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      1 /*
      2  * libjingle
      3  * Copyright 2013 Google Inc.
      4  *
      5  * Redistribution and use in source and binary forms, with or without
      6  * modification, are permitted provided that the following conditions are met:
      7  *
      8  *  1. Redistributions of source code must retain the above copyright notice,
      9  *     this list of conditions and the following disclaimer.
     10  *  2. Redistributions in binary form must reproduce the above copyright notice,
     11  *     this list of conditions and the following disclaimer in the documentation
     12  *     and/or other materials provided with the distribution.
     13  *  3. The name of the author may not be used to endorse or promote products
     14  *     derived from this software without specific prior written permission.
     15  *
     16  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
     17  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
     18  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
     19  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
     20  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
     21  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
     22  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
     23  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
     24  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
     25  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
     26  */
     27 
     28 #include <utility>
     29 
     30 #include "talk/app/webrtc/test/fakedtlsidentitystore.h"
     31 #include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
     32 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
     33 #include "talk/app/webrtc/test/peerconnectiontestwrapper.h"
     34 #include "talk/app/webrtc/videosourceinterface.h"
     35 #include "webrtc/base/gunit.h"
     36 #include "webrtc/p2p/client/fakeportallocator.h"
     37 
     38 static const char kStreamLabelBase[] = "stream_label";
     39 static const char kVideoTrackLabelBase[] = "video_track";
     40 static const char kAudioTrackLabelBase[] = "audio_track";
     41 static const int kMaxWait = 10000;
     42 static const int kTestAudioFrameCount = 3;
     43 static const int kTestVideoFrameCount = 3;
     44 
     45 using webrtc::FakeConstraints;
     46 using webrtc::FakeVideoTrackRenderer;
     47 using webrtc::IceCandidateInterface;
     48 using webrtc::MediaConstraintsInterface;
     49 using webrtc::MediaStreamInterface;
     50 using webrtc::MockSetSessionDescriptionObserver;
     51 using webrtc::PeerConnectionInterface;
     52 using webrtc::SessionDescriptionInterface;
     53 using webrtc::VideoTrackInterface;
     54 
     55 void PeerConnectionTestWrapper::Connect(PeerConnectionTestWrapper* caller,
     56                                         PeerConnectionTestWrapper* callee) {
     57   caller->SignalOnIceCandidateReady.connect(
     58       callee, &PeerConnectionTestWrapper::AddIceCandidate);
     59   callee->SignalOnIceCandidateReady.connect(
     60       caller, &PeerConnectionTestWrapper::AddIceCandidate);
     61 
     62   caller->SignalOnSdpReady.connect(
     63       callee, &PeerConnectionTestWrapper::ReceiveOfferSdp);
     64   callee->SignalOnSdpReady.connect(
     65       caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
     66 }
     67 
     68 PeerConnectionTestWrapper::PeerConnectionTestWrapper(const std::string& name)
     69     : name_(name) {}
     70 
     71 PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
     72 
     73 bool PeerConnectionTestWrapper::CreatePc(
     74   const MediaConstraintsInterface* constraints) {
     75   rtc::scoped_ptr<cricket::PortAllocator> port_allocator(
     76       new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
     77 
     78   fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
     79   if (fake_audio_capture_module_ == NULL) {
     80     return false;
     81   }
     82 
     83   peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
     84       rtc::Thread::Current(), rtc::Thread::Current(),
     85       fake_audio_capture_module_, NULL, NULL);
     86   if (!peer_connection_factory_) {
     87     return false;
     88   }
     89 
     90   // CreatePeerConnection with RTCConfiguration.
