Contents
PageSectionTitle
51.Introduction
62.Features of the MSP 3400C
62.1.Features of the Demodulator and Decoder Sections
62.2.Features of the DSP-Section
62.3.Features of the Analog Section
73.Application Fields of the MSP 3400C
73.1.German 2-Carrier System (DUAL FM System)
94.Architecture of the MSP 3400C
94.1.Demodulator Block
94.1.1.Analog Sound IF – Input Section
94.1.2.Quadrature Mixers
104.1.3. Lowpass Filtering Block for Mixed Sound IF Signals
104.1.4.Phase and AM Discrimination
104.1.5.Differentiators
104.1.6.Lowpass Filter Block for Demodulated Signals
104.1.7.High Deviation FM Mode
104.1.8.MSPC-Mute Function in the Dual Carrier FM Mode
1 14.2.Analog Section and SCART Switching Facilities
1 14.3.MSP 3400C Audio Baseband Processing
114.3.1.Dual Carrier FM Stereo/Bilingual Detection
134.4.Audio PLL and Crystal Specifications
134.5.ADR Bus
144.6.S-Bus Interface
154.7.I
2
S Bus Interface
PRELIMINARY DAT A SHEET
165.I
175.1.Protocol Description
185.2.Proposal for MSP 3400C I2C Telegrams
185.2.1.Symbols
185.2.2.Write Telegrams
185.2.3.Read Telegrams
185.2.4.Examples
195.3.Start Up Sequence
206.Programming the Demodulator Part
206.1.Registers: Table and Addresses
216.2.Registers: Functions and Values
216.2.1.Setting of Parameter AD_CV
236.2.2.Control Register ‘MODE_REG’
246.2.3.FIR-Filter Switches
246.2.4.FIR-Parameter
266.2.5.DCO-Increments
2
C Bus Interface: Device and Subaddresses
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Contents, continued
PageSectionTitle
276.3.Sequences to Transmit Parameters and to Start Processing
276.4.Software Proposals for Multistandard TV-Sets
276.4.1.Multistandard System B/G German DUAL FM
286.4.2.Satellite Mode
286.4.3.Automatic Search Function for FM-Carrier Detection
286.4.4.Automatic Standard Detection
297.Programming the Audio Processing Part
297.1.Summary of the DSP Control Registers
317.1.1.Volume Loudspeaker Channel and Headphone Channel
327.1.2.Balance Loudspeaker and Headphone Channel
337.1.3.Bass Loudspeaker and Headphone Channel
337.1.4.Treble Loudspeaker and Headphone Channel
347.1.5.Loudness Loudspeaker and Headphone Channel
347.1.6.Spatial Effects Loudspeaker Channel
357.1.7.Volume SCAR T
357.1.8.Channel Source Modes
367.1.9.Channel Matrix Modes
367.1.10.SCART Prescale
367.1.11.FM Prescale
377.1.12.FM Matrix Modes
377.1.13.FM Fixed Deemphasis
377.1.14.FM Adaptive Deemphasis
377.1.15.I
377.1.16.ACB Register, Definition of the SCART-Switches and DIG_CTR_OUT Pins
387.1.17.Beeper
387.1.18.Identification Mode
387.1.19.FM DC Notch
387.1.20.Mode Tone Control
397.1.21.Equalizer Loudspeaker Channel
397.1.22.Automatic Volume Correction (AVC)
407.1.23.Subwoofer on Headphone Output
407.2.Exclusions
417.3.Summary of Readable Registers
417.3.1.Stereo Detection Register
417.3.2.Quasi Peak Detector
427.3.3.DC Level Register
427.3.4.MSP Hardware Version Code
427.3.5.MSP Major Revision Code
427.3.6.MSP Product Code
427.3.7.MSP ROM Version Code
2
S1 and I2S2 Prescale
MSP 3400C
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MSP 3400C
Contents, continued
PageSectionTitle
438.Specifications
438.1.Outline Dimensions
448.2.Pin Connections and Descriptions
488.3.Pin Configuration
518.4.Pin Circuits
538.5.Electrical Characteristics
538.5.1.Absolute Maximum Ratings
548.5.2.Recommended Operating Conditions
588.5.3.Characteristics
649.Application of the MSP 3400C
6510.DMA Application
6711.MSP Application with External Clock
PRELIMINARY DAT A SHEET
6712.ADR Application
6813.I
6914.APPENDIX A: T echnical Code History
6915.APPENDIX B: Documentation History
2
S Bus in Master/Slave Configuration with Standby Mode
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PRELIMINARY DAT A SHEET
MSP 3400C
Multistandard Sound Processor
sound IF signal-in, down to processed analog AF-out, is
performed in a single chip. The IC is produced in 0.8 µm
Release Notes: The hardware description in this
document is valid for the MSP 3400C – C8 and newer
CMOS technology, combined with high performance
digital signal processing.
codes. Revision bars indicate significant changes
to the previous version.
The MSP 3400C 0.8 µ CMOS version is fully pin and
software compatible to the 1.0 µ MSP 3400 and MSP
1. Introduction
The MSP 3400C is designed as single-chip Multistan-
3410. The main difference between the MSP 3400C and
the MSP 3410, consists of the MSP 3410 being able to
decode NICAM signals.
dard Sound Processor for applications in analog and
digital TV sets, satellite receivers and video recorders.
The MSP-family , which is based on the MSP 2400, dem-
The MSP 3400C is available in PLCC68, PSDIP64,
PSDIP52, and PQFP80 package.
onstrates the progressive development towards highly
integrated multi-functional ICs.
Note: T o achieve compatibility with the functions of MSP
3400 and MSP 3410 (except NICAM), the load seThe MSP 3400C, again, improves function integration:
The full TV sound processing, starting with analog
quences must be programmed as described in the data
sheet of MSP 3410.
MSP 3400C Integrated Functions:
– FM-demodulation of all terrestrial standards (incl. identification decoding)
– FM-demodulation of all satellite standards
– various deemphasis types (incl. Panda1)
– volume, balance, bass, treble, loudness for loudspeaker and headphone output
– automatic volume correction (A.V.C.)
– 5 band graphic equalizer
– subwoofer output alternatively with headphone output
– spatial effect (pseudostereo/basewidth enlargement)
– ADR together with DRP 3510 A
– Dolby ProLogic together with DPL 3418/19/20 A
– 3 pairs of D/A converters
– 1 pair of A/D converters
– SCART switches
I2SI2C
25
MSP 3400C
2
2
2
2
LOUDSPEAKER OUT
HEADPHONE OUT
SCART1 OUT
SCART2 OUT
Sound IF 1
Sound IF 2
MONO IN
SCART1 IN
SCART2 IN
SCART3 IN
ADR/SBus
3
2
2
2
Fig. 1–1: Main I/O Signals MSP 3400C
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MSP 3400C
PRELIMINARY DAT A SHEET
2. Features of the MSP 3400C
2.1. Features of the Demodulator and Decoder
Sections
The MSP 3400C is designed to perform demodulation
of FM-mono TV sound and two carrier FM systems according to the German or Korean terrestrial specs. With
certain constraints, it is also possible to do AM-demodulation according to the SECAM system. Alternatively , the
satellite specs can be processed with the MSP 3400C.
For FM carrier detection in satellite operation, the AMdemodulation offers a powerful feature to calculate the
carrier field strength, which can be used for automatic
search algorithms. So, the IC facilitates a first step towards multistandard capability with its very flexible
application and may be used in TV -sets, satellite tuners,
and video recorders.
The MSP 3400C facilitates profitable multistandard capability, offering the following advantages:
– two selectable analog inputs (TV and SA T-IF sources)
– Automatic Gain Control (AGC) for analog input: input
range: 0.14 – 3 Vpp
– integrated A/D converter for sound-IF inputs
2.2. Features of the DSP-Section
– flexible selection of audio sources to be processed
– digital input and output interfaces via I
nal DSP-processors, surround sound, ADR etc.
– digital interface to process ADR (Astra Digital Radio)
together with DRP 3510 A
– performance of all deemphasis systems including
adaptive Wegener Panda 1 without external components or controlling
– digitally performed FM-identification decoding and de-
matrixing
– digital baseband processing: volume, bass, treble,
5-band equalizer, loudness, pseudostereo, and basewidth enlargement
– simple controlling of volume, bass, treble, equalizer
etc.
– increased audio bandwidth for FM-Audio-signals
(20 Hz – 15 kHz, 1 dB)
2.3. Features of the Analog Section
– three selectable analog pairs of audio baseband in-
puts (= three SCART inputs)
input level: ≤2 V RMS,
input impedance: ≥25 kΩ
2
S-Bus for exter-
– all demodulation and filtering is performed on chip and
is individually programmable
– no external filter hardware is required
– only one crystal clock (18.432 MHz) is necessary
– FM carrier level calculation for automatic search algo-
rithms and carrier mute function
– high deviation FM-mono mode (max. deviation:
approx. 360 kHz)
– one selectable analog mono input (i.e. AM sound),
input level: ≤2 V RMS,
input impedance: ≥10 kΩ
– two high quality A/D converters, S/N-Ratio: ≥85 dB
– 20 Hz to 20 kHz Bandwidth for SCART-to-SCART-
Copy facilities
– MAIN (loudspeaker) and AUX (headphones): two
pairs of 4-fold oversampled D/A-converters
output level per channel: max. 1.4 V RMS
output resistance: max. 5 kΩ
S/N-Ratio: ≥85 dB at maximum volume
max. noise voltage in mute mode: ≤10 µV (BW: 20 Hz
...16 kHz)
– one pair of four-fold oversampled D/A-converters sup-
plying two selectable pairs of SCART -Outputs. Output
level per channel: max. 2 V RMS, output resistance:
max. 0.5 kΩ, S/N-Ratio: ≥85 dB
(20 Hz...16 kHz)
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MSP 3400C
3. Application Fields of the MSP 3400C
The MSP 3400C processes TV sound according to the
German and Korean two carrier system and the commonly used satellite systems. In the following sections,
a brief overview on the German FM-Stereo system
shows what is required of a multistandard audio IC.
3.1. German 2-Carrier System (DUAL FM System)
Since September 1981, stereo and dual sound programs have been transmitted in Germany using the
2-carrier system. Sound transmission consists of the already existing first sound carrier and a second sound
carrier additionally containing an identification signal.
More details of this standard are given in Table 3–1.
the Sound-IF-mixer, the Sound-IF filter
may be omitted.
optional
Feature
Processor
AMU and DMA
or DRP
Fig. 3–1: Typical MSP 3400C application
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MSP 3400C
Table 3–2: Key parameters for B/G, D/K, and M 2-carrier sound system
Sound CarriersCarrier FM1Carrier FM2
B/GD/KMB/GD/KM
Vision/sound power difference13 dB20 dB
Sound bandwidth40 Hz to 15 kHz
Pre-emphasis50 µs75 µs50 µs75 µs
Frequency deviation±50 kHz±25 kHz±50 kHz±25 kHz
Sound Signal Components
Mono transmissionmonomono
Stereo transmission(L+R)/2(L+R)/2R(L–R)/2
Dual sound transmissionlanguage Alanguage B
PRELIMINARY DAT A SHEET
Identification of Transmission Mode on Carrier FM2
Pilot carrier frequency in kHz54.687555.0699
Type of modulationAM
Modulation depth50%
Modulation frequencymono: unmodulated
stereo: 117.5 Hz
dual:274.1 Hz
Note: NICAM decoding can be achieved by using the
MSP 3410 instead of the MSP 3400C. Since the
MSP 3400C and the MSP 3410 are fully pin and software downwards compatible (concerning all features of
MSP 3410), it is possible to decide in the assembly line,
whether the application should be able to decode
NICAM or not.
149.9 Hz
276.0 Hz
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MSP 3400C
4. Architecture of the MSP 3400C
Fig. 4–1 shows a simplified block diagram of the IC. Its
architecture is split into three functional blocks:
1. demodulator section
2. digital signal processing (DSP) section performing
audio baseband processing
3. analog section containing two A/D-converters,
6 D/A-converters, and SCART switching facilities
4.1. Demodulator Block
4.1.1. Analog Sound IF – Input Section
The input pins ANA_IN1+, ANA_IN2+, and ANA_IN–
offer the possibility to connect two different sound IF
sources to the MSP 3400C. By means of bit [8] of
AD_CV (see Table 6–3), either terrestrial or satellite
sound IF signals can be selected. The analog-to-digital
conversion of the preselected sound IF signal is done by
a flash-converter, whose output can be used to control
an automatic gain circuit (AGC), providing optimum level
for a wide range of input levels. It is possible to switch
between automatic gain control and a fixed (setable) input gain. In the optimum case, the input range of the A/D
converter is completely covered by the sound IF source.
