Move talk/app/webrtc to webrtc/api

talk/app/webrtc/test/peerconnectiontestwrapper.cc

-/*
- * libjingle
- * Copyright 2013 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 <utility>
-
-#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
-#include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
-#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
-#include "talk/app/webrtc/test/peerconnectiontestwrapper.h"
-#include "talk/app/webrtc/videosourceinterface.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/p2p/client/fakeportallocator.h"
-
-static const char kStreamLabelBase[] = "stream_label";
-static const char kVideoTrackLabelBase[] = "video_track";
-static const char kAudioTrackLabelBase[] = "audio_track";
-static const int kMaxWait = 10000;
-static const int kTestAudioFrameCount = 3;
-static const int kTestVideoFrameCount = 3;
-
-using webrtc::FakeConstraints;
-using webrtc::FakeVideoTrackRenderer;
-using webrtc::IceCandidateInterface;
-using webrtc::MediaConstraintsInterface;
-using webrtc::MediaStreamInterface;
-using webrtc::MockSetSessionDescriptionObserver;
-using webrtc::PeerConnectionInterface;
-using webrtc::SessionDescriptionInterface;
-using webrtc::VideoTrackInterface;
-
-void PeerConnectionTestWrapper::Connect(PeerConnectionTestWrapper* caller,
-                                        PeerConnectionTestWrapper* callee) {
-  caller->SignalOnIceCandidateReady.connect(
-      callee, &PeerConnectionTestWrapper::AddIceCandidate);
-  callee->SignalOnIceCandidateReady.connect(
-      caller, &PeerConnectionTestWrapper::AddIceCandidate);
-
-  caller->SignalOnSdpReady.connect(
-      callee, &PeerConnectionTestWrapper::ReceiveOfferSdp);
-  callee->SignalOnSdpReady.connect(
-      caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
-}
-
-PeerConnectionTestWrapper::PeerConnectionTestWrapper(const std::string& name)
-    : name_(name) {}
-
-PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
-
-bool PeerConnectionTestWrapper::CreatePc(
-  const MediaConstraintsInterface* constraints) {
-  rtc::scoped_ptr<cricket::PortAllocator> port_allocator(
-      new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
-
-  fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
-  if (fake_audio_capture_module_ == NULL) {
-    return false;
-  }
-
-  peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
-      rtc::Thread::Current(), rtc::Thread::Current(),
-      fake_audio_capture_module_, NULL, NULL);
-  if (!peer_connection_factory_) {
-    return false;
-  }
-
-  // CreatePeerConnection with RTCConfiguration.
-  webrtc::PeerConnectionInterface::RTCConfiguration config;
-  webrtc::PeerConnectionInterface::IceServer ice_server;
-  ice_server.uri = "stun:stun.l.google.com:19302";
-  config.servers.push_back(ice_server);
-  rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store(
-      rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
-      new FakeDtlsIdentityStore() : nullptr);
-  peer_connection_ = peer_connection_factory_->CreatePeerConnection(
-      config, constraints, std::move(port_allocator),
-      std::move(dtls_identity_store), this);
-
-  return peer_connection_.get() != NULL;
-}
-
-rtc::scoped_refptr<webrtc::DataChannelInterface>
-PeerConnectionTestWrapper::CreateDataChannel(
-    const std::string& label,
-    const webrtc::DataChannelInit& init) {
-  return peer_connection_->CreateDataChannel(label, &init);
-}
-
-void PeerConnectionTestWrapper::OnAddStream(MediaStreamInterface* stream) {
-  LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
-               << ": OnAddStream";
-  // TODO(ronghuawu): support multiple streams.
-  if (stream->GetVideoTracks().size() > 0) {
-    renderer_.reset(new FakeVideoTrackRenderer(stream->GetVideoTracks()[0]));
-  }
-}
-
-void PeerConnectionTestWrapper::OnIceCandidate(
-    const IceCandidateInterface* candidate) {
-  std::string sdp;
-  EXPECT_TRUE(candidate->ToString(&sdp));
-  // Give the user a chance to modify sdp for testing.
-  SignalOnIceCandidateCreated(&sdp);
-  SignalOnIceCandidateReady(candidate->sdp_mid(), candidate->sdp_mline_index(),
-                            sdp);
-}
-
-void PeerConnectionTestWrapper::OnDataChannel(
-    webrtc::DataChannelInterface* data_channel) {
-  SignalOnDataChannel(data_channel);
-}
-
-void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
-  // This callback should take the ownership of |desc|.
-  rtc::scoped_ptr<SessionDescriptionInterface> owned_desc(desc);
-  std::string sdp;
-  EXPECT_TRUE(desc->ToString(&sdp));
-
-  LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
-               << ": " << desc->type() << " sdp created: " << sdp;
-
-  // Give the user a chance to modify sdp for testing.
