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
diff --git a/webrtc/modules/audio_coding/neteq/accelerate.h b/webrtc/modules/audio_coding/neteq/accelerate.h
index 2da9993..6e3aa46 100644
--- a/webrtc/modules/audio_coding/neteq/accelerate.h
+++ b/webrtc/modules/audio_coding/neteq/accelerate.h
@@ -49,16 +49,18 @@
protected:
// Sets the parameters |best_correlation| and |peak_index| to suitable
// values when the signal contains no active speech.
- virtual void SetParametersForPassiveSpeech(size_t len,
- int16_t* best_correlation,
- int* peak_index) const OVERRIDE;
+ void SetParametersForPassiveSpeech(size_t len,
+ int16_t* best_correlation,
+ int* peak_index) const override;
// Checks the criteria for performing the time-stretching operation and,
// if possible, performs the time-stretching.
- virtual ReturnCodes CheckCriteriaAndStretch(
- const int16_t* input, size_t input_length, size_t peak_index,
- int16_t best_correlation, bool active_speech,
- AudioMultiVector* output) const OVERRIDE;
+ ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
+ size_t input_length,
+ size_t peak_index,
+ int16_t best_correlation,
+ bool active_speech,
+ AudioMultiVector* output) const override;
private:
DISALLOW_COPY_AND_ASSIGN(Accelerate);
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
index 7d36a39..7e80a36 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -233,7 +233,7 @@
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) { return -1; }
- virtual CNG_dec_inst* CngDecoderInstance() OVERRIDE { return dec_state_; }
+ CNG_dec_inst* CngDecoderInstance() override { return dec_state_; }
private:
CNG_dec_inst* dec_state_;
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_fax.h b/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
index 01a948f..97c481d 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
@@ -47,13 +47,13 @@
// should be set to true. The output variable |reset_decoder| will be set to
// true if a reset is required; otherwise it is left unchanged (i.e., it can
// remain true if it was true before the call).
- virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
- const Expand& expand,
- int decoder_frame_length,
- const RTPHeader* packet_header,
- Modes prev_mode,
- bool play_dtmf,
- bool* reset_decoder) OVERRIDE;
+ Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
+ const Expand& expand,
+ int decoder_frame_length,
+ const RTPHeader* packet_header,
+ Modes prev_mode,
+ bool play_dtmf,
+ bool* reset_decoder) override;
private:
DISALLOW_COPY_AND_ASSIGN(DecisionLogicFax);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index 8a382e9..28e901e 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -378,7 +378,7 @@
class ShortTimestampJumpTest : public LargeTimestampJumpTest {
protected:
- void UpdateState(NetEqOutputType output_type) OVERRIDE {
+ void UpdateState(NetEqOutputType output_type) override {
switch (test_state_) {
case kInitialPhase: {
if (output_type == kOutputNormal) {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index b82b43e..ac4689b 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -79,10 +79,10 @@
// of the time when the packet was received, and should be measured with
// the same tick rate as the RTP timestamp of the current payload.
// Returns 0 on success, -1 on failure.
- virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
- const uint8_t* payload,
- size_t length_bytes,
- uint32_t receive_timestamp) OVERRIDE;
+ int InsertPacket(const WebRtcRTPHeader& rtp_header,
+ const uint8_t* payload,
+ size_t length_bytes,
+ uint32_t receive_timestamp) override;
// Inserts a sync-packet into packet queue. Sync-packets are decoded to
// silence and are intended to keep AV-sync intact in an event of long packet
@@ -93,8 +93,8 @@
// type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
// can be implied by inserting a sync-packet.
// Returns kOk on success, kFail on failure.
- virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
- uint32_t receive_timestamp) OVERRIDE;
+ int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
+ uint32_t receive_timestamp) override;
// Instructs NetEq to deliver 10 ms of audio data. The data is written to
// |output_audio|, which can hold (at least) |max_length| elements.
@@ -104,97 +104,98 @@
// the samples are interleaved.
// The speech type is written to |type|, if |type| is not NULL.
// Returns kOK on success, or kFail in case of an error.
