l 28-Pin CASON package
l 1.8 to 3.3 Volt supply
l 24-Bit conversion / 96 kHz sample rate
l 96 dB ADC/DAC dynamic range at 3 V supply
l -88/-85 dB ADC/DAC THD+N
l 19 mW playback power consumption @ 1.8 V
l Microphone or Line input amplifier with up to
32dB of gain
l 2:1 stereo mux
l Digital volume control on inputs and outputs
– 96 dB attenuation, 1 dB step size
l Digital bass and treble boost on outputs
– Selectable corner frequencies
l Dynamic range compression and limiting
l De-emphasis for 32 kHz, 44.1 kHz, and 48 kHz
l Headphone amplifier
– 26 mW power output into 16 W load @ 3.0V
supply
– -80 dB THD+N
– 34 dB analog attenuation and mute
l ATAPI mixing functions
I I
SCLSDA
Description
The CS42L50 is a highly integrated, 24-bit, 96 kHz audio
codec.
This device is based on delta-sigma modulation allowing
infinite adjustment of the sample rate between 8 kHz and
100 kHz simply by changing the master clock frequency.
The CS42L50 contains a 2:1 stereo mux, programmable
analog gain control, and digital attenuation on the analog
inputs. The output D/A converters include digital bass
and treble boost, dynamic range compression, limiting,
mixing, volume control and de-emphasis.
The CS42L50 operates from a +1.8 V to +3.3 V supply.
These features are ideal for portable MP3 and MD recorders, CD and DVD recorders, digital camcorders, and
other portable systems that require extremely low power
consumption in a minimal amount of space.
ORDERING INFORMATION
CS42L50-KN 28-pin CASON, -10 to 70 °C
CDB42L50Evaluation Board
Purchase of I2C components of Cirrus Logic, Inc., or one of its sublicensed Associated Companies conveys a license under the Philips I2C Patent Rights to use those
components in a standard I2C system.
Preliminary product information describes products which are in production, but for which full characterization data is not yet available. Advance product information describes products which are in development and subject to development changes. Cirrus Logic, Inc. has made best efforts to ensure that the information contained in this
document is accurate and reliable. However, the information is subject to change without notice and is provided “AS IS” without warranty 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 conditions of sale supplied at the time of order acknowledgment, including those pertaining to warranty, patent infringement, and
limitation of liability. No responsibility is assumed by Cirrus Logic, Inc. for the use of this information, including use of this information as the basis for manufacture or sale
of any items, nor for infringements of patents or other rights of third parties. This document is the property of Cirrus Logic, Inc. and by furnishing this information, Cirrus
Logic, Inc. grants no license, express or implied under any patents, mask work rights, copyrights, trademarks, trade secrets or other intellectual property rights of Cirrus
Logic, Inc. Cirrus Logic, Inc., copyright owner of the information contained herein, gives consent for copies to be made of the information only for use within your organization
with respect to Cirrus Logic integrated circuits or other parts of Cirrus Logic, Inc. The same consent is given for similar information contained on any Cirrus Logic website
or disk. 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.
The names of products of Cirrus Logic, Inc. or other vendors and suppliers appearing in this document may be trademarks or service marks of their respective owners which
may be registered in some jurisdictions. A list of Cirrus Logic, Inc. trademarks and service marks can be found
2DS544PP1
at http://www.cirrus.com
.
CS42L50
4.2.5 Power Down Line Amplifier (PDNLN)............................................................................ 24
4.2.6 Power Down (PDN) ....................................................................................................... 24
4.2.7 Control Port Enable (CP_EN) ........................................................................................ 24
4.2.8 Channel A Analog Headphone Attenuation Control (address 02h) (HVOLA) .............. 25
4.2.9 Channel B Analog Headphone Attenuation Control (address 03h) (HVOLB) .............. 25
4.2.10 Channel A Digital Volume Control (address 04h) (DVOLA) ....................................... 25
4.2.11 Channel B Digital Volume Control (address 05h) (DVOLB) ....................................... 25
4.2.12 Tone Control (address 06h)........................................................................................ 26
Full Scale Input VoltageTBDVA/3.6TBDTBDVA/3.6 TBDVrms
Gain Drift-100--100-ppm/°C
Input Resistance10--10--kΩ
Input Capacitance--15--15pF
Programmable Gain Characteristics
Gain Step Size-1.0--1.0-dB
Absolute Gain Step Error--TBD--TBDdB
A/D Decimation Filter Characteristics (Note 2)
Passband(Note 3)0-23.50-47.5kHz
Passband Ripple-0.08-+0.17 -0.09-0dB
Stopband(Note 3)27.5--64.1--kHz
Stopband Attenuation(Note 4)-60.3---48.4--dB
Group Delay (Fs = Output Sample Rate)(Note 5)t
Group Delay Variation vs. Frequency∆t
High Pass Filter Characteristics
Frequency Response-3 dB(Note 2)
-0.1 dB
Phase Deviation@ 20 Hz(Note 2)-10--10-Degree
Passband Ripple (Note 2)--0.17--0.09dB
*PGA : Programmable Gain Amplifier
gd
gd
-10/Fs--2.7/Fs-s
--0.03--0.007µs
-
-
3.7
24.2
-
-
-
-
3.7
24.2
UnitMinTypMaxMinTypMax
-
Hz
-
Hz
6DS544PP1
CS42L50
ANALOG OUTPUT CHARACTERISTICS (T
GND = 0 V; Full-Scale Output Sine Wave, 997 Hz; MCLK = 12.288 MHz; Measurement Bandwidth 10 Hz to
20 kHz, unless otherwise specified; Fs for Single Speed Mode = 48 kHz, SCLK = 3.072 MHz. Fs for Double Speed
Mode = 96 kHz, SCLK = 6.144 MHz. Test load R
C
= 10 pF (see Figure 24) for headphone out.
