MSP 34x0G Multistandard Sound Processor Family: Micronas
MSP 34x0G Multistandard Sound Processor Family: Micronas
MSP 34x0G Multistandard Sound Processor Family: Micronas
MICRONAS
Edition May 27, 2003
6251-476-1DS
MSP 34x0G DATA SHEET
Contents
6 1. Introduction
7 1.1. Features of the MSP 34x0G Family
7 1.2. MSP 34x0G Version List
8 1.3. MSP 34x0G Versions and their Application Fields
9 2. Functional Description
10 2.1. Architecture of the MSP 34x0G Family
10 2.2. Sound IF Processing
10 2.2.1. Analog Sound IF Input
10 2.2.2. Demodulator: Standards and Features
11 2.2.3. Preprocessing of Demodulator Signals
11 2.2.4. Automatic Sound Select
11 2.2.5. Manual Mode
11 2.3. Preprocessing for SCART and I2S Input Signals
13 2.4. Source Selection and Output Channel Matrix
13 2.5. Audio Baseband Processing
13 2.5.1. SRS WOW
13 2.5.2. BBE High Definition Sound
13 2.5.3. Micronas VOICE
14 2.5.4. Automatic Volume Correction (AVC)
14 2.5.5. Loudspeaker and Headphone Outputs
14 2.5.6. Subwoofer Output
15 2.5.7. Quasi-Peak Detector
15 2.5.8. Micronas BASS (MB)
15 2.5.8.1. Dynamic Amplification
15 2.5.8.2. Adding Harmonics
15 2.5.8.3. Micronas BASS Parameters
15 2.6. SCART Signal Routing
15 2.6.1. SCART DSP In and SCART Out Select
16 2.6.2. Stand-by Mode
16 2.7. I2S Bus Interface
16 2.8. ADR Bus Interface
16 2.9. Digital Control I/O Pins and Status Change Indication
16 2.10. Clock PLL Oscillator and Crystal Specifications
17 3. Control Interface
17 3.1. I2C Bus Interface
17 3.1.1. Internal Hardware Error Handling
18 3.1.2. Description of CONTROL Register
18 3.1.3. Protocol Description
19 3.1.4. Proposals for General MSP 34x0G I2C Telegrams
19 3.1.4.1. Symbols
19 3.1.4.2. Write Telegrams
19 3.1.4.3. Read Telegrams
19 3.1.4.4. Examples
19 3.2. Start-Up Sequence: Power-Up and I2C-Controlling
19 3.3. MSP 34x0G Programming Interface
Contents, continued
46 4. Specifications
46 4.1. Outline Dimensions
51 4.2. Pin Connections and Short Descriptions
54 4.3. Pin Descriptions
57 4.4. Pin Configurations
60 4.5. Pin Circuits
62 4.6. Electrical Characteristics
62 4.6.1. Absolute Maximum Ratings
64 4.6.2. Recommended Operating Conditions
64 4.6.2.1. General Recommended Operating Conditions
65 4.6.2.2. Analog Input and Output Recommendations
66 4.6.2.3. Recommendations for Analog Sound IF Input Signal
67 4.6.2.4. Crystal Recommendations
68 4.6.3. Characteristics
68 4.6.3.1. General Characteristics
69 4.6.3.2. Digital Inputs, Digital Outputs
70 4.6.3.3. Reset Input and Power-Up
71 4.6.3.4. I2C-Bus Characteristics
72 4.6.3.5. I2S-Bus Characteristics
74 4.6.3.6. Analog Baseband Inputs and Outputs, AGNDC
75 4.6.3.7. Sound IF Inputs
75 4.6.3.8. Power Supply Rejection
76 4.6.3.9. Analog Performance
79 4.6.3.10. Sound Standard Dependent Characteristics
Contents, continued
Contents, continued
License Notice:
Supply of this implementation of Dolby Technology does not convey a license nor imply a right under any patent, or any other industrial or intellec-
tual property right of Dolby Laboratories, to use this implementation in any finished end-user or ready-to-use final product. Companies planning to
use this implementation in products must obtain a license from Dolby Laboratories Licensing Corporation before designing such products.
Multistandard Sound Processor Family Current ICs have to perform adjustment procedures in
order to achieve good stereo separation for BTSC and
Release Note: Revision bars indicate significant EIA-J. The MSP 34x0G has optimum stereo perfor-
changes to the previous edition. The hardware and mance without any adjustments.
software description in this document is valid for
the MSP 34x0G version C12 and following ver- All MSP 34xxG versions are pin compatible to the
sions. MSP 34xxD. Only minor modifications are necessary
to adapt a MSP 34xxD controlling software to the
MSP 34xxG. The MSP 34x0G further simplifies con-
1. Introduction trolling software. Standard selection requires a single
I2C transmission only.
The MSP 34x0G family of single-chip Multistandard
Sound Processors covers the sound processing of all The MSP 34x0G has built-in automatic functions: The
analog TV-Standards worldwide, as well as the NICAM IC is able to detect the actual sound standard automat-
digital sound standards. The full TV sound processing, ically (Automatic Standard Detection). Furthermore,
starting with analog sound IF signal-in, down to pro- pilot levels and identification signals can be evaluated
cessed analog AF-out, is performed on a single chip. internally with subsequent switching between mono/
Figure 11 shows a simplified functional block diagram stereo/bilingual; no I2C interaction is necessary (Auto-
of the MSP 34x0G. matic Sound Selection).
These TV sound processing ICs now include versions The MSP 34x0G can handle very high FM deviations
for processing the multichannel television sound even in conjunction with NICAM processing. This is
(MTS) signal conforming to the standard recom- especially important for the introduction of NICAM in
mended by the Broadcast Television Systems Com- China.
mittee (BTSC). The DBX noise reduction, or alterna-
tively, Micronas Noise Reduction (MNR) is performed The ICs are produced in submicron CMOS technology.
alignment free. The MSP 34x0G is available in the following pack-
ages: PSDIP64-1, PSDIP52-1/-2, PMQFP80-11, and
Other processed standards are the Japanese FM-FM PMQFP64-2.
multiplex standard (EIA-J) and the FM Stereo Radio
standard.
Headphone
Source Select
SCART1
DAC
SCART2 SCART
DSP SCART1
SCART3 Input ADC Prescale
Select SCART
DAC Output
SCART4 Select
MONO
SCART2
Micronas BASS (MB) and 5-band graphic equalizer for loudspeaker channel X X X X X X
Four Stereo SCART (line) inputs, one Mono input; two Stereo SCART outputs X X X X X X
Alignment free digital DBX noise reduction for BTSC Stereo and SAP X X
Alignment free digital Micronas Noise Reduction (MNR) for BTSC Stereo and SAP X
BTSC stereo separation (MSP 3420/40G also EIA-J) significantly better than spec. X X X
MSP 3420G available NTSC Version (A2 Korea, BTSC with Micronas Noise Reduction (MNR), Japanese EIA-J system)
MSP 3440G available NTSC Version (A2 Korea, BTSC with DBX noise reduction, Japanese EIA-J system)
Table 11 provides an overview of TV sound stan- multimedia receiver covering all TV sound standards
dards that can be processed by the MSP 34x0G fam- together with terrestrial/cable and satellite radio sound
ily. In addition, the MSP 34x0G is able to handle the can be built; even ASTRA Digital Radio can be pro-
FM-Radio standard. With the MSP 34x0G, a complete cessed (with a DRP 3510A coprocessor).
Table 11: TV Stereo Sound Standards covered by the MSP 34x0G IC Family (details see Appendix A)
3410
6.5 FM-Mono
7.02/7.2 FM-Stereo Europe Sat.
Satellite 7.38/7.56 PAL
3400
SAW Filter
Sound
Tuner IF Loudspeaker
Mixer
1
Mono Subwoofer
Vision
Demo- MSP 34x0G
dulator 2
SCART1
Headphone
2
SCART SCART2
Composite Inputs 2 2
SCART3 SCART1 SCART
Video
2 2 Outputs
SCART4 SCART2
Dolby ADR
Pro Logic Decoder
Processor DRP 3510A
DPL 351xA
DATA SHEET
Automatic
ANA_IN1+ Standard Selection Sound Select
AGC A Deemphasis: FM/AM 0 Loud- Bass/
FM/AM Loud- Comple- D DACM_L
50/75 s, J17 speaker
DEMODULATOR
DBX/MNR Channel AVC
Treble
or ness mentary Spatial Balance
Highpass Effects
Volume
D (incl. Carrier Mute) Matrix
Panda1 Prescale Stereo or A/B 1 Equalize
(0Ehex)
DACM_R
(08hex) (29hex) (02hex) (04hex) 0.5 (2Dhex) (05hex) (01hex)
ANA_IN2+ Decoded (03hex)
Standards: NICAM Stereo or A 3 Level A
Lowpass MB
NICAM Deemphasis
Beeper
Adjust DACM_SUB
A2 J17
4 (2Dhex) (2C hex) (00hex)
Stereo or B
AM Prescale (14hex)
BTSC (10hex)
EIA-J
SAT Standard I2C
FM-Radio and Sound Read
ADR-Bus
Interface Detection Register
Volume D
Headphone DACA_L
Bass/ Loudness
Channel Balance
2
I S1 Treble
5 Matrix A
I2S_DA_IN1 I2S
DACA_R
Source Select
Interface (09hex) (31/32hex) (33hex) (30hex) (06hex)
Prescale
May 27, 2003; 6251-476-1DS
(16hex)
I2S
I2S I2S_DA_OUT
2 Channel
I S2 Interface
Matrix
I2S_DA_IN2 I2S 6
Interface (0Bhex)
Prescale
(12hex)
Quasi-Peak I2C
Quasi-Peak
Channel Read
Detector
Matrix Register (19hex)
(1Ahex)
(0Chex)
A SCART
SCART DSP Input Select
2 Volume
SCART1 D
D Prescale Channel SCART1_L/R
(0Dhex)
Matrix A
(0Ahex) (07hex)
Volume SC1_OUT_L
SCART2 D
Channel SCART2_L/R
Matrix A
SC1_OUT_R
MSP 34x0G
SC2_IN_L
SC2_OUT_L
SC2_IN_R
SC3_IN_L
SC2_OUT_R
SC3_IN_R
SC4_IN_L
SC4_IN_R
MONO_IN
(13hex)
Fig. 21: Signal flow block diagram of the MSP 34x0G (input and output names correspond to pin names)
9
MSP 34x0G DATA SHEET
2.1. Architecture of the MSP 34x0G Family BTSC-Mono + SAP: Detection and FM demodulation
of the aural carrier resulting in the MTS/MPX signal.
