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Audio analyzers generate sine waves, square waves, random noise, and combinations of these signals. For optimum performance, some products use multiple oscillators or waveforms in the analog domain. Typically, converter-based audio analyzers are used when analog methods are unable to generate specific waveforms. By using DACs, these devices can produce not only multi-tone waveforms and shaped burst, but also sine waves with inter-channel phase shifts—a technique that is especially useful in testing surround-sound decoders. Balancing sharpness of filtering against distortion reduction is difficult, however, and sharper filters may introduce more response ripples. Therefore, most high-performance audio analyzers have a dedicated analog oscillator for total harmonic distortion plus noise (THD+N) testing.
Audio analyzers carry product specifications for both generator and analyzer performance. Generator performance specifications consist of sine frequency range, frequency accuracy, intermodulation distortion (IMD) test-signal type, maximum amplitude, amplitude accuracy, flatness, and digital-output sampling rate. The analog output configuration for audio analyzers is either balanced or unbalanced. Analyzer specifications include maximum rated input voltage, maximum bandwidth, amplitude accuracy, amplitude flatness, residual input noise, fast Fourier transform (FFT) resolution, IMD measurement capability, and DC voltage measurement. Mounting style and form factor are also important to consider. Audio engineers and audio technicians use audio analyzers in a variety of settings, including design laboratories, broadcast facilities and production lines.
Components of Audio Analyzer
User interface
Audio analyzer uses an integrated computer running the windows xp embedded operating system, so operation will be immediately familiar and intuitive. Depending on the application, audio analyzer can be operated with an external mouse and keyboard, or by using the front-panel knob, keypad and touchpad. Seven on-screen tabbed pages are available for arranging panels, graphs, and displays. Screen setups, data, and instrument configurations can be quickly saved and recalled to either the internal hard disk or to a flash drive connected to one of the two front-panel usb connectors. An optional 1024 x 768 xvga monitor provides better resolution and allows more information to be displayed.
Analog signal generator
At the heart of audio analyzer is a uniquely flexible analog signal generator. All of the standard audio waveforms are available including sine, log-swept sine chirp, synchronous burst sine, noise (white, pink, and filtered), standard intermodulation test signals (smpte, ccif, dim), square waves, arbitrary waveforms (ascii and .Wav), ramps and multitone waveforms. Many of these signals can be combined in the generator allowing you to create an unlimited number of test waveforms. But the analog signal generator doesn't sacrifice performance for flexibility. With a flatness of ±0.008 db (20 hz to 20 khz) and a residual thd+n of -106 db (20 hz to 20 khz), audio analyzer's low distortion sine rivals the performance of any analyzer.
Digital audio signal generator
The same flexibility and performance is found in audio analyzer's digital audio signal generator. Almost all the same waveforms found in the analog generator are available in the digital generator with the addition of several special digital test waveforms including digital constant, walking bits, and a staircase waveform (for d/a testing). The digital audio output sampling rate is continuously adjustable from 24 khz to 216 khz (single and dual connector). Full control over transmitted status bits (in both professional and consumer formats), user bits, and validity bits is provided.
Digital i/o panels
For digital interface testing, a variety of impairment signals can be imposed on the digital audio carrier. Carrier impairments include variable rise time (5 ns to 400 ns), common mode sine waves, normal mode noise, and several jitter waveforms (sine, square, and noise).
Timebase
All of audio analyzer's sampling clocks are derived from an internal timebase with 5 ppm accuracy. For the most demanding applications, an optional atomic rubidium (perf) timebase is available with an accuracy at shipment of ±5 x 10-11, and a 20-year aging specification of )less than 5 ppb. Additionally, the timebase may be synchronized to an external clock, an aes11 reference signal, or any standard video signal.
Analyzers
The heart of audio analyzer's measurement abilities is its versatile set of analyzers which operate symmetrically on both analog and digital audio signals with no need to purchase additional options. Up to two analyzers can be run simultaneously on either the analog or digital inputs. The Time Domain Detector makes all of the standard audio measurements including Amplitude, Crosstalk, and THD+N. Continuously variable bandwidth limiting and standard weighting filters are included. The post notch-filter distortion signal can be fed to an FFT analyzer for a live spectral display of distortion, or to the rear-panel monitor output or speaker.
Digital Audio Interface
Audio analyzer provides a complete set of measurements for digital interface testing. Carrier level and sampling frequency are measured directly. Status bits are fully decoded in both professional and consumer formats, and user bits are displayed as well. audio analyzer's Jitter Analyzer measures jitter in both the time and frequency domain, including continuously variable bandwidth limiting and weighting in both domains.
Digitizer
An optional 80 MHz transient digitizer (opt. 01) provides additional digital audio carrier analysis. Operating on a record of up to 2M samples, the digitizer computes and displays the time record of the input signal and its jitter, input spectrum, jitter spectrum, and the probability distributions of the input and jitter amplitudes as well as the pulse width and pulse rate. Full color eye-diagrams can be generated allowing easy testing against user-configurable eye limits.
