98 dB Dynamic Range (A-wtd)
-86 dB THD+N
Headphone Amplifier - GND Centered
–On-Chip Charge Pump Provides -VA_HP
–No DC-Blocking Capacitor Required
–46 mW Power Into Stereo 16 Ω @ 1.8 V
–88 mW Power Into Stereo 16 Ω @ 2.5 V
–-75 dB THD+N
Digital Signal Processing Engine
–Bass & Treble Tone Control, De-Emphasis
–PCM + ADC Mix w/Independent Vol Control
–Master Digital Volume Control
–Soft Ramp & Zero Cross Transitions
Beep Generator
–Tone Selections Across Two Octaves
–Separate Volume Control
–Programmable On & Off Time Intervals
–Continuous, Periodic or One-Shot Beep
Selections
Programmable Peak-Detect and Limiter
Pop and Click Suppression
ANALOG to DIGITAL FEATURES
98 dB Dynamic Range (A-wtd)
-88 dB THD+N
Analog Gain Controls
–+32 dB or +16 dB MIC Pre-Amplifiers
–Analog Programmable Gain Amplifier
(PGA)
+20 dB Digital Boost
Programmable Automatic Level Control (ALC)
–Noise Gate for Noise Suppression
–Programmable Threshold and
Attack/Release Rates
Independent Channel Control
Digital Volume Control
High-Pass Filter Disable for DC Measurements
Stereo 3:1 Analog Input MUX
Dual MIC Inputs
–Programmable, Low Noise MIC Bias Levels
–Differential MIC Mix for Common Mode
–Analog 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)
APPLICATIONS
HDD & Flash-Based Portable Audio Players
MD Players/Recorders
PDAs
Personal Media Players
Portable Game Consoles
Digital Voice Recorders
Digital Camcorders
Digital Cameras
Smart Phones
®
& SPI™ Control)
GENERAL DESCRIPTION
The CS42L51 is a highly integrated, 24-bit, 96 kHz, low
power stereo CODEC. Based on multi-bit, delta-sigma
modulation, it allows infinite sample rate adjustment between 4 kHz and 96 kHz. Both the ADC and DAC offer
many features suitable for low power, portable system
applications.
The ADC input path allows independent channel control
of a number of features. An input multiplexer selects between line-level or microphone level inputs for each
channel. The microphone input path includes a selectable programmable-gain pre-a mplifier 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 attenuator with soft ramp
transitions. A programmable ALC and Noise Gate monitor the input signals and adjust the volume levels
appropriately.
The DAC output path includes a digital signal pro cessing engine. Tone Control provides bass and treble
adjustment of four selectable corner frequencies. The
Mixer allows independent volume control for both the
ADC mix and the PCM mix, as well as a master digital
volume control for the analog output. All volume level
changes may be configured to occur on soft ramp and
zero cross transitions. The DAC also includes de-emphasis, limiting functions and a beep generator
delivering tones selectable across a range of two full
octaves.
The stereo headphone amplifier is powered from a separate positive supply and the integrated charge pump
provides a negative supply. This allows a gr ound-centered analog output with a wide signal swing and
eliminates external DC-blocking capacitors.
In addition to its many features, the CS42L51 operates
from a low-voltage analog and digital core, making this
CODEC ideal for portable systems that require extremely low power consumption in a minimal amount of
space.
The CS42L51 is available in a 32-pin QFN package in
both Commercial (-10 to +70° C) and Automotive
grades (-40 to +85° C). The CDB42L51 Customer Demonstration board is also available for device evaluation
and implementation suggestions. Please see “Ordering
Figure 19.I²S Format .................................................................................................................................40
Figure 20.Left-Justified Format .................................................................................................................41
Figure 21.Right-Justified Format (DAC only) ............................................................................................ 41
Left Right Clock (Input/Output) - Determines which channel, Left or Right, is currently active on the
LRCK
SDA/CDIN
(MCLKDIV2)
SCL/CCLK
)
(I²S/LJ
AD0/CS
(DEM)
VA_HP
FLYP
GND_HP
FLYN
VSS_HP
1
serial audio data line.
SerialControl Data (Input/Output) - SDA is a data I/O in I²C Mode. CDIN is the input data line for the
control port interface in SPI Mode.
2
MCLK Divide by 2 (Input) - Hardware Mode: Divides the MCLK by 2 prior to all internal circuitry.
Serial Control Port Clock (Input) - Serial clock for the serial control port.
3
Interface Format Selection (Input) - Hardware Mode: Selects between I²S & Left-Justified interface for-
mats for the ADC & DAC.
Address Bit 0 (I²C) / Control Port Chip Select (SPI) (Input) - AD0 is a chip address pin in I²C Mode; CS
is the chip-select signal for SPI format.
4
De-Emphasis(Input) - Hardware Mode: Enables/disables the de-emphasis filter.
5
Analog Power For Headphone (Input) - Positive power for the internal analog headphone section.
6
Charge Pump Cap Positive Node (Input) - Positive node for the external charge pump capacitor.
7
Analog Ground (Input) - Ground reference for the internal headphone/charge pump section.
8
Charge Pump Cap Negative Node (Input) - Negative node for the external charge pump capacitor.
Negative Voltage From Charge Pump (Output) - Negative voltage rail for the internal analog head-
9
phone section.
25
RESET
AIN1B
24
AIN1A
23
AFILTB
22
AFILTA
21
20
AIN2B/BIAS
AIN2A
19
MICIN2/BIAS/AIN3B
18
MICIN1/AIN3A
17
ADC_FILT+
DS679F17
CS42L51
AOUTB
AOUTA
VA
AGND
DAC_FILT+
VQ
ADC_FILT+
MICIN1/
AIN3A
MICIN2/
BIAS/AIN3B
AIN2A
AIN2B/BIAS
AFILTA
AFILTB
AIN1A
AIN1B
RESET
VL
VD
DGND
SDOUT
)
(M/S
MCLK
SCLK
SDIN
Thermal Pad
10
Analog Audio Output (Output) - The full-scale output level is specified in the DAC Analog Characteris-
11
tics specification table
12
Analog Power (Input) - Positive power for the internal analog section.
13
Analog Ground (Input) - Ground reference for the internal analog section.
14
Positive Voltage Reference (Output) - Positive reference voltage for the internal sampling circuits.
15
Quiescent Voltage (Output) - Filter connection for internal quiescent voltage.
16
Positive Voltage Reference (Output) - Positive reference voltage for the internal sampling circuits.
Microphone Input 1 (Input) - The full-scale level is specified in th e ADC Analog Characteristics specifi-
17
cation table.
Microphone Input 2 (Input/Output) - The full-scale level is specified in the ADC Analog Characteristics
18
specification table. This pin can also be configured as an output to provide a low noise bias supply for an
external microphone. Electrical characteristics are specified in the DC Electrical Characteristics table.
Analog Input (Input) - The full-scale level is specified in the ADC Analog Characteristics specification
19
table.
Analog Input (Input/Output) -
20
cation table. This pin can also be configured as an output to provide a low noise bias supply for an external microphone. Electrical characteristics are specified in the DC Electrical Characteristics table.
21
Filter Connection (Output) - Filter connection for the ADC inputs.
22
23
Analog Input (Input) - The full-scale level is specified in the ADC Analog Characteristics specification
24
table.
25
Reset (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
26
host control port. Refer to the Recommended Operating Conditions for appropriate voltages.
27
Digital Power (Input) - Positive power for the internal digital section.
28
Digital Ground (Input) - Ground reference for the internal digital section.
Serial Audio Data Output (Output) - Output for two’s complement serial au d io da ta.
29
Serial Port Master/Slave (Input/Output) - Hardware Mode Startup Option: Selects between Master and
Slave Mode for the serial port.
30
Master Clock (Input) - Clock source for the delta-sigma modulators.
31
Serial Clock (Input/Output) - Serial clock for the serial audio interface.
32
Serial Audio Data Input (Input) - Input for two’s complement serial audio data.
-
Thermal relief pad for optimize d he a t di ssi pation. See “QFN Thermal Pad” on page 81.
The full-scale level is specified in the ADC Analog Characteristics specifi-
8DS679F1
1.1Digital I/O Pin Characteristics
The logic level for each input should not exceed the maximum ratings for the VL power supply.
CS42L51
Pin Name
SW/(HW)
RESETInput
SCL/CCLK
)
(I²S/LJ
SDA/CDIN
(MCLKDIV2)
AD0/CS
(DEM)
MCLKInput
LRCKInput/Output
SCLKInput/Output
SDOUT
)
(M/S
SDINInput
Input
Input/Output
Input
Input/Output
I/ODriverReceiver
-1.8 V - 3.3 V
-1.8 V - 3.3 V, with Hysteresis
1.8 V - 3.3 V, CMOS/Open Drain1.8 V - 3.3 V, with Hysteresis
-1.8 V - 3.3 V
-1.8 V - 3.3 V
1.8 V - 3.3 V, CMOS1.8 V - 3.3 V
1.8 V - 3.3 V, CMOS1.8 V - 3.3 V
1.8 V - 3.3 V, CMOS1.8 V - 3.3 V
-1.8 V - 3.3 V
Table 1. I/O Power Rails
DS679F19
2. TYPICAL CONNECTION DIAGRAMS
+1.8 V o r +2.5 V
See Note 5
* *Use low ESR ceramic capacitors.
Note 2 :
For best response to Fs/2 :
C
This circuitry is intended fo r applications where the
CS42L51 connects directly to an unbalanced output of the
device. For internal routing applications please see the
DAC Analog Output Cha racteristics section for loading
limitations.
Note 5 :
Larger capacitors, such as 1.5 µF, improves the charge
pump performance (and subsequent THD+N) at the full
scale output power achieved with gain (G) settings
greater than default.
470×+
R
ext
=
()
RFs
π
ext
1.5 µF
1.5 µF
4704
**
**
1 µF
1 µF
1 µF
Digital Audio
Processor
2 k
2 k
Ω
+1.8 V, +2.5 V
See Note 1
or +3.3 V
Note 1:
Resistors are required for I²C
control port operation
Note 4:
Series resistance in the path of the power supplies must
be avoided. Any voltage drop on VA_HP will dir ectly
impact the negative charge pump supply (VSS_HP) and
result in clipping on the audio output .
Headphone Out
Left & Right
470
Ω
C
Line Level Out
Left & Right
See Note 2
C
470
Ω
Speaker Driver
Left Analog Input 1
100 Ω
100 kΩ
100 kΩ
100 Ω
100 Ω
100 Ω
100 kΩ
R
L
Note 3: The value of RL is dictated
by the microphone cartridge.
Note 4:
Series resistance in the path of the power supplies (typically
used for added filtering) must be avoided. Any voltage drop
on VA_HP will directly impact the negative charge pump
supply (VSS_HP) and result in clipping on the audio output .
470Ω
470Ω
100 Ω
1 µF
100 Ω
1 µF
+1.8V or +2.5V
C
Line Level Out
Left & Right
See Note 2
C
Speaker Driver
Left Analog Input 1
100 kΩ
100 kΩ
Right Analog Input 1
Headphone Out
Left & Right
R
ext
R
ext
ADC_FILT+
DAC_FILT+
AGND
AFILTA
AFILTB
VQ
1 µF
**
* Capacitors must be C0G or equivalent
150 pF
150 pF
10 µF
1 µF
Note 2 :
This circuitry is intended for applications where the CS 42L51 connects directly to an unbalanced output of the device . For
internal routing applications please see the DAC Analog Output Characteristics section for loading limitations .
(All Min/Max characteristics and specifications are guaranteed over the Specified Operating Conditions. Typical performance characteristics and specifications are derived from measurements taken at nominal supply voltages and
T
= 25° C.)
A
SPECIFIED OPERATING CONDITIONS
(AGND=DGND=0 V, all voltages with respect to ground.)
ParametersSymbol Min NomMaxUnits
DC Power Supply (Note 1)
Analog Core
Headphone Amplifier
Digital Core
Serial/Control Port Interface
Ambient Temperature Commercial - CNZ
Automotive - DNZ
VA
VA_HP
VD
VL
T
A
1.65
2.37
1.65
2.37
1.65
2.37
1.65
2.37
3.14
-10
-40
1.8
2.5
1.8
2.5
1.8
2.5
1.8
2.5
3.3
1.89
2.63
1.89
2.63
1.89
2.63
1.89
2.63
3.47
-
-
+70
+85
V
V
V
V
V
V
V
V
V
°C
°C
ABSOLUTE MAXIMUM RATINGS
(AGND = DGND = 0 V; all voltages with respect to ground.)
ParametersSymbolMinMaxUnits
DC Power SupplyAnalog
Digital
Serial/Control Port Interface
Input Current(Note 2)
Analog Input Voltage
Digital Input Voltage
(Note 3)
Ambient Operating Temperature (power applied)
Storage Temperature
VA, VA_HP
VD
VL
I
in
V
IN
V
IND
T
A
T
stg
WARNING:Operation at or beyond these limits may result in permanent damage to the device. Normal operation
is not guaranteed at these extremes.
Notes:
1. The device will operate properly over the full range of the analog, headphone amplifier, digital core and
serial/control port interface supplies.
2. Any pin except supplies. Transient currents of up to ±100 mA on the analog input pins will not cause
SCR latch-up.
