Datasheet TDA9875A Datasheet (Philips)

Page 1
INTEGRATED CIRCUITS
DATA SH EET
TDA9875A
Digital TV Sound Processor (DTVSP)
Product specification Supersedes data of 1998 Aug 13 File under Integrated Circuits, IC02
1999 Dec 20
Page 2
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
CONTENTS
1 FEATURES
1.1 Demodulator and decoder section
1.2 DSP section
1.3 Analog audio section 2 GENERAL DESCRIPTION
2.1 Supported standards 3 ORDERING INFORMATION 4 BLOCK DIAGRAM 5 PINNING 6 FUNCTIONAL DESCRIPTION
6.1 Demodulator and decoder section
6.2 Digital signal processing
6.3 Analog audio section 7 LIMITING VALUES 8 THERMAL CHARACTERISTICS 9 CHARACTERISTICS
10 I2C-BUS CONTROL
10.1 Introduction
10.2 Power-up state
10.3 Slave receiver mode
10.4 Slave transmitter mode
10.5 Expert mode 11 I2S-BUS DESCRIPTION 12 APPLICATION INFORMATION 13 PACKAGE OUTLINES 14 SOLDERING
14.1 Introduction
14.2 Through-hole mount packages
14.3 Surface mount packages
14.4 Suitability of IC packages for wave,reflow and dipping soldering methods
15 DEFINITIONS 16 LIFE SUPPORT APPLICATIONS 17 PURCHASE OF PHILIPS I2C COMPONENTS
1999 Dec 20 2
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

1 FEATURES

1.1 Demodulator and decoder section

Sound IF (SIF) input switch e.g. to select between terrestrial TV SIF and SAT SIF sources
SIF AGC with 24 dB control range
SIF 8-bit Analog-to-Digital Converter (ADC)
Differential Quadrature Phase Shift Keying (DQPSK)
demodulation for different standards, simultaneously with 1-channel FM demodulation
Near Instantaneous Companded Audio Multiplex (NICAM) decoding (B/G, I and L standard)
Two-carrier multistandard FM demodulation (B/G, D/K and M standard)
Decoding for three analog multi-channel systems (A2, A2+ and A2*) and satellite sound
OptionalAM demodulation for system L, simultaneously with NICAM
Programmable identification (B/G, D/K and M standard) and different identification times.

1.2 DSP section

Digital crossbar switch for all digital signal sources and destinations
Control of volume, balance, contour, bass, treble, pseudo stereo, spatial, bass boost and soft mute
Plop-free volume control
Automatic Volume Level (AVL) control
Adaptive de-emphasis for satellite
Programmable beeper
Monitor selection for FM/AM DC values and signals,
with peak detection option
I2S-bus interface for a feature extension (e.g. Dolby Pro Logic) with matrix, level adjust and mute.
DualaudioDigital-to-AnalogConverter(DAC)fromDSP to analog crossbar switch, bandwidth 15 kHz
Dual audio ADC from analog inputs to DSP
Two dual audio DACs for loudspeaker (Main) and
headphone (Auxiliary) outputs; also applicable for L, R, C and S in the Dolby Pro Logic mode with feature extension.

2 GENERAL DESCRIPTION

The TDA9875A is a single-chip Digital TV Sound Processor (DTVSP) for analog and digital multi-channel sound systems in TV sets and satellite receivers.

2.1 Supported standards

The multistandard/multi-stereo capability of the TDA9875A is mainly of interest in Europe, but also in Hong Kong/Peoples Republic of China and South East Asia. This includes B/G, D/K, I, M and L standards. In other application areas there exists only subsets of these standard combinations otherwise only single standards are transmitted.
M standard is transmitted in Europe by the American Forces Network (AFN) with European channel spacing (7 MHz VHF and 8 MHz UHF) and monaural sound.
The AM sound of L/L accent standard is normally demodulated in the first sound IF. The resulting AF signal has to be entered into the mono audio input of the TDA9875A. A second possibility is to use the internal AM demodulator stage, however this gives limited performance.

1.3 Analog audio section

Analog crossbar switch with inputs for mono and stereo (also applicable as SCART 3 input), SCART 1 input/output, SCART 2 input/output and line output
User defined full-level/3 dB scaling for SCART outputs
Output selection of mono, stereo, dual A/B, dual A or
dual B
20 kHz bandwidth for SCART-to-SCART copies
Standby mode with function for SCART copies
1999 Dec 20 3
Korea has a stereo sound system similar to Europe and is supported by the TDA9875A. The differences include deviation, modulation contents and identification. It is based on M standard.
An overview of the supported standards and sound systems and their key parameters is given in Table 1.
The analog multi-channel sound systems (A2, A2+ and A2*) are 2-Carrier Systems (2CS).
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
2.1.1 ANALOG 2-CARRIER SYSTEMS
Table 1 Frequency modulation
STANDARD
SOUND
SYSTEM
CARRIER
FREQUENCY
(MHz)
FM DEVIATION (kHz) MODULATION BANDWIDTH/
DE-EMPHASIS
NOM. MAX. OVER SC1 SC2
(kHz/µs)
M mono 4.5 15 25 50 mono 15/75 M A2+ 4.5/4.724 15 25 50 B/G A2 5.5/5.742 27 50 80
1
⁄2(L+R)1⁄2(L − R) 15/75 (Korea)
1
⁄2(L + R) R 15/50
I mono 6.0 27 50 80 mono 15/50 D/K A2 6.5/6.742 27 50 801⁄2(L + R) R 15/50 D/K A2* 6.5/6.258 27 50 80
1
⁄2(L + R) R 15/50
Table 2 Identification for A2 systems
PARAMETER A2/A2* A2+ (KOREA)
Pilot frequency 54.6875 kHz = 3.5 × line frequency 55.0699 kHz = 3.5 × line frequency Stereo identification frequency
Dual identification frequency
117.5 Hz
274.1 Hz
line frequency
= 149.9 Hz
------------------------------------­133
line frequency
= 276.0 Hz
-------------------------------------
57
line frequency
=
------------------------------------­105
line frequency
=
-------------------------------------
57
AM modulation depth 50% 50%
2.1.2 2-CARRIER SYSTEMS WITH NICAM
Table 3 NICAM
SC1
STANDARD
FREQUENCY
(MHz)
TYPE
MODULATION
INDEX
(%)
DEVIATION
(kHz)
SC2
NICAM
(MHz)
DE-
EMPHASIS
ROLL-
OFF (%)
NICAM
CODING
NOM. MAX. NOM. MAX.
B/G 5.5 FM −−27 50 5.85 J17 40 note 1 I 6.0 FM −−27 50 6.552 J17 100 note 1 D/K 6.5 FM −−27 50 5.85 J17 40 note 2 L 6.5 AM 54 100 −− 5.85 J17 40 note 1
Notes
1. See
“EBU specification”
or equivalent specification.
2. Not yet defined.
1999 Dec 20 4
Page 5
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
2.1.3 SATELLITE SYSTEMS An important specification for satellite TV reception is the
‘Astra specification’
. The TDA9875A is suited for the reception
of Astra and other satellite signals.
Table 4 FM satellite sound
CARRIER
CARRIER TYPE
FREQUENCY
(MHz)
Main 6.50
(1)
MODULATION
INDEX
0.26 85 mono 15/50
Sub 7.02/7.20 0.15 50 m/st/d
MAXIMUM
FM DEVIATION
(kHz)
MODULATION
(3)
BANDWIDTH/
DE-EMPHASIS
(kHz/µs)
15/adaptive Sub 7.38/7.56 Sub 7.74/7.92 Sub 8.10/8.28
Notes
1. For other satellite systems, frequencies of e.g. 5.80, 6.60 or 6.65 MHz can also be received.
2. A de-emphasis of 60 µs, or in accordance with J17, is available.
3. m/st/d = mono, stereo or dual language sound.
4. Adaptive de-emphasis is compatible to transmitter specification.

3 ORDERING INFORMATION

PACKAGE
TYPE NUMBER
NAME DESCRIPTION VERSION
TDA9875A SDIP64 plastic shrink dual in-line package; 64 leads (750 mil) SOT274-1 TDA9875AH QFP64 plastic quad flat package; 64 leads (lead length 1.6 mm);
SOT393-1
body 14 × 14 × 2.7 mm
(2)
(4)
1999 Dec 20 5
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

4 BLOCK DIAGRAM

handbook, full pagewidth
ADDR1 ADDR2
SCL SDA
XTALI
XTALO
SYSCLK
SDI1
SDI2 SDO1 SDO2
SCK
WS
SIF2 SIF1
9 (1)
P1
20 (12)
P2
3 (59) 13 (5) 4 (60) 5 (61)
18 (10) 19 (11) 21 (13)
27 (19) 26 (18) 25 (17) 24 (16) 22 (14) 23 (15)
I2C-BUS
INTERFACE
IDENTIFICATION
CLOCK
PEAK
DETECTION
I2S-BUS
INTERFACE
10 (2) 12 (4)
INPUT SWITCH
AGC, ADC
FM (AM)
DEMODULATION
A2/SATELLITE
DECODER
LEVEL
ADJUST
DIGITAL SELECT
NICAM
DEMODULATION
NICAM
DECODER
LEVEL
ADJUST
ADC (2)
SUPPLY
SOUND IF
(SIF)
ANALOG
CROSSBAR
SWITCH
(63) 7 (62) 6 (3) 11 (64) 8
(58) 2 (57) 1
(25) 33 (26) 34
(28) 36 (29) 37
(23) 31 (24) 32
(21) 29
(39) 47 (40) 48
(43) 51 (44) 52 (55) 63 (54) 62
(33) 41 (34) 42 (36) 44 (37) 45
V
DEC1
V
SSA1
V
ref1
I
ref
NICAM PCLK
SCIR1 SCIL1 SCIR2 SCIL2 EXTIR EXTIL MONOIN
SCOR1 SCOL1 SCOR2 SCOL2 LOR LOL
i.c. i.c. i.c. i.c.
V
DDD1
V
DDD2
V
SSD1
V
SSD2
V
SSD3
V
SSD4
CRESET
TEST1 TEST2
15 (7) 64 (56) 14 (6) 49 (41) 35 (27) 17 (9) 16 (8)
28 (20) 30 (22)
DIGITAL SUPPLY
TDA9875A
(
TDA9875AH
TEST
AUDIO PROCESSING
)
DAC (2)
(52)
(53)
60
61
MOR
MOL
The pin numbers given in parenthesis refer to the TDA9875AH version.
Fig.1 Block diagram.
1999 Dec 20 6
AUXOL
DAC (2)
DAC (2)
(50) 58
(49) 57
AUXOR
SUPPLY
SCART,
DAC, ADC
(46) 54
(47) 55
(51) 59 (30) 38
(31) 39 (32) 40
(38) 46 (45) 53
(35) 43 (48) 56 (42) 50
MHB598
PCAPR PCAPL
V
DDA
V
DEC2
V
ref(p)
V
ref(n)
V
ref2
V
ref3
V
SSA2
V
SSA3
V
SSA4
Page 7
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

5 PINNING

SYMBOL
PIN
TDA9875A TDA9875AH
PIN
TYPE
(1)
DESCRIPTION
PCLK 1 57 O NICAM clock output at 728 kHz NICAM 2 58 O serial NICAM data output at 728 kHz ADDR1 3 59 I I SCL 4 60 I I SDA 5 61 I/O I V V I
ref
SSA1 DEC1
6 62 S supply ground 1; analog front-end circuitry 763supply voltage decoupling 1; analog front-end circuitry 864resistor for reference current generator; analog front-end circuitry
2
C-bus slave address input 1
2
C-bus clock input
2
C-bus data input/output
P1 9 1 I/O general purpose input/output pin 1 SIF2 10 2 I sound IF input 2 V
ref1
11 3 reference voltage 1; analog front-end circuitry
SIF1 12 4 I sound IF input 1 ADDR2 13 5 I I V V
SSD1 DDD1
14 6 S supply ground 1; digital circuitry
15 7 S digital supply voltage 1; digital circuitry
2
C-bus slave address input 2
CRESET 16 8 capacitor for Power-on reset V
SSD4
17 9 S supply ground 4; digital circuitry
XTALI 18 10 I crystal oscillator input XTALO 19 11 O crystal oscillator output P2 20 12 I/O general purpose input/output pin 2 SYSCLK 21 13 O system clock output SCK 22 14 I/O I WS 23 15 I/O I SDO2 24 16 O I SDO1 25 17 O I SDI2 26 18 I I SDI1 27 19 I I TEST1 28 20 I test pin 1; connected to V
2
S-bus clock input/output
2
S-bus word select input/output
2
S-bus data output 2 (I2S2 output)
2
S-bus data output 1 (I2S1 output)
2
S-bus data input 2 (I2S2 input)
2
S-bus data input 1 (I2S1 input)
for normal operating mode
SSD1
MONOIN 29 21 I audio mono input TEST2 30 22 I test pin 2; connected to V
for normal operating mode
SSD1
EXTIR 31 23 I external audio input right channel EXTIL 32 24 I external audio input left channel SCIR1 33 25 I SCART 1 input right channel SCIL1 34 26 I SCART 1 input left channel V
SSD3
35 27 S supply ground 3; digital circuitry
SCIR2 36 28 I SCART 2 input right channel SCIL2 37 29 I SCART 2 input left channel V
DEC2
38 30 supply voltage decoupling 2; audio analog-to-digital converter
circuitry
1999 Dec 20 7
Page 8
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
SYMBOL
V
ref(p)
PIN
TDA9875A TDA9875AH
39 31 positive reference voltage; audio analog-to-digital converter
PIN
TYPE
(1)
DESCRIPTION
circuitry
V
ref(n)
40 32 reference voltage ground; audio analog-to-digital converter
circuitry i.c. 41 33 internally connected; note 2 i.c. 42 34 internally connected; note 3 V
SSA2
43 35 S supply ground 2; audio analog-to-digital converter circuitry i.c. 44 36 internally connected; note 3 i.c. 45 37 internally connected; note 2 V
ref2
46 38 reference voltage 2; audio analog-to-digital converter circuitry SCOR1 47 39 O SCART 1 output right channel SCOL1 48 40 O SCART 1 output left channel V V
SSD2 SSA4
49 41 S supply ground 2; digital circuitry
50 42 S supply ground 4; audio operational amplifier circuitry SCOR2 51 43 O SCART 2 output right channel SCOL2 52 44 O SCART 2 output left channel V
ref3
53 45 reference voltage 3; audio digital-to-analog converter and
operational amplifier circuitry
PCAPR 54 46 post-filter capacitor pin right channel; audio digital-to-analog
converter
PCAPL 55 47 post-filter capacitor pin left channel; audio digital-to-analog
converter
V
SSA3
56 48 S supply ground 3; audio digital-to-analog converter circuitry AUXOR 57 49 O headphone (Auxiliary) output right channel AUXOL 58 50 O headphone (Auxiliary) output left channel V
DDA
59 51 S analog supply voltage; analog circuitry MOR 60 52 O loudspeaker (Main) output right channel MOL 61 53 O loudspeaker (Main) output left channel LOL 62 54 O line output left channel LOR 63 55 O line output right channel V
DDD2
64 56 S digital supply voltage 2; digital circuitry
Notes
1. Pin type: I = input, O = output, S = supply.
2. Test pin: CMOS level input; pull-up resistor; can be connected to VSS.
3. Test pin: CMOS 3-state stage; can be connected to VSS.
1999 Dec 20 8
Page 9
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
handbook, halfpage
PCLK
NICAM
ADDR1
SCL
SDA
V
SSA1
V
DEC1
I
ref P1
SIF2
V
ref1
SIF1
ADDR2
V
SSD1
V
DDD1
CRESET
V
SSD4
XTALI
XTALO
P2
SYSCLK
SCK
WS SDO2 SDO1
SDI2 SDI1
TEST1
MONOIN
TEST2
EXTIR
EXTIL
1 2 3 4 5 6 7 8 9
10
11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32
TDA9875A
MHB071
64 63 62 61 60 59 58 57 56 55 54 53 52 51 50 49 48 47 46 45 44 43 42 41 40 39 38 37 36 35 34 33
V
DDD2
LOR LOL MOL MOR V
DDA
AUXOL AUXOR V
SSA3
PCAPL PCAPR V
ref3
SCOL2 SCOR2 V
SSA4
V
SSD2
SCOL1 SCOR1 V
ref2
i.c. i.c. V
SSA2
i.c. i.c. V
ref(n)
V
ref(p)
V
DEC2
SCIL2 SCIR2 V
SSD3
SCIL1 SCIR1
Fig.2 Pin configuration (TDA9875A).
1999 Dec 20 9
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
handbook, full pagewidth
ref
I
64
DEC1VSSA1
V
63
62
SDA 61
SCL 60
ADDR1
NICAM
59
58
PCLK 57
DDD2
V
56
LOR 55
LOL 54
MOL 53
MOR 52
DDA
V
51
AUXOL 50
AUXOR 49
P1
1
SIF2
2
V
3
ref1
SIF1
4
ADDR2
V
V
CRESET
V
XTALO
SYSCLK
5 6
SSD1
7
DDD1
8 9
SSD4
XTALI
10 11
P2
12 13
SCK
14
WS
15
SDO2 33
16
17
SDO1
18
SDI2
19
SDI1
20
TEST1
21
22
TEST2
MONOIN
TDA9875AH
23
24
25
EXTIL
EXTIR
SCIR1
26
SCIL1
27
SSD3
V
28
SCIR2
29
SCIL2
30
DEC2
V
31
32
ref(p)Vref(n)
V
48 47 46 45 44 43 42 41 40 39 38 37 36 35 34
V
SSA3
PCAPL PCAPR V
ref3
SCOL2 SCOR2 V
SSA4
V
SSD2
SCOL1 SCOR1 V
ref2
i.c. i.c. V
SSA2
i.c. i.c.
MHB599
Fig.3 Pin configuration (TDA9875AH).
1999 Dec 20 10
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

