Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/
Bug: webrtc:13555, webrtc:13082
Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35771}
diff --git a/modules/audio_coding/neteq/tools/audio_checksum.h b/modules/audio_coding/neteq/tools/audio_checksum.h
index e4306fa..9d6f343 100644
--- a/modules/audio_coding/neteq/tools/audio_checksum.h
+++ b/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -16,7 +16,6 @@
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/message_digest.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/system/arch.h"
@@ -31,6 +30,9 @@
checksum_result_(checksum_->Size()),
finished_(false) {}
+ AudioChecksum(const AudioChecksum&) = delete;
+ AudioChecksum& operator=(const AudioChecksum&) = delete;
+
bool WriteArray(const int16_t* audio, size_t num_samples) override {
if (finished_)
return false;
@@ -56,8 +58,6 @@
std::unique_ptr<rtc::MessageDigest> checksum_;
rtc::Buffer checksum_result_;
bool finished_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/audio_loop.h b/modules/audio_coding/neteq/tools/audio_loop.h
index 25da463..a73be2d 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/modules/audio_coding/neteq/tools/audio_loop.h
@@ -15,7 +15,6 @@
#include <string>
#include "api/array_view.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@@ -29,6 +28,9 @@
virtual ~AudioLoop() {}
+ AudioLoop(const AudioLoop&) = delete;
+ AudioLoop& operator=(const AudioLoop&) = delete;
+
// Initializes the AudioLoop by reading from `file_name`. The loop will be no
// longer than `max_loop_length_samples`, if the length of the file is
// greater. Otherwise, the loop length is the same as the file length.
@@ -47,8 +49,6 @@
size_t loop_length_samples_;
size_t block_length_samples_;
std::unique_ptr<int16_t[]> audio_array_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(AudioLoop);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/audio_sink.h b/modules/audio_coding/neteq/tools/audio_sink.h
index cd6733b..53729fa 100644
--- a/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/modules/audio_coding/neteq/tools/audio_sink.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
#include "api/audio/audio_frame.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@@ -24,6 +23,9 @@
AudioSink() {}
virtual ~AudioSink() {}
+ AudioSink(const AudioSink&) = delete;
+ AudioSink& operator=(const AudioSink&) = delete;
+
// Writes `num_samples` from `audio` to the AudioSink. Returns true if
// successful, otherwise false.
virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
@@ -34,9 +36,6 @@
return WriteArray(audio_frame.data(), audio_frame.samples_per_channel_ *
audio_frame.num_channels_);
}
-
- private:
- RTC_DISALLOW_COPY_AND_ASSIGN(AudioSink);
};
// Forks the output audio to two AudioSink objects.
@@ -45,23 +44,25 @@
AudioSinkFork(AudioSink* left, AudioSink* right)
: left_sink_(left), right_sink_(right) {}
+ AudioSinkFork(const AudioSinkFork&) = delete;
+ AudioSinkFork& operator=(const AudioSinkFork&) = delete;
+
bool WriteArray(const int16_t* audio, size_t num_samples) override;
private:
AudioSink* left_sink_;
AudioSink* right_sink_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
};
// An AudioSink implementation that does nothing.
class VoidAudioSink : public AudioSink {
public:
VoidAudioSink() = default;
- bool WriteArray(const int16_t* audio, size_t num_samples) override;
- private:
- RTC_DISALLOW_COPY_AND_ASSIGN(VoidAudioSink);
+ VoidAudioSink(const VoidAudioSink&) = delete;
+ VoidAudioSink& operator=(const VoidAudioSink&) = delete;
+
+ bool WriteArray(const int16_t* audio, size_t num_samples) override;
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
index 6a79ce4..ab4f5c2 100644
--- a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
+++ b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
@@ -16,7 +16,6 @@
#include <string>
#include "modules/audio_coding/neteq/tools/packet_source.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@@ -31,6 +30,9 @@
int sample_rate_hz,
int payload_type);
+ ConstantPcmPacketSource(const ConstantPcmPacketSource&) = delete;
+ ConstantPcmPacketSource& operator=(const ConstantPcmPacketSource&) = delete;
+
std::unique_ptr<Packet> NextPacket() override;
private:
@@ -46,8 +48,6 @@
uint16_t seq_number_;
uint32_t timestamp_;
const uint32_t payload_ssrc_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(ConstantPcmPacketSource);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/input_audio_file.h b/modules/audio_coding/neteq/tools/input_audio_file.h
index 010d8cc..c6e65a0 100644
--- a/modules/audio_coding/neteq/tools/input_audio_file.h
+++ b/modules/audio_coding/neteq/tools/input_audio_file.h
@@ -15,8 +15,6 @@
#include <string>
-#include "rtc_base/constructor_magic.h"
-
namespace webrtc {
namespace test {
@@ -27,6 +25,9 @@
virtual ~InputAudioFile();
+ InputAudioFile(const InputAudioFile&) = delete;
+ InputAudioFile& operator=(const InputAudioFile&) = delete;
+
// Reads `samples` elements from source file to `destination`. Returns true
// if the read was successful, otherwise false. If the file end is reached,
// the file is rewound and reading continues from the beginning.
