View SAA7709HN1B_4472590.PDF datasheet online --- IC-ON-LINE

Datasheet File OCR Text:

user manual hardware part saa7709h/n1b car radio digital signal processor product development cardsp, mainstream consumer systems nijmegen, the netherlands. version nr. : 4.0 author : g. laarhoven status : accepted date : january 8, 2002 report nr. : rnb-c/3272/2002i-0005 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 2 revision history manual version date remarks 1.0 march 31, 2000 n101a initial version 2.0 august 11, 2000 n102b initial version 3.0 october 30, 2000 n103b initial version ? radio features adapted ? rds description adapted 4.0 january 4, 2002 n1b final version ? fig. 8.15 : changed. resistor r16/r18 is added ? fig. 8.23 : changed. resistor r5 is added ? table 8.2 : timing (unit) rds changed ? chapter 8.8 : chapter ?interface with tuner tea6840 is added www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 3 summary. the purpose of this manual is to give all hardware application information of thesaa7709h, being a car radio digital signal processor (cdsp), needed to make a hardware application. also the audio and radio features are described. the audio and software part of the saa7709h/n103b is decribed in a other document named ?software audio and radio part saa7709h/n103b?. before reading this report it is necessary to read first the data sheet of the saa7709h. in this manual all the pins are described with additional information on the input- and output circuits. the blockdiagram is given and all functions are explained. all necessary coefficient settings and tables for several selections are given. the application diagram is explained in detail. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 4 table of contents 1 introduction. 7 2. general description. 8 3 hardware/software features. 8 3.1 hardware features 8 3.2 software features 9 4 quick reference specification. 9 4.1 fm reception 9 4.2 am reception 10 4.3 analogue tape input 10 4.4 analogue cd input 10 4.5 rds reception 10 4.6 cd i2s / spdif input 11 4.7 audio output performance 11 4.8 audio processing 11 4.8.1 volume/balance/fader/tone/loudness/dynamic bass boost control 11 4.8.2 equalisation 11 5 block diagrams. 12 5.1 total block diagram 12 5.2 cdsp block diagram. 13 6 pinning diagram. 14 6.1 pin description. 15 7 functional description of the cdsp modes and functions. 18 7.1 audio processing 18 7.2 fm mode 19 7.4 general purpose tone generator 21 7.5 tone sequencer 21 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 5 7.6 noise generator 22 7.7 mss function 22 7.8 radio data system (rds) function 22 7.9 second processor extension function 22 8 hardware application of the cdsp 23 8.1 audio processing in dsp 23 8.1.1 analogue outputs (pin 8, 9, 13 and 15) 23 8.1.2. internal reference voltage sources vrefda (pin 12) and vrefad (pin 77) 24 8.1.2.3 ref. voltages for the ad convertors vdacp (pin 75) and vdacn (pin 76) 25 8.1.3 power on/off mute 26 8.1.4 dsp reset (pin 42) 28 8.2 am / fm signal quality processing 30 8.3 fm mode 31 8.3.1 fm input pins 31 8.3.2 fm input sensitivity selection 32 8.3.3 fm iac 34 8.4 am, aux, tape, phone, nav and cd_a inputs 40 8.4.1 am inputs 42 8.4.1.1 am inputs for am-mono mode and am-stereo mode with external decoder 42 8.4.2 tape / aux input 44 8.4.3 analogue cd input 45 8.5 phone and navigation inputs 47 8.6 cd-digital inputs (i2s or spdif) 52 8.6.1 general 52 8.6.2 i2s input 52 8.6.3 spdif input 53 8.7 radio data system (rds) function 53 8.7.1 general description 53 8.7.3 direct rds inputs/outputs in davd mode (dac0=1, dac1=1, rds decoder bypass mode) 56 8.7.4 direct rds timing of clock and data signals in davd mode (dac0=1, dac1=1, rds decoder bypass mode) 56 8.7.5 buffering of rds data 57 8.7.6 buffer interface 57 8.7.3 fast rds detection with the rds-band signal level detector 58 8.7.4 bitslip 60 8.7.5 rds/rbds decoder 61 8.7.6 rds/rbds block detection 62 8.7.7 error detection and correction 62 8.7.8 synchronization 63 8.7.9 flywheel for synchronization hold 63 8.7.10 bit slip correction 63 8.7.11 data processing control 63 8.7.12 restart of synchronization mode: 64 8.7.13 error correction control mode for synchronization: 64 8.7.14 rbds processing mode: 64 8.7.15 data available control modes: 65 8.7.16 data output of rds/rbds information 65 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 6 8.8 interface with tuner tea6840 (nice) 66 8.9 i2c interface 68 8.10 second processor extension function 69 8.11 digital subwoofer and center output 69 8.12 external dac output (subwoofer) 69 8.13 x-tal oscillator circuit 70 8.13 emc saa7709h application 72 8.14 changing the clock frequency of the dsp 73 8.14.1 procedure for increasing the clock frequency of the dsp 73 appendix 1 : application diagram 74 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 7 1 introduction. digital techniques have been widely accepted in the audio world including in the car radio. the cd was the first example, nowadays the digitisation of the car radio is a logical continuation of this trend. furthermore, in the car radio world there is a growing demand not only for a good radio perception but also for a better sound quality. people want the sound of their living room in their vehicle. this led not only to the digitisation of analogue designs of the car radio, but also to the incorporation of a customised dsp on board of a car radio i.c., delivering many possibilities for sound quality improvement. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 8 2. general description. the cdsp-chip performs all the signal functions in front of the power amplifiers and behind the am and fm_mpx demodulation of a car radio or the tape input. these functions are: interference absorption, stereo decoding, rds demodulation and decoding, fm and am weak signal processing (soft-mute, sliding stereo, etc.), dolby-b tape noise reduction and the audio controls (volume, balance, fader and tone). some functions have been implemented in hardware (stereo decoder, rds decoding and iac for fm_mpx) and are not freely programmable. digital audio signals from external sources with the philips i 2 s and the lsb 16, 18, 20 and 24 bit justified format or spdif format up to fs = 48 khz are accepted. there are four independent analogue output channels. the saa7709h/n103b is the final version. all audio and radio software features are in the n103 romcode available. the hardware of the ic between the n102b and n103b are the same. the dsp contains a basic program which enables a set with am/fm reception, compressor function for all audio modes on the primary channel (channel 1) and fader/balance control. a hardware 5 band per channel parametric equalizer is also implemented. with some restrictions also 2 different stereo channels can be processed. 3 hardware/software features. 3.1 hardware features ? = 1 bit stream 1st order sigma-delta a/d converter with anti aliasing broadband input filter ? = 4 bit stream 3rd order sigma-delta a/d converters with anti aliasing broadband input filter ? = 4 bitstream d/a converters with 128-fold oversampling and noise shaping ? = 4 channel 5 band i 2 c controlled parametric equalizer ? = integrated semi-digital filter, no external post filter required for d/a. ? = a stereo i 2 s output with 256 fs clock for connection to an external da converter. ? = limited dual media support, allowing limited separate front-seat and rear-seat signal sources and separate control. ? = digital fm stereo decoder. ? = digital fm interference suppression. ? = rds demodulation decoding via separate adc, with buffered output option on the demodulator and decoder i 2 c accessible. ? = two mono cmrr or differential input high performance stages for voice signals from phone and navigation inputs via 3rd order sigma-delta a/d converter. ? = four switchable stereo cmrr or differential input stages. (cd-walkman, cd-changer etc.) ? = analogue single ended tape input ? = a 5120 x 32 dsp program rom, a 1024 x 24 data ram and a 640 x 12 coefficient ram. ? = separate am-left and am-right inputs in case of use of external am stereo decoder. ? = one digital input: i 2 s or lsb justified format. ? = two digital inputs: spdif format at fs = 48 khz maximum. ? = audio output short circuit protected ? = i 2 c bus controlled (including fast mode) www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 9 ? = a phase lock loop derives the internal clock for the dsp from one common fundamental crystal oscillator ? = a phase lock loop derives the internal clock for the dac and other parts of the ic from a digital input word select ? = combined am/fm level input ? = relative high pin compatibility with saa7704, saa7705, saa7706, saa7708 (incidental minor replacements needed) ? = low number of external components required ? = -40 to +85 o c operating temperature range ? = easy applicable 3.2 software features ? = fm de-matrixing ? = am brick wall filter ? = baseband audio processing (balance/fader/volume) ? = soft audio mute ? = large volume jumps e-power interpolated for smooth volume steps ? = general purpose tone generator 4 quick reference specification. important: this overview shows the best specification which can be obtained with an 'ideal' receiver in combination with the saa7709h. however the specification points 4.1 and 4.2 will be limited by the front-end receiver and not by the saa7709h. 4.1 fm reception frequency response(+/-1 db) 20 hz - 17 khz s/n (mono, 1khz, 22.5 khz dev.) > 69 db ; typical 72 db (deemphasis 50 s) s/n (stereo, 1 khz, 22.5 khz dev.) > 60 db ; typical 63 db (deemphasis 50 s) max. deviation (at thd < 1%) at 1 khz > 120 khz mono distortion, 1 khz at 75 khz deviation < 0.2 % at 22.5 khz deviation < 0.1 % stereo distortion, 1khz, 1 channel at 22.5 khz deviation < 0.2 % stereo channel separation, 1 khz > 40 db ; typical 45 db www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 10 4.2 am reception frequency response with tuner 20 hz - 2 khz frequency response (+/-1 db) with dsp software brickwall 20 hz - 4.5 khz filter (without tuner) frequency response (+/-1 db) without brickwall filter 20 hz - 15 khz (without tuner) s/n at 1 khz, 30 % am > 70 db ; typical 75 db distortion, 400 hz, bw 5 khz 80 % am < 0.2 % 30 % am < 0.1 % 4.3 analogue tape input frequency response (+/-3 db) 20 hz - 18 khz typ. s/n at 1 khz, 0 ref. db 84 db typ. thd+n, 1 khz (0.55 vrms) -85 db typ. channel separation, 1 khz 65 db rds traffic information reception from radio signals in this mode is possible; the decoder is still operating 4.4 analogue cd input frequency response (+/-3 db) 20 hz - 18 khz typ. s/n at 1 khz, 0 db ref. 84 db typ. thd+n, 1 khz (0.5 vrms) -85 db typ. channel separation, 1 khz 65 db rds traffic information reception from radio signals in this mode is possible; the decoder is still operating 4.5 rds reception min. nearby selectivity 61 db (neighbour ch at 200 khz) min. pilot attenuation 50 db www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 11 4.6 cd i2s / spdif input the performance of these input signals is actually limited by the dac output, as described in chapter 4.7. the digital cd input can be used as an alternative input for the analogue cd. rds traffic information reception from radio signals in this mode is possible; the decoder is still operating 4.7 audio output performance typ. output level 1 vrms bandwidth (fs=44.1 khz, -3db) 20 hz - 22 khz typ. s/n 105 dba typ. output noise 3 v ; a-weighted typ. thd+n, 1 khz, 0 db -90 db typ. dynamic range, 1 khz (-60 db) 97 dba 4.8 audio processing 4.8.1 volume/balance/fader/tone/loudness/dynamic bass boost control (figures count for default coefficient set) volume control range -66 db --> +24 db balance attenuation range (left/right) 0 db --> -66 db fader attenuation range (front/rear) 0 db --> -66 db 4.8.2 equalisation number of bands 20 bands filter order 2nd order bp centre frequency 20 hz --> 18 khz gain control range -30 db --> +12 db quality factor 0.01 --> 100 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 12 5 block diagrams. 5.1 total block diagram the total block diagram indicates a possible application in which the cdsp can be used. fig 5.1 total block diagram am/fm-rf tea6811 am/fm-if tea6824 am fm rds level saa7709h p lr rr lf rf audio co-processor power amp cd_d spdif-1 spdif-2 aux cd_a i2c rds display i2c tape external dac i2s phone/nav www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 13 5.2 cdsp block diagram. fig 5.2 block diagram cdsp analog source selector stereo adc1 stereo adc2 digital source selector dsp quad fsdac level adc rds demodulator xtal osc i2s spdif i2c saa7709h digital i/o i2s out phone phone_gnd nav nav_gnd cd_r cr_gndr cd_l cd_gndl tape_r tape_l am_l am_r aux_r aux_l fm_mpx fm_rds sel_fr level rds_data rds_clock osc_in osc_out cd_data cd_ws cd_clk spdif1 spdif2 scl sda ao iis_out1 iis_out2 iis_clk iis_ws iis_in1 iis_in2 ws_dac data_dac clk_dac fs_sys rear-right rear-left front-right front-left dsp_io(1..8) www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 14 6 pinning diagram. fig 6.1 pinning diagram 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 64 59 60 61 58 57 62 63 51 52 53 54 55 56 50 49 45 44 41 43 42 46 47 48 40 39 38 37 36 35 34 33 32 31 30 29 28 27 26 25 1 6 5 4 7 8 3 2 14 13 12 11 10 9 15 16 20 21 24 22 23 19 18 17 saa7709h vdd_osc am_r/am_mono am_l tape_r tape_l cd_r cd_l cd_gndr cd_gndl vddaad vdacp vdacn vrefad aux_r aux_l fm_rds dsp_io3 dsp_io2 dsp_io1 vssq2 vddq2 iis_out2 iis_out1 iis_ws iis_in2 iis_in1 iis_clk cd_clk cd_dat a cd_ws dsp_io5 spdif1 osc_out osc_in vss_osc sel_fr rds_data rds_clock sda scl ao vsss6 vsss5 vsss4 vsss3 vddd2 vsss2 vsss1 vddd1 vssq3 vddq3 tscan shtcb rtcb dsp_reset dsp_io4 spdif2 vssq1 vddq1 dsp_io6 clk_dac data_dac ws_dac fs_sys dsp_io7 flv dsp_io8 frv vrefda vddda vssda rlv rrv pom nrv_gnd nav phone_gnd phone level fm_mpx www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 15 6.1 pin description. table 1 pin list saa7709h symbol pin description fm_mpx 1 analogue input pin for fm-multiplex signal level 2 fm/am-level input pin. via this pin the level of the fm signal or level of the am signal is fed to the cdsp. the level information is used in the dsp for signal correction phone 3 input of common mode phone signal phone_gnd 4 by i2c switchable common mode reference pin to enable an arbitrary high common mode analogue input for all 4 ads. nav 5 input of common mode navigation signal nav_gnd 6 by i2c switchable common mode reference pin to