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1 /* | 1 /* |
2 * Copyright 2017 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2017 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 <utility> // For std::pair, std::move. | 12 #include <utility> // For std::pair, std::move. |
13 | 13 |
14 #include "webrtc/api/ortc/ortcfactoryinterface.h" | 14 #include "webrtc/api/ortc/ortcfactoryinterface.h" |
15 #include "webrtc/base/criticalsection.h" | |
16 #include "webrtc/base/fakenetwork.h" | |
17 #include "webrtc/base/gunit.h" | |
18 #include "webrtc/base/virtualsocketserver.h" | |
19 #include "webrtc/ortc/testrtpparameters.h" | 15 #include "webrtc/ortc/testrtpparameters.h" |
20 #include "webrtc/p2p/base/udptransport.h" | 16 #include "webrtc/p2p/base/udptransport.h" |
21 #include "webrtc/pc/test/fakeaudiocapturemodule.h" | 17 #include "webrtc/pc/test/fakeaudiocapturemodule.h" |
22 #include "webrtc/pc/test/fakeperiodicvideocapturer.h" | 18 #include "webrtc/pc/test/fakeperiodicvideocapturer.h" |
23 #include "webrtc/pc/test/fakevideotrackrenderer.h" | 19 #include "webrtc/pc/test/fakevideotrackrenderer.h" |
| 20 #include "webrtc/rtc_base/criticalsection.h" |
| 21 #include "webrtc/rtc_base/fakenetwork.h" |
| 22 #include "webrtc/rtc_base/gunit.h" |
| 23 #include "webrtc/rtc_base/virtualsocketserver.h" |
24 | 24 |
25 namespace { | 25 namespace { |
26 | 26 |
27 const int kDefaultTimeout = 10000; // 10 seconds. | 27 const int kDefaultTimeout = 10000; // 10 seconds. |
28 const int kReceivingDuration = 1000; // 1 second. | 28 const int kReceivingDuration = 1000; // 1 second. |
29 // Default number of audio/video frames to wait for before considering a test a | 29 // Default number of audio/video frames to wait for before considering a test a |
30 // success. | 30 // success. |
31 const int kDefaultNumFrames = 3; | 31 const int kDefaultNumFrames = 3; |
32 const rtc::IPAddress kIPv4LocalHostAddress = | 32 const rtc::IPAddress kIPv4LocalHostAddress = |
33 rtc::IPAddress(0x7F000001); // 127.0.0.1 | 33 rtc::IPAddress(0x7F000001); // 127.0.0.1 |
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672 // BaseChannel model relies on there being a single VoiceChannel and | 672 // BaseChannel model relies on there being a single VoiceChannel and |
673 // VideoChannel, and these only support a single set of codecs/etc. per | 673 // VideoChannel, and these only support a single set of codecs/etc. per |
674 // send/receive direction. | 674 // send/receive direction. |
675 | 675 |
676 // TODO(deadbeef): End-to-end test for simulcast, once that's supported by this | 676 // TODO(deadbeef): End-to-end test for simulcast, once that's supported by this |
677 // API. | 677 // API. |
678 | 678 |
679 #endif // if !defined(THREAD_SANITIZER) | 679 #endif // if !defined(THREAD_SANITIZER) |
680 | 680 |
681 } // namespace webrtc | 681 } // namespace webrtc |
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