Danville Signal Processing dspMusik 2/8 User manual

Danville Signal Processing, Inc.
dspMusik™ 2/8
High Perfor ance DSP Audio Processor
P/N A.03708D
P/N A.03708E
User Manual
Version 2.10

Danville Signal Processing, Inc.
dspMusik™ 2/8
User Manual
Copyright © 2017 Danville Signal Processing, Inc. All rights reserved. Printed in the USA.
Under the copyright la s, this manual may not be reproduced in any form ithout prior ritten permission
from Danville Signal Processing, Inc.
Danville Signal Processing, Inc. strives to deliver the best product to our customers. As part of this goal, e
are constantly trying to improve our products. Danville Signal Processing, Inc., therefore, reserves the right to
make changes to product specification or documentation ithout prior notice. Updated operating manuals
and product specification sheets are available at our ebsite for do nloading.
This manual may contain errors, omissions or “typo’s”. Please send your comments, suggestions and
corrections to:
Danville Signal Processing, Inc.
29687 82nd Avenue Way
Cannon Falls, MN 55009
Trade ark Notice
dspblok and dspBootloader are trademarks of Danville Signal Processing, Inc. VisualDSP++, CrossCore, and
SHARC are trademarks of Analog Devices, Inc.
Contact Infor ation
Danville Signal Processing, Inc.
29687 82nd Avenue Way
Cannon Falls, MN 55009
E-mail: [email protected]
Web Site: http:// .danvillesignal.com
Voice: 507-263-5854
Fax: [phone]
dspMusik™ 2/8 User Manual Page B

Table of Contents
Overvie ...................................................................................................................1
Hard are...................................................................................................................1
Soft are.....................................................................................................................2
I/O Configuration.......................................................................................................3
Balanced Audio - ADC & DAC Level Setting:.............................................................................................3
dspMusik Display/Encoder Operation........................................................................6
dspblok Options.........................................................................................................9
dspblok 21469uac2...................................................................................................................................9
dspblok 21489uac2...................................................................................................................................9
Product Warranty.....................................................................................................10
dspMusik™ 2/8 User Manual Page C

Overview
The dspMusik 2/8 is intended for applications here audio quality is of paramount importance.
Applications include high performance DSP based crossovers for studio monitors and high end consumer
audio systems. It is also ell suited for multichannel decoders and algorithm development and
demonstration.
Loudspeaker manufacturers can use the dspMusik for as an external standalone crossover or as a
development system for Danville's embedded dspCrossover semi custom embedded DSP crossovers used
in po ered monitors
One of the advantages of the dspMusik 2/8 for algorithm developers is that the hard are performance does
not mask the quality of the algorithm. This allo s you to demonstrate your intellectual property at its best.
This manual covers the dspMusik 2/8 as implemented as a complete ready to operate product. It is housed in
an extruded aluminum box ith an LCD display and encoder. The 2x16 character alpha-numeric LCD
display and companion encoder are used for volume control, source selection and adjustment of other
parameters. It can also be controlled via IR using an Apple Remote. The Apple Remote is available from
Apple resellers for about $20.
The dspMusik 2/8 is also available in bare board configurations for integration by OEMs into custom
housings. There are many options including s itching amplifier interfaces, LCD displays and 8 channel
expansion. This manual covers the complete standalone version. The OEM integration manual covers
bareboard configurations ith iring details, physical dimensions and similar information needed to
implement the boards into a custom enclosure.
Most dspMusik 2/8 systems are resold by OEMs ith optimized soft are that supports their product needs.
This manual covers most of the basic features of the dspMusik 2/8, ho ever there might be custom features
implemented in the version of the dspMusik that you are using that are specific to the OEM provider. If you
have an OEM version, the splash screen ill typically have their name and product.
Hardware
The dspMusik 2/8 is po ered by an Analog Devices’ SHARC floating point DSP as implemented on a
Danville dspblok DSP module. There are a number of pin compatible dspbloks that ill ork ith the
dspMusik. These choices ill be discussed later in this manual.
As important as the DSP is to the platform, the audio quality is ultimately determined by the analog
electronics, the data converters and the clock sources. The dspMusik 2/8 supports 2 analog inputs, an S/PDIF
receiver, USB Audio Class 2 and 8 analog outputs. All analog I/O is available as balanced +4dBu professional
or balanced/single ended –10dBV consumer audio levels. The choice is based on soft are configuration. The
sample rate can be as high as 384k ith 24 bit conversions.
Four state-of-the-art AKM AK4495 DACs are used for the 8 analog outputs. Each channel has balanced anti-
imaging filters to attenuate clock feed through and high frequency images. The output level is balanced (or
single ended) ith levels as high as +4dBu (+22dB FS). Each DAC has analog attenuators to optimize the
full scale performance of the converter.
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The analog input circuit is a lo noise balanced input ith very high common mode rejection. The optional
ADC module uses an AKM AK5397, considered by many to be the best ADC on the market. It has a S/N in
excess of 127dB. This is 20-30dB better that most home theater oriented converters.
All of the data converters have a variety of internal filter choices. This include both sharp cutoff and
minimum phase types. Minimum phase filters are favored for most applications, because they avoid pre echo
artifacts.
Minimal clock jitter is of prime importance for high quality audio. If a clock has excessive jitter, a perfect data
converter ould be transformed into something mediocre. The right data at the rong time is the rong
data! The dspMusik 2/8 uses a very lo phase jitter clock ith very careful routing to minimize reflections to
each device. The SHARC DSP has eight ASRCs that provide jitter attenuation far better than a lo noise PLL
is capable of achieving. This is very helpful for rate conversions (generally upsampling) and is absolutely
imperative for S/PDIF streams that are intrinsically noisy from a jitter perspective.
The dspMusik 2/8 uses a standard IEC connector and can be po ered ith any 100-240 VAC 50/60 Hz
mains source. All that is needed is the appropriate mains cord set. This makes the dspMusik 2/8 functional in
international markets ithout special configuration settings.
Software
There are t o primary ays to program the dspMusik 2/8.
Most customers ill opt for DSP Concepts' Audio Weaver. This is a graphical design program that includes as
many as 400 optimized modules. Systems are designed in a drag and drop fashion to create signal flo .
Audio Weaver runs natively on the SHARC DSP ith optimized production quality modules. You design the
signal processing system ithout the need for custom programming. Once you have your layout, you can
adjust parameters in real time. For example, a loudspeaker designer can create a crossover design, make
some measurements and listening tests, and then adjust parameters to fine tune the system ithout
interruption of the test source or music. A run time version of Audio Weaver is included ith the dspMusik
2/8. This lets you design and distribute your application ithout buying additional tools or licensing. All you
need is a Windo s computer.
If you have an OEM version of the dspMusik 2/8, chances are this is ho most of your system soft are as
implemented. The specific layout and tuning as done by the OEM, so that their system performs optimally.
The alternative method ould be traditional programming in C or assembly. In this case, you ould use
either Analog Devices' Visual DSP++ or CrossCore Embedded Studio. You ould also ant an emulator,
either an ICE-1000 or ICE-2000. The older ADI emulators ill also ork ith any of the dspblok 214xx
boards. Danville supplies example code ith device drivers for the converters and a basic talk thru shell
program to get you started.
dspMusik™ 2/8 User Manual Page 2

