Prism sound DREAM DA-1 User Manual

Prism

Dream

Operation Manual

Issue 1.1 12 February 1996

Prism Media Products Limited

William James House

Cowley Road

Cambridge

CB4 4WX

UK

Phone: [phone]

Fax: [phone]

Email: [e-mail]

Prism Media Products Limited Dream
Operation Manual

CONTENTS

1. INTRODUCTION ............................................. 4

2. SUMMARY OF DA-1 FEATURES ................................ 4

3. GETTING STARTED .......................................... 5

3.1. Un-packing your DA-1 .................................... 5

3.2. Using the DA-1 for the first time ............................ 5

3.3. DA-1 product concept and capabilities ....................... 7

3.4. Dynamic Range Enhancement (DRE) ........................ 7

4. OPERATION ................................................ 7

4.1. Normal mode controls .................................... 8

4.2. Set inputs mode ........................................ 9

4.3. Set level mode ......................................... 9

4.4. Set master mode ....................................... 10

5. FRONT-PANEL INDICATORS ................................. 11

5.1. Input selection or maximum output level setting indicators ....... 11

5.2. 32kHz, 44.1kHz and 48kHz indicators ...................... 12

5.3. `Other' indicator ........................................ 12

5.4. `Master' indicator ....................................... 12

6. CONNECTORS ............................................. 13

6.1. Analogue output connections ............................. 13

6.2. Interconnect screens - the pin 1 conditions ................... 13

6.3. Connector table ........................................ 14

7. SPECIFICATION ............................................ 15

7.1. Digital Inputs .......................................... 15

7.2. Synchronization ........................................ 18

7.3. Digital Outputs ......................................... 19

7.4. Analogue Outputs ...................................... 23

7.5. Performance Specification ............................... 23

7.6. Power ............................................... 26

7.7. Physical Dimensions .................................... 26

8. INTERNAL CONFIGURATION OPTIONS AND FUSES .............. 26

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8.1. Fuses ............................................... 26

8.2. PCB jumper links ....................................... 27

8.3. Mains transformer voltage selection ........................ 27

9. ELECTROMAGNETIC COMPATIBILITY .......................... 27

10. REFERENCES ............................................. 28

Manual Revision History

6 September 1995 Version 1.0 First issue
12 February 1996 Version 1.1 Hardware revision C

Product Revision History

February 1996 Hardware revision C
Note that prior to revision C the DA-1 had a different output configuration from that

described in this manual. If your unit serial number is less than 090 and the serial number
label does not state "Rev C" (indicating that the unit has been upgraded to hardware
revision C) then this manual is not appropriate. Please ask your supplier for a version 1.0
manual. If in doubt as to the revision status of your unit the older configuration can be
identified by a short circuit between the analogue output XLR pin 1 and the chassis.

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1. INTRODUCTION

The Dream converter is intended for professional audio use in applications where
a very high performance is required for monitoring; for example in the production of high
quality 16-bit CD masters - where noise shaping can provide a weighted dynamic range
of more than 110dB, or for conditions such as live recording where a large amount of
headroom may be required.

The Prism Dynamic Range Enhancement (DRE) system is also incorporated for use with
monitoring and/or decoding a DRE encoded source.

2. SUMMARY OF DA-1 FEATURES

Ž 24 bit input resolution
Ž 115 dB A-weighted dynamic range
Ž -104 dB THD+N at full scale (0.0006%)
Ž Flat low frequency response (-0.05 dB at 5.0 Hz)
Ž Floating balanced outputs
Ž Calibrated maximum output level adjustment:

+5 dBu to +28 dBu in 1 dB steps
fine trim in steps of 0.05dB

Ž DRE decoding to analogue and digital outputs
Ž Input selection from 7 inputs
Ž Digital inputs in AES3, SPDIF, Optical, AES-3ID coax, and SDIF-2 formats
Ž Digital outputs in AES3, SPDIF, Optical, and SDIF-2 formats simultaneously
Ž Digital interface format conversion
Ž Mute control on front panel
Ž Invert control on front panel
Ž De-emphasis of emphasised signals (Both CD and J17 characteristics)
Ž Low jitter, high precision clock with:

Capture range better than IEC958 consumer specification (±1500ppm)
Excellent jitter tolerance ( >200ns at 2kHz)
Excellent jitter rejection (-60dB at 500Hz)

Ž Clock master mode for use with digital audio workstations

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3. GETTING STARTED

It is not necessary to read all the manual before being able to use the Dream . This
section contains all the information required for you to get going straight away.

3.1. Un-packing your DA-1

Check that you have the following items and that they are undamaged:

" Dream converter unit
" Dream operation manual (this book)
" IEC320 type mains lead with appropriate plug for your supply

Check that your unit carries a label on the rear panel indicating the correct mains voltag e
for your application area and that the plug fitted to the mains lead is of the correct type. If
not, DO NOT CONNECT THE MAINS SUPPLY, but contact your distributor.

Please keep the packaging for re-use in the event that the unit should be shipped to
another location or in the event that it should ever need to be returned to the manufacturer
for repair or upgrade.

3.2. Using the DA-1 for the first time

Connect a source of digital audio to an appropriate input connector on the DA-1. Note that
if the source is in the Sony Digital Interface Format (SDIF-2) then the three BNC inputs,
labelled L, R and WCK need to be used, otherwise only one connector is required.