L
Parameter
Line Output Dynamic Performance for VA = 1.8 V
Dynamic Range(Note 6)
18 to 24-Bitunweighted
A-Weighted
16-Bitunweighted
A-Weighted
Total Harmonic Distortion + Noise(Note 6)
18 to 24-Bit0 dB
-20 dB
-60 dB
16-Bit0 dB
-20 dB
-60 dB
Interchannel Isolation(1 kHz)-100--100-dB
Headphone Output Dynamic Performance for VA = VA_HP = 1.8 V
Dynamic Range(Note 6)
18 to 24-Bitunweighted
A-Weighted
16-Bitunweighted
A-Weighted
Total Harmonic Distortion + Noise(Note 6)
18 to 24-Bit0 dB
-20 dB
-60 dB
16-Bit0 dB
-20 dB
-60 dB
Interchannel Isolation(1 kHz)-66--66-dB
=10kΩ, CL= 10 pF (see Figure 23) for line out, RL=16Ω,
L
SymbolMinTypMaxMinTypMaxUnit
THD+N
THD+N
= 25° C; Logic "1" = VL = 1.8 V; Logic "0" =
A
Single Speed ModeDouble Speed Mode
TBD
TBD
-
-
-
-
-
-
-
-
TBD
TBD
-
-
-
-
-
-
-
-
91
94
89
92
-80
-71
-31
-78
-69
-29
88
91
86
89
-82
-68
-28
-80
-66
-26
-
-
-
-
TBD
-
-
-
-
-
-
-
-
-
TBD
-
-
-
-
-
TBD
TBD
-
-
-
-
-
-
-
-
TBD
TBD
-
-
-
-
-
-
-
-
89
92
87
90
-80
-69
-29
-78
-67
-27
88
91
86
89
-85
-68
-28
-83
-66
-26
TBD
TBD
dB
dB
dB
dB
-
dB
dB
dB
dB
dB
dB
-
dB
dB
dB
dB
-
dB
dB
dB
dB
dB
dB
-
Notes: 1. Referenced to typical full-scale input voltage.
2. Filter response is not tested but is guaranteed by design.
3. Filter characteristics scale with output sample rate. For output sample rates, Fs, other than 48 kHz, the
0.01 dB passband edge is 0.4535x Fs and the stopband edge is 0.625x Fs.
4. The analog modulator samples the input at 6.144 MHz for an Fs equal to 48 kHz. There is no rejection
of input signals which are multiples of the sampling frequency ( n x 6.144 MHz ±21.8 kHz where
n = 0,1,2,3...).
5. Group delay for Fs = 48 kHz, t
6. One-half LSB of triangular PDF dither is added to data.
DS544PP17
= 10/48 kHz = 208 µs.
gd
CS42L50
ANALOG OUTPUT CHARACTERISTICS (Continued)
Single Speed ModeDouble Speed Mode
Parameter
Line Output Dynamic Performance for VA = 3.0 V
Dynamic Range (Note 6)
18 to 24-Bit unweighted
A-Weighted
16-Bit unweighted
A-Weighted
Total Harmonic Distortion + Noise (Note 6)
18 to 24 Bit 0dB
-20dB
-60dB
16-Bit 0dB
-20dB
-60dB
Interchannel Isolation (1 kHz)-100--100-dB
Headphone Output Dynamic Performance for VA = VA_HP = 3.0 V
Dynamic Range (Note 6)
18 to 24-Bit unweighted
A-Weighted
16-Bit unweighted
A-Weighted
Total Harmonic Distortion + Noise (Note 6)
18 to 24 Bit 0dB
-20dB
-60dB
16-Bit 0dB
-20dB
-60dB
Interchannel Isolation(1 kHz)-66--66-dB
SymbolMinTypMaxMinTypMaxUnit
dB
-
-
-
-
TBD
-
-
-
-
-
-
-
-
-
TBD
-
-
-
-
-
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
dB
THD+N
THD+N
TBD
TBD
-
-
-
-
-
-
-
-
TBD
TBD
-
-
-
-
-
-
-
-
93
96
91
94
-85
-73
-33
-83
-71
-31
90
93
88
91
-76
-70
-30
-74
-68
-28
-
-
-
-
TBD
-
-
-
-
-
-
-
-
-
TBD
-
-
-
-
-
TBD
TBD
-
-
-
-
-
-
-
-
TBD
TBD
-
-
-
-
-
-
-
-
93
96
91
94
-85
-73
-33
-83
-71
-31
90
93
88
91
-73
-70
-30
-71
-68
-28
8DS544PP1
CS42L50
ANALOG OUTPUT CHARACTERISTICS (Continued)
ParametersSymbolMinTypMaxUnits
Analog Output
Full Scale Line Output Voltage (Note 7) V
Line Output Quiescent VoltageV
Full Scale Headphone Output VoltageV
Headphone Output Quiescent VoltageV
FS_LINE
Q_LINE
FS_HP
Q_HP
Interchannel Gain Mismatch-0.1-dB
Gain Drift-100-ppm/°C
Maximum Line Output AC-CurrentVA=1.8 V
I
LINE
VA= 3 . 0 V
Maximum Headphone Output VA=VA_HP=1.8 V
I
HP
AC-CurrentVA=VA_HP=3.0 V
Single Speed ModeDouble Speed Mode
Parameter
Combined Digital and On-chip Analog Filter Response (Note 8)
8. Filter response is not tested but is guaranteed by design.
9. Response is clock dependent and will scale with Fs. Note that the response plots (Figures 14 through 21)
have been normalized to Fs and can be de-normalized by multiplying the X-axis scale by Fs.
