Lavry Engineering - Lavry Gold - LE3000S - finalizer

• AVAILABILITY/DELIVERY: On Order/Usually 10 days

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Lavry Engineering LE3000S is a powerful multi-function digital audio processor (finalizer) for sample rate/word length conversion, monitoring, and measurement. Includes multiple dither and noise shaping settings, plus reference quality metering.

Features
for Lavry Engineering - Lavry Gold LE3000S audio processor/finalizer

• Sample Rate Converter - highest performance in the industry 96Kbytes of algorithm coefficients for superior accuracy
• High Density Downsample, provides 2:1 sample rate change for the finest audio performance
• Proprietary jitter attenuation process, with high quality tuned LC circuit
• Acoustic Bit Correction™ - dynamic range increase to 18-19 bits on a 16-bit format
• AES and word clock external synchronization
• Selectable dither types (Flat, High-Pass Triangular, OFF)
• Data Format Converter - displays all input parameters and allows editing of output settings, including word length
• High-resolution reference meter - (Adjustable Scaling to include noise floor) Peak, Peak-Hold, Clip and all Zero detection. Fine mode for reference level settings
• Digital Tone Generator (16-24 bits, -122 THD+ N) Fully programmable amplitude, frequency and channel selection
• Measures THD+N, frequency and amplitude, with test bench accuracy
• Signal Analysis - verifies exact number of bits being produced

Additional Features
for Lavry Engineering - Lavry Gold LE3000S audio processor/finalizer

• 96kHz, 88.skHz, 48kHz and 44.1kHz conversion frequencies
• Built-in Acoustic Bit Correction™ re-dithering to 16-20 bits
• Precision reference meter bridge
• Fully programmable digital test tones for system alignment
• AES and word clock external synchronization
• Programmable Digital delay 0.2 frames to 10 frames
• 96K or 88.2K to 48K or 44.1K sample rate down conversion
• 48K or 44.1K to 96K or 88.2K sample rate up conversion
• Bit-packing function for 96K DAT -- 48K 24-bits slit to packed 16-bit 96K stream with decoding on playback
• DC removal function
• Normalization function (boost or cut) +/-60dB
• Polarity Inversion
• Digital Bits display on meter

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SPECIFICATIONS and DOWNLOADS
for Lavry Engineering - Lavry Gold LE3000S audio processor/finalizer

• User Manual for Lavry LE3000S audio processor
  

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Signal Analysis of THD+N with dB3000S

Introduction

Lavry Engineering's latest revision to the Model 3000S Sample-Rate Data Format Converter consists of THD+N (total harmonic distortion and noise) measurement capabilities. The new function expands the usefulness of this "digital audio all in one tool". The new feature offers the high performance and ease of use found in all the other features (data format conversion, sample rate conversion, Acoustic Bit Correction, reference meter bridge and test tone generation). 24 bit tone generation and THD+N measurement accuracy in excess of 122ÝdB provide sufficient margin for the next generation of digital audio equipment.

The Model 3000S's test tone generation and Signal analysis are independent of each other. This flexibility allows performing either function separately or both simultaneously. Distortion and noise

Real hardware generated tones contain both distortion and noise. Harmonic distortion is referred to as energy residing at multiples of the tone frequency. Noise is all other undesirable energy.

Some common sources of distortion:

• Amplifiers: non linear signal processing causes distortion. While mostly corrected for by use of negative feedback, amplifiers tend to degrade at higher frequencies.
• Component imperfection: distortions occur when component values depend on the signal. For example, capacitors tend to counteract a changing signal (dielectric absorption).
• Digital truncation: limitations of word length (number of bits) is a non linear process. Such distortions increase at lower signal levels.

By definition, noise is not all random (tones occurring at non harmonic frequencies are considered noise). Well known and understood is the critical noise requirement associated with signal amplification. Often overlooked is the accumulated noise due to connecting of many units in series.

