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.
1.2DSP 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.
2GENERAL 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.
2.1Supported 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.
1.3Analog 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 203
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).
The pin numbers given in parenthesis refer to the TDA9875AH version.
Fig.1 Block diagram.
1999 Dec 206
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 SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
5PINNING
SYMBOL
PIN
TDA9875ATDA9875AH
PIN
TYPE
(1)
DESCRIPTION
PCLK157ONICAM clock output at 728 kHz
NICAM258Oserial NICAM data output at 728 kHz
ADDR1359II
SCL460II
SDA561I/OI
V
V
I
ref
SSA1
DEC1
662Ssupply ground 1; analog front-end circuitry
763−supply voltage decoupling 1; analog front-end circuitry
864−resistor 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
P191I/Ogeneral purpose input/output pin 1
SIF2102Isound IF input 2
V
ref1
113−reference voltage 1; analog front-end circuitry
MONOIN2921Iaudio mono input
TEST23022Itest pin 2; connected to V
for normal operating mode
SSD1
EXTIR3123Iexternal audio input right channel
EXTIL3224Iexternal audio input left channel
SCIR13325ISCART 1 input right channel
SCIL13426ISCART 1 input left channel
V
SSD3
3527Ssupply ground 3; digital circuitry
SCIR23628ISCART 2 input right channel
SCIL23729ISCART 2 input left channel
V
DEC2
3830−supply voltage decoupling 2; audio analog-to-digital converter
4638−reference voltage 2; audio analog-to-digital converter circuitry
SCOR14739OSCART 1 output right channel
SCOL14840OSCART 1 output left channel
V
V
SSD2
SSA4
4941Ssupply ground 2; digital circuitry
5042Ssupply ground 4; audio operational amplifier circuitry
SCOR25143OSCART 2 output right channel
SCOL25244OSCART 2 output left channel
V
ref3
5345−reference voltage 3; audio digital-to-analog converter and
operational amplifier circuitry
PCAPR5446−post-filter capacitor pin right channel; audio digital-to-analog
converter
PCAPL5547−post-filter capacitor pin left channel; audio digital-to-analog
converter
V
SSA3
5648Ssupply ground 3; audio digital-to-analog converter circuitry
AUXOR5749Oheadphone (Auxiliary) output right channel
AUXOL5850Oheadphone (Auxiliary) output left channel
V
DDA
5951Sanalog supply voltage; analog circuitry
MOR6052Oloudspeaker (Main) output right channel
MOL6153Oloudspeaker (Main) output left channel
LOL6254Oline output left channel
LOR6355Oline output right channel
V
DDD2
6456Sdigital 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.
6.1.1SIF 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
switchable−10 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.
6.1.2AGC
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).
6.1.3MIXER
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.
6.1.4FM 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.
6.1.5FM 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.
6.1.6NICAM 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.
6.1.7NICAM 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 2011
Page 12
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.1.8NICAM 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.
6.1.9CRYSTAL OSCILLATOR
The circuitry of the crystal oscillator is fully integrated, only
the external 24.576 MHz crystal is needed (see Fig.10).
6.1.10TEST 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.
6.1.12POWER-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 kΩ, 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.
6.1.11POWER 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 2012
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 2013
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
222
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
6.2Digital signal processing
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
2
S1
2
S2
24
12
16
Fig.5 DSP data flow diagram.
MONITOR
SELECT
PEAK
DETECTION
1
2
I
C-bus
MGK108
Page 14
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.2.1LEVEL 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.
6.2.2NICAM PATH
The NICAM path has a switchable J17 de-emphasis.
6.2.3FM (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.
6.2.4NICAM 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.
6.2.5MONITOR
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.
6.2.6LOUDSPEAKER (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 2014
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 SemiconductorsProduct 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.
6.2.7HEADPHONE (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.
6.2.8FEATURE 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.
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.
6.2.9CHANNEL 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.
6.2.10CHANNEL TO THE ANALOG CROSSBAR PATH
Level adjust with control positions 0, +3, +6 and +9 dB.
6.2.11DIGITAL 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.
