Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(73)

Side by Side Diff: webrtc/api/test/peerconnectiontestwrapper.cc

Issue 1844803002: Modify PeerConnection for end-to-end QuicDataChannel usage (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Sync to upstream after landing QUIC data channel and QUIC transport CLs Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/api/test/peerconnectiontestwrapper.h ('k') | webrtc/api/webrtcsession.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 36 matching lines...) Expand 10 before | Expand all | Expand 10 after
47 caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp); 47 caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
48 } 48 }
49 49
50 PeerConnectionTestWrapper::PeerConnectionTestWrapper(const std::string& name, 50 PeerConnectionTestWrapper::PeerConnectionTestWrapper(const std::string& name,
51 rtc::Thread* worker_thread) 51 rtc::Thread* worker_thread)
52 : name_(name), worker_thread_(worker_thread) {} 52 : name_(name), worker_thread_(worker_thread) {}
53 53
54 PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {} 54 PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
55 55
56 bool PeerConnectionTestWrapper::CreatePc( 56 bool PeerConnectionTestWrapper::CreatePc(
57 const MediaConstraintsInterface* constraints) { 57 const MediaConstraintsInterface* constraints,
58 const webrtc::PeerConnectionInterface::RTCConfiguration& config) {
58 std::unique_ptr<cricket::PortAllocator> port_allocator( 59 std::unique_ptr<cricket::PortAllocator> port_allocator(
59 new cricket::FakePortAllocator(worker_thread_, nullptr)); 60 new cricket::FakePortAllocator(worker_thread_, nullptr));
60 61
61 fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); 62 fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
62 if (fake_audio_capture_module_ == NULL) { 63 if (fake_audio_capture_module_ == NULL) {
63 return false; 64 return false;
64 } 65 }
65 66
66 peer_connection_factory_ = webrtc::CreatePeerConnectionFactory( 67 peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
67 worker_thread_, rtc::Thread::Current(), fake_audio_capture_module_, NULL, 68 worker_thread_, rtc::Thread::Current(), fake_audio_capture_module_, NULL,
68 NULL); 69 NULL);
69 if (!peer_connection_factory_) { 70 if (!peer_connection_factory_) {
70 return false; 71 return false;
71 } 72 }
72 73
73 // CreatePeerConnection with RTCConfiguration.
74 webrtc::PeerConnectionInterface::RTCConfiguration config;
75 webrtc::PeerConnectionInterface::IceServer ice_server;
76 ice_server.uri = "stun:stun.l.google.com:19302";
77 config.servers.push_back(ice_server);
78 std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store( 74 std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store(
79 rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore() 75 rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
80 : nullptr); 76 : nullptr);
81 peer_connection_ = peer_connection_factory_->CreatePeerConnection( 77 peer_connection_ = peer_connection_factory_->CreatePeerConnection(
82 config, constraints, std::move(port_allocator), 78 config, constraints, std::move(port_allocator),
83 std::move(dtls_identity_store), this); 79 std::move(dtls_identity_store), this);
84 80
85 return peer_connection_.get() != NULL; 81 return peer_connection_.get() != NULL;
86 } 82 }
87 83
(...skipping 183 matching lines...) Expand 10 before | Expand all | Expand 10 after
271 peer_connection_factory_->CreateVideoSource( 267 peer_connection_factory_->CreateVideoSource(
272 new webrtc::FakePeriodicVideoCapturer(), &constraints); 268 new webrtc::FakePeriodicVideoCapturer(), &constraints);
273 std::string videotrack_label = label + kVideoTrackLabelBase; 269 std::string videotrack_label = label + kVideoTrackLabelBase;
274 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track( 270 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
275 peer_connection_factory_->CreateVideoTrack(videotrack_label, source)); 271 peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
276 272
277 stream->AddTrack(video_track); 273 stream->AddTrack(video_track);
278 } 274 }
279 return stream; 275 return stream;
280 } 276 }
OLDNEW
« no previous file with comments | « webrtc/api/test/peerconnectiontestwrapper.h ('k') | webrtc/api/webrtcsession.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698