Common sources of random noise:

• Resistor noise (flat frequency distribution) increases for larger value of resistance.
• Semiconductor noise, mostly flat frequency distribution. Increased noise levels at very low frequencies usually occurs below audible frequencies.
• Capacitor noise, inconsequential for higher values, has become a performance limiting factor with the introduction of very small capacitor values incorporated in modern semiconductors such as oversampling sigma delta converters.
• Digital truncation: limitations of word length (number of bits) in A/D converters, signal processors and more. The problem grows with increased amount of processing. Common sources of non-random noise:
• Coupling to analog signal path: AC power line, RFI/EMI, coupling of digital signals to analog path, inadequate power supply rails and more.
• Digital truncation: limitations of word length (number of bits) in A/D converters, signal processors and more. The problem increases for low level signals.
• Limit cycles: cyclical patterns behavior in feedback based digital signal processing (such as sigma delta converters and IIR filter structures).

Listening tests and measurement of individual equipment in the audio chain does not guarantee optimum THD+N performance. Setup optimization is a very complex subject. Top recording engineers blend artistic considerations and engineering know-how into the process. The following discussion does not deal with artistic aspects. Lavry acknowledges the great importance of artistry in music production, but is bound to limit discussion to measurable and objective phenomena.


Optimizing THD+N (some engineering considerations):

Analog amplification: good signal to noise ratio requires "early" signal amplification, but with careful attention to tradeoffs between distortions and noise.

Analog attenuation: undesirable from noise standpoint, may be required to accommodate signal range limitations of various gear.

Jitter: clock jitter in A/D, D/A and Sample Rate Converters degrades THD+N. Reference D/A clock jitter in a studio may have little to do with the end product quality, but may make the monitoring process difficult. Sample rate converters perform best with low jitter on both incoming and outgoing clocks.

Configuring proper digital chain: whenever unit A may drive unit B or visa versa. A good "rule of thumb" is to have the better performer drive the lower performer. A quality digital device utilizing 24 bit words is limited to 16 bits when driven by a 16 bit device. The compounded outcome is that of "2 x 16 bit devices". Reversing the order allows the first process to retain its high accuracy, leaving a compounded outcome of one 16 bit device.


Measuring THD+N

The common method for measuring THD+N is based on feeding a "device under test" with a quality reference test tone and measuring the undesirable energy (THD+N) at its output. Lavry Engineering's Model 3000S provides the user with a reference test tone. The processed tone (or any other source) may be fed back to Model 3000S input. The input signal is filtered by a very sharp notch to separate the desired signal component from the undesirable energy (THD+N). The undesirable energy (THD+N) is then displayed inÝdB (referenced to full scale).

The notch filter must be very deep and narrow. Notch depth assures that no energy at the fundamental frequency "leaks" to the meter. Narrow notch is required to leave noise and harmonics intact. How steep should the notch be? For theoretical tones one may strive for the steepest notch possible. Real applications require full attenuation over a slightly wider frequency range to accommodate possible small jitter of both sampling clock and the tone itself. Model 3000S provides sufficient attenuation over about .1% (allowing about 20 nsec jitter).

Additional 20 Hz high pass filtering ensures that very low frequency components (such as DC and other low frequency inaudible energy) does not alter the outcome.

Model 3000S begins its measurement by locking to the test tone frequency. There are 2 modes (user selectable) for achieving lock:

• Auto mode: Proper locking in the presence of noise and distortion requires "reasonable" signal to noise and distortion ratio. When using Auto mode we recommend starting the test with a large enough signal (full scale signals are ideal, but locking will take place at 40ÝdB over the noise and distortions floor). Pressing the "Go" button sets the notch in place, allowing THD+N measurements at any level.
• Normal mode: when using Model 3000S tone generator, the notch frequency follows the tone frequency settings. Normal mode frees the lock mechanism from any signal to noise restrictions. This mode requires the tone sample rate and incoming signal sample rate to match within +/- .1% of each other (testing Sample rate converters requires Auto mode).

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Testing Of Asynchronous Sample Rate Converters

The theoretically ideal sample rate converter is a device that converts the data transfer rate without changing the content of the material.

