LM49352
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LM49352 Boomer™ Mono Class D Audio Codec Subsystem with Ground Referenced
Headphone Amplifiers, Earpiece Driver, and Audio DSP
Check for Samples: LM49352
1FEATURES
23 Ultra Efficient, Spread Spectrum Class D Micro-Power Shutdown Mode
Loudspeaker Amplifier that Operates at 93% Available in the 3.3 x 3.3 mm 36 bump DSBGA
Efficiency package
Low Voltage, True Ground Headphone
Amplifier Operation APPLICATIONS
High Performance 103dB SNR Stereo DAC Smart Phones
High Performance 97dB SNR Stereo ADC Mobile Phones and VOIP Phones
Up to 96kHz Stereo Audio Playback Portable GPS Navigator and Portable Gaming
Devices
Up to 48kHz Stereo Recording Portable DVD/CD/AAC/MP3/MP4 Players
Dual Bidirectional I2S or PCM Compatible
Audio Interfaces Digital Cameras/Camcorders
Read/Write I2C Compatible Control Interface KEY SPECIFICATIONS
Flexible Digital Mixer with Sample Rate
Conversion Class D Amplifier Efficiency 93% (Typ)
Sigma-Delta PLL Clock Network that Supports PEP at A_VDD = 3.3V, 32, 1% THD 58mW (Typ)
System Clocks up to 50MHz Including 13MHz, PHP at HP_VDD = 2.8V, Stereo 32, 1% THD
19.2MHz, and 26MHz 65mW/ch (Typ)
Dual stereo 5 band parametric equalizers PLS at LS_VDD = 5V, 8, 1% THD 1.4W (Typ)
Cascadable DSP Effects that Allow Stereo 10 PLS at LS_VDD = 4.2V, 8, 1% THD 970 mW
Band Parametric Equalization (Typ)
ALC/Limiter/Compressor on Both DAC and PLS at LS_VDD = 3.3V, 8, 1% THD 590 mW
ADC Paths (Typ)
Dedicated Earpiece Speaker Amplifier SNR (Stereo DAC at 48kHz) 103 dB (Typ)
Stereo Auxiliary Inputs and Mono Differential PSRR at 217 Hz, A_VDD = 3.3V, (HP from AUX)
Input 100dB (Typ)
Differential Microphone Input with Single-
Ended Option DESCRIPTION
Automatic Level Control for Digital Audio The LM49352 is a high performance mixed signal
audio subsystem. The LM49352 includes a high
Inputs, Mono Differential Input, Microphone quality stereo DAC, a high quality stereo ADC, a
Input, and Stereo Auxiliary Inputs stereo headphone amplifier, which supports True
Flexible Audio Routing from Input to Output Ground operation, a low EMI Class D loudspeaker
16 Step Volume Control for Microphone with amplifier, and an earpiece speaker amplifier. It
2dB Steps combines advanced audio processing, conversion,
mixing, and amplification in the smallest possible
32 Step Volume Control for Auxiliary Inputs in footprint while extending the battery life of feature rich
1.5dB Steps portable devices.
4 Step Volume Control for Class D
Loudspeaker Amplifier
8 Step Volume Control for Headphone
Amplifier
1Please be aware that an important notice concerning availability, standard warranty, and use in critical applications of
Texas Instruments semiconductor products and disclaimers thereto appears at the end of this data sheet.
2Boomer is a trademark of Texas Instruments.
3All other trademarks are the property of their respective owners.
PRODUCTION DATA information is current as of publication date. Copyright © 2009–2013, Texas Instruments Incorporated
Products conform to specifications per the terms of the Texas
Instruments standard warranty. Production processing does not
necessarily include testing of all parameters.
EP+/
AUX_OUT+
LS_OUT+
HPL_OUT
HPR_OUT
DACREF
MCLK
A_VDD
D_VDD
SCL
SDA
PORT1_CLK
PORT1_WS
PORT1_SDI
PORT1_SDO
PORT2_CLK
PORT2_WS
PORT2_SDI
MONO_IN-
MONO_IN+
POWER MANAGEMENT
and CONTROL
CLOCK
NETWORK
with
6'PLL
REGISTERS
I2C
SLAVE
BG
AB
AB
DIGITAL MIXER & AUDIO PORTS
AUX
MONO
-46.5 dB to 18 dB
6 dB to 36 dB
PORT2_SDO/GPIO
RIGHT
ADC
6'
6'
LEFT
ADC
DAC
EFFECTS
VOL CTRL
5-BAND EQ
ALC
AB
CHARGE
PUMP
CP+
CP-
LS_VDD LSGND
AGND
DGND
I/O_VDD
6'
LEFT
DAC
6'
RIGHT
DAC
MIC
HPVSS
MIC+
VCM
DAC_LEFT
DAC_RIGHT
EP-/
AUX_OUT-
LS_OUT-
MIC-
VCM
ADC
EFFECTS
VOL CTRL
5-BAND EQ
ALC
AUXR / AUX+
AUXL / AUX- VCM
HP_VSS
HP_VSS
HP_VDD
HP_VDD
HP_VDD
0 dB to 12 dB
-18 dB
to
0 dB
D
-18 dB
to
0 dB
ADCREF
0 dB to 6 dB
LM49352
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DESCRIPTION (CONTINUED)
The LM49352 features dual bi-directional I2S or PCM audio interfaces and an I2C compatible interface for control.
The stereo DAC path features an SNR of 103dB with 24-bit 48 kHz input. The headphone amplifier delivers
65mWRMS (typ) to a 32single-ended stereo load with less than 1% distortion (THD+N) when HP_VDD = 2.8V.
The loudspeaker amplifier delivers up to 970mW into an 8load with less than 1% distortion when LS_VDD =
4.2V.
The LM49352 employs advanced techniques to extend battery life, to reduce controller overhead, to speed
development time, and to eliminate click and pop artifacts. Boomer audio power amplifiers are designed
specifically for mobile devices and require minimal PCB area and external components.
LM49352 Overview
Figure 1. LM49352 Block Diagram
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LM49352
I2C
Baseband
Controller
CP+
HP_VSS
MCLK
EP+
LS+
32O
AUX_RAUX_L
0.5 - 50
MHz
I2S / PCM
(PORT1)
Multimedia
IC I2S / PCM
(PORT2)
HPL
HPR
CP-
DACREF
AGND
A_VDD
D_VDD LS_VDD
I/O_VDD
LSGNDDGND
MIC+
EP-
LS-
MIC-
8O
MONO+
MONO-
HP_VDD
Synthesized
FM Radio / TV Tuner
ADC
REF
LM49352
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Typical Application
Figure 2. Example Application in Multimedia Phone with Dedicated Earpiece Speaker and Mono
Loudspeaker
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LM49352
I2C
Baseband
Controller
CP+
HP_VSS
MCLK
EP+
LS+
32O
AUX_RAUX_L
0.5 - 50
MHz
I2S / PCM
(PORT1)
FM /
Bluetooth
Radio I2S / PCM
(PORT2)
HPL
HPR
CP-
DAC REF
AGND
LSGNDDGND
MIC+
EP-
LS-
MIC-
8O
MONO+
MONO-
Synthesized
TV Tuner
Multimedia
IC
A_VDD
D_VDD LS_VDD
I/O_VDD HP_VDD
ADC
REF
LM49352
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Figure 3. Example Application in Multimedia Phone Using Multiple Media Sources
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LM49352
I2C
Baseband
Controller
CP+
HP_VSS
MCLK
AUX_OUT+
LS+
8O
AUX_RAUX_L
0.5 - 50
MHz
I2S / PCM
(PORT1)
Multimedia
IC I2S / PCM
(PORT2)
HPL
HPR
CP-
DACREF
AGND
LSGNDDGND
MIC+
AUX_OUT-
LS-
MIC-
8O
MONO+
MONO-
Synthesized
FM Radio / TV Tuner
LM48310
A_VDD
D_VDD LS_VDD
I/O_VDD HP_VDD
ADC
REF
LM49352
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Figure 4. Example Application in a Multimedia Phone with Stereo Loudspeaker
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LM49352
I2C
Multimedia
Processor
CP+
HP_VSS
MCLK
AUX_OUT+
LS+
8O
+ AUX -
0.5 - 50
MHz
I2S / PCM
(PORT1)
Bluetooth
Radio I2S / PCM
(PORT2)
HPL
HPR
CP-
DACREF
AGND
LSGNDDGND
MIC+
AUX_OUT-
LS-
MIC-
8O
FM Radio / TV Tuner
w/ Stereo Differential
Outputs
LM48310
+ MONO -
A_VDD
D_VDD LS_VDD
I/O_VDD HP_VDD
ADC
REF
LM49352
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Figure 5. Example Application in a Portable Media Player with Stereo Loudspeakers
Connection Diagrams
Figure 6. 36 Bump DSBGA
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HPVSS
CP-
LSGND
LS+
C
D
E
F
1 2 3 4 5 6
HPL
B
AHPR
HPVDD
CP+
LSVDD
LS-
AVDD
AUXR
EP-
EP+
MONO-
MONO+
AGND
AUXL
PORT2
SDO
PORT1
SYNC
MIC-
MIC+
DAC
REF
PORT2
SYNC
PORT2
CLK
PORT1
SDO
PORT1
SDI
PORT1
CLK
ADC
REF
PORT2
SDI
MCLK
DGND
DVDD
IOVDD
SDA
SCL
LM49352
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Figure 7. LM49352RL Pinout Diagram
Top View (Bump Side Down)
PIN DESCRIPTIONS
Pin Pin Name Type Direction Description
A1 HPR Analog Output Headphone right output
DAC (Analog), ADC (Analog), PLL (Analog), input stages, analog mixer
A2 A_VDD Supply Input (AUX and class D), and Earpiece amplifier power supply input
DAC (Analog), ADC (Analog), PLL (Analog), input stages, analog mixer
A3 AGND Supply Input (AUX and class D), and Earpiece amplifier ground
A4 DAC REF Analog Input/Output Filter point for the DAC reference
A5 ADC REF Analog Input Filter point for the ADC reference. Connect this pin to A_VDD.
A6 SDA Digital Input/Output I2C interface data line
B1 HPL Analog Output Headphone left output
B2 AUX_R/AUX+ Analog Input Right analog input or positive differential auxiliary input
B3 AUX_L/AUX- Analog Input Left analog input or negative differential auxiliary input
B4 PORT2_SYNC Digital Input/Output Audio Port 2 sync signal (can be master or slave)
B5 PORT2_SDI Digital Input Audio Port 2 serial data input
B6 SCL Digital Input I2C interface clock line
C1 HP_VSS Analog Output Negative power supply pin for the headphone amplifier
C2 HP_VDD Supply Input Headphone amplifier power supply pin
C3 EP-/AUXOUT- Analog Output Earpiece negative output or Auxiliary negative output
PORT2_SDO /
C4 Digital Input / Output Audio port 2 serial data output or General Purpose Input Output
GPIO
C5 PORT2_CLK Digital Input/Output Audio port 2 clock signal (can be master or slave)
C6 MCLK Digital Input Input clock from 0.5MHz to 50 MHz
D1 CP- Analog Input/Output Fly capacitor negative input
D2 CP+ Analog Input/Output Fly capacitor positive input
D3 EP+/AUXOUT+ Analog Output Earpiece positive output or Auxiliary positive output
D4 PORT1_SYNC Digital Input/Output Audio Port 1 sync signal (can be master or slave)
D5 PORT1_SDO Digital Output Audio Port 1 serial data output
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PIN DESCRIPTIONS (continued)
Pin Pin Name Type Direction Description
D6 DGND Supply Input Digital ground
E1 LSGND Supply Input Loudspeaker ground
E2 LS_VDD Supply Input Loudspeaker amplifier and analog mixer (headphone) supply input
E3 MONO- Analog Input Mono differential negative input
E4 MIC- Analog Input Microphone negative input
E5 PORT1_SDI Digital Input Audio Port 1 serial data input
DAC (Digital), ADC (Digital), PLL (Digital), digital mixer, DSP core, and I2C
E6 D_VDD Supply Input register power supply input
F1 LS + Analog Output Loudspeaker positive output
F2 LS - Analog Output Loudspeaker negative output
F3 MONO+ Analog Input Mono differential positive input
F4 MIC + Analog Input Microphone positive input
F5 PORT1_CLK Digital Input/Output Audio Port 1 clock signal (can be master or slave)
F6 I/O_VDD Supply Input Digital I/O (MCLK, I2S/PCM, I2C) interface power supply input
PIN TYPE DEFINITIONS
Analog Input A pin that is used by the analog and is never driven by the device. Supplies are part of this
classification.
Analog Output A pin that is driven by the device and should not be driven by external sources.
Analog Input/Output A pin that is typically used for filtering a DC signal within the device. Passive
components can be connected to these pins.
Digital Input A pin that is used by the digital but is never driven by the device.
Digital Output A pin that is driven by the device and should not be driven by another device to avoid
contention.
Digital Input/Output A pin that is either open drain (SDA) or a bidirectional CMOS in/out. In the latter case the
direction is selected by a control register within the LM49352.
These devices have limited built-in ESD protection. The leads should be shorted together or the device placed in conductive foam
during storage or handling to prevent electrostatic damage to the MOS gates.
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Absolute Maximum Ratings(1)(2)(3)
Analog Supply Voltage
(A_VDD and LS_VDD) 6.0V
Digital Supply Voltage
D_VDD 2.2V
I/O Supply Voltage
I/O_VDD 5.5V
Headphone Supply Voltage
HP_VDD 3.0V
Storage Temperature 65°C to +150°C
Power Dissipation(4) Internally Limited
Human Body Model(5) HPR and HPL pins 8kV
ESD Ratings All other pins 2.5kV
Machine Model(6) 200V
Junction Temperature 150°C
θJA RLA36 (soldered down to PCB
Thermal Resistance 60°C/W
with 2in21oz. copper plane)
Soldering Information See Applications Note AN-1112 (Literature Number SNVA009).
(1) “Absolute Maximum Ratings” indicate limits beyond which damage to the device may occur, including inoperability and degradation of
device reliability and/or performance. Functional operation of the device and/or non-degradation at the Absolute Maximum Ratings or
other conditions beyond those indicated in the Recommended Operating Conditions is not implied. The Recommended Operating
Conditions indicate conditions at which the device is functional and the device should not be operated beyond such conditions. All
voltages are measured with respect to the ground pin, unless otherwise specified.
(2) The Electrical Characteristics tables list ensured specifications under the listed Recommended Operating Conditions except as
otherwise modified or specified by the Electrical Characteristics Conditions and/or Notes. Typical specifications are estimations only and
are not ensured.
(3) If Military/Aerospace specified devices are required, please contact the Texas Instruments Sales Office/Distributors for availability and
specifications.
(4) The maximum power dissipation must be derated at elevated temperatures and is dictated by TJMAX,θJA, and the ambient temperature,
TA. The maximum allowable power dissipation is PDMAX = (TJMAX - TA) / θJA or the number given in Absolute Maximum Ratings,
whichever is lower.
(5) Human body model, applicable std. JESD22-A114C.
(6) Machine model, applicable std. JESD22-A115-A.
Operating Ratings
Temperature Range 40°C to +85°C
A_VDD, LS_VDD, and AVDD_REF 2.7V to 5.5V
D_VDD 1.6V to 2.0V
Supply Voltage I/O_VDD 1.6V to 4.5V
HP_VDD 1.7V to 2.8V
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Electrical Characteristics
A_VDD = LS_VDD = 3.3V; HP_VDD = D_VDD = I/O_VDD = 1.8V(1)(2)
The following specifications apply for RL(LS) = 8, RL(HP) = 32, f = 1kHz, unless otherwise specified. Limits apply for TA=
25°C. LM49352 Units
Symbol Parameter Conditions Typical Limit (Limit)
(3) (4)
DC CHARACTERISTICS (Digital current combines D_VDD and I/O_VDD. Analog current combines A_VDD, HP_VDD, and LS_VDD)
Shutdown Mode,
DISD Digital Shutdown Current 2 µA
fMCLK = 13MHz, PLL Off
fMCLK = 11.2896MHz, fS= 44.1kHz,
Digital Active Current (MP3 Mode) Stereo DAC On, OSRDAC = 64, 1.2 1.4 mA (max)
PLL Off, HP On
fMCLK = 13MHz
Digital Active Current (FM Mode) 0.2 0.5 mA (max)
Analog Audio modes
fMCLK = 12.288MHz, fS= 48kHz,
DIDD Digital Active Current Stereo ADC On, OSRADC = 128, 1.3 1.5 mA (max)
(FM Record Mode) PLL Off, Stereo Analog Inputs On
fMCLK = 12.288MHz, fS= 8kHz,
Digital Active Current Mono ADC On, Mono DAC On, 0.5 0.8 mA (max)
(CODEC Mode) OSRDAC = 64
OSRADC = 128, PLL Off, MIC On
AISD Analog Shutdown Current Shutdown Mode 0.1 5 μA (max)
fMCLK = 11.2896MHz, fS= 44.1kHz
Stereo DAC On, OSRDAC = 64
PLL Off, Stereo HP On
Analog Supply Current (MP3 Mode) From A_VDD 4.3 6 mA (max)
From HP_VDD 1.5 2.7 mA (max)
fMCLK = 13MHz, PLL Off
Stereo Auxiliary Inputs On,
PLL Off, Stereo HP On
Analog Supply Current (FM Mode) From A_VDD 1.7 2.6 mA (max)
AIDD From HP_VDD 1.5 2.7 mA (max)
fMCLK = 12.288MHz, fS= 48kHz,
Analog Supply Current Stereo ADC On, OSRADC = 128, 7.2 9.3 mA (max)
(FM Record Mode) PLL Off, Stereo Analog Inputs On
fMCLK = 12.288MHz, fS= 8kHz,
Mono ADC On, Mono DAC On,
Analog Supply Current OSRDAC = 64 6.6 8.7 mA (max)
(CODEC Mode) OSRADC = 128, PLL Off, MIC On,
EP On
fMCLK = 13MHz,
fPLLOUT = 12MHz, PLL On only
PLLIDD PLL Total Active Current From A_VDD 1.9 2.7 mA (max)
From D_VDD 1.4 2 mA (max)
HPIDD Headphone Quiescent Current Stereo HP On only 1.5 mA
LSIDD Loudspeaker Quiescent Current LS On only 2.2 mA
MICIDD Microphone Quiescent Current Mono MIC 0.4 mA
(1) “Absolute Maximum Ratings” indicate limits beyond which damage to the device may occur, including inoperability and degradation of
device reliability and/or performance. Functional operation of the device and/or non-degradation at the Absolute Maximum Ratings or
other conditions beyond those indicated in the Recommended Operating Conditions is not implied. The Recommended Operating
Conditions indicate conditions at which the device is functional and the device should not be operated beyond such conditions. All
voltages are measured with respect to the ground pin, unless otherwise specified.
(2) The Electrical Characteristics tables list ensured specifications under the listed Recommended Operating Conditions except as
otherwise modified or specified by the Electrical Characteristics Conditions and/or Notes. Typical specifications are estimations only and
are not ensured.
