and 24 bits formats supported on digital I/O interface.
1.2Control
• Controlled by means of static pins or microcontroller
(L3-bus or I2C-bus) interface.
1.3IEC 60958 input
• On-chip amplifier for converting IEC 60958 input to
CMOS levels
• Supports level I, II and III timing
• Selectable IEC 60958 input channel, one of four
• Supports input frequencies from 28 to 96 kHz
• Lock indication signal available on pin LOCK
• 40 status bits can be read for left and right channel via
L3-bus or I2C-bus
• ChannelstatusbitsavailableviaL3-busorI2C-bus:lock,
pre-emphasis, audio sample frequency, two channel
Pulse Code Modulation (PCM) indication and clock
accuracy
• Pre-emphasis information of incoming IEC 60958
bitstream available in register
• Detection of digital data preamble, such as AC3,
available on pin in microcontroller mode.
1.4IEC 60958 output
• 32, 44.1 and 48 kHz output frequencies (including
double and half of these frequencies) supported in
microcontroller mode
• Via microcontroller, 40 status bits can be set for left and
right channel.
1.5Digital I/O interface
• Supports sampling frequencies from 16 to 100 kHz
• Supported static mode:
–I2S-bus format
– LSB-justified 16 and 24 bits format
– MSB-justified format.
• Supported microcontroller mode:
–I2S-bus format
– LSB-justified 16, 18, 20 or 24 bits format
– MSB-justified format.
• BCKand WS signals can be slave or master,depending
on application mode.
1.6ADC digital sound processing
• Supports sampling frequencies from 16 to 100 kHz
• Analogfront-end includes a 0 to +24 dB PGA in steps of
3 dB, selectable via microcontroller interface
• Digital independent left and right volume control of
+24 to −63.5 dB in steps of 0.5 dB via microcontroller
interface
• Bitstream ADC operating at 64f
• Comb filter decreases sample rate from 64fsto 8f
• Decimator filter (8fsto fs) made of a cascade of three
FIR half-band filters.
s
s
• CMOS output level converted to IEC 60958 output
signal
• Full-swing digital signal, with level II timing using crystal
oscillator clock
• 32, 44.1 and 48 kHz output frequencies supported in
static mode
2003 Apr 103
1.7DAC digital sound processing
• Digital de-emphasis for 32, 44.1, 48 and 96 kHz audio
sampling frequencies
• Automatic de-emphasis when using IEC 60958 to DAC
• Soft mute made of a cosine roll-off circuit selectable via
pin MUTE or L3-bus interface
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
• Programmable digital silence detector
• Interpolating filter (fsto 64fs or fsto 128fs) comprising a
recursive and a FIR filter in cascade
• Selectable fifth-order noise shaper operating at 64fs or
third-order noise shaper operating at 128fs(specially for
low sampling frequencies, e.g. 16 kHz) generating
bitstream for DAC
• Filter Stream DAC (FSDAC)
• In microcontroller mode:
– Left and right volume control (for balance control)
0to−78 dB and −∞
– Left and right bass boost and treble control
– Optional resonant bass boost control
– Mixing possibility of two data streams.
2GENERAL DESCRIPTION
The UDA1355H is a single-chip IEC 60958 decoder and
encoderwith integrated stereo digital-to-analogconverters
and analog-to-digital converters employing bitstream
conversion techniques.
The UDA1355H has a selectable one-of-four SPDIF input
(accepting level I, II and III timing) and one SPDIF output
which can generate level II output signals with CMOS
levels. In microcontroller mode the UDA1355H offers a
large variety of possibilities for defining signal flows
throughtheIC, offering a flexible analog, digital and SPDIF
converter chip with possibilities for off-chip sound
processing via the digital input and output interface.
A lock indicator is available on pin LOCK when the
IEC 60958 decoder and the clock regeneration
mechanism is in lock. By default the DAC output and the
digital data interface output are muted when the decoder
is not in lock.
TheUDA1355H contains two clock systems which can run
at independent frequencies, allowing to lock-on to an
incoming SPDIF or digital audio signal, and in the mean
time generating a stable signal by means of the crystal
oscillator for driving, for example, the ADC or SPDIF
output signal.
Using the crystal oscillator (which requires a 12.288 MHz
crystal) and the on-chip low jitter PLL, all standard audio
sampling frequencies (fs= 32, 44.1 and 48 kHz including
half and double these frequencies) can be generated.
DAC in playback mode−74−mW
DAC in Power-down mode−63−mW
2003 Apr 106
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
5BLOCK DIAGRAM
VOUTL
VOUTR
MUTE
WSO
DATAO
DDA1
V
40
DAC
NOISE
INTER-
42
DAC
SHAPER
POLATOR
BCKO
9
44
8
10
DATA OUT
5
IEC 60958
ENCODER
SPDIFOUT
UDA1355H
41
MGU826
V
V
SSA1
SSE
DDE
V
ndbook, full pagewidth
REF
V
DDI
V
CLK_OUT
DDA2
V
ADCP
V
SSX
V
DDX
V
3237273863911
15
12
TIMING
CLOCK AND
XTAL
13
14
AUDIO
AUDIO
ADC
34
FEATURE
PROCESSOR
FEATURE
PROCESSOR
DECI-
MATOR
COMB
FILTER
ADC
36
INPUT
16432
AND
OUTPUT
SELECT
DATA IN
3
1
IEC 60958
SLICER
232425
DECODER
26
21224
CONTROL
INTERFACE
MODE2
MODE1
MODE0
Fig.1 Block diagram.
SEL_STATIC
MP2
MP0
SSIS
V
ADCN
V
MP1
SSA2
V
29 30 312017 18 197
333528
This text is here in white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here in
_white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here inThis text is here in
white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader. white to force landscape pages to be ...
2003 Apr 107
XTALIN
XTALOUT
VINL
VINR
RTCB
RESET
WSI
BCKI
DATAI
SPDIF0
SPDIF1
SPDIF2
SPDIF3
LOCK
SLICER_SEL0
SLICER_SEL1
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
6PINNING
SYMBOLPINPAD
(1)
DESCRIPTION
BCKI1bpt4mtht5vbit clock input (master or slave)
WSI2bpt4mtht5vword select input (master or slave)
DATAI3iptht5vdigital data input
LOCK4op4mcPLL lock indicator output
SPDIFOUT5op4mcSPDIF output
V
V
DDE
SSE
6vddedigital pad supply voltage
7vssedigital pad ground
DATAO8ops5cdigital data output
WSO9bpt4mtht5vword select output (master or slave)
BCKO10bpt4mtht5vbit clock output (master or slave)
CLK_OUT11op4mcclock output; 256f
V
DDX
12vddcocrystal oscillator and PLL supply voltage
or 384f
s
s
XTALIN13apiocrystal oscillator input
XTALOUT14apiocrystal oscillator output
V
SSX
15vsscocrystal oscillator and PLL ground
RESET16ipthdt5vreset input
MODE017apiomode selection input 0 for static mode or microcontroller mode (grounded
2
for I
C-bus)
MODE118bpts5tht5vmode selection input 1 for static mode or AO address input and output for
microcontroller mode
MODE219bpts5tht5vmode selection input 2 for static mode or U_RDY output for microcontroller
mode
SEL_STATIC20apioselection input for static mode, I
2
C-bus mode or L3-bus mode
SLICER_SEL021bpts5tht5vSPDIF slicer selection input 0 for static mode and USER bit output for
microcontroller mode
SLICER_SEL122bpts5tht5vSPDIF slicer selection input 1 for static mode and AC3 preamble detect
output for microcontroller mode
SPDIF023apioSPDIF input 0
SPDIF124apioSPDIF input 1
SPDIF225apioSPDIF input 2
SPDIF326apioSPDIF input 3
V
V
DDI
SSIS
27vddidigital core supply voltage
28vssisdigital core ground
MP029apiomulti-purpose pin 0: frequency select for static mode, not used for
microcontroller mode
MP130iptht5vmulti-purpose pin 1: SFOR1 for static mode, SCL for I2C-bus mode and
L3CLOCK for L3-bus mode
2
MP231iic400kt5vmulti-purpose pin 2: SFOR0 for static mode, SDA for I
C-bus mode and
L3DATA for L3-bus mode
V
ADCP
V
ADCN
32vddcopositive ADC reference voltage
33vssconegative ADC reference voltage
2003 Apr 108
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
SYMBOLPINPAD
(1)
DESCRIPTION
VINL34apioADC left channel input
V
SSA2
35vsscoADC ground
VINR36apioADC right channel input
V
V
V
DDA2
REF
DDA1
37vddcoADC supply voltage
38apioreference voltage for ADC and DAC
39vddcoDAC supply voltage
VOUTL40apioDAC left channel output
V
SSA1
41vsscoDAC ground
VOUTR42apioDAC right channel output
RTCB43ipthdt5vtest control input
MUTE44iipthdt5vDAC mute input
Note
1. See Table 1.
Table 1 Pad description
PADDESCRIPTION
iptht5vinput pad; push-pull; TTL with hysteresis; 5 V tolerant
ipthdt5vinput pad; push-pull; TTL with hysteresis; pull-down; 5 V tolerant
op4mcoutput pad; push-pull; 4 mA output drive; CMOS
ops5coutput pad; push-pull; 5 ns slew rate control; CMOS
bpt4mtht5vbidirectional pad; push-pull input; 3-state output; 4 mA output drive; TTL with hysteresis;
C-bus pad; 400 kHz I2C-bus specification with open drain; 5 V tolerant
apioanalog pad; analog input or output
vddcoanalog supply pad
vsscoanalog ground pad
vddedigital supply pad
vssedigital ground pad
vddidigital core supply pad
vssisdigital core ground pad
2003 Apr 109
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
handbook, full pagewidth
DDA1
REF
V
38
DDA2
V
37
VINR
36
SSA2
V
35
VINL
34
33
32
31
30
29
28
27
26
25
24
23
V
ADCN
V
ADCP
MP2
MP1
MP0
V
SSIS
V
DDI
SPDIF3
SPDIF2
SPDIF1
SPDIF0
BCKI
WSI
DATAI
LOCK
SPDIFOUT
V
DDE
V
SSE
DATAO
WSO
BCKO
CLK_OUT
RTCB
43
42
V
VOUTL
41
40
UDA1355H
V
39
MUTE
44
1
2
3
4
5
6
7
8
9
10
11
SSA1
VOUTR
12
13
14
15
XTALIN
XTALOUT
SSX
V
DDX
V
Fig.2 Pin configuration.
7FUNCTIONAL DESCRIPTION
7.1IC control
The UDA1355H can be controlled either via static pins or
via the microcontroller serial hardware interface being the
I2C-bus with a clock up to 400 kHz or the L3-bus with a
clock up to 2 MHz. It is recommended to use the
microcontroller interface since this gives full access to all
the IC features.
The two microcontroller interfaces only differ in interface
format. The register addresses and features that can be
controlled are identical for L3-bus mode and I2C-bus
mode.
The UDA1355H can operate in three control modes:
• Static mode with limited features
• L3-bus mode with full featuring
• I2C-bus mode with full featuring.
The modes are selected via the 3-level pin SEL_STATIC
according to Table 2.
21
16
17
RESET
MODE0
18
19
MODE2
MODE1
22
20
SEL_STATIC
SLICER_SEL1
SLICER_SEL0
MGU828
Table 2 Control mode selection via pin SEL_STATIC
LEVELMODE
HIGHstatic mode
MIDI2C-bus mode
LOWL3-bus mode
7.2Microcontroller interface
The UDA1355H has a microcontroller interface and all the
sound processing features and system settings can be
controlled by the microcontroller.
The controllable settings are:
• Restoring L3-bus defaults
• Power-on settings for all blocks
• Digital interface input and output formats
• Volume settings for the decimator
• PGA gain settings
2003 Apr 1010
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
• Set two times 40 bits of channel status bits of the SPDIF
output
• Select one of four SPDIF input sources
• Enable digital mixer inside interpolator
• Control mute and mixer volumes of digital mixer
• Selection of filter mode and settings of treble and bass
boost for the interpolator (DAC) section
• Volume settings of interpolator
• Selectionof soft mute via cosine roll-off (only effectivein
L3-bus control mode) and bypass of auto mute
• Selection of de-emphasis
• Enable and control of digital mixer inside interpolator.
The readable settings are:
• Mute status of interpolator
• PLL lock and adaptive lock
• Two times 40 bits of channels status bits of the SPDIF
input signal.
7.3Clock systems
The UDA1355H has two clock systems.
The first system uses an external crystal of 12.288 MHz to
generate the audio related system clocks. Only a crystal
with a frequency of 12.288 MHz is allowed.
The second system is a PLL which locks on the SPDIF or
incoming digital audio signal (e.g. I2S-bus) and recovers
the system clock.
7.3.1CRYSTAL OSCILLATOR CLOCK SYSTEM
The crystal oscillator and the on-chip PLL and divider
circuit can be used to generate internal and external clock
signals related to standard audio sampling frequencies
(such as 32, 44.1 and 48 kHz including half and double of
these frequencies).
The audio frequencies supported in either microcontroller
mode or static mode are given in Table 3.
