K6JRF's Page
formerly W6FZC
ESSB Audio Techniques Page
(Updated: Sep 27, 2014)

This FT2000 Page discusses the techniques to attain clean ESSB (Extended SSB) audio to 4Khz SSB BW in the 'ttbf' mode by exloring fundamental radio and audio characteristics required to produce this audio along with how-to-do-this using both the radio's internal EQ plus external audio equipment.

External Audio Equipment
The next choice concerns the audio dynamics. The term 'dynamics' means all steps normally associated with processing of the audio signal. These are: gating, compression, limiting, de-essing and equalization. In order to more comprehend these, click on the many links in the aforementioned section. Take the time to read the details contained on these sites. It will help you understand the 'terminology' and give you ideas as to what is needed in your equipment lineup.

The chart to the right shows my current audio system equipment and the "block" diagram shows all of the major interconnects between\ these equipments. The connections are standard and you will find that the interconnections shown will be required. Of course, these do change from time to time but the basic "ordering" concept has not changed. The numbers in blue at the top right side of each equipment, refer to the input impedance of the stage except for the microphone where this is its output impedance. The balanced interconnections are employed between all units using either XLR or TRS connectors except for the Alesis Microverb. If you aren't careful, you will get "hum" and that's indicates that you have a "ground-loop". The use of balanced interconnections (XLR) will ensure that these are minimized.

K6JRF's Current Audio System Interconnection The output from the dBX DDP is balanced and is transformed in the Ebtech isolator to unbalanced to match the FT2000 mic input connection. One channel of the Ebtech Hum Eliminator is used to ensure a ground loop free connection. The second channel is used for the Sony E10 MDS playback EQ, ensuring complete isolation between the audio processing system and the FT2000.

Audio Equipment Hierarchy
I firmly believe that the 'cleanest' sound is produced by;
1) attenuating any 'unwanted' frequencies before they enter the main processing chain,
2) using equalization (DEQ2496) processing ahead of any Gating, Compression and De-essing (or Limiting),
3) always adding reverberation effects as the last piece of equipment (if you use reverb; I do not use it any longer).


Therefore the my audio processing system reflects this philosophy.
1) For my voice, '400hz' is a no-no. Thus the Behringer MIC2200 preamp's EQ stage is used to remove this frequency before it enters the processing.
2) The Behringer's Ultra Curve DEQ2496 is placed in front of the dBx DDP, Digital Dynamics Processor. This gives the DDP's compressor, limiter and de-esser the ability to compress, limit and de-ess the audio signal so that there is no chance of overdrive, harshness or clipping to the audio signal that drives the mic input of the FT2000.
3) The Alesis MicroVerb III which was at the end of the chain has been removed. The reverb effects are not normally heard unless signals are S9 + 30db or more so this function is mostly useless. However, if you use one, it should be at the end of the chain since if a Gate is use, it will close very quickly so the reverberation effect is 'cutoff'. If you make the Gate unaturally long, you are defeating the reason for the gate!


Real ESSB Example
K6JRF on his FT1000D radio. Spectral plot from VE6CQ
In order to see what's involved, what's better than a picture. These are spectrum plots made with SpectraPlus, a software program that turns your computer's sound card into a low frequency spectrum analyzer. The picture below shows the transmit audio response of my former radio, FT1000D taken by VE6CQ. The Tx bw is basically flat from 50hz to 3.1Khz. The key is the balance that it possess. The difference from top to bottom is less than 3dB - 4dB. Also the mid range frequencies are slightly attenuated so as to bring out the resonant bottom and top frequencies, where the 'articulation' lies.


FT2000 Internal Menus Settings
The following sections are specific to the FT2000 radio and show my current alignments and adjustments along with the external audio equipment and its settings. Unlike my former FT-1000D that required "manual" adjustments inside the radio, most of the FT2000 adjustments can be made via the menus accessible from the front panel! This truly makes it very much easier to adjust the radio's parameters.