6 FUNCTIONAL DESCRIPTION

6.1 Demodulator and decoder section

6.1.1 SIF INPUT Two input pins are provided: SIF1 e.g. for terrestrial TV
and SIF2 e.g. for a satellite tuner. For higher SIF signal levels the SIF input can be attenuated with an internal switchable10 dB resistor divider. As nospecificfiltersare integrated, both inputs have the same specification giving flexibility in application. The selected signal is passed through an AGC circuit and then digitized by an 8-bit ADC operating at 24.576 MHz.
6.1.2 AGC The gain of the AGC amplifier is controlled from the ADC
output by means of a digital control loop employing hysteresis. The AGC has a fast attack behaviour to prevent ADC overloads and a slow decay behaviour to prevent AGC oscillations. For AM demodulation the AGC must be switched off. When switched off, the control loop is reset and fixed gain settings can be chosen (see Table 15).
The AGC can be controlled via the I2C-bus. Details can be found in the I2C-bus register definitions (see Chapter 10).
6.1.3 MIXER The digitized input signal is fed to the mixers, which mix
one or both input sound carriers down to zero IF. A 24-bit control word for each carrier sets the required frequency. Access to the mixer control word registers is via the I2C-bus. When receiving NICAM programs, a feedback signal is added to the control word of the second carrier mixer to establish a carrier-frequency loop.
6.1.4 FM AND AM DEMODULATION An FM or AM input signal is fed via a band-limiting filter to
a demodulator that can be used for either FM or AM demodulation. Apart from the standard (fixed) de-emphasis characteristic, an adaptive de-emphasis is availableforencodedsatelliteprograms.Astereodecoder recovers the left and right signal channels from the demodulated sound carriers. Both the European and Korean stereo systems are supported.
6.1.5 FM IDENTIFICATION The identification of the FM sound mode is performed by
AM synchronous demodulation of the pilot signal and narrow-band detection of the identification frequencies. Theresultisavailableviathe I2C-businterface.Aselection can be made via the I2C-bus for B/G, D/K and M standard and for three different modes that represent different trade-offs between speed and reliability of identification.
6.1.6 NICAM DEMODULATION The NICAM signal is transmitted in a DQPSK code at a bit
rate of 728 kbit/s. The NICAM demodulator performs DQPSK demodulation and feeds the resulting bitstream and clock signal onto the NICAM decoder and, for evaluation purposes, to pins PCLK and NICAM.
Atimingloopcontrolsthefrequencyof the crystal oscillator to lock the sampling rate to the symbol timing of the NICAM data.
6.1.7 NICAM DECODER The device performs all decoding functions in accordance
with the the frame alignment word, the data is descrambled by applyingthedefinedpseudo-randombinarysequenceand the device will then synchronize to the periodic frame flag bit C0.
Bit VDSP (see Section 10.4.1) indicates that the decoder has locked to the NICAM data and that the data is valid sound data.
The status of the NICAM decoder can be read outfrom the NICAM status register by the user (see Section 10.4.2). Bit OSB indicates that the decoder has locked to the NICAM data. Bit C4 indicates that the sound conveyed by the FM mono channel is identical to the sound signal conveyed by the NICAM channel.
The error byte contains the number of sound sample errors, resulting from parity checking, that occurred in the past 128 ms period. The Bit Error Rate (BER) can be calculated using the following equation:
BER
“EBU NICAM 728 specification”
bit errors
----------------------­total bits
error byte 1.74× 10
×=
. After locking to
5–
1999 Dec 20 11
Page 12
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
6.1.8 NICAM AUTO-MUTE This function is enabled by setting bit AMUTE to logic 0
(see Section 10.3.11). Upper and lower error limits may be defined by writing
appropriate values to two registers in the I2C-bus section (see Sections 10.3.13 and 10.3.14). When the number of errors in a 128 ms period exceeds the upper error limit the auto-mute function will switch the output sound from NICAM to whatever sound is on the first sound carrier (FM orAM).Whentheerrorcountissmallerthanthelower error limit the NICAM sound is restored.
The auto-mute function can be disabled by setting bit AMUTE to logic 1. In this condition clicks become audible when the error count increases; the user will hear a signal of degrading quality.
A decision to enable/disable the auto-muting is taken by the microcontroller based on an interpretation of the application control bits C1, C2, C3 and C4 and, possibly, any additional strategy implemented by the set maker in the microcontroller software.
For NICAM L applications, it is recommended to demodulate AM sound in the first sound IF and connect the audio signal to the mono input of the TDA9875A. By setting bit AMSEL (see Section 10.3.11), the auto-mute function will switch to the audio ADC instead of switching to the first sound carrier. The ADC source selector (see Section 10.3.20) should be set to mono input, where the AM sound signal should be connected.
6.1.9 CRYSTAL OSCILLATOR The circuitry of the crystal oscillator is fully integrated, only
the external 24.576 MHz crystal is needed (see Fig.10).
6.1.10 TEST PINS
Bit CLRPOR (see Section 10.3.2) resets the Power-on reset flip-flop to LOW. If this is detected, an initialization of the TDA9875A has to be carried out to ensure reliable operation.
6.1.12 POWER-ON RESET The reset is active LOW. In order to perform a reset at
power-up, a simple RC circuit may be used which consists of the integrated passive pull-up resistor and an external capacitor connected to ground. The pull-up resistor has a nominal value of 50 k, which can easily be measured between pins CRESET and V
. Before the supply
DDD2
voltage has reached a certain minimum, the state of the circuit is completely undefined, and it remains in this undefined state unless a reset is applied.
The reset is guaranteed to be active when:
The power supply is within the specified limits (4.75 and 5.5 V)
The crystal oscillator is functioning
The voltage at pin CRESET is below 0.3V
V
= 5.0 V, typically below 1.8 V).
DDD
DDD
(1.5 V if
The required capacitor value depends on the gradient of the rising power supply voltage. The time constant of the RC circuit should be clearly larger than the rise time of the power supply, to make sure that the reset condition is always satisfied (see Fig.4), even considering the tolerance spread. To avoid problems with a too slow discharging of the capacitor at power-down, it may be helpful to add a diode from pin CRESET to V
DDD
. It should be noted that the internal ESD protection diode does not help here as it only conducts at higher voltages. Under difficult power supply conditions (e.g. very slow or non-monotonic ramp-up), it is recommended to drive the reset line from a microcontroller port or the like.
Test pins TEST1 and TEST2 are active HIGH and in the normal operating mode of the device they are connected to V
. Test functions are for manufacturing tests only
SSD1
and are not available to customers. Without external circuitry these pins are pulled down to a LOW level with internal resistors.
6.1.11 POWER FAIL DETECTOR The power fail detector monitors the internal power supply
for the digital part of the device. If the supply has temporarily been lower than the specified lower limit, the Power-onresetbit POR (see Section 10.4.1), will be set to logic 1.
1999 Dec 20 12
handbook, halfpage
5
voltage
(V)
1.5
V
> 4.75 V
DDD
V
CRESET
reset active guaranteed
Fig.4 Reset at power-on.
MHB595
< 0.3V
DDD
t
Page 13
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1999 Dec 20 13
FM
2
DC
FILTER
from ADC
2
I
2
I
NICAM
ADAPTIVE
DE-EMPHASIS
S1
S2
2
DC
FILTER
2
2
2
FIXED
DE-EMPHASIS
FIXED
DE-EMPHASIS
LEVEL ADJUST
2
LEVEL ADJUST
LEVEL ADJUST
LEVEL ADJUST
LEVEL ADJUST
MATRIX
2 2 2
2
4
6
8
10
handbook, full pagewidth
DIGITAL
CROSSBAR
SELECT
2
2
2
2
2
MATRIX
MATRIX
MATRIX
MATRIX
MATRIX
AUTOMATIC
VOLUME
LEVEL
VOLUME
SOFT-MUTE
BASS/TREBLE
BEEPER
LEVEL ADJUST AND MUTE
LEVEL ADJUST AND MUTE
LEVEL ADJUST
BASS/TREBLE
BASS BOOST
SPATIAL PSEUDO VOLUME
CONTOUR
SOFT-MUTE
BEEPER
2
2
2
2
2
Main
Auxiliary
I
I
DAC

6.2 Digital signal processing

Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
2
S1
2
S2
2 4
12
16
Fig.5 DSP data flow diagram.
MONITOR
SELECT
PEAK
DETECTION
1
2
I
C-bus
MGK108
Page 14
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
6.2.1 LEVEL SCALING All input channels to the digital crossbar switch (except for
the loudspeaker feedback path) are equipped with a level adjust facility to change the signal level in a range from +15 to 15 dB (see Fig.5). It is recommended to scale all input channels to be 15 dB below full-scale (15 dB full-scale) under nominal conditions.
6.2.2 NICAM PATH The NICAM path has a switchable J17 de-emphasis.
6.2.3 FM (AM) PATH A high-pass filter suppresses DC offsets from the
FM demodulator due to carrier frequency offsets and supplies the monitor/peak function with DC values and an unfiltered signal, e.g. for the purpose of carrier detection.
The de-emphasis function offers fixed settings for the supported standards (50, 60 or 75 µs and J17).
An adaptive de-emphasis is available for Wegener-Panda 1 encoded programs.
A matrix performs the dematrixing of the A2 stereo, dual and mono signals.
6.2.4 NICAM AUTO-MUTE If NICAM B/G, I or D/K is received, the auto-mute is
enabled and the signal quality becomes poor, the digital crossbar switch switches automatically to FM and switches the matrix to channel 1. The automatic switching depends on the NICAM bit error rate.
The auto-mute function can be disabled via the I2C-bus. For NICAM L applications, it is recommended to
demodulateAMsoundinthe first sound IF andconnectthe audio signal to the mono input of the TDA9875A. By setting bit AMSEL (see Section 10.3.11), the auto-mute function will switch to the audio ADC instead of switching to the first sound carrier. The ADC source selector bits (see Section 10.3.20) should be set to mono input, where the AM sound signal should be connected.
6.2.5 MONITOR
Optionally, the peak value can be measured instead of simply taking samples. The internally stored peak value is reset to zero when the data is read via the I2C-bus. The monitor function may be used, for example, for signal level measurements or carrier detection.
6.2.6 LOUDSPEAKER (MAIN) CHANNEL The matrix provides the following functions: forced mono,
stereo, channel swap, channel 1, channel 2 and spatial effects.
There are fixed coefficient sets for spatial settings of 30%, 40% and 52%.
The Automatic Volume Level (AVL) function provides a constant output level of 23 dB (full-scale) for input levels between 0 and 29 dB (full-scale). There are some fixed decay time constants to choose from, i.e. 2, 4 and 8 s.
Pseudostereoisbasedonaphaseshiftinonechannelvia a second-order all-pass filter. There are fixed coefficient sets to provide 90 degrees phase shift at frequencies of 150, 200 and 300 Hz.
Volume is controlled individually for each channel ranging from +24 to 83 dB with 1 dB resolution. There is also a muteposition.Forthepurposeofasimplecontrolsoftware in the microcontroller, the decimal number that is sent as an I2C-bus data byte for volume control is identical to the volume setting in dB (e.g. the I2C-bus data byte +10 sets the new volume value to +10 dB).
Balance can be realized by independent control of the left and right channel volume settings.
Contour is adjustable between 0 and +18 dB with 1 dB resolution. This function is linked to the volume setting by means of microcontroller software.
Bass is adjustable between +15 and 12 dB with 1 dB resolution and treble is adjustable between +12 and 12 dB with 1 dB resolution.
For the purpose of a simple control software in the microcontroller, the decimal number that is sent as an I2C-bus data byte for contour, bass or treble is identical to the new contour, bass or treble setting in dB (e.g. the I2C-bus data byte +8 sets the new value to +8 dB).
This function provides data words from a number of locations in the signal processing paths to the I2C-bus interface (2 data bytes). Signal sources include the FM demodulator outputs, most inputs to the digital crossbar switch and the outputs of the ADC. Source selection and data read-out is performed via the I2C-bus.
1999 Dec 20 14
Extra bass boost is provided up to 20 dB with 2 dB resolution. The implemented coefficient set serves merely as an example on how to use this filter.
The beeper provides tones in a range from approximately 400 Hz to 30 kHz. The frequency can be selected via the I2C-bus. The beeper output signal is added to the loudspeaker and headphone channel signals.
Page 15
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
The beeper volume is adjustable with respect to full-scale between 0 and 93 dB with 3 dB resolution. The beeper is not effected by mute.
Soft mute provides a mute ability in addition to volume control with a well defined time (32 ms) after which thesoft mute is completed. A smooth fading is achieved by a cosine masking.
6.2.7 HEADPHONE (AUXILIARY) CHANNEL The matrix provides the following functions: forced mono,
stereo, channel swap, channel 1 and channel 2 (or C and S in Dolby Surround Pro Logic mode).
Volume is controlled individually for each channel in a range from +24 to 83 dB with 1 dB resolution. There is also a mute position. For the purpose of a simple control software in the microcontroller, the decimal number that is sent as an I2C-bus data byte for volume control is identical to the volume setting in dB (e.g. the I2C-bus data byte +10 sets the new volume value to +10 dB).
Balance can be realized by independent control of the left and right channel volume settings.
Bass is adjustable between +15 and 12 dB with 1 dB resolution and treble is adjustable between +12 and 12 dB with 1 dB resolution. For the purpose of a simple control software in the microcontroller, the decimal number that is sent as an I2C-bus data byte for bass or treble is identical to the new bass or treble setting in dB (e.g. the I2C-bus data byte +8 sets the new value to +8 dB).
The beeper provides tones in a range from approximately 400 Hz to 30 kHz. The frequency can be selected via the I2C-bus. The beeper output signal is added to the loudspeaker and headphone channel signals. The beeper volume is adjustable with respect to full-scale between 0 and 93 dB with 3 dB resolution. The beeper is not effected by mute.
Soft mute provides a mute ability in addition to volume control with a well defined time (32 ms) after which thesoft mute is completed. A smooth fading is achieved by a cosine masking.
6.2.8 FEATURE INTERFACE The feature interface comprises two I2S-bus input/output
ports and a system clock output. Each I2S-bus port is equipped with level adjust facilities that can change the signal level in a range from +15 to 15 dB with 1 dB resolution. Outputs can be disabled to improve EMC performance.
TheI2S-busoutputmatrixprovidesthefollowingfunctions: forced mono, stereo, channel swap, channel 1 and channel 2.
One example of how the feature interface can be used in a TV set is to connect an external Dolby Surround Pro Logic DSP, such as the SAA7710, to the I2S-bus ports. Outputs must be enabled and a suitable master clock signal for the DSP can be taken from pin SYSCLK. A stereo signal from any source will be output on one of the I2S-bus serial data outputs and the four processed signal channels will be entered at both I2S-bus serial data inputs. Left and right could then be output to the power amplifiers via the Main channel, centre and surround via the Auxiliary channel.
6.2.9 CHANNEL FROM THE AUDIO ADC The signal level at the output of the ADC can be adjusted
in a range from +15 to 15 dB with 1 dB resolution. The audio ADC itself is scaled to a gain of 6 dB.
6.2.10 CHANNEL TO THE ANALOG CROSSBAR PATH Level adjust with control positions 0, +3, +6 and +9 dB.
6.2.11 DIGITAL CROSSBAR SWITCH Input channels to the crossbar switch are from the audio
ADC, I2S1, I2S2, FM path, NICAM path and from the loudspeaker channel path after matrix and AVL (see Fig.8).
Outputchannelscomprise loudspeaker, headphone, I2S1, I2S2 and audio DACs for line output and SCART. I2S1 and I2S2 outputs also provide digital outputs from the loudspeaker and headphone channels, but without the beeper signals.
6.2.12 SIGNAL GAIN There are a number of functions that can provide signal
gain, e.g. volume, bass and treble control. Great care has to be taken when using gain with large input signals in order not to exceed the maximum possible signal swing, which would cause severe signal distortion. The nominal signal level of the various signal sources to the digital crossbar switch should be 15 dB below digital full-scale (15 dB full-scale). This means that a volume setting of, say, +15 dB would just produce a full-scale output signal and not cause clipping, if the signal level is nominal.
Sending illegal data patterns via the I2C-bus will not cause any changes of the current setting for the volume, bass, treble, bass boost and level adjust functions.
1999 Dec 20 15
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
6.2.13 EXPERT MODE The TDA9875A provides a special expert mode that gives
directwriteaccesstotheinternalCoefficientRAM(CRAM) of the DSP. It can be used to create user-defined characteristics, such as a tone control with different corner frequencies or special boost/cut characteristics to correct the low-frequency loudspeaker and/or cabinet frequency responsesby means of the bass boost filter. However, this mode must be used with great care.
6.2.14 DSP FUNCTIONS
Table 5 Overview of DSP functions
FUNCTION
Bass control for loudspeaker and headphone output
Treble control for loudspeaker and headphone output
Contour for loudspeaker output yes control range 0to +18 dB
Bass boost for loudspeaker output yes control range 0 to +20 dB
Volume control for each separate channel in loudspeaker and headphone output
Soft mute for loudspeaker and headphone output
Spatial effects yes anti-phase crosstalk positions 30, 40 and 52 % Pseudo stereo yes 90 degrees phase shift at frequency 150, 200 and 300 Hz Beeper additional to the signal in
the loudspeaker and headphone channel
Automatic Volume Level (AVL) yes step width quasi continuously
EXPERT
MODE
yes control range 12 to +15 dB
resolution 1 dB resolution at frequency 40 Hz
yes control range 12 to +12 dB
resolution 1 dB resolution at frequency 14 kHz
resolution 1 dB resolution at frequency 40 Hz
resolution 2 dB resolution at frequency 20 Hz corner frequency 350 Hz
no control range 83 to +24 dB
resolution 1 dB mute position at step 1010 1100
no processing time 32 ms
yes beep frequencies see Section 10.3.38
control range 0 to 93 dB resolution 3 dB mute position at step 0010 0000
AVL output level for an input level between 0 and 29 dB (full-scale)
attack time 10 ms decay time constant 2, 4 and 8 s
More information on the functions of this device, such as the number of coefficients per function, their default values, memory addresses, etc., can be made available on request.
PARAMETER VALUE UNIT
23 dB
1999 Dec 20 16
Page 17
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
FUNCTION
EXPERT
MODE
PARAMETER VALUE UNIT
General no 3 dB lower corner frequency of DSP 10 Hz
1 dB bandwidth of DSP 14.5 kHz
Level adjust I
2
S1 and I2S2 inputs yes control range 15 to +15 dB
resolution 1 dB
Level adjust I
2
S1 and I2S2 outputs yes control range 15 to +15 dB
resolution 1 dB
mute position at step 0001 0000 Level adjust analog crossbar path no control positions 0, 3, 6 and 9 dB Level adjust audio ADC outputs yes control range +15 to 15 dB
resolution 1 dB Level adjust NICAM path yes control range +15 to 15 dB
resolution 1 dB Level adjust FM path yes control range +15 to 15 dB
resolution 1 dB

