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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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 <vector> | 12 #include <vector> |
13 | 13 |
| 14 #include "webrtc/base/buffer.h" |
| 15 #include "webrtc/base/rate_limiter.h" |
14 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" | 16 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
15 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" | 17 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" |
16 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h" | 18 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h" |
17 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 19 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" | 21 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
20 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" | 22 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" |
21 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" | 23 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
22 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" | 24 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
23 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" | 25 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" |
24 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" | 26 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
25 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" | 27 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" |
26 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 28 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
27 #include "webrtc/rtc_base/buffer.h" | |
28 #include "webrtc/rtc_base/rate_limiter.h" | |
29 #include "webrtc/test/field_trial.h" | 29 #include "webrtc/test/field_trial.h" |
30 #include "webrtc/test/gmock.h" | 30 #include "webrtc/test/gmock.h" |
31 #include "webrtc/test/gtest.h" | 31 #include "webrtc/test/gtest.h" |
32 #include "webrtc/test/mock_transport.h" | 32 #include "webrtc/test/mock_transport.h" |
33 #include "webrtc/typedefs.h" | 33 #include "webrtc/typedefs.h" |
34 | 34 |
35 namespace webrtc { | 35 namespace webrtc { |
36 | 36 |
37 namespace { | 37 namespace { |
38 const int kTransmissionTimeOffsetExtensionId = 1; | 38 const int kTransmissionTimeOffsetExtensionId = 1; |
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1717 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead, | 1717 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead, |
1718 RtpSenderTestWithoutPacer, | 1718 RtpSenderTestWithoutPacer, |
1719 ::testing::Bool()); | 1719 ::testing::Bool()); |
1720 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead, | 1720 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead, |
1721 RtpSenderVideoTest, | 1721 RtpSenderVideoTest, |
1722 ::testing::Bool()); | 1722 ::testing::Bool()); |
1723 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead, | 1723 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead, |
1724 RtpSenderAudioTest, | 1724 RtpSenderAudioTest, |
1725 ::testing::Bool()); | 1725 ::testing::Bool()); |
1726 } // namespace webrtc | 1726 } // namespace webrtc |
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