Recommended Menu Settings
To access the internal FT2000 menus, press the "MENU" button. Then rotate the main VFO - "A" knob until the desired number is shown. To change the value, rotate the VFO - "B" knob until your choice is shown. Then press "MENU" for 2 - 3 seconds until a 'beep' is heard. The new parameter is now stored.

The settings here reflect the lastest FT2000 FW update: 11.54 + V1.58/V1.59

The following settings cover both "Rx" and "Tx". Some are "default" settings but are included for completeness [1/09]
Menu# Settings Function Comments
001 500 msec AGC Fst Delay for Rx 500 msec - add more delay
002 200 msec AGC Fst Hold for Rx 200 msec - more hold time
003 800 msec AGC Mid Delay for Rx 800 msec - more delay
004 500 msec AGC Mid Hold for Rx 500 msec - more hold time
005 2800 msec AGC Slw Dly for Rx 2800 msec delays rcvr's AGC setting
006 2000 msec AGC Slow Hold for Rx 2000 msec holds current AGC level; quiets the rcvr so noise is not recorded
063 Dir A1A Frequency Display; keeps same frequency when switching from LSB to USB Fixes offset when changing sidebands
084 Front SSB Mic Connection Point Default but needs to be set to "Front"
085 ttbf Audio passband for SSB Tx ttbf = 50hz to 4Khz; widest mode for ESSB Tx
086 0 hz Carrier offset for SSB LSB Tx Set to "0" (default): affects low frequency response
087 0 hz Carrier offset on SSB USB Tx Set to "0" (default): affects low frequency response
090 nor DSP filter passband shape nor keeps more constant AGC level resulting in better AF frequency response recovery
092 +5dB Parametric EQ DSP Gain Flat AF response for SSB Rx signal
093 5 Parametric EQ DSP "Q" factor Wide BW for flat SSB AF response
104 ShAP DSP receiver passband for SSB Tx Primary importance is attached to the "phase" of the filter factor
105 Scp SSB Slope for DSP filter Gives the narrowest passband at filter bottom
125* 100 EQ1: EQ low frequency range Set to 100hz
126* 5 EQ1: Gain of "low" range Minimum gain to boost low-freqs
127* 1 EQ1: Q-factor for "low" EQ range Set to one octave bw
128* 700 EQ2: Center Freq of "lo-mid" range Set to 700hz of EQ2 bandwidth
129* -12 EQ2: Gain of "mid" range Set gain to attenuate lo-mid range
130* 2 EQ2: Q-factor for "mid" EQ range Set to 2 octave bw
131* 3000 EQ3: high frequency EQ Set to near-MAX to EQ corner frequency
132* 10 EQ3: Gain of EQ3 range Set to MAX gain
133* 1 EQ3: Q-factor for EQ3 range Set to one octave of the EQ3 range
134** 100 PE1: EQ low frequency range Set to 100hz
135** 5 PE1: Gain of "low" range Set for mid gain to boost low frequencies
136** 1 PE1: Q-factor for "low" PE1 range Set to 1 octave bw
137** 700 PE2: Center freq of "lo-mid" range Set to 700hz = center of PE2 bw
138** -12 PE2: Gain of "lo-mid" range Set to attenuate lo-mid freqs
139** 2 PE2: Q-factor for "lo-mid" EQ range Set for 2 octave bw
140** 3000 PE3: high frequency EQ Set to upper hi-frequency
141** 10 PE3: Gain of "high" PE3 range Set to high gain
142** 1 PE3: Q-factor for "high" PE3 range Set to 1 octave bw
I've have changed the EQ settings (125 thru 142) for both Mic Eq and Proc to an ESSB type EQ to bring out the 4Khz sound of the FT2000 radio. This represents a departure from my previous settings which were designed for narrow band hi-freq DX type sound. Make sure that Menu #85 is set to "ttbf".