6.3 Analog audio section

handbook, full pagewidth
SCART 1
SCART 2
external
mono
NICAM
I I I I
2 2 2 2
FM
S1 S2 S1 S2
2
3 dB
2
3 dB
2
2
D
2
2
2
A
2 2 2 2 2 2
ANALOG
CROSSBAR
SWITCH
DSP AND
DIGITAL
CROSSBAR
SWITCH
2
2
2
2
2
2
ANALOG
MATRIX
ANALOG
MATRIX
ANALOG
MATRIX
A
D
D
A
D
A
3 dB
2 2
0 dB
3 dB
2
0 dB
3 dB
2
0 dB
2
SCART 1
2
SCART 2
2
Line output
2
Main
2
Auxiliary
MGK109
Fig.6 Block diagram for the audio section.
1999 Dec 20 17
Page 18
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
6.3.1 ANALOG CROSSBAR SWITCH AND ANALOG MATRIX There are a number of analog input and output ports with
theTDA9875A (see Figs 6 and 8). Analog source selector switches are employed to provide the desired analog signal routing capability. The analog signal routing is performed by the analog crossbar switch section. A dual audio ADC provides the connection to the DSP section and a dual audio DAC provides the connection from the DSP section to the analog crossbar switch. The digital signal routing is performed by a digital crossbar switch.
The basic signal routing philosophy of the TDA9875A is that each switch handles two signal channels at the same time, e.g. left and right, language A and B, directly at the source.
Each source selector switch is followed by an analog matrix to perform further selection tasks, such as putting a signal from one input channel, say language A, to both output channels or for swapping left and right channels (see Fig.7).
handbook, halfpage
left input
right input
ANALOG
MATRIX
left output right output
MGK110
Fig.7 Analog matrix.
The analog matrix provides the functions given in Table 6.
6.3.2 SCART INPUTS The SCART specification allows for a signal level of up to
2 V (RMS). Because of signal handling limitations, due to the 5 V supply voltage of the TDA9875A, it is necessary to have fixed 3 dB attenuators at the SCART inputs to obtain a 2 V input. This results in a 3 dB SCART-to-SCART copy gain. If 0 dB copy gain is preferred (with a maximum inputof 1.4 V), there are 0/3 dB amplifiers at the outputs of SCART 1 and SCART 2 and at the line output.
The input attenuator is realized by an external series resistor in combination with the input impedance, both of which form a voltage divider. With this voltage divider the maximum SCART signal level of 2 V (RMS) is scaled down to 1.4 V (RMS) at the input pin.
6.3.3 EXTERNAL AND MONO INPUTS The3 dB input attenuators are not requiredfortheexternal
and mono inputs, because those signal levels are under control of the TV designer. The maximum allowed input level is 1.4 V (RMS). By adding external series resistors, the external inputs can be used as an additional SCART input.
6.3.4 SCART OUTPUTS The SCART outputs employ amplifiers with two gain
settings. The gain can be set to 3 or 0 dB via the I2C-bus. The 3 dB position is needed to compensate for the 3 dB attenuation at the SCART inputs should SCART-to-SCART copies with 0 dB gain be preferred [under the condition of 1.4 V (RMS) maximum input level]. The 0 dB position is needed, for example, for an external-to-SCART copy with 0 dB gain.
Table 6 Analog matrix functions
MATRIX OUTPUT
MODE
LEFT OUTPUT RIGHT OUTPUT
1 left input right input 2 right input left input 3 left input left input 4 right input right input
All switches and matrices are controlled via the I2C-bus.
1999 Dec 20 18
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
6.3.5 LINE OUTPUT The line output can provide an unprocessed copy of the
audio signal in the loudspeaker channels. This can be either an external signal that comes from the dual audio ADC, or a signal from an internal digital audio source that comes from the dual audio DAC. The line output employs amplifiers with two gain settings. The 3 dB position is needed to compensate for the attenuation at the SCART inputs, while the 0 dB position is needed, for example, for non-attenuated external or internal digital signals (see Section 6.3.4).
6.3.6 LOUDSPEAKER (MAIN) AND HEADPHONE (AUXILIARY) OUTPUTS
Signals from any audio source can be applied to the loudspeakerandtotheheadphone output channels via the digital crossbar switch and the DSP.
6.3.7 DUAL AUDIO DAC
The TDA9875A contains three dual audio DACs, one for theconnectionfromtheDSPtotheanalogcrossbarswitch section and two for the loudspeaker and headphone outputs. Each of the three dual low-noise high-dynamic range DACs consists of two 15-bit DACs with current outputs, followed by a buffer operational amplifier. The audio DACs operate with four-fold oversampling and noise shaping.
6.3.8 DUAL AUDIO ADC There is one dual audio ADC in the TDA9875A for the
connection of the analog crossbar switch section to the DSP. The dual audio ADC consists of two bitstream third-order sigma-delta audio ADCs and a high-order decimation filter.
6.3.9 STANDBY MODE The standby mode, selected by setting bit STDBY to
logic 1 (see Section 10.3.2) disables most functions and reduces power dissipation. The analog crossbar switch and the SCART section remain operational and can be controlled by the I2C-bus to support copying of analog signals from SCART-to-SCART.
Unused internal registers may lose their information in the standby mode. Therefore, the device needs to be initialized on returning to the normal operating mode. This can be accomplished in the same way as after a Power-on reset.
6.3.10 SUPPLY GROUND The different supply grounds VSSare internally connected
via the substrate. It is recommended to connect all ground pins by means of a copper plane close to the pins.
1999 Dec 20 19
Page 20
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1999 Dec 20 20
SCART 1
handbook, full pagewidth
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
FM/AM
part
NICAM
part
I2S1
2
I
S2
SCART 2
external
mono
FM/AM
DEMODULATOR
NICAM
DECODER
ADAPTIVE
DE-EMPHASIS
FIXED
DE-EMPHASIS
DE-EMPHASIS
ADC
6 dB
STEREO
DECODER
ADC
LEVEL
ADJUST
FM
LEVEL
ADJUST
NICAM LEVEL
ADJUST
I2S1 INPUT LEVEL
ADJUST
I2S2 INPUT LEVEL
ADJUST
DIGITAL MATRIX
DIGITAL MATRIX
DIGITAL MATRIX
DIGITAL MATRIX
DIGITAL MATRIX
AUTOMATIC
VOLUME
LEVEL
I2S1
OUTPUT
LEVEL
ADJUST
I2S2
OUTPUT
LEVEL
ADJUST
DAC
GAIN
LOUDSPEAKER
PROCESSING
HEADPHONE
PROCESSING
DAC
CHANNEL
CHANNEL
DAC
DAC
ANALOG
MATRIX
ANALOG
MATRIX
ANALOG
MATRIX
Main
Auxiliary
2
I
S1
2
I
S2
BUFFER
0/+3 dB
BUFFER
0/+3 dB
BUFFER
0/+3 dB
Line
SCART 1
SCART 2
MHB600
Fig.8 Audio signal flow diagram.
Page 21
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

7 LIMITING VALUES

In accordance with the Absolute Maximum Rating System (IEC 134).
SYMBOL PARAMETER CONDITIONS MIN. MAX. UNIT
V
DD
V
DD
V
n
I
, I
DDD
SSD
I
lu(prot)
P
tot
T
stg
T
amb
V
es
Notes
1. Human body model: C = 100 pF; R = 1.5 k.
2. Machine model: C = 200 pF; L = 0.75 µH; R = 0 .
DC supply voltage 0.5 +6.0 V voltage differences between two VDD pins 550 mV voltage on any other pin 0.5 VDD+ 0.5 V DC current per digital supply pin −±180 mA latch-up protection current 100 mA total power dissipation 1.0 W storage temperature 55 +125 °C ambient temperature 20 +70 °C electrostatic handling voltage note 1 2000 +2000 V
note 2 200 +200 V

8 THERMAL CHARACTERISTICS

SYMBOL PARAMETER CONDITIONS VALUE UNIT
R
th(j-a)
thermal resistance from junction to ambient in free air
TDA9875A (SDIP64) 40 K/W TDA9875AH (QFP64) 50 K/W
1999 Dec 20 21
Page 22
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

9 CHARACTERISTICS

V V parameters in accordance with system A2; NICAM in accordance with resistance for AF inputs; with external components of Fig.10; unless otherwise specified.
Supplies
V V I V V I
V V V I
V
V
V
V
Demodulator supply decoupling and references
V
V
I
Audio supply decoupling and references
V
V
Z Z V
Z Z
= 300 mV; AGCOFF= 0; AGCSLOW = 0; AGCLEV = 0; level and gain settings in accordance with note 1;
SIF(p-p)
=5V; T
DD
=25°C; settings in accordance with B/G standard; FM deviation ±50 kHz; f
amb
“EBU specification”
= 1 kHz; FM sound
mod
; 1 k measurement source
SYMBOL PARAMETER CONDITIONS MIN. TYP. MAX. UNIT
DDD1 SSD1
DDD1
DDD2 SSD2
DDD2
digital supply voltage 1 4.75 5.0 5.5 V digital supply ground 1 note 2 0.0 V digital supply current 1 V
=5.0V 5873 88mA
DDD1
digital supply voltage 2 4.75 5.0 5.5 V digital supply ground 2 note 2 0.0 V digital supply current 2 V
= 5.0 V; system clock
DDD2
0.1 0.4 2 mA
output disabled
SSD3 SSD4 DDA
DDA
digital supply ground 3 note 2 0.0 V digital supply ground 4 note 2 0.0 V analog supply voltage 4.75 5.0 5.5 V analog supply current for
V
= 5.0 V; digital silence 44 56 68 mA
DDA
DACpart
SSA1
analog ground for analog
note 2 0.0 V
front-end
SSA2
analog ground for audio ADC
note 2 0.0 V
part
SSA3
analog ground for audio DAC
note 2 0.0 V
part
SSA4
DEC1
analog ground for SCART 0.0 V
analog supply decoupling
3.0 3.3 3.6 V
voltage for demodulator part
ref1
analog reference voltage for
2 V
demodulator part
ref1(sink)
DEC2
sink current at pin V
ref1
analog supply decoupling
200 −µA
3.0 3.3 3.6 V
voltage for audio ADC part
ref2
Vref2-VDEC2 Vref2-VSSA2 ref3
reference voltage ratio for audio ADCs
impedance pins V impedance pins V
ref2 ref2
to V to V
reference voltage ratio for audio DAC and operational
referenced to V V
DEC2 SSA2
referenced to V V
SSA2
SSA3
DEC2
DDA
and
and
50 %
20 k
20 k
50 %
amplifier
Vref3-VDDA Vref3-VSSA3
impedance pins V impedance pins V
ref3 ref3
to V to V
DDA SSA3
20 k
20 k
1999 Dec 20 22
Page 23
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
SYMBOL PARAMETER CONDITIONS MIN. TYP. MAX. UNIT
Power fail detector
V
th(pf)
Digital inputs and outputs
INPUTS
CMOS level input, pull-down (pins TEST1 and TEST2)
V
IL
V
IH
C
i
Z
i
CMOS level input, hysteresis, pull-up (pin CRESET)
V
IL
V
IH
V
hys
C
i
Z
i
INPUTS/OUTPUTS
power fail threshold voltage 3.9 V
LOW-level input voltage −− 0.3V HIGH-level input voltage 0.7V
−−V
DDD
DDD
V
input capacitance −− 10 pF input impedance 50 k
LOW-level input voltage −− 0.3V HIGH-level input voltage 0.7V
−−V
DDD
DDD
V
hysteresis voltage 1.3 V input capacitance −− 10 pF input impedance 30 50 k
I2C-bus level input with Schmitt trigger, open-drain output stage, 400 kHz I2C-bus operation (pins SCL and SDA)
V
IL
V
IH
V
hys
I
LI
C
i
V
OL
C
L
LOW-level input voltage −− 0.3V HIGH-level input voltage 0.7V hysteresis voltage 0.05V
−−V
DDD
V
DDD
DDD
V
input leakage current −− ±10 µA input capacitance −− 10 pF LOW-level output voltage −− 0.6 V load capacitance −− 400 pF
TTL/CMOS level, 4 mA 3-state output stage, pull-up (pins PCLK, NICAM, ADDR1, ADDR2, P1, P2, SCK, WS, SDO1, SDO2, SDI1 and SDI2)
V
IL
V
IH
C
i
V
OL
V
OH
C
L
Z
i
LOW-level input voltage −− 0.8 V HIGH-level input voltage 2.0 −−V input capacitance −− 10 pF LOW-level output voltage −− 0.4 V HIGH-level output voltage 2.4 −−V load capacitance −− 100 pF input impedance 50 k
OUTPUTS
CMOS level output, 4 mA 3-state output stage, slew rate controlled (pin SYSCLK)
V
OL
V
OH
C
L
I
LIZ
LOW-level output voltage −− 0.3V HIGH-level output voltage 0.7V
−−V
DDD
DDD
V
load capacitance −− 100 pF 3-state leakage current Vi= 0 to V
DDD
−− ±10 µA
1999 Dec 20 23
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
SYMBOL PARAMETER CONDITIONS MIN. TYP. MAX. UNIT
SIF1 and SIF2 analog inputs
V
SIF(max)(p-p)
maximum composite SIF input voltage for clipping (peak-to-peak value)
V
SIF(min)(p-p)
minimum composite SIF input voltage for lower limit of AGC
(peak-to-peak value) AGC AGC range 24 dB f
i
R
i
C
i
f
FM
f
FM(FS)
C/N
FM
C/N
N
α
ct
input frequency 4 9.2 MHz
input resistance AGCLEV = 0 10 −−k
input capacitance 7.5 11 pF
FM deviation B/G standard; THD < 1% ±100 −−kHz
FM deviation full-scale level terrestrial FM;
FM carrier-to-noise ratio NFM bandwidth = 6 MHz;
NICAM carrier-to-noise ratio NN bandwidth = 6 MHz;
crosstalk attenuation
SIF1 to SIF2
Demodulator performance
THD + N total harmonic distortion plus
noise
S/N signal-to-noise ratio SC1 from FM source to any
B
3dB
f
res
3 dB bandwidth from FM source to any
frequency response
20 Hz to 14 kHz
SIF input level adjust 0 dB 941 mV SIF input level adjust 10 dB 2976 mV
SIF input level adjust 0 dB 59 mV SIF input level adjust 10 dB 188 mV
±150 −−kHz
level adjust 0 dB
white noise for S/N = 40 dB;
“CCIR468”
; quasi peak
bit error rate = 10−3;
77
66
dB
------ ­Hz
dB
------ ­Hz
white noise fi= 4 to 9.2 MHz; note 3 50 −−dB
from FM source to any output; V
= 1 V (RMS) with
o
0.3 0.5 %
low-pass filter from NICAM source to any
output; V
= 1 V (RMS) with
o
0.1 0.3 %
low-pass filter
64 70 dB
output; V
“CCIR468”
SC2 from FM source to any output; V
“CCIR468”
NICAM source; V
= 1 V (RMS); note 4
o
= 1 V (RMS);
o
; quasi peak
= 1 V (RMS);
o
; quasi peak
60 66 dB
−− −
14.5 15 kHz
output from NICAM source to any
14.5 15 kHz
output from FM or NICAM to any
output; f
= 1 kHz;
ref
−±2dB
inclusive pre-emphasis and de-emphasis
1999 Dec 20 24
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
SYMBOL PARAMETER CONDITIONS MIN. TYP. MAX. UNIT
α
cs(dual)
α
cs(stereo)
α
AM
S/N
AM
IDENTIFICATION FOR FM SYSTEMS mod
p
C/N
p
f
ident
t
ident(on)
t
ident(off)
Analog audio inputs
dual signal channel separation note 5 65 70 dB
stereo channel separation note 6 40 45 dB
AM suppression for FM AM: 1 kHz, 30% modulation;
50 −−dB reference: 1 kHz, 50 kHz deviation
AM demodulation SIF level 100 mV (RMS);
36 45 dB 54% AM; 1 kHz AF;
“CCIR468”
pilot modulation for
; quasi peak
25 50 75 %
identification pilot sideband carrier-to-noise
ratio for identification start
27
dB
------ ­Hz
identification window B/G stereo
slow mode 116.85 118.12 Hz medium mode 116.11 118.89 Hz fast mode 114.65 120.46 Hz
B/G dual
slow mode 273.44 274.81 Hz medium mode 272.07 276.20 Hz fast mode 270.73 277.60 Hz
total identification time ON slow mode −− 2s
medium mode −− 1s fast mode −− 0.5 s
total identification time OFF slow mode −− 2s
medium mode −− 1s fast mode −− 0.5 s
MONO INPUT AND EXTERNAL INPUT V
i(nom)(rms)
nominal level input voltage (RMS value)
V
i(clip)(rms)
clipping level input voltage
THD < 3%; note 7 1250 1400 mV
(RMS value)
R
i
input resistance note 7 28 35 42 k SCART INPUTS V
i(nom)(rms)
V
i(clip)(rms)
nominal level input voltage at
input pin (RMS value)
clipping level input voltage at
input pin (RMS value)
3 dB divider with external 15 k resistor; note 8
3 dB divider with external 15 k resistor; THD < 3%; notes 7 and 8
R
i
input resistance note 7 28 35 42 k
1999 Dec 20 25
500 mV
350 mV
1250 1400 mV
Page 26
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
SYMBOL PARAMETER CONDITIONS MIN. TYP. MAX. UNIT
Analog audio outputs
LOUDSPEAKER (MAIN) AND HEADPHONE (AUXILIARY) OUTPUTS V
o(clip)(rms)
R
o
R
L(AC)
R
L(DC)
C
L
V
offset(DC)
α
mute
G
ro(main,aux)
PSRR
main,aux
SCART OUTPUTS AND LINE OUTPUT V
o(nom)(rms)
V
o(clip)(rms)
R
o
R
L(AC)
R
L(DC)
C
L
V
offset(DC)
α
mute
B bandwidth from SCART, external and
PSRR power supply ripple rejection f
clipping level output voltage
THD < 3% 1250 1400 mV
(RMS value)
output resistance 150 250 375
AC load resistance 10 −−k
DC load resistance 10 −−k
load capacitance 10 12 nF
static DC offset voltage 30 70 mV
mute suppression nominal input signal from
80 −−dB
any source; fi= 1 kHz
roll-off gain at 14.5 kHz for
from any source 3 2 dB
Main and Auxiliary channels
power supply ripple rejection
for Main and Auxiliary
channels
f
= 70 Hz;
ripple
V
= 100 mV (peak);
ripple
C
=47µF;
Vref
40 45 dB
signal from I2S-bus
nominal level output voltage
3 dB amplification 500 mV
(RMS value)
clipping level output voltage
THD < 3% 1250 1400 mV
(RMS value)
output resistance 150 250 375
AC load resistance 10 −−k
DC load resistance 10 −−k
load capacitance −− 2.5 nF
static DC offset voltage output amplifiers at 3 dB
30 50 mV
position
mute suppression nominal input signal from
80 −−dB
any source; fi= 1 kHz
20 −−kHz
mono sources;
3 dB bandwidth from DSP sources;
14.5 −−kHz
3 dB bandwidth = 70 Hz;
ripple
V
= 100 mV (peak);
ripple
C
=47µF;
Vref
40 45 dB
signal from I2S-bus
1999 Dec 20 26
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
SYMBOL PARAMETER CONDITIONS MIN. TYP. MAX. UNIT
Audio performance
THD + N total harmonic distortion plus
noise
V fi= 1 kHz; bandwidth 20 Hz to 15 kHz; note 9
S/N signal-to-noise ratio reference voltage
V
“CCIR468”
note 9
α
ct
crosstalk attenuation between any analog input
pairs; fi= 1 kHz between any analog output
pairs; f
α
cs
channel separation between left and right of any
input pair between left and right of any
output pair
G
A
gain from SCART-to-SCART with 3 dB input voltage divider
output amplifier in 3 dB position; R
output amplifier in 0 dB position; R
Crystal specification (fundamental mode)
f
xtal
C
L
C
1
C
0
Φ
pull
crystal frequency 24.576 MHz load capacitance 20 pF series capacitance 20 fF parallel capacitance −− 7pF pulling sensitivity CLchanged from
18 to 16 pF
R
R
R
N
equivalent series resistance at nominal frequency −− 30 equivalentseries resistance of
unwanted mode
T temperature range 20 +25 +70 °C
= 1 V (RMS);
i=Vo
from any analog audio input to I
fromI
2
S-bus
2
S-bustoanyanalog
0.1 0.3 %
0.1 0.3 %
audio output SCART-to-SCART copy 0.1 0.3 % SCART-to-Main copy 0.2 0.5 %
= 1.4 V (RMS);fi= 1 kHz;
o
; quasi peak;
from any analog audio input to I
fromI
2
S-bus
2
S-bustoanyanalog
73 77 dB
78 85 dB
audio output SCART-to-SCART copy 78 85 dB SCART-to-Main copy 73 77 dB
70 −−dB
65 −−dB
=10kHz
i
65 −−dB
60 −−dB
1.5 0 +1.1 dB
=15kΩ±10%
ext
4.5 3.0 1.9 dB
=15kΩ±10%
ext
25
2R
−−Ω
R
10
----------­pF
6–
1999 Dec 20 27
Page 28
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
SYMBOL PARAMETER CONDITIONS MIN. TYP. MAX. UNIT
X
J
X
D
X
A
adjustment tolerance −− ±30 10 drift across temperature range −− ±30 10 ageing −− ±5
Notes
1. Definitions of levels and level setting: a) The full-scale level for analog audio signals is 1.4 V (RMS). b) The nominal level at the digital crossbar switch is defined at 15 dB (full-scale). c) Nominal audio input levels for external and mono: 500 mV (RMS) at 9 dB (full-scale). d) See also Tables 7 and 8.
2. All analog and digital supply ground pins are connected internally.
3. Set demodulator to AM mode. Apply an AM carrier (with 1 kHz and 100%) to one channel. Check AGC step. Switch AGC offandsetAGCtothegainstepfound.Measurethe1 kHz signal level of this channel and take it as a reference. Switch to the other SIF input to which no signal is connected and which is terminated with 50 . Now measure the 1 kHz crosstalk signal level. The SIF source resistance should be low (50 ).
4. NICAM in accordance with
“EBU specification”
. Audio performance is limited by the dynamic range of the NICAM728 system. Due to compansion, the quantization noise is never lower than 62 dB (unweighted RMS) with respect to the input level.
5. FM source; in dual mode only A (respectively B) signal modulated; measured at B (respectively A) channel output; Vo= 1 V (RMS) of modulated channel.
6. FM source; in stereo mode only L (respectively R) signal modulated; measured at R (respectively L) channel output; Vo= 1 V (RMS) of modulated channel. The stereo channel separation may be limited by adjustment tolerances of the transmitter.
7. If the supply voltage for the TDA9875A is switched off, because of the ESD protection circuitry, all audio input pins are short-circuited. To avoid a short-circuit at the SCART inputs a 15 k resistor (3 dB divider) has to be used.
8. The SCART specification allows a signal level of up to 2 V (RMS). Because of signal handling limitations due to the 5 V supply voltage for the TDA9875A, there is a need for fixed 3 dB attenuators at the SCART inputs. To achieve SCART-to-SCART copies with 0 dB gain, there are 3 dB/0 dB amplifiers at the outputs of SCART 1 and SCART 2 and at the line output. The attenuator is realized by an internal resistor that works together with an external series resistor as a voltage divider. With this voltage divider the maximum SCART input signal level of 2 V (RMS) is scaled down to 1.4 V (RMS) at the input pin. To avoid clipping, the 3 dB gain must not be used if the SCART input signal is larger than 1.4 V (RMS).
9. ADC level adjust is 6 dB, all other level adjusts are 0 dB. If an external 3 dB divider is used set output buffer gain to 3 dB, tone control to 0 dB, AVL off and volume control to 0 dB.
6
6
10
----------­year
6–
1999 Dec 20 28
Page 29
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1999 Dec 20 29
Table 7 Level setting FM, AM and NICAM at 0 dB (full-scale) = 1.4 V (RMS); note 1
TRANSMITTER
STANDARD MODE
M 2 channel 15 kHz deviation 24 dB (full-scale);
B/G 2 channel 27 kHz deviation 19 dB (full-scale) 1 5.5 MHz FM 50 µs+4dB
NICAM 11.2 dB (full-scale) 18 dB (full-scale) 1 5.5 MHz FM 50 µs+4dB
I NICAM 15.8 dB (full-scale) 23 dB (full-scale) 1 6.0 MHz FM 50 µs+4dB
D/K 2 channel 27 kHz deviation 19 dB (full-scale) 1 6.5MHz FM 50 µs+4dB
2 channel 27 kHz deviation 19 dB (full-scale) 1 6.5 MHz FM 50 µs+4dB
2 channel 27 kHz deviation 19 dB (full-scale) 1 6.5 MHz FM 50 µs+4dB
NICAM 11.2 dB (full-scale) 18 dB (full-scale) 1 6.5 MHz FM 50 µs+4dB
L/L accent NICAM 54% AM 19 dB (full-scale) 1 6.5 MHz AM 50 µs+5dB
NOMINAL
MODULATION
DEPTH
NOMINALLEVELAT
DEMODULATOR
OUTPUT
note 2
FM/NICAM
CARRIER FREQUENCY MODE IDENT DE-EMPHASIS
1 4.5 MHz FM 75 µs+9dB 2 4.724 MHz FM on 75 µs+9dB
2 5.742 MHz FM on 50 µs+4dB
2 5.85 MHz NICAM off J17 +3 dB
2 6.552 MHz NICAM off J17 +8 dB
2 6.742 MHz FM on 50 µs+4dB
2 6.25 MHz FM on 50 µs+4dB
2 5.742 MHz FM on 50 µs+4dB
2 5.85 MHz NICAM off J17 +3 dB
2 5.85 MHz NICAM off J17 +3 dB
LEVEL
ADJUST
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
Notes
1. Nominal level at digital crossbar is defined at 15 dB (full-scale). DAC gain setting 6 dB. Output buffer setting 0 dB. Nominal SCART output level 500 mV (RMS).
2. For stereo signals the output level is 6 dB lower. The level adjust has to be increased by 6 dB.
Page 30
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1999 Dec 20 30
Table 8 Level setting SAT FM at 0 dB (full-scale) = 1.4 V (RMS)
TRANSMITTER
SOURCE
SAT FM, stereo 50 kHz deviation 13 dB (full-scale) +4 dB 9 dB (full-scale) +6 dB 0 dB 1 V (RMS) SAT FM, mono 85 kHz deviation 9 dB (full-scale) 0 dB
MAXIMUM
MODULATION
DEPTH
NOMINAL LEVEL AT
DEMODULATOR
OUTPUT
FM LEVEL
ADJUST
SETTING
MAXIMUM
LEVEL AT
CROSSBAR
DAC GAIN
SETTING
OUTPUT BUFFER
NOMINAL SCART
OUTPUT VOLTAGE
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
Page 31
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

