
CreateAndSetRemoteOffer(kSdpStringWithStream1);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
rtc::scoped_refptr<webrtc::VideoTrackInterface> remote_video =
remote_stream->GetVideoTracks()[0];
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_video->state());
rtc::scoped_refptr<webrtc::AudioTrackInterface> remote_audio =
remote_stream->GetAudioTracks()[0];
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
std::unique_ptr<SessionDescriptionInterface> local_answer;
EXPECT_TRUE(DoCreateAnswer(&local_answer, nullptr));
cricket::ContentInfo* video_info =
local_answer->description()->GetContentByName("video");
video_info->rejected = true;
EXPECT_TRUE(DoSetLocalDescription(local_answer.release()));
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_video->state());
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
// Now create an offer where we reject both video and audio.
std::unique_ptr<SessionDescriptionInterface> local_offer;
EXPECT_TRUE(DoCreateOffer(&local_offer, nullptr));
video_info = local_offer->description()->GetContentByName("video");
ASSERT_TRUE(video_info != nullptr);
video_info->rejected = true;
cricket::ContentInfo* audio_info =
local_offer->description()->GetContentByName("audio");
ASSERT_TRUE(audio_info != nullptr);
audio_info->rejected = true;
EXPECT_TRUE(DoSetLocalDescription(local_offer.release()));
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
remote_audio->state(), kTimeout);
EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
remote_video->state(), kTimeout);
}
// This tests that we won't crash if the remote track has been removed outside
// of PeerConnection and then PeerConnection tries to reject the track.
TEST_F(PeerConnectionInterfaceTest, RemoveTrackThenRejectMediaContent) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
std::unique_ptr<SessionDescriptionInterface> local_answer(
webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
kSdpStringWithStream1, nullptr));
cricket::ContentInfo* video_info =
local_answer->description()->GetContentByName("video");
video_info->rejected = true;
cricket::ContentInfo* audio_info =
local_answer->description()->GetContentByName("audio");
audio_info->rejected = true;
EXPECT_TRUE(DoSetLocalDescription(local_answer.release()));
// No crash is a pass.
}
// This tests that if a recvonly remote description is set, no remote streams
// will be created, even if the description contains SSRCs/MSIDs.
// See: https://code.google.com/p/webrtc/issues/detail?id=5054
TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
std::string recvonly_offer = kSdpStringWithStream1;
rtc::replace_substrs(kSendrecv, strlen(kSendrecv), kRecvonly,
strlen(kRecvonly), &recvonly_offer);
CreateAndSetRemoteOffer(recvonly_offer);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
// This tests that a default MediaStream is created if a remote session
// description doesn't contain any streams and no MSID support.
// It also tests that the default stream is updated if a video m-line is added
// in a subsequent session description.
TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ(0u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("default", remote_stream->label());
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ("defaulta0", remote_stream->GetAudioTracks()[0]->id());
EXPECT_EQ(MediaStreamTrackInterface::kLive,
remote_stream->GetAudioTracks()[0]->state());
ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("defaultv0", remote_stream->GetVideoTracks()[0]->id());
EXPECT_EQ(MediaStreamTrackInterface::kLive,
remote_stream->GetVideoTracks()[0]->state());
}
// This tests that a default MediaStream is created if a remote session
// description doesn't contain any streams and media direction is send only.
TEST_F(PeerConnectionInterfaceTest,
SendOnlySdpWithoutMsidCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringSendOnlyWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("default", remote_stream->label());
}
// This tests that it won't crash when PeerConnection tries to remove
// a remote track that as already been removed from the MediaStream.
TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
// No crash is a pass.
}
// This tests that a default MediaStream is created if the remote session
// description doesn't contain any streams and don't contain an indication if
// MSID is supported.
TEST_F(PeerConnectionInterfaceTest,
SdpWithoutMsidAndStreamsCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
}
// This tests that a default MediaStream is not created if the remote session
// description doesn't contain any streams but does support MSID.
TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithMsidWithoutStreams);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
// This tests that when setting a new description, the old default tracks are
// not destroyed and recreated.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
TEST_F(PeerConnectionInterfaceTest,
DefaultTracksNotDestroyedAndRecreated) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
// Set the track to "disabled", then set a new description and ensure the
// track is still disabled, which ensures it hasn't been recreated.
remote_stream->GetAudioTracks()[0]->set_enabled(false);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_FALSE(remote_stream->GetAudioTracks()[0]->enabled());
}
// This tests that a default MediaStream is not created if a remote session
// description is updated to not have any MediaStreams.
TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference.get()));
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
// This tests that an RtpSender is created when the local description is set
// after adding a local stream.
// TODO(deadbeef): This test and the one below it need to be updated when
// an RtpSender's lifetime isn't determined by when a local description is set.
TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
// Create an offer with 1 stream with 2 tracks of each type.
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(1, 2);
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(offer.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(4u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
// Remove an audio and video track.
pc_->RemoveStream(stream_collection->at(0));
stream_collection = CreateStreamCollection(1, 1);
pc_->AddStream(stream_collection->at(0));
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(offer.release()));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
EXPECT_FALSE(ContainsSender(senders, kAudioTracks[1]));
EXPECT_FALSE(ContainsSender(senders, kVideoTracks[1]));
}
// This tests that an RtpSender is created when the local description is set
// before adding a local stream.
TEST_F(PeerConnectionInterfaceTest,
AddLocalStreamAfterLocalDescriptionChanged) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(1, 2);
// Add a stream to create the offer, but remove it afterwards.
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
pc_->RemoveStream(stream_collection->at(0));
EXPECT_TRUE(DoSetLocalDescription(offer.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(0u, senders.size());
pc_->AddStream(stream_collection->at(0));
senders = pc_->GetSenders();
EXPECT_EQ(4u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
}
// This tests that the expected behavior occurs if the SSRC on a local track is
// changed when SetLocalDescription is called.
TEST_F(PeerConnectionInterfaceTest,
ChangeSsrcOnTrackInLocalSessionDescription) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(2, 1);
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
// Grab a copy of the offer before it gets passed into the PC.
std::unique_ptr<JsepSessionDescription> modified_offer(
new JsepSessionDescription(JsepSessionDescription::kOffer));
modified_offer->Initialize(offer->description()->Copy(), offer->session_id(),
offer->session_version());
EXPECT_TRUE(DoSetLocalDescription(offer.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
// Change the ssrc of the audio and video track.
cricket::MediaContentDescription* desc =
cricket::GetFirstAudioContentDescription(modified_offer->description());
ASSERT_TRUE(desc != NULL);
for (StreamParams& stream : desc->mutable_streams()) {
for (unsigned int& ssrc : stream.ssrcs) {
++ssrc;
}
}
desc =
cricket::GetFirstVideoContentDescription(modified_offer->description());
ASSERT_TRUE(desc != NULL);
for (StreamParams& stream : desc->mutable_streams()) {
for (unsigned int& ssrc : stream.ssrcs) {
++ssrc;
}
}
EXPECT_TRUE(DoSetLocalDescription(modified_offer.release()));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
// TODO(deadbeef): Once RtpSenders expose parameters, check that the SSRC
// changed.
}
// This tests that the expected behavior occurs if a new session description is
// set with the same tracks, but on a different MediaStream.
TEST_F(PeerConnectionInterfaceTest,
SignalSameTracksInSeparateMediaStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(2, 1);
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(offer.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0], kStreams[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0]));
// Add a new MediaStream but with the same tracks as in the first stream.
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1(
webrtc::MediaStream::Create(kStreams[1]));
stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]);
stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]);
pc_->AddStream(stream_1);
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(offer.release()));
auto new_senders = pc_->GetSenders();
// Should be the same senders as before, but with updated stream id.
// Note that this behavior is subject to change in the future.
// We may decide the PC should ignore existing tracks in AddStream.
EXPECT_EQ(senders, new_senders);
EXPECT_TRUE(ContainsSender(new_senders, kAudioTracks[0], kStreams[1]));
EXPECT_TRUE(ContainsSender(new_senders, kVideoTracks[0], kStreams[1]));
}
class PeerConnectionMediaConfigTest : public testing::Test {
protected:
void SetUp() override {
pcf_ = new rtc::RefCountedObject<PeerConnectionFactoryForTest>();
pcf_->Initialize();
}
const cricket::MediaConfig& TestCreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& config,
const MediaConstraintsInterface *constraints) {
pcf_->create_media_controller_called_ = false;
rtc::scoped_refptr<PeerConnectionInterface> pc(pcf_->CreatePeerConnection(
config, constraints, nullptr, nullptr, &observer_));
EXPECT_TRUE(pc.get());
EXPECT_TRUE(pcf_->create_media_controller_called_);
return pcf_->create_media_controller_config_;
}
rtc::scoped_refptr<PeerConnectionFactoryForTest> pcf_;
MockPeerConnectionObserver observer_;
};
// This test verifies the default behaviour with no constraints and a
// default RTCConfiguration.