Some combinations of SAW filters and sound IF mixer
ICs, however, show large picture components on their
outputs. In this case, filtering is recommended. It was
found that the high pass filters formed by the coupling
capacitors at pins ANA_IN1+ and ANA_IN2+ as shown
in the application diagram are sufficient in most cases.
4.1.2. Quadrature Mixers
The digital input coming from the integrated A/D converter may contain audio information at a frequency range
of theoretically 0 to 9 MHz corresponding to the selected
standards. By means of two programmable quadrature
mixers two different audio sources, for example FM1
and FM2, may be shifted into baseband position. In the
following, the two main channels are provided to process either:
– FM mono (channel 2) or
– FM2 (channel 1) and FM1 (channel 2).
Two independent digital oscillators are provided to generate two pairs of sin/cos-functions. T wo programmable
increments, to be divided up into Low- and High Part, determine frequency of the oscillator, which corresponds
to the frequency of the desired audio carrier. In section
6.1., format and values of the increments are listed.
Sound IF
ANA_IN1+
ANA_IN2+
Mono
MONO_IN
SC1_IN_L
SCART1
SC1_IN_R
SC2_IN_L
SCART2
SC2_IN_R
SC3_IN_L
SCART3
SC3_IN_R
S_DA_IN / ADR_DA
S_CL / ADR_CL
SBUS/ADR Interface
Demodulator
S_ID / ADR_WS
A/D
A/D
SCART Switching Facilities
I2S_DA_OUT
S1..4
FM1 / AM
FM2
IDENT
SCART_L
SCART_R
I2S_DA_IN_1/2
I2S Interface
I2S1/2L/R
LOUDSPEAKER L
LOUDSPEAKER R
DFP
HEADPHONE L
HEADPHONE R
SCART_L
SCART_R
I2SL/R
I2S_CL
I2S_WS
D/A
D/A
D/A
D/A
D/A
D/A
AUD_CL_OUT
XTAL_IN
Audio PLL
XTAL_OUT
DACM_L
Loudspeaker
DACM_R
DACA_L
Headphone
DACA_R
SC1_OUT_L
SCART 1
SC1_OUT_R
SC2_OUT_L
SCART 2
SC2_OUT_R
Fig. 4–1: Architecture of the MSP 3400C
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MSP 3400C
PRELIMINARY DAT A SHEET
DCO1
VREFTOP
AD_CV[7:1]
ANA_IN1+
ANA_IN2+
AD_CV[8]
ANA_IN–
FRAME
FM2
DCO2
AGCAD
Pins
Internal signal lines
Control registers
Fig. 4–2: Demodulator architecture
Oscillator
FIR_REG_1
MixerLowpass
MSPC sound IF channel 1
(MSP-CH1: FM2)
MSPC sound IF channel 2
(MSP-CH2: FM1, AM)
MixerLowpass
FIR_REG_2
Oscillator
DCO2
Phase and
AM Discrimination
Amplitude
Amplitude
Phase and
AM Discrimination
Phase
Phase
Differentiator
Carrier
Detect
AD_CV[9,10,11]
Carrier
Detect
Differentiator
MODE_REG[8]
MODE_REG[8]
LowpassMute
MixerIDENT
LowpassMute
FM1/AM
ADR_DA
FM2
4.1.3. Lowpass Filtering Block for Mixed Sound IF
Signals
FM bandwidth limitation is performed by a linear phase
Finite Impulse Response (FIR-filter). Just like the oscillators’ increments, the filter coefficients are programmable and are written into the IC by the CCU via the control bus. Two not necessarily dif ferent sets of coefficients
are required, one for channel 1 (FM2) and one for channel 2 (FM1=FM-mono). In section 6.2.4., several coefficient sets are proposed.
4.1.4. Phase and AM Discrimination
The filtered sound IF signals are demodulated by means
of the phase and amplitude discriminator block. On the
output, the phase and amplitude is available for further
processing. AM signals are derived from the amplitude
information, whereas the phase information serves for
FM demodulation.
4.1.5. Differentiators
FM demodulation is completed by differentiating the
phase information output.
4.1.6. Lowpass Filter Block for Demodulated
Signals
The demodulated FM and AM signals are further lowpass filtered and decimated to a final sampling frequen-
cy of 32 kHz. The usable bandwidth of the final baseband signals is about 15 kHz.
4.1.7. High Deviation FM Mode
By means of MODE_REG [9], the maximum FM-deviation can be extended to approximately 360 kHz.
Since this mode can be applied only for the MSPC sound
IF channel 2, the corresponding matrices in the baseband processing must be set to sound A. Apart from this,
the coefficient sets 380 kHz FIR_REG2 or 500 kHz
FIR_REG2 must be chosen for the FIR_REG_2. For a
given deviation, in relation to the normal FM-mode, the
audio level of the high-deviation mode is reduced by
6 dB.
4.1.8. MSPC-Mute Function in the Dual Carrier FM
Mode
T o prevent noise effects or FM identification problems in
the absence of one of the two FM carriers, the
MSP 3400 C offers a carrier detection feature, which
must be activated by means of AD_CV[9]. The mute level may be programmed by means of AD_CV[10,11].
(see section 6.2.1.) If no FM carrier is available at the
MSPC channel 1, the corresponding channel FM2 is
muted. If no FM carrier is available at the MSPC channel
2, the corresponding channel FM1 is muted. In case of
the absence of both FM carriers, pure noise will be amplified by the input AGC. Therefore, a proper mute function depends on the noise quality of the TV set’s IF part
and cannot be guaranteed. The mute function is not recommended for the satellite mode.
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PRELIMINARY DAT A SHEET
MSP 3400C
4.2. Analog Section and SCART Switching Facilities
The analog input and output sections offer a wide range
of switching facilities, which are shown in Fig. 4–3. To
design a TV-set with 3 pairs of SCART-inputs and two
pairs of SCART -outputs, no external switching hardware
is required.
The switches are controlled by the ACB bits defined in
the audio processing interface (see section 7. Programming the Audio Processing Part).
If the MSP 3400C is switched off by first pulling ST ANDBYQ low, and then disconnecting the 5 V, but keeping
the 8 V power supply (‘Standby’-mode), the switches
S1, S2, and S3 maintain their position and function. This
facilitates the copying from selected SCART-inputs to
SCART-outputs in the TV-sets standby mode.
SCART_IN
SC1_IN_L/R
MONO
SC2_IN_L/R
SC3_IN_L/R
from Audio Baseband
Processing (DFP)
SCARTL/R
2
2
2
2
D
A
2
ACB[1:0]
00
01
10
11
S1
ACB[3:2]
2
00
2
01
2
10
2
11
S2
ACB[5:4]
00
2
2
01
S3
10
2
to Audio Baseband
Processing (DFP)
A
2
D
SCART_OUT
2
SC1_OUT_L/R
2
SC2_OUT_L/R
SCARTL/R
4.3. MSP 3400C Audio Baseband Processing
By means of the DFP processor, all audio baseband
functions are performed by digital signal processing
(DSP). The DSP functions are grouped into three processing parts: input preprocessing, channel selection,
and channel postprocessing.
The input preprocessing is intended to prepare the various signals of all input sources in order to form a standardized signal at the input to the channel selector. The
signals can be adjusted in volume, are processed with
the appropriate deemphasis, and are dematrixed if necessary .
Having prepared the signals that way , the channel selector makes it possible to distribute all possible source signals to the desired output channels.
The ability to route in an external coprocessor for special
effects like surround and sound field processing is of
special importance. Routing can be done with each input
source and output channel via the I
2
S inputs and out-
puts.
All input and output signals can be processed simulta-
neously. Note that the NICAM input signals are only
available in the MSP 3410 version. While processing the
adaptive deemphasis, no dual carrier stereo (German or
Korean) is possible. Identification values are not valid either.
4.3.1. Dual Carrier FM Stereo/Bilingual Detection
In the German and Korean TV standard, audio information can be transmitted in three modes: mono, stereo, or
bilingual. To obtain information about the current audio
operation mode, the MSP 3400C detects the so-called
identification signal. Information is supplied via the Stereo Detection Register to an external CCU.
Fig. 4–3: SCART-Switching Facilities
Bold lines determine the default configuration
In case of power-on start or starting from standby , the IC
switches automatically to the default configuration,
shown in Fig. 4–3. This takes place after the first I2C
IDENT
AM
Demodu-
lation
Stereo
Detection
Filter
Bilingual
Detection
Filter
Level
Detect
Level
Detect
Stereo
–
Detection
Register
transmission into the DFP part. By transmitting the ACB
register first, the default setting mode can be changed.
Table 4–1: Several examples for recommended channel assignments for demodulator and audio processing part
ModeMSPC Sound IF-
Channel 1 / FM2
B/G-StereoFM2 (5.74 MHz): RFM1 (5.5 MHz): (L+R)/2B/G StereoSpeakers: FMStereo
B/G-BilingualFM2 (5.74 MHz): Sound BFM1 (5.5 MHz): Sound ANo MatrixSpeakers: FM
Sat-Mononot usedFM (6.5 MHz): monoNo MatrixSpeakers: FMSound A
Sat-Stereo7.20 MHz: R7.02 MHz: LNo MatrixSpeakers: FMStereo
Sat-Bilingual7.38 MHz: Sound C7.02 MHz: Sound ANo MatrixSpeakers: FM
Sat High Dev.
Mode (e.g.
EutelSat)
don’t care6.552 MHzNo MatrixSpeakers: FM
4.4. Audio PLL and Crystal Specifications
MSPC Sound IFChannel 2 / FM1
FMMatrix
Channel
Select
H.Phone : FM
H.Phone : FM
H.Phone : FM
nominal free running frequency should match the center
of the tolerance range between 18.433 and 18.431 MHz
The MSP 3400C runs at 18.432 MHz. A detailed specification of the required crystal for different packages and
master/slave applications can be found in Table 8.5.2.
as closely as possible. Due to different layouts of cus-
tomer PCBs, the matching capacitor size should be de-
fined in the application (see also Table 8.5.2.).
The clock supply of the entire system depends on the
MSP 3400C operation mode:
1. FM-Stereo/I
2
S Master operation:
The system clock runs free on the crystal’s 18.432 MHz.
2. I2S Slave operation:
In this case, the system clock is synchronizing on the
I2S_WS signal, which is fed into the MSP 3400C
(Mode_Reg[3] = 1).
4.5. ADR Bus
T o be able to process ADR, the MSPC has a special de-
signed interface to work together with DRP 3510A. T o be
prepared for an upgrade to ADR with an additional DRP
board, the following lines of MSP 3400C should be pro-
vided on a feature connector:
Channel
Matrix
Speakers: Sound A
H.Phone : Sound B
Speakers: Sound A
H.Phone :Sound B=C
Speakers: Sound A
H.Phone : Sound A
3. D2-MAC operation:
In this case, the system clock is locked to a synchronizing signal (DMA_SYNC) supplied by the D2-MAC chip
(Mode_Reg[0] = 1). The DMA and the AMU chips can be
driven by the MSP 3400C audio clock (AUD_CL_OUT).
Remark on using the crystal:
External capacitors at each crystal pin to ground are required. They are necessary for tuning the open-loop frequency of the internal PLL and for stabilizing the frequency in closed-loop operation. The higher the
capacitors, the lower the clock frequency results. The
Digital audio information provided by the DMA 2381 via
the AMU is serially transmitted to the MSP 3400C via the
S-Bus. The MSP 3400C is always in S-Bus master
mode.
The S-Bus interface consists of three pins:
1. S_DA_IN:
Four channels (4*16 bits) per sampling cycle (32 kHz)
are transmitted.
2. S_CL:
Gives the timing for the transmission of S-DATA
(4.608 MHz).
3. S_ID:
After 64 S-CLOCK cycles, the S_ID determines the end
of one sampling period.
A detailed timing diagram is shown in Fig. 4–6.
