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3DQDVRQLF
Operating mode Characteristic Frequency
Mono un-modulated
Stereo
117.5Hz
Dual Tone
274.1Hz
These characteristic frequencies are fed to the signal
identification
identification
circuit,
here
the
type
of
signal
transmission is identified, with the following switching
being made available :
being made available :
FM - Stereo transmission
Dual tone transmission (where available)
FM. - Mono
This identification signal is also used to inform the
MSP as to which of the above three modes is being
transmitted, this information being fed via the I
MSP as to which of the above three modes is being
transmitted, this information being fed via the I
2
C bus.
The UOC IC IC601 then uses this information to
trigger the required switching and OSD display for
user information.
trigger the required switching and OSD display for
user information.
When NICAM / FM or Wagner stereo IF sound signals
are input to the MSP IC2001, the signals are firstly fed
via an analogue AGC circuit. This AGC circuit is used
to provide an optimum signal level for a wide range of
input levels. This AGC circuit can also be set to a fixed
input range. To provide the optimum level the input
range of the A/D converter should be completely
covered by the sound source. From the output of the
AGC circuit the SIF signal is then fed to an A/D
converter, where the signal is converted to digital.
From the output of the A/D converter, the signal
processing splits into two paths.
are input to the MSP IC2001, the signals are firstly fed
via an analogue AGC circuit. This AGC circuit is used
to provide an optimum signal level for a wide range of
input levels. This AGC circuit can also be set to a fixed
input range. To provide the optimum level the input
range of the A/D converter should be completely
covered by the sound source. From the output of the
AGC circuit the SIF signal is then fed to an A/D
converter, where the signal is converted to digital.
From the output of the A/D converter, the signal
processing splits into two paths.
SIF channel 1
SIF channel 1 is used to process NICAM or FM2,
this being sound carrier 2 of the FM stereo
system.
this being sound carrier 2 of the FM stereo
system.
SIF channel 2
SIF channel 2 is used to process FM mono or
FM1, again this being sound carrier 1 of the FM
stereo system.
stereo system.
10.2.2. Clock Generator
To aid processing an external crystal is connected to
pins 51 and 52 of IC2001. This provides the required
audio clock frequency for audio processing.
For NICAM / FM Mono processing, the MSP requires
a clock frequency of 18.432MHz, which it uses to lock
to the sampling rate of the NICAM signal.
When processing FM-stereo, the system clock runs
free on the crystal’s 18.432MHz signal.
pins 51 and 52 of IC2001. This provides the required
audio clock frequency for audio processing.
For NICAM / FM Mono processing, the MSP requires
a clock frequency of 18.432MHz, which it uses to lock
to the sampling rate of the NICAM signal.
When processing FM-stereo, the system clock runs
free on the crystal’s 18.432MHz signal.
10.2.3. Demodulation Stage
The digitised IF sound signals fed from the A/D
converter are then fed to two Quadrature mixer
circuits. By means of the two programmable
quadrature mixers, two different audio signals e.g.
NICAM and FM-mono signals can be input.
Depending on the selected standards, this audio
information could have a frequency range of 0 to
9MHz.
From the Quadrature mixers the signals are then fed
via a lowpass filter, these filters being programmable
make it possible to process both NICAM standards.
Control of these filters are carried out by the UOC IC
via the I
converter are then fed to two Quadrature mixer
circuits. By means of the two programmable
quadrature mixers, two different audio signals e.g.
NICAM and FM-mono signals can be input.
Depending on the selected standards, this audio
information could have a frequency range of 0 to
9MHz.
From the Quadrature mixers the signals are then fed
via a lowpass filter, these filters being programmable
make it possible to process both NICAM standards.
Control of these filters are carried out by the UOC IC
via the I
2
C bus.
From the output of the low pass filters the signals are
then fed to the Phase and AM Discriminator stage.
From the output of this stage the FM and NICAM
processing follow different paths.
The NICAM signal is fed via the DQPSK decoder,
from the output of the decoder we now have a data
stream of 728k bits/sec which is fed to the NICAM
decoder.
For FM processing the demodulated signals are fed
to two differentiator circuits which differentiate the
phase information output from the demodulator
circuit to complete the FM demodulation.
then fed to the Phase and AM Discriminator stage.
From the output of this stage the FM and NICAM
processing follow different paths.
The NICAM signal is fed via the DQPSK decoder,
from the output of the decoder we now have a data
stream of 728k bits/sec which is fed to the NICAM
decoder.
