10.5Expert mode
11I2S-BUS DESCRIPTION
12APPLICATION INFORMATION
13PACKAGE OUTLINES
14SOLDERING
14.1Introduction
14.2Through-hole mount packages
14.3Surface mount packages
14.4Suitability of IC packages for wave,reflow and
dipping soldering methods
15DEFINITIONS
16LIFE SUPPORT APPLICATIONS
17PURCHASE OF PHILIPS I2C COMPONENTS
1999 Dec 202
Page 3
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
1FEATURES
1.1Demodulator and decoder section
• Sound IF (SIF) input switch e.g. to select between
terrestrial TV SIF and SAT SIF sources
• SIF AGC with 24 dB control range
• SIF 8-bit Analog-to-Digital Converter (ADC)
• 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
• Programmableidentification(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
surround) with matrix, level adjust and mute.
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 functionality for SCART copies
headphone (Auxiliary) outputs; also applicable for
L, R, C and S in the Dolby Pro Logic mode with feature
extension.
2GENERAL DESCRIPTION
The TDA9870A is a single-chip Digital TV Sound
Processor (DTVSP) for analog multi-channel sound
systems in TV sets and satellite receivers.
2.1Supported standards
The multistandard/multi-stereo capability of the
TDA9870A is mainly of interest in Europe, but also in
Hong Kong/Peoples Republic of China and South East
Asia.This includesB/G, D/K, I, M and Lstandard.Inother
application areas there exists only subsets of those
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, 8 MHz UHF) and monaural sound.
Korea has a stereo sound system similar to Europe and is
supported by the TDA9870A. 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 TDA9870AH version.
SIF2SIF1
10 (2)12 (4)
INPUT SWITCH
AGC, ADC
FM (AM)
DEMODULATION
A2/SATELLITE
DECODER
LEVEL
ADJUST
DIGITAL
SELECT
AUDIO PROCESSING
DAC (2)
(52)
(53)
60
61
MOR
AUXOL
ADC (2)
DAC (2)
DAC (2)
(50)
58
(49)
57
AUXOR
SUPPLY
SOUND IF
(SIF)
TDA9870A
(
TDA9870AH
ANALOG
CROSSBAR
SWITCH
SUPPLY
SCART,
DAC,
ADC
(63) 7
V
V
V
I
ref
DEC1
SSA1
ref1
(62) 6
(3) 11
(64) 8
)
(25) 33
(26) 34
(28) 36
(29) 37
(23) 31
(24) 32
(21) 29
(39) 47
(40) 48
(43) 51
(44) 52
(55) 63
(54) 62
(33) 41
(34) 42
(36) 44
(37) 45
(46) 54
(47) 55
(51) 59
(30) 38
(31) 39
(32) 40
(38) 46
(45) 53
(35) 43
(48) 56
(42) 50
MHB593
(57) 1
(58) 2
SCIR1
SCIL1
SCIR2
SCIL2
EXTIR
EXTIL
MONOIN
SCOR1
SCOL1
SCOR2
SCOL2
LOR
LOL
i.c.
i.c.
i.c.
i.c.
i.c.
i.c.
PCAPR
PCAPL
V
DDA
V
DEC2
V
ref(p)
V
ref(n)
V
ref2
V
ref3
V
SSA2
V
SSA3
V
SSA4
Fig.1 Block diagram.
1999 Dec 206
Page 7
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
5PINNING
SYMBOL
PIN
TDA9870ATDA9870AH
PIN
TYPE
(1)
DESCRIPTION
i.c.157−internally connected; note 2
i.c.258−internally connected; note 2
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
SIF1124Isound IF input 1
ADDR2135II
V
V
MONOIN2921Iaudio mono input
TEST23022Itest pin 2; connected to V
for normal operation
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 right channel output
SCOL14840OSCART 1 left channel output
V
V
SSD2
SSA4
4941Ssupply ground 2; digital circuitry
5042Ssupply ground 4; audio operational amplifier circuitry
SCOR25143OSCART 2 right channel output
SCOL25244OSCART 2 left channel output
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) right channel output
AUXOL5850Oheadphone (Auxiliary) left channel output
V
DDA
5951Sanalog power supply voltage; analog circuitry
MOR6052Oloudspeaker (Main) right channel output
MOL6153Oloudspeaker (Main) left channel output
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 3-state stage, pull-up resistor, can be connected to VSS.
