View SAA2502H_1744188.PDF datasheet online --- IC-ON-LINE

Datasheet File OCR Text:

d a t a sh eet preliminary speci?cation supersedes data of 1997 apr 18 file under integrated circuits, ic01 1997 nov 17 integrated circuits saa2502 iso/mpeg audio source decoder
1997 nov 17 2 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 contents 1 features 2 applications 3 general description 4 ordering information 5 block diagram 6 pinning 7 functional description 7.1 basic functionality 7.2 clock generator module 7.2.1 external sample clock 7.2.2 free running internal sample clock 7.2.3 locked internal sample clock 7.2.4 limited sampling frequency support for internal sampling clocks 7.3 input interface module 7.3.1 master input mode 7.3.2 slave input mode 7.3.3 buffer controlled input mode 7.4 decoder core 7.4.1 frame synchronization to input data streams 7.4.2 master input mode bit rate generation 7.4.3 sample clock generation 7.4.4 decoder precision 7.4.5 scale factor crc protection 7.4.6 handling of errors in the coded input data 7.4.7 dynamic range compression 7.4.8 baseband audio processing 7.4.9 decoder latency time 7.5 output interface module 7.5.1 i 2 s output 7.5.2 spidf output 7.5.3 bit serial analog output 7.6 control interface module 7.6.1 resetting 7.6.2 interrupts 7.6.3 microcontroller interface 7.6.4 initialization 7.6.5 transfer protocols 7.6.6 local registers 8 appendix 8.1 l3 interface specification 8.1.1 introduction 8.1.2 example of a data transfer 8.1.3 timing requirements 8.1.4 timing 9 limiting values 10 dc characteristics 11 ac characteristics 11.1 host interface: cdata, cclk and cmode 12 application information 13 package outline 14 soldering 14.1 introduction 14.2 reflow soldering 14.3 wave soldering 14.4 repairing soldered joints 15 definitions 16 life support applications
1997 nov 17 3 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 1 features low sampling frequency decoding possibilities (24 khz, 22.05 khz and 16 khz) of mpeg2 are supported a variety of output formats are supported: i 2 s, spdif and 256 or more times oversampled bit serial analog stereo automatic internal dynamic range compression algorithm using programmable compression parameters non byte-aligned coded input data is handled built-in provisions to generate high quality sampling clocks for all six supported sampling frequencies; these sampling clocks may locked to an external pll to support an extensive list of input data reference clock frequencies bit-rate and sampling-rate settings may be overruled by the microcontroller while the saa2502 is trying to establish frame synchronization input interface mode which requests data based on input buffer content, enables the handling of variable bit-rate input streams and input data offered in (fixed length) bursts an interrupt output pin which can generate interrupt requests at the occurrence of various events; consequently polling by the microcontroller is not needed in most situations l3 and the i 2 c-bus microcontroller interface protocols are supported the control interface is always fully operational (also while stop is asserted) crc protection of scale factors is provided for all supported sample frequencies. 2 applications astra digital radio (adr) digital audio broadcast (dab) digital versatile disc (dvd) digital video broadcast (dvb) general purpose mpeg2 audio decoding. 3 general description the saa2502 is a second generation iso/mpeg audio source decoder. the device specification has been enhanced with respect to the saa2500 and saa2501 ics and therefore it offers in principle all features of its predecessors. it supports layer i and ii of mpeg1 and the mpeg2 requirements for a stereo decoder. 4 ordering information type number package name description version SAA2502H qfp44 plastic quad flat package; 44 leads (lead length 1.3 mm); body 10 10 1.75 mm sot307-2
1997 nov 17 4 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 5 block diagram fig.1 block diagram. handbook, full pagewidth mge469 input interface demulti- plexer dequanti- zation and scaling synthesis sub-band filter digital-to- analog converter spdif encoder clock generator phase compa- rator divider decoding control 1 25 40 43 15 17 19 14 20 13 16 28 42 37 36 35 34 38 39 6 4 2 3 5 21 24 41 33 11 12 10 9 7 8 22 23 26 27 29 32 31 18 30 44 saa2502 mclkout mclkin mclk24 x22out x22in phdif vdd1 vdd2 vdd3 refclk cdata cclk cmode int stop reset tdi tdo tck tms trst sd sck ws spdif lftpos lftneg rgtpos rgtneg refn refp gnd1 gnd2 gnd3 cdrq cdval cdcl cdsy cdef cd tc1 tc0 fsclkin fsclk
1997 nov 17 5 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 6 pinning symbol pin description fsclk 1 sample rate clock output; buffered signal sck 2 baseband audio data i 2 s clock output sd 3 baseband audio i 2 s data output ws 4 baseband audio data i 2 s word select output trst 5 boundary scan test reset input spdif 6 spdif baseband audio output cclk 7 l3 clock/i 2 c-bus bit clock input cdata 8 l3 data/i 2 c-bus serial data input/output; note 1 cmode 9 l3 mode (address/data select input) int 10 interrupt request output; active low; note 1 reset 11 master reset input stop 12 soft reset/stop decoding input cdrq 13 coded data request output cdcl 14 coded data bit clock input/output; note 2 cd 15 mpeg coded data input gnd1 16 ground 1 cdef 17 coded data error ?ag input v dd1 18 supply voltage 1 cdsy 19 coded data byte or frame sync input cdval 20 coded data valid ?ag input tms 21 boundary scan test mode select input refclk 22 pll reference clock input phdif 23 pll phase comparator output; note 2 tck 24 boundary scan test clock input fsclkin 25 sample rate clock input x22in 26 22.579 mhz clock oscillator input or signal input x22out 27 22.579 mhz clock oscillator output gnd2 28 ground 2 mclk24 29 master clock frequency indication input v dd2 30 supply voltage 2 mclkout 31 master clock oscillator output mclkin 32 master clock oscillator input or signal input tdi 33 boundary scan test data input rgtpos 34 analog right channel positive output rgtneg 35 analog right channel negative output refn 36 low reference voltage input for analog outputs refp 37 high reference voltage input for analog outputs lftneg 38 analog left channel negative output lftpos 39 analog left channel positive output tc0 40 factory test scan chain control 0 input
1997 nov 17 6 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 notes 1. output type is: open-drain. 2. output type is: 3-state. tdo 41 boundary scan test data output gnd3 42 ground 3 tc1 43 factory test scan chain control 1 input v dd3 44 supply voltage 3 symbol pin description fig.2 pin configuration. 1 2 3 4 5 6 7 8 9 10 11 33 32 31 30 29 28 27 26 25 24 23 12 13 14 15 16 17 18 19 20 21 22 44 43 42 41 40 39 38 37 36 35 34 saa2502 mge468 tdi mclkin mclkout v dd2 gnd2 x22out x22in fsclkin tck phdif fsclk sck sd ws trst spdif cdata cmode reset mclk24 tc1 gnd3 tdo tc0 lftpos lftneg refn rgtneg rgtpos v dd3 refp cdrq cdcl cd gnd1 cdef v dd1 cdval tms refclk stop cdsy cclk int
1997 nov 17 7 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7 functional description 7.1 basic functionality from a functional point of view, several blocks can be distinguished in the saa2502. a clock generator section derives the internally and externally required clock signals from its clock inputs. the input interface section receives or requests coded input data in one of the supported input interface modes. the demultiplexer processor handles frame synchronization, parsing, demultiplexing and error concealment of the input data stream the de-quantization and scaling processor performs the transformation and scaling operations on the (demultiplexed) coded sample representations in the input bitstream to yield sub-band domain samples. the sub-band samples are transferred to the synthesis sub-band filter bank processor which reconstructs the baseband audio samples. the output interface block transforms the audio samples to the output formats required by the different output ports. the decoding control block houses the i 2 c-bus/l3 microcontroller interface, and handles the response to external control signals. this section enables the application to configure the saa2502, to read its decoding status, to read ancillary data and so on. several pins are reserved for boundary scan test (5 pins) and factory test scan chain control (2 pins). 7.2 clock generator module the saa2502 clock interfacing is designed for application versatility. it consists of 9 signals (see table 1). the clock generator provides the following clock signals: internal sample clocks external buffered sample clock fsclk processor master clock coded input data bit clock coded input data request clock the module can be configured to operate in 3 different modes of operation: external sample clock mode free running internal sample clock mode locked internal sample clock mode. clock generator operation mode must be stationary while the device is in normal operation. changing mode should always be followed by a (soft) reset. f input bit rate 32 ---------------------------------- - = table 1 clock interfacing signals signal direction function mclkin input master clock oscillator input or signal input mclkout output master clock oscillator output mclk24 input master clock frequency indication x22in input 22.5792 mhz clock oscillator input or signal input x22out output 22.5792 mhz clock oscillator output fsclkin input external sample rate clock signal input fsclk output sample rate clock signal output refclk input coded input data rate reference clock phdif output phase difference indication output between reference clock and sample clock
1997 nov 17 8 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.2.1 e xternal sample clock in applications where a 256 f s sample clock is available, the use of external crystals may be avoided by putting the saa2502 clock generator module in external sample clock mode. such mode setting may be realized by setting control flag fscinp of the control interface. in this event the sample clock has to be provided to the fsclkin clock input. if sample rate switching should be supported, required clock frequency changes are the responsibility of the application. after such a clock frequency change, enforcement of a soft reset is advised. in external sample clock mode (and only in that mode) the clock generator module is able to accept a 384 f s sample clock input. if that mode of operation is desired the control flag fsc384 should be set. the fsclk output is normally disabled in this mode. if enabled (by setting control flag fscena) fsclk will produce a buffered copy of fsclkin. x22in, x22out, refclk and phdif are not used in this mode. x22in and refclk should be connected to gnd or v dd . mclkin is used to provide the (free running) master clock. this may either be achieved by applying a correct clock signal to mclkin or by connecting a crystal between mclkin and mclkout. in external sample clock mode (and only in that mode) the master clock may deviate from 24.576 mhz. the master clock frequency value required depends on the state of pin mclk24 (see table 2). table 2 master clock frequency setting by mclk24 7.2.2 f ree running internal sample clock this is the default mode of operation: 256 f s for all six supported sample rates is generated internally from the clock frequencies supplied to mclkin (24.576 mhz) and x22in (22.5792 mhz) as shown in table 3. mclk24 frequency minimum maximum gnd 256 f s 12.288 mhz (256 48 khz) v dd 512 f s 24.576 mhz (512 48 khz) table 3 internal sample clock (default mode) note 1. asymmetrical fsclk. the main advantage of this mode is that the saa2502 determines automatically which sampling rate is active from the sampling rate setting of the input data bit stream, and then selects either mclkin or x22in divided by the correct number as the sample clock source. therefore this mode is particularly suited in applications supporting dynamically varying sampling rates. the required clocks may either be applied to mclkin (respectively to x22in) or be generated by connecting a crystal between mclkin and mclkout (respectively between x22in and x22out). the recommended crystal oscillator configuration is shown in fig.3. the specified component values only apply to crystals with a low equivalent series resistance of <40 w . fsclkin, refclk and phdif are not used in this mode (fsclkin and refclk should be connected to v ss or v dd ). mclk24 has to be connected to v dd , while the control flags fscinp and fsc384 should be left in their default (cleared) states. if the fsclk output is enabled (by setting control flag fscena) fsclk will produce a buffered version of 256 f s . sample frequency resultant frequencies (mhz) 256 48 khz 12.288 256 44.1 khz 11.2896 256 32 khz 8.192 (1) 256 24 khz 6.144 256 22.05 khz 5.6448 256 16 khz 4.096 24.576 2 ----------------- - 22.5792 2 --------------------- 24.576 3 ----------------- - 24.576 4 ----------------- - 22.5792 4 --------------------- 24.576 6 ----------------- -
1997 nov 17 9 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.2.3 l ocked internal sample clock this mode differs from the previous one in just a single aspect: the refclk and phdif pins are used to realize a phase-locked loop (pll) which locks the 256 f s sample clock to the refclk reference clock. because the real goal is locking sample clock and bit rate, a reference clock should be used which has a fixed relation to the input bit rate. an example of such a pll realization is shown in fig.4. the phase comparator output phdif generates a signal with a dc component proportional to the phase difference between the internal signals sig and ref (see fig.5). the 22.5792 mhz signal x22in is divided by 147 and the 24.576 mhz signal mclkin is divided by 160. this results in the same frequency (153.6 khz) in both events. one of the two signals is selected as input for the programmable divide by n 1 unit. the selector is controlled handbook, halfpage mge470 c2 c1 c3 c4 x1 x2 r1 r4 r2 r3 26 27 32 31 saa2502 fig.3 crystal oscillator components. c1 = c2 = c3 = c4 = 10 pf; r1 = r4 = 100 k w ; r2=r3=1k w ; x1 = 22.5792 mhz; x2 = 24.5760 mhz. in such a way that sig and 256 f s will stem from the same source. the divisor n 1 is programmable with (1 to 16) 8 as possible values. ref on the other hand is derived from the refclk input. two programmable dividers in series are used here. n 2 may adopt one of 4 possible values: 5, 25, 125 or 625 while n 3 can be programmed to be 1 to 32. because both inputs of the phase comparator have to operate at identical frequencies the next equation has to be obeyed: or, written differently: for a list of supported refclk frequency values see chapter 8. the mode of operation of the phase comparator in fig.5 is programmable via the control flag phsmod: refclk n 2 n 3 ------------------------- - 156.6 khz n 1 --------------------------- = refclk 153.6 khz n 2 n 3 n 1 ----------------------------------------------------- - = fig.4 external pll components. handbook, halfpage mge471 low- pass filter 24.576 mhz vcxo 22.5792 mhz vcxo phdif mclkin mclkout x22in x22out saa2502 fig.5 saa2502 phase comparator. handbook, full pagewidth mge472 divide by 147 divide by 160 divide by n 2 divide by n 3 divide by n 1 phase compa- rator x22in mclkin refclk 153.6 khz sig ref phdif
