1 INTRODUCTION … 5
1.1 REQUIREMENTS … 6
1.1.1 Soundcards … 6
1.2 MEASUREMENT SETUP … 8
1.3 A First Touch … 12
1.4 Audio Devices Setup. … 14
1.4.1 WDM Audio Driver Setup … 15
1.4.2 ASIO Driver Setup … 18
1.5 CALIBRATION … 19
1.5.1 Calibration of Soundcard Output Left Channel … 20
1.5.2 Calibration of Soundcard Input Channels. … 20
1.5.3 Calibration of the Microphone … 21
1.5.4 Frequency Response Compensation … 21
1.6 Rotating TURNTAble DRIVER SEtuP … 22
1.6.1 External .exe file driver … 23
1.6.2 Internal driver for Outline turntable ET 250-3D … 23
1.6.3 Testing of turntable driver … 23
1.7 GETTING IMAGES OF GRAPHS AND WINDOWS … 24
2 THE SPECTRUM ANALYZER … 26
2.1 Soundcard testing … 26
2.2 The SPECTRUM Estimation Procedure … 31
2.2.1 Spectrum Averaging … 33
2.2.2 Signal Windowing … 33
2.2.3 Spectrum Graph Setup. … 35
2.2.4 Graph Colors and Grid Style Setup. … 36
2.3 Frequency Resolution of DFT and Octave-BAnd AnAlyZers … 37
2.4 RMS LEVEL … 40
2.5 The Time ReCord … 41
2.6 Monitoring Spectra of Wideband Signals … 43
2.7 The Periodic Noise … 45
2.8 Testing WITH Two Sine SignAL … 47
2.8.1 Intermodulation distortion definitions. … 48
2.9 The Multitone Testing … 50
2.10 MONITORING MEASUREMENT DYNAMICs … 53
2.11 SPECTRUM OVERLAY AND TaRGET CURVES … 54
2.12 SAVING GENERATOR SIGNALS IN A .WAV FILE … 57
3 THEORY OF THE FREQUENCY RESPONSE MEASUREMENTS … 58
3.1 LTI INPUT / OUTPUT RELATIONSHIP … 58
3.2 DUAL CHANNEL SYSTEM WITH CONTINUOUS NOISE EXCITATION … 60
3.3 DUAL ChanNEl SYSTEM WITH PERIODIC NOISE EXCITATION … 61
3.4 SINGLE CHANNEL SySTEM FOR FREQUENCY RESPONSE ESTIMATION … 63
4 REAL-TIME FREQUENCY RESPONSE MEASUREMENT … 64
4.1 USER INTERFACE FOR REAL-TIME MEASUREMENT OF FREQUENCY RESPONSE … 64
4.2 Dynamic Range In Frequency Response MEASUREMENTS … 67
4.3 FR Overlay and Target Curves. … 70
4.4 GETTING IMPULSE RESPONSE FROM MEASURED FREQUENCY RESPONSE … 72
4.5 SYSTEM DELAY ESTIMATION. … 75
4.6 PIR Files … 76
4.6.1 PIR file format … 76
4.6.2 PIR file export and import … 78
4.6.3 Export of (spatial group) of frequency responses. … 80
5 IMPULSE RESPONSE MEASUREMENT AND SIGNAL RECORDING … 81
5.1 IMPULSE RESPONSE MEASUREMENT WITH PERIODIC NOISE EXCITATION … 81
5.2 IMPULSE RESPONSE MEASUREMENT WITH SWEPT-SINE EXCITATION … 83
5.3 IMPULSE RESPONSE MEASUREMENT WITH MLS EXCITATION … 85
5.4 IMPULSE RESPONSE MEASUREMENT WITH EXTERNAL EXCITATION AND TRIGGERED RECORDING … 87
5.5 TRIGGERED SIGNAL RECORDING … 88
5.5.1 Triggered signal recording with external excitation and spectrum estimation … 88
5.5.2 Triggered signal recording with internal signal and trigger generation … 90
5.6 BASIC EdITING PROCEDURES … 91
6 SYSTEM ANALYSIS FROM IMPULSE RESPONSE … 93
6.1 GATED IMPULSE AND FREQUENCY RESPONSE … 93
6.1.1 Gated Impulse Response. … 93
6.1.2 Gated Frequency Response … 96
6.1.3 Minimum Phase, Group Delay and Phase Intercept Distortion … 99
6.1.4 Managing Overlays … 100
6.1.5 Editing Smoothed Frequency Response. … 102
6.1.6 Low Frequency Loudspeaker Box Diffraction Scaling … 103
6.1.7 Repeated Measurement … 104
6.1.8 Simultaneous Measurement of Frequency Response and Harmonic Distortion … 104
6.1.9 Sound card transient time estimation using frequency response and distortion measurements … 107
6.2 STEP RESPONSE … 109
6.3 Impulse RESPONSE ENVELoPe (ETC - EnERGY TIME CURVE). … 110
6.4 CumULATIVE SPECTRUM. … 112
6.4.1 Cumulative spectral decay . … 112
6.4.2 Short-time Fourier transform … 116
6.5 BURST DECAY WITH A PERIOD BASED TIME SCALE … 118
6.5.1 Classical Sine-burst Testing … 118
6.5.2 The Importance of the Period Based Time Scale … 118
6.5.3 Wavelet Analysis for the Fast Estimation of Bursts Decay Envelopes … 119
6.5.4 Procedure for Obtaining the Burst Decay Graph … 120
6.5.4 Comparison of Burst Decay Graphs and CSD Graphs … 122
6.5.5 Time-frequency Resolution. … 124
7 ESTIMATION OF ROOM ACOUSTICAL PARAMETERS … 125
7.1 ISO 3382 PARAMETERS … 125
7.2 Acoustical Energy DECAy … 132
7.3 Spatial Acoustical Parameters … 135
Early Lateral Energy Measures … 135
Interaural Cross Correlation - IACC … 135
Measurement of Spatial Parameters … 136
Estimation of Spatial Parameters from Previously Measured Impulse Responses … 138
8 SPEECH INTELLIGIBILITY … 139
8.1 MTF - MODULATION TRANSFER FUNCTION. … 139
8.2 STI - SpeECH TRANSMISSION InDEX … 141
8.2.1 How IEC standard defines STI. … 141
8.2.2 Measurement of STI for unamplified speech signal. … 144
8.2.3 Measurement of STI for amplified speech signal. … 150
8.2.4 Measurement of STI at large distances … 151
8.3 STI AND %ALCONS … 151
9 TOOLS … 153
9.1 DiReCTIVITY PatTERNs … 153
9.1.1 Basic Definitions … 153
9.1.2 Types of Directivity Patterns in ARTA … 154
9.1.3 Creation of Directivity Patterns in ARTA. … 158
9.14 Format of DPF files … 162
9.15 Automatic recording of spatial group of impulse responses … 162
9.16 Exporting (spatial group) of frequency responses. … 164
9.2 INTEGRATING SPL MEASUREMENTS AND DATA LOGGING … 166
9.2.1 Basic Definitions of an Integrating SPL meter … 166
9.2.2 Working with ARTA SPL-Meter … 169
9.3 Octave Band SPL METER and NoISE Rating … 172
Noise Rating in Buildings … 175