enable an arbitrary high common mode analogue input for all 4 ads. pom 7 power on mute of the fsdac. timing is determined by an external capacitor and the internal current sources. rrv 8 rear right audio voltage output of the fsdac rlv 9 rear left audio voltage output of the fsdac vssda 10 ground supply analogue part of the fsdac and spdif bitslicer vddda 11 3v3 positive supply analogue part of the fsdac and spdif bitslicer vrefda 12 voltage reference of the analogue part of the fsdac frv 13 front right audio voltage output of the fsdac dsp_io8 14 digital in/output 8 of the dsp-core (f7 of the status register) flv 15 front left audio voltage output of the fsdac dsp_io7 16 digital in/output 7 of the dsp-core (f6 of the status register) fs_sys 17 256- or n x fs clock output to be used together with an external da converter ws_dac 18 word select signal to be used for an external da converter data_dac 19 data signal to be used for an external da converter clk_dac 20 clock signal to be used for an external da converter dsp_io6 21 digital in/output 6 of the dsp-core (f5 of the status register) vddq1 22 3v3 positive supply 1 peripheral cells only vssq1 23 ground supply 1 of peripheral cells only spdif2 24 analogue bitslicer input2 for spdif, can be selected i.s.o. spdif1 via i2c bit spdif1 25 analogue bitslicer input1 for spdif, can be selected i.s.o. spdif2 via i2c bit www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 16 dsp_io5 26 digital in/output 5 of the dsp-core (f4 of the status register) cd_ws 27 i 2 s or lsb justified format word select input from a digital audio source cd_data 28 i 2 s or lsb justified format left-right data input from a digital audio source cd_cl 29 i 2 s clock or lsb justified format input from a digital audio source iis_clk 30 clock output for external i 2 s receiver. for example headphone/ subwoofer iis_in1 31 data 1 input for external i 2 s transmitter, e.g. audio co-processor iis_in2 32 data 2 input for external i 2 s transmitter, e.g. audio co-processor iis_ws 33 word select output for external i 2 s receiver. for example headphone/ subwoofer iis_out1 34 data1 in/output for external i 2 s receiver. for example headphone/ subwoofer iis_out2 35 data2 in/output for external i 2 s receiver. for example headphone/ subwoofer vddq2 36 3.3 v positive supply 2 peripheral cells only vssq2 37 ground supply 2 of peripheral cells only dsp_io1 38 digital in/output 1 of the dsp-core (f0 of the status register). input level must always be defined externally in the application dsp_io2 39 digital in/output 2 of the dsp-core (f1 of the status register). input level must always be defined externally in the application dsp_io3 40 digital in/output 3 of the dsp-core (f2 of the status register) dsp_io4 41 digital in/output 4 of the dsp-core (f3 of the status register) dsp_reset 42 reset of the dsp core (active low) rtcb 43 asynchronous reset test control block active low, may not be connected in the application shtcb 44 shift clock test control block, may not be connected in the application tscan 45 scan control active high, may not be connected in the application vddq3 46 3v3 positive supply 3 peripheral cells only vssq3 47 ground supply 3 peripheral cells only vddd1 48 3v3 positive supply 1 core and internal supply io ring vsss1 49 ground supply 1 of 3.3 volt core only vsss2 50 ground supply 2 of 3.3 volt core only vddd2 51 3v3 positive supply 2 core only vsss3 52 ground supply 3 of core, internal ground supply ring and substrate vsss4 53 ground supply 4 of core, internal ground supply ring and substrate vsss5 54 ground supply 5 of core only www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 17 vsss6 55 ground supply 6 of core only a0 56 slave sub-address i 2 c selection / serial data input test control block scl 57 serial clock input i 2 c bus. must always be defined by application. sda 58 serial data input / output i 2 c bus rds_clock 59 radio data system bit clock output / rds external clock input rds_data 60 radio data system data output sel_fr 61 ad input selection switch to enable high ohmic fm_mpx input at fast tuner search on fm_rds input. must always be defined by application. vss_osc 62 ground supply crystal oscillator circuit osc_in 63 crystal oscillator input: crystal oscillator sense for gain control or forced input in slave mode osc_out 64 crystal oscillator output: drive output to 11.2896 mhz crystal vdd_osc 65 3v positive supply crystal oscillator circuit am_r/am_mono 66 analogue input pin for am audio frequency right channel am_l 67 analogue input pin for am audio frequency left channel or am mono tape_r 68 input of the analogue tape right signal tape_l 69 input of the analogue tape left signal cd_r 70 input of the analogue cd right signal cd_gndr 71 by i2c switchable common mode reference pin to enable high common mode analogue input for the cd_r input or a high common mode analogue input for all 2 ads for right channel processing cd_l 72 input of the analogue cd left signal cd_gndl 73 by i2c switchable common mode reference pin to enable high common mode analogue input for the cd_l input or a high common mode analogue input for all 2 ads for left channel processing vddaad 74 positive supply analogue ad1..4 and level ad. vdacp 75 positive reference voltage adc1..4 and level ad vdacn 76 ground supply analogue ad1..4 and level ad. vrefad 77 common mode reference voltage ad1..4 and level ad aux_r 78 input of the analogue auxiliary right signal aux_l 79 input of the analogue auxiliary left signal fm_rds 80 analogue input pin for fm rds signal www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 18 7 functional description of the cdsp modes and functions. introduction. the cdsp block diagram is depicted in figure 5.1. for a thorough description of the cdsp block diagram see the data sheet of the saa7709h. in this chapter a general overview is given of all modes and functions. the cdsp can be set in several operation modes. each mode executes functions which are required for that particular mode, furthermore the cdsp can process two independent sources simultaneously (dual media modes), note that not all combinations are possible, for example am-mode and fm-mode, two digital input modes are only possible when the the (external) digital sources are synchronous and locked to each other. the selection of a particular mode is software controlled (via the i2c bus) and described in chapter 9. the required inputs for each mode can be selected by the analogue or digital source selectors which are software controlled (via the i2c bus). the source selection is also described in chapter 9. 7.1 audio processing all basic audio processing in the cdsp chip is performed by the integrated digital signal processor (dsp). the signal flow is more or less fixed and the functions are controlled by sending coefficient values to the appropriate places in the dsp processor coefficient memory via the i2c bus. the functions of the audio processing block are always executed independent of the mode. the audio processing block consists of two parts, the audio processing functions for the primary channel and the audio processing functions for the secondary channel. the following functions have been implemented: - volume control the volume control function determines the output voltage of the cdsp. the volume control is split into a gain and a attenuation section which acts equal for both channels. the volume control contains also a prescaling which ensures that for the various input signals the same sound pressure level (for a fixed volume setting) can be obtained at the output of the cdsp. the primary- and secondary channel have independent volume control. - balance the balance function controls the attenuation of either the left or the right channel while the other channel is kept constant. separate balance functions are available in the primary- and secondary channel. - fader this function is only implemented for the primary channel the fader function controls the attenuation of either the front or rear channels while the other channels are kept constant. the fader is controlled via the i2c bus. - soft audio mute the soft audio mute function enables the user to generate a gradual mute or de-mute function without www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 19 undesired clicks. the soft audio mute is implemented as a linear ramp. there is one sam function for the primary channel and the subwoofer output, the secondary channel has a separate sam function. - parametric equaliser 2 sections of 2x5 bands each are available, they can be used in the primary and/or the secondary channel in the main audio program. 7.2 fm mode this is the mode for fm reception and runs in the dsp. the selected input is fm-mpx or fm-rds the program in the fm mode offers the following functions: enhanced fm dynamic signal processing the fm dynamic signal processing adapts the fm audio characteristics depending on the quality of the received station. as criterium to judge this quality the following parameters are used: - the level signal as a measure for the fieldstrength - the multipath detector output as a measure for the multipath distortion - the noise above 60 khz of the mpx signal as a measure for the adjacent channel interference from the audio characteristics the output level (softmute), the stereo image (sliding stereo to mono) and the audio frequency response (high cut control) are adapted. the following functions are implemented: ? softmute as a funtion of level and noise: - fast attack and recovery at level dips with a low repetition rate - fast attack with slow recovery at dips with a high repetition rate or with a long duration - fast attack and recovery at adjacent channel breakthrough ? stereo control (sliding stereo) as a function of level, noise and multipath: - fast attack and recovery at level dips, noise or multipath bursts with a low repetition rate - fast attack with slow recovery at events with a high repetition rate or with a long duration ? audio frequency response control as a function of level, noise and multipath: - fast attack and recovery at level dips, noise or multipath bursts with a low repetition rate - fast attack with slow recovery at events with a high repetition rate or with a long duration. adjustment of channel separation the purpose of this function is to compensate for the non flat frequency response around 38 khz of the fm tuner which causes extra cross-talk. fm de-emphasis filter and 19 khz mpx filter the purpose of the de-emphasis filter is to compensate the pre-emphasized fm signal with a filter with a time constant of 50 s or 75 s. the notch filter at 19 khz is used to protect tweeters in high power applications from overload by the stereo pilot. fm audio filter the purpose of the fm audio filter is to set the audio bandwidth in fm mode independent from the www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 20 other modes. stereo detection the purpose of the stereo detector is to indicate the presence of a pilot tone and that the stereo decoder is in lock. noise filter the noise level is detected in a band from 60 khz till 120 khz with an envelope detector (see data sheet). the noise level is used as adjacent channel information for the controller and for the fm dynamic signal processing. rds updates this function offers the following features: - pause detection the purpose of the pause detection is to search for a pause in the fm signal. a pause is detected when the fm signal is below a pre-defined level for a certain amount of time. the output of the pause detector is the dsp_io4 pin (pin 41). "high" indicates pause. -mute the purpose of the mute is to mute the fm signal that goes to the audio processing block. this mute is activated by the external control pin dsp_io1 (pin 38). "low" is mute. - hold function the purpose of the hold function is to prevent that the information retrieved during an rds update can disturb the filters in the fm signal processing block. the hold function is activated by the external control pin dsp_io1 (pin 38). "low" is hold. - freeze function the purpose of the freeze function is to freeze the level, noise and multipath values measured during an rds update and to read them out after the update. the freeze function is activated by the external control pin dsp_io2 (pin 39). "low" is freeze. interference absorption circuit (iac) the interference absorption circuit (iac) detects and suppresses ignition interference. the characteristics of the iac can be adapted to the properties of different fm tuners by means of the predefined coefficients in the iac control register. the values can be changed via the i2c bus. on power on the nominal setting for a good performing iac is selected (all iac control bits are set to their default value, according the i2c hardware register definition). www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 21 7.3 am-mono mode this is the mode for mono mw, lw or sw reception, the selected input is am_r. the program in the am-mono mode runs in the dsp and offers the following functions: am dynamic signal processing the am dynamic signal processing adapts the am audio characteristics depending on the quality of the received station. as criterium to judge this quality the level signal as a measure for the fieldstrength is used. from the audio characteristics the output level (softmute) and the audio frequency response (high cut control) are adapted. the following functions are implemented: ? soft mute as a function of level ? audio frequency response control as a function of level 6th order low-pass filter the purpose of the low-pass filter is to suppress interference whistles from adjacent channels and noise. am iac the am iac (interference absorption circuitry) detects and eliminates audible clicks caused by impulsive interference, such as caused by engine ignition or fan, on am reception. the characteristics of the am iac can be adapted to the properties of different am tuners by means of coefficients in the yram of the dsp am quality detection the am quality feature detects interfering signals caused by adjacent- and co-channel interference. this feature is available only during search mode of the am-tuner. the audio output is muted during search mode. 7.4 general purpose tone generator the tone generator generates a sinewave signals on the left- and right audio channel and can be selected as main audio source. the tone generator can be used f.i. to test the speaker outputs in the car radio during production. the tone generator function is part of the audio program in the dsp and is therefor always available. 