- virtual int GetAudio(size_t max_length, int16_t* output_audio,
- int* samples_per_channel, int* num_channels,
- NetEqOutputType* type) OVERRIDE;
+ int GetAudio(size_t max_length,
+ int16_t* output_audio,
+ int* samples_per_channel,
+ int* num_channels,
+ NetEqOutputType* type) override;
// Associates |rtp_payload_type| with |codec| and stores the information in
// the codec database. Returns kOK on success, kFail on failure.
- virtual int RegisterPayloadType(enum NetEqDecoder codec,
- uint8_t rtp_payload_type) OVERRIDE;
+ int RegisterPayloadType(enum NetEqDecoder codec,
+ uint8_t rtp_payload_type) override;
// Provides an externally created decoder object |decoder| to insert in the
// decoder database. The decoder implements a decoder of type |codec| and
// associates it with |rtp_payload_type|. Returns kOK on success, kFail on
// failure.
- virtual int RegisterExternalDecoder(AudioDecoder* decoder,
- enum NetEqDecoder codec,
- uint8_t rtp_payload_type) OVERRIDE;
+ int RegisterExternalDecoder(AudioDecoder* decoder,
+ enum NetEqDecoder codec,
+ uint8_t rtp_payload_type) override;
// Removes |rtp_payload_type| from the codec database. Returns 0 on success,
// -1 on failure.
- virtual int RemovePayloadType(uint8_t rtp_payload_type) OVERRIDE;
+ int RemovePayloadType(uint8_t rtp_payload_type) override;
- virtual bool SetMinimumDelay(int delay_ms) OVERRIDE;
+ bool SetMinimumDelay(int delay_ms) override;
- virtual bool SetMaximumDelay(int delay_ms) OVERRIDE;
+ bool SetMaximumDelay(int delay_ms) override;
- virtual int LeastRequiredDelayMs() const OVERRIDE;
+ int LeastRequiredDelayMs() const override;
- virtual int SetTargetDelay() OVERRIDE { return kNotImplemented; }
+ int SetTargetDelay() override { return kNotImplemented; }
- virtual int TargetDelay() OVERRIDE { return kNotImplemented; }
+ int TargetDelay() override { return kNotImplemented; }
- virtual int CurrentDelay() OVERRIDE { return kNotImplemented; }
+ int CurrentDelay() override { return kNotImplemented; }
// Sets the playout mode to |mode|.
// Deprecated.
// TODO(henrik.lundin) Delete.
- virtual void SetPlayoutMode(NetEqPlayoutMode mode) OVERRIDE;
+ void SetPlayoutMode(NetEqPlayoutMode mode) override;
// Returns the current playout mode.
// Deprecated.
// TODO(henrik.lundin) Delete.
- virtual NetEqPlayoutMode PlayoutMode() const OVERRIDE;
+ NetEqPlayoutMode PlayoutMode() const override;
// Writes the current network statistics to |stats|. The statistics are reset
// after the call.
- virtual int NetworkStatistics(NetEqNetworkStatistics* stats) OVERRIDE;
+ int NetworkStatistics(NetEqNetworkStatistics* stats) override;
// Writes the last packet waiting times (in ms) to |waiting_times|. The number
// of values written is no more than 100, but may be smaller if the interface
// is polled again before 100 packets has arrived.
- virtual void WaitingTimes(std::vector<int>* waiting_times) OVERRIDE;
+ void WaitingTimes(std::vector<int>* waiting_times) override;
// Writes the current RTCP statistics to |stats|. The statistics are reset
// and a new report period is started with the call.
- virtual void GetRtcpStatistics(RtcpStatistics* stats) OVERRIDE;
+ void GetRtcpStatistics(RtcpStatistics* stats) override;
// Same as RtcpStatistics(), but does not reset anything.
- virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats) OVERRIDE;
+ void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
// Enables post-decode VAD. When enabled, GetAudio() will return
// kOutputVADPassive when the signal contains no speech.
- virtual void EnableVad() OVERRIDE;
+ void EnableVad() override;
// Disables post-decode VAD.