10 I2C-BUS CONTROL

10.1 Introduction

The TDA9875A is fully controlled via the I2C-bus. Control is exercised by writing data to one or more internal registers. Status information can be read from an array of registers to enable the controlling microcontroller to determine whether any action is required.
The device has an I2C-bus slave transceiver, in accordance with the fast-mode specification, with a maximumspeedof400 kbits/s. Information concerningthe I2C-bus can be found in brochure
it”
(order number 9398 393 40011). To avoid conflicts in a real application with other ICs providing similar or complementary functions, there are four possible slave addresses available which can be selected by pins ADDR1 and ADDR2 (see Table 9).
Table 9 Possible slave addresses
ADDR2 ADDR1
LOW LOW 1011000
LOW HIGH 1011001 HIGH LOW 1011010 HIGH HIGH 1011011
The I2C-bus interface remains operational in the standby mode of the TDA9875A to allow control of the analog source selectors with regard to SCART-to-SCART copying.
The device will not respond to a ‘general call’ on the I2C-bus, i.e. when a slave address of 0000000 is sent by a master.
The data transmission between the microcontroller and the other I2C-bus controlled ICs is not disturbed when the supply voltage of the TDA9875A is not connected.
A6 A5 A4 A3 A2 A1 A0
“I2C-bus and how to use
SLAVE ADDRESS

10.2 Power-up state

At power-up the device is in the following state:
All outputs muted
No sound carrier frequency loaded
General-purpose I/O pins ready for input (HIGH)
Input SIF1 selected with:
– AGC on – Small hysteresis – SIF input level shift 0 dB.
Demodulators for both sound carriers set to FM with: – Identification for B/G and D/K, response time 1 s – Level adjust set to 0 dB – De-emphasis 50 µs – Matrix set to mono.
Main channel set to FM input with: – Spatial off – Pseudo off – AVL off – Volume mute – Bass flat – Treble flat – Contour off – Bass boost flat.
Auxiliary channel set to FM input with: – Volume mute – Bass flat – Treble flat.
Feature interface all outputs off
Beeper off
Monitoring of carrier 1 FM demodulator DC output.
After power-up a device initialization has to be performed via the I of operation, in accordance with the desired TV standard, audio control settings, etc.
2
C-bus to put the TDA9875A into the proper mode
1999 Dec 20 31
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

10.3 Slave receiver mode

As a slave receiver, the TDA9875A provides 46 registers for storing commands and data. These registers are accessed via so-called subaddresses. A subaddress can be thought of as a pointer to an internal memory location.
2
Table 10 I
S SLAVE ADDRESS 0 ACK SUBADDRESS ACK DATA ACK P
Table 11 Explanation of Table 10
S START condition SLAVE ADDRESS 7-bit device address 0 data direction bit (write to device) ACK acknowledge by slave SUBADDRESS address of register to write to DATA data byte to be written into register P STOP condition
C-bus; slave address, subaddress and data format
BIT FUNCTION
It is allowed to send more than one data byte per transmission to the TDA9875A. In this event, the subaddress is automatically incremented after each data byte, resulting in storing the sequence of data bytes at successive register locations, starting at SUBADDRESS. A transmission can start at any valid subaddress. Each byte is acknowledged with ACK (acknowledge).
There is no ‘wrap-around’ of subaddresses. Commands and data are processed as soon as they have been completely received. Functions requiring more than one
byte will, thus, be executed only after all bytes for that function have been received. If the transmission is terminated (STOP condition) before all bytes have been received, the incomplete data for that function are ignored.
Table 12 Format for a transmission employing auto-increment of subaddresses
S SLAVE ADDRESS 0 ACK SUBADDRESS ACK DATA
BYTE A
Note
1. n data bytes with auto-increment of subaddresses.
Data patterns sent to the various subaddresses are not checked for being illegal or not at that address, except for the functions of volume, bass, treble control, bass boost and level adjust.
Detection of a STOP condition without a preceding acknowledge bit is regarded as a bus error. The last operation will then not be executed.
DATA ACK P
(1)
1999 Dec 20 32
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
Table 13 Overview of the slave receiver registers
SUBADDRESS
(DECIMAL)
0 00sgggggAGC level shift, AGC gain selection 1 ccccccccgeneral configuration 2 p 0 0 m m s s s monitor select, peak detector on/off 3 ffffffffcarrier 1 frequency; most significant part 4 ffffffffcarrier 1 frequency 5 ffffffffcarrier 1 frequency; least significant part 6 ffffffffcarrier 2 frequency; most significant part 7 ffffffffcarrier 2 frequency 8 ffffffffcarrier 2 frequency; least significant part
9 ccccccccdemodulator configuration 10 ddddddddFMde-emphasis 11 00000mmmFMmatrix 12 000lllllchannel 1 output level adjust 13 000lllllchannel 2 output level adjust 14 t t 0 c 0 c c c NICAM configuration 15 000lllllNICAM output level adjust 16 llllllllNICAM lower error limit 17 uuuuuuuuNICAM upper error limit 18 mmmmmmmmaudio mute control 19 g m m m g s s s DAC output select 20 0 g m m 0 s s s SCART 1 output select 21 0 g m m 0 s s s SCART 2 output select 22 0 g m m 0 0 0 s line output select 23 ssslllllADC output select 24 0 m m m 0 s s s Main channel select 25 00ssppaaaudio effects (AVL, pseudo and spatial) 26 vvvvvvvvvolume control, Main left 27 vvvvvvvvvolume control, Main right 28 000ccccccontour control, Main 29 000bbbbbbass control, Main 30 000ttttttreble control, Main 31 0 m m m 0 s s s Auxiliary channel select 32 vvvvvvvvvolume control, Auxiliary left 33 vvvvvvvvvolume control, Auxiliary right 34 000bbbbbbass control, Auxiliary 35 000ttttttreble control, Auxiliary 36 000cccccfeature interface configuration 37 0mmm0 s s sI
MSB LSB
DATA
2
S1 output select
FUNCTION
1999 Dec 20 33
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
SUBADDRESS
(DECIMAL)
MSB LSB
38 000iiiiiI 39 000oooooI 40 0mmm0 s s sI 41 000iiiiiI 42 000oooooI
DATA
FUNCTION
2
S1 input level adjust
2
S1 output level adjust
2
S2 output select
2
S2 input level adjust
2
S2 output level adjust 43 00000f f fbeeper frequency 44 00vvvvvvbeeper volume, Main and Auxiliary 45 bbbbbbbbbass boost, Main left and right
The following sub-sections provide a detailed description of the slave receiver registers.

