PHILIPS SAA2520

INTEGRATED CIRCUITS
DATA SHEET
SAA2520
Stereo filter and codec for MPEG
layer 1 audio applications
Preliminary specification
File under Integrated Circuits, IC01
August 1993
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
FEATURES
GENERAL DESCRIPTION
• Stereo filtering and codec functions in a single chip
The SAA2520 performs the sub-band filtering and audio
frame codec functions to provide efficient audio
compression/decompression for MPEG (11172-3) Layer1
applications. It is capable of functioning as a stand-alone
decoder but requires the addition of an adaptive masking
threshold processor (SAA2521) in order to function as a
highly efficient encoder.
• MPEG coded interface
• Filtered data interface
• Baseband audio data interface
• LT interface to microcontroller
• Clock generator
• Low operating voltage capability.
ORDERING INFORMATION
PACKAGE
EXTENDED TYPE
NUMBER
SAA2520GP(1)
PINS
PIN POSITION
MATERIAL
CODE
44
QFP
plastic
SOT205AG
Note
1. SOT205-1; 1996 August 26.
CLK24
X22OUT
X24OUT
X22IN
X24IN
handbook, full pagewidth CLK22
38
39
40
41
42
43
V DD
28,44
8
FS256
SBDIR
SBEF
1
CLOCK GENERATOR
SAA2520
9
SUB-BAND
SERIAL
INTERFACE
7
10
12
11
2
STEREO
SUB-BAND
FILTER
PROCESSOR
CODEC
3
4
SWS
SCL
SDA
SBCL
SBWS
SBMCLK
MUTEDAC
DEEMDAC
ATTDAC
20
19
21
BASEBAND
SERIAL
INTERFACE
5,37
FILTERED
DATA
INTERFACE
18
17
FDAF
V
SS
FDAC
MICROPROCESSOR
INTERFACE & CONTROL
16
15
14
FRESET
FSYNC
13
SYNCDAI
FDIR
36
35
LTCNT0
LTCNT1
Fig.1 Block diagram.
August 1993
SBDA
2
34
33
LTCLK
LTENA
32
22
PWRDWN
LTDATA
29
6
URDA
RESET
MLB125
Philips Semiconductors
Preliminary specification
34 LTENA
35 LTCNT0
36 LTCNT1
SAA2520
37 V SS
38 CLK22
39 CLK24
40 X22IN
41 X22OUT
42 X24IN
handbook, full pagewidth
43 X24OUT
44 VDD
Stereo filter and codec for MPEG layer 1
audio applications
FS256
1
33 LTCLK
MUTEDAC
2
32 LTDATA
DEEMDAC
3
31 T0
ATTDAC
4
30 T1
V SS
5
URDA
6
SBDIR
7
27 DSC0
SBDA
8
26 DSC1
SBCL
9
25 DSC2
29 RESET
SAA2520
28 VDD
20
21
SCL
SWS
SDA
22
19
FDAC
MLB126
PWRDWN
18
17
FDAF
15
FSYNC 16
12
SBMCLK
FRESET
23 DSC4
14
11
SBEF
FDIR
24 DSC3
SYNCDAI 13
SBWS 10
Fig.2 Pin configuration.
handbook, full pagewidth
AUDIO
AMPLIFIER
digital audio interface
DAC
control
SAA2520
MPEG interface
MPEG
source
system micro interface
MICROCONTROLLER
power down
reset
MLB127
Fig.3 MPEG decoder system data flow diagram.
August 1993
3
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
PINNING
SYMBOL
PIN
DESCRIPTION
TYPE
FS256
1
(Filtered)-I2S
clock; 256 × sample frequency. 12 mA 3-state output + CMOS
input with pull-down
I/O
MUTEDAC
2
DAC control/output expander
O
DEEMDAC
3
DAC control/output expander
O
O
ATTDAC
4
DAC control/output expander
VSS
5
supply ground (0 V)
URDA
6
unreliable drive processing data; CMOS level
I2S
I
SBDIR
7
sub-band
SBDA
8
sub-band I2S data; 4 mA, 3-state output + CMOS input with pull-down
I/O
SBCL
9
sub-band I2S bit clock; 4 mA, 3-state output + CMOS input with pull-down
I/O
SBWS
10
direction: (SWBS, SBCL, SBDA); CMOS level
sub-band
I2S
word select; 4 mA, 3-state output + CMOS input with pull-down
I2S
byte error flag; CMOS level
I
I/O
SBEF
11
sub-band
SBMCLK
12
sub-band I2S clock, 6.144 MHz locked to FS256; 8 mA, 3-state output + CMOS
input with pull-down
O
SYNCDAI
13
DAI synchronization pulse
O
I
FDIR
14
(Filtered)-I2S
FRESET
15
reset signal for SAA2521
O
FSYNC
16
Filtered-I2S sync signal for SAA2521
O
17
Filtered-I2S
I/O
FDAF
direction: (FDAC, FDAF, SDA);
sub-band filter data; 4 mA, 3-state output + CMOS input with
O
pull-down
FDAC
18
Filtered-I2S sub-band codec data; 4 mA, 3-state output + CMOS input with
pull-down
I/O
SCL
19
I2S bit clock; 4 mA, 3-state output + CMOS input with pull-down
I/O
SWS
20
I2S-word
select; 4 mA, 3-state output + CMOS input with pull-down
I/O
SDA
21
I2S baseband data filter; 4 mA, 3-state output + CMOS input with pull-down
I/O
PWRDWN
22
power-down mode; CMOS level
DSC4
23
test pin
DSC3
24
test pin
DSC2
25
test pin
DSC1
26
test pin
DSC0
27
test pin
VDD
28
positive supply voltage (+5 V)
RESET
29
system reset; CMOS level with pull-down and hysteresis
T1
30
test pin; do not connect
T0
31
test pin; do not connect
LTDATA
32
LT interface data; 4 mA, 3-state output + CMOS input with pull-down
LTCLK
33
LT interface bit clock; CMOS level
August 1993
4
I
I
I/O
I
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SYMBOL
PIN
DESCRIPTION
SAA2520
TYPE
LTENA
34
LT interface enable; CMOS level
I
LTCNT0
35
LT interface control; CMOS level
I
LTCNT1
36
LT interface control; CMOS level
I
VSS
37
supply ground (0 V)
CLK22
38
22.5792 MHz buffered output
O
CLK24
39
24.576 MHz buffered output
O
X22IN
40
22.5792 MHz crystal input
I
X22OUT
41
22.5792 MHz crystal output
O
X24IN
42
24.576 MHZ crystal input
I
X24OUT
43
24.576 MHz crystal output
O
VDD
44
positive supply voltage (+5 V)
August 1993
5
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
handbook, full pagewidth
from SAA2521
allocation information
and scale factor indices
ALLOCATION &
SCALE FACTOR
INFORMATION
TABLE
SYNC AND
CODING
INFORMATION
base band
samples
SUB-BAND
FILTER
sub - band
samples
SCALING &
QUANTIZATION
FORMATTER
MPEG
OUTPUT
DATA
quantized samples
MLB128
Fig.4 Encoding mode.
handbook, full pagewidth
sync/coding
allocation
scale factor
MPEG
input
data
DE–
FORMATTER
quantized
samples
CONTROL
SCALE
FACTOR
ARRAY
& ALLOCATION
MULTIPLY
DEQUANTIZATION
OUTPUT
CONTROL
sub-band
samples
MLB129
Fig.5 Decoding mode.
