Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(197)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc

Issue 1460043002: Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Restore the Pass methods Created 5 years 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) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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/source/rtcp_packet/extended_jitter_report.h" 11 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
12 12
13 #include <limits> 13 #include <limits>
14 14
15 #include "testing/gtest/include/gtest/gtest.h" 15 #include "testing/gtest/include/gtest/gtest.h"
16 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" 16 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
17 17
18 using webrtc::rtcp::RawPacket; 18 using webrtc::rtcp::RawPacket;
19 using webrtc::rtcp::ExtendedJitterReport; 19 using webrtc::rtcp::ExtendedJitterReport;
20 using webrtc::RTCPUtility::RtcpCommonHeader; 20 using webrtc::RTCPUtility::RtcpCommonHeader;
21 using webrtc::RTCPUtility::RtcpParseCommonHeader; 21 using webrtc::RTCPUtility::RtcpParseCommonHeader;
22 22
23 namespace webrtc { 23 namespace webrtc {
24 namespace { 24 namespace {
25 25
26 class RtcpPacketExtendedJitterReportTest : public ::testing::Test { 26 class RtcpPacketExtendedJitterReportTest : public ::testing::Test {
27 protected: 27 protected:
28 void BuildPacket() { packet = ij.Build().Pass(); } 28 void BuildPacket() { packet = ij.Build(); }
29 void ParsePacket() { 29 void ParsePacket() {
30 RtcpCommonHeader header; 30 RtcpCommonHeader header;
31 EXPECT_TRUE( 31 EXPECT_TRUE(
32 RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header)); 32 RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
33 EXPECT_EQ(header.BlockSize(), packet->Length()); 33 EXPECT_EQ(header.BlockSize(), packet->Length());
34 EXPECT_TRUE(parsed_.Parse( 34 EXPECT_TRUE(parsed_.Parse(
35 header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes)); 35 header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
36 } 36 }
37 37
38 ExtendedJitterReport ij; 38 ExtendedJitterReport ij;
(...skipping 50 matching lines...) Expand 10 before | Expand all | Expand 10 after
89 header.count_or_format++; // Damage package. 89 header.count_or_format++; // Damage package.
90 90
91 ExtendedJitterReport parsed; 91 ExtendedJitterReport parsed;
92 92
93 EXPECT_FALSE(parsed.Parse( 93 EXPECT_FALSE(parsed.Parse(
94 header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes)); 94 header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
95 } 95 }
96 96
97 } // namespace 97 } // namespace
98 } // namespace webrtc 98 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698