10. Referenced to a 1 kHz, full-scale sine wave.
11. For Single Speed Mode, the measurement bandwidth is 0.5465 Fs to 3 Fs.
For Double Speed Mode, the measurement bandwidth is 0.577 Fs to 1.4 Fs.
12. De-emphasis is not available in Double Speed Mode.
DS544PP19
CS42L50
POWER AND THERMAL CHARACTERISTICS (GND = 0 V; All voltages with respect to
ground. All measurements taken with all zeros input and open outputs, unless otherwise specified.)
ParametersSymbolMinTypMaxUnits
Power Supplies
Power Supply Current-VA=1.8 V
Normal OperationVA_HP=1.8 V
VL=1.8 V
Power Supply Current-VA=3.0 V
Normal OperationVA_HP=3.0 V
VL=3.0 V
Power Supply Current-All Supplies=1.8 V
Power Down Mode (Note 13)All Supplies =3.0V
Total Power Dissipation-All Supplies=1.8 V
Normal OperationAll Supplies=3.0 V
Maximum Headphone Power Dissipation
(1 kHz full-scale sine wave VA=1.8 V
into 16 ohm load)VA=3.0 V
Package Thermal Resistanceθ
Power Supply Rejection Ratio (Note 14) (1 kHz)
(60 Hz)
I
A
I
A_HP
I
D_L
I
A
I
A_HP
I
D_L
JA
PSRR-
-
-
-
-
-
-
-
-
-
-
-
-
13.3
1.5
154
20
1.5
270
150
350
27
65
15
26
-
-
-
-
-
-
-
-
TBD
TBD
-
-
mA
mA
µA
mA
mA
µA
µA
µA
mW
mW
mW
mW
-55-°C/Watt
60
-
40
-
-
dB
dB
Notes: 13. Power Down Mode is defined as RST
14. Valid with the recommended capacitor values on FILT+_ADC, FILT+_DAC, VQ_DAC, and VQ_ADC as
shown in Figure 4. Increasing the capacitance will also increase the PSRR. Note that care should be
taken when selecting capacitor type, as any leakage current in excess of 1.0 µA will cause degradation
in analog performance. A small ceramic capacitor in parallel with a larger electrolytic is recommended.
Applies a 20dB digital gain to the input signal, regardless of the input path.
4.1.3ANALOG INPUT MULTIPLEXER (AINMUX)
Default = 00
00 - Channel 1 direct to A/D
01 - Channel 2 direct to A/D
10 - Channel 2 through PGA to A/D
11 - Reserved
Function:
The analog input multiplexer selects the input channel as well as the input path associated with various
gain stages.
4.1.4POWER DOWN (PDN)
Default - 1
0 - Disabled
1 - Enabled
Function:
The entire ADC device will enter a low-power state whenever this function is activated. The power-down
bit defaults to ‘enabled’ on power-up and must be disabled before normal operation will begin. The contents of the control registers are retained when this mode is enabled.
4.1.5CONTROL PORT ENABLE (CP_EN)
Default = 0
0 - Disabled
1 - Enabled
Function:
The ADC will enter control port mode when this bit is enabled. This bit must be set prior to writing to the
control port.
18DS544PP1
CS42L50
4.1.6Interface Control (address 02h)
76543210
RESERVEDMCLKDIVRATIO1RATIO0MASTERDIF2DIF1DIF0
00000000
4.1.7 MASTER CLOCK DIVIDE (MCLKDIV)
Default = 0
0 - Disabled
1 - Enabled
Function:
Divides ADC MCLK by two prior to all other chip circuitry.
4.1.8 MASTER CLOCK RATIO (RATIO)
Default = 00
00 - 128x
01 - 192x
10 - 256x
11 - 384x
Function:
Sets the ratio of MCLK to LRCK for the ADC.
4.1.9 MASTER MODE (MASTER)
Default = 0
0 - Slave Mode
1 - Master Mode
Function:
Configures the CS42L50 for master or slave operation.
4.1.10 DIGITAL INTERFACE FORMAT (DIF)
Default = 000
2
000 - I
001 - Left Justified, up to 24-bit data, data valid on positive edge of SCLK
010 - Reserved
011 - Right Justified, 16-bit data, data valid on positive edge of SCLK
100 - Right Justifed, 24-bit data, data valid on positive edge of SCLK
101 - Right Justified, 18-bit data, data valid on positive edge of SCLK
110 - Right Justified, 20-bit data, data valid on positive edge of SCLK
111 - Reserved
Function:
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 Figures 25 through 30. It is recommended that the ADC
and the DAC are configured for the same Digital Interface Format.
S, up to 24-bit data, data valid on positive edge of SCLK
DS544PP119
CS42L50
4.1.11 Analog I/O Control (address 03h)
76543210
MUTELMUTERSZC1SZC0RESERVEDINDVCL=RHPFREEZE
00110000
4.1.12 LEFT/RIGHT CHANNEL MUTE (MUTE)
Default = 0
0 - Disabled
1 - Enabled
Function:
Digital mute of the left and right ADC channels.
4.1.13 SOFT RAMP AND ZERO CROSS ENABLE (SOFT/ZC)
Default = 11
00 - Change volume immediately
01 - Change volume at next zero cross time
10 - Change volume in 1dB steps
11 - Change volume in 1dB steps at every zero cross time
Function:
Soft Ramp Enable :
Soft Ramp allows level changes, both muting and attenuation, to be implemented via an incremental
ramp. Digital volume control is ramped from the current level to the new level at a rate of 1/8 dB per
left/right clock period. Analog volume control is ramped in 1 dB steps every 8 left/right clock periods in
Single Speed mode, and 1dB every 16 left/right clock periods in Double Speed mode.