August 1993
6
SUB-BAND
FILTER
base band
samples
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
After sync and coding information, allocation data and the
scale factors are used to correctly fill the scale factor array.
FUNCTIONAL DESCRIPTION
Coding System
This is followed by a process of multiplication to provide
de-quantization and de-scaling of the samples.
The decoded sub-band samples, which are represented in
24-bit two's complement notation, are processed by the
sub-band filters and reconstituted into a single digital audio
signal.
MPEG coding achieves highly efficient digital encoding of
audio signals by using an algorithm based on the
characteristics of the human auditory system.
The broad-band audio signal is split into 32 sub-band
signals during encoding. For each of the sub-band signals
the masking threshold is calculated. The samples of the
sub-bands are incorporated in the signal with an accuracy
that is determined by the signal to masking threshold ratio
for that sub-band.
RESET
Reset must be active under the following conditions:
1. From system power-up until CLK24 has executed
more than 24 clock cycles.
During decoding, the sub-band signals are reconstructed
and combined into a broadband audio signal. The
integrated filter processor performs the splitting (encoding)
and joining (decoding) including the corresponding
formatting functions.
2. From the falling edge of PWRDWN for a period
equivalent to 24 cycles of CLK24 + oscillator start-up
time. This is typically >1 ms, however, this value is
crystal dependent.
For encoding, a SAA2521 is necessary to calculate the
masking threshold and required accuracy of the sub-band
samples.
PWRDWN
A HIGH input applied to this pin will halt all internally
generated clock signals. As a result, chip activity will halt
completely with outputs frozen in the state which was
current at the time of PWRDWN activation.
Encoding (See Fig.4)
An encoding algorithm table is used during the coding
process but, due to the Adaptive Allocation functions of the
SAA2521, this may change with every frame. The table is
therefore calculated for each frame by the SAA2521 and
then transferred to the SAA2520.
The bi-directional outputs: LTDATA, FDAC, FDAF, SDA,
SBWS, SBCL and SBDA will be 3-stated.
A frame contains 2 × 384 samples of Left and Right audio
data. This results in 12 samples per sub-band
(32 sub-bands). The samples of the greatest amplitude are
used to determine the scale factor for a given sub-band.
All samples are then scaled to represent a fraction of the
greatest amplitude.
Crystal Oscillators
A 24.576 MHz crystal together with some external
components form the 24.576 MHz oscillator (pins 42 and
43). Similarly a 22.5792 MHz oscillator (pins 40 and 41) is
formed by similar peripheral components together with an
appropriate crystal (see Fig.6).
Once scaled, the samples are quantized to reduce the
number of bits to correspond with the allocation table as
calculated by the SAA2521. Synchronization and coding
information data is then added to result in a fully encoded
MPEG signal.
The component values shown apply only to crystals from
the Philips 4322 156 series which exhibit an equivalent
series resistance of ≤ 40 Ω.
Decoding (See Fig.5)
All essential information (synchronization, system
information, scale factors and encoded sub-band
samples) are conveyed by incoming data. Decoding is
repeated for every frame.
August 1993
SAA2520
7
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
handbook, full pagewidth
C2 33 pF
X22IN
22.5792
MHz
X1
C1 33 pF
X22OUT
R2 1 kΩ
24.576
MHz
X2
C3 33 pF
40
R1
1 MΩ
X24IN
41
42
SAA2520
R4
1 MΩ
X24OUT
43
R3 1 kΩ
C4 33 pF
MLB130
Component values apply only to crystals from the Philips 4322 156 series.
Fig.6 Crystal oscillator components.
channel
right
left 32 bits
SWS
SCL
18 bits
1
13 bits
SDA
bit :
1
7
MSB
1
6
1
5
1
4
0
2
0
1
0
0
1
7
LSB
1
6
MSB
Fig.7 Transfer of SDA data (Standard I2S default format).
August 1993
8
1
5
1
4
MLA923 - 2
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
channel
SAA2520
right
left 32 bits
SWS
SCL
18 bits
14 bits
SDA
bit :
1
7
1
6
1
5
1
4
1
3
0
2
0
1
MSB
0
0
1
7
LSB
1
6
1
5
1
4
MLA924 - 2
MSB
Fig.8 Transfer of SDA data (alternative format).
channel
right
left 32 bits
SWS
SCL
1
7 bits
FDAC/
FDAF
bit :
2
3
MSB
2
2
2
1
2
0
0
2
0
1
2
3
0
0
LSB
MSB
Fig.9 Transfer of FDAF and FDAC (filtered) data.
August 1993
9
2
2
2
1
2
0
MLA925 - 2
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
channel
L
R
L
R
L
R
L
SAA2520
R
L
R
L
R
L
R
SWS
FSYNC
sub-band
31
0
1
31
0
1
MBC148 - 1
Fig.10 SWS related to phase of FSYNC.
Baseband Interface Signals
The interface between the SAA2520 and the baseband input/output circuitry consists of the following signals:
SWS
bi-directional
word (channel) select
FS
SCL
bi-directional
bit clock
64FS
SDA
bi-directional
baseband data
FDIR
output
decoding mode (direction control)
The SWS signal indicates the channel of the sample signal
(either LEFT or RIGHT) and is equal to the sampling
frequency FS.
Operating at a frequency of 64 times that is used for
sampling, the bit clock dictates that each SWS period
contains 64 SDA data bits. Of these, a maximum of 36 are
used to transfer data (samples may have a length up to
18-bits). Samples are transferred most significant bit first.
Both SWS and SDA change state at the negative edge of
SCL.
This baseband data is transferred between the SAA2520
and the input/output using either Standard I2S (default) or
the alternative format shown in Fig.8.