3. Test pin: CMOS level input, pull-up resistor, can be connected to VSS.
4. 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 internally
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 14; subaddress 0).
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.
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.
6.1.6CRYSTAL OSCILLATOR
The circuitry of the crystal oscillator is fully integrated, only
the external 24.576 MHz crystal is needed (see Fig.10).
6.1.7TEST PINS
Test pins TEST1 and TEST2 are active HIGH and in
normal operating mode of the device they are connected
to V
and are not available to customers. Without external
circuitry these pins are pulled down to LOW level with
internal resistors.
6.1.8POWER 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-onreset bit POR (see Section 10.4.1), will be set to
logic 1. Bit CLRPOR (see Section 10.3.2) resets the
Power-on reset flip-flop to LOW. If this is detected, an
initialization of the TDA9870A has to be carried out to
ensure reliable operation.
. Test functions are for manufacturing tests only
SSD1
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.
1999 Dec 2011
Page 12
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
6.1.9POWER-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
. It should
DDD
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.
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
1999 Dec 2012
Page 13
This text is here in white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here in
ha
_white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here inThis text is here in
white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader. white to force landscape pages to be ...
1999 Dec 2013
from ADC
2
S1
I
2
S2
I
222
LEVEL ADJUST
2
FILTER
2
2
DC
LEVEL ADJUST
LEVEL ADJUST
2
2
4
6
8
ndbook, full pagewidth
DIGITAL
CROSSBAR
SELECT
2
2
2
2
MATRIX
MATRIX
MATRIX
MATRIX
AUTOMATIC
VOLUME
LEVEL
VOLUME
SOFT-MUTE
BASS/TREBLE
BEEPER
LEVEL ADJUST AND MUTE
LEVEL ADJUST AND MUTE
BASS/TREBLE
BASS BOOST
SPATIAL
PSEUDO
VOLUME
CONTOUR
SOFT-MUTE
BEEPER
2
2
2
2
Main
Auxiliary
I
I
6.2Digital signal processing
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
2
S1
2
S2
FM
2
DC
FILTER
24
ADAPTIVE
DE-EMPHASIS
FIXED
DE-EMPHASIS
LEVEL ADJUST
MATRIX
2
10
14
Fig.5 DSP data flow diagram.
MATRIX
DETECTION
MONITOR
SELECT
PEAK
LEVEL ADJUST
2
1
I
MHB112
DAC
2
C-bus
Page 14
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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.
6.2.2FM (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 and 75 µs).
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.
6.2.3MONITOR
This function provides data words from a number of
locations of 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.
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.
6.2.4LOUDSPEAKER (MAIN) CHANNEL
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 dBs (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 dBs (e.g. the
I2C-bus data byte +8 sets the new value to +8 dB).
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. 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.
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.
1999 Dec 2014
6.2.5HEADPHONE (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).
Page 15
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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.
6.2.6FEATURE 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.
6.2.8CHANNEL TO THE ANALOG CROSSBAR PATH
Level adjust with control positions 0, +3, +6 and +9 dB.
6.2.9DIGITAL CROSSBAR SWITCH
Input channels to the crossbar switch are from the audio
ADC, I2S1, I2S2, FM path and from the loudspeaker
channel path after matrix and AVL (see Fig.6).
Outputchannelscomprise loudspeaker, headphone, I2S1,
I2S2 and the audio DACs for line output and SCART.
TheI2S1andI2S2outputsalsoprovidedigitaloutputsfrom
the loudspeaker and headphone channels, but without the
beeper signals.