PRELIMINARY DAT A SHEET
(Data: MSB first)
S-Ident
S-Clock
S-Data
H
L
H
L
H
L
Section A
S-Ident
S-Clock
4.608 MHz
H
L
H
L
64 Clock Cycles
16 Bit Sound 1
A
16 Bit Sound 216 Bit Sound 316 Bit Sound 4
B
Section B
t
S6
S-Ident
t
S1
t
S4
t
S2
S-Clock
4.608 MHz
t
S5
H
L
t
S3
H
L
S-Data
H
L
LSB of Sound 1
Fig. 4–6: S-Bus timing diagram
S-Data
H
MSB of Sound 4
L
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4.7. I2S Bus Interface
By means of this standardized interface, additional feature processors can be connected to the MSP 3400C.
Two possible formats are supported: The standard
mode (MODE_REG[4]=0) selects the SONY format,
where the I2S_WS signal changes at the word boundaries. The so-called PHILIPS format, which is characterized by a change of the I2S_WS signal, one I2S_CL period before the word boundaries, is selected by setting
MODE_REG[4]=1.
MSP 3400C
2
The I
S bus interface consists of five pins:
1. I2S_DA_IN1:
For input, two channels (2*16 bits) per sampling cycle
(32 kHz) are transmitted.
2. I2S_DA_IN2:
For input, two channels (2*16 bits) per sampling cycle
(32 kHz) are transmitted.
3. I2S_DA_OUT:
For output, two channels (2*16 bits) per sampling cycle
(32 kHz) are transmitted.
The MSP 3400C normally serves as the master on the
2
I
S interface. Here, the clock and word strobe lines are
driven by the MSP 3400C. By setting MODE_REG[3]=1,
the MSP 3400C is switched to a slave mode. Now, these
lines are input to the MSP 3400 C, and the master clock
is synchronized to 576 times the I2S_WS rate (32 kHz).
No D2MAC operation is possible in this mode.
Data: MSB first)
2
I
S_WS
SONY ModeSONY Mode
2
S_CL
I
I2S_DAIN
PHILIPS Mode
R LSB L MSB
PHILIPS/SONY Mode programmable by MODE_REG[4]
Detail A
16 bit left channel
4. I2S_CL:
Gives the timing for the transmission of I
2
(1.024 MHz).
5. I2S_WS:
The I2S_WS word strobe line defines the left and right
sample.
A detailed timing diagram is shown in Fig. 4–7.
F
I2SWS
PHILIPS Mode
Detail C
L LSB R MSB
16 bit right channel
S serial data
R LSB L LSB
2
S_DAOUT
I
R LSB L MSB
Detail CDetail A,B
2
I
S_CL
2
I
S_WS as INPUT
I2S_WS as OUTPUT
Detail B
16 bit left channel16 bit right channel
F
I2SCL
T
I2SWS1
T
I2S5
T
I2SWS2
T
I2S6
L LSB R MSB
I2S_CL
2
I
S_DA_IN
I2S_DA_OUT
T
I2S1
T
I2S3
T
I2S2
T
I2S4
R LSB L LSB
Fig. 4–7: I2S Bus timing diagram
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MSP 3400C
PRELIMINARY DAT A SHEET
5. I2C Bus Interface: Device and Subaddresses
As a slave receiver, the MSP 3400C can be controlled
2
via I
C bus. Access to internal memory locations is
achieved by subaddressing. The demodulator part and
the audio processor part (DFP) have two separate subaddressing register banks.
In order to allow for more MSP 3400C ICs to be connected to the control bus, an ADR_SEL pin has been implemented. With ADR_SEL pulled to high, the MSP
3400C responds to changed device addresses, thus two
identical devices can be selected. Other devices of the
same family will have different subaddresses (e.g. 34x0)
By means of the RESET bit in the CONTROL register,
all devices with the same device address are reset.
The IC is selected by asserting a special device address
in the address part of an I
2
C transmission. A device address pair is defined as a write address (80 hex or 84
hex) and a read address (81 hex or 85 hex). Writing is
done by sending the device write address first, followed
by the subaddress byte, two address bytes, and two
data bytes. For reading, the read address has to be
transmitted first by sending the device write address (80
hex or 84 hex), followed by the subaddress byte, and two
address bytes. Without sending a stop condition, reading of the addressed data is done by sending the device
read address (81 hex or 85 hex) and reading two bytes
of data. Refer to Fig. 5–1 I
2
C Bus Protocol and section
5.2. Proposal for MSP 3400C I2C Telegrams.
Due to the internal architecture of the MSP 3400C, the
IC cannot react immediately to an I2C request. The typical response time is about 0.3 ms. If the addressed processor is not ready for further transmissions on the I
2
bus, the clock line I2C_CL is pulled low. This puts the
current transmission into a wait state. After a certain period of time, the MSP 3400C releases the clock, and the
interrupted transmission is carried on.
2
The I
C Bus lines can be set tristate by switching the IC
into “Standby”-mode.
I2C-Bus error conditions:
In case of any internal error, the MSP’ s wait-period is extended to 1.77 ms. Afterwards, the MSP does not acknowledge (NAK) the device address. The data line will
be left HIGH by the MSP, and the clock line will be released. The master can then generate a STOP condition
to abort the transfer.
By means of NAK, the master is able to recognize the error state and to reset the IC via I
2
C-Bus. While transmitting the reset protocol (section. 5.2.4.) to ‘CONTROL’,
the master must ignore the not acknowledge bits (NAK)
of the MSP.
A detailed timing diagram is shown in Fig. 5–1 and
Fig. 5–2.
C
Table 5–1: I
2
C Bus Device Addresses
ADR_SELLowHighLeft Open
ModeWriteReadWriteReadWriteRead
After power on or RESET, the IC is in an inactive state.
The CCU has to transmit the required coefficient set for
a given operation via the I
with the demodulator part. If required for any reason, the
audio processing part can be loaded before the demodulator part.
The reset pin should not be >0.45 DVSUP (see recommended operation conditions) before the 5 Volt digital
power supply (DVSUP) and the analog power supply
(AVSUP) are >4.75 Volt and the MSP-Clock is running
(Delay: 2 ms max, 0.5 ms typ.).
This means, if the reset low-high edge starts with a delay
of 2 ms after DVSUP>4.75 Volt and AVSUP>4.75 V olt,
even under worst case conditions, the reset is ok.
2
C bus. Initialization must start
MSP 3400C
DVSUP/V
AVSUP/V
4.75
Oscillator
RESETQ
0.45 * DVSUP
Fig. 5–3: Power-up sequence
max. 2
min. 2
time / ms
time / ms
time / ms
Note: The reset should
not reach high level before the oscillator has
started. This requires a
reset delay of >2 ms
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MSP 3400C
6. Programming the Demodulator Part
6.1. Registers: Table and Addresses
In Table 6–1, all Write Registers are listed.
All transmissions on the control bus are 16 bits wide.
Data for the demodulator part has 8 or 12 significant bits.
These data have to be inserted LSB bound and filled
with zero bits into the 16 bit transmission word. If channel 1 or channel 2 is selected in the channel matrix while
any of the parameters are changed, the corresponding
output must be muted. Click and crack noise may occur
during coefficient changes. Table 4–1 explains how to
assign FM carriers to the MSPC-Sound IF channels and
the corresponding matrix modes in the audio processing
part.
PRELIMINARY DAT A SHEET
Table 6–1: MSP 3400C demodulator write registers
RegisterProtocolWrite
Address
(hex)
AD_CVlong00BBinput selection, configuration of AGC and Mute Function,
MODE_REGlong0083mode register
FIR_REG_1
FIR_REG_2
DCO1_LO
DCO1_HI
DCO2_LO
DCO2_HI
PLL_CAP
Table 6–2: MSP 3400C demodulator read registers
1)
long
long
long
long
long
long
short1Fswitchable PLL capacities
0001
0005
0093
009B
00A3
00AB
Function
and selection of A/D-converter
serial shift register for 6 ⋅ 8 bit, filter coefficient channel 1 (48 bit)
serial shift register for 6 ⋅ 8 bit, + 2 ⋅ 12 bit off set (total 72 bit)
increment channel 1 Low Part
increment channel 1 High Part
increment channel 2 Low Part
increment channel 2 High Part
RegisterProtocolRead
Address
(hex)
PLL_CAP
AGC_RMS
C_AD_BITSlong0023A read from this address always responds with 0. This ensures
1)
The registers PLL_CAP and AGC_RMS are only available in MSP 3400C. In MSP 3410 and MSP 34x0D, this register
cannot be accessed.
1)
1)
short1Fswitchable PLL capacities
short1ERMS value, comparable with reference value
Function
software compatibility with the MSP 3410 readout. Reading 0 from
this register signals “No NICAM”.
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PRELIMINARY DAT A SHEET
6.2. Registers: Functions and Values
In the following, the functions of several registers are explained and their (default) values are defined.
6.2.1. Setting of Parameter AD_CV
Table 6–3: AD_CV Register
AD_CV Bit RangeMeaningSettings
AD_CV [0]not usedmust be set to 0
MSP 3400C
AD_CV [6:1]Reference level in case of Automatic Gain
see Table 6–5
Control = on.
Constant gain factor when Automatic Gain
see Table 6–6
Control = off .
AD_CV [7]Determination of Automatic Gain or Constant
Gain
0 = constant gain
1 = automatic gain
AD_CV [8]Selection of analog input0 = ANALOG IN1
1 = ANALOG IN2
AD_CV [9]MSPC-Carrier-Mute Function0 = off: no mute
1 = on: mute (see section 4.1.8.)
AD_CV [1 1–10]Programmable Carrier-Mute Levelsee Table 6–4
AD_CV [15–12]not usedmust be set to 0
Table 6–4: Carrier Mute Level
StepAD_CV [11:10]
binary
0
1
2
3
00
01
10
11
AD_CV [11:10]
decimal
0
1
2
3
Internal reference level for mute active
(dBr: relative to MSP 3410 )
0 dBr
–3 dBr
–6 dBr
–12 dBr
Table 6–5: Reference valuesAD_CV [6:1] for active AGC (AD_CV[7] = 1)
ApplicationInput Signal ContainsRef. Value
binary
Ref. Value
decimal
Range of Input Signal
at pin ANA_IN_1+ and
ANA_IN_2+
Terrestrial TV2 FM Carriers101000400.14 – 3 V
SAT1 or more FM Carriers100011350.14 – 3 V
ADR1 or more FM Carriers and
010100200.14 – 3 V
pp
pp
pp
1)
1)
1)
1 or more ADR Carriers
1)
For signals above 1.4 Vpp, the minimum gain of 3 dB is switched and overflow of the AD converter may result. Due
to the robustness of the internal processing in FM mode, the IC works properly up to and even more than 3 Vpp. In AM
mode, of course, no AD converter overflow is allowed. As a consequence, in the AM-mode, the maximum input at pins
41 or 43 must not exceed 1.4 Vpp.
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MSP 3400C
Table 6–6: AD_CV parameters for constant input gain (AD_CV[7]=0)
to the robustness of the internal processing in FM mode, the IC works properly up to and even more than 3 Vpp. In AM
mode, of course, no AD converter overflow is allowed. As a consequence, in the AM-mode, the maximum input at pins
41 or 43 must not exceed 1.4 Vpp.
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PRELIMINARY DAT A SHEET
MSP 3400C
6.2.2. Control Register ‘MODE_REG’
The register ‘MODE_REG’ contains the control bits determining the operation mode of the MSP 3400C; Table
6–7 explains all bit positions.
Table 6–7: Control word ‘MODE_REG’: All bits are “0” after power-on-reset
BitFunctionCommentDefinitionRecom-
mendation
[0]DMA_SYNC
1)
Synchronization to DMA0 : off
X
1 : on
[1]DCTR_TRIDigital control out 0/1 tristate0 : active
0
1 : tristate
[2]I2S_TRII2S outputs tristate (I2S_CL,
I2S_WS, I2S_DA_OUT)
[3]I2S Mode
1)
Master/Slave mode of the
2
I
S bus
[4]I2S_WS ModeWS due to the Sony or
Philips-Format
[5]Audio_CL_OUTswitch Audio_Clock_Output
to tristate
0 : active
1 : tristate
0 : Master
1 : Slave
0 : Sony
1 : Philips
0 : on
1 : tristate
0
X
X
X
[6]not usedmust be 00
[7]FM1 FM2MSPC-channel 1 modes.Table 6–8
[8]AMMSPC-channel 1/2 mode0 : FM
s.Table 6–8
1 : AM
[9]HDEVHigh Deviation Mode
(channel matrix must be
0 : normal mode
1 : high deviation mode
s.Table 6–8
sound A )
[10]not usedmust be 11
[11]S-Bus Mode
2)
mode of Pins S_CL and S_ID0 : Tristate
0
1 : Active
[12]FM2 FIR Filter Gain
(FM2 = Ch1)
[13]FM2 FIR Filter Coeff. Set
(FM2 = Ch1)
[14]ADRMode of ADR Interface0 : normal mode
see table 6–100 : Gain = 6 dB
1 : Gain = 0 dB
see table 6–100 : use FIR_REG_1
1 : use FIR_REG_2
0
0
X
1 : ADR mode
[15]AM-Gainadditional gain in AM-mode0 : 0 dB
0
1 : +12 dB
1)
In case of synchronization to DMA, no I2S-slave mode possible.
