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SR1 Operations Manual - Stanford Research Systems

1. If the user presses OK the function returns the string corresponding the selected item i e A B or C in the example If the window times out the string timedout is returned If the user presses Cancel the string cancelled is returned UserChoiceMulti The UserChoiceMulti function also lets the user select among different choices but using a checkbox style selection User Select Some Choices a Multiple selections are possible In the example above for instance the function would return the string A B As with UserChoice the function returns cancelled if cancelled is pressed and timedout if the timeout limit is reached before the user takes any action UserLaunchChoice Presents a series of buttons with corresponding explanations x SR1l Instrument UserLaunchChoice Some Choices A B C Performs A Performs B 2014 Stanford Research Systems SR1 Operation 249 User Launch Some Choices Performs 4 Performs B Performs The function returns A B or C if the corresponding button is pressed and cancelled or timedout if the window is closed or no action is taken within the timeout interval UserLoadFile and UserSaveFile These functions display the standard Windows Load File and Save File dialogs x SR1l Instrument UserLoadFile Select File 200 COMGUID dtd EQcurve dtd aimdhvays xml HE j A ajmIFOFfline xml O
2. Switcher Config Daisy Chains Networks BNC Oukput G OutpukSwitcho 4 OutputSwitch1 EMC Input SLR Oukput ALR Input 7 BalancedOutputSwitch Mew Switch Close The Networks tab of the Switcher Configuration Panel displays the attached switchers by function BNC ouput BNC input XLR output and XLR input SR10 switchers are used for XLR inputs and SR11 switchers for XLR outputs while SR12 switchers can be used for BNC inputs and ouputs Opening the node corresponding to one of the switches displays the state of each connector on that switch box OukputSwiktchi i 4 gt 4 gt il gt 11 i2 gt 12 In this output switch example output A is connected to connectors 1 2 and 3 and output B is connected to connectors 4 5 and 6 Double clicking on the one of the connectors toggles its status between no connection icon is dark connected to the A output icon is green and connected to the B output icon is red The connection status can also be set by right clicking on one of the connector icons Note that for an output switch the output can be simultaneously connected to several of the switchbox connectors For an input switcher only one of the connectors can be connected to the input at a time The numbers to the right of the connector icons identify the physical connector ID for that switch 2014 Stanford Research Systems SR1 Operation 22 the first number and the logical
3. 2014 Stanford Research Systems e SR1 Operation Manual Phased Sines Config Phased Sine aveform On iw a ine AME feo o0 mFFS Fre 0 997000 kHz Phase 0 000 3 The phased sine waveform consists of two sines one on channel A and one on Channel B with a specified phase difference between them This waveform may not be combined with other waveforms Phases may be entered in any 360 interval for instance 0 to 360 or 180 to 180 specified on the Preferences Panel Noise Config Moise aveform ilter On fw 1 3 Oct x gise fi 00000 kHz Arig s00 00 mFFS eer Repeat 7 Ripe j100 00 msec The noise waveform outputs random noise with an almost gaussian amplitude probability density function with several options A true gaussian probability density function has finite probability for any amplitude no matter how large which is impractical for a physical device Amplitude controls as with all SR1 waveforms the peak value of the noise output The crest factor of the noise waveform i e the ratio of the peak value to the rms value of the noise is approximately 4 The Repeat checkbox governs the repetition interval of the generated noise If not checked the repeat interval is sufficiently long that it won t observable under most conditions If however it is desired for the noise to repeat after a fixed interval check the Repeat box and enter the Repeat interval below i
4. The size of the digitizer record specified in 80MHz samples is set with the Record Length control Values from 8 kSamples to 2 MSamples can be selected To set a scale for the record length recall that the frame rate for a digital audio signal is equivalent to the embedded audio sample rate So for a 48 kHz sampled digital audio signal each frame takes about 20 8 us or about 1 6 kSamples Thus there are about 4 9 48 kHz frames in an 8 kSample digitizer record and 1258 48 kHz frames in the longest 2M sample record Each frame contains two sub frames of 32 bits for a total of 64 bits and each bit cell contains 2 unit intervals Uls for a total of 128 Uls in each frame Therefore at a 48 kHz sample rate each UI contains about 13 digitizer samples This number grows proportionally larger 26 samples UI at Fs 24 kHz at lower sample rates and smaller 3 25 samples UI at Fs 192 kHz at higher sample rates Choosing longer records increases the acquisition and analysis time for each record but provides more frequency resolution in spectral measurements Digitizer Input Gain The digitizer has a selectable front end gain of x1 x2 and x4 The magnitude of the input relative to the digitizer s full scale is shown by the colored bars at the bottom of the Acquisition box Blue indicates the Signal is below 1 2 of full scale Green indicates the signal level is optimally adjusted while red indicates the digitizer is overloaded Usually setting the
5. 2014 Stanford Research Systems SR1 Operation 113 Signal there are 2 pulses used to represent each bit and there are 32 bits allocated for each sample and 2 channels there are a total of 128 Uls in the time required to transmit one sample of both channels Therefore the relationship between the UI and the digital audio sampling frequency is 1 UI sec 1 128 Fs Hz For frequency domain jitter measurements power spectral density PSD units are available as well The jitter PSD units are s VHz dBs VHz and dBc Hz The relationship between the jitter expressed in dBc VHz and s VHz is given by the following relation See AES 11id jitter density in s VHz V2 2rrf_ 10 0tter in dBc Hz 20 where f is the carrier frequency Analyzer Frequency Units The following table describes the units used by SR1 in display measured freqeuncies All frequency units except Hz make use of the Frequency Reference which is set in the References Tab of each analyzer The fundamental unit of frequency 1 Hz 1 cycle per second F R Ratio relative to the Frequency Reference A value of 3 F R with reference of 2 kHz gives a waveform frequency of 6 kHz Difference relative the Frequency Reference A value of 500 dHz with a reference of 2 kHz gives a waveform frequency of 2 5 kHz Percent of the Frequency Reference A frequency value of 50 Fref with a reference value of 10 KHz gives a waveform frequency of 5 KHz A cent is a logarithmic
6. 101 dB 1 uV 22 kHz BW O to 360 0 001 resolution SMPTE DIN CCIF DFD DIM TIM White Pink Filtered White Pink USASI Maximum Length Sequence from 2 to 2 samples repetition interval 1 to 50 tones individually adjustable in amplitude phase and frequency Equal power in each FFT bin Frequency response can be modified with an EQ file Swept sine with a logarithmically increasing frequency Used for impulse response measurements 298 SR1 Operation Manual Square 10 Hz to 50 kHz frequency range Ramp Fs N frequency range N 20 adjustable rise fall fraction Arbitrary 256 Samples to 136k Samples Polarity 10 Hz to Fs 4 frequency range Constant Offset DC to 20 Vp unbal 40 Vp bal Burst Types Timed ext triggered ext gated All waveforms can be bursted asynchronously Sines can be bursted synchronous with sine zero crossings Digital Audio Signal Generator Digital Audio Carrier Characteristics Output Amplitude Balanced Range 16 mV to 10 24 V 110 load Accuracy 10 80 mV Unbalanced Range 4 mV to 2 55 V 75 load Accuracy 10 20 mV Output Format Balanced XLR AES EBU dual connector XLR unbalanced BNC SPDIF EIAJ dual connector BNC Optical Toslink Output Sample Rate 24 kHz to 216 kHz Sample Rate Accuracy 5 ppm Output Impedance 110 balanced 75 unbalanced Digital Audio Waveforms Sine Frequency Range 10 Hz to Fs 2 Frequency Resolution lt p Flat
7. Log sine Chirp Config a Wiaveror Om Ww Chirp Amp 1 0000 rms Compliance She AD FFT1 w The log sine chirp waveform is designed to work in combination the dual channel FFT analyzer to make impulse response measurements The log sine chirp waveform is a sinusoid whose frequency is swept in a logarithmic fashion over the frequency span of its associated FFT analyzer The log sine chirp waveform has a pink frequency roll off of 3 dB oct The log sine chirp waveform is synchronized to the settings of a particular FFT analyzer If the resolution or frequency span of that analyzer changes the log sine chirp waveform automatically reconfigures to 2014 Stanford Research Systems SR1 Operation sweep over the frequency span of the selected analyzer Because of the synchronous nature of the chirp signal a uniform window should be selected in the analyzer when using the log sine chirp waveform Selection of the associated FFT Analyzer is done with the Chirp Source control Chirp Source Associated Analyzer AO FFT1 Analyzer 0 Single Channel FFT Analyzer AO FFT2 Analyzer 0 Dual Channel FFT Analyzer A1 FFTO Analyzer 1 Single Channel FFT Analyzer A1 FFT1 Analyzer 1 Dual Channel FFT Analyzer If the analyzer span is set to frequencies which are unreachable with the current generator sampling the Compliance led will glow red and no waveform will be output Be sure to select the Uniform window whe
8. Bar Chart Settings Panel Settings Chart Meas X axis Min 47 969 kHz Max 46 035 kHz E SF le Log 9 Num Ony J Bin Mini Mase wt Avgisdew w Range Readout Min 0 0000 Hz fe MiniMax Co Avgrsdey Max 0 0000 Hz On Alarm Ck The Minimum and Maximum controls set the bar chart scale as well as setting the units for both the numerical display and the bar chart The SF significant figures control sets the number of significant figures in the numerical display Log sets a logarithmic scale for the bar chart Selecting Num Only drops the unit string from the numerical display e g instead of 1 043 kHz the display will read 1 043 k Minimum Maximum LA N 4736k 4 5 00k 43 02k Red Bar shows current value 1 Std Dev Min Max and Avg Sdev turn on and off the respective markers on the bar chart display When Min 2014 Stanford Research Systems 204 SR1 Operation Manual Max is selected the minimum and maximum recorded values of the measurement will be displayed on the bar chart as pink lines When Avg Sdev is checked the average value of the measurement mean will be displayed as a solid black line with the range of 1 standard deviation shown as a dotted blue line The current value of the measurement is always shown as a red bar Alarm Range The bar chart display has the ability to check if the displayed measurement falls outside a pre set range The range i
9. Optional Input Filters Analog Inputs Hi Res Converter Optional Filters h 4 Filter Selection None Filter 1 Filter 2 Fiter3 Filter 4 None Fiter 1 Filter 2 Filter 3 i Filter 4 h B Filter Selection The final tab on the Analog Inputs Panel contains the selection controls for the Optional Filters on the analog input boards See the SR1 Hardware Reference for details on the optional filters 2014 Stanford Research Systems SR1 Operation 2 3 4 Digital I O Panel SR1 s digital audio inputs and outputs work with both the AES EBU and S PDIF standards for digital audio over a wide range of sampling rates from 24 kHz to 216 kHZ SR1 also makes a variety of digital audio carrier measurements including carrier amplitude sampling freqency and output input delay Finally to test susceptibility of devices to carrier degradation SR1 provides number of different types of Carrier impairments including jitter common and normal mode interference and variable rise times All these functions are controlled from the three tabs of the Digital I O panel eI Carrier Status Qukouk Impairment Output onfig SOUrCe BMC t Dual conn Optical BMC Amp 2 000 Vpp FS Invert 7 ample Rate reEmphasis Resolution j48 000 kHz None z 24 E bits Input onfig E ctr BNC i E Dual Conn EQT Sgr Ww Term E ample Rate ignal Generator Fs z Cpl oc z Resi fe
10. ato sl TE E Tn cat Cole a OS PreRecorded PreRecorded Source MA Source Channel Na Channel Make Sampling Freq level o Clock Accuracy Not Indicated Word Length b k Original Fs Sampling Freq Clock Accuracy Word Length Original Fs jz 1s a H a Channel Status Panel Consumer Mode SR1 Operation Manual Channel Status Bits m is mB fe ANB Mode Invalid 7 ee ModefInvalid W Audio Sample Linear PCM Audio Sample C Consumer f Professional i i Emphasis Mo Emphasis Emphasis Linear PCM E fio emphasis ock Ce Lock Lock Mot Indic ch Mode Mok Sampling Freq kak Sampling Freq 48 0 kH 1 001 z Channel Mode Mot Indicated Channel Mode Mo User Info 20 bit Mot Def l l a E eae User Bits 20 bits Nat a Referer la A AER uxiliary Bits 20 bits Not Def uxiliary Bits Under Mich Md li Audio Word Len 20 bits Audio Word Len Src sri Dest T Add Code Inc Time Code Tnc Mutlichannel Md Undefined no MutlChannel Md a a a Reference Sig Not a Reference Reference Sig v OST cate sc MMN EE cs A con Conf 14 17 0 Conf 16 21 CREO Correct SE OEE ET E E e EE E ee E ggs jo da ge do e e e e e e re ee geo de e e o e o o e o e bjs joo js foo foo foo fs fe fat fou fe fe p fon foo foo foo fo fon fon o fo fo fo Rb SSCS Ee ee ee ee eee eee l2 l3 Is 16 17 1b 19 z l oe ai a ES
11. 20 SR1 Operation Manual submenu the corresponding panel appears For instance after selecting the Linearity function the Linearity panel will be displayed Linearity Trace Al FFT Power Spectrum 4 Stark End lt r 0 0000 Hz 23 977 kHz Cursors Close Calculate This panel displays several important features common to all the trace calculator functions The single input trace is displayed at the top of the panel Because linearity takes only one input argument no mechanism for selecting the input trace is provded it is simply the active trace If the selected function requires a second input trace a drop down list appears showing the available selections The controls below the input trace determine the range of data within the trace on which the calculation will be performed The default is to operate on the entire data range However the user may enter a specific starting and ending X values for the calculation range Pressing the Cursors button sets the starting and ending values according to the position of the two cursors Pressing Close aborts the calculation Pressing Calculate performs the calculation The result of the calculation will always be a new offline trace For instance after pressing Calculate in the example above a new trace appears in the trace listing box as shown below SFR Power Soectrum amp SA ean Se ea wer Saeckrian A Note that the trace calculator does not
12. Aliasing arises in sampled data systems because the real world filters used to protect A D converters have finite cutoff slopes Each of the SR1 s A D converters is protected with either analog or a combination of analog and digital anti aliasing filtering For each converter there is some range of frequency below Fs 2 that is not fully protected to the level of the ultimate attenuation of the anti aliasing filter When using the zoom feature of the FFT3 analyzer the analyzer applies digital anti aliasing filter after each decimation stage to meaning that there is a small region at either end of the analysis range that is not alias protected to the full attenuation of the digital anti aliasing filter When M Show Aliased Lines jig checked the analyzer displays all lines in the full analysis range of the FFT without regard to whether they are fully alias protected This setting is often perfectly useful for many typical audio measurements However when aliasing is potentially a problem the box can be unchecked in which case only FFT lines that are fully alias protected are displayed Areas of the spectrum that are not fully protected are shown in red in the graphical frequency indicator on the FFT panel Resolution Resolution 4cq Timet ik 16 0 msec ad 32k 512 msec 4 16k f 256 msec 5k 128 msec 4k 64 0 msec k lk i 16 0 msec 200 i 4 00 msec The resolution control determines the number of lines in the computed
13. Enable Rise Fall Time 5 0000 nsec CRE Lock Validity Conf Coding Parity Invert inverts the polarity of the digital audio output carrier signals Properly operating digital audio equipment should be immune to an overall polarity inversion 2014 Stanford Research Systems SR1 Operation 85 Invalid A or B sets the Validity bit bit 28 in each corresponding subframe of the digital audio output signal The validity bit has no fixed meaning but has been used to signal conditions including non audio data non linear pcm data overload etc Cable Sim applies the cable simulator circuit to the digital audio output BNC and XLR The cable simulator is a low pass filter designed to mimic the signal degradation caused by approximately 1000m of cable See the discussion of EQ for an illustration of the effect of the cable simulator Checking Common Mode Sine applies a sinewave to both pins 2 and 3 of the XLR outputs The BNC outputs are not affected The sinewave is variable in frequency from 10 Hz to 100 kHz and in amplitude from 0 Vpp to 20 Vpp Such a signal is useful for testing the immunity of digital audio equipment with balanced inputs to common mode pickup Normal Mode Noise can be applied to the XLR and BNR outputs Normal mode noise is white noise with a bandwidth of approximately 20 MHz For BNC outputs the noise is simply added to the output For XLR signals the signal is added as a balanced signal with positive polarit
14. Occurs when the Unlocked status of the Occurs when the Unlocked status of the digital audio input changes audio input changes BiPhase Change Occurs when the BiPhase Error status of the digital audio input changes Parity Change Occurs when the Parity Error status of the digital audio input changes Digital Channel Status Pro Consumer Occurs when the received channel status format changes from Professional to Change Consumer or vice versa Copyright Bit Occurs when there is a change in the status of the Copyright bit in the received Change consumer channel status Emphasis Occurs when there change in the emphasis setting for the received consumer or Change professional channel status CRC Change Occurs when there is a change in the CRC error status for the received professional channel status CSA Byte0 4 Channel status is sent in both consumer and professional formats in 23 byte Change blocks SR1 allows up to 5 bytes to be watched i e any change in the received channel status in that byte triggers the corresponding event The watched bytes are configured in the Config tab of the channel status panel These event occurs when there is a change in the received channel status for channel A corresponding to the configured byte Analog HiV Trip CSB Byte0 4 Channel status is sent in both consumer and professional formats in 23 byte Change blocks SR1 allows up to 5 bytes to be watched i e any change in the received
15. SRI Audio Analyzer Operation Manual R 5 Stanford Research Systems Audio Revision 3 0 0 January 2014 2 SR1 Operation Manual Table of Contents Foreword 0 Part Getting Started 6 t Unpackingand Sate ly asiar E a E E econ een user 6 2 Manual Revision FISCONY ascienca a a AA 8 9 OVEIVIE Wisco naus a E Oe A E E 9 A User IME ACO sie coat satiate ate ee a aas E K A E a A aa Eaa 13 5 A QUICK Eamplesnouadunen a a a a a a a a et nate aca 19 Part Il SR1 Operation 28 1 PIS IIG Uso cote ated a a amore tees euaneediae yeu teusiinee a A a toasters 28 DAVE SRI CONFIGURATION s a a aaa a a a a a a 29 Sav Partial C Onftigu ratio Me rena a a ee N ETa E a iad 30 Load Configuratio ssns e a E E R a a i 31 PrN SRI SGEE Prici E E S E sean euveeneeasies 32 PrINCSCTUD yemon a r a a 33 EGI ME MU eessen a E ea a e a E A e a TENE Ea 34 Panels MENU siosio a a a we cual aa a ulae oni 35 Analog Gen rator Pane licin i a E a aaa aasa iaaa 36 Analg Generator UNI Sessa a A E E A T 41 Anal g Generator Wavel ol MS niceriir neier n a ee A cece 43 Digital G nerator Pane blossen a aaa aE a vets Ea a aAa a Sa a EAEan 57 DiditalGenerator Unila a A eee aeage eee 60 Digital Generator Wav Cl OMn ss ssscraserestach sites cede ianenasgarsdetoese Ge oii gid e2 98 Reacisacbec Ricans aaseasaeasnddedeaedeaastenesaaseasiae 62 Analog Inputs Pane linii Ieeaedueweenndatouieblaaniadesvenmerumennemaaaieen mee 76 Digital VO Panel oriniai a a eaaa
16. 1 3 Octave None ad a Averaging Exponential Time 100 00 msec Octave Mode Selection Octave Mode 1 3 Octave Full Octave 1 3 Octave 1 12 Octave The octave mode selection box determines the fractional ocatve bands used to measure power The allowed choices are full octave 1 3 octave and 1 12 octave Octave Averaging Averaging Cr _ There are two averaging selections available for the octave analyzer exponential time averaging and peak hold averaging Exponential time averaging applies and exponentially weighted more recent samples are weighted more highly moving average filter with the specified time constant to the output of each bin Peak hold averaging maintains the highest reading obtained for each bin 2014 Stanford Research Systems SR1 Operation 177 2 4 10 Jitter Analyzer The Jitter analyzer provides realtime detection and analysis of jitter in digital audio carrier and clock signals The jitter analyzer uses analog Phase Locked Loop hardware to extract the jitter signal from any one of SR1 s digital audio inputs The demodulated jitter signal is digitized and passed to the DSP processor which analyzes the jitter signal either in the time domain or frequency domain depending on the user s selection Jitter Analyzer Block Diagram Clock Lock Demod Lock Indicator Indicator AES EBU SIPOIF Signals m Clock Recovery Jitter Demodulator PLL A PLL quare wave Demodulated Jitter
17. C D E and F characters above the micro milli kilo meta Exp and Page keys are active allowing the entry of a hexadecimal value Hex Entry mode is terminated when Enter is pressed Generator After pressing this key press lt to open the analog generator panel or L gt to open the digital generator panel After pressing Analyzer press lt J to open the AO panel or L to open the A1 panel Alpha Calls up the virtual keyboard t Translate graph up down Once pressed the lt LD keys will move the data in the current graph up and down Press Esc to exit this mode T Zoom Y axis Once pressed the E E e keys will zoom the Y axis in and out Press Esc to exit this mode Autoscale X Y After pressing the key use the L keys to select autoscale of either the the Xor Y axis WMin Moves the active cursor in the current graph to the position corresponding to the minimum Fay a e ooo value in the graph Hep sd Brings up the online help up the online help Using the Knob Turning the knob modifies the currently focused control If the control is a unit entry control the knob increments and decrements its value with an appropriate resolution If the control is a drop down list the knob cycles through the entries in the list Three different knob acceleration profiles can be selected from the preferences panel A control can be selected as Sticky in which case the knob will continue to control it even though focus is
18. Rise Fall w Re analyze Advanced TI G T Asymmetry Jitter Fs 2 dspley tern v ES SE After acquisition the digitizer examines the input time record and makes a preliminary estimate of the zero crossing positions and the unit interval time For a digital audio signal all transitions should either Measure 2014 Stanford Research Systems 186 SR1 Operation Manual be one two or three unit intervals long If the one two three pattern is completely unrecognizable the digitizer will abort processing at this point If the one two three UI pattern is recognized the digitizer does a more careful measurement of the zero crossing positions and attempts to recreate the original clock signal by minimizing the jitter at one of three user selectable points throughout the input record Selecting Preambles in the Jitter Detection control instructs the Digitizer to reconstruct the clock based on minimizing the jitter at the trailing transition of the 3 Ul pulse found in every preamble of the digital audio signal Because this transition occurs at the end of the longest stable interval in the signal remember that all pulses in the digital audio signal are one two or three Uls long and the 3 UI pulses only occur in preambles pulse pile up effects due to limited bandwidth are at a minimum at this transition and the Preambles selection yields the jitter that is most representative of the intrinsic jitter of the transmitting
19. Some experimentation may be required depending on the characteristics of the EUT to find the optimal values of the settling parameters 2014 Stanford Research Systems SR1 Operation 269 2 8 9 Crosstalk Panel E Measure Crosstalk Input Oukput Measurement Settling Signal Waveform Sine o Start Stop Steps Log Freq 20 000H2 oo ho i Level 100 00 mems o alr Analysis Highpass 10 Hz ha Lowpass Fsf2 ad Measure Ratio Append Traces Free Run Sweep The Crosstalk Quick Measurment panel uses the Time Domain Detector TDD to measure the crosstalk from one input channel to the other output channel of the EUT A signal is applied to one input channel of the EUT and the TDD with a bandpass filter is used to measure the output signal from the other channel to get the crosstalk amplitude The final crosstalk measurement is the dB ratio of this amplitude to the output level on the channel to which the input is applied Cabling Configuration The output of SR1 s analog or digital generator is connected to the EUT s input The output of the EUT is connected to SR1 s analog or digital inputs The number of inputs and outputs and the connectors used are specified in the Setup Panel Generator Configuration The output waveform is set to either Sine or Low Distortion Sine as specified in the panel For free run operation the amplitude and frequency of the generator is set to the frequency and level in the
20. The Analyzer References tab has fields which contain the reference quantities for the analyzer units Note that all analyzers share one set of analyzer units there are not separate sets of references for the FFT analyzer and the Time Domain detector nor are there separate references for the AO analyzer and the A1 analyzer 2014 Stanford Research Systems SR1 Operation 115 2 4 2 Time Domain Detector The Time Domain Detector TDD performs the classic audio analyzer measurements amplitude ratio crest factor and THD N on both analog and digital audio signals The Time Domain Detector operates much the same way that traditional analog audio analyzers worked a notch filter is used to remove the fundamental to allow amplification of noise and distortion products bandwidth limiting and weighting filters are applied and a precision RMS detector is used to measure the amplitude of the resulting signal Analyzer Output ff RMS PKGPK Amplitude Chain Selection Measurements on Selected Channel Bandwidth Limiting Filtering Weighting Amplitude p e Bandpass Hotch Filtering Filters d Measurement Selected Ch Level Chain Measurements on Both Channels j Meas Rate N Selection f Ch A Frequency RMS Level Measurement Measurement Ch E Time Domain Detector Generic Block Diagram Two sets of measurements are made by the TDD The first the Level Chain is made for both channels in the
21. Unlike noise signals which are not deterministic and require long averaging times to measure frequency response a chirp signal can precisely measure frequency response in a single FFT record The Chirp waveform is synchronized to the settings of a particular FFT analyzer If the resolution or frequency span of that analyzer changes the chirp waveform automatically reconfigures to provide a full set of bin center tones If the corresponding analyzer is set to show aliased lines the chirp outputs a tone in every single FFT bin from near DC to Fs 2 If show aliased lines is off the chirp outputs tones from near DC up to the alias limit for the analyzer s selected converter and frequency span See the FFT Analyzer section for more details on these topics Because of the synchronous nature of the chirp signal a uniform window should be selected in the analyzer when using the chirp waveform Selection of the associated FFT Analyzer is done with the Chirp Source control Chirp Source Associated Analyzer AO FFT1 Analyzer 0 Single Channel FFT Analyzer AO FFT2 Analyzer 0 Dual Channel FFT Analyzer A1 FFTO Analyzer 1 Single Channel FFT Analyzer A1 FFT1 Analyzer 1 Dual Channel FFT Analyzer Synchronous chirp generation requires there to be a relationship between the selected generator sampling frequency and the associated analyzer sampling frequency In general the two frequencies need to be either exactly the same or some integer multi
22. and the row and column of the current cursor location The Scripting Window Speedbar Icon Description S Do Creates a new script A dialog will prompt to save any existing script before it is cleared Loads a script from a file Saves the current script Print Setup Offers printer and page layout options for printing the script Prints the script Save As Saves the current script to a different file 2014 Stanford Research Systems 244 SR1 Operation Manual SE EX ER Cut Copy and Paste Performs the normal editing options on selected text Cut Copy and Paste Performs the normal editing options on selected text Copy and Paste Performs the normal editing options on selected text DE Undo and Redo The first button undoes the last action performed If actions have been undone the second button will redo the undone action ml Fra Finds selected text in the Find Finds selected text in the script window window 7 Replace Finds selected text in the script window and replaces the text with the user selection sets a bookmark at the current cursor location Bookmarks are indicated by the numbered icons El at the left of the main window Go to Moves the cursor to a given line number and column or to a bookmarked location Bl Begins execution of the script This button is also found on the main SR1 speedbar Stops Scripts Events Disables trapping of any events which have been set up in the curren
23. bits EE onon rone CRE Lock walidity Conf Coding Parity Output Configuration Controls Select the output source as either BNC Unbalanced or XLR balanced Although the digital audio signal is always output to both connectors the displayed amplitude is only calibrated for chosen connector Amplitude ranges from 2 mVpp to 2 55 Vpp can be selected for the BNC outputs while amplitudes from 10 mVpp to 10 2 Vpp are available for the XLR outputs Output amplitudes are calibrated only when the outputs are correctly terminated i e 75Q for unbalanced outputs and 110Q for balanced outputs Selecting Dual Connector sets the digital audio output to dual connector mode In normal digital audio data streams each frame contains a pair of samples one left and one right In dual connector mode each data stream outputs a set of successive samples for a single channel with an effective single channel sampling rate of twice the frame rate In dual connector mode the left channel is output on C1 and the right channel on C2 The Optical indicator indicates whether the TOSLINK Optical output is active The indicator glows green to indicate that the optical output is active and red to indicate that optical output may not work correctly FS Invert controls the polarity of the of the rear panel Frame Sync signal This signal is normally high during the first left subframe of each digital audio frame Checking the box inverts this po
24. feature of the SR1 generator s architecture the ability to combine several waveforms in the generator The 1 KHz 1 Vrms signal and the 2 kHz 1 mVrms signals are added in the generator Using this technique an almost infinite variety of waveforms can be created in the generator to suit almost any test situation Now let s measure the properties of the signals just created Look at the Time Domain Detector panel and change the source from Digital A to Analog A The Converter field will read HiRes indicating that the analyzer is using SR1 s 24 bit High Resolution converter The current sampling rate for this converter 64 kHz is displayed in the Fs field next to the converter selection SR1 uses two high quality analog to digital converters ADCs for analysis of analog signals a 24bit high resolution converter that can operate over a variety of sampling rates and a 16 bit high bandwidth converter that operates at a fixed sampling rate of 512 KHz giving the instrument an analog bandwidth of 200 kHz The differences between the two converters will be discussed in detail later but for now we can leave the converter selection at HiRes Press the gt l button at the top left of the screen to start the measurements Note that the Status Indicator at the bottom left of the screen SR1 operates in two distinct measurement modes In Free Run mode which we just started all the analyzers make continuous measurements and continuously update the re
25. remove the mapping 2014 Stanford Research Systems 234 SR1 Operation Manual a Disconnect Network Drives Select the network drivets vou want to disconnect then click OK Network Drives r WoonomasiRO Home r apas Corp Data on l am F r VWiapasisrsprog r Sonoma AO Publ Projects 4 cae U gt e Weonomas RADO Publ 2 6 5 5 Share SR1 Selecting Share SR1 allows sharing of the user folder and its subfolders on the SR1 hard disk with other network users The user folder contains subfolders in which configuration files EQ files eye diagram limit specifications and logfiles are stored 2014 Stanford Research Systems SR1 Operation 235 Local Disk C Properties General Hardware sharing Security Quota Tou can share this folder with other users on Your network To enable sharing for this folder click Share this folder Do not share this folder Share this folder Share name User Comment User limit Maimun allowed Allow this number of users To set permissions for users who access this ages Permizzion folder over the network click Permissions To configure settings for offline access click Caching sasis Mew Share Windows Firewall will be configured to allow this folder to be shared with other computers on the network View pour Windows Firewall settings Click Share this folder to enable sharing of the folder Use the Permissions
26. 16 1m 16 2m 16 3m 72 Time Record without Time Domain Interpolation 15 7m 15 8m 15 9m 16 0mm 16 1m 16 2m 16 3m 72 Same Signal with Time Domain Interpolation Using the FFT1 Analyzer With the FFT Chirp Source Both the analog and digital generators have a FFT chirp source which generates a signal which has equal power in each FFT bin For the digital chirp the output is flat to within fractions of a mdB for an analog chirp the signal is typically flat to 5 mdB when used with the Hi Resolution ADC and 10 mdB out to the 200 kHz limit of the Hi Bandwidth ADC The plots below show FFTs for the analog chirp source for both input converters 2014 Stanford Research Systems 138 SR1 Operation Manual SP SNS 0 2 5k 5 0k 7 5k 10 0k 12 5k 15 0k 17 5k 20 0k 22 5k 25 0k 27 5k Chirp Hi Resolution ADC Y Scale 10 m Db div E E E E S E PO ee E VO A ee ee TEE VE E ee OO E O OA a e FON aee e a TO e er a aa ea M anne ee nn a a a ee a a 29 96 O nui E onan ere tee te ee aaan a Ce ee ee ee ee er ne en enn Se Sennen gt ae ee eet eee en ee es E ee a a S ee sn ae cause re ee ae a pos 20k 40k 60k 80k 100k 170k 140k 160k 180k Hz Chirp Hi Bandwidth ADC Y scale 10 mdB div The unique property of the synchronous chirp equal power in each FFT bin makes it a powerful tool for quickly measuring the frequency response of audio devices Unlike noise stimulus which although having a uniform frequency co
27. 20 Hz to 20 kHz 10 Hz to 64 kHz 10 Hz to 200 kHz Residual THD N Hi BW DAC Fs 512 kHz 1 kHz 4 Vrms 20 Hz to 20 kHz 10 Hz to 100 kHz Hi Res DAC Fs 128 kHz 1 kHz 4 Vrms 20 Hz to 20 kHz 0 020 dB typ 0 012 dB 0 025 dB 0 050 dB 0 25 Vrms lt Amplitude lt 4 0 Vrrms 112 dB 22 kHz BW 105 dB 1 uV 22 kHz BW 100 5 dB 1 7 pV 80 kHz BW 97 dB 2 5 uV 200 kHz BW 89 dB 2 5 uV 200 kHz BW 112 dB 22 kHz BW 106 dB 1 pV 22 kHz BW 2014 Stanford Research Systems 20 Hz to 57 6 kHz Hi Res DAC Fs 64 kHz 1 kHz 4 Vrms 20 Hz to 20 kHz Normal Sine Flatness rel 1 kHz 20 Hz to 20 kHz 10 Hz to 64 kHz 10 Hz to 200 kHz Residual THD N Hi BW DAC 1 kHz 4Vrms 20 Hz to 20 kHz 10 Hz to 100 kHz Hi Res DAC Fs 128 kHz 1 kHz 4Vrms 20 Hz to 20 kHz 10 Hz to 57 6 kHz Hi Res DAC Fs 64 kHz 1 kHz 4Vrms 20 Hz to 20 kHz Other Waveforms Phased Sines IMD Noise MLS Multitone FFT Chirp Log sine Chirp 2014 Stanford Research Systems SR1 Reference 297 102 dB 1 4 uV 57 6 kHz BW 112 dB 22 kHz BW 106 dB 1 uV 22 kHz BW Amp lt 4 Vrms 0 008 dB typ 0 003 dB 0 020 dB 0 030 dB Measured w HiBW Analyzer 86 dB 22 kHz BW 85 dB 1 uV 22 kHz BW 84 5 dB 1 7 uV 80 kHz BW 82 dB 2 5 uV 200 kHz BW 93 dB 200 kHz BW 99 dB 22 kHz BW 98 dB 1 uV 22 kHz BW 96 5 dB 1 4 uV 57 6 kHz BW 106 dB 22 kHz BW
28. 2009 2014 Stanford Research Systems Inc All rights reserved No part of this manual may be reproduced or transmitted in any form or by any means electronic or mechanical including photocopying recording or by any information storage and retrieval system without permission in writing from Stanford Research Systems Inc Manual Revision Histo Version Date Author Comments T a 20 709 am Preliminary Version shipped with frst SRIs_ 21 1009 am 2nd Preliminary version S O 30 uta faim Upoateg for SR1 x24113 oS O Copyright and Trademark Acknowledgements Windows is a trademark of the Microsoft Corporation Apache Xerces Copyright 1999 2007 The Apache Software Foundation This product includes software developed at The Apache Software Foundation http www apache org Portions of this software were originally based on the following Software copyright c 1999 IBM Corporation http www ibm com Software copyright c 1999 Sun Microsystems http www sun com Voluntary contributions made by Paul Eng on behalf of the Apache Software Foundation that were originally developed at iClick Inc software copyright c 1999 NI Device Copyright 2009 National Instruments Corporation All Rights Reserved FastReport Copyright 2009 Fast Reports Inc All Rights Reserved 2014 Stanford Research Systems Getting Started 9 1 3 Overview SR1 is versatile and complex instrument capable of making
29. A value of 3 F R with reference of 2 kHz gives a waveform frequency of 6 KHz Difference relative the Frequency Reference A value of 500 dHz with a reference of 2 kHz gives a waveform frequency of 2 5 KHz Percent of the Frequency Reference A frequency value of 50 Fref with a reference value of 10 kHz gives a waveform frequency of 5 KHz A cent is a logarithmic unit which represents 1 100 of a semitone of the musical scale 12 semitones make up an octave Thus a cent is 1 1200 of an octave An octave is a factor of 2 in frequency Thus a frequency value of 3 octaves with a reference of 1 KHz gives a waveform frequency of 8 kHz An decade is a factor of 10 in frequency Thus a frequency value of 2 decades with a reference of 2 KHz gives a waveform frequency of 200 kHz Digital Generator Waveforms oe o D Q S o 3 S o 3 N SR1 s Digital Generator is capable of generating an enormous variety of different audio waveforms from simple sines to complex synchronous multitone waveforms Because of its unique architecture which allows different waveforms to be combined the generator offers almost limitless flexibility in providing the perfect audio test output In this section each waveform the building blocks of the generator output will be described in detail Config 40 Sine aveform On fw EQ T ine Arig 0 0000 WETS Freq fi 00000 kHz Sine Waveform Tab When a waveform is added to the generator outpu
30. CCIR unwtd Filter defined in ITU CCIR Rec 468 4 Annex Il for making unweighted noise measurments CCIR 2 kHz This filter is identical to the CCIR wtd filter except the normalization is changed so that the filter has unity gain at 2 kHz instead of 1 kHz Hardware Filters Up to 4 optional hardware filters may be installed on each analog input board Only 1 filter may be selected at any given time These filters are applied only with the analog HiBw input selection The filters are inserted in the amplitude chain after the notch bandpass filter and before the postfilter analog gain 2014 Stanford Research Systems 126 SR1 Operation Manual Post Filter Gain Post Filter Level sain Auto k For analog HiBw inputs the TDD offers variable postfilter gain Typically the Gain selection can be left on Auto and SR1 will automatically optimize the postfilter gain The level indicator above the gain selection indicates the current level of the postfilter signal blue for less than half scale green for greater than half scale and red for overloaded The AutoGain setting should keep the indicator in the green range If the level indicator jumps around too much the gain can be set manually to a setting which maintains the level in the green range Time Domain Detector and the Trigger When the trigger is enabled for the TDD each measurement interval begins synchronously with the receipt of a trigger While the trigger is not n
31. Cabling Configuration The output of SR1 s analog or digital generator is connected to the EUT s input The output of the EUT is connected to SR1 s analog or digital inputs The number of inputs and outputs and the connectors used are specified in the Setup Panel Generator Configuration The output waveform is set to either Sine or Low Distortion Sine as specified in the panel For free run operation the amplitude and frequency of the generator is set to the frequency and level in the start column of the panel For swept operation the frequency is swept from the start to stop value for each amplitude value specified Analyzer Configuration One or both analyzers depending on the number of output channels specified in the Setup Panel are configured as Time Domain Detectors Free Run Configuration A barchart display is created for each output channel showing the instantaneous SNR Sweep Configuration A graph will be created for each channel showing a graph of SNR vs Freq for the selected number of amplitude steps Log spacing can be selected for both the frequency and amplitude sweeps The settling parameters of the SNR measurement can be adjusted directly from the Settling tab of the panel 2014 Stanford Research Systems 256 SR1 Operation Manual Settling Profile Delay Exponential 20 000 msec Precesion nPoints Threshold p A000 So 3 p 000D mrm Refer to the Settling Panel chapter for a more detailed discu
32. Clock Signals 0 30dE Variable Gain Jitter Analyzer Block Diagram A block diagram of the analog section of the Jitter Analyzer is shown above If the selected input signal is a professional or consumer digital audio carrier the signal is first passed through a clock recovery circuit which extracts the underlying clock signal with any accompanying jitter Clock inputs bypass this stage and are applied directly to the input of a second Phase Locked Loop which extracts the jitter from the clock Both the clock recovery and jitter demodulator PLLs have corresponding lock indicators on the Jitter Analyzer Panel indicating they are locked to their respective input signals The demodulated jitter signal is amplified and digitized at a sample rate of 256 kHz providing a maximum jitter bandwidth of approximately 113 KHz Further processing is done in the DSP processor and depends on the user s selection of Time Domain or Frequency Domain analysis When Time Domain analysis is selected the jitter filter is passed successively through 4th order butterworth highpass and lowpass filters to limit the bandwidth of the signal and then through a user selectable weighing filter See the discussion of the Time Domain Detector for details of the available weighting filters The resulting signal is then detected with either an RMS or Peak response much like the Time Domain Detector When Frequency Domain analysis is selected the analyzer works like the sing
33. Cursor Tab Scale Cursors FET Cursors Limits esos Pease Pe os Basco Pind 81 FFT Power Spect E A1 FFT Power Spectr The cursor tab displays statistics for the two cursors The left side of the tab shows value for cursor 1 the right side for cursor 2 Pressing Start or Stop toggles the collection of statistics Pressing Reset resets the statistical history The four statistical values collected are Maximum Max Minimum Min Average Avg and Standard Deviation SDev Note that the statistics are calculated in the units of the Y axis i e the average of a quantity displayed in dB is the average of the dB values not the average of the underlying value expressed in dB Independent Cursors Ind Normally both cursors are attached to the active trace Checking Ind Independent Cursors allows the two cursors to be attached to any of the traces on the graph Use the two drop down lists to select the trace attached to each cursor Checking Lock X ensures that the two cursors maintain the same X value so that the Y values of different traces at the same X value can be easily compared 2014 Stanford Research Systems 198 SR1 Operation Manual FFT Cursors Tab Scale_ Cursors FFT Cursors Limits if Calc Power W Calc THD Ratio W Calc SNR ind FRESE E C gt c aT For FF T type traces the FFT cursors tab offers some additional cursor measurements Checking Calc Power displays the RMS integr
34. Demo X User Message And then displays the second message if the user does not act within the 5 second time out period UserOKCancel Displays a dialog box with a user message and an OK and Cancel button For instance x SR1 Instrument UserOKCancel Proceed 10 creates the window Confirm P Proceed Cancel The function returns 1 if OK is pressed O if Cancel is pressed or the window is closed and 1 if the window times out UserChoice The UserChoice function displays a window with a drop down box displaying a list of choices dim x x SR1l Instrument UserChoice Some Choices A B C 5 if x A then Call SR1 Instrument UserMessage User Picked A 10 elseif x B then Call SR1 Instrument UserMessage User Picked B 10 elseif x C then Call SR1 Instrument UserMessage User Picked C 10 elseif x timedout then Call SR1l Instrument UserMessage Window Timed Out 10 2014 Stanford Research Systems 248 SR1 Operation Manual elseif x cancelled then Call SR1 Instrument UserMessage User Cancelled 10 end if Displays the following window User Select Some Choices B OK Cancel The label above the drop down box is specified in the first argument The choices displayed in the drop down box are given in the second argument as a single string with the choices separated by commas e g A B C yields User Select Some Choices
35. E 2014 Stanford Research Systems SR1 Operation m 2 7 Automation Menu Automation refers to controlling the operation of SR1 through some means other than the front panel interface There are two basic types of automation remote and local Remote automation refers to control of the instrument from a remote computer through one of SR1 s 3 physical interface ports IEEE 488 GPIB Serial RS 232 and ethernet VXI 11 There are two methods by which the instrument can be controlled remotely the GPIB command interface and the SR1 Basic command interface The GPIB command interface consists a set of GPIB commands which are sent as ASCII text strings over one of the three physical interfaces Arguments to the commands and responses from SR1 are also sent as ASCII text strings The SR1 Basic interface a binary interface based on the Microsoft COM standard presents the instrument as a set of objects containing properties and actions which can be manipulated by the user The SR1 Basic interface can only be accessed wa the ethernet port and is particularly suited to control by programs written in Visual Basic Microsoft Office and other COM enabled languages Local Automation or scripting refers to control of the instrument via scripts executed locally on the instrument Scripts interact with the instrument only through the SR1 Basic interface The Automation Menu provides access to the Scripting Window a complete development envronment
36. Interfacing SR1 Supports a variety of remote interfaces IEEE 488 2 GPIB SR1 has a rear panel IEEE 488 connector and fully supports the IEEE 488 2 standard All instrument features can be set and queried from the remote interface Serial GPIB commands may also be sent over the rear panel serial connector which supports baud rates up to 115 2 kBaud TCP IP GPIB commands may also be sent over a TCP IP network via the rear panel ethernet connector SR1 follows the VXI 11 standard for the transmission of commands over TCP IP COM SR1 is fully COM enabled allowing applications such as Visual Basic to set and query instrument internals COM enabled applications may also control and query the instrument remotely over ethernet Not that firewall settings may need to changed to fully enable this functionality 2014 Stanford Research Systems 12 SR1 Operation Manual 2014 Stanford Research Systems 1 4 Getting Started User Interface The SR1 audio analyzer user interface is based on software that runs on the Microsoft Windows operating system As such the basic user interface based on standard Windows menus and controls should be familiar to most users who have used Microsoft Windows The same software that runs on the instrument is available for free download from Stanford Research Systems www thinksrs com and can be run on any Windows PC While no measurements can be made while running on a PC this mode is perfe
37. Irw Red Outer Lowe tock x Outer Lower The eye diagram is a plot of probability vs amplitude and time with probability coded as color The Digitizer Display assembles overlays the signal transitions corresponding to the Jitter Detection selection on the digitizer panel and calculates the probability of the signal appearing at a given time with colored point is plotted on the display If the licon is active in the speedbar the probability can be directly examined by moving the mouse over the desired point By examining the probabilities the Display computes four envelope traces corresponding to the inner and outer limits of both the upper and lower portions of each transition Selecting a trace from the Cursor 1 or Cursor 2 control links that cursor to the selected trace Note that unlike the graph clicking on a trace in the trace listing window of the Digitizer Display does not link the cursor to that trace Checking Lock X locks the X value of the two cursors to the same value so that differences between any of the envelopes can be read off directly on the cursor display bar Z scale and Color Selection The mapping from probability to color can be selected as linear square root or logarithmic with the Z Scale control Linear mapping tends to produce the most washed out color display while logarithmic produces the most intense color The Square Root selection tends to be a good compromise between the two
38. Manual dEYrms P S ia oe es eee eee Offset 75 ME Paii SPA A Correction aa E AT E EEATT E e EE E ES m E E A E E EAT With Average SS nme eee a O OC Correction l 10 0 50 100 00 s00 ik zk Hz Effect of DC Correction on Spectrum An example of the us of DC correction is shown above The original spectrum has some DC offset which has leaked into adjacent low frequency bins because of windowing The red trace shows the same signal with Average DC correction applied Spectrum Weighting Spectrum Weighting None Invert The two spectral outputs of the FFT1 analyzer the Power Spectrum and Linear Spectrum can have weighting curves applied to them Weighting curves are represented by EQ files The standard_EQ files supplied with SR1 include Weighting Filters None No weighting filter is applied A weighting filter is applied A weighting is specified in ANSI standard S1 4 1983 and is typically used for noise and THD N measurements with audio applications C Msg Wt The C Msg weighting filter specified in IEEE Std 743 1995 is intended to be used for noise measurements associated with voice transmission telecommunications CCITT The CCITT weighting filter defined by ITU T Recommendation O 41 is another telecommunication noise weighting filter CCIR wtd Filter defined in CCIR Rec 468 4 for audio noise measurements Designed to be used with the Quasi Peak setting on the Time
39. Notes field and press Update Report To begin the sequence of automated tests press Autom Meas 2014 Stanford Research Systems 2 8 SR1 Operation Manual 2 9 Setups Menu The Setups menu allows SR1 to quickly be configured for many common audio measurements These pre programmed setups can be used as a Starting point to customize your own configuration files See Saving SR1 Configurations for details The setups are categorized by the domain of the stimulus signal and the domain of the analysis signal as follows Analog Analog Measurement setups entirely in the Analog Domain Analog Digital Measurement setups with an analog stimulus and a digital response ADC measurements for example Digital Analog Measurement setups with a digital stimulus and an analog response DAC measurements for example Digital Digital Measurement setups entirely in the digital signal domain i e a digital effects processor for instance Digital IO Measurements entirely in the digital carrier domain i e jitter measurements 2014 Stanford Research Systems SR1 Operation 279 2 9 1 Analog Analog The Analog Analog selection lists basic Audio Analyzer measurements in the analog domain All the Analog analog setups default the A and B input channels to the BNC inputs DC coupled autoranged What you ll need to configure for all these setups Change the analog input connector and coupling settings on page 1 to match the type of equipmen
40. Of 1 00000 VRMS uf 1000 00 H21 of The top portion of the panel records the GPIB commands received over the three physical interfaces Commands received over the IEEE 488 port are displayed in black commands received over the ethernet interface in blue and commands received over the serial port RS 232 are shown in green Commands which cannot be parsed by SR1 are underlined to indicate an error Interface messages such as lt Local gt or lt Device Clear gt are placed in lt angle brackets gt to differentiate them from literal text The bottom window of the panel shows SR1 s responses to the received commands Responses that are queued for sending but have not yet been sent are shown in italics once sent they are shown in normal type The same color coding for physical interface is used in the output window that is used in the input window Press Clear to clear both the input and output windows of the Remote Interface Panel 2014 Stanford Research Systems SR1 Operation 243 2 2 Scripting Window The scripting window provides a complete development environment for writing and debugging SR1 scripts Scripts are small programs written either in VBscript or JScript which can automate small tasks or entire test sequences Scripts interact with SR1 through the SR1 Basic interface which divides the instrument into a heirarchical collection of objects each of which can contain properties that can be set and read and act
41. Output input Steps 5 4 Metwork BMC r Bus 4 stark ch 1 ah in steps of 2 iw B stark ch 2 There are two tabs on the switcher sweep source panel one for inputs and one for outputs Inputs and outputs can be swept simultaneously if the corresponding switcher hardware has been configured on the Switcher Configuration Panel On each tab the network selection determines whether unbalanced BNC or balanced XLR inputs or outputs will be swept The number of steps determines how many points will be taken The number of points is the number of steps plus one i e a sweep of one step contains two points the initial and final ones On both the input and output tabs the checkboxes for A and B determine whether the A channel B channel or both will be swept If the checkbox is checked then the corresponding up down control will be enabled allowing setting of the initial logical channel for each channel At each step in the sweep the logical channel connected to each checked input or output will be incremented by the amount specified by the in steps of control For example consider the output sweep specified by the panel illustrated above At each point in the sweep the logical channel connections to the A and B outputs will be 0 Initial 5 Final owecp rom State ae fs fa State A Logical agga hoo fs is Pr fo fe B Logical aegea fe fa fo fe fole This represents a typical stereo sweep of a single switcher wi
42. Panel are configured as Time Domain Detectors The bandwidth limits for the measurement is selected on the Analysis section of the panel Free Run Configuration A barchart display is created for each output channel showing the instantaneous amplitude for the two inputs There is no sweep configuration for the Reference measurement 2014 Stanford Research Systems 258 SR1 Operation Manual 2 Reference B 2014 Stanford Research Systems SR1 Operation 259 2 8 4 Level Panel E Measure Level Inpukiutouk Measurement Settling Signal Waveform Sine Start Stop Steps Log Freq 20 000 Hz 20 000 kHz od o ail Level 100 00 mivrms 1 0000 Vrms O Sil Analysis Highpass 10 Hz ad Lowpass Fs 2 ad Measure Level sd Append Traces Free Run Sweep The Level quick measurement measures the output level of the EUT either absolutely or relative to the reference values computed with the Reference measurement The source can be swept over frequency and amplitude to yield the frequency response of the EUT for a range of input amplitudes Cabling Configuration The output of SR1 s analog or digital generator is connected to the EUT s input The output of the EUT is connected to SR1 s analog or digital inputs The number of inputs and outputs and the connectors used are specified in the Setup Panel Generator Configuration The output waveform is set to either Sine or Low Dist
43. SinefO Amp On ChB Gain E ClockRef E Digsen o E Digho H External aa The different sweep sources are organized in a tree structure Click on the appropriate source Time Internal External or Switcher Channels to display the options relevant for that particular source 2014 Stanford Research Systems SR1 Operation Configuring Internal Sweeps Internal sweeps are perhaps the most common type of sweep used in audio measurements In an internal sweep SR1 sweeps some internal parameter for instance the analog generator sine frequency for a fixed number of points between two endpoint values The progression of the sweep can be linear or logarithmic When the sweep source selection window is displayed clicking on Internal displays the major subsections of SR1 containing sweepable parameters Only the currently active generator and analyzer parameters are visible so it s important to set up any generators and analyzers before configuring the sweep Clicking on an individual parameter sets that parameter as the sweep source Inner Sweep Outer SEED External Settings SOUrCE JanlgGen ch Sine O Freq imiks teps Stark 20 000 kHz kai Steps 30 F Step Size e66 00 Hz Stop 20 000 Hz 7 Log Step Size Table Sweep None O C Internal Sweep Controls When a source parameter is selected the limits controls will display the default limits for that parameter Any limits within the allowed
44. THD 157 THD Analyzer 157 Fundamental Selection 157 Harmonic Selection 158 Measurement Speed 158 Weighting 159 THD Ratio 198 THD N 115 261 Time Domain Detector 115 Bandwidth Limiting 124 Filter 120 Measurement Rate 119 Post Filter Gain 126 Response 119 Time Sweeps 93 2014 Stanford Research Systems Index 317 Tools Menu 213 TOSLINK 79 Trace Calculator 199 Trackpad 13 U Units 110 Unpacking 6 Updating SR1 238 USASI Noise 43 62 User Bits 79 User Interface 13 User Status Bits 90 V Validity 79 VBScript 243 Video Reference 105 Video Reference Input 293 Virtual Keyboard 238 Virtual Keypad 13 Visual Basic 243 Volume Knob 291 VXI 11 218 241 W Weighting Filters 164 X XLR Connector 79 2014 Stanford Research Systems 318 SR1 Operation Manual v2 Endnotes 2 after index 2014 Stanford Research Systems Back Cover
45. The ETC is the magnitude of the analytic impulse SR1 calculates the ETC from the anechoic frequency response A frequency domain window is applied to the anechoic frequency response to reduce alias effects The result is transformed back to the time domain to obtain the real part of the analytic impulse response A phase shifted Hilbert transformed version is transformed to obtain the imaginary part of the analytic impulse The magnitude of the resulting function is the ETC 2014 Stanford Research Systems SR1 Operation 147 Summary of FFT2 Outputs Level A B Peak based levels of both channels of the selected input domain Time Record A B The underlying time data used to compute spectra When displayed on a graph this measurement produces an oscilloscope type display Power Spectrum The amplitude of the power averaged spectra for each channel Amplitude A B Linear Spectrum The amplitude of the synchronously averaged spectra The amplitude of signals Amplitude A B that are synchronous with the time record is preserved other uncorrelated signals average away Frequency The magnitude of the averaged frequency response Response Mag Frequency The phase of the averaged frequency response Response Phase Coherence 2 channel coherence of the A and B signals Values near one indicate good input output correlation while values near zero indicate the output consists mostly of uncorrelated noise Impulse The Inverse FFT of
46. The touch pad can be used as a mouse if an external mouse is not connected to the instrument Drag your finger over the touch pad to move the mouse cursor To click the mouse tap once on the dark grey area To double click tap twice on the dark grey area To right click the mouse tap in the lighter grey area at the top right hand corner of the touch pad L Digital Audio Output Section Contains connectors for digital audio output There are two pairs of connectors labeled C1 and C2 Each pair contains a balanced XLR connector and an unbalanced BNC connector as well as an optical TOSLINK connector Configuration of the output connectors is done on the Digital I O panel M Digital Audio Input Section Contains connectors for digital audio input There are two pairs of connectors labeled C1 and C2 Each pair contains a balanced XLR connector and an unbalanced BNC connector as well as an optical TOSLINK connector Configuration of the input connectors is done on the Digital I O panel N Analog Output Connectors Contains the analog output connectors The two channels of the analog generator can be output either as a balanced signal on the XLR connectors or as an unbalanced signal on either the XLR or BNC connectors See the Analog Generator panel description for more details O Analog Input Connectors Contains the analog intput connectors SR1 accepts analog inputs on either the XLR or BNC connectors Selection of the input co
47. Time Domain Analysis Detector Rate Reading RMS 16 s Filters Hi Pass M n Lopass weighting 700 00 Hz 100 00 kHz None r The Time Domain jitter analyzer has a selectable peak or RMS response which is selected with the Detector control In the RMS mode the analyzer calculates the RMS value of the jitter over an interval whose length is the reciprocal of the specified Measurement Rate The Peak response mode simply chooses the peak value of the Jitter over the same interval as the jitter reading The rate is adjustable from 1 to 512 measurement intervals per second Two additional rate settings are available When the rate is set to Dig Fs the measurement interval is set to an integral number of samples of the detected digital audio input sample rate When the rate is set to Jitter Gen the measurement interval is set to an integral number of cycles of the current jitter impairment frequency The detected jitter amplitude is shown on the Reading control Three separate filters can be applied to the jitter signal before the amplitude detector The highpass and lowpass filters are both 4th order butterworth filters The highpass has an adjustable corner frequency from 50 Hz to 20 kHz The lowpass filter has a corner frequency between 2 kHz and 100 kHz The selection of weighting filter is the same as for the Time Domain Analyzer See the discussion there for details of each filter The Jitter Analyzer Panel Frequency Do
48. a wide a variety of precision audio measurements in both the analog and digital domains The following is a brief introduction to the different pieces of SR1 and their capabilities Measurement Setups SR1 includes pre configured setups for many common audio measurements To get up and running quickly see the Setups Menu Generators SR1 contains precise and flexible analog and digital audio generators Each generator is built around several core waveforms which can be either generated individually or combined in the generator and output as a composite signal For the Analog Generator the core waveforms include Sine Low Disortion Sine Phased Sine Noise USASI Noise Squarewave Ramp triangle FFT Chirp Multitone Arbitrary and Constant offset The analog generator can be output either as a balanced or unbalanced signal and ouputs a maximum peak output voltage of 20 V unbalanced or 40 V balanced The Digital Generator offers the same core waveforms as the analog generator and additionally offers a number of waveforms optimized for digital interface testing including Digital Constant Digital Counter Walking Bits and a J test waveform designed to test the jitter susceptibility of devices Analog Inputs SR1 offers both balanced and unbalanced analog inputs with full scale input ranges from 160 Vrms down to 62 mVrms SR1 s analog inputs are autoranging meaning that for most input signals the analyzer automatically sets the input
49. amplitudes of each component of the composite signal are adjustable separately as is the overall amplitude of the combined signal Some waveforms USASI noise for instance are special purpose test signals and may not be combined with other waveforms The tabs corresponding to each particular waveform are described in the Digital Generator Waveforms section This section describes the controls and settings of the digital generator that are relevant to all waveforms Note that the controls on the generator panel and the waveform tabs are mainly concerned with the properties of the embedded digital audio output controls which govern the properties of the digital audio carrier signal are found on the Digital I O panel The SR1 Digital Generator is separate and independent from the Analog Generator The two generators operate simultaneously and independently with different waveforms The two generators however do share memory used for arbitrary waveforms and FFT chirps Deleting an arbitrary waveform in the analog generator makes memory available for the digital generator and vice versa 210 x E A aveform utput Mew Mode Mono J Fs Delete hA o p Ch B 100 0 AIB E Lock 100 0 Ho Abo E On Config Sine eferences Freq Ref lt 00000 kHz Y FS j1 0000 Yms EQ None i Invert Output Controls Mode controls the output mode of the Digital Generator The same waveform is output to bo
50. audio sampling rate For non digital audio signals Square waves the available selections are Square Rising Square Falling or Square Both Square Rising or Falling calculate the jitter on the respective edges of the clock signal The Jitter bandwidth for these selections is 1 2 the square wave frequency Square Both calculates the jitter on both clock edges leading to a jitter bandwidth equal to the square wave frequency After reconstructing the clock the digitizer calculates the jitter at the user specified points in the input record by comparing the zero crossing of the reconstructed clock with the actual measured zero crossings in the input record The total rms jitter and the effective embedded audio sampling rate are calculated and displayed in the Measure tab of the Analysis box The jitter sampling frequency and the number of points at which the jitter is calculated are shown in the Advanced tab Asymmetry Several physical processes can lead to a situation where negative going pulses in the carrier signal have systematically different widths from the positive going pulses For pulses that are roughly trapezoidal in shape an offset in the signal will cause this effect as shown below 2014 Stanford Research Systems SR1 Operation 187 longer COOP eee Perey TPC PEELE CLP SEERE ETETETT rPererrererey Voltage Offset Leads to Different Positive and Negative Pulse Widths The same effect can by caused by a trans