The analogue outputs of the DA-1 are wired as follows:

pin 1 Screen and mid-point of balanced output
pin 2 Balanced output (Hot or `+')
pin 3 Balanced output (Cold or `-')

Conventional connections to a balanced analogue input should use a screened twisted
pair lead. The DA-1 output pins 2 and 3 should be wired to the two conductors of the pair
and pin 1 to the cable screen. The other end of the cable should be connected in a similar
manner.

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The DA-1 can also connect to unbalanced inputs. For optimum performance the DA-1
should be connected to an unbalanced input using only pins 1 and 2. Pin 2 should connect
to the signal conductor and pin 1 should connect to the ground or return. The signal level
is then halved. (It is possible to drive unbalanced inputs from pins 2 and 3 but this
unbalances the output and can lead to hum. In any case do not connect the three output
pins to each other).

Connect the mains supply and switch on the DA-1 unit.
The front-panel lights should all illuminate momentarily; if any of them fails to light it may be

faulty. If so then check again by turning the unit off and on. If the problem is still evident
contact your distributor.

After the unit has initialised one of the seven input selection lights on the left of the front
panel will illuminate to indicate the selected input. This indicator will be steadily on if the
signal at the selected input is being decoded without error. It will flash if a signal is not
detected, and briefly extinguish when individual errors are detected. To select a different
input press the round select push-button on the left hand side of the front panel.

When you have selected the input you require then, if the signal is decoded correctly and
the unit is configured to the factory default settings, then only the appropriate sample rate
indicator and the input selection indicator will be steadily on. This indicates that the unit
is now operating correctly. If the output level needs adjustment refer to section 4.3.

NOTE: If you are having problems with any of these controls it may be because the configuration has been altered
from the factory defaults. If you wish to return all the DA-1 settings to the factory defaults then switch the unit on
while pressing all four of the right hand buttons during the whole of the start-up cycle - this takes approximately
six seconds.

In the default configuration the indicators for the four right hand side controls should not be on. If the set button
indicator is on then press the button to leave `set' mode. If any of the other control button indicators are on then
press the respective control buttons to toggle them off.

If you are still having problems setting the unit into the standard configuration described above, then refer to the
descriptions of the set inputs (section 4.2), or set master (section 4.4) controls.

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3.3. DA-1 product concept and capabilities

The Dream is designed to provide a very high quality of digital to analogue
conversion without sampling jitter or data truncation - whatever the resolution or timing
accuracy of the source.

Two other useful features are also provided. These are a set of jitter filtered digital outputs
and a Dynamic Range Enhancement (DRE) decoder.

The digital outputs normally provide the same data as present on the selected digital input,
translated into the appropriate format, and any incoming jitter is filtered off. This means
that the Dream is also a digital audio format convertor, corrector and de-jitterer.

3.4. Dynamic Range Enhancement (DRE)

DRE is a process designed for the increasing the dynamic range of 16 bit recording
channels, such as DAT, CD-R, or 1630+U-matic, when further post-processing is
required. It requires an encode process on recording and a decode process on playback.
It is suitable for applications where 20-bit dynamic performance is desired of the recorder
but only a 16-bit recorder is available, and where the requirement for a decode process
is not a problem. The Prism Sound Dream AD-1 20-bit analogue to digital converter can
encode DRE signals from digital or analogue sources, and can also decode them (in
digital to digital mode) to transfer the high resolution signal, encoded on a 16-bit tape, onto
a 20 bit digital audio workstation. The Dream can be used with the AD-1 to provide
analogue monitoring of the encoded signal, as well as to perform the digital to digital
decoding function required for transfers to the `linear' or `non-DRE' domain.

4. OPERATION

There are five control buttons. Four of them operate in two ways, and the fifth, SET, button
switches their function. For normal use they operate in isolation to select the digital
source, MUTE the output, INVert the output and turn on the DRE decode function. The
configuration settings - which are intended to be set and then left alone - can be modified
from one of the three `set' modes: set inputs, set level and set master . The SET button
is illuminated to indicate one of these modes.

All the control settings are saved in permanent memory so that they are remembered when
the unit is off.

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4.1. Normal mode controls

The following functions apply to the controls only if the SET control is not illuminated. This
shows that the unit controls are in the normal mode. The unit is always in normal mode
after being switched on.

4.1.1. Select
Pressing this button causes the selected digital input to switch to the next enabled input.

NOTE If this control does not have any effect, or it will not select the required input, this may be because some
inputs have been disabled. Refer to the description of the SET inputs control, section 4.2, to change this.

4.1.2. MUTE
This control is used to mute the analogue output. The digital outputs are unaffected.

4.1.3. INV
This control is used to invert the polarity of both analogue outputs. The digital outputs are

unaffected.

4.1.4. DRE
This control is used to switch on the DRE decoding function on the digital input (DRE is

described in section 3.4). Both the analogue and digital outputs are affected by this
control.

4.1.5. SET
Holding this button down while pressing one of the three adjacent switches puts the unit

into one of the following control modes: Either Set Inputs, Set Level, or Set Master, as
described by the red text. These modes allow the user to modify the configuration of the
DA-1. In normal use they are not required.