* the "low" [EQ1], "mid" [EQ2] and "high" [EQ3] EQ ranges should be set at shown. If so, you can be assured that the Tx BW is flat from 50hz to 4Khz as shown in the white noise Tx sweep chart below.

** the PE1, PE2 and PE3 ranges are used with the "PROC" = ON and set to 9 - 10 oclock. For details, please see the table "Proper Use of the SSB Speech Processor" down the page.


FT2000 EQ in the 'TTBF' Mode [50hz - 4Khz]
This section treats the methods for FT2000 Rx and Tx EQ in the 'ttbf' mode. The "key" is that the response is tailored by the radio's DSP and the white noise response is the same for Tx as for Rx!. In other words, the response is identical in either mode so only "one" master EQ setup is required to level the response in both modes. That was discovered today and has lead me to this revise the "TTBF" EQ section.

So, in summary, a master EQ can effectively render the radio flat from 50hz to 4Khz on BOTH Rx and Tx. This still implies that two (2) EQ sources are required; one for Rx and another for Tx. The former is accomplished by the DEQ2496 (Rx) and the latter by the DDP (Tx). For normal ESSB microphone EQ, special EQ tailoring is still required. Nothing has changed for that mode.

FT2000 Rx & Tx Sweep Tests

This section shows the FT2000 no-EQ response in either Rx or Tx modes!. This discovery has made it quite easy to 'flatten' the radio's response from 50hz to 4Khz with just one (1) EQ curve resident in the DEQ2496 and DDP. Using WWV @ 10Mhz, the following chart was made using SpectraPlus. It's the raw response from the FT2000 in 'ttbf' mode.

WWV Rx Sweep w/ EDSP 11.54

The overall Rx BW measures from 40Hz to 4KHz. Note that w/ the latest FW updates, the Rx response is much smoother and the upper cutoff is now 4Khz. The droop, starting at 1Khz is apx -8dB and is identical to the response in Tx mode. Since the same DSP shapes and controls the Tx as well as the Rx BW, it's logical that it would exhibit a similar response in either mode.

FT2000 Rx and Tx White Noise Response

The overall response is shown in the blue chart; note the rolloff starting at 3Khz and attaining -8dB @ 4Khz. It's typical for the way ALL Yaesu radios are built. Note that the SECOND blue chart was recorded from the radio's PHONE jack. It shouldn't be used b/c of the extra rolloff (-20dB @ 4Khz). This may be possible to 'fix' by replacing a 'capacitor'. This was done during the FT1000 re-design effort. Hopefully, the same thing may happen as I start to dig into the FT2000 schematics. Until that is accomplished, the PHONES jack output is not very useful b/c of the extreme frequency response rolloff.

The green chart shows first corrected response in the DEQ2496. Note that it's essentially flat and all recordings sent to either the computer or MiniDisc (MD) are recorded with a flat response. Note that when it's re-transmitted additional EQ is needed to compensate for the Tx BW rolloff. Once done, the re-transmitted "copy" of the original signal will be identical to the way it was received. Can't ask for anything better! The "block-diagram" interconnection sketch shows the details of how-to-do-this. For Rx, this is accomplished by use of the DEQ2496.

The red chart shows the extremely flat response generated by the DDP during "playback". It's as flat as flat can be! A ESSB signal playback is indistinguishable from the original. For Tx, this is accomplished by use of the DDP.

The main settings for Rx and Tx BW measurements are: IPO = ON, RFlt = 15Khz; VRF = OFF; Width = Max. Receiving and/ or recording ESSB signals s/b use these settings. This will guarantee the maximum response from the radio.

FT2000 Playback EQ Sweep Tests - "Flat Response"
Most ESSB enthusiaists have the capability to capture Tx audio, display its BW using SpectraPlus (or similiar) and finally, play it back to you as a faithful reproduction of what was captured. This latter function requires that your radio's playback capability be compensated to be 'flat'.

