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1 /* | 1 /* |
2 * Copyright 2016 The WebRTC Project Authors. All rights reserved. | 2 * Copyright 2016 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 "webrtc/pc/trackmediainfomap.h" | 11 #include "webrtc/pc/trackmediainfomap.h" |
12 | 12 |
13 #include <initializer_list> | 13 #include <initializer_list> |
14 #include <memory> | 14 #include <memory> |
15 #include <utility> | 15 #include <utility> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/api/rtpreceiverinterface.h" | 18 #include "webrtc/api/rtpreceiverinterface.h" |
19 #include "webrtc/api/rtpsenderinterface.h" | 19 #include "webrtc/api/rtpsenderinterface.h" |
20 #include "webrtc/api/test/mock_rtpreceiver.h" | 20 #include "webrtc/api/test/mock_rtpreceiver.h" |
21 #include "webrtc/api/test/mock_rtpsender.h" | 21 #include "webrtc/api/test/mock_rtpsender.h" |
22 #include "webrtc/base/refcount.h" | |
23 #include "webrtc/media/base/mediachannel.h" | 22 #include "webrtc/media/base/mediachannel.h" |
24 #include "webrtc/pc/audiotrack.h" | 23 #include "webrtc/pc/audiotrack.h" |
25 #include "webrtc/pc/test/fakevideotracksource.h" | 24 #include "webrtc/pc/test/fakevideotracksource.h" |
26 #include "webrtc/pc/videotrack.h" | 25 #include "webrtc/pc/videotrack.h" |
| 26 #include "webrtc/rtc_base/refcount.h" |
27 #include "webrtc/test/gtest.h" | 27 #include "webrtc/test/gtest.h" |
28 | 28 |
29 namespace webrtc { | 29 namespace webrtc { |
30 | 30 |
31 namespace { | 31 namespace { |
32 | 32 |
33 RtpParameters CreateRtpParametersWithSsrcs( | 33 RtpParameters CreateRtpParametersWithSsrcs( |
34 std::initializer_list<uint32_t> ssrcs) { | 34 std::initializer_list<uint32_t> ssrcs) { |
35 RtpParameters params; | 35 RtpParameters params; |
36 for (uint32_t ssrc : ssrcs) { | 36 for (uint32_t ssrc : ssrcs) { |
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401 AddRtpReceiverWithSsrcs({1, 2}, remote_audio_track_); | 401 AddRtpReceiverWithSsrcs({1, 2}, remote_audio_track_); |
402 AddRtpReceiverWithSsrcs({3, 4}, remote_audio_track_); | 402 AddRtpReceiverWithSsrcs({3, 4}, remote_audio_track_); |
403 AddRtpReceiverWithSsrcs({5, 6}, remote_video_track_); | 403 AddRtpReceiverWithSsrcs({5, 6}, remote_video_track_); |
404 AddRtpReceiverWithSsrcs({7, 8}, remote_video_track_); | 404 AddRtpReceiverWithSsrcs({7, 8}, remote_video_track_); |
405 EXPECT_DEATH(CreateMap(), ""); | 405 EXPECT_DEATH(CreateMap(), ""); |
406 } | 406 } |
407 | 407 |
408 #endif // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) | 408 #endif // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) |
409 | 409 |
410 } // namespace webrtc | 410 } // namespace webrtc |
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