Fig. 21 on page 9 shows a simplified block diagram of Detection and evaluation of the pilot carrier, detection
the IC. The block diagram contains all features of the and FM demodulation of the SAP subcarrier. Process-
MSP 3450G. Other members of the MSP 34x0G fam- ing of DBX noise reduction or Micronas Noise Reduc-
ily do not have the complete set of features: The tion (MNR).
demodulator handles only a subset of the standards
presented in the demodulator block; NICAM process- Japan Stereo: Detection and FM demodulation of the
ing is only possible in the MSP 3410G and aural carrier resulting in the MPX signal. Demodulation
MSP 3450G. and evaluation of the identification signal and FM
demodulation of the (LR)-carrier.
2.2.2. Demodulator: Standards and Features Standard Selection: The controlling of the demodula-
tor is minimized: All parameters, such as tuning fre-
The MSP 34x0G is able to demodulate all TV-sound quencies or filter bandwidth, are adjusted automati-
standards worldwide including the digital NICAM sys- cally by transmitting one single value to the
tem. Depending on the MSP 34x0G version, the fol- STANDARD SELECT register. For all standards, spe-
lowing demodulation modes can be performed: cific MSP standard codes are defined.
A2 Systems: Detection and demodulation of two sep- Automatic Standard Detection: If the TV sound stan-
arate FM carriers (FM1 and FM2), demodulation and dard is unknown, the MSP 34x0G can automatically
evaluation of the identification signal of carrier FM2. detect the actual standard, switch to that standard, and
respond the actual MSP standard code.
NICAM Systems: Demodulation and decoding of the
NICAM carrier, detection and demodulation of the ana- Automatic Carrier Mute: To prevent noise effects or
log (FM or AM) carrier. For D/K-NICAM, the FM carrier FM identification problems in the absence of an FM
may have a maximum deviation of 384 kHz. carrier, the MSP 34x0G offers a configurable carrier
mute feature, which is activated automatically if the TV
Very high deviation FM-Mono: Detection and robust sound standard is selected by means of the STAN-
demodulation of one FM carrier with a maximum devi- DARD SELECT register. If no FM carrier is detected at
ation of 540 kHz. one of the two MSP demodulator channels, the corre-
sponding demodulator output is muted. This is indi-
BTSC-Stereo: Detection and FM demodulation of the cated in the STATUS register.
aural carrier resulting in the MTS/MPX signal. Detec-
tion and evaluation of the pilot carrier, AM demodula-
tion of the (LR)-carrier and detection of the SAP sub-
carrier. Processing of DBX noise reduction or Micronas
Noise Reduction (MNR).
Source Select
secondary Prescale
channel Stereo or A/B 1
Automatic Output-Ch.
interaction is necessary when the broadcasted sound Sound matrices
mode changes (e.g. from mono to stereo). NICAM A NICAM
Select must be set
Stereo or A 3 once to
stereo.
The demodulator supports the identification check by NICAM B Prescale Stereo or B 4
switching between mono-compatible standards (stan-
dards that have the same FM-Mono carrier) automati-
cally and non-audible. If B/G-FM or B/G-NICAM is Fig. 22: Source channel assignment of demodulated
selected, the MSP will switch between these stan- signals in Automatic Sound Select Mode
dards. The same action is performed for the stan-
dards: D/K1-FM, D/K2-FM, D/K3-FM and D/K-NICAM.
Switching is only done in the absence of any stereo or 2.2.5. Manual Mode
bilingual identification. If identification is found, the
MSP keeps the detected standard. Fig. 23 shows the source channel assignment of
demodulated signals in case of manual mode. If man-
In case of high bit-error rates, the MSP 34x0G auto- ual mode is required, more information can be found in
matically falls back from digital NICAM sound to ana- Section 6.7. Demodulator Source Channels in Manual
log FM or AM mono. Mode on page 101.
secondary
block prepares four different source channels of channel
Prescale
Output-Ch.
demodulated sound (Fig. 22). By choosing one of the matrices
must be set
four demodulator channels, the preferred sound mode NICAM A NICAM
according to
NICAM the standard.
can be selected for each of the output channels (loud- (Stereo or A/B)
1
The following source channels of demodulated sound Fig. 23: Source channel assignment of demodulated
are defined: signals in Manual Mode
B/G-FM, D/K-FM, M-Korea, Evaluation of the identification signal and automatic switching to mono, stereo, or bilingual. Preparing four
and M-Japan demodulator source channels according to Table 22.
B/G-NICAM, L-NICAM, I-NICAM, Evaluation of NICAM-C-bits and automatic switching to mono, stereo, or bilingual. Preparing four
D/K-NICAM demodulator source channels according to Table 22.
In case of bad or no NICAM reception, the MSP switches automatically to FM/AM mono and switches
back to NICAM if possible. A hysteresis prevents periodical switching.
B/G-FM, B/G-NICAM Automatic searching for stereo/bilingual-identification in case of mono transmission. Automatic and non-
audible changes between Dual-FM and FM-NICAM standards while listening to the basic FM-mono sound
or
carrier.
D/K1-FM, D/K2-FM, D/K3-FM, Example: If starting with B/G-FM-Stereo, there will be a periodical alternation to B/G-NICAM in the
and D/K-NICAM absence of FM-Stereo/Bilingual or NICAM-identification. Once an identification is detected, the MSP
keeps the corresponding standard.
BTSC-STEREO, FM Radio Evaluation of the pilot signal and automatic switching to mono or stereo. Preparing four demodulator
source channels according to Table 22. Detection of the SAP carrier.
M-BTSC-SAP In the absence of SAP, the MSP switches to BTSC-stereo if available. If SAP is detected, the MSP
switches automatically to SAP (see Table 22).
Table 22: Sound modes for the demodulator source channels with Automatic Sound Select
B/G-NICAM 08, 032) NICAM not available or analog Mono analog Mono analog Mono analog Mono
L-NICAM 09 error rate too high
I-NICAM 0A
D/K-NICAM 0B, 042), 052) MONO analog Mono NICAM Mono NICAM Mono NICAM Mono
D/K-NICAM 0C, 0D
(with high STEREO analog Mono NICAM Stereo NICAM Stereo NICAM Stereo
deviation FM)
BILINGUAL: analog Mono Left = NICAM A NICAM A NICAM B
Languages A and B Right = NICAM B
2.4. Source Selection and Output Channel Matrix 2.5.2. BBE High Definition Sound
The Source Selector makes it possible to distribute all License Notice: BBE is a registered trademark of
source signals (one of the demodulator source chan- BBE Sound Inc., the BBE Logo is a trademark of BBE
nels, SCART, or I2S input) to the desired output chan- Sound Inc. A license from BBE Sound Inc. is required
nels (loudspeaker, headphone, etc.). All input and out- before a BBE-version of the MSP 34x0G can be pur-
put signals can be processed simultaneously. Each chased.
source channel is identified by a unique source
address. BBE High Definition Sound or, also called, Sonic Maxi-
mizer technology improves the clarity of music when
For each output channel, the sound mode can be set played back via loudspeakers. A more life like feeling
to sound A, sound B, stereo, or mono by means of the is created by BBE. The BBE-approved Micronas
output channel matrix. implementation works in the digital domain and thus
needs no external components and does not suffer
If Automatic Sound Select is on, the output channel from tolerances and aging effects.
matrix can stay fixed to stereo (transparent) for
demodulated signals. All MSP 34x0G are shipped without BBE except other-
wise ordered. When a BBE-version of the MSP 34x0G
is ordered, it carries a special marking on the chip for
2.5. Audio Baseband Processing identification. The BBE Sonic Maximizer functionality
must be enabled by writing a "license key" into the
2.5.1. SRS WOW MSP 34x0G. For information on how to obtain this
license key from Micronas, please contact your Micro-
License Notice: SRS, SRS WOW, and the SRS Logo nas sales representative.
are trademarks of SRS Labs, Inc. A license from SRS
Labs, Inc. is required before an SRS-version of the
MSP 34x0G can be purchased. 2.5.3. Micronas VOICE
SRS Labs WOW technology enlarges the sound Micronas VOICE was developed to add the following
image field and improves the bass performance of improvements to speech signals:
television speakers. Manufacturers can save costs by
Increase speech signal over background noise to
licensing WOW while utilizing smaller speakers and
increase intelligibility in noisy environments
still provide a higher quality audio experience.
Move voice to the foreground, closer to the listener,
WOW consists of three sections: while other sounds are moved to the back
Clarity Improvement, Improve voices that are hard to understand, leave
clear voices largely undisturbed
3D-Audio
(SRS, Sound Retrieval System),
Micronas VOICE dynamically enhances those portions
and
of speech that are important for intelligibility while at
Bass Enhancement (TruBass). the same time decreasing portions of the signal that
disturb intelligibility. The average amplitude of the sig-
nal is not changed.
Key features of WOW include:
According to speech theory, there are two main effects
Wider and taller sound image field
that affect the intelligibility of speech. Micronas VOICE
Larger sweet spot combines both effects to achieve a maximum
enhancement of intelligibility.
Deep, rich bass tones
Forward and backward masking: For intelligibility,
Quality improvements to audio listening experience consonants are more important than vowels, but the
Improved clarity of speech amplitude of consonants is much lower than that of
vowels. The consonants are masked by the vowels.
All MSP 34x0G are shipped without SRS except other- Therefore, the amplitude of consonants is increased
wise ordered. When an SRS-version of MSP 34x0G is and the amplitude of vowels decreased.
ordered, it carries a special marking on the chip for Phonemes and formants: Most important for intelli-
identification. The SRS WOW functionality must be gibility are the second to fourth formants of speech.
enabled by writing a "license key" into the
These formants are detected and increased, while
MSP 34x0G. For information on how to obtain this
other parts of the signal are decreased.
license key from Micronas, please contact your Micro-
nas sales representative.
All MSP 34x0G are shipped without Micronas VOICE 2.5.5. Loudspeaker and Headphone Outputs
except otherwise ordered. When a Micronas VOICE -
version of the MSP 34x0G is ordered, it carries a spe- The following baseband features are implemented in
cial marking on the chip for identification. The Micro- the loudspeaker and headphone output channels:
nas VOICE functionality must be enabled by writing a bass/treble, loudness, balance, and volume. A square
"license key" into the MSP 34x0G. For information on wave beeper can be added to the loudspeaker and
how to obtain this license key from Micronas, please headphone channel. The loudspeaker channel addi-
contact your Micronas sales representative. tionally performs: equalizer (not simultaneously with
bass/treble), spatial effects, and a subwoofer cross-
over filter.