9.4 THIRd Octave Band SPL and LOUDNESS METER … 179
A Zwicker Loudness Model … 182
9.5 Third Octave SPL and Loudness Time Record … 183
9.6 Two-CHAnNel Voltage LeVEl Meter and Third Octave Analyzer … 185
LITERATURE … 188
APPENDIX - MENUS, TOOLBARS AND SHORTCUTS … 191
1 Introduction
ARTA is a program for impulse response measurement, real-time spectrum analysis and real-time measurement of the frequency response. It is a tool for acoustical measurements and “point to point” testing of the audio quality in communication systems.
ARTA has functions of following measurement systems:
Impulse response measurement system with signal generators: periodic white noise, periodic pink noise, MLS, linear and logarithmic swept-sine.
Dual channel Fourier analyzer with signal generators: white noise, pink noise, periodic white noise and periodic pink noise.
Single channel Fourier analyzer with signal generators: periodic white noise and periodic pink noise.
Spectrum, octave band and THD analyzer with signal generators: sine, two sine, multitone, white noise, pink noise, periodic white noise and periodic pink noise.
Triggered storage scope with gated spectrum analysis and short-time Fourier transform.
Two-channel voltage level meter and third octave analyzer.
Note: Mode 2 and 3 can be also used for the estimation of the impulse response.
With calibrated microphone, ARTA can be used as virtual IEC class 1 SPL meter with real time modes:
Integrating SPL meter with 24 hours data logging,
Octave SPL meter with noise rating (NR, NC, PNC, RC, NCB),
Third octave SPL meter with report of specific loudness, loudness in Sone and loudness level in Phone.
ARTA is also a powerful analyzer of:
Gated frequency response,
Smoothed frequency response (in 1//n1 / n-octave bands),
Step response,
Impulse response envelope (ETC - curve),
Cumulative spectrum and burst decay waterfall curves and sonograms,
Operating systems: Windows Vista / 7 / 8 / 10 / 11
Processor class Pentium, clock frequency 1 GHz or higher, memory 2 GB for 32bit Windows or 4GB for 64bit Windows.
Full duplex soundcard with synchronous clock for AD and DA converters
WDM or ASIO soundcard driver (ASIO is trademark and software of Steinberg Media Technologies GmbH).
The installation of this software is simple: Take ARTA setup program and execute it or just copy the files “ARTA.exe” and “ARTA.chm” to some folder and make a shortcut to “ARTA.exe”. All registry data will be saved automatically at the first program execution.
Files with extension “.PIR” are registered to be opened with ARTA. They contain the data of the periodic impulse response (PIR) or signal time record. Results of other types of measurements (frequency response and spectrum) may be saved in ASCII formatted file, or as an overlay file. ARTA can export and import file in various formats (.wav, .tim and .txt).
ARTA does not dump graphs to the printer, instead, all graphs could be copied to the Clipboard and pasted to other Windows applications or saved as graphic files (.bmp, .png).
Windows treats a standard computer display as 96 DPI (dot per inch) device. Many modern displays have higher DPI resolution and Windows can treat them as 96 to 300 DPI device by allowing user to setup display scaling from 100%100 \% to 300%300 \%. If changes in DPI are not adequately implemented in an application, Windows scale application graphic size, by roughly scaling application window bitmap. From version 1.9.2 ARTA software is “High DPI aware” which means that DPI settings on windows startup determine size of ARTA windows elements. If Windows user changes display scaling, a restart or re-login is required, and ARTA will accept new DPI setting in all graphic operations.
1.1.1 Soundcards
ARTA has been used successfully with all soundcards that has WDM or ASIO driver. The compatibility with different operating system versions is driver dependent.
Soundcards are classified into three groups:
standard sound systems that are incorporated in the computer motherboard,
add-on sound cards for PCI or ISA bus,
sound systems that connects to the computer USB or Firewire interface.
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balanced microphone cables
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balanced cables or unbalanced stereo cables
unbalanced cables (coaxial cable)
unbalanced cables (coaxial cable)
XLR- female XLR - male TRS 6.3mm and 3.5 mm TS 6.3 mm and 3.5 mm RCA
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Table 1.1 Connectors and cables used in audio systems
Depending of the target user group, soundcards differs in type of input/output connectors and necessary cabling. Basic characteristics of connectors and cabling are given in Table 1.1.
Standard PC soundcards use stereo cables and mini-TRS connectors (Fig. 1.1).
Semi-professional high quality soundcards use RCA connectors and unbalanced connections (Fig. 1.2).
Professional soundcards use TRS 6.3 mm connectors for balanced connection, TS 6.3 mm connectors for unbalanced connection, and XLR (Cannon) connectors for balanced microphone connections (Fig 1.3).