Table 3 Output frequencies
OUTPUT FREQUENCY
BASIC AUDIO
FREQUENCY
32 kHz256 × 16 kHz
44.1 kHz256 × 22.05 kHz
48 kHz256 × 24 kHz
Remarks:
• If an application mode is selected which does not need
a crystal oscillator, the crystal oscillator cannot be
omitted. The reason is that the interpolator switches to
the crystal clock when an SPDIF input signal is
removed. This switch prevents the noise shaper noise
from moving inside the audio band as the PLL gradually
decreases in frequency.
• If no accurate output frequency is needed, the crystal
can be replaced with a resonator.
• Instead of the crystal, a 12.288 MHz system clock can
be applied to pin XTALIN.
The block diagram of the crystal oscillator and the PLL
circuit is given in Fig.3.
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
7.3.4CLOCK OUTPUT
handbook, halfpage
12.288 MHz
XTALIN
XTALOUT
CLK_OUT
13
CRYSTAL
14
OSCILLATOR
11
L3-bus or I
register setting
256fs or 384fs clock
PLL clock
2
C-bus
UDA1355H
PLL
MODULE
MGU830
Fig.3 Crystal oscillator clock system.
7.3.2PLL CLOCK SYSTEM
The PLL locks on the incoming digital data of the SPDIF or
WS input signal. The PLL recovers the clock from the
SPDIForWSI signal and removes jitter toproduceastable
system clock (see Fig.4).
The UDA1355H has a clock output pin (pin CLK_OUT),
which can be used to drive other audio devices in the
system. In microcontroller mode the output clock is
256fsor 384fs. In static mode the output clock is 256 times
32, 44.1 and 48 kHz.
The source of the output clock is either the crystal
oscillator or the PLL, depending on the selected
application and control mode.
7.4IEC 60958 decoder
The UDA1355H IEC 60958 decoder can selectone of four
SPDIFinputchannels. An on-chip amplifier with hysteresis
amplifies the SPDIF input signal to CMOS level, making it
possible to accept both analog and digital SPDIF signals
(see Fig.5).
handbook, halfpage
23
24
25
26
75 Ω
10 nF
180 pF
SPDIF0
SPDIF1
SPDIF2
SPDIF3
SPDIF0
SPDIF1
SPDIF2
SPDIF3
WSI
select SPDIF source
23
24
25
26
SLICER
2
IEC 60958
DECODER
PLL
UDA1355H
256f
384f
s
or
s
MGU827
Fig.4 PLL clock system.
7.3.3WORD SELECTION DETECTION CIRCUIT
This circuit is clocked by the 12.288 MHz crystal oscillator
clock and generates a Word Selection (WS) detection
signal. If the WS detector does not detect any WS edge,
defined as 7 times LOW and 7 times HIGH, then the
WS detection signal is LOW. This information can be used
tosetthe clock for the noise shaper in theinterpolator.This
will prevent noise shaper noise in the audio band.
UDA1355H
MGU829
Fig.5 IEC 60958 input circuit.
7.4.1AUDIO DATA
From the incoming SPDIF bitstream 24 bits of data for the
left and right channel are extracted.
There is a hard mute (not a cosine roll-off mute) if the
IEC 60958 decoder is out of lock or detects bi-mark phase
encoding violations. The lock indicator and the key
channel status bits are accessible in L3-bus mode.
The UDA1355H supports the following sample
frequencies and data rates, including half and double of
these frequencies:
• fs= 32 kHz; resulting in a data rate of 2.048 Mbit/s
• fs= 44.1 kHz; resulting in a data rate of 2.8224 Mbit/s
• fs= 48 kHz; resulting in a data rate of 3.072 Mbit/s.
2003 Apr 1012
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
7.4.2CHANNEL STATUS AND USER BITS
As well as the data bits there are several IEC 60958 key
channel status bits:
• Pre-emphasis and audio sampling frequency bits
• Two channel PCM indicator bits
• Clock accuracy bits.
In total 40 status bits per channel are recovered from the
incomingIEC 60958bitstream.Thesearereadableviathe
microcontroller interface.
User bits, which can contain a large variety of data, such
asCD text, are output to pin SLICER_SEL0 (see Table 4).
In microcontroller mode this signal contains the raw user
bits extracted from the SPDIF bitstream. Signal U_RDY
gives a pulse on pin MODE2 each time there is a new user
bit available. Both signals can be used by an external
microcontroller to grab and decode the user bits.
Table 4 Signal names in microcontroller mode
PIN NAMESIGNAL NAME
SLICER_SEL0USER
MODE2U_RDY
SLICER_SEL1AC3
7.4.3D
Audio and digital data can be transmitted in the SPDIF
bitstream. The PCM channel status bit should be set to
logic 1 if the SPDIF bitstream is carrying digital data
instead of audio data, but in practice it proves that not all
equipment handles these channel status bits properly.
In the UDA1355H, digital data is detected via bit PCM, or
via the sync bytes as specified by IEC. These sync bytes
are two sync words, F872H and 4E1FH (two subframes)
preceded by four or more subframes filled with zeros.
Signal AC3 is kept HIGH for 4096 frames when the
UDA1355H detects this burst preamble. Signal AC3 is
present on pin SLICER_SEL1 in microcontroller mode
(see Table 4).
IGITAL DATA
7.5IEC 60958 encoder
When using the crystal oscillator clock, the IEC 60958
encoder output is a full-swing digital signal with level II
timing.
When the recovered clock from the PLL is used the
IEC 60958 encoder will function correctly but will not meet
level II timing requirements.
7.5.1S
All user and channel status bits are set to logic 0. This is
default value specified by IEC.
In static mode 0 and 2, the selected SPDIF input channel
can be looped through to pin SPDIFOUT (see Fig.6).
7.5.2MICROCONTROLLER MODE
Two times 40 channel status bits can be set. Default value
for each status bit is logic 0. When setting the channel
status bits, it is possible to set only the left channel status
bits and have the bits copied to the right channel.
The procedure of writing the channel status bits is as
follows:
1. Set bit SPDO_VALID = 0 to prevent immediately
2. Set bit l_r_copy = 1 if the right channel needs the
3. Write the left and right channel status bits.
4. Set bit SPDO_VALID = 1 after writing all channel
In microcontroller modes 2 and 13, the selected SPDIF
input channel can be looped through to pin SPDIFOUT
(see Fig.6).
TATIC MODE
sending the status bits during writing.
same status bits as the left channel or set
bit l_r_copy = 0 if the right channel needs different
status bits to the left channel.
statusbitstotheregister.StartingfromthenextSPDIF
block the IEC 60958 encoder will use the new status
bits.
2003 Apr 1013
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
handbook, full pagewidth
SPDIF0
SPDIF1
SPDIF2
SPDIF3
SPDOUT_SEL1
SPDOUT_SEL0
23
24
25
26
SLICER_SEL[1:0
SLICER
select
SPDIF source
21, 22
]
SPDIF
source
UDA1355H
IEC 60958
DECODER
IEC 60958
ENCODER
Fig.6 Selection options for SPDIF output.
7.6Analog input
7.6.1ADC
The analog input is equipped with a Programmable Gain
Amplifier (PGA) which can be controlled via the
microcontroller interface. The control range is from
0 to 24 dB gain in 3 dB steps independent for the left and
right channels.
In applications in with a 2 V (RMS) input signal, a 12 kΩ
resistor must be used in series with the input of the ADC.
The 12 kΩ resistor forms a voltage divider together with
the internal ADC resistor and ensures that the voltage,
applied to the input of the IC, never exceeds 1 V (RMS).
In the application for a 2 V (RMS) input signal, the PGA
must be set to 0 dB. When a 1 V (RMS) input signal is
applied to the ADC in the same application, the PGA gain
must be set to 6 dB.
An overview of the maximum input voltages allowed with
and without an external resistor and the PGA gain setting
is given in Table 5.
Table 5 Maximum input voltage; VDD=3V
EXTERNAL
RESISTOR
(12 kΩ)
PGA GAIN
SETTING
MAXIMUM
INPUT
VOLTAGE
Present0 dB2 V (RMS)
6 dB1 V (RMS)
Absent0 dB1 V (RMS)
6 dB0.5 V (RMS)
SPDOUT_SEL2
MODE[3:0
17 to 1920
MODE[2:0]SEL_STATIC
]
5
SPDIF OUT
MGU833
7.6.2DECIMATION
Thedecimation from 64fsisperformed in two stages: comb
filter and decimation filter. The first stage realizes a
fourth-order characteristic with a decimation factor
sin x
----------- x
of eight. The second stage consists of three half-band
filters each decimating by a factor of two. Table 6 shows
the characteristics.
Table 6 Decimation filter characteristics
ITEMCONDITIONSVALUE (dB)
Pass-band ripple0 to 0.45f
Stop band>0.55f
Dynamic range0 to 0.45f
Overallgain from ADC
input to digital output
DC; V
note 1
= 0 dB;
I
s
s
s
±0.02
−60
140
−1.16
Note
1. Theoutput is not 0 dB when V
=1VatVDD=3V.
I(rms)
This is because the analog components can spread
over the process. When there is no external resistor,
the −1.16 dB scaling prevents clipping caused by
process mismatch.
In the ADC path there are left and right independent digital
volume controls with a range from +24 to −63.5 dB
and −∞ dB. This volume control is also used as a digital
linear mute that can be used to prevent plops when
powering-up or powering down the ADC front path.
2003 Apr 1014
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
7.6.3DC FILTERING
In the decimator there are two digital DC blocking circuits.
The first blocking circuit is in front of the volume control to
remove DC bias from the ADC output. The DC bias is
added in the ADC to prevent audio band Idle tones
occurring in the noise shaper. With the DC components
removed, a signal gain of 24 dB can be achieved.
The second blocking circuit removes the DC components
introduced by the decimator stage.
7.6.4OVERLOAD DETECTION
Bit OVERFLOW = 1 when the output data in the left or
right channel is larger than −1.16 dB of the maximum
possible digital swing. This condition is set for at least
512fs cycles (that is 11.6 ms at fs= 44.1 kHz). This
time-out is reset for each infringement.
7.7Analog output
7.7.1AUDIO FEATURE PROCESSOR
The audio feature processor provides automatic
de-emphasis for the IEC 60958 bitstream.
In microcontroller mode all features are available and
there is a default mute on start up.
7.7.2INTERPOLATING FILTER
The digital filter interpolates from 1fsto 64fs, or from
1fsto 128fs, by cascading a half-band filter and a FIR filter.
The stereo interpolator has the following basic features:
• 24-bit data path
• Mixing of two channels:
– To prevent clipping inside the core, there is an
automatic signal level correction of −6 dB scaling
before mixing and +6 dB gain after digital volume
control
– Position of mixing can be set before or after bass
boost and treble
– Mastervolume control andmutewithindependent left
and right channel settings for balance control
– Independently left and right channel de-emphasis,
volume control and mute (no left or right)
– Output of the mixer is to the I2S-bus or IEC 60958
decoder.
• Full FIR filter implementation for all the upsampling
filters
• Integrated digital silence detection for left and right
channels with selectable silence detection time
• Support for 1fsand 2fs input data rate and 192 kHz
audio via I2S-bus.
The stereo interpolator has the following sound features:
• Linear volume control using 14-bit coefficients with
0.25 dB steps: range 0 to −78 dB and −∞ dB; hold for
master volume and mixing volume control
• A cosine roll-off soft mute with 32 coefficients; each
coefficientis used for four samples, in total 128 samples
are needed to fully mute or de-mute (approximately
3 ms at fs= 44.1 kHz)
• Independent selectable de-emphasis for 32, 44.1, 48
and 96 kHz for both channels
• Treble is the selectable positive gain for high
frequencies. The edge frequency of the treble is fixed
and depends on the sampling frequency. Treble can be
set independently for left and right channel with two
settings:
–fc= 1.5 kHz; fs= 44.1 kHz; 0 to 6 dB gain range with
2 dB steps
–fc= 3 kHz; fs= 44.1 kHz; 0 to 6 dB gain range with
2 dB steps.
• Normal bass boost is the selectable positive gain for low
frequencies. The edge frequency of the bass boost is
fixed and depends on the sampling frequency. Normal
bassboostcan be set independently for the left and right
channel with two sets:
–fc= 250 Hz;fs= 44.1 kHz;0 to 18 dBgainrangewith
2 dB steps
–fc= 300 Hz;fs= 44.1 kHz;0 to 24 dBgainrangewith
2 dB steps.
• Resonant bass boost optional function is selected if
bit BASS_SEL = 1. When selected, the characteristics
aredeterminedbysix14-bitcoefficients.Resonantbass
boost controls the left and right channel with the same
characteristics. When resonant bass boost is selected,
the treble control also changes to a single control for
both channels following the gain setting of the left
channel.
Asoftware program is available foruserstogenerate the
required six 14-bit coefficients by entering the desired
centre frequency (fc), positive or negative peak gain,
sampling frequency (fs) and shape factor (see
Figs 7 and 8).
2003 Apr 1015
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Table 7 Interpolation filter characteristics
ITEMCONDITIONSVALUE (dB)
Pass-band ripple0 to 0.45f
Stop band>0.55f
Dynamic range0 to 0.4535f
7.7.3D
IGITAL MIXER
s
s
s
±0.035
−60
140
The UDA1355H has a digital mixer inside the interpolator.
Thedigitalmixer can be used as a cross over or a selector.
A functional block diagram of the mixer mode is shown in
Fig.9. This mixer can be used in microcontroller mode
only.
