demodulation for different standards, simultaneously
with 1-channel FM demodulation
• Near Instantaneous Companded Audio Multiplex
(NICAM) decoding (B/G, I and L standard)
• Two-carrier multistandard FM demodulation
(B/G, D/K and M standard)
• Decoding for three analog multi-channel systems
(A2, A2+ and A2*) and satellite sound
• OptionalAM demodulation for system L, simultaneously
with NICAM
• Programmable identification (B/G, D/K and M standard)
and different identification times.
1.2DSP section
• Digital crossbar switch for all digital signal sources and
destinations
• Control of volume, balance, contour, bass, treble,
pseudo stereo, spatial, bass boost and soft mute
• Plop-free volume control
• Automatic Volume Level (AVL) control
• Adaptive de-emphasis for satellite
• Programmable beeper
• Monitor selection for FM/AM DC values and signals,
with peak detection option
• I2S-bus interface for a feature extension (e.g. Dolby Pro
Logic) with matrix, level adjust and mute.
• DualaudioDigital-to-AnalogConverter(DAC)fromDSP
to analog crossbar switch, bandwidth 15 kHz
• Dual audio ADC from analog inputs to DSP
• Two dual audio DACs for loudspeaker (Main) and
headphone (Auxiliary) outputs; also applicable for
L, R, C and S in the Dolby Pro Logic mode with feature
extension.
2GENERAL DESCRIPTION
The TDA9875A is a single-chip Digital TV Sound
Processor (DTVSP) for analog and digital multi-channel
sound systems in TV sets and satellite receivers.
2.1Supported standards
The multistandard/multi-stereo capability of the
TDA9875A is mainly of interest in Europe, but also in
Hong Kong/Peoples Republic of China and
South East Asia. This includes B/G, D/K, I, M and L
standards. In other application areas there exists only
subsets of these standard combinations otherwise only
single standards are transmitted.
M standard is transmitted in Europe by the American
Forces Network (AFN) with European channel spacing
(7 MHz VHF and 8 MHz UHF) and monaural sound.
The AM sound of L/L accent standard is normally
demodulated in the first sound IF. The resulting AF signal
has to be entered into the mono audio input of the
TDA9875A. A second possibility is to use the internal
AM demodulator stage, however this gives limited
performance.
1.3Analog audio section
• Analog crossbar switch with inputs for mono and stereo
(also applicable as SCART 3 input), SCART 1
input/output, SCART 2 input/output and line output
• User defined full-level/−3 dB scaling for SCART outputs
• Output selection of mono, stereo, dual A/B, dual A or
dual B
• 20 kHz bandwidth for SCART-to-SCART copies
• Standby mode with function for SCART copies
1999 Dec 203
Korea has a stereo sound system similar to Europe and is
supported by the TDA9875A. The differences include
deviation, modulation contents and identification. It is
based on M standard.
An overview of the supported standards and sound
systems and their key parameters is given in Table 1.
The analog multi-channel sound systems (A2, A2+ and
A2*) are 2-Carrier Systems (2CS).
The pin numbers given in parenthesis refer to the TDA9875AH version.
Fig.1 Block diagram.
1999 Dec 206
AUXOL
DAC (2)
DAC (2)
(50)
58
(49)
57
AUXOR
SUPPLY
SCART,
DAC,
ADC
(46) 54
(47) 55
(51) 59
(30) 38
(31) 39
(32) 40
(38) 46
(45) 53
(35) 43
(48) 56
(42) 50
MHB598
PCAPR
PCAPL
V
DDA
V
DEC2
V
ref(p)
V
ref(n)
V
ref2
V
ref3
V
SSA2
V
SSA3
V
SSA4
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
5PINNING
SYMBOL
PIN
TDA9875ATDA9875AH
PIN
TYPE
(1)