2.5.4. Automatic Volume Correction (AVC)
Different sound sources (e.g. terrestrial channels, SAT 2.5.6. Subwoofer Output
channels, or SCART) fairly often do not have the same
volume level. Advertisements during movies usually The subwoofer signal is created by combining the left
have a higher volume level than the movie itself. This and right channels directly behind the loudness block
results in annoying volume changes. The Automatic using the formula (L+R)/2. Due to the division by 2, the
Volume Correction (AVC) solves this problem by D/A converter will not be overloaded, even with full
equalizing the volume level. scale input signals. The subwoofer signal is filtered by
a third-order low-pass with programmable corner fre-
To prevent clipping, the AVCs gain decreases quickly quency followed by a level adjustment. At the loud-
in dynamic boost conditions. To suppress oscillation speaker channels, a complementary high-pass filter
effects, the gain increases rather slowly for low level can be switched on. Subwoofer and loudspeaker out-
inputs. The decay time is programmable by means of put use the same volume (Loudspeaker Volume Reg-
the AVC register (see page 33). ister).
output level
15
[dBr]
20
18
25
24
Fig. 24: Simplified AVC characteristics Fig. 25: Frequency response of subwoofer filter
0: sharp edge
1: medium edge
2: soft edge
3: very soft edge
The quasi-peak readout register can be used to read Micronas BASS exploits the psychoacoustic phenome-
out the quasi-peak level of any input source. The fea- non of the missing fundamental. Adding harmonics of
ture is based on following filter time constants: the frequency components below the cutoff frequency
gives the impression of actually hearing the low fre-
attack time: 1.3 ms quency fundamental. In other words: The listener has
decay time: 37 ms the impression that a loudspeaker system seems to
reproduce frequencies although physically not possi-
ble.
2.5.8. Micronas BASS (MB)
Amplitude (db)
The Micronas Bass system extends the frequency
range of loudspeakers or headphones.
Low frequency signals can be boosted while the output Several parameters allow tuning the characteristics of
signal amplitude is measured. If the amplitude comes Micronas BASS according to the TV loudspeaker, the
close to a definable limit, the gain is reduced automati- cabinet, and personal preferences (see Table 311).
cally in dynamic Volume mode. Therefore, the system For more detailed information on how to set up MB,
adapts to the signal amplitude which is really present please refer to the corresponding application note.
at the output of the MSP device. Clipping effects are
avoided.
2.6. SCART Signal Routing
Amplitude
(db)
If the MSP 34x0G is switched off by first pulling For the ASTRA Digital Radio System (ADR), the
STANDBYQ low and then (after >1 s delay) switching MSP 3400G, MSP 3410G, and MSP 3450G performs
off DVSUP and AVSUP, but keeping AHVSUP preprocessing such as carrier selection and filtering.
(Stand-by-mode), the SCART switches maintain Via the 3-line ADR-bus, the resulting signals are trans-
their position and function. This allows the copying ferred to the DRP 3510A coprocessor, where the
from SCART-input to SCART-output in the TV sets source decoding is performed. To be prepared for an
stand-by mode. upgrade to ADR with an additional DRP board, the fol-
lowing lines of MSP 34x0G should be provided on a
In case of power on or starting from stand-by (switch- feature connector:
ing on the DVSUP and AVSUP, RESETQ going high
AUD_CL_OUT
2 ms later), all internal registers except the ACB regis-
ter (see page 41) are reset to the default configuration I2S_DA_IN1 or I2S_DA_IN2
(see Table 35 on page 20). The reset position of the
I2S_DA_OUT
ACB register becomes active after the first I2C trans-
mission into the Baseband Processing part. By trans- I2S_WS
mitting the ACB register first, the reset state can be
redefined. I2S_CL
ADR_CL, ADR_WS, ADR_DA
2.7. I2S Bus Interface For more details, please refer to the DRP 3510A data
sheet.
The MSP 34x0G has a synchronous master/slave
input/output interface running on 32 kHz.
2.9. Digital Control I/O Pins and
The interface accepts two formats: Status Change Indication
1. I2S_WS changes at the word boundary
The static level of the digital input/output pins
2. I2S_WS changes one I2S-clock period before the D_CTR_I/O_0/1 is switchable between HIGH and
word boundaries. LOW via the I2C-bus by means of the ACB register
(see page 41). This enables the controlling of external
All I2S options are set by means of the MODUS and hardware switches or other devices via I2C-bus.
the I2S_CONFIGURATION registers.
The digital input/output pins can be set to high imped-
The I2S bus interface consists of five pins: ance by means of the MODUS register (see page 26).
I2S_DA_IN1, I2S_DA_IN2: In this mode, the pins can be used as input. The cur-
I2S serial data input: 16, 18....32 bits per sample rent state can be read out of the STATUS register (see
page 28).
I2S_DA_OUT:
I2S serial data output: 16, 18...32 bits per sample Optionally, the pin D_CTR_I/O_1 can be used as an
interrupt request signal to the controller, indicating any
I2S_CL:
changes in the read register STATUS. This makes poll-
I2S serial clock
ing unnecessary, I2C bus interactions are reduced to a
I2S_WS: minimum (see STATUS register on page 28 and
I2S word strobe signal defines the left and right MODUS register on page 26).
sample
If the MSP 34x0G serves as the master on the I2S 2.10. Clock PLL Oscillator and Crystal Specifications
interface, the clock and word strobe lines are driven by
the IC. In this mode, only 16 or 32 bits per sample can The MSP 34x0G derives all internal system clocks
be selected. In slave mode, these lines are input to the from the 18.432 MHz oscillator. In NICAM or in I2S-
IC and the MSP clock is synchronized to 576 times the Slave mode, the clock is phase-locked to the corre-
I2S_WS rate (32 kHz). NICAM operation is not possi- sponding source. Therefore, it is not possible to use
ble in slave mode. NICAM and I2S-Slave mode at the same time.
An I2S timing diagram is shown in Fig. 427 on For proper performance, the MSP clock oscillator
page 73. requires a 18.432 MHz crystal. Note that for the
phase-locked modes (NICAM, I2S-Slave), crystals with
tighter tolerance are required.
3. Control Interface response time is about 0.3 ms. If the MSP cannot
accept another byte of data (e.g. while servicing an
3.1. I2C Bus Interface internal interrupt), it holds the clock line I2C_CL low to
force the transmitter into a wait state. The I2C Bus
The MSP 34x0G is controlled via the I2C bus slave Master must read back the clock line to detect when
interface. the MSP is ready to receive the next I2C transmission.
The positions within a transmission where this may
The IC is selected by transmitting one of the happen are indicated by Wait in Section 3.1.3. The
MSP 34x0G device addresses. In order to allow up to maximum wait period of the MSP during normal opera-
three MSP ICs to be connected to a single bus, an tion mode is less than 1 ms.
address select pin (ADR_SEL) has been implemented.
With ADR_SEL pulled to high, low, or left open, the
MSP 34x0G responds to different device addresses. A 3.1.1. Internal Hardware Error Handling
device address pair is defined as a write address and
a read address (see Table 31). In case of any hardware problems (e.g. interruption of
the power supply of the MSP), the MSPs wait period is
Writing is done by sending the write device address, extended to 1.8 ms. After this time period elapses, the
followed by the subaddress byte, two address bytes, MSP releases data and clock lines.
and two data bytes.
Reading is done by sending the write device address, Indication and solving the error status:
followed by the subaddress byte and two address
bytes. Without sending a stop condition, reading of the To indicate the error status, the remaining acknowl-
addressed data is completed by sending the device edge bits of the actual I2C-protocol will be left high.
read address and reading two bytes of data. Additionally, bit[14] of CONTROL is set to one. The
MSP can then be reset via the I2C bus by transmitting
Refer to Section 3.1.3. for the I2C bus protocol and to the RESET condition to CONTROL.
Section 3.4. Programming Tips on page 43 for pro-
posals of MSP 34x0G I2C telegrams. See Table 32
for a list of available subaddresses. Indication of reset:
Besides the possibility of hardware reset, the MSP can Any reset, even caused by an unstable reset line etc.,
also be reset by means of the RESET bit in the CON- is indicated in bit[15] of CONTROL.
TROL register by the controller via I2C bus.
A general timing diagram of the I2C bus is shown in
Due to the architecture of the MSP 34x0G, the IC can- Fig. 425 on page 71.
not react immediately to an I2C request. The typical
CONTROL 0000 0000 00 Read/Write Write: Software reset of MSP (see Table 33)
Read: Hardware error status of MSP
CONTROL 00hex RESET status after last reading of Internal hardware status: not of interest
CONTROL: 0 : no error occured
1 : internal error occured
0 : no reset occured
1 : reset occured
Reading of CONTROL will reset the bits[15,14] of CONTROL. After Power-on, bit[15] of CONTROL will be set; it must be
read once to be reset.
S write Wait ACK sub-addr ACK addr-byte ACK addr-byte ACK data-byte ACK data-byte ACK P
device high low high low
address
S write Wait ACK sub-addr ACK addr-byte ACK addr-byte ACK S read Wait ACK data-byte- ACK data-byte NAK P
device high low device high low
address address
S write Wait ACK 00hex ACK S read Wait ACK data-byte- ACK data-byte NAK P
device device high low
address address
1
I2C_DA
0
S P
I2C_CL
Fig. 31: I2C bus protocol (MSB first; data must be stable while clock is high)
<daw 00 <dar dd dd> read data from Write and read registers are 16 bit wide, whereby the
CONTROL register MSB is denoted bit[15]. Transmissions via I2C bus
<daw 11 aa aa <dar dd dd> read data from demodulator have to take place in 16-bit words (two byte transfers, with
<daw 13 aa aa <dar dd dd> read data from DSP the most significant byte transferred first). All write regis-
ters, except the demodulator write registers are readable.
3.1.4.4. Examples Unused parts of the 16-bit write registers must be zero.
Addresses not given in this table must not be
<80 00 80 00> RESET MSP statically accessed.
<80 00 00 00> Clear RESET
<80 10 00 30 00 01> Automatic Sound Select = ON For reasons of software compatibility to the
<80 10 00 20 00 03> Set demodulator to stand. 03hex MSP 34xxD, a Manual/Compatibility Mode is available.
<80 11 02 00 <81 dd dd> Read STATUS More read and write registers together with a detailed
<80 12 00 08 01 20> Set loudspeaker channel description can be found in Appendix B: Manual/Com-
source to Stereo or A/B and patibility Mode on page 87.
