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ANAN-100D SDR Audio Processing Guide

The document discusses software-based audio processing for the Apache Labs ANAN-100D SDR, highlighting successful tools like Stereo Tool and JB Broadcast for enhancing microphone audio. It provides detailed setup instructions for audio processing, including EQ adjustments and multiband compression techniques. Additionally, it mentions the use of iZotope Ozone 5 for advanced audio processing and EQ matching to achieve optimal sound quality.

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Patrick Favre
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0% found this document useful (0 votes)
12 views5 pages

ANAN-100D SDR Audio Processing Guide

The document discusses software-based audio processing for the Apache Labs ANAN-100D SDR, highlighting successful tools like Stereo Tool and JB Broadcast for enhancing microphone audio. It provides detailed setup instructions for audio processing, including EQ adjustments and multiband compression techniques. Additionally, it mentions the use of iZotope Ozone 5 for advanced audio processing and EQ matching to achieve optimal sound quality.

Uploaded by

Patrick Favre
Copyright
© All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd

Software Based Audio Processing - Apache Labs ANAN-100D SDR [Link]

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DDC/DUC 160-6M Transceiver

Software Based Audio Processing


Site Navigation
(this page is unpublished and mainly documents some past audio experimentation)
ANAN Hardware
Users are currently having great success processing their microphone audio in their PC with
Stereo Tool, Breakaway Broadcast, and various DAWs (digital audio workstations) or live VST
Fan Replacement & Cooling
hosting applications (e.g. AudioMulch, Live Professor). Audio is then routed into the ANAN by
RFI - Ungrounded Case
soundcard output cabling or VAC. Using VAC introduces more latency.
Panels
RFI - DC Power Cable Stereo Tool is a very cost-effective software solution and is available as either a stand-alone
Case Disassembly application or as a VST plug-in. There is a free version and a paid version ($40) with a few extra
Using the speaker jack features. HB9UQX is interested in getting more ANAN owners using Stereo Tool.
Fuse Replacement
PiHPSDR front panel JB Broadcast is a free VST plug-in that does an excellent job, is easy to use, and includes AGC,
multiband compressor, ambiance and limiter.

Transmit Also, there's a free repository of VST plug-ins to play with - look under "VST Effects"

QRO Pure Signal If you want to invest in a professional level VST plug-in package, iZotope Ozone series of plug-
NR0V on AM Transmission ins can be found for a reasonable price and offer great performance, low latency, excellent GUIs,
Latency Measured and one of the niftiest features that I've ever used (see "EQ Matching" below).
Opposite SB Suppression
Smooth T/R setup
W1AEX Restore Profiles after Initial Audio Setup
upgrading
(not necessary if you're using iZoptope Ozone 5 Matching EQ - see below)

Transmit Audio Connect your microphone to your computer - an audio interface (USB, firewire) with high quality
preamp is recommended (Focusrite, Presonus, Steinberg, etc) but many will use a separate mic
NR0V on TX Audio Level preamp and/or equalizer connected to a soundcard.
NR0V on TX Audio Latency
NR0V Pure Signal As a first step, consider EQ'ing your mic audio to flatten out the microphone response curve.
Presentation Doing this first will simplify adjustments downstream and also will allow many users to share
Software Audio Processing downstream settings to get similar results, discounting only the differences in each person's
Connecting Line-In Audio voice characteristics (and NOT microphone response differences).
ASIO and DAW Setup
1. EQ your mic audio: As the first processing step, pass the mic audio through an EQ stage
WU2O Virtual Audio
that delivers a flat response from the mic audio. To determine the EQ needed to flatten your mic
W1AEX Audio Setup
response, place a high quality HiFi speaker one foot from the microphone and play pink noise
through the speaker. Record 20 seconds with Audacity and plot the spectrum (Log frequency,
Receive sample size 4096). Make EQ adjustments and repeat until the resulting spectrum is reasonably
flat. Ignore the spectrum above 5KHz. Save the final EQ settings as specific for the microphone
Diversity Reception used. Alternately, use the EQ Matching function built into many high-end VST packages (see
Filter Response Curves Ozone 5 EQ Matching below).
Latency Measured
Attenuator Action Here's the resulting EQ for my Rode Procaster:

Software/Firmware

PowerSDR Release Notes


Angelia FW Releases
DigiMode/CWSkimmer Config
Maintaining settings between
versions
Unbricking your ANAN
PSDR Wisdom files
OpenHPSDR downloads
Android OpenHPSDR

Buffer Size

Buffer size setup

CAT/COM Interfacing
2. Make a WAV recording of your mic audio: Because of the impossibility of making audio
adjustment as you speak live into a microphone (due to latency), make a WAV recording which
Port Configuration
will be used as the audio source (looped playing) for your software audio chain as you monitor
Flex Tuning Knob
while adjusting (either audio into headphones directly or transmitted into a dummy load while
Configuration
monitoring on a second rx). One phrase that is commonly used for audio adjustment since it
fully exercises the frequency spectrum is: "Please orange, I insist that you take the brown zipper
Networking off the blue boom," which can be spoken twice to make a 10 sec WAV recording.

