Datasheet MSP 3400 C Datasheet

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PRELIMINARY DATA SHEET
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MSP 3400 C Multistandard Sound Processor
Edition Dec. 8, 1997 6251-377-3PD
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MSP 3400C
Contents Page Section Title 5 1. Introduction 6 2. Features of the MSP 3400C
6 2.1. Features of the Demodulator and Decoder Sections 6 2.2. Features of the DSP-Section 6 2.3. Features of the Analog Section
7 3. Application Fields of the MSP 3400C
7 3.1. German 2-Carrier System (DUAL FM System)
9 4. Architecture of the MSP 3400C
9 4.1. Demodulator Block 9 4.1.1. Analog Sound IF – Input Section 9 4.1.2. Quadrature Mixers 10 4.1.3. Lowpass Filtering Block for Mixed Sound IF Signals 10 4.1.4. Phase and AM Discrimination 10 4.1.5. Differentiators 10 4.1.6. Lowpass Filter Block for Demodulated Signals 10 4.1.7. High Deviation FM Mode 10 4.1.8. MSPC-Mute Function in the Dual Carrier FM Mode 1 1 4.2. Analog Section and SCART Switching Facilities 1 1 4.3. MSP 3400C Audio Baseband Processing 11 4.3.1. Dual Carrier FM Stereo/Bilingual Detection 13 4.4. Audio PLL and Crystal Specifications 13 4.5. ADR Bus 14 4.6. S-Bus Interface 15 4.7. I
2
S Bus Interface
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16 5. I
17 5.1. Protocol Description 18 5.2. Proposal for MSP 3400C I2C Telegrams 18 5.2.1. Symbols 18 5.2.2. Write Telegrams 18 5.2.3. Read Telegrams 18 5.2.4. Examples 19 5.3. Start Up Sequence
20 6. Programming the Demodulator Part
20 6.1. Registers: Table and Addresses 21 6.2. Registers: Functions and Values 21 6.2.1. Setting of Parameter AD_CV 23 6.2.2. Control Register ‘MODE_REG’ 24 6.2.3. FIR-Filter Switches 24 6.2.4. FIR-Parameter 26 6.2.5. DCO-Increments
2
C Bus Interface: Device and Subaddresses
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Contents, continued Page Section Title
27 6.3. Sequences to Transmit Parameters and to Start Processing 27 6.4. Software Proposals for Multistandard TV-Sets 27 6.4.1. Multistandard System B/G German DUAL FM 28 6.4.2. Satellite Mode 28 6.4.3. Automatic Search Function for FM-Carrier Detection 28 6.4.4. Automatic Standard Detection
29 7. Programming the Audio Processing Part
29 7.1. Summary of the DSP Control Registers 31 7.1.1. Volume Loudspeaker Channel and Headphone Channel 32 7.1.2. Balance Loudspeaker and Headphone Channel 33 7.1.3. Bass Loudspeaker and Headphone Channel 33 7.1.4. Treble Loudspeaker and Headphone Channel 34 7.1.5. Loudness Loudspeaker and Headphone Channel 34 7.1.6. Spatial Effects Loudspeaker Channel 35 7.1.7. Volume SCAR T 35 7.1.8. Channel Source Modes 36 7.1.9. Channel Matrix Modes 36 7.1.10. SCART Prescale 36 7.1.11. FM Prescale 37 7.1.12. FM Matrix Modes 37 7.1.13. FM Fixed Deemphasis 37 7.1.14. FM Adaptive Deemphasis 37 7.1.15. I 37 7.1.16. ACB Register, Definition of the SCART-Switches and DIG_CTR_OUT Pins 38 7.1.17. Beeper 38 7.1.18. Identification Mode 38 7.1.19. FM DC Notch 38 7.1.20. Mode Tone Control 39 7.1.21. Equalizer Loudspeaker Channel 39 7.1.22. Automatic Volume Correction (AVC) 40 7.1.23. Subwoofer on Headphone Output 40 7.2. Exclusions 41 7.3. Summary of Readable Registers 41 7.3.1. Stereo Detection Register 41 7.3.2. Quasi Peak Detector 42 7.3.3. DC Level Register 42 7.3.4. MSP Hardware Version Code 42 7.3.5. MSP Major Revision Code 42 7.3.6. MSP Product Code 42 7.3.7. MSP ROM Version Code
2
S1 and I2S2 Prescale
MSP 3400C
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MSP 3400C
Contents, continued Page Section Title 43 8. Specifications
43 8.1. Outline Dimensions 44 8.2. Pin Connections and Descriptions 48 8.3. Pin Configuration 51 8.4. Pin Circuits 53 8.5. Electrical Characteristics 53 8.5.1. Absolute Maximum Ratings 54 8.5.2. Recommended Operating Conditions 58 8.5.3. Characteristics
64 9. Application of the MSP 3400C 65 10. DMA Application 67 11. MSP Application with External Clock
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67 12. ADR Application 68 13. I 69 14. APPENDIX A: T echnical Code History 69 15. APPENDIX B: Documentation History
2
S Bus in Master/Slave Configuration with Standby Mode
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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
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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 k
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 k – 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 k
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 k, 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.
Table 3–1: European TV standards
TV-System Position of Sound
Carrier /MHz
Sound Modulation
Color System Country
B/G 5.5/5.7421875 FM-Stereo PAL Germany B/G 5.5/5.85 FM-Mono/NICAM PAL Scandinavia,Spain L 6.5/5.85 AM-Mono/NICAM SECAM-L France I 6.0/6.552 FM-Mono/NICAM PAL UK D/K 6.5 /6.2578125 D/K1
FM-Stereo
SECAM-East USSR
6.5/6.7421875 D/K2
6.5/5.85 D/K-NICAM
FM-Mono/NICAM
Hungary
M M-Korea
Satellite Satellite
Tuner
4.5
4.5/4.724212
6.5
7.02/7.2
33 34 39 MHz 5 9 MHz
SAW Filter Sound IF Filter
Sound IF Mixer
Vision Demo­dulator
Composite Video
SCART Inputs
FM-Mono FM-Stereo
FM-Mono FM-Stereo
AM Sound
SCART1
SCART2
SCART3
NTSC USA
Korea
PAL PAL
MSP 3400C
2
2
2
2
2
SCART1
SCART2
Europe (ASTRA) Europe (ASTRA)
Loudspeaker
Headphone
SCART Outputs
According to the mixing characteristics of
I2S SBUS / ADR
I2S
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 Carriers Carrier FM1 Carrier FM2
B/G D/K M B/G D/K M Vision/sound power difference 13 dB 20 dB Sound bandwidth 40 Hz to 15 kHz Pre-emphasis 50 µs 75 µs 50 µs 75 µs Frequency deviation ±50 kHz ±25 kHz ±50 kHz ±25 kHz
Sound Signal Components
Mono transmission mono mono Stereo transmission (L+R)/2 (L+R)/2 R (L–R)/2 Dual sound transmission language A language B
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Identification of Transmission Mode on Carrier FM2
Pilot carrier frequency in kHz 54.6875 55.0699 Type of modulation AM Modulation depth 50% Modulation frequency mono: 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
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DCO1
VREFTOP
AD_CV[7:1]
ANA_IN1+
ANA_IN2+
AD_CV[8]
ANA_IN–
FRAME
FM2
DCO2
AGC AD
Pins
Internal signal lines
Control registers
Fig. 4–2: Demodulator architecture
Oscillator
FIR_REG_1
Mixer Lowpass
MSPC sound IF channel 1 (MSP-CH1: FM2)
MSPC sound IF channel 2 (MSP-CH2: FM1, AM)
Mixer Lowpass
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
Mixer IDENT
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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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.
