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Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/

Browse Source 
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}
This commit is contained in:
Byoungchan Lee
2022-01-21 09:49:39 +09:00
committed by WebRTC LUCI CQ
parent ce6170fcdf
commit 604fd2f1ab
127 changed files with 466 additions and 393 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
modules/audio_coding/neteq/accelerate.h
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@ -15,7 +15,6 @@
#include <stdint.h>
#include "modules/audio_coding/neteq/time_stretch.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -33,6 +32,9 @@ class Accelerate : public TimeStretch {
const BackgroundNoise& background_noise)
: TimeStretch(sample_rate_hz, num_channels, background_noise) {}
Accelerate(const Accelerate&) = delete;
Accelerate& operator=(const Accelerate&) = delete;
// This method performs the actual Accelerate operation. The samples are
// read from `input`, of length `input_length` elements, and are written to
// `output`. The number of samples removed through time-stretching is
@ -62,9 +64,6 @@ class Accelerate : public TimeStretch {
bool active_speech,
bool fast_mode,
AudioMultiVector* output) const override;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(Accelerate);
};
struct AccelerateFactory {

7
modules/audio_coding/neteq/audio_multi_vector.h
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@ -18,7 +18,6 @@
#include "api/array_view.h"
#include "modules/audio_coding/neteq/audio_vector.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -34,6 +33,9 @@ class AudioMultiVector {
virtual ~AudioMultiVector();
AudioMultiVector(const AudioMultiVector&) = delete;
AudioMultiVector& operator=(const AudioMultiVector&) = delete;
// Deletes all values and make the vector empty.
virtual void Clear();
@ -130,9 +132,6 @@ class AudioMultiVector {
protected:
std::vector<AudioVector*> channels_;
size_t num_channels_;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioMultiVector);
};
} // namespace webrtc

6
modules/audio_coding/neteq/audio_vector.h
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@ -17,7 +17,6 @@
#include <memory>
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -31,6 +30,9 @@ class AudioVector {
virtual ~AudioVector();
AudioVector(const AudioVector&) = delete;
AudioVector& operator=(const AudioVector&) = delete;
// Deletes all values and make the vector empty.
virtual void Clear();
@ -164,8 +166,6 @@ class AudioVector {
// The index of the sample after the last sample in `array_`.
size_t end_index_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioVector);
};
} // namespace webrtc

6
modules/audio_coding/neteq/background_noise.h
View File

@ -16,7 +16,6 @@
#include <memory>
#include "api/array_view.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -34,6 +33,9 @@ class BackgroundNoise {
explicit BackgroundNoise(size_t num_channels);
virtual ~BackgroundNoise();
BackgroundNoise(const BackgroundNoise&) = delete;
BackgroundNoise& operator=(const BackgroundNoise&) = delete;
void Reset();
// Updates the parameter estimates based on the signal currently in the
@ -130,8 +132,6 @@ class BackgroundNoise {
size_t num_channels_;
std::unique_ptr<ChannelParameters[]> channel_parameters_;
bool initialized_;
RTC_DISALLOW_COPY_AND_ASSIGN(BackgroundNoise);
};
} // namespace webrtc

8
modules/audio_coding/neteq/buffer_level_filter.h
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@ -14,14 +14,16 @@
#include <stddef.h>
#include <stdint.h>
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class BufferLevelFilter {
public:
BufferLevelFilter();
virtual ~BufferLevelFilter() {}
BufferLevelFilter(const BufferLevelFilter&) = delete;
BufferLevelFilter& operator=(const BufferLevelFilter&) = delete;
virtual void Reset();
// Updates the filter. Current buffer size is `buffer_size_samples`.
@ -46,8 +48,6 @@ class BufferLevelFilter {
private:
int level_factor_; // Filter factor for the buffer level filter in Q8.
int filtered_current_level_; // Filtered current buffer level in Q8.
RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
};
} // namespace webrtc

