–Stereo Analog Passthrough: 10 mW @ 1.8 V
–Stereo Playback: 14 mW @ 1.8 V
–Stereo Rec. and Playback: 23 mW @ 1.8 V
Variable Power Supplies
–1.8 V to 2.5 V Digital and Analog
–1.6 V to 5 V Class D Amplifier
–1.8 V to 2.5 V Headphone Amplifier
–1.8 V to 3.3 V Interface Logic
Power-down Management
–ADC, DAC, CODEC, MIC Pre-Amplifier, PGA,
Headphone Amplifier, Speaker Amplifier
Analog and Digital Routing/Mixes:
–Line/Headphone Out = Analog In (ADC
Bypassed)
–Line/Headphone/Speaker
Out = ADC + Digital In
–Digital Out = ADC + Digital In
–Internal Digital Loopback
–Mono Mixes
Flexible Clocking Options
–Master or Slave Operation
–High-impedance Digital Output Option (for easy
MUXing between CODEC and other data
sources)
–Quarter-speed Mode - (i.e. allows 8 kHz Fs
while maintaining a flat noise floor up to 16 kHz)
–4 kHz to 96 kHz Sample Rates
I²C™ Control Port Operation
Headphone/Speaker Detection Input
Pop and Click Suppression
Applications
Digital Voice Recorders, Digital Cameras, and
Camcorders
PDA’s
Personal Media Players
Portable Game Consoles
General Description
The CS42L52 is a highly integrated, low-power stereo CODEC with headphone and Class D speaker amplifiers. The
CS42L52 offers many features suitable for low-power, portable system applications.
The ADCinput path allows independent channel control of a
number of features. Input summing amplifiers mix and select
line-level and/or microphone-level inputs for each channel.
The microphone input path includes a selectable programmable-gain pre-amplifier stage and a low-noise MIC bias voltage
supply. A PGA is available for line or microphone inputs and
provides analog gain with soft-ramp and zero-cross transitions. The ADC also features a digital volume control with soft
ramp transitions. A programmable ALC and Noise Gate monitor the input signals and adjust the volume levels
appropriately. To conserve power, the ADC may be bypassed
while still allowing full analog volume control.
The DAC output path includes a digital signal processing engine with various fixed-function controls. Tone Control
provides bass and treble adjustment of four selectable corner
frequencies. The Digital Mixer provides independent volume
control for both the ADC output and PCM input signal paths,
as well as a master volume control. Digital Volume controls
may be configured to change on soft-ramp transitions while
the analog controls can be configured to occur on every zero
crossing. The DAC also includes de-emphasis, limiting functions and a BEEP generator, delivering tones selectable
across a range of two full octaves.
The stereo hea dphone amplifier is powered from a separate
positive supply and the integrated charge pump provides a
negative supply. This allows a ground-centered, analog output
with a wide signal swing and eliminates external DC-blocking
capacitors.
The Class D stereo s peaker amplifier does not require an
external filter and provides the high-efficiency amplification required by power-sensitive portable applications. The speaker
amplifier may be powered directly from a battery while the internal DC supply monitoring and compensation provides a
constant gain level as the battery’s voltage decays.
In addition to its many features, the CS42L52 operates from a
low-voltage analog and digital core making it ideal for portable
systems that require extremely low power consumption in a
minimal amount of space.
The CS42L52 is available in a 40-pin QFN package in Commercial (-40 to +85 °C) grade. The CS42L52 Customer
Demonstration board is also available for device evaluation
and implementation suggestions. Refer to “Ordering Informa-
tion” on page 81 for complete ordering information.
Figure 32. DAC Transition Band ............................................................................................................... 78
Figure 33. DAC Transition Band (Detail) ................................................................................................... 78
3/1/13
CS42L52
DS680F27
1. PIN DESCRIPTIONS
12
11
13
14
15
16
17
18
19
20
29
30
28
27
26
25
24
23
22
21
39
40
38
37
36
35
34
33
32
31
2
1
3
4
5
6
7
8
9
10
GND/Thermal Pad
SDOUT
MCLK
SCLK
SDIN
SDA
LRCK
FLYN
+VHP
HP/LINE_OUTB
HP/LINE_OUTA
VQ
MICBIAS
AIN4A/MIC1+/MIC2A
AIN2A
AD0
SPKR_OUTA+
VP
VP
VD
SPKR_OUTB-
-VHPFILT
AIN4B/MIC2+/MIC2B
AIN1B
AIN2B
AFILTB
AIN3B/MIC2-/MIC1B
AFILTA
AIN1A
AIN3A/MIC1-/MIC1A
SPKR_OUTB+
SCL
DGND
SPKR_OUTA-
FLYP
VA
AGND
FILT+
RESET
VL
SPKR/HP
Top-Down (Through-Package) View
40-Pin QFN Package
3/1/13
CS42L52
Pin Name#Pin Description
SDA1SerialControl Data (Input/Output) - SDA is a data I/O in I²C Mode.
SCL2Serial Control Port Clock (Input) - Serial clock for the serial control port.
AD03
SPKR_OUTA+
SPKR_OUTASPKR_OUTB+
SPKR_OUTB-
VP
-VHPFILT10
FLYN11
FLYP12
+VHP13
HP/LINE_OUTB, A14,15 Headphone/Line Audio Output (Output) - Stereo headphone or line level analog outputs.
VA16Analog Power (Input) - Positive power for the internal analog section.
Power for PWM Drivers (Input) - Power supply for the PWM output driver stages.
8
Inverting Charge Pump Filter Connection (Output) - Power supply from the inverting charge
pump that provides the negative rail for the headphone/line amplifi e rs.
Charge Pump Cap Negative Node (Output) - Negative node for the inverting charge pump’s fly-
ing capacitor.
Charge Pump Cap Positive Node (Output) - Positive node for the inverting charge pump’s flying
capacitor.
Positive Analog Power for Headphone (Input) - Positive voltage rail and power for the internal
headphone amplifiers and inverting charge pump.
3/1/13
CS42L52
AGND17Analog Ground (Input) - Ground reference for the internal analog section.
FILT+18
VQ19Quiescent Voltage (Output) - Filter connection for the internal quiescent voltage.
MICBIAS20
AIN4A,B
AIN3A,B
MIC1+,MIC2+,MIC2A,B
MIC1A,B
AIN2A,B
AIN1A,B
AFILTA,B27,28 Anti-alias Filter Connection (Output) - Anti-alias filter connection for the ADC inputs.
SPKR/HP31
RESET
VL33
VD34Digital Power (Input) - Positive power for the internal digital section.
DGND35Digital Ground (Input) - Ground reference for the internal digital section.
SDOUT36Serial Audio Data Output (Output) - Output for two’s complement serial audio data.
MCLK37Master Clock (Input) - Clock source for the delta-sigma modulators.
SCLK38Serial Clock (Input/Output) - Serial clock for the serial audio interface.
SDIN39Serial Audio Data Input (Input) - Input for two’s complement serial audio data.
LRCK40
GND/Thermal Pad-
Positive Voltage Reference (Output) - Positive reference voltage for the internal sampling cir-
cuits.
Microphone Bias (Output) - Low noise bias supply for an external microphone. Electrical characteristics are specified in the DC Electrical Characteristics table.
21,22
Line-Level Analog Inputs (Input) - Single-ended stereo line-level analog inputs.
Line-Level Analog Inputs (Input) - Single-ended stereo line-level analog inputs.
29,30
Speaker/Headphone Switch (Input) - Powers down the left and/or right channel of the speaker
and/or headphone outputs.
32Reset (Input) - The device enters a low power mode when this pin is driven low.
Digital Interface Power (Input) - Determines the required signal level for the serial audio interface and host control port.
Left Right Clock (Input/Output) - Determines which channel, Left or Right, is currently active on
the serial audio data line.
Ground reference for PWM power FETs and charge pump; thermal relief pad for optimized heat
dissipation.
DS680F29
1.1I/O Pin Characteristics
Input and output levels and associated power supply voltage are shown in the table below. Logic levels
should not exceed the corresponding power supply voltage.
3/1/13
CS42L52
Power
Supply
VL
VASPKR/HPInput--1.65 V - 2.63 V
VP
Pin NameI/OInternal
DriverReceiver
Connections
RESETInput--1.65 V - 3.47 V, with Hysteresis
AD0Input--1.65 V - 3.47 V, with Hysteresis
SCLInput--1.65 V - 3.47 V, with Hysteresis
SDAInput/
Output
MCLKInput--1.65 V - 3.47 V
LRCKInput/
Output
SCLKInput/
Output
SDOUTOutputWeak Pullup
SDINInput--1.65 V - 3.47 V
SPKR_OUTA+ Output-1.6 V - 5.25 V Power MOSFET-
SPKR_OUTA- Output-1.6 V - 5.25 V Power MOSFET-
SPKR_OUTB+ Output-1.6 V - 5.25 V Power MOSFET-
SPKR_OUTB- Output-1.6 V - 5.25 V Power MOSFET-
-1.65 V - 3.47 V, CMOS/Open
Drain
Weak Pullup
(~1 M)
Weak Pullup
(~1 M)
(~1 M)
1.65 V - 3.47 V, CMOS1.65 V - 3.47 V
1.65 V - 3.47 V, CMOS1.65 V - 3.47 V
1.65 V - 3.47 V, CMOS
1.65 V - 3.47 V, with Hysteresis
10DS680F2
2. TYPICAL CONNECTION DIAGRAM
Note 4
Note 3
Note 2
Note 1
1 µF
+1.8 V to +2.5 V
0.1 µF
1 µF
DGND
VL
0.1 µF
+1.8 V to + 3 .3 V
SCL
SDA
RESET
2 k
LRCK
Digital Audio
Processor
MCLK
SCLK
VD
MIC1-
AIN3A/MIC1A
Microphone 1
SDIN
SDOUT
CS42L52
2 k
MICBIAS
+1.8 V to + 2 .5 V
HP/LIN E_OUTB
HP/LINE_OUTA
AIN1A
Left 1
1800 pF
1800 pF
100 k
100
AIN1B
Right 1
*
*
R
L
0.1 µF
VA
Headphone Out
Left & Right
Line Level Out
Left & Right
FLYP
FLYN
-VHPFILT
0.1 µF
51.1
0.022 µF
100 k
100
SPKR_OUTA+
SPKR_OUTA-
SPKR/HP
51.1
0.022 µF
MIC1+
AIN4A/MIC2A
MIC2+
AIN4B/MIC2B
Microphone 2
MIC2-
AIN3B/MIC1B
100 k
R
L
100 k
1 µF
1 µF
0.1 µF
+VHP
1 µF
10 µF
VQ
AGND
* Capacitors must be C0G or equivalent
150 pF
AFILTA
AFILTB
150 pF
1 µF
**
FILT+
1 µF
1 µF
1 µF
1 µF
* *Use low ESR ceramic capacitors.
**
**
See Note 5
SPKR_OUTB+
SPKR_OUTB-
1 µF
VP
VP
+1.6 V to
+5 V
Stereo Speakers
AIN2A
Left 2
1800 pF
1800 pF
100 k
100
AIN2B
Right 2
*
*
100 k
100
1 µF
1 µF
0.1 µF
0.1 µF
Analog
Input 1
Analog
Input 2
10 µF
Mic-Level
Inputs
47 k
Notes:
1. Recommended values for the default charge pump switching
frequency. The required capacitance follows an inverse
relationship with the charge pump’s switching frequency. When
increasing the switching frequency, the capacitance may
decrease; when lowering the switching frequency, the
capacitance must increase.
2. Larger capacitance reduces the ripple on the internal
amplifier’s supply. This may reduce the distortion at higher
output power levels.
3. Additional bulk capacitance may be added to improve PSRR
at low frequencies.
4. These capacitors serve as a charge reservoir for the internal
switched capacitor ADC modulators. They are only needed
when the PGA (Programmable Gain Amplifier) is bypassed.
5. Series resistance in the path of the power supplies must be
avoided. Any voltage drop on VHP will dire c tly impact the
negative charge pump supply (-VHPFILT) and clip the audio
output.
6. The value of R
L
, a current-limiting resistor used with electret
condenser microphones, is dictated by the microphone
cartridge.
7. The negative terminal of the MICx inputs connects to the
ground pin of the microphone cartridge. Gain is applied only to
the positive te rminal.
Note 6
Note 7
Note 7
AD0
Figure 1. Typical Connection Diagram
3/1/13
CS42L52
DS680F211
3/1/13
CS42L52
3. CHARACTERISTIC AND SPECIFICATIONS
RECOMMENDED OPERATING CONDITIONS
AGND=DGND=0 V, All voltages with respect to ground.
ParametersSymbol Min MaxUnits
DC Power Supply
AnalogVA1.652.63V
Headphone Amplifier+VHP1.652.63V
Speaker AmplifierVP1.605.25V
DigitalVD 1.652.63V
Serial/Control Port InterfaceVL1.653.47V
Ambient Temperature Commercial - CNZT
A
-40+85C
ABSOLUTE MAXIMUM RATINGS
AGND = DGND = 0 V; All voltages with respect to ground.
ParametersSymbolMinMaxUnits
DC Power SupplyAnalog
Speaker
Digital
Serial/Control Port Interface
Input Current(Note 1)I
External Voltage Applied to Analog Input(Note 2)
External Voltage Applied to Analog Output
External Voltage Applied to Digital Input(Note 2)V
WARNING:Operation at or beyond these limits may result in permanent damage to the device. Normal operation
is not guaranteed at these extremes.