6.2.12SIGNAL 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 2015
Page 16
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.2.13EXPERT 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.
6.2.14DSP 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 outputyescontrol range0to +18dB
Bass boost for loudspeaker outputyescontrol range0 to +20dB
Volume control for each separate
channel in loudspeaker and
headphone output
Soft mute for loudspeaker and
headphone output
Spatial effectsyesanti-phase crosstalk positions30, 40 and 52%
Pseudo stereoyes90 degrees phase shift at frequency150, 200 and 300Hz
Beeper additional to the signal in
resolution2dB
resolution at frequency20Hz
corner frequency350Hz
nocontrol range−83 to +24dB
resolution1dB
mute position at step1010 1100
noprocessing time32ms
yesbeep frequenciessee Section 10.3.38
control range0 to −93dB
resolution3dB
mute position at step0010 0000
AVL output level for an input level
between 0 and −29 dB (full-scale)
attack time10ms
decay time constant2, 4 and 8s
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.
PARAMETERVALUEUNIT
−23dB
1999 Dec 2016
Page 17
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
FUNCTION
EXPERT
MODE
PARAMETERVALUEUNIT
Generalno−3 dB lower corner frequency of DSP 10Hz
−1 dB bandwidth of DSP14.5kHz
Level adjust I
2
S1 and I2S2 inputsyescontrol range−15 to +15dB
resolution1dB
Level adjust I
2
S1 and I2S2 outputsyescontrol range−15 to +15dB
resolution1dB
mute position at step0001 0000
Level adjust analog crossbar pathnocontrol positions0, 3, 6 and 9dB
Level adjust audio ADC outputsyescontrol range+15 to −15dB
resolution1dB
Level adjust NICAM pathyescontrol range+15 to −15dB
resolution1dB
Level adjust FM pathyescontrol range+15 to −15dB
resolution1dB
6.3Analog 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
22
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 2017
Page 18
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.3.1ANALOG 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.
6.3.2SCART 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.
6.3.3EXTERNAL 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.
6.3.4SCART 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.
All switches and matrices are controlled via the I2C-bus.
1999 Dec 2018
Page 19
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.3.5LINE 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).
6.3.6LOUDSPEAKER (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.
6.3.7DUAL 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.
6.3.8DUAL 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.
6.3.9STANDBY 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.
6.3.10SUPPLY 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 2019
Page 20
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1999 Dec 2020
SCART 1
handbook, full pagewidth
Philips SemiconductorsProduct 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 SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
7LIMITING VALUES
In accordance with the Absolute Maximum Rating System (IEC 134).
SYMBOLPARAMETERCONDITIONSMIN.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 kΩ.
2. Machine model: C = 200 pF; L = 0.75 µH; R = 0 Ω.
DC supply voltage−0.5+6.0V
voltage differences between two VDD pins−550mV
voltage on any other pin−0.5VDD+ 0.5 V
DC current per digital supply pin−±180mA
latch-up protection current100−mA
total power dissipation−1.0W
storage temperature−55+125°C
ambient temperature−20+70°C
electrostatic handling voltagenote 1−2000+2000V
note 2−200+200V
8THERMAL CHARACTERISTICS
SYMBOLPARAMETERCONDITIONSVALUEUNIT
R
th(j-a)
thermal resistance from junction to ambientin free air
TDA9875A (SDIP64)40K/W
TDA9875AH (QFP64)50K/W
1999 Dec 2021
Page 22
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
9CHARACTERISTICS
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 kΩ measurement source
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
DDD1
SSD1
DDD1
DDD2
SSD2
DDD2
digital supply voltage 14.755.05.5V
digital supply ground 1note 2−0.0−V
digital supply current 1V
=5.0V5873 88mA
DDD1
digital supply voltage 24.755.05.5V
digital supply ground 2note 2−0.0−V
digital supply current 2V
= 5.0 V; system clock
DDD2
0.10.42mA
output disabled
SSD3
SSD4
DDA
DDA
digital supply ground 3note 2−0.0−V
digital supply ground 4note 2−0.0−V
analog supply voltage4.755.05.5V
analog supply current for
V
= 5.0 V; digital silence445668mA
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
3.03.33.6V
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
3.03.33.6V
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−kΩ
−20−kΩ
−50−%
amplifier
Vref3-VDDA
Vref3-VSSA3
impedance pins V
impedance pins V
ref3
ref3
to V
to V
DDA
SSA3
−20−kΩ
−20−kΩ
1999 Dec 2022
Page 23
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
Power fail detector
V
th(pf)
Digital inputs and outputs
INPUTS
CMOS level input, pull-down (pins TEST1 and TEST2)
adjustment tolerance−− ±3010
driftacross temperature range−− ±3010
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 Ω. Now measure the
1 kHz crosstalk signal level. The SIF source resistance should be low (50 Ω).