6.2.10SIGNAL 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.
6.2.11EXPERT MODE
The TDA9870A 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.
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.
6.2.7CHANNEL 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.
1999 Dec 2015
Page 16
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
6.2.12DSP FUNCTIONS
Table 4 Overview of DSP functions
FUNCTION
Bass control for loudspeaker and
headphone output
Treble control for loudspeaker and
headphone output
Contour for loudspeaker outputyescontrol range0 to +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 the
Level adjust I2S1 and I2S2 outputsyescontrol range−15 to +15dB
Level adjust analog crossbar pathnocontrol positions0, 3, 6 and 9dB
2
S1 and I2S2 inputsyescontrol range−15 to +15dB
EXPERT
MODE
yescontrol range−12 to +15dB
resolution1dB
resolution at frequency40Hz
yescontrol range−12 to +12dB
resolution1dB
resolution at frequency14kHz
resolution1dB
resolution at frequency40Hz
resolution2dB
resolution at frequency20Hz
corner frequency350Hz
nocontrol range−83 to +24dB
resolution1dB
mute position at step10101100
noprocessing time32ms
yesbeep frequenciessee Section 10.3.38
control range0 to −93dB
resolution3dB
mute position at step00100000
AVL output level for an input level
between 0 and −29 dB full-scale
attack time10ms
decay time constant2, 4 and 8s
−1 dB bandwidth of DSP14.5kHz
resolution1dB
resolution1dB
mute position at step00010000
PARAMETERVALUEUNIT
−23dB
1999 Dec 2016
Page 17
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
FUNCTION
EXPERT
MODE
PARAMETERVALUEUNIT
Level adjust audio ADC outputsyescontrol 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
2
−3 dB
2
−3 dB
2
2
D
2
2
2
A
ANALOG
CROSSBAR
SWITCH
2
2
2
2
ANALOG
MATRIX
ANALOG
MATRIX
ANALOG
MATRIX
A
D
3 dB
22
0 dB
3 dB
2
0 dB
3 dB
2
0 dB
2
SCART 1
2
SCART 2
2
Line output
FM
2
I
S1
2
S2
I
2
S1
I
2
S2
I
2
2
2
2
2
DSP
AND
DIGITAL
CROSSBAR
SWITCH
2
D
A
2
D
A
2
Main
2
Auxiliary
MHB113
Fig.6 Block diagram for the audio section.
1999 Dec 2017
Page 18
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
6.3.1ANALOG CROSSBAR SWITCH AND ANALOG MATRIX
There are a number of analog input and output ports with
the TDA9870A (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 TDA9870A 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 5.
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 TDA9870A, 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 maximum
1.4 V input), there are 3 and 0 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)TDA9870A
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
loudspeakerandtotheheadphoneoutput channels via the
digital crossbar switch and the DSP.
6.3.7DUAL AUDIO DAC
The TDA9870A 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 TDA9870A for the
connection of the analog crossbar switch section to the
DSP. The dual audio ADC consists of two bitstream
3rd-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 a copper plane close to the pins.
1999 Dec 2019
Page 20
This text is here in white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here in
a
_white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here inThis text is here in
white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader. white to force landscape pages to be ...
1999 Dec 2020
SCART 1
ndbook, full pagewidth
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
FM/AM
part
I2S1
2
I
S2
SCART 2
external
mono
FM/AM
DEMODULATOR
ADAPTIVE
DE-EMPHASIS
FIXED
DE-EMPHASIS
ADC
−6 dB
STEREO
DECODER
ADC
LEVEL
ADJUST
FM
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
S2
I
BUFFER
0/+3 dB
BUFFER
0/+3 dB
BUFFER
0/+3 dB
Line
SCART 1
SCART 2
MHB114
Fig.8 Audio signal flow diagram.
Page 21
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
7LIMITING VALUES
In accordance with the Absolute Maximum Rating System (IEC 134).