VA, VHP
VP
VD
VL
in
V
IN
V
IN
IND
A
stg
-0.3
-0.3
-0.3
-0.3
-±10mA
AGND-0.3VA+0.3
-VHP - 0.3+VHP + 0.3
-0.3VL+ 0.3V
-50+115°C
-65+150°C
3.0
6.0
3.0
4.0
V
V
V
V
V
V
Notes:
1. Any pin except supplies. Transient currents of up to ±100 mA on the analog input pins will not cause
SCR latch-up.
2. The maximum over/under voltage is limited by the input current.
12DS680F2
3/1/13
CS42L52
ANALOG INPUT CHARACTERISTICS
Test Conditions (unless otherwise specified): Input sine wave (relative to digital full scale): 1 kHz through passive input filter; All
Supplies = VA; T
fied; “Required Initialization Settings” on page 37 written on power up.
Analog In to ADC (PGA bypassed)
Dynamic Range A-weighted
Total Harmonic Distortion + Noise -1 dBFS
Analog In to PGA to ADC
Dynamic Range
PGA Setting: 0 dBA-weighted
PGA Setting: +12 dBA-weighted
Total Harmonic Distortion + Noise
PGA Setting: 0 dB -1 dBFS
-60 dBFS
PGA Setting: +12 dB -1 dBFS--85-79--83-77dB
Analog In to MIC Pre-Amp (+16 dB) to PGA to ADC
Dynamic Range
PGA Setting: 0 dBA-weighted
Total Harmonic Distortion + Noise
PGA Setting: 0 dB -1 dBFS--76---74-dB
Analog In to MIC Pre-Amp (+32 dB) to PGA to ADC
Dynamic Range
PGA Setting: 0 dBA-weighted
Total Harmonic Distortion + Noise
PGA Setting: 0 dB -2 dBFS--74---71-dB
Other Characteristics
DC Accuracy
Interchannel Gain Mismatch-0.2--0.2-dB
Gain Drift-±100--±100-p pm/°C
Offset ErrorSDOUT Code with HPF On-352--352-LSB
Input
Interchannel Isolation-90--90-dB
HP Amp to Analog Input IsolationR
(Note 3)R
Speaker Amp to Analog Input Isolation-60--60-dB
Full-scale Input VoltageADC
Input Impedance (Note 4)ADC
3. Measured with DAC delivering full-scale output into specified load.
4. Measured between analog input and AGND.
= +25C; Sample Frequency = 48 kHz; Measurement Bandwidth is 20 Hz to 20 kHz unless otherwise speci-
A
VA = 2.5VVA = 1.8V
Parameters MinTypMaxMinTypMaxUnit
unweighted9390
-20 dBFS
-60 dBFS
unweighted9289
unweighted8582
unweighted
unweighted
= 10 k
L
= 16
L
PGA (0 dB)
PGA (+12 dB)
MIC (+16 dB)
MIC (+32 dB)
PGA
MIC
-
-
-
-
-
-
-
-
-
-
-
0.73•VA
0.73•VA
-
-
-
99
96
-86
-76
-36
96
93
91
88
-88
-33
86
83
76
73
100
70
0.769•VA
0.770•VA
0.194•VA
0.115•VA
0.019•VA
20
39
50
-
-
-80
-
-30
-
-
-
-
-82
-27
-
-
-
-
-
-
0.83•VA
0.83•VA
-
-
-
90
87
-
-
-
89
86
82
79
-
-
-
-
-
-
-
-
0.73•VA
0.73•VA
-
-
-
96
93
-84
-73
-33
95
92
88
85
-86
-32
83
80
74
71
100
70
0.769•VA
0.770•VA
0.194•VA
0.1 15•VA
0.019•VA
20
39
50
-
-
-78
-
-27
-
-
-
-
-80
-26
-
-
-
-
-
-
0.83•VA
0.83•VA
-
-
-
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
Vpp
Vpp
Vpp
Vpp
Vpp
k
k
k
DS680F213
3/1/13
CS42L52
ADC DIGITAL FILTER CHARACTERISTICS
Parameters (Note 5)MinTypMaxUnit
Passband (Frequency Response) to -0.1 dB corner0-0.4948Fs
Passband Ripple-0.09-0.17dB
Stopband0.6--Fs
Stopband Attenuation33--dB
Total Group Delay -7.6/Fs- s
5. Response is clock-dependent and will scale with Fs. Note that the response plots (Figures 26 to 29 on
page 78) have been normalized to Fs and can be de-normalized by multiplying the X-axis scale by Fs.
HPF parameters are for Fs = 48 kHz.
-
-
3.6
24.2
5
/Fs0s
-
-
Hz
Hz
14DS680F2
3/1/13
CS42L52
ANALOG OUTPUT CHARACTERISTICS
Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; All Supplies = VA; TA = +25C;
Sample Frequency = 48 kHz; Measurement bandwidth is 20 Hz to 20 kHz; Test load R
(see Figure 2); Test load RL = 16 CL = 10 pF (see Figure 2) for the headphone output; HP_GAIN[2:0] = 011; “Required Initial-
ization Settings” on page 37 written on power up.
VA = 2.5 VVA = 1.8 V
Parameters
R
= 10 k
L
Dynamic Range
18- to 24-BitA-weighted
unweighted
16-BitA-weighted
Total Harmonic Distortion + Noise
18- to 24-Bit0 dB
16-Bit0 dB
RL = 16
Dynamic Range
18- to 24-Bit A-weighted
16-BitA-weighted
Total Harmonic Distortion + Noise
18- to 24-Bit0 dB
16-Bit0 dB
Other Characteristics for R
Output ParametersModulation Index (MI)
(Note 7)Analog Gain Multiplier (G)
Full-scale Output Voltage (2•G•MI•VA) (Note 7)See “Line Output Voltage Level Characteristics” on
Full-scale Output Power (Note 7)See “Headphone Output Power Characteristics” on page 18
Interchannel Isolation (1 kHz)16
Speaker Amp to HP Amp Isolation-80--80-dB
Interchannel Gain Mismatch-0.10.25-0.10.25dB
Gain Drift-±100--±100-ppm/°C
AC Load Resistance (R
Load Capacitance (C
L
)(Note 8)--150--150pF
L
(Note 6)MinTypMaxMinTypMaxUnit
92
89
-
unweighted
-20 dB
-60 dB
-20 dB
-60 dB
unweighted
unweighted
-20 dB
-60 dB
-20 dB
-60 dB
= 16 or 10 k
L
)(Note 8)16--16--
10 k
-
-
-
-
-
-
-
92
89
-
-
-
-
-
-
-
-
-
-
page 19
-
-
98
95
96
93
-86
-75
-35
-86
-73
-33
98
95
96
93
-75
-75
-35
-75
-73
-33
0.6787
0.6047
80
95
= 10 k CL = 10 pFfor the line output
L
-
-
-
-
-80
-
-29
-
-
-
-
-
-
-
-69
-
-29
-
-
-
-
-
-
-
89
86
89
86
95
92
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
93
90
-88
-72
-32
-88
-70
-30
95
92
93
90
-75
-72
-32
-75
-70
-30
0.6787
0.6047
80
93
-
-
-
-
-82
-
-26
-
-
-
-
-
-
-
-69
-
-26
-
-
-
-
-
-
-
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
Vpp
dB
dB
6. One (least-significant bit) LSB of triangular PDF dither is added to data.
7. Full-scale output voltage and power is determined by the gain setting, G, in register “Headphone Analog
Gain” on page 51. High gain settings at certain VA and VHP supply levels may cause clipping when the
audio signal approaches full-scale, maximum power output, as shown in Figures 22 - 25 on page 75.
DS680F215
3/1/13
CS42L52
8. See Figure 2. RL and CL reflect the recommended minimum resistance and maximum capacitance required for the internal op-amp's stability and signal integrity. In this circuit topology, C
will effectively
L
move the band-limiting pole of the amp in the output stage. Increasing this value beyond the recommended 150 pF can cause the internal op-amp to become unstable.
ANALOG PASSTHROUGH CHARACTERISTICS
Test Conditions (unless otherwise specified): Input sine wave (relative to full-scale): 1 kHz through passive input filter; PGA and
HP/Line Gain = 0 dB; All Supplies = VA; T
“Required Initialization Settings” on page 37 written on power up.
= +25C; Sample Frequency = 48 kHz; Measurement Bandwidth is 20 Hz to 20 kHz;
A
VA = 2.5 VVA = 1.8 V
unweighted
-60 dBFS
unweighted
-60 dBFS
-
-
-
-
-
-
-
-
-
-
-96
-93
-70
-73
-33
-96
-93
-70
-73
-33
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-94
-91
-70
-71
-31
-94
-91
-70
-71
-31
-
-
-
-
-
-
-
-
-
-
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
16DS680F2
3/1/13
CS42L52
PWM OUTPUT CHARACTERISTICS
Test conditions (unless otherwise specified): Input test signal is a full scale 997 Hz signal; MCLK = 12.2880 MHz; Measurement
Bandwidth is 20 Hz to 20 kHz; Sample Frequency = 48 kHz; Test load R
full-bridge; VD = VL = VA = VHP = 1.8 V; PWM Modulation Index of 0.85; PWM Switch Rate = 384 kHz; “Required Initialization
Settings” on page 37 written on power up.
(Note 9)
Parameters (Note 10)Symbol ConditionsMin TypMax Units
Parameters (Note 10)Symbol ConditionsMin TypMax Units
MOSFET On Resistance R
MOSFET On ResistanceR
DS(ON)
DS(ON)
EfficiencyVP = 5.0 V, P
Output Operating Peak CurrentI
VP Input Current During ResetI
PC
VP
VP = 3.7V, Id = 0.5 A-640-m
VP = 2.5V, Id = 0.5 A-760-m
= 2 x 0.8 W, RL = 8 -81-%
O
--1.5A
RESET, pin 32, is held low
-0.85.0µA
9. The PWM driver should be used in captive speaker systems only.
10. Optimal PWM performance is achieved when MCLK > 12 MHz.
HEADPHONE OUTPUT POWER CHARACTERISTICS
Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; Sample Frequency = 48 kHz;
Measurement Bandwidth is 20 Hz to 20 kHz; Test load RL = 16 CL = 10 pF (see Figure 2); “Required Initialization Settings”
on page 37 written on power up.
Parameters VA = 2.5V
AOUTx Power Into R
= 16
L
Min Typ Max
Min Typ Max
VA = 1.8V
Unit
HP_GAIN[2:0] Analog Gain (G) VHP
0000.39591.8 V -14--7 -mW
2.5 V -14--7 -mW
0010.45711.8 V -19--10 -mW
2.5 V -19--10 -mW
0100.51111.8 V -23--12 -mW
2.5 V -23--12 -mW
011 (default)0.60471.8 V (Note 11)-17 -mW
2.5 V -32--17 -mW
1000.70991.8 V (Note 11)-23 -mW
2.5 V -44--23 -mW
1010.83991.8 V (Note 7), Figure 22 on page 74 mW
2.5 V -32 -mW
1101.00001.8 V (Note 7, 11) See Figures 22 an d 23 on page 74mW
11. VHP settings lower than VA reduces the headroom of the headphone amplifier. As a result, the DAC
may not achieve the full THD+N performance at full-scale output voltage and power.
18DS680F2
3/1/13
CS42L52
LINE OUTPUT VOLTAGE LEVEL CHARACTERISTICS
Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; measurement bandwidth is 20 Hz
to 20 kHz; Sample Frequency = 48 kHz; Test load R
page 37 written on power up.
Parameters VA = 2.5V
AOUTx Voltage Into RL = 10 k
HP_GAIN[2:0] Analog
Gain (G)
0000.39591.8 V -1.34--0.97-V
0010.45711.8 V -1.55--1.12-V
0100.51111.8 V -1.73--1.25-V
011 (default)0.60471.8 V -2.05-1.411.48 1.55V
1000.70991.8 V -2.41--1.73-V
1010.83991.8 V -2.85-2.05V
1 1 01.00001.8 V -3.39--2.44-V
1111.14301.8 V (See (Note 11)-2.79-V
VHP
2.5 V -1.34--0.97-V
2.5 V -1.55--1.12-V
2.5 V -1.73--1.25-V
2.5 V 1.952.052.15-1.48-V
2.5 V -2.41--1.73-V
2.5 V -2.85--2.05-V
2.5 V -3.39--2.44-V
2.5 V -3.88--2.79-V
= 10 k CL = 10 pF (see Figure 2); “Required Initialization Settings” on
L
Min Typ Max
VA = 1.8V
Min Typ Max
Unit
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
COMBINED DAC INTERPOLATION AND ONCHIP ANALOG FILTER RESPONSE
Parameters (Note 12)MinTypMaxUnit
Frequency Response 10 Hz to 20 kHz-0.01-+0.08dB
Passband to -0.05 dB corner
to -3 dB corner
StopBand0.5465--Fs
StopBand Attenuation (Note 13)50--dB
Group Delay-9/Fs-s
De-emphasis Error Fs = 32 kHz
Fs = 44.1 kHz
Fs = 48 kHz
12. Response is clock dependent and scales with Fs. Note that the response plots (Figures 30 and 33 on
page 78) have been normalized to Fs and can be de-normalized by multiplying the X-axis scale by Fs.