1997 nov 17 10 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.2.3.1 xor mode phdif is the xor function of sig and ref. the frequency is twice the frequency of sig and ref. the phdif output carries a signal, switching between gnd and v dd , with an average value v avg which is a function of the phase difference between sig and ref (see left part of fig.6). the locking range in this mode of operation is maximum for even values of n 3 (180 degrees phase difference) but less for odd values of n 3 . it is minimum for n 3 =3 (120 degrees phase difference). 7.2.3.2 edge triggered mode phdif is only influenced by the rising edges of sig and ref. consequently its frequency is equal to the sig and ref frequency. the electrical behaviour of the phdif output pin in this mode is special: phdif is high from the rising edge of ref to the rising edge of sig and 3-stated elsewhere if ref is leading and phdif is low from rising edge of sig to rising edge of ref and 3-stated elsewhere if ref is trailing. therefore phdif is not 3-stated during a portion t up of each cycle when it acts as a pull-up device or during a portion t down of each cycle when it acts as a pull-down device (see right part of fig.6). as a result the locking range is always 360 degrees phase difference. the output behaviour as function of phase difference is non-symmetrical with reference to the vertical axis, but a reversed mode is also available (by setting the control flag phsrvs). fig.6 phdif output behaviour. handbook, full pagewidth mge473 ref to sig phase difference ref to sig phase difference max min 0 o 180 o 360 o - 180 o 0 o + 180 o 0 1 5/6 1/6 v avg v dd 100% 100% 0% t up t down t t 1 t 2 t t phdif 3-stated xor mode edge triggered mode
1997 nov 17 11 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.2.4 l imited sampling frequency support for internal sampling clocks 7.2.4.1 when sampling frequency is limited to 44.1 and/or 22.05 khz: in this event mclkin is only required to generate the master clock frequency. consequently the remarks on mclkin frequency also apply in this special case. 7.2.4.2 when sampling frequency is limited to 48, 32, 24 and/or 16 khz: in this event x22in is not required. therefore x22in should be connected to v ss or v dd , but it is more efficient to apply any available clock signal to x22in. because 44.1 khz is the default initial sampling frequency it may also be advisable to over-rule the sampling frequency after a hard reset. 7.3 input interface module the input interface module handles the reception of the coded input data stream. the module can be configured to operate in 3 distinct modes of operation: the master input mode the slave input mode the buffer controlled input mode. input interface mode must be stationary while the device is in normal operation. changing mode will result in an (automatically generated) internal soft reset. the inputs cd, cdval, cdef and cdsy are all clocked at the rising edge of the cdcl bit clock. cdrq changes at the falling edge of cdcl. cdval = logic 0 indicates that cd and cdef should be ignored while cdval = logic 1 indicates that cd is a valid coded input stream data bit (cdef is then its error attribute). cdef = logic 0 means that the value of cd may be assumed to be reliable while cdef = logic 1 means that the value of cd is flagged as insecure (e.g. due to erratic non-correctable channel behaviour). the value of cdef may be different for each data bit, but is combined by the saa2502 for every group of 8 (byte aligned) valid coded input bits. cdsy will only have effect when the symod control flags are set to 10 or 11. when symod = 10 the valid input bit at a rising edge of cdsy marks the start of a new byte (when symod = 11 it marks the start of a new mpeg audio frame). note that just the rising edge of cdsy is important, the falling edge has no meaning. if cdsy is used with symod = 10 leading edges must be frequent enough to assure fast byte alignment, if used with symod = 11 a leading edge must be present every frame. leading edges of cdsy may occur while cdval is (implicitly) high. alternatively, a situation as shown in fig.8 is also allowed, where cdsy has a rising edge while cdval is low, i.e. during invalid data. the first valid cd bit after the rising edge of cdval is then interpreted as the first byte or frame bit. the output pin cdrq is used to request new coded input data. table 4 signals of coded data input interface signal direction function cd input coded data input bit cdval input coded data bit valid ?ag cdef input coded data bit error ?ag cdsy input coded data sync (start of byte/frame) indication cdcl input/output coded data bit clock cdrq output coded data request
1997 nov 17 12 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.3.1 m aster input mode master input mode is the default mode of operation. this mode may also be enforced by setting the inmod control flags to 00. which means that the saa2502 will generate requests for input data at regular intervals. cdval is not used in this mode (it should be connected to v ss or v dd ). cdval is implicitly assumed to be logic 1 during the 2nd up to (and including) the 17th bit slot after a rising or a falling edge of cdrq (see fig.7). thus signal cd should carry the coded data in bursts of 16 valid bits. in this mode the cdrq frequency is locked to (i.e. derived from) the 256 f s clock. its average value equals the bit rate divided by 32. the bit clock cdcl is output, its frequency is fixed: when mclk24 = logic 1 when mclk24 = logic 0. mclk 32 ----------------- - mclk 16 ----------------- - mpeg free format bit rate is not allowed in this mode. assume n is the number of cdcl periods between two transitions of cdrq, and r is the number of cdcl periods to obtain the effective bit rate e (in kbits/s) at a cdcl frequency of 768 khz, i.e. . the saa2502 keeps the average value of n exactly at r, but individual values of n may vary between n = round (r) - 2 and n = round (r) +2. 7.3.2 s lave input mode slave input mode is activated by setting the inmod control flags to 0 1. which means that the saa2502 will accept input data as presented by the application. in this mode it is the responsibility of the application to maintain locking between the 256 f s sample clock and the average bit rate. r 16 768 e ---------------------- = fig.7 master mode input data format. handbook, full pagewidth mge474 cdcl cd cdef cdrq cdsy unreliable data bit (example) start of byte or frame 1 1 2 2 14 valid data valid but unreliable data invalid data 15 16
1997 nov 17 13 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 the bit clock cdcl is input, its frequency is determined by the application, however certain minimum and maximum values have to be obeyed. mpeg free format bit rate is allowed in this mode. cdval = logic 1 indicates valid data. in this way, burst input data is supported. the speed at which data may be transferred to the input interface is restricted. transfer of an mpeg frame is illustrated in fig.9. it shows the transfer of all nf bits of one frame between time 0 and tf, where tf corresponds to 384 sample periods (mpeg layer i input data) or 1152 sample periods (mpeg layer ii input data). in the figure, an example of an actual transfer characteristic is drawn. input data may be transferred at a speed higher than bit rate (i.e. cdcl may have a frequency higher than bit rate). ideally the data transfer of the first frame is in a single burst. in practice multiple bursts are allowed, provided that the data transfer is always within 128 cdcl cycles of the ideal data transfer. subsequent frames may also have multiple bursts, but the data transfer must always be within 128 cdcl cycles of both the first frame data transfer and the ideal single burst transfer characteristics. all frames must start within the first four bytes of a data burst. the transfer characteristic has a slope equal to cdcl frequency during the bursts (when cdval is high) and is horizontal outside the bursts (when cdval is low; no bits are transferred). the frequency of cdcl has to be constant (except when cdval is low) in normal operation; any change of cdcl frequency should be followed by a (soft) reset. for dab applications there is an exception to the rule that data transfer is always within 128 cdcl cycles of the ideal single burst characteristic. when the sampling frequency is 24 khz and the cdcl frequency is 384 kbits/s, it is allowed to send an input frame in two bursts of equal length. the first bit of a frame must be the first bit of a burst, while the last bit of a frame must be the last bit of a burst. fig.8 slave mode input data format. handbook, full pagewidth mge475 cdcl cd cdef cdval cdsy unreliable data bits (example) start of byte or frame valid data valid but unreliable data invalid data
1997 nov 17 14 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 fig.9 slave input data transferring speed. (1) ideal frame transfer characteristics are restricted to this area. (2) ideal frame transfer characteristic (example). handbook, full pagewidth mge476 slope = cdcl frequency slope = input bit rate (2) jitter limits slope = maximum cdcl frequency nf transferred input bits 0 0 time tf (1) the shaded area in fig.9 represents the restrictions to the transfer characteristic of a frame. the characteristic may not cross the shown upper limit of the shaded area in order to prevent input buffer underflow and/or overflow. the slope of the upper limit is determined by the sample frequency as shown in table 5. table 5 slope of the upper limit determined by sampling frequency sample frequency (khz) maximum cdcl frequency (kbits/s) 48 768 44.1 705.6 32 512 24 384 22.05 352.8 16 256
1997 nov 17 15 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.3.3 b uffer controlled input mode (see fig.10) buffer controlled input mode is activated by setting the inmod control flags to 1x, which means that the saa2502 will request data based on the amount of input bytes currently residing in the input buffer. the bit clock cdcl is output, its frequency is fixed: when mclk24 = logic 1 when mclk24 = logic 0. in this mode cdrq = logic 1 is an indication that new input data is required. cdval = logic 1 indicates the delivery of valid data. the application should react to the event of an input data request as follows: one byte of input data should be delivered within 16 cdcl cycles. if cdrq remains high the next byte mclk 32 ----------------- - mclk 16 ----------------- - should be delivered and so on until cdrq is dropped. delivery of subsequent bytes while cdrq remains high should be uninterrupted (cdval should stay high) there is also an option for the application to deliver part of the input data later. despite violating the conditions in the previous paragraph, this is allowed, but with consequences for the input buffer latency time. mpeg free format bit rate is allowed in this mode. dynamically varying bit rate may be supported in this mode. whether such support is desired or not is indicated by the following input mode bits: inpmod = 10 means bit rate is assumed to be (quasi) static inpmod = 11 means bit rate is assumed to be dynamic. fig.10 buffer controlled mode input data format. handbook, full pagewidth mge477 cdcl cd cdef cdrq cdval cdsy unreliable data bit (example) start of byte or frame valid data valid but unreliable data invalid data
1997 nov 17 16 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.4 decoder core the saa2502 fully complies with mpeg1 (layer i and ii) and mpeg2 (layer i and ii, l 0 and r 0 channels). also some dab specific features are supported. free format bit rate is not supported in master input mode. several aspects of the decoding process and audio post-processing features are offered. 7.4.1 f rame synchronization to input data streams the saa2502 has to localize the start of a frame before decoding may begin. the process of locating the start of a frame is called frame synchronization. there are 4 different modes of frame synchronization available. these modes are in order of decreasing speed of frame synchronization. 7.4.1.1 frame sync pulse mode in this mode the start of each frame is marked by a rising edge of the cdsy input pin. it is the fastest and most reliable method of frame synchronization. it is activated by loading 11 into the symod control flags. 7.4.1.2 byte aligned mode this default mode may also be enforced by loading 10 into the symod control flags. the start of a frame is located by detection of the 14-bit sync pattern 111111111111x1. the probability of correct sync detection is enhanced by the fact that a rising edge of the cdsy input pin marks a location which is byte aligned with frame bounds. a rising edge of cdsy is not required at every byte edge but should occur at regular intervals for reliable frame synchronization. 7.4.1.3 layer ii non-byte aligned mode this mode may be entered by loading 01 into the symod control flags. frame start is found by detection of the 15-bit sync pattern 111111111111x10. as this pattern is slightly longer than the previous one and also contains at least one 1-to-0 transition, it may be used to obtain frame synchronization in the absence of any external alignment indication (cdsy is ignored and therefore may be left floating). 7.4.1.4 general non-byte aligned mode this mode may be entered by loading 00 into the symod control flags. frame start is detected by alternating searches for a 15-bit sync pattern 111111111111x10 (identical to the layer ii mode search pattern) and a15-bit sync pattern 0111111111111x1. because valid mpeg streams exist that do not contain the first pattern while other valid mpeg streams do not contain the second pattern a time-out counter will always be active in this mode. time-out length is set to slightly more then 72 ms which is the length of the longest audio frame. the second pattern operates for layer i and layer ii, but successful synchronization is only guaranteed when the last bit of the previous frame equals logic 0. consequently this mode synchronizes to layer i input bit streams only if frames at least sometimes end with a logic 0 bit. both patterns contain the 1-to-0 or 0-to-1 transition required for a reliable start-of-frame detection in the absence of external alignment information. if the saa2502 starts at a random place in the bit stream, it may take up to one frame before a sync pattern or sync pulse is encountered. because sync patterns may be emulated by frame content, detection of a sync is always followed by a verification period to check whether the sync is located at the start of a frame. the length of the verification period depends on the presence of crc protection and/or a free format bit rate index. during sync search and verification the baseband audio outputs are muted. if verification fails the synchronization process is restarted. table 6 frame sync veri?cation input data format length of verification period free format bit rate non-free format bit rate mpeg; no crc 2 frame bit rate 1 frame mpeg with crc 1 frame 0 frame