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 kΩ 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 2028
Page 29
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1999 Dec 2029
Table 7 Level setting FM, AM and NICAM at 0 dB (full-scale) = 1.4 V (RMS); note 1
TRANSMITTER
STANDARDMODE
M2 channel 15 kHz deviation−24 dB (full-scale);
B/G2 channel 27 kHz deviation−19 dB (full-scale)15.5 MHzFM−50 µs+4dB
NICAM−11.2 dB (full-scale) −18 dB (full-scale)15.5 MHzFM−50 µs+4dB
INICAM−15.8 dB (full-scale) −23 dB (full-scale)16.0 MHzFM−50 µs+4dB
D/K2 channel 27 kHz deviation−19 dB (full-scale)16.5MHzFM−50 µs+4dB
2 channel 27 kHz deviation−19 dB (full-scale)16.5 MHzFM−50 µs+4dB
2 channel 27 kHz deviation−19 dB (full-scale)16.5 MHzFM−50 µs+4dB
NICAM−11.2 dB (full-scale) −18 dB (full-scale)16.5 MHzFM−50 µs+4dB
L/L accentNICAM54% AM−19 dB (full-scale)16.5 MHzAM−50 µs+5dB
NOMINAL
MODULATION
DEPTH
NOMINALLEVELAT
DEMODULATOR
OUTPUT
note 2
FM/NICAM
CARRIER FREQUENCY MODE IDENT DE-EMPHASIS
14.5 MHzFM−75 µs+9dB
24.724 MHzFMon75 µs+9dB
25.742 MHzFMon50 µs+4dB
25.85 MHzNICAM offJ17+3 dB
26.552 MHzNICAM offJ17+8 dB
26.742 MHzFMon50 µs+4dB
26.25 MHzFMon50 µs+4dB
25.742 MHzFMon50 µs+4dB
25.85 MHzNICAM offJ17+3 dB
25.85 MHzNICAM offJ17+3 dB
LEVEL
ADJUST
Philips SemiconductorsProduct 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 2030
Table 8 Level setting SAT FM at 0 dB (full-scale) = 1.4 V (RMS)
TRANSMITTER
SOURCE
SAT FM, stereo50 kHz deviation−13 dB (full-scale)+4 dB−9 dB (full-scale)+6 dB0 dB1 V (RMS)
SAT FM, mono85 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 SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
Page 31
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
10 I2C-BUS CONTROL
10.1Introduction
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
ADDR2ADDR1
LOW LOW 1011000
LOWHIGH1011001
HIGHLOW1011010
HIGHHIGH1011011
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
10.2Power-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 2031
Page 32
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
10.3Slave 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
SSLAVE ADDRESS0ACKSUBADDRESSACKDATAACKP
Table 11 Explanation of Table 10
SSTART condition
SLAVE ADDRESS7-bit device address
0data direction bit (write to device)
ACKacknowledge by slave
SUBADDRESSaddress of register to write to
DATAdata byte to be written into register
PSTOP condition
C-bus; slave address, subaddress and data format
BITFUNCTION
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
SSLAVE ADDRESS0ACKSUBADDRESSACKDATA
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.