13. Measurement Bandwidth is from Stopband to 3 Fs.
0
0
-
-
-
-
-
-
-
-
0.4780
0.4996
+1.5/+0
+0.05/-0.25
-0.2/-0.4
Fs
Fs
dB
dB
dB
DS680F219
3/1/13
t
h(SK-SDO)
//
//
//
//
//
//
//
//
t
s(SD-SK)
MSB
MSB
MSB-1
MSB-1
LRCK
SCLK
SDOUT
SDIN
t
d(MSB)
t
s(LK-SK)
t
P
t
h
t
s(SDO-SK)
Figure 3. Serial Audio Interface Timing
SWITCHING SPECIFICATIONS - SERIAL PORT
Inputs: Logic 0 = DGND, Logic 1 = VL, SDOUT C
ParametersSymbol Min MaxUnits
RESET
MCLK Frequency (Note 15)
MCLK Duty Cycle
pin Low Pulse Width(Note 14)
Slave Mode
Input Sample Rate (LRCK)
LRCK Duty Cycle
SCLK Frequency
SCLK Duty Cycle
LRCK Setup Time Before SCLK Rising Edge
LRCK Edge to SDOUT MSB Output Delay
SDOUT Setup Time Before SCLK Rising Edge
SDOUT Hold Time After SCLK Rising Edge
SDIN Setup Time Before SCLK Rising Edge
SDIN Hold Time After SCLK Rising Edge
Master Mode
Output Sample Rate (LRCK) All Speed Modes
LRCK Duty Cycle
SCLK Frequency SCLK=MCLK mode
SCLK Duty Cycle
LRCK Edge to SDOUT MSB Output Delay
SDOUT Setup Time Before SCLK Rising Edge
SDOUT Hold Time After SCLK Rising Edge
SDIN Setup Time Before SCLK Rising Edge
SDIN Hold Time After SCLK Rising Edge
14. After powering up the CS42L52, RESET should be held low after the power supplies and clocks are
settled.
15. See “Example System Clock Frequencies” on page 76 for typical MCLK frequencies.
Bus Free Time Between Transmissions
Start Condition Hold Time (prior to first clock pulse)
Clock Low time
Clock High Time
Setup Time for Repeated Start Condition
SDA Hold Time from SCL Falling(Note 16)
SDA Setup time to SCL Rising
Rise Time of SCL and SDA
Fall Time SCL and SDA
Setup Time for Stop Condition
Acknowledge Delay from SCL Falling
16. Data must be held for sufficient time to bridge the transition time, tfc, of SCL.
f
t
t
t
hdst
t
low
t
high
t
sust
t
hdd
t
sud
t
t
t
susp
t
ack
scl
irs
buf
rc
fc
CS42L52
-100kHz
550-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
DS680F221
3/1/13
CS42L52
DC ELECTRICAL CHARACTERISTICS
AGND = 0 V; All voltages with respect to ground.
ParametersMinTypMaxUnits
VQ Characteristics
Nominal Voltage
Output Impedance
DC Current Source/Sink
BIASLVL[2:0] = 101
DC Output Current
Power Supply Rejection Ratio (PSRR)1 kHz
Power Supply Rejection Ratio Characteristics
PSRR @1 kHz (Note 17)PGA to ADC
ADC
DAC (HP and Line Amps)
PSRR @60 Hz (Note 17)PGA to ADC
DAC (HP and Line Amps)
PSRR @217 Hz Full-Bridge PWM Outputs-56-dB
(Note 18)
ADC
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
0.5•VA
23
-
0.5•VA
0.6•VA
0.7•VA
0.8•VA
0.83•VA
0.91•VA
-
50
44
60
60
22
42
60
-
-
1
-
-
-
-
-
-
1
-
-
-
-
-
-
-
V
kA
V
V
V
V
V
V
mA
dB
dB
dB
dB
dB
dB
dB
17. Valid with the recommended capacitor values on FILT+ and VQ. Increasing the capacitance will also
increase the PSRR.
18. The PGA is biased with VQ, created from a resistor divider from the VA supply. Increasing the capacitance on VQ will also increase the PSRR at low frequencies. A 10 µF capacitor on VQ improves the
PSRR to 42 dB.
DIGITAL INTERFACE SPECIFICATIONS AND CHARACTERISTICS
Parameters (Note 19)Symbol Min MaxUnits
Input Leakage CurrentI
Input Capacitance -10pF
1.8 V - 3.3 V Logic
High-Level Output Voltage (I
Low-Level Output Voltage (I
High-Level Input Voltage VL = 1.65 V
Low-Level Input Voltage V
= -100 A)V
OH
= 100 A)V
OL
VL = 1.8 V
VL = 2.0 V
VL > 2.0 V
in
OH
OL
V
IH
IL
19. See “I/O Pin Characteristics” on page 10 for serial and control port power rails.
-±10A
VL - 0.2-V
-0.2V
0.85•VL
0.77•VL
0.68•VL
0.65•VL
-0.30•VLV
-
-
-
-
V
V
V
V
22DS680F2
3/1/13
POWER CONSUMPTION See (Note 20).
Power Ctl. RegistersTypical Current (mA)
Operation
1
Off (Note 21)
Standby (Note 22)xxxx1xxxx x x x
2
Mono RecordADC 1 110011111111111
3
PGA to ADC1010011111111111
MIC to PGA to ADC
MIC to PGA to ADC
Stereo RecordADC1100011111111111
4
MIC to PGA to ADC
Mono Playback to Headphone1 111011110111111
5
Mono Playback to Speaker1 111011111111010
6
Stereo Playback to Headphone1 111011110101111
7
Stereo Playback to Speaker1111011111111010
8
Stereo Passthrough to Head-
9
phone
Mono Record and Playback
10
PGA in (no MIC) to Mono HP
Phone Monitor
11
MIC (w/bias) in to Mono Out
Stereo Record and Playback
12
PGA in (no MIC) to St. HP Out
Stereo Rec. and Full Playback
13
PGA (no MIC) to St. HP and SPK
(with Bias)
(no Bias)
PGA to ADC0000011111111111
(no Bias)
02h03h04h
PDN_PGAB
PDN_PGAA
PDN_ADCB
PDN_ADCA
PDN
PDN_MICB
PDN_MICA
PDN_MICBIAS
PDN_HPB[1:0]
xxxxxxxxx x x x
1010010011111111
1010010111111111
0000000111111111
1111011110101111
1010011111101111
1010010011101111
0000011110101111
0000011110101010
PDN_HPA[1:0]
i
VHPiVA
PDN_SPKB[1:0]
PDN_SPKA[1:0]
V
1.8 0.00 0.000.00
2.5 0.000.000.000.00
1.8 0.00 0.000.01
2.5 0.000.000.020.05
1.8 0.00 1.672.32
2.5 0.001.873.7214.05
1.8 0.002.12.31
2.5 0.002.33.7215.13
1.8 0.00 3.482.32
2.5 0.00 3.713.7218.65
1.8 0.00 3.152.32
2.5 0.003.373.7317.83
1.8 0.00 2.312.37
2.5 0.002.533.8215.95
1.8 0.00 3.182.37
2.5 0.003.423.8118.15
1.8 0.00 5.322.37
2.5 0.00 5.573.8123.53
1.8 1.59 1.992.72
2.5 2.072.624.2722.43
1.8 0.000.204.42
2.5 0.000.226.7721.21
1.8 2.77 2.002.91
2.5 3.272.634.2825.48
1.8 0.000.204.38
2.5 0.000.226.8021.28
1.8 2.791.911.06
2.5 3.182.141.8117.85
1.8 1.77 3.954.28
2.5 2.134.776.6333.90
1.8 1.76 5.334.28
2.5 2.156.196.6937.65
1.8 2.76 5.054.64
2.5 3.215.907.1740.78
1.8 3.49 5.247.20
2.5 3.956.10 10.4655.07
i
VD
i
VL
VL=3.3V
(Note 23)
0.000.00
0.000.00
0.030.00
0.030.00
0.030.00
0.030.00
0.030.00
0.030.00
0.030.00
0.010.00
0.011.00
0.010.00
0.011.00
0.010.00
0.030.00
0.030.00
0.030.00
0.031.00
CS42L52
i
VP
VP=3.7V
Total
Power
(mW
0.00
0.02
7.24
7.99
10.49
9.90
8.48
10.04
13.90
11.36
12.05
13.84
11.98
10.39
18.05
20.52
22.46
32.47
rms
)
20. Unless otherwise noted, test conditions are as follows: All zeros input, slave mode, sample
rate = 48 kHz; No load. Digital (VD) and logic (VL) supply current will vary depending on speed mode
and master/slave operation. “Required Initialization Settings” on page 37 written on power up.
21. RESET
22. RESET
pin 25 held LO, all clocks and data lines are held LO.
pin 25 held HI, all clocks and data lines are held HI.
23. VL current will slightly increase in master mode.
DS680F223
4. APPLICATIONS
4.1Overview
4.1.1Basic Architecture
The CS42L52 is a highly integrated, low-power, 24-bit audio CODEC comprised of a stereo analog-todigital converter (ADC), a stereo digital-to-analog converter (DAC), a digital PWM modulator and two fullbridge power back-ends. The ADC and DAC are designed using multibit delta-sigma techniques - the
DAC operates at an oversampling ratio of 128Fs and the ADC oper ates at 64Fs, where Fs is equal to the
system sample rate.
The different clock rates maximize power savings while maintaining high performance. The PWM modulator operates at a fixed frequency of 384 kHz. The power FETs are configure d for either stereo full-bridge
or mono parallel full-bridge output. The CODEC operates in one of four sample rate speed modes: Quar ter, Half, Single, and Double. It accepts and is capable of generating serial port clocks (SCLK, LRCK) derived from an input Master Clock (MCLK).
4.1.2Line and MIC Inputs
The analog input portion of the CODEC allows selection from and configuration of multiple combinations
of stereo and microphone (MIC) sources. Eight line inputs with an option for two balanced MIC inputs, a
MIC bias output, and a Programmable Gain Amplifier (PGA) comprise the analog front- end.
3/1/13
CS42L52
4.1.3Line and Headphone Outputs
The analog output portion of the CODEC includes a headphone amplifier capable of driving headphone
and line-level loads. An on-chip charge pump creates a negative headphone supply allowing a full-scale
output swing centered around groun d. Th is elim in at es the need for large DC-Blocking capacitors and allows the amplifier to deliver more power to headphone loads at lower supply voltages.
4.1.4Speaker Driver Outputs
The Class D power amplifiers drive 8 ohm (stereo) and 4 ohm (mono ) speakers d irectly, without the ne ed
for an external filter. The power MOSFETS are powered directly from a battery eliminating the efficiency
loss associated with an external regulator. Battery level monitoring and compensation maintains a stead y
output as battery levels fall. NOTE: The CS42L52 should only be used in captive speaker systems where
the outputs are permanently tied to the speaker terminals.
4.1.5Fixed Function DSP Engine
The fixed-function digital signal processing engine processes both the PCM serial input data and ADC
output data, allowing a mix between the two. Independent volume control, le ft/right channel swaps, mono
mixes, tone control, and limiting functions also comprise the DSP engine.
4.1.6Beep Generator
The beep generator delivers tones at select frequencies across approximately two octave major scales.
With independent volume control, beeps may be configured to occur continuously, perio dically, or at single time intervals.
4.1.7Power Management
Three control registers provide independent power-down control of the ADC, DAC, PGA, MIC pre-amp,
MIC bias, Headphone, and Speaker outputs, allowing operation in select applications with minimal power
consumption.
“Power Down PGAx” on page 42
“PGAx Volume” on page 56
“Analog Front-End Volume Setting B=A” on page 50
“Ch. x Analog Soft Ramp” on page 49
“Ch. x Analog Zero Cross” on page 49
“ADC Input Select” on page 48
“PGA Input Mapping” on page 49
“MIC Bias Level” on page 48
“Power Down MIC Bias” on page 43
“Power Down ADCx” on page 43
“Power Down ADC Charge Pump” on page 42
“Invert ADC Signal Polarity” on page 51
“ADCx High-Pass Filter Freeze” on page 49
“ADCx High-Pass Filter” on page 49
“HPF x Corner Frequency” on page 50
“ADCx Overflow (Read Only)” on page 71
“ADC Mute” on page 51
“ADCx Volume” on page 57
“ALCx Enable” on page 67
“ALCx Soft Ramp Disable” on page 55
“ALCx Zero Cross Disable” on page 56
“ALC Attack Rate” on page 67
“ALC Release Rate” on page 68
“ALC Maximum Threshold” on page 68
“ALC Minimum Threshold” on page 69
“Noise Gate All Channels” on page 69
“Noise Gate Enable” on page 69
“Noise Gate Threshold and Boost” on page 70
“Noise Gate Delay Timing” on page 70
“Digital Sum” on page 50
“Digital MUX” on page 50
4.2.1MIC Inputs
MIC1-
-
+
+
MIC1+
MIC2-
-
+
-
+
MIC2+
23
21
24
22
MICACFG=’1'b
MICBCFG=’1'b
MICAGAIN[4:0]
MICBGAIN[4:0]
16..32 dB/
1 dB steps
16..32 dB/
1 dB steps
PDN_MICA=’0'b
PDN_MICB=’0'b
to summing
PGA A
Note: Output to PGA = (MIC+ - MIC-)*gain + MIC
-
to summing
PGA B
MIC1A
-
+
MIC2A
MIC1B
-
+
MIC2B
23
21
24
22
MICACFG=’0'b
MICBCFG=’0'b
MICAGAIN[4:0]
MICBGAIN[4:0]
16..32 dB/
1 dB steps
16..32 dB/
1 dB steps
PDN_MICA=’0'b
PDN_MICB=’0'b
MICASEL
MICBSEL
to summing
PGA A
to summing
PGA B
VQ
VQ
Figure 6. Single-Ended MIC ConfigurationFigure 7. Differential MIC Configuration
The input pins 21, 22, 23, and 24 accept stereo line-level or microphone signals. For microphone inputs,
either single-ended or differential configuration is allowed, providing programmable pre-amplification of
low-level signals. In the single-ended configuration, an internal MUX chooses one of two stereo sets (s election is made independently on channels A and B). In the differential configuration, an internal voltage
follower cascaded with the pre-amplifier maintains high input impedance and provides noise rejection
above the MICxGAIN setting. The pre-amps are biased to VQ in both configurations.
