z Multi-bit Delta Sigma modulator
z 105 dB Dynamic Range
z -95 dB THD+N
z Up to 192 kHz Sampling Rates
z Single-ended Analog Architecture
z Volume Control with Soft Ramp
– 0.5 dB Step Size
– Zero Crossing Click-free Transitions
z Popguard
– Minimizes the effects of output transients.
z Filtered Line-level Outputs
z Selectable Serial Audio Interface Formats
– Left Justified up to 24-bit
2
–I
S up to 24-bit
– Right Justified 16, 18, 20 and 24-bit
z Selectable 50/15 µs De-emphasis
TM
Technology
A/D Features
z Multi-bit Delta Sigma Modulator
z 105 dB Dynamic Range
z -95 dB THD+N
z Stereo 2:1 Input Multiplexer
z Programmable Gain Amplifier (PGA)
– +/- 12 dB gain, 0.5 dB Step Size
– Zero Crossing, Click-free Transitions
z Pseudo-differential Stereo Line Inputs
z Stereo Microphone Inputs
– +32 dB Gain Stage
– Low-noise Bias Supply
z Up to 192 kHz Sampling Rates
z Selectable Serial Audio Interface Formats
– Left Justified up to 24-bit
– I²S up to 24-bit
z High-pass Filter or DC Offset Calibration
1.8 V to 5 V
Volume
PC M S erial Interfa ce / Loopb ac k
C ontrol
Volume
C ontrol
IEC 60958-3 Transmitter
High Pass
F ilte r
High Pass
F ilte r
Serial
Audio
Input
Serial
Audio
O utput
I2C C ontrol
Data
Reset
Level TranslatorLevel Translator
Re gister Configuration
Advance Product Information
Cirrus Logic, Inc.
www.cirrus.com
3.3 V to 5 V3.3 V to 5 V
Inte rpolation
Filter
Inte rpolation
Filter
Linear Phase
Anti-Alias Filter
Linear Phase
Anti-Alias Filter
M ultib it
∆Σ Modulator
M u ltibit
∆Σ Modulator
O ve rsa m p ling
O ve rsa m p ling
Switched Capacitor
DA C and F ilter
Switched Capacitor
DA C and F ilter
Internal Voltage
Reference
M u ltibit
ADC
M u ltibit
ADC
PGA
PGA
Control
Mic Bias
MUX
Mute
+32 dB
+32 dB
This document contains information for a new product.
Cirrus Logic reserves the right to modify this product without notice.
– Up to 192 kHz Sampling Rates
–75 Ω Drive Capability
z Serial Audio Data Input Multiplexer
z Internal Digital Loopback
z Supports Master or Slave Operation
z Mute Output Control
z Power Down Mode
– Available for A/D, D/A, CODEC, Mic
Preamplifier
z +3.3 V to +5 V Analog Power Supply
z +3.3 V to +5 V digital Power Supply
z Direct Interface with 1.8 V to 5 V Logic
Levels
z Supports I²C Control Port Interface
General Description
The CS4265 is a highly integrated stereo audio CODEC. The CS4265 performs stereo analog-to-digital
(A/D) and digital-to-analog (D/A) conversion of up to
24-bit serial values at sample rates up to 192 kHz.
A 2:1 stereo input multiplexer is included for selecting
between line level or microphone level inputs. The microphone input path includes a +32 dB gain stage and a
low noise bias voltage supply. The PGA is available for
line or microphone inputs and provides gain or attenuation of ±12 dB in 0.5 dB steps.
The output of the PGA is followed by an advanced 5thorder, multi-bit delta sigma modulator and digital filtering/decimation. Sampled data is transmitted by the
serial audio interface at rates from 4 kHz to 192 kHz in
either slave or master mode.
The D/A converter is based on a 4th-order multi-bit delta
sigma modulator with an ultra-linear low pass filter and
offers a volume control that operates with a 0.5 dB step
size. It incorporates selectable soft ramp and zero
crossing transition functions to eliminate clicks and
pops.
Standard 50/15 µs de-emphasis is available for a
44.1 kHz sample rate for compatibility with digital audio
programs mastered using the 50/15 µs pre-emphasis
technique.
Integrated level translators allow easy interfacing between the CS4265 and other devices operating over a
wide range of logic levels.
ORDERING INFORMATION
CS4265-CNZ, Lead Free-10° to 70° C 32-pin QFN
CDB4265Evaluation Board
are specified in the DC Electrical Characteristics table.
Analog Power (Input) - Positive power for the internal analog section.
17
Analog Ground (Input) - Ground reference for the internal analog section.
18
19,
Analog Audio Output (Output) - The full scale output level is specified in the DAC Analog Characteris-
tics specification table.
20
Mute Control (Output) - This pin is active during power-up initialization, reset, muting, when master
21
clock left/right clock frequency ratio is incorrect, or power-down.
Serial Audio Interface Power (Input) - Determines the required signal level for the serial audio inter-
22
face. Refer to the Recommended Operating Conditions for appropriate voltages.
Transmitter Serial Audio Data Input (Input) - Input for two’s complement serial audio data.
23
Serial Audio Data Input 2 (Input) - Input for two’s complement serial audio data.
24
Serial Audio Data Input 1 (Input) - Input for two’s complement serial audio data.
25
Serial Audio Data Output (Output) - Output for two’s complement serial audio data.
26
Serial Clock (Input/Output) - Serial clock for the serial audio interface.
27
Left Right Clock (Input/Output) - Determines which channel, Left or Right, is currently active on the
28
serial audio data line.
Master Clock (Input) -Clock source for the delta-sigma modulators.
29
Digital Ground (Input) - Ground reference for the internal digital section.
30
Digital Power (Input) - Positive power for the internal digital section.
31
Transm itter Line Driver Output (Output) - IEC60958-3 driver output.
32
Thermal Pad - Thermal relief pad for optimized heat dissipation.
-
6DS657A2
CS4265
2. CHARACTERISTICS AND SPECIFICATIONS
(All Min/Max characteristics and specifications are guaranteed over the Specified Operating Conditions. Typical
performance characteristics and specifications are derived from measurements taken at nominal supply voltages
and T
SPECIFIED OPERATING CONDITIONS (AGND = DGND = 0 V; All voltages with respect to
ground.)
ABSOLUTE MAXIMUM RATINGS (AGND= DGND = 0 V All voltages with respect to ground.) (Note
1)
= 25°C.)
A
ParametersSymbol Min NomMaxUnits
DC Power Supplies:Analog
Digital
Logic - Serial Port
Logic - Control Port
Ambient Operating Temperature (Power Applied)T
ParameterSymbolMinTypMaxUnits
DC Power Supplies:Analog
Digital
Logic - Serial Port
Logic - Control Port
Input Current(Note 2)I
Analog Input Voltage V
Digital Input VoltageLogic - Serial Port
Logic - Control Port
Ambient Operating Temperature (Power Applied)T
Storage TemperatureT
VA
VD
VLS
VLC
VD
VLS
VLC
V
IND-S
V
IND-C
A
VA
in
INA
stg
3.1
3.1
1.71
1.71
-10-+70°C
-0.3
-0.3
-0.3
-0.3
AGND-0.3-VA+0.3V
-0.3
-0.3
A
-20-+85°C
-65-+150°C
5.0
3.3
3.3
3.3
--±10mA
5.25
5.25
5.25
5.25
-
-
-
-
--VLS+0.3
+6.0
+6.0
+6.0
+6.0
VLC+0.3VV
V
V
V
V
V
V
V
V
Notes: 1. Operation beyond these limits may result in permanent damage to the device.
Normal operation is not guaranteed at these extremes.
2. Any pin except supplies. Transient currents of up to ±100 mA on the analog input pins will not cause
SCR latch-up.
DS657A27
CS4265
DAC ANALOG CHARACTERISTICS (Full-Scale Output Sine Wave, 997 Hz; Test load R
C
= 10 pF (see Figure 1), Fs = 48/96/192 kHz. Measurement Bandwidth 10 Hz to 20 kHz, unless otherwise speci-
L
= 3 kΩ,
L
fied.)
