15DEFINITIONS
16LIFE SUPPORT APPLICATIONS
17PURCHASE OF PHILIPS I2C COMPONENTS
1998 Feb 132
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
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 21 dB control range
• SIF 8-bit Analog-to-Digital Converter (ADC)
• DQPSK demodulation for different standards,
simultaneously with 1-channel FM demodulation
• 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
• Optional AM 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
2
• I
S-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
• Dual audio digital-to-analog converter from
DSP to analog crossbar switch, bandwidth 15 kHz
• Dual audio ADC from analog inputs to DSP
• Two dual audio Digital-to-Analog Converters (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 TDA9875 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 TDA9875
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 standard. In other
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.
The AM sound of L/L’ standard is normally demodulated in
the 1st sound IF. The resulting AF signal has to be entered
into the mono audio input of the TDA9875. A second
possibility is to use the internal AM demodulator stage,
however this gives limited performance.
Korea has a stereo sound system similar to Europe and is
supported by the TDA9875. 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.
1. For other satellite systems, frequencies of, for example, 5.80, 6.60 or 6.65 MHz can also be received. A de-emphasis
of 60 µs, or in accordance with J17, is available.
2. m/st/d = mono or stereo or dual language sound.
3. Adaptive de-emphasis = compatible to transmitter specification.
PCLK1ONICAM clock output at 728 kHz
NICAM2Oserial NICAM data output at 728 kHz
ADDR13Ifirst I
SCL4II
SDA5I/OI
V
V
I
ref
SSA1
DDA1
6supplysupply ground 1; analog front-end circuitry
7supplyanalog supply voltage 1; analog front-end circuitry
8−resistor for reference current generator; analog front-end circuitry
P19I/Ofirst general purpose I/O pin
SIF210Isound IF input 2
V
ref1
11−reference voltage; analog front-end circuitry
SIF112Isound IF input 1
ADDR213Isecond I
V
V
SSD1
DDD1
14supplysupply ground 1; digital circuitry
15supplydigital supply voltage 1; digital circuitry
CRESET16−capacitor for power-on reset
XTALO17Ocrystal oscillator output
XTALI18Icrystal oscillator input
V
tune
19Otuning voltage output for crystal oscillator
P220I/Osecond general purpose I/O pin
SYSCLK21Osystem clock output
SCK22I/OI
WS23I/OI
SDO224OI
SDO125OI
SDI226II
SDI127II
TEST128Ifirst test pin; connected to V
MONOIN29Iaudio mono input
TEST230Isecond test pin; connected to V
EXTIR31Iexternal audio input right channel
EXTIL32Iexternal audio input left channel
SCIR133ISCART 1 input right channel
SCIL134ISCART 1 input left channel
V
SSG
35−ground guards; audio analog-to-digital converter circuitry
SCIR236ISCART 2 input right channel
SCIL237ISCART 2 input left channel
V
V
V
DDA2
ref(p)
ref(n)
38supplyanalog supply voltage 2; audio analog-to-digital converter circuitry
40−reference voltage ground; audio analog-to-digital converter circuitry
2
C-bus slave address modifier
2
C-bus clock
2
C-bus data
2
C-bus slave address modifier
2
S-bus clock
2
S-bus word select
2
S-bus data output 2
2
S-bus data output 1
2
S-bus data input 2
2
S-bus data input 1
for normal operation
SSD1
for normal operation
SSD1
1998 Feb 137
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
SYMBOLPINI/ODESCRIPTION
CAPL141−filter capacitor pin 1; audio analog-to-digital converter, left channel
CAPL242−filter capacitor pin 2; audio analog-to-digital converter, left channel
V
SSA2
CAPR244−filter capacitor pin 2; audio analog-to-digital converter, right channel
CAPR145−filter capacitor pin 1; audio analog-to-digital converter, right channel
V
ref2
SCOR147OSCART 1 output right channel
SCOL148OSCART 1 output left channel
V
SSD2
V
SSA4
SCOR251OSCART 2 output right channel
SCOL252OSCART 2 output left channel
V
ref3
PCAPR54−post-filter capacitor pin right channel, audio digital-to-analog converter
PCAPL55−post-filter capacitor pin left channel, audio digital-to-analog converter
V
SSA3
AUXOR57Oheadphone (Auxiliary) output right channel
AUXOL58Oheadphone (Auxiliary) output left channel
V
DDA3
MOR60Oloudspeaker (Main) output right channel
MOL61Oloudspeaker (Main) output left channel
LOL62Oline output left channel
LOR63Oline output right channel
V
6.1Description of the demodulator and decoder
section
6.1.1SIF
INPUT
Two input pins are provided, SIF1 e.g. for terrestrial TV
and SIF2 e.g. for a satellite tuner. As no specific filters are
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 from
Table 12 (subaddress 0).
