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Issue 2969623003: Update includes for webrtc/{base => rtc_base} rename (2/3) (Closed)
Patch Set: Rebased onto 224e65939af87443addfc5bb500fbf434728bd1c and restored sorting in clock.cc Created 3 years, 5 months ago
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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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