TI LM49350 Lm49350 boomerâ® audio power amplifier series high performance audio codec sub-system with a ground-referenced stereo headphone amplifier & an ultra low emi class d loudspeaker amplifier with dual i2s/pcm digital audio interface Datasheet

LM49350, LM49350RLEVAL
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
LM49350 Boomer® Audio Power Amplifier Series High Performance Audio Codec SubSystem with a Ground-Referenced Stereo Headphone Amplifier & an Ultra Low EMI Class
D Loudspeaker Amplifier with Dual I2S/PCM Digital Audio Interfaces
Check for Samples: LM49350, LM49350RLEVAL
FEATURES
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High Performance 96dB SNR Stereo DAC
High Performance 94dB SNR Stereo ADC
Up to 192kHz 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
Dual Sigma-Delta PLLs for Operation from any
Clock at Any Sample rate
Digital 3D Stereo Enhancement
Dual 5 Band Parametric Equalizers
Cascadable DSP Effects that Allow 10 Band
Parametric Equalization
ALC/Compressor/Limiter on Both DAC and
ADC Paths
Ultra Low EMI, Class D Loudspeaker Amplifier
with Spread Spectrum Control
Ground Referenced Output Cap-Less
Headphone Amplifier Operation
Earpiece Speaker Amplifier with Reduced
Power Consumption Mode for Mono
Differential Line out Applications
Stereo Auxiliary Inputs or Mono Differential
Input
Differential Stereo Microphone Inputs with
Single-Ended Option
Automatic Level Control for Digital Audio
Inputs, Stereo Microphone Inputs, and Stereo
Auxiliary Inputs
Flexible Audio Routing from Input to Output
16 Step Volume Control for Microphones with
2dB Steps
32 Step Volume Control for Auxiliary Inputs in
1.5dB Steps
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Micro-Power Shutdown Mode
Available in the 3.5 x 3.5 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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PHP at A_VDD = 3.3V, Stereo 32Ω, 1% THD
69mW/ch (typ)
PLS at LS_VDD = 5V, 8Ω, 1% THD 1.2W (typ)
PLS at LS_VDD = 4.2V, 8Ω, 1% THD 825mW (typ)
PLS at LS_VDD = 3.3V, 8Ω, 1% THD 495mW (typ)
PEP at A_VDD = 3.3V, 32Ω BTL, 1% THD 58mW
(typ)
Supply Voltage Range
– D_VDD = 1.7V to 2.0V
– LS_VDD and A_VDD = 2.7V to 5.5V
– I/O_VDD = 1.6V to 4.5V
SNR (Stereo DAC at 48kHz) 96dB (typ)
SNR (Stereo ADC at 48kHz) 94dB (typ)
PSRR at 217 Hz, A_VDD = 3.3V, (HP from AUX)
97dB (typ)
DESCRIPTION
The LM49350 is a high performance audio subsystem
that supports both analog and digital audio functions.
The LM49350 includes a high quality stereo DAC, a
high quality stereo ADC, a stereo headphone
amplifier that supports ground referenced output capless operation, a dual mode earpiece speaker
amplifier, and a low EMI Class D loudspeaker
amplifier. It is designed for demanding applications in
mobile phones and other 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.
All 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 © 2008–2012, Texas Instruments Incorporated
LM49350, LM49350RLEVAL
SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
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DESCRIPTION CONTINUED
The LM49350 features dual bi-directional I2S or PCM audio interfaces for full range audio and an I2C compatible
interface for control. The stereo DAC path features an SNR of 96dB with 24-bit 48 kHz input. The headphone
amplifier delivers 69mWRMS (typ) to a 32Ω single-ended stereo load with less than 1% distortion (THD+N) when
A_VDD = 3.3V. The earpiece speaker amplifier delivers 58mWRMS (typ) to a 32Ω bridged-tied load with less than
1% distortion (THD+N) when A_VDD = 3.3V. The loudspeaker amplifier delivers up to 495mW into an 8Ω load
with less than 1% distortion when LS_VDD = 3.3V and up to 1.2W when LS_VDD = 5.0V.