Zero Cross Enable :
Zero Cross enable dictates that signal level changes, either by attenuation changes or muting, will occur
on a signal zero crossing to minimize audible artifacts. The requested level change will occur after a timeout period of 512 sample periods in Single Speed mode or 1024 sample periods in Double Speed mode
(approximately 10.7ms at 48kHz 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 and Zero Cross Enable :
Soft Ramp and Zero Cross enable dictates that signal level changes, either by attenuation changes or
muting, will occur in 1 dB steps and be implemented on a signal zero crossing. The level change will occur
after a timeout period of 512 sample periods in Single Speed mode or 1024 sample periods in Double
Speed mode (approximately 10.7 ms at 48kHz sample rate) if the signal does not encounter a zero crossing. The zero cross function is independently monitored and implemented for each channel.
20DS544PP1
4.1.14 INDEPENDENT VOLUME CONTROL ENABLE (INDVC)
Default = 0
0 - Disabled
1 - Enabled
Function:
When this function is disabled, the AIN_L and AIN_R volume levels are controlled by the Left and Right
Volume Control registers and the Independent Analog Gain Control registers are ignored.
When this function is enabled, the volume levels are determined by both the Volume Control registers and
the Independent Analog Gain Control registers.
4.1.15 LEFT CHANNEL VOLUME = RIGHT CHANNEL VOLUME (L=R)
Default = 0
0 - Disabled
1 - Enabled
Function:
When this function is disabled, the left channel volume is determined by the left channel volume control
register and right channel volume is determined by the right channel volume control register.
CS42L50
When enabled, the left and right channel volumes are determined by the left channel volume control register and the right channel volume control register is ignored.
4.1.16 HIGH-PASS FILTER FREEZE (HPFREEZE)
Default = 0
0 - Frozen
1 - Enabled
Function:
The high-pass filter works by continuously subtracting a measure of the dc offset from the output of the
decimation filter. If the HPFREEZE bit is taken low 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.
DS544PP121
CS42L50
4.1.17 Volume Control: Left Channel (address 04h) & Right Channel (address 05h)
76543210
VOL7VOL6VOL5VOL4VOL3VOL2VOL1VOL0
00000000
Default = 0 (No attenuation)
Binary CodeDecimal ValueVolume Setting
0000101012+12 dB
000001117+7 dB
0000000000 dB
11000100-60-60 dB
10100110-90-90 dB
Function:
The volume control allows the user to alter the signal level in 1 dB increments from +12 to -96 dB, when
the INDVC bit is disabled. When INDVC is enabled, the volume control can be altered in 1 dB increments
from 0 to -96dB. Volume settings are decoded as shown above, using a 2’s complement code. The volume changes are implemented as dictated by the Soft and Zero Cross bits in the Analog I/O Control register. All volume settings less than -96dB are equivalent to muting the channel.
4.1.18 Left/Right Analog Gain (address 06h)
76543210
LVOL3LVOL2LVOL1LVOL0RVOL3RVOL2RVOL1RVOL0
00000000
Default = 0 (No gain)
Binary CodeDecimal ValueVolume Setting
000000 dB
00102+2 dB
10106+6 dB
10019+9 dB
110012+12 dB
Function:
The level of the left and right analog channels can be adjusted in 1dB increments as dictated by the Soft
Ramp and Zero Cross bits from 0 to +12dB when routed through the PGA via the AINMUX bits. Levels
are decoded as shown above. Levels above +12dB are interpreted as +12dB.
The Clip Flags indicate when there is an over-range condition anywhere in the CS42L50 internal signal
path. These bits are “sticky”. They constantly monitor the ADC signal path and are set to 1 when an overrange condition occurs. They are reset to 0 when read.
4.2DAC (Address = 0010001)
4.2.1Power and Muting Control (address 01h)
76543210
AMUTESZC1SZC0PORPDNHPPDNLNPDNCP_EN
11010010
4.2.2 AUTO-MUTE (AMUTE)
Default = 1
0 - Disabled
1 - Enabled
Function:
The Digital-to-Analog converter output will mute following the reception of 8192 consecutive audio samples of static 0 or -1. A single sample of non-static data will release the mute. Detection and muting is
done independently for each channel. The quiescent voltage on the output will be retained and the Mute
Control pin will go active during the mute period. The muting function is affected, similar to volume control
changes, by the Soft and Zero Cross bits in the Power and Muting Control register.
4.2.3SOFT RAMP AND ZERO CROSS CONTROL (SZC)
Default = 10
00 - Immediate Change
01 - Zero Cross Digital and Analog
10 - Ramped Digital and Analog
11 - Reserved
Function:
Immediate Change
When Immediate Change is selected all level changes will take effect immediately in one step.
Zero Cross Digital and Analog
Zero Cross Enable dictates that signal level changes, either by attenuation changes or muting, will occur
on a signal zero crossing to minimize audible artifacts. The requested level change will occur after a
timeout period of 512 sample periods (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.
DS544PP123
Ramped Digital and Analog
Soft Ramp allows digital level changes, both muting and attenuation, to be implemented by incrementally
ramping, in 1/8 dB steps, from the current level to the new level at a rate of 1 dB per 8 left/right clock periods. Analog level changes will occur in 1 dB steps on a signal zero crossing. The analog level change
will occur after a timeout period of 512 sample periods (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.
Note:Ramped Digital and Analog is not available in Double Speed Mode.
4.2.4POWER DOWN HEADPHONE AMPLIFIER (PDNHP)
Default = 0
0 - Disabled
1 - Enabled
Function:
The headphone amplifier will independently enter a low-power state when this function is enabled.
4.2.5POWER DOWN LINE AMPLIFIER (PDNLN)
CS42L50
Default = 0
0 - Disabled
1 - Enabled
Function:
The line output amplifier will independently enter a low-power state when this function is enabled.
