Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(143)

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

Issue 2206793007: Revert of Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 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 40 matching lines...) Expand 10 before | Expand all | Expand 10 after
51 const std::string& name, 51 const std::string& name,
52 rtc::Thread* network_thread, 52 rtc::Thread* network_thread,
53 rtc::Thread* worker_thread) 53 rtc::Thread* worker_thread)
54 : name_(name), 54 : name_(name),
55 network_thread_(network_thread), 55 network_thread_(network_thread),
56 worker_thread_(worker_thread) {} 56 worker_thread_(worker_thread) {}
57 57
58 PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {} 58 PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
59 59
60 bool PeerConnectionTestWrapper::CreatePc( 60 bool PeerConnectionTestWrapper::CreatePc(
61 const MediaConstraintsInterface* constraints, 61 const MediaConstraintsInterface* constraints) {
62 const webrtc::PeerConnectionInterface::RTCConfiguration& config) {
63 std::unique_ptr<cricket::PortAllocator> port_allocator( 62 std::unique_ptr<cricket::PortAllocator> port_allocator(
64 new cricket::FakePortAllocator(network_thread_, nullptr)); 63 new cricket::FakePortAllocator(network_thread_, nullptr));
65 64
66 fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); 65 fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
67 if (fake_audio_capture_module_ == NULL) { 66 if (fake_audio_capture_module_ == NULL) {
68 return false; 67 return false;
69 } 68 }
70 69
71 peer_connection_factory_ = webrtc::CreatePeerConnectionFactory( 70 peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
72 network_thread_, worker_thread_, rtc::Thread::Current(), 71 network_thread_, worker_thread_, rtc::Thread::Current(),
73 fake_audio_capture_module_, NULL, NULL); 72 fake_audio_capture_module_, NULL, NULL);
74 if (!peer_connection_factory_) { 73 if (!peer_connection_factory_) {
75 return false; 74 return false;
76 } 75 }
77 76
77 // CreatePeerConnection with RTCConfiguration.
78 webrtc::PeerConnectionInterface::RTCConfiguration config;
79 webrtc::PeerConnectionInterface::IceServer ice_server;
80 ice_server.uri = "stun:stun.l.google.com:19302";
81 config.servers.push_back(ice_server);
78 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator( 82 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator(
79 rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeRTCCertificateGenerator() 83 rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeRTCCertificateGenerator()
80 : nullptr); 84 : nullptr);
81 peer_connection_ = peer_connection_factory_->CreatePeerConnection( 85 peer_connection_ = peer_connection_factory_->CreatePeerConnection(
82 config, constraints, std::move(port_allocator), std::move(cert_generator), 86 config, constraints, std::move(port_allocator), std::move(cert_generator),
83 this); 87 this);
84 88
85 return peer_connection_.get() != NULL; 89 return peer_connection_.get() != NULL;
86 } 90 }
87 91
(...skipping 184 matching lines...) Expand 10 before | Expand all | Expand 10 after
272 peer_connection_factory_->CreateVideoSource( 276 peer_connection_factory_->CreateVideoSource(
273 new webrtc::FakePeriodicVideoCapturer(), &constraints); 277 new webrtc::FakePeriodicVideoCapturer(), &constraints);
274 std::string videotrack_label = label + kVideoTrackLabelBase; 278 std::string videotrack_label = label + kVideoTrackLabelBase;
275 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track( 279 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
276 peer_connection_factory_->CreateVideoTrack(videotrack_label, source)); 280 peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
277 281
278 stream->AddTrack(video_track); 282 stream->AddTrack(video_track);
279 } 283 }
280 return stream; 284 return stream;
281 } 285 }
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