6
modules/audio_coding/neteq/comfort_noise.h
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@ -13,8 +13,6 @@
#include <stddef.h>
#include "rtc_base/constructor_magic.h"
namespace webrtc {
// Forward declarations.
@ -42,6 +40,9 @@ class ComfortNoise {
decoder_database_(decoder_database),
sync_buffer_(sync_buffer) {}
ComfortNoise(const ComfortNoise&) = delete;
ComfortNoise& operator=(const ComfortNoise&) = delete;
// Resets the state. Should be called before each new comfort noise period.
void Reset();
@ -65,7 +66,6 @@ class ComfortNoise {
DecoderDatabase* decoder_database_;
SyncBuffer* sync_buffer_;
int internal_error_code_;
RTC_DISALLOW_COPY_AND_ASSIGN(ComfortNoise);
};
} // namespace webrtc

6
modules/audio_coding/neteq/decision_logic.h
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@ -18,7 +18,6 @@
#include "api/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/buffer_level_filter.h"
#include "modules/audio_coding/neteq/delay_manager.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/experiments/field_trial_parser.h"
namespace webrtc {
@ -37,6 +36,9 @@ class DecisionLogic : public NetEqController {
~DecisionLogic() override;
DecisionLogic(const DecisionLogic&) = delete;
DecisionLogic& operator=(const DecisionLogic&) = delete;
// Resets object to a clean state.
void Reset() override;
@ -192,8 +194,6 @@ class DecisionLogic : public NetEqController {
FieldTrialParameter<bool> estimate_dtx_delay_;
FieldTrialParameter<bool> time_stretch_cn_;
FieldTrialConstrained<int> target_level_window_ms_;
RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
};
} // namespace webrtc

6
modules/audio_coding/neteq/decoder_database.h
View File

@ -20,7 +20,6 @@
#include "api/scoped_refptr.h"
#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
#include "modules/audio_coding/neteq/packet.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -122,6 +121,9 @@ class DecoderDatabase {
virtual ~DecoderDatabase();
DecoderDatabase(const DecoderDatabase&) = delete;
DecoderDatabase& operator=(const DecoderDatabase&) = delete;
// Returns true if the database is empty.
virtual bool Empty() const;
@ -208,8 +210,6 @@ class DecoderDatabase {
mutable std::unique_ptr<ComfortNoiseDecoder> active_cng_decoder_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
const absl::optional<AudioCodecPairId> codec_pair_id_;
RTC_DISALLOW_COPY_AND_ASSIGN(DecoderDatabase);
};
} // namespace webrtc

8
modules/audio_coding/neteq/delay_manager.h
View File

@ -22,7 +22,6 @@
#include "modules/audio_coding/neteq/relative_arrival_delay_tracker.h"
#include "modules/audio_coding/neteq/reorder_optimizer.h"
#include "modules/audio_coding/neteq/underrun_optimizer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -52,6 +51,9 @@ class DelayManager {
virtual ~DelayManager();
DelayManager(const DelayManager&) = delete;
DelayManager& operator=(const DelayManager&) = delete;
// Updates the delay manager with a new incoming packet, with `timestamp` from
// the RTP header. This updates the statistics and a new target buffer level
// is calculated. Returns the relative delay if it can be calculated. If
@ -111,9 +113,7 @@ class DelayManager {
int maximum_delay_ms_; // Externally set maximum allowed delay.
int packet_len_ms_ = 0;
int target_level_ms_; // Currently preferred buffer level.
RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager);
int target_level_ms_; // Currently preferred buffer level.
};
} // namespace webrtc

6
modules/audio_coding/neteq/dsp_helper.h
View File

@ -16,7 +16,6 @@
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/audio_vector.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -150,11 +149,12 @@ class DspHelper {
bool compensate_delay,
int16_t* output);
DspHelper(const DspHelper&) = delete;
DspHelper& operator=(const DspHelper&) = delete;
private:
// Table of constants used in method DspHelper::ParabolicFit().
static const int16_t kParabolaCoefficients[17][3];
RTC_DISALLOW_COPY_AND_ASSIGN(DspHelper);
};
} // namespace webrtc