The LM49350 employs advanced techniques to reduce power consumption, to reduce controller overhead, to
speed development time, and to eliminate click and pop. Boomer audio power amplifiers were designed
specifically to provide high quality output power with a minimal amount of external components. It is therefore
ideally suited for mobile phone and other low voltage applications where minimal power consumption, PCB area
and cost are primary requirements.
2
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
LM49350 Overview
I/O_VDD D_VDD
AUX_OUT+
PORT1_SDO
PORT2_CLK
PORT2_SYNC
PORT2_SDI
PORT2_SDO
6'
LEFT
ADC
ADC
EFFECTS
VOL CTRL
5-BAND EQ
SOFT CLIP
ALC
DAC
EFFECTS
LS -
CP+
CP-
CHARGE
PUMP
HP_VSS
6'
RIGHT
ADC
6'
LEFT
DAC
VOL CTRL
DIGITAL 3D
5-BAND EQ
SOFT CLIP
ALC
LS +
D
AUX_RIGHT
REGISTERS
AUX_OUT-
MIC_RIGHT
DIGITAL MIXER and AUDIO PORT INTERFACE
PORT1_SDI
AB
MIC_LEFT
2
GPIO
PORT1_SYNC
LSGND
POWER MANAGEMENT
and CONTROL
I C
SLAVE
PORT1_CLK
LS_VDD
AGND
DAC_LEFT
SCL
SDA
A_VDD
DAC_RIGHT
AUX_LEFT
MCLK
CLOCK
NETWORK
with DUAL
6' PLLs
DGND
AB
HPL
AB
HPR
6'
RIGHT
DAC
AUX_L / MONO_IN+
VCM
-46.5 dB to 12 dB
AUX_R / MONO_INVCM
LEFT_MIC/LINE+
LEFT_MIC/LINE-
BG
VCM
6 dB to 36 dB
RIGHT_MIC/LINE+
RIGHT_MIC/LINE-
VREF_FLT
MIC_BIAS
Figure 1. LM49350 Block Diagram
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Typical Application
D_VDD
I/O_VDD A_VDD
LS_VDD
LEFT_MIC-
HP_VSS
LEFT_MIC+
VREF
0.5 - 50
MHz
CP+
MIC_BIAS
CP-
RIGHT_MIC+
RIGHT_MIC-
MCLK
2
I C
LM49350
Baseband
Controller
2
I S/PCM
(PORT1)
GPIO
HPL
HPR
LS+
LS8O
AUX_OUT+
Bluetooth
Transceiver
2
I S/PCM
(PORT2)
AUX_OUT-
32O
DGND LSGND AGND AUX_L AUX_R
Synthesized
FM Radio/TV Tuner
Figure 2. Example Application in Multimedia Phone with a Dedicated Earpiece and Mono Loudspeaker
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D_VDD
I/O_VDD
A_VDD
LS_VDD
LEFT_MIC-
HP_VSS
LEFT_MIC+
VREF
0.5 - 50
MHz
CP+
MIC_BIAS
CP-
RIGHT_MIC+
RIGHT_MIC-
MCLK
2
I C
LM49350
Baseband
Controller
2
I S/PCM
(PORT1)
HPL
HPR
LS+
LS-
AUX_OUT+
Bluetooth
Transceiver
2
I S/PCM
(PORT2)
LM4675
AUX_OUTGPIO
DGND LSGND AGND AUX_L AUX_R
LM4675 Can Be Used
for Stereo 8: Loudspeakers
Synthesized
FM Radio/TV Tuner
Figure 3. Example Application in Multimedia Phone Using Stereo Loudspeaker
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D_VDD
I/O_VDD
A_VDD
LEFT_MIC-
HP_VSS
LEFT_MIC+
VREF
0.5 - 50
MHz
LS_VDD
CP+
MIC_BIAS
CP-
RIGHT _MIC+
RIGHT _MIC-
MCLK
2
I C
Baseband
Controller
HPL
LM49350
2
I S/PCM
(PORT1)
GPIO
HPR
LS+
LS-
AUX_OUT+
Bluetooth
Transceiver
2
I S/PCM
(PORT2)
AUX_OUTMONO_IN+
Voice
Modem
MONO_INDGND LSGND AGND
Figure 4. Example Application in a Multimedia Phone Using a Dedicated RF Module for Voice Modern
Functions
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D_VDD
I/O_VDD
A_VDD
LS_VDD
LEFT_MIC-
HP_VSS
LEFT_MIC+
VREF
0.5 - 50
MHz
CP+
MIC_BIAS
