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

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
View File

@ -46,7 +46,7 @@ class AudioEncoderCngTest : public ::testing::Test {
config_.payload_type = kCngPayloadType;
}
virtual void TearDown() OVERRIDE {
void TearDown() override {
EXPECT_CALL(*mock_vad_, Die()).Times(1);
cng_.reset();
// Don't expect the cng_ object to delete the AudioEncoder object. But it
@ -407,7 +407,7 @@ class AudioEncoderCngDeathTest : public AudioEncoderCngTest {
// Override AudioEncoderCngTest::TearDown, since that one expects a call to
// the destructor of |mock_vad_|. In this case, that object is already
// deleted.
virtual void TearDown() OVERRIDE {
void TearDown() override {
cng_.reset();
// Don't expect the cng_ object to delete the AudioEncoder object. But it
// will be deleted with the test fixture. This is why we explicitly delete

20
webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
View File

@ -44,22 +44,22 @@ class AudioEncoderCng final : public AudioEncoder {
explicit AudioEncoderCng(const Config& config);
virtual ~AudioEncoderCng();
~AudioEncoderCng() override;
virtual int SampleRateHz() const OVERRIDE;
virtual int NumChannels() const OVERRIDE;
int SampleRateHz() const override;
int NumChannels() const override;
int RtpTimestampRateHz() const override;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
virtual int Max10MsFramesInAPacket() const OVERRIDE;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
void SetTargetBitrate(int bits_per_second) override;
void SetProjectedPacketLossRate(double fraction) override;
protected:
virtual void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) OVERRIDE;
void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) override;
private:
// Deleter for use with scoped_ptr. E.g., use as

32
webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
View File

@ -30,21 +30,21 @@ class AudioEncoderPcm : public AudioEncoder {
: frame_size_ms(20), num_channels(1), payload_type(pt) {}
};
virtual ~AudioEncoderPcm();
~AudioEncoderPcm() override;
virtual int SampleRateHz() const OVERRIDE;
virtual int NumChannels() const OVERRIDE;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
virtual int Max10MsFramesInAPacket() const OVERRIDE;
int SampleRateHz() const override;
int NumChannels() const override;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
protected:
AudioEncoderPcm(const Config& config, int sample_rate_hz);
virtual void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) OVERRIDE;
void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) override;
virtual int16_t EncodeCall(const int16_t* audio,
size_t input_len,
@ -70,9 +70,9 @@ class AudioEncoderPcmA : public AudioEncoderPcm {
: AudioEncoderPcm(config, kSampleRateHz) {}
protected:
virtual int16_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) OVERRIDE;
int16_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
private:
static const int kSampleRateHz = 8000;
@ -88,9 +88,9 @@ class AudioEncoderPcmU : public AudioEncoderPcm {
: AudioEncoderPcm(config, kSampleRateHz) {}
protected:
virtual int16_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) OVERRIDE;
int16_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
private:
static const int kSampleRateHz = 8000;

20
webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
View File

@ -28,20 +28,20 @@ class AudioEncoderG722 : public AudioEncoder {
};
explicit AudioEncoderG722(const Config& config);
virtual ~AudioEncoderG722();
~AudioEncoderG722() override;
virtual int SampleRateHz() const OVERRIDE;
int SampleRateHz() const override;
int RtpTimestampRateHz() const override;
virtual int NumChannels() const OVERRIDE;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
virtual int Max10MsFramesInAPacket() const OVERRIDE;
int NumChannels() const override;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
protected:
virtual void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) OVERRIDE;
void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) override;
private:
// The encoder state for one channel.

20
webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h
View File

@ -29,19 +29,19 @@ class AudioEncoderIlbc : public AudioEncoder {
};
explicit AudioEncoderIlbc(const Config& config);
virtual ~AudioEncoderIlbc();
~AudioEncoderIlbc() override;
virtual int SampleRateHz() const OVERRIDE;
virtual int NumChannels() const OVERRIDE;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
virtual int Max10MsFramesInAPacket() const OVERRIDE;
int SampleRateHz() const override;
int NumChannels() const override;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
protected:
virtual void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) OVERRIDE;
void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) override;
private:
static const int kMaxSamplesPerPacket = 480;