For FM processing the demodulated signals are fed
to two differentiator circuits which differentiate the
phase information output from the demodulator
circuit to complete the FM demodulation.
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10.2.4. NICAM Decoder
Before any NICAM decoding can start, IC2001 must
first lock to the NICAM frame structure by searching
and synchronizing to the Frame Alignment Word
(FAW).
first lock to the NICAM frame structure by searching
and synchronizing to the Frame Alignment Word
(FAW).
To reconstruct the original digital sound samples, the
NICAM
NICAM
-
bitstream
has
to
be descrambled,
deinterleaved and rescaled, as well as performing bit
error detection and correction, all of which are carried
out in this section.
error detection and correction, all of which are carried
out in this section.
To switch the TV set to the actual sound mode, control
information on the NICAM mode and bit error rate are
supplied by the NICAM decoder via the I
information on the NICAM mode and bit error rate are
supplied by the NICAM decoder via the I
2
C bus to the
UOC IC IC601. The UOC IC then, as mentioned for
FM processing, initiates the required switching and
OSD display for user information. From the output of
the NICAM decoder the left and right channels in
digital format are fed to the DSP processor stage.
FM processing, initiates the required switching and
OSD display for user information. From the output of
the NICAM decoder the left and right channels in
digital format are fed to the DSP processor stage.
10.2.5. FM Processing
After the FM signal has been demodulated the signal
is fed to a Mute stage which is controlled by the carrier
detector circuit. If no FM carrier is detected in channel
2 of the MSP then the subsequent FM1 output is
muted by the muting stage in that channel. Likewise,
if no FM2 carrier is detected in channel 1, the FM2
output will be muted. This muting stage is provided to
prevent the processing of a noise signal to the
loudspeaker system.
From the output of the muting circuit, the signal is also
fed to the DSP after passing a lowpass filter, here the
demodulated FM / AM signals are decimated to a final
sampling frequency of 32KHz.
The usable bandwidth of the baseband signal being
about 15KHz.
is fed to a Mute stage which is controlled by the carrier
detector circuit. If no FM carrier is detected in channel
2 of the MSP then the subsequent FM1 output is
muted by the muting stage in that channel. Likewise,
if no FM2 carrier is detected in channel 1, the FM2
output will be muted. This muting stage is provided to
prevent the processing of a noise signal to the
loudspeaker system.
From the output of the muting circuit, the signal is also
fed to the DSP after passing a lowpass filter, here the
demodulated FM / AM signals are decimated to a final
sampling frequency of 32KHz.
The usable bandwidth of the baseband signal being
about 15KHz.
10.2.6. MSP3415D Audio Baseband
Processing
By means of the DSP processor all audio functions
are performed by digital signal processing. The DSP
functions being grouped into three processing parts
are performed by digital signal processing. The DSP
functions being grouped into three processing parts
Input Pre-processing
Channel Selection
Channel Post-processing
The input pre-processing is intended to prepare the
various signals of all input sources in order to form a
standardised signal at the input to the channel
selector. The signals are adjusted in volume, by the
prescaler,
various signals of all input sources in order to form a
standardised signal at the input to the channel
selector. The signals are adjusted in volume, by the
prescaler,
before
being
processed
with
the
appropriate de-emphasis. When transmitting an FM
stereo signal, sound carrier1 is made up of L + R/2
while the second sound carrier FM2 is made up of just
right hand signal. To produce a stereo signal the R/2
has to be removed from the ’Left’, this is performed in
the FM matrix stage.
stereo signal, sound carrier1 is made up of L + R/2
while the second sound carrier FM2 is made up of just
right hand signal. To produce a stereo signal the R/2
has to be removed from the ’Left’, this is performed in
the FM matrix stage.
Having
prepared
all
the
input
signals
to
a
standardised level it is now possible with the aid of the
channel selector to distribute all possible signal
sources to the desired outputs.
channel selector to distribute all possible signal
sources to the desired outputs.
All input and output signals can be processed
simultaneously with the exception of FM2, which
cannot be processed at the same time as NICAM.
This is due to the fact that NICAM and FM2
processing use SIF channel 1 in the demodulator
section. The processing of the NICAM and FM2
signals follow different paths controlled by an internal
switch, used to switch between the two processing
paths. This causes a delay during switching between
the two signal processing paths.
simultaneously with the exception of FM2, which
cannot be processed at the same time as NICAM.
This is due to the fact that NICAM and FM2
processing use SIF channel 1 in the demodulator
section. The processing of the NICAM and FM2
signals follow different paths controlled by an internal
switch, used to switch between the two processing
paths. This causes a delay during switching between
the two signal processing paths.
