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MICRONAS
MSP34x2G Multistandard Sound Processor Family with Dolby Surround Pro Logic
Edition May 22, 2000 6251-520-1PD
MICRONAS
MSP 34x2G
Contents Page 6 6 7 7 8 9 11 11 11 11 12 12 12 14 14 14 14 14 14 14 15 15 15 15 15 15 16 16 16 17 18 18 18 18 18 18 19 19 19 19 19 19 20 20 20 Section 1. 1.1. 1.2. 1.3. 1.4. 2. 2.1. 2.2. 2.2.1. 2.2.2. 2.2.3. 2.2.4. 2.2.5. 2.3. 2.4. 2.5. 2.5.1. 2.5.2. 2.5.3. 2.5.4. 2.6. 2.6.1. 2.6.1.1. 2.6.1.2. 2.6.2. 2.6.2.1. 2.6.2.2. 2.6.2.3. 2.6.2.4. 2.6.3. 2.6.4. 2.6.4.1. 2.6.4.2. 2.6.4.3. 2.6.4.4. 2.6.5. 2.6.6. 2.6.7. 2.7. 2.7.1. 2.7.2. 2.8. 2.9. 2.10. 2.11. Title Introduction Features Features of the MSP 34x2G Family MSP 34x2G Version List MSP 34x2G Versions and their Application Fields Functional Description Architecture of the MSP 34x2G Family Sound IF Processing Analog Sound IF Input Demodulator: Standards and Features Preprocessing of Demodulator Signals Automatic Sound Select Manual Mode Preprocessing for SCART and I2S Input Signals Source Selection and Output Channel Matrix Audio Baseband Processing Automatic Volume Correction (AVC) Loudspeaker and Headphone Outputs Subwoofer Output Quasi-Peak Detector Surround Processing Output Configuration HP/CS Switch Channel Configuration Surround Processing Mode Decoder Matrix Surround Reproduction Center Modes Useful Combinations of Surround Processing Modes Examples Application Tips for using 3D-PANORAMA Sweet Spot Clipping Loudspeaker Requirements Cabinet Requirements Input and Output Levels in Dolby Surround Pro Logic Mode Subwoofer in Surround Mode Equalizer in Surround Mode SCART Signal Routing SCART DSP In and SCART Out Select Stand-by Mode I2S Bus Interface ADR Bus Interface Digital Control I/O Pins and Status Change Indication Clock PLL Oscillator and Crystal Specifications
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Contents, continued Page 21 21 21 21 22 22 23 23 23 23 23 23 23 23 27 28 28 28 30 31 32 46 47 47 47 47 47 48 48 48 48 48 48 48 50 50 51 54 57 60 62 62 63 63 63 64 Section 3. 3.1. 3.1.1. 3.1.2. 3.1.3. 3.1.4. 3.1.5. 3.1.5.1. 3.1.5.2. 3.1.5.3. 3.1.5.4. 3.2. 3.3. 3.3.1. 3.3.2. 3.3.2.1. 3.3.2.2. 3.3.2.3. 3.3.2.4. 3.3.2.5. 3.3.2.6. 3.3.2.7. 3.4. 3.5. 3.5.1. 3.5.2. 3.5.3. 3.5.4. 3.5.5. 3.5.6. 3.5.7. 3.5.8. 3.5.9. 3.5.10. 4. 4.1. 4.2. 4.3. 4.4. 4.5. 4.6. 4.6.1. 4.6.2. 4.6.2.1. 4.6.2.2. 4.6.2.3. Title Control Interface I2C Bus Interface Device and Subaddresses Internal Hardware Error Handling Description of CONTROL Register Protocol Description Proposals for General MSP 34x2G I2C Telegrams Symbols Write Telegrams Read Telegrams Examples Start-Up Sequence: Power-Up and I2C Controlling MSP 34x2G Programming Interface User Registers Overview Description of User Registers STANDARD SELECT Register Refresh of STANDARD SELECT Register STANDARD RESULT Register Write Registers on I2C Subaddress 10hex Read Registers on I2C Subaddress 11hex Write Registers on I2C Subaddress 12hex Read Registers on I2C Subaddress 13hex Programming Tips Examples of Minimum Initialization Codes SCART1 Input to Loudspeaker in Stereo Sound B/G-FM (A2 or NICAM) BTSC-Stereo BTSC-SAP with SAP at Loudspeaker Channel FM-Stereo Radio Automatic Standard Detection Dolby Surround Pro Logic Example Virtual Dolby Surround Example Noise Sequencer for Dolby Pro Logic Software Flow for Interrupt driven STATUS Check Specifications Outline Dimensions Pin Connections and Short Descriptions Pin Descriptions Pin Configurations Pin Circuits Electrical Characteristics Absolute Maximum Ratings Recommended Operating Conditions (TA = 0 to 70 °C) General Recommended Operating Conditions Analog Input and Output Recommendations Recommendations for Analog Sound IF Input Signal
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Contents, continued Page 65 66 66 67 68 69 70 72 74 74 75 78 81 81 82 83 83 84 84 85 86 87 87 87 87 88 89 89 90 91 93 93 95 95 95 95 96 96 96 96 97 97 97 97 97 Section 4.6.2.4. 4.6.3. 4.6.3.1. 4.6.3.2. 4.6.3.3. 4.6.3.4. 4.6.3.5. 4.6.3.6. 4.6.3.7. 4.6.3.8. 4.6.3.9. 4.6.3.10. 5. 5.1. 5.2. 5.3. 5.4. 5.5. 5.6. 6. 6.1. 6.2. 6.3. 6.3.1. 6.3.1.1. 6.3.1.2. 6.3.2. 6.3.3. 6.3.4. 6.3.5. 6.3.6. 6.3.7. 6.4. 6.4.1. 6.4.2. 6.4.3. 6.4.4. 6.4.5. 6.4.6. 6.4.7. 6.5. 6.5.1. 6.5.2. 6.5.3. 6.5.4. Title Crystal Recommendations Characteristics General Characteristics Digital Inputs, Digital Outputs Reset Input and Power-Up I2C-Bus Characteristics I2S-Bus Characteristics Analog Baseband Inputs and Outputs, AGNDC Sound IF Inputs Power Supply Rejection Analog Performance Sound Standard Dependent Characteristics Appendix A: Overview of TV-Sound Standards NICAM 728 A2-Systems BTSC-Sound System Japanese FM Stereo System (EIA-J) FM Satellite Sound FM-Stereo Radio
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Appendix B: Manual/Compatibility Mode Demodulator Write and Read Registers for Manual/Compatibility Mode DSP Write and Read Registers for Manual/Compatibility Mode Manual/Compatibility Mode: Description of Demodulator Write Registers Automatic Switching between NICAM and Analog Sound Function in Automatic Sound Select Mode Function in Manual Mode A2 Threshold Carrier-Mute Threshold Register AD_CV Register MODE_REG FIR-Parameter, Registers FIR1 and FIR2 DCO-Registers Manual/Compatibility Mode: Description of Demodulator Read Registers NICAM Mode Control/Additional Data Bits Register Additional Data Bits Register CIB Bits Register NICAM Error Rate Register PLL_CAPS Readback Register AGC_GAIN Readback Register Automatic Search Function for FM-Carrier Detection in Satellite Mode Manual/Compatibility Mode: Description of DSP Write Registers Additional Channel Matrix Modes Volume Modes of SCART1/2 Outputs FM Fixed Deemphasis FM Adaptive Deemphasis
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Contents, continued Page 97 98 98 98 98 98 99 99 99 99 99 101 101 102 104 104 Section 6.5.5. 6.5.6. 6.5.7. 6.6. 6.6.1. 6.6.2. 6.7. 6.7.1. 6.7.2. 6.8. 6.9. 7. 7.1. 7.2. 8. 9. Title NICAM Deemphasis Identification Mode for A2 Stereo Systems FM DC Notch Manual/Compatibility Mode: Description of DSP Read Registers Stereo Detection Register for A2 Stereo Systems DC Level Register Demodulator Source Channels in Manual Mode Terrestric Sound Standards SAT Sound Standards Exclusions of Audio Baseband Features Compatibility Restrictions to MSP 34x0D Appendix D: Application Information Phase Relationship of Analog Outputs Application Circuit Appendix E: MSP 34x2G Version History Data Sheet History
License Notice:
1)
“Dolby”, “Virtual Dolby Surround”, and the double-D Symbol are trademarks of Dolby Laboratories.
Supply of this implementation of Dolby Technology does not convey a license nor imply a right under any patent, or any other industrial or intellectual 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.
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Multistandard Sound Processor Family with Dolby Surround Pro Logic The hardware and software description in this document is valid for the MSP 34x2G version A1 and following versions.
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Surround sound can be reproduced to a certain extent with only two loudspeakers. The MSP 3452G includes a Micronas virtualizer algorithm which has been approved by the Dolby1) Laboratories for compliance with the "Virtual Dolby Surround" technology. This algorithm is called “3D-PANORAMA” and enables convincing acoustical sensations. Virtual Dolby Surround can be processed together with headphone signals. The ICs are produced in submicron CMOS technology. The MSP 34x2G is available in the following packages: PSDIP64, PQFP80, and PLQFP64.
1. Introduction The MSP 34x2G family of single-chip Multistandard Sound Processors covers the sound processing of all analog TV-Standards worldwide, as well as the NICAM digital sound standards. The full TV sound processing, starting with analog sound IF signal-in, down to processed analog AF-out, is performed on a single chip. The family’s latest member, the MSP 3452G has all functions of the MSP 3450G with the addition of Dolby Surround Pro Logic and Virtual Dolby Surround sound processing (See License Notice on page 5). The MSP 3452G forms a superset of the functions of the MSP 3451G, which contains the virtualizer algorithms but does not contain any multi-channel processing. Additional output pins DACM_C and DACM_S have been defined which deliver the Dolby Surround Pro Logic processed Center and Surround channels. When DACM_C and DACM_S are active, the headphone outputs DACA_L and DACA_R are muted and vice versa. Simultaneous processing of Headphone signals and Dolby Surround Pro Logic is not possible.
1.1. Features – All MSP 3450G features – All MSP 3451G features as there are - the 3D-PANORAMA virtualizer algorithm - the PANORAMA virtualizer algorithm - Noise Generator – Dolby Surround Pro Logic processing – Various other multichannel sound modes – Additional pins for Center and Surround channels – Virtualizer able to work with 2 or 3 front loudspeakers – Pin and software compatible to MSP 34x0G
Sound IF1 ADC Sound IF2
Demodulator
Preprocessing
Loudspeaker Sound Processing Headphone/ Surround Sound Processing
DAC
Loudspeaker Subwoofer Center Surround
Source Select
DAC Headphone I2S
I2S1 I2S2 SCART1 SCART2 SCART3 SCART4 MONO SCART DSP Input Select
Prescale
DAC SCART1 ADC Prescale DAC SCART Output Select SCART2
Fig. 1–1: Block diagram of the MSP 34x2G
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MSP 34x2G
1.2. Features of the MSP 34x2G Family
Feature Dolby Surround Pro Logic 3D-PANORAMA virtualizer (approved by Dolby Laboratories) with noise generator PANORAMA virtualizer algorithm Standard Selection with single I2C transmission Automatic Standard Detection of terrestrial TV standards Automatic Sound Selection (mono/stereo/bilingual) Two selectable sound IF (SIF) inputs Automatic Carrier Mute function Interrupt output programmable (indicating status change) Loudspeaker / Headphone channel with volume, balance, bass, treble, loudness AVC: Automatic Volume Correction Subwoofer output with programmable low-pass and complementary high-pass filter 5-band graphic equalizer for loudspeaker channel Spatial effect for loudspeaker channel Four Stereo SCART (line) inputs, one Mono input; two Stereo SCART outputs Complete SCART in/out switching matrix Two I2S inputs; one I2S output All analog FM-Stereo A2 and satellite standards; AM-SECAM L standard Simultaneous demodulation of (very) high-deviation FM-Mono and NICAM Adaptive deemphasis for satellite (Wegener-Panda, acc. to ASTRA specification) ASTRA Digital Radio (ADR) together with DRP 3510A All NICAM standards Demodulation of the BTSC multiplex signal and the SAP channel Alignment free digital DBX noise reduction for BTSC Stereo and SAP Alignment free digital Micronas Noise Reduction (MNR) for BTSC Stereo and SAP BTSC stereo separation (MSP 3422/42G also EIA-J) significantly better than spec. SAP and stereo detection for BTSC system Korean FM-Stereo A2 standard Alignment-free Japanese standard EIA-J Demodulation of the FM-Radio multiplex signal X X X X X X X X X X X X X X X X X X X X 3402 X X X X X X X X X X X X X X X X X X 3412 X X X X X X X X X X X X X X X X X X X X X X X X X 3422 X X X X X X X X X X X X X X X X X 3442 X X X X X X X X X X X X X X X X X 3452 X X X X X X X X X X X X X X X X X X X X X X X X
1.3. MSP 34x2G Version List
Version MSP 3402G MSP 3412G MSP 3422G MSP 3442G MSP 3452G Status not confirmed planned not confirmed not confirmed available Description FM Stereo (A2) Version NICAM and FM Stereo (A2) Version NTSC Version (A2 Korea, BTSC with Micronas Noise Reduction (MNR), and Japanese EIA-J system) NTSC Version (A2 Korea, BTSC with DBX noise reduction, and Japanese EIA-J system) Global Version (all sound standards)
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1.4. MSP 34x2G Versions and their Application Fields Table 1–1 provides an overview of TV sound standards that can be processed by the MSP 34x2G family. In addition, the MSP 34x2G is able to handle the terrestrial FM-Radio standard. With the MSP 34x2G, a com-
PRELIMINARY DATA SHEET
plete multimedia receiver covering all TV sound standards together with terrestrial and satellite radio sound can be built; even ASTRA Digital Radio can be processed (with a DRP 3510A coprocessor).