SYMBOLPARAMETERCONDITIONSMIN.MAX.UNIT
V
DD
∆V
DD
V
n
I
DDD,ISSD
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 ambient in free air
TDA9870A (SDIP64)40K/W
TDA9870AH (QFP64)50K/W
1999 Dec 2021
Page 22
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
9CHARACTERISTICS
V
V
parameters in accordance with system A2; 1 kΩ measurement source 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
= 1 kHz; FM sound
mod
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
=5.0V5368 83mA
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)TDA9870A
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
Power fail detector
V
th(pf)
Digital inputs and outputs
INPUTS
CMOS level input, pull-down (pins TEST1 and TEST2)
LOUDSPEAKER (MAIN) AND HEADPHONE (AUXILIARY) OUTPUTS
V
o(clip)(rms)
R
o
R
L(AC)
R
L(DC)
C
L
V
offset(DC)
α
mute
G
ro(main,aux)
PSRR
main,aux
SCART OUTPUTS AND LINE OUTPUT
V
o(nom)(rms)
V
o(clip)(rms)
R
o
R
L(AC)
R
L(DC)
C
L
V
offset(DC)
α
mute
Bbandwidthfrom SCART, external and
PSRRpower supply ripple rejectionf
clipping level output voltage
THD < 3%12501400−mV
(RMS value)
output resistance150250375Ω
AC load resistance10−−kΩ
DC load resistance10−−kΩ
load capacitance−1012nF
static DC offset voltage−3070mV
mute suppressionnominal input signal from
80−−dB
any source; fi= 1 kHz
roll-off gain at 14.5 kHz for
from any source−3−2−dB
Main and Auxiliary channels
power supply ripple rejection
for Main and Auxiliary
channels
f
= 70 Hz;
ripple
V
= 100 mV (peak);
ripple
C
=47µF;
Vref
4045−dB
signal from I2S-bus
nominal level output voltage
3 dB amplification−500−mV
(RMS value)
clipping level output voltage
THD < 3%12501400−mV
(RMS value)
output resistance150250375Ω
AC load resistance10−−kΩ
DC load resistance10−−kΩ
load capacitance−− 2.5nF
static DC offset voltageoutput amplifiers at 3 dB
−3050mV
position
mute suppressionnominal input signal from
80−−dB
any source; fi= 1 kHz
20−−kHz
mono sources;
−3 dB bandwidth
from DSP sources;
14.5−−kHz
−3 dB bandwidth
= 70 Hz;
ripple
V
= 100 mV (peak);
ripple
C
=47µF;
Vref
4045−dB
signal from I2S-bus
1999 Dec 2026
Page 27
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
Audio performance
THD + Ntotal harmonic distortion plus
noise
V
fi= 1 kHz; bandwidth
20 Hz to 15 kHz; note 8
S/Nsignal-to-noise ratioreference voltage
V
“CCIR468”
note 8
α
ct
crosstalk attenuationbetween any analog input
pairs; fi= 1 kHz
between any analog output
pairs; f
α
cs
channel separationbetween left and right of any
input pair
between left and right of any
output pair
G
A
gain from SCART-to-SCART
with −3 dB input voltage
divider
output amplifier in 3 dB
position; R
output amplifier in 0 dB
position; R
Crystal specification (fundamental mode)
f
xtal
C
L
C
1
C
0
Φ
pull
crystal frequency−24.576−MHz
load capacitance−20−pF
series capacitance−20−fF
parallel capacitance−− 7pF
pulling sensitivityCLchanged from
18 to 16 pF
R
R
R
N
equivalent series resistanceat nominal frequency−− 30Ω
equivalentseries resistance of
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 6 and 7.
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
AGCoffandset AGC to the gain step found. Measure the 1 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. 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.
5. 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.
6. If the supply voltage for the TDA9870A 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.
7. 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 TDA9870A, 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 and 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).
