2000 Jul 27 10
Philips Semiconductors Product specification
96 kHz IEC 958 audio DAC UDA1351H
8.2 Clock regeneration and lock detection
The UDA1351H contains an on-board PLL for
regenerating a system clock from the IEC 958 input
bitstream or the incoming digital data stream via the data
input interface. In addition to the system clock for the
on-board digital sound processing the PLL also generates
a 256f
s
clock output for use in the application. In the
absence of an input signal the clock will generate
a minimum frequency to warrant system functionality.
Remark: in case of no input signal, the PLL generates
a minimum frequency and the output spectrum shifts
accordingly. Since the analog output does not have
a analog mute, this means noise which is out of band noise
under normal operation conditions, can move into the
audio band.
When the on-board clock has locked to the incoming
frequency the lock indicator bit will be set and can be read
via the L3 interface. Internally the PLL lock indication is
combined with the PCM status bit of the input data stream.
When both the IEC 958 decoder and the on-board clock
have locked to the incoming signal and the input data
stream is PCM data, then pin LOCK will be asserted.
However, when the IC is locked but the PCM status bit
reports non-PCM data then pin LOCK is returned to LOW
level.
The lock indication output can be used, for example, for
muting purposes. The lock signal can be used to drive an
external analog mute circuit to prevent out of band noise to
become audible in case the PLL runs at its minimum
frequency (e.g. when there is no SPDIF input signal).
An example is given in Fig.3 where V
DD
is the positive
power supply and V
SS
is the negative power supply.
8.3 Mute
The UDA1351H is equipped with a cosine roll-off mute in
the DSP data path of the DAC part. Muting the DAC, by
pin MUTE (in static mode) or via bit MT (in L3 mode) will
result in a soft mute as presented in Fig.4. The cosine
roll-off soft mute takes 32 × 32 samples = 24 ms at a
sampling frequency of 44.1 kHz.
When operating in the L3 control mode the device will
mute on start-up. In L3 mode it is necessary to explicitly
switch off the mute for audio output by means of the MT bit
in the L3 register.
In the L3 mode pin MUTE does not have any function (the
same holds for several other pins) and can either be left
open-circuit (since it has an internal pull-down resistor) or
be connected to ground.
handbook, halfpage
MBL213
V
DD
V
SS
LOCK
21
18
22
VOUTL
VOUTR
DAC
RIGHT
UDA1351H
DAC
LEFT
Fig.3 Example of external analog mute circuit.
handbook, halfpage
01 3
1
0
0.8
MGS755
2
0.6
0.4
0.2
t (ms)
mute
factor
Fig.4 Mute as a function of raised cosine roll-off.
2000 Jul 27 11
Philips Semiconductors Product specification
96 kHz IEC 958 audio DAC UDA1351H
8.4 Auto mute
By default the outputs of the digital data output interface
and the DAC will be muted until the IC is locked,
regardless the level on pin MUTE (in static mode) or the
state of bit MT of the sound feature register (in L3 mode).
In this way only valid data will be passed to the outputs.
This mute is done in the SPDIF interface and is a hard
mute, not a cosine roll-off mute.
If needed, this muting can be bypassed by setting
bit AutoMT to logic 0 via the L3 interface. As a result the IC
will no longer mute during out-of-lock situations.
8.5 Data path
The UDA1351H data path consists of the slicer and the
IEC 958 decoder, the digital data output and input
interfaces, the audio feature processor, digital interpolator
and noise shaper and the digital-to-analog converters.
8.5.1 IEC 958 INPUT
The UDA1351H IEC 958 decoder can select 1 out of 2
IEC 958 input channels. An on-chip amplifier with
hysteresis amplifies the IEC 958 input signal to CMOS
level (see Fig.5).
All 24 bits of data for left and right are extracted from the
input bitstream as well as several of the IEC 958 key
channel-status bits.
The extracted key parameters are:
Pre-emphasis
Audio sample frequency
Two-channel PCM indicator
Clock accuracy.
Both the lock indicator and the key channel status bits are
accessible via the L3 interface.
The UDA1351H supports the following sample
frequencies and data bit rates:
f
s
= 32.0 kHz, resulting in a data rate of 2.048 Mbits/s
f
s
= 44.1 kHz, resulting in a data rate of 2.8224 Mbits/s
f
s
= 48.0 kHz, resulting in a data rate of 3.072 Mbits/s
f
s
= 64.0 kHz, resulting in a data rate of 4.096 Mbits/s
f
s
= 88.2 kHz, resulting in a data rate of 5.6448 Mbits/s
f
s
= 96.0 kHz, resulting in a data rate of 6.144 Mbits/s.
The UDA1351H supports timing level I, II and III as
specified by the IEC 958 standard.