The AGC can be controlled via the I
2
C-bus. Details can be
found in the I2C-bus register definitions (see Chapter 10).
6.1.3M
IXER
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
available for encoded satellite programs. A stereo decoder
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.
The result is available via the I2C-bus interface. A selection
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 PCLK (pin 1) and NICAM (pin 2).
A timing loop controls the frequency of the crystal oscillator
to lock the sampling rate to the symbol timing of the
NICAM data. The polarity of the control signal is selectable
to support applications in which external circuitry is used to
boost the tuning voltage of the oscillator.
6.1.7NICAM
DECODER
The device performs all decoding functions in accordance
with the
“EBU NICAM 728 specification”
. After locking to
the frame alignment word, the data is descrambled by
applying the defined pseudo-random binary sequence;
the device will then synchronize to the periodic frame flag
bit C0.
The status of the NICAM decoder can be read out from the
2
NICAM status register by the user (see the I
C-bus
register description in Section 10.4.2). The OSB bit
indicates that the decoder has locked to the NICAM data.
The VDSP bit indicates that the decoder has locked to the
NICAM data and that the data is valid sound data. The C4
bit indicates that the sound conveyed by the FM mono
channel is identical to the sound 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
bit errors
----------------------total bits
error byte 1.74×10
5–
×≈=
1998 Feb 1310
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
6.1.8NICAM AUTO-MUTE
This function is enabled by setting bit AMUTE LOW
subaddress 14 (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 (subaddresses 16 and
17; 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 or AM). When the error count is smaller than the lower
error limit the NICAM sound is restored.
The auto-mute function can be disabled by setting bit
AMUTE HIGH. 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.
When the AM sound in NICAM L systems is demodulated
in the 1st sound IF and the audio signal connected to the
mono input of the TDA9875, the controlling microcontroller
must implement the switching from NICAM reception to
mono input, if auto-muting is desired.
6.1.9CRYSTAL OSCILLATOR
A circuit diagram of the external components of the
voltage-controlled crystal oscillator is illustrated in Fig.8
(see Chapter 12).
6.1.10T
Both test pins are active HIGH, in normal operation of the
device they are wired to V
manufacturing tests only and are not available to
customers. Without external circuitry these pads are pulled
down to LOW level with internal resistors.
EST PINS
. Test functions are for
SSD1
1998 Feb 1311
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1998 Feb 1312
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
book, 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
SPA TIAL
PSEUDO
VOLUME
BASS BOOST
CONTOUR
SOFT-MUTE
BEEPER
2
2
2
2
2
LS
HP
2
S1
I
2
I
S2
DAC
6.2Description of the DSP
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
24
12
Fig.3 DSP data flow diagram.
16
MONITOR
SELECT
PEAK
DETECTION
1
2
I
C-bus
MGK108
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
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 of
±15 dB. 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 µs, 60 µs, 75 µs and J17).
An adaptive de-emphasis is available for
Wegener-Panda 1 encoded programs.
