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 DUETTM DIGITAL Advanced DSP System with FRONTWAVE(R)
GB3212 PRELIMINARY DATA SHEET FEATURES * 4-channel WDRC compression * 16-band frequency shaping * 16-band adaptive noise reduction * adaptive feedback cancellation * FRONTWAVE(R) directional processing * high fidelity CODEC - dual A/D's;D/A * 16-bit DSP core processor * 95dB input dynamic range with HRXTM Headroom Extension * drives zero-bias 2-terminal receivers * thinSTAXTM packaging * 4 fully configurable memories with audible memory change indicator * 2 memory select pads * internal/external volume control * AGCo with variable threshold and time constants thinSTAXTM PACKAGING Hybrid typical dimensions: 0.217 x 0.129 x 0.087in. (5.51 x 3.28 x 2.21mm) DESCRIPTION DUETTM DIGITAL is a high end DSP system with advanced adaptive algorithms. The signal processing algorithms run on a hardware platform which is a combination of a highfidelity CODEC and a general purpose DSP core. Algorithms developed and optimized by Gennum, running on this powerful platform, offer true speech processing. The reflowable thinSTAXTM packaging enables easy integration into a wide range of applications, from CIC to BTE. As shown in the block diagram below, some of the audio DSP functions are implemented in hardware as a part of our high fidelity CODEC while other adaptive algorithms such as Noise Reduction and Feedback Cancellation use the DSP core. Pre-processing blocks include FRONTWAVE(R) directional processing and programmable filters. Postprocessing blocks include tone generation, volume control, AGCo and programmable filters. The GB3212 hybrid code programmed into the EEPROM is "40". This data sheet is part of a set of documents available for this product. Please refer to Getting Started with DUETTM DIGITAL, Document #29231 for a list of other documents.
GB3212
VB 5
VC 3
CODEC VREG 14 50n TONE GENERATOR FMIC 15 70n BIQUAD FILTERS BIQUAD FILTERS A/D VC A/D REGULATOR
10n
6
VBP
FRONTWAVE 70n RMIC T 16 A/D 2 60n MGND 1
AGC-O
BIQUAD FILTERS
D/A HBRIDGE
8 7
OUTOUT+
9
PGND
EEPROM Adaptive Feedback Cancellation band 1
Frequency Band Analysis
band 16
Frequency Band Synthesis WDRC 4 channels Frequency Shaping 16 bands Noise Reduction 16 bands
DSP
10 GND
13 MS1
11 N/A
12 N/A
17 MS2
4 SDA
BLOCK DIAGRAM
Doc.No. 20352 - 2 [Rev. July 2004]
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ABSOLUTE MAXIMUM RATINGS
PARAMETER Operating Temperature Range Storage Temperature Range Absolute Maximum Power Dissipation Maximum Operating Supply Voltage Absolute Maximum Supply Voltage VALUE/UNITS -10C to 40C -20C to 70C
PAD CONNECTION
PGND 9 GND 10 N/A 11 N/A 12 MS1 13 VREG 14
OUT-
8
25mW 1.5VDC 2VDC
6 5 4 OUT+ 7
17
MS2
15 16
FMIC
GB3212
RMIC
3
2
1
VBP
VB
SDA
VC
T
MGND
CAUTION
ELECTROSTATIC SENSITIVE DEVICES
DO NOT OPEN PACKAGES OR HANDLE EXCEPT AT A STATIC-FREE WORKSTATION
CAUTION
LEVEL 3 MOISTURE SENSITIVE DEVICES
DO NOT OPEN PACKAGES EXCEPT UNDER CONTROLLED CONDITIONS
ELECTRICAL CHARACTERISTICS
Conditions: Supply Voltage VB = 1.3V; Temperature = 25C