The UDA1355H can be set to the mixer mode by setting
bit MIX = 1.In the mixer mode, there are three volume and
10
handbook, halfpage
gain
8
(dB)
6
4
2
0
−2
−4
−6
−8
−10
11010
MGU832
2103
f (Hz))
mute controls available: for source 1, for source 2 and for
the master (sum) signal. All three volume ranges can be
controlled in 0.25 dB steps.
Topreventclipping inside the mixer, the signals are scaled
with −6 dB before mixing, therefore the sum of the two
signals is always equal to or lower than 0 dB. After the
mixing there is a 6 dB gain in the master volume control.
This means that at the analog output the signal can clip,
but the clipping can be undone by decreasing the master
volume control.
2
The output of the mixer is available via the I
S-bus output
or via the SPDIF output. The output signal of the mixer is
scaled to a maximum of 0 dB, so the digital output can
never clip.
10
handbook, halfpage
gain
8
(dB)
6
4
2
0
−2
−4
−6
−8
−10
11010
MGU831
2103
f (Hz))
fc=70Hz
fs= 44.1 kHz
Peak gain = 10 dB
Shape factor = 1.4142
Fig.7 Resonant bass boost example 1.
2003 Apr 1016
fc=70Hz
fs= 44.1 kHz
Peak gain = 10 dB
Shape factor = 1.4142
Fig.8 Resonant bass boost example 2.
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
handbook, full pagewidth
channel 2
VOLUME
DE-EMPHASIS
VOLUME
DE-EMPHASIS
channel 1
AND
MUTE
AND
MUTE
mixing before
sound features
BASS-BOOST
AND
TREBLE
Fig.9 Digital mixer (DAC) inside the interpolator DSP.
7.7.4DIGITAL SILENCE DETECTOR
The UDA1355H is equipped with adigital silence detector.
This detects whether a certain amount of consecutive
samples are 0. The number of samples can be set with
bits SD_VALUE[1:0] to 3200, 4800, 9600 or 19600
samples.
The digital silence detection status can be read via the
microcontroller interface.
7.7.5NOISE SHAPER (DAC)
The noise shaper shifts in-band quantization noise to
frequencies above the audio band. The noise shaper
output is converted into an analog signal using a Filter
Stream Digital-to-Analog Converter (FSDAC). This noise
shaping technique enables high signal-to-noise ratios to
be achieved.
The UDA1355H is equipped with two noise shapers:
• A third-order noise shaper operating at 128fs. Which is
used at low sampling frequencies (8 to 16 kHz) to
prevent noise shaper noise shifting into the audio band
for the fifth-order noise shaper
• A fifth-order noise shaper operating at 64fs. Which is
used at high sampling frequencies (from 32 kHz
upwards).
When the noise shaper changes, the clock to the FSDAC
changes and the filter characteristic of the FSDAC also
changes. The effect on the roll of is compensated by
selecting the filter matching speed and order of the noise
shaper.
mixing after
sound features
1f
s
output of mixer
L3/I
2
C bit
FILTER
INT.
UDA1355H
MASTER
2f
s
VOLUME
AND
MUTE
to
interpolation
filter and
DAC output
MGU834
7.7.6FILTER STREAM DAC
The FSDAC is a semi digital reconstruction filter that
converts the 1-bit data bitstream of the noise shaper to an
analog output voltage. The filter coefficients are
implemented as current sources and are summed at
virtual ground of the operational amplifier output. In this
way, very high signal-to-noise performance and low clock
jitter sensitivity are achieved. A post filter is not needed
due to the inherent filter function of the FSDAC. On-chip
amplifiers convert the FSDAC output current to an output
voltage signal capable of driving a line output. The output
voltageoftheFSDAC scales proportionally with the supply
voltage.
7.7.7DAC MUTE
The DAC and interpolator can be muted by setting
pin MUTE to a HIGH level. The output signal is muted to
zero via a cosine roll-off curve and the DAC is powered
down. When pin MUTE is at LOW level the signal rise
follows the same cosine curve.
To prevent plops in case of changing inputs, clock to the
DAC or application modes, a special mute circuit for the
DAC is implemented (see Table 8).
In all application modes in which the DAC is active the
DAC can be muted by pin MUTE. The microcontroller
mute bits and pin MUTE act as an OR function.
2003 Apr 1017
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Table 8 Muting to prevent plopping
OCCASION
MT1MT2MTM
Input selection
Select channel 1 sourcex−−no mute after selection
Select channel 2 source−x−no mute after selection
Select chip mode
PLL is source for the DAC−−xwait until PLL is locked again
Crystal is source for the DAC−−xno mute after selection
Select between microcontroller mode and static mode
PLL is source for the DAC−−xwait until PLL is locked again
Crystal is source for the DAC−−xno mute after selection
Audio features
Select noise shaper order−−xno mute after selection
Select FSDAC output polarity−−xno mute after selection
Select SPDIF input−−xPLL is locked again
Select mixer−−−no mute needed
Select mixer position−−−no mute needed
Select crystal clock source−−xno mute after selection
BIT
DE-MUTE CONDITION
7.8Digital audio input and output
The selection of the digital audio input and output formats
and master or slave modes differ for static and
microcontroller mode.
In master mode, when 256fs output clock is selected and
the digital interface is master, the BCK output clock will be
64fs.Incase384fsoutputclockisselected,theBCK output
clock will be 48fs.
In the static mode the digital audio input formats are:
• I2S-bus
• LSB-justified; 16 bits
• LSB-justified; 24 bits
• MSB-justified.
The digital audio output formats are:
• I2S-bus
• MSB-justified.
In the microcontroller mode, the following formats are
independently selectable:
• I2S-bus
• LSB-justified; 16 bits
• LSB-justified; 18 bits
• LSB-justified; 20 bits
• LSB-justified; 24 bits
• MSB-justified.
7.9Power-on reset
The UDA1355H has a dedicated reset pin with an internal
pull-down resistor. In this way a Power-on reset circuit can
bemadewithacapacitorandaresistoratpin RESET.The
external resistor is needed since the pad is 5 V tolerant.
This means that there is a transmission gate in series with
the input and the resistor inside the pad cannot be seen
from the outside world (see Fig.10).
The reset timing is determined by the external pull-down
resistor and the external capacitor which is connected to
pin RESET. At Power-on reset, all the digital sound
processing features and the system controlling features
are set to the default setting of the microcontroller mode.
Since the bit controlling the clock of the synchronous
registers is set to enable, the synchronous registers are
also reset.
2003 Apr 1018
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
8APPLICATION MODES
handbook, halfpage
RESET
Transmission gate
for 5V tolerance
16
UDA1355H
V
SS
MGU835
Fig.10 5 V tolerant pull-down input pad.
The clock should be running during the reset time. When
noclockcanbe guaranteed in microcontroller mode, a soft
reset should be given when the system is running by
writing to register 7FH.
Table 9 Static mode pin assignment
In this chapter the application modes for static mode and
microcontroller mode are described.
The UDA1355H can be controlled by static pins, the
L3-busorI2C-businterface.Duetothelimitationsimposed
bythe pin count, only basic functions are available in static
mode. For optimum use of the UDA1355H features, the
microcontroller mode is strongly recommended.
There are 11 application modes available in the static
mode and 14 application modes in microcontroller mode.
The application modes are explained in the two sections:
Section 8.2 explains the application modes 0 to 10.
Section 8.4 explains the more advanced features of
modes 0 to 10 and modes 12 to 14 available in the
microcontroller mode.
8.1Static mode pin assignment
The default values for all non-pin controlled settings are
identical to the start-up defaults from the microcontroller
mode.WhetherBCK and WS are master or slave depends
on the selected application mode.
Table 9 defines the pin functions in static mode.
PIN
STATIC MODE
SYMBOL
LEVELDESCRIPTION
4LOCKLOWIEC 60958 decoder out of lock (when SPDIF input) or clock
regeneration out of lock (I
2
S-bus input)
HIGHIEC 60958 decoder in lock (when SPDIF input) or clock
regeneration in lock (I
2
S-bus input)
16RESETLOWnormal operation
HIGHreset
17, 18,19MODE0, MODE1,
−select application mode; see Table 10
MODE2
20SEL_STATICHIGHstatic pin control
LOWmicrocontroller mode
22, 21SLICER_SEL1,
SLICER_SEL0
LOW, LOWIEC 60958 input from pin SPDIF0
LOW, HIGHIEC 60958 input from pin SPDIF1
HIGH, LOWIEC 60958 input from pin SPDIF2
HIGH, HIGHIEC 60958 input from pin SPDIF3
29FREQ_SELLOWselect 44.1 kHz sampling frequency for the crystal oscillator,
note 1
MIDselect 32 kHz sampling frequency for the crystal oscillator, note 1
HIGHselect 48 kHz sampling frequency for the crystal oscillator, note 1
2003 Apr 1019
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
PIN
30, 31SFOR1, SFOR0LOW, LOWset I
STATIC MODE
SYMBOL
LEVELDESCRIPTION
2
S-bus format for digital data input and output interface
LOW, HIGHset LSB-justified 16 bits format for digital data input interface and
MSB-justified format for digital data output interface
HIGH, LOWset LSB-justified 24 bits format for digital data input interface and
MSB-justified format for digital data output interface
HIGH, HIGHset MSB-justified format for digital data input and output interface
44MUTELOWnormal operation
HIGHmute active
Note
1. FPLL 256fs is output from pin CLKOUT in PLL locked static mode.
8.2Static mode basic applications
The static application modes are selected with the pins MODE2, MODE1 and MODE0, with pin MODE0 being a 3-level
pin. In Table 10, the encoding of the pins MODE[2:0] is given.
Table 10 Static mode basic applications
MODE
MODE SELECTION PINS
(1)
MODE2MODE1MODE0
SPDIF
INPUT
SPDIF
OUTPUT
CLOCK
ADCDAC
(2)
I2S-BUS
INPUT
SLAVE
I2S-BUS
OUTPUT
MASTER
PLL
LOCKS
ON
INPUT
0LLLPLLPLL−PLL−PLLSPDIF
1L L M − PLL−PLLPLL−I2S-bus
2LLHPLLPLL−PLLPLLPLLSPDIF
3L H L − xtalxtal−−xtal−
4L H M − xtalxtalxtalxtalxtal−
5L H H − xtalxtalxtalxtalxtal−
6H L L − PLLxtalPLLPLLxtalI
2
S-bus
7HLMPLLxtalxtalPLL−xtalSPDIF
8H L H − xtalxtalPLLPLLxtalI2S-bus
9HHLPLLxtal−xtalxtalPLLSPDIF
10HHMPLLxtal−PLLxtalPLLSPDIF
11HHHnot used
Notes
1. In column mode selection pins means:
L: pin at 0 V; M: pin at half V
; H: pin at V
DDD
DDD
.
2. In column clock means:
xtal: the clock is based on the crystal oscillator; PLL: the clock is based on the PLL.
2003 Apr 1020
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
The first 11 application modes are given in this section. Schematic diagrams of these application modes are given in
Table 11. In this table the basic features are mentioned and also the extra features in case of microcontroller mode are
given. It should be noted that the blocks running at the crystal clock (XTAL) are marked unshaded while the blocks
running at the PLL clock are shaded.
Table 11 Overview of static mode basic applications
MODEFEATURESSCHEMATIC
0Data path:
2
• Input SPDIF to outputs DAC, I
SPDIFOUT via loop through.
Features:
• System locks onto the SPDIF input
signal
• BCK and WS are master
• Microcontroller mode:
– DAC sound features can be used
– SPDIF input channel status bits
(two times 40 bits) can be read.
S or
SPDIF IN
SPDIF LOCK
PLL
MUTE
DAC
SPDIFOUT
I2S OUTPUTI2S master
MGU836
1Data path:
• Input I
(level II not guaranteed: depends on
I2S-bus clock).
Features:
• System locks onto the WSI signal
• BCKI and WSI are slave
• Microcontroller mode:
– DAC sound features can be used
– SPDIF output channel status bits
(two times 40 bits) setting.
2
S to outputs DAC or SPDIF
I2S slave
I2S INPUT
PLL
I2S LOCK
MUTE
DAC
SPDIF OUT
MGU837
2003 Apr 1021
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
MODEFEATURESSCHEMATIC
2Data path:
• Input SPDIF to outputs I
SPDIFOUT via loop through
• Input I2S to output DAC.
Features:
• Possibility to process input SPDIF via
2
S-bus using an external DSP and
I
then to output DAC
• System locks onto the SPDIF input
signal
• I2S input and output with BCK and WS
are master
• Microcontroller mode: see Section 8.4.
2
S or
SPDIF IN
I2S INPUT
I2S slave
EXTERNAL DSP
(e.g. equalizing, spatializing)
(SAA7715)
PLL
I2S OUTPUT
SPDIF LOCK
DAC
2
S master
I
MGU838
MUTE
SPDIFOUT
3Data path:
• Input ADC to outputs I
Features:
• Crystal oscillator generates the clocks
• Microcontroller mode:
– PGA gain setting
– Volume control in decimator setting
– SPDIF output channel status bits
(two times 40 bits) setting.
4Data path:
• Input ADC to output I
• Input I2S to outputs DAC or SPDIF.
Features:
• Possibility to process input ADC via
2
S-bus using a external DSP and then
I
to outputs DAC or SPDIF
• Crystal oscillator generates the clocks
• I2S input and output with BCK and WS
are master
• Microcontroller mode: see Section 8.4.