DATAACKP
(1)
1999 Dec 2032
Page 33
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
Table 13 Overview of the slave receiver registers
SUBADDRESS
(DECIMAL)
000sgggggAGC level shift, AGC gain selection
1ccccccccgeneral configuration
2p00mmsssmonitor select, peak detector on/off
3ffffffffcarrier 1 frequency; most significant part
4ffffffffcarrier 1 frequency
5ffffffffcarrier 1 frequency; least significant part
6ffffffffcarrier 2 frequency; most significant part
7ffffffffcarrier 2 frequency
8ffffffffcarrier 2 frequency; least significant part
9ccccccccdemodulator configuration
10ddddddddFMde-emphasis
1100000mmmFMmatrix
12 000lllllchannel 1 output level adjust
13 000lllllchannel 2 output level adjust
14tt0c0cccNICAM configuration
15 000lllllNICAM output level adjust
16llllllllNICAM lower error limit
17uuuuuuuuNICAM upper error limit
18mmmmmmmmaudio mute control
19gmmmgsssDAC output select
200gmm0sssSCART 1 output select
210gmm0sssSCART 2 output select
220gmm000sline output select
23ssslllllADC output select
240mmm0sssMain channel select
2500ssppaaaudio effects (AVL, pseudo and spatial)
26vvvvvvvvvolume control, Main left
27vvvvvvvvvolume control, Main right
28000ccccccontour control, Main
29000bbbbbbass control, Main
30000ttttttreble control, Main
310mmm0sssAuxiliary channel select
32vvvvvvvvvolume control, Auxiliary left
33vvvvvvvvvolume control, Auxiliary right
34000bbbbbbass control, Auxiliary
35000ttttttreble control, Auxiliary
36000cccccfeature interface configuration
370mmm0 s s sI
MSBLSB
DATA
2
S1 output select
FUNCTION
1999 Dec 2033
Page 34
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
SUBADDRESS
(DECIMAL)
MSBLSB
38 000iiiiiI
39000oooooI
400mmm0 s s sI
41 000iiiiiI
42000oooooI
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
4300000f f fbeeper frequency
4400vvvvvvbeeper volume, Main and Auxiliary
45bbbbbbbbbass boost, Main left and right
The following sub-sections provide a detailed description of the slave receiver registers.
10.3.1AGC 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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70set to logic 0
6B60set to logic 0
5AGCLEV1input signal scaled with −10 dB
0input signal not scaled
4AGCGAIN−gain control bits (see Table 15)
3
2
1
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).
6P1OUT−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.
5STDBY1The 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.
0The 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.
4INIT1Causes 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.
0Automatically reset to logic 0 after initialization. When set to logic 0, the
TDA9875A is in its normal operating mode.
3CLRPOR1Resets the power fail detector to LOW.
0This bit is automatically reset to logic 0 after bit POR in the device status register
has been reset.
2AGCSLOW1A longer decay time is selected for input signals with strong video modulation
(intercarrier). This bit only has an effect when bit AGCOFF = 0.
0Selects normal attack and decay times for the AGC.
1AGCOFF1Forces the AGC block to a fixed gain as defined in the AGC gain register.
0The automatic gain control function is enabled and the contents of the AGC gain
register is ignored.
0 (LSB)SIFSEL1Selects pin SIF2 for input (recommended for satellite tuner).
0Selects pin SIF1 for input (terrestrial TV).
Note
1. The default setting at power-up is 1100 0000.
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Digital TV Sound Processor (DTVSP)TDA9875A
10.3.3MONITOR 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)
BITNAMEVALUEDESCRIPTION
7 (MSB)PEAKMON1selects the peak level of a source to be monitored
0the last sample will be supplied
6B60default value
5B50default value
4B4−monitor output (see Table 18)
3B3
2B2−signal source (see Table 19)
1B1
000DC output of FM demodulator
001audio magnitude/phase, FM demodulator output
010crossbar input from FM/AM channel
011crossbar input from NICAM channel
100crossbar input from I
101crossbar input from I
110crossbar input from audio ADC channel
111input 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.