7.5 tone sequencer the tone sequencer generates a wide range of bleeps and chime sounds with selectable frequency and wave form. these sounds can be used for audio feedback or for test purposes and can be added to the primary and/or secundary channel outputs. the tone sequencer function is part of the audio program in the dsp and is therefor always available. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 22 7.6 noise generator the noise generator produces white noise, the purpose of this function is automatic car aucoustic measurements. the noise generator has an optional octave-band filter. 7.7 mss function the purpose of the music search (mss) function is to search for the next pause on a cassette tape. the output of the mss mode is the dsp_io5 pin (pin 26). this pin is "high" when the level of the input signals remain below a pre-defined level for a certain amount of time. 7.8 radio data system (rds) function the selected input for this function is either fm-rds or fm-mpx. this function offers the following features: - demodulation of the inaudible rds information, which is transmitted by fm broadcasting and is sent it to a suitable external decoder. also a internal rds decoder is available to decode the demodulated rds information. rds information is then available via i2c communication. - two tuners concept. there are two different input pins from which the rds information can be retrieved. the demodulated rds information is available by each bit or buffered by 16 bits. 7.9 second processor extension function this function offers the possibility of the addition of a second dsp which offers special, more sophisticated features such as acoustic and room effects. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 23 8 hardware application of the cdsp general in this chapter the external components are discussed, sometimes in combination with the on chip input/output circuits. how to select specific inputs and operating modes is described in chapter 9 of this manual. the external components are depicted in the cdsp application diagram which can be found in appendix 1 . it must be stressed here that this application diagram is an example, not the ultimate application diagram. there is also an application diagram of the application board (not in the usermanual) and that contains much more components in order to optimise the emc. 8.1 audio processing in dsp in this chapter the following hardware functions will be covered - analogue outputs (d/a convertors) - voltage and current reference sources - power on/off mute -dsp reset 8.1.1 analogue outputs (pin 8, 9, 13 and 15) there are 4 analogue outputs namely those which make the outputs of the 4 bitstream dacs (front left/right and rear left/right). the d/a convertors contain an internal filter so no external filter is required. the basic block diagram of one analogue output is depicted in figure 8.1. fig. 8.1 analogue output block diagram i dac r int vref - + 1 f + 100 ohm 10 kohm 10 nf analog out cdsp www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 24 an analogue output consists mainly of a current to voltage convertor. a current to voltage convertor is constructed by using the internal cmos op-amp and resistor as depicted in figure 8.1. the current from the bistream dac is converted into a voltage via rint. the full scale output voltage of the dac is 1 vrms. the dac outputs require a ac load that may not drop under the 2 k ? . in the application 10 k ? resistors are used (r26, r28, r30 and r32). the dc output voltage is the same as vrefda (typ. 1.65v). this dc is removed by the electrolytic capacitors c34, c36, c38 and c40. the cut-off frequency (and phase non-linearity) of these high pass filters depends on the dac load resistance and/or input impedance of the equipment behind the cdsp. the extra 1st order rc filter is to suppress radiation from the analogue output to the outside world (fc = 160 khz) (r27, r29, r31, r33, c35, c37, c39, c41). the cut-off frequency is not critical (component tolerance of 20% is tolerable). these filters may be omitted if considered not necessary. 8.1.2. internal reference voltage sources vrefda (pin 12) and vrefad (pin 77) the block diagram of the reference voltage sources vrefda and vrefad is depicted in figure 8.2. fig. 8.2 internal reference voltage sources block diagram the supply voltage vdda2 is divided by two internal 10 k ? resisters and buffered. the output of the first buffer is called the internal vref and is used as the reference voltage for the d/a convertors. the output of the second buffer is connected to pin vrefad (pin 77) and is internally used as the 1.65 v reference voltage of the switch capacitor d/a convertors (and buffers) of the level a/d convertor, adc1/2 and adc3/4. as filtering for the internal reference voltages a capacitor is added at the vrefda pin. in the cdsp application we use: f >= 1 khz (vripple=100 mv), ripple rejection (psrr)= typ. 60 db, crefda=22 f (c42). the vrefad voltage is also used as a dc-bias for the analogue am, tape and cd inputs via 82 k ? / 100 k ? resistors. due to the low output impedance of the buffer, the crosstalk between the analogue inputs is -74 db at dc, the external 22 f elco (c12) is added to further improve the crosstalk rejection to -80 db at 1 khz. the 47 nf capacitor (c11) is added to remove high frequency noise on the midref voltage for the a/d convertors. 10k vdda2 10k + buffer buffer int. vref for dac vrefda vrefad cdsp c refda (c42) + c11 47 nf c12 22 f dc-bias analog inputs 100 k 100 k 100 k ? ? www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 25 8.1.2.3 ref. voltages for the ad convertors vdacp (pin 75) and vdacn (pin 76) the block diagram of the reference voltages for the ad convertors is depicted in figure 8.3. fig. 8.3 block diagram reference voltages ad convertors. the voltages at the pins vdacp and vdacn1 are the reference voltages for the ad convertors, for good performing ad?s it is important that these reference voltages are clean. the external 10 ? resistor (r34) and 100 f elco (c43) filter the analogue supply voltage vddaad and are added to improve the power supply rejection ratio (psrr). in the cdsp application we use: f >= 1 khz (vripple=100 mv), ripple rejection (psrr)= typ. 39 db, c refda =100 f (c43). r i = 40 k c43 100 f + r34 10 ? www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 26 8.1.3 power on/off mute the block diagram of the power on mute (pom) circuit is depicted in figure 8.4. fig. 8.4 power on mute block diagram power on mute (pin 7) to avoid any uncontrolled noise at the audio outputs after power-on/off of the ic, the internal reference current source of the d/a converter is controlled. the capacitor on the pom pin (c32) determines the switch-on timing of this current. see figure 8.5 at power-on the the switch s is closed and the current out of the pom pin (i pom ) is 16 a, the capacitor cpom gets charged by i pom and the voltage at the pom pin (v pom ) increases linearly until v pom = 0.5 v, at this point the comparator is triggered and switch s is opened. as a result i pom becomes 112 a and v pom increases fast until it reaches vdda. as a result of this pom control the vout-ac at the dac outputs (in db) increases almost db linear from -100 dbfs till 0 dbfs. at time=t pom the dac output vout-ac = -25 dbfs, before this output voltage is reached the chip must be resetted. after the reset the chip comes automatically in the idle mode via the dsp program (see also chapter 9). this dsp program sets the outputs of the digital upsampling filters to digital silence and therefore the ac output current of the dac's to 0 a (see also in the next chapter). the time t pom as function of cpom is : t pom = 0.03125 * c pom ( c pom in f) the time t s during which the output voltage further increases from -25 dbfs to 0 dbfs is : t s = 0.8 * t pom in the cdsp application we use: tpom = 690 msec ( 20%) cpom = 22 f (c32) + c pom (c32) 22f 16a 96a comparator < 500 mv i pom pom s www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 27 fig. 8.5 switch-on timing dac tpom time [sec] vdda v pom [v] 0.5 0 -25 -100 tpom time [sec] dac v out - ac [dbfs] www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 28 power off plop suppression to avoid plops in a power amplifier, the supply voltage of the analogue part of the d/a converter and the op-amps are fed by an external voltage regulation circuit (r9, r10 and tr2) and an extra capacitor (c31) as indicated in the application diagram. during power-off the output voltage will decrease gradually, allowing the power amplifier some extra time to switch off without audible plops. in addition the pom should be pulled to zero by the processor before the power supply of the digital circuitry (vdddd1, vdddd2) is below 2.2v. below this value the digital circuitry is undefined and can therefore cause extra undesirable clicks during power-off. 8.1.4 dsp reset (pin 42) the reset pin is active low and requires an external pull-up resistor of 47 k ? . between the reset pin and the ground a capacitor should be connected to allow a proper switch-on of the supply voltage. the capacitor value is such that the chip is in reset as long as the power supply is not stabilised. a more or less fixed relationship between the dsp reset (pin 42) time constant and the pom (pin 7) time is obligatory. the voltage on the pom pin determines the current flowing in the dacs. at 0 v at the pom pin the dac currents are zero and so are the dac output voltages. at 3.3 v the dac currents are at their nominal value. long before the dac outputs get to their nominal output voltages, the dsp must be in working mode to reset the output register of the digital filter, therefore the dsp reset time constant must be shorter than the pom time (t pom ). the dsp reset input is a digital input (with hysteresis) and that means that the dsp reset circuit is enabled when vc dspres =80%vdd. in the cdsp application we use: tdspres = 68 - 92 msec (tolerance is due to tolerance of the external pull-up resistor and the capacitor tolerance (10%) cdspres = 1 f) in calculating tdspres it is assumed that vc dspres =80%vddd and the formula to calculate tdspres is: the reset has the following functions: - the bits of the i2c hardware register are set to their preset values - the program counter is set to address $0000 - dsp_io1 .. dsp_io7 when the level on the reset pin is at logical high (vc dspres =80%vddd), the dsp program in the dsp starts to run from the idle mode, resets the output registers of the digital filters and the processor can start sending commands. general timing requirements are: tdspres > tpower tpom > tdspres t pcomm > tdspres figure 8.6 gives an overview of the several time constants. vcdspres vddd e tdspres rc =?? ? () / 1 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 29 fig. 8.6 power-up timing diagram to avoid clicks . 0.7 v 80% vddd vpom vdspres pcomm tdspres= 68 - 92 msec > 150 msec tpom = 690 msec www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 30 8.2 am / fm signal quality processing level (pin 2) the basic circuit diagram of the level input is depicted in figure 8.7. fig. 8.7 level input the level signal from the tuner is divided with resistors r4 and r5 in order to match the conversion range of the level a/d convertor to the tuner properties. with r4, r5 and c4 a first order low pass filter is realised at the level input. the cut-off frequency of this filter is : the low pass filter at the input of the fm level pin has two functions: to avoid aliasing in the level a/d convertor, that means that frequency components of f > 1/2 fs a/d < 54 db below the maximum input of the level a/d convertor (figure for s/n for level a/d convertor, mentioned in data sheet) to create a filter for the multipath detector with a cut-off frequency of 34 khz ( 20%) these functions are realised with r4=27 k ? ( 10%), r3=100 k ? ( 10%) and c1= 220 pf ( 10%). the capacitor is connected to the ground plane. remarks: a) the source resistance is not taken into account because this r source << (127 k ? ) otherwise the cut-off frequency is affected. b) the input resistance of the cdsp is not taken into account because this rin is big (1.5 m ? min.). the conversion range of the a/d convertor is from 0 v to vdda1 (see spec in the data sheet). the voltage range of the fm level information has to be within this range. the total voltage range of the level information has to be >= 1/2 vdda1 to meet the minimum resolution. c4 r4 cdsp r5 ri > 1.5 m ? level fc rr r r c = ?? ? + ? 1 2 45 45 4 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 31 8.3 fm mode 8.3.1 fm input pins sel_fr (pin 61) the function of the sel_fr pin is to select in fm mode between the fm-mpx and fm-rds input. this pin has an schmitt trigger input and needs no external components because the applied signal is static (0v or 3.3v). note that this pin is not used in the application diagram (appendix 1) and therefor connected to ground. fm-mpx (pin 1) and fm-rds (pin 80) the basic circuit diagram of the fm-mpx and fm-rds inputs is depicted in figure 8.8. fig. 8.8 fm-mpx and fm-rds input fm_mpx fm_rds sel_fr b0 + - 0.189r r r r 2r 2r 2r b2 b3 b4 b5 to ad midref b0 + - 0.189r r r 2r 2r 2r b2 b3 b4 b5 to ad b1 0.707r* 2.26 r* 2.26 b1 0.707r* 2.26 r* 2.26 cdsp r19 c22 c21 fm r 1 0 0 1 0 00 0 1 1 1 1 1 0 0 1 0 0 0 0 0 11 1 1 1 c23 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 32 the 1st order rc filter (r19 and c21) at the input of the fm-mpx pin has two functions: to avoid aliasing in the a/d convertor, that means that frequency components of f > 1/2 fsa/d < 83 db below the maximum input of the a/d convertor (figure for s/n for adc1 and adc2 convertors, mentioned in data sheet) to create a filter for the fm-mpx input with a cut-off frequency > 250 khz this filter is realised with r19=1.8 k ? ( 10%) and c21=270 pf ( 10%), in combination with the fm_mpx input resistance of > 48 k ? this results in a cut-off frequency of 340 khz ( 20%). remarks: a) the source resistance is not taken into account because this r source << 27k otherwise the cut-off frequency is affected. b) the input resistance of the cdsp (> 48 k ? ) is taken into account for the cut-off frequency. c) concerning c22 and c23, x7r smd capacitors are not allowed because they show some voltage dependency which causes extra distortion, therefore np0 smd capacitors are recommended. the capacitor c22 is applied to block any dc content of the incoming signal. the capacitor c22 forms with the rin of the cdsp a high pass filter which must fulfil the following requirements: - maximum leakage current < 0.5 a, otherwise the specified dynamic range of the a/d convertor is limited by an offset voltage (i leak * rin max =v offset ); 0.5 a * 60k = 30 mv offset (in case of 200 mv input sensitivity), compared to 1.229 vrms input voltage this results in a loss of dynamic range of 0.2 db. - the cut-off frequency <= 5 hz, a higher fc limits the maximum channel separation of fm due to phase shift at 19 khz. in the cdsp application we use c22 = 1 f (mkt). - during a rds update the switch is pointed to the fm_rds pin. for a fast update it is necessary to determine the value of c23 much smaller then c22. in the application we use c23 = 330 pf (mkt). 