3. The maximum over/under voltage is limited by the input current.
-0.3
-0.3
-0.3
-±10mA
AGND-0.7VA+0.7
-0.3VL+ 0.4V
-50+115°C
-65+150°C
3.0
3.0
4.0
V
V
V
V
12DS679F1
CS42L51
ANALOG INPUT CHARACTERISTICS (COMMERCIAL - CNZ)
(Test Conditions (unless otherwise specified): Input sine wave (relative to digital full scale): 1 kHz through passive
input filter; Measurement Bandwidth is 10 Hz to 20 kHz unless otherwise specified. Sample Frequency = 48 kHz)
VA = 2.5 V (nominal)VA = 1.8 V (nominal)
Parameter (Note 5)
Analog In to ADC (PGA bypassed)
Dynamic Range A-weighted
unweighted
Total Harmonic Distortion + Noise -1 dBFS
-20 dBFS
-60 dBFS
Analog In to PGA to ADC
Dynamic Range
PGA Setting: 0 dBA-weighted
unweighted
PGA Setting: +12 dBA-weighted
unweighted
Total Harmonic Distortion + Noise
PGA Setting: 0 dB -1 dBFS
-60 dBFS
PGA Setting: +12 dB -1 dBFS
Analog In to MIC Pre-Amp (+16 dB) to PGA to ADC
Dynamic Range
PGA Setting: 0 dBA-weighted
unweighted
Total Harmonic Distortion + Noise
PGA Setting: 0 dB -1 dBFS
Analog In to MIC Pre-Amp (+32 dB) to PGA to ADC
Dynamic Range
PGA Setting: 0 dBA-weighted
unweighted
Total Harmonic Distortion + Noise
PGA Setting: 0 dB -1 dBFS
Other Characteristics
DC Accuracy
Interchannel Gain Mismatch
Gain Drift
Offset ErrorSDOUT Code with HPF On
Input
Interchannel Isolation
DAC Isolation (Note 4)
Full-scale Input VoltageADC
PGA (0 dB)
MIC (+16 dB)
MIC (+32 dB)
Input Impedance (Note 6)ADC
PGA
MIC
MinTypMaxMinTypMaxUnit
93
90
-
-
-
92
89
85
82
-
-
--85-79--83-77dB
-
-
--76- --74-dB
-
-
--74- --71-dB
-0.2- -0.2-dB
-±100--±100-ppm/°C
-352--352-LSB
-90--90-dB
-70--70-dB
0.74•VA
0.75•VA
-
-
-
99
96
-86
-76
-36
98
95
91
88
-88
-35
86
83
78
74
0.78•VA
0.794•VA
0.129•VA
0.022•VA
20
39
50
-
-
-80
-
-
-
-
-
-
-81
-
-
-
-
-
0.82•VA
0.83•VA
-
-
-
90
87
-
-
-
89
86
82
79
-
-
-
-
-
-
0.74•VA
0.75•VA
-
-
-
96
93
-84
-73
-33
95
92
88
85
-86
-32
83
80
75
71
0.78•VA
0.794•VA
0.129•VA
0.022•VA
20
39
50
-
-
-78
-
-
-
-
-
-
-80
-
-
-
-
-
0.82•VA
0.83•VA
-
-
-
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
Vpp
Vpp
Vpp
Vpp
kΩ
kΩ
kΩ
DS679F113
CS42L51
ANALOG INPUT CHARACTERISTICS (AUTOMOTIVE - DNZ)
(Test Conditions (unless otherwise specified): Input sine wave (relative to full scale): 1 kHz through passive input
filter; Measurement Bandwidth is 10 Hz to 20 kHz unless otherwise specified. Sample Frequency = 48 kHz)
VA = 2.5 V (nominal)VA = 1.8 V (nominal)
Parameter (Note 5)
Analog In to ADC
Dynamic Range A-weighted
unweighted
Total Harmonic Distortion + Noise -1 dBFS
-20 dBFS
-60 dBFS
Analog In to PGA to ADC
Dynamic Range
PGA Setting: 0 dBA-weighted
unweighted
PGA Setting: +12 dBA-weighted
unweighted
Total Harmonic Distortion + Noise
PGA Setting: 0 dB -1 dBFS
-60 dBFS
PGA Setting: +12 dB -1 dBFS
Analog In to MIC Pre-Amp (+16 dB) to PGA to ADC
Dynamic Range
PGA Setting: 0 dBA-weighted
unweighted
Total Harmonic Distortion + Noise
PGA Setting: 0 dB -1 dBFS
Analog In to MIC Pre-Amp (+32 dB) to PGA to ADC
Dynamic Range
PGA Setting: 0 dBA-weighted
unweighted
Total Harmonic Distortion + Noise
PGA Setting: 0 dB -1 dBFS
Other Characteristics
DC Accuracy
Interchannel Gain Mismatch
Gain Drift
Offset ErrorSDOUT Code with HPF On
Input
Interchannel Isolation
DAC Isolation (Note 4)
Full-scale Input VoltageADC
PGA (0 dB)
MIC (+16 dB)
MIC (+32 dB)
Input Impedance (Note 6)ADC
PGA
MIC
4. Measured with DAC delivering full-scale output power into 16 Ω.
MinTypMaxMinTypMaxUnit
91
78
-
-
-
90
87
83
80
-
-
--85-77--83-75dB
-
-
--76---74-dB
-
-
--74---71-dB
-0.1--0.1-dB
-±100--±100-ppm/°C
-352--352-LSB
-90--90-dB
-70--70-dB
0.74•VA
0.75•VA
18
40
50
99
96
-86
-76
-36
98
95
91
88
-88
-35
86
83
78
74
0.78•VA
0.794•VA
0.129•VA
0.022•VA
-
-
-
-
-
-78
-
-
-
-
-
-
-80
-
-
-
-
-
0.82•VA
0.83•VA
-
-
-
88
85
-
-
-
87
84
80
77
-
-
-
-
-
-
0.74•VA
0.75•VA
18
40
50
96
93
-84
-73
-33
95
92
88
85
-86
-32
83
80
75
71
0.78•VA
0.794•VA
0.129•VA
0.022•VA
-
-
-
-
-
-76
-
-
-
-
-
-
-78
-
-
-
-
-
0.82•VA
0.83•VA
-
-
-
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
Vpp
Vpp
Vpp
Vpp
kΩ
kΩ
kΩ
14DS679F1
5. Referred to the typical full-scale voltage. Applies to all THD+N and Dynamic Range values in the table.
6. Measured between AINxx and AGND.
ADC DIGITAL FILTER CHARACTERISTICS
Parameter (Note 7)MinTypMaxUnit
Passband (Frequency Response) to -0.1 dB corner
Passband Ripple
Stopband
Stopband Attenuation
Total Group Delay
High-Pass Filter Characteristics (48 kHz Fs)
Frequency Response-3.0 dB
-0.13 dB
Phase Deviation@ 20 Hz
Passband Ripple
Filter Settling Time
CS42L51
0-0.4948Fs
-0.09-0.17dB
0.6--Fs
33--dB
-7.6/Fs-s
-
-
-10-Deg
--0.17dB
-10
3.7
24.2
5
/Fs0s
-
-
Hz
Hz
7. Response is clock-dependent and will scale with Fs. Note that the response plots (Figure 33 to Figure 41)
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.
DS679F115
CS42L51
ANALOG OUTPUT CHARACTERISTICS (COMMERCIAL - CNZ)
(Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; measurement
bandwidth is 10 Hz to 20 kHz; Sample Frequency = 48 kHz; test load R
(see Figure 3), and test load R
= 16 Ω, CL = 10 pF (see Figure 3) for the headphone output. HP_GAIN[2:0] = 011.)
L
= 10 kΩ, CL = 10 pF for the line output
L
Parameter(Note 8)
RL = 10 kΩ
Dynamic Range
18 to 24-BitA-weighted
unweighted
16-BitA-weighted
unweighted
Total Harmonic Distortion + Noise
18 to 24-Bit0 dB
-20 dB
-60 dB
16-Bit0 dB
-20 dB
-60 dB
RL = 16 Ω
Dynamic Range
18 to 24-Bit A-weighted
unweighted
16-BitA-weighted
unweighted
Total Harmonic Distortion + Noise
18 to 24-Bit0 dB
-20 dB
-60 dB
16-Bit0 dB
-20 dB
-60 dB
Other Characteristics for RL = 16 Ω or 10 k
Output ParametersModulation Index (MI)
(Note 9)Analog Gain Multiplier (G)
Full-scale Output Voltage (2•G•MI•VA) (Note 9)
Full-scale Output Power (Note 9)
Interchannel Isolation (1 kHz)16 Ω
Interchannel Gain Mismatch
Gain Drift
AC-Load Resistance (R
Load Capacitance (C
)(Note 10)
L
)(Note 10)
L
Ω
10 kΩ
VA = 2.5V (nominal)
Min Typ Max
92
89
-
-
-
-
-
-
-
-
92
89
-
-
-
-
-
-
-
-
-
Refer to Table “Line Output Voltage Characteristics” on
Refer to Table “Headphone Output Power Characteristics”
-
-
-0.10.25-0.10.25dB
-±100--±100-
16--16--Ω
--150--150pF
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
-
-
-
-
-78
-
-
-
-
-
-
-
-
-
-69
-
-
-
-
-
--
page 18
on page 19
-
-
VA = 1.8V
(nominal)
Min Typ Max Unit
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
-
-
-
-
-
-
-
-
-
-69
-
-
-
-
-
-
-
-
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
Vpp
mW
dB
dB
ppm/°
C
16DS679F1
CS42L51
ANALOG OUTPUT CHARACTERISTICS (AUTOMOTIVE - DNZ)
(Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; measurement
bandwidth is 10 Hz to 20 kHz; Sample Frequency = 48 kHz and 96 kHz; test load R
line output (see Figure 3), and test load R
= 16 Ω, CL = 10 pF (see Figure 3) for the headphone output.
L
HP_GAIN[2:0] = 011.)
= 10 kΩ, CL = 10 pF for the
L
Parameter(Note 8)
RL = 10 kΩ
Dynamic Range
18 to 24-BitA-weighted
unweighted
16-BitA-weighted
unweighted
Total Harmonic Distortion + Noise
18 to 24-Bit0 dB
-20 dB
-60 dB
16-Bit0 dB
-20 dB
-60 dB
RL = 16
Ω
Dynamic Range
18 to 24-Bit A-weighted
unweighted
16-BitA-weighted
unweighted
Total Harmonic Distortion + Noise
18 to 24-Bit0 dB
-20 dB
-60 dB
16-Bit0 dB
-20 dB
-60 dB
Other Characteristics for RL = 16 Ω or 10 k
Output ParametersModulation Index (MI)
(Note 9)Analog Gain Multiplier (G)
Full-scale Output Voltage (2•G•MI•VA) (Note 9)
Full-scale Output Power (Note 9)
Interchannel Isolation (1 kHz)16 Ω
Interchannel Gain Mismatch
Gain Drift
AC-Load Resistance (R
Load Capacitance (C
)(Note 10)
L
)(Note 10)
L
Ω
10 kΩ
VA = 2.5V (nominal)
Min Typ Max
90
87
-
-
-
-
-
-
-
-
90
87
-
-
-
-
-
-
-
-
-
Refer to Table “Line Output Voltage Characteristics” on
Refer to Table “Headphone Output Power Characteristics”
-
-
-0.10.25-0.10.25dB
-±100--±100-
16--16--Ω
--150--150pF
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
-
-
-
-
-73
-
-
-
-
-
-
-
-
-
-67
-
-
-
-
-
--
page 18
on page 19
-
-
VA = 1.8V
(nominal)
Min Typ Max Unit
87
84
-
-
-
-
-
-
-
-
87
84
-
-
-
-
-
-
-
-
-
-
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
-
-
-
-
-80
-
-
-
-
-
-
-
-
-
-67
-
-
-
-
-
-
-
-
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
Vpp
mW
dB
dB
ppm/°
C
DS679F117
CS42L51
LINE OUTPUT VOLTAGE CHARACTERISTICS
Test conditions (unless otherwise specified): Inp ut test signal is a fu ll-scale 997 Hz sine wave; measurement bandwidth is 10 Hz to 20 kHz; Sample Frequency = 48 kHz; test load R
= 10 kΩ, CL = 10 pF (see Figure 3).
L
Parameter
AOUTx Voltage Into RL = 10 k
HP_GAIN[2:0]
0000.3959
0010.4571
0100.511 1
011 (default)0.6047
1000.7099
1010.8399
1101.0000
1111.1430
Analog
Gain (G)
VA = 2.5V (nominal)
Min Typ Max
VA = 1.8V
(nominal)
Min Typ Max Unit
Ω
VA_HP
1.8 V -1.34--0.97-V
2.5 V -1.34--0.97-V
1.8 V -1.55--1.12-V
2.5 V -1.55--1.12-V
1.8 V -1.73--1.25-V
2.5 V -1.73--1.25-V
1.8 V -2.05-1.411.48 1.55V
2.5 V 1.952.052.15-1.48-V
1.8 V -2.41--1.73-V
2.5 V -2.41--1.73-V
1.8 V -2.85-2.05V
2.5 V -2.85--2.05-V
1.8 V -3.39--2.44-V
2.5 V -3.39--2.44-V
1.8 V
2.5 V -3.88--2.79-V
(See (Note 11)2.79V
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
pp
18DS679F1
CS42L51
HEADPHONE OUTPUT POWER CHARACTERISTICS
Test conditions (unless otherwise specified): Input test signal is a full-scale 997 Hz sine wave; measurement bandwidth is 10 Hz to 20 kHz; Sample Frequency = 48 kHz; test load R
= 16 Ω, CL = 10 pF (see Figure 3).
L
Parameter
AOUTx Power Into RL = 16
HP_GAIN[2:0]
0000.3959
0010.4571
0100.511 1
011 (default)0.6 047
1000.7099
1010.8399
1101.0000
1111.1430
Analog
Gain (G)
8. One-half LSB of triangular PDF dither is added to data.
9. Full-scale output voltage and power is determined by the gain setting, G, in register “Headphone Analog
Gain (HP_GAIN[2:0])” on page 57. High gain settings at certain VA and VA_HP supply levels may
cause clipping when the audio signal approaches full-scale, maximum power output, as shown in
Figures 27 - 30 on page 76.
10. See Figure 3. R
quired for the internal op-amp's stability and signal integrity. In this circuit topology, C
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.
11. VA_HP settings lower than VA reduces the headroom of the h eadphone amplifier. As a resu lt, the DAC
may not achieve the full THD+N performance at full-scale output voltage and power.
