The TDA9614H is an audio processing IC for VHS hi-fi and
linear audio, digitally controlled via the I2C-bus. The
FM (de)modulator and peak noise reduction functions are
highly integrated, resulting in few external components
and no external adjustments.
In addition special functions for audio mixing, dubbing and
descrambling have been implemented.
3ORDERING INFORMATION
TYPE NUMBER
TDA9614HQFP64
Note
1. When using IR reflow soldering it is recommended that the Drypack instructions in the
(order number 9398 510 63011) are followed.
1995 Mar 213
NAMEDESCRIPTIONVERSION
(1)
plastic quad flat package; 64 leads (lead length 1.95 mm);
body 14 × 20 × 2.8 mm
SDA14Data input/output for I2C-bus.
SCL15Clock input for I2C-bus.
V
SSD
RFCAGC17RF converter AGC-time constant.
RFCOUT18RF converter drive output.
LINE2L19Line 2 left output (e.g. decoder output).
LINE2R20Line 2 right output (e.g. decoder output).
LINEL21Line output left.
LINER22Line output right.
DCOUTL23VU meter drive output left.
DCOUTR24VU meter drive output right.
HPOUTL25Headphone drive output left.
HPOUTR26Headphone drive output right.
LINAGC27Linear audio AGC-time constant.
LININ28Audio input for linear audio to REC equalizer and output select.
LINOUT29Audio output from AGC or PB equalizer.
EQSW30Long Play (LP) equalization switch; 15 Ω on resistance and 150 kΩ input impedance.
RECEQ31Linear audio recording amplifier negative feedback input for connecting a record
LINREC32Digital output controlled by I2C-bit RN; can be used to drive an external (high voltage)
RECOUT33Linear audio recording amplifier output.
PBIN34Linear audio playback amplifier input; during playback the impedance is 100 kΩ;
RESSW35Long Play equalization switch 50 Ω on and 150 kΩ off impedance to PBIN.
PBDC36Linear audio playback amplifier DC decoupling.
13Digital supply voltage for I2C-bus (+5 V).
16Digital ground for I2C-bus.
equalization network.
head switch and possibly the bias oscillator.
during record the impedance is 7 Ω.
1995 Mar 215
Philips SemiconductorsPreliminary specification
Audio processor for VHS hi-fi
TDA9614H
and linear audio
SYMBOLPINDESCRIPTION
DCFBL37DC feedback left.
DCREFL38DC reference left.
EMPHL39Total emphasis left (240 to 20 µs).
RECTL40Rectifier DC decoupling left.
DETL41Attack/recovery timing left.
V
ref
V
SSA1
DETR44Attack/recovery timing right.
RECTR45Rectifier DC decoupling right.
EMPHR46Total emphasis right (240 to 20 µs).
DCREFR47DC reference right.
DCFBR48DC feedback right.
RFIX49Fixed bias current generation circuit by using an external 180 kΩ resistor to ground.
V
DDA1
AFNR51Audio output from noise reduction of right channel (record and loop-through) or audio
AFMR52Audio input for audio clipper of right channel (record and loop-through) or audio
RCCOR53Voltage-to-current transfer for right channel oscillator by means of an external resistor
V
SSA2
RBPF55Bias current generation for the internal band-pass filters by means of an external
FMOUT56FM output.
FMIN57FM input.
V
DDA2
RCCOL59Voltage-to-current transfer for left channel oscillator by means of an external resistor
ENVOUT60Level detector output (external capacitor required for filtering).
HID61Head Identification Pulse (HID) input for sample-and-hold circuits.
CMUTE62Mute timing (external capacitor required for playback mute).
AFML63Audio input for audio clipper of left channel (record and loop-through) or audio output
AFNL64Audio output from noise reduction of left channel (record and loop-through) or audio
42Noise filtering of 3.8 V reference voltage (external capacitor required for filtering).
43Analog ground 1 for LF circuits.
50Analog supply voltage 1 for LF circuits (+12 V).
input for noise reduction of right channel (playback).
output from sample-and-hold (playback).
to ground.
54Analog ground 2 for HF circuits.
resistor to ground.
58Analog supply voltage 2 for HF circuits (+5 V).
to ground.
from sample-and-hold circuit of left channel (playback).
input for noise reduction of left channel (playback).
6.1.1INPUT SELECT (INPUTSEL)
For the audio FM circuitry eight different audio signals can
be selected. The selection made here for hi-fi is also
available for the linear audio circuitry.