8.3.2 fm input sensitivity selection the fm input sensitivity is designed for tuner front ends which deliver an output voltage in the range of 60 mv to 237 mv (af = 1 khz, 22.5 khz deviation). the input sensitivity can be changed in steps of 1.5 db by means of the 6 volfm bits (bits 7 ..12 of register $0ff8). the maximum gain of the fm-mpx and fm-rds input circuit is 2.26 (7.08 db), the switches in fig. 8.8 are drawn in the 7.08 db gain position. in this gain position the input sensitivity is 60 mv at 22.5 khz, the full scale input level in this case is 340 mv (0 dbfs at the input of dsp1) corresponding with a deviation of 138 khz. the input impedance of the fm-mpx and fm-rds input circuits depends on the setting of bit b0 ; in case b0 = 0 the input impedance is ?r? in case b0 = 1 the input impedance = 1.189 r. the value of the internal resistor r = 50k ? typical. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 33 the input sensitivity as function of the setting of the volfm bits is given in table 8.1. the input sensitivity of the fm-rds input can be set independent from the fm-mpx input with the 6 volrds bits (bits 1 ..6 of register $0ff8), table 8.1 also applies for the fm-rds input. the gain settings -6.4 db ... -15.4 db are not applicable for the mpx inputs because signal levels in the opamp input stage in front of the ad would exceed the rail to rail levels far before the 0 dbfs input level is reached at the dsp1 input. tuner output voltage (mv) at ? f = 22.5 khz gain (db) volfm i2c bits (11:6) 0 dbfs input level (mv) of fm-mpx and/or fm-rds via stereo decoder to dsp1 typical input impedance [ohms] 60 +7.1 $00 340 50 k 71 +5.6 $01 404 60 k 85 +4.1 $02 480 50 k 100 +2.6 $03 571 60 k 120 +1.1 $04 678 50 k 142 -0.4 $05 806 60 k 170 -1.9 $06 961 50 k 200 (default position) -3.4 $07 1135 60 k 237 -4.9 $0c 1350 50 k n.a. -6.4 $0d 60 k n.a. -7.9 $0e 50 k n.a. -9.4 $0f 60 k n.a. -10.9 $1c 50 k n.a. -12.4 $1d 60 k n.a. -13.9 $1e 50 k n.a. -15.4 $1f 60 k n.a. mute $3f 60 k table 8.1 fm input sensitivity and impedance www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 34 8.3.3 fm iac the interference absorption circuit (iac) detects and eliminates audible clicks caused by impulsive interference on fm reception. the block diagram of the iac is depicted in figure 8.9. fig. 8.9 block diagram of iac a gc feed_forward + - or a nd mpx dela y mpx_delay agc threshold monostable multivibrator suppression gate mpx input deviation detector dyn_threshold a nd en_dyn_iac + - iac on/off dynamic iac mpx iac high pass + - iac_threshold monostable multivibrator iac_stretch fm-level monostable multivibrator iac_feed_forward level iac www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 35 the iac consists of three circuits : mpx iac level iac dynamic iac the input signal of the mpx-iac circuit is the mpx signal derived from the decimated output signal of the a/d convertor. the mpx signal is fed to a delay circuit followed a gate switch. this gate is activated by the interference detector which consists of a feed forward path, an agc circuit, a comparator and a monostable multivibrator. the interference detector analyses the high frequency content of the mpx signal and discriminates between interference pulses and other signals. the mute switch interrupts the normal signal flow, during mute switch activation the output is held at a constant level which is obtained from a lpf. the mpx-iac circuit performs optimally in higher antenna voltage circumstances. the input signal of the level iac circuit is the fm level signal. this detector is added to the iac circuit in order to further optimise the iac performance at lower antenna voltage circumstances. this detection circuit is complementary to the mpx-iac detection circuit. the third iac function is the dynamic iac circuit. this function is intended to switch off the iac completely at the moment that the mpx signal has a too high frequency deviation. in case the frontend tuner has narrow if filters a too high frequency deviation will result in am modulation that could be interpreted by the iac circuitry as interference caused by the car's engine. by enabling the dynamic iac function this false triggering will be avoided. the characteristics of the iac can be adapted to the properties of different fm tuners by means of the predefined coefficients in the hardware i2c registers iac settings . the values can be changed via the i2c bus. on power on the nominal setting for a good performing iac is selected (all iac control bits set to there prefix value). note that the level iac and the dynamic iac functions are switched on after power on. there are in total 9 different coefficients which will be described in short. agc (bit 11 of iac settings register) in case the sensitivity and feed forward factor are out of range in a certain application, the set point of the agc can be shifted. threshold (sensitivity offset) (bit 2,1,0 of iac settings register) sets the threshold sensitivity of the comparator in the interference detector. it also influences the amount of unwanted triggering. feed_forward (bit 5,4,3 of iac settings register) determines the reduction of the detector sensitivity. this mechanism prevents the detector from unwanted triggering at noise with modulation peaks. suppression (bit 8,7,6 of iac settings register) sets the duration of the pulse suppression after the detector has stopped sending trigger pulses. mpx_delay (bit 10,9 of iac settings register) sets the delay time (between 2 and 5 samples of fs=304 khz) depending on the used front end of the car radio. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 36 lev_iac_threshold (bit 16,15,14,13 of iac settings register) sets the sensitivity of the comparator in the ignition interference pulse detector. it also influences the amount of unwanted triggering. with the value '0000' the level iac function is switched off. lev_iac_feed_forward (bit 18,17 of iac settings register) this parameter allows to adjust for delay differences in the signal paths from the fm antenna to the mpx mute, namely via the fm level adc and level iac detection and via the fm demodulator and mpx conversion and filtering. these differences depend on the used frontend tuner in the car radio. lev_iac_stretch (bit 20,19 of iac settings register) sets the duration of the pulse suppression after the fm level input has stopped exceeding the threshold level. lev_dyn_threshold (bit 22,21 of iac settings register) if enabled by the en_dyn_iac bit (bit 23 of the iac settings register), this block will disable temporarily all iac action in case the mpx signal exceeds a threshold deviation for a certain period of time. a higher mpx iac threshold means a lower overall trigger sensitivity, a higher deviation feed forward factor causes a lower trigger sensitivity at a non zero fm deviation. when the mpx iac suppression stretch time is increased, the suppression of the mpx signal will last longer after the last pulse detection. increasing the mpx delay causes that the suppression of the mpx signal begins and ends earlier, relative to the mpx signal itself. in case the sensitivity and feed forward factor are out of range in a certain application, the set point of the agc can be shifted with parameter agc set point; this decreases the overall sensitivity of the iac circuit. the more often and the longer the mpx signal is suppressed, the more distortion of the audio signal will be the result. in practice, the best setting of the parameters is obtained when the annoying interference pulses are eliminated and when the iac reacts only little at noise and audio signals. for the tea6811/6824 tuner, the value codes $f4caed for the iac settings register (address $0ffc) showed a good performance, also on the road. iac testing the internal trigger is visible on dsp-io4 (pin 41) if the iac_trigger bit of the iac settings register is set (bit 12). in this mode the parameter settings on the iac performance can be verified. the iac can be tested with the setup given in figure 8.10. the schematics of the interference simulation network (isn) and the dummy antenna are given in figure 8.11 and 8.12 respectively. the rise time of the pulse generator has to be faster than 5 nanoseconds. the loaded voltage amplitude must be circa 10 volts. note that the isn attenuates the fm signal by circa 20 db, so the rf signal output of the generator should be compensated for this. ignition interference of a four cylinder engine running at 6000 rev/min can be simulated by setting the frequency of the pulse generator at 100 hz and the duty cycle at 50%. note that the rising edge as well as the falling edge of the square wave causes a pulse in the rf signal to the receiver. when the iac is switched off, each 5 milliseconds a pulse can be expected in the audio signal. however, because of the random phase relation between the square wave and the fm signal, the amplitude of the pulses in the audio signal varies and can even be zero. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 37 fig. 8.10 iac test setup fig. 8.11 isn schematics fig. 8.12 dummy antenna schematics receiver with cdsp dummy antenna interference simulating network -20 db 50o rf gen pulse generator iac trigger pulse to oscilloscope 6.8 pf 50 ? 3.3 pf from rf-gen to dummy ant interf. sim. network from pulse-gen metal box 55 ? from isn to receiver dummy antenna 4.3 ? 120 ? metal box www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 38 optimisation procedure iac parameters using the measurement set-up in figure 8.10, the parameters of the iac can be optimised. (all rf-voltages measured at the input of the dummy antenna. signal to noise is relative to 22.5 khz deviation). the iac characteristics can be adapted by means of the iac settings registers. this register contains the control bits that define the iac parameters. for monitoring the iac trigger pulse at pin dsp-io4 (pin 41), bit iac_trigger of the iac settings register ($0ffc) should be set to 1. see also the datasheet of the saa7709h. the complete iac function can be optimised in three steps : 1) optimise the mpx iac 2) optimise the dynamic iac 3) optimise the level iac 1. optimisation of the mpx iac figure 8.13 gives a graphical presentation of the effect that the mpx iac parameters have on the interference noise and mpx signal. the procedure to optimise the mpx iac parameters is described below. fig. 8.13 graphical presentation mpx iac parameters switch off the level iac and dynamic iac functions (set bits 13,14,15,16 and 12 of the iac settings register $0ffc to '0'). the optimisation of the mpx iac can be done by listening, optimise the parameters in the order as given below. note that the optimisation procedure is iterative, in some circumstances it is needed to re-adjust an already optimised parameter in order to get the overall optimal results. interference puls detected gate c5 : mpx_delay c4 : suppression stretch time c2 : threshold (sensitivity) high audio modulation noise c3 : deviation feed_forward factor www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 39 adjustment of iac parameter : threshold (sensitivity) use no isn. no modulation. set feed_forward to minimum (0.00000). by decreasing the rf voltage, the snr decreases (some sputtering noise can be heard now). changing threshold causes unwanted trigger pulses. after increasing the rf voltage (which causes also snr to increase), the unwanted trigger pulses should disappear. increase of snr due to unwanted trigger pulses has to be less than1 db, this requirement has to be met for any fieldstrength, this can be verified by comparing the audible noise in case of mpx iac switched on with the audible noise in case of mpx iac switched off. the audible noise in case of switching on the mpx iac should not increase significantly. adjustment of iac parameter : feed_forward use no isn. modulation: fmod=1 khz, deviation=75 khz. choose the rf voltage amplitude so that some trigger pulses occur around the zero crossings of the audio signal. adjust feed_forward to the value at which the number of unwanted trigger pulses around the peaks of the audio signal is about the same as the number around the zero crossings. adjustment of iac parameters suppression and mpx_delay use the isn and create interfering pulses. make the rf voltage so low that the iac is still sensitive for pulses. (at lower voltages the iac will be too sensitive for pulses because of the noise). at this rf voltage, the suppression time is shortest, so the timing of the beginning and the end of the suppression period is most critical. set suppression and mpx_delay to maximum. the beginning of an interference pulse or maybe the whole pulse is suppressed now. reduce mpx_delay to the value at which the beginning of the pulse is still just eliminated. then, reduce suppression so much that the tail of the pulse is just suppressed well. the adjustment of suppression and mpx_delay can be done by listening. 2. optimisation of the dynamic iac the level iac function should remain switched off; mpx iac adjusted and switched on. adjustment of iac parameter : dyn_threshold use no isn. set the rf voltage to 200 v. modulation: fmod=10 khz, deviation= 22.5 khz. first switch off the dynamic iac. increase the deviation until the mpx iac starts (unwanted) triggering. now switch on the dynamic iac and set dyn_threshold to maximum deviation (65 khz). adjust (decrease) dyn_threshold to the value at which the unwanted trigger pulses just disappear, now the highest deviation at which no unwanted triggering occurs is achieved. 