VA = 2.5V (nominal)
Min Typ Max
VA = 1.8V
(nominal)
Min Typ Max Unit
Ω
VA_HP
1.8 V -14--7 -mW
2.5 V -14--7 -mW
1.8 V -19--10 -mW
2.5 V -19--10 -mW
1.8 V -23--12 -mW
2.5 V -23--12 -mW
1.8 V (Note 11)-17 -mW
2.5 V -32--17 -mW
1.8 V (Note 11)-23 -mW
2.5 V -44--23 -mW
1.8 V (Note 9)mW
2.5 V -32 -mW
1.8 V
2.5 V mW
(Note 9, 11)
1.8 V mW
2.5 V mW
and CL reflect the recommended minimum resistance and maximum capacitance re-
COMBINED DAC INTERPOLATION & ON-CHIP ANALOG FILTER RESPONSE
Parameter (Note 12)MinTypMaxUnit
Frequency Response 10 Hz to 20 kHz
Passband to -0.05 dB corner
to -3 dB corner
StopBand
StopBand Attenuation (Note 13)
Group Delay
De-emphasis Error Fs = 32 kHz
Fs = 44.1 kHz
Fs = 48 kHz
-0.01-+0.08dB
0
0
0.5465--Fs
50--dB
-10.4/Fs-s
-
-
-
-
-
-
-
-
0.4780
0.4996
+1.5/+0
+0.05/-0.25
-0.2/-0.4
Fs
Fs
dB
dB
dB
Notes:
12. Response is clock dependent and will scale with Fs. Note that the response plots (Figure 38 to Figure 41
on page 82) 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.
SWITCHING SPECIFICATIONS - SERIAL PORT
(Inputs: Logic 0 = DGND, Logic 1 = VL, SDOUT C
LOAD
= 15 pF.)
ParametersSymbol Min MaxUnits
RESET pin Low Pulse Width(Not e 14 )
MCLK Frequency
MCLK Duty Cycle(Note 15)
Slave Mode
Input Sample Rate (LRCK)Quarter-Speed Mode
Half-Speed Mode
Single-Speed Mode
Double-Speed Mode
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
F
s
F
s
F
s
F
s
1/t
P
t
s(LK-SK)
t
d(MSB)
t
s(SDO-SK)
t
h(SK-SDO)
t
s(SD-SK)
t
h
1-ms
1.02438.4MHz
4555%
4
8
4
50
4555%
-64•FsHz
4555%
40-ns
-52ns
20-ns
30-ns
20-ns
20-ns
12.5
25
50
100
kHz
kHz
kHz
kHz
20DS679F1
Master Mode (Note 17)
CS42L51
ParametersSymbol Min Max Units
Output Sample Rate (LRCK) All Speed Modes
(Note 17)
LRCK Duty Cycle
SCLK Frequency
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 CS42L51, RESET should be held low after the power supplies and clocks are
settled.
15. See “Example System Clock Frequencies” on page 79 for typical MCLK frequencies.
16. See
17. “Master” on page 39
18. “MCLK” refers to the external master clock applied.
LRCK
//
t
s(LK-SK)
//
t
P
//
SCLK
//
SDOUT
SDIN
t
d(MSB)
t
s(SD-SK)
MSB
MSB
t
h(SK-SDO)
//
//
t
//
//
h
Figure 4. Serial Audio Interface Slave Mode Timing
F
s
1/t
P
t
d(MSB)
t
s(SDO-SK)
t
h(SK-SDO)
t
s(SD-SK)
t
h
t
s(SDO-SK)
MSB-1
MSB-1
MCLK
-Hz
----------------128
4555%
-64•F
Hz
s
4555%
-52ns
20-ns
30-ns
20-ns
20-ns
LRCK
//
//
t
P
//
SCLK
//
SDOUT
SDIN
t
d(MSB)
t
s(SD-SK)
MSB
MSB
t
h(SK-SDO)
//
//
t
//
//
t
s(SDO-SK)
MSB-1
h
MSB-1
Figure 5. Serial Audio Interface Master Mode Timing
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 19)
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
f
t
t
t
hdst
t
low
t
high
t
sust
t
hdd
t
sud
t
t
susp
t
ack
scl
irs
buf
rc
t
fc
CS42L51
-100kHz
500-ns
4.7-µs
4.0-µs
4.7-µs
4.0-µs
4.7-µs
0-µs
250-ns
-1µs
-300ns
4.7-µs
3003450ns
19. Data must be held for sufficient time to bridge the transition time, t
RST
t
SDA
SCL
irs
StopStart
t
buf
t
t
hdst
low
t
high
t
hdd
Figure 6. Control Port Timing - I²C
t
sud
Repeated
Start
t
sust
t
hdst
, of SCL.
fc
t
t
r
Stop
f
t
susp
22DS679F1
SWITCHING CHARACTERISTICS - SPI™ CONTROL PORT
(Inputs: Logic 0 = DGN D, Lo gic 1 = VL)
ParameterSymbol Min MaxUnits
CS42L51
CCLK Clock Frequency
RESET Rising Edge to CS Falling
Falling to CCLK Edge
CS
CS
High Time Between Transmissions
CCLK Low Time
CCLK High Time
CDIN to CCLK Rising Setup Time
CCLK Rising to DATA Hold Time(Note 20)
Rise Time of CCLK and CDIN(Note 21)
Fall Time of CCLK and CDIN(Note 21)
20. Data must be held for sufficient time to bridge the transition time of CCLK.
21. For f
<1 MHz.
sck
RST
CS
CCLK
t
srs
t
t
sch
css
t
scl
t
dsu
f
sck
t
srs
t
css
t
csh
t
scl
t
sch
t
dsu
t
dh
t
r2
t
f2
t
f2
t
dh
06.0MHz
20-ns
20-ns
1.0-µs
66-ns
66-ns
40-ns
15-ns
-100ns
-100ns
t
csh
t
r2
CDIN
Figure 7. Control Port Timing - SPI Format
DS679F123
DC ELECTRICAL CHARACTERISTICS
(AGND = 0 V; all voltages with respect to ground.)
ParametersMinTypMaxUnits
VQ Characteristics
Nominal Voltage
Output Impedance
DC Current Source/Sink(Note 22)
DAC_FILT+ Nominal Voltage
ADC_FILT+ Nominal Voltage
VSS_HP Characteristics
Nominal Voltage
DC Current Source
MIC BIAS Characteristics
Nominal VoltageMICBIAS_LVL[1:0] = 00
MICBIAS_LVL[1:0] = 01
MICBIAS_LVL[1:0] = 10
MICBIAS_LVL[1:0] = 11
DC Current Source
Power Supply Rejection Ratio (PSRR)1 kHz
Power Supply Rejection Ratio (PSRR) (Note 23)1 kHz
CS42L51
-
-
-
-
-
-
-
-
-
-
-
-
-
-60-dB
0.5•VA
23
-
VA
VA
-0.8•(VA_HP)-
0.8•VA
0.7•VA
0.6•VA
0.5•VA
-
50
10
10
-
-
-
-
-
-
-
-
1
-
V
kΩµA
V
V
V
µA
V
V
V
V
mA
dB
22. The DC current draw represents the allowed current draw from the VQ pin due to typical leakage
through electrolytic de-coupling capacitors.
23. Valid with the recommended capacitor values on DAC_FILT+, ADC_FILT+ and VQ. In crea sing the capacitance will also increase the PSRR.
DIGITAL INTERFACE SPECIFICATIONS & CHARACTERISTICS
Parameters (Note 24)Symbol Min MaxUnits
Input Leakage Current
Input Capacitance
1.8 V - 3.3 V Logic
High-Level Output Voltage
Low-Level Output Voltage
High-Level Input Voltage
Low-Level Input Voltage
24. See “Digital I/O Pin Characteristics” on page 9 for serial and control port power rails.
(IOH = -100 µA)
(IOL = 100 µA)
I
in
V
OH
V
OL
V
IH
V
IL
-±10µA
-10pF
VL - 0.2-V
-0.2V
0.68•VL-V
-0.32•VLV
24DS679F1
POWER CONSUMPTION
See (Note 25)
CS42L51
Power Ctl. RegistersTypical Current (mA)
02h03h
Operation
PDN_DACB
PDN_DACA
PDN_PGAB
PDN_PGAA
PDN_ADCB
PDN_ADCA
PDN
PDN_MICB
PDN_MICA
1
Off
(Note 26)
2Standby (Note 27)
3 Mono RecordADC1111100111
PGA to ADC
MIC to PGA to ADC
(with Bias)
MIC to PGA to ADC
(no Bias)
4 Stereo RecordADC1111000111
PGA to ADC
MIC to PGA to ADC
(no Bias)
5 Mono Playback1011110111
6 Stereo Playback
Mono Record & Playback
7
PGA in (no MIC) to Mono Out
Phone Monitor
8
MIC (w/bias) in to Mono Out
Stereo Record & Playback
9
PGA in (no MIC) to Stereo Out
xxxxxxxxxx
xxxxxx1xxx
1110100111
1110100100
1110100101
1100000111
1100000001
0011110111
1010100111
1010100100
0000000111
i
i
VA_HP
V
PDN_MICBIAS
1.8 00000
2.5 00000
1.8 00.010.0200.05
2.5 00.010.0300.10
1.8 01.852.030.037.05
2.5 02.073.050.0512.94
1.8 02.352.030.037.95
2.5 02.583.080.0514.29
1.8 03.672.050.0310.36
2.5 03.953.090.0517.71
1.8 03.272.030.039.61
2.5 03.523.080.0516.62
1.8 02.692.120.038.72
2.5 02.933.180.0415.40
1.8 03.652.120.0310.45
2.5 03.913.170.0417.84
1.8 05.482.110.0313.73
2.5 05.763.170.0422.45
1.8 1.661.402.350.019.74
2.5 2.031.713.480.0218.08
1.8 2.772.052.350.0112.93
2.5 3.212.503.490.0223.02
1.8 1.663.632.730.0314.49
2.5 2.034.164.080.0525.79
1.8 1.664.952.750.0316.90
2.5 2.035.524.080.0529.20
1.8 2.775.592.820.0320.18
2.5 3.216.284.190.0434.30
i
VA
i
VD
VL
(Note 28)
Total
Power
(mW
rms
)
25. 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.
26. RESET
27. 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.
28. VL current will slightly increase in master mode.
DS679F125
4. APPLICATIONS
4.1Overview
4.1.1Architecture
The CS42L51 is a highly integrated, low power, 24-bit au dio CODEC comprised of stereo analog-to-digital
converters (ADC), and stereo digital-to-analog converters (DAC) designed using multi-bit delta-sigma
techniques. The DAC operates at an oversampling ratio of 128Fs and the ADC operates at 64Fs, where
Fs is equal to the system sample rate. The different clock rates ma ximize power savings while maintaining
high performance. The CODEC operates in one of four sample rate speed modes: Quarter, 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 & MIC Inputs
The analog input portion of the CODEC allows selection from and configuration of multiple combinations
of stereo and microphone (MIC) sources. Six line inputs with configuration for two MIC inputs (or one MIC
input with common mode rejection), two MIC bias outputs and independent channel control (including a
high-pass filter disable function) are available. A Programmable Gain Amplifier (PGA), MIC boost, and Automatic Level Control (ALC), with noise gate settings, provide analog gain and adjustment. Digital volume
controls, including gain, boost, attenuation and inversion are also available.
CS42L51
4.1.3Line & Headphone Outputs
The analog output portion of the D/A 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 ground. This eliminates the need for large DC-Blocking capacitors and allows
the amplifier to deliver more power to headphone loads at lower supply voltages. Eight gain settings for
the headphone amplifier are available.
4.1.4Signal Processing Eng ine
A signal processing engine is available to process serial input D/A data be fore output to the DAC. The
D/A data has independent volume controls and mixing functions such as mo no mixes a nd left/right cha nnel swaps. A Tone Control provides bass and treble at four selectable corner frequencies. An automatic
level control provides limiting capabilities at programmable attack and release rates, maximum thresholds
and soft ramping. A 15/50 µs de-emphasis filter is also available at a 44.1 kHz sample rate.
4.1.5Beep Generator
A beep may be generated internally at select frequencies across approximately two octave major scales
and configured to occur continuously, periodically or at single time interva ls controlled by the user. Volume
may be controlled independently.
4.1.6Device Control (Hardware or Software Mode)
In Software Mode, all functions and features may be controlled via a two-wire I²C or three-wire SPI control
port interface. In Hardware Mode, a limited feature set may be controlled via stand-alone control pins.
4.1.7Power Management
Two Software Mode control registers provide independent power-down control of the ADC, DAC, PGA,
MIC pre-amp and MIC bias, allowing operation in select applications with minimal power consumption.
26DS679F1
4.2Hardware Mode
A limited feature-set is available when the CODEC powers up in Hardware Mode (see “Recommended Pow-
er-Up Sequence” on page 41) and may be controlled via stand-alone control pins. Table 2 shows a list of
functions/features, the default configuration and the associated stand-alone control available.
Zero Cross
DAC De-Emphasis
Signal Processing Engine (SPE)Mix
Beep
Tone Control
Peak Detect and Limiter
Data Selection
Channel MixADC
DAC
Charge Pump Frequency
Table 2. Hardware Mode Feature Summary
CS42L51
Powered Up
Powered Up
Powered Up
Powered Up
Powered Down
Powered Down
Enabled--
Auto-Detect Speed Mode
Single-Speed Mode
(Selectable)“MCLKDIV2” pin 2
(Selectable)“M/S” pin 29
(Selectable)“I²S/LJ” pin 3
Disabled
Disabled
Disabled
Disabled
0 dB
0 dB
Disabled
Disabled
Enabled
Continuous DC Subtraction
AIN1A to PGAA
AIN1B to PGAB
G = 0.6047
0 dB
Disabled
Enabled
Disabled
(Selectable)“DEM” pin 4
Disabled
Disabled
Disabled
Disabled
Data Input (PCM) to DAC--
ADCA = L; ADCB = R
PCMA = L; PCMB = R
(64xFs)/7--
--
-see Section
4.5 on page 38
see Section
4.5 on page 38
see Section
4.6 on page 40
--
--
--
--
see Section
on page 34
--
--
DS679F127
4.3 Analog Inputs
AINxA and AINxB are the analog inputs, internally biased to VQ, that accepts line-level and MIC-level signals, allowing various gain and signal adjustments for each channel.