Not all radios have the same playback BW; the FT2000 captures the signal from 50hz to 4Khz and can playback from 50hz to 4Khz. That's where the brick wall formed by the DSP third stage "IF" sets in . However, even with that minor limitation, the EQ must be adjusted to be flat.

To that end, the chart shows the FT2000 playback response after using the dbx DDP's channe1 for the (see block diagram) EQ.
Playback EQ for FT2000 using CH2 of DDP

The DEQ2496 has 31 and the DDP has 3 stages of parametric EQ that can be employed. The values for the center frequency, bandwidth (BW) or "Q" plus the level (in dB) are shown that flatten the FT2000's playback response.

FT2000 TX Equalization: "TTBF_EQ"

Properly EQing for a 4Khz Tx BW is different than for a 3Khz BW. One realization is that b/c of the higher upper frequency repsonse, the low-frequency response must be increased to 'balance' the sound. It's ok to have a slight favoring of the hi-freqs since it will make the sound brighter which is always a good thing.

This section shows the EQ for the FT200's "TTBF" mode. This represents 50hz to 4Khz Tx BW.
K6JRF EQ using DEQ2496 Ultra Curve Pro

The green line shows the FT2000 TTBF Tx No-EQ response to a white noise sweep. As you see, it droops but the amount is not significant since this is just the "cal" run to establish a 'baseline' WN sweep to 4Khz.

The blue line represents the overall Tx white noise EQ response in the TTBF mode. For my voice, I need to remove the 130hz to 300hz low-mid freqs. Starting around 600hz, the response is basically flat to 3.5Khz, then drops at 4Khz, the DSP 'wall'. Overall this EQ has a lot of 'fullness' but is well balanced w/ the upper high freqs.

The MP3 below was recored by Duke, NA1A using a FTdx9000 on 75mtrs. A little band noise but you get the idea!

"TTBF EQ" captured by NA1A
MiniDisk capture by NA1A on FTdx9000

The SP chart made from an MP3 file captured by NA1A shows that it extends to 4Khz as is the "TTBF" mode. The mid-section is basically flat from 200hz to 3.8Khz and drops in level to 4Khz at the DSP's cutoff frequency.

Raw Mic Audio Captured - No EQ
Raw Voice Audio Captured w/ No EQ

DEQ2496 PEQ Settings

Detailed EQ Settings for DEQ2496's PEQ
The EQ was born by capturing my voice via the TLM-103 w/ no EQ onto a Mini Disk. The recording was then captured by SP and a "raw" chart emerged. From this the "bad" spots were attenuated where needed including the low-mids and even the low-bass frequencies. Only areas that needed EQ were 'touched' to remove "peaks". The high frequency area needed some added gain.


Octave BW vs Q
Detailed EQ Settings for DEQ2496's PEQ
In the application here, it's relatively easy to determine the "Q" of the area that needs attention but the value that must be programmed into DEQ2496 has to be entered in "octave bandwidth". The chart shows how to translate "Q" into octave bandwidth.

An example best shows how this is done.
The area at "1497hz" in the above chart has a "Q" of 3.52. This is determined from this formula:

     Q = fo / (f2 - f1)

   where fo = center frequency; f2 = upper frequency; f1 = lower frequency.

So here the Q = 1497 / (1710 - 1285) = 3.52.    Referring to the chart, the BW required to cover that area lies between 1/2 octave and 1/3 octave. The best choice is 1/2 octave since that ensures best "coverage" of the center frequency and all surrounding frequencies between f2 and f1. Of course, you could use 1/3 octave if the ends points don't require a lot of attenuation.

Proper Use of the SSB Speech Processor
A word . . .or two . . .about the use of the FT2000 speech processor. It's pretty simple but worth explaining for those who might not know esp some of the newcomers.

The "only" rule is to . . . NOT . . . use the same setting you use for "hi-fi" SSB. Using the "wide" [1-30 or 3000WB] setting for Dx chasing does not work. The reason is simple; the SSB speech processor clipping products for low frequencies end up lying in the audio passband. This causes the SSB Tx audio to sound very muddy and distorted.

