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/codecs/cng/audio_encoder_cng_unittest.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index c2a9ce3..03e7ba0 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -46,7 +46,7 @@
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 @@
// 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
diff --git a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
index 2f51676..1d3c2f3 100644
--- a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
@@ -44,22 +44,22 @@
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
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
index 903b220..3967c5e 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
@@ -30,21 +30,21 @@
: 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 @@
: 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 @@
: 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;
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
index 65a0d4b..229c06e 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
@@ -28,20 +28,20 @@
};
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.
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h
index 25c397a..fc3aa0d 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h
@@ -29,19 +29,19 @@
};
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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 38d5903..c918eaf 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -66,42 +66,42 @@
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
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
index d93fae3..bb76257 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -23,8 +23,8 @@
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,
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
index 93aca6c..417faf8 100644
--- a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
@@ -43,23 +43,23 @@
};
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_;
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
index ebe4a35..b39de49 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -21,8 +21,8 @@
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,
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h b/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h
index a5a040f..99ecd24 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h
@@ -28,9 +28,9 @@
: 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
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index beea1cf..39f1615 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -33,22 +33,22 @@
// 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_;
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index b3a76a5..ea3ad43 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -46,7 +46,7 @@
.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
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h b/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h
index 63c35e4..5e5ff9a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h
@@ -75,7 +75,7 @@
NumOutputChannels exptected_output_channels);
protected:
- void AfterGetAudio() OVERRIDE;
+ void AfterGetAudio() override;
const int output_freq_hz_1_;
const int output_freq_hz_2_;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
index 2eb1bf9..602c31a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -55,7 +55,7 @@
~AcmReceiverTest() {}
- virtual void SetUp() OVERRIDE {
+ void SetUp() override {
ASSERT_TRUE(receiver_.get() != NULL);
ASSERT_TRUE(acm_.get() != NULL);
for (int n = 0; n < ACMCodecDB::kNumCodecs; n++) {
@@ -72,8 +72,7 @@
rtp_header_.type.Audio.isCNG = false;
}
- virtual void TearDown() OVERRIDE {
- }
+ void TearDown() override {}
void InsertOnePacketOfSilence(int codec_id) {
CodecInst codec;
@@ -115,13 +114,12 @@
}
}
- virtual int32_t SendData(
- FrameType frame_type,
- uint8_t payload_type,
- uint32_t timestamp,
- const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) OVERRIDE {
+ int32_t SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_len_bytes,
+ const RTPFragmentationHeader* fragmentation) override {
if (frame_type == kFrameEmpty)
return 0;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
index e9096ff..aeacc06 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
@@ -54,7 +54,7 @@
~AcmReceiverTestOldApi() {}
- virtual void SetUp() OVERRIDE {
+ void SetUp() override {
ASSERT_TRUE(receiver_.get() != NULL);
ASSERT_TRUE(acm_.get() != NULL);
for (int n = 0; n < ACMCodecDB::kNumCodecs; n++) {
@@ -75,8 +75,7 @@
rtp_header_.type.Audio.isCNG = false;
}
- virtual void TearDown() OVERRIDE {
- }
+ void TearDown() override {}
void InsertOnePacketOfSilence(int codec_id) {
CodecInst codec;
@@ -115,13 +114,12 @@
}
}
- virtual int SendData(
- FrameType frame_type,
- uint8_t payload_type,
- uint32_t timestamp,
- const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) OVERRIDE {
+ int SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_len_bytes,
+ const RTPFragmentationHeader* fragmentation) override {
if (frame_type == kFrameEmpty)
return 0;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
index 769a327..4c4db5b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
@@ -41,16 +41,15 @@
// Returns the next encoded packet. Returns NULL if the test duration was
// exceeded. Ownership of the packet is handed over to the caller.
// Inherited from PacketSource.
- virtual Packet* NextPacket() OVERRIDE;
+ Packet* NextPacket() override;
// Inherited from AudioPacketizationCallback.