CS42L51
ADCA_MUTE
+20dB
Digital
Boost
ADCA_ATT[7:0]
0/-96dB
1dB steps
Attenuator
SOFTA
MUX
DIGMIX
MUX
MICMIX
ADCA_HPF FREEZE
ADCA_HPF ENABLE
ALC_ARATE[5:0]
ALC_RRATE[5:0]
Σ
ALCA_SRDIS
ALCA_ZCDIS
ALC_ENA
MAX[2:0]
MIN[2:0]
ALC_ENB
ALCB_SRDIS
ALCB_ZCDIS
ADCA_DBOOST
ALC
PCM Serial Interface
MUX
MUX
ADCB_HPF FREEZE
ADCB_HPF ENABLE
ADCB_DBOOST
TO SIGNAL PROCESSING
ENGINE (SPE)
FROM SIGNAL
PROCESSING ENGINE
(SPE)
+20dB
Digital
Boost
SOFTB
Attenuator
ADCB_MUTE
ADCB_ATT[7:0]
0/-96dB
1dB steps
Figure 8. Analog Input Architecture
4.3.1Digital Code, Offset & DC Measurement
Noise Gate
PDN_ADCA
Multibit
Oversampling
ADC
INV_ADCA
NG_ALL
NG_EN
THRESH[3:0]
NGDELAY[1:0]
PDN_ADCB
Multibit
Oversampling
ADC
INV_ADCB
PGAA_VOL[5:0]
ADC_SNGVOL
SOFTA
ZCROSSA
+12/-3dB
0.5dB steps
PGA
PDN_PGAA
AINA_MUX[1:0]
MICBIAS_LVL[1:0]
PDN_MICBIAS
PGAB_VOL[5:0]
ADC_SNGVOL
SOFTB
ZCROSSB
+12/-3dB
0.5dB steps
PGA
PDN_PGAB
AINB_MUX[1:0]
MUX
MUX
+16/
32 dB
MICA_BOOST
PDN_MICA
MICBIAS
MICBIAS_SEL
+16/
32dB
MICB_BOOST
PDN_MICB
AIN1A
AIN2A
AIN3A/ MICIN1
AIN1B
AIN2B/MICBIAS
AIN3B/ MICIN2/
MICBIAS
The ADC output data is in two’s complement binary format. For inputs above positive full scale or below
negative full scale, the ADC will output 7FFFFFH or 800000H, respectively and cause the ADC overflow
bit to be set to a ‘1’.
Given the two’s complement format, low-level signals may cause the MSB of the se rial data to periodically
toggle between ‘1’ and ‘0’, poss ibly int roducing noise in to the sys tem as the bit switches back an d fort h.
To prevent this phenomena, a constant DC offset is a dded to th e ser ial d ata brin ging th e low- level sig nal
just above the point at which the MSB would normally toggle, thus reducing the noise introduced. Note
that this offset is not removed (refer to “Analog Input Characteristics (Commercial - CNZ)” on page 13
and/or “Analog Input Characteristics (Automotive - DNZ)” on page 14 for the specified offset level).
The CODEC may be used to measure DC voltages by disabling the high-pass filter for the designated
channel. DC levels are measured relative to VQ and will be decoded as positive two’s complement binary
numbers above VQ and negative two’s complement binary numbers below VQ.
Software
Controls:
“Status (Address 20h) (Read Only)” on page 73, “ADC Control (Address 06h)” on page 54.
28DS679F1
4.3.2High-Pass Filter and DC Offset Calibration
The high-pass filter continuously subtracts a measure of the DC offset from the output of the decimation
filter. If the high-pass filter is “frozen” during normal operation, the current value of the DC offset for the
corresponding channel is held. It is this DC offset that will continue to be subtracted from the conversion
result. This feature makes it possible to perform a system DC offset calibration by:
1. Running the CODECwith the high-pass filter enabled and the DC offset not “frozen” until the filter
settles. See the Digital Filter Characteristics for filter settling time.
2. Freezing the DC offset.
The high-pass filters are controlled using the ADCx_HPFRZ and ADCx_HPFEN bits.
If a particular ADC channel is used to measure DC voltages, the high-pass filter may be disabled using
the ADCx_HPFEN bit.
CS42L51
Software
Controls:
“ADC Control (Address 06h)” on page 54.
4.3.3Digital Routing
The digital output of the ADC may be internally routed to the signal processing engi ne (SPE) for playba ck
of analog input signals. Volume to the DAC may be controlled using the ADCMIX[6:0] bits. The serial input
data may also be routed to the ADC serial interface using the DIGMIX bit. This is useful for recording a
digital mix along with the analog input.
Software
Controls:
“ADCx Mixer Volume Control: ADCA (Address 0Eh) & ADCB (Address 0Fh)” on page 61, “Inter-
face Control (Address 04h)” on page 52.
4.3.4Differential Inputs
The stereo pair inputs act as a single differential input when the MICMIX bit is enabled. This p rovides common mode rejection of noise in digitally intense PCB’s where the microphone signal traverses long traces,
or across long microphone cables as illustrated in Figure 9.
Since the mixer provides a differential combination of the two signals, the potential input mix may exceed
the maximum full-scale input and result in clipping. The level out of the mixer, therefore, is automatically
attenuated 6 dB. Gain may be applied using either the analog PGA o r MIC Pre-amp or the digital ADCMIX
volume control to re-adjust a small signal to desired levels.
The analog inputs may also be used as a differential input pair as illustrated in Figure 10. The two channels are differentially combined when the MICMIX bit is enabled.
4.3.4.1External Passive Components
The microphone input is internally biased to VQ. Input signals must be AC coupled using external capacitors with values consistent with the desired high-pass filter design. The MICINx input resistance of 50 kΩ
may be combined with an external capacitor of 1 µF to achieve the cutoff frequency defined by the equation,
An electrolytic capacitor must be placed such that the positive terminal is positioned relative to the side with
the greater bias voltage. The MICBIAS voltage level is controlled by the MICBIAS_LVL[1:0] bits.
DS679F129
CS42L51
fc
-----------------------------------------------3.18 H z==
1
2π 50 k Ω()1 µF()
The MICBIAS series resistor must be selected based on the requirements of the particular microphone
used. The MICBIAS output pin is selected using the MICBIAS_SEL bit.
Software
Controls:
“Interface Control (Address 04h)” on page 52, “MIC Control (Address 05h)” on page 53.
MICBIAS
20
MICIN1
//
+
17
Σ
MICIN2
//
Figure 9. MIC Input Mix w/Common Mode Rejection
+
18
2.5 V
2.15 V
1.25 V
0.35 V
2.15 V
1.25 V
0.35 V
Full-Scale Differential Input Level (MICMIX=1)
= (AINxA - AINxB) = 3.6 V
PP
Figure 10. Differential Input
= 1.27 V
AINxA
AINxB
RMS
VA
30DS679F1
4.3.5Analog Input Multiplexer
A stereo 4-to-1 analog input multiplexer selects between a line-level input source, or a mic-level input
source, depending on the PDN_PGAx and AINx_MUX[1:0] bit settings. Signals may be routed to or bypassed around the PGA. To conserve power, the PGA’s may be powered down allowing the u ser to select
from multiple line-level sources and route the stereo signal directly to the ADC. When using the MIC preamp, however, the PGA must be powered up.
Analog input channel B may also be used as an output for the MIC bias voltage. The MICBIAS_SEL bit
routes the bias voltage to either of two pins. The multiplexer must then select from the remainder of the
two input channels.
The ADC, PGA and MIC pre-amplifier each has an associated input resistance. When selecting between
these paths, the input resistance to the CODEC will change accordingly. Refer to the input resistance
characteristics in the Characteristic and Specification Tables for the input resistance of each path.
CS42L51
Software
Controls:
“Power Control 1 (Address 02h)” on page 49, “MIC Control (Ad dress 05h)” on page 53“ADCx
Input Select, Invert & Mute (Address 07h)” on page 56.
4.3.6MIC & PGA Gain
The MIC-level input passes through a +16 dB or +32 dB analog gain stage prior to the input multiplexer,
allowing it to be used for microphone level signals without the need for any e x terna l ga in. The PGA mu st
be powered up when using the MIC pre-amp.
The PGA stage provides an additional +12 dB to -3 dB of analog gain in 0.5 dB steps.
Software
Controls:
“Power Control 1 (Address 02h)” on page 49, “ADCx Input Select, Invert & Mute (Address 07h)” on
page 56, “ALCX & PGAX Control: ALCA, PGAA (Address 0Ah) & ALCB, PGAB (Address 0Bh)” on
page 59, “MIC Control (Address 05h)” on page 53.
DS679F131
4.3.7Automatic Level Control (ALC)
When enabled, the ALC monitors the analog input signal after the digital attenuator, detects when peak
levels exceed the maximum threshold settings and lowers , first, the PGA gain settings and then increases
the digital attenuation levels at a programmable attack rate and maintains the resulting level below the
maximum threshold.
When input signal levels fall below the minimum threshold, digital attenuation levels are decreased first
and the PGA gain is then increased at a programmable release rate and maintains the resulting level
above the minimum threshold.
Attack and release rates are affected by the ADC soft ramp/zero cross settings and sample rate, 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.
in the PGAx_VOL register. The ALC will only apply the gain set in the PGAx_VOL.
the output signal between the MIN and MAX thresholds. As the input signal level changes, the level-controlled output may not always be the same but will always fall within the thresholds.
PGAx_VOL[4:0] volume
controls should NOT be
adjusted manually when
ALCx is enabled.
MAX[2:0]
below full scale
Figure 11. ALC
32DS679F1
4.3.8Noise 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.
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.
Ramp-down time to the maximum setting is affected by the SOFTx bit.
Recommended settings: For best results, enable soft ramp for the digital attenuator. When the analog in-
puts are configured for differential signals (see “Differential Inputs” on page 29), enable the NG_ALL bit
to trigger the noise gate only when both inputs fall below the threshold.
CS42L51
Software
Controls:
“Noise Gate Configuration & Misc. (Address 1Fh)” on page 72, “ADC Control (Address 06h)” on
page 54.
Output
(dB)
1
=
N
E
G
N
-52 dB
0
=
N
E
G
N
-96-40
THRESH[2:0]
-64 dB
-80 dB
Maximum Attenuation*
Input (dB)
Figure 12. Noise Gate Attenuation
DS679F133
4.4Analog Outputs
AOUTA and AOUTB are the ground-centered line or headpho ne outputs. Various signal processing options
are available, including digital mixes with the ADC si gnal and an interna l Beep Generator . The desire d path
to the DAC must be selected using the DATA_SEL[1:0] bits.
The CODEC includes on-chip digital de-emphasis optimized for a sample rate of 44.1 kHz. The filter response is shown in Figure 14. The de-emphasis feature is included to accommodate audio recordings
that utilize 50/15 µs pre-emphasis equalization as a means of noise reduction. De-emphasis is only available in Single-Speed Mode.
Software
Controls:
Hardware
Control:
“DAC Control (Address 09h)” on page 58.
Pin
“DEM” pin 4.
Figure 13. Output Architecture
SettingSelection
LO
HI
No De-Emphasis
De-Emphasis Applied
34DS679F1
Gain
0dB
CS42L51
dB
T1=50 µs
-10dB
4.4.2Volume Controls
Three digital volume control functions are implemented, offering independent control over the ADC and
PCM signal paths into the mixer as well as a combined control over the mixed signals. All volume controls
are programmable to ramp in increments o f 0.125 dB at a rate controlled by the DAC soft ramp/zero cross
settings.
All signal paths may also be independently muted via mute control bits. When enabled, each bit attenuates the signal to its maximum value. When the mute bit is disabled, the signal returns to the attenuation
level set in the respective volume control register. The attenuation is ramped up and down at the rate
specified by the DAC_SZC[1:0] bits.
ume Control: AOUTA (Address 16h) & AOUTB (Address 17h)” on page 66, “DAC Output Control
(Address 08h)” on page 57.
4.4.3Mono Channel Mixer
A channel mixer may be used to create a mix of the left and right channels for either the PCM or ADC
signals. This mix allows the user to produce a MONO signal from a stereo source. The mixer may also be
used to implement a left/right channel swap.
T2 = 15 µs
F1F2
3.183 kHz10.61 kHz
Figure 14. De-Emphasis Curve
Frequency
Software
Controls:
“PCM Channel Mixer (Address 18h)” on page 67.
4.4.4Beep Generato r
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 ma y affe ct desire d limitin g perfo rmance. If th e 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, AOUTx_VOL[7:0], will affect the Beep volume, DAC volume may alternatively be controlled using the PCMMIXx_VOL[6:0] bits.
Software
Controls:
“Beep Frequency & Timing Configuration (Address 12h)” on page 62, “Beep Off Time & Volume
(Address 13h)” on page 63, “Beep Configuration & Tone Configuration (Address 14h)” on page 64
DS679F135
CS42L51
REPEAT = '1'
BEEP = '1'
REPEAT = '1'
BEEP = '0'
REPEAT = '0'
BEEP = '1'
BPVOL[4:0]
FREQ[3:0]
4.4.5Tone Control
Shelving filters are used to implement bass and treble (boost and cut) with four selectable corner frequencies. Boosting will affect peak detect and limiting when levels exceed the maximum threshold settings.
Software
Controls:
CONTINUOUS BEEP: Beep turns on at a configurable frequency (FREQ) and volume (BPVOL) and remains
on until REPEAT is cleared.
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
REPEAT is cleared.
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.
ONTIME[3:0]OFFTIME[2:0]
Figure 15. Beep Configuration Options
“Tone Control (Address 15h)” on page 65.
...
4.4.6Limiter
When enabled, the limiter monitors the digital input signal before th e DAC modulator, detects when leve ls
exceed the maximum threshold settings and lowers the AOUT volume at a programmable attack rate below the maximum threshold. When the input signal level falls below the maximum threshold, the AOUT
volume returns to its original level set in the Volume Control register at a programmable release rate. Attack and release rates are affected by the DAC soft ramp/zero cross settings and sample rate, Fs. Limiter
soft ramp and zero cross dependency may be independently enabled/disabled.
Recommended settings: Best limiting performance may be realized with the fastest attack and slowest
release setting with soft ramp enabled in th e control registe rs. The “cushio n” bits allow the user to set a
threshold slightly below the maximum threshold for hysteresis control - this cushions the sound as the limiter attacks and releases.
Note:
1. When the Limiter is enab led, the AOUT Volume is automatically controlled and should not be adjusted
manually. Alternative volume control may be realized using the PCMMIXx_VOL[6:0] bits.
2. The Limiter maintains the output signal between the CUSH and MAX thresholds. As the digita l input
signal level changes, the level-controlled output may not always be the same but will always fall within
the thresholds.
Software
Controls:
“Limiter Release Rate Register (Address 1Ah)” on page 69, “Limiter Attack Rate Register (Address
1Bh)” on page 70, “DAC Control (Address 09h)” on page 58
36DS679F1
Input
MAX[2:0]
Limiter
Output
(after Limiter)
MAX[2:0]
Volume
ATTACK/RELEASE SOUND
CUSHION
CS42L51
AOUTx_VOL[7:0] volume
control should NOT be
adjusted manually when
Limiter is enabled.