2
S or SPDIF.
2
S
ADC
ADC
I
2
XTAL
XTAL
S INPUT
I2S slave
EXTERNAL DSP
(e.g. equalizing, spatializing)
(SAA7715)
SPDIF OUT
I2S OUTPUT
MGU839
DAC
SPDIF OUT
2
I
S OUTPUT
I2S master
MGU840
I2S master
MUTE
2003 Apr 1022
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
MODEFEATURESSCHEMATIC
5Data path:
• Input ADC to outputs I
• Input I2S to output DAC.
Features:
• Possibility to process input ADC via
2
I
S-bus using an external DSP and
then to output DAC
• Crystal oscillator generates the clocks
• I2S input and output with BCK and WS
are master
• Microcontroller mode: see Section 8.4.
2
S or SPDIF
XTAL
ADC
2
I
S INPUT
I2S slave
EXTERNAL DSP
(e.g. equalizing, spatializing)
(SAA7715)
DAC
SPDIF OUT
2
I
S OUTPUT
I2S master
MGU841
MUTE
6Data path:
• Input ADC to output I
• Input I2S to outputs DAC or SPDIF
(level II not guaranteed: depends on
I2S-bus clock).
Features:
• Possibility to process input ADC via
2
I
S-bus using an external DSP and
then to outputs DAC or SPDIF
• Crystal oscillator generates the clocks
for input ADC and output I2S
• WSI is slave
• WSO is master
• Microcontroller mode: see Section 8.4.
2
S
I2S LOCK
XTAL
I2S INPUT
I2S slave
EXTERNAL DSP
(SAA7715)
PLL
SPDIF OUT
I
DACADC
2
S OUTPUT
I2S master
MGU842
MUTE
2003 Apr 1023
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
MODEFEATURESSCHEMATIC
7Data path:
• Input SPDIF to output DAC
• Input ADC to outputs SPDIF or I
Features:
• Crystal oscillator generates the clocks
for outputs SPDIF and I
2
S
• PLL locks onto the SPDIF input signal
• WS of I2S output is master
• Microcontroller mode:
– Decimator features can be used
– DAC sound features can be used
– SPDIF input channel status bits
(two times 40 bits) can be read
– SPDIF output channel status bits
(two times 40 bits) setting.
2
S.
SPDIF LOCK
XTAL
ADC
SPDIF IN
PLL
I
DAC
SPDIF OUT
2
S OUTPUT
MGU843
MUTE
I2S master
8Data path:
• Input ADC to outputs SPDIF or I
• Input I2S to output DAC.
Features:
• Possibility to process input ADC, via
2
I
S-bus using an external DSP and
then to output DAC
• Crystal oscillator generates the clocks
for outputs SPDIF and I2S
• WSI is slave
• WSO master
• Microcontroller mode:
– Decimator features can be used
– DAC sound features can be used
– SPDIF output channel status bits
(two times 40 bits) setting.
2
S
I2S LOCK
XTAL
ADC
I2S INPUT
I2S slave
EXTERNAL DSP
(e.g. Sample Rate Convertor)
(SAA7715)
PLL
SPDIF OUT
2
I
MUTE
DAC
S OUTPUT
I2S master
MGU844
2003 Apr 1024
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
MODEFEATURESSCHEMATIC
9Data path:
• Input SPDIF to output I
• Input I2S to outputs DAC or SPDIF.
Features:
• Possibility to process input SPDIF, via
2
I
S-bus using an external DSP and
then to outputs DAC or SPDIF
• BCKand WS being master for both I2S
input and output (different clocks)
• Input I2S to outputs DAC and SPDIF;
BCK and WS being master; clocks
based on crystal oscillator
• Microcontroller mode:
– DAC sound features can be used
– SPDIF output channel status bits
(two times 40) setting.
2
S
SPDIF LOCK
XTAL
SPDIF IN
2
I
S INPUT
I2S slave
EXTERNAL DSP
(e.g. Sample Rate Convertor)
(SAA7715)
PLL
SPDIF OUT
I2S OUTPUT
MGU845
DAC
2
S master
I
MUTE
10Data path:
• Input SPDIF to output DAC or I
• Input I2S-bus to output SPDIF.
Features:
• Possibility to process input SPDIF, via
2
I
S-bus using an external DSP and
then to output SPDIF
• Input SPDIF to outputs I2S and DAC;
locking onto the SPDIF input signal;
BCK and WS being master
• Input I2S to output SPDIF; BCK and
WS being master; clocks are
generated by the crystal oscillator
• Microcontroller mode:
– DAC sound features can be used
– SPDIF input channel status bits
(two times 40) can be read
– SPDIF output channel status bits
(two times 40) setting.
11Not used
12See microcontroller mode
13See microcontroller mode
14See microcontroller mode
15Not used
2
S
SPDIF LOCK
XTAL
SPDIF IN
2
I
S INPUT
I2S slave
EXTERNAL DSP
(e.g. Sample Rate Convertor)
(SAA7715)
PLL
SPDIF OUT
I2S OUTPUT
MGU846
DAC
2
S master
I
MUTE
2003 Apr 1025
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
8.3Microcontroller mode pin assignment
In microcontroller mode all features become available, such as volume control, PGA gain and mixing (in some modes).
The pin functions are defined in Table 12.
Table 12 Microcontroller mode pin assignment
PIN
SYMBOL
L3-BUS
SYMBOL
I2C-BUS
LEVELDESCRIPTION
4LOCKLOCKLOWFPLL and SPDIF are out of LOCK
HIGHFPLL in lock when SPDIF is not used; FPLL or SPDIF in lock when
SPDIF is used
16RESETRESETLOWnormal operation
HIGHreset
17no functionno functionLOWconnect to ground
18A0A0−A0 address input/output bit (for microcontroller register)
19U_RDYU_RDYLOWuser bit stable
HIGHnew user bit
20SEL_STATICSEL_STATICMIDI
2
C-bus mode
LOWL3-bus mode
HIGHstatic mode
21USERUSER−user bit output (new bit every SPDIF sub-frame)
22AC3AC3LOWno I
HIGHI
29L3MODEno function−L3MODE for L3-bus mode; no function for I
30L3CLOCKSCL−L3CLOCK for L3-bus mode or SCL for I
31L3DATASDA−L3DATA for L3-bus mode or SDA for I
2
S-bus data preamble detected
2
S-bus data preamble detected
2
C-bus
2
C-bus mode
2
C-bus mode
44MUTEMUTELOWno mute
HIGHmute active
2003 Apr 1026
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
8.4Microcontroller mode applications
In Table 13, the encoding of bits MODE[3:0] in the microcontroller mode is given.
91001PLLxtalxtalxtalxtalPLLSPDIF
101010PLLxtalPLLPLLxtalPLLSPDIF
111011not used
121100PLLxtalxtalPLLPLLxtalSPDIF
131101PLLPLLxtalPLLPLLxtalSPDIF
141110−PLLPLLPLLPLLPLLI
151111not used
(1)
I2S-BUS
INPUT
SLAVE
I2S-BUS
OUTPUT
MASTER
PLL
LOCKS
ON
INPUT
2
S
2
S
2
S
Note
1. In column clock means:
xtal: the clock is based on the crystal oscillator; PLL: the clock is based on the PLL.
2003 Apr 1027
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
In the microcontroller mode, more features are available. The application modes are given in Table 14. Some modes are
the same in terms of data path as for the static mode. These modes are already explained in Section 8.2. Some modes
are combined into one mode (like modes 4 and 5).
Table 14 Overview of microcontroller modes
MODEFEATURESCHEMATIC
0See static mode
1See static mode
2Data path:
2
• Inputs ADC, I
DAC, I2S or SPDIF.
Features:
• All clocks are related to the SPDIF
clock
2
S input and output have master BCK
• I
and WS
• SPDIF input channel status bits (two
times 40) can be read
• OutputSPDIFsupported but the timing
not according to level II: depends on
I2S-bus clock
• Output SPDIFOUT loop through can
be selected with independent SPDIF
input channel select.
S and SPDIF to outputs
ADC
SPDIF IN
I2S INPUT
PLL
I2S slave
EXTERNAL DSP
(e.g. equalizing, spatializing)
(SAA7715)
SPDIF LOCK
DAC
SPDIF
OUT
I2S OUTPUT
2
I
MGU847
MUTE
SPDIF OUT
S master
3See static mode
4 + 5Data path:
• Inputs ADC and I2S to outputs DAC,
I2S or SPDIF.
Features:
• Mode 4 and 5 are combined in
microcontroller mode
• Crystal oscillator generates the clocks
2
S input and output have master BCK
• I
and WS
• SPDIF output channel status bits (two
times 40) setting.
• Input SPDIF to output I
and WS being master; the clocks for
this are recovered from the SPDIF
input signal
• Therestoftheclocksaregeneratedby
the crystal oscillator
• SPDIF input channel status bits (two
times 40) can be read
• SPDIF output channel status bits (two
times 40) setting
• Possibility to process input SPDIF, via
I2S-bus using an external DSP and
then to outputs DAC or SPDIF.
2
S to outputs DAC or
2
S with BCK
XTAL
ADC
SPDIF IN
2
I
S INPUT
I2S slave
EXTERNAL DSP
(e.g. Sample Rate Convertor)
(SAA7715)
PLL
SPDIF OUT
I2S OUTPUT
SPDIF LOCK
DAC
2
S master
I
MGU849
MUTE
10Data path:
• InputsADC and SPDIF to outputs DAC
or I2S
• Input I2S to output SPDIF.
Features:
• BCK and WS are master
• SPDIF input channel status bits (two
times 40) can be read
• SPDIF output channel status bits (two
times 40) setting
• Possibility to process inputs ADC or
SPDIF, via I
2
S-bus using an external
DSP and then to output SPDIF.
11Not used
XTAL
ADC
SPDIF IN
2
I
S INPUT
I2S slave
EXTERNAL DSP
(e.g. Sample Rate Convertor)
(SAA7715)
PLL
SPDIF OUT
I2S OUTPUT
SPDIF LOCK
DAC
2
I
S master
MGU850
MUTE
2003 Apr 1029
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
MODEFEATURESCHEMATIC
12Data path:
• Input ADC to outputs I
• Inputs I2S and SPDIF to output DAC.
Features:
• BCK and WS of I
• Inputs SPDIF and I2S to output DAC
withmixing/selection possibility; clocks
aregenerated fromSPDIFinput signal,
and BCK and WS are master
• SPDIF input channel status bits (two
times 40) can be read
• SPDIF output channel status bits (two
times 40) setting.
2
S or SPDIF
2
S output are master
XTAL
ADC
SPDIF IN
I2S INPUT
PLL
I
SPDIF LOCK
DAC
SPDIF OUT
2
S OUTPUT
MGU851
MUTE
I2S masterI2S slave
13Data path:
• Input ADC to output I
2
S
• Inputs I2S and SPDIF to outputs DAC
or SPDIF.
Features
• BCK and WS being master
• SPDIF input channel status bits (two
times 40) can be read
• OutputSPDIFsupported but the timing
not according to level II
• Output SPDIFOUT loop through can
be selected with independent SPDIF
input channel select.
14Data path:
2
• Inputs ADC and I
S to outputs DAC
SPDIF and I2S.
Features:
• All clocks are related to WS signal of
2
I
S-bus input
• Master BCK and WS for I2S output;
slave BCK and WS for I2S input
• SPDIF output channel status bits (two
times 40) can be set; level II timing
depends on the I2S-bus clocks.
2
S slave
I
XTAL
ADC
SPDIF IN
I2S INPUT
ADC
I2S INPUT
PLL
SPDIF
OUT
2
I
S OUTPUT
PLL
SPDIF OUT
I2S OUTPUT
SPDIF LOCK
MUTE
DAC
SPDIF OUT
I2S master
MGU852
I2S LOCK
MUTE
DAC
I2S masterI2S slave
15Not used
2003 Apr 1030
MGU853
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
9SPDIF SIGNAL FORMAT
9.1SPDIF channel encoding
The digital signal is coded using Biphase Mark Code
(BMC), which is a kind of phase modulation. In this
scheme, a logic 1 in the data corresponds to two
zero-crossings in the coded signal, and a logic 0 to one
zero-crossing. An example of the encoding is given in
Fig.11.
handbook, halfpage
clock
data
BMC
MGU606
Fig.11 Biphase mark encoding.
9.2SPDIF hierarchical layers
The SPDIF signal format is shown in Fig.12. A PCM signal
is transmitted in sequential blocks. Each block consists
of 192 frames. Each frame contains two sub-frames, one
for each channel. Each subframe is preceded by a
preamble. There are three types of preambles: B, M
and W. Preambles can be spotted easily in an SPDIF
bitstream because these sequences never occur in the
channel parts of a valid SPDIF bitstream.
The sub-frame format is represented by Fig.13.
A sub-frame contains a single audio sample word which
may be 24 bits wide, a validity bit which indicates whether
the sample is valid, a bit containing user data, a bit
indicating the channel status and a parity bit for this
sub-frame.
Thedata bits 31 to 4 in each sub-frame are encodedusing
a BMC scheme. The sync preamble contains a violation of
the BMC scheme and can be detected. Table 15 indicates
the values of the preambles.