All Speed Modes
Parameter
SymbolMinTypMaxUnit
Dynamic Performance for VA = 5 V
Dynamic Range(Note 3)
18 to 24-Bitunweighted
A-Weighted
16-Bitunweighted
A-Weighted
Total Harmonic Distortion + Noise(Note 3)
18 to 24-Bit0 dB
-20 dB
-60 dB
16-Bit0 dB
-20 dB
-60 dB
THD+N
96
99
87
90
102
105
93
96
-
-
-
-
-
-
-95
-82
-42
-93
-73
-33
-
-
-
-
-89
-76
-36
-87
-67
-27
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
Dynamic Performance for VA = 3.3 V
Dynamic Range(Note 3)
18 to 24-Bitunweighted
A-Weighted
16-Bitunweighted
A-Weighted
Total Harmonic Distortion + Noise(Note 3)
18 to 24-Bit0 dB
-20 dB
-60 dB
16-Bit0 dB
-20 dB
-60 dB
THD+N
93
96
85
88
99
102
90
93
-
-
-
-
-
-
-92
-79
-39
-90
-70
-30
-
-
-
-
-84
-71
-31
-82
-62
-22
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
Interchannel Isolation(1 kHz)-100-dB
DC Accuracy
Interchannel Gain Mismatch-0.10.25dB
Gain Drift-100-ppm/°C
Analog Output
Full Scale Output Voltage0.60*VA0. 65 *VA0.70*VAV
DC Current draw from an AOUT pin(Note 4)I
Note:3. One-half LSB of triangular PDF dither added to data.
4. Guaranteed by design. The DC current draw represents the allowed current draw from the AOUT pin
due to typical leakage through the electrolytic DC blocking capacitors.
5. Guaranteed by design. See Figure 2. R
maximum capacitance required for the internal op-amp’s stability. C
internal output amp; increasing C
L
and CL reflect the recommended minimum resistance and
L
affects the dominant pole of the
L
beyond 100 pF can cause the internal op-amp to become unstable.
8DS657A2
CS4265
DAC COMBINED INTERPOLATION & ON-CHIP ANALOG FILTER RESPONSE
Parameter (Note 6,9)SymbolMinTypMaxUnit
Combined Digital and On-chip Analog Filter Response Single Speed Mode
Passband (Note 6)to -0.05 dB corner
to -3 dB corner
Frequency Response 10 Hz to 20 kHz-.01-+.08dB
StopBand.5465--Fs
StopBand Attenuation(Note 7)50--dB
Group Delaytgd-10/Fs-s
De-emphasis Error (Note 8) Fs = 44.1 kHz--+.05/-.25dB
Combined Digital and On-chip Analog Filter Response Double Speed Mode
Passband (Note 6)to -0.1 dB corner
to -3 dB corner
Frequency Response 10 Hz to 20 kHz-.05-+.2dB
StopBand.5770--Fs
StopBand Attenuation(Note 7)55--dB
Group Delaytgd-5/Fs-s
Combined Digital and On-chip Analog Filter Response Quad Speed Mode
Passband (Note 6)to -0.1 dB corner
to -3 dB corner
Frequency Response 10 Hz to 20 kHz0-+0.00004dB
StopBand0.7--Fs
StopBand Attenuation(Note 7)51--dB
Group Delaytgd-2.5/Fs-s
0
0
0
0
0
0
-
-
-
-
-
-
.4780
.4996
.4650
.4982
0.397
0.476
Fs
Fs
Fs
Fs
Fs
Fs
Notes: 6. Filter response is guaranteed by design.
7. For Single Speed Mode, the Measurement Bandwidth is 0.5465 Fs to 3 Fs.
For Double Speed Mode, the Measurement Bandwidth is 0.577 Fs to 1.4 Fs.
For Quad Speed Mode, the Measurement Bandwidth is 0.7 Fs to 1 Fs.
8. De-emphasis is available only in Single Speed Mode.
9. Response is clock dependent and will scale with Fs. Note that the amplitude vs. frequency plots of this
data (Figures 18 to 27) have been normalized to Fs and can be de-normalized by multiplying the X-axis
scale by Fs.
DS657A29
CS4265
125
3.3 µF
AOUTx
AGND
Figure 1. DAC Output Test Load
100
L
V
out
R
L
C
L
75
50
25
Capacitive Load -- C (pF)
2.5
51015
3
Safe Operating
Region
Resistive Load -- R (kΩ)
L
20
Figure 2. Maximum DAC Loading
10DS657A2
CS4265
ADC ANALOG CHARACTERISTICS Test conditions (unless otherwise specified): Input test sig-
nal is a 1 kHz sine wave; measurement bandwidth is 10 Hz to 20 kHz. Fs = 48/96/192 kHz.
Note: 14. Filter response is guaranteed by design.
15. Response shown is for Fs = 48 kHz.
16. Response is clock dependent and will scale with Fs. Note that the response plots (Figures 30 to 41) are
normalized to Fs and can be de-normalized by multiplying the X-axis scale by Fs.
14DS657A2
CS4265
DC ELECTRICAL CHARACTERISTICS (AGND = DGND = 0 V, all voltages with respect to
ground. MCLK=12.288 MHz; Fs=48 kHz, Master Mode)
ParameterSymbolMinTypMaxUnit
Power Supply Current VA = 5 V
(Normal Operation) VA = 3.3 V
VD, VLS, VLC = 5 V
VD, VLS, VLC = 3.3 V
Power Supply Current. VA = 5 V
(Power-Down Mode) (Note 17). VLS, VLC, VD=5 V
I
A
I
A
I
D
I
D
I
A
I
D
-
-
-
-
-
-
41
37
39
23
0.50
0.54
50
45
47
28
-
-
Power Consumption
(Normal Operation). VA, VD, VLS, VLC = 5 V
VA, VD, VLS, VLC = 3.3 V
(Power-Down Mode). VA, VD, VLS, VLC = 5 V
-
-
-
-
-
-
400
198
4.2
485
241
-
Power Supply Rejection Ratio (1 kHz)(Note 18)PSRR-60-dB
VQ Characteristics
Quiescent VoltageVQ-0.5 x VA-VDC
DC Current from VQ(Note 19)I
VQ Output ImpedanceZ
Q
Q
-- 1µA
-23 -kΩ
FILT+ Nominal VoltageFILT+-VA-VDC
Microphone Bias VoltageMICBIAS-0.8 x VA-VDC
Current from MICBIASI
MB
-- 2mA
mA
mA
mA
mA
mA
mA
mW
mW
mW
Notes: 17. Power Down Mode is defines as RESET
input.
18. Valid with the recommended capacitor values on FILT+ and VQ as shown in the Typical Connection
Diagram.
19. Guaranteed by design. The DC current draw represents the allowed current draw due to typical leakage
through the electrolytic de-coupling capacitors.
= Low with all clock and data lines held static and no analog
DS657A215
CS4265
DIGITAL INTERFACE CHARACTERISTICS
Parameters (Note 20)Symbol Min TypMaxUnits
High-Level Input VoltageSerial Port
Control Port
Low-Level Input VoltageSerial Port
Control Port
High-Level Output Voltage at I
=2 mASerial Port
o
Control Port
MUTEC
TXOUT
Low-Level Output Voltage at I
=2 mASerial Port
o
Control Port
MUTEC
TXOUT
Input Leakage CurrentI
V
IH
V
IH
V
IL
V
IL
V
OH
V
OH
V
OH
V
OH
V
OL
V
OL
V
OL
V
OL
in
Input Capacitance(Note 21)--1pF
Maximum MUTEC Drive Current-3-mA
Notes: 20. Serial Port signals include: MCLK, SCLK, LRCK, SDIN1, SDIN2, TXSDIN, SDOUT.
Control Port signals include: SCL, SDA, RESET
.