     91   webrtc::PeerConnectionInterface::RTCConfiguration config;
     92   webrtc::PeerConnectionInterface::IceServer ice_server;
     93   ice_server.uri = "stun:stun.l.google.com:19302";
     94   config.servers.push_back(ice_server);
     95   rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store(
     96       rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
     97       new FakeDtlsIdentityStore() : nullptr);
     98   peer_connection_ = peer_connection_factory_->CreatePeerConnection(
     99       config, constraints, std::move(port_allocator),
    100       std::move(dtls_identity_store), this);
    101 
    102   return peer_connection_.get() != NULL;
    103 }
    104 
    105 rtc::scoped_refptr<webrtc::DataChannelInterface>
    106 PeerConnectionTestWrapper::CreateDataChannel(
    107     const std::string& label,
    108     const webrtc::DataChannelInit& init) {
    109   return peer_connection_->CreateDataChannel(label, &init);
    110 }
    111 
    112 void PeerConnectionTestWrapper::OnAddStream(MediaStreamInterface* stream) {
    113   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
    114                << ": OnAddStream";
    115   // TODO(ronghuawu): support multiple streams.
    116   if (stream->GetVideoTracks().size() > 0) {
    117     renderer_.reset(new FakeVideoTrackRenderer(stream->GetVideoTracks()[0]));
    118   }
    119 }
    120 
    121 void PeerConnectionTestWrapper::OnIceCandidate(
    122     const IceCandidateInterface* candidate) {
    123   std::string sdp;
    124   EXPECT_TRUE(candidate->ToString(&sdp));
    125   // Give the user a chance to modify sdp for testing.
    126   SignalOnIceCandidateCreated(&sdp);
    127   SignalOnIceCandidateReady(candidate->sdp_mid(), candidate->sdp_mline_index(),
    128                             sdp);
    129 }
    130 
    131 void PeerConnectionTestWrapper::OnDataChannel(
    132     webrtc::DataChannelInterface* data_channel) {
    133   SignalOnDataChannel(data_channel);
    134 }
    135 
    136 void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
    137   // This callback should take the ownership of |desc|.
    138   rtc::scoped_ptr<SessionDescriptionInterface> owned_desc(desc);
    139   std::string sdp;
    140   EXPECT_TRUE(desc->ToString(&sdp));
    141 
    142   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
    143                << ": " << desc->type() << " sdp created: " << sdp;
    144 
    145   // Give the user a chance to modify sdp for testing.
    146   SignalOnSdpCreated(&sdp);
    147 
    148   SetLocalDescription(desc->type(), sdp);
    149 
    150   SignalOnSdpReady(sdp);
    151 }
    152 
    153 void PeerConnectionTestWrapper::CreateOffer(
    154     const MediaConstraintsInterface* constraints) {
    155   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
    156                << ": CreateOffer.";
    157   peer_connection_->CreateOffer(this, constraints);
    158 }
    159 
    160 void PeerConnectionTestWrapper::CreateAnswer(
    161     const MediaConstraintsInterface* constraints) {
    162   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
    163                << ": CreateAnswer.";
    164   peer_connection_->CreateAnswer(this, constraints);
    165 }
    166 
    167 void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) {
    168   SetRemoteDescription(SessionDescriptionInterface::kOffer, sdp);
    169   CreateAnswer(NULL);
    170 }
    171 
    172 void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) {
    173   SetRemoteDescription(SessionDescriptionInterface::kAnswer, sdp);
    174 }
    175 
    176 void PeerConnectionTestWrapper::SetLocalDescription(const std::string& type,
    177                                                     const std::string& sdp) {
    178   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
    179                << ": SetLocalDescription " << type << " " << sdp;
    180 
    181   rtc::scoped_refptr<MockSetSessionDescriptionObserver>
    182       observer(new rtc::RefCountedObject<
    183                    MockSetSessionDescriptionObserver>());
    184   peer_connection_->SetLocalDescription(
    185       observer, webrtc::CreateSessionDescription(type, sdp, NULL));
    186 }
    187 
    188 void PeerConnectionTestWrapper::SetRemoteDescription(const std::string& type,
    189                                                      const std::string& sdp) {
    190   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
    191                << ": SetRemoteDescription " << type << " " << sdp;
    192 
    193   rtc::scoped_refptr<MockSetSessionDescriptionObserver>
    194       observer(new rtc::RefCountedObject<
    195                    MockSetSessionDescriptionObserver>());
