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

webrtc/modules/audio_coding/neteq/tools/audio_checksum.h

@@ -26,7 +26,7 @@ class AudioChecksum : public AudioSink {
  public:
   AudioChecksum() : finished_(false) {}
 
-  virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
     if (finished_)
       return false;
 

webrtc/modules/audio_coding/neteq/tools/audio_sink.h

@@ -47,7 +47,7 @@ class AudioSinkFork : public AudioSink {
   AudioSinkFork(AudioSink* left, AudioSink* right)
       : left_sink_(left), right_sink_(right) {}
 
-  virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
     return left_sink_->WriteArray(audio, num_samples) &&
            right_sink_->WriteArray(audio, num_samples);
   }

webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h

@@ -33,7 +33,7 @@ class ConstantPcmPacketSource : public PacketSource {
 
   // Returns a pointer to the next packet. Will never return NULL. That is,
   // the source is infinite.
-  Packet* NextPacket() OVERRIDE;
+  Packet* NextPacket() override;
 
  private:
   void WriteHeader(uint8_t* packet_memory);

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

@@ -33,13 +33,13 @@ class LossModel {
 
 class NoLoss : public LossModel {
  public:
-  virtual bool Lost() OVERRIDE;
+  bool Lost() override;
 };
 
 class UniformLoss : public LossModel {
  public:
   UniformLoss(double loss_rate);
-  virtual bool Lost() OVERRIDE;
+  bool Lost() override;
   void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
 
  private:
@@ -49,7 +49,7 @@ class UniformLoss : public LossModel {
 class GilbertElliotLoss : public LossModel {
  public:
   GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
-  virtual bool Lost() OVERRIDE;
+  bool Lost() override;
 
  private:
   // Prob. of losing current packet, when previous packet is lost.
@@ -69,8 +69,8 @@ class NetEqQualityTest : public ::testing::Test {
                    int channels,
                    std::string in_filename,
                    std::string out_filename);
-  virtual void SetUp() OVERRIDE;
-  virtual void TearDown() OVERRIDE;
+  void SetUp() override;
+  void TearDown() override;
 
   // EncodeBlock(...) does the following:
   // 1. encodes a block of audio, saved in |in_data| and has a length of

webrtc/modules/audio_coding/neteq/tools/output_audio_file.h

@@ -34,7 +34,7 @@ class OutputAudioFile : public AudioSink {
       fclose(out_file_);
   }
 
-  virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
     assert(out_file_);
     return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
   }

webrtc/modules/audio_coding/neteq/tools/output_wav_file.h

@@ -27,7 +27,7 @@ class OutputWavFile : public AudioSink {
   OutputWavFile(const std::string& file_name, int sample_rate_hz)
       : wav_writer_(file_name, sample_rate_hz, 1) {}
 
-  virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
     wav_writer_.WriteSamples(audio, num_samples);
     return true;
   }

webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h

@@ -41,7 +41,7 @@ class RtpFileSource : public PacketSource {
 
   // Returns a pointer to the next packet. Returns NULL if end of file was
   // reached, or if a the data was corrupt.
-  virtual Packet* NextPacket() OVERRIDE;
+  Packet* NextPacket() override;
 
  private:
   static const int kFirstLineLength = 40;

webrtc/modules/audio_coding/neteq/tools/rtp_generator.h

@@ -70,7 +70,7 @@ class TimestampJumpRtpGenerator : public RtpGenerator {
 
   uint32_t GetRtpHeader(uint8_t payload_type,
                         size_t payload_length_samples,
-                        WebRtcRTPHeader* rtp_header) OVERRIDE;
+                        WebRtcRTPHeader* rtp_header) override;
 
  private:
   uint32_t jump_from_timestamp_;