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Denon Original Technologies

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Advanced AL32 Processing

Denon led the world in developing a practical PCM recorder in 1972. The shift to digital technology solved numerous problems such as noise, wow, flutter, and frequency response. The delicate nuances of sound from CD up to the least significant bit (LSB) are lost, however, due to the 16-bit quantization and band cutting at fs/2 (half the sampling frequency) during 44.1-kHz sampling. Denon was determined to tackle the problem of quantization noise that remained in digital audio, and developed ALPHA (adaptive line pattern harmonized algorithm), a technology that reproduced 16-bit data with 32-bit quality and became a favorite with audiophiles around the world.

Advanced AL32 Processing uses a unique data interpolation algorithm to achieve high-bit, high-sampling output performance. The volume of information has been dramatically improved without any loss in the original data. The clean playback sound free of interference makes it possible to enjoy delicate details, accurate localization, and rich expression in the lower range.

Advanced AL32 Processing has three functions:

Function 1. High-bit up-conversion (Adaptive Line Pattern Harmonized Algorithm)

By installing arithmetic circuitry for lower-order data generation that is four times greater than conventional arithmetic circuitry and using that to process calculations, upper-order 16-bit data is added to the lower-order 16-bit data that is generated and output data of 32-bit quality is obtained.

Audio signals that previously could only be output in a 1LSB staircase pattern are reproduced in a smooth waveform of 32-bit precision, reducing distortion at micro levels.

AL32_Sine_wave_of_waveform_reproduced_cd_diagram.jpg

Function 2. Advanced ALPHA Processing

The 44.1-kHz sampling signals of a CD are oversampled by a factor of 16 to produce a smoother waveform. At this time, simply performing linear interpolation and increasing the data will not produce a signal waveform that exists in the natural world.
A waveform close to that of the original signal is achieved by inferring data interpolated from a large volume of data that should be reproduced before and after the data read from the CD.

AL32_Conceptual_diagram.jpg

By analyzing large volumes of sampling before and after data read from the CD and inferring and interpolating the points that should exist, it is possible to produce a smooth signal that is closer to the original sound.


Function 3. Adaptive digital filter (Automatic Low-Pass filter Harmonic Adjustment)

In conventional ALPHA Processing, an adaptive digital filter was used to widen the passband for pulse data and prevent the occurrence of ringing. Advanced AL32 Processing, however, uses a filter algorithm with even greater adaptive capability. Since filtering is conducted using the optimum algorithm even for pulsating signals or continuous high-frequency sound, a natural sound, unaffected by aliasing noise or lower high-range response, is produced.

AL32_impulse_response_attack_sound.jpg