DESCRIPTION
PCLK157ONICAM clock output at 728 kHz
NICAM258Oserial NICAM data output at 728 kHz
ADDR1359II
SCL460II
SDA561I/OI
V
V
I
ref
SSA1
DEC1
662Ssupply ground 1; analog front-end circuitry
763−supply voltage decoupling 1; analog front-end circuitry
864−resistor for reference current generator; analog front-end circuitry
2
C-bus slave address input 1
2
C-bus clock input
2
C-bus data input/output
P191I/Ogeneral purpose input/output pin 1
SIF2102Isound IF input 2
V
ref1
113−reference voltage 1; analog front-end circuitry
MONOIN2921Iaudio mono input
TEST23022Itest pin 2; connected to V
for normal operating mode
SSD1
EXTIR3123Iexternal audio input right channel
EXTIL3224Iexternal audio input left channel
SCIR13325ISCART 1 input right channel
SCIL13426ISCART 1 input left channel
V
SSD3
3527Ssupply ground 3; digital circuitry
SCIR23628ISCART 2 input right channel
SCIL23729ISCART 2 input left channel
V
DEC2
3830−supply voltage decoupling 2; audio analog-to-digital converter
4638−reference voltage 2; audio analog-to-digital converter circuitry
SCOR14739OSCART 1 output right channel
SCOL14840OSCART 1 output left channel
V
V
SSD2
SSA4
4941Ssupply ground 2; digital circuitry
5042Ssupply ground 4; audio operational amplifier circuitry
SCOR25143OSCART 2 output right channel
SCOL25244OSCART 2 output left channel
V
ref3
5345−reference voltage 3; audio digital-to-analog converter and
operational amplifier circuitry
PCAPR5446−post-filter capacitor pin right channel; audio digital-to-analog
converter
PCAPL5547−post-filter capacitor pin left channel; audio digital-to-analog
converter
V
SSA3
5648Ssupply ground 3; audio digital-to-analog converter circuitry
AUXOR5749Oheadphone (Auxiliary) output right channel
AUXOL5850Oheadphone (Auxiliary) output left channel
V
DDA
5951Sanalog supply voltage; analog circuitry
MOR6052Oloudspeaker (Main) output right channel
MOL6153Oloudspeaker (Main) output left channel
LOL6254Oline output left channel
LOR6355Oline output right channel
V
DDD2
6456Sdigital supply voltage 2; digital circuitry
Notes
1. Pin type: I = input, O = output, S = supply.
2. Test pin: CMOS level input; pull-up resistor; can be connected to VSS.
3. Test pin: CMOS 3-state stage; can be connected to VSS.
6.1.1SIF INPUT
Two input pins are provided: SIF1 e.g. for terrestrial TV
and SIF2 e.g. for a satellite tuner. For higher SIF signal
levels the SIF input can be attenuated with an internal
switchable−10 dB resistor divider. As nospecificfiltersare
integrated, both inputs have the same specification giving
flexibility in application. The selected signal is passed
through an AGC circuit and then digitized by an 8-bit ADC
operating at 24.576 MHz.
6.1.2AGC
The gain of the AGC amplifier is controlled from the ADC
output by means of a digital control loop employing
hysteresis. The AGC has a fast attack behaviour to
prevent ADC overloads and a slow decay behaviour to
prevent AGC oscillations. For AM demodulation the AGC
must be switched off. When switched off, the control loop
is reset and fixed gain settings can be chosen
(see Table 15).
The AGC can be controlled via the I2C-bus. Details can be
found in the I2C-bus register definitions (see Chapter 10).
6.1.3MIXER
The digitized input signal is fed to the mixers, which mix
one or both input sound carriers down to zero IF. A 24-bit
control word for each carrier sets the required frequency.
Access to the mixer control word registers is via the
I2C-bus. When receiving NICAM programs, a feedback
signal is added to the control word of the second carrier
mixer to establish a carrier-frequency loop.
6.1.4FM AND AM DEMODULATION
An FM or AM input signal is fed via a band-limiting filter to
a demodulator that can be used for either FM or AM
demodulation. Apart from the standard (fixed)
de-emphasis characteristic, an adaptive de-emphasis is
availableforencodedsatelliteprograms.Astereodecoder
recovers the left and right signal channels from the
demodulated sound carriers. Both the European and
Korean stereo systems are supported.
6.1.5FM IDENTIFICATION
The identification of the FM sound mode is performed by
AM synchronous demodulation of the pilot signal and
narrow-band detection of the identification frequencies.
Theresultisavailableviathe I2C-businterface.Aselection
can be made via the I2C-bus for B/G, D/K and M standard
and for three different modes that represent different
trade-offs between speed and reliability of identification.
6.1.6NICAM DEMODULATION
The NICAM signal is transmitted in a DQPSK code at a bit
rate of 728 kbit/s. The NICAM demodulator performs
DQPSK demodulation and feeds the resulting bitstream
and clock signal onto the NICAM decoder and, for
evaluation purposes, to pins PCLK and NICAM.
Atimingloopcontrolsthefrequencyof the crystal oscillator
to lock the sampling rate to the symbol timing of the
NICAM data.
6.1.7NICAM DECODER
The device performs all decoding functions in accordance
with the
the frame alignment word, the data is descrambled by
applyingthedefinedpseudo-randombinarysequenceand
the device will then synchronize to the periodic frame flag
bit C0.
Bit VDSP (see Section 10.4.1) indicates that the decoder
has locked to the NICAM data and that the data is valid
sound data.
The status of the NICAM decoder can be read outfrom the
NICAM status register by the user (see Section 10.4.2).
Bit OSB indicates that the decoder has locked to the
NICAM data. Bit C4 indicates that the sound conveyed by
the FM mono channel is identical to the sound signal
conveyed by the NICAM channel.
The error byte contains the number of sound sample
errors, resulting from parity checking, that occurred in the
past 128 ms period. The Bit Error Rate (BER) can be
calculated using the following equation:
BER
“EBU NICAM 728 specification”
bit errors
----------------------total bits
error byte 1.74×10
×≈=
. After locking to
5–
1999 Dec 2011
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.1.8NICAM AUTO-MUTE
This function is enabled by setting bit AMUTE to logic 0
(see Section 10.3.11).
Upper and lower error limits may be defined by writing
appropriate values to two registers in the I2C-bus section
(see Sections 10.3.13 and 10.3.14). When the number of
errors in a 128 ms period exceeds the upper error limit the
auto-mute function will switch the output sound from
NICAM to whatever sound is on the first sound carrier
(FM orAM).Whentheerrorcountissmallerthanthelower
error limit the NICAM sound is restored.