3/1/13
CS42L52
Referenced ControlRegister Location
MICxCFG............................
PDN_MICx..........................
MICxGAIN...........................
“MICx Configuration” on page 55
“Power Down MICx” on page 43
“MICx Gain” on page 55
4.2.2Automatic Level Control (ALC)
When enabled, the ALC monitors the analog input signal after the digital attenuator, detects when peak
levels exceed the maximum (MAX) threshold settings, and responds by applying attenuation as necessary to maintain the resulting level below the MAX threshold. To apply this attenuation, the ALC first lowers the PGA gain settings and then increases the digital attenuation levels. All attenuation is applied at a
programmable attack rate.
When input signal levels fall below the minimum (MIN) threshold, the ALC responds by removing any attenuation that it has previously applied until all ALC-applied attenuation has been removed or until the
MAX threshold is again crossed. To remov e th is at te nu at ion , th e AL C firs t de cr ea se s the dig ita l att en uation levels and then increases the PGA gain. All attenuation is removed at a programmable release rate.
It should be noted that the ALC is applied independently to channels A and B with one exception: the input
signals on both channels A and B must be below the MIN threshold in order for the ALC attenuation to be
released on channel B.
Attack and release rates are affected by the ADC soft-r amp/zero-cross settings and sample ra te, Fs. ALC
soft-ramp and zero-cross dependency may be independently enabled/disabled.
Recommended settings: Best level control may be realized with the fastest attack and slowest release
setting with soft ramp enabled in the control registers.
Notes:
1. When ALC x is enabled and the PGAxVOL[5:0] is set above 12 dB, the ADCxVOL[7:0] should not be
set below 0 dB.
2. The maximum realized gain must be set in the PGAxVOL register. The ALC will only apply the gain
set in the PGAxVOL.
3. The ALC maintains the output signal between the MIN/MAX thresholds. As input signal level changes,
the level-controlled output may not always be the same but always falls within the thresholds.
26DS680F2
Output
(after A L C )
Input (before ALC)
RRATE[5:0]
PGA G ain and/or
Attenuator
ALC
Response
MAX[2:0]
ARATE[5:0
]
below f u ll s c ale
MIN[2:0]
below fu ll s cale
MIN[2:0]
below f u ll s c a le
MAX[2:0]
below f u ll s c a le
Figure 8. ALC
-96-40
THRESH[2:0]
Maximum Attenuation*
-52 dB
Output
(dB)
Input (dB)
N
G
E
N
=
1
N
G
E
N
=
0
-80 dB
-64 dB
Figure 9. Noise Gate Attenuation
Referenced Control Register Location
PGAxVOL[5:0
MAX[2:0], MIN[2:0]
“PGAx Vol. and ALCx Transition Ctl.: ALC, PGA A (Address 12h) and ALC, PGA B (Address 13h)” on
“ALC Threshold (Address 2Ch)” on page 68
3/1/13
CS42L52
page 55
4.2.3Noise Gate
The noise gate may be used to mute signal levels that fall below a programmable thre shold. This prevents
the ALC from applying gain to noise. A programma ble delay may be used to set the minimum time before
the noise gate attacks the signal.
Note:Maximum noise gate attenuation levels will depend on the gain applied in either the PGA or MIC
pre-amplifier. For example: If both +32 dB pre- amplification and +12 dB programmable gain is applied,
the maximum attenuation that the noise gate achieves will be 52 dB (-96 + 32 + 12) below full-scale.
Referenced ControlRegister Location
Noise Gate Controls............ “Noise Gate Control (Address 2Dh)” on page 69
“HP/Speaker De-emphasis” on page 53
“PCM Mixer Channel x Mute” on page 58
“PCM Mixer Channel x Volume” on page 58
“Invert PCM Signal Polarity” on page 52
“PCM Mix Channel Swap” on page 64
“ADC Mixer Channel x Mute” on page 58
“ADC Mixer Channel x Volume” on page 58
“ADC Mix Channel Swap” on page 64
“Master Volume Control” on page 63
“Master Playback Mute” on page 52
“Digital Soft Ramp” on page 53
“Digital Zero Cross” on page 53
“Playback Volume Setting B=A” on page 51
“Tone Control Enable” on page 62
“Bass Corner Frequency” on page 62
“Treble Corner Frequency” on page 62
“Bass Gain” on page 63
“Treble Gain” on page 62
“Peak Detect and Limiter” on page 66
“Limiter Soft Ramp Disable” on page 65
“Limiter Zero Cross Disable” on page 66
“Limiter Maximum Threshold” on page 65
“Limiter Cushion Threshold” on page 65
“Limiter Attack Rate” on page 67
“Limiter Release Rate” on page 66
Figure 11. PWM Output StageFigure 12. Analog Output Stage
PWM Control
SPKxMUTE.........................
MUTE50/50.........................
SPKMONO..........................
SPKxVOL[7:0].....................
SPKSWAP...........................
SPKB=A ..............................
BATTCMP...........................
VPREF[3:0] .........................
VPLVL[7:0]..........................
PDN_SPKx[1:0]...................
SPKxSHRT..........................
“Speaker Mute” on page 54
“Speaker Mute 50/50 Control” on page 54
“Speaker MONO Control” on page 54
“Speaker Volume Control” on page 64
“Speaker Channel Swap” on page 54
“Speaker Volume Setting B=A” on page 54
“Battery Compensation” on page 71
“VP Reference” on page 72
“VP Voltage Level (Read Only)” on page 72
“Speaker Power Control” on page 44
“Speaker Current Load Status (Read Only)” on page 72
3/1/13
CS42L52
Referenced ControlRegister Location
Analog Output
HPxMUTE ...........................
HPxVOL[7:0] .......................
PDN_HPx[1:0].....................
HPGAIN[2:0]........................
PASSTHRUx.......................
PASSxMUTE.......................
PASSxVOL[7:0]...................
CHGFREQ ..........................
“Headphone Mute” on page 54
“Headphone Volume Control” on page 63
“Headphone Power Control” on page 44
“Headphone Analog Gain” on page 51
“Passthrough Analog” on page 52
“Passthrough Mute” on page 52
“Passthrough x Volume” on page 57
“Charge Pump Frequency” on page 73
4.3.1Beep Generator
The Beep Generator generates audio frequencies across approximately two octave major sca les. It offers
three modes of operation: Continuous, multiple, and single (one-shot) beeps. Sixteen on and eight off
times are available.
Note:The Beep is generated before the limiter and may affect desired limiting performance. If the limiter function is used, it may be required to set the beep volume sufficiently below the threshold to prevent
the peak detect from triggering. Since the master volume control, MSTxVOL[7:0], will affect the beep vol-
DS680F229
ume, DAC volume may alternatively be controlled using the PMIXxVOL[6:0] bits.
FREQ[3:0]
...
BPVOL[4:0]
ONTIME[3:0]OFFTIME[2:0]
BEEP[1:0] =
'01'
BEEP[1:0] =
'10'
BEEP[1:0] =
'11'
SINGLE-BEEP: Beep turns on at a
configurable frequency (FREQ) and
volume (BPVOL) for the duration of
ONTIME. BEEP must be cleared
and set for additional beeps.
MULTI-BEEP: Beep turns on at a configurable frequency (FREQ)
and volume (BPVOL) for the duration of ONTIME and turns off for
the duration of OFFTIME. On and off cycles are repeated until
BEEP is cleared.
CONTINUOUS BEEP: Beep turns on at a configurable frequency (FREQ) and volume (BPVOL) and remains on
until BEEP is cleared.
Figure 13. Beep Configuration Options
Referenced ControlRegister Location
MSTxVOL[7:0].....................
PMIXxVOL[6:0] ...................
OFFTIME[2:0] .....................
ONTIME[3:0].......................
FREQ[3:0]...........................
BEEP[1:0]............................
BEEPMIXDIS......................
BPVOL[4:0].........................
“Master Volume Control: MSTA (Address 20h) and MSTB (Address 21h)” on page 63
“PCMx Mixer Volume: PCMA (Address 1Ah) and PCMB (Address 1Bh)” on page 58
“Beep Off Time” on page 60
“Beep On Time” on page 60
“Beep Frequency” on page 59
“Beep Configuration” on page 61
“Beep Mix Disable” on page 61
“Beep Volume” on page 61
3/1/13
CS42L52
4.3.2Limiter
When enabled, the limiter monitors the digital input signal before th e DAC and PWM mo dulators, detects
when levels exceed the maximum threshold settings, and lowers the master volume at a pr ogram mable
attack rate below the maximum threshold. When the input signa l level falls below the maximum threshold,
the AOUT volume returns to its original level set in the Master Volume Control register at a programmable
release rate. Attack and release rates are affe cted by the DAC soft-r amp/zero -cross settin gs and sample
rate, Fs. Limiter soft-ramp and zero-cross dependency may be independently enabled/di sabled.
Notes:
1. Recommended settings: Best limiting performance may be realized with the fastest attack and
slowest release setting with soft ramp enabled in th e contro l registers. The MIN bits allow the user to
set a threshold slightly below the maximum thresh old for hystere sis control - this cushions the sound
as the limiter attacks and releases.
2. T h e Lim ite r ma in tains th e ou tp ut sign al be twe en the MIN and MAX thresholds. As the digital input
signal level changes, the level-controlled output may not always be the same but will always fall within
the thresholds.
Referenced ControlRegister Location
Limiter Controls...................
Master Volume Control........
“Limiter Control 2, Release Rate (Address 28h)” on page 66, “Limiter Attack Rate (Address 29h)” on page 67
“Master Volume Control: MSTA (Address 20h) and MSTB (Address 21h)” on page 63
30DS680F2
3/1/13
MAX[2:0]
Output
(after Limiter)
Inp u t
RRATE[5:0]ARATE[5:0]
Volume
Limiter
CUSH[2:0]
ATTACK/RELEASE SOUND
CUS HIO N
MAX[2:0]
Figure 14. Peak Detect and Limiter
CS42L52
4.4Analog In to Analog Out Passthrough
The CS42L52 accommodates analog routing of the analog inp ut signal directly to the headphone amplifiers.
This feature is useful in applications that utilize an FM tuner where audio recovered over-the-air must be
transmitted to the headphone amplifier without digital conversion in the ADC and DAC. This analog
passthrough path reduces power consumption and is immune to modulator switching noise that could
interfere with some tuners.
4.4.1Overriding the ADC Power Down
To accommodate automatic activation of the speaker amplifier when the SPK/HP_SW switch pin changes, the CS42L52 provides the option to automatically power up the ADC whenever the analog signal must
route to the digital PWM modulator, regardless of the PDN_ADC bit. Refer to the table below for details
on how this ADC power-down override functions in accordance with the state of the speaker channels.
The shaded cells represent normal ADC operation when passthrough is disabled.
PDN_ADCPASSTHRUPDN_OVRDSpeaker ChannelADC Status
0xxxPowered UP
0xxPowered DOWN
1
1
When PASSTHRU and PDN_OVRD are enabled, turning the speaker channel ON (by writing ‘11’b to SPKx_PDN[1:0] or by automatic activation of the headphone detect switch, SPK/HP_SW) will automatically
disable the ADCx_PDN in order to convert the analog input to a digital signal for the PWM modulator. This
allows automatic analog input routing to the speaker amplifiers.
DS680F231
0xPowered DOWN
1
OFFPowered DOWN
ONPowered UP
Referenced ControlRegister Location
PDN_ADCx.........................
PASSTHRU.........................
PDN_OVRD........................
SPKx_PDN[1:0]...................
“Power Down ADCx” on page 43
“Passthrough Analog” on page 52
“Power Down ADC Override” on page 43
“Speaker Power Control” on page 44
4.4.2Overriding the PGA Power Down
To accommodate automatic activation of the headphone amplifier when the SPK/HP_SW switch pin
changes, the CS42L52 will automatically power up the PGA whenever passthrough is enabled, regardless of the PDN_PGA setting. Refer to the table below for details on how this PGA power-down override
functions in accordance with the state of the headphone channels. The shaded cells represent normal
PGA operation when passthrough is disabled.
PDN_PGAPASSTHRUHP ChannelPGA Status
0xxPowered UP
0xPowered DOWN
1
When passthrough is enabled, turning the headphone channel ON (by writing ‘11’b to HPx_PDN[1:0] or
by automatic activation of the headphone detect switch, SPK/HP_SW) will automatically disable the
PGAx_PDN in order to transmit the analog signal to the headphone.