Fig. 4–4: Stereo/bilingual detection
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MSP 3400C
r
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Analog Inputs
Demodulated IF Inputs
SBUS Inputs
2
I
S Bus
Inputs
SCARTL
SCARTR
FM1
FM2
SBUS1
SBUS2
SBUS3
SBUS4
2
S1L
I
I2S1R
2
I
S2L
2
S2R
I
Adaptive Deemphasis
DC level readout FM1
Deemphasis
50/75 µs
J17
DC level readout FM2
SCART
Prescale
FM
Prescale
I2S1
Prescale
I2S2
Prescale
FM-Matrix
Loudspeaker
Headphone
Channel Select
Quasi-Peak-
Channel
Matrix
Channel
Matrix
SCART
Channel
Matrix
I2S
Channel
Matrix
Channel
Matrix
AVC
Quasi-Peak
Detector
Bass/
Treble
or
Equalizer
Bass/
Treble
Quasi peak readout L
Quasi peak readout R
Beeper
Loudness
Loudness
Comple-
mentary
Highpass
Lowpass
Spatial Effects
Balance
Balance
Level
Adjust
Volume
Volume
Volume
Loudspeaker L
Loudspeaker R
Subwoofer
Headphone L
Headphone R
SCART_L
SCART_R
2
I
SL
I2SR
Loudspeake Outputs
Headphone Outputs
SCART Outputs
2
I
S
Outputs
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Fig. 4–5: Audio baseband processing (DSP-Firmware)
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MSP 3400C
Table 4–1: Several examples for recommended channel assignments for demodulator and audio processing part
Mode MSPC Sound IF-
Channel 1 / FM2
B/G-Stereo FM2 (5.74 MHz): R FM1 (5.5 MHz): (L+R)/2 B/G Stereo Speakers: FM Stereo B/G-Bilingual FM2 (5.74 MHz): Sound B FM1 (5.5 MHz): Sound A No Matrix Speakers: FM
Sat-Mono not used FM (6.5 MHz): mono No Matrix Speakers: FM Sound A Sat-Stereo 7.20 MHz: R 7.02 MHz: L No Matrix Speakers: FM Stereo Sat-Bilingual 7.38 MHz: Sound C 7.02 MHz: Sound A No Matrix Speakers: FM
Sat High Dev. Mode (e.g. EutelSat)
don’t care 6.552 MHz No Matrix Speakers: 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
– AUD_CL_OUT
2
–I
S_DA_IN1 or I2S_DA_IN2 –I2S_DA_OUT –I2S_WS –I2S_CLK – S_CL = ADR_CL – S_ID = ADR_WS – S_DA_IN = ADR_DA
MICRONAS INTERMETALL 13
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MSP 3400C
4.6. S-Bus Interface
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 2 16 Bit Sound 3 16 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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PRELIMINARY DAT A SHEET
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 Mode SONY 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 C Detail A,B
2
I
S_CL
2
I
S_WS as INPUT
I2S_WS as OUTPUT
Detail B
16 bit left channel 16 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_SEL Low High Left Open Mode Write Read Write Read Write Read
MSP device address 80 hex 81 hex 84 hex 85 hex 88 hex 89 hex
Table 5–2: I2C Bus Device and Subaddresses
Name Binary Value Hex Value Function
CONTROL 0000 0000 00 software reset TEST1 0000 0001 01 only for internal use TEST2 0000 0010 02 only for internal use WR_DEM 0001 0000 10 write address demodulator RD_DEM 0001 0001 11 read address demodulator WR_DFP 0001 0010 12 write address DFP RD_DFP 0001 0011 13 read address DFP AGC 0001 1110 1E read AGC RMS PLL_CAP 0001 1111 1F read / write PLL_Cap
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PRELIMINARY DAT A SHEET
Ç
Table 5–3: Control Register
Name 15 14..0
CONTROL RESET 0
5.1. Protocol Description
Write to DFP or Demodulator Part (long protocol)
MSP 3400C
S daw Wait ACK sub-addr ACK addr-byte
high
ACK addr-byte low ACK data-byte high ACK data-byte low ACK P
Read from DFP Part (long protocol)
S daw Wait ACK sub-addr ACK addr-byte
high
ACK addr-byte
low
ACK S dar Wait ACK data-byte
high
ACK data-byte
Ç
low
NAK P
Write to Control / Test / AGC / PLL_Cap Registers (short protocol)
S daw Wait ACK sub-addr ACK data-byte high ACK data-byte low ACK P
Read from Control / Test / AGC / PLL_Cap Registers (short protocol)
S daw Wait ACK sub-addr ACK S dar Wait ACK data-byte high
ACK data-byte low
NAK P
Note: S = I2C-Bus Start Condition from master
P = I2C-Bus Stop Condition from master daw = Device Address Write dar = Device Address Read ACK = Acknowledge-Bit: LOW on I2C_DA from slave (= MSPC, grey)
or master (= CCU, hatched)
NAK = Not Acknowledge-Bit: HIGH on I2C_DA from master (= CCU, hatched) to indicate
‘End of Read’ or from MSPC indicating internal error state (not illustrated)
Wait = I
2
C-Clock line held low by the slave (= MSPC) while interrupt is serviced (<1.77 ms)
I2C_DA
1
0
SP
I2C_CL
Fig. 5–1: I2C bus protocol
(MSB first; data must be stable while clock is high)
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MSP 3400C
PRELIMINARY DAT A SHEET
(Data: LSB first)
I2C_CL
T
I2C1
I2C_DA as input
2
C_DA as output
I
Fig. 5–2: I2C bus timing diagram
T
I2C5
T
IMOL2
T
I2C4
F
IM
T
I2C3
T
I2C6
T
IMOL1
T
I2C2
5.2. Proposal for MSP 3400C I2C Telegrams
5.2.1. Symbols
daw device address write dar device address read < Start Condition > Stop Condition aa Address Byte dd Data Byte
5.2.2.
Write Telegrams
<daw 00 dd dd> software RESET <daw 10 aa aa dd dd> write data into demodulator register <daw 12 aa aa dd dd> write data into DFP register
5.2.3. Read Telegrams
<daw 11 aa aa <dar dd dd> read data from demodulator <daw 13 aa aa <dar dd dd> read data from DSP
5.2.4. Examples
<daw 00 80 00> RESET MSPC statically <daw 00 00 00> clear RESET <daw 12 00 08 00 20> set loudspeaker channel source
to FM and Matrix to STEREO
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PRELIMINARY DAT A SHEET
5.3. Start Up Sequence
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
Register Protocol Write
Address (hex)
AD_CV long 00BB input selection, configuration of AGC and Mute Function,
MODE_REG long 0083 mode 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
short 1F switchable 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
Register Protocol Read
Address (hex)
PLL_CAP AGC_RMS C_AD_BITS long 0023 A 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)
short 1F switchable PLL capacities short 1E RMS 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 Range Meaning Settings
AD_CV [0] not used must 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 input 0 = ANALOG IN1
1 = ANALOG IN2
AD_CV [9] MSPC-Carrier-Mute Function 0 = off: no mute
1 = on: mute (see section 4.1.8.)
AD_CV [1 1–10] Programmable Carrier-Mute Level see Table 6–4 AD_CV [15–12] not used must be set to 0
Table 6–4: Carrier Mute Level
Step AD_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 values AD_CV [6:1] for active AGC (AD_CV[7] = 1)
Application Input Signal Contains Ref. Value
binary
Ref. Value decimal
Range of Input Signal at pin ANA_IN_1+ and ANA_IN_2+
Terrestrial TV 2 FM Carriers 101000 40 0.14 – 3 V SAT 1 or more FM Carriers 100011 35 0.14 – 3 V ADR 1 or more FM Carriers and
010100 20 0.14 – 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)
PRELIMINARY DAT A SHEET
Step AD_CV [6:1]
Gain Input Level at pin ANA_IN1+ and ANA_IN2+
Constant Gain
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20
1)
For signals above 1.4 Vpp, the minimum gain of 3 dB is switched and overflow of the AD converter may result. Due
000000 000001 000010 000011 000100 000101 000110 000111 001000 001001 001010 001011 001100 001101 001110 001111 010000 010001 010010 010011 010100
3.00 dB
3.85 dB
4.70 dB
5.55 dB
6.40 dB
7.25 dB
8.10 dB
8.95 dB
9.80 dB
10.65 dB
11.50 dB
12.35 dB
13.20 dB
14.05 dB
14.90 dB
15.75 dB
16.60 dB
17.45 dB
18.30 dB
19.15 dB
20.00 dB
maximum input level1): 3 Vpp (FM) or 1.4 V
maximum input level: 0.14 V
pp
1)
pp
(AM)
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
Bit Function Comment Definition Recom-
mendation
[0] DMA_SYNC
1)
Synchronization to DMA 0 : off
X
1 : on
[1] DCTR_TRI Digital control out 0/1 tristate 0 : active
0
1 : tristate
[2] I2S_TRI I2S 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 Mode WS due to the Sony or
Philips-Format
[5] Audio_CL_OUT switch 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 used must be 0 0 [7] FM1 FM2 MSPC-channel 1 mode s.Table 6–8 [8] AM MSPC-channel 1/2 mode 0 : FM
s.Table 6–8
1 : AM
[9] HDEV High Deviation Mode
(channel matrix must be
0 : normal mode 1 : high deviation mode
s.Table 6–8
sound A ) [10] not used must be 1 1 [11] S-Bus Mode
2)
mode of Pins S_CL and S_ID 0 : Tristate
0
1 : Active
[12] FM2 FIR Filter Gain
(FM2 = Ch1)
[13] FM2 FIR Filter Coeff. Set
(FM2 = Ch1)
[14] ADR Mode of ADR Interface 0 : normal mode
see table 6–10 0 : Gain = 6 dB
1 : Gain = 0 dB
see table 6–10 0 : use FIR_REG_1
1 : use FIR_REG_2
0
0
X
1 : ADR mode
[15] AM-Gain additional gain in AM-mode 0 : 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.
X:Depend­ing on mode
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MSP 3400C
Table 6–8: Channel modes ‘MODE_REG [7–9]‘
PRELIMINARY DAT A SHEET
FM1 FM2 bit[7]
AM bit[8]
HDEV bit[9]
channel 1 channel 2
0 0 0 mute FM-Mono (FM1) 1 0 0 FM2 FM1 X 1 0 AM AM X X 1 FM-Mono (high deviation) FM-Mono (high deviation)
6.2.3. FIR-Filter Switches
To simplify programming of the MSP 3400C, two addi­tional switches have been implemented.
Table 6–9: Loading sequence for FIR-coefficients
WRITE_ADR = FIR_REG_1(Channel 1: FM2) No. Symbol Name Bits Value
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-
1 FM2_Coeff. (5) 8 see Table 6–10. 2 FM2_Coeff. (4) 8
3 FM2_Coeff. (3) 8
cient.
4 FM2_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
5 FM2_Coeff. (1) 8 6 FM2_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 Name Bits Value
The following data values (see Table 6–9) are to be transferred 8 bits at a time embedded LSB-bound in a 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-
1 * IMREG1 (8 LSBS) 8 04 HEX 2 * IMREG1 / IMREG2
(4 MSBs / 4 LSBs)
3 * IMREG2 (8 MSBs) 8 00 HEX
8 40 HEX
tine.