PARAMETER Hybrid Current Minimum Operating Supply Voltage Supply Voltage Turn On Threshold Low Frequency System Bandwidth High Frequency System Bandwidth Converter Gain Total Harmonic Distortion THD at Maximum Input Clock Frequency INPUT Input Referred Noise Input Impedance Anti-alias Filter Rejection (input referred) Maximum Input Level Input Dynamic Range A/D Dynamic Range OUTPUT Maximum RMS Output Voltage D/A Dynamic Range
SYMBOL AMP VBOFF VBON
CONDITIONS All adaptive features enabled. Ramp down Ramp up
MIN 0.94 -
TYP 1.1 0.97 1.10 125 8 29 2.048
MAX 1.0 1 3 2.15
UNITS mA V V Hz kHz dB % % MHz
ACONV THD THDM clk
A/D + D/A gain. VIN = -40 dBV VIN = -15 dBV, HRX - ON
1.945
IRN ZIN
Bandwidth 100Hz - 8kHz
-
16 80 -15 95 86
-106 -
dBV k dB dBV dB dB
=clk-8kHz,
VIN = -40dBV
-
HRX - ON, Bandwidth 100Hz - 8kHz Bandwidth 100Hz - 8kHz
-
0dBFS = 1kHz Bandwidth 100Hz - 8kHz
-
-1 83
-
dBV dB
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ELECTRICAL CHARACTERISTICS
PARAMETER
(CONTINUED)
Conditions: Supply Voltage VB = 1.3V; Temperature = 25C
RANGE MIN MAX
UNITS
FRONTWAVE Time Delay
(R)
0.1 0.05
50 8
ms kHz
Low Frequency Equalizer Corner Frequency FREQUENCY SHAPING Pre1 and Pre2 Biquad Filter PostA and PostB Biquad Filter Graphic EQ Band Gain WIDE DYNAMIC RANGE COMPRESSION Lower Threshold Upper Threshold Low Level Gain High Level Gain Compression Ratio Fast Detector Time Constant Slow Detector Time Constant AGCo AGCo Output Limiting AGCo Compression Ratio AGCo Attack Time Constant AGCo Release Time Constant WIDEBAND SYSTEM GAIN Wideband System Gain External Volume Control Internal Volume Control Attenuator
GB3212
Design Specific Design Specific -42 0
N/A N/A dB
-100 -90 -18 -18 1:1 4 4
-30 -20 42 42 100:1 8188 8188
dBFS dBFS dB dB Ratio ms ms
-40 :1 0.25 0.25
0
dBFS* Ratio
8192 8192
ms ms
-36 -48 -48
12 0 0
dB dB dB
* peak output is defined as largest sine wave possible at the resonant frequency of the receiver
SUPPORT SOFTWARE All support software for the GB3212 is available from the Gennum Web site, http://www.gennum.com/hip/software/index.html.
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VB
200k 5 3 10n
14 50n
REGULATOR
CODEC
TONE GENERATOR 15 3k9 70n BIQUAD FILTERS BIQUAD FILTERS A/D
VC A/D
GB3212
6 OUT FRONTWAVE
AGC-O
BIQUAD FILTERS
D/A HBRIDGE
8 LP FILTER 7
70n 16 3k9 1k 2 60n EEPROM Adaptive Feedback Cancellation band 1 band 16 A/D
9
1
Frequency Band Analysis
Frequency Band Synthesis WDRC 4 channels Frequency Shaping 16 bands Noise Reduction 16 bands
DSP
10
13
11
12
17
4
All resistors in ohms, all capacitors in farads unless otherwise stated.
Figure 1: Test circuit
VB
5
3 10n
14 50n
REGULATOR
CODEC
TONE GENERATOR 15 70n BIQUAD FILTERS BIQUAD FILTERS A/D
VC A/D 6
FRONTWAVE 70n 16 2 60n A/D
AGC-O
BIQUAD FILTERS
D/A HBRIDGE
8 7 Knowles or Microtronic zero-bias receiver 9
EEPROM Adaptive Feedback Cancellation band 1 band 16
1
Frequency Band Analysis
Frequency Band Synthesis WDRC 4 channels Frequency Shaping 16 bands Noise Reduction 16 bands
DSP
10
13
11
12
17
4
All resistors in ohms, all capacitors in farads unless otherwise stated.
