ASI5111 - AudioScience
January 06
ASI5111
LINEAR PCI AUDIO ADAPTER
DESCRIPTION
FEATURES
The ASI5111 is a professional PCI audio adapter designed for use in
radio broadcast production.
•
Four stereo streams of PCM playback
•
Two stereo streams of PCM record.
•
Balanced stereo analog input and output
•
AES/EBU or S/PDIF digital input and output (software switchable).
•
Low noise microphone input with 48V phantom supply and DSP
based compressor/limiter and 3 band equaliser.
•
24bit analog-to-digital and digital-to-analog converters - 100dB
SNR and 0.0025% THD+N.
•
11 to 96kHz sample rates.
•
MRX™ multi rate mixing technology supports digital mixing of
multiple sample rates.
•
SoundGuard™ transient voltage suppression protects against
lightning and other high voltage surges on all I/O
•
Up to 8 cards in one system.
•
Windows 98/NT/2000/XP and Linux software drivers available.
The adapter offers two stereo record stream from either a balanced
analog input or AES/EBU digital input and four stereo play streams
mixed to both a balanced analog output and an AES/EBU digital output.
Also included is a microphone input, with low noise pre-amp and a 48V
phantom supply.
ASI5111 Connectors
ASI5111
¼” Stereo Jack
R ecord Stream 2
PCM
Peak Meter
M
SRC
U
X
R ecord Stream 1
PCM
SRC
Sam ple R ate
C onverter
M
U
X
AES3
Rx
AES/EBU
input
48V
EQ
CLE
M
U
X
Mic
am p
24 bit
A/D
Microphone
input
Stereo balanced
input
Level 0 to 20dBu
6-input
Mixer
DB-9 Analog
(sam e as 4113)
DB-9 Digital
(sam e as 4113)
Play Stream 1
PCM
SRC
Play Stream 2
PCM
SRC
Play Stream 3
PCM
SRC
Play Stream 4
PCM
SRC
Level 0 to 20dBu
+
24 bit
D/A
AES3
Tx
Stereo balanced
output
AES/EBU
output
PCI Bus
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ASI5111
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SPECIFICATIONS
BALANCED INPUT/OUTPUT
Connector
Input Level
Input Impedance
Output Level
Load Impedance
S/N Ratio[1]
THD+N[2]
Sample Precision
Frequency Response
MICROPHONE INPUT
Connector
Input Gain
Input Impedance
Phantom Power
S/N Ratio[1]
THD+N[2]
Frequency Response
DIGITAL INPUT/OUTPUT
Type
DB-9 Female
-10 to +20dBu in 1dBu steps
20K ohms
-10 to +20dBu in 1dBu steps
600ohms or greater
> 100dB (record or play)
< 0.0025% (record or play)
24bit Oversampling
20Hz to 20kHz +/-0.25dB
20Hz to 40kHz +0.25/-5dB[3]
¼” TRS jack
20, 40 and 60dB software adjustable
11K ohms (+ or – to ground)
48V +/- 4V, software switchable on and off.
90dB @ 40dB gain
0.005% @ 40dB gain
20Hz to 20kHz +/-0.5dB
20Hz to 40kHz +0.5/-5dB [3]
Connector
Sample Rates
Sample Precision
AES/EBU (EIAJ CP-340 TypeI / IEC-958 Professional)
S/PDIF (EIAJ CP-340 TypeII / IEC-958 Consumer) (software selectable)
DB-9 Male
32, 44.1, 48, 64, 88.2 and 96kHz
24bit
SAMPLE RATE CLOCK
Internal
AES/EBU In
32, 44.1, 48, 64, 88.2 and 96kHz
32, 44.1, 48, 64, 88.2 and 96kHz
SIGNAL PROCESSING
DSP
Memory
Audio Formats
Texas Instruments TMS320C6711@135MHz
8MB
8 bit unsigned PCM
16bit signed PCM
32bit IEEE floating point PCM
BREAKOUT CABLES (INCLUDED)
Analog
Digital
CBL1001 : DB-9 to 2 in and 2 out XLR
CBL1003: DB-9 to 1 in and 1 out XLR
GENERAL
Bus
Dimensions
Weight
Operating Temperature
Power Requirements
Universal 32bit PCI (3.3V or 5V signaling)
PCI form factor – 6.75" x 3.9" x 0.6" (172mm x 100mm x 15mm)
8 oz (227g) max
0C to 70C
+5V @ 600mA, +12V @ 150mA, -12V @ 70mA
[1] - S/N Ratio is the difference between a 1kHz digital fullscale sinewave and digital zero using an A weighting filter
[2] - THD+N measured using a +20dBu 1kHz sinewave sampled at 48kHz and A weighting filter
[3] – Using a 96kHz sampling rate
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ASI5111
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CONNECTORS
Analog DB9 (Female)
Digital DB9 (Male)
Microphone input jack – Tip=+, Ring=-, Shield=GND.