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Software Based Audio Processing - Apache Labs ANAN-100D SDR [Link]

3. Loop Play the WAV recording while transmitting and making audio adjustment:
W1AEX Networking Info
Use either the built-in file player in your DAW if it has one (like Audiomulch does) or play in
Recovering Network
Audacity connected to your DAW via virtual audio cable. Use either PowerSDR MON or a second
Connectivity
receiver for monitoring.

Wish List

Improvement Wish List


JB Broadcast Processor Setup (free)
I'll describe a simple setup routine that will get you going quickly with good results. Use just two
User Tips VST components: the EQ you used to flatten the microphone response and JB Broadcast
processor (JBBP).
G7CNF Shielding/Cooling Mod
External Software Config
VK1HW PureSignal Coupler
Hercules DJ Console Mods
Voicemeeter Pro Video

General Links

Apache Labs Forum


Apache Yahoo Group
KC9XG Yahoo Group
SDR Console v3
OpenHPSDR TeamSpeak
Of all the multiband compressors I've used, this is one of the easiest to use.
Recent changes
Site Manager

edit this panel

For initial setup, I suggest the guidelines from NU9N. You can download the JBBP manual if
desired.

Set crossover frequencies approximately X1=300hz and X2=2500hz


Begin loop play feeding your EQ'd recording through the JBBP. AudioMulch makes this very
easy with its built-in FilePlay. Otherwise use Audacity and connect to your VST Host
software with a virtual audio cable. VB-Audio Virtual Cable is a good free VAC. If you can,
monitor your audio on a second receiver while transmitting into a dummy load. Otherwise,
listen through your soundcard output.
Enable AGC in top right
Set 3 RATIO sliders - 3:1 for gentle compression or 5:1 for moderate compression are
good starting points - the value of any slider that you are adjusting is viewable in the
VALUE box in the top right under AGC
Set the SIDE and AMB sliders to zero (these set ambience and can be experimented with
later)
Turn on the output brickwall limiter, the square immediately to the left ot he PREGAIN
slider
Set the DENS slider to zero (not used)
Set the OUT slider to maximum
Turn off the WLIM (width limiter, for stereo use)
Turn on all 3 bands - white squares next to the THRESHOLD sliders
Adjust the THRESHOLD sliders (above each frequency band) while watching the
compression level meters (0 to -20) initially for peak compression readings of approx 6dB
on each band.
After thresholds are set, carefully adjust the GAIN sliders of each band while listening for
desired tonal balance.

Initial setup is complete. If you want, go back in and add a little ambiance with the SIDE and/or
AMB sliders.

It IS possible to dispense with the EQ stage and feed your mic audio directly into JBBP, and still
get acceptable results but you need to use some extreme settings. Set the low band threshold to
minimum, ratio to maximum, and gain to minimum to start. Also on the high band, set the
threshold to maximum and gain to about 3/4 for starters. These settings give you zero
compression for the highs to help boost them and maximal compression to lows to tame them.

iZotope Ozone 5 VST Plug-Ins


I started using the iZotope Ozone 5 VST plug-ins recently, and have been extremely impressed
with both features and sound quality. There's some very good tutorials on YouTube which help to
quickly get up the learning curve plus the context sensitive help within Ozone 5 is excellent. I
will be regularly updating this section to pass along tips. Within your DAW, I recommend using

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Software Based Audio Processing - Apache Labs ANAN-100D SDR [Link]

the central management plug-in (iZotope Ozone [Link]) which contains all the individual plug-ins
(rather than loading the individual plug-ins themselves into the DAW), and also allows you to
change their order under the "Graph" option.

Multiband Compressor ("Dynamics")


If your multiband processor can display a histogram of the audio (e.g., iZotope Ozone 5), a good
starting point is to adjust the compressor threshold for the "meat" (midpoint) of the histogram
plot, and adjust your limiter threshold so it's just touching the upward tail of the histogram plot.
Solo each of the bands, one at a time, and make these adjustments.