Figure 2: Typical application circuit
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VC
MS switch (N.O.)
+ Rear Mic
GB3212
Zero Biased Receiver
+ Front Mic
INTRODUCTION The GB3212 hybrid comprises a highly versatile, advanced digital signal processing system. Configuration data stored in non-volatile memory defines hearing aid parameters. This data needs to be uploaded to the hybrid before the circuit becomes functional. The GB3212 hybrid is programmed via the SDA pin using industry-standard programming boxes. Configuration data is generated by an ARK product component library (DLL). Like Gennum's other digital products, the GB3212 is fully supported by Gennum's software tools available from the Gennum ARK web site http://ark.gennum.com. SIGNAL PATH There are two main audio input signal paths. The first path contains the Front Microphone and second path contains the Rear Microphone or Telecoil input as selected by a programmable MUX. The front microphone input is intended as the main Microphone audio input for single microphone applications. In FRONTWAVE(R) operation, a multimicrophone signal is used to produce a directional hearing instrument response. The two audio inputs are buffered, sampled and converted into digital form using dual A/D converters. The digital outputs are converted into a 32kHz, 20-bit digital audio signal. Further IIR filter blocks process the front microphone and rear microphone signals. Two biquad filters, "miccomp1" and "miccomp2", are used to match the rear microphone's gain and phase to that of the front microphone. After the miccomp filters, more filters are used to provide an adjustable group delay to create the desired polar response pattern during the calibration process. In the Telecoil mode gains are trimmed during Cal/Config process to compensate for microphone/telecoil mismatches. The FRONTWAVE(R) block is followed by two cascaded biquad filters, "pre1" and "pre2". These filters can be used for frequency response shaping before the signal goes from the CODEC chip into the DSP chip. When FRONTWAVE(R) is not enabled, the miccomp filters can be used for frequency response shaping also. After passing through the biquad filters the signal enters the DSP chip. At this point, the signal is converted to 16kHz and 16-bit. The DSP chip runs the following signal processing algorithms: * * * * * * frequency analysis 4 channel WDRC 16 band frequency shaping 16 band noise reduction frequency band synthesis adaptive feedback cancellation
GENNUM CORPORATION
+ CS44 T-coil
-
Figure 3: Assembly diagram
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Once the signal has been processed by the DSP chip it goes back into the CODEC chip. On the CODEC chip there are four more cascaded biquad filters -- "post1", "post2", "post3" and "post4". These biquad filters are followed by the tone generator, AGCo block and two more biquad filters -- "postagco1" and "postagco2". The last stage is in the signal path is the D/A H-bridge. FUNCTIONAL BLOCK DESCRIPTION
ADAPTIVE FEEDBACK CANCELLER
A/D AND D/A CONVERTERS
The system's two A/D converters are 2nd-order sigma-delta modulators, which operate at a 2.048MHz sample rate. The system's two audio inputs are pre-conditioned with antialias filtering and programmable gain pre-amplifiers. These analog outputs are over sampled and modulated to produce two, 1-bit pulse density modulated (PDM) data streams. The digital PDM data is then decimated down to pulse-code modulated (PCM) digital words at the system sampling rate of 32kHz. The D/A is comprised of a digital, 3rd-order sigma-delta modulator and an H-bridge. The modulator accepts PCM audio data from the DSP path and converts it into a 32times over-sampled, 1-bit PDM data stream, which is then supplied to the H-bridge. The H-bridge is a specialized CMOS output driver used to convert the 1-bit data stream into a low-impedance, differential output voltage waveform suitable for driving zero-biased hearing aid receivers.