4.2.6POWER DOWN (PDN)
Default = 1
0 - Disabled
1 - Enabled
Function:
The entire DAC device will enter a low-power state when this function is enabled, and the contents of the
control registers are retained in this mode. The power-down bit defaults to ‘enabled’ on power-up and
must be disabled before normal operation will begin.
4.2.7CONTROL PORT ENABLE (CP_EN)
Default = 0
0 - Disabled
1 - Enabled
Function:
The DAC will enter control port mode when this bit is enabled. This bit must be set prior to writing to the
control port.
24DS544PP1
CS42L50
4.2.8Channel A Analog Headphone Attenuation Control (address 02h) (HVOLA)
4.2.9Channel B Analog Headphone Attenuation Control (address 03h) (HVOLB)
76543210
HVOLx7HVOLx6HVOLx5HVOLx4HVOLx3HVOLx2HVOLx1HVOLx0
00000000
Default = 0 dB (No attenuation)
Function:
The Analog Headphone Attenuation Control operates independently from the Digital Volume Control. The
Analog Headphone Attenuation Control registers allow attenuation of the headphone output signal for
each channel in 1 dB increments from 0 to -25 dB. Attenuation settings are decoded using a 2’s complement code, as shown in Table 1. The volume changes are implemented as dictated by the Soft and Zero
Cross bits in the Power and Muting Control register. All volume settings greater than zero are interpreted
as zero.
Note:The Analog Headphone Attenuation only affects the headphone outputs. When set for levels
greater than -10dB, the actual attenuation deviates from the register setting by more than 1 dB.
Binary CodeDecimal ValueVolume Setting
0000000000 dB
11110110-10-10 dB
11110 0 01-15-1 5 dB
Table 1. Example Analog Volume Settings
4.2.10 Channel A Digital Volume Control (address 04h) (DVOLA)
4.2.11 Channel B Digital Volume Control (address 05h) (DVOLB)
76543210
DVOLx7DVOLx6DVOLx5DVOLx4DVOLx3DVOLx2DVOLx1DVOLx0
00000000
Default = 0 dB (No attenuation)
Function:
The Digital Volume Control registers allow independent control of the signal levels in 1 dB increments
from +18 to -96 dB. Volume settings are decoded using a 2’s complement code, as shown in Table 2.
The volume changes are implemented as dictated by the Soft and Zero Cross bits in the Power and Muting Control register. All volume settings less than -96 dB are equivalent to muting the channel via the
ATAPI bits (see Section 4.46).
Note:The digital volume control affects both the line outputs and the headphone outputs. Setting this
register to values greater than +18 dB will cause distortion in the audio outputs.
DS544PP125
CS42L50
Binary CodeDecimal ValueVolume Setting
0000101012+12 dB
000001117+7 dB
0000000000 dB
11000100-60-60 dB
10100110-90-90 dB
Table 2. Example Digital Volume Settings
4.2.12 Tone Control (address 06h)
76543210
BB3BB2BB1BB0TB3TB2TB1TB0
00000000
4.2.13 BASS BOOST LEVEL (BB)
Default = 0 dB (No Bass Boost)
Function:
The level of the shelving bass boost filter is set by Bass Boost Level. The level can be adjusted in 1 dB
increments from 0 to +12 dB of boost. Boost levels are decoded as shown in Table 3. Levels above
+12 dB are interpreted as +12 dB.
Binary CodeDecimal ValueBoost Setting
000000 dB
00102+2 dB
10106+6 dB
10019+9 dB
110012+12 dB
Table 3. Example Bass Boost Settings
4.2.14 TREBLE BOOST LEVEL (TB)
Default = 0 dB (No Treble Boost)
Function:
The level of the shelving treble boost filter is set by Treble Boost Level. The level can be adjusted in 1 dB
increments from 0 to +12 dB of boost. Boost levels are decoded as shown in Table 4. Levels above
+12 dB are interpreted as +12 dB.
Note:Treble Boost is not available in Double Speed Mode.
26DS544PP1
CS42L50
Binary CodeDecimal ValueBoost Setting
000000 dB
00102+2 dB
10106+6 dB
10019+9 dB
110012+12 dB
Table 4. Example Treble Boost Settings
4.2.15 Mode Control (address 07h)
76543210
BBCF1BBCF0TBCF1TBCF0A=BDEM1DEM0VCBYP
00000000
4.2.16 BASS BOOST CORNER FREQUENCY (BBCF)
Default = 00
00 - 50 Hz
01 - 100 Hz
10 - 200 Hz
11 - Reserved
Function:
The bass boost corner frequency is user selectable as shown above.
4.2.17 TREBLE BOOST CORNER FREQUENCY (TBCF)
Default = 00
00 - 2 kHz
01 - 4 kHz
10 - 7 kHz
11 - Reserved
Function:
The treble boost corner frequency is user selectable as shown above.
Note: Treble Boost is not available in Double Speed Mode.
4.2.18 CHANNEL A VOLUME = CHANNEL B VOLUME (A=B)
Default = 0
0 - Disabled
1 - Enabled
Function:
The AOUTA/HP_A and AOUTB/HP_B volume levels are independently controlled by the A and the B
Channel Volume Control Bytes when this function is disabled. The volume on both AOUTA/HP_A and
AOUTB/HP_B are determined by the A Channel Attenuation and Volume Control Bytes, and the B Channel Bytes are ignored when this function is enabled.
Selects the appropriate digital filter to maintain the standard 15 µs/50 µs digital de-emphasis filter response at 32, 44.1 or 48 kHz sample rates. (see Figure 31)
Note:De-emphasis is not available in Double Speed Mode.