8.5.2 SPDIF SELECTION PROCEDURE
In order to prevent noise at the FSDAC output when
switching between the SPDIF inputs, the following
procedures are recommended. This procedure uses an
external analog mute circuit as shown in Fig.3.
Static mode:
Activate the external analog mute circuit
Select the proper SPDIF input signal
Activate pin RESET to reset the PLL settings and the
PLL will synchronize again to the new input signal
De-activate the external analog mute circuit.
L3 mode:
Activate the external analog mute circuit
Select the proper SPDIF input signal via the
L3 interface
Toggle bit RST_PLL of the L3 interface to reset the
PLL and the PLL will synchronize again to the new
input signal
De-activate the external analog mute circuit.
handbook, halfpage
MGL975
15,
16
SPDIF0,
SPDIF1
75
180 pF
10 nF
UDA1351H
Fig.5 IEC 958 input circuit and typical application.
WARNING
At switching between the two SPDIF inputs, the
switching inside the UDA1351H is done instantly. It may
occur that SPDIF words inside the SPDIF decoder of the
UDA1351H get corrupted. When no action is taken,
corrupted data can reach the FSDAC output.
2000 Jul 27 12
Philips Semiconductors Product specification
96 kHz IEC 958 audio DAC UDA1351H
8.5.3 DIGITAL DATA OUTPUT AND INPUT INTERFACE
The digital data interface enables the exchange of digital
data to and from an external signal processing device.
The digital output and input formats are identical by
design. The possible formats are (see Fig.6):
I
2
S-bus with a word length of up to 24 bits
LSB-justified with a word length of 16 bits
LSB-justified with a word length of 20 bits
LSB-justified with a word length of 24 bits.
Important: the edge of the WS signal must fall on the
negative edge of the BCK signal at all times for proper
operation of the input and output interface (see Fig.9).
In the static pin control mode the format is selected by
means of pins L3MODE and L3DATA. In the L3 control
mode the format defaults to the I
2
S-bus settings and is
programmable via the L3 interface.
The IEC 958 decoder provides the pre-emphasis
information from the IEC 958 input bitstream to pins
PREEM0 and PREEM1 and to the L3 interface register.
Controlling the de-emphasis is different for the 2 modes:
Static pin control mode:
For IEC 958 input de-emphasis is automatically
done, but for I
2
S-bus input de-emphasis is not
possible.
L3 control mode:
IEC 958 input: bit SPDSEL must be set to logic 1 and
de-emphasis is done automatically
–I
2
S-bus input: bit SPDSEL must be set to logic 0 and
de-emphasis can be controlled via bits DE0
and DE1.
8.5.4 AUDIO FEATURE PROCESSOR
The audio feature processor automatically provides
de-emphasis for the IEC 958 data stream in the static pin
control mode and default mute at start-up in the L3 control
mode. When used in the L3 control mode it provides the
following additional features:
Volume control using 6 bits
Bass boost control using 4 bits
Treble control using 2 bits
Mode selection of the sound processing bass boost and
treble filters: flat, minimum and maximum
Soft mute control with raised cosine roll-off
De-emphasis selection of the incoming data stream for
f
s
= 32.0, 44.1 and 48.0 kHz.
8.5.5 INTERPOLATOR
The UDA1351H includes an on-board interpolating filter
which converts the incoming data stream from 1f
s
to 128f
s
by cascading a recursive filter and a FIR filter.
Table 2 Interpolator characteristics
8.5.6 NOISE SHAPER
The third-order noise shaper operates at 128f
s
. It shifts
in-band quantization noise to frequencies well above the
audio band. This noise shaping technique enables high
signal-to-noise ratios to be achieved. The noise shaper
output is converted into an analog signal using a filter
stream digital-to-analog converter.
8.5.7 FILTER STREAM DAC (FSDAC)
The FSDAC is a semi-digital reconstruction filter that
converts the 1-bit data stream 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 output operational
amplifier. In this way very high signal-to-noise
performance and low clock jitter sensitivity is achieved.
A post filter is not needed due to the inherent filter function
of the DAC. On-board amplifiers convert the FSDAC
output current to an output voltage signal capable of
driving a line output.
The output voltage of the FSDAC is scaled proportionally
with the power supply voltage.
PARAMETER CONDITIONS VALUE (dB)
Pass-band ripple 0f
s
to 0.45f
s
±0.03
Stop band >0.65f
s
50
Dynamic range 0f
s
to 0.45f
s
115
DC gain −−3.5

UDA1351H/N1,551

Mfr. #:
Manufacturer:
NXP Semiconductors
Description:
IC DAC/AUDIO 24BIT 100K 44PQFP
Lifecycle:
New from this manufacturer.
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