1. TUNER (stereo input; internal tuner of VCR).
2. CINCH (stereo input; connection to hi-fi set).
3. EXT1 (stereo input; TV SCART1).
4. EXT2 (stereo input; decoder, 2nd VCR SCART2).
5. EXT3 (stereo input; front CINCH for e.g. camcorder).
6. SAP (mono input; ‘Second Audio Program’ audio
carrier for NTSC or conventional FM carrier with
NICAM reception).
7. NORMAL (linear audio; [playback] signal of linear
audio).
When inserting a new video signal in an old
audio/video recording the hi-fi audio track is erased.
This setting can keep the current audio available on
the hi-fi track by copying the (playback) linear audio
signal to the hi-fi track during video insert.
8. EXT3/ST (input mix; mono EXT3 on left, mono hi-fi on
right channel).
For linear audio use see Section 6.1.2.
6.1.2NORMAL SELECT (NORMSEL)
For linear audio four settings can be selected:
1. INPUTSEL (mono signal from input select).
This is the ‘standard’ setting. The signal source
selected with the input select is led to the linear audio
circuit. The linear audio AGC should be switched on.
2. VOLUME (mono signal from [hi-fi] volume control).
The signal source selected with the input select has
now volume control, the linear audio AGC should be
switched off. This setting is for use with ‘audio
dubbing’: inserting a new audio signal on the linear
track in an old audio/video recording. A special audio
dubbing feature is now possible by selecting ‘input mix’
EXT3/ST in the input select. This setting enables us to
use the VCR as a mixer console: an audio source
connected to EXT3 (front CINCH) can be adjusted and
mixed together with the original hi-fi playback signal,
using the left (EXT3) and right (hi-fi playback) volume
controls.
TDA9614H
3. SAP (SAP input).
This setting enables simultaneous recording of the
NTSC SAP signal on the linear track and TUNER
stereo on the audio FM track. When receiving a
NICAM audio signal, the linear audio can be used as a
‘backup’ track when received television signals are
weak.
4. TUNER LEFT (left channel [language 1] of TUNER
input).
When receiving dual language transmissions the main
language (language 1) can be recorded on linear
audio. Note that if the SAP input is not used it can be
connected to the right TUNER channel, enabling
language selection for linear audio.
6.1.3VOLUME L/R; AGC
The volume controls are mainly intended for level
adjusting of the audio signals to be recorded on the hi-fi
track, however using the ‘VOLUME’ setting in the Normal
Select it can also be used to control linear audio. The
volume controls have a control range of +14 dB to −49 dB
in 1 dB steps and a full mute. Because the volume controls
are I2C-bus controlled their actual behaviour is defined by
the VCR’s software. For instance user control can be
‘volume only’, ‘left + right’, ‘volume + balance’ or the
setting can be defined by a ‘digital AGC’ software loop
using the signals at the DC (VU meter) outputs.
The linear audio level can be controlled by an AGC circuit,
which can be switched off when desired. In most cases the
AGC should be used at all settings of the Normal Select
except for ‘VOLUME’.
6.1.4AUDIO FM OUTPUT MUTE (AFOMUTE);
NORMAL OUTPUT MUTE (NOMUTE)
The audio output signal of the audio FM circuit can be
muted with AFOMute, the linear output signal can be
muted using NOMute. If one of these signals is not used
as an output (or input) signal it is best muted to further
minimize crosstalk.
6.1.5OUTPUT SELECT (OUTSEL)
This block is the ‘main’ output select function, possibly
functional on all outputs. Each output however has some
means to override this selection for its own output signal,
to implement extra features. Eight selections are possible
and they are shown in Table 1.