    196   peer_connection_->SetRemoteDescription(
    197       observer, webrtc::CreateSessionDescription(type, sdp, NULL));
    198 }
    199 
    200 void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
    201                                                 int sdp_mline_index,
    202                                                 const std::string& candidate) {
    203   rtc::scoped_ptr<webrtc::IceCandidateInterface> owned_candidate(
    204       webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL));
    205   EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get()));
    206 }
    207 
    208 void PeerConnectionTestWrapper::WaitForCallEstablished() {
    209   WaitForConnection();
    210   WaitForAudio();
    211   WaitForVideo();
    212 }
    213 
    214 void PeerConnectionTestWrapper::WaitForConnection() {
    215   EXPECT_TRUE_WAIT(CheckForConnection(), kMaxWait);
    216   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
    217                << ": Connected.";
    218 }
    219 
    220 bool PeerConnectionTestWrapper::CheckForConnection() {
    221   return (peer_connection_->ice_connection_state() ==
    222           PeerConnectionInterface::kIceConnectionConnected) ||
    223          (peer_connection_->ice_connection_state() ==
    224           PeerConnectionInterface::kIceConnectionCompleted);
    225 }
    226 
    227 void PeerConnectionTestWrapper::WaitForAudio() {
    228   EXPECT_TRUE_WAIT(CheckForAudio(), kMaxWait);
    229   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
    230                << ": Got enough audio frames.";
    231 }
    232 
    233 bool PeerConnectionTestWrapper::CheckForAudio() {
    234   return (fake_audio_capture_module_->frames_received() >=
    235           kTestAudioFrameCount);
    236 }
    237 
    238 void PeerConnectionTestWrapper::WaitForVideo() {
    239   EXPECT_TRUE_WAIT(CheckForVideo(), kMaxWait);
    240   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
    241                << ": Got enough video frames.";
    242 }
    243 
    244 bool PeerConnectionTestWrapper::CheckForVideo() {
    245   if (!renderer_) {
    246     return false;
    247   }
    248   return (renderer_->num_rendered_frames() >= kTestVideoFrameCount);
    249 }
    250 
    251 void PeerConnectionTestWrapper::GetAndAddUserMedia(
    252     bool audio, const webrtc::FakeConstraints& audio_constraints,
    253     bool video, const webrtc::FakeConstraints& video_constraints) {
    254   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
    255       GetUserMedia(audio, audio_constraints, video, video_constraints);
    256   EXPECT_TRUE(peer_connection_->AddStream(stream));
    257 }
    258 
    259 rtc::scoped_refptr<webrtc::MediaStreamInterface>
    260     PeerConnectionTestWrapper::GetUserMedia(
    261         bool audio, const webrtc::FakeConstraints& audio_constraints,
    262         bool video, const webrtc::FakeConstraints& video_constraints) {
    263   std::string label = kStreamLabelBase +
    264       rtc::ToString<int>(
    265           static_cast<int>(peer_connection_->local_streams()->count()));
    266   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
    267       peer_connection_factory_->CreateLocalMediaStream(label);
    268 
    269   if (audio) {
    270     FakeConstraints constraints = audio_constraints;
    271     // Disable highpass filter so that we can get all the test audio frames.
    272     constraints.AddMandatory(
    273         MediaConstraintsInterface::kHighpassFilter, false);
    274     rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
    275         peer_connection_factory_->CreateAudioSource(&constraints);
    276     rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
    277         peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
    278                                                    source));
    279     stream->AddTrack(audio_track);
    280   }
    281 
    282   if (video) {
    283     // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
    284     FakeConstraints constraints = video_constraints;
    285     constraints.SetMandatoryMaxFrameRate(10);
    286 
    287     rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
    288         peer_connection_factory_->CreateVideoSource(
    289             new webrtc::FakePeriodicVideoCapturer(), &constraints);
    290     std::string videotrack_label = label + kVideoTrackLabelBase;
    291     rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
    292         peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
    293 
    294     stream->AddTrack(video_track);
    295   }
    296   return stream;
    297 }
    298