8. 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
This text is here in white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here in
_white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here inThis text is here in
white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader. white to force landscape pages to be ...
1999 Dec 2029
Table 6 Level setting FM at 0 dB (full-scale) = 1.4 V (RMS); note 1
2 CHANNEL
FM
STANDARD
M15 kHz deviation −24 dB (full-scale);
B/G27 kHz deviation −19 dB (full-scale)5.5 MHz5.742 MHzon50 µs+4dB
D/K27 kHz deviation −19 dB (full-scale)6.5 MHz6.742 MHzon50 µs+4dB
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.
Table 7 Level setting SAT FM at 0 dB (full-scale) = 1.4 V (RMS)
TRANSMITTER
NOMINAL
MODULATION
DEPTH
27 kHz deviation −19 dB (full-scale)6.5 MHz6.25 MHzon50 µs+4dB
27 kHz deviation −19 dB (full-scale)6.5 MHz5.742 MHzon50 µs+4dB
NOMINAL LEVEL AT
DEMODULATOR
OUTPUT
note 2
CARRIER 1
FREQUENCY
4.5 MHz4.724 MHzon75 µs+9dB
FREQUENCYIDENT
CARRIER 2
DE-EMPHASISOF
CARRIER 1 AND
CARRIER 2
FM LEVEL
ADJUSTSETTING
OF CARRIER 1
AND CARRIER 2
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
TRANSMITTER
SOURCE
SAT FM, stereo 50 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
DACGAIN
SETTING
OUTPUT
BUFFER
NOMINAL SCART
OUTPUT VOLTAGE
Page 30
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
10 I2C-BUS CONTROL
10.1Introduction
The TDA9870A 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 8).
Table 8 Possible slave addresses
ADDR2ADDR1
LOW LOW 1011000
LOWHIGH1011001
HIGHLOW1011010
HIGHHIGH1011011
The I2C-bus interface remains operational in the standby
mode of the TDA9870A 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 TDA9870A 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, 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 TDA9870A into the proper mode
1999 Dec 2030
Page 31
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
10.3Slave receiver mode
As a slave receiver, the TDA9870A 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 9 I
SSLAVE ADDRESS0ACKSUBADDRESSACKDATAACKP
Table 10 Explanation of Table 9
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 TDA9870A. 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 11 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 and 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 2031
Page 32
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
Table 12 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
12000lllllchannel 1 output level adjust
13000lllllchannel 2 output level adjust
1400000000set to logic 0; note 1
1500000000set to logic 0; note 1
1600000000set to logic 0; note 1
1700000000set to logic 0; note 1
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 2032
Page 33
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
SUBADDRESS
(DECIMAL)
MSBLSB
38000iiiiiI
39000oooooI
400mmm0 s s sI
41000iiiiiI
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
Note
1. These bits have not been assigned to a function.
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 13 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 14).
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, subaddress 0).
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, subaddress 0). P1OUT is
recommended to be used for switching an SIF trap for the adjacent picture carrier
in designs that employ such a trap.
5STDBY1The TDA9870A 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 TDA9870A is in 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 TDA9870A 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
TDA9870A is in 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 11000000.
1999 Dec 2035
Page 36
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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
(subaddresses 5 and 6, see Section 10.4), the current peak level will be reset.
Table 16 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 17)
3B3
2B2−signal source (see Table 18)
1B1
000DC output of FM demodulator
001audio magnitude/phase, FM demodulator output
010crossbar input from FM/AM channel
011don’t care
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.
1999 Dec 2036
Page 37
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 TDA9870A (data = 7509333 in decimal
notation or 729555 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 00000000 for all three
bytes.
Most significant part at subaddress 3 and least significant
part at subaddress 5 (see Table 19).
Table 19 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)
1999 Dec 2037
Page 38
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 the second FM sound carrier of a terrestrial or satellite FM program.