In case of I2S-slave mode, no synchronization to DMA allowed. I2S-Slave mode dominates.
2)
The normal operation mode is ‘Tristate’; SBUS is only used in conjunction with DMA.
The FIR filter for channel1/FM2 can use either
FIR_REG_1 coefficients or FIR_REG_2 coefficients by
means of MODE_REG[13]. Herewith, it is no longer necessary to transmit both coefficient sets in FM-terrestrial
mode. The loading sequence for FIR_REG_2 is suffi-
1FM2_Coeff. (5)8see Table 6–10.
2FM2_Coeff. (4)8
3FM2_Coeff. (3)8
cient.
4FM2_Coeff. (2)8
The additional gain of +6 dB in channel1/FM2 can be
switched to 0 dB by means of MODE_REG[12]. T ogether with MODE_REG[13] set to 1, in satellite mode, it is
no longer necessary to transmit both FIR filter coefficient
5FM2_Coeff. (1)8
6FM2_Coeff. (0)8
sets. The loading sequence for FIR_REG_2 is sufficient.
WRITE_ADR = FIR_REG_2 (Channel 2: FM1/FM
6.2.4. FIR-Parameter
mono)
No.Symbol NameBitsValue
The following data values (see Table 6–9) are to be
transferred 8 bits at a time embedded LSB-bound ina 16 bit word. These sequences must be obeyed. To
change a coefficient set, the complete block
FIR_REG_1 or FIR_REG_2 must be transmitted. The
new coefficient set will be active without a load_reg rou-
MODE_REG[12] should be set to 0 (= 6 dB gain) if the level of the FM2-carrier processed in MSP-Ch1 is appr. 7 dB below the FM1-carrier of
MSP-Ch2. If both carriers have the same level, MODE_REG[12] must be set to 1 (=0 dB gain).
and FIR2 0005
hex
hex
FIR filter corresponds to a
bandpass with a bandwidth of B = 130 to 500 kHz
B
frequencyf
c
–8
36
MODE_REG[13]: If in MSP-Channel 1 and 2 the same bandwidth is required, it is sufficient to transmit FIR_REG2 only and to set
MODE_REG[13] to 1.
For compatibility (besides the above programming), the FIR-filter programming as used for the MSP 3410B is also possible.
ADR coefficients are listed in the DRP-data sheet.
The 130 kHz coefficients are based on subcarriers, which are 7 dB below an existent main carrier.
MICRONAS INTERMETALL25
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MSP 3400C
6.2.5. DCO-Increments
For a chosen TV standard, a corresponding set of 24-bit
increments determining the mixing frequencies of the
quadrature mixers, has to be written into the IC. In T able
6–11, several examples of DCO increments are listed.
It is necessary to divide them into low part and high part.
The formula for the calculation of the increments for any
chosen IF-Frequency is as follows:
PRELIMINARY DAT A SHEET
INCR
with: int = integer function
Conversion of INCR into hex-format and separation of
the 12-bit low and high parts lead to the required increments. (DCO1_HI or _LO for channel 1, DCO2_HI or LO
for channel 2).
Table 6–11: DCO increments for the MSP 3400C; frequency in MHz, increments in Hex
= int(f/fs ⋅ 224)
dez
f= IF-frequency in MHz
fS= sampling frequency (18.432 MHz)
Frq. MHzDCO_HIDCO_LOFrq. MHzDCO_HIDCO_LO
4.503E80000
5.04
5.5
5.58
5.7421875
6.0
6.2
6.5
6.552
0460
04C6
04D8
04FC
0535
0561
05A4
05B0
0000
038E
0000
00AA
0555
0C71
071C
0000
5.76
5.85
5.94
6.6
6.65
6.8
0500
0514
0528
05BA
05C5
05E7
0000
0000
0000
0AAA
0C71
01C7
7.02061800007.206400000
7.38066800007.5606900000
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PRELIMINARY DAT A SHEET
MSP 3400C
6.3. Sequences to Transmit Parameters and to Start
Processing
After having been switched on, the MSPC must be initialized by transmitting the parameters according to the
LOAD_SEQ_1/2 of T able 6–12. In the MSPC, the initialization sequence must no longer be terminated by transmitting LOAD_REG_1/2. The transmitted data are active as soon as the corresponding I
finished. Therefore, while changing parameters of the
demodulator section, a mute is recommended for the affected channel (LOAD_SEQ_1/2: mute all FM,
LOAD_SEQ_1: switch audio processing to channel2/FM1 or mute channel1/FM2). Otherwise, distorted
sound may occur while switching.
For FM-stereo operation, the evaluation of the identification signal must be performed. For positive identification
check, the MSP 3400C sound channels have to be
switched corresponding to the detected operation
mode.
2
C telegram has
6.4. Software Proposals for Multistandard TV-Sets
To familiarize the reader with the programming scheme
of the MSP 3400C, two examples in the shape of flow
diagrams are shown in the following sections.
6.4.1. Multistandard System B/G German DUAL FM
Fig. 6–1 shows a flow diagram for the CCU software,
applied for the MSP 3400C in a TV set, which facilitates
all standards according to System B/G. For the instructions used in the diagram, please refer to Table 6–12.
After having switched on the TV-set and having initialized the MSP 3400C (LOAD_SEQ_1/2), FM-mono
sound is available.
Fig. 6–1 shows how to check for any stereo or bilingual
audio information in channel 1. If successful, the
MSP 3400C must be switched to the desired audio
mode.
Table 6–12: Sequences to initialize and start the MSP 3400C
LOAD_SEQ_1/2: General Initialization
1. AD_CV
2. FIR_REG_1
3. FIR_REG_2
4. MODE_REG
5. DCO1_LO
6. DCO1_HI
7. DCO2_LO
8. DCO2_HI
FM_IDENT_CHECK: Decoding of the identification signal
1. Evaluation of the stereo detection register (DFP register 0018
2. If necessary, switch the corresponding sound channels within the audio processing part
LOAD_SEQ_1: Reinitialization of Channel 1 without affecting Channel 2
1. FIR_REG_1
2. MODE_REG
3. DCO1_LO
4. DCO1_HI
(6 ⋅ 8 bit)
(12 bit)
(12 bit)
, high part)
hex
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MSP 3400C
PRELIMINARY DAT A SHEET
Pause
START
LOAD_SEQ_1/2
Channel 1:
FM2 Parameter
Channel 2:
FM1 Parameter
Audio Processing Init
Bilingual
Set FM Matrix:
To NO_MATRIX
Set Channel Matrix:
To SOUND A or B
Mono
Set FM Matrix:
To NO_MATRIX
Set Channel Matrix:
To SOUNDA
Stereo
Set FM Matrix:
To G/KMATRIX
Set Channel Matrix:
To STEREO
< –t
IDENT_CHECK
0x0018
> –t & < t
> t
FM_
6.4.3. Automatic Search Function for FM-Carrier Detection
The AM demodulation ability of the MSP 3400C offers
the possibility to calculate the “field strength” of the momentarily selected FM carrier, which can be read out by
the CCU. In SAT receivers, this feature can be used to
make automatic FM carrier search possible.
Therefore, the MSPC has to be switched to AM-mode
(MODE_REG[8]), FM-Prescale must be set to
7F
=+127
hex
off. The sound-IF frequency range must now be
“scanned” in the MSPC-channel 2 by means of the programmable quadrature mixer with an appropriate incremental frequency (i.e. 10 kHz).
After each incrementation, a field strength value is available at the quasi-peak detector output (quasi-peak detector source must be set to FM), which must be examined for relative maxima by the CCU. This results in
either continuing search or switching the MSP 3400C
back to FM demodulation mode.
During the search process, the FIR_REG_2 must be
loaded with the coefficient set “AUTOSEARCH”, which
enables small bandwidth, resulting in appropriate field
strength characteristics. The absolute field strength value (can be read out of “quasi peak detector output FM1”)
also gives information on whether a main FM carrier or
a subcarrier was detected, and as a practical consequence, the FM bandwidth (FIR_REG_1/2) and the
deemphasis (50 µs or adaptive) can be switched automatically.
, and the FM DC Notch must be switched
dez
Fig. 6–1: CCU software flow diagram: Standard B/G,
t = threshold value for stereo/bilingual detection
START
LOAD_SEQ_1/2
MSP-Channel 1:
FM2-Parameter
MSP-Channel 2:
FM1-Parameter
Audio Processing Init
STOP
Fig. 6–2: CCU software flow diagram: SAT-mode
6.4.2. Satellite Mode
Fig. 6–2 shows the simple flow diagram to be used for
the MSP 3400C in a satellite receiver. For FM-mono
operation, the corresponding FM carrier should preferably be processed at the MSPC-channel 2 or at the
MSPC-channel 1 with FIR gain = 0 dB.
Due to the fact that a constant demodulation frequency
offset of a few kHz, leads to a DC-level in the demodulated signal, a further fine tuning of the found carrier can
be achieved by evaluating the “DC Level Readout FM1”.
Therefore, the FM DC Notch must be switched on, and
the demodulator part must be switched back to FM-demodulation mode.
For a detailed description of the automatic search function, please refer to the corresponding MSP 3400C Windows software.
Note: The automatic search is still possible by evaluating only the DC Level Readout FM1 (DC Notch On) as
it is described with the MSP 3410, but the above mentioned method is faster.
6.4.4. Automatic Standard Detection
The AM demodulation ability of the MSP 3400 C enables a simple method of deciding between standard
B/G (FM-carrier at 5.5 MHz) and standard I (FM-carrier
at 6.0 MHz). It is achieved by tuning the MSP 3400C in
the AM-mode to the two discrete frequencies and evaluating the field strength via the DC level register or the
quasi-peak detector output (Mode_Reg, DC Notch, FM
Prescale as described in section 6.4.3.).
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PRELIMINARY DAT A SHEET
7. Programming the Audio Processing Part
7.1. Summary of the DSP Control Registers
Control registers are 16 bit wide. Transmissions via I
2
bus have to take place in 16 bit words. Single data entries are 8 bit. Some of the defined 16 bit words are divided into low and high byte, thus holding two different
control entities. All control registers are readable.
Note: Unused parts of the 16 bit registers must be zero.
Table 7–1: DSP Control Registers
MSP 3400C
C
NameI2C Bus
Address
Volume loudspeaker channel0000
Volume / Mode loudspeaker channelL1/8 dB Steps, Reduce Volume / Tone Control00
Balance loudspeaker channel [L/R]0001
DIG_OUT Pins)
Beeper0014
Identification Mode0015
Prescale I2S10016
FM DC Notch0017
Mode Tone Control0020
Equalizer loudspeaker ch. band 10021
Equalizer loudspeaker ch. band 20022
Equalizer loudspeaker ch. band 30023
Equalizer loudspeaker ch. band 40024
Equalizer loudspeaker ch. band 50025
Loudness filter characteristicL[NORMAL, SUPER_BASS]NORMAL
Note: For compatibility to new technical codes of the MSP 3400C, please consider the following compatibility restrictions:
If adaptive deemphasis is switched on, 75 µs deemphasis must be activated.
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PRELIMINARY DAT A SHEET
MSP 3400C
7.1.1. Volume Loudspeaker Channel and Headphone Channel
RESET
Reduce Tone Controlx0011
Compromise Modex0102
hex
hex
If the clipping mode is set to “Reduce Volume”, the following clipping procedure is used: To prevent severe
clipping effects with bass, treble, or equalizer boosts, the
internal volume is automatically limited to a level where,
in combination with either bass, treble, or equalizer setting, the amplification does not exceed 12 dB.