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4.2. Set inputs mode

The set inputs mode is used to enable or disable the selection of digital inputs. This
avoids cycling through unused inputs when switching between sources. An input will be
skipped if it is "disabled". This mode is indicated by both the MUTE/Inputs and the SET
controls being lit. (Note in this mode the MUTE button has the function of enabling the
inputs. For that reason it is referred to as the MUTE/Inputs control.)

To enter set inputs mode from the normal mode (ie. if the SET control is not illuminated)
hold the SET button down and press the MUTE/Inputs button.

To enter set inputs mode from one of the other set modes (ie. if the SET control is
illuminated) just press the MUTE/Inputs button.

The seven input indicators on the left of the unit illuminate to indicate the inputs that are
enabled. One of the indicators will be flickering to show the input that would be affected
by the MUTE/Inputs control. Press the MUTE/Inputs control to toggle the enabled state
of this input. The flickering will toggle between fast and mostly on, indicating enabled, to
slow and mostly off, indicating disabled.

Press the small round select button to cycle through the other inputs.
To leave set inputs mode then either select another of set modes by pressing INV/Level

or DRE/Master, or change to normal mode by pressing SET.

4.3. Set level mode

Set level mode is used to adjust the maximum analogue output level. This is the output
level corresponding to an digital input at full scale. It can be set to any value between
+5dBu to +28dBu in 1dB steps. A fine trim, in 0.05dB steps, is also possible.

Set level mode is indicated by both the INV/Level and the SET controls being illuminated.
(Note that in this mode the INV button has the function of enabling the inputs. For that
reason it is referred to as the INV/Level control.)

To enter set level mode from the normal mode (ie. if the SET control is not illuminated)

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hold the SET button down and press the INV/Level button.
To enter set level mode from one of the other set modes (ie. if the SET control is

illuminated) just press the INV/Level button.
The seven lights at the left hand side will show a pattern that relates to the output level. The

text in red under the LEDs indicates the value of each LED. The maximum output level (in
dBu) is calculated by adding the value of all the LEDs that are on. (Remember that when
connecting to an unbalanced input as described in section 3.2 the level will be attenuated
by 6dB.)

Pressing the select button will cycle through the possible maximum level settings. Each
press will raise the level by 1dB until the highest level is reached. The next press will then
set the minimum level.

Fine level adjustment can be achieved by holding down the INV/Level control while
pressing the select button. This puts in a level trim adjustment of -0.05dB for each press
of the select button until -0.95dB. The next press removes the trim adjustment. The
amount of trim that is applied is not displayed. If the trim is active the control will flash when
in set level mode. The trim can be removed by pressing the left hand button once. (In this
event the gain does not go up by the normal 1dB step. The first press will only disable the
gain trim function. This is shown by the INV/Level light staying on rather than flashing)

Note that the best dynamic range is achieved when one of the standard levels (+12, +18, +22, +28) is used and
the other LEDs (-4, -2, -1) are not illuminated. The latter LEDs show the amount of digital attenuation that has
been applied. As this attenuation lowers the maximum level but not the noise floor it reduces the dynamic range
by that amount. For example, with a level setting of +13 dBu the analogue level is set to +18dBu with a digital
attenuation of 5dBu. Therefore the dynamic range is reduced by 5dB compared with the 18dBu setting.

To leave set level mode then either select another of set modes by pressing MUTE/Inputs
or DRE/Master, or change to normal mode by pressing SET.

4.4. Set master mode

Set master mode is used to set the DA-1 to be clock master and change the sample rate
it uses when master. As a clock master the DA-1 requires incoming data to be
synchronized to the DA-1 internal clock. Any of the DA-1 digital outputs can be used as
a synchronization reference to the source of the data. It is used in conjunction with
equipment that requires, or benefits from, external synchronization from a high quality
clock. This mode is indicated by both the DRE/Master and the SET controls being

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illuminated. (Note that in this mode the DRE button has the function of enabling the inputs.
For that reason it is referred to as the DRE/Master control.)

To enter set master mode from the normal mode (ie. if the SET control is not illuminated)
hold the SET button down and press the DRE/Master button.

To enter set master mode from one of the other set modes (ie. if the SET control is
illuminated) just press the DRE/Master button.

Pressing the DRE/Master control toggles between master (internal) and external
synchronization. This is indicated by the amber Master light situated beneath the
frequency lights at the left of the front panel. If this light is on then the DA-1 is synchronized
to a precise free-running internal oscillator otherwise it is synchronized externally. (If it is
flashing then one of two error states is indicated as described in section 5.4).

If the DA-1 is configured as a clock master then the sampling frequency can be changed
by pressing the select button. It will cycle between the available sample rates. If not
configured as clock master then the select button has no effect.

To leave set master mode then either select another of set modes by pressing
MUTE/Inputs or INV/Level, or change to normal mode by pressing SET.

5. FRONT-PANEL INDICATORS

5.1. Input selection or maximum output level setting indicators

The horizontal bank of seven green LEDs on the left side of the front panel performs three
functions:

5.1.1. Selected input indication
In normal mode (the SET control not illuminated) one of the LEDs will be illuminated to

show the currently selected input. It will flash if that input is not present or incorrect. On
receipt of single errors it will briefly extinguish.