21. Guaranteed by design.
0.7xVLS
0.7xVLC
-
-
VLS-1.0
VLC-1.0
VA-1.0
VD-1.0
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
0.2xVLS
0.2xVLC
-
-
-
-
0.4
0.4
0.4
0.4
--±10µA
V
V
V
V
V
V
V
V
V
V
V
V
16DS657A2
CS4265
SWITCHING CHARACTERISTICS - SERIAL AUDIO PORT (Logic ‘0’ = DGND = 0 V;
Logic ‘1’ = VL, C
Sample RateSingle Speed Mode
MCLK Specifications
MCLK Frequency f
MCLK Input Pulse Width High/Lowt
MCLK Output Duty Cycle455055%
Master Mode
LRCK Duty Cycle-50-%
SCLK Duty Cycle-50-%
SCLK falling to LRCK edget
SCLK falling to SDOUT validt
SDIN valid to SCLK rising setup timet
SCLK rising to SDIN hold timet
Slave Mode
LRCK Duty Cycle405060%
SCLK Period
= 20 pF) (Note 22)
L
ParameterSymbolMinTypMaxUnit
Double Speed Mode
Quad Speed Mode
Single Speed Mode
Double Speed Mode
Quad Speed Mode
Fs
Fs
Fs
mclk1.024-51.200MHz
clkhl8--ns
slr
sdo
sdis
sdih
t
sclkw
t
sclkw
t
sclkw
4
50
100
-
-
-
50
100
200
-10-10ns
0-32ns
16--ns
20--ns
9
10
-------------------- -
128()Fs
9
10
----------------- -
64()Fs
9
10
----------------- 64()Fs
-
-
-
-
-
-
kHz
kHz
kHz
ns
ns
ns
SCLK Pulse Width Hight
SCLK Pulse Width Lowt
SCLK falling to LRCK edget
SCLK falling to SDOUT validt
SDIN valid to SCLK rising setup timet
SCLK rising to SDIN hold timet
sclkh
sclkl
slr
sdo
sdis
sdih
30--ns
48--ns
-10-10ns
0-32ns
16--ns
20--ns
Notes: 22. See figures 3 and 4 on page 18.
DS657A217
LRCK
Output
SCLK
Output
SDOUT
t
t
sdo
CS4265
slr
SDIN
LRCK
Input
SCLK
Input
SDOUT
t
sdis
Figure 3. Master Mode Serial Audio Port Timing
t
sclkh
t
sclkw
t
t
slr
sdo
t
sdih
t
sclkl
t
sdis
SDIN
Figure 4. Slave Mode Serial Audio Port Timing
18DS657A2
t
sdih
CS4265
Left Channel
LRCK
SCLK
SDATA+3 +2 +1
MSB
-1 -2 -3 -4 -5
Channel A - Left
+5 +4
Figure 5. Format 0, Left Justified up to 24-Bit Data
Channel A - Left
LRCK
SCLK
SDATA+3 +2 +1
MSB
-1 -2 -3 -4 -5
Left Channel
+5 +4
Figure 6. Format 1, I²S up to 24-Bit Data
LRCK
SCLK
Channel A - Left
LeftChannel
LSB
LSB
MSB
-1 -2 -3 -4
MSB
-1 -2 -3 -4
Channel B - Right
Right Channel
+3 +2 +1
+5 +4
Right Channel
Channel B - Right
+3 +2 +1
+5 +4
Channel B - Right
RightChannel
LSB
LSB
SDATA
+6
+4 +3 +2
LS B+5
MSB -1 -2 -3 -4 -5
32 clocks
-6
+5
+1 LSB
MSB -1 -2 -3 -4
-5
-6
+6
Figure 7. Format 2, Right Justified 16-Bit Data.
Format 3, Right Justified 24-Bit Data.
+4 +3 +2
+1 LSB
DS657A219
SWITCHING CHARACTERISTICS - I²C CONTROL PORT
(Inputs: Logic 0 = DGND, Logic 1 = VLC, C
Parameter SymbolMinMaxUnit
SCL Clock Frequencyf
RESET
Rising Edge to Startt
Bus Free Time Between Transmissionst
Start Condition Hold Time (prior to first clock pulse)t
Clock Low timet
Clock High Timet
Setup Time for Repeated Start Conditiont
SDA Hold Time from SCL Falling(Note 23)t
SDA Setup time to SCL Risingt
Rise Time of SCL and SDA(Note 24)t
Fall Time SCL and SDA(Note 24)t
Setup Time for Stop Conditiont
Acknowledge Delay from SCL Fallingt
=30pF)
L
scl
irs
buf
hdst
low
high
sust
hdd
sud
rc
fc
susp
ack
-100kHz
500-ns
4.7-µs
4.0-µs
4.7-µs
4.0-µs
4.7-µs
0-µs
250-ns
-1µs
-300ns
4.7-µs
3001000ns
CS4265
Notes: 23. Data must be held for sufficient time to bridge the transition time, t
24. Guaranteed by design.
RST
t
irs
StopStart
SDA
SCL
t
buf
t
t
hdst
lo w
t
t
high
hdd
t
sud
t
ack
Figure 8. Control Port Timing - I²C Format
, of SCL.
fc
Repeated
Start
t
t
sust
t
hdst
Stop
rd
t
t
rc
t
fd
fc
t
susp
20DS657A2
3.TYPICAL CONNECTION DIAGRAM
CS4265
+3.3V to +5V
+1.8V
to +5V
Digital Audio
Processor
Digital Audio
Output
Micro-
Controller
2 kΩ
+1.8V
Note 1
to +5V
Note 1: Resistors are required for I²C control
port operation
Note 3: The value of RL is dictated by the
microphone carteridge.
Note 4: Sets the LSB of the 7-bit chip address.
See the I²C Control Port Description and
Timing section.
0.1 µF
47 kΩ
Note 4
2 kΩ
10 µF
0.1 µF
0.1 µF
VLS
SDOUT
SDIN1
SDIN2
TXSDIN
MCLK
SCLK
LRCK
TXOUT
RST
SCL
SDA
VLC
VA
CS4265
DGND
0.1 µF
0.1 µF
10 µF
+3.3V to +5V
VAVD
10 kΩ
10 kΩ
=
C
*
1800 pF
Mute
Drive
470Ω
*
C
Optional
Analog
Muting
*
C
470Ω
470
+
R
ext
()
4704
×
RFs
π
ext
10 µF
100 Ω
100 kΩ
Left Analog Input 1
MUTEC
AOUTA
AOUTB
AIN1A
3.3 µF
3.3 µF
Note 2 :
For best response to Fs/2 :
This circuitry is intended for applications where the CS4265
connects directly to an unbalanced output of the design. For
internal routing applications please see the DAC Analog Output
Characteristics section for loading limitations.
10 µF
SGNDS i gnal Ground
*
1800 pF
100 kΩ
AIN1B
MICIN1
MICIN2
MICBIAS
10 µF
10 µF
Note 3
10 µF
100 Ω
R
LRL
Right Analog Input 1
Microphone Input 1
Microphone Input 2
FILT+
10 µF
0.1 µF
AGND
AGND
AFILTA
0.1 µF
**
2.2nF
47 µF
2.2nF
AFILTB
VQ
* Capacitors must be C0G or equivalent
10 µF
See Note 2
R
ext
R
ext
Figure 9. Typical Connection Diagram
DS657A221
CS4265
4. APPLICATIONS
4.1Recommended Power-Up Sequence
1) Hold RESET low until the power supply, MCLK, and LRCK are stable. In this state, the Control Port is reset to
its default settings.
2) Bring RESET
will be accessible.
3) The desired register settings can be loaded while the PDN bit remains set.
4) Clear the PDN bit to initiate the power-up sequence.
4.2System Clocking
The CS4265 will operate at sampling frequencies from 4 kHz to 200 k Hz. This range is divided into three speed
modes as shown in Table 1 below.
high. The device will remain in a low power state with the PDN bit set by default. The control port
ModeSampling Frequency
Single Speed4-50 kHz
Double Speed50-100 kHz
Quad Speed100-200 kHz
Table 1. Speed Modes
4.2.1Master Clock
MCLK/LRCK must maintain an integer ratio as shown in Table 2. The LRCK frequency is equal to Fs, the frequency
at which audio samples for each channel are clocked into or out of the device. The FM bits (see page 35) and the
MCLK Freq bits (see page 36) configure the device to generate the proper clocks in Master Mode and receive the
proper clocks in Slave Mode. Table 2 illustrates several standard audio sample rates and the required MCLK and
LRCK frequencies.
22DS657A2
CS4265
LRCK
(kHz)
32
44.1
48
64
88.2
96
128
176.4
192
Mode
64x96x128x192x256x384x512x768x1024x
----8.192012.288016.384024.576032.7680
----11.289616.934422.579233.868045.1584
----12.288018.432024.576036.864049.1520
--8.192012.288016.384024.576032.7680--
--11.289616.934422.579233.868045.1584--
--12.288018.432024.576036.864049.1520--
8.192012.288016.384024.576032.7680----
11.289616.934422.579233.868045.1584----
12.288018.432024.576036.864049.1520----
QSM
MCLK (MHz)
DSMSSM
Table 2. Common Clock Frequencies
4.2.2Master Mode
As a clock master, LRCK and SCLK will operate as outputs. LRCK and SCLK are internally derived from MCLK with
LRCK equal to Fs and SCLK equal to 64 x Fs as shown in Figure 10.