7
modules/audio_coding/neteq/dtmf_buffer.h
View File

@ -16,8 +16,6 @@
#include <list>
#include "rtc_base/constructor_magic.h"
namespace webrtc {
struct DtmfEvent {
@ -50,6 +48,9 @@ class DtmfBuffer {
virtual ~DtmfBuffer();
DtmfBuffer(const DtmfBuffer&) = delete;
DtmfBuffer& operator=(const DtmfBuffer&) = delete;
// Flushes the buffer.
virtual void Flush();
@ -97,8 +98,6 @@ class DtmfBuffer {
static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b);
DtmfList buffer_;
RTC_DISALLOW_COPY_AND_ASSIGN(DtmfBuffer);
};
} // namespace webrtc

7
modules/audio_coding/neteq/dtmf_tone_generator.h
View File

@ -15,7 +15,6 @@
#include <stdint.h>
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -29,6 +28,10 @@ class DtmfToneGenerator {
DtmfToneGenerator();
virtual ~DtmfToneGenerator() {}
DtmfToneGenerator(const DtmfToneGenerator&) = delete;
DtmfToneGenerator& operator=(const DtmfToneGenerator&) = delete;
virtual int Init(int fs, int event, int attenuation);
virtual void Reset();
virtual int Generate(size_t num_samples, AudioMultiVector* output);
@ -48,8 +51,6 @@ class DtmfToneGenerator {
int amplitude_; // Amplitude for this event.
int16_t sample_history1_[2]; // Last 2 samples for the 1st oscillator.
int16_t sample_history2_[2]; // Last 2 samples for the 2nd oscillator.
RTC_DISALLOW_COPY_AND_ASSIGN(DtmfToneGenerator);
};
} // namespace webrtc

6
modules/audio_coding/neteq/expand.h
View File

@ -15,7 +15,6 @@
#include <memory>
#include "modules/audio_coding/neteq/audio_vector.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -41,6 +40,9 @@ class Expand {
virtual ~Expand();
Expand(const Expand&) = delete;
Expand& operator=(const Expand&) = delete;
// Resets the object.
virtual void Reset();
@ -134,8 +136,6 @@ class Expand {
bool stop_muting_;
size_t expand_duration_samples_;
std::unique_ptr<ChannelParameters[]> channel_parameters_;
RTC_DISALLOW_COPY_AND_ASSIGN(Expand);
};
struct ExpandFactory {

6
modules/audio_coding/neteq/expand_uma_logger.h
View File

@ -17,7 +17,6 @@
#include "absl/types/optional.h"
#include "api/neteq/tick_timer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -36,6 +35,9 @@ class ExpandUmaLogger {
~ExpandUmaLogger();
ExpandUmaLogger(const ExpandUmaLogger&) = delete;
ExpandUmaLogger& operator=(const ExpandUmaLogger&) = delete;
// In this call, value should be an incremental sample counter. The sample
// rate must be strictly positive.
void UpdateSampleCounter(uint64_t value, int sample_rate_hz);
@ -48,8 +50,6 @@ class ExpandUmaLogger {
absl::optional<uint64_t> last_logged_value_;
uint64_t last_value_ = 0;
int sample_rate_hz_ = 0;
RTC_DISALLOW_COPY_AND_ASSIGN(ExpandUmaLogger);
};
} // namespace webrtc

6
modules/audio_coding/neteq/merge.h
View File

@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_MERGE_H_
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -36,6 +35,9 @@ class Merge {
SyncBuffer* sync_buffer);
virtual ~Merge();
Merge(const Merge&) = delete;
Merge& operator=(const Merge&) = delete;
// The main method to produce the audio data. The decoded data is supplied in
// `input`, having `input_length` samples in total for all channels
// (interleaved). The result is written to `output`. The number of channels
@ -93,8 +95,6 @@ class Merge {
int16_t input_downsampled_[kInputDownsampLength];
AudioMultiVector expanded_;
std::vector<int16_t> temp_data_;
RTC_DISALLOW_COPY_AND_ASSIGN(Merge);
};
} // namespace webrtc

