blob: 5e88658a4e25ce3c464853dfbba53a1335829c63 [file] [log] [blame]
/*
* libjingle
* Copyright 2012 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include <string>
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/fakeportallocatorfactory.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/mediastream.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/peerconnection.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/rtpreceiverinterface.h"
#include "talk/app/webrtc/rtpsenderinterface.h"
#include "talk/app/webrtc/streamcollection.h"
#include "talk/app/webrtc/test/fakeconstraints.h"
#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
#include "talk/app/webrtc/test/testsdpstrings.h"
#include "talk/app/webrtc/videosource.h"
#include "talk/app/webrtc/videotrack.h"
#include "talk/media/base/fakevideocapturer.h"
#include "talk/media/sctp/sctpdataengine.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/thread.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"};
// 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=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=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=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=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=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\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=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\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=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=sendonly\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\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=rtpmap:103 ISAC/16000\r\n";
static const char kSdpStringVideo[] =
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\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 rtc::scoped_ptr;
using rtc::scoped_refptr;
using webrtc::AudioSourceInterface;
using webrtc::AudioTrack;
using webrtc::AudioTrackInterface;
using webrtc::DataBuffer;
using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
using webrtc::FakePortAllocatorFactory;
using webrtc::IceCandidateInterface;
using webrtc::MediaStream;
using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
using webrtc::MockCreateSessionDescriptionObserver;
using webrtc::MockDataChannelObserver;
using webrtc::MockSetSessionDescriptionObserver;
using webrtc::MockStatsObserver;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
using webrtc::PortAllocatorFactoryInterface;
using webrtc::RtpReceiverInterface;
using webrtc::RtpSenderInterface;
using webrtc::SdpParseError;
using webrtc::SessionDescriptionInterface;
using webrtc::StreamCollection;
using webrtc::StreamCollectionInterface;
using webrtc::VideoSourceInterface;
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;
}
// 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) {
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]));
// Add a local audio track.
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(kAudioTracks[i], nullptr));
stream->AddTrack(audio_track);
// Add a local video track.
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
webrtc::VideoTrack::Create(kVideoTracks[i], nullptr));
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;
}
class MockPeerConnectionObserver : public PeerConnectionObserver {
public:
MockPeerConnectionObserver() : remote_streams_(StreamCollection::Create()) {}
~MockPeerConnectionObserver() {
}
void SetPeerConnectionInterface(PeerConnectionInterface* pc) {
pc_ = pc;
if (pc) {
state_ = pc_->signaling_state();
}
}
virtual void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) {
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_; }
virtual void OnAddStream(MediaStreamInterface* stream) {
last_added_stream_ = stream;
remote_streams_->AddStream(stream);
}
virtual void OnRemoveStream(MediaStreamInterface* stream) {
last_removed_stream_ = stream;
remote_streams_->RemoveStream(stream);
}
virtual void OnRenegotiationNeeded() {
renegotiation_needed_ = true;
}
virtual void OnDataChannel(DataChannelInterface* data_channel) {
last_datachannel_ = data_channel;
}
virtual void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {
EXPECT_EQ(pc_->ice_connection_state(), new_state);
}
virtual void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {
EXPECT_EQ(pc_->ice_gathering_state(), new_state);
}
virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
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);
}
// TODO(bemasc): Remove this once callers transition to OnSignalingChange.
virtual void OnIceComplete() {
ice_complete_ = true;
// OnIceGatheringChange(IceGatheringCompleted) and OnIceComplete() should
// be called approximately simultaneously. For ease of testing, this
// check additionally requires that they be called in the above order.
EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete,
pc_->ice_gathering_state());
}
// 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 "";
}
scoped_refptr<PeerConnectionInterface> pc_;
PeerConnectionInterface::SignalingState state_;
scoped_ptr<IceCandidateInterface> last_candidate_;
scoped_refptr<DataChannelInterface> last_datachannel_;
rtc::scoped_refptr<StreamCollection> remote_streams_;
bool renegotiation_needed_ = false;
bool ice_complete_ = false;
private:
scoped_refptr<MediaStreamInterface> last_added_stream_;
scoped_refptr<MediaStreamInterface> last_removed_stream_;
};
} // namespace
class PeerConnectionInterfaceTest : public testing::Test {
protected:
virtual void SetUp() {
pc_factory_ = webrtc::CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), NULL, NULL,
NULL);
ASSERT_TRUE(pc_factory_.get() != NULL);
}
void CreatePeerConnection() {
CreatePeerConnection("", "", NULL);
}
void CreatePeerConnection(webrtc::MediaConstraintsInterface* constraints) {
CreatePeerConnection("", "", constraints);
}
void CreatePeerConnection(const std::string& uri,
const std::string& password,
webrtc::MediaConstraintsInterface* constraints) {
PeerConnectionInterface::IceServer server;
PeerConnectionInterface::IceServers servers;
if (!uri.empty()) {
server.uri = uri;
server.password = password;
servers.push_back(server);
}
port_allocator_factory_ = FakePortAllocatorFactory::Create();
// 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);
}
scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store;
bool dtls;
if (FindConstraint(constraints,
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
&dtls,
nullptr) && dtls) {
dtls_identity_store.reset(new FakeDtlsIdentityStore());
}
pc_ = pc_factory_->CreatePeerConnection(servers, constraints,
port_allocator_factory_.get(),
dtls_identity_store.Pass(),
&observer_);
ASSERT_TRUE(pc_.get() != NULL);
observer_.SetPeerConnectionInterface(pc_.get());
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePeerConnectionExpectFail(const std::string& uri) {
PeerConnectionInterface::IceServer server;
PeerConnectionInterface::IceServers servers;
server.uri = uri;
servers.push_back(server);
scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store;
port_allocator_factory_ = FakePortAllocatorFactory::Create();
scoped_refptr<PeerConnectionInterface> pc;
pc = pc_factory_->CreatePeerConnection(
servers, nullptr, port_allocator_factory_.get(),
dtls_identity_store.Pass(), &observer_);
ASSERT_EQ(nullptr, pc);
}
void CreatePeerConnectionWithDifferentConfigurations() {
CreatePeerConnection(kStunAddressOnly, "", NULL);
EXPECT_EQ(1u, port_allocator_factory_->stun_configs().size());
EXPECT_EQ(0u, port_allocator_factory_->turn_configs().size());
EXPECT_EQ("address",
port_allocator_factory_->stun_configs()[0].server.hostname());
EXPECT_EQ(kDefaultStunPort,
port_allocator_factory_->stun_configs()[0].server.port());
CreatePeerConnectionExpectFail(kStunInvalidPort);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore1);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore2);
CreatePeerConnection(kTurnIceServerUri, kTurnPassword, NULL);
EXPECT_EQ(0u, port_allocator_factory_->stun_configs().size());
EXPECT_EQ(1u, port_allocator_factory_->turn_configs().size());
EXPECT_EQ(kTurnUsername,
port_allocator_factory_->turn_configs()[0].username);
EXPECT_EQ(kTurnPassword,
port_allocator_factory_->turn_configs()[0].password);
EXPECT_EQ(kTurnHostname,
port_allocator_factory_->turn_configs()[0].server.hostname());
}
void ReleasePeerConnection() {
pc_ = NULL;
observer_.SetPeerConnectionInterface(NULL);
}
void AddVideoStream(const std::string& label) {
// Create a local stream.
scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
scoped_refptr<VideoSourceInterface> video_source(
pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer(), NULL));
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.
scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
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.
scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(stream_label));
scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack(
audio_track_label, static_cast<AudioSourceInterface*>(NULL)));
stream->AddTrack(audio_track.get());
scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(video_track_label, NULL));
stream->AddTrack(video_track.get());
EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
bool DoCreateOfferAnswer(SessionDescriptionInterface** desc, bool offer) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockCreateSessionDescriptionObserver>());
if (offer) {
pc_->CreateOffer(observer, NULL);
} else {
pc_->CreateAnswer(observer, NULL);
}
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
*desc = observer->release_desc();
return observer->result();
}
bool DoCreateOffer(SessionDescriptionInterface** desc) {
return DoCreateOfferAnswer(desc, true);
}
bool DoCreateAnswer(SessionDescriptionInterface** desc) {
return DoCreateOfferAnswer(desc, false);
}
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);
}
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() {
rtc::scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.use()));
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() {
scoped_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(answer.use()));
// 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() {
scoped_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(answer.use()));
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() {
rtc::scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.use()));
// 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 in |desc| and the MediaStream is stored in
// |reference_collection_|
void CreateSessionDescriptionAndReference(
size_t number_of_audio_tracks,
size_t number_of_video_tracks,
SessionDescriptionInterface** desc) {
ASSERT_TRUE(desc != nullptr);
ASSERT_LE(number_of_audio_tracks, 2u);
ASSERT_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);
}
*desc = 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, nullptr));
ASSERT_TRUE(stream->AddTrack(video_track));
}
scoped_refptr<FakePortAllocatorFactory> port_allocator_factory_;
scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
scoped_refptr<PeerConnectionInterface> pc_;
MockPeerConnectionObserver observer_;
rtc::scoped_refptr<StreamCollection> reference_collection_;
};
TEST_F(PeerConnectionInterfaceTest,
CreatePeerConnectionWithDifferentConfigurations) {
CreatePeerConnectionWithDifferentConfigurations();
}
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.
scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamLabel3));
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");
scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.accept()));
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.accept()));
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_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.
SessionDescriptionInterface* offer = NULL;
AddVideoStream(kStreamLabel1);
EXPECT_TRUE(DoCreateOffer(&offer));
EXPECT_TRUE(DoSetRemoteDescription(offer));
// SetLocalDescription takes ownership of answer.
SessionDescriptionInterface* answer = NULL;
EXPECT_TRUE(DoCreateAnswer(&answer));
EXPECT_TRUE(DoSetLocalDescription(answer));
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.
SessionDescriptionInterface* offer = NULL;
EXPECT_TRUE(DoCreateOffer(&offer));
EXPECT_TRUE(offer != NULL);
delete offer;
offer = NULL;
// Create a local stream with audio&video tracks having same label.
AddAudioVideoStream(kStreamLabel1, "track_label", "track_label");
// Test CreateOffer
EXPECT_FALSE(DoCreateOffer(&offer));
// Test CreateAnswer
SessionDescriptionInterface* answer = NULL;
EXPECT_FALSE(DoCreateAnswer(&answer));
}
// 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
scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.use()));
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()));
scoped_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(answer.use()));
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 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());
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());
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();
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);
scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
rtc::scoped_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
rtc::scoped_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);
scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
rtc::scoped_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
rtc::scoped_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::Buffer 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);
scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_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";
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;
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";
scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(channel, nullptr);
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;
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;
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;
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";
scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(channel, nullptr);
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);
scoped_refptr<DataChannelInterface> dc1 =
pc_->CreateDataChannel("test1", NULL);
EXPECT_TRUE(observer_.renegotiation_needed_);
observer_.renegotiation_needed_ = false;
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);
scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
rtc::scoped_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
rtc::scoped_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);
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 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());
scoped_refptr<MediaStreamInterface> remote_stream =
pc_->remote_streams()->at(0);
EXPECT_EQ(MediaStreamTrackInterface::kEnded,
remote_stream->GetVideoTracks()[0]->state());
EXPECT_EQ(MediaStreamTrackInterface::kEnded,
remote_stream->GetAudioTracks()[0]->state());
}
// 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());
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);
rtc::scoped_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(offer.use()));
rtc::scoped_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(answer.use()));
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));
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));
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);
rtc::scoped_ptr<SessionDescriptionInterface> desc_ms1;
CreateSessionDescriptionAndReference(1, 1, desc_ms1.accept());
EXPECT_TRUE(DoSetRemoteDescription(desc_ms1.release()));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
// Add extra audio and video tracks to the same MediaStream.
