TI LM49352RLX/NOPB Boomer mono class d audio codec subsystem with ground referenced headphone amplifiers, earpiece driver, and audio dsp Datasheet

LM49352
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SNAS467E – MARCH 2009 – REVISED MARCH 2013
LM49352 Boomer™ Mono Class D Audio Codec Subsystem with Ground Referenced
Headphone Amplifiers, Earpiece Driver, and Audio DSP
Check for Samples: LM49352
FEATURES
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Ultra Efficient, Spread Spectrum Class D
Loudspeaker Amplifier that Operates at 93%
Efficiency
Low Voltage, True Ground Headphone
Amplifier Operation
High Performance 103dB SNR Stereo DAC
High Performance 97dB SNR Stereo ADC
Up to 96kHz Stereo Audio Playback
Up to 48kHz Stereo Recording
Dual Bidirectional I2S or PCM Compatible
Audio Interfaces
Read/Write I2C Compatible Control Interface
Flexible Digital Mixer with Sample Rate
Conversion
Sigma-Delta PLL Clock Network that Supports
System Clocks up to 50MHz Including 13MHz,
19.2MHz, and 26MHz
Dual stereo 5 band parametric equalizers
Cascadable DSP Effects that Allow Stereo 10
Band Parametric Equalization
ALC/Limiter/Compressor on Both DAC and
ADC Paths
Dedicated Earpiece Speaker Amplifier
Stereo Auxiliary Inputs and Mono Differential
Input
Differential Microphone Input with SingleEnded Option
Automatic Level Control for Digital Audio
Inputs, Mono Differential Input, Microphone
Input, and Stereo Auxiliary Inputs
Flexible Audio Routing from Input to Output
16 Step Volume Control for Microphone with
2dB Steps
32 Step Volume Control for Auxiliary Inputs in
1.5dB Steps
4 Step Volume Control for Class D
Loudspeaker Amplifier
8 Step Volume Control for Headphone
Amplifier
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Micro-Power Shutdown Mode
Available in the 3.3 x 3.3 mm 36 bump DSBGA
package
APPLICATIONS
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Smart Phones
Mobile Phones and VOIP Phones
Portable GPS Navigator and Portable Gaming
Devices
Portable DVD/CD/AAC/MP3/MP4 Players
Digital Cameras/Camcorders
KEY SPECIFICATIONS
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Class D Amplifier Efficiency 93% (Typ)
PEP at A_VDD = 3.3V, 32Ω, 1% THD 58mW (Typ)
PHP at HP_VDD = 2.8V, Stereo 32Ω, 1% THD
65mW/ch (Typ)
PLS at LS_VDD = 5V, 8Ω, 1% THD 1.4W (Typ)
PLS at LS_VDD = 4.2V, 8Ω, 1% THD 970 mW
(Typ)
PLS at LS_VDD = 3.3V, 8Ω, 1% THD 590 mW
(Typ)
SNR (Stereo DAC at 48kHz) 103 dB (Typ)
PSRR at 217 Hz, A_VDD = 3.3V, (HP from AUX)
100dB (Typ)
DESCRIPTION
The LM49352 is a high performance mixed signal
audio subsystem. The LM49352 includes a high
quality stereo DAC, a high quality stereo ADC, a
stereo headphone amplifier, which supports True
Ground operation, a low EMI Class D loudspeaker
amplifier, and an earpiece speaker amplifier. It
combines advanced audio processing, conversion,
mixing, and amplification in the smallest possible
footprint while extending the battery life of feature rich
portable devices.
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Please 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.
Boomer is a trademark of Texas Instruments.
All other trademarks are the property of their respective owners.
PRODUCTION DATA information is current as of publication date.
Products conform to specifications per the terms of the Texas
Instruments standard warranty. Production processing does not
necessarily include testing of all parameters.
Copyright © 2009–2013, Texas Instruments Incorporated
LM49352
SNAS467E – MARCH 2009 – REVISED MARCH 2013
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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 32Ω single-ended stereo load with less than 1% distortion (THD+N) when HP_VDD = 2.8V.
The loudspeaker amplifier delivers up to 970mW into an 8Ω load 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
I/O_VDD D_VDD DGND
ADCREF A_VDD AGND LS_VDD LSGND
HP_VDD
EP+/
AUX_OUT+
SCL
SDA
0 dB to 6 dB
2
I C
SLAVE
REGISTERS
PORT1_CLK
ADC
EFFECTS
PORT1_SDI
PORT1_SDO
PORT2_CLK
PORT2_WS
PORT2_SDI
DIGITAL MIXER & AUDIO PORTS
PORT1_WS
PORT2_SDO/GPIO
EP-/
AUX_OUT-
POWER MANAGEMENT
and CONTROL
VOL CTRL
5-BAND EQ
ALC
DAC_LEFT
CLOCK
NETWORK
with
6' PLL
VOL CTRL
5-BAND EQ
ALC
LS_OUT-
0 dB to 12 dB
6'
LEFT
ADC
CP+
CP-
CHARGE
PUMP
HPVSS
6'
RIGHT
ADC
HP_VDD
-18 dB
to
0 dB
DAC
EFFECTS
LS_OUT+
D
DAC_RIGHT
MONO
AUX
MIC
MCLK
AB
6'
LEFT
DAC
AB
HPL_OUT
HP_VSS
HP_VDD
6'
RIGHT
DAC
-18 dB
to
0 dB
MONO_IN+
AB
HPR_OUT
HP_VSS
MONO_INVCM
-46.5 dB to 18 dB
VCM
DACREF
BG
AUXL / AUX-
VCM
AUXR / AUX+
6 dB to 36 dB
MIC+
MIC-
Figure 1. LM49352 Block Diagram
2
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Typical Application
D_VDD I/O_VDD ADC
REF
HP_VSS
A_VDD LS_VDD HP_VDD
MIC+
DACREF
MIC-
CP+
CP0.5 - 50
MHz
MCLK
HPL
2
I C
HPR
Baseband
Controller
2
I S / PCM
(PORT1)
LM49352
LS+
LS-
MONO+
8O
MONOEP+
EPMultimedia
IC
32O
2
I S / PCM
(PORT2)
DGND LSGND AGND
AUX_L AUX_R
Synthesized
FM Radio / TV Tuner
Figure 2. Example Application in Multimedia Phone with Dedicated Earpiece Speaker and Mono
Loudspeaker
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D_VDD I/O_VDD ADC
REF
HP_VSS
A_VDD LS_VDD HP_VDD
MIC+
DAC REF
MIC-
CP+
CP0.5 - 50
MHz
MCLK
HPL
Baseband
Controller
2
I C
MONO+
LM49352
HPR
MONO-
Multimedia
IC
LS+
LS-
2
8O
I S / PCM
(PORT1)
EP+
EP-
FM /
Bluetooth
Radio
32O
2
I S / PCM
(PORT2)
DGND LSGND AGND
AUX_L AUX_R
Synthesized
TV Tuner
Figure 3. Example Application in Multimedia Phone Using Multiple Media Sources
4
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D_VDD I/O_VDD ADC
REF
HP_VSS
A_VDD LS_VDD HP_VDD
MIC+
DACREF
MIC-
CP+
CP0.5 - 50
MHz
MCLK
HPL
2
I C
HPR
Baseband
Controller
2
I S / PCM
(PORT1)
LM49352
LS+
LS-
MONO+
8O
MONOAUX_OUT+
AUX_OUTMultimedia
IC
2
8O
I S / PCM
(PORT2)
DGND LSGND AGND
LM48310
AUX_L AUX_R
Synthesized
FM Radio / TV Tuner
Figure 4. Example Application in a Multimedia Phone with Stereo Loudspeaker
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D_VDD I/O_VDD ADC
REF
HP_VSS
A_VDD LS_VDD HP_VDD
MIC+
DACREF
MIC-
CP+
CP0.5 - 50
MHz
MCLK
HPL
2
I C
HPR
Multimedia
Processor
2
I S / PCM
(PORT1)
LM49352
LS+
LS8O
AUX_OUT+
AUX_OUT-
Bluetooth
Radio
2
8O
I S / PCM
(PORT2)
DGND LSGND AGND
LM48310
+ AUX -
+ MONO -
FM Radio / TV Tuner
w/ Stereo Differential
Outputs
Figure 5. Example Application in a Portable Media Player with Stereo Loudspeakers
Connection Diagrams
Figure 6. 36 Bump DSBGA
6
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1
2
3
4
5
6
A
HPR
AVDD
AGND
DAC
REF
ADC
REF
SDA
B
HPL
AUXR
AUXL
PORT2
SYNC
PORT2
SDI
SCL
C
HPVSS
HPVDD
EP-
PORT2
SDO
PORT2
CLK
MCLK
D
CP-
CP+
EP+
PORT1
SYNC
PORT1
SDO
DGND
E
LSGND
LSVDD
MONO-
MIC-
PORT1
SDI
DVDD
F
LS+
LS-
MONO+
MIC+
PORT1
CLK
IOVDD
Figure 7. LM49352RL Pinout Diagram
Top View (Bump Side Down)
PIN DESCRIPTIONS
Pin
Pin Name
Type
Direction
A1
HPR
Analog
Output
Description
A2
A_VDD
Supply
Input
DAC (Analog), ADC (Analog), PLL (Analog), input stages, analog mixer
(AUX and class D), and Earpiece amplifier power supply input
A3
AGND
Supply
Input
DAC (Analog), ADC (Analog), PLL (Analog), input stages, analog mixer
(AUX and class D), and Earpiece amplifier ground
A4
DAC REF
Analog
Input/Output
A5
ADC REF
Analog
Input
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
B5
PORT2_SDI
Digital
Input
Audio Port 2 serial data input
B6
SCL
Digital
Input
I2C interface clock line
C1
HP_VSS
Analog
Output
C2
HP_VDD
Supply
Input
C3
EP-/AUXOUT-
Analog
Output
C4
PORT2_SDO /
GPIO
Digital
Input / Output
C5
PORT2_CLK
Digital
Input/Output
C6
MCLK
Digital
Input
D1
CP-
Analog
Input/Output
Fly capacitor negative input
D2
CP+
Analog
Input/Output
Fly capacitor positive input
D3
EP+/AUXOUT+
Analog
Output
D4
PORT1_SYNC
Digital
Input/Output
D5
PORT1_SDO
Digital
Output
Headphone right output
Filter point for the DAC reference
Filter point for the ADC reference. Connect this pin to A_VDD.