Matrix to Stereo
(transparent mode)
Write Register Address Bits Description and Adjustable Range Reset See
(hex) Page
Volume loudspeaker channel 00 00 [15:8] [+12 dB ... 114 dB, MUTE] MUTE 32
Balance loudspeaker channel [L/R] 00 01 [15:8] [0...100 / 100% and 100 / 0...100%] 100%/100% 33
[127...0 / 0 dB and 0 / 128...0 dB]
Volume headphone channel 00 06 [15:8] [+12 dB ... 114 dB, MUTE] MUTE 32
Volume SCART1 output channel 00 07 [15:8] [+12 dB ... 114 dB, MUTE] MUTE 40
Loudspeaker source select 00 08 [15:8] [FM/AM, NICAM, SCART, I2S1, I2S2] FM/AM 31
Headphone source select 00 09 [15:8] [FM/AM, NICAM, SCART, I2S1, I2S2] FM/AM 31
I2S source select 00 0B [15:8] [FM/AM, NICAM, SCART, I2S1, I2S2] FM/AM 31
2
I S channel matrix [7:0] [SOUNDA, SOUNDB, STEREO, MONO...] SOUNDA 31
Quasi-peak detector source select 00 0C [15:8] [FM/AM, NICAM, SCART, I2S1, I2S2] FM/AM 31
Prescale NICAM 00 10 [15:8] [00hex ... 7Fhex] (MSP 3410G, MSP 3450G only) 00hex 30
Write Register Address Bits Description and Adjustable Range Reset See
(hex) Page
Subwoofer complementary high-pass [3:0] [off, on, Micronas BASS to Main] off 38
Balance headphone channel [L/R] 00 30 [15:8] [0...100 / 100% and 100 / 0...100%] 100 %/100 % 33
[127...0 / 0 dB and 0 / 128...0 dB]]
Balance mode headphone [7:0] [Linear mode / logarithmic mode] linear mode
Volume SCART2 output channel 00 40 [15:8] [+12 dB ... 114 dB, MUTE] 00hex 40
Micronas BASS Effect Strength 00 68 [15:8] [0 dB ... 127 dB, off] off 38
Micronas BASS Low Pass Corner 00 6B [15:8] [50 Hz ... 300 Hz] 0 Hz 39
Frequency
Micronas BASS High Pass Corner 00 6C [15:8] [20 Hz ... 300 Hz] 0 Hz 39
Frequency
STANDARD RESULT 00 7E [15:0] Result of Automatic Standard Detection (see Table 38 on page 25) 28
STATUS 02 00 [15:0] Monitoring of internal settings e.g. Stereo, Mono, Mute etc. . 28
Quasi peak readout left 00 19 [15:0] [00hex ... 7FFFhex]16 bit twos complement 42
Quasi peak readout right 00 1A [15:0] [00hex ... 7FFFhex]16 bit twos complement 42
MSP Standard Code TV Sound Standard Sound Carrier MSP 34x0G Version
(Data in hex) Frequencies in MHz
Standard Selection
00 09 L -NICAM-AM 6.5/5.85
00 0A I -NICAM-FM 6.0/6.552
00 21 BTSC-Mono + SAP
The TV sound standard of the MSP 34x0G demodula- A general refresh of the STANDARD SELECT register
tor is determined by the STANDARD SELECT register. is not allowed. However, the following method
There are two ways to use the STANDARD SELECT enables watching the MSP 34x0G alive status and
register: detection of accidental resets (only versions B6 and
later):
Setting up the demodulator for a TV sound standard
by sending the corresponding standard code with a After Power-on, bit[15] of CONTROL will be set; it
single I2C bus transmission. must be read once to enable the reset-detection
feature.
Starting the Automatic Standard Detection for ter-
restrial TV standards. This is the most comfortable Reading of the CONTROL register and checking
way to set up the demodulator. Within 0.5 s, the the reset indicator bit[15] .
detection and setup of the actual TV sound standard
If bit[15] is 0, any refresh of the STANDARD
is performed. The detected standard can be read
SELECT register is not allowed.
out of the STANDARD RESULT register by the con-
trol processor. This feature is recommended for the If bit[15] is 1, indicating a reset, a refresh of the
primary setup of a TV set. Outputs should be muted STANDARD SELECT register and all other MSPG
during Automatic Standard Detection. registers is required.
For reasons of software compatibility to the As long as the STANDARD RESULT register contains
MSP 34xxD, a Manual/Compatibility mode is available. a value greater than 07 FFhex, the Automatic Standard
A detailed description of this mode can be found on Detection is still active. During this period, the MODUS
page 87. and STANDARD SELECT register must not be written.
The STATUS register will be updated when the Auto-
matic Standard Detection has finished.
Example:
The MSPs 3420G and 3440G will detect a B/G-NICAM
signal as standard 3 and will switch to the analog FM-
Mono sound.
B/G-FM 0003hex
B/G-NICAM 0008hex
I 000Ahex
FM-Radio 0040hex
Contains all user relevant internal information about the status of the MSP
bit[15:10] undefined
bit[8] 0/1 1 indicates bilingual sound mode or SAP present
(internally evaluated from received analog or digital
identification signals)
bit[7] 0/1 1 indicates independent mono sound (only for
NICAM)
bit[6] 0/1 mono/stereo indication
(internally evaluated from received analog or digital
identification signals)
bit[5,9] 00 analog sound standard (FM or AM) active
01 this pattern will not occur
10 digital sound (NICAM) available
11 bad reception condition of digital sound (NICAM) due
to:
a. high error rate
b. unimplemented sound code
c. data transmission only
bit[4] 0/1 low/high level of digital I/O pin D_CTR_I/O_1
bit[3] 0/1 low/high level of digital I/O pin D_CTR_I/O_0
bit[2] 0 detected secondary carrier (2nd A2 or SAP sub-carrier)
1 no secondary carrier detected
bit[1] 0 detected primary carrier (Mono or MPX carrier)
1 no primary carrier detected
bit[0] undefined
PREPROCESSING
For all FM modes except satellite FM and AM-mode, the combinations of pres-
cale value and FM deviation listed below lead to internal full scale with 1 kHz
test signal and 50 s emphasis.
FM mode
bit[15:8] 7Fhex 28 kHz FM deviation
48hex 50 kHz FM deviation
30hex 75 kHz FM deviation
24hex 100 kHz FM deviation
18hex 150 kHz FM deviation
13hex 180 kHz FM deviation (limit)
FM very high deviation mode (HDEV3, MSP Standard Code = 6 and Dhex)
bit[15:8] 20hex 450 kHz FM deviation
1Ahex 540 kHz FM deviation (limit)
In case of Automatic Sound Select = on, the FM Matrix Mode is set automati-
cally. Writing to the FM/AM prescale register (00 0Ehex high part) is still allowed.
In order not to disturb the automatic process, the low part of any I2C transmis-
sion to this register is ignored. Therefore, any FM-Matrix readback values may
differ from data written previously.
In case of Automatic Sound Select = off, the FM Matrix Mode must be set as
shown in Table 617 of Appendix B.
Defines the input prescale value for the digital NICAM signal
bit[15:8] 00hex ... 7Fhex prescale gain
examples:
00hex off
20hex 0 dB gain
5Ahex 9 dB gain (recommendation)
7Fhex +12 dB gain (maximum gain)
Defines the input prescale value for digital I2S input signals
bit[15:8] 00hex ... 7Fhex prescale gain
examples:
00hex off
10hex 0 dB gain (recommendation)
7Fhex +18 dB gain (maximum gain)
Defines the input prescale value for the analog SCART input signal
bit[15:8] 00hex ... 7Fhex prescale gain
examples:
00hex off
19hex 0 dB gain (2 VRMS input leads to digital full scale)
7Fhex +14 dB gain (400 mVRMS input leads to digital full scale)
Source for:
00 08hex Loudspeaker Output SRC_MAIN
00 09hex Headphone Output SRC_AUX
00 0Ahex SCART1 DA Output SRC_SCART1
00 41hex SCART2 DA Output SRC_SCART2
00 0Bhex I2S Output SRC_I2S
00 0Chex Quasi-Peak Detector SRC_QPEAK
In Automatic Sound Select mode, the demodulator source channels are set
according to Table 22. Therefore, the matrix modes of the corresponding out-
put channels should be set to Stereo (transparent).
Note: AVC should not be used in any Dolby Prologic mode (with DPL 35xx),
except in PANORAMA or 3D-PANORAMA mode, when only the loudspeaker
output is active.
Positive balance settings reduce the left channel without affecting the right
channel; negative settings reduce the right channel leaving the left channel
unaffected.
Higher resolution is possible: an LSB step in the normal range results in a gain
step of about 1/8 dB, in the extended range about 1/4 dB.
With positive bass settings, internal clipping may occur even with overall volume
less than 0 dB. This will lead to a clipped output signal. Therefore, it is not rec-
ommended to set bass to a value that, in conjunction with volume, would result
in an overall positive gain.
Higher resolution is possible: an LSB step results in a gain step of about 1/8 dB.
With positive treble settings, internal clipping may occur even with overall vol-
ume 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.
Higher resolution is possible: an LSB step results in a gain step of about 1/8 dB.
With positive equalizer settings, internal clipping 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 vol-
ume, would result in an overall positive gain.
Loudness increases the volume of low- and high-frequency signals, while keep-
ing the amplitude of the reference frequency constant. The intended loudness
has to be set according to the actual volume setting. Because loudness intro-
duces gain, it is not recommended to set loudness to a value that, in conjunction
with volume, would result in an overall positive gain.
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 vol-
ume is shifted from 1 kHz to 2 kHz.
In mode A (low byte = 00hex), the spatial effect depends on the source mode. If
the incoming signal is mono, 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 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 B, only Stereo Basewidth Enlargement is effective. For mono input sig-
nals, the Pseudo Stereo Effect has to be switched on.
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 0hex yields a flat response for center signals (L = R), but a high-pass
function for L or R only signals. A value of 6hex has a flat response for L or R
only signals, but a low-pass function for center signals. By using 8hex, the fre-
quency response is automatically adapted to the sound material by choosing an
optimal high-pass gain.
If Micronas BASS is added onto the main channel, this register should be set to
00hex
An adjustable subwoofer filter characterisitic with four different filter sets can be
selected. Due to the complementary filter design, the output of high- and low-
pass filter add up to 0db at the crossover region for all filter sets.
The Micronas BASS effect strength can be adjusted in 1dB steps. A value of
44hex will yield a medium Micronas BASS effect.
The Micronas BASS Amplitude Limit defines the maximum allowed amplitude at
the output of the MB relative to 0 dbFS. If the amplitude exceeds MB_LIM, the
gain of the MB is automatically reduced. Note that the Volume Clipping Mode
must be set to dynamic (see page 32).
The MB lowpass corner frequency (range 50...300 Hz) defines the upper corner
frequency of the MB bandpass filter. Recommended values are the same as for
the MB highpass corner frequency (MB_HP).
The MB highpass corner frequency defines the lower corner frequency of the
MB bandpass filter. The highpass filter avoids loading the loudspeakers with low
frequency components that are below the speakers cut off frequency. Recom-
mended values for subwoofer systems are around 5dec (=50 Hz), for regular TV
sets around 10dec (=100 Hz).