HRX HEAD ROOM EXPANDER
GB3212
The adaptive feedback canceller (AFC) reduces acoustic feedback by forming an estimate of the hearing aid feedback signal and then subtracting this estimate from the hearing aid input. Therefore the forward path of the hearing is not affected. Unlike adaptive notch filter approaches, DUET's AFC does not reduce the hearing aid's gain. The AFC is based on a time-domain model of the feedback path.
Feedback path H
+ -
G
The GB3212 has an enhanced Head Room Expander (HRX) circuit, which increases the input dynamic range of the DUETTM DIGITAL without any unwanted audible artifacts. This is accomplished by dynamically adjusting the preamplifier's gain and the post-A/D attenuation depending on the input level.
FRONTWAVE(R) DIRECTIONALITY
H' Estimated feedback
Figure 4: Adaptive Feedback Canceller (AFC) block diagram ADAPTIVE NOISE REDUCTION
Noise reduction is applied independently in each of 16 frequency bands. The algorithm utilizes perceptual criteria to determine the audibility of individual noise bands. More attenuation is applied to those bands where the noise is most audible. Less attenuation is applied where the noise is inaudible. This maximizes the perceptual benefit of noise reduction and also reduces the audible artifacts that are often associated with adaptive noise reduction algorithms. The attenuation applied to a given band is determined by a combination of two factors: the SNR and the masking threshold. The SNR estimate in each band determines the maximum amount of attenuation that will be applied to that band (the poorer the SNR the greater the amount of attenuation). At the same time the masking threshold resulting from the energy in adjacent bands is also estimated. Only enough attenuation is applied to bring the energy in each 'noise' band to just below the masking threshold. This prevents excessive amounts of attenuation from being applied and thereby reduces unwanted artifacts and distortion.
The FRONTWAVE(R) block provides the resources necessary to implement directional microphone processing. The block accepts inputs from both a front and rear microphone and provides a synthesized directional microphone signal as its output. The directional microphone output is obtained by delaying the rear microphone signal and subtracting it from the front microphone signal. Various microphone response patterns can be obtained by adjusting the time delay. The FRONTWAVE(R) circuit also provides a fixed filter for compensating the sensitivity and frequency response differences between microphones. The filter parameters are adjusted during product calibration. One of the generic IIR filters following the FRONTWAVE(R) block ("pre1") has been allocated for low frequency equalization to compensate for the 6dB/octave roll off in frequency response that occurs in directional mode. The amount of low frequency equalization that is applied can be determined during product calibration. Gennum recommends using matched microphones with FRONTWAVE(R), although calibration is fully possible using unmatched microphones. Initially, calibration using unmatched microphones will result in no difference in directionality. However, over a longer period of time unmatched microphones are more likely to drift apart and result in poor directional characteristics.
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GENERIC BIQUAD FILTERS
AGCO
Frequency shaping can be achieved by configuring generic biquad filters. The transfer function for each of the biquad filters is as follows: b0 + b1 * z-1 + b2 * z-2 H(z) = _________________________ 1 + a1 * z-1 + a2 * z-2 Note that the a0 coefficient is hard-wired to always be a 1. The coefficients are each 16 bits in length and include one sign bit, one bit to the left of the decimal point, and 14 bits to the right of the decimal point. Thus, before quantization, the floating-point coefficients must be in the range -2.0 <= x < 2.0 and quantized with the function: round(x * 214) After designing a filter, the quantized coefficients can be entered into the PreBiquads or PostBiquads tab in the Interactive Data Sheet. The coefficients b0, b1, b2, a1, and a2 are as defined in the transfer function above. The parameters meta0 and meta1 do not have any effect on the signal processing, but can be used to store additional information related to the biquad with which they are associated. The underlying code in the product components automatically checks all of the filters in the system for stability (that is, the poles have to be within the unit circle) before updating the graphs on the screen or programming the coefficients into the hybrid. If the Interactive Data Sheet receives an exception from the underlying stability checking code, it will automatically disable the biquad being modified and display a warning message. When the filter is made stable again, it can be re-enabled. Note also that in some configurations some of these filters may be used by the product component for microphone/telecoil compensation, low-frequency EQ, etc. If this is the case, the coefficients the user enters into IDS will be ignored and the filter designed by the software will be programmed instead. For more information on filter design refer to Biquad Filters In PARAGONTM Digital Hybrid information note, Document # 20205.