1995 Mar 219
Philips SemiconductorsPreliminary specification
Audio processor for VHS hi-fi
TDA9614H
and linear audio
Table 1Output select possibilities
SELECTIONREMARKDESCRIPTION
STEREO
LEFThi-fi leftaudio FM output signal left channel (language 1)
RIGHThi-fi rightaudio FM output signal right channel (language 2)
NORMALlinear audiolinear audio
NOR + STlinear and hi-fimix of audio FM stereo and linear audio
NOR + Llinear and leftmix of audio FM left channel and linear audio
NOR + Rlinear and rightmix of audio FM right channel and linear audio
MUTEmute
Note
1. STEREO is the ‘standard’ setting, LEFT and RIGHT are for language selection. Using the tuner as input or output
(1)
hi-fi stereoaudio FM output signal left and right channel
signal (record or loop-through of audio FM) the VCR can automatically switch from stereo to left (language 1) when
a ‘dual language’ transmission is received. If the VCR is in playback mode and no hi-fi track is present (checking bit
AFOM via the I2C-bus or using the level at pin 11) the VCR can switch the output select to NORMAL (or let the IC
do this automatically by setting bit AUTN). If a hi-fi track is present the user may want to switch to LEFT and RIGHT
if the recording made was a ‘dual language’ recording.
Furthermore the user may want to switch from a hi-fi selection to NORMAL, e.g. when a complete audio remix has
been created on the linear track using ‘audio dubbing’ (see Section 6.7). ‘Audio dubbing’ can also be used to
re-record the linear track with e.g. only commentary. Combined playback of this commentary together with the
original hi-fi sound is then possible by means of a mix mode.
6.1.6RF CONVERTER MUTE (RFCMUTE)
The RFC output is a mono output derived from the output select, its output signal can be muted.
6.1.7LINE SELECT (LINESEL)
The line output select is normally connected to both CINCH (hi-fi set) and television (SCART1). Normally the audio signal
from the output select will be available at this output, but three extra selections are possible and they are shown in
Table 2.
Table 2Extra line select possibilities
SELECTIONDESCRIPTION
NORMALlinear audio, useful for monitoring during ‘audio dubbing’
(1)
EXT2
(2)
EXT3
Notes
1. A signal from a source at EXT2 (laser disc or second VCR) can be connected to a television set at line out. Because
this is a direct connection from the input to the line output buffers it is fully independent of any mode setting of the
IC. Connection of a ‘Pay-TV’ decoder box to EXT2 and using this selection together with selecting EXT1 in the
decoder select enables combined use of the decoder box by both TV and VCR. In Europe, use of the SCART
connector ‘status line’ can automatize this switching.
2. Direct audio connection from input EXT3. If the line 2 outputs and EXT3 inputs are not used for external connections
a built-in karaoke unit or sound processor can be inserted between line 2 and EXT3.
audio from input EXT2
audio from input EXT3
1995 Mar 2110
Philips SemiconductorsPreliminary specification
+()
Z
------------------------------
×=
Audio processor for VHS hi-fi
and linear audio
6.1.8DECODER SELECT (DECODESEL)
The line 2 output is normally connected to a decoder box
or e.g. a second VCR (SCART2). Normally the audio
signal from the output select will be available at this output,
but three extra selections are shown in Table 3.
Table 3Extra decoder select possibilities
SELECTIONDESCRIPTION
TUNER
EXT1
EXT3audio from input EXT3
Notes
1. Enables the use of a decoder box (connected to line 2
2. Decoder box connection to television set
6.1.9HEADPHONE SELECT (HPSEL)
The headphone output will normally carry the output select
signal, however three more selections are possible and
shown in Table 4.
(1)
(2)
[EXT2]) by the VCR.
(See Section 6.1.7).
audio from input TUNER
audio from input EXT1
TDA9614H
both hi-fi audio channels, independent from the setting of
output or headphone select. For this mode the selection
STEREO can be made.
6.2Linear audio circuits
6.2.1RECORD/LOOP-THROUGH
The signal selected with the normal select can be level
controlled in the AGC block. If wanted this AGC can be
switched off using bit AGCN. The audio signal is thereafter
DC decoupled using a capacitor between pin 29 (LINOUT)
and pin 28 (LININ).
From here on the signal goes to the output select and (in
the event of record mode) to the record equalizer. The
record equalizer is a non-inverting amplifier with a gain of
+12 dB, consisting of an opamp with feedback resistors,
and acts as a pre-correction filter for the tape and head
characteristics. The inverting input of the opamp is
connected to pin 31 (RECEQ), its output to pin 33
(RECOUT). The circuit externally connected between
pin 31 and ground defines the transfer of the record
equalizer with equation:
Where: Z is the complex impedance of the external circuit.
A3.98
1200Z
Table 4Extra headphone select possibilities
SELECTIONDESCRIPTION
(1)
LEFT
(1)
RIGHT
NORMALlinear audio, useful for monitoring the
Note
1. The selections LEFT and RIGHT make it possible to
select a different language for the headphone as for
the line outputs. A possible implementation is to invert
always a language selection made at the output select.