Defines the level of the digital output pins and the position of the SCART
switches
bit[15] 0/1 low/high of digital output pin D_CTR_I/O_1
(MODUS[3]=0)
bit[14] 0/1 low/high of digital output pin D_CTR_I/O_0
(MODUS[3]=0)
bit[13:5] SCART DSP Input Select
xxxx00xx0 SCART1 to DSP input (RESET position)
xxxx01xx0 MONO to DSP input (set Sound A Mono in the channel
matrix mode for the corresponding output channels)
xxxx10xx0 SCART2 to DSP input
xxxx11xx0 SCART3 to DSP input
xxxx00xx1 SCART4 to DSP input
xxxx11xx1 mute DSP input
bit[13:5] SCART1 Output Select
xx00xxx0x SCART3 input to SCART1 output (RESET position)
xx01xxx0x SCART2 input to SCART1 output
xx10xxx0x MONO input to SCART1 output
xx11xxx0x SCART1 DA to SCART1 output
xx00xxx1x SCART2 DA to SCART1 output
xx01xxx1x SCART1 input to SCART1 output
xx10xxx1x SCART4 input to SCART1 output
xx11xxx1x mute SCART1 output
bit[13:5] SCART2 Output Select
00xxxx0xxSCART1 DA to SCART2 output (RESET position)
01xxxx0xxSCART1 input to SCART2 output
10xxxx0xxMONO input to SCART2 output
00xxxx1xxSCART2 DA to SCART2 output
01xxxx1xxSCART2 input to SCART2 output
10xxxx1xxSCART3 input to SCART2 output
11xxxx1xxSCART4 input to SCART2 output
11xxxx0xxmute SCART2 output
bit[4:0] must be zero
The RESET position becomes active at the time of the first write transmission
on the control bus to the audio processing part. By writing to the ACB register
first, the RESET state can be redefined.
BEEPER
To avoid compatibility problems with MSP 3410B and MSP 34x0D, an offset of
40hex is added to the ROM version code of the chips imprint.
This section describes the preferred method for initial- Initialization of the MSP 34x0G according to these list-
izing the MSP 34x0G. The initialization is grouped into ings reproduces sound of the selected standard on the
four sections: loudspeaker output. All numbers are hexadecimal. The
examples have the following structure:
SCART Signal Path (analog signal path)
1. Perform an I2C controlled reset of the IC.
Demodulator
2. Write MODUS register
SCART and I2S Inputs
(with Automatic Sound Select).
Output Channels
3. Set Source Selection for loudspeaker channel
(with matrix set to STEREO).
See Fig. 21 on page 9 for a complete signal flow.
4. Set Prescale
(FM and/or NICAM and dummy FM matrix).
SCART Signal Path
5. Write STANDARD SELECT register.
1. Select analog input for the SCART baseband pro-
6. Set Volume loudspeaker channel to 0 dB.
cessing (SCART DSP Input Select) by means of the
ACB register.
2. Select the source for each analog SCART output 3.5.1. B/G-FM (A2 or NICAM)
(SCART Output Select) by means of the ACB regis- <80 00 80 00> // Softreset
ter. <80 00 00 00>
<80 10 00 30 20 03> // MODUS-Register: Automatic = on
<80 12 00 08 03 20> // Source Sel. = (St or A) & Ch. Matr. = St
Demodulator
<80 12 00 0E 24 03> // FM/AM-Prescale = 24hex,
FM-Matrix = MONO/SOUNDA
For a complete setup of the TV sound processing from
analog IF input to the source selection, the following <80 12 00 10 5A 00> // NICAM-Prescale = 5Ahex
steps must be performed: <80 10 00 20 00 03> // Standard Select: A2 B/G or NICAM B/G
or
1. Set MODUS register to the preferred mode and <80 10 00 20 00 08>
Sound IF input. <80 12 00 00 73 00> // Loudspeaker Volume 0 dB
:ULWH02'865HJLVWHU:
([DPSOH for the essential bits:
>@ $XWRPDWLF6RXQG6HOHFW RQ
[1] = 1 Enable interrupt if STATUS changes
[8] = 0 ANA_IN1+ is selected
Define Preference for Automatic Standard
Detection:
[12] = 0 If 6.5 MHz, set SECAM-L
[14:13] = 3 Ignore 4.5 MHz carrier
:ULWH6285&(6(/(&76HWWLQJV
([DPSOH
set loudspeaker Source Select to "Stereo or A"
set headphone Source Select to "Stereo or B"
set SCART_Out Source Select to "Stereo or A/B"
Write FM/AM-Prescale
Write NICAM-Prescale
:ULWHLQWR
67$1'$5'6(/(&75HJLVWHU
(Start Automatic Standard Detection)
no
expecting MSPG-interrupt
,QFDVHRI063*
,QWHUUXSWWR&RQWUROOHU Read STATUS
Adjust TV-Display
Fig. 32: Software flow diagram for a Minimum demodulator setup for a European Multistandard TV set applying the
Automatic Sound Select feature
4. Specifications
Fig. 41:
PSDIP64-1: Plastic Shrink Dual In-line Package, 64 leads, 750 mil
Ordering code: PP
Weight approximately 8.77 g
Fig. 42:
PSDIP52-1: Plastic Shrink Dual In-line Package, 52 leads, 600 mil
Ordering code: PO1)
Weight approximately 5.13 g
1)
Micronas delivers two types of PSDIP52 packages (-1, -2). The packages have slightly different outline dimen-
sions, but are considered identical. For logistics reasons, the customer cannot choose the package to be delivered.
Fig. 43:
PSDIP52-2: Plastic Shrink Dual In-line Package, 52 leads, 600 mil
Ordering code: PO1)
Weight approximately 5.92 g
1)
Micronas delivers two types of PSDIP52 packages (-1, -2). The packages have slightly different outline dimen-
sions, but are considered identical. For logistics reasons, the customer cannot choose the package to be delivered.
Fig. 44:
PMQFP80-11: Plastic Metric Quad Flat Package, 80 leads, 14 20 2.7 mm3, high standoff
Ordering code: QA
Weight approximately 1.68 g
Fig. 45:
PMQFP64-2: Plastic Metric Quad Flat Package, 64 leads, 10 10 2 mm3
Ordering code: QI
Weight approximately 0.5 g
1 64 8 NC LV Not connected
18 13 21 19 NC LV Not connected
19 14 22 NC LV Not connected
20 15 23 NC LV Not connected
22 NC LV Not connected
23 NC LV Not connected
29 22 30 NC LV Not connected
31 24 32 NC LV Not connected
32 NC LV Not connected
41 NC LV Not connected
42 NC LV Not connected
59 NC LV Not connected
63 NC LV Not connected
64 NC LV Not connected
67 50 58 47 ANA_IN1+ IN LV IF input 1
73 56 64 1 TP LV Test pin
75 58 2 NC LV Not connected
76 59 3 NC LV Not connected
Pin numbers refer to the PMQFP80-11 package. Pins 24, 25, DACA_R/L Headphone Outputs
(Fig. 420)
Pin 1, NC Pin not connected. Output of the headphone signal. A 1-nF capacitor to
AHVSS must be connected to these pins. The DC off-
Pin 2, I2C_CL I2C Clock Input/Output (Fig. 417) set on these pins depends on the selected headphone
Via this pin, the I2C-bus clock signal has to be sup- volume.
plied. The signal can be pulled down by the MSP in
case of wait conditions. Pin 26, VREF2 Reference Ground 2
Reference analog ground. This pin must be connected
Pin 3, I2C_DA I2C Data Input/Output (Fig. 417) separately to ground (AHVSS). VREF2 serves as a
Via this pin, the I2C-bus data is written to or read from clean ground and should be used as the reference for
the MSP. analog connections to the loudspeaker and head-
phone outputs.
Pin 4, I2S_CL I2S Clock Input/Output (Fig. 418)
Clock line for the I2S bus. In master mode, this line is Pins 27, 28, DACM_R/L Loudspeaker Outputs
driven by the MSP; in slave mode, an external I2S (Fig. 420)
clock has to be supplied. Output of the loudspeaker signal. A 1-nF capacitor to
AHVSS must be connected to these pins. The DC off-
Pin 5, I2S_WS I2S Word Strobe Input/Output set on these pins depends on the selected loud-
(Fig. 418) speaker volume.
Word strobe line for the I2S bus. In master mode, this
line is driven by the MSP; in slave mode, an external Pin 29, NC Pin not connected.
I2S word strobe has to be supplied.
Pin 30, DACM_SUB Subwoofer Output (Fig. 420)
Pin 6, I2S_DA_OUT I2S Data Output (Fig. 422) Output of the subwoofer signal. A 1-nF capacitor to
Output of digital serial sound data of the MSP on the AHVSS must be connected to this pin. Due to the low
I2S bus. frequency content of the subwoofer output, the value
of the capacitor may be increased for better suppres-
Pin 7, I2S_DA_IN1 I2S Data Input 1 (Fig. 414) sion of high-frequency noise. The DC offset on this pin
First input of digital serial sound data to the MSP via depends on the selected loudspeaker volume.
the I2S bus.
Pins 31, 32, NC Pin not connected.
Pin 8, ADR_DA ADR Bus Data Output (Fig. 422)
Output of digital serial data to the DRP 3510A via the Pins 33, 34, SC2_OUT_R/L SCART2 Outputs
ADR bus. (Fig. 421)
Output of the SCART2 signal. Connections to these
Pin 9, ADR_WS ADR Bus Word Strobe Output pins must use a 100- series resistor and are intended
(Fig. 422) to be AC-coupled.
Word strobe output for the ADR bus.
Pin 35, VREF1 Reference Ground 1
Pin 10, ADR_CL ADR Bus Clock Output (Fig. 422) Reference analog ground. This pin must be connected
Clock line for the ADR bus. separately to ground (AHVSS). VREF1 serves as a
clean ground and should be used as the reference for
Pins 11, 12, 13, DVSUP* Digital Supply Voltage analog connections to the SCART outputs.
Power supply for the digital circuitry of the MSP. Must
be connected to a +5 V or +3.3 V power supply. Pins 36, 37, SC1_OUT_R/L SCART1 Outputs
(Fig. 421)
Pins 14, 15, 16, DVSS* Digital Ground Output of the SCART1 signal. Connections to these
Ground connection for the digital circuitry of the MSP. pins must use a 100- series resistor and are intended
to be AC-coupled.
Pin 17, I2S_DA_IN2 I2S Data Input 2 (Fig. 414)
Second input of digital serial sound data to the MSP
via the I2S bus.
Pin 38, CAPL_A Volume Capacitor Headphone Pins 53, 54, SC2_IN_L/R SCART2 Inputs
(Fig. 423) (Fig. 413)
A 10-F capacitor to AHVSUP must be connected to The analog input signal for SCART2 is fed to this pin.
this pin. It serves as a smoothing filter for headphone Analog input connection must be AC-coupled.
volume changes in order to suppress audible plops.
The value of the capacitor can be lowered to 1-F if Pin 55, ASG Analog Shield Ground
faster response is required. The area encircled by the Analog ground (AHVSS) should be connected to this
trace lines should be minimized; keep traces as short pin to reduce cross-coupling between SCART inputs.
as possible. This input is sensitive for magnetic induc-
tion. Pins 56, 57, SC1_IN_L/R SCART1 Inputs
(Fig. 413)
Pin 39, AHVSUP* Analog Power Supply High Volt- The analog input signal for SCART1 is fed to this pin.
age Analog input connection must be AC-coupled.