CUSH[2:0]
Figure 16. Peak Detect & Limiter
4.4.7Line-Level Outputs and Filtering
The CODEC contains on-chip buffer amplifiers capable of producing line level sing le-ended outputs on
AOUTA and AOUTB. These amplifiers are ground centere d and do not have any DC of fset. A load stabilizer circuit, shown in the “Typical Connection Diagram (Software Mode)” on page 10 and the “Typical
Connection Diagram (Hardware Mode)” on page 11, is required on the analog outputs. This allows the
DAC amplifiers to drive line or headphone outputs.
Also shown in the Typical Connection diagrams is the r ecommended passive output filter to support high-
er impedances such as those found on the inputs to operational amplifiers. “Rext”, shown in the typical
connection diagrams, is the input impedanc e of the re ce ivin g de vice .
The invert and digital gain controls may be used to provide phase and/or amplitude compensation for an
external filter.
The delta-sigma conversion process produces high fr equency noise beyond the audio passband, most of
which is removed by the on-chip analog filters. The remaining out-of-band noise can be attenu ated using
an off-chip low pass filter.
Software
Controls:
“DAC Output Control (Address 08h)” on page 57, “AOUTx Volume Control: AOUTA (Address 16h)
& AOUTB (Address 17h)” on page 66.
RRATE[5:0]ARATE[5:0]
DS679F137
4.4.8On-Chip Charge Pump
An on-chip charge pump derives a negative supply voltage from the VA_HP supply. This provides dual
rail supplies allowing a full-scale output swing centered around ground and eliminates the need for large,
DC-blocking capacitors. Added benefits include greater pop suppression and improved low frequency
(bass) response.
drop on the VA_HP supply will directly impact the derived negative voltage on the charge pump supply,
VSS_HP, and may result in clipping.
The FLYN and FLYP pins connect to internal switches that char ges and discharges the externa l capacitor
attached, at a default switching frequency. This frequency may be adjusted in the control port registers.
Increasing the charge-pumping capacitor will slightly decease the pumping frequency. The capacitor connected to VSS_HP acts as a charge reservoir for the negative supply as well as a filter for the ripple induced by the charge pump. Increasing this capacitor will decrease the ripple on VSS_HP. Refer to the
typical connection diagrams in Figure 1 on page 10 or Figure 2 on page 11 for the recommended capacitor
values for the charge pump circuitry.
Note: Series resistance in the path of the power supplies must be avoided. Any voltage
CS42L51
Software
Controls:
4.5Serial Port Clocking
The CODEC serial audio interface port operates either as a slave or master. It accepts externally generated
clocks in slave mode and will generate synchronous clocks derived from an input master clock (MCLK) in
master mode.
The frequency of the MCLK must be an integer multiple of, and synchronous with, the system sample rate,
Fs. The LRCK frequency is equal to Fs, the frequency at which audi o samples for each cha nnel are clocked
into or out of the device.
The SPEED and MCLKDIV2 software control bits or the SDOUT/(M/S
pins, configure the device to generate the proper clocks in Master Mode and receive the proper clocks in
Slave Mode. The value on the SDOUT pin is latched immediately after powering up in Hardware Mode.
Software
Control:
Hardware
Control:
“Charge Pump Frequency (Address 21h)” on page 74.
) and MCLKDIV2 stand-alone control
“MIC Power Control & Speed Control (Address 03h)” on page 50, “DAC Control
(Address 09h)” on page 58.
PinSettingSelection
“SDOUT, M/S” pin 29
“MCLKDIV2” pin 2
47 kΩ Pull-down
47 kΩ Pull-up
LO
HI
Slave
Master
No Divide
MCLK is divided by 2 prior
to all internal circuitry.
38DS679F1
4.5.1Slave
LRCK and SCLK are inputs in Slave Mode. The speed of the CODEC is automatically determined based
on the input MCLK/LRCK ratio when the Auto-Detect function is enabled. Certain input clock ratios will
then require an internal divide-by-two of MCLK* using either the MCLKDIV2 bit or the MCLKDIV2 standalone control pin.
Additional clock ratios are allowed when the Auto-Detect function is disabled; but the appropriate speed
mode must be selected using the SPEED[1:0] bits.
Auto-DetectQSMHSMSSMDSM
Disabled
(Software
Mode only)
Enabled
*MCLKDIV2 must be enabled.
4.5.2Master
LRCK and SCLK are internally derived from the internal MCLK (after the divide, if MCLKDIV2 is enabled).
In Hardware Mode the CODEC operates in single-speed only. In Software Mode, the CODEC operates
in either quarter-, half-, single- or double-speed depending on the setting of the SPEED[1:0] bits.
The serial port may be placed on a clock/data bus that allows multiple masters for the serial port I/O without the need for external buffers. The 3ST_SP bit places the in ternal buffers for these I/O in a high-impedance state, allowing another device to transmit serial port data without bus contention.
CS42L51
CS42L51
Transmitting Device #1
3ST_SP
Figure 18. Tri-State Serial Port
4.5.4Quarter- and Half-Speed Mode
Quarter-Speed Mode (QSM) and Half-Speed Mode (HSM) allow lower sample rates while maintaining a
relatively flat noise floor in the typical audio band of 20 Hz - 20 kHz. Single-Speed Mode (SSM) will allow
lower frequency sample rates; however, the DAC's noise floor, that normally rises out-of-band, will scale
with the lower sample rate and begin to rise within the audio band. QSM and HSM corrects for most of
this scaling, effectively increasing the dynamic range of the CODEC at lower sample rates, relative to
SSM.
4.6Digital Interface Formats
Transmitting Device #2
SDOUT
SCLK/LRCK
Receiving Device
The serial port operates in standard I²S, Left-Ju stified or Rig ht-Justified (DAC only) digital interface formats
with varying bit depths from 16 to 24. Data is clocked out of the ADC or into the DAC on the rising edge of
SCLK. Figures 19-21 illustrate the general structure of each format. Refer to “Switching Specifications - Se-
rial Port” on page 20 for exact timing relationship between clocks and data.
Software
Control:
“Interface Control (Address 04h)” on page 52.
PinSettingSelection
Hardware
Control:
“I²S/LJ” pin 3
LO
HI
Left-Justified Interface
I²S Interface
LRCK
SCLK
SDIN
MSBLSB
40DS679F1
Left ChannelRight Channel
MSB
AOUTA / AINxA
Figure 19. I²S Format
AOUTB / AINxB
LSB
MSB
CS42L51
LRCK
SCLK
SDIN
LRCK
SCLK
SDIN
MSBLSB
4.7Initialization
The initialization and Power-Down sequence flowchart is shown in Figure 22 on page 42. The CODEC enters a Power-Down state upon initial power-up. The inter polation and decima tion filters, de lta-sigma mo dulators and control port registers are reset. The internal voltage reference, multi-bit DAC and ADC and
switched-capacitor low-pass filters are powered down.
The device will remain in the Power-Down state until the RESET
cessible once RESET
in “Softwar e Mode” on page 43. If a valid write sequence to the control port is not made within appro ximately
10 ms, the CODEC will enter Hardware Mode.
Left ChannelRight Channel
MSB
AOUTA / AINxA
Figure 20. Left-Justified Format
Left ChannelRight Chan nel
MSBLS B
AOUTA
Figure 21. Right-Justified Format (DAC only)
AOUTB / AINxB
AOUTB
MSB
LSB
pin is brought high. The control port is ac-
is high and the desired register settings can be loaded per the interface descrip tions
MSB
LSB
Once MCLK is valid, the quiescent voltage, VQ, and the internal voltage references, DAC_FILT+ and
ADC_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 state. Once LRCK is valid, MCLK occurrences are counted over one LRCK
period t o d etermine th e M CLK/LRCK fr equency ratio and normal operation begins.
4.8Recommended Power-Up Sequence
1. Hold RESET low until the power supplies are stable.
2. Bring RESET
3. For Software Mode operation, set the PDN bit to ‘1’b in under 10 ms. This will place the device in “standby”.
4. Load the desired register settings while keeping the PDN bit set to ‘1’b.
5. Start MCLK to the appropriate frequency, as discussed in Section 4.5.
6. Set the PDN bit to ‘0’b.
7. Apply LRCK,SCLK and SDIN for normal operation to begin.
8. Bring RESET
prevent power glitch related issues.
high. After approximately 10 ms, the device will enter Hard wa re Mod e .
low if the analog or digital supplies drop below the recommended operating condition to
DS679F141
4.9Recommended Power-Down Sequence
To minimize audible pops when turning off or placing the CODEC in standby,
1. Mute the DAC’s and ADC’s.
2. Set the PDN bit in the power control register to ‘1’b. The CODEC will not power down until it reaches a
fully muted sate. Do not remove MCLK until after the part has fully muted. Note that it may be necessary
to disable the soft ramp and/or zero cross volume transitions to achieve faster muting/power down.
3. Bring RESET
low.
No Power
1. No audio signal
generated.
CS42L51
Power Off Transition
1. Audible pops.
Reset Transition
1. Pops suppressed.
ERROR: Power removed
Off Mode (Power Applied)
1. No audio signal generated.
2. Control Port Registers reset
to default.
RESET = Low?
Control Port
Control Port Valid
No
Write Seq. within
Hardware Mode
Minimal feature
set support.
Yes
No
Active
Yes
10 ms?
Software Mode
Registers setup to
desired settings.
ERROR: MCLK/LRCK ratio change
RESET = Low
PDN bit = '1'b?
MCLK Applied?
20 ms delay
Charge Caps
1. VQ Charged to
quiescent voltage.
2. Filtx+ Charged.
ADC Initialization
2048 internal
MCLK cycle delay
Digital/Analog
Output Muted
Sub-Clocks Applied
1. LRCK valid.
2. SCLK valid.
3. Audio samples
processed.
No
MCLK/LRCK
Ratio?
Audio signal generated per control port or stand-
Normal Operation
alone settings.
Yes
No
No
Valid
Valid
Yes
DAC Initialization
50 ms delay
Charge Pump
Powered Up
20 µs delay
Headphone Amp
Powered Up
PDN bit set to '1'b
(software mode only)
Standby Mode
1. No audio signal generated.
2. Control Port Registers retain
settings.
Headphone Amp
Powered Down
20 µs delay (DAC
only)
Stand-By
Transition
1. Pops suppressed.
ERROR: MCLK removed
Analog Output Freeze
1. Aout bias = last audio sample.
2. DAC Modulators stop operation.
3. Audible pops.
Figure 22. Initialization Flowchart
42DS679F1
4.10Software Mode
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 asynchronous with r espect
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 in two modes: SPI and I²C, with the CODEC acting as a slave device. Software
Mode is selected if there is a high-to-low transition on the AD0/CS
high. I²C Mode is selected by connecting the AD0/CS
manently selecting the desired AD0 bit address state.
4.10.1SPI Control
In Software Mode, CS is the CS42L51 ch ip-select sign al, CCLK is the contro l port bit clock (input into the
CS42L51 from the microcontroller), CDIN is the input data line from the microcontroller. Data is clocked
in on the rising edge of CCLK. The CODEC will only support write operations. Read request will be ignored.
CS42L51
pin after the RESET pin has been brought
pin through a resistor to VL o r DGND, thereby pe r-
Figure 23 shows the operation of the control port in Software Mode. To write to a register, bring CS
The first seven bits on CDIN form the chip address and must be 1001010. The eighth bit is a read/write
indicator (R/W
which is set to the address of the register that is to be updated. The next eight bits are the data which will
be placed into the register designated by the MAP.
There is MAP auto-increment capability, enabled by the INCR bit in the MAP register. If INCR is a zero,
the MAP will stay constant for successive read or writes. If INCR is set to a 1, the MAP will auto-increment
after each byte is read or written, allowing block reads or writes of successive registers.
CS
0 1 2 38 91216 1710 1113 14 15
CCLK
CDIN
4.10.2I²C Control
In I²C Mode, SDA is a bidirectional data line. Data is clocked into and out of the part by the clock, SCL.
There is no CS
through a resistor to VL or DGND as desired. The state of the pin is sensed while the
reset.
), which should be low to write. The next eight bits form the Memory Addr ess Pointer (MAP),
4 5 6 7
CHIP ADDRESS (WRITE)MAP BYTEDATA
1 0 0 1 0 1 0 0
Figure 23. Control Port Timing in SPI Mode
INCR 6 5 4 3 2 1 0 7 6 1 0
DATA +n
7 6 1 0
pin. Pin AD0 forms the least significant bit of the chip address and should be connected
CS42L51 is being
low.
The signal timings for a read and write cycle are shown in Figure 24 and Figure 25. A Start condition is
defined as a falling transition of SDA while the clock is high. A Stop condition is a rising transition while
the clock is high. All other transitions of SDA occur while the clock is low. The first byte sent to the
CS42L51
after a Start condition consists of a 7-bit chip address field and a R/W bit (high for a read, low for a write).
The upper 6 bits of the 7-bit address field are fixed at 100101. To communicate with a
address field, which is the first byte sent to the
the AD0 pin. The eighth bit of the address is the R/W
CS42L51, should match 100101 followed by the setting of
bit. If the operation is a write, the next byte is the
CS42L51, the chip
Memory Address Pointer (MAP) which selects the register to be read or written. If the op eration is a read,
DS679F143
CS42L51
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
the microcontroller after each transmitted byte.
CS42L51 after each input byte is read and is input to the CS42L51 from
SCL
SDA
0 1 2 38 91216 17 18 1910 1113 14 1527 28
4 5 6 724 25
26
SCL
DATA +n
SDA
CHIP ADDRESS (WRITE)MAP BYTEDATA
1 0 0 1 0 1 AD0 0
START
INCR 6 5 4 3 2 1 07 6 1 07 6 1 07 6 1 0
ACK
DATA +1
Figure 24. Control Port Timing, I²C Write
2 310 1117 18 1925
CHIP ADDRESS (WRITE)
1 0 0 1 0 1 AD0 0
START
MAP BYTE
INCR 6 5 4 3 2 1 0
ACK
168 912 13 14 154 5 6 7 0 120 21 22 23 24
STOP
ACK
START
CHIP ADDRESS (READ)
1 0 0 1 0 1 AD0 1
26 27 28
DATA
7 07 07 0
ACK
DATA +1
ACK
DATA + n
Figure 25. Control Port Timing, I²C Read
Since the read operation cannot set the MAP, an aborted write operation is used as a preamble. As shown
in Figure 25, 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.