58
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
View File

@ -66,42 +66,42 @@ class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder {
explicit AudioEncoderDecoderIsacT(const Config& config);
explicit AudioEncoderDecoderIsacT(const ConfigAdaptive& config);
virtual ~AudioEncoderDecoderIsacT() OVERRIDE;
~AudioEncoderDecoderIsacT() override;
// AudioEncoder public methods.
virtual int SampleRateHz() const OVERRIDE;
virtual int NumChannels() const OVERRIDE;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
virtual int Max10MsFramesInAPacket() const OVERRIDE;
int SampleRateHz() const override;
int NumChannels() const override;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
// AudioDecoder methods.
virtual int Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) OVERRIDE;
virtual int DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) OVERRIDE;
virtual bool HasDecodePlc() const OVERRIDE;
virtual int DecodePlc(int num_frames, int16_t* decoded) OVERRIDE;
virtual int Init() OVERRIDE;
virtual int IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) OVERRIDE;
virtual int ErrorCode() OVERRIDE;
int Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
int DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
bool HasDecodePlc() const override;
int DecodePlc(int num_frames, int16_t* decoded) override;
int Init() override;
int IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) override;
int ErrorCode() override;
protected:
// AudioEncoder protected method.
virtual void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) OVERRIDE;
void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) override;
private:
// This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and

4
webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
View File

@ -23,8 +23,8 @@ static const int kIsacOutputSamplingKhz = 16;
class IsacSpeedTest : public AudioCodecSpeedTest {
protected:
IsacSpeedTest();
virtual void SetUp() OVERRIDE;
virtual void TearDown() OVERRIDE;
void SetUp() override;
void TearDown() override;
virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
int max_bytes, int* encoded_bytes);
virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,

20
webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
View File

@ -43,23 +43,23 @@ class AudioEncoderOpus final : public AudioEncoder {
};
explicit AudioEncoderOpus(const Config& config);
virtual ~AudioEncoderOpus() OVERRIDE;
~AudioEncoderOpus() override;
virtual int SampleRateHz() const OVERRIDE;
virtual int NumChannels() const OVERRIDE;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
virtual int Max10MsFramesInAPacket() const OVERRIDE;
int SampleRateHz() const override;
int NumChannels() const override;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
void SetTargetBitrate(int bits_per_second) override;
void SetProjectedPacketLossRate(double fraction) override;
double packet_loss_rate() const { return packet_loss_rate_; }
ApplicationMode application() const { return application_; }
protected:
virtual void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) OVERRIDE;
void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) override;
private:
const int num_10ms_frames_per_packet_;

4
webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
View File

@ -21,8 +21,8 @@ static const int kOpusSamplingKhz = 48;
class OpusSpeedTest : public AudioCodecSpeedTest {
protected:
OpusSpeedTest();
virtual void SetUp() OVERRIDE;
virtual void TearDown() OVERRIDE;
void SetUp() override;
void TearDown() override;
virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
int max_bytes, int* encoded_bytes);
virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,

6
webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h
View File

@ -28,9 +28,9 @@ class AudioEncoderPcm16B : public AudioEncoderPcm {
: AudioEncoderPcm(config, config.sample_rate_hz) {}
protected:
virtual int16_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) OVERRIDE;
int16_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
};
} // namespace webrtc

20
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
View File

@ -33,22 +33,22 @@ class AudioEncoderCopyRed : public AudioEncoder {
// Caller keeps ownership of the AudioEncoder object.
explicit AudioEncoderCopyRed(const Config& config);
virtual ~AudioEncoderCopyRed();
~AudioEncoderCopyRed() override;
virtual int SampleRateHz() const OVERRIDE;
int SampleRateHz() const override;
int RtpTimestampRateHz() const override;
virtual int NumChannels() const OVERRIDE;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
virtual int Max10MsFramesInAPacket() const OVERRIDE;
int NumChannels() const override;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
void SetTargetBitrate(int bits_per_second) override;
void SetProjectedPacketLossRate(double fraction) override;
protected:
virtual void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) OVERRIDE;
void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) override;
private:
AudioEncoder* speech_encoder_;

2
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
View File

@ -46,7 +46,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
.WillRepeatedly(Return(sample_rate_hz_));
}
virtual void TearDown() OVERRIDE {
void TearDown() override {
red_.reset();
// Don't expect the red_ object to delete the AudioEncoder object. But it
// will be deleted with the test fixture. This is why we explicitly delete