- virtual int32_t SendData(
- FrameType frame_type,
- uint8_t payload_type,
- uint32_t timestamp,
- const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) OVERRIDE;
+ int32_t SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_len_bytes,
+ const RTPFragmentationHeader* fragmentation) override;
private:
static const int kBlockSizeMs = 10;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
index 05a29df..52cb415 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
@@ -46,13 +46,12 @@
Packet* NextPacket();
// Inherited from AudioPacketizationCallback.
- virtual int32_t SendData(
- FrameType frame_type,
- uint8_t payload_type,
- uint32_t timestamp,
- const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) OVERRIDE;
+ int32_t SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_len_bytes,
+ const RTPFragmentationHeader* fragmentation) override;
AudioCodingModule* acm() { return acm_.get(); }
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 10423a7..ef69ba5 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -43,59 +43,58 @@
//
// Initialize send codec.
- virtual int InitializeSender() OVERRIDE;
+ int InitializeSender() override;
// Reset send codec.
- virtual int ResetEncoder() OVERRIDE;
+ int ResetEncoder() override;
// Can be called multiple times for Codec, CNG, RED.
- virtual int RegisterSendCodec(const CodecInst& send_codec) OVERRIDE;
+ int RegisterSendCodec(const CodecInst& send_codec) override;
// Get current send codec.
- virtual int SendCodec(CodecInst* current_codec) const OVERRIDE;
+ int SendCodec(CodecInst* current_codec) const override;
// Get current send frequency.
- virtual int SendFrequency() const OVERRIDE;
+ int SendFrequency() const override;
// Get encode bit-rate.
// Adaptive rate codecs return their current encode target rate, while other
// codecs return there long-term average or their fixed rate.
- virtual int SendBitrate() const OVERRIDE;
+ int SendBitrate() const override;
// Set available bandwidth, inform the encoder about the
// estimated bandwidth received from the remote party.
- virtual int SetReceivedEstimatedBandwidth(int bw) OVERRIDE;
+ int SetReceivedEstimatedBandwidth(int bw) override;
// Register a transport callback which will be
// called to deliver the encoded buffers.
- virtual int RegisterTransportCallback(
- AudioPacketizationCallback* transport) OVERRIDE;
+ int RegisterTransportCallback(AudioPacketizationCallback* transport) override;
// Add 10 ms of raw (PCM) audio data to the encoder.
- virtual int Add10MsData(const AudioFrame& audio_frame) OVERRIDE;
+ int Add10MsData(const AudioFrame& audio_frame) override;
/////////////////////////////////////////
// (RED) Redundant Coding
//
// Configure RED status i.e. on/off.
- virtual int SetREDStatus(bool enable_red) OVERRIDE;
+ int SetREDStatus(bool enable_red) override;
// Get RED status.
- virtual bool REDStatus() const OVERRIDE;
+ bool REDStatus() const override;
/////////////////////////////////////////
// (FEC) Forward Error Correction (codec internal)
//
// Configure FEC status i.e. on/off.
- virtual int SetCodecFEC(bool enabled_codec_fec) OVERRIDE;
+ int SetCodecFEC(bool enabled_codec_fec) override;
// Get FEC status.
- virtual bool CodecFEC() const OVERRIDE;
+ bool CodecFEC() const override;
// Set target packet loss rate
- virtual int SetPacketLossRate(int loss_rate) OVERRIDE;
+ int SetPacketLossRate(int loss_rate) override;
/////////////////////////////////////////
// (VAD) Voice Activity Detection
@@ -103,98 +102,97 @@
// (CNG) Comfort Noise Generation
//
- virtual int SetVAD(bool enable_dtx = true,
- bool enable_vad = false,
- ACMVADMode mode = VADNormal) OVERRIDE;
+ int SetVAD(bool enable_dtx = true,
+ bool enable_vad = false,
+ ACMVADMode mode = VADNormal) override;
- virtual int VAD(bool* dtx_enabled,
- bool* vad_enabled,
- ACMVADMode* mode) const OVERRIDE;
+ int VAD(bool* dtx_enabled,
+ bool* vad_enabled,
+ ACMVADMode* mode) const override;
- virtual int RegisterVADCallback(ACMVADCallback* vad_callback) OVERRIDE;
+ int RegisterVADCallback(ACMVADCallback* vad_callback) override;
/////////////////////////////////////////
// Receiver
//
// Initialize receiver, resets codec database etc.