VOLUME CONTROL
The AGCo module is an output limiting circuit whose compression ratio is fixed at infinity:1. The threshold level is programmable. The AGCo module has its own twin level detector, with programmable attack and release time constants.
MS1 AND MS2 SWITCHES
There are two, two-pole Memory Select switches available on the GB3212, which allows the user tremendous flexibility in switching between configurations. These switches may be either momentary or static as set up in the Interactive Data Sheet. Up to four memories can be configured. Enabled (valid) memories must be sequential. For example, if three memories were required, memories A, B and C would be enabled. Memory A must always be valid.
Momentary Switch on MS1
GB3212
This mode uses a single momentary switch on MS (Pin 13) to change memories. Using this mode will cause the part to start in Memory A and whenever the button is pressed the next valid memory will be loaded. When the user is in the last valid memory, a button press will cause memory A to be loaded. Examples: If 4 valid memories ABCDABCDA... If 3 valid memories ABCABCA... If 2 valid memories ABABA... If 1 valid memories AAA...
Static Switch on MS1 and MS2
The volume control (VC) can be either external or programmable. If VC is programmed for external operation, a 200k variable resistor should be connected to the 9bit A/D converter. Hysteresis is built into the Volume Control circuitry to prevent unintentional volume level toggling. A log taper potentiometer is recommended so that gain in dB will be linear with potentiometer rotation.
This mode uses two static switches to change memories. The following table describes which memory is selected depending on the state of the switches. In this mode it is possible to jump from any memory to any other memory simply by changing the state of both switches. If both switches are changed simultaneously then the transition will be smooth, otherwise, if one switch is changed and then the other, the part will transition to an intermediate memory before reaching the final memory. The part will start in whatever memory the switches are selecting. If a memory is invalid the part will not switch to the invalid memory, but stay in the current memory.
MS1 LOW LOW OPEN OPEN
MS2 LOW OPEN LOW OPEN A
Memory
B (if valid otherwise no change) C (if valid otherwise no change) D (if valid otherwise no change)
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Static Switch on MS1 Static Switch on MS2 (jump to memory D)
This mode uses two static switches to change memories. Unlike in the previous example, this mode will switch to memory D when the static switch on MS2 is OPEN. This means that this mode will only use a maximum of three memories (even if four valid memories are programmed). The following table describes which memory is selected depending on the state of the switches.
SELECTED MEMORY A B C D TONE GENERATOR
# OF BEEPS 1 2 3 4
GB3212
MS1 LOW LOW OPEN OPEN
MS2 LOW OPEN LOW OPEN A
Memory
D if valid otherwise no change) B (if valid otherwise no change) D (if valid otherwise no change)
The programmable tone generator is capable of producing programmable tones. Upon reception of the tone enable instruction, the DUETTM DIGITAL connects the output of the tone generator to the input of the D/A converter. The programmed tone is then output until a tone disable instruction is issued. When disabled, the normal audio signal is again connected.
WIDE DYNAMIC RANGE COMPRESSION
In this mode it is possible to jump from any memory to any other memory simply by changing the state of both switches. If both switches are changed simultaneously then the transition will be smooth, otherwise, if one switch is changed and then the other, the part will transition to an intermediate memory before reaching the final memory. The part will start in whatever memory the switches are selecting. If the device starts up in a memory other than A, and the memory beep tones are enabled, the device will emit the corresponding tones for that memory. If a memory is invalid and the part starts up with the switches indicating this memory, the part will stay in memory A.