1997 nov 17 17 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.4.2 m aster input mode bit rate generation when master input mode is used, the saa2502 fetches input data at the effective bit rate. however after a hard reset the input requests input data at the default bit rate until synchronization has been established as shown in table 7. when the clock generator mode is free running internal sample clock or locked internal sample clock the default input bit rate is always 384 kbits/s. when the mode is external sample clock the saa2502 derives the selected bit rate from the signal fsclkin. but initially it has no indication of the current sampling rate corresponding to fsclkin. therefore the bit rate of 384 kbits/s is generated at an assumed sampling frequency of 44.1 khz. for different sample rates, the bit rate changes proportionally. the consequence is that while the saa2502 is synchronizing after a hard reset, the application should be able to supply input data at the given default bit rate until synchronization is established. alternatively there is also the possibility to overrule default bit rate setting and sample rate setting using the control interface while synchronization has not been established. the speed at which input data is requested by the input in master mode is changed in one of the following events: when input synchronization is established at the end of the verification phase and the bit rate index of the decoded bit stream indicates a bit rate different from the one currently selected. in this event, the bit rate is adapted to the new index. when the signal stop is raised while the stoprq control flag = logic 1, input requesting is halted. requesting resumes at the last selected input bit rate when the stop signal is dropped. in all other events (including when the saa2502 loses synchronization), the last selected input bit rate is maintained. whenever the selected bit rate changes while dynamic bit rate is not enabled, the saa2502 will generate internally a soft reset resulting in a soft mute of the output interfaces and a decoder restart in order to re-initialize internal buffer settings. table 7 establishment of default bit rate clock generator mode fsclkin (khz) default bit rate (kbits/s) free running internal clock dont care 384 locked internal clock dont care 384 external sample clock 256 or 384 48 417.96 256 or 384 44.1 384 256 or 384 32 278.64 256 or 384 24 208.98 256 or 384 22.05 192 256 or 384 16 139.32
1997 nov 17 18 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.4.3 s ample clock generation when the external sample clock mode of the clock generator is used, the application must know the sample rate. fsclkin has to be applied, with a frequency which is a multiple of the sample rate. the (sample rate dependent) output interface timing signals will be generated from fsclkin. this mode will normally be used in applications with a fixed sample rate. should the sample rate change, then a soft reset is strongly advised. when one of the remaining clock generator modes is used, the saa2502 selects the active sample rate automatically, and generates the required sample rate related timing signals from its mclkin and x22in clock inputs. soft resets at sample rate changes are generated automatically. after a hard reset, a sample rate of 44.1 khz by default is selected. such default setting may be overruled using the control interface. sck, ws and spdif will show frequency changes in any of the following 3 situations: when the saa2502 establishes synchronization to the coded data input bit stream at a sample rate different from the one previously selected when the current (default) sample rate is overruled by the control interface when the clock generator mode is changed, resulting in a switch from or to the external sample clock mode. in all those situation the phase of ws and the data content of spdif will be continuous. in all other events sck, ws and spdif remain operating without phase or frequency changes and the sample rate selection remains unchanged. 7.4.4 d ecoder precision during decoding several multiply operations are carried out on coded samples. the results of these operations have to be rounded in order to keep the word length required for internal number representation within reasonable limits. accumulation of these rounding errors is kept at a very low level in order to assure precise audio output samples. saa2502 precision is specified using the output of the mpeg reference decoder based on double precision floating point calculations as a reference. differences between that reference decoder and saa2502 output manifest themselves as white noise. two contributions to this noise may be identified: noise resulting from internal rounding on intermediate results noise resulting from rounding of final output samples to 16, 18, 20 or 22 bits (depending on selected output accuracy). table 8 shows the effective noise level figures. (unit is 1 lsb of 22-bit accuracy output). except for 22-bit accuracy, output rounding is by far the dominant effect. consequently the saa2502 may be considered a professional level high precision decoder. table 8 effective noise level ?gures note 1. the output rounding part of this precision is valid only for i 2 s and spdif outputs. output accuracy (bits) intermediate rounding output rounding (1) total noise level 22 0.6 0.3 0.7 20 0.6 1.2 1.3 18 0.6 4.6 4.7 16 0.6 18.5 18.5
1997 nov 17 19 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.4.5 s cale factor crc protection mpeg specifies an optional 16 bit crc that may be used to verify whether an important part of each audio frame is received correctly. the following data items is protected by this crc: bytes 3 and 4 of the first 4 bytes of each frame, containing most of the frame header information allocation information scale factor select information (layer ii only). the scale factors are not protected by this scheme. the dab specification includes crc protection for scale factors. the 32 sub-bands are divided into the following 4 blocks: block 0 = sub-bands 0 to 3 block 1 = sub-bands 4 to 7 block 2 = sub-bands 8 to 15 block 3 = sub-bands 16 to 31. each block is protected by an 8-bit crc if that block of sub-bands is (partly) inside the current sub-band limit. the required scale factor crcs are stored in the last bytes of the previous audio frame: the last two bytes of each frame are reserved for ancillary data; dab specification calls this fixed program associated data (fpad) minimum 2 and maximum 4 bytes before fpad are reserved for scale factor crcs. the number of crc bytes present is be derived from the sub-band limit of the following audio frame bytes before the crcs are available for more ancillary data; dab specification calls this extended program associated data (xpad), as far as not occupied by mpeg coded input data. the dab type of scale factor crc protection, extended to all valid sample frequency plus bit rate combinations of mpeg1 and mpeg2, and to layer i, is fully supported by the saa2502. (dab is restricted to mpeg1 layer ii, to 48 khz sample frequency and does not support free format bit rate). requirements for scale factor crc handling is indicated by the sfcrc control flag. 7.4.6 h andling of errors in the coded input data the saa2502 is able to handle certain types of errors in the input data. three error categories will be handled: errors flagged by the coded input data error flag cdef crc failures (if mpeg and/or scale factor error protection is active) mpeg audio frame syntax errors. error flags in the input data will effect the decoding process if the corrupted data is inside the header, bit allocation or scale factor select information part of a frame (then the saa2502 will soft mute that frame) or inside the scale factor field (then the most recent valid scale factor of the same sub-band will be copied). error flags in other data fields will be ignored. if mpeg and/or scale factor crcs are active the crc result has priority over cdef flags inside the protected fields. in applications where the mpeg crc is always present, the protection bit (which is not crc protected) in the mpeg header may be overruled by setting control flag crcact. thus the saa2502 is robust for data errors in the protection bit.
1997 nov 17 20 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.4.7 d ynamic range compression the baseband audio output resulting from mpeg decoding has a high dynamic range (theoretically >200 db, practically up to 120 db for the 22-bit output mode).this feature is very attractive from the high quality audio standpoint of view, but such high dynamic range is undesirable when there is a relatively high level of background noise (e.g. for car radio). for those applications the saa2502 offers the possibility of built in dynamic range compression: internal dynamic range compression is offered. thus any standard mpeg encoded bit stream may be compressed i.e. no added compression information is required. the dynamic range compression algorithm is fully parameterised. all major characteristics are programmable through the control interface: C level of compression C maximum compression C compression offset C compression release rate (compression attack rate has to be fixed). the dynamic range compression algorithm is based on a (in time varying) amplification factor, which is equally applied to all audio output samples. the value of the amplification factor is calculated on basis of the current audio output power level for each (sub)frame of 384 output samples. the applied power to amplification curve is shown in fig.11. all characteristics of the curve are programmable: compression slope minimum = 0, maximum = 0.996 maximum amplification minimum = 0 db, maximum = 23.81 db offset minimum = 0 db, maximum = 47.81 db. offset values close to 0 db may result in clipped output signals. this is especially true for signals with a high amplitude-to-power ratio (an extreme example of such a signal is a maximum amplitude unit impulse). the occurrence of this effect can be avoided by selecting an offset value close to or greater than 15 db. in the context of dynamic range compression definition, the 0 db power reference level is defined as a sine wave shaped output signal with maximum amplitude in just one (right or left) channel. the calculation will result in an new amplification factor every 384 samples (i.e. from 8 ms at 48 khz to 24 ms at 16 khz sample rate). subsequent amplification factors may vary considerably. an example showing two large step type discontinuation is shown in fig.12. it is undesirable to apply large increasing amplification steps immediately. consequently increasing the amplification factor is limited to the release rate which is also programmable: minimum release rate = (1.46 db/s at 48 khz; 0.488 db/s at 16 khz) maximum release rate = (46.87 db/s at 48 khz; 15.625 db/s at 16 khz). decreasing amplification factors, must be applied almost immediately to avoid overflow when the audio power increases rapidly; thus attack rate is non-programmable and fast. 0.0117 db 384 samples ---------------------------------- 0.375 db 384 samples ---------------------------------- fig.11 dynamic range compression characteristic. handbook, halfpage mge478 maximum amplification compression slope amplification (db) 0 db power (db) offset fig.12 amplification change rates. handbook, halfpage mge479 audio signal power amplifi- cation release rate attack rate time
1997 nov 17 21 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.4.8 b aseband audio processing baseband audio de-emphasis as indicated in the mpeg input data stream is performed digitally inside the saa2502. the included audio processing unit (apu) see fig.19, may be used to apply programmable inter-channel crosstalk or independent channel volume control. the apu attenuation coefficients ll, lr, rl and rr may be changed dynamically by the microcontroller, writing their 8-bit indices to the saa2502 through its control interface. the coefficient changes become effective within one sample period after writing. to avoid audible clicks at coefficient changes, the transition from the current attenuation to the next is smoothed. the relationship between the apu coefficient index and the actual coefficient (i.e. the gain) is shown in fig.14 and in table 9 for coefficient index 0 to 64 the step size is - 3 16 db and for coefficient index 64 to 255 the step size is - 3 8 db. the apu has no built-in overflow protection, so the application must assure that the output signals of the apu cannot exceed the 0 db level. for an update of the apu coefficients, it may be required to increase some of the coefficients and decrease some others. the apu coefficients are always written sequentially in a fixed sequence ll, lr, rl and rr. therefore, to prevent (temporary) internal apu data overflow, the following sequence of steps may be necessary: 1. write ll, lr, rl and rr, but change only decreasing coefficients. overwrite increasing coefficients with their old value (therefore do not change these yet). 2. write ll, lr, rl and rr again, but now change increasing coefficients, keeping the other ones unchanged. table 9 apu coef?cient index and actual coef?cient. apu coefficient index c apu coefficient binary decimal 00000000 to 00111111 0 to 63 01000000 to 11111110 64 to 254 11111111 255 0 2 c 32 ------ C 2 c32 C () 16 ----------------------- C fig.13 audio processing unit (apu). handbook, halfpage left decoded audio samples lr ll rl left output audio samples right decoded audio samples rr right output audio samples mgb493 fig.14 relation between apu coefficient index and gain. (1) step - 3 16 db per coefficient increment. (2) step - 3 8 db per coefficient increment. handbook, halfpage mge480 0 0 - 12 - 83.25 254 255 (2) (1) apu coefficient index gain (db) 64
