Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(8)

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

Issue 2969653002: Update includes for webrtc/{base => rtc_base} rename (1/3) (Closed)
Patch Set: Rebased onto 89c4a7e57d524b13fbe0c823a83a4c10c2e63bd0 Created 3 years, 5 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/base/arraysize.h"
14 #include "webrtc/base/array_view.h"
15 #include "webrtc/base/random.h"
16 #include "webrtc/common_types.h" 13 #include "webrtc/common_types.h"
17 #include "webrtc/common_video/include/video_bitrate_allocator.h" 14 #include "webrtc/common_video/include/video_bitrate_allocator.h"
18 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 15 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" 16 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
20 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" 17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
21 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" 18 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
22 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h" 19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
23 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" 20 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
24 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" 21 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
25 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h" 22 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h"
26 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h" 23 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
27 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
28 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h" 25 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
29 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" 26 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
30 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h" 27 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
31 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h" 28 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
32 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" 29 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
33 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h" 30 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
34 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" 31 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
35 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" 32 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
36 #include "webrtc/modules/rtp_rtcp/source/time_util.h" 33 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
34 #include "webrtc/rtc_base/array_view.h"
35 #include "webrtc/rtc_base/arraysize.h"
36 #include "webrtc/rtc_base/random.h"
37 #include "webrtc/system_wrappers/include/ntp_time.h" 37 #include "webrtc/system_wrappers/include/ntp_time.h"
38 #include "webrtc/test/gmock.h" 38 #include "webrtc/test/gmock.h"
39 #include "webrtc/test/gtest.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;
(...skipping 1196 matching lines...) Expand 10 before | Expand all | Expand 10 after
1243 1243
1244 rtcp::ExtendedReports xr; 1244 rtcp::ExtendedReports xr;
1245 xr.SetTargetBitrate(bitrate); 1245 xr.SetTargetBitrate(bitrate);
1246 1246
1247 EXPECT_CALL(bitrate_allocation_observer_, 1247 EXPECT_CALL(bitrate_allocation_observer_,
1248 OnBitrateAllocationUpdated(expected_allocation)); 1248 OnBitrateAllocationUpdated(expected_allocation));
1249 InjectRtcpPacket(xr); 1249 InjectRtcpPacket(xr);
1250 } 1250 }
1251 1251
1252 } // namespace webrtc 1252 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc ('k') | webrtc/modules/rtp_rtcp/source/rtcp_sender.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698