6.1.10H
The headphone volume control has a control range of
+16 dB to −47 dB in 1 dB steps and a full mute.
6.1.11DC SELECT (DCSEL)
The VU meter output is normally connected to the
headphone select, but can also be switched to STEREO
(hi-fi stereo, before the output select). When using the
VU meter output signal to implement a ‘digital AGC’ (read
the level at the DC output, and adjust the VOLUME L/R
controls accordingly) it is necessary to have information of
left hi-fi channel
right hi-fi channel
recording by ‘audio dubbing’
EADPHONE VOLUME
Two equalizings can be used, switching additional circuitry
to ground with pin 30 (EQSW) in long play or extended
play (I2C-bus: LP = 1).
Instead of the hi-fi circuit, linear audio has two
loop-through modes:
• Loop-through playback
• Loop-through record.
The only difference between Loop-through record and
record is the RECOUT output, which is muted.
The LINREC output (pin 32) is used to drive the external
high-voltage switch and possibly the bias oscillator. The
LINREC voltage has a small internal delay to be
non-overlapping with the internal switching to playback
mode. This ensures that no bias voltage will appear at the
PBIN input while the IC is in playback mode.
6.2.2P
The signal coming from the head is amplified and can be
level controlled in 16 steps of 1 dB. This setting can be
used to adjust for spread or nominal difference in head
output level. Afterwards, the signal is frequency corrected
in the playback equalizer. This equalizer includes the two
standard VHS equalizing settings of 3180 and 120 µs
LAYBACK
1995 Mar 2111
Philips SemiconductorsPreliminary specification
Audio processor for VHS hi-fi
and linear audio
(standard play: LP = 0) and 3180 and 170 µs (Long Play:
LP = 1).
The signal is DC decoupled and routed to the output select
via pins 29 and 28. When switching to or from playback
the signal should be muted shortly using the Normal
Output Mute (NOM) to prevent audible clicks caused by
offset differences of the playback amplifier and AGC.
6.3Audio FM circuits
6.3.1RECORD/LOOP-THROUGH
After low-pass filtering and signal compression in the
Noise Reduction (NR) the audio signal is available at
AFNL and AFNR. Through a DC-decoupling capacitor the
compressed audio signal is fed to an audio clipper at
AFML and AFMR. Subsequently, the audio signal is
FM-modulated on a RF-carrier using an integrated
current-controlled oscillator (CCO). Each CCO (L and R)
requires an external resistor (RCCOL/R) for temperature
stability. Through automatic calibration both the oscillator
centre frequency and FM-deviation are adjusted. After
low-pass filtering and summation of the two RF-carriers,
the FM signal is available at FMOUT during record. During
loop-through the FMOUT pin is inactive.
The RAF pin reflects the status of the RAF-bit when it is
used as output pin. In this case it is meant to switch the
head amplifier between record and playback position. It
can also be used as an input pin to overrule the RAF-bit
thereby forcing the IC in record or playback/loop-through
position.
6.3.2PLAYBACK
The two FM modulated RF carriers, presented at FMIN,
are fed to an AGC circuit. The AGC operates on the peak
level of the added BPF output signals. As a result, the input
signal of the BPFs is kept constant for FMIN voltages
>30 mV (RMS value) both carriers added. In this way the
dynamic range of the integrated BPFs is optimally used
over a large input signal range. Subsequently, the
RF carriers are bandpass filtered using two integrated
BPFs. These bandpass filters are automatically calibrated
as described in Section 6.3.3. An external resistor (RBPF)
is required for temperature stability.
The filtered input signals are fed to a limiter. The limiter
output signal is demodulated by means of a
PLL demodulator. The LF audio signal is fed through a
sample-and-hold circuit to suppress head-switching noise.
The demodulated audio is available at AFML and AFMR.
Through a decoupling capacitor the audio signal is applied
to an audio low-pass filter at AFNL and AFNR. The
TDA9614H
low-pass filtered audio signal is expanded in the noise
reduction.
If the left channel carrier at FMIN falls below the mute level
of 4.2 mV (RMS value) the audio signal is muted in front of
the low-pass filter. This is achieved by comparing the left
channel level detector signal with an internal reference.
The mute timing is fixed by a capacitor at the CMUTE pin.