1997 nov 17 22 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.4.9 d ecoder latency time latency time is defined as elapsed time between the moment that the first byte of an audio frame is delivered to the saa2502 and the moment that the output response resulting from the first (sub-band) sample of the same frame reaches its maximum. latency time results from the addition of two internal latency contributions: t latency =t proc +t buf . the processing latency time (t proc ) is sample frequency dependent (see table 10). the input buffer latency time (t buf ) is input interface mode dependent. precision of latency time calculation is sampling rate and bit rate dependent. maximum deviation is roughly plus or minus 4 sample periods. 7.4.9.1 master and slave input interface modes input buffer latency time t buf = (minimum of t buf1 and t buf2 )+cr 3.52 ms: t buf1 is sample frequency dependent (see table 10) t buf2 is input bit rate dependent (see table 11 and table 12) cr is the ratio between maximum and actual value of mclkin frequency. for slave input interface mode not the average input bit rate should be used for table look-up, but cdcl frequency (input bit rate during the burst). for free format bit rates the table should be interpolated (t buf2 is proportional to 1/bit rate). table 10 processing latency time sample frequency (khz) t proc (ms) t buf1 layer i (ms) t buf1 layer ii (ms) 48 6.67 8.00 24.00 44.1 7.26 8.71 26.12 32 10.00 12.00 36.00 24 13.33 16.00 48.00 22.05 14.51 17.41 52.24 16 20.00 24.00 72.00 table 11 buffer latency time; high bit rate bit rate (kbits/s) t buf2 (ms) 448 5.52 384 6.44 320 7.73 256 9.66 192 12.88 160 15.45 128 19.31 96 25.75 64 38.63 48 51.50 32 77.25 16 154.50 table 12 buffer latency time; low bit rate bit rate (kbits/s) t buf2 (ms) 416 5.94 352 7.02 288 8.58 224 11.04 176 14.05 144 17.17 112 22.07 80 30.90 56 44.14 40 61.80 24 103.00 8 309.00
1997 nov 17 23 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.4.9.2 buffer controlled input mode input buffer latency time behaviour is relatively complex in this mode. at start-up (i.e. during the search-for-frame sync) latency time is very small (t buf < 2 ms) because the input buffer remains empty. after a frame sync is detected, normal decoding starts and the buffer fills up to its desired fill level. that level will result in a buffer latency time t buf2 (see tables 11 and 12, t buf1 plays no role) for constant bit rate operation. it is more complex for variable bit rates, at high bit rates the buffer will hold only a fraction of a frame, while at low bit rates it may hold many frames (each possibly of a different bit rate). also input buffer content may deviate from the desired level because data consumption rate at the output of the buffer may be high during short periods while replenishing is limited by cdcl frequency. as a result buffer latency time in buffer controlled input mode may be predicted more or less accurately only at (re)start time. another consequence of buffer behaviour at very low bit rates in this mode is that buffer latency time values may become large. therefore it might be possible that the saa2502 will request data, which is not (yet) available. in those situations the saa2502 is requesting more data than required; storage of more than one complete frame in the input buffer is never necessary. consequently the application may delay delivery of requested data until it becomes available without any effect on correct saa2502 operation. this option constitutes delayed delivery possibility. 7.5 output interface module the output interface module produces stereo baseband output samples in three different formats at the same time: i 2 s spdif 256 times oversampled bit serial analog. any of the three outputs may be enabled or disabled in order to save dissipation and minimize emc generation in applications that do not need all of them. decoded mono streams and the (user) selected channel of dual channel streams are presented at both (left and right) output channels. if indicated in the coded input data, de-emphasis filtering is performed digitally on the output data, thus avoiding the need of external analog de-emphasis filter circuitry. 7.5.1 i 2 s output this output interface section generates decoded baseband audio data in i 2 s format (see fig.15). the i 2 s output interface section consists of 3 signals (see table 13). fig.15 i 2 s output serial data transfer format. handbook, full pagewidth ws sck sd msb left sample lsb msb right sample lsb 1 16/18/20/22 32 1 16/18/20/22 32 valid data mgb502
1997 nov 17 24 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 13 signals of output interfacing the frequency of clock sck is 64 times the sample frequency. the signal sd is the serial baseband audio data, sample by sample (left/right interleaved; the left sample and the right immediately following it form one stereo pair). 32 bits are transferred per sample per channel. the samples are transmitted in twos complement, msb first. the output samples are rounded to either 16, 18, 20 or 22 bit precision, selectable by the control interface flags rnd1 and rnd0. the remainder of the 32 transferred bits per sample per channel are zero. the word select signal ws indicates the channel of the output samples (low if left, high if right). 7.5.2 spidf output 7.5.2.1 spidf format the spdif data format is frame based. one spdif frame represents one audio sampling period. complete frames must be transmitted at the audio sample rate. every frame comprises two sub-frames, each of 32 bits. the data is transmitted in bi-phase mark modulated format to ensure a zero dc component. four bits of data at the beginning of each sub-frame are assigned to frame and sub-frame synchronization, which is achieved using a set of 3 output sequences which violate the bi-phase mark rules. the audio samples occupy 24 bits (bits 4 to 27), transmitted lsb first. depending on the selected accuracy the 2, 4, 6 or 8 lsbs will be logic 0. bits 28 to 31 are occupied by the validity flag for the audio sample, a channel status bit (each super-frame of 192 frames contains two groups of 192 channel status bits, one for each channel), a user data bit, and a parity bit (even parity for bits 4 to 31). these bits are described respectively as v, u, c and p in the spdif specification. the synchronization for the channel status frame is achieved by a pair of preamble violation sequences. the synchronization for the user channel data is embedded within the data. signal direction function sck output data clock sd output baseband audio data ws output word select 7.5.2.2 frame synchronization patterns (bits 0 to 3, spdif subframe) the frame synchronization patterns are based on bi-phase violations. they are sent as shown in table 14 the sequences are sent in place of 4 bi-phase coded bits 0 to 3. they are not bi-phase coded, but are sent as they are. table 14 frame synchronization patterns 7.5.2.3 validity ?ag (bit 28, spdif subframe, v bit) the v bit is intended to indicate an invalid data sample. equipment connected to the interface is expected to perform interpolations across small numbers of invalid (v = logic 1) samples. owing to the manner in which data is decoded in the saa2502, and the sub-band processing of the signal, an input data error affects output audio signals in a complex way. there is not a simple relationship between input errors and damaged audio samples. therefore the validity flag value is made programmable (through the control interface unit) control software can use this bit in any way required. 7.5.2.4 user channel data (bit 29, spdif subframe, u bit) there is a single user data channel. two bits of data in this channel are transmitted in each frame. for this minimum implementation only the possibility to send single byte user messages to the user channel is offered. each byte sent will be preceded by a single logic 1 valued start bit. the 8 bits of the user message are then sent lsb first. 7.5.2.5 channel status data (bit 30, spdif sub-frame, c bit) a group of c channel status bits consists of 192 bits. two groups of channel status bits are transmitted every super-frame (one group for each channel) at a rate of one bit per sub-frame. in this application, both channel status words will be identical. binary pattern description 11101000 b left sub-frame follows. spdif super-frame starts. bit 0 of left c channel will be sent in this subframe 11100100 w right subframe follows 11100010 m left subframe follows
1997 nov 17 25 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 15 channel status data notes 1. this field is filled according to clause 4.2.2.2 of the spdif standard channel status data format for digital audio equipment for consumer use (mode 0). 2. the low sample frequencies of mpeg2 are not defined yet. in order to be able to follow future standardization, the code sent for the three remaining sampling frequencies (24, 22.05 and 16 khz) is programmable through the controller interface. 3. the remaining 162 bits of each channel status word will all be logic 0. individual bits of the status channel will be sent bit 0 first. description bits field indication control ?eld; note 1 0 0 indicates consumer use 1 0 logic 1 reserved for digital data and further standardization 2 c logic 0 = copy prohibited; logic 1 = copy permitted 3 and 4 00 no pre-emphasis (saa2502 has automatic de-emphasis) 5 0 2 channel audio data 6 and 7 00 mode 0 indication category code 8 to 15 00000000 2 channel source number 16 to 19 0000 dont care channel number 20 to 23 0000 dont care sample frequency; note 2 24 to 27 ?eld ?lled in accordance with clause 4.2.2.2 of the spdif standard: 0100 = 48 khz 0000 = 44.1 khz 1100 = 32 khz clock accuracy; note 3 28 and 29 ?eld ?lled in accordance with clause 4.2.2.2 of the spdif standard: 00= level ii (normal accuracy of 0.1%) 7.5.2.6 parity (bit 31, spdif sub-frame, p bit) even parity is generated on the 28 sub-frame data bits (4 to 31) in bit 31. 7.5.2.7 spdif control the spdif interface will be controlled by the microcontroller via the control interface. the v bit is copied into each spdif subframe (once for each data sample). the c bit is inserted twice per spdif super-frame into the channel status data (bit 2 in each c channel). the user byte is inserted into the user channel (preceded by a start bit) immediately after reception through the control interface, otherwise the user channel is filled with logic 0s. table 16 spdif interface control 7.5.2.8 channel status the sampling frequency bits (bits 24 to 27) are derived from the sampling frequency index bits of the input data stream 7.5.2.9 user data only single 8 bit messages are sent. individual messages should be time separated far enough to insert at least 9 logic 0s in between (for easy synchronization at the receiver end at random entry points in the stream). bit/byte default result v bit default = logic 0 valid audio data c bit default = logic 1 digital copy permitted u byte uuuuuuuu 8 bits user byte
1997 nov 17 26 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.5.3 b it serial analog output in order to serve applications which require low to medium performance stereo audio output, two bit serial analog outputs are provided (one for each channel). the on-chip dacs each consist of three functional blocks in series: 4 f s up-sampling filter ac and dc dithering block n f s noise shaper; see table 17. table 17 value of n for n f s noise shaper mode sample rate values external sample clock mode fsc384 = 0 n = 256 fsc384 = 1 n = 384 other clock generator modes f s = 48 khz n = 256 f s = 44.1 khz n = 256 f s = 32 khz n = 384 f s = 24 khz n = 512 f s = 22.05 khz n = 512 f s = 16 khz n = 768 the two analog outputs deliver a pulse density modulated signal, switching between refn and refp. the format is programmable (through the control interface): non return-to-zero format (subsequent logic 1 pulses are merged) return-to-zero format (subsequent logic 1 pulses are separated by logic 0 levels). the quality of the analog output signal depends on several external factors: stability and decoupling of the analog supply absence of jitter on the sample clock which external low-pass filter circuit is used the layout of the low-pass filter. the recommended external low-pass filter is shown in fig.17. with this circuit the dacs performance is < - 75 db (thd + n)/s with a 1 khz sine wave, measured over the bandwidth 20 hz to 20 khz. the amplifier in the low-pass filter circuit is the class ab stereo headphone driver tda1308. the recommended dac output format is non return-to-zero, this has a better signal-to-noise ratio than the return-to-zero format. fig.16 bit serial output formats. handbook, full pagewidth mge481 lftpos rgtpos lftneg rgtneg bit serial data 011001 011001 non-return-to-zero (recommended) return-to-zero
1997 nov 17 27 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 handbook, full pagewidth mbh974 220 pf + 2.5 v 220 pf 390 pf tda1308t SAA2502H neg pos output 11 k w 10 k w 10 k w 10 k w 10 k w 100 m f 11 k w 10 k w fig.17 external low-pass filters 7.6 control interface module 7.6.1 r esetting table 18 resetting is performed by 2 signals a rising edge of the signal stop triggers the next event. the decoding process is interrupted and the input buffer is flushed. consequently audio frame synchronization is abandoned and the decoder starts searching for a new sync in the coded input data stream. in the meantime the output interface is soft muted (i.e. the output signal fades away in approximately 500 samples). there are several other events that have the same effect as a rising edge of the stop signal: change of the current mpeg layer in the input stream change of the current sampling frequency in the input stream change of the current bit rate in the input stream (variable bit rate is not supported) change of current input interface mode (inmod1 and 0) and/or audio frame synchronization mode (symod1 and 0) setting enforcement of a soft reset through the control interface. there is also a level triggered effect which remains provided stop is asserted. when the stoprq control flag is set input data requesting will be halted, otherwise normal input interface behaviour will continue at the bit rate that was valid before stop assertion but all data is signal direction function stop input soft reset and stop decoding reset input hard reset: force default settings considered to be unreliable (as if cdef were asserted). consequently frame synchronization and decoding will not resume until stop is de-asserted. the hard reset signal reset has the same effect as stop but it will also force the control interface settings into their default states. reset must stay high during at least 24 mclkin periods if mclk24 = logic 1 or 12 mclkin periods if mclk24 = logic 0. 7.6.2 i nterrupts the saa2502 is able to generate an interrupt upon the occurrence of one or more of the following events: status bit dst0 has been set (i.e. ancillary/pad data, frame headers and error report are available) rising edge of stop input signal mpeg crc check failed status bit insync has been set status bit insync has been cleared. for more information on these items see sections 7.6.6.1 and 7.6.6.9. each of these interrupts sources may be enabled or disabled as required by the application. after a hard reset all interrupt sources are disabled. when the host processor is interrupted by the saa2502 it should read the interrupt event register to find out which event or events caused the interrupt. reading this register will also clear all pending interrupts. the interrupt pin is active low ( int = logic 0 indicates an interrupt) and it is of the open drain type. consequently it is allowed to wire or this pin with interrupt pins of the same type of other devices. for correct operation an external pull-up resistor should be provided.