I/O Configuration
The dspMusik has three stereo input sources, Analog, USB Audio Class 2, and S/PDIF. Each input can be
selected from the display/encoder or Apple remote and have independent gain settings. This is helpful hen
the levels from each source can be quite diverse.
For example, you might have the Analog input connected to a phono preamplifier, the USB connected to
your music server and the S/PDIF to a television. Each of these sources could have very different output
levels. When you s itch bet een each source, the volume level can remain consistent. This also means that
turning the volume do n for one source does not adjust the others. Mute is universal. All sources are muted
together.
There are 8 balanced Analog outputs. These are usually assigned for specific purposes by the soft are. For
example, in a DSP crossover, some outputs are assigned to t eeters, others, to oofers, etc. If you have an
OEM version of the dspMusik, make sure you consult the OEM’s instructions for the correct assignments of
each output.
Balanced Audio - ADC & DAC Level Setting:
The dspMusik 2/8 has both input PGAs (programmable gain amplifiers) and output attenuators to adjust the
audio signals to optimum levels. The goal is to maximize signal level ithout clipping. These s itch in
approximately 6dB steps (a little different for the first step to accommodate dBu to dBV differences).
Nominal audio levels are defined by standard in only t o ays:
Professional level (Balanced) -4dBu nominal (+22dBu full scale)
Consumer level (Unbalanced) –10dBV nominal (+8dBV full scale)
This means that signal levels are much higher ith professional audio equipment (about 11.75dB). We don’t
amplify inputs and e amplify outputs to create these levels.
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Since professional level is defined as balanced and consumer level is defined as unbalanced, it gets a little
more complicated.
Consumer level is usually unbalanced. This means that a proper input cable (maybe from a phono
preamplifier) should connect Center (source) to Hot (dspMusik, tip), and Ground (source) to Ground and
Cold (ring) on the dspMusik TRS. The best ay is ith a three conductor cable ith separate ires (shield &
t isted pair) connected at the source end.
Unbalanced outputs ONLY use the Hot and Gnd on the dspMusik. The Cold connection is left open.
Other ise, The Cold output ill drive into a short (Ground).
When making input connections, a less optimum but certainly convenient interconnect method is to use a
phono to 2 conductor (mono) phone plug adapter. This shorts the Cold and Ground connections at the TRS
input of the dspMusik. DO NOT DO THIS FOR UNBALANCED OUT CONNECTIONS! It ill short the
Cold Output to Ground! Output cables for unbalanced circuits leave Cold open.
In the professional audio case, levels and interfaces are consistent. With consumer gear, everything gets a
little sketchy. Some consumer gear no has balanced audio connections available. The problem is that
–10dBV is no longer is clearly defined. If balanced connections are used, should levels be measured from
Hot to Cold or alternatively, Hot to Ground? This is a 6 dB difference and different manufacturers have
different ideas.
Another issue that comes up is po er amplifier gain. Most po er amplifiers have a gain of bet een 20 and
30 (26dB to 31dB). This can be referenced from balanced or unbalanced inputs. This means that a large
po er amplifier might need to about 3.5V peak to s ing to full range. If the output gain is insufficient, the
amplifier ill not s ing to maximum po er. In general, you use the settings that allo the amplifier to go full
range and not more. This ill usually be the Consumer –10dBV output settings ired as balanced if the
amplifier has balanced inputs and single ended (unbalanced) if it does not.
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dspMusik Naming Conventions
Input Levels:
Pro Audio +4 dBu
Balanced –4 dBV 5.75 dB Gain
Consumer –10 dBV 11.75 dB Gain
Balanced –16 dBV 17.75 dB Gain
Output Levels:
Pro Audio +4dBu
Consumer –10dBV 5.75 dB Attenuation (same level as Balanced –4 dBV)
Balanced –10dBV 11.75 dB Attenuation
Balanced –16dBV 17.75 dB Attenuation
On some displays, the levels may be labeled as +6 dB, +12 dB, etc for Inputs and –6 dB, -12dB etc for
Outputs.
When e are referring to Consumer Input and Output, the assumption is that e are iring the cables in an
unbalanced configuration. For inputs, this means that e do not have an out of phase signal available for the
Cold input. A +4dBu (Pro) balanced signal is the nominal level bet een Hot and Cold. A –10dBV Consumer
level is the nominal level bet een Hot to Gnd.
The “Balanced” labels are expected nominal levels from Hot to Cold. A Balanced –10dBV signal ould be
6dB lo er than the Consumer –10dBV level since Hot to Gnd is only half of the signal.
It is also possible that a manufacturer decided that it ould provide a balanced output here the level is
–4 dBV. This ould allo the unbalanced circuit (leaving Cold disconnected) to be –10dBV. It ould have
t ice the voltage s ing hen the balanced ires are used. This means that the input circuit should have less
gain in this iring configuration.
This is admittedly confusing. The good ne s is that if you are interfacing pro audio equipment, you just set
the levels to the Pro settings. If you are interfacing to most consumer equipment (RCA Phono) or PC
soundcards; make your cables as discussed above and use the consumer settings.
dspMusik™ 2/8 User Manual Page 5