4 FM_Coef (5) 8 see Table 6–10. 5 FM_Coef (4) 8
6 FM_Coef (3) 8 7 FM_Coef (2) 8 8 FM_Coef (1) 8 9 FM_Coef (0) 8 * IMREG_1/2: Two 12-bit off-set constants
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PRELIMINARY DAT A SHEET
MSP 3400C
Table 6–10: 8-bit FIR-coefficients (decimal integer) for MSP 3410D; reset status: all coefficients are “0”
Coefficients for FIR1 0001
Terrestrial TV-Standards FM - Satellite
B/G-,D/K-,M-Dual FM 130 kHz 180 kHz 200 kHz 280 kHz 380 kHz 500 kHz Autosearch
Coef(i) FIR2 FIR2 FIR2 FIR2 FIR2 FIR2 FIR2 FIR2 0 3 73 9 3 –8 –1 –1 75 1 18 53 18 18 –8 –9 –1 19 2 27 64 28 27 4 –16
3 48 119 47 48 6 5 2 35 4 66 101 55 66 78 65 59 39 5 72 127 64 72 107 123 126 40 MODE-REG[12] 0 1 1 1 1 1 1 0 MODE-REG[13] 1 1 1 1 1 1 1 0
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.
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Page 26
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. MHz DCO_HI DCO_LO Frq. MHz DCO_HI DCO_LO
4.5 03E8 0000
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
7.02 0618 0000 7.2 0640 0000
7.38 0668 0000 7.56 0690 0000
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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
MICRONAS INTERMETALL 27
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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.).
MICRONAS INTERMETALL28
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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
Name I2C Bus
Address
Volume loudspeaker channel 0000 Volume / Mode loudspeaker channel L 1/8 dB Steps, Reduce Volume / Tone Control 00 Balance loudspeaker channel [L/R] 0001
High/
Adjustable Range, Operational Modes Reset Mode
Low
H [+12 dB ... –114 dB, MUTE] MUTE
hex
H [0..100 / 100 % and vv][–127..0 / 0 dB and vv]
hex
hex
100%/100%
Balance Mode loudspeaker L [Linear mode / logarithmic mode] linear mode Bass loudspeaker channel 0002 Treble loudspeaker channel 0003 Loudness loudspeaker channel 0004 Loudness Filter Characteristic Spatial effect strength loudspeaker ch. 0005
H [+20 dB ... –12 dB] 0 dB
hex
H [+15 dB ... –12 dB] 0 dB
hex
H [0 dB ... +17 dB] 0 dB
hex
L [NORMAL, SUPER_BASS] NORMAL H [–100%...OFF...+100%] OFF
hex
Spatial effect mode/customize L [SBE, SBE+PSE] SBE+PSE Volume headphone channel 0006 Volume / Mode headphone channel L 1/8 dB Steps, Reduce Volume / Tone Control 00 Volume SCAR T channel 0007
H [+12 dB ... –114 dB, MUTE] MUTE
hex
hex
H [00
hex
... 7F
],[+12 dB ... –1 14 dB, MUTE] 00
hex
hex
hex
Volume / Mode SCAR T channel L [Linear mode / logarithmic mode] linear mode Loudspeaker channel source 0008
H [FM, NICAM, SCART, I2S1, I2S2] FM
hex
Loudspeaker channel matrix L [SOUNDA, SOUNDB, STEREO, MONO...] SOUNDA Headphone channel source 0009
H [FM, NICAM, SCART, I2S1, I2S2] FM
hex
Headphone channel matrix L [SOUNDA, SOUNDB, STEREO, MONO...] SOUNDA SCART1 channel source 000a
H [FM, NICAM, SCART, I2S1, I2S2] FM
hex
SCART1 channel matrix L [SOUNDA, SOUNDB, STEREO, MONO...] SOUNDA I2S channel source 000b
H [FM, NICAM, SCART, I2S1, I2S2] FM
hex
I2S channel matrix L [SOUNDA, SOUNDB, STEREO, MONO...] SOUNDA Quasi-peak detector source
000c
hex
H
[FM, NICAM, SCART, I2S1, I2S2]
FM Quasi-peak detector matrix L [SOUNDA, SOUNDB, STEREO, MONO...] SOUNDA Prescale SCART 000d Prescale FM 000e
hex
hex
H [00 H [00
hex
hex
... 7F ... 7F
] 00
hex
] 00
hex
hex
hex
FM matrix L [NO_MAT, GSTEREO, KSTEREO] NO_MAT
MICRONAS INTERMETALL 29
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MSP 3400C
PRELIMINARY DAT A SHEET
Name
I2C Bus Address
Deemphasis FM 000f
hex
Low
H [OFF , 50 µs, 75 µs, J17] 50 µs
Reset ModeAdjustable Range, Operational ModesHigh/
Adaptive Deemphasis FM L [OFF, WP1] OFF Prescale I2S2 0012 ACB Register (SCART Switches and
0013
H [00
hex
H/L Bits [15..0] 00
hex
hex
... 7F
] 10
hex
DIG_OUT Pins) Beeper 0014 Identification Mode 0015 Prescale I2S1 0016 FM DC Notch 0017 Mode Tone Control 0020 Equalizer loudspeaker ch. band 1 0021 Equalizer loudspeaker ch. band 2 0022 Equalizer loudspeaker ch. band 3 0023 Equalizer loudspeaker ch. band 4 0024 Equalizer loudspeaker ch. band 5 0025
H/L [00
hex
L [B/G, M] B/G
hex
H [00
hex
L [ON, OFF] ON
hex
H [BASS/TREBLE, EQUALIZER] BASS/TREB
hex
H [+12 dB ... –12 dB] 0 dB
hex
H [+12 dB ... –12 dB] 0 dB
hex
H [+12 dB ... –12 dB] 0 dB
hex
H [+12 dB ... –12 dB] 0 dB
hex
H [+12 dB ... –12 dB] 0 dB
hex
hex
hex
... 7F
... 7F
]/[00
hex
hex
... 7F
hex
] 10
] 0/0
hex
hex
hex
hex
Automatic Volume Correction 0029
H [off, on, decay time] off
hex
Volume Subwoofer channel 002Chex H [0dB ... –30 dB, mute] 0 dB Subwoofer Channel Corner Frequency 002Dhex H [50 Hz ... 400 Hz] Subwoofer: Complementary Highpass L [off, on] off Balance headphone channel [L/R] 0030
H [0...100 / 100% and vv][–127...0 / 0 dB and vv]
hex
100%/100%
Balance Mode headphone L [Linear mode / logarithmic mode] linear mode Bass headphone channel 0031 Treble headphone channel 0032 Loudness headphone channel 0033
H [+20 dB ... –12 dB] 0 dB
hex
H [+15 dB ... –12 dB] 0 dB
hex
H [0 dB ... +17 dB] 0 dB
hex
Loudness filter characteristic L [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.
MICRONAS INTERMETALL30
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PRELIMINARY DAT A SHEET
MSP 3400C
7.1.1. Volume Loudspeaker Channel and Head­phone Channel
Volume
0000
hex
11 MSBs
loudspeaker Volume
0006
hex
11 MSBs
headphone
+12 dB 0111 1111 000x 7F0 +11.875 dB 0111 1110 111x 7EE
+0.125 dB 0111 0011 001x 732 0 dB 0111 0011 000x 730 –0.125 dB 0111 0010 1 11x 72E
–113.875dB 0000 0001 001x 012 –114 dB 0000 0001 000x 010 Mute 0000 0000 xxxx 00x
hex
hex
hex
hex
hex
hex
hex
hex
RESET
Fast Mute 1111 1111 111x FFE
hex
Clipping Mode
0000
hex
3 LSBs
loudspeaker Clipping Mode
0006
hex
3 LSBs
headphone
Reduce Volume x000 0
hex
RESET Reduce Tone Control x001 1 Compromise Mode x010 2
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. Volume 3 9 5 Red. Tone Con. 6 6 5 Compromise 4.5 7.5 5
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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
linear xxx0 0
hex
RESET
logarithmic xxx1 1
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 dB 0111 1111 7F Left –126 dB, Right 0 dB 0111 1110 7E
Left –1 dB, Right 0 dB 0000 0001 01 Left 0 dB, Right 0 dB 0000 0000 00
RESET
Left 0 dB, Right –1 dB 1111 1111 FF Left 0 dB, Right –127 dB 1000 0001 81
Left 0 dB, Right –128 dB 1000 0000 80
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 1111 7F Left 0.8%, Right 100% 0111 1110 7E
Left 99.2%, Right 100% 0000 0001 01 Left 100%, Right 100% 0000 0000 00
RESET
Left 100%, Right 99.2% 1111 1111 FF Left 100%, Right 0.8% 1000 0010 82
Left 100%, Right muted 1000 0001 81
hex
hex
hex
hex
hex
hex
hex
MICRONAS INTERMETALL32
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PRELIMINARY DAT A SHEET
MSP 3400C
7.1.3. Bass Loudspeaker and Headphone Channel
Bass loudspeaker 0002 Bass headphone 0031
hex
hex
+20 dB 0111 1111 7F +18 dB 0111 1000 78 +16 dB 0111 0000 70 +14 dB 0110 1000 68 +12 dB 0110 0000 60 +11 dB 0101 1000 58
+1 dB 0000 1000 08 +1/8 dB 0000 0001 01
0 dB 0000 0000 00
H H
hex
hex
hex
hex
hex
hex
hex
hex
hex
RESET
–1/8 dB 1111 1111 FF
hex
7.1.4. Tr e bl e Loudspeaker and Headphone Channel
Treble loudspeaker 0003 Treble headphone 0032
hex
hex
+15 dB 0111 1000 78 +14 dB 0111 0000 70
+1 dB 0000 1000 08 +1/8 dB 0000 0001 01
0 dB 0000 0000 00
H H
hex
hex
hex
hex
hex
RESET
–1/8 dB 1111 1111 FF –1 dB 1111 1000 F8 –11 dB 1010 1000 A8
–12 dB 1010 0000 A0
hex
hex
hex
hex
–1 dB 1111 1000 F8 –11 dB 1010 1000 A8
–12 dB 1010 0000 A0
hex
hex
hex
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.