Line-In (light blue)
Line-Out - Front speaker (green)
Mic In - Mono microphone (pink)
Out - Central speaker and Sub-bass (orange)
Out-Back speaker (black)
Out - Side speaker (grey)
Figure 1.1 Audio connectors on the PC motherboard (example for 5+15+1 surround sound system).
Standard PC stereo systems have three connectors ( 1,2 , and 3 on the motherboard). Surround 5+1 sound systems have additional three connectors ( 4,5 , and 6 on motherboard). One of the outputs is designed to drive headphones with nominal 32 Omega32 \Omega impedance. For soundcard testing we will use loopback connection of Line-In (blue) and Line-Out (green) using stereo cable terminated with miniTRS connectors. Input impedance of Line-In input on most PC soundcards is 10-20kOmega10-20 \mathrm{k} \Omega.
On laptops and notebooks, usually there are only headphone output and microphone input. Those systems are not appropriate for use with ARTA, as they cannot enable measurements in dual-channel mode because microphone input is a mono channel.
Figure 1.2 PCI card with RCA connectors (i.e. Terratec EWX24/96 or M-Audio Audiophile 24/96). There are separate connectors for left channel (in white color) and for right channel (in red color).
Figure 1.3 Professional sound system with Firewire interface, TRS and XLR connectors
Fig. 1.3 shows an example of the high quality Firewire professional sound system. On the front panel, there are two XLR microphone inputs. In the center of the XLR connector, a TS connector is inserted. It serves as music instrument input. Input impedance of instrument input is from 470kOmega470 \mathrm{k} \Omega to 1MOmega1 \mathrm{M} \Omega. Both inputs have volume control. Microphone inputs can be switched to phantom power, which gives power supply of 48 V to pins 2 and 3 of XLR microphone connector. Next, there is a master volume control for adjusting output level and input monitor level. Finally, there is a headphone volume control and a headphone stereo TRS connector. On the back panel, there are two balanced inputs, two balanced outputs, SPDIF optical connectors and two Firewire connectors.
1.2 Measurement Setup
In this document, we refer to following measurement setups:
Dual channel measurement setup
Single channel measurement setup
Semi dual channel measurement setup
Loopback for soundcard testing
A general measurement setup for system testing is shown in Fig. 1.4. The soundcard left line-output channel is used as a signal generator output. The left line-input is used for recording a D.U.T. output voltage and the right line-input is used for recording a D.U.T. input voltage. In a single channel setup, only a D.U.T. output voltage is recorded. In a semi dual channel setup the right line-input is used to measures the right line-output voltage. In a loopback setup, the left line-output is connected to the left line-input and the right line-output is connected to the right line-input.
Figure 1.4 General measurement setup for system response testing (D.U.T = device under test)
Setups for acoustical measurements are shown in figures 1.5, 1.6, 1.7 and 1.8.
Figure 1.5 Dual channel measurement setup for acoustical measurements
To protect the soundcard input from a high voltage that may be generated by the power amplifier, it is recommended to use a voltage probe circuit, as shown in Fig. 1.6. Values of resistors R1 and R2 have to be chosen for arbitrary attenuation (i.e. R1=8200 Omega\mathrm{R} 1=8200 \Omega and R2 =910 Omega=910 \Omega gives probe with -20.7 dB (0.0923)(0.0923) attenuation if the soundcard has usual input impedance -10kOmega-10 \mathrm{k} \Omega ).
Figure 1.6 Voltage probe with soundcard input channel overload protection
Figure 1.7 Single channel measurement setup for acoustical measurements
Figure 1.8 Semi-dual channel measurement setup for acoustical measurements
Figure 1.9 Loopback setup for soundcard testing
ARTA is also targeted for “point-to-point” testing of audio quality in communication systems. Figure 1.10 shows setup for testing such systems. Interface to mobile phones can be realized by using a headset I/O. Interface to the standard phone line (POTS) is shown in Fig. 1.11.
Figure 1.10 Measurement setup for testing communication systems
Figure 1.11 Interface from the soundcard I/O to the standard phone line (POTS)
ARTA can measure frequency and impulse response, distortions of sine, two-sine and multitone signals, estimate delays, echoes and speech transmission index. A special measurement technique, with an interrupted noise excitation, is applied to circumvent time-variant behavior of these systems (automatic gain control, noise reduction, voice activation).
1.3 A First Touch
When you start ARTA you will see the program window as shown in Figure 1.12. This window is called Impulse response window (Imp window). It will be primarily used to show the impulse response, also it will be used to show the time record of captured signals.
Figure 1.12 Impulse response window
By using the menu Mode, you can switch to three frequency domain windows for the real-time analysis:
Dual channel frequency response measurement window
Single channel frequency response measurement window
Spectrum analyzer window
The measurement mode may be chosen also by clicking the following toolbar icons:
IMP. Impulse response / Signal recording window
FR2- Dual channel frequency response measurement window
FR1- Single channel frequency response measurement window
SPA - Spectrum analyzer window
The Impulse response window is most important for a system response analysis. It will be described in more detail after we show how to analyze the spectrum and the system frequency response.
Now click these menus or toolbar icons to see how the measurement windows are working.
Figure 1.13 Dual-channel frequency response window - FR2 (the single channel frequency response window - FR1 looks the same)
Figure 1.14 Spectrum analyzer window
1.4 Audio Devices Setup
Before you start measuring you have to setup your hardware and audio devices by clicking the menu Setup->Audio Devices or by clicking the toolbar icon /_\0\boldsymbol{\triangle} \boldsymbol{0}. You will get the dialog box for the audio devices setup shown in the Fig. 1.15.
Figure 1.15 Dialog box for audio devices setup
The ‘Audio Device Setup’ dialog box has following controls:
In section Sound Card:
Soundcard driver - chooses the type of soundcard driver (WDM - windows multimedia driver or one of installed ASIO drivers).
Input channels - chooses the soundcard input stereo channels. ASIO driver can have large number of channels.
Output Device - chooses the soundcard output stereo channels.