51. based meter The phase indicator shows the relative phase between the A and B channels of the analog or digital audio input signals depending on the selected input source The phase indicator reading is only meaningful when A and B channels have the same frequency Analyzer Trigger Measurement Meas 2 References Trigger SOuUrce Input Channel iw Enabled Level Polarity fe Rising 10 000 FS f Falling fen ratar Trig Source Manual Analog A Trigger Most analyzers can operate in a triggered mode The FFT analyzers for instance use the trigger to synchronize the start of the FFT time record The Time Domain detector uses the trigger to start the beginning of each measurement interval To select triggered mode for the analyzer check the Enabled box on the trigger tab Although each analyzer uses the trigger differently the method by which triggers are generated is common to all analyzers Trigger Source Input Channel The trigger source is the selected input signal of the analyzer Other Channel The trigger source is the channel not selected as the input of the analyzer For instance if the analyzer source is Analog A then selecting other channel as the trigger source selects Analog B Selects a TTL signal on the rear panel Ext Trigger BNC connector as the Only allows manual triggers Manual triggers are generated by clicking on the Manual Trigger button Selects the generator as the trigger source
52. be configured with a different waveform Output Configuration selects the output connector configuration See the diagram below Outputs to the both the XLR and BNC connectors The BNC shield and XLR pin 3 are connected to chassis ground through a 5Q resistor XLR pin 1 is connected directly to chassis ground The BNC center pin and XLR pin 2 both are connected to the unbalanced signal Unbal Float Outputs to the both the XLR and BNC connectors The BNC shield and XLR pin 3 are connected to chassis ground through a high impedance 100k 1uF XLR pin 1 is connected directly to chassis ground The BNC center pin and XLR pin 2 both are connected to the unbalanced signal Bal Gnd Outputs to only the XLR connectors XLR pin 1 is connected directly to chassis ground XLR pins 2 and 3 carry the balanced signal which is symmetric around chassis ground Bal Float Outputs to only the XLR connectors XLR pin 1 is connected directly to chassis ground XLR pins 2 and 3 carry the balanced signal which is unreferenced to chassis B ground al Common Similar to Balanced Ground except that the same signal is present on pins 2 and 3 This allows testing the Common Mode Rejection Ratio of external devices 2014 Stanford Research Systems SR1 Operation Manual out XLR ENC 5 Ohms 100k Unbalanced Float Z out I Z out Balanced Float Analog Generator Output Connections Common Mode
53. be minimized maximized or closed with the standard Windows tools 2 5 x Multiple copies of the same panel may be maintained on different pages of the main display In general changes made to a panel on one page are automatically updated on the other pages Panels are color coded according to the domain to which they apply Panels relevant to the digital domain have a light blue bar under the title bar Tene Panels which are relevant to the analog domain have an orange bar under the title 15 xi Analyzer Panels which can operate in either domain have either the light blue or orange bar depending on whether the analyzer is operating in the digital or analog domain 683 AD FFT Analyzer X S 10O x ource Lonverter Fs Digital A bia Aud 32 000 kHz The following panel selections are available from the panels menu Analog Generator Controls the operation of SR1 s Analog Generator Panel Digital Generator Controls the operation of SR1 s Digital Audio Generator eo Analog Inputs Panel Inputs Panel Configures the analog inputs ss ss i CS S the analog inputs Digital I O Panel Configures the Digital Audio Carrier Input and Output configuration Sweep Panel Sets up one or two dimensional sweeps 2014 Stanford Research Systems 36 SR1 Operation Manual 2 3 1 Analog Generator Panel The Analog Generator Panel controls the operation of SR1 s analog gene
54. by the user in order to protect the instrument If this occurs the input voltage range will be shown in red as shown below Range if Auto 1160 0 rms f Remove the input and reset the range manually to restore normal operation Hi Res Converter Sampling Rate Selection The second tab of the Analog Input Panel contains a single control to select the sampling rate of the Hi Resolution ADC SR1 s analyzers use two different high quality audio ADC s the 16 bit high bandwidth converter operating at a fixed sampling rate of 512 kHz and the 24 bit high resolution converter which can operate at a variety of sampling rates Analog Inputs Hi Resolution Converter Sampling Rate 428kHz The converter runs at a fixed 128 kHz sampling rate 64kHz si The converter runs at a fixed 64 kHz sampling rate Digital OSR The converter runs at the Digital Audio Output Sampling rate that is set in the Digital I O panel Digital OSRx2 The converter runs at twice the Digital Audio Output Sampling rate that is set in the Digital I O panel The last two settings are useful for making cross domain measurements of D A converters using FFT Chirp or Multitone signals where the stimulus is generated in the digital domain but the measured signal is in the analog domain Using Digital OSR or Digital OSRx2 as the converter sample rate maintains synchronicity between the generator and the analyzer 2014 Stanford Research Systems SR1 Operation Manual
55. byte of the channel status block A red indicator indicates a CRC mismatch Lock Green Indicates that the digital audio receiver is locked to an input data stream Red indicates no lock Validity Red indicates that the validity bit has been set in one of the received sub frames This indicator glows red when either the Unlock or Bi phase error is detected Coding Bi phase error A valid digital audio signal remains at the same amplitude for no longer than two Unit Intervals Uls Except during preambles A red indicator indicates that this condition has been violated Parity Each digital audio subframe contains a parity bit which summarizes the parity of the remainder of the bits in the subframe A red parity indicator means that the received parity bit does not match the parity computed in the remainder of the subframe Carrier Status Tab 2014 Stanford Research Systems SR1 Operation iss Distal I0 m QuUcouUeE Impairment Carrier Status Bits Level MEEME chn status Measured Fs IME ser Status Delay AMIE Highiight i Delay Mode DigQut Digin Mone Data Aactive Biks TAMU COUR MU 23 16 z 0 TAMU UUM KOMMUNAN 23 16 5 0 CRE Lock Validity Conf Coding Parity Carrier Level displays the measured peak to peak amplitude of the digital audio carrier The displayed value is only meaningful when the input connector is set to BNC or XLR for optical or GenMon inputs the display shows the ampli
56. channel status in that byte triggers the corresponding event The watched bytes are configured in the Config tab of the channel status panel These event occurs when there is a change in the received channel status for channel B corresponding to the configured byte Ch A B User Bit This event occurs if there is any change in any bit of the the 24 bytes of the Activity received user bits Sweep Started Occurs when a sweep is started Sweep New Point Occurs at the beginning of each sweep point before settled data has been Start obtained Sweep New Point Occurs when a sweep point times out due to inability to obtain settled data Timeout Sweep New Point Occurs at the end of each sweep point after settled data has been acquired Done 2014 Stanford Research Systems SR1 Operation Manual Sweep Finished Occurs when the sweep finishes Trigger AO Trigger Occurs when the AO analyzer is triggered A1 Trigger Occurs when the A1 analyzer is triggered New Meas0 4 Up to 5 measurements may be watched by the events system These measurements are configured on the Config tab of the events panel The New Meas0 4 events are triggered when a new value for the corresponding measurements is calculated Digitizer Finished Occurs when the optional digitizer has finished analyzing a record and creating the Analysis digitizer measurements Displays Bar Limit Occurs when the value displayed in any bar chart exceed
57. converter s range must be set large enough to not overload on the the fundamental of the input signal When using the TDD as an input to the THD Analyzer the analog notch filter and post filter gain eliminate the fundamental and amplify the remaining distortion products so that the full range of the input converter can be applied to harmonics rather than the fundamental To use the THD analyzer with the Time Domain Detector select the type of one of the analyzers say AO to be TDD Set the input converter of the TDD to Hi Bandwdith and set the measurement type of the TDD to THD N in order to enable the notch filter Set the type of the A1 analyzer to be THD Set the input of the THD analyzer to Other Analyzer The THD will now use as its input the notch filtered signal from the TDD When using the THD analyzer in combination with the TDD to measure low levels of THD N be sure the TDD is set to the Hi Bandwidth converter Only the Hi Bandwidth converter signal chain employs the analog notch filter and and analog post filter gain that enables making the most sensitive THD measurements 2014 Stanford Research Systems SR1 Operation 161 2 4 6 IMD Analyzer The IMD Analyzer works in concert with the IMD generator to make three classic audio intermodulation distortion measurements The SMPTE Society of Motion Picture and Television Engineers standard RP 120 1994 standard also similar to the German DIN standard 45403 uses a signal c
58. d3 d3 Usasa Uso ong U2 d2 d3 d4 4 U 2 3 f1 iu U 2 3 f 1 U2 CCIF DFD fc center frequency fd difference da 2 Uz U frequency U average power of fundamental das Upsrsy Us 157 U components 3 7 Urea sta t Urosa l U DIM TIM fs Sine Frequency 15 kHz fq Square Freq 3 15 kHz Square Wave 3 kHz Sine Wave 15 kHz Note that the listed distortion products are not the complete set of distortion products for each type of measurement For instance in the CCIF DFD measurment there is a second order distortion product at the sum of the input frequencies as well as the difference However the listed distortion products represent the ones typically measured for each type of IMD measurement Averaging Averaging Speed very Fast l Synchronous Avg Clear The IMD analyzer implements a user selectable tradeoff between measurement speed and measurement precision Internally this is accomplished by varying both the number of FFT averages performed and the resolution of the FFT spectra When using the precise and very precise settings measurements will be noticeably slower but the measurement results will exhibit less variability Pressing Clear clears the average buffer and is useful for eliminating the transients caused for example by switching input ranges 2014 Stanford Research Systems 164 SR1 Operation Manual Weighting Weighting None sd When summing the harmonic amplitudes
59. dBvrms o 5k 10k 15k 20k 25k 30k Unaveraged Power Spectrum of Sine Noise 0 Sk 10k 15k 20k 25k 30k Averaged N 10 Power Spectrum of Sine Noise The second spectral output computed by the FFT1 analyzer is the Linear Spectrum The Linear Spectrum is computed by averaging the real and imaginary parts of each FFT separately The average of the real and imaginary parts are then used to compute the Linear Spectrum amplitude and phase In the Linear Spectrum unlike the Power Spectrum noise that is uncorrelated to the signal is actually reduced by further averaging Because of this use of the Linear Spectrum unlike the Power Spectrum requires that the time record be triggered so that the signal waveform will have the same phase relative to the beginning of the time record for each averaged FFT The linear spectrum has phase information associated with it It is important to note that the phase of the Linear Spectrum for the single channel FFT analyzer is only meaningful if the time record is triggered so that the signal has a constant phase relationship to the beginning of the time record Below the averaged Linear Spectrum Navg 100 is shown plotted with the averaged Power Spectrum for the same number of averages Note that averaging the linear spectrum does not reduce the variation in the noise floor but does reduce the amplitude of the noise In this case averaging 100 spectra has reduced the noise floor by about 20 dB dBYrmns 10k 15
60. extremes A range of color palettes are available Color palettes beginning with Inverse display on a white background while the remaining palettes display on a black graph background Eye Limits The Digitizer Display can test both the width of the eye opening and the maximum positive and negative voltage of the input signal against pre determined limits To enable limit testing press the Limits button on the Eye Diagram tab There are three sets of limits for the eye diagram The Outer Limits not to be confused with the classic television series of the same name are two voltage values which specify the maximum positive and negative values of the input signal 2014 Stanford Research Systems 210 SR1 Operation Manual Inner Upper Limit Outer Limit La 200m 0 200m 400r onim e00rm 1 0 Outer Limit Inner Lower Limit The Inner Upper Limit and Inner Lower Limit determine the minimum width of the eye opening The limit box can be forced to be symmetrical around the time axis by checking the Mirror Up Down box on the Inner Lower Limit tab The Inner Upper Limit can be made to be symmetric around the center of the eye opening by checking the Mirror Left Right box on the Inner Upper Limit tab Arbitrarily complex limit shapes can be created for the Inner limits by entering a sequence of Ul and Voltage values into the list on the Inner Limit tabs The Replace Add and Delete buttons can be used to modify the list and the shap
61. fho r Level 100 00 mitms jo ar Analysis Analysis Harmonics Lines Measure Append Traces Free Run Sweep The Distortion Quick Measurement panel offers fast access to three distortion related measurements THD Individual Harmonic Amplitude and an FFT of distortion THD measured using the THD Analyzer expresses the sum of all harmonically related distortion products either as an absolute level or as a ratio to the fundamental Individual Harmonic Amplitude Selected in the Analysis window by choosing Harmonics displays the amplitude of each individual harmonic as a function of harmonic number Finally the Distortion FFT uses the time domain detector and FFT analyzer in tandem to examine the frequency spectrum of the EUT output with the fundamental notched out Cabling Configuration The output of SR1 s analog or digital generator is connected to the EUT s input The output of the EUT is connected to SR1 s analog or digital inputs The number of inputs and outputs and the connectors used are specified in the Setup Panel Generator Configuration The output waveform is set to Sine or Low Distortion Sine For free run operation the amplitude of the generator is set to the value in the start column of the panel For swept operation the amplitude is swept from the start to stop value with the number of steps specified Analyzer Configuration The analyzer configuration depends on the the Analysis selected Fo
62. for writing and debugging scripts that provides simple visual access to the SR1 Basic interface A complete description of the GPIB command interface is given in the SR1 GPIB Programming Reference A complete description of the SR1 Basic interface is given in the SR1 Basic Programming Reference Refer to these documents for details of individual commands and functions The automation menu contains the following options Remote lfc Opens the remote interface debug panel This panel allows monitoring of remote commands sent to and by SR1 over each of the remote interfaces Allows a script to be run directly without opening the script development window pting Opens the script development window Scripting Log Opens a scripting log panel The scripting log is a form which scripts can write to Enable Learning Starts Learning Mode Keypresses will automatically be recorded in the current script 2014 Stanford Research Systems 242 SR1 Operation Manual 2 1 1 Remote Interface Panel The Remote Interface Panel is a valuable tool for developing and debugging remote applications using the GPIB command interface 225 Remote Interface Alzi AnladErE 1 00000 Yrms ESR Alzi Oi anigFredg alyvzre Oi anigFreg 1000 00 Hz E SR lt Local gt lt WeI 11 closed abort channel gt feWI 11 closed connection to ICHAN DEY 172 25 96 226 a Seared Error GPIB WxT 11 Clear Pending Macro Serial User 1 00000 YAMS
63. frequency domain analyzers with digital audio inputs 2014 Stanford Research Systems s2 SR1 Operation Manual Digital Audio Input Sampling Rate Digital Audio Input Sampling Rate ISR is set to the Digital Audio Output Sampling Rate Fs OSR ISR is set to the value measured by SR1 Fs Status Bits The sample rate embedded in the digital audio channel status is used as the ISR User The ISR is set to a fixed user entered value Reference Digital Input Signal Controls ignal cpl oc g Resi fet bits DeEmph None The embedded digital audio signal may be AC or DC coupled AC coupling inserts a DC blocking filter into the signal path with a pole at 4 HZ The Deemphasis control is included for future expansion and is not implemented in the current version of SR1 Input Resolution may be set between 8 and 24 bits The input data is truncated to the selected value Digital Audio Status Panel The Digital Audio Status Panel at the bottom indicates the current status of the received digital audio signal It consists of 6 indicators each of which glows green when the corresponding status condition is good and red when the status condition changes to bad The 6 status conditions detected are Status Condition CRC For AES EBU digital audio streams an 8 bit CRC code is generated from the received channel status bits in each block and compared to the value stored in the last
64. generator panel the actual generated amplitude for the waveform will be the nominal amplitude multiplied by the frequency response of the EQ file at the current frequency 2014 Stanford Research Systems 44 SR1 Operation Manual Amp o o000 WETS ka Freq fi 00000 kHz The Waveform Amplitude control sets the peak amplitude of most waveforms The Waveform Frequency control sets the frequency of many waveforms Generator Trigger Certain generator waveforms can generate a trigger known as a generator trigger which can be used by the analyzers to synchronize the analyzer to a certain portion of the waveform Triggering is a complex subject which is fully described in the Analyzers section and there are many different possible analyzer trigger sources besides generator trigger so in this section the discussion will be limited to those waveforms that provide generator triggers and where in the waveform the trigger generated When the generator is configured with multiple waveforms the first waveform which is generator trigger capable will be the source of all generator triggers The Analog Generator Waveforms In the following list waveforms that may not be combined with other waveforms are marked with an asterisk next to their names Normal Sine Low Distortion Sine Config 40 Sine aveform On fw EQT ine Aig 0 0000 WETS Freq fi 00000 kHz The sine wave is the most basic audio test wavefor
65. given to another control When cursors on a graph are active the knob controls the active cursor position Using the Virtual Keyboard For situations where it s necessary to enter text into a control and an external keyboard is not available the Virtual Keyboard can be selected from the SR1 keypad or from the Tools Menu The virtual keyboard offers the full functionality of a standard PC keyboard but can be operated with just a mouse or the trackpad 2014 Stanford Research Systems 16 SR1 Operation Manual On Screen Keyboard a Fa rid Fri r12 poof ok PTilelsl lslel7 e o o tse ACTE T ta ajwie rjtly ufijto e r i s feel end pda 7 lock EVES EIT FIGS st BABE ctrl Keyboard Shortcuts A number of keyboard shortcuts are available using either an external keyboard or the virtual keyboard to simplify the execution of common functions Key Sequence Action S O Opens the analog generator panel Opens the digital i o panel Opens the analog inputs panel Opens the monitors panel Opens the sweep coniroller panel Opens the scripting window Opens the AO analyzer panel Opens the A1 analyzer panel Auto reference analyzer AO Moves the current AO level A and B values to the dBrA and dBrB reference values Auto reference analyzer A1 Moves the current A1 level A and B values to the dBrA and dBrB reference values Emergency shutoff Turns off all channels of the analog and digital generators Restore g
66. headphone channels are fed the analog input A and B channels B the speaker is fed the summed A B signal Digital Level A The speaker and both headphone channels are fed the embedded digital audio input signal channel A Digital Level B The speaker and both headphone channels are fed the embedded digital audio input signal channel B Digital Level A The left and right headphone channels are fed the A and B channels of the B embedded digital audio input signal the speaker is fed the summed A B signal 2014 Stanford Research Systems SR1 Operation tor Monitor AO The speaker and both headphone channels are fed the AO monitor signal See the discussion below on the monitor signals The speaker and both headphone channels are fed the A1 monitor signal See the discussion below on the monitor signals Monitor A0 A1 The left and right headphone channels are fed the AO and A1 monitor signals the speaker is fed the summed AO A1 monitor signal Jitter The speaker and both headphone channels are fed a signal proportional to the current output of the Jitter Demodulator See the Jitter Analyzer section for a description of the function and controls pertaining to the Jitter Demodulator System Sound Level controls the volume of sound generated by the Windows operating system Note that sounds generated by the Windows operating system are not affected by the the volume control knob Checking Mute turns off Windows sounds altog
67. in impulse response measurements where several special properties of the MLS the autocorrelation of an MLS sequence is a delta function simplify the calculation of the impulse response Because SR1 uses a full 2 channel FFT analyzer to make impulse response measurements it is not necessary to use MLS noise as a stimulus any broadband signal will work Nevertheless SR1 includes the MLS waveform because of its historical association with impulse response measurements Amplitude controls as with all SR1 waveforms the peak value of the noise output The crest factor of the MLS waveform which is essentially a square wave is close to 1 The Length selection controls the length of the MLS sequence Selecting 13 for example chooses an MLS sequence whose repetition interval is 213 1 samples The Pink checkbox specifies that the noise output should be filtered by a 3 dB octave pinking filter While the power contained in a white noise signal is linearly proportional to the measurement bandwidth pink noise will have equal power in equal logarithmic frequency intervals e g the power contained in the 100 Hz to 200 Hz interval will be the same as the power contained in the 10 kHz to 20 kHz interval USASI Noise Config USASI Noise aveform On fw USASI noise is a special type of filtered noise designed to mimic the content of audio program material USASI noise is typically used in testing broadcast transmitters to measure compliance
68. input gain to Auto will yield the best results Occasionally with signals that have some transient component it may be necessary to manually set 2014 Stanford Research Systems 184 SR1 Operation Manual the gain to one of the three settings Digitizer Acquisition To begin digitizer acquisition press the Acquire button on the digitizer panel or the l icon on the SR1 speedbar Pressing the button opens a file dialog allowing the current digitizer record to be saved to disk or to load a previously saved digitizer record The format of digitizer files is detailed in the SR1 File Reference The Digitizer Trigger Once acquisition has begun the digitizer waits for the specified trigger event to occur The digitizer trigger can be set to trigger on a variety of events summarized in the table below Digitizer Trigger Receive Preamble A Preamble B Preamble A Ref Preamble B Ref Triggers on any channel A subframe preamble X preamble on the C1 input connector Triggers on any channel B subframe preamble Y preamble on the C1 input connector Triggers on any channel A subframe preamble X preamble on the C2 input connector Triggers on any channel B subframe preamble Y preamble on the C2 input connector Triggers on any channel A subframe preamble X preamble on the rear panel reference input connector Triggers on any channel B subframe preamble Y preamble on the rear panel reference input c
69. instance one may want to measure the characteristics of an external sweep generator not controlled by SR1 When performing an external sweep SR1 measures the frequency of the external source and then determines when that frequency meets preset criteria for starting the sweep ending the sweep or beginning a new sweep point When SR1 measures the external Sweep source it requires that the reading be Settled Refer to the Settling Panel for information on the different types of settling To configure an external sweep first make sure that the appropriate analyzers are active to make the measurement on which the external sweep will be based For instance to sweep based on a measured amplitude the Time Domain Detector should be selected Now click on the External node in the sweep source selection window When the External branch of the sweep source tree is opened the various measurements that can be used as external sweep sources are displayed Click on the desired measurement lolx E Time E Internal External Analog Freq Analog Freg B Analog Phase AB E Digital Freq E AC FFT lL FFT me The sweep source selection parameters are now displayed Inner Sweep outer SEED External Settings SOUPCE va Analog Freg Freg 4 imits teps Start 20 000 kHz sl Spacing 5 0000 Ho Stop 20 000 Hz Y in Level eas Hone fi 0000 Hz In an external sweep the Start Stop and Spac
70. is available for an accurate jitter FFT spectrum Probability Tab fessor AN 2 2 lt co E SRN E cock EEE Eye Diagram Probability Spectrum Time Rec 500m 0 v m log i Input Amplitude Jitter Amplitude Pulse Width ki E Pulse Rate The probability tab shows the histograms for input amplitude jitter amplitude pulse width and pulse rate All 4 histograms are normalized as probabilities i e the y value for each bin represents the fraction of the total events amplitude samples or pulses that fall in that bin A typical input amplitude histogram shown above shows two peaks corresponding to the positive and negative values of the input carrier signal For a professional or consumer digital audio signal the pulse width histogram typically shows two peaks corresponding to the numerous 1 and 2 UI wide pulses in the signal and a smaller third peak corresponding to the 3 UI pulses found only in the preambles of the digital audio signals The pulse rate histogram is simply the histogram of the reciprocal of the pulse width values 2014 Stanford Research Systems SR1 Operation 209 Eye Diagram Tab Outer Upper 7 Eye Diagram Probability Spectrum Time Rec Ci O oir 100n 150n smir Ama z scale cursor 1 Intensity 39 270 nsec 353 43 nsec E Root Outer Uppe Limits Outer Upper iM Inner Upper Ymir Ymax z palette cursor 2 Inner ee Outer Lower 3 4095 Y 3 4191
71. is by o 7 lt F s Fi gt definition a positive real quantity The numerator IS a complex quantity known as the cross spectrum When SR1 calculates the magnitude and phase of the frequency response it uses the following definitions 7 z Mag Cross Spectrum TrEQWETLCY HESPOTLSE IY SS cial aig i Power Spectrum A Frequency Response Phase 4 Cross Spectrum This technique provides a significantly more stable Frequency Response than if SR1 were to simply average the shot by shot frequency response Coherence The FFT2 analyzer also calculates another 2 channel scalar measurement the coherence Coherence is measure of the fraction of the output power at a frequency that is phase coherent with the input A 2014 Stanford Research Systems 144 SR1 Operation Manual coherence value of 1 means all of the output is phase coherent with the input while a value of 0 indicates the output is completely uncorrelated with the input Because only averaging over several FF Is reveals which portions of the spectrum are phase coherent and which are not the coherence measurement is valid only when averaging Mathematically the coherence is defined as CrossSpectrum x CrossSpectrum PwrSpecAPwrSpecB As an example consider the spectra below showing the frequency response and coherence of an 8 pole 6 zero elliptical filter with a 5 kHz passband frequency The spectra were taken with the FF T2 analyzer with the hi band
72. linearly scaled input data the inverted value is equal to the reciprocal of the ratio of the original y value to the invariant y value multiplied by the invariant y value After selecting a target X and Y value this routine returns a scaled version of the input data such that the new trace has the selected Y value at the selected X value Returns a trace containing the difference between the two selected input traces over the input data range If the input data is in dB units the difference between the dB values is calculated Multiply Returns a trace containing the product of the two selected input traces Equalize Returns a trace containing the input data weighted by the selected EQ file Ratio Returns a trace containing the ratio of the two selected input traces Trim Trims the input data do the selected input data range The returned trace has identical Y Values to the input trace but only contains points within the specified X range Returns a trace with the unwrapped version of the input phase frequency trace Unwrapping is done by inserting 360 jumps at appropriate break points in the phase vs frequency curve Group Returns a trace equal to the Group Delay of the input phase vs frequency trace Group Delay delay is defined as the derivative of the radian phase with respect to frequency Parametric Displays a parametric plot of the two selected input traces with their common X axis as a parameter Each pair of Y
73. lt EQdata Name ZerosImag Type DblArray Value 0 0 0 0 gt These two lines give us the 4 zeros at zero frequency specified in the transfer function Finally the last line in the file lt EQdata Name Gain Type 3 Value 7397235900 gt Specifies the overall multiplier for this transfer function A number of EQ files are supplied with SR1 By examining these files and using the information given above it should be possible to synthesize new EQ files corresponding to any pole zero type transfer function Sampled EQ Files Sampled EQ files represent a real magnitude only frequency response by storing a list of frequencies and responses and using spline interpolation to interpolate the values at other frequency points The list of y values in the files may be either pure numbers or dB ratios As an example the same A Weighting function that was described in pole zero format above can be represented in sampled format with the file below lt xml version 1 0 encoding UTF 8 gt lt DOCTYPE EQcurve SYSTEM EQcurve dtd gt lt EHOcurve gt lt MetaData gt lt Source Value C Program Files Borland CBuilder5 Projects Audio an EBOgen EQg en exe gt lt DocType Value EQcurve XML gt lt Version Value 1 0 0 gt lt Date Value 2005 05 05T15 54 12 gt lt MetaData gt lt EQdata Name Style Type 6 Value Freq Response gt lt EQdata Name Freq Type DblArray Value Oe OT OO 7 OS Ce OA 4 OS Oe Gy Os OF 0s O70
74. measurement must be known separately Thus in the following table only the sine relationships between the units is considered Analyzer Amplitude Units Analog Inputs dBVrms Decibels relative to 1 Vrms A signal with an amplitude of 20 dBVrms has a peak amplitude of 100 mVrms or 141 4 mVp Decibels relative to 0 7746 Vrms Historically the value of 0 7746 Vrms was chosen because it represents the voltage required to dissipate 1 mW in a 600 Q load back when 600 loads were used in audio signal chains Even though low output impedance high input impedance signal chains are more typical in today s audio circuits the dBu retains its importance as a unit of measurement Decibels relative to the dBrA Reference specified in the References Tab Decibels relative to the dBrB Reference specified in the References Tab Decibels relative to 1 mW into a load specified by the dBm Reference in the References Tab dBm 20 0 log Vrms 1mW Load Z Note that the load is for calculation purposes only the actual input impedance is unrelated to this value Watts into a load specified by the Watts Reference in the References Tab Watts Vrms Load Z Note that the load is for calculation purposes only the actual input impedance is unrelated to this value Analyzer Amplitude Units Digital Audio Inputs According to AES17 1998 r2004 Full scale amplitude is the amplitude of a 997 Hz sinewave whose positive peak value reaches the positive d
75. nalog Processing C Range Control Control stereo Level Heas E Digital Processing Hikes ADC Atte ater AA Gain Q aa a Ch E Level Chain Detailed TDD Block Diagram Analog HiRes Inputs Shown for Ch A Selected The analog front end is identical for both HiRes and HiBw converter selections The autorange control 2014 Stanford Research Systems ons SR1 Operation Manual automatically adjusts the input attenuation and gain and an analog frequency measurement is made on both input channels When the HiRes converter is selected the two signals are digitized by a 24 bit stereo ADC As in the HiBw case the phase and level measurements are computed by the DSP The Amplitude Chain unlike the HiBw case is implemented purely in the DSP For the HiRes converter the Bandpass Notch filter is implemented digitally by the DSP There is no post filter gain Bandwidth limiting and weighting filters are implemented digitally The DSP measures the amplitude signal with a selectable RMS Peak Quasi Peak response The amplitude signal is reconverted to an analog signal by a DAC and is output on the rear panel AO Monitor Out or A1 Monitor Out depending on which analyzer is being used BNC connector Finally the amplitude signal is sent to the Other Analyzer e g A1 if the TDD is active on AO or AO if the TDD is active on A1 If the Other Analyzer is an FFT analyzer whose input is set to Other Analyzer the FFT analyzer will display the
76. number of points in the sweep can be changed to give faster sweeps or higher resolution sweeps The settling parameters can be adjusted to fit the noise levels of the signals being measured THD N vs Amplitude Sweep Performs a THD N vs amplitude stereo sweep The AO and A1 analyzers are set to Time Domain Detector in order to measure THD N A digital 1 kHz sine waveform is setup with an amplitude sweep range of 1 mFFS to 1 FFS in 22 logarithmic steps Page 2 displays a graph of the A and B channel THD N vs Amplitude What you ll need to configure The default frequency is 1 kHz This should be adjusted as needed The settling parameters can be adjusted to fit the noise levels of the signals being measured The bandwidth of TDD can be adjusted to suit the measurement requirements THD N vs Frequency Sweep Performs a THD N vs frequency stereo sweep The AO and A1 analyzers are set to Time Domain 2014 Stanford Research Systems 286 SR1 Operation Manual Detector in order to measure THD N A low distortion 1 FFS sine waveform is setup with an frequency sweep range of 20 Hz to 20 kHz in 22 logarithmic steps Page 2 displays a graph of the A and B channel THD N vs Frequency What you ll need to configure The default sine amplitude is 1 FFS This should be adjusted as needed 2014 Stanford Research Systems SR1 Operation 287 2 9 5 Digital IO The Digital IO selection provides 3 measurement setups dealing with jitter
77. of 512 kHz and a 24 bit converter which operates at a variety of sampling rates Each converter has advantages depending on the specific application See the Specifications section for detailed information on each of the converters 512 kHz Selects the Hi Bandwidth 16 bit converter operating at a fixed output rate of 512 kHz This setting allows a maximum waveform frequency of 200 kHz 128 kHz Selects the Hi Resolution 24 bit converter operating at a fixed sample rate of 128 kHz providing a maximum waveform frequency of 57 6 kHz 64 kHz Selects the Hi Resolution 24 bit converter operating at a fixed sample rate of 64 kHz 2014 Stanford Research Systems SR1 Operation a providing a maximum waveform frequency of 28 8 kHz Synchronizes the sampling rate of the analog generator to the digital audio output sampling rate set in the Digital I O panel This setting is useful for performing cross domain measurements using the FFT Chirp waveform or Multitone techniques Maximum waveform frequency is 45 of the digital audio sampling rate Synchronizes the sampling rate of the analog generator to the sample rate of the received Digital Audio signal Maximum waveform frequency is 45 of the digital audio sampling frequency Mode controls the output mode of the Analog Generator Mono The same waveform is output to both the A and B channels The amplitude of each channel is still separately adjustable but the waveform is the same Each channel can
78. overwrite existing trace data a new trace is always created to contain the calculation results Trace Calculator Functions Returns a trace containing a constant value equal to the maximum value of the input trace over the input data range Returns a trace containing a constant value equal to the minimum value of the input trace over the input data range Returns a trace Retums a trace containing a constant value equal to the average value of the input trace a constant value equal to the average value of the input trace 2014 Stanford Research Systems SR1 Operation 201 over the input data range RMS or Linear averaging can be selected A third option variance does not return the average but rather the variance the RMS average deviation from the mean Returns a trace containing a constant value which exceeds 95 2o of the y values within the input data range Linearity Calculates a linear fit to the data within the input range and then returns a trace equal to the difference of the input data and the linear fit Returns a smoothed version of the data in the input range The smoothing algorithm is selected from the calculator panel Invert Returns a trace containing the inverted y data within the input rage For data displayed in logarithmic units the y values are sign inverted and translated so that the original data and the inverted data have the same value at the invariant X Value selected on the panel For
79. plus frequency response 2014 Stanford Research Systems 10 SR1 Operation Manual transfer function and interchannel phase Zoom changing the FFT frequency range is supported in the dual channel analyzer THD Analyzer Makes frequency selective Total Harmonic Distortion THD measurements both ratio and absolute Includes the ability to measure only selected harmonics IMD Analyzer Makes standard Intermodulation Distortion measurements including SMPTE DIN CCIF difference frequency and DIM TIM Multitone Analyzer Makes single shot multitone measurements allowing fast measurements of common audio parameters including noise distortion and level Jitter Analyzer Measures the jitter of the digital audio carrier in both the time and frequency domain including variable high and low pass filtering Histogram Analyzer Creates histograms of the analog and digital audio input signals Octave Analyzer Displays 1 1 3 and 1 12 fractional octave spectra Each type of analyzer has a corresponding panel with the appropriate controls for that analyzer and readouts for the analyzers measurements At any instant there are two active analyzers designated AO and A1 The user can select the type of A0 and A1 from any of the allowed analyzer types Each analyzer has controls which set the input domain to analog or digital audio and select the appropriate input channel For analog inputs SR1 offers an additional choice between two Ana
80. points adjust this to obtain the proper balance of update rate vs frequency resolution THD N vs Amplitude Sweep Performs a THD N vs amplitude stereo sweep The AO and A1 analyzers are set to Time Domain Detector in order to measure THD N A 1 kHz digital sine waveform is setup with an amplitude sweep range of 1 mFFS to 1 FFS in 22 logarithmic steps Page 2 displays a graph of the A and B channel results What you ll need to configure The default frequency is 1 kHz This should be adjusted as needed The settling parameters can be adjusted to fit the noise levels of the signals being measured The bandwidth of TDD can be adjusted to suit the measurement requirements THD N vs Frequency Sweep Performs a THD N vs frequency stereo sweep The AO and A1 analyzers are set to Time Domain Detector in order to measure THD N A low distortion 1 FFS sine waveform is setup with an frequency sweep range of 20 Hz to 20 kHz in 22 logarithmic steps Page 2 displays a graph of the A and B channel results What you ll need to configure The default sine amplitdue is 1 Vrms This should be adjusted as needed 2014 Stanford Research Systems SR1 Operation 285 2 9 4 Digital Digital The Digital Digital selection lists measurements with both a digital stimulus and response The outputs and inputs are set to the XLR connector The output sampling rate is defaulted to 48 kHz What you ll need to configure for all these setups Change the d
81. points in the sweep can be changed to give faster sweeps or higher resolution sweeps The settling parameters can be adjusted to fit the noise levels of the signals being measured Wideband FFT This setup is similar to the audio band FFT setup described above except that the FFT frequency range is set to the maximum 200 kHz value The AO analyzer is configured as a dual channel FFT analyzer with a span of 28 8 kHz and continuous averaging The Analog Generator is loaded with a low distortion sine waveform set to a default frequency of 1 kHz and an amplitude of 1 Vrms Page2 contains a graph displaying the A and B channel power spectra while Page 3 contains a graph displaying the A and B channel time records What you ll need to configure If your e using the analog generator adjust the amplitude and frequency of the signal to match your requirements Otherwise you can turn the analog generator off The amount of averaging in the FFT analyzer can be adjusted to suit the noise level of your signal The FFT resolution is defaulted to 4k points this can be adjusted 2014 Stanford Research Systems 280 SR1 Operation Manual THD N vs Amplitude Sweep Performs a THD N vs amplitude stereo sweep The AO and A1 analyzers are set to Time Domain Detector in order to measure THD N A low distortion 1 kHz sine waveform is setup with an amplitude sweep range of 50 mVrms to 5 Vrms in 22 logarithmic steps Page 2 displays a graph of the A and B ch
82. pole elliptic low pass filter with a passband edge of 80 kHz that satisfies the requirements of AES17 1998 r2004 section 4 2 1 1 Passband ripple is lt 1 dB Stopband attenuation is gt 60 dB for f 96 KHz Note that the availability of some of the bandwidth limiting filters depends on the current analyzer sampling rate For instance if the digital audio inputs are selected and the digital audio sampling rate is 48 kHz giving a nominal bandwidth of 24 kHz an 80 kHz high pass filter doesn t make sense and that selection will be unavailable Weighting Filters Weighting CCITT The following standard weighting filters may be applied to the amplitude measurement These filters are applied to the amplitude signal only they do not affect the level measurements Weighting Filters None No weighting filter is applied A weighting filter is applied A weighting is specified in ANSI standard S1 4 1983 and is typically used for noise and THD N measurements with audio applications C Msg Wt The C Msg weighting filter specified in IEEE Std 743 1995 is intended to be used for noise measurements associated with voice transmission telecommunications CCITT The CCITT weighting filter defined by ITU T Recommendation O 41 is another telecommunication noise weighting filter CCIR wtd Filter defined in ITU CCIR Rec 468 4 for audio noise measurements Designed to be used with the Quasi Peak setting on the Time Domain Detector
83. pressing Resize For instance to create a limit that consists of a single line segment resize the limit to 2 and then adjust the starting and ending X and Y values for the single limit segment Upper or lower limits can be deleted with the Delete button on the limit tab and can be saved to disk with the Save button To recall a previously saved limit use the button next to the the New button and select the previously saved file from the file dialog A limit can be made a single limit in which case it has no X axis and only a single Y value that is applied to all the points in the trace A single limit can be thought of as a horizontal line that limits the entire trace Once a limit has been converted to a single limit it cannot be undone 2 5 1 1 Trace Calculator Trace Calculator The trace calculator is a powerful tool for manipulating data contained in graph traces Each calculator function takes as input one or more input traces and produces a new trace which contains the specified calculator function output If the function requires a single input the active trace is used by default If a second trace input is required the user can select from among all available graph traces The Trace Calculator can be accessed in two ways by right clicking the active trace in the trace listing box or by clicking the calculator icon on the speedbar After selecting the desired function from the 2014 Stanford Research Systems
84. range can be selected When the sweep is started the selected source parameter will be set to the Start value and when the sweep is finished the source parameter will have the Stop value The Stop value can be greater or less than the Start value i e both forwards and backwards sweeps are allowed Steps sets the number of steps in the sweep the number of points in the sweep is the number of steps 1 since the sweep always includes the start and end points When it is selected the Step Size parameter will automatically adjust to a step which covers the selected source parameter range in that number of steps If Log Step Size is unchecked then the sweep progression is linear the source parameter is incremented by the Step Size after each sweep point If Log Step Size is checked the sweep progression is logarithmic and the source parameter is multiplied by the sweep increment after each sweep point If the log or linear step size is entered the Steps parameter will be adjusted to a value which covers the selected range with the chosen sweep increment up to a maximum number of 10 000 sweep points When an internal sweep is finished the values of all swept parameters are returned to what they were before the sweep was started Internal Table Sweeps In addition to setting the sweep limits and step size the X axis points for an internal sweep can be explicitly set using a Table Sweep To setup a table sweep create an ASCII text fil
85. resolution relative to the number of tones in the stimulus Signal even a single FFT can yield a wealth of audio information by examining the amplitudes in three categories of FFT bins bins in which a generator tone is present the amplitudes of which can be used to derive the frequency response of the DUT bins in which harmonic and intermodulation distortion products of the tones in the stimulus signal are present which can be used to compute the THD and IMD characteristics of the DUT and bins in which their are no signal tone or distortion products whose amplitude represents the noise of the DUT By carefully examining the amplitudes of these three types of FFT bins it is possible to simultaneously measure Frequency Response THD N THD IMD and noise vs frequency for a device based on a single captured FFT record measurements that would otherwise take several separate swept measurements Multitone Analysis is typically a synchronous measurement tones are generated exactly on bin frequencies and the multitone FFT analyzer is run without a window uniform window ensuring that each tone will occupy a single bin in the analyzer spectrum For this to be true the device under test cannot shift the frequencies in the stimulus signals When this condition is met the multitone analyzer MTA can use an interesting technique to measure the noise of the DUT even close to tone and distortion frequencies When Noise Analysis is enabled the MTA sets the n
86. setting of each transmitted user bit The bottom line of hex numbers indicates the current values for each of the received user bits Received user bits can be highlighted in yellow when they differ from the corresponding transmitted bit or when they differ from the corresponding received bit on the other channel The Highlight Differences control on the 2nd tab of the Digital I O panel controls highlighting in the user status panel and the channel status panel 2014 Stanford Research Systems SR1 Operation oot 2 3 7 Sweep Panel Sweep Concepts SR1 operates in two different modes Free Run and Sweep In Free Run mode the analyzers make measurements continuously and update measurement results on displays and panels as each new measurement is available This is convenient way to use the instrument in a benchtop setting while debugging hardware or quickly measuring the performance of a new device Sweep is a more structured operating mode suitable for formal repetitive testing to standards Sweeps involve choosing a sweep source and sweep data The sweep source defines the X axis of the sweep it specifies a series of points at which SR1 will take measurements There are four types of sweep sources time sweeps in which measurements are made at specified time intervals internal sweeps in which measurements are made at fixed values of some internal parameter such as generator frequency external sweeps where measurements are made
87. spectra Values from 256 lines to 32k lines can be selected In the resolution control the time to acquire a time record for the selected resolution and current bandwidth selection is shown alongside the number of lines Obviously the higher the selected spectral resolution the longer it will take to acquire the time record for that spectrum Averaging Averagin sirg Ayigs Clear Avg Done Exponential 1 _ Fixed Length Continuous Cont PkHold The Power Spectrum Linear Spectrum and Frequency Response are all averaged quantities The amount and type of averaging for both spectra is specified by the controls above The averaging type controls determines how each of the averaged spectra is averaged None implies that no averaging is performed In the magnitude of the Power Spectrum and Linear Spectrum are the same and reduce to the magnitude of the last individual FFT Fixed Length averaging means that the analyzer will average the selected number of spectra and then stop Continuous averaging continuously averages the spectra weighting more recent results exponentially more than older spectra The two Peak Hold selections only affect the averaging of the Power Spectra When Peak Hold is selected instead of 2014 Stanford Research Systems 150 SR1 Operation Manual averaging successive power spectra each bin of the new spectrum is compared to the current buffer if the value in the new spectrum is great
88. switch must be on the same ethernet network as SR1 Enter the IP Address and port that matches the address and port that the switch box was configured with The default port for the SR10 11 12 switch boxes is 600 If desired a switch connected to SR1 via ethernet can in turn be daisy chained via RS 232 with up to 15 additional switchers The first switch chain address 0 receives commands from SR1 via TCP IP and the retransmits the commands via RS 232 down the daisy chain Configuring the Switcher Network The Switcher Configuration Panel displays two different views of the network of attached switchers Switcher Config Daisy Chains Networks Serial F COMI 7 Outputswitch1 7 OutputSwitcho TCP IP 192 168 1 18 600 4 BalancedOutputSwwitch Mew Switch Close The Daisy Chain tab shows the attached switchers organized according to how they are connected to SR1 The Serial node of the tree shows all the COM ports found in the instrument and the switchers connected to each COM port Each switcher must be assigned a unique chain address which identifies it The chain address is assigned by setting DIP switches on the back of each switcher module The TCP IP node shows the IP addresses and port assignments of each switcher attached via 2014 Stanford Research Systems 228 SR1 Operation Manual the ethernet network Ethernet addresses and port assignments are configured using the switchers web interface
89. the Digital I O panel For the default 24 bit output resolution setting 1FFS 279 1 8 388 607 dec The peak value of the waveform expressed as a hexadecimal code The conversion of hexidecimal code to FFS depends on the setting of the Output Resolution in the Digital O panel For the default 24 bit output resolution setting 1FFS 223 1 Ox7fffff hex Decibels relative to the Vrms value calculated with the V FS Volts Full Scale reference set in the digital generator panel For example with a V FS value of 2 Vrms an amplitude of 20 dBVrms corresponds to 1 Vrms which in turn corresponds to an amplitude of 0 05 FFS Decibels relative to 0 7746 Vrms Hiistorically the value of 0 7746 Vrms represents the voltage required to dissipate 1 mW ina 600 Q load If V FS is set to 1 Vrms than an amplidue of 0 dBu correesponds to 0 7746 which in turn corresponds to 0 7746 FFS Decibels relative to the dBr reference set in the references section of the digital generator panel The dBr reference is always set in units of FFS Thus with the dBr reference set to 5 FFS an amplitude of 0 dBr corresponds to 5 FFS 2014 Stanford Research Systems 2 3 2 2 SR1 Operation Manual waveforms All frequency units except Hz make use of the Frequency Reference which is set in the References Box on the_Digital Generator Panel Description The fundamental unit of frequency 1 Hz 1 cycle per second F R Ratio relative to the Frequency Reference
90. the FFT analyzer is also 64 KHz Just underneath is the control for setting the number of FFT lines resolution The resolution can be set to values between 256 lines and 32k lines Unlike most Audio Analyzers SR1 s FFT analyzer doesn t operate at a fixed frequency range from DC to Fs 2 The Bandwidth control allows setting the measurement range to Fs 2 Fs 4 etc all the way down to Fs 2048 The full FFT resolution is applied to this narrower frequency range which can be moved to any position in the range of DC to Fs 2 using the Start Center and End controls We will illustrate this shortly Averaging can be applied to the FFT to lower the shot to shot variation in the noise or in some cases to actually lower the amount of noise Select 5 averages in the Avgs field to create a nice stable FFT display To display the FFT results we ll need to create a graph Click on Page 2 of the page control to give us some room for the graph and click on the a icon to create a new graph Maximize the graph with the standard maximize control in the upper right Now we need to add some data to the graph Click the Plus sign icon at the top left of the graph to add a trace to the graph The Add Trace menu appears Open the A1 FFT node to see the measurements produced by the A1 FFT analyzer A Time Record MN Power Spectrum MN Linear Magnitude MN Linear Phase Analog In DigAud In EQ Curve Sweep OK Cancel Show Automat
91. the Frequency Response This time domain signal represents Response the response of the system to a narrow impulse excitation even though it can be measured using any broadband source Anechoic The frequency response magnitude calculated from the time gated no reflections Response impulse response Multiple reflection paths introduce can produce interference Magnitude which obscures the true frequency response of the DUT Anechoic The frequency response phase calculated from the time gated no reflections Response impulse response Multiple reflection paths introduce can produce interference Phase which obscures the true frequency response of the DUT Energy Time The magnitude of the analytic impulse response or the envelope of the impulse Curve response 2014 Stanford Research Systems 148 SR1 Operation Manual The FFT2 Analyzer Panel SOUrCE Converter Fs Measurement Meas References Trigger Bandwidth Resolution 4cq Time Baseband ik 32 0 msec 32 000 kHz Iw Show Aliased Lines Stark Center End 0000 Hz F 16 000 kHz oF 32 000 kHz a Fai Averadgin aig Ayigs Clear Avg Done Exponential 5 _ a Because both the A and B inputs of the selected input domain are always used by the FFT2 analyzer the Source selection for FFT only offers the choice of Analog or Digital The Converter and Fs controls are common to all analyzers Like the FFT1 analyzer th
92. the analyzer can apply any of the standard weighing filters to the individual harmonic amplitudes The table below lists the available weighting filter and their typical applications Weighting Filters None No weighting filter is applied A weighing filter is applied A weighting is specified in ANSI standard 81 4 1983 and is typically used for noise and THD N measurements with audio applications C Msg Wt The C Msg weighting filter specified in IEEE Std 743 1995 is intended to be used for noise measurements associated with voice transmission telecommunications CCITT The CCITT weighting filter defined by ITU T Recommendation O 41 is another telecommunication noise weighting filter CCIR wtd Filter defined in CCIR Rec 468 4 for audio noise measurements Designed to be used with the Quasi Peak setting on the Time Domain Detector CCIR unwtd Filter defined in CCIR Rec 468 4 Annex Il for making unweighted noise measurments CCIR 2 kHz This filter is identical to the CCIR wtd filter except the normalization is changed so that the filter has unity gain at 2 kHz instead of 1 kHz 2014 Stanford Research Systems SR1 Operation 165 2 4 7 Multitone Analyzer Multitone analysis is a technique whereby a device under test is subject to a stimulus signal containing a number of discrete tones whose frequencies are adjusted to fall exactly on fft bin frequencies of the multitone analyzer If the analyzer has sufficient
93. the generator The first Sine 0 refers to the 1 kHz fundamental The second Sine 1 is the distortion sinewave whose amplitude we re going to sweep Double click on ChA Sine 1 Amp to select it and return to the sweep panel 2014 Stanford Research Systems Getting Started 25 alas weep Controller Inner Sweep Outer Sweep External Settings Pre swp Delay 200 msec Meas Timeout 10 0 sec Rpt Select the start amplitude as 100 uVrms and the stop amplitude as 100 mVrms Leave the number of sweep steps at 30 but select Log Step Size Next we need to select that data that will be measured in the sweep Up to 6 separate measurements can be recorded at each sweep point Click on the Sweep Data to specify the sweep data From the Sweep Data Selection menu open the AO node since the measurement we re going to record THD N is made by the currently active Time Domain Detector on AO E poweep Data Selection H Analog Freg H Digital Freq 2014 Stanford Research Systems SR1 Operation Manual Select Analog THD Ratio A by double clicking it We re now ready to set up the graph to display our sweep Return to Page 2 of the page control In the trace listing panel at the bottom of the graph unclick the Power Spectrum trace that we were looking at previously Now use the Plus button at the top of the display to add another trace to graph This time open the Sweep node and select THD Ratio A S
94. the outer sweep axis while external sweeps can only be selected on the inner sweep axis Starting and Stopping Sweeps and Free Run To start free run mode press the free run start button gt l or the lt Run gt key on the keypad While in free run mode press the pause button which then changes to the resume button which resumes free run measurements This is equivalent to using the lt Pause gt button on the keypad To start a sweep press the sweep start button A or press the lt Sweep gt key on the keypad During the sweep the pause resume function is controlled by the and buttons as it is for free run When the sweep is complete the sweep done indicator kaa is displayed The current sweep free run status is always shown at the bottom of the screen Free Run Active 2014 Stanford Research Systems 92 SR1 Operation Manual The Sweep Panel ioi Inner Sweep outer Sweep External Settings SOURCE anlgGen Ch Sine O Freq imiks Stark 20 000 kHz Y F Stop 20 000 Hz Y ultiply By 794 33 ri Log Step Size r Table Sweep Mone ween Data 40 Time Dom Det Analog Amplitude A Mone Mone Mone Mone Mone Fre swp Delay 200 msec Meas Timeout 10 0 sec Rpp Configuring the Sweep Source Pressing the Source button opens the Sweep Source Selection Window Sweep Source Selection oxi El Time Internal E Alyzr Ol Anlgaen et ThanGain Cha
95. the panel For swept operation the frequency is swept from the start to stop value for each amplitude value specified Analyzer Configuration One or both analyzers depending on the number of output channels specified in the Setup Panel are configured as Time Domain Detectors The upper and lower bandwdith limits are adjustable in the Analysis section of the panel The THD N can either be displayed as the absolute amplitude of the noise plus distortion or the Free Run Configuration A barchart display is created for each output channel showing the instantaneous THD N of the EUT Sweep Configuration A graph will be created for each channel showing a graph of THD N vs Freq for the selected range of amplitudess Log spacing can be selected for both the frequency and amplitude sweeps The settling parameters of the THD N measurement can be adjusted directly from the Settling tab of the panel 2014 Stanford Research Systems 262 SR1 Operation Manual Settling Profile Delay Exponential 20 000 msec Precesion nPoints Threshold p A000 So 3 p 000D mrm Refer to the Settling Panel chapter for a more detailed discussion of what each of the controls means n general specifying a smaller precision window and larger value for nPoints will decrease the noise and glitches in the sweep at the expense of increasing the time required for the sweep Some experimentation may be required depending on the characteristics of the EUT
96. the panel shown above a CSA Byte event will be fired every time the received channel status in byte 2 of channel A changes Watched Measurements Up to 5 measurements may be monitored by the events system The 5 events New Meas0 4 are fired when SR1 computes a new value for the associated measurement Use the ellipsis button next to the event and navigate the tree to associate a measurement with one the 5 events 2014 Stanford Research Systems 226 SR1 Operation Manual 2 6 3 Switcher Configuration Panel SR1 is designed to work with the SR10 SR11 and SR12 switching systems from Stanford Research Systems to provide a flexible system for multiplexing SR1 s inputs and outputs both XLR and BNC SR10 is an XLR input switcher capable of connecting any two of its twelve XLR inputs to SR1 s analog inputs SR11 is an XLR output switcher capable of routing SR1 s two analog output channels to any or all of its 12 XLR output connectors SR12 is an BNC switcher capable of being configured as either an input or output switcher All three switchers can operated in stereo mode connecting a pair of inputs or outputs to one of 6 pairs of switcher connectors or mono mode connecting a single input or output to one of 12 switcher connectors All three switchers can be controlled using a serial RS 232 connection or va TCP IP over ethernet Up to 16 switchers can be daisy chained together providing a switching capability of 192 channels U
97. this use of the Linear Spectrum unlike the Power Spectrum requires that the jitter time record be triggered so that the signal waveform will have the same phase relative to the beginning of the time record for each averaged FFT As an example consider the spectra below both of the same 48k digital audio carrier with 200 mUI of added sine jitter at 10 kHz SGU 0 5k 10k 15k 20k 25k 30k H The top trace shows the averaged N 100 Power Spectrum The bottom trace is the N 100 averaged Linear Spectrum taken with the trigger enabled Note the substantial improvement in the noise floor and the additional spectral detail revealed in the linear spectrum Highpass lowpass and weighting filter selections are available in Frequency Domain analysis but these filters are applied in the frequency domain after the spectra are computed Jitter Analyzer Measurements Measurement Description Time Domain Amplitude The amplitude of the jitter signal calculated from an RMS sum of the jitter input samples Though the amplitude is calculated with an RMS sum it is expressed in equivalent peak units of sec or Ul Physical Sample The effective sampling frequency of the input digital audio signal or in the case of Rate a square wawe input the square wawe frequency Frequency Domain Time Record The amplitude vs time record of the jitter signal When displayed on a graph this produces an Oscilloscope type display Power Spectrum The p
98. to find the optimal values of the settling parameters 2014 Stanford Research Systems SR1 Operation 263 2 8 6 Frequency Response Panel E Measure Freq Resp aag Input Output Measurement Settling Signal waveform Chirp ad Start Stop Steps Log 100 00 mrs 1 0000 rms Level Analysis Lines tk Measure Ratio ki Append Traces Free Run Sweep The Frequency Response Quick Measurements uses the FFT analyzer and the chirp waveform to rapidly measure the frequency response of the EUT The chirp waveform outputs equal power in each FFT bin Thus a single FFT using this waveform as a stimulus reveals the frequency response of the EUT Cabling Configuration The output of SR1 s analog or digital generator is connected to the EUT s input The output of the EUT is connected to SR1 s analog or digital inputs The number of inputs and outputs and the connectors used are specified in the Setup Panel Generator Configuration The output waveform is set to Chirp For free run operation the amplitude of the generator is set to the value in the start column of the panel For swept operation the amplitude is swept from the start to stop value with the number of steps specified Analyzer Configuration One or both analyzers depending on the number of output channels specified in the Setup Panel are configured as Single Channel FFT Analyzers If level is specified in the Analysis sectio
99. trace Any subsequent scaling operation such as manually setting the scales or pressing one of the autoscale buttons will scale all the traces with compatible axes rather than just the active trace X Axis and Sweep Controls SAIS Index zl Scale l Sweep ppend E The X axis controls only apply to strip chart type traces The X axis for such traces can be plotted as a function of Index which is simply a sequential integer label that is applied to each new point or as a function of real time elapsed since the Start button was pressed If index is selected all the points will appear to have equal spacing even though the actual measurement intervals may vary widely The index numbering or the time origin may always be reset by pressing Start on the speedbar or on the keypad again The second control governs the scaling of strip chart traces If Scale is selected the graph 2014 Stanford Research Systems SR1 Operation 197 automatically autoscales the Xdata each time a new point is added to the trace so that the full range of data is always shown on the graph If Fixed is selected the graph scale is not updated automatically so that any new points that are added with X values exceeding the current range will not be seen Pan maintains the X axis range until new points reach the right hand edge of the graph At that point the maximum X value pans to keep up with the new data while maintaining the total width of the X axi