MICRONAS INTERMETALL 33
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MSP 3400C
PRELIMINARY DAT A SHEET
7.1.5. Loudness Loudspeaker and Headphone Channel
Loudness
0004
hex
H
loudspeaker Loudness
0033
hex
H
headphone
+17 dB 0100 0100 44 +16 dB 0100 0000 40
+1 dB 0000 0100 04 0 dB 0000 0000 00
hex
hex
hex
hex
RESET
Mode Loudness
0004
hex
L
loudspeaker Mode Loudness
0033
hex
L
headphone
Normal (constant volume at 1 kHz)
Super Bass (constant
0000 0000 00 RESET
0000 0100 04
hex
hex
volume at 2 kHz)
7.1.6. Spatial Effects Loudspeaker Channel
Spatial effect strength
0005
hex
loudspeaker channel
Enlargement 100% 0111 1111 7F Enlargement 50% 0011 1111 3F
Enlargement 1.5% 0000 0001 01 Effect off 0000 0000 00
RESET
Reduction 1.5% 1111 1111 FF Reduction 50% 1100 0000 C0 Reduction 100% 1000 0000 80
Spatial Effect Mode 0005
Stereo Basewidth En­largement (SBE) and Pseudo Stereo Effect
hex
0000 0 RESET 0000 0
(PSE). (Mode A) Stereo Basewidth En-
0010 2 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 gain 0000 0
hex
RESET 2/3 high pass gain 0010 2 1/3 high pass gain 0100 4 min high pass gain 0110 6 automatic 1000 8
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.
MICRONAS INTERMETALL34
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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.
7.1.7. Volume SCART
Volume Mode SCART 0007
hex
linear xxx0 0
LSB
hex
RESET
logarithmic xxx1 1
hex
Linear Mode V olume SCART 0007
hex
H
7.1.8. Channel Source Modes
Loudspeaker channel
0008
hex
source Headphone channel
0009
hex
source SCART channel
000a
hex
source I2S channel source 000b Quasi-peak detector
000c
hex
hex
source
FM 0000 0000 00
RESET
NONE
(MSP3410: NICAM)
0000 0001 01 SCART 0000 0010 02 SBUS12 0000 0011 03 SBUS34 0000 0100 04
H
H
H
H H
hex
hex
hex
hex
hex
OFF 0000 0000 00
RESET
0 dB gain
0100 0000 40 (digital full scale (FS) to 2 V
+6 dB gain (–6 dBFS to 2 V
RMS
RMS
output)
0111 1111 7F
output)
Logarithmic Mode V olume SCART 0007
hex
11 MSBs
+12 dB 0111 1111 000x 7F0 +11.875 dB 0111 1110 111x 7EE
+0.125 dB 0111 0011 001x 732 0 dB 0111 0011 000x 730 –0.125 dB 0111 0010 111x 72E
–113.875 dB 0000 0001 001x 012
hex
hex
hex
hex
hex
hex
hex
hex
hex
I2S1 0000 0101 05 I2S2 0000 0110 06
hex
hex
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.
–114 dB 0000 0001 000x 010 Mute 0000 0000 0000 000
hex
hex
RESET
MICRONAS INTERMETALL 35
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MSP 3400C
PRELIMINARY DAT A SHEET
7.1.9. Channel Matrix Modes (see also Table 4–1)
Loudspeaker channel
0008
hex
L
matrix Headphone channel
0009
hex
L
matrix SCART channel
000a
hex
L
matrix I2S channel matrix 000b Quasi-peak detector-
000c
hex
hex
L L
matrix
SOUNDA / LEFT / MSP-IF-CHANNEL2
SOUNDB / RIGHT /
0000 0000 00 RESET
0001 0000 10
hex
hex
MSP-IF-CHANNEL1 STEREO 0010 0000 20 MONO 001 1 0000 30 SUM/DIFF 0100 0000 40 AB_XCHANGE 0101 0000 50 INVERT_B 0110 0000 60
hex
hex
hex
hex
hex
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)
OFF 0000 0000 00
RESET
Maximum Volume (28 kHz deviation
1)
0111 1111 7F
recommended FIR­bandwidth: 130 kHz)
Deviation 50 kHz
1)
0100 1000 48 recommended FIR­bandwidth: 200 kHz
Deviation 75 kHz
1)
0011 0000 30 recommended FIR­bandwidth: 200 or 280 kHz
Deviation 150 kHz
1)
0001 1000 18 recommended FIR­bandwidth: 380 kHz
Maximum deviation
0001 0011 13 192 kHz1) recommended FIR­bandwidth: 380 kHz
Prescale for adaptive
0001 0000 10 deemphasis WP1 recommended FIR­bandwidth: 130 kHz
Volume Prescale FM
000e
hex
(High Deviation Mode)
Deviation 150 kHz
1)
0011 0000 30 recommended FIR­bandwidth: 380 kHz
H
hex
hex
hex
hex
hex
hex
hex
H
hex
V olume Prescale
000d
hex
H
SCART
OFF 0000 0000 00
RESET
0 dB gain (2 V
RMS
in-
0001 1001 19
put to digital full scale) +14 dB gain
(400 mV
RMS
input to
0111 1111 7F
digital full scale)
hex
hex
hex
Maximum deviation
0001 0011 13
hex
384 kHz1) recommended FIR­bandwidth: 500 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.
MICRONAS INTERMETALL36
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PRELIMINARY DAT A SHEET
MSP 3400C
7.1.12. FM Matrix Modes (see also Table 4–1)
FM matrix 000e
hex
NO MATRIX 0000 0000 00
L
hex
RESET GSTEREO 0000 0001 01 KSTEREO 0000 0010 02
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).
7.1.13. FM Fixed Deemphasis
Deemphasis FM 000f
hex
50 µs 0000 0000 00
H
hex
RESET 75 µs 0000 0001 01 J17 0000 0100 04 OFF 0011 1111 3F
hex
hex
hex
7.1.15. I
2
S1 and I2S2 Prescale
Prescale I2S1 0016 Prescale I2S2 0012
OFF 00 0 dB gain 10
hex
hex
hex
hex
H H
RESET
+18 dB gain 7F
hex
7.1.16. ACB Register, Definition of the SCART­Switches and DIG_CTR_OUT Pins
ACB Register 0013
hex
H
DSP In
Selection of Source:
SC_1_IN MONO_IN SC_2_IN SC_3_IN
xxxx xx00 RESET 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 00xx RESET xxxx 01xx xxxx 10xx xxxx 11xx
7.1.14. FM Adaptive Deemphasis
FM Adaptive
000f
hex
L
Deemphasis WP1
OFF 0000 0000 00
hex
RESET WP1 0011 1111 3F
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 xxxx RESET xx01 xxxx xx10 xxxx
DIG_CTR_OUT1
low high
x0xx xxxx RESET x1xx xxxx
DIG_CTR_OUT2
low high
0xxx xxxx RESET 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.
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MSP 3400C
PRELIMINARY DAT A SHEET
7.1.17. Beeper
Beeper V olume 0014
hex
OFF 0000 0000 00
H
hex
RESET
Maximum Volume (full
0111 1111 7F
hex
digital scale FDS)
Beeper Frequency 0014
hex
16 Hz (lowest) 0000 0001 01 1 kHz 0100 0000 40 4 kHz (highest) 1111 1111 FF
L
hex
hex
hex
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 Mode 0015
Standard B/G (German Stereo)
0000 0000 00 RESET
hex
L
hex
7.1.19. FM DC Notch
FM DC Notch 0017
hex
ON 0000 0000 00
L
hex
Reset
OFF 0011 1111 3F
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 Control 00020
hex
Bass and Treble 0000 0000 00
H
hex
RESET
Equalizer 1111 1111 FF
hex
By means of ‘Mode T one Control’, Bass/Treble or Equal­izer may be activated.
Standard M (Korean
0000 0001 01
hex
Stereo) Reset of Ident-Filter 0011 1111 3F
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
MICRONAS INTERMETALL38
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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 dB 0110 0000 60 +11 dB 0101 1000 58
+1 dB 0000 1000 08 +1/8 dB 0000 0001 01
0 dB 0000 0000 00
RESET
–1/8 dB 1111 1111 FF –1 dB 1111 1000 F8
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
–11dB 1010 1000 A8 –12 dB 1010 0000 A0
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).
7.1.22. Automatic Volume Correction (AVC)
AVC on/off 0029hex [15:12]
AVC off and Reset
of int. variables
0000 0hex
RESET AVC on 1000 8hex
AVC Decay Time 0029hex [1 1:8]
8 sec (long) 4 sec (middle) 2 sec (short) 20 ms (very short)
1000 8hex
0100 4hex
0010 2hex
0001 1hex
Scart Input 0dbr = 2 Vrms
Volume Correc-
Main Output 0dBr = 1.4 Vrms
tion
0 dBr –18 dB –18 dBr –6 dBr –12 dB –18 dBr –12 dBr –6 dB –18 dBr –18 dBr –0 dB –18 dBr –24 dBr + 6 dB –18 dBr –30 dBr + 6 dB –24 dBr Loudspeaker Volume = 73h = 0 dBFS
Scart Prescale = 20h i.e. 2.0 Vrms = 0dBFS
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 dB 0000 0000 00hex
–1 dB 1111 1111 FFhex –29 dB 1110 001 1 E3hex
–30 dB 1110 0010 E2hex
002Chex H
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
Mute 1000 0000 80hex
Subwoofer Channel Corner Frequency
50 Hz .... 400 Hz
e.g. 50 Hz = 5 int
400 Hz = 40int
Headphone Output 002Dhex [7:4]
Headphone 0000 0hex Subwoofer 1000 8hex
Subwoofer: Comple­mentary Highpass
HP off 0000 0hex HP on 0001 1hex
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.
