This is the characteristic of the audibility compensation filter used for noise level measurement.

"The noise level" is a measured value with insertion of an A-weighted sound pressure level filter.

Sensitivity of human hearing varies by the frequency (e.g., it is hard to hear the sound of the low frequency) so that it is necessary for noise level measurement to compensate the input signal of a microphone.

The A-weighted sound pressure level filter is used for the compensation.

This is a unit to express the level of the true peak meter using the peak level measurement algorithm by oversampling specified in Rec. ITU-R BS.1770.

The precision of the measurement of true peak improves along with the increase of a multiple of oversampling, but it is appropriate to use 4 times oversampling considering the complexity of calculation.

This is one of the digital filters.

An output signal is fed back to input by a recursive filter. The reply of the impulse becomes infinite. This is an infinite impulse reply filter.

This is the characteristic of the audibility compensation filter specified in Rec. ITU-R BS.1770.

It consists of two pre-filters. The first step is the filter which simulates head sound effects, and the second step is the RLB properties filter. The measurement that is near to human audibility is enabled by using a K-weighted sound pressure level filter.

This is the weighting of the second step filter of K-weighted sound pressure level defined in Rec. ITU-R BS.1770.

It is a modified B-weighted sound pressure level, so that it is called modified B-weighted sound pressure level.

This is a meter which measures the volume of an audio signal.

It is designed to replace the audio signal with a unit of volume called Volume Unit so that the meter shows volume level that is near to the feeling of a person.

It shows 99% indication in 0.3 seconds for the response time of the meter for the input signal. There is a problem that it does not show accurate response for the peak signal with rapid increase, but it has been used for many years.

This is to separate measured values by a certain threshold.

The calculation algorithm of a loudness level in this technical report removes the value that is smaller than the threshold to obtain the loudness level by using absolute gating and relative gating

By doing so, the loudness level which is nearer to subjectivity can be obtained.

In this technical report, an input signal is divided into the block of 400ms length in order to apply gating in loudness calculation.

One of the block of 400ms length is called a gating block.

This is the peak of an encoded digital signal.

The normal peak meter (sample peak meter) displays the maximum of the digital signal in a certain block as a peak level.

This is to reconstitute the sound with more than 3 channels into the sound with 2 (or 1) channels.

Generally, it is used when 5.1ch surround-sound is converted into 2 channels (stereo).

This is the peak of the analog signal and called "the true peak".

The sample peak meter in general use may not display the true peak that could exist between a sample and a sample.

A peak after oversampling is often called the true peak for convenience. Please mind not to confuse it.

This is the maximum coded level of a digital signal.

It may be called 0dBFS or Full Scale Digital.

In this technical report, this is the data added in an audio signal, which describes the control information of the signal.

The sound quality can be controlled by adding metadata to an audio signal to be played out.

In this technical report, this is the common name of the calculated value of loudness which is calculated from a digital recording level based on loudness measurement algorithm.

Calculated values with different calculation method including the calculated value without gating processing, the calculated value in the gating block (400ms length), and the calculated value after absolute gating and the relative gating are all called “loudness level” unless otherwise distinguished.

LKFS is used for a unit

This is one of the pulse code modulation.

It has basic process of sampling, quantization, and encoding, but does not employ data compression for encoding.

Generally PCM refers to linear PCM. There are also APCM, ADPCM.

This is to adjust the difference in standard level between equipment and system.

The level set is required for analog recording, when digital signals are converted to analog signals (D/A conversion).

This is the displayed value of noise (fluctuated noise) level fluctuated time to time by averaging total energy of measurement section T in time.

The average energy of measurement section T is obtained by averaging squared sound pressure (spontaneous pressure).

The equivalent noise level is expressed as noise level of the steady noise with average energy, which is equivalent to average energy in fluctuated noise section T.

Officially, the equivalent noise level uses LAeq,T for a quantifier and dB for a unit.

As a custom, LAeq, LAeq and Leq are used for a quantifier, or in some cases only dB(A) and dBA are used to express.

To measure "noise level" including an equivalent noise level, an A-weighting audibility compensation filter is used.

This is the loudness level of any measurement section which is calculated using loudness measurement algorithm including all gating processing described in Chapter 3.

Alternatively, it is called long term loudness or integrated loudness.

"The average loudness level of the program" stands for the average loudness level with a measurement section from the start of the program and the end of it.