100. unit which represents 1 100 of a semitone of the musical scale 12 semitones make up an octave Thus a cent is 1 1200 of an octave octaves An octave is a factor of 2 in frequency Thus a frequency value of 3 octaves with a reference of 1 kHz gives a waveform frequency of 8 kHz decades An decade is a factor of 10 in frequency Thus a frequency value of 2 decades with a reference of 2 kHz gives a waveform frequency of 200 kHz Engineering Units ence EU Vrms A 1 0000 B 1 0000 EU Label EU A ELI B When a transducer is connected to the input of SR1 it is convenient to have measurement results displayed in the units of what the transducer is measuring rather than Volts For instance a microphone might be calibrated in V Pa or an accelerometer in V g The Engineering Units selection allows SR1 to display measurement results in Pa or g in these cases rather than Volts Transducer units are known as Engineering Units To use Engineering units enter the Engineering units Vrms conversion factor for the A and B channels as well as the label corresponding to the Engineering Units Engineering units will then show up when appropriate in all analyzer unit lists 2014 Stanford Research Systems oa SR1 Operation Manual Analyzer References Analog Digital der 4 1 0000 VIS oo WIFS 1 0000 VEINS OBrB 1 0000 Yrms der4 1 0000 FFS dBrri 600 00 ohms dErE 1 0000 FFS watts 2 0000 ohms Freg 1 0000 kHz Freg i 0000 kHz
101. values from the two input traces are used to create an XY pair in the returned trace For instance if the same trace is selected for both inputs the result will be a straight line with a slope of one whose extent in X and Y is equal to the Y range of the input data The selected trace Y values become the Y values for each point the second trace Y values furnish the values for the displayed points Create an saves an EQ file corresponding to the selected trace The selected X value is assigned a value of 1 in the EQ file All other points are scaled relative to that point Make ARB _ Creates and saves an ARB file corresponding to the selected trace The resulting file can be used by the arbitrary waveform generator 2014 Stanford Research Systems 22 SR1 Operation Manual 2 5 2 Bar Chart The Bar Chart is optimized for the display of a single measurement The chart consists of two main areas the top portion consists of a large numerical display of the selected measurement and the bottom portion contains a bar chart showing the current value of the measurement along with statistics of the measurement mean maximum minimum and standard deviation DigAud Carrier Freq i ES The Barchart Speedbar i a a The bar chart speedbar contains the controls necessary to configure the display The button displays the Add Measurement panel which allows selection of the measurement displayed on the bar char
102. waveform with a definite frequency i e not noise is chosen Sweep 1 follows the frequency of the Inner Sweep axis while Sweep 2 follows the frequency of the outer sweep axis In general it is better to select one of the generator or sweep tuning sources instead of measured frequency when possible to eliminate possible jitter in the filter frequency 2014 Stanford Research Systems 12 4 SR1 Operation Manual Bandwidth Limiting Filters Var LP 22 000 kHz 756 kHz Fs einer Hiv Filters Var LP 20 kHz 4E51 7 40 kHz AES17 Several different high and low pass filters to limit the bandwidth of the amplitude measurement These filters are only applied to the amplitude signal they do not affect the level measurements For low frequency high pass bandwidth limiting the following filters can be selected BW Limit High Pass selection No high pass filters are applied Selects a fourth order butterworth high pass filter with a corner frequency of 22 Hz Selects a fourth order butterworth high pass filter with a corner frequency of 100 Hz Selects a fourth order butterworth high pass filter with a corner frequency of 400 Hz Sharp 400 Hz Selects a sharp 10th order elliptical highpass filter with a cutoff of 400 Hz passband ripple of lt 1 dB and a stopband attenuation of 125 dB Selects a 4 pole elliptic high pass filter with a passband edge of 20 kHz that satisfies the requirements of AES17 1998 r200
103. with scripts gives the user a window into the internals of SR1 and can offer almost unlimited flexibility if used correctly However care must be used when customizing events with scripts Associating time consuming scripts with frequently occurring events can adversely affect SR1 s response time and even hang the instrument Be extremely careful when writing event scripts 2014 Stanford Research Systems SR1 Operation 225 Configuration Tab Events Contig Tones Log File EEE o mE E mm Watched Channel Status Hide 0 Byte 1 l Byte 2 z Off od SRi Log Opened On 12 15 2009 3 04 12 PM 12 15 2009 4 10 18 PM Sweep Started 3 off 4 off 12 15 2009 4 10 30 PM Sweep Finished Watched Measurements Tones Pressing the buttons labeled 1 5 previews the tones that can be associated with each event Log File Use the ellipsis button to open a file dialog to select a log file The default is SR1Log txt Pressing View displays the current contents of the file Note that the form is not updated automatically pressing View displays a new snapshot of the file Watched Channel Status Up to 5 bytes of channel status may be monitored by the events system There are 5 events for channel A CSA Byte0 4 Change and 5 events for channel B CSB Byte0 4 Change Each event is associated with one of the bytes in the received channel status by setting the appropriate control to the desired byte For instance in
104. 00 n W Relative value tart sweep at Within tolerance within tol or beyond First settled reading top sweep at f Within tolerance Return to within tol The Start Sweep At parameter controls the starting of the external sweep Start Sweep At Within Tolerance The external sweep starts when a settled measurement of the sweep source is within tolerance of the Start value of the sweep Within Tolerance The external sweep starts when a settled measurement of the sweep source is or Beyond within tolerance of the Start value of the sweep or greater than the Start value First Settled The sweep starts at the first settled measurement of the sweep source regardless Reading of the relation of that value to the Start parameter Once the external sweep is started SR1 monitors the sweep source for a settled measurement The value of the settled measurement is compared to the last sweep point and if it exceeds the Spacing value a new point is declared External sweeps are unidirectional The direction is determined by the relative values of the Start and Stop parameters Only settled readings that exceed the spacing in the direction of the sweep will result in a new sweep point The external sweep ends when Stop Sweep At Within Tolerance The external sweep ends after a settled measurement of the sweep source is within tolerance of the Stop value of the sweep Retu
105. 09 Ooh 0u 27037 Ue 05060077 Da ey O s Oe hp 2g Os Oy OF Vp Op Dp 20 y 30 4 Oy 5 Oe 00 2107 80 007 LOD R200 7 gt lt EQdata Name Resp Type DblArray Value 0 000300773933872953 0 00302174093251185 0 009325402777401384 0 0716 7511398949463 030636637564135 0 0444168356181221 0 0596245322111266 0 075876407055739 0 0928635700095015 110342389581799 0 286838481522557 0 443881655056602 0 577159672164201 0 688018968891406 778916516580281 0 852785568914054 0 91256030198955 0 960885423147423 1 00000000003858 14837827316987 1 15195959556707 1 11740718036267 1 06603938913942 1 00595593739291 941761216135588 0 876545831584705 0 8124486702114 0 750902168765253 0 341162925654635 178593707588384 0 107123333734722 0 0707303220476656 0 0499780417068754 0 037110212737056 0 0286105664352016 0 0227144341264837 0 01846210568194 0 00466695600216974 gt lt EQcurve gt Oo OF OO O 2014 Stanford Research Systems 308 SR1 Operation Manual The line lt EQdata Name Style Type 6 Value Freq Response gt identifies the file as a sampled EQ file with the values given as pure numbers To specify a file where the values are given in dB use lt EQdata Name Style Type 6 Value Freq Response dB gt The next line lt EQdata Name Freq Type DblArray Value OO gO nO 5 Oe E O47 0 07 0206 0207 008 0 097 Oe Lye Zn E P N ss 1 ee Terora oror Oe 0710 72070 407 007 60 10 80 90 100 200 7 gives the frequency values for each
106. 1 SR1 will not response to further VX 11 commands until the correct password has been entered 2014 Stanford Research Systems 22 SR1 Operation Manual 2 6 2 Events Panel The events panel allows the user to link the occurrence of certain events within SR1 to a variety of user specified actions file logging audio alarms running of scripts or the firing of COM events Configuration of the events panel is not necessary during ordinary use of SR1 but can be used to create highly customizable test configurations The panel shows each of the events that can be tracked arranged in functional groups The leftmost checkbox enables tracking of the selected event this box must be checked for all the other configured actions to occur The next check box to the right of the event description enables logging of the event to a text file The Tone control allows selection of an audio tone which is played when the selected event occurs The ellipsis button to the right brings up a file diaglog allowing selection of a script which is executed when the event occurs Finally the last control on the right allows firing a COM event when the event occurs which can be trapped and acted upon by a local script or a remote program written in Visual Basic Microsoft Office or other COM enabled programming environments Events Events Config Enable Logging Tone Script Com Event nalog Inputs i Analog Ch A Scale Change Analog Ch 6 Scale Ch
107. 1 Operation Manual arbitrary waveform at each sampling interval Use the a button to open a file dialog to select an arbitrary waveform file If multiple columns are detected in the file SR1 displays a dialog asking which column to load The number of points read from the table is then displayed in the corresponding control The amplitude entered in the Arb amplitude control is assigned to the maximum value found in the table The absolute scaling of the table values does not affect the output waveform Thus the following table of values in the file 0 0 1 0 2 0 3 0 4 0 5 produces a linear ramp from 0 to 2 0 Vrms if the amplitude control is set to 2 0 Vrms The maximum amplitude for arbitrary waveforms is half that of sinewaves due to the fact that non bandlimited arbitrary waveforms may exhibit overshoot The Output Rate control governs how fast table points are output At 100 the output rate is 1 table point per output sample At 200 the generator skips a table point and outputs every other table point each output sample For fractional output rates the waveform is interpolated So at 50 Output Rate the generator outputs a table point then and interpolated point and then the next table point Bandlimited interpolation is used where memory allows Otherwise linear interpolation is used A maximum table length of 128 kpoints is allowed subject to other waveforms or instrument features using the DSP memory The arbitrary waveform gene
108. 1 kHz LPF Without Unwrapping 2014 Stanford Research Systems 152 srt Operation Manual 100 200 500 ik 2k He Same Spectrum with Phase Unw rapping On The FFT2 analyzer uses the coherence as a threshold value for calculating phase Coherence is a measure of the phase stability between the input and output channels thus regions of high coherence will yield stable phase measurements and regions of low coherence will likely yield noisy phase values Setting a threshold value instructs the FFT2 analyzer to only calculate phase for frequency bins where the coherence exceeds the threshold value Setting a small threshold can often clean up the phase response spectrum considerably DC Correction OL Correction Average Average Small amounts of DC in the FFT time record can be removed using the DC Correction control Selecting Average will subtract the average value of each time record from the time record before taking the FFT 1 2 Pk Pk will subtract the average of the maximum and minimum values found in each time record dEYrms oo ss ccncecisiedesiyiaibacss i arreen No DE e TRONS TT i ee re aarre Offset 75 destin PED EAA EEES Correction eG AA E e e a P pies SO EE E PSE E EAT Na agate E EE SS h Average a PNN eres ee OC Corecton 5 10 0 50 100 00 s00 ik zk Hz Effect of DC Correction on Spectrum An example of the us of DC correction is shown above The original spectrum has some DC offset which h
109. 1 to a typical PC serial port usually configured as a DTE use a straight through serial cable The Serial Port can also be used to connect SR1 to a network of SR10 11 12 switching modules See the section on configuring the switchers for more details D Keyboard and Mouse Port These connectors accept a standard PS 2 style mouse and keyboard For the unit to recognize the external mouse and keyboard they must be plugged in before SR1 is powered on E External Video Connector The external video connector is a VGA style connector which allows connection of an external monitor to SR1 Note that the external video resolution is the same as the resolution used on the SR1 s main LCD panel 2014 Stanford Research Systems 294 SR1 Operation Manual F Ethernet Connector The standard RJ 11 ethernet connector allows the SR1 to be connected to an ethernet network Configuration of network parameters IP address DNS servers etc is done using the Network Setup option of the Tools menu G Analog Signal Monitor Out Chs A and B The analog monitors output an amplified version of the input signals found at Analog Inputs A and B The scaling of these signals is adjustable using the Monitor Panel The analog monitor signals can also be routed to the speaker or headphone output H Analyzer Monitor Out AO and A1 Certain analyzers have a realtime output signal For instance the Time Domain Detector output signal consists of the
110. 2014 Stanford Research Systems 146 SR1 Operation Manual 20 50 100 200 s00 ik 2k Sk 10k 20k Anechoic Frequency Response Energy Time Curve The energy time curve ETC is an attempt to find an envelope function for the impulse response that attempts to illuminate features of the impulse response that may be obscured by interference effects For example the synthesized impulse response shown on the left below consists of a high amplitude sinewave with a fast decay time constant mixed with a delayed lower amplitude sign with a slower decay time constant a Es 5E Left Im pulse response consisting of a high amplitude sinusoid with a fast decay time constant and a delayed lower amplitude sine with a slower decay time constant Right ETC of the impulse response From Andrew Duncan The Analytic Impulse Presented at the 81st AES Convention November 1986 While this may not be immediately obvious from the impulse response graph it is more evident in the computed ETC The ETC is calculated by finding an appropriate imaginary part for the impulse response in the same way that sine is the appropriate imaginary part for a uniform phasor whose real part is cosine Technically the imaginary part is found by computing the Hilbert transform of the impulse response The complex function whose real part is the impulse response and whose imaginary part is its Hilbert transform is known as the analytic impulse response of the system
111. 20k 50k 100k Hz 500 ik 2k Sk 10k 20k 50k 100k Hz As an example the 2 plots above show the single shot magnitude and phase of the frequency response of an 8 pole 6 zero elliptical filter with a 5 kHz pass band edge The phase distortion departure from linear phase characteristic of elliptical filters near the passband edge is apparent Using the Graph Calculator s group delay function we can directly calculate the group delay from the phase curve 2014 Stanford Research Systems 156 SR1 Operation Manual sec 575 550p 525p 500p 475u 450p 425 400 3754 350p 3254 300 275 250 225u 200 1754 150p 1254 100 75y 500 1 0k 1 5k 2 0k 2 5K 3 0k 3 5k 4 0k 4 5k 5 0k Hz Passband Group Delay of Elliptical Filter 2014 Stanford Research Systems SR1 Operation 157 2 4 5 THD Analyzer The THD Total Harmonic Distortion analyzer uses FFT techniques to measure the total or relative amplitude in two groups of user specified group of harmonics Unlike the Time Domain Detector which uses time domain techniques to integrate the total noise harmonic power outside the fundamental the THD analyzer uses the FFT internally to selectively measure only the amplitudes of harmonics The THD Analyzer Panel A1 THD Analyzer SOUrCE Converter Fs Measurement References Trigger Fundamental Speed Synch avg Clear f Tuned Fixed very Fast E _ Weighting Measured Freg fone ad Measurement 1 Measuremen
112. 4 section 4 2 2 2 Passband ripple is lt 2 dB Stopband attenuation is gt 60 dB and the ratio of passband edge to stopband edge is 1 2 Selects a 4 pole elliptic high pass filter with a passband edge of 40 kHz that satisfies the requirements of AES17 1998 r2004 section 4 2 2 2 Passband ripple is lt 2 dB stopband attenuation is gt 60 dB and the ratio of passband edge to stopband edge is 1 2 AES 80 kHz Selects a 4 pole elliptic high pass filter with a passband edge of 80 kHz that satisfies the requirements of AES17 1998 r2004 section 4 2 2 2 Passband ripple is lt 2 dB stopband attenuation is gt 60 dB and the ratio of passband edge to stopband edge is 1 2 For high frequency low pass bandwidth limiting the following filters can be selected BW Limit Low Pass selection No Low pass filters are applied Variable Low Variable Cutoff 4th order Butterworth Filter Pass Selects a 4 pole elliptic low pass filter with a passband edge of 20 kHz that satisfies the requirements of AES17 1998 r2004 section 4 2 1 1 Passband ripple is lt 1 dB Stopband attenuation is gt 60 dB for f gt 24 KHz AES 40 kHz Selects a 4 pole elliptic low pass filter with a passband edge of 40 kHz that 2014 Stanford Research Systems SR1 Operation 125 satisfies the requirements of AES17 1998 r2004 section 4 2 1 1 Passband ripple is lt 1 dB Stopband attenuation is gt 60 dB for f 48 KHz AES 80 kHz Selects a 4
113. 4 1 5 Notch Filter for Digital and Analog HiRes Inputs For HiRes analog and digital audio inputs the bandpass filters with 1 3 1 6 1 12 and 1 24 Octave responses can be selected 2014 Stanford Research Systems 122 srt Operation Manual Response dB 40 10 10 10 Normalized Freq Software Bandpass Filters 1 3 1 6 1 12 1 24 Octave Digital and Analog HiRes Inputs for HiBw analog inputs a single analog bandpass filter may be selected 2014 Stanford Research Systems SR1 Operation 123 Hardware BP Filter for Analog HiBW Inputs Notch Filter Tuning Motch BF Filter C Fixed Mone ka f Tuned The frequency of the tunable notch bandpass filter can be set to a fixed frequency or tuned to one of several sources To select a fixed notch bandpass frequency select the fixed radio button at the right of the Notch BP panel The fixed frequency is entered in the control which appears to the left of the radio buttons To select a tuned notch bandpass filter press the tuned button and select the tuning source from the drop down control When the tuning source is set to Measured Freq the notch bandpass filter tracks the measured frequency of the selected input When set to one of the generator channels the filter frequency follows the frequency of the selected channel When the generator is in mono mode choose the A channel if there are more than one waveform on the selected channel the first
114. 5 File Format 306 Input Gain 183 Input Selection 183 Measurements 188 Overview 9 Trigger 184 Digitizer Display 2014 Stanford Research Systems Index 313 Digitizer Display Eye Diagrams 209 5 Eye Limits 209 Probability Tab 208 Spectrum 207 Speedbar 205 Time Record Tab 207 FFT 127 141 FFT Dual Ch 141 FFT Analyzer 1 Ch 127 FFT Chirp 48 62 DIM 62 FFT1 Analyzer DIM TIM 161 267 Aliasing 131 Displays Averaging 132 Bandwidth 131 Block Diagram 127 DC Correction 135 Phase 134 Resolution 132 Weighting 136 Bar Chart 202 Digitizer Display 205 Graphs 192 Overview 9 Displays Menu 191 Distortion 265 Distortion Spectra 139 FFT2 141 Dual Connector 79 FFT2 Analyzer Aliasing 149 Averaging 149 Bandwidth 148 Edit Block Diagram 141 Copy 34 Coherence 143 Cut 34 DC Correction 152 Delete 34 Resolution 149 Paste 34 Weighting 152 Edit Menu 34 Windows 150 Edit Tones 104 FFT2 Analzyer Ejecting a USB Drive 237 Phase 151 Energy Time Cure 144 File Menu 28 EQ 36 57 File Types 306 EQ Files 306 Filtered Noise 43 62 Ethernet 218 241 Frame Synch Output 293 Events 220 Frame Synch Invert 79 and Scripting 223 Free run 91 Configuration 225 Frequency Response 263 Definitions 220 Frequency Units 41 60 Example Measurement 19 Front Panel 13 External Monitor Setup 310 Front Panel Lockout 216 External Reference Input 293 Front Panel Reference 291 External Sweeps 94 Fuse 6 External Trigger Input 293 External Video C
115. A Aa AAA RAAEN 242 SCD UES WIN AOW Serera Ea a 243 Leao POG E E a aa 252 8 Quick Measurement Men siniiersi aas ene aa ea ee eee ee 253 Set Pane a 254 SNR Panel saioei A aAA a aE E EA AAAA 255 FRE FETS GO Pah lonriiaaaa i a a aE dean nuabeaeemenamwaanes 257 Level Pane l a a A a a a A Aa A A A AAA ATEAN 259 THOEN PINE T orr a A Gecsuaboutacntuenauceuneges 261 FFEQUeNCY Response Pane binisikan aa a A aa Ea aAa aE 263 DiStorion Pane acsi i A 265 IMD Fanell soinaren aaae AAA exceed CAE E AA AAA ON ANAK O AAA E ANAE E EA TAA ANA EONA EENE 267 Crosstalk Pant lunnir a a a RE 269 interch nnel Phase Panel siiin nades aa aa a a A aa A aa AKAA ALARA NEATA 271 MOU PNIS E Triin A E A tees be seinn en euabeaneeuaaamwaanes 273 Automated Measurements and Reports ssssassssessssssnunsnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnn nennu nn nnnn nnne n nnmnnn 275 9 SEUS MENU ri a a a a a a a bed a cel tie E 278 PNGIOG ANAlOG ire a E aae EEEE A EKRE a 279 Anal g Digtal oeiia E EEEE E E Ea A E E AA A 281 DIG ta FANO Gie a aa a a a A a 283 Oke t key Bs o gt zrenie a e a E A A a a E 285 DiOHALIO adarr o E a e E ede ee 287 10 HEID MENU cossiensa EE EE OEE 288 Part Ill SR1 Reference 291 1 Front Panel DESCH DUOM wos csueuewrur tween e anu as aa ia ue mub Meno aa E 291 2014 Stanford Research Systems 4 SR1 Operation Manual 2 Rear Panel DESErlpl On aerisiiasiceasctovssstndecscsansigsueosionansuivasavensdncascanciacbensentan
116. Channel Status Panel Professional Mode The channel status panel is divided into 3 main areas On the upper left are the controls governing the transmitted status information On the upper right are the displays indicating the received status information for channels A and B The bottom of the panel contains the hexadecimal representation of each of the 23 transmitted and received status bytes for both channel A and B Transmitted Status Bits The top 3 radio buttons determine whether the remainder of the transmit status controls refer to channel A B or both channels The next control down determines whether the status bytes are transmitted in professional or consumer standards The panel redraws itself as shown above according to the selected standard Changes made to the transmit status controls are immediately reflected in the hexadecimal status display along the bottom of the panel However changes made to the hexadecimal transmit panel are not reflected back in the status controls The status controls and the hexadecimal status panel remain out of sync until the next time a status control is changed when transmitted status reverts to the state reflected by the controls Professional Transmit Status Bits For professional mode only bytes 14 to 17 specify the local sample address code and bytes 18 21 specify the time of day sample address code These 4 byte counter values can be specified and additionally if the increment checkbox to the rig
117. Create Default Tones is pressed Configuration The Load and Save buttons call up file dialogs allowing the user to save and recall only the configuration information contained in the multitone configuration panel including any manually edited tone parameters Active Tones The two indicators display the number of active tones the the A and B channel Usually this will be the same number of tones in the Desired of Tones control The Edit Tones button displays the Edit Tones panel The Edit Tones Panel E Edit Multitone Tones Frequency Phase Amplitude ro sooo ooo fio0 00 jv 1 sooo foo fioo 00 aliro faze 10 00 alezo oa s75 100 00 alpo feza t00 00 5 f11 250 kHz 69 300 100 00 y iw 6 13 500 kHz 39 615 100 00 Iv 7 f15 500 kHz 115 53 100 00 g iw a i7 750 kHz 17 1 46 100 00 a 20 000 kHz li70 25 100 00 3 l ie Channel C E Recalc Phase The Edit Tones panel allows modification of individual tone amplitudes frequencies and phase The Edit Tones panel is also the only method by which tones can be placed in different positions on the A and B channels Frequencies entered will be rounded to the nearest bin frequency Amplitudes are all relative the overall amplitude of the signal is set in the generator waveform panel Each tone may be turned on and off individually with the checkbox at the right hand side of the panel Pressing So
118. Domain Detector CCIR unwtd Filter defined in CCIR Rec 468 4 Annex ll for making unweighted noise measurments CCIR 2 kHz This filter is identical to the CCIR wtd filter except the normalization is changed so that the filter has unity gain at 2 kHz instead of 1 kHz Note that the spectral weighting selected with this option is done in the the DSP after FFT computation the filtering does not affect the dynamic range of the measurement Checking the invert box applies the inverse of the specified weighting to the FFT measurements Time Display Interpolation Near the top of the FFT frequency range of DC to Fs 2 there are relative few sampled data points per 2014 Stanford Research Systems SR1 Operation 137 cycle in the FFT time record If the points are plotted by simply connecting adjacent samples a distorted picture of the actual time data may result When Time Display Interpolation is turned on band limited interpolation is applied to the time record display to oversample the displayed points and produce a more accurate visual representation of the original time signal For instance the graph below on the left shows the time record of a 19 57 kHz sinewave input to the FFT1 with an analyzer sample rate of 64 kHz and with Time Display Interpolation turned off Turning on Time Display Interpolation shown below on the right gives a much more accurate representation of the original signal Vins 15 7m 15 8m 15 9m 16 0mm
119. Duty Cycle 25 Lo Amplitude 10 EQ Controls EQ DeEmphesisl 0 Certain waveforms can have their amplitudes scaled as a function of frequency according to the information contained in an EQ File EQ files are XML files which specifiy a relative frequency response as a function of frequency by either interpolating a table of frequency response pairs or by calculation from a set of pole and zero locations The structure of EQ Files is detailed in the File Reference section Use the a button to open a file dialog to specify the EQ File Check the Invert EQ box to have the amplitude scaled by the inverse of the EQ file response Waveforms that are capable of being used with EQ files will have an EQ checkbox in their waveform tab This box must be checked for EQ to be active regardless of whether an EQ file is selected in the file selection control When an EQ file is selected and the checkbox is checked the amplitude control in the waveform tab will continue to show the constant user selected waveform amplitude however the Total Channel Amplitude control will display the amplitude with the EQ response included Invert Digital Generator Units The amplitude and frequency of generator waveforms can be specified using a variety of units all of which are useful in different audio test scenarios Because of the large number of waveforms that SR1 can generate and because it s useful to define amplitude in a way that simplifies the coupli
120. EEE A ARE CEAO E EAE E EEEE 202 DIONIZ Er DIS lay aa E aie a E O a a 205 Other Display Menu Options siinirsnoissavai aaa a ae ave eee ne eee 212 6 TOOLS MENU arima ates cient ete a ce atcha enews ernest ects ean ew eae ace aeisae 213 Preferences Pane Wl iccceccce ccc creda AEEA EE KENA Ea a ELATAS Aa A AAAA EEEE TEE EAA AEA ENT EKRE 214 Events Pane ecis E a a a ae ea aa R 220 SWitcher Configuration Panelo aisa eaa r a a a a a a a aariaid 226 Hardware StatUS iinei a e aeaa a e a Eaa R 230 Ne tWOTK INO cae a E eee AEA vipeed vey ntane ah cocectcuneteeuRutevesues raamecunente 231 NOW OK OOU Dein eaa ee a a ele E AA E A A ET 231 INE TWOP O ACC Sy EEEE E E EEEE E EEA NEOON S EE EEEN EREE OTE EE EEES 232 Mab ING VON DIV Eseri e o EE EE EE E EE EAE E he ea ia 233 Remove Netw OF KDFIV Ge oiats enn cevicaxGuhtesehecidwcied sion tnancneabecwsceatanvadene loahS puscslvantiewsteateee nirestoucdestsbesnaveneteanueedbanendoes 233 SARC SES I scareeciche bee aaa a a lives ee tnt cba ask int a eerie drink a ents eee 234 COM BULGER FUNCIONS soria aaa sos sbeuea aE A a E a a a aa 236 Primers Panel cadre chee casa e E ee a A E A A 236 Kec rO Ene r aoa a eee eee ene 237 POWORODIONS raei a a E A eT eee ee 237 Besoni A EA S EE A EEE ESE EEE NEOA S EE VEENRE BOR ISEE 238 NAPE IS OY OAR as E SEE N A ENEA 238 Updating SRIren a a a sl cane han ahve Mh 238 T Aoma Uon Me Moineau anra a a E OS 241 Remote Interface Pane bioure anaa O aAa EAA A eaaa AAAA AE
121. EOE e a S EEEE ae aE ee a a anne ne NIEA E a re a SEENE J PAAR TANE E E E T E ERE E A E E T E E E E A EE Seas cence peace nena ea nenaeenene seca eenecenenemeneneeenenereneneeanenatensnaeanenaransnassneneneangan 2OID YY penne a oem e e a eee a st efector 4004 E E E ET E A A SOE 2nd Refecti fai te E EEEE EE E TE 600p F PECE a atest EES e e a Seer ae ee i OES EOE ET E 5004 ee T WEEE E EES EE EEE O E E A E E ETE E E E EE T E E E E E PTEE E EE E E E 2m 4m m am 10m 12m 14m 16m 18m ia Loudspeaker Impulse Response with Reflections In addition to the response due to the sound directly transmitted from the loudspeaker to the microphone the response includes several reflections from nearby walls The calculated frequency response including the direct sound and reflections is shown below Interference between the direct paths and the reflected paths cause oscillations in the frequency response and obscure the true frequency response of the loudspeaker S00 ik Frequency Response Inluding Reflections In order to avoid this problem the FFT2 analyzer offers an Anechoic Frequency Response measurement The anechoic frequency response is calculated by transforming only the direct sound portion of the impulse response back to the frequency domain The impulse response is suitably windowed to avoid problems with abrupt transitions When this is done for the case above the true frequency response emerges much more clearly
122. It will in general agree with the Reading calculated by the Time Domain jitter analyzer The sum is performed after all filtering and weighting is applied Filtering Filters Hi Pass M On Lopass weighting 20 000 kHz 2 0000 kHz Adjustable highpass lowpass and weighting filters can be applied to the jitter spectra and the RMS sum calculation These filters are applied by the DSP after the jitter FFT has been computed The highpass and lowpass filters are 4th order Butterworth filters Each filter is enabled or disabled by the corresponding checkbox The weighting filter choices are same as for the Time Domain Detector See the discussion there for the types and intended applications of each of the weighting filters Averaging Averaging vgs Clear Avg Done Exponential p E Mone Linear Exponential 2014 Stanford Research Systems 182 SR1 Operation Manual Exponential or continuous averaging continuously updates the averaged FFT displays weighting recent spectra more heavily than older spectra The number of averages in this case is only an approximation of how many spectra are included in the average Linear averaging averages the specified number of spectra all equally weighted and then stops None means that no averaging is applied In this case the magnitude of the Linear Spectrum and Power Spectrum are the same The Clear button clears the Current average buffer and restarts the averaging proc
123. Preferences panel Zooms the active display in and out in the Xand Y direction Toggles the sense of mouse zooming gt Shows eye diagram intensity values as the mouse passes over a point Toggles the graph area between the standard display and a larger graph which covers the scaling controls and trace listing Toggles the cursors on and off ie Moves the active cursor to the maximum or minimum displayed point on the active mores trace Toggles Autoscale on Acquire When unlocked the display autoscales each time a new digitizer record is acquired When locked the current scaling is maintained Toggles online offline status When the door is closed the graph is online and new E digitizer measurements will be updated to the display When the door is open the graph is offline and the display data will not be updated when new digitizer records are taken Add Annotation Opens a text window and allows placement of text on the graph area Es Change Graph Title 2014 Stanford Research Systems SR1 Operation 207 Time Record Tab Eve Diagram Probability Spectrum Time Rec a E E S S S S N E E E E S 45H 46H 47H 48H min max K 44 173 Hsec 55 651 Hsec iT 3 3825 Y 3 3825 The Time Record tab of the Digitizer Display shows the two amplitude vs time measurements of the digitizer input amplitude vs time and jitter amplitude vs time The input amplitude vs time trace shows the raw digitizer data i e ess
124. Print Screen 32 Print Setup 33 Printers 236 Q Quick Measurements 253 Automated Measurements 2 75 Crosstalk 269 Distortion 265 IMD 267 In Out Phase 273 Level 259 316 SR1 Operation Manual v2 Quick Measurements 253 Reference 25 7 Reports 2 75 Setup 254 SNR 255 THD N 261 Quick Measurments Interchannel Phase 271 Quick Stat 19 R Rackmount Setup 310 Ramp 483 62 References 40 Remote Interface Panel 242 Remote Interfaces Overvew 9 Remote Interfacing 241 Remove Network Drive 233 Risetime 79 RS 232 218 241 RS 232 Connector 293 eit Safety 6 Save Configuration 29 Save Displays 212 Save Partial Configuration 30 Scripting Window 243 Speedbar 243 SR1 Basic Interface 244 Scripts 250 Serial Port 218 Service 6 Settling 98 Settling Options Delay 99 Precision 98 Profile 98 Threshold 99 Sharing SR1 234 Signal to Noise 255 Sine 62 Sine Low Distortion 43 Sine Normal 43 SMPTE DIN 62 161 267 SNR 255 Software Updates 238 Speaker 100 Specifications 296 Square Wave 43 62 SR1 Basic 244 Actions 246 Properties 245 SR10 226 SR11 226 SR12 226 SRS Contact Information 6 Startup Configuration 214 Status Bits 79 Sweep 91 Sweep Append 197 Sweep Axes 91 Sweep Data 97 Sweep Settling 98 Sweep Source 92 Sweeps and Free Run Overview 9 Switcher Configuration Adding a Switcher 226 Switcher Sweeps 96 Switchers 226 Synchronous Burst Sine 43 T Table Sweeps 93 TCP IP 218
125. R1 audio analyzer in an EIA standard 487 mm 19 rack cabinet Two brackets two handles two washers and assorted screws are supplied with this kit When installed the instrument will occupy 5 U or 5 rack units increments 8 75 in 222 mm To install the rack mount kit you will need a 2 Phillips screw driver Remember that SR1 weighs about 50 Ibs so use caution 2014 Stanford Research Systems SR1 Reference 311 when installing it into a rack See the diagram below for details on installing the rack mount brackets 1 Optional Attach the handles to the rack mount brackets with two 8 32 x 3 8 flat head screws for each bracket 2 Optional Remove the feet and bail from the bottom of the instrument Turn the instrument onto its left side and remove the 6 screws that attach the feet to the bottom of the instrument Rotate the instrument back onto its bottom 3 Remove the front two 10 32 x 3 8 truss head screws on both the left and right side of the instrument 4 Install one bracket on the right side of the unit using two 10 32 x 1 2 truss head screws Install the other bracket on the left side of the unit with the washers between the bracket and SR1 using two 10 32 x 1 2 truss head screws 10 32 x If2 TRUSS HEAD 4 PLACES 8 32 x 3 8 FLAT HEAD 4 PLACES n e a WUR 10 WASHER 2 PLACES SRS P N 7 021031 2 PLACES SRS PZN 0 00506 2 PLACES 2014 Stanford Research Systems 312 SR1 Oper
126. Research Systems Getting Started 19 1 5 A Quick Example In this example we ll use the analog generator to create a signal with a known amount of distortion and we ll use the Time Domain Detector to measure the distortion and the FFT analyzer to look at the distortion in the frequency domain To begin turn on SR1 and wait for the instrument to finish its startup sequence No external cables will be necessary for this demo Feel free to use the front panel trackpad and keypad keys to control the instrument or if you ve got an external mouse and keyboard you can those We won t go into excruciating detail on how to select items and use the controls because they re standard Windows controls and their operation should be familiar Start out by using the speedbar at the top of the SR1 screen to open the Analog Inputs panel and the Analog Generator panel and from the Analyzers menu set Analyzer 0 to Time Domain Detector These three panels may automatically open when SR1 is powered up Analog Analyzer 0 Inputs ar E CEA aa F AE EE enerator Note that the icons for the Analog Inputs panels and the Analog Generator panel are both orange SR1 uses orange to denote items associated with the analog domain and blue to denote items associated with the digital audio domain Most panels have color bars under the title of the panel indicating the domain associated with the panel SR1 Speedbar The three panels you ve just opened should look li
127. T computation the filtering does not affect the dynamic range of the measurement Time Display Interpolation Near the top of the FFT frequency range of DC to Fs 2 there are relative few sampled data points per cycle in the FFT time record If the points are plotted by simply connecting adjacent samples a distorted picture of the actual time data may result When Time Display Interpolation is turned on band limited interpolation is applied to the time record display to oversample the displayed points and produce a more accurate visual representation of the original time signal For instance the first graph below shows the time record of a 19 57 kHz sinewave with a FFT analyzer sample rate of 64 kHz with Time Display Interpolation turned off Turning on Time Display Interpolation shown in the second graph gives a much more accurate representation of the original signal rms 15 7mm 15 8mm 15 9mm 16 0mm 16 1mm 16 2mm ieam Time Record without Time Domain Interpolation 2014 Stanford Research Systems 154 SR1 Operation Manual 15 7m 15 8mm 15 9m 16 0mm 16 1m 16 2m 16 3m 72 Same Signal with Time Domain Interpolation The FFT2 Impulse Panel Measurement Meas2 Impulse References Trigger 4nechoic Frequency Response Start Stop Data Points Time 2 0000 msec 1 2980 msec Window 5 ki 5 T Energy Time CUrvYE Window HalF Hann The Impulse tab on the FFT2 analyzer panel contains controls go
128. THD HY Rato SINAD Ratio Crest Factor Measurement selects the quantity displayed on the TDD panel Amplitude The selected channel amplitude is displayed All filtering options are available The ratio of the selected channel amplitude to the level of the other channel is displayed This mode can be used for instance to make crosstalk measurements THD N Amplitude The selected channel amplitude is displayed The notch filter is always selected to filter the fundamental The amplitude signal then represents the total noise and distortion present in the input signal THD N Ratio The ratio of the amplitude to the level of the selected channel is displayed The notch filter is always selected This selection displays THD N amplitude relative to the total amplitude of the signal Crest Factor The ration of the peak amplitude of the selected channel to the RMS amplitude of the same channel is displayed Time Domain Detector Analog Hi Bandwidth Inputs To fully exploit the capabilites of SR1 s analyzers i is useful to understand exactly how SR1 functions for each of the three classes of inputs analog signals using the Hi Bandwidth converter HiBw analog signals using the Hi Resolution converter HiRes and digital audio signals Below is a block diagram of the TDD for analog signals when the HiBw converter is selected Analyzer Out To Other Analyzer Amplitude Chain HW Notch EP Optional HW Filter Filters Attenuator G
129. Test Output Impedance selects the Analog Generator output impedance 2014 Stanford Research Systems SR1 Operation 39 259 75Q 6000 Allowed impedance values for unbalanced outputs 500 1500 600Q Allowed impedance values for balanced outputs Waveform Controls The New button displays the Waveform Selection Submenu Sine Normal Sine Noise Low Distortion Sine US 457 Noise Phased Sines Synchronous Burst Sine Ramp Arb FFT Chirp MultiTone IMC Polarity Const Offset The selected waveform will either be added to the output for one or both channels depending on the Mode setting Certain waveforms for instance Low Distortion Sine cannot be combined with other waveforms When one of these waveforms is selected all of the current waveforms are deleted Other waveforms simply add to the current output when selected The Delete button deletes the currently selected waveform The A B selection buttons only appear when the generator Mode is set to Stereo The buttons determine which channel a newly added waveform will appear on When the mode is changed to stereo any waveforms present will be assigned to channel A When the mode is changed from stereo to mono all channel B waveforms are deleted and the channel A waveforms are output on both channels Amplitude Controls hA n Eh BB 100 0 Sty aje Lock 100 0 6 0000 ins N Inve Inve a m The Channel Gain control varies modifies the total ou
130. The input is connected directly to the corresponding channel of the Analog Generator Digital Audio This special setting allows the analog input to look at the common mode signal Common Mode present at the balanced digital audio connector The common mode signal can then be measured or displayed This can be useful in diagnosing noise problems with the digital audio input 2014 Stanford Research Systems SR1 Operation Input Configuration Coupling DC coupling of the analog inputs AC coupling of the analog inputs with 1 7 Hz corner frequency To the left of the controls for each channel is the visual level indicator for that channel The blue led glows when the input level is below approximately half scale The green led glows when the input level is between half scale and full scale and the red led glows when the input exceeds full scale If the red led glows continuously then the signal either exceeds the input range of the instrument or in the case of no autoranging the range needs to be manually adjusted Note that the input level indicators are also displayed at the bottom of SR1 s main window so that they are visible even when the Analog Inputs Panel is not displayed ChA ChB Hil LiL norm Analog Level Indicators When a potentially damaging high voltage is detected at the input SR1 automatically switches the input range to the largest possible value 160 Vrms and removes any termination resistors selected
131. Use the Generator Trig Source _ os source 2014 Stanford Research Systems no SR1 Operation Manual 2 4 1 1 control to select which generator will be the source of triggers Only certain waveforms generate triggers See the Analog and Digital Generators section for more information about generator triggers Digital Audio Block Selects the digital audio carrier Z Preamble the beginning of a block as the trigger source This option is only allowed when the analyzer source is set to digital audio External Burst A B Selects the burst as the trigger source The Input Channel and Other Channel triggers are level triggers i e they trigger when the selected Signal passes through a level with a specific polarity chosen with the Level and Polarity controls All the other trigger sources are binary they either occur or not Trigger Indicators aki tlt The two trigger indicators on the speedbar at the top of the SR1 screen flash green when the corresponding analyzer is triggered When an analyzer s trigger is enabled and the corresponding analyzer is waiting for a trigger the indicator is drawn with a yellow background Analyzer Units The relation between various amplitude units in SR1 is always fixed at the same ratio that applies to a sinewave This can cause some confusion due to the fact that in normal usage an amplitude of 1 Vrms implies not just a magnitude and unit of measurement but a method
132. aaa e O a Aa E aA aaan 79 Channel Status Pane lonssso n a aaa a E a aa 87 USEF Status Panel E E E E E E E EE E EE E E a E E A E E E E E E ies 90 SWED Pan Ol orse a a a aa 91 Setting Panel aioa eE a a eE Aa E AAE aaa 98 Monitors Panelas a a EE a a aE aa at 100 Multitone PANG sioria aiae le aa aE a aKa eKA aaO la he EACK a a AAE AKAA EESE CA AERAR ESNS 102 Clock Reference Pane lnice rE 105 Ae Analyzers Me NMU na a a E E A E E AR 107 Common Analyzer Featul s ccrich A a E a EALE Aae SAO ASE A 108 AnaNzer Unless Dee ce E a a E E E O 110 TMe Doman Detector erraia eae a EE a 115 FET SINGIS Chane rsrsrs a a cece dtavedaeaductandeis 127 FFE DU ar carn cscs osiscee nctsea tswwcc a dan tee te cau anenatlendienhrmuneneaunsaaeueaa sal trmctearebtiamecenuausire ccna 141 TAD SAA ANY 2 ON n a O a AE 157 IMD ANaIVZ OF aeaa aaaea e vi act cuceneutta a cute ssenaeumontuateeseaenseetetis N aa 161 NIUITIEONG ANALY ZOD dagien n a Eae aia Ean aa aE Ka TER 165 FISTOGK all ANalyZE T se N E E Ea 171 OGtAVE ANGI ZO T na a a aa a E 175 RECO FAM ALY ZC T aeaa aa a a E E i 177 DIGHIZEr dokira oa EEE E sosbiveveddnteveteceaetnelsauesnes 183 2014 Stanford Research Systems Contents D DISDIAYS MeO MU isi venice dasscivinsksncesacscaresiwiesdocasssiciaibiagianaseaniedaciescusacsandasdi asoezoeuemialaesseecacuacede 191 GRAD aoia a Ri a aa E a araa da aa aaa A Ta fae dea wec viata AEEA 192 Mace Call CU ALON eenaa A E a a S 199 Bar VA aan A O E E E
133. above 5 Hz AES Reference Controls The Terminate checkbox terminates the rear panel AES Ref In XLR connector with a110Q resistor Reference In Digital Input Delay displays the measured time interval between the z preambles of the AES reference input rear panel AES Ref In and the front panel digital audio input The AES Reference Status indicators indicate the current status of the AES Reference input Status Condition CRC For AES EBU digital audio streams an 8 bit CRC code is generated from the received channel status bits in each block and compared to the value stored in the last byte of the channel status block Red indicates a CRC mismatch Lock Green indicates that the digital audio receiver is locked to an input data stream Red indicates no lock Red when either the Unlock or Bi phase error is detected Coding Bi phase error A valid digital audio signal remains at the same amplitude for no greater than two Unit Intervals Uls except during preambles A red indicator indicates that this condition has been wolated Parity Each digital audio subframe contains a parity bit which summarizes the parity of the remainder of the bits in the subframe Red indicates that the received parity bit does not match the parity computed in the remainder of the subframe Validity Red indicates that the validity bit has been set in one of the received sub frames Reference Out gt Main Output delay sets the delay between the AES Refere
134. ain ka _ iBw cha ae i Attenu ater A Gain k OP Che oe Lev el Chain Detailed TDD Block Diagram HiBW Analog Inputs Shown for ChA Selected Hibw ADG k EW Limit Weighting Filtering Filters RMS SP Amplitude Quasi Pk Heas Mt Analog Processing PostFilter Gain Level Meas ME Z Digital Processing 2014 Stanford Research Systems SR1 Operation 117 Because of the wide range of input amplitudes handled by SR1 160 Vrms full scale to 62 mVrms full scale the input signal may need to be either attenuated or amplified For most signals the autorange control can automatically select the appropriate attenuation or gain without user interaction The level chain starts with a frequency measurement of both channels For analog signals the frequency is measured using hardware The two level signals are then digitized with identical 16 bit converters running at a sampling rate of 512 KHz and the resulting digital signal is sent to a high speed DSP processor The DSP measures the phase between the two signals and does an RMS integration of the signals to calculate the level of each channel The analog signal corresponding to the selected channel A or B is sent to to the amplitude chain First the signal is sent through the tunable hardware Notch BP filter if the notch or bandpass is selected on the analyzer panel Next the signal is can be passed through one of the 4 optional hardware filter
135. al scripts terminate when flow of control reaches the end of the script Since in long complex scripts it may be difficult to provide flow control to the end of the script the script can be terminated at any time using the VBScipt err raise function call err raise errNo int errorSource string errorDescription string The SR1 Basic Interface includes a built in flag SR1 Scripting Terminate which is set true when the user presses the stop fal h button on the scripting panel A subroutine can be included in user programs which checks the flag and ends the program sub checkterm if SR1 Scripting Terminate then call err raise 1 checkterm Manual Program Termination end if end sub By sprinkling calls to checkterm throughout a script the programmer can ensure that the script will always terminate when the stop button on the scripting panel is pressed 2014 Stanford Research Systems 252 srt Operation Manual 2 7 3 Learning Mode SR1 incorporates a learning mode to facilitate script development When learning mode is enabled if a script is not currently open a new script will be created The corresponding scripting commands for all subsequent keypresses menu selections panel entries etc will then be incorporated into the script window This lets you do most of the programming by simply making a measurement normally on the instrument Exit learning mode by selecting Disable Learning Mode on the Aut
136. and headphone using the Monitors Panel Does not control the volume of system sounds generated by the Windows operating system The Windows sound level is controlled separately on the Monitors Panel or by the standard Windows volume control D USB Connectors The two USB connectors may be used to connect a USB drive a USB printer or a USB mouse or keyboard E Main Screen Color LCD screen on which the SR1 software runs SR1 is sold with two screens the standard VGA 640x480 display and the optional XGA 1024x768 screen F Knob Use the knob to modify the value of the screen control which currently has focus When graphs are 2014 Stanford Research Systems 292 SR1 Operation Manual displayed the knob can be used to mowe the graph cursor G Enter Key The Enter key confirms numeric and text entry into controls See the User Interface chapter for more information on how to use the Enter Key H Tab Keys The tab keys are used to move focus from one control to another within a window or to move focus between windows on the screen I Action Keys These keys are associated with a specific action such as autoscaling the displays or starting and stopping sweeps See the User Interface chapter for a detailed descriptions of each key function J Numeric Entry Keys These keys are used to enter numeric data into controls See the User Interface chapter for a detailed descriptions of each key function K Touch Pad
137. anford Research Systems 24 SR1 Operation Manual Note how more detail is resolved inside the notch Go back to the FFT analyzer on page 1 and press the Baseband button to return the analyzer to it s full measurment range We re now ready for the last step in the demonstration Setting Up a Sweep On page 1 click on the zj icon to open the sweep controller panel We re going to sweep the amplitude of our distortion sine wave and record the measured THD N from the Time Domain Detector Click on the Sweep Source button to bring up the sweep source selection menu 225 Sweep Source Selection BEA None Time Internal Alyzr 0 AnigGen ChA Gain ChaA Sine 0 Amp ChaA Sine 0 Freq ChaA Sine 1 Freq ChB Gain ClockRef DigGen DigiO External coe SR1 has 3 main types of sweep sources Time Sweeps where the sweep occurs at fixed time intervals Internal Sweeps where the sweep occurs at fixed values of some internal parameter and External Sweeps where the sweep occurs at fixed values of some externally measured parameter Since we re going to sweeping the amplitude of the sinewave we created in the generator this will be an Internal sweep Open the Internal node on the tree and then open the AnlgGen Analog Generator node to show the associated parameters that can be swept Note that there are two sine amplitudes and frequencies shown Sine O and Sine 1 That s because we created two sinewaves in
138. ange Analog Ch A Hi Trip Analog Ch 6 Hi Trip Tone im scrtipti vbs Event 2 No Tone nf Fed No Event No Tone bi bia No Event bi No Tone Poo No Event i Tone 2 bi E No Event X No Tone Fa No Event bd No Tone bia No Event bi No Tone bi a No Event b No Tone bd E No Event No Tone z f No Event hi fa No Event 7 Mo Event l E Oa igital Inputs Dig Ch 4 Validity Change Dig Ch B Validity Change x Dig Unlock Change Dig BiPhase Change W E i Dig Parity Change igital Channel Status ProConsumer Change Copyright Bit Change Mo Tone Emphasis Change No Tone E E Event Definitions The following table lists the events shown on the Event Panel along with a description when the event occurs Analog Inputs Analog Scale Occurs when the input range changes for the specified analog input change The Change A B change can occur as a result of a manual range change or range change due to 2014 Stanford Research Systems SR1 Operation 221 o autorais Occurs when the analog input High Voltage protection triggers due to a potentially AB damaging input voltage on the specified channel The high voltage protection will automatically clear itself once the high voltage is removed Digital Inputs Validity Change Occurs when the status of the validity bit for the specified channel of the digital amt B audio input E Unlock Unlock Change
139. ange the default color for the n fh trace select the desired trace and click on the color bar to bring up a color dialog and choose a new color Note that the set of default colors is different depending on whether a black or white graph background is chosen Typically brighter colors work well on the dark background and darker trace colors show up better on the white graph background 2014 Stanford Research Systems z6 SR1 Operation Manual Autoscale Options When Autoscale on Span Change is selected graph traces that are linked to live FFT measurements will have their X axis automatically autoscaled to the FFT span whenever the FFT span is changed Normally the autoscale algorithm looks at all the points in a trace and chooses a graph scale to includes all points When Autoscale Ignore Outliers is selected the autoscale algorithm ignores points that it considers outliers This is useful to avoid having a relatively few stray points distort the graph scale Trace Initialization Selecting SR1 Default means that each time a new trace is added to a graph the X and Y axis units will be set to the SR1 default for that type of trace For example an FFT power spectrum trace will always be added with the X axis in Hz and the Y axis in Vrms Selecting Last Used means that SR1 will use the most recent user units for the particular type of trace Screen Size When equipped with the optional XGA display the SR1 s default panel size may be
140. annel results What you ll need to configure The default frequency is 1 kHz This should be adjusted as needed The settling parameters can be adjusted to fit the noise levels of the signals being measured The bandwidth of TDD can be adjusted to suit the measurement requirements THD N vs Frequency Sweep Performs a THD N vs frequency stereo sweep The AO and A1 analyzers are set to Time Domain Detector in order to measure THD N A low distortion 1 Vrms sine waveform is setup with an frequency sweep range of 20 Hz to 20 kHz in 22 logarithmic steps Page 2 displays a graph of the A and B channel results What you ll need to configure The default sine amplitdue is 1 Vrms This should be adjusted as needed 2014 Stanford Research Systems SR1 Operation 2e1 2 9 2 Analog Digital The Analog Digital selection lists Audio Analyzer measurements with an analog stimulus and a digital response such as ADC measurements All the Analog Digital setups default the digital input to the XLR C1 connector a sample rate of 48 kHz and 24 bit resolution In all of these measurements the sampling rate of the analog generator is set to the digital ISR allowing synchronous cross domain measuements What you ll need to configure for all these setups Configure the Digital IO Input panel to match the sampling rate and resolution of your ADC or other digital output equipment FFT This setup configures SR1 for a basic audio band FFT The AO ana
141. as leaked into adjacent low frequency bins because of windowing The red trace shows the same signal with Average DC correction applied Spectrum Weighting Spectrum Weighting None ie Invert The spectral outputs of the FFT2 analyzer can have weighting curves applied to them Weighting curves are represented by EQ files The standard EQ files supplied with SR1 include 2014 Stanford Research Systems SR1 Operation 153 Weighting Filters None No weighting filter is applied A weighing filter is applied A weighting is specified in ANSI standard 81 4 1983 and is typically used for noise and THD N measurements with audio applications C Msg Wt The C Msg weighting filter specified in IEEE Std 743 1995 is intended to be used for noise measurements associated with voice transmission telecommunications CCITT The CCITT weighting filter defined by ITU T Recommendation O 41 is another telecommunication noise weighting filter CCIR wtd Filter defined in CCIR Rec 468 4 for audio noise measurements Designed to be used with the Quasi Peak setting on the Time Domain Detector CCIR unwtd Filter defined in CCIR Rec 468 4 Annex Il for making unweighted noise measurments CCIR 2 kHz This filter is identical to the CCIR wtd filter except the normalization is changed so that the filter has unity gain at 2 kHz instead of 1 kHz Note that the spectral weighting selected with this option is done in the the DSP after FF
142. at specified values of some externally measured parameter and finally switcher sweeps where measurements are made over some set of inputs and outputs switched by external hardware Configuring a sweep also involves selecting the sweep data Unlike free run mode where all measurements are continuously updated in a sweep SR1 only computes the set of measurements selected as sweep data Additionally in a sweep each sweep data measurement needs to settle before the instrument will move to the next sweep point A settled measurement is a measurement whose variability has decreased below a user configured threshold and is considered suitable for final results Each SR1 measurement can have completely different settling parameters all configured on the Settling Panel Note that settling is never applied to measurements when SR1 is running in free run mode The Inner and Outer Sweep Axes SR1 allows both one and two dimensional sweeps with a separate sweep source for each dimension The two sweep dimensions are referred to as the outer and inner sweeps Both the inner and outer sweeps may be swept alone with no sweep source set for the other axes When used together the inner sweep is performed for each value of the outer sweep This allows for instance a distortion sweep to be made over both frequency and amplitude Certain restrictions apply to the type of sweep source that can be used for each sweep dimension Switcher sweeps can only be used on
143. at the instrument starts up in the same state that it was turned off in The Autosave file can also be used to recreate the instrument s state in the event of a crash The autosave configuration file is called autosave xml Phase Select any 360 interval All phase controls and measurements will be shifted into this interval when displayed Generator Signal Initialization This control governs the parameter values for newly selected generator signals for both the analog and digital generators SR1 Default means that all signals will be created with fixed default values for amplitude frequency etc Typically the default value for amplitude is 0 and the frequency is 1 kHz Last Used means that signals will be created with the frequency and amplitude values set to the last used values for that particular signal Last Used Except Amp is similar but guarantees that the signal amplitude is set to 0 on creation Last Used signal off is again similar but creates the signal with the waveform on box unchecked 2014 Stanford Research Systems SR1 Operation 215 Analog Generator Maximum Output This selection limits the maximum analog generator peak voltage to the specified value This can be useful in avoiding accidental over voltages when the generator is connected to sensitive equipment Knob amp Keypad The two checkboxes turn on and off the sounds produced when the knob is turned or a key on the front panel keypad is p
144. ata Name Gain Type 3 Value 7397235900 gt lt EQcurve gt The important lines from the point of view of modifying or synthesizing a new EQ files are lt EQdata Name Style Type 6 Value PZmodel gt This line indicates that the file is an Pole Zero EQ file It must be present exactly as shown in all pole zero EQ files The next two lines in the file lt EQdata Name PolesReal Type DblArray Value 20 6 20 6 107 7 737 9 12200 12200 gt lt EQdata Name PolesImag Type DblArray Value 0 0 0 0 0 0 gt lists the real and the imaginary parts of each of the poles The two lists must have the same size Since 2014 Stanford Research Systems SR1 Reference 307 the poles in the A Weighting transfer function are purely real the imaginary part of each pole in this file is 0 Note that the two repeated poles are simply repeated in the list Note that the poles in the list are given with values in Hz Since this is an s plane transfer function the actual poles should be given in rad sec This is handled by the line lt EQdata Name PZmultiplier Type 3 Value 6 283185307 gt This specifies that all pole and zero values should be multiplied by 2z This converts the Hz values into rad sec values and also moves poles into the left hand plane where stable poles are located The real and imaginary parts of zeros are specified in a similar manner by lt EQdata Name ZerosReal Type DblArray Value 0 0 0 0 gt
145. ated MLS Noise Config MLS waveform k3 Omn iw MLS Amp i O00 Yrms oF Length E Pink w The MLS noise waveform outputs a Maximum Length Sequence with a specified repetition interval and frequency profile MLS waveforms are typically used as a stimulus in impulse response measurements where several special properties of the MLS the autocorrelation of an MLS sequence is a delta function simplify the calculation of the impulse response Because SR1 uses a full 2 channel FFT analyzer to make impulse response measurements it is not necessary to use MLS noise as a stimulus any broadband signal will work Nevertheless SR1 includes the MLS waveform because of its historical association with impulse response measurements Amplitude controls as with all SR1 waveforms the peak value of the noise output The crest factor of the MLS waveform which is essentially a square wave is close to 1 The Length selection controls the length of the MLS sequence Selecting 13 for example chooses an MLS sequence whose repetition interval is 213 1 samples The Pink checkbox specifies that the noise output should be filtered by a 3 dB octave pinking filter While the power contained in a white noise signal is linearly proportional to the measurement bandwidth pink noise will have equal power in equal logarithmic frequency intervals e g the power contained in the 100 Hz to 200 Hz interval will be the same as the power contai
146. ated power in the spectrum between the two cursors as well as the RMS integrated power outside of the two cursors Checking Calc THD Ratio calculates the ratio of power in harmonics of the fundamental to the power in the fundamental The fundamental is marked using the left most cursor The maximum harmonic frequency included in the measurement is marked by the right most cursor All harmonics odd and even up to the maximum frequency are used in calculating the ratio For a more sophisticated THD measurement that allows the choice of specific harmonics use the THD Analyzer Checking Calc SNR calculates the Signal to noise ratio of the signal Signal to noise is defined as the ratio to the peak signal contained within the cursor region to the integrated power in the cursor region excluding the peak signal Limits Tab Scale Cursors FFT Cursors E Limits Compatible Traces ie Upper Limit Lower Limit W Enable Single Limit Mew E Edit Delete Jave OI FFT Power Spectrum 4 Graphs have the capability to check trace data against predefined upper and lower limits To create an upper or lower limit click the appropriate radio button and click on New A dialog appears allowing selection of an existing trace on which to base the limit The selected trace determines which of the graph traces can by limit tested only traces with X and Y axes compatible with the limit trace can be tested For instance if the t
147. atio is selected the analyzer inserts a deep notch filter at the frequency of the fundamental In this case because the notch filter is set to be tuned to the measured frequency it is set at the 1 KHz dominant frequency of our generator signal The Time Domain detector then performs an RMS integration of the remaining signal which is distortion plus noise and displays the ratio of that Signal to the total amplitude of the input Note that the measured THD N ratio is 60 dB 1mV 1V 60 dB To show how much more SR1 brings to a standard distortion measurement select the second analyzer A1 as the FFT1 analyzer When the panel is displayed change the source of the Analyzer to Other Analyzer In this mode the FFT analyzer and the Time Domain Analyzer work as a team the FFT analyzer looking at the notch filtered signal from the Time Domain Detector Take a look at the information displayed on the FFT1 analyzer panel Ek 1 FFT Analyzer Other Analyze Measurement Meas2 References Trigger Bandwidth Resolution Acq Time Baseband 32 000 kHz start 0 0000 Hz 0 SSS SSS SSS Fay2 Averaging Avgs Clear Avg Done Like all analyzers the input source is displayed at the upper left The analyzer sampling rate is displayed at the upper right Because we ve connected the FFT analyzer to the output of the Time Domain Detector 2014 Stanford Research Systems 22 SR1 Operation Manual the sampling rate for
148. ation Manual v2 Index _A Abitrary 49 About SR1 288 AES Reference 105 293 Amplitude Units 41 60 Analog Inputs AutoRange 76 Connector 76 Coupling 76 Overload 76 Overview 9 Range 76 Analyzer Monitors 100 Analyzer Units 110 Analyzers 107 Overview 9 Anechoic frequency response 144 Arbitrary Waveform 43 62 Arbitrary Waveform Files 306 Automated Measurements 275 Automation 241 250 B Bar Chart 202 BNC Connector 79 Burst 36 57 Burst Modes 36 C Cable Sim 79 Carrier Level 79 Carrier Status Tab 79 CCIF DFD 62 161 267 Channel Status Consumer 87 Professional 87 Chirp 43 Clock 105 Coding 79 Coherence 141 143 Colors Analog Colors 35 Digital Colors 35 Common Analyzer Features 108 Common Mode Sine 79 Computer Functions 236 Confidence 79 Configuration Files 28 29 30 31 Constant Offset 43 Controls Unit Display 13 Unit Entry 13 Converter 108 Copyright and Trademarks 8 CRC 79 CRC Transmission 87 Crosstalk 115 269 Cursors 197 D Databits Display 79 Date Time Settings 238 Debugging Interface Problems 242 DeEmphasis 79 Digital Audio Carrier Measurements 183 Digital Audio Inputs and Outputs Overview 9 Digital Carrier Amplitude 82 Digital Input Resolution 80 Digital Output Connectors 79 Digital Output Resolution 79 Digital Test Waveforms Digital Constant 62 Digital Count 62 Digital Staircase 62 J Test 62 Rotate Bits 62 Digitizer 183 205 Acquisition 183 Analysis 18
149. ation panel the user needs to select the Multi Tone waveform on the analog or digital audio generator in order to begin generation of the multitone stimulus signal Unlike many other audio analyzers SR1 does not require the user to employ a offline program to calculate a multitone stimulus signal and then load it into an arbitrary waveform generator Once the multitone waveform is running on SR1 any changes to the waveform made on the Multitone configuration panel are immediately reflected in the generator output The Multitone Configuration Panel Multitone Configuration aag Generator Output Mode Mono sd Domain Analog ki Signal Length z045 r Fs 512 kHz Repeat Count F l Sample Rate Headroom Tone Generation Desired of Tones 10 Start Freq 20 000 Hz Freg Dist Linear End Freg 20 000 kHz Phase Dist zvYgmund sd Create Defaulk Tones A ho B ho Edit Tones Active Tones Configuration Load Save Generator The Domain control selects whether the Analog or Digital generator will be used For the Analog Generator a choice of fixed sampling rates 512 kHz 128 kHz and 64 kHz as well as variable rates digital audio output sampling OSR and input sampling rate ISR can be selected using the Fs 2014 Stanford Research Systems SR1 Operation 103 control For the variable cases the Sample Control will be initialized with the current OSR or ISR the user