If the clipping mode is “Reduce Tone Control”, the bass
or treble value is reduced if amplification exceeds 12 dB.
If the equalizer is switched on, the gain of those bands
is reduced, where amplification together with volume exceeds 12 dB.
The highest given positive 1 1-bit number (7F0
hex
) yields
in a maximum possible gain of 12 dB. Decreasing the
volume register by 1 LSB decreases the volume by
0.125 dB. Volume settings lower than the given minimum mute the output. With large scale input signals,
positive volume settings may lead to signal clipping.
With Fast Mute, volume is reduced to mute position by
digital volume only . Analog volume is not changed. This
reduces any audible DC plops. Going back from Fast
Mute should be done to the volume step before Fast
Mute was activated.
If the clipping mode is “Compromise Mode”, the bass or
treble value and volume are reduced half and half if amplification exceeds 12 dB (see example below). If the
equalizer is switched on, the gain of those bands is reduced half and half, where amplification together with
volume exceeds 12 dB.
Example:Vol.:
+6 dB
Bass:
+9 dB
Treble:
+5 dB
Red. Volume395
Red. Tone Con.665
Compromise4.57.55
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MSP 3400C
PRELIMINARY DAT A SHEET
7.1.2. Balance Loudspeaker and Headphone
Channel
Positive balance settings reduce the left channel without
affecting the right channel; negative settings reduce the
right channel leaving the left channel unaffected. In linear mode, a step by 1 LSB decreases or increases the
balance by about 0.8% (exact figure: 100/127). In logarithmic mode, a step by 1 LSB decreases or increases
the balance by 1 dB.
Balance Mode
0001
hex
LSB
loudspeaker
Balance Mode
0030
hex
LSB
headphone
linearxxx00
hex
RESET
logarithmicxxx11
hex
Linear Mode
Logarithmic Mode
Balance loudspeaker
0001
hex
H
channel [L/R]
Balance headphone
0030
hex
H
channel [L/R]
Left –127 dB, Right 0 dB0111 11117F
Left –126 dB, Right 0 dB0111 11107E
Left –1 dB, Right 0 dB0000 000101
Left 0 dB, Right 0 dB0000 000000
RESET
Left 0 dB, Right –1 dB1111 1111FF
Left 0 dB, Right –127 dB1000 000181
Left 0 dB, Right –128 dB1000 000080
hex
hex
hex
hex
hex
hex
hex
Balance loudspeaker
0001
hex
H
channel [L/R]
Balance headphone
0030
hex
H
channel [L/R]
Left muted, Right 100%0111 11117F
Left 0.8%, Right 100%0111 11107E
Left 99.2%, Right 100%0000 000101
Left 100%, Right 100%0000 000000
RESET
Left 100%, Right 99.2%1111 1111FF
Left 100%, Right 0.8%1000 001082
With positive bass settings, internal overflow may occur
even with overall volume less than 0 dB. This will lead to
a clipped output signal. Therefore, it is not recommended to set bass to a value that, in conjunction with
volume, would result in an overall positive gain.
Loudspeaker channel: Bass and Equalizer cannot work
simultaneously (see Table: Mode Tone Control). If
Equalizer is used, Bass and Treble coef ficients must be
set to zero and vice versa.
With positive treble settings, internal overflow may occur
even with overall volume less than 0 dB. This will lead to
a clipped output signal. Therefore, it is not recommended to set treble to a value that, in conjunction with
volume, would result in an overall positive gain.
Loudspeaker channel: Treble and Equalizer cannot
work simultaneously (see Table: Mode T one Control). If
Equalizer is used, Bass and Treble coef ficients must be
set to zero and vice versa.
Stereo Basewidth Enlargement (SBE) and
Pseudo Stereo Effect
hex
00000
RESET
00000
(PSE). (Mode A)
Stereo Basewidth En-
00102
largement (SBE) only.
(Mode B)
Spatial Effect Cus-
0005
hex
tomize Coefficient
H
hex
hex
hex
hex
hex
hex
hex
[7:4]
hex
hex
hex
[3:0]
Loudness increases the volume of low and high frequency signals, while keeping the amplitude of the 1 kHz reference frequency constant. The intended loudness has
to be set according to the actual volume setting. Because loudness introduces gain, it is not recommended
to set loudness to a value that ,in conjunction with volume, would result in an overall positive gain.
By means of ‘Mode Loudness’, the corner frequency for
bass amplification can be set to two different values. In
Super Bass mode, the corner frequency is shifted up.
The point of constant volume is shifted from 1 kHz to
2 kHz.
max high pass gain00000
hex
RESET
2/3 high pass gain00102
1/3 high pass gain01004
min high pass gain01106
automatic10008
hex
hex
hex
hex
There are several spatial effect modes available:
Mode A (low byte = 00
) is compatible to the formerly
hex
used spatial effect. Here, the kind of spatial effect depends on the source mode. If the incoming signal is in
mono mode, Pseudo Stereo Effect is active; for stereo
signals, Pseudo Stereo Effect and Stereo Basewidth
Enlargement is effective. The strength of the effect is
controllable by the upper byte. A negative value reduces
the stereo image. A rather strong spatial effect is recommended for small TV sets where loudspeaker spacing is
rather close. For large screen TV sets, a more moderate
spatial effect is recommended. In mode A, even in case
of stereo input signals, Pseudo Stereo Effect is active,
which reduces the center image.
In Mode B, only Stereo Basewidth Enlargement is effective. For mono input signals, the Pseudo Stereo Effect
has to be switched on.
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PRELIMINARY DAT A SHEET
MSP 3400C
It is worth mentioning, that all spatial effects affect amplitude and phase response. With the lower 4 bits, the frequency response can be customized. A value of 0000
bin
yields a flat response for center signals (L = R) but a high
pass function of L or R only signals. A value of 0110
bin
has a flat response for L or R only signals but a lowpass
function for center signals. By using 1000
, the fre-
bin
quency response is automatically adapted to the sound
material by choosing an optimal high pass gain.
Note: For Headphone output it is also possible to select
a subwoofer signal derived from the Loudspeaker channel. For more details see section 7.1.23.
The sum/difference mode can be used together with the
quasi-peak detector to determine the sound material
mode. If the difference signal on channel B (right) is near
to zero, and the sum signal on channel A (left) is high,
the incoming audio signal is mono. If there is a significant
level on the difference signal, the incoming audio is stereo.
7.1.10. SCART Prescale
7.1.11. FM Prescale
Volume Prescale FM
000e
hex
(normal FM mode)
OFF0000 000000
RESET
Maximum Volume
(28 kHz deviation
1)
0111 11117F
recommended FIRbandwidth: 130 kHz)
Deviation 50 kHz
1)
0100 100048
recommended FIRbandwidth: 200 kHz
Deviation 75 kHz
1)
0011 000030
recommended FIRbandwidth: 200 or
280 kHz
For the High Deviation Mode, the FM prescaling values
can be used in the range between 13
hex
to 30
. Please
hex
consider the internal reduction of 6 dB for this mode. The
FIR-bandwidth should be selected to 500 kHz.
1)
Given deviations will result in internal digital full scale
signals. Appropriate clipping headroom has to be set by
the customer. This can be done by decreasing the listed
values by a specific factor.
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PRELIMINARY DAT A SHEET
MSP 3400C
7.1.12. FM Matrix Modes (see also Table 4–1)
FM matrix000e
hex
NO MATRIX0000 000000
L
hex
RESET
GSTEREO0000 000101
KSTEREO0000 001002
hex
hex
NO_MA TRIX is used for terrestrial mono or satellite stereo sound. GSTEREO dematrixes (L+R, 2R) to (2L, 2R)
and is used for German dual carrier stereo system
(Standard B/G). KSTEREO dematrixes (L+R, L–R) to
(2L, 2R) and is used for the Korean dual carrier stereo
system (Standard M).
7.1.16. ACB Register, Definition of the SCARTSwitches and DIG_CTR_OUT Pins
ACB Register0013
hex
H
DSP In
Selection of Source:
SC_1_IN
MONO_IN
SC_2_IN
SC_3_IN
xxxx xx00RESET
xxxx xx01
xxxx xx10
xxxx xx11
SC_1_OUT_L/R
Selection of Source:
SC_3_IN
SC_2_IN
MONO_IN
DA_SCART
xxxx 00xxRESET
xxxx 01xx
xxxx 10xx
xxxx 11xx
7.1.14. FM Adaptive Deemphasis
FM Adaptive
000f
hex
L
Deemphasis WP1
OFF0000 000000
hex
RESET
WP10011 11113F
hex
Must be set to ‘OFF’ in case of dual carrier stereo (German or Korean). If ‘ON’, FM fixed deemphasis must be
set to 75 µs.
SC_2_OUT_L/R
Selection of Source:
DA_SCART
SC_1_IN
MONO_IN
xx00 xxxxRESET
xx01 xxxx
xx10 xxxx
DIG_CTR_OUT1
low
high
x0xx xxxxRESET
x1xx xxxx
DIG_CTR_OUT2
low
high
0xxx xxxxRESET
1xxx xxxx
RESET: The RESET state is taken at the time of
the first write transmission on the control bus to
the audio processing part (DSP). By writing to the
ACB register first, the RESET state can be redefined.
A squarewave beeper can be added to the loudspeaker
channel and the headphone channel. The addition point
is just before loudness and volume adjustment.
7.1.18. Identification Mode
Identification Mode0015
Standard B/G (German
Stereo)
0000 000000
RESET
hex
L
hex
7.1.19. FM DC Notch
FM DC Notch0017
hex
ON0000 000000
L
hex
Reset
OFF0011 11113F
hex
The DC compensation filter (FM DC Notch) for FM input
can be switched off. This is used to speed up the automatic search function (see sector 6.4.3.). In normal FMmode, the FM DC Notch should be switched on.
7.1.20. Mode Tone Control
Mode Tone Control00020
hex
Bass and Treble0000 000000
H
hex
RESET
Equalizer1111 1111FF
hex
By means of ‘Mode T one Control’, Bass/Treble or Equalizer may be activated.
Standard M (Korean
0000 000101
hex
Stereo)
Reset of Ident-Filter0011 11113F
hex
To shorten the response time of the identification algorithm after a program change between two FM-stereo
capable programs, the reset of ident-filter can be applied.
Sequence:
1. Program change
2. Reset ident-filter
3. Wait at least 1 msec.
4. Set identification mode back to standard B/G or M
5. Wait approx. 1 sec.
6. Read stereo detection register
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PRELIMINARY DAT A SHEET
MSP 3400C
7.1.21. Equalizer Loudspeaker Channel
Band 1 (below 120 Hz)0021
Band 2 (Center: 500 Hz)0022
Band 3 (Center: 1.5 kHz)0023
Band 4 (Center: 5 kHz)0024
Band 5 (above 10kHz)0025
hex
hex
hex
hex
hex
+12 dB0110 000060
+11 dB0101 100058
+1 dB0000 100008
+1/8 dB0000 000101
0 dB0000 000000
RESET
–1/8 dB1111 1111FF
–1 dB1111 1000F8
Different sound sources (e.g. Terrestrial channels, SA T
channels or SCART) fairly often don’t have the same
H
H
volume level. Advertisement during movies as well has
mostly a different (higher) volume level, than the movie
itself. The Automatic Volume Correction (AVC) solves
this problem and equalizes the volume levels.
H
H
H
hex
hex
hex
The absolute value of the incoming signal is fed into a
filter with 16ms attack time and selectable decay time.
The decay time must be adjusted as shown in the table
above. This attack/decay filter block works similar to a
peak hold function. The volume correction value with it’s
quasi continuous step width is calculated using the attack/decay filter output.
The Automatic Volume Correction works with an internal
reference level of –18 dBFS. This means, input signals
hex
with a volume level of –18 dBFS will not be affected by
the A VC. If the input signals vary in a range of –24 dB to
hex
hex
0 dB the AVC compensates this.
Example: A static input signal of 1 kHz on Scart has an
output level as shown in the table below.
hex
–11dB1010 1000A8
–12 dB1010 0000A0
hex
hex
With positive equalizer settings, internal overflow may
occur even with overall volume less than 0 dB. This will
lead to a clipped output signal. Therefore, it is not recommended to set equalizer bands to a value that, in conjunction with volume, would result in an overall positive
gain.
Equalizer must not be used simultaneously with Bass
and Treble (Mode Tone Control must be set to FF to use
the Equalizer).