-  SignalOnSdpCreated(&sdp);
-
-  SetLocalDescription(desc->type(), sdp);
-
-  SignalOnSdpReady(sdp);
-}
-
-void PeerConnectionTestWrapper::CreateOffer(
-    const MediaConstraintsInterface* constraints) {
-  LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
-               << ": CreateOffer.";
-  peer_connection_->CreateOffer(this, constraints);
-}
-
-void PeerConnectionTestWrapper::CreateAnswer(
-    const MediaConstraintsInterface* constraints) {
-  LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
-               << ": CreateAnswer.";
-  peer_connection_->CreateAnswer(this, constraints);
-}
-
-void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) {
-  SetRemoteDescription(SessionDescriptionInterface::kOffer, sdp);
-  CreateAnswer(NULL);
-}
-
-void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) {
-  SetRemoteDescription(SessionDescriptionInterface::kAnswer, sdp);
-}
-
-void PeerConnectionTestWrapper::SetLocalDescription(const std::string& type,
-                                                    const std::string& sdp) {
-  LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
-               << ": SetLocalDescription " << type << " " << sdp;
-
-  rtc::scoped_refptr<MockSetSessionDescriptionObserver>
-      observer(new rtc::RefCountedObject<
-                   MockSetSessionDescriptionObserver>());
-  peer_connection_->SetLocalDescription(
-      observer, webrtc::CreateSessionDescription(type, sdp, NULL));
-}
-
-void PeerConnectionTestWrapper::SetRemoteDescription(const std::string& type,
-                                                     const std::string& sdp) {
-  LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
-               << ": SetRemoteDescription " << type << " " << sdp;
-
-  rtc::scoped_refptr<MockSetSessionDescriptionObserver>
-      observer(new rtc::RefCountedObject<
-                   MockSetSessionDescriptionObserver>());
-  peer_connection_->SetRemoteDescription(
-      observer, webrtc::CreateSessionDescription(type, sdp, NULL));
-}
-
-void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
-                                                int sdp_mline_index,
-                                                const std::string& candidate) {
-  rtc::scoped_ptr<webrtc::IceCandidateInterface> owned_candidate(
-      webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL));
-  EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get()));
-}
-
-void PeerConnectionTestWrapper::WaitForCallEstablished() {
-  WaitForConnection();
-  WaitForAudio();
-  WaitForVideo();
-}
-
-void PeerConnectionTestWrapper::WaitForConnection() {
-  EXPECT_TRUE_WAIT(CheckForConnection(), kMaxWait);
-  LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
-               << ": Connected.";
-}
-
-bool PeerConnectionTestWrapper::CheckForConnection() {
-  return (peer_connection_->ice_connection_state() ==
-          PeerConnectionInterface::kIceConnectionConnected) ||
-         (peer_connection_->ice_connection_state() ==
-          PeerConnectionInterface::kIceConnectionCompleted);
-}
-
-void PeerConnectionTestWrapper::WaitForAudio() {
-  EXPECT_TRUE_WAIT(CheckForAudio(), kMaxWait);
-  LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
-               << ": Got enough audio frames.";
-}
-
-bool PeerConnectionTestWrapper::CheckForAudio() {
-  return (fake_audio_capture_module_->frames_received() >=
-          kTestAudioFrameCount);
-}
-
-void PeerConnectionTestWrapper::WaitForVideo() {
-  EXPECT_TRUE_WAIT(CheckForVideo(), kMaxWait);
-  LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
-               << ": Got enough video frames.";
-}
-
-bool PeerConnectionTestWrapper::CheckForVideo() {
-  if (!renderer_) {
-    return false;
-  }
-  return (renderer_->num_rendered_frames() >= kTestVideoFrameCount);
-}
-
-void PeerConnectionTestWrapper::GetAndAddUserMedia(
-    bool audio, const webrtc::FakeConstraints& audio_constraints,
-    bool video, const webrtc::FakeConstraints& video_constraints) {
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
-      GetUserMedia(audio, audio_constraints, video, video_constraints);
-  EXPECT_TRUE(peer_connection_->AddStream(stream));
-}
-
-rtc::scoped_refptr<webrtc::MediaStreamInterface>
-    PeerConnectionTestWrapper::GetUserMedia(
-        bool audio, const webrtc::FakeConstraints& audio_constraints,
-        bool video, const webrtc::FakeConstraints& video_constraints) {
-  std::string label = kStreamLabelBase +
-      rtc::ToString<int>(
-          static_cast<int>(peer_connection_->local_streams()->count()));
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
-      peer_connection_factory_->CreateLocalMediaStream(label);
-
-  if (audio) {
-    FakeConstraints constraints = audio_constraints;
-    // Disable highpass filter so that we can get all the test audio frames.
-    constraints.AddMandatory(
-        MediaConstraintsInterface::kHighpassFilter, false);
-    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
-        peer_connection_factory_->CreateAudioSource(&constraints);
-    rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-        peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
-                                                   source));
-    stream->AddTrack(audio_track);
-  }
-
-  if (video) {
-    // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
-    FakeConstraints constraints = video_constraints;
-    constraints.SetMandatoryMaxFrameRate(10);
-
-    rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
-        peer_connection_factory_->CreateVideoSource(
-            new webrtc::FakePeriodicVideoCapturer(), &constraints);
-    std::string videotrack_label = label + kVideoTrackLabelBase;
-    rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
-        peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
-
-    stream->AddTrack(video_track);
-  }
-  return stream;
-}