Audio Port 2 sync signal (can be master or slave)
Negative power supply pin for the headphone amplifier
Headphone amplifier power supply pin
Earpiece negative output or Auxiliary negative output
Audio port 2 serial data output or General Purpose Input Output
Audio port 2 clock signal (can be master or slave)
Input clock from 0.5MHz to 50 MHz
Earpiece positive output or Auxiliary positive output
Audio Port 1 sync signal (can be master or slave)
Audio Port 1 serial data output
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PIN DESCRIPTIONS (continued)
Pin
Pin Name
Type
Direction
D6
DGND
Supply
Input
Digital ground
Description
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
E6
D_VDD
Supply
Input
DAC (Digital), ADC (Digital), PLL (Digital), digital mixer, DSP core, and I2C
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
F6
I/O_VDD
Supply
Input
Audio Port 1 clock signal (can be master or slave)
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.
8
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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
−65°C to +150°C
Storage Temperature
Power Dissipation (4)
Internally Limited
Human Body Model (5)
ESD Ratings
HPR and HPL pins
8kV
All other pins
2.5kV
Machine Model (6)
200V
Junction Temperature
150°C
Thermal Resistance
θJA – RLA36 (soldered down to PCB
with 2in2 1oz. copper plane)
Soldering Information
See Applications Note AN-1112 (Literature Number SNVA009).
(1)
(2)
(3)
(4)
(5)
(6)
60°C/W
“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.
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.
If Military/Aerospace specified devices are required, please contact the Texas Instruments Sales Office/Distributors for availability and
specifications.
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.
Human body model, applicable std. JESD22-A114C.
Machine model, applicable std. JESD22-A115-A.
Operating Ratings
−40°C to +85°C
Temperature Range
Supply Voltage
A_VDD, LS_VDD, and AVDD_REF
2.7V to 5.5V
D_VDD
1.6V to 2.0V
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) = 8Ω, RL(HP) = 32Ω, f = 1kHz, unless otherwise specified. Limits apply for TA =
25°C.
LM49352
Symbol
Parameter
Conditions
Typical
(3)
Limit
(4)
Units
(Limit)
DC CHARACTERISTICS (Digital current combines D_VDD and I/O_VDD. Analog current combines A_VDD, HP_VDD, and LS_VDD)
DISD
DIDD
AISD
Digital Shutdown Current
Shutdown Mode,
fMCLK = 13MHz, PLL Off
Digital Active Current (MP3 Mode)
fMCLK = 11.2896MHz, fS = 44.1kHz,
Stereo DAC On, OSRDAC = 64,
PLL Off, HP On
1.2
1.4
mA (max)
Digital Active Current (FM Mode)
fMCLK = 13MHz
Analog Audio modes
0.2
0.5
mA (max)
Digital Active Current
(FM Record Mode)
fMCLK = 12.288MHz, fS = 48kHz,
Stereo ADC On, OSRADC = 128,
PLL Off, Stereo Analog Inputs On
1.3
1.5
mA (max)
Digital Active Current
(CODEC Mode)
fMCLK = 12.288MHz, fS = 8kHz,
Mono ADC On, Mono DAC On,
OSRDAC = 64
OSRADC = 128, PLL Off, MIC On
0.5
0.8
mA (max)
Analog Shutdown Current
Shutdown Mode
0.1
5
μA (max)
Analog Supply Current (MP3 Mode)
2
µA
fMCLK = 11.2896MHz, fS = 44.1kHz
Stereo DAC On, OSRDAC = 64
PLL Off, Stereo HP On
From A_VDD
4.3
6
mA (max)
From HP_VDD
1.5
2.7
mA (max)
From A_VDD
1.7
2.6
mA (max)
From HP_VDD
1.5
2.7
mA (max)
Analog Supply Current
(FM Record Mode)
fMCLK = 12.288MHz, fS = 48kHz,
Stereo ADC On, OSRADC = 128,
PLL Off, Stereo Analog Inputs On
7.2
9.3
mA (max)
Analog Supply Current
(CODEC Mode)
fMCLK = 12.288MHz, fS = 8kHz,
Mono ADC On, Mono DAC On,
OSRDAC = 64
OSRADC = 128, PLL Off, MIC On,
EP On
6.6
8.7
mA (max)
From A_VDD
1.9
2.7
mA (max)
2
mA (max)
Analog Supply Current (FM Mode)
AIDD
fMCLK = 13MHz, PLL Off
Stereo Auxiliary Inputs On,
PLL Off, Stereo HP On
fMCLK = 13MHz,
fPLLOUT = 12MHz, PLL On only
PLLIDD
PLL Total Active Current
From D_VDD
1.4
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)
(2)
(3)
(4)
10
“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.
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.
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.
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) = 8Ω, RL(HP) = 32Ω, f = 1kHz, unless otherwise specified. Limits apply for TA =
25°C.
LM49352
Symbol
Parameter
Conditions
Typical
(3)
Limit
(4)
Units
(Limit)
fS = 48kHz, Stereo
ADCIDD
ADC Total Active Current
From A_VDD
5.8
mA
From D_VDD
1.3
mA
From A_VDD
3.1
mA
From D_VDD
1
mA
fS = 48kHz, Stereo
DACIDD
DAC Total Active Current
AUXINIDD
Mono/Auxiliary Input Amplifier
Quiescent Current
Mono and AUX Input Amplifiers
enabled
0.6
mA
AUXOUTIDD
Auxiliary Output Amplifier Quiescent
Current
AUXOUT enabled
0.4
mA
Earpiece Mode
1.1
mA
93
%
0.03
%
LOUDSPEAKER AMPLIFIER
LSEFF
Loudspeaker Efficiency
PO = 970mW, RL = 8Ω,
LS_VDD = 4.2V
THD+N
Total Harmonic Distortion + Noise
PO = 300mW, f = 1kHz,
RL = 8Ω, Mono Input Signal
RL = 8Ω, f = 1kHz, THD+N = 1%, Mono Input Signal
PO
PSRR
Output Power
Power Supply Rejection Ratio
LS_VDD = 5V
1.4
LS_VDD = 4.2V
970
LS_VDD = 3.3V
590
W
mW
510
mW (min)
RL = 4Ω, f = 1kHz, THD+N = 1%,
Mono Input Signal
LS_VDD = 5V
2.4
LS_VDD = 4.2V
1.65
W
W
LS_VDD = 3.3V
980
mW
VRIPPLE = 200mVP-P
fRIPPLE = 217Hz
Mono Input Terminated
VREF = 1.0μF, Input Referred
LS Gain = 12dB
75
VRIPPLE = 200mVP-P
fRIPPLE = 217Hz
From DAC, DAC gain = 0dB
74
dB
LS_VDD = 4.2V
95
dB
LS_VDD = 3.3V
93
60
dB (min)
Reference = VOUT (1% THD+N )
Mono gain = 0dB, A-weighted
Mono Input Terminated, LS Gain = 8dB
SNR
Signal-to-Noise Ratio
dB (min)
LS_VDD = 4.2V
91
dB
LS_VDD = 3.3V
89
dB
Output Noise
Mono gain = 0dB, A-weighted,
Mono Input Terminated, Input
Referred
43
µV
Offset Voltage
Mono gain = 0dB, from Mono Input
10
eOS
VOS
88
Reference = VOUT (1% THD+N )
DAC Gain = 0dB, A-weighted
fS = 48kHz, OSR = 128
LS Gain = 8dB
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) = 8Ω, RL(HP) = 32Ω, f = 1kHz, unless otherwise specified. Limits apply for TA =
25°C.