10.3.1 AGC GAIN REGISTER If the automatic gain control function is switched off in the general configuration register, the contents of this register will

define a fixed gain of the AGC stage. The input voltages given are meant to generate a full-scale output from the SIF ADC. If automatic gain control is on, the AGCGAIN setting is ignored. After switching off the automatic gain control function, the latest gain control setting is copied to the AGC gain register.
If the AGC input level shift bit AGCLEV is set to logic 1 the input signal is scaled with 10 dB. The AGCLEV bit is also active if the automatic gain function is enabled.
It should be noted that the input voltages should be considered as approximate target values.
Table 14 Subaddress 0 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 set to logic 0
6 B6 0 set to logic 0 5 AGCLEV 1 input signal scaled with 10 dB
0 input signal not scaled 4 AGCGAIN gain control bits (see Table 15) 3 2 1
0 (LSB)
Note
1. The default setting at power-up is 0000 0000.
1999 Dec 20 34
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
Table 15 Gain control bits
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
000/111111 0.0 941/2976 000/111110 0.8 861/2723 000/111101 1.5 788/2490 000/111100 2.3 720/2278 000/111011 3.1 659/2084 000/111010 3.9 603/1906 000/111001 4.6 551/1744 000/111000 5.4 504/1595 000/110111 6.2 461/1459 000/110110 7.0 422/1334 000/110101 7.7 386/1221 000/110100 8.5 353/1117 000/110011 9.3 323/1021 000/110010 10.1 295/934 000/110001 10.8 270/855 000/110000 11.6 247/782 000/101111 12.4 226/715 000/101110 13.2 207/654 000/101101 13.9 189/598 000/101100 14.7 173/547 000/101011 15.5 158/501 000/101010 16.3 145/458 000/101001 17.0 132/419 000/101000 17.8 121/383 000/100111 18.6 111/350 000/100110 19.4 101/321 000/100101 20.1 93/293 000/100100 20.9 85/268 000/100011 21.7 78/245 000/100010 22.5 71/224 000/100001 23.2 65/205 000/100000 24.0 59/188 (note 1)
AGC GAIN
(dB)
SIF INPUT VOLTAGE
[mV (p-p)]
Note
1. The default setting at power-up is 0000 0000.
1999 Dec 20 35
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

10.3.2 GENERAL CONFIGURATION REGISTER

Table 16 Subaddress 1 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) P2OUT This bit controls the general purpose input/output pin P2. The contents of this bit
is written directly to the corresponding pin. If input is desired, the bit must be set to logic 1 to allow the pin to be pulled LOW externally. Input from the pin is reflected in the device status register (see Section 10.4.1).
6 P1OUT This bit controls the general purpose input/output pin P1. The contents of this bit
is written directly to the corresponding pin. If input is desired, the bit must be set to logic 1 to allow the pin to be pulled LOW externally. Input from the pin is reflected in the device status register (see Section 10.4.1). P1OUT is recommended to be used for switching an SIF trap for the adjacent picture carrier in designs that employ such a trap.
5 STDBY 1 The IC is in the standby mode. Most functions are disabled and power dissipation
is somewhat reduced, but the analog selectors/matrices remain operational to support analog copying from SCART-to-SCART.
0 The IC is in the normal operating mode. On return from standby mode, the device
is in its Power-on reset mode and needs to be re-initialized.
4 INIT 1 Causes initialization of the TDA9875A to its default settings. This has the same
effect as a Power-on reset. If there is a conflict between the default settings and any bit set to logic 1 in this register, the bits of this register have priority over the corresponding default setting.
0 Automatically reset to logic 0 after initialization. When set to logic 0, the
TDA9875A is in its normal operating mode.
3 CLRPOR 1 Resets the power fail detector to LOW.
0 This bit is automatically reset to logic 0 after bit POR in the device status register
has been reset.
2 AGCSLOW 1 A longer decay time is selected for input signals with strong video modulation
(intercarrier). This bit only has an effect when bit AGCOFF = 0.
0 Selects normal attack and decay times for the AGC.
1 AGCOFF 1 Forces the AGC block to a fixed gain as defined in the AGC gain register.
0 The automatic gain control function is enabled and the contents of the AGC gain
register is ignored.
0 (LSB) SIFSEL 1 Selects pin SIF2 for input (recommended for satellite tuner).
0 Selects pin SIF1 for input (terrestrial TV).
Note
1. The default setting at power-up is 1100 0000.
1999 Dec 20 36
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

10.3.3 MONITOR SELECT REGISTER This register is used to define the signal source, the level of which is to be monitored, and if the peak level is to be

monitored. Peak level refers to the magnitude of the maximum excursion of a signal. Audio magnitude/phase is related to the FM demodulator output. Phase information is provided, when it operates in
FM mode, while magnitude is supplied in AM mode. Data can be read-out in the I2C-bus slave transmitter mode. By reading out level read-out registers (see Section 10.4)
the current peak level will be reset.
Table 17 Subaddress 2 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) PEAKMON 1 selects the peak level of a source to be monitored
0 the last sample will be supplied 6 B6 0 default value 5 B5 0 default value 4B4monitor output (see Table 18) 3B3 2B2signal source (see Table 19) 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 18 Monitor output
B4 B3 MONITOR OUTPUT
0 1 L input (channel 1, respectively) 1 0 R input (channel 2, respectively)
Table 19 Signal source (note 1)
B2 B1 B0 SIGNAL SOURCE
0 0 0 DC output of FM demodulator 0 0 1 audio magnitude/phase, FM demodulator output 0 1 0 crossbar input from FM/AM channel 0 1 1 crossbar input from NICAM channel 1 0 0 crossbar input from I 1 0 1 crossbar input from I 1 1 0 crossbar input from audio ADC channel 1 1 1 input to Main channel DAC (without beeper)
L input R input+
------------------------------------------­2
2
S1 channel
2
S2 channel
Note
1. The term ‘crossbar’ refers to the digital selector, where level-adjusted signals from various sources are available.
1999 Dec 20 37
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

10.3.4 CARRIER 1 FREQUENCY REGISTER The three bytes together constitute a 24-bit frequency

control word to represent the sound carrier (i.e. mixer) frequency in accordance with the following formula:
f
data
mix
--------­f
clk
224×=
where:
data = 24-bit frequency control word f
= desired sound carrier frequency
mix
f
= 12.288 MHz (clock frequency of mixer)
clk
224= 16777216 (number of steps in a 24-bit word size).
Example: A 5.5 MHz sound carrier frequency will be generated by sending the following sequence of data bytes to the TDA9875A (data = 7509333 in decimal notation or 72555 in hexadecimal): 01110010 10010101 01010101.
As three bytes are required to define a carrier frequency, execution of this command starts only after all bytes have been received. If an error occurs, e.g. a premature STOP condition, partial data for this function is ignored.
The default setting at power-up is 0000 0000 for all three bytes.
Most significant part at subaddress 3 and least significant part at subaddress 5 (see Table 20).
Table 20 Subaddresses 3 to 5
SUB-
ADDRESS
BIT DESCRIPTION
3 7 (MSB) carrier 1 frequency;
6
most significant part
5 4 3 2 1 0
4 7 carrier 1 frequency
6 5 4 3 2 1 0
5 7 carrier 1 frequency;
6
least significant part
5 4 3 2 1
0 (LSB)
1999 Dec 20 38
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Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

10.3.5 CARRIER 2 FREQUENCY REGISTER Same as for sound carrier 1, except for subaddresses (subaddresses 6 to 8). If the carrier 2 frequency register is used,

it will be for either the second FM sound carrier of a terrestrial or satellite FM program or the NICAM sound carrier.

10.3.6 DEMODULATOR CONFIGURATION REGISTER It is recommended to switch the FM sound mode identification off whenever the received program is not a terrestrial

2-carrier sound. Switching the identification off will reset the associated hardware to a defined state.
Table 21 Subaddress 9 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) IDMOD1 these bits define the response time after which a FM sound mode identification
6 IDMOD0
5 IDAREA 1 selects FM identification frequencies in accordance with the specification for
4 FILTBW1 selects filter bandwidth (see Table 23) 3 CH2MOD1 channel 2 receive mode: these bits control the hardware for the second sound 2 CH2MOD0
1 FILTBW0 selects the filter bandwidth (see Table 23)
0 (LSB) CH1MODE 1 selects the hardware for the first sound carrier to operate in AM mode
result may be expected; the longer the time, the more reliable the identification (see Table 22)
Korea
0 selects frequencies for Europe (B/G and D/K standard)
carrier (see Table 24); the NICAM mode employs a wider bandwidth of the decimation filters than the FM mode
0 FM mode is assumed; this applies to both terrestrial and satellite FM reception
Notes
1. The default setting at power-up is 0000 0000.
Table 22 Identification mode
B7 B6 IDENT MODE
0 0 slow 0 1 medium 1 0 fast 1 1 off/reset
Table 23 Filter bandwidth channel 1 and channel 2
FILTER
B4 B1
0 0 narrow narrow recommended for nominal terrestrial broadcast conditions and
0 1 extra wide narrow recommended only for high-deviation SAT mono carriers
1 0 medium medium recommended for moderately overmodulated broadcast conditions 1 1 wide wide recommended for strongly overmodulated broadcast conditions
BANDWIDTH
CH1 CH2
FILTER MODES
SAT with 2 carriers
(e.g. obsolete Main channel on Astra)
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Digital TV Sound Processor (DTVSP) TDA9875A
Table 24 Channel 2 receive mode
B3 B2 CHANNEL 2
00 FM 01 AM 1 0 NICAM
10.3.7 FM This register is used to select the proper de-emphasis characteristics as appropriate for the standard of the received
carrier. Bits B3 to B0 apply to sound carrier 1, bits B7 to B4 apply to sound carrier 2. In the event of A2 reception, both groups must be set to the same characteristics.
Table 25 Subaddress 10 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) ADEEM2 1 Activates the adaptive de-emphasis function, which is required for certain satellite
6B6Time constant selection for FM de-emphasis (see Table 26). 5B5 4B4 3 ADEEM1 1 Activates the adaptive de-emphasis function, which is required for certain satellite
2B2Time constant selection for FM de-emphasis (see Table 27). 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
2. The FM de-emphasis gain is 0 dB at 40 Hz.
DE-EMPHASIS REGISTER
FM channels. The standard FM de-emphasis must then be set to 75 µs (note 2).
0 The adaptive de-emphasis is off.
FM channels. The standard FM de-emphasis must then be set to 75 µs (note 2).
0 The adaptive de-emphasis is off.
Table 26 De-emphasis sound carrier 2
B6 B5 B4 DE-EMPHASIS
000 50µs (Europe) 001 60µs 010 75µs (M standard) 011 J17 100 off
Table 27 De-emphasis sound carrier 1
B2 B1 B0 DE-EMPHASIS
000 50µs (Europe) 001 60µs
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Digital TV Sound Processor (DTVSP) TDA9875A
B2 B1 B0 DE-EMPHASIS
010 75µs (M standard) 011 J17 100 off

10.3.8 FM MATRIX REGISTER This register is used to select the proper dematrixing characteristics as appropriate for the standard of the received

carrier and the related sound mode identification.
Table 28 Subaddress 11 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 default value
6 B6 0 default value 5 B5 0 default value 4 B4 0 default value 3 B3 0 default value 2B2dematrixing characteristics (see Table 29) 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 29 Dematrixing characteristics
B2 B1 B0 L OUTPUT R OUTPUT MODE
0 0 0 CH1 input; note 1 CH1 input; note 1 mono 1 0 0 1 CH2 input; note 2 CH2 input; note 2 mono 2 0 1 0 CH1 input; note 1 CH2 input; note 2 dual 0 1 1 CH2 input; note 2 CH1 input; note 1 dual swapped 1 0 0 2CH1 input CH2 input CH2 input; note 2 stereo Europe 1 0 1 stereo Korea; note 3
Notes
1. CH1 input: audio signal from FM channel 1.
2. CH2 input: audio signal from FM channel 2.
3. For stereo Korea the dematrix applies 6 dB attenuation (see Table 7).
CH1 input CH2 input+
----------------------------------------------------------­2
CH1 input CH2 input
----------------------------------------------------------­2
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.9 FM CHANNEL 1 LEVEL ADJUST REGISTER This register is used to correct for standard and station-dependent differences of signal levels. Table 30 applies to sound carrier 1.

Table 30 Subaddress 12
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12 00010011 13 00010010 14 00010001 15 00010000 mute
GAIN SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.10 FM CHANNEL 2 LEVEL ADJUST REGISTER This register is used to correct for standard and station-dependent differences of signal levels. Table 31 applies to sound

carrier 2 in its FM and AM modes. In the event of A2, channels 1 and 2 should be adjusted to the same level.
Table 31 Subaddress 13
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12 00010011 13 00010010 14 00010001 15 00010000 mute
GAIN SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.11 NICAM CONFIGURATION REGISTER The decision of whether auto-muting is permitted shall be taken by the controlling microcontroller based on information

contained in the TDA9875A’s status registers. Thus, it depends on the strategy implemented in the software whether the auto-mute function is in accordance with
The NICAM de-emphasis gain is 0 dB at 40 Hz.
Table 32 Subaddress 14 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) DCXOPULL 1 Set to lower DCXO frequency during DCXO test mode.
6 DCXOTEST 1 DCXO test mode on (available only during FM mode); note 2
5 B5 0 Set logic to 0. 4 DOUTEN 1 Enables the output of the NICAM serial data stream from the DQPSK
3 0 Set logic to 0. 2 AMSEL 1 The auto-mute function will switch the output sound from NICAM L to the
1 NDEEM 1 Switches the NICAM J17 de-emphasis off.
0 (LSB) AMUTE 1 Automatic muting is disabled. This bit has only an effect when the second
“NICAM 728 ETS Revised for Data Applications”
0 Set to higher DCXO frequency during DCXO test mode.
0 DCXO normal mode on
demodulator on pin NICAM and of the associated clock on pin PCLK
0 Both outputs will be 3-stated.
ADC output select register. With the ADC output select register the wanted signal source, e.g. the mono input, can be pre-set (see Section 10.3.20). This is useful, if the AM sound NICAM L system is demodulated externally.
0 The auto-mute function will switch the output sound from NICAM L to the
AM program on the internal first sound carrier.
0 Switches the NICAM J17 de-emphasis on.
sound carrier is set to NICAM.
0 Enables the automatic switching between NICAM and the program on the
first sound carrier (i.e. FM mono or AM), dependent on the NICAM bit error rate.
or any other preference.
Notes
1. The default setting at power-up is 0000 0000.
2. The DCXO test mode is intended for checking the DCXO control range with the actually used PCB layout and crystal type. During the normal operating mode, the DCXO test mode should not be used.
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.12 NICAM LEVEL ADJUST REGISTER

This register is used to correct for standard and station-dependent differences of signal levels. Table 33 applies to both NICAM sound outputs.
Table 33 Subaddress 15 (note 1)
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12 00010011 13 00010010 14 00010001 15 00010000 mute
GAIN SETTING (dB)
Note
1. The default setting at power-up is 000 00000.
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.13 NICAM LOWER ERROR LIMIT REGISTER

When the auto-mute function is enabled (bit AMUTE in the NICAM configuration register) and the NICAM bit error count is lower than the value contained in this register, the NICAM signal is selected (again) for reproduction (see Section 10.3.14).
Table 34 Subaddress 16 (notes 1 and 2)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 lower error limit value
6B6 5B5 4B4 3B3 2B2 1B1
0 (LSB) B0
Notes
1. The default setting at power-up is 0001 0100.
2. The lower bit error rate limit subaddress 16 × 1.74 × 10−5.
Table 35 Subaddress 17 (notes 1 and 2)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 upper error limit value
6B6 5B5 4B4 3B3 2B2 1B1
0 (LSB) B0
Notes
1. The default setting at power-up is 0101 0000.
2. The upper bit error rate limit subaddress 17 × 1.74 × 10−5.
10.3.14 NICAM UPPER ERROR LIMIT REGISTER When the auto-mute function is enabled (bit AMUTE in the
NICAM configuration register) and the NICAM bit error countishigherthanthevaluecontainedin this register, the signalofthefirstsound carrier (i.e. FM monoorAM sound) or the external mono input (depending on bit AMSEL and ADC output selection) is selected for reproduction.
The difference between upper and lower error limit constitutes a hysteresis to avoid frequent switching between NICAM and the program on the first sound carrier.
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.15 AUDIO MUTE CONTROL REGISTER When any of these bits are set to logic 1, the corresponding pair of output channels will be muted. A bit set to logic 0

allows normal signal output. There is a soft mute facility for the Main and Auxiliary output channels to provide click-free muting independent of the
volume control. This is switched on/off by bits MUTMAIN and MUTAUX.
Table 36 Subaddress 18 (note 1)
BIT NAME VALUE DESCRIPTION
2
7 (MSB) MUTI
6 MUTI
5 MUTDAC 1 mute internal DAC
4 MUTLINE 1 mute line outputs
3 MUTSC2 1 mute SCART 2 outputs
2 MUTSC1 1 mute SCART 1 outputs
1 MUTAUX 1 mute Auxiliary outputs
0 (LSB) MUTMAIN 1 mute Main outputs
S2 1 mute I2S2 output
0 normal I
2
S1 1 mute I2S1 output
0 normal I
2
S2 output
2
S1 output
0 normal internal DAC
0 normal line outputs
0 normal SCART 2 outputs
0 normal SCART 1 outputs
0 normal Auxiliary outputs
0 normal Main outputs
Note
1. The default setting at power-up is 1111 1111.
10.3.16 DAC OUTPUT SELECT REGISTER This register is used to define both the signal source to be entered into the DAC and the mode of the digital matrix for
signal selection. The DAC is used for signal output from digital sources at analog outputs. The bits DACGAIN1 and DACGAIN2 can introduce some extra gain at the input to the DAC; DACGAIN1 adds 3 dB and
DACGAIN2 adds 6 dB of gain, respectively. The two combinations of FM and NICAM apply to the (rare) condition that three different languages are being broadcast
in an FM + NICAM system. They allow for a two-out-of-three selection for external use, such as recording.
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Digital TV Sound Processor (DTVSP) TDA9875A
Table 37 Subaddress 19 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) DACGAIN2 extra gain setting (see Table 38)
6B6DAC output selection (seeTable 39) 5B5 4B4 3 DACGAIN1 extra gain setting (see Table 38) 2B2signal source selection (see Table 40) 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 38 Extra gain setting
B7 B3 GAIN (dB)
00 0 01 3 10 6 11 9
Table 39 DAC output selection
B6 B5 B4 L OUTPUT R OUTPUT
0 0 0 L input R input 0 0 1 L input L input 0 1 0 R input R input 0 1 1 R input L input 100
Table 40 Signal source selection
B2 B1 B0
0 0 0 FM left FM right 0 0 1 NICAM left NICAM right 010 I 011 I 1 0 0 ADC left ADC right 1 0 1 AVL left AVL right 1 1 0 FM mono NICAM M1 1 1 1 FM mono NICAM M2
LR+
------------- ­2
SIGNAL SOURCE
LEFT RIGHT
2
S1 left I2S1 right
2
S2 left I2S2 right
LR+
------------- ­2
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.17 SCART 1 OUTPUT SELECT REGISTER This register is used to define both the signal source to be output at SCART 1 and the output channel selector mode.

Table 41 Subaddress 20 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 default value
6 SC1GAIN 1 Activates the 3 dB gain stage at the SCART 1 output buffers.As any SCART input
passes a 3 dB attenuator, this gain stage can be used to compensate that attenuation, resulting in a 0 dB insertion loss when copying from SCART 2 input to SCART 1 output. However,that gain must be used with great care, as it will cause signal clipping at high input levels.
0 the audio signal output will be unchanged (0 dB gain) 5B5output channel selection (see Table 42) 4B4 3 B3 0 default value 2B2signal source selection (see Table 43) 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0001.
Table 42 Output channel selection
B5 B4 L OUTPUT R OUTPUT
0 0 L input R input 0 1 L input L input 1 0 R input R input 1 1 R input L input
Table 43 Signal source selection
B2 B1 B0 SIGNAL SOURCE
0 0 0 SCART 1 input 0 0 1 SCART 2 input 0 1 0 external input 0 1 1 mono input 1 0 0 DAC input
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.18 SCART 2 OUTPUT SELECT REGISTER This register is used to define both the signal source to be output at SCART 2 and the output channel selector mode.