A matrix performs the dematrixing of1⁄2(L + R) and
R to L and R signals, of1⁄2(L + R) and1⁄2(L − R) to L and R
signals or of channel 1 and channel 2 to L and R signals.
6.2.4NICAM
AUTO-MUTE
If NICAM is received and the signal quality becomes poor,
the digital crossbar switch switches automatically to FM
and switches the matrix to channel 1. The automatic
switching between NICAM and channel 1 (FM or AM)
reception depends on the NICAM bit error rate.
The auto-mute function can be disabled via the I2C-bus.
6.2.5M
ONITOR
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 inputs to the loudspeaker channel
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.
6.2.6L
OUDSPEAKER (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 dBFS and −29 dB full-scale. There are some
fixed decay time constants to choose from, i.e. 2, 4 and
8 seconds.
Pseudo stereo is based on a phase shift in one channel via
a 2nd-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 dB 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
2
sent as an I
C-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 dB and +18 dB with 1 dB
resolution.
Bass is adjustable between +15 dB and −12 dB with 1 dB
resolution and treble is adjustable between ±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 dB and −93 dB with a 3 dB step resolution. The beeper
is not effected by mute.
1998 Feb 1313
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
Soft-mute provides a mute ability in addition to volume
control with a well defined time (32 ms) after which the
soft-mute is completed. A smooth fading is achieved by a
cosine masking.
6.2.7H
EADPHONE (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
2
sent as an I
C-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.
Bass is adjustable between +15 dB and −12 dB with 1 dB
resolution and treble is adjustable between ±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 dB and −93 dB with a 3 dB step 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 the
soft-mute is completed. A smooth fading is achieved by a
cosine masking.
6.2.8F
EATURE 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 ±15 dB range in 1 dB steps. Outputs can
be disabled to improve EMC performance.
The I2S-bus output matrix provides the following functions;
forced mono, stereo, channel swap, channel 1 and
channel 2.
One example of how the feature interface can be used in
a TV set is to connect an external Dolby Surround Pro
Logic DSP, such as the SAA7710, to the I
2
S-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.
6.2.9C
HANNEL FROM THE AUDIO ANALOG-TO-DIGITAL
CONVERTER
The signal level at the output of the ADC can be adjusted
in a range of±15 dB with a 1 dB step resolution. The audio
ADC itself is scaled to a gain of −6 dB.
6.2.10C
HANNEL TO THE ANALOG CROSSBAR PATH
Level adjust with control positions 0 dB, +3 dB, +6 dB
and +9 dB.
6.2.11D
IGITAL CROSSBAR SWITCH (SEE Fig.6)
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.
Output channels comprise loudspeaker, headphone, I2S1,
I2S2 and the audio DACs for line output and SCART.
The I2S1 and I2S2 outputs also provide digital outputs from
the loudspeaker and headphone channels, but without the
beeper signals.
6.2.12G
ENERAL
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.
1998 Feb 1314
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
6.2.13EXPERT MODE
The TDA9875 provides a special expert mode that gives direct write access to the internal Coefficient RAM (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 responses by
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 filter structures, the number of coefficients per function, their
default values, memory addresses, etc., can be made available on request.