Referenced ControlRegister Location
PDN_PGAx.........................
PASSTHRU.........................
HPx_PDN[1:0].....................
“Power Down PGAx” on page 42
“Passthrough Analog” on page 52
“Headphone Power Control” on page 44
1
3/1/13
CS42L52
OFFPowered DOWN
ONPowered UP
4.5PWM Outputs
Note:The PWM speaker amplifiers should not be used in the 384x MCLK modes (18.4320 and
16.9344 MHz).
4.5.1Mono Speaker Output Configuration
The CS42L52 accommodates a stereo as well as a mono speaker output configuration. In mono mode
the output drivers of each channel are connected in parallel to deliver maximum power to a 4 ohm speaker. Refer to the table below for pin mapping in mono configuration.
“Speaker MONO Control” on page 54
“Speaker Channel Swap” on page 54
SPKMONO=0SPKMONO=1
Speaker Output
32DS680F2
4.5.2VP Battery Compensation
-24
-22
-20
-18
-16
-14
-12
-10
-8
-6
1.61.92.22.52.83.13.43.744.34.64.9
Uncompensated
PWM Output
Level
Battery Compensated
PWM Output Level
VP Supply (V)
PWM Output Level (dB)
Figure 15. Battery Compensation
The CS42L52 provides the option to maintain a desired power output level, independent of the VP supp ly.
When enabled, this feature works by monitoring the voltage on the VP supply and reducing the attenua-tion on the speaker outputs when VP voltage levels fall.
Note:The internal ADC that monitors the VP supply operates from the VA supply. Calculations are
based on typical VA levels of 1.8 V and 2.5 V using the VPREF bits.
4.5.2.1Maintaining a Desired Output Level
Using SPKxVOL, the speaker output level must first be attenuated by the decibel equivalent of the expected VP supply range (MAX relative to MIN). The CS42L52 then gradually reduces the attenuation as the
VP supply drops from its maximum level, maintaining a nearly constant power output.
Compensation Example 1 (VP Battery supply ranges from 4.5 V to 3.0 V)
1. Set speaker attenuation (SPKxVOL) to -3.5 dB. The VP supply changes ~3.5 dB.
2. Set the reference VP supply (VPREF) to 4.5 V.
3. Enable battery compensation (BATTC MP ).
The CS42L52 automatically adjusts the output level as the battery discharges.
3/1/13
CS42L52
Compensation Example 2 (VP Battery supply ranges from 5.0 V to 1.6 V)
1. Set speaker attenuation (SPKxVOL) to -10 dB. The VP supply changes ~9.9 dB.
2. Set the reference VP supply (VPREF) to 5.0 V.
3. Enable battery compensation (BATTC MP ).
The CS42L52 automatically adjusts the output level as the battery discharges. Refer to Figure 1 5 on page
33. In this example, the VP supply changes over a wide range, illustrating the accuracy of the CS42L52’s
battery compensation.
4.6Serial Port Clocking
Referenced ControlRegister Location
VPREF................................
SPKxVOL............................
The CODEC serial audio interface port ope rates either as a slav e or master, determ ined by the M/S bit. It
“VP Reference” on page 72
“Speaker Volume Control” on page 64
accepts externally generated clocks in Slave Mode and will generate synchronous clocks derived from an
DS680F233
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CS42L52
input master clock (MCLK) in Master Mode. Refer to the tables below for the required setting in register 05h
and 06h associated with a given MCLK and sample rate.
Referenced ControlRegister Location
M/S
Register 05h
Register 06h
“Master/Slave Mode” on page 46
“Clocking Control (Address 05h)” on page 44
“Interface Control 1 (Address 06h)” on page 46
MCLK
(MHz)
12.2880
11.2896
18.4320
(Slave
Mode
ONLY)
16.9344
(Slave
Mode
ONLY)
12.0000
24.0000
Sample Rate,
Fs (kHz)
8.00001110000
12.00001100000
16.00001010000
24.00001000000
32.00000110000
48.00000100000
96.00000000000
1 1 .02501100000
22.05001000000
44.10000100000
88.20000000000
8.00001110000
12.00001100000
16.00001010000
24.00001000000
32.00000110000
48.00000100000
96.00000000000
8.01821100100
1 1 .02501100000
22.05001000000
44.10000100000
88.20000000000
8.00001110010
11.029411 0 0110
12.00001100010
16.00001010010
22.05881000110
24.00001000010
32.00000110010
44.11760100110
48.00000100010
88.23530000110
96.00000000010
8.00001110011
11.029411 0 0111
12.00001100011
16.00001010011
22.05881000111
24.00001000011
32.00000110011
44.11760100111
48.00000100011
88.23530000111
96.00000000011
SPEED[1:0]
(AUTO=’0’b)
32kGROUPVIDEOCLKRATIO[1:0]MCLKDIV2
Table 1. MCLK, LRCK Quick Decode
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LRCK
SCLK
MSBLSB
MSB
LSB
AOUTA / AINxA
Left C hannelRight C hannel
SDOUT
SDIN
AOUTB / AINxB
MSB
Figure 16. I²S Format
LRCK
SCLK
MSBLSB
MSB
LSB
Left ChannelRight Channel
SDOUT
SDIN
MSB
AOUTA / AINxA
AOUTB / AINxB
Figure 17. Left-Justified Format
LRCK
SCLK
MSBLSB
MSBLSB
Left ChannelRight Chan nel
SDIN
AOUTLAOUTR
Audio Word Length (AWL)
Figure 18. Right-Justified Format (DAC only)
CS42L52
MCLK
(MHz)
Sample Rate,
Fs (kHz)
SPEED[1:0]
(AUTO=’0’b)
8.00001111010
12.00001101010
24.00001001010
32.00000111010
27.0000
44.117601 0 1110
48.00000101010
11.029411 0 1110
22.058810 0 1110
16.00001011010
4.7Digital Interface Formats
The serial port operates in standard I²S, Left-justified, Right-justified (DAC only), or DSP Mode digital interface formats with varying bit dep ths from 16 to 24. Data is cloc ked out of t he ADC or into t he DAC on the
rising edge of SCLK.
32kGROUPVIDEOCLKRATIO[1:0]MCLKDIV2
Table 1. MCLK, LRCK Quick Decode
4.7.1DSP Mode
In DSP Mode, the LRCK acts as a frame sync for 2 data-pac ked words (left and right channel) input on
SDIN and output on SDOUT. The MSB is input/output on the first SCLK rising edge after the frame sync
rising edge. The right channel immediately follows the left channel.
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LRCK
SCLK
MSBLS B
SDIN
HP/LINE OUTB
LSB
Lef t Channel
Right Channel
MSBLSB MSB
Audio Word Length (AWL)
1/fs
HP/LINE OUTA
Figure 19. DSP Mode Format)
CS42L52
When configuring the 16-bit SDOUT word length with an 8 kHz sample rate in master mode and when
SCLK is set equal to MCLK, perform the following write sequences:
Register commands ONLY when entering DSP 16-bit, 8 kHz Fs, SCLK=MCLK, master mode:
Register[Bits] ValueDescription
10x0C[1:0]0x03 Mute the ADC outputs to ensure no audible artifacts are transmitted when changing modes.
2Refer to Section 4.10 Follow the recommended power down sequence for the HP and PWM outputs.
30x02[0]0x01 Power down the CODEC.
40x05[7:0]0x72 Enable 8 kHz Fs for MCLK=12.000 MHz.
50x06[7:0]0x93 Enable DSP 16-bit master mode.
60x07[6]0x01 Enable SCLK=MCLK.
70x33[6]0x01 Undisclosed register command for enabling mode mentioned above.
8Refer to Section 4.9Follow the recommended power up sequence for the HP and PWM outputs.
90x02[0]0x00 Power up the CODEC.
10x0C[1:0]0x03 Mute the ADC outputs to ensure no audible artifacts are transmitted when changing modes.
2Refer to Section 4.10 Follow the recommended power down sequence for the HP and PWM outputs.
30x02[0]0x01 Power down the CODEC.
40x05[7:0]0x20 Enable 48 kHz Fs for MCLK = 12.2880 MHz or re-establish original settings.
50x06[7:0]0x00 Enable Left-Justified 24-bit slave mode or re-establish original settings.
60x07[6]0x00 Disable SCLK=MCLK or re-establish original settings.
70x33[6]0x00 Undisclosed register command for disabling mode mentioned above.
8Refer to Section 4.9Follow the recommended power up sequence for the HP and PWM outputs.
90x02[0]0x00 Power up the CODEC.
100x0C[1:0]0x00 Unmute the ADC outputs.
4.8Initialization
The CODEC enters a Power-down state on initial power-up. The interpolation and decimation filters, deltasigma and PWM modulators, and control port registers are reset. The internal voltage reference, and
switched-capacitor low-pass filters are powered down.
The device remains in Power-down state until the RESET
once RESET
is high and the desired register settings can be loaded per the descriptions in the Section 6.
Once MCLK is valid, the quiescent voltage, VQ, and the internal voltage reference, FILT+, will begin powering up to normal operation. The charge pump slowly powers up and charges the capacitors. Power is then
applied to the headphone amplifiers and switched-capacitor filters, and the analog/digital outputs enter a muted stat e . Once LRCK is valid, MCLK occurre nces are counted over one LRCK period to de termine the MCLK/LRCK frequency ratio and normal operation begins.
36DS680F2
pin is brought high. The control port is accessible
3/1/13
4.9Recommended Power-up Sequence
1. Hold RESET low until the power supplies are stable.
2. Bring RESET
3. The default state of the PDN bit is 1. Load the desired register settings while keeping the PDN bit set
to 11.
4. Load the required initialization settings listed in Section 4.11.
5. Apply MCLK at the appropriate frequency, as discussed in Section 4.6. SCLK may be applied or set to
master at any time; LRCK may only be applied or set to master while the PDN bit is set to 1.
6. Set the PDN bit to 0.
7. Bring RESET
prevent power glitch related issues.
4.10Recommended Power-Down Sequence
To minimize audible pops when turning off or placing the CODEC in standby:
1. Mute the DACs, PWM outputs and ADCs.
2. Disable soft ramp and zero cross volume transitions.
3. Set the PDN bit to 1.
4. Wait at least 100 µs.
The CODEC will be fully powered down after this 100 µs delay. Prior to the removal of the master clock
(MCLK), this delay of at least 100 µs must be implemented after step 3 to avoid premature disruption
of the CODEC’s power down sequence.
high.
low if the analog or digital supplies drop below the recommended operating condition to
CS42L52
A disruption in the CODEC’s power down sequence (i.e. removing the MCLK signal before this 100 µs
delay) has consequences on both the headphone and speaker amplifiers: The charge pump may stop
abruptly, causing the headphone amplifiers to drive the outputs up to the +VHP supply. Also, the last
state of each ‘+’ and ‘-’ PWM output terminal before the premature removal of MCLK could randomly
be held at either VP or AGND. When this event occurs, it is possible for each PWM terminal to output
opposing potentials, creating a DC source into the speaker voice coil.
The disruption of the CODEC’s power down sequence may also cause clicks and pops on the output
of the DACs as the modulator holds the last output level before the MCLK signal was removed.
5. Optionally, MCLK may be removed at this time.
6. To achieve the lowest operating quiescent current, bring RESET
reset to their default state.
7. Power Supply Removal (Option 1): Switch power supplies to a high impedance state.
8. Power Supply Removal (Option 2): To minimize pops when the power supplies are pulled to ground, a
discharge resistor must be added in parallel with the capacitor on the FILT+ pin. With a 1 M resistor
and a 2.2 µF capacitor on FILT+, FILT+ will ramp to ground in approximately 5 seconds. A 1 M resistor
on FILT+ reduces the full scale input/output voltage by approximately 0.25 dB.
After step 5, wait the required time for FILT+ to ramp to ground before pulling VA to ground.
4.11Required Initialization Settings
The current and thresholds required for various sections in the CODEC must be adjusted by implementing
the initialization settings shown below after power-up sequence step 3. All performance and power consumption measurements were taken with the following settings:
low. All control port registers will be
DS680F237
1. Write 0x99 to register 0x00.
4 5 6 7 24 25
SCL
CHIP ADDRESS (WRITE)MAP BYTEDATA
DATA +1
START
ACK
STOP
ACKACKACK
1 0 0 1 0 1 AD0 0
SDA
INCR 6 5 4 3 2 1 0 7 6 1 07 6 1 07 6 1 0
0 1 2 3 8 9 12 16 17 18 19 10 11 13 14 15 27 28
26
DATA +n
Figure 20. Control Port Timing, I²C Write
2. Write 0xBA to register 0x3E.
3. Write 0x80 to register 0x47.
4. Write 1 to bit 7 in register 0x32.
5. Write 00 to bit 7 in register 0x32.
6. Write 0x00 to register 0x00.
4.12Control Port Operation
The control port is used to access the registers, allowing the CODEC to be configured for the desired operational modes and formats. The operation of the control port may be completely asynchronou s 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.
The control port operates using an I²C interface with the CODEC acting as a slave device.
4.12.1I²C Control
SDA is a bidirectional data line. Data is clocked into and out of the device by the clock, SCL. The AD0 pin
sets the LSB of the chip address; ‘0’ when connec ted to DGND, ‘1’ w hen connected to VL. This pin may
be driven by a host controller or directly connected to VL or DGND. The AD0 pin state is sensed and the
LSB of the chip address is set upon the release of the RESET
3/1/13
CS42L52
signal (a low-to-high transition).