rtc::scoped_ptr<SessionDescriptionInterface> desc_ms1_two_tracks;
CreateSessionDescriptionAndReference(2, 2, desc_ms1_two_tracks.accept());
EXPECT_TRUE(DoSetRemoteDescription(desc_ms1_two_tracks.release()));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
// Remove the extra audio and video tracks.
rtc::scoped_ptr<SessionDescriptionInterface> desc_ms2;
CreateSessionDescriptionAndReference(1, 1, desc_ms2.accept());
EXPECT_TRUE(DoSetRemoteDescription(desc_ms2.release()));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
}
// 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());
rtc::scoped_ptr<SessionDescriptionInterface> local_answer;
EXPECT_TRUE(DoCreateAnswer(local_answer.accept()));
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.
rtc::scoped_ptr<SessionDescriptionInterface> local_offer;
EXPECT_TRUE(DoCreateOffer(local_offer.accept()));
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()));
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_video->state());
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_audio->state());
}
// 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]);
rtc::scoped_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 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());
ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("defaultv0", remote_stream->GetVideoTracks()[0]->id());
}
// 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 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));
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 just to ensure we have an identity before we manually
// call SetLocalDescription.
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
ASSERT_TRUE(DoCreateOffer(throwaway.accept()));
rtc::scoped_ptr<SessionDescriptionInterface> desc_1;
CreateSessionDescriptionAndReference(2, 2, desc_1.accept());
pc_->AddStream(reference_collection_->at(0));
EXPECT_TRUE(DoSetLocalDescription(desc_1.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.
rtc::scoped_ptr<SessionDescriptionInterface> desc_2;
CreateSessionDescriptionAndReference(1, 1, desc_2.accept());
EXPECT_TRUE(DoSetLocalDescription(desc_2.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);
// Create an offer just to ensure we have an identity before we manually
// call SetLocalDescription.
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
ASSERT_TRUE(DoCreateOffer(throwaway.accept()));
rtc::scoped_ptr<SessionDescriptionInterface> desc_1;
CreateSessionDescriptionAndReference(2, 2, desc_1.accept());
EXPECT_TRUE(DoSetLocalDescription(desc_1.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(0u, senders.size());
pc_->AddStream(reference_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);
// Create an offer just to ensure we have an identity before we manually
// call SetLocalDescription.
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
ASSERT_TRUE(DoCreateOffer(throwaway.accept()));
rtc::scoped_ptr<SessionDescriptionInterface> desc;
CreateSessionDescriptionAndReference(1, 1, desc.accept());
std::string sdp;
desc->ToString(&sdp);
pc_->AddStream(reference_collection_->at(0));
EXPECT_TRUE(DoSetLocalDescription(desc.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.
std::string ssrc_org = "a=ssrc:1";
std::string ssrc_to = "a=ssrc:97";
rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(), ssrc_to.c_str(),
ssrc_to.length(), &sdp);
ssrc_org = "a=ssrc:2";
ssrc_to = "a=ssrc:98";
rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(), ssrc_to.c_str(),
ssrc_to.length(), &sdp);
rtc::scoped_ptr<SessionDescriptionInterface> updated_desc(
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer, sdp,
nullptr));
EXPECT_TRUE(DoSetLocalDescription(updated_desc.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);
// Create an offer just to ensure we have an identity before we manually
// call SetLocalDescription.