Automation and Programming
Audio analyzer offers unprecedented flexibility for user scripting and remote programming. On-board scripts can be written in VBScript, Jscript, or Python with full access to all of the instrument's capabilities as well as the ability to create simple user-interfaces for running tests. audio analyzer has a complete hierarchical GPIB command set, and GPIB commands can be sent over the standard IEE-488 interface, RS-232 port, or over the Ethernet on a TCP/IP network (VXI-11). Finally, audio analyzer has a complete COM interface allowing instrument operation to be automated from any COM capable application such as Visual Basic, LabView, or Microsoft Office.
Level and gain
Level describes the magnitude of a signal, and may be expressed in absolute or relative terms. Common absolute units may be volts, watts, dbv and dbu, while relative measurements are expressed most commonly in db. Level may also be conditioned as a peak measurement or an rms measurement. Gain is the ratio of signal level at a dut's output divided by the signal level at the input, usually expressed in db.
Frequency response
Measures the output level of a dut as a function of frequency. Level is expressed in the same units as above, typically dbv and dbu.
Total harmonic distortion plus noise (thd+n)
Harmonic distortion products are multiples of stimulus frequencies, while noise is energy that is mathematically unrelated to the input signal. As a signal result, thd+n can be considered all signal content in the dut response that is not contained in the stimulus.
Signal-to-noise ratio (snr)
The ratio of desired signal to unwanted noise coming from a dut, expressed in db.
Crosstalk
The unwanted presence of a signal from one audio channel as it appears in other audio channels of a dut. Since this is a ratio, it is expressed in db.
Phase
The relationship in time between two signals of identical frequency, expressed as a fraction of the period of the signal. This is usually expressed in degrees, with one complete cycle of a sinusoidal signal being 360 degrees.
Intermodulation distortion (imd)
Distortion that is the result of non-linear mixing of two or more signals, typically two sine-waves at different frequencies or the sum of a sine-wave and square-wave. In addition to distortion products at harmonic multiples of the frequencies, products are also found at multiples of the sums and differences of the original frequencies.
Time domain display
Equivalent to an oscilloscope display of the signal, showing instantaneous amplitude as a function of time.
Tips on Getting Good Data with Your Audio Analyzer
Double check all connections
This is the most common problem I run into. You accidentally plug the microphone into the reference channel. If checking the connections doesn’t help and you are doing some fancy routing with your mix console, go back to a simple Y split from the output of the audio analyzer instead. If everything is working, you should be able to make a change on your system EQ and see it reflected in the analyzer.
Measure at an appropriate level
The great thing about the audio analyzer is that you don’t need to blast everyone with pink noise, but you do need to get it up above the noise floor by 20 dB at all frequencies you want to measure for good coherence.
Use coherence blanking
Sometimes you’ll take a measurement that will go all the way down to 20 Hz, but you know the speaker only really goes down to 80 Hz. What’s happening? The rest of the measurement is noise, which you can see in the coherence and phase trace. Bring up the coherence threshold until the noise disappears.
Use more averages
You’ll need about 1 second for normal conditions and 3 for noisy conditions. If that doesn’t work, set your averages to infinite, then just keep measuring until the coherence stops improving.
Do your Main+Sub alignment in the nearfield
I have had so many headaches trying to complete a Main+Sub alignment in the far field. It’s tough to get good data because you are no longer measuring direct sound from the sub, you are measuring a million other reflections that are just as loud or louder than the direct sound. After you do your alignment in the nearfield, you can adjust it for any distance offset with a laser distance measure.
Measure subs at ground plane
If you’re doing an LF measurement and having trouble getting a stable trace, try a ground plane mic position. This will remove the floor bounce.
Audio signals are essential to many modern electronic systems, as a means of communicating test results and often serving as a warning. Audio analysis is the process of understanding information from sound and what it might say about equipment under test. Testing helps to discover unwanted distortion and noise made by the audio equipment and possible how to remove or at least reduce the levels of noise compared to the audio signal levels. It requires a reliable sound source and a signal analyzer at least capable of operating within the audible frequency range.
Fine-tuning electronic equipment for audio performance whether for use or for production can be accomplished by a variety of techniques, in the frequency domain or the time domain and usually to gauge the consistency of the signal amplitude and/or phase response as a function of frequency across the operating bandwidth. Typically, the human hearing range extends from 20 Hz to 20 kHz although some test setups are geared for DC-coupled measurements, and some extend beyond the range of human hearing. Audio signals are subject to various forms of distortion and noise from other energy sources and must be tested against the effects of unwanted signal-to-noise radio (SNR) or even unwanted crosstalk between channels in a multichannel audio communications system. With appropriate software, analysis allows a visual depiction of the sounds under investigation to be viewed on a computer screen.
Audio signals serve many different purposes in electronic equipment, to draw the attention of human hearing and other listeners, such as a family pet, who may be warned away by the sounds. The sounds may be as simple as a ringing alarm or a caller’s message on a telephone answering machine. For an important message, clarity and accuracy are essential, rather than trying to decipher the message from a background of distortion and noise. Poor audio performance can make the difference a message that is fully understood and a number that was “lost in translation.”
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Analyzers, Audio(0/10)