4.2.20 DIGITAL VOLUME CONTROL BYPASS (VCBYP)
Default = 0
0 - Disabled
1 - Enabled
Function:
The digital volume control section is bypassed when this function is enabled. This disables the digital volume control, muting, bass boost, treble boost, limiting and ATAPI functions. The analog headphone attenuation control will remain functional.
CS42L50
4.2.21 Volume and Mixing Control (address 0Ah)
76543210
TC1TC0TC_ENLIM_ENATAPI3ATAPI2ATAPI1ATAPI0
00001001
4.2.22 TONE CONTROL MODE (TC)
Default = 00
00 - All settings are taken from user registers
01 - 12 dB of Bass Boost at 100 Hz and 6 dB of Treble Boost at 7 kHz
10 - 8 dB of Bass Boost at 100 Hz and 4 dB of Treble Boost at 7 kHz
11 - 4 dB of Bass Boost at 100 Hz and 2 dB of Treble Boost at 7 kHz
Function:
The Tone Control Mode bits determine how the Bass Boost and Treble Boost features are configured.
The user defined settings from the Bass and Treble Boost Level and Corner Frequency registers are used
when these bits are set to ‘00’. Alternately, one of three pre-defined settings may be used.
4.2.23 TONE CONTROL ENABLE (TC_EN)
Default = 0
0 - Disabled
1 - Enabled
Function:
The Bass Boost and Treble Boost features are active when this function is enabled.
The MCLKDIV bit enables a circuit which divides the externally applied MCLK signal by 2 prior to all other
DAC circuitry.
4.2.27 LINE AMPLIFIER GAIN COMPENSATION (LINE)
Default = 00
00 - 0.785 x VA
01 - 0.943 x VA
10 - Reserved
DS544PP129
11 - Line Mute
Function:
The Line Amplifier Gain Compensation bits allow the user to scale the full-scale line output level according
to the power supply voltage used. The full-scale line output level will be equal to {gain factor}xVA, where
{gain factor} is selected from options above.
The Line Mute option is available to allow muting of the line output when the headphone output is still in
use and the line amp is still powered up. To use this feature, first mute the outputs via the ATAPI bits.
Next, set the LINE GAIN to Line Mute. Finally, un-mute the outputs with the ATAPI bits. Following these
steps will ensure a click free mute.
4.2.28 DIGITAL INTERFACE FORMAT (DIF)
Default = 000 - Format 0 (I2S, up to 24-bit data)
Function:
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 Figures 25-30. It is recommended that the ADC and the
DAC is configured for the same Digital Interface Format.
CS42L50
DIF2DIF1DIF0DESCRIPTIONFormatFIGURE
000
001
010
011
100
101
110
111
I2S, up to 24-bit data
Reserved
Left Justified, up to 24-bit data,
Right Justified, 24-bit data
Right Justified, 20-bit data
Right Justified, 16-bit data
Right Justified, 18-bit data
Identical to Format 0
030
-225
327
429
526
628
030
Table 6. Digital Interface Format
30DS544PP1
CS42L50
5. PIN DESCRIPTIONS
1
Filter CapacitorAFLTRRST Reset
Filter CapacitorAFLTLSCLControl Port Clock
Voltage ReferenceFILT+_ADCSDAControl Port Data
Analog Input 2 RightAIN_R2LRCKLeft/Right Clock
Analog Input 2 LeftAIN_L2SDOUTSerial Audio Data Output
Ground ReferenceREF_GNDVAAnalog Power
Analog Input 1 RightAIN_R1GNDGround Reference
Analog Input 1 LeftAIN_L1VLInterface Power
Quiescent VoltageVQ_ADCSDINSerial Audio Data Input
Voltage ReferenceFILT+_DACMCLKMaster Clock
Quiescent VoltageVQ_DACSCLKSerial Clock
Headphone A OutputHP_AMUTECExternal Mute Control
Headphone Amp PowerVA_HPAOUTLAnalog Output Left
Headphone B OutputHP_BAOUTRAnalog Output Right
28
2
27
3
26
4
25
5
24
6
23
7
22
821
9
20
10
19
11
18
1217
13
16
1415
Pin Name#Pin Description
VA
VL
VA_ H P
VQ_ADC
VQ_DAC
REF_GND
GND
23Analog Power (Input) - Positive power supply for the analog section. Refer to the Recommended Oper-
ating Conditions for appropriate voltages.
Logic Power (Input) - Determines the required signal level for the digital input/output. Refer to the Rec-
21
ommended Operating Conditions for appropriate voltages.
13Headphone Amp Power (Input) - Positive power supply for the headphone amplifier. Refer to the Rec-
ommended Operating Conditions for appropriate voltages.
9,11 Quiescent Voltage (Output) - Filter connection for internal quiescent voltage. VQ must be capacitively
coupled to analog ground, as shown in the Typical Connection Diagram. The nominal voltage level is
specified in the Analog Characteristics and Specifications section. VQ presents an appreciable source
impedance and any current drawn from this pin will alter device performance. However, VQ can be used
to bias the analog circuitry assuming there is no AC signal component and the DC current is less than
10uA.
Reference Ground (Input) - Ground reference for the internal sampling circuits and must be connected
6
to analog ground.
Ground (Input) - Ground reference. Should be connected to analog ground.
22
Serial Audio
Interface
MCLK
SCLK
LRCK
SDIN
SDOUT
Master Clock (Input) - Clock source for the delta-sigma modulator and digital filters.
19
Serial Clock (Input/Output) - Serial clock for the serial audio interface.
18
Left Right Clock (Input/Output) - Determines which channel, Left or Right, is currently active on the
25
serial audio data line. The frequency of the left/right clock must be at the audio sample rate, Fs.