150. ave is filtered with a single pole 30 kHz filter Outputs a 15 kHz sinewave and 3 15 kHz squarewave Squarewave is filtered with a single pole 100 kHz filter Config IMO aveform IMD On iw Total Amp ft o000 Vrms Type Sine Freg 114 9576 kHz SMPTEIDIN CCIF DFD Sor Freg 3 14110 kHz C DME 0 DIM 30 Amp Ratio 4 1 A C DIM 100 IMD tab for DIM selection For all the DIM options the ratio of the Squarewave to sinewave peak to peak amplitudes is 4 1 Because the squarewave needs to be perfect frequency with an exactly equal number of positive and negative samples the exact squarewave frequency is chosen by the generator depending on the generator Fs to be as close as possible to the Squarewave frequency given by the standard The sinewave frequency is then set in the same ratio to the squarewave frequency as would be for the signals in the standard In all cases the Total Amplitude control sets the combined amplitude of all signals The IMD waveform is designed to be used with the IMD analyzer The analyzer automatically determines the type of IMD signal is being generated and automatically configures itself for the correct analysis See the IMD Analyzer section for more details Polarity Check Waveform Config Polarity avefarm Om Ww olarity Amp fo 0000 Vrms Freq 1 0000 kHz 2014 Stanford Research Systems s SR1 Operation Manual The polarit
151. ays calculated by the digitizer Optionally three additional sets of measurements can be selected If these additional measurements are not needed the corresponding check boxes may be left unchecked to save computation time Checking the Spectrum box instructs the digitizer to compute the FFT of the input signal and the jitter spectrum The window function applied before calculating the spectra can be selected from the same list of window functions available in the FFT analyzers The spectral resolution for the input amplitude spectrum can be selected between 512 and 16k lines The number of lines in the Jitter spectrum is 2014 Stanford Research Systems SR1 Operation 189 adjustable between 256 and 8k lines ik zk Sk 10k 20k 50k 100k 200k ii Spectrum of 200 mUI 10 kHz Sinusoidal Jitter signal Checking Probability computes the histograms of the input amplitude jitter amplitude input pulse width and input pulse rate For a digital audio or clock signal the input amplitude histogram is dominated by two peaks corresponding to the positive and negative voltages of the signal Pr 0 iu ZH 3H 4u SH H fu Bu ou Typical Digital Audio Carrier Signal Om 4 sn Se ree eee ers E A PEO EE eee keane E e PE PE Ea oe ee a e a Pia ee ee ee Peak e E E E S S Corespondinc to Masinmurn Positive 260m pec i m a cans endcasevbccnennteaass Corresponding to Maximu
152. ays time which increases monotonically in intervals of the sampling period For digitizer files created by SR1 s optional digitizer the sampling period is always 12 5 ns 80 MHz However it is possible to create files with different sampling rates and import them into the digitizer for display and analysis The time difference between the first two points is used to compute the sampling interval when a digitizer file is read in The second column gives the magnitude of each sample in Volts Columns are separated by a Tab character When a digitizer file is loaded the X axis values in the file are ignored and the Y values are simply loaded as sequential samples The X axis is aligned so that the first transition of the type specified in the Jitter Detection control on the Digitizer Panel Preamble all bits etc is at time 0 A fragment of an example digitizer file is shown below Users are encouraged to modify digitizer files taken by SR1 or to synthesize their own to experiment with the capabilities of the Digitizer SR1 Digitizer Time Record 17 x UNITES 6 Y Unita y O Oa UTI 1 25e 08 2a020 24 DE 06 ars as Je T SCUL Laaa 2014 Stanford Research Systems SR1 Reference 309 5e 08 AE 6 25e 08 ZOO 7 5e 08 22475 8 75e 08 225 l1e 07 TAE 1 125e 07 ZOD 1 25e 07 TER RS 1 375e 07 POPS 1 5e 07 Ge aS Arbitrary Waveform Files ARB SR1 arbitrary waveform files are ASCII text files containing columns of floating point nu
153. be used to connect an external monitor If the SR1 is equipped with the optional XGA 1024x768 display then the external monitor will run in 1024x768 resolution If SR1 is equipped with the standard VGA 640x480 display then the external monitor can be run either in VGA mode or in XGA mode If a VGA SR1 is operated with an external monitor in XGA mode the front panel LCD display will no longer show the entire screen but instead will show a 640x480 portion of the screen that will pan as the mouse is moved To enable operation with an external monitor 1 Using an external keyboard or the virtual keyboard bring up the windows start menu by pressing the ey 2 Choose Settings Control Panel from the Windows start menu 3 Double click the Display icon and choose the settings tab Click on the Advanced Button 4 Click on the S3Display tab You should see the following panel 5 Click on the CRT box to enable the external monitor 6 Click OK to return to the Display control panel To enable XGA resolution on a VGA SRT 1 Follow steps 1 6 above 2 In the Screen Resolution control move the slider to select 1024x768 resolution Screen req lution Less More 1024 by 768 peels 3 The external monitor will now display the entire SR1 screen while the front panel monitor will display a 640x480 portion of it which will pan in response to mouse movements Rack mounting SR1 The optional O1RM rack mount kit allow mounting the S
154. button to set up a list of network users who will be allowed to access SR1 2014 Stanford Research Systems 236 SR1 Operation Manual 2 6 6 Computer Functions The Tools menu contains a number of options related to the configuration of SR1 s internal computer These are Printer Panel Allows configuration of existing printer and adding new printers Eject Drive Safely shuts off removable drives prior to removal Power Options Sets idle time before shutting off the display and hard drive Date Time Allows setting of the internal clock Virtual Keyboard Brings up the virtual keyboard for text entry without a physical keyboard Update SR1 Updates SR1 to the latest version of firmware 2 6 6 1 Printers Panel SR1 can print to a USB printer directly connected to the front panel USB port or to any printer on the network Click Add a printer to configure a new printer gt Printers and Faxes Eei gt alg p Search B Folders E Address Printer Tasks 4dd a printer Send a fax h Ap deskjet 5100 See Also 3 HPSSSO on Modoc 2 Troubleshoot printing o Get help with printing 2 HP9040 on Modoc Other Places E Control Panel h Microsoft XPS Document Writer gt Scanners and Cameras My Documents fe My Pictures H My Computer Details When installing a printer on SR1 the printer driver files must be either in an accessible network location or on a USB drive connected to the instrume
155. ce Denied IP Addresses Check Denied List First 197 165 1 14 192 168 1 29 25 Password clear text stl iw Require Password Basic VXI 11 security measures can be configured on the VXI 11 Security Panel which is accessed from the VXI 11 Options tab by pressing Security The panel includes controls for creating a list of IP addresses that will be allowed to connect to SR1 as well as a list of IP addresses that will be denied access to SR1 By default SR1 checks the allowed list first but if Check Denied List First is checked then SR1 will check the denied list first The wildcard character 192 168 1 as well as explicit ranges of addresses 192 168 1 30 55 are allowed in both lists As an example to allow only a specific range of IP addresses to connect to SR1 one would specify in the denied list and the required allowed address range in the allowed range Check Denied List First should remain unchecked in this case Otherwise SR1 would deny access to all IP addresses Alternatively to allow access to all computers except those in a specific address range one could enter in the allowed list and the range of addresses to be denied in the denied list with Check Denied List First checked in this case The optional password is a clear text message that must be sent via VXI 11 immediately after connecting to SR1 The user has 3 tries to enter the password after connecting before the connection is closed by SR
156. ch Systems 128 SR1 Operation Manual Heterodyne Selected Input _ x a 3 Time Record Buffer Time Record Outputs Time Record i _ Power Spectrum ji i Averaging Magnitude Computation Linear Spectrum Magnitude Phase Levels co Peak Based p Other Input Level Meas FFT1 Analyzer Block Diagram A block diagram of the FFT1 analyzer is shown above The selected input signal is first optionally heterodyned to move the selected center frequency to the center of the FFT analysis range The signal is then optionally decimated by up to a factor of 21 in order to reduce the sample rate when using the zoom feature Each stage of decimation includes filtering to eliminate alias effects from the discarded portions of the frequency spectrum The output is sent to a buffer which serves as the time record for the FFT analyzer Each time record starts with a trigger If the analyzer trigger is not enabled then a trigger is automatically generated as soon as the DSP has finished processing the previous time record Otherwise the analyzer waits for a trigger which matches the specified trigger criteria and begins the time record at the trigger point After a trigger occurs and enough time record points have been accumulated to compute a spectrum of the specified resolution the DSP applies a windowing function to the time domain data See Window Selection Windowing is necessary due to the finite length of the FFT
157. ch case only FFT lines that are fully alias protected are displayed Areas of the spectrum that are not fully protected are shown in red in the graphical frequency indicator on the FFT panel 2014 Stanford Research Systems 132 SR1 Operation Manual Resolution Resolution 4cq Timet ik 16 0 msec r 32k 512 msec 16k f 256 msec j ok 128 msec 4k 64 0 msec k ik 16 0 msec 256 i 4 00 msec The resolution control determines the number of lines in the FFT spectra Values from 256 lines to 32k lines can be selected In the resolution control the time to acquire a time record for the selected resolution and current bandwidth selection is shown alongside the number of lines Obviously the higher the selected spectral resolution the longer it will take to acquire the time record for that spectrum Averaging Ayveragin sig vgs Clear Avg Done Exponential 1 _ Fixed Length Continuous Cont PkHald Both the Power Spectrum and Linear Spectrum are averaged in the case of the Power Spectrum the power in each bin is averaged while for the Linear Spectrum the complex spectrum is averaged in each bin The amount and type of averaging for both spectra is specified by the controls above The averaging type controls determines how each of the spectra is averaged None implies that no averaging is performed In this case the magnitude of the Power Spectrum and Linear Spectrum are the same a
158. computes frequency response using a relative measurement it can make highly accurate measurements of frequency response magnitude and phase without the need for a perfectly flat reference source Freq Resp Fb s Fa s A E Device Under Test Typical Dual Channel Response Measurement The diagram above shows a typical two channel measurement setup SR1 s Generator provides the stimulus signal which can be a broadband source such as noise or the FFT chirp signal or can be narrowband The generator output is split and sent to both the DUT and the A channel input of the analyzer The output of the DUT is sent to the B channel input of the analyzer The analyzer then computes the quotient of the two complex FFTs which yields the magnitude and phase of the frequency response FFT2 Analyzer Block Diagram Selected Inputs Trigger Contral ATE Decimati i Time Record Buffers 2 Outputs Time Record Time Records B Power Spectra AtB m Magnitude Averaging Mag l FFT i Linear Spectra B E Computation Freq Resp P Computation Magnitude Phase Frequency Response Magnitude Phase i Peak Based Levels A B Level Meas A block diagram of the FFT2 analyzer is shown above Both input channels of the selected domain are first optionally decimated by up to a factor of 21 depending on the user s bandwidth selection Each 2014 Stanford Research Systems 142 SR1 Operation Manual s
159. connector ID assigned to that connector The physical connector IDs always range from 1 12 whereas the logical IDs can range from 1 to 192 depending on the number of connected switchers Logical connector IDs are used when setting up a sweep over the switcher inputs or outputs Right clicking on a switch displays the switch options submenu Delete deletes the selected switch from the configuration Reset resets all connectors to their default positions no connection Identify causes relays to click on the selected switcher so that the physical switch box can be easily identified Edit displays the Switch Info panel 5 Switch Info Physical Mame utputSwitcht Switched Ch El Dest ch E Control Chain Addr 5 Test Change Network Order F The Switch Info panel allows renaming of the switch and displays the communication parameters for the switch Pressing Test tests communication with the switcher the button turns green if SR1 successfully communicates with the switch and red if not 2014 Stanford Research Systems 230 SR1 Operation Manual 2 6 4 Hardware Status The Hardware Status panel is a diagnostic panel that is not needed for normal operation of SR1 During normal operation of the instrument all the status indicators for all of the boards should display green Should a red indicator appear indicating a problem with the hardware contact Stanford Research Systems for technical support a
160. ct for offline viewing of saved datafiles or simply becoming familiar with the software Like all Windows software operating SR1 requires a pointing device and a means of entering text and numeric data The instrument provides several options depending on the intended environment External Mouse and Keyboard Control SR1 can be used with an external mouse and keyboard The mouse and keyboard must be connected to the rear panel connectors before the instrument is turned on to be properly recognized by the instrument Operation with an external mouse and keyboard is convenient when SR1 will be used in a benchtop environment Using the Trackpad For situations where it s inconvenient to use an external mouse e g rackmount SR1 has a front panel trackpad which can be used as a pointing device Note that the trackpad is always active even when an external mouse is connected To use the trackpad drag your finger over the surface of the pad to move the cursor on the screen To left click gently tap your finger To double click tap your finger twice Finally to right click tap your finger in the upper right triangle colored in a lighter shade of gray Using the Front Panel Keypad The front panel keypad provides a convenient way to enter numeric values as well to access some commonly used SR1 functions without having to reach for the mouse or keyboard 2014 Stanford Research Systems 14 SR1 Operation Manual Standard Functi
161. ct squarewave frequency is chosen by the generator depending on the generator Fs to be as close as possible to the squarewave frequency specified by the standard The sinewave frequency is then set in the same ratio to the square wave frequency as would be for the signals in the standard Note that the DIM waveform cannot be generated if the digital audio generator bandwidth is sufficiently low OSR lt 30 kHz 2014 Stanford Research Systems SR1 Operation Manual Config IMD Waveform 7 OMD On lw Total Amp 1 0000 Vrms Type Sine Freq 9576 C SMPTE DIN C CCIFIDFD Sor Freg 3 14110 kHz DIME i pM 30 Amp Ratio DIM 100 IMD tab for DIM selection In all cases the Total Amplitude control sets the combined amplitude of all signals The IMD waveform is designed to be used with the IMD analyzer The analyzer automatically determines the type of IMD signal is being generated and automatically configures itself for the correct analysis See the IMD Analyzer section for more details Polarity Check Waveform Config Polarity Waverorn On fw Polarity Amp 1 0000FFS Freq i 000 kHz The polarity check waveform uses a phased combination of two sine waves to produce a deliberately asymmetric waveform that poin
162. ctionality of a PC keyboard using only the mouse Fie Keyboard Settings Help proccmm jiftetstats e 7s s o j bksp fins tm puf E E GAppEpnApNipH0ONE OCT 00nn TDpDEnnnpn0nnnnEE noo fee l lel lel l gt n DBA PLCI ff ied Use the virtual keyboard for text entry e g filenames in situations where SR1 is not connected to a PC keyboard The virtual keyboard can be selected via the main menu under the Tools category or using the front panel keypad by pressing lt Alt Alpha gt 2 6 6 6 Updating SR1 Stanford Research Systems is continually updating SR1 s firmware to provide new functionality and fix bugs There are two methods for updating the firmware depending on whether the instrument is 2014 Stanford Research Systems SR1 Operation 239 connected directly to the internet or not If SR1 is connected to the internet simply select Update SR1 from the Tools menu and the instrument will automatically contact the update server at Stanford Research Systems SR1 Audio Analyzer Update Welcome Welcome to the SAT Audio Analyzer update This program will connect to a Tre Update server to find out if a new version of SAT Audio Analyzer it available Please make sure that you are connected to the Internet and then click Next to continue Follow the onscreen directions The update server will determine if the installed firmware version is the most recent and if necessary download and instal
163. ctor that contain no tone harmonic of a tone or IMD product of tones This vector measurement is only active when Noise Analysis is selected For Total Distortion each non tone bin pair the power in the odd bin noise is subtracted from the Bins A B power in the corresponding even bin noisetdistortion The resulting vector is a good indicator of the total distortion at each frequency Frequency The vector containing the ratio of the received amplitudes at the tone locations Response Mag A relative to the generated tone amplitudes B The vector containing the phase at the tone locations relative to the phase of each generated tone For the phase measurement to be meaningful several conditions must be met First the MTA must be running in synchronous mode Frequency Second the analyzer must be triggered so that the analyzer s FFT time record Response Phase _ Maintains a consistent position with respect to the generator signal The A B frequency response phase is calibrated only when using the generator trigger source Be sure to set the trigger source on the Trigger tab of the analyzer to Generator and set the Generator Trigger Source to either Analog or Digital depending on which generator is being used The vector of the ratios of the sum of the harmonics for each tone and the noise THD N vs Freq A amplitude to the received amplitude at the tone frequency For this measurement it useful to select a tone frequency d
164. d A weighting filter is applied A weighting is specified in ANSI standard S1 4 1983 and is typically used for noise and THD N measurements with audio applications C Msg Wt The C Msg weighting filter specified in IEEE Std 743 1995 is intended to be used for noise measurements associated with voice transmission telecommunications CCITT The CCITT weighting filter defined by ITU T Recommendation O 41 is another telecommunication noise weighting filter CCIR wtd Filter defined in CCIR Rec 468 4 for audio noise measurements Designed to be used with the Quasi Peak setting on the Time Domain Detector CCIR unwtd Filter defined in CCIR Rec 468 4 Annex ll for making unweighted noise measurments CCIR 2 kHz This filter is identical to the CCIR wtd filter except the normalization is changed so that the filter has unity gain at 2 kHz instead of 1 kHz Vector Harmonic Measurement In addition to the scalar THD measurement the THD analyzer produces a vector measurement which contains the relative amplitude of all harmonics regardless of the harmonics selected in the panel 67 5 ASRS 100 0 2014 Stanford Research Systems 160 SR1 Operation Manual The X axis for this measurement is simply the harmonic number Using the THD Analyzer with the Time Domain Detector The THD analyzer can be used together with the Time Domain Detector to make the most precise THD measurements possible Normally the selected input
165. d gaussian fit 100m ee sieectacadeacccans e wasGusuansssuseasiecsusnsaceeesueuatecedsisesactessteusueuateseducnecsduiesseraseusserssesssasunvasersseucssass m e r ee ei een AEAEE PETNE IE er OIE PARINAAN IAI IOIO I IE TIIE IA deur E TONE AVAE IOA AE AAAA a EE PEIN ee ee ee e eee e raa a ee E RUT e E E E F E E E E TEI EOE E EEE AE N E ETEEN OUT e E E E e E E E E z i Piece sense ened theca a tans aa aAa E a a A A A 1 25 1 00 750m 500m 250m 0 250m 500m 750m 1 00 1 25 p Histogram of Analog Noise w Gaussian Fit 2014 Stanford Research Systems SR1 Operation 175 2 4 9 Octave Analyzer The Octave analyzer Sometimes called an RTA or Real time analyzer measures the power present in full octave 1 3 octave and 1 12 octave frequency bins from 20 Hz to 20 kHz The bin centers and shapes follow ANSI Std S1 11 2004 In the analog domain the octave analyzer always uses the Hi Resolution converter Octave Analyzer Outputs Octave A B The fractional octave spectrum for each input channel Octave Delta The difference between the A and B channel octave spectra Example Octave Spectra dEYrms Hz 20 a0 100 200 S010 ik zk Sk 10k The figure shows a 1 3 octave spectrum of A weighted white noise Octave Analyzer Panel 2014 Stanford Research Systems 176 SR1 Operation Manual 5 Al OCT Analyzer SOUrce Conver ter Fs araos zines l Measurement References Trigger Octave Mode Weighting
166. de 50 FS 5 FFS This digital waveform amplitude unit is directly convertible to FFS depending on the value of the V FS Volts Full Scale set in the references section of the digital generator panel If the value of V FS is 2 Vrms for instance then an amplitude of 1 Vrms corresponds to an amplitude of 5 FFS These units allow digital amplitudes to be expressed as analog voltages as is often useful when working with D A converters where the V FS value can be thought of as the DAC s full scale output voltage Vrms This unit of digital wave form amplitude is convertible to FFS depending on the value of the V FS Volts Full Scale reference and the fixed relationship of peak voltage to rms voltage for a sinewave If the value of V FS is 1 Vrms for instance then an amplitude of 1 414 Vp would correspond to an amplitude of 1 Vrms which in turn would correspond to an amplitude of 1 FFS pp This unit of digital wave form amplitude is convertible to FFS depending on the value of the V FS Volts Full Scale reference and the fixed relationship of peak voltage to peak to peak voltage for a sinewave If the value of V FS is 1 Vrms for instance then an amplitude of 2 0 Vpp would correspond to an amplitude of 1 Vrms which in turn would correspond to an amplitude of 1 FFS h i lt The peak value of the waveform expressed as a decimal code The conversion of decimal code to FFS depends on the setting of the Output Resolution in
167. device rather than cabling effects Since preambles occur only once every 32 bit cells in the digital audio signal the sampling frequency for the calculated jitter will be 1 32 of the bit cell rate and the number of calculated jitter points will be equal to the number of preambles in the input record This implies that the bandwidth for jitter measured with this selection is 1 2 of 1 32 or 1 64th of the original bit cell rate which is equal to the sampling rate of the embedded digital audio signal Selecting All Bits reconstructs the clock by minimizing the jitter at each bit cell transition This method has the maximum measurement bandwidth for the calculated jitter signal The jitter bandwidth will be 1 2 of the bit cell rate or 32 times the embedded digital audio sampling rate Because the jitter is being sampled at each bit cell transition the calculated jitter will include contributions from both the intrinsic jitter of the transmitter as well as pulse pile up effects due to bandwidth limiting in the transmission channel Since during preambles there are bit cell boundaries where no actual transition occurs the digitizer interpolates the position of these virtual transitions The Stable Bits selection is a midpoint between the Preambles and All Bits The jitter for this setting is calculated at every 4th bit cell starting from the first transition in the preamble The jitter bandwidth is thus 1 8th the bit cell rate or 8 times the embedded
168. during a sweep If Auto On is checked the generator output will automatically turn on at the beginning of a sweep if it was turned off and turn off when the sweep is completed The large green red On Off buttons turn on and off their respective channels while the Invert buttons invert the output for each channel Reference Controls eferences der Ref lt 0000 FFS Freq Ret ii 00000 kHz vY FS i o000 Yms F The Reference controls allow setting several parameters used in the computation of different generator amplitude units See the Digital Generator Units section for a complete description of all these units Note that there is one set of references for both generator channels Burst Controls urs Lo Amp Ji0 00 To Period J10 00 Cycd z Dry CY 50 00 Yo The SR1 Digital Generator implements a Timed Burst mode which switches the overall amplitude of the combined waveform output between two different values Burst Mode turns the burst feature on and off Lo Amp selects the lo burst amplitude as a fraction of the original generator amplitude Burst Period controls the total hi lo period of the burst cycle This value may be entered in seconds or in cycles of the current waveform Burst Duty Cycle controls the fraction of the burst period during which the generator output is at the hi amplitude 2014 Stanford Research Systems eo SR1 Operation Manual 2 3 2 1 Burst Digital Noise Period 20 ms
169. e Hi BW ADC 1 3 Octave Class II 4 pole Hi RES ADC 1 3 1 6 1 12 1 24 Octave Class Ill 6 pole Digital Audio 1 3 1 6 1 12 1 24 Octave Class Ill 6 pole Tuning Range Hi BW ADC 10 Hz to 200 kHz Hi RES ADC 10 Hz to 0 44 Fs Digital Audio 10 Hz to 0 44 Fs Tuning Accuracy 2 5 Amplitude Accuracy 0 5 weighting Filters A wt C Msg wt CCITT CCIR weighted unweighted amp 2 kHz norm Optional Filters Up to 4 per channel Hi BW only Detector Type RMS Peak Quasi Peak CCIR 468 Single Channel and Dual Channel FFT Analyzers Frequency Range Hi BW ADC DC to 200 kHz Hi Res ADC DC to 0 45 Fs Digital Audio DC to Fs 2 2014 Stanford Research Systems Number of Lines Processing Windows Zoom Heterodyne Averaging Dual Ch Measurements Other Analyzers THD Analyzer IMD Analyzer Histogram Analyzer Multitone Analyzer SR1 Reference 303 256 512 32k 40 bit floating point Blackman Harris Hanning Hamming Equiripple Flattop Gaussian Kaiser Uniform Rife Vincent 4 5 and 10 term Span can be narrowed by up to 512x Narrowed span can be centered anywhere in the measurement range Single Ch FFT only Continuous and fixed length Frequency response coherence Measures two independent sets of user selectable harmonics 2x to 14x SMPTE DIN CCIF DFD DIM TIM Time vs amplitude Histogram PDF Gaussian fit to PDF Level Frequency Response THD N noise THD IMD Crosstalk Digital Audi
170. e Analyzer panel The rms sum of all non tone bins between the the frequency limits If noise THD N Sum A B analysis is enabled this includes all odd noise bins and all even bins without tones THD Sum A B The rms sum of all bins containing harmonics of tones but not tones The rms sum of all bins between the specified frequency limits containing IMD IMD Sum A B products up to the selected order of all tones l The rms sum of all noise bins between the frequency limits If Noise Analysis is Total Noise A B enabled this is simply the sum of all the odd bins within the frequency range Total Distortion A The rms sum of the total distortion bins between the frequency limits This B measurement is only active when Noise Analysis is selected The ratio of the highest amplitude received tone to the lowest amplitude received Lowest Tone A B The frequency of the lowest amplitude received tone Highest Tone A B he frequency of the highest amplitude received tone Because the multitone source has a periodic time dependent amplitude profile it may fool the input autoranging control into switching ranges over the duration of the signal It may be necessary to turn off input autoranging while using the Multitone Analyzer and manually set the input range The Multitone Analyzer Panel 2014 Stanford Research Systems 168 SR1 Operation Manual AQ Multitone Analyzer EID Input Source Converter Fs anoo He
171. e Digital Generator is loaded with a sine waveform Page2 contains a graph displaying the A and B channel frequency response The sweep controller is programmed to sweep the sine frequency from 20 Hz to 20 kHz in 22 logarithmic steps What you ll need to configure Adjust the the sine amplitude to match the equipment being tested The number of points in the sweep can be changed to give faster sweeps or higher resolution sweeps The settling parameters can be adjusted to fit the noise levels of the signals being measured Wideband FFT This setup is similar to the audio band FFT setup described above except that the analog input sampling rate is set to OSRx2 so that the FFT bandwidth is twice as large The AO analyzer is configured as a dual channel FFT analyzer with continuous averaging The Digital Generator is loaded with a low distortion sine waveform set to a default frequency of 1 KHz and an amplitude of 1 Vrms Page2 contains a graph displaying the A and B channel power spectra while Page 3 contains a graph displaying the A and B channel time records What you ll need to configure If your e using the digital generator adjust the amplitude and frequency of the signal to match your requirements Otherwise you can turn the generator off The amount of averaging in the FFT analyzer can 2014 Stanford Research Systems 284 SR1 Operation Manual be adjusted to suit the noise level of your signal The FFT resolution is defaulted to 4k
172. e instrument has a 3 wire power cord Be sure to connect it to a properly grounded outlet to guard against electrical shock After connecting the power cord to an AC power source the unit can be turned on by pressing the power button at the lower left of the front panel The green power LED above the power button illuminates to indicate the unit is on After turning the power on the unit will take about 50 seconds to run through its boot up sequence before it is ready for use Safety Dangerous voltages capable of causing injury or death are present in this instrument Use extreme caution whenever the instrument cover is removed Do not remove the instrument s cover while the power cord is connected to a live outlet Do not expose the instrument to rain or excessive moisture Do not attempt to disconnect the internal cooling fans or block the fan vents Service Refer all servicing to qualified SRS authorized service personnel Do not attempt to substitute parts or perform any unauthorized modification to the instrument Contact SRS for instructions on how to return the instrument for authorized service calibration or adjustment Stanford Research Systems 1290 D Reamwood Ave Sunnyvale CA 94089 USA phone 408 744 9040 email info thinksrs com web www thinksrs com 2014 Stanford Research Systems Getting Started 2014 Stanford Research Systems 8 SR1 Operation Manual 1 2 Manual Revision History Copyright
173. e levels displayed on the FFT2 analyzer panel are peak based levels not RMS levels Remember the units the levels may be displayed in are independent of the method of computation Because the FFT2 analyzer needs to take time to compute FFTs the continuous input data stream necessary to compute RMS levels is not be available Likewise the fact that continuous data is unavailable means that the A B phase computation which also requires continuous input data is not available for the FFT2 analyzer Frequency data is available for analog inputs which use a hardware based frequency measurement technique but not for digital audio inputs Bandwidth Bandwidth Baseband 256 00 kHz Y 756 00 kHz The maximum bandwidth of the FFT is 1 2 of the sampling rate for the selected input source To instantly switch to full bandwidth click the Baseband button The Zoom feature of the FFT2 analyzer allows selection of narrower bandwidths in factor of 2 increments as well When the bandwidth is decreased the effective sampling rate is decreased and therefore the amount of time necessary to 2014 Stanford Research Systems SR1 Operation 149 acquire a time record of a given length will increase Therefore the FFT display will update more slowly as the bandwidth is narrowed Unlike the FFT1 analyzer the FF I2 analyzer has no heterodyne capability so that when the bandwidth is decreased the frequency range always starts at DC Aliasing
174. e of a Square wave is equal to the digital audio sampling interval times the value of Samples Point parameter on the waveform panel Thus in the example below each half cycle of the square wave is 100 48 kHz 2 083 msec and therefore the time spent at each code is 5x2x2 083 msec 20 83 msec as can be seen in the graph hex wee UU Meee cee a SE TTO annan a aa Oxo Ox4 Nee ee eee ee ee j tld ceded E PE oxa HHH EHHH oe a E oaan a 0 25m 50m 75m 100m 125m 150m 175m 200m 38 Digital Staircase Fs 48kHz Samples Point 100 When observing the digital staircase using the FFT time record be sure to enable the analyzer trigger and set the trigger source to generator This will ensure that the beginning of the time record is synchronized with the beginning of the staircase J Test Config J Test J Test On w 2014 Stanford Research Systems SR1 Operation The JTest waveform conceived of by the late Julian Dunn is designed to excite jitter due intersymbol interference in digital audio signal paths with reduced bandwdiths such as long cables The Jtest waveform is primarily a square wave at OSR 4 but the dc level is slightly shifted every 192 samples The resulting jitter will occur at OSR 192 and can be seen with the Jitter Analyzer or the Digitizer 2014 Stanford Research Systems SR1 Operation Manual 2 3 3 Analog Inputs Panel The analog inputs panel controls and monitors the configurat
175. e of the limit box Eye Limits Enable Limits Scale By Presets iw 1 0000 Load Save Inner Upper Inner Lower Outer Limits W Mirror Left Right 0 0000 UI O 0000 w 290 00 mL O 0000 w 250 00 ml 1 0000 0 0000 x 250 00 mur Y 1 0000 Replace Add Delete The Scale Y By control can be used to scale the Y values of all the limit segments simultaneously so 2014 Stanford Research Systems SR1 Operation 211 that the entire limit box can be quickly scaled up or down Limit settings can be saved to disk and recalled with the Load and Save buttons on the Eye Limits panel When Eye Limit Testing is enabled the text on the Limits button is drawn in Red when the eye diagram plot fails any of the limit tests 2014 Stanford Research Systems 212 srt Operation Manual 2 5 4 Other Display Menu Options Save All Displays Saves all displays Graphs Bar Charts and Digitizer Displays to an XML configuration file When this option is selected a file dialog appears which allows selection of the file and directory The file dialog contains an additional option which controls whether data is saved with the file or whether the display is saved as a placeholder with all options and scales preserved but no trace data Never Save saves none of the trace data Always Save saves all of the trace data Save if Offline saves trace data only for offline traces but not for live traces Load D
176. e selected tone is the first lowest frequency tone Relative to Tone The absolute voltage for each channel is divided by the received amplitude of the Selected Frequency selected tone in the other channel The selected tone is selected on the Edit Other Channel Multitone Tones panel which can be accessed from the Multitone Configuration panel The default for the selected tone is the first lowest frequency tone Relative to RMS The absolute voltage for each measurment is divided by the RMS voltage of the Amplitude entire received signal Noise Analysis The Noise Analysis checkbox enables noise related measurements in the MTA When noise analysis is selected the MTA uses an FFT record which is twice the resolution of the corresponding generator signal length that is selected with the Multitone Configuration Panel Tones harmonics of tones and IMD products of tones will then fall only on even bins in the received FFT while the odd bins will contain only noise Noise Analysis is only available in the synchronous processing mode since the use of a window function would cause the precise division of the odd and even bins to be lost The Noise and Total Distortion measurements are only available when Noise Analysis is checked The Scalar Measurement Panel Scalar Meas References Trigger Debug irt 20 000 Hz End j22 000 kHz IME 32 10 prs l Moise To Fipple Loudest Cuiebest The scalar mea
177. e with the desired sweep parameter values in one column Then select the file using the file dialog box opened by the button When a table sweep is selected the Start Stop values and the Steps Step Size entries are ignored the sweep simply proceeds from one table value to the next until the table ends Configuring Time Sweeps In a time sweep the sweep x axis is based on the passage of time rather than the progression of some parameter value The start and stop time determine the time of the first and last measurements taken in the sweep The step size determines the amount of time between sweep points If the amount of time necessary to obtain settled sweep data exceeds the step size SR1 can take two different actions depending on whether the sweep source was chosen as Intersample Delay or Absolute Time If the source is Intersample Delay SR1 always waits at least a step size after getting settled sweep data before beginning a new measurement If Absolute Time is chosen the analyzer will begin a new measurement immediately after finishing the previous measurement if necessary Note that in both 2014 Stanford Research Systems 94 SR1 Operation Manual cases the final number of points in the sweep may be less than Steps if the amount of time necessary to obtain settled data is significant Configuring External Sweeps External sweeps allow SR1 s measurement of an external parameter to determine the the sweep X axis For
178. ecessary for most audio signals its use can improve the Stability of amplitude measurements when using burst or signals of a transient nature 2014 Stanford Research Systems SR1 Operation 127 2 4 3 FFT Single Channel The Single Channel FFT Analyzer FF 1T1 computes the frequency spectrum of the selected input signal The input to the FFT1 analyzer can be either channel of the analog or digital audio input or can be the amplitude output of the Time Domain Detector SR1 s FFT analyzer can operate over the range from DC to Fs 2 with a resolution of 32k lines or the spectrum can be zoomed by up to a factor of 1024 to apply the full 32k line resolution to a smaller portion of the frequency located anywhere within the full range For example the graph below shows the output of a typical device when subjected to a test Signal consisting of two closely spaced sines near 10 kHz delta 60 Hz dByrrns 0 5k 10k 15k 20k 25k 30k Fe Full Range Fs 64 kHz Spectrum of IMD signal Resolution 4k lines dB yrs 3 5k 9 6k g 7k 3 8k 3 9k 10 0k 10 ik 10 2k 10 3k 10 4k Hz Same Spectrum Zoomed x32 and centered at 10k Hz Resolution 4k lines The top diagram shows the original spectrum showing 4k lines over the range from DC to Fs 2 The bottom spectrum has been zoomed to a range of 1kHz and centered at the input frequency of 10 kHz to reveal additional spectral detail FFT1 Analyzer Block Diagram 2014 Stanford Resear
179. ed in their respective input streams CRC For AES EBU digital audio streams an 8 bit CRC code is generated from the received channel status bits in each block and compared to the value stored in the last byte of the channel status block A CRC error occurs if these values do not match Lock An error occurs if the digital audio receiver on the given connector cannot lock to the input signal Validity An error occurs if the validity bit is set on either subframe of the received or transmitted Signal Coding Bi phase coding error The error occurs if the carrier signal remains at the same level for greater than 2 unit intervals other than during preambles Parity An error occurs if the computed parity of each subframe does not match the parity bit contained in that subframe Post Trigger Length For all trigger sources except auto trigger the user can specify the fraction of the total digitizer record that occurs after the occurrence of the trigger A post trigger fraction of 100 means that the trigger occurs at the very beginning of the record while a fraction of 50 means that the trigger occurs precisely in the middle of the record This features enables examination of signal features that occur pre trigger For the Auto trigger selection the post trigger length is always set to 100 Digitizer Analysis Controls Jitter Detection Stable Bits Re analyze mu 5 A Jit Rec Len Clock Freq RMS Jitter Fast Interp
180. eed 1 FFS regardless of waveform The Waveform Frequency control sets the frequency of many waveforms in one of the digital generator frequency units Digital generator frequencies range from 10 Hz up to Fs 2 where Fs is the current digital output sampling rate OSR The current OSR is displayed at the top of the digital generator panel Generator Trigger Certain generator waveforms can generate a trigger known as a generator trigger which can be used by the analyzers to synchronize the analyzer to a certain portion of the waveform Triggering is a complex subject which is fully described in the Analyzers section and there are many different possible analyzer trigger sources besides generator trigger In this section the discussion of triggering will be limited to those waveforms which provide generator triggers and where in those waveform is the trigger occurs When the generator is configured with multiple waveforms the first waveform which is generator trigger capable will be the source of all generator triggers The Digital Generator Waveforms In the following list waveforms that may not be combined with other waveforms are marked with an asterisk next to their names Sine Config Sine aveform On fw EQ ine mp feo0 o0 mFFS The sine wave is the most basic audio test waveform Sine waveforms are specified only by their amplitude and frequency The Normal Sine waveform may be combined with other waveforms
181. eep Some experimentation may be required depending on the characteristics of the EUT to find the optimal values of the settling parameters 2014 Stanford Research Systems SR1 Operation 261 2 8 5 THD N Panel E Measure THD N Inpukiutouk Measurement Settling Signal Waveform LD Sine o Start Stop Steps Log Freq 20 000H2 oo ho i Level 100 00 mems f o alr Analysis Highpass 10 Hz ha Lowpass 22 kHz ad Measure Ratio Append Traces Free Run Sweep The THD N Quick Measurment panel uses the Time Domain Detector to measure the THD N Total Harmonic Distortion plus Noise of the EUT In free run mode the measurement displays the instantaneous THD N of the EUT at a single frequency and amplitude in addtion the source can be swept over frequency and amplitude to yield the THD N vs frequency of the EUT for a range of input amplitudes Cabling Configuration The output of SR1 s analog or digital generator is connected to the EUT s input The output of the EUT is connected to SR1 s analog or digital inputs The number of inputs and outputs and the connectors used are specified in the Setup Panel Generator Configuration The output waveform is set to either Sine or Low Distortion Sine as specified in the panel For THD N values below about 90 dB use the Low Distortion sine For free run operation the amplitude and frequency of the generator is set to the frequency and level in the start column of
182. elect a log X axis to match the log sweep we just specified and select dB as the Y axis units with the unit entry controls Select a Y axis range from 95 to 10 dB Now start the sweep by pressing the Sweep button on the front panel After a few seconds you ll see the results of the sweep This of course is exactly what we expect At an amplitude of 100 uVrms the THD N is simply 20 log 100 uV 1V 80 dB At the other of the sweep the the THD N is 20 log 1V 1V 20 dB This quick example only scratches the surface of SR1 s capabilities but should give you a feeling for how the instrument operates 2014 Stanford Research Systems SR1 Operation Part SRS Audio 28 SR1 Operation Manual 2 SR1 Operation 2 1 File Menu The File Menu contains options for saving and recalling instrument configurations and also for printing e Save SR1 Saves the entire instrument setup to a configuration file Configuration e Save Partial Saves the entire instrument setup to a configuration file Configuration e Load Configuration Loads setup information from a configuration file e Print SR1 Screen Prints the currently displayed screen to the currently selected printer e Print Setup Displays the standard Windows Print Setup dialog allowing the selection of an installed printer and paper options 2014 Stanford Research Systems SR1 Operation 29 2 1 1 Save SR1 Configuration Selecting File gt Save SR1 Con
183. ency Domain Analysis 180 Resolution 181 Time Domain Analysis 180 Jitter Generation Amplitude 79 EQ 79 Waveform 79 Jittter Analyzer Input Selection 179 JScript 243 2014 Stanford Research Systems K Keyboard 13 Keyboard and Mouse Ports 293 Keypad 13 Knob 13 291 Ea e Limit Testing 198 Line Voltage Selector 6 Load Configuration 31 Load Displays 212 Lock 79 Locking the Clock to an External Source 105 Logging 250 Log sine chirp 43 62 Macros 216 Manual Revision History 8 Mapped Drives 233 Master Clock Output 293 MLS 43 47 62 Monitors 100 293 Mouse 13 Multitone 62 102 Multitone Analyzer 165 Distortion Products 168 Equalization 168 Measurements 165 Noise Analysis 169 Panel 167 Processing Modes 168 Trigger Panel 170 Validation 170 N Network Drives 233 Network Places 232 Network Setup 231 Networking Options 231 Noise 43 46 62 Normal Mode Noise 79 2014 Stanford Research Systems 0 Octave Anallyzer Panel 175 Octave Analyzer Averaging 176 Mode Selection 176 Optional Filters 76 305 OSR 79 Output Impairment 79 Overview 9 P Panels 35 Parity 79 Phase 271 273 Phase Limits 214 Phased Sine 43 Phased Sines 62 Polarity Check 62 Polarity Check Waveform 43 Power Button 291 Power Cord 6 Power Entry Module 293 Power Options 23 7 Pre emphasis 79 Preferences 214 Display Options 215 Phase Limits 214 Remote Interfacing 216 Startup Configuration 214 Print
184. enerator Panel Description The fundamental unit of frequency 1 Hz 1 cycle per second F R Ratio relative to the Frequency Reference A value of 3 F R with reference of 2 kHz gives a waveform frequency of 6 KHz Difference relative to the Frequency Reference A value of 500 dHz with a reference of 2 kHz gives a waveform frequency of 2 5 KHz Fret Percent of the Frequency Reference A frequency value of 50 Fref with a reference value of 10 KHz gives a waveform frequency of 5 KHz A cent is a logarithmic unit which represents 1 100 of a semitone of the musical scale 12 semitones make up an octave Thus a cent is 1 1200 of an octave octaves An octave is a factor of 2 in frequency Thus a frequency value of 3 octaves with a reference of 1 KHz gives a waveform frequency of 8 kHz 2014 Stanford Research Systems 2 3 1 2 SR1 Operation decades An decade is a factor of 10 in frequency Thus a frequency value of 2 decades with a reference of 2 KHz gives a waveform frequency of 200 KHz Analog Generator Waveforms SR1 s Analog Generator is capable of generating an enormous variety of different audio waveforms from simple ultra lo distortion sines to complex synchronous multitone waveforms Because of its unique architecture which allows different waveforms to be combined the generator offers almost limitless flexibility in providing the perfect audio test output In this section each waveform the bu
185. enerators Turns on generator channels that were turned off with F12 AK A m A TINIAN TAINANIAITAIA O O Olololglglg e a FAI SP Ye m n V V V IVIVIVIVIV T T 4 s gt NO O O ct 5 Q Q o O O O er 7 _ Save the current script Open a script Create a new script Copy selected text to the clipboard Cut selected text and copy to the clipboard Paste clipboard text at the cursor location Brings up the dialog for finding text in the script Brings up the dialog for finding and replacing text in the script lalalalalalala lAa O19 19 919 19 919 e vivivivivivivivly VIAINI lt xX QIZIO VN Prints the current script 2014 Stanford Research Systems Getting Started Unit Entries The Unit Entry is a type of control used extensively throughout SR1 that allows entry of numeric data in a range of different units The example below shows the generator amplitude control for example When the down arrow is clicked a the current value of the Sine mMmp p A000 Yrm w 1 0000 Vrms Freq fi 4142 vp 2 0204 Yop 0 0000 dBvrms 2 2155 dBu 0 0000 dEr 1 5639 dEm othe mi entry is shown in the drop down list expressed in each of the allowed units Selecting a different unit makes it the current unit for subsequent entries To enter a new value into a unit entry first click anywhere on the current text and then enter a new numeric value followed by an optional
186. ent is the RMS sum of the harmonic amplitudes for each of the selected harmonics Note that it is possible to select a harmonic which is outside the frequency range of the selected converter It is the user s responsibility to ensure that each of the selected harmonics is within the analyzer s frequency range If the Ratio box is checked the summed amplitude will be divided by the amplitude of the fundamental for the selected channel and the result will be reported as a ratio The definitions of THD N and THD both involve the ratio of distortion products to the amplitude of the fundamental However most time domain based analyzers including SR1 s TDD cannot separately identify the amplitude of the fundamental and instead measure the ratio of the distortion products to the total power in the signal For small distortion the difference is insignificant The THD Analyzer which is frequency selective correctly displays the ration of the sum of the harmonics to the fundamental amplitude The only exception to this is when the THD analyzer source is set to Other Analyzer and the THD analyzer is looking at the notch filtered output of the TDD In this case the Ratio result is divided by the total power in the signal as measured by the TDD Since the Other Analyzer input is only needed when measuring the smallest levels of distortion lt 95 dB this will not signficantly impact the measurement result Measurement Speed Speed Synch Avg C
187. ental lt 140 dB 0 25k 50k 75k 100k 125k 150k 175k 200k 225k 130 140 150 160 170 Hz Sweep Configuration For THD analysis a graph is created showing THD vs frequency A new trace is created for each amplitude specified in the amplitude sweep For the other analysis modes graphs similar to the free run graphs are created the difference being that as the amplitude sweep progresses a new trace is created for each amplitude value in the sweep 2014 Stanford Research Systems SR1 Operation 267 2 8 8 IMD Panel E Measure IMD Input Oukput Measurement Settling Signal Waveform SMPTE IMC Start Stop Steps Log Freq 7 0000 kHz Level 100 00 mvins oF o iia Analysis Products dz Append Traces Free Run The IMD Quick Measurment panel sets up SR1 s analyzers generators and displays to perform any one of the three standard intermodulation distortion IMD measurements SMPTE CCIF and DIM Cabling Configuration The output of SR1 s analog or digital generator is connected to the EUT s input The output of the EUT is connected to SR1 s analog or digital inputs The number of inputs and outputs and the connectors used are specified in the Setup Panel Generator Configuration The output waveform is set to the IMD waveform of the specified type For SMPTE measurements the high frequency is set to the value entered in the panel while the low frequency defaults to 60 Hz Fo
188. entially an oscilloscope display of the input signal The jitter vs time trace shows the jitter amplitude calculated by the digitizer as a function of time Jitter amplitude may be displayed in units of seconds or Uls If either trace is sufficiently zoomed in the display interpolates points between the actual measured points to provide a smoother visual presentation The X axes of the input amplitude and jitter amplitude traces are always the same Spectrum Tab Eve Diagram Probability Spectrum Time Rec a Ss S E E E y 0 oM 10m 15M UM 25M 30M 35M min log liv Input Spectrum max x 0 0000 Hz 39 961 MHz i Jitter Spectrum y 24 834 mv 553 19 mv m 2014 Stanford Research Systems 208 SR1 Operation Manual The two traces on the spectrum tab show the FFTs of the input signal and the jitter signal Each FFT is calculated using all the data in the input or jitter time records The resolution of these spectra is set on the Spectrum tab of the Digitizer panel Note that the FFT of the input signal shows the essentially square wave characteristic of the digital audio carrier signal while the spectrum of the jitter signal will depend on the the nature of the detected jitter If there are fewer jitter data points than FFT lines SR1 will zero pad the jitter data when computing the jitter FFT spectrum The jitter record length display on the digitizer panel will be shown in yellow to indicate that insufficient data
189. ep Finished Sweep Started Sweep Finished Sweep Started Sweep Finished Sweep Started Sweep Finished Sweep Started Sweep Finished Sweep Started Sweep Finished Events and Scripting If a script file is configured in the script column that script will be executed when the event occurs To associate a script with an event press the ellipsis button in the script column and use the file dialog to navigate to the appropriate script vbs file To clear a script from an event double click on the name of the script file For example we can write the simple script below which simply displays a message box with the text Sweep Done when it is run Scripting swp2 vbs BS ba S Fa ARDE 4 alt DI 1 z call 5El Instrument UserMessage Sweep Done 10 When this script is associated with the Sweep Finished event the panel looks like v No Tone swipe wbs No Event Now each time a sweep is finished the swp2 vbs script will run and the message Sweep Done will be displayed on the SR1 screen W Sweep Finished A Windows COM event may be fired in response to the event COM events allow local or remote programs to respond to the event including programs written in Visual Basic MS Office or SR1 s own local scripting environment Some events are associated with their own COM Event for example the Sweep Finished event can fire its own COM Event called unsurprisingly Sweep Finished Other events can f
190. er the value in the buffer is replaced This hold the maximum value in each bin and is useful for detecting unwanted transient events The Clear button clears the average buffer and re starts averaging The Avg Done indicator lights when the required number of averages have been accumulated The FFT2 Meas2 Panel This panel contains additional configuration controls for the FFT2 analyzer Measurement Meas2 References Trigger Window DC Correction Rife Vincent term Average ad Phase Spectrum Weighting Unwrap None fed Threshold Invert 120 00 dBFS Time Display Interpolation C OF f On Variable Time Window Limits start 0 0000 Yo stop 100 00 Yo Window Selection It is well known that the application of a window function is typically necessary to obtain maximum dynamic range for FFT measurements The discrete Fourier Transform calculated by SR1 calculates a spectrum assuming the time record repeats continuously Thus if the signal being analyzed is not perfectly repetitive in the time record interval the the calculated spectrum will include the discontinuities between the beginning and end of the time record which show up in the frequency domain as wide skirts and a high noise floor around the actual spectrum The first spectrum shown below is the unwindowed spectrum of two sinewaves of arbitrary frequency The spectrum is completely distorted by the artifacts associated with the time record discontinuitie
191. erpolates the user supplied table and normalizes to the peak interpolated value thereby insuring that the peak amplitude remains constant at different output rates Thus the following table of values in the file 0 0 1 0 2 0 3 0 4 0 5 produces a linear ramp from 0 to 0 2 FFS if the amplitude control is set to 0 2 FFS The Output Rate control governs how fast table points are output At 100 the output rate is 1 table point per output sample At 200 the generator outputs every other table point each output sample For fractional output rates the waveform is interpolated So at 50 Output Rate the generator outputs a table point then and interpolated point and then the next table point The arbitrary waveform generates a generator trigger each time the output returns to the beginning of the table 2014 Stanford Research Systems e SR1 Operation Manual FFT Chirp Config chirp Waveform On iw EQ Chirp Amp 1 0000 FFS Compliance SFE AL FFT 1 l The FFT Chirp waveform is designed to work in combination with one of SR1 s FFT Analyzers The chirp waveform provides a tone exactly at the bin center of each the FFT Analyzer s analysis bins In the default case each tone has equal amplitude however the Chirp waveform can be used with generator EQ to generate chirp signals with custom tailored frequency response Chirp signals are useful for quickly measuring the frequency response of a device under test
192. ershoot depending on the frequency and the selected D A converter 2014 Stanford Research Systems SR1 Operation 49 Ramp Config Ramp averorm On fw Frequency Frac RiseTime fi 00000 kHz 25 000 To Low Amplitude High Amplitude fo oo00 Vo 0 0000 Yp The ramp waveform consists of repetitive runs of integer numbers of rising and falling samples to produce triangle like output waveforms Because the runs are restricted to integer number of samples the Ramp Frequency and Ramp Fractional Rise Time have limited resolution which is a function of the selected generator sample rate The lowest amplitude sample has the value assigned in the Low Amplitude control The highest amplitude sample has the value given in the High Amplitude control y 1 o0 4 four 2 n m ee Ce Ss cc coc Sc cee See ec Sc te ek TE ee ee ee ee ee eee ere re Lene 0 500p 1 0m 1 5m 2 0m 2 5m 3 0m 3 5m i 1 kHz Ramp with Vmax 1 V min 1 25 RiseTime The ramp waveform generates a generator trigger each time the ramp begins its rising segment Arbitrary Waveform Config rb averorm On fw Arb CHIRP1K j 16 Ths Output Rate ji00 0 Yo The arbitrary waveform plays a sequence of values found in a user supplied table Arbitrary waveform files are simple ASCII files with one or more columns of floating point numbers representing the values of the 2014 Stanford Research Systems 50 SR
193. es Trigger Measurement Post Filter Level onami gt M a a Rate Response ain Auto Fast RMS Auto kai Motch BF Filter y 1 0000 kH a Tuned Biv Limit 1 Var LP lt 10 Hz Fs Ad Weighting amp Wk Ad H Filters Rate Rate Auto Fast Auto Fast The rate control sets the time interval over which the amplitude and levels are computed Six fixed rates are available from 1 sec to 32 sec the measurement interval is simply the inverse of the rate The Auto Fast and Auto Precise options compute an optimum measurement rate based on a specified frequency The frequency on which the rates are based is selected on the Meas 2 tab of the TDD panel Auto Meas Frequency SME Measured Frea E f Tuned Measured Freq Analog Sen 4 Analog Gen B Digital Gen 4 Digital Gen B Sweep 1 SMWIBED Z The frequency can be fixed or tuned to the measured input frequency any of the generator frequencies or the current sweep frequency When possible Sweep or Generator should be chosen as relying on the measured frequency while sweeping introduces a small but noticeable delay Using Measured Frequency on a signal with noticeable frequency jitter may also result in glitches in sweeps as the filter is constantly being moved to follow the input signal Response The amplitude measurement can be made with a choice of RMS Peak or Quasi Peak response RMS response calculates t
194. esearch Systems SR1 Operation 33 2 1 5 Print Setup File gt Print Setup displays the standard Windows print setup dialog box Print Setup ed bs Properties Status Type hp deskjet 5100 seres Where IP_17 2 25 96 16 Comment Paper Onentation SIZE Letter 8 5 s 11 in 7 f Portrait SOUICE Upper Tray C Landscape Hetwork Cancel When running the SR1 program from a Windows computer use whatever printers have already been installed When using the SR1 instrument use the Tools menu to connect SR1 to a network and install and configure network printers 2014 Stanford Research Systems SR1 Operation Manual 2 2 Edit Menu The Edit menu supplies the standard Windows editing functions Cut Copy Paste and Delete Cut Deletes the currently selected text and copies it to the clipboard Copies the currently selected text to the clipboard without deleting it Paste Copies the clipboard contents to the current cursor location Deletes the currently selected text without copying it to the clipboard Note that in addition to being useful for transferring text the edit commands may be used to transfer Graph Traces between different graphs 2014 Stanford Research Systems SR1 Operation 35 2 3 Panels Menu The Panels menu provides access to the various panels which control the operation of the instrument SR1 panels are fixed size windows they re not resizable They can
195. ess The Average Done indicator lights when the required number of averages has been accumulated Other Jitter Analyzer Controls Time Display Interpolation Co on Or Correction Mone 1 2 Pk Pk Turning on Time Domain Interpolation applies band limited interpolation to the time record of the Jitter Analyzer This is useful when looking at the time record display for jitter frequencies near the top of the range where there are only a few samples per cycle As an example the plots below show time record displays for a 100 mUI of 75 kHz sinusoidal jitter with and without time domain interpolation Oo 10 20H 30h 40H p 60H F SOU 90 SEC 75 kHz Sinusoidal Jitter Recorded without Interpolation 0 10u 20 30 40 Soy 60 7D BO 90u ace and with Time Domain Interpolation Selecting one of the DC Correction options either Average or 1 2 Pk Pk can reduce the DC offset in the each jitter time record and reduce the leakage from the DC bin in the spectra which can obscure low frequency details Since jitter has intrinsically no DC component the observed DC components are all due to measurement artifacts 2014 Stanford Research Systems SR1 Operation 183 2 4 11 Digitizer SR1 s optional Digitizer is a sophisticated tool for the analysis of digital audio carrier signals The Digitizer is designed to complement the real time measurement capabilities of the Jitter Analyzer with off line analysis capable of revealing additional sig
196. ether Analyzer Monitor Output Controls Two identical groups of controls control the AO and A1 Monitor outputs The monitor signals are only active if the corresponding analyzer is set to Time Domain Detector The AO Time Domain Detector outputs a realtime analog signal to the rear panel AO Monitor connector which corresponds to the input signal after the notch bandpass filter and any bandwidth limiting or weighting filters have been applied In a typical THD N measurement signal the monitor signal corresponds the distortion noise signal after the fundamental has been notched out Sending the monitor signal to the headphones speaker allows one to hear the distortion The Gain controls govern the scaling of the monitor output signals If the corresponding analyzer has an analog input signal then the gain is displayed in units of Volts Volt Volts output per Volts Input If the corresponding analyzer has a digital input signal then the gain is displayed in units of Volts FFS Volts output per FFS input In either case the maximum output signal from the monitor outputs is about 2 Vrms Checking Auto gain automatically sets the gain such that the monitor output signal remains near mid scale If the signal has significant time variation auto gain should be disabled so that the gain control doesn t chase the variation in the monitor signal The Monitor Level indicators display the current level of the monitor outputs Blue indicates the moni
197. external square wave reference signal Configuration of the clock reference is done on the Clock Reference panel R Video Reference In Use this input to lock the SR1 s internal clock to an external video NTSC PAL or SECAM signal Configuration of the clock reference is done on the Clock Reference panel S Ext Trigger In This TTL input triggers any analyzer configured to use External Trigger See Common Analyzer Features for a complete description of the SR1 s analyzer trigger modes Note that there is only one input for both analyzers 2014 Stanford Research Systems 296 SR1 Operation Manual 3 3 Specifications Analog Signal Generator General Characteristics Amplitude Range rms Amplitude Accuracy Frequency Range High BW DAC High Res DAC Frequency Accuracy Frequency Resolution Output Configuration Source Impedance Max Power 600 load Balanced Unbalanced Float Voltage Crosstalk 10 Hz to 20 kHz gt 20 kHz Waveforms 10 uV to 28 3 V balanced 10 uV to 14 1 V unbalanced 0 5 0 043 dB at 1 kHz 10 Hz to 200 kHz 10 Hz to 0 45 Fs Fs 128 kHz or 64 kHz fixed 24 kHz to 216 kHz adj 0 0005 5 ppm lt F 12 Balanced Ground Balanced Float Unbalanced Ground Unbalanced Float Common Mode Test 50 150 600 balanced 25 75 600 unbalanced 30 5 dBm 24 9 dBm 40 V 125 dB 100 dB Low Distortion Sine Hi BW DAC Flatness rel 1 kHz
198. f one were not interested in the new trace ID would be modified to Call SR1 Graph graphId LoadTrace C myTraceFile xml The same syntax applies to actions with no return value Call SR1 Displays Graph graphId AutoScalex Input and Output with Scripts SR1 scripts can interact with the user in a variety of ways The built in VBscript function MsgBox displays a message window containing a user specified string and an OK button For instance the line MsgBox MsgBox Argument produces the following output MsgBox Argument The message window remains up until the user presses either the OK button or the Close Window button The Instrument section of the SR1 Basic Interface contains several other functions designed to provide flexible input output from scripts UserM essage The UserMessage function is similar to the built in MsgBox function except that it contains an additional parameter which specifies a time out period in seconds If the user does not press the OK or Close Window boxes within the time out period the window will disappear and the function returns 1 If the user does take action within the time out period the function returns 1 For instance dim x x SRl Instrument UserMessage User Message 5 if x 1 then 2014 Stanford Research Systems SR1 Operation 247 Call SR1l Instrument UserMessage User Did Not Respond 10 end if first displays the message window shown below SR1
199. figuration saves the entire instrument setup to an SR1 configuration file Configuration files are XML files whose structure is detailed in the SR1 File Reference The default extension for configuration files is XML AIl operating parameters as well as the the position of all panels and displays are recorded in the configuration file After loading the file the instrument state will be exactly what it was when saved Save Instrument Save rr 4 config 46517 File name Gave Save as type KML file 1ml Cancel Trace Data Save IF Offline Always Save Save IF Offline The trace data option governs how SR1 will save data stored in graphs Never Save means that no graph data will be saved along with the configuration file The file in this case is a pure settings file Always Save means that all graph data including live measurements will be saved with the configuration file Save if Offline means that only offline graph data such as reference curves or limits will be saved with the configuration 2014 Stanford Research Systems 30 SR1 Operation Manual 2 1 2 Save Partial Configuration Selecting File gt Save Partial Configuration allows a choice of which portions of the instrument setup will be saved to the SR1 configuration file After selecting this option the Save Partial Configuration dialog box is diplayed Select Items i E A Instrument H Analog Generator E Digital Generat