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 20 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 22)
3CH2MOD1−channel 2 receive mode: these bits control the hardware for the second sound
2CH2MOD0
1FILTBW0−selects filter bandwidth (see Table 22)
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 21)
Korea
0selects frequencies for Europe (B/G and D/K standard)
carrier (see Table 23)
0FM mode is assumed; this applies to both terrestrial and satellite FM reception
Note
1. The default setting at power-up is 00000000.
Table 21 Identification mode
B7B6IDENT MODE
00slow
01medium
10fast
11off/reset
Table 22 Filter bandwidth channel 1 and channel 2
FILTER
B4B1
00narrownarrowrecommended for nominal terrestrial broadcast conditions and
01extrawidenarrowrecommended 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)
1999 Dec 2038
Page 39
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
Table 23 Channel 2 receive mode
B3B2CHANNEL 2
00FM
01AM
10don’t care
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 24 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 25).
5B5
4B4
3ADEEM11Activates the adaptive de-emphasis function, which is required for certain satellite
2B2−Time constant selection for FM de-emphasis (see Table 26).
1B1
0 (LSB)B0
Notes
1. The default setting at power-up is 00000000.
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).
3. For stereo Korea the dematrix applies 6 dB attenuation (see Table 6).
1999 Dec 2040
Page 41
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
10.3.9FM CHANNEL 1 LEVEL ADJUST REGISTER
This register is used to correct for standard and station-dependent differences of signal levels.
Table 29 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 30 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 REGISTER 14
Set to logic 0. These bits have not been assigned to a function.
10.3.12 REGISTER 15
Set to logic 0. These bits have not been assigned to a function.
10.3.13 REGISTER 16
Set to logic 0. These bits have not been assigned to a function.
10.3.14 REGISTER 17
Set to logic 0. These bits have not been assigned to a function.
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 31 Subaddress 18 (note 1)
BITNAMEVALUEDESCRIPTION
7 (MSB)MUTI
6MUTI
2
S21mute I2S2 outputs
0normal I
2
S11mute I2S1 outputs
0normal I
2
S2 outputs
2
S1 outputs
5MUTDAC1mute internal DAC
0normal internal DAC
4MUTLINE1mute line outputs
0normal line outputs
3MUTSC21mute SCART 2 outputs
0normal SCART 2 outputs
2MUTSC11mute SCART 1 outputs
0normal SCART 1 outputs
1MUTAUX1mute Auxiliary outputs
0normal Auxiliary outputs
0 (LSB)MUTMAIN1mute Main outputs
0normal Main outputs
Note
1. The default setting at power-up is 11111111.
1999 Dec 2043
Page 44
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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.
Table 32 Subaddress 19 (note 1)
BITNAMEVALUEDESCRIPTION
7 (MSB)DACGAIN2−extra gain setting (see Table 33)
6B6−DAC output selection (see Table 34)
5B5
4B4
3DACGAIN1−extra gain setting (see Table 33)
2B2−signal source selection (see Table 35)
1B1
000FM leftFM right
001don’t care
010I
011I
100ADC leftADC right
101AVL leftAVL right
110don’t care
111don’t care
LEFTRIGHT
2
S1 leftI2S1 right
2
S2 leftI2S2 right
SIGNAL SOURCE
1999 Dec 2045
Page 46
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 36 Subaddress 20 (note 1)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6SC1GAIN1Activatesthe 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 37)
4B4
3B30default value
2B2−signal source selection (see Table 38)
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 39 Subaddress 21 (note 1)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6SC2GAIN1Activatesthe 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 unchanged (0 dB gain)
5B5−output channel selection (see Table 40)
4B4
3B30default value
2B2−signal source selection (see Table 41)
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 42 Subaddress 22 (note 1)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70set to logic 0
6LINGAIN1activates the 3 dB gain stage at the line output buffers
0the audio signal output will be unchanged (0 dB gain)
5B5−output channel selection (see Table43)
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
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 level of −6 dB full-scale at the
digital crossbar.