Generally, user chooses input and output channels of the same soundcard (mandatory in ASIO driver mode).
Control panel button - if WDM driver is chosen, it opens Window Sound control panel. If ASIO driver is chosen, it opens ASIO control panel.
Wave format - chooses16-bit, 24-bit, 32-bit or Float. Float means IEEE floating point single precision 32-bit format. It is recommended to use 24-bit or 32-bit modes when using high quality soundcard (many soundcards are declared as 24 -bit, but their real bit-resolution is less than 16-bits). It is recommended to choose resolution type Float. This control has no effect in the ASIO mode, where a bit resolution has to be setup in the ASIO control panel.
Abstract
AD/DA transient time on startup and on sampling rate change - chooses time in milliseconds that sound capture will be delayed in order to eliminate transient signals and distortion that can arise on startup or on sampling rate change. After the first measurement with this delay, it will not be used until sampling rate changes. User decides on transient time value by monitoring low-frequency excess noise in sine spectrum, or by monitoring distortion in impulse response measured with swept-sine (see chapter 6.1.9). For those that just started using ARTA it is recommended to use the transient time value of 1500 ms in ASIO mode or 900 ms in WDM mode… Some high quality USB sound systems, like RME UC or Babyface Pro, have a very small transient time and values bellow or equal to 300ms are acceptable. If user set transient time to zero, it is recommended that on startup or after sample rate change user make one probe (false) measurement.
In section Input:
LineIn sensitivity - enters the sensitivity of the line input (i.e. peak voltage in mV that corresponds to the full excitation of the line input).
LR channel diff. - enters the difference between the level of the left and the right input channels in dB.
The best way to enter these values is to follow the calibration procedure as described in the next chapter.
In section Output:
LineOut sensitivity - enters the sensitivity of the left line output (i.e. peak voltage in mV that corresponds to the full excitation of the line output).
Power amplifier gain - if you connect the power amplifier to the line-output, and you need calibrated results in a single channel setup, you have to enter the power amplifier voltage gain, otherwise set gain to 1.
In section External preamplifier:
Left preamplifier gain - If you connect a preamplifier or voltage probe to the Left line input you should enter the gain of the preamplifier or probe attenuation in the edit box, otherwise set it to unity gain. Right preamplifier gain - If you connect a preamplifier or voltage probe to the Right line input you should enter the gain of the preamplifier or probe attenuation in the edit box, otherwise set it to unity gain.
In section Microphone:
Microphone used - check box if you use the microphone and want the plot to be scaled in dB re 20 muPa20 \mu \mathrm{~Pa} or dB re 1Pa. Also, use combo box to choose the channel where the microphone is connected (we strongly recommend to use the soundcard left channel as a microphone input channel).
Sensitivity - enters the sensitivity of the microphone in mV//Pa\mathrm{mV} / \mathrm{Pa}.
The setup data may be saved and loaded, by pressing the buttons ‘Save setup’ and ‘Load setup’. The setup-files have the extension ‘.cal’
Important notice: Please mute the line and microphone channels at the output mixer of the soundcard; otherwise, you might have a positive feedback during measurements. If you use a professional audio soundcard, switch off the direct or zero-latency monitoring of the line inputs. Many professional audio soundcards have their own program for adjustment of input and output channel, or have hardware control of input monitoring, and input and output volume controls.
1.4.1 WDM Audio Driver Setup
The operating system (also, sometimes in conjunction with control programs of professional soundcards) is responsible for setting soundcard native sampling rate and bit resolution. Operating system changes native resolution to floating point format for high quality mixing and eventually for the sample rate conversion.
For ARTA this means that it is strongly recommended to use resolution ‘Float’ and set the sampling rate to the native format. Access to these values is in ‘Windows sound control panel’, which user gets by clicking on button ‘Control Panel’ in ‘Audio Device Setup’ dialog.
Fig. 1.16 shows Vista/Win 7 control panel, that has four property pages.
As first step, user has to adjust Playback page and later repeat the same procedure for ‘Recording’ page. Adjustment steps are:
Click on channel info to choose the playback channel. It is not recommended to use the measurement channel as a default audio channel.
Click on button ‘Properties’ to opens channel ‘Sound properties’ dialog.
Click on the tab ‘Levels’ to open the output mixer (as in Fig. 1.17). Then mute Line In and Mic channels, if exist.
Click on the tab ‘Advanced’ to set the channel resolution and a sample rate (as in Fig. 1.20)
Repeat previous procedure 1) to 4) for recording channel and choose the same sampling rate as in the playback channel.
Figure 1.18 Setting the native bit resolution and sampling rate
1.4.2 ASIO Driver Setup
ASIO drivers are decoupled from the operating system control. They have their own control panel to adjust native resolution and memory buffer size. The buffer is used for the transfer of sampled data from the driver to the user program. User opens the ASIO control panel by clicking button ‘Control Panel’ in ARTA ‘Audio Device Setup’ dialog. Fig. 1.19 shows an example of ASIO control panel.
톨MU
Figure 1.19 E-MU Tracker Pre ASIO Control panel for setting bit-resolution and buffer size
In music applications user usually sets buffer size as small as it is possible for stable work. That gives the lowest input/output latency (system-introduced delay).
In ARTA, the latency is not problem, as it is encountered in software, but it is not recommended to use buffer with size larger than 2048 samples, or smaller than 256 samples. Some ASIO control panels express the buffer size in samples, while other express the buffer size in time [ms]. In that case we can calculate the size in samples using following expression :
buffer_size [samples] = buffer_size[ms] x samplerate[kHz] / number_of_channels.
ARTA automatically sets the buffer size for signal duration of 10 ms (i.e. 512 samples for sample rate 48kHz,102448 \mathrm{kHz}, 1024 samples for sample rate 96 kHz and 2048 samples for sample rate 192kHz).
ARTA always works with two input channels, and two output channels, treating them as stereo left and right channels. As ASIO support multichannel devices, user has to choose in a dialog box ‘Audio Device Setup’’ which pair of channels will be used in ARTA (1/2, 3//43 / 4, …).