Table 1–1: TV Stereo Sound Standards covered by the MSP 34x2G IC Family (details see Appendix A)
MSP Version 3402 TVSystem B/G 5.5/5.85 L I 6.5/5.85 6.0/6.552 6.5/6.2578125 3402 3412 6.5/6.7421875 D/K 6.5/5.7421875 3452 6.5/5.85 6.5 7.02/7.2 7.38/7.56 etc. 4.5/4.724212 3422, 3442 M/N 4.5 4.5 FM-Radio 10.7 FM-Stereo (A2, D/K3) FM-Mono/NICAM (D/K, NICAM) FM-Mono FM-Stereo ASTRA Digital Radio (ADR) with DRP 3510A FM-Stereo (A2) FM-FM (EIA-J) BTSC-Stereo + SAP FM-Stereo Radio NTSC NTSC NTSC, PAL Korea Japan USA, Argentina USA, Europe SECAM-East PAL PAL Poland China, Hungary Europe Sat. ASTRA FM-Mono/NICAM AM-Mono/NICAM FM-Mono/NICAM FM-Stereo (A2, D/K1) FM-Stereo (A2, D/K2) PAL SECAM-L PAL SECAM-East PAL Scandinavia, Spain France UK, Hong Kong Slovak. Rep. currently no broadcast Position of Sound Carrier /MHz 5.5/5.7421875 Sound Modulation FM-Stereo (A2) Color System PAL Broadcast e.g. in: Germany
3402
Satellite
33 34 39 MHz
4.5 9 MHz
Loudspeaker SAW Filter Tuner Sound IF Mixer Subwoofer Center
1
Mono Vision Demodulator SCART Inputs Composite Video SCART1 SCART2 SCART3 SCART4
MSP 34x2G
Surround
2 2 2 2 ADR I2S2 2 2
Headphone
SCART1 SCART2
SCART Outputs
ADR Decoder DRP 3510A
Fig. 1–2: Typical MSP 34x2G application
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Interface Prescale
(16hex)
Source Select
SCART DSP Input Select
(13hex)
SC1_IN_L SC1_IN_R SC2_IN_L SC2_IN_R SC3_IN_L SC3_IN_R SC4_IN_L SC4_IN_R MONO_IN
(41hex)
(40hex)
SCART Output Select
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2. Functional Description
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Standard Selection ANA_IN1+ AGC A D DEMODULATOR (incl. Carrier Mute) Deemphasis: 50/75 µs DBX Panda1 FM/AM
Automatic Sound Select
FM/AM
Configurable Output Section
Loudspeaker Channel Matrix
(08hex)
AVC
(29hex)
ANA_IN2+
Prescale
(0Ehex)
Stereo or A/B
Bass/ Treble or Equalize
(02hex) (03hex)
Σ
Loudness
(04hex)
Complementary Highpass
0.5 (20hex)
Spatial Effects
(05hex)
D Balance Volume
DACM_L
(01hex)
DACM_R A DACM_SUB
(00hex)
ADR-Bus Interface
Decoded Standards: − NICAM − A2 − AM − BTSC − EIA-J − SAT − FM-Radio
NICAM Deemphasis J17 Prescale
(10hex)
Stereo or A
Lowpass Beeper
(14hex) (2Dhex)
Level Adjust
(2Chex)
Stereo or B
DACM_C
Standard and Sound Detection
I2 C Read Register
DACM_S
I2S1 I2S_DA_IN1 I2 S
Headphone Channel Matrix
(09hex)
Volume Bass/ Treble
(32hex)
Σ
D A
Loudness
Balance
DACA_L
(33hex)
(30hex)
(06hex)
DACA_R
I2S1 I2S_DA_IN2 I2 S Interface Prescale
(12hex)
I2S Channel Matrix
(0Bhex)
I2S Interface
I2S_DA_OUT
Quasi-Peak Channel Matrix
(0Chex)
Quasi-Peak Detector
I2C Read Register
(19hex) (1Ahex)
A D
SCART SCART1 Channel Matrix
(0Ahex)
Volume
D SCART1_L/R A
Prescale
(0Dhex)
(07hex)
SCART2 Channel Matrix
Volume
D SCART2_L/R A
SC1_OUT_L
SC1_OUT_R
SC2_OUT_L
MSP 34x2G
SC2_OUT_R
(13 hex)
Fig. 2–1: Signal flow block diagram of the MSP 3452G without any surround processing: Output Configuration (register 48hex) = 0000hex
MSP 34x2G
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Configurable Output Section
Loudspeaker Channel Matrix
(08hex)
Virtualizer
AVC
(29hex)
Bass/ Treble or Equalize
(02hex) (03hex)
Σ
Loudness
(04hex) 0.5
Complementary Highpass
(20hex)
DACM_L D Balance Volume DACM_R
(01hex)
Noise Generator
Lowpass Beeper
(14hex) (2Dhex)
Level Adjust
(2Chex) (00hex)
A
DACM_SUB
DACM_C
DACM_S
Headphone Channel Matrix
(09hex)
Volume Bass/ Treble
(32hex)
Σ
D A
DACA_L
Loudness
Balance DACA_R
(33hex)
(30hex)
(06hex)
Fig. 2–2: Output section in virtual mode: Output Configuration (register 48hex) = 0100hex
Configurable Output Section
Loudspeaker Channel Matrix
(08hex)
Bass/ Treble
(02hex)
Σ
Loudness
(04hex)
Complementary Highpass
(20hex)
DACM_L D Balance Volume DACM_R
(01hex)
Noise Generator
Dolby Pro Logic and optional Virtualizer
Lowpass AVC Beeper
(14hex) 0.5 (2Dhex)
Level Adjust
(2Chex) (00hex)
A
DACM_SUB
Bass/ Treble
(29hex) (32hex)
Σ
Loudness
Complementary Highpass
Volume Balance
D A
DACM_C
DACM_S DACA_L
(33hex)
(30hex)
(06hex)
DACA_R
Fig. 2–3: Output section with multi-channel surround: Output Configuration (register 48hex) = 8200hex
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MSP 34x2G
BTSC-Stereo: Detection and FM demodulation of the aural carrier resulting in the MTS/MPX signal. Detection and evaluation of the pilot carrier, AM demodulation of the (L−R)-carrier and detection of the SAP subcarrier. Processing of DBX noise reduction or Micronas Noise Reduction (MNR). BTSC-Mono + SAP: Detection and FM demodulation of the aural carrier resulting in the MTS/MPX signal. Detection and evaluation of the pilot carrier, detection and FM demodulation of the SAP subcarrier. Processing of DBX noise reduction or Micronas Noise Reduction (MNR). Japan Stereo: Detection and FM demodulation of the aural carrier resulting in the MPX signal. Demodulation and evaluation of the identification signal and FM demodulation of the (L−R)-carrier. FM-Satellite Sound: Demodulation of one or two FM carriers. Processing of high-deviation mono or narrow bandwidth mono, stereo, or bilingual satellite sound according to the ASTRA specification. FM-Stereo-Radio: Detection and FM demodulation of the aural carrier resulting in the MPX signal. Detection and evaluation of the pilot carrier and AM demodulation of the (L−R)-carrier. The demodulator blocks of all MSP 34x2G versions have identical user interfaces. Even completely different systems like the BTSC and NICAM systems are controlled the same way. Standards are selected by means of MSP Standard Codes. Automatic processes handle standard detection and identification without controller interaction. The key features of the MSP 34x2G demodulator blocks are Standard Selection: The controlling of the demodulator is minimized: All parameters, such as tuning frequencies or filter bandwidth, are adjusted automatically by transmitting one single value to the STANDARD SELECT register. For all standards, specific MSP standard codes are defined. Automatic Standard Detection: If the TV sound standard is unknown, the MSP 34x2G can automatically detect the actual standard, switch to that standard, and respond the actual MSP standard code. Automatic Carrier Mute: To prevent noise effects or FM identification problems in the absence of an FM carrier, the MSP 34x2G offers a configurable carrier mute feature, which is activated automatically if the TV sound standard is selected by means of the STANDARD SELECT register. If no FM carrier is detected at one of the two MSP demodulator channels, the corresponding demodulator output is muted. This is indicated in the STATUS register.
2.1. Architecture of the MSP 34x2G Family The block diagrams in Fig. 2–1, Fig. 2–2, and Fig. 2–3 show the signal flow in the MSP 34x2G in three modes that can be set in the Output Configuration register. – Standard mode (see Fig. 2–1). The IC is compatible to the MSP 34x0 family. – Virtual mode (see Fig. 2–2). The IC is compatible to the Virtual Dolby MSP 34x1 family. – Dolby Surround Pro Logic mode (see Fig. 2–3). The three block diagrams show the features of the MSP 3452G family member. Other members of the MSP 34x2G family do not have the complete set of features: The demodulator handles only a subset of the standards presented in the demodulator block; NICAM processing is only possible in the MSP 3412G and MSP 3452G.
2.2. Sound IF Processing 2.2.1. Analog Sound IF Input The input pins ANA_IN1+, ANA_IN2+, and ANA_IN− offer the possibility to connect two different sound IF (SIF) sources to the MSP 34x2G. The analog-to-digital conversion of the preselected sound IF signal is done by an A/D-converter. An analog automatic gain circuit (AGC) allows a wide range of input levels. The highpass filters formed by the coupling capacitors at pins ANA_IN1+ and ANA_IN2+ see Section 7.2. “Application Circuit” on page 102 are sufficient in most cases to suppress video components. Some combinations of SAW filters and sound IF mixer ICs, however, show large picture components on their outputs. In this case, further filtering is recommended.
2.2.2. Demodulator: Standards and Features The MSP 34x2G is able to demodulate all TV-sound standards worldwide including the digital NICAM system. Depending on the MSP 34x2G version, the following demodulation modes can be performed: A2 Systems: Detection and demodulation of two separate FM carriers (FM1 and FM2), demodulation and evaluation of the identification signal of carrier FM2. NICAM Systems: Demodulation and decoding of the NICAM carrier, detection and demodulation of the analog (FM or AM) carrier. For D/K-NICAM, the FM carrier may have a maximum deviation of 384 kHz. Very high deviation FM-Mono: Detection and robust demodulation of one FM carrier with a maximum deviation of 540 kHz.
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2.2.3. Preprocessing of Demodulator Signals The NICAM signals must be processed by a deemphasis filter and adjusted in level. The analog demodulated signals must be processed by a deemphasis filter, adjusted in level, and dematrixed. The correct deemphasis filters are already selected by setting the standard in the STANDARD SELECT register. The level adjustment has to be done by means of the FM/ AM and NICAM prescale registers. The necessary dematrix function depends on the selected sound standard and the actual broadcasted sound mode (mono, stereo, or bilingual). It can be manually set by the FM Matrix Mode register or automatically set by the Automatic Sound Selection.
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– “Stereo or A” channel: Analog or digital mono sound, stereo if available. In case of bilingual broadcast, it contains language A (on left and right). – “Stereo or B” channel: Analog or digital mono sound, stereo if available. In case of bilingual broadcast, it contains language B (on left and right). Fig. 2–4 and Table 2–2 show the source channel assignment of the demodulated signals in case of Automatic Sound Select mode for all sound standards. Note: The analog primary input channel contains the signal of the mono FM/AM carrier or the L+R signal of the MPX carrier. The secondary input channel contains the signal of the second FM carrier, the L−R signal of the MPX carrier, or the SAP signal.
2.2.4. Automatic Sound Select In the Automatic Sound Select mode, the dematrix function is automatically selected based on the identification information in the STATUS register. No I2C interaction is necessary when the broadcasted sound mode changes (e.g. from mono to stereo). The demodulator supports the identification check by switching between mono compatible standards (standards that have the same FM mono carrier) automatically and non-audible. If B/G-FM or B/G-NICAM is selected, the MSP will switch between these standards. The same action is performed for the standards: D/K1-FM, D/K2-FM, and D/K-NICAM. Switching is only done in the absence of any stereo or bilingual identification. If identification is found, the MSP keeps the detected standard. In case of high bit-error rates, the MSP 34x2G automatically falls back from digital NICAM sound to analog FM or AM mono. Table 2–1 summarizes all actions that take place when Automatic Sound Select is switched on. To provide more flexibility, the Automatic Sound Select block prepares four different source channels of demodulated sound (see Fig. 2–4). By choosing one of the four demodulator channels, the preferred sound mode can be selected for each of the output channels (loudspeaker, headphone, etc.). This is done by means of the Source Select registers. The following source channels of demodulated sound are defined: – “FM/AM” channel: Analog mono sound, stereo if available. In case of NICAM, analog mono only (FM or AM mono). – “Stereo or A/B” channel: Analog or digital mono sound, stereo if available. In case of bilingual broadcast, it contains both languages A (left) and B (right).
primary channel secondary channel FM/AM FM-Matrix Prescale FM/AM 0 Source Select Output-Ch. Matrices must be set according the standard SC2 Ch. Matrix LS Ch. Matrix primary channel secondary channel NICAM A FM/AM FM/AM 0 Source Select LS Ch. Matrix Output-Ch. Matrices must be set once to stereo
Prescale
NICAM
Automatic Sound Select
Stereo or A/B
1
Stereo or A
3
NICAM
Prescale
Stereo or B
4
SC2 Ch. Matrix
Fig. 2–4: Source channel assignment of demodulated signals in Automatic Sound Select Mode
2.2.5. Manual Mode Fig. 2–5 shows the source channel assignment of demodulated signals in case of manual mode. If manual mode is required, more information can be found in Section 6.7. “Demodulator Source Channels in Manual Mode” on page 99.
NICAM A
NICAM NICAM (Stereo or A/B) 1
NICAM
Prescale
Fig. 2–5: Source channel assignment of demodulated signals in Manual Mode
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MSP 34x2G
Table 2–1: Performed actions of the Automatic Sound Selection
Selected TV Sound Standard B/G-FM, D/K-FM, M-Korea, and M-Japan B/G-NICAM, L-NICAM, I-NICAM, and D/K-NICAM Performed Actions Evaluation of the identification signal and automatic switching to mono, stereo, or bilingual. Preparing four demodulator source channels according to Table 2–2. Identification is acquired after 500 ms. Evaluation of NICAM-C-bits and automatic switching to mono, stereo, or bilingual. Preparing four demodulator source channels according to Table 2–2. NICAM detection is acquired within 150 ms. 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 or D/K1-FM, D/K2-FM, D/K3-FM, and D/K-NICAM Automatic searching for stereo/bilingual-identification in case of mono transmission. Automatic and nonaudible changes between Dual-FM and FM-NICAM standards while listening to the basic FM-Mono sound carrier. Example: If starting with B/G-FM-Stereo, there will be a periodical alternation to B/G-NICAM in the absence of FM-Stereo/Bilingual or NICAM-identification. Once an identification is detected, the MSP keeps the corresponding standard. Evaluation of the pilot signal and automatic switching to mono or stereo. Preparing four demodulator source channels according to Table 2–2. Detection of the SAP carrier. Pilot detection is acquired after 200 ms. 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 2–2).
BTSC-STEREO, FM Radio
BTSC-SAP
Table 2–2: Sound modes for the demodulator source channels with Automatic Sound Select
Source Channels in Automatic Sound Select Mode Broadcasted Sound Standard M-Korea B/G-FM D/K-FM M-Japan Selected MSP Standard Code3) 02 03, 081) 04, 05, 07, 0B1) 30 Broadcasted Sound Mode MONO STEREO BILINGUAL: Languages A and B B/G-NICAM L-NICAM I-NICAM D/K-NICAM D/K-NICAM
(with high deviation FM)
FM/AM
(source select: 0)
Stereo or A/B
(source select: 1)
Stereo or A
(source select: 3)
Stereo or B
(source select: 4)
Mono Stereo Left = A Right = B analog Mono analog Mono analog Mono analog Mono Mono Stereo Mono Stereo Left = Mono Right = SAP Left = Mono Right = SAP Mono Stereo
Mono Stereo Left = A Right = B analog Mono NICAM Mono NICAM Stereo Left = NICAM A Right = NICAM B Mono Stereo Mono Stereo Left = Mono Right = SAP Left = Mono Right = SAP Mono Stereo
Mono Stereo A analog Mono NICAM Mono NICAM Stereo NICAM A Mono Stereo Mono Stereo Mono Mono Mono Stereo
Mono Stereo B analog Mono NICAM Mono NICAM Stereo NICAM B Mono Stereo Mono Stereo SAP SAP Mono Stereo
08, 032) 09 0A 0B, 042), 052) 0C
NICAM not available or error rate too high MONO STEREO BILINGUAL: Languages A and B
BTSC
20, 21
MONO STEREO
20
MONO+SAP STEREO+SAP
21
MONO+SAP STEREO+SAP
FM Radio
40
MONO STEREO
1) 2) 3)
The Automatic Sound Select process will automatically switch to the mono compatible analog standard. The Automatic Sound Select process will automatically switch to the mono compatible digital standard. The MSP Standard Codes are defined in Table 3–7 on page 27.
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MSP 34x2G
2.3. Preprocessing for SCART and I2S Input Signals The SCART and I2S inputs need only be adjusted in level by means of the SCART and I2S prescale registers.
PRELIMINARY DATA SHEET
output level [dBr]
−18 −24
2.4. Source Selection and Output Channel Matrix The Source Selector makes it possible to distribute all source signals (one of the demodulator source channels, SCART, or I2S input) to the desired output channels (loudspeaker, headphone, etc.). All input and output signals can be processed simultaneously. Each source channel is identified by a unique source address. For each output channel, the sound mode can be set to sound A, sound B, stereo, or mono by means of the output channel matrix. If Automatic Sound Select is on, the output channel matrix can stay fixed to stereo (transparent) for demodulated signals.
−30 −24 −18 −12 −6 0
input level [dBr]
Fig. 2–6: Simplified AVC characteristics
2.5.2. Loudspeaker and Headphone Outputs The following baseband features are implemented in the loudspeaker and headphone output channels: bass/treble, loudness, balance, and volume. A square wave beeper can be added to the loudspeaker and headphone channel. The loudspeaker channel additionally performs: equalizer (not simultaneously with bass/treble), spatial effects, and a subwoofer crossover filter.