So make sure that audio sound that you want for "DX" chasing is tailored by removing all frequencies below 300hz to 400hz. Such tailoring is setup in the PE1 thru PE3 [#134 thru #142] values in the FT2000 setup table above. If you use these values, you can be assured of high-penetration SSB audio that is "loud" and "clear"!

To use the processor, select the "3-27" [300hz - 2700hz] setting and press the "PROC" button until "MIC EQ" and "PROC" are displayed in the VFO window. Adjust the PROC knob until 5 - 7dB is displayed on the meter with the selection set for "COMP". Listen with the headphones so that you can adjust it properly.

. . How to compensate SpectraPlus for BW Rolloff . .
SpectraPlus (S/P) is a very powerful and user friendly spectral analysis program that can correct the BW rolloff present in all radios. Some are much flatter than others but all need compensation. This note shows an easy way to accomplish this compensation.
Text File Format
Using the above chart (Rx EQ) as a reference, focus on the "blue" Rx EQ plots. There's two (2) plots showing the non-EQ and EQ response. Here's how to generate the compensation file that will 'correct' the sweep for lack of flatness. First, make af "text" file (using NotePad) that looks like the file at the right. This file is called a "MIC" compensation file and is located in the "miccomp" folder in the SP folder. There are some files that can be edited and used as a starting point.

Referring to the file at the right, the file contains lines that represent "frequencies" and the amount of compensation that is required. For example, "2500" represents 2.5Khz and "-6.0" represents the amount of correction, in dB, that's required to flatten the response. After adding the frequencies that are needed for your application, save the file giving it a name such as "Rx_Cal.MIC". The suffix "MIC" is mandatory. Save it in the "miccomp" folder.

Next, with SP active, press "F5" and place a check mark in the lower box labeled, "Enable Compensation". Then use "Select" to point to the file that you created. Once found, click "OK" to enable its use. Now run the SP program.

Select "Rec" and make sure that the input to your sound card is connected to the radio's output such as "Phones" or "Audio Out". When the values are correct, the Rx EQ should make a "straight" line across the SP screen.

Audio Dynamics Processors and Settings:
As you have seen, my current goal to minimize the use of external equipment so as to produce a more natural sound. As a consquence three (3) audio dynamics boxes have been removed and the "rack" repacked to contain all of my current "audio" devices. This section details the current ones in use along w/ their strong points. At the end the current settings are shown for each.
Current Audio Dynamics Equipment in the
MIC2200 Preamp:
A key box that will never (??) be replaced is the "preamp". It's the top box in the photo above. The current one is the Behringer UltraGain 2200. It contains a tube ("GT" Electronics) preamp that produces a warm sound. The main advantage is a one stage parametric (boxed in red) EQ that allows full control of the mic's input signal and pre-processes any frequency from 20hz to 10Khz with "attenuation" or "gain". I use it to remove the heavier low-mid frequency range (150hz to 300hz) BEFORE it gets into the following stages.

Behringer Pro Mic 2200 Preamp
The picture shows a closeup of the controls of one of the two separate preamp sections. The preamp features tunable low cut (from 32hz to 320hz) for removing low "rumble". Currently it's set to 60hz. The parametric equalizer is fully tunable from 20hz to 20Khz with Q and level control. Currently set to 150hz with a Q = 1.0 (BW of 1 octave) and level of -12.5dB.

DEQ2496 Ultra Curve:
The single replacement for the removed audio dynamics processors is the Behringer DEQ2496.   It's the second box down from the top in the photo above. This is now my primary audio processor and I can say that it's a marvelous piece of DSP hardware and software. It tailors the audio dynamics not only based on "frequency" but also "level", so the EQ can change as a function of the input signal level! It can raise or lower the level of a specific frequency (or a band of frequencies) based on two parameters, "Gain" and "Threshold". Other parameters control the speed of how this is done.