@@ -52,7 +53,6 @@
private:
FILE* fp_;
const bool loop_at_end_;
- RTC_DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/output_audio_file.h b/modules/audio_coding/neteq/tools/output_audio_file.h
index ad97722..491cbd0 100644
--- a/modules/audio_coding/neteq/tools/output_audio_file.h
+++ b/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -16,7 +16,6 @@
#include <string>
#include "modules/audio_coding/neteq/tools/audio_sink.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@@ -34,6 +33,9 @@
fclose(out_file_);
}
+ OutputAudioFile(const OutputAudioFile&) = delete;
+ OutputAudioFile& operator=(const OutputAudioFile&) = delete;
+
bool WriteArray(const int16_t* audio, size_t num_samples) override {
RTC_DCHECK(out_file_);
return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
@@ -41,8 +43,6 @@
private:
FILE* out_file_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/output_wav_file.h b/modules/audio_coding/neteq/tools/output_wav_file.h
index ae2e970..1485f4e 100644
--- a/modules/audio_coding/neteq/tools/output_wav_file.h
+++ b/modules/audio_coding/neteq/tools/output_wav_file.h
@@ -15,7 +15,6 @@
#include "common_audio/wav_file.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@@ -29,6 +28,9 @@
int num_channels = 1)
: wav_writer_(file_name, sample_rate_hz, num_channels) {}
+ OutputWavFile(const OutputWavFile&) = delete;
+ OutputWavFile& operator=(const OutputWavFile&) = delete;
+
bool WriteArray(const int16_t* audio, size_t num_samples) override {
wav_writer_.WriteSamples(audio, num_samples);
return true;
@@ -36,8 +38,6 @@
private:
WavWriter wav_writer_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h
index 92e5ee9..9671090 100644
--- a/modules/audio_coding/neteq/tools/packet.h
+++ b/modules/audio_coding/neteq/tools/packet.h
@@ -16,7 +16,6 @@
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "rtc_base/constructor_magic.h"
#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
@@ -54,6 +53,9 @@
virtual ~Packet();
+ Packet(const Packet&) = delete;
+ Packet& operator=(const Packet&) = delete;
+
// Parses the first bytes of the RTP payload, interpreting them as RED headers
// according to RFC 2198. The headers will be inserted into `headers`. The
// caller of the method assumes ownership of the objects in the list, and
@@ -95,8 +97,6 @@
size_t virtual_payload_length_bytes_ = 0;
const double time_ms_; // Used to denote a packet's arrival time.
const bool valid_header_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(Packet);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/packet_source.h b/modules/audio_coding/neteq/tools/packet_source.h
index 975680f..be1705c 100644
--- a/modules/audio_coding/neteq/tools/packet_source.h
+++ b/modules/audio_coding/neteq/tools/packet_source.h
@@ -15,7 +15,6 @@
#include <memory>
#include "modules/audio_coding/neteq/tools/packet.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@@ -26,6 +25,9 @@
PacketSource();
virtual ~PacketSource();
+ PacketSource(const PacketSource&) = delete;
+ PacketSource& operator=(const PacketSource&) = delete;
+
// Returns next packet. Returns nullptr if the source is depleted, or if an
// error occurred.
virtual std::unique_ptr<Packet> NextPacket() = 0;
@@ -34,9 +36,6 @@
protected:
std::bitset<128> filter_; // Payload type is 7 bits in the RFC.