The signal timings for a read and write cycle are shown in Figure 20 and Figure 21. A St art condition is
defined as a falling transition of SDA while the clock is high. A Stop condition is defined as a rising transition of SDA while the clock is high. All other transitions of SDA occur while the clock is low. The first byte
sent to the CS42L52 after a Start condition consists of a 7-bit chip address fiel d and a R/W
bit (high for a
read, low for a write).
The upper 6 bits of the address field are fixed at 100101. To communicate with the CS42L52, the chip
address field, which is the first byte sent to the CS42L52, should m atch 100101 follo wed by th e setting of
the AD0 pin. The eighth bit of the address is the R/W
bit. If the operation is a write, the next byte is the
Memory Address Pointer (MAP), which selects the r egister to be read or written. If the opera tion 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 CS42L52 after each input byte is read and is input to the CS42L52 from
the microcontroller after each transmitted byte.
38DS680F2
3/1/13
SCL
CHIP ADDRESS (WRITE)
MAP BYTE
DATA
DATA +1
START
ACK
STOP
ACK
ACK
ACK
1 0 0 1 0 1 AD0 0
SDA
1 0 0 1 0 1 AD0 1
CHIP ADDRESS (READ)
START
INCR 6 5 4 3 2 1 0
7 07 07 0
NO
16 8 9 12 13 14 15 4 5 6 7 0 1 20 21 22 23 24
26 27 28
2 3 10 11 17 18 19 25
ACK
DATA + n
STOP
Figure 21. Control Port Timing, I²C Read
CS42L52
Since the read operation cannot set the MAP, an aborted write operation is used as a preamble. As shown
in Figure21, 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.
4.12.2Memory Address Pointer (MAP)
The MAP byte comes after the address byte and selects the register to be read or written. Refer to the
pseudo code above for implementation details.
4.12.2.1 Map Increment (INCR)
The device has MAP auto-increment capability enabled by the INCR bit (the MSB) of the MAP. If INCR is
set to 0, MAP will stay constant for successive I²C writes or reads. If INCR is set to 1, MAP will auto-increment after each byte is read or written, allowing block reads or writes of successive registers.
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3/1/13
CS42L52
5. REGISTER QUICK REFERENCE
Default values are shown below the bit names. Unless otherwise specified, all “Reserved” bits must maintain their
default values.
All registers are read/write except for the Chip I.D. and Revision Register and Interrupt Status Register, which are
read only. See the following bit definition tables for bit assignment information. The default stat e of ea ch b it afte r a
power-up sequence or reset is listed in each bit description. Unless otherwise specified, all “Reserved” b its must
maintain their default value.
6.1Chip I.D. and Revision Register (Address 01h) (Read Only)
Configures the power state of the ADC charge pump.
PDN_CHRGADC Charge Pump Status
0 Powered Up
1Powered Down
6.2.2Power Down PGAx
Configures the power state of PGA channel x.
PDN_PGAxPGA Status
0Powered Up (ONLY when the ADC or the analog passthru is used)
1Powered Down
Application“Analog In to Analog Out Passthrough” on page 31
Notes:
1. The CS42L52 employs a scheme for controlling the power to the PGA when PASSTHRU (“Passthrough
Analog” on page 52) is enabled. Refer to the referenced application for more information.
2. This bit should be used in conjunction with ADCxSEL and PGAxSEL bits to determine the analog
42DS680F2
input path. The PGAxSEL bits may be used to isolate the input signal(s) from the PGA outputs. When
the PGA is powered down, no input should be selected. Refer to “ADC Input Select” on page 48 and
“PGA Input Mapping” on page 49 for the required settings.
6.2.3Power Down ADCx
Configures the power state of ADC channel x.
PDN_ADCxADC Status
0Powered Up
1Powered Down
Application“Analog In to Analog Out Passthrough” on page 31
Notes:
1. The CS42L52 employs a scheme for controlling the power to the ADC when PASSTHRU
(“Passthrough Analog” on page 52) and PDN_OVRD (“Power Down ADC Override” on page 43) are
enabled. Refer to the referenced application.
Configures the auto-detect circuitry for detecting the speed mode of the CODEC when operating as a
slave.
AUTOAuto-detection of Speed Mode
0 Disabled
1Enabled
Application:“Serial Port Clocking” on page 33
Notes:
1. The SPEED[1:0] bits are ignored and speed is determined by the MCLK/LRCK ratio.
2. Certain sample and MCLK frequencies require setting the SPEED[1:0] bits, the 32k_GROUP bit
(“32kHz Sample Rate Group” on page 45) and/or the VIDEOCLK bit (“27 MHz Video Clock” on
page 45) and RATIO[1:0] bits (“Internal MCLK/LRCK Ratio” on page 45). Low sample rates may also
affect dynamic range performance in the typical audio band. Refer to the referenced application for
more information.
44DS680F2
6.5.2Speed Mode
Configures the speed mode of the CODEC in slave mode and sets the appropriate MCLK divide ratio for
LRCK and SCLK in master mode.
2. Certain sample and MCLK frequencies require setting the SPEED[1:0] bits, the 32k_GROUP bit
(“32kHz Sample Rate Group” on page 45) and/or the VIDEOCLK bit (“27 MHz Video Clock” on
page 45) and RATIO[1:0] bits (“Internal MCLK/LRCK Ratio” on page 45). Low sample rates may also
affect dynamic range performance in the typical audio band. Refer to the referenced application for
more information.
3. These bits are ignored when the AUTO bit (“Auto-Detect” on page 44) is enabled.
3/1/13
CS42L52
Slave ModeMaster Mode
Serial Port SpeedMCLK/LRCK RatioSCLK/LRCK Ratio
1024
bit in “Master/Slave Mode” on page 46.
64
6.5.332kHz Sample Rate Group
Specifies whether or not the input/output sample rate is 8 kHz, 16 kHz or 32 kHz.
32kGROUP8 kHz, 16 kHz or 32 kHz sample rate?
0No
1Yes
Application:“Serial Port Clocking” on page 33
6.5.427 MHz Video Clock
Specifies whether or not the external MCLK frequency is 27 MHz
VIDEOCLK27 MHz MCLK?
0No
1Yes
Application:“Serial Port Clocking” on page 33
6.5.5Internal MCLK/LRCK Ratio
Configures the internal MCLK/LRCK ratio.
RATIO[1:0]Internal MCLK Cycles per LRCKSCLK/LRCK Ratio in Master Mode
0012864
0112562
1013266
1113668
Application:“Serial Port Clocking” on page 33
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3/1/13
CS42L52
6.5.6MCLK Divide By 2
Divides the input MCLK by 2 prior to all internal circuitry.
MCLKDIV2MCLK signal into CODEC
0No divide
1Divided by 2
Application:“Serial Port Clocking” on page 33
Note:In slave mode, this bit is ignored when the AUTO bit (“Auto-Detect” on page 44) is disabled.
6.6Interface Control 1 (Address 06h)
76543210
M/S
6.6.1Master/Slave Mode
INV_SCLKADCDIFDSPDACDIF1DACDIF0AWL1AWL0
Configures the serial port I/O clocking.
M/SSerial Port Clocks
0Slave (input ONLY)
1Master (output ONLY)
6.6.2SCLK Polarity
Configures the polarity of the SCLK signal.
INV_SCLKSCLK Polarity
0Not Inverted
1Inverted
6.6.3ADC Interface Format
Configures the digital interface format for data on SDOUT.
ADCDIFADC Interface Format
0Left Justified
1I²S
Application:“Digital Interface Formats” on page 35
6.6.4DSP Mode
Configures a data-packed interface format for both the ADC and DAC.
DSPDSP Mode
0Disabled
1Enabled
Application:“DSP Mode” on page 35
Notes:
1. Select the audio word length using the AWL[1:0] bits (“Audio Word Length” on page 47).
2. The interface format for both the ADC and the DAC must be set to “Left-Justified” when DSP Mode
is enabled.
46DS680F2
6.6.5DAC Interface Format
Configures the digital interface format for data on SDIN.
DACDIF[1:0]DAC Interface Format
00Left Justified, up to 24-bit data
01I²S, up to 24-bit data
10Right Justified
11Reserved
Application:“Digital Interface Formats” on page 35
Note:Select the audio word length for Right Justified using the AWL[1:0] bits (“Audio Word Length” on
page 47).
6.6.6Audio Word Length
Configures the audio sample word length used for the data into SDIN and out of SDOUT.
AWL[1:0]
0032-bit data24-bit data
0124-bit data20-bit data
1020-bit data18-bit data
1116-bit data16-bit data
Application:“DSP Mode” on page 35
Audio Word Length
DSP ModeRight Justified (DAC ONLY)
3/1/13
CS42L52
Note:When the internal MCLK/LRCK ratio is set to 125 in master mode, the 32-bit data width option
Configures the SCLK signal source for master mode.
SCLK=MCLKOutput SCLK
0Re-timed signal, synchronously derived from MCLK
1Non-retimed, MCLK signal
Note:This bit is only valid for MCLK = 12.0000 MHz.
6.7.2SDOUT to SDIN Digital Loopback
Configures an internal loops the signal on the SDOUT pin to SDIN.
DIGLOOPInternal Loopback
0Disabled; SDOUT internally disconnected from SDIN
1Enabled; SDOUT internally connected to SDIN
DS680F247
6.7.3Tri-State Serial Port Interface
Determines the state of the serial port drivers.
3ST_SP
0Serial Port clocks are inputs and SDOUT is outputSerial Port clocks and SDOUT are outputs
1Serial Port clocks are inputs and SDOUT is HI-ZSerial Port clocks and SDOUT are HI-Z
Serial Port Status
Slave ModeMaster Mode
Notes:
1. Slave/Master Mode is determined by the M/S
2. When the serial port is tri-stated in master mode, the ADC and DAC serial ports are clocked internally.
6.7.4Speaker/Headphone Switch Invert
Determines the control signal polarity of the SPK/HP_SW pin.
INV_SWCHSPK/HP_SW pin 6 Control
0Not inverted
1Inverted
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bit in “Master/Slave Mode” on page 46.
6.7.5MIC Bias Level
Sets the output voltage level on the MICBIAS output pin.
BIASLVL[2:0]Output Bias Level
0000.5 x VA
0010.6 x VA
0100.7 x VA
0110.8 x VA
1000.83 x VA
1010.91 x VA
110Reserved
111Reserved
6.8Input x Select: ADCA and PGAA (Address 08h), ADCB and PGAB (Address 09h)
Selects the specified analog input signal into ADCx.
ADCxSEL[2:0]Selected Input to ADCx
000AIN1x
001AIN2x
010AIN3x
011AIN4x
100PGAx - Use PGAxSEL bits (“PGA Input Mapping” on page 49) to select input channels
101Reserved
110Reserved
111Reserved
Application:“Analog Inputs” on page 25
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6.8.2PGA Input Mapping
Selects one or sums/mixes the analog input signal into the PGA. Each bit of the PGAx_SEL[5:1] word
corresponds to individual channels (i.e. PGAx_SEL1 selects AIN1x, PGAx_SEL2 selects AIN2x, etc.).
Configures the internal high-pass filter after ADCx.
HPFxHigh Pass Filter Status
0Disabled
1Enabled
6.9.2ADCx High-Pass Filter Freeze
Configures the high pass filter’s digital DC subtraction and/or calibration after ADCx.
HPFRZxHigh Pass Filter Digital Subtraction
0Continuous DC Subtraction
1Frozen DC Subtraction
6.9.3Ch. x Analog Soft Ramp
Configures an incremental volume ramp from the current level to the new level at the specified rate.
ANLGSFTxVolume ChangesAffected Analog Volume Controls
0Do not occur with a soft ramp
1Occur with a soft ramp
Ramp Rate:1/2 dB every 16 LRCK cycles
6.9.4Ch. x Analog Zero Cross
Configures when the signal level changes occur for the analog volume controls.
ANLGZCxVolume ChangesAffected Analog Volume Controls
0
1Occur on a zero crossing
Do not occur on a zero crossing
MICxGAIN[4:0] (“MICx Gain” on page 55), PGAxVOL[5:0] (“PGAx Volume”
on page 56), and PASSxVOL[7:0] (“Passthrough x Volume” on page 57)
MICxGAIN[4:0] (“MICx Gain” on page 55), PGAxVOL[5:0] (“PGAx Volume”
on page 56), and PASSxVOL[7:0] (“Passthrough x Volume” on page 57)
Note:If the signal does not encounter a zero crossing, the requested volume change will occur after a
timeout period of 1024 sample periods (approximately 10.7 ms at 48 kHz sample rate).
ADCxMUTE (“ADC Mute” on page 51),
ALC and Limiter Attack/Release (page 66 to page 68)
MICxGAIN[4:0] (“MICx Gain” on page 55),
PGAxVOL[5:0] (“PGAx Volume” on page 56),
PASSxVOL[7:0] (“Passthrough x Volume” on page 57)
Serial Output Signal
Left ChannelRight Channel
50DS680F2
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6.11.4Invert ADC Signal Polarity
Configures the polarity of the ADC signal.
INV_ADCxADC Signal Polarity
0Not Inverted
1Inverted
6.11.5ADC Mute
Configures a digital mute on ADC channel x.