Table 44 Subaddress 21 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0
6 SC2GAIN 1 Activates the 3 dB gain stage at the SCART 2 output buffers.As any SCART input
passes a 3 dB attenuator, this gain stage can be used to compensate that attenuation, resulting in a 0 dB insertion loss when copying from SCART 1 input to SCART 2 output. However,that gain must be used with great care, as it will cause signal clipping at high input levels.
0 the audio signal output will be output (0 dB gain) 5B5output channel selection (see Table 45) 4B4 3 B3 0 default value 2B2signal source selection (see Table 46) 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 45 Output channel selection
B5 B4 L OUTPUT R OUTPUT
0 0 L input R input 0 1 L input L input 1 0 R input R input 1 1 R input L input
Table 46 Signal source selection
B2 B1 B0 SIGNAL SOURCE
0 0 0 SCART 1 input 0 0 1 SCART 2 input 0 1 0 external input 0 1 1 mono input 1 0 0 DAC input
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10.3.19 LINE OUTPUT SELECT REGISTER By definition, the line output conveys the same signal as the Main (loudspeaker) channel, but in a non-processed form.

This register is used to characterize the signal to be output at the line output and define the output channel selector mode.
Table 47 Subaddress 22 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 set to logic 0
6 LINGAIN 1 activates the 3 dB gain stage at the line output buffers
0 audio signal will be output unchanged (0 dB gain) 5B5output channel selection (see Table 48) 4B4 3 B3 0 set to logic 0 2 B2 0 set to logic 0 1 B1 0 set to logic 0
0 (LSB) LINSEL 1 A signal from an analog source is being processed in the Main channel for line
output. Analog signal sources comprise SCART 1 input, SCART 2 input, external input and mono input, i.e. any input to the ADC.
0 A signal from a digital source is being processed in the Main channel for line
output. Digital signal sources comprise FM, NICAM, I
2
S1 input and I2S2 input.
Note
1. The default setting at power-up is 0000 0000.
Table 48 Output channel selection
B5 B4 L OUTPUT R OUTPUT
0 0 L input Rinput 0 1 L input L input 1 0 R input R input 1 1 R input L input
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10.3.20 ADC OUTPUT SELECT REGISTER This register is used to define the signal source for the ADC. There is no output channel selector, because all digital

signal sinks of the ADC have their own matrix. Instead, a level adjustment facility for the ADC output is provided.
Table 49 Subaddress 23 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 signal source selection (see Table 50)
6B6 5B5 4B4ADC level adjust (see Table 51) 3B3 2B2 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 50 Signal source selection
B7 B6 B5 SIGNAL SOURCE
0 0 0 SCART 1 input 0 0 1 SCART 2 input 0 1 0 external input 0 1 1 mono input
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Table 51 ADC level adjust (note 1)
B4 B3 B2 B1 B0 GAIN SETTING (dB)
01111 +15 01110 +14 01101 +13 01100 +12 01011 +11 01010 +10 01001 +9 01000 +8 00111 +7 00110 +6 00101 +5 00100 +4 00011 +3 00010 +2 00001 +1 00000 0 11111 1 11110 2 11101 3 11100 4 11011 5 11010 6 11001 7 11000 8 10111 9 10110 10 10101 11 10100 12 10011 13 10010 14 10001 15 10000 mute
Note
1. If the ADC level adjust is set to 0 dB a full-scale input signal to the ADC results into a full-scale level of 6dBatthe digital crossbar.
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.21 MAIN CHANNEL SELECT REGISTER

This register is used to define both the signal source to be processed in the Main (loudspeaker) channel and the mode of the digital matrix for signal selection.
Table 52 Subaddress 24 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 default value
6B6output channel selection (see Table 53) 5B5 4B4 3 B3 0 default value 2B2signal source selection (see Table 54) 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 53 Output channel selection
B6 B5 B4 L OUTPUT R OUTPUT
0 0 0 L input R input 0 0 1 L input L input 0 1 0 R input R input 0 1 1 R input L input 100
Table 54 Signal source selection
B2 B1 B0 SIGNAL SOURCE
0 0 0 FM input 0 0 1 NICAM input 010 I 011 I 1 0 0 ADC input
LR+
------------- ­2
2
S1 input
2
S2 input
LR+
------------- ­2
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.22 AUDIO EFFECTS REGISTER (MAIN) Switching the AVL off will reset the associated hardware to a defined state. When the signal source for the Main channel

ischangedwhiletheAVLis on, the AVL needs to be reset in order to avoid excessive settling times. This can be achieved by switching the AVL off and on again.
The pseudo stereo function is based on an all-pass filter. A 90 degrees phase shift occurs at the frequencies stated in Table 57. There is a gain of 3 dB in the left audio channel.
Table 55 Subaddress 25 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 Default value.
6 B6 0 Default value. 5 SPATIAL1 These bits set the amount of the effect function (stereo base width expansion) 4 SPATIAL0
3 PSEUDO1 These bits set the amount of the effect function (pseudo stereo) for mono 2 PSEUDO0
1 AVL1 These bits set the mode of operation of the automatic volume level control
0 (LSB) AVL0
for stereo signals in the Main channel (see Table 56). This function should be activated only in accordance with the result of the sound mode identification.
signals in the Main channel (see Table 57). This function should be activated only in accordance with the result of the sound mode identification.
function at the entrance to the Main (loudspeaker) channel (see Table 58).
Note
1. The default setting at power-up is 0000 0000.
Table 56 Spatial control setting
B5 B4 SPATIAL SETTING (%)
0 0 off 01 30 10 40 11 52
Table 57 Pseudo control setting
B3 B2 PSEUDO SETTING (Hz)
0 0 off 0 1 300 1 0 200 1 1 150
Table 58 AVL control mode
B1 B0 AVL MODE
0 0 off/reset 0 1 short decay 1 0 medium decay 1 1 long decay
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10.3.23 VOLUME CONTROL REGISTERS (MAIN) These two registers control the volume setting of the Main (loudspeaker) channel. The register at subaddress 26 applies

to the left channel signal, while the register at subaddress 27 applies to the right channel signal. Balance control is exercised by offsetting the left and right channel volume settings.
Table 59 Subaddresses 26 and 27
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00011000 +24 00010111 +23 00010110 +22 00010101 +21 00010100 +20 00010011 +19 00010010 +18 00010001 +17 00010000 +16 00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0 11111111 1 11111110 2 11111101 3 11111100 4 11111011 5 11111010 6 11111001 7 11111000 8 11110111 9
VOLUME SETTING (dB)
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MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
11110110 10 11110101 11 11110100 12 11110011 13 11110010 14 11110001 15 11110000 16 11101111 17 11101110 18 11101101 19 11101100 20 11101011 21 11101010 22 11101001 23 11101000 24 11100111 25 11100110 26 11100101 27 11100100 28 11100011 29 11100010 30 11100001 31 11100000 32 11011111 33 11011110 34 11011101 35 11011100 36 11011011 37 11011010 38 11011001 39 11011000 40 11010111 41 11010110 42 11010101 43 11010100 44 11010011 45 11010010 46 11010001 47 11010000 48
VOLUME SETTING (dB)
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Digital TV Sound Processor (DTVSP) TDA9875A
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
11001111 49 11001110 50 11001101 51 11001100 52 11001011 53 11001010 54 11001001 55 11001000 56 11000111 57 11000110 58 11000101 59 11000100 60 11000011 61 11000010 62 11000001 63 11000000 64 10111111 65 10111110 66 10111101 67 10111100 68 10111011 69 10111010 70 10111001 71 10111000 72 10110111 73 10110110 74 10110101 75 10110100 76 10110011 77 10110010 78 10110001 79 10110000 80 10101111 81 10101110 82 10101101 83 10101100 mute (note 1)
VOLUME SETTING (dB)
Note
1. The default setting at power-up is 1010 1100.
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10.3.24 CONTOUR CONTROL REGISTER (MAIN) This register is used to apply the contour or loudness function (physiological volume control) to the left and right signal

channels of the Main channel by means of an extra bass boost. The gain setting must be chosen in accordance with the volume control setting for the Main channel. For example, the contour gain could be incremented for every 5 dB, or so, of decrease of the volume setting. This needs to be done by the microcontroller. The 0 dB contour setting is equal to contour off.
Table 60 Subaddress 28
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00010010 18 00010001 17 00010000 16 00001111 15 00001110 14 00001101 13 00001100 12 00001011 11 00001010 10 00001001 9 00001000 8 00000111 7 00000110 6 00000101 5 00000100 4 00000011 3 00000010 2 00000001 1 00000000 0(note 1)
CONTOUR GAIN (dB)
Note
1. The default setting at power-up is 0000 0000.
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10.3.25 BASS CONTROL REGISTER (MAIN) This register is used to apply bass control to the left and right signal channels of the Main channel.

Table 61 Subaddress 29
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12
BASS SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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10.3.26 TREBLE CONTROL REGISTER (MAIN) This register is used to apply treble control to the left and right signal channels of the Main channel.

Table 62 Subaddress 30
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12
TREBLE SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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10.3.27 AUXILIARY CHANNEL SELECT REGISTER This register is used to define both the signal source to be processed in the Auxiliary (headphone) channel and the mode

of the digital matrix for signal selection.
Table 63 Subaddress 31 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 default value
6B6output channel selection (see Table 64) 5B5 4B4 3 B3 0 default value 2B2signal source selection (see Table 65) 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 64 Output channel selection
B6 B5 B4 L OUTPUT R OUTPUT
0 0 0 L input R input 0 0 1 L input L input 0 1 0 R input R input 0 1 1 R input L input 100
Table 65 Signal source selection
B2 B1 B0 SIGNAL SOURCE
0 0 0 FM input 0 0 1 NICAM input 010 I 011 I 1 0 0 ADC input 1 0 1 AVL input
LR+
------------- ­2
2
S1 input
2
S2 input
LR+
------------- ­2
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10.3.28 VOLUME CONTROL REGISTERS (AUXILIARY) These two registers control the volume setting of the Auxiliary (headphone) channel. The register at subaddress 32

applies to the left channel signal, while the register at subaddress 33 applies to the right channel signal. Balance control is exercised by offsetting the left and right channel volume settings.
Table 66 Subaddresses 32 and 33
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00011000 +24 00010111 +23 00010110 +22 00010101 +21 00010100 +20 00010011 +19 00010010 +18 00010001 +17 00010000 +16 00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0 11111111 1 11111110 2 11111101 3 11111100 4 11111011 5 11111010 6 11111001 7 11111000 8 11110111 9 11110110 10
VOLUME SETTING (dB)
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MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
11110101 11 11110100 12 11110011 13 11110010 14 11110001 15 11110000 16 11101111 17 11101110 18 11101101 19 11101100 20 11101011 21 11101010 22 11101001 23 11101000 24 11100111 25 11100110 26 11100101 27 11100100 28 11100011 29 11100010 30 11100001 31 11100000 32 11011111 33 11011110 34 11011101 35 11011100 36 11011011 37 11011010 38 11011001 39 11011000 40 11010111 41 11010110 42 11010101 43 11010100 44 11010011 45 11010010 46 11010001 47 11010000 48 11001111 49 11001110 50 11001101 51
VOLUME SETTING (dB)
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Digital TV Sound Processor (DTVSP) TDA9875A
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
11001100 52 11001011 53 11001010 54 11001001 55 11001000 56 11000111 57 11000110 58 11000101 59 11000100 60 11000011 61 11000010 62 11000001 63 11000000 64 10111111 65 10111110 66 10111101 67 10111100 68 10111011 69 10111010 70 10111001 71 10111000 72 10110111 73 10110110 74 10110101 75 10110100 76 10110011 77 10110010 78 10110001 79 10110000 80 10101111 81 10101110 82 10101101 83 10101100 mute (note 1)
VOLUME SETTING (dB)
Note
1. The default setting at power-up is 1010 1100.
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10.3.29 BASS CONTROL REGISTER (AUXILIARY) This register is used to apply bass control to the left and right signal channels of the Auxiliary channel.

Table 67 Subaddress 34
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12
BASS SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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10.3.30 TREBLE CONTROL REGISTER (AUXILIARY) This register is used to apply treble control to the left and right signal channels of the Auxiliary channel.

Table 68 Subaddress 35
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
XXX01100 +12 XXX01011 +11 XXX01010 +10 XXX01001 +9 XXX01000 +8 XXX00111 +7 XXX00110 +6 XXX00101 +5 XXX00100 +4 XXX00011 +3 XXX00010 +2 XXX00001 +1 XXX00000 0(note 1) XXX11111 1 XXX11110 2 XXX11101 3 XXX11100 4 XXX11011 5 XXX11010 6 XXX11001 7 XXX11000 8 XXX10111 9 XXX10110 10 XXX10101 11 XXX10100 12
TREBLE SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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10.3.31 FEATURE INTERFACE CONFIGURATION REGISTER

Table 69 Subaddress 36 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 default value
6 B6 0 default value 5 B5 0 default value 4 SYSCL1 system clock frequency selection (see Table70) 3 SYSCL0 2 SYSOUT 1 enables the output of a system (or master) clock signal at pin SYSCLK
0 the output will be off, thereby improving the EMC performance
1I
0 (LSB) I
2
SFORM 1 an MSB-aligned (MSB-first) serial output format is selected, i.e. a level change at
pin WS indicates the beginning of a new audio sample
0 the standard I
2
SOUT 1 enables the I2S-bus outputs (both serial data outputs plus serial bit clock and word
2
S-bus output format is selected
select) in a format determined by bit I2SFORM; the TDA9875A is then an I2S-bus master
0 the outputs mentioned will be 3-stated, thereby improving the EMC performance
Note
1. The default setting at power-up is 0000 0000.
Table 70 System clock frequency selection
B4 B3 SYSCLK OUTPUT FREQUENCY (MHz)
0 0 256f 0 1 384f 1 0 512f 1 1 768f
Note
1. With 16.384 MHz the duty cycle is 33%.
s s s s
8.192
12.288
16.384
24.576
(1)
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10.3.32 I2S1 OUTPUT SELECT REGISTER This register is used to define both the signal source to be output at I2S1 and the mode of the digital matrix for signal

selection.
Table 71 Subaddress 37 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 default value
6B6output selection (see Table 72) 5B5 4B4 3 B3 0 default value 2B2signal source selection (see Table 73) 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 72 Output selection
B6 B5 B4 L OUTPUT R OUTPUT
0 0 0 L input R input 0 0 1 L input L input 0 1 0 R input R input 0 1 1 R input L input 100
Table 73 Signal source selection (note 1)
B2 B1 B0 SIGNAL SOURCE
0 0 0 FM output 0 0 1 NICAM output 010 I 011 I 1 0 0 ADC output 1 0 1 AVL output 1 1 0 Auxiliary output 1 1 1 Main output
LR+
------------- ­2
2
S1 input
2
S2 input
LR+
------------- ­2
Note
1. The Main and Auxiliary channel outputs will not contain the beeper signal.
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10.3.33 I2S1 INPUT LEVEL ADJUST REGISTER This register is used to adjust the input level at the I2S1 interface. Left and right signal channel are treated identically.

Table 74 Subaddress 38
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12 00010011 13 00010010 14 00010001 15 00010000 mute
GAIN SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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10.3.34 I2S1 OUTPUT LEVEL ADJUST REGISTER This register is used to adjust the output level at the I2S1 interface. Left and right signal channel are treated identically.

Table 75 Subaddress 39
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12 00010011 13 00010010 14 00010001 15 00010000 mute
GAIN SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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10.3.35 I2S2 OUTPUT SELECT REGISTER This register is used to define both the signal source to be output at I2S2 and the mode of the digital matrix for signal

selection.
Table 76 Subaddress 40 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 0 default value
6B6output selection (see Table 77) 5B5 4B4 3 B3 0 default value 2B2signal source selection (see Table 78) 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 77 Output selection
B6 B5 B4 L OUTPUT R OUTPUT
0 0 0 L input R input 0 0 1 L input L input 0 1 0 R input R input 0 1 1 R input L input 100
Table 78 Signal source selection (note 1)
B2 B1 B0 SIGNAL SOURCE
0 0 0 FM output 0 0 1 NICAM output 010 I 011 I 1 0 0 ADC output 1 0 1 AVL output 1 1 0 Auxiliary output 1 1 1 Main output
LR+
------------- ­2
2
S1 input
2
S2 input
LR+
------------- ­2
Note
1. The Main and Auxiliary channel outputs will not contain the beeper signal.
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Digital TV Sound Processor (DTVSP) TDA9875A

10.3.36 I2S2 INPUT LEVEL ADJUST REGISTER This register is used to adjust the input level at the I2S2 interface. Left and right signal channel are treated identically.

Table 79 Subaddress 41
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12 00010011 13 00010010 14 00010001 15 00010000 mute
GAIN SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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10.3.37 I2S2 OUTPUT LEVEL ADJUST REGISTER This register is used to adjust the output level at the I2S2 interface. Left and right signal channel are treated identically.

