Files
platform-external-webrtc/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
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

79 lines
2.6 KiB
C++

/*
* Copyright (c) 2013 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.
*/
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
namespace webrtc {
class RtpHeaderParserImpl : public RtpHeaderParser {
public:
RtpHeaderParserImpl();
virtual ~RtpHeaderParserImpl() {}
bool Parse(const uint8_t* packet,
size_t length,
RTPHeader* header) const override;
bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id) override;
bool DeregisterRtpHeaderExtension(RTPExtensionType type) override;
private:
rtc::scoped_ptr<CriticalSectionWrapper> critical_section_;
RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(critical_section_);
};
RtpHeaderParser* RtpHeaderParser::Create() {
return new RtpHeaderParserImpl;
}
RtpHeaderParserImpl::RtpHeaderParserImpl()
: critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {}
bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) {
RtpUtility::RtpHeaderParser rtp_parser(packet, length);
return rtp_parser.RTCP();
}
bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
size_t length,
RTPHeader* header) const {
RtpUtility::RtpHeaderParser rtp_parser(packet, length);
memset(header, 0, sizeof(*header));
RtpHeaderExtensionMap map;
{
CriticalSectionScoped cs(critical_section_.get());
rtp_header_extension_map_.GetCopy(&map);
}
const bool valid_rtpheader = rtp_parser.Parse(*header, &map);
if (!valid_rtpheader) {
return false;
}
return true;
}
bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
CriticalSectionScoped cs(critical_section_.get());
return rtp_header_extension_map_.Register(type, id) == 0;
}
bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
CriticalSectionScoped cs(critical_section_.get());
return rtp_header_extension_map_.Deregister(type) == 0;
}
} // namespace webrtc