The SPDIF specification IEC 60958 supports three levels
of clock accuracy:
• Level I high accuracy: Tolerance of transmitting
sampling frequency shall be within 50 × 10
−6
• Level II, normal accuracy: All receivers should receive asignal of 1000 × 10−6 of nominal sampling frequency
• Level III, variable pitch shifted clock mode: A deviation
of 12.5% of the nominal sampling frequency is possible.
The UDA1355H inputs support level I, II, and III as
specified by the IEC 60958 standard.
9.3.2RISE AND FALL TIMES
Rise and fall times (see Fig.14) are defined as:
t
Rise time =
Fall time =
--------------tLtH+
t
f
--------------tLtH+
r
100%×
100%×
Rise and fall times should be in the range:
• 0% to 20% when the data bit is a logic 1
• 0% to 10% when the data bits are two succeeding
logic 0.
9.3.3D
UTY CYCLE
The duty cycle (see Fig.14) is defined as:
t
Duty cycle =
H
--------------tLtH+
100%×
The duty cycle should be in the range:
• 40% to 60% when the data bit is a logic 1
• 45% to 55% when the data bits are two succeeding
logic 0.
10 L3-BUS DESCRIPTION
Theexchange of data and control information between the
microcontroller and the UDA1355H is accomplished
through a serial hardware L3-bus interface comprising the
following pins:
• MP0: mode line with signal L3MODE
• MP1: clock line with signal L3CLOCK
• MP2: data line with signal L3DATA.
The exchange of bytes in L3-bus mode is LSB first.
The L3-bus format has two modes of operation:
• Address mode
• Data transfer mode.
The address mode is used to select a device for a
subsequent data transfer. The address mode is
characterized by L3MODE being LOW and a burst of
8 pulses on L3CLOCK, accompanied by 8 bits (see
Fig.15). The data transfer mode is characterized by
L3MODE being HIGH and is used to transfer one or more
bytes representing a register address, instruction or data.
Basically two types of data transfers can be defined:
• Write action: data transfer to the device
• Read action: data transfer from the device.
handbook, halfpage
90%
50%
10%
t
H
t
r
t
L
t
f
MGU612
Fig.14 Rise, fall time and duty cycle.
2003 Apr 1032
10.1Device addressing
The device address consists of one byte with:
• Data Operating Mode (DOM) bits 0 and 1 representing
the type of data transfer (see Table 16)
• Address bits 2 to 7 representing a 6-bit device address.
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Table 16 Selection of data transfer
DOM BITS
BIT 0BIT 1
00not used
10not used
01write data or prepare read
11read data
ThedeviceaddressoftheUDA1355Hisgiven in Table 17,
being the first 6 bits of the device address byte. The
address can be set one of two by using pin MODE1
(pin A0 in microcontroller mode).
Table 17 L3-bus device address
MSBADDRESSLSB
00001A0
Remark: When using the device address, there is often
misunderstanding. This is caused by the fact that the data
is sent LSB first. This means that when we use the device
address in, for example the Philips L3-bus/I
bithacker’, we have to use the address like LSB → MSB.
For the UDA1355H this means that the device address to
be used is either 10H (010000) or 30H (110000).
10.2Register addressing
After sending the device address, including Data
Operating Mode (DOM) bits indicating whether the
information is to be read or written, one data byte is sent
using bit 0 to indicate whether the information will be read
or written and bits 1 to 7 for the destination register
address.
Basically there are three methods for register addressing:
• Addressing for write data: bit 0 is logic 0 indicating a
write action to the destination register, followed by
bits 1 to 7 indicating the register address (see Fig.15)
• Addressing for prepare read: bit 0 is logic 1 indicating
that data will be read from the register (see Fig.16)
• Addressing for data read action: in this case the device
returnsaregisteraddresspriortosendingdata from that
register. When bit 0 is logic 0, the register address is
valid; in case bit 0 is logic 1 the register address is
invalid.
TRANSFER
2
C-bus
10.3Data write mode
The data write mode is explained in the signal diagram of
Fig.15.
For writing data to a device, 4 bytes must be sent (see
Table 18):
• Byte 1 starting with 01 for signalling the write action to
the device, followed by the device address
• Byte 2 starting with 0 for signalling the write action,
followed by 7 bits indicating the destination address in
binary format with A6 being the MSB and A0 being the
LSB
• Byte 3 with bit D15 being the MSB
• Byte 4 with bit D0 being the LSB.
Itshould be noted that each time anewdestinationregister
address needs to be written, the device address must be
sent again.
10.4Data read mode
For reading data from the device, first a prepare read must
be done and then data read. The data read mode is
explained in the signal diagram of Fig.16.
For reading data from a device, the following 6 bytes are
involved (see Table 19):
• Byte 1 with the device address including 01 for
signalling the write action to the device
• Byte 2 is sent with the register address from which data
needstobe read. This byte starts with 1, which indicates
thattherewillbe a read action from the register, followed
again by 7 bits for the destination address in binary
format with A6 being the MSB and A0 being the LSB
• Byte 3with the device address including 11issentto the
device. The 11 indicates that the device must write data
to the microcontroller
• Byte 4, sent by the device to the bus, with the
(requested) register address and a flag bit indicating
whether the requested register was valid (bit is logic 0)
or invalid (bit is logic 1)
• Byte 5, sent by the device to the bus, with the data
information in binary format with D15 being the MSB
• Byte 6, sent by the device to the bus, with the data
information in binary format with D0 being the LSB.
2003 Apr 1033
Philips SemiconductorsPreliminary specification
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Stereo audio codec with SPDIF interfaceUDA1355H
MGS754
MGS753
data byte 1data byte 2
data byte 1data byte 2
register address
write
device address
10
0
DOM bits
Fig.15 Data write mode.
0/1
register addressdevice addressregister address
1
valid/non-valid
Fig.16 Data read mode.
read
prepare readsend by the device
This text is here in white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here in
_white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here inThis text is here in
white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader. white to force landscape pages to be ...
The bus is for 2-way, 2-line communication between
different ICs or modules. The two lines are a Serial Data
Line (SDA) and a Serial Clock Line (SCL). Both lines must
be connected to the supply voltage (VDD) via a pull-up
resistor when connected to the output stages of a
microcontroller. For a 400 kHz IC the recommendation for
this type of bus from Philips Semiconductors must be
11.2Bit transfer
One data bit is transferred during each clock pulse (see
Fig.17). The data on the SDA line must remain stable
during the HIGH period of the clock pulse as changes in
the data line at this time will be interpreted as control
signals. The maximum clock frequency is 400 kHz. To be
abletorunonthis high frequency all the inputs and outputs
connectedtothisbusmustbedesignedforthishighspeed
2
I
C-bus according the Philips specification.
followed (e.g. up to loads of 200 pF on the bus a pull-up
resistor can be used, between 200 to 400 pF a current
source or switched resistor must be used). Data transfer
can only be initiated when the bus is not busy.
handbook, full pagewidth
SDA
SCL
data line
stable;
data valid
Fig.17 Bit transfer on the I2C-bus.
2003 Apr 1035
change
of data
allowed
MBC621
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
11.3Byte transfer
Each byte (8 bits) is transferred with the MSB first (see
Table 20).
Table 20 Byte transfer
MSBBITLSB
76543210
11.4Data transfer
A device generating a message is a transmitter; a device
receiving a message is the receiver. The device that
controls the message is the master and the devices which
are controlled by the master are the slaves.
11.5Register address
2
The register addresses in the I
C-bus mode are the same
as in the L3-bus mode.
11.6Device address
Before any data is transmitted on the I2C-bus, the device
whichshouldrespondis addressed first. The addressing is
always done with the first byte transmitted after the start
procedure. The device address can be one of two, being
set by bit A0 which corresponds to pin MODE1.
The UDA1355H acts as a slave receiver or a slave
transmitter. Therefore, the clock signal SCL is only an
input signal. The data signal SDA is a bidirectional line.
The UDA1355H slave address is shown in Table 21.
11.7Start and stop conditions
Both data and clock line will remain HIGH when the bus in
not busy. A HIGH-to-LOW transition of the data line, while
the clock is HIGH, is defined as a start condition (S).
A LOW-to-HIGH transition of the data line while the clock
is HIGH is defined as a stop condition (P); (see Fig.18).
11.8Acknowledgment
The number of data bits transferred between the start and
stop conditions from the transmitter to receiver is not
limited. Each byte of eight bits is followed by one
acknowledge bit (see Fig.19). At the acknowledge bit the
data line is released by the master and the master
generates an extra acknowledge related clock pulse.
A slave receiver which is addressed must generate an
acknowledge after the reception of each byte. Also a
master must generate an acknowledge after the reception
of each byte that has been clocked out of the slave
transmitter.
The device that acknowledges has to pull-down the SDA
line during the acknowledge clock pulse, so that the SDA
line is stable LOW during the HIGH period of the
acknowledge related clock pulse. Set-up and hold times
must be taken into account. A master receiver must signal
an end of data to the transmitter by not generating an
acknowledge on the last byte that has been clocked out of
the slave. In this event the transmitter must leave the data
line HIGH to enable the master to generate a stop
condition.
2
Table 21 I
C-bus slave address
DEVICE ADDRESSR/
W
A6A5A4A3A2A1A0−
001101A00/1
handbook, full pagewidth
SDA
SCL
S
START condition
Fig.18 START and STOP conditions on the I2C-bus.
2003 Apr 1036
P
STOP condition
SDA
SCL
MBC622
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
handbook, full pagewidth
DATA OUTPUT
BY TRANSMITTER
not acknowledge
DATA OUTPUT
BY RECEIVER
acknowledge
SCL FROM
MASTER
S
START
condition
Fig.19 Acknowledge on the I2C-bus.
11.9Write cycle
The write cycle is used to write groups of two bytes to the
internal registers for the digital sound feature control and
system setting. It is also possible to read these locations
for chip status information.
The I2C-bus configuration for a write cycle is shown in
Table 22. The write cycle is used to write the data to the
internal registers. The device and register addresses are
one byte each, the setting data is always a couple of two
bytes.
The format of the write cycle is as follows:
1. The microcontroller starts with a start condition (S).
2. The first byte (8 bits) contains the device address
0011010 and a logic 0 (write) for the R/W bit.
3. This is followed by an acknowledge (A) from the
UDA1355H.
9821
clock pulse for
acknowledgement
MBC602
4. After this the microcontroller writes the 8-bit register
address (ADDR) where the writing of the register
content of the UDA1355H must start.
5. The UDA1355H acknowledges this register address
(A).
6. The microcontroller sends two bytes data with the
Most Significant (MS) byte first and then the Least
Significant (LS) byte. After each byte an acknowledge
is followed from the UDA1355H.
7. If repeated groups of two bytes are transmitted, then
the register address is auto incremented. After each
byte an acknowledge is followed from the
microcontroller.
8. Finally, the UDA1355H frees the I2C-bus and the
microcontroller can generate a stop condition (P).
2
Table 22 Master transmitter writes to the UDA1355H registers in the I
DEVICE
ADDRESS
R/W
REGISTER
ADDRESS
DATA 1DATA 2
C mode.
(1)
DATA n
(1)
S00110100AADDRAMS1ALS1A....A.....AMSnALSnAP
acknowledge from UDA1355H
Note
1. Auto increment of register address.
2003 Apr 1037
Philips SemiconductorsPreliminary specification
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Stereo audio codec with SPDIF interfaceUDA1355H
(1)
DATA n
(1)
C-bus configuration for a read cycle is shown in Table 23
2
R/WDATA 1DATA 2
C-bus mode
2
DEVICE
ADDRESS
C-bus and the microcontroller can generate a stop condition (P).
2
ADDRESS
REGISTER
acknowledge from UDA1355Hacknowledge from master
R/W
This text is here in white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here in
_white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here inThis text is here in
white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader. white to force landscape pages to be ...
11.10 Read cycle
The read cycle is used to read the data values from the internal registers. The I
2003 Apr 1038
2003 Apr 1038
(A) follows from the UDA1355H.
acknowledged follows from the microcontroller.
microcontroller.
The format of the read cycle is as follows:
1. The microcontroller starts with a start condition (S).
2. The first byte (8 bits) contains the device address 0011010 and a logic 0 (write) for the R/W bit.
3. This is followed by an acknowledge (A) from the UDA1355H.
4. After this microcontroller writes the 8-bit register address (ADDR) where the reading of the register content of the UDA1355H must start.
5. The UDA1355H acknowledges this register address (A).
6. Then the microcontroller generates a repeated start (Sr).
7. Then the microcontroller generates the device address 0011010 again, but this time followed by a logic 1 (read) of the R/W bit. An acknowledge
8. The UDA1355H sends two bytes data with the Most Significant (MS) byte first and then the Least Significant (LS) byte. After each byte an
9. If repeated groups of two bytes are transmitted, then the register address is auto incremented. After each byte an acknowledge follows from the
10. The microcontroller stops this cycle by generating a Negative Acknowledge (NA).
DEVICE
ADDRESS
Table 23 Master transmitter reads from the UDA1355H registers in the I
11. Finally, the UDA1355H frees the I
S00110100AADDRASr00110101AMS1 A LS1A ... A ... A MSn ALSnNAP
Note
1. Auto increment of register address.
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
12 REGISTER MAPPING
In this chapter the register addressing of the microcontroller interface of the UDA1355H is given. In Section 12.1, the
mapping of the readable and writable registers is given. The explanation of the register definitions are explained in
Sections 12.2 and 12.3.