August 1993
10
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
Interface between SAA2520 and SAA2521 consists of the following signals:
FILTERED-I2S INTERFACE
SWS
bi-directional
word select (common to I2S)
SCL
bi-directional
bit clock (common to
FDAC
bi-directional
codec data
FDAF
bi-directional
filter data
FSYNC
output
synchronization
MPEG CODED INTERFACE
The interface that carries the MPEG coded signal uses
the following signals:
The MPEG I2S interface
The frequency of the SWS signal is equal to the sample
frequency FS and the bit clock SCL is 64 times the sample
frequency. Each period of SWS contains 64 data-bits, 48
of which are used to transfer data. The half period in which
SWS is LOW is used to transfer the information of the
LEFT channel while the following half period during which
SWS is HIGH carries the data of the RIGHT channel.
The 24-bit samples are transferred most significant bit first.
This bit is transferred in the bit clock period with a 1-bit
delay following the change in SWS. Both SWS and
FDAF/FDAC change state at the negative edge of SCL.
The operation of SAA2521 and the input/output circuitry is
controlled by three signals shown in Table 1.
FRESET and SYNCDAI are given whenever:
FS256 frequency is changed
(12.288/11.2896/8.192 MHz)
−
FDIR is switching
−
bit rate is changing
−
system reset is active
August 1993
bi-directional
word selection
SBCL
bi-directional
bit clock
SBDA
bi-directional
sub-band coded data
SBEF
input
error signal
SBDIR
input
direction of data flow
URDA
input
unreliable encoded data signal
The SBMCLK signal is the main frequency from which
other clock signals are derived. In encode mode this
division is performed internally. In decode mode the
external source should provide SBWS and SBCL.
The frequency of the signal is equal to 1/32nd of the bit
rate. The frequency of the bit clock SBCL is twice that of
the bit rate. Some examples of the frequencies are given
in Table 2.
SAA2521 AND INPUT/OUTPUT MODE CONTROL
−
SBWS
Operation is further controlled by:
The SAA2521 may be synchronized to the sub-band
codec using the FSYNC signal, which defines the SWS
period in which the samples of sub-band 0 (containing the
lowest frequency components) are transferred
(see Fig.10).
FS256, SCL and SWS outputs switch between
high and low impedance
64FS
FS/32
Filtered data is transferred between SAA2520 filter/codec
functions and the SAA2521 using the format shown in
Fig.9.
−
FS
I2S)
11
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
Table 1
SAA2520
SAA2521 input/output control.
FRESET
output
request a general reset of SAA2521
FDIR
output
'1' for decoding and '0' for encoding mode (common to I2S)
SYNCDAI
output
pulse for synchronization of digital input/output (TDA1315)
Table 2
Frequency examples.
BIT RATE
(k BITS/s)
SBWS FREQUENCY
(kHz)
SBCL FREQUENCY
(kHz)
384
12
768
256
8
512
192
6
384
128
4
256
ENCODE MODE
The following modes are supported:
Stereo or 2-channel mono with allowable bit rates of 384,
256, 192 and 128 kbits/s; audio sampling frequencies of
48, 44.1 and 32 kHz.
DECODE MODE
The following modes are supported:
Stereo and joint stereo, 2-channel mono and 1-channel
mono with allowable bit rates in the range 448 to 32 k
bits/s; audio sampling frequencies of 48, 44.1 and 32 kHz.
August 1993
12
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
T
t fH
t fL
FS256
t d1
t d1
SCL
t sL
t sH
Tc
t h2
SWS, SDA, FDAF, FDAC, FSYNC
output
t d2
t su
t h1
SDA, FDAF, FDAC
input
MEA642 - 3
Fig.11 Filtered I2S interface timing (master mode - FS256, SCL and SWS are input).
Notes to Fig.11
T
FS256 cycle time (fs = 48 kHz)
FS256 cycle time (fs = 44.1 kHz)
FS256 cycle time (fs = 32 kHz)
81.4 ns nominal
88.6 ns nominal
122.1 ns nominal
Tc
SCL cycle time
4T ns nominal
tfH
FS256 HIGH time (fs = 48 kHz)
FS256 HIGH time (fs = 44.1 kHz)
FS256 HIGH time (fs = 32 kHz)
≥ 35 ns
≥ 38 ns
≥ 35 ns
tfL
FS256 LOW time (fs = 48 kHz)
FS256 LOW time (fs = 44.1 kHz)
FS256 LOW time (fs = 32 kHz)
≥ 35 ns
≥ 38 ns
≥ 75 ns
tSH
SCL HIGH time
≥ 2T - 20 ns
tSL
SCL LOW time
≥ 2T - 20 ns
tS
SDA, FDAF, FDAC input set-up
before FS256 HIGH
≥ 20 ns
tH1
SDA, FDAF, FDAC input hold
after FS256 HIGH
≥ 30 ns
tH2
SDA, FDAF, FDAC output hold
after FS256 HIGH
≤ 0 ns
tD1, 2
FS256 HIGH to SCL, SWS,
SDA, FDAF, FDAC output valid
≤ 50 ns
August 1993
13
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
t fL
T
SAA2520
t fH
FS256
t sH
Tc
t sL
SCL
t h1
td
SDA, FDAF, FDAC, FSYNC
output
t su
t h2
SWS, SDA, FDAF, FDAC
input
MEA644 - 3
Fig.12 Filtered I2S interface timing (slave mode - FS256, SCL and SWS are input).
Notes to Fig.12
tfH
FS256 HIGH time
≥ 35 ns
tfL
FS256 LOW time
≥ 35 ns
tsH
SCL HIGH time
≥ T + 35 ns
tsL
SCL LOW time
≥ T + 35 ns
tH1
SDA, FDAF, FDAC output
hold after SCL HIGH
≥ 2T - 15 ns
tD
SCL HIGH to SDA, FDAF
FDAC output valid
≤ 3T + 60 ns
ts
SDA, FDAF, FDAC input
valid after SCL HIGH
≥ 20 ns
tH2
SDA, FDAF, FDAC input
hold after SCL HIGH
≥ T + 20 ns
August 1993
14
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
,
SAA2520
FRESET
t d0
t sH
t d1
SYNCDAI
t d2
t d3
FDIR
t d4
t d5
HIGH Z
SDA
HIGH Z
SDA
t d6
HIGH Z SWS
FS256
SCL
FDAF
FDAC
HIGH Z
t d8
t d7
HIGH Z
FDAF
FDAC
t d9
HIGH Z
Fig.13 Mode switch timing.