6.2.14DSP
Table 5 DSP characteristics
Bass control for loudspeaker and
headphone output
Treble control for loudspeaker
and headphone output
Contour for loudspeaker output
Bass boost for loudspeaker
output
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 degree phase shift at frequency150, 200 and 300Hz
Beeper additional to the signal in
the loudspeaker and headphone
channel
AVL
General
CHARACTERISTICS
FUNCTION
EXPERT
MODE
yes
yes
yes
yes
no
no
yes
no
no
PARAMETERVALUEUNIT
control range−12 to +15dB
step resolution1dB
step resolution at frequency40Hz
control range−12 to +12dB
step resolution1dB
step resolution at frequency14kHz
control range0 to +18dB
step resolution1dB
step resolution at frequency40Hz
control range0 to +20dB
step resolution2dB
step resolution at frequency20Hz
corner frequency350Hz
control range−83 to +24dB
step resolution1dB
mute position at step10101100
processing time32ms
beep frequenciessee Section 10.3.38
control range0 to −93dB
step resolution3dB
mute position at step00100000
step widthquasi continuously
AVL output level for an input level
between 0 dB and −29 dB (full-scale)
attack time10ms
decay time constant2, 4 and 8s
−3 dB lower corner frequency of DSP 10Hz
−1 dB bandwidth of DSP14.5kHz
−23dB
1998 Feb 1315
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
FUNCTION
Level adjust I
Level adjust I
outputs
2
S1 and I2S2 inputs
2
S1 and I2S2
EXPERT
MODE
yes
yes
PARAMETERVALUEUNIT
control range−15 to +15dB
step resolution1dB
control range−15 to +15dB
step resolution1dB
mute position at step00010000
Level adjust analog crossbar pathnocontrol positions0, 3, 6 and 9dB
Level adjust audio ADC outputs
Level adjust NICAM path
Level adjust FM path
yes
yes
yes
control range+15 to −15dB
step resolution1dB
control range+15 to −15dB
step resolution1dB
control range+15 to −15dB
step resolution1dB
6.3Description of the analog audio section
handbook, full pagewidth
SCART 1
SCART 2
external
mono
NICAM
I
I
I
I
2
2
2
2
FM
S1
S2
S1
S2
2
−3 dB
2
−3 dB
2
2
D
2
2
2
A
2
2
2
2
2
2
ANALOG
CROSSBAR
SWITCH
DSP
AND
DIGITAL
CROSSBAR
SWITCH
2
2
2
2
2
2
ANALOG
MATRIX
ANALOG
MATRIX
ANALOG
MATRIX
A
D
D
A
D
A
3 dB
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.4 Block diagram for the audio section.
1998 Feb 1316
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
6.3.1ANALOG CROSSBAR SWITCH AND ANALOG MATRIX
(see also Fig.6)
There are a number of analog input and output ports with
the TDA9875. 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 TDA9875 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.
The analog matrix provides the functions given in Table 6
(see also Fig.5).
All switches and matrices are controlled via the I
2
C-bus.
There is one restriction for switching signals at inputs and
outputs for SCART 1 and SCART 2. At these ports, an
input signal cannot be copied to its own output, i.e. it is not
possible to make a copy from SCART 1 input to SCART 1
output.
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 TDA9875, 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 dB/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. If it is known for
certain applications that the input signal level is always
below 1.4 V (RMS), the SCART inputs can be used
without external resistors.
6.3.3E
XTERNAL AND MONO INPUTS
The 3 dB input attenuators are not required for the external
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 dB or to 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.
6.3.5L
INE 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).
1998 Feb 1317
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
6.3.6LOUDSPEAKER (MAIN) AND HEADPHONE
(AUXILIARY) OUTPUTS
Signals from any audio source can be applied to the
loudspeaker and to the headphone output channels via the
digital crossbar switch and the DSP.
6.3.7D
UAL AUDIO DAC
The TDA9875 contains three dual audio DACs, one for the
connection from the DSP to the analog crossbar switch
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.
6.3.8D
UAL AUDIO ADC
There is one dual audio ADC in the TDA9875 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.
6.3.9S
TANDBY MODE
The standby mode (subaddress 1, bit 5) disables most
functions and reduces power dissipation. The analog
crossbar switch and the SCART section remains
operational and can be controlled by the I2C-bus to
support copying of analog signals from SCART 1 to
SCART 2 and vice versa.
Unused internal registers may lose their information in
standby mode. Therefore, the device needs to be
initialized on returning to normal operation. This can be
accomplished in the same way as after a power-on reset.