5.1.2. Enabled inputs indication
In set inputs mode (the SET and inputs controls being illuminated) the LEDs

corresponding to the enabled inputs will be illuminated. The LEDs corresponding to the

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disabled inputs will be off. One LED will be flashing to indicate the input that will be
enabled or disabled if the inputs control is pressed. (See section 4.2)

5.1.3. Peak output level indication
In set level mode (the SET and level controls being illuminated) the LEDs give an

indication of the output level corresponding to peak code. This level, in dBu, is calculated
from the sum of the illuminated LEDs. (When connecting to an unbalanced input as
described in section 3.2 the output level is 6dB less than this value.)

For example: +20dBu is shown by only the +22 and -2 LEDs being illuminated; +28dBu
is shown by only the +28 LED being illuminated; A typical consumer unbalanced level of
8dBu, which is equivalent to balanced 14dBu, needs to be indicated by the +18 and -4
LEDs being indicated.

5.2. 32kHz, 44.1kHz and 48kHz indicators

If the operating sample rate is within the capture range of the high precision PLL then one
of these indicators will indicate the sampling frequency. The indicator will flash if the unit
has lost the source of synchronization and defaulted to free-run from the internal clock.

5.3. `Other' indicator

This indicator will illuminate if the unit is externally synchronized but the sample rate is
outside the range of the high precision PLL.

5.4. `Master' indicator

LED off: Sync OK

- external synchronization selected with no errors

LED flashing in time with one of the frequency indicators: No Sync

- external synchronization selected but failed

LED on: Clock master mode

- either no input or the input is correctly synchronized to the DA-1 master

LED mostly on but occasionally blinking off: Sample slipping in master mode

- each blink corresponds with a missed or repeated audio sample

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6. CONNECTORS

6.1. Analogue output connections

Conventional connections to a balanced analogue input should use a screened twisted
pair lead. The DA-1 output pins 2 and 3 should be wired to the two conductors of the pair
and pin 1 to the cable screen. The other end of the cable should be connected in a similar
manner with the screen connecting to the chassis of the analogue input.

The DA-1 should be connected to an unbalanced input using only pins 1 and 2 of the output
XLR. Pin 2 should connect to the signal input and pin 1 should connect to the input ground.
Using this method the signal level is half the nominal value.

NOTE: It is possible to drive unbalanced inputs from pins 2 and 3 of the DA-1 analogue output but, because the
connection has been unbalanced, this will give slightly worse performance. However, if the output is connected
in that way, it is important that pin 1 is not connected at the unbalanced input end as this will short circuit one leg
of the floating output driver.

6.2. Interconnect screens - the pin 1 conditions

`Screen' connects directly to the chassis for the digital XLR and BNC connectors.
The RCA digital coaxial connections are transformer isolated with the screen coupled to
the chassis using a capacitor at the connector.

The analogue output screens are the mid points of the floating differential outputs. They
are coupled to the chassis using a capacitor from pin 1 to the chassis.

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6.3. Connector table

Viewed from the rear the connectors are as follows:

Left Hand Side IEC320 Mains power inlet

XLR-Male Analogue Analogue out left/A
XLR-Male Outputs Analogue out right/B
XLR-Female DI1 - AES3
XLR-Female DI2 - AES3
XLR-Female DI3 - AES3
OPTICAL Digital DI4 - IEC958 optical
RCA-phono Inputs DI5 - IEC958 coaxial
BNC-75S DI6 - AES-3ID or

L - SDIF-2 input left/A

BNC-75S DI7 - AES-31D or

R - SDIF-2 input right/B
BNC-75S WCK - Word-clock input
BNC-75S L - SDIF-2 output left/A
BNC-75S R - SDIF-2 output right/B
BNC-75S Digital WCK - Word-clock output
RCA-phono Outputs IEC958 consumer format
OPTICAL IEC958 consumer format
XLR-Male DO1 - AES3 output 1

Right Hand Side XLR-Male DO2 - AES3 output 2

XLR wiring conventions for both analogue and digital connections are:

pin 1 Screen
pin 2 Balanced input or output (Hot or `+')
pin 3 Balanced input or output (Cold or `-')

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7. SPECIFICATION

7.1. Digital Inputs

7.1.1. General (except SDIF-2)
Jitter Tolerance

Above 12kHz: 0.5UI (This is 88.6ns at Fs = 44.1kHz)
50Hz to 12kHz: Increases with falling jitter frequency.

eg: 0.75 UI at 8kHz

30 UI at 200Hz

Below 50Hz: > 128UI (22.7 us at Fs = 44.1kHz)

Note: Jitter tolerance is a measure of the ability of a receiver to correctly decode a digital interface signal which
has jitter. Like most other jitter parameters it is normally measured in unit intervals. A unit interval (UI)
corresponds to the smallest nominal time interval in the interface signal, for the AES3 or IEC958 interface formats
there are 128 unit intervals per sample. Therefore at a sample frequency of 44.1kHz the unit interval is 177 ns.
References 5 and 6 explain interface jitter in more detail.

Digital Input Resolution/Word-Length: 24 bits
The Dream resolves all the data bits of the digital interface formats. Input signals

are not truncated and hence truncation distortion does not occur. For example: a sine
wave only modulating the lowest 4 bits of the 24 bit interface signal is reproduced at the
correct amplitude.

