Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(78)

Side by Side Diff: webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc

Issue 2964773002: Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" (Closed)
Patch Set: 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 "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h" 11 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 #include <memory> 14 #include <memory>
15 #include <vector> 15 #include <vector>
16 16
17 #include "webrtc/rtc_base/checks.h" 17 #include "webrtc/base/checks.h"
18 #include "webrtc/rtc_base/rate_limiter.h" 18 #include "webrtc/base/rate_limiter.h"
19 #include "webrtc/test/null_transport.h" 19 #include "webrtc/test/null_transport.h"
20 20
21 namespace webrtc { 21 namespace webrtc {
22 22
23 void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module, 23 void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
24 RTPPayloadRegistry* payload_registry, 24 RTPPayloadRegistry* payload_registry,
25 RtpReceiver* receiver, 25 RtpReceiver* receiver,
26 ReceiveStatistics* receive_statistics) { 26 ReceiveStatistics* receive_statistics) {
27 rtp_rtcp_module_ = rtp_rtcp_module; 27 rtp_rtcp_module_ = rtp_rtcp_module;
28 rtp_payload_registry_ = payload_registry; 28 rtp_payload_registry_ = payload_registry;
(...skipping 146 matching lines...) Expand 10 before | Expand all | Expand 10 after
175 rtx_header.payloadType = kRtxPayloadType; 175 rtx_header.payloadType = kRtxPayloadType;
176 EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header)); 176 EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
177 rtx_header.ssrc = 0; 177 rtx_header.ssrc = 0;
178 EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header)); 178 EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
179 rtx_header.ssrc = kRtxSsrc; 179 rtx_header.ssrc = kRtxSsrc;
180 rtx_header.payloadType = 0; 180 rtx_header.payloadType = 0;
181 EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header)); 181 EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
182 } 182 }
183 183
184 } // namespace webrtc 184 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc ('k') | webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698