1998 Feb 1318
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1998 Feb 1319
SCART 1
ndbook, full pagewidth
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
FM/AM
part
NICAM
part
I2S1
2
I
S2
SCART 2
external
mono
FM/AM
DEMODULATOR
NICAM
DECODER
ADAPTIVE
DE-EMPHASIS
SW4
FIXED
DE-EMPHASIS
DE-EMPHASIS
ADC
STEREO
DECODER
LEVEL
ADJUST
LEVEL
ADJUST
LEVEL
ADJUST
LEVEL
ADJUST
LEVEL
ADJUST
SW9
SW5
SW6
SW7
SW8
DIGITAL
MATRIX
DIGITAL
MATRIX
DIGITAL
MATRIX
DIGITAL
MATRIX
DIGITAL
MATRIX
AUTOMATIC
VOLUME
LEVEL
LEVEL
ADJUST
LEVEL
ADJUST
LEVEL
ADJUST
LOUDSPEAKER
PROCESSING
HEADPHONE
PROCESSING
DAC
CHANNEL
CHANNEL
SW3
SW1
SW2
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
MGK111
Fig.6 Audio signal flow diagram.
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
7LIMITING VALUES
In accordance with the Absolute Maximum Rating System (IEC 134).
SYMBOLPARAMETERCONDITIONSMIN.MAX.UNIT
V
DD
∆V
DD
I
IK
I
OK
I
O
I
, I
DDD
SSD
, I
I
DDA
SSA
I
lu(prot)
P/outpower dissipation per output−100mW
P
tot
T
stg
T
amb
V
es
DC supply voltage−0.5+6.5V
voltage differences between two VDD pins−550mV
DC input clamping diode currentVI< −0.5 V or
−±10mA
VI>VDD+ 0.5 V
DC output clamping diode current
(output type 4 mA)
DC output source or sink current
VO< −0.5 V or
−±20mA
VO>VDD+ 0.5 V
−0.5V<VO<VDD+ 0.5 V−±20mA
(output type 4 mA)
DC VDD or VSS current per digital supply pin−±100mA
DC VDD or VSS current per analog supply pin−±50mA
latch-up protection100−mA
total power dissipation−1.3W
storage temperature−55+125°C
operating ambient temperature−20+70°C
electrostatic handlingnote 12000−V
note 2200−V
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.
8THERMAL CHARACTERISTICS
SYMBOLPARAMETERCONDITIONSVALUEUNIT
R
th j-a
thermal resistance from junction to ambient in free air31K/W
1998 Feb 1320
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
9CHARACTERISTICS
V
=5V; T
DD
parameters in accordance with system A2; NICAM in accordance with
resistance for AF inputs; unless otherwise specified.
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
Digital supplies
V
DDD1
V
SSD1
I
DDD1
V
DDD2
V
SSD2
I
DDD2
Demodulator supplies and references
V
DDA1
V
SSA1
I
DDA1
V
ref1
I
ref1(sink)
Audio supplies and references
V
DDA2
V
SSA2
I
DDA2
V
DDA3
V
SSA3
I
DDA3
V
SSA4
V
SSG
V
ref2
=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
digital supply voltage 14.755.05.5V
digital supply ground 1−0.0−V
digital supply current 1V
=5.5V658095mA
DDD1
V
=5.0V557085mA
DDD1
digital supply voltage 24.755.05.5V
digital supply ground 2−0.0−V
digital supply current 2V
analog supply voltage for
=5.5V506580mA
DDD2
V
=5.0V455565mA
DDD2
4.755.05.5V
demodulator part
analog ground for demodulator
−0.0−V
part
analog supply current for
demodulator part
analog reference voltage for
V
=5.5V202631mA
DDA
V
=5.0V192428mA
DDA
referenced to V
DDA1/VSSA1
355065%
demodulator part
V
sink current170220260µA
ref1
analog supply voltage for audio
4.755.05.5V
ADC part
analog ground for audio ADC
−0.0−V
part
analog supply current for audio
ADC part
analog supply voltage for audio
V
= 5.5 V111417mA
DDA
V
=5.0V101316mA
DDA
4.755.05.5V
DAC part