3
MIXER MAPS
3.1 HPI Mixer
The mixer layout for the ASI5111 as represented by the HPI is as follows. For details on each HPI control type, see
the HPI specification (SPCHPI.PDF).
* SRC = Sample Rate Converter, not visible as an HPI object
INSTREAM [0] SRC* METER CH MODE MUX
EQUALIZER
COMPANDER
METER
LEVEL LINEIN [0]
MUX
VOLUME
INSTREAM [1] SRC*
MICROPHONE MICROPHONE [0]
AESEBU_RECEIVER AESEBU_IN [0]
METER CH MODE MUX
VOLUME
VOLUME
VOLUME +
METER
LEVEL
LINEOUT [0]
VOLUME
VOLUME
AESEBU_TRANSMITTER AESEBU_OUT [0]
VOLUME
OUTSTREAM [0] SRC*
METER
VOLUME
CH MODE
OUTSTREAM [1] SRC*
METER
VOLUME
CH MODE
OUTSTREAM [2] SRC*
METER
VOLUME
CH MODE
OUTSTREAM [3] SRC*
METER
VOLUME
CH MODE
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SAMPLE_CLOCK CLOCK_SOURCE
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ASI5111
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AUDIO FORMATS
The ASI5111 supports record and play of the following formats:
Format
8 bit unsigned PCM
16 bit signed PCM
32 bit signed PCM
32 bit floating point PCM (+/-1.0)
5
HPI format
HPI_FORMAT_PCM8_UNSIGNED
HPI_FORMAT_PCM16_SIGNED
HPI_FORMAT_PCM32_SIGNED
HPI_FORMAT_PCM32_FLOAT
Windows format
WAVE_FORMAT_PCM, wBitsPerSample=8
WAVE_FORMAT_PCM, wBitsPerSample=16
WAVE_FORMAT_PCM, wBitsPerSample=32
WAVE_FORMAT_IEEE_FLOAT
MICROPHONE INPUT
The ASI5111 has a balanced microphone input using a ¼” stereo jack.
5.1 Phantom Power
When phantom power is enabled, +48V is present on both the + and – signal inputs (tip and ring of ¼” jack). This is
used to drive professional condenser type microphones. If you are using a dynamic microphone, make sure that the
phantom power is off as it may damage the mic.
User
Phantom power is turned on and off using the following control in the ASI Mixer on the Microphone panel:
Developer
Windows – Phantom power is controlled using….
HPI – Phantom power is controlled using the HPI_Microphone_SetPhantomPower() API
5.2 Programmable Gain
The microphone preamp has a software programmable gain of +20, +40 or +60dB.
User
Microphone gain is adjusted using the following control in the ASI Mixer:
Developer
Windows – Microphone gain is controlled using….
HPI – Microphone is controlled using a Volume control on the MICROPHONE source node. Use
HPI_VolumeSetGain() API.
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ASI5111
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BALANCED ANALOG I/O
The ASI5111 has a stereo balanced analog input and output on a DB-9 female connector.
6.1 Analog I/O Level
The analog Level (or Trim) is software programmable independantly for the input and output. It can be set from –10
to +20dBu in 1dB increments.
User
Analog levels are adjusted using the Trim/Level controls located on the LineIn and LineOut panels in the ASI
Mixer:
Developer
Windows – Analog levels are controlled using….
HPI – Analog levels controlled using the HPI_LevelSet() API
7
AES/EBU I/O
The ASI5111 has an AES/EBU digital audio input and output on a DB-9 male connector. This maybe also operated as
S/PDIF. The AES/EBU I/O operates at either 32, 44.1, 48, 64, 88.2 or 96kHz. The bitstream contains samples of
24bit precision. When a valid AES/EBU source is connected to the ASI5111, the card will automatically generate the
sample clock from that source (see Sample Clock section)
7.1 Operating as S/PDIF
The AES/EBU I/O can be operated as S/PDIF (IEC958). When this happens, the impedance of the I/O changes to
75ohms and the signal level becomes ~0.5Vpp. As well as programming the correct settings in the card, the
AES/EBU signals must be connected as follows. For S/PDIF output, connect the "-" side of the AES signal to the
S/PDIF shield. The "+" side becomes the S/PDIF signal.
output
shielded cable
RCA jack
AESO+
AESO-
2
For S/PDIF input, connect the "-" side of the AES signal to the shield and ground. The "+" side becomes the signal.