1997 nov 17 28 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.6.3 m icrocontroller interface the microcontroller interface operates in one of two distinct modes of operation: l3 or i 2 c-bus. mode setting is determined at initialization. the interface uses 3 signals. the function of these signals in the two modes is indicated in table 19: typical advantages of the use of the l3 protocol are: high speed protocol (normally the speed of the microcontroller will be the limiting factor) the protocol may be implemented using microcontrollers featuring only standard i/o ports. the implemented i 2 c-bus interface is of the 400 kbits/s, 7-bit address, emc improved type. typical advantages of the use of the i 2 c-bus protocol are: standardized protocol which is implemented in hardware in many existing microcontrollers good robustness against external disturbances on interconnecting lines may be applied in multi-master configurations. the cdata output driver is of the open drain type in order to be compliant with the i 2 c-bus specification. during a hard reset of the device, the microcontroller interface mode is determined. as a consequence, the interface cannot be used while the reset signal is asserted. table 19 bus modes 7.6.4 i nitialization mandatory actions that must be taken for correct microcontroller interface start-up at a hard reset (see fig.18). signal l3 mode i 2 c-bus mode direction description cdata l3data sda input/output microcontroller interface serial data cclk l3clk sck input microcontroller interface bit clock cmode l3mode none input microcontroller interface mode select
1997 nov 17 29 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.6.5 t ransfer protocols 7.6.5.1 l3 transfer protocol the protocol enables writing of settings and reading of status and/or data. in this protocol, the host first issues a 6-bit wide device address on cdata while cmode = logic 0. all devices connected to the bus read this address. then data transfers to or from the host are carried out while cmode = logic 1. all devices with a different device address must neglect these data transfers until the next address is issued. only the device with an address equal to the issued device address performs the transfer. table 20 l3 device address. note 1. the data operation mode bits dom1 and dom0 define the current sub-mode of the control interface until the next time a device address is issued (see table 21). table 21 dom1 and dom0 bits bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 011000 dom1 (1) dom0 (1) dom1 dom0 function 0 0 data (new local register contents) sent to the saa2502 0 1 data (current local register contents) sent to the microcontroller 1 0 local register address sent to the saa2502 1 1 short (1 byte) saa2502 status report sent to the microcontroller fig.18 microcontroller interface initialization procedure. (1) the value of the cmode signal while reset is asserted defines the microcontroller interface mode; cmode = logic 1 = i 2 c-bus, cmode = logic 0 = l3. no transfers can be performed (cclk stays high). (2) l3 mode of operation only. for a correct initialization of the interface unit, it is mandatory to make cmode high and low ag ain after reset has been de-asserted. this must occur before any l3 transfer (even to or from other devices) is performed. as shown cclk should sta y high during this step. (3) now the first transfer can be performed on the microcontroller bus. any deviation from these steps may result in undefined behaviour of the microcontroller interface, even with the possibility of disturbing transfers to other devices connected to the control bus. at a hard reset, all writeable data items are forced to their default values. handbook, full pagewidth mge482 reset cmode cclk i 2 c-bus mode l3 mode (1) (2) (3)
1997 nov 17 30 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 fig.19 l3 transfer protocol. (1) halt mode. (2) addressing mode. (3) data from microcontroller to saa2502. (4) data from saa2502 to microcontroller. handbook, full pagewidth mge483 1 1 1 0 0 1 1 0 0 1 saa2502 address saa2502 address saa2502 address saa2502 address saa2502 address h h h h h h h h h h h h h status to microcontroller local register address local register data to microcontroller local register address microcontroller data to local register (2) (2) (2) (2) (2) (1) (1) (1) (1) (1) (1) (1) (1) (1) (1) (1) (1) (1) (4) (3) (3) (3) (4) read status read (block) data write (block) data
1997 nov 17 31 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.6.5.2 i 2 c-bus transfer protocol (see fig.20) the protocol enables reading of data and writing of settings. in this protocol, the host first issues a 7-bit wide device address on cdata immediately after the generation of a start condition. all devices connected to the bus read this address. data transfers to or from the host are then carried out. all devices with a different device address must neglect these data transfers until the next address is issued. only the device with an address equal to the issued device address performs the transfer. table 22 i 2 c-bus device address notes 1. r/ w determines the direction of the subsequent data transfer(s): logic 0 = write, data is sent to the saa2502; logic 0 = read, data is sent to the microcontroller. 2. for further description of the acknowledge bit ack consult the i 2 c-bus specification. bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 - 0011101r/ w (1) ack (2) fig.20 i 2 c-bus transfer protocol. (1) start condition. (2) stop condition. (3) transfer from microcontroller to saa2502. (4) transfer from saa2502 to microcontroller. handbook, full pagewidth mge484 0 0 1 0 0 0 saa2502 address saa2502 address saa2502 address p p s s s local register address local register data to microcontroller local register data to microcontroller local register address microcontroller data to local register (2) (1) (1) (1) (2) (4) (4) (4) (4) (4) (4) (3) (3) (2) (3) (3) (3) (3) (3) (4) (4) read (block) data write (block) data 0 0 1 0 0
1997 nov 17 32 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 data is transferred to or from the saa2502 in local register units (1 byte). local registers may be of readable and/or of writeable type. a local register transfer is initiated by writing the corresponding local register address. the local register unit content is then transferred. 7.6.5.3 register block type some sets of local registers are organized in blocks. one local register address is assigned to a complete block. the local register block address points to the first local register of the block. blocks may be accessed only sequentially by reading or writing successively to the individual members of the block. reading or writing a restricted type block may be interrupted if desired by stopping at any location in the block. transferring may then continue later via a new block operation using a special local address (provided that no other restricted type local saa2502 address has been sent since). this special address is labelled continue block (see section 7.6.6.11). the set of four apu registers is a special type that has an auto increment option. the local addresses of these registers are adjacent to each other. to save time there is an option to programme them in sequence, in one i 2 c-bus transmission. after an initial local address (14h to 17h) the data for each apu coefficient follows in sequence, without the need for transmitting other local addresses. the auto increment will (if required) scroll round from the last local address (17h) back to the first local address (14h). only the apu registers have local addresses that provide the auto increment option. several individual registers store more than one byte of data. to program them, transmit their local address, followed by all the data bytes, in sequence. 7.6.5.4 restricted type registers some local registers and/or local register blocks are of the so-called restricted type. access of such registers is subject to the following limitations: transfer speed in l3 mode is limited to 800 kbits/s. there are no special speed limitations in i 2 c-bus mode other than the 400 kbits/s specification limit. both maximum speeds are scaled down proportionally when the mclk24 frequency is below maximum. restricted registers should not be accessed more frequently than once per audio frame. section 7.6.6 describes the category of each local register/block. 7.6.6 l ocal registers 7.6.6.1 status the host may check the saa2502 status by reading the one byte status word. reading status may be accomplished in two ways: using the special read status protocol of the l3 mode using the normal data exchange protocol. the status byte read branch of the protocol may be looped an arbitrary number of times. if read is looped, status is updated between individual readings. the status bits are shown in table 23. table 23 status register: status is 1 byte (read-only, unrestricted type, local address = 1ah) bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 dst1 dst1 unde?ned unde?ned unde?ned unde?ned insync unde?ned
1997 nov 17 33 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 24 explanation of bits in table 23 notes 1. dst0 values in general do not have a determined duration. however, dst0 = logic 1 lasts at least 0.4 frame period when mpeg layer i data is decoded, and 0.8 frame period when mpeg layer ii data is decoded. table 25 indicates the validity of the saa2502 readable data items with respect to the decoding subprocess. 2. some of the readable local register bits only have significance if insync is logic 1. table 25 validity of the saa2502 readable data items with respect to the decoding subprocess note 1. reading of a data item in a period when it is not valid renders undefined data 7.6.6.2 clock generator control table 26 clock generator control 1: 1 byte (write-only, unrestricted type, local address = 11h) table 27 clock generator control 2: 1 byte (write-only, unrestricted type, local address = 12h) bit description dst1 and dst0 by interpreting dst1 and 0, the host can synchronize to the input frame frequency and also determine at which moment specific data items are available to be read. the value of dst1 and 0 is only valid if flag insync is set. dst1 this is a modulo 2 frame counter, i.e. dst1 inverts at the moment the decoding of a new frame is started. dst1 enables the host to sample the data items available flag dst0 less frequently, meanwhile enabling the host to see if it missed a state. dst0 bit indicates whether data items are available to be read; note 1: logic 0 indicates updating of data items is in progress (consequently they are invalid) logic 1 indicates ancillary (or pad) data, frame headers and error report are valid. insync synchronization indication: logic 0 indicates not synchronized to input audio frame borders logic 1 indicates synchronized to input audio frame borders; note 2. decoding frame n decoding frame n + 1 dst1 = 0 dst1 = 1 dst0 = 0 dst0 = 1 dst0 = 0 dst0 = 1 not valid; note 1 ancillary data (frame n - 1) not valid; note 1 ancillary data (frame n) frame headers (frame n) frame headers (fram en+1) error report (frame n) error report (frame n+1) bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 fscinp fsc384 fscena n3b4 n3b3 n3b2 n3b1 n3b0 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 n2b1 n2b0 n1b3 n1b2 n1b1 n1b0 phsrvs phsmod
1997 nov 17 34 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 28 explanation of bits in tables 26 and 27 note 1. default settings (settings value after a hard reset). 7.6.6.3 input and decoding control table 29 input and decoding control: 1 byte (write-only, restricted type, local address = 33h) bit description fscinp external sample clock mode: logic 0 (1) : internal sample clock mode (sample clock derived from mclkin and x22in clock inputs) logic 1: external sample clock mode (fsclkin is sample clock input) fsc384 external sample clock frequency indication: logic 0 (1) : fsclkin is 256 f s logic 1: fsclkin is 384 f s fscena fsclk output enable flag: logic 0 (1) : fsclk output is disabled logic 1: fsclk output is enabled phsmod phase detector mode of operation: logic 0 (1) : edge triggered mode of operation logic 1: xor mode of operation phsrvs reversed phase detection: logic 0 (1) : normal phase detection logic 1: reversed phase detection (characteristics mirrored with reference to vertical axis) n1b3 to 0 code for n1 value: 0 (1) n1 = 8; 1 n1 = 16; 2 n1 = 24; 3 n1 = 32; 4 n1 = 40; 5 n1 = 48; 6 n1 = 56; 7 n1 = 64 n2b1 to 0 code for n2 value: 0 (1) n2 = 5; 1 n2 = 25; 2 n2 = 125; 3 n2 = 625 n3b4 to 0 n3 - 1; range 0 (1) to 31 (n3 is 1 to 32) bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 symod1 symod0 inmod1 inmod0 stoprq crcact selch2 sfcrc
1997 nov 17 35 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 30 explanation of bits in table 29 notes 1. default settings (settings value after a hard reset). 2. the saa2502 can only decode one of the dual channels, at a time. both left and right audio outputs then play the selected channel. table 31 sampling rate and bit rate: 1 byte (write-only, unrestricted type, local address = 1bh) table 32 soft reset: 1 byte (write-only, unrestricted type, local address = 1eh) bit description symod1 and symod0 audio frame synchronization mode: 00 (1) : general non-byte aligned frame synchronization 01: mpeg layer ii non-byte aligned frame synchronization 10: byte aligned frame synchronization 11: sync pulse frame synchronization inmod1 and inmod0 input interface mode of operation: 00 (1) : master input mode for static bit rates 01: slave input mode for static bit rates 10: buffer controlled input mode for static bit rates 11: buffer controlled input mode for variable bit rates stoprq enable stop requesting flag: 0 (1) : input requesting continues when stop = logic 1 1: input requesting stops when stop = logic 1 crcact crc presence: 0 (1) : protection bit in the mpeg frame header is used to determine crc presence 1: crc is assumed be present by definition (the protection bit is overruled) selch2 (2) dual channel mode channel select (with other modes of input data = dont care): 0 (1) : select channel i 1: select channel ii sfcrc enable scale factor crc protection: 0 (1) : no scale factor protection 1: scale factor crc protection enabled bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 sfx2 sfx1 sfx0 brx4 brx3 brx2 brx1 brx0 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 00000000
1997 nov 17 36 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 33 sample frequency index setting notes 1. modification of sfx values is only possible while insync = logic 0. writing the sample rate control word while insync = logic 1 will have no effect. 2. default settings (settings value after a hard reset). sfx2 to sfx0 (1) sample frequency (khz) 000 22.05 001 24 010 16 011 - 100 44.1 (2) 101 48 110 32 111 - table 34 input bit rate index setting notes 1. modification of brx values is only possible while insync = logic 0. writing the bit rate control word while insync = logic 1 will have no effect. 2. default settings (settings value after a hard reset). brx4 to brx0 (1) bit rate (kbits/s) 00000 - 00001 8 00010 16 00011 24 00100 32 00101 40 00110 48 00111 56 01000 - 01001 16 01010 32 01011 48 01100 64 01101 80 01110 96 01111 112 10000 128 10001 144 10010 160 10011 176 10100 192 10101 - 10110 224 10111 - 11000 256 11001 288 11010 320 11011 352 11100 384 (2) 11101 416 11110 448 11111 -