Notes to Fig.13
tDO
FRESET HIGH to SYNCDAI HIGH
≥ 300 ns
tSH
SYNCDAI HIGH time
≥ 1280 ns
tD1
SYNCDAI LOW to FRESET LOW
≥ 790 ns
tD2
FDIR hold to FRESET HIGH
≤ 20 ns
tD3
FRESET HIGH to FDIR valid
≤ 20 ns
tD4
SDA change to high impedance
after FRESET HIGH
tD5
SDA remains high impedance
after FRESET LOW
tD6
FDAF, FDAC change to high impedance
after FRESET HIGH
tD7
FDAF, FDAC remain high impedance
August 1993
≥ 0 ns
≤ 170 ns
≥ 0 ns
≤ 170 ns
≤ 20 ns
15
SWS
FS256
SCL
MEA646 - 1
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
Notes to Fig.13
after FRESET HIGH
≥ 460 ns
tD8
FS256, SWS, SCL change
to high impedance before SYNCDAI HIGH
≥ 140 ns
tD9
FS256, SWS, SCL remain HIGH
impedance after SYNCDAI HIGH
≥ 140 ns
32 bits
SBWS
SBCL
1
15 bits
1
SBDA
bit :
0 0
0 1
MSB
0
2
0
3
1
0
1
1
1
2
1
3
1
4
1
5
LSB
1 1
6 7
MSB
1
8
1
9
2
0
2
1
2
2
SBEF
byte 0
byte 1
byte 2
MEA649 - 2
Fig.14 Transferring MPEG data to and from the SAA2520.
August 1993
16
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
MPEG Coded Interface (Sub-band I2S)
Encoding mode
The MPEG coded data is transferred to and from the
SAA2520 using the format shown in Fig.14.
SBCL, SBWS and SBDA are generated by the SAA2520.
However, if the SBDIR signal is logic 1, the output buffers
are not enabled and these signals do not appear on the
pins. This mode is available to permit a change of
operating mode whilst the bus signals are driven from an
external source.
Each period of SBWS contains 64 data bits, 32 of which
are used to convey data. The half-period during which
SBWS is logic 0 is used to transfer the first 16-bits (0 to 15)
of a sub-band slot. The remaining half-period during which
SBWS is logic 1 carries the remaining 16-bits (16 to 31).
Thus one period of SBWS corresponds with one slot of the
sub-band signal.
Decoding mode
SBCL, SBWS and SBDA are generated by an external
source.
Bits 0 and 16 are transferred in the bit clock period, one
bit-time after the change in SBWS. Both SBWS and SBDA
change state during the negative edge of SBCL.
Table 3 contains a summary of the source signals in the
various modes.
In decode mode a byte error flag SBEF is also transferred.
This occurs approximately in the middle of the
corresponding byte (byte 0 = bits 0 to 7,
byte 1 = bits 8 to 15 etc).
Table 3
Modes and source signals.
source of:
Mode
FDIR
SBDIR
SBWS
SBCL
SBDA
SBEF
SBMCLK
Encode
0
0
INT
INT
INT
----
INT
note 1
Encode
0
1
EXT
EXT
EXT
----
INT
note 2
Decode
1
0
INT
INT
INT
EXT
INT
note 3
Decode
1
1
EXT
EXT
EXT
EXT
INT
Notes
1. During encoding the SBEF signal is ‘don’t care’.
2. Incoming data is not decoded. The SAA2520 operates in the encoding mode and the data does not enter the
interface.
3. Operation is undefined. The SAA2520 is in decoding mode whilst the SBWS, SBCL and SBDA output drivers are
enabled.
August 1993
17
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
t mL
Ts
SAA2520
t mH
SBMCLK
Tsc
t cL
t cH
SBCL
t d1
t d2
SBWS
SBDA
MEA645 - 2
Fig.15 Sub-band I2S interface timing (master mode - SBCL, SBWS and SBDA are output).
Notes to Fig.15
T
SBMCLK cycle time
120 to 205 ns (163 ns nominal)
tmH
SBMCLK HIGH time
≥ 35 ns
tmL
SBMCLK LOW time
≥ 75 ns
Tc
SBCL cycle time (384 kB/s)
SBCL cycle time (256 kB/s)
SBCL cycle time (192 kB/s)
SBCL cycle time (128 kB/s)
8T ns nominal
12T ns nominal
16T ns nominal
24T ns nominal
tCH
SBCL HIGH time (384 kB/s)
SBCL HIGH time (256 kB/s)
SBCL HIGH time (192 kB/s)
SBCL HIGH time (128 kB/s)
≥ 4T - 20 ns
≥ 6T - 20 ns
≥ 8T - 20 ns
≥ 12T - 20 ns
tCL
SBCL LOW time (384 kB/s)
SBCL LOW time (256 k/Bs)
SBCL LOW time (192 kB/s)
SBCL LOW time (128 kB/s)
≥ 4T - 20 ns
≥ 6T - 20 ns
≥ 8T - 20 ns
≥ 12T - 20 ns
tD1
SBWS, SBDA hold to SBCL LOW
≤ 20 ns
tD2
SBCL LOW to SBWS, SBDA valid
≤ 20 ns
August 1993
18
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
,,
,,,
,
,
SBWS
SBCL
SBDA
0
1
2
3
4
5
6
7
8
9
10
11
12
SBEF
t cL
Tsc
t cH
SBCL
t su1
t h1
SBWS
SBDA
t h2
SBEF
MEA648 - 2
t su2
Fig.16 Sub-band I2S interface timing (slave mode - SBCL, SBWS and SBDA are input).
Notes to Fig.16
TC
SBCL cycle time (see note 1)
6.86T to 96T ns (8T ns nominal)
tCH
SBCL HIGH time
≥ T + 30 ns
tCL
SBCL LOW time
≥ T + 30 ns
tS1
SBWS, SBDA input set-up
before SBCL HIGH
≥ T + 30 ns
tH1
SBWS, SBDA input hold
after SBCL HIGH
≥ 30 ns
tS2
SBCL HIGH to SBEF valid
≤ T - 30 ns
tH2
SBEF hold after SBCL HIGH
Note 1:
Minimum at bit rate = 448 kB/s
Nominal at bit rate = 384 kB/s
Maximum at bit rate = 32 kB/s
≥ 2T- 30 ns
August 1993
19
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
SBDIR
t d1
t d2
HIGH Z
SBCL
SBWS
SBDA
HIGH Z
MEA647 - 1
Fig.17 Sub-band I2S mode switch timing.