Note that, in conformance with the digital audio interface specifications, all signals with consumer channel status,
and signals with professional channel status that indicate that the auxiliary audio data bits do not carry the main
audio signal will have the lower 4 bits masked off to avoid non-audio data adding to the audio. The SDIF-2 format
is also limited to a maximum resolution of 20 bits for the audio data.

7.1.2. Balanced AES-3 XLR inputs: DI1, DI2, DI3
Input impedance: 110S.
These conform with the professional interface, AES3-1992 [Ref. 1], with transformers for

improved isolation. However if the data is encoded according to the consumer format of
IEC958 [Ref. 4] it is also decoded correctly.

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Pin 1 of the XLR connector is connected to the chassis via a capacitor and the signal is
presented differentially across pins 2 and 3.

7.1.3. Optical input: DI4
This input is physically compatible with the Toshiba `TosLink' fibre-optic connector, and

data encoded according to professional (AES3-1992) or consumer (IEC958) formats is
correctly decoded.

7.1.4. Coaxial (RCA) consumer input: DI5
Input impedance: 75S.
This conforms to the coaxial specification of IEC958 and, for improved isolation, has

transformer isolation with the screen capacitatively coupled to the chassis. Data encoded
according to professional (AES3-1992) or consumer (IEC958) formats is correctly
decoded.

7.1.5. Dual-function coaxial (BNC) professional inputs: DI6, DI7
Input impedance: 75S.
The connectors either operate as coaxial AES3 (AES-3ID) or SDIF-2 inputs. An auto-

detection algorithm decides which format to use as follows.
If the unit is in clock master mode (Master light on):

If a valid SDIF signal is detected on the selected input then both inputs will be
treated as SDIF-2 format.
If a valid AES or IEC958 signal is detected on the selected input then that input will
be decoded as that format.

Otherwise, if the unit is not in clock master mode (Master light off):

If a valid word clock signal is present on the WCK input then both inputs will be
treated as SDIF-2 format. Without a valid WCK then any signal on the selected
input will be treated as AES3 or IEC958 format.

7.1.5.1. DI6, DI7 operating as AES3 or IEC958 format inputs.

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These conform to the AES-3ID recommendation [Ref. 28]. However data encoded
according to professional (AES3-1992) or consumer (IEC958) formats is correctly
decoded.

7.1.5.2. DI6, DI7 operating as SDIF-2 (Sony) format inputs
The input selection indicators for DI6 and DI7 both illuminate if the inputs are being handled

in this format. (If one of the inputs is not present then that indicator will flash.)
In this mode the inputs conform with the interfaces initially used by the Sony

PCM-1610/1630 PCM encoder. The signal on DI6 is the left channel and on DI7 is the
right channel.

Note: Signals in this format are D.C. coupled at TTL levels and transmitted most significant bit (MSB) first in non­return to zero (NRZ) format. A synchronization pattern identifies the start of a new sample word. Some
implementations of this interface do not use the synchronization pattern to identify the sample word boundary.
Such equipment will often require delays to be carefully matched before they will work correctly. The Dream
uses a rugged algorithm to correctly decode SDIF data independent of delays or phase.

7.1.6. Coaxial word-clock (WCK) input
This input can be used to lock the DA-1 to a sample rate square wave, or word-clock. It

is used in conjunction with the SDIF-2 inputs to synchronize the Dream internal clock
to the SDIF-2 source equipment.

Input impedance: 75S
Input level: 200mV to 10V pk-pk
Mark-space ratio: 40:60 to 60:40
Sync reference point: Rising edge

This input is ac coupled so that in addition to operation with the TTL levels typical of SDIF­2 word clock sources it will also work with lower signal levels, such as from a double or
triple terminated source, with an optimum noise margin.

Note: When using WCK synchronization the digital outputs of the Dream are timed so that the start of the
sample frame is coincident with the timing reference point (the rising edge of WCK). Specified delay
measurements are also made from the timing reference point. This input is not required if the Dream is
configured as clock master.

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7.2. Synchronization

The Dream has two methods of synchronization. In almost all applications the
device will operate as a clock slave so that it derives it's internal clocks from the incoming
signal. It can also be operated as a clock master. In that case care must be taken to
ensure that the digital source supplying the input signals is `locked', or synchronized, to the
DA-1.

7.2.1. Clock slave synchronization
The Dream will normally be used synchronized with an external clock source. In this

slave configuration the internal clocks will be derived from the input signal (or, in the case
of SDIF-2, from the WCK input). The Dream can operate as a slave if the incoming
sample frequency is within one of the capture ranges of the high precision clock. If the
wide range option is fitted then the unit can also slave, without the same jitter attenuation
performance, over the full 30kHz to 50kHz range if the external clock is not within one of the
high precision ranges.

High precision clock capture range: ±0.15%

Nominal Minimum Range
32 kHz 31.96 to 32.04 kHz * (see note)

44.1 kHz 44.03 to 44.17 kHz
48 kHz 47.93 to 48.07 kHz

Normal clock capture range: 40kHz ±25%

`Other' 30.0 to 50.0 kHz * (see note)
* Note: The 32 kHz and 30 to 50 kHz modes are only available with the wide-range option fitted.
Lock-up time:

Signal present on output 0.1 s

High precision PLL locked 2 s
Digital input to digital output timing difference: < 1% of sample period

Note: This is referred to the first transition of an AES3 or IEC958 frame or the rising edge of word clock.