To reset the internal variables, the AVC should be
switched off and on during any channel or source
change. For standard applications, the recommended
decay time is 4sec.
Note: AVC should not be used in any Dolby Prologic
modes, except PANORAMA, where no other than the
loudspeaker output is active.
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MSP 3400C
PRELIMINARY DAT A SHEET
7.1.23. Subwoofer on Headphone Output
The subwoofer channel is created by combining the left
and right loudspeaker channels ( (L+R)/2 ) directly behind the tone control filter block. A third order lowpass filter with programmable corner frequency and volume adjustment respectively to the loudspeaker channel output
is performed to the bass-signal. Additionally , at the loudspeaker channels, a complementary high pass filter can
be switched on. The subwoofer channel output can be
switched to the headphone D/A converter alternatively
with the headphone output.
Subwoofer Channel
Volume Adjust
0 dB0000 000000hex
–1 dB1111 1111FFhex
–29 dB1110 001 1E3hex
–30 dB1110 0010E2hex
002ChexH
RESET
7.2. Exclusions
In general, all functions can be switched independently
of the others. One exception exists:
1. If the adaptive deemphasis is activated (Reg. 000f
L), the FM fixed deemphasis (Reg. 000f
set to 75 µs.
H) must be
hex
hex
Mute1000 000080hex
Subwoofer Channel
Corner Frequency
50 Hz .... 400 Hz
e.g. 50 Hz = 5 int
400 Hz = 40int
Headphone Output002Dhex[7:4]
Headphone0000 0hex
Subwoofer10008hex
Subwoofer: Complementary Highpass
HP off00000hex
HP on00011hex
Note: If subwoofer is chosen for headphone output, the
corner frequency must be set to the desired value, before the loudspeaker volume is set. This is to avoid plop
noise.
002DhexH
0000 010105hex
0010 100028hex
002Dhex[3:0]
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PRELIMINARY DAT A SHEET
7.3. Summary of Readable Registers
All readable registers are 16 bit wide. Transmissions via
I2C bus have to take place in 16 bit words. Single data
entries are 8 bit. Some of the defined 16 bit words are
divided into low and high byte, thus holding two different
control entities.
These registers are not writeable.
NameAddressHigh/LowOutput Range
MSP 3400C
Stereo detection register0018
Quasi peak readout left0019
Quasi peak readout right001a
DC level readout FM1/Ch2–L001b
DC level readout FM2/Ch1–R001c
MSP hardware version code001e
MSP major revision code
MSP product code001f
hex
hex
hex
hex
hex
hex
hex
H[80
H&L[00
H&L[00
H&L[00
H&L[00
H[00
L[00
H[00
MSP ROM version codeL[00
7.3.1. Stereo Detection Register
Stereo Detection
0018
hex
H
7.3.2. Quasi Peak Detector
Register
Stereo ModeReading
(two’s complement)
... 7F
hex
... 7FFF
hex
... 7FFF
hex
... 7FFF
hex
... 7FFF
hex
... FF
hex
... FF
hex
... 0A
hex
... FF
hex
]8 bit two’s complement
hex
hex
hex
hex
hex
]
hex
]
hex
]
hex
]
hex
Quasi peak readout
left
Quasi peak readout
right
]16 bit binary
]16 bit binary
]16 bit binary
]16 bit binary
0019
hex
001a
hex
H+L
H+L
MONOnear zero
Quasi peak readout[0
... 7FFF
hex
hex
]
values are 16 bit binary
STEREOpositive value (ideal
reception: 7F
BILINGUALnegative value (ideal
reception: 80
hex
hex)
)
The quasi peak readout register can be used to read out
the quasi peak level of any input source, in order to adjust all inputs to the same normalized listening level. The
refresh rate is 32 kHz. The feature is based on a filter
time constant:
attack-time: 1.3 ms
decay-time: 37 ms
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MSP 3400C
PRELIMINARY DAT A SHEET
7.3.3. DC Level Register
DC level readout FM1001b
DC level readout FM2001c
DC Level[0
hex
hex
hex
... 7FFF
hex
H+L
H+L
]
values are 16 bit binary
The DC level register measures the DC component of
the incoming FM signals (FM1 and FM2). This can be
used for seek functions in satellite receivers and for IF
FM frequencies fine tuning. For further processing, the
DC content of the demodulated FM signals is suppressed. The time constant τ, defining the transition time
of the DC Level Register, is approximately 28 ms.
A change in the hardware version code defines hardware optimizations that may have influence on the chip’s
behavior. The readout of this register is identical to the
hardware version code in the chip’s imprint.
7.3.5. MSP Major Revision Code
Major Revision001e
MSP 3400C03
hex
hex
L
The MSP 3400C is the third generation of ICs in the MSP
family .
A change in the ROM version code defines internal software optimizations, that may have influence on the
chip’s behavior, e.g. new features may have been included. While a software change is intended to create no
compatibility problems, customers that want to use the
new functions can identify new MSP 3400C versions according to this number. The readout of this register is
identical to the ROM version code in the chip’s imprint.
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PRELIMINARY DAT A SHEET
8. Specifications
8.1. Outline Dimensions
MSP 3400C
619
60
44
4327
+0.25
25
1
10
2
9
9
26
+0.25
25
Fig. 8–1:
68-Pin Plastic Leaded Chip Carrier Package
(PLCC68)
Weight approximately 4.8 g
Dimensions in mm
+0.2
x 45 °
1
1.9 1.5
4.05
±0.15
4.75
0.4570.2
0.711
0.9
23.4
0.1
2.4
1.27
2
0.1±
24.2
0.1±0.1±
16 x 1.27= 20.32
0.1±
15
0.1±
24.2
1.2 x 45°
2.4
0.1±
1.27
0.1±0.1±
16 x 1.27= 20.32
SPGS7004-3/4E
3364
1.9
3
±0.1
0.3
3.8
±0.4
4.8
0.3
2.5
132
±0.1
57.7
(1)
±0.4
1.29
3.2
1.778
31 x 1.778 = 55.118
±0.05
±0.1
1
0.457
±0.1
Fig. 8–2:
64-Pin Plastic Shrink Dual Inline Package
(PSDIP64)
Weight approximately 9.0 g
Dimensions in mm
0.27
19.3
18
±0.06
20.1
SPGS0016-4/2E
±0.1
±0.1
±0.6
2752
126
±0.1
47
±0.1
1
±0.05
1.778
25 x 1.778 = 44.47
0.457
±0.1
±0.2
0.4
±0.1
4
0.3
0.27
0.24
±0.2
3.2
Fig. 8–3:
52-Pin Plastic Shrink Dual In Line Package
(PSDIP52)
Weight approximately 5.5 g
Dimensions in mm
SPGS0015-1/2E
±0.1
15.6
±0.1
14
±0.06
0°...15°
MICRONAS INTERMETALL43
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MSP 3400C
PRELIMINARY DAT A SHEET
23 x 0.8 = 18.4
±0.03
4164
0.17
0.8
40
25
241
1.28
17.2
65
1.8
10.3
9.8
80
16
8
23.2
Fig. 8–4:
80-Pin Plastic Quad Flat Pack Package
(PQFP80)
Weight approximately 1.61 g
Dimensions in mm
8.2. Pin Connections and Short Descriptions
NC = not connected; leave vacant
LV = if not used, leave vacant
X = obligatory; connect as described
in circuit diagram
3
2.70
±0.2
1.8
14
0.1
8
5
20
AHVSS = connect to AHVSS
DVSS = if not used, connect to DVSS
–= pin does not exist in this package
0.8
15 x 0.8 = 12.0
SPGS0025-1/1E
Pin No.Pin NameTypeConnectionShort Description
PLCC
68-pin
116149S_ID
PSDIP
64-pin
PSDIP
52-pin
PQFP
80-pin
3410D in ( )(if not used)
OUTLVSBUS Ident or ADR
(ADR_WS)
wordstrobe
1)
2–––NCLVNot connected
315138S_DA_IN
(ADR_DA)
OUTLVSBUS Data input or ADR
data output
1)
414127I2S_DA_IN1INLVI2S1 data input
513116I2S_DA_OUTOUTLVI2S data output
612105I2S_WSIN/OUTLVI2S wordstrobe
71194I2S_CLIN/OUTLVI2S clock
81083I2C_DAIN/OUTXI2C data
9972I2C_CLIN/OUTXI2C clock
108–1NCLVNot connected
117680STANDBYQINXStandby (low-active)
126579ADR_SELINXI2C Bus address select
1)
Depending on MODE_REG[14], the SBUS Interface can be switched into ADR_MODE with S_CL
becoming ADR_CL, S_ID becoming ADR_WS and S_DA_IN becoming ADR_DA (see also section 4.5.).
2)
Due to compatibility with MSP 3410, it is possible to connect with DVSS as well.
tor)
24594868ANA_IN–INLVIF common
25584767ANA_IN1+INLVIF input 1
26574666AVSUPXAnalog power supply +5 V
–––65AVSUPXAnalog power supply +5 V
–––64NCLVNot connected
–––63NCLVNot connected
27564562AVSSXAnalog ground
–––61AVSSXAnalog ground
28554460MONO_ININLVMono input
–––59NCLVNot connected
29544358VREFTOPXReference voltage IF A/D
converter
30534257SC1_IN_RINLVScart input 1 in, right
31524156SC1_IN_LINLVScart input 1 in, left
3251–55ASG1AHVSSAnalog Shield Ground 1
33504054SC2_IN_RINLVScart input 2 in, right
34493953SC2_IN_LINLVScart input 2 in, left
1)
Depending on MODE_REG[14], the SBUS Interface can be switched into ADR_MODE with S_CL
becoming ADR_CL, S_ID becoming ADR_WS and S_DA_IN becoming ADR_DA (see also section 4.5.).
2)
Due to compatibility with MSP 3410, it is possible to connect with DVSS as well.
MICRONAS INTERMETALL45
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MSP 3400C
PRELIMINARY DAT A SHEET
Short DescriptionConnectionTypePin NamePin No.
PLCC
68-pin
PSDIP
64-pin
PSDIP
52-pin
80-pin
(if not used)3410D in ( )PQFP
3548–52ASG2AHVSSAnalog Shield Ground 2
36473851SC3_IN_RINLVScart input 3 in, right
37463750SC3_IN_LINLVScart input 3 in, left
3845–49NC (ASG4)LVNot connected
3944–48NC
LVNot connected
(SC4_IN_R)
4043–47NC
LVNot connected
(SC4_IN_L)
41––46NCLV or
Not connected
AHVSS
42423645AGNDCXAnalog reference voltage
high voltage part
43413544AHVSSXAnalog ground
–––43AHVSSXAnalog ground
–––42NCLVNot connected
–––41NCLVNot connected
44403440CAPL_MXVolume capacitor MAIN
45393339AHVSUPXAnalog power supply
8.0 V
46383238CAPL_AXVolume capacitor AUX
47373137SC1_OUT_LOUTLVScart output 1, left
48363036SC1_OUT_ROUTLVScart output 1, right
49352935VREF1XReference ground 1 high
voltage part
50342834SC2_OUT_LOUTLVScart output 2, left
51332733SC2_OUT_ROUTLVScart output 2, right
52––32ASG3AHVSS
2)
Analog Shield Ground 3
5332–31NCLVNot connected
54312630NC
LVNot connected
(DACM_SUB)
5530–29NCLVNot connected
56292528DACM_LOUTLVAnalog output MAIN, left
1)
Depending on MODE_REG[14], the SBUS Interface can be switched into ADR_MODE with S_CL
becoming ADR_CL, S_ID becoming ADR_WS and S_DA_IN becoming ADR_DA (see also section 4.5.).
2)
Due to compatibility with MSP 3410, it is possible to connect with DVSS as well.
MICRONAS INTERMETALL46
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PRELIMINARY DAT A SHEET
MSP 3400C
Short DescriptionConnectionTypePin NamePin No.