Serial Audio Data Input (Input) - Input for two’s complement serial audio data.
20
Serial Audio Data Output (Output) - Output for two’s complement serial audio data.
24
DS544PP131
Analog
Input/Output
AIN_Rx
AIN_Lx
AOUTL
AOUTR
HP_A
HP_B
4, 5,
7,8
15,
16
12,
14
Control Port
Interface
SCL
SDA
27
26
Control & Misc.
AFLTR
AFLTL
FILT+_ADC
FILT+_DAC
MUTEC
RST
1,2
10
17Mute Control (Output) - The Mute Control pin goes low during power-up initialization, reset, muting,
28
CS42L50
Analog Inputs (Input) - The full scale analog input level is specified in the Analog Input Characteristics
specification table.
Analog Outputs (Output) - The full scale analog line output level is specified in the Analog Output Char-
acteristics specifications table.
Headphone Outputs (Output) - The full scale analog headphone output level is specified in the Analog
Output Characteristics specifications table.
Serial Control Port Clock (Input) - Serial clock for the serial control port. Requires an external pull-up
resistor to the logic interface voltage as shown in the Typical Connection Diagram.
Serial Control Data (Input/Output) - SDA is a data I/O line and requires an external pull-up resistor to
the logic interface voltage, as shown in the Typical Connection Diagram.
Anti-Aliasing Capacitors (Output) - Anti-aliasing capacitors for the left and right channels. An external
capacitor is required from AFLTR and AFLTL to ground, as shown in the Typical Connections Diagram.
AFLTR and AFLTL are not intended to supply external current, and any current drawn from these pins
will alter device performance.
Positive Voltage Reference (Output) - Positive reference voltage for the internal sampling circuits.
3
Requires the capacitive decoupling to AGND as shown in the Typical Connection Diagram.
Positive Voltage Reference (Output) - Positive reference voltage for the internal sampling circuits.
Requires the capacitive decoupling to AGND as shown in the Typical Connection Diagram.
power-down or if the master clock to left/right clock frequency ratio is incorrect. This pin is intended to be
used as a control for an external mute circuit to prevent the clicks and pops that can occur in any single
supply system. The use of an external mute circuit is not mandatory but may be desired for designs
requiring the absolute minimum in extraneous clicks and pops.
Reset (Input) - The device enters a low power mode and all internal registers are reset to their default
settings when low. When high, the control port becomes operational and the CP_EN bits must be set and
the PDN bits must be cleared before normal operation will occur. The control port cannot be accessed
when Reset is low.
32DS544PP1
CS42L50
6. APPLICATIONS
6.1Grounding and Power Supply
Decoupling
As with any high resolution converter, the
CS42L50 requires careful attention to power supply and grounding arrangements to optimize performance. Figure 4 shows the recommended power
arrangement with VA, VA_HP, and VL connected
to clean supplies. Decoupling capacitors should be
located as close to the device package as possible.
If desired, all supply pins may be connected to the
same supply, but a decoupling capacitor should still
be used on each supply pin.
6.2Clock Modes
The CS42L50 operates in one of two clocking
modes. Single Speed Mode supports input sample
rates up to 50 kHz, and Double Speed Mode supports input sample rates up to 100 kHz. All clock
modes use 64x oversampling.
6.3EP73xx Serial Port Interface
Special considerations must be made when interfacing the CS42L50 with the EP73xx series of
ARM processors. To receive stereo data from the
ADC, connect the MCLK pin (pin 19) of the
CS42L50 to the BUZ pin (pin 93) of the EP73xx,
and run the serial port in 64Fs mode with MCLK
generation enabled on the EP73xx. Any other configuration, either hardware or software modes, will
result in mono data being produced from the ADC
of the CS42L50.
6.4De-Emphasis
The CS42L50 includes on-chip digital de-emphasis. Figure 31 shows the de-emphasis curve. The
frequency response of the de-emphasis curve will
scale proportionally with changes in sample rate,
Fs.
The de-emphasis feature is included to accommodate older audio recordings that utilize pre-emphasis equalization as a means of noise reduction.
6.5Recommended Power-up Sequence
1) Hold RST low until the power supply, master
clock and left/right clock are stable. In this
state, the control port is reset to its default settings and VQ_ADC and VQ_DAC will remain
low.
2) Bring RST high. The device will remain in a
low power state and VQ_ADC and VQ_DAC
remain low. The control port will be accessible
at this time and the desired register settings can
be loaded after setting the CP_EN bits and
while keeping the PDN bits set to 1.
3) Once the registers are configured as desired, set
the PDN bits to 0, initiating the power-up sequence.
6.6Optional External Headphone Mute
An external headphone mute circuit, as shown in
the CDB42L50 datasheet schematic, is recommended to minimize the effects of output transients
during power-up and power-down. This technique
minimizes the audio transients commonly produced by single-ended, single-supply converters
when it is implemented with external DC-blocking
capacitors connected in series with the audio outputs.
Use of the Mute Control function on the line outputs is recommended for designs requiring the absolute minimum in extraneous clicks and pops.
Also, use of the Mute Control function can enable
the system designer to achieve idle channel
noise/signal-to-noise ratios only limited by the external mute circuit. See the CDB42L50 datasheet
for a suggested mute circuit.
7. CONTROL PORT INTERFACE
The control port is used to load all the internal settings. The operation of the control port may be
completely asynchronous with the audio sample
rate. However, to avoid potential interference problems, the control port pins should remain static if
DS544PP133
CS42L50
no operation is required. Please note that the internal registers are separated into two unique chip address blocks, one for the control of the ADC and
one for the control of the DAC portion of the codec.
SDA is a bidirectional data line. Data is clocked
into and out of the part by the clock, SCL, with the
clock to data relationship as shown in Figure 5.