The auto-mute function can be disabled by setting
bit AMUTE to logic 1. In this condition clicks become
audible when the error count increases; the user will hear
a signal of degrading quality.
A decision to enable/disable the auto-muting is taken by
the microcontroller based on an interpretation of the
application control bits C1, C2, C3 and C4 and, possibly,
any additional strategy implemented by the set maker in
the microcontroller software.
For NICAM L applications, it is recommended to
demodulate AM sound in the first sound IF and connect
the audio signal to the mono input of the TDA9875A.
By setting bit AMSEL (see Section 10.3.11), the
auto-mute function will switch to the audio ADC instead of
switching to the first sound carrier. The ADC source
selector (see Section 10.3.20) should be set to mono
input, where the AM sound signal should be connected.
6.1.9CRYSTAL OSCILLATOR
The circuitry of the crystal oscillator is fully integrated, only
the external 24.576 MHz crystal is needed (see Fig.10).
6.1.10TEST PINS
Bit CLRPOR (see Section 10.3.2) resets the Power-on
reset flip-flop to LOW. If this is detected, an initialization of
the TDA9875A has to be carried out to ensure reliable
operation.
6.1.12POWER-ON RESET
The reset is active LOW. In order to perform a reset at
power-up, a simple RC circuit may be used which consists
of the integrated passive pull-up resistor and an external
capacitor connected to ground. The pull-up resistor has a
nominal value of 50 kΩ, which can easily be measured
between pins CRESET and V
. Before the supply
DDD2
voltage has reached a certain minimum, the state of the
circuit is completely undefined, and it remains in this
undefined state unless a reset is applied.
The reset is guaranteed to be active when:
• The power supply is within the specified limits
(4.75 and 5.5 V)
• The crystal oscillator is functioning
• The voltage at pin CRESET is below 0.3V
V
= 5.0 V, typically below 1.8 V).
DDD
DDD
(1.5 V if
The required capacitor value depends on the gradient of
the rising power supply voltage. The time constant of the
RC circuit should be clearly larger than the rise time of the
power supply, to make sure that the reset condition is
always satisfied (see Fig.4), even considering the
tolerance spread. To avoid problems with a too slow
discharging of the capacitor at power-down, it may be
helpful to add a diode from pin CRESET to V
DDD
. It should
be noted that the internal ESD protection diode does not
help here as it only conducts at higher voltages. Under
difficult power supply conditions (e.g. very slow or
non-monotonic ramp-up), it is recommended to drive the
reset line from a microcontroller port or the like.
Test pins TEST1 and TEST2 are active HIGH and in the
normal operating mode of the device they are connected
to V
. Test functions are for manufacturing tests only
SSD1
and are not available to customers. Without external
circuitry these pins are pulled down to a LOW level with
internal resistors.
6.1.11POWER FAIL DETECTOR
The power fail detector monitors the internal power supply
for the digital part of the device. If the supply has
temporarily been lower than the specified lower limit, the
Power-onresetbit POR (see Section 10.4.1), will be set to
logic 1.
1999 Dec 2012
handbook, halfpage
5
voltage
(V)
1.5
V
> 4.75 V
DDD
V
CRESET
reset active
guaranteed
Fig.4 Reset at power-on.
MHB595
< 0.3V
DDD
t
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1999 Dec 2013
FM
2
DC
FILTER
from ADC
2
I
2
I
NICAM
ADAPTIVE
DE-EMPHASIS
S1
S2
2
DC
FILTER
2
2
2
FIXED
DE-EMPHASIS
FIXED
DE-EMPHASIS
LEVEL ADJUST
2
LEVEL ADJUST
LEVEL ADJUST
LEVEL ADJUST
LEVEL ADJUST
MATRIX
222
2
4
6
8
10
handbook, full pagewidth
DIGITAL
CROSSBAR
SELECT
2
2
2
2
2
MATRIX
MATRIX
MATRIX
MATRIX
MATRIX
AUTOMATIC
VOLUME
LEVEL
VOLUME
SOFT-MUTE
BASS/TREBLE
BEEPER
LEVEL ADJUST AND MUTE
LEVEL ADJUST AND MUTE
LEVEL ADJUST
BASS/TREBLE
BASS BOOST
SPATIAL
PSEUDO
VOLUME
CONTOUR
SOFT-MUTE
BEEPER
2
2
2
2
2
Main
Auxiliary
I
I
DAC
6.2Digital signal processing
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
2
S1
2
S2
24
12
16
Fig.5 DSP data flow diagram.
MONITOR
SELECT
PEAK
DETECTION
1
2
I
C-bus
MGK108
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.2.1LEVEL SCALING
All input channels to the digital crossbar switch (except for
the loudspeaker feedback path) are equipped with a level
adjust facility to change the signal level in a range from
+15 to −15 dB (see Fig.5). It is recommended to scale all
input channels to be 15 dB below full-scale (−15 dB
full-scale) under nominal conditions.
6.2.2NICAM PATH
The NICAM path has a switchable J17 de-emphasis.
6.2.3FM (AM) PATH
A high-pass filter suppresses DC offsets from the
FM demodulator due to carrier frequency offsets and
supplies the monitor/peak function with DC values and an
unfiltered signal, e.g. for the purpose of carrier detection.