44DS679F1
4.10.3Memory 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.
4.10.3.1 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 and SPI writes. 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.
CS42L51
DS679F145
CS42L51
5. REGISTER QUICK REFERENCE
Software mode register defaults are as shown. “Reserved” registers must maintain their default state.
All registers are read/write except for the ch ip I.D. and Revision Register and In terrupt Status Register which are
read only. See the following bit definition tables for bit assignment information. The default state of each bit after a
power-up sequence or reset is listed in each bit description.
All “Reserved” registers must maintain their default state.
Note:Certain functions are only available when the “Signal Processing Engine to DAC” option is selected using
the DATA_SEL[1:0] bits, as described in section “DAC Data Selection (DATA_SEL[1:0])” on page 58.
6.1Chip I.D. and Revision Register (Address 01h) (Read Only)
1. To activate the power-down sequence fo r individual channels (A or B,) both channels must first be po wered down either by enabling the PDN bit or by enabling the power-down bits for both channels. Enabling the power-down bit on an individual channel basis after the CODEC has fully powered up will
mute the selected channel without achieving any po we r sav ing s.
Recommended channel power-down sequence: 1.) Ena ble the PDN bit, 2.) enable power-do wn for the se-
lect channels, 3.) disable the PDN bit.
Power Down DAC X (PDN_DACX)
Default: 0
0 - Disable
1 - Enable
Function:
DAC channel x will either enter a power-down or muted state when this bit is enabled. See Note 1 above.
DS679F149
CS42L51
Power Down PGA X (PDN_PGAX)
Default: 0
0 - Disable
1 - Enable
Function:
PGA channel x will either enter a power-down or muted state when this bit is enabled. See Power Control
1 (Address 02h) on page 49 above.
This bit is used in conjunction with AINx_MUX bits to determine the analog input path to the ADC. Refer to
“ADCX Input Select Bits (AINX_MUX[1:0])” on page 56 for the required settings.
Power Down ADC X (PDN_ADCX)
Default: 0
0 - Disable
1 - Enable
Function:
ADC channel x will either enter a power-down or muted state when this bit is enabled. See Note 1 on page
49.
Power Down (PDN)
Default: 0
0 - Disable
1 - Enable
Function:
The entire CODEC will enter a low-power state when this function is enabled. The contents of the control
port registers are retained in this mode.
6.3MIC Power Control & Speed Control (Address 03h)
1 - Enable
Function:
Enables the auto-detect circuitry for detecting the speed mode of the CODEC when o perating a s a slave.
When AUTO is enabled, the MCLK/LRCK ratio must be implemented according to Table 3 on page 39. The
SPEED[1:0] bits are ignored when this bit is enabled. Speed is determined by the MCLK/LRCK ratio.
Function:
Sets the appropriate speed mode for the CODEC in Master or Slave Mode. QSM is optimized for 8 kHz sam-
ple rate and HSM is optimized for 16 kHz sample rate . These bits are ignored when the AUTO bit is enabled
(see Auto-Detect Speed Mode (AUTO) above).
Tri-State Serial Port Interface (3ST_SP)
Default: 0
0 - Disable
1 - Enable
Function:
When enabled and the device is configured as a master, all serial port outputs are place in a high impedan ce
state. If the serial port is configured as a slave, only the SDOUT pin will be placed in a high-impedance state.
The other signals will remain as inputs.
Power Down MIC X (PDN_MICX)
Default: 1
0 - Disable
1 - Enable
Function:
When enabled, the microphone pre-amplifier for channel x will be in a power-down state.
Power Down MIC BIAS (PDN_MICBIAS)
Default: 1
0 - Disable
1 - Enable
Function:
When enabled, the microphone bias circuit will be in a power-down state.
MCLK Divide By 2 (MCLKDIV2)
Default: 0
0 - Disabled
1 - Divide by 2
Function:
Divides the input MCLK by 2 prior to all internal circuitry. This bit is ignored when the AUTO bit is disabled
in Slave Mode.
DS679F151
CS42L51
6.4Interface Control (Address 04h)
76543210
SDOUT->SDINM/S
SDOUT to SDIN Loopback (SDOUT->SDIN)
Default: 0
0 - Disabled; SDOUT internally disconnected from SDIN
1 - Enabled; SDOUT internally connected to SDIN
Function:
Internally loops the signal on the SDOUT pin to SDIN.
Master/Slave Mode (M/S)
Default: 0
0 - Slave
1 - Master
Function:
Selects either master or slave operation for the serial port.
DAC Digital Interface Format (DAC_DIF[2:0])
DAC_DIF2DAC_DIF1DAC_DIF0ADC_I²S/LJDIGMIXMICMIX
Default = 000
DAC_DIF[2:0]DescriptionFigure
000Left-Justified, up to 24-bit data20 on page 41
001I²S, up to 24-bit data 19 on page 40
010Right-Justified, 24-bit data21 on page 41
011Right-Justified, 20-bit data21 on page 41
100Right-Justified, 18-bit data21 on page 41
101Right-Justified, 16-bit data21 on page 41
110Reserved100Reserved-
Function:
Selects the digital interface format used for the data in on SDIN. The required relationship between the
Left/Right clock, serial clock and serial data is defined by the Digital Interface Format and the options are
detailed in the section “Digital Interface Formats” on page 40.
52DS679F1
CS42L51
ADC I²S or Left-Justified (ADC_I²S/LJ)
Default: 0
0 - Left-Justified
1 - I²S
Function:
Selects either the I²S or Left-Justified digital interface format for the data on SDOUT. The required relation-
ship between the Left/Right clock, serial clock and serial data is defined by the Digital Interface Format and
the options are detailed in this section .
Digital Mix (DIGMIX)
Default: 0
DIGMIXDATA_SEL[1:0]Mix Selected
0xxNo Mix: ADC to ADC serial port, SDOUT data.
00No Mix: SDIN data to ADC serial port, SDOUT data.
1
Function:
01Mix: ADC + SDIN data to ADC serial port, SDOUT data.
10No Mix: ADC to ADC serial port, SDOUT data.
11Reserved
Selects between the ADC or a di gital mix of the ADC and DAC into th e seria l port to t he SDOUT pin. Th is
mix function is affected by the data select bits DATA_SEL[1:0].
Microphone Mix (MICMIX)
Default: 0
0 - Disabled; No Mix: Left/Right Channel to ADC serial port, SDOUT.
1 - Enabled; Mix: Differential mix ((A-B)/2)to ADC serial port, SDOUT.
Function:
Selects between the ADC stereo mix or a differential mix of analog inputs A and B.
1 - Enabled
Function:
The individual PGA Volume (PGAx_VOLx) and ADC channel attenuation (ADCx_ATTx) levels as well as
the ALC A and B enable (ALC_ENx) are independ ently controlled by their respecti ve control registers when
this function is disabled. When enabled, the volume on both channels is determined by the ADCA Attenuator
Control register, or the PGAA Control register, and the ADCB Attenuator and PGAB Control registers are
ignored. The ALC enable control for channel B is controlled by the ALC A enable whe n the ADC_SNGVOL
bit is enabled and the ALC_ENB control register is ignored.
DS679F153
CS42L51
ADCx 20 dB Digital Boost (ADCx_DBOOST)
Default: 0
0 - Disabled
1 - Enabled
Function:
Applies a 20 dB digital gain to the input signal on ADC channel x, regardless of the input path.
MIC Bias Select (MICBIAS_SEL)
Default: 0
0 - MICBIAS on AIN3B/MICIN2 pin
1 - MICBIAS on AIN2B pin
Function:
Determines the output pin for the internally generated MICBIAS signal. If set to ‘0’b, the MICBIAS is output
on the AIN3B/MICIN2 pin. If set to ‘1’b, the MICBIAS is output on the AIN2B pin.
MIC Bias Level (MICBIAS_LVL[1:0])
Default: 00
00 - 0.8 x VA
01 - 0.7 x VA
10 - 0.6 x VA
11 - 0.5 x VA
Function:
Determines the output voltage level of the MICBIAS output.
MIC X Preamplifier Boost (MICX_BOOST)
Default: 0
0 - +16 d B Gain
1 - +32 d B Gain
Function:
Determines the amount of gain applied to the microphone preamplifier for channel x.
When this bit is set, the internal high-pass filter will be enabled for ADCx. When set to ‘0’, the high-pass filter
will be disabled. For DC measurements, this bit must be cleared to ‘0’. See “ADC Digital Filter Characteris-
tics” on page 15.
54DS679F1
CS42L51
ADCX High-Pass Filter Freeze (ADCX_HPFRZ)
Default: 0
0 - Continuous DC Subtraction
1 - Frozen DC Subtraction
Function:
The high-pass filter works by continuously subtracting a measure of the DC offset from the output of the
decimation filter. If the ADCx_HPFRZ bit is taken high during normal operation, the current value of the DC
offset is frozen, and this DC offset will continue to be subtracted from the conversion result. For DC measurements, this bit must be set to ‘1’. See “ADC Digital Filter Characteristics” on page 15.
Soft Ramp CHX Control (SOFTX)
Default: 0
0 - Disabled
1 - Enabled
Function:
Soft Ramp allows level changes to be implemented via an incremental ramp. ADCx_ATT[7:0] digital atten-
uation changes are ramped from the current level to the new level at a rate of 0.125 dB per LRCK period.
PGAx_VOL[4:0] gain changes are ramped in 0.5 dB steps every 16 LRCK periods.
Soft Ramp & Zero Cross Enabled
When used in conjunction with the ZCROSSx bit, the PGAx_VOL[4:0] gain changes will occur in 0.5 dB
steps and be implemented on a signal zero crossing.
Zero Cross CHX Control (ZCROSSX)
Default: 0
0 - Disabled
1 - Enabled
Function:
Zero Cross Enable dictates that signal level changes will occur on a signal zero crossing to minimize audible
artifacts. The requested level change will occur after a timeout period of 1024 sample periods (approximately 10.7 ms at 48 kHz sample rate) if the signal does not encounter a zero crossing. The zero cross function
is independently monitored and implemented for each channel.
Soft Ramp & Zero Cross Enabled
When used in conjunction with the SOFTx bit, the PGAx_VOL[4:0] gain changes will occur in 0.5 dB steps
and be implemented on a signal zero crossing.
The ADC Attenuator ADCx_ATT[7:0] is not affected by the ZCROSSx bit.
SOFTxZCROSSxAnalog PGA Volume
Digital Attenuator (ADCx_ATT[7:0])
(PGAx_VOL[4:0])
00
01
10
11
Volume changes immediately.Volume changes immediately.
Volume changes at next zero cross time.Volume changes immediately.
Volume changes in 0.5 dB steps.Change volume in 0.125 dB steps.
Volume changes in 0.5 dB steps at every
Function:
These bits select the gain multiplier for the headphone/line outputs. See “Line Output Voltage Characteri s-
tics” on page 18 and “Headphone Output Power Characteristics” on page 19.
DAC_
SNGVOL
INV_PCMBINV_PCMADACB_MUTE DACA_MUTE
DAC Single Volume Control (DAC_SNGVOL)
Default: 0
Function:
The individual channel volume levels are independently controlled by their respective Volume Control reg-
isters when this function is disabled. When enabled, the volume on all channels is determined by the AOUTA Volume Control register and the AOUTB Volume Control register is ignored.
PCMX Invert Signal Polarity (INV_PCMX)
Default: 0
0 - Disabled
1 - Enabled
Function:
When enabled, this bit will invert the signal polarity of the PCM x channel.
DACX Channel Mute (DACX_MUTE)
Default: 0
0 - Disabled
1 - Enabled
Function:
The output of channel x DAC will mute when enabled. The muting function is affected by the DACx Soft and
01 - Signal Processing Engine to DAC
10 - ADC Serial Port to DAC
11 - Reserved
Function:
Selects the digital signal source for the DAC.
Processing Engine to DAC” option is selected using these bits.
Freeze Controls (FREEZE)
Default: 0
Function:
This function will freeze the previous settings of, and allow modifications to be made to all control port reg-
isters without the changes taking effect until the FREEZE is disabled. To have multiple chang es in the control port registers take effect simultaneously, enable the FREEZE bit, make all register changes, then
disable the FREEZE bit.
DAC De-Emphasis Control (DEEMPH)
Default: 0
0 - No De-Emphasis
1 - De-Emphasis Enabled
Function:
Note: The DATA_SEL[1:0] bits in reg09h must be set to ‘01’b to enable function control.
Enables the digital filter to apply the standard 15µs/50µs digital de-emphasis filter response for a sample
rate of 44.1 kHz.
Note: Certain functions are only available when the “Signal
Analog Output Auto MUTE (AMUTE)
Default: 0
0 - Auto Mute Disabled
1 - Auto Mute Enabled
Function:
Enables (or disables) Automatic Mute of the analog outputs after 8192 “0” samples on each digital input
channel.
58DS679F1
CS42L51
DAC Soft Ramp and Zero Cross Control (DAC_SZC[1:0])
Default = 10
00 - Immediate Change
01 - Zero Cross
10 - Soft Ramp
11 - Soft Ramp on Zero Crossings
Function:
Note: The DATA_SEL[1:0] bits in reg09h must be set to ‘01’b to enable function control
Immediate Change
When Immediate Change is selected all volume-level changes will take effect immediately in one step.
Zero Cross
This setting dictates that signal-level changes, either by gain changes, attenuation changes or muting, will
occur on a signal zero crossing to minimize audible artifacts. The requested level change will occur after a
timeout period between 1024 and 2048 sample periods (21.3 ms to 42.7 ms at 48 kHz sample rate) if the
signal does not encounter a zero crossing. The zero cross function is independently monitored and implemented for each channel.
Soft Ramp
Note: The LIM_SRDIS bit is ignored.
Soft Ramp allows level changes, either by gain changes, attenuati on changes or muting, to be implemente d
by incrementally ramping, in 1/8 dB steps, from the current level to the new level at a rate of 0.5 dB per 4
left/right clock periods.