(3) Typical values represent most likely parametric norms at TA= +25ºC, and at the Recommended Operation Conditions at the time of
product characterization and are not ensured.
(4) Datasheet min/max specification limits are specified by test or statistical analysis.
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Electrical Characteristics
A_VDD = LS_VDD = 3.3V; HP_VDD = D_VDD = I/O_VDD = 1.8V(1)(2) (continued)
The following specifications apply for RL(LS) = 8, RL(HP) = 32, f = 1kHz, unless otherwise specified. Limits apply for TA=
25°C. LM49352 Units
Symbol Parameter Conditions Typical Limit (Limit)
(3) (4)
fS= 48kHz, Stereo
ADCIDD ADC Total Active Current From A_VDD 5.8 mA
From D_VDD 1.3 mA
fS= 48kHz, Stereo
DACIDD DAC Total Active Current From A_VDD 3.1 mA
From D_VDD 1 mA
Mono/Auxiliary Input Amplifier Mono and AUX Input Amplifiers
AUXINIDD 0.6 mA
Quiescent Current enabled
AUXOUT enabled 0.4 mA
Auxiliary Output Amplifier Quiescent
AUXOUTIDD Current Earpiece Mode 1.1 mA
LOUDSPEAKER AMPLIFIER
PO= 970mW, RL= 8,
LSEFF Loudspeaker Efficiency 93 %
LS_VDD = 4.2V
PO= 300mW, f = 1kHz,
THD+N Total Harmonic Distortion + Noise 0.03 %
RL= 8, Mono Input Signal
RL= 8, f = 1kHz, THD+N = 1%, Mono Input Signal
LS_VDD = 5V 1.4 W
LS_VDD = 4.2V 970 mW
LS_VDD = 3.3V 590 510 mW (min)
POOutput Power RL= 4, f = 1kHz, THD+N = 1%,
Mono Input Signal
LS_VDD = 5V 2.4 W
LS_VDD = 4.2V 1.65 W
LS_VDD = 3.3V 980 mW
VRIPPLE = 200mVP-P
fRIPPLE = 217Hz
Mono Input Terminated 75 60 dB (min)
VREF = 1.0μF, Input Referred
PSRR Power Supply Rejection Ratio LS Gain = 12dB
VRIPPLE = 200mVP-P
fRIPPLE = 217Hz 74 dB
From DAC, DAC gain = 0dB
Reference = VOUT (1% THD+N )
Mono gain = 0dB, A-weighted
Mono Input Terminated, LS Gain = 8dB
LS_VDD = 4.2V 95 dB
LS_VDD = 3.3V 93 88 dB (min)
SNR Signal-to-Noise Ratio Reference = VOUT (1% THD+N )
DAC Gain = 0dB, A-weighted
fS= 48kHz, OSR = 128
LS Gain = 8dB
LS_VDD = 4.2V 91 dB
LS_VDD = 3.3V 89 dB
eOS Mono gain = 0dB, A-weighted,
Output Noise Mono Input Terminated, Input 43 µV
Referred
VOS Offset Voltage Mono gain = 0dB, from Mono Input 10 50 mV (max)
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Electrical Characteristics
A_VDD = LS_VDD = 3.3V; HP_VDD = D_VDD = I/O_VDD = 1.8V(1)(2) (continued)
The following specifications apply for RL(LS) = 8, RL(HP) = 32, f = 1kHz, unless otherwise specified. Limits apply for TA=
25°C. LM49352 Units
Symbol Parameter Conditions Typical Limit (Limit)
(3) (4)
HEADPHONE AMPLIFIERS
PO= 15mW, f = 1kHz,
THD+N Total Harmonic Distortion + Noise RL= 320.025 0.1 % (max)
Stereo Analog Input Signal
RL= 32, f = 1kHz, THD+N = 1%,
Stereo Analog Input Signal, In phase
HP_VDD = 2.8V 65 mW
HP_VDD = 1.8V 24 20 mW (min)
POHeadphone Output Power RL= 16, f = 1kHz, THD+N = 1%,
Stereo Analog Input Signal, In phase
HP_VDD = 2.8V 73 mW
HP_VDD = 1.8V 24 mW
VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
Mono Input Terminated, Mono gain = 0dB
VREF = 1.0μF, Mono Differential Input Mode,
Ripple applied to AVDD only 100 85 dB (min)
PSRR Power Supply Rejection Ratio Ripple applied to AVDD and HPVDD 88 dB
VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
From DAC, DAC gain = 0dB
Ripple applied to AVDD 81 dB
HPVDD, and DVDD
Reference = VOUT (1% THD+N )
Gain = 0dB, A-weighted 98 93 dB (min)
Stereo Inputs Terminated
SNR Signal to Noise Ratio Reference = VOUT (1% THD+N)
Gain = 0dB, 97 93 dB (min)
A-weighted, I2S Input = Digital Zero
Gain = 0dB, A-weighted, 11 µV
Stereo Inputs Terminated
eOS Output Noise Gain = 0dB, A-weighted, 12 µV
I2S Input = Digital Zero
PO= 7.5mW, f = 1kHz,
XTALK Crosstalk RL= 3285 dB
Stereo Analog Input Signal
ΔACH-CH Channel-to-Channel Gain Matching 0.03 dB
AUX Gain = 0dB 1.4 6 mV (max)
From mono Input
VOS Output Offset Voltage DAC Gain = 0dB 1.6 6 mV (max)
From DAC Input, fMCLK = 12.288MHz
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Electrical Characteristics
A_VDD = LS_VDD = 3.3V; HP_VDD = D_VDD = I/O_VDD = 1.8V(1)(2) (continued)
The following specifications apply for RL(LS) = 8, RL(HP) = 32, f = 1kHz, unless otherwise specified. Limits apply for TA=
25°C. LM49352 Units
Symbol Parameter Conditions Typical Limit (Limit)
(3) (4)
AUXILIARY OUTPUT/EARPIECE AMPLIFIER
AUX_LINE_OUT, f = 1kHz
From Mono In 0.002 %
RL= 5k, VOUT = 1VRMS
THD+N Total Harmonic Distortion Earpiece mode, f = 1kHz
From Mono In 0.02 %
RL= 32BTL, POUT = 20mW
Earpiece mode, f = 1kHz
POUT Output Power 58 50 mW (min)
RL= 32BTL, THD+N = 1%
VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
Mono Input terminated, CREF = 1.0μF 100 dB
AUX_LINE_OUT
PSRR Power Supply Rejection Ratio VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
Mono Input terminated, CREF = 1.0μF100 90 dB
Earpiece mode,
–6dB cut enabled
Gain = 0dB, VREF = VOUT (1% THD+N)
SNR Signal to Noise Ratio 105 dB
A-weighted, Mono Input Terminated
Gain = 0dB, VREF = VOUT (1% THD+N)
OUT Output Noise 7 μV
A-weighted, Mono Input Terminated
MONO gain = 0dB, From Mono Input 4 mV
AUX_LINE_OUT
VOS Output Offset Voltage Gain = 0dB, From Mono Input 4 12 mV (max)
Earpiece mode
TWU Turn-On time PMC Clock = 300kHz 28 ms
STEREO ADC
Mono Differential Input
THD+NADC ADC Total Harmonic Distortion + Noise VIN = 1VRMS, f = 1kHz 0.007 %
Gain = 0dB, fS= 48kHz
HPF On, fS= 48kHz 220 Hz
Lower -3dB Point
PBADC ADC Passband HPF On, Upper -3dB Point 0.41*fSkHz
RADC ADC Ripple OSRDAC = 128 0.1 dB
Reference = VOUT (0dBFS )
Gain = 6dB, 98 dB
A-weighted From MIC, fS= 8kHz
SNRADC ADC Signal to Noise Ratio Reference = VOUT (0dBFS )
Gain = 0dB, 97 dB
A-weighted From Stereo Input,
fS= 48kHz
ADCLEVEL ADC Full Scale Input Level 1.6 VRMS
STEREO DAC
I2S Input, AUXOUT, OSRDAC = 64
THD+NDAC DAC Total Harmonic Distortion + Noise VIN = 500mFFSRMS, f = 1kHz 0.01 %
Gain = 0dB
DACLEVEL DAC Full Scale Output Level 1.08 VRMS
RDAC DAC Ripple 0.1 dB
PBDAC DAC Passband Upper –3dB Point 0.45*fSkHz
SNRDAC DAC Signal to Noise Ratio fS= 48kHz, A-weighted, AUXOUT 103 dB
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Electrical Characteristics
A_VDD = LS_VDD = 3.3V; HP_VDD = D_VDD = I/O_VDD = 1.8V(1)(2) (continued)
The following specifications apply for RL(LS) = 8, RL(HP) = 32, f = 1kHz, unless otherwise specified. Limits apply for TA=
25°C. LM49352 Units
Symbol Parameter Conditions Typical Limit (Limit)
(3) (4)
VOLUME CONTROL
Minimum Gain –46.5 dB
VCRAUX Stereo Input Volume Control Range Maximum Gain 18 dB
Minimum Gain –46.5 dB
VCRMONO MONO Input Volume Control Range Maximum Gain 18 dB
Minimum Gain –76.5 dB
VCRDAC DAC Volume Control Range Maximum Gain 18 dB
Minimum Gain –76.5 dB
VCRADC ADC Volume Control Range Maximum Gain 18 dB
Minimum Gain 6 dB
VCRMIC MIC Volume Control Range Maximum Gain 36 dB
Minimum Gain 0 dB
Loudspeaker Amplifier Volume Control
VCRLS Range Maximum Gain 12 dB
Minimum Gain –18 dB
Headphone Amplifier Volume Control
VCRHP Range Maximum Gain 0 dB
Loudspeaker Amplifier Volume Control 4 dB
SSLS Stepsize
Headphone Amplifier Volume Control Refer to
SSHP dB
Stepsize Table 19
SSAUX AUX Input Volume Control Stepsize 1.5 dB
SSMONO MONO Input Volume Control Stepsize 1.5 dB
SSDAC DAC Volume Control Stepsize 1.5 dB
SSADC ADC Volume Control Stepsize 1.5 dB
SSMIC MIC Volume Control Stepsize 2 dB
SVAUX AUX Volume Setting Variation ±1 dB (max)
SVMONO MONO Volume Setting Variation ±1 dB (min)
SVMIC MIC Volume Setting Variation ±1 dB (max)
ANALOG INPUTS
AUX Gain = 18dB 10 k
AUX_RIN Auxiliary Input Impedance AUX Gain = 0dB 38 k
AUX Gain = –46.5dB 64 k
MONO Gain = 18dB 10 k
MONO_RIN Mono Input Impedance MONO Gain = 0dB 38 k
MONO Gain = –46.5dB 64 k
MIC_RIN Microphone Input Impedance All MIC gain settings 50 k
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FREQUENCY (Hz)
-1
+1
-0.5
+0
+0.5
20 4k50 100 200 500 1k 2k
MAGNITUDE (dB)
10
0.001
0.01
0.1
1
THD+N (%)
FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
0 400 800 1200 1600 2000
0
10
20
30
40
50
60
70
80
90
100
EFFICIENCY (%)
OUTPUT POWER (mW)
LM49352
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Typical Performance Characteristics
Class D Loudspeaker Amplifier Efficiency vs Output Power
THD+N < 10%, RL= 8DAC Frequency Response
Green >> LSVDD = 3.3V fS= 48kHz
Gray >> LSVDD = 4.2V Blue >> OSR = 64
Blue >> LSVDD = 5V Light Blue >> OSR = 128
Figure 8. Figure 9.
DAC Frequency Response DAC THD+N vs Frequency
fS= 8kHz fS= 8kHz, OSR = 128
Blue >> OSR = 64
Light Blue >> OSR = 128 I2S Input = 500mFFS
Figure 10. Figure 11.
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FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
+1
-3
-2
-1
+0
MAGNITUDE (dB)
FREQUENCY (Hz)
-3
+1
-2
-1
+0
20 4 k50 100 200 500 1k 2 k
MAGNITUDE (dB)
10m 120m 50m 100m 200m 500m
INPUT VOLTAGE (FFS)
10
0.001
0.01
0.1
1
THD+N (%)
10
0.001
0.01
0.1
1
THD+N (%)
FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
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Typical Performance Characteristics (continued)
DAC THD+N
vs
DAC THD+N vs Frequency Input Level
fS= 8kHz, OSR = 64 fS= 48kHz, OSR = 64
I2S Input = 500mFFS I2S Input = 1kHz
Figure 12. Figure 13.
Stereo Audio ADC Frequency Response Stereo Audio ADC Frequency Response
fS= 48kHz, OSR = 64 fS= 8kHz, OSR = 128
From MIC, MIC Gain = 6dB, CIN = F From MIC, MIC Gain = 6dB, CIN = F
Figure 14. Figure 15.
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FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
+1
-3
-2
-1
+0
MAGNITUDE (dB)
FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
+1
-3
-2
-1
+0
MAGNITUDE (dB)
FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
+1
-3
-2
-1
+0
MAGNITUDE (dB)
FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
+1
-3
-2
-1
+0
MAGNITUDE (dB)
LM49352
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SNAS467E MARCH 2009REVISED MARCH 2013
Typical Performance Characteristics (continued)
Mono Voice ADC Frequency Response Mono Voice ADC Frequency Response
fS= 48kHz, OSR = 128 fS= 8kHz, OSR = 128
From MIC, MIC Gain = 6dB, CIN = F From MIC, MIC Gain = 6dB, CIN = F
Figure 16. Figure 17.
Mono Voice HPF ADC Frequency Response
fS= 48kHz, OSR = 128
Stereo Audio HPF ADC Frequency Response From MIC, MIC Gain = 6dB, CIN = 1µF
fS= 48kHz, OSR = 128 Gray >> No HPF
From MONO/AUX, MONO>AUX Gain = 0dB, CIN = F Blue >> HPF Mode = '000'
Blue >> No HPF Light Blue >> HPF Mode = '001'
Light Blue >> HPF Mode = '101' Green >> HPF Mode = '010'
Green >> HPF Mode = '110' Light Green >> HPF Mode = '011'
Light Green >> HPF Mode = '111' Yellow >> HPF Mode = '100'
Figure 18. Figure 19.
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10
0.001
0.01
0.1
1
THD+N (%)
1m 25m 10m 50m100m 1
INPUT VOLTAGE (Vrms)
500m
10
0.001
0.01
0.1
1
THD+N (%)
FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
10
0.001
0.01
0.1
1
THD+N (%)
FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
FREQUENCY (Hz)
20 20k50 100 200 500 1k 2k 5k 10k
+1
-3
-2
-1
+0
MAGNITUDE (dB)
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Typical Performance Characteristics (continued)
Mono Voice HPF ADC Frequency Response
fS= 8kHz, OSR = 128
From MIC, MIC Gain = 6dB, CIN = F
Gray >> No HPF
Blue >> HPF Mode = '000'
Light Blue >> HPF Mode = '001'
Green >> HPF Mode = '010' ADC THD+N vs Frequency
Light Green >> HPF Mode = '011' fS= 48kHz, OSR = 128
Yellow >> HPF Mode = '100' From MONO/AUX, MONO/AUX Gain = 6dB, VIN = 1VRMS
Figure 20. Figure 21.
ADC THD+N vs Frequency ADC THD+N vs Input Voltage
fS= 48kHz, OSR = 128 fS= 48kHz, OSR = 128
From MIC, MIC Gain = 6dB, VIN = 500mVRMS From MONO/AUX, MONO/AUX Gain = 0dB, fIN = 1kHz
Figure 22. Figure 23.
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20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
LM49352
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Typical Performance Characteristics (continued)
ADC THD+N vs Input Voltage Loudspeaker THD+N vs Input Voltage
fS= 48kHz, OSR = 128 From MONO/AUX Input, MONO/AUX Gain = 0dB
From MIC, MIC Gain = 6dB, VIN = 1kHz LS Gain = 8dB, LSVDD = 3.3V, POUT = 300VRMS, RL= 8
Figure 24. Figure 25.
Loudspeaker THD+N vs Input Voltage Loudspeaker THD+N vs Input Voltage
From MONO/AUX Input, MONO/AUX Gain = 0dB From MONO/AUX Input, MONO/AUX Gain = 0dB
LS Gain = 8dB, LSVDD = 4.2V, POUT = 300VRMS, RL= 8LS Gain = 8dB, LSVDD = 5V, POUT = 300VRMS, RL= 8
Figure 26. Figure 27.
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-100
0
-90
-80
-70
-60
-50
-40
-30
-20
-10
PSRR (dB)
20 100k100 1k 10k
FREQUENCY (Hz)
0.001
10
0.01
0.1
1
THD+N (%)
10m 3100m 1 2
OUTPUT POWER (W)
20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
0.001
10
0.01
0.1
1
THD+N (%)
10m 3100m 1 2
OUTPUT POWER (W)
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Typical Performance Characteristics (continued)
Loudspeaker THD+N vs Output Power
From MONO/AUX Input, MONO/AUX Gain = 0dB
LS Gain = 8dB, fIN = 1kHz, RL= 8
Loudspeaker THD+N vs Input Voltage Blue >> LSVDD = 3.3V
From MONO/AUX Input, MONO/AUX Gain = 0dB Light Blue >> LSVDD = 4.2V
LS Gain = 8dB, LSVDD = 3.3V, POUT = 300VRMS, RL= 4Green >> LSVDD = 5V
Figure 28. Figure 29.
Loudspeaker THD+N vs Output Power
fS= 48kHz, OSR = 128
From MONO/AUX Input, MONO/AUX Gain = 0dB
LS Gain = 8dB, fIN = 1kHz, RL= 4
Blue >> LSVDD = 3.3V Loudspeaker PSRR vs Frequency
Light Blue >> LSVDD = 4.2V MONO/AUX Input, MONO/AUX Gain = 0dB, LS Gain = 12dB
Green >> LSVDD = 5V LSVDD = 3.3V, VRIPPLE = 200mVPP, Input Referred
Figure 30. Figure 31.
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-100
0
-90
-80
-70
-60
-50
-40
-30
-20
-10
PSRR (dB)
20 100k100 1k 10k
FREQUENCY (Hz)
20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
-100
0
-90
-80
-70
-60
-50
-40
-30
-20
-10
PSRR (dB)
20 100k100 1k 10k
FREQUENCY (Hz)
-100
0
-90
-80
-70
-60
-50
-40
-30
-20
-10
PSRR (dB)
20 100k100 1k 10k
FREQUENCY (Hz)
LM49352
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Typical Performance Characteristics (continued)
Loudspeaker PSRR vs Frequency
fS= 48kHz, OSR = 128 Loudspeaker PSRR vs Frequency
MONO/AUX Input, MONO/AUX Gain = 0dB, LS Gain = 12dB MONO/AUX Input, MONO/AUX Gain = 0dB, LS Gain = 12dB
LSVDD = 4.2V, VRIPPLE = 200mVPP, Input Referred LSVDD = 5V, VRIPPLE = 200mVPP, Input Referred
Figure 32. Figure 33.