MC LK F req Bits
MC LK
÷1
÷1.5
÷2
÷3
÷4
000
001
010
011
100
÷256
÷128
÷64
÷4
÷2
÷1
FM Bits
00
01
10
00
01
10
LRC K
SCLK
Figure 10. Master Mode Clocking
4.2.3Slave Mode
In Slave mode, SCLK and LRCK operate as inputs. The Left/Right clock signal must be equal to the sample rate,
Fs, and must be synchronously derived from the supplied master clock, MCLK.
The serial bit clock, SCLK, must be synchronously derived from the master clock, MCLK, and be equal to 128x, 64x,
48x or 32x Fs depending on the desired speed mode. Refer to Table 3 for required clock ratios.
When using operational amplifiers in the input circuitry driving the CS4265, a small DC offset may be driven into the
A/D converter. The CS4265 includes a high pass filter after the decimator to remove any DC offset which could result
in recording a DC level, possibly yielding clicks when switching between devices in a multichannel system.
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CS4265
The high pass filter continuously subtracts a measure of the DC offset from the output of the decimation filter. If the
HPFFreeze bit (see page 35) is set during normal o peration, the current value of the DC offset for the each channe l
is frozen and this DC offset will continue to be subtracted from the conversion result. This feature makes it possible
to perform a system DC offset calibration by:
1) Running the CS4265 with the high pass filter enabled until the filter settles. See the ADC Digital Filter Characteristics section for filter settling time.
2) Disabling the high pass filter and freezing the stored DC offset.
A system calibration performed in this way will eliminate offsets anywhere in the signal path between the calibration
point and the CS4265.
4.4Analog Input Multiplexer, PGA, and Mic Gain
The CS4265 contains a stereo 2-to-1 analog input multiplexer followed by a programmable gain amplifier (PGA).
The input multiplexer is able to select either a line-level input source, or a mic-level input source and route it to the
PGA. The mic-level input passes through a +32 dB gain stage prior to the input multiplexer, allowing it to be used
for microphone level signals without the need for any external gain. The PGA sta ge provides 12 dB of gain or attenuation in 0.5 dB steps. Figure 11 shows the architecture of the input multiplexer, PGA, and mic gain stages.
AINA
MICIN1
AINB
MICIN2
+32 dB
+32 dB
MUX
Analog Input
Selection Bits
MUX
PGA
Channel A
PGA Gain Bits
Channel B
PGA Gain Bits
PGA
Out to ADC
Channel A
Out to ADC
Channel B
Figure 11. Analog Input Architecture
The “Analog Input Selection (Bit 0)” section on page 38 outlines the bit settings necessary to control the input multiplexer and mic gain. “Channel A PGA Control - Address 07h” on page 36 and “Channel B PGA Control - Address
08h” on page 37 outlines the register settings necessary to control the PGA. By default, the line level input is selected by the input multiplexer, and the PGA is set to 0 dB.
4.5Input Connections
The analog modulator samples the input at 6.144 MHz (MCLK=12.288 MHz). The digital filter will reject signals within the stopband of the filter. However, there is no rejection for input signals which are (n
passband frequency, where n=0,1,2,... Refer to the Typical Connection Diagram for the recommended analog input
circuit that will attenuate noise energy at 6.144 MHz. The use of capacitors which have a large voltage coefficient
(such as general purpose ceramics) must be avoided since these can degrade signal linearity. Any unused analog
input pairs should be left unconnected.
× 6.144 MHz) the digital
4.5.1Pseudo-Differential Input
The CS4265 implements a pseudo-differential input stage. The SGND input is intended to be used as a pseudodifferential reference signal. This feature allows for common mode noise rejection with single-ended signals.
Figure 12 shows a basic diagram outlining the internal implementation of the pseudo-differential input stage. The
Typical Connection Diagram shows the recommended pseudo-d ifferential input topology. If pseudo-differential inp ut
24DS657A2
CS4265
functionality is not required, simply connect the SGND pin to AGND through the parallel combination of a 10 µF and
a 0.1 µF capacitor.
CS4265
AINA
VA
SGND
AINB
Note: If pseudo-differential input functionality is not required, the
10 µF
0.1 µF
connections shown with dashed line should be added.
Figure 12. Pseudo-Differential Input Stage
-
+
+
-
In to P GA
In to P GA
4.6Output Connections
The CS4265 DAC’s implement a switched-capacitor filter followed by a continuous time low pass filter. Its response,
combined with that of the digital interpolator, is shown in the “DAC Filter Plots” section beginning on page 45. The
recommended external analog circuitry is shown in the Typical Connection Diagram.
The CS4265 DAC is a linear phase design and does not include phase or amplitude compensation for an external
filter. Therefore, the DAC system phase and amplitude response will be dependent on the external analog circuitry.
4.7Output Transient Control
The CS4265 uses Popguard™ technology to minimize the effects of output transients during power-up and powerdown. This technique eliminates the audio transients commonly prod uced by single-ended single- supply converters
when it is implemented with external DC-blocking capacitors connected in series with the audio outputs. To make
best use of this feature, it is necessary to understand its operation.
4.7.1Power-up
When the device is initially powered-up, the DAC outputs AOUTA and AOUTB are clamped to VQ which is initially
low. After the PDN bit is released (set to ‘0’), the outputs begin to ramp with VQ towards the nominal quiescent vo ltage. This ramp takes approximately 200 ms to complete. The gradual voltage ramping allows time for the external
DC-blocking capacitors to charge to VQ, effectively blocking the quiescent DC voltage. Audio output will begin after
approximately 2000 sample periods.
4.7.2Power-down
To prevent audio transients at power-down the DC-blocking capacitors must fully discharge before turning off the
power. In order to do this, either the PDN should be set or the device should be reset about 250 ms before removing
power. During this time, the voltage on VQ and the DAC outputs will gradually discharge to GND. If power is removed
before this 250 ms time period has passed a transient will occur when the VA supply drops below that of VQ. There
is no minimum time for a power cycle, power may be re-applied at any time.
4.7.3Serial Interface Clock Changes
When changing the clock ratio or sample rate it is recommended that zero data (or near zero data) be present on
the selected SDIN pin for at least 10 LRCK samples before the change is made. During the clocking change the
DAC outputs will always be in a zero data state. If non-zero serial audio input is present at the time of switching, a
slight click or pop may be heard as the DAC output automatically goes to it’s zero data state.
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CS4265
4.8DAC Serial Data Input Multiplexer
The CS4265 contains a 2-to-1 serial data input multiplexer. This allows two separate data sources to be input into
the DAC without the use of any external multiplexing components. The “DAC SDIN Source (Bit 7)” section on
page 36 describes the control port settings necessary to control the multiplexer.
4.9De-Emphasis Filter
The CS4265 includes on-chip digital de -emphasis optimized for a sample rate of 44 .1 kHz. The filter response is
shown in Figure 13. The frequency response of the de-emphasis curve will scale proportionally with changes in sample rate, Fs. Please see section 6.3.3 for de-emphasis control.
The de-emphasis feature is included to accommodate audio recordings that utilize 50/15 µs pre-emphasis equalization as a means of noise reduction.
De-emphasis is only available in Single Speed Mode.
Gain
dB
T1=50 µs
0dB
T2 = 15 µs
-10dB
F1F2
3.183 kHz10.61 kHz
Figure 13. De-Emphasis Curve
Frequency
4.10Internal Digital Loopback
The CS4265 supports an internal digital loopback mode in which the output of the ADC is routed to the input of the
DAC. This mode may be activated by setting the LOOP bit in the Signal Selection register (06h - See page 36).
When this bit is set, the status of the DAC_DIF[1:0] bits in register 03h will be disregarded by the CS4265. Any
changes made to the DAC_DIF[1:0] bits while the LOOP bit is set will have no impact on operation until the LOOP
bit is cleared, at which time the Digital Interface Format of the DAC will operate according to the format selected by
the DAC_DIF[1:0] bits. While the LOOP bit is set, data will be present on the SDOUT pin in the format selected by
the ADC_DIF bit in register 04h.
4.11Mute Control
The MUTEC pin becomes active during power-up initialization, reset, muting, if the MCLK to LRCK ratio is incorrect,
and during power-down. The M UTEC
add off-chip mute capability.