The de-emphasis function offers fixed settings for the
supported standards (50, 60 or 75 µs and J17).
An adaptive de-emphasis is available for
Wegener-Panda 1 encoded programs.
A matrix performs the dematrixing of the A2 stereo, dual
and mono signals.
6.2.4NICAM AUTO-MUTE
If NICAM B/G, I or D/K is received, the auto-mute is
enabled and the signal quality becomes poor, the digital
crossbar switch switches automatically to FM and
switches the matrix to channel 1. The automatic switching
depends on the NICAM bit error rate.
The auto-mute function can be disabled via the I2C-bus.
For NICAM L applications, it is recommended to
demodulateAMsoundinthe first sound IF andconnectthe
audio signal to the mono input of the TDA9875A.
By setting bit AMSEL (see Section 10.3.11), the
auto-mute function will switch to the audio ADC instead of
switching to the first sound carrier. The ADC source
selector bits (see Section 10.3.20) should be set to mono
input, where the AM sound signal should be connected.
6.2.5MONITOR
Optionally, the peak value can be measured instead of
simply taking samples. The internally stored peak value is
reset to zero when the data is read via the I2C-bus.
The monitor function may be used, for example, for signal
level measurements or carrier detection.
6.2.6LOUDSPEAKER (MAIN) CHANNEL
The matrix provides the following functions: forced mono,
stereo, channel swap, channel 1, channel 2 and spatial
effects.
There are fixed coefficient sets for spatial settings of 30%,
40% and 52%.
The Automatic Volume Level (AVL) function provides a
constant output level of −23 dB (full-scale) for input levels
between 0 and −29 dB (full-scale). There are some fixed
decay time constants to choose from, i.e. 2, 4 and 8 s.
Pseudostereoisbasedonaphaseshiftinonechannelvia
a second-order all-pass filter. There are fixed coefficient
sets to provide 90 degrees phase shift at frequencies of
150, 200 and 300 Hz.
Volume is controlled individually for each channel ranging
from +24 to −83 dB with 1 dB resolution. There is also a
muteposition.Forthepurposeofasimplecontrolsoftware
in the microcontroller, the decimal number that is sent as
an I2C-bus data byte for volume control is identical to the
volume setting in dB (e.g. the I2C-bus data byte +10 sets
the new volume value to +10 dB).
Balance can be realized by independent control of the left
and right channel volume settings.
Contour is adjustable between 0 and +18 dB with 1 dB
resolution. This function is linked to the volume setting by
means of microcontroller software.
Bass is adjustable between +15 and −12 dB with 1 dB
resolution and treble is adjustable between
+12 and −12 dB with 1 dB resolution.
For the purpose of a simple control software in the
microcontroller, the decimal number that is sent as an
I2C-bus data byte for contour, bass or treble is identical to
the new contour, bass or treble setting in dB (e.g. the
I2C-bus data byte +8 sets the new value to +8 dB).
This function provides data words from a number of
locations in the signal processing paths to the I2C-bus
interface (2 data bytes). Signal sources include the
FM demodulator outputs, most inputs to the digital
crossbar switch and the outputs of the ADC. Source
selection and data read-out is performed via the I2C-bus.
1999 Dec 2014
Extra bass boost is provided up to 20 dB with 2 dB
resolution. The implemented coefficient set serves merely
as an example on how to use this filter.
The beeper provides tones in a range from approximately
400 Hz to 30 kHz. The frequency can be selected via the
I2C-bus. The beeper output signal is added to the
loudspeaker and headphone channel signals.
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
The beeper volume is adjustable with respect to full-scale
between 0 and −93 dB with 3 dB resolution. The beeper is
not effected by mute.
Soft mute provides a mute ability in addition to volume
control with a well defined time (32 ms) after which thesoft
mute is completed. A smooth fading is achieved by a
cosine masking.
6.2.7HEADPHONE (AUXILIARY) CHANNEL
The matrix provides the following functions: forced mono,
stereo, channel swap, channel 1 and channel 2
(or C and S in Dolby Surround Pro Logic mode).
Volume is controlled individually for each channel in a
range from +24 to −83 dB with 1 dB resolution. There is
also a mute position. For the purpose of a simple control
software in the microcontroller, the decimal number that is
sent as an I2C-bus data byte for volume control is identical
to the volume setting in dB (e.g. the I2C-bus data byte +10
sets the new volume value to +10 dB).
Balance can be realized by independent control of the left
and right channel volume settings.
Bass is adjustable between +15 and −12 dB with 1 dB
resolution and treble is adjustable between
+12 and −12 dB with 1 dB resolution. For the purpose of a
simple control software in the microcontroller, the decimal
number that is sent as an I2C-bus data byte for bass or
treble is identical to the new bass or treble setting in dB
(e.g. the I2C-bus data byte +8 sets the new value
to +8 dB).
The beeper provides tones in a range from approximately
400 Hz to 30 kHz. The frequency can be selected via the
I2C-bus. The beeper output signal is added to the
loudspeaker and headphone channel signals. The beeper
volume is adjustable with respect to full-scale between
0 and −93 dB with 3 dB resolution. The beeper is not
effected by mute.