Soft Ramp on Zero Crossing
This setting dictates that signal-level changes, either by gain changes, attenuation changes or muting, will
occur in 1/8 dB steps and be implemented on a signal zero crossing. The 1/8 dB level change will occur
after a timeout period between 512 and 1024 sample periods (10.7 ms to 21.3 ms at 48 kHz sample rate) if
the signal does not encounter a zero crossing. The zero cross fu nctio n is indepe nde ntly mon ito red an d implemented for each channel.
Overrides the SOFTx bit setting for the ADC. When this bit is set, the ALC attack rate in the PGA will not be
dictated by the soft ramp setting. ALC volume-level changes will take effect in one step.
DS679F159
CS42L51
ALCX Zero Cross Disable (ALCX_ZCDIS)
Default: 0
0 - Off
1 - On
Function:
Overrides the ZCROSSx bit setting for the ADC. When this bit is set, the ALC attack rate in the PGA will not
be dictated by the zero cross setting. ALC volume-level changes will take effect immediately in one step.
PGA X Gain Control (PGAX_VOL[4:0])
Default: 00000
Binary CodeVolume Setting
11000+12 dB
······
01010+5 dB
······
000000 dB
11111-0.5 dB
11110-1 dB
······
11001-3 dB
11010-3 dB
Function:
The PGAx Gain Control register allows independent setting of the sig nal levels in 0.5 dB increments as dic-
tated by the ADCx Soft and Zero Cross bits (SOFTx & ZCROSSx) from +12 dB to -3 dB. Gain settings are
decoded as shown in the table above. The gain changes are implemented as dictated by the ALCX Soft &
Zero Cross bits (ALCX_SZC). Levels are decoded as described in the table above.
Note:When the ALC is enabled, the PGA is automatic ally controlled and should no t be adjusted manu-
Note: The DATA_SEL[1:0] bits in reg09h must be set to ‘01’b to enable function control in this register.
Beep Off Time (OFFTIME[2:0])
Default: 000
OFFTIME[2:0]Off Time
Fs = 12, 24, 48 or
96 kHz
0001.23 s
0012.58 s
0103.90 s
0115.20 s
1006.60 s
1018.05 s
DS679F163
CS42L51
OFFTIME[2:0]Off Time
Fs = 12, 24, 48 or
96 kHz
1109.35 s
11110.80 s
Function:
The off-duration of the beep signal can be adjusted from approximately 75 ms to 680 ms. The off-duration
will scale inversely with sample rate, Fs, but is fixed at the nominal Fs within each speed mode. Refer to
Figure 15 on page 36 for single-, multiple- and continuous-beep configurations using the REPEAT and
BEEP bits.
Beep Volume (BPVOL[4:0])
Default: 00000
Binary CodeVolume Setting
00110+12.0 dB
······
000000 dB
11111-2 dB
11110-4 dB
······
00111-50 dB
Function:
The level of the beep into the output mixer can be adjusted in 2.0 dB increments from +12 dB to -50 dB.
Refer to Figure 15 on page 36 for single-, multiple- and continu ous-beep co nfigurations using the REPEAT
and BEEP bits. Levels are decoded as described in the table above.
6.16Beep Configuration & Tone Configuration (Address 14h)
Note: The DATA_SEL[1:0] bits in reg09h must be set to ‘01’b to enable function control in this register.
Repeat Beep (REPEAT)
Default: 0
0 - Disabled
1 - Enabled
Function:
This bit is used in conjunction with the BEEP bit to mix a continuous or periodic beep with the analog output.
Refer to Figure 15 on page 36 for a description of each configuration option.
Beep (BEEP)
Default: 0
0 - Disabled
1 - Enabled
Function:
64DS679F1
CS42L51
This bit is used in conjunction with the REPEAT bit to mix a continuous or periodic beep with the analog
output.
remain ON for the maximum ONTIME duration. Refer to Figu re 15 on page 36 for a d escription of each configuration option.
Treble Corner Frequency (TREB_CF[1:0])
Default: 00
00 - 5 kHz
01 - 7 kHz
10 - 10 kHz
11 - 15 kHz
Function:
The treble corner frequency is user selectable as shown above.
Note: The DATA_SEL[1:0] bits in reg09h must be set to ‘01’b to enable function control in this register.
AOUTX Volume Control (AOUTX_VOL[7:0])
Default = 00h
Binary CodeVolume Setting
0001 1000+12.0 dB
······
0000 00000 dB
1111 1111-0.5 dB
1111 1110-1.0 dB
······
0011 0100-102 dB
······
0001 1001-102 dB
Function:
The analog output levels can be adjusted in 0.5 dB increments from +12 to - 102 dB as dictated by the DAC
Soft and Zero Cross bits (DACX_SZC[1:0]). Levels are decoded in unsigned binary as described in the table
above.
Note:When the limiter is enabled, the AOUT Volume is automatically controlled and should not be ad-
justed manually. Alternative volume control may be achieved using the PCMMIXx_VOL[6:0] bits.
66DS679F1
CS42L51
6.19PCM Channel Mixer (Address 18h)
76543210
PCMA1PCMA0PCMB1PCMB0ADCA1ADCA0ADCB1ADCB0
Note: The DATA_SEL[1:0] bits in reg09h must be set to ‘01’b to enable function control in this register.
Channel Mixer (PCMx[1:0] & ADCx[1:0])
Default: 00
PCMA[1:0]
and/or
AOUTA PCMB[1:0]
ADCB[1:0]
and/or
AOUTB
ADCA[1:0]
00L00R
0101
1010
11R11L
LR+
------------
2
LR+
------------
2
Function:
Implements mono mixes of the left and right channels as well as a left/right channel swap.
6.20Limiter Threshold SZC Disable (Address 19h)
76543210
MAX2MAX1MAX0CUSH2CUSH1CUSH0LIM_SRDISLIM_ZCDIS
Note: The DATA_SEL[1:0] bits in reg09h must be set to ‘01’b to enable function control in this register.
Maximum Threshold (MAX[2:0])
Default: 000
MAX[2:0] Threshold
Setting
(dB)
0000
001-3
010-6
011-9
101-12
101-18
110-24
111-30
Function:
Sets the maximum level, below full scale, at which to limit and attenuate the output signal at the attack rate.
Bass, Treble and digital gain settings that boost the signal beyond the maximum threshold may trigger an
attack.
DS679F167
CS42L51
Cushion Threshold (CUSH[2:0])
Default: 000
CUSH[2:0] Threshold
Setting
(dB)
0000
001-3
010-6
011-9
101-12
101-18
110-24
111-30
Function:
Sets a cushion level below full scale. This setting is usually set slightly below the maximum (MAX[2:0])
threshold. The Limiter uses this cushion as a hysteresis point for the input signal as it maintains the signal
below the maximum as well as below the cushion setting. This provides a more natural sound as the limiter
attacks and releases.
Limiter Soft Ramp Disable (LIM_SRDIS)
Default: 0
0 - Off
1 - On
Function:
Overrides the DAC_SZC setting. When this bit is set, the Limiter attack and release rate will not be dictated
by the soft ramp setting.
Note: This bit is ignored when the zero-cross function is enabled (i.e. when
DAC_SZC[1:0] = ‘01’b or ‘11’b.)
Limiter Zero Cross Disable (LIM_ZCDIS)
Default: 0
0 - Off
1 - On
Function:
Overrides the DAC_SZC setting. When this bit is set, the Limiter attack and release rate will not be dictated
Note: The DATA_SEL[1:0] bits in reg09h must be set to ‘01’b to enable function control in this register.
Peak Detect and Limiter Enable (LIMIT_EN)
Default: 0
0 - Disabled
1 - Enabled
Function:
Limits the maximum signal amplitude to prevent clipping when this function is enabled. Peak Signal Limiting
is performed by digital attenu ation.
controlled and should not be adjusted manually. Alternative volume control may be realized using the
PCMMIXx_VOL[6:0] bits.
Peak Signal Limit All Channels (LIMIT_ALL)
Default: 1
0 - Individual Channel
1 - Both channel A & B
Note: When the limiter is enabled, the AOUT Volume is automatically
Function:
When set to 0, the peak signal limiter will limit the maximum signal amplitude to prevent clipping on the spe-
cific channel indicating clipping. The other channels will not be affected.
When set to 1, the peak signal limiter will limit the maximum signal amplitude to prevent clipping on both
channels in response to any single channel indicating clipping.
Limiter RELEASE Rate (RRATE[5:0])
Default: 111111
Binary CodeRelease Time
000000Fastest Release
······
111111Slowest Release
Function:
Sets the rate at which the limiter releases the digital attenuation from levels below the minimum setting in
the limiter threshold register, and returns the analog output level to the AOUTx_VOL[7:0] setting.
The limiter release rate is user selectable but is also a function of the sampling frequency, Fs, and the
DAC_SZC setting unless the disable bit is enabled.
Function:
Sets the rate at which the ALC releases the PGA & digital attenuation from levels below the minimum setting
in the ALC threshold register, and returns the input level to the PGA_VOL[4:0] & ADCx_ATT[7:0] setting.
The ALC release rate is user selectable, but is also a function of th e sampling frequency, Fs, and the SOFTx
& ZCROSS bit settings unless the disable bit for each function is enabled.
6.25ALC Threshold (Address 1Eh)
76543210
MAX2MAX1MAX0MIN2MIN1MIN0ReservedReserved
Maximum Threshold (MAX[2:0])
Default: 000
MAX[2:0] Threshold
Setting
(dB)
0000
001-3
010-6
011-9
100-12
101-18
110-24
111-30
Function:
Sets the maximum level, relative to full scale, at which to limit and attenuate the input signal at the attack
rate.
Minimum Threshold (MIN[2:0])
Default: 000
Threshold
MIN[2:0]
Setting
(dB)
0000
001-3
010-6
DS679F171
CS42L51
Threshold
MIN[2:0]
011-9
100-12
101-18
110-24
111-30
Function:
Sets the minimum level at which to disengage the ALC’s attenuation or amplify the input signal at a rate se t
in the release rate register until levels again reach this minimum threshold. The ALC uses this minimum as
a hysteresis point for the input signal a s it maintains the signal below the maximum as well as below the
minimum setting. This provides a more natural sound as the ALC attacks and releases.
1 - Enabled
Function:
Gangs the noise gate function for channel A and B. When enable d, both channels must fall below the thresh-
old setting for the noise gate attenuation to take effect.
Noise Gate Enable (NG_EN)
Default: 0
0 - Disabled
1 - Enabled
Function:
Enables the noise gate. Maximum attenuation is relative to all gain settings applied.
Noise Gate Boost (NG_BOOST) and Threshold (THRESH[3:0])
Default: 000
THRESH[2:0]
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
1 11Reserved-64 dB
Minimum Setting
(NG_BOOST = ‘0’b)
Minimum Setting
(NG_BOOST = ‘1’b)
72DS679F1
CS42L51
Function:
Sets the threshold level of the noise gate. Input signals below the threshold level will be attenuated to -96
dB. NG_BOOST = ‘1’b adds 30 dB to the threshold settings.
Noise Gate Delay Timing (NGDELAY[1:0])
Default: 00
00 - 50 ms
01 - 100 ms
10 - 150 ms
11 - 200 ms
Function:
Sets the delay time before the noise gate attacks. Noise gate attenuation is dictated by the SOFTx &
ZCROSS bit settings unless the disable bit for each function is enabled.
For all bits in this register, a “1” means the associated error condition has occurred at least once since 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.
Serial Port Clock Error (SP_CLK Error)
Default: 0
Function:
Indicates an invalid MCLK to LRCK ratio. See “Serial Port Clocking” on page 38 for valid clock ratios.
Note:On initial power up and application of clocks, this bit will be high as the serial port re-synchronizes.
Signal Processing Engine Overflow (SPEX_OVFL)
Default: 0
Function:
Indicates a digital overflow condition within the data path after the signal processing engine.
PCMX Overflow (PCMX_OVFL)
Default: 0
Function:
Indicates a digital overflow condition within the data path of the PCM mix.
ADC Overflow (ADCX_OVFL)
Default = 0
Function:
Indicates that there is an over-range cond ition anywhere in the CS42L51 ADC signal path of each of the
associated ADC’s.
DS679F173
CS42L51
6.28Charge Pump Frequency (Address 21h)
76543210
CHRG_FREQ3CHRG_FREQ2CHRG_FREQ1CHRG_FREQ
0
Charge Pump Frequency (CHRG_FREQ[3:0])
Default: 0101
NCHRG_FREQ[3:0]Frequency
00000
......
151111
64xFs
---------------- N2+
Function:
Alters the clocking frequency of the charge pump in 1/(N+2) fractions of the DAC oversampling rate, 128Fs,
should the switching frequency interfere with other system frequencies such as those in the AM radio band.
Note:Distortion performance may be affected.
ReservedReservedReservedReserved
74DS679F1
7. ANALOG PERFORMANCE PLOTS
7.1Headphone THD+N versus Output Power Plots
Test conditions (unless otherwise spec ified): Input test signal is a 99 7 Hz sine wave; measurement bandwidth is 10 Hz to 20 kHz; Fs = 48 kHz. Plots were taken from the CDB42L51 using an Audio Precision analyzer.
-10
-15
VA_HP = VA = 1.8 V
-20
-25
-30
-35
-40
-45
-50
d
B
-55
r
A
-60
-65
-70
-75
-80
-85
-90
-95
-100
080m10m20m30m40m50m60m70m
W
Figure 27. THD+N vs. Output Power per Channel at 1.8 V (16 Ω load)
CS42L51
G = 0.6047
G = 0.7099
G = 0.8399
G = 1.0000
G = 1.1430
Legend
NOTE: Graph shows the out-
put power per channel (i.e.
Output Power = 23 mW into
single 16 Ω and 46 mW into
stereo 16 Ω with THD+N = 75 dB).
-10
-15
VA_HP = VA = 2.5 V
-20
-25
-30
-35
-40
-45
-50
d
B
-55
r
A
-60
-65
-70
-75
-80
-85
-90
-95
-100
080m10m20m30m40m50m60m70m
Figure 28. THD+N vs. Output Power per Channel at 2.5 V (16 Ω load)
W
G = 0.6047
G = 0.7099
G = 0.8399
G = 1.0000
G = 1.1430
Legend
NOTE: Graph shows the out-
put power per channel (i.e.