3. optimisation of the level iac switch off the dynamic iac. mpx iac adjusted and switched on. adjustment of iac parameter : iac_threshold use isn and create interfering pulses. no modulation. set iac_threshold to maximum (0.5), iac_feed_forward to minimum (-2) and iac_stretch to maximum (15). apply a low rf voltage amplitude such that the mpx iac stops triggering. decrease iac_threshold until the level iac starts triggering. the sensitivity can be increased by further decreasing iac_threshold but then also false triggering on audio can occur; this can be verified by removing the interfering pulses. choose iac_threshold such that there is no false triggering and that the level iac sensitivity is sufficient at a lower fieldstrength. it is recommended not to set the level iac threshold lower than 0.05 , otherwise false triggering can occur at certain fm-level dc values. adjustment of iac parameter : iac_feed_forward and iac_stretch create interfering pulses and verify the audio output signal. increase iac_feed_forward to the value at which the beginning of the interfering pulse is still just eliminated. then, reduce iac_stretch so much that the tail of the pulse is just suppressed well. the adjustment of iac_feed_forward and iac_stretch can be done by listening. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 40 8.4 am, aux, tape, phone, nav and cd_a inputs from a hardware point of view the input circuits in front of the ad?s are the same for these inputs. first the internal input circuit common for the analogue inputs will be discussed here, the specific external components for the different analogue inputs will be discussed in the following sections. the internal input circuit common for the analogue inputs is given in figure 8.14. fig. 8.14 input circuit cd_a / tape / am / phone / nav / aux the switches are controlled via hardware i2c registers with the following bits : for adc1 : ? ansel1 (bits 0, 1 and 2 of register $0ffa) selects which input source is connected ? gnd_sel1 (bits 12 and 13 of register $0ffa) selects between internal ground (midref) or an external ground (cd_gndl, nav_gnd or phone_gnd) ? dif_sw1 (bit 1 of register $0ff9) selects a high common mode or fully differential input mode. for adc2 : ? ansel2 (bits 3, 4 and 5 of register $0ffa) selects which input source is connected ? gnd_sel2 (bits 14 and 15 of register $0ffa) selects between internal ground (midref) or an external ground (cd_gndl, nav_gnd or phone_gnd) ? dif_sw2 (bit 2 of register $0ff9) selects a high common mode or fully differential input mode. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 41 for adc3 : ? ansel3 (bits 6, 7 and 8 of register $0ffa) selects which input source is connected ? gnd_sel3 (bits 16 and 17 of register $0ffa) selects between internal ground (midref) or an external ground (cd_gndl, nav_gnd or phone_gnd) ? dif_sw3 (bit 3 of register $0ff9) selects a high common mode or fully differential input mode. for adc4 : ? ansel4 (bits 9, 10 and 11 of register $0ffa) selects which input source is connected ? gnd_sel4 (bits 18 and 19 of register $0ffa) selects between internal ground (midref) or an external ground (cd_gndl, nav_gnd or phone_gnd) ? dif_sw4 (bit 4 of register $0ff9) selects a high common mode or fully differential input mode. notes : 1. the input selection related i2c bits are automatically set by the easy programming source switching commands as described in chapter 9.0.3. 2. the full scale input level of 660 mvrms at the cd_a, aux, nav, phone, tape and am inputs corresponds with an input level of 0 dbfs at the input of the dsp. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 42 8.4.1 am inputs am-r/am_mono and am_l inputs (pin 66 and pin 67) the function of the am inputs can be either : 1. input for am mono : am mono applied to am_r/am_mono input pin. 2. input for am stereo with external am-stereo decoder : am left applied to am-l pin and am right applied to am_r/am_mono input pin. 8.4.1.1 am inputs for am-mono mode and am-stereo mode with external decoder the basic circuit diagram for the am-l and am-r/am_mono input (for am mono or am stereo with external stereo decoder is depicted in figure 8.15. the use of the internal ground connection is required for the am inputs. fig. 8.15 am left/right input in case of am-mono mode the am-af output of the tuner has to be connected to the am-r/am input of the cardsp, this section deals with both the left- and right input because they are identical from a hardware point of view. the resistor combinations r11 and r16 (or r13 and r18) attenuates the input signal in order to match the am output voltage of the tuner to the input range of the a/d convertor. biasing is done via the resistor r16/r18. for the cdsp application the am output voltage of the tuner is assumed to be about 1 vrms max, r11 and r13 are 100 k ? , this results in a voltage of 545 mvrms at the af-am left/right pins of the cdsp. with r11, c14 and r13, c16 a first order low pass filter is realised at the left and right inputs respectively. the cut-off frequency of this filter (left channel example) is : r11/r13 cdsp r16/r18 a m_l/r c14/c16 a m-l/r vrefad c13/c15 ri > 1 m ? 14 16 11 16 11 2 1 c r r r r fc ? + ? ? ? = www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 43 the 1st order filter at the input of the am left/right pins is to avoid aliasing in the audio a/d convertors, that means that frequency components of f > 1/2 fsa/d < 88 db below the maximum input of the a/d convertor (figure for s/n for am-inputs mentioned in the data sheet). the requirements are not critical though. if r11/r13 = 45 k ? (+/-10%) , ri = 1 m ? (+/-20%)and c14/c16 = 100 pf (+/-10%), then the cut-off frequency fc = 36.9 khz. remarks: a) the source resistance is not taken into account because this r source << 220 k ? otherwise the cut-off frequency is affected. b) concerning c14/c16, x7r smd capacitors are not allowed because they show some voltage dependency which causes extra distortion, therefore np0 smd capacitors are recommended. the capacitor c13/c15 is applied to block any dc content of the incoming signal. the capacitor c13/c15 forms with r11+ri / r13+ri a high pass filter but there are no critical requirements. for the cdsp application we use : c13/c15 = 220nf (+/-10%) and rin = 220 k ? (+/-10%), resulting in a fc = 3.3 hz (+/-20%). local oscillator frequency accuracy the cquam am-stereo decoder algorithm that runs in the dsp locks and tracks the incoming 9.5 khz if signal and allows a maximum tolerance of +/- 250 hz on the 9.5 khz if frequency. the north american broadcast specification allows for 20 hz broadcast frequency error, so this leaves a maximum tolerance of +/- 230 hz for the if tuner and if downconvertor in front of the saa7709h. if for example the if-mixer inside the am-stereo tuner has a tolerance of +/- 40 hz than the maximum tolerance of the local oscillator in fig 8.17 above is 459.5 khz +/- 190 hz ; this means that the overall accuracy of the 7.353 mhz xtal in fig 8.17 is 400 ppm. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 44 8.4.2 tape / aux input the tape and aux input can configured the same. below there is an example given of the tape input. tape left / right (pin 68 and pin 69) the tape input is for connecting a cassette deck. the basic circuit diagram of the tape-l and tape-r input is depicted in figure 8.18 ; it is assumed that the internal ground is used for the tape inputs (single ended inputs). fig. 8.18 tape left/right input with r15/r17 = 56 k ? ( 10%), r16/r18 = 100 k ? ( 10%) and c18/c20 = 100 pf ( 10%) a low pass input filter with cut-off frequency fc = 44.3 khz is realised. the capacitor c17/c19 is applied to block any dc content of the incoming signal. the capacitor c17/c19 forms with r15+r16 / r17+r18 a high pass filter but there are no critical requirements. for the cdsp application we use : c17/c19 = 100pf ( 10%) and rin = 156 k ? ( 10%), resulting in a fc = 10.2 hz ( 20%). remarks: a) the source resistance is not taken into account because this r source << 150 k ? otherwise the cut-off frequency is affected. b) the input resistance of the cdsp is not taken into account because this rin 1 m ? . c) concerning c18/c20, x7r smd capacitors are not allowed because they show some voltage dependency which causes extra distortion, therefore np0 smd capacitors are recommended. r15/r17 cdsp r16/r18 tape l/r c18/c20 tape-l/r vrefad c17/c19 ri > 1 m ? www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 45 8.4.3 analogue cd input cd left/right and cd-ground left/right (pin 70, 71, 72 and 73) the analogue cd input is for connecting the analogue output signal of a cd player or cd-changer. the saa7709h handles fully differential and/or high common mode cd players/changers. high common mode cd player/changer: in order to have a high level of common-mode rejection and thus eliminates ground noise it is required to use the external ground (cd_gndl and cd_gndr) pins. the ground wire of the cd-cable has a separate input at the cdsp in order to realise the required level of common mode rejection ratio. the basic circuit diagram of the analogue cd-input with high common mode inputs is depicted in figure 8.19a. fig. 8.19a high common mode analogue cd input low pass filter the filtering of the incoming signal is a first order rc low pass filter realised with the resistors r6, r8 and the capacitor c9 for the left channel input and with r7, r9 and the capacitor c10 for the right channel input. the cut-off frequency (fc) of this filter (left channel) is : in the cdsp application we use r6 = r7 = 8.2 k ? ( 10%), r8 = r9 = 10 k ? ( 10%). in combination with the capacitor c9 = c10 = 1 nf ( 10%), this results in a cut-off frequency fc = 35 khz ( 20%). fc rr r r c = ?? ? + ? 1 2 68 68 9 + - + - + - + - a d1 a d2 0 1 2 3 4 5 0 1 2 3 4 5 0 1 2 3 0 1 2 3 ansel_sel1 ansel_sel2 gnd_sel1 gnd_sel2 dif_sw1 dif_sw2 cd_l cd-cable ground cd_r cd_gndr cd_r cd_gndl cd_l c6 r6 r8 c9 c7 c8 + r7 cdsp r9 c10 vrefad r10 c12 + www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 46 variable gain the gain of the left- and right input stages is adjustable and equals r8/(r6+r8) and r9/(r7+r9) respectively. the 0 db output level of the input stage opamps is 0.55 vrms. in the cdsp application we assume a level of 1 vrms at the cd-l and cd-r application input. it is recommended to change the resistor values if a different input gain is required, such that the equivalent parallel resistance r3//r4 and r5//r6 remains < 5 k ? for optimal cmrr characteristics. remarks: the capacitors c6 and c8 are applied to block any dc content of the incoming signal. the capacitors c6/c8 forms with (r6+r8) and (r7+r9) a high pass filter. the cut-off frequency of this filter must be 15 hz. with r6 = r7 = 8.2 k ? and r8= r9= 10 k ? this means c2,c3 583 nf. capacitor c7 is applied to block the dc-bias voltage at the cd-gnd pin, in the cdsp application we use c7 = 47 f ( 10%). fully differential player/changer: the basic circuit diagram of the analogue cd-input with fully differential inputs is depicted in figure 8.19b. fig. 8.19b full differential analogue cd input - + - ad1 0 1 2 3 4 5 0 1 2 3 ansel_sel1 gnd_sel1 dif_sw1 cdl_pos cdl_neg cd_gndl cd_l c6 r6 r8 c9 c8 r7 cdsp r9 c10 vrefad + - + - ad2 0 1 2 3 4 5 0 1 2 3 ansel_sel2 gnd_sel2 dif_sw2 cdr_pos c dr_neg cd_gndr cd_r c6 r6 r8 c9 c8 r7 r9 c10 c12 r10 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 47 analogue cd input with floating source in case the signal ground of the cd player is floating (e.g. a battery supplied cd player) the input circuit as given in figure 8.20 does not work due to the high input impedance of the circuit. this problem can be solved by replacing resistor r10 with two diodes connected in anti-parallel, see figure 8.21. recommended diode types are bas216 or baw62 . fig. 8.21 analogue cd input with floating cd player 8.5 phone and navigation inputs the saa7709h has separate inputs for phone and navigation. these inputs have their own ground input therefor several different configurations are possible, such as : single ended, high common mode and full differential mode. the basic circuit diagram is given in figure 8.22. as example the phone input is a differential input and the nav input is single ended. + - + - + - + - a d1 a d2 0 1 2 3 4 5 0 1 2 3 4 5 0 1 2 3 0 1 2 3 ansel_sel1 ansel_sel2 gnd_sel1 gnd_sel2 dif_sw1 dif_sw2 cd_l cd-cable ground cd_r cd_gndr cd_r cd_gndl cd_l c6 r6 r8 c9 c7 c8 + r7 cdsp r9 c10 vrefad c12 bas216 baw62 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 48 fig. 8.22 phone and navigation inputs the input circuit diagram as given in fig. 8.22 shows that both the phone and navigation input have a separate ground input pin. in this case it is assumed that the ground wire phone input source is connected with the phone_gnd pin in order to realise a differential input, for the navigation input it is assumed that the nav_gnd pin is not used (single ended input) and therefor connected via c5 to ground. it is of course possible to connect the ground wire of the navigation input source with the nav_gnd pin, similar as for the phone input if desired. the external components of the phone and navigation inputs have the following functions : ? adapt the source signal amplitude to the maximum input voltage of the cdsp ? input filtering input sensitivity of the phone input in the cdsp application we assume that the external phone source delivers a signal of 1 vrms maximum. the full scale input level (0 db) of the a/d convertor is 660 mvrms, the phone source voltage has to be attenuated accordingly. c2 r2 r1 c1 r3 c3 phone phone_gnd c5 r11 c13 c14 nav_gnd nav phone navigation vrefad c4 r4 r5 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 49 this results in the equivalent circuit diagram for determining the overall gain of the phone input according the diagram below. in the cdsp application we use ra=3.9 k ? (+/- 1%) and rb=10 k ? (+/- 10%); rv=1 m ? . also high impedance inputs are possible. the overall input attenuation is: the overall gain of the phone-input now matches the 1 vrms phone source to the full scale input level of the a/d convertor (=0.66 vrms). cmmr the cmmr of the phone input depends on the matching of the internal resistors and the external resistors r1 and r2, for this reason the maximum tolerance of these resistors must be 1% in order to achieve typical 50 db cmmr as specified in the saa7709h datasheet. input filtering the equivalent electrical diagram for determining the phone input filtering is given in the figure below : low pass filter ra cdsp rv phone phone gnd rb ra 1 vrms full scale = 0.66 vrms att r b r v rb rv ra = +? = (//) (//) ) . 