Files
platform-external-webrtc/webrtc/modules/utility/source/rtp_dump_impl.h
kjellander@webrtc.org 14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00

50 lines
1.6 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
#define WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
#include "webrtc/modules/utility/interface/rtp_dump.h"
namespace webrtc {
class CriticalSectionWrapper;
class FileWrapper;
class RtpDumpImpl : public RtpDump
{
public:
RtpDumpImpl();
virtual ~RtpDumpImpl();
int32_t Start(const char* fileNameUTF8) override;
int32_t Stop() override;
bool IsActive() const override;
int32_t DumpPacket(const uint8_t* packet, size_t packetLength) override;
private:
// Return the system time in ms.
inline uint32_t GetTimeInMS() const;
// Return x in network byte order (big endian).
inline uint32_t RtpDumpHtonl(uint32_t x) const;
// Return x in network byte order (big endian).
inline uint16_t RtpDumpHtons(uint16_t x) const;
// Return true if the packet starts with a valid RTCP header.
// Note: See RtpUtility::RtpHeaderParser::RTCP() for details on how
// to determine if the packet is an RTCP packet.
bool RTCP(const uint8_t* packet) const;
private:
CriticalSectionWrapper* _critSect;
FileWrapper& _file;
uint32_t _startTime;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_