Listening tests:

Many devices are judged by their particular sonic quality. A recording engineer may prefer, for example, a small amount of distortion to add some "color" to the sound. The same engineer, testing a number of processing devices separately, may choose his building blocks to achieve a certain characteristic sound. Let us assume that the desired characteristic is based on some small amount of second order distortion which imparts a characteristically "warm" sound. A problem may arise when processing the sound through more than one unit. The "desired distortion" may be compounded beyond the desired level (in our example, the second order distortion may increase each time the sound is processed to an unacceptable end result).

Sample rate converters may serve to reduce excessive clock jitter. This improvement can take place for any sampling-rate ratio (including 1:1). Jittery incoming data introduces signal dependent noise and distortions (increasing with signal amplitude and frequency). Such jitter reduction yields noticeable sonic improvement, thus complicating the objectivity of listening tests. Some manufacturers' comparison tests inappropriately compare a high jitter input to a low jitter output. Further confusion is due to the fact that digital domain FFT tests do not adequately show the effects of input jitter.

Listening tests should be based on comparing the audio of a directly applied signal against the converted version of the same material. The greater the difference, the less ideal the converter.

The sample rate converter should receive a low jitter data source, and drive a reference grade D/A converter, a high quality power amplifier and top grade speakers, all matched to 0.1 dB. "Blind" listening comparisons (A/B/X tests) by recording professionals yield the best unbiased results.


Measuring performance:

The most commonly used measurements are based on a standard FFT, THD plus Noise testing (in the digital domain) and phase linearity. Interpreting measurements performed on asynchronous sample rate converters is less straightforward. The asynchronous sample rate converter can not control the input and output rates (these rates are forced by the driving source and required destination devices). The converter is required to reconstruct the data content while accommodating receiver and transmitter clock rates.

The digital nature of the process introduces quantization effect into the conversion. The ratio does not change smoothly. It tracks the clock rate variations in a quantized fashion (small incremental jumps). Proper design requires that the quantized ratio changes fall below the ear sensitivity levels. Tracking the clock rates sets restrictions on the maximum tolerable ratio step size and the manner in which the ratio tracks the clocks rates. Let us focus on the two extreme cases for ratio tracking:

• a. Steady clocks: The converter is adjusting the ratio up and down by a small amount around the correct average ratio.
• b. Fast "varispeed": The adjustment accuracy is reduced with fast varispeed affecting the accuracy of ratio adjustment.

Technological limitations require careful consideration for optimizing both ratio step size and the tracking mechanism. Psychoacoustic considerations (listening tests) and practical limitations of varispeed should form the basis for proper performance criteria. Measuring asynchronous sample rate converters may reveal some of these compromises.

While the input and output of theoretical converters measure identically, real converters continuously track and adjust internal coefficients. Such ratio modulation appears on FFT measurements as a "widening of the main lobe" of a sinusoidal test tone. The amount of widening depends greatly on variables such as ratio step size, ratio tracking, FFT size and type of FFT window used.

Common digital domain measurements (FFT based measurements) do not show the effects of low levels of incoming jitter. A common indirect approach is based on measuring the THD plus noise reduction at the output of a reference DAC (driven by full scale high frequency tone). While such a measurement does not quantify jitter, it yields (in principle) the desired end result. Real world limitations of reference DAC performance set limits to such measurements (DAC performance is typically lower at high amplitudes and frequencies). Model 3000 utilizes a high Q (steep resonance) LC circuit for de-jittering incoming data. Further jitter reduction may be achieved with a 1:1 sample rate conversion, using a low jitter crystal clock oscillator for the output data clock.

Sample rate converter performance should be measured over the usable audio range. Poor performance at high frequencies cannot be dismissed as inaudible noise. A typical justification for high frequency performance degradation is based on the fact that real music contains less energy at this frequency range. This could allow, at most, a few dB reduction in THD+N. Unlike many unsampled analog circuits which tend to generate higher frequency distortions, sampling folds back distortion energy to lower frequencies, including the ear's most sensitive mid-range region.

When processing long words a desirable performance specification should exceed the limitations of a 16 bit word by a significant margin. This will ensure that overall performance is limited almost completely by word length bottleneck. (Truncation of a long word to 16 bits yields theoretical results of approximately 98 dB THD+N). The combination of long word format and good performance specifications is even more desirable when additional processing may take place. Premature truncation amounts to loss of detail.

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