ADCxMUTEADC Mute
0Disabled
1Enabled
Note:When the ADCxMUTE bit is enabled, the PGA will automatically apply 6 dB of attenuation.
Configures an analog passthrough from the PGA inputs to the headphone/line outputs.
PASSTHRUxAnalog In Routed to HP/Line Output
0Disabled
1Enabled
Notes:
1. The Passthrough volume control is realized using a combination of the PGA volume control settings
(“PGAx Volume” on page 56) and the headphone amplifier volume control settings (hidden). When
passthrough is enabled and the PGA to ADC path is selected, the signal seen by the ADC will change
depending on the passthrough volume setting.
6.13.2Passthrough Mute
Configures an analog mute on the channel x analog in to analog out passthrough.
PASSxMUTEPassthrough Mute
0Disabled
1Enabled
6.13.3Freeze Registers
Configures a hold on all register settings.
FREEZEControl Port Status
0Register changes take effect immediately
1
Modifications may be made to all control port registers without the changes taking effect until after the
FREEZE is disabled.
Notes:
1. Use this bit only to synchronize run-time controls, such as volume and mute, during normal operation .
52DS680F2
Using this bit before the relevant circuitry begins normal operation could cause the change to take
effect immediately, ignoring the FREEZE bit.
6.13.4HP/Speaker De-emphasis
Configures a 15s/50s digital de-emphasis filter response on the headphone/line and speaker outputs .
DEEMPHASISControl Port Status
0Disabled
1Enabled
6.13.5Digital Soft Ramp
Configures an incremental volume ramp from the current level to the new level at the specified rate.
DIGSFTVolume ChangesAffected Digital Volume Controls
0Do not occur with a soft rampMSTxMUTE (“Master Playback Mute” on page 52),
1Occur with a soft ramp
Ramp Rate:1/8 dB every LRCK cycle
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HPxMUTE, SPKxMUTE (“Playback Control 2 (Address 0Fh)” on page 54),
ADCxVOL[7:0] (“ADCx Volume” on page 57),
AMIXxMUTE, AMIXxVOL[7:0] (“ADC Mixer Channel x Volume” on page 58),
PMIXxMUTE, PMIXxVOL[7:0] (“PCM Mixer Channel x Volume” on page 58),
MSTxVOL[7:0] (“Master Volume Control” on page 63),
HPxVOL[7:0] (“Headphone Volume Control” on page 63),
SPKxVOL[7:0] (“Speaker Volume Control” on page 64),
ALC and Limiter Attack/Release (page 66 to page 68)
Beep Volume (“Beep Volume” on page 61)
Note:When the DIGSFT bit is enabled, the Master Volume (MSTxVOL[7:0]) transitions are guaranteed
to occur with a soft ramp only when
6.13.6Digital Zero Cross
Configures when the signal level changes occur for the digital volume controls.
DIGZCVolume ChangesAffected Digital Volume Controls
0
1Occur on a zero crossing
Notes:
1. If the signal does not encounter a zero crossing, the requested volume change will occur after a
timeout period between 1024 and 2048 sa mple pe riods (21.3 ms to 42.7 ms at 4 8 kHz sample r ate).
2. The zero cross function is independently monitored and implemented for each channel.
3. The DIS_LIMSFT bit (“Limiter Soft Ramp Disable” on page 65) is ignored when zero cross is enabled.
4. Wh en the DIGZ C bit is en abled , th e Ma st er Volume (MSTxVOL[7:0]) transitions are guaranteed to
occur on a zero cross only if bits 7 and 6 in register 29h are set to '00'b
Do not occur on a zero crossing
bits 7 and 6 in register 29h are set to ‘00’b.
MSTxMUTE (“Master Playback Mute” on page 52),
AMIXxMUTE, AMIXxVOL[7:0] (“ADC Mixer Channel x Volume” on page 58),
PMIXxMUTE, PMIXxVOL[7:0] (“PCM Mixer Channel x Volume” on page 58),
MSTxVOL[7:0] (“Master Volume Control” on page 63),
ALC and Limiter Attack/Release (page 66 to page 68)
Beep Volume (“Beep Volume” on page 61)
Configures an override of the analog soft ram p se ttin g.
ALCxSRDISALC Soft Ramp Disable
0OFF; ALC Attack Rate is dictated by the ANLGSFT (“Ch. x Analog Soft Ramp” on page 49) setting
1ON; ALC volume changes take effect in one step, regardless of the ANLGSFT setting.
Application:“Automatic Level Control (ALC)” on page 26
DS680F255
6.16.2ALCx Zero Cross Disable
Configures an override of the analog zero cross setting.
ALCxZCDISALC Zero Cross Disable
0OFF; ALC Attack Rate is dictated by the ANLGZC (“Ch. x Analog Zero Cross” on page 49) setting
1ON; ALC volume changes take effect at any time, regardless of the ANLGZC setting.
Application:“Automatic Level Control (ALC)” on page 26
6.16.3PGAx Volume
Sets the volume/gain of the Programmable Gain Amplifier (PGA).
PGAxVOL[5:0]Volume
01 111112 dB
......
01 100012 dB
......
00 0001+0.5 dB
00 00000 dB
11 1111-0.5 dB
......
10 1000-6.0 dB
......
10 0000-6.0 dB
Step Size:0.5 dB
3/1/13
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Note:The PGAxVOL bits are ignored when the PASSTHRUx bit (“Passthrough Analog” on page 52) is
enabled.
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6.17Passthrough x Volume: PASSAVOL (Address 14h) and PASSBVOL (Address 15h)
1. This setting must not change when BEEP is enabled.
2. Beep frequency will scale directly with sample rate, Fs, but is fixed at the nominal Fs within each
speed mode.
DS680F259
6.21.2Beep On Time
Sets the on duration of the beep signal.
ONTIME[3:0]On Time (Fs = 12, 24, 48 or 96 kHz)
0000~86 ms
0001~430 ms
0010~780 ms
0011~1.20 s
0100~1.50 s
0101~1.80 s
0110~2.20 s
0111~2.50 s
1000~2.80 s
1001~3.20 s
1010~3.50 s
1011~3.80 s
1100~4.20 s
1101~4.50 s
1110~4.80 s
1111~5.20 s
Application:“Beep Generator” on page 29
3/1/13
CS42L52
Notes:
1. This setting must not change when BEEP is enabled.
2. Beep on time will scale inversely with sample rate, Fs, but is fixed at the nominal Fs within each speed
mode.
Configures a beep mixed with the HP/Line and SPK output.
BEEP[1:0] Beep Occurrence
00Off
01Single
10Multiple
11Continuous
Application:“Beep Generator” on page 29
Notes:
1. When used in analog pass-through mode, the output alternates between the signal from the PGA and
the beep signal. The beep signal does not mix with the analog signal from the PGA.
2. Re-engaging the beep before it has completed its initial cycle will cause the beep signal to remain ON
for the maximum ONTIME duration.
6.23.2Beep Mix Disable
Configures how the beep mixes with the serial data input.
BEEPMIXDISBeep Output to HP/Line and Speaker
0Mix Enabled; The beep signal mixes with the digital signal from the serial data input.
1
Application:“Beep Generator” on page 29
Mix Disabled; The output alternates between the signal from the serial data input and the beep signal. The
beep signal does not mix with the digital signal from the serial data input.
Note:This setting must not change when BEEP is enabled.
DS680F261
6.23.3Treble Corner Frequency
Sets the corner frequency (-3 dB point) for the treble shelving filter.
TREBCF[1:0]Treble Corner Frequency Setting
005 kHz
017 kHz
1010 kHz
1115 kHz
6.23.4Bass Corner Frequency
Sets the corner frequency (-3 dB point) for the bass shelving filter.
BASSCF[1:0]Bass Corner Frequency Setting
0050 Hz
01100 Hz
10200 Hz
11250 Hz
6.23.5Tone Control Enable
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CS42L52
Configures the treble and bass activation.
TCENBass and Treble Control
0Disabled
1Enabled
Application:“Beep Generator” on page 29
6.24Tone Control (Address 1Fh)
76543210
TREB3TREB2TREB1TREB0BASS3BASS2BASS1BASS0
6.24.1Treble Gain
Sets the gain of the treble shelving filter.
TREB[3:0]Gain Setting
0000+12.0 dB
······
0111+1.5 dB
10000 dB
1001-1.5 dB
······
1111-10.5 dB
Step Size:1.5 dB
62DS680F2
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CS42L52
6.24.2Bass Gain
Sets the gain of the bass shelving filter.
TREB[3:0]Gain Setting
0000+12.0 dB
······
0111+1.5 dB
10000 dB
1001-1.5 dB
······
1111-10.5 dB
Step Size:1.5 dB
6.25Master Volume Control: MSTA (Address 20h) and MSTB (Address 21h)
Configures a mix/swap of the PCM Mix to the headphone/line or speaker outputs.
PCMxSWP[1:0]PCM Mix to HP/LINEOUTAPCM Mix to HP/LINEOUTB
00LeftRight
01
10
11RightLeft
(Left + Right)/2(Left + Right)/2
6.28.2ADC Mix Channel Swap
Configures a mix/swap of the ADC Mix to the headphone/line or speaker outputs.
ADCxSWP[1:0]ADC Mix to HP/LINEOUTA ChannelADC Mix to HP/LINEOUTB Channel
00LeftRight
01
10
11RightLeft
(Left + Right)/2(Left + Right)/2
64DS680F2
3/1/13
CS42L52
6.29Limiter Control 1, Min/Max Thresholds (Address 27h)
76543210
LMAX2LMAX1LMAX0CUSH2CUSH1CUSH0LIMSRDISLIMZCDIS
6.29.1Limiter Maximum Threshold
Sets the maximum level, below full scale, at which to limit and attenuate the output signal at the attack
rate (LIMARATE - “Limiter Release Rate” on page 66).
LMAX[2:0]Threshold Setting
0000 dB
001-3 dB
010-6 dB
011-9 dB
100-12 dB
101-18 dB
110-24 dB
111-30 dB
Application:“Limiter” on page 30
Note:Bass, Treble, and digital gain settings that boost the signal beyond the maximu m thresho ld may
trigger an attack.
6.29.2Limiter Cushion Threshold
Sets the minimum level at which to disengage the Limiter’s attenuation at the release rate (LIMRRATE -
“Limiter Release Rate” on page 66) until levels lie between the LMAX and CUSH thresholds.
CUSH[2:0]Threshold Setting
0000 dB
001-3 dB
010-6 dB
011-9 dB
100-12 dB
101-18 dB
110-24 dB
111-30 dB
Application:“Limiter” on page 30
Note:This setting is usually set slightly below the LMAX th re sh old .
6.29.3Limiter Soft Ramp Disable
Configures an override of the digital soft ramp settin g.
LIMSRDISLimiter Soft Ramp Disable
0OFF; Limiter Attack Rate is dictated by the DIGSFT (“Digital Soft Ramp” on page 53) setting
1ON; Limiter volume changes take effect in one step, regardless of the DIGSFT setting.
Application:“Limiter” on page 30
Note:This bit is ignored when the DIGZC (“Digital Zero Cross” on page 53) is enabled.
DS680F265
3/1/13
CS42L52
6.29.4Limiter Zero Cross Disable
Configures an override of the digital zero-cross setting.
LIMZCDISLimiter Zero Cross Disable
0OFF; Limiter Attack Rate is dictated by the DIGZC (“Digital Zero Cross” on page 53) setting
1ON; Limiter volume changes take effect in one step, regardless of the DIGZC setting.
Sets how channels are attenuated when the limiter is enabled.
LIMIT_ALLLimiter action:
Apply the necessary attenuation on a specific channel only when the signal amplitude on
0
1
Application:“Limiter” on page 30
nel rises above LMAX.
Remove attenuation on a specific channel only when the signal amplitude on
CUSH.
Apply the necessary attenuation on BOTH channels when the signal amplitude on any ONE channel rises
above LMAX.
Remove attenuation on BOTH channels only when the signal amplitude on BOTH channels fall below CUSH.
6.30.3Limiter Release Rate
Sets the rate at which the limiter releases the digita l attenuation fro m levels below the CUSH[2:0] thre shold (“Limiter Cushion Threshold” on page 65) and returns t he analog output level t o the MSTxVOL[7:0]
(“Master Volume Control” on page 63) setting.
LIMRRATE[5:0]Release Time
00 0000Fastest Release
······
11 1111Slowest Release
Application:“Limiter” on page 30
Note:The limiter release rate is user-selectable but is also a function of the sampling frequency, Fs,
and the DIGSFT (“Digital Soft Ramp” on page 53) and DIGZC (“Digital Zero Cross” on page 53) s et tin g.
Sets the rate at which the limiter applies digital attenuation from levels above the MAX[2:0] threshold
(“Limiter Maximum Threshold” on page 65).
LIMARATE[5:0]Attack Time
00 0000Fastest Attack
······
11 1111Slowest Attack
Application:“Limiter” on page 30
Note:The limiter attack rate is user-selectable but is also a function of the sampling frequency, Fs, and
the DIGSFT (“Digital Soft Ramp” on page 53) and DIGZC (“Digital Zero Cross” on page 53) setting unless
the respective disable bit (“Limiter Soft Ramp Disable” on page 65 or “Limiter Zero Cross Disable” on
Application:“Automatic Level Control (ALC)” on page 26
Notes:
1. When the ALC is enabled, the digital volume and PGA volume is automatically controlled and should
not be adjusted manually.