002Dhex H
0000 0101 05hex 0010 1000 28hex
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.
Name Address High/Low Output Range
MSP 3400C
Stereo detection register 0018 Quasi peak readout left 0019 Quasi peak readout right 001a DC level readout FM1/Ch2–L 001b DC level readout FM2/Ch1–R 001c MSP hardware version code 001e MSP major revision code MSP product code 001f
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 code L [00
7.3.1. Stereo Detection Register
Stereo Detection
0018
hex
H
7.3.2. Quasi Peak Detector
Register
Stereo Mode Reading
(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
MONO near zero
Quasi peak readout [0
... 7FFF
hex
hex
]
values are 16 bit binary
STEREO positive value (ideal
reception: 7F BILINGUAL negative 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 FM1 001b DC level readout FM2 001c
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.
7.3.4. MSP Hardware Version Code
Hardware Version 001e
Hardware Version [00 MSP 3400C – C8 03
hex
hex
hex
... FF
hex
H
]
7.3.6. MSP Product Code
Product 001f
hex
MSP 3400C 0000 0000 00 MSP 3400 0000 1010 0A MSP 3410 0000 1010 0A
1)
Note: The MSP 3400 hardware is identical to the
H
hex
1)
hex
hex
MSP 3410. Therefore, the family code readout will show ‘MSP 3410’ instead of its label ‘MSP 3400’.
7.3.7. MSP ROM Version Code
ROM Version 001f
Major software revision [00
hex
hex
... FF MSP 3400C – B5 0000 0101 05 MSP 3400C – C6 0000 0110 06 MSP 3400C – C8 0000 1000 08
hex
L
]
hex
hex
hex
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 Revision 001e
MSP 3400C 03
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 INTERMETALL 43
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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 Name Type Connection Short Description
PLCC 68-pin
1 16 14 9 S_ID
PSDIP 64-pin
PSDIP 52-pin
PQFP 80-pin
3410D in ( ) (if not used)
OUT LV SBUS Ident or ADR
(ADR_WS)
wordstrobe
1)
2 NC LV Not connected 3 15 13 8 S_DA_IN
(ADR_DA)
OUT LV SBUS Data input or ADR
data output
1)
4 14 12 7 I2S_DA_IN1 IN LV I2S1 data input 5 13 11 6 I2S_DA_OUT OUT LV I2S data output 6 12 10 5 I2S_WS IN/OUT LV I2S wordstrobe 7 11 9 4 I2S_CL IN/OUT LV I2S clock 8 10 8 3 I2C_DA IN/OUT X I2C data 9 9 7 2 I2C_CL IN/OUT X I2C clock 10 8 1 NC LV Not connected 11 7 6 80 STANDBYQ IN X Standby (low-active) 12 6 5 79 ADR_SEL IN X I2C 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.
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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
13 5 4 78 D_CTR_OUT0 OUT LV Digital control output 0 14 4 3 77 D_CTR_OUT1 OUT LV Digital control output 1 15 3 76 NC LV Not connected 16 2 NC LV Not connected 17 75 NC LV Not connected 18 1 2 74 AUD_CL_OUT OUT LV Audio clock output 19 64 1 73 DMA_SYNC IN LV DMA-Sync. Input 20 63 52 72 XT AL_OUT OUT X Crystal oscillator 21 62 51 71 XTAL_IN IN X Crystal oscillator 22 61 50 70 TESTEN IN X Test pin 23 60 49 69 ANA_IN2+ IN LV IF input 2 (if ANA_IN1+ is
used only, connect to AVSS with 50 pF capaci-
tor) 24 59 48 68 ANA_IN– IN LV IF common 25 58 47 67 ANA_IN1+ IN LV IF input 1 26 57 46 66 AVSUP X Analog power supply +5 V – 65 AVSUP X Analog power supply +5 V – 64 NC LV Not connected – 63 NC LV Not connected 27 56 45 62 AVSS X Analog ground – 61 AVSS X Analog ground 28 55 44 60 MONO_IN IN LV Mono input – 59 NC LV Not connected 29 54 43 58 VREFTOP X Reference voltage IF A/D
converter 30 53 42 57 SC1_IN_R IN LV Scart input 1 in, right 31 52 41 56 SC1_IN_L IN LV Scart input 1 in, left 32 51 55 ASG1 AHVSS Analog Shield Ground 1 33 50 40 54 SC2_IN_R IN LV Scart input 2 in, right 34 49 39 53 SC2_IN_L IN LV Scart 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 INTERMETALL 45
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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
35 48 52 ASG2 AHVSS Analog Shield Ground 2 36 47 38 51 SC3_IN_R IN LV Scart input 3 in, right 37 46 37 50 SC3_IN_L IN LV Scart input 3 in, left 38 45 49 NC (ASG4) LV Not connected 39 44 48 NC
LV Not connected
(SC4_IN_R)
40 43 47 NC
LV Not connected
(SC4_IN_L)
41 46 NC LV or
Not connected
AHVSS
42 42 36 45 AGNDC X Analog reference voltage
high voltage part 43 41 35 44 AHVSS X Analog ground – 43 AHVSS X Analog ground – 42 NC LV Not connected – 41 NC LV Not connected 44 40 34 40 CAPL_M X Volume capacitor MAIN 45 39 33 39 AHVSUP X Analog power supply
8.0 V 46 38 32 38 CAPL_A X Volume capacitor AUX 47 37 31 37 SC1_OUT_L OUT LV Scart output 1, left 48 36 30 36 SC1_OUT_R OUT LV Scart output 1, right 49 35 29 35 VREF1 X Reference ground 1 high
voltage part 50 34 28 34 SC2_OUT_L OUT LV Scart output 2, left 51 33 27 33 SC2_OUT_R OUT LV Scart output 2, right 52 32 ASG3 AHVSS
2)
Analog Shield Ground 3 53 32 31 NC LV Not connected 54 31 26 30 NC
LV Not connected
(DACM_SUB) 55 30 29 NC LV Not connected 56 29 25 28 DACM_L OUT LV Analog 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
57 28 24 27 DACM_R OUT LV Analog output MAIN,
right
58 27 23 26 VREF2 X Reference ground 2 high
voltage part 59 26 22 25 DACA_L OUT LV Analog output AUX, left 60 25 21 24 DACA_R OUT LV Analog output AUX, right – 23 NC LV Not connected – 22 NC LV Not connected 61 24 20 21 RESETQ IN X Power-on-reset 62 23 20 NC LV Not connected 63 22 19 NC LV Not connected 64 21 19 18 NC LV Not connected 65 20 18 17 I2S_DA_IN2 IN LV I2S2-data input 66 19 17 16 DVSS X Digital ground – 15 DVSS X Digital ground – 14 DVSS X Digital ground 67 18 16 13 DVSUP X Digital power supply +5 V – 12 DVSUP X Digital power supply +5 V – 11 DVSUP X Digital power supply +5 V 68 17 15 10 S_CL
(ADR_CL)
1)
Depending on MODE_REG[14], the SBUS Interface can be switched into ADR_MODE with S_CL
OUT LV SBUS 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.