2.5. Audio Baseband Processing 2.5.1. Automatic Volume Correction (AVC) Different sound sources (e.g. terrestrial channels, SAT channels, or SCART) fairly often do not have the same volume level. Advertisements during movies usually have a higher volume level than the movie itself. This results in annoying volume changes. The AVC solves this problem by equalizing the volume level. To prevent clipping, the AVC’s gain decreases quickly in dynamic boost conditions. To suppress oscillation effects, the gain increases rather slowly for low-level inputs. The decay time is programmable by means of the AVC register (see page page 36). For input signals ranging from −24 dBr to 0 dBr, the AVC maintains a fixed output level of −18 dBr. Fig. 2–6 shows the AVC output level versus its input level. For prescale and volume registers set to 0 dB, a level of 0 dBr corresponds to full scale input/output. This is – SCART input/output 0 dBr = 2.0 Vrms – Loudspeaker and Aux output 0 dBr = 1.4 Vrms
2.5.3. Subwoofer Output The subwoofer signal is created by combining the left and right channels directly behind the loudness block using the formula (L+R)/2. Due to the division by 2, the D/A converter will not be overloaded, even with full scale input signals. The subwoofer signal is filtered by a third-order low-pass with programmable corner frequency followed by a level adjustment. At the loudspeaker channels, a complementary high-pass filter can be switched on. Subwoofer and loudspeaker output use the same volume (Loudspeaker Volume Register).
2.5.4. Quasi-Peak Detector The quasi-peak readout register can be used to read out the quasi-peak level of any input source. The feature is based on following filter time constants: attack time: 1.3 ms decay time: 37 ms
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PRELIMINARY DATA SHEET
MSP 34x2G
2.6.2. Surround Processing Mode Surround sound processing is controlled by three functions: The "Decoder Matrix" defines which method should be used to create a multichannel signal (L, C, R, S) out of a stereo input. The "Surround Reproduction" determines whether the surround signal “S” is fed to surround speakers. If no surround speaker is actually connected, it defines the method that should be used to create surround effects. The “Center Mode” determines how the center signal “C” is to be processed. It can be left unmodified, distributed to left and right, discarded or high pass filtered, whereby the low pass signals are distributed to left and right. The surround processing mode is controlled by means of register 4Bhex on I2C subaddress 12hex. 2.6.2.1. Decoder Matrix The Decoder Matrix allows three settings: – ADAPTIVE: The adaptive matrix is used for Dolby Surround Pro Logic. Even sound material not encoded in Dolby Surround will produce good surround effects in this mode. The use of the adaptive matrix requires a license from Dolby Laboratories (See License Notice on page 5). – PASSIVE: A simple fixed matrix is used for surround sound. – EFFECT: A fixed matrix that is used for mono sound and special effects. In adaptive or passive mode no surround signal is present in case of mono, moreover in adaptive mode even the left and right output channels carry no signal (or just low frequency signals in case of Center Mode = NORMAL). If surround sound is still required for mono signals, the effect mode can be used. This forces the surround channel to be active. The effect mode can be used together with 3D-PANORAMA. The result will be a pseudo stereo effect or a broadened stereo image respectively.
2.6. Surround Processing 2.6.1. Output Configuration Like the MSP 34x1G ICs, the MSP 34x2G can be used for virtual surround sound on the left and right loudspeaker outputs. For multichannel outputs (more than 2 channels), extra output pins have been defined (DACM_C and DACM_S pins). For processing of these output channels, internal resources are shared with the headphone processing. As a result, headphone output is not possible together with multi-channel surround processing. When headphone output pins are active, the surround outputs are muted and vice versa. There are two options: the HP/CS switch and the channel configuration. The output configuration is controlled by means of register 48hex on I2C subaddress 12hex. 2.6.1.1. HP/CS Switch This switch defines which output pin pair is driven by the D/A converters that are used for headphone or surround processing. The unselected pins are muted. This makes it convenient to connect the center/surround amplifiers or outputs to the MSP 34x2 without external switches. Mute the Headphone/Surround channel by setting register 06hex to 0000hex before switching. Allow at least 2 s for settling to avoid audible plops.
2.6.1.2. Channel Configuration The channel configuration defines whether surround processing is switched on and what resources of the IC are to be used for surround sound processing. There are 3 options: – STEREO: The IC is in the normal stereo processing mode. No surround processing takes place. In this mode, the IC is compatible to the MSP 34x0G. – TWO_CHANNEL: Surround sound processing is switched on, but only left and right loudspeaker channels are used for output. This mode is used for virtual surround sound. – MULTI_CHANNEL: Surround sound processing is switched on, left and right loudspeaker channels together with left and right headphone channels are used for output. The following relationship applies: Center corresponds to the left headphone channel; Surround corresponds to the right headphone channel.
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MSP 34x2G
2.6.2.2. Surround Reproduction Surround sound can be reproduced with four choices: – REAR_SPEAKER: If there are any surround speakers connected to the system, this mode should be used. Useful loudspeaker combinations are: (L, C, R, S) or (L, R, S). – FRONT_SPEAKER: If there is no surround speaker connected, this mode can be used. Surround information is mixed to left and right output but without creating the illusion of a virtual speaker. It is similar to stereo but an additional center speaker can be used. This mode should be used with the adaptive decoder matrix only. Useful loudspeaker combinations are: (L, C, R) (Note: the surround output channel is muted). – PANORAMA: The surround information is mixed to left and right in order to create the illusion of a virtual surround speaker. Useful loudspeaker combinations are: (L, C, R) or (L, R) (Note: the surround output channel is muted). – 3D-PANORAMA: Like PANORAMA with improved effect. This algorithm has been approved by the Dolby Laboratories for compliance with the "Virtual Dolby Surround" technology. Useful loudspeaker combinations are: (L, C, R) or (L, R) (Note: the surround output channel is muted).
PRELIMINARY DATA SHEET
2.6.2.4. Useful Combinations of Surround Processing Modes In principle, "Decoder Matrix", "Surround Reproduction", and "Center Modes" are independent settings (all "Decoder Matrix" settings can be used with all "Surround Reproduction" and "Center Modes") but there are some combinations that do not create "good" sound. Useful combinations are Surround Reproduction and Center Modes – REAR_SPEAKER: This mode is used if surround speakers are available. Useful center modes are NORMAL, WIDE, PHANTOM, and OFF. – FRONT_SPEAKER: This mode can be used if no surround speaker but a center speaker is connected. Useful center modes are NORMAL and WIDE. – PANORAMA or 3D-PANORAMA: No surround speaker used. Two (L and R) or three (L, R, and C) loudspeakers can be used. Useful center modes are NORMAL, WIDE, PHANTOM, and OFF. Center Modes and Decoder Matrix – PHANTOM: Should only be used together with ADAPTIVE Decoder Matrix. – NORMAL and WIDE: Can be used together with any Surround Decoder Matrix. – OFF: In special cases, this mode can be used together with the PASSIVE and EFFECT Decoder Matrix (no center speaker connected).
2.6.2.3. Center Modes Four center modes are supported: – NORMAL: small center speaker connected, L and R speakers have better bass capability. – WIDE: L,R, and C speakers all have good bass capability. – PHANTOM: No center speaker used. Center signal is distributed to L and R (Note: the center output channel C is muted). – OFF: No center speaker used. Center signal C is discarded (Note: the center output channel C is muted).
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PRELIMINARY DATA SHEET
MSP 34x2G
2.6.3. Examples Table 2–3 shows some examples of how these modes can be used to configure the IC. The list is not intended to be complete, more modes are possible.
Table 2–3: Examples of Surround Configurations
Configurations
Speaker Configuration1)
Output Configuration
Register (48hex) HP/CS Switch [15] Channel Configuration [14:8]
Surround Processing Mode
Register (4Bhex) Decoder Matrix [15:8] Surround Reproduction [7:4] Center Mode [3:0]
Stereo IC is compatible to the MSP34x0G. Stereo
(L,R) HP STEREO − − −
Surround Modes as defined by Dolby Laboratories 2) Dolby Surround Pro Logic
(L,C,R,S) CS MULTI_CHANNEL ADAPTIVE REAR_ SPEAKER REAR_ SPEAKER FRONT_ SPEAKER 3D_PANORAMA NORMAL WIDE PHANTOM
(L,R,S)
CS
MULTI_CHANNEL
ADAPTIVE
Dolby 3 Stereo Virtual Dolby Surround
(L,C,R)
CS
MULTI_CHANNEL
ADAPTIVE
NORMAL WIDE PHANTOM
(L,R)
HP
TWO_CHANNEL
ADAPTIVE
Surround Modes that use the Dolby Pro Logic Matrix2) 3-Channel Virtual Surround Passive Matrix Surround Sound 4-Channel Surround 3-Channel Surround 2-Channel Micronas Perfect 3D Sound 3-Channel Micronas Perfect 3D Sound Special Effects Surround Sound 4-Channel Surround for mono 2-Channel Virtual Surround for mono 3-Channel Virtual Surround for mono
1) 2)
(L,C,R)
CS
MULTI_CHANNEL
ADAPTIVE
3D_PANORAMA
NORMAL WIDE
(L,C,R,S)
CS
MULTI_CHANNEL
PASSIVE
REAR_ SPEAKER REAR_ SPEAKER 3D_PANORAMA 3D_PANORAMA
NORMAL WIDE OFF
(L,R,S)
CS
MULTI_CHANNEL
PASSIVE
(L,R) (L,C,R)
HP CS
TWO_CHANNEL MULTI_CHANNEL
PASSIVE PASSIVE
OFF NORMAL WIDE
(L,C,R,S)
CS
MULTI_CHANNEL
EFFECT
REAR_ SPEAKER 3D_PANORAMA 3D_PANORAMA
NORMAL WIDE OFF NORMAL WIDE
(L,R) (L,C,R)
HP CS
TWO_CHANNEL MULTI_CHANNEL
EFFECT EFFECT
Speakers not in use are muted automatically. The implementation in products requires a license from Dolby Laboratories Licensing Corporation (see note on page 5).
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2.6.4. Application Tips for using 3D-PANORAMA 2.6.4.1. Sweet Spot Good results are only obtained in a rather close area along the middle axis between the two loudspeakers: the sweet spot. Moving away from this position degrades the effect.
PRELIMINARY DATA SHEET
Great care has to be taken with systems that use one common subwoofer: A single loudspeaker cannot reproduce virtual sound locations. The crossover frequency must be lower than 120 Hz.
2.6.4.4. Cabinet Requirements During listening tests at Dolby Laboratories, no resonances in the cabinet should occur.
2.6.4.2. Clipping For the test at Dolby Labs, it is very important to have no clipping effects even with worst case signals. That is, 2 Vrms input signal must not clip. The SCART input prescale register has to be set to values of max 19hex (25dec). This is sufficient in terms of clipping. However, it was found, that by reducing the prescale to a value lower than 25dec more convincing effects are generated in case of very high dynamic signals. A value of 18dec is a good compromise between overall volume and additional headroom. Test signals: sine sweep with 2 VRMS; L only, R only, L&R equal phase, L&R anti phase. Listening tests: Dolby Trailers (train trailer, city trailer, canyon trailer...) Good material to check for resonances are the Dolby Trailers or other dynamic sound tracks.
2.6.5. Input and Output Levels in Dolby Surround Pro Logic Mode The analog inputs are able to accept 2 Vrms input level without overloading any stage before the volume control. The nominal input level (input sensitivity) is 350 mV. This gives 15 dB headroom. The scart prescale value should be set to max 0 dB (max 25dec). I2S-Inputs should have the same headroom (15 dB) when entering the MSP 3452G. The highest possible input level of 0 dBFS is accepted without internal overflow. The I2S-prescale value should be set to 0 dB (16dec). With higher prescale values lower input sensitivities can be accommodated. A higher input sensitivity is not possible, because at least 15 dB headroom is required for every input according to the Dolby specifications. A full-scale left only input (2 Vrms) will produce a fullscale left only output (at 0 dB volume). The typical output level is 1.37 Vrms for DACM_L. The same holds true for right only signals (1.37 Vrms for DACM_R). A full-scale input level on both inputs (Lin=Rin=2 Vrms) will give a center only output with maximum level. The typical output level is 1.37 Vrms for DACM_C. A fullscale input level on both inputs (but Lin and Rin with inverted phases) will give a surround-only signal with maximum level (1.37 Vrms for DACM_S). For reproducing Dolby Pro Logic according to its specifications, the center and surround outputs must be amplified by 3 dB with respect to the L and R output signals. This can be done in two ways: 1. By implementing 3 dB more amplification for center and surround loudspeaker outputs. 2. By always selecting volume for L and R 3 dB lower than center and surround. Method 1 is preferable, as method 2 lowers the achievable SNR for left and right signals by 3 dB.
2.6.4.3. Loudspeaker Requirements The loudspeakers used and their positioning inside the TV set will greatly influence the performance of the virtualizer. The algorithm works with the direct sound path. Reflected sound waves reduce the effect. So it’s most important to have as much direct sound as possible, compared to indirect sound. To obtain the approval for a TV set, Dolby Laboratories require mounting the loudspeakers at the front of the set. Loudspeakers radiating to the side of the TV set will not produce convincing effects. Good directionality of the loudspeakers towards the listener is optimal. The virtualizer was specially developed for implementation in TV sets. Even for rather small stereo TV's, sufficient sound effects can be obtained. For small sets, the loudspeaker placement should be to the side of the CRT; for large screen sets (or 16:9 sets), mounting the loudspeakers below the CRT is acceptable (large separation is preferred, low frequency speakers should be outmost to avoid cancellation effects). Using external loudspeakers with a large stereo base will not create optimal effects. The loudspeakers should be able to reproduce a wide frequency range. The most important frequency range starts from 160 Hz and ranges up to 5 kHz.
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PRELIMINARY DATA SHEET
MSP 34x2G
2.8. I2S Bus Interface The MSP 34x2G has a synchronous master/slave input/output interface running on 32 kHz. The interface accepts two formats: 1. I2S_WS changes at the word boundary 2. I2S_WS changes one I2S-clock period before the word boundaries. All I2S options are set by means of the MODUS and the I2S_CONFIG registers. The synchronous I2S bus interface consists of five pins: – I2S_DA_IN1, I2S_DA_IN2: I2S serial data input: 16, 18....32 bits per sample – I2S_DA_OUT: I2S serial data output: 16, 18...32 bits per sample – I2S_CL: I2S serial clock – I2S_WS: I2S word strobe signal defines the left and right sample If the MSP 34x2G serves as the master on the I2S interface, the clock and word strobe lines are driven by the IC. In this mode, only 16 or 32 bits per sample can be selected. In slave mode, these lines are input to the IC and the MSP clock is synchronized to 576 times the I2S_WS rate (32 kHz). NICAM operation is not possible in slave mode. An I2S timing diagram is shown in Fig. 4–22 on page 71.
2.6.6. Subwoofer in Surround Mode If the channel configuration is set to OFF or TWO_CHANNEL, the subwoofer signal is created by combining the left and right channels directly behind the loudness block using the formula (L+R)/2. Note: This is identical to the MSP 34x0G. If the channel configuration is MULTI_CHANNEL, the subwoofer signal is created by combining the left and right channels of the loudspeaker channel and the center signal (= headphone left) directly behind the loudness block using the formula (L+R+C)/2. Due to the fact, that the subwoofer is formed behind all bass/ treble/loudness filters, it is strongly recommended to have exactly the same setting for these filters in both, the loudspeaker and center/surround channels when using the subwoofer output. Any mismatch in these settings will result in an unbalanced mix of L, C and R for the subwoofer signal.
2.6.7. Equalizer in Surround Mode In the MULTI_CHANNEL mode, the equalizer cannot be used.
2.7. SCART Signal Routing 2.7.1. SCART DSP In and SCART Out Select The SCART DSP Input Select and SCART Output Select blocks include full matrix switching facilities. To design a TV set with four pairs of SCART-inputs and two pairs of SCART-outputs, no external switching hardware is required. The switches are controlled by the ACB user register (see Table 3–11on page 42).