Soft mute provides a mute ability in addition to volume
control with a well defined time (32 ms) after which thesoft
mute is completed. A smooth fading is achieved by a
cosine masking.
6.2.8FEATURE INTERFACE
The feature interface comprises two I2S-bus input/output
ports and a system clock output. Each I2S-bus port is
equipped with level adjust facilities that can change the
signal level in a range from +15 to −15 dB with 1 dB
resolution. Outputs can be disabled to improve EMC
performance.
One example of how the feature interface can be used in
a TV set is to connect an external Dolby Surround Pro
Logic DSP, such as the SAA7710, to the I2S-bus ports.
Outputs must be enabled and a suitable master clock
signal for the DSP can be taken from pin SYSCLK.
A stereo signal from any source will be output on one of
the I2S-bus serial data outputs and the four processed
signal channels will be entered at both I2S-bus serial data
inputs. Left and right could then be output to the power
amplifiers via the Main channel, centre and surround via
the Auxiliary channel.
6.2.9CHANNEL FROM THE AUDIO ADC
The signal level at the output of the ADC can be adjusted
in a range from +15 to −15 dB with 1 dB resolution.
The audio ADC itself is scaled to a gain of −6 dB.
6.2.10CHANNEL TO THE ANALOG CROSSBAR PATH
Level adjust with control positions 0, +3, +6 and +9 dB.
6.2.11DIGITAL CROSSBAR SWITCH
Input channels to the crossbar switch are from the audio
ADC, I2S1, I2S2, FM path, NICAM path and from the
loudspeaker channel path after matrix and AVL
(see Fig.8).
Outputchannelscomprise loudspeaker, headphone, I2S1,
I2S2 and audio DACs for line output and SCART. I2S1 and
I2S2 outputs also provide digital outputs from the
loudspeaker and headphone channels, but without the
beeper signals.
6.2.12SIGNAL GAIN
There are a number of functions that can provide signal
gain, e.g. volume, bass and treble control. Great care has
to be taken when using gain with large input signals in
order not to exceed the maximum possible signal swing,
which would cause severe signal distortion. The nominal
signal level of the various signal sources to the digital
crossbar switch should be 15 dB below digital full-scale
(−15 dB full-scale). This means that a volume setting of,
say, +15 dB would just produce a full-scale output signal
and not cause clipping, if the signal level is nominal.
Sending illegal data patterns via the I2C-bus will not cause
any changes of the current setting for the volume, bass,
treble, bass boost and level adjust functions.
1999 Dec 2015
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.2.13EXPERT MODE
The TDA9875A provides a special expert mode that gives
directwriteaccesstotheinternalCoefficientRAM(CRAM)
of the DSP. It can be used to create user-defined
characteristics, such as a tone control with different corner
frequencies or special boost/cut characteristics to correct
the low-frequency loudspeaker and/or cabinet frequency
responsesby means of the bass boost filter. However, this
mode must be used with great care.
6.2.14DSP FUNCTIONS
Table 5 Overview of DSP functions
FUNCTION
Bass control for loudspeaker and
headphone output
Treble control for loudspeaker and
headphone output
Contour for loudspeaker outputyescontrol range0to +18dB
Bass boost for loudspeaker outputyescontrol range0 to +20dB
Volume control for each separate
channel in loudspeaker and
headphone output
Soft mute for loudspeaker and
headphone output
Spatial effectsyesanti-phase crosstalk positions30, 40 and 52%
Pseudo stereoyes90 degrees phase shift at frequency150, 200 and 300Hz
Beeper additional to the signal in
resolution2dB
resolution at frequency20Hz
corner frequency350Hz
nocontrol range−83 to +24dB
resolution1dB
mute position at step1010 1100
noprocessing time32ms
yesbeep frequenciessee Section 10.3.38
control range0 to −93dB
resolution3dB
mute position at step0010 0000
AVL output level for an input level
between 0 and −29 dB (full-scale)
attack time10ms
decay time constant2, 4 and 8s
More information on the functions of this device, such as
the number of coefficients per function, their default
values, memory addresses, etc., can be made available
on request.
PARAMETERVALUEUNIT
−23dB
1999 Dec 2016
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
FUNCTION
EXPERT
MODE
PARAMETERVALUEUNIT
Generalno−3 dB lower corner frequency of DSP 10Hz
−1 dB bandwidth of DSP14.5kHz
Level adjust I
2
S1 and I2S2 inputsyescontrol range−15 to +15dB
resolution1dB
Level adjust I
2
S1 and I2S2 outputsyescontrol range−15 to +15dB
resolution1dB
mute position at step0001 0000
Level adjust analog crossbar pathnocontrol positions0, 3, 6 and 9dB
Level adjust audio ADC outputsyescontrol range+15 to −15dB
resolution1dB
Level adjust NICAM pathyescontrol range+15 to −15dB
resolution1dB
Level adjust FM pathyescontrol range+15 to −15dB
resolution1dB
6.3Analog audio section
handbook, full pagewidth
SCART 1
SCART 2
external
mono
NICAM
I
I
I
I
2
2
2
2
FM
S1
S2
S1
S2
2
−3 dB
2
−3 dB
2
2
D
2
2
2
A
2
2
2
2
2
2
ANALOG
CROSSBAR
SWITCH
DSP
AND
DIGITAL
CROSSBAR
SWITCH
2
2
2
2
2
2
ANALOG
MATRIX
ANALOG
MATRIX
ANALOG
MATRIX
A
D
D
A
D
A
3 dB
22
0 dB
3 dB
2
0 dB
3 dB
2
0 dB
2
SCART 1
2
SCART 2
2
Line output
2
Main
2
Auxiliary
MGK109
Fig.6 Block diagram for the audio section.