MICRONAS INTERMETALL 47
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MSP 3400C
8.3. Pin Configurations
PRELIMINARY DAT A SHEET
S_ID
I2C_CL
NC
STANDBYQ
ADR_SEL D_CTR_OUT0 D_CTR_OUT1
NC NC NC
AUD_CL_OUT
DMA_SYNC
XTAL_OUT
XTAL_IN TESTEN
ANA_IN2+
ANA_IN–
ANA_IN1+
AVSUP
I2S_CL
I2C_DA
10 11 12 13 14 15 16 17 18 19
20 21 22 23 24 25 26
NC
S_DA_IN
I2S_DA_IN1
I2S_DA_OUT
I2S_WS
654321
789
MSP 3400C
27 28
29 30 31 32 33 34 35 36 37 38 39
S_CL
DVSUP
DVSS
I2S_DA_IN2
NC
NC
68 67 66 65 64 63 62 61
43424140
NC
RESETQ
60 59 58 57 56 55 54 53 52 51 50 49 48 47 46 45 44
DACA_R DACA_L VREF2 DACM_R DACM_L NC NC NC ASG3
SC2_OUT_R SC2_OUT_L VREF1 SC1_OUT_R SC1_OUT_L
CAPL_A AHVSUP CAPL_M
AVSS
MONO_IN
VREFTOP
SC1_IN_R
SC1_IN_L
ASG1
SC2_IN_R
SC2_IN_L
Fig. 8–5: 68-pin PLCC package
AHVSS
AGNDC
NC
NC
NC
NC
SC3_IN_L
SC3_IN_R
ASG2
MICRONAS INTERMETALL48
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PRELIMINARY DAT A SHEET
MSP 3400C
NC NC
NC
S_ID
S_CL
DVSS
NC NC
NC
ASG3
NC NC
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32
AUD_CL_OUT
D_CTR_OUT1 D_CTR_OUT0
ADR_SEL
STANDBYQ
I2C_CL
I2C_DA
I2S_CL
I2S_WS
I2S_DA_OUT
I2S_DA_IN1
S_DA_IN
DVSUP
I2S_DA_IN2 NC
RESETQ DACA_R
DACA_L
VREF2
DACM_R
DACM_L
64 63 62 61 60 59 58 57 56 55 54 53 52 51 50 49
48 47 46
MSP 3400C
45 44
43 42 41 40 39 38 37 36 35 34 33
DMA_SYNC XTAL_OUT XTAL_IN TESTEN ANA_IN2+ ANA_IN– ANA_IN1+ AVSUP AVSS MONO_IN VREFTOP SC1_IN_R SC1_IN_L ASG1 SC2_IN_R SC2_IN_L ASG2 SC3_IN_R SC3_IN_L
NC NC AGNDC AHVSS CAPL_M AHVSUP CAPL_A SC1_OUT_L SC1_OUT_R VREF1 SC2_OUT_L SC2_OUT_R
NC
AUD_CL_OUT
D_CTR_OUT1 D_CTR_OUT0
ADR_SEL
STANDBYQ
I2C_CL
I2C_DA
I2S_CL
I2S_WS
I2S_DA_OUT
I2S_DA_IN1
S_DA_IN
S_ID
S_CL
DVSUP
DVSS
I2S_DA_IN2
NC RESETQ DACA_R
DACA_L
VREF2 DACM_R DACM_L
NC
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26
MSP 3400C
52 51 50 49 48 47 46 45 44 43 42 41 40 39 38 37
36 35 34 33 32 31 30 29 28 27
XTAL_OUT XTAL_IN TESTEN ANA_IN2+ ANA_IN– ANA_IN1+ AVSUP AVSS MONO_IN VREFTOP SC1_IN_R SC1_IN_L SC2_IN_R SC2_IN_L SC3_IN_R SC3_IN_L AGNDC AHVSS CAPL_M AHVSUP CAPL_A SC1_OUT_L SC1_OUT_R VREF1 SC2_OUT_L SC2_OUT_R
Fig. 8–6: 64-pin shrink PSDIP package
Fig. 8–7: 52-pin shrink PSDIP package
MICRONAS INTERMETALL 49
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MSP 3400C
PRELIMINARY DAT A SHEET
AVSUP AVSUP
ANA_IN1+
ANA_IN–
ANA_IN2+
TESTEN XTAL_IN
XTAL_OUT
DMA_SYNC
AUD_CL_OUT
D_CTR_OUT1 D_CTR_OUT0
ADR_SEL
STANDBY_Q
NC NC
NC
65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80
SC2_IN_R
ASG1
SC1_IN_L
SC1_IN_R
VREFTOP
NC
MONO_IN
AVSS
AVSS
NC
626364
61 60 59 58 57 56
345678910111213
12
SC2_IN_L
55 54 53 52 51 50 49 48
ASG2
MSP 3400C
SC3_IN_R
SC3_IN_L
NC
NC
47 46 45 44 43 42 41
18 19 20 21 22 23 24
17161514
NC
NC
AGNDC
AHVSS
AHVSS
NC
NC
40 39 38 37 36 35 34 33 32 31 30 29 28 27 26 25
CAPL_M AHVSUP CAPL_A SC1_OUT_L SC1_OUT_R VREF1 SC2_OUT_L SC2_OUT_R ASG3
NC NC NC DACM_L DACM_R
VREF2 DACA_L
NC I2C_CL
I2C_DA
I2S_CL
I2S_WS
I2S_DA_OUT
I2S_DA_IN1
S_DA_IN I2S_DA_IN2
S_ID
Fig. 8–8: 80-pin PQFP package
S_CL
DVSUP
DVSUP
DACA_R
NC
NC
RESETQ
NC
NC
NC
DVSS
DVSS
DVSS
DVSUP
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PRELIMINARY DAT A SHEET
8.4. Pin Circuits
DV
SUP
P
MSP 3400C
N
GND
Fig. 8–9: Output Pins 1, 5, 13, 14, and 68 (S_ID, I
2
S_DA_OUT, D_CTR_OUT0/1, S_CL)
DV
SUP
P
N
GND
Fig. 8–10: Input Pins 4 and 65
2
(I
S_DA_IN1/2)
2.5 V
Fig. 8–14: Input Pin 19 (DMA_SYNC)
DV
SUP
P
N
GND
Fig. 8–15: Input Pin 3 (S_DA_IN)
P
N
GND
Fig. 8–11: Input/Output Pins 8 and 9 (I2C_DA, I2C_CL)
Fig. 8–12: Input Pins 11, 12, 61, and 62 (ST ANDBYQ, ADR_SEL, RESETQ, TESTEN)
DV
SUP
P
N
3–30 pF
3–30 pF
500 k
N
2.5 V
Fig. 8–16: Output/Input Pins 18, 20, and 21 (AUD_CL_OUT, XTALIN/OUT)
ANAIN1+ ANAIN2+
ANAIN– VREFTOP
A
D
GND
Fig. 8–13: Input/Output Pins 6 and 7
2
(I
S_WS, I2S_CL)
Fig. 8–17: Input Pins 23–25 and 29 (ANA_IN2+, ANA_IN–, ANA_IN1+, VREFTOP)
MICRONAS INTERMETALL 51
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MSP 3400C
16 K
3.75 V
AHV
SUP
0...1.2 mA
PRELIMINARY DAT A SHEET
Fig. 8–18: Input Pin 28 (MONO_IN)
0...2 V
Fig. 8–19: Capacitor Pins 44 and 46 (CAPL_M, CAPL_A)
40 K
3.75 V
3.3 K
Fig. 8–21: Output Pins 56, 57, 59, and 60 (DACA_L/R, DACM_L/R)
125 K
3.75 V
Fig. 8–22: Pin 42 (AGNDC)
40 pF
80 K
300
3.75 V
Fig. 8–20: Input Pins 30, 31, 33, 34, 36, and 37 (SC1–3_IN_L/R)
Fig. 8–23: Output Pins 47, 48, 50 and 51 (SC_1/2_OUT_L/R)
MICRONAS INTERMETALL52
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PRELIMINARY DAT A SHEET
MSP 3400C
8.5. Electrical Characteristics
8.5.1. Absolute Maximum Ratings
Symbol Parameter Pin Name Min. 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 0 70 °C Storage Temperature –40 125 °C First Supply Voltage AHVSUP –0.3 9.0 V Second Supply Voltage DVSUP –0.3 6.0 V Third Supply Voltage AVSUP –0.3 6.0 V Voltage between AVSUP
and DVSUP Chip Power Dissipation
PLCC68 without Heat Spreader Input Voltage, all Digital Inputs –0.3 V
A VSUP,
–0.5 0.5 V
DVSUP AHVSUP,
DVSUP, AVSUP 1100 mW
+0.3 V
SUP2
Input Current, all Digital Pins –20 +20 mA Input Voltage, all Analog Inputs SCn_IN_s,
2)
–0.3 V
SUP1
+0.3 V
MONO_IN
Input Current, all Analog Inputs SCn_IN_s,
2)
–5 +5 mA
MONO_IN Output Current, all SCART Outputs SCn_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 INTERMETALL 53
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
Symbol Parameter
V V V
V V t
REIL
SUP1
SUP2
SUP3
REIL
REIH
First Supply Voltage AHVSUP 7.6 8.0 8.4 V Second Supply Voltage DVSUP 4.75 5.0 5.25 V Third Supply Voltage AVSUP 4.75 5.0 5.25 V
RESET Input Low Voltage RESETQ 0.45 V RESET Input High Voltage 0.8 V RESET Low Time after DVSUP
Stable and Oscillator Startup
V
DMAIL
V
DMAIH
t
DMA
R
DMA
V
DIGIL
V
DIGIH
Sync Input Low Voltage DMA_SYNC 0.44 V Sync Input High Voltage 0.56 V Sync Input Frequency 18.0 kHz Sync Input Clock High-Level Time 500 ns
Digital Input Low Voltage STANDBYQ, Digital Input High Voltage
Pin Name
TESTEN
Min. Typ. Max. Unit
SUP2
SUP2
5 µs
SUP1
SUP1
0.25 V
SUP2
,
0.75 V
SUP2
t
STBYQ1
STANDBYQ Setup Time before Turn-off of Second Supply Voltage
I2C-Bus Recommendations V V f
IM
t
I2C1
t
I2C2
t
I2C3
t
I2C4
t
I2C5
IMIL
IMIH
I2C-BUS Input Low Voltage I2C_CL, I2C-BUS Input High Voltage I2C-BUS Frequency I2C_CL 1.0 MHz I2C START Condition Setup Time I2C_CL, I2C STOP Condition Setup Time I2C-Clock Low Pulse Time I2C_CL 500 ns I2C-Clock High Pulse Time 500 ns I2C-Data Setup Time Before
Rising Edge of Clock
t
I2C6
I2C-Data Hold Time after Falling
Edge of Clock V V
I2SIL
I2SIH
I2S-Data Input Low Voltage I2S_DA_IN1/2 0.25 V
I2S-Data Input High Voltage 0.75 V
ST ANDBYQ, DVSUP
I2C_CL,
2
I
C_DA
1 µs
0.3 V
0.6 V
120 ns 120 ns
55 ns
55 ns
SUP2
SUP2
SUP2
SUP2
MICRONAS INTERMETALL54
Page 55
PRELIMINARY DAT A SHEET
MSP 3400C
t
I2S1
t
I2S2
V
I2SIDL
V
I2SIDH
f
I2SCL
R
I2SCL
f
I2SWS
t
I2SWS1
ParameterSymbol
I2S-Data Input Setup Time before Rising Edge of Clock
I2S-Data Input Hold Time after Falling Edge of Clock
I2S-Input Low Voltage when MSP 3400C in I2S-Slave-Mode
I2S-Input High Voltage when MSP 3400C in I2S-Slave-Mode
I2S-Clock Input Frequency when MSP 3400C in I2S-Slave-Mode
I2S-Clock Input Ratio when MSP 3400C in I2S-Slave-Mode
I2S-Wordstrobe Input Frequency when MSP 3400C in I2S-Slave­Mode
I2S-Wordstrobe Input Setup Time before Rising Edge of Clock when MSP 3400C in I2S-Slave-Mode
Pin Name
I2S_DA_IN1/2,
20 ns
UnitMax.T yp.Min.
