Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(20)

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

Issue 1922103002: Remove "This file includes unit tests" comments. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 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) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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 *
10 * This file includes unit tests for the RtcpPacket.
11 */ 9 */
12 10
13 #include "testing/gmock/include/gmock/gmock.h" 11 #include "testing/gmock/include/gmock/gmock.h"
14 #include "testing/gtest/include/gtest/gtest.h" 12 #include "testing/gtest/include/gtest/gtest.h"
15 13
16 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" 14 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" 15 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
18 16
19 using webrtc::rtcp::ReceiverReport; 17 using webrtc::rtcp::ReceiverReport;
20 using webrtc::rtcp::ReportBlock; 18 using webrtc::rtcp::ReportBlock;
(...skipping 15 matching lines...) Expand all
36 class Verifier : public rtcp::RtcpPacket::PacketReadyCallback { 34 class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
37 void OnPacketReady(uint8_t* data, size_t length) override { 35 void OnPacketReady(uint8_t* data, size_t length) override {
38 ADD_FAILURE() << "Packet should not fit within max size."; 36 ADD_FAILURE() << "Packet should not fit within max size.";
39 } 37 }
40 } verifier; 38 } verifier;
41 const size_t kBufferSize = kRrLength + kReportBlockLength - 1; 39 const size_t kBufferSize = kRrLength + kReportBlockLength - 1;
42 uint8_t buffer[kBufferSize]; 40 uint8_t buffer[kBufferSize];
43 EXPECT_FALSE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier)); 41 EXPECT_FALSE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
44 } 42 }
45 } // namespace webrtc 43 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/neteq/neteq_unittest.cc ('k') | webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698