LM49352
Symbol
Parameter
Conditions
Typical
Limit
(4)
Units
(Limit)
0.025
0.1
% (max)
(3)
HEADPHONE AMPLIFIERS
THD+N
Total Harmonic Distortion + Noise
PO = 15mW, f = 1kHz,
RL = 32Ω
Stereo Analog Input Signal
RL = 32Ω, f = 1kHz, THD+N = 1%,
Stereo Analog Input Signal, In phase
PO
Headphone Output Power
HP_VDD = 2.8V
65
HP_VDD = 1.8V
24
mW
20
mW (min)
RL = 16Ω, 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,
PSRR
Power Supply Rejection Ratio
Ripple applied to AVDD only
100
Ripple applied to AVDD and HPVDD
88
85
dB (min)
dB
Ripple applied to AVDD
HPVDD, and DVDD
81
dB
Reference = VOUT (1% THD+N )
Gain = 0dB, A-weighted
Stereo Inputs Terminated
98
93
dB (min)
Reference = VOUT (1% THD+N)
Gain = 0dB,
A-weighted, I2S Input = Digital Zero
97
93
dB (min)
Gain = 0dB, A-weighted,
Stereo Inputs Terminated
11
µV
Gain = 0dB, A-weighted,
I2S Input = Digital Zero
12
µV
PO = 7.5mW, f = 1kHz,
RL = 32Ω
Stereo Analog Input Signal
85
dB
0.03
dB
VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
From DAC, DAC gain = 0dB
SNR
Signal to Noise Ratio
eOS
Output Noise
XTALK
Crosstalk
ΔACH-CH
Channel-to-Channel Gain Matching
VOS
12
Output Offset Voltage
AUX Gain = 0dB
From mono Input
1.4
6
mV (max)
DAC Gain = 0dB
From DAC Input, fMCLK = 12.288MHz
1.6
6
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) = 8Ω, RL(HP) = 32Ω, f = 1kHz, unless otherwise specified. Limits apply for TA =
25°C.
LM49352
Symbol
Parameter
Conditions
Typical
(3)
Limit
(4)
Units
(Limit)
AUXILIARY OUTPUT/EARPIECE AMPLIFIER
THD+N
POUT
PSRR
Total Harmonic Distortion
Output Power
Power Supply Rejection Ratio
AUX_LINE_OUT, f = 1kHz
From Mono In
RL = 5kΩ, VOUT = 1VRMS
0.002
%
Earpiece mode, f = 1kHz
From Mono In
RL = 32Ω BTL, POUT = 20mW
0.02
%
Earpiece mode, f = 1kHz
RL = 32Ω BTL, THD+N = 1%
58
VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
Mono Input terminated, CREF = 1.0μF
AUX_LINE_OUT
100
VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
Mono Input terminated, CREF = 1.0μF
Earpiece mode,
–6dB cut enabled
100
50
mW (min)
dB
90
dB
SNR
Signal to Noise Ratio
Gain = 0dB, VREF = VOUT (1% THD+N)
A-weighted, Mono Input Terminated
105
dB
∈OUT
Output Noise
Gain = 0dB, VREF = VOUT (1% THD+N)
A-weighted, Mono Input Terminated
7
μV
MONO gain = 0dB, From Mono Input
AUX_LINE_OUT
4
mV
Gain = 0dB, From Mono Input
Earpiece mode
4
Turn-On time
PMC Clock = 300kHz
28
ms
THD+NADC
ADC Total Harmonic Distortion + Noise
Mono Differential Input
VIN = 1VRMS, f = 1kHz
Gain = 0dB, fS = 48kHz
0.007
%
PBADC
ADC Passband
220
Hz
RADC
ADC Ripple
VOS
TWU
Output Offset Voltage
12
mV (max)
STEREO ADC
HPF On, fS = 48kHz
Lower -3dB Point
HPF On, Upper -3dB Point
SNRADC
ADCLEVEL
ADC Signal to Noise Ratio
0.41*fS
kHz
OSRDAC = 128
0.1
dB
Reference = VOUT (0dBFS )
Gain = 6dB,
A-weighted From MIC, fS = 8kHz
98
dB
Reference = VOUT (0dBFS )
Gain = 0dB,
A-weighted From Stereo Input,
fS = 48kHz
97
dB
1.6
VRMS
0.01
%
VRMS
ADC Full Scale Input Level
STEREO DAC
I2S Input, AUXOUT, OSRDAC = 64
VIN = 500mFFSRMS, f = 1kHz
Gain = 0dB
THD+NDAC
DAC Total Harmonic Distortion + Noise
DACLEVEL
DAC Full Scale Output Level
1.08
RDAC
DAC Ripple
0.1
dB
PBDAC
DAC Passband
Upper –3dB Point
0.45*fS
kHz
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) = 8Ω, RL(HP) = 32Ω, f = 1kHz, unless otherwise specified. Limits apply for TA =
25°C.
LM49352
Symbol
Parameter
Conditions
Typical
(3)
Limit
(4)
Units
(Limit)
VOLUME CONTROL
VCRAUX
VCRMONO
Stereo Input Volume Control Range
MONO Input Volume Control Range
VCRDAC
DAC Volume Control Range
VCRADC
ADC Volume Control Range
Minimum Gain
–46.5
dB
Maximum Gain
18
dB
Minimum Gain
–46.5
dB
Maximum Gain
18
dB
Minimum Gain
–76.5
dB
Maximum Gain
18
dB
Minimum Gain
–76.5
dB
Maximum Gain
18
dB
Minimum Gain
6
dB
VCRMIC
MIC Volume Control Range
Maximum Gain
36
dB
VCRLS
Loudspeaker Amplifier Volume Control
Range
Minimum Gain
0
dB
Maximum Gain
12
dB
VCRHP
Headphone Amplifier Volume Control
Range
Minimum Gain
–18
dB
Maximum Gain
0
dB
SSLS
Loudspeaker Amplifier Volume Control
Stepsize
4
dB
SSHP
Headphone Amplifier Volume Control
Stepsize
Refer to
Table 19
dB
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
SVMONO
MONO Volume Setting Variation
±1
dB (min)
SVMIC
MIC Volume Setting Variation
±1
dB (max)
dB (max)
ANALOG INPUTS
AUX_RIN
MONO_RIN
MIC_RIN
14
Auxiliary Input Impedance
Mono Input Impedance
Microphone Input Impedance
AUX Gain = 18dB
10
kΩ
AUX Gain = 0dB
38
kΩ
AUX Gain = –46.5dB
64
kΩ
MONO Gain = 18dB
10
kΩ
MONO Gain = 0dB
38
kΩ
MONO Gain = –46.5dB
64
kΩ
All MIC gain settings
50
kΩ
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Typical Performance Characteristics
Class D Loudspeaker Amplifier Efficiency vs Output Power
THD+N < 10%, RL = 8Ω
Green >> LSVDD = 3.3V
Gray >> LSVDD = 4.2V
Blue >> LSVDD = 5V
DAC Frequency Response
fS = 48kHz
Blue >> OSR = 64
Light Blue >> OSR = 128
+1
100
90
+0.5
70
MAGNITUDE (dB)
EFFICIENCY (%)
80
60
50
40
30
+0
-0.5
20
10
0
-1
0
400
800
1200
1600
2000
20
FREQUENCY (Hz)
Figure 9.
DAC Frequency Response
fS = 8kHz
Blue >> OSR = 64
Light Blue >> OSR = 128
DAC THD+N vs Frequency
fS = 8kHz, OSR = 128
I2S Input = 500mFFS
+1
10
+0.5
1
+0
0.1
0.01
-0.5
-1
20
5k 10k 20k
Figure 8.
THD+N (%)
MAGNITUDE (dB)
OUTPUT POWER (mW)
50 100 200 500 1k 2k
0.001
50
100
200
500
1k
2k
4k
20
FREQUENCY (Hz)
50 100 200 500 1k 2k
5k 10k 20k
FREQUENCY (Hz)
Figure 10.
Figure 11.
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Typical Performance Characteristics (continued)
DAC THD+N
vs
Input Level
fS = 48kHz, OSR = 64
I2S Input = 1kHz
10
10
1
1
THD+N (%)
THD+N (%)
DAC THD+N vs Frequency
fS = 8kHz, OSR = 64
I2S Input = 500mFFS
0.1
0.01
0.1
0.01
0.001
20
50 100 200 500 1k 2k
0.001
10m
5k 10k 20k
20m
FREQUENCY (Hz)
1
INPUT VOLTAGE (FFS)
Figure 13.
Stereo Audio ADC Frequency Response
fS = 48kHz, OSR = 64
From MIC, MIC Gain = 6dB, CIN = 1µF
Stereo Audio ADC Frequency Response
fS = 8kHz, OSR = 128
From MIC, MIC Gain = 6dB, CIN = 1µF
+1
+0
MAGNITUDE (dB)
+0
MAGNITUDE (dB)
500m
Figure 12.