- virtual void DisableVad() OVERRIDE;
+ void DisableVad() override;
- virtual bool GetPlayoutTimestamp(uint32_t* timestamp) OVERRIDE;
+ bool GetPlayoutTimestamp(uint32_t* timestamp) override;
- virtual int SetTargetNumberOfChannels() OVERRIDE { return kNotImplemented; }
+ int SetTargetNumberOfChannels() override { return kNotImplemented; }
- virtual int SetTargetSampleRate() OVERRIDE { return kNotImplemented; }
+ int SetTargetSampleRate() override { return kNotImplemented; }
// Returns the error code for the last occurred error. If no error has
// occurred, 0 is returned.
- virtual int LastError() const OVERRIDE;
+ int LastError() const override;
// Returns the error code last returned by a decoder (audio or comfort noise).
// When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
// this method to get the decoder's error code.
- virtual int LastDecoderError() OVERRIDE;
+ int LastDecoderError() override;
// Flushes both the packet buffer and the sync buffer.
- virtual void FlushBuffers() OVERRIDE;
+ void FlushBuffers() override;
- virtual void PacketBufferStatistics(int* current_num_packets,
- int* max_num_packets) const OVERRIDE;
+ void PacketBufferStatistics(int* current_num_packets,
+ int* max_num_packets) const override;
// Get sequence number and timestamp of the latest RTP.
// This method is to facilitate NACK.
- virtual int DecodedRtpInfo(int* sequence_number,
- uint32_t* timestamp) const OVERRIDE;
+ int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const override;
// This accessor method is only intended for testing purposes.
const SyncBuffer* sync_buffer_for_test() const;
diff --git a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h
index 9f09c94..1ac6c9a 100644
--- a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h
+++ b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h
@@ -15,9 +15,9 @@
class NETEQTEST_DummyRTPpacket : public NETEQTEST_RTPpacket {
public:
- virtual int readFromFile(FILE* fp) OVERRIDE;
- virtual int writeToFile(FILE* fp) OVERRIDE;
- virtual void parseHeader() OVERRIDE;
+ int readFromFile(FILE* fp) override;
+ int writeToFile(FILE* fp) override;
+ void parseHeader() override;
};
#endif // NETEQTEST_DUMMYRTPPACKET_H
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
index b672a0c..7abf5a1 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
@@ -88,8 +88,8 @@
class NetEqIsacQualityTest : public NetEqQualityTest {
protected:
NetEqIsacQualityTest();
- virtual void SetUp() OVERRIDE;
- virtual void TearDown() OVERRIDE;
+ void SetUp() override;
+ void TearDown() override;
virtual int EncodeBlock(int16_t* in_data, int block_size_samples,
uint8_t* payload, int max_bytes);
private:
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc
index e1f53af..55ef0c7 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc
@@ -116,8 +116,8 @@
class NetEqOpusFecQualityTest : public NetEqQualityTest {
protected:
NetEqOpusFecQualityTest();
- virtual void SetUp() OVERRIDE;
- virtual void TearDown() OVERRIDE;
+ void SetUp() override;
+ void TearDown() override;
virtual int EncodeBlock(int16_t* in_data, int block_size_samples,
uint8_t* payload, int max_bytes);
private:
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h b/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
index 173713a..b4a6a81 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -26,7 +26,7 @@
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;
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
index 474ec1c..b7b3ed1 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
@@ -47,7 +47,7 @@
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);
}
diff --git a/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
index fcf4f13..b780fbf 100644
--- a/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
@@ -33,7 +33,7 @@
// 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);
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
index 6207fde..cb40b1c 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -33,13 +33,13 @@
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 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 @@
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
diff --git a/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h b/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h
index 1d61280..ff30f67 100644
--- a/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h
+++ b/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -34,7 +34,7 @@
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;
}
diff --git a/webrtc/modules/audio_coding/neteq/tools/output_wav_file.h b/webrtc/modules/audio_coding/neteq/tools/output_wav_file.h
index 1f72162..1b1ed42 100644
--- a/webrtc/modules/audio_coding/neteq/tools/output_wav_file.h
+++ b/webrtc/modules/audio_coding/neteq/tools/output_wav_file.h
@@ -27,7 +27,7 @@
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;
}
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
index 70b5216..d711685 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -41,7 +41,7 @@
// 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;
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
index 2280436..e09f6e4 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -70,7 +70,7 @@
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_;