1999 Dec 2050
Page 51
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 47 Subaddress 24 (note 1)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6B6−output channel selection (see Table 48)
5B5
4B4
3B30default value
2B2−signal source selection (see Table 49)
1B1
10.3.22 AUDIO EFFECTS REGISTER
Switching the AVL off will reset the associated hardware to a defined state.
When the signal source for the Main channel is changed while the AVL is 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 52. There is a gain of 3 dB in the left audio channel.
Table 50 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 51). This function should be
activated only in accordance with the result of the sound mode identification.
signals in the Main channel (see Table 52). 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 53).
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
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 58 Subaddress 31 (note 1)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6B6−output channel selection (see Table 59)
5B5
4B4
3B30default value
2B2−signal source selection (see Table 60)
1B1
000FM input
001don’t care
010I
011I
100ADC input
101AVL input
LR+
------------- 2
2
S1 input
2
S2 input
LR+
------------- 2
1999 Dec 2059
Page 60
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 Table 65)
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 TDA9870A 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 00000000.
Table 65 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)
1999 Dec 2065
Page 66
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 66 Subaddress 37 (note 1)
BITNAMEVALUEDESCRIPTION
7 (MSB)B70default value
6B6−output selection (see Table 67)
5B5
4B4
3B30default value
2B2−signal source selection (see Table 68)
1B1
1. The Main and Auxiliary channel outputs will not contain the beeper signal.
1999 Dec 2066
Page 67
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 71 Subaddress 40 (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.
1999 Dec 2069
Page 70
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 78 Subaddress 45 (note 1)
BITNAMEVALUEDESCRIPTION
7 (MSB)B7−gain setting of right channel (see Table 79)
6B6
5B5
4B4
3B3−gain setting of left channel (see Table 80)
2B2
1B1
As a slave transmitter, the TDA9870A 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 81 General format for reading data from the TDA9870A
S SLAVE ADDRESS 0 ACK SUBADDRESS ACK SrSLAVE ADDRESS1ACKDATA NAm P
Table 82 Explanation of Tables 81 and 83
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
TDA9870A. 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 83 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
1999 Dec 2076
Page 77
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
Table 84 Overview of the slave transmitter registers; note 1
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.
1999 Dec 2077
Page 78
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
10.4.1DEVICE STATUS REGISTER
Table 85 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).
5B5−don’t care
4B4−don’t care
3B3−don’t care
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 TDA9870A has to be carried out to ensure a reliable
operation.
DDD2
, has
10.4.2REGISTER 1
Subaddress 1: These bits have not been assigned to a function. These bits are reserved for future extensions.
10.4.3REGISTER 2
Subaddress 2: These bits have not been assigned to a function. These bits are reserved for future extensions.
10.4.4REGISTER 3
Subaddress 3: These bits have not been assigned to a function. These bits are reserved for future extensions.
10.4.5REGISTER 4
Subaddress 4: These bits have not been assigned to a function. These bits are reserved for future extensions.
1999 Dec 2078
Page 79
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
10.4.6LEVEL 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 86 Subaddresses 5 and 6
SUB-
ADDRESS
57 (MSB) most significant bit or sign bit
67 (MSB)
10.4.7SIF 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 87 Subaddress 7
BITNAMEVALUEDESCRIPTION
7 (MSB)B7Xbit not assigned
6B6Xbit not assigned
5B5Xbit not assigned
4B4 −indication of SIF
3B3
2B2
1B1
0 (LSB)B0
10.4.8TEST REGISTER 3
This register contains, as a binary number, the highest
memory address used for the Coefficient RAM (CRAM,
expert mode).
Table 88 Subaddress 251
MSBLSB
B7B6B5B4B3B2B1B0
01111111
10.4.9T
This register contains, as a binary number, the highest
subaddress used for slave receiver registers.
Table 89 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 2079
Page 80
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
10.4.10 TEST REGISTER 1
This register contains, as a binary number, the highest
subaddress used for slave transmitter (status) registers.