+1
-1
-1
-2
-2
-3
20
50 100 200 500 1k 2k
5k 10k 20k
-3
20
50
100 200
500
1k
2k
4k
FREQUENCY (Hz)
FREQUENCY (Hz)
Figure 14.
16
50m 100m 200m
Figure 15.
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Typical Performance Characteristics (continued)
+1
Mono Voice ADC Frequency Response
fS = 48kHz, OSR = 128
From MIC, MIC Gain = 6dB, CIN = 1µF
+1
+0
MAGNITUDE (dB)
MAGNITUDE (dB)
+0
-1
-1
-2
-2
-3
20
-3
20
50 100 200 500 1k 2k
5k 10k 20k
50 100 200 500 1k 2k
5k 10k 20k
Figure 16.
FREQUENCY (Hz)
Figure 17.
Stereo Audio HPF ADC Frequency Response
fS = 48kHz, OSR = 128
From MONO/AUX, MONO>AUX Gain = 0dB, CIN = 1µF
Blue >> No HPF
Light Blue >> HPF Mode = '101'
Green >> HPF Mode = '110'
Light Green >> HPF Mode = '111'
Mono Voice HPF ADC Frequency Response
fS = 48kHz, OSR = 128
From MIC, MIC Gain = 6dB, CIN = 1µF
Gray >> No HPF
Blue >> HPF Mode = '000'
Light Blue >> HPF Mode = '001'
Green >> HPF Mode = '010'
Light Green >> HPF Mode = '011'
Yellow >> HPF Mode = '100'
FREQUENCY (Hz)
+1
+1
+0
+0
MAGNITUDE (dB)
MAGNITUDE (dB)
Mono Voice ADC Frequency Response
fS = 8kHz, OSR = 128
From MIC, MIC Gain = 6dB, CIN = 1µF
-1
-2
-3
20
-1
-2
50 100 200 500 1k 2k
5k 10k 20k
-3
20
FREQUENCY (Hz)
50 100 200 500 1k 2k
5k 10k 20k
FREQUENCY (Hz)
Figure 18.
Figure 19.
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Typical Performance Characteristics (continued)
Mono Voice HPF ADC Frequency Response
fS = 8kHz, OSR = 128
From MIC, MIC Gain = 6dB, CIN = 1µF
Gray >> No HPF
Blue >> HPF Mode = '000'
Light Blue >> HPF Mode = '001'
Green >> HPF Mode = '010'
Light Green >> HPF Mode = '011'
Yellow >> HPF Mode = '100'
ADC THD+N vs Frequency
fS = 48kHz, OSR = 128
From MONO/AUX, MONO/AUX Gain = 6dB, VIN = 1VRMS
+1
10
1
THD+N (%)
MAGNITUDE (dB)
+0
-1
-2
-3
20
0.1
0.01
0.001
50 100 200 500 1k 2k
5k 10k 20k
20
FREQUENCY (Hz)
FREQUENCY (Hz)
Figure 21.
ADC THD+N vs Frequency
fS = 48kHz, OSR = 128
From MIC, MIC Gain = 6dB, VIN = 500mVRMS
ADC THD+N vs Input Voltage
fS = 48kHz, OSR = 128
From MONO/AUX, MONO/AUX Gain = 0dB, fIN = 1kHz
10
1
1
THD+N (%)
THD+N (%)
5k 10k 20k
Figure 20.
10
0.1
0.01
0.1
0.01
0.001
20
50 100 200 500 1k 2k
5k 10k 20k
0.001
1m
FREQUENCY (Hz)
5m 10m
50m100m
500m 1
2
INPUT VOLTAGE (Vrms)
Figure 22.
18
50 100 200 500 1k 2k
Figure 23.
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Typical Performance Characteristics (continued)
ADC THD+N vs Input Voltage
fS = 48kHz, OSR = 128
From MIC, MIC Gain = 6dB, VIN = 1kHz
10
Loudspeaker THD+N vs Input Voltage
From MONO/AUX Input, MONO/AUX Gain = 0dB
LS Gain = 8dB, LSVDD = 3.3V, POUT = 300VRMS, RL = 8Ω
10
1
THD+N (%)
THD+N (%)
1
0.1
.01
.001
20
0.1
.01
100
1k
.001
20
10k 20k
100
FREQUENCY (Hz)
Figure 24.
Loudspeaker THD+N vs Input Voltage
From MONO/AUX Input, MONO/AUX Gain = 0dB
LS Gain = 8dB, LSVDD = 5V, POUT = 300VRMS, RL = 8Ω
10
1
1
THD+N (%)
THD+N (%)
10k 20k
FREQUENCY (Hz)
Figure 25.
Loudspeaker THD+N vs Input Voltage
From MONO/AUX Input, MONO/AUX Gain = 0dB
LS Gain = 8dB, LSVDD = 4.2V, POUT = 300VRMS, RL = 8Ω
10
0.1
.01
.001
20
1k
0.1
.01
100
1k
10k 20k
.001
FREQUENCY (Hz)
Figure 26.
20
100
1k
10k 20k
FREQUENCY (Hz)
Figure 27.
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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 = 8Ω
Blue >> LSVDD = 3.3V
Light Blue >> LSVDD = 4.2V
Green >> LSVDD = 5V
10
10
1
1
THD+N (%)
THD+N (%)
Loudspeaker THD+N vs Input Voltage
From MONO/AUX Input, MONO/AUX Gain = 0dB
LS Gain = 8dB, LSVDD = 3.3V, POUT = 300VRMS, RL = 4Ω
0.1
0.1
0.01
.01
0.001
10m
.001
20
100
1k
10k 20k
100m
1
2 3
OUTPUT POWER (W)
FREQUENCY (Hz)
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 = 4Ω
Blue >> LSVDD = 3.3V
Light Blue >> LSVDD = 4.2V
Green >> LSVDD = 5V
Loudspeaker PSRR vs Frequency
MONO/AUX Input, MONO/AUX Gain = 0dB, LS Gain = 12dB
LSVDD = 3.3V, VRIPPLE = 200mVPP, Input Referred
10
0
-10
-20
-30
PSRR (dB)
THD+N (%)
1
0.1
-40
-50
-60
-70
0.01
-80
-90
0.001
10m
100m
1
2 3
-100
OUTPUT POWER (W)
100
1k
10k
100k
FREQUENCY (Hz)
Figure 30.
20
20
Figure 31.
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Typical Performance Characteristics (continued)
Loudspeaker PSRR vs Frequency
fS = 48kHz, OSR = 128
MONO/AUX Input, MONO/AUX Gain = 0dB, LS Gain = 12dB
LSVDD = 4.2V, VRIPPLE = 200mVPP, Input Referred
0
-10
-10
-20
-20
-30
-30
PSRR (dB)
PSRR (dB)
0
Loudspeaker PSRR vs Frequency
MONO/AUX Input, MONO/AUX Gain = 0dB, LS Gain = 12dB
LSVDD = 5V, VRIPPLE = 200mVPP, Input Referred
-40
-50
-60
-40
-50
-60
-70
-70
-80
-80
-90
-90
-100
-100
20
100
1k
10k
100k
20
FREQUENCY (Hz)
100
1k
10k
100k
FREQUENCY (Hz)
Figure 32.
Figure 33.
Headphone PSRR vs Frequency
DAC Input, DAC Gain = 0dB, LS Gain = 12dB
LSVDD = 3,3V, AVDD = 3.3V, DVDD = 1.8V, VRIPPLE =
200mVPP
Ripple on LSVDD, AVDD, DVDD, Input Referred
HeadphoneTHD+N vs Frequency
MONO/AUX, MONO/AUX Gain = 0dB, HP Gain = 0dB
HPVDD = 3.3V, POUT = 15mW, RL = 32Ω
Stereo In Phase
0
10
-10
-20
1
THD+N (%)
PSRR (dB)
-30
-40
-50
-60
0.1
-70
.01
-80
-90
-100
20
100
1k
10k
100k
.001
20
100
1k
10k 20k
FREQUENCY (Hz)
FREQUENCY (Hz)
Figure 34.
Figure 35.
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Typical Performance Characteristics (continued)
Headphone THD+N vs Frequency
MONO/AUX, MONO/AUX Gain = 0dB, HP Gain = 0dB
HPVDD = 1.8V, POUT = 15mW, RL = 16Ω
Stereo In Phase
10
10
1
1
THD+N (%)
THD+N (%)
Headphone THD+N vs Frequency
MONO/AUX, MONO/AUX Gain = 0dB, HP Gain = 0dB
HPVDD = 2.8V, POUT = 15mW, RL = 32Ω
Stereo In Phase
0.1
0.1
.01
.01
.001
20
.001
100
1k
20
10k 20k
100
10k 20k
FREQUENCY (Hz)
Figure 36.
Figure 37.