Notes to Fig.17
tD1
SBDIR HIGH to SBCL, SBWS, SBDA high impedance
≤ 50 ns
tD2
SBCL, SBWS, SBDA after SBDIR LOW high impedance
≥ 240 ns
August 1993
20
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
Microcontroller interface
The SAA2520 has an interface connection to the serial interface of a microcontroller. The following signals are used:
LTCLK
input
bit clock
LTDATA
bi-directional
serial data
LTCNT0
input
control line 0
LTCNT1
input
control line 1
LTENA
input
enable
The SAA2520 microcontroller interface is enabled only if
LTENA (pin 34) is logic 1. Information to or from the
SAA2520 is conveyed in serial 8 or 16-bit units, whilst the
type of information is controlled by LTCNT0 (pin 35) and
LTCNT1 (pin 36).
During the transfer of 8-bit units, the least significant bit is
first to be transferred. When 16-bit units are transferred the
most significant byte is sent first.
A transfer commences when the microcontroller sets the
control lines to the correct combination for the required
action. LTENA is set to logic 1. The SAA2520 determines
its required action and prepares to transfer data. When the
microcontroller supplies the LTCLK, data is transferred to
or from the SAA2520 in units of 8-bits. 16-bit transfers are
conveyed as two 8-bit units during which LTENA remains
high.
Four information bits together with four address bits are
transferred in this mode. The order in which the bits appear
on the interface is:
Table 4
EXTENDED SETTINGS (LTCNT1 = 0, LTCNT0 = 0)
D0..D1..D2..D3..A0..A1..A2..A3
Extended Settings.
BIT
A3
BIT
A2
BIT
A1
BIT
A0
0
0
0
0
CODEC external settings (see Table 5)
0
0
0
1
FILTER settings (see note 1)
0
0
1
0
not used
..
..
..
..
..
1
1
1
1
not used
Table 5
DESCRIPTION
Extended Settings.
BIT
DESIGNATION
DEFAULT
FUNCTION
D0
MUTEDAC
1
connected to DAC mute input
D1
ATTDAC
0
connected to DAC attenuation input
D2
DEEMDAC
0
emphasis control for DAC circuit
D3
HOLDCLKOK
0
selects CLKOK hold mode
Note
If not used for DAC control, the MUTEDAC, ATTDAC and DEEMDAC can be used as general purpose output expanders.
August 1993
21
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
Bits D0 to D3 are copied directly to the corresponding
output pins/mode flip-flop.
SAA2520
This is then followed by the scale factor information.
In the event that only internal settings information is sent,
then a default allocation of logic 0 will be assigned to all
sub-bands. If in addition no internal settings are sent then
the previous settings remain valid.
For HOLDCLKOK = logic 1. When CLKOK drops it will
remain LOW until set by an encode/decode mode, sample
frequency, external 256FS or bit rate index change.
The allocation information is transferred in 4-bit units.
Each of these units contains the number of bits allocated
to the sub-band, MINUS 1, except in the case of a logic 0
value, which indicates that no bits are allocated to that
sub-band.
Note 1.
When D0 = logic 1 (default) I2S mode is selected. For D0
= logic 0 the alternative mode is selected. The setting of D0
remains dormant until activated by the occurrence of
FRESET.
Scale factor information is transferred in units of 8-bits,
containing the 6-bit scale factor which is extended to 8-bits
by adding two logic 0’s at the most significant end.
ALLOCATION/SCALE FACTOR INFORMATION (LTCNT1 =
0, LTCNT0 = LOGIC 1)
LOGIC
In the case of stereo encoding the channels are indicated
by L (left) and R (right). This changes to I and II in the case
of 2 channel mono encoding.
For encoding, the allocation and scale factor arrays can be
filled using this mode. To completely fill the allocation array
16 complete transfers of 16-bits are required. After the first
transfer of allocation information a check must be made to
determine when the SAA2520 is ready to receive the
remaining information. This will ensure synchronization
with the internal program of the SAA2520. Transfer of the
allocation information is completed by sending the internal
settings.
Table 6
Allocation information format.
msb
bits
B13
-
channel
sub-band
B12
L or I
0 .. 30 (even)
B15
-
B14
B11
-
B10
-
B9
-
B8
R or II
0 .. 30 (even)
B7
-
B6
-
B5
-
B4
L or I
1 .. 31 (odd)
B3
-
B2
-
B1
-
B0
R or II
1 .. 31 (odd)
Table 7
-
lsb
Scalefactor information format.
msb
bits
lsb
channel
B15
................
B8
L or I
0 .. 31
B7
................
B0
R or II
0 .. 31
August 1993
22
sub-band
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
INTERNAL SETTINGS (LTCNT1 = LOGIC 1, LTNCT 0 = LOGIC 0)
The operation of the codec is controlled by the bits transferred in this mode.
Table 8
Internal Settings (LTCNT1 = logic 1, LTNCT 0 = logic 0).
msb
lsb
name
function
valid in
S15
...
S12
bit rate index
bit rate indication
encode
S11
...
S10
sample frequency
44.1, 48 or 32 kHz
indication
encode
S9
decode
1 = decode; 0 = encode
encode/decode
S8
EXT 256FS
1 = external;
0 = internal 256FS
encode/decode
S7
2-channel mono
1 = 2-CH mono;
0 = stereo
encode
S6
MUTE
1 = mute; 0 = no mute
encode/decode
S5
not used
S4
CH1
1 = CH1; 0 = CH2
decode
S3
...
S2
Tr0 to Tr1
transparent bits
encode
S1
...
S0
EMPHASIS
emphasis indication
encode
Table 9
Internal Settings (LTCNT1 = logic 1, LTCNT0 = logic 0).
msb
lsb
bit rate
1
1
0
0
384 kbits/s
1
0
0
0
256 kbits/s
0
1
1
0
192 kbits/s
0
1
0
0
128 kbits/s
CH1 is utilized in the decoding mode to select one of the
2-channel mono signals to be decoded (default is
I - channel 1). A value of 0 results in channel 2 being
decoded).
The bit rate index indicates the bit rate of the encoded
signal and is only effective in the encode mode.
The decode bit determines the operation mode of the
SAA2520. The default value is logic 1 (decoding mode).
The transparent bits are copied in the sub-band signal,
default is 00.
EXT 256FS in the encoding mode determines whether or
not the SAA2520 is master or slave of the Filtered-I2S
interface (default is logic 0, master mode).
The information from S15 to S10, S7 and S3 to S0 will be
copied into the sub-band signal.