Note: ARTA closes and releases ASIO driver when measurement stops, but if driver needs long time to be loaded in memory, ARTA keeps driver open all the time.
1.5 Calibration
Calibration is a process which defines mapping of internal digital values D[i]D[i] to external analog voltage values V[i]. Index ii denotes signal value sampled at time i / samplerate. For linear system this mapping is defined with single factor called sensitivity;
" Sensitivity "=V[i]//D[i],quad" where "|D[i]|<=1,quad i=0,1,2,dots\text { Sensitivity }=V[i] / D[i], \quad \text { where }|D[i]|<=1, \quad i=0,1,2, \ldots
In ARTA software discrete values D[i]D[i] are floating point values in the range from -1 to 1 . The unit of sensitivity is the Volt as D[i]D[i] is dimensionless. We will also use the unit mV . The above definition is also valid for RMS values of periodic signals;
Maximum possible value of discrete sequence D[i]D[i] is 1 . That gives us alternative definition of sensitivity as maximum peak value of voltage (or full scale value) that can be recorded (or generated) by digital instrumentation;
Sensitivity = Vpeak_max (Volts).
Menu command Setup->Calibrate devices opens the dialog box ‘Soundcard and Microphone Calibration’ shown on Fig. 1.22. That dialog enables setup of sensitivity for soundcard input and output channels. The same dialog box serves for calibration of microphone sensitivity. Microphone sensitivity defines mapping of sound pressure on microphone membrane to voltage generated by microphone. It has unit mV/Pa.
During calibration sampling rate can be set to 44100 or 48000 Hz , by using combo box at bottom of dialog box.
1.5.1 Calibration of Soundcard Output Left Channel
It is recommended to follow this procedure:
Connect the electronic voltmeter to the left line output channel.
Set ‘Output level’ control to -3dB or less.
Click the button ‘Generate sine ( 400Hz\mathbf{4 0 0} \mathbf{~ H z} )’ and program generates output sine signal with peak value that is 3 dB below full scale value (or other value set with Output level control). The button label changes to ‘Stop generator’.
Enter the voltmeter readout in edit box (in mVrms\mathbf{m V} \mathbf{~ r m s} ). (Note that rms value is 3 dB , or 1.414 times, lower than peak value). If you read the peak value from scope in the combo box choose ’ mV peak’.
Click the button ‘Stop generator’, then click the button ‘Estimate Peak Output mV’.
The estimated value will be shown in the box ‘Estimated’. Following equation is used for sensitivity:
If the generator output level was set to -3dB this value will be twice the rms voltmeter readout.
7. If you are satisfied with the measurement, click the button ‘Accept’, and the estimated value will become the current value of the ‘LineOut Sensitivity’. Also, it will be entered as a value for the input channel calibration.
Important note: Calibration is valid until we change the output volume control.
1.5.2 Calibration of Soundcard Input Channels
You can use an external sine generator or the output channel of the soundcard to calibrate the input channels. In both cases you should measure the value of generator output voltage.
If you are using the output channel of the soundcard as a calibrated generator:
Set the left line input volume to some value. Start with maximum volume or minimum gain if your soundcard has a built in preamplifier. Later you can calibrate for different preamplifier gain.
Connect the left output to the left line input channel.
Click the button ‘Generate sine ( 400Hz\mathbf{4 0 0} \mathbf{~ H z} )’ and monitor the input level at bottom peak-meters. If the soundcard input is clipping, lower the level of input volume. Alternatively, you can lower the generator level (but, then you need to measure output voltage Vrms again).
Enter the value of generator voltage in the edit box (but only if it differs from value used during output channel calibration (1.5.1)).
Click the button 'Estimate Peak Input mV\mathbf{m V} ', and program calculate sensitivity as ratio of Vrms / Drms.
If you are satisfied with measurements, click the button ‘Accept’, and estimated value will become the current value of the ‘LineIn Sensitivity’.
Repeat steps 1-6 for the right input channel.
This is the recommended procedure as it guarantees that you can connect the soundcard in loopback mode. If you want to calibrate the input channels with input volume control set to maximum, many soundcards require a reduction of the level of the output channel.
Important note: Calibration is valid until we change the input volume control.
1.5.3 Calibration of the Microphone
For microphone calibration you must have a sound calibrator. Then:
Connect the microphone preamplifier to the soundcard input (left or right).
Enter the preamplifier gain.
Attach the sound calibrator on the microphone.
Press the button ‘Estimate mic sensitivity’.
If you are satisfied with a measurement, press the button ‘Accept’.
Note: If you don’t know the preamplifier gain, you can set some arbitrary gain value (i.e. 1), but that value must be used as a preamplifier gain in the ‘Audio Devices Setup’ dialog box.
1.5.4 Frequency Response Compensation
The quality of the measurements depends on the quality of used sensors, i.e. microphones. It is possible to enter the frequency response of sensor in ARTA and make the compensation of their frequency response (by applying the inverse of sensor FR to measured FR).
The menu command Setup->FR compensation or click on icon gets the dialog box “Frequency Response for Compensation”, shown in Fig. 1.23. The dialog has a few controls and a graph that shows the frequency response which will be used for FR compensation.
The button Load opens the dialog for loading ASCII files that contain frequency response data. The file name must have extensions .MIC, .TXT or .FRD, and data entered in lines of text. Lines that start with a digit or dot characters must contain at least two values: first value is frequency in Hz and the second value is magnitude of frequency response in dB . The third value is optional. It may be the value of phase or any other text that will be treated as comment. All other lines are treated as comment. After successfully reading of the compensation file, the path of the file will be shown in the box below the graph.
For example, file “MB550-B.mic” (shown in Fig. 1.23) has content:
Figure 1.23 Typical frequency response of an electret microphone.
The check box ‘Show interpolated values’ enables us to see the interpolated FR curve that will be used in FR compensation.
The button ‘Copy’ copies current graph picture on Windows clipboard.
The combo list box ‘Range ( dB\mathbf{d B} )’ sets graph magnitude dynamic range (10-100dB).