AUDIBLE MEMORY CHANGE INDICATOR
Any combination of adjacent frequency bands can be grouped to form four independent channels of compression. The I/O curve of each channel is divided into up to four regions (linear, compression, return to linear, clipping). The thresholds between these regions are adjustable over a wide range. Each channel has twin average detectors: a fast detector with a configurable time constant and a slow detector with a configurable time constant.
FREQUENCY SHAPING
The DUETTM DIGITAL can be programmed to produce tones to indicate a memory change. Using the Interactive Data Sheet the GB3212 can be configured to either enable or disable the Memory Change Indicator. When the Memory Change Indicator is enabled, there is an option to have a single beep for each memory change or multiple beeps. The amplitude and frequency of the memory change tone can be selected independent of the Tone Generator settings and can be individually selected for each memory. When the memory change multiple beep is enabled and the memory change tone is enabled, then during a memory change operation the selected tone will beep a code to indicate which memory has been selected. The beep sequence will be 150ms ON followed by a 150ms OFF time between the beeps. The memory change beeping code is deciphered in the table below.
The 16-band signal processor acts as a graphic equalizer. The gain of each band can be adjusted over 0 to -42dB range. The width of each band is 500Hz. The bands can be selected to have either an even or odd stacking arrangement. Selecting even stacking shifts the bands in unison by one half-band width (250Hz) effectively doubling the number of potential band edges. The default setting will be even stacking as this effectively results in one "extra" band since the nyquist band is "split" into two 250Hz bands, one from 0 to 250Hz the other from 7750Hz to 8000Hz.
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LOW BATTERY INDICATOR (POWER-ON/POWER-OFF)
POWER MANAGEMENT
The DUETTM DIGITAL hybrids have two power management components on their controller chips: the Power-On-Reset sequence and turn off/end of battery life system. The Power-On-Reset block's purpose is to ensure that a stable turn-on state is achieved. The blocks that are kept OFF are the A/D and preamp channels (both front and rear), the controller, the DSP chip, and the EEPROM power. A small portion of the controller is enabled to monitor the signals coming from the analog POR block. The audio output is muted when the supply voltage is below the turnon threshold and during the power-on sequence. An analog voltage comparator monitors the supply voltage and feeds its output to a digital timer whose purpose is to deglitch bouncy turn-ons. When the supply crosses the 1.1Vdc level (VBON) the timer starts and if the supply voltage maintains a level above 1.0Vdc (VBOFF) for at least 30ms, the disabled blocks will be enabled, else the timer is reset and waits for the analog comparator to signal that supply is again above VBON. This ensures rejection of any turn on transients. Only after this 30ms finishes can the part start to down load EEPROM configuration data, then configure and activate the DSP. Once the part is ON, dropping the supply below VBON causes the Lowbat signal to become active but otherwise the part continues to operate as normal. The Lowbat signal true condition requires that the supply voltage remain below VBON for at least 30ms. Once the Lowbat signal becomes active, the audible low battery voltage indicator will produce two consecutive beeps, 0.45 second long. These two beeps will repeat every 30 seconds. The communication with the hybrid is not possible when the beeps are being produced by the hybrid. The frequency and the amplitude of the beeps are programmable. If the supply drops below 1.0Vdc (VBOFF), then the part is put into an OFF state, there is no debouncing of this signal, and the action is immediate. This level was chosen since the regulator has a 950mV regulation voltage. The regulator needs some headroom to ensure that it maintains good supply rejection, which is critical in high gain, high power applications to prevent system instability. The GB3212 operates in shallow-reset mode, during the power-on sequence circuit starts when the supply voltage rises above the turn-on threshold (VBON) after shutdown. The device will function until the supply voltage drops below the turn-off threshold (VBOFF) but will recover once the supply voltage rises above the turn-on threshold (VBON) again.