analog ground for audio DAC
−0.0−V
part
analog supply current for audio
DAC part
analog ground for operational
V
= 5.5 V; digital silence 91214mA
DDA
V
= 5.0 V; digital silence 81113mA
DDA
−0.0−V
amplifier
ground, guard rings for
−0.0−V
analog-to-digital circuitry
reference voltage for audio
referenced to V
DDA2/VSSA2
−50−%
ADCs
1998 Feb 1321
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
Z
Vref2-VDDA2
Z
Vref2-VSSA2
V
ref3
Z
Vref3-VDDA3
Z
Vref3-VSSA3
Digital inputs and outputs
I
NPUTS
CMOS level input, high drive, pull-down (pins TEST1 and TEST2)
V
IL
V
IH
C
i
Z
i
CMOS level input, hysteresis, high drive, pull-up (pin CRESET)
V
IL
V
IH
V
hys
C
i
Z
i
INPUTS/OUTPUTS
impedance V
impedance V
reference voltage for audio DAC
pilot modulation for identification255075%
pilot sideband C/N for
identification start
−32−
dB
------ Hz
hysteresis−−2dB
identification windowB/G stereo
slow mode116.85−118.12Hz
medium mode116.11−118.89Hz
fast mode114.65−120.46Hz
B/G dual
slow mode273.44−274.81Hz
medium mode272.07 −276.20Hz
fast mode270.73−277.60Hz
total identification time ONslow mode−−2s
medium mode−−1s
fast mode−−0.5s
total identification time OFFslow mode−−2s
medium mode−−1s
fast mode−−0.5s
Analog audio inputs
M
ONO INPUT
V
i(nom)(rms)
nominal level input voltage
note 3−500−mV
(RMS value)
V
i(clip)(rms)
clipping level input voltage
THD < 3%; note 412501400−mV
(RMS value)
R
i
input resistancenote 4283542kΩ
1998 Feb 1324
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
SCART AND EXTERNAL INPUTS
V
i(nom)(rms)
V
i(clip)(rms)
R
i
Analog audio outputs
L
OUDSPEAKER (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
nominal level input voltage at
input pin (RMS value)
−3 dB divider with external
15 kΩ resistor;
−350−mV
notes 3 and 5
clipping level input voltage at
input pin (RMS value)
−3 dB divider with external
15 kΩ resistor; THD < 3%;
12501400−mV
notes 4 and 5
input resistancenote 4283542kΩ
clipping level output voltage
THD < 3%1400−−mV
(RMS value)
output resistance150250375Ω
AC load resistance10−−kΩ
DC load resistance10−−kΩ
output load capacitance−1012nF
static DC offset voltage−3070mV
mute suppressionnominal input signal from
80−−dB
any source; fi= 1 kHz;
note 3
)roll-off gain at 14.5 kHz for Main
from any source−3−2−dB
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; note 3−500−mV
(RMS value)
clipping level output voltage
THD < 3%−1400−mV
(RMS value)
output resistance150250375Ω
AC load resistance10−−kΩ
DC load resistance10−−kΩ
output 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;
note 3
1998 Feb 1325
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
B
line
PSRR
line
Audio performance
THD + Ntotal harmonic distortion plus
S/Nsignal-to-noise ratioreference voltage
α
ct
α
cs
G
A
line bandwidthfrom SCART, external,
Auxiliary and mono
sources; −3 dB bandwidth
from DSP sources;
−3 dB bandwidth
power supply ripple rejectionf
= 70 Hz;
ripple
V
= 100 mV (peak);
ripple
C
=47µF;
Vref
signal from I2S-bus
noise
V
fi= 1 kHz; bandwidth
i=Vo
= 1 V (RMS);
20 Hz to 15 kHz; note 1
from any analog audio
input to I
from I
2
S-bus
2
S-bus to any
analog audio output
SCART-to-SCART copy−0.10.3%