2CH MONO is used in the encoding mode to determine
whether the sub-band signal is generated as a stereo or
2-channel mono signal. Default value is logic 0.
MUTE is used in both the encoding and decoding modes
to mute the information to or from the Filtered-I2S interface
(the default value is logic 0).
August 1993
default value
23
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
Table 10 Sample frequency indication.
msb
lsb
sample frequency
0
0
44.1 kHz
0
1
48 kHz
1
0
32 kHz
1
1
not used
default value
Table 11 EMPHASIS indication.
msb
lsb
emphasis
0
0
no emphasis
0
1
50/15 µs
1
0
reserved
1
1
CCITT J.17
default value
Before sending internal settings the microcontroller should
check whether or not the SAA2520 is ready-to-receive.
However, this does not apply for the transfer of internal
settings to end a transfer of allocation information.
STATUS (LTCNT = LOGIC 1, LTNCT0 = LOGIC 1)
Table 12 Status information 16-bit units.
msb
lsb
name
function
valid in
T15
...
T12
bit rate index
bit rate indication
encode/decode
T11
...
T10
sample frequency
44.1, 48 or 32 kHz indication
encode/decode
ready-to-receive
1 = ready; 0 = not ready
encode/decode
MODE
sub-band signal mode indication
encode/decode
SYNC
synchronization indication
decode
T9
T8
not used
T7
T6
T5
T4
T3
T1
...
CLKOK
1 = o.k.; 0 = not o.k.
encode/decode
T2
Tr0 to Tr1
transparent bits
encode/decode
T0
EMPHASIS
emphasis indication
encode/decode
The bit rate index indicates the bit rate of the sub-band
signal in units of 32 kbits/s. bit rate index 0000 indicates
the ‘free format’ condition. bit rate 1111 is illegal and
should not be found.
The coding of the sample frequency indication is equal
to the one in the internal settings.
August 1993
24
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
Table 13 MODE identification.
msb
lsb
mode
0
0
stereo
L and R
0
1
joint stereo
L and R
1
0
2 channel mono
I or II as selected
1
1
1 channel mono
mono; no selection
Ready-to-receive indicates whether the SAA2520 is ready
to receive allocation, scale factor or internal setting
transfers. This should be checked in order to synchronize
the transfer of such information.
In 2 channel mono decode mode the selected samples are
transferred to both output channels. The same occurs with
all samples in 1-channel mono decode mode. In both of
these instances the L and R filter output channels are
identical.
In decode mode the SYNC bit is logic 0 when the SAA2520
is unable to decode the sub-band frames. This will occur in
the following situations:
• with the loss of synchronisation
• when in correct allocation information is received for two
or more subsequent frames (SBEF was HIGH).
• when the URDA input pin is HIGH
In these situations the SAA2520 data output will be muted.
The SYNC bit will return to logic 1 as soon as the decoder
is resynchronized to the incoming sub-band data.
CLKOK indicates whether the 256FS clock corresponds to
specified sample frequency. The CLKOK bit is set to logic
1 after a change in sample frequency, operation mode or
EXT256FS setting. It drops to logic 0 as soon as the
256FS clock deviates from the nominal frequency by more
than approximately 0.2%. Return to logic 1 will only occur
automatically when the extended setting
CLKOK-hold-mode is logic 0.
The transparent bits are copied from the MPEG coded
signal.
The EMPHASIS indication is as defined in the internal
settings. It can be used to apply the correct de-emphasis.
Note: the two bytes of the status are 'sampled' at different
moments so the information may not result from the same
sub-band frame.
August 1993
25
output
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
handbook, full pagewidth
LTENA
LTCNT0/1
LTCLK
LTDATA
0
lsb
1
2
3
4
5
6
7
msb
MLB131
Fig.18 Transfer of data on SAA2520 microcontroller interface.
handbook, full pagewidth
LTENA
16 bits allocation / scale factor information
16 bits allocation / scale factor information
LTCLKC
MLB132
Fig.19 The LTENA line must return to logic 0 between information transfers.
August 1993
26
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
handbook, full pagewidth
LTENA
LTCNT0/1
LTCLK
LTDATA
8
bit :
10
9
12
11
14
13
0
2
15
4
1
6
3
MLB133
5
7
Fig.20 Order of settings and status bits on the SAA2520 microcontroller interface.
handbook, full LTENA
pagewidth
LTENA must remain HIGH
LTCNT0/1
LTCLK
t D6
LTDATA
OUTPUT
8
9 10 11 12 13 14 15
0
tD6 delay LTCLK HIGH to LTDATA valid output for bit 0 in 16-bit transfers
Fig.21 16-bit transfers.
August 1993
27
1
2
3
4
5
6
7
MLB134
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
t el
handbook, full pagewidth
LTENA
t S1
t D1
t H1
LTCNT0
LTCNT1
t D5
t CL
t CH
t H4
LTCLK
t S2
t H2
t D4
LTCDATA
input
t D2
LTCDATA
output
t D3
t H3
t D6
hiZ
hiZ
MLB135
Fig.22 Microcontroller interface timing.
Notes to Fig.22
teL
LTENA LOW time
≥ 190 ns
tCH
LTCLK HIGH time
≥ 190 ns
tCL
LTCLK LOW time
≥ 190 ns
tD1
LTENA HIGH to LTCLK HIGH
≥ 190 ns
tD2
LTENA HIGH to LTDATA
output low impedance
≥ 0 ns
tD3
LTENA HIGH to LTDATA output valid
≤ 380 ns
tD4
LTENA LOW to LTDATA high impedance
≤ 50 ns
tH4
LTENA hold after LTCLK HIGH
≥ 355 ns
tD5
LTCLK HIGH to LTENA HIGH
≥ 190 ns
tD6
LTCLK HIGH to LTDATA output valid
for bit 0 (see Fig.21)
for first bit in the second 8-bit unit
≤ 355 ns
≤ 520 ns
tS1
LTCNT0/1 set-up before LTENA HIGH
≥ 190 ns
tH1
LTCNT0/1 hold after LTENA HIGH
≥ 190 ns
tS2
LTDATA set-up before LTCLK HIGH
≥ 190 ns
tH2
LTDATA input hold after LTCLK HIGH
≥ 30 ns
tH3
LTDATA output hold after LTCLK HIGH
≥ 145 ns
tH4
LTENA hold after LTCLK HIGH
≥ 355 ns
August 1993
28
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
LIMITING VALUES
In accordance with the Absolute Maximum System (IEC 134).
SYMBOL
PARAMETER
CONDITIONS
MIN.
MAX.