The standard usage of this device is to "lower" the gain above a certain level however many are not aware of another way of using the DEQ processor. That is to use it to "RAISE" the level rather than lower it. It has some advantages that would be hard to accomplish with a 'standard' audio dynamics equipment over a wide frequency band.

DEQ Screen #1 Settings An example shows how this happens. Setting the "M-Gain" to a positive number (+10dB) means that as soon as the level drops BELOW the "Threshold" (-23dB), the gain is increased by the amount of the M-Gain number, 10dB. When the input signal level goes above the "Threshold", it is reduced by -10db to the "normal" value.  The results are a louder audio almost as if it was 'compressed' for low (soft) mic input levels.

Another way is take advantage of the DEQ's power is to use the shelving filter, "H12" (high-pass, 12dB/oct) or the "L12" (low-pass, 12dB/oct) rather than the bandpass filter, "BP". This results in a larger region of controlled gain audio.

DEQ Screen #3 Settings In DEQ #1, the H12 shelving filter attenuates all frequencies above 100hz. In DEQ#3 as shown on the right, the L12 shelving filter results in all frequencies below 2000hz being attenuated when signal level crosses the threshold level. This is a novel way of damping frequencies across a large frequency spectrum, 50hz to 2000hz.

Currently using the H12 settings for DEQ #1(Lo Freq) and DEQ #3 (Hi Freq). DEQ#2 (Mid Freq) uses the BP function. Check out my "Current Audio Settings" table below. Try them, they should at least get your SSB audio into the ballpark.


dbX Digital Dynamics Processor (DDP):
My seccondary processor is the DDP. This is the third box down from the top in the above photo. This is an excellent DP and there's no plans to replace or remove it. It has the necessary audio dynamics functions such as Gate, Equalizer, Compressor and Limiter. Currently, just the Gate and Compressor are being used. The DDP is a DSP device and it has an extremely clean response with no audio coloration which is key for any good audio dynamics processor.

DDP Transient Capture Mode
Unlike analog technology with its response speed limitations regarding changes in amplitude, digital signal processing permits differences in amplitude to be identified in advance but you must use a bit of signal delay. Increasing this delay also increases the potential for the intelligent control. Even “looking ahead” by only a few samples is sufficient to ensure the intelligent application of dynamic processing – such as limiting, which ensures an absolutely reliable signal ceiling – without clipping.

The DDP's Transient Capture Mode (TCM) works on the principle of delaying the audio signal and letting a 'control' signal begin to activate the response of the VCA (Voltage Control Amplifier). The overall result is the perception of a very fast moving compressor that is able to catch the front of almost every transient signal, but not delaying the audio signal enough to produce any phase correlation errors. The DDP can vary the delay with the range from 0us to 3ms. This gives the processor enough time to react before the signal arrives at the point of processing.

The controls for TCM lie in the GATE parameter area. This effect is global but the GATE MUST be turned on in order for the TCM module to work. The implementation in the current DDP allows the signal to sound smoother and makes it much easier for the DDP to process the COMPRESSOR, LIMITER and DE-ESSER functions b/c it can capture all transients.

Current Dynamics Settings:
The table above shows my current audio dynamics settings for easy reference. As you see, all settings are changed regularly b/c there are no "sacred" positions. These settings may need to be 'tweaked' abit for your particular radio and microphone but they should get you in the ballpark.

Please note that my FT2000D has the low frequency mod shown
here that is reflected in the DEQ2496 settings below.