2.7.2. Stand-by Mode If the MSP 34x2G is switched off by first pulling STANDBYQ low and then (after >1 µs delay) switching off the 5-V, but keeping the 8-V power supply (‘Standby’-mode), the SCART switches maintain their position and function. This allows the copying from selected SCART-inputs to SCART-outputs in the TV set’s stand-by mode. In case of power on or starting from stand-by (switching on the 5-V supply, RESETQ going high 2 ms later), all internal registers except the ACB register (page page 42) are reset to the default configuration (see Table 3–5 on page 24). The reset position of the ACB register becomes active after the first I2C transmission into the Baseband Processing part (subaddress 12hex). By transmitting the ACB register first, the reset state can be redefined.
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MSP 34x2G
2.9. ADR Bus Interface For the ASTRA Digital Radio System (ADR), the MSP 3402G, MSP 3412G and MSP 3452G performs preprocessing such as carrier selection and filtering. Via the 3-line ADR-bus, the resulting signals are transferred to the DRP 3510A coprocessor, where the source decoding is performed. To be prepared for an upgrade to ADR with an additional DRP board, the following lines of MSP 34x2G should be provided on a feature connector: – AUD_CL_OUT – I2S_DA_IN1 or I2S_DA_IN2 – I2S_DA_OUT – I2S_WS – I2S_CL – ADR_CL, ADR_WS, ADR_DA For more details, please refer to the DRP 3510A data sheet.
PRELIMINARY DATA SHEET
2.11.Clock PLL Oscillator and Crystal Specifications The MSP 34x2G derives all internal system clocks from the 18.432 MHz oscillator. In NICAM or in I2SSlave mode, the clock is phase-locked to the corresponding source. Therefore, it is not possible to use NICAM and I2S-Slave mode at the same time. For proper performance, the on-chip clock oscillator requires a 18.432 MHz crystal. Note that for the phase-locked modes (NICAM, I2S-Slave), crystals with tighter tolerance are required.
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 resulting clock frequency. The nominal free running frequency should match 18.432 MHz as closely as possible. Clock measurements should be done at pin AUD_CL_OUT. This pin must be activated for this purpose (see Table 3–9 on page 30).
2.10.Digital Control I/O Pins and Status Change Indication The static level of the digital input/output pins D_CTR_I/O_0/1 is switchable between HIGH and LOW via the I2C-bus by means of the ACB register (see Table 3–11on page page 42). This enables the controlling of external hardware switches or other devices via I2C-bus. The digital input/output pins can be set to high impedance by means of the MODUS register (see Table 3–9 on page 30). In this mode, the pins can be used as input. The current state can be read out of the STATUS register (see Table 3–9 on page page 31). Optionally, the pin D_CTR_I/O_1 can be used as an interrupt request signal to the controller, indicating any changes in the read register STATUS. This makes polling unnecessary, I2C bus interactions are reduced to a minimum.
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PRELIMINARY DATA SHEET
MSP 34x2G
typical response time is about 0.3 ms. If the MSP cannot accept another complete byte of data until it has performed some other function (for example, servicing an internal interrupt), it will hold the clock line I2C_CL LOW to force the transmitter into a wait state. The positions within a transmission where this may happen are indicated by “Wait” in Section 3.1.4. The maximum wait period of the MSP during normal operation mode is less than 1 ms.
3. Control Interface 3.1. I2C Bus Interface 3.1.1. Device and Subaddresses The MSP 34x2G is controlled via the I2C bus slave interface. The IC is selected by transmitting one of the MSP 34x2G device addresses. In order to allow up to three MSP ICs to be connected to a single bus, an address select pin (ADR_SEL) has been implemented. With ADR_SEL pulled to high, low, or left open, the MSP 34x2G responds to different device addresses. A device address pair is defined as a write address and a read address (see Table 3–1). Writing is done by sending the device write address, followed by the subaddress byte, two address bytes, and two data bytes. Reading is done by sending the write device address, followed by the subaddress byte and two address bytes. Without sending a stop condition, reading of the addressed data is completed by sending the device read address and reading two bytes of data. Refer to Section 3.1.4. for the I2C bus protocol and to Section 3.4. “Programming Tips” on page 47 for proposals of MSP 34x2G I2C telegrams. See Table 3–2 for a list of available subaddresses. Besides the possibility of hardware reset, the MSP can also be reset by means of the RESET bit in the CONTROL register by the controller via I2C bus. Due to the internal architecture of the MSP 34x2G, the IC cannot react immediately to an I2C request. The Table 3–1: I2C Bus Device Addresses
ADR_SEL Mode MSP device address Write 80hex Low Read 81hex
3.1.2. Internal Hardware Error Handling In case of any internal hardware error (e.g. interruption of the power supply of the MSP), the MSP’s wait period is extended to 1.8 ms. After this time period elapses, the MSP releases data and clock lines.
Indicating and solving the error status: To indicate the error status, the remaining acknowledge bits of the actual I2C-protocol will be left high. Additionally, bit[14] of CONTROL is set to one. The MSP can then be reset via the I2C bus by transmitting the reset condition to CONTROL.
Indication of reset: Any reset, even caused by an unstable reset line etc., is indicated in bit[15] of CONTROL. A general timing diagram of the I2C bus is shown in Fig. 4–21 on page 69.
High Write 84hex Read 85hex Write 88hex
Left Open Read 89hex
Table 3–2: I2C Bus Subaddresses
Name CONTROL TEST WR_DEM RD_DEM WR_DSP RD_DSP Binary Value 0000 0000 0000 0001 0001 0000 0001 0001 0001 0010 0001 0011 Hex Value 00 01 10 11 12 13 Mode Read/Write Write Write Write Write Write Function Write: Software reset of MSP (see Table 3–3) Read: Hardware error status of MSP only for internal use write address demodulator read address demodulator write address DSP read address DSP
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3.1.3. Description of CONTROL Register
PRELIMINARY DATA SHEET
Table 3–3: CONTROL as a Write Register
Name CONTROL Subaddress 00 hex Bit[15] (MSB) 1 : RESET 0 : normal Bits[14:0] 0
Table 3–4: CONTROL as a Read Register (only MSP 34x2G-versions from A2 on)
Name CONTROL Subaddress 00 hex Bit[15] (MSB) Reset status after last reading of CONTROL: 0 : no reset occured 1 : reset occured Bit[14] Internal hardware status: 0 : no error occured 1 : internal error occured Bits[13:0] not of interest
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 resetted.
3.1.4. Protocol Description Write to DSP or Demodulator
S Wait write device address ACK sub-addr ACK addr-byte ACK addr-byte ACK data-byte- ACK data-byte ACK P high low high low
Read from DSP or Demodulator
S Wait write device address ACK sub-addr ACK addr-byte ACK addr-byte ACK S high low read device address Wait ACK data-byte- ACK data-byte NAK P high low
Write to Control or Test Registers
S Wait write device address ACK sub-addr ACK data-byte ACK data-byte ACK P high low
Read from Control Register
S Wait write device address ACK 00hex ACK S read device address Wait ACK data-byte- ACK data-byte NAK P high low
Note: S = P= ACK = NAK = Wait =
I2C-Bus Start Condition from master I2C-Bus Stop Condition from master Acknowledge-Bit: LOW on I2C_DA from slave (= MSP, light gray) or master (= controller dark gray) Not Acknowledge-Bit: HIGH on I2C_DA from master (dark gray) to indicate ‘End of Read’ or from MSP indicating internal error state I2C-Clock line is held low, while the MSP is processing the I2C command. This waiting time is max. 1 ms
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PRELIMINARY DATA SHEET
MSP 34x2G
I2C_DA S I2C_CL
1 0 P
Fig. 3–1: I2C bus protocol (MSB first; data must be stable while clock is high) 3.1.5. Proposals for General MSP 34x2G I2C Telegrams
3.2. Start-Up Sequence: Power-Up and I2C Controlling After POWER ON or RESET (see Fig. 4–20), the IC is in an inactive state. All registers are in the reset position (see Table 3–5 and Table 3–6), the analog outputs are muted. The controller has to initialize all registers for which a non-default setting is necessary.
3.1.5.1. Symbols
daw dar
< > aa dd
write device address (80hex, 84hex or 88hex) read device address (81hex, 85hex or 89hex) Start Condition Stop Condition Address Byte Data Byte
3.3. MSP 34x2G Programming Interface 3.3.1. User Registers Overview
3.1.5.2. Write Telegrams
write to CONTROL register write data into demodulator write data into DSP
The MSP 34x2G is controlled by means of user registers. The complete list of all user registers is given in the following tables. The registers are partitioned into the Demodulator section (Subaddress 10hex for writing, 11hex for reading) and the Baseband Processing sections (Subaddress 12hex for writing, 13hex for reading). Write and read registers are 16-bit wide, whereby the MSB is denoted bit[15]. Transmissions via I2C bus have to take place in 16-bit words (two byte transfers, with the most significant byte transferred first). All write registers, except the demodulator write registers, are readable. Unused parts of the 16-bit write registers must be zero. Addresses not given in this table must not be accessed. For reasons of software compatibility to the MSP 34x0D, an Manual/Compatibility Mode is available. More read and write registers together with a detailed description of this mode can be found in the “Appendix B: Manual/Compatibility Mode” on page 85. An overview of all MSP 34x2G Write Registers is shown in Table 3–5; all Read Registers are given in Table 3–6.
3.1.5.3. Read Telegrams
read data from CONTROL register 2 ms
Internal Reset
High
Low
t/ms Fig. 4–20: Power-up sequence
68
Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
4.6.3.4. I2C-Bus Characteristics
Symbol VI2CIL VI2CIH tI2C1 tI2C2 tI2C5 tI2C6 tI2C3 tI2C4 fI2C VI2COL II2COH tI2COL1 tI2COL2 Parameter I C-Bus Input Low Voltage I2C-Bus Input High Voltage I2C Start Condition Setup Time I2C Stop Condition Setup Time I2C-Data Setup Time before Rising Edge of Clock I2C-Data Hold Time after Falling Edge of Clock I2C-Clock Low Pulse Time I2C-Clock High Pulse Time I2C-BUS Frequency I2C-Data Output Low Voltage I2C-Data Output High Leakage Current I2C-Data Output Hold Time after Falling Edge of Clock I2C-Data Output Setup Time before Rising Edge of Clock 15 I2C_CL, I2C_DA I2C_CL
2
Pin Name I2C_CL, I2C_DA
Min.
Typ.
Max. 0.3
Unit VSUP2 VSUP2 ns ns ns
Test Conditions
0.6 120 120 55
55
ns
500 500 1.0 0.4 1.0
ns ns MHz V µA II2COL = 3 mA VI2COH = 5 V
ns
100
ns
fI2C = 1 MHz
1/FI2C I2C_CL TI2C4 TI2C3
TI2C1 I2C_DA as input
TI2C5
TI2C6
TI2C2
TI2COL2 I2C_DA as output
TI2COL1
Fig. 4–21: I2C bus timing diagram
Micronas
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MSP 34x2G
4.6.3.5. I2S-Bus Characteristics
Symbol VI2SIL VI2SIH ZI2SI ILEAKI2S VI2SOL VI2SOH fI2SOWS fI2SOCL RI2S10/I2S20 ts_I2S th_I2S td_I2S Parameter Input Low Voltage Input High Voltage Input Impedance Input Leakage Current I2S Output Low Voltage I2S_CL I2S_WS I2S_DA_OUT I2S_WS I2S_CL 0.9 I2S_CL I2S_DA_IN1/2 12 40 I2S_CL I2S_WS I2S_DA_OUT I2S_WS I2S_CL 0.9 32.0 1.024 1.1 28 −1 Pin Name I2S_CL I2S_WS I2S_DA_IN1/2 Min. Typ. Max. 0.2 0.5 5 1 0.4 VSUP2 − 0.3 32.0 1.024 1.0 1.1 ns ns ns
PRELIMINARY DATA SHEET
Unit VSUP2 VSUP2 pF µA V V kHz MHz
Test Conditions
0 V < UINPUT< DVSUP II2SOL = 1 mA II2SOH = −1 mA
I2S Output High Voltage I2S-Word Strobe Output Frequency I2S-Clock Output Frequency I S-Clock Output High/Low-Ratio I S Input Setup Time before Rising Edge of Clock I2S Input Hold Time after Rising Edge of Clock I2S Output Delay Time after Falling Edge of Clock I2S-Word Strobe Input Frequency I2S-Clock Input Frequency I S-Clock Input Ratio
2 2 2
for details see Fig. 4–22 “I2S bus timing diagram”
CL = 30 pF
fI2SWS fI2SCL RI2SCL
kHz MHz
70
Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
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
16/32 bit left channel Detail B I2S_DA_OUT R LSB
L MSB L LSB R MSB
16/32 bit right channel
R LSB L LSB
16/32 bit left channel
16/32 bit right channel
Data: MSB first, I2S master
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
16,18...32 bit left channel Detail B I2S_DA_OUT R LSB
L MSB
16, 18...32 bit right channel
16, 18...32 bit left channel
L LSB R MSB R LSB L LSB
16, 18...32 bit right channel
Data: MSB first, I2S slave
Detail C
I2S_CL
1/FI2SCL
Detail A,B
I2S_CL
Ts_I2S Ts_I2S I2S_DA_IN1/2 I2S_WS as INPUT
Th_I2S
Td_I2S
Td_I2S
I2S_WS as OUTPUT
I2S_DA_OUT
Fig. 4–22: I2S bus timing diagram
Micronas
71
MSP 34x2G
4.6.3.6. Analog Baseband Inputs and Outputs, AGNDC
Symbol Parameter Pin Name Min. Typ. Max.
PRELIMINARY DATA SHEET
Unit
Test Conditions
Analog Ground VAGNDC0 AGNDC Open Circuit Voltage (AHVSUP = 8 V) AGNDC Open Circuit Voltage (AHVSUP = 8 V) RoutAGN AGNDC Output Resistance (AHVSUP = 8 V) AGNDC Output Resistance (AHVSUP = 8 V) Analog Input Resistance RinSC RinMONO
1)
AGNDC
3.67
3.77
3.87
V
Rload ≥ 10 MΩ
2.41
2.51
2.61
V kΩ kΩ 3 V ≤ VAGNDC ≤ 4 V
70
125
180
47
83
120
SCART Input Resistance from TA = 0 to 70 °C MONO Input Resistance from TA = 0 to 70 °C
SCn_IN_s1)
25
40
58
kΩ kΩ
fsignal = 1 kHz, I = 0.05 mA fsignal = 1 kHz, I = 0.1 mA
MONO_IN
15
24
35
“n” means “1”, “2”, “3”, or “4”; “s” means “L” or “R”
72
Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
Symbol
Parameter
Pin Name
Min.
Typ.
Max.