200. ford Research Systems 100 SR1 Operation Manual 2 3 9 Monitors Panel 2151 eadphone Speaker wstem Sound Ly SOUrce Steren analog Gen O B ad W Auto f 0000 Wiw fi 0000 WIFFS The Monitor Panel controls a number of functions related to SR1 s speaker headphone output and the rear panel analyzer monitor outputs Several different signals can be routed to the speaker headphones Some of these sources a monophonic and are sent to both channels of the headphones and to the single speaker Stereo sources are routed to both channels of the headphone output and the left and right channels are summed and sent to the speaker The volume of both the headphone and speaker outputs is controlled from the front panel volume knob Headphone Speaker Source indicates stereo source Analog Gen A The speaker and both headphone channels are fed the current output of the analog generator A channel Analog Gen B The speaker and both headphone channels are fed the current output of the analog generator B channel Analog Gen A The left and right headphone channels are fed the A and B channels of the analog B generator the speaker is fed the summed A B signal Analog Level A The speaker and both headphone channels are fed the signal input on the analog A channel Analog Level B The speaker and both headphone channels are fed the signal input on the analog B channel Analog Level A The left and right
201. hardware Networking Customize the network environment Computer Functions Provides standard Windows computer options printers disk ejection etc 2014 Stanford Research Systems za SR1 Operation Manual 2 6 1 Preferences Panel Unlike most SR1 controls items selected on the preferences panel retain their value each time SR1 is started Preferences General Display Remote Startup Config Last Autosawed a Gubosave Interval F minutes Aubosave Mow Phase 180 00 lt Phase 180 00 Generator Signal Initialization Last Used Analog Generator Max Gukpuk 40 000 YP Knob amp keypad Knob Sound Knob Accel Cursor Pos iw Keypad Sound Startup Configuration The Startup Configuration parameter controls SR1 s behavior on startup SR1 Default means SR1 will startup in its fixed default configuration each time the instrument is turned on regardless of the configuration when the instrument was last turned off User Default means the instrument will load a user specified configuration file when the instrument is started Use the file selection button to select any SR1 configuration file to load at startup Last AutoSaved tells SR1 to use the last Auto Saved configuration file at startup SR1 periodically saves a configuration file at the interval specified by Autosave Interval If Last Autosaved is selected SR1 will load this configuration file ensuring th
202. he transisions found This range is then binned into the number of bins specified in the Y Resolution control The number of points falling into each bin is then mapped into a color indicating the relative probability of that particular voltage time point A description of the various color mapping options is given in the Digitizer Display section 12 Ul 200m 0 200m 400m 600m e00m 1 0 Eye Diagram of Carrier with 0 2UI Square Wave Jitter As an example consider the eye diagram above produced by the digitizer looking at SR1 s digital audio output with 200 mUI of added square wave jitter For square wave jitter the pulse edges will occur at one of two times corresponding to the high and low portion of the square wave jitter signal This is clearly evident in the eye diagram as the lines of high intensity dark green in this color mapping are separated by the 200 mUl jitter amplitude 2014 Stanford Research Systems SR1 Operation 191 2 5 Displays Menu SR1 offers three different types of displays which present visually the measurements made by the various analyzers The Displays menu offers options for creating and managing the different types of displays used by the instrument ph Opens a new Graph on the current page Bar Chart Opens a new Bar Chart on the current page Digitizer Display Opens a new digitizer display on the current page Other Display Menu Saving and Recalling of displays Options for managing the page co
203. he current input signal Note that the ranges are set to their minimum value 62 5 mVrms and the Input Level indicators are showing blue below half scale That s because the generator hasn t been set up yet 2014 Stanford Research Systems 20 SR1 Operation Manual Take a look at the Analog Generator panel We haven t selected any waveforms yet so the tab control at the bottom only contains the configuration tab By default the generator is in Mono mode which means that the same waveform will be output on the A and B channels We can adjust the channel amplitudes separately but the waveform is the same In Stereo mode we can select different waveforms for A and B but we don t need to do that for this example Now press the New button on the Analog Generator panel This brings up the menu of available waveforms From the Sine submenu choose Normal Sine After the Sine waveform shows up in the tab control set an amplitude of 1Vrms Note how the range controls on the Analog Input panel both move to the 1 Vrms input range and the Input Level indicators turn green indicating the ranges are optimally adjusted The Input Level indicators are also visible at the bottom right of the SR1 screen Now we can add a little bit of distortion to the signal Press the New button again on the Analog Generator panel and once again select a Normal Sine This time set the frequency to 2 kHz and the amplitude to 1 mVrms We ve just illustrated a key
204. he true root mean square RMS value of the amplitude signal over the 2014 Stanford Research Systems 120 SR1 Operation Manual measurement interval selected with the Rate control Peak response calculates the peak value of the amplitude signal over the measurement interval Quasi peak filters the amplitude signal with a dual time constant response to provide the dynamic characteristics specified in ITU R BS 468 Notch BP Filter ae Filter m Fixed als f Tuned The amplitude chain of the TDD contains a selectable notch bandpass filter For HiBandwidth Analog inputs there are actually two notch filters an analog notch which cuts the fundamental level enough so that the postfilter gain can amplify the resulting noise distortion signal to full scale on the amplitude ADC and a second notch implemented by the DSP processor which cuts the fundamental even further so that residual fundamental is not a measurable component of the final residual THD N specification The combined response of the two filters is shown below Response dB 0 5 0 6 0 7 0 8 0 9 1 1 1 1 2 1 3 1 4 1 5 Normalized Frequency Combined Notch for Analog HiBw Inputs Because of the wide dynamic range of the 24 bit hi resolution converter an analog filter is not necessary So for HiRes analog inputs and digital inputs a single notch filter is applied by the DSP 2014 Stanford Research Systems SR1 Operation 121 0 5 0 6 0 7 0 8 0 9 1 1 1 1 2 1 3 1
205. hich are unreachable with the current generator sampling the Compliance led will glow red and no waveform will be output 2014 Stanford Research Systems SR1 Operation Manual Be sure to select the Uniform window when using the log sine chirp Window functions attenuate the beginning and the end of the time record which means that some frequencies will be attenuated more than others by the window The advantages using log sine chirps for impulse response measurements are detailed by Miller and Massarini Transfer Function Measurements with Sweeps J Audio Eng Soc vol 49 pp 443 471 June 2001 They report better results using log sine stimulus compared to MLS stimulus for a wide range of audio DUTs One particular advantage concerns the unique ability of the log sine chirp to differentiate between the linear and non linear portions of the DUT response Because of the properties of the group delay of the log sine chirp waveform harmonic responses appear offset in time relative to the linear response This can be clearly seen in this impulse response measurement of a consumer stereo amplifier shown below Harmonic Response o E EE Ei A eae eee Pentre e joven commas Pien T Linear Response 50m 25m 0 25m Som 75m 100m sE When the purely linear response is gated and transformed back into the frequency domain a much better measurement of the DUT frequency response is obtained This is detailed in the FFT2 a
206. hosen measurement Most of the Quick Measurements can be run in either free run mode in which the measurement result is continuously displayed or in swept mode in which the measurement result is recorded as a function of one or two swept parameters All the necessary controls to configure sweeps and settling are on the Quick Measurement panel Sequences of multiple quick measurements can be programmed with the Automated Measurements panel Each of the selected measurements is performed sequentially and the results are included in a formatted report which can be printed or saved as a PDF or HTML file In the following sections each quick measurement will be described For each measurement the cabling requirements from SR1 to the EUT will be given followed by a description of the analyzer and generator configuration for that measurement Finally the displays created for each measurement in both free run and swept mode are detailed 2014 Stanford Research Systems 254 SR1 Operation Manual 2 8 1 Setup Panel Quick Setup Mutou Domain Analog Z Connector LR z Channels 2 Bandwidth 50 kHz x Input Domain Analog Connector IR l Bandwidth 50 kHz Coupling AT or Channels 2 3 Terminate W Gukoscale w asi The Setup Panel records basic input and output information for all subsequent quick measurements First select the domain analog or digital for the measurement input and outputs The input and outpu
207. ht of each entry is checked SR1 will increment the transmitted value by 1 each digital audio block The confidence flags are defined by the professional standard to indicate whether the information carried by the channel status data is reliable Like the validity flag their sense is reversed from the name i e setting the confidence flag to 1 indicates lack of confidence The grouping of the 4 confidence flags is 2014 Stanford Research Systems SR1 Operation s mirrored on the professional receive panel by 4 indicators which glow red if the corresponding flag is set and green if it is not The professional standard defines byte 23 as a cyclic redundancy check CRC character SR1 allows several options for the transmission of the CRC byte CRC controls the transmission of byte 23 The same byte is transmitted with every digital audio block The value of the byte is whatever was being transmitted when the CRC mode was set to satatic The correct CRC byte is sent with each digital audio block Incorrect A changing incorrect CRC byte is sent with each digital audio block Zero i The CRC byte is set to zero Received Status Bits The grouping of the received status display for both professional and consumer modes is shown above The topmost display shows the current mode alongside an indicator which glows red if the validity flag for that channel is set In professional mode CRC errors are not shown on the status panel but i
208. igital full scale leaving the negative maximum code unused The FFS Fraction Full Scale unit expresses the peak amplitude of any SR1 generator waveform relative to this definition of full scale amplitude When dither is on full scale amplitude is reduced symmetrically by 1 bit to allow for dither Small values of FFS may be expressed as mFFS milli FFS or uFFS microF FS Same as FFS above but expressed as a percentage of full scale amplitude 50 FS 0 5 FFS This digital waveform amplitude unit is directly convertible to FFS depending on the value of the V FS Volts Full Scale set in the analyzer references tab If the value of V FS is 2 Vrms for instance then an amplitude of 1 Vrms corresponds to an amplitude of 0 5 FFS These units allow digital amplitudes to be expressed as analog voltages as is often useful when working with A D converters where the V FS value can be thought of as the 2014 Stanford Research Systems SR1 Operation Manual ADC s full scale inpu voltage This unit of digital waveform amplitude is convertible to FFS depending on the value of the V FS Volts Full Scale set in the analyzer references tab and the fixed relationship of peak voltage to rms voltage for a sinewave If the value of V FS is 1 Vrms for instance then an amplitude of 1 414 Vp would correspond to an amplitude of 1 Vrms which in turn would correspond to an amplitude of 1 FFS This unit of digital waveform amplitude is convertible t
209. igital input and output settings to match the equipment being tested FFT This setup configures SR1 for a basic audio band FFT The Hi resolution converter is selected for maximum dynamic range The AO analyzer is configured as a dual channel FFT analyzer with a span of 28 8 kHz and continuous averaging The Analog Generator is loaded with a low distortion sine waveform set to a default frequency of 1 kHz and an amplitude of 1 FFS Page2 contains a graph displaying the A and B channel power spectra while Page 3 contains a graph displaying the A and B channel time records What you ll need to configure If you re using the digital generator adjust the amplitude and frequency of the signal to match your requirements Otherwise you can turn the generator off The amount of averaging in the FFT analyzer can be adjusted to suit the noise level of your signal The FFT resolution is defaulted to 2k points this can be adjusted Frequency Response This setup configures SR1 for a stereo audio band frequency response sweep The Time Domain Detector is selected for the AO analyzer The Digital Generator is loaded with a sine waveform Page2 contains a graph displaying the A and B channel frequency response The sweep controller is programmed to sweep the sine frequency from 20 Hz to 20 kHz in 22 logarithmic steps and record the A and B channel levels What you ll need to configure Adjust the the sine amplitude to match the equipment being tested The
210. ilding blocks of the generator output will be described in detail Config 40 Sine aveform n fw EQT ine Arig 0 0000 WETS Freq fi 00000 kHz Sine Waveform Tab When a waveform is added to the generator output using the Waveform Controls the associated waveform tab shows up on the generator panel For instance the Sine Waveform tab is shown above If the Generator is in the stereo output mode Separate waveforms for each channel the tab title contains a channel designator indicating which channel the waveform is associated with If the generator is in mono output mode the channel designator is omitted and the title contains just the name of the waveform some of the controls found on the waveform tabs are common to many waveforms These will be described first to avoid repetition Controls Common to Most Waveforms aveform On fw EO The Waveform On checkbox turns the selected waveform on and off If the selected waveform is currently the only waveform in the generator checking and unchecking this box has the same effect as turning on and off the generator using the Amplitude Controls or by simply setting the waveform amplitude to zero and back When the generator is outputting a combined waveform this checkbox allows the selected waveform to be toggled on and off while still outputting the remainder of the waveforms The EQ checkbox appears for only certain waveforms If EQ is checked and an EQ file is selected on the
211. ile floating point array data is transmitted and received as arrays of 64 bit double precision data Big Endian Binary The IEEE 488 standard specifies that binary data is sent in Big Endian fromat i e most significant byte first However most PCs natively store floating point data in memory in little endian format i e least significant byte first This would normally then require byte swapping before sending or receiving binary data Un checking this box instructs SR1 to send and receive binary arrays in little endian format This option has no effect unless Binary Array In Out is selected Input Options Ignore Case Checking this box instructs SR1 to ignore the case of incoming remote commands as specified in the IEEE 488 2 standard When un checked all commands must be sent with the exact case specified in the SR1 Remote Programming Reference Parse Absolute When checked SR1 requires that a full path specifier be sent with each command Un checked the object referenced by the previous command will be considered the default for the next command For instance the command AnlgGen AGenChA Gain 0 45 sets the gain of the Analog Generator channel A to 45 When Parse Absolute is selected this command could be followed for instance by the command AnlgGen AGenChA On 1 to turn on the generator channel Un checking parse absolute means that the second command could be abbreviated as On 1 All commands subsequen
212. ing parameters can all be thought of as desired values rather than guaranteed values because SR1 has no control over the external sweep source However 2014 Stanford Research Systems SR1 Operation 95 SR1 monitors the swept measurement and attempts to make the actual sweep X axis correspond as closely as possible to these values The spacing of the steps in an external sweep can either be specified as a percentage Relative Value checked or as an absolute value Relative Value unchecked It can be useful to condition the external sweep s search for a new sweep point on the value of another measurement For instance if both the amplitude and frequency of the external source varies there may be no point in searching for a new frequency sweep point if the amplitude is zero SR1 handles this situation by allowing the setting of a Minimum Level measurement Click the Minimum Level Meas button to open a window allowing the selection of any SR1 measurement to act as a qualifier for external sweeps The analyzer will then need to obtain a settled value of the selected measurement that is greater than the specified minimum value before finding a new value of the actual sweep parameter The External Settings tab on the Sweep Panel contains a number of additional controls which govern the starting and stopping of external sweeps Inner Sweep l Outer Sweep External Settings Tolerance Start 5 0000 Ys W Relative Value Stop 5 o0
213. int or unit ed quantities Integer quantities are read and set as follows dim intVal intVal SR1 AnlgInputs HiResSampleRate read the hi res sample rate SR1 AnlgIinputs HiResSampleRate intVal set the hi res sample rate Some integer properties have enumerations associated with them Enumerations are short mnemonic text strings which can be used in place of integer values Any enumerations associated with a property are shown underneath the property with the icon So for instance the two lines below are interchangeable SR1 AnlgIinputs HiResSampleRate 1 set to an explicit integer value SR1 AnlgIinputs HiResSampleRate srHz128k set to the equivalent enumeration These enumerations exist in the SR1 internal scripting environment To specifiy these enumerations in VisualBasic MS Office or other scripting environments they must be referred to as SR1 enumtype enum e g SR1 HRSR srHz128k Enumeration types are found in the file SR1 tlb SR1 type library Unit ed properties require an additional argument to specify the units the property should be expressed in For instance x SR1 AnlgGen AGenChA Gain db assigns to the variable x the value of the Analog Generator channel A gain in units of db To set the corresponding property SR1 AnlgGen AGenChA Gain db 10 0 To read the value of a unit ed property in current units simply use an empty string for the unit specifier x SR1 AnlgGen AGenChA Gain Un
214. internally based on FFT analysis levels displayed are peak levels Frequency is calculated for analog but not digital audio inputs Phase is not calculated with this analyzer 2014 Stanford Research Systems 162 SR1 Operation Manual The IMD Analyzer Panel A1 IMD Analyzer SUF Ce Converter Fs Measurement References Trigger Generator IMD Product Analog dd z Averaging Speed very Fast _ Synchronous Awg Weighting Generator Selection Generator Analog ka The Generator control selects the generator that will be used for the test Typically the analog generator will be selected with analog inputs and vice versa however cross domain measurements are possible on ADC and DAC devices by selecting a different generator The selected generator must be using the IMD waveform for the analyzer to function properly The choice of generator waveform SMTPE CCIF or DIM dictates the analyzer configuration and is displayed in the readout below the generator selection IMD Product Selection IMO Product dd 2 E CETE For each type of IMD Measurements the IMD Product control determines the particular IMD products that will be included in the measurement The table below summarizes the available choices 2014 Stanford Research Systems SR1 Operation 163 SMPTE DIN f1 low frequency f2 high frequency do fotf U2 amplitude of high freq component 2 dy fpt3 f d2
215. ion IDs Note that scalar measurements like frequency or level are denoted with the EH icon while vector measurements like spectra are accompanied by the Lal icon Select Power Spectrum click ok and the FFT will appear Since we re looking at a very wide dynamic range some logarithmic units will help Click on the Ymax Unit Entry and change the Y units to dBVrms Now click the Log box in the X axis row to select a logarithmic X axis Click on the I icon autoscale on the graph speedbar to id autoscale the display Now turn on the graph cursors by clicking the ie icon Use the knob or the mouse to drag cursors as shown below The graph should look like this 2014 Stanford Research Systems Getting Started K x ADH HOM 1t AARAA MEA AY AmE 2 0000 EES 2 00 E A ie Bp sicss Joo Bd Scale Cursors FFT Cursors X Axis Min Max Log de X 31 250 Hz _x 31 969kHz M B Y 167 36 dBVrms _ 54 892 dBVrms aa z Lock Axes E pe r Now we re ready to zoom in on a portion of the the distortion product spectrum Go back to Page 1 of the page control and in the bandwidth control of the FFT analyzer select 500 Hz In the Center Frequency field select 1 KHz We re narrowing the bandwidth by a factor of 64 still using 1k FFT lines and centering the new bandwidth on the notched out fundamental Go back to Page 2 autoscale the spectrum which should now look like this 2014 St
216. ion of SR1 s XLR and BNC analog inputs 2101 x SS SS SS SS Analog Inputs Hi Res Converter Optional Filters hA ange Iv Suto 1 000 Yrms Input Config BNC hiz DC w Input Lvl hE ange W Auto ji o00 rms Input Config The first tab on the panel Analog Inputs contains most of the configuration controls Note that Channel A and Channel B analog inputs can be configured separately The Range box contains a checkbox to enable autoranging of the corresponding input In general autoranging should be left on however for certain low frequencies or slowly varying input signals it can be useful to turn off autoranging and set the range manually to avoid unnecessary range jumping The Input Range control displays the current input range when autoranging is on and allows entry of the range in Vrms or dBVrms when autoranging is off The smallest input range allowed by SR1 is 62 5 mVrms the largest is 160 Vrms In the Input Configuration box is the control to set the input connector Input Configuration Connector Select the BNC connector for the corresponding channel Do not apply balanced inputs to the BNC connectors as the BNC outer conductor is connected to analyzer ground Select the XLR connector for the corresponding channel The impedance control allows the input impedance to be varied between 3000 6000 and Hi Z 100 kQ It is possible to use unbalanced inputs with the XLR connectors
217. ions SAA File Edit view Favorites Tools Advanced Help ae as ie Fa T r E i a Pi Search Folders k 3 x e Address Network Connections LAN or High Speed Internet Local Area Connection Connected Firewalled a a Realtek RTL8139 Family PCI F Wizard New Connection Wizard The connection which appears in the LAN or High Speed Internet section corresponds to SR1 s rear panel ethernet connector To configure this connection right click the Local Area Connection icon and select Properties Scroll down the list of items which appears 2014 Stanford Research Systems 232 srt Operation Manual El os Packet Scheduler Network Monitor Driver Internet Protocol TCP IP lt till to Internet Protocol and right click on Internet Protocol and then click the Properties button below the list General lk E ou can get F settings assigned automatically if your network supports this capability Othermize you need to ask your network administrator for the appropriate IP settings O Obtain an IF address automatically Use the following IP address IF address Tf2 25 128 16 Subnet mask 255 255 0 OU Default gateway Tf 5 0 7 Use the following ONS server addresses Prefered DHS server Tf2 2 9 2 Alternate DNS server i2 25 9b GF For networks supporting automatic assignment of IP addresses select Obtain an IP address aut
218. ions which can be performed A complete description of the SR1 Basic functions can be found in the SR1 Basic Programming Reference A good recent introduction to both scripting languages can be found in William R Stanek s Microsoft PowerShell VBScript amp JScript Bible published by Wiley All examples in this manual will be shown in VBscript Scripting sr10 test vbs Py 2 ARR D G ej ahh Bl j loop UserChoice UserchoiceMulti do while isEmpty 5N UserInput SN inputhox Please input serial number switcher box UserLaunchchoice if SNe then UserLoadFile if ismmeric 5N then UserMessage oM fixi 5N Message exit do Timeout end if fi Result end if UserokCancel SHN empt Y UsersayveFile loop Uservesho Cancel aW cstr 1x 5N Version end sub eh Wy ait Message ByVal Message As String 41 load pre sared setting for amp rs fred 42 sub load mpysFresettings i VBScript 10 43 0 U Frojectstaudioswitcheriswitcher testing scriptisr10 test vbs The scripting window has several sections At the top is a speedbar providing fast access to commonly used functions The main window area is used for actually writing the script Note the line numbers at the left The two windows to the right show the functions available in the SR1 Basic interface along with a description of function arguments At the bottom a status bar shows the currently selected scripting language the script timeout setting
219. ipting Opens the script logging window on the active page of the page OpenScriptLogForm control Opens a script logging window and returns a formID for the opened window See the SR1 Basic manual for more information on formIDs SR1 Scripting OpenScriptLogFormwID C1 Scripting Closes the log form with the specified formID Close ScriptLogForm formld SR1 Scripting Closes all script logging windows on all pages of the page control Close ScriptLogForms SR1 Scripting ClearLines Clears the contents of the scripting window SR1 Scripting WriteLine Writes the string specified by the argument to the scripting window Text SR1 Scripting PrintLog Prints the contents of the log Using the Bar Chart Display in Scripts Scripts can use the Bar Chart Display to output messages to users in scripts using the WriteMsg function The Bar Chart display must be offline i e not displaying a live measurement to do this dim bearid barId SR1 Displays NewBar Call SR1 Displays Bar barId GoOffline Call SR1 Displays Bar barId WriteMsg rHello The color of the display can be optionally changed by including the escape sequence followed by 2014 Stanford Research Systems SR1 Operation 251 either r red g green or y yellow at the beginning of the string These characters are not displayed they only indicate the color of the text Vicia HX 0m 400m 600m S Terminating Scripts In gener
220. ire one of 5 generic COM Events labeled Event1 through Event5 To set up a subroutine in a SR1 s scripting environment which traps a COM Event navigate the tree in the right hand panel in the scripting window to the Events node and double click the event to be trapped A subroutine declaration is started in the script window which marks the routine that will be called when the COM event is fired For instance in the script below 2014 Stanford Research Systems 224 SR1 Operation Manual Scripting swpdone vbs G EB Pa Sy 9S BR ER D C 4 alt S Dei Call SRL Sweep tarti Switcher Events o BarLimitExceeded EveLimitExceeded a GaraphLimitExceeded o OnCriticalError m Onkeypad o Onknob w OnScriptError o OnSweepFinished sub Events OUniweepFinished Call 5R1 Instrument UserMessage Sweep Done 10 end sub Jnn bk Ww hm Fe F E the main program simply starts a sweep The subroutine Events OnSweepFinished was created by double clicking the OnSweepFinished node in the tree in the right hand panel It traps the Sweep Finished COM event The event subroutine continues to trap events until the Hj button on the scipting panel is pressed To enable firing of the Sweep Finished COM event configure the event panel as shown below Iv Sweep Finished iw No Tone Sweep Finished More details on scripting can be found in the scripting section of this manual Using Events
221. isplays Recalls displays saved with either the Save All Displays main menu selection or with the Save button at the top of the graph speedbar Recalled displays are placed on the same page and position on the page control that they were originally on when saved Delete all Displays Deletes all displays Graphs Bar Charts and Digitizer Displays from all tabs of the page control Goto Page This options selects a new active page on the page control Select a page from the submenu and the active page will be changed The same action can be accomplished by simply clicking on the desired page tab at the right of the SR1 screen Move Form To Page Moves the active panel or graph on the current page to the page selected from the submenu Tile Forms Moves all panels and displays on the active page of the page control to minimize overlap between windows 2014 Stanford Research Systems SR1 Operation 213 2 6 Tools Menu The Tools menu provides access to options for customizing the operation and appearance of SR1 integrating SR1 into a network environment and accessing standard Windows computer functionality The Tools menu options are Preferences Panel Opens the SR1 User Preferences panel Events Panel Opens the SR1 Events panel Links SR1 events with user configurable actions Switcher Opens the Switcher Configuration panel Configure networks of I O Configuration switchers Hardware Info Display the status of SR1 s
222. istribution such as log prime or linear prime so that harmonics of different tones do not fall on the same frequencies The vector of the ratios of the sum of the harmonics amplitudes for each tone to the received amplitude at the tone frequency For this measurement it useful to THD vs Freq A B select a tone frequency distribution such as log prime or linear prime so that harmonics of different tones do not fall on the same frequencies The vector of the ratios of the sum of all IMD products involving a given tone to the IMD vs Freq A B received tone amplitude at the tone frequency This measurement is only active when the stereo mode is enabled on the Multitone Configuration Panel since it relies on tones being present at different Crosstalk A B frequencies for the two channels The Crosstalk vector is the ratio of received amplitudes in bins that do not have tones in the measurement channel to the amplitudes of corresponding bins in the other channel that do have tones 2014 Stanford Research Systems SR1 Operation 167 Freq Response The vector of the ratios of the B channel rmagnitude response to the A channel Magnitude magnitude response Ratio B A Freq Response The vector of the differences between the B channel phase and the A channel Phase phase Delta B A Scalar Measurements All scalar measurements are taken between the frequency limits specified on the Scalar Measurements tab of the Multiton
223. it ed properties typically have an associated property which reads the current units of the property For instance stringvar SR1 AnlgGen AGenChA GainUnit assigns to the variable stringvar the unit string for the Gain property When setting a unit ed property the specified units become the current units For instance SR1 AnlgGen AGenChA Gain S 50 0 sets percent as the current units of the gain property An object can be assigned to a variable using the set keyword 2014 Stanford Research Systems 246 SR1 Operation Manual set chna SR1 AnlgGen AGenChA chna Gain db 10 0 x chna FregqRdg Actions Actions are functions associated with objects which can return values and take parameters Parameters are shown underneath the associated action with the P icon Return values from actions are denoted by the icon Integer parameters like properties may have enumerations which are listed underneath the parameter The syntax for using actions in scripts is similar to that for properties For instance the action LoadTrace associated with a graph takes a string specifying a trace file and returns an integer which is the id of the new trace In a script intMyTraceId SR1 Graph graphId LoadTrace C myTraceFile xml One peculiarity of the syntax is the necessity to use the keyword Call for actions that have no return values or when the return value is not assigned to a variable So the example above i
224. itch See the SR12 manuals for details The number 2014 Stanford Research Systems SR1 Operation 227 of switched channels 12 and destination channels 2 is currently fixed these fields are included for future operation with other switches The next set of controls specifies how SR1 will control the switchers via the serial port RS 232 or over the ethernet network If Serial communications is selected then a serial cable must be connected from SR1 s rear panel serial port to the switcher s serial connector The COM port control will display the available COM port on SR1 s internal computer no user adjustment of this field is necessary The Chain Address identifies individual switches in a daisy chain topology The chain address entered on the New Switch panel must match the chain address set on the switcher with the rear panel DIP switch settings In this configuration the serial cable is connected from SR1 to the Serial In port on the first switcher The Serial Out port of the first switcher is connected to the Serial In port of the second switcher and so on Commands are sent from SR1 to the first switch which retransmits the command down the daisy chain until it received by the switch whose chain address matches the destination of the command See the SR10 11 12 manual for details on how to set the chain address and how to setup switches in a daisy chain configuration If TCP IP operation over ethernet is selected then the
225. ith the zero crossings of the sinewave In addition to the standard amplitude and frequency controls the Synchronous Burst Sine tab contains several controls which govern the burst options Burst Type selects the burst triggering mode The sine output alternates between the hi and lo amplitudes Both the hi and lo intervals are integer numbers of cycles of the sine The total period is determined by the Burst Rep Rate control The Hi amplitude interval is determined by the Burst On Time control The waveform amplitude is determined by the TTL gating signal applied to the rear panel TTL burst trigger input When the external signal is high the generator output is set to the high amplitude when the external signal is lo the generator switches to the lo amplitude Amplitude switching is performed synchronous with the sine zero crossings regardless of the actual moment at which the external gating signal switches Ext The output is at the lo amplitude until a TTL rising edge is detected at the rear panel Triggered burst trigger input At the next zero crossing the output then goes high for the number of cycles specified by Burst On Time Burst Repetition Rate is only valid for internal bursts It sets the total on off period of the burst cycle The rate may be entered as a number of cycles a frequency or a time interval In the latter two cases the value will be rounded to the nearest integer number of cycles Burst On Time For inter
226. k 20k 25k 30k Hz Comparison of Averaged Power Spectrum and Linear Spectrum 2014 Stanford Research Systems 130 SR1 Operation Manual Summary of FFT1 Analyzer Outputs Measurement Description Time Record The underlying time data used to compute spectra When displayed on a graph this measurement produces an oscilloscope type display Power Spectrum The amplitude of the power averaged spectrum Amplitude Linear Spectrum The amplitude of the synchronously averaged spectrum The amplitude of signals Amplitude that are synchronous with the time record is preserved other uncorrelated Signals average away Linear Spectrum The phase of the synchronously averaged spectrum Tr ae Level B A B Peak based level Peak based level computation for both input channels 0 for both input channels The FFT1 Analyzer Panel 5 AO FFT Analyzer Source Converter mnoga imow ME Measurement Meas References Trigger Bandwidth Resolution Acq Timet _ Baseband ik 4 00 msec 256 00 kHz iw Show Aliased Lines Stark Center End o 0000 Hz z 128 00 kHz oF 256 00 kHz oF a a mj a ae Averagin a Aygs Clear 4g Done Exponential 1 Ej The FFT1 panel contains the normal input source selection controls and level indicators common to all analyzers However a few items unique to the FFI1 analyzer are worth noting Unlike the RMS levels computed by the Time Domain Detector levels compu
227. ke this 25 Analyzer Analog Inputs aaa i Analog Generator Analog Inputs Hi Res Converter Optional Filters Waveform Source Fs sa i Res Conver ptonal Fi rs New Fs Bum TTE a T Digital gt InputLvl Pange Delete Mode Mono ki 250hms l W Auto 62 50 mYrms Input Config BNC DC Ch B Input Lvl Input Config BNC T Hi Fa T DC Ch A Ch B 100 0 A Lock 100 0 0 0000 vems nei Invt Invt M EE a Config References Burst dEr Ref 1 0000 Vrms _ ModelOfF gt l Freg Ref 1 00000 kHz Lo Ampl Watts Ref 8 0000 ohms Period gt dBm Ref 600 00 ohms Dty Cy _ Trg EQ None T fal InvertEQ Measurement Meas 2 References Trigger Measurement Post Filter aT Level fanoitde hes Rate Response Gain Auto fast bd RMS bd Notch BP Filter C Fined None Measured Freg f Tuned BW Limit y lt 10Hz _ 24kHz Fs 2 _ 0 Weighting None h HW Filters Now let s configure the instrument for this example On the Analog Inputs panel change the Input Source for both channels from BNC to GenMon With the inputs set to GenMon the Ch A analog input is directly connected to the Ch A generator output and the Ch B analog input is directly connected to the Ch B generator output Leave the Auto box checked for each of the range controls With Auto Range on SR1 automatically adjusts the range to the optimum value for t
228. l all required files to update the firmware If the instrument is not directly connected to the internet the update files can be downloaded separately placed on a USB drive and then executed To download the required installer file use a browser to access the URL hitp sr1update thinksrs com The server will display a list of patch files 2014 Stanford Research Systems 24 SR1 Operation Manual SRI Audio Analyzer Software Updates Verson 1 0 14 0 patch download 12 1MB 10 30 4009 This paick updates ali earlier versions of SAS fo version ILO 4 04 Version 1 0 20 0 patch download 10 4MB 12 14 2009 This paick updates version JL 4 O of SAS and later fo version i0200 Verson 1 0 40 0 demo installer download 40 4MB 1414 2009 For other SES products please visit our main site Note that separate installers are used to patch the instrument software and install the demo software on a PC The currently installed firmware version can be determined by selecting About SR1 from the Help menu After determining whether a more recent firmware version exists download the appropriate patch from the website and place the file on a USB drive connected to SR1 To execute the patch open a Windows explorer window and navigate to the USB drive with the patch file Double click on the patch file to begin the update process A windows explorer window can be opened while SR1 is running by using a an external keyboard and pressing Ae
229. l be simply Analog or Digital Some FFT based analyzers Single Channel FFT THD IMD can also use as input the output of the time domain detector analyzer allowing these analyzers to examine the post notch filtered noise and distortion signal Converter For analog signals SR1 offers a choice of two analog to digital converters each optimized for different measurements The high bandwidth Hi BW converter is a 16 bit converter operating at a fixed sampling rate of 512 kHz The high resolution Hi Res converter is a 24 bit converter which operates at fixed sampling rates of 128 kHz and 64 kHz and variable rates which can be synchronized with the digital audio output or input signal to perform cross domain measurements Selection of the Hi Res converter Sampling rate is made on the Analog Inputs Panel Digital Audio inputs do not involve a choice of converter and the converter field is fixed at Dig Aud Fs The current analyzer sampling rate is displayed in the Fs field in the upper right of the analyzer panel The sampling rate is dependent on the source converter combination For analog inputs the sampling rate is dependent on the converter selection as described above For digital audio inputs the sampling rate depends on the sampling rate embedded in the digital audio input signal Note that for dual connector digital audio inputs the sampling rate shown is the logical sampling rate representing the actual spacing of the digital a
230. larity 2014 Stanford Research Systems s SR1 Operation Manual Output Sample Rates can be chosen for single connector outputs between 24 kHz and 216 kHz For dual connector outputs the range is 54 kHz to 216 kHz Note that for dual connector outputs the chosen sample rate is the effective sample rate of the two digital audio streams not the physical frame rate for each connector The pre emphasis control is included for future expansion In the current version of SR1 pre emphasis is not implemented and is fixed at None Output resolution can be set between 8 and 24 bits The embedded digital audio signal is truncated to the specified number of bits after any dither selected in the Digital Generator panel is applied Input Configuration Controls Select the input connector from one of the following Digital Audio Input Connector Selects the balanced XLR connectors as the input source BNC Selects the unabalanced BNC connectors as the input source Optical Selects the TOSLINK optical connector as the input source GenMon The digital audio input is connected directly to the output of the digital generator The Dual connector checkbox selects dual connector input mode In dual connector mode SR1 expects two digital audio data streams left on C1 right on C2 on the selected connectors either XLR or BNC with each frame containing 2 successive samples of the same channel If the input is set to GenMon then the input d
231. le channel FFT analyzer FF 11 The Jitter signal is first optionally decimated to reduce the bandwidth and increase the frequency resolution The output of the decimator is sent to a buffer which stores the incoming data until an analyzer trigger is received Upon receipt of a trigger the time record data is multiplied by a window function which is necessary to attain good dynamic range in the FFT spectrum After windowing the DSP computes the FFT of the jitter time record For an FFT resolution of N lines 2N real time record points are needed Each FFT is then averaged in two different ways The Power Spectrum is computed by computing the power for each spectrum taking the absolute value of the 2014 Stanford Research Systems 178 SR1 Operation Manual complex FFT points and averaging that power into the power computed for previous FFTs This type of averaging does not reduce the noise floor of the spectrum but it does reduce the variation of the noise floor making it easier to see spectral details on the order of the noise amplitude The Jitter Analyzer also computes the Linear Spectrum The Linear Spectrum is computed by averaging the real and imaginary parts of each jitter FFT separately The average of the real and imaginary parts are then used to compute the Linear Spectrum amplitude and phase In the Linear Spectrum unlike the Power Spectrum noise that is uncorrelated to the signal is actually reduced by further averaging Because of
232. lear Fast C Fast precise VErY precise The THD analyzer allows a tradeoff between measurement speed and measurement precision Internally this is accomplished by varying both the number of FFT averages performed and the resolution of the FFT spectra When using the precise and very precise settings sweep speeds will be noticeably slower but the measurement results will exhibit less variability Enabling Synchronous Averaging can sometimes be useful when the harmonic amplitudes are close to the noise floor When Synchronous Averaging is enabled the THD analyzer uses the averaged Linear Spectrum to compute the harmonic amplitudes As discussed in the FFT Analyzer section averaging the linear spectrum reduces the amplitude of uncorrelated noise which allows a more accurate measurement of harmonic amplitudes which are phase correlated to the fundamental and therefore maintain their amplitude in the linear spectrum Pressing Clear clears the current average buffer This is useful for reducing the transients caused by 2014 Stanford Research Systems SR1 Operation 159 for instance changing input ranges Weighting Weighting Aweighting sd When summing the harmonic amplitudes the analyzer can apply any of the standard weighing filters to the individual harmonic amplitudes The table below lists the available weighting filter and their typical applications Weighting Filters None No weighting filter is applie
233. letes the active trace from the graph Right clicking this button displays a submenu allowing fast deletion of all traces all unused traces traces that are unchecked in the trace listing or all sweep traces Saves the graph to a file The graph can be recalled using the Load Displays option from the Displays selection of the main menu Exports the graph to one of several file formats Graphical file formats include JPEG jpg bitmap omp and enhanced windows metafile emf Saving the graph to text txt format saves a comma delimited listing of each trace in X Y format Prints the graph After pressing the button a print preview is displayed The user can select among the installed printers and then press the print button to finalize printing Graphs are always printed in landscape mode Autoscales the active trace The first button autoscales only the X axis the second only the Y axis The third button autoscales both the X and Y axes t Translates the active trace right left up and down The direction of translation is adjustable on the Preferences panel Zooms the active trace in and out in the Xand Y direction Ca Toggles the graph area between the standard display and a larger graph which covers the 2014 Stanford Research Systems 196 SR1 Operation Manual 4 scaling controls and trace listing Wf Toggles the cursors on and of Toggles the cursors on and of the cursors on and off se Moves
234. ling of these Signals is adjustable using the Monitor Panel These signals can also be routed to the SR1 s internal speaker or to the headphone output N AES Reference Out This AES 11 2003 compatible digital audio reference signal is generated at the frequency specified on the Digital I O panel O Master Clock Out The master clock signal is a TTL squarewave whose frequency is related to the digital audio output sampling frequency OSR as follows OSR Master Clock Frequency 24 kHz 56 OSR x 512 kHz 56 kHz 108 OSR x 256 kHz 108 kHz 216 OSR x128 2014 Stanford Research Systems SR1 Reference 29 kHz If Master Clock Jitter is enabled on the Digital I O panel than the master clock output will be jittered as specified in the Output Impairment tab of the Digital I O panel If Master Clock Jitter is not selected the master clock output will reflect the nominal Output Sample Rate regardless of any jitter specified on the Digital I O panel P Frame Sync Out This TTL squarewave output signal is high when the digital audio output is outputting the first subframe of each digital audio sample and lo when the output is outputting the second subframe of the sample Thus it is a sSquarewave with the same frequency as the digital audio sampling frequency This signal is unaffected by any jitter selected on the impairment tab of the Digital I O panel Q Ext Ref In Use this input to lock the SR1 s internal clock to an
235. listing Each trace is represented by a line in the listing such as O0 A1 FFT Power Spectrum A The color of the line in the trace listing corresponds to the color of the graphed data The checkbox at the left allows each trace to be separately turned on and off in the graph The listing shows the source of the measurement in this case analyzer A1 which is configured as an FFT analyzer and the measurement in this case Power Spectrum A When a trace is offline as opposed to live the trace listing display is in italics e g O0 A1 FFT Power Spectrum A To add a trace to the graph press the button at the top left of the graph speedbar After pressing the Add Measurement form appears containing a tree listing all currently available measurements Double clicking on a measurement creates a trace in the graph corresponding to the selected measurement 2014 Stanford Research Systems 194 SR1 Operation Manual Add Measurement 40 Time Dom Det AL FFT Level 4 Level B MN Time Record A RN Power Spectrum A MN Linear Magnitude 4 MN Linear Phase 4 Analog In Dig4ud In Sweep OK Cancel Show Automation IDs Add Measurement Form The first two nodes of the tree represent the measurements available from the two analyzers AO and A1 The second two nodes of the tree contain measurements related to generic properties of the corresponding input signal level frequency and phase The fi
236. log to Digital converters each optimized for different measurements The high bandwidth Hi BW 16 bit ADC operates at a fixed sampling rate of 512 kHz The high resolution Hi Res ADC is a 24 bit converter which operates at fixed sampling rates of 128 kHz and 64 kHz and and variable rates which can be synchronized with the digital audio output or input signal to perform cross domain measurements Sweeps and Free Run SR1 operates in two different measurement modes free run and sweep In free run mode the analyzers make continuous measurements and continually updates the measurement results on the analyzer panels Ihe second mode sweep mode requires that several options be set First a sweep source must be configured which determines whether SR1 will be sweeping an internal parameter e g generator frequency over a specified range an externally measured parameter e g input frequency or at regular time intervals Once the sweep source is configured the user must select the measurements up to 6 that will be recorded during the sweep Swept measurements must be settled to be included in the sweep meaning that the transient variability of the measurement must drop below a user set value before being included in the sweep result Sweep sources and data are setup on the Sweep Panel while Settling parameters are set in the Settling panel Displays Three different types of displays are available for graphically displaying measurement data The g
237. lue ewes in the grap this key toggles the active cursor between the two cursors keys Alt Function Keys Menu Moves focus to the main menu Once Menu is pressed the knob and enter keys can be used to access the main menu selections without using a pointing device Sticky Normally the knob varies whatever control on the screen currently has focus Press Sticky to Stick the knob to the current control Even when focus is moved to a different control the knob continues to modify the sticky control The sticky control is drawn with a yellow background Press sticky again to 2014 Stanford Research Systems Getting Started pT exit stickymode S Upon receipt of a command from one of the remote interfaces SR1 is placed in Remote mode Using this key returns the unit to local control AutoReference The contents of the currently focused control is examined If it is a frequency the frequency is moved to the frequency reference of the selected analyzer The analyzer AO or A1 is selected by pressing the lt Lo keys after pressing Ref If the contents of the currently focused window is not a frequency then the contents of the A and B levels for the selected analyzer will be transferred to the dBrA and dBrB references either analog or digital depending on the analyzer input Hex Hex Entry mode Places a Ox character in the currently focused control to Hex Digits A F begin hexadecimal entry In this mode the A B
238. lyzer is configured as a dual channel FFT analyzer with a span of 28 8 kHz and continuous averaging The Analog Generator is loaded with a low distortion sine waveform set to a default frequency of 1 kHz and an amplitude of 1 Vrms Page2 contains a graph displaying the A and B channel power spectra while Page 3 contains a graph displaying the A and B channel time records What you ll need to configure Adjust the amplitude and frequency of the analog output signal to match your requirements The amount of averaging in the FFT analyzer can be adjusted to suit the noise level of your signal The FFT resolution is defaulted to 2k points this can be adjusted Frequency Response This setup configures SR1 for a stereo audio band frequency response sweep The Hi resolution converter is selected for maximum dynamic range The Time Domain Detector is selected for the AO analyzer The Analog Generator is loaded with a sine waveform Page2 contains a graph displaying the A and B channel frequency response The sweep controller is programmed to sweep the analog sine frequency from 20 Hz to 20 kHz in 22 logarithmic steps What you ll need to configure Adjust the the sine amplitude to match the equipment being tested The number of points in the sweep can be changed to give faster sweeps or higher resolution sweeps The settling parameters can be adjusted to fit the noise levels of the signals being measured THD N vs Amplitude Sweep Performs a THD N v
239. m E E T A T E Cher Cee eery o00m perna isus ven sinasiusenepseateans ded i Negative i i 150m ere ima none ad bor eeERNeRRNTTERNRNE naha I had aupassur i ionK estra E anani hither monet vanctunecdennemaedmnniaataceen ac ree Creer i a a oe Se ET a eee a ee i ae oe ee aeons E crane SEN eee eee E 2 0 1 5 1 0 500m a 500m 1 0 1 5 N And its Input Amplitude Histogram For a digital audio signal the pulse width and pulse rate histogram will show peaks at the one two and three Ul transition widths that make up a legitimate digital audio carrier 2014 Stanford Research Systems 190 SR1 Operation Manual Yo E E T CEE T E papas _ a S sn S FE OE EEE a E SE O 7 _ 20 SU Peak Preambles 10 a E ss a a L E TT OT E ANAKE EN NENONEN KEATONA sssrini Onli T 0 500m 1 0 1 5 2 0 2 5 3 0 3 5 Pulse Width Histogram of Digitial Audio Carrier The Eye Diagram can be thought of as a two dimensional histogram The eye diagram is constructed by overlaying the transitions corresponding to the selected jitter detection option preambles stable bits or all bits on a two dimensional grid The x axis of this grid consists of the specified number of Uls binned into the number of bins specified in the X Resolution control under the Eye Diagram heading The y axis includes the range from the maximum positive to the maximum negative extent of t
240. m Sine waveforms are specified only by their amplitude and frequency The Normal Sine waveform may be combined with other waveforms The Low Distortion Sine waveform uses additional signal processing to obtain ultra low distortion sine waves Because this extra signal processing would interfere with other waveforms the Low Distortion Sine may not be combined with other waveforms Low Distortion Sine and Phase Due to the additional signal processing used to obtain ultra low distortion outputs significant 10 phase differences may exist between the A and B channel outputs when the low distortion waveform is selected For this reason it is not recommended to use low distortion sine for testing phase 2014 Stanford Research Systems SR1 Operation 45 Phased Sine Config Phased Sine averorm On Iw EQ T ine Amp 0 0000 VETS Freq 1 00000 kHz Phase 0 0000 2 The phased sine waveform consists of two sines one on channel A and one on Channel B with a specified phase difference between them This waveform may not be combined with other waveforms Synchronous Burst Sine Config Synch Burst Sine aveform On v EQ 7 ut Type Internal n Rep Rate 10 00 CYC Arig 0 0000 WETS On Time 2 000 CYC sd Freq 1 00000 kH z l LoAmp i0 00 l The synchronous burst sine is a sinewave capable of fast switching between two amplitude levels Amplitude shifts occur synchronously w
241. main Frequency Domain Trigger Resolution Bandwidth 4k 128 00 kHz Iw Show Aliased Window Phase Thresh Unwrap RMS Sum Rife Vincent 4 100 00 mUI E Filters Hi Pass M n Lopass weighting 20 000 kHz 2 0000 kHz None bl Skart Center End 0 0000 Hz 64 000 kHz 128 00 kHz E oO Ji Time Display Interpolation DE Correction C Off On None Averaging Ayigs Clear Avg Done Exponential F a 2014 Stanford Research Systems SR1 Operation tet Resolution and Bandwidth Resolution Bandwidth 128 00 kHz i Show Aliased 126 00 kHz The number of lines in the jitter FFT can be set from 256 to 32k lines using the Resolution control These lines span the frequency range from DC to the value set in the Bandwidth control The ADC used by the jitter detector like all of SR1 s converters is preceded by an anti aliasing filter with a finite attenuation slope As a result some lines at the upper edge of the frequency range may not be protected against aliases to the full attenuation of the filter If Show Aliased Lines is checked SR1 will display the entire spectrum up to Fs 2 including lines that may not be fully alias protected If the control is unchecked only fully alias protected lines will be displayed and the bandwidths will be adjusted accordingly Total Jitter RMS Sum The RMS Sum control displays the total jitter found by RMS summing all the points in the jitter FFT
242. mbers For instance 0000000e 000 s0001 00000006004 0000000e 004 0000000e 004 0000000e 004 0000000e 004 0000000e 004 0000000e 004 0000000e 004 Oo oo x7 HD oO SB WN OC OC Multiple tab separated columns of values may be present in the file If so SR1 will prompt the user for the correct column to use when loading the arbitrary waveform file Note that the absolute value of the sample values are disregarded when outputting an arbitrary waveform only the relative magnitudes of the sample values are important To arrive at the absolute scaling SR first estimates any overshoot that may occur due to the fact that the file samples may not be band limited The maximum value of the file waveform including overshoot is then scaled to the amplitude value entered on the arbitrary waveform panel For instance if the file above was loaded and output with an amplitude of 1 Vp the actual output would show a linear ramp from O Vp to 1 Vp Configuration and Display Files XML Configuration and Display files are XML files that specify instrument configuration and display data Because the data in these files contains multiple settings with many complex interactions it is not recommended or necessary for users to modify configuration or display files 2014 Stanford Research Systems sto SR1 Operation Manual 3 6 Hardware Reference Using SR1 with an External Monitor SR1 has a VGA connector on the rear panel which can
243. measurement What you ll need to configure for all these setups Change the digital input and output settings connector OSR bit resolution to match the equipment being tested Jitter Chirp This measurment uses a jitter chirp to measure the jitter transfer function of the equipment being tested A jitter chirp is a waveform synchronized to the jitter analyzer that contains equal amounts of jitter in each frequency bin When used as the input to a DUT the output jitter spectrum is proportional to the jitter transfer function of the DUT The AO analyzer is setup as a frequency domain jitter analyzer Pages 2 and 3 contain the frequency domain and time domain jitter results What you ll need to configure Adjust the amplitude of the jitter chirp on the_Digital IO Panel Jitter Sweep This setup measures jitter amplitude vs frequency using a sweep The frequency of a jitter sine is swept from 20 Hz to 100 kHz and the resulting jitter output from the DUT is recorded on Page2 What you ll need to configure Adjust the the sine amplitude to match the equipment being tested The number of points in the sweep can be changed to give faster sweeps or higher resolution sweeps The settling parameters can be adjusted to fit the noise levels of the signals being measured Jitter Measures both RMS Jitter and the Jitter spectrum with SR1 s jitter generator off This setup is used to measure the intrinsic jitter output from a DUT AO is setup a
244. mined and the relevant file options will be explained Pole Zero EQ Files Pole Zero EQ files contain a list of s plane poles and zeros which form a complex function of frequency Thus pole zero EQ files have both a magnitude and phase response as a function of frequency Consider for instance the s plane transfer function corresponding to the well known A Weighting function ka g T 5 O O a s s 2r 20 6 s 27 107 7 s 27 737 9 s 27 12200 where k 7397235900 0 Note that the function involves an overall constant multiplier 4 zeros at zero frequency and 6 real poles This A Weighting transfer function is represented by the EQ file shown below lt xml version 1 0 encoding UTF 8 gt lt DOCTYPE EQcurve SYSTEM EQcurve dtd gt lt EQcurve gt lt MetaData gt lt Source Value C Program Files Borland CBuilder5 Projects Audio an EQgen EQgen exe gt lt DocType Value EQcurve XML gt lt Version Value 1 0 0 gt lt Date Value 2005 05 05T15 54 12 gt lt MetaData gt lt EQdata Name Style Type 6 Value PZmodel gt lt EQdata Name PolesReal Type DblArray Value 20 6 20 6 107 7 737 9 12200 12200 gt lt EQdata Name PolesImag Type DblArray Value 0 0 0 0 0 0 gt lt EQdata Name ZerosReal Type DblArray Value 0 0 0 0 gt lt EQdata Name ZerosImag Type DblArray Value 0 0 0 0 gt lt EQdata Name PZmultiplier Type 3 Value 6 283185307 gt lt EQd
245. mitter with different time constants for rising and falling edges On an eye diagram an offset can be detected by observing that the points at which the positive and negative going pulses cross is above the 0 Volts line The eye diagram for an actual digital audio signal with a small offset is shown below NA 500m 200r T 200m 400m 600m 500m 1 0 12 Ul Eye Diagram of Offset Digital Audio Carrier Another way of observing the effect of asymmetry is to examine the histogram of pulse widths For an ideal square wave signal there would be a single peak in the histogram of pulse widths corresponding to 1 2 the square wave period For the digital audio signal above we can see that the 2 Ul peak has been split into two peaks the higher one corresponding to the slightly longer positive going pulses and the peak on the left corresponding to the slightly shorter negative going pulses 1 90 1 92 1 94 1 96 1 98 2 00 2 02 2 04 2 06 2 05 I Splitting of 2 Ul Peak Caused by Pulse Asymmetry 2014 Stanford Research Systems 188 SR1 Operation Manual To account for these processes the digitizer includes an asymmetry parameter when reconstructing the original clock Instead of a single Unit Interval length the digitizer assumes 2 unit interval lengths one for positive pulses and one for negative pulses The difference in the lengths of positive going pulses U and negative going pulses U is calculated and displayed as an asymmetr
246. modifier character The allowed modifiers are Character Modifier Numeric Vaus pico hoe P ito C e T e he ito Note that the front panel keypad contains dedicated keys for the unit modifiers just to the right of the Enter key To complete the entry press the Enter key on the front panel or an external keyboard To abort the entry press the Esc key on the keypad or keyboard Sometimes the precision of the value represented by a unit entry may exceed the number of digits displayed To see the value of any unit entry displayed with full double precision hold the cursor over the unit entry the full double precision value will be shown as a hint Unit Displays Unit Displays are the controls with a black background and green text which are used to display measurement results and other quantities whose value can t be changed by the user As with unit entries the units of the unit display can be changed by using the mouse or trackpad to click on the Ea at the right of the control and select a unit from the drop down list Right clicking the Unit Display brings up a menu which offers some more options 2014 Stanford Research Systems 8 SR1 Operation Manual w Auto Vrms Mvrms EM rns VEINS mrs UVES nVrrns w Standard Max Min Delta Normally the Unit Display auto selects the appropriate unit modifier for the currently displayed value For instance 1 2x10 Vrms would be displayed as 1 2 uVrms There are situatio
247. n interval of the generated noise If not checked the repeat interval is sufficiently long that it won t observable under most conditions If however it is desired for the noise to repeat after a fixed interval check the Repeat box and enter the Repeat interval below it If Repeat is selected the noise waveform produces a generator trigger each time a waveform repeat is initiated The Pink checkbox specifies that the noise output should be filtered by a 3 dB octave pinking filter While the power contained in a white noise signal is linearly proportional to the measurement bandwidth pink noise will have equal power in equal logarithmic frequency intervals e g the power contained in the 100 Hz to 200 Hz interval will be the same as the power contained in the 10 kHz to 20 kHz interval White or Pink noise can be further filtered according to the controls in the filter group 2014 Stanford Research Systems SR1 Operation None s_ i White or Pink noise is directly output without further filtering frequency specified in the filter frequency control ee ee frequency specified in the filter frequency control seis fi l l i 1 3 Octave The white or pink noise is filtered with a 4th order 1 3 octave bandwidth Butterworth bandpass filter at the frequency specified in the filter frequency control The noise waveform outputs a generator trigger when only when Repeat is checked The trigger occurs each time the waveform is repe
248. n the results will be displayed as an absolute voltage If Ratio is specified the results are shown relative to the dBrA and dBrB references acquired using the Reference quick measurement The number of lines of resolution is adjustable from 256 to 32k Free Run Configuration A single graph is created with two traces showing the instantaneous frequency response of the EUT Sweep Configuration A graph will be created for each channel As the sweep progresses a trace will be created on each graph for each amplitude value specified in the sweep The settling parameters of the Frequency Response measurement can be adjusted directly from the Settling tab of the panel 2014 Stanford Research Systems 264 SR1 Operation Manual Settling Profile Delay average 20 000 msec Frecesion nPoinks Threshold Because the frequency response quick measurement is a vector measurement it has different settling options compared to other quick measurements The only settling profile available for vector measurements is average The nPoints control in this case dictates how many FFTs will be averaged before the result is displayed Refer to the Settling Panel chapter for a more detailed discussion of settling 2014 Stanford Research Systems SR1 Operation 265 2 8 7 Distortion Panel E Measure Distortion Inpukiutouk Measurement Settling Signal waveform Sine a Start Stop Steps Log Freq 20 000Hz l20 000kHz