I2S_CL
0 ns
I2S_CL,
0.25 V
SUP2
I2S_WS
0.75 V
SUP2
I2S_CL 1.024 MHz
0.9 1.1
I2S_WS 32.0 kHz
I2S_WS,
60 ns
I2S_CL
t
I2SWS2
I2S-Wordstrobe Input Hold Time after Falling Edge of Clock when MSP 3400C in I2S-Slave-Mode
V
SBUSIL
I
SBUSIL
V
SBUSTRIG
t
SBUS1
SBUS-Data Input Low Voltage S_DA_IN 0.6 V SBUS-Data Input Low Current 0.9 1.7 3.2 mA SBUS-Data Input Trigger Voltage 0.8 1.2 V SBUS-Data Input Setup Time
before Rising Edge of Clock
t
SBUS2
SBUS-Data Input Hold Time after Falling Edge of Clock
Crystal Recommendations for Master-Slave Application f
P
Parallel Resonance Frequency at 12 pF Load Capacitance
f
TOL
D
TEM
Accuracy of Adjustment –20 +20 ppm Frequency Variation versus
Temperature
R
R
Series Resistance 8 25
S_DA_IN, S_CL
0 ns
10 ns
0 ns
18.432 MHz
–20 +20 ppm
C
0
C
1
Shunt (Parallel) Capacitance 6.2 7.0 pF Motional (Dynamic) Capacitance 19 24 fF
MICRONAS INTERMETALL 55
Page 56
MSP 3400C
1)
1)
PRELIMINARY DAT A SHEET
ParameterSymbol
Pin Name
Load Capacitance Recommendations for Master-Slave Applications C
L
f
CL
External Load Capacitance
Required Open Loop Clock
Frequency (T
amb
= 25°C)
2)
XTAL_IN, XTAL_OUT
PSDIP 1.5 PLCC 3.3
18.431 18.433 MHz
Crystal Recommendations for FM Application (No Master-Slave Mode possible) f
P
Parallel Resonance Frequency at
18.432 MHz
12 pF Load Capacitance f
TOL
D
TEM
Accuracy of Adjustment –100 +100 ppm
Frequency Variation versus
–50 +50 ppm
Temperature R
R
C
0
Series Resistance 8 25
Shunt (Parallel) Capacitance 6.2 7.0 pF Load Capacitance Recommendations for FM Application (No Master-Slave Mode possible)
C
L
External Load Capacitance
2)
XTAL_IN, XTAL_OUT
PSDIP 1.5 PLCC 3.3
UnitMax.Typ.Min.
pF pF
pF pF
Amplitude Recommendation for Operation with External Clock Input (C V
XCA
External Clock Amplitude XTAL_IN 0.7 V
after reset = 22 pF)
load
Analog Input and Output Recommendations C
AGNDC
AGNDC-Filter-Capacitor AGNDC –20% 3.3 µF
Ceramic Capacitor in Parallel –20% 100 nF C
inSC
DC-Decoupling Capacitor in front
SCn_IN_s
–20% 330 +20% nF
of SCART Inputs V
inSC
V
inMONO
R
LSC
C
LSC
C
VMA
SCART Input Level 2.0 V
Input Level, Mono Input MONO_IN 2.0 V
SCART Load Resistance SCn_OUT_s
10 k SCART Load Capacitance 6.0 nF Main/AUX Volume Capacitor CAPL_M,
10 µF
CAPL_A
C
FMA
1)
“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 Capacitor DACM_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-Capacitor VREFTOP –20% 10 µF Ceramic Capacitor in Parallel –20% 100 nF
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 dB dB
Suppression of Spectrum
Above 9.0 MHz
Pin Name
ANA_IN1+, ANA_IN2+,
UnitMax.T yp.Min.
0.14 0.8 3 Vpp
7 dB
7 dB
15 dB
15 dB
15 dB
FM
MAX
Maximum FM-Deviation (apprx.)
normal mode
high deviation mode
±192 ±360
kHz
MICRONAS INTERMETALL 57
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.)
Symbol Parameter Pin Name Min. Typ. Max. Unit Test Conditions
DCO f
CLOCK
D
CLOCK
t
JITTER
V
xtalDC
t
Startup
Power Supply I
SUP1A
I
SUP2A
Clock Input Frequency XTAL_IN 18.432 MHz Clock High to Low Ratio 45 55 % Clock Jitter (verification not
50 ps
provided in production test) DC-Voltage Oscillator 2.5 V Oscillator Startup Time at
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
0.4 2.0 ms
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) DVSUP 60 65 70 mA f = 18.432 MHz
DVSUP = 5 V
I
SUP3A
I
SUP1S
Third Supply Current (active) AVSUP 25 mA f = 18.432 MHz
First Supply Current (standby mode) at T
Audio Clock Output V
APUAC
V
APUDC
Audio Clock Output AC Voltage AUD_CL_OUT 1.2 V Audio Clock Output DC Voltage 0.4 0.6 V
Digital Output V
DCTROL
V
DCTROH
Digital Output Low Voltage D_CTR_OUT0 Digital Output High Voltage
I2C Bus V
IMOL
I
IMOH
t
IMOL1
t
IMOL2
I2C-Data Output Low Voltage I2C_DA 0.4 V I I2C-Data Output High Current 1 µA V 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
AHVSUP 2.8 5.0 7.2 mA ST 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
0.4 V I
4.0 V I
15 ns
100 ns fIM = 1 MHz
DVSUP = 5 V
SBus f
SB
t
S1/S2
t
S3
f
SIO
t
S6
SBUS-Clock Frequency S_CL 4608 kHz DVSUP = 5 V SBUS-Clock High/Low-Ratio 0.9 1.0 1.1 ns SBUS Setup Time before
Ident End Pulse
S_CL, S_ID
210 ns DVSUP = 5.25 V
SBUS Ident frequency S_ID 32 kHz SBUS-Ident End Pulse Time 210 ns DVSUP = 5.25 V
MICRONAS INTERMETALL58
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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 Voltage I2S_WS, I2S Output High Voltage
I2S_DA_OUT
,
4.0 V I
0.4 V I
I2SOL
I2SOH
= 1 mA
= –1 mA I2S-Clock Output Frequency I2S_CL 1204 kHz DVSUP = 5 V I2S-Wordstrobe Output Frequency I2S_WS 32.0 kHz DVSUP = 5 V I2S-Clock High/Low-Ratio I2S_CL 0.9 1.0 1.1 I2S-Data Setup Time
before Rising Edge of Clock I2S-Data Hold Time after Falling
I2S_CL, I2S_DA_OUT
200 ns DVSUP = 4.75 V
12 ns DVSUP = 5.25 V
Edge of Clock I2S-Wordstrobe Setup Time
before Rising Edge of Clock I2S-Wordstrobe Hold Time after
I2S_CL, I2S_WS
100 ns DVSUP = 4.75 V
50 ns DVSUP = 5.25 V
Falling Edge of Clock
AGNDC Open Circuit Voltage AGNDC 3.64 3.73 3.84 V R AGNDC Output Resistance
at Tj = 27 °C from T
= 0 to 70 °C
A
70 70
125 180
180
k k
load
3 V ≤ V
10 M
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)
2.02 2.12 2.22 V
0.20
0.20
40 58
58
16 23
23
0.33 0.46
0.5
k k
k k
RMS
k k
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 +70 mV
f
= 1kHz
–1.0 0 +0.5 dB
1)
signal
with respect to 1 kHz
–0.5 0 +0.5 dB
V
outSC
Signal Level at SCART-Output during full-scale digital input signal
SCn_OUT_s
1)
1.8 1.9 2.0 V
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 INTERMETALL 59
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 SNR Signal-to-Noise Ratio
from Analog Input to DSP MONO_IN,
from Analog Input to SCART Output
from DSP to SCART Output SCn_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
3.3 4.6
5.0
k k
signal
1.74–1.94612.14–V mV
1.23 1.37 1.51 V
RMS
f
signal
= 1 kHz
85 88 dB Input Level = –20 dB with
resp. to V kHz, equally weighted 20 Hz...16 kHz
93 96 dB Input Level = –20 dB,
f
= 1 kHz,
sig
1)
1)
85 88 dB Input 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)
THD Total Harmonic Distortion
from Analog Input to DSP MONO_IN,
SCn_IN_s
from Analog Input to SCART Output
MONO_IN, SCn_IN_s
1)
SCn_OUT_s
from DSP to SCART Output SCn_OUT_s
from DSP to Main or AUX Output DACA_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 k
R
Load
AICL
, f
=1kHz,
sig
2)
0.01 0.03 % Input Level = –3 dBr, f
= 1 kHz, equally
sig
1)
1)
0.01 0.03 % Input Level = –3 dBr,
weighted 20 Hz...20 kHz,
= 30 k
R
Load
f
= 1 kHz, equally
sig
weighted 20 Hz...16 kHz, R
Load
= 30 k
3)
0.01 0.03 % Input Level = –3 dBr, f
= 1 kHz, equally
sig
weighted 20 Hz...16 kHz, R
Load
= 30 k
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
XTALK Crosstalk attenuation
– PLCC68 – PSDIP64
between left and right channel within SCART Input/Out­put pair (LR, 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 k
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_OUT PLCC68 O: or unsel. MONO/SCn_IN → DSP
D: MONO/SCn_IN SC1_OUT PLCC68 O: DSP SCn_OUT
D: MONO/SCn_IN unselected PLCC68 O: DSP SC1_OUT
Crosstalk between Main and AUX Output pairs DSP DACp
1)
1)
1)
1)
PSDIP64
PSDIP64
PSDIP64
PLCC68
PSDIP649590
Crosstalk from Main or AUX Output to SCART Output and vice versa