Headphone PSRR vs Frequency HeadphoneTHD+N vs Frequency
DAC Input, DAC Gain = 0dB, LS Gain = 12dB MONO/AUX, MONO/AUX Gain = 0dB, HP Gain = 0dB
LSVDD = 3,3V, AVDD = 3.3V, DVDD = 1.8V, VRIPPLE =
200mVPP HPVDD = 3.3V, POUT = 15mW, RL= 32
Ripple on LSVDD, AVDD, DVDD, Input Referred Stereo In Phase
Figure 34. Figure 35.
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20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
0.001 100m
OUTPUT POWER (W)
10
1
10m1m 2m 5m
0.01
0.1
THD + N (%)
20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
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Typical Performance Characteristics (continued)
Headphone THD+N vs Frequency Headphone THD+N vs Frequency
MONO/AUX, MONO/AUX Gain = 0dB, HP Gain = 0dB MONO/AUX, MONO/AUX Gain = 0dB, HP Gain = 0dB
HPVDD = 2.8V, POUT = 15mW, RL= 32HPVDD = 1.8V, POUT = 15mW, RL= 16
Stereo In Phase Stereo In Phase
Figure 36. Figure 37.
Headphone THD+N vs Output Power
Headphone THD+N vs Frequency MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB
MONO/AUX, MONO/AUX Gain = 0dB, HP Gain = 0dB Stereo In Phase, fIN = 1kHz, RL= 32
HPVDD = 2.8V, POUT = 15mW, RL= 16Blue >> HPVDD = 1.8V
Stereo in Phase Light Blue >> HPVDD = 2.8V
Figure 38. Figure 39.
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-120
-20
-110
-100
-90
-80
-70
-60
-50
-40
-30
PSRR (dB)
20 100k100 1k 10k
FREQUENCY (Hz)
-100
0
-90
-80
-70
-60
-50
-40
-30
-20
-10
PSRR (dB)
20 100k100 1k 10k
FREQUENCY (Hz)
-100
0
-90
-80
-70
-60
-50
-40
-30
-20
-10
PSRR (dB)
20 100k100 1k 10k
FREQUENCY (Hz)
0.001 100m
OUTPUT POWER (W)
10
1
10m1m 2m 5m
0.01
0.1
THD + N (%)
LM49352
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Typical Performance Characteristics (continued)
Headphone THD+N vs Output Power
MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB Headphone PSRR vs Frequency
Stereo In Phase, fIN = 1kHz, RL= 16MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB
Blue >> HPVDD = 1.8V HPVDD = 1.8V, AVDD = 3.3V, VRIPPLE = 200mVPP
Light Blue >> HPVDD = 2.8V Ripple on HPVDD, AVDD
Figure 40. Figure 41.
Headphone PSRR vs Frequency
Headphone PSRR vs Frequency DAC Input, DAC Gain = 0dB, HP Gain = 0dB
MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB HPVDD = 1.8V, AVDD = 3.3V, DVDD = 1.8V, VRIPPLE =
HPVDD = 1.8V, AVDD = 3.3V, VRIPPLE = 200mVPP 200mVPP
Ripple on AVDD only Ripple on HPVDD, AVDD, DVDD
Figure 42. Figure 43.
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-120
-20
-110
-100
-90
-80
-70
-60
-50
-40
-30
PSRR (dB)
20 100k100 1k 10k
FREQUENCY (Hz)
0.001 100m
OUTPUT POWER (W)
10
1
10m1m 2m 5m
0.01
0.1
THD + N (%)
20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
20 100 1k 10k 20k
-100
-90
-80
-70
-60
-50
-40
-30
-20
-10
0
CROSSTALK (dB)
FREQUENCY (Hz)
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Typical Performance Characteristics (continued)
Earpiece THD+N vs Frequency
Headphone Crosstalk vs Frequency MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = 0dB
MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB AVDD = 3.3V, POUT = 20mW, RL= 32
HPVDD = 1.8V, AVDD = 3.3V, POUT = 15mW, RL= 32Earpiece Mode
Figure 44. Figure 45.
Earpiece THD+N vs Output Power
MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = 0dB Earpiece PSRR vs Frequency
AVDD = 3.3V, fIN = 1kHz, RL= 32MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = –6dB
Earpiece Mode AVDD = 3.3V, VRIPPLE = 200mVPP, Earpiece Mode
Figure 46. Figure 47.
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Product Folder Links: LM49352
-120
-20
-110
-100
-90
-80
-70
-60
-50
-40
-30
PSRR (dB)
20 100k100 1k 10k
FREQUENCY (Hz)
20 100 1k 20k
FREQUENCY (Hz)
.001
.01
0.1
1
10
THD+N (%)
10k
0.001
10
0.01
0.1
1
THD+N (%)
10m 3100m 1 2
OUTPUT VOLTAGE (VRMS)
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Typical Performance Characteristics (continued)
Auxiliary Output THD+N vs Frequency Auxiliary Output THD+N vs Output Voltage
MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = 0dB MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = 0dB
AVDD = 3.3V, VOUT = 1VRMS, RL= 5kAVDD = 3.3V, fIN = 1kHz, RL= 5k
AUXOUT Mode AUXOUT Mode
Figure 48. Figure 49.
Auxiliary Output PSRR vs FRequency
MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = 0dB
AVDD = 3.3V, VRIPPLE = 200mVPP, AUXOUT Mode
Figure 50.
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SDA
SCL SP
START condition STOP condition
SCL
SDA
data
change
allowed
data
valid
data
change
allowed
data
valid
data
change
allowed
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SYSTEM CONTROL
Method 1. I2C Compatible Interface
I2C SIGNALS
In I2C mode the LM49352 pin SCL is used for the I2C clock SCL and the pin SDA is used for the I2C data signal
SDA. Both these signals need a pull-up resistor according to I2C specification. The I2C slave address for
LM49352 is 00110102.
I2C DATA VALIDITY
The data on SDA line must be stable during the HIGH period of the clock signal (SCL). In other words, state of
the data line can only be changed when SCL is LOW.
Figure 51. I2C Signals: Data Validity
I2C START AND STOP CONDITIONS
START and STOP bits classify the beginning and the end of the I2C session. START condition is defined as SDA
signal transitioning from HIGH to LOW while SCL line is HIGH. STOP condition is defined as the SDA
transitioning from LOW to HIGH while SCL is HIGH. The I2C master always generates START and STOP bits.
The I2C bus is considered to be busy after START condition and free after STOP condition. During data
transmission, I2C master can generate repeated START conditions. First START and repeated START
conditions are equivalent, function-wise.
Figure 52. I2C Start and Stop Conditions
TRANSFERRING DATA
Every byte put on the SDA line must be eight bits long, with the most significant bit (MSB) being transferred first.
Each byte of data has to be followed by an acknowledge bit. The acknowledge related clock pulse is generated
by the master. The transmitter releases the SDA line (HIGH) during the acknowledge clock pulse. The receiver
must pull down the SDA line during the 9th clock pulse, signifying an acknowledge. A receiver which has been
addressed must generate an acknowledge after each byte has been received.
After the START condition, the I2C master sends a chip address. This address is seven bits long followed by an
eight bit which is a data direction bit (R/W). The LM49352 address is 00110102. For the eighth bit, a “0” indicates
a WRITE and a “1” indicates a READ. The second byte selects the register to which the data will be written. The
third byte contains data to write to the selected register.
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SDA
SCL
1
82
3
76
5
8
10
4 9
1 7
ack from slave
ack from slave
w rs r stop
ack from slave ack from masterrepeated start data from slave
start w ack ack rs r ack ack stop
start
SCL
SDA
MSB Chip Address LSB
slave address =
00110102register address = 0x00h
MSB Register 0x00h LSB MSB Data LSB
MSB Chip Address LSB
slave address =
00110102register 0x00h data
ack ack ack ack
start MSB Chip Address LSB w ack MSB Register 0x02h LSB ack MSB Data LSB ack stop
ack from slave ack from slave ack from slave
SCL
SDA
start slave address =
00110102w ack register address = 0x02h ack ack
register 0x02h data stop
ADR6
Bit7 ADR5
bit6 ADR4
bit5 ADR3
bit4 ADR2
bit3 ADR1
bit2 ADR0
bit1 R/W
bit0
MSB LSB
I2C SLAVE address (chip address)
LM49352
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Figure 53. I2C Chip Address
Register changes take effect at the SCL rising edge during the last ACK from slave.
w = write (SDA = “0”)
r = read (SDA = “1”)
ack = acknowledge (SDA pulled down by slave)
rs = repeated start
Figure 54. Example I2C Write Cycle
When a READ function is to be accomplished, a WRITE function must precede the READ function, as shown in
the Read Cycle waveform.
Figure 55. Example I2C Read Cycle
Figure 56. I2C Timing Diagram
I2C TIMING PARAMETERS
Limit
Symbol Parameter Units
Min Typ Max
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1 Hold Time (repeated) START Condition 0.6 µs
2 Clock Low Time 1.3 µs
3 Clock High Time 600 ns
4 Setup Time for a Repeated START Condition 600 ns
Data Hold Time (Output direction, delay generated by LM49352) 50 See(1) ns
5Data Hold Time (Input direction, delay generated by the Master) 50 ns
6 Data Setup Time 100 ns
7 Rise Time of SDA 300 ns
8 Fall Time of SDA 300 ns
9 Set-up Time for STOP condition 600 ns
10 Bus Free Time between a STOP and a START Condition 1.3 µs
(1) Data ensured by fall time.
Device Register Map
Table 1. Device Register Map
Address Register 7 6 5 4 3 2 1 0
BASIC SETUP
PMC CHIP PORT2 PORT1 MCLK OSC PLL CHIP
PLL_P2
0x00h ENB
SETUP ACTIVE CLK OVR CLK OVR OVR ENB ENB ENABLE
PMC
0x01h PMC_CLK_SEL
CLOCKS
PMC
0x02h PMC_CLK_DIV(R)
CLK_DIV
PLL
0x03h PLL_CLK_SEL
0x04h PLL M PLL M
0x05h PLL N PLL N
PLL
0x06h PLL P2[8] PLL P1[8] PLL N_MOD
N_MOD
0x07h PLL P PLL P1 [7:0]
0x08h PLL P2 PLL P2[7:0]
ANALOG MIXER
0x10h CLASSD AUX_LS MONO_LS DACL_LS DACR_LS
HEAD MONO
0x11h AUX_HPL DACL_HPL DACR_HPL
PHONESL _HPL
HEAD MONO DACL DACR
0x12h AUX_HPR
PHONESR _HPR _HPR _HPR
MONO DACR
0x13h AUX_OUT AUX_AUX MIC_AUX DACL_AUX
_AUX _AUX
OUTPUT AUX_LINE_ AUX_NEG_
0x14h LS_LEVEL LR_HP_LEVEL RSVD
OPTIONS OUT 6dB
MONO AUX DACL DACR
0x15h ADC MIC_ADCL MIC_ADCR
_ADCL _ADCR _ADCL _ADCR
0x16h MIC_LVL MUTE SE/DIFF MIC_LEVEL
0x18h AUXL_LVL SE/DIFF AUX_LEVEL
AUXL_MON
0x19h MONO_LV SE/DIFF MONO_LEVEL
O_IN
HP HP SENSE HP SENSE HP
0x1Bh HP SENSE
_SENSE _AUX_D _AUX SENSE_D
ADC
0x20h ADC BASIC DSPONLY ADC_CLK_SEL MUTE_R MUTE_L ADC_OSR MONO
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Table 1. Device Register Map (continued)
Address Register 7 6 5 4 3 2 1 0
ADC
0x21h ADC_CLK_DIV (T)
CLOCK
ADC STEREO
0x23h ADC_MIX_LEVEL_R ADC_MIX_LEVEL_L
_MIXER _LINK
DAC
DAC
0x30h DSPONLY DAC_CLK_SEL MUTE_R MUTE_L DAC_OSR
_BASIC
DAC
0x31h DAC_CLK_DIV (S)
_CLOCK
DIGITAL MIXER
0x40h IPLVL1 PORT2_RX_R_LVL PORT2_RX_L_LVL PORT1_RX_R_LVL PORT1_RX_L_LVL
0x41h IPLVL2 INTERP_L_LVL INTERP_R_LVL ADC_R_LVL ADC_L_LVL
0x42h OPPORT1 MONO SWAP R_SEL L_SEL
0x43h OPPORT2 MONO SWAP R_SEL L_SEL
0x44h OPDAC SWAP ADCR PORT2R PORT1R ADCL PORT2L PORT1L
0x45h OPDECI MXRCLK_SEL R_SEL L_SEL
AUDIO PORT 1
STEREO STEREO
0x50h BASIC _SYNC _SYNC CLK_PH SYNC_MS CLK_MS TX_ENB RX_ENB STEREO
_MODE _PHASE
0x51h CLK_GEN1 CLK_SEL HALF_CYCLE_DIVDER
SYNTH
0x52h CLK_GEN2 SYNTH_NUM
_DENOM
SYNC
0x53h SYNC_WIDTH(MONO MODE) SYNC_RATE
_GEN
DATA
0x54h TX_EXTRA_BITS TX_WIDTH RX_WIDTH
_WIDTH
0x55h RX_MODE A/ULAW COMPAND MSB_POSITION RX_MODE
0x56h TX_MODE A/ULAW COMPAND MSB_POSITION TX_MODE
AUDIO PORT 2
STEREO STEREO
0x60h BASIC _SYNC _SYNC CLK_PH SYNC_MS CLK_MS TX_ENB RX_ENB STEREO
_MODE _PHASE
0x61h CLK_GEN1 CLK_SEL HALF_CYCLE_DIVDER
SYNTH_DE
0x62h CLK_GEN2 SYNTH_NUM
NOM
SYNC
0x63h SYNC_WIDTH (MONO MODE) SYNC_RATE
_GEN
DATA
0x64h TX_EXTRA_BITS TX_WIDTH RX_WIDTH
_WIDTH
0x65h RX_MODE A/ULAW COMPAND MSB_POSITION RX_MODE
0x66h TX_MODE A/ULAW COMPAND MSB_POSITION TX_MODE
EFFECTS ENGINE
ADC ADC ADC ADC ADC
0x70h ADC FX SCLP ENB EQ ENB PK ENB ALC ENB HPF_ENB
DAC DAC DAC DAC
0x71h DAC FX RSVD
SCLP ENB EQ ENB PK ENB ALC ENB
ADC EFFECTS
0x80h HPF HPF MODE
ADC SOURCE SOURCE SOURCE STEREO
0x81h LIMITER ADC_SAMPLE
ALC 1 OVR RSEL LSEL LINK
ADC
0x82h NG_ENB NOISE_FLOOR
ALC 2
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Table 1. Device Register Map (continued)
Address Register 7 6 5 4 3 2 1 0
ADC
0x83h ALC_TARGET_LEVEL
ALC 3
ADC
0x84h ATTACK_RATE
ALC 4
ADC
0x85h PK_DECAY_RATE DECAY_RATE/RELEASE_RATE
ALC 5
ADC
0x86h HOLDTIME
ALC 6
ADC
0x87h MAX_LEVEL
ALC 7
ADC
0x88h MIN_LEVEL
ALC 8
ADC L STEREO
0x89h ADC_L_LEVEL
LEVEL LINK
ADC R
0x8Ah ADC_R_LEVEL
LEVEL
0x8Bh EQ BAND 1 LEVEL FREQ
0x8Ch EQ BAND 2 Q LEVEL FREQ
0x8Dh EQ BAND 3 Q LEVEL FREQ
0x8Eh EQ BAND 4 Q LEVEL FREQ
0x8Fh EQ BAND 5 LEVEL FREQ
SOFTCLIP SOFT
0x90h THRESHOLD
1 KNEE
SOFTCLIP
0x91h RATIO
2
SOFTCLIP
0x92h LEVEL
3
ADC EFFECT MONITORS
0x98h LVLMONL ADC LEFT LEVEL MONITOR
0x99h LVLMONR ADC RIGHT LEVEL MONITOR
SCLP SCLP EQ EQ GAIN GAIN ADC ADC
0x9Ah FXCLIP _R CLIP _L CLIP _R CLIP _L CLIP _R CLIP _L CLIP _R CLIP _L CLIP
SCLP_R SCLP_L
0x9Bh ALCMONL ADC LEFT ALC MONITOR
DISTORT DISTORT
SCLP_L SCLP_R
0x9Ch ALCMONR ADC RIGHT ALC MONITOR
DISTORT DISTORT
DAC EFFECTS
DAC STEREO
0xA0h LIMITER DAC_SAMPLE
ALC 1 LINK
DAC
0xA1h NG_ENB NOISE_FLOOR
ALC 2
DAC
0xA2h AGC_TARGET_LEVEL
ALC 3
DAC
0xA3h ATTACK_RATE
ALC 4
DAC
0xA4h PK_DECAY_RATE DECAY_RATE/RELEASE_RATE
ALC 5
DAC
0xA5h HOLDTIME
ALC 6
DAC
0xA6h MAX_LEVEL
ALC 7
DAC
0xA7h MIN_LEVEL
ALC 8
DAC L STEREO
0xA8h DAC_L_LEVEL
LEVEL LINK
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Table 1. Device Register Map (continued)
Address Register 7 6 5 4 3 2 1 0
DAC R
0xA9h DAC_R_LEVEL
LEVEL
0xABh EQ BAND 1 LEVEL FREQ
0xACh EQ BAND 2 Q LEVEL FREQ
0xADh EQ BAND 3 Q LEVEL FREQ
0xAEh EQ BAND 4 Q LEVEL FREQ
0xAFh EQ BAND 5 LEVEL FREQ
SOFTCLIP SOFT
0xB0h THRESHOLD
1 KNEE
SOFTCLIP
0xB1h RATIO
2
SOFTCLIP
0xB2h LEVEL
3
DAC EFFECT MONITORS
0xB8h LVLMONL DAC LEFT LEVEL MONITOR
0xB9h LVLMONR DAC RIGHT LEVEL MONITOR
SCLP SCLP EQ EQ GAIN GAIN
0xBAh FXCLIP RSVD RSVD
_R CLIP _L CLIP _R CLIP _L CLIP _R CLIP _L CLIP
SCLP_R SCLP_L
0xBBh ALCMONL DAC LEFT ALC MONITOR
DISTORT DISTORT
SCLP_L SCLP_R
0xBCh ALCMONR DAC RIGHT ALC MONITOR
DISTORT DISTORT
GPIO
0xE0h GPIO1 GPIO_RX GPIO_TX GPIO_MODE
0xE1h GPIO2 TEMP SHORT
SPREAD SPECTRUM
SOFT
0xF0h RESET RSVD RSVD RSVD RSVD RSVD
_RESET SS
0xF1h SS RSVD RSVD
_DISABLE
0xFEh FORCE CPFORCE DACREF RSVD
Unless otherwise specified, the default values of the I2C registers is 0x00h.