-
- private:
- RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/resample_input_audio_file.h b/modules/audio_coding/neteq/tools/resample_input_audio_file.h
index 9106d5b..497a410 100644
--- a/modules/audio_coding/neteq/tools/resample_input_audio_file.h
+++ b/modules/audio_coding/neteq/tools/resample_input_audio_file.h
@@ -15,7 +15,6 @@
#include "common_audio/resampler/include/resampler.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@@ -37,6 +36,9 @@
file_rate_hz_(file_rate_hz),
output_rate_hz_(output_rate_hz) {}
+ ResampleInputAudioFile(const ResampleInputAudioFile&) = delete;
+ ResampleInputAudioFile& operator=(const ResampleInputAudioFile&) = delete;
+
bool Read(size_t samples, int output_rate_hz, int16_t* destination);
bool Read(size_t samples, int16_t* destination) override;
void set_output_rate_hz(int rate_hz);
@@ -45,7 +47,6 @@
const int file_rate_hz_;
int output_rate_hz_;
Resampler resampler_;
- RTC_DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/modules/audio_coding/neteq/tools/rtc_event_log_source.h
index d4be2a7..e2d0f61 100644
--- a/modules/audio_coding/neteq/tools/rtc_event_log_source.h
+++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -19,7 +19,6 @@
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
@@ -43,6 +42,9 @@
virtual ~RtcEventLogSource();
+ RtcEventLogSource(const RtcEventLogSource&) = delete;
+ RtcEventLogSource& operator=(const RtcEventLogSource&) = delete;
+
std::unique_ptr<Packet> NextPacket() override;
// Returns the timestamp of the next audio output event, in milliseconds. The
@@ -60,8 +62,6 @@
size_t rtp_packet_index_ = 0;
std::vector<int64_t> audio_outputs_;
size_t audio_output_index_ = 0;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h
index d6aab24..7e284ac 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -19,7 +19,6 @@
#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
@@ -41,6 +40,9 @@
~RtpFileSource() override;
+ RtpFileSource(const RtpFileSource&) = delete;
+ RtpFileSource& operator=(const RtpFileSource&) = delete;
+
// Registers an RTP header extension and binds it to `id`.
virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
@@ -58,8 +60,6 @@
std::unique_ptr<RtpFileReader> rtp_reader_;
const absl::optional<uint32_t> ssrc_filter_;
RtpHeaderExtensionMap rtp_header_extension_map_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/rtp_generator.h b/modules/audio_coding/neteq/tools/rtp_generator.h
index 6ca6e1b..2e615ad 100644
--- a/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
#include "api/rtp_headers.h"
-#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@@ -34,6 +33,9 @@
virtual ~RtpGenerator() {}
+ RtpGenerator(const RtpGenerator&) = delete;
+ RtpGenerator& operator=(const RtpGenerator&) = delete;
+
// Writes the next RTP header to `rtp_header`, which will be of type
// `payload_type`. Returns the send time for this packet (in ms). The value of
// `payload_length_samples` determines the send time for the next packet.
@@ -50,9 +52,6 @@
const uint32_t ssrc_;
const int samples_per_ms_;
double drift_factor_;
-
- private:
- RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
};
class TimestampJumpRtpGenerator : public RtpGenerator {
@@ -66,6 +65,10 @@
jump_from_timestamp_(jump_from_timestamp),
jump_to_timestamp_(jump_to_timestamp) {}
+ TimestampJumpRtpGenerator(const TimestampJumpRtpGenerator&) = delete;
+ TimestampJumpRtpGenerator& operator=(const TimestampJumpRtpGenerator&) =
+ delete;
+
uint32_t GetRtpHeader(uint8_t payload_type,
size_t payload_length_samples,
RTPHeader* rtp_header) override;
@@ -73,7 +76,6 @@
private:
uint32_t jump_from_timestamp_;
uint32_t jump_to_timestamp_;
- RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
};
} // namespace test