12.1Address mapping
Table 24 Register map settings
ADDRESSR/WDESCRIPTION
System settings
00HR/Wcrystal clock power-on setting; crystal clock and PLL divider settings; MODE and WS detector
settings; clock output setting
01HR/WI2S-bus output format settings
02HR/WI
03HR/Wreserved for manufacturers evaluation and should be kept untouched for normal operation
04HR/Wanalog power and clock settings
Interpolator
10HR/Wmaster volume control settings
11HR/Wmixer volume settings
12HR/Wsound feature and bass boost and treble settings
13HR/Wgain select; de-emphasis and mute settings
14HR/WDACpolarity; noise shaper selection; mixer; source selection; silence detector and interpolator
20HR/WADC gain settings
21HR/WADC mute and PGA gain settings;
22HR/WADC polarity and DC cancellation settings
28HRmute status and overflow ADC read-out
SPDIF input
30HR/WSPDIF power control and SPDIF input settings
40HR/Wreserved for manufacturers evaluation and should be kept untouched for normal operation
59HRSPDIF LOCK; bit error information and SPDIF encoder output status read-out
5AHRSPDIF input status bits 15 to 0 left channel read-out
5BHRSPDIF input status bits 31 to 16 left channel read-out
5CHRSPDIF input status bits 39 to 32 left channel read-out
2
S-bus input format settings
oversampling settings
2003 Apr 1039
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
ADDRESSR/WDESCRIPTION
5DHRSPDIF input status bits 15 to 0 right channel read-out
5EHRSPDIF input status bits 31 to 16 right channel read-out
5FHRSPDIF input status bits 39 to 32 right channel read-out
SPDIF output
50HR/WSPDIF output valid; left to right channel status bit copy; power control and SPDIF output
selection setting
51HR/WSPDIF output status bits 39 to 24 left channel setting
52HR/WSPDIF output status bits 23 to 8 left channel setting
53HR/WSPDIF output status bits 7 to 0 left channel setting
54HR/WSPDIF output status bits 39 to 24 right channel setting
55HR/WSPDIF output status bits 23 to 8 right channel setting
56HR/WSPDIF output status bits 7 to 0 right channel setting
60HR/Wreserved for manufacturers evaluation and should be kept untouched for normal operation
61HR/Wreserved for manufacturers evaluation and should be kept untouched for normal operation
62HR/Wreserved for manufacturers evaluation and should be kept untouched for normal operation
63HR/Wreserved for manufacturers evaluation and should be kept untouched for normal operation
64HR/Wreserved for manufacturers evaluation and should be kept untouched for normal operation
Table 26 Description of register bits (address 00H)
BITSYMBOLDESCRIPTION
15EXPUEXPU. Bit EXPU is reserved for manufacturers evaluation and should be kept
untouched for normal operation of UDA1355H.
14−reserved
13PON_XTALPLLPower control crystal oscillator and PLL. If this bit is logic 0, then the crystal
oscillator and PLL are turnedoff; if this bit is logic 1, then the crystal oscillator and PLL
are running.
12 to 8XTL_DIV[4:0]Crystal oscillator clock divider setting. Value to select the sampling frequency and
the system clock output frequency (256f
BCKI and BCKO clock frequency of digital interface running with crystal oscillator clock
will be 64fs; when 384fs is selected, it will be 48fs (see Table 27).
7 to 4MODE[3:0]Microcontroller application mode setting. Value to select the microcontroller
application mode (see Table 28).
3ws_detct_ENWord select detector enable.If this bit is logic 0, then WS detector is disabled; if this
bit is logic 1, then WS detector is enabled.
2ws_detct_setWord select detector limit setting. If this bit is logic 0, then the lower frequency limit
of the WS detector is 4095 clock cycles (3 kHz); if this bit is logic 1, then the lower
frequency limit of the WS detector is 2047 clock cycles (6 kHz).
1 and 0 CLKOUT_SEL[1:0] Clock output select. If these bits are 00 or 10, then the BCKI and BCKO clock
frequency of digital interface running with FPLL clock will be 64f
48fs. The selection between 256fsand 384fs for the crystal clock output is set via the
Table 30 Description of register bits (address 01H)
BITSYMBOLDESCRIPTION
15 to 9−reserved
8MUTE_DAODigital mute setting. If this bit is logic 0, then the digital output is not muted; if this bit is
logic 1, then the digital output is muted.
7PON_DIGOPower control digital output. If this bit is logic 0, then the digital output is in Power-down
mode; if this bit is logic 1, then the digital output is in power-on mode. The registers have
their own clock, which means that there cannot be a dead-lock situation.
6−reserved
5 and 4 DIGOUT[1:0]Input selector for digital output. Value to select the input signal for the digital output. The
default input will be chosen if in an application an invalid data signal is selected:
Table 32 Description of register bits (address 02H)
BITSYMBOLDESCRIPTION
15 to 8−reserved
7PON_DIGIPower control digital input. If this bit is logic 0, then the digital input is in Power-down
mode; if this bit is logic 1, then the digital input is in power-on mode. The registers have their
own clock, which means that there cannot be a dead-lock situation.
6to3 −reserved
2003 Apr 1043
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
BITSYMBOLDESCRIPTION
2 to 0SFORI[2:0]Digital input format. Value to set the digital input format:
2
000 = I
001 = LSB-justified; 16 bits
010 = LSB-justified; 18 bits
011 = LSB-justified; 20 bits
100 = LSB-justified; 24 bits
101 = MSB-justified
110 = not used; input is default value
111 = not used; input is default value
Table 34 Description of register bits (address 04H)
BITSYMBOLDESCRIPTION
15PON_DACPower control DAC. If this bit is logic 0, then the DAC is in Power-down mode; if this bit
14 to 11 −reserved
10PON_ADCLPower control ADC left channel. Value to set power on the ADC left channel (see
9PON_ADCRPower control ADC right channel. Value to set power on the ADC right channel (see
8PON_ADC_biasPower control ADC bias. Value to set power on the ADCs (see Table 35).
7DACLK_OFFDAC clock enable. If this bit is logic 0, then the DAC clock is disabled; if this bit is
6DACLK_AUTODAC clock auto function. If this bit is logic 0, then the DAC clock auto function is
5to3 −reserved
2EN_DECDecimator and ADC clock enable. If this bit is logic 0, then the clock to decimator and
1−reserved
0EN_INTInterpolator clock enable. If this bit is logic 0, then the clock to interpolator and FSDAC
7654321
is logic 1, then the DACis in power-on mode. This bit is only connected to the DACinput
and is not combined with mute status or other signals.
Table 35).
Table 35).
logic 1, then the DAC clock is enabled.
disabled; if this bit is logic 1, then the DAC clock auto function is enabled. If the FPLL is
unlocked, the interpolator will be muted and the DAC clock is automatically disabled.
ADC is disabled; if this bit is logic 1, then the clock to decimator and ADC is running.
is disabled; if this bit is logic 1, then the clock to the interpolator and FSDAC is running.
0
2003 Apr 1044
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Table 35 ADC power control
PON_ADC_BIASPON_ADCRPON_ADCLDESCRIPTION
0XXno power on both ADCs
100no power on both ADCs
110only power on right channel ADC
101only power on left channel ADC
111power on both ADCs
Table 43 Description of register bits (address 12H)
BITSYMBOLDESCRIPTION
15 and 14M[1:0]Sound feature mode. Value to program the sound processing filter sets (modes) of bass
boost and treble:
00 = flat set
01 = minimum set
10 = minimum set
11 = maximum set
13 and 12TRL[1:0]Treble settings left. Value to program the left channel treble setting. Both left and right
channels will follow the left channel setting when bit BASS_SEL = 1. The used filter set is
selected with the sound feature mode bits M1 and M2 (see Table 44).
11 to 8BBL[3:0]Normal bass boost settings left. Value to program the left bass boost settings. The
used filter set is selected by the sound feature mode bits M1 and M2 (see Table 45).
7BB_OFFResonant bass boost. If this bit is logic 0 then the resonant bass boost is enabled; if this
bit is logic 1 then the resonant bass boost is disabled.
2003 Apr 1047
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
BITSYMBOLDESCRIPTION
6BB_FIXResonant bass boost coefficient. If this bit is logic 0 then the resonant bass boost
coefficient is finished loading; if this bit is logic 1 then the resonant bass boost coefficient
is loading to register.
5 and 4TRR[1:0]Treble settings right. Value to program the right treble setting. The used filter set is
selected by the sound feature mode bits M1 and M2 (see Table 44).
3 to 0BBR[3:0]Normal bass boost settings right. Value to program the right bass boost settings. The
used filter set is selected by the sound feature mode bits M1 and M2 (see Table 45).
Table 47 Description of register bits (address 13H)
BITSYMBOLDESCRIPTION
15-reserved
14MTMMaster mute. If this bit is logic 0 then there is no master mute or the master de-mute is in
progress; if this bit is logic 1 then the master mute is in progress or muted.
13GSGain select. See Table 48.
12MIXGAINMixer gain select. See Tables 48 and 49.
11MT2Channel 2 mute. If this bit is logic 0 then channel 2 is not muted or the de-mute is in
progress; if this bit is logic 1 then channel 2 is muted or the muting is in progress.
10 to 8DE2_[2:0]De-emphasis setting for channel 2. See Table 50.
7 and 6MTNS[1:0]Interpolator mute. Selection:
00 = no mute
01 = if no WS signal is detected, the noise shaper of the interpolator mute
1x = the noise shaper of the interpolator mute
5WS_SELWS signal select. If this bit is logic 0 then WS_DET is selected for the WS detection; if
this bit is logic 1 then FPLL is selected for the WS detection.
4DE_SWDe-emphasis select. If this bit is logic 0 then SPDIF pre-emphasis information is
selected; if this bit is logic 1 then the de-emphasis setting is selected.
3MT1Channel 1 mute. If this bit is logic 0 then channel 1 is not muted or the de-mute is in
progress; if this bit is logic 1 then channel 1 is muted or the muting is in progress.
2 to 0DE1_[2:0]De-emphasis setting for channel 1. See Table 50.
Table 48 DAC gain setting
GSMIX
0X
(1)
(2)
1006
1100
1016
1116
Notes
1. See Table 52.
2. X = don’t care
2003 Apr 1049
MIX_GAINDAC GAIN (dB)
(2)
X
0
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Table 49 Mixer gain setting
(1)
MIX
10DAC output gain is set to 0 dB and mixer signal output gain is set −6dB
11DAC output gain and mixer signal output gain are set to 0 dB
Note
1. See Table 52.
Table 50 De-emphasis setting for the incoming signal
MIX_GAINMIXER OUTPUT GAIN
DE2_2DE2_1DE2_0
DE1_2DE1_1DE1_0
000off
00132 kHz
01044.1 kHz
01148 kHz
10096 kHz
Table 51 Register address 14H
BIT15141312111098
SymbolDA_POL_
INV
Default01001101
BIT765 4 3210
SymbolSILENCESDET_ONSD_
Default00000000
Table 52 Description of register bits (address 14H)
SEL_NSMIX_POSMIXDAC_CH2_
SEL1
VALUE1
SD_
VALUE0
BASS_SELBYPASSOS_IN1OS_IN0
FUNCTION
DAC_CH2_
SEL0
DAC_CH1_
SEL1
DAC_CH1_
SEL0
BITSYMBOLDESCRIPTION
15DA_POL_INVDAC polarity control. If this bit is logic 0 then the DAC output is not inverted; if this
bit is logic 1 then the DAC output is inverted.
14SEL_NSSelect noise shaper. If this bit is logic 0 then the third order noise shaper is
selected; if this bit is logic 1 then the fifth order noise shaper is selected.
13MIX_POSMixer position. Mixing is done before or after the sound processing unit (see
Table 53).
12MIXMixer. If this bit is logic 0 then the mixer is disabled; if this bit is logic 1 then the mixer
is enabled (see Tables 48, 49 and 53).
11 and10DAC_CH2_SEL[1:0]DAC channel 2 input selection. Value to select the input mode to channel 2 of the
interpolator (see Table 54).
9 and 8 DAC_CH1_SEL[1:0]DAC channel 1 input selection. Value to select the input mode to channel 1 of the
interpolator (see Table 54).
2003 Apr 1050
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
BITSYMBOLDESCRIPTION
7SILENCESilence detector overrule. Value to force the DAC output to silence. This will give a
plop at the output of the DAC because of mismatch in offsets and the DC offset
added to the signal in the interpolator itself. If this bit is logic 0 then there is no
overruling and the FSDAC silence switch setting depends on the silence detector
circuit and on the status of bit MTM; if this bit is logic 1 then there is overruling and
the FSDAC silence switch is activated independent of the status of the digital silence
detector circuit or the status of bit MTM.
6SDET_ONSilence detector enable. If this bit is logic 0 then the silence detection circuit is
disabled; if this bit is logic 1 then the silence detection circuit is enabled.
5 and 4 SD_VALUE[1:0]Silence detector setting. Value to program the silence detector. The number of zero
samples counted before the silence detector signals whether there has been digital
silence:
3BASS_SELBass boost select. If this bit is logic 0 then the normal bass boost function is
selected; if this bit is logic 1 then the resonant bass boost function is selected.