Table 2. Nonzero I2C Default Registers
Address Register Default Data Value
0x02h PMC_CLK_DIV 0x50h
0x30h DAC_BASIC 0x02h
0x31h DAC_CLOCK 0x03h
0x84h ADC_ALC_4 0x0Ah
0x85h ADC_ALC_5 0x0Ah
0x86h ADC_ALC_6 0x0Ah
0x87h ADC_ALC_7 0x1Fh
0x89h ADC_L_LEVEL 0x33h
0x8Ah ADC_R_LEVEL 0x33h
0xA3h DAC_ALC_4 0x0Ah
0xA4h DAC_ALC_5 0x0Ah
0xA5h DAC_ALC_6 0x0Ah
0xA6h DAC_ALC_7 0x33h
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Table 2. Nonzero I2C Default Registers (continued)
Address Register Default Data Value
0xA8h DAC_L_LEVEL 0x33h
0xA9h DAC_R_LEVEL 0x33h
0xF0h RESET 0x02h
Basic PMC Setup Register
This register is used to control the LM49352's Basic Power Management Setup:
Table 3. PMC_SETUP (0x00h)
Bits Field Description
When this bit is set the power management will enable the MCLK I/O or internal oscillator(1). It will then
use this clock to sequence the enabling of the analog references and bias points. When this bit is
cleared the PMC will bring the analog down gently and disable the MCLK or oscillator.
0 CHIP_ENABLE CHIP _ENABLE Chip Status
0 Turn Chip Off
1 Turn Chip On
This enables the PLL.
PLL_ENABLE PLL Status
1 PLL_ENB 0 PLL Off
1 PLL On
This enables the P2 output of the PLL.
PLL_P2ENB PLL P2 Status
2 PLL_P2ENB 0 PLL P2 Off
1 PLL P2 On
This enables the internal 300kHz Oscillator. For analog only chip modes, the oscillator can be used
instead of an external system clock to drive the chip's power management (PMC).
OSC_ENABLE Oscillator Status
3 OSC_ENB 0 Oscillator Off
1 Oscillator On
This forces the MCLK input to enable, regardless of requirement. If set, the audio ports and digital mixer
can be activated even if the chip is in shutdown mode. This assumes that MCLK is selected as the PMC
clock source (reg 0x01h) and that there is an active clock signal driving the MCLK pin. Setting this bit
reduces power consumption, by allowing audio ports and digital mixer to operate while the analog
sections of the chip are powered down.
4 MCLK_OVR MCLK_OVR Comment
0 I/O control is automatic
1 MCLK input forced on.
This forces the clock input of Audio Port 1 input to enable, regardless of other port settings.
PORT1_CLK_OVR Comment
5 PORT1_CLK_OVR 0 I/O control is automatic
1 PORT_CLK input forced on
This forces the clock input of Audio Port 2 input to enable, regardless of other port settings.
PORT2_CLK_OVR Comment
6 PORT2_CLK_OVR 0 I/O control is automatic
1 PORT_CLK input forced on
7 CHIP_ACTIVE This bit is used to readback the enable status of the chip.
(1) If the PMC is set to operate from one of the audio ports then it will wait for the port to be enabled or the relevant override bit to be set,
forcing the port clock input to enable.
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PMC Clocks Register
This register is used to control the LM49352's Basic Power Management Clock:
Table 4. PMC_SETUP (0x01h)
Bits Field Description
1:0 PMC_CLK_SEL This selects the source of the PMC input clock.
PMC_CLK_SEL PMC Input Clock Source
00 MCLK (Default divide is 40.5)
01 Internal 300kHz Oscillator
10 DAC SOURCE CLOCK
11 ADC SOURCE CLOCK
PMC Clock Divide Register
This register is used to control the LM49352's Power Management Circuit Clock Divider:
Table 5. PMC_SETUP (0x02h)
Bits Field Description
7:0 PMC_CLK_DIV This programs the half cycle divider that precedes the PMC. The PMC should run from a 300kHz
clock. The default of this divider is 0x50h (divide by 40.5) to get a 300kHz PMC clock from a 12MHz
or 12.288MHz MCLK.
Program this divider with the required division, multiplied by 2, and subtract 1.
PMC_CLK_DIV Divide by
00000000 1
00000001 1
00000010 1.5
00000011 2
00000100 2.5
00000101 3
11111101 126
11111110 127.5
11111111 128
LM49352 Clock Network
(Refer to Figure 57)
The audio DAC and ADC operate at a clock frequency of 2*OSR*fSwhere OSR is the oversampling ratio and fS
is the sampling frequency of the DAC or ADC. The DAC can operate at three different OSR settings (128, 125,
64). The ADC can operate at two different OSR settings (128, 125). For example, if the stereo DAC or ADC is
set at OSR = 128, a 12.288MHz clock is required for 48kHz data. If a 12.288MHz clock is not available, then the
internal PLL can be used to generate the desired clock frequency. Otherwise, if a 12.288MHz is available, the
PLL can be bypassed to reduce power consumption. The DAC clock divider or ADC clock divider can also be
used to generate the correct clock. If an 18.432 MHz clock is available, the DAC or ADC clock divider could be
set to 1.5 in order to generate a 12.288MHz clock from 18.432MHz without using a PLL.
The DAC path clock (DAC_SOURCE_CLK) and ADC path clock (ADC_SOURCE_CLK) can be driven directly by
the MCLK input, the PORT1_CLK input, the PORT2_CLK input, or PLL output.
For instances where a PLL must be used, the PLL input clock can come from three sources. The clock input to
the PLL can come from the MCLK input, the PORT1_CLK input, or the PORT2_CLK input.
The LM49352's Power Management Circuit (PMC) requires a clock that is independent from the DAC or ADC. It
is recommended to provide a 300kHz clock at Point C. The PMC clock divider is available to generate the
correct clock to the PMC block. The PMC clock path can be driven directly by the MCLK input, the internal
300kHz oscillator, the DAC_SOURCE_CLK, or the ADC_SOURCE_CLK.
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300 kHz
MCLK
0_> 50 MHz
0_50 MHz
OSCILLATOR
INTERNAL
PORT2_CLK
AUDIO
PORT 1
PMC
Stereo DAC
MIXER
Stereo ADC
PLL
% DAC
1 ± 25 MHz
PMC, DAC, ADC divides by 1, 1.5, 2,
2.5 ==> 128
% ADC
% PMC
AUDIO
PORT 2
PORT1_CLK
0_50 MHz
A
B
C
% 1/2
Cycle
% 1/2
Cycle Half Cycle divides by 1, 1.5, 2, 2.5 ==> 32
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Table 6. DAC Clock Requirements
DAC Sample Rate Clock Required at A Clock Required at A Clock Required at A Clock Required at A
(kHz) (OSR = 128) (OSR = 125) (OSR = 64) (OSR = 32)
8 2.048 MHz 2 MHz 1.024 MHz 0.512 MHz
11.025 2.8224 MHz 2.75625 MHz 1.4112 MHz 0.7056 MHz
12 3.072 MHz 3 MHz 1.536 MHz 0.768 MHz
16 4.096 MHz 4 MHz 2.048 MHz 1.024 MHz
22.05 5.6448 MHz 5.5125 MHz 2.8224 MHz 1.4112 MHz
24 6.144 MHz 6 MHz 3.072 MHz 1.536 MHz
32 8.192 MHz 8 MHz 4.096 MHz 2.048MHz
44.1 11.2896 MHz 11.025 MHz 5.6448 MHz 2.8224 MHz
48 12.288 MHz 12 MHz 6.144 MHz 3.072 MHz
96 24.576 MHz 24 MHz 12.288 MHz 6.144 MHz
Table 7. ADC Clock Requirements
ADC Sample Rate Clock Required at B Clock Required at B
(kHz) (OSR = 128) (OSR = 125)
8 2.048 MHz 2 MHz
11.025 2.8224 MHz 2.75625 MHz
12 3.072 MHz 3 MHz
16 4.096 MHz 4 MHz
22.05 5.6448 MHz 5.5125 MHz
24 6.144 MHz 6 MHz
32 8.192 MHz 8 MHz
44.1 11.2896 MHz 11.025 MHz
48 12.288 MHz 12 MHz
Figure 57. Internal Clock Network
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9
PLL_P2
1 ± 25 MHz
1 ± 25 MHz
PLL_P1
8
PLL_M PLL_N_MOD
7
5
9
% N
% P1
% P2
VCO
6'M
140 to
210 MHz
0.7 < 5 MHz
% M
N = 10, 11, 12 _> 250
Phase Comparator
and Charge Pump
M = 1, 1.5, 2 _> 64
0.5 - 50 MHz
PLL_N
I
8
P1 = 1,1.5, 2 _> 256
P2 = 1,1.5, 2 _> 256
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PLL Setup Registers
Figure 58. PLL Loop
The LM49352 contains a PLL for flexible operation of its dual audio ports. The PLL has a P1 and P2 output
divider thereby allowing the PLL to generate two distinct clock outputs. The equations for the PLL's generated
output clocks are as follows:
fOUT1 = (fIN . N / M . P1)
fOUT2 = (fIN . N / M . P2)
where: N = PLL_N + PLL_N_MOD
M = (PLL_M + 1) / 2
P1= (PLL_P1 + 1) / 2
P2= (PLL_P2 + 1) / 2
The VCO frequency and comparison frequencies are as follows:
fVCO = fOUT.P
fCOMP = fIN/M
Keep fVCO between 140MHz to 240MHz and keep fCOMP between 700KHz to 5MHz.
Table 8. PLL Settings for Common System Clock Frequencies
fIN (MHz) M N N_MOD P fOUT (Hz) Error (Hz)
12 2.5 32 0 12.5 12288000 0
13 15.5 175 26 12 12287970 30
14.4 12.5 128 0 12 12288000 0
16.2 13.5 128 0 12.5 12288000 0
16.8 3.5 32 0 12.5 12288000 0
19.2 12.5 96 0 12 12288000 0
19.68 20.5 160 0 12.5 12288000 0
19.8 16.5 128 0 12.5 12288000 0
26 32.5 192 0 12.5 12288000 0
27 22.5 128 0 12.5 12288000 0
12 12.5 147 0 12.5 11289600 0
12.288 10 147 0 16 11289600 0
13 9 144 19 18.5 11289603 +3
14.4 12.5 147 0 15 11289600 0
16.2 22.5 196 0 12.5 11289600 0
16.8 12.5 126 0 15 11289600 0
19.2 20 147 0 12.5 11289600 0
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Table 8. PLL Settings for Common System Clock Frequencies (continued)
fIN (MHz) M N N_MOD P fOUT (Hz) Error (Hz)
19.68 20.5 147 0 12.5 11289600 0
19.8 27.5 196 0 12.5 11289600 0
26 18.5 144 19 18 11289602.1 2.1
27 37.5 196 0 12.5 12289600 0
11.2896 10.5 195 0 17.5 12000000 0
12.288 8 125 0 16 12000000 0
13 6.5 102 0 17 12000000 0
13.5 4.5 68 0 17 12000000 0
14.4 6 85 0 17 12000000 0
16.2 13.5 170 0 17 12000000 0
16.8 7 85 0 17 12000000 0
19.2 8 85 0 17 12000000 0
19.68 20.5 200 0 16 12000000 0
19.8 16.5 170 0 17 12000000 0
26 6.5 36 0 12 12000000 0
11.2896 8 125 0 16 11025000 0
12 10 147 0 16 11025000 0
12.288 8 114 27 16 11025000 0
13 6.5 96 15 17.5 11025000 0
13.5 10 147 0 18 11025000 0
14.4 4 49 0 16 11025000 0
16.2 4 49 0 18 11025000 0
16.8 16 189 0 18 11025000 0
19.2 16 147 0 16 11025000 0
19.68 16 189 0 18 11025000 0
19.8 16 147 0 16.5 11025000 0
26 5 27 18 13 1102500 0
Table 9. PLL_CLOCK_SOURCE (0x03h)
Bits Field Description
1:0 PLL_CLK_SEL This selects the source of the input clock to the PLL.
PLL_CLK_SEL PLL Input Clock Source
00 MCLK
01 PORT1_CLK
10 PORT2_CLK
11 RESERVED
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Table 10. PLL_M (0x04h)
Bits Field Description
6:0 PLL_M This programs the PLL's M divider to divide from 1 to 64.
PLL_M PLL Input Divider Value
000000 1
000001 1
000010 1.5
000011 2
000100 2.5
000101 3
1111101 63
1111110 63.5
1111111 64
Table 11. PLL_N (0x05h)
Bits Field Description
7:0 PLL_N This programs the PLL N divider to divide from 1 to 250.
PLL_N Feedback Divider Value
00000000 to 00001010 10
00001011 11
00001100 12
00001101 13
00001110 14
00001111 15
11111000 248
11111001 249
11111010 to 11111111 250
Table 12. PLL_N_MOD (0x06h)
Bits Field Description
4:0 PLL_N_MOD This programs the sigma-delta modulator in the PLL.
PLL_N_MOD Fractional Part of N
00000 0
00001 1/32
00010 2/32
00011 3/32
00100 4/32
00101 5/32
11101 20/32
11110 30/32
11111 31/32
5 PLL_P1[8] This sets the MSB of the 1st P Divider on the PLL which is part of a standard half-cycle divider control.
6 PLL_P2[8] This sets the MSB of the 2nd P Divider on PLL which is part of a standard half-cycle divider control.
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Table 13. PLL_P1 (0x07h)
Bits Field Description
7:0 PLL_P1[7:0] This programs the 8 LSBs of the PLL's P1 Divider. These LSBs combine with PL1_P1[8] which allows
the P1 divider to divide by up to 256.
PLL_P1 [8:0] P1 Divider Value
000000000 1
000000001 1
000000010 1.5
000000011 2
000000100 2.5
000000101 3
111111101 255
111111110 255.5
111111111 256
Table 14. PLL_P2 (0x08h)
Bits Field Description
7:0 PLL_P2[7:0] This programs 8 LSBs of the PLL's P2 Divider. These LSBs combine with PLL_P2[8] which allows the
P2 divider to divide by up to 256.
PLL_P2 [8:0] P2 Divider Value
000000000 1
000000001 1
000000010 1.5
000000011 2
000000100 2.5
000000101 3
111111101 255
111111110 255.5
111111111 256
Analog Mixer Control Registers
This register is used to control the LM49352's Analog Mixer:
Table 15. CLASS_D_OUTPUT (0x10h)
Bits Field Description
0 DACR_LS The right DAC output is added to the loudspeaker output.
1 DACL_LS The left DAC output is added to the loudspeaker output.
2 RSVD Reserved
3 RSVD Reserved
4 MONO_LS The MONO input is added to the loudspeaker output.
5 AUX_LS The AUX input is added to the loudspeaker output.
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Class D Loudspeaker Amplifier
The LM49352 features a filterless modulation scheme. The differential outputs of the device switch at 300kHz
from VDD to GND. When there is no input signal applied, the two outputs (LS+ and LS-) switch with a 50% duty
cycle, with both outputs in phase. Because the outputs of the LM49352 are differential, the two signals cancel
each other. This results in no net voltage across the speaker, thus there is no load current during an idle state,
conserving power.
With an input signal applied, the duty cycle (pulse width) of the LM49352 outputs changes. For increasing output
voltages, the duty cycle of LS+ increases, while the duty cycle of LS- decreases. For decreasing output voltages,
the converse occurs, the duty cycle of LS- increases while the duty cycle of LS+ decreases. The difference
between the two pulse widths yields the differential output voltage.
Spread Spectrum Modulation
The LM49352 features a fitlerless spread spectrum modulation scheme that eliminates the need for output filters,
ferrite beads or chokes. The switching frequency varies by ±30% about a 300kHz center frequency, reducing the
wideband spectral content, improving EMI emissions radiated by the speaker and associated cables and traces.
Where a fixed frequency class D exhibits large amounts of spectral energy at multiples of the switching
frequency, the spread spectrum architecture of the LM49352 spreads that energy over a larger bandwidth. The
cycle-to-cycle variation of the switching period does not affect the audio reproduction or efficiency.
Class D Power Dissipation and Efficiency
In general terms, efficiency is considered to be the ratio of useful work output divided by the total energy required
to produce it with the difference being the power dissipated, typically, in the IC. The key here is “useful” work. For
audio systems, the energy delivered in the audible bands is considered useful including the distortion products of
the input signal. Sub-sonic (DC) and super-sonic components (>22kHz) are not useful. The difference between
the power flowing from the power supply and the audio band power being transduced is dissipated in the
LM49352 and in the transducer load. The amount of power dissipation in the LM49352's class D amplifier is very
low. This is because the ON resistance of the switches used to form the output waveforms is typically less than
0.25. This leaves only the transducer load as a potential "sink" for the small excess of input power over audio
band output power. The LM49352 dissipates only a fraction of the excess power requiring no additional PCB
area or copper plane to act as a heat sink.
EMI/RFI Filtering
If system level PCB layout constraints require the LM49352’s Class D output bumps to be placed far away from
the speaker or the Class D output traces to be routed near EMI/RFI sensitive components, an external EMI/RFI
filter should be used. A series ferrite bead placed close to the Class D output bumps along with a shunt capacitor
to ground placed close to the ferrite bead will reduce the EMI/RFI emissions of the Class D amplifier’s switching
outputs. The ferrite bead must be rated with a current rating high enough to properly drive the loudspeaker. The
ferrite bead that is rated for 1A or greater is recommended. The DC resistance of the ferrite bead is another
important specification that must be taken into consideration. A low DC resistance will minimize any power losses
dissipated by the EMI/RFI filter thereby preserving the power efficiency advantages of the Class D amplifier.
Selecting a ferrite bead with high DC resistance will decrease output power delivered to speaker and reduce the
Class D amplifier’s efficiency. The shunt capacitor needs to have low ESR. A 10pF ceramic capacitor with a X7R
dielectric is recommended as a starting point. Care needs to be taken to ensure that the value of the shunt
capacitor does not exceed 47pF when using a low resistance ferrite bead in order to prevent permanent damage
to the low side FETs of the Class D output stage.
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10 pF Speaker
Ferrite Bead
LSGND
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Class D LS-
LS+
10 pF
Ferrite Bead
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Figure 59. EMI/RFI Filter for the Class D Amplifier
Table 16. LEFT HEADPHONE_OUTPUT (0x11h)
Bits Field Description
0 DACR_HPL The right DAC output is added to the left headphone output.
1 DACL_HPL The left DAC output is added to the left headphone output.
2 RSVD Reserved
3 RSVD Reserved
4 MONO_HPL The MONO input is added to the left headphone output.
5 AUX_HPL The AUX input is added to the left headphone output.
Table 17. RIGHT HEADPHONE_OUTPUT (0x12h)
Bits Field Description
0 DACR_HPR The right DAC output is added to the right headphone output.
1 DACL_HPR The left DAC output is added to the right headphone output.
2 RSVD Reserved
3 RSVD Reserved
4 MONO_HPR The MONO input is added to the right headphone output.
5 AUX _HPR The AUX input is added to the right headphone output.
Headphone Amplifier Function
The LM49352 headphone amplifier features TI’s ground referenced architecture that eliminates the large DC-
blocking capacitors required at the outputs of traditional headphone amplifiers. A low-noise inverting charge
pump creates a negative supply (HP_VSS) from the positive supply voltage (LS_VDD). The headphone amplifiers
operate from these bipolar supplies, with the amplifier outputs biased about GND, instead of a nominal DC
voltage (typically VDD/2), like traditional amplifiers. Because there is no DC component to the headphone output
signals, the large DC-blocking capacitors (typically 220μF) are not necessary, conserving board space and
system cost, while improving frequency response.