CP-
HPL
MCLK
HPR
2
I C
LM49350
MP3/MP4
CODEC
2
LS+
LS-
I S/PCM
(PORT1)
AUX_OUT+
LM4675
AUX_OUTGPIO
LM4675 Can Be Used
for Stereo Loudspeakers
Bluetooth
Transceiver
2
I S/PCM
(PORT2)
RIGHT_LINE+
RIGHT_LINEAUX_IN+
DGND LSGND AGND
AUX_IN-
Synthesized
FM Radio/
TV Tuner
(Stereo
Differential)
Figure 5. Example Application in a Portable Media Player with a Differential Stereo Line Input
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Connection Diagram
Figure 6. 36 Bump DSBGA
Top View (Bump Side Down)
See Package Number YPG0036TTA
PIN DESCRIPTIONS
8
Pin
Pin Name
Type
Direction
A1
HPR
Analog
Output
A2
A_VDD
Supply
Input
Headphone and mixer power supply input
A3
AGND
Supply
Input
Headphone and mixer ground
A4
VREF_FLT
Analog
Input/Output
Filter point for the microphone power supply and internal references
A5
GPIO
Digital
Input/Output
General purpose input or output
A6
SDA
Digital
Input/Output
I2C interface data line
B1
HPL
Analog
Output
Headphone left output
B2
AUX_R
Analog
Input
Right analog input
B3
AUX_L
Analog
Input
Left analog 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
Negative power supply pin for the headphone amplifier
C2
AUX_OUT+
Analog
Output
Auxiliary positive output
C3
AUX_OUT-
Analog
Output
Auxiliary negative output
C4
PORT2_SDO
Digital
Output
Audio port 2 serial data out
C5
PORT2_CLK
Digital
Input/Output
C6
MCLK
Digital
Input
D1
CP-
Analog
Input/Output
Charge pump flying capacitor negative input
D2
CP+
Analog
Input/Output
Charge pump flying capacitor positive input
D3
MIC_BIAS
Analog
Output
D4
PORT1_SYNC
Digital
Input/Output
D5
PORT1_SDO
Digital
Output
D6
DGND
Supply
Input
Digital ground
E1
LSGND
Supply
Input
Loudspeaker ground
E2
LS_VDD
Supply
Input
Loudspeaker power supply input
E3
RIGHT_MIC-
Analog
Input
Right microphone negative input
E4
LEFT_MIC-
Analog
Output
E5
PORT1_SDI
Digital
Input
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Description
Headphone right output
Audio Port 2 SYNC Signal (can be master or slave)
Audio port 2 clock signal (can be master or slave)
Input clock from 0.5MHz to 50 MHz
Microphone ultra clean supply (2.2V)
Audio Port 1 sync signal (can be master or slave)
Audio Port 1 serial data output
Left microphone negative input
Audio Port 1 serial data input
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PIN DESCRIPTIONS (continued)
Pin
Pin Name
Type
Direction
E6
D_VDD
Supply
Input
Description
F1
LS +
Analog
Output
Loudspeaker positive output
F2
LS -
Analog
Output
Loudspeaker negative output
F3
RIGHT_MIC +
Analog
Input
Right microphone positive input
F4
LEFT_MIC +
Analog
Input
Left microphone positive input
F5
PORT1_CLK
Digital
Input/Output
F6
I/O_VDD
Supply
Input
Digital power supply input
Audio Port 1 clock signal (can be master or slave)
Digital 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 LM49350.