SCART-to-Main copy−0.20.5%
V
= 1.4 V (RMS);
o
fi= 1 kHz;
“CCIR468”
;
quasi peak; note 1
from any analog audio
input to I
from I
2
S-bus
2
S-bus to any
analog audio output
SCART-to-SCART copy7890−dB
SCART-to-Main copy7377−dB
crosstalk attenuationbetween any analog input
pairs; fi= 1 kHz
between any analog output
pairs; f
=10kHz
i
channel separationbetween left and right of
any input pair
between left and right of
any output pair
from SCART-to-SCART with
−3 dB input voltage divider
output amplifier in +3 dB
position;
R
=15kΩ±10%
ext
output amplifier in 0 dB
position;
R
=15kΩ±10%
ext
from external input to
=15kΩ±10%; note 1−1.50+1.5dB
R
ext
loudspeaker
20−−kHz
14.5−−kHz
4045−dB
−0.10.3%
−0.10.3%
7377−dB
7890−dB
70−−dB
65−−dB
65−−dB
60−−dB
−1.50+1.1dB
−4.5−3.0−1.9dB
1998 Feb 1326
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
VCXO and clock generation
VCXO
Crystal input
C
i
V
bias(DC)
Crystal output
V
osc(p-p)
V
bias(DC)
G
m
C
o
CRYSTAL SPECIFICATION (FUNDAMENTAL MODE)
f
xtal
C
L
C
1
C
0
Φ
pull
R
R
R
N
∆Ttemperature range−20+25+70°C
X
J
X
D
X
A
input capacitance−−10pF
DC bias voltageRi= 100 kΩ3.53.633.7V
oscillation amplitude
−1.4−V
(peak-to-peak value)
DC bias voltage2.02.42.8V
mutual conductance at
24.576 MHz
16.617.618.8
mA
---------
V
output capacitance−−10pF
crystal frequency−24.576−MHz
load capacitance−20−pF
series capacitance−20−fF
parallel capacitance−−7pF
pulling sensitivityCL changed from
18 to 16 pF
−25−
10
----------pF
6–
equivalent series resistanceat nominal frequency−−30Ω
equivalent series resistance of
2R
−−Ω
R
unwanted mode
adjustment tolerance−−±3010
driftacross temperature range−−±3010
ageing−−±5
−6
−6
10
-----------
year
6–
1998 Feb 1327
Philips SemiconductorsPreliminary specification
Digital TV Sound Processor (DTVSP)TDA9875
Notes to the characteristics
1. ADC level adjust = +6 dB, all other level adjusts = 0 dB, if 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.
2. Due to companding, the quantization noise of the NICAM system limits signal-to-noise ratio to 62 dB
(unweighted; RMS value).
3. Definition of nominal levels:
The full-scale level for analog audio signals is VFS= 1.4 V (RMS). The nominal level at the digital crossbar switch is
defined at −15 dB (full-scale).
a) Audio input nominal levels:
SCART: 350 mV (at pin); −12 dB (full-scale)
external, mono: 500 mV; −9 dB (full-scale)
b) FM/AM path nominal (maximum) levels:
system M: 15 kHz deviation; −23.7 dB (full-scale)
system B/G, D/K, I: 27 kHz deviation; −18.6 dB (full-scale)
SAT stereo (maximum): 50 kHz deviation; −13.3 dB (full-scale)
SAT mono (maximum): 85 kHz deviation; −8.7 dB (full-scale)
AM: 54% modulation; full-scale SIF ADC; −20 dB (full-scale)
c) NICAM path nominal (maximum) levels:
system B/G: −18.2 dB (full-scale)
system I: −22.8 dB (full-scale)
maximum level: 0.0 dB (full-scale).
4. If the supply voltage for the TDA9875 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.
5. 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 TDA9875, 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).
1998 Feb 1328
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