Table 80 Subaddress 42
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00001111 +15 00001110 +14 00001101 +13 00001100 +12 00001011 +11 00001010 +10 00001001 +9 00001000 +8 00000111 +7 00000110 +6 00000101 +5 00000100 +4 00000011 +3 00000010 +2 00000001 +1 00000000 0(note 1) 00011111 1 00011110 2 00011101 3 00011100 4 00011011 5 00011010 6 00011001 7 00011000 8 00010111 9 00010110 10 00010101 11 00010100 12 00010011 13 00010010 14 00010001 15 00010000 mute
GAIN SETTING (dB)
Note
1. The default setting at power-up is 0000 0000.
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10.3.38 BEEPER FREQUENCY CONTROL REGISTER This register is used to select from sample beeper oscillator frequencies. The beeper output signal is added to the Main

and Auxiliary channel output DAC. Due to the frequency response of the audio DACs upsampling filters, the 25 kHz beep is approximately 5 dB louder than
the 390 Hz beep.
Table 81 Subaddress 43 (note 1)
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00000111 25000 00000110 7040 00000101 3580 00000100 1770 00000011 1270 00000010 900 00000001 640 00000000 390
Note
1. The default setting at power-up is 0000 0000.
10.3.39 BEEPER VOLUME CONTROL REGISTER This register is used to set the beeper volume. The gain setting is relative to digital full-scale at the input to the Main and
Auxiliary channel output DACs. The beeper volume is independent of any other volume setting. The beeper signal is added to the Main and Auxiliary channel output signals in the 2 × fs domain. The beeper volume
should be set with great care, when the audio signals in the Main and Auxiliary channels are close to digital full-scale, to avoid output signal distortion due to overload.
GENERATED FREQUENCY (Hz)
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Table 82 Subaddress 44
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00000000 0 00111111 3 00111110 6 00111101 9 00111100 12 00111011 15 00111010 18 00111001 21 00111000 24 00110111 27 00110110 30 00110101 33 00110100 36 00110011 39 00110010 42 00110001 45 00110000 48 00101111 51 00101110 54 00101101 57 00101100 60 00101011 63 00101010 66 00101001 69 00101000 72 00100111 75 00100110 78 00100101 81 00100100 84 00100011 87 00100010 90 00100001 93 00100000 mute (note 1)
GAIN SETTING (dB)
Note
1. The default setting at power-up is 0010 0000.
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10.3.40 BASS BOOST CONTROL REGISTER This register is used to select from a few sample bass boost settings to modify the frequency characteristics of the Main

channel (shelving filter). Bits B3 to B0 apply to the left channel, bits B7 to B4 apply to the right channel. This function must be used with care in order to avoid clipping distortion at high volume settings.
More sophisticated control of the bass boost filter can be exercised in the expert mode (see Section 10.5). The user then has full control over this second-order filter and can, within limits, realize bass equalizers with arbitrary centre frequencies, Q factors and boost/cut settings.
Table 83 Subaddress 45 (note 1)
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 gain setting of right channel (see Table 84)
6B6 5B5 4B4 3B3gain setting of left channel (see Table 85) 2B2 1B1
0 (LSB) B0
Note
1. The default setting at power-up is 0000 0000.
Table 84 Gain setting right channel
B7 B6 B5 B4 GAIN SETTING (dB) CORNER FREQUENCY (Hz)
1010 20 350 1001 18 350 1000 16 350 0111 14 350 0110 12 350 0101 10 350 0100 8 350 0011 6 350 0010 4 350 0001 2 350 0000 0 350
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Table 85 Gain setting left channel
B3 B2 B1 B0 GAIN SETTING (dB) CORNER FREQUENCY (Hz)
1010 20 350 1001 18 350 1000 16 350 0111 14 350 0110 12 350 0101 10 350 0100 8 350 0011 6 350 0010 4 350 0001 2 350 0000 0 350
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Digital TV Sound Processor (DTVSP) TDA9875A

10.4 Slave transmitter mode

As a slave transmitter, the TDA9875A provides 13 registers with status information and data, a part of which is for Philips internal purposes only. These registers can be accessed by means of subaddresses.
Table 86 General format for reading data from the TDA9875A
S SLAVE ADDRESS 0 ACK SUBADDRESS ACK Sr SLAVE ADDRESS 1 ACK DATA NAm P
Table 87 Explanation of Tables 86 and 88
BIT FUNCTION
S START condition SLAVE ADDRESS 7-bit device address 0 data direction bit (write to device) ACK acknowledge (by the slave) SUBADDRESS address of register to read from Sr repeated START condition 1 data direction bit (read from device) DATA data byte read from register NAm not acknowledge (by the master) Am acknowledge (by the master) P STOP condition
Reading of data can start at any valid subaddress. It is allowed to read more than 1 data byte per transmission from the TDA9875A. In this situation, the subaddress is automatically incremented after each data byte, which results in reading the sequence of data bytes from successive register locations, starting at SUBADDRESS.
Table 88 Format of a transmission using automatic incrementing of subaddresses
S SLAVE ADDRESS 0 ACK SUBADDRESS ACK Sr SLAVE
ADDRESS
Note
1. n data bytes with auto-increment of subaddresses.
Each data byte in a read sequence, except for the last one, is acknowledged with Am (acknowledge by the master). The subaddresses ‘wrap around’ from decimal 255 to 0. If an attempt is made to read from a non-existing subaddress, the device will send a data pattern of all ones, i.e. FF in hexadecimal notation.
1 ACK DATA BYTE
Am
(1)
DATA NAm P
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Table 89 Overview of the slave transmitter registers (note 1)
SUBADDRESS
(DECIMAL)
0 ssssssssdevice status (power-on, identification, etc.) 1 ssssssssNICAM status 2 eeeeeeeeNICAM error count 3 ddddddddadditional data (LSB) 4 c c X c c d d d additional data (MSB) 5 lllllllllevel read-out (MSB) 6 lllllllllevel read-out (LSB)
7 XXXcccccSIF level 251 aaaaaaaatest register 3; note 2 252 aaaaaaaatest register 2; note 2 253 aaaaaaaatest register 1; note 2 254 dddddddddevice identification code 255 sssssssssoftware identification code
Notes
1. X indicates a bit that has not been assigned to a function. This bit is reserved for future extensions.
2. Registers from subaddress 251 to 255 are for Philips internal purposes only. They are considered as a set of registers for the identification of individual members and some key parameters in a family of devices.
MSB LSB
DATA
FUNCTION
The following sub-sections provide a detailed description of the slave transmitter registers.
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10.4.1 DEVICE STATUS REGISTER
Table 90 Subaddress 0
BIT NAME VALUE DESCRIPTION
7 (MSB) P2IN This bit reflects the status of the corresponding general purpose port of pin P2
(see Section 10.3.2).
6 P1IN This bit reflects the status of the corresponding general purpose port of pin P1
(see Section 10.3.2).
5 RSSF 1 Reserve sound switching flag: this bit is a copy of the C4 bit in the NICAM status
register. It indicates that the FM (or AM for standard L) sound matches the digital transmission and auto-muting should be enabled.
0 Auto-muting should be disabled, as analog and digital sound are different.
4 AMSTAT 1 Auto-mute status: it indicates that the auto-muting function has switched from
NICAM to the program of the first sound carrier (i.e. FM mono or AM in the NICAM L system) or to the ADC (depending on bit AMSEL).
0 Auto-muting function has not switched.
3 VDSP 1 Indicates that digital transmission is a sound source (NICAM).
0 The transmission is either data or currently undefined format (NICAM).
2 IDDUA This bit is logic 1 if an FM dual-language signal has been identified. When
neither IDSTE nor IDDUA are set, the received signal has to be assumed to be FM mono.
1 IDSTE This bit is logic 1 if an FM stereo signal has been identified.
0 (LSB) POR Power fail bit: the power supply for the digital part of the device, V
temporarily been lower than the specified lower limit. If this is detected an initialization of the TDA9875A has to be carried out to ensure a reliable operation.
DDD2
, has
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Digital TV Sound Processor (DTVSP) TDA9875A
10.4.2 NICAM STATUS REGISTER
The TDA9875A does not support the Extended Control Modes. Therefore, the program of the first sound carrier (i.e. FM mono or AM) is selected for reproduction in case bit C3 is set to logic 1, independent of bit AMUTE in the NICAM configuration register being set or not.
When a NICAM transmitter is switched off, the device will lose synchronization. In this situation the program of the first sound carrier is selected for reproduction, independent of bit AMUTE being set or not.
Table 91 Subaddress 1
BIT NAME VALUE DESCRIPTION
7 (MSB) C4 application control bits (C1 to C4 in the NICAM transmission)
0 the audio output from the NICAM part should be digital silence
2 CFC 1 indication of a configuration change at the 16 frame (C0) boundary
0 no configuration change
1 S/MB 1 indication of NICAM stereo mode
0 no NICAM stereo mode
0 (LSB) D/SB 1 indication NICAM dual mono mode
0 no NICAM dual mono mode
10.4.3 NICAN ERROR COUNT REGISTER
BitsB7 to B0 contain the number of errors occurring in the previous 128 ms period. The registeris updated every 128 ms.
Table 92 Subaddress 2
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 number of errors
6B6 5B5 4B4 3B3 2B2 1B1
0 (LSB) B0
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10.4.4 ADDITIONAL DATA REGISTERS
These two bytes provide information on the additional data bits.
Table 93 Subaddress 3
BIT NAME VALUE DESCRIPTION
7 (MSB) AD7 comprise the additional data word
6 AD6 5 AD5 4 AD4 3 AD3 2 AD2 1 AD1
0 (LSB) AD0
Table 94 Subaddress 4
BIT NAME VALUE DESCRIPTION
7 (MSB) OVW 1 new additional data bits are written to the IC without the previous bits being
read
0 no bits are written
6 SAD 1 new additional data is written into the IC
0 this bit is set to logic 0 when the additional data bits are read 5Xdon’t care 4 CI1 these are CI bits decoded by majority logic from the parity checks of the last 3 CI2 2 AD10 comprise the additional data word 1 AD9
0 (LSB) AD8
ten samples in a frame
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10.4.5 LEVEL READ-OUT REGISTERS These two bytes constitute a word that provides data from
a location that has been specified with the monitor select register. The most significant byte of the data is stored at subaddress 5.
If peak-level monitoring has been selected, the peak-level monitoring register is cleared and monitoring resumes after its contents has been transferred to these two bytes.
Table 95 Subaddresses 5 and 6
SUB-
ADDRESS
5 7 (MSB) most significant bit or sign bit
6 7 (MSB)
10.4.6 SIF LEVEL REGISTER When the SIF AGC is on, bits B4 to B0 of this register
contain a number that gives an indication of the SIF input level. That number corresponds to the AGC gain register setting (see Section 10.3.1).
BIT DESCRIPTION
6 5 4 3 2 1
0 (LSB)
6 5 4 3 2 1
0 (LSB) least significant bit
Table 96 Subaddress 7
BIT NAME VALUE DESCRIPTION
7 (MSB) B7 X bit not assigned
6 B6 X bit not assigned 5 B5 X bit not assigned 4B4indication of SIF 3B3 2B2 1B1
0 (LSB) B0
10.4.7 TEST REGISTER 3 This register contains, as a binary number, the highest
memory address used for the Coefficient RAM (CRAM, expert mode).
Table 97 Subaddress 251
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
01111111
10.4.8 T This register contains, as a binary number, the highest
subaddress used for slave receiver registers.
Table 98 Subaddress 252
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00101101
EST REGISTER 2
input level
When the SIF AGC is off, this register returns the contents of the AGC gain register.
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10.4.9 TEST REGISTER 1 This register contains, as a binary number, the highest
subaddress used for slave transmitter (status) registers.
Table 99 Subaddress 253
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00000111
10.4.10 D There will be several devices in the digital TV sound
processorfamily.Thisbyteisusedtoidentifytheindividual family members.
Table 100 Subaddress 254
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00000010
10.4.11 S It is likely that during the life time of this family of devices
several versions of the DSP software will be made, e.g., to accommodate new application concepts, respond to customer wishes, etc. This byte is used to identify the different releases.
Table 101 Subaddress 255
MSB LSB
B7 B6 B5 B4 B3 B2 B1 B0
00000010

10.5 Expert mode

In addition to the slave receiver and slave transmitter modes previously described, there is a special ‘expert’ mode that gives direct write access to the internal CRAM of the DSP.
In this mode, transferred data contain 12-bit coefficients. As these coefficients bypass on-chip coefficient look-up tables for many functions, they directly influence the processing of signals within the DSP.
EVICE IDENTIFICATION CODE
OFTWARE IDENTIFICATION CODE
As the coefficients do not fit into one data byte, they have to be split and arranged (see Table 104). The most significant bit is transferred first.
The general format described in Table 104 shows the minimum number of data bytes required, i.e. two bytes for the transfer of a single coefficient.
Should more than one coefficient be sent, then the CRAM address will be automatically incremented after each coefficient, resulting in writing the sequence of coefficients into successive memory locations, starting at CRAM ADDRESS. A transmission can start with any valid CRAM address. If two coefficients are to be transferred, they are arranged as shown in Table 105.
With any odd number of coefficients to be transferred, the least significant nibble of the last byte is regarded as containing don’t care data.
As the transfer of coefficients cannot be accomplished within one audio sample period, it is necessary that receivedcoefficientsbebufferedandmade active all at the same time to avoid audio signal transients. The receive buffer is designed to store up to 8 coefficients in addition to the CRAM address. Each byte that fits into the buffer is acknowledged with ACK (acknowledge). If an attempt is made to write more coefficients than the buffer can store, the device acknowledges with NACK (not acknowledge) and any further coefficients are ignored. Coefficients that are already in the receive buffer remain intact.
An expert mode transfer ends when the I2C-bus STOP condition or a repeated START condition has been detected. Only those coefficients that have been received during the last transmission will then be copied from the buffer to the CRAM.
To make efficient and correct use of the expert mode, it is recommended to transfer all coefficients for any one function in a single transmission.
There is no checking of memory addresses and the automatic incrementing of addresses does not stop at the highestusedCRAMaddress.Theuserofthisexpertmode must be fully acquainted with the relevant procedures.
More information concerning the functions of this device, such as the number of coefficients per function, their default values, memory addresses, etc., can be supplied on request at a later date.
This mode must be used with great care. It can be used to create user-defined characteristics, such as a tone control with different corner frequencies or special boost/cut characteristics to correct the low-frequency loudspeaker and/or cabinet frequency responses.
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Table 102 General format for entering the expert mode and writing coefficients into the TDA9875A
S SLAVE
ADDRESS
Table 103 Explanation of Table 102
BIT FUNCTION
S START condition SLAVE ADDRESS 7-bit device address 0 data direction bit (write to device) ACK acknowledge 10000000 pattern to enter the expert mode CRAM ADDRESS start address of coefficient RAM to write to DATA data byte containing part of a coefficient P STOP condition
Table 104 General format (notes 1, 2 and 3)
BYTE DATA DESCRIPTION
1 data byte aaaaaaaa2MST of 1st coefficient 2 data byte aaaaXXXX1LST of 1st coefficient
Notes
1. X = don’t care.
2. MST = most significant third.
3. LST = least significant third.
0 ACK 10000000 ACK CRAM
ADDRESS
ACK DATA ACK DATA ACK P
Table 105 Transfer of two coefficients
BYTE DATA DESCRIPTION
1 data byte aaaaaaaa2MST of 1st coefficient 2 data byte aaaabbbb1LST of 1st coefficient + 1 MST of 2nd coefficient 3 data byte bbbbbbbb2LST of 2nd coefficient
1999 Dec 20 86
Page 87
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

11 I2S-BUS DESCRIPTION

The feature interface of the TDA9875A contains two serial audio inputs and outputs and associated clock signals. It can be used to supply, for example, audio signals from received TV programs to a digital audio output device (AES/EBU format), or import serial audio signals from other sources for reproduction through the TV set’s loudspeakerand/orheadphonechannels.Apartfromsuch simple data input or output, it is also possible to run audio signals through an external DSP, which performs some additional functions, such as room simulation, Dolby Surround Pro Logic etc. and feed those signals back into the loudspeaker and/or headphone channels of the TDA9875A.
Two serial audio formats are supported at the feature interface, i.e. the I2S-bus format and a very similar MSB-aligned format. The difference is illustrated in Fig.9.
In both formats the left audio channel of a stereo sample pair is output first and is placed on the serial data line (SDI for input, SDO for output) when the Word Select line (WS) is LOW. Data is written at the trailing edge of SCK and read at the leading edge of SCK. The most significant bit is sent first.
At power-up, the outputs of the feature interface are 3-stated to reduce EMC and allow for combinations with other ICs. If output is desired, it has to be activated by means of an I2C-bus command.
When the output is enabled, the serial audio data can be taken from pins SDO1 and SDO2. Depending on the signal source, switch and matrix positions, the output can be either mono, stereo or dual language sound on either output.
Apart from just feeding a digital audio device, such as a DAC or an AES/EBU transmitter, the serial data outputs can be connected directly to the serial inputs (loop-back connection) or first to an external device, e.g. a feature DSP such as the SAA7710 and then back to the serial inputs. In all of these configurations, the SCK and WS clocks will be generated by the TDA9875A, which then is
2
the I
S-bus master.
The serial data inputs, SDI1 and SDI2, are active at all times, independent of the serial data outputs being on or off. When the serial data outputs are off (either after power-up or via the appropriate I2C-bus command) serial dataandclocksWS and SCK from a separate digital audio source can be fed into the TDA9875A, be processed and output in accordance with internal selector positions, provided that the following criteria are met:
32 kHz audio sample frequency
32 clock bits per sample
External timing and data synchronized to TDA9875A.
In such cases, the external source is the I2S-bus master and the TDA9875A is the I2S-bus slave.
To support synchronization of external devices or as a master clock for them, a system clock output, SYSCLK, is available from the TDA9875A. At power-up it is off. It can be enabled and the output frequency set via an I2C-bus command. Available output frequencies are
8.192, 12.288, 16.384 and 24.576 MHz.
The word select output is clocked with the audio sample frequency at 32 kHz. The serial clock output (SCK) is clocked at a frequency of 2.048 MHz. This means, that there are 64 clock pulses per pair of stereo output samples, or 32 clock pulses per sample. Depending again on the signal source, the number of significant bits on the serial data outputs, SDO1 and SDO2, is between 14 and 18.
1999 Dec 20 87
Page 88
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
handbook, full pagewidth
SCK
WS
SD LSB MSB
handbook, full pagewidth
SCK
WS
SD LSB MSB
LSB MSB
MGK112
one sample
a. I2S-bus format.
LSB MSB
MGK113
one sample
b. MSB-aligned format.
Fig.9 Serial audio interface formats.
1999 Dec 20 88
Page 89
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