7
modules/audio_coding/neteq/neteq_impl.h
View File

@ -29,7 +29,6 @@
#include "modules/audio_coding/neteq/packet.h"
#include "modules/audio_coding/neteq/random_vector.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@ -124,6 +123,9 @@ class NetEqImpl : public webrtc::NetEq {
~NetEqImpl() override;
NetEqImpl(const NetEqImpl&) = delete;
NetEqImpl& operator=(const NetEqImpl&) = delete;
// Inserts a new packet into NetEq. Returns 0 on success, -1 on failure.
int InsertPacket(const RTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload) override;
@ -399,9 +401,6 @@ class NetEqImpl : public webrtc::NetEq {
ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(mutex_);
bool no_time_stretching_ RTC_GUARDED_BY(mutex_); // Only used for test.
rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(mutex_);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
};
} // namespace webrtc

6
modules/audio_coding/neteq/normal.h
View File

@ -17,7 +17,6 @@
#include "api/neteq/neteq.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
@ -49,6 +48,9 @@ class Normal {
virtual ~Normal() {}
Normal(const Normal&) = delete;
Normal& operator=(const Normal&) = delete;
// Performs the "Normal" operation. The decoder data is supplied in `input`,
// having `length` samples in total for all channels (interleaved). The
// result is written to `output`. The number of channels allocated in
@ -68,8 +70,6 @@ class Normal {
const size_t samples_per_ms_;
const int16_t default_win_slope_Q14_;
StatisticsCalculator* const statistics_;
RTC_DISALLOW_COPY_AND_ASSIGN(Normal);
};
} // namespace webrtc

5
modules/audio_coding/neteq/packet_buffer.h
View File

@ -15,7 +15,6 @@
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/packet.h"
#include "modules/include/module_common_types_public.h" // IsNewerTimestamp
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -51,6 +50,9 @@ class PacketBuffer {
// Deletes all packets in the buffer before destroying the buffer.
virtual ~PacketBuffer();
PacketBuffer(const PacketBuffer&) = delete;
PacketBuffer& operator=(const PacketBuffer&) = delete;
// Flushes the buffer and deletes all packets in it.
virtual void Flush(StatisticsCalculator* stats);
@ -173,7 +175,6 @@ class PacketBuffer {
size_t max_number_of_packets_;
PacketList buffer_;
const TickTimer* tick_timer_;
RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
};
} // namespace webrtc

6
modules/audio_coding/neteq/post_decode_vad.h
View File

@ -16,7 +16,6 @@
#include "api/audio_codecs/audio_decoder.h"
#include "common_audio/vad/include/webrtc_vad.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -31,6 +30,9 @@ class PostDecodeVad {
virtual ~PostDecodeVad();
PostDecodeVad(const PostDecodeVad&) = delete;
PostDecodeVad& operator=(const PostDecodeVad&) = delete;
// Enables post-decode VAD.
void Enable();
@ -63,8 +65,6 @@ class PostDecodeVad {
bool active_speech_;
int sid_interval_counter_;
::VadInst* vad_instance_;
RTC_DISALLOW_COPY_AND_ASSIGN(PostDecodeVad);
};
} // namespace webrtc

6
modules/audio_coding/neteq/preemptive_expand.h
View File

@ -15,7 +15,6 @@
#include <stdint.h>
#include "modules/audio_coding/neteq/time_stretch.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -36,6 +35,9 @@ class PreemptiveExpand : public TimeStretch {
old_data_length_per_channel_(0),
overlap_samples_(overlap_samples) {}
PreemptiveExpand(const PreemptiveExpand&) = delete;
PreemptiveExpand& operator=(const PreemptiveExpand&) = delete;
// This method performs the actual PreemptiveExpand operation. The samples are
// read from `input`, of length `input_length` elements, and are written to
// `output`. The number of samples added through time-stretching is
@ -67,8 +69,6 @@ class PreemptiveExpand : public TimeStretch {
private:
size_t old_data_length_per_channel_;
size_t overlap_samples_;
RTC_DISALLOW_COPY_AND_ASSIGN(PreemptiveExpand);
};
struct PreemptiveExpandFactory {