PLCC
68-pin
PSDIP
64-pin
PSDIP
52-pin
80-pin
(if not used)3410D in ( )PQFP
57282427DACM_ROUTLVAnalog output MAIN,
right
58272326VREF2XReference ground 2 high
voltage part
59262225DACA_LOUTLVAnalog output AUX, left
60252124DACA_ROUTLVAnalog output AUX, right
–––23NCLVNot connected
–––22NCLVNot connected
61242021RESETQINXPower-on-reset
6223–20NCLVNot connected
6322–19NCLVNot connected
64211918NCLVNot connected
65201817I2S_DA_IN2INLVI2S2-data input
66191716DVSSXDigital ground
–––15DVSSXDigital ground
–––14DVSSXDigital ground
67181613DVSUPXDigital power supply +5 V
–––12DVSUPXDigital power supply +5 V
–––11DVSUPXDigital power supply +5 V
68171510S_CL
(ADR_CL)
1)
Depending on MODE_REG[14], the SBUS Interface can be switched into ADR_MODE with S_CL
OUTLVSBUS clock or ADR
clock
1)
becoming ADR_CL, S_ID becoming ADR_WS and S_DA_IN becoming ADR_DA (see also section 4.5.).
2)
Due to compatibility with MSP 3410, it is possible to connect with DVSS as well.
Fig. 8–23: Output Pins 47, 48, 50 and 51
(SC_1/2_OUT_L/R)
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PRELIMINARY DAT A SHEET
MSP 3400C
8.5. Electrical Characteristics
8.5.1. Absolute Maximum Ratings
SymbolParameterPin NameMin.Max.Unit
T
A
T
S
V
SUP1
V
SUP2
V
SUP3
dV
P
TOT
V
Idig
I
Idig
V
Iana
I
Iana
I
Oana
I
Oana
SUP23
Ambient Operating Temperature–070°C
Storage Temperature––40125°C
First Supply VoltageAHVSUP–0.39.0V
Second Supply VoltageDVSUP–0.36.0V
Third Supply VoltageAVSUP–0.36.0V
Voltage between AVSUP
and DVSUP
Chip Power Dissipation
PLCC68 without Heat Spreader
Input Voltage, all Digital Inputs–0.3V
A VSUP,
–0.50.5V
DVSUP
AHVSUP,
DVSUP, AVSUP1100mW
+0.3V
SUP2
Input Current, all Digital Pins––20+20mA
Input Voltage, all Analog InputsSCn_IN_s,
2)
–0.3V
SUP1
+0.3V
MONO_IN
Input Current, all Analog InputsSCn_IN_s,
2)
–5+5mA
MONO_IN
Output Current, all SCART OutputsSCn_OUT_s
Output Current, all Analog Outputs
DACp_s
2)3), 4)3), 4)
2)3)3)
except SCART Outputs
1)
1)
I
Cana
1)
positive value means current flowing into the circuit
2)
“n” means “1”, “2” or “3”, “s” means “L” or “R”, “p” means “M” or “A”
3)
The Analog Outputs are short circuit proof with respect to First Supply Voltage and Ground.
4)
Total chip power dissipation must not exceed absolute maximum rating.
Output Current, other pins
connected to capacitors
CAPL_p,
AGNDC
2)
3)3)
Stresses beyond those listed in the “Absolute Maximum Ratings” may cause permanent damage to the device. This
is a stress rating only . Functional operation of the device at these or any other conditions beyond those indicated in the
“Recommended Operating Conditions/Characteristics” of this specification is not implied. Exposure to absolute maximum ratings conditions for extended periods may affect device reliability.
MICRONAS INTERMETALL53
Page 54
MSP 3400C
ADR_SEL
I2C_DA
I2C_DA
8.5.2. Recommended Operating Conditions
(at TA = 0 to 70 °C)
PRELIMINARY DAT A SHEET
SymbolParameter
V
V
V
V
V
t
REIL
SUP1
SUP2
SUP3
REIL
REIH
First Supply VoltageAHVSUP7.68.08.4V
Second Supply VoltageDVSUP4.755.05.25V
Third Supply VoltageAVSUP4.755.05.25V
RESET Input Low VoltageRESETQ0.45V
RESET Input High Voltage0.8V
RESET Low Time after DVSUP
“n” means “1”, “2” or “3”, “s” means “L” or “R”, “p” means “M” or “A”
2)
External capacitors at each crystal pin to ground are required. They are necessary to tune the open-loop fre-
Main/AUX Filter CapacitorDACM_s,
DACA_s
1)
–10%1+10%nF
quency of the internal PLL and to stabilize the frequency in closed-loop operation. The higher the capacitors,
the lower the clock frequency results. The nominal free running frequency should match 18.432 MHz as closely
as possible. Due to different layouts of customer PCBs, the matching capacitor size should be defined in the
application. The suggested values (1.5 pF/3.3 pF) are figures based on experience with various PCB layouts.
pp
RMS
RMS
MICRONAS INTERMETALL56
Page 57
PRELIMINARY DAT A SHEET
ANA_IN
MSP 3400C
ParameterSymbol
Recommendations for Analog Sound IF Input Signal
C
VREFTOP
VREFTOP-Filter-CapacitorVREFTOP–20%10µF
Ceramic Capacitor in Parallel–20%100nF
V
IF
Analog Input Range
(Complete Sound IF, 0 – 9 MHz)
R
FM
Ratio: FM-Main/FM-Sub
Satellite
R
FM1/FM2
Ratio: FM1/FM2
German FM-System
R
FC
Ratio: Main FM Carrier/Color
Carrier
R
FV
Ratio: Main FM Carrier/Luma
Components
PR
SUP
IF
HF
Passband Ripple––±2 dBdB
Suppression of Spectrum
Above 9.0 MHz
Pin Name
ANA_IN1+,
ANA_IN2+,
–
UnitMax.T yp.Min.
0.140.83Vpp
7dB
7dB
15––dB
15––dB
15–dB
FM
MAX
Maximum FM-Deviation (apprx.)
normal mode
high deviation mode
±192
±360
kHz
MICRONAS INTERMETALL57
Page 58
MSP 3400C
D_CTR_OUT1
PRELIMINARY DAT A SHEET
8.5.3. Characteristics at TA = 0 to 70 °C, f
= 18.432 MHz
CLOCK
(Typical values are measured at TA = 25 °C, AHVSUP = 8 V, DVSUP = 5 V, AVSUP = 5 V.)
Clock Input FrequencyXTAL_IN18.432MHz
Clock High to Low Ratio4555%
Clock Jitter (verification not
50ps
provided in production test)
DC-Voltage Oscillator2.5V
OscillatorStartup Timeat
VDD Slew-rate of 1 V / 1 µs
First Supply Current (active)
Analog Volume for Main and Aux at 0dB
Analog Volume for Main and Aux at –30dB
at Tj = 27 °C
XTAL_IN,
XTAL_OUT
AHVSUP
8.2
5.6
0.42.0ms
14.8
10.0
22.0
15.0mAmA
f = 18.432 MHz
AHVSUP = 8 V
DVSUP = 5 V
AVSUP = 5 V
Second Supply Current (active)DVSUP606570mAf = 18.432 MHz
DVSUP = 5 V
I
SUP3A
I
SUP1S
Third Supply Current (active)AVSUP 25mAf = 18.432 MHz
First Supply Current
(standby mode) at T
Audio Clock Output
V
APUAC
V
APUDC
Audio Clock Output AC VoltageAUD_CL_OUT1.2V
Audio Clock Output DC Voltage0.40.6V
Digital Output
V
DCTROL
V
DCTROH
Digital Output Low VoltageD_CTR_OUT0
Digital Output High Voltage
I2C Bus
V
IMOL
I
IMOH
t
IMOL1
t
IMOL2
I2C-Data Output Low VoltageI2C_DA0.4VI
I2C-Data Output High Current1µAV
I2C-Data Output Hold Time after
Falling Edge of Clock
I2C-Data Output Setup Time
before Rising Edge of Clock
= 27 °C
j
AVSUP = 5 V
AHVSUP2.85.07.2mAST ANDBYQ = low
VSUP = 8 V
40 pF load
= 1 mA
DDCTR
= –1 mA
DDCTR
= 3 mA
iMOL
= 5 V
IMOH
I2C_DA,
2
I
C_CL
pp
SUP1
0.4VI
4.0VI
15ns
100nsfIM = 1 MHz
DVSUP = 5 V
SBus
f
SB
t
S1/S2
t
S3
f
SIO
t
S6
SBUS-Clock FrequencyS_CL4608kHzDVSUP = 5 V
SBUS-Clock High/Low-Ratio0.91.01.1ns
SBUS Setup Time before
Ident End Pulse
S_CL,
S_ID
210nsDVSUP = 5.25 V
SBUS Ident frequencyS_ID32kHz
SBUS-Ident End Pulse Time210nsDVSUP = 5.25 V
MICRONAS INTERMETALL58
Page 59
PRELIMINARY DAT A SHEET
I2S_CL
1)
1)
→
I2S Bus
MSP 3400C
Test ConditionsUnitMax.Typ.Min.Pin NameParameterSymbol
V
I2SOL
V
I2SOH
f
I2SCL
f
I2SWS
t
I2S1/I2S2
t
I2S3
t
I2S4
t
I2S5
t
I2S6
Analog Ground
V
AGNDC0
R
outAGN
I2S Output Low VoltageI2S_WS,
I2S Output High Voltage
I2S_DA_OUT
,
4.0VI
0.4VI
I2SOL
I2SOH
= 1 mA
= –1 mA
I2S-Clock Output FrequencyI2S_CL1204kHzDVSUP = 5 V
I2S-Wordstrobe Output FrequencyI2S_WS32.0kHzDVSUP = 5 V
I2S-Clock High/Low-RatioI2S_CL0.91.01.1
I2S-Data Setup Time
before Rising Edge of Clock
I2S-Data Hold Time after Falling
I2S_CL,
I2S_DA_OUT
200nsDVSUP = 4.75 V
12nsDVSUP = 5.25 V
Edge of Clock
I2S-Wordstrobe Setup Time
before Rising Edge of Clock
I2S-Wordstrobe Hold Time after
I2S_CL,
I2S_WS
100nsDVSUP = 4.75 V
50nsDVSUP = 5.25 V
Falling Edge of Clock
AGNDC Open Circuit VoltageAGNDC3.643.733.84VR
AGNDC Output Resistance
at Tj = 27 °C
from T
= 0 to 70 °C
A
70
70
125180
180
kΩ
kΩ
load
3 V ≤ V
≥ 10 MΩ
AGNDC
≤ 4 V
Analog Input Resistance
R
inSC
R
inMONO
SCART Input Resistance
at Tj = 27 °C
from T
= 0 to 70 °C
A
MONO Input Resistance
at T
= 27 °C
j
from T
= 0 to 70 °C
A
Audio Analog-to-Digital-Converter
V
AICL
Analog Input Clipping Level for
Analog-to-Digital-Conversion
SCART Outputs
R
outSC
dV
OUTSC
A
SCtoSC
f
rSCtoSC
SCART Output Resistance
at T
= 27 °C
j
from T
= 0 to 70 °C
A
Deviation of DC-Level at SCART
Output from AGNDC Voltage
Gain from Analog Input to SCART
Output
Frequency Response from Analog
Input to SCART Output
bandwidth: 0 to 20000 Hz
SCn_IN_s
1)
MONO_IN
SCn_IN_s,
MONO_IN
SCn_OUT_s
SCn_IN_s
MONO_IN
SCn_OUT_s
f
= 1 kHz,
signal
25
25
10
10
1)
2.022.122.22V
0.20
0.20
4058
58
1623
23
0.330.46
0.5
kΩ
kΩ
kΩ
kΩ
RMS
kΩ
kΩ
I ≤ 0.05 mA
f
= 1 kHz,
signal
I ≤ 0.1 mA
f
= 1 kHz
signal
f
= 1 kHz, I = 0.1 mA
signal
–70+70mV
f
= 1kHz
–1.00+0.5dB
1)
signal
with respect to 1 kHz
–0.50+0.5dB
V
outSC
Signal Level at SCART-Output
during full-scale digital input signal
SCn_OUT_s
1)
1.81.92.0V
RMS
f
signal
= 1 kHz
from DSP
1)
“n” means “1”, “2” or “3”, “s” means “L” or “R”, “p” means “M” or “A”
MICRONAS INTERMETALL59
Page 60
MSP 3400C
1)
Main and AUX Outputs
PRELIMINARY DAT A SHEET
Test ConditionsUnitMax.Typ.Min.Pin NameParameterSymbol
R
outMA
V
outDCMA
Main/AUX Output Resistance
at T
= 27 °C
j
from T
= 0 to 70 °C
A
DC-Level at Main/AUX-Output
for Analog Volume at 0 dB
for Analog Volume at –30 dB
V
outMA
Signal Level at Main/AUX-Output
during full-scale digital input signal
from DSP for Analog Volume at
0 dB
Analog Performance
SNRSignal-to-Noise Ratio
from Analog Input to DSPMONO_IN,
from Analog Input to
SCART Output
from DSP to SCART OutputSCn_OUT_s
from DSP to Main/AUX-Output