2 065 r1 cdsp rv c3 c1 phone phone_gnd r3 r2 c2 vi www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 50 with r1, r2, r3 and c3 a first order low pass filter is realised at the phone input. the cut-off frequency of this filter is : the 1st order filter at the phone input is to avoid aliasing in the audio a/d convertor. the requirements are not critical though. remark: concerning c59, a x7r smd capacitor is not allowed because it shows some voltage dependency which causes extra distortion, therefore np0 smd capacitors are recommended. high pass filter the capacitors c1 and c2 are applied to block any dc content of the incoming signal. the capacitors c1 and c2 forms with r1, r2 and r3 a first order high pass filter but there are no critical requirements. the cut-off frequency of this filter is : for the cdsp application we use : () () fc ri c ri rr rrv rrv rr rrv r r v = ?? ? = +? ? + ++ ? + 1 23 12 3 3 12 3 3 with () fc rc rrr rrv rrv c cc c c hh h h = ?? ? =+ + ? + = ? + 1 2 12 3 3 12 12 with and www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 51 input sensitivity of the navigation input in the cdsp application we assume that the external phone source delivers a signal of 1 vrms maximum. the full scale input level (0 db) of the a/d convertor is 660 mvrms, the navigation source voltage has to be attenuated accordingly, see figure 8.23. fig. 8.23 navigation input r11 cdsp r5 nav c14 nav vrefad c13 ri > 1 m ? www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 52 8.6 cd-digital inputs (i2s or spdif) 8.6.1 general the dsp runs at the sample frequency of the selected digital input (i2s or spdif), supported sample frequencies are 44.1 khz and 48 khz ; sample frequency of 32 khz is not supported . the saa7709h detects when the selected digital source becomes disconnected for some reason and as a result dsp continues running at 44.1 khz sampling frequency (derived from xtal oscillator). 8.6.2 i2s input i2s inputs cd-cl (pin 29), cd-ws (pin 27) and cd-data (pin 28) the digital inputs digin is capable of handling multiple input formats (i2s and lsb-justified). if the i2s input pins are not used they must be connected to ground as is indicated in the application diagram. if they are used then in every input line a t-filter can be used, see figure below. the t-filter is used to avoid incoming and outgoing radiation (component tolerance of the t-filter is 20%). in case the i2s signals come from a device with slew rate controlled outputs, this t-filter might be unnecessary. please note that this filter is optimised for a bitrate of 64*fs ; the rise- and fall times of the i2s signals will be too high when a higher bitrate is used, in this case the component values of the filter should be adapted in order to compensate for this. if the i2s driver outputs of the external digital source ic's have tri-state outputs, they can all be connected on one single i2s input. (not used outputs must be put in the high impedance mode). 100 ? from i 2 s source 100 pf 100 ? to cdsp www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 53 8.6.3 spdif input spdif inputs spdif1 (pin 25) and spdif2 (pin 24) the spdif input can be used as an alternative for the i 2 s input. the two spdif inputs are connected to the internal spdif receiver via an analogue multiplexer (switch) that selects between spdif1 and spdif2. the recommended input circuit is given in figure 8.24, see the saa7709h datasheet for additional information. fig. 8.24 spdif input circuit 8.7 radio data system (rds) function 8.7.1 general description the rds function recovers the additional inaudible rds information which is transmitted by fm radio broadcasting. the operational functions of the demodulator and decoder are in accordance with ebu specification en 50067. the rds function processes the rds signal that is frequency multiplexed in the stereo-multiplex signal, to recover the information transmitted over the rds data channel. this processing consists of band- pass filtering, rds demodulation and rds/rbds decoding. under control of iic bit rds_clkin, an internal buffer can be used to read out the raw rds stream in bursts of 16 bits. with the iic-bit rds_clkout the rds clock can be enabled or switched off. the rds-band signal level in iic bits rds_det of register iic_rds_detection supports fast rds presence detection. the rds band-pass filter discards the audio content from the input signal and reduces the bandwidth. the rds-band signal level detector removes a possible ari signal from the rds band-pass filter output and measures the level of the remaining signal. the rds demodulator regenerates the raw rds bit stream (bit rate = 1187.5 hz) from the modulated rds signal in two steps. the first step is the demodulation of the double-side-band suppressed- carrier signal around 57 khz into a baseband signal, by carrier extraction and down- mixing. the second step is the bpsk demodulation of the biphase coded baseband signal, by clock extraction and correlation. 100nf 100pf 75 ohm spdif input spdif input pin 24/25 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 54 fig. 8.25 : rds/rbds block diagram the rds/rbds decoder provides block synchronization, error detection, error correction, complex flywheel function and programmable block data output. new processed rds/rbds block information is signalled to the main microcontroller as ?new data available? by use of the davn output. the block data itself and the corresponding status information can be read out via iic-bus request. 8.7.2 rds bandpass filtering the rds-chain has a separate input fm_rds. this enables rds updates during tape or other analog source play. the rds chain contains a third order sigma-delta ad convertor, followed by two decimation filters. the first filter passes the multiplex band including the signals around 57 khz and reduces the sigma- delta noise. the second filter reduces the rds bandwidth around 57 khz. the overall filter curve is shown in fig. 8.26 and a more detailed curve of the rds 57khz band in fig. 8.27. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 55 fig. 8.26 : overall frequency response curve decimation filters fig. 8.27 : detailed frequency response curve rds channel in case of fm-stereo reception the clock of the total chip is locked to the stereo pilot (19 khz multiple). in case of fm-mono the dcs loop keeps the dcs clock around the same 19 khz multiple. in all other cases like am reception or tape, the dcs circuit has to be set in a preset position by means of the locked_preset bit of the iic_dcs control register. under these conditions the rds system is always clocked by the dcs clock in a 38 khz (4*9.5 khz) based sequence. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 56 8.7.3 direct rds inputs/outputs in davd mode (dac0=1, dac1=1, rds decoder bypass mode) apart from control inputs and data outputs via iic, the following inputs and outputs are related to the rds function (see fig. 8.25) . ? unbuffered raw rds output mode (rds_clkin = 0, rds_clkout = 1) rds_clk: clock of the raw rds bit stream, extracted from the biphase coded baseband signal by the rds demodulator. clock period 1.1875 khz (= 8192 clock cycles of the dcs system clock), 50% duty cycle. the positive edge can be used to sample the rds_data output with. ? rds_data: raw rds bit stream, generated by the demodulator, detection of a positive going edge on the rdcl input signal. the data output is changing 100 ms (= 1/8 of the rds_bck period) after the falling edge of rds_bck. this allows for external receivers of the rds data to clock the data on the rds_bck signal as well as on its inverse. buffered raw rds output mode (rds_clkin = 1, rds_clkout = 0) ? rds_clk: burst clock, generated by the mp. bursts of 16 clock cycles are expected. the average time between bursts = 13.5 ms. ? rds_data: bursts of 16 raw rds bits are put out under control of the burst clock input. after a data burst, this output is high. it is pulled low when 16 new bits are available and a new clock burst is awaited.the microprocessor has to monitor this line at least every 13.4 ms. 8.7.4 direct rds timing of clock and data signals in davd mode (dac0=1, dac1=1, rds decoder bypass mode ) the timing of the clock and data output is derived from the incoming data signal. under stable conditions the data will remain valid for 400 ms after the clock transition. the timing of the data change is 100 ms before a positive clock change. this timing is suited for positive as well as negative triggered interrupts on a microprocessor. the rds timing is shown in fig. 8.28. during poor reception it is possible that faults in phase occur, then the duty cycle of the clock and data signals will vary from minimum 0.5 times to a maximum of 1.5 times the standard clock periods. normally, faults in phase do not occur on a cyclic basis. fig. 8.28 :rds timing in the direct output mode www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 57 8.7.5 buffering of rds data the repetition of the rds data is around the 1187 hz. this results in an interrupt on the microprocessor for every 842 us. in a second mode, the rds interface has a double 16 bit buffer. 8.7.6 buffer interface the rds interface buffers 16 data bits. every time 16 bits are received, the data line in pulled down and the buffer is overwritten. the control microprocessor has to monitor the data line in at most every 13.5 msec. this mode is selected by setting the rds_clkin iic bit to ?1? and rds_clkout to ?0? (see ?iic_rds_control register ($6005)). in fig. 8.29 the interface signals from the rds demodulator and the microcomputer in buffer mode are shown. when the buffer is filled with 16 bit the data line is pulled down. the data line will remain low until reading of the buffer is started by pulling down the clock line. the first bit is clocked out. after 16 clock pulses the reading of the buffer is ready and the data line is set high until the buffer is filled again. the microprocessor stops communication by pulling the line high. the data is written out just after the clock high-low transition. the data is valid when the clock is high. when a new 16 bit buffer is filled before the other buffer is read, that buffer will be overwritten and the old data is lost. fig. 8.29 : interface signals rds demodulator and microcomputer the timing figures can be found in table 8.2. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 58 table 8.2 : timing of the rds symbol parameter conditions min typ max unit no frdscl nominal clock frequency rds clock - 1187.5 - hz 8.01 tsr clock set-up time 100 - - s 8.02 tpr periodic time - 842 - s 8.03 thr clock high time 220 - 640 s 8.04 tir clock low time 220 - 640 s 8.05 tdr data hold time 100 - - s 8.06 twb wait time 1 - - s 8.09 tpb periodic time 2 - - s 8.10 thb clock high time 1 - - s 8.11 tlb clock low time 1 - - s 8.12 fexcl input frequency extern rds- clock --22mhz8.13 8.7.3 fast rds detection with the rds-band signal level detector rds presence detection after tuning to a new fm station using only the rds demodulator/decoder will take 130 ms. the special for fast rds detection designed rds-band signal level detector supports rds presence detection in 10 ms after the front-end is tuned to the selected frequency. the band-pass filtered input of the rds demodulator is first passed through a notch filter to discard a possible ari signal in the band [57 khz- 54 hz, 57 khz + 54 hz]. the designed filter has 2 passbands with 3 db frequencies at 55.4 khz and 56.6 khz and at 57.4 khz and 58.6 khz respectively. the overall filter characteristic around 57 khz, from the output of the src to the output of the ari notch, is shown in fig 8.30 compares the detector filter characteristic with the spectra of rds signals (deviation = 0.8 khz) with zero, random, one and toggled messages and the spectrum of an ari signal (deviation = 7.5 khz) with worst case subcarriers (sk + dk + bk area f) and with a 12 hz skew between the centre frequency of the filter and the ari arrier. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 59 fig. 8.30 : rds detector: band-pass filter characteristics fig. 8.31 : rds detector filter versus various signal spectra the remaining signal is rectified and averaged with a first order low-pass filter with a time constant of 6.75 ms. the output of this filter is a measure for the signal and noise content in the rds band. the rds-band signal level in the iic_rds_detector register, is an 8-bit unsigned number between 0 and 0.996. for an rds deviation below 4 khz, the 8-bit output rds_det is approximately detector output = rds_det value = rds deviation [in khz] / 4 www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 60 for all zero or all one messages, the stationary detector output differs approximately 20% from this value. for random messages, the detector output varies between 10% bounds. so, rds_det = 0.2 +/- 0.02 for a deviation of 0.8 khz, 0.5 +/- 0.05 for a deviation of 2.0 khz in case of random messages. for rds deviations above 4 khz, the detector output saturates to the maximum value. simulations show that the detector output reaches 90% of its stationary value after less than 14 ms. an ari signal with maximal signal content away from 57 khz (sk + dk + bk area f and 12 hz carrier frequency versus filter centre frequency skew) and a deviation of 7.5 khz, causes a wiggling detector output with maxima below 0.14. if a threshold of 0.18 is used to detect rds presence, detection takes 12.6 ms for an rds deviation of 0.8 khz and 3.4 ms for an rds deviation of 2.0 khz. fig. 8.32 shows various detector output transients after the selection of the input stereo-mpx signals containing rds or ari signals. fig. 8.32 : rds output for various signals the noise in the rds band adds to the detector output. consider the case of a nominal rds deviation of 2.0 khz and white noise measured over the band of the rds main lobe, i.e. between 54.6 khz and 59.4 khz. a noise power of 25 db below the rds power level adds 0.043 to the detector output. every 6 db more noise doubles the noise contribution to the detector output. a threshold for rds presence indication can be made dependent on the noise level measured with the wide-band or narrow-band noise detector. 