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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 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 |
11 #include <memory> | 11 #include <memory> |
12 #include <sstream> | 12 #include <sstream> |
13 #include <string> | 13 #include <string> |
14 #include <utility> | 14 #include <utility> |
15 | 15 |
16 #include "webrtc/api/jsepsessiondescription.h" | 16 #include "webrtc/api/jsepsessiondescription.h" |
17 #include "webrtc/api/mediastreaminterface.h" | 17 #include "webrtc/api/mediastreaminterface.h" |
18 #include "webrtc/api/peerconnectioninterface.h" | 18 #include "webrtc/api/peerconnectioninterface.h" |
19 #include "webrtc/api/rtpreceiverinterface.h" | 19 #include "webrtc/api/rtpreceiverinterface.h" |
20 #include "webrtc/api/rtpsenderinterface.h" | 20 #include "webrtc/api/rtpsenderinterface.h" |
21 #include "webrtc/api/test/fakeconstraints.h" | 21 #include "webrtc/api/test/fakeconstraints.h" |
22 #include "webrtc/base/gunit.h" | 22 #include "webrtc/base/gunit.h" |
23 #include "webrtc/base/ssladapter.h" | 23 #include "webrtc/base/ssladapter.h" |
24 #include "webrtc/base/sslstreamadapter.h" | 24 #include "webrtc/base/sslstreamadapter.h" |
25 #include "webrtc/base/stringutils.h" | 25 #include "webrtc/base/stringutils.h" |
26 #include "webrtc/base/thread.h" | 26 #include "webrtc/base/thread.h" |
27 #include "webrtc/media/base/fakevideocapturer.h" | 27 #include "webrtc/media/base/fakevideocapturer.h" |
28 #include "webrtc/media/sctp/sctptransportinternal.h" | 28 #include "webrtc/media/sctp/sctptransportinternal.h" |
| 29 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" |
29 #include "webrtc/p2p/base/fakeportallocator.h" | 30 #include "webrtc/p2p/base/fakeportallocator.h" |
30 #include "webrtc/pc/audiotrack.h" | 31 #include "webrtc/pc/audiotrack.h" |
31 #include "webrtc/pc/mediasession.h" | 32 #include "webrtc/pc/mediasession.h" |
32 #include "webrtc/pc/mediastream.h" | 33 #include "webrtc/pc/mediastream.h" |
33 #include "webrtc/pc/peerconnection.h" | 34 #include "webrtc/pc/peerconnection.h" |
34 #include "webrtc/pc/streamcollection.h" | 35 #include "webrtc/pc/streamcollection.h" |
35 #include "webrtc/pc/test/fakertccertificategenerator.h" | 36 #include "webrtc/pc/test/fakertccertificategenerator.h" |
36 #include "webrtc/pc/test/fakevideotracksource.h" | 37 #include "webrtc/pc/test/fakevideotracksource.h" |
37 #include "webrtc/pc/test/mockpeerconnectionobservers.h" | 38 #include "webrtc/pc/test/mockpeerconnectionobservers.h" |
38 #include "webrtc/pc/test/testsdpstrings.h" | 39 #include "webrtc/pc/test/testsdpstrings.h" |
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658 }; | 659 }; |
659 | 660 |
660 } // namespace | 661 } // namespace |
661 | 662 |
662 // The PeerConnectionMediaConfig tests below verify that configuration | 663 // The PeerConnectionMediaConfig tests below verify that configuration |
663 // and constraints are propagated into the MediaConfig passed to | 664 // and constraints are propagated into the MediaConfig passed to |
664 // CreateMediaController. These settings are intended for MediaChannel | 665 // CreateMediaController. These settings are intended for MediaChannel |
665 // constructors, but that is not exercised by these unittest. | 666 // constructors, but that is not exercised by these unittest. |
666 class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory { | 667 class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory { |
667 public: | 668 public: |
| 669 PeerConnectionFactoryForTest() |
| 670 : webrtc::PeerConnectionFactory( |
| 671 webrtc::CreateBuiltinAudioEncoderFactory(), |
| 672 webrtc::CreateBuiltinAudioDecoderFactory()) {} |
| 673 |
668 webrtc::MediaControllerInterface* CreateMediaController( | 674 webrtc::MediaControllerInterface* CreateMediaController( |
669 const cricket::MediaConfig& config, | 675 const cricket::MediaConfig& config, |
670 webrtc::RtcEventLog* event_log) const override { | 676 webrtc::RtcEventLog* event_log) const override { |
671 create_media_controller_called_ = true; | 677 create_media_controller_called_ = true; |
672 create_media_controller_config_ = config; | 678 create_media_controller_config_ = config; |
673 | 679 |
674 webrtc::MediaControllerInterface* mc = | 680 webrtc::MediaControllerInterface* mc = |
675 PeerConnectionFactory::CreateMediaController(config, event_log); | 681 PeerConnectionFactory::CreateMediaController(config, event_log); |
676 EXPECT_TRUE(mc != nullptr); | 682 EXPECT_TRUE(mc != nullptr); |
677 return mc; | 683 return mc; |
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3395 EXPECT_NE(a, f); | 3401 EXPECT_NE(a, f); |
3396 | 3402 |
3397 PeerConnectionInterface::RTCConfiguration g; | 3403 PeerConnectionInterface::RTCConfiguration g; |
3398 g.disable_ipv6 = true; | 3404 g.disable_ipv6 = true; |
3399 EXPECT_NE(a, g); | 3405 EXPECT_NE(a, g); |
3400 | 3406 |
3401 PeerConnectionInterface::RTCConfiguration h( | 3407 PeerConnectionInterface::RTCConfiguration h( |
3402 PeerConnectionInterface::RTCConfigurationType::kAggressive); | 3408 PeerConnectionInterface::RTCConfigurationType::kAggressive); |
3403 EXPECT_NE(a, h); | 3409 EXPECT_NE(a, h); |
3404 } | 3410 } |
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