UNIT
−0.5
6.5
V
−0.5
VDD + 0.5
V
supply current from VSS
−
160
mA
IDD
supply current in VDD
−
160
mA
II
input current
−10
10
mA
IO
output current
−20
20
mA
Ptot
total power dissipation
−
880
mW
Tstg
storage temperature range
−55
150
°C
Tamb
operating ambient temperature range
- 40
85
°C
Ves1
electrostatic handling
note 2
−1500
1500
V
Ves2
electrostatic handling
note 3
−70
70
V
VDD
supply voltage
VI
input voltage
ISS
note 1
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 kΩ series resistor
3. Equivalent to discharging a 200 pF capacitor through a 0 Ω series resistor.
DC CHARACTERISTICS
Tamb = −40 to 85 °C; VDD = 3.8 to 5.5 V unless otherwise specified.
SYMBOL
PARAMETER
CONDITIONS
MIN.
TYP.
MAX.
UNIT
Supply
VDD
supply voltage range
IDD
operating current
IDD
operating current
3.8
5.0
5.5
V
VDD = 5 V (note 1)
−
82
110
mA
VDD = 3.8 V (note 1)
−
58
80
mA
Inputs URDA, SBDIR, SBEF, LTCLK, LTCNT0, LTNCT1, X22IN, X24IN
VIH
HIGH level input voltage
0.7VDD
−
−
V
VIL
LOW level input voltage
−
−
0.3VDD
V
−II
input current
Vi = 0 V;
Tamb = 25 °C
−
−
10
µA
+II
input current
Vi = 5.5 V;
Tamb = 25 °C
−
−
10
µA
Inputs PWRDWN, LTENA
VIH
HIGH level input voltage
0.7VDD
−
−
V
VIL
LOW level input voltage
−
−
0.3VDD
V
+II
input current
40
−
250
µA
August 1993
Vi = VDD;
Tamb = 25 °C
29
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SYMBOL
PARAMETER
SAA2520
CONDITIONS
MIN.
TYP.
MAX.
UNIT
Input RESET
Vtlh
positive-going threshold
−
−
0.8VDD
V
Vthl
negative-going threshold
0.2VDD
−
−
V
Vhys
hystersis
(Vtlh - Vthl)
−
1.5
−
V
+II
input current
Vi = VDD;
Tamb = 25 °C
40
−
250
µA
Outputs MUTEDAC, DEEMDAC, ATTDAC, SYNCDAI, FDIR, FRESET, FSYNC, CLK22
VOH
HIGH level output voltage
+Io = 2 mA
VDD−0.5
−
−
V
VOL
LOW level output voltage
−Io = 2 mA
−
−
0.4
V
Outputs CLK24
VOH
HIGH level output voltage
+Io = 8 mA
VDD−0.5
−
−
V
VOL
LOW level output voltage
−Io = 8 mA
−
−
0.4
V
Inputs/outputs SBDA, SBCL, SBWS, FDAF, FDAC, SCL, SWS, SDA, LTDATA
VOH
HIGH level output voltage
+Io = 2 mA
VDD−0.5
−
−
V
VOL
LOW level output voltage
−Io = 2 mA
−
−
0.4
V
−
V
Outputs SBDA, SBCL, SBWS, FDAF, FDAC, SCL, SWS, SDA, LTDATA in 3-state
VIH
HIGH level input voltage
VIL
LOW level input voltage
II
input current
0.7VDD
Vi = VDD;
Tamb = 25 °C
−
−
−
0.3VDD
V
40
−
250
µA
Input/output SBMCLK
VOH
HIGH level output voltage
+Io = 8 mA
VDD−0.5
−
−
V
VOL
LOW level output voltage
−Io = 8 mA
−
−
0.4
V
Output SBMCLK in 3-state
VIH
HIGH level input voltage
0.7VDD
−
−
V
VIL
LOW level input voltage
−
−
0.3VDD
V
II
input current
Vi = VDD;
Tamb = 25 °C
40
−
250
µA
Input/output FS256
VOH
HIGH level output voltage
+Io = 12 mA
VDD−0.5
−
−
V
VOL
LOW level output voltage
−Io = 12 mA
−
−
0.4
V
Output FS256 in 3-state
VIH
HIGH level input voltage
0.7VDD
−
−
V
VIL
LOW level input voltage
−
−
0.3VDD
V
II
input current
40
−
250
µA
Vi = VDD;
Tamb = 25 °C
Note
1. For load impedances representative of the application.
August 1993
30
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
AC CHARACTERISTICS
Tamb = −40 to 85 °C; VDD = 3.8 to 5.5 V unless otherwise specified.
SYMBOL
PARAMETER
CONDITIONS
MIN.
TYP.
MAX.
UNIT
Inputs
CI
input capacitance
−
−
10
pF
X24IN and X22IN
f
crystal frequency at X22OUT,
CLK22
note 1
21
22.5792
24
MHz
f
crystal frequency at X24OUT,
CLK24
note 1
23
24.576
26
MHz
gm
mutual conductance
100 kHz
1.5
−
−
mA/V
Av
small signal gain
Av = gm.Ro
3.5
−
−
V/V
Cfb
feedback capacitance
−
−
5
pF
CO
output capacitance
−
−
10
pF
output capacitance
−
−
10
pF
Outputs
CO
Inputs URDA, RESET, LTDATA, LTCLK, LTENA, LTCNT0, LTCNT1
tSU
setup time to X24IN
15
−
−
ns
tHD
hold time to X24IN
60
−
−
ns
Outputs LTDATA, MUTEDAC, DEEMDAC, ATTDAC, SYNCDAI, FDIR, FRESET
td
propagation delay from X24IN
−
−
80
ns
Inputs FDAF, FDAC, SDA, SCL, SWS
tSU
setup time to FS256
15
−
−
ns
tHD
hold time to FS256
25
−
−
ns
−
−
50
ns
Outputs FDAF, FDAC, SDA, SCL, SWS, FSYNC
td
propagation delay from FS256
Inputs SBDA, SBCL, SBWS, URDA, SBDIR, SBEF
tSU
setup time to SBMCLK
15
−
−
ns
tHD
hold time to SBMCLK
25
−
−
ns
−
−
50
ns
Outputs SBDA, SBCL, SBWS
td
propagation delay from SBMCLK
FS256
T
FS256 cycle time
fs = 48 kHz
−
81.4
−
ns
T
FS256 cycle time
fs = 44.1 kHz
−
88.6
−
ns
T
FS256 cycle time
fs = 32 kHz
−
122.1
−
ns
TC
SCL cycle time
−
4T
−
ns
August 1993
31
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SYMBOL
PARAMETER
SAA2520
CONDITIONS
MIN.