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10.3. AM and SCART Processing
As mentioned at the start of the section on MSP3415
IC2001
IC2001
processing,
when
receiving
an
AM
modulated signal, as is system ’L’, the signal is
demodulated in the IF stage. The audio signal is then
fed to IC2001 pin 44 mono input. The demodulated
AM audio signal is then fed to an internal analogue
scart switching circuit which is used to select between
demodulated in the IF stage. The audio signal is then
fed to IC2001 pin 44 mono input. The demodulated
AM audio signal is then fed to an internal analogue
scart switching circuit which is used to select between
the scart signals input via pins 41 and 42. The
selected audio signals are then fed via an A/D
converter before being input to the DSP processing
stage. Once the audio signals are fed to the DSP
stage, the signals are fed via a prescaler circuit which,
as mentioned previously, adjusts the volume of the
audio signals, thus ensuring the signals are at the
same standard as the other signal sources at the
channel select circuit.
selected audio signals are then fed via an A/D
converter before being input to the DSP processing
stage. Once the audio signals are fed to the DSP
stage, the signals are fed via a prescaler circuit which,
as mentioned previously, adjusts the volume of the
audio signals, thus ensuring the signals are at the
same standard as the other signal sources at the
channel select circuit.
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10.4. Audio Output
10.4.1. Loudspeaker Output Path
When a signal source has been selected by the user
for output via the loudspeaker the signal is fed via a
stereo separation matrix circuit which is used only
when processing an FM stereo signal, at any other
time the circuit is switched off. From the output of this
circuit the signal is then fed to an Automatic Volume
Correction circuit, as well as Bass, Treble and
Loudness stages.
for output via the loudspeaker the signal is fed via a
stereo separation matrix circuit which is used only
when processing an FM stereo signal, at any other
time the circuit is switched off. From the output of this
circuit the signal is then fed to an Automatic Volume
Correction circuit, as well as Bass, Treble and
Loudness stages.
10.4.2. Automatic Volume Correction (AVC)
Different sound sources such as terrestrial and
satellite channels often have different volume levels.
Advertisements during movies usually have a higher
volume level than the movie itself. The Automatic
Volume Correction (AVC) solves this problem and
equalises the volume levels.
satellite channels often have different volume levels.
Advertisements during movies usually have a higher
volume level than the movie itself. The Automatic
Volume Correction (AVC) solves this problem and
equalises the volume levels.
10.4.3. Bass and Treble adjustments
The Bass and Treble adjustments consist of two
separate filters. The control range is +20dB to -12dB
for Bass adjustment, and +15dB to -12dB for Treble
adjustment. These adjustments are carried out via
the OSD. The two filters coefficients for the selected
range are set to the required value via the I
separate filters. The control range is +20dB to -12dB
for Bass adjustment, and +15dB to -12dB for Treble
adjustment. These adjustments are carried out via
the OSD. The two filters coefficients for the selected
range are set to the required value via the I
2
C bus.
The volume modifications which occur during bass
and treble adjustments are stabilised by limiting the
internal volume, to prevent clipping, this limitation is
carried out via software.
internal volume, to prevent clipping, this limitation is
carried out via software.
10.4.4. Loudness
The audio signal fed to the loudness stage is
examined for loudness and aurally compensated for.
Loudness increases the volume of low and high
frequency signals while keeping the amplitude of the
1KHz reference frequency constant.
As the loudness is set according to the actual volume
setting, the required filter coefficients must be
established first, before the signals are acted upon
with varying degrees of severity according to the
volume.
examined for loudness and aurally compensated for.
Loudness increases the volume of low and high
frequency signals while keeping the amplitude of the
1KHz reference frequency constant.
As the loudness is set according to the actual volume
setting, the required filter coefficients must be
established first, before the signals are acted upon
with varying degrees of severity according to the
volume.
10.4.5. AV Scart Output
Signals selected for outputing to the 21 pin scart
terminals, are fed via a stereo separation matrix
circuit as in the loudspeaker path. However there are
no bass, treble or balance controls but rather, just a
preset volume control circuit. The audio signals
output from IC2001 via pins 30 and 31 are then fed to
the 21 pin AV socket pins 1 and 3.
terminals, are fed via a stereo separation matrix
circuit as in the loudspeaker path. However there are
no bass, treble or balance controls but rather, just a
preset volume control circuit. The audio signals
output from IC2001 via pins 30 and 31 are then fed to
the 21 pin AV socket pins 1 and 3.
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