[Current Audio Settings for FT2000D: 09/27/14]
These settings are for the "Full_Rack_EQ" using the updated FW: 11.54 EDSP; Main 1.58/1.59
All settings here require the use of the "TTBF" [50hz to 4Khz] mode [Menu#85]. The DEQ LO (low freq) EQ uses the L12 filter and the DEQ HI (hi freq) EQ uses the H12. The BP mode is now used for DEQ MID (mid freq) EQ. Most DEQ2496 settings have changed. The "ugly" 1824hz area has been attenuated by an added filter. Also the Dyn Expansion and Limiter settings have been re-adjusted.
Behringer Ultra Gain Pro2200 PreAmp:
Phantom: On (+48V); [Phase Reverse; OUT]; [Gain= +37dB];
[Cut: On, 50hz];
[Parametric Eq: Fr= 150, x1 (150hz), BW= 0.96, Level= -10dB];
[Output= +4dB]


Behringer Ultra Curve DEQ2496: - [All PEQs are "Param"]
[PEQ #1; Not used; PEQ #2; Not used; PEQ #3; Not used; PEQ #4; Freq=164hz, BW=1/2, Gain=-5.5dB; PEQ #5; Freq=351hz, BW=1/3, Gain=-5.5dB;PEQ #6; Not used; PEQ #7; Freq=1824hz, BW=1/8, Gain=-14.5dB;PEQ #8; Freq=3597hz, BW=1/2, Gain=+3.5dB];
[FBD #1; Freq=452hz, BW=2/60, Gn=-48dB; FBD #2; Freq=1061hz, BW=3/60, Gn=-48dB]
[DEQ: #1- Gn=-+1.5db; Thr=-28dB; Rat=1:5.0; Attk=0.00ms; Thrs=-28dB; Rel=227.4ms; Mode=L12; Freq=62.5hz <-- Lo Freq
#2- Gn=-15.0db; Thr=-42.0dB; Rat=1:5.0; Attk=0.00ms; Thrs=-42.0dB; Rel=227.4ms; Mode=BP; Freq=162hz; BW=1 <-- Mid Freq
#3- Gn=+4.0db; Thr=-26dB; Rat=1:5.0; Attk=0.37ms; Thrs=-26dB; Rel=227.4ms; Mode=H12; Freq=2118hz] <-- Hi Freq
[DYN: Expander; Gn=0.0dB, Thr=-28dB, Rat=1:3.5, Attk=0.00ms, Rel=200.7ms; Limiter; Hold=1.0ms, Thrs=-16.0dB, Rel=227.4ms]


dBx DDP:
[Common for Compressor: Atk: 0.1us; Hld: 180ms, Rel: 360db/sec]
[Compr: OverEsy: Off, Auto: Off, Thrs: -30dB, Rat: 1.3:1, Gain: 3.0db]
[Gate: Thrs: -28db, Rat:1:2.4, Atk: 0.1us; Hld: 180ms, Rel: 360dB/sec, TCM:On, Time:3 msec,]
[Equalizer: OFF]; [De-esser: Not Used]; [Limiter: OFF]
K6JRF Audio Equipment
FBD #1 and FBD #2 are "feedback destroyer" filters used to attenuate blower noise from linear amplifier.
Note: The settings here are for use with the "Full_Rack_EQ". Make sure to turn OFF the radio's EQ settings.

Check back regularly for the latest updates.

. . Standard Sweep Test Setup . .
Pink Noise Setup The photo at the right shows the hookup that I use to sweep my audio rack. The inset photo shows the output signal into my laptop for recording. A MP3, WAV files are generated as well as a SpectraPlus Pro chart.

The Sony MiniDisk (MD) recorder previously recorded all of the needed "stimulus" signals such as sine (20hz to 6Khz) sweep, pink noise and white noise (20hz to 20Khz) sweep signals. Then these are selected and the MD outputs the selected one through a 40dB attenuator into the 'MIC' input of the Behringer MIC2200 Preamp. From there, by activating the appropriate 'bypass' buttons on each equipment, it is possible to sweep each piece of equipment separately to determine its individual response as well as the overall composite response. Then by adding each back 'inline', you can see how each stage tailors the final response.

This is a very staightforward way to use a MD as an accurate "signal generator". The MD is flat from 20hz to 20hz so it serves as an accurate "signal generator".

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