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7.2.2. Clock master synchronization
Clock accuracy: ±10 ppm
Reference to input timing variation, from ideal, before sample slip:

±75% of sample period

Sample slip hysteresis: 50% of sample period

Note: The AES11-1991 synchronization standard [2] requires an input synchronization window of ±25% of a
sample period without hysteresis. The Dream exceeds this specification. This is to allow it to work properly
with equipment that does not conform with the AES11 standard. This hysteresis avoids the possibility of repeated
sample slips which can occur as a result of small timing variations about some static offset.

7.3. Digital Outputs

The digital outputs provide the same audio data as present on the selected digital input,
except when DRE decode is active. When the unit is operating as a clock master, or when
an implicit format conversion is taking place, such as from the professional (AES3) format
to the consumer (SPDIF), or from the SDIF-2 format to AES3, then the channel status data
is not copied from the input and is regenerated by translating from the input format to the
output format. This means that the various output connectors always have the appropriate
data format for the connector style.

NOTE: The input format is determined by the data content at the input, and not the connector. Therefore when
connecting an AES3 data format, but consumer (SPDIF) electrical format signal to the RCA coaxial input then the
signal channel status format will be translated to consumer format at the RCA coaxial output and remain in
professional format at the XLR output. This means that the Dream is also a digital audio format convertor
and corrector.

7.3.1. Jitter characteristics common to all digital outputs
Jitter peaking: less than 1dB in both ranges
High precision clock:

Jitter attenuation slope: 80dB per decade

Corner frequency: approx 80Hz

Attenuation above 500Hz: > 60dB

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Wide range clock:

Jitter attenuation slope: 40dB per decade

Corner frequency: approx 2 kHz

Attenuation above 8kHz: > 24dB
Intrinsic jitter: < 0.005 UI pk-pk (< 1ns at Fs = 44.1kHz)

Note: Intrinsic jitter is measured using a low frequency cut-off of 200Hz, see references 5 and 6 for a more
detailed explanation of these interface jitter parameters.

7.3.2. Audio data on digital outputs
Unless the DRE decoder function is operating all the audio data on the input is passed

through to the digital outputs without modification. However the SDIF-2 output is limited
to a maximum of 20 bits.

If the DRE decoder function is on then the digital output word length is 18 bits.

Note: To avoid truncation distortion care must always be taken to match the signal word length to the input word
length of following equipment. If the DA-1 is passing through 24-bit data then truncation distortion will occur in
the signal at the SDIF-2 outputs. Often equipment is limited to a 16-bit input resolution and further truncation will
occur at that equipment if the signal is not re-dithered. This can be done by setting the output word-length of the
source device appropriately or by using a proprietary re-dither or noise-shaping device. (For example, the Prism
Sound SNS-4 dedicated noise shaper, or the Prism Sound Dream AD-1. The latter is an analogue to digital
converter which can also operate as a re-ditherer or noise-shaper when fed with a digital source.)

7.3.3. Balanced AES3 (AES/EBU) professional format: DO1 and DO2
These conform with the professional interface, AES3-1992, with transformers for improved

isolation.
Pin 1 of the XLR connector is connected to the chassis and the signal is output

differentially across pins 2 and 3.
Output impedance: 110 S ± 20%

Output level: 4 V ± 10% (terminated)
Rise/fall time: 20 ns ± 20% (terminated)

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If the selected input carries professional format channel status, and the unit is not set to
clock master mode then that status is passed through. Otherwise a professional channel
status pattern is generated, carrying the emphasis and word-length information transcoded
from the input.

Valid and user-data bits are passed through from the selected digital input (not present in
SDIF-2). Reception errors are indicated by setting the validity flag.

7.3.4. Optical consumer format
This output is physically compatible with the Toshiba `TosLink' fibre-optic connector.
If the selected input carries consumer format channel status, and the unit is not set to clock

master mode then that status is passed through. Otherwise a consumer channel status
pattern is generated, carrying the emphasis information transcoded from the input.

Note: In IEC958 there is no category code for DAC so with a non-consumer status input the category code
transmitted in the consumer channel status output is set to `ADC'.

Valid and user-data bits are passed through from the selected digital input (not present in
SDIF-2). Reception errors are indicated by setting the validity flag.

7.3.5. Coaxial consumer format (IEC958)
This conforms with the unbalanced interface defined in IEC958 clause 5.3 and carries

consumer format channels status information as defined in IEC958 clause 4.4.2. This
output is floating with the screen capacitatively coupled to the chassis.

Output impedance: 75 S ± 10%
Output level: 0.5 V ± 10% (terminated)
Rise/fall time: 22 ns ± 20% (terminated)

The data carried on this output is identical to the data on the optical consumer format
output (see above).