2BYPASSMixer bypass mode. If this bit is logic 0 then the mixer is in mixer mode; if this bit is
logic 1 then the mixer is in mixer bypass mode.
1 and 0 OS_IN[1:0]Oversampling ratio select. Valueto select the oversampling input mode. This mode
is only for I2S-bus input:
00 = single speed input; normal input; mixing possible
01 = double speed input; after first half-band filter; no mixing possible but volume
and mute still possible
10 = quad speed input; in front of noise shaper; no mixing possible; no volume
control possible
11 = reserved.
Table 53 Mixer signal control signals
MIXMIX_POSFUNCTION
0X
10mixing is done before the sound processing; input signals are automatically scaled
11mixing is done after the sound processing; input signals are automatically scaled in
Note
1. X = don’t care
2003 Apr 1051
(1)
this is the default setting: no mixing, volume of channel 1 is forced to 0 dB and
volume of channel 2 is forced to −∞ dB
by 6 dB in order to prevent clipping during adding; after the addition, the 6 dB scaling
is compensated
order to prevent clipping during adding
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Table 54 Data source selector DAC channel 1 and 2; note 1
DAC_CH2_SEL1DAC_CH2_SEL0
DATA OUTPUT DAC
DAC_CH1_SEL1DAC_CH1_SEL0
00ADC input
01I
2
S-bus input
10IEC 60958 input
2
11I
S-bus input
Note
1. The change of the data source will take place only when the mix mode is turned on (bit MIX = 1).
The channel 2 input selection is valid only when the channel 1 data source is correct.
Table 56 Description of register bits (addresses 19H, 1AH, 1BH, 1CH, 1DH and 1EH)
BITSYMBOLDESCRIPTION
15 and 14 −reserved
13 to 0BASS_x_[13:0]Resonant bass boost coefficient x. Six 14-bit registers are used as the filter
coefficients to specify the bass boost characteristics. The six coefficients are k1, km,
a1, a2, b1 and b2m. A software program is available for users to generate these six
14-bit coefficients by entering the desired centre frequency, peak gain, sampling
frequency and shape factor (default flat response).
12.2.3D
ECIMATOR SETTINGS
Table 57 Register address 20H
BIT15141312111098
SymbolMA_
DECL7
MA_
DECL6
MA_
DECL5
MA_
DECL4
MA_
DECL3
MA_
DECL2
MA_
DECL1
MA_
DECL0
Default00000000
BIT76543210
SymbolMA_
DECR7
MA_
DECR6
MA_
DECR5
MA_
DECR4
MA_
DECR3
MA_
DECR2
MA_
DECR1
MA_
DECR0
Default00000000
2003 Apr 1052
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Table 58 Description of register bits (address 20H)
BITSYMBOLDESCRIPTION
15 to 8MA_DECL[7:0]ADC volume setting left channel. Value to program the ADC gain setting for the left
channel. The range is from +24 to −63 dB and −∞ dB (see Table 59).
7 to 0MA_DECR[7:0]ADC volume setting right channel. Valueto program the ADC gain setting for the right
channel. The range is from +24 to −63 dB and −∞ dB (see Table 59).
Table 61 Description of register bits (address 21H)
BITSYMBOLDESCRIPTION
15MT_ADCMute ADC. If this bit is logic 0 then the ADC is not muted; if this bit is logic 1 then
the ADC is muted.
14 to 12 −reserved
11 to 8PGA_GAIN_CTRLL[3:0]PGA gain control left channel. Value to program the gain of the left input
amplifier. There are nine settings (see Table 62).
7to4 −reserved
3 to 0PGA_GAIN_CTRLR[3:0]PGA gain control right channel. Value to program the gain of the right input
Table 64 Description of register bits (address 22H)
BITSYMBOLDESCRIPTION
15 to 13 −reserved
12ADCPOL_INVADC polarity control. If this bit is logic 0 then the ADC input is not inverted; if this bit is
logic 1 then the ADC input is inverted.
2003 Apr 1054
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
BITSYMBOLDESCRIPTION
11 to 2−reserved
1DC_SKIPDC filter skip. If this bit is logic 0 then the DC filter is enabled; if this bit is logic 1 then the
DC filter is disabled. The DC filter is at the output of the comb filter just before the
decimator.This DC filter compensates for the DC offset added in the ADC (to remove idle
tones from the audio band). This DC offset must not be amplified in order to prevent
clipping.
0HP_EN_DECHigh-pass enable. If this bit is logic 0 then the high-pass is disabled; if this bit is logic 1
then the high-pass is enabled. The high-pass is a DC filter which is at the output of the
decimation filter (running at f
Table 68 Description of register bits (address 50H)
BITSYMBOLDESCRIPTION
15 to 9 −reserved
8SPDO_VALIDSDPDIF output valid. If this bit is logic 0 then the SPDIF output is invalid; if this bit is
logic 1 then the SPDIF output is valid.
7−reserved
6L_r_copySPDIF channel status copy. If this bit is logic 0 then the status bits of the left channel
are not copied to the right channel; if this bit is logic 1 then the status bits of the left
channel are copied to the right channel.
5−reserved
4PON_SPDOPowercontrol of SPDIF output. If this bit is logic 0 then the SPDIF output is switched
to Power-down mode; if this bit is logic 1 then the SPDIF output is switched to
power-on mode.
3DIS_SPDOSPDIF encoder enable. If this bit is logic 0 then the SPDIF encoder is enabled; if this
bit is logic 1 then the SPDIF encoder is disabled.
2 to 0SPDOUT_SEL[2:0]SPDIF output source selector. Valueto select the input source forSPDIF output. The
selection option to select the SPDIF input just after the slicer was already there. Added
is an independent selection of the input signals SPDIF0 to SPDIF3:
000 = ADC
2
001 = I
010 = not used
011 = interpolator mix output
100 = SPDIF0 loop through
101 = SPDIF1 loop through
110 = SPDIF2 loop through
111 = SPDIF3 loop through
S-bus input
2003 Apr 1056
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Table 69 Register addresses 51H (left) and 54H (right)
BIT15141312111098
SymbolSPDO_
BIT39
Default00000000
BIT76543210
SymbolSPDO_
BIT31
Default00000000
Table 70 Register addresses 52H (left) and 55H (right)
BIT15141312111098
SymbolSPDO_
BIT23
Default00000000
SPDO_
BIT38
SPDO_
BIT30
SPDO_
BIT22
SPDO_
BIT37
SPDO_
BIT29
SPDO_
BIT21
SPDO_
BIT36
SPDO_
BIT28
SPDO_
BIT20
SPDO_
BIT35
SPDO_
BIT27
SPDO_
BIT19
SPDO_
BIT34
SPDO_
BIT26
SPDO_
BIT18
SPDO_
BIT33
SPDO_
BIT25
SPDO_
BIT17
SPDO_
BIT32
SPDO_
BIT24
SPDO_
BIT16
BIT76543210
SymbolSPDO_
BIT15
Default00000000
Table 71 Register addresses 53H (left) and 56H (right)
BIT15141312111098
Symbol−−−−−−−−
Default00000000
BIT76543210
SymbolSPDO_
BIT7
Default00000000
Table 72 Description of register bits
BITSYMBOLDESCRIPTION
39 to 36SPDO_BIT[39:36]reserved
35 to 33SPDO_BIT[35:33]Word length. Value indicating the word length (see Table 73).
32SPDO_BIT[32]Audio sample word length. Valueto signal the maximum audio sample word length.
31 to 30SPDO_BIT[31:30]reserved
SPDO_
BIT14
SPDO_
BIT6
If bit 32 is logic 0, then the maximum length is 20 bits; if bit 32 is logic 1, then the
maximum length is 24 bits (see Table 73).
SPDO_
BIT13
SPDO_
BIT5
SPDO_
BIT12
SPDO_
BIT4
SPDO_
BIT11
SPDO_
BIT3
SPDO_
BIT10
SPDO_
BIT2
SPDO_
BIT9
SPDO_
BIT1
SPDO_
BIT8
SPDO_
BIT0
2003 Apr 1057
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
BITSYMBOLDESCRIPTION
29 to 28SPDO_BIT[29:28]Clock accuracy. Value indicating the clock accuracy:
00 = level II
01 = level I
10 = level III
11 = reserved
27 to 24SPDO_BIT[27:24]Sample frequency. Value indicating the sampling frequency:
0000 = 44.1 kHz
0001 = 48 kHz
0010 = 32 kHz
other states = reserved
23 to 20SPDO_BIT[23:20]Channel number. Value indicating the channel number (see Table 74).
19 to 16SPDO_BIT[19:16]Source number. Value indicating the source number (see Table 75).
15 to 8SPDO_BIT[15:8]General information. Value indicating general information (see Table 76).
7 to 6SPDO_BIT[7:6]Mode. Value indicating mode 0:
00 = mode 0
other states = reserved
5 to 3SPDO_BIT[5:3]Audio sampling. Value indicating the type of audio sampling (linear PCM). For
bit SPDO_BIT1 = 0:
000 = two audio samples without pre-emphasis
001 = two audio samples with 50/15 µs pre-emphasis
010 = reserved (two audio samples with pre-emphasis)
011 = reserved (two audio samples with pre-emphasis)
other states = reserved
2SPDO_BIT2Software copyright. Value indicating software for which copyright is asserted or not.
If this bit is logic 0, then copyright is asserted; if this bit is logic 1, then no copyright is
asserted.
1SPDO_BIT1Audio sample word. Value indicating the type of audio sample word. If this bit is
logic 0, then the audio sample word represents linear PCM samples; if this bit is
logic 1, then the audio sample word is used for other purposes.
0SPDO_BIT0Channel status. Value indicating the consumer use of the status block. This bit is
SPDO_BIT19SPDO_BIT18SPDO_BIT17SPDO_BIT16SOURCE NUMBER
101111
110012
110113
111014
111115
Table 76 General information
SPDO_BIT[15:8]FUNCTION
00000 000general
Lxxxx 001laser optical products
Lxxxx 010digital-to-digital converters and signal processing products
Lxxxx 011magnetic tape or disc based products
Lxxxx 100broadcast reception of digitally encoded audio signals with video signals
Lxxxx 110broadcast reception of digitally encoded audio signals without video signals
Lxxxx 101musical instruments, microphones and other sources without copyright
information
Lxx00 110analog-to-digital converters for analog signals without copyright information
Lxx10 110analog-to-digital converters for analog signals which include copyright
information in the form of Cp- and L-bit status
Lxxx1 000solid state memory based products
L1000 000experimental products not for commercial sale
Lxxxx 111reserved
Lxxx0 000reserved, except 000 0000 and 000 0001L
(rms)input voltage (RMS value)Vo= −1.16 dBFS digital
V
i
∆V
i
input voltage unbalance−0.1−dB
(THD+N)/Stotal harmonic
distortion-plus-noise to
signal ratio
S/Nsignal-to-noise ratiocode = 0; A-weighted
αcschannel separation−100−dB
with respect to V
IEC 60958 input; f
SSA
= 48 kHz
s
0.45V
DD
0.5V
DD
0.55V
DD
at 0 dB−−88−dB
at −20 dB−−75−dB
at −60 dB; A-weighted−−37−dB
IEC 60958 input; fs= 96 kHz
at 0 dB−−83−dB
at −60 dB; A-weighted−−37−dB
A-weighted
f
= 48 kHz−98−dB
s
f
= 96 kHz−96−dB
s
−tbf−mA
RL=5kΩ
−V
DDA2
−V
−0.0−V
−1.0−V
output
f
= 48 kHz
s
at 0 dB−−85−dB
at −60 dB; A-weighted−−35−dB
f
= 96 kHz
s
at 0 dB−−85−dB
at −60 dB; A-weighted−−35−dB
fs= 48 kHz−97−dB
f
= 96 kHz−95−dB
s
V
2003 Apr 1065
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX.UNIT
IEC 60958 inputs
V
i(p-p)
input voltage (peak-to-peak
value)
R
i
V
hys
I
DD(diff)
input resistance−6−kΩ
hysteresis voltage−40−mV
I
DD(DAC,input)/IDD(DAC,no input)
Power consumption
P
tot
total power consumptionIEC 60958 input; fs= 48 kHz
DAC in playback mode−74−mW
DAC in Power-down mode −63−mW
Notes
1. All power supply pins (VDD and VSS) must be connected to the same external power supply unit.
2. When the DAC must drive a higher capacitive load (above 50 pF), then a series resistor of 100 Ω must be used in
order to prevent oscillations in the output.
