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.
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

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: 11.53 + V1.49
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
|
2200 msec
|
AGC Slw Dly for Rx
|
2200 msec delays rcvr's AGC setting
|
|
006
|
1500 msec
|
AGC Slow Hold for Rx
|
1500 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
|
1-30
|
Audio passband on SSB Tx
|
1-30 (50hz - 3.0Khz) has less rolloff; for ESSB Tx
|
|
086
|
0 hz
|
Carrier offset on 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
|
Slp
|
DSP filter passband shape
|
Slp increases as AGC level is present resulting in further rcvr quieting
|
|
092
|
-20
|
Parametric EQ DSP Gain
|
Lower value removes "noise" from Rx signal
|
|
093
|
11
|
Parametric EQ DSP "Q" factor
|
Wider Q = more BW which removes "noise" from Rx signal
|
|
104
|
ShAP
|
DSP receiver passband for SSB Tx
|
Primary importance is attached to the "phase" of the filter factor
|
|
105
|
GEnt
|
SSB Slope for DSP filter
|
Allows for widest passband at filter bottom
|
|
125*
|
100
|
EQ1: EQ low frequency range
|
Set to 100hz
|
|
126*
|
2
|
EQ1: Gain of "low" range
|
Set gain to EQ low frequencies
|
|
127*
|
1
|
EQ1: Q-factor for "low" EQ range
|
Set to lowest level
|
|
128*
|
1500
|
EQ2: Center Freq of "mid" range
|
Set to 1500hz as center of EQ2 bandwidth
|
|
129*
|
3
|
EQ2: Gain of "mid" range
|
Set gain to flatten mid-range
|
|
130*
|
1
|
EQ2: Q-factor for "mid" EQ range
|
Set to mid-range to effect large bw
|
|
131*
|
2900
|
EQ3: high frequency EQ
|
Set to MAX to EQ corner frequency
|
|
132*
|
10
|
EQ3: Gain of EQ3 range
|
Set to MAX gain
|
|
133*
|
3
|
EQ3: Q-factor for EQ3 range
|
Set to level the EQ3 range
|
|
134**
|
200
|
PE1: EQ low frequency range
|
Set to 200hz
|
|
135**
|
-20
|
PE1: Gain of "low" range
|
Set gain to attenuate low frequencies
|
|
136**
|
1
|
PE1: Q-factor for "low" PE1 range
|
Set to widest bw (=1)
|
|
137**
|
1200
|
PE2: Center freq of "mid" range
|
Set to 1200hz = center of PE2 bw
|
|
138**
|
8
|
PE2: Gain of "mid" range
|
Set gain to boost uppper-mid-freqs
|
|
139**
|
2
|
PE2: Q-factor for "mid" EQ range
|
Set to middle to effect largesr bw
|
|
140**
|
2200
|
PE3: high frequency EQ
|
Set to upper mid-frequency
|
|
141**
|
+10
|
PE3: Gain of "high" PE3 range
|
Set to high gain
|
|
142**
|
1
|
PE3: Q-factor for "high" PE3 range
|
Set to widest bw on PE3 range
|
* 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 90hz to 3Khz 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 12 noon; the values emulate a "HC5" mic element
for DX operation. 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.

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.

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.
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.
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
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
DEQ2496 PEQ Settings
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
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 too . . . 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.

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.
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.
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.
An example shows how this happens. Setting the "M-Gain" to a positive number (+15dB) means that the as soon as the level drops
BELOW the "Threshold" (-35dB), the gain is increased by the amount of the M-Gain number, 15dB. When the input signal
level goes above the "Threshold", it is reduced by -15db 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" (short for high-pass) or the "L12"
(short of low-pass) rather than the bandpass filter, "BP". This results in a larger region of controlled gain audio. To
this end, check out the DEQ #1 for L12 and DEQ#3 for H12 usage in the "Current Audio Settings" table below. The M-Gain
parameter is "negative" however. The frequency is set to 2349hz so by use of the shelving filter, H12, all frequencies
from 2349hz to 3000hz (DSP "wall") are attenuated when signal level crosses the threshold level. In DEQ #1, the low pass
shelving filter attenuates all low frequencies from 20hz to 121hz. Again a novel way of damping the low frequencies
across a large spectrum. Try both, they might work for you!
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.
[Current Audio Settings for FT2000D: 1/11/10]
These settings are for the "Full_Rack_EQ" using the updated FW: 11.53 EDSP; Main 1.50
All settings here represent a change for the "TTBF" Tx [50hz to 4Khz] setting and the change from "shelving" (L12) to
"bandpass" filters for LO and HI frequency equalization. Many DEQ2496 settings have changed. Low end boost @ 50hz
has been added and the "ugly" (1.5Khz area) hi-frequencies have been attenuated by the BP filter instead of the L12
shelving filter.
Behringer Ultra Gain Pro2200 PreAmp:
Phantom: On (+48V); [Phase Reverse; OUT]; [Gain= +37dB];
[Cut: On, 50hz];
[Parametric Eq: Fr= 160, x1 (160hz), BW= 0.96, Level= -12.5dB];
[Output= +4dB]
Behringer Ultra Curve DEQ2496: - [All PEQs are "Param"]
[PEQ #1; Freq=52.0hz, BW=1/5, Gain=+4.0dB; PEQ #2; Freq=82.4hz, BW=1/4, Gain=+7.5dB;
PEQ #3; Freq=108.0hz, BW=1/6, Gain=-2.5dB; PEQ #4; Freq=164hz, BW=1/2, Gain=-6.0dB;
PEQ #5; Freq=351hz, BW=1/2, Gain=-5.5dB; PEQ #6; Freq=669hz, BW=1/8, Gain=-5.5dB;
PEQ #7; Freq=3855hz, BW=3/4, Gain=+6.5dB]
[FBD #1; Freq= not used . . . .]
[DEQ: #1 Gn=-6.5db; Thr=-30dB; Rat=1:3.0; Attk=0.37ms; Thrs=-30dB; Rel=135.3ms; Mode=BP; Freq=170hz; BW=1
<-- Lo Freq
#2- Gn=-12.5db; Thr=-30dB; Rat=1:5.0; Attk=20.1ms; Thrs=-30B; Rel=304.2ms; Mode=BP; Freq=335hz; BW=3/2 <-- Mid Freq
#3- Gn=-9.0db; Thr=-31dB; Rat=1:5.0; Attk=0.34ms; Thrs=-31dB; Rel=107.6ms; Mode=BP; Freq=1570hz; BW=3/4] <-- Hi Freq
[DYN: Expander; Gn=0.0dB, Thr=-26dB, Rat=1:2.5, Attk=0.08ms, Rel=330.3ms; Limiter; Hold=1.0ms,
Thrs=-11.0dB, Rel=56.5ms]
dBx DDP:
[Common for Compressor: Attack: 0.1us; Hld: 100ms, Rel: 360db/sec]
[Compr: OverEsy: Off, Auto: Off, Thrs: -27dB, Rat: 1.3:1, Gain:3.0db]
[Gate: Thrs: -27db, Rat:1:3.5, Hld: 205ms, Rel: 360dB/sec, TCM:On, Time:1 msec,]
[Equalizer: OFF]; [De-esser: Not Used]; [Limiter: OFF]
|
 |
FBD #1 is a "feedback destroyer" filter derived from a FBD scan. Not currently used
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 . .
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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