Use of the Mute Control function is not mandatory but recommended for designs requiri ng the a bsolute m inimum in
extraneous clicks and pops. Also, use of the Mute Control function can enable the system designer to achieve idle
pin is intended to be used as control for an external mute circuit in order to
26DS657A2
CS4265
channel noise/signal-to-noise ratios which are only limited by the external mute circuit. The MUTEC pin is an activelow CMOS driver. See Figure 14 below for a suggested active-low mute circuit.
+V
EE
AOUT
CS4265
MUT E C
LPF
AC
Couple
-V
EE
+V
A
MMUN2111LT1
47 kΩ
-V
EE
560 Ω
10 kΩ
2 kΩ
Audio
Out
Figure 14. Suggested Active-Low Mute Circuit
4.12AES3 Transmitter
The CS4265 includes an IEC60958-3 digital audio transmitter. A comprehensive buffering scheme provides write
access to the channel status data. This buffering scheme is described in “Channel Status Buffer Management” on
page 50.
The IEC60958-3 transmitter encodes and transmits audio and digital data according to the IEC60958-3 (S/PDIF)
interface standard. Audio and control data are multiplexed together and bi-phase mark encoded. The resulting bit
stream is driven to an output connector either directly or through a transformer. Th e tr a nsm itt er is clocked from the
clock input pin MCLK.
The channel status (C) bits in the transmitted data stream are taken from stor age areas within the CS4265. The user
can manually access the internal storage of the CS4265 to configure the transmitted channel status data. The section “Channel Status Buffer Management” on page 50 describes the method of manually accessing the storage areas. The CS4265 transmits all ‘0’s in the user (U) data fields.
4.12.1 TxOut Driver
The line driver is a low skew, low impedance, single-ended output capable of driving cables directly. The driver is
set to ground during reset (RESET
= LOW), when no transmit clock is provided, and optionally under the control of
a register bit. The CS4265 also allows immediate muting of the IEC60958-3 transmitter a udio data through a control
register bit.
External components are used to terminate and isolate the external cable from the CS4265 . These components are
detailed in “External IEC60958-3 Transmitter Components” on page 49.
4.12.2 Mono Mode Operation
An IEC60958-3 stream may be used in more than on e way to tr ansm it 192 kHz sample rate data. One method is to
double the frame rate of the current format. This results is a stereo signal with a sample rate of 192 kHz. An alternate
method is implemented using the two sub-frames in a 96 kHz frame rate IEC60958-3 signal to carry consecutive
samples of a mono signal, resulting in a 192 kHz sample rate stream. This allows older equipment, whose
IEC60958-3 transmitters and receivers are not rated for 192kHz frame rate operation, to handle 192 kHz sample
DS657A227
CS4265
rate information. In this “mono mode”, two cables are needed for stereo data transfer. The CS4265 offers mono
mode operation. The CS4265 is set placed into and out of mono mode with the MMT control bit.
In mono mode, the input port will run at the audio sample rate (Fs), while the IEC60958-3 transmitter frame rate will
be at Fs/2. Consecutive left or right channel serial audio data samples may be selected for transmission on the A
and B sub-frames, and the channel status block transmitted is also selectable.
Using mono mode is only necessary if the incoming audio sample rate is already at 192 kHz and contains both left
and right audio data words. The “mono mode” IEC60958-3 output stream may also be achieved by keeping the
CS4265 in normal stereo mode, and placing consecutive audio samples in the left and ri ght positions in an incoming
96 kHz word rate data stream.
4.13I²C Control Port Description and Timing
The control port is used to access the registers, allowing the CS4265 to be configured for the desired operational
modes and formats. The operation of the control port may be completely asynchronous with respect to the audio
sample rates. However, to avoid potential interference problems, the control port pins should remain static if no operation is required.
SDA is a bidirectional data line. Data is clocked into and out of the part by the clock, SCL. A 47 kΩ pull-up or pulldown on the SDOUT pin will set AD0, the least significant bit of the chip address. A pull-up to VLS will set AD0 to ‘1’
and a pull-down to DGND will set AD0 to ‘0’. The state of SDOUT is sensed and AD0 is set upon the release of
RESET.
The signal timings for a read and write cycle are shown in Figure 15 and Figure 16. A Start condition is defined as
a falling transition of SDA while the clock is high. A Stop condition is a rising transition while the clock is high. All
other transitions of SDA occur while the clock is low. The first byte sent to the CS4265 after a Start condition consists
of a 7 bit chip address field and a R/W
are fixed at 100111. To communicate with a CS4265, the chip address field, which is the first byte sent to the
CS4265, should match 100111 followed by the setting of AD0. The eighth bit of the address is the R/W
operation is a write, the next byte is the Memory Address Pointer (MAP) which selects the register to be read or
written. If the operation is a read, the contents of the register pointed to by the MAP will be output. Setting the auto
increment bit in MAP allows successive reads or writes of consecutive registers. Each byte is separated by an acknowledge bit. The ACK bit is output from the CS4265 after each input byte is read, and is input to the CS4265 fro m
the microcontroller after each transmitted byte.
bit (high for a read, low for a write). The uppe r 6 bits o f the 7-b it addre ss field
bit. If the
26
DATA +1
DATA +n
ACKACKACK
STOP
SCL
SDA
0 1 2 38 91 216 17 18 1910 1113 14 1527 28
CHIP ADDRESS (WRITE)MAP BYTEDATA
1 0 0 1 1 1 AD0 0
START
4 5 6 724 25
INCR 6 5 4 3 2 1 07 6 1 07 6 1 07 6 1 0
ACK
Figure 15. Control Port Timing, I²C Write
28DS657A2
CS4265
168 912 13 14 154 5 6 7 0 120 21 22 23 24
STOP
ACK
START
CHIP ADDRESS (READ)
1 0 0 1 1 1 AD0 1
26 27 28
DATA
7 07 07 0
ACK
DATA +1
ACK
DATA + n
NO
ACK
STOP
SCL
SDA
2 310 1117 18 1925
CHIP ADDRESS (WRITE)
1 0 0 1 1 1 AD0 0
START
MAP BYTE
INCR 6 5 4 3 2 1 0
ACK
Figure 16. Control Port Timing, I²C Read
Since the read operation can n ot set the MAP, an aborted write operation is used as a preamble. As shown in
Figure 16, the write operation is aborted after the acknowledge for the MAP byte by sending a stop condition. The
following pseudocode illustrates an aborted write operation followed by a read operation.
Setting the auto increment bit in the MAP allows successive reads or writes of consecutive registers. Each
byte is separated by an acknowledge bit.
4.14Status Reporting
The CS4265 has comprehensive status reporting capabilities. Many conditions can be reported in the status register, as listed in the status register descriptions. See “ Status - Address 0Dh” on page 39. Each source may be masked
off through mask register bits. In addition, each source may be set to rising edge, falling edge, or level sensitive.
Combined with the option of level sensitive or edge sensitive modes within the microcontroller, many different configurations are possible, depending on the needs of the equipment designer.
4.15 Reset
When RESET is low, the CS4265 enters a low power mode and all internal states are reset, including the control
port and registers, and the outputs are muted. When RESET
desired settings should be loaded into the control registers. Writing a 0 to the PDN bit in the Power Control register
will then cause the part to leave the low power state and begin operation.
The delta-sigma modulators settle in a matter of microseconds after the analog section is powered, either through
the application of power or by setting the RESET
pin high. However, the voltage reference will take much longer to
reach a final value due to the presence of external capacitance on the FILT+ pin. During this voltage reference ramp
delay, both SDOUT and DAC outputs will be automatically muted.
is high, the control port becomes operational and the
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CS4265
It is recommended that RESET be activated if the analog or digital supplies drop below th e recommended operating
condition to prevent power glitch related issues.
4.16Synchronization of Multiple Devices
In systems where multiple ADCs are required, care must be taken to achieve simultaneous sampling. To ensure
synchronous sampling, the master clocks and left/right clocks must be the same for all of the CS4265’s in the system. If only one master clock source is needed, on e so lu tion is to pl ace one CS4 2 65 in Ma st er Mode , an d slave all
of the other CS4265’s to the one master. If multiple master clock sources are needed, a possible solution would b e
to supply all clocks from the same external source and time the CS4265 reset with the inactive edge o f master clock.
This will ensure that all converters begin sampling on the same clock edge.