0.20.53.3V
−tbf−−
16 TIMING CHARACTERISTICS
VDD= 2.7 to 3.6 V; T
= −20 to +85 °C; RL=5kΩ; unless otherwise specified.
amb
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX. UNIT
Device reset
t
rst
reset time−250−µs
PLL lock time
t
lock
2
S-bus interface (see Fig.20)
I
T
cy(BCK)
t
BCKH
t
BCKL
t
r
t
f
t
su(DATAI)
t
h(DATAI)
t
d(DATAO-BCK)
time-to-lockfs= 32 kHz−85.0−ms
bit clock period
bit clock HIGH time30−−ns
bit clock LOW time30−−ns
rise time−−20ns
fall time−−20ns
data input set-up time10−−ns
data input hold time10−−ns
data output to bit clock
delay
t
d(DATAO-WS)
data output to word
select delay
t
h(DATAO)
t
su(WS)
t
h(WS)
data output hold time0−−ns
word select set-up time10−−ns
word select hold time10−−ns
f
= 44.1 kHz−63.0−ms
s
= 48 kHz−60.0−ms
f
s
f
= 96 kHz−40.0−ms
s
1
/
128fs
−−ms
−−30ns
−−30ns
2003 Apr 1066
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX. UNIT
L3-bus interface (see Figs 21 and 22)
t
r
t
f
T
cy(CLK)L3
t
CLK(L3)H
t
CLK(L3)L
t
su(L3)A
t
h(L3)A
t
su(L3)D
t
h(L3)D
t
stp(L3)
t
su(L3)DA
t
h(L3)DA
t
d(L3)R
t
dis(L3)R
2
C-bus interface (see Fig.23)
I
f
SCL
t
LOW
t
HIGH
t
r
t
f
t
HD;STA
t
SU;STA
t
SU;STO
t
BUF
t
SU;DAT
rise timenote 1−−10ns/V
fall timenote 1−−10ns/V
L3CLOCK cycle timenote 2500−−ns
L3CLOCK HIGH timenote 2250−−ns
L3CLOCK LOW timenote 2250−−ns
L3MODE set-up time in
190−−ns
address mode
L3MODE hold time in
190−−ns
address mode
L3MODE set-up time in
190−−ns
data transfer mode
L3MODE hold time in
190−−ns
data transfer mode
L3MODE stop time in
190−−ns
data transfer mode
L3DATA set-up time in
190−−ns
address and data
transfer mode
L3DATA hold time in
30−−ns
address and data
transfer mode
L3DATA delay time in
0−50ns
data transfer mode
L3DATAdisable time for
0−50ns
read data
SCL clock frequency0−400kHz
SCL LOW time1.3−−µs
SCL HIGH time0.6−−µs
rise time SDA and SCL note 320 + 0.1Cb−300ns
fall time SDA and SCLnote 320 + 0.1Cb−300ns
hold time START
note 40.6−−µs
condition
set-up time repeated
0.6−−µs
START
set-up time STOP
0.6−−µs
condition
bus free timebetween a STOP and START
1.3−−µs
condition
data set-up time100−−ns
2003 Apr 1067
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
SYMBOLPARAMETERCONDITIONSMIN.TYP.MAX. UNIT
t
HD;DAT
t
SP
C
L
Notes
1. In order to prevent digital noise interfering with the L3-bus communication, the rise and fall times should be as small
as possible.
2. When the sampling frequency is below 32 kHz, the L3CLOCK cycle must be limited to1⁄
3. Cb is the total capacitance of one bus line in pF. The maximum capacitive load for each bus line is 400 pF.
4. After this period, the first clock pulse is generated.
5. To be suppressed by the input filter.
data hold time0−−µs
pulse width of spikesnote 50−50ns
load capacitancefor each bus line−−400pF
cycle.
64fs
handbook, full pagewidth
WS
BCK
DATAO
DATAI
t
BCKH
t
r
T
cy(BCK)
t
f
t
BCKL
t
h(WS)
t
d(DATAO-WS)
t
su(WS)
t
h(DATAO)
t
su(DATAI)
t
d(DATAO-BCK)
t
h(DATAI)
MGS756
Fig.20 I2S-bus interface timing.
2003 Apr 1068
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
handbook, full pagewidth
L3MODE
L3CLOCK
L3DATA
t
h(L3)A
t
BIT 0
su(L3)A
t
su(L3)DA
t
CLK(L3)L
t
CLK(L3)H
t
h(L3)DA
Fig.21 L3-bus interface timing for address mode.
t
su(L3)A
t
h(L3)A
T
cy(CLK)(L3)
BIT 7
MGL723
handbook, full pagewidth
L3MODE
L3CLOCK
L3DATA
L3DATA
write
read
t
t
su(L3)D
stp(L3)
t
h(L3)DA
BIT 0
t
CLK(L3)H
t
CLK(L3)L
t
d(L3)R
Fig.22 L3-bus interface timing for data transfer mode (write and read).
2003 Apr 1069
t
su(L3)DA
T
cy(CLK)L3
t
BIT 7
t
dis(L3)R
h(L3)D
MBL566
Philips SemiconductorsPreliminary specification
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
Stereo audio codec with SPDIF interfaceUDA1355H
P
MBC611
SU;STO
SP
t
HD;STA
t
t
f
t
r
t
LOW
t
Sr
SU;STA
t
SU;DAT
t
HIGH
t
HD;DAT
t
HD;STA
t
handbook, full pagewidth
C-bus interface timing.
2
Fig.23 I
This text is here in white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here in
_white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader.This text is here inThis text is here in
white to force landscape pages to be rotated correctly when browsing through the pdf in the Acrobat reader. white to force landscape pages to be ...
2003 Apr 1070
2003 Apr 1070
SDA
BUF
t
S
P
SCL
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
17 PACKAGE OUTLINE
QFP44: plastic quad flat package; 44 leads (lead length 1.3 mm); body 10 x 10 x 1.75 mm
c
y
X
A
3323
34
pin 1 index
44
1
22
Z
E
e
H
E
E
w M
b
p
12
11
A
2
A
A
1
detail X
SOT307-2
(A )
3
θ
L
p
L
w M
b
e
DIMENSIONS (mm are the original dimensions)
mm
OUTLINE
VERSION
SOT307-2
A
max.
2.1
0.25
0.05
1.85
1.65
UNITA1A2A3bpcE
Note
1. Plastic or metal protrusions of 0.25 mm maximum per side are not included.
p
D
H
D
0.4
0.2
0.25
0.14
0.25
IEC JEDEC JEITA
Z
D
B
02.55 mm
scale
(1)
(1)(1)(1)
D
10.1
9.9
REFERENCES
eH
10.1
9.9
12.9
0.81.3
12.3
2003 Apr 1071
v M
H
v M
D
A
B
E
12.9
12.3
LL
p
0.95
0.55
0.150.10.15
EUROPEAN
PROJECTION
Z
D
1.2
0.8
Zywvθ
E
1.2
0.8
o
10
o
0
ISSUE DATE
97-08-01
03-02-25
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
18 SOLDERING
18.1Introduction to soldering surface mount
packages
Thistextgivesa very brief insight to a complex technology.
A more in-depth account of soldering ICs can be found in
our
“Data Handbook IC26; Integrated Circuit Packages”
(document order number 9398 652 90011).
There is no soldering method that is ideal for all surface
mount IC packages. Wave soldering can still be used for
certainsurfacemount ICs, but it is not suitable forfinepitch
SMDs. In these situations reflow soldering is
recommended.
18.2Reflow soldering
Reflow soldering requires solder paste (a suspension of
fine solder particles, flux and binding agent) to be applied
totheprinted-circuitboardbyscreen printing, stencilling or
pressure-syringe dispensing before package placement.
Several methods exist for reflowing; for example,
convection or convection/infrared heating in a conveyor
type oven. Throughput times (preheating, soldering and
cooling) vary between 100 and 200 seconds depending
on heating method.
Typical reflow peak temperatures range from
215 to 250 °C. The top-surface temperature of the
packages should preferably be kept:
• below 220 °C for all the BGA packages and packages
with a thickness ≥ 2.5mm and packages with a
thickness <2.5 mm and a volume ≥350 mm3 so called
thick/large packages
• below 235 °C for packages with a thickness <2.5 mm
and a volume <350 mm3 so called small/thin packages.
18.3Wave soldering
Conventional single wave soldering is not recommended
forsurfacemount devices (SMDs) or printed-circuit boards
with a high component density, as solder bridging and
non-wetting can present major problems.
If wave soldering is used the following conditions must be
observed for optimal results:
• Use a double-wave soldering method comprising a
turbulent wave with high upward pressure followed by a
smooth laminar wave.
• For packages with leads on two sides and a pitch (e):
– larger than or equal to 1.27 mm, the footprint
longitudinal axis is preferred to be parallel to the
transport direction of the printed-circuit board;
– smaller than 1.27 mm, the footprint longitudinal axis
must be parallel to the transport direction of the
printed-circuit board.
The footprint must incorporate solder thieves at the
downstream end.
• Forpackageswith leads on four sides, the footprint must
be placed at a 45° angle to the transport direction of the
printed-circuit board. The footprint must incorporate
solder thieves downstream and at the side corners.
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.
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.
18.4Manual soldering
Fix the component by first soldering two
diagonally-opposite end leads. Use a low voltage (24 V or
less) soldering iron 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.
To overcome these problems the double-wave soldering
method was specifically developed.
2003 Apr 1072
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
18.5Suitability of surface mount IC packages for wave and reflow soldering methods
1. For more detailed information on the BGA packages refer to the
“(LF)BGAApplicationNote
”(AN01026);orderacopy
from your Philips Semiconductors sales office.
2. All surface mount (SMD) packages are moisture sensitive. Depending upon the moisture content, the maximum
temperature (with respect to time) and body size of the package, there is a risk that internal or external package
cracks may occur due to vaporization of the moisture in them (the so called popcorn effect). For details, refer to the
Drypack information in the
3. These packages are not suitable for wave soldering. On versions with the heatsink on the bottom side, the solder
cannot penetrate between the printed-circuit board and the heatsink. On versions with the heatsink on the top side,
the solder might be deposited on the heatsink surface.
4. If wave soldering is considered, then the package must be placed at a 45° angle to the solder wave direction.
The package footprint must incorporate solder thieves downstream and at the side corners.
5. Wave soldering is suitable for LQFP, TQFP and QFP packages with a pitch (e) larger than 0.8 mm; it is definitely not
suitable for packages with a pitch (e) equal to or smaller than 0.65 mm.
6. Wave soldering is suitable for SSOP, TSSOP, VSO and VSSOP packages with a pitch (e) equal to or larger than
0.65 mm; it is definitely not suitable for packages with a pitch (e) equal to or smaller than 0.5 mm.
2003 Apr 1073
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
19 DATA SHEET STATUS
LEVEL
DATA SHEET
STATUS
(1)
PRODUCT
STATUS
(2)(3)
DEFINITION
IObjective dataDevelopmentThis data sheet contains data from the objective specification for product
development. Philips Semiconductors reserves the right to change the
specification in any manner without notice.
IIPreliminary data QualificationThis data sheet contains data from the preliminary specification.
Supplementary data will be published at a later date. Philips
Semiconductors reserves the right to change the specification without
notice, in order to improve the design and supply the best possible
product.
IIIProduct dataProductionThis data sheet contains data from the product specification. Philips
Semiconductors reserves the right to make changes at any time in order
to improve the design, manufacturing and supply. Relevant changes will
be communicated via a Customer Product/Process Change Notification
(CPCN).
Notes
1. Please consult the most recently issued data sheet before initiating or completing a design.
2. The product status of the device(s) described in this data sheet may have changed since this data sheet was
published. The latest information is available on the Internet at URL http://www.semiconductors.philips.com.
3. For data sheets describing multiple type numbers, the highest-level product status determines the data sheet status.
20 DEFINITIONS
21 DISCLAIMERS
Short-form specification The data in a short-form
specification is extracted from a full data sheet with the
same type number and title. For detailed information see
the relevant data sheet or data handbook.
Limiting values definition Limiting values given are in
accordance with the Absolute Maximum Rating System
(IEC 60134). 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
attheseoratany 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 Applications that are
described herein for any of these products are for
illustrative purposes only. Philips Semiconductors make
norepresentation or warranty that such applicationswillbe
suitable for the specified use without further testing or
modification.
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 inpersonal injury. Philips
Semiconductorscustomers using or selling these products
for use in such applications do so at their own risk and
agree to fully indemnify Philips Semiconductors for any
damages resulting from such application.
Right to make changes Philips Semiconductors
reserves the right to make changes in the products including circuits, standard cells, and/or software described or contained herein in order to improve design
and/or performance. When the product is in full production
(status ‘Production’), relevant changes will be
communicated via a Customer Product/Process Change
Notification (CPCN). Philips Semiconductors assumes no
responsibility or liability for the use of any of these
products, conveys no licence or title under any patent,
copyright, or mask work right to these products, and
makes no representations or warranties that these
products are free from patent, copyright, or mask work
right infringement, unless otherwise specified.
2003 Apr 1074
Philips SemiconductorsPreliminary specification
Stereo audio codec with SPDIF interfaceUDA1355H
22 PURCHASE OF PHILIPS I2C COMPONENTS
Purchase of Philips I
components in the I2C system provided the system conforms to the I2C specification defined by
Philips. This specification can be ordered using the code 9398 393 40011.
2
C components conveys a license under the Philips’ I2C patent to use the
2003 Apr 1075
Philips Semiconductors – a w orldwide compan y
Contact information
For additional information please visit http://www.semiconductors.philips.com.Fax: +31 40 27 24825
For sales offices addresses send e-mail to: sales.addresses@www.semiconductors.philips.com.
The information presented in this document does not form part of any quotation or contract, is believed to be accurate and reliable and may be changed
without notice. No liability will be accepted by the publisher for any consequence of its use. Publication thereof does not convey nor imply any license
under patent- or other industrial or intellectual property rights.
Printed in The Netherlands753503/01/pp76 Date of release: 2003 Apr 10Document order number: 9397 750 09925
SCA75
Loading...
+ hidden pages
You need points to download manuals.
1 point = 1 manual.
You can buy points or you can get point for every manual you upload.