D = disturbing program O = observed program
D: MONO/SCn_IN/DSP → SCn_OUT PLCC68 O: DSP DACp
D: MONO/SCn_IN/DSP SCn_OUT PLCC68 O: DSP DACp
D: DSP DACp PLCC68 O: MONO/SCn_IN SCn_OUT
1)
1)
1)
PSDIP64
PSDIP64
PSDIP64
95 95
100 100
100 100
90 85
95 85
100 95
dB dB
dB dB
dB dB
dB dB
dB dB
dB dB
dB dB
2)
3)
3)
(equally weighted 20 Hz...16 kHz)
3)
same signal source on left and right disturbing channel, effect on each observed output channel
(equally weighted 20 Hz...20 kHz) same signal source on left and right disturbing channel, effect on each observed output channel
SCART output load resis­tance 10 k
SCART output load resis­tance 30 k
3)
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
MICRONAS INTERMETALL 61
D: DSP DACp PLCC68 O: DSP SCn_OUT
1)
PSDIP64
100 95
dB dB
Page 62
MSP 3400C
PSRR: rejection of noise on AHVSUP at 1 kHz PSRR AGNDC AGNDC 80 dB
PRELIMINARY DAT A SHEET
Test ConditionsUnitMax.Typ.Min.Pin NameParameterSymbol
From analog Input to DSP MONO_IN
SCn_IN_s
From analog Input to SCART Output
MONO_IN SCn_IN_s,
1)
SCn_OUT_s
From DSP to SCART Output SCn_OUT_s From DSP to MAIN/AUX Output DACp_s
1)
1)
1)
1)
69 dB
74 dB
70 dB
80 dB Sound IF Input Section DC
VREFTOP
R
IFIN
DC voltage at VREFTOP VREFTOP 2.4 2.6 2.7 V V
Input Impedance ANA_IN1+,
ANA_IN2+,
1.5
10.5214.1
2.5
17.6
kOhm AGC = +20 dB
ANA_IN–
DC
ANA_IN
XTALK
BW
IF
IF
DC voltage on IF inputs 1.3 1.5 1.7 V AVSUP = 5 V
Crosstalk attenuation 40 t.b.d. dB f
3 dB Bandwidth 10 MHz Input Level = –2 dBr
AGC AGC step width t.b.d. 0.85 t.b.d. dB f
SUPANALOG
R
Load
= 5 V
10 M
AGC = +3 dB
10 M
R
Load
R
10 M
Load
= 1 MHz,
sig
Input Level = –2 dBr
= 1 MHz,
sig
Input Level = –2 dBr
1)
“n” means “1”, “2” or “3”, “s” means “L” or “R”, “p” means “M” or “A”
MICRONAS INTERMETALL62
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PRELIMINARY DAT A SHEET
Overall Performance
MSP 3400C
Test ConditionsUnitMax.Typ.Min.Pin NameParameterSymbol
S/N
FM
S/N
D2MAC
THD
FM
THD
D2MAC
dV
FMOUT
dV-
D2MACOUT
fR
FM
fR
D2MAC
FM input to Main/AUX/SCART Output
Signal to Noise ratio of D2MAC baseband signal on Main/AUX/ SCART outputs
Total Harmonic Distortion + Noise of FM demodulated signal on Main/AUX/SCART output
Total Harmonic Distortion + Noise of D2MAC baseband signal for Main/AUX/SCART output
Tolerance of output voltage of FM demodulated signal
Tolerance of output voltage of D2MAC baseband signal
FM frequency response on Main/ AUX/SCART outputs, bandwidth 20 to 15000 Hz
D2MAC frequency response on Main/AUX/SCART outputs, band­width 20 to 15000 Hz
DACp_s, SCn_OUT_s
70 dB 1 FM-carrier 5.5 MHz,
1)
50 µs, 1 kHz, 40 kHz de­viation; RMS, unweighted 0 to 15 kHz; full input range
TBD dB
0.3 % 1 FM-carrier 5.5 MHz,
1kHz, 50 µs; 40 kHz devi­ation; full input range
0.01 0.1 % 2.12 kHz, Modulator input
level = 0 dBref
–1.5 +1.5 dB 1 FM-carrier, 50 µs,
1 kHz 40 kHz deviation; RMS
–1.5 +1.5 dB 2.12 kHz, Modulator input
level = 0 dBref
–1.0 +1.0 dB 1 FM-carrier 5.5 MHz,
50 µs, Modulator input level = –14.6 dBref; RMS
–1.0 +1.0 dB Modulator input level =
–12 dB dBref; RMS
SEP
FM
SEP
D2MAC
XTALK
FM
XTALK-
D2MAC
1)
“n” means “1”, “2” or “3”, “s” means “L” or “R”, “p” means “M” or “A”
FM channel separation (Stereo) 50 dB 2 FM-carriers 5.5/5.74
D2MAC channel separation (Stereo)
FM crosstalk attenuation (Dual) 80 dB 2 FM-carriers 5.5/5.74
D2MAC crosstalk attenuation (Dual)
MHz, 50 µs, 1 kHz, 40 kHz deviation; RMS
80 dB
MHz, 50 µs, 1 kHz, 40 kHz deviation; RMS
80 dB
MICRONAS INTERMETALL 63
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MSP 3400C
9. Application of the MSP 3400C
PRELIMINARY DAT A SHEET
5V
DVSS
AHVSS
AHVSS
AHVSS
5V
DVSS
Tuner 2
Tuner 1
330 nF
330 nF 330 nF
330 nF 330 nF
330 nF 330 nF
IF 2 IN
Signal GND
IF 1 IN
50pF 50pF
50pF
Ana_IN1+ (58) 25
28 (55) MONO_IN 52 (30) ASG3 30 (53) SC1_IN_R 31 (52) SC1_IN_L 32 (51) ASG1
33 (50) SC2_IN_R 34 (49) SC2_IN_L
35 (48) ASG2 36 (47) SC3_IN_R 37 (46) SC3_IN_L
11 (7) STANDBY Q
12 (6) ADR_SEL
2
9 (9) I
C-CL
8 (10) I2C-DA 1 (16) S_ID
68 (17) S_CL 3 (15) S_DA_IN
6 (12) I2S_WS 7 (11) I2S_CL 4 (14) I2S_DA_IN1 65 (20) I2S_DA_IN2 5 (13) I2S_DA_OUT
10
100
µF
nF
-
+
3.3 µF
+
Ana_IN– (59) 24
Ana_IN2+ (60) 23
VREFTOP (54) 29
MSP 3400C
18.432
100
MHz
nF
AGNDC (42) 42
XTAL_IN (62) 21
0.1 pF
+8.0 V
++
10 µF
10 µF
CAPL_A (46) 38
CAPL_M (40) 44
XTAL_OUT (63) 20
DACM_L (29) 56
DACM_R (28) 57
DACA_L (26) 59
DACA_R (25) 60
SC1_OUT_L (37) 47
SC1_OUT_R (36) 48
SC2_OUT_L (34) 50
SC2_OUT_R (33) 51
D_CTR_OUT0 (5) 13 D_CTR_OUT1 (4) 14
AUD_CL_OUT (1) 18
DMA_SYNC (64) 19
TESTEN (61) 22
1 nF
1 nF
1 nF
1 nF
100 100
100 100
1µF
1µF
1µF
1µF
+
+
+
+
22 µF 22 µF
22 µF 22 µF
DVSS
MAIN
HEAD­PHONE
45 (39) AHVSUP
100 nF
43 (41) AHVSS
49 (35) VREF1
58 (27) VREF2
67 (18) DVSUP
61 (24) RESETQ
100 nF
+
10 µF
26 (57) AVSUP
66 (19) DVSS
100 nF
27 (56) AVSSAVSS
5V 5V 8.0 V
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
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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
65 17 16
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
65 66 64
ACLK S_DATA S_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
120 6k8
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 INTERMETALL 65
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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_OUT min with C load
Rout (HF) typ.
>1.2 Vpp 40 pF
150
>1.2 Vpp 40 pF
120
Table 10–2: Clock input and output specification for ICs connected to MSP
DMA 2381 DMA 2386 AMU2481
XTAL_IN min
>0.7 Vpp
>0.7 Vpp Clock-in min (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
>0.7 Vpp
7pF
DMAIH
.”
MICRONAS INTERMETALL66
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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 INTERMETALL 67
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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 (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 (330–470 Ω)
Standby control
DVSUP STANDBYQ
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
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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
New Features:
1. Equalizer
2. Improved identification
3. Improved adaptive deemphasis Version C6
New Features:
1. Adjustable Stereo Basewidth Enlargement (SBE) and switchable Pseudo Stereo Effect (SBE)
2. New Channel Matrix Modes (Mono, Sum/Dif, etc)
3. New Audio Clock Output Driver
4. Fast mute (Volume)
5. Clipping mode (Volume)
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.
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MSP 3400C
PRELIMINARY DAT A SHEET
MICRONAS INTERMETALL70
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PRELIMINARY DAT A SHEET
MSP 3400C
MICRONAS INTERMETALL 71
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MSP 3400C
PRELIMINARY DAT A SHEET
MICRONAS INTERMETALL GmbH Hans-Bunte-Strasse 19 D-79108 Freiburg (Germany) P.O. Box 840 D-79008 Freiburg (Germany) Tel. +49-761-517-0 Fax +49-761-517-2174 E-mail: docservice@intermetall.de Internet: http://www.intermetall.de
Printed in Germany Order No. 6251-377-3PD
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
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MultimediaICs
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