Headphone THD+N vs Frequency
MONO/AUX, MONO/AUX Gain = 0dB, HP Gain = 0dB
HPVDD = 2.8V, POUT = 15mW, RL = 16Ω
Stereo in Phase
Headphone THD+N vs Output Power
MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB
Stereo In Phase, fIN = 1kHz, RL = 32Ω
Blue >> HPVDD = 1.8V
Light Blue >> HPVDD = 2.8V
10
10
1
1
THD + N (%)
THD+N (%)
FREQUENCY (Hz)
1k
0.1
0.1
0.01
.01
.001
20
100
1k
10k 20k
0.001
1m
2m
5m
10m
100m
OUTPUT POWER (W)
FREQUENCY (Hz)
Figure 38.
22
Figure 39.
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Typical Performance Characteristics (continued)
Headphone THD+N vs Output Power
MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB
Stereo In Phase, fIN = 1kHz, RL = 16Ω
Blue >> HPVDD = 1.8V
Light Blue >> HPVDD = 2.8V
Headphone PSRR vs Frequency
MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB
HPVDD = 1.8V, AVDD = 3.3V, VRIPPLE = 200mVPP
Ripple on HPVDD, AVDD
0
10
-10
-20
-30
PSRR (dB)
THD + N (%)
1
0.1
-40
-50
-60
-70
0.01
-80
-90
0.001
1m
2m
5m
10m
-100
20
100m
100
1k
10k
100k
OUTPUT POWER (W)
FREQUENCY (Hz)
Figure 41.
Headphone PSRR vs Frequency
MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB
HPVDD = 1.8V, AVDD = 3.3V, VRIPPLE = 200mVPP
Ripple on AVDD only
Headphone PSRR vs Frequency
DAC Input, DAC Gain = 0dB, HP Gain = 0dB
HPVDD = 1.8V, AVDD = 3.3V, DVDD = 1.8V, VRIPPLE =
200mVPP
Ripple on HPVDD, AVDD, DVDD
-20
0
-30
-10
-40
-20
-50
-30
PSRR (dB)
PSRR (dB)
Figure 40.
-60
-70
-80
-40
-50
-60
-90
-70
-100
-80
-110
-90
-120
20
100
1k
10k
100k
-100
20
100
1k
10k
100k
FREQUENCY (Hz)
FREQUENCY (Hz)
Figure 42.
Figure 43.
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Typical Performance Characteristics (continued)
Headphone Crosstalk vs Frequency
MONO/AUX Input, MONO/AUX Gain = 0dB, HP Gain = 0dB
HPVDD = 1.8V, AVDD = 3.3V, POUT = 15mW, RL = 32Ω
0
Earpiece THD+N vs Frequency
MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = 0dB
AVDD = 3.3V, POUT = 20mW, RL = 32Ω
Earpiece Mode
10
-10
-20
1
THD+N (%)
CROSSTALK (dB)
-30
-40
-50
-60
0.1
-70
.01
-80
-90
-100
20
.001
100
1k
20
10k 20k
100
FREQUENCY (Hz)
1k
10k 20k
FREQUENCY (Hz)
Figure 44.
Figure 45.
Earpiece THD+N vs Output Power
MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = 0dB
AVDD = 3.3V, fIN = 1kHz, RL = 32Ω
Earpiece Mode
10
Earpiece PSRR vs Frequency
MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = –6dB
AVDD = 3.3V, VRIPPLE = 200mVPP, Earpiece Mode
-20
-30
-40
-50
PSRR (dB)
THD + N (%)
1
0.1
-60
-70
-80
-90
0.01
-100
-110
0.001
1m
2m
5m
10m
100m
-120
20
100
1k
10k
100k
OUTPUT POWER (W)
FREQUENCY (Hz)
Figure 47.
Figure 46.
24
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Typical Performance Characteristics (continued)
Auxiliary Output THD+N vs Output Voltage
MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = 0dB
AVDD = 3.3V, fIN = 1kHz, RL = 5kΩ
AUXOUT Mode
10
10
1
1
THD+N (%)
THD+N (%)
Auxiliary Output THD+N vs Frequency
MONO/AUX Input, MONO/AUX Gain = 0dB, EP Gain = 0dB
AVDD = 3.3V, VOUT = 1VRMS, RL = 5kΩ
AUXOUT Mode
0.1
0.01
.01
.001
20
0.1
100
0.001
10m
10k 20k
1k
100m
1
2 3
OUTPUT VOLTAGE (VRMS)
FREQUENCY (Hz)
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
-20
-30
-40
PSRR (dB)
-50
-60
-70
-80
-90
-100
-110
-120
20
100
1k
10k
100k
FREQUENCY (Hz)
Figure 50.
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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.
SCL
SDA
data
change
allowed
data
valid
data
change
allowed
data
valid
data
change
allowed
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.
SDA
SCL
S
P
START condition
STOP condition
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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MSB
LSB
ADR6
Bit7
ADR5
bit6
ADR4
bit5
ADR3
bit4
ADR2
bit3
ADR1
bit2
ADR0
bit1
R/W
bit0
2
I C SLAVE address (chip address)
Figure 53. I2C Chip Address
Register changes take effect at the SCL rising edge during the last ACK from slave.
ack from slave
ack from slave
start
MSB Chip Address LSB
w
ack MSB Register 0x02h LSB ack
start
slave address =
00110102
w
ack
ack from slave
MSB
Data
LSB
ack
stop
ack
stop
SCL
SDA
register address = 0x02h
ack
register 0x02h data
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.
ack from slave repeated start
ack from slave
start MSB Chip Address LSB w ack MSB Register 0x00h LSB ack rs
ack from slave data from slave ack from master
MSB Chip Address LSB r ack MSB
Data
LSB ack stop
SCL
SDA
start
slave address =
00110102
w ack register address = 0x00h ack
rs
slave address =
00110102
r ack
register 0x00h data
ack stop
Figure 55. Example I2C Read Cycle
SDA
10
8
7
6
1
8
2
7
SCL
5
1
3
4
9
Figure 56. I2C Timing Diagram
I2C TIMING PARAMETERS
Symbol
Parameter
Limit
Min
Typ
Max
Units
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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
5
(1)
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ns
Data Hold Time (Output direction, delay generated by LM49352)
See (1)
50
ns
Data Hold Time (Input direction, delay generated by the Master)
50
ns
6
Data Setup Time
100
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
ns
Data ensured by fall time.
Device Register Map
Table 1. Device Register Map
Address
Register
7
6
5
4
PMC
CHIP
PORT2
PORT1
SETUP
ACTIVE
CLK OVR
CLK OVR
3
2
1
0
MCLK
OSC
CHIP
ENB
PLL_P2
ENB
PLL
OVR
ENB
ENABLE
BASIC SETUP
0x00h
0x01h
PMC
CLOCKS
0x02h
PMC
CLK_DIV
PMC_CLK_SEL
PMC_CLK_DIV(R)
PLL
0x03h
PLL_CLK_SEL
0x04h
PLL M
0x05h
PLL N
0x06h
PLL
N_MOD
0x07h
PLL P
0x08h
PLL P2
PLL M
PLL N
PLL P2[8]
PLL P1[8]
PLL N_MOD
PLL P1 [7:0]
PLL P2[7:0]
ANALOG MIXER
0x10h
CLASSD
AUX_LS
MONO_LS
DACL_LS
DACR_LS
0x11h
HEAD
PHONESL
AUX_HPL
MONO
_HPL
DACL_HPL
DACR_HPL
0x12h
HEAD
PHONESR
AUX_HPR
MONO
_HPR
DACL
_HPR
DACR
_HPR
0x13h
AUX_OUT
AUX_AUX
MONO
_AUX
DACL_AUX
DACR
_AUX
0x14h
OUTPUT
OPTIONS
0x15h
ADC
MONO
_ADCL
AUX
_ADCR
0x16h
MIC_LVL
MUTE
SE/DIFF
0x18h
AUXL_LVL
0x19h
MONO_LV
0x1Bh
HP
_SENSE
0x20h
ADC BASIC
LS_LEVEL
MIC_AUX
AUX_LINE_ AUX_NEG_
OUT
6dB
LR_HP_LEVEL
MIC_ADCL
MIC_ADCR
RSVD
DACL
_ADCL
DACR
_ADCR
MIC_LEVEL
SE/DIFF
AUX_LEVEL
AUXL_MON
SE/DIFF
O_IN
MONO_LEVEL
HP SENSE
_AUX_D
HP SENSE
_AUX
HP
SENSE_D
HP SENSE
MUTE_R
MUTE_L
ADC_OSR
MONO
ADC
28
DSPONLY
ADC_CLK_SEL
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Table 1. Device Register Map (continued)