1999 Dec 2017
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.3.1ANALOG CROSSBAR SWITCH AND ANALOG MATRIX
There are a number of analog input and output ports with
theTDA9875A (see Figs 6 and 8). Analog source selector
switches are employed to provide the desired analog
signal routing capability. The analog signal routing is
performed by the analog crossbar switch section. A dual
audio ADC provides the connection to the DSP section
and a dual audio DAC provides the connection from the
DSP section to the analog crossbar switch. The digital
signal routing is performed by a digital crossbar switch.
The basic signal routing philosophy of the TDA9875A is
that each switch handles two signal channels at the same
time, e.g. left and right, language A and B, directly at the
source.
Each source selector switch is followed by an analog
matrix to perform further selection tasks, such as putting a
signal from one input channel, say language A, to both
output channels or for swapping left and right channels
(see Fig.7).
handbook, halfpage
left input
right input
ANALOG
MATRIX
left output
right output
MGK110
Fig.7 Analog matrix.
The analog matrix provides the functions given in Table 6.
6.3.2SCART INPUTS
The SCART specification allows for a signal level of up to
2 V (RMS). Because of signal handling limitations, due to
the 5 V supply voltage of the TDA9875A, it is necessary to
have fixed 3 dB attenuators at the SCART inputs to obtain
a 2 V input. This results in a −3 dB SCART-to-SCART
copy gain. If 0 dB copy gain is preferred (with a maximum
inputof 1.4 V), there are 0/3 dB amplifiers at the outputs of
SCART 1 and SCART 2 and at the line output.
The input attenuator is realized by an external series
resistor in combination with the input impedance, both of
which form a voltage divider. With this voltage divider the
maximum SCART signal level of 2 V (RMS) is scaled
down to 1.4 V (RMS) at the input pin.
6.3.3EXTERNAL AND MONO INPUTS
The3 dB input attenuators are not requiredfortheexternal
and mono inputs, because those signal levels are under
control of the TV designer. The maximum allowed input
level is 1.4 V (RMS). By adding external series resistors,
the external inputs can be used as an additional SCART
input.
6.3.4SCART OUTPUTS
The SCART outputs employ amplifiers with two gain
settings. The gain can be set to 3 or 0 dB via the I2C-bus.
The 3 dB position is needed to compensate for the 3 dB
attenuation at the SCART inputs should
SCART-to-SCART copies with 0 dB gain be preferred
[under the condition of 1.4 V (RMS) maximum input level].
The 0 dB position is needed, for example, for an
external-to-SCART copy with 0 dB gain.
All switches and matrices are controlled via the I2C-bus.
1999 Dec 2018
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
6.3.5LINE OUTPUT
The line output can provide an unprocessed copy of the
audio signal in the loudspeaker channels. This can be
either an external signal that comes from the dual audio
ADC, or a signal from an internal digital audio source that
comes from the dual audio DAC. The line output employs
amplifiers with two gain settings. The 3 dB position is
needed to compensate for the attenuation at the SCART
inputs, while the 0 dB position is needed, for example, for
non-attenuated external or internal digital signals
(see Section 6.3.4).
6.3.6LOUDSPEAKER (MAIN) AND HEADPHONE
(AUXILIARY) OUTPUTS
Signals from any audio source can be applied to the
loudspeakerandtotheheadphone output channels via the
digital crossbar switch and the DSP.
6.3.7DUAL AUDIO DAC
The TDA9875A contains three dual audio DACs, one for
theconnectionfromtheDSPtotheanalogcrossbarswitch
section and two for the loudspeaker and headphone
outputs. Each of the three dual low-noise high-dynamic
range DACs consists of two 15-bit DACs with current
outputs, followed by a buffer operational amplifier.
The audio DACs operate with four-fold oversampling and
noise shaping.
6.3.8DUAL AUDIO ADC
There is one dual audio ADC in the TDA9875A for the
connection of the analog crossbar switch section to the
DSP. The dual audio ADC consists of two bitstream
third-order sigma-delta audio ADCs and a high-order
decimation filter.
6.3.9STANDBY MODE
The standby mode, selected by setting bit STDBY to
logic 1 (see Section 10.3.2) disables most functions and
reduces power dissipation. The analog crossbar switch
and the SCART section remain operational and can be
controlled by the I2C-bus to support copying of analog
signals from SCART-to-SCART.
Unused internal registers may lose their information in the
standby mode. Therefore, the device needs to be
initialized on returning to the normal operating mode. This
can be accomplished in the same way as after a Power-on
reset.
6.3.10SUPPLY GROUND
The different supply grounds VSSare internally connected
via the substrate. It is recommended to connect all ground
pins by means of a copper plane close to the pins.