4.17Grounding and Power Supply Decoupling
As with any high resolution converter, the CS4265 requires careful attention to power supply and grounding arrangements if its potential performance is to be realized. Figure 9 shows the recommended power arrange ments, with VA
connected to a clean supply. VD, which powers the digital filter, may be run from the syst em logic supply (VLS or
VLC) or may be powered from the analog supply (VA) via a resistor. In this case, no additional devices should be
powered from VD. Power supply decoupling capacitors should be as near to the CS4265 as possible, with the low
value ceramic capacitor being the nearest. All signals, especially clocks, should be kept away from the FILT+ and
VQ pins in order to avoid unwanted coupling into the modulators. The FILT+ and VQ decoupling capacitors, particularly the 0.1 µF, must be positioned to minimize the electrical path from FILT+ and AGND. The CS4265 evaluation
board demonstrates the optimum layout and power supply arrangements. To minimize digital noise, connect the
CS4265 digital outputs only to CMOS inputs.
4.18Package Considerations
The CS4265 is available in the compact QFN package. The under side of the QFN package reveals a large metal
pad that serves as a thermal relief to provide for maximum heat dissipation. This pad must mate with an equally
dimensioned copper pad on the PCB and must be electrically connected to ground. A series of vias should be used
to connect this copper pad to one or more larger ground planes on other PCB layers. In split ground systems, it is
recommended that this thermal pad be connected to AGND for best performance. The CS4265 evaluation board
demonstrates the optimum thermal pad and via configuration.
30DS657A2
CS4265
5. REGISTER QUICK REFERENCE
This table shows the register names and their associated default values.
AddrFunction76543210
01h Chip ID
02h Power ControlFreezeReserved Reserved ReservedPDN_MICPDN_ADCPDN_DACPDN
03h DAC Control 1
04h ADC Control
05h MCLK
Frequency
06h Signal Selec-
tion
07h PGA Ch B Gain
Control
08h PGA Ch A Gain
Control
09h Analog Input
Control
0Ah DAC Ch A Vol-
ume Control
0Bh DAC Ch B Vol-
ume Control
0Ch DAC Contro l 2DACSoftDACZero InvertDAC ReservedReservedReservedReservedReserved
This register is Read-Only. Bits 7 through 4 are the part number ID which is 1101b (0Dh) and the remaining bits (3 through 0) are for the chip revision.
This function allows modifications to be made to certain control port bits without the changes taking
effect until the Freeze bit is disabled. To make multiple changes to these bits take effect simultaneously, set the Freeze bit, make all changes, then clear the Freeze bit. The bits affected by the
Freeze function are listed in Table 4 below.
NameRegisterBit(s)
MuteDAC03h2
MuteADC04h2
Gain[5:0]07h5:0
Gain[5:0]08h5:0
Vol[7:0]0Ah7:0
Vol[7:0]0Bh7:0
TxMute0Eh4
6.2.2Power Down MIC (Bit 3)
Function:
The microphone preamplifier block will enter a low-power state whenever this bit is set.
6.2.3Power Down ADC (Bit 2)
Function:
The ADC pair will remain in a reset state whenever this bit is set.
6.2.4Power Down DAC (Bit 1)
Function:
The DAC pair will remain in a reset state whenever this bit is set.
Table 4. Freeze-able Bits
6.2.5Power Down Device (Bit 0)
Function:
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CS4265
The device will enter a low-power state whenever this bit is set. The power-down bit is set by default
and must be cleared before normal operation can oc cur. The contents of the control re gisters are retained when the device is in power-down.
The required relationship between LRCK, SCLK and SDIN for the DAC is defined by the DAC Digital
Interface Format and the options are detailed in Table 5 and Figures 5-7.
Table 5. DAC Digital Interface Formats
DAC_DIF1 DAC_DIF0DescriptionFormatFigure
00Left Justified, up to 24-bit data (default)05
01
The DAC outputs will mute and the MUTEC pin will become active when this bit is set. Though this
bit is active high, it should be noted that the MUTEC pin is active low. Th e common mode voltage on
the outputs will be retained when this bit is set. The muting function is effected, similar to attenuation
changes, by the DACSoft and DACZero bits in the DAC Control 2 register.
6.3.3De-Emphasis Control (Bit 1)
Function:
The standard 50/15 µs digital de-emphasis filter response, Figure 17, may be implemented for a sample rate of 44.1 kHz when the DeEmph bit is configured as shown in Table 6 below. NOTE: De-emphasis is available only in Single-Speed Mode.
Table 6. De-Emphasis Control
DeEmphDescription
0Disabled (default)
144.1 kHz de-em ph as is
6.4ADC Control - Address 04h
76543210
FM1FM0ReservedADC_DIFReservedMuteADCHPFFreezeM/S
34DS657A2
Gain
dB
0dB
-10dB
CS4265
T1=50 µs
T2 = 15 µs
F1F2
3.183 kHz10.61 kHz
Figure 17. De-Emphasis Curve
6.4.1Functional Mode (Bits 7:6)
Function:
Selects the required range of sample rates.
Table 7. Functional Mode Selection
FM1FM0Mode
00Single-Speed Mode: 4 to 50 kHz sample rates
01Double-Speed Mode: 50 to 100 kHz sample rates
10Quad-Speed Mode: 100 to 200 kHz sample rates
11Reserved
6.4.2ADC Digital Interface Format (Bit 4)
Function:
The required relationship between LRCK, SCLK and SDOUT is defined by the ADC Digital Interface
Format bit. The options are detailed in Table 8 and may be seen in Figure 5 and 6.
Table 8. ADC Digital Interface Formats
Frequency
ADC_DIFDescriptionFormatFigure
0Left Justified, up to 24-bit data (default)05
1
2
I
S, up to 24-bit data
16
6.4.3Mute ADC (Bit 2)
Function:
When this bit is set, the serial audio output of the both ADC channels will be muted.
6.4.4ADC High Pass Filter Freeze (Bit 1)
Function:
When this bit is set, the internal high-pass filter will be disabled.The current DC offset value will be
frozen and continue to be subtracted from the conversion result. See “High Pass Filter and DC Offset
Calibration” on page 23.
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CS4265
6.4.5Master / Slave Mode (Bit 0)
Function:
This bit selects either master or slave operation for the serial audio port. Setting this bit will select master mode, while clearing this bit will select slave mode.
6.5MCLK Frequency - Address 05h
76543210
Reserved
MCLK
Freq2
6.5.1Master Clock Dividers (Bits 6:4)
Function:
Sets the frequency of the supplied MCLK signal. See Table 9 below for the appropriate settings.
This bit is used to select the serial audio data source for the DAC as shown in Table 10 below.
Table 10. DAC SDIN Source Selection
SDINSel SettingDAC Data Source
0SDIN1
1SDIN2
6.6.2Digital Loopback (Bit 1)
Function:
When this bit is set, an internal digital loopback from the ADC to the DAC will be enabled. Please refer
to “Internal Digital Loopback” on page 26.
6.7Channel A PGA Control - Address 07h
76543210
ReservedReservedGain5Gain4Gain3Gain2Gain1Gain0
36DS657A2
CS4265
6.7.1Channel A PGA Gain (Bits 5:0)
Function:
See “Channel B PGA Gain (Bits 5:0)” on page 37.
6.8Channel B PGA Control - Address 08h
76543210
ReservedReservedGain5Gain4Gain3Gain2Gain1Gain0
6.8.1Channel B PGA Gain (Bits 5:0)
Function:
Sets the gain or attenuation for the ADC input PGA stage. The gain may be adjusted from -12 dB to
+12 dB in 0.5 dB steps. The gain bits are in two’s complement with the Gain0 bit set for a 0.5 dB step.
Register settings outside of the ±12 dB range are reserved and must not be used. See Table 11 for
example settings.
6.9.1PGA Soft Ramp or Zero Cross Enable (Bits 4:3)
Function:
Soft Ramp Enable
Soft Ramp allows level changes, both muting and attenuation, to be implemented by incrementally
ramping, in 1/8 dB steps, from the current level to the new level at a rate of 1 dB per 8 left/right clock
periods. See Table 12 on page 38.
Zero Cross Enable
Zero Cross Enable dictates that signal level changes, either by attenuation changes or muting, will
occur on a signal zero crossing to minimize audible artifacts. The requested level change will occur
after a time-out period between 512 and 1 024 sample periods ( 10.7 ms to 21.3 ms at 48 kHz sample
rate) if the signal does not encounter a ze ro cro s sing. The ze ro cross functi on is independ ently monitored and implemented for each channel. See Table 12 on page 38.