TYP.
MAX.
UNIT
FS256 master mode (FS256,SCL and SWS are output)
tfH
FS256 HIGH time
fs = 48 kHz
35
−
−
ns
tfH
FS256 HIGH time
fs = 44.1 kHz
38
−
−
ns
tfH
FS256 HIGH time
fs = 32 kHz
75
−
−
ns
tfL
FS256 LOW time
fs = 48 kHz
35
−
−
ns
tfL
FS256 LOW time
fs = 44.1 kHz
38
−
−
ns
tfL
FS256 LOW time
fs = 32 kHz
75
−
−
ns
tsH
SCL HIGH time
2T-20
−
−
ns
tsL
SCL LOW time
2T-20
−
−
ns
ts
SDA, FDAF, FDAC input setup
time before FS256 HIGH
20
−
−
ns
tH1
SDA, FDAF, FDAC input hold time
after FS256 HIGH
30
−
−
ns
tH2
SDA, FDAF, FDAC output hold
time after FS256 HIGH
0
−
−
ns
tD1,2
FS256 HIGH-to SCL, SWS, SDA,
FDAF, FDAC output valid
−
−
50
ns
FS256 slave mode (FS256, SCL and SWS are input)
tfH
FS256 HIGH time
35
−
−
ns
tfL
FS256 LOW time
35
−
−
ns
tsH
SCL HIGH time
T+35
−
−
ns
tsL
SCL LOW time
T+35
−
−
ns
tH1
SDA, FDAF, FDAC output hold
time after SCL HIGH
2T-15
−
−
ns
tD
SCL HIGH-to SDA, FDAF, FDAC
output valid
−
−
3T+60
ns
tS
SDA, FDAF, FDAC input valid
after SCL HIGH
20
−
−
ns
tH2
SDA, FDAF, FDAC input hold time
after SCL HIGH
T+20
−
−
ns
T
SBMCLK cycle time
120
163
205
ns
tmH
SBMCLK HIGH time
35
−
−
ns
tmL
SBMCLK LOW time
75
−
−
ns
SBMCLK
August 1993
32
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SYMBOL
PARAMETER
SAA2520
CONDITIONS
MIN.
TYP.
MAX.
UNIT
SBMCLK master mode (SBCL, SBWS and SBDA are output)
TC
SBCL cycle time
384 kB/s
−
8T
−
ns
TC
SBCL cycle time
256 kB/s
−
12T
−
ns
TC
SBCL cycle time
192 kB/s
−
16T
−
ns
TC
SBCL cycle time
128 kB/s
−
24T
−
ns
tcH
SBCL HIGH time
384 kB/s
4T − 20
−
−
ns
tcH
SBCL HIGH time
256 kB/s
6T − 20
−
−
ns
tcH
SBCL HIGH time
192 kB/s
8T − 20
−
−
ns
tcH
SBCL HIGH time
128 kB/s
12T − 20
−
−
ns
tcL
SBCL LOW time
384 kB/s
4T − 20
−
−
ns
tcL
SBCL LOW time
256 kB/s
6T − 20
−
−
ns
tcL
SBCL LOW time
192 kB/s
8T − 20
−
−
ns
tcL
SBCL LOW time
128 kB/s
12T − 20
−
−
ns
tD1
SBWS, SBDA hold
to SBCL LOW 20
−
−
ns
tD2
SBWS, SBDA valid
after SBCL 0
−
−
20
ns
6.86T
8T
96T
ns
SBMCLK slave mode (SBCL, SBWS and SBDA are input)
TC
SBCL cycle time
note 2
tcH
SBCL HIGH time
T + 30
−
−
ns
tcL
SBCL LOW time
T + 30
−
−
ns
tS1
SBWS, SBDA setup time
before SBCL
HIGH
T + 30
−
−
ns
tH1
SBWS, SBDA hold time
after SBCL
HIGH
30
−
−
ns
tS2
delay before SBEF valid
after SBCL
HIGH
−
−
T − 30
ns
tH2
SBEF hold time
after SBCL
HIGH
2T − 30
−
−
ns
Notes
1. % deviation from nominal frequency must be the same for X24, X22, and FS256 inputs to within 0.2%
2. Minimum value for bit rate = 448 kB/s
Typical value for bit rate = 384 kB/s
Maximum value for bit rate = 32 kB/s
August 1993
33
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
PACKAGE OUTLINE
QFP44: plastic quad flat package; 44 leads (lead length 2.35 mm); body 14 x 14 x 2.2 mm
SOT205-1
c
y
X
33
A
23
34
22
ZE
e
E HE
A
A2
(A 3)
A1
wM
θ
bp
Lp
pin 1 index
44
L
12
detail X
1
11
ZD
e
v M A
wM
bp
D
B
HD
v M B
0
5
10 mm
scale
DIMENSIONS (mm are the original dimensions)
UNIT
A
max.
A1
A2
A3
bp
c
D (1)
E (1)
e
HD
HE
L
Lp
v
w
y
mm
2.60
0.25
0.05
2.3
2.1
0.25
0.50
0.35
0.25
0.14
14.1
13.9
14.1
13.9
1
19.2
18.2
19.2
18.2
2.35
2.0
1.2
0.3
0.15
0.1
Z D (1) Z E (1)
2.4
1.8
2.4
1.8
θ
o
7
0o
Note
1. Plastic or metal protrusions of 0.25 mm maximum per side are not included.
REFERENCES
OUTLINE
VERSION
IEC
SOT205-1
133E01A
August 1993
JEDEC
EIAJ
EUROPEAN
PROJECTION
ISSUE DATE
95-02-04
97-08-01
34
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
If wave soldering cannot be avoided, the following
conditions must be observed:
SOLDERING
Introduction
• A double-wave (a turbulent wave with high upward
pressure followed by a smooth laminar wave)
soldering technique should be used.
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.
• 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).
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).
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.
Reflow 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).
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.
Repairing soldered joints
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.
Fix the component by first soldering two diagonallyopposite 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.
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.
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.
August 1993
SAA2520
35
Philips Semiconductors
Preliminary specification
Stereo filter and codec for MPEG layer 1
audio applications
SAA2520
DEFINITIONS
Data sheet status
Objective specification
This data sheet contains target or goal specifications for product development.
Preliminary specification
This data sheet contains preliminary data; supplementary data may be published later.
Product specification
This data sheet contains final product specifications.
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 specification
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 specification.
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.
August 1993
36