The check box ‘Use for frequency response compensation’ enables/disables frequency response compensation.
The check box ‘Use for spectrum’ enables/disables spectrum magnitude compensation. This compensation is also used in harmonic and intermodulation distortions calculations.
For compensation of FR or Spectrum levels we use the equation:
Corrected level (dB)=(\mathrm{dB})= Measured level (dB)-(\mathrm{dB})- Compensation level (dB)(\mathrm{dB})
1.6 Rotating Turntable Driver Setup
The menu command Setup->Rotating Turntable opens the dialog box ‘Rotating Turntable Driver Setup’ shown in Fig. 1.24. It is used for setup of DIY-made turntable or Outline ET250-3D turntable that are usually used for automated polar diagram measurements.
Two types of drivers can be used:
External .exe file DIY-turntable driver,
Internal driver for Outline turntable ET 250-3D
Figure 1.24 Dialog box for setup of rotating turntable driver
1.6.1 External .exe file driver
A button command ’ <<<< Browse file’ opens a dialog for choosing the path and name of DIY driver .exe file. It is required that driver .exe file is a program that accept two types of command line arguments:
First type of command has an argument denoted -r and it resets turntable and sets current position as zero angle position.
Second type of command has argument an integer in range - 360 to 360 and it represent a command to rotate turntable to angle given by that argument.
1.6.2 Internal driver for Outline turntable ET 250-3D
ARTA has built-in driver for the ET250-3D Outline turntable. This turntable has to be connected to the ethernet network port. To setup turntable driver user has to enter three strings: ET 250 network IP address, PC local IP address and port number. Values for IP addresses can be obtained with command line program that is delivered with the turntable setup program.
After entering IP addresses user should press button ‘Init ET250 network connection’ and optionally check the box ‘Use acceleration’ if needs faster (but more demanding) turntable rotation.
Note: Manufacturer of Outline turntable changed port numbers for units sold after January, 2021. Allowed port numbers are: 6667, 6668 and 6669. Ports numbers 6665 and 6667 were valid for older units.
Note: Before use in ARTA execute the delivered setup program and assure that turntable is working.
1.6.3 Testing of turntable driver
To test turntable driver two commands are available:
Resetting turntable is done by pressing the button ‘Set current position as zero degree position’.
Rotating turntable to angle (value -360 to 360 ), that is entered in the edit box, is done by pressing the button ‘Rotate to angle’.
1.7 Getting Images of Graphs and Windows
Images of graphs and windows can be copied to Windows clipboard or saved to the file in a three image formats: .png, .bmp and .jpg. It is recommended to use .png format.
Obtaining copy of the full window picture is simple. The user needs to simultaneously press keys Ctrl+P. After that command the window picture will be saved in the System Clipboard. From there the user can paste it in other opened Windows applications (MS Word, MS Paint). Keys Ctrl+Alt+P activate command to save that image in the file.
To copy or save the graph picture, that is shown inside the window, user needs to simultaneously press keys Ctrl+C\mathbf{C t r l}+\mathbf{C} or activate the menu command ‘Edit->Copy’, or press appropriate ‘Copy’ button. In the main window toolbar, the ‘Copy’ button is shown as toolbar icon 1 .
Figure 1.25 Dialog box ‘Copy / Save Image with Extended Information’
The Copy command opens the dialog box ‘Copy/Save Image with Extended Information’, shown in Figure 1.25. Here the user has to set up the following options:
By using the combo box above ‘OK’ button, user chooses one of three modes of saving the image: Copy to Clipboard, Save to File and Save to File + Copy to Clipboard.
In the Edit box user optionally enters the text that will be appended at the bottom of the graph.
Check box ’ Add filename and date’ enables adding text to the graph that shows file name, date and time. If overlay curves exist their names and line color signs are added at the bottom of the graph.
Check box ‘Save text’ enables saving entered text for the next copy operation.
Combo box ‘Aspect ratio’ enables copying of graphs with fixed aspect ratios: 3:2 and 2:1.
Bitmap size is chosen by selecting one of following combo box items:
Current screen size - user adjusts graph width and height
Smallest - graph width is 500 points
Small - graph width is 600 points
Medium - graph width is 800 points
Large - graph width is 1000 points
X Large - graph width is 1200 points
XX Large - graph width is 1500 points
XXX Large - graph width is 2000 points
The size greater than Large will give publication style quality if graphs are drawn with thick lines and grids. Thick lines and grids are drawn with width of 2 points. User selects thickness in every graph window by menu command Edit->Thick lines and Edit->Thick grid.
The button ‘OK’ copies the graph to the system clipboard or opens dialog to enter name of file in which picture will be saved. The button ‘Cancel’ cancels the copy operation.
2 The Spectrum Analyzer
The spectrum analyzer of ARTA is implemented as a real-time FFT based spectrum analyzer. A builtin generator provides the following signals: sine, two sine, square, triangle, multitone, white noise, pink noise, periodic white noise (PN white), periodic pink noise (PN pink) and periodic speech noise (PN speech).
Working with the spectrum analyzer will be explained through the soundcard testing procedure.
2.1 Soundcard testing
The easiest way to test the quality of the soundcard is in the Spectrum analyzer mode.
Make the loopback connection for the soundcard testing.
Click the menu Mode -> Spectrum Analyzer or click the toolbar icon SPA.
Click the menu Generator->Setup or click the toolbar icon ∼\sim. You will get the dialog box shown in Fig. 2.1.
Figure 2.1 The dialog box for the signal generator setup
This dialog box has following controls:
Two sine generator section allows the choice of three possible combinations of frequencies and magnitude ratios:
Def1 - sets f1=13kHz,f2=14kHz\mathrm{f} 1=13 \mathrm{kHz}, \mathrm{f} 2=14 \mathrm{kHz}, amplitude ratio 1:1.
Def2 - sets f1=100Hz,f2=8kHz\mathrm{f} 1=100 \mathrm{~Hz}, \mathrm{f} 2=8 \mathrm{kHz}, amplitude ratio 1:4.