Unit
Test Conditions
Audio Analog-to-Digital-Converter VAICL Effective Analog Input Clipping Level for Analog-to-DigitalConversion (AHVSUP = 8 V) Effective Analog Input Clipping Level for Analog-to-DigitalConversion (AHVSUP = 5 V) SCART Outputs RoutSC SCART Output Resistance at Tj = 27 °C from TA = 0 to 70 °C 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 Effective Signal Level at SCART-Output during full-scale Digital Input Signal from I2S (AHVSUP = 8 V) Effective Signal Level at SCART-Output during full-scale Digital Input Signal from I2S (AHVSUP = 5 V) Main, AUX, and CS Outputs RoutMACS Main/AUX Output Resistance at Tj = 27 °C from TA = 0 to 70 °C DC-Level at Main/AUX-Output for Analog Volume at 0 dB for Analog Volume at −30 dB DC-Level, not selected CS-Output (AHVSUP = 8 V) DC-Level at Main/AUX-Output for Analog Volume at 0 dB for Analog Volume at −30 dB DC-Level, not selected CS-Output (AHVSUP = 5 V) VoutMACS Effective Signal Level at Main/AUX-Output during full-scale Digital Input Signal from I2S for Analog Volume at 0 dB (AHVSUP = 8 V) Effective Signal Level at Main/AUX-Output during full-scale Digital Input Signal from I2S for Analog Volume at 0 dB (AHVSUP = 5 V)
1)
SCn_IN_s,1) MONO_IN
2.00
2.25
VRMS
fsignal = 1 kHz
1.13
1.51
VRMS
SCn_OUT_s1) 200 200 −70 SCn_IN_s,1) MONO_IN → SCn_OUT_s1) −1.0 −0.5 330 460 500 +70 +0.5 +0.5
Ω Ω mV
fsignal = 1 kHz, I = 0.1 mA
dVOUTSC ASCtoSC frSCtoSC
dB
fsignal = 1 kHz with resp. to 1 kHz
dB
VoutSC
SCn_OUT_s1)
1.8
1.9
2.0
VRMS
fsignal = 1 kHz
1.17
1.27
1.37
VRMS
DACM_r,1) DACA_s
2.1 2.1
3.3
4.6 5.0
kΩ kΩ
fsignal = 1 kHz, I = 0.1 mA
VoutDCMACS
1.80
2.04 61 0
2.28
V mV V
1.12
1.36 40 0
1.60
V mV V
1.23
1.37
1.51
VRMS
fsignal = 1 kHz
0.76
0.90
1.04
VRMS
“n” means “1”, “2”, “3”, or “4”; “r” means “L”, “R”, “C”, or “S”; “s” means “L” or “R”
Micronas
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MSP 34x2G
4.6.3.7. Sound IF Inputs
Symbol RIFIN Parameter Input Impedance Pin Name ANA_IN1+, ANA_IN2+, ANA_IN− VREFTOP ANA_IN1+, ANA_IN2+, ANA_IN− ANA_IN1+, ANA_IN2+, ANA_IN− Min. 1.5 6.8 Typ. 2 9.1 Max. 2.5 11.4
PRELIMINARY DATA SHEET
Unit kΩ kΩ
Test Conditions Gain AGC = 20 dB Gain AGC = 3 dB
DCVREFTOP DCANA_IN
DC Voltage at VREFTOP DC Voltage on IF Inputs
2.45 1.3
2.65 1.5
2.75 1.7
V V
XTALKIF BWIF AGC
Crosstalk Attenuation 3 dB Bandwidth AGC Step Width
40 10 0.85
dB MHz dB
fsignal = 1 MHz Input Level = −2 dBr
4.6.3.8. Power Supply Rejection
Symbol Parameter Pin Name Min. Typ. Max. Unit Test Conditions
PSRR: Rejection of Noise on AHVSUP at 1 kHz PSRR AGNDC From Analog Input to I S Output
2
AGNDC MONO_IN, SCn_IN_s1) MONO_IN, SCn_IN_s1) SCn_OUT_s1) SCn_OUT_s1) DACM_r, DACA_s
1)
80 70
dB dB
From Analog Input to SCART Output From I2S Input to SCART Output From I S Input to Main or AUX Output
1) 2
70
dB
60 80
dB dB
“n” means “1”, “2”, “3”, or “4”; “r” means “L”, “R”, “C”, or “S”; “s” means “L” or “R”
74
Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
4.6.3.9. Analog Performance
Symbol Parameter Pin Name Min. Typ. Max. Unit Test Conditions
Specifications for AHVSUP = 8 V SNR Signal-to-Noise Ratio from Analog Input to I2S Output MONO_IN, SCn_IN_s1) 85 88 dB Input Level = −20 dB with resp. to VAICL, fsig = 1 kHz, unweighted 20 Hz...16 kHz Input Level = −20 dB, fsig = 1 kHz, unweighted 20 Hz...20 kHz Input Level = −20 dB, fsig = 1 kHz, unweighted 20 Hz...15 kHz Input Level = −20 dB, fsig = 1 kHz, unweighted 20 Hz...15 kHz
from Analog Input to SCART Output
MONO_IN, SCn_IN_s1) → SCn_OUT_s1) SCn_OUT_s1)
93
96
dB
from I2S Input to SCART Output
85
88
dB
from I2S Input to Main/AUX-Output for Analog Volume at 0 dB for Analog Volume at −30 dB
DACM_r,1) DACA_s
85 78
88 83
dB dB
THD
Total Harmonic Distortion from Analog Input to I2S Output MONO_IN, SCn_IN_s1) 0.01 0.03 % Input Level = −3 dBr with resp. to VAICL, fsig = 1 kHz, unweighted 20 Hz...16 kHz Input Level = −3 dBr, fsig = 1 kHz, unweighted 20 Hz...20 kHz Input Level = −3 dBr, fsig = 1 kHz, unweighted 20 Hz...16 kHz Input Level = −3 dBr, fsig = 1 kHz, unweighted 20 Hz...16 kHz
from Analog Input to SCART Output
MONO_IN, SCn_IN_s → SCn_OUT_s1) SCn_OUT_s1)
0.01
0.03
%
from I2S Input to SCART Output
0.01
0.03
%
from I2S Input to Main or AUX Output
DACM_r,1) DACA_s
0.01
0.03
%
1)
“n” means “1”, “2”, “3”, or “4”; “r” means “L”, “R”, “C”, or “S”; “s” means “L” or “R”
Micronas
75
MSP 34x2G
PRELIMINARY DATA SHEET
Symbol
Parameter
Pin Name
Min.
Typ.
Max.
Unit
Test Conditions
Specifications for AHVSUP = 5 V SNR Signal-to-Noise Ratio from Analog Input to I2S Output MONO_IN, SCn_IN_s1) 82 85 dB Input Level = −20 dB with resp. to VAICL, fsig = 1 kHz, unweighted 20 Hz...16 kHz Input Level = −20 dB, fsig = 1 kHz, unweighted 20 Hz...20 kHz Input Level = −20 dB, fsig = 1 kHz, unweighted 20 Hz...15 kHz Input Level = −20 dB, fsig = 1 kHz, unweighted 20 Hz...15 kHz
from Analog Input to SCART Output
MONO_IN, SCn_IN_s1) → SCn_OUT_s1) SCn_OUT_s1)
90
93
dB
from I2S Input to SCART Output
82
85
dB
from I2S Input to Main/AUX-Output for Analog Volume at 0 dB for Analog Volume at −30 dB
DACM_r,1) DACA_s
82 75
85 80
dB dB
THD
Total Harmonic Distortion from Analog Input to I2S Output MONO_IN, SCn_IN_s1) 0.03 0.1 % Input Level = −3 dBr with resp. to VAICL, fsig = 1 kHz, unweighted 20 Hz...16 kHz Input Level = −3 dBr, fsig = 1 kHz, unweighted 20 Hz...20 kHz Input Level = −3 dBr, fsig = 1 kHz, unweighted 20 Hz...16 kHz Input Level = −3 dBr, fsig = 1 kHz, unweighted 20 Hz...16 kHz
from Analog Input to SCART Output
MONO_IN, SCn_IN_s → SCn_OUT_s1) SCn_OUT_s1)
0.1
%
from I2S Input to SCART Output
0.1
%
from I2S Input to Main or AUX Output
DACA_s, DACM_s1)
0.1
%
1)
“n” means “1”, “2”, “3”, or “4”; “r” means “L”, “R”, “C”, or “S”; “s” means “L” or “R”
76
Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
Symbol
Parameter
Pin Name
Min.
Typ.
Max.
Unit
Test Conditions
CROSSTALK Specifications XTALK Crosstalk Attenuation Input Level = −3 dB, fsig = 1 kHz, unused analog inputs connected to ground by Z < 1 kΩ unweighted 20 Hz...20 kHz 80 80 80 80 dB dB dB dB unweighted 20 Hz...16 kHz 75 dB
between left and right channel within SCART Input/Output pair (L→R, R→L) SCn_IN → SCn_OUT1) SC1_IN or SC2_IN → I2S Output SC3_IN → I2S Output I2S Input → SCn_OUT1) between left and right channel within Main or AUX Output pair I2S Input → DACM I2S Input → DACA between SCART Input/Output pairs D = disturbing program O = observed program D: MONO/SCn_IN → SCn_OUT O: MONO/SCn_IN → SCn_OUT1) D: MONO/SCn_IN → SCn_OUT or unsel. O: MONO/SCn_IN → I2S Output D: MONO/SCn_IN → SCn_OUT O: I2S Input → SCn_OUT1) D: MONO/SCn_IN → unselected O: I2S Input → SC1_OUT1) Crosstalk between Main and AUX Output pairs I2S Input → DACM I2S Input → DACA 90 dB 100 95 100 100 dB dB dB dB
unweighted 20 Hz...20 kHz same signal source on left and right disturbing channel, effect on each observed output channel
unweighted 20 Hz...16 kHz same signal source on left and right disturbing channel, effect on each observed output channel unweighted 20 Hz...20 kHz same signal source on left and right disturbing channel, effect on each observed output channel
XTALK
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 O: I2S Input → DACM O: I2S Input → DACA D: MONO/SCn_IN/DSP → SCn_OUT O: I2S Input → DACM O: I2S Input → DACA D: I2S Input → DACM D: I2S Input → DACA O: MONO/SCn_IN → SCn_OUT1) D: I2S Input → DACM D: I2S Input → DACA O: I2S Input → SCn_OUT1) 80 dB
SCART output load resistance 10 kΩ SCART output load resistance 30 kΩ
85
dB
95
dB
95
dB
1)
“n” means “1”, “2”, “3”, or “4”
Micronas
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MSP 34x2G
4.6.3.10. Sound Standard Dependent Characteristics
Symbol Parameter Pin Name Min. Typ. Max.
PRELIMINARY DATA SHEET
Unit
Test Conditions
NICAM Characteristics (MSP Standard Code = 8) dVNICAMOUT S/NNICAM Tolerance of Output Voltage of NICAM Baseband Signal S/N of NICAM Baseband Signal DACM_r,1) DACA_s, SCn_OUT_s −1.5 +1.5 dB 2.12 kHz, Modulator input level = 0 dBref NICAM: −6 dB, 1 kHz, RMS unweighted 0 to 15 kHz, Vol = 9 dB NIC_Presc = 7Fhex Output level 1 VRMS at DACp_s 2.12 kHz, Modulator input level = 0 dBref FM+NICAM, norm conditions Modulator input level = −12 dB dBref; RMS
72
dB
THDNICAM BERNICAM fRNICAM XTALKNICAM SEPNICAM
Total Harmonic Distortion + Noise of NICAM Baseband Signal NICAM: Bit Error Rate NICAM Frequency Response , 20...15000 Hz NICAM Crosstalk Attenuation (Dual) NICAM Channel Separation (Stereo) −1.0
0.1
% 10−7 dB
1 +1.0
80 80
dB dB
FM Characteristics (MSP Standard Code = 3) dVFMOUT S/NFM THDFM Tolerance of Output Voltage of FM Demodulated Signal S/N of FM Demodulated Signal Total Harmonic Distortion + Noise of FM Demodulated Signal DACM_r,1) DACA_s, SCn_OUT_s1) −1.5 +1.5 dB 1 FM-carrier, 50 µs, 1 kHz, 40 kHz deviation; RMS 1 FM-carrier 5.5 MHz, 50 µs, 1 kHz, 40 kHz deviation; RMS, unweighted 0 to 15 kHz (for S/N); full input range, FM-Prescale = 46hex, Vol = 0 dB → Output Level 1 VRMS at DACp_s 1 FM-carrier 5.5 MHz, 50 µs, Modulator input level = −14.6 dBref; RMS 2 FM-carriers 5.5/5.74 MHz, 50 µs, 1 kHz, 40 kHz deviation; Bandpass 1 kHz 2 FM-carriers 5.5/5.74 MHz, 50 µs, 1 kHz, 40 kHz deviation; RMS
73 0.1
dB %
fRFM
FM Frequency Response 20...15000 Hz
−1.0
+1.0
dB
XTALKFM
FM Crosstalk Attenuation (Dual)
80
dB
SEPFM
FM Channel Separation (Stereo)
DACM_r,1) DACA_s, SCn_OUT_s
50
dB
AM Characteristics (MSP Standard Code = 9) S/NAM(1) S/NAM(2) THDAM fRRM S/N of AM Demodulated Signal measurement condition: RMS/Flat S/N of AM Demodulated Signal measurement condition: QP/CCIR Total Harmonic Distortion + Noise of AM Demodulated Signal RM Frequency Response 50...12000 Hz −2.5 DACM_r,1) DACA_s, SCn_OUT_s 55 dB SIF level: 0.1−0.8 Vpp AM-carrier 54% at 6.5 MHz Vol = 0 dB, FM/AM prescaler set for output = 0.5 VRMS at Loudspeaker out; Standard Code = 09hex no video/chroma components
45
dB
0.6 +1.0
%
dB
1) “n” means “1”, “2”, “3”, or “4”; “r” means “L”, “R”, “C”, or “S”; “s” means “L” or “R”
78
Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
Symbol
Parameter
Pin Name
Min.
Typ.
Max.
Unit
Test Conditions
BTSC Characteristics (MSP Standard Code = 20hex, 21hex) S/NBTSC S/N of BTSC Stereo Signal S/N of BTSC-SAP Signal DACM_r,1) DACA_s, SCn_OUT_s 68 57 dB dB 1 kHz L or R or SAP, 100% modulation, 75 µs deemphasis, RMS unweighted 0 to 15 kHz 1 kHz L or R or SAP, 100% 75 µs EIM2), DBX NR, RMS unweighted 0 to 15 kHz L or R or SAP, 1%...66% EIM2), DBX NR
THDBTSC
THD+N of BTSC Stereo Signal THD+N of BTSC SAP Signal
0.1 0.5
% %
fRBTSC
Frequency Response of BTSC Stereo, 50 Hz...12 kHz Frequency Response of BTSCSAP, 50 Hz...9 kHz
−1.0 −1.0
1.0 1.0
dB dB
XTALKBTSC
Stereo → SAP SAP → Stereo
76 80
dB dB
1 kHz L or R or SAP, 100% modulation, 75 µs deemphasis, Bandpass 1 kHz L or R 1%...66% EIM2), DBX NR
SEPBTSC
Stereo Separation 50 Hz...10 kHz 50 Hz...12 kHz Pilot deviation threshold Stereo off → on Stereo on → off ANA_IN1+, ANA_IN2+
35 30
dB dB
FMpil
3.2 1.2
3.5 1.5 15.843
kHz kHz kHz
4.5 MHz carrier modulated with fh = 15.743 kHz SIF level = 100 mVpp indication: STATUS Bit[6] standard BTSC stereo signal, sound carrier only
fPilot
Pilot Frequency Range
ANA_IN1+ ANA_IN2+
15.563
BTSC Characteristics (MSP Standard Code = 20hex, 21hex) with a minimum IF input signal level of 70 mVpp (measured without any video/chroma signal components) S/NBTSC S/N of BTSC Stereo Signal S/N of BTSC-SAP Signal DACM_r,1) DACA_s, SCn_OUT_s 64 55 dB dB 1 kHz L or R or SAP, 100% modulation, 75 µs deemphasis, RMS unweighted 0 to 15 kHz 1 kHz L or R or SAP, 100% 75 µs EIM2), DBX NR, RMS unweighted 0 to 15 kHz L or R or SAP, 1%...66% EIM2), DBX NR
THDBTSC
THD+N of BTSC Stereo Signal THD+N of BTSC SAP Signal
0.15 0.8
% %
fRBTSC
Frequency Response of BTSC Stereo, 50 Hz...12 kHz Frequency Response of BTSCSAP, 50 Hz...9 kHz
−1.0 −1.0
1.0 1.0
dB dB
XTALKBTSC
Stereo → SAP SAP → Stereo
75 75
dB dB
1 kHz L or R or SAP, 100% modulation, 75 µs deemphasis, Bandpass 1 kHz L or R 1%...66% EIM2), DBX NR
SEPBTSC
Stereo Separation 50 Hz...10 kHz 50 Hz...12 kHz
35 30
dB dB
1) 2)
“n” means “1”, “2”, “3”, or “4”; “r” means “L”, “R”, “C”, or “S”; “s” means “L” or “R” EIM refers to 75-µs 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-µs preemphasis network.
Micronas
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MSP 34x2G
PRELIMINARY DATA SHEET
Symbol
Parameter
Pin Name
Min.
Typ.
Max.