2. The ALC should only be configured while the power down bit is enabled.
3. The ALC is not available in passthrough mode.
6.32.2ALC Attack Rate
Sets the rate at which the ALC applies analog and/or digital attenuation from levels above the AMAX[2:0]
threshold (“ALC Maximum Threshold” on page 68).
LIMARATE[5:0]Attack Time
00 0000Fastest Attack
······
11 1111Slowest Attack
Application:“Automatic Level Control (ALC)” on page 26
Note:The ALC attack rate is user-se lecta ble bu t is al so a function of the sampling frequency, Fs, and
the ANLGSFTx (“Ch. x Analog Soft Ramp” on page 49) and ANLGZCx (“Ch. x Analog Zero Cross” on
page 49) setting unless the respective disable bit (“ALCx Soft Ramp Disable” on page 55 or “ALCx Zero
Cross Disable” on page 56) is enabled.
Sets the rate at which the ALC releases the analog and/or digital attenuation from levels below the
MIN[2:0] threshold (“ALC Minimum Threshold” on page 69) and returns the signal level to the PGAxVOL[5:0] (“PGAx Volume” on page 56) and ADCxVOL[7:0] (“ADCx Volume” on page 57) setting.
ALCRRATE[5:0]Release Time
00 0000Fastest Release
······
11 1111Slowest Release
Application:“Automatic Level Control (ALC)” on page 26
Notes:
1. The ALC release rate is user-selectable but is also a function of the sampling frequency, Fs, and the
ANLGSFTx (“Ch. x Analog Soft Ramp” on page 49) and ANLGZCx (“Ch. x Analog Zero Cross” on
page 49) setting.
2. The Release Rate setting must always be slower than the Attack Rate.
Sets the maximum level, below full scale, at which to limit and attenuate the input signal at the atta ck rate
(ALCARATE - “ALC Attack Rate” on page 67).
MAX[2:0]Threshold Setting
0000 dB
001-3 dB
010-6 dB
011-9 dB
100-12 dB
101-18 dB
110-24 dB
111-30 dB
Application:“Automatic Level Control (ALC)” on page 26
68DS680F2
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CS42L52
6.34.2ALC Minimum Threshold
Sets the minimum level at which to disengage the ALC’s attenuation or amplify the input signal at the release rate (ALCRRATE - “ALC Release Rate” on page 68) until levels lie between the ALCMAX and ALCMIN thresholds.
ALCMIN[2:0]Threshold Setting
0000 dB
001-3 dB
010-6 dB
011-9 dB
100-12 dB
101-18 dB
110-24 dB
111-30 dB
Application:“Automatic Level Control (ALC)” on page 26
Notes:
1. This setting is usually set slightly below the ALCMAX threshold.
Sets which channels are attenuated when clipping on any single channel occurs.
NGALLNoise Gate triggered by:
0
1
Application:“Noise Gate” on page 27
Individual channel; Any channel that falls below the threshold setting triggers the noise gate attenuation for
both channels.
Both channels A and B; Both channels must fall below the threshold setting for the noise gate attenuation to
take effect.
6.35.2Noise Gate Enable
Configures the noise gate.
NGNoise Gate Status
0Disabled
1Enabled
Application:“Noise Gate” on page 27
DS680F269
6.35.3Noise Gate Threshold and Boost
THRESH sets the threshold level of the noise gate. Input signals below the threshold level will be attenuated to -96 dB. NG_BOOST configures a +30 dB boost to the threshold settings.
000-64 dB-34 dB
001-67 dB-37 dB
010-70 dB-40 dB
011-73 dB-43 dB
100-76 dB-46 dB
101-82 dB-52 dB
110Reserved-58 dB
111Reserved-64 dB
Application:“Noise Gate” on page 27
6.35.4Noise Gate Delay Timing
Sets the delay time before the noise gate attacks.
NGDELAY[1:0]Delay Setting
0050 ms
01100 ms
10150 ms
11200 ms
Application:“Noise Gate” on page 27
3/1/13
CS42L52
Note:The Noise Gate attack rate is a function of the sampling frequency, Fs, and the ANLGSFTx (“Ch.
x Analog Soft Ramp” on page 49) and ANLGZCx (“Ch. x Analog Zero Cross” on page 49) setting unless
the respective disable bit (“ALCx Soft Ramp Disable” on page 55 or “ALCx Zero Cross Disable” on
page 56) is enabled.
6.36Status (Address 2Eh) (Read Only)
For all bits in this register, a “1” means the associated error condition has occurred at least on ce sinc e the
register was last read. A”0” means the associated error condition has NOT occurred since the last reading
of the register. Reading the register resets all bits to 0.
Configures automatic adjustment of the speaker volume when VP deviates from VPREF[3:0].
BATTCMPAutomatic Battery Compensation
0Disabled
1Enabled
Application:“Maintaining a Desired Output Level” on page 33
6.37.2VP Monitor
Configures the internal ADC that monitors the VP voltage level.
VPMONITORVP ADC Status
0Disabled
1Enabled
Notes:
1. The internal ADC that monitors the VP supply is enabled automatically when BATTCMP is enabled,
regardless of the VPMONITOR setting. Conversely, when BATTCMP is disabled, the ADC may be
enabled by enabling VPMONITOR; this provides a convenient battery monitor without enabling
battery compensation.
2. When enabled, VPMONITOR remains enabled regardless of the PDN bit setting.
DS680F271
6.37.3VP Reference
Sets the desired VP reference used for battery compensation.
VPREF[3:0]Desired VP used to calculate the required attenuation on the speaker output:
00001.5 V
00012.0 V
00102.5 V
00113.0 V
01003.5 V
01014.0 V
01104.5 V
01115.0 V
10001.5 V
10012.0 V
10102.5 V
10113.0 V
11003.5 V
11014.0 V
11104.5 V
11115.0 V
Application:“VP Battery Compensation” on page 33
3/1/13
CS42L52
(for VA = 1.8 V)
(for VA = 2.5 V)
6.38VP Battery Level (Address 30h) (Read Only)
76543210
VPLVL7VPLVL6VPLVL5VPLVL4VPLVL3VPLVL2VPLVL1VPLVL0
6.38.1VP Voltage Level (Read Only)
Indicates the unsigned VP voltage level.
VPLVL[7:0]VP Voltage
...
0101 11103.0 V (for VA = 2.0 V); apply formula using actual VA voltage to calculate VP voltage.
...
0111 00103.7 V (for VA = 2.0 V); apply formula using actual VA voltage to calculate VP voltage.
...
Formula:VP Voltage = (Binary representation of VPLVL[7:0]) * VA / 63.3
As with any high-resolution converter, the CS42L52 req uires ca reful atte ntion to power su pply and gr ounding arrangements if its potential performance is to be realized. Figu re 1 o n page 11 shows the recomme nd-
ed power arrangements, with VA and VHP connected to clean supplies VD, which powers the digital
circuitry, may be run from the system logic supply. Alternatively, VD ma y be powered from the analog supply
via a ferrite bead. In this case, no additional devices should be powered from VD.
Extensive use of power and ground planes, ground plane fill in unused areas and surface mount decoupling
capacitors are recommended. Decoupling capacitors should be as close to the pins of the CS42L52 as possible. The low value ceramic capacitor should be closest to the pin and should be mounted on the same
side of the board as the CS42L52 to minimize inductance effects.
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 cap acitors, particularly the 0.1 µF, must be positioned to minimize the electrical path from FILT+ and AGND. The CDB42L52 evaluation board demonstrates the optimum layout and power supply arrangements.
9.2QFN Thermal Pad
3/1/13
CS42L52
The CS42L52 is available in a compact QFN package. The underside 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 CS42L52 evaluation board demonstrates the op timum thermal pad and via configuration.
Figure 32. DAC Transition BandFigure 33. DAC Transition Band (Detail)
3/1/13
CS42L52
78DS680F2
11.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 band width 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 ar e below the n oise level an d do n ot affect the m easu rement. 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
band width (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 k Hz. Units in decibels.
Interchannel Isolation
3/1/13
CS42L52
A measure of crosstalk between the left and right channel pairs. Mea sured for ea ch channel at the co nverter's output with no signal to the input under test and a full-scale signa l applied to the other channel. Units in
decibels.
Interchannel Gain Mismatch
The gain difference between left and right channel pairs. Units in decibels.
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.
1. Dimensioning and tolerance per ASME Y 14.5M-1995.
2. Dimensioning lead width applies to the plated terminal and is measu red between 0.20 mm and 0.25 mm
from the terminal tip.
THERMAL CHARACTERISTICS
Junction to Ambient Thermal Impedance2 Layer Board
ParameterSymbolMinTypMaxUnits
4 Layer Board
JA
-
-
44
19
-
°C/Watt
-
80DS680F2
3/1/13
CS42L52
13.ORDERING INFORMATION
ProductDescriptionPackage Pb-FreeGradeTemp Range ContainerOrder #
CS42L52
CDB42L52
CRD42L52
Low-Power, Stereo
CODEC with Headphone
and Speaker Amps
CS42L52 Evaluation
Board
CS42L52 Reference
Design
40L-QFNYesCommercial -40 to +85° C
-No---CDB42L52
-No---CRD42L52
RailCS42L52-CNZ
Tape and Reel CS42L52-CNZR
14.REFERENCES
1. Philips Semiconductor, The I²C-Bus Specification: Version 2.1, January 2000.
http://www.semiconductors.philips.com
.
15.REVISION HISTORY
RevisionChanges
F1Initial draft
Removed the Automotive specification.
Added AD0 characteristics to “I/O Pin Characteristics” on page 10.
Added AD0 pin to Figure 1. Typical Connection Diagram on page 11.
Updated Note 6 on page 15.
specification for VL = 1.8 V in “Digital Interface Specifications and Characteristics” on page 22.
IH
F2
Updated the V
Updated “PWM Outputs” on page 32 to exclude support of a 384x MCLK to LR CK ratio.
Added register commands for entering and exiting DSP 16-bit, 8 kHz Fs, SCLK = MCLK, master mode in section
“DSP Mode” on page 35.
Updated Section 4.9 “Recommended Power-up Sequence” on page 37.
Updated Section 4.10 “Recommended Power-Down Sequence” on page 37.
Added a description of the AD0 pin to “I²C Control” on page 38.
Added AD0 detail to Figure 20. Control Port Timing, I²C Write on page 38 and Figure 21. Control Port Timing, I²C
Read on page 39.
Updated the first paragraph in “Register Quick Reference” on page 40 to allow for data sheet-specified controlwrites to reserved registers.
Removed I²C address heading row from “Register Quick Reference” on page 40.
Updated notes in “Auto-Detect” on page 44.
Updated table in “Speed Mode” on page 45.
Added note 1 in “Freeze Registers” on page 52.
Updated notes in “Digital Soft Ramp” on page 53.
Added note 4 to “Digital Zero Cross” on page 53.
Added notes 1 and 2 for ALC configuration in “ALC Enable and Attack Rate (Address 2Ah)” on page 67.
Corrected the E2 scale in the package drawing in “Package Dimensions” on page 80.
DS680F281
3/1/13
Contacting Cirrus Logic Support
For all product questions and inquiries, contact a Cirrus Logic Sales Representative.
To find one nearest you, go to www.cirrus.com.
IMPORTANT NOTICE
Cirrus Logic, Inc. and its subsidiaries (“Cirrus”) believe that the information contained in this document is accurate and reliable. However, the information is subject
to change without noti ce and is provided “AS IS” wi thout war ranty of any kind (express or impli ed). Cust omers ar e advise d to ob tain the latest version of relevant
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supplied at the time of order acknowled gment, includin g those pertaining to wa rranty, indemnificatio n, and limitation of liability. No responsibility is assumed by Cirrus
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CERTAIN APPLICATIONS USING SEMICONDUCTOR PRODUCTS MAY INVOLVE POTENTIAL RISKS OF DEATH, PERSONAL INJURY, OR SEVERE PROPERTY OR ENVIRONMENTAL DAMAGE (“CRITICAL APPLICATIONS”). CIRRUS PRODUCTS ARE NOT DESIGNED, AUTHORIZED OR WARRANTED FOR USE
IN PRODUCTS SURGICALLY IMPLANTED INTO THE BODY, AUTOMOTIVE SAFETY OR SECURITY DEVICES, LIFE SUPPORT PRODUCTS OR OTHER CRITICAL APPLICATIONS. INCLUSION OF CIRRUS PRODUCTS IN SUCH APPLICATIONS IS UNDERSTOOD TO BE FULLY AT THE CUSTOMER’S RISK AND CIRRUS DISCLAIMS AND MAKES NO WARRANTY, EXPRESS, STATUTORY OR IMPLIED, INCLUDING THE IMPLIED WARRANTIES OF MERCHANTABILITY AND
FITNESS FOR PARTICULAR PURPOSE, WITH REGARD TO ANY CIRRUS PRODUCT THAT IS USED IN SUCH A MANNER. IF THE CUSTOMER OR CUSTOMER’S CUSTOMER USES OR PERMITS THE USE OF CIRRUS PRODUCTS IN CRITICAL APPLICATIONS, CUSTOMER AGREES, BY SUCH USE, TO FULLY
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or service marks of their respective owners.
I²C is a trademark of Philips Semiconductor.
CS42L52
82DS680F2
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