1997 nov 17 37 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.6.6.4 soft reset writing to this local address has the same effect as a rising edge at the stop input (pin 12). 7.6.6.5 dynamic range compression control table 35 drc control registers: 4 bytes (read/write, restricted block type, local address = 20h) table 36 explanation of bits in table 35 note 1. default settings (settings value after a hard reset). 7.6.6.6 output control the output interface is controlled by 4 local registers and a register block. table 37 output control register: 1 byte (write-only, unrestricted type, local address = 10h) table 38 spdif sf code 1: 1 byte (write-only, unrestricted type, local address = 18h) table 39 spdif sf code 2: 1 byte (write-only, unrestricted type, local address = 19h) subsequent bytes 76543210 compression slope cslp7 cslp6 cslp5 cslp4 cslp3 cslp2 cslp1 cslp0 maximum compression 0 cmax6 cmax5 cmax4 cmax3 cmax2 cmax1 cmax0 compression offset cofs7 cofs6 cofs5 cofs4 cofs3 cofs2 cofs1 cofs0 release rate 0 0 0 crrt4 crrt3 crrt2 crrt1 crrt0 bit description cslp7 to cslp0 compression slope range 0 (1) to 255; unit = 1 256 db per db cmax6 to cmax0 maximum amplification range 0 (1) to 127; unit = 3 16 db cofs7 to cofs0 compression offset range 0 (1) to 255; unit = 3 16 db crrt4 to crrt0 release rate range 1 (1) to 31; unit = 3 256 db per 384 samples bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 spdena i2sena anaena anartz rnd1 rnd0 spd_v spd_c bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 22c3 22c2 22c1 22c0 24c3 24c2 24c1 24c0 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 0000 16c3 16c2 16c1 16c0
1997 nov 17 38 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 40 spdif user byte: 1 byte (write-only, unrestricted type, local address = 1fh) table 41 explanation of bits in tables 37, 38, 39 and 40 notes 1. default settings (settings value after a hard reset). 2. anartz = logic 1 is only allowed in internal sample clock mode; fscinp = logic 0 in clock generator control word 1. apu coefficients are set by writing their 8-bit indices to the 4-byte apu coefficient local register block. at a hard reset, indices ll and rr are set to 0 (no attenuation) and indices lr and rl to 255 (infinite attenuation; no crosstalk). bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 spdu7 spdu7 spdu5 spdu4 spdu3 spdu2 spdu1 spdu0 bit description spdena enable spdif output pin: logic 0 (1) : spdif output pin is disabled logic 1: spdif output pin is enabled i2sena enable i 2 s output: logic 0 (1) : i 2 s output is disabled logic 1: i 2 s output is enabled anaena enable analog output: logic 0: analog output is disabled logic 1 (1) : analog output is enabled anartz analog output return-to-zero mode: logic 0 (1) : non return-to-zero mode; subsequent logic 1s in analog outputs are merged logic 1 (2) : return-to-zero mode; subsequent logic 1s in analog outputs are separated rnd1 and 0 i 2 s and spdif output sample rounding control: 00 (1) : output rounded to 16 bits 01: output rounded to 18 bits 10: output rounded to 20 bits 11: output rounded to 22 bits spd_v value of validity flag (v bit) in spdif output format: logic 0 (1) : valid logic 1: not valid spd_c value of copy permission flag (c bit) in spdif output format: logic 0 (1) : copy prohibited logic 1: copy permitted 22c3 to 22c0 spdif code used for 22.05 khz sample frequency; default = 0100 (1) 24c3 to 24c0 spdif code used for 24 khz sample frequency; default = 0110 (1) 16c3 to 16c0 spdif code used for 16 khz sample frequency; default = 0111 (1) spdu7 to spdu0 spdif user byte (content of byte is sent on spdif user channel); default = inactive (1)
1997 nov 17 39 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 42 apu coef?cients: 4 bytes (read/write, unrestricted special block type table 43 explanation of bits in table 42 note 1. default settings (settings value after a hard reset). the apu coefficient block type is a special one: block accesses may start at any individual coefficient (each has its own local address) block accesses may also extent past rr (the block access will wrap around to ll). 7.6.6.7 interrupt control interrupt generation is controlled using two separate single byte local registers. table 44 interrupt event register: 1 byte (read-only, unrestricted type, local address = 1ch) the separate bits of the interrupt event register indicate the occurrence of the events shown in table 44 subsequent bytes bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 local address apu coef?cient ll ll7 ll6 ll5 ll4 ll3 ll2 ll1 ll0 14h apu coef?cient lr lr7 lr6 lr5 lr4 lr3 lr2 lr1 lr0 15h apu coef?cient rl rl7 rl6 rl5 rl4 rl3 rl2 rl1 rl0 16h apu coef?cient rr rr7 rr6 rr5 rr4 rr3 rr2 rr1 rr0 17h bit description ll7 to ll0 left channel in to left channel out attenuation index range 0 (1) to 255; see fig.14 lr7 to lr0 left channel in to right channel out attenuation index range 0 to 255 (1) ; see fig.14 rl7 to rl0 right channel in to left channel out attenuation index range 0 to 255 (1) ; see fig.14 rr7 to rr0 right channel in to right channel out attenuation index range 0 (1) to 255; see fig.14 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 unde?ned unde?ned unde?ned dst0u stop crcerr insnc nosnc
1997 nov 17 40 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 45 explanation of bits in table 44 note 1. default settings (settings value after a hard reset). table 46 interrupt masking register: 1 byte (write-only, unrestricted type, local address = 1dh) the individual bits of the interrupt masking register (table 46) may mask the interrupt events at the same bit location in the interrupt event register (table 44): logic 0 (default setting, setting value after a hard reset); interrupt event is masked. logic 1; interrupt event is not masked. masked interrupt are still flagged in the interrupt event register, they just do not have an effect on the intrpt interrupt pin (thus polling of masked interrupts is possible). bit description dst0u dst0 has been set (valid ancillary/pad data, headers and error report) stop rising edge of stop input signal crcerr mpeg crc check failed insnc status bit insync was set nosnc status bit insync was cleared logic 0 (1) ; no interrupt for this event logic 1; interrupt for this event bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 0 0 0 imsk4 imsk3 imsk2 imsk1 imsk0
1997 nov 17 41 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.6.6.8 frame headers information about input data, derived by the saa2502 from the input data frame headers, may be read from the frame header items. both the frame header bytes decoded from the input bit stream and the header bytes used for the actual decoding may be read. the decoded frame header item is valid independent of the value of status flag insync. it shows, for example, the decoded headers while the saa2500 is in the process of synchronizing. the used frame header item is only valid if status flag insync is set. the used header bytes are derived by the saa2502 from the decoded header bytes by filling in known header fields (e.g. those that have a fixed value) and overruling detected errors. table 47 decoded frame header: 3 bytes (read-only, restricted block type, local address = 21h) note 1. the emph1 and emph0 bits may only be used to monitor the current de-emphasis indication. corresponding de-emphasis is performed automatically before outputting the baseband audio signal. table 48 used frame header: 3 bytes (read-only, restricted block type, local address = 22h) note 1. the emph1 and emph0 bits may only be used to monitor the current de-emphasis indication. corresponding de-emphasis is performed automatically before outputting the baseband audio signal. table 49 explanation of bits in tables 47 and 48 subsequent bytes bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 decoded header byte 1 sy3 sy2 sy1 sy0 id lay1 lay0 nopr decoded header byte 2 br3 br2 br1 br0 fs1 fs0 unde?ned unde?ned decoded header byte 3 mod1 mod0 modx1 modx0 copr orig emph1 (1) emph0 (1) subsequent bytes bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 used header byte 1 1111id1lay0 nopr used header byte 2 br3 br2 br1 br0 fs1 fs0 unde?ned unde?ned used header byte 3 mod1 mod0 modx1 modx0 copr orig emph1 (1) emph0 (1) bit description sy3 to sy0 last 4 bits of the synchronization word id algorithm identification lay1 and lay0 layer nopr flag for crc on header plus bit allocation plus scale factor select information br3 to br0 bit rate index fs1 and fs0 sample rate index mod1 and mod0 mode modx1 and modx0 mode extension copr copyright flag orig original or home copy flag emph1 and emph0 audio de-emphasis indication
1997 nov 17 42 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.6.6.9 error report the validity of bit allocation plus scale factor select information and the result of the scale factor crcs (only when scale factor crcs are enabled) may be read from the error report register. the error report is only valid when status flag insync is set. table 50 error report register: 1 byte (read-only, restricted type, local address = 24h) table 51 explanation of bits in table 50 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 balok decfm unde?ned unde?ned sf3ok sf2ok sf1ok sf0ok bit description balok bit allocation and scale factor select information validity indication: logic 0; bit allocation or scale factor select information are incorrect or the crc over header plus bit allocation plus scale factor select information has failed logic 1; bit allocation and scale factor select information are correct and crc over header plus bit allocation plus scale factor select information is correct or not active decfm frame skipping or frame decoding indication: logic 0; current input data frame is skipped, and the corresponding baseband audio output frame is muted due to input data errors or inconsistencies; audio frame synchronization is maintained logic 1; current frame is decoded normally sf3ok to sf0ok scale factor crcs not enabled; bits are invalid scale factor crcs enabled: logic 0; one or more scale factors have been concealed in sub-band block 0 to 3 logic 1; no scale factor concealment in sub-band block 0 to 3 (crc check was ok) block 0; sub-bands 0 to 3 block 1; sub-bands 4 to 7 block 2; sub-bands 8 to 15 block 3; sub-bands 16 to 31
1997 nov 17 43 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 7.6.6.10 ancillary data and program associated data with standard mpeg input data, the last 54 bytes of each frame, which may carry ancillary data (ad), are buffered by the saa2502 to be read by the host. subsequent ancillary data bytes are read in reversed order with respect to their order in the input data bit stream; the first item data byte is the last frame byte in the input bit stream. the ancillary data block of local registers is refilled for every frame. the host must either know or determine itself how many of the ancillary data bytes are valid per frame. the ancillary data block contains only valid data when status flag insync is set. table 52 ancillary data: 54 bytes (read-only, restricted block type, local address = 25h) similarly when scale factor crcs are enabled, the fixed program associated data (fpad) and extended program associated data (xpad) bytes contained in each frame may be read, with the 2 fpad bytes first, followed by maximum 52 xpad bytes. subsequent fpad and xpad bytes are read in reversed order with respect to their order in the input data bit stream; the first item data byte is the last pad byte in the input bit stream. the host must determine itself how many of the xpad bytes are valid per frame by interpretation of the fpad content. the pad data block contains only valid data when status flag insync is set. table 53 xpad plus fpad: 54 bytes (read-only, restricted block type, local address = 25h) 7.6.6.11 continue block operation local address 00h is reserved for continuation of restricted type block operations. whenever this local address is used, it will result in continuation of any restricted type block transfer at the point where it was interrupted (provided that no other restricted type saa2502 transfer was carried out since). subsequent bytes 7654321 0 ad byte 1 to byte 54 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 subsequent bytes 7654321 0 fpad bytes 1 and 2; xpad byte 1 to byte 52 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0
1997 nov 17 44 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 8 appendix 8.1 l3 interface speci?cation 8.1.1 i ntroduction the main purpose of the interface definition is to define a protocol that allows for the transfer of control information and operational details between a microcontroller and a number of slave devices, at a rate that exceeds other common interfaces, but with a sufficient low complexity for application in consumer products. it should be clearly noted that the current interface definition is intended for use in a single apparatus, preferably restricted to a single printed circuit board. the interface requires 3 signal lines (apart from a return ground) between the microcontroller and the slave devices (from this the name l3 is derived). these 3-lines are common to all ics connected to the bus: 1. l3mode 2. l3data 3. l3clk. l3mode and l3clk are always driven by the microcontroller, l3data is bidirectional: table 54 the 3-lines common to all ics; l3mode, l3clk and l3data notes 1. l3mode is used for the identification of the operation mode. 2. l3clk is the bit clock to which the information transfer will be synchronized. 3. l3data will carry the information to be transferred. signal microcontroller slave device l3mode (1) output input l3clk (2) output input l3data (3) output/input input/output all slave devices in the system can be addressed using a 6 bit address. this allows for up to 63 different slave devices, as the all 0 address is reserved for special purposes. in operation 2 modes can be identified: 1. addressing mode (am). during addressing mode a single byte is sent by the microcontroller. this byte consists of 2 data operation mode (dom) bits and 6 operational address (oa) bits. each of the slave devices evaluates the operational address. only the device that has been issued the same operational address will become active during the following data mode. the operation to be executed during the data mode is indicated by the two data operation mode bits. 2. data mode (dm). during data mode information is transferred between microcontroller and slave device. the transfer direction may be from microcontroller to slave (write) or from slave to microcontroller (read). however, during one data mode the transfer direction can not change. 8.1.1.1 addressing mode in order to start an addressing mode the microcontroller will make the l3mode line low. the l3clk line is lowered 8 times during which the l3data line transfers 8 bits. the information is presented lsb first and remains stable during the low phase of the l3clk signal. the addressing mode is ended by making the l3mode line high.
1997 nov 17 45 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 55 preferred allocations dom1 dom0 function remarks 0 0 data from microcontroller to saa2500 general purpose data transfer 0 1 data from saa2500 to microcontroller general purpose data transfer 1 0 control from microcontroller to saa2500 register selection for data transfer 1 1 status from saa2500 to microcontroller short device status message handbook, halfpage 01234567 l3mode l3clk l3data mgb505 fig.21 addressing mode. the meaning of the bits on l3data. bit 0 and bit 1; these are the data operation mode (dom) bits that indicate the nature of the following data transfer. the prefer red allocations are given in table 55. bit 2 to bit 7; these bits act as 6 bit operational ic address, with bit 7 as msb and bit 2 as lsb. 8.1.1.2 data mode in the data mode the microcontroller sends or receives information to or from the selected device. during data transfer the l3mode line is high. the l3clk line is lowered 8 times during which the l3data line carries 8 bits. the information is presented lsb first and remains stable during the low phase of the l3clk signal. the basic data transfer unit is an 8-bit byte. no other basic data transfer unit is allowed. 8.1.1.3 halt mode in between transfer units the l3mode line will be driven low by the microcontroller to indicate the completion of a unit transfer. this is called halt mode (hm). during halt mode the l3clk line remains high (to distinguish it from an addressing mode). handbook, halfpage 01234567 l3mode l3clk l3data mgb504 fig.22 data transfer mode.