Output Power = 44 mW into
single 16 Ω and 88 mW into
stereo 16 Ω with THD+N = 75 dB).
DS679F175
VA_HP = VA = 1.8
-20
-30
-35
-40
-45
-50
-55
d
B
-60
r
A
-65
-70
-75
-80
-85
-90
-95
-100
060m6m12m18m24m30m36m42m48m54m
W
CS42L51
G = 0.6047
G = 0.7099
G = 0.8399
G = 1.0000
G = 1.1430
Legend
NOTE: Graph shows the out-
put power per channel (i.e.
Output Power = 22 mW into
single 32 Ω and 44 mW into
stereo 32 Ω with THD+N = 75 dB).
Figure 29. THD+N vs. Output Power per Channel at 1.8 V (32 Ω load)
-20
VA_HP = VA = 2.5 V
-25
-30
-35
-40
-45
-50
-55
d
B
-60
r
A
-65
-70
-75
-80
-85
-90
-95
-100
060m5m10m15m20m25m30m35m40m45m50m55m
Figure 30. THD+N vs. Output Power per Channel at 2.5 V (32 Ω load)
W
G = 0.6047
G = 0.7099
G = 0.8399
G = 1.0000
G = 1.1430
Legend
NOTE: Graph shows the out-
put power per channel (i.e.
Output Power = 42 mW into
single 32 Ω and 84 mW into
stereo 32 Ω with THD+N = 75 dB).
76DS679F1
7.2Headphone Amplifier Efficiency
The architecture of the headphone amplifier is that of typical class AB amplifiers. Test conditions (unless
otherwise specified): Input test signal is a 997 Hz sine wave; Power Consumption Mode 6 - Stereo Playback
w/16 Ω load. HP_GAIN = 1.1430. Best efficiency is realized when the amplifier outputs maximum power.
VA_HP = VA = 1.8 V
CS42L51
Figure 31. Power Dissipation vs. Output Power into Stereo 16 Ω
VA_HP = VA = 1.8 V
Figure 32. Power Dissipation vs. Output Power into Stereo 16 Ω (Log Detail)
DS679F177
7.3ADC_FILT+ Capacitor Effects on THD+N
60
The value of the capacitor on the ADC_FILT+ pin, 16, affects the low frequency total harmonic distortion +
noise (THD+N) performance of the ADC. Larger capacitor values yield significant improvement in THD+N
at low frequencies. Figure 33 shows the THD+N versus frequency for the ADC analog input. Plots were taken from the CDB42L51 using an Audio Precision analyzer.
-
CS42L51
-64
-68
-72
-76
d
B
-80
F
S
-84
-88
-92
-96
-100
2020k501002005001k2k5k10k
Hz
Figure 33. ADC THD+N vs. Frequency w/Capacitor Effects
As with any high-resolution converter, the CS42L51 requires careful attention to power supply and grounding
arrangements if its potential performance is to be realized. Figure 1 on page 10 shows the recommended
power arrangements, with VA and VA_HP connected to clean supplies. VD, which powers the digital circui try, may be run from the system logic supply. Alternatively, VD may 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 capa citors sh ould be as close to the pins of the
sible. The low value ceramic capacitor should be closest to the pin and should be mounted on the same
side of the board as the
kept away from the DAC_FILT+/ADC_FILT+ and VQ pins in order to avoid unwanted coupling into the modulators. The DAC_FILT+/ADC_FILT+ an d VQ de co up lin g capa cit or s, p ar ticul ar ly the 0 .1 µF, must be positioned to minimize the electrical path from DAC_FILT+/ADC_FILT+ and AGND. The CDB42L51 e valuation
board demonstrates the optimum layout and power supply arrangements.
9.2QFN Thermal Pad
The CS42L51 is available in a compact QFN package. The under side o f the QFN packag e reveals a larg e
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 CS42L51 evaluation board demonstrates the op timum thermal pad and via configuration.
CS42L51 to minimize inductance effects . All signals, especially clocks, should be
Figure 40. DAC Transition BandFigure 41. DAC Transition Band (Detail)
82DS679F1
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
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.
CS42L51
Interchannel Gain Mismatch
The gain difference between left and right channel pairs. Units in decibels.
Gain Error
The deviation from the nominal full-scale analog output for a full-scale digital input.
Gain Drift
The change in gain value with temperature. Units in ppm/°C.
Offset Error
The deviation of the mid-scale transition (111...111 to 000...000) from the ideal. Units in mV.
1. Dimensioning and tolerance per ASME Y 14.5M-1995.
2. Dimensioning lead width applies to the plated terminal and is m easured between 0.20 mm and 0.25 mm
from the terminal tip.
THERMAL CHARACTERISTICS
ParameterSymbolMinTypMaxUnits
Junction to Ambient Thermal Impedance2 Layer Board
4 Layer Board
θ
JA
-
-
84DS679F1
52
38
-
-
°C/Watt
CS42L51
13.ORDERING INFORMATION
ProductDescriptionPackage Pb-FreeGradeTemp Range ContainerOrder #
RailCS42L51-CNZ
Tape & Reel CS42L51-CNZR
RailCS42L51-DNZ
Tape & Reel CS42L51-DNZR
CS42L51
CDB42L51
CRD42L51
Low-Power Stereo
CODEC w/HP Amp for
Portable Apps
CS42L51 Evaluation
Board
CS42L51 Reference
Design
Commercial -10 to +70° C
32L-QFNYes
Automotive -40 to +85° C
-No---CDB42L51
-No---CRD42L51
14.REFERENCES
1. Cirrus Logic, AN18: Layout and Design Rules for Data Converters and Other Mixed Signal Devices,
Version 6.0, February 1998.
2. Cirrus Logic, Techniques to Measure and Maximize the Performance of a 120 dB, 96 kHz A/D ConverterIntegrated Circuit, by Steven Harris, Steven Green and Ka Leung. Presented at the 103rd Co nvention of the
Audio Engineering Society, September 1997.
3. Cirrus Logic, A Stereo 16-bit Delta-Sigma A/D Conv erter for Digital Audio, by D.R. Welland, B.P. Del Signore, E.J. Swanson, T. Tanaka, K. Hamashita, S. Hara, K. Takasuka. Paper presented at the 85th Convention
of the Audio Engineering Society, November 1988.
4. Cirrus Logic, The Effects of Sampling Clock Jitter on Nyquist Sampling Analog-to-Digital Converters, andon Oversampling Delta Sigma ADC's, by Steven Harris. Paper presented at the 87th Convention of the Audio Engineering Society, October 1989.
5. Cirrus Logic, An 18-Bit Dual-Channel Oversampling Delta-Sigma A/D Conver ter, with 19-Bit Mono Applica-tion Example, by Clif Sanchez. Paper presented at the 87th Convention of the Audio Engineering Society,
October 1989.
6. Cirrus Logic, How to Achieve Optimum Performance from Delta-Sigma A/D and D/A Converters, by Steven
Harris. Presented at the 93rd Convention of the Audio Engineering Society, October 1992.
7. Cirrus Logic, A Fifth-Order Delta-Sigma Modula tor with 110 dB Audio Dynamic Range, by I. Fujimori, K. Hamashita and E.J. Swanson. Paper presented at the 93rd Convention of the Aud io Engineering Society, October 1992.
8. Philips Semiconductor, The I²C-Bus Specification: Version 2.1, January 2000.
http://www.semiconductors.philips.com
DS679F185
15.REVISION HISTORY
RevisionChanges
A1
A2
Initial Release
Renamed pin 14, FILT1+, to DAC_FILT+ and pin 16, FILT2+, to ADC_FILT+.
Added 1.5 µF capacitor recommendation to figu re “Typical Connection Diagram (Software Mode)” on page 10.
Removed the 0.1µF capacitors from pins DAC_FILT+, ADC_FILT+ and VQ on the figures “Typical Connection
Diagram (Software Mode)” on page 10 and “Typical Connection Diagram (Hardware Mode)” on page 11.
Added DAC Isolation specific ation to “Analog Input Characteristics (Commercial - CNZ)” on page 13 and “Analog
Input Characteristics (Automotive - DNZ)” on page 14.
Corrected specification table “Headphone Output Power Characteristics” on page 19.
Removed t
Added t
page 20.
Adjusted timing specifications t
Specifications - Serial Port” on page 20.
Added MIC Bias PSRR specification to “DC Electrical Characteristics” on page 24.
Adjusted specification table “Power Consumption” on page 25.
Removed QSM clock ratios 128, 192, 256, 384 and HSM ratios 128, 192 from Table 3 on page 39.
Modified Digital Mix description in section “Digital Mix (DIGMIX)” on page 53.
Corrected DAC Zero Cross timeout period in section “Zero Cross” on page 59.
Adjusted BEEP off time settings in section “Beep Off Time (OFFTIME[2:0])” on page 63.
Modified BEEP description in section “Beep (BEEP)” on page 64.
Adjusted the minimum settings for the “Noise Gate Boost (NG_BOOST) and Threshold (THRESH[3:0])” on
page 72.
Swapped bits PCMA_OVFL w/PCMB_OVFL and ADCA_OVFL w/ADCB_OVFL in register “Status (Address 20h)
(Read Only)” on page 73.
Corrected Charge Pump Frequency setting in section “Charge Pump Frequency (CHRG_FREQ[3:0])” on
page 74.
Added sections “Headphone THD+N versus Output Power Plots” on page 75 and “ADC_FILT+ Capacitor Effects
on THD+N” on page 78.
timing specification from table in section “Switching Specifications - Serial Port” on page 20.
d
s(SDO-SK)
and t
h(SK-SDO)
timing specification to table in section “Switching Specifications - Serial Port” on
s(SD-SK)
from 0 ns to 20 ns and th from 50 ns to 20 ns in table in section “Switching
CS42L51
86DS679F1
RevisionChanges
Adjusted the minimum voltage specification in “Specified Operating Conditions” section on page 12.
Adjusted Ambient Operating Temp. specification in “Absolute Maximum Ratings” section on page 12.
Adjusted maximum “Analog In to PGA to ADC” THD+N performance specification in “Analog Input Characteris-
tics (Commercial - CNZ)” on page 13.
Added Offset Error specification to “Analog Input Characteristics (Commercial - CNZ)” on page 13 and “Analog
Input Characteristics (Automotive - DNZ)” on page 14.
Corrected Interchannel Gain Mismatch specification in “Anal og Input Characteristics (Commercial - CNZ)” on
page 13 and “Analog Input Characteristics (Automotive - DNZ)” on page 14.
Adjusted ADC full scale input voltage specification in “Analog Input Characteristics (Commercial - CNZ)” on
page 13 and “Analog Input Characteristics (Automotive - DNZ)” on page 14.
Corrected Group Delay characteristic in table in section “ADC Digital Filter Characteristics” on page 15.
= 10kΩ” THD+N performance specification in “Analog Output Characteristics (Commer-
L
from 40 ns to 52 ns and t
d(MSB)
from 1000 ns to 3450 ns in table in section “” on page 21.
ack
PP1
F1
Adjusted maximum “R
cial - CNZ)” on page 16 and “Analog Output Characteristics (Automotive - DNZ)” on page 17.
Corrected Group Delay characteristic in table in section “Combined DAC Interpolation & on-Chip Analog FIlter
Response” on page 20.
Adjusted timing specifications t
“Switching Specifications - Serial Port” on page 20 .
Adjusted I²C timing specification t
Adjusted High-Level Input Voltage specifications VIH from 0.65VL to 0.68VL and V
table in section “Digital Interface Specifications & Characteristics” on page 24.
Adjusted the +20 dB Digital Boost block before the ALC feedback path in Figure 8 on page 28.
Modified ALC Recommended Settings in section “Automatic Level Control (ALC)” on page 32.
Modified step 2 of the “Recommended Power-Down Sequence” on page 42.
Corrected default values for ALC and Limiter Release Rates shown in “Register Quick Reference” on page 46.
Corrected default value for the DAC_SZC bits and Added AMUTE bit and description in “DAC Control (Address
09h)” on page 58.
Added section “Headphone Amplifier Efficiency” on page 77.
Corrected ADC Filter Response shown in Figures 34, 35, 36, and 37 on page 82.
Corrected ADC_SNGVOL description in “MIC Control (Address 05h)” on page 53.
Final Release
s(SDO-SK)
CS42L51
from 30 ns to 20 ns in table in section
from 0.35VL to 0.32VL in
IL
DS679F187
CS42L51
Contacting Cirrus Logic Support
For all product questions and inquiries, contact a Cirrus Logic Sales Representative.
To find the one nearest to you, go to www.cirrus.com.
IMPORTANT NOTICE
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 not ice and is pr ovided "AS IS" witho ut warr anty of any kind (express or implied). Customers are advised to obtain the latest version of relevant
information to verify, before placing orders, that information being relied on is current and complete. All products are sold subject to the terms and conditio ns of sa le
supplied at the time of order acknowledgment, including those pertaining to warranty, indemnification, and limitation of liability. No responsibility is assumed by Cirrus
for the use of this information, including use of this information as the basis for manufacture or sale of any items, or for infringement of patents or other rights of third
parties. This document is the property of Cirrus and by furnishing this information, Cirrus grants no license, express or implied under any patents, mask work rights,
copyrights, trademarks, trade secrets or other inte llectual property rig hts. Cirrus owns the copyrights associated with the information contained herein and gives consent for copies to be made of the information only for use within your organization with respect to Cirrus integrated circuits or other products of Cirrus. This consent
does not extend to other copying such as copying for general distribution, advertising or promotional purposes, or for creating any work for resale.
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 AIRCRAFT SYSTEMS, MILITARY APPLICATIONS, 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, T O FULLY INDEMNIF Y CIRRUS, ITS OF FICE RS, DI RECTORS, EMPLOYEES, DISTRI BUTORS AND
OTHER AGENTS FROM ANY AND ALL LIABILITY, INCLUDING ATTORNEYS’ FEES AND COSTS, THAT MAY RESULT FROM OR ARISE IN CONNECTION
WITH THESE USES.
Cirrus Logic, Cirrus, and the Cirrus Logic logo designs are trademarks of Cirrus Logic, Inc. All other brand and product names in this document may be trademarks
or service marks of their respective owners.
I²C is a registered trademark of Philips Semiconductor.
SPI is a trademark of Motorola, Inc.
88DS679F1
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