for Analog Volume at 0 dB
for Analog Volume at –30 dB
DACp_s
SCn_IN_s
MONO_IN,
SCn_IN_s
1)
1)
→
SCn_OUT_s
DACp_s
1)
f
= 1 kHz, I = 0.1 mA
2.1
2.1
3.34.6
5.0
kΩ
kΩ
signal
1.74–1.94612.14–V
mV
1.231.371.51V
RMS
f
signal
= 1 kHz
8588dBInput Level = –20 dB with
resp. to V
kHz, equally weighted
20 Hz...16 kHz
9396dBInput Level = –20 dB,
f
= 1 kHz,
sig
1)
1)
8588dBInput Level = –20 dB,
equally weighted
20 Hz...20 kHz
f
= 1 kHz,
sig
equally weighted
20 Hz...15 kHz
Input Level = –20 dB,
f
= 1 kHz,
sig
85
78
88
83
dB
dB
equally weighted
20 Hz...15 kHz
AICL
, f
= 1
sig
2)
3)
3)
THDTotal Harmonic Distortion
from Analog Input to DSPMONO_IN,
SCn_IN_s
from Analog Input to
SCART Output
MONO_IN,
SCn_IN_s
1)
→
SCn_OUT_s
from DSP to SCART OutputSCn_OUT_s
from DSP to Main or AUX OutputDACA_s,
DACM_s
1)
“n” means “1”, “2” or “3”, “s” means “L” or “R”, “p” means “M” or “A”
2)
DSP measured at I2S-Output
3)
DSP Input at I2S-Input
1)
0.05%Input Level = –3 dBr with
resp. to V
equally weighted
20 Hz...16 kHz,
= 30 kΩ
R
Load
AICL
, f
=1kHz,
sig
2)
0.010.03%Input Level = –3 dBr,
f
= 1 kHz, equally
sig
1)
1)
0.010.03%Input Level = –3 dBr,
weighted 20 Hz...20 kHz,
= 30 kΩ
R
Load
f
= 1 kHz, equally
sig
weighted 20 Hz...16 kHz,
R
Load
= 30 kΩ
3)
0.010.03%Input Level = –3 dBr,
f
= 1 kHz, equally
sig
weighted 20 Hz...16 kHz,
R
Load
= 30 kΩ
3)
MICRONAS INTERMETALL60
Page 61
PRELIMINARY DAT A SHEET
20 Hz
kHz)
channel, effect on each
g
MSP 3400C
Test ConditionsUnitMax.Typ.Min.Pin NameParameterSymbol
XTALKCrosstalk attenuation
– PLCC68
– PSDIP64
between left and right channel within SCART Input/Output pair (L→R, R→L)
SCn_IN → SCn_OUT
SCn_IN → DSP
DSP → SCn_OUT
1)
1)
between left and right channel within Main or AUX
Output pair
DSP → DACp
1)
between SCART Input/Output pairs
D = disturbing program
O = observed program
D: MONO/SCn_IN → SCn_OUT PLCC68
O: MONO/SCn_IN → SCn_OUT
Input Level = –3 dB,
= 1 kHz, unused ana-
f
sig
log inputs connected to
ground by Z<1 kΩ
equally weighted
20 Hz...20 kHz
1)
PLCC68
PSDIP648080
PLCC68
PSDIP648080
PLCC68
PSDIP648080
dB
dB
dB
dB
dB
dB
2)
3)
equally weighted
20 Hz...16 kHz
3)
(equally weighted
1)
PLCC68
PSDIP648075
dB
dB
...20
same signal source on
left and right disturbing
1)
PSDIP64
100
100
dB
dB
observed output channel
D: MONO/SCn_IN → SCn_OUTPLCC68
O: or unsel. MONO/SCn_IN → DSP
Note: Pin numbers refer to PLCC packages, pin numbers for PSDIP packages in brackets.
not connected pins are 2,10,15,16,17,38,39,40,41,53,54,55,62,63,64 (2,3,8,21,22,23,31,32,43,44,45)
MICRONAS INTERMETALL64
Page 65
PRELIMINARY DAT A SHEET
MSP 3400C
10. DMA Application
Fig. 10–1 shows an example for the D2MAC application
with the MSP 3400 or MSP 3400C. T o obtain the optimal
amplitude and phase conditions for the clock input of
+ 5 Volt
S_DATA 66
S_IDENT 64
S_CLOCK 67
open
DMA 2381
Software:
SBS = 1
ACS = 1
ACF = 0
DCOF= 1
(addr. 204, 214)
ACLK
651716
AMU, DMA 2386, and DMA 2381, it is recommended to
use a clock inverter circuit, as shown below right, a minimum gain of 1.0 at 18.432 MHz and an output phase as
specified in Fig. 10–2.
5 K
9 S_DATA_IN
15 S_IDENT
8 S_CLOCK
AMU 2481
S_Bus
Slave_mode
S_DATA_OUT 6
13 AUDIO_CLOCK
18.432 MHz
1 nF
4.7 nF
656664
ACLKS_DATAS_IDENT
DMA 2386
Clock
Inverter
(see below)
+2...3 V
DMA 2381/86
and AMU 2481
68 S_CL
1 S_ID
MSP 3400C C6...
MSP 3410/00
TC15/F7
MODE_REG[0] = 1
18 AUD_CL_OUT
19 DMA_SYNC
To
3
S_DA_IN
Clock Inverter
+5 V
100 nF
1206k8
10 nF
BC 848C
3k882
Fig. 10–1: DMA application with MSP 3410 TC15 or F7
Note: Pin numbers refer to PLCC packages for DMA 2381 and MSP 3400C and to PSDIP package for AMU 2481
MICRONAS INTERMETALL65
Page 66
MSP 3400C
MSP Clock Output
PRELIMINARY DAT A SHEET
Clock Inverter Output
Timing window
typ. 20 ns
at inverter output
>10 ns
for the low to high edge at
pin 17 of DMA 2381 (XTAL2)
<42 ns
Fig. 10–2: Timing requirements for the clock signal at the DMA 2381 clock input
In the following table, the input/output clock-specification of the D2MAC circuit is shown.
Table 10–1: Clock input and output specification for MSPs
XTAL_IN min
MSP 3400C >C6
new Version
>0.7 Vpp
MSP 3410/00 TC27
new Version
>0.7 Vpp
(minimum amplitude)
C input
22 pF
22 pF
(after Reset)
MSP 3410/00 TC15
actual Version
>0.7 Vpp
31 pF
AUD_CL_OUTmin
with C load
Rout (HF) typ.
>1.2 Vpp
40 pF
150 Ω
>1.2 Vpp
40 pF
120 Ω
Table 10–2: Clock input and output specification for ICs connected to MSP
DMA 2381DMA 2386AMU2481
XTAL_IN min
>0.7 Vpp
>0.7 Vpp
Clock-inmin
(minimal amplitude)
C input
24 pF
7pF
10 pF with: Adr.
204,14=1
For the DMA_SYNC input specification of the MSP, please refer to page 54 “V
DMAIL
, V
>1.0 Vpp
43 pF
120 Ω
>0.7 Vpp
7pF
DMAIH
.”
MICRONAS INTERMETALL66
Page 67
PRELIMINARY DAT A SHEET
11. MSP Application with External Clock
If for some reason, e.g. to spare the cost of an additional
crystal, the MSP receives the 18.432 MHz clock from an
external source, for example from an other MSP , the following circuit can be used. For input/output specification
see also Table 10–1.
18.432 MHz
6362
MSP 3400C
MSP 3400C or
MSP 3410B
AUD_CL_OUT 18
Fig. 11–1: MSP 3400C with external clock
12. ADR Application
18.432 MHz
Tuner
(Sat)
MSP 3400C
(in I2S Slave Mode)
I2S_DA_OUT
S_CL
S_ID
S_DA_IN
2
I
S_CL
2
S_WS
I
I2S_DA_IN
10 nF
LV
ADR-Interface
2
S-Interface
I
62 XTAL_IN
63 XTAL_OUT
18.432 MHz
SI1C
SI1I
SI1D
PI16
PI15
SO1C
SO1I
SO1D
PI14
MSP 3400C
DRP 3510A
MICRONAS INTERMETALL67
Page 68
MSP 3400C
13. I2S Bus in Master/Slave Configuration with
Standby Mode
In a master/slave application, both MSP, after power up
and reset, will start as master by default. This means that
before the slave MSP is set to slave-mode, relatively
large current-pulses (~20 mA) in the I2S_CL and
I2S_WS lines can cause some crackling noise during
startup time, if the the MSP is demuted before the slave
MSP is set to slave mode.
These high current pulses are also possible, if the active
I2S_CL and I2S_WS outputs of the master MSP are
clipped by the correspondent inputs of the slave MSP,
which is switched to standby mode.
To avoid this, it is recommended, that the I2S-bus lines
I2S_CL and I2S_WS are current-limited to about 5 mA
with series resistors of about 390 Ω (330...470 Ω).
Fig. 13–1 depicts the recommended application circuit
for two MSP 3410/00 or MSP 3400C, which are connected via I2S Bus in a master/slave configuration, and
where the slave MSP can be switched in standby mode
(+5 Volt power is switched off).
PRELIMINARY DAT A SHEET
18.432 MHz
6362
MSP 3410/00
MSP 3400C
(master)
+5 V
I2S_DA_OUT 13
I2S_DA_IN 14
I2S_WS 12
I2S_CL 11
R
C
minimal corner frequency = 4 MHz
with R = 390 Ω (330–470 Ω)
Standby control
DVSUPSTANDBYQ
13 I2S_DA_OUT
14 I2S_DA_IN
12 I2S_WS
11 I2S_CL
718
MSP 3410/00
MSP 3400C
(slave)
18.432 MHz
6362
Fig. 13–1: I2S master/slave application
MICRONAS INTERMETALL68
Page 69
PRELIMINARY DAT A SHEET
MSP 3400C
14. APPENDIX A: Technical Code History
TC01
First Release, compatible with MSP3410 and
MSP 3400. Date: June 1994.
TC04
Emulator version for software development.
Version B5
6. Sub dB steps for Volume, Bass, Treble, Equalizer
15. APPENDIX B: Documentation History
1. Advance Information: “MSP 3400C Multistandard
Sound Processor”, Apr. 14, 1994, 6251-377-1AI.
First release of the advance information.
2. MSP 3400C Data Sheet: “MSP 3400C Multistandard
Sound Processor”, Dec. 14, 1994, 6251-377-1PD.
First release of the preliminary data sheet.
3. MSP 3400C Data Sheet: “MSP 3400C Multistandard
Sound Processor”, Oct. 6, 1996, 6251-377-2PD.
Second release of the preliminary data sheet.
Major changes: see Appendix A: Version C6
4. MSP 3400C Data Sheet: “MSP 3400C Multistandard
Sound Processor”, Dec. 8, 1997, 6251-377-3PD.
Third release of the preliminary data sheet.
Major changes: see Appendix A: Version C7 and C8
– new PQFP80 package
Version C7
New Features:
1. Balance, Bass, Treble and Loudness for Headphone
output
2. Prescale for I2S1 and I2S2 inputs
3. Balance in dB units and linear mode
4. SCART volume in dB units and linear mode
5. Increased range for Bass/Treble
Version C8
New Features:
1. Automatic Volume Control A.V.C.
2. Subwoofer Output alternatively with Headphone Output.
All information and data contained in this data sheet are without any commitment, are not to be considered as an offer for
conclusion of a contract nor shall they be construed as to
create any liability . Any new issue of this data sheet invalidates
previous issues. Product availability and delivery dates are exclusively subject to our respective order confirmation form; the
same applies to orders based on development samples delivered. By this publication, MICRONAS INTERMET ALL GmbH
does not assume responsibility for patent infringements or
other rights of third parties which may result from its use.
Reprinting is generally permitted, indicating the source. However, our prior consent must be obtained in all cases.
MICRONAS INTERMETALL72
Page 73
EndofDataSheet
MultimediaICs
MICRONAS
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