The DUETTM DIGITAL was designed to accommodate high power applications. AC ripple on the supply can cause instantaneous reduction of the battery's voltage, potentially disruption the circuit's function. The GB3212 has a separate power supply and ground connection for the output stage. This allows hearing instrument designers to accommodate external RC filters in order to minimize any AC ripple from the supply line. Reducing this AC ripple greatly improves the stability of the circuit and prevents unwanted reset of the circuit caused by spikes on the supply line. For more information on properly designing a filter to reduce supply ripple, please refer to information note Using the GB3211 PARAGON Digital in High Power Application Initial Design Tips document #24561.
GB3212
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PACKAGE DIMENSIONS
PAD LOCATION
0.217 (5.51)
PAD NO. 1
PAD POSITION X 0 -36 -70.25 -102.75 -137.25 -175.5 -180.25 -180.25 -180.25 -139 -97 -65 -30.75 2.5 0 0 -95 0 -0.914 -1.784 -2.610 -3.486 -4.458 -4.578 -4.578 -4.578 -3.531 -2.464 -1.651 -0.781 0.064 0 0 -2.413 Y 0 0 0 0 0 0 32 59.5 90 90 90 90 90 89.75 57.75 30.5 52 0 0 0 0 0 0 0.813 1.511 2.286 2.286 2.286 2.286 2.286 2.280 1.467 0.775 1.321 Xdim 26.5 27.5 23 24 27 31.5 21 21 21 43.5 22.5 23.5 27 21.5 26.5 26.5 24.5
PAD DIMENSION Ydim 24.5 24.5 24.5
0.129 (3.28)
GB 3212
2 3 4 5 6 7 8
0.092 MAX (2.34)
GB3212
24.5 24.5 24.5 21.5 15.5 27.5 27.5 27.5 27.5 27.5 28 18 18.5 24.5 0.622 0.622 0.622 0.622 0.622 0.546 0.394 0.699 0.699 0.699 0.699 0.699 0.711 0.457 0.470 0.622 0.622
9 10 11 12 13
9 8 7 6
10
11
12
13
14 15 16
14 15 16 17 1
0.01875 (0.476)
17
5
4
3
2
1
0.673 0.699 0.584 0.610 0.686 0.800 0.533 0.533 0.533 1.105 0.572 0.597 0.686 0.546 0.673 0.673 0.622
2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17
0.01925 (0.489)
Dimension units are in inches. Dimensions in parentheses are in millimetres, converted from inches and include minor rounding errors. 1.0000 inches = 25.400mm Dimension tolerances: 0.003 (0.08) unless otherwise stated. Work order number: XXXXXX This Hybrid is designed for either point-to-point manual soldering or for reflow according to Gennum's reflow process (Information Note 521-45).
DOCUMENT IDENTIFICATION
PRELIMINARY DATA SHEET The product is in a preproduction phase and specifications are subject to change without notice.
REVISION NOTES:
Corrected Package Dimensions drawing.
For latest product information, visit www.gennum.com
GENNUM CORPORATION
MAILING ADDRESS: P.O. Box 489, Stn. A, Burlington, Ontario, Canada L7R 3Y3 Tel. +1 (905) 632-2996 Fax. +1 (905) 632-5946 SHIPPING ADDRESS: 970 Fraser Drive, Burlington, Ontario, Canada L7L 5P5
GENNUM JAPAN CORPORATION Shinjuku Green Tower Building 27F, 6-14-1, Nishi Shinjuku, Shinjuku-ku, Tokyo, 160-0023 Japan Tel. +81 (03) 3349-5501, Fax. +81 (03) 3349-5505
Gennum Corporation assumes no responsibility for the use of any circuits described herein and makes no representations that they are free from patent infringement. (c) Copyright May 2002 Gennum Corporation. All rights reserved. Printed in Canada.
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MIL


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