Charge Pump Capacitor Selection
Use low ESR ceramic capacitors (less than 100m) for optimum performance.
Charge Pump Flying Capacitor (C6)
The flying capacitor (C6) affects the load regulation and output impedance of the charge pump. A C6 value that
is too low results in a loss of current drive, leading to a loss of amplifier headroom. A higher valued C6 improves
load regulation and lowers charge pump output impedance to an extent. Above 2.2μF, the RDS(ON) of the charge
pump switches and the ESR of C6 and C5 dominate the output impedance. A lower value capacitor can be used
in systems with low maximum output power requirements. Please refer to the demonstration board schematic
shown in Figure 72.
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Charge Pump Flying Capacitor (C5)
The value and ESR of the hold capacitor (C5) directly affects the ripple on CPVSS. Increasing the value of C5
reduces output ripple. Decreasing the ESR of C5 reduces both output ripple and charge pump output impedance.
A lower value capacitor can be used in systems with low maximum output power requirements. Please refer to
the demonstration board schematic shown in Figure 72.
Table 18. AUX_OUTPUT (0x13h)
Bits Field Description
0 DACR_AUX The right DAC output is added to the AUX output.
1 DACL_AUX The left DAC output is added to the AUX output.
2 MIC_AUX The MIC input is added to the AUX output.
3 RSVD Reserved
4 MONO_AUX The MONO input is added to the AUX output.
5 AUX_AUX The AUX input is added to the AUX output.
Auxiliary Output Amplifier
The LM49352’s auxiliary output (AUXOUT) amplifier provides differential drive capability to loads that are
connected across its outputs. This results in output signals at the AUX_OUT+ and AUX_OUT- pins that are 180
degrees out of phase with respect to each other. This effectively doubles the maximum possible output swing for
a specific supply voltage when compared to single-ended output configurations. The differential output
configuration also allows the load to be isolated from ground since both the AUX_OUT+ and AUX_OUT- pins are
biased at the same DC potential. This eliminates the need for any large and expensive DC blocking capacitors at
the AUXOUT amplifier outputs. The load can then be directly connected to the positive and negative outputs of
the AUXOUT amplifier which then isolates it from any ground noise, thereby improving signal to noise ratio
(SNR) and power supply rejection ratio (PSRR).
The AUXOUT amplifier has two modes of operation. The primary mode of operation is high current drive mode
(Earpiece Mode) where the AUXOUT amplifier can be used to differentially drive a mono earpiece speaker. The
secondary mode of operation is low current drive mode where the AUXOUT amplifier operates in a power saving
mode (AUX_LINE_OUT Mode) to provide a differential output that is used as a mono differential line level input
to a standalone mono differential input class D amplifier (LM4675) for stereo loudspeaker applications.
Table 19. OUTPUT_OPTIONS (0x14h)
Bits Field Description
0 RSVD Reserved
This sets the gain of the left and right headphone amplifiers. The gain of the left and right headphone
amplifiers are always set to the same level.
LR_HP_LEVEL Gain (dB)
000 0
001 –1.5
010 –3
3:1 LR_HP_LEVEL 011 –6
100 –9
101 –12
110 –15
111 –18
This sets the gain of the Auxiliary output amplifier.
AUX_NEG_6dB Gain (dB)
4 AUX_NEG_6dB 0 0
1 –6
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Table 19. OUTPUT_OPTIONS (0x14h) (continued)
Bits Field Description
This sets the Auxiliary output amplifier mode of operation.
AUX_LINE_OUT Auxiliary Output Mode
5 AUX_LINE_OUT 0 Earpiece Amplifier
1 AUX_LINE_OUT
This sets the gain of the Class D loudspeaker amplifier.
LS_LEVEL Gain (dB)
00 0
7:6 LS_LEVEL 01 4
10 8
11 12
Table 20. ADC_INPUT (0x15h)
Bits Field Description
0 DACR_ADCR The right DAC output is added to the ADC right input.
1 DACL_ADCL The left DAC output is added to the ADC left input.
2 MIC_ADCR The MIC input is added to the ADC right input.
3 MIC_ADCL The MIC input is added to the ADC left input.
4 AUX_ADCR The AUX input is added to the ADC right input.
5 MONO_ADCL The MONO input is added to the ADC left input.
Table 21. MIC_INPUT (0x16h)
Bits Field Description
3:0 MIC_LEVEL This sets the gain of the microphone preamp.
MIC_LEVEL Gain
0000 6dB
0001 8dB
0010 10dB
0011 12dB
0100 14dB
0101 16dB
0110 18dB
0111 20dB
1000 22dB
1001 24dB
1010 26dB
1011 28dB
1100 30dB
1101 32dB
1110 34dB
1111 36dB
4 SE_DIFF If set, the MIC negative input is ignored. In single-ended mode, the MIC negative input pin should
left floating.
5 MUTE If set, the microphone preamp is muted.
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Table 22. AUX_LEVEL (0x18h)
Bits Field Description
5:0 AUX_LEVEL This programs the AUX input level. All gain changes are performed at zero crossings.
AUX_LEVEL Level AUX_LEVEL Level
000000 –46.5dB 100000 1.5dB
000001 –45dB 100001 3dB
000010 –43.5dB 100010 4.5dB
000011 –42dB 100011 6dB
000100 –40.5dB 100100 7.5dB
000101 –39dB 100101 9dB
000110 –37.5dB 100110 10.5dB
000111 –36dB 100111 12dB
001000 –34.5dB 101000 13.5dB
001001 –33dB 101001 15dB
001010 –31.5dB 101010 16.5dB
001011 –30dB 101011 18dB
001100 –28.5dB
001101 –27dB
001110 –25.5dB
001111 –24dB
010000 –22.5dB
010001 –21dB
010010 –19.5dB
010011 –18dB
010100 –16.5dB
010101 –15dB
010110 –13.5dB
010111 –12dB
011000 –10.5dB
011000 –9dB
011001 –7.5dB
011010 –6dB
011100 –4.5dB
011101 –3dB
011110 –1.5dB
011111 0dB
6 SE/DIFF If set, the AUXL input is ignored. In single-ended mode, the AUXL input pin should be left floating.
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Table 23. MONO_LEVEL (0x19h)
Bits Field Description
5:0 MONO_LEVEL This programs the MONO input level. All gain changes are performed at zero crossings.
MONO_LEVEL Level MONO_LEVEL Level
000000 –46.5dB 100000 1.5dB
000001 –45dB 100001 3dB
000010 –43.5dB 100010 4.5dB
000011 –42dB 100011 6dB
000100 –40.5dB 100100 7.5dB
000101 –39dB 100101 9dB
000110 –37.5dB 100110 10.5dB
000111 –36dB 100111 12dB
001000 –34.5dB 101000 13.5dB
001001 –33dB 101001 15dB
001010 –31.5dB 101010 16.5dB
001011 –30dB 101011 18dB
001100 –28.5dB
001101 –27dB
001110 –25.5dB
001111 –24dB
010000 –22.5dB
010001 –21dB
010010 –19.5dB
010011 –18dB
010100 –16.5dB
010101 –15dB
010110 –13.5dB
010111 –12dB
011000 –10.5dB
011000 –9dB
011001 –7.5dB
011010 –6dB
011100 –4.5dB
011101 –3dB
011110 –1.5dB
011111 0dB
6 SE/DIFF If set, the MONO– input is ignored. In single-ended mode, the MONO- input pin should be left floating.
7 AUXL_MONO If set, AUXL is routed to the MONO Input Amplifier.
Headphone Detection Circuit
The LM49352 features a headphone detection circuit (HDC) that automatically enables the headphone amplifier
whenever the insertion of a headphone plug is detected and disables the headphone amplifier during the removal
of a headphone plug. The HDC optimizes power management by automatically disabling any output amplifier
that is not in use. The HDC eliminates the necessity of polling the I2C bus for status changes. However, since the
HDC requires the use of the GPIO pin, the PORT2_SDO functionality sensing is required.
The HDC requires a headphone jack with a normally closed mechanical switch and a pullup resistor, RPU, tied
between the mechanical switch and I/O_VDD (Refer to Figure 60). Choosing a RPU value of at least 500k
ensures minimal current draw through the pullup resistor. When the headphone amplifier is disabled, an internal
50kpulldown, RPD, is connected to each headphone amplifier output. Without the presence of a headphone
plug, the headphone jack’s mechanical switch is closed thereby connecting the right headphone amplifier output
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HPR
HPL
RPU
500 k:
RPD
50 k:
I/OVDD
GPIO
RPD
50 k:
Headphone Jack
with a Normally Closed
(NC)
Mechanical Switch
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to RPU. The GPIO pin detects a logic low level due to the voltage division between RPU and RPD. When the GPIO
pin is set to HPSENSE mode, a logic low voltage reading causes the HDC to disable the headphone amplifier.
When a headphone plug is inserted, the mechanical connection between RPU and RPD is broken, resulting in a
logic high level detected by the GPIO pin. A logic high voltage reading causes the HDC to enable the headphone
amplifier.
The HDC has four modes of operation that automatically enable/disable different combinations of the audio
output amplifiers contained within the LM49352. Having the choice of four different HDC settings maximizes
power management flexibility to suit a particular application. Please refer to the HP_SENSE (reg 0x1Bh) register
table for a detailed discussion on the different HDC modes of operation.
Figure 60. Application Circuit for Headphone Detection
Table 24. HP_SENSE (0x1Bh)
Bits Field Description
0 HP SENSE This bit enables the headphone sense circuit. If enabled, the headphone amplifier will automatically
turn on/off based on the logic level of the GPIO pin whenever GPIO is selected as a headphone sense
input. If the presence of a headphone insertion is detected, the headphone amplifier will automatically
turn on. If a headphone removal is detected the headphone amplifier will automatically turn off.
HPSENSE Headphone Sense Status
0 Off
1 On
1 HPSENSE_D This bit enables the headphone sense circuit. If enabled, the headphone amplifier will automatically
turn on/off based on the logic level of the GPIO pin whenever GPIO is selected as a headphone sense
input. If the presence of a headphone insertion is detected, the headphone amplifier will automatically
turn on and the Class D loudspeaker amplifier will turn off. If a headphone removal is detected the
headphone amplifier will automatically turn off and the Class D loudspeaker amplifier will turn on. This
bit overrides bit 0 of this register.
HPSENSE_D Headphone Sense Status
0 Off
1 On
2 HPSENSE_AUX This bit enables the headphone sense circuit. If enabled, the headphone amplifier will automatically
turn on/off based on the logic level of the GPIO pin whenever GPIO is selected as a headphone sense
input. If the presence of a headphone insertion is detected, the headphone amplifier will automatically
turn on and the Earpiece / Auxout amplifier will turn off. If a headphone removal is detected the
headphone amplifier will automatically turn off and the Earpiece / Auxout amplifier will turn on. This bit
overrides bit 0 and bit 1 of this register.
HPSENSE_AUX Headphone Sense Status
0 Off
1 On
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Table 24. HP_SENSE (0x1Bh) (continued)
Bits Field Description
3 HPSENSE_AUX_D This bit enables the headphone sense circuit. If enabled, the headphone amplifier will automatically
turn on/off based on the logic level of the GPIO pin whenever GPIO is selected as a headphone sense
input. If the presence of a headphone insertion is detected, the headphone amplifier will automatically
turn on and the Class D loudspeaker amplifier along with the Earpiece / Auxout amplifier will turn off. If
a headphone removal is detected the headphone amplifier will automatically turn off and the Class D
loudspeaker amplifier along with the Earpiece / Auxout amplifier will turn on. This bit overrides bit 0, bit
1, and bit 2 of this register.
HPSENSE_AUX_D Headphone Sense Status
0 Off
1 On
ADC Control Registers
This register is used to control the LM49352's ADC:
Table 25. ADC Basic (0x20h)
Bits Field Description
0 MONO This sets mono or stereo operation of the ADC.
MONO ADC Operation
0 Stereo Audio
1 Mono Voice (Right ADC channel disabled, Left ADC channel active)
1 OSR This sets the oversampling ratio of the ADC.
OSR Stereo Audio ADC Mono Voice ADC Oversampling Ratio
Oversampling Ratio
0 128 125
1 128 128
2 MUTE_L If set, a digital mute is applied to the Left (or mono) ADC output.
3 MUTE_R If set, a digital mute is applied to the Right ADC output.
6:4 ADC_CLK_SEL This selects the source of the ADC clock domain, ADC_SOURCE_CLK.
ADC_CLK_SEL Source
000 MCLK
001 PORT1_RX_CLK
010 PORT2_RX_CLK
011 PLL_OUTPUT1
100 PLL_OUTPUT2
7 ADC_DSP_ONLY If set, the ADC's analog circuitry is disabled to reduce power consumption, however, ADC DSP
functionality is maintained. This can be used to perform asynchronous resampling between audio
rates of a common family. Setting this bit is also useful whenever applying Automatic Level Control
(ALC) to an analog only audio path.
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Table 26. ADC_CLK_DIV (0x21h)
Bits Field Description
7:0 ADC_CLK_DIV This programs the half cycle divider that preceeds the ADC. The input of this divider should be around
12MHz. The default of this divider is 0x00.
Program this divider with the division you want, multiplied by 2, and subtract 1.
ADC_CLK_DIV Divides by
00000000 1
00000001 1
00000010 1.5
00000011 2
11111101 127
11111110 127.5
11111111 128
Table 27. ADC_MIXER (0x23h)
Bits Field Description
1:0 ADC_MIX_LEVEL_L This sets the input level to the left ADC channel.
ADC_MIX_LEVEL_L Level
00 0dB
01 1.35dB
10 3.5dB
11 6dB
3.2 ADC_MIX_LEVEL_R This sets the input level to the right ADC channel.
ADC_MIX_LEVEL_R Level
00 0dB
01 1.35dB
10 3.5dB
11 6dB
4 STEREO_LINK If set, this links ADC_MIX_LEVEL_R with ADC_MIX_LEVEL_L.
STEREO_LINK Status
0 Off
1 On
DAC Control Registers
This register is used to control the LM49352's DAC:
Table 28. DAC Basic (0x30h)
Bits Field Description
1:0 MODE This programs the over sampling ratio of the stereo DAC.
MODE DAC Oversampling Ratio
00 125
01 128
10 64 (Default)
11 RSVD
2 MUTE_L This digitally mutes the Left DAC output.
3 MUTE_R This digitally mutes the Right DAC output.
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Table 28. DAC Basic (0x30h) (continued)
Bits Field Description
6:4 DAC_CLK_SEL This selects the source of the DAC clock domain, DAC_SOURCE_CLK.
DAC_CLK_SEL Source
000 MCLK
001 PORT1_RX_CLK
010 PORT2_RX_CLK
011 PLL_OUTPUT1
100 PLL_OUTPUT2
7 DSP_ONLY If set, the DAC's analog circuitry is disabled to reduce power consumption, however DAC DSP
functionality is maintained. This can be used to perform asyncronous resampling between audio rates of
a common family.
Table 29. DAC_CLK_DIV (0x31h)
Bits Field Description
7:0 DAC_CLK_DIV This programs the half cycle divider that precedes the DAC. The input of this divider should be around
12MHz. The default of this divider is 0x03 which gives a division by 2.
Program this divider with the division you want, multiplied by 2, and subtract 1.
DAC_CLK_DIV Divides by
00000000 1
00000001 1
00000010 1.5
00000011 2 (Default)
11111101 127
11111110 127.5
11111111 128
Digital Mixer Control Registers
Digital Mixer
The LM49352’s digital mixer allows for flexible routing of digital audio signals between both audio ports, DAC,
and ADC. This mixer handles which digital data path (Port1 RX data, Port2 RX data, or ADC output) is routed to
the DAC input. The digital mixer also selects the appropriate digital data path (Port1 RX data, Port2 RX data,
ADC output, or DAC DSP (Interpolator) output) that is used for data transmission on Audio Port 1 and 2. Audio
inputs to the digital mixer can be attenuated down to -18dB to avoid clipping conditions. The digital mixer also
allows direct routing from the DAC interpolator output to the ADC decimator input which allows the DAC and
ADC DSP blocks to be cascaded witjhout having to enable the analog of the DAC and ADC in order to save
power.
Another key feature of the digital mixer is sample rate conversion (SRC) between audio ports. This allows
simultaneous operation of the dual audio ports even if each port is operating at a different sample rate. The
LM49352 can be used as an audio port bridge with SRC capability. The digital mixer allows either straight pass
through between audio ports or, if desired, DSP effects can be added to the digital audio signal during audio port
bridge operation. The digital mixer automatically handles stereo I2S to mono PCM conversion between audio
ports and vice versa.
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Figure 61. Digital Mixer
The LM49352 includes two separate and independent DSP blocks, one for the DAC and the other for the ADC.
The digital mixer also allows both DSP blocks to be cascaded together in either order so that the DSP effects
from both blocks can be combined into the same signal path. For example, the 5 band parametric EQ of each
DSP block can be combined together to form a 10 band parametric EQ for added flexibility.
This register is used to control the LM49352's digital mixer:
Table 30. Input Levels 1 (0x40h)
Bits Field Description
1:0 PORT1_RX_L This programs the input level of the data arriving from the left receive channel of Audio Port 1.
_LVL PORT1_RX_L_LVL Level
00 0dB
01 –6dB
10 –12dB
11 –18dB
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Table 30. Input Levels 1 (0x40h) (continued)
Bits Field Description
3:2 PORT1_RX_R This programs the input level of the data arriving from the right receive channel of Audio Port 1.
_LVL PORT1_RX_R_LVL Level
00 0dB
01 –6dB
10 –12dB
11 –18dB
5:4 PORT2_RX_L This programs the input level of the data arriving from the left receive channel of Audio Port 2.
_LVL PORT2_RX_L_LVL Level
00 0dB
01 –6dB
10 –12dB
11 –18dB
7:6 PORT2_RX_R This programs the input level of the data arriving from the right receive channel of Audio Port 2.
_LVL PORT2_RX_R_LVL Level
00 0dB
01 –6dB
10 –12dB
11 –18dB
Table 31. Input Levels 2 (0x41h)
Bits Field Description
1:0 ADC_L_LVL This programs the input level of the data arriving from the left ADC channel.
ADC_L_LVL Level
00 0dB
01 –6dB
10 –12dB
11 –18dB
3:2 ADC_R_LVL This programs the input level of the data arriving from the right ADC channel.