If the level at CMUTE rises above a certain threshold, the
PBMUTE bit is set. This will mute the audio signal. If bit
AUTN has been set, the output selection will automatically
switch to NORMAL mode. The left channel level detector
output is also available at pin ENVOUT. An AGC
correction voltage is added at this pin, such that the
ENVOUT voltage indicates the left channel FM-carrier
level even within the AGC-range. The ENVOUT voltage
can be used for auto-tracking of the FM-audio heads.
In the event of drop-outs in the incoming FM signal, the left
channel level detector is also used to activate the
sample-and-hold. This drop-out cancellation is only active
if bit DOC has been set.
6.3.3AUTOMATIC CALIBRATION
By means of bit CALS the two integrated oscillators and
bandpass filters can be calibrated in Loop-Through (LT) or
record (REC) mode. Normally this will be done after
start-up of the VCR. After setting of bit CALS the
calibration procedure is automatically executed as shown
in Fig.4. The calibration starts at the first negative edge of
HID after CALS has been set. During the LOW period of
HID (20 ms for PAL) the number of oscillator cycles is
compared to a certain value stored in a Read Only Memory
(ROM).
Therefore, the final oscillator frequency is proportional to
the inverse of the HID LOW-time. This means that an error
of 1 µs in the HID LOW-time will result in an additional
frequency error of approximately 100 Hz. The maximum
number of required HID cycles for one complete oscillator
calibration is six, which is equal to 240 ms.
After both oscillators have been calibrated the bandpass
filters are calibrated using the right channel oscillator as
reference frequency generator (1.6 MHz for PAL). This will
take <10 ms.
Bit CALR is set to 1 if the calibration has been executed
successfully. CALR is 0 if:
• A Power-On Reset (POR) has occurred.
• The calibration is running.
• The initial oscillator and bandpass frequencies are too
low or too high for a successful calibration.
1995 Mar 2112
Philips SemiconductorsPreliminary specification
Fig.4 Timing diagram of automatic calibration.
MBE281
max 240 ms
max 240 ms
max 10 ms
HID
CALS
LEFT
OSCILLATOR
RIGHT
OSCILLATOR
CALR
BANDPASS
FILTERS
W bit = 0, a subaddress byte and eight data bytes.
If more than one data byte follows the subaddress, these
bytes are stored in the successive registers by the
automatic address increment feature. As slave transmitter
(R/W bit = 1) there is one read byte.
Audio processor for VHS hi-fi
and linear audio
TDA9614H
If CALS is reset during CALR = 0, the calibration will be
stopped if it was not finished yet. If required, a new
calibration can be started using CALS.
WARNING: the IC should not be calibrated in a test or
standby mode.
6.3.4PAL/NTSC MODE
Depending on bit NTSC the IC is calibrated in PAL or
NTSC mode using bit CALS. After an automatic calibration
the oscillator frequencies can be switched between PAL
and NTSC using bit NTSC. This centre frequency shift is
done without automatic calibration. During playback (PB)
or loop-through (LT) a change in bit NTSC will
automatically calibrate the bandpass filters again. This will
take <10 ms.
1995 Mar 2113
WARNING: A change of PB to REC, LT to REC and
PAL to NTSC should not be combined in one transmitted
byte. The PAL to NTSC transition should be preceded by
the PB to REC or LT to REC transition.
6.4I2C-bus
The TDA9614H is controlled via the 2-wire I2C-bus, in
accordance with the I2C-bus specification. As slave
receiver for programming there is one module address,
with R/
Philips SemiconductorsPreliminary specification
Audio processor for VHS hi-fi
TDA9614H
and linear audio
Table 5TDA9614H addresses and data bytes
NAMEADDRESS
Slave address byte1011100R/
Subaddress bytes 00 to 0700000X
Control byte (subaddress 00)RAFIPAFAFOMNTSCRNIPNNOMLP
Main select byte
6.5Power-On Reset (POR); derived from digital supply voltage V
In the data byte descriptions [por] indicates the mode after POR. The status of the data bytes after POR is shown in
Table 6.
Table 6Status of data bytes after POR
DATA BYTEADDRESS
Control byte01100110
Main select byte00000110
Secondary select byte11000000
Left volume byte00110001
Right volume byte00110001
Headphone volume byte00101111
Install byte01011101
Test byte00000000
A POR occurrence is signalled by bit POR in the read byte (see Table 39).