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.
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
−65°C to +150°C
Storage Temperature
Power Dissipation (4)
ESD Ratings
Internally Limited
Human Body Model (5)
Machine Model
2000V
(6)
200V
Junction Temperature
150°C
Thermal Resistance θJA – YPG36 (soldered down to PCB with 2in2 1oz. copper plane)
Soldering Information
(1)
(2)
(3)
(4)
(5)
(6)
60°C/W
See Applications Note AN-1112 (SNVA009).
“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.
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OPERATING RATINGS
−40°C to +85°C
Temperature Range
Supply Voltage
(1)
A_VDD and LS_VDD (1)
2.7V to 5.5V
D_VDD
1.7V to 2.0V
I/O_VDD
1.6V to 4.5V
LS_VDD need to be the highest voltage than A_VDD, D_VDD, and I/O_VDD. For proper power supply sequence, LS_VDD need to be
applied first.
ELECTRICAL CHARACTERISTICS: A_VDD = LS_VDD = 3.3V; 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.
Symbol
Parameter
LM49350
Conditions
Typical (3)
Limit (4)
Units
(Limits)
DC CHARACTERISTICS (Digital current combines D_VDD and I/O_VDD. Analog current combines A_VDD and LS_VDD)
DISD
Digital Shutdown Current
Shutdown Mode,
fMCLK = 13MHz, PLL Off
DIST
Digital Standby Current
Digital Active Current (MP3 Mode)
2
15
µA (max)
fMCLK = 12.288MHz, PMC On only
0.25
1
mA (max)
fMCLK = 11.2896MHz, fS = 44.1kHz,
Stereo DAC On, OSRDAC = 128,
PLL Off, HP On
0.9
2
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
1.5
2
mA (max)
Digital Active Current (CODEC
Mode)-
fMCLK = 11.2896MHz, fS = 44.1kHz,
Mono ADC On, Stereo DAC On,
OSR = 128, PLL Off, MIC On
2.7
3.8
mA (max)
AISD
Analog Shutdown Current
Shutdown Mode
0.3
5
μA (max)
AIST
Analog Standby Quiescent Current
Reference Voltages On only
0.85
1.5
mA (max)
Analog Supply Current (MP3 Mode)
fMCLK = 11.2896MHz, fS = 44.1kHz,
Stereo DAC On, OSRDAC = 128,
PLL Off, HP On
7.8
10
mA (max)
Analog Supply Current (FM Mode)
Stereo Analog Inputs On, HP On
5.3
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
9.8
12
mA (max)
Analog Supply Current (CODEC
Mode)
fMCLK = 11.2896MHz, fS = 44.1kHz,
Mono ADC On, Stereo DAC On,
OSR = 128, PLL Off, MIC On
13
15
mA (max)
PLLIDD
PLL Total Active Current
fMCLK = 13MHz,
fPLLOUT = 12MHz, PLL On only
2.9
5.5
mA (max)
HPIDD
Headphone Quiescent Current
Stereo HP On only
3.5
mA
LSIDD
Loudspeaker Quiescent Current
LS On only
2.9
mA
MICIDD
Microphone Quiescent Current
mono MIC + MIC Bias On
0.5
mA
ADCIDD
ADC Total Active Current
fS = 48kHz, Stereo
9
mA
DACIDD
DAC Total Active Current
fS = 48kHz, Stereo
5.5
mA
DIDD
AIDD
(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; 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.