Unit
Test Conditions
EIA-J Characteristics (MSP Standard Code = 30hex) S/NEIAJ S/N of EIA-J Stereo Signal S/N of EIA-J Sub-Channel THDEIAJ THD+N of EIA-J Stereo Signal THD+N of EIA-J Sub-Channel fREIAJ Frequency Response of EIA-J Stereo, 50 Hz...12 kHz Frequency Response of EIA-J Sub-Channel, 50 Hz...12 kHz XTALKEIAJ Main → SUB Sub → MAIN SEPEIAJ Stereo Separation 50 Hz...5 kHz 50 Hz...10 kHz −1.0 −1.0 DACM_r,1) DACA_s, SCn_OUT_s 60 60 0.2 0.3 1.0 1.0 dB dB % % dB dB 100% modulation, 75 µs deemphasis 1 kHz L or R, 100% modulation, 75 µs deemphasis, RMS unweighted 0 to 15 kHz
66 80
dB dB
1 kHz L or R, 100% modulation, 75 µs deemphasis, Bandpass 1 kHz EIA-J Stereo Signal, L or R 100% modulation
35 28
dB dB
FM-Radio Characteristics (MSP Standard Code = 40hex) S/NUKW THDUKW fRUKW S/N of FM-Radio Stereo Signal THD+N of FM-Radio Stereo Signal Frequency Response of FM-Radio Stereo 50 Hz...15 kHz Stereo Separation 50 Hz...15 kHz Pilot Frequency Range ANA_IN1+ ANA_IN2+ DACM_r,1) DACA_s, SCn_OUT_s 68 0.1 dB % 1 kHz L or R, 100% modulation, 75 µs deemphasis, RMS unweighted 0 to 15 kHz L or R, 1%...100% modulation, 75 µs deemphasis 1.0 dB dB 19.125 kHz standard FM radio stereo signal
−1.0 45 18.844
SEPUKW fPilot
1)
“n” means “1”, “2”, “3”, or “4”; “r” means “L”, “R”, “C”, or “S”; “s” means “L” or “R”
80
Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
5. Appendix A: Overview of TV-Sound Standards 5.1. NICAM 728
Table 5–1: Summary of NICAM 728 sound modulation parameters
Specification Carrier frequency of digital sound Transmission rate Type of modulation Spectrum shaping Roll-off factor 1.0 Carrier frequency of analog sound component Power ratio between vision carrier and analog sound carrier Power ratio between analog and modulated digital sound carrier 6.0 MHz FM mono 10 dB 0.4 5.5 MHz FM mono 13 dB I 6.552 MHz B/G 5.85 MHz L 5.85 MHz 728 kbit/s Differentially encoded quadrature phase shift keying (DQPSK) by means of Roll-off filters 0.4 6.5 MHz AM mono terrestrial 10 dB cable 16 dB 13 dB 0.4 6.5 MHz FM mono D/K 5.85 MHz
10 dB
7 dB
17 dB
11 dB
China/ Hungary 12 dB
Poland 7 dB
Table 5–2: Summary of NICAM 728 sound coding characteristics
Characteristics Audio sampling frequency Number of channels Initial resolution Companding characteristics Coding for compressed samples Preemphasis Audio overload level Values 32 kHz 2 14 bit/sample near instantaneous, with compression to 10 bits/sample in 32-samples (1 ms) blocks 2’s complement CCITT Recommendation J.17 (6.5 dB attenuation at 800 Hz) +12 dBm measured at the unity gain frequency of the preemphasis network (2 kHz)
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MSP 34x2G
5.2. A2-Systems
PRELIMINARY DATA SHEET
Table 5–3: Key parameters for A2 Systems of Standards B/G, D/K, and M
Characteristics TV-Sound Standard Carrier frequency in MHz B/G 5.5 Sound Carrier FM1 D/K 6.5 M 4.5 B/G 5.7421875 Sound Carrier FM2 D/K 6.2578125 6.7421875 5.7421875 20 dB 40 Hz to 15 kHz 50 µs ±27/±50 kHz 75 µs ±17/±25 kHz 50 µs ±27/±50 kHz 75 µs ±15/±25 kHz M 4.724212
Vision/sound power difference Sound bandwidth Preemphasis Frequency deviation (nom/max) Transmission Modes Mono transmission Stereo transmission Dual sound transmission Identification of Transmission Mode Pilot carrier frequency Max. deviation portion Type of modulation / modulation depth Modulation frequency (L+R)/2
13 dB
mono (L+R)/2 R
mono (L−R)/2 language B
language A
54.6875 kHz
55.0699 kHz
±2.5 kHz
AM / 50% mono: unmodulated stereo: 117.5 Hz dual: 274.1 Hz 149.9 Hz 276.0 Hz
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PRELIMINARY DATA SHEET
MSP 34x2G
5.3. BTSC-Sound System
Table 5–4: Key parameters for BTSC-Sound Systems
Aural Carrier Carrier frequency (fhNTSC = 15.734 kHz) (fhPAL = 15.625 kHz) Sound bandwidth in kHz Preemphasis Max. deviation to Aural Carrier Max. Freq. Deviation of Subcarrier Modulation Type
1)
BTSC-MPX-Components (L+R) Baseband Pilot fh (L−R) 2 fh SAP 5 fh Prof. Ch. 6.5 fh
4.5 MHz
0.05 - 15 75 µs 73 kHz (total) 25 kHz1) 5 kHz
0.05 - 15 DBX 50 kHz1)
0.05 - 12 DBX 15 kHz 10 kHz FM
0.05 - 3.4 150 µs 3 kHz 3 kHz FM
AM
Sum does not exceed 50 kHz due to interleaving effects
5.4. Japanese FM Stereo System (EIA-J)
Table 5–5: Key parameters for Japanese FM-Stereo Sound System EIA-J
Aural Carrier FM Carrier frequency (fh = 15.734 kHz) Sound bandwidth Preemphasis Max. deviation portion to Aural Carrier Max. Freq. Deviation of Subcarrier Modulation Type Transmitter-sided delay Mono transmission Stereo transmission Bilingual transmission 20 µs L+R L+R Language A 47 kHz 4.5 MHz EIA-J-MPX-Components (L+R) Baseband 0.05 - 15 kHz 75 µs 25 kHz (L−R) 2 fh 0.05 - 15 kHz 75 µs 20 kHz 10 kHz FM 0 µs − L−R Language B Identification 3.5 fh − none 2 kHz 60% AM 0 µs unmodulated 982.5 Hz 922.5 Hz
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MSP 34x2G
5.5. FM Satellite Sound
PRELIMINARY DATA SHEET
Table 5–6: Key parameters for FM Satellite Sound
Carrier Frequency 6.5 MHz 7.02/7.20 MHz 7.38/7.56 MHz 7.74/7.92 MHz Maximum FM Deviation 85 kHz 50 kHz 50 kHz 50 kHz Sound Mode Mono Mono/Stereo/Bilingual Mono/Stereo/Bilingual Mono/Stereo/Bilingual Bandwidth 15 kHz 15 kHz 15 kHz 15 kHz Deemphasis 50 µs adaptive adaptive adaptive
5.6. FM-Stereo Radio
Table 5–7: Key parameters for FM-Stereo Radio Systems
Aural Carrier Carrier frequency (fp = 19 kHz) Sound bandwidth in kHz Preemphasis: − USA − Europe Max. deviation to Aural Carrier
1)
FM-Radio-MPX-Components (L+R) Baseband 0.05 - 15 75 µs 50 µs Pilot fp (L−R) 2 fp 0.05 - 15 75 µs 50 µs 10% 90%1) 5% RDS/ARI 3 fh
10.7 MHz
75 kHz (100%)
90%1)
Sum does not exceed 90% due to interleaving effects.
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PRELIMINARY DATA SHEET
MSP 34x2G
6. Appendix B: Manual/Compatibility Mode To adapt the modes of the STANDARD SELECT register to individual requirements and for reasons of compatibility to the MSP 34x0D, the MSP 34x2G offers an Manual/Compatibility Mode, which provides sophisticated programming of the MSP 34x2G. Using the STANDARD SELECT register generally provides a more economic way to program the MSP 34x2G and will result in optimal behavior. Therefore, it is not recommended to use the Manual/ Compatibility mode. In those cases, where the MSP 34x0D is to be substituted by the MSP 34x2G, the tips given in section 6.9. have to be obeyed by the controller software.
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MSP 34x2G
6.1. Demodulator Write and Read Registers for Manual/Compatibility Mode
PRELIMINARY DATA SHEET
Table 6–1: Demodulator Write Registers; Subaddress: 10hex; these registers are not readable!
Demodulator Write Registers AUTO_FM/AM Address (hex) 00 21 MSPVersion 3411, 34511) Description 1. MODUS[0]=1 (Automatic Sound Select): Switching Level threshold of Automatic Switching between NICAM and FM/AM in case of bad NICAM reception 2. MODUS[0]=0 (Manual Mode): Activation and configuration of Automatic Switching between NICAM and FM/AM in case of bad NICAM reception A2_Threshold CM_Threshold AD_CV MODE_REG 00 22 00 24 00 BB 00 83 all all all 3411, 34511) A2 Stereo Identification Threshold Carrier-Mute Threshold SIF-input selection, configuration of AGC, and Carrier-Mute Function Controlling of MSP-Demodulator and Interface options. As soon as this register is applied, the MSP 34x2G works in the MSP 34x0D Compatibility Mode. Warning: In this mode, BTSC, EIA-J, and FM-Radio are disabled. Only MSP 34x0D features are available; the use of MODUS and STATUS register is not allowed. The MSP 34x2G is reset to the normal mode by first programming the MODUS register followed by transmitting a valid standard code to the STANDARD SELECTION register. FIR1 FIR2 DCO1_LO DCO1_HI DCO2_LO DCO2_HI PLL_CAPS
1)
Reset Mode 00 00
Page 87
00 19hex 00 2Ahex 00 00 00 00
89 89 90 91
00 01 00 05 00 93 00 9B 00 A3 00 AB 00 1F
FIR1-filter coefficients channel 1 (6 ⋅ 8 bit) FIR2-filter coefficients channel 2 (6 ⋅ 8 bit), + 3 ⋅ 8 bit offset (total 72 bit) Increment channel 1 Low Part Increment channel 1 High Part Increment channel 2 Low Part Increment channel 2 High Part Not of interest for the customer Switchable PLL capacitors to tune open-loop frequency
00 00 00 00
93 93
00 56
96
not in BTSC, EIA-J, and FM-Radio mode
Note: All registers except AUTO_FM/AM, A2_Threshold and CM-Threshold are initialised during STANDARD SELECTION and are automatically updated when Automatic Sound Select (MODUS[0]=1) is on.
Table 6–2: Demodulator Read Registers; Subaddress: 11hex; these registers are not writable!
Demodulator Read Registers C_AD_BITS ADD_BITS CIB_BITS ERROR_RATE PLL_CAPS AGC_GAIN Address (hex) 00 23 00 38 00 3E 00 57 02 1F 02 1E MSPVersion 3411, 3451 Description NICAM-Sync bit, NICAM-C-Bits, and three LSBs of additional data bits NICAM: bit [10:3] of additional data bits NICAM: CIB1 and CIB2 control bits NICAM error rate, updated with 182 ms Not for customer use Not for customer use Page 95 95 95 96 96 96
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PRELIMINARY DATA SHEET
MSP 34x2G
6.2. DSP Write and Read Registers for Manual/Compatibility Mode Table 6–3: DSP-Write Registers; Subaddress: 12hex, all registers are readable as well
Write Register Volume SCART1 channel: Ctrl. mode FM Fixed Deemphasis FM Adaptive Deemphasis Identification Mode FM DC Notch Volume SCART2 channel: Ctrl. mode 00 15 00 17 00 40 Address (hex) 00 07 00 0F Bits [7..0] [15..8] [7..0] [7..0] [7..0] [7..0] Operational Modes and Adjustable Range [Linear mode / logarithmic mode] [50 µs, 75 µs, OFF] [OFF, WP1] [B/G, M] [ON, OFF] [Linear mode / logarithmic mode] Reset Mode 00hex 50 µs OFF B/G ON 00hex Page 97 97 97 98 98 97
Table 6–4: DSP Read Registers; Subaddress: 13hex, all registers are not writable
Additional Read Registers Stereo detection register for A2 Stereo Systems DC level readout FM1/Ch2-L DC level readout FM2/Ch1-R Address (hex) 00 18 00 1B 00 1C Bits [15..8] [15..0] [15..0] Output Range [80hex ... 7Fhex] [8000hex ... 7FFFhex] [8000hex ... 7FFFhex] 8 bit two’s complement 16 bit two’s complement 16 bit two’s complement Page 98 98 98
6.3. Manual/Compatibility Mode: Description of Demodulator Write Registers 6.3.1. Automatic Switching between NICAM and Analog Sound In case of bad NICAM reception or loss of the NICAM-carrier, the MSP 34x2G offers an Automatic Switching (fall back) to the analog sound (FM/AMMono), without the necessity of the controller reading and evaluating any parameters. If a proper NICAM signal returns, switching back to this source is performed automatically as well. The feature evaluates the NICAM ERROR_RATE and switches, if necessary, all output channels which are assigned to the NICAM source, to the analog source, and vice versa. An appropriate hysteresis algorithm avoids oscillating effects (see Fig. 6–1). STATUS[9] and C_AD_BITS[11] (Addr: 0023 hex) provide information about the actual NICAM-FM/AM-status.
Selected Sound
NICAM
analog Sound
ERROR_RATE threshold/2 threshold
Fig. 6–1: Hysteresis for Automatic Switching
6.3.1.1. Function in Automatic Sound Select Mode The Automatic Sound Select feature (MODUS[0]=1) includes the procedure mentioned above. By default, the internal ERROR_RATE threshold is set to 700dec. i.e. : – NICAM → analog Sound if ERROR_RATE > 700 – analog Sound → NICAM if ERROR_RATE < 700/2 The ERROR_RATE value of 700 corresponds to a BER of approximately 5.46*10-3/s.
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MSP 34x2G
Individual configuration of the threshold can be done using Table 6–5, whereby the bits 0 and 11 of AUTO_FM are ignored. It is recommended to use the internal setting used by the standard selection. The optimum NICAM sound can be assigned to the MSP output channels by selecting one of the “Stereo or A/B”, “Stereo or A”, or “Stereo or B” source channels.
PRELIMINARY DATA SHEET
6.3.1.2. Function in Manual Mode If the manual mode (MODUS[0]=0) is required, the activation and configuration of the Automatic Switching feature has to be done as described in Table 6–5. Note, that the channel matrix of the corresponding output channels must be set according to the NICAM mode and need not to be changed in the FM/ AM-fallback case. Example: Required threshold = 500: bits [10:1]=00 1111 1010
Table 6–5: Coding of Automatic NICAM/Analog Sound Switching; Reset Status: Mode 0; Automatic Sound Select is on (MODUS[0] = 1)
Mode 1 Description Automatic Switching with internal threshold (Default, if Automatic Sound Select is on) Automatic Switching with external threshold (Customizing of Automatic Sound Select) AUTO_FM [11:0] Addr. = 00 21hex bit[11] = ignored bit[10:1] = 0 bit[0] = ignored bit[11] = ignored bit[10:1] = 25...1000 = threshold/2 bit[0] = ignored ERROR_RATEThreshold/dec 700 Source Select: Input at NICAM Path1) NICAM or FM/AM, depending on ERROR_RATE
2
set by customer; recommended range: 50...2000
1)
The NICAM path may be assigned to “Stereo or A/B”, “Stereo or A”, or “Stereo or B” source channels (see Table 2–2 on page 13).