249. n to open a file dialog to specify the EQ File Check the Invert EQ box to have the amplitude scaled by the inverse of the EQ file response Waveforms that are capable of being used with EQ files will have an EQ checkbox in their waveform tab This box must be checked for EQ to be active regardless of whether an EQ file is selected in the file selection control When an EQ file is selected and the checkbox is checked the amplitude control in the waveform tab will continue to show the constant user selected waveform amplitude however the Total Channel Amplitude control will display the amplitude with the EQ response included 2 3 1 1 Analog Generator Units The amplitude and frequency of generator waveforms can be specified using a variety of units all of which are useful in different audio test scenarios Because of the large number of waveforms that SR1 can generate and because it s useful to define amplitude in a way that simplifies the coupling between the details of the waveform and its amplitude SR1 uses the following two conventions for analog generator amplitudes 1 Analog generator amplitudes regardless of the waveform or the units they are expressed in refer to the peak value of the waveform When waveforms are combined in the generator the amplitudes add simply regardless of the phase relation of the waveforms 2 RMS units always regardless of waveform have the same relation to peak units that they do for a sine wave Thus a
250. n twice the tolerance of of the current point the 3rd previous point within 3 times the tolerance etc This is the most lenient of the settling profiles and reflects the fact that the variability of many measurements decreases as a function of time after a transient change The measurement is considered settled using the flat profile when the previous nPoints measurements are all within tolerance of the current measurement This is the strictist of all the settling profiles Sequential The measurement is considered settled if the current measurement is within tolerance of the previous point the previous point is within the tolerance of the next previous point etc for a total of nPoints Average The result of the last nPoints measurements are averaged together and the average result is considered the settled value Delay The delay parameters sets the amount of time after the sweep source has changed to a new value before the analyzer begins to look for a settled measurement If the sweep appears glitchy due to transients from the moving sweep source increasing the delay value can sometimes help to clean up the sweep Threshold At very low levels the tolerance window yx 7 precision becomes very small and it is possible that some measurements will not settle The threshold parameter sets a minimum value for the tolerance window such that if yxtolerance lt threshold the tolerance window becomes y threshold 2014 Stan
251. n using the log sine chirp Window functions attenuate the beginning and the end of the time record which means that some frequencies will be attenuated more than others by the window The advantages using log sine chirps for impulse response measurements are detailed by Miller and Massarini Transfer Function Measurements with Sweeps J Audio Eng Soc vol 49 pp 443 471 June 2001 They report better results using log sine stimulus compared to MLS stimulus for a wide range of audio DUTs One particular advantage concerns the unique ability of the log sine chirp to differentiate between the linear and non linear portions of the DUT response Because of the properties of the group delay of the log sine chirp waveform harmonic responses appear offset in time relative to the linear response This can be clearly seen in this impulse response measurement of a consumer stereo amplifier Harmonic Response 2 i Te Se eee ee ee IEEE EEE EE 3 POEET 3 EREA eee ay Linear Response 50m 25m 0 25m Som 75m 100m a When the purely linear response is gated and transformed back into the frequency domain a much better measurement of the DUT frequency response is obtained This is detailed in the FFT2 analyzer discussion See Measurement of audio equipment with log swept sine chirps by Thomas Kite Presented at the 117th AES Convention San Francisco CA October 2004 for a concise explanation of the unique group delay properties
252. nal bursts Burst On Time specifies the number of cycles the sine is at the 2014 Stanford Research Systems 46 SR1 Operation Manual high amplitude during the burst cycle For externally triggered bursts the Burst On Time specifies the number of cycles the sine is output at the high amplitude following a trigger Lo Amplitude specifies the burst Lo Amplitude as a fraction of the nominal waveform amplitude Synchronous Burst Sine Repetition Rate 6 cycles Burst On Time 2cycles The green burst indicator will flash when the burst is triggered in either the Ext Gated or Ext Triggered modes The red burst error indicator will flash in the externally triggered mode to indicate a rate error condition Noise Config Moise mw aveform q Filter 7 On Iw 1 3 Oct i D0000 kHz Fink J Repeat Rpt 100 00 msec C ie The noise waveform outputs random noise with an almost gaussian amplitude probability density function with several options A true gaussian probability density function has finite probability for any amplitude no matter how large which is impractical for a physical device Amplitude controls as with all SR1 waveforms the peak value of the noise output The crest factor of the noise waveform the ratio of the peak value to the rms value of the noise is approximately 4 The Repeat checkbox governs the repetitio
253. nal details The Digitizer is designed to work with the Digitizer Display to produce full color eye diagrams histograms of carrier amplitude jitter amplitude pulse width and rate and spectra of both the carrier and the computed jitter signal The heart of SR1 s digitizer is an 80 MHz 8 bit transient digitizer with an effective analog bandwidth of approximately 20 MHz The digitizer can store up to 2 Msamples of data in each record A flexible trigger generator allows synchronization of the digitizer record with a variety of points on the carrier Signal as well as external events After acquiring the record the digitizer analyzes the zero crossings of the digitized signal reconstructs the original clock and then calculates the jitter of the signal relative to the reconstructed clock Digitizer Input Selection Input Contig Cnictr exc ea Dual Conn EQ Sor w Term Selection of the input signal for the digitizer is done on the Input section of the Digital I O panel Select either the BNC or XLR connectors optical connector or the digital audio output monitor For BNC and XLR connector C1 or C2 must also be specified Checking Term applies the appropriate termination for the selected connector which can substantially improve the quality of the digitized signal Digitizer Acquisition cguisition Record Len ek Acquire Trigger Recv Preamble 4 C1 Repeat Post Trigger 100 00 Ho Input Range a Input ain Auto
254. nal tree node sweep is only active when a sweep is configured In that case the node will contain a separate entry for each sweep data that has been selected on the sweep controller panel Clicking on the sweep trace will produce a trace who s x axis corresponds to the configured sweep source and y axis corresponds to the selected sweep data Note that measurements accompanied by the icon are scalar measurements which will be represented by strip chart type traces and measurements shown with the icon are vector measurements which will be shown on FF T type traces Clicking on any line in the Trace Listing makes that trace the active trace which connects it to the cursor and allows control of its scaling Right clicking on the trace listing line brings up a submenu offering the following choices Trace Listing Right Click Menu Makes the trace offline The current trace data is saved After being taken offline the trace can still be scaled renamed and saved and recalled It s data however is fixed when the trace is taken offline Saves the trace to a disk file A trace file can be added to any graph by right clicking in the graph area and selecting Load Trace from the submenu Removes the trace from the graph and stores it in memory The stored trace can be pasted back in the current graph or a different graph by right clicking in the graph area and selecting Paste Trace from the submenu 2014 Stanford Research Sys
255. nal waveform will have the same 2014 Stanford Research Systems SR1 Operation 143 phase relative to the beginning of the time record for each averaged FFT Below the averaged Linear Spectrum Navg 100 is shown plotted with the averaged Power Spectrum for the same number of averages Note that averaging the linear spectrum does not reduce the variation in the noise floor but does reduce the amount of noise In this case averaging 100 spectra has reduced the noise floor by about 20 dB dByrmns 10k 15k 20k 25k 30k Hz Comparison of Averaged Power Spectrum and Linear Spectrum Computing the 2 Channel Frequency Response After computing the Power Spectrum and Linear Spectrum for both channels DSP computes the Frequency Response The complex frequency response is defined in terms of the FFTs for the two channels as Fy s MS F is a mathematically volatile expression because it diverges when the A channel input is small To minimize this volatility SR1 uses a standard technique called the Tri Spectral Average when computing the dual channel frequency response If we multiply the numerator and denominator of the equation above by the complex conjugate of the A channel FFT and average the numerator and denominator separately we get the following expression for the averaged frequency response Pe we LPO EG gt lt T s gt SATE The denominator of this expression is simply the Power spectrum of the input channel which
256. nalog input boards have provision for the installation of up to 4 optional filters Each filter can be inserted in the signal chain after the Notch Bandpass filter and before the postfilter gain elements Note that the optional filters only affect Hi Bandwidth analog measurements Optional Filters Amplitude Chain HW Notch EP Optional HW Filter Filters Attenuator Gain FP _ _ to HiBw ADC Range Control Freq Meas Phase Meas Attenuator kA Galih k 9 Ge Ch E e Level Chain a gt Out z m EW Limit Weighting HiBw ADG PostFilter Gain RMS Pk Amplitude uasi Pk Heas Mt Analog Processing Level Meas SOOO S Digital Processing At the current time no optional filters are available As they are developed detailed information on the response of the optional filters will be included in this section of the manual 2014 Stanford Research Systems 306 3 9 SR1 Operation Manual File Reference EQ Files EQ EQ files are XML files that represent a real or complex magnitude or magnitude and phase frequency response There are two types of EQ files pole zero files which contain a set of s plane poles and zeros which represent a complex frequency response and sampled EQ files which contain a set of pairs of frequencies and response values at those frequencies with other frequency values calculated by interpolation In the examples below an example of each type of file will be exa
257. nalyzer discussion See Measurement of audio equipment with log swept sine chirps by Thomas Kite Presented at the 117th AES Convention San Francisco CA October 2004 for a concise explanation of the unique group delay properties of the log sine chirp The log sine chirp waveform outputs a generator trigger at the beginning of each FFT time record Be sure to use generator trigger as the analyzer trigger source when using the log sine chirp waveform with the FFT analyzer MultiTone Config MultiTone aveform On fw Eo Export Like FFT Chirp the MultiTone Waveform is a special waveform containing a series of tones that generated in synchronous fashion to ensure bin center placement in the MultiTone Analyzer Using the MultiTone generator and analyzer many different characteristics of a device e g noise distortion frequency response cans be measured quickly and simultaneously without the need for multiple swept 2014 Stanford Research Systems SR1 Operation measurements All of the details of the multitone waveform tone frequency and amplitude phase etc are configured using the MultiTone Configuration Panel As a result only the amplitude is specified on the waveform tab The Export button allows the current multitone waveform to be saved in arbitrary waveform format for off line analysis Because of the special properties of the MultiTone waveform it may not be combined with other waveforms If the genera
258. nalyzer section for more details on these topics Because of the synchronous nature of the chirp signal a uniform window should be selected in the analyzer when using the chirp waveform Selection of the associated FFT Analyzer is done with the Chirp Source control Chirp Source Associated Analyzer AO FFT1 Analyzer 0 Single Channel FFT Analyzer AO FFT2 Analyzer 0 Dual Channel FFT Analyzer A1 FFTO Analyzer 1 Single Channel FFT Analyzer A1 FFT1 Analyzer 1 Dual Channel FFT Analyzer Synchronous chirp generation requires there to be a relationship between the selected generator sampling frequency and the associated analyzer sampling frequency In general the two frequencies need to be either exactly the same or some integer multiple of one another If the generator and analyzer sampling frequencies are incompatible the Compliance led will glow red and no waveform will be output The compliance led may also show red if SR1 runs out of table memory to create a long chirp Signal say for a large number of FFT lines or when large amounts of fft zoom are used The chirp waveform may attempt to use interpolation in situations when not enough memory is available for the complete table in which case the Compliance led will glow yellow Some degradation of flatness can be expected in this case Using the chirp signal it is possible to make cross domain measurements of both amplitude and phase For instance when testing a D A converter a chi
259. nce Out z preamble and the main Digital Audio output z preamble The delay may be set between 0 and 127 5 UI 128 UI would be 1 complete digital audio frame in 5 UI intervals 2014 Stanford Research Systems SR1 Operation 107 2 4 Analyzers Menu The core of SR1 s measurement capabilities is found in its collection of analyzers Each analyzer is represents a functionally related group of measurements and the controls related to those measurements At any given time there are two active analyzers denoted by AO and A1 To change the type of the two active analyzers use the following selections from the Analyzers menu Makes wideband amplitude ratio SNR and THD N measurements The Time Domain Detector time domain detector signal chain includes bandpass or notch filters TDD bandwidth limiting filters and a variety of different weighting filters Peak RMS and Quasi Peak responses are selectable Provides single channel FFT functionality Measurements include power spectrum time record phase Both zoom changing the FFT frequency range and heterodyne moving the FFT frequency range are included Single Channel FFT FFT1 Provides dual channel FFT functionality Measurements include power Dual Channel FFT spectrum and time record for both channels plus frequency response FFT 2 transfer function and interchannel phase Zoom changing the FFT frequency range is supported in the dual channel analyzer Makes frequency selec
260. nd simply become the magnitude of the last indivdual FFT Fixed Length averaging means that the analyzer will average the selected number of spectra and then stop Continuous averaging continuously averages the spectra weighting more recent results exponentially more than older spectra The two Peak Hold selections only affect the averaging of the Power Spectra When Peak Hold is selected instead of averaging successive power spectra each bin of the new spectrum is compared to the current buffer if the value in the new spectrum is greater the value in the buffer is replaced This hold the maximum value in each bin and is useful for detecting unwanted transient events The Clear button clears the average buffer and re starts averaging The Avg Done indicator lights when the required number of averages have been accumulated The FFT1 Meas2 Panel This panel contains additional configuration controls for the FFT1 analyzer 2014 Stanford Research Systems SR1 Operation 133 Measurement Meas2 References Trigger Window DC Correction Rife Vincent term Average ad Phase spectrum Weighting Unwrap None nd Threshold Invert 120 00 dBFS Time Display Interpolation t Off m On Variable Time Window Limits start 0 0000 Yo stop 100 00 vo Window Selection It is well known that the application of a window functions is typically necessary to obtain maximum dynamic range for FFT measurements The di
261. nd report which items are red in the hardware status panel Hardware Status Anidin B Board Hot Found B Curent Temp 0 Seral No 0 Version E D Supported Version es B Vers NH Source Constructor Cal File C Program Files SA1 Audio Analyzercal Cal Date 0700 B Cal Source Nominal Cals B Cal Flags Ox00000000 Anigln B Board Hot Found B Current Temp 0 Seral Ho 0 Version E D Supported Version es B Ver5 N Source Constructor jiii 2014 Stanford Research Systems SR1 Operation 231 2 6 5 Networking The following sections describe the controls used to configure operation of SR1 in a networked environment The controls and dialog boxes described are not unique to SR1 they are from the Windows XP operating system which underlies the instrument As such these controls will not be described in complete detail Only the features relevant to normal use of SR1 will be discussed The following networking options are available Network Setup Set the instrument s IP address Network Places Display a list of files folders and computers on the network Map Network Drive Map a network location to a driver letter Remove Network Drive Remove a mapped network drive Allow other users on the network access to a portion of SR1 s disk drive 2 6 5 1 Network Setup To configure SR1 in a network select Network Setup from the Networking option in the Tools menu a Network Cprmect
262. ndividual tone Linear The desired number of tones are placed on prime numbered bin frequencies within the Prime selected range with an approximately uniform frequency distribution Log The desired number of tones are placed on prime numbered bin frequencies within the Prime selected range with an approximately uniform ratio between adjacent tone frequencies The phase distribution of tones in a multitone signal directly affects the crest factor of the signal Crest factor is important because it limits the amount of power that can be placed in each tone for a given peak amplitude of the resultant signal SR1 offers several phase distribution choices for multitone signals Zero simply sets all tone phases to 0 which usually results in a signal with very high crest factor but which may be useful for comparison purposes Random assigns random phases to each tone The remaining choices Newman Schroeder Zygmund and Rudin are various 2014 Stanford Research Systems 104 SR1 Operation Manual approximations to the full crest factor minimization problem which has no closed form solution The various algorithms are described in the paper Low Crest Factor Multitone Test Signals for Audio Testing by Alexander Potchinkov JAES v50 9 p681 Once the tone placement options are selected press Create Default Tones to update the waveform Note that none of the tone placement options will be reflected in the actual output until
263. ned in the 10 kHz to 20 kHz interval The MLS noise waveform outputs a generator trigger each time the MLS cycle repeats 2014 Stanford Research Systems 48 SR1 Operation Manual USASI Noise Config USASI Noise Waveform On fw MUSASI Noise Amp 0 0000 Vrms USASI noise is a special type of filtered noise designed to mimic the content of audio program material USASI noise is typically used in testing broadcast transmitters to measure compliance with transmission bandwidth requirements Because the frequency content of USASI noise is fixed only an amplitude control appears on the waveform tab USASI noise may be used in conjunction with generator bursting to generate a burst USASI signal suitable for transmitter testing Frequency Spectrum of USASI Noise 100 Hz 6dB oct hipass 320 Hz 6 dB oct lopass Square Wave Config Square waveform gt On W Square Amp 0 0000 rms Freq 1 00000 kHz SR1 square waves uses special hardware to generate clean analog square waves As a result square waves may not be combined with other waveforms in the analog generator The square wave frequency is limited to a maximum of 50 kHz regardless of the selected analog generator sampling rate Because of the finite bandwidth of the analog generator square waves will have finite risetime and ov
264. nel converter Fs to Digital OSR using the Analog Inputs panel The chirp waveform generates a generator trigger once each cycle Be sure to use the generator trigger as the analyzer trigger source when using FFT chirp Log sine Chirp conf Chirp waveform On e EQ Chirp Amp 1 0000 FFS Compliance She AL FFT 2 r l The log sine chirp waveform is designed to work in combination the dual channel FFT analyzer to make impulse response measurements The log sine chirp waveform is a sinusoid whose frequency is swept in a logarithmic fashion over the frequency span of its associated FFT analyzer The log sine chirp waveform has a pink frequency roll off of 3 dB oct The log sine chirp waveform is synchronized to the settings of a particular FFT analyzer If the resolution or frequency span of that analyzer changes the log sine chirp waveform automatically reconfigures to sweep over the frequency span of the selected analyzer Because of the synchronous nature of the chirp signal a uniform window should be selected in the analyzer when using the log sine chirp waveform Selection of the associated FFT Analyzer is done with the Chirp Source control Chirp Source Associated Analyzer AO FFT1 Analyzer 0 Single Channel FFT Analyzer A0 FFT2 Analyzer 0 Dual Channel FFT Analyzer A1 FFTO Analyzer 1 Single Channel FFT Analyzer A1 FFT1 Analyzer 1 Dual Channel FFT Analyzer If the analyzer span is set to frequencies w
265. nential j100 00 nFFS 30 000 msec Precision The precision parameter sets the amplitude scale for what is considered a settled measurement If the value of the most recent measurement is y then another measurement is considered to be within tolerance if its value falls in the range of yx 1 precision Tightening the precision window increases the precision of the final sweep data at the expense of having to possibly wait longer for a settled measurement The correct value of the precision depends on the measurement and application and can usually only be determined by experimentation the tight tolerance that might be appropriate for measuring the flatness of a filter passband with sub dB precision is probably too restrictive for broadband noise measurements Profile The settling profile along with the value of nPoints defines an algorithm which determines whether an measurement is settled Each of the settling profiles is described below Settling Profile No settling is performed when this profile is selected The most recent measurement is always considered settled and added to the sweep Exponential In exponential settling the most recent measurement point is compared with the last nPoints 1 previous measurements For the measurement to be considered settled the previous point must be within tolerance of the current point The 2nd 2014 Stanford Research Systems SR1 Operation 99 previous point must be withi
266. ness 0 001 dB Harmonic Spurious 160 dB Phased Sine 0 to 360 range 0 001 resolution Square 10 Hz to F _ 2 frequency range 2014 Stanford Research Systems SR1 Reference 299 IMD SMPTE DIN CCIF DFD DIM TIM Noise White Pink Filtered White Pink USASI MLS Maximum Length Sequence from 2 to 2 samples repetition interval Ramp F N frequency range N 20 adjustable rise fall fraction Arbitrary 256 Samples to 136k Samples FFT Chirp Equal power in each FFT bin Frequency response can be modified with an EQ file Log sine Chirp Swept sine with a logarithmically increasing frequency Used for impulse response measurements Polarity 10 Hz to Fs 4 frequency range Bursts All allowed waveforms Digital Test Waveforms Digital Constant Count Rotating Bits Staircase J Test Dither None triangle and rectangular probability distribution Digital Audio Carrier Impairments Jitter Waveforms Sine square uniform noise BP filtered noise Chirp Frequency Range 2 Hz to 200 kHz Amplitude Range O Ul to 13 Ul Normal Mode Noise Amplitude Range Unbalanced 0 to 637 mVpp Balanced 0 to 2 55 Vpp Common Mode Sine Amplitude Range 0 to 20 Vpp balanced only Frequency Range 10 Hz to 100 kHz Cable Simulation Simulates 100 m of digital audio cable Variable Rise Time 5 ns 10 ns 20 ns 30 ns or variable from 40 ns to 400 ns 2014 Stanford Research Systems SR1 Operation Manual Signal Measurements General Analog In
267. ng between the details of the waveform and its amplitude SR1 uses the following two conventions for digital generator amplitudes 1 Digital generator amplitudes regardless of the waveform or the units they are expressed in refer to the peak value of the waveform When waveforms are combined in the generator the amplitudes add simply regardless of the phase relation of the waveforms 2 RMS and Peak to peak units always regardless of waveform have the same relation to peak units that they do for a sine wave In other words think of rms and peak to peak units as simply units with a a fixed scale relative to peak units rather than as a quantity derived through a calculation on the waveform The following table describes the units available for setting the amplitude of digital generator waveforms 2014 Stanford Research Systems SR1 Operation EEE According to AES17 1998 r2004 Full scale amplitude is the amplitude of a 997 Hz sinewave whose positive peak value reaches the positive digital full scale leaving the negative maximum code unused The FFS Fraction Full Scale unit expresses the peak amplitude of any SR1 generator waveform relative to this definition of full scale amplitude When dither is on full scale amplitude is reduced symmetrically by 1 bit to allow for dither Small values of FFS may be expressed as mFFS milli FFS or uFFS microFFS AFS Same as FFS above but expressed as a percentage of full scale amplitu
268. ng the first data point Measurement Timeout sets the maximum time that the analyzer waits for settled measurements if a settled measurement is not returned in this time interval the analyzer skips that point and moves on Checking Rpt Repeat causes the sweep to automatically restart as soon as it finishes the current sweep Sweeps and Events SR1 s_Event panel allows a number of user defined actions to be associated with the occurrence of certain events Actions include logging the occurrence to a text file running a script or triggering a COM event Sweeps define several such events Sweep Start Sweep New Point Start Sweep New Point Timeout Sweep New Point Done and Sweep Finished 2014 Stanford Research Systems o SR1 Operation Manual 2 3 8 Settling Panel While in sweep mode SR1 requires all measurements to settle before adding them to the sweep Settling insures that the variability for the measurement whether intrinsic to the measurement or due to transients arising from the sweep is reduced to a predetermined level Each measurement can have a different settling profile each characterized by a precision number of points profile threshold and delay iol x E aeiee aA A Precision nPoints Profile Threshold Delay Ratio fi OOOO f E Exponential j10 000 m 30 000 msec Amplitude Analog fi A000 f E Exponential j100 00 nvr 30 000 msec Amplitude Digital fi A000 i f E Expo
269. nnector as well as other input parameters coupling range etc is made on the Analog Inputs panel 2014 Stanford Research Systems SR1 Reference 293 3 2 Rear Panel Descritpion The following signals and connectors are found on SR1 s rear panel oh o f phere ae her Line 48 Hz to 63 Hz Ax vrsable Fuse 4A 100 120 VAC pe gt R D FIN a b te be Me wit rae a D Dirt rr lt an ot ee Des JD MC es G AUC P DVRA Rie SSN see LD em e e o e eee 0 80 0 0 0 0 80 000 0 88 ANA e e e o e 0000o ARAA ome m Hl sa SRS seontor reser ch Systems MADE IN USA Sec a ee eee pi 0 00 0 e000 0000000 0 0 6 0 0 00000 6 6 6 6 6 6 6 6 00 008 8 MO USN SEC sm ee ee Pe SEEEN CO tees tern OTIO 7 d A Power Entry Module Connect the supplied power cord to the power entry module to provide AC power to SR1 Make sure that the card on the power entry module shows the correct AC line voltage 100 120 200 240 for your locale B lIEEE 488 Connector The 24 pin IEEE 488 connector allows a host computer to control the SR1 via the IEE 488 GPIB instrument bus The GPIB Address of the unit is set va software on the Remote tab of the Preferences Panel C RS 232 Connector The RS232 serial interface connector is configured as a DCE transmit on pin 3 receive on pin 20 The Baud Rate Parity and Word Length are all configured on the Remote tab of the Preferences Panel To connect the SR
270. nnel Probability A B The histogram normalized to the total number of bins in the histogram Each bin amplitude then represents the probability that the input amplitude falls within the limits of that bin FtAB O The best fit gaussian to the the probability histograms The Average value of the input samples for each channel Sigma A B The Standard Deviation of the input samples for each channel Example Histograms The histogram provides a different perspective on audio signals compared FFTs or standard Time Domain techniques As an example consider the histograms below 1 25 1 00 750m 500m 250m 0 250m 500m 750m 1 00 1 25 F5 Histogram of 997 Hz Digital Sine 2014 Stanford Research Systems 172 SR1 Operation Manual 2 5k 1 5k 1 0k 500 1 25 1 00 750m 500m 250m 0 250m 500m 750m 1 00 1 25 F5 Histogram of 1 kHz Digital Sine The first histogram shows a 997 Hz digital audio sine wave with a sampling frequency of 48 kHz Because the sampling frequency is not a multiple of the signal frequency each cycle of the digital audio sine uses slightly different values and the result is a histogram with a smooth continuum of amplitudes between 1 FFS If the sine frequency is shifted slightly to 1 kHz the histogram changes radically Now the sampling is a multiple of the signal frequency and each cycle of the sine repeats a finite set of exactly the same values The discrete nature of the 1 kHz sine is easily seen on the hi
271. notch filtered input signal i e the distortion signal Like the other monitor outputs the scaling of the Analyzer monitors can be set using the Monitor Panel I Burst Trigger In Chs A and B These TTL inputs trigger the corresponding Analog Generator bursts when the generator is configured for either Gated Burst or Triggered Burst See the Analog Generator chapter for details of the SR1 s analog generator burst modes J Sync Out Chs A and B The sync connectors output a square wave signal with TTL levels that is phase synchronous with the outputs from the Analog Generator These signals are only meaningful if the Analog Generator Waveform has periodic zero crossings i e is not noise K Burst Monitor Out Chs A and B The burst monitors output a TTL high level when the burst of the corresponding Analog Generator channel is in its high state and a TTL lo level when the corresponding burst is in a low state These Signals are valid for both synchronous bursts and generator bursts See the Analog Generator chapter for details of the SR1 s analog generator burst modes L AES Reference In Use this input to lock the SR1 s internal clock to an external digital audio reference signal as defined by AES11 2003 Configuration of the clock reference is done on the Clock Reference panel M Digital Signal Monitor Out Chs A and B The digital monitors output an amplified version of the digital audio input signal The sca
272. ns where it might be desirable to lock the unit modifier selection For instance we might want the display to readout as 0 0012 mVrms To do this select the appropriate modifier from the right click menu Select Auto from the right click menu to return to normal operation The second feature offered by Unit Displays is selected in the bottom half of the right click menu In Standard mode the Unit Display displays each new value that is sent to it If one of the other modes are selected the current value of the display is saved as the reference and subsequent updates are displayed as follows Mode Operation S Standard Each update to the Unit Display s value is displayed Data is displayed in Green Updates are displayed only if they are greater than the reference value When the data exceeds the reference that data becomes the new reference value Data is displayed in Red Updates are displayed only if they are less than the reference value When the data is less than the reference that data becomes the new reference Data is displayed in Blue Delta The difference between the update value and the reference value is displayed Data is displayed in Purple To reset the reference double click the display Like unit entries it is possible to see the value represented by a unit display with full double precision by holding the cursor over the unit display the double precision value will be shown as a hint 2014 Stanford
273. nstead on the Digital Audio Status Panel at the bottom of the Digital I O panel Received status bits can be highlighted to call out differences between transmit and receive status or to indicate differences in received status between the two channels Highlighting is controlled from the Digital I O panel Differences are highlighted in yellow 2014 Stanford Research Systems o SR1 Operation Manual 2 3 6 User Status Panel The user status bits panel displays the transmitted and displayed user status bits for both channels grouped as 23 bytes Status Bits m ai o fox foo a fou Joo fox foo oo foo Joo ov foo oo foo Joo foo oo on fm Joo on fo Fem foo foo fe foo foo foo foo foo foo fo foo foo fon foo foo foo foo fon foo fon fs fon foo bja joo joo foo fa fon foo foo foo fog foo foo foo fog foo foo fon fon foo fon fo fo fon fo Rb SSG Ee Cee ee ee eee l 13 ee eee ee ee eed See ee a Since no universal standard for the formatting or meaning of the user status bits transmitted with digital audio data the status bits are presented as raw hexadecimal numbers The byte number starting from the beginning of the digital audio block is given in red on the lowest line Within each byte the most Significant bit of the hexadecimal word corresponds to the user bit furthest from the start of the block Thus setting the value in byte 1 to 80 sets user bit number 15 out of 184 high The top line of values for each channel allows
274. nt 2014 Stanford Research Systems 2 6 6 2 2 6 6 3 SR1 Operation 237 Eject Drive Select Eject Drive to safely shut down a USB drive connected to SR1 before removing it Drive Eject Drive List E USB Mass Storage Device Kingston DataTraveler 2 0 USB Device Ge Refresh Select the drive to be shut down and press Eject before removing the drive Power Options Selecting Power Options allows setting the idle time interval before SR1 s computer shuts off the monitor or hard disk Any mouse movement or keyboard activity will turn on the monitor and or hard drive after they have been turned off Select the power scheme with the most appropriate settings for this computer Mote that changing the settings below will modity the selected scheme Power schemes Home Office Desk Settings for Home Office Desk power scheme Turm off monitor After 30 mins Ww Turn off hard disks After 30 mins Ww 2014 Stanford Research Systems 238 SR1 Operation Manual It is recommended that users do not modify any of the other power settings on this panel 2 6 6 4 Date Time Select Date Time to set the date time and time zone for SR1 s internal computer Date and Time Properties Date amp Time Time Zone Internet Time 13 14 15 16 20 21 23 27 28 29 30 i2 08 05PM Current time zone Pacific Standard Time 2 6 6 5 Virtual Keyboard The virtual keyboard provides the fun
275. ntent takes many repeated shots to converge to an accurate result or swept measurements which require a separate measurement at each frequency point the chirp sources allows accurate measurement of frequency response in one single FFT acquisition For instance the plot at left below shows the single shot FFT response of the chirp source when passed through a 8 pole 6 zero analog filter with a 5 kHz cutoff frequency dBYrms 30 40 50 60 70 80 90 100 110 120 130 0 2 5k 5 0k 7 5k 10 0k 12 5k 15 0k 17 5K 20 0k 22 5k 25 0k 27 5k M2 Elliptical Filter DUT Output with Chirp Input 2014 Stanford Research Systems SR1 Operation 139 2 5k 5 0k 7 5k 10 0k 12 5k 15 0k 17 5k 20 0k 22 5k 25 0k Hz Normalized Frequency Response of Elliptical Filter If desired the original chirp signal can be acquired first without passing it through the DUT and saved as an Offline trace Then after the spectrum through the DUT is acquired the graph s Ratio function can be used to normalize the output to the input producing a true frequency response plot such as the one seen above right The same process of acquiring a reference input plot and then normalizing the output to it can be used to measure the phase response of devices in a single FFT shot however the process is so much simpler using the dual channel FFT analyzer FFT2 that we ll postpone the discussion of phase response until then Because the synchronous chi
276. ntly implemented Video NTSC Selects a standard NISC composite video signal connected to the rear panel Video Ref In BNC connector The expected frequency is 15 7343 KHz Video SECAM Selects a standard SECAM composite video signal connected to the rear panel Video Ref In BNC connector The expected frequency is 15 625 KHz Video PAL Selects a standard PAL composite video signal connected to the rear panel Video Ref In BNC connector The expected frequency is 15 625 kHz Expected Frequency displays the nominal frequency of the selected source For some sources such as the video signals the expected freqeuncy is preset For the other sources the correct value of the external reference must be entered Reference Frequency displays the current frequency of the selected input clock source When locked the Reference Frequency display will match the expected frequency value Checking Lock begins the process of locking to the selected external source After a few seconds the 2014 Stanford Research Systems 106 SR1 Operation Manual Lock Indicator should glow green indicating a successful lock If the indicator does not glow green check that the correct clock is selected the expected frequency is correct and the external clock Signal is free of excessive noise and jitter If lock is not checked SR1 uses its own internal crystal oscillator as a clock source The phase lock loop attenuates jitter of the reference signal by 6 dB octave
277. ntrol Options 2014 Stanford Research Systems 192 SR1 Operation Manual 2 5 1 Graph SR1 s graphs are sophisticated tools for displaying and analyzing SR1 s measurement results Any number of graphs with unique content can be placed on different pages of SR1 s page control Each measurement is represented on the graph by a trace Each trace can have its own unique Xand Y axis and scaling parameters Although there is no limit to the number of traces that can be displayed on the graph only two pairs of Xand Y axes corresponding to the last two selected traces are shown at once so to avoid cluttering the display There are four basic types of graph traces FFT type traces show a vector measurement such as FFT amplitude or phase with the measurement defining the X axis Thus an FFT amplitude trace will be shown with frequency on the X axis while an FFT time record trace will be displayed with time on the X axis Stripchart traces show a scalar measurement such as THD N as a function of time with a scrolling stripchart type display Sweep traces show the data from an SR sweep with a fixed X axis that is defined by the sweep and data values that are defined by on of the selected Sweep Data measurments Finally EQ traces show the frequency response magnitude or phase of an EQ file Each trace on the graph is assigned a unique color when it is created the color may be changed later The color of the trace data matches the col
278. o Carrier Measurements Measurements Sample Rate Sample Rate Accuracy Carrier Amplitude Measurements Balanced XLR Unbalanced BNC Optical Output to Input Delay Range Resolution Residual Jitter 50 Hz to 100 KHz Optional Digitizer Sampling Rate Acquisition Length Measurements 2014 Stanford Research Systems Carrier amplitude sample rate jitter amplitude jitter soectrum 24 kHz to 216 kHz 5 ppm 5 80 mV 5 20 mV Displays voltage of Toslink receiver Measures delay from Digital Audio Output or AES11 reference output to Digital Audio Input 12 7 Ul to 115 1 Ul in seconds 60 ns 600 ps 80 MHz Ak 8k 16k 128k 256k 512k 1M 2M samples Input vs time jitter vs time input spectrum jitter spectrum pulse width rate histograms jitter probability histogram eye diagrams 304 SR1 Operation Manual General Computer Interfaces Reference Input Sources Reference Output Format Video Out Power Dimensions Weight Warranty GPIB RS 232 Ethernet COM All instrument functions can be controlled AES3 24 Hz to 216 kHz sine or TTL 8 kHz to 32 MHz video NTSC PAL SECAM AES3 24 Hz to 216 kHz VGA output for driving external monitor lt 275 W 90 VACto 264 VAC 47 Hz to 63 Hz 17 x8 5 x 20 25 WHD 40 Ibs One year parts and labor on defects in materials and workmanship 2014 Stanford Research Systems SR1 Reference 305 3 4 Filter Reference SR1 s a
279. o FFS depending on the value of the V FS Volts Full Scale set in the analyzer references tab and the fixed relationship of peak voltage to peak to peak voltage for a sinewave If the value of V FS is 1 Vrms for instance then an amplitude of 2 0 Vpp would correspond to an amplitude of 1 Vrms which in turn would correspond to an amplitude of 1 FFS The peak value of the waveform expressed as a decimal code The conversion of decimal code to FFS depends on the setting of the Input Resolution in the Digital I O panel For the default 24 bit input resolution setting 1FFS 223 1 8 388 607 dec The peak value of the waveform expressed as a hexadecimal code The conversion of hexidecimal code to FFS depends on the setting of the Input Resolution in the Digital I O panel For the default 24 bit output resolution setting 1FFS 279 1 Ox fffff hex Decibels relative to the full scale amplitude definition See FFS above For instance 0 1 FFS 20 dBFS 20 log10 0 1 Decibels relative to the Vrms value calculated with the V FS Volts Full Scale reference set in the analyzer references tab For example with a V FS value of 2 Vrms an amplitude of 20 dBVrms corresponds to 0 1 Vrms which in turn corresponds to an amplitude of 0 05 FFS Decibels relative to 0 7746 Vrms Hiistorically the value of 0 7746 Vrms represents the voltage required to dissipate 1 mW ina 600 Q load If V FS is set to 1 Vrms then an amplitude of 0 dBu corresponds to 0 7746
280. o work with the frequency domain Jitter Analyzer The chirp signal outputs an equal amount of power in each FFT bin of the Jitter Analyzer yielding a flat jitter spectrum When a device is placed between the digital audio output and input the jitter susceptibility at all frequencies can be simultaneously measured eliminating the need for time consuming swept measurements The maximum amplitude for all jitter waveforms is 13 UI Note that all jitter amplitudes are set in peak seconds or peak Uls The Ul is the smallest time scale on which the digital audio carrier signal changes The length of a Ul can be derived from the sampling frequency by the formula Ul 1 Fs 128 The number 128 can be arrived at by remembering that each digital audio frame contains 2 subframes left 2014 Stanford Research Systems s SR1 Operation Manual and right each with 32 bits and that each bit represents 2 Uls in biphase encoding 2 32 2 128 Jitter EQ Sine and square jitter can be set to have a variable amplitude as a function of frequency by specifying a Jitter EQ file EQ files are XML files which specifiy a relative frequency response as a function of frequency by either interpolating a table of frequency response pairs or by calculation from a set of pole and zero locations The structure of EQ Files is detailed in the File Reference section Use the oa button to open a file dialog to specify the EQ File Although the jitter amplitude contr
281. oduces both scalar and vector output measurements which are described below Measurement Description Vector Measurements This is the raw time record received by the MTA While not useful in and of itself Time Record A B Viewing the time record can be useful in diagnosing measurement setup issues Linear Magnitude This vector is the complete FFT of the input signal to the analyzer including all A B bins THD Noise Bins This measurement is the vector of all bins that that do not contain a tone in the AIB i generator signal The amplitude in these bins represents the sum of noise and distortion products 2014 Stanford Research Systems 166 SR1 Operation Manual Harmonic The vector containing only the bins that represent harmonics of tones present in Distortion Bins A the generator signal B The vector containing intermodulation products of the tones up to the order IMD Distortion specified on the analyzer panel 2nd order IMD products for instance fall at Bins A B frequencies equal to f f where f and f are tones in the generator signal Note that not all IMD products of a given order may fall within the analysis range This vector contains bins that contain no tones or distortion products If Noise Analysis is enabled this vector will contain all the odd bins in the analyzer FFT Noise Bins A B If noise analysis is not enabled the analyzer examines each bin and only includes bins in the noise ve
282. of rising and falling samples to produce triangle like output waveforms Because the runs are restricted to integer number of samples the Ramp Frequency and Ramp Fractional Rise Time have limited resolution which is a function of the selected generator sample rate The lowest amplitude sample has the value assigned in the Low 2014 Stanford Research Systems SR1 Operation Amplitude control The highest amplitude sample has the value given in the High Amplitude control The ramp waveform generates a generator trigger each time the ramp begins its rising segment Arbitrary Waveform Config rb aveform Arb On o cHIRPIK r Amp 500 00 mFFS Output Rate e R g The arbitrary waveform plays a sequence of values found in a user supplied table Arbitrary waveform files are simple ASCII files with one or more columns of floating point numbers representing the values of the arbitrary waveform at each sampling interval Use the a button to open a file dialog to select an arbitrary waveform file If multiple columns are detected in file SR1 displays a dialog asking which column to load The number of table points read from the table is then displayed in the corresponding control The amplitude entered in the Arb amplitude control is assigned to the maximum value found in the table The absolute scaling of the table values does not affect the output waveform However when operating at arbitrary output rates the SR1 int
283. of measurement i e an rms integration over the waveform In SR1 this is not the case the measurement method is specified separately and the units of measurement always have the same fixed sinewave relationship An example will help to clarify this Consider the following square wave 1F 1 With an RMS voltmeter the square wave shown would have an amplitude of 1 Vrms With a peak responding meter the square wave would be found to have an amplitude of 1 Vp or 2 Vpp However with SR1 things are slightly more complicated in that it s necessary to know the measurement method as well as the units to correctly interpret the answer If SR1 s Time Domain Detector was configured for and RMS response and the amplitude units set to Vrms SR1 would perform the RMS amplitude calculation and display an answer of 1 Vrms However if the units of the answer were changed to Vp the answer would be 1 414 Vp because that s the sinewave relationship between Vp and Vrms Likewise if the Time Domain Detector is configured for Peak response SR1 will display a result of 1 Vp but if the units of the display are changed to Vrms the display will read 0 707 Vrms 2014 Stanford Research Systems SR1 Operation In other words an answer displayed by SR1 in Vp does not imply that a peak measurement is being performed nor does an answer in units of Vrms imply that an rms measurement is being made The units will always have their sinewave ratios but the method of
284. of the frequency response points This value must be listed in kHz not Hz Finally the y values corresponding to each of the frequencies listed is described by lt EQdata Name Resp Type DblArray Value 0 0003007759332 72953 0 003021 74095251185 0200932540277 40164 0 016 7511398949463 030636637564135 0 0444168356181221 0 0596245322111266 0 075876407055739 0 0928635700095015 110342389581799 0 286838481522557 0 443881655056602 0 577159672164201 0 688018968891406 778916516580281 0 852785568914054 0 91256030198955 0 960885423147423 1 00000000003858 14837827316987 1 15195959556707 1 11740718036267 1 06603938913942 1 00595593739291 941761216135588 0 876545831584705 0 8124486702114 0 750902168765253 0 341162925654635 178593707588384 0 107123333734722 0 0707303220476656 0 0499780417068754 0 037110212737056 0286105664352016 0 0227144341264837 0 01846210568194 0 00466695600216974 gt oOo OO OF CO CO O Obviously the lists of frequency values and response values must be the same length When interpolating in a sampled EQ file if the required frequency is less than the value of the lowest frequency point in the list the response corresponding to this lowest frequency point is given Likewise if the response is required at a value greater than the highest point in the list the value for the highest point in the list is returned Digitizer Files TXT SR1 digitizer files are two column ASCII text files The first column displ
285. of the log sine chirp The log sine chirp waveform outputs a generator trigger at the beginning of each FFT time record Be sure to use generator trigger as the analyzer trigger source when using the log sine chirp waveform with the FFT analyzer 2014 Stanford Research Systems 54 SR1 Operation Manual MultiTone Config MultiTone aveformi On fw EG Export Like FFT Chirp the MultiTone Waveform is a special waveform containing a series of tones that are generated in synchronous fashion to ensure bin center placement in the MultiTone Analyzer Using the Multi Tone generator and analyzer many different characteristics of a devce noise distortion frequency response can be measured quickly and simultaneously without the need for multiple swept measurements All of the details of the multitone waveform tone frequency and amplitude phase etc are configured using the MultiTone Configuration Panel As a result only the amplitude is specified on the waveform tab The Export button allows the current multitone waveform to be saved in arbitrary waveform format for off line analysis Because of the special properties of the MultiTone waveform it may not be combined with other waveforms The Multitone waveform cannot be selected from the analog generator panel unless the MultiTone configuration panel specifies the use of the analog generator Multitone Configuration Generator Domain Analog Cute The M
286. ol will continue to display the nominal amplitude value the actual output jitter amplitude will be the nominal value multiplied by the EQ file response at the current sine or squarewave frequency Rear Panel Clock Jitter if enabled applies the currently selected jitter to the rear panel Master Clock Output If disabled the Master Clock outputs the un jittered digital audio bit clock 2014 Stanford Research Systems 2 3 5 Channel Status Panel SR1 Operation The channel status panel displays received channel status information and controls the transmitted channel status information Channel status information is organized according to either the AES EBU professional standard or the SPDIF consumer standard SR1 relies on AES3 2003 and AES 2id 2006 for the professional standard and IEC60958 3 as the source for the consumer standard 5 Channel Stat Bits gt A Bi f AIB f Consumer f Professional Sample Not Linear PCM ha Cp bit Copyright Mot 4sserted Emphasis No Pre emphasis sd Category General Prerecorded Indicator Src NS Chat Mya Fs 48 kHz ad Clock Acc Level II ad Word Len Not Indicated Orig Fs E kHz sd aje fo fo fe fm fo fm fo fo mo fo foo foo foo ooo ooo foo oo foo om oo N E E E E a a E E a E E a a E a bj foo foo fe fa fo fa fa foo fou m fo fo fon foo foo fo fa foo foo foo fa foo foo b ERCECE Cae Ee Ce ee eee 2014 Stanford Research Systems mmx nein m E oeoa a
287. omatically Otherwise enter the IP address subnet mask gateway and DNS information manually If none of this makes any sense contact your network administrator for the correct settings 2 6 5 2 Network Places Selecting Network Places displays folders files and computers located on the network Click Add Network Place to specify new network locations Select Entire Network to browse the entire accessible network 2014 Stanford Research Systems SR1 Operation 233 Local Network srsprinters on Modoc modach Unspecified Add Wetwork Place Entire Network 2 6 5 3 Map Network Drive Select Map Network Drive to assign a drive letter to a folder on the network Enter a drive letter name and browse the network to the desired folder If Reconnect at logon is checked the mapping will be made permanent Press Finish to confirm the drive mapping Map Network Drive Windows can help you connect to a shared network Folder and assign a drive letter to the connection so that you can oe access the Folder using My Computer ae Specify the drive letter For the connection and the folder that vou wank bo connect to Example server share Reconnect at logon Connect using 4 different user name Sign up For online storage or connect bo a network server 2 6 5 4 Remove Network Drive Select Remove Network Drive to undo a drive mapping Click on the drive mapping and press OK to
288. omation Menu Scripts generated with Learning Mode may require some manual polishing to be completely reusable but learning mode can still save considerable time when developing a substantial script 2014 Stanford Research Systems 2 8 SR1 Operation 253 Quick Measurement Menu Most of SR1 s panels are function centric in the sense that they group controls related to a single function or feature of the instrument Thus the analog generator panel contains controls relevant to the operation of the analog generator the settling panel groups settling controls for all measurements etc The Quick Measurements menu is different in that includes a collection of panels that are measurement centric rather than function centric i e each panel contains controls that pertain to a common audio measurement and allows for setup of all the different aspects of the instrument related to that measurement from one convenient location Quick Measurements enable the user to quickly setup a measurement with little experience with SR1 but does not expose the full possibilities power of all of SR1 s features and controls All Quick Measurements rely on the Quick Measurement Setup Panel which records the basic format for the tests whether the inputs and outputs are analog or digital the bandwidth of the measurement mono or stero etc Based on these settings each measurement panel then configures the inputs generators and analyzers appropriately for the c
289. omposed of a low frequency sine wave and a higher frequency sinewave at 1 4 or equal amplitude The measured distortion products are the sidebands of the high frequency sine at multiples of the low frequency Difference frequency distortion DFD the subject of an old standard by the CCIF the predecessor of the ITU R consists of applying two equal amplitude high frequency sines separated by a small frequency difference to the device under test and measuring the nonlinear distortion products at the difference frequency and higher order combinations of the two frequencies The modern standard for DFD is IEC 60268 3 Finally DIM Dynamic Transient Intermodulation distortion also known as TIM Transient Intermodulation Distortion uses a square wave near 3 kHz and a sine wave near 15 kHz and examines the distortion products at various combinations of the two frequencies All of these measurements have a long history and many pages have been written for and against each of them SR1 can perform all these measurements by first generating the appropriate stimulus using the generator IMD waveform and then selecting the IMD analyzer to perform the measurement Summary of IMD Analyzer Outputs Measurement Description Ratio Selected Ch The relative IMD distortion for the selected measurement type and distortion products Displayed in either percent or dB Level A B Peak based level computation for both input channels Because the IMD Analyzer is
290. omputing the power for each spectrum taking the absolute value of the complex FFT points and averaging that power into the power computed for previous FFTs This type of averaging does not reduce the noise floor of the spectrum but it does reduce the variation of the noise floor making it easier to see spectral details on the order of the noise amplitude Phase information is lost when computing the Power Spectrum In the example below the unaveraged power spectrum is shown for a signal composed of a 1 kHz sine wave with added white noise The second spectrum shows averaging on and Navg 10 Note the substantial reduction in the variation of the noise floor and note also that the average value of the noise floor is the same dBvrns 0 Sk 10k 15k 20k 25k sok Hz Unaveraged Power Spectrum of Sine Noise dEYrms o 5k 10k 15k 20k 25k 30k Me Averaged N 10 Power Spectrum of Sine Noise The second spectral output computed by the FF T2 analyzer is the Linear Spectrum for each channel The Linear Spectrum is computed by averaging the real and imaginary parts of each FFT separately The average of the real and imaginary parts are then used to compute the Linear Spectrum amplitude and phase In the Linear Spectrum unlike the Power Spectrum noise that is uncorrelated to the signal is actually reduced by further averaging Because of this use of the Linear Spectrum unlike the Power Spectrum requires that the time record be triggered so that the sig
291. on of Keypad Keys Key Funcion s Numeric Entry Keys O Backspace Deletes the character to the left of the cursor woo Inserts the E character at the current cursor position Sewing entry in ed Hoos notation e g 1 2E3 Aborts numeric entry and returns the value in the selected field to the previous value inserts the u modifier at the cursor position interpreted as x10 inserts the m modifier at the cursor position interpreted as x10 kilo inserts the k modifier at the cursor position interpreted as x10 Mega inserts the M modifier at the cursor position interpreted as x10 Finishes entering the new value Function Keys Rotates through the seven pages of the page control on the SR1 screen Cah i Moves the control with focus either to the next or previous control Tab Left Right Sweep l l A Starts a Sweep Equivalent to pressing the button on the speedbar Pause Pauses and resumes both sweeps and free run mode Equivalent to pressing the CJ and te buttons on the speedbar o Starts free run mode Equivalent to pressing gt l on the speedbar Translate graph left right Once pressed the lt LD keys will move the data in the current graph left and right Press Esc to exit this mode Zoom X axis Once pressed the Ca Ca keys will zoom the X axis in and out Press Esc to exit this mode AutoScale Autoscales the Xand Y axes of the currently selected graph the maximum va
292. onnector Receive Block Triggers on any block preamble Z preamble on the rear panel reference input Ref Ref A B Triggers on the occurrence on the rear panel reference input of any of the error conditions listed in the table below Triggers on the transmission of a channel A subframe preamble X preamble by the digital audio output generator Triggers on the transmission of a channel B subframe preamble Y preamble by the digital audio output generator 2014 Stanford Research Systems SR1 Operation 185 Xmit Block Triggers on the transmission of a block preamble Z preamble by the digital audio output generator Ref Out Triggers on the transmission of a channel A subframe preamble X preamble at the Preamble A digital audio reference output Ref Out Triggers on the transmission of a channel B subframe preamble Y preamble at the Preamble B digital audio reference output Ref Out Block Triggers on the transmission of a block preamble Z preamble at the digital audio reference output Triggers on the receipt of a generator trigger See the Analog and Digital generator sections External rising Triggers on the rising edge of a signal at the rear panel external trigger in connector Triggers on the falling edge of a signal at the rear panel external trigger in connector falling Error Trigger Conditions The Error Trigger triggers on the receipt of any of the following error conditions detect
293. onnector 293 n G n Eye Diagrams 209 Generator References 36 57 Generator Trigger 43 62 Generator Analog 2014 Stanford Research Systems 314 SR1 Operation Manual v2 Generator Analog Amplitude Controls 36 Maximum Output Limit 215 Mono Stereo Selection 36 Output Configuration 36 Panel 36 Sampling Rate 36 Units 41 Waveforms 43 Generator Digital Audio Adding Waveforms 5 7 Amplitude Controls 5 7 Mono Stereo Selection 57 Panel 5 7 Units 60 Waveforms 62 GPIB 218 241 GPIB Connector 293 Graph Adding a Trace 193 Cursor Display Bar 197 Cursor Tab 197 FFT Cursors 198 Limits 198 Panel 193 Scaling 196 Speedbar 195 Trace Calculator 199 Trace Listing 193 X Axis 196 Graphs 192 H Hardware Status 230 Harmonic Analyzer 157 Headphone 100 Headphone Jack 291 Help 288 Hi Bandwdith 108 Hi Res Converter 76 Hi Resolution 108 Histogram Analyzer 171 Averaging 173 Bins 173 Fit 174 Measurements 1 71 Panel 1 72 Range 173 Sample Rate 173 Scaling 173 Size 173 l IEEE 488 241 Setting Address 218 IMD 62 IMD Analyzer 161 Averaging 163 Generator Selection 162 IMD Products 162 Panel 162 Weighting 164 IMD Distortion 267 IMD Waveform 43 Impulse Response 141 144 Input Autorange 76 Input Connectors 76 Input Impedance 76 Input Range 76 Input Sampling Rate 76 79 Internal Sweeps 93 ISR 79 J Jitter Analyzer 177 Averaging 181 Bandwidth 181 Block Diagram 1 7 Filtering 181 Frequ
294. ons the analyzer provides a validation option which is a specialized form of analyzer trigger applicable only to the Multitone Analyzer The validator works by assigning a score to each received FFT based on the ratio of the amplitudes of tone bins to non tone bins Only received FFTs that score higher than the selected validation criteria Loose normal or tight will be processed by the analyzer In general it is best to use the tightest validation criteria that still finds the multitone signal as it will generally produce the best measurement results 2014 Stanford Research Systems SR1 Operation 171 2 4 8 Histogram Analyzer The Histogram Analyzer calculates input amplitude histograms of both channels of the selected input domain The histogram analyzer divides the input amplitude range into a user specified number of bins The histogram is formed by examining the amplitude of samples of the input data stream determining the bin the samples belongs in and incrementing the amplitude of that bin This process continues for a predetermined number of input samples after which the histogram is displayed and the process repeats SR1 s Histogram Analyzer also has the capability of calculating a real time gaussian fit to the histogram data and displaying it alongside the histogram Histogram Analyzer Outputs Time Record A B The sequence of input samples for each input channel Histogram A B The input amplitude histograms for each input cha
295. or EJ Multitone Configuration Select the portions of the instrument configuration to save and click OK to display the standard SR1 file save dialog box 2014 Stanford Research Systems SR1 Operation 31 2 1 3 Load Configuration File gt Load Configuration loads the instrument s configuration based on the values found in the file selected with the standard Windows file open dialog box After selecting the file a dialog is displayed allowing selection of individual configuration areas The default is to load all the configuration information in the file Only configuration areas found in the selected file have a clickable checkbox the rest are grayed out Sit instrument gt E Analog Generator E Digital Generator O Multitone Configuration ED Digitizer z O 4nalog Input z O Snalyzer References O Clock References E Digital 1 0 Cancel Select the portions of the instrument configuration to load and click OK 2014 Stanford Research Systems SR1 Operation Manual 2 1 4 Print SR1 Screen File gt Print SR1 Screen prints the full main window of the program including menus and borders to the current printer Printer and paper options are selected from the File gt Print Setup dialog Note that File gt Print SR1 Screen always prints the entire screen exactly as it appears Other options are available for more formatted printing and exporting of data from displays 2014 Stanford R
296. or of the corresponding axes as well as the color of the trace listing in panel on the lower right of the graph When the cursor is clicked on the trace data or on the corresponding line in the trace listing panel the trace becomes the active trace and it s axis becomes visible if it had not been previously visible and the trace listing line is highlighted The active trace is the trace whose scaling data is shown in the scaling tab at the lower left of the graph and is connected to the cursor display Traces on a graph can be live i e connected to a currently updating measurement or offline where the data in the trace is static Offline traces are useful for storing reference data or for creating limit traces which can be compared to live measurement values and produce a pass fail type result 2014 Stanford Research Systems SR1 Operation 193 The Graph Display Panel Graph Cursor Display Bar Speedbar Graph O i amp 5 x ASS HDE Tti ALAA MAY SmE 21575k SSE 27o I A G eee ES 2 Active Cursor 0 2 ok 5 0k 75k 10 0k 12 5k 15 0k 17 5k 20 0k 225k Hz Scale cursors FFT Cursors Limits Al FFT Power Spectrum 4 41 FFT Time Record 4 Min m ax hi 0 00000 Hz ba 24 0000 kHz J Y 70 0000 dBFS 10 0000 dBF5 ba E Lock Axes S Sweep Append E Scaling X axis and Sweep Controls Controls Trace Listing Trace Listing The panel at the lower right of the graph shows the trace
297. ortion Sine as specified in the panel For free run operation the amplitude and frequency of the generator is set to the frequency and level in the start column of the panel For swept operation the frequency is swept from the start to stop value for each amplitude value specified Analyzer Configuration One or both analyzers depending on the number of output channels specified in the Setup Panel are configured as Time Domain Detectors The upper and lower bandwdith limits are adjustable in the Analysis section of the panel Free Run Configuration A barchart display is created for each output channel showing the instantaneous output level of the EUT Sweep Configuration A graph will be created for each channel showing a graph of output level vs Freq for the selected number of amplitude steps Log spacing can be selected for both the frequency and amplitude sweeps The settling parameters of the Level measurement can be adjusted directly from the Settling tab of the panel 2014 Stanford Research Systems 260 SR1 Operation Manual Settling Profile Delay Exponential e 20 000 msec Precesion nPoints Threshald p A000 So 3 p 000D mrm Refer to the Settling Panel chapter for a more detailed discussion of what each of the controls means n general specifying a smaller precision window and larger value for nPoints will decrease the noise and glitches in the sweep at the expense of increasing the time required for the sw
298. ower averaged FFT of the jitter signal Linear Magnitude The amplitude of the synchronously averaged jitter spectrum The amplitude of signals that are synchronous with the time record is preserved other 2014 Stanford Research Systems SR1 Operation 179 uncorrelated signals average away Note that the signal must either be naturally repetitive within the time record or a trigger must be used for the linear magnitude spectrum to be meaningful The phase of the synchronously averaged jitter spectrum The amplitude of the jitter signal calculated from an RMS sum of the FFT bins Though the amplitude is calculated with an RMS sum it is expressed in equivalent peak units of sec or Ul Physical Sample The effective sampling frequency of the input digital audio signal or in the case of Rate a square wave input the square wave frequency Jitter Analyzer Input Selection Input Contig Cnctr Buc elca Dual Conn EQ Sgr w Term Unlike other SR1 analyzers selection of the input signal for the Jitter Analyzer is done on the Input section of the Digital I O panel Select either the BNC or XLR connectors optical connector or the digital audio output monitor For BNC and XLR connector C1 or C2 must also be specified Checking Term applies the appropriate termination for the selected connector Selecting Square Wave instructs the Jitter Analyzer that the signal at the selected input is not a AES EBU or consumer digi