dspMusik Display/Encoder Operation
The display/encoder consists of a 2x16 alphanumeric display and a rotary encoder for control. The encoder
rotates continuously and also has a pushbutton shaft s itch.
There is also an IR receiver that allo s alternative control from an Apple Remote.
The Apple remote is assumed to be the later style (silver) that has 7 buttons.
Ergonomically, the Encoder and Apple Remote ork in very similar fashion. Both controls can operate at the
same time.
The encoder functions are as follo s:
Rotating clock ise increases the selection or moves to ard the end of a list.
Rotating counterclock ise decreases the selection or moves to ard the start of a list.
Normal Click (pushing like a mouse click) is a normal selection, orks like an Enter key.
Doubleclicking (similar to a mouse) toggles mute.
Long Click (holding the encoder button do n for 2 seconds).
The Apple Remote functions are as follo s:
Up or Right Arro s function the same as a clock ise rotation of the encoder.
Do n or Left Arro s function the same as a counterclock ise rotation of the encoder.
Center functions like a normal click or long click. There is no doubleclick functionality.
The Pause Button functions like a doubleclick (mute).
Menu is a hot key that eliminates the need for the Source Selection mode discussed belo .
There are three main modes and one temporary mode:
Splash
Volume
Source Selection
Settings
Splash Mode
The splash screen typically displays company name and product and lasts for about 5 seconds.
During this time, all the nonvolatile settings from the dspMusik are transferred to the display
processor. This is ho the display/encoder kno s the current source selection, volume settings, etc.
This process is invisible on the display. During this time the encoder is ignored.
Volume Mode
The volume mode is the normal operating mode. It displays the volume setting in 0.5dB steps. 0dB is
maximum setting. It also displays the selected source. Volume level is independent for each source.
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Rotating the encoder changes the volume for the selected source. Alternatively, you can use the Up,
Do n, Left or Right buttons of the Apple Remote. Doubleclicking the encoder or pressing the Pause
button of the remote causes the system to mute or unmute for all sources. When the system is
muted, it is sho n on the display.
Example Screen:
Volume -100.0 dB
S/PDIF Muted
Source Selection Mode
Source selection is invoked by a normal click of the encoder hen starting in Volume Mode. Rotate
the encoder to the desired selection and press the encoder again. This ill accept the ne source
setting and return to Volume Mode. Alternatively, you can use the Apple Remote Menu button to
toggle through the selections. This is quicker and bypasses the Source Selection Mode (or returns to
Volume mode if already in Source Selection mode).
Example Screen:
Source Selection
S/PDIF
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