Daewoo DTL- 25G6F, Service Manual

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26
(3) Description
• Analog Sound IF - Input Section
The input pins , and ANN_IN-offer the possibility to connect two different sound IF sources to
the MSP 3415D. By means of bit [8] of AD_CV either terrestrial or satellite sound IF signals can be selected.
The analog-to-digital conversion of the preselected sound IF signal is done by a flash-converter, whose output can be
used to con-trol an analog automatic gain circuit (AGC), providing optimum level for a wide range of input levels. It is
possible to switch between automatic gain control and a fixed (setable) input gain. In the optimum case, the input range
of the AD converter is completely covered by the sound if source. Some combinations of SAW filters and sound
IF mixer IC’s however show large picture components on their outputs. In this case filtering is recommended.
It was found, that the high pass filters formed by the coupling capacitors at pins and are sufficient
in most cases.
• Quadrature Mixers
The digital input coming from the integrated A/D converter may contain audio information at a frequency
range of theoretically 0 to 9 MHz corresponding to the selected standards. By means of two
programmable quadrature mixers two different audio sources, for example NICAM and FM-mono, may be
shifted into baseband position. In the following the two main channels are provided to process either:
- NICAM (channel 1) and FM mono (channel 2) simultaneously or alternatively
- FM2 (channel 1) and FM1 (channel2).
Two independent digital oscillators are provided to generate two pairs of sin/cos-functions. Two
programmable increments, to be divided up into Low- and High part, determine frequency of the oscillator,
which corresponds to the frequency of the desired audio carrier.
• Lowpass Filtering Block for Mixed Sound IF Signals
By means of decimation filters the sampling rate is reduced. Then, data shaping and/or FM bandwidth limitation is
performed by a linear phase Finite Impulse Response (FIR-filter). Just like the oscillators’ increments the filter
NICAM versions can easily be implemented. Two not necessarily different sets of coefficients are required, one for
channel 1 (NICAM or FM2) and one for channel 2 (FM1=FM-mono).
Since both MSP channels are designed to process the German FM Stereo System with the same FIR coefficient set
(despite 7 dB power level difference of the two sound carriers), the MSP channel 1 has an internal overall gain of 6
dB. To process two carriers of identical power level these 6 dBs have to be taken into account by decreasing the
values of the channel 1 coefficient set.
• CORDIC Block
The filtered sound IF signals are demodulated by transforming the incoming signals from Cartesian into polar format
by means of a CORDIC processor block. On the output, the phase and amplitude is available for further processing.
AM signals are derived from the amplitude information whereas the phase information serves for FM and NICAM
(DQPSK) demodulation.
• Differentiators
FM demodulation is completed by differentiation the phase information output of the CORDIC block.
• Lowpass Filer Block for Demodulated Signals
The demodulated FM and AM signals are further lowpass filtered and decimated to a final sampling
frequency of 32 kHz. The usable bandwidth of the final baseband signals is about 15 kHz.