8.7.4 bitslip phase jumps of the extracted rds clock are detected and accumulated. if the accumulated phase shift exceeds a threshold, the rds/rbds decoder is informed by the bslp signal (see fig. , page ). if the rds/rbds decoder detects a bitslip, the rds demodulator is informed by the bpsa signal. this causes the accumulator of rds clock phase shifts to be cleared. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 61 8.7.5 rds/rbds decoder the rds/rbds decoder handles the complete data processing and decoding of the continuously received serial rds/rbds demodulator output data stream (rdda,rdcl). different data processing modes are software controllable by the external main controller via iic-bus request. all control-signals are direct inputs to the decoder and are connected to the outputs of the iic memory map interface. processed rds/rbds data blocks with corresponding decoder status information are available via iic- bus. also the output signals of the decoder are direct outputs and are connected to the inputs of the iic- bus memory map interface. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 62 as already mentioned, the rds/rbds decoder contains the following functions. ? rds and rbds block detection ? error detection and correction ? fast block synchronization ? synchronization control (flywheel) ? mode control for rds/rbds processing ? different rds/rbds block information output modes (e.g. a / c? block output mode) the functions which are realized in the decoder are described in detail within the next sections. 8.7.6 rds/rbds block detection the rds/rbds block detection is always active. for a received sequence of 26 data bits a valid block and corresponding offset are identified via syndrome calculation. during synchronization search, the syndrome is calculated with every new received data bit (bit-by-bit) for a received 26-bit sequence. if the decoder is synchronized, syndrome calculation is activated only after 26 data bits for each new block received. under rbds reception situation, besides the rds block sequences with (a, b, c/c', d) offset also block sequences of 4 blocks with offset e may be received. if the decoder detects an 'e- block', this block is marked in the block identification number (blnr<2:0>) and is available via iic-bus request. in rbds processing mode the block is signalled as valid 'e-block' and in rds processing mode, where only rds blocks are expected, signalled as invalid 'e-block'. this information can be used by the main controller to detect 'e-block' sequences and identify rds or rbds transmitter stations. 8.7.7 error detection and correction the rds/rbds error detection and correction recognizes and corrects potential transmission errors within a received block via parity-check in consideration of the offset word of the expected block. burst errors with a maximum length of 5 bits are corrected with this method. after synchronization has been found the error correction is always active depending on the pre- selected 'error correction mode for synchronization' (mode synca... syncd), but cannot be carried out in every reception situation. during synchronization search, the error correction is disabled for detection of the first block and is enabled for processing of the second block depending on the pre-selected 'error correction mode for synchronization' (mode synca... syncd). the processed block data and the status of error correction are available for data request via iic- bus for the last two blocks. table 8.3 :rds processed error correction exb1 exb0 description 0 0 no errors detected 0 1 burst error of max. 2 bits corrected 1 0 burst error of max. 5 bits corrected 1 1 uncorrectable block processed blocks are characterized as uncorrectable under the following conditions: ? during synchronization search, if the burst error (for the second block) is higher than allowed by the pre-selected correction mode synca... syncd. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 63 ? after synchronization has been found, if the burst error exceeds the correctable max. 5 bit burst error or if errors are detected but error correction is not possible. 8.7.8 synchronization the decoder is synchronized if two valid blocks in a valid sequence are detected by the block detection. the search for the first block is done by a bit-by-bit syndrome calculation, starting after the first 26 bits have been received. this bit-by-bit syndrome calculation is carried out until the first valid and error free block has been received. then the next expected block is calculated and syndrome calculation is done after the next 26 bits have been received. the block-span in which the second valid and expected block can be received is selectable via previously setting of the max_bad_blocks_gain (mbbg<4:0>). if the second received block is an invalid block, then the bad_blocks_counter is incremented and again the new next expected block is calculated. if the bad_blocks_counter value reaches the pre-selected max_bad_blocks_gain, then the bit-by- bit search for the first block is started again. if synchronization is found, the synchronization status flag (sync) is set and available via iic- bus request.the synchronization is held until the bad_blocks_counter value reaches the pre-selected max_bad_blocks_lose value (used for synchronization hold) or an external restart of synchronization is performed (nwsy=1; or power-on reset). 8.7.9 flywheel for synchronization hold for a fast detection of loss of synchronization an internal flywheel is implemented. therefore one counter (bad_blocks_counter) checks the number of uncorrectable blocks and a second counter (good_blocks_counter) checks the number of error free or correctable blocks. error blocks increment the bad_blocks_counter and valid blocks increment the good_blocks_counter. if the counter value of the good_blocks_counter reaches the pre-selected max_good_blocks_lose value (mgbl<5:0>) then good_blocks_counter and bad_blocks_counter are reset to zero. but if the bad_blocks_counter reaches the pre-selected max_bad_blocks_lose value (mbbl<5:0>) then new synchronization search (bit-by- bit) is started (sync=0) and both counters are reset to zero. the flywheel function is only activated if the decoder is synchronized. the synchronization is held until the bad_blocks_counter reaches the pre-selected max_bad_blocks_lose value (loss of synchronization) or an external forced start of new synchronization search (nwsy=1) is performed. the maximum values for the flywheel counters are both adjustable via iic-bus in a range of 0 to 63. 8.7.10 bit slip correction during poor reception situation phase shifts of one bit to the left or right /- 1 bit slip) between the rds/rbds clock and data may occur, depending on the lock conditions of the demodulators clock regeneration. if the decoder is synchronized and detects a bit slip (bslp=1), the synchronization is corrected by +1, 0 or -1 bit via block detection on the respectively shifted expected new block. 8.7.11 data processing control the decoder provides different operating modes selectable by nwsy, sym0, sym1, dac0 and dac1 inputs via the external iic-bus. the data processing control performs the pre-selected operating modes and controls the requested output of the rds/rbds information. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 64 8.7.12 restart of synchronization mode: the 'restart synchronization' (nwsy) control mode immediately terminates the actual synchronization and restarts a new synchronization search procedure (nwsy=1). the nwsy flag is automatically reset after the restart of synchronization by the decoder. this mode is required for a fast new synchronization on the rds/rbds data from a new transmitter station if the tuning frequency is changed by the radio set. restart of synchronization search is furthermore automatically carried out if the internal flywheel signals a loss of synchronization. 8.7.13 error correction control mode for synchronization: for error correction and identification of valid blocks during synchronization search as well as synchronization hold, four different modes are selectable (sym1, sym0). ? mode synca (sym0=0, sym1=0): no error correction; blocks detected as correctable are treated as invalid blocks, internal bad_blocks_counter still incremented even if correctable errors detected. if synchronized only error free blocks increment the good_blocks_counter. all blocks except error free blocks increment the bad_blocks_counter. ? mode syncb: (sym0=1, sym1=0) error correction of burst error max. 2 bits; blocks corrected are treated as valid blocks, all other errors detected are treated as invalid blocks. if synchronized error free and correctable max. 2 bit errors increment the good_blocks_counter. ? mode syncc: (sym0=0, sym1=1) error correction of burst error max. 5 bits; blocks corrected are treated as valid blocks, all other errors detected are treated as invalid blocks. if synchronized error free and correctable max. 5 bit errors increment the good_blocks_counter. ? mode syncd: (sym0=1, sym1=1) no error correction; blocks detected as correctable are treated as invalid blocks if in synchronization search mode. internal bad_block_counter is always incremented even if correctable errors detected. if synchronized error free blocks and correctable max. 5 bit errors increment the good_blocks_counter. only uncorrectable blocks increment the bad_blocks_counter. 8.7.14 rbds processing mode: the decoder is suitable for receivers intended for the european (rds) as well as for the usa (rbds) standard. if rbds mode is selected (rbds=1) via the iic-bus, the block detection and the error detection and correction are adjusted to rbds data processing. that is, also e blocks are treated as valid blocks. if rbds is reset to zero (0), rds mode is selected. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 65 8.7.15 data available control modes: the decoder provides three different rds/rbds data output processing modes plus one decoder bypass mode selectable via the 'data available' control mode inputs dac0 and dac1. table 8.4 : dav modes mode dava: (dac0=0, dac1=0) standard output mode: if the decoder is synchronized and a new block is received (every 26 bits), the actual rds/rbds information of the last two blocks is available with every new received block (approx. every 21.9ms). mode davb: (dac0=1, dac1=0) fast pi search mode: during synchronization search and if a new a or c? block is received, the actual rds/rbds information of this or the last two a or c? blocks respectively is available with every new received a or c? block. if the decoder is synchronized, the ?standard output mode? is active. mode davc: (dac0=0, dac1=1) reduced data request output mode: if the decoder is synchronized and two new blocks are received (every 52 bits), the actual rds/rbds information of the last two blocks is available with every two new received blocks (approx. every 43.8ms). mode davd: (dac0=1, dac1=1) decoder bypassed mode: if this mode is selected then the outmux output of the decoder is reset to low (outmux=0). then the internal row buffer output is active and the decoder is bypassed. the decoder provides: data output of the block-identification of the last and previously processed blocks, the rds/rbds information words and error detection / correction status of the last two blocks as well as general decoder status information. in addition the decoder output is controlled indirectly by the data request from the external main controller. the decoder receives a 'data overflow' (dofl) signal controlled by the iic-bus register- interface. this dofl is set to high (dofl=1) if the decoder is synchronized and a new rds/rbds block is received before the previously processed block was completely transmitted via iic-bus. after detection of data overflow the interface-registers are not updated until reset of the data overflow flag (dofl=0) by reading via the iic-bus or if nwsy=1 which results in start of new synchronization search (sync=0). 8.7.16 data output of rds/rbds information the decoded rds/rbds block information and the current decoder status is available via the iic-bus. for synchronization of data request between main controller and decoder the additional data available output (davn) is used. if the decoder has processed new information for the main controller the data available signal (davn) is activated (low) under the following conditions: during synchronization search in davb mode if a valid a or c' block has been detected. this mode can be used for fast search tuning (detection and comparison of the pi code contained in the a and c' blocks). during synchronization search in any dav mode except davd mode, if two blocks in the correct sequence have been detected (synchronization criterion fulfilled). if the decoder is synchronized and in mode dava and davb a new block has been processed. this mode is the if the decoder is synchronized. if the decoder is synchronized and in davc mode two new blocks have been processed. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 66 if the decoder is synchronized and in any dav mode except davd mode loss of synchronization is detected (flywheel loss of synchronization, resulting in restart of synchronization search). in any dav mode except davd mode, if a reset caused by power-on or voltage-drop is detected. remark: if the decoder is synchronized, the davn signal is always activated after 21.9ms in dava or davb mode and after 43.8 ms in davc mode independent of valid or unvalid blocks are received. the processed rds/rbds data are available for iic-bus request for at least 20 ms after the davn signal was activated. the davn signal is always automatically de-activated (high) after ~10 ms or almost immediately after the main controller has read the rds/rbds status byte via iic-bus (see davn timing). the decoder ignores new processed rds/rbds blocks if the davn signal is active or if data overflow occurs (dofl=1). the following tables show the block identification number and processed error status outputs of the decoder and how to interpret the output data. rds block identification number blnr<2> blnr<1> blnr<0> block identification 0 0 0 block a 0 0 1 block b 0 1 0 block c 0 1 1 block d 1 0 0 block c? 1 0 1 block e (rbds mode) 1 1 0 invalid block e (rds mode) 1 1 1 invalid block rds processed error correction exb1 exb0 description 0 0 no errors detected 0 1 burst error of max. 2 bits corrected 1 0 burst error of max. 5 bits corrected 1 1 uncorrectable block 8.8 interface with tuner tea6840 (nice) the tuner ic tea6840 allows for a fast rds update sequence of about 7ms. the ic has an rds update timing sequencer on board which performs the following tasks: ? mute of the fm-mpx signal with a slope of 1 ms for fade out and fade in of the mpx signal ? tuning to the alternative frequency and back to the main frequency ? generating of two timing signals afhold and afsample to control the cdsp figure 8.33 shows the interface diagram between tuner and cdsp and figure 8.34 the timing diagram. the tea6840 delivers two mpx signals to the cdsp, one with mute, the fmmpx and one without mute, the rdsmpx. the rdsmpx signal enables the possibility to take also a noise sample x:noisflt_u from the alternative frequency. this is realised by switching the input of the a/d converter from fmmpx to rdsmpx during the rds update with sel_fr. an internal mute in the cdsp is initiated with afsample to suppress the modulation from the rdsmpx signal. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 67 figure 8.33 : interface diagram between tuner tea6840 and cdsp figure 8.34 :timing diagram rds update with tea6840 remark : due to the fast update the allowed settling time for the sensors is very short, about 2ms. in this time the sampled sensor values are not fully stabilised. therefor the values taken during an update for x:leva_u and x:noisflt can differ slightly (+/- 3db) from the real values. the multipath value x:mltflim_u is not reliable after a jump from a frequency with high fieldstrength to a frequency with low fieldstrength. tea6840 nice saa7709h a/d am/fm level fmmpx with mute rdsmpx no mute afsample afhold iic bus sel_fr dsp_in2 dsp_in1 pause dsp_out2 micro controller internal mute cdsp afhold fmmpx rdsmpx afsample 7ms 2 ms 2 ms freeze sensor signals www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 68 8.9 i2c interface scl (pin 57), sda (pin 58) these two pins needs a rc filter in the input/output lines for emc reasons, as is indicated in the application diagram. the components are not critical so a tolerance of 20% is tolerable. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 69 8.10 second processor extension function for communication with external processors, delay lines or other i2s controllable devices a complete dual channel 18 bit output bus is implemented. the cdsp is acting as the master transmitter and the external device has to be synchronised with the word select line. as input for the processed data two data input lines have been implemented which are processed synchronously with the data output to the external processor. this enables in total a feedback of two stereo audio channels. to the external processor, the dsp program should move data to the two ext iis data output registers, and read it back from the two or four ext iis data 1/2 input registers. the hardware of the bus can be enabled/disabled with bit 11 (en_host_io) of the selector registers (address $0ff9), by default the i2s outputs are disabled. to minimise emc, the output has to be disabled (= default) in case the output is not used. i2s input/outputs: iis_in1 (pin 31), iis_in2 (pin 32), iis_out1 (pin34), iis_out2 (pin 35), iis_clk (pin 30), iis_ws (pin33) 8.11 digital subwoofer and center output the cdsp offers an additional dual channel 18 bit digital output for the use of a subwoofer and center output. a choose can be made between ext. dac outputs, iis-out1 and iis-out2 for the subwoofer and center output. similar as the second processor extension function, the digital subwoofer output is capable of generating multiple output formats (i2s and lsb justified data formats). it is however not possible to select different data formats for the second processor and the subwoofer output. as also mentioned for the second processor outputs, the hardware of the bus can be enabled/disabled with the en_host_io bit in register $0ff9. to minimise emc, the output has to be disabled (= default) in case the output is not used. i2s outputs: iis_out1 (pin 34) or iis_out2 (pin 35), iis_clk (pin 30) and iis_ws (pin33) 8.12 external dac output (subwoofer) the saa7709h consists over i2s outputs that could be connected to an external dac with a own clock (fs_sys). this external dac could be applied to convert the digital subwoofer/center signal to an audible signal. the external dac output can be enabled/disabled with bit 15 (en_dac_out) of the selector registers (address $0ff9), by default the external dac output is disabled. to minimise emc, the output has to be disabled (= default) in case the output is not used. the uda1320 or uda1330 (filter stream dac) can be applied to convert the digital subwoofer and center signal, this dac type is compatible with the 3.3v output levels of the saa7709h. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 70 8.13 x-tal oscillator circuit osc_in and osc_out (pin 63 and pin 64) the on chip crystal oscillator is a pierce oscillator and is described in the data sheet. the crystal is running in fundamental mode on 11.2896 mhz. although a multiple of the crystal frequency falls within the fm reception band, this will not influence the reception because the crystal is driven in a controlled way. the crystal oscillator circuit can operate both in master mode and in slave mode. the blockdiagram of the x-tal oscillator circuit in master mode is depicted in figure 8.25. the active element gm compensates for the loss resistance of the crystal. the agc circuit controls the gain of the oscillator and prevents clipping of the generated sine-wave and therefor minimises the higher harmonics. the blockdiagram of the x-tal oscillator circuit in slave mode is depicted in figure 8.26. in order to minimise feedback due to ground bounce the power supply connections of the crystal oscillator circuit are separated from the other power supply lines. fig. 8.25 block diagram oscillator in master mode fig. 8.26 block diagram oscillator in slave mode rbias 100 k clkout gm agc + - xtal cx1 cx2 osc_in vss_osc vdd_osc osc_out cdsp rbias 100 k clkout gm agc + - 1 nf osc_in vss_osc vdd_osc osc_out cdsp 47 nf www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 71 x-tal calculations it can be shown that in order to start-up the transconductance of the active element must have a certain value g m,a where r is the loss resistance of the crystal and c l is the load capacitance: c x1 and c x2 are the capacitors connected to either side of the crystal and c p is the parasitic shunt capacitance of the crystal. in the cdsp the minimum transconductance is 4 ma/v. in order to ensure start-up, the following inequality must hold: filling in the oscillation frequency of the cdsp (f 0 = 11.2896mhz) and g m,min one obtains: for example, if c x1 = c x2 = 18pf and c p = 5pf so that c l = 14pf, the loss resistance r of the crystal must be smaller than 1000 ? . however it is wise to take a safety margin of 30% because the above equations are approximations. in this example that would mean that the maximum loss resistance of the crystal (which is specified by the manufacturer) should not exceed 80 ? . the internal bias resistor r bias is chosen high enough (100 k ? ) in order to prevent start-up problems. a safe value for r bias is where 0 is the oscillation frequency, r s is the resonator series resistance and c p its parallel capacitance. this means : r bias >> 8 k ? ; the value of 100 k ? is sufficiently large. grc ma l , = 4 22 c cc c c c l x xx p x = ? + + 1 2 12 22 4 ?? < rc g l m ,min r cl 219 199 10 <. ? ? r rc bias sp >> 1 0 22 ? ? www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 72 8.13 emc saa7709h application in order to optimise the emc behaviour of the saa7709h device, some measures are taken in the saa7709h design : 1. on-chip decoupling capacitor ; this reduces the high frequency components of the supply currents. 2. distributed clock ; the switching moments of the digital circuitry is spread in time. 3. adjustable pll frequency for the dsp ; this enables to change core frequency under microprocessor control in case fm reception is interfered by the saa7709h emission. 4. edge controlled digital outputs (controlled rise- and fall times) ; this reduces the high frequency components that could interfere with radio reception. to further optimise the emc behaviour of the saa7709h in a cdsp application, some additional measures are required, see also the saa7709h application diagram. 1. supply filtering digital circuitry dsp core the emc most critical supply pins are vddd1 and vddd2. via these pins the core is supplied, so the bulk of the digital current is flowing through these pins. to suppress interferences a choke blm21a10 should be added in the supply line. 2. supply filtering analog to digital convertors the analog to digital convertors are supplied via the vddaad pin. in order to decrease the groundbounce in the chip and to increase the analog performance will increase) a choke blm21a10 must be added in the supply line (vddaad). it?s important that no decoupling capacitor is connected to the vddaad pin, see appendix 1. 3. supply filtering peripheral supply the peripheral i/o circuitry is supplied via vddq1, vddq2 and vddq3. in order to suppress interferences a 10 ohm series resistor and a decoupling capacitor of 22 nf are added to these pins (see appendix 1). 4. main ground plane the pinning of the cdsp chip has been chosen in such a way that the lay-out is possible with a double sided pcb. it is recommended to create a ground plane on the non-smd side of the pcb. this main ground plane provides a low inductance ground return for the power supply and signal currents. this plane act as an equipotential point for the digital as well as for the analogue parts of the cdsp circuitry. the emc critical peripheral components should be above the plane. 5. ground plane under the cdsp chip it is recommended to provide a ground plane under the cdsp chip at the smd side of the pcb. the six ground pins (vsssx) of the digital related signals and the oscillator ground pin (vss_osc) have to be connected directly to this plane in order to reduce the loop area of the digital supply, this reduces the emc emission and ground bounce. this small ground plane has to be connected to the main ground plane with sufficient vias. do not use the small ground plane under the chip for the other ground pins, these have to be directly connected to the main ground plane. 6. oscillator circuit mount the oscillator peripheral components (xtal, cx1 and cx2, see figure 8.25) as close as possible to the cdsp chip. the oscillator supply is separately filtered with components a capacitor of 100nf and a choke (blm21a10), see appendix 1. www.datasheet.co.kr datasheet pdf - http://www..net/
philips semiconductors usermanual saa7709h/n1b 73 7. filtering dac outputs - on all four analogue outputs the same filtering are used. this is described in figure 8.1. first order rc-filtering of the analogue output signals will be done with a 100 ohm resistor and a 10nf capacitor. furthermore it is important to separate the supply of the digital circuitry from the supply of the analogue circuitry. in case an analogue +5v supply is used for generating the +3.3v for vdda2 (supply of fsdac) it is recommended to add a 100 h coil in series with the +5v analogue supply line. 8.14 changing the clock frequency of the dsp by default the dsp in the saa7709h are running on a frequency of: dsp clock frequency = 69.854 mhz the emi behaviour of the saa7709h is very good and in normal cases there is no interference with fm reception, however if desired it is possible to slightly increase the clock frequency of the dsp by changing the divide factor of the pll that generates the dsp clock signal. a decrease of the dsp clock frequency is possible but not allowed because this decreases the number of dsp program cycles below the number of required program cycles. an increase of one of the dsp clock frequencies is of course only applicable if the second harmonic of the dsp clock interferes with the frequency of the selected fm station. 8.14.1 procedure for increasing the clock frequency of the dsp the dsp clock signal is generated by pll1, the default divide factor of pll1 is 198, this result in a clock frequency of 69.854 mhz. the divide factor is set via bits 1 .. 5 of the iic_dsp_cntr register (address $602f). www.datasheet.co.kr datasheet pdf - http://www..net/
- 74 - appendix 1 : application diagram vddaad 75 vdacn 100 ? + 100f vddq1 vddq3 vddq2 22nf 220pf 100k ? 27k ? 220nf nav_gnd 6 3.9k ? 10k ? 3.9k ? 1nf 3 phone phone_gnd 4 8.2k ? 10k ? 10k ? 8.2k ? 72 71 cd_l cd_gndl vrefad 77 + 22f 47nf 100k ? 100k ? 100k ? 100k ? 82k ? 100pf 220nf 82k ? 100pf 220nf 56k ? 100pf 220nf 56k ? 100pf 220nf 67 66 69 68 1 am-l am-r tape-l fm-mpx tape-r fm-rds 80 sel-fr 61 1f rtcb shtcb tscan 43 44 45 rds-data rds-clock 60 59 100nf 65 62 vss-osc vdd-osc x1 18pf 18pf 63 64 osc_in osc_out vsss1 vsss2 vsss3 vsss4 74 76 22 36 46 49 53 52 50 vsss5 vsss6 54 55 48 51 vddd1 vddd2 38 39 10 ? dsp-io1 dsp-io2 40 41 dsp-io3 dsp-io4 26 21 dsp-io5 dsp-io6 vdda vssa 11 10 100 ? 10nf 10k ? + 1f 22f vrefda 12 + rrv rlv frv flv 15 13 9 8 34 iis-out1 35 iis-out2 30 iis-clk 33 iis-ws 31 iis-in1 32 iis-in2 dsp-reset 1f 10k ? 10k ? a0 sda scl 58 57 56 42 28 27 29 26 cd-data cd-ws cd-cl sysfs spdif1 spdif2 25 24 scl sda fl fr rl rr micro-control 100nf 100pf spdif-1 + 3.3v ana level phone_pos cdl_neg am-l am-r tape-l tape-r + 3.3v ana + 3.3v dig + 3.3v dig dsp-flags 910 ? 4.7k ? 100nf 22f + + 5v ana t1 75 ? quad fsdac analog source selector ad1 left ad2 right ad3 left level adc rds demodulator xtal osc i 2 s spdif i 2 c saa7709h + 5v dig + 5v dig 100 ? 10nf ? + 1f 100 ? 10nf ? + 1f 100 ? 10nf ? + 1f 10k 10k 10k 11.2896 mhz 220nf 220nf 220nf 220nf digital i/o 16 14 dsp-io7 dsp-io8 vssq3 vssq2 vssq1 vdacp nav 82k ? 100pf 1.8k ? 330pf fm aux-l aux-r 79 78 cd_r cd_gndr ad4 right iac digital source selector 22f micro-control ext.dac 17 20 19 18 level 2 70 71 10k ? 10k ? 8.2k 8.2k ? ? 220nf 220nf 100k ? 1nf 1nf 1nf 1nf blm21a10 blm21a10 330pf 220 220 ?? micro-controller 100pf 100pf ++ ? 1m phone_neg blm21a10 23 36 47 220nf nav 5 cdr_pos isn qmf stereo decoder dsp pom 7 ws_dac data_dac clk_dac fs_sys differential input differential input 100pf cdr_neg cdl_pos diff. input www.datasheet.co.kr datasheet pdf - http://www..net/

▲Up To Search▲

Price & Availability of SAA7709HN1B

	To Download SAA7709HN1B Datasheet File
If you can't view the Datasheet, Please click here to try to view without PDF Reader .