- virtual int InitializeReceiver() OVERRIDE;
+ int InitializeReceiver() override;
// Reset the decoder state.
- virtual int ResetDecoder() OVERRIDE;
+ int ResetDecoder() override;
// Get current receive frequency.
- virtual int ReceiveFrequency() const OVERRIDE;
+ int ReceiveFrequency() const override;
// Get current playout frequency.
- virtual int PlayoutFrequency() const OVERRIDE;
+ int PlayoutFrequency() const override;
// Register possible receive codecs, can be called multiple times,
// for codecs, CNG, DTMF, RED.
- virtual int RegisterReceiveCodec(const CodecInst& receive_codec) OVERRIDE;
+ int RegisterReceiveCodec(const CodecInst& receive_codec) override;
// Get current received codec.
- virtual int ReceiveCodec(CodecInst* current_codec) const OVERRIDE;
+ int ReceiveCodec(CodecInst* current_codec) const override;
// Incoming packet from network parsed and ready for decode.
- virtual int IncomingPacket(const uint8_t* incoming_payload,
- const size_t payload_length,
- const WebRtcRTPHeader& rtp_info) OVERRIDE;
+ int IncomingPacket(const uint8_t* incoming_payload,
+ const size_t payload_length,
+ const WebRtcRTPHeader& rtp_info) override;
// Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
// One usage for this API is when pre-encoded files are pushed in ACM.
- virtual int IncomingPayload(const uint8_t* incoming_payload,
- const size_t payload_length,
- uint8_t payload_type,
- uint32_t timestamp) OVERRIDE;
+ int IncomingPayload(const uint8_t* incoming_payload,
+ const size_t payload_length,
+ uint8_t payload_type,
+ uint32_t timestamp) override;
// Minimum playout delay.
- virtual int SetMinimumPlayoutDelay(int time_ms) OVERRIDE;
+ int SetMinimumPlayoutDelay(int time_ms) override;
// Maximum playout delay.
- virtual int SetMaximumPlayoutDelay(int time_ms) OVERRIDE;
+ int SetMaximumPlayoutDelay(int time_ms) override;
// Smallest latency NetEq will maintain.
- virtual int LeastRequiredDelayMs() const OVERRIDE;
+ int LeastRequiredDelayMs() const override;
// Impose an initial delay on playout. ACM plays silence until |delay_ms|
// audio is accumulated in NetEq buffer, then starts decoding payloads.
- virtual int SetInitialPlayoutDelay(int delay_ms) OVERRIDE;
+ int SetInitialPlayoutDelay(int delay_ms) override;
// TODO(turajs): DTMF playout is always activated in NetEq these APIs should
// be removed, as well as all VoE related APIs and methods.
//
// Configure Dtmf playout status i.e on/off playout the incoming outband Dtmf
// tone.
- virtual int SetDtmfPlayoutStatus(bool enable) OVERRIDE { return 0; }
+ int SetDtmfPlayoutStatus(bool enable) override { return 0; }
// Get Dtmf playout status.
- virtual bool DtmfPlayoutStatus() const OVERRIDE { return true; }
+ bool DtmfPlayoutStatus() const override { return true; }
// Estimate the Bandwidth based on the incoming stream, needed
// for one way audio where the RTCP send the BW estimate.
// This is also done in the RTP module .
- virtual int DecoderEstimatedBandwidth() const OVERRIDE;
+ int DecoderEstimatedBandwidth() const override;
// Set playout mode voice, fax.
- virtual int SetPlayoutMode(AudioPlayoutMode mode) OVERRIDE;
+ int SetPlayoutMode(AudioPlayoutMode mode) override;
// Get playout mode voice, fax.
- virtual AudioPlayoutMode PlayoutMode() const OVERRIDE;
+ AudioPlayoutMode PlayoutMode() const override;
// Get playout timestamp.
- virtual int PlayoutTimestamp(uint32_t* timestamp) OVERRIDE;
+ int PlayoutTimestamp(uint32_t* timestamp) override;
// Get 10 milliseconds of raw audio data to play out, and
// automatic resample to the requested frequency if > 0.