Address
Register
0x21h
ADC
CLOCK
7
6
5
4
3
2
0x23h
ADC
_MIXER
0x30h
DAC
_BASIC
0x31h
DAC
_CLOCK
0x40h
IPLVL1
PORT2_RX_R_LVL
0x41h
IPLVL2
INTERP_L_LVL
0x42h
OPPORT1
MONO
SWAP
0x43h
OPPORT2
MONO
SWAP
R_SEL
0x44h
OPDAC
ADCR
PORT2R
0x45h
OPDECI
1
0
ADC_CLK_DIV (T)
STEREO
_LINK
ADC_MIX_LEVEL_R
ADC_MIX_LEVEL_L
DAC
DSPONLY
DAC_CLK_SEL
MUTE_R
MUTE_L
DAC_OSR
DAC_CLK_DIV (S)
DIGITAL MIXER
SWAP
PORT2_RX_L_LVL
PORT1_RX_R_LVL
PORT1_RX_L_LVL
INTERP_R_LVL
ADC_R_LVL
ADC_L_LVL
R_SEL
L_SEL
PORT1R
MXRCLK_SEL
L_SEL
ADCL
PORT2L
R_SEL
PORT1L
L_SEL
AUDIO PORT 1
STEREO
_SYNC
_MODE
STEREO
_SYNC
_PHASE
0x50h
BASIC
CLK_PH
SYNC_MS
CLK_MS
0x51h
CLK_GEN1
0x52h
CLK_GEN2
0x53h
SYNC
_GEN
0x54h
DATA
_WIDTH
0x55h
RX_MODE
A/ULAW
COMPAND
MSB_POSITION
RX_MODE
0x56h
TX_MODE
A/ULAW
COMPAND
MSB_POSITION
TX_MODE
0x60h
BASIC
STEREO
_SYNC
_MODE
STEREO
_SYNC
_PHASE
0x61h
CLK_GEN1
CLK_SEL
TX_ENB
RX_ENB
STEREO
HALF_CYCLE_DIVDER
SYNTH
_DENOM
SYNTH_NUM
SYNC_WIDTH(MONO MODE)
SYNC_RATE
TX_WIDTH
RX_WIDTH
TX_EXTRA_BITS
AUDIO PORT 2
CLK_PH
SYNC_MS
CLK_MS
CLK_SEL
TX_ENB
RX_ENB
STEREO
HALF_CYCLE_DIVDER
SYNTH_DE
NOM
0x62h
CLK_GEN2
SYNTH_NUM
0x63h
SYNC
_GEN
0x64h
DATA
_WIDTH
0x65h
RX_MODE
A/ULAW
COMPAND
MSB_POSITION
RX_MODE
0x66h
TX_MODE
A/ULAW
COMPAND
MSB_POSITION
TX_MODE
SYNC_WIDTH (MONO MODE)
SYNC_RATE
TX_WIDTH
RX_WIDTH
TX_EXTRA_BITS
EFFECTS ENGINE
0x70h
ADC FX
ADC
SCLP ENB
ADC
EQ ENB
ADC
PK ENB
ADC
ALC ENB
ADC
HPF_ENB
0x71h
DAC FX
DAC
SCLP ENB
RSVD
DAC
EQ ENB
DAC
PK ENB
DAC
ALC ENB
0x80h
HPF
0x81h
ADC
ALC 1
0x82h
ADC
ALC 2
ADC EFFECTS
HPF MODE
SOURCE
OVR
SOURCE
RSEL
SOURCE
LSEL
STEREO
LINK
NG_ENB
LIMITER
ADC_SAMPLE
NOISE_FLOOR
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Table 1. Device Register Map (continued)
Address
Register
0x83h
ADC
ALC 3
7
6
5
4
3
ALC_TARGET_LEVEL
0x84h
ADC
ALC 4
ATTACK_RATE
0x85h
ADC
ALC 5
0x86h
ADC
ALC 6
0x87h
ADC
ALC 7
MAX_LEVEL
0x88h
ADC
ALC 8
MIN_LEVEL
0x89h
ADC L
LEVEL
0x8Ah
ADC R
LEVEL
PK_DECAY_RATE
2
1
0
DECAY_RATE/RELEASE_RATE
HOLDTIME
STEREO
LINK
ADC_L_LEVEL
ADC_R_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
0x90h
SOFTCLIP
1
SOFT
KNEE
0x91h
SOFTCLIP
2
RATIO
0x92h
SOFTCLIP
3
LEVEL
0x98h
LVLMONL
0x99h
LVLMONR
THRESHOLD
ADC EFFECT MONITORS
ADC LEFT LEVEL MONITOR
ADC RIGHT LEVEL MONITOR
0x9Ah
FXCLIP
SCLP
_R CLIP
SCLP
_L CLIP
EQ
_R CLIP
EQ
_L CLIP
GAIN
_R CLIP
GAIN
_L CLIP
ADC
_R CLIP
0x9Bh
ALCMONL
SCLP_R
DISTORT
SCLP_L
DISTORT
ADC LEFT ALC MONITOR
0x9Ch
ALCMONR
SCLP_L
DISTORT
SCLP_R
DISTORT
ADC RIGHT ALC MONITOR
0xA0h
DAC
ALC 1
STEREO
LINK
0xA1h
DAC
ALC 2
NG_ENB
0xA2h
DAC
ALC 3
AGC_TARGET_LEVEL
0xA3h
DAC
ALC 4
ATTACK_RATE
0xA4h
DAC
ALC 5
0xA5h
DAC
ALC 6
0xA6h
DAC
ALC 7
MAX_LEVEL
0xA7h
DAC
ALC 8
MIN_LEVEL
0xA8h
DAC L
LEVEL
ADC
_L CLIP
DAC EFFECTS
30
PK_DECAY_RATE
LIMITER
DAC_SAMPLE
NOISE_FLOOR
DECAY_RATE/RELEASE_RATE
HOLDTIME
STEREO
LINK
DAC_L_LEVEL
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Table 1. Device Register Map (continued)
Address
Register
0xA9h
DAC R
LEVEL
7
6
5
4
3
2
1
0
DAC_R_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
0xB0h
SOFTCLIP
1
SOFT
KNEE
0xB1h
SOFTCLIP
2
RATIO
0xB2h
SOFTCLIP
3
LEVEL
0xB8h
LVLMONL
0xB9h
LVLMONR
THRESHOLD
DAC EFFECT MONITORS
DAC LEFT LEVEL MONITOR
DAC RIGHT LEVEL MONITOR
0xBAh
FXCLIP
SCLP
_R CLIP
SCLP
_L CLIP
EQ
_R CLIP
EQ
_L CLIP
0xBBh
ALCMONL
SCLP_R
DISTORT
SCLP_L
DISTORT
DAC LEFT ALC MONITOR
0xBCh
ALCMONR
SCLP_L
DISTORT
SCLP_R
DISTORT
DAC RIGHT ALC MONITOR
0xE0h
GPIO1
GPIO_RX
GPIO_TX
0xE1h
GPIO2
RSVD
GAIN
_R CLIP
GAIN
_L CLIP
TEMP
SHORT
RSVD
RSVD
RSVD
SS
_DISABLE
RSVD
RSVD
CPFORCE
DACREF
RSVD
GPIO
GPIO_MODE
SPREAD SPECTRUM
0xF0h
RESET
0xF1h
SS
0xFEh
FORCE
SOFT
_RESET
RSVD
RSVD
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.
1
PLL_ENB
PLL_ENABLE
PLL Status
0
PLL Off
1
PLL On
This enables the P2 output of the PLL.
2
PLL_P2ENB
PLL_P2ENB
PLL P2 Status
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).
3
4
OSC_ENB
MCLK_OVR
OSC_ENABLE
Oscillator Status
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.
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.
5
PORT1_CLK_OVR
PORT1_CLK_OVR
Comment
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.
(1)
32
6
PORT2_CLK_OVR
7
CHIP_ACTIVE
PORT2_CLK_OVR
Comment
0
I/O control is automatic
1
PORT_CLK input forced on
This bit is used to readback the enable status of the chip.
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
1:0
PMC_CLK_SEL
Description
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
7:0
PMC_CLK_DIV
Description
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*fS where 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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Table 6. DAC Clock Requirements
DAC Sample Rate
(kHz)
Clock Required at A
(OSR = 128)
Clock Required at A
(OSR = 125)
Clock Required at A
(OSR = 64)
Clock Required at A
(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
(kHz)
Clock Required at B
(OSR = 128)
Clock Required at B
(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
INTERNAL
OSCILLATOR
300 kHz
% PMC
C
PMC
A
% DAC
Stereo DAC
1 ± 25 MHz
0 _> 50 MHz
PLL
MCLK
0 _ 50 MHz
% ADC
AUDIO
PORT 1
PORT1_CLK
0 _ 50 MHz
AUDIO
PORT 2
PORT2_CLK
MIXER
Stereo ADC
B
% 1/2
Cycle
PMC, DAC, ADC divides by 1, 1.5, 2,
2.5 ==> 128
% 1/2
Cycle
Half Cycle divides by 1, 1.5, 2, 2.5 ==> 32
Figure 57. Internal Clock Network
34
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PLL Setup Registers
PLL_P1
1 ± 25 MHz
% P1
9
P1 = 1,1.5, 2 _> 256
M = 1, 1.5, 2 _> 64
0.5 - 50 MHz
Phase Comparator
and Charge Pump
9
I
%M
VCO
PLL_P2
140 to
210 MHz
1 ± 25 MHz
% P2
P2 = 1,1.5, 2 _> 256
0.7 < 5 MHz
7
%N
N = 10, 11, 12 _> 250
PLL_M
6'M
8
8
5
PLL_N PLL_N_MOD
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)
12
2.5
32
0
12.5
12288000
Error (Hz)
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)
36
Bits
Field
1:0
PLL_CLK_SEL
Description
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
6:0
PLL_M
Description
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
7:0
PLL_N
Description
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
4:0
PLL_N_MOD
Description
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)
38
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
5
AUX_LS
The MONO input is added to the loudspeaker output.