299. perates between two user specified values The generator outputs each value for the number of samples specified in the Samples Point control before incrementing the value by one When the final value has been output for Samples Point samples the counter output resets to the initial value Rotate Bits Config Potatebits Waveform Omn a Rotate Bits Zeros Ones Dwell 10000 The Rotating Bits waveform outputs a pattern composed of all zeros and a single one if Ones is selected or all ones and a single zero if Zeros is selected The pattern is output for the number of 2014 Stanford Research Systems SR1 Operation Manual samples specified in the Dwell control and the the pattern is shifted left by one bit When the one zero is shifted out of the leftmost bit of the digital audio word it rotates back into the rightmost bit of the word Using the Rotating Bits generator and the Active Bits display on the Digital I O panel is a simple method for detecting stuck or cross linked bits in digital audio equipment Digital Staircase Config DigitalStair waveform Omn ly sStaicase Samples Point 10000 The Digital Staircase waveform is a special digital only waveform which is useful for testing D A converters The staircase waveform outputs 5 complete square wave cycles of the ten smallest digital codes in succession followed by an equivalent interval of digital zero The length of each half cylc
300. ple of one another If the generator and analyzer sampling frequencies are incompatible the Compliance led will glow red and no waveform will be output The compliance led may also show red if SR1 runs out of table memory to create a long chirp Signal say for a large number of FFT lines or when large amounts of fft zoom are used The chirp waveform may attempt to use interpolation in situations when not enough memory is available for the complete table in which case the Compliance led will glow yellow Some degradation of flatness can be expected in this case Using the chirp signal it is possible to do cross domain measurements of both amplitude and phase For instance when testing a D A converter a chirp can be output in the digital domain and the D A frequency response and phase recorded in the analog domain See the SR1 Applications Manual for more information To ensure cross domain sampling rate compatibility use the following sampling rate selections for the generator and analyzer 2014 Stanford Research Systems SR1 Operation 6 Type of Measurement Analyzer Fs Analog Source Select Digital ISR as the analog Select Digital as the Analyzer source Digital Measurement generator Fs which automatically chooses ISR as ADC the analyzer Fs Digital Source Analog Select the desired digital audio Select the Analog Hi Res converter on Measurement D A output frequency using the Digital the analyzer panel and set the Hi Res I O pa
301. plied to the rear panel TTL burst trigger input When the external signal is lo the generator output is at the high amplitude when the external signal is high the generator runs at the lo amplitude Burst Period and Burst Duty Cycle are ignored Shaped Similar to Timed Burst except that a cosine squared window is applied to the signal through the hi portion of the burst Triggered The output is zero until a TTL rising edge is detected at the rear panel burst trigger input The output then goes high for the interval specified by Burst Period Lo Amp selects the lo burst amplitude as a fraction of the original generator amplitude For Triggered 2014 Stanford Research Systems SR1 Operation oat Bursts the lo amplitude is fixed at zero Burst Period controls the duration of triggered bursts and the total on off period for Timed and Shaped bursts Burst Duty Cycle controls the fraction of the burst period during which the generator output is at the hi amplitude EQ Controls EQ None l o Invert EQ Certain waveforms can have their amplitudes scaled as a function of frequency according to the information contained in an EQ File EQ files are XML files which specify a relative frequency response as a function of frequency by either interpolating a table of frequency response pairs or by calculation from a set of pole and zero locations The structure of EQ Files is detailed in the File Reference section Use the a butto
302. positioning controls Stark Center End 103 00 kHz oF 128 00 kHz oF 153 00 kHz I ee eee The position of the FFT analysis range can be set with any of the Start Center or End controls the values of the other two will automatically adjust according to the selected bandwidth The graphical indicator below the controls shows the position of the analysis range relative to the full DC to Fs 2 frequency range Aliasing Aliasing arises in sampled data systems because the real world filters used to protect A D converters have finite cutoff slopes Each of the SR1 s A D converters is protected with either analog or a combination of analog and digital anti aliasing filtering For each converter there is some range of frequency below Fs 2 that is not fully protected to the level of the ultimate attenuation of the anti aliasing filter When using the zoom feature of the FFT1 analyzer the analyzer applies digital anti aliasing filter after each decimation stage meaning that there is a small region at either end of the analysis range that is not alias protected to the full attenuation of the digital anti aliasing filter When Show Aliased Lines jig checked the analyzer displays all lines in the full analysis range of the FFT without regard to whether they are fully alias protected This setting is often perfectly useful for many typical audio measurements However when aliasing is potentially a problem the box can be unchecked in whi
303. put Board B 30161 Analog Input Board A 20238 Analog Input Board B 20231 Reference Waveform LD Sine Freq 1 0000 kHz Level 1 0000 Vrms High Pass lt 10 Hz Low Pass Fs 2 Reference A 0 99793 Vrms Reference B 0 99775 Vrms Level Waveform LD Sine Start Freq 10 000 Hz Stop Freq 50 000 kHz Start Level 1 0000 Vrms High Pass lt 10 Hz Low Pass Fs 2 Level A amp B vs Frequency dBrA dBrB e plitude A Sweep A1 Time Dom Det A 0 0 LOM 10m 20m 20m 30M 30m 40m 40m 10 20 50 100 200 500 1k 2k 5k 10k 20k Hz 10 20 50 100 200 500 1k 2k 5k 10k 20k Hz To display the report after measurements have been made select Report from the Quick Measurements menu The report preview window includes options for page navigation zooming and scaling printing and exporting the report to PDF or HTML format The Automated Measurement Panel Autom Meas Actions Setup w SMR Reference Level w THOM Freg Resp w Distortion w IMO Crosstalk Interch Phase InfOut Phase pen Report Update Report Clear Report 2014 Stanford Research Systems SR1 Operation 277 The Automated Measurement Panel allows selection of a group of quick measurements to be performed sequentially The measurement results for each selected measurement are accumulated in a report To clear the report in preparation for a new round of measurements press Clear Report To include a text note in the report enter the text in the
304. put Characteristics Amplitude Range rms Input Configuration Input Impedance Balanced Unbalanced Input Termination bal Crosstalk 10 Hz to 20 kHz gt 20 kHz Hi BW ADC Type Sampling Frequency Frequency Range Hi Resolution ADC Type Sampling Frequency Frequency Range 62 5 mV to 160 V XLR BNC Generator Monitor Digital Audio Common Mode 200 k 95 pF 100 k 185 pF 300 600 none 140 dB 135 dB 16 bit sigma delta 512 kHz DC to 228 kHz 24 bit sigma delta 128 kHz or 64 kHz fixed 24 kHz to 216 kHz adj DC to 0 45 Fs General Digital Input Characteristics Input Format Input Sample Rate Input Impedance Balanced XLR AES EBU dual connector XLR unbalanced BNC SPDIF EIAJ dual connector BNC Optical Toslink 24 kHz to 216 kHz Hi Zor 110 balanced Hi Z or 75 unbalanced 2014 Stanford Research Systems SR1 Reference 301 Analog Signal Meters RMS Level Meter Accuracy 1 kHz ref 0 5 40 043 dB Flatness rel 1 KHz Amp 4 Vrms 20 Hz to 20 kHz lt 0 008 dB typ lt 0 003 dB 10 Hz to 64 kHz lt 0 020 dB 10 Hz to 200 KHZ lt 0 030 dB Frequency Meter Range 8 Hz to 300 kHz Accuracy 2 ppm 10 mHz Phase Meter 1 0 accuracy Digital Signal Meters Frequency Meter 10 Hzto 0 45 F_ 100 ppm accuracy Phase Meter 0 5 accuracy f 50 Hz Analyzers Analog and Digital Audio Time Domain Analyzer Measurements Amplitude amplitude ratio THD N THD N
305. r CCIF measurements the center frequency is set to the specified value while the difference frequency defaults to 1 kHz For DIM measurements the tone frequencies are fixed by the standard The default tone frequencies set up by the quick measurement can always be manually overridden on the appropriate panel Analyzer Configuration One or both analyzers depending on the number of output channels specified in the Setup Panel are configured as IMD Analyzers The measured distortion products are selectable from the Analysis section of the panel Free Run Configuration A barchart display is created for each output channel showing the instantaneous IMD distortion of the EUT Sweep Configuration A graph will be created for each channel showing IMD vs amplitude for the selected range of amplitudes Log spacing can be selected for both the frequency and amplitude sweeps The settling parameters of the THD N measurement can be adjusted directly from the Settling tab of the panel 2014 Stanford Research Systems 268 SR1 Operation Manual Settling Profile Delay Exponential e 20 000 msec Precesion nPoints Threshald p A000 So 3 p 000D mrm Refer to the Settling Panel chapter for a more detailed discussion of what each of the controls means n general specifying a smaller precision window and larger value for nPoints will decrease the noise and glitches in the sweep at the expense of increasing the time required for the sweep
306. r THD or individual Harmonic Amplitude one or both analyzers are configured as THD analyzers depending on the number of output channels selected in the setup panel For this analysis the default is to select all harmonics to be included in the THD measurement If individual harmonic measurement is required the change must be manually made on the THD analyzer panel For Distortion FFT AO is set as a Time Domain Detector and A1 is setup as an FFT analyzer Note that in this configuration only one channel can be analyzed at a time Channel A is the default In order to analyze channel B the source for the AO analyzer must be manually changed to Analog B or press sweep with no freq or level steps This will essentially perform a free run measurement on both 2014 Stanford Research Systems 266 SR1 Operation Manual channels Free Run Configuration For THD analysis a barchart display is created for each output channel showing the instantaneous THD of the EUT For individual harmonic amplitude a graph is created with a trace for each output channel showing the relative harmonic amplitude as a function of harmonic number dB 67 5 SRS 70 0 i i i i i i i 72 5 75 0 100 0 For distortion FFT a graph is created showing the spectrum of the EUT output with the fundamental removed dBYrms 30 40 i Large 50 Subharmonic i i 60 70 80 Notch Filter Profile 90 100 110 120 Residual Fundam
307. r a complete description of all these units Note that there is one set of references for both generator channels Burst Controls Lo amp o o00 To bl Period 000 sec Dty Cy 50 00 Yo Trg The SR1 Analog Generator implements two different types of burst functionality Synchronous Burst provides a burst sine wave with a variety of triggering options where the burst transitions are guaranteed to be synchronous with the zero crossings of the sine wave This type of burst is implemented in SR1 as a separate waveform SR1 also offers the capability of bursting any waveform that can be configured in the generator although with no guarantee that bursting will occur at zero crossings This type of burst will be referred to as generator burst as opposed to synchronous burst Generator Burst Mode selects the burst triggering mode The waveform outputs at the high amplitude then the low amplitude and repeats The total period is determined by the Burst Period control The high amplitude fraction of the period is determined by the Burst Duty Cycle Gated hi The waveform amplitude is determined by the TTL gating signal applied to the rear panel TTL burst trigger input When the external signal is high the generator output is at the high amplitude when the external signal is lo the generator runs at the lo amplitude Burst Period and Burst Duty Cycle are ignored Gated lo The waveform amplitude is determined by the TTL gating signal ap
308. r is encountered during the execution of a script User Event The User event only occurs when it triggered programmatically in a script Use the line Call SR1 EventMgr FireUserEvent to fire the user event Event Tab The events tab on the events panel lists all of the defined events organized by category Tracking of each event is enabled by checking the corresponding check box When being tracked the occurrence of an event will trigger one or more of the actions configured on the events tab If a tone is selected in the 2014 Stanford Research Systems SR1 Operation 223 Tone column the specified tone will be played when the event occurs If Logging is checked the occurrence of the event will be recorded with a time stamp in a log file The logfile can be set and viewed on the Config tab of the events panel An example logfile is shown below for the case where Sweep Start and Sweep Finished are the events being logged SRL Log Opened On 12 16 2009 9 49 02 AM 12 16 2009 10 34 50 AM 12 16 2009 10 35 03 AM 12 16 2009 10 41 23 AM 12 16 2009 10 41 36 4M 12 16 2009 10 41 49 AM 12 16 2009 10 42 01 AM 12 16 2009 10 42 11 AM 12 16 2009 10 42 21 AM 12 16 2009 10 42 59 AM 12 16 2009 10 43 12 AM 12 16 2009 10 43 55 AM 12 16 2009 10 44 09 AM 12 16 2009 10 47 21 AM 12 16 2009 10 47 32 AM 12 16 2009 10 47 47 AM 12 16 2009 10 48 01 AM Sweep Started Sweep Finished Sweep Started Sweep Finished Sweep Started Swe
309. r it is possible to tailor a stimulus with a precisely defined frequency profile EQ files specify a relative frequency response as a function of frequency The format of EQ files is described in the SR1 File Reference When an EQ file is on the Configuration tab of the generator panel the amplitude in each frequency bin will be multiplied by the frequency response of the EQ file at that frequency If the Invert EQ checkbox is checked the amplitude in each bin will be 2014 Stanford Research Systems 52 SR1 Operation Manual multiplied by the inverse of the frequency response of the EQ file When EQ is selected with chirp the Variable Sweep Rate feature is enabled Selecting Variable Sweep Rate changes the chirp sweep rate to equalize the time domain amplitude in a chirp with variable frequency domain amplitude For instance the graph above shows the time response of a chirp weighted with the CCIR EQ curve The yellow trace was taken with Variable Sweep rate off and the green with Variable Sweep Rate on Even though the two chirps have identical amplitudes in the frequency domain notice how more uniform and how much lower the crest factor the variable sweep rate chirp has The lower crest factor makes it much more suitable waveform for audio testing The chirp waveform outputs a generator trigger once each cycle Be sure to use generator trigger as the analyzer trigger source when using the chirp waveform with the FFT analyzer
310. r panel serial port connector may also be used to communicate with SR1 The COM port selection is only relevant when SR1 is being run in demo mode on a PC On the instrument all choice will be grayed out except the the port that is actually connected to the rear panel connector The other serial port options such as Baud Rate Data bits etc should be set according to the device that will be communicating with SR1 Checking the On box enables sending and receiving data over the serial interface VXI 11 Options GPIB VXI L1 serial Core Pork Abort Port Max Links Active Links 1000 ajho ai co o secur F The Core Port is the main TCP IP port on which VXI 11 commands are sent and received The abort port is a secondary port used to send abort commands only to the instrument SR1 does not respond to commands on the abort port The core port and abort port must not conflict with other applications on the computer used to communicate with SR1 Any firewall on the communicating computer must open the main port the abort port and the Sun RPC port port 111 for VXl 11 communications to work Max Links determines the maximum number of simultaneous VXI 11 connections that SR1 will allow Commands received over multiple connections are processed on a first come first served basis 2014 Stanford Research Systems SR1 Operation 219 VXI Security Panel VXI Security Allowed IP Addresses HEE HEHE 192 168 1 25 35 add del repla
311. r selected measurements the 4 tabs of the Digitizer Display always display the same digitizer measurements Each digitizer display can be live meaning that it is updated each time a new digitizer record is acquired or it can be offline meaning that the displayed measurement results are not updated Offline Digitizer Displays can be useful for displaying reference data for comparison with live data E Digitizer Display 0 22222 See 147 993 usec MEA 2 51 231 psec FREER A U GEE a y Li T oe 1 E 7 iW a D mi T E D pa a a m ae o 1 Te i 46u 47u 48u 43u SOU Siu min max x 44 173 Hsec 55 651 HSE im 3 3825 i 3 3825 if The digitizer display has a similar layout to the Graph At the top of the display is a speedbar with buttons corresponding to frequently used functions Below that is a cursor bar with readouts for the two cursors The calculated effective digital audio sampling frequency is shown at the right hand edge of the cursor bar The central portion of the display is a tab control with 4 tabs each of which displays a different set of digitizer measurements The Time Rec tab show the two measurements which are functions of time Input Amplitude and Jitter Amplitude The Spectrum tab displays the spectrum of the input signal and the jitter The Probability tab shows amplitude histograms of the input and jitter amplitudes as well as the distribution of pulse widths and p
312. r voltage limit for the histogram There is no need for the limits to be symmetric Pressing autoscale automatically sets the upper and lower histogram limits to the current analog input range if analog inputs are selected If the selected domain is digital the limits are set to 1 414 FFS The amplitude space between the upper and lower limits is divided into the number of bins selected with the Bins drop down Values between 16 and 512 bins are allowed Histogram Averaging Overagin aig vgs Clear Avg Done Continuous 10 _ Successive histograms may be averaged When Continuous averaging is selected each new histogram is averaged with the previous results in a manner that weights more recent histograms more strongly than older ones The averaging process is updated continuously Single averaging averages N histograms all weighted equally where N is selected with the Avgs control and then stops acquisition Press Clear at any point to clear the averaging buffer and re start the averaging process 2014 Stanford Research Systems 174 SR1 Operation Manual Histogram Fit lw Do Fit Channel 4 Channel B Mean Mean Sigma Sigma When Do Fit is checked the Histogram Analyzer fits a gaussian curve to the to each computed probability histogram The parameters of the gaussian fit the mean and standard dewation are displayed in the panel The graph below shows the histogram of analog white noise along with the generate
313. race selected as the limit is an FFT trace then all FFT traces on the graph may be limit tested but a time record trace cannot be The X axis of the time record is incompatible with the frequency X axis of the FFT Once the trace for the limit has been selected a list of compatible traces appears in the Limit Tab Each trace can be selected for limit testing by checking the box adjacent to the trace name Once selected a green or red box appears indicating whether the trace data passes or fails the limit test If both an upper and lower limit is present the trace must pass both limits in order to pass The red box on the tab indicates the AND of all the pass fail results for the individual traces When first created a limit trace contains identical data to the trace that it was created from Press Edit to change the limit data Note that X values must be in increasing order 2014 Stanford Research Systems SR1 Operation 199 Upper Limit Sele pe o gt ja tla o The limit X and Y values can be all moved together with the arrow buttons at the bottom of the panel This can be useful for creating a limit equal to an existing trace plus some error margin Individual X and Y values in the limit can be changed as well If it is not necessary for the limit to have the same number of points as the original trace the limit can be resized by entering the new number of limit points in the up down control at the bottom of the panel and
314. range without any user interaction Digital Audio Inputs and Outputs SR1 has 2 sets of XLR and BNC connectors and is compatible with both consumer and professional digital audio signals from 50 mVpp up to 10 2 Vpp All significant parameters of the digital audio carrier Signal are measured including signal amplitude digital audio effective sampling frequency and input output delay Status bits are fully decoded in both the the professional and consumer standards User bits are also displayed in raw binary format Several impairment signals can be applied to the output digital audio carrier including Common Mode Sinewave Normal Mode Noise and Jitter Jitter waveforms include Sine Square Noise and Bandpass Noise with a peak jitter amplitude of 13UI Analyzers There are 8 different types of Analyzers in SR1 summarized below Time Domain Detector Makes wideband amplitude ratio and THD N measurements The time domain detector signal chain includes bandpass or notch filters bandwidth limiting filters and a variety of different weighting filters Peak RMS and Quasi Peak responses are selectable FFT 1 ch Provides single channel FFT functionality Measurements include power spectrum time record phase Both zoom changing the FFT frequency range and heterodyne moving the FFT frequency range are included FFT 2 ch Provides dual channel FFT functionality Measurements include power spectrum and time record for both channels
315. raph is the main display type and is capable of displaying many different live and off line data traces both Graphs include cursors for reading out absolute and delta values the ability to export to bitmap and Windows Metafiles movable onscreen annotations and a variety of data calculations smoothing linearity maximum and minimum etc for manipulating data The Bar Chart is a simpler display optimized for displaying a single measurement value which reports the instantaneous value of the measurement as well as maximum minimum and standard dewation The Digitizer Display is designed to work with the optional digitizer and displays various measurements calculated by the digitizer including full color eye diagrams 2014 Stanford Research Systems Getting Started 11 Digitizer Optional The optional digitizer digitizes the digital audio carrier signal at an 80 MHz sampling rate The digitizer calculates the overall clock rate of the signal as well as jitter as a function of time and probability density for pulse width pulse amplitude input amplitude and jitter amplitude The digitizer also calculates the spectra of both the carrier signal as well as the jitter signal Finally the digitizer calculates an eye diagram which shows the probability of the carrier signal as a function of amplitude and time The off line measurements made by the digitizer complement the real time jitter measurement capabilities of the Jitter Analyzer Remote
316. rate a reference phase curve Next the analyzer inputs are switched to measure the phase response of the signal from the EUT The phase reponse of the EUT is then generated by taking the difference between the total phase and the reference phase Cabling Configuration The output of SR1 s analog or digital generator is connected to the EUT s input The output of the EUT is connected to SR1 s analog or digital inputs The number of inputs and outputs and the connectors used are specified in the Setup Panel Generator Configuration The output waveform is set to the FFT Chrip waveform For free run operation the amplitude the generator is set to the frequency and level in the start column of the panel For swept operation the amplitude is swept from the start to stop value in the number of steps specified Analyzer Configuration One or both analyzers depending on the number of output channels specified in the Setup Panel are configured as FFT single channel analyzers The number of lines of resolution is adjustable in the analysis section of the panel Free Run Configuration A graph is created for each output channel Traces are created for the reference phase GenMon phase the total phase and the computed phase difference which is the phase response of the EUT If the delay calculation is selected an additional trace is created for the group delay measurement Group delay is calculated by the trace calculator and is the derivati
317. rates a generator trigger each time the output returns to the beginning of the table FFT Chirp Config Chirp Waveform On w v EQ Chirp Amp 2 0000 Vrms Compliance ac AO FFT ba EJ Var Sweep Rate E The FFT Chirp waveform is designed to work in combination with one of SR1 s FFT Analyzers The chirp waveform provides a tone exactly at the bin center of each the FFT Analyzer s analysis bins In the default case each tone has equal amplitude however the Chirp waveform can be used with generator EQ to generate chirp signals with custom tailored frequency response Chirp signals are useful for quickly measuring the frequency response of a device under test Unlike noise signals which are not deterministic and require long averaging times to measure frequency response a chirp signal can precisely measure frequency response in a single FFT record The Chirp waveform is synchronized to the settings of a particular FFT analyzer If the resolution or frequency span of that analyzer changes the chirp waveform automatically reconfigures to provide a full set of bin center tones If the corresponding analyzer is set to show aliased lines the chirp outputs a tone in every single FFT bin from near DC to Fs 2 If show aliased lines is off the chirp outputs tones from near DC up to the alias limit for the analyzer s selected converter and frequency span See the FFT 2014 Stanford Research Systems SR1 Operation 51 A
318. ratio SINAD Analog Inputs Amplitude Accuracy 0 5 40 043 dB Flatness 1 kHz ref 50 Hz to 20 kHz lt 0 008 dB typ 0 003 dB 20 Hz to 64 kHz lt 0 02 dB 10 Hz to 200 kHz lt 0 05 dB Residual Noise 62 5 mVrms Input Range Shorted Input High Res ADC Fs 128 kHz 22 Hz to 22 kHz 117 5 dBu 22 kHz to 57 6 KHz 115 dBu A Weighted 120 dBu High BW ADC 22 Hz to 22 kHz 118 dBu 22 kHz to 80 kHz 113 dBu 10 Hz to 200 kHz 110 dBu A Weighted 120 dBu Residual THD N High Res ADC Fs 128 kHz 2014 Stanford Research Systems 302 SR1 Operation Manual 1 kHz 4Vrms 111 dB 22 kHz BW 20 Hz to 20 kHz 10 7 dB 0 8 uV 22 kHz BW 101 dB 1 3 uV 57 6 kHz BW High Res ADC Fs 64 kHz 1 kHz 4Vrms 111 dB 22 kHz BW 20 Hz to 20 kHz 107 dB 0 8 uV 22 kHz BW High BW ADC 1 kHz 4 Vrms 113 dB 22 kHz BW 20 Hz to20kHz 109 dB 0 8 pV 22 kHz BW 102 dB 1 5 pV 80 kHz BW 98 dB 2 5 uV 200 kHz BW 10 Hz to 100 kHz 91 dB 200 kHz BW Digital Inputs Amplitude Accuracy 0 001 dB at 1 kHz Flatness 0 001 dB 15 Hz to 22 kHz Residual THD N 140 dBFS Bandwidth Limiting Filters Low Pass Filter 4th order Butterworth adj from Fs 40 to 0 45 Fs 20 kHz 40 kHz and 80 kHz fixed elliptical filters per AES17 High Pass Filter 4th order Butterworth 22 Hz 100 Hz and 400 Hz 20 kHz 40 kHz and 80 kHz fixed elliptical filters per AES17 Band Pass Filter Respons
319. rator The generator can be populated with many different waveforms sines square waves ramps etc Many of the waveforms can be combined by the generator For instance if the generator is populated with sinewave and noise than the output will be the sum of the sinewave and noise signals The amplitudes of each component of the composite signal are adjustable separately as is the overall amplitude of the combined signal Some waveforms USASI noise for instance are special purpose test signals and may not be combined with other waveforms The tabs corresponding to each particular waveform are described in Analog Generator Waveforms section This section describes the controls and settings of the analog generator that are relevant to all waveforms Note that the SR1 Analog Generator is completely separate and independent of the Digital Audio Generator The two generators operate simultaneously and independently with different waveforms EE Suro Fs 512 kHz Unbal and Mode Mona h B A rN 100 0 m aig Lock j100 0 fy Auta E On dEr Ref lt 000 Vins Freq Ret lt 00000 kHz Watts Ref 18 0000 ohms dBm Ref 600 00 ohms EQ None z El Invert EQ Output Controls Fs controls the output sampling rate and D A converter selection for the analog generator SR1 uses two different types of D A converters to generate high quality analog waveforms a 16 bit converter operating at a fixed output sampling rate
320. red as a Two Channel FFT analyzer The two channel FFT analyzer divides the B channel complex FFT spectrum by the A channel spectrum to get the AB phase vs frequency response Free Run Configuration For sine stimulus a barchart display is created displaying the interchannel phase at the selected amplitude and frequency For chirp stimulus a graph is created showing the instantaneous AB phase difference as a function of frequency Sweep Configuration A graph is created showing the interchannel phase vs frequency If the stimulus is sine the data is obtained by sweeping the sine frequency over the specified limits If chirp is selected the trace corresponds to a single FFT measurement In each case a new trace is created for each point specified 2014 Stanford Research Systems 2 2 SR1 Operation Manual on the amplitude sweep 2014 Stanford Research Systems SR1 Operation 273 2 8 11 In Out Phase E Measure InputOutputPhase oy Inpukiutouk Measurement Settling Signal waveform Chirp ad Start Stop Steps Log 100 00 mivrmns o a e Level Analysis Lines ik ad Measure Phase Only Append Traces Free Run The In Out Phase Measurement panel uses the FFT analyzer and the FFT Chirp waveform to measure the input output phase vs frequency response of one or both channels of the EUT The measurement is made in two parts first the analyzer inputs are switched to Generator Monitor to gene
321. ressed The Knob Acceleration control determines how the selected parameter changes when the knob is turned Exponential and Power Law are continuous acceleration profiles which smoothly increase the amount by which the selected parameter changes as the knob turns When Cursor Position is selected turning the knob increments or decrements the digit just to the left of the current cursor position Display Preferences Tab Joi General Display Remote zoom and Translate A zooms In ssi Background a Prints white l On White Jon white Bkard Vera Calla Click To Change EE Ea E raph Display Trace Initialization MW dutoscale On Span Chg SR 1 Default Autoscl Ignore Outliers Zoom and Translate These controls select the sense of the Zoom and Translate controls on both the graph and digitizer graph displays Depending on which feels more natural the left arrow can be selected to move either the data to the left or the axes to the left Likewise the icon can be configured to zoom the display in or out Graph Background Select a White or Black background for the graph displays This selection only affects new graph displays not displays already on the screen For printed graphs the Graph Prints control selects whether the printed background is the same as the screen background color or is always white Trace Colors All graph displays use the same set of default colors for each trace To ch
322. rn to Within The external sweep stops when a settled measurement of the sweep source is Tolerance within tolerance of the Stop value of the sweep or greater than the Stop value 2014 Stanford Research Systems SR1 Operation Manual Configuring Switcher Sweeps SR1 is designed to work with the SR10 SR11 and SR12 Audio Switchers to allow users to switch both outputs from and inputs to SR1 during a sweep to enable testing of multiple devices Configuration of a switch network including identification of the physical switch boxes and specification of communication protocols is performed on the Switcher Configuration Panel After configuring the details of the switcher network each input and output is assigned a logical channel number For purposes of discussing the switcher sweep all we need to know is the range of logical channel numbers for both the inputs and outputs the details of the switcher network configuration will be left to the discussion of the Switcher Configuration Panel Switcher sweeps can only be configured on the outer sweep axis The typical measurement configuration will be to first sweep the input or output to a particular device outer axis and then perform the sweep which represents the actual measurement being performed on the inner axis To select a switcher sweep double click on the Switcher node in the sweep source selection for the outer sweep axis This displays the switcher sweep source panel
323. roprietary improved version of the traditional Blackman Harris window These two windows have sufficient dynamic range for most measurements The other windows are included for historical interest and comparison purposes The FFT chirp and multitone sources are synchronous with the FFT time record and have different frequency content over different parts of the time record Therefore the Uniform window should be selected when the FFT analyzer is being used with the FFT chirp or Multitone source The Variable Time window is on 1 during the portion of the time record specified by the Variable Time Window Limits Outside of that interval the window is 0 The transitions between the off and on segments of the window are done using a raised cosine function to minimize disturbance in the frequency domain The Variable Time window is useful for isolating analysis to a particular region in the FFT time record Obviously the time record needs to be repetitive or the Analyzer Trigger needs to be used to obtain meaningful results Phase Controls Phase Unwrap Threshold 1 0000 prrs The linear spectrum has phase as well as magnitude associated with it Checking Unwrap Phase instructs SR1 to unwrap the phase by adding or subtracting 360 at appropriate break points to create a continuous phase curve 100 2000 500 ik 2k Hz 100 200 500 ik 2k Hz Same Spectrum with Phase Unw rapping On 2014 Stanford Research Systems SR1 Opera
324. rp can be output in the digital domain and the D A frequency response and phase recorded in the analog domain To ensure cross domain sampling rate compatibility use the following sampling rate selections for the generator and analyzer Type of Measurement Analyzer Fs Analog Source Select Digital ISR as the analog Select Digital as the Analyzer source Digital Measurement generator Fs which automatically chooses ISR as ADC the analyzer Fs Digital Source Analog Select the desired digital audio Select the Analog Hi Res converter on Measurement D A output frequency using the Digital the analyzer panel and set the Hi Res O panel converter Fs to Digital OSR or digital OSRx2 using the Analog Inputs panel Because the synchronous chirp source has a time dependent amplitude profile it may fool the input autoranging control into switching ranges too frequently Depending on the FFT bandwidth which determines the chirp length it may be useful to turn off input autoranging when using the f synchronous chirp signal For the same reason be sure to select the Uniform window when using the synchronous chirp Window functions attenuate the beginning and the end of the timre record for the FFT analyzers which when using the chirp means that certain frequencies will be attenuated more than others This destroys the flat nature of the synchronous chirp signal Using Chirp With Eq and Inverse Eg By using the EQ feature of the generato
325. rp source has a periodic time dependent amplitude profile it may fool the input autoranging control into switching ranges too frequently Depending on the FFT bandwidth which determines the chirp length it may be useful to turn off input autoranging when using the synchronous chirp signal For the same reason be sure to select the Uniform window when using the synchronous chirp Window functions attenuate the beginning and the end of the timre record for the FFT analyzers which when using the chirp means that certain frequencies will be attenuated more than others This destroys the flat nature of the synchronous chirp signal Using the FFT1 Analyzer With the Time Domain Detector The FFT1 analyzer can be used together with the Time Domain Detector to create a powerful tool for analyzing noise and distortion spectra in realtime The Time Domain Detector outputs a digital signal which represents the input signal with the fundamental notched out and any bandwidth limiting and weighting filters applied This signal can be routed directly to the input of the FFT analyzer To implement this setup select the type of the AO analyzer as Time Domain Detector Set the TDD input to analog Hi Bandwidth and the TDD function to THD N Ratio Now set the type of the A1 analyzer to FFT1 and set its input to Other Analyzer The spectrum displayed is the spectrum of the input signal with the fundamental removed by the TDD s notch filter and the remaining noi
326. rresponding output status fields Diff from Other Highlights received status fields that differ from the corresponding received status Ch field in the other channel 2014 Stanford Research Systems osa SR1 Operation Manual Data Active Bits Display Data active Bits A OM CONNANAN ao 16 5 O MMM UUM MU 2d 16 5 O The Data Active Bits indicator displays 2 rows of indicators corresponding to the 24 data bits of each digital audio channel The color of each indicator indicates the status of that bit over 1 digital audio block 1 block contains 192 frames Red indicates that the bit does not change value over one block This can be an indication that the bit is stuck Dark Green indicates that the bit changes value over the block Bright green indicates that the bit is on in the first frame of the block it is only active if one of the active analyzers AO or A1 currently has digital audio selected as its input If none of the analyzer is currently looking at digital audio the indicator will be grayed and inactive Because the data active bits display requires examination of the embedded digital audio signal Output Impairment Tab ss Digital lo a vere Common Mode Sine Invalid A 1 0000 kHz 0 0000 pp Invalid B M Normal Mode Moise Cable Sim 0 0000 Vp unbal bal x4 Clock Jitter Freq Amp peak OFF ka o 0000 SEC Jitter EQ Curve Rear Pol Clack Jitter None
327. rt Tones sorts the tones in frequency order while Recalc Phase recomputes the phase distribution according to the algorithm selected on the Multitone Configuration Panel The buttons on the left allow one tone to be the selected tone The amplitude of this tone is used for normalization of measurements computed by the Multitone Analyzer 2014 Stanford Research Systems SR1 Operation 105 2 3 11 Clock Reference Panel Clock Reference Source Lock AES Rr Pril LF Term Expected Frequency 48 00000 kHz Reference Frequency HES Reference Status Ref In Dig Input Delay Terr CRE Lock Validity Conf Coding Parity Ref Qut gt Main Gut Delay 0 0000 Sec i All of SR1 s clocks for both analog and digital audio can be slaved to an external clock signal specified on the Clock Reference panel In the absence of an external clock source the sampling clocks are derived from an internal high quality crystal oscillator Source selects the external clock source to lock to Selects an AES EBU signal connected to the rear panel AES Ref In XLR connector SR1 can lock to signals representing frame rates from 24 kHz to 216 kHz External Selects a Square or sine clock signal connected to the rear panel Ext Ref In Reference BNC connector The expected frequency can range from 10 KHz to 29 MHz Internal Reference Selects the optional internal atomic clock oscillator as the reference signal Not curre
328. rved verion 1 1 5 New Features Full 15 digit precision of SAT settings and readouts made available as hint Added new remote commands for manipulating SAT Forme Improvements FFT Chirp signal automatically sets analyzer window to Unifarm when sweeping generator frequency the tuning of TOD and THD analyzers i set to generator Auto Precise setting of TOD analyzer made more precise Bug Fines Resolved issue when exiting SAT with a generator signal on che Resolved issue of certain private remote commands not executing properly Version 1 1 4 New Features Added peak finder to graph display Added the ability to run a script directly from the main mena Added the ability to print the script log Added the ability to set the text of a Barchart Display trom automation Improvements 2014 Stanford Research Systems SR1 Reference Part SRS Audio SR1 Reference 21 3 SR1 Reference 3 1 Front Panel Description A Power Button and Indicator Turns the instrument power on and off To turn the instrument off select Exit from the file menu When the screen displays the message It is now safe to power off your computer press the button to turn the power off B Headphone Jack Standard 1 4 jack accepts all types of headphones The speaker is muted automatically when headphones are plugged in C Volume Knob Controls the volume of signals routed to the speaker
329. s Sweep Append When sweep traces are present in the graph the default behavior when a new sweep is started is to clear the data from each sweep trace and fill them with data from the new sweep When Sweep Append is checked each sweep traces is copied off line before the sweep traces are zeroed so that data from each iteration of a sweep is maintained in the graph Cursor Display Bar i1931 EEE 2 o 75000 ke SE A ESET i Ee Ee The graph has two independent cursors cursor 1 drawn in blue and cursor 2 drawn in green The X and Y values for the two cursors are displayed in the corresponding displays on the Cursor Display Bar Note that the units of the cursor readout are always the same as the units selected in the scaling tab The displays to the right of the A symbol show the difference between X and Y values of the two cursors Moving the Cursors The two cursors may be moved by grabbing them with the mouse When the mouse arrow is moved over the cursor the arrow icon changes to lt At this point click the mouse button and drag the cursor to the desired location The cursors can also be moved by explicitly entering an X value into either the cursor 1 or cursor 2 X value control on the cursor display bar Finally the active cursor may be moved by turning the knob when the graph has focus Click on one of the cursors to make it the active cursor or press the lt Cursor gt key on the keypad to toggle the active cursor
330. s dEvrms 0 Sk 10k 15k 20k 25k 30k Unwindowed Spectrum 2014 Stanford Research Systems SR1 Operation 151 dEvrms 0 Sk 10k 15k 25k 30k Spectrum with BlackmanHarris Window To eliminate the discontinuities between the beginning and end of the time record the time record is typically multiplied by a window function which is large in the middle of the time record and tapers off near the beginning and end of the time record The second spectrum shows the same two sine waves after application of a window function Many window functions have been developed over the years but few have the gt 120 dB dynamic range necessary for modern audio measurements For most measurements with SR1 the Rife Vincent 4 term or the Blackman Harris window are the best windows to use The BlackmanHarris window is a proprietary improved version of the traditional Blackman Harris window These two windows have sufficient dynamic range for most measurements The other windows are included for historical interest and comparison purposes The Uniform window should be used when use the FFT chirp log sine chirp or Multitone source is used Phase Controls Phase Unwrap Threshold we The linear spectrum has phase as well as magnitude associated with it Checking Unwrap Phase instructs SR1 to unwrap the phase by adding or subtracting 360 at appropriate break points to create a continuous phase curve 100 200 500 ik 2k Phase Spectrum of
331. s amplitude sweep The AO and A1 analyzers are set to Time Domain Detector in order to measure THD N of both channels A low distortion 1 KHz sine waveform is setup with an amplitude sweep range of 50 mVrms to 5 Vrms in 22 logarithmic steps Page 2 displays a graph of the A and B channel results What you ll need to configure The default frequency is 1 KHz This should be adjusted as needed The settling parameters can be adjusted to fit the noise levels of the signals being measured The bandwidth of TDD can be adjusted to suit the measurement requirements 2014 Stanford Research Systems 282 SR1 Operation Manual THD N vs Frequency Sweep Performs a THD N vs frequency sweep The AO and A1 analyzers are set to Time Domain Detector in order to measure THD N of both channels A low distortion 1 Vrms sine waveform is setup with an frequency sweep range of 20 Hz to 20 kHz in 22 logarithmic steps Page 2 displays a graph of the A and B channel results What you ll need to configure The default sine amplitdue is 1 Vrms This should be adjusted as needed 2014 Stanford Research Systems SR1 Operation 283 2 9 3 Digital Analog The Digital Analog selection lists basic Audio Analyzer measurements with a digital stimulus and an analog response i e a DAC The digital generator is configured with the stimulus signal The sampling rate of the high resolution analog converter is set to the digital OSR so both domains are operating wi
332. s installed on the analog input board corresponding to the selected channel Next adjustable analog gain from O to 66 dB is applied to the post filter signal The purpose of this gain is to amplify the post filter signal which in the case of a THD N measurement is typically small after the fundamental is removed by the notch filter so that it will be close to full scale of the third 16 bit ADC which digitizes the amplitude signal with a sampling rate of 512 kHz After digitizing the DSP applies the bandwidth limiting and weighting filters The amplitude of the resulting signal is computed by the DSP with selectable RMS Peak Quasi Peak response and is sent to the host computer for display The amplitude signal is reconverted to an analog signal by a DAC and is output on the rear panel AO Monitor Out or A1 Monitor Out depending on which analyzer is being used BNC connector Finally the amplitude signal is sent to the Other Analyzer e g A1 if the TDD is active on AO or AO if the TDD is active on A1 If the Other Analyzer is an FFT analyzer whose input is set to Other Analyzer the FFT analyzer will display the spectrum of the post Notch BP filtered bandwidth limited weighted amplitude signal Time Domain Detector Analog Hi Resolution Inputs ie To Other Analyzer Amplitude Chain BP Hotch BW Limit Filter Filters Weighting Amplitude Filters Meas ee Att t i a le Sam Coa Pk O ro Cha E A l Ml A
333. s a time domain jitter analyzer to measure rms jitter while A1 is setup to measure the jitter frequency spectrum What you ll need to configure The default frequency and amplitude of the digital sine is 1 kHz 1FFS This should be adjusted as needed 2014 Stanford Research Systems 288 SR1 Operation Manual 2 10 Help Menu SR1 Help Opens the help browser containing the SR1 help file The contents of the helpfile and printed manual are the same About SR1 Displays the About SR1 panel listing the version of instrument software If the software is running on the instrument as opposed to demo mode the serial number of the instrument is also displayed About SR1 Stanford Research Systems SR1 udio Analyzer Version 1 0 2 0 Audio Copyright 2006 2009 Stanford Research Systems Inc Serial Number 104299 Always record the instrument software version and serial number before contacting Stanford Research Systems for technical support View Release Notes Opens a window displaying version information Each version is listed with features that have been added improvements to existing features and bug fixes The information displayed in this window may be more recent than information contained in the manual or help files 2014 Stanford Research Systems SR1 Operation 289 SR1 Release Notes M Ox SRI Release Motes A jMarch 2010 Copiright Stanford Research Systems 2008 2010 All rights rese
334. s set with the Min and Max controls in the Range box The On box must be checked to enable range checking Checking Alarm will sound a tone each time the measured value is outside the defined range Meas X Axis Connecting the bar chart to a scalar measurement such as frequency is straightforward but what does it mean to connect the bar chart to an FFT for instance The answer is supplied by the selection made in the Meas X Axis box on the bar chart settings panel Selecting Max or Min tells the bar chart to display the maximum or minimum value found in the vector measurement Selecting RMS avg displays the RMS average of all of the vector values For an FFT for instance the displayed value would be the RMS average of each of the FFT bin amplitudes Finally Specific Bin displays the value from one particular bin of the vector measurement The bin number is simply an integer index into the vector it is not the actual x axis value for the vector measurement Readout The Readout control determines whether the unit displays in the speedbar at the top of the bar chart display the maximum and minimum values of the measurement or the mean and standard dewation 2014 Stanford Research Systems SR1 Operation 205 2 5 3 Digitizer Display The digitizer display is designed to display the measurements calculated by SR1 s optional Digitizer Unlike the Graph which can display many different traces corresponding to use
335. s the limits set in the Exceeded Range Alarm settings for that bar chart The event only occurs if the Range Alarm is turned on The COM event fired by this event includes an argument identifying the particular bar graph which triggered the event Graph Limit Occurs when trace limit testing fails for any trace on any graph for which it is Exceeded enabled The COM event fired by this event includes an argument identifying the particular graph which triggered the event Eye Limit Occurs when the eye diagram limit testing fails on any digitizer display on which it Exceeded is enabled The COM event fired by this event includes an argument identifying the particular digitizer display which triggered the event Keypad Occurs when any button on the front panel is pressed The COM Event which can by fired by this event contains a keycode argument identifying the pressed key Knob Occurs when the front panel knob is turned The corresponding COM Event includes an argument identifying the direction and amount the knob has turned Warning Occurs whenever SR1 issues a warning The text of the warning is included as an argument by the corresponding COM event Critical Error Occurs when SR1 encounters a critical error The text of the error message is included as an Saa by the corresponding COM event Script timeout timeout Occurs when the execution of a Occurs when the execution of a script times out times out Script Error Occurs when an erro
336. s the profile of the notch filter with the residual fundamental poking through bottom at about 144 dBVrms The THD N is clearly dominated by discrete frequencies rather than noise In this case interestingly the dominant distortion product is not a harmonic but the 12 5 kHz subharmonic As it turns out this function generator generates it s signals at 1 2 the output frequency and employs a frequency doubler to create the final signal the subharmonic is the residual of the original pre doubler signal By using the FFT Cursors to display the integrated power in the residual spectrum it is possible to directly read out the THD N due to any part of the residual spectrum between the cursors This information can be invaluable in deciding which portions of a circuit to optimize When using the FFT1 analyzer in combination with the TDD to measure low levels of THD N be sure the TDD is set to the Hi Bandwidth converter Only the Hi Bandwidth converter signal chain employs the analog notch filter and and analog post filter gain that enables making the most sensitive THD N measurements 2014 Stanford Research Systems SR1 Operation omo 2 4 4 FFT Dual Channel The Dual Channel FFT Analyzer FFT2 computes the frequency spectra of both channels of the selected input domain analog or digital and additionally computes the magnitude and phase of the dual channel frequency response function and impulse response Because the FF T2 analyzer
337. screte Fourier Transform implemented by SR1 calculates a spectrum assuming the time record repeats continuously Thus if the signal being analyzed is not perfectly repetitive in the time record interval the the calculated spectrum will include the discontinuities between the beginning and end of the time record which show up in the frequency domain as wide skirts and a high noise floor around the actual spectrum The spectrum shown below on the left is the unwindowed spectrum of two sinewaves of arbitrary frequency The spectrum is completely distorted by the artifacts associated with the time record discontinuities dEvrms 0 Sk 10k 15k 20k 25k 30k Unwindowed Spectrum 0 5k 10k 15k 25k 30k He Spectrum with BlackmanHarris Window To eliminate the discontinuities between the beginning and end of the time record the time record is typically multiplied by a window function which is large in the middle of the time record and tapers off near the beginning and end of the time record The spectrum on the right shows the spectrum of the same two sine waves after application of a window function Many window functions have been developed over the years but few have the gt 120 dB dynamic range necessary for modern audio measurements For most measurements with SR1 the Rife Vincent 4 2014 Stanford Research Systems 134 SR1 Operation Manual term or the Blackman Harris window are the best windows to use The BlackmanHarris window is a p
338. se and distortion amplified by the TDD s post filter gain 2014 Stanford Research Systems 140 SR1 Operation Manual dByirms Eci eee sayudiud au bu uuu youu E A O E TTT TAO A con enesauswuecat ay aue amex TTOTTE A A E e peers etecteeeteeeeenedy S T A EA AA E ETE E San duaucsendi A T E EE E A E sess sirasees spins guvaseuseis sey I A Large 50 mennsnnennennannnnnnnn Subharmonic see deoedeeenaeeneeeneneneneoaeseneennecnnecnaecnsensoen onneenneeneenaeeneeenenssen oeeseeneeenaeenaeenensnenssensaeeeGeeeeneennenssensseessmaseeeecenessaeesaeeneesseensanneenneenaeenaeenenenensnenineneas Sei commer Ss i Motch Filter Profile E r a a a a T a a E a E a L a a T a a a a T T E TT Fesidual Fundamental 170 deieceancensassseexsvevaxs i uasduedsuteavessiewaree A z 440 dE l i i i 0 25k Sok 75k 100k 125k 150k 175k 200k 225k Hz Residual Spectrum of 25 kHz 1 Vrms Sine from Commercial Function Generator cPrrCrererrrer er rere el eee rerecretecereoree renee Pe cerere T tr eCerererere rere ererci tie Cecccce ee cee cer ecer reer ecrer As an example the spectrum above shows the residual spectrum from a 25 kHz 1 Vrms sine wave generated by acommercial function generator The THD N in a 200 kHz bandwidth as measured by the TDD is 58 6 dB hardly audio quality but within the specifications of this function generator The residual spectrum reveals much more information than the single THD N value does The first thing we notice i
339. selected domain and includes an RMS level measurement a frequency measurement and a measurement of the phase between the two channels These values are displayed at the bottom of the analyzer panel evela Freq hE n ES a TA ERE A The second set of operations the Amplitude Chain is performed only on the selected channel The signal is first passed through a selectable notch or bandpass filter The notch filter is used to remove the fundamental of the signal to allow measurement of noise and distortion while the bandpass filter can be selected to measure narrowband signals or noise The filtered signal is passed through a series of high pass and low pass filters to limit the bandwidth of the amplitude measurement Finally a variety of standard weighting filters can be applied to the signal The output of the weighting filters is available as an analog signal on the rear panel AO Monitor Out or A1 Monitor Out BNC connector This signal can also be passed to the FFT analyzer for real time spectral analysis of the TDD amplitude signal The amplitude of the weighting filter output is measured with a selectable rms peak quasi peak response 2014 Stanford Research Systems ne SR1 Operation Manual and a variable measurement rate Depending on the selected measurement one of the following functions of the amplitude and levels will be displayed Measurement Anpitude aaa Response Ratio RMS THD N Ampl
340. should enter the actual sample rate that will be used during testing The multitone generator can output either a mono signal in which the A and B channel output signals are the same or a stereo signal in which the tone placements for the two channels can differ The latter mode is useful for measuring crosstalk by the examining the amplitude in bins containing a tone in one channel but not in the other Note that the default tones created by the Multitone Configuration panel are always the same on both channels even if stereo mode is selected The user must explicitly change the tone placement on one of the channels using the Edit Tones panel This is especially important when configuring crosstalk measurements where the two channels need to have different tone placements The signal length determines the frequency resolution of the generated signal For a signal length of N and a sampling rate of Fs there are N 2 bin frequencies each separated by Fs N ranging from DC to just under Fs 2 Increasing the signal length increases the number of possible tone positions and allows for more differentiation of tones harmonics and IMD products but increases the amount of time necessary to acquire the FFT record The Repeat Count is only used when the generator is operated in burst mode In this mode each burst will contain the indicated number of repetitions of the complete multitone signal Increasing the repeat count beyond one is useful in b
341. sing the switchers with SR1 is a two part process First the switcher network must be configured which involves informing SR1 about the number and type of switchers and how they will be controlled as well as using the web based control interface of the switchers to assign them unique IP or serial addresses Details of how to configure the switchers can be found in the user manual for the SR10 11 12 which can be downloaded at www thinksrs com Once the switch network is configured the switched inputs and or outputs will have a sequential range of logical channel numbers which identifies each potential input or output connection It is then possible to configure a sweep over these logical channel numbers in order to repeat measurements over multiple devices connected to the switch network Instructions for configuring a switcher sweep can be found in the sweep panel section Adding a Switcher to the Network To configure SR1 with a new switch press the New Switch button at the bottom of the Switcher Configuration panel E New Switch Physical Name NewSiwitchd Connector BNC Output z Switched Ch Ex Dest ch A Control Communication Serial COM Port p Chain Addr p Cancel Enter a descriptive name for the new switch and select the type of switch from the drop down list BNC Input BNC Output XLR Input or XLR Output Note that SR12 BNC Switchers may be used as either BNC Output or Input by setting a jumper on the sw
342. spectrum of the post Notch BP filtered bandwidth limited weighted amplitude signal Time Domain Detector Digital Audio Inputs ri Analyzer Out Amplitude Chain BP Hoich BW Limit Weighting Filter Filters Filters CGhanneluUser Status Channel A Digital Audio Recen er Amplitude Meas RHS Pk Quasi Pk i To Other Analyzer Me Analog Process ing Level Chain Os Digital Processing Digital Audio Input Channel B Detailed TDD Block Diagram Digital Audio Inputs Shown for Ch A Selected For digital signals all calculations are performed by the DSP A digital audio receiver decodes and demultiplexes the multiplexed bi phase encoded digital audio signal and the decoded amplitude data for both channel is sent to the DSP The DSP calculates the RMS levels for both channels the frequency and relative phase of the input signals For the selected channel a 140dB deep notch or selectable width bandpass filter is applied followed by bandwidth limiting and weighing filters The resulting amplitude signal is sent to a DAC where it is transformed into an analog signal available at the rear panel AO Monitor Out or A1 Monitor Out connector The DSP computes the amplitude of the amplitude signal with a selectable RMS Peak or Quasi Peak response 2014 Stanford Research Systems SR1 Operation ng Time Domain Detector Panel Measurement Meas 2 Referenc
343. sponse as a function of frequency EQ files are detailed in the SR1 File Reference When an EQ file is specified the MTA weights all bins in the received FFT by the frequency response of the EQ file before computing measurements Relative Measurements Certain MTA measurements can be expressed either as an absolute voltage or relative to various tone amplitudes The measurements affected by this choice are THD N Bins and Scalar Measurement THD Bins and Scalar Measurement Noise Bins and Scalar Measurement Total Distortion Bins and Scalar Measurement and IMD Bins and Scalar Measurement The allowed relative values are Measurement Relation selects the relative mode for the measurements mentioned above Absolute Amplitude The measurements are expressed as absolute voltages Relative to The absolute voltage for each measurement is divided by the peak voltage of the 2014 Stanford Research Systems SR1 Operation 169 Generator corresponding generator signal Relative to The absolute voltage for each measurement is divided by the peak voltage of the Generator Other generator signal for the other channel Channel Relative to Tone The absolute voltage for each channel is divided by the received amplitude of the Selected Frequency selected tone in the measurement channel The selected tone is selected on the Edit Multitone Tones panel which can be accessed from the Multitone Configuration panel The default for th
344. square wave with instantaneous amplitude of 1V is said to have an amplitude of 1 Vp or 707 Vrms even though this is not the actual value of the RMS amplitude derived from a computation While this may seem counterintuitive it simplifies the specification of amplitude units and is the convention used by most audio test equipment The following table describes the units available for setting the amplitude of analog generator waveforms p Volts Peak or simply volts The p is added to reduce ambiguity with other volt oriented units An analog waveform with an amplitude of 1 Vp has an instantaneous peak value of 1 Volt A sine wawe with an amplitude of 1 414 Vp has an RMS amplitude of 1 Vrms In SR1 2014 Stanford Research Systems lt lt O a2 SR1 Operation Manual Vrms and Vp always have this fixed relationship Irrespective of waveform 1 Vrms 1 414 Vp pp Similar to rms Vpp Volts peak to peak has the same relation to Vp that it does for a sinewave Thus 1Vpp 5 Vp i lt Decibels relative to 1 Vrms A signal with an amplitude of 20 dBVrms has a peak amplitude of 100 mVrms or 141 4 mVp Decibels relative to 0 7746 Vrms Note that even though historically the value of 0 7746 Vrms was chosen because it represents the voltage required to dissipate 1 mW ina 600 Q load in SR1 dBu specifies an open circuit voltage without regard to the actual load connected to the output Decibels relati
345. ssion of what each of the controls means n general specifying a smaller precision window and larger value for nPoints will decrease the noise and glitches in the sweep at the expense of increasing the time required for the sweep Some experimentation may be required depending on the characteristics of the EUT to find the optimal values of the settling parameters 2014 Stanford Research Systems SR1 Operation 257 2 8 3 Reference Panel Measure Reference Input Oukput Measurement Signal Waveform LD Sine ki Start Freq 1 0000 kHz Level 100 00 mvinis oF Analysis Highpass 22 Hz ad Lowpass 22 kHz ad Append Traces Free Run The Reference Quick Measurement uses the Time Domain Detector to measure the output of the EUT at a specified amplitude and frequency and record the result in the dBr references for each output channel Cabling Configuration The output of SR1 s analog or digital generator is connected to the EUT s input The output of the EUT is connected to SR1 s analog or digital inputs The number of inputs and outputs and the connectors used are specified in the Setup Panel Generator Configuration The output waveform is set to either Sine or Low Distortion Sine as specified in the panel The amplitude and frequency of the generator is set to the frequency and level Analyzer Configuration One or both analyzers depending on the number of output channels specified in the Setup
346. st broadband sources will work In particular noise MLS FFT chirp and log sine chirp are all suitable waveforms for impulse response measurements Random noise is not a particularly good choice and has a relatively poor crest factor but is interesting for comparison purposes The MLS waveform has a good crest factor and sounds much like random noise but can produce misleading results in systems with transfer function nonlinearities The FFT chiro waveform has a comparable crest factor to MLS and has the additional advantage of being able to completely tailor the frequency content of the chirp to suit the device under test The FFT chirp however does sound decidedly non random Finally the log 2014 Stanford Research Systems SR1 Operation 145 sine chirp stimulus has the fascinating property of being able to temporally separate the response due to the linear and non linear portions of the DUT transfer functions See Measurement of audio equipment with log swept sine chirps by Thomas Kite AES Convention Paper 6269 presented at the 117th AES Convention October 2004 for a succinct summary of the properties and advantages of using the log sine Chirp Anechoic Frequency Response A typical impulse response measurement made with a small room with a microphone and loudspeaker is shown below vp 6001 N E EE E T A a A a A A S A ait saseepeisereeceecenene tes AT Direct Sound o 400p aS Diet canna APEE EEE EEEN A EE
347. start column of the panel For swept operation the frequency is swept from the start to stop value for each amplitude value specified Analyzer Configuration One or both analyzers depending on the number of output channels specified in the Setup Panel are configured as Time Domain Detectors The upper and lower bandwdith limits are adjustable in the Analysis section of the panel Even though both analyzers are configured only one channel of crosstalk can be measured at a time Free Run Configuration A barchart display is created for each output channel showing the instantaneous Crosstalk ratio Sweep Configuration A graph will be created for each channel showing a graph of Crosstalk vs Freq for the selected range of amplitudes Log spacing can be selected for both the frequency and amplitude sweeps The settling parameter of the THD N measurement can be adjusted directly from the Settling tab of the panel 2014 Stanford Research Systems 2 0 SR1 Operation Manual Settling Profile Delay Exponential 20 000 msec Precesion nPoints Threshold p A000 So 3 p 000D mrm Refer to the Settling Panel chapter for a more detailed discussion of what each of the controls means n general specifying a smaller precision window and larger value for nPoints will decrease the noise and glitches in the sweep at the expense of increasing the time required for the sweep Some experimentation may be required depending on the characteri