- virtual int PlayoutData10Ms(int desired_freq_hz,
- AudioFrame* audio_frame) OVERRIDE;
+ int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override;
/////////////////////////////////////////
// Statistics
//
- virtual int GetNetworkStatistics(NetworkStatistics* statistics) OVERRIDE;
+ int GetNetworkStatistics(NetworkStatistics* statistics) override;
// GET RED payload for iSAC. The method id called when 'this' ACM is
// the default ACM.
@@ -204,40 +202,37 @@
uint8_t* payload,
int16_t* length_bytes);
- virtual int ReplaceInternalDTXWithWebRtc(bool use_webrtc_dtx) OVERRIDE;
+ int ReplaceInternalDTXWithWebRtc(bool use_webrtc_dtx) override;
- virtual int IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx) OVERRIDE;
+ int IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx) override;
- virtual int SetISACMaxRate(int max_bit_per_sec) OVERRIDE;
+ int SetISACMaxRate(int max_bit_per_sec) override;
- virtual int SetISACMaxPayloadSize(int max_size_bytes) OVERRIDE;
+ int SetISACMaxPayloadSize(int max_size_bytes) override;
- virtual int ConfigISACBandwidthEstimator(
- int frame_size_ms,
- int rate_bit_per_sec,
- bool enforce_frame_size = false) OVERRIDE;
+ int ConfigISACBandwidthEstimator(int frame_size_ms,
+ int rate_bit_per_sec,
+ bool enforce_frame_size = false) override;
int SetOpusApplication(OpusApplicationMode application) override;
// If current send codec is Opus, informs it about the maximum playback rate
// the receiver will render.
- virtual int SetOpusMaxPlaybackRate(int frequency_hz) OVERRIDE;
+ int SetOpusMaxPlaybackRate(int frequency_hz) override;
int EnableOpusDtx() override;
int DisableOpusDtx() override;
- virtual int UnregisterReceiveCodec(uint8_t payload_type) OVERRIDE;
+ int UnregisterReceiveCodec(uint8_t payload_type) override;
- virtual int EnableNack(size_t max_nack_list_size) OVERRIDE;
+ int EnableNack(size_t max_nack_list_size) override;
- virtual void DisableNack() OVERRIDE;
+ void DisableNack() override;
- virtual std::vector<uint16_t> GetNackList(
- int64_t round_trip_time_ms) const OVERRIDE;
+ std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
- virtual void GetDecodingCallStatistics(
- AudioDecodingCallStats* stats) const OVERRIDE;
+ void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override;
private:
struct InputData {
@@ -372,62 +367,57 @@
playout_frequency_hz_ = config.playout_frequency_hz;
}
- virtual ~AudioCodingImpl() OVERRIDE {};
+ ~AudioCodingImpl() override{};
- virtual bool RegisterSendCodec(AudioEncoder* send_codec) OVERRIDE;
+ bool RegisterSendCodec(AudioEncoder* send_codec) override;
- virtual bool RegisterSendCodec(int encoder_type,
- uint8_t payload_type,
- int frame_size_samples = 0) OVERRIDE;
+ bool RegisterSendCodec(int encoder_type,
+ uint8_t payload_type,
+ int frame_size_samples = 0) override;
- virtual const AudioEncoder* GetSenderInfo() const OVERRIDE;
+ const AudioEncoder* GetSenderInfo() const override;
- virtual const CodecInst* GetSenderCodecInst() OVERRIDE;
+ const CodecInst* GetSenderCodecInst() override;
- virtual int Add10MsAudio(const AudioFrame& audio_frame) OVERRIDE;
+ int Add10MsAudio(const AudioFrame& audio_frame) override;
- virtual const ReceiverInfo* GetReceiverInfo() const OVERRIDE;
+ const ReceiverInfo* GetReceiverInfo() const override;
- virtual bool RegisterReceiveCodec(AudioDecoder* receive_codec) OVERRIDE;
+ bool RegisterReceiveCodec(AudioDecoder* receive_codec) override;
- virtual bool RegisterReceiveCodec(int decoder_type,
- uint8_t payload_type) OVERRIDE;
+ bool RegisterReceiveCodec(int decoder_type, uint8_t payload_type) override;
- virtual bool InsertPacket(const uint8_t* incoming_payload,
- size_t payload_len_bytes,
- const WebRtcRTPHeader& rtp_info) OVERRIDE;
