| /* |
| * 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"}; |
| |
| 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 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::MediaConstraintsInterface; |
| 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, |
| 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 = observer->release_desc(); |
| return observer->result(); |
| } |
| |
| bool DoCreateOffer(SessionDescriptionInterface** desc, |
| MediaConstraintsInterface* constraints) { |
| return DoCreateOfferAnswer(desc, true, constraints); |
| } |
| |
| bool DoCreateAnswer(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); |
| } |
| 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(), 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() { |
| scoped_ptr<SessionDescriptionInterface> answer; |
| ASSERT_TRUE(DoCreateAnswer(answer.use(), 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() { |
| scoped_ptr<SessionDescriptionInterface> answer; |
| ASSERT_TRUE(DoCreateAnswer(answer.use(), 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() { |
| rtc::scoped_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(offer.use(), 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 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(), 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.accept(), 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_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, nullptr)); |
| EXPECT_TRUE(DoSetRemoteDescription(offer)); |
| |
| // SetLocalDescription takes ownership of answer. |
| SessionDescriptionInterface* answer = NULL; |
| EXPECT_TRUE(DoCreateAnswer(&answer, nullptr)); |
| 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, nullptr)); |
| 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, nullptr)); |
| |
| // Test CreateAnswer |
| SessionDescriptionInterface* answer = NULL; |
| 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 |
| scoped_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(offer.use(), 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())); |
| scoped_ptr<SessionDescriptionInterface> answer; |
| ASSERT_TRUE(DoCreateAnswer(answer.use(), 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 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 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. |
| 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. |
| 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 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(), nullptr)); |
| rtc::scoped_ptr<SessionDescriptionInterface> answer; |
| EXPECT_TRUE(DoCreateAnswer(answer.use(), 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)); |
| 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(), 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. |
| rtc::scoped_ptr<SessionDescriptionInterface> local_offer; |
| EXPECT_TRUE(DoCreateOffer(local_offer.accept(), 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())); |
| 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 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()); |
| 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(), nullptr)); |
| |
| 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(), nullptr)); |
| |
| 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(), nullptr)); |
| |
| 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(), nullptr)); |
| |
| 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. |
| // 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(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. |
| 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. |
| 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 OfferOptions are used. |
| TEST(CreateSessionOptionsTest, GetDefaultMediaSessionOptionsForOffer) { |
| RTCOfferAnswerOptions rtc_options; |
| |
| cricket::MediaSessionOptions options; |
| EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &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.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(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_options; |
| |
| cricket::MediaSessionOptions offer_options; |
| EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_offer_options, &offer_options)); |
| EXPECT_TRUE(offer_options.has_audio()); |
| EXPECT_FALSE(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(ConvertRtcOptionsForOffer(updated_rtc_offer_options, |
| &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()); |
| } |
