commit 14665ff7d4024d07e58622f498b23fd980001871
author kjellander@webrtc.org <kjellander@webrtc.org> Wed Mar 04 12:58:35 2015 +0000
committer kjellander@webrtc.org <kjellander@webrtc.org> Wed Mar 04 13:04:54 2015 +0000
tree 7548808544f25988d15e2d0172e92474a228756a
parent 792f1a14e2b62382c5c6080d0b8fdc5c89d27bc6

Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro

Clang version changed 223108:230914
Details: https://chromium.googlesource.com/chromium/src/+/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

webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h

diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
index 4ee4fa2..44fb0b2 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
@@ -29,13 +29,12 @@
  public:
   TestPacketization(RTPStream *rtpStream, uint16_t frequency);
   ~TestPacketization();
-  virtual int32_t SendData(
-      const FrameType frameType,
-      const uint8_t payloadType,
-      const uint32_t timeStamp,
-      const uint8_t* payloadData,
-      const size_t payloadSize,
-      const RTPFragmentationHeader* fragmentation) OVERRIDE;
+  int32_t SendData(const FrameType frameType,
+                   const uint8_t payloadType,
+                   const uint32_t timeStamp,
+                   const uint8_t* payloadData,
+                   const size_t payloadSize,
+                   const RTPFragmentationHeader* fragmentation) override;
 
  private:
   static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
@@ -103,7 +102,7 @@
  public:
   EncodeDecodeTest();
   explicit EncodeDecodeTest(int testMode);
-  virtual void Perform() OVERRIDE;
+  void Perform() override;
 
   uint16_t _playoutFreq;
   uint8_t _testMode;