Table 90 Subaddress 253
MSBLSB
B7B6B5B4B3B2B1B0
00000111
10.4.11 D
EVICE IDENTIFICATION CODE
There will be several devices in the digital TV sound
processorfamily.Thisbyteisusedtoidentifytheindividual
family members.
Table 91 Subaddress 254
MSBLSB
B7B6B5B4B3B2B1B0
00100010
10.4.12 S
OFTWARE IDENTIFICATION CODE
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 92 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 contains 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.
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.
As the coefficients do not fit into one data byte, they have
to be split and arranged (see Table 95). The most
significant bit is transferred first.
The general format described in Table 95 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 96.
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
receivedcoefficientsbebufferedandmadeactive 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 I
2
C-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.
1999 Dec 2080
Page 81
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
Table 93 General format for entering the expert mode and writing coefficients into the TDA9870A
SSLAVE
ADDRESS
Table 94 Explanation of Table 93
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 95 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 96 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 2081
Page 82
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
11 I2S-BUS DESCRIPTION
The feature interface of the TDA9870A 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
TDA9870A.
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 TDA9870A, 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 TDA9870A, 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 TDA9870A.
In such cases, the external source is the I2S-bus master
and the TDA9870A 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 TDA9870A. 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 2082
Page 83
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
handbook, full pagewidth
SCK
WS
SDLSB MSB
handbook, full pagewidth
SCK
WS
SDLSB MSB
LSB MSB
MGK112
one sample
a. I2S-bus format.
LSB MSB
MGK113
one sample
b. MSB-aligned format.
Fig.9 Serial audio interface formats.
1999 Dec 2083
Page 84
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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
C9
470 nF
10 kΩ
C3
100 nF
C6
1 µF
ADDR1
V
V
R1
ADDR2
V
V
CRESET
V
XTALO
SYSCLK
TEST1
MONOIN
TEST2
EXTIR
SCL
SDA
SSA1
DEC1
SIF2
V
ref1
SIF1
SSD1
DDD1
SSD4
XTALI
SCK
WS
SDO2
SDO1
SDI2
SDI1
EXTIL
i.c.
1 (57)
1
2 (58)
2
3 (59)
3
4 (60)
4
5 (61)
5
6 (62)
6
7 (63)
7
I
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 TDA9870AH version.
TDA9870A
(TDA9870AH)
(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
MHB596
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
LORi.c.
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
10 nF
47 µF
C16
47 µF
C15
47 µF
15 kΩ
C23
C21
R5
15 kΩ
R4
47 µF
10 nF
10 nF
47 µF
10 nF
10 nF
2.2 µF
270 Ω
2.2 µF
2.2 µF
2.2 µF
C19
C17
2.2 µF
R7
C13
330 nF
C11
330 nF
R19
1.5 Ω
C33
C30
C27
C22
10 nF
C34
2.2 µF
2.2 µF
2.2 Ω
2.2 µF
C20
2.2 µF
C18
2.2 µF
C12
330 nF
C10
330 nF
C32
R8
C25
C14
4.7 µF
+5 V
+5 V
Fig.10 Schematic for measurements.
1999 Dec 2084
Page 85
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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
i.c.
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 TDA9870AH 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 2086
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 87
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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
scale
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 2087
v M
A
B
E
17.45
16.95
LL
p
1.03
0.73
0.16 0.100.161.60
H
D
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 88
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
14 SOLDERING
14.1Introduction
Thistextgivesaverybriefinsighttoa 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-holeandsurfacemountcomponents 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 2088
Page 89
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 2089
Page 90
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
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 2090
Page 91
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9870A
NOTES
1999 Dec 2091
Page 92
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,isbelievedtobeaccurateand 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/pp92 Date of release: 1999 Dec 20Document order number: 9397 750 06064
Loading...
+ hidden pages
You need points to download manuals.
1 point = 1 manual.
You can buy points or you can get point for every manual you upload.