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
10.3.4CARRIER 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
BITDESCRIPTION
37 (MSB)carrier 1 frequency;
6
most significant part
5
4
3
2
1
0
47carrier 1 frequency
6
5
4
3
2
1
0
57carrier 1 frequency;
6
least significant part
5
4
3
2
1
0 (LSB)
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
10.3.5CARRIER 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.
10.3.6DEMODULATOR 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)
BITNAMEVALUEDESCRIPTION
7 (MSB)IDMOD1−these bits define the response time after which a FM sound mode identification
6IDMOD0
5IDAREA1selects FM identification frequencies in accordance with the specification for
4FILTBW1−selects filter bandwidth (see Table 23)
3CH2MOD1−channel 2 receive mode: these bits control the hardware for the second sound
2CH2MOD0
1FILTBW0−selects the filter bandwidth (see Table 23)
0 (LSB)CH1MODE1selects 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
0selects 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
0FM 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
B7B6IDENT MODE
00slow
01medium
10fast
11off/reset
Table 23 Filter bandwidth channel 1 and channel 2
FILTER
B4B1
00narrownarrowrecommended for nominal terrestrial broadcast conditions and
01extra widenarrowrecommended only for high-deviation SAT mono carriers
10mediummediumrecommended for moderately overmodulated broadcast conditions
11widewiderecommended for strongly overmodulated broadcast conditions
BANDWIDTH
CH1CH2
FILTER MODES
SAT with 2 carriers
(e.g. obsolete Main channel on Astra)
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
Table 24 Channel 2 receive mode
B3B2CHANNEL 2
00FM
01AM
10NICAM
10.3.7FM
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)ADEEM21Activates the adaptive de-emphasis function, which is required for certain satellite
6B6−Time constant selection for FM de-emphasis (see Table 26).
5B5
4B4
3ADEEM11Activates the adaptive de-emphasis function, which is required for certain satellite
2B2−Time 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).
0The adaptive de-emphasis is off.
FM channels. The standard FM de-emphasis must then be set to 75 µs (note 2).
10.3.9FM 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.
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.
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)DCXOPULL1Set to lower DCXO frequency during DCXO test mode.
6DCXOTEST1DCXO test mode on (available only during FM mode); note 2
5B50Set logic to 0.
4DOUTEN1Enables the output of the NICAM serial data stream from the DQPSK
30Set logic to 0.
2AMSEL1The auto-mute function will switch the output sound from NICAM L to the
1NDEEM1Switches the NICAM J17 de-emphasis off.
0 (LSB)AMUTE1Automatic muting is disabled. This bit has only an effect when the second
“NICAM 728 ETS Revised for Data Applications”
0Set to higher DCXO frequency during DCXO test mode.
0DCXO normal mode on
demodulator on pin NICAM and of the associated clock on pin PCLK
0Both 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.
0The auto-mute function will switch the output sound from NICAM L to the
AM program on the internal first sound carrier.
0Switches the NICAM J17 de-emphasis on.
sound carrier is set to NICAM.
0Enables 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.
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)
BITNAMEVALUEDESCRIPTION
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)
BITNAMEVALUEDESCRIPTION
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)
BITNAMEVALUEDESCRIPTION
2
7 (MSB)MUTI
6MUTI
5MUTDAC1mute internal DAC
4MUTLINE1mute line outputs
3MUTSC21mute SCART 2 outputs
2MUTSC11mute SCART 1 outputs
1MUTAUX1mute Auxiliary outputs
0 (LSB)MUTMAIN1mute Main outputs
S21mute I2S2 output
0normal I
2
S11mute I2S1 output
0normal I
2
S2 output
2
S1 output
0normal internal DAC
0normal line outputs
0normal SCART 2 outputs
0normal SCART 1 outputs
0normal Auxiliary outputs
0normal 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)
BITNAMEVALUEDESCRIPTION
7 (MSB)DACGAIN2−extra gain setting (see Table 38)
6B6−DAC output selection (seeTable 39)
5B5
4B4
3DACGAIN1−extra gain setting (see Table 38)
2B2−signal source selection (see Table 40)
1B1
000FM leftFM right
001NICAM leftNICAM right
010I
011I
100ADC leftADC right
101AVL leftAVL right
110FM monoNICAM M1
111FM monoNICAM M2
LR+
------------- 2
SIGNAL SOURCE
LEFTRIGHT
2
S1 leftI2S1 right
2
S2 leftI2S2 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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6SC1GAIN1Activates 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.