DDD
1995 Mar 2114
Philips SemiconductorsPreliminary specification
Audio processor for VHS hi-fi
and linear audio
6.6Control byte
Table 7Bits of control byte
BITDESCRIPTION
RAFRecord Audio FM; see Table 8
IPAFInverse Playback Audio FM; see Table 8
AFOMAudio FM Output Mute; see Table 9
NTSCNTSC; television standard; see Table 10
RNRecord Normal; see Table 11
IPNInverse Playback Normal; see Table 11
NOMNormal Output Mute; see Table 12
LPLong Play mode; see Table 13
Table 8Bits RAF and IPAF
(1)
RAF
00playbackNR and modem in playback mode
01loop-throughNR in record mode; modem not active [por]
10record
11record
IPAFMODEDESCRIPTION
(2)
(2)
NR and modem in record mode
NR and modem in record mode
TDA9614H
Notes
1. The RAF bit can be overruled externally by applying a low-ohmic voltage to the RAF I/O (pin 12) either logic 0 or
logic 1 (0 or +5 V). The actual mode of the IC is determined by the level measured at this pin, enabling fast switching
between record and playback/loop-through.
2. The two record modes are equal, only differing in their reaction to forcing RAF LOW at the RAF I/O pin; the status of
the IPAF bit determines whether the IC is switched to the playback or loop-through mode.
Table 9Bit AFOM
AFOMMODEDESCRIPTION
0−−
1mute
Note
1. Audio coming from the audio FM circuit is muted, either the signal from tape in playback or the loop-through signal
in record or loop-through modes.
Table 10 Bit NTSC
NTSCMODEDESCRIPTION
0PALmodem and BPF set to PAL carrier frequencies [por]
1NTSCmodem and BPF set to NTSC carrier frequencies
(1)
output from audio FM (NR) circuit is muted [por]
1995 Mar 2115
Philips SemiconductorsPreliminary specification
Audio processor for VHS hi-fi
TDA9614H
and linear audio
Table 11 Bits RN and IPN
(1)
RN
00playbacklinear audio circuit in playback mode
01loop-through Plinear audio circuit in loop-through mode (playback ready) [por]
10loop-through R
11recordlinear audio circuit in record mode
Notes
1. Bit RN is output at LINREC (pin 32), which is used to drive an external (high voltage) head switch and possibly the
bias oscillator. The two loop-through modes are equal in signal flow, however they differ in the state of the head
switches (the internal switch at PBIN [pin 34] and the external head switch driven by pin LINREC).
2. Loop-through R is equal to record except for the RECOUT output (pin 33) which is muted.
Table 12 Bit NOM
NOMMODEDESCRIPTION
0−−
1mute
IPNMODEDESCRIPTION
(1)
(2)
output from linear audio circuit is muted [por]
linear audio circuit in loop-through mode (record ready)
Note
1. Audio coming from the linear audio circuit is muted, either the signal from tape in playback or the loop-through signal
in record or loop-through modes. To avoid clicks when changing the mode of the linear audio circuit to or from
playback, the output should be muted shortly, using this bit.
Table 13 Bit LP
LPMODEDESCRIPTION
0SPRecord and playback equalizing set for Standard Play [por].
Record equalization switch at EQSW (pin 30) and playback equalization switch at RESSW
(pin 35) are open (high impedance). The internal playback equalizing is 3180 and 120 µs.
It can be changed to 2544 and 96 µs by selecting test number 31 (see Section 6.12).
1LPRecord and playback equalizing set for Long Play.
Record equalization switch at EQSW (pin 30) and playback equalization switch at RESSW
(pin 35) are closed. The internal playback equalizing is 3180 and 170 µs. It can be
changed to 2731 and 146 µs by selecting test number 31 (see Section 6.12).