User - enters two sine frequencies and the amplitude ratio .
Note: the two sine signal has the peak level defined in the Sine generator section - Peak Level control.
Multitone and noise generator section:
Output volume - chooses the output level re full scale level in range 0 dB to -50dB.
PN Pink cut off - enters the low frequency cut-off in Hz , for the periodic pink noise.
Speech - chooses the type of speech spectra: Male[2011], Female[2011] and Male[2020]
(number in brackets denotes year of publishing IEC standard. 60268-16).
Multitone - chooses the type of multitone signal:
Audio range -1//-1 / R octave spaced sine signals in range up to 20 kHz , A//D\mathrm{A} / \mathrm{D} sampling range -1//R-1 / \mathrm{R} octave spaced sine signals in range up to half sampling rate,
Speech range - composite signal for testing GSM audio in range 100 to 8000 Hz ,
ITU-T O. 81 - signal for testing telephone audio quality,
Sine + Square - signal for testing transient intermodulation distortion.
Resolution - chooses tone density from 1 to 12 per octave in Audio and A/D sampling range.
Range (Hz) - sets Audio range lowest and highest frequency;
Lowest frequency can be set from 5 Hz to 1000 Hz .
Highest frequency can be set from 2000 Hz to 20000 Hz .
Use output filter - if checked, the output filter is applied on generated Audio range multitone signal.
Set filter - opens dialog for the definition of output filter.
Note 1: PN (periodic noise) is a periodic, noise-like signal with a controlled spectrum level and a random phase. The periodic noise and multitone are belonging to the class of multisine signals (to be explained later).
Note 2: Jitter test signal is a sine signal with a frequency equal to 1//41 / 4 of the sampling rate, and with a LSB bit toggled with a frequency equal to 1//1921 / 192 of the sampling rate.
Note 3: Multitone test signals contain mix of sine signals with different amplitudes and phases. Their use will be explained later.
Now choose parameters of sine generator:
Frequency: 1000 Hz
Peak level: -3 dB
Dither level: 16 bit.
4. Using the dialog bar choose:
Gen: Sine
Fs (Hz): 48000
(sampling frequency or sampling rate)
FFT: 16384
Wnd: Kaiser5
Avg: None
(number of samples in FFT analysis frame)
(signal window to suppress leakage in FFT analysis)
(averaging of the signal)
The same parameters can be set up in a dialog box ‘Spectrum Analysis Setup’ shown in Fig. 2.2. (you get it by clicking the menu Setup->Measurement). By using this dialog box you set (1) the preferred input channel, (2) averaging parameters and (3) the FFT resolution.
Figure 2.2 The spectrum analysis setup
This dialog box has following controls:
Input channel section:
Combo box chooses left or right channel as active soundcard input channel.
Averaging section sets:
Type: None, Linear, Exponential or Peak Hold.
Max. averages: the maximum number of averages.
FFT resolution section sets:
FFT size: number of samples in FFT block (4096, 8192, 16384, 32768, 65536 and 131072),
Window: Uniform, Hanning, Blackman3, Blackman4, Kaiser5, Kaiser7 or Flat Top window. Sampling rate: 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200 or 96000 Hz .
5. Choose: Input channel: Left.
Prepare the Windows sound mixer:
Enable the line-input channel
Mute the line-input channel in the output mixer.
Set the line-out volume for maximum output sensitivity.
Set the line-input volume near minimal input sensitivity.
By menu command Setup->Spectrum Scaling ( bar(ABC)\overline{\mathrm{ABC}} ), or by clicking right mouse button in graph title area, you get the dialog box ‘Spectrum scaling’ (shown in Fig. 2.3). Use this dialog box to set (1) the magnitude scaling, (2) the power weighting and (3) distortion measures.
Figure 2.3 The spectrum scaling
Scaling section:
Magnitude scaling: dBFS (dB re full scale),
dBV or SPL (sound pressure level),
PSD (power spectral density mode in dBV/\sqrt{}{Hz}\mathrm{ ).}\\mathrm{ .}.
Voltage units: dBV or dBu,
Pressure units: dB re 20u Pa or
dB re 1 Pa (valid only if microphone is connected and enabled).
Power section:
Power Weighting combo box - chooses: None, A, B or C filter for weighted signal power estimation. Show RMS level - check to show the power level at the bottom of the graph.
Distortion section:
THD - check to show total harmonic distortion (THD) in sine response testing. THD+N\mathbf{T H D}+\mathbf{N} - check to show total harmonic distortion + noise in sine response testing.
IMD - check to show intermodulation distortion (IMD) in two sine response testing, or transient intermodulation distortion (DIM) in square+sine multitone testing.
Multitone TD+N - check to show total distortion + noise (TD+N)(\mathrm{TD}+\mathrm{N}) in multitone response testing.
Normalize with full power - check to get THD normalized with signal power including higher harmonics. Low cut-off (Hz) combo box - sets low frequency cut-off in THD+N measurements. 2^("nd ")2^{\text {nd }} and 3^("rd ")3^{\text {rd }} order IMD - check to show 2^("nd ")2^{\text {nd }} and 3^("rd ")3^{\text {rd }} order IMD defined in SMPTE, DIN, CCIF and IEC standards.
Frequency weighting - check to use frequency weighting (A,B,C) in THD +N and TD +N measurements.
8. Check following check boxes: THD, THD+N and Show RMS level .
9. Start recording by clicking the toolbar icon (or via menu Recorder->Run). You should get a response like the one shown in Fig. 2.4. This figure can be obtained by the copy/paste operation (menu Edit->Copy).
Slowly increase the volume of the line input channel (using the soundcard mixer) until you get the peak level close to -3dB FS.
Figure 2.4 Spectrum of 1 kHz sine generator of the soundcard Terratec EWX 24/96 in loopback setup. Signal window: Kaiser5, FFT size: 16384, Fs: 48000 Hz.