Power is supplied via this pin for the analog circuitry of
the MSP (except IF input). This pin must be connected Pin 58, VREFTOP Reference Voltage IF A/D Con-
to the +8 V supply. verter (Fig. 415)
Via this pin, the reference voltage for the IF A/D con-
Pin 40, CAPL_M Volume Capacitor Loudspeaker verter is decoupled. It must be connected to AVSS
(Fig. 423) pins with a 10-F and a 100-nF capacitor in parallel.
A 10-F capacitor to AHVSUP must be connected to Traces must be kept short.
this pin. It serves as a smoothing filter for loudspeaker
volume changes in order to suppress audible plops. Pin 59, NC Pin not connected.
The value of the capacitor can be lowered to 1 F if
faster response is required. The area encircled by the Pin 60, MONO_IN Mono Input (Fig. 413)
trace lines should be minimized; keep traces as short The analog mono input signal is fed to this pin. Analog
as possible. This input is sensitive for magnetic induc- input connection must be AC-coupled.
tion.
Pins 61, 62, AVSS* Ground for Analog Power
Pins 41, 42, NC Pins not connected. Supply Voltage
Ground connection for the analog IF input circuitry of
Pins 43, 44, AHVSS* Ground for Analog Power the MSP.
Supply High Voltage
Ground connection for the analog circuitry of the MSP Pins 63, 64, NC Pins not connected.
(except IF input).
Pins 65, 66, AVSUP* Analog Power Supply Voltage
Pin 45, AGNDC Internal Analog Reference Voltage Power is supplied via this pin for the analog IF input
This pin serves as the internal ground connection for circuitry of the MSP. This pin must be connected to the
the analog circuitry (except IF input). It must be con- +5 V supply.
nected to the VREF pins with a 3.3-F and a 100-nF
capacitor in parallel. This pins shows a DC level of typ- Pin 67, ANA_IN1+ IF Input 1 (Fig. 415)
ically 3.73 V (with AHVSUP = 8 V). The analog sound IF signal is supplied to this pin.
Inputs must be AC-coupled. This pin is designed as
Pin 46, NC Pin not connected. symmetrical input: ANA_IN1+ is internally connected
to one input of a symmetrical op amp, ANA_IN- to the
Pins 47, 48, SC4_IN_L/R SCART4 Inputs other.
(Fig. 413)
The analog input signal for SCART4 is fed to this pin. Pin 68, ANA_IN IF Common (Fig. 415)
Analog input connection must be AC-coupled. This pins serves as a common reference for ANA_IN1/
2+ inputs.
Pin 49, ASG Analog Shield Ground
Analog ground (AHVSS) should be connected to this Pin 69, ANA_IN2+ IF Input 2 (Fig. 415)
pin to reduce cross-coupling between SCART inputs. The analog sound if signal is supplied to this pin.
Inputs must be AC-coupled. This pin is designed as
Pins 50, 51, SC3_IN_L/R SCART3 Inputs symmetrical input: ANA_IN2+ is internally connected
(Fig. 413) to one input of a symmetrical op amp, ANA_IN to the
The analog input signal for SCART3 is fed to this pin. other.
Analog input connection must be AC-coupled.
Pin 70, TESTEN Test Enable Pin (Fig. 411)
Pin 52, ASG Analog Shield Ground This pin enables factory test modes. For normal opera-
Analog ground (AHVSS) should be connected to this tion, it must be connected to ground.
pin to reduce cross-coupling between SCART inputs.
* Application Note:
All ground pins should be connected to one low-resis-
tive ground plane. All supply pins should be connected
separately with short and low-resistive lines to the
power supply. Decoupling capacitors from DVSUP to
DVSS, AVSUP to AVSS, and AHVSUP to AHVSS are
recommended as closely as possible to these pins.
Decoupling of DVSUP and DVSS is most important.
We recommend using more than one capacitor. By
choosing different values, the frequency range of
active decoupling can be extended. In our application
boards we use: 220 pF, 470 pF, 1.5 nF, and 10 F. The
capacitor with the lowest value should be placed near-
est to the DVSUP and DVSS pins.
AUD_CL_OUT 1 64 TP TP 1 52 XTAL_OUT
NC 2 63 XTAL_OUT AUD_CL_OUT 2 51 XTAL_IN
NC 3 62 XTAL_IN D_CTR_I/O_1 3 50 TESTEN
D_CTR_I/O_1 4 61 TESTEN D_CTR_I/O_0 4 49 ANA_IN2+
D_CTR_I/O_0 5 60 ANA_IN2+ ADR_SEL 5 48 ANA_IN
ADR_SEL 6 59 ANA_IN STANDBYQ 6 47 ANA_IN1+
STANDBYQ 7 58 ANA_IN+ I2C_CL 7 46 AVSUP
NC 8 57 AVSUP I2C_DA 8 45 AVSS
I2C_CL 9 56 AVSS I2S_CL 9 44 MONO_IN
I2C_DA 10 55 MONO_IN I2S_WS 10 43 VREFTOP
MSP 34x0G
I2S_CL 11 54 VREFTOP I2S_DA_OUT 11 42 SC1_IN_R
I2S_WS 12 53 SC1_IN_R I2S_DA_IN1 12 41 SC1_IN_L
I2S_DA_OUT 13 52 SC1_IN_L ADR_DA 13 40 SC2_IN_R
MSP 34x0G
SC2_IN_L ASG
SC2_IN_R SC3_IN_R
ASG SC3_IN_L
SC1_IN_L ASG
SC1_IN_R SC4_IN_R
VREFTOP SC4_IN_L
NC NC
MONO_IN AGNDC
AVSS AHVSS
AVSS AHVSS
NC NC
NC NC
64 63 62 61 60 59 58 57 56 55 54 53 52 51 50 49 48 47 46 45 44 43 42 41
AVSUP 65 40 CAPL_M
AVSUP 66 39 AHVSUP
ANA_IN1+ 67 38 CAPL_A
ANA_IN 68 37 SC1_OUT_L
ANA_IN2+ 69 36 SC1_OUT_R
TESTEN 70 35 VREF1
XTAL_IN 71 34 SC2_OUT_L
XTAL_OUT 72 33 SC2_OUT_R
TP 73
MSP 34x0G 32 NC
AUD_CL_OUT 74 31 NC
NC 75 30 DACM_SUB
NC 76 29 NC
D_CTR_I/O_1 77 28 DACM_L
D_CTR_I/O_0 78 27 DACM_R
ADR_SEL 79 26 VREF2
STANDBYQ 80 25 DACA_L
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24
NC DACA_R
I2C_CL NC
I2C_DA NC
I2S_CL RESETQ
I2S_WS NC
I2S_DA_OUT NC
I2S_DA_IN1 NC
ADR_DA I2S_DA_IN2
ADR_WS DVSS
ADR_CL DVSS
DVSUP DVSS
DVSUP DVSUP
SC2_IN_L ASG
SC2_IN_R SC3_IN_R
ASG SC3_IN_L
SC1_IN_L ASG
SC1_IN_R SC4_IN_R
VREFTOP SC4_IN_L
MONO_IN AGNDC
AVSS AHVSS
48 47 46 45 44 43 42 41 40 39 38 37 36 35 34 33
AVSUP 49 32 CAPL_M
ANA_IN1+ 50 31 AHVSUP
ANA_IN 51 30 CAPL_A
ANA_IN2+ 52 29 SC1_OUT_L
TESTEN 53 28 SC1_OUT_R
XTAL_IN 54 27 VREF1
XTAL_OUT 55 26 SC2_OUT_L
TP 56 25 SC2_OUT_R
AUD_CL_OUT 57
MSP 34x0G 24 NC
NC 58 23 DACM_SUB
NC 59 22 NC
D_CTR_I/OUT1 60 21 DACM_L
D_CTR_I/OUT0 61 20 DACM_R
ADR_SEL 62 19 VREF2
STANDBYQ 63 18 DACA_L
NC 64 17 DACA_R
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
I2C_CL RESETQ
I2C_DA NC
I2S_CL NC
I2S_WS NC
I2S_DA_OUT I2S_DA_IN2
I2S_DA_IN1 DVSS
ADR_DA DVSUP
ADR_WS ADR_CL
DVSUP
23 k
>300 k
DVSS
23 k
Fig. 410: Input Pin: RESETQ
GND
ADR_SEL
200 k
N
Fig. 411: Input Pin TESTEN
GND
24 k
3.75 V
DVSUP
P
Fig. 412: Input Pin: MONO_IN
N
GND
40 k
3.75 V
Fig. 418: Input/Output Pins:
I2S_CL, I2S_WS, D_CTR_I/O_1, D_CTR_I/O_0
Fig. 413: Input Pins: SC4-1_IN_L/R
ANA_IN1+
330 pF
ANA_IN2+
A
Fig. 419: Input/Output Pins:
D XTAL_IN, XTAL_OUT, AUD_CL_OUT
ANA_IN
VREFTOP
AHVSUP
0...1.2 mA
3.3 k
26 pF
120 k
300
3.75 V
DVSUP
P
N
GND
0...2 V
125 k
3.75 V
Abbreviations:
tbd = to be defined
vacant = not applicable
positive current values mean current flowing into the chip
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 conditions is not implied. Exposure to absolut max-
imum rating conditions for extended periods will affect device reliability.
All voltages listed are referenced to ground (0 V, VSS ), except where noted.
This device contains circuitry to protect the inputs and outputs against damage due to high static voltages or electric
fields; however, it is advised that normal precautions be taken to avoid application of any voltage higher than absolut
maximum-rated voltages to this high-impedance circuit.
Min. Max.
Min. Max.
Functional operation of the device beyond those indicated in the "Recommended Operating Conditions/Characteris-
tics" is not implied and will result in unpredictable behaviour of the device and may result in device destruction.
All voltages listed are referenced to ground (VSS = 0 V) except where noted.
Do not insert the device into a live socket. Instead, apply power by switching on the external power supply. For
power-up/-down sequences, see the instructions in Section 4.6.3.3. of this document.
Crystal Recommendations for Master-Slave Applications (MSP-clock must perform synchronization to I2S clock)
Crystal Recommendations for FM / NICAM Applications (No MSP-clock synchronization to I2S clock possible)
Crystal Recommendations for all analog FM/AM Applications (No MSP-clock synchronization to I2S clock possible)
Amplitude Recommendation for Operation with External Clock Input (Cload after reset typ. 22 pF)
Note: To minimize adjustment tolerances for all MSP-generations, it is strongly recommended to use the so-
called MSP-XTAL-REF ICs (available in all packages) for the capacitor adjustment.