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7.3.6. Coaxial SDIF-2 (Sony) format, L and R.
These outputs conform with the interfaces initially used by the Sony PCM-1610/1630 PC M

encoder and often called SDIF-2. Signals in this format are at TTL levels and transmitted
most significant bit (MSB) first in non-return to zero (NRZ) format. A synchronization
pattern identifies the start of a new sample word.
Output impedance: approximately 30 S
Load impedance: 75 S
Output level: 2.8 V ± 0.5 V (when terminated in 75 S)
Rise/fall time: 60 ns

The embedded flags carried on the SDIF-2 outputs are set as follows:
Emphasis: Set if CD (15/50µs) emphasis is indicated on the selected input.

Note: J17 emphasis cannot be indicated on the SDIF-2 outputs. This bit is toggled on and off in the unlikely event
that J17 emphasis is carried in the signal.

Dub prohibit: Always clear.

7.3.7. Coaxial SDIF-2 (Sony) word-clock format
This output carries a TTL level square wave at the sample rate. The rising edge coincides

with the start of a sample frame on the digital outputs.
Electrically it is identical to the SDIF-2 L and R outputs.
This output is normally used to synchronize equipment that is using the SDIF-2 data

outputs.

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7.4. Analogue Outputs

The analogue outputs are on a three pin XLR connector with positive and negative signal
polarities on pins 2 and 3 respectively, and the mid-point on pin 1. These pins float
together with respect to the chassis earth. This high common mode impedance means
that the pin 1 screen connection to following equipment will eliminate any common mode
component on the output without introducing significant cable screen currents.

Differential output impedance: 47 S ± 1 S (pin 2 to pin 3)
Single-ended output impedance: 24 S ± 1 S (pin 1 to 2 or 3)
Coupling impedance to chassis: 4.7 kS (chassis to pin 1,2 or 3)
Common-mode range: 70 V

Note: Units manufactured before revision C had a different output configuration (see page 3). This differs in that
pin 1 is connected to the chassis and not to the mid-point of the balanced output. The mid-point of the balanced
output was not accessible in the earlier units.

7.5. Performance Specification

Specifications quoted in this section are to AES17-1991 (ANSI S4.51-1991) [Ref. 3], at

44.1kHz with output level set to +22dBu (for digital full scale input) except where stated.
Frequency response:

5 Hz - 22 kHz ± 0.05 dB Fs = 48 kHz

5 Hz - 20 kHz ± 0.05 dB Fs = 44.1 kHz

5 Hz - 15 kHz ± 0.05 dB Fs = 32 kHz
Group delay: 51.5 samples

1.07 ms (Fs = 48 kHz)

1.17 ms (Fs = 44.1 kHz)

1.61 ms (Fs = 32 kHz)

Phase linearity error < 0.8E within the following frequency ranges:

5 Hz - 20 kHz (Fs = 48 kHz)

5 Hz - 18 kHz (Fs = 44.1 kHz) (1.6E over 2.5 Hz - 20 kHz)

5 Hz - 15 kHz (Fs = 32 kHz)

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Inter-channel phase deviation

without de-emphasis < 0.5E

with de-emphasis < 0.5E
Absolute phase: Correct unless invert function on.

Note: Unless invert is on negative digital inputs produce a negative analogue output. This makes the pin 2 of
the output connector negative with respect to pin 3. This is in accordance with the standards IEC268­12(1969) and AES14.

Crosstalk: -100 dB (20 Hz to 2O kHz)
Sampling jitter rejection:

High precision clock (used if clock within 0.15% of standard frequencies).

Jitter attenuation slope: 80dB per decade

Corner frequency: approx 80Hz

Attenuation above 500Hz: > 60dB
Wide range clock (used if clock outside normal frequency tolerances):

Jitter attenuation slope: 40dB per decade

Corner frequency: approx 800 Hz

Attenuation above 8kHz: > 30dB
Maximum output signal level:

Gain is adjustable so that the output level for 0dB FS (full scale) input can be

preset from +5 dBu to +28 dBu in 1dB steps with fine trim in steps of 0.05dB. At

sample rates below 44kHz the maximum level is +25dBu.

This is made up from four analogue gain settings (corresponding to +12 dBu, +18 dBu, +22 dBu, +28 dBu)
and a variable attenuation in the digital domain. As the digital attenuator reduces the maximum level without
lowering the noise floor it has the effect of lowering the dynamic range. For optimum dynamic range the
digital attenuation should be minimised. The digital attenuation is inactive at level settings of +12 dBu, +18
dBu, +22 dBu and +28 dBu. (For samples rates below 44kHz the four analogue gain settings are 3dB lower
so the digital attenuation is inactive at settings of +9dBu, +15dBu, +19dBu and +25dBu. However the final
gain is shown correctly unless the gain is set to more than the maximum of +25dBu)

Gain error: < 0.05 dB for all settings
Gain trim step size: 0.05 dB

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Level-dependent logarithmic gain linearity: << 3dB at -144dB FS

Note: This is a measurement of the output level of a 996.115Hz tone with respect to the input level. The output
level at 997Hz is measured using 32 averages of an 8192 point FFT with a rectangular window. This method
has much higher resolution than the test in AES17, which specifies a third-octave band-pass filter to eliminate
noise. Even so, the signal level measurements are still dominated by the noise floor at -144dB FS.

Harmonic distortion and noise <-104dB (997 Hz at -1dB FS)
Intermodulation distortion <-90dB

Note: The intermodulation test from AES17 uses 18kHz and 20kHz signals at -6.03dB FS each. The result is
the ratio of the total output rms signal level to the rms sum of the 2nd and 3rd order modulation products at
2kHz and 16kHz.