1997 nov 17 46 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 8.1.2 e xample of a data transfer fig.23 example of transfer of 4 bytes. handbook, full pagewidth mge485 l3 mode l3clk l3data address data byte1 data byte2 data byte3 data byte4 address a data transfer starts when the microcontroller sends an address on the bus. all ics will evaluate this address, but only the ic addressed will be an active partner for the microcontroller in the following data transfer mode. during the data transfer mode bytes will be sent from or to the microcontroller. the l3mode line is made low (halt mode) in between byte transfers. only bytes should be used as basic data transfer units. after the data transfer the microcontroller does not need to send a new address until a new data transfer is necessary. 8.1.3 t iming requirements these are requirements for the slave devices designed in accordance with the l3 interface definitions. 8.1.3.1 addressing mode fig.24 timing (addressing mode). handbook, full pagewidth mgb507 t d1 h2 cl t ch t h1 t su t t l3clk l3mode l3data
1997 nov 17 47 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 8.1.3.2 data transfer 8.1.3.3 halt mode fig.25 timing (data transfer). an dbook, full pagewidth mgb508 l3data microcontroller to ic t d1 h2 cl t ch t h1 t su t t d2 t d3 t h3 t d4 t d5 t l3data ic to microcontroller l3clk l3mode fig.26 timing (halt mode). handbook, full pagewidth mgb509 t l l3clk l3mode d5 t h2 t t d1 d2 t l3data ic to microcontroller
1997 nov 17 48 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 56 requirements for timing; note 1 notes 1. l3data output timing is given with 0 pf external load (derating of maximum delay = 0.5 ns/pf). maximum external l3data load = 50 pf. 2. t = 4 mclkin cycle time if mclk24 = logic 1; t = 2 mclkin cycle time if mclk24 = logic 0. 8.1.4 t iming 8.1.4.1 general ancillary data if the last part of an audio frame is not occupied by encoded sub-band samples, it may be used to transfer any other data. definition of size, format and meaning of this so called ancillary data is completely up to the application (there are no mpeg requirements). non-byte aligned layer i coded input audio frames should however preferably not (always) end with a logic 1 valued bit. in practice there are two common ways to define the size of ancillary data: the number of ancillary data bytes per frame is fixed and known by the application. there is a fixed minimum size of the ancillary data block (usually this size is small; one or two bytes). the fixed part of the block then contains an indication of the actual size of the ancillary data block. if room for ancillary data is present the content will be stored to be read by the microcontroller (up to a maximum of 54 bytes). symbol parameter min. max. unit microcontroller to slave device; note 2 t cl l3clk low time t + 10 - ns t ch l3clk high time t + 10 - ns t d1 l3mode set-up time before ?rst l3clk low 10 - ns t h1 l3data hold time after l3clk high 10 - ns t h2 l3mode hold time after last l3clk high 15 - ns t su l3data set-up time before l3clk high t + 10 - ns t l l3mode low time t + 10 - ns slave device to microcontroller; note 2 t d2 l3mode high to l3data enabled time 0 20 ns t d3 l3mode high to l3data stable time - 20 ns t d4 l3clk high to l3data stable time - 2t + 30 ns t d5 l3mode low to l3data disabled time 0 20 ns t h3 l3data hold time after l3clk high t - ns
1997 nov 17 49 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 8.1.4.2 ancillary data containing scale factor crcs if scale factor crc protection is enabled, the required crc values for each audio frame are carried among the ancillary data of the previous frame. this approach ensures mpeg compatibility for encoded streams with scale factor protection. the saa2502 assumes the next ancillary data format when scale factor crc protection is enabled: the last 2 bytes of each audio frame carry the minimum ancillary data. these two bytes are called fpad (fixed program associated data) bytes. definition of the content of fpad is up to the application but should contain information on the length of the remainder of the ancillary data if that length is variable. fpad bytes are stored to be read by the microcontroller. the byte before the fpad bytes is called crc0 and contains the scale factor crc for sub-bands 0 to 3. the byte before crc0 is called crc1 and contains the scale factor crc for sub-bands 4 to 7. an optional byte crc2 may precede crc1. it contains the scale factor crc for sub-bands 8 to 15 and is present only for sub-band limits greater than 8. there may be an optional byte crc3 before crc2. it contains the scale factor crc for sub-bands 16 to 31 and will be present only for sub-band limits greater than 16. before the sub-band crcs more ancillary data may be present. this extra ancillary data is called xpad (extended program associated data). if xpad is present it will be stored to be read by the microcontroller (up to a maximum of 52 bytes). 8.1.4.3 boundary scan test provision the saa2502 contains a 5-pin interface for boundary scan test (bst): table 57 boundary scan test in normal use trst must be low, tck must be low or high while tdi and tms must be high or not connected. otherwise when any of these pins is used in a way not designed correctly for boundary scan test purposes in the application, damaging of the saa2502 and/or the components surrounding it may occur. signal direction function tdi input boundary scan test data input tdo output boundary scan test data output tms input boundary scan test mode select tck input boundary scan test clock trst input boundary scan test reset fig.27 ancillary data containing scale factor crcs. handbook, full pagewidth mge486 xpad 52 fpad 2 fpad 1 xpad 1 crc 3 crc 2 crc 1 crc 0 - - - optional optional optional optional optional end of frame n start of frame n + 1
1997 nov 17 50 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 table 58 boundary scan controller instructions opcode instruction binary 000 extest binary 001 idcode binary 010 sample binary 011 clamp binary 100 intest binary 101 stctest binary 110 unde?ned binary 111 bypass table 59 boundary scan register definition notes 1. low or high control of 2 state output. 2. low or high control of 3 state output. 3. low or high impedance control of 3 state output. 4. low or high impedance control of open drain output. number port function 1 fsclk output 2; note 1 2 sck output 2; note 1 3 sd output 2; note 1 4 ws output 2; note 1 5 spdif output 2; note 1 6 tc0 input 7 tc1 input 8 fsclkin input 9 refclk input 10 x22in input 11 mclk24 input 12 mclkin input 13 phdif output 3; note 2 14 phdif control; note 3 15 int open drain; note 4 16 reset input 17 stop input 18 cdrq output 2; note 1 19 cdef input 20 cdcl input 21 cdcl output 3; note 2 22 cdcl control; note 3 23 cd input 24 cdsy input 25 cdval input 26 cclk input 27 cdata input 28 cdata open drain; note 4 29 cdata control; note 3 30 cmode input
1997 nov 17 51 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 8.1.4.4 factory test scan chain provision table 60 signals provided for factory test scan chain control in normal use factory test scan chain control pins must be not connected or kept low. if any of these pins are pulled high in the application, damage to the saa2502 and/or the surrounding components may occur. 8.1.4.5 provision to read internal status the following internal status information is made available for reading. it provides designers additional information on status and/or progress of internal processes. this information has no meaning for the application. table 61 transcoder program counter register: 1 byte (read-only, unrestricted type, local address = 10h) table 62 decoder program counter register: 1 byte (read-only, unrestricted type, local address = 11h) table 63 transcoder flag register: 1 byte (read-only, unrestricted type, local address = 12h) table 64 transcoder and decoder branch conditions register: 1 byte (read-only, unrestricted type, local address = 13h) signal direction function tc0 input factory test scan chain control 0 tc1 input factory test scan chain control 1 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 tpc11 tpc9 tpc8 tpc7 tpc6 tpc5 tpc4 tpc3 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 tpc7 tpc6 tpc5 tpc4 tpc3 tpc2 tpc1 tpc0 bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 unde?ned tnsync torenb tirenb tcrc16 tcrcf terrf tskf bit 7 bit 6 bit 5 bit 4 bit 3 bit 2 bit 1 bit 0 doful dirdy doreq tiemt toreq tupreq tireq tpor
1997 nov 17 52 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 9 limiting values in accordance with the absolute maximum rating system (iec 134). notes 1. input voltage should not exceed 6.5 v unless otherwise specified. 2. equivalent to discharging a 100 pf capacitor through a 1.5 k w series resistor. 3. equivalent to discharging a 200 pf capacitor through a 0 w series resistor. symbol parameter conditions min. max. unit v dd supply voltage - 0.5 +6.5 v v i input voltage note 1 - 0.5 v dd + 0.5 v i dd supply current - 100 ma i ss supply current - 100 ma i i input current - 10 +10 ma i o output current - 20 +20 ma p tot total power dissipation - 163 mw t stg storage temperature - 65 +150 c t amb operating ambient temperature - 40 +85 c v es electrostatic handling note 2 - 2000 +2000 v note 3 - 200 +200 v
1997 nov 17 53 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 10 dc characteristics v dd = 4.5 to 5.5 v; t amb = - 40 to +85 c; unless otherwise speci?ed. notes 1. only applies to pin 25 (fsclkin). 2. boundary scan test inputs. 3. all inputs except for tc0, tc1, fsclkin, mclkin, x22in, refp and refn. 4. dac outputs i oh = 2 ma. typical dac output impedance = 125 w . symbol parameter conditions min. typ. max. unit inputs v ih high level input voltage (cmos) note 1 0.7v dd -- v v il low level input voltage (cmos) note 1 -- 0.3v dd v v ih high level input voltage (ttl) note 2 2 -- v v il low level input voltage (ttl) note 2 -- 0.8 v v tlh rising edge threshold voltage (cmos hysteresis) note 3 -- 0.8v dd v v thl falling edge threshold voltage (cmos hysteresis) note 3 0.2v dd -- v v hys hysteresis voltage (cmos hysteresis) - 0.3v dd - v i i input current (all input types) - 5 - +5 m a r pull pull-up or pull-down resistance 14 - 140 k w outputs v oh high level output voltage note 4 v dd - 0.5 -- v v ol low level output voltage note 4 -- 0.5 v i lo leakage current of a disabled output -- 5 m a
1997 nov 17 54 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 11 ac characteristics v dd = 4.5 to 5.5 v; t amb = - 40 to +85 c; unless otherwise speci?ed. symbol parameter conditions min. typ. max. unit clock inputs mclkin t cy cycle time 40 -- ns t h high time 12 -- ns t l low time 12 -- ns x22in t cy cycle time 44 -- ns t h high time 12 -- ns t l low time 12 -- ns fsclkin t cy cycle time 54 -- ns t h high time 12 -- ns t l low time 12 -- ns refclk t cy cycle time 33 -- ns t h high time 12 -- ns t l low time 12 -- ns cdcl t cy cycle time note 1 8 t -- ns t h high time note 1 t + 10 -- ns t l low time note 1 t + 10 -- ns clock outputs fsclk t cy cycle time 54 -- ns t h high time 10 -- ns t l low time 10 -- ns cdcl t cy cycle time note 1 8 t -- ns t h high time note 1 4 t - 10 -- ns t l low time note 1 4 t - 10 -- ns sck t cy cycle time note 2 2 s -- ns t h high time note 2 s - 10 -- ns t l low time note 2 s - 10 -- ns
1997 nov 17 55 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 notes 1. t = 4 mclkin cycle time if mclk24 = logic 1; t = 2 mclkin cycle time if mclk24 = logic 0. 2. s is the audio sample time divided by 128. a) maximum external clock output load = 25 pf. 3. when cdcl is output (input master mode or buffer controlled mode). 4. when cdcl is input (input slave mode). 5. a negative value of t pd means that the output changes before the falling edge of the clock. a) propagation delay times are given with an external load of 0 pf. b) maximum external output load = 50 pf. c) output load derating of maximum propagation delay time is 0.5 ns per pf. 6. sample frequency = 44.1 khz. data inputs: cd, cdef, cdsy, cdval, tdi and tms t su set-up time cd, cdef, cdsy and cdval cdcl clock; note 3 42 -- ns tdi and tms 50 -- ns t h hold time cd, cdef, cdsy and cdval cdcl clock; note 3 0 -- ns t su set-up time cd, cdef, cdsy and cdval cdcl clock; notes 1 and 4 t+10 -- ns t h hold time cd, cdef, cdsy and cdval cdcl clock; note 4 10 -- ns t h tdi and tms high time 50 -- ns data outputs cdrq t pd propagation delay time cdrq cdcl clock; note 5 - 22 - +10 ns sd and ws t pd propagation delay time sd and ws sck clock; note 5 - 22 - +10 ns tdo t pd propagation delay time tdo tck clock; note 5 0 - 100 ns analog output performance; note 6 thd + n total harmonic distortion plus noise -- 75 - db dr dynamic range - 75 - db a cs channel separation -- 92 - db symbol parameter conditions min. typ. max. unit
1997 nov 17 56 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 11.1 host interface: cdata, cclk and cmode for l3 mode host interface timing information is detailed in the section 8.1. the i 2 c-bus mode host interface timing is master clock dependent, adherence to this specification is only guaranteed for the maximum mclkin frequency. if mclkin frequency is below maximum in principle all timing figures should be increased proportionally. table 65 supported refclk frequencies refclk (khz) 6 6.4 8 9.6 12 12.8 16 18 19.2 24 25.6 28.8 30 32 36 38.4 40 42 44.8 48 51.2 54 56 57.6 60 64 66 67.2 70.4 72 76.8 78 80 83.2 84 86.4 88 89.6 90 96 102 102.4 104 105.6 108 108.8 112 114 115.2 120 121.6 124.8 126 128 132 134.4 136 138 140.8 144 147.2 150 152 153.6 156 160 162 163.2 166.4 168 172.8 174 176 179.2 180 182.4 184 185.6 186 192 198.4 200 201.6 204 204.8 208 210 211.2 216 220.8 224 228 230.4 232 240 248 249.6 252 256 259.2 264 268.8 270 272 276 278.4 280 288 297.6 300 304 307.2 312 320 324 326.4 330 336 345.6 348 fig.28 timing diagram. handbook, full pagewidth mge487 clock input output t pd t su t h t su t h t h t l t cy
1997 nov 17 57 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 352 360 364.8 368 372 384 390 400 403.2 408 416 420 422.4 432 440 441.6 448 450 456 460.8 464 480 496 499.2 504 510 512 518.4 520 528 537.6 540 544 552 556.8 560 570 576 595.2 600 608 614.4 624 630 640 648 660 672 680 690 696 704 720 736 744 750 760 768 780 800 810 816 832 840 864 870 880 896 900 912 920 928 930 960 992 1000 1008 1020 1024 1040 1050 1056 1080 1104 1120 1140 1152 1160 1200 1240 1248 1260 1280 1296 1320 1344 1350 1360 1380 1392 1400 1440 1488 1500 1520 1536 1560 1600 1620 1632 1650 1680 1728 1740 1760 1800 1824 1840 1860 1920 1950 2000 2016 2040 2080 2100 2112 2160 2200 2208 2240 2250 2280 2304 2320 2400 2480 2496 2520 2550 2560 2592 2600 2640 2688 2700 2720 2760 2784 2800 2850 2880 2976 3000 3040 3072 3120 3150 3200 3240 3300 3360 3400 3450 3480 3520 3600 3680 3720 3750 3800 3840 3900 4000 4050 4080 4160 4200 4320 4350 4400 4480 4500 4560 4600 4640 4650 4800 4960 5000 5040 5100 5120 5200 5250 5280 5400 5520 5600 5700 5760 5800 6000 6200 6240 6300 6400 6480 6600 6720 6750 6800 6900 6960 7000 7200 7440 7500 7600 7680 7800 8000 8100 8160 8250 8400 8640 8700 8800 9000 9120 9200 9300 9600 9750 10000 10080 10200 10400 10500 10560 10800 11000 11040 11200 11250 11400 11520 11600 12000 12400 12480 12600 12750 12800 12960 13000 13200 13440 13500 13600 13800 13920 14000 14250 14400 14880 15000 15200 15360 15600 15750 16000 16200 16500 16800 17000 17250 17400 17600 18000 18400 18600 18750 19000 19200 19500 20000 20250 20400 20800 21000 21600 21750 22000 22400 22500 22800 23000 23200 23250 24000 24800 25000 25200 25500 25600 26000 26400 27000 27600 28000 28500 28800 29000 30000 ----
1997 nov 17 58 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 12 application information k , full pagewidth mgg817 10 k w 4.7 m h c13 100 m f 11 k w r14 10 k w r7 10 k w c2 100 m f c1 10 nf c3 390 pf c4 220 pf u2 tda1308t c10 220 pf c14 100 nf c9 390 pf stop cdrq cdcl cd gnd1 cdef cdsy v dd1 cdval tms refclk phdif reset int cmode cdata cclk spdif trst ws sd sck fsclk tck fsclkin x22in x22out gnd2 mclk24 mlckout tdi mlckin reset 12.228 mhz i 2 c-bus data i 2 c-bus clock 11 12 13 14 15 16 17 18 19 20 21 22 44 43 42 41 40 39 38 37 36 35 34 v dd3 va1 l1 r16 r15 r1 10 k w 11 k w r2 c6 10 nf 10 k w r3 10 k w r8 10 k w r9 r5 3 2 8 4 1 7 5 6 c11 c5 r6 va1 220 pf right left analog r13 220 pf 11 k w r12 10 k w r10 10 k w r11 10 k w 11 k w r4 + 5 v + 5 v + 5 v + 5 v + 2.5 v 4.7 w 4.7 w spdif o/p data clock sync signal data tc1 gnd3 tdo tc0 lftpos lftneg refp refn rgtneg rgtpos 109876 u1 saa2502 54321 23 24 25 26 27 28 29 30 31 32 33 c8 100 m f 100 m f c12 c7 100 m f v dd2 fig.29 application circuit for adr.