ADC_R_LVL Level
00 0dB
01 –6dB
10 –12dB
11 –18dB
5:4 INTERP_L_LVL This programs the input level of the data arriving from the left DAC's interpolator output.
INTERP_L_LVL Level
00 0dB
01 –6dB
10 –12dB
11 –18dB
7:6 INTERP_R_LVL This programs the input level of the data arriving from the right DAC's interpolator output.
INTERP_R_LVL Level
00 0dB
01 –6dB
10 –12dB
11 –18dB
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Table 32. Audio Port 1 Input (0x42h)
Bits Field Description
1:0 L_SEL This selects which input is fed to the Left TX Channel of Audio Port 1.
L_SEL Selected Input
00 None
01 ADC_L
10 PORT2_RX_L
11 DAC_INTERP_L
3:2 R_SEL This selects which input is fed to the Right TX Channel of Audio Port 1.
R_SEL Selected Input
00 None
01 ADC_R
10 PORT2_RX_R
11 DAC_INTERP_R
4 SWAP If set, this swaps the Left and Right outputs to Audio Port 1.
5 MONO If set, the right channel is ignored and the left channel becomes (left+right)/2.
Table 33. Audio Port 2 Input (0x43h)
Bits Field Description
1:0 L_SEL This selects which input is fed to Audio Port 2's Left TX Channel.
L_SEL Selected Input
00 None
01 ADC_L
10 PORT1_RX_L
11 DAC_INTERP_L
3:2 R_SEL This selects which input is fed to Audio Port 2's Right TX Channel.
R_SEL Selected Input
00 None
01 ADC_R
10 PORT1_RX_R
11 DAC_INTERP_R
4 SWAP If set, this swaps the Left and Right outputs to Audio Port 2.
5 MONO If set, the right channel is ignored and the left channel becomes (left+right)/2.
Table 34. DAC Input Select (0x44h)
Bits Field Description
0 PORT1_L This adds Audio Port 1's left RX channel to the DAC's left input.
1 PORT2_L This adds Audio Port 2's left RX channel to the DAC's left input.
2 ADC_L This adds the ADC's left output to the DAC's left input.
3 PORT1_R This adds Audio Port 1's right RX channel to the DAC's right input.
4 PORT2_R This adds Audio Port 2's right RX channel to the DAC's right input.
5 ADC_R This adds the ADC's right output to the DAC's right input.
6 SWAP If set, this swaps the Left and Right inputs to the DAC.
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23 22 21 20 2 1 0 23 22 21 20 2 1 0 23
I2S_CLK
I2S_SDO/
I2S_SDI
I2S_SYNC
Left Word Right Word
23 22 21 2 1 0 23 22 21 2 1 0
I2S_CLK
I2S_SDO/
I2S_SDI
I2S_SYNC
Left Word Right Word
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Table 35. Decimator Input Select (0x45h)
Bits Field Description
1:0 L_SEL This selects which input is fed to the left ADC's decimator input.
L_SEL Selected Input
00 None
01 PORT1_RX_L
10 PORT2_RX_L
11 DAC_INTERP_L
3:2 R_SEL This selects which input is fed to the right ADC's decimator input.
R_SEL Selected Input
00 None
01 PORT1_RX_R
10 PORT2_RX_R
11 DAC_INTERP_R
5:4 MXR_CLK_SEL This selects sets the source of the Digital Mixer Clock. The 'Auto' setting will automatically select the source
with the highest clock frequency. If the DAC interpolator output (DAC_OSR_L or DAC_OSR_R) is selected,
then MXR_CLK_SEL should be set to '10'.
MXR_CLK_SEL Selected Input
00 Auto
01 MCLK
10 DAC
11 ADC
Audio Port Control Registers
Figure 62. I2S Serial Data Format (24 bit example)
Figure 63. Left Justified Data Format (24 bit example)
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0 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 15 14 13 12 11 10 9
PCM_CLK
PCM_SDO/
PCM_SDI
PCM_SYNC
Short frame sync mode
Long frame sync mode
0 23 22 21 3 2 1 0 23 22 21 3 2 1 0
I2S_CLK
I2S_SDO/
I2S_SDI
I2S_SYNC
Left Word Right Word
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Figure 64. Right Justified Data Format (24 bit example)
Figure 65. PCM Serial Data Format (16 bit example)
The following registers are used to control the LM49352's audio ports. Audio Port 1 and Audio Port 2 are
identical. Port 1 is programmed through the (0x5Xh) registers. Port 2 is programmed through the (0x6Xh)
registers.
Table 36. BASIC_SETUP (0x50h/0x60h)
Bits Field Description
0 STEREO If set, the audio port will receive and transmit stereo data.
1 RX_ENABLE If set, the input is enabled (enables the SDI port and input shift register and any clock
generation required).
2 TX_ENABLE If set, the output is enabled (enables the SDO port and output shift register and any clock
generation required).
3 CLOCK_MS If set, the audio port will transmit the clock when either the RX or TX is enabled.
4 SYNC_MS If set, the audio port will transmit the sync signal when either the RX or TX is enabled.
5 CLOCK_PHASE This sets how data is clocked by the Audio Port.
CLOCK_PHASE Audio Data Mode
0 I2S (TX on falling edge, RX on rising edge)
1 PCM (TX on rising edge, RX on falling edge)
6 STEREO_SYNC_PHASE If set, this reverses the left and right channel data of the Audio Port.
STEREO_SYNC_PHASE Audio Port Data Orientation
Left channel data goes to left channel output.
0Right channel data goes to right channel output.
Right channel data goes to left channel output.
1Left channel data goes to right channel output.
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Table 36. BASIC_SETUP (0x50h/0x60h) (continued)
Bits Field Description
7 SYNC_INVERT If this bit is set the SYNC is inverted before the receiver and transmitter.
SYNC_INVERT SYNC ORIENTATION
0 SYNC Low = Left, SYNC High = Right
1 SYNC Low = Right, SYNC High = Left
Table 37. CLK_GEN_1 (0x51h/0x61h)
Bits Field Description
5:0 HALF_CYCLE_CLK_DI This programs the half-cycle divider that generates the master clocks in the audio port. The default of
V this divider is 0x00, i.e. bypassed.
Program this divider with the required division multiplied by 2, and subtract 1.
HALF_CYCLE_CLK_DIV Divides By
000000 BYPASS
000001 1
000010 1.5
000011 2
111101 31
111110 31.5
11111 32
6 CLOCK_SEL This selects the clock source of the master mode Audio Port Clock generator's half-cycle divider.
0 = DAC_SOURCE_CLK
1 = ADC_SOURCE_CLK
Table 38. CLK_GEN_1 (0x52h/62h)
Bits Field Description
2:0 SYNTH_NUM Along with SYNTH_DENOM, this sets the clock divider that generates the Port 1 or Port 2 clock in master
mode. SYNTH_NUM Numerator
000 SYNTH_DENOM (1/1)
001 100/SYNTH_DENOM
010 96/SYNTH_DENOM
011 80/SYNTH_DENOM
100 72/SYNTH_DENOM
101 64/SYNTH_DENOM
110 48/SYNTH_DENOM
111 0/SYNTH_DENOM
3 SYNTH_DENOM Along with SYNTH_NUM, this sets the clock divider that generates the Port 1 or Port 2 clock in master
mode. SYNTH_DENOM Denominator
0 128
1 125
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Table 39. CLK_GEN_1 (0x53h/63h)
Bits Field Description
2:0 SYNC_RATE This sets the number of clock cycles before the sync pattern repeats. This depends if the audio port data
is mono or stereo.
In MONO mode:
SYNC_RATE Number of Clock Cycles
000 8
001 12
010 16
011 18
100 20
101 24
110 25
111 32
In STEREO mode:
SYNC_RATE Number of Clock Cycles
000 16
001 24
010 32
011 36
100 40
101 48
110 50
111 64
5:3 SYNC_WIDTH In MONO mode, this programs the width (in number of bits) of the SYNC signal.
SYNC_WIDTH Width of SYNC (in bits)
000 1
001 2
010 4
011 7
100 8
101 11
110 15
111 16
Table 40. DATA_WIDTHS (0x54h/64h)
Bits Field Description
2:0 RX_WIDTH This programs the expected bits per word of the serial data input SDI.
RX_WIDTH Bits
000 24
001 20
010 18
011 16
100 14
101 13
110 12
111 8
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Table 40. DATA_WIDTHS (0x54h/64h) (continued)
Bits Field Description
5:3 TX_WIDTH This programs the bits per word of the serial data output SDO.
TX_WIDTH Description
000 24
001 20
010 18
011 16
100 14
101 13
110 12
111 8
7:6 TX_EXTRA_BITS This programs the TX data output padding.
TX_EXTRA_BITS Description
00 0
01 1
10 High-Z
11 High-Z
Table 41. RX_MODE (0x55h/x65h)
Bits Field Description
0 RX_MODE This sets the RX data input justification with respect to the SYNC signal.
RX_MODE Description
0 MSB Justified
1 LSB Justified
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Table 41. RX_MODE (0x55h/x65h) (continued)
Bits Field Description
5:1 MSB_POSITION This specifies the bit location of the MSB from the start of the frame (MSB Justified) or from the end of the
frame (LSB Justified).
MSB_POSITION Description
00000 0(Left Justified/PCM Long)
00001 1(I2S/PCM Short)
00010 2
00011 3
00100 4
00101 5
00110 6
00111 7
01000 8
01001 9
01010 10
01011 11
01100 12
01101 13
01110 14
01111 15
10000 16
10001 17
10010 18
10011 19
10100 20
10101 21
10110 22
10111 23
11000 24
11001 25
11010 26
11011 27
11100 28
11101 29
11110 30
11111 31
6 COMPAND If set, audio data will be companded.
7μLaw/A-Law This sets the audio companding mode.
μLaw/A-Law Compand Mode
0μLaw
1 A-Law
Table 42. TX_MODE (0x56h/x66h)
Bits Field Description
0 TX_MODE This sets the TX data output justification with respect to the SYNC signal.
TX_MODE Description
0 MSB Justified
1 LSB Justified
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Table 42. TX_MODE (0x56h/x66h) (continued)
Bits Field Description
5:1 MSB_POSITION This specifies the bit location of the MSB from the start of the frame (MSB Justified) or from the end of the
frame (LSB Justified).
MSB_POSITION Description
00000 0(Left Justified/PCM Long)
00001 1(I2S/PCM Short)
00010 2
00011 3
00100 4
00101 5
00110 6
00111 7
01000 8
01001 9
01010 10
01011 11
01100 12
01101 13
01110 14
01111 15
10000 16
10001 17
10010 18
10011 19
10100 20
10101 21
10110 22
10111 23
11000 24
11001 25
11010 26
11011 27
11100 28
11101 29
11110 30
11111 31
6 COMPAND If set, audio data will be companded.
7μLaw/A-Law This sets the audio companding mode.
μLaw/A-Law Compand Mode
0μLaw
1 A-Law
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RIGHT
DAC
LEFT
DAC
SOFT CLIP
5 Band EQ
Distort
DIG MIXER
DIG Gain &
NG Mutes
ALC
-78 dB to +18 dB
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Digital Effects Engine
Digital Signal Processor (DSP)
The LM49352 is designed to handle the entire audio signal conditioning and processing within the audio system,
thereby freeing up the workload of any other applications processor contained within the system. The LM49352
features two independent DSPs, one for the DAC and the other for the ADC. Each DSP is fully featured and
performs as a professional quality digital audio effects engine. Both DSP engines feature digital volume control,
automatic level control (ALC), digital soft clip compression, and a 5-band parametric EQ. The effects chain of
each DSP engine is shown by the diagrams below.
Figure 66. ADC DSP Effects Chain
Figure 67. DAC DSP Effects Chain
The ADC and DAC DSP engines can be cascaded together in any order via the digital mixer to combine different
audio effects to the same signal path. For example, a signal can be processed with high-pass filtering from the
ADC effects engine with ALC from the DAC effects engine. The 5-band parametric EQs from each DSP engine
can be combined to form a single 10-band parametric EQ or a single 5-band parametric EQ with ±30dB (instead
of ±15dB) gain control for each band.
Table 43. ADC EFFECTS (0x70h)
Bits Field Description
0 ADC_HPF_ENB This enables the ADC's High Pass Filter.
1 ADC_ALC_ENB This enables the ADC's Automatic Level Control.
2 ADC_PK_ENB This enables the ADC's Peak Detector.
3 ADC_EQ_ENB This enables the ADC's 5-band Parametric EQ.
4 ADC_SCLP_ENB This enables the ADC's Soft Clip Feature.
Table 44. DAC EFFECTS (0x71h)
Bits Field Description
0 DAC_ALC_ENB This enables the DAC's Automatic Level Control.
1 DAC_PK_ENB This enables the DAC's Peak Detector.
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Table 44. DAC EFFECTS (0x71h) (continued)
Bits Field Description
2 DAC_EQ_ENB This enables the DAC's 5-band Parametric EQ.
3 RSVD Reserved
4 ADC_SCLP_ENB This enables the DAC's Soft Clip Feature.
Table 45. HPF MODE (0x80h)
Bits Field Description
2:0 HPF_MODE This configures the ADC's High Pass Filter (HPF). To calculate the –3dB cutoff frequency, multiply the
coefficient by the sample rate (Hz): fC= Xn.fS(Hz)
HPF_MODE Coefficient Filter Characteristics
fC= 220Hz for:
000 X0= 0.0275 8kHz Voice
001 X1= 0.01833 12kHz Voice
010 X2= 0.01375 16kHz Voice
011 X3= 0.009166 24kHz Voice
100 X4= 0.006875 32kHz Voice
fC= 100Hz for:
101 X5= 0.003125 32kHz Audio
110 X6= 0.0020833 48kHz Audio
fC= 150Hz for:
111 X7= 0.0015625 96kHz Audio
ALC Overview
The Automatic Level Control (ALC) system can be used to regulate the audio output level to a user defined
target level. The ALC feature is especially useful whenever the level of the audio input is unknown,
unpredictable, or has a large dynamic range. The main purpose of the ALC is to optimize the dynamic range of
the audio input to audio output path.
There are two separate and independent ALC circuits in the LM49352. One of the ALC circuits is located within
the DAC DSP effects block. The other ALC circuit is integrated into the ADC DSP effects block. The DAC ALC
controls the DAC digital gain. The ADC ALC controls the mono/auxiliary input amplifier gain or microphone
preamplifier gain. The dual ALCs can be used to regulate the level of the analog (AUX, MONO, MIC) and digital
(Port1 Data In, Port2 Data In) audio inputs. The ALC regulated output can be routed to any of the LM49352’s
amplifier outputs for playback. The ALC regulated output can also be routed to Audio Port1 or Audio Port2 for
digital data transmission via I2S or PCM.
Only audio inputs that are considered signals (rather than noise) are sent to the ALC’s peak detector block. The
peak detector compares the level of the audio input versus the ALC target level (TARGET_LEVEL). Signals
lower than the target level will be amplified and signals higher than the target level will be attenuated. Any audio
input that is lower than the level specified by the noise floor level (NOISE_FLOOR) will be considered as noise
and will be gated from the ALC’s peak detector in order to avoid noise pumping. So it is important to set
NOISE_FLOOR to correlate with the signal to noise ratio of the corresponding audio path. In some instances (ie.
Conference calls), it may be desirable to mute audio input signals that consist solely of background noise from
the audio output. This is accomplished by enabling the ALC’s noise gate (NG_ENB). When the noise gate is
enabled, signals lower than the noise floor level will be muted from the audio output.
If the audio input signal is below the target level, the ALC will increase the gain of the corresponding volume
control until the signal reaches the target level. The rate at which the ALC performs gain increases is known as
decay rate (DECAY RATE). But before each ALC gain increase the ALC must wait a predetermined amount of
time (HOLD TIME). If the audio input signal is above the target level, the ALC will decrease the gain of the
corresponding volume control until the signal reaches the target level. The rate at which the ALC performs
attenuation is known as attack rate (ATTACK RATE). The ALC’s peak detector tracks increases in audio input
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Output TARGET
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signal amplitude instantaneously, but tracks decreases in audio input signal amplitude at programmable rate
(PEAK DECAY TIME). ATTACK RATE, DECAY RATE, HOLD TIME, and PEAK DECAY TIME are fully
adjustable which allows flexible operation of the ALC circuit. The ALC’s timers are based on the sample rate of
the DAC or ADC, so the closest corresponding sample rate must be programmed into the DAC SAMPLE setting
(for DAC ALC) or the ADC SAMPLE (for ADC ALC).
Figure 68. ALC Example
Limiter
The LM49352’s ALC features a limiter function. The purpose of the limiter is to limit the maximum level of the
audio signal to the specified ALC target level. When the limiter is enabled, the ALC will decrease the gain of the
volume control whenever the audio signal is higher than the specified target level. The programmed I2C gain
setting when the limiter is first enabled is the maximum gain setting that the ALC limiter will apply to the audio
signal. Gain increases beyond the original I2C gain setting are disabled. This is in contrast to ALC operation with
the limiter disabled, where the ALC may increase gain of audio signals below target level using gain settings
beyond the original I2C gain setting. Therefore, it is important to set the gain of the audio path to the desired
setting before enabling the ALC limiter function.
The limiter’s target level can be set just below the clipping level of the output amplifier or ADC in order to prevent
harsh distortions delivered to the loudspeaker or headphone on the receiving end. This method of ALC limiter
operation is also known as “no clip” mode. Operating the ALC limiter in “no clip” mode maximizes the dynamic
range of the audio amplifier or ADC while ensuring that the audio signal will never clip. Utilizing the ALC limiter in
“no clip” mode also protects the loudspeaker from damage due to harmful overdriven conditions.
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Input
ALC
Output
with
Limiter
Enabled
TARGET
ALC
Output
with
Limiter
Disabled
TARGET
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The ALC limiter’s target level can also be set for a predetermined maximum output power or voltage level. This
method of ALC limiter operation is known as “power limit” mode. Operating the ALC limiter in “power limit” mode
prevents the speaker or headphone from playing at unsafe hearing levels that can permanently damage the end
user’s ears. “Power limit” operation is especially useful for applications such as listening to music through a set of
headphones. Another benefit of using the ALC limit in “power limit” mode is to extend battery life by reducing
power consumption of the output amplifiers during audio playback.
Figure 69. ALC Limiter
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Table 46. ADC_ALC_1 (0x81h)
Bits Field Description
2:0 ADC_SAMPLE This programs the timers on the ALC with the closest sample rate of the ADC.
ADC_SAMPLE Expected ADC fS
000 8kHz
001 12kHz
010 16kHz
011 24kHz
100 32kHz
101 48kHz
110 96kHz
111 192kHz
3 LIMITER If set, the circuit will never apply gain to the signal, no matter how small, but it will attenuate the signal
as soon as it reaches target and release it at the decay rate, once signal level reduces below target.