348. stics of the EUT to find the optimal values of the settling parameters 2014 Stanford Research Systems SR1 Operation 271 2 8 10 Interchannel Phase Panel Measure Interchannel Phase BAA Input output Measurement Settling Signal waveform Sine ki Start Stop Steps Log Freq 20 000 Hz o Level 100 00 mivrms o j Analysis Highpass 10 Hz ha Lowpass Fs 2 ad Lines Append Traces Free Run The Interchannel Phase panel configures a measurement of the phase difference between the A and B output channels of the EUT The phase difference can be measured using either a sinewave at a sweepable specific frequency and amplitude or by using the FFT chirp stimulus to provide a complete measurement of phase difference vs frequency in a single FFT record Cabling Configuration For analog measurements the A channel of the analog generator should be teed and connected to both the A and B inputs of the EUT For digital measurements simply connect the SR1 generator to the EUT inputs In both cases the EUT outputs should be connected to the appropriate analog or digital inputs of SR1 Generator Configuration Depending on the selected stimulus waveform the generator is configured with either a sine or FFT chirp waveform Analyzer Configuration Ifa sine stimulus is selected the AO and A1 analyzers are configured as Time Domain Detectors When a chirp source is selected the AO analyzer is configu
349. stogram even though the two signals would have similar spectra if vewed with the FFT analyzer In general the histogram analyzer can reveal details of a signal s quantization that are not apparent with time or frequency domain analysis Histogram Analyzer Panel E AQ Histogram Analyzer Source Converter Measurement Trigger Sample Rate Histogram Size jE EE am Scale Bins 2 0000 vp gt Autoscale ea 64 2 0000 p Averaging Ayvgs Clear Avg Done Continuous 2014 Stanford Research Systems SR1 Operation 173 Sample Rate Sample Rate The Histogram Analyzer can examine each sample in the input data stream or it can examine only every other every 4th etc up to every 512th sample Use the Sample Rate drop down to control the fraction of points in the input record which are included in the histogram Histogram Size Histogram Size The Histogram Size control determines how many input points are examined for each histogram This number does not include points which are skipped as a result of the Sample Rate selection Remember that this control only determines how many input points are looked at if all the points lie outside of the histogram range the histogram can still have 0 counts in each bin even though the histogram size is set to 16k Histogram Scale Histogram Scale Bins 10 000 p Auboscale 4 10 000 p The scale controls determine the upper and lowe
350. sults on the analyzer panels and displays This mode is useful for benchtop exploration like we re doing now Sweep mode is a more structured measurement mode in which the instrument sweeps a certain parameter and only measures data at certain defined values of that parameter Sweep mode is more useful for repetitive testing to standards We ll do a sweep at the end of this demo as an example For now observe how in Free Run mode the analyzer s level and amplitude displays are continuously updated To measure the distorted sine signal we ve created select THD N Ratio as the Measurement in the Time Domain Detector Change the units of the Amplitude display to dB and the three panels should appear as they do below 2014 Stanford Research Systems Getting Started 21 28 Analyzer Analog Inputs agag 25 Analog Generator Analog Inputs Hi Res Converter Optional Filters aan ante tha New Fs 512 kHz r Unbal Gnd Analog A Inputid Range Delete Mode Mono kai 25 Ohms hal Measurement Meas 2 References Trigger W Auto 1 000 Vrms r Ch A Ch B esaeran Post Filter Input Config 100 0 to AJB Lok 100 0 o EET eer i z nO A Rall aa mial On m m GenMon Hj H o RE ate Response Gain DC z si Notch BP Filter Config Sine sine he Fixed d vefo f Tuned Range Waveform InputLvl P Auto 1 000 Vrms EQ T Input Config Amp 1 0000 Vrms Freq 1 00000 kHz When THD N r
351. surement panel displays the current values of the scalar measurements calculated by the MTA The controls at the top of the panel select the minimum and maximum frequencies between which the scalar measurements will be calculated 2014 Stanford Research Systems 170 SR1 Operation Manual The Multitone Trigger Panel Scalar Meas References Trigger Debug FCE Trigger alidation Generator i i Validation Channel Level Polarity Generator Trig Source gt Analog ki Manual Delay 0 0000 sec Trigger The analyzer trigger is useful in making several different types of multitone measurements For measurements where the SR1 generator is locally producing the stimulus signal as opposed to a recorded or broadcast stimulus signal selecting the Generator Trigger option will maintain a constant time relation between the generated signal and the received FFT frame This calibrates the MTA s Frequency Response Phase measurement which will have zero phase when a wire is connected from the input to the output When a device is inserted between the input and the output the phase and group delay can be directly read off using the MTA For broadcast or recorded multitone stimuli the generator trigger is obviously no longer useful as the SR1 s generator no longer has any relation to the actual occurrence of the stimulus signal Likewise amplitude triggers may trigger on noise and miss the multitone signal For these situati
352. t Note that both scalar and vector measurements can be displayed on the bar chart for vector measurements additional configuration is necessary on the bar chart settings panel to determine which X axis point on the vector measurement will be displayed Add Measurement G0 Time Dom Det 41 FFT Level 4 Level E MN Time Record A RN Power Spectrum 4 MN Linear Magnitude 4 MN Linear Phase A Analog In Dig4ud In SWEER OK Cancel Show Automation IDs The Add Measurement panel displays a tree containing all the available measurements Open the tree 2014 Stanford Research Systems SR1 Operation 23 and double click on a measurement to select it The Settings button displays the bar chart settings panel which contains additional configuration options which will be discussed below The H button rescales the bar chart display setting the minimum and maximum of the display to the minimum and maximum observed value of the measurement The Ed button resets computation of measurement statistics maximum minimum mean and standard deviation will be computed from measurements made after the button press The two displays to the right of the button can be configured on the bar chart settings panel to display the maximum and minimum of the observed measurement or the mean and standard deviation Finally the m button toggles the display between numerical display only and the numerical display plus bar chart
353. t If Repeat is selected the noise waveform produces a generator trigger each time a waveform repeat is initiated The Pink checkbox specifies that the noise output should be filtered by a 3 dB octave pinking filter While the power contained in a white noise signal is linearly proportional to the measurement bandwidth pink noise will have equal power in equal logarithmic frequency intervals e g the power contained in the 100 Hz to 200 Hz interval will be the same as the power contained in the 10 kHz to 20 kHz interval White or Pink noise can be further filtered according to the controls in the filter group None White or Pink noise is directly output without further filtering The white or pink noise is filtered with a 4th order Butterworth lowpass filter at the frequency specified in the filter frequency control 2014 Stanford Research Systems SR1 Operation e The white or pink noise is filtered with a 4th order Butterworth hipass filter at the frequency specified in the filter frequency control 1 3 Octave The white or pink noise is filtered with a 4th order 1 3 octave bandwidth Butterworth bandpass filter at the frequency specified in the filter frequency control MLS Noise Config MLS waveform b Omn i MLS Amp ZAT iae Length 15 2i Fink The MLS noise waveform outputs a Maximum Length Sequence with a specified repetition interval and frequency profile MLS waveforms are typically used as a stimulus
354. t domains do not need to be the same i e cross domain measurements may be made Next select the number of input and output channels and the connector BNC XLR or for digital audio signals Optical for the inputs and outputs For digital outputs the output sampling rate needs to be configured For analog inputs and outputs the Bandwidth control dictates which DAC and ADC high bandwidth or high resolution will be used in the measurements Pressing Setup performs the configuration 2014 Stanford Research Systems SR1 Operation 255 2 8 2 SNR Panel E Measure SNR Inpukiutouk Measurement Settling Signal Waveform LD Sine Start Stop Steps Log Freq 20 000 Hz 20 000 kHz od O oS iw Level 100 00 mivrms 1 0000 vrs O i Analysis Weighting A Wk ad Append Traces Free Run Sweep The SNR Signal to Noise Quick Measurement uses the Time Domain Detector to measure the Signal to Noise ratio of the EUT Noise is measured first by measuring the integrated output of the EUT with the generator off This effectively puts a resistance equal to the output impedance of the generator across the input of the EUT The noise value then stored in the dBr reference for each channel If a weighting curve is specified the noise measurement is made with the selected weighting filter in place Subsequent measurements are made with the generator on and the results are expressed as a dB Signal to Noise ratio
355. t you re using 28 kHz FFT This setup configures SR1 for a basic audio band FFT The Hi resolution converter is selected for maximum dynamic range The AO analyzer is configured as a dual channel FFT analyzer with a span of 28 8 kHz and continuous averaging The Analog Generator is loaded with a low distortion sine waveform set to a default frequency of 1 kHz and an amplitude of 1 Vrms Page2 contains a graph displaying the A and B channel power spectra while Page 3 contains a graph displaying the A and B channel time records What you ll need to configure lf your e using the analog generator adjust the amplitude and frequency of the signal to match your requirements Otherwise you can turn the analog generator off The amount of averaging in the FFT analyzer can be adjusted to suit the noise level of your signal The FFT resolution is defaulted to 4k points this can be adjusted Frequency Response This setup configures SR1 for a stereo audio band frequency response sweep The Hi resolution converter is selected for maximum dynamic range The Time Domain Detector is selected for the AO analyzer The Analog Generator is loaded with a sine waveform Page2 contains a graph displaying the A and B channel frequency response The sweep controller is programmed to sweep the sine frequency from 20 Hz to 20 kHz in 22 logarithmic steps What you ll need to configure Adjust the the sine amplitude to match the equipment being tested The number of
356. t 2 Odd 1 3 5 7 91113 Odd We o e e e e e oa H EEEE EEE Even 4 6 6 101214 Fundamental Frequency Selection Fundamental f Tuned Fixed Analog Gen Analog Gen 4 Analog Gen B Digital Gen A Digital Gen B Sweep Sweep The THD Analyzer computes the power in two sets of harmonics both relative to the same fundamental frequency The fundamental can either be a fixed frequency or it can be tuned to any of the choices listed in the tuning control Selecting Analog Gen A B or Digital Gen A B tune the fundamental the frequency of the selected channel of the specified generator The Sweep 0 1 selections choose the current sweep frequency of the inner or outer sweep Note that when possible the Generator or the Sweep settings should be used over the Measured Freq setting as they will often result in faster response and better measurement stability 2014 Stanford Research Systems 158 SR1 Operation Manual Harmonic Selection Measurement 1 Qdd 13 5 91113 We O e e e Coad Cal Ca me r r r r r ca Ca Even 4 6 6 101214 Ratio E These checkboxes select the set of harmonics included in each measurement Clicking on the boxes next to the Odd and Even labels will automatically select or unselect all the corresponding odd or even harmonics Alternatively the individual harmonic can be selected or unselected by checking or unchecking the corresponding box The amplitude reported by the measurem
357. t script This button is also found on the main SR1 speedbar SR1 Basic Interface Window SR gt AlnCh A B Alyzrti AlyzrReferences 3 Anlagen nlgInputs H HiRessampleRate srHz64k srH21 28k srOSk srOSRXZ a CloseForms S GetID ch ID OpenForm Hl ClockRef EN AESFreg AESFreqUnit EN AESTerm False True ts DelayOutFromRef The right hand window shows the SR1 Basic interface organized in a tree The top two branches of the tree are SR1 and Events The SR1 branch shows the instrument objects and their properties through which the script interacts with SR1 The Events branch contains the different events which can be trapped in a script See the Events Panel description for a more detailed description how events and scripts interact x Within the SR1 branch the interface is organized as a hierarchical collection of objects each denoted by the icon Each object may contain several properties shown with the z l icon and actions shown with the NN icon Properties are single values which can be read or set in a script Actions are more 2014 Stanford Research Systems SR1 Operation 245 complex functions which can return values and take multiple arguments In general double clicking on any item in the SR1 Basic interface window inserts that text at the cursor location in the main script window Properties Properties are either integer floating po
358. t to the first would be understood to point to the AnlgGen AGenChA object until another explicit object reference is sent Although Parse Absolute is the IEEE 488 2 standard it is often more convenient when configuring the instrument which often requires sending many commands to the same object to use the abbrewated form Arb Block Linefeed By default the linefeed character is a required terminator for GPIB arbitrary length block data When un checked the linefeed is not required Output Options Include Header Checking this box causes SR1 to include the path header in its response to query commands For instance with the box checked the response to AnlgGen AGenChA Gain is AnlgGen AGenChA Gain 0 45 2014 Stanford Research Systems zis SR1 Operation Manual while if Include Header is off the response is simply 045 GPIB Options GPIB yxq 11 Serial ddress T1 Delay On E 500ns E The address control selects the primary GPIB address for the instrument The T1 delay sets the interval between the time the data lines are set on the bus and the time that Data Valid is asserted The default value of 500 ns is usually sufficient Checking the On box turns on the GPIB interface and enables sending and receiving of remote commands over the GPIB bus Serial Port Options GPIB yxI 11 Serial COM Baud Rate Data Parity Stop Handshake On 2 5600 s n E gt RTS cTS j The rea
359. t using the Waveform Controls the associated waveform tab shows up on the generator panel For instance the Sine Waveform tab is shown above If the Generator is in the stereo output mode Separate waveforms for each channel the tab title contains a channel designator indicating which channel the waveform is associated with If the generator is in mono output mode the channel designator is omitted and the title contains just the name of the waveform Some of the controls found on the waveform tabs are common to many waveforms These will be described first to avoid repetition Controls Common to Most Waveforms aveform On fw EO The Waveform On checkbox turns the selected waveform on and off When the generator is outputting 2014 Stanford Research Systems SR1 Operation a combined waveform this checkbox allows the selected waveform to be toggled on and off while still outputting the remainder of the waveforms The EQ checkbox appears for only certain waveforms If EQ is checked and an EQ file is selected on the generator panel the actual generated amplitude for the waveform will be the nominal amplitude multiplied by the frequency response of the EQ file at the current frequency Amp o o000 HIE Freq fi 00000 kHz The Waveform Amplitude control sets the peak amplitude of most waveforms in one of the digital generator amplitude units Because generator amplitudes refer to peak waveform values amplitudes cannot exc
360. tage of decimation includes filtering to eliminate alias effects from the discarded portions of the frequency spectrum The outputs are sent to two buffers which serves as the time records for the FFT2 analyzer The time records for each channel are synchronized to the occurrence of a trigger If the analyzer trigger is not enabled then a trigger is automatically generated as soon as the DSP has finished processing the previous time record Otherwise the analyzer waits for a trigger which matches the specified trigger criteria and begins both time records at the trigger point After a trigger occurs and enough time record points have been accumulated to compute a spectrum of the specified resolution the DSP applies a windowing function to the time domain data See Window Selection Windowing is necessary due to the finite length of the FFT time record Unless the input signal happens to be periodic in the time record discontinuities at the beginning and end of the time record will appear as significant broadening of the true spectrum of the input signal Typical window functions are large in the middle of the time record and taper off at the beginning and ends thus minimizing the discontinuities After windowing the DSP computes the FFT of the windowed time records For a resolution of N lines 2N real time record points are used to compute an FFT of N complex points Each FFT is then averaged in two different ways The Power Spectrum is computed by c
361. tal audio carrier but instead a square wave or clock signal Clock signals bypass the clock recovery PLL and are applied directly to the Jitter Demodulator PLL The Jitter Analyzer Panel Common Controls Analysis Mode Units Lock Gain vid Time Domain Jur 9 joc JE The controls shown above are common to the Time Domain and Frequency Domain analysis modes Selection of the Time Domain analysis mode is made using the Analysis Mode drop down control The Units control selects whether the Jitter amplitude will be reported in seconds or in Unit Intervals Uls The Unit Interval is the shortest pulse width found in the digital audio carrer signal and is related to the embedded digital audio sampling rate by UI sec 1 0 Fs Hz 128 The first of the two lock indicators indicate whether the Digital Audio clock recovery phased lock loop is locked to the input digital audio signal This indicator is absent when the Square Wave selection is made on the Digital I O panel The second indicates whether the jitter recovery PLL has locked to the recovered clock signal In general both indicators should be glowing green for the Jitter Anlayzer to produce reliable results Up to 30 dB of gain is available before the analog jitter signal is digitized The Gain control should be set to the highest value that does not cause the analyzer to overload 2014 Stanford Research Systems 180 SR1 Operation Manual The Jitter Analyzer Panel
362. ted by the FFT1 analyzer are peak based levels Remember the units the levels may be displayed in are independent of the method of computation While the TDD has a continuous input data stream available to it allowing RMS computations the FFT analyzer cannot compute FFTs and maintain a continuous input stream simultaneously So the levels are based on peak values Likewise the fact that continuous data is unavailable means that the A B phase computation is not available for the FFT1 analyzer Frequency data is available for analog inputs which use a hardware based frequency measurement technique but not for digital audio inputs 2014 Stanford Research Systems SR1 Operation 131 Bandwidth Bandwidth Baseband 256 00 kHz 756 00 kHz The maximum bandwidth of the FFT is 1 2 of the sampling rate for the selected input source To instantly switch to full bandwidth click the Baseband button The Zoom feature of the FFT1 analyzer allows selection of narrower bandwidths in factor of 2 increments as well When the bandwidth is decreased the effective sampling rate is decreased and therefore the amount of time necessary to acquire a time record of a given length will increase Therefore updates to the FFT display may slow down as the bandwidth is narrowed When the bandwidth is set to any value other than the maximum the FFT analysis range can be set anywhere within the frequency interval from DC to Fs 2 using the spectrum
363. tems SR1 Operation 195 Stores the trace in memory without removing it from the current graph The stored trace can be pasted back in the current graph or a different graph by right clicking in the graph area and selecting Paste Trace from the submenu Creates a copy of the current trace and makes the copy offline Offline Opens a window which allows manual editing of the trace data This is only useful for offline traces as any edits made on live traces will be overwritten when the next am a is received Color Opens a color selection dialog which allows selection of a new color for the trace Selects the width of the trace 1 2 or 3 pixels For optimum printed output select a width greater than 1 pixel Rename Allows renaming the trace in the trace listing A1 FFT Power Spectrum A can be renamed Reference Amp or some more descriptive phrase Only offline traces can be renamed Calculate Provides access to the Trace Calculator Each calculator option is explained in the trace calculator section Graph Speedbar The speedbar at the top of the graph provides quick access to the following functions Graph Speed Bar Functions icon Description o Adds a trace to the graph After pressing the button the Add Measurement form is displayed showing a tree containing all currently available measurements Double clicking on a measurement creates a trace in the graph corresponding to the selected measurement De
364. tencadavensuasaceceres 293 3 ODECINGCAUONS ai a oat toes e eed a seats sear E e a A ASA 296 A Fiter Refere NGC acs o5 22 pats rece tees a eaten ce tc eee decane eae eee 305 5 Pil sRelerenGeectesecSesas see iotiwentets a a a 306 GO Hardware ROTC VC INC esoteric east peace ee a a a e a aE 310 Index 312 2014 Stanford Research Systems Getting Started Part SRS Audio 6 SR1 Operation Manual 1 1 Getting Started Unpacking and Safety Removing the Instrument From Its Shipping Container Use care in removing the instrument from its shipping container The SR1 weighs approximately 50 Ibs and can be awkward to handle while being removed from the box In particular please take care that the front panel LCD screen is not damaged while unpacking Retain the original packing materials in case the unit ever needs to be returned for service Connecting The Power Cord Your SR1 was shipped with a power cord appropriate to your location SR1 operates from a 100V 120V 220V or 240V nominal AC power source with a line frequency of 50 or 60 Hz Before connecting the power cord to the rear panel power entry module please ensure the the LINE VOLTAGE SELECTOR card located in the rear panel fuse holder is set so that the correct AC input voltage value is visible The fuse rating should also be checked to see if it matches the line voltage setting For 100V 120V operation a 4A fuse is used For 220V 240V a 2A should be installed Th
365. th the same samplling rate What you ll need to configure for all these setups Change the digital output parameters connector sample rate bit resolution to match the input of the DACt being tested Set the analog inputs to the appropriate connector and input coupling FFT This setup configures SR1 for a basic audio band FFT The Hi resolution converter is selected for maximum dynamic range The frequency range of the FFT is 1 2 the sampling rate of the digital output The AO analyzer is configured as a dual channel FFT analyze continuous averaging The Digital Generator is loaded with a Isine waveform set to a default frequency of 1 KHz and an amplitude of 1 FFS Page2 contains a graph displaying the A and B channel power spectra while Page 3 contains a graph displaying the A and B channel time records What you ll need to configure lf your e using the digital generator adjust the amplitude and frequency of the signal to match your requirements Otherwise you can turn the generator off The amount of averaging in the FFT analyzer can be adjusted to suit the noise level of your signal The FFT resolution is defaulted to 4k points adjust this to obtain the proper balance of update rate vs frequency resolution Frequency Response This setup configures SR1 for a stereo audio band frequency response sweep The Hi resolution converter is selected for maximum input dynamic range The Time Domain Detector is selected for the AO analyzer Th
366. th 12 channels If a single channel sweep were desired the B sweep could be unchecked and the channel increment reduced to one thereby enabling up to 11 sweep steps on a single device Adding more devices in the Switcher Configuration Panel will increase the available range of logical channel numbers and increase the number of possible steps in the sweep When the Bus box is checked output switcher sweeps only all the output channels are simultaneously connected to SR1 s output except the one channel that is being sweep This is useful for testing crosstalk of multiple devices 2014 Stanford Research Systems SR1 Operation Configuring Sweep Data After a new sweep X axis point is determined according to the selected Sweep Source SR1 begins to search for settled values of each of the sweep data measurements Up to 6 measurements may be selected for each sweep To select the sweep data measurements click on the numbered box in the sweep data section of the sweep panel This opens a tree display of all the available measurement selections Click on the measurement to add it to the sweep or click on None to remove the current measurement from the sweep Only measurements from the currently active analyzers will be shown so it s important to set up the analyzers before configuring the sweep Other Sweep Parameters Pre Sweep Delay sets a fixed delay between the time when the sweep starts and the time when the analyzer begins measrui
367. th the A and B channels The amplitude of each channel is still separately adjustable but the waveform is the same Each channel can be configured with a different waveform 2014 Stanford Research Systems 58 SR1 Operation Manual Dither controls the type of dither used by the Digital Generator The digital generator generates the waveform internally with higher precision than the maximum 24 bit digital output word Noise with the selected probability distribution is added to the internal representation and the result is truncated to the width specified in the Digital I O Output Resolution control of No dither is used in calculating the output word Triangular Dither with a triangular probability distribution and a width of 1 Isb is added to the signal Rectangular Dither with a rectangular probability distribution and a width of 1 2 Isb is added to the signal Fs displays the current generator output sampling rate This value is chosen on the Digital I O panel Waveform Controls Delete CB The New button displays the Waveform Selection Submenu Moise Phased Sines USASI Moise Square Ramp Arb FFT Chirp Digital Constant MultiTane Count IMC Rotating Bits Polarity Staircase Digital Test Signals Test The selected waveform will either be added to the output for one or both channels depending on the Mode setting Certain waveforms for instance USASI Noise cannot be combined with other
368. the active cursor to the next peak right or left o a Annotation Opens a text window and allows placement of text on the graph area e peee amanea Graph Title g o the Trace The trace calculator functions are fully described in the trace calculator section Scaling Tab Min Max Log x 23 4375 Hz 24 0000 kHz m Y 20 0000 dEFS 20 0000 dBF5 E Lock Axes The graph scaling tab controls the scaling and unit selection for the active trace Each axis can be plotted logarithmically or linearly by clicking the Log checkbox for that Axis Note that when the units for an axis are already intrinsically logarithmic such as dBVrms selecting Log will change the units to their corresponding linear value Vrms and plot the data on a logarithmic scale Some unit selections in the scaling controls depend on analyzer settings For instance power spectral density units such as V v Hz depend on the FFT analyzer s resolution and window settings When traces with these units are taken offline the units are frozen and conversion to other units may be limited Lock Axes Normally each trace has its own pair of Xand Y axes and each trace can be scaled separately However there are times when it is useful to be able to view several traces all plotted with the same scaling on the same axes Clicking Lock Axes causes SR1 to plot all traces that have Xand Y axes compatible with the active trace on the same axes as the active
369. time record Unless the input signal happens to be periodic in the time record discontinuities at the beginning and end of the time record will appear as significant broadening of the true spectrum of the input signal Window functions are large in the middle of the time record and taper off at the beginning at the end in order to minimize the offending discontinuities After windowing the DSP computes the FFT of the windowed time record For a resolution of N lines 2N real time record points are used to compute an FFT of N complex points Each FFT is then averaged in two different ways The Power Spectrum is computed by computing the power for each spectrum taking the absolute value of the complex FFT points and averaging that power into the power computed for previous FFTs This type of averaging does not reduce the noise floor of the spectrum but it does reduce the variation of the noise floor making it easier to see spectral details on the order of the noise amplitude Phase information is lost when computing the Power Spectrum In the example below the Uunaveraged power spectrum is shown for a signal composed of a 1 kHz sine wave with added white noise The second spectrum shows the power spectrum with averaging on and Navg 10 Note the substantial reduction in the variation of the noise floor and note also that the average value of the noise floor is is the same in the two spectra 2014 Stanford Research Systems SR1 Operation 129
370. tion 135 The Phase Threshold specifies a minimum amplitude at which to compute phase This can be useful when the spectrum only has amplitude at discrete frequency points If a phase threshold is not specified important phase information can be lost in the phase noise that is generated by the noise floor In the example below the phase spectrum of a 3 KHz square wave is shown The spectrum contains power only at the fundamental odd harmonics 3 kHz 9 kHz 15 kHz 21 kHz etc The phase of the odd harmonics alternates between 0 and 180 However in the first spectrum the phase pattern of the harmonics is obscured by the noise generated by bins without any amplitude ih iil alll TTA A ee iV i ae ili i lk j i Il 0 10k 20k 30k 40k 50k 60k Phase Spectrum of Square Wave Obscured by Noise T 10k 20k 30k 40k 50k 60k H With Phase Threshold set to 100uV Adding a relatively small phase threshold value 100 uV suppresses phase calculation for spectral bins with no amplitude and allows the true phase spectrum to emerge DC Correction OC Correction mii Bi average Small amounts of DC in the FFT time record can be removed using the DC Correction control Selecting Average will subtract the average value of each time record from the time record before taking the FFT 1 2 Pk Pk will subtract the average of the maximum and minimum values found in each time record 2014 Stanford Research Systems 136 SR1 Operation
371. tive Total Harmonic Distortion THD measurements THD Analyzer l a l THD both ratio and absolute Includes the ability to measure specific sets of harmonics IMD Analyzer Makes standard Intermodulation Distortion measurements including IMD SMPTE DIN CCIF difference frequency and DIM TIM Multitone Analyzer Makes single shot multitone measurements allowing fast measurements of MTA common audio parameters including noise distortion Jitter Analyzer Measures the jitter of the digital audio carrier in both the time and frequency JITT domain including variable high and low pass filtering a ee aL i Creates amplitude histograms of the analog and digital audio input signals Octave Anayzer Measures power in fractional octave bins RTA 2014 Stanford Research Systems 108 SR1 Operation Manual 2 4 1 Common Analyzer Features Analyzer Input Selection Most of the analyzers share common controls that select the analyzer input SOUrCE Converter Fs Source Most analyzers take a single input source either analog channel A or B or digital audio channel A or B Details of the analog input connections such as connector coupling etc are set on the Analog Inputs Panel Details of the Digital Audio Input connection are set on the Digital I O Panel Some analyzers such as dual channel FFT intrinsically take a pair of inputs as their input source For these analyzers the input source selection wil
372. too small for some users to read comfortably Changing the screen size allows a uniform magnification of all panels and displays on the screen Note that the main menu at the top of the screen is unaffected by this setting Remote Commands Preferences Tab x General Display Remote Enable Macros Big Endian Binary W Ignore Case M Parse 4bsolute I rb Block Linefeed Include Header Jf Verbose Use Enum Append Units fe i Sig Figs GPIB yxt 11 Serial Address T1 Delay On la 500s z E Front Panel Lockout The IEEE 488 standard specifies that after receipt of a command the instrument is placed in Remote state in which the front panel controls are locked out This can be inconvenient during development Un checking this box allows the user to have front panel access without having to press the Local key SR1 never enables the Local Lockout state Enable Macros By default GPIB macros are disabled on startup as specified in the IEEE 488 2 standard Selecting this option enables GPIB macros when SR1 starts up 2014 Stanford Research Systems SR1 Operation 217 Binary Array In Out By default all remote data interchange with SR1 is done in ASCII format To maximize transfer rate for large arrays Binary Array In Out can be selected which instructs SR1 to transfer all array data in binary When selected integer arrays are transmitted and received as arrays of 32 bit binary values wh
373. tor domain on the MultiTone configuration panel is not set as Digital then the MultiTone waveform cannot be selected in the digital generator IMD Waveform Config IMD Config IMD aveform aveform on W Total Amp E On W Total Amp ME Type High Freg 3 00000 kHz Type Center Freq 3 00000 kHz SMPTEIDIN SMPTEIDIN C CCIFIDFD IM Freq fe0 0000 Hz IM Freq fe0 0000 Hz CCIFIDFD C DIME C DIME DIM 30 Amp Ratio a i 0 DIM 30 Amp Ratio E al C DIM 100 C DIM 100 IMD Tab for SMPTE DIN And for CCIF The IMD Waveform can be configured to output the three classic audio IMD test signals SMPTE CCIF and DIM SMPTE DIN Combines a High Frequency Sinewave with a low frequency sinewave For a generator Fs of 192 kHz the low frequency can be set between 10 Hz and 1 kHz The High Frequency can be anywhere down to 5x the low frequency The Amplitude Ratio low freq high freq can be set to either 1 1 or 4 1 with the Amplitude Ratio control CCIF DFD Outputs two sines centered around the Center Frequency separated in frequency by the IM frequency The amplitude ratio is fixed at 1 1 Outputs a 14 kHz sinewave and 2 96 For all the DIM options the ratio of the kHz squarewave Squarewave is filtered Ssquarewave to sinewave peak to peak with a single pole 30 kHz filter amplitudes is 4 1 Because the square needs to be perfect frequency with an exactly equal number of positive and negative samples the exa
374. tor level is below half scale red indicates the monitor output is overloaded and may be clipping Green indicates the monitor level is between half scale and full scale 2014 Stanford Research Systems 102 SR1 Operation Manual 2 3 10 Multitone Panel Multitone Testing is a relatively new technique compared to traditional audio measurements many of which have their roots in the 1920s and 1930s which enables testing of multiple audio parameters using the information contained in the FFT of a short record of audio data By using a stimulus signal which contains discrete tones at only few of the bin frequencies of the FFT the Multitone analzyer can examine the amplitude of bins in the FFT which correspond to tones in the original signal frequency response harmonics of tones in the original signal THD and THD N intermodulation products of tones in the original signal IMD and finally bins where none of the above are present Noise All these measurements can be made simultaneously on the data found in a single FFT instead of by making multiple laborious swept measurements Setting up multitone testing on SR1 involves configuring the generator and the analyzer Details of the analyzer configuration are discussed in the Multilone Analyzer section The MultiTone configuration panel discussed here is concerned with details of the generator configuration Once the tone placement signal length etc are configured on the Multitone Configur
375. tput amplitude for the channel from 0 to 1000 of the sum of the waveform amplitudes for that channel The Total Channel Amplitude control displays that value For instance if the channel has 2 sine waveforms one with an amplitude of 1 Vp and the second with an amplitude of 3 Vp and if the Channel Gain control is set to 50 then Total Channel Amplitude will display 2 Vp In general The A and B channels can have separate Channel Gains however if the A B Lock checkbox is checked the A and B values are always the same If the sum of the waveform amplitudes exceeds the maximum output voltage of the generator then the Channel Gain will automatically adjust to a value such that the total output amplitude reamains within range Auto On affects the behavior of the generator output during a sweep If Auto On is checked the 2014 Stanford Research Systems 40 SR1 Operation Manual generator output will automatically turn on if it was off at the beginning of a sweep and turn off when the sweep is completed The large green red On Off buttons turn on and off their respective channels while the Invert buttons invert the output for each channel Reference Controls eferences der Ref lt 000 Vries Freg Ref lt 00000 kHz Watts Ref 18 0000 ohms dBm Fef 600 00 ohms The Reference controls allow setting several parameters used in the computation of different generator amplitude units See the Generator Units section fo
376. ts up When this waveform is applied to a device under test it is easy to see if the device properly maintains or inverts polarity by checking the output waveform using the time record of the FFT Analyzer If the output waveform still points up the device maintains polarity If the waveform points down the device is inverting Polarity Waveform Non Iinverted And Inverted 2014 Stanford Research Systems SR1 Operation Special Digital Test Waveforms SR1 s digital audio generator includes several waveforms that are specific to the digital audio domain i e they have no analog counterparts These waveforms generate specific bit patterns in the embedded digital audio output signal As such they cannot be combined with other signals Dither is not added to the special digital test waveforms Digital Constant Config DigitalConst averorm On fw onst Offset Amp 0x7FFF0E hex oF olarity f Pos C Meg The digital constant outputs a fixed bit pattern to the audio data portion of each digital output sample The constant can be specified as a hex word or in any of the standard digital generator amplitude units The polarity of the constant can be controlled with the radio buttons at the bottom of the panel Digital Count Config Digital Count waveform On w Digital Counter Meg Initial Value 0x0 hex eE Final value 0x7FFF hex I 100 The Digital Count waveform implements a counter which o
377. ts no frequency window This will result in significant aliasing in the ETC especially if the signal contains energy near the Nyquist frequency The Hann windows suppresses both low frequencies and frequencies near the Nyquist frequency and emphasizes frequencies near Fs 4 The paper mentioned above contains a detailed discussion of how this window can distort the ETC but it is included for comparison purposes The Half Hann is window recommended by Lipshitz and Vanderkooy which eliminates frequencies near 2014 Stanford Research Systems SR1 Operation 155 Nyquist but retains low frequencies The 240Hz 8kHz selection removes frequencies below 240Hz and above 8 kHz providing an approximately 5 octave analysis range The 120Hz 16kHz selection removes frequencies below 120Hz and above 16 kHz providing an approximately 7 octave analysis range Using the FFT2 Analyzer With the FFT Chirp Source Like the FFT1 analyzer the FFT2 analyzer can be used with SR1 s generator FFT Chirp waveform which produces a signal with uniform power in each FFT bin The FFT2 analyzer makes one shot frequency response measurements using the chirp source even easier it s not necessary to store a reference input curve for later normalization as is the case with FFT1 because the FF I2 analyzer normalizes each measurement individually The magnitude and phase of the frequency response is immediately available with no further computation 500 ik 2k Sk 10k
378. tude of an internal signal Measured Fs displays the measured frame rate of the digital audio signal For single connector digital audio inputs this will be equal to the effective sampling rate of the digital audio data For dual connector digital inputs this value will be half the effective sampling rate When Square Wave is checked on the input configuration panel the frequency reported is simply the measured frequency of the input square wave Delay measures the delay specified by the Delay Mode control When Delay Mode is set to DigOut gt Digln the Delay field displays the delay from the z preamble beginning of digital audio block of the digital audio output signal to the z preamble of the digital audio input signal When When Delay Mode is set to RefOut gt Digln the Delay field displays the delay from the z preamble of the rear panel digital audio reference output to the z preamble of the digital audio input signal Both these values can be used to measure the transmission delay through a device depending on whether the rear panel reference or the main digital audio output is used as a source Status Bits Controls The Channel Status Button displays the Channel Status Panel The User Status Button displays the User Status Panel Highlight Differences controls how status bits are displayed on the User and Channel Status panels None No fields are highlighted Diff from Output Highlights received status fields that differ from the co
379. ual connector mode is slaved to the output dual connector mode Checking the Square Wave box tells SR1 that the input signal is not an AES EBU or consumer digital audio signal but simply a square wave signal When square wave is checked none of the analysis capabilities related to the embedded digital audio data stream or the channel status bits are active However the amplitude and frequency of square wave signals is measured and square wave signals can be used as an input to the Digitizer or Jitter Analyzer for jitter analysis Checking Terminate terminates the XLR inputs with 110Q and the BNC inputs with 75Q Checking EQ inserts a high pass equalization circuit into the signal path EQ can be selected only with XLR and BNC inputs The frequency response of the EQ circuit is given in AES3 2003 in section 8 3 4 12 10 Relative gain 6 dB i 2 0 0 1 03 1 3 10 Frequency MHZ Suggested Eq Characteristic for 48 kHz OSR 2014 Stanford Research Systems SR1 Operation at The purpose of equalization is to reverse the lo pass degradation of the digital audio carrier which results from transmission over long cable runs So0n 3 0 3 5 4 u Soon 1 0 S00n 1 04 3 0 4 0 Long Cable Run Input EQ Digital Input Sample Rate Controls Sample Rate Generator Fs m Select the Digital Audio Input Sampling Rate ISR used by all digital domain analyzers The ISR determines frequency range and resolution for all
380. udio samples rather than the physical sampling rate on each connector When To Use the Hi Bandwidth and Hi Resolution Converters Use the Hi Bandwidth Converter when e The measurement requires a bandwidth greater than the maximum bandwidth of the Hi Res converter e When making THD and THD N measurements of ultra low distortion devices lt 100 dB The Hi Bandwidth converter signal chain includes hardware notch filters and hardware post filter gain to provide the best dynamic range for THD and THD N measurements Use the Hi Resolution Converter when e When making FFT measurements within the bandwidth of the Hi Resolution converter FFT measurements made with the Hi Resolution converter can have an up to a 15 dB lower noise floor than equivalent measurements with the hi bandwidth ADC 2014 Stanford Research Systems SR1 Operation 109 e When making cross domain measurements that require the input analog sampling rate to be synchronized to the digital audio output e When making IMD measurements The Hi Resolution converter has lower residual IMD distortion Level Indicators For all analyzers except the Jitter analyzer the bottom portion of the analyzer panel displays the levels frequencies and relative phase associated with the selected inputs For the Time Domain Detector the indicated level is a highly accurate RMS measurement of the two input channels For all other analyzers the level meter is a less accurate peak
381. ulse rates in the digitizer record Finally the eye diagram displays probability vs amplitude and time using a color coded 2 dimensional plot Note that all the traces in a digitizer display reflect measurements computed from the same underlying digitizer record Therefore traces may not be copied and pasted between digitizer displays as they may be on the normal graph Digitizer Display Speedbar The speedbar at the top of the Digitizer Display provides quick access to the following functions Graph Speed Bar Functions 2014 Stanford Research Systems 206 SR1 Operation Manual Icon Description Load a previously saved Digitizer Display a Saves the Digitizer Display to a file Exports the currently displayed tab of the Digitizer Display to one of several file formats The output file can bye a text file a bitmap BMP file a Windows Enhanced Metafile EMF or a JPEG JPG file Prints the display After pressing the button a print preview is displayed The user can select among the installed printers and then press the print button to finalizing printing The currently selected tab Time Record Spectrum Probability or Eye Diagram is printed Autoscales the active display The first button autoscales only the X axis the second only the Y axis The third button autoscales both the X and Y axes t Translates the active display right left up and down The direction of translation is adjustable on the
382. ultiTone waveform outputs a generator trigger once each cycle When performing multitone analysis with the local SR1 generator be sure to use generator trigger as the analyzer trigger source to ensure proper phase calibration IMD Waveform Config IMD Config MD aveform aveform n wW Total Amp e n W Total Amp e ies Type High Freq 3 00000 kHz Type Center Freg 3 00000 kHz SMPTE DIN SMPTE DIN CCIFIDFD IM Freq fe0 0000 Hz CCIFIDFD IM Freq fe0 0000 Hz DIM B DIME l DIM 30 Amp Ratio t1 DIM 30 Amp Ratio E al C DIM 100 C DIM 100 IMD Tab for SMPTE DIN And for CCIF The IMD Waveform can be configured to output the three classic audio IMD test signals SMPTE CCIF and DIM 2014 Stanford Research Systems SR1 Operation IMD Type can be anywhere down to 5x the low frequency The Amplitude Ratio low freq high freq SMPTE DIN Combines a High Frequency Sinewave with a low frequency sinewave For a generator Fs of 512 kHz the low frequency can be set between 10 Hz and 1 kHz The High Frequency can be set to either 1 1 or 4 1 with the Amplitude Ratio control CCIF DFD Two sines centered around the Center Frequency separated in frequency by the IM frequency The amplitude ratio is fixed at 1 1 Outputs a 14 kHz sinewave and 2 96 kHz squarewave Squarewave is filtered with a single pole 30 kHz filter Outputs a 15 kHz sinewave and 3 15 kHz squarewave Squarew
383. umber of lines in the analyzer FFT to twice the length of the stimulus signal By making the frequency resolution of the received spectrum twice the resolution of the stimulus it is ensured that all tones harmonics of tones and IMD products of tones will fall on even bins in the received spectrum while the odd bins will contain only noise For some test situations the signal chain does shift the frequencies For instance the multitone stimulus signal can be played back on a tape player with a speed error In this situation the exact bin frequencies of the stimulus will be smeared over many bins in the received spectra and the MTA should be operated in windowed mode In windowed mode a window is applied to the received signal to limit the smearing of the tone frequencies and the noise analysis feature described above for synchronous mode is not available In windowed mode a parameter can be entered which describes the maximum extent of the frequency shift and assists the analyzer in locating regions of tones distortion and noise Multitone measurements with SR1 first require the configuration of the multitone generator using the Multitone Configuration Panel The Multitone Configuration Panel contains options for setting the number of tones tone frequency and phase and the length of the stimulus signal Once the generator is configured the Multitone Analyzer MTA can be selected Summary of Multitone Analyzer Outputs The multitone analyzer pr
384. urst situations as it gives the analyzer more time to recognize the multitone stimulus and increases the dynamic range of the final measurements Tone Generation Tones are placed at bin frequencies between the values entered in the Start Frequency and End Frequency fields according to the algorithm selected in the frequency distribution control Multitone Frequency Distribution Linear The desired number of tones will be distributed between the frequency limits with an approximately uniform frequency separation a i desired number of tones will be distributed between the frequency limits with an a oe constant ratio between the frequencies of adjacent tones Octave Starting with the first bin with a frequency greater than or equal to the Start Frequency tones will be placed with each tone having a frequency two times greater than the previous tone The final tone has a frequency less than or equal to the End Frequency Decade Starting with the first bin with a frequency greater than or equal to the Start Frequency tones will be placed with each tone having a frequency ten times greater than the previous tone The final tone has a frequency less than or equal to the End Frequency Tones are placed at each prime numbered bin whose frequency falls within the selected range This placement algorithm guarantees that no tones will fall on harmonics of other tones which is useful in separating the harmonics distortion due to each i
385. ve of the phase with respect to frequency Sweep Configuration Same as the free run configuration except that total phase and phase response traces and optionally 2014 Stanford Research Systems 274 SR1 Operation Manual group delay traces are added for each amplitude setting in the sweep 2014 Stanford Research Systems SR1 Operation 275 2 8 12 Automated Measurements and Reports Each time a quick measurement is performed either in free run or sweep mode the results are accumulated in a formatted report The report includes a header page which details the setup configuration entered in the quick measurement setup panel Measurement Report Signal Setup Output Analog XLR Channels 2 Impedence 600 ohms Output Bil 200 kHz Input Analog XLR Channels 2 Termination 600 ohm Coupling AC Input B i 00 kHz Motes Example notes for SRL manual Subsequent pages give information on the results of each test Free run tests simply record the first value measured For instance Measurement Report IMD Wave for rm SMPTE Start Freq 7 0000 kHz IM Freg 60 000 Hz Start Level 100 00 mr ms Products d2 IMD A 79 34 dB IMD E 74 48 dE Swept measurements record a graph of the measurement in the report 2014 Stanford Research Systems 2 6 SR1 Operation Manual Self Test 2 Report SRS Signal Setup Output Analog XLR Output BW 50 kHz Input GenMon Input BW 200 kHz Notes Analog Output Board A 30145 Analog Out
386. ve to the dBr Reference specified in the References Box Decibels relative to 1 mW into a load specified by the dBm Reference in the References Box Specifying the waveform amplitude in dBm asks SR1 to set the amplitude necessary to deliver a certain amount of power to a specified load taking into account the output impedance of the generator If the amplitude is measured across the open circuited output terminals with a high impedance meter get a different result Watts into a load specified by the dBm Reference in the References Box Specifying the waveform amplitude in dBm asks SR1 to set the amplitude necessary to deliver a certain amount of power to a specified load taking into account the output impedance of the generator If the amplitude is measured across the open circuited output terminals with a high impedance meter get a different result Q W N The value of 0 has no exact representation in dB units When a generator amplitude of 0 is shown in a control displaying dB units the value 1000 dB is shown which may display as 1 0 kdB This is not meant to imply an actual value of negative 1000 dB but merely to provide a convenient way to represent O in dB units Analog Generator Frequency Units The following table describes the units used by SR1 in setting the frequency of digital generator waveforms All frequency units except Hz make use of the Frequency Reference which is set in the References Box on the Analog G
387. verning the computation of the impulse response and related measurements The Calc Impulse Response checkbox must be checked for SR1 to compute impulse response anechoic frequency response and the Energy Time curve The calculation of these measurements is time consuming so the box should be left un checked to obtain the maximum FFT update rate in situations where they are not needed The Anechoic Frequency Repsonse controls select the portion of the total impulse response used in the calculation of the anechoic frequency response and energy time curve The start value should be set just before the main peak while the stop value should be set to just before the first reflection The Data Points display shows the number of points within the selected range The two window selections control the width of the raised cosine window that smooths the transitions to the selected portion of the impulse reponse Some experimentation may be necessary to find the settings that produce the best anechoic transfer function The Energy Time Curve window selection controls the window applied in the frequency domain to the anechoic frequency response before obtaining the real and imaginary parts of the analytic impulse response An excellent discussion of the effects of windowing on the energy time curve can be found in Lipshitz and Vanderkooy Uses and Abuses of the Energy Time Curve Journal of the AES Vol 38 No 11 November 1990 pp 819 836 None selec
388. w l Measurement Scalar Meas References Trigger Processing Synchronous FM Correction 0 0000 hy o FSOK a E Distortion Products Harmonics S ThrujlO IMC Products Thru 2 Equalization None Invert Eg Measurement Relation Absolute Amplitude Noise Analysis Processing The MTA has two distinct processing modes Synchronous and Windowed Synchronous processing assumes that the DUT does not shift frequencies at all and that therefore all the tone frequencies and harmonics will fall on exact bin frequencies in the received FFT When operating in the synchronous mode the MTA does not use a window function which would otherwise smear the bin frequency tones over several lines in the received spectrum When the DUT does shift the frequencies of the multitone generator signal the MTA must apply a window function to the received signal to limit the smearing of the received tones In windowed mode certain MTA measurements such as phase and noise measurement are disabled Distortion Products This panel allows selection of the minimum and maximum harmonics that will be included in measurements like THD and THD N as well as the maximum order of the intermodulation products that will be included in the IMD measurements Equalization The file selection box allows specification of an EQ file which is then applied as a weighting function to all MTA measurements EQ files are files which specify a relative re
389. wPSettings dtd ajmMewer xml ProtoCom dtd Aliasaraph dtd S termination vbs File name IntertaceFunctions vbs Upen Files of type Any file Cancel The return value is the full path extended filename In the example above the return value was C Program Files SR1 Audio Analyzer user config InterfaceFunctions vbs The function also returns cancelled and timedout on the occurrence of the corresponding events Userinput The Userinput function displays a simple dialog allowing the user to enter a value A default value can be shown For instance x SR1l Instrument UserInput Input a numeric value 122 25 creates the dialog box 2014 Stanford Research Systems 250 SR1 Operation Manual User Input Input a numeric value The function returns the string entered in the editbox The function also returns cancelled and timedout on the occurrence of the corresponding events Script Logging The functions described above use dialog boxes to interact with the user In some situations it is useful to be able to provide output to a form that remains on the screen and does not require user interaction This can be done using Script Logging The script log form can be opened from the Automation entry of the SR1 main menu When opened the log is blank Scripts can interact with the log using the following functions from the Scripting section of the SR1 Basic interface SR1 Scr
390. waveforms When one of these waveforms is selected all of the current waveforms are deleted Other waveforms simply add to the current output when selected See the Digital Audio Waveform section for a detailed description of all available waveforms The Delete button deletes the currently selected waveform The A B selection buttons only appear when the generator Mode is set to Stereo The buttons determine which channel a newly added waveform will appear on When the mode is changed to stereo any waveforms present will be assigned to channel A When the mode is changed from stereo to mono all channel B waveforms are deleted and the channel A waveforms are output on both channels 2014 Stanford Research Systems SR1 Operation os Amplitude Controls h ACh B J100 0 AjB Lock 60 00 Abo On The Channel Gain control varies modifies the total output amplitude for the channel from 0 to 1000 of the sum of the waveform amplitudes for that channel The Total Channel Amplitude control displays that value For instance if the channel has 2 sine waveforms one with an amplitude of 3 FFS and the second with an amplitude of 5 FFS and if the Channel Gain control is set to 50 then Total Channel Amplitude will display 4 FFS In general the A and B channels can have separate Channel Gains however if the A B Lock checkbox is checked the A and B values are always the same Auto On affects the behavior of the generator output
391. which in turn corresponds to 0 7746 FFS Q uy lt Decibels relative to the dBr reference set in the references section of the analyzer references tab The dBr reference is always set in units of FFS Thus with the dBr reference set to 5 FFS an amplitude of 0 dBr corresponds to 5 FFS The Bits unit is related to dBFS by the relationship bits 1 76 dBFS 6 02 Power Spectral Density Units In addition to the normal analyzer amplitude units FFT results can be displayed in power spectral density PSD units The available PSD units are V vHz and dBV vHz PSD units are useful when examining the amplitude of signals that are spread out in the frequency domain If the amplitude of such signals is reported in absolute units such as wolts the amplitude will change depending on the FFT number of lines the FFT frequency range and the FFT window Since these factors are normalized out when using PSD units the measured amplitude reflects only the properties of the input signal rather than the particular FFT setup used Jitter Units Jitter represents the deviation in time between the nominal and actual occurrence of zero crossings in a signal As such time units are used to measure jitter amplitude In SR1 jitter amplitudes can be expressed in seconds or in Ul s The Ul or unit interval is for a digital audio signal the smallest pulse width present in the digital audio signal Since for a normal bi phase encoded 2 channel digital audio
392. width ADC and a 1 Vrms synchronous chirp source outputting equal power into each FFT bin up to 200 kHz Note that throughout the passband of the filter and into the transition region the coherence is unity In the stopband however the output signal is gt 80 dB below the input and at those small levels there exists enough uncorrelated output noise in the filter output to reduce the coherence to a value below one At the zeros of the filter there is virtually no output from the filter and all the SR1 is measuring is uncorrelated noise hence the zeros of the filter correspond to the regions of lowest coherence Coherence 20 40 0 50 100 500 ik 2k Sk 10k 20k 50k 100k He Frequency Response and Coherence of Elliptical Filter Impulse Response SR1 computes the impulse response by taking the inverse FFT of the complex magnitude and phase frequency response Only portions of the frequency response which exceed the coherence threshold See Coherence are used when computing the impulse response Traditionally impulse response measurements have been associated with the use of Maximum Length Sequences as a stimulus The MLS waveform has several useful properties which simplify the calculation of the impulse response and has a crest factor close to 1 providing good signal to noise However because SR1 uses a full dual channel FFT to calculate impulse response it is not necessary to use MLS waveforms as a stimulus in fact mo
393. with transmission bandwidth requirements Because the frequency content of USASI noise is fixed only an 2014 Stanford Research Systems e SR1 Operation Manual amplitude control appears on the waveform tab USASI noise may be used in conjunction with generator bursting to generate a burst USASI signal suitable for transmitter testing Frequency Spectrum of USASI Noise 100 Hz 6dB oct hipass 320 Hz 6 dB oct lopass Square Wave Config Square Mm aveform on fw Square p Precision Frequencies Amp 00 00 mFFS J Limit gii 00000 kHz Perfect digital square waves equal integer numbers of up and down samples are possible only for a limited number of frequencies for each digital OSR If Precision Frequencies is checked square wave frequencies will be limited to these perfect frequencies If Precision Frequencies is unchecked the digital generator will interpolate a square wave at any frequency with significantly less fidelity to an ideal Square wave Ramp Config Ramp Waveform On fw Ramp Frequency Frac RiseTime i 0000 kHz 25 000 7o Low Amplitude 0 90000 FFS ii The ramp waveform consists of repetitive runs of integer numbers
394. y parameter defined as asymmetry U U U U The asymmetry calculated for each input record is displayed on the Advanced tab of the digitizer panel If desired the peak splitting produced by asymmetry in the input record can be removed before calculating the pulse width histogram by checking the Rise Fall checkbox on the Advanced Measure tab of the digitizer panel When the box is checked the digitizer will correct the measured pulse widths by the calculated asymmetry before compiling the pulse width histogram The histogram above when recalculated with Rise Fall checked yields the result shown below 1 90 1 92 1 94 1 96 1 96 2 00 202 2 04 2 06 2 05 I Additional Analysis Controls The Re analyze button causes the digitizer to reanalyze the current digitizer record with possibly different analysis settings without first acquiring a new record The Fast Interpolation check box selects the use a faster but less accurate interpolation algorithm spline interpolation when calculating the zero crossing positions in the input record The standard interpolation algorithm band limited interpolation yields the highest accuracy Digitizer Measurements Spectrum ra Window Amp Lines Jit Lines Blackman Harris 1024 512 ki Probability l iw Aum Bins 256 Eve Diagram r UIs Res Y Res 2 512 256 The input amplitude vs time and jitter amplitude vs time measurements are alw
395. y check waveform uses a phased combination of two sine waves to produce a deliberately asymmetric waveform that points up When this waveform is applied to a device under test it is easy to see if the device properly maintains or inverts polarity by checking the output waveform using the time record of the FFT Analyzer If the output waveform still points up the device maintains polarity If the waveform points down the device is inverting Polarity Waveform Non Iinverted And Inverted Constant Offset Config anst AW aveform 7 On iw Const Offset The constant waveform simply adds a DC offset to the output The offset can be set anywhere over the full positive and negative range of the generator output Offset waveforms take up space in the generator s D A converter and reduce the resolution available to other waveforms For the best distortion and noise keep the offset as small as possible 2014 Stanford Research Systems SR1 Operation 2 3 2 Digital Generator Panel The Digital Generator Panel controls the operation of SR1 s digital audio generator The generator can be populated with many different waveforms sines square waves ramps etc Many of the waveforms can be combined by the generator For instance if the generator is populated with sinewave and noise than the output will be the sum of the sinewave and noise signals The
396. y to pin 2 and with negative polarity to pin 3 The maximum noise amplitude is 640 mV for BNC outputs and 2 5 Vpp for XLR outputs Rise Fall Time varies the of the digital audio outputs BNC and XLR The risetime may be varied between 5 ns 10 ns 20 ns 30ns and continuously from 40 ns to 400 ns Jitter Controls SR1 is capable of generating a wide variety of jitter waveforms frequencies and amplitudes Jitter represents the dewations of transitions in the digital audio carrier signal from their ideal times SR1 has two main methods for viewing jitter The Jitter Analyzer uses a phased lock loop to demodulate the jitter present in the digital audio input carrier and display it in both the time and frequency domains The Digitizer records a digitized version of the input carrier and uses mathematical techniques to extract the jitter Note that it is easily possible to generate more jitter than can be tolerated at SR1 s digital audio inputs The digital audio inputs will not lock if too much jitter is applied Jitter Waveform None si The digital audio output is un jittered The digital audio carrier is modulated with sinusoidal jitter with a frequency between 2 Hz and 200 kHz Applies squarewave jitter with frequencies between 2 Hz and 40 kHz Jitter Applies white gaussian noise jitter Bandpass Noise Applies 1 3 Octave bandpass noise jitter at the selected frequency Chirp The chirp is a special jitter source designed t

SR1 Operations Manual - Stanford Research Systems

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