1995 Mar 2116
Philips SemiconductorsPreliminary specification
Audio processor for VHS hi-fi
and linear audio
6.7Main select byte
Table 14 Bits of main select byte
BITDESCRIPTION
IS2 to IS0Input Select 2 to Input Select 0; see Table 15
NS1 and NS0Normal Select 1 and Normal Select 0; see Table 16
OSLOutput Select Left; see Table 17
OSROutput Select Right; see Table 17
OSNOutput Select Normal; see Table 17
Table 15 Bits IS2 to IS0; note 1
IS2IS1IS0MODESELECTED INPUT SOURCE
000TunerTUNL and TUNR [por]
001CinchCINL and CINR
010Ext1EXT1L and EXT1R (e.g. SCART1; TV connector)
011Ext2EXT2L and EXT2R (e.g. SCART2; decoder connector)
100Ext3EXT3L and EXT3R (e.g. front cinch)
101SAPadditional mono input (SAP; pin 1)
110Normal
111Dub Mix
(2)
(2)(3)
output from linear audio (LININ; pin 28)
mixing of EXT3 input with hi-fi output signal
TDA9614H
Notes
1. With bits IS2 to IS0, the stereo input signal is selected which is led to the hi-fi processing. One out of five stereo
sources can be selected. The five stereo inputs differ in their connectivity; with bits NS1 and NS0 tuner left
(TUNL; pin 2) can be selected as linear audio input source, with bits LOS1 and LOS0 Normal, Ext2 and Ext3 are
directly connectable to Line out and to DOS1 and DOS0 Tuner, Ext1 and Ext3 are directly connectable to decoder
out (2nd Line Out).
2. Remark: when using the selections Normal or Dub Mix be aware of signal loops (which should be avoided) because
this can lead to large audio oscillations.
3. The selection Dub Mix is a special function in which a mono signal derived from the Ext3 inputs
(1⁄2EXT3L +1⁄2EXT3R) is led to the left input and a mono signal derived from the audio FM output (1⁄2L +1⁄2R) is led
to the right input channel. This function can be used for audio dubbing using the volume controls as a mixing desk.
Table 16 Bits NS1 and NS0; note 1
NS1NS0MODEDESCRIPTION
00Inputlinear audio input source is equal to hi-fi input [por]
01Volumelinear audio input source is equal to hi-fi input, inclusive volume control
10SAPadditional mono input (SAP; pin 1) is selected
11Tuner Lleft channel of tuner input is selected
Note
1. With bits NS1 and NS0 the input signal is selected for the linear audio circuit. When a stereo input source is selected,
a mono signal is made by adding the left and right channel. Furthermore two independent selections can be made
(the mono input SAP or the left channel of tuner e.g. for dual language). If in the volume mode the built-in AGC circuit
is switched off by using bit AGCN, the audio level can be controlled by the left and right volume controls
(VLx and VRx).
1995 Mar 2117
Philips SemiconductorsPreliminary specification
Audio processor for VHS hi-fi
TDA9614H
and linear audio
Table 17 Bits OSL, OSR and OSN; note 1
OSLOSROSNMODEDESCRIPTION
110stereoLEFT at left channel; RIGHT at right channel [por].
100leftLEFT at both left and right channels.
010rightRIGHT at both left and right channels.
000muteNo selection.
111mixed stereoLEFT + NORMAL added at left channel; RIGHT + NORMAL added at
right channel.
101mixed leftLEFT + NORMAL added at both left and right channels.
011mixed rightRIGHT + NORMAL added at both left and right channels.
001normalNORMAL (is linear audio) at both left and right channels.
Note
1. The bits OSL, OSR and OSN provide eight output select functions. LEFT and RIGHT are the left and right hi-fi
channels; NORMAL is the linear audio channel (LININ; pin 28). This selection is normally output at line out (LINEL
and LINER), RFCOUT (pin 18), decoder out (LINE2L and LINE2R), headphone out (HPOUTL and HPOUTR) and
DC out (DCOUTL and DCOUTR). However line out can be overruled by bits LOS1 and LOS0; decoder out can be
overruled by bits DOS1 and DOS0, headphone out (and DC out) can be overruled by bits HSL and HSR and DC out
can be overruled by bit DCS.
Remark: if AUTN = 1 (see Section 6.11) the output select modes stereo, left and right will be forced to normal as long
as the audio FM circuit is in playback mode and the internal PB mute signal is generated (no hi-fi track on tape). The
status of this internal mute signal can be read by bit PAFM (see Section 6.13).
6.8Secondary select byte
Table 18 Bits of secondary select byte
BITDESCRIPTION
HSLHeadphone Select Left; see Table 19
HSRHeadphone Select Right; see Table 19
DCSDC Select; see Table 20
RFCMRFC Mute; see Table 21
LOS1 and LOS0Line Output Select 1 and Line Output Select 0; see Table 22
DOS1 and DOS0Decoder Output Select 1 and Decoder Output Select 0; see Table 23
1995 Mar 2118
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