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
0.25Ω. 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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LS+
Ferrite Bead
LM49352
Class D
10 pF
Speaker
LSFerrite Bead
10 pF
LSGND
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
5
AUX_HPL
The MONO input is added to the left headphone output.
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
5
AUX _HPR
The MONO input is added to the right headphone output.
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 DCblocking 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 100mΩ) 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.
40
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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
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
3
RSVD
4
MONO_AUX
5
AUX_AUX
Description
The MIC input is added to the AUX output.
Reserved
The MONO input is added to the AUX output.
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
0
RSVD
Description
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
3:1
LR_HP_LEVEL
Gain (dB)
000
0
001
–1.5
010
–3
011
–6
100
–9
101
–12
110
–15
111
–18
This sets the gain of the Auxiliary output amplifier.
4
AUX_NEG_6dB
AUX_NEG_6dB
Gain (dB)
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.
5
AUX_LINE_OUT
AUX_LINE_OUT
Auxiliary Output Mode
0
Earpiece Amplifier
1
AUX_LINE_OUT
This sets the gain of the Class D loudspeaker amplifier.
7:6
LS_LEVEL
LS_LEVEL
Gain (dB)
00
0
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)
42
Bits
Field
3:0
MIC_LEVEL
4
SE_DIFF
5
MUTE
Description
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
If set, the MIC negative input is ignored. In single-ended mode, the MIC negative input pin should
left floating.
If set, the microphone preamp is muted.
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Table 22. AUX_LEVEL (0x18h)
Bits
Field
5:0
AUX_LEVEL
6
SE/DIFF
Description
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
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
5:0
MONO_LEVEL
6
SE/DIFF
7
AUXL_MONO
Description
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
If set, the MONO– input is ignored. In single-ended mode, the MONO- input pin should be left floating.
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 500kΩ
ensures minimal current draw through the pullup resistor. When the headphone amplifier is disabled, an internal
50kΩ pulldown, 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
44
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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.
I/OVDD
RPU
500 k:
GPIO
HPR
RPD
50 k:
Headphone Jack
with a Normally Closed
(NC)
Mechanical Switch
LM49352
HPL
RPD
50 k:
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.
1
2
HPSENSE_D
HPSENSE_AUX
HPSENSE
Headphone Sense Status
0
Off
1
On
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
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
0
MONO
Description
This sets mono or stereo operation of the ADC.
MONO
1
Stereo Audio
1
Mono Voice (Right ADC channel disabled, Left ADC channel active)
This sets the oversampling ratio of the ADC.
OSR
Stereo Audio ADC
Oversampling Ratio
Mono Voice ADC 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
7
46
OSR
ADC Operation
0
ADC_DSP_ONLY
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
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
1:0
ADC_MIX_LEVEL_L
Description
This sets the input level to the left ADC channel.
ADC_MIX_LEVEL_L
3.2
4
ADC_MIX_LEVEL_R
STEREO_LINK
Level
00
0dB
01
1.35dB
10
3.5dB
11
6dB
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
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
1:0
MODE
Description
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
6:4
DAC_CLK_SEL
Description
This selects the source of the DAC clock domain, DAC_SOURCE_CLK.
DAC_CLK_SEL
7
DSP_ONLY
Source
000
MCLK
001
PORT1_RX_CLK
010
PORT2_RX_CLK
011
PLL_OUTPUT1
100
PLL_OUTPUT2
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.
48
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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
1:0
PORT1_RX_L
_LVL
Description
This programs the input level of the data arriving from the left receive channel of Audio Port 1.
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
3:2
PORT1_RX_R
_LVL
5:4
7:6
PORT2_RX_L
_LVL
PORT2_RX_R
_LVL
Description
This programs the input level of the data arriving from the right receive channel of Audio Port 1.
PORT1_RX_R_LVL
Level
00
0dB
01
–6dB
10
–12dB
11
–18dB
This programs the input level of the data arriving from the left receive channel of Audio Port 2.
PORT2_RX_L_LVL
Level
00
0dB
01
–6dB
10
–12dB
11
–18dB
This programs the input level of the data arriving from the right receive channel of Audio Port 2.
PORT2_RX_R_LVL
Level
00
0dB
01
–6dB
10
–12dB
11
–18dB
Table 31. Input Levels 2 (0x41h)
Bits
Field
1:0
ADC_L_LVL
Description
This programs the input level of the data arriving from the left ADC channel.
ADC_L_LVL
3:2
ADC_R_LVL
00
0dB
01
–6dB
10
–12dB
11
–18dB
This programs the input level of the data arriving from the right ADC channel.
ADC_R_LVL
5:4
7:6
INTERP_L_LVL
INTERP_R_LVL
Level
00
0dB
01
–6dB
10
–12dB
11
–18dB
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
This programs the input level of the data arriving from the right DAC's interpolator output.
INTERP_R_LVL
50
Level
Level
00
0dB
01
–6dB
10
–12dB
11
–18dB
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Table 32. Audio Port 1 Input (0x42h)
Bits
Field
1:0
L_SEL
Description
This selects which input is fed to the Left TX Channel of Audio Port 1.
L_SEL
3:2
R_SEL
Selected Input
00
None
01
ADC_L
10
PORT2_RX_L
11
DAC_INTERP_L
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.
Bits
Field
1:0
L_SEL
Table 33. Audio Port 2 Input (0x43h)
3:2
R_SEL
Description
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
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
0
PORT1_L
This adds Audio Port 1's left RX channel to the DAC's left input.
Description
1
PORT2_L
This adds Audio Port 2's left RX channel to the DAC's left input.
2
ADC_L
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.
This adds the ADC's left output to the DAC's left input.
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Table 35. Decimator Input Select (0x45h)
Bits
Field
1:0
L_SEL
Description
This selects which input is fed to the left ADC's decimator input.
L_SEL
3:2
R_SEL
Selected Input
00
None
01
PORT1_RX_L
10
PORT2_RX_L
11
DAC_INTERP_L
This selects which input is fed to the right ADC's decimator input.
R_SEL
5:4
MXR_CLK_SEL
Selected Input
00
None
01
PORT1_RX_R
10
PORT2_RX_R
11
DAC_INTERP_R
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
I2S_CLK
I2S_SYNC
I2S_SDO/
I2S_SDI
23
22
21
2
1
0
23
22
21
Left Word
2
1
0
1
0
Right Word
2
Figure 62. I S Serial Data Format (24 bit example)
I2S_CLK
I2S_SYNC
I2S_SDO/
I2S_SDI
23
22
21
20
2
1
0
23
22
21
Left Word
20
2
23
Right Word
Figure 63. Left Justified Data Format (24 bit example)
52
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I2S_CLK
I2S_SYNC
I2S_SDO/
I2S_SDI
0
23
22
21
3
2
1
0
23
22
21
Left Word
3
2
1
0
11
10
Right Word
Figure 64. Right Justified Data Format (24 bit example)
PCM_CLK
PCM_SYNC
PCM_SDO/
PCM_SDI
0
15
14
13
12
11
10
9
8
7
6
5
4
3
2
1
0
15
14
13
12
9
Short frame sync mode
Long frame sync mode
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
0
STEREO
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
5
CLOCK_PHASE
6
STEREO_SYNC_PHASE
Description
If set, the audio port will receive and transmit stereo data.
If set, the audio port will transmit the sync signal when either the RX or TX is enabled.
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)
If set, this reverses the left and right channel data of the Audio Port.
STEREO_SYNC_PHASE
Audio Port Data Orientation
0
Left channel data goes to left channel output.
Right channel data goes to right channel output.
1
Right channel data goes to left channel output.
Left channel data goes to right channel output.
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Table 36. BASIC_SETUP (0x50h/0x60h) (continued)
Bits
Field
7
SYNC_INVERT
Description
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
5:0
6
Field
Description
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.
CLOCK_SEL
HALF_CYCLE_CLK_DIV
Divides By
000000
BYPASS
000001
1
000010
1.5
000011
2
—
—
111101
31
111110
31.5
11111
32
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.
3
54
SYNTH_DENOM
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
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:
5:3
SYNC_WIDTH
SYNC_RATE
Number of Clock Cycles
000
16
001
24
010
32
011
36
100
40
101
48
110
50
111
64
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
2:0
RX_WIDTH
Description
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
5:3
TX_WIDTH
7:6
TX_EXTRA_BITS
Description
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
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)
56
Bits
Field
0
RX_MODE
Description
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
5:1
MSB_POSITION
Description
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.