0the audio signal output will be unchanged (0 dB gain)
5B5−output channel selection (see Table 42)
4B4
3B30default value
2B2−signal source selection (see Table 43)
1B1
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70
6SC2GAIN1Activates 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.
0the audio signal output will be output (0 dB gain)
5B5−output channel selection (see Table 45)
4B4
3B30default value
2B2−signal source selection (see Table 46)
1B1
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70set to logic 0
6LINGAIN1activates the 3 dB gain stage at the line output buffers
0audio signal will be output unchanged (0 dB gain)
5B5−output channel selection (see Table 48)
4B4
3B30set to logic 0
2B20set to logic 0
1B10set to logic 0
0 (LSB)LINSEL1A 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.
0A signal from a digital source is being processed in the Main channel for line
output. Digital signal sources comprise FM, NICAM, I
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.
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6B6−output channel selection (see Table 53)
5B5
4B4
3B30default value
2B2−signal source selection (see Table 54)
1B1
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70Default value.
6B60Default value.
5SPATIAL1−These bits set the amount of the effect function (stereo base width expansion)
4SPATIAL0
3PSEUDO1−These bits set the amount of the effect function (pseudo stereo) for mono
2PSEUDO0
1AVL1−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).
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.
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.
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6B6−output channel selection (see Table 64)
5B5
4B4
3B30default value
2B2−signal source selection (see Table 65)
1B1
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.
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.
6B60default value
5B50default value
4SYSCL1−system clock frequency selection (see Table70)
3SYSCL0
2SYSOUT1enables the output of a system (or master) clock signal at pin SYSCLK
0the output will be off, thereby improving the EMC performance
1I
0 (LSB)I
2
SFORM1an 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
0the standard I
2
SOUT1enables 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
0the 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
B4B3SYSCLK OUTPUTFREQUENCY (MHz)
00256f
01384f
10512f
11768f
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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Digital TV Sound Processor (DTVSP)TDA9875A
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6B6−output selection (see Table 72)
5B5
4B4
3B30default value
2B2−signal source selection (see Table 73)
1B1
1. The Main and Auxiliary channel outputs will not contain the beeper signal.
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Digital TV Sound Processor (DTVSP)TDA9875A
10.3.33 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.
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.
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6B6−output selection (see Table 77)
5B5
4B4
3B30default value
2B2−signal source selection (see Table 78)
1B1
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.
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.
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
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.
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)
BITNAMEVALUEDESCRIPTION
7 (MSB)B7−gain setting of right channel (see Table 84)
6B6
5B5
4B4
3B3−gain setting of left channel (see Table 85)
2B2
1B1
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 SrSLAVE ADDRESS1ACKDATA NAm P
Table 87 Explanation of Tables 86 and 88
BITFUNCTION
SSTART condition
SLAVE ADDRESS7-bit device address
0data direction bit (write to device)
ACKacknowledge (by the slave)
SUBADDRESSaddress of register to read from
Srrepeated START condition
1data direction bit (read from device)
DATAdata byte read from register
NAmnot acknowledge (by the master)
Amacknowledge (by the master)
PSTOP 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 SrSLAVE
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 ACKDATA BYTE
Am
(1)
DATA NAm P
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Digital TV Sound Processor (DTVSP)TDA9875A
Table 89 Overview of the slave transmitter registers (note 1)
SUBADDRESS
(DECIMAL)
0ssssssssdevice status (power-on, identification, etc.)
1ssssssssNICAM status
2 eeeeeeeeNICAM error count
3 ddddddddadditional data (LSB)
4ccXccdddadditional data (MSB)
5lllllllllevel read-out (MSB)
6lllllllllevel read-out (LSB)
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.
MSBLSB
DATA
FUNCTION
The following sub-sections provide a detailed description of the slave transmitter registers.