TEST_F(PeerConnectionMediaConfigTest, TestDefaults) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
const cricket::MediaConfig& media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_FALSE(media_config.enable_dscp);
EXPECT_TRUE(media_config.video.enable_cpu_overuse_detection);
EXPECT_FALSE(media_config.video.disable_prerenderer_smoothing);
EXPECT_FALSE(media_config.video.suspend_below_min_bitrate);
}
// This test verifies the DSCP constraint is recognized and passed to
// the CreateMediaController call.
TEST_F(PeerConnectionMediaConfigTest, TestDscpConstraintTrue) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
constraints.AddOptional(webrtc::MediaConstraintsInterface::kEnableDscp, true);
const cricket::MediaConfig& media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_TRUE(media_config.enable_dscp);
}
// This test verifies the cpu overuse detection constraint is
// recognized and passed to the CreateMediaController call.
TEST_F(PeerConnectionMediaConfigTest, TestCpuOveruseConstraintFalse) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
constraints.AddOptional(
webrtc::MediaConstraintsInterface::kCpuOveruseDetection, false);
const cricket::MediaConfig media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_FALSE(media_config.video.enable_cpu_overuse_detection);
}
// This test verifies that the disable_prerenderer_smoothing flag is
// propagated from RTCConfiguration to the CreateMediaController call.
TEST_F(PeerConnectionMediaConfigTest, TestDisablePrerendererSmoothingTrue) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
config.set_prerenderer_smoothing(false);
const cricket::MediaConfig& media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_TRUE(media_config.video.disable_prerenderer_smoothing);
}
// This test verifies the suspend below min bitrate constraint is
// recognized and passed to the CreateMediaController call.
TEST_F(PeerConnectionMediaConfigTest,
TestSuspendBelowMinBitrateConstraintTrue) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
constraints.AddOptional(
webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
true);
const cricket::MediaConfig media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_TRUE(media_config.video.suspend_below_min_bitrate);
}
// The following tests verify that session options are created correctly.
// TODO(deadbeef): Convert these tests to be more end-to-end. Instead of
// "verify options are converted correctly", should be "pass options into
// CreateOffer and verify the correct offer is produced."
TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidAudioOption) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = RTCOfferAnswerOptions::kUndefined - 1;
cricket::MediaSessionOptions options;
EXPECT_FALSE(ExtractMediaSessionOptions(rtc_options, true, &options));
rtc_options.offer_to_receive_audio =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(ExtractMediaSessionOptions(rtc_options, true, &options));
}
TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidVideoOption) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_video = RTCOfferAnswerOptions::kUndefined - 1;
cricket::MediaSessionOptions options;
EXPECT_FALSE(ExtractMediaSessionOptions(rtc_options, true, &options));
rtc_options.offer_to_receive_video =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(ExtractMediaSessionOptions(rtc_options, true, &options));
}
// Test that a MediaSessionOptions is created for an offer if
// OfferToReceiveAudio and OfferToReceiveVideo options are set.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudioVideo) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 1;
cricket::MediaSessionOptions options;
EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
EXPECT_TRUE(options.has_audio());
EXPECT_TRUE(options.has_video());
EXPECT_TRUE(options.bundle_enabled);
}
// Test that a correct MediaSessionOptions is created for an offer if
// OfferToReceiveAudio is set.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudio) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
cricket::MediaSessionOptions options;
EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
EXPECT_TRUE(options.has_audio());
EXPECT_FALSE(options.has_video());
EXPECT_TRUE(options.bundle_enabled);
}
// Test that a correct MediaSessionOptions is created for an offer if
// the default OfferOptions are used.