12 APPLICATION INFORMATION

handbook, full pagewidth
SIFSAT
SIFTV
+5 V
R2
1.5
C1
4.7 µF
C2
47 pF
C4
47 pF
C5 47 µF
24.576 MHz
C7
470 nF
C8
470 nF
1 µF
C9
470 nF
R1
10 k
C3
100 nF
C6
PCLK
ADDR1
SCL
SDA
V
SSA1
V
DEC1
I
SIF2
V
ref1
SIF1
ADDR2
V
SSD1
V
DDD1
CRESET
V
SSD4
XTALI
XTALO
SYSCLK
SCK
WS
SDO2
SDO1
SDI2
SDI1
TEST1
MONOIN
TEST2
EXTIR
EXTIL
1 (57)
1
2 (58)
2
3 (59)
3
4 (60)
4
5 (61)
5
6 (62)
6
7 (63)
7
ref
8 (64)
8
P1
9 (1)
9
10 (2)
10
11 (3)
11
12 (4)
12
13 (5)
13
14 (6)
14
15 (7)
15
16 (8)
16
17 (9)
17
18 (10)
18
19 (11)
19
P2
20 (12)
20
21 (13)
21
22 (14)
22
23 (15)
23
24 (16)
24
25 (17)
25
26 (18)
26
27 (19)
27
28 (20)
28
29 (21)
29
30 (22)
30
31 (23)
31
32 (24)
32
The pin numbers given in parenthesis refer to the TDA9875AH version.
TDA9875A
(TDA9875AH)
(56) 64
(55) 63
(54) 62
(53) 61
(52) 60
(51) 59
(50) 58
(49) 57
(48) 56
(47) 55
(46) 54
(45) 53
(44) 52
(43) 51
(42) 50
(41) 49
(40) 48
(39) 47
(38) 46
(37) 45
(36) 44
(35) 43
(34) 42
(33) 41
(32) 40
(31) 39
(30) 38
(29) 37
(28) 36
(27) 35
(26) 34
(25) 33
MHB601
64
63
62
61
60
59
58
57
56
55
54
53
52
51
50
49
48
47
46
45
44
43
42
41
40
39
38
37
36
35
34
33
V
DDD2
LORNICAM
LOL
MOL
MOR
V
DDA
AUXOL
AUXOR
V
SSA3
PCAPL
PCAPR
V
ref3
SCOL2
SCOR2
V
SSA4
V
SSD2
SCOL1
SCOR1
V
ref2
i.c.
i.c.
V
SSA2
i.c.
i.c.
V
ref(n)
V
ref(p)
V
DEC2
SCIL2
SCIR2
V
SSD3
SCIL1
SCIR1
C35
C31
C29
C28
C26
C24
R6
15 k
R3
15 k
C23
10 nF
C21
47 µF
C16
47 µF
C15 47 µF
15 k
15 k
R19
47 µF
1.5
2.2 µF
C33
2.2 µF
10 nF
C30
10 nF
2.2 µF
47 µF
10 nF
10 nF
R5
R4
C19
2.2 µF
C17
2.2 µF
R7
270
330 nF
330 nF
2.2 µF
C13
C11
C27
C22
10 nF
2.2
C20
2.2 µF
C18
2.2 µF
C12
330 nF
C10
330 nF
C34
C32
2.2 µF
R8
C25
2.2 µF
C14
4.7 µF
+5 V
+5 V
Fig.10 Schematic for measurements.
1999 Dec 20 89
Page 90
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
handbook, full pagewidth
POWER
0 V
SIFSAT
SIFTV
+5 V
100
C2 470 nF
C3
47 pF
C5
47 pF
R5
1
C8
470 nF
C9
470 nF
+5 V
47 µFC1
R1
R2
100
R4
2.2 k L3
L4
C6 470 nF
24.576 MHz
R6
2.2 k
L6
L7
C10
470 nF
PCLK
1 (57)
2 (58)
ADDR1
SCL
SDA
V
SSA1
V
DEC1
I
SIF2
V
ref1
SIF1
ADDR2
V
SSD1
V
DDD1
CRESET
V
SSD4
XTALI
XTALO
SYSCLK
SCK
WS
SDO2
SDO1
SDI2
SDI1
TEST1
MONOIN
TEST2
EXTIR
EXTIL
3 (59)
4 (60)
5 (61)
6 (62)
7 (63)
ref
8 (64)
P1
9 (1)
10 (2)
11 (3)
12 (4)
13 (5)
14 (6)
15 (7)
16 (8)
17 (9)
18 (10)
19 (11)
P2
20 (12)
21 (13)
22 (14)
23 (15)
24 (16)
25 (17)
26 (18)
27 (19)
28 (20)
29 (21)
30 (22)
31 (23)
32 (24)
L1
L2
R3
10 k
C4
100 nF
L5
C7
1 µF
L8
L1 to L9 are ferrite beads. The pin numbers given in parenthesis refer to the TDA9875AH version.
TDA9875A
(TDA9875AH)
(56) 64
(55) 63
(54) 62
(53) 61
(52) 60
(51) 59
(50) 58
(49) 57
(48) 56
(47) 55
(46) 54
(45) 53
(44) 52
(43) 51
(42) 50
(41) 49
(40) 48
(39) 47
(38) 46
(37) 45
(36) 44
(35) 43
(34) 42
(33) 41
(32) 40
(31) 39
(30) 38
(29) 37
(28) 36
(27) 35
(26) 34
(25) 33
MHB602
V
DDD2
LORNICAM
LOL
MOL
MOR
V
DDA
AUXOL
AUXOR
V
SSA3
PCAPL
PCAPR
V
ref3
SCOL2
SCOR2
V
SSA4
V
SSD2
SCOL1
SCOR1
V
ref2
i.c.
i.c.
V
SSA2
i.c.
i.c.
V
ref(n)
V
ref(p)
V
DEC2
SCIL2
SCIR2
V
SSD3
SCIL1
SCIR1
L9
C42
C40
C38
C36
C34 10 nF
C32
C31 10 nF
C30
C28
10 nF
C26
47 µF
C24
C22
C20
C18
C16 47 µF
R10
15 k
R9
15 k
R8
15 k
R7
15 k
470 pF
470 pF
10 nF
470 nF
10 nF
470 pF
470 pF
470 pF
470 pF
R11
270
330 nF
330 nF
470
nF
C14
C12
C39
2.2 µF
C35
2.2 µF
C33
2.2 µF
C27
10 nF
C23
2.2 µF
C19
2.2 µF
330 nF
330 nF
R13 1
C13
C11
C41
2.2 µF
C37
2.2 µF
R12
2.2
C29
2.2 µF
C25
2.2 µF
C21
2.2 µF
C17
47 µF
C15 470 nF
+5 V
+5 V
Fig.11 Schematic for application.
1999 Dec 20 90
Page 91
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

13 PACKAGE OUTLINES

SDIP64: plastic shrink dual in-line package; 64 leads (750 mil)

SOT274-1

seating plane
L
Z
64
1
pin 1 index
D
A
2
A
A
1
e
b
w
b
1
33
32
M
E
c
M
(e )
M
E
1
H
0 5 10 mm
scale
DIMENSIONS (mm are the original dimensions)
A
A
A
UNIT b
Note
1. Plastic or metal protrusions of 0.25 mm maximum per side are not included.
max.
mm
5.84
OUTLINE VERSION
SOT274-1 MS-021
12
min.
max.
4.57
0.51
IEC JEDEC EIAJ
1.3
0.8
b
1
0.53
0.40
REFERENCES
0.32
0.23
cEe M
(1) (1)
D
58.67
57.70
1999 Dec 20 91
17.2
16.9
1
L
M
E
3.2
19.61
2.8
19.05
EUROPEAN
PROJECTION
20.96
19.71
e
w
H
0.181.778 19.05
ISSUE DATE
95-02-04 99-12-27
max.
1.73
(1)
Z
Page 92
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
QFP64: plastic quad flat package; 64 leads (lead length 1.6 mm); body 14 x 14 x 2.7 mm
c
y
X
A
48 33
49
pin 1 index
64
1
32
Z
E
e
A
H
E
E
2
A
A
1
w M
b
p
17
16
detail X

SOT393-1

(A )
3
θ
L
p
L
w M
b
e
p
D
H
D
Z
D
B
v M
0 5 10 mm
DIMENSIONS (mm are the original dimensions)
mm
A
max.
3.00
0.25
0.10
2.75
2.55
0.25
UNIT A1A2A3b
cE
p
0.45
0.23
0.30
0.13
(1)
(1) (1)(1)
D
14.1
13.9
eH
14.1
13.9
0.8
17.45
16.95
Note
1. Plastic or metal protrusions of 0.25 mm maximum per side are not included.
OUTLINE
VERSION
IEC JEDEC EIAJ
REFERENCES
SOT393-1 MS-022
1999 Dec 20 92
v M
scale
H
D
A
B
E
17.45
16.95
LL
p
1.03
0.73
0.16 0.100.161.60
EUROPEAN
PROJECTION
Z
D
1.2
0.8
Zywv θ
E
o
1.2
7
o
0.8
0
ISSUE DATE
97-08-04 99-12-27
Page 93
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

14 SOLDERING

14.1 Introduction

Thistextgivesaverybriefinsightto a complex technology. A more in-depth account of soldering ICs can be found in our
“Data Handbook IC26; Integrated Circuit Packages”
(document order number 9398 652 90011). There is no soldering method that is ideal for all IC
packages. Wave soldering is often preferred when through-holeandsurfacemount components are mixedon one printed-circuit board. However, wave soldering is not always suitable for surface mount ICs, or for printed-circuit boards with high population densities. In these situations reflow soldering is often used.

14.2 Through-hole mount packages

14.2.1 SOLDERING BY DIPPING OR BY SOLDER WAVE The maximum permissible temperature of the solder is
260 °C; solder at this temperature must not be in contact with the joints for more than 5 seconds. The total contact time of successive solder waves must not exceed 5 seconds.
The device may be mounted up to the seating plane, but the temperature of the plastic body must not exceed the specified maximum storage temperature (T printed-circuit board has been pre-heated, forced cooling may be necessary immediately after soldering to keep the temperature within the permissible limit.
14.2.2 MANUAL SOLDERING Apply the soldering iron (24 V or less) to the lead(s) of the
package, either below the seating plane or not more than 2 mm above it. If the temperature of the soldering iron bit is less than 300 °C it may remain in contact for up to 10 seconds. If the bit temperature is between 300 and 400 °C, contact may be up to 5 seconds.

14.3 Surface mount packages

14.3.1 REFLOW SOLDERING Reflow soldering requires solder paste (a suspension of
fine solder particles, flux and binding agent) to be applied totheprinted-circuitboardbyscreenprinting,stencillingor pressure-syringe dispensing before package placement.
Several methods exist for reflowing; for example, infrared/convection heating in a conveyor type oven. Throughput times (preheating, soldering and cooling) vary between 100 and 200 seconds depending on heating method.
stg(max)
). If the
Typical reflow peak temperatures range from 215 to 250 °C. The top-surface temperature of the packages should preferable be kept below 230 °C.
14.3.2 WAVE SOLDERING Conventional single wave soldering is not recommended
forsurfacemountdevices(SMDs)orprinted-circuit boards with a high component density, as solder bridging and non-wetting can present major problems.
To overcome these problems the double-wave soldering method was specifically developed.
If wave soldering is used the following conditions must be observed for optimal results:
Use a double-wave soldering method comprising a turbulent wave with high upward pressure followed by a smooth laminar wave.
For packages with leads on two sides and a pitch (e): – larger than or equal to 1.27 mm, the footprint
longitudinal axis is preferred to be parallel to the transport direction of the printed-circuit board;
– smaller than 1.27 mm, the footprint longitudinal axis
must be parallel to the transport direction of the printed-circuit board.
The footprint must incorporate solder thieves at the downstream end.
Forpackageswithleadsonfoursides, the footprint must be placed at a 45° angle to the transport direction of the printed-circuit board. The footprint must incorporate solder thieves downstream and at the side corners.
During placement and before soldering, the package must be fixed with a droplet of adhesive. The adhesive can be applied by screen printing, pin transfer or syringe dispensing. The package can be soldered after the adhesive is cured.
Typical dwell time is 4 seconds at 250 °C. A mildly-activated flux will eliminate the need for removal of corrosive residues in most applications.
14.3.3 MANUAL SOLDERING
Fix the component by first soldering two diagonally-opposite end leads. Use a low voltage (24 V or less) soldering iron applied to the flat part of the lead. Contact time must be limited to 10 seconds at up to 300 °C.
When using a dedicated tool, all other leads can be soldered in one operation within 2 to 5 seconds between 270 and 320 °C.
1999 Dec 20 93
Page 94
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A
14.4 Suitability of IC packages for wave, reflow and dipping soldering methods
MOUNTING PACKAGE
Through-hole mount DBS, DIP, HDIP, SDIP, SIL suitable
WAVE REFLOW
(2)
suitable
(1)
DIPPING
Surface mount BGA, SQFP not suitable suitable
SOLDERING METHOD
HLQFP, HSQFP, HSOP, HTQFP, HTSSOP,
not suitable
(3)
suitable
SMS
(4)
PLCC LQFP, QFP, TQFP not recommended SSOP, TSSOP, VSO not recommended
, SO, SOJ suitable suitable
(4)(5)
suitable
(6)
suitable
Notes
1. All surface mount (SMD) packages are moisture sensitive. Depending upon the moisture content, the maximum temperature (with respect to time) and body size of the package, there is a risk that internal or external package cracks may occur due to vaporization of the moisture in them (the so called popcorn effect). For details, refer to the Drypack information in the
“Data Handbook IC26; Integrated Circuit Packages; Section: Packing Methods”
.
2. For SDIP packages, the longitudinal axis must be parallel to the transport direction of the printed-circuit board.
3. These packages are not suitable for wave soldering as a solder joint between the printed-circuit board and heatsink (at bottom version) can not be achieved, and as solder may stick to the heatsink (on top version).
4. If wave soldering is considered, then the package must be placed at a 45° angle to the solder wave direction. The package footprint must incorporate solder thieves downstream and at the side corners.
5. Wave soldering is only suitable for LQFP, QFP and TQFP packages with a pitch (e) equal to or larger than 0.8 mm; it is definitely not suitable for packages with a pitch (e) equal to or smaller than 0.65 mm.
6. Wave soldering is only suitable for SSOP and TSSOP packages with a pitch (e) equal to or larger than 0.65 mm; it is definitely not suitable for packages with a pitch (e) equal to or smaller than 0.5 mm.
1999 Dec 20 94
Page 95
Philips Semiconductors Product specification
Digital TV Sound Processor (DTVSP) TDA9875A

15 DEFINITIONS

Data sheet status
Objective specification This data sheet contains target or goal specifications for product development. Preliminary specification This data sheet contains preliminary data; supplementary data may be published later. Product specification This data sheet contains final product specifications.
Limiting values
Limiting values given are in accordance with the Absolute Maximum Rating System (IEC 134). Stress above one or more of the limiting values may cause permanent damage to the device. These are stress ratings only and operation of the device at these or at any other conditions above those given in the Characteristics sections of the specification is not implied. Exposure to limiting values for extended periods may affect device reliability.
Application information
Where application information is given, it is advisory and does not form part of the specification.

16 LIFE SUPPORT APPLICATIONS

These products are not designed for use in life support appliances, devices, or systems where malfunction of these products can reasonably be expected to result in personal injury. Philips customers using or selling these products for use in such applications do so at their own risk and agree to fully indemnify Philips for any damages resulting from such improper use or sale.
17 PURCHASE OF PHILIPS I
Purchase of Philips I components in the I2C system provided the system conforms to the I2C specification defined by Philips. This specification can be ordered using the code 9398 393 40011.
2
C COMPONENTS
2
C components conveys a license under the Philips’ I2C patent to use the
1999 Dec 20 95
Page 96
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Poland: Al.Jerozolimskie 195 B, 02-222 WARSAW, Tel. +48 22 5710 000, Fax. +48 22 5710 001
Portugal: see Spain Romania: see Italy Russia: Philips Russia, Ul. Usatcheva 35A, 119048 MOSCOW,
Tel. +7 095 755 6918, Fax. +7 095 755 6919 Singapore: Lorong 1, Toa Payoh, SINGAPORE 319762,
Tel. +65 350 2538, Fax. +65 251 6500
Slovakia: see Austria Slovenia: see Italy South Africa: S.A. PHILIPS Pty Ltd., 195-215 Main Road Martindale,
2092 JOHANNESBURG, P.O. Box 58088 Newville 2114, Tel. +27 11 471 5401, Fax. +27 11 471 5398
South America: Al. Vicente Pinzon, 173, 6th floor, 04547-130 SÃO PAULO, SP, Brazil, Tel. +55 11 821 2333, Fax. +55 11 821 2382
Spain: Balmes 22, 08007 BARCELONA, Tel. +34 93 301 6312, Fax. +34 93 301 4107
Sweden: Kottbygatan 7, Akalla, S-16485 STOCKHOLM, Tel. +46 8 5985 2000, Fax. +46 8 5985 2745
Switzerland: Allmendstrasse 140, CH-8027 ZÜRICH, Tel. +41 1 488 2741 Fax. +41 1 488 3263
Taiwan: Philips Semiconductors, 6F, No. 96, Chien Kuo N. Rd., Sec. 1, TAIPEI, Taiwan Tel. +886 2 2134 2886, Fax. +886 2 2134 2874
Thailand: PHILIPS ELECTRONICS (THAILAND) Ltd., 209/2 Sanpavuth-Bangna Road Prakanong, BANGKOK 10260, Tel. +66 2 745 4090, Fax. +66 2 398 0793
Turkey: Yukari Dudullu, Org. San. Blg., 2.Cad. Nr. 28 81260 Umraniye, ISTANBUL, Tel. +90 216 522 1500, Fax. +90 216 522 1813
Ukraine: PHILIPS UKRAINE, 4 Patrice Lumumba str., Building B, Floor 7, 252042 KIEV, Tel. +380 44 264 2776, Fax. +380 44 268 0461
United Kingdom: Philips Semiconductors Ltd., 276 Bath Road, Hayes, MIDDLESEX UB3 5BX, Tel. +44 208 730 5000, Fax. +44 208 754 8421
United States: 811 East Arques Avenue, SUNNYVALE, CA 94088-3409, Tel. +1 800 234 7381, Fax. +1 800 943 0087
Uruguay: see South America Vietnam: see Singapore Yugoslavia: PHILIPS, Trg N. Pasica 5/v, 11000 BEOGRAD,
Tel. +381 11 62 5344, Fax.+381 11 63 5777
For all other countries apply to: Philips Semiconductors, International Marketing & Sales Communications, Building BE-p, P.O. Box 218, 5600 MD EINDHOVEN, The Netherlands, Fax. +31 40 27 24825
© Philips Electronics N.V. SCA All rights are reserved. Reproduction in whole or in part is prohibited without the prior written consent of the copyright owner.
The information presented in this document does not form part of any quotation or contract,isbelievedtobe accurate and reliable and may be changed without notice. No liability will be accepted by the publisher for any consequence of its use. Publication thereof does not convey nor imply any license under patent- or other industrial or intellectual property rights.
1999
Internet: http://www.semiconductors.philips.com
68
Printed in The Netherlands 545004/02/pp96 Date of release: 1999 Dec 20 Document order number: 9397 750 06065
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