+ bool InsertPacket(const uint8_t* incoming_payload,
+ size_t payload_len_bytes,
+ const WebRtcRTPHeader& rtp_info) override;
- virtual bool InsertPayload(const uint8_t* incoming_payload,
- size_t payload_len_byte,
- uint8_t payload_type,
- uint32_t timestamp) OVERRIDE;
+ bool InsertPayload(const uint8_t* incoming_payload,
+ size_t payload_len_byte,
+ uint8_t payload_type,
+ uint32_t timestamp) override;
- virtual bool SetMinimumPlayoutDelay(int time_ms) OVERRIDE;
+ bool SetMinimumPlayoutDelay(int time_ms) override;
- virtual bool SetMaximumPlayoutDelay(int time_ms) OVERRIDE;
+ bool SetMaximumPlayoutDelay(int time_ms) override;
- virtual int LeastRequiredDelayMs() const OVERRIDE;
+ int LeastRequiredDelayMs() const override;
- virtual bool PlayoutTimestamp(uint32_t* timestamp) OVERRIDE;
+ bool PlayoutTimestamp(uint32_t* timestamp) override;
- virtual bool Get10MsAudio(AudioFrame* audio_frame) OVERRIDE;
+ bool Get10MsAudio(AudioFrame* audio_frame) override;
- virtual bool GetNetworkStatistics(
- NetworkStatistics* network_statistics) OVERRIDE;
+ bool GetNetworkStatistics(NetworkStatistics* network_statistics) override;
- virtual bool EnableNack(size_t max_nack_list_size) OVERRIDE;
+ bool EnableNack(size_t max_nack_list_size) override;
- virtual void DisableNack() OVERRIDE;
+ void DisableNack() override;
- virtual bool SetVad(bool enable_dtx,
- bool enable_vad,
- ACMVADMode vad_mode) OVERRIDE;
+ bool SetVad(bool enable_dtx, bool enable_vad, ACMVADMode vad_mode) override;
- virtual std::vector<uint16_t> GetNackList(
- int round_trip_time_ms) const OVERRIDE;
+ std::vector<uint16_t> GetNackList(int round_trip_time_ms) const override;
- virtual void GetDecodingCallStatistics(
- AudioDecodingCallStats* call_stats) const OVERRIDE;
+ void GetDecodingCallStatistics(
+ AudioDecodingCallStats* call_stats) const override;
private:
// Temporary method to be used during redesign phase.
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
index b37ef9d..8527767 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
@@ -81,13 +81,12 @@
: num_calls_(0),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
- virtual int32_t SendData(
- FrameType frame_type,
- uint8_t payload_type,
- uint32_t timestamp,
- const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) OVERRIDE {
+ int32_t SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_len_bytes,
+ const RTPFragmentationHeader* fragmentation) override {
CriticalSectionScoped lock(crit_sect_.get());
++num_calls_;
last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
@@ -124,9 +123,9 @@
~AudioCodingModuleTest() {}
- void TearDown() OVERRIDE {}
+ void TearDown() override {}
- void SetUp() OVERRIDE {
+ void SetUp() override {
rtp_utility_->Populate(&rtp_header_);
input_frame_.sample_rate_hz_ = kSampleRateHz;
@@ -308,7 +307,7 @@
config_.clock = fake_clock_.get();
}
- virtual void SetUp() OVERRIDE {
+ void SetUp() override {
AudioCodingModuleTest::SetUp();
CreateAcm();
StartThreads();
@@ -321,7 +320,7 @@
ASSERT_TRUE(pull_audio_thread_->Start(thread_id));
}
- virtual void TearDown() OVERRIDE {
+ void TearDown() override {
AudioCodingModuleTest::TearDown();
pull_audio_thread_->Stop();
send_thread_->Stop();
@@ -436,7 +435,7 @@
~AcmIsacMtTest() {}
- virtual void SetUp() OVERRIDE {
+ void SetUp() override {
AudioCodingModuleTest::SetUp();
CreateAcm();
@@ -459,7 +458,7 @@
StartThreads();
}
- virtual void RegisterCodec() OVERRIDE {
+ void RegisterCodec() override {
static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
@@ -469,7 +468,7 @@
acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, kPayloadType));
}
- virtual void InsertPacket() OVERRIDE {
+ void InsertPacket() override {
int num_calls = packet_cb_.num_calls(); // Store locally for thread safety.