Table 6–6: Coding of Automatic NICAM/Analog Sound Switching; Reset Status: Mode 0; Automatic Sound Select is off (MODUS[0] = 0)
Mode 0 Description Forced NICAM (Automatic Switching disabled) Automatic Switching with internal threshold (Default, if Automatic Sound Select is on) Automatic Switching with external threshold (Customizing of Automatic Sound Select) Forced Analog Mono (Automatic Switching disabled) AUTO_FM [11:0] Addr. = 00 21hex bit[11] =0 bit[10:1] = 0 bit[0] =0 bit[11] =0 bit[10:1] = 0 bit[0] =1 bit[11] =0 bit[10:1] = 25...1000 = threshold/2 bit[0] =1 bit[11] =1 bit[10:1] = 0 bit[0] =1 ERROR_RATEThreshold/dec none Source Select: Input at NICAM Path always NICAM; Mute in case of no NICAM available NICAM or FM/AM, depending on ERROR_RATE
1
700
2
set by customer; recommended range: 50...2000 none always FM/AM
3
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PRELIMINARY DATA SHEET
MSP 34x2G
6.3.2. A2 Threshold The threshold between Stereo/Bilingual and Mono Identification for the A2 Standard has been made programmable according to the user’s preferences. An internal hysteresis ensures robustness and stability. Table 6–7: Write Register on I2C Subaddress 10hex : A2 Threshold Register Address THRESHOLDS 00 22hex (write) A2 THRESHOLD Register Defines threshold of all A2 and EIA_J standards for Stereo and Bilingual detection bit[15:12] bit[11:0] 7F0hex ... 190hex ... 0A0hex must be set to 0 force Mono Identification default setting after reset minimum Threshold for stable detection A2_THRESH Function Name
recommended range : 0A0hex...3C0hex
6.3.3. Carrier-Mute Threshold The Carrier-Mute threshold has been made programmable according to the user’s preferences. An internal hysteresis ensures stable behavior. Table 6–8: Write Register on I2C Subaddress 10hex : Carrier-Mute Threshold Register Address THRESHOLDS 00 24hex (write) Carrier-Mute THRESHOLD Register Defines threshold for the carrier mute feature bit[15:12] bit[11:0] 000hex ... 02Ahex ... 7FFhex must be set to 0 Carrier-Mute always ON (both channels muted) default setting after reset Carrier-Mute always OFF (both channels forced on) CM_THRESH Function Name
recommended range : 14hex...50hex
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6.3.4. Register AD_CV The use of this register is no longer recommended. Use it only in cases where compatibility to the MSP 34x0D is required. Using the STANDARD SELECTION register together with the MODUS register provides a more economic way to program the MSP 34x2G. Table 6–9: AD_CV Register; reset status: all bits are “0”
AD_CV (00 BBhex) Bit [0] [1−6] Function not used Reference level in case of Automatic Gain Control = on (see Table 6–10). Constant gain factor when Automatic Gain Control = off (see Table 6–11). Determination of Automatic Gain or Constant Gain Selection of Sound IF source (identical to MODUS[8]) MSP-Carrier-Mute Feature 0 = constant gain 1 = automatic gain 0 = ANA_IN1+ 1 = ANA_IN2+ 0 = off: no mute 1 = on: mute as described in section 2.2.2. must be set to 0 Settings must be set to 0
PRELIMINARY DATA SHEET
Automatic setting by STANDARD SELECT Register 2-8, 0A-60hex 0 101000 9 0 100011
[7] [8] [9]
1 X 1
1 X 1
[10−15]
not used
0
0
X : not affected while choosing the TV sound standard by means of the STANDARD SELECT Register
Table 6–10: Reference Values for Active AGC (AD_CV[7] = 1)
Application Input Signal Contains AD_CV [6:1] Ref. Value AD_CV [6:1] in integer Range of Input Signal at pin ANA_IN1+ and ANA_IN2+
Terrestrial TV − FM Standards − NICAM/FM − NICAM/AM − NICAM only SAT ADR
1)
1 or 2 FM Carriers 1 FM and 1 NICAM Carrier 1 AM and 1 NICAM Carrier 1 NICAM Carrier only 1 or more FM Carriers FM and ADR carriers
101000 101000 100011 010100 100011
40 40 35 20 35
0.10 − 3 Vpp1) 0.10 − 3 Vpp1) 0.10 − 1.4 Vpp (recommended: 0.10 − 0.8 Vpp) 0.05 − 1.0 Vpp 0.10 − 3 Vpp1)
see DRP 3510A data sheet
For signals above 1.4 Vpp, the minimum gain of 3 dB is switched, and overflow of the A/D converter may result. Due to the robustness of the internal processing, the IC works up to and even more than 3 Vpp, if norm conditions of FM/NICAM or FM1/FM2 ratio are supposed. In this overflow case, a loss of FM-S/N ratio of about 10 dB may appear.
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PRELIMINARY DATA SHEET
MSP 34x2G
Table 6–11: AD_CV parameters for Constant Input Gain (AD_CV[7]=0)
Step 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20
1)
AD_CV [6:1] Constant Gain 000000 000001 000010 000011 000100 000101 000110 000111 001000 001001 001010 001011 001100 001101 001110 001111 010000 010001 010010 010011 010100
Gain 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
Input Level at pin ANA_IN1+ and ANA_IN2+ maximum input level: 3 Vpp (FM) or 1 Vpp (NICAM)1)
maximum input level: 0.14 Vpp
For signals above 1.4 Vpp, the minimum gain of 3 dB is switched and overflow of the A/D converter may result. Due to the robustness of the internal processing, the IC works up to and even more than 3 Vpp, if norm conditions of FM/NICAM or FM1/FM2 ratio are supposed. In this overflow case, a loss of FM-S/N ratio of about 10 dB may appear.
6.3.5. Register MODE_REG Note: The use of this register is no longer recommended. It should be used only in cases where software compatibility to the MSP 34x0D is required. Using the STANDARD SELECTION register together with the MODUS register provides a more economic way to program the MSP 34x2G. As soon as this register is applied, the MSP 34x2G works in the MSP 34x0D Compatibility Mode. In this mode: BTSC, EIA-J, and FM-Radio are disabled. Only MSP 34x0D features are available; the use of MODUS and STATUS register is not allowed. The MSP 34x2G is reset to the normal mode by first programming the MODUS register, followed by transmitting a valid standard code to the STANDARD SELECTION register. The register ‘MODE_REG’ contains the control bits determining the operation mode of the MSP 34x2G in the MSP 34x0D Compatibility Mode; Table 6–12 explains all bit positions.
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MSP 34x2G
PRELIMINARY DATA SHEET
Table 6–12: Control word ‘MODE_REG’; reset status: all bits are “0”
MODE_REG 00 83hex Bit [0] [1] [2] Function not used DCTR_TRI I2S_TRI Digital control out 0/1 tri-state I2S outputs tri-state (I2S_CL, I2S_WS, I2S_DA_OUT) Master/Slave mode of the I2S bus WS due to the Sony or Philips-Format Switch Audio_Clock_Output to tri-state Mode of MSP-Ch1 Comment Definition 0 : must be used 0 : active 1 : tri-state 0 : active 1 : tri-state 0 : Master 1 : Slave 0 : Sony 1 : Philips 0 : on 1 : tri-state 0 : FM 1 : Nicam 0 : must be used Mode of MSP-Ch2 High Deviation Mode (channel matrix must be sound A) 0 : FM 1 : AM 0 : normal 1 : high deviation mode 0 : must be used see also Table 6–14 see also Table 6–14 Mode of MSP-Ch1/ ADR-Interface Gain for AM Demodulation 0 : Gain = 6 dB 1 : Gain = 0 dB 0 : use FIR1 1 : use FIR2 0 : normal mode/tri-state 1 : ADR-mode/active 0 : 0 dB (default. of MSPB) 1 :12 dB (recommended) Automatic setting by STANDARD SELECT Register 2-5 0 X X 8, A, B 0 X X 9 0 X X
[3] [4] [5]
I2S Mode1) I2S_WS Mode Audio_CL_OUT
X X X
X X X
X X X
[6] [7] [8] [9]
NICAM1) not used FM AM HDEV
0 0 0 0
1 0 0 0
1 0 1 0
[11:10] [12] [13] [14] [15]
not used MSP-Ch1 Gain FIR1-Filter Coeff. Set ADR AM-Gain
0 0 1 0 1
0 0 0 0 1
0 0 0 0 1
X: not affected by short-programming
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PRELIMINARY DATA SHEET
MSP 34x2G
The loading sequences must be obeyed. To change a coefficient set, the complete block FIR1 or FIR2 must be transmitted. Note: For compatibility with MSP 3410B, IMREG1 and IMREG2 have to be transmitted. The value for IMREG1 and IMREG2 is 004. Due to the partitioning to 8-bit units, the values 04hex, 40hex, and 00hex arise. 6.3.7. DCO-Registers Note: The use of this register is no longer recommended. It should be used only in cases where software-compatibility to the MSP 34x0D is required. Using the STANDARD SELECTION register together with the MODUS register provides a more economic way to program the MSP 34x2G.
Value 04hex 40hex 00hex
Table 6–13: Loading sequence for FIR-coefficients
FIR1 00 01hex (MSP-Ch1: NICAM/FM2) No. 1 2 3 4 5 6 Symbol Name NICAM/FM2_Coeff. (5) NICAM/FM2_Coeff. (4) NICAM/FM2_Coeff. (3) NICAM/FM2_Coeff. (2) NICAM/FM2_Coeff. (1) NICAM/FM2_Coeff. (0) Bits 8 8 8 see Table 6–14 8 8 8 Value
FIR2 00 05hex (MSP-Ch2: FM1/AM) No. 1 2 3 4 5 6 7 8 9 Symbol Name IMREG1 IMREG1/ IMREG2 IMREG2 FM/AM_Coef (5) FM/AM_Coef (4) FM/AM_Coef (3) FM/AM_Coef (2) FM/AM_Coef (1) FM/AM_Coef (0) Bits 8 8 8 8 8 8 see Table 6–14 8 8 8
When selecting a TV-sound standard by means of the STANDARD SELECT register, all frequency tuning is performed automatically. If manual setting of the tuning frequency is required, a set of 24-bit registers determining the mixing frequencies of the quadrature mixers can be written manually into the IC. In Table 6–15, some examples of DCO registers are listed. It is necessary to divide them up into low part and high part. The formula for the calculation of the registers for any chosen IF frequency is as follows: INCRdec = int(f/fs ⋅ 224) with: int = integer function f = IF frequency in MHz fS = sampling frequency (18.432 MHz) Conversion of INCR into hex-format and separation of the 12-bit low and high parts lead to the required register values (DCO1_HI or _LO for MSP-Ch1, DCO2_HI or LO for MSP-Ch2).
6.3.6. FIR-Parameter, Registers FIR1 and FIR2 Note: The use of this register is no longer recommended. It should be used only in cases where software compatibility to the MSP 34x0D is required. Using the STANDARD SELECTION register together with the MODUS register provides a more economic way to program the MSP 34x2G. Data-shaping and/or FM/AM bandwidth limitation is performed by a pair of linear phase Finite Impulse Response filters (FIR-filter). The filter coefficients are programmable and are either configured automatically by the STANDARD SELECT register or written manually 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 6–14 several coefficient sets are proposed. To load the FIR-filters, the following data values are to be transferred 8 bits at a time embedded LSB-bound in a 16-bit word.
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MSP 34x2G
PRELIMINARY DATA SHEET
Table 6–14: 8-bit FIR-coefficients (decimal integer); reset status: all coefficients are “0”
Coefficients for FIR1 00 01hex and FIR2 00 05hex
Terrestrial TV Standards
FM - Satellite FIR filter corresponds to a band-pass with a bandwidth of B = 130 to 500 kHz
B fc frequency
B/G-, D/KNICAM-FM
INICAM-FM FIR1 2 4 −6 −4 40 94 0 FIR2 3 18 27 48 66 72
LNICAM-AM FIR1 −2 −8 −10 10 50 86 0 FIR2 −4 −12 −9 23 79 126
B/G-, D/K-, M-Dual FM FIR2 3 18 27 48 66 72 0
130 kHz FIR2 73 53 64 119 101 127 1
180 kHz FIR2 9 18 28 47 55 64 1
200 kHz FIR2 3 18 27 48 66 72 1
280 kHz FIR2 −8 −8 4 36 78 107 1
380 kHz FIR2 −1 −9 −16 5 65 123 1
500 kHz FIR2 −1 −1 −8 2 59 126 1
Autosearch FIR2 −1 −1 −8 2 59 126 0
Coef(i) 0 1 2 3 4 5 ModeREG[12] ModeREG[13]
FIR1 −2 −8 −10 10 50 86 0
FIR2 3 18 27 48 66 72
0
0
0
1
1
1
1
1
1
1
0
For compatibility, except for the FIR2-AM and the Autosearch-sets, the FIR-filter programming as used for the MSP 3410B is also possible. ADR coefficients are listed in the DRP data sheet.
Table 6–15: DCO registers for the MSP 34x2G; reset status: DCO_HI/LO = “0000”
DCO1_LO 00 93hex, DCO1_HI 00 9Bhex; DCO2_LO 00 A3hex, DCO2_HI 00 ABhex Freq. MHz 4.5 5.04 5.5 5.58 5.7421875 6.0 6.2 6.5 6.552 7.02 7.38 DCO_HI/hex 03E8 0460 04C6 04D8 04FC 0535 0561 05A4 05B0 0618 0668 DCO_LO/hex 000 0000 038E 0000 00AA 0555 0C71 071C 0000 0000 0000 5.76 5.85 5.94 6.6 6.65 6.8 7.2 7.56 0500 0514 0528 05BA 05C5 05E7 0640 0690 0000 0000 0000 0AAA 0C71 01C7 0000 0000 Freq. MHz DCO_HI/hex DCO_LO/hex
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PRELIMINARY DATA SHEET
MSP 34x2G
Table 6–16: NICAM operation modes as defined by the EBU NICAM 728 specification
C4 0 0 0 C3 0 0 0 0 0 0 0 C2 0 0 1 1 0 0 1 C1 0 1 0 1 0 1 0 Operation Mode Stereo sound (NICAMA/B), independent mono sound (FM1) Two independent mono signals (NICAMA, FM1) Three independent mono channels (NICAMA, NICAMB, FM1) Data transmission only; no audio Stereo sound (NICAMA/B), FM1 carries same channel One mono signal (NICAMA). FM1 carries same channel as NICAMA Two independent mono channels (NICAMA, NICAMB). FM1 carries same channel as NICAMA Data transmission only; no audio Unimplemented sound coding option (not yet defined by EBU NICAM 728 specification)
6.4. Manual/Compatibility Mode: Description of Demodulator Read Registers Note: The use of these register is no longer recommended. It should be used only in cases where software compatibility to the MSP 34x0D is required. Using the STANDARD SELECTION register together with the STATUS register provides a more economic way to program the MSP 34x2G and to retrieve information from the IC. All registers except C_AD_BITs are 8 bits wide. They can be read out of the RAM of the MSP 34x2G if the MSP 34x0D Compatibility Mode is required. All transmissions take place in 16-bit words. The valid 8-bit data are the 8 LSBs of the received data word. If the Automatic Sound Select feature is not used, the NICAM or FM-identification parameters must be read and evaluated by the controller in order to enable appropriate switching of the channel select matrix of the baseband processing part. The FM-identification registers are described in section 6.6.1. To handle the NICAM-sound and to observe the NICAM-quality, at least the registers C_AD_BITS and ERROR_RATE must be read and evaluated by the controller. Additional data bits and CIB bits, if supplied by the NICAM transmitter, can be obtained by reading the registers ADD_BITS and CIB_BITS.
0 1 1 1
1 x
0 1
1 x
1 x
AUTO_FM: monitor bit for the AUTO_FM Status: 0: NICAM source is NICAM 1: NICAM source is FM
6.4.1. NICAM Mode Control/Additional Data Bits Register NICAM operation mode control bits and A[2:0] of the additional data bits. Format:
MSB
11 Auto _FM ... ... 7 A[2]
Note: It is no longer necessary to read out and evaluate the C_AD_BITS. All evaluation is performed in the MSP and indicated in the STATUS register.
6.4.2. Additional Data Bits Register Contains the remaining 8 of the 11 additional data bits. The additional data bits are not yet defined by the NICAM 728 system. Format:
MSB
7 6 A[9] 5 A[8]
C_AD_BITS 00 23hex
6 A[1] 5 A[0] 4 C4 3 C3 2 C2 1 C1
LSB
0 S
ADD_BITS 00 38hex
4 A[7] 3 A[6] 2 A[5] 1 A[4]
LSB
0 A[3]
Important: “S” = Bit[0] indicates correct NICAM-synchronization (S = 1). If S = 0, the MSP 3411/3451G has not yet synchronized correctly to frame and sequence, or has lost synchronization. The remaining read registers are therefore not valid. The MSP mutes the NICAM output automatically and tries to synchronize again as long as MODE_REG[6] is set. The operation mode is coded by C4-C1 as shown in Table 6–16.