Symbol
Parameter
Conditions
AUXINIDD
Auxiliary Input Amplifier Quiescent
Current
AUXOUTIDD
Auxiliary Output Amplifier Quiescent
Current
LM49350
Typical (3)
Limit (4)
Units
(Limits)
Stereo Auxiliary Inputs enabled
0.7
mA
AUX_LINE_OUT enabled
0.5
mA
Earpiece mode enabled
1.0
mA
Loudspeaker Efficiency
PO = 400mW, RL = 8Ω
83
%
Total Harmonic Distortion + Noise
PO = 400mW, f = 1kHz,
RL = 8Ω, Mono Input Signal
0.07
%
LOUDSPEAKER AMPLIFIER
LSEFF
THD+N
RL = 8Ω, f = 1kHz, THD+N = 1%, Mono Input Signal
PO
LS_VDD = 3.3V
LS_VDD = 4.2V
LS_VDD = 5V
Output Power
495
825
1.2
400
mW (min)
mW
W
RL = 4Ω, f = 1kHz, THD+N = 1%, Mono Input Signal
LS_VDD = 3.3V
LS_VDD = 4.2V
LS_VDD = 5V
800
1.4
2
mW
W
W
PSRR
Power Supply Rejection Ration
VRIPPLE = 200mVP-P
fRIPPLE = 217Hz
Mono Input Terminated
VREF = 1.0μF
73
55
dB (min)
SNR
Signal-to-Noise Ratio
Reference = VOUT (1% THD+N )
Gain = 0dB, A-weighted
Mono Input Terminated
95
85
dB (min)
eOS
Output Noise
Gain = 0dB, A-weighted,
Mono Input Terminated
35
VOS
Offset Voltage
Gain = 0dB, form Mono Input
10
TWU
Turn-On Time
PMC Clock = 300kHz
28
µV
50
mV (max)
ms
HEADPHONE AMPLIFIERS
THD+N
Total Harmonic Distortion + Noise
PO = 7.5mW, f = 1kHz,
RL = 32Ω
Stereo Analog Input Signal
PO
Headphone Output Power
Power Supply Rejection Ratio
PSRR
SNR
eOS
Signal-to-Noise Ratio
Output Noise
XTALK
Crosstalk
ΔACH-CH
Channel-to-Channel Gain Matching
0.025
0.1
% (max)
RL = 32Ω, f = 1kHz, THD+N = 1%,
Stereo Analog Input Signal
69
60
mW (min)
VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
Stereo Analog Inputs Terminated,
VREF = 1.0μF, Mono Differential Input
Mode
97
75
dB (min)
Reference = VOUT (1% THD+N )
Gain = 0dB, A-weighted
Stereo Inputs Terminated
106
98
dB (min)
Reference = VOUT (0dBFS ) Gain =
0dB,
A-weighted, I2S Input = Digital Zero
96
90
dB (min)
Gain = 0dB, A-weighted,
Stereo Inputs Terminated
8
µV
Gain = 0dB, A-weighted,
I2S Input = Digital Zero
16
µV
PO = 60mW, f = 1kHz,
RL = 32Ω
Stereo Analog Input Signal
71
dB
0.03
dB
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ELECTRICAL CHARACTERISTICS: A_VDD = LS_VDD = 3.3V; 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.