7
modules/audio_coding/neteq/random_vector.h
View File

@ -14,8 +14,6 @@
#include <stddef.h>
#include <stdint.h>
#include "rtc_base/constructor_magic.h"
namespace webrtc {
// This class generates pseudo-random samples.
@ -26,6 +24,9 @@ class RandomVector {
RandomVector() : seed_(777), seed_increment_(1) {}
RandomVector(const RandomVector&) = delete;
RandomVector& operator=(const RandomVector&) = delete;
void Reset();
void Generate(size_t length, int16_t* output);
@ -39,8 +40,6 @@ class RandomVector {
private:
uint32_t seed_;
int16_t seed_increment_;
RTC_DISALLOW_COPY_AND_ASSIGN(RandomVector);
};
} // namespace webrtc

7
modules/audio_coding/neteq/red_payload_splitter.h
View File

@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
#include "modules/audio_coding/neteq/packet.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -30,6 +29,9 @@ class RedPayloadSplitter {
virtual ~RedPayloadSplitter() {}
RedPayloadSplitter(const RedPayloadSplitter&) = delete;
RedPayloadSplitter& operator=(const RedPayloadSplitter&) = delete;
// Splits each packet in `packet_list` into its separate RED payloads. Each
// RED payload is packetized into a Packet. The original elements in
// `packet_list` are properly deleted, and replaced by the new packets.
@ -43,9 +45,6 @@ class RedPayloadSplitter {
// is accepted. Any packet with another payload type is discarded.
virtual void CheckRedPayloads(PacketList* packet_list,
const DecoderDatabase& decoder_database);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(RedPayloadSplitter);
};
} // namespace webrtc

6
modules/audio_coding/neteq/statistics_calculator.h
View File

@ -15,7 +15,6 @@
#include <string>
#include "api/neteq/neteq.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -28,6 +27,9 @@ class StatisticsCalculator {
virtual ~StatisticsCalculator();
StatisticsCalculator(const StatisticsCalculator&) = delete;
StatisticsCalculator& operator=(const StatisticsCalculator&) = delete;
// Resets most of the counters.
void Reset();
@ -197,8 +199,6 @@ class StatisticsCalculator {
PeriodicUmaAverage excess_buffer_delay_;
PeriodicUmaCount buffer_full_counter_;
bool decoded_output_played_ = false;
RTC_DISALLOW_COPY_AND_ASSIGN(StatisticsCalculator);
};
} // namespace webrtc

6
modules/audio_coding/neteq/sync_buffer.h
View File

@ -20,7 +20,6 @@
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/audio_vector.h"
#include "rtc_base/buffer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -32,6 +31,9 @@ class SyncBuffer : public AudioMultiVector {
end_timestamp_(0),
dtmf_index_(0) {}
SyncBuffer(const SyncBuffer&) = delete;
SyncBuffer& operator=(const SyncBuffer&) = delete;
// Returns the number of samples yet to play out from the buffer.
size_t FutureLength() const;
@ -102,8 +104,6 @@ class SyncBuffer : public AudioMultiVector {
size_t next_index_;
uint32_t end_timestamp_; // The timestamp of the last sample in the buffer.
size_t dtmf_index_; // Index to the first non-DTMF sample in the buffer.
RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
};
} // namespace webrtc

6
modules/audio_coding/neteq/time_stretch.h
View File

@ -14,7 +14,6 @@
#include <string.h> // memset, size_t
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -49,6 +48,9 @@ class TimeStretch {
virtual ~TimeStretch() {}
TimeStretch(const TimeStretch&) = delete;
TimeStretch& operator=(const TimeStretch&) = delete;
// This method performs the processing common to both Accelerate and
// PreemptiveExpand.
ReturnCodes Process(const int16_t* input,
@ -105,8 +107,6 @@ class TimeStretch {
int32_t vec2_energy,
size_t peak_index,
int scaling) const;
RTC_DISALLOW_COPY_AND_ASSIGN(TimeStretch);
};
} // namespace webrtc

