14665ff7d4
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
64 lines
1.9 KiB
C++
64 lines
1.9 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
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#include <stdio.h>
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#include <string>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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namespace webrtc {
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class RtpHeaderParser;
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namespace test {
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class RtpFileReader;
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class RtpFileSource : public PacketSource {
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public:
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// Creates an RtpFileSource reading from |file_name|. If the file cannot be
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// opened, or has the wrong format, NULL will be returned.
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static RtpFileSource* Create(const std::string& file_name);
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virtual ~RtpFileSource();
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// Registers an RTP header extension and binds it to |id|.
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virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
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// Returns a pointer to the next packet. Returns NULL if end of file was
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// reached, or if a the data was corrupt.
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Packet* NextPacket() override;
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private:
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static const int kFirstLineLength = 40;
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static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
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static const size_t kPacketHeaderSize = 8;
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RtpFileSource();
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bool OpenFile(const std::string& file_name);
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rtc::scoped_ptr<RtpFileReader> rtp_reader_;
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rtc::scoped_ptr<RtpHeaderParser> parser_;
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DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
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