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
ASSERT_TRUE(DoCreateOffer(throwaway.accept()));
rtc::scoped_ptr<SessionDescriptionInterface> desc;
CreateSessionDescriptionAndReference(1, 1, desc.accept());
std::string sdp;
desc->ToString(&sdp);
pc_->AddStream(reference_collection_->at(0));
EXPECT_TRUE(DoSetLocalDescription(desc.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[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(reference_collection_->at(0)->GetVideoTracks()[0]);
stream_1->AddTrack(reference_collection_->at(0)->GetAudioTracks()[0]);
pc_->AddStream(stream_1);
// Replace msid in the original SDP.
rtc::replace_substrs(kStreams[0], strlen(kStreams[0]), kStreams[1],
strlen(kStreams[1]), &sdp);
rtc::scoped_ptr<SessionDescriptionInterface> updated_desc(
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer, sdp,
nullptr));
EXPECT_TRUE(DoSetLocalDescription(updated_desc.release()));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
}
// The following tests verify that session options are created correctly.
TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidAudioOption) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = RTCOfferAnswerOptions::kUndefined - 1;
cricket::MediaSessionOptions options;
EXPECT_FALSE(ConvertRtcOptionsForOffer(rtc_options, &options));
rtc_options.offer_to_receive_audio =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(ConvertRtcOptionsForOffer(rtc_options, &options));
}
TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidVideoOption) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_video = RTCOfferAnswerOptions::kUndefined - 1;
cricket::MediaSessionOptions options;
EXPECT_FALSE(ConvertRtcOptionsForOffer(rtc_options, &options));
rtc_options.offer_to_receive_video =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(ConvertRtcOptionsForOffer(rtc_options, &options));
}
// Test that a MediaSessionOptions is created for an offer if
// OfferToReceiveAudio and OfferToReceiveVideo options are set but no
// MediaStreams are sent.
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(ConvertRtcOptionsForOffer(rtc_options, &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 but no MediaStreams are sent.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudio) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
cricket::MediaSessionOptions options;
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &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 OfferOptons is used or MediaStreams are sent.
TEST(CreateSessionOptionsTest, GetDefaultMediaSessionOptionsForOffer) {
RTCOfferAnswerOptions rtc_options;
cricket::MediaSessionOptions options;
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
EXPECT_FALSE(options.has_audio());
EXPECT_FALSE(options.has_video());
EXPECT_FALSE(options.bundle_enabled);
EXPECT_TRUE(options.vad_enabled);
EXPECT_FALSE(options.transport_options.ice_restart);
}
// Test that a correct MediaSessionOptions is created for an offer if
// OfferToReceiveVideo is set but no MediaStreams are sent.
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(ConvertRtcOptionsForOffer(rtc_options, &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(ConvertRtcOptionsForOffer(rtc_options, &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 |transport_options.ice_restart| set.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithIceRestart) {
RTCOfferAnswerOptions rtc_options;
rtc_options.ice_restart = true;
cricket::MediaSessionOptions options;
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
EXPECT_TRUE(options.transport_options.ice_restart);
rtc_options = RTCOfferAnswerOptions();
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
EXPECT_FALSE(options.transport_options.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_optoins;
cricket::MediaSessionOptions offer_options;
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_offer_optoins, &offer_options));
EXPECT_FALSE(offer_options.has_audio());
EXPECT_FALSE(offer_options.has_video());
RTCOfferAnswerOptions updated_rtc_offer_optoins;
updated_rtc_offer_optoins.offer_to_receive_audio = 1;
updated_rtc_offer_optoins.offer_to_receive_video = 1;
cricket::MediaSessionOptions updated_offer_options;
EXPECT_TRUE(ConvertRtcOptionsForOffer(updated_rtc_offer_optoins,
&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());
RTCOfferAnswerOptions default_rtc_options;
EXPECT_TRUE(
ConvertRtcOptionsForOffer(default_rtc_options, &updated_offer_options));
// By default, |has_audio| or |has_video| are false if there is no media
// track.
EXPECT_FALSE(updated_offer_options.has_audio());
EXPECT_FALSE(updated_offer_options.has_video());
}