Input
shielded cable
RCA jack
AESI+
AESIGND
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ASI5111
User
Use the following controls in ASI Mixer to toggle between AES/EBU and S/PDIF
Developer
Windows – Use Digital I/O controls – see the “AudioScience WavX Specification” (SPCWAVX.PDF)
HPI – Use the HPI_AESEBU_Receiver_SetSource() and HPI_AESEBU_Transmitter_SetFormat() API
7.2 Channel Status and User Data
The ASI Mixer does not setup the Channel Status and User Data in the AES/EBU output. This must be done by the
application using the following APIs:
Windows – Use Digital I/O controls – see the “AudioScience WavX Specification” (SPCWAVX.PDF)
HPI – Use HPI_AESEBU_Transmitter_SetChannelStatus() and HPI_AESEBU_Transmitter_SetUserData()
APIs
Your application can also read the Channel Status and User Data of the AES/EBU input using the following APIs:
Windows – Use Digital I/O controls – see the “AudioScience WavX Specification” (SPCWAVX.PDF)
HPI – Use HPI_AESEBU_Receiver_GetChannelStatus() and HPI_AESEBU_Receiver_GetUserData() APIs
8
COMPANDER
The ASI5111 contains a compressor/expander (Compander), which is used to reduce or expand the dynamic range of
the signal it acts on. It is located on the LineIn input and maybe used on both the Line In and Microphone signals.
User
The ASI5111’s Compander is accessed from the ASI Mixer by clicking on the “Compander” button on the
LineIn panel. The following parameters can be set:
Compression Threshold – the input signal level at which the compression starts.
Compression Ratio – The ratio of the input signal level to the output signal level
Makeup Gain – additional gain applied the the compressed/expanded signal
Attack - Attack time of compander in milliseconds. Sets the time that the compressor takes to act.
Decay - Decay time of compander in milliseconds. Sets the time for the signal gain to return to normal after
compression.
Developer
Windows – Use the Compandor control – see the “AudioScience WavX Specification” (SPCWAVX.PDF)
HPI – Use the HPI_Compandor_XXXX APIs - see the “AudioScience HPI Specification” (SPCHPI.PDF)
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ASI5111
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PARAMETRIC EQUALIZER
The ASI5111 contains a 5 band parametric equalizer. It is located on the LineIn input and maybe used on both the
Line In and Microphone signals. Each of the equalizers 5 bands maybe be individualy programmed with filter type
(eq, low-pass, high-shelf etc), Q (sharpness) and center frequency.
User
The ASI5111’s Parametric Equalizer is accessed from the ASI Mixer by clicking on the “EQ” button on the
LineIn panel. The EQ window contains controls for setting the filter parameters of each of the 5 bands, with a
graph showing the combined frequency response of the 5 bands.
Each filter band has the following parameters:
Filter Type – The shape of the filter. Can be Eq (default), Lowpass, Highpass, Bandpass, Lowshelf,
Highshelf.
Filter Freq – The center frequency of the filter.
Filter Q – The sharpness of the filter. The higher the Q, the more selective the filter is.
Filter Gain – The gain of the filter at the center frequency.
Developer
Windows – Use the equalizer mixer control – see the “AudioScience WavX Specification” (SPCWAVX.PDF)
HPI – Use the HPI_ParametricEQ_XXXX APIs – see the “AudioScience HPI Specification” (SPCHPI.PDF)
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ASI5111
10 SAMPLE RATE CLOCK and MRX MIXER
The ASI5111 sample rate clock is used to drive the MRX digital mixer, Analog to Digital Converter (ADC), Digital to
Analog Converter (DAC) and AES/EBU output. There are two sources of sample rate clock – internal and the
AES/EBU input.
The internal adapter clock is generated from a low jitter frequency synthesizer and may be set to 32, 44.1, 48, 64,
88.2 and 96kHz. When a valid AES/EBU bitstream is connected to the AES/EBU input, the ASI5111 will
automatically switch to using this as the sample rate clock. This is needed so that digital audio from the AES/EBU
input can be synchronised with the other audio streams present in the mixer. There is no way to overide this.
Note that the sample rate clock does not determine the sample rates of the audio streams that may be played and
recorded. These are independantly set using the MRX multi rate mixer, so that, for instance, you can have the
adapter running at 96kHz, but be playing files of 44.1 and 48kHz and recording files of 32 and 88.2khz.
User
Use the following controls in ASI Mixer to select the internal adapter rate. Note the SampleClk source control
is not user selectable as the adapter automatically switches depending whether a valid AES/EBU input is
present.
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ASI5111
Developer
Windows –
HPI – Use the HPI_SampleClock_XXXX APIs.
11 CABLES
The ASI5111 comes with XLR breakout cables for both the analog and digital connectors.
12 REFERENCES
Specifications
SPCWAVX.PDF - WavX - AudioScience Windows Multimedia Extensions
SPCHPI.PDF - Hardware Programming Interface (HPI) Specification
All these documents are available from www.audioscience.com in the Technical Info section
[end]
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