TEST(CreateSessionOptionsTest, GetDefaultMediaSessionOptionsForOffer) {
RTCOfferAnswerOptions rtc_options;
cricket::MediaSessionOptions options;
options.transport_options["audio"] = cricket::TransportOptions();
options.transport_options["video"] = cricket::TransportOptions();
EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
EXPECT_TRUE(options.has_audio());
EXPECT_FALSE(options.has_video());
EXPECT_TRUE(options.bundle_enabled);
EXPECT_TRUE(options.vad_enabled);
EXPECT_FALSE(options.transport_options["audio"].ice_restart);
EXPECT_FALSE(options.transport_options["video"].ice_restart);
}
// Test that a correct MediaSessionOptions is created for an offer if
// OfferToReceiveVideo is set.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithVideo) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 0;
rtc_options.offer_to_receive_video = 1;
cricket::MediaSessionOptions options;
EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
EXPECT_FALSE(options.has_audio());
EXPECT_TRUE(options.has_video());
EXPECT_TRUE(options.bundle_enabled);
}
// Test that a correct MediaSessionOptions is created for an offer if
// UseRtpMux is set to false.
TEST(CreateSessionOptionsTest,
GetMediaSessionOptionsForOfferWithBundleDisabled) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 1;
rtc_options.use_rtp_mux = false;
cricket::MediaSessionOptions options;
EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
EXPECT_TRUE(options.has_audio());
EXPECT_TRUE(options.has_video());
EXPECT_FALSE(options.bundle_enabled);
}
// Test that a correct MediaSessionOptions is created to restart ice if
// IceRestart is set. It also tests that subsequent MediaSessionOptions don't
// have |audio_transport_options.ice_restart| etc. set.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithIceRestart) {
RTCOfferAnswerOptions rtc_options;
rtc_options.ice_restart = true;
cricket::MediaSessionOptions options;
options.transport_options["audio"] = cricket::TransportOptions();
options.transport_options["video"] = cricket::TransportOptions();
EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
EXPECT_TRUE(options.transport_options["audio"].ice_restart);
EXPECT_TRUE(options.transport_options["video"].ice_restart);
rtc_options = RTCOfferAnswerOptions();
EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
EXPECT_FALSE(options.transport_options["audio"].ice_restart);
EXPECT_FALSE(options.transport_options["video"].ice_restart);
}
// Test that the MediaConstraints in an answer don't affect if audio and video
// is offered in an offer but that if kOfferToReceiveAudio or
// kOfferToReceiveVideo constraints are true in an offer, the media type will be
// included in subsequent answers.
TEST(CreateSessionOptionsTest, MediaConstraintsInAnswer) {
FakeConstraints answer_c;
answer_c.SetMandatoryReceiveAudio(true);
answer_c.SetMandatoryReceiveVideo(true);
cricket::MediaSessionOptions answer_options;
EXPECT_TRUE(ParseConstraintsForAnswer(&answer_c, &answer_options));
EXPECT_TRUE(answer_options.has_audio());
EXPECT_TRUE(answer_options.has_video());
RTCOfferAnswerOptions rtc_offer_options;
cricket::MediaSessionOptions offer_options;
EXPECT_TRUE(
ExtractMediaSessionOptions(rtc_offer_options, false, &offer_options));
EXPECT_TRUE(offer_options.has_audio());
EXPECT_TRUE(offer_options.has_video());
RTCOfferAnswerOptions updated_rtc_offer_options;
updated_rtc_offer_options.offer_to_receive_audio = 1;
updated_rtc_offer_options.offer_to_receive_video = 1;
cricket::MediaSessionOptions updated_offer_options;
EXPECT_TRUE(ExtractMediaSessionOptions(updated_rtc_offer_options, false,
&updated_offer_options));
EXPECT_TRUE(updated_offer_options.has_audio());
EXPECT_TRUE(updated_offer_options.has_video());
// Since an offer has been created with both audio and video, subsequent
// offers and answers should contain both audio and video.
// Answers will only contain the media types that exist in the offer
// regardless of the value of |updated_answer_options.has_audio| and
// |updated_answer_options.has_video|.
FakeConstraints updated_answer_c;
answer_c.SetMandatoryReceiveAudio(false);
answer_c.SetMandatoryReceiveVideo(false);
cricket::MediaSessionOptions updated_answer_options;
EXPECT_TRUE(
ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options));
EXPECT_TRUE(updated_answer_options.has_audio());
EXPECT_TRUE(updated_answer_options.has_video());
}