7
μLaw/A-Law
This sets the audio companding mode.
μLaw/A-Law
Compand Mode
0
μLaw
1
A-Law
Table 42. TX_MODE (0x56h/x66h)
Bits
Field
0
TX_MODE
Description
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
5:1
MSB_POSITION
Description
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.
7
μLaw/A-Law
This sets the audio companding mode.
μLaw/A-Law
58
Compand Mode
0
μLaw
1
A-Law
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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
-78 dB to +18 dB
ALC
SOFT CLIP
DIG Gain &
NG Mutes
5 Band EQ
DIG MIXER
LEFT
DAC
RIGHT
DAC
Distort
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
0
ADC_HPF_ENB
This enables the ADC's High Pass Filter.
Description
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
0
DAC_ALC_ENB
1
DAC_PK_ENB
Description
This enables the DAC's Automatic Level Control.
This enables the DAC's Peak Detector.
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Table 44. DAC EFFECTS (0x71h) (continued)
Bits
Field
2
DAC_EQ_ENB
3
RSVD
4
ADC_SCLP_ENB
Description
This enables the DAC's 5-band Parametric EQ.
Reserved
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
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
fC = 220Hz for:
32kHz Voice
fC = 100Hz for:
101
X5 = 0.003125
110
X6 = 0.0020833
32kHz Audio
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
60
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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).
Audio
Input
ALC
Output
TARGET
HOLD
DECAY
ATTACK
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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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.
Audio
Input
ALC
Output
with
Limiter
Enabled
TARGET
ALC
Output
with
Limiter
Disabled
TARGET
Figure 69. ALC Limiter
62
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Table 46. ADC_ALC_1 (0x81h)
Bits
Field
2:0
ADC_SAMPLE
Description
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.
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.
Table 47. ADC_ALC_2 (0x82h)
4
NG_ENB
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
This enables the Noise Gate.
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Table 48. ADC_ALC_3 (0x83h)
64
Bits
Field
4:0
TARGET_LEVEL
Description
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
4:0
ATTACK_RATE
Description
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.
7:5
66
PK_DECAY_RATE
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)
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
4:0
HOLD_TIME
Description
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
5:0
MIN_LEVEL
Description
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
5:0
ADC_L_LEVEL
6
68
STEREO_LINK
Description
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
If set, this links the ADC_R_LEVEL with ADC_L_LEVEL.
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Table 55. ADC_R_LEVEL (0x8Ah)
Bits
Field
5:0
ADC_R_LEVEL
Description
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
6:2
LEVEL
Description
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)
70
Bits
Field
1:0
FREQ
Description
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
6:2
LEVEL
7
Q
Description
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
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.
6:2
7
72
LEVEL
Q
FREQ
Frequency (Hz)
00
600
01
800
10
1k
11
1.2k
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
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.
6:2
7
LEVEL
Q
FREQ
Frequency (Hz)
00
2k
01
2.7k
10
3.4k
11
4.1k
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
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.
6:2
74
LEVEL
FREQ
Frequency (Hz)
00
7k
01
9k
10
11k
11
13k
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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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.
1:1 Compression
Ratio
Audio
Compressor
Threshold
Level
1:2 Compression
Ratio
1:4 Compression
Ratio
Audio
Compressor
Output Level
(dB)
1:215
Compression
Ratio
Audio Compressor Input
Level
(dB)
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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Soft Knee
Enabled
Audio
Compressor
Threshold
Level
1:3.4
1:2.8
1:2.4
1:2
1:1.7
1:1.4
Audio
Compressor
Output Level
(dB)
1:3.4
1:1.2
Soft Knee
Disabled
Audio Compressor Input
Level
(dB)
Figure 71. Soft Knee Example with Compression Ratio Setting of 1:3.4
Table 61. SOFTCLIP1 (0x90h)
76
Bits
Field
3:0
THRESHOLD
Description
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
4
SOFT_KNEE
Description
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
4:0
RATIO
Description
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)
78
Bits
Field
3:0
LEVEL
Description
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
2:0
DAC_SAMPLE
Description
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
3:0
NOISE_FLOOR
4
NG_ENB
Description
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
This enables the Noise Gate
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Table 66. DAC_ALC_3 (0xA2h)
80
Bits
Field
4:0
TARGET_LEVEL
Description
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
4:0
DECAY_RATE
7:5
PK_DECAY_RATE
Description
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)
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
82
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
4:0
HOLD_TIME
Description
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
5:0
MAX_LEVEL
Description
This sets the maximum allowed gain to the digital level control when the ALC is used.
Table 71. DAC_ALC_8 (0xA7h)
Bits
Field
5:0
MIN_LEVEL
Description
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
5:0
DAC_L_LEVEL
6
84
STEREO_LINK
Description
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
If set, this links DAC_R_LEVEL with DAC_L_LEVEL.
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Table 73. DAC_R_LEVEL (0xA9h)
Bits
Field
5:0
DAC_R_LEVEL
Description
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
6:2
LEVEL
Description
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)
86
Bits
Field
1:0
FREQ
Description
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
6:2
LEVEL
7
Q
Description
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
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.
6:2
7
88
LEVEL
Q
FREQ
Frequency (Hz)
00
600
01
800
10
1k
11
1.2k
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
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.
6:2
7
LEVEL
Q
FREQ
Frequency (Hz)
00
2k
01
2.7k
10
3.4k
11
4.1k
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
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.
6:2
90
LEVEL
FREQ
Frequency (Hz)
00
7k
01
9k
10
11k
11
13k
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.
4
SOFT_KNEE
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
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)
92
Bits
Field
4:0
RATIO
Description
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
4:0
LEVEL
Description
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
5:0
GPIO_MODE
5:0
GPIO_MODE
6
94
GPIO_TX
Description
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)
100000
LS SHORT CCT (out)
100001
ANALOG ERROR
Thermal or LS CCT condition (out)
100010
ANALOG ERROR (out)
100011
ERROR (out)
Thermal or LS CCT or Clipping
100100
ERROR (out)
100101 – 111111
RESERVED
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
7
GPIO_RX
Description
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
4:0
RSVD
5
SOFT_RESET
Description
Reserved.
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
1:0
RSVD
2
SS_DISABLE
Description
Reserved
If this bit is set, Spread Spectrum mode will be disabled from the Class D amplifier.
Table 86. FORCE (0xFE)
Bits
Field
0
RSVD
1
DACREF
Description
Reserved
This bit determines whether the DAC reference voltage is internally generated or externally
driven.
DACREF
2
CP_FORCE
STATUS
0
DACREF uses an internal bandgap reference.
1
DACREF is driven by an external voltage reference.
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
96
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Demonstration Board Layout
Figure 73. Top Silkscreen
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Figure 74. Top Layer
Figure 75. Inner Layer 2
98
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Figure 76. Inner Layer 3
Figure 77. Bottom Layer
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Figure 78. Bottom Silkscreen
Revision History
100
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
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11-Apr-2013
PACKAGING INFORMATION
Orderable Device
Status
(1)
Package Type Package Pins Package
Drawing
Qty
Eco Plan
Lead/Ball Finish
(2)
MSL Peak Temp
Op Temp (°C)
Top-Side Markings
(3)
(4)
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.
Addendum-Page 1
Samples
PACKAGE MATERIALS INFORMATION
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11-Oct-2013
TAPE AND REEL INFORMATION
*All dimensions are nominal
Device
Package Package Pins
Type Drawing
SPQ
Reel
Reel
A0
Diameter Width (mm)
(mm) W1 (mm)
LM49352RL/NOPB
DSBGA
YPG
36
250
178.0
12.4
LM49352RLX/NOPB
DSBGA
YPG
36
1000
178.0
12.4
Pack Materials-Page 1
B0
(mm)
K0
(mm)
P1
(mm)
W
Pin1
(mm) Quadrant
3.43
3.59
0.76
8.0
12.0
Q1
3.43
3.59
0.76
8.0
12.0
Q1
PACKAGE MATERIALS INFORMATION
www.ti.com
11-Oct-2013
*All dimensions are nominal
Device
Package Type
Package Drawing
Pins
SPQ
Length (mm)
Width (mm)
Height (mm)
LM49352RL/NOPB
DSBGA
YPG
LM49352RLX/NOPB
DSBGA
YPG
36
250
210.0
185.0
35.0
36
1000
210.0
185.0
35.0
Pack Materials-Page 2
MECHANICAL DATA
YPG0036xxx
D
0.650±0.075
E
RLA36XXX (Rev A)
D: Max = 3.305 mm, Min =3.245 mm
E: Max = 3.305 mm, Min =3.245 mm
4214895/A
NOTES:
A. All linear dimensions are in millimeters. Dimensioning and tolerancing per ASME Y14.5M-1994.
B. This drawing is subject to change without notice.
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12/12
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