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
This commit is contained in:
kjellander@webrtc.org
2015-03-04 12:58:35 +00:00
parent 792f1a14e2
commit 14665ff7d4
286 changed files with 3546 additions and 3920 deletions

View File

@ -89,7 +89,7 @@ class RtpRtcpAPITest : public ::testing::Test {
}
~RtpRtcpAPITest() {}
virtual void SetUp() OVERRIDE {
void SetUp() override {
RtpRtcp::Configuration configuration;
configuration.id = test_id;
configuration.audio = true;

View File

@ -35,10 +35,8 @@ class LoopBackTransport : public webrtc::Transport {
RtpReceiver* receiver,
ReceiveStatistics* receive_statistics);
void DropEveryNthPacket(int n);
virtual int SendPacket(int channel, const void* data, size_t len) OVERRIDE;
virtual int SendRTCPPacket(int channel,
const void* data,
size_t len) OVERRIDE;
int SendPacket(int channel, const void* data, size_t len) override;
int SendRTCPPacket(int channel, const void* data, size_t len) override;
private:
int count_;
@ -51,10 +49,10 @@ class LoopBackTransport : public webrtc::Transport {
class TestRtpReceiver : public NullRtpData {
public:
virtual int32_t OnReceivedPayloadData(
int32_t OnReceivedPayloadData(
const uint8_t* payload_data,
const size_t payload_size,
const webrtc::WebRtcRTPHeader* rtp_header) OVERRIDE;
const webrtc::WebRtcRTPHeader* rtp_header) override;
const uint8_t* payload_data() const { return payload_data_; }
size_t payload_size() const { return payload_size_; }

View File

@ -25,10 +25,10 @@ using namespace webrtc;
class VerifyingAudioReceiver : public NullRtpData {
public:
virtual int32_t OnReceivedPayloadData(
int32_t OnReceivedPayloadData(
const uint8_t* payloadData,
const size_t payloadSize,
const webrtc::WebRtcRTPHeader* rtpHeader) OVERRIDE {
const webrtc::WebRtcRTPHeader* rtpHeader) override {
if (rtpHeader->header.payloadType == 98 ||
rtpHeader->header.payloadType == 99) {
EXPECT_EQ(4u, payloadSize);
@ -61,13 +61,12 @@ class VerifyingAudioReceiver : public NullRtpData {
class RTPCallback : public NullRtpFeedback {
public:
virtual int32_t OnInitializeDecoder(
const int32_t id,
const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const uint8_t channels,
const uint32_t rate) OVERRIDE {
int32_t OnInitializeDecoder(const int32_t id,
const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const uint8_t channels,
const uint32_t rate) override {
if (payloadType == 96) {
EXPECT_EQ(test_rate, rate) <<
"The rate should be 64K for this payloadType";
@ -88,7 +87,7 @@ class RtpRtcpAudioTest : public ::testing::Test {
}
~RtpRtcpAudioTest() {}
virtual void SetUp() OVERRIDE {
void SetUp() override {
audioFeedback = new NullRtpAudioFeedback();
data_receiver1 = new VerifyingAudioReceiver();
data_receiver2 = new VerifyingAudioReceiver();
@ -133,7 +132,7 @@ class RtpRtcpAudioTest : public ::testing::Test {
rtp_receiver1_.get(), receive_statistics1_.get());
}
virtual void TearDown() OVERRIDE {
void TearDown() override {
delete module1;
delete module2;
delete transport1;