1999 Dec 2019
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1999 Dec 2020
SCART 1
handbook, full pagewidth
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
FM/AM
part
NICAM
part
I2S1
2
I
S2
SCART 2
external
mono
FM/AM
DEMODULATOR
NICAM
DECODER
ADAPTIVE
DE-EMPHASIS
FIXED
DE-EMPHASIS
DE-EMPHASIS
ADC
−6 dB
STEREO
DECODER
ADC
LEVEL
ADJUST
FM
LEVEL
ADJUST
NICAM
LEVEL
ADJUST
I2S1
INPUT
LEVEL
ADJUST
I2S2
INPUT
LEVEL
ADJUST
DIGITAL
MATRIX
DIGITAL
MATRIX
DIGITAL
MATRIX
DIGITAL
MATRIX
DIGITAL
MATRIX
AUTOMATIC
VOLUME
LEVEL
I2S1
OUTPUT
LEVEL
ADJUST
I2S2
OUTPUT
LEVEL
ADJUST
DAC
GAIN
LOUDSPEAKER
PROCESSING
HEADPHONE
PROCESSING
DAC
CHANNEL
CHANNEL
DAC
DAC
ANALOG
MATRIX
ANALOG
MATRIX
ANALOG
MATRIX
Main
Auxiliary
2
I
S1
2
I
S2
BUFFER
0/+3 dB
BUFFER
0/+3 dB
BUFFER
0/+3 dB
Line
SCART 1
SCART 2
MHB600
Fig.8 Audio signal flow diagram.
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
7LIMITING VALUES
In accordance with the Absolute Maximum Rating System (IEC 134).
SYMBOLPARAMETERCONDITIONSMIN.MAX.UNIT
V
DD
∆V
DD
V
n
I
, I
DDD
SSD
I
lu(prot)
P
tot
T
stg
T
amb
V
es
Notes
1. Human body model: C = 100 pF; R = 1.5 kΩ.
2. Machine model: C = 200 pF; L = 0.75 µH; R = 0 Ω.
DC supply voltage−0.5+6.0V
voltage differences between two VDD pins−550mV
voltage on any other pin−0.5VDD+ 0.5 V
DC current per digital supply pin−±180mA
latch-up protection current100−mA
total power dissipation−1.0W
storage temperature−55+125°C
ambient temperature−20+70°C
electrostatic handling voltagenote 1−2000+2000V
note 2−200+200V
8THERMAL CHARACTERISTICS
SYMBOLPARAMETERCONDITIONSVALUEUNIT
R
th(j-a)
thermal resistance from junction to ambientin free air
TDA9875A (SDIP64)40K/W
TDA9875AH (QFP64)50K/W
1999 Dec 2021
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
9CHARACTERISTICS
V
V
parameters in accordance with system A2; NICAM in accordance with
resistance for AF inputs; with external components of Fig.10; unless otherwise specified.
Supplies
V
V
I
V
V
I
V
V
V
I
V
V
V
V
Demodulator supply decoupling and references
V
V
I
Audio supply decoupling and references
V
V
Z
Z
V
Z
Z
= 300 mV; AGCOFF= 0; AGCSLOW = 0; AGCLEV = 0; level and gain settings in accordance with note 1;
SIF(p-p)
=5V; T
DD
=25°C; settings in accordance with B/G standard; FM deviation ±50 kHz; f
amb
“EBU specification”
= 1 kHz; FM sound
mod
; 1 kΩ measurement source
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
DDD1
SSD1
DDD1
DDD2
SSD2
DDD2
digital supply voltage 14.755.05.5V
digital supply ground 1note 2−0.0−V
digital supply current 1V
=5.0V5873 88mA
DDD1
digital supply voltage 24.755.05.5V
digital supply ground 2note 2−0.0−V
digital supply current 2V
= 5.0 V; system clock
DDD2
0.10.42mA
output disabled
SSD3
SSD4
DDA
DDA
digital supply ground 3note 2−0.0−V
digital supply ground 4note 2−0.0−V
analog supply voltage4.755.05.5V
analog supply current for
V
= 5.0 V; digital silence445668mA
DDA
DACpart
SSA1
analog ground for analog
note 2−0.0−V
front-end
SSA2
analog ground for audio ADC
note 2−0.0−V
part
SSA3
analog ground for audio DAC
note 2−0.0−V
part
SSA4
DEC1
analog ground for SCART−0.0−V
analog supply decoupling
3.03.33.6V
voltage for demodulator part
ref1
analog reference voltage for
−2−V
demodulator part
ref1(sink)
DEC2
sink current at pin V
ref1
analog supply decoupling
−200−µA
3.03.33.6V
voltage for audio ADC part
ref2
Vref2-VDEC2
Vref2-VSSA2
ref3
reference voltage ratio for
audio ADCs
impedance pins V
impedance pins V
ref2
ref2
to V
to V
reference voltage ratio for
audio DAC and operational
referenced to V
V
DEC2
SSA2
referenced to V
V
SSA2
SSA3
DEC2
DDA
and
and
−50−%
−20−kΩ
−20−kΩ
−50−%
amplifier
Vref3-VDDA
Vref3-VSSA3
impedance pins V
impedance pins V
ref3
ref3
to V
to V
DDA
SSA3
−20−kΩ
−20−kΩ
1999 Dec 2022
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
Power fail detector
V
th(pf)
Digital inputs and outputs
INPUTS
CMOS level input, pull-down (pins TEST1 and TEST2)
adjustment tolerance−− ±3010
driftacross temperature range−− ±3010
ageing−− ±5
Notes
1. Definitions of levels and level setting:
a) The full-scale level for analog audio signals is 1.4 V (RMS).
b) The nominal level at the digital crossbar switch is defined at −15 dB (full-scale).
c) Nominal audio input levels for external and mono: 500 mV (RMS) at −9 dB (full-scale).
d) See also Tables 7 and 8.