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10.4.1DEVICE STATUS REGISTER
Table 90 Subaddress 0
BITNAMEVALUEDESCRIPTION
7 (MSB)P2IN−This bit reflects the status of the corresponding general purpose port of pin P2
(see Section 10.3.2).
6P1IN−This bit reflects the status of the corresponding general purpose port of pin P1
(see Section 10.3.2).
5RSSF1Reserve 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.
0Auto-muting should be disabled, as analog and digital sound are different.
4AMSTAT1Auto-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).
0Auto-muting function has not switched.
3VDSP1Indicates that digital transmission is a sound source (NICAM).
0The transmission is either data or currently undefined format (NICAM).
2IDDUA−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.
1IDSTE−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
1999 Dec 2081
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
10.4.2NICAM 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
BITNAMEVALUEDESCRIPTION
7 (MSB)C4−application control bits (C1 to C4 in the NICAM transmission)
6C3
5C2
4C1
3OSB1indication that the device has both frame and C0 (16 frame) synchronization
0the audio output from the NICAM part should be digital silence
2CFC1indication of a configuration change at the 16 frame (C0) boundary
0no configuration change
1S/MB1indication of NICAM stereo mode
0no NICAM stereo mode
0 (LSB)D/SB1indication NICAM dual mono mode
0no NICAM dual mono mode
10.4.3NICAN 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
BITNAMEVALUEDESCRIPTION
7 (MSB)B7−number of errors
6B6
5B5
4B4
3B3
2B2
1B1
0 (LSB)B0
1999 Dec 2082
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
10.4.4ADDITIONAL DATA REGISTERS
These two bytes provide information on the additional data bits.
Table 93 Subaddress 3
BITNAMEVALUEDESCRIPTION
7 (MSB)AD7−comprise the additional data word
6AD6
5AD5
4AD4
3AD3
2AD2
1AD1
0 (LSB)AD0
Table 94 Subaddress 4
BITNAMEVALUEDESCRIPTION
7 (MSB)OVW1new additional data bits are written to the IC without the previous bits being
read
0no bits are written
6SAD1new additional data is written into the IC
0this bit is set to logic 0 when the additional data bits are read
5X−don’t care
4CI1−these are CI bits decoded by majority logic from the parity checks of the last
3CI2
2AD10−comprise the additional data word
1AD9
0 (LSB)AD8
ten samples in a frame
1999 Dec 2083
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
10.4.5LEVEL 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
57 (MSB) most significant bit or sign bit
67 (MSB)
10.4.6SIF 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).
BITDESCRIPTION
6
5
4
3
2
1
0 (LSB)
6
5
4
3
2
1
0 (LSB) least significant bit
Table 96 Subaddress 7
BITNAMEVALUEDESCRIPTION
7 (MSB)B7Xbit not assigned
6B6Xbit not assigned
5B5Xbit not assigned
4B4−indication of SIF
3B3
2B2
1B1
0 (LSB)B0
10.4.7TEST 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
MSBLSB
B7B6B5B4B3B2B1B0
01111111
10.4.8T
This register contains, as a binary number, the highest
subaddress used for slave receiver registers.
Table 98 Subaddress 252
MSBLSB
B7B6B5B4B3B2B1B0
00101101
EST REGISTER 2
input level
When the SIF AGC is off, this register returns the contents
of the AGC gain register.
1999 Dec 2084
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
10.4.9TEST REGISTER 1
This register contains, as a binary number, the highest
subaddress used for slave transmitter (status) registers.
Table 99 Subaddress 253
MSBLSB
B7B6B5B4B3B2B1B0
00000111
10.4.10 D
There will be several devices in the digital TV sound
processorfamily.Thisbyteisusedtoidentifytheindividual
family members.
Table 100 Subaddress 254
MSBLSB
B7B6B5B4B3B2B1B0
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
MSBLSB
B7B6B5B4B3B2B1B0
00000010
10.5Expert 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.