Here is my latest configuration using 4 bands of compression. The gain settings on each band
are a major determinant of the final EQ of the tx audio.

WZ5Q MultiBand Compressor Setup Instructions


Basically, the first thing you need to do is set the Crossover Points on the Low and High Split
Frequencies. Mine wound up being set to around 160cps on the Low and 2.5kc on the High. Our
Input Gain and Output Gains are set to around –3dB. The Peak Limiter is set to around +2dB.
The Gain reduction lights just occasionally show –1dB gain reduction with a –3dB on overshoot
peaks.

The next thing is to set the compressors. The Low Band Compressor is set up to smooth the
lows. This is accomplished by using a slow Attack of about 50ms, a very slow Release of about 2
seconds, a Ratio of about 4:1, and by tuning the Threshold to achieve about 2dB – 3dB of Gain
Reduction. The Make up gain is set to achieve a tonal Mix balance along with the Mids and Highs.

The Mid Band Compressor is set up to give the Mids some density. This is accomplished by using
a moderate Attack of about 20ms to 30ms, a Release of about 1/2 second, a Ratio of about 4:1
to 5:1, and by tuning the Threshold to achieve about 3dB – 5dB of Gain Reduction. The Make Up
Gain is set to achieve a tonal Mix balance along with the Lows and Highs.

The High Band is set up for de-essing and sibilance. This is accomplished by using a very fast
Attack of about .5ms, a very fast Release of about .1 second, a Ratio of about 4:1 to 5:1, and by
tuning the Threshold to achieve about 6dB of Gain Reduction on peaks while talking normally,
and about 20dB on a hard "ssss". The Make up gain is set to achieve a tonal Mix balance along
with the Lows and Mids.

Now that this setting is close, you have to listen to the TX Monitor (Off Air Monitor) and adjust
for effect. While listening, tweak the Low Split Frequency control back and forth to find the best
sounding setting. Now tweak the Threshold settings on the Low and Mid compressors along with
the Make Up gains to get the best balanced sound. Do the same thing with the High Split
Frequency control and the Mid and High Thresholds and Make Up Gains. Readjust the Lows, then
back to the Highs. The closer you get, the more of a difference a small adjustment will make.
When you think your close enough, now you can start to fine tune the Ratio, Attack and Release
timings on the different compressors.

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Software Based Audio Processing - Apache Labs ANAN-100D SDR [Link]

As you can see, this is a very touchy operation. You must be able to hear what you are doing.
We say to listen the the TX Monitor or an Off Air Monitor while you adjust as that is actually your
"Final Mix" and you are tuning for that to sound the best.

Equalizer
EQ Matching - the greatest invention since sliced bread?

There is an alternative to the painstaking initial effort to optimize your EQ from scratch. Use the
"Matching" feature of the Ozone 5 EQ.

Optional first step - first make a WAV recording while playing pink noise out high quality
(flat response) speakers through your preferred microphone
Gather one or more good WAV recordings ("targets") over the air of audio EQ you wish to
emulate.
Then, capture your own raw microphone audio WAV recording with Audacity while
speaking for 10-20 seconds into your microphone preamp fed into your soundcard.
For best results, you should normalize the amplitudes of all recordings to the same peak
level using Audacity's Effect-Amplify function.

Here are spectra of the excellent AM audio of G4BIM and G8NOF (notice their similarity!) plus
my unequalized mic audio into my soundcard. FWIW, I determined (via pink noise tests) that my
Rode Procaster microphone response is almost completely flat.
For ragchew audio, simply use the target soundclip that you wish to emulate as the matching
model ("Apply To"), use Match amount of 100 and Smooth of 0.5. For dx audio, simply use the
built-in "Pink Guide" as the matching target. In all matching cases, you should go back in to the
"EQ Bands" table and enable a HighPass filter (48dB/octave, starting at 100hz or so) to roll off
the low bass.

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iZotope Ozone 5 Equalizer Matching function - my raw audio snapshot in yellow, snapshot from
G4BIM audio clip in white (target), derived matching EQ in red (with manually added 92hz high
pass filter "1"), and resulting audio in green.

You can load multiple target snapshots, and just click on them to instantly change your EQ on
the fly!

You can match your EQ to a different microphone nearly instantaneously - just speak into the
microphone for ten seconds while snapshotting it, and click the match button and the new EQ is
created.