Soft Ramp and Zero Cross Enable
Soft Ramp and Zero Cross Enable dictate that signal level ch anges, either by attenuation changes or
muting, will occur in 1/8 dB steps and be implemented on a signal zero crossing. The 1/8 dB level
change will occur after a time-out period between 51 2 and 1024 sample per iods (10.7 ms to 21.3 ms
at 48 kHz sample rate) if the signal does not encounter a zero crossing. The zero cross function is
independently monitored and implemented for each channel. See Table 12 on page 38.
DS657A237
Table 12. PGA Soft Cross or Zero Cross Mode Selection
PGASoftPGAZeroCrossMode
00Change s to affect immediate ly
01Zero Cross enabled
10Soft Ramp enabled
11Soft Ramp and Zero Cross enabled (default)
6.9.2Analog Input Selection (Bit 0)
Function:
These bits are used to select the input source for the PGA and ADC. Please see Table 13 below.
Table 13. Analog Input Selection
SelectPGA/ADC Input
0Microphone Level Input
1Line Level Input
6.10DAC Channel A Volume Control - Address 0Ah
CS4265
See 6.11 DAC Channel B Volume Control - Address 0Bh
6.11DAC Channel B Volume Control - Address 0Bh
76543210
Vol7Vol6Vol5Vol4Vol3Vol2Vol1V ol0
6.11.1 Volume Control (Bits 7:0)
Function:
The digital volume control allows the user to attenuate the signal in 0.5 dB increments from 0 to
-127 dB. The Vol0 bit activates a 0.5 dB attenuation when set, and no attenuation when cleared. The
Vol[7:1] bits activate attenuation equal to their decimal equivalent (in dB). Example volume settings
are decoded as shown in Table Table 14. The volume changes are implemented as dictated by the
DACSoft and DACZeroCross bits in the DAC Control 2 register (see section 6.12.1).
Table 14. Digital Vo lume Control Example Settings
Binary CodeVolume Setting
000000000 dB
00000001-0.5 dB
00101000-20 dB
00101001-20.5 dB
6.12.1 DAC Soft Ramp or Zero Cross Enable (Bits 7:6)
Function:
Soft Ramp Enable
Soft Ramp allows level changes, both muting and attenuation, to be implemented by incrementally
ramping, in 1/8 dB steps, from the current level to the new level at a rate of 1 dB per 8 left/right clock
periods. See Table 15 on page 39.
Zero Cross Enable
Zero Cross Enable dictates that signal level changes, either by attenuation changes or muting, will
occur on a signal zero crossing to minimize audible artifacts. The requested level change will occur
after a time-out period between 512 and 1 024 sample periods ( 10.7 ms to 21.3 ms at 48 kHz sample
rate) if the signal does not encounter a ze ro cro s sing. The ze ro cross functi on is independ ently monitored and implemented for each channel. See Table 15 on page 39.
Soft Ramp and Zero Cross Enable
Soft Ramp and Zero Cross Enable dictate that signal level ch anges, either by attenuation changes or
muting, will occur in 1/8 dB steps and be implemented on a signal zero crossing. The 1/8 dB level
change will occur after a time-out period between 51 2 and 1024 sample per iods (10.7 ms to 21.3 ms
at 48 kHz sample rate) if the signal does not encounter a zero crossing. The zero cross function is
independently monitored and implemented for each channel. See Table 15 on page 39.
CS4265
Table 15. DAC Soft Cross or Zero Cross Mode Selection
DACSoftDACZeroCrossMode
00Changes to affect immediately
01Zero Cross enabled
10Soft Ramp enabled
11Soft Ramp and Zero Cross enabled (default)
6.12.2 Invert DAC Output (Bit 5)
Function:
When this bit is set, the output of the DAC will be inverted.
For all bits in this register, a ‘1’ means the associated condition has occur red at least once since the
register was last read. A ‘0’ means the associated condition has NOT occurred since the la st reading
of the register. Status bits that are masked off in the associated mask register will always be ‘0’ in this
register. This register defaults to 00h.
6.13.1 E to F C-buffer Transfer
Function:
Indicates the completion of an E to F C-buffer transfer. See “Channe l Status Buffer Management” on
page 50 for more information.
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6.13.2 Clock Error (Bit 3)
Function:
Indicates the occurrence of a clock error condition.
6.13.3 ADC Overflow (Bit 1)
Function:
Indicates the occurrence of an ADC overflow condition.
6.13.4 ADC Underflow (Bit 0)
Function:
Indicates the occurrence of an ADC underflow condition.
The bits of this register serve as a mask for the Status source s found in the register “Status - Address
0Dh” on page 39. If a mask bit is set to 1, the error is unmasked, meaning that its occurrence will affect
the status register. If a mask bit is set to 0, the error is masked, meaning that its occurrence will not
affect the status register. The bit positions align with the corresponding bits in the Status register.
The two Status Mode registers form a 2-bit code for each Status register function. There are three
ways to update the Status register in accordance with the status condition. In the Rising edge active
mode, the status bit becomes active on the arrival of the condition. In the Falling edge active mode,
the status bit becomes active on the removal of the condition. In Level active mode, the status bit is
active during the condition.
00 - Rising edge active
01 - Falling edge active
10 - Level active
11 - Reserved
6.17.1 E to F C-Data Buffer Transfer Inhibit (Bit 6)
Function:
When cleared, C-data E to F buffer transfers are allowed. When set, C-data E to F buffer transfers
are inhibited. See “IEC60958-3 Channel Status (C) Bit Management” on page 50.
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6.17.2 C-Data Access Mode (Bit 5)
Function:
When cleared, the C-data buffer will operate in One-byte control port access mode. When set, the Cdata buffer will operate in Two-byte control port access mode. See “IEC60958-3 Channel Status (C)
Bit Management” on page 50.
6.18Transmitter Control 2 - Address 12h
76543210
Tx_DIF1Tx_DIF0TxOffTxMuteVMMTMMTCSMMTLR
6.18.1 Transmitter Digital Interface Format (Bits 7:6)
Function:
The required relationship between LRCK, SCLK and SDIN for the transmitter is defined by the T ransmitter Digital Interface Format and the options are detailed in Table 16 and Figures 5-7.
Table 16. Transmitter Digital Interface Formats
Tx_DIF1Tx_DIF0DescriptionFormatFigure
00Left Justified, up to 24-bit data (default)05
01
When this bit is cleared, the transmitter output pin driver will be in the normal operational mode. When
set, the transmitter output pin driver will drive to a constant 0 V.
6.18.3 Transmitter Mute Control (Bit 4)
Function:
When this bit is cleared, the transmitter data will be in the normal operational mode. When set, the
transmitter will output all zero data.
6.18.4 Transmitted Validity Bit Control (Bit 3)
Function:
This bit sets the transmitted Validity bit level.
When this bit is cleared, valid linear PCM audio data is indicated. When this bit is set, invalid or non-
linear PCM audio data is indicated.
6.18.5 Transmitter Mono/Stereo Operation Control (Bit 2)
Function:
When this bit is cleared, the transmitter will operate in stereo mode. When set, the transmitter will operate in mono mode with one input channel’ s data output in both A a nd B subframes (see “IEC609583 Channel Status (C) Bit Management” on page 50) and the CS data defined by the MMTCS bit (see
section 6.18.6).
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6.18.6 Mono Mode CS Data Source (Bit 1)
Function:
When this bit is cleared, the transmitter will transmit the channel A CS data in the A subframe and the
channel B CS data in the B subframe.
When this bit is set, the transmitter will transmit the CS data defined for the channel selected by the
MMTLR bit in both the A and B subframes.
6.18.7 Mono Mode Channel Selection (Bit 0)
Function:
When this bit is cleared, channel A input data will be transmitted in both channel A and B subframes
in mono mode. When this bit is set, channel B input data will be transmitted in both channel A and B
subframes in mono mode.
CS4265
42DS657A2
7.PARAMETER DEFINITIONS
Dynamic Range
The ratio of the rms value of the signal to the rms sum of all other spectral components over the specified
bandwidth. Dynamic Range is a signal-to-noise ratio measurement over the specified bandwidth made
with a -60 dBFS signal. 60 dB is added to resulting measurement to refer the measurement to full- scale.
This technique ensures that the distortion components are below the noise level and do not affect the
measurement. This measurement technique has been accepted by the Audio Engineering Society,
AES17-1991, and the Electronic Industries Association of Japan, EIAJ CP-307. Expressed in decibels.