if (num_calls > last_packet_number_) {
// Get the new payload out from the callback handler.
@@ -486,14 +485,14 @@
&last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
}
- virtual void InsertAudio() OVERRIDE {
+ void InsertAudio() override {
memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms);
AudioCodingModuleTest::InsertAudio();
}
// This method is the same as AudioCodingModuleMtTest::TestDone(), but here
// it is using the constants defined in this class (i.e., shorter test run).
- virtual bool TestDone() OVERRIDE {
+ bool TestDone() override {
if (packet_cb_.num_calls() > kNumPackets) {
CriticalSectionScoped lock(crit_sect_.get());
if (pull_audio_count_ > kNumPullCalls) {
@@ -708,7 +707,7 @@
// Returns a pointer to the next packet. Returns NULL if the source is
// depleted (i.e., the test duration is exceeded), or if an error occurred.
// Inherited from test::PacketSource.
- virtual test::Packet* NextPacket() OVERRIDE {
+ test::Packet* NextPacket() override {
// Get the next packet from AcmSendTest. Ownership of |packet| is
// transferred to this method.
test::Packet* packet = send_test_->NextPacket();
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index 122d027..14a73fd 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -86,13 +86,12 @@
last_payload_type_(-1),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
- virtual int32_t SendData(
- FrameType frame_type,
- uint8_t payload_type,
- uint32_t timestamp,
- const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) OVERRIDE {
+ int32_t SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_len_bytes,
+ const RTPFragmentationHeader* fragmentation) override {
CriticalSectionScoped lock(crit_sect_.get());
++num_calls_;
last_frame_type_ = frame_type;
@@ -855,7 +854,7 @@
// Returns a pointer to the next packet. Returns NULL if the source is
// depleted (i.e., the test duration is exceeded), or if an error occurred.
// Inherited from test::PacketSource.
- test::Packet* NextPacket() OVERRIDE {
+ test::Packet* NextPacket() override {
// Get the next packet from AcmSendTest. Ownership of |packet| is
// transferred to this method.
test::Packet* packet = send_test_->NextPacket();
@@ -1185,7 +1184,7 @@
}
// Inherited from test::PacketSource.
- test::Packet* NextPacket() OVERRIDE {
+ test::Packet* NextPacket() override {
// Check if it is time to terminate the test. The packet source is of type
// ConstantPcmPacketSource, which is infinite, so we must end the test
// "manually".
diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/main/test/Channel.h
index 55167d3..c4ad7d1 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.h
+++ b/webrtc/modules/audio_coding/main/test/Channel.h
@@ -50,13 +50,12 @@
Channel(int16_t chID = -1);
~Channel();
- virtual int32_t SendData(
- FrameType frameType,
- uint8_t payloadType,
- uint32_t timeStamp,
- const uint8_t* payloadData,
- size_t payloadSize,
- const RTPFragmentationHeader* fragmentation) OVERRIDE;
+ int32_t SendData(FrameType frameType,
+ uint8_t payloadType,
+ uint32_t timeStamp,
+ const uint8_t* payloadData,
+ size_t payloadSize,
+ const RTPFragmentationHeader* fragmentation) override;
void RegisterReceiverACM(AudioCodingModule *acm);
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;
diff --git a/webrtc/modules/audio_coding/main/test/PacketLossTest.h b/webrtc/modules/audio_coding/main/test/PacketLossTest.h
index 70fa9ff..d25dea2 100644
--- a/webrtc/modules/audio_coding/main/test/PacketLossTest.h
+++ b/webrtc/modules/audio_coding/main/test/PacketLossTest.h
@@ -23,7 +23,8 @@
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string out_file_name, int channels, int loss_rate,
int burst_length);
- bool IncomingPacket() OVERRIDE;
+ bool IncomingPacket() override;
+
protected:
bool PacketLost();
int loss_rate_;
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/main/test/RTPFile.h
index 9a2d43a..346440b 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.h
+++ b/webrtc/modules/audio_coding/main/test/RTPFile.h
@@ -65,14 +65,19 @@
~RTPBuffer();
- virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
- const int16_t seqNo, const uint8_t* payloadData,
- const size_t payloadSize, uint32_t frequency) OVERRIDE;
+ void Write(const uint8_t payloadType,
+ const uint32_t timeStamp,
+ const int16_t seqNo,
+ const uint8_t* payloadData,
+ const size_t payloadSize,
+ uint32_t frequency) override;
- virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
- size_t payloadSize, uint32_t* offset) OVERRIDE;
+ size_t Read(WebRtcRTPHeader* rtpInfo,
+ uint8_t* payloadData,
+ size_t payloadSize,
+ uint32_t* offset) override;
- virtual bool EndOfFile() const OVERRIDE;
+ bool EndOfFile() const override;
private:
RWLockWrapper* _queueRWLock;
@@ -97,16 +102,19 @@
void ReadHeader();
- virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
- const int16_t seqNo, const uint8_t* payloadData,
- const size_t payloadSize, uint32_t frequency) OVERRIDE;
+ void Write(const uint8_t payloadType,
+ const uint32_t timeStamp,
+ const int16_t seqNo,
+ const uint8_t* payloadData,
+ const size_t payloadSize,
+ uint32_t frequency) override;
- virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
- size_t payloadSize, uint32_t* offset) OVERRIDE;
+ size_t Read(WebRtcRTPHeader* rtpInfo,
+ uint8_t* payloadData,
+ size_t payloadSize,
+ uint32_t* offset) override;
- virtual bool EndOfFile() const OVERRIDE {
- return _rtpEOF;
- }
+ bool EndOfFile() const override { return _rtpEOF; }
private:
FILE* _rtpFile;
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
index 4292d77..1cdc0cb 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
@@ -28,13 +28,12 @@
void RegisterReceiverACM(AudioCodingModule* acm);
- virtual int32_t SendData(
- FrameType frame_type,
- uint8_t payload_type,
- uint32_t timestamp,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation) OVERRIDE;
+ int32_t SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_size,
+ const RTPFragmentationHeader* fragmentation) override;
size_t payload_size();
uint32_t timestamp_diff();
@@ -55,7 +54,7 @@
explicit TestAllCodecs(int test_mode);
~TestAllCodecs();
- virtual void Perform() OVERRIDE;
+ void Perform() override;
private:
// The default value of '-1' indicates that the registration is based only on
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.h b/webrtc/modules/audio_coding/main/test/TestStereo.h
index 89914cc..c6412c7 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.h
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.h
@@ -35,13 +35,12 @@
void RegisterReceiverACM(AudioCodingModule* acm);
- virtual int32_t SendData(
- const FrameType frame_type,
- const uint8_t payload_type,
- const uint32_t timestamp,
- const uint8_t* payload_data,
- const size_t payload_size,
- const RTPFragmentationHeader* fragmentation) OVERRIDE;
+ int32_t SendData(const FrameType frame_type,
+ const uint8_t payload_type,
+ const uint32_t timestamp,
+ const uint8_t* payload_data,
+ const size_t payload_size,
+ const RTPFragmentationHeader* fragmentation) override;
uint16_t payload_size();
uint32_t timestamp_diff();
@@ -66,7 +65,8 @@
explicit TestStereo(int test_mode);
~TestStereo();
- virtual void Perform() OVERRIDE;
+ void Perform() override;
+
private:
// The default value of '-1' indicates that the registration is based only on
// codec name and a sampling frequncy matching is not required. This is useful
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_;