One caveat: your target snapshot should have equal or wider passband than what you want to
realize. It is difficult/impossible to edit the resultant matching EQ (red line) to edit a sharp roll-
off in bass or treble that the matcher creates. This is the one limitation I've found in the
matching EQ.

The Matching function can be a great help to get you a running start (before perhaps some final
adjustments) to reach a preferred EQ.

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Common questions

Powered by AI

Using Virtual Audio Cable (VAC) for routing audio into the ANAN transceiver offers the advantage of flexibility in software audio processing by allowing multiple audio applications to interact without needing physical cable connections . However, the primary disadvantage is the introduction of additional latency due to the digital routing process, which can complicate real-time audio adjustments when using these paths for microphone audio .

To achieve a flat frequency response curve, place a high-quality HiFi speaker one foot from the microphone and play pink noise through it. Record 20 seconds with Audacity and plot the spectrum. Adjust the EQ until the recorded spectrum is reasonably flat, ignoring frequencies above 5KHz . This optimization ensures that any subsequent adjustments focus more on the voice's characteristics rather than hardware response discrepancies.

The iZotope Ozone 5's EQ Matching feature is used to quickly emulate the EQ profile of a target audio clip. To use this, first record your own raw microphone audio. Capture spectra from both your recording and a target audio that exemplifies the desired EQ. Normalize these recordings in Audacity and apply the EQ Match function. This process provides a baseline EQ setup that can facilitate achieving high-quality transmission audio without manually adjusting each EQ parameter from scratch .

Latency impacts live audio adjustments for SDR transmitters by delaying the feedback that operators receive from their transmitted audio, complicating real-time monitoring and correction. This delay can result from processing time in software and routing via VAC, reducing the accuracy of live tuning decisions. Solutions include minimizing digital routing pathways, using low-latency ASIO drivers, and monitoring directly through hardware where feasible, thus enabling more immediate audio feedback and adjustment .

Multiband compression applies different levels of gain reduction across divided frequency bands, which allows for precise control over the audio signal dynamics. Low band settings can smooth lows using slow attack and release times, while mid bands add density with moderate timings. High bands reduce sibilance with fast attack and release. Proper adjustment ensures balanced tonal quality and intelligibility during transmission, but incorrect settings can lead to pumping effects or loss of clarity, thus making the off-air monitor critical for accurate adjustments .

Audiophile-grade equipment, like high-quality preamps and equalizers connected to soundcards, combined with advanced VST plug-ins, enhance SDR audio by offering precise control over sound characteristics. High-quality gear ensures accurate capture and transmission of audio signals, while VSTs like iZotope Ozone provide detailed EQ, compression, and matching capabilities that refine audio output. This integration results in improved clarity and fidelity by compensating for constraints in standard SDR hardware setups .

For real-time monitoring during audio adjustments in ANAN SDR setups, options include using PowerSDR MON or a second receiver to monitor the transmitted audio into a dummy load or direct listening through soundcard outputs. These methods are crucial for ensuring that the audio adjustments made in the software, like EQ or compression settings, translate into the intended audio output, avoiding unintended distortions or artifacts in transmitted signals .

Automatic Gain Control (AGC) in software-based audio processing for SDRs stabilizes output levels across varying audio signals. By dynamically adjusting gain based on input signal levels, AGC ensures that output remains clear and consistent, preventing signal overload or distortion during transmission. Enabling AGC helps maintain the balance between different vocal tones and background noise, enhancing overall communication clarity. However, it requires careful calibration to avoid artifacts like clipping or unnatural audio levels, which can degrade audio quality .

Crossover frequency settings in multiband compressors define the boundaries between different frequency bands, allowing specific control over separate tonal regions. For optimal SDR audio adjustment, each band's frequency settings must be tuned based on the desired tonal balance and dynamic characteristics. Ideally, low crossover is set around the vocal range (e.g., 160cps), and high around 2.5kc. Proper setting ensures smooth transitions and appropriate compression in each band, facilitating a clear, coherent overall sound .

Compression ratio settings in SDR audio processing affect how aggressively signal peaks are reduced, impacting both dynamic range and clarity. A ratio that's too high can cause over-compression, reducing dynamic range and making audio sound flat; too low, and excessive peaks can lead to clipping. Balancing involves setting ratios that maintain a natural sound, typically starting with gentle settings like 3:1 for minimal compression and adjusting based on feedback from the TX monitor. Ensuring proper threshold settings complements these adjustments, allowing for dynamic energy while preserving clarity .

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