Dynamic range or signal to noise ratio:

112dB CCIR-RMS
115dB (A weighted RMS)
112dB (unweighted RMS)

Note: Measured at output settings of +18,+22 and +28dBu at a level of -60 dB FS.

Idle channel noise: -112dB FS CCIR-RMS

-115dB FS (A weighted RMS)

-112dB FS (unweighted RMS)

Note: Measured at output settings of +18,+22 and +28dBu

Spurious levels: -105dB FS full scale signals

-130dB FS signals below -20dB FS

-144dB FS signals below -60dB FS

Note: Highest level of any spurious frequency component

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7.6. Power

7.6.1. Mains voltage: Internally set for 90-120V (nominally 110V) or 195-

250V (nominally 230V) operation. The supply
voltage is indicated on the rear panel.

7.6.2. Consumption: 30W

7.7. Physical Dimensions

Weight: 8.8 lb (4 Kg)
Width: 19 inch (483mm) (rack-mountable)
Height: 1U (44mm)
Depth: 10.25 inches (260mm)

8. INTERNAL CONFIGURATION OPTIONS AND FUSES

Other than the external mains fuse, the following parts can only be accessed by
removing the equipment top cover. This should only be undertaken by qualified
personnel. There are no user serviceable parts inside this unit.

8.1. Fuses

There is one mains fuse in the IEC320 mains inlet. If this fuse is blown it should be
replaced by a similar value and type. (20x5mm 250V 2AT anti-surge)

There are seven fuses internal to the unit. If these fuses have blown a fault has occurred
and the unit should be returned to the manufacturer or agent for service. The fuse types
and application are:

F1 - Digital processing, input and output
Type: 2AT Wickman TR5 subminiature type no. 19372K/2A

F5,F6,F7,F9 - Analogue output channel A
F2,F3,F4,F8 - Analogue output channel B
Type: 0.5AT Wickman TR5 subminiature type no. 19372K/500mA

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8.2. PCB jumper links

Link Purpose Factory setting

Main board (PREV051/1)
LK1 (Option select - not used) No links fitted
LK2 Do not fit Not fitted
LK3 Do not fit Not fitted
CN4 Future compatibility Link pins 15-16, 17-18

Sub Board (PREV051/2)
LK1 (Option select - not used) Not fitted

8.3. Mains transformer voltage selection

The mains transformer has a tapped primary to allow operation at nominal voltages of
230V or 110V. The brown lead is connected to the lower terminal on the mains inlet.
The upper terminal is connected as follows:

110V Connect red primary lead to the upper terminal of mains inlet.
230V Connect orange primary lead to the upper terminal of mains inlet.

The unused primary lead terminal should be firmly secured to the other primary leads.
If the mains operating voltage has changed then the rear panel label should be changed

to reflect this.

9. ELECTROMAGNETIC COMPATIBILITY

This equipment is intended for use in an electromagnetically controlled environment. To
maintain the performance specification it should not be subject to strong magnetic
fields (such as in the immediate vicinity of a power amplifier or cathode ray tube) and all
connections should be terminated as described below. This is also required to ensure
that emissions are within applicable norms and that it does not interfere with other
equipment.

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All coaxial connections should be made using a properly screened 75S cable with the
screen connected to the outer of the connector at both ends. All XLR connections
should use a screened twisted pair cable with the screen connected to pin 1 of the XLR
connector at both ends. In the case of the digital XLR connections this cable should be
of 110S impedance.

10. REFERENCES

[1] AES3-1992 - ` Recommended Practice for Digital Audio Engineering - Serial Transmission Format for Two-Channel

[2] AES11-1991 - `AES Recommended Practice for Digital Audio Engineering - Synchronization of Digital Audio

[3] AES17-1991 - `AES Standard Method for Digital Audio Engineering - Measurement of Digital Audio Equipment' J.

[4] IEC958 - `Digital Audio Interface' International Electrotechnical Commission 1989

[5] Julian Dunn, Barry McKibben, Roger Taylor and Chris Travis - `Towards Common Specifications for Digital Audio

[6] Julian Dunn and Ian Dennis - `The Diagnosis and Solution of Jitter-Related Problems in Digital Audio Systems'

[7] AES-3id-1995 - `AES information document for digital audio engineering - Transmission of AES3 formatted data by

Linearly Represented Digital Audio Data' J. Audio Eng. Soc., Vol 40 No. 3, pp 147-165 (June 1992)

Equipment in Studio Operations' J. Audio Eng. Soc., Vol. 39 No. 3, pp 155-162 (March 1991)

Audio Eng. Soc., Vol. 39 No. 12, pp 961-975 (December 1991)

Interface Jitter' Preprint 3705, presented at the 95th AES Convention, New York, October 1993

Preprint 3868, presented at the 96th AES Convention, Amsterdam, February 1994.

unbalanced coaxial cable' J. Audio Eng. Soc., Vol. 43 No. 10, pp 827-844 (October 1995)

FURTHER INFORMATION

Enquiries about this product should be addressed to :

Prism Media Products Limited

William James House

Cowley Road

Cambridge

CB4 4WX

UK

Phone: [phone]

Fax: [phone]

Email: [e-mail]

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