A[10]
6.4.3. CIB Bits Register CIB bits 1 and 2 (see NICAM 728 specifications). Format:
MSB
7 x 6 x 5 x
CIB_BITS 00 3Ehex
4 x 3 x 2 x 1 CIB1
LSB
0 CIB2
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MSP 34x2G
6.4.4. NICAM Error Rate Register
ERROR_RATE Error free maximum error rate 00 57hex 0000hex 07FFhex
PRELIMINARY DATA SHEET
6.4.7. Automatic Search Function for FM-Carrier Detection in Satellite Mode The AM demodulation ability of the MSP 34x2G offers the possibility to calculate the “field strength” of the momentarily selected FM carrier, which can be read out by the controller. In SAT receivers, this feature can be used to make automatic FM carrier search possible. For this, the MSP has to be switched to AM-mode (MODE_REG[8]), FM-Prescale must be set to 7Fhex = +127dec, and the FM DC notch (see section 6.5.7.) must be switched off. The sound-IF frequency range must now be “scanned” in the MSP-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 controller. This results in either continuing search or switching the MSP back to FM demodulation mode. During the search process, the FIR2 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 (FIR1/2) and the deemphasis (50 µs or adaptive) can be switched accordingly. Due to the fact that a constant demodulation frequency offset of a few kHz leads to a DC level in the demodulated signal, 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 Windows software.
Average error rate of the NICAM reception in a time interval of 182 ms, which should be close to 0. The initial and maximum value of ERROR_RATE is 2047. This value is also active if the NICAM bit of MODE_REG is not set. Since the value is achieved by filtering, a certain transition time (approx. 0.5 sec) is unavoidable. Acceptable audio may have error rates up to a value of 700 int. Individual evaluation of this value by the controller and an appropriate threshold may define the fallback mode from NICAM to FM/ AM-Mono in case of poor NICAM reception. The bit error rate per second (BER) can be calculated by means of the following formula: BER = ERROR_RATE * 12.3*10−6 /s
6.4.5. PLL_CAPS Readback Register It is possible to read out the actual setting of the PLL_CAPS. In standard applications, this register is not of interest for the customer.
PLL_CAPS minimum frequency nominal frequency maximum frequency PLL_CAPS PLL open PLL closed 02 1Fhex L 1111 1111 0101 0110 RESET 0000 0000 02 1Fhex H xxxx xxx0 xxxx xxx1 FFhex 56hex 00hex
6.4.6. AGC_GAIN Readback Register It is possible to read out the actual setting of AGC_GAIN in Automatic Gain Mode. In standard applications, this register is not of interest for the customer.
AGC_GAIN max. amplification (20 dB) min. amplification (3 dB) 02 1Ehex 0001 0100 0000 0000 14hex 00hex
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Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
6.5.2. Volume Modes of SCART1/2 Outputs
Volume Mode SCART1 Volume Mode SCART2 00 07hex 00 40hex 0000 RESET 0001 [3:0] [3:0] 0hex 1hex
6.5. Manual/Compatibility Mode: Description of DSP Write Registers 6.5.1. Additional Channel Matrix Modes
Loudspeaker Matrix Headphone Matrix SCART1 Matrix SCART2 Matrix I2S Matrix Quasi-Peak Detector Matrix SUM/DIFF AB_XCHANGE PHASE_CHANGE_B PHASE_CHANGE_A A_ONLY B_ONLY 00 08hex 00 09hex 00 0Ahex 00 41hex 00 0Bhex 00 0Chex 0100 0000 0101 0000 0110 0000 0111 0000 1000 0000 1001 0000 L L L L
linear logarithmic
Linear Mode L Volume SCART1 L Volume SCART2 40hex 50hex 60hex 70hex 80hex 90hex OFF 0 dB gain (digital full scale (FS) to 2 VRMS output) +6 dB gain (−6 dBFS to 2 VRMS output) 00 40hex 0000 0000 RESET 0100 0000 H 00hex 40hex 00 07hex H
0111 1111
7Fhex
6.5.3. FM Fixed Deemphasis This table shows additional modes for the channel matrix registers. 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.
FM Deemphasis 50 µs 75 µs OFF 00 0Fhex 0000 0000 RESET 0000 0001 0011 1111 H 00hex 01hex 3Fhex
Note: This register is initialised during STANDARD SELECTION and is automatically updated when Automatic Sound Select (MODUS[0]=1) is on. 6.5.4. FM Adaptive Deemphasis
FM Adaptive Deemphasis WP1 OFF WP1 00 0Fhex 0000 0000 RESET 0011 1111 L 00hex 3Fhex
Note: This register is initialised during STANDARD SELECTION and is automatically updated when Automatic Sound Select (MODUS[0]=1) is on. 6.5.5. NICAM Deemphasis A J17 Deemphasis is always applied to the NICAM signal. It is not switchable.
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MSP 34x2G
6.5.6. Identification Mode for A2 Stereo Systems
Identification Mode Standard B/G (German Stereo) Standard M (Korean Stereo) Reset of Ident-Filter 00 15hex 0000 0000 RESET 0000 0001 0011 1111 L 00hex 01hex
PRELIMINARY DATA SHEET
6.6. Manual/Compatibility Mode: Description of DSP Read Registers All readable registers are 16-bit wide. Transmissions via I2C bus have to take place in 16-bit words. Some of the defined 16-bit words are divided into low and high byte, thus holding two different control entities. These registers are not writable.
3Fhex
To shorten the response time of the identification algorithm after a program change between two FM-Stereo capable programs, the reset of the ident-filter can be applied. Sequence: 1. Program change 2. Reset ident-filter 3. Set identification mode back to standard B/G or M 4. Wait approx. 500 ms 5. Read stereo detection register Note: This register is initialized during STANDARD SELECTION and is automatically updated when Automatic Sound Select (MODUS[0]=1) is on.
6.6.1. Stereo Detection Register for A2 Stereo Systems
Stereo Detection Register Stereo Mode MONO STEREO BILINGUAL 00 18hex H
Reading (two’s complement) near zero positive value (ideal reception: 7Fhex) negative value (ideal reception: 80hex)
Note: It is no longer necessary to read out and evaluate the A2 identification level. All evaluation is performed in the MSP and indicated in the STATUS register.
6.5.7. FM DC Notch 6.6.2. DC Level Register 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 Section 6.4.7.). In normal FM-mode, the FM DC Notch should be switched on.
FM DC Notch ON OFF 00 17hex 0000 0000 Reset 0011 1111 L 00hex 3Fhex DC Level Readout FM1 (MSP-Ch2) DC Level Readout FM2 (MSP-Ch1) DC Level 00 1Bhex 00 1Chex H+L H+L
[8000hex ... 7FFFhex] values are 16 bit two’s complement
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. A too low demodulation frequency (DCO) results in a positive DC-level and vice versa. 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.
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PRELIMINARY DATA SHEET
MSP 34x2G
6.8. Exclusions of Audio Baseband Features In general, all functions can be switched independently. Two exceptions exist: 1. NICAM cannot be processed simultaneously with the FM2 channel. 2. FM adaptive deemphasis cannot be processed simultaneously with FM-identification.
6.7. Demodulator Source Channels in Manual Mode 6.7.1. Terrestric Sound Standards Table 6–17 shows the source channel assignment of the demodulated signals in case of manual mode for all terrestric sound standards. See Table 2–2 for the assignment in the Automatic Sound Select mode. In manual mode for terrestric sound standards, only two demodulator sources are defined.
6.9. Compatibility Restrictions to MSP 34x0D 6.7.2. SAT Sound Standards Table 6–18 shows the source channel assignment of the demodulated signals for SAT sound standards. The MSP 34x2G is fully hardware compatible to the MSP 34x0D. However, to substitute a MSP 34x0D by the corresponding MSP 34x2G, the controller software has to be adapted slightly: 1. The register FM-Matrix (00 0Ehex low part) must be changed from “no matrix (00hex)” to “sound A mono (03hex)” during mono transmission of all TV-sound standards (see also Table 6–17). 2. With the MSP 34x2G, the STANDARD SELECTION initializes the FM-deemphasis, which is not the case for the MSP 34x0D. So, if STANDARD SELECTION is applied, this I2C instruction can be omitted.
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MSP 34x2G
Table 6–17: Manual Sound Select Mode for Terrestric Sound Standards
PRELIMINARY DATA SHEET
Source Channels of Sound Select Block Broadcasted Sound Standard B/G-FM D/K-FM M-Korea M-Japan Selected MSP Standard Code 03 04, 05 02 30 Broadcasted Sound Mode MONO STEREO BILINGUAL, Languages A and B B/G-NICAM L-NICAM I-NICAM D/K-NICAM D/K-NICAM
(with high deviation FM)
FM Matrix
FM/AM
(use 0 for channel select)
Stereo or A/B
(use 1 for channel select)
Sound A Mono German Stereo Korean Stereo No Matrix Sound A Mono1)
Mono Stereo Left = A Right = B analog Mono
Mono Stereo Left = A Right = B no sound with AUTO_FM: analog Mono
08 09 0A 0B 0C
NICAM not available or NICAM error rate too high MONO STEREO BILINGUAL, Languages A and B MONO
Sound A Mono1) Sound A Mono1) Sound A Mono1) Sound A Mono Korean Stereo Sound A Mono Korean Stereo Sound A Mono
analog Mono analog Mono analog Mono Mono Stereo Mono Stereo Mono
NICAM Mono NICAM Stereo Left = NICAM A Right = NICAM B Mono Stereo Mono Stereo Mono
20
STEREO MONO + SAP
BTSC
STEREO + SAP MONO 21 STEREO MONO + SAP STEREO + SAP
No Matrix Sound A Mono Korean Stereo
Left = Mono Right = SAP Mono Stereo
Left = Mono Right = SAP Mono Stereo
FM-Radio
1)
40
MONO STEREO
Automatic refresh to Sound A Mono, do not write any other value to the register FM Matrix!
Table 6–18: Manual Sound Select Modes for SAT-Modes
Source Channels of Sound Select Block for SAT-Modes Broadcasted Sound Standard Selected MSP Standard Code 6, 50hex FM SAT 51hex Broadcasted Sound Mode FM Matrix FM/AM
(source select: 0)
Stereo or A/B
(source select: 1)
Stereo or A
(source select: 3)
Stereo or B
(source select: 4)
MONO STEREO BILINGUAL
Sound A Mono No Matrix No Matrix
Mono Stereo Left = A (FM1) Right = B (FM2)
Mono Stereo Left = A (FM1) Right = B (FM2)
Mono Stereo A (FM1)
Mono Stereo B (FM2)
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Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
7. Appendix D: Application Information 7.1. Phase Relationship of Analog Outputs The analog output signals: Loudspeaker, headphone, and SCART2 all have the same phases. The user does not need to correct output phases when using these analog outputs directly. The SCART1 output has opposite phase. Using the I2S-outputs for other DSPs or D/A converters, care must be taken to adjust for the correct phase. If the attached coprocessor is one of the MSP family, the following schematics help to determine the phase relationship.
I2S_IN1/2 I2S_OUT1/2
Loudspeaker Center/Surround Headphone SCART1-Ch. SCART1
Audio Baseband Processing SCART1 SCART2 SCART3 SCART4 MONO SCART DSP Input Select
SCART2-Ch.
SCART2
MONO, SCART1...4
SCART Output Select
Fig. 7–1: Phase diagram of the MSP 34x2G
Micronas
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MSP 34x2G
7.2. Application Circuit Tuner 2
IF 2 IN
if ANA_IN2+ not used
PRELIMINARY DATA SHEET
Signal GND IF 1 IN
10 µF + 3.3 µF 56 pF 56 pF 56 pF + 100 nF 100 nF 18.432 MHz
C s. section 4.6.2.
8 V(5 V)
100 pF
56 pF
Tuner 1
ANA_IN1+
+ 10 µF + 10 µF 1 KΩ
Alternative circuit for ANA_IN1+for more attenuation of video components:
AGNDC (42) 45
XTAL_OUT (63) 72
VREFTOP (54) 58
XTAL_IN (62) 71
ANA_IN1+ (58) 67
ANA_IN− (59) 68
ANA_IN2+ (60) 69
CAPL_M (40) 40
CAPL_A (38) 38
1 µF 1 µF
DACM_L (29) 28 1 nF DACM_R (28) 27
330 nF
60 (55) MONO_IN
Loudspeaker
1 nF
330 nF 330 nF AHVSS 330 nF 330 nF AHVSS 330 nF 330 nF AHVSS 330 nF
56 (52) SC1_IN_L 57 (53) SC1_IN_R 55 (51) ASG 53 (49) SC2_IN_L 54 (50) SC2_IN_R 52 (48) ASG 50 (46) SC3_IN_L 51 (47) SC3_IN_R 49 (45) ASG 47 (43) SC4_IN_L 48 (44) SC4_IN_R DACA_R (25) 24 DACA_L (26) 25 DACM_S (32) 31 DACM_C (30) 29 DACM_SUB (31) 30
1 µF 1 µF
1 nF
Center
1 nF
1 µF
Surround
1 µF 1 µF
1 nF
1 nF
Headphone
5V 5V
DVSS
330 nF
MSP 34x2G
SC1_OUT_L (37) 37
1 nF
80 (7) STANDBYQ 79 (6) ADR_SEL
100 Ω 22 µF
+
SC1_OUT_R (36) 36
100 Ω 22 µF
+
DVSS 3 (10) I2C_DA 2 (9) I2C_CL 9 (16) ADR_WS 10 (17) ADR_CL 8 (15) ADR_DA 5 (12) I2S_WS 4 (11) I2S_CL 7 (14) I2S_DA_IN1 17 (20) I2S_DA_IN2 6 (13) I2S_DA_OUT 21 (24) RESETQ 39 (39) AHVSUP 13 (18) DVSUP 66 (57) AVSUP 16 (19) DVSS 62 (56) AVSS TESTEN (61) 70 43 (41) AHVSS 35 (35) VREF1 26 (27) VREF2 AUD_CL_OUT (1) 74 D_CTR_I/O_0 (5) 78 D_CTR_I/O_1 (4) 77 SC2_OUT_L (34) 34
100 Ω 22 µF
+
SC2_OUT_R (33) 33
100 Ω 22 µF
+
AHVSS
RESETQ (from Controller, see section 4.6.3.3.)
220 pF 470 pF 1.5 nF 10 µF
470 pF 1.5 nF 10 µF
470 pF 1.5 nF 10 µF
Note:
Decoupling capacitors from − DVSUP to DVSS, − AVSUP to AVSS, and − AHVSUP to AHVSS are recommended as closely as possible to supply pins (see note on page 56).
AHVSS
AHVSS
5V
5V
8V ( 5 V)
Note: Pin numbers refer to the PQFP80 package, numbers in brackets refer to the PSDIP64 package.
102
AHVSS
AVSS
Micronas
PRELIMINARY DATA SHEET
MSP 34x2G
Micronas
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MSP 34x2G
8. Appendix E: MSP 34x2G Version History 9. Data Sheet History
PRELIMINARY DATA SHEET
MSP 3452G-A1 First release
1. Preliminary data sheet: “MSP 34x2G Multistandard Sound Processor Family with Dolby Surround Pro Logic”, May 22, 2000, 6251-520-1PD. First release of the preliminary data sheet.
Micronas 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@micronas.com Internet: www.micronas.com Printed in Germany Order No. 6251-520-1PD
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 are exclusively subject to our respective order confirmation form; the same applies to orders based on development samples delivered. By this publication, Micronas GmbH does not assume responsibility for patent infringements or other rights of third parties which may result from its use. Further, Micronas GmbH reserves the right to revise this publication and to make changes to its content, at any time, without obligation to notify any person or entity of such revisions or changes. No part of this publication may be reproduced, photocopied, stored on a retrieval system, or transmitted without the express written consent of Micronas GmbH.
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