1997 nov 17 59 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 13 package outline unit a 1 a 2 a 3 b p ce (1) eh e ll p z y w v q references outline version european projection issue date iec jedec eiaj mm 0.25 0.05 1.85 1.65 0.25 0.40 0.20 0.25 0.14 10.1 9.9 0.8 1.3 12.9 12.3 1.2 0.8 10 0 o o 0.15 0.1 0.15 dimensions (mm are the original dimensions) note 1. plastic or metal protrusions of 0.25 mm maximum per side are not included. 0.95 0.55 sot307-2 95-02-04 97-08-01 d (1) (1) (1) 10.1 9.9 h d 12.9 12.3 e z 1.2 0.8 d e e b 11 c e h d z d a z e e v m a x 1 44 34 33 23 22 12 y q a 1 a l p detail x l (a ) 3 a 2 pin 1 index d h v m b b p b p w m w m 0 2.5 5 mm scale qfp44: plastic quad flat package; 44 leads (lead length 1.3 mm); body 10 x 10 x 1.75 mm sot307-2 a max. 2.10
1997 nov 17 60 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 14 soldering 14.1 introduction there is no soldering method that is ideal for all ic packages. wave soldering is often preferred when through-hole and surface mounted components are mixed on one printed-circuit board. however, wave soldering is not always suitable for surface mounted ics, or for printed-circuits with high population densities. in these situations reflow soldering is often used. this text gives a very brief insight to a complex technology. a more in-depth account of soldering ics can be found in our ic package databook (order code 9398 652 90011). 14.2 re?ow soldering reflow soldering techniques are suitable for all qfp packages. the choice of heating method may be influenced by larger plastic qfp packages (44 leads, or more). if infrared or vapour phase heating is used and the large packages are not absolutely dry (less than 0.1% moisture content by weight), vaporization of the small amount of moisture in them can cause cracking of the plastic body. for more information, refer to the drypack chapter in our quality reference handbook (order code 9397 750 00192). reflow soldering requires solder paste (a suspension of fine solder particles, flux and binding agent) to be applied to the printed-circuit board by screen printing, stencilling or pressure-syringe dispensing before package placement. several techniques exist for reflowing; for example, thermal conduction by heated belt. dwell times vary between 50 and 300 seconds depending on heating method. typical reflow temperatures range from 215 to 250 c. preheating is necessary to dry the paste and evaporate the binding agent. preheating duration: 45 minutes at 45 c. 14.3 wave soldering wave soldering is not recommended for qfp packages. this is because of the likelihood of solder bridging due to closely-spaced leads and the possibility of incomplete solder penetration in multi-lead devices. if wave soldering cannot be avoided, the following conditions must be observed: a double-wave (a turbulent wave with high upward pressure followed by a smooth laminar wave) soldering technique should be used. the footprint must be at an angle of 45 to the board direction and must incorporate solder thieves downstream and at the side corners. even with these conditions, do not consider wave soldering the following packages: qfp52 (sot379-1), qfp100 (sot317-1), qfp100 (sot317-2), qfp100 (sot382-1) or qfp160 (sot322-1). during placement and before soldering, the package must be fixed with a droplet of adhesive. the adhesive can be applied by screen printing, pin transfer or syringe dispensing. the package can be soldered after the adhesive is cured. maximum permissible solder temperature is 260 c, and maximum duration of package immersion in solder is 10 seconds, if cooled to less than 150 c within 6 seconds. typical dwell time is 4 seconds at 250 c. a mildly-activated flux will eliminate the need for removal of corrosive residues in most applications. 14.4 repairing soldered joints fix the component by first soldering two diagonally- opposite end leads. use only a low voltage soldering iron (less than 24 v) applied to the flat part of the lead. contact time must be limited to 10 seconds at up to 300 c. when using a dedicated tool, all other leads can be soldered in one operation within 2 to 5 seconds between 270 and 320 c.
1997 nov 17 61 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 15 definitions 16 life support applications these products are not designed for use in life support appliances, devices, or systems where malfunction of these products can reasonably be expected to result in personal injury. philips customers using or selling these products for use in such applications do so at their own risk and agree to fully indemnify philips for any damages resulting from such improper use or sale. data sheet status objective speci?cation this data sheet contains target or goal speci?cations for product development. preliminary speci?cation this data sheet contains preliminary data; supplementary data may be published later. product speci?cation this data sheet contains ?nal product speci?cations. limiting values limiting values given are in accordance with the absolute maximum rating system (iec 134). stress above one or more of the limiting values may cause permanent damage to the device. these are stress ratings only and operation of the device at these or at any other conditions above those given in the characteristics sections of the speci?cation is not implied. exposure to limiting values for extended periods may affect device reliability. application information where application information is given, it is advisory and does not form part of the speci?cation.
1997 nov 17 62 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 notes
1997 nov 17 63 philips semiconductors preliminary speci?cation iso/mpeg audio source decoder saa2502 notes
internet: http://www.semiconductors.philips.com philips semiconductors C a worldwide company ? philips electronics n.v. 1997 sca56 all rights are reserved. reproduction in whole or in part is prohibited without the prior written consent of the copyright owne r. the information presented in this document does not form part of any quotation or contract, is believed to be accurate and reli able and may be changed without notice. no liability will be accepted by the publisher for any consequence of its use. publication thereof does not con vey nor imply any license under patent- or other industrial or intellectual property rights. netherlands: postbus 90050, 5600 pb eindhoven, bldg. vb, tel. +31 40 27 82785, fax. +31 40 27 88399 new zealand: 2 wagener place, c.p.o. box 1041, auckland, tel. +64 9 849 4160, fax. +64 9 849 7811 norway: box 1, manglerud 0612, oslo, tel. +47 22 74 8000, fax. +47 22 74 8341 philippines: philips semiconductors philippines inc., 106 valero st. salcedo village, p.o. box 2108 mcc, makati, metro manila, tel. +63 2 816 6380, fax. +63 2 817 3474 poland: ul. lukiska 10, pl 04-123 warszawa, tel. +48 22 612 2831, fax. +48 22 612 2327 portugal: see spain romania: see italy russia: philips russia, ul. usatcheva 35a, 119048 moscow, tel. +7 095 755 6918, fax. +7 095 755 6919 singapore: lorong 1, toa payoh, singapore 1231, tel. +65 350 2538, fax. +65 251 6500 slovakia: see austria slovenia: see italy south africa: s.a. philips pty ltd., 195-215 main road martindale, 2092 johannesburg, p.o. box 7430 johannesburg 2000, tel. +27 11 470 5911, fax. +27 11 470 5494 south america: al. vicente pinzon, 173, 6th floor, 04547-130 s?o paulo, sp, brazil, tel. +55 11 821 2333, fax. +55 11 821 2382 spain: balmes 22, 08007 barcelona, tel. +34 3 301 6312, fax. +34 3 301 4107 sweden: kottbygatan 7, akalla, s-16485 stockholm, tel. +46 8 632 2000, fax. +46 8 632 2745 switzerland: allmendstrasse 140, ch-8027 zrich, tel. +41 1 488 2686, fax. +41 1 481 7730 taiwan: philips semiconductors, 6f, no. 96, chien kuo n. rd., sec. 1, taipei, taiwan tel. +886 2 2134 2865, fax. +886 2 2134 2874 thailand: philips electronics (thailand) ltd., 209/2 sanpavuth-bangna road prakanong, bangkok 10260, tel. +66 2 745 4090, fax. +66 2 398 0793 turkey: talatpasa cad. no. 5, 80640 gltepe/istanbul, tel. +90 212 279 2770, fax. +90 212 282 6707 ukraine : philips ukraine, 4 patrice lumumba str., building b, floor 7, 252042 kiev, tel. +380 44 264 2776, fax. +380 44 268 0461 united kingdom: philips semiconductors ltd., 276 bath road, hayes, middlesex ub3 5bx, tel. +44 181 730 5000, fax. +44 181 754 8421 united states: 811 east arques avenue, sunnyvale, ca 94088-3409, tel. +1 800 234 7381 uruguay: see south america vietnam: see singapore yugoslavia: philips, trg n. pasica 5/v, 11000 beograd, tel. +381 11 625 344, fax.+381 11 635 777 for all other countries apply to: philips semiconductors, international marketing & sales communications, building be-p, p.o. box 218, 5600 md eindhoven, the netherlands, fax. +31 40 27 24825 argentina: see south america australia: 34 waterloo road, north ryde, nsw 2113, tel. +61 2 9805 4455, fax. +61 2 9805 4466 austria: computerstr. 6, a-1101 wien, p.o. box 213, tel. +43 160 1010, fax. +43 160 101 1210 belarus: hotel minsk business center, bld. 3, r. 1211, volodarski str. 6, 220050 minsk, tel. +375 172 200 733, fax. +375 172 200 773 belgium: see the netherlands brazil: see south america bulgaria: philips bulgaria ltd., energoproject, 15th floor, 51 james bourchier blvd., 1407 sofia, tel. +359 2 689 211, fax. +359 2 689 102 canada: philips semiconductors/components, tel. +1 800 234 7381 china/hong kong: 501 hong kong industrial technology centre, 72 tat chee avenue, kowloon tong, hong kong, tel. +852 2319 7888, fax. +852 2319 7700 colombia: see south america czech republic: see austria denmark: prags boulevard 80, pb 1919, dk-2300 copenhagen s, tel. +45 32 88 2636, fax. +45 31 57 0044 finland: sinikalliontie 3, fin-02630 espoo, tel. +358 9 615800, fax. +358 9 61580920 france: 51 rue carnot, bp317, 92156 suresnes cedex, tel. +33 1 40 99 6161, fax. +33 1 40 99 6427 germany: hammerbrookstra?e 69, d-20097 hamburg, tel. +49 40 23 53 60, fax. +49 40 23 536 300 greece: no. 15, 25th march street, gr 17778 tavros/athens, tel. +30 1 4894 339/239, fax. +30 1 4814 240 hungary: see austria india: philips india ltd, band box building, 2nd floor, 254-d, dr. annie besant road, worli, mumbai 400 025, tel. +91 22 493 8541, fax. +91 22 493 0966 indonesia: see singapore ireland: newstead, clonskeagh, dublin 14, tel. +353 1 7640 000, fax. +353 1 7640 200 israel: rapac electronics, 7 kehilat saloniki st, po box 18053, tel aviv 61180, tel. +972 3 645 0444, fax. +972 3 649 1007 italy: philips semiconductors, piazza iv novembre 3, 20124 milano, tel. +39 2 6752 2531, fax. +39 2 6752 2557 japan: philips bldg 13-37, kohnan 2-chome, minato-ku, tokyo 108, tel. +81 3 3740 5130, fax. +81 3 3740 5077 korea: philips house, 260-199 itaewon-dong, yongsan-ku, seoul, tel. +82 2 709 1412, fax. +82 2 709 1415 malaysia: no. 76 jalan universiti, 46200 petaling jaya, selangor, tel. +60 3 750 5214, fax. +60 3 757 4880 mexico: 5900 gateway east, suite 200, el paso, texas 79905, tel. +9-5 800 234 7381 middle east: see italy printed in the netherlands 547027/00/02/pp64 date of release: 1997 nov 17 document order number: 9397 750 03068

▲Up To Search▲

Price & Availability of SAA2502H

	To Download SAA2502H Datasheet File
If you can't view the Datasheet, Please click here to try to view without PDF Reader .