6
modules/audio_coding/neteq/timestamp_scaler.h
View File

@ -12,7 +12,6 @@
#define MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
#include "modules/audio_coding/neteq/packet.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -34,6 +33,9 @@ class TimestampScaler {
virtual ~TimestampScaler() {}
TimestampScaler(const TimestampScaler&) = delete;
TimestampScaler& operator=(const TimestampScaler&) = delete;
// Start over.
virtual void Reset();
@ -59,8 +61,6 @@ class TimestampScaler {
uint32_t external_ref_;
uint32_t internal_ref_;
const DecoderDatabase& decoder_database_;
RTC_DISALLOW_COPY_AND_ASSIGN(TimestampScaler);
};
} // namespace webrtc

6
modules/audio_coding/neteq/tools/audio_checksum.h
View File

@ -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 @@ class AudioChecksum : public AudioSink {
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 @@ class AudioChecksum : public AudioSink {
std::unique_ptr<rtc::MessageDigest> checksum_;
rtc::Buffer checksum_result_;
bool finished_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
};
} // namespace test

6
modules/audio_coding/neteq/tools/audio_loop.h
View File

@ -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 @@ class AudioLoop {
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 @@ class AudioLoop {
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

19
modules/audio_coding/neteq/tools/audio_sink.h
View File

@ -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 @@ class AudioSink {
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 @@ class AudioSink {
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 @@ class AudioSinkFork : public AudioSink {
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

6
modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
View File

@ -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 @@ class ConstantPcmPacketSource : public PacketSource {
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 @@ class ConstantPcmPacketSource : public PacketSource {
uint16_t seq_number_;
uint32_t timestamp_;
const uint32_t payload_ssrc_;
RTC_DISALLOW_COPY_AND_ASSIGN(ConstantPcmPacketSource);
};
} // namespace test

6
modules/audio_coding/neteq/tools/input_audio_file.h
View File

@ -15,8 +15,6 @@
#include <string>
#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace test {
@ -27,6 +25,9 @@ class InputAudioFile {
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 @@ class InputAudioFile {
private:
FILE* fp_;
const bool loop_at_end_;
RTC_DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
};
} // namespace test

6
modules/audio_coding/neteq/tools/output_audio_file.h
View File

@ -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 @@ class OutputAudioFile : public AudioSink {
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 @@ class OutputAudioFile : public AudioSink {
private:
FILE* out_file_;
RTC_DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
};
} // namespace test

6
modules/audio_coding/neteq/tools/output_wav_file.h
View File

@ -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 @@ class OutputWavFile : public AudioSink {
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 @@ class OutputWavFile : public AudioSink {
private:
WavWriter wav_writer_;
RTC_DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
};
} // namespace test

6
modules/audio_coding/neteq/tools/packet.h
View File

@ -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 @@ class Packet {
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 @@ class Packet {
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

7
modules/audio_coding/neteq/tools/packet_source.h
View File

@ -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 @@ class PacketSource {
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 @@ class PacketSource {
protected:
std::bitset<128> filter_; // Payload type is 7 bits in the RFC.
private:
RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource);
};
} // namespace test

5
modules/audio_coding/neteq/tools/resample_input_audio_file.h
View File

@ -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 @@ class ResampleInputAudioFile : public InputAudioFile {
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 @@ class ResampleInputAudioFile : public InputAudioFile {
const int file_rate_hz_;
int output_rate_hz_;
Resampler resampler_;
RTC_DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
};
} // namespace test

6
modules/audio_coding/neteq/tools/rtc_event_log_source.h
View File

@ -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 @@ class RtcEventLogSource : public PacketSource {
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 @@ class RtcEventLogSource : public PacketSource {
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

6
modules/audio_coding/neteq/tools/rtp_file_source.h
View File

@ -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 @@ class RtpFileSource : public PacketSource {
~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 @@ class RtpFileSource : public PacketSource {
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

12
modules/audio_coding/neteq/tools/rtp_generator.h
View File

@ -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 @@ class RtpGenerator {
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 @@ class RtpGenerator {
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 @@ class TimestampJumpRtpGenerator : public RtpGenerator {
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 @@ class TimestampJumpRtpGenerator : public RtpGenerator {
private:
uint32_t jump_from_timestamp_;
uint32_t jump_to_timestamp_;
RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
};
} // namespace test