Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(344)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc

Issue 2381013002: Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ (Closed)
Patch Set: rebase Created 4 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 12
13 #include "webrtc/test/gmock.h"
14 #include "webrtc/test/gtest.h"
15 #include "webrtc/base/array_view.h" 13 #include "webrtc/base/array_view.h"
16 #include "webrtc/base/random.h" 14 #include "webrtc/base/random.h"
17 #include "webrtc/common_types.h" 15 #include "webrtc/common_types.h"
18 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 16 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" 17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
20 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" 18 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
21 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" 19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
22 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h" 20 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
23 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" 21 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
24 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" 22 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
25 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h" 23 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h"
26 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
27 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h" 25 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
28 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h" 26 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
29 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" 27 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
30 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h" 28 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
31 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rpsi.h" 29 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rpsi.h"
32 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h" 30 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
33 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" 31 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
34 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h" 32 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
35 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h" 33 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
36 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" 34 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
37 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" 35 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
38 #include "webrtc/modules/rtp_rtcp/source/time_util.h" 36 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
39 #include "webrtc/system_wrappers/include/ntp_time.h" 37 #include "webrtc/system_wrappers/include/ntp_time.h"
38 #include "webrtc/test/gmock.h"
39 #include "webrtc/test/gtest.h"
40 40
41 namespace webrtc { 41 namespace webrtc {
42 namespace { 42 namespace {
43 43
44 using ::testing::_; 44 using ::testing::_;
45 using ::testing::AllOf; 45 using ::testing::AllOf;
46 using ::testing::ElementsAreArray; 46 using ::testing::ElementsAreArray;
47 using ::testing::Field; 47 using ::testing::Field;
48 using ::testing::IsEmpty; 48 using ::testing::IsEmpty;
49 using ::testing::NiceMock; 49 using ::testing::NiceMock;
(...skipping 1211 matching lines...) Expand 10 before | Expand all | Expand 10 after
1261 TEST_F(RtcpReceiverTest, ForceSenderReport) { 1261 TEST_F(RtcpReceiverTest, ForceSenderReport) {
1262 rtcp::RapidResyncRequest rr; 1262 rtcp::RapidResyncRequest rr;
1263 rr.SetSenderSsrc(kSenderSsrc); 1263 rr.SetSenderSsrc(kSenderSsrc);
1264 rr.SetMediaSsrc(kReceiverMainSsrc); 1264 rr.SetMediaSsrc(kReceiverMainSsrc);
1265 1265
1266 EXPECT_CALL(rtp_rtcp_impl_, OnRequestSendReport()); 1266 EXPECT_CALL(rtp_rtcp_impl_, OnRequestSendReport());
1267 InjectRtcpPacket(rr); 1267 InjectRtcpPacket(rr);
1268 } 1268 }
1269 1269
1270 } // namespace webrtc 1270 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc ('k') | webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698