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Add RTC_ prefix to contructormagic macros.

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We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

Related CL: https://codereview.webrtc.org/1335923002/

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1345433002

Cr-Commit-Position: refs/heads/master@{#9953}
This commit is contained in:
henrikg
2015-09-16 05:37:44 -07:00
committed by Commit bot
parent afb6b5e3e0
commit 3c089d751e
258 changed files with 411 additions and 418 deletions
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2
webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
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@ -51,7 +51,7 @@ class AudioChecksum : public AudioSink {
char checksum_result_[rtc::Md5Digest::kSize];
bool finished_;
DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
RTC_DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/audio_loop.h
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@ -51,7 +51,7 @@ class AudioLoop {
size_t block_length_samples_;
rtc::scoped_ptr<int16_t[]> audio_array_;
DISALLOW_COPY_AND_ASSIGN(AudioLoop);
RTC_DISALLOW_COPY_AND_ASSIGN(AudioLoop);
};
} // namespace test

4
webrtc/modules/audio_coding/neteq/tools/audio_sink.h
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@ -38,7 +38,7 @@ class AudioSink {
}
private:
DISALLOW_COPY_AND_ASSIGN(AudioSink);
RTC_DISALLOW_COPY_AND_ASSIGN(AudioSink);
};
// Forks the output audio to two AudioSink objects.
@ -56,7 +56,7 @@ class AudioSinkFork : public AudioSink {
AudioSink* left_sink_;
AudioSink* right_sink_;
DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
RTC_DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
};
} // namespace test
} // namespace webrtc

2
webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
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@ -49,7 +49,7 @@ class ConstantPcmPacketSource : public PacketSource {
uint32_t timestamp_;
const uint32_t payload_ssrc_;
DISALLOW_COPY_AND_ASSIGN(ConstantPcmPacketSource);
RTC_DISALLOW_COPY_AND_ASSIGN(ConstantPcmPacketSource);
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/input_audio_file.h
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@ -50,7 +50,7 @@ class InputAudioFile {
private:
FILE* fp_;
DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
RTC_DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/output_audio_file.h
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@ -42,7 +42,7 @@ class OutputAudioFile : public AudioSink {
private:
FILE* out_file_;
DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
RTC_DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/output_wav_file.h
View File

@ -35,7 +35,7 @@ class OutputWavFile : public AudioSink {
private:
WavWriter wav_writer_;
DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
RTC_DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/packet.h
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@ -114,7 +114,7 @@ class Packet {
double time_ms_; // Used to denote a packet's arrival time.
bool valid_header_; // Set by the RtpHeaderParser.
DISALLOW_COPY_AND_ASSIGN(Packet);
RTC_DISALLOW_COPY_AND_ASSIGN(Packet);
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/packet_source.h
View File

@ -46,7 +46,7 @@ class PacketSource {
uint32_t ssrc_; // The selected SSRC. All other SSRCs will be discarded.
private:
DISALLOW_COPY_AND_ASSIGN(PacketSource);
RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource);
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
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@ -43,7 +43,7 @@ class ResampleInputAudioFile : public InputAudioFile {
const int file_rate_hz_;
int output_rate_hz_;
Resampler resampler_;
DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
RTC_DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
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@ -61,7 +61,7 @@ class RtcEventLogSource : public PacketSource {
rtc::scoped_ptr<rtclog::EventStream> event_log_;
rtc::scoped_ptr<RtpHeaderParser> parser_;
DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
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@ -55,7 +55,7 @@ class RtpFileSource : public PacketSource {
rtc::scoped_ptr<RtpFileReader> rtp_reader_;
rtc::scoped_ptr<RtpHeaderParser> parser_;
DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
};
} // namespace test

4
webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
View File

@ -54,7 +54,7 @@ class RtpGenerator {
double drift_factor_;
private:
DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
};
class TimestampJumpRtpGenerator : public RtpGenerator {
@ -75,7 +75,7 @@ class TimestampJumpRtpGenerator : public RtpGenerator {
private:
uint32_t jump_from_timestamp_;
uint32_t jump_to_timestamp_;
DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
};
} // namespace test