The I2C gain setting (at the time the LIMITER is enabled) is the maximum gain that the ALC will apply.
Care should be taken when choosing the optimum I2C gain setting whenever enabling the Limiter.
4 STEREO LINK If set, the ALC circuit uses the stereo average of the input signals to control the gain of the stereo
output. This maintains stereo imaging. If this bit is cleared, then both channels operate as dual mono.
5 SOURCE_RSEL If both SOURCE_OVR and this bit is set, the right ADC ALC channel will be active.
6 SOURCE_LSEL If both SOURCE_OVR and this bit is set, the left ADC ALC channel will be active.
7 SOURCE_OVR If set, the active channel of the ADC ALC is determined by SOURCE_RSEL and SOURCE_LSEL. If
cleared, the active channel of the ADC ALC is determined by the selected input to the ADC. MONO
enables left ALC, AUX enables right ALC, MIC enables left and / or right ALC depending on which
ADC channel MIC is selected to.
Table 47. ADC_ALC_2 (0x82h)
Bits Field Description
3:0 NOISE_FLOOR This sets the anticipated noise floor. Signals lower than the noise floor specified will be gated from the
ALC to avoid noise pumping.
NOISE_FLOOR Noise Floor (dB)
0000 –39
0001 –42
0010 –45
0011 –48
0100 –51
0101 –54
0110 –57
0111 –60
1000 –63
1001 –66
1010 –69
1011 –72
1100 –75
1101 –78
1110 –81
1111 –84
4 NG_ENB This enables the Noise Gate.
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Table 48. ADC_ALC_3 (0x83h)
Bits Field Description
4:0 TARGET_LEVEL This sets the desired target output level. Signals lower than this will be amplified and signals larger
than this will be attenuated.
TARGET_LEVEL Target Level (dB)
00000 –1.5
00001 –3
00010 –4.5
00011 –6
00100 –7.5
00101 –9
00110 –10.5
00111 –12
01000 –13.5
01001 –15
01010 –16.5
01011 –18
01100 –19.5
01101 –21
01110 –22.5
01111 –24
10000 –25.5
10001 –27
10010 –28.5
10011 –30
10100 –31.5
10101 –33
10110 –34.5
10111 –36
11000 –37.5
11001 –39
11010 –40.5
11011 –42
11100 –43.5
11101 –45
11110 –46.5
11111 –48
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Table 49. ADC_ALC_4 (0x84h)
Bits Field Description
4:0 ATTACK_RATE This sets the rate at which the ALC will reduce gain if it detects the input signal is large.
ATTACK_RATE Time between gain steps (μs)
00000 21
00001 42
00010 83
00011 167
00100 250
00101 333
00110 417
00111 542
01000 729
01001 958
01010 1250 (Default)
01011 1604
01100 1896
01101 2208
01110 2792
01111 3708
10000 4792
10001 5688
10010 6563
10011 8396
10100 11000
10101 14167
10110 17083
10111 20000
11000 25000
11001 32000
11010 45000
11011 60000
11100 75000
11101 87500
11110 100000
11111 114583
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Table 50. ADC_ALC_5 (0x85h)
Bits Field Description
4:0 DECAY_RATE This sets the rate at which the ALC will increase gain if it detects the input signal is too small.
DECAY_RATE Time between gain steps (μs)
00000 104
00001 125
00010 167
00011 250
00100 292
00101 396
00110 500
00111 708
01000 896
01001 1250
01010 1396 (Default)
01011 2000
01100 2708
01101 3500
01110 4750
01111 6250
10000 8000
10001 11000
10010 14000
10011 18500
10100 25000
10101 32000
10110 42000
10111 55000
11000 72500
11001 100000
11010 125000
11011 160000
11100 225000
11101 300000
11110 375000
11111 500000 (0.5s)
7:5 PK_DECAY_RATE PK_DECAY_RATE Max Time to track decay
000 1.3ms (Default)
001 2.6ms
010 5.3ms
011 10.6ms
100 21.3ms
101 42.6.3ms
110 85.5ms
111 2.73 secs
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Table 51. ADC_ALC_6 (0x86h)
Bits Field Description
4:0 HOLD_TIME This sets how long the ALC circuit waits before increasing the gain.
HOLD_TIME Time (ms)
00000 1
00001 1.25
00010 1.6
00011 2
00100 2.5
00101 3.2
00110 4
00111 5
01000 6.25
01001 8
01010 10 (Default)
01011 12.5
01100 16
01101 20
01110 25
01111 32
10000 40
10001 50
10010 64
10011 80
10100 100
10101 125
10110 160
10111 200
11000 250
11001 320
11010 400
11011 500
11100 640
11101 800
11110 1000
11111 1250
Table 52. ADC_ALC_7 (0x87h)
Bits Field Description
5:0 MAX_LEVEL This sets the maximum allowed gain of the volume control to the output amplifier whenever
the ALC is use. If the volume control is less than 6 bits the relevant LSBs are used as the
limit and the MSBs are ignored.
Table 53. ADC_ALC_8 (0x88h)
Bits Field Description
5:0 MIN_LEVEL This sets the minimum allowed gain of the volume control to the output amplifier whenever
the ALC is use. If the volume control is less than 6 bits the relevant LSBs are used as the
limit and the MSBs are ignored.
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Table 54. ADC_L_LEVEL (0x89h)
Bits Field Description
5:0 ADC_L_LEVEL This sets the post ADC digital gain of the left channel.
ADC_L_LEVEL Level ADC_L_LEVEL Level
000000 -76.5dB 100000 -28.5dB
000001 -75dB 100001 -27dB
000010 -73.5dB 100010 -25.5dB
000011 -72dB 100011 -24dB
000100 -70.5dB 100100 -22.5dB
000101 -69dB 100101 -21dB
000110 -67.5dB 100110 -20.5dB
000111 -66dB 100111 -18dB
001000 -64.5dB 101000 -16.5dB
001001 -63dB 101001 -15dB
001010 -61.5dB 101010 -13.5dB
001011 -60dB 101011 -12dB
001100 -58.5dB 101100 -10.5dB
001101 -57dB 101101 -9dB
001110 -55.5dB 101110 -7.5dB
001111 -54dB 101111 -6dB
010000 -52.5dB 110000 -4.5dB
010001 -51dB 110001 -3dB
010010 -49.5dB 110010 -1.5dB
010011 -48dB 110011 0dB
010100 -46.5dB 110100 1.5dB
010101 -45dB 110101 3dB
010110 -43.5dB 110110 4.5dB
010111 -42dB 110111 6dB
011000 -40.5dB 111000 7.5dB
011001 -39dB 111001 9dB
011010 -37.5dB 111010 10.5dB
011011 -36dB 111011 12dB
011100 -34.5dB 111100 13.5dB
011101 -33dB 111101 15dB
011110 -31.5dB 111110 16.5dB
011111 -30dB 111111 18dB
6 STEREO_LINK If set, this links the ADC_R_LEVEL with ADC_L_LEVEL.
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Table 55. ADC_R_LEVEL (0x8Ah)
Bits Field Description
5:0 ADC_R_LEVEL This sets the post ADC digital gain of the right channel.
ADC_R_LEVEL Level ADC_R_LEVEL Level
000000 -76.5dB 100000 -28.5dB
000001 -75dB 100001 -27dB
000010 -73.5dB 100010 -25.5dB
000011 -72dB 100011 -24dB
000100 -70.5dB 100100 -22.5dB
000101 -69dB 100101 -21dB
000110 -67.5dB 100110 -20.5dB
000111 -66dB 100111 -18dB
001000 -64.5dB 101000 -16.5dB
001001 -63dB 101001 -15dB
001010 -61.5dB 101010 -13.5dB
001011 -60dB 101011 -12dB
001100 -58.5dB 101100 -10.5dB
001101 -57dB 101101 -9dB
001110 -55.5dB 101110 -7.5dB
001111 -54dB 101111 -6dB
010000 -52.5dB 110000 -4.5dB
010001 -51dB 110001 -3dB
010010 -49.5dB 110010 -1.5dB
010011 -48dB 110011 0dB
010100 -46.5dB 110100 1.5dB
010101 -45dB 110101 3dB
010110 -43.5dB 110110 4.5dB
010111 -42dB 110111 6dB
011000 -40.5dB 111000 7.5dB
011001 -39dB 111001 9dB
011010 -37.5dB 111010 10.5dB
011011 -36dB 111011 12dB
011100 -34.5dB 111100 13.5dB
011101 -33dB 111101 15dB
011110 -31.5dB 111110 16.5dB
011111 -30dB 111111 18dB
Table 56. EQ_BAND_1 (0x8Bh)
Bits Field Description
1:0 FREQ This sets the Sub-bass shelving filter's cut-off frequency. The cut-off frequencies shown are
based on a 48kHz sample rate. Using lower sample rates will scale down the cut-off
frequencies proportionately.
FREQ Frequency (Hz)
00 60
01 80
10 100
11 120
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Table 56. EQ_BAND_1 (0x8Bh) (continued)
Bits Field Description
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
Table 57. EQ_BAND_2 (0x8Ch)
Bits Field Description
1:0 FREQ This sets the Bass peak filter's center frequency. The cut-off frequencies shown are based
on a 48kHz sample rate. Using lower sample rates will scale down the cut-off frequencies
proportionately.
FREQ Frequency (Hz)
00 150
01 200
10 250
11 300
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Table 57. EQ_BAND_2 (0x8Ch) (continued)
Bits Field Description
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
7 Q Programs the width of the peak filter.
Q Bandwidth
0 2/3 Octave
1 4/3 Octave
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Table 58. EQ_BAND_3 (0x8Dh)
Bits Field Description
1:0 FREQ This sets the Mid peak filter's center frequency. The cut-off frequencies shown are based on
a 48kHz sample rate. Using lower sample rates will scale down the cut-off frequencies
proportionately.
FREQ Frequency (Hz)
00 600
01 800
10 1k
11 1.2k
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
7 Q This programs the width of the peak filter.
Q Bandwidth
0 2/3 Octave
1 4/3 Octave
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Table 59. EQ_BAND_4 (0x8Eh)
Bits Field Description
1:0 FREQ This sets the Treble peak filter's center frequency. The cut-off frequencies shown are based
on a 48kHz sample rate. Using lower sample rates will scale down the cut-off frequencies
proportionately.
FREQ Frequency (Hz)
00 2k
01 2.7k
10 3.4k
11 4.1k
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
7 Q This programs the width of the peak filter.
Q Bandwidth
0 2/3 Octave
1 4/3 Octave
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Table 60. EQ_BAND_5 (0x8Fh)
Bits Field Description
1:0 FREQ This sets the presence shelving filter's cut-off frequency. The cut-off frequencies shown are
based on a 48kHz sample rate. Using lower sample rates will scale down the cut-off
frequencies proportionately.
FREQ Frequency (Hz)
00 7k
01 9k
10 11k
11 13k
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
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Audio
Compressor
Output Level
(dB)
Audio Compressor Input
Level
(dB)
Audio
Compressor
Threshold
Level 1:2 Compression
Ratio
1:215
Compression
Ratio
1:4 Compression
Ratio
1:1 Compression
Ratio
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Digital Audio Compressor
The LM49352 features a digital audio compressor on both the DAC and ADC paths. The compressor works by
reducing the level of the audio signal that is higher than the level set by the audio compressor threshold level
(THRESHOLD) by a fixed ratio (compressor output / compressor input) that is set by a predetermined audio
compression ratio (RATIO). Higher compression ratios result in more compression as shown in Figure 70. The
audio compressor can be used in conjunction with the ALC to limit audio peaks that the ALC may not be fast
enough to react to.
Figure 70. Audio Compressor Effect
Soft Knee Function
The LM49352’s audio compressor also features a soft knee function that smooths the harsh edges found during
clipping of an audio signal. For audio signals higher than the compressor threshold level, the soft knee function
gradually increases the compression ratio for increasing levels of audio signal beyond the compressor threshold.
To achieve the smoothing effect to prevent hard clipping, the soft knee function initially compresses the audio
signal at the smallest ratio and then incrementally increases the compression ratio if required. The highest level
of compression applied by the soft knee function is set by the compressor ratio. The effect of the soft knee
function is shown in Figure 71.
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Audio
Compressor
Output Level
(dB)
Audio Compressor Input
Level
(dB)
Audio
Compressor
Threshold
Level
Soft Knee
Disabled
1:1.4
1:1.2
1:1.7 1:2
1:2.4 1:2.8 1:3.4
Soft Knee
Enabled
1:3.4
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Figure 71. Soft Knee Example with Compression Ratio Setting of 1:3.4
Table 61. SOFTCLIP1 (0x90h)
Bits Field Description
3:0 THRESHOLD This sets the threshold level of the audio compressor. Audio signals above the threshold
will be compressed.
THRESHOLD Threshold Level (dB)
0000 -36dB
0001 -30dB
0010 -24dB
0011 -20dB
0100 -18dB
0101 -17dB
0110 -16dB
0111 -15dB
1000 -14dB
1001 -12dB
1010 -10dB
1011 -8dB
1100 -6dB
1101 -4dB
1110 -2.5dB
1111 -1dB
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Table 61. SOFTCLIP1 (0x90h) (continued)
Bits Field Description
4 SOFT_KNEE If set, the audio compressor will automatically apply higher compression ratios to audio
signals higher than the threshold level. As the audio signal approaches levels higher than
the threshold, SOFT_KNEE will increase the compression RATIO. The highest
compression that the SOFT_KNEE algorithm will apply is the compression that is set by
RATIO.
Table 62. SOFTCLIP2 (0x91h)
Bits Field Description
4:0 RATIO This sets the ratio at which the audio is compressed to when it passes beyond the
threshold. In SOFT_KNEE mode this is the final level of compression.
RATIO Ratio
00000 1:1 (Bypass)
00001 1:1.2
00010 1:1.4
00011 1:1.7
00100 1:2.0
00101 1:2.4
00110 1:2.8
00111 1:3.4
01000 1:4.0
01001 1:4.7
01010 1:5.7
01011 1:6.7
01100 1:8.0
01101 1:9.5
01110 1:11.3
01111 1:13.5
10000 1:16.0
10001 1:19.0
10010 1:22.8
10011 1:27.0
10100 1:32.0
10101 1:37.9
10110 1:45.5
10111 1:53.9
11000 1:64.0
11001 1:75.0
11010 1:91.0
11011 1:108
11100 1:128
11101 1:152
11110 1:182
11111 1:215
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Table 63. SOFTCLIP3 (0x92h)
Bits Field Description
3:0 LEVEL This sets the post compressor gain level.
LEVEL Level (dB)
00000 -22.5dB
00001 -21dB
00010 -19.5dB
00011 -18dB
00100 -16.5dB
00101 -15dB
00110 -13.5dB
00111 -12dB
01000 -10.5dB
01001 -9dB
01010 -7.5dB
01011 -6dB
01100 -4.5dB
01101 -3dB
01110 -1.5dB
01111 0dB
10000 1.5dB
10001 3dB
10010 4.5dB
10011 6dB
10100 7.5dB
10101 9dB
10110 10.5dB
10111 12dB
11000 13.5dB
11001 15dB
11010 16.5dB
11011 18dB
11100 19.5dB
11101 21dB
11110 22.5dB
11111 24dB
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DAC Effects Registers
Table 64. DAC_ALC_1 (0xA0h)
Bits Field Description
2:0 DAC_SAMPLE This programs the timers on the ALC with the closest DAC sample rate.
DAC_SAMPLE Expected DAC fS
000 8kHz
001 12kHz
010 16kHz
011 24kHz
100 32kHz
101 48kHz
110 96kHz
111 192kHz
3 LIMITER If set, the circuit will never apply gain to the signal, no matter how small, but it will attenuate
the signal as soon as it reaches target and release it at the decay rate, once signal level
reduces below target. The I2C gain setting (at the time the LIMITER is enabled) is the
maximum gain that the ALC will apply. Care should be taken when choosing the optimum
I2C gain setting whenever enabling the Limiter.
4 STEREO LINK If set, the ALC circuit uses the stereo average of the input signals to control the gain of the
stereo output. This maintains stereo imaging. If this bit is cleared, then both channels
operate as dual mono.
Table 65. DAC_ALC_2 (0xA1h)
Bits Field Description
3:0 NOISE_FLOOR This sets the anticipated noise floor. Signals lower than the specified noise floor will be
gated from the ALC to avoid noise pumping.
NOISE_FLOOR Noise Floor (dB)
0000 -39
0001 -42
0010 -45
0011 -48
0100 -51
0101 -54
0110 -57
0111 -60
1000 -63
1001 -66
1010 -69
1011 -72
1100 -75
1101 -78
1110 -81
1111 -84
4 NG_ENB This enables the Noise Gate
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Table 66. DAC_ALC_3 (0xA2h)
Bits Field Description
4:0 TARGET_LEVEL This sets the desired output level. Signals lower than this will be amplified and signals
larger than this will be attenuated.
TARGET_LEVEL Target Level (dB)
00000 -1.5
00001 -3
00010 -4.5
00011 -6
00100 -7.5
00101 -9
00110 -10.5
00111 -12
01000 -13.5
01001 -15
01010 -16.5
01011 -18
01100 -19.5
01101 -21
01110 -22.5
01111 -24
10000 -25.5
10001 -27
10010 -28.5
10011 -30
10100 -31.5
10101 -33
10110 -34.5
10111 -36
11000 -37.5
11001 -39
11010 -40.5
11011 -42
11100 -43.5
11101 -45
11110 -46.5
11111 -48
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Table 67. DAC_ALC_4 (0xA3h)
Bits Field Description
4:0 ATTACK_RATE This sets the rate at which the ALC will reduce gain if it detects the input signal is too large.
ATTACK_RATE Time between gain steps (μs)
00000 21
00001 42
00010 83
00011 167
00100 250
00101 333
00110 417
00111 542
01000 729
01001 958
01010 1250 (Default)
01011 1604
01100 1896
01101 2208
01110 2792
01111 3708
10000 4792
10001 5688
10010 6563
10011 8396
10100 11000
10101 14167
10110 17083
10111 20000
11000 25000
11001 32000
11010 45000
11011 60000
11100 75000
11101 87500
11110 100000
11111 114583
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Table 68. DAC_ALC_5 (0xA4h)
Bits Field Description
4:0 DECAY_RATE This sets the rate at which the ALC will increase gain if it detects the input signal is too
small.DECAY_RATE Time between gain steps(us)
00000 104
00001 125
00010 167
00011 250
00100 292
00101 396
00110 500
00111 708
01000 896
01001 1250
01010 1396 (Default)
01011 2000
01100 2708
01101 3500
01110 4750
01111 6250
10000 8000
10001 11000
10010 14000
10011 18500
10100 25000
10101 32000
10110 42000
10111 55000
11000 72500
11001 100000
11010 125000
11011 160000
11100 225000
11101 300000
11110 375000
11111 500000 (0.5s)
7:5 PK_DECAY_RATE This sets how precise the ALC will track amplitude reductions of the audio input. The
shorter the length of time for PK_DECAY_RATE, the more responsive the ALC will be
when applying gain increases whenever the audio falls below target level.