4.6.3. Characteristics
TJ = Junction Temperature
Supply
ISUP1A First Supply Current (active) AHVSUP 17 25 mA Vol. Main and Aux = 0 dB
(AHVSUP = 8 V) 11 16 mA Vol. Main and Aux = -30dB
Clock
VACLKAC Audio Clock Output AC Voltage AUD_CL_OUT 1.2 1.8 Vpp load = 40 pF
VACLKDC Audio Clock Output DC Voltage 0.4 0.6 VSUP3 Imax = 0.2 mA
IADRSEL Input Current Address Select Pin 500 220 A UADR_SEL= DVSS
DVSUP
AVSUP
0.9DVSUP
t/ms
Internal
Reset High
Low
t/ms
1/FI2C
TI2C4 TI2C3
I2C_CL
I2C_DA as input
TI2COL2 TI2COL1
I2C_DA as output
ts_I2S I2SInput Setup Time I2S_CL 12 ns for details see Fig. 427
before Rising Edge of Clock I2S_DA_IN1/2 I2S bus timing diagram
1/FI2SWS
I2S_WS
MODUS[6] = 0
MODUS[6] = 1 Detail C
I2S_CL
Detail A
I2S_DA_IN R LSB L MSB L LSB R MSB R LSB L LSB
Detail B
I2S_DA_OUT R LSB L MSB L LSB R MSB R LSB L LSB
1/FI2SWS
I2S_WS
MODUS[6] = 0
MODUS[6] = 1 Detail C
I2S_CL
Detail A
I2S_DA_IN R LSB L MSB L LSB R MSB R LSB L LSB
Ts_I2S Th_I2S
Ts_I2S
I2S_DA_IN1/2
I2S_WS as INPUT
Td_I2S
Td_I2S
Analog Ground
Audio Analog-to-Digital-Converter
VAICL Analog Input Clipping Level for SCn_IN_s,1) 2.00 2.25 VRMS fsignal = 1 kHz
Analog-to-Digital- MONO_IN
Conversion
(AHVSUP = 8 V)
SCART Outputs
ASCtoSC Gain from Analog Input SCn_IN_s,1) 1.0 +0.5 dB fsignal = 1 kHz
to SCART Output MONO_IN
frSCtoSC Frequency Response from Analog SCn_OUT_s1) 0.5 +0.5 dB with resp. to 1 kHz
Input to SCART Output Bandwidth: 0 to 20000 Hz
VoutSC Signal Level at SCART Output SCn_OUT_s1) 1.8 1.9 2.0 VRMS fsignal = 1 kHz
(AHVSUP = 8 V) Volume 0 dB
Full Scale input from I2S
Signal Level at SCART Output 1.17 1.27 1.37 VRMS
(AHVSUP = 5V)
1)
n means 1, 2, 3, or 4; s means L or R
VoutMA Signal Level at Main/AUX-Output 1.23 1.37 1.51 VRMS fsignal = 1 kHz
(AHVSUP = 8 V) Volume 0 dB
Full scale input from I2S
Signal Level at Main/AUX-Output 0.76 0.90 1.04 VRMS
(AHVSUP = 5 V)
1)
s means L or R; p means M or A
from Analog Input to I2S Output MONO_IN, 0.008 0.03 % Input Level = 3 dBr with
SCn_IN_s1) resp. to VAICL, fsig = 1 kHz,
unweighted
20 Hz...16 kHz
from I2S Input to SCART Output SCn_OUT_s1) 0.008 0.03 % Input Level = 3 dBr,
fsig = 1 kHz,
unweighted
20 Hz...16 kHz
from I2S Input to Main or AUX Out- DACp_s1) 0.008 0.03 % Input Level = 3 dBr,
put fsig = 1 kHz,
unweighted
20 Hz...16 kHz
1)
n means 1, 2, 3, or 4; s means L or R; p means M or A
from Analog Input to I2S Output MONO_IN, 0.03 0.1 % Input Level = 3 dBr with
SCn_IN_s1) resp. to VAICL, fsig = 1 kHz,
unweighted
20 Hz...16 kHz
from I2S Input to SCART Output SCn_OUT_s1) 0.1 % Input Level = 3 dBr,
fsig = 1 kHz,
unweighted
20 Hz...16 kHz
from I2S Input to Main or AUX Out- DACp_s1) 0.1 % Input Level = 3 dBr,
put fsig = 1 kHz,
unweighted
20 Hz...16 kHz
1)
n means 1, 2, 3, or 4; s means L or R; p means M or A
CROSSTALK Specifications
dVNICAMOUT Tolerance of Output Voltage DACp_s 1.5 +1.5 dB 2.12 kHz, Modulator input
of NICAM Baseband Signal SCn_OUT_s1 level = 0 dBref
THDNICAM Total Harmonic Distortion + Noise 0.1 % 2.12 kHz, Modulator input
of NICAM Baseband Signal level = 0 dBref
fRMNR Frequency Response of BTSC Ste- 2.0 2.0 dB L or R 5%...66% EIM2), MNR
reo, 50 Hz...12 kHz
Frequency Response of BTSC- 2.0 2.0 dB SAP, white noise, 10% Modu-
SAP, 50 Hz...9 kHz lation, MNR
fPilot Pilot Frequency Range 15.563 15.843 kHz standard BTSC stereo sig-
nal, sound carrier only
1)
n means 1 or 2; s means L or R; p means M or A
2)
EIM refers to 75-s Equivalent Input Modulation. It is defined as the audio-signal level which results in a stated percentage modulation,
when the DBX encoding process is replaced by a 75-s preemphasis network.
fRMNR Frequency Response of BTSC Ste- 2.0 2.0 dB L or R 5%...66% EIM2), MNR
reo, 50 Hz...12 kHz
Frequency Response of BTSC- 2.0 2.0 dB SAP, white noise, 10% Modu-
SAP, 50 Hz...9 kHz lation, MNR
fPilot Pilot Frequency Range ANA_IN1+ 18.844 19.125 kHz standard FM radio
ANA_IN2+ stereo signal
1)
n means 1 or 2; s means L or R; p means M or A
Carrier frequency of 6.552 MHz 5.85 MHz 5.85 MHz 5.85 MHz
digital sound
Carrier frequency of 6.0 MHz 5.5 MHz 6.5 MHz AM mono 6.5 MHz
analog sound component FM mono FM mono FM mono
terrestrial cable
Characteristics Values
Number of channels 2
Companding characteristics near instantaneous, with compression to 10 bits/sample in 32-samples (1 ms) blocks
Audio overload level +12 dBm measured at the unity gain frequency of the preemphasis network (2 kHz)
5.2. A2-Systems
Table 53: Key parameters for A2 Systems of Standards B/G, D/K, and M
Preemphasis 50 s 75 s 50 s 75 s
Frequency deviation (nom/max) 27/50 kHz 17/25 kHz 27/50 kHz 15/25 kHz
Transmission Modes
Aural BTSC-MPX-Components
Carrier
(L+R) Pilot (LR) SAP Prof. Ch.
Max. deviation to Aural Carrier 73 kHz 25 kHz1) 5 kHz 50 kHz1) 15 kHz 3 kHz
(total)
Table 55: Key parameters for Japanese FM-Stereo Sound System EIA-J
Aural EIA-J-MPX-Components
Carrier
FM (L+R) (LR) Identification
Preemphasis 75 s 75 s none
Transmitter-sided delay 20 s 0 s 0 s
Aural FM-Radio-MPX-Components
Carrier
(L+R) Pilot (LR) RDS/ARI
Preemphasis:
USA 75 s 75 s
Europe 50 s 50 s
Table 61: Demodulator Write Registers; Subaddress: 10hex; these registers are not readable!
Note: All registers except AUTO_FM/AM, A2_Threshold, and CM_Threshold are initialized during STANDARD SELECTION and are
automatically updated when Automatic Sound Select (MODUS[0]=1) is on.
Table 62: Demodulator Read Registers; Subaddress: 11hex; these registers are not writable!
C_AD_BITS 00 23 3410, NICAM-Sync bit, NICAM-C-Bits, and three LSBs of additional data bits 97
3450
ADD_BITS 00 38 NICAM: bit [10:3] of additional data bits 97
Table 63: DSP-Write Registers; Subaddress: 12hex, all registers are readable as well
Write Register Address Bits Operational Modes and Adjustable Range Reset Page
(hex) Mode
Volume SCART1 channel: Ctrl. mode 00 07 [7:0] [Linear mode / logarithmic mode] 00hex 99
Volume SCART2 channel: Ctrl. mode 00 40 [7:0] [Linear mode / logarithmic mode] 00hex 99
Table 64: DSP Read Registers; Subaddress: 13hex, all registers are not writable
Stereo detection register for 00 18 [15:8] [80hex ... 7Fhex] 8 bit twos complement 100
A2 Stereo Systems
DC level readout FM1/Ch2-L 00 1B [15:0] [8000hex ... 7FFFhex] 16 bit twos complement 100
DC level readout FM2/Ch1-R 00 1C [15:0] [8000hex ... 7FFFhex] 16 bit twos complement 100
6.3.2. A2 Threshold
THRESHOLDS
THRESHOLDS
X : not affected while choosing the TV sound standard by means of the STANDARD SELECT Register
Note: This register is initialized during STANDARD SELECTION and is automatically updated when Automatic
Sound Select (MODUS[0]=1) is on.
Application Input Signal Contains AD_CV [6:1] AD_CV [6:1] Range of Input Signal
Ref. Value in integer at pin ANA_IN1+
and ANA_IN2+
Terrestrial TV
FM Standards 1 or 2 FM Carriers 101000 40 0.10 3 Vpp1)
NICAM/FM 1 FM and 1 NICAM Carrier 101000 40 0.10 3 Vpp1)
NICAM/AM 1 AM and 1 NICAM Carrier 100011 35 0.10 1.4 Vpp
(recommended: 0.10 0.8 Vpp)
NICAM only 1 NICAM Carrier only 010100 20 0.05 1.0 Vpp
Step AD_CV [6:1] Gain Input Level at pin ANA_IN1+ and ANA_IN2+
Constant Gain
Table 612: Control word MODE_REG; reset status: all bits are 0
Table 613: Loading sequence for FIR-coefficients To load the FIR-filters, the following data values are to
be transferred 8 bits at a time embedded
FIR1 00 01hex (MSP-Ch1: NICAM/FM2)
LSB-bound in a 16-bit word.
No. Symbol Name Bits Value The loading sequences must be obeyed. To change a
coefficient set, the complete block FIR1 or FIR2 must
1 NICAM/FM2_Coeff. (5) 8 be transmitted.
2 NICAM/FM2_Coeff. (4) 8
3 NICAM/FM2_Coeff. (3) 8
Note: For compatibility with MSP 3410B, IMREG1 and
see Table 614 IMREG2 have to be transmitted. The value for
4 NICAM/FM2_Coeff. (2) 8 IMREG1 and IMREG2 is 004. Due to the parti-
tioning to 8-bit units, the values 04hex, 40hex,
5 NICAM/FM2_Coeff. (1) 8 and 00hex arise.
6 NICAM/FM2_Coeff. (0) 8
6 FM/AM_Coef (3) 8
see Table 614
When selecting a TV-sound standard by means of the
7 FM/AM_Coef (2) 8 STANDARD SELECT register, all frequency tuning is
performed automatically.