2. All analog and digital supply ground pins are connected internally.
3. Set demodulator to AM mode. Apply an AM carrier (with 1 kHz and 100%) to one channel. Check AGC step. Switch
AGC offandsetAGCtothegainstepfound.Measurethe1 kHz signal level of this channel and take it as a reference.
Switch to the other SIF input to which no signal is connected and which is terminated with 50 Ω. Now measure the
1 kHz crosstalk signal level. The SIF source resistance should be low (50 Ω).
4. NICAM in accordance with
“EBU specification”
. Audio performance is limited by the dynamic range of the NICAM728
system. Due to compansion, the quantization noise is never lower than −62 dB (unweighted RMS) with respect to
the input level.
5. FM source; in dual mode only A (respectively B) signal modulated; measured at B (respectively A) channel output;
Vo= 1 V (RMS) of modulated channel.
6. FM source; in stereo mode only L (respectively R) signal modulated; measured at R (respectively L) channel output;
Vo= 1 V (RMS) of modulated channel. The stereo channel separation may be limited by adjustment tolerances of
the transmitter.
7. If the supply voltage for the TDA9875A is switched off, because of the ESD protection circuitry, all audio input pins
are short-circuited. To avoid a short-circuit at the SCART inputs a 15 kΩ resistor (−3 dB divider) has to be used.
8. The SCART specification allows a signal level of up to 2 V (RMS). Because of signal handling limitations due to the
5 V supply voltage for the TDA9875A, there is a need for fixed 3 dB attenuators at the SCART inputs. To achieve
SCART-to-SCART copies with 0 dB gain, there are 3 dB/0 dB amplifiers at the outputs of SCART 1 and SCART 2
and at the line output. The attenuator is realized by an internal resistor that works together with an external series
resistor as a voltage divider. With this voltage divider the maximum SCART input signal level of 2 V (RMS) is scaled
down to 1.4 V (RMS) at the input pin. To avoid clipping, the 3 dB gain must not be used if the SCART input signal is
larger than 1.4 V (RMS).
9. ADC level adjust is 6 dB, all other level adjusts are 0 dB. If an external −3 dB divider is used set output buffer gain
to 3 dB, tone control to 0 dB, AVL off and volume control to 0 dB.
−6
−6
10
----------year
6–
1999 Dec 2028
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1999 Dec 2029
Table 7 Level setting FM, AM and NICAM at 0 dB (full-scale) = 1.4 V (RMS); note 1
TRANSMITTER
STANDARDMODE
M2 channel 15 kHz deviation−24 dB (full-scale);
B/G2 channel 27 kHz deviation−19 dB (full-scale)15.5 MHzFM−50 µs+4dB
NICAM−11.2 dB (full-scale) −18 dB (full-scale)15.5 MHzFM−50 µs+4dB
INICAM−15.8 dB (full-scale) −23 dB (full-scale)16.0 MHzFM−50 µs+4dB
D/K2 channel 27 kHz deviation−19 dB (full-scale)16.5MHzFM−50 µs+4dB
2 channel 27 kHz deviation−19 dB (full-scale)16.5 MHzFM−50 µs+4dB
2 channel 27 kHz deviation−19 dB (full-scale)16.5 MHzFM−50 µs+4dB
NICAM−11.2 dB (full-scale) −18 dB (full-scale)16.5 MHzFM−50 µs+4dB
L/L accentNICAM54% AM−19 dB (full-scale)16.5 MHzAM−50 µs+5dB
NOMINAL
MODULATION
DEPTH
NOMINALLEVELAT
DEMODULATOR
OUTPUT
note 2
FM/NICAM
CARRIER FREQUENCY MODE IDENT DE-EMPHASIS
14.5 MHzFM−75 µs+9dB
24.724 MHzFMon75 µs+9dB
25.742 MHzFMon50 µs+4dB
25.85 MHzNICAM offJ17+3 dB
26.552 MHzNICAM offJ17+8 dB
26.742 MHzFMon50 µs+4dB
26.25 MHzFMon50 µs+4dB
25.742 MHzFMon50 µs+4dB
25.85 MHzNICAM offJ17+3 dB
25.85 MHzNICAM offJ17+3 dB
LEVEL
ADJUST
Philips SemiconductorsProduct specification
Digital TV Sound Processor (DTVSP)TDA9875A
Notes
1. Nominal level at digital crossbar is defined at −15 dB (full-scale). DAC gain setting 6 dB. Output buffer setting 0 dB. Nominal SCART output level
500 mV (RMS).
2. For stereo signals the output level is 6 dB lower. The level adjust has to be increased by 6 dB.
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