PK_DECAY_RATE Time
000 1.3ms (Default)
001 2.6ms
010 5.3ms
011 10.6ms
100 21.3ms
101 42.6ms
110 85.5ms
111 2.73secs
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Table 69. DAC_ALC_6 (0xA5h)
Bits Field Description
4:0 HOLD_TIME This sets how long the ALC circuit waits before increasing the gain.
HOLDTIME Time (ms)
00000 1
00001 1.25
00010 1.6
00011 2
00100 2.5
00101 3.2
00110 4
00111 5
01000 6.25
01001 8
01010 10 (Default)
01011 12.5
01100 16
01101 20
01110 25
01111 32
10000 40
10001 50
10010 64
10011 80
10100 100
10101 125
10110 160
10111 200
11000 250
11001 320
11010 400
11011 500
11100 640
11101 800
11110 1000
11111 1250
Table 70. DAC_ALC_7 (0xA6h)
Bits Field Description
5:0 MAX_LEVEL This sets the maximum allowed gain to the digital level control when the ALC is used.
Table 71. DAC_ALC_8 (0xA7h)
Bits Field Description
5:0 MIN_LEVEL This sets the minimum allowed gain to the digital level control when the ALC is used.
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Table 72. DAC_L_LEVEL (0xA8h)
Bits Field Description
5:0 DAC_L_LEVEL This sets the pre DAC digital gain.
DAC_L_LEVEL Level DAC_L_LEVEL Level
000000 -76.5dB 100000 -28.5dB
000001 -75dB 100001 -27dB
000010 -73.5dB 100010 -25.5dB
000011 -72dB 100011 -24dB
000100 -70.5dB 100100 -22.5dB
000101 -69dB 100101 -21dB
000110 -67.5dB 100110 -20.5dB
000111 -66dB 100111 -18dB
001000 -64.5dB 101000 -16.5dB
001001 -63dB 101001 -15dB
001010 -61.5dB 101010 -13.5dB
001011 -60dB 101011 -12dB
001100 -58.5dB 101100 -10.5dB
001101 -57dB 101101 -9dB
001110 -55.5dB 101110 -7.5dB
001111 -54dB 101111 -6dB
010000 -52.5dB 110000 -4.5dB
010001 -51dB 110001 -3dB
010010 -49.5dB 110010 -1.5dB
010011 -48dB 110011 0dB
010100 -46.5dB 110100 1.5dB
010101 -45dB 110101 3dB
010110 -43.5dB 110110 4.5dB
010111 -42dB 110111 6dB
011000 -40.5dB 111000 7.5dB
011001 -39dB 111001 9dB
011010 -37.5dB 111010 10.5dB
011011 -36dB 111011 12dB
011100 -34.5dB 111100 13.5dB
011101 -33dB 111101 15dB
011110 -31.5dB 111110 16.5dB
011111 -30dB 111111 18dB
6 STEREO_LINK If set, this links DAC_R_LEVEL with DAC_L_LEVEL.
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Table 73. DAC_R_LEVEL (0xA9h)
Bits Field Description
5:0 DAC_R_LEVEL This sets the pre DAC digital gain.
DAC_R_LEVEL Level DAC_R_LEVEL Level
000000 -76.5dB 100000 -28.5dB
000001 -75dB 100001 -27dB
000010 -73.5dB 100010 -25.5dB
000011 -72dB 100011 -24dB
000100 -70.5dB 100100 -22.5dB
000101 -69dB 100101 -21dB
000110 -67.5dB 100110 -20.5dB
000111 -66dB 100111 -18dB
001000 -64.5dB 101000 -16.5dB
001001 -63dB 101001 -15dB
001010 -61.5dB 101010 -13.5dB
001011 -60dB 101011 -12dB
001100 -58.5dB 101100 -10.5dB
001101 -57dB 101101 -9dB
001110 -55.5dB 101110 -7.5dB
001111 -54dB 101111 -6dB
010000 -52.5dB 110000 -4.5dB
010001 -51dB 110001 -3dB
010010 -49.5dB 110010 -1.5dB
010011 -48dB 110011 0dB
010100 -46.5dB 110100 1.5dB
010101 -45dB 110101 3dB
010110 -43.5dB 110110 4.5dB
010111 -42dB 110111 6dB
011000 -40.5dB 111000 7.5dB
011001 -39dB 111001 9dB
011010 -37.5dB 111010 10.5dB
011011 -36dB 111011 12dB
011100 -34.5dB 111100 13.5dB
011101 -33dB 111101 15dB
011110 -31.5dB 111110 16.5dB
011111 -30dB 111111 18dB
Table 74. EQ_BAND_1 (0xABh)
Bits Field Description
1:0 FREQ This sets the Sub-bass shelving filter's cut-off frequency. The cut-off frequencies shown are
based on a 48kHz sample rate. Using lower sample rates will scale down the cut-off
frequencies proportionately.
FREQ Frequency (Hz)
00 60
01 80
10 100
11 120
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Table 74. EQ_BAND_1 (0xABh) (continued)
Bits Field Description
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
Table 75. EQ_BAND_2 (0xACh)
Bits Field Description
1:0 FREQ This sets the Bass peak filter's center frequency. The cut-off frequencies shown are based
on a 48kHz sample rate. Using lower sample rates will scale down the cut-off frequencies
proportionately.
FREQ Frequency (Hz)
00 150
01 200
10 250
11 300
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Table 75. EQ_BAND_2 (0xACh) (continued)
Bits Field Description
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
7 Q This programs the width of the peak filter.
Q Bandwidth
0 2/3 Octave
1 4/3 Octave
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Table 76. EQ_BAND_3 (0xADh)
Bits Field Description
1:0 FREQ This sets the Mid peak filter's center frequency. The cut-off frequencies shown are based on
a 48kHz sample rate. Using lower sample rates will scale down the cut-off frequencies
proportionately.
FREQ Frequency (Hz)
00 600
01 800
10 1k
11 1.2k
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
7 Q This programs the width of the peak filter.
Q Bandwidth
0 2/3 Octave
1 4/3 Octave
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Table 77. EQ_BAND_4 (0xAEh)
Bits Field Description
1:0 FREQ This sets the Treble peak filter's center frequency. The cut-off frequencies shown are based
on a 48kHz sample rate. Using lower sample rates will scale down the cut-off frequencies
proportionately.
FREQ Frequency (Hz)
00 2k
01 2.7k
10 3.4k
11 4.1k
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
7 Q This programs the width of the peak filter.
Q Bandwidth
0 2/3 Octave
1 4/3 Octave
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Table 78. EQ_BAND_5 (0xAFh)
Bits Field Description
1:0 FREQ This sets the presence shelving filter's cut-off frequency. The cut-off frequencies shown are
based on a 48kHz sample rate. Using lower sample rates will scale down the cut-off
frequencies proportionately.
FREQ Frequency (Hz)
00 7k
01 9k
10 11k
11 13k
6:2 LEVEL This sets the gain at fC.
LEVEL Effect
00000 Off (0dB)
00001 -15dB
00010 -14dB
00011 -13dB
00100 -12dB
00101 -11dB
00110 -10dB
00111 -9dB
01000 -8dB
01001 -7dB
01010 -6dB
01011 -5dB
01100 -4dB
01101 -3dB
01110 -2dB
01111 -1dB
10000 0dB
10001 1dB
10010 2dB
10011 3dB
10100 4dB
10101 5dB
10110 6dB
10111 7dB
11000 8dB
11001 9dB
11010 10dB
11011 11dB
11100 12dB
11101 13dB
11110 14dB
11111 15dB
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Table 79. SOFTCLIP1 (0xB0h)
Bits Field Description
3:0 TRESHOLD This sets the threshold level of the audio compressor. Audio signals above the threshold will
be compressed.
THRESHOLD Threshold Level (dB)
0000 -36dB
0001 -30dB
0010 -24dB
0011 -20dB
0100 -18dB
0101 -17dB
0110 -16dB
0111 -15dB
1000 -14dB
1001 -12dB
1010 -10dB
1011 -8dB
1100 -6dB
1101 -4dB
1110 -2.5dB
1111 -1dB
4 SOFT_KNEE If set, the audio compressor will automatically apply higher compression ratios to audio
signals higher than the threshold level. As the audio signal approaches levels higher than
the threshold, SOFT_KNEE will increase the compression RATIO. The highest compression
that the SOFT_KNEE algorithm will apply is the compression that is set by RATIO.
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Table 80. SOFTCLIP2 (0xB1h)
Bits Field Description
4:0 RATIO This sets the ratio at which the audio is compressed to when it passes beyond the
threshold. In soft clip mode this is the final level of compression.
RATIO Ratio
00000 1:1 (Bypass)
00001 1:1.2
00010 1:1.4
00011 1:1.7
00100 1:2.0
00101 1:2.4
00110 1:2.8
00111 1:3.4
01000 1:4.0
01001 1:4.7
01010 1:5.7
01011 1:6.7
01100 1:8.0
01101 1:9.5
01110 1:11.3
01111 1:13.5
10000 1:16.0
10001 1:19.0
10010 1:22.8
10011 1:27.0
10100 1:32.0
10101 1:37.9
10110 1:45.5
10111 1:53.9
11000 1:64
11001 1:75.9
11010 1:91.0
11011 1:108
11100 1:128
11101 1:152
11110 1:182
11111 1:215
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Table 81. SOFTCLIP3 (0xB2h)
Bits Field Description
4:0 LEVEL This sets the post compressor gain level.
LEVEL Level (dB)
00000 -22.5dB
00001 -21dB
00010 -19.5dB
00011 -18dB
00100 -16.5dB
00101 -15dB
00110 -13.5dB
00111 -12dB
01000 -10.5dB
01001 -9dB
01010 -7.5dB
01011 -6dB
01100 -4.5dB
01101 -3dB
01110 -1.5dB
01111 0dB
10000 1.5dB
10001 3dB
10010 4.5dB
10011 6dB
10100 7.5dB
10101 9dB
10110 10.5dB
10111 12dB
11000 13.5dB
11001 15dB
11010 16.5dB
11011 18dB
11100 19.5dB
11101 21dB
11110 22.5dB
11111 24dB
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GPIO Registers
Table 82. GPIO1 (0xE0h)
Bits Field Description
5:0 GPIO_MODE This sets the mode of the GPIO Pin.
GPIO_MODE GPIO STATUS
000000 GPIO Mode is disabled .PORT2_SDO is controlled by the Port2 serial
interface configuration. In all the other modes PORT2_SDO is configured
as the GPIO pin.
000001 GPIO_RX (in)
000010 CHIP ENABLE (in)
000011 CHIP ENABLE (in)
000100 ADC MUTE (in)
000101 ADC MUTE (in)
000110 HP SENSE (in)
000111 HP SENSE (in)
001000 SPARE (in)
001001 SPARE (in)
001010 GPIO TX (out)
001011 CHIP ACTIVE (out)
001100 CHIP ACTIVE (out)
001101 HP ENABLE (out)
001110 HP ENABLE (out)
001111 LS ENABLE (out)
010000 LS ENABLE (out)
010001 EP ENABLE (out)
010010 EP ENABLE (out)
010011 ADC CLIPPED (out)
010100 ADC CLIPPED (out)
010101 DAC CLIPPED (out)
010110 DAC CLIPPED (out)
010111 SOMETHING CLIPPED (out)
011000 SOMETHING CLIPPED (out)
011001 ADC NG ACTIVE (out)
011010 ADC NG ACTIVE (out)
011011 DAC NG ACTIVE (out)
011100 DAC NG ACTIVE (out)
011101 THERMAL (out)
011110 THERMAL (out)
011111 LS SHORT CCT (out)
5:0 GPIO_MODE 100000 LS SHORT CCT (out)
ANALOG ERROR
100001 Thermal or LS CCT condition (out)
100010 ANALOG ERROR (out)
ERROR (out)
100011 Thermal or LS CCT or Clipping
100100 ERROR (out)
100101 111111 RESERVED
6 GPIO_TX Whenever GPIO_MODE is set to '001010', the GPIO pin will output a logic level based on this bit
setting. Setting this bit high will result in a logic high GPIO output.
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Table 82. GPIO1 (0xE0h) (continued)
Bits Field Description
7 GPIO_RX This bit reports the logic level is present on the GPIO pin.
Table 83. GPIO2 (0xE1h)
Bits Field Description
0 SHORT This bit will go high whenever a short circuit condition occurs on the Class D loudspeaker
amplifier outputs. Once triggered by a short circuit event, an I2C write of 1 to this bit clear this bit.
1 TEMP This bit will go high whenever the temperature of the LM49352 reaches a critical temperature.
Once triggered by a thermal event, an I2C write of 1 to this bit clear this bit.
Table 84. RESET (0xF0h)
Bits Field Description
4:0 RSVD Reserved.
5 SOFT_RESET Setting this bit resets the digital core of LM49352. SOFT_RESET does not affect the current I2C
register settings.
Table 85. Spread Spectrum (0xF1h)
Bits Field Description
1:0 RSVD Reserved
2 SS_DISABLE If this bit is set, Spread Spectrum mode will be disabled from the Class D amplifier.
Table 86. FORCE (0xFE)
Bits Field Description
0 RSVD Reserved
1 DACREF This bit determines whether the DAC reference voltage is internally generated or externally
driven.DACREF STATUS
0 DACREF uses an internal bandgap reference.
1 DACREF is driven by an external voltage reference.
2 CP_FORCE If set, a -LS_VDD rail will be generated on HP_VSS, even if the headphone output stage is not
required.
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Schematic Diagram
Figure 72. Demo Board Schematic
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Demonstration Board Layout
Figure 73. Top Silkscreen
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Figure 74. Top Layer
Figure 75. Inner Layer 2
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Figure 76. Inner Layer 3
Figure 77. Bottom Layer
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Figure 78. Bottom Silkscreen
Revision History
Rev Date Description
1.0 05/03/10 Initial released.
1.01 06/30/10 Fixed a typo in the I2C Timing Parameters table.
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REVISION HISTORY
Changes from Revision D (March 2013) to Revision E Page
Changed layout of National Data Sheet to TI format ........................................................................................................ 100
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PACKAGE OPTION ADDENDUM
www.ti.com 11-Apr-2013
Addendum-Page 1
PACKAGING INFORMATION
Orderable Device Status
(1)
Package Type Package
Drawing Pins Package
Qty Eco Plan
(2)
Lead/Ball Finish MSL Peak Temp
(3)
Op Temp (°C) Top-Side Markings
(4)
Samples
LM49352RL/NOPB ACTIVE DSBGA YPG 36 250 Green (RoHS
& no Sb/Br) SNAG Level-1-260C-UNLIM -40 to 85 GL5
LM49352RLX/NOPB ACTIVE DSBGA YPG 36 1000 Green (RoHS
& no Sb/Br) SNAG Level-1-260C-UNLIM -40 to 85 GL5
(1) The marketing status values are defined as follows:
ACTIVE: Product device recommended for new designs.
LIFEBUY: TI has announced that the device will be discontinued, and a lifetime-buy period is in effect.
NRND: Not recommended for new designs. Device is in production to support existing customers, but TI does not recommend using this part in a new design.
PREVIEW: Device has been announced but is not in production. Samples may or may not be available.
OBSOLETE: TI has discontinued the production of the device.
(2) Eco Plan - The planned eco-friendly classification: Pb-Free (RoHS), Pb-Free (RoHS Exempt), or Green (RoHS & no Sb/Br) - please check http://www.ti.com/productcontent for the latest availability
information and additional product content details.
TBD: The Pb-Free/Green conversion plan has not been defined.
Pb-Free (RoHS): TI's terms "Lead-Free" or "Pb-Free" mean semiconductor products that are compatible with the current RoHS requirements for all 6 substances, including the requirement that
lead not exceed 0.1% by weight in homogeneous materials. Where designed to be soldered at high temperatures, TI Pb-Free products are suitable for use in specified lead-free processes.
Pb-Free (RoHS Exempt): This component has a RoHS exemption for either 1) lead-based flip-chip solder bumps used between the die and package, or 2) lead-based die adhesive used between
the die and leadframe. The component is otherwise considered Pb-Free (RoHS compatible) as defined above.
Green (RoHS & no Sb/Br): TI defines "Green" to mean Pb-Free (RoHS compatible), and free of Bromine (Br) and Antimony (Sb) based flame retardants (Br or Sb do not exceed 0.1% by weight
in homogeneous material)
(3) MSL, Peak Temp. -- The Moisture Sensitivity Level rating according to the JEDEC industry standard classifications, and peak solder temperature.
(4) Multiple Top-Side Markings will be inside parentheses. Only one Top-Side Marking contained in parentheses and separated by a "~" will appear on a device. If a line is indented then it is a
continuation of the previous line and the two combined represent the entire Top-Side Marking for that device.
Important Information and Disclaimer:The information provided on this page represents TI's knowledge and belief as of the date that it is provided. TI bases its knowledge and belief on information
provided by third parties, and makes no representation or warranty as to the accuracy of such information. Efforts are underway to better integrate information from third parties. TI has taken and
continues to take reasonable steps to provide representative and accurate information but may not have conducted destructive testing or chemical analysis on incoming materials and chemicals.
TI and TI suppliers consider certain information to be proprietary, and thus CAS numbers and other limited information may not be available for release.
In no event shall TI's liability arising out of such information exceed the total purchase price of the TI part(s) at issue in this document sold by TI to Customer on an annual basis.
TAPE AND REEL INFORMATION
*All dimensions are nominal
Device Package
Type Package
Drawing Pins SPQ Reel
Diameter
(mm)
Reel
Width
W1 (mm)
A0
(mm) B0
(mm) K0
(mm) P1
(mm) W
(mm) Pin1
Quadrant
LM49352RL/NOPB DSBGA YPG 36 250 178.0 12.4 3.43 3.59 0.76 8.0 12.0 Q1
LM49352RLX/NOPB DSBGA YPG 36 1000 178.0 12.4 3.43 3.59 0.76 8.0 12.0 Q1
PACKAGE MATERIALS INFORMATION
www.ti.com 11-Oct-2013
Pack Materials-Page 1
*All dimensions are nominal
Device Package Type Package Drawing Pins SPQ Length (mm) Width (mm) Height (mm)
LM49352RL/NOPB DSBGA YPG 36 250 210.0 185.0 35.0
LM49352RLX/NOPB DSBGA YPG 36 1000 210.0 185.0 35.0
PACKAGE MATERIALS INFORMATION
www.ti.com 11-Oct-2013
Pack Materials-Page 2
MECHANICAL DATA
YPG0036xxx
www.ti.com
RLA36XXX (Rev A)
0.650±0.075
D
E
4214895/A 12/12
A
. All linear dimensions are in millimeters. Dimensioning and tolerancing per ASME Y14.5M-1994.
B. This drawing is subject to change without notice.
NOTES:
D: Max =
E: Max =
3.305 mm, Min =
3.305 mm, Min =
3.245 mm
3.245 mm
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