8 FM/AM_Coef (1) 8
If manual setting of the tuning frequency is required, a
9 FM/AM_Coef (0) 8 set of 24-bit registers determining the mixing frequen-
cies of the quadrature mixers can be written manually
into the IC. In Table 615, some examples of DCO reg-
isters are listed. It is necessary to divide them up into
6.3.6. FIR-Parameter, Registers FIR1 and FIR2 low part and high part. The formula for the calculation
of the registers for any chosen IF frequency is as fol-
lows:
Note: The use of this register is no longer recom-
mended. It should be used only in cases where INCRdec = int(f/fs 224)
software compatibility to the MSP 34x0D is
required. Using the STANDARD SELECTION with: int = integer function
register together with the MODUS register pro- f = IF frequency in MHz
vides a more economic way to program the fS = sampling frequency (18.432 MHz)
MSP 34x0G.
Conversion of INCR into hex-format and separation of
the 12-bit low and high parts lead to the required regis-
Data-shaping and/or FM/AM bandwidth limitation is ter values (DCO1_HI or _LO for MSP-Ch1, DCO2_HI
performed by a pair of linear phase Finite Impulse or LO for MSP-Ch2).
Response filters (FIR-filter). The filter coefficients are
programmable and are either configured automatically
by the STANDARD SELECT register or written manu-
ally by the control processor via the control bus. Two
not necessarily different sets of coefficients are
required: one for MSP-Ch1 (NICAM or FM2) and one
for MSP-Ch2 (FM1 = FM-mono). In Table 614 several
coefficient sets are proposed.
Table 614: 8-bit FIR-coefficients (decimal integer) for MSP 34x0D; reset status: all coefficients are 0
B/G-, D/K- I- L- B/G-, D/K-, 130 180 200 280 380 500 Auto-
NICAM-FM NICAM-FM NICAM-AM M-Dual FM kHz kHz kHz kHz kHz kHz search
Coef(i) FIR1 FIR2 FIR1 FIR2 FIR1 FIR2 FIR2 FIR2 FIR2 FIR2 FIR2 FIR2 FIR2 FIR2
0 2 3 2 3 2 4 3 73 9 3 8 1 1 1
1 8 18 4 18 8 12 18 53 18 18 8 9 1 1
2 10 27 6 27 10 9 27 64 28 27 4 16 8 8
3 10 48 4 48 10 23 48 119 47 48 36 5 2 2
4 50 66 40 66 50 79 66 101 55 66 78 65 59 59
Mode- 0 0 0 0 1 1 1 1 1 1 0
REG[12]
Mode- 0 0 0 1 1 1 1 1 1 1 0
REG[13]
For compatibility, except for the FIR2-AM and the Autosearch-sets, the FIR-filter programming as used for the MSP 3410B is also possible.
Table 615: DCO registers for the MSP 34x0G; reset status: DCO_HI/LO = 0000
C4 C3 C2 C1 Operation Mode
Note: The use of these register is no longer recom-
mended. It should be used only in cases where 0 0 0 0 Stereo sound (NICAMA/B),
software compatibility to the MSP 34x0D is independent mono sound (FM1)
required. Using the STANDARD SELECTION
0 0 0 1 Two independent mono signals
register together with the STATUS register pro-
(NICAMA, FM1)
vides a more economic way to program the
MSP 34x0G and to retrieve information from the 0 0 1 0 Three independent mono channels
IC. (NICAMA, NICAMB, FM1)
6.4.4. NICAM Error Rate Register 6.4.7. Automatic Search Function for FM-Carrier
Detection in Satellite Mode
ERROR_RATE 00 57hex
The AM demodulation ability of the MSP 3410G and
Error free 0000hex MSP 3450G offers the possibility to calculate the field
strength of the momentarily selected FM carrier,
maximum error rate 07FFhex which can be read out by the controller. In SAT receiv-
ers, this feature can be used to make automatic FM
carrier search possible.
Average error rate of the NICAM reception in a time
interval of 182 ms, which should be close to 0. The ini- For this, the MSP has to be switched to AM-mode
tial and maximum value of ERROR_RATE is 2047. (MODE_REG[8]), FM-Prescale must be set to
This value is also active if the NICAM bit of 7Fhex = +127dec, and the FM DC notch (see section
MODE_REG is not set. Since the value is achieved by 6.5.7.) must be switched off. The sound-IF frequency
filtering, a certain transition time (approx. 0.5 sec) is range must now be scanned in the MSP-channel 2
unavoidable. Acceptable audio may have error rates by means of the programmable quadrature mixer with
up to a value of 700 int. Individual evaluation of this an appropriate incremental frequency (i.e. 10 kHz).
value by the controller and an appropriate threshold After each incrementation, a field strength value is
may define the fallback mode from NICAM to FM/ available at the quasi-peak detector output (quasi-
AM-Mono in case of poor NICAM reception. peak detector source must be set to FM), which must
be examined for relative maxima by the controller. This
The bit error rate per second (BER) can be calculated results in either continuing search or switching the
by means of the following formula: MSP back to FM demodulation mode.
BER = ERROR_RATE * 12.3*106 /s During the search process, the FIR2 must be loaded
with the coefficient set AUTOSEARCH, which
enables small bandwidth, resulting in appropriate field
6.4.5. PLL_CAPS Readback Register strength characteristics. The absolute field strength
value (can be read out of quasi-peak detector output
It is possible to read out the actual setting of the FM1) also gives information on whether a main FM
PLL_CAPS. In standard applications, this register is carrier or a subcarrier was detected; and as a practical
not of interest for the customer. consequence, the FM bandwidth (FIR1/2) and the
deemphasis (50 s or adaptive) can be switched
PLL_CAPS 02 1Fhex L accordingly.
minimum frequency 1111 1111 FFhex Due to the fact that a constant demodulation frequency
offset of a few kHz leads to a DC level in the demodu-
nominal frequency 0101 0110 56hex lated signal, further fine tuning of the found carrier can
RESET
be achieved by evaluating the DC Level Readout
maximum frequency 0000 0000 00hex FM1. Therefore, the FM DC Notch must be switched
on, and the demodulator part must be switched back to
PLL_CAPS 02 1Fhex H FM-demodulation mode.
PLL open xxxx xxx0
AGC_GAIN 02 1Ehex
SUM/DIFF 0100 0000 40hex Note: SCART Volume linear mode will not be sup-
AB_XCHANGE 0101 0000 50hex ported in the future (documented for compatibil-
ity reasons only).
PHASE_CHANGE_B 0110 0000 60hex
The sum/difference mode can be used together with J17 0000 0100 04hex
the quasi-peak detector to determine the sound mate-
OFF 0011 1111 3Fhex
rial 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 Note: This register is initialized during STANDARD
audio is stereo. SELECTION and is automatically updated when
Auto-matic Sound Select (MODUS[0]=1) is on.
2. Reset ident-filter
3. Set identification mode back to standard B/G or M Note: It is no longer necessary to read out and evalu-
4. Wait approx. 500 ms ate the A2 identification level. All evaluation is
performed in the MSP and indicated in the STA-
5. Read stereo detection register TUS register.
6.7. Demodulator Source Channels in Manual Mode 6.8. Exclusions of Audio Baseband Features
6.7.1. Terrestric Sound Standards In general, all functions can be switched independently.
Two exceptions exist:
Table 617 shows the source channel assignment of
1. NICAM cannot be processed simultaneously with
the demodulated signals in case of manual mode for
the FM2 channel.
all terrestric sound standards. See Table 22 for the
assignment in the Automatic Sound Select mode. In 2. FM adaptive deemphasis cannot be processed
manual mode for terrestric sound standards, only two simultaneously with FM-identification.
demodulator sources are defined.
Table 617: Manual Sound Select Mode for Terrestric Sound Standards
I2S_IN1/2 I2S_OUT
Loudspeaker
Headphone
SCART1-Ch.
Audio
Baseband
SCART1
Processing
SCART1
SCART2 SCART
DSP
SCART3 Input
Select SCART2-Ch.
SCART2
SCART4
MONO
SIF 2 IN
Tuner 2 if ANA_IN2+ not used
10 100
SIF 1 IN F - nF 8 V(5 V) 100 pF 56 pF
Tuner 1 + ANA_IN1/2+
18.432
3.3 100 MHz
F nF + + Alternative circuit for
1 k
56 pF 56 pF 56 pF + 10 F 10 F SIF-inputs for more
attenuation of video
components:
ANA_IN-
CAPL_M
XTAL_OUT
VREFTOP
CAPL_A
ANA_IN1+
ANA_IN2+
AGNDC
XTAL_IN
1 F
DACM_L
MONO_IN
330 nF
1 nF 1 F
DACM_R
LOUD
SC1_IN_L
SPEAKER
330 nF 1 nF 1 F
SC1_IN_R
330 nF DACM_SUB
AHVSS ASG
1 nF
SC2_IN_L
330 nF
330 nF
SC2_IN_R 1 F
AHVSS ASG DACA_L
330 nF
SC3_IN_L
1 F HEAD
1 nF
SC3_IN_R DACA_R
PHONE
330 nF
AHVSS ASG
1 nF
SC4_IN_L
330 nF
330 nF
SC4_IN_R
MSP 34x0G 100 22 F
5V SC1_OUT_L
+
5V
STANDBYQ 100 22 F
DVSS SC1_OUT_R
+
ADR_SEL
100 22 F
DVSS SC2_OUT_L
+
I2C_DA
I2C_CL 100 22 F
SC2_OUT_R
+
ADR_WS
ADR_CL
ADR_DA D_CTR_I/O_0
I2S_WS D_CTR_I/O_1
I2S_CL
I2S_DA_IN1 AUD_CL_OUT
I2S_DA_IN2
I2S_DA_OUT TESTEN
AHVSS
RESETQ
AHVSUP
DVSUP
AVSUP
AHVSS
VREF1
VREF2
DVSS
AVSS
AHVSS
AHVSS
AVSS
5V 5V 8V
(3.3 V) (5 V)
MSP 34x0G-B8
fine-tuning of A2-identification and carrier mute
EIA-J identification: faster transition time stereo/
bilingual to mono
J17 FM-deemphasis implemented
input specification for RESETQ and TESTEN
changed
MB implemented
MSP 34x0G-B8V3
The Automatic Sound Select (ASS) malfunction has
been corrected. In the previous version, under cer-
tain circumstances and depending on the baseband
features used, e.g.: Micronas BASS, Virtual Sur-
round Sound, Equalizer... etc., the Automatic Sound
Select Feature (ASS) did not work correctly.
Pseudo Stereo Effect (PSE) malfunction has been
corrected
MSP 34x0G-C12
FM-Radio available in non-BTSC versions with acti-
vation key MSP 3400G/3410G
implementation of SRS WOW
implementation of BBE
implementation of Micronas VOICE
correction of default Scart-Switch configuration dur-
ing power-up
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