The upper 6 bits of the 7 bit address field must be
001000. To communicate with the CS42L50, the
chip address should match that of the ADC
(0010000) or DAC (0010001) address. The eighth
bit of the address byte is the R/W bit (high for a
read, low for a write). If the operation is a write, the
next byte is the Memory Address Pointer, MAP,
which selects the register to be read or written. The
MAP is then followed by the data to be written. If
the operation is a read, the contents of the register
pointed to by the MAP will be output after the chip
address.
The CS42L50 has MAP auto increment capability,
enabled by the INCR bit in the MAP register. If
INCR is 0, then the MAP will stay constant for successive writes. If INCR is set to 1, then MAP will
auto increment after each byte is written, allowing
block reads or writes of successive registers.
34DS544PP1
7.1Memory Address Pointer (MAP)
76543210
INCRReservedReservedReservedMAP3MAP2MAP1MAP0
00000000
7.2INCR (AUTO MAP INCREMENT ENABLE)
Default = ‘0’
0 - Disabled
1 - Enabled
7.3MAP0-3 (MEMORY ADDRESS POINTER)
Default = ‘0000’
Note 1
SDA
001000
ADDR
AD0
R/W
ACK
DATA
1-8
ACK
DATA
1-8
ACK
CS42L50
SCL
Start
Note: If o peration is a w rite , th is byte co ntains the M em ory A d dress P o in ter, M AP .
Left Justified, up to 24-Bit Data. Data Valid on Rising
Edge of SCLK.
Figure 25. Left Justified, up to 24-bit data
LRCK
SCLK
SDATA
Left Channel
15 14 13 12 11 10
Right Justified, 16-Bit Data. Data Valid on Rising Edge of
32 clocks
SCLK. SCLK Must Have at Least 32 Cycles per LRCK
Period.
Figure 26. Right Justified, 16-bit data
LSB
Right Channel
+3 +2 +1
MSB
-1 -2 -3 -4
6543210987
15 14 13 12 11 10
+5 +4
Right Channel
LSB
6543210987
LRCK
SCLK
SDATA
Left Channel
0
23 22 21 20 19 18
32 clocks
Right Justified, 24-Bit Data. Data Valid on Rising Edge of
65432107
23 22 21 20 19 18
Right Channel
65432107
SCLK. SCLK Must Have at Least 48 Cycles per LRCK
Period.
Figure 27. Right Justified, 24-bit data
DS544PP141
CS42L50
LRCK
SCLK
SDATA
LRCK
SCLK
SDATA
Right C hannel
10
Left Channel
17 1617 16
15 14 13 12 11 10
32 clocks
6543210987
15 14 13 12 11 10
Right Justified, 18-Bit Data. Data Valid on Rising Edge of
SCLK. SCLK Must Have at Least 36 Cycles per LRCK
Period.
Figure 28. Right Justified, 18-bit data
Left Channel
106543210987
17 1617 16
19 1819 18
15 14 13 12 11 10
32 clocks
6543210987
15 14 13 12 11 10
Right Channel
Right Justified, 20-Bit Data. Data Valid on Rising Edge of
SCLK. SCLK Must Have at Least 40 Cycles per LRCK
Period.
6543210987
Figure 29. Right Justified, 20-bit data
LRCK
SCLK
SDATA+3 +2 +1
MSB
-1 -2 -3 -4 -5
Left Channel
+5 +4
I2S, up to 24-Bit Data. Data Valid on Rising Edge of
SCLK
LSB
MSB
-1 -2 -3 -4
Figure 30. I2S, up to 24-bit data
Right Channel
+3 +2 +1
+5 +4
LSB
42DS544PP1
CS42L50
Gain
dB
T1=50 µs
0dB
T2 = 15 µs
-10dB
Left Channel
Audio Data
F1F2
3.183 kHz10.61 kHz
Figure 31. De-Emphasis Curve
A Channel
Digital
Volume
Control
Σ
Frequency
EQ
MUTE
AoutA/HP_A
B Channel
Right Channel
Audio Data
Digital
Volume
Control
EQ
MUTE
AoutB/HP_B
Figure 32. ATAPI Block Diagram
DS544PP143
8. PARAMETER DEFINITIONS
Total Harmonic Distortion + Noise (THD+N)
The ratio of the rms value of the signal to the rms sum of all other spectral components over the specified
bandwidth (typically 10 Hz to 20 kHz), including distortion components. Expressed in decibels.
Dynamic Range
The ratio of the full scale 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 measurement over the specified bandwidth
made with a -60 dBFS signal. 60 dB is then added to the resulting measurement to refer the measurement
to full scale. This technique ensures that the distortion components are below the noise level and do not
affect the measurement. This measurement technique has been accepted by the Audio Engineering Society, AES17-1991, and the Electronic Industries Association of Japan, EIAJ CP-307.
Interchannel Isolation
A measure of crosstalk between the left and right channels. Measured for each channel at the converter's
output with all zeros to the input under test and a full-scale signal applied to the other channel. Units in
decibels.
Interchannel Gain Mismatch
The gain difference between left and right channels. Units in decibels.
CS42L50
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.
9. REFERENCES
1) "How to Achieve Optimum Performance from Delta-Sigma A/D & D/A Converters" by Steven Harris.
Paper presented at the 93rd Convention of the Audio Engineering Society, October 1992.
1) "The I2C-Bus Specification: Version 2.1" Philips Semiconductors, January 2000.
http://semiconductors.philips.com
44DS544PP1
10. PACKAGE DIMENSIONS
CS42L50
Figure 33. Package Dimensions
DS544PP145
CS42L50
Figure 34. Package Top and Side Views
46DS544PP1
CS42L50
Figure 35. Package Bottom View
DS544PP147
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