Total Harmonic Distortion + Noise
The ratio of the rms value of the signal to the rms sum of all other spectral components over the specified
bandwidth (typically 10 Hz to 20 kHz), including distortion components. Expressed in decibels. Measured
at -1 and -20 dBFS as suggested in AES17-1991 Annex A.
Frequency Response
A measure of the amplitude response variation from 10 Hz to 20 kHz relative to the amplitude response
at 1 kHz. Units in decibels.
Interchannel Isolation
A measure of crosstalk between the left and right channels. Measured for each channel at the conver ter's
output with no signal to the input under test and a full-scale signal applied to the other channel. Units in
decibels.
Interchannel Gain Mismatch
The gain difference between left and right channels. Units in decibels.
CS4265
Gain Error
The deviation from the nominal full-scale analog output for a full-scale digital input.
Gain Drift
The change in gain value with temperature. Units in ppm/°C.
Offset Error
The deviation of the mid-scale transition (111...111 to 000...000) from the ideal. Units in mV.
This section details the external components required to interface the IEC60958-3 transmitter to cables and fiberoptic components.
C.1IEC60958-3 Transmitter External Components
The IEC60958-3 specifications call for an unbalanced drive circuit with an output impedance of 75 Ω ± 20% and a
output drive level of 0.5 volts peak-to-peak ±20% when measured across a 75 Ω load using no cable. The circuit
shown in Figure 42 provides the proper output impedance and drive level using standard 1% resistor s. If VD is driven
from +3.3 V, use re sist or va lu es o f 24 3 Ω in place of the 374 Ω resistor and a 107 Ω resistor in place of the 90 .9 Ω
resistor. The standard connector for a consumer application is an RCA phono socke t.
The TXOUT pin may be used to drive TTL or CMOS gates as shown in Figure 43. This circuit may be used for optical
connectors for digital audio as they typically implement TTL or CMOS compatible inputs. This circuit is also useful
when driving multiple digital audio outputs as RS422 line drivers typically implement TTL compatible inputs.
374-R
90.9
TXP
Ω
RCA
Phono
TXOUT
CS4265
Figure 42. Consumer Output Circuit (VD = 5 V)
TXOUT
CS4265
Figure 43. TTL/CMOS Output Circuit
TTL or
CMOS Gate
C.2Isolating Transformer Requirements
Please refer to the application note AN134: AES and SPDIF Recommended Transformers for resources on transformer selection.
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APPENDIX D: CHANNEL STATUS BUFFER MANAGEMENT
The CS4265 has a comprehensive channel status (C) data buffering scheme which allows the user to manage the
C data through the control port.
D.1IEC60958-3 Channel Status (C) Bit Management
The CS4265 contains sufficient RAM to store a full block of C data for both A and B channels (192x2 = 384 bits).
The user may read from or write to these RAM buffers through the control port.
The CS4265 manages the flow of channel status data at the b lock level, meaning that entire blocks of channel status
information are buffered at the input, synchronized to the output time base, and then transmitted. The buffering
scheme involves a cascade of 2 block-sized buffers, named E and F, as show n in Figure 44. The MSB of each byte
represents the first bit in the serial C data stream. For example, the MSB of byte 0 (which is at control port address
13h) is the consumer/professional bit for channel status block A.
The E buffer is accessible from the control port, allowing read and writing of the C data. Th e F buffer is u sed as th e
source of C data for the IEC60958-3 transmitter. The F buffer accepts block transfers from the E buffer.
AB
8-bits8-bits
E
24
words
F
Transmit
Data
Buffer
To
AES3
Transmitter
Control Port
Figure 44. Channel Status Data Buffer Structure
D.1.1Accessing the E buffer
The user can monitor the data being tran sferre d by re ading the E b uffer, which is mapped into th e regi ster space of
the CS4265, through the control port. The user can modify the data to be transmitted by writing to the E buffer.
The user can configure the status register such that EFTC bit is set whenever an E to F transfer completes. With
this configuration in place, periodic polling of the status register allows determination of the time periods acceptable
for E buffer interaction.
Also provided is an “E to F” inhibit bit. The “E to F” buffer transfer is disabled whenever the user sets this bit. This
may be used whenever “long” control port interactions are occurring.
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A flowchart for reading and writing to the E buffer is shown in Figure 45. For writing, the sequence starts after an E
to F transfer, which is based on the output timebase .
Begin
Configure the EFTC status bit as
Rising Edge active.
Read the Status Register
(Reg 0Dh)
Is the EFTC bit set?
Yes
Optionally set E to F inhibit
Write E data
If set, clear E to F inhibit
Figure 45. Flowchart for Writing the E Buffer
No
D.1.2Serial Copy Management System (SCMS)
The CS4265 allows read/modify/write access to all the channel status bits. For consumer mode SCMS compliance,
the host microcontroller needs to manipulate the Category Code, Copy bit and L bit appropriately.
D.1.3Channel Status Data E Buffer Access
The E buffer is organized as 24 x 16-bit words. For each word the most significant byte is the A channel data, and
the least significant byte is the B channel data (see Figure 44).
There are two methods of accessing this memory, known as one byte mode and two byte mode. The desired mode
is selected through a control register bit.
D.1.3.1One Byte mode
In many applications, the channel status blocks for the A and B channels will be identical. In this situation, if the user
reads a byte from one of the channel's blocks, the corresponding byte for the other channel will be the same. Similarly, if the user wrote a byte to one channel's block, it would be ne cessary to write the same byte to the other block.
One byte mode takes advantage of the often identical nature of A and B channel status data.
When reading data in one byte mode, a single byte is returned, which can be from channel A or B data, depending
on a register control bit. If a write is being done, the CS4265 expects a single byte to be input to its control port. This
byte will be written to both the A and B locations in the addressed word.
One byte mode saves the user substantial control port access time, as it effectively accesses 2 bytes worth of information in 1 byte's worth of access t ime. If th e cont rol port's a uto increm ent ad dressing is used in combin ation with
this mode, multi-byte accesses such as full-block reads or writes can be done especially efficiently.
D.1.3.2Two Byte mode
There are those applications in which the A and B channel status blocks will not be the same, and the user is interested in accessing both blocks. In these situations, two byte mode should be used to access the E buffer.
In this mode, a read will cause the CS4265 to output two bytes from its control port. The first byte out will represent
the A channel status data, and the 2nd byte will represent the B channel status data. Writing is similar, in that two
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bytes must now be input to the CS4265's control port. The A channel status data is first, B channel status data second.
– Updated descriptions of pins 3, 4, 5, and 6 on page 5.
– Removed specifications for SPI control port.
– Added specification for AD0 selection in the I²C Control Port
Description and Timing section on page 28.
– Updated the typical connection diagram on page 21 to reflect the
pin changes and AD0 selection method.
– Added thermal pad to pin descriptions on page 6.
– Added Package Considerations section on page 30.
– Updated the Mic level input impedance specification on page 13.
Table 17. Revision History
CS4265
Contacting Cirrus Logic Support
For all product questions and inquiries contact a Cirrus Logic Sales Representative.
To find the one nearest to you go to www.cirrus.com
IMPORTANT NOTICE
"Advance" product informat ion de scrib es produc ts that are in developmen t and subje ct to d evelopmen t changes . Cirr us Logic , Inc. and its subsidiaries ("Cirrus")
believe that the inf ormation contained in this document is accurate and relia ble. However, the information is subject to change without notice and is provided "AS
IS" without warranty of any kind (express or implied). Customers are ad vised to obtain the latest version of relevant information to verify, before placing orders, that
information being relied on is current an d complete. All pr oducts are sold subject to the te rms and conditio ns of sa le supp lied at the time of order acknowledgment,
including those pertaining to warranty, patent infr ingemen t, and limitation of lia bility. No responsib ility is assumed by Ci rrus for the use of th is informat ion, includi ng
use of this information as the basis for manufacture or sale of any items, or for infringement of patents or other rights of third parties. This document is the property
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or other intellect ual prop erty rig hts. Cir rus owns t he copyri ghts asso ciated wit h the in formati on conta ined her ein and giv es consent for copies to be made of the
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Cirrus Logic, Cirrus, the Cirrus Logic logo designs, and Popguard are trademarks of Cirrus Logic, Inc. All other brand and product names in this document may be
trademarks or service marks of their respective owners.
I2C is a registered trademark of Philips Semicondu ctor. Purchase of I2C C omponents of Cirrus L ogic, Inc., or one of its sublice nsed Associated Companies conveys
a license under the Philips I2C Patent Rights to use those components in a standard I2C system.
DS657A253
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