1999 Dec 2085
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
Table 102 General format for entering the expert mode and writing coefficients into the TDA9875A
SSLAVE
ADDRESS
Table 103 Explanation of Table 102
BITFUNCTION
SSTART condition
SLAVE ADDRESS7-bit device address
0data direction bit (write to device)
ACKacknowledge
10000000pattern to enter the expert mode
CRAM ADDRESSstart address of coefficient RAM to write to
DATAdata byte containing part of a coefficient
PSTOP condition
Table 104 General format (notes 1, 2 and 3)
BYTEDATADESCRIPTION
1 data byteaaaaaaaa2MST of 1st coefficient
2 data byteaaaaXXXX1LST of 1st coefficient
Notes
1. X = don’t care.
2. MST = most significant third.
3. LST = least significant third.
0ACK10000000ACKCRAM
ADDRESS
ACKDATAACKDATAACKP
Table 105 Transfer of two coefficients
BYTEDATADESCRIPTION
1 data byteaaaaaaaa2MST of 1st coefficient
2 data byteaaaabbbb1LST of 1st coefficient + 1 MST of 2nd coefficient
3 data bytebbbbbbbb2LST of 2nd coefficient
1999 Dec 2086
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Philips SemiconductorsProduct 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 2087
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
handbook, full pagewidth
SCK
WS
SDLSBMSB
handbook, full pagewidth
SCK
WS
SDLSBMSB
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 2088
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
12 APPLICATION INFORMATION
handbook, full pagewidth
SIFSAT
SIFTV
+5 V
R2
1.5 Ω
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 kΩ
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 kΩ
R3
15 kΩ
C23
10 nF
C21
47 µF
C16
47 µF
C15
47 µF
15 kΩ
15 kΩ
R19
47 µF
1.5 Ω
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 Ω
330 nF
330 nF
2.2 µF
C13
C11
C27
C22
10 nF
2.2 Ω
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 2089
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Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
handbook, full pagewidth
POWER
0 V
SIFSAT
SIFTV
+5 V
100 Ω
C2
470 nF
C3
47 pF
C5
47 pF
R5
1 Ω
C8
470 nF
C9
470 nF
+5 V
47 µFC1
R1
R2
100 Ω
R4
2.2 kΩ
L3
L4
C6
470 nF
24.576 MHz
R6
2.2 kΩ
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 kΩ
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.
1. Plastic or metal protrusions of 0.25 mm maximum per side are not included.
max.
mm
5.84
OUTLINE
VERSION
SOT274-1MS-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
cEeM
(1)(1)
D
58.67
57.70
1999 Dec 2091
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
0.181.77819.05
ISSUE DATE
95-02-04
99-12-27
max.
1.73
(1)
Z
Page 92
Philips SemiconductorsProduct 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
4833
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
0510 mm
DIMENSIONS (mm are the original dimensions)
mm
A
max.
3.00
0.25
0.10
2.75
2.55
0.25
UNITA1A2A3b
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 2092
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
Zywvθ
E
o
1.2
7
o
0.8
0
ISSUE DATE
97-08-04
99-12-27
Page 93
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
14 SOLDERING
14.1Introduction
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.
14.2Through-hole mount packages
14.2.1SOLDERING 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.
14.2.2MANUAL 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.
14.3Surface mount packages
14.3.1REFLOW 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.
14.3.2WAVE 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.
14.3.3MANUAL 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 2093
Page 94
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
14.4Suitability of IC packages for wave, reflow and dipping soldering methods
MOUNTINGPACKAGE
Through-hole mount DBS, DIP, HDIP, SDIP, SILsuitable
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
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 2094
Page 95
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
15 DEFINITIONS
Data sheet status
Objective specificationThis data sheet contains target or goal specifications for product development.
Preliminary specificationThis data sheet contains preliminary data; supplementary data may be published later.
Product specificationThis 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 2095
Page 96
Philips Semiconductors – a w orldwide compan y
Argentina: see South America
Australia: 3 Figtree Drive, HOMEBUSH, NSW 2140,
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
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 Netherlands545004/02/pp96 Date of release: 1999 Dec 20Document order number: 9397 750 06065
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