The bottom of Fig. 2.4 shows the spectrum value at the cursor position (frequency and magnitude), RMS level and distortions. The cursor is drawn as a thin line that can be moved by pressing left mouse button or by pressing keyboard’s left and right keys.
If you get THD+N lower than 0.1%0.1 \% you have a usable soundcard.
If you get THD+N lower than 0.01%0.01 \% you have a good soundcard.
Note: During the measurement you can use the control bar to change the averaging type, reset averaging, change the sampling frequency, change the type of an excitation signal and an FFT size. You can change any plot parameters (dynamic range, frequency range and axis) from dialog box ‘Spectrum graph setup’ (you get it by clicking the menu Setup-> Graph setup or by clicking right mouse button in the plot area). The easiest way to adjust graph margins is by using Right Control bar. Functions of bar buttons are explained in Figure 2.5.
Top
-
Changes graph top magnitude margin
✓\checkmark
Fit
Fit plotted curve to graph top magnitude margin
Range
-
Changes graph magnitude range
Set
Opens dialog box for the setup of graph margins
FrHigh
- *\cdot
Changes graph high-frequency margin
FrLow
*\cdot*\cdot
Changes graph low-frequency margin
Top
- Changes graph top magnitude margin
✓
Fit Fit plotted curve to graph top magnitude margin
Range
- Changes graph magnitude range
Set Opens dialog box for the setup of graph margins
FrHigh
- * Changes graph high-frequency margin
FrLow
* * Changes graph low-frequency margin| Top | |
| :---: | :---: |
| - | Changes graph top magnitude margin |
| $\checkmark$ | |
| Fit | Fit plotted curve to graph top magnitude margin |
| Range | |
| - | Changes graph magnitude range |
| Set | Opens dialog box for the setup of graph margins |
| FrHigh | |
| - $\cdot$ | Changes graph high-frequency margin |
| FrLow | |
| $\cdot$ $\cdot$ | Changes graph low-frequency margin |
Figure 2.5 Control bar for graph margins setup (also used for Frequency Response windows)
Note: Useful shortcuts to change the top graph magnitude margin are “Up” and “Down” keys and the mouse scroll wheel (they move the plot up and down).
2.2 The Spectrum Estimation Procedure
The spectrum shown in Fig. 2.4 is obtained by following procedure:
An input signal is sampled with frequency f_(s)f_{s} and transformed into discrete sequence x_(n)x_{n} of length N=FFTN=F F T size (the number of samples in the acquisition window is equal to the ‘FFT size’, and can be set to: 4096, 8192, 16384, 32768, 65536 or 131072).
The discrete input sequence is multiplied with a window sequence w_(n)w_{n} (will be explained later)
The Discrete Fourier Transform
X_(k)=sum_(n=0)^(N-1)w_(n)x_(n)e^(-j2pi kn//N)X_{k}=\sum_{n=0}^{N-1} w_{n} x_{n} e^{-j 2 \pi k n / N}
is calculated using the FFT algorithm. It gives spectral components as complex values at discrete frequencies
f_(k)=k Delta f,f_{k}=k \Delta f,
where Delta f\Delta f is a DFT spectral resolution
Delta f=f_(s)//N.\Delta f=f_{s} / \mathrm{N} .
For real signals, there are N//2N / 2 single sided power spectral components G_(k)G_{k} :
{:[G_(0)=|X_(0)//N|^(2)-quad dc" component "],[G_(k)=2|X_(k)//N|^(2)","quad k=1","2","..N//2-1]:}\begin{aligned}
& G_{0}=\left|X_{0} / N\right|^{2}-\quad d c \text { component } \\
& G_{k}=2\left|X_{k} / N\right|^{2}, \quad k=1,2, . . N / 2-1
\end{aligned}
The magnitude spectrum is shown in one of following scaling modes:
Scaling mode
Level
Units
Peak level
(ref. full scale)
Peak level
(ref. full scale)| Peak level |
| :--- |
| (ref. full scale) |
Note 1: If a signal window w_(n)w_{n} is applied, then spectrum values X_(k)X_{\mathrm{k}} are divided by a scale factor that is equal to window w_(n)w_{n} average value w_(AVG)w_{A V G} - in a RMS level mode, or a window w_(n)rmsw_{n} \mathrm{rms} value w_(RMS)w_{R M S} - in a power spectral density mode.
w_(AVG)=(1)/(N)sum_(n=0)^(N-1)w_(n),quadw_(RMS)=sqrt((1)/(N)sum_(n=0)^(N-1)w_(n)^(2))w_{A V G}=\frac{1}{N} \sum_{n=0}^{N-1} w_{n}, \quad w_{R M S}=\sqrt{\frac{1}{N} \sum_{n=0}^{N-1} w_{n}^{2}}
Note 2: If the check box Use Microphone is enabled in dialog box ‘Audio device setup’, then RMS or PSD levels are raised by 20log_(10)(2xx10^(-5)(Pa))xx20 \log _{10}\left(2 \times 10^{-5} \mathrm{~Pa}\right) \times microphone_sensitivity (mV//Pa)(\mathrm{mV} / \mathrm{Pa}) ).
5. The spectrum plot shows levels of spectral magnitudes as line-graph.
Note: A DFT spectrum is defined at discrete set of frequencies, so it would be more appropriate to show the spectrum as a discrete bar-graph, but when we deal with large number of spectral components, as is the case in ARTA, a line-graph gives better visual insight of spectral magnitudes.
The bottom of Fig. 2.4 shows:
RMS - RMS level of an input signal - defined as 10log_(10)10 \log _{10} (sum of all DFT power spectrum components). If the power weighting, in the ‘Spectrum Scaling’ dialog box, is set to A, B or C filter, then each spectral component is weighted, before the spectrum summation, with a magnitude response of A, B or C filters (for definition of these filters see section 2.4).
THD - total harmonic distortion - defined as percentage of the square root of ratio of power sum of higher harmonics (H_(2),H_(3),..)\left(H_{2}, H_{3}, ..\right) to the power of fundamental signal harmonic (H_(1))\left(H_{1}\right).