Symbol
VOS
Parameter
Limit (4)
Units
(Limits)
0.5
6
mV (max)
DAC Gain = 0dB, From DAC Input
fMCLK = 12.288MHz, PLL off
1
6
mV (max)
PMC Clock = 300kHz
28
ms
AUX_LINE_OUT
RL = 5kΩ, VOUT = 1VRMS
0.004
%
Earpiece mode, f = 1kHz
RL = 32Ω BTL, POUT = 20mW
0.08
%
Earpiece mode, f = 1kHz
RL = 32Ω BTL, THD+N = 1%
58
VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
Mono Input terminated, CREF = 1μF
AUX_LINE_OUT
100
VRIPPLE = 200mVP-P, fRIPPLE = 217Hz
Mono Input terminated, CREF = 1μF
Earpiece mode
94
AUX Gain = 0dB
From Differential Mono Input
Output Offset Voltage
TWU
LM49350
Typical (3)
Conditions
Turn-On Time
AUXILIARY OUTPUTS
THD+N
Total Harmonic Distortion + Noise
POUT
Output Power
PSRR
Power Supply Rejection Ratio
45
mW (min)
dB
62
dB (min)
SNR
Signal-to-Noise Ratio
Gain = 0dB, VREF = VOUT (1%THD+N)
A-weighted, Mono Input Terminated
100
dB
∈OUT
Output Noise
Gain = 0dB, VREF = VOUT (1%THD+N)
A-weighted, Mono Input Terminated
13
μV
Gain = 0dB, From Mono Input
AUX_LINE_OUT
7
mV
Gain = 0dB, From Mono Input
Earpiece mode
3
Turn-On Time
PMC Clock = 300kHz
28
ms
THD+NADC
ADC Total Harmonic Distortion +
Noise
Differential Line Input
VIN = 200mVRMS, f = 1kHz
Gain = 0dB
0.03
%
PBADC
ADC Passband
HPF On, fS = 48kHz
Lower -3dB Point
300
Hz
0.41*fS
kHz
ADC Compensated
0.1
dB
Reference = VOUT (0dBFS )
Gain = 6dB,
A-weighted From MIC, fS = 8kHz
90
dB
Reference = VOUT (0dBFS )
Gain = 0dB,
A-weighted From Stereo Input,
fS = 48kHz
94
dB
1
VRMS
0.05
%
1
VRMS
VOS
Output Offset Voltage
TWU
15
mV (max)
STEREO ADC
HPF On, Upper -3dB Point
RADC
ADC Ripple
SNRADC
ADCLEVEL
ADC Signal-to-Noise Ratio
ADC Full Scale Input Level
STEREO DAC
I2S Input
VIN = 500mFFSRMS, f = 1kHz
Gain = 0dB
THD+NDAC
DAC Total Harmonic Distortion +
Noise
DACLEVEL
DAC Full Scale Output Level
RDAC
DAC Ripple
PBDAC
DAC Passband
Upper –3dB Point
SNRDAC
DAC Signal-to-Noise Ratio
fS = 48kHz, A-weighted
12
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0.1
dB
0.45*fS
kHz
96
dB
Copyright © 2008–2012, Texas Instruments Incorporated
Product Folder Links: LM49350 LM49350RLEVAL
LM49350, LM49350RLEVAL
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
ELECTRICAL CHARACTERISTICS: A_VDD = LS_VDD = 3.3V; 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.
Symbol
Parameter
Conditions
LM49350
Typical (3)
Limit (4)
Units
(Limits)
MIC BIAS
VBIAS
Microphone Bias Voltage
MIC input selected
2.2
V
–46.5
dB
Maximum Gain
12
dB
Minimum Gain
–76.5
dB
Maximum Gain
18
dB
Minimum Gain
–76.5
dB
Maximum Gain
18
dB
Minimum Gain
6
dB
Maximum Gain
36
dB
VOLUME CONTROL
VCRAUX
VCRDAC
VCRADC
Stereo Input Volume Control Range
DAC Volume Control Range
ADC Volume Control Range
Minimum Gain
VCRMIC
MIC Volume Control Range
SSAUX
AUX Volume Control Stepsize
1.5
dB
SSDAC
DAC Volume Control Stepsize
1.5
dB
SSADC
DAC Volume Control Stepsize
1.5
dB
SSMIC
MIC Volume Control Stepsize
2
SVAUX
AUX Volume Setting Variation
±1
dB (max)
SVMIC
MIC Volume Setting Variation
±1
dB (max)
dB
ANALOG INPUTS
AUXR_RIN
AUXL_RIN
Right Auxiliary Input Impedance
Right Auxiliary Input Impedance
AUXR Gain = 12dB
17.5
kΩ
AUXR Gain = 0dB
38
kΩ
AUXR Gain = –46.5dB
64
kΩ
AUXL Gain = 12dB
17.5
kΩ
AUXL Gain = 0dB
38
kΩ
AUXL Gain = –46.5dB
64
kΩ
MICR_RIN
Right Microphone Input Impedance
All MIC gain settings
50
kΩ
MICL_RIN
Left Microphone Input Impedance
All MIC gain settings
50
kΩ
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LM49350, LM49350RLEVAL
SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
www.ti.com
TIMING CHARACTERISTICS: DVDD = I/OVDD = 1.8V (1) (2)
The following specifications apply for RL(SP) = 8Ω, RL(HP) = 32Ω, f = 1kHz, unless otherwise specified. Limits apply for TA =
25°C.
LM49350
Symbol
Parameter
Conditions
Typical (3
)
Limit (4)
Units
(Limits)
PLL
fIN
PLL Input Frequency Range
Minimum MCLK Frequency
0.5
MHz (min)
Maximum MCLK Frequency
50
MHz (max)
DIGITAL AUDIO INTERFACE TIMING
tBCLKR
BCK rise time
3
ns (max)
tBCLKCF
BCK fall time
3
ns (max)
tBCLKDS
BCK duty cycle
tDL
WS Propagation Delay from BCK
falling edge
10
ns (max)
tDST
DATA Setup Time to BCK Rising Edge
10
ns (min)
tDHT
DATA Hold Time from BCK Rising
Edge
10
ns (min)
SCL Frequency
400
kHz (max)
1
Hold Time (repeated START
Condition)
0.6
μs (min)
2
Clock Low Time
1.3
μs (min)
3
Clock High Time
600
ns (min)
4
Setup Time for a Repeated START
Condition
600
ns (min)
Output
300
900
ns (min)
ns (max)
Input
0
900
ns (min)
ns (max)
50
%
CONTROL INTERFACE TIMING
5
6
Data Hold Time
Data Setup Time
100
ns (min)
ns (min)
ns (max)
7
Rise Time of SDA and SCL
20+0.1CB
300
8
Fall Time SDA and SCL
15+0.1CB
300
ns (min)
ns (max)
9
Setup Time for STOP Condition
600
ns (min)
10
Bus Free Time Between a STOP and
START Condition
1.3
μs (min)
CB
Bus Capacitance
10
200
pF (min)
pF(max)
(1)
(2)
(3)
(4)
14
“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.
Submit Documentation Feedback
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LM49350, LM49350RLEVAL
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
TIMING CHARACTERISTICS: DVDD = I/OVDD = 1.8V (1) (2)
The following specifications apply for RL(SP) = 8Ω, RL(HP) = 32Ω, f = 1kHz, unless otherwise specified. Limits apply for TA =
25°C.
LM49350
Symbol
Parameter
Conditions
Typical (
3)
Limit (4)
Units
(Limit)
PLL
fIN
PLL Input Frequency Range
Minimum MCLK Frequency
0.5
MHz (min)
Maximum MCLK Frequency
50
MHz (max)
I2S MASTER TIMING
I2S_CLKPER
I2S_CLK Period
I2S Master
81.38
ns
tCLK_L
I2S_CLK Low Time
I2S Master
37
ns
2
37
ns
2
tCLK_H
I2S_CLK High Time
tWS_DLY
WS Propagation Delay from I2S_CLK
falling edge
I S Master
I S Master
21
ns
tSDO_DLY
SDO Propagation Delay from I2S_CLK
falling edge
I2S Master
21
ns
tDST
SDI Setup Time to I2S_CLK Rising
Edge
I2S Master
20
ns
tDHT
SDI Hold Time to I2S_CLK Rising Edge
I2S Master
20
ns
2
I S SLAVE TIMING
I2S_CLKPER
I2S_CLK Period
I2S Slave
81.38
ns (min)
tCLK_L
I2S_CLK Low Time
I2S Slave
37
ns (min)
37
ns (min)
2
tCLK_H
I2S_CLK High Time
tSDO_DLY
SDO Propagation Delay from I2S_CLK
falling edge
I S Slave
I S Slave
tDST
SDI Setup Time to I2S_CLK Rising
Edge
I2S Slave
2
2
21
ns
20
ns (min)
tDHT
SDI Hold Time to I2S_CLK Rising Edge
I S Slave
20
ns (min)
tWS_ST
WS Setup Time to I2S_CLK Rising
Edge
I2S Slave
20
ns (min)
20
ns (min)
tWS_HT
(1)
(2)
(3)
(4)
WS Hold Time to I2S_CLK Rising Edge
2
I S Slave
“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.
Copyright © 2008–2012, Texas Instruments Incorporated
Product Folder Links: LM49350 LM49350RLEVAL
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LM49350, LM49350RLEVAL
SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
www.ti.com
TCLK_H
TCLK_L
I2S_CLKPER
I2S_CLK
(CLK_PHASE = 0)
I2S_WS
TWS_DLY
I2S_SDI
tDST
tDHT
I2S_SDO
TSDO_DLY
w = write (SDA = “0”)
r = read (SDA = “1”)
ack = acknowledge (SDA pulled down by slave)
rs = repeated start
Figure 7. Timing for I2S Master
TCLK_H
TCLK_L
I2S_CLKPER
I2S_CLK
(CLK_PHASE = 0)
TWS_ST
I2S_WS
TWS_HT
I2S_SDI
tDST
tDHT
I2S_SDO
TSDO_DLY
w = write (SDA = “0”)
r = read (SDA = “1”)
ack = acknowledge (SDA pulled down by slave)
rs = repeated start
Figure 8. Timing for I2S Slave
16
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LM49350, LM49350RLEVAL
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
TYPICAL PERFORMANCE CHARACTERISTICS
+1
DAC Frequency Response
fS = 48kHz, OSR = 128
+1
+0.8
DAC Frequency Response
fS = 8kHz, OSR = 128
+0.5
+0.6
MAGNITUDE (dB)
MAGNITUDE (dB)
-0
+0.4
+0.2
+0
-0.2
-0.4
-0.5
-1
-1.5
-2
-0.6
-2.5
-0.8
-1
20
-3
50 100 200 500 1k 2k
20
5k 10k 20k
50
100 200
500
1k
2k
4k
FREQUENCY (Hz)
FREQUENCY (Hz)
Figure 9.
Figure 10.
Stereo Audio ADC Frequency Response
fS = 48kHz, OSR = 128, CIN = 1μF, MIC gain = 6dB
Stereo Audio ADC Frequency Response
fS = 8kHz, OSR = 128, CIN = 1μF, MIC gain = 6dB
+1
+0.5
+0.5
-0
-0
MAGNITUDE (dB)
MAGNITUDE (dB)
+1
-0.5
-1
-1.5
-0.5
-1
-1.5
-2
-2
-2.5
-2.5
-3
20
50 100 200 500 1k
2k
-3
5k 10k 20k
20
50
100 200
500
1k
2k
4k
FREQUENCY (Hz)
FREQUENCY (Hz)
Figure 11.
Figure 12.
Stereo Audio ADC HPF Frequency Response
fS = 48kHz, OSR = 128, CIN = 1μF, MIC gain = 6dB
(Top-No HPF, Upper-HPF_Mode = '101',
Lower-HPF_Mode = '110)'
Bottom-HPF_Mode = '111'
Mono Voice ADC Frequency Response
fS = 48kHz, OSR = 128, CIN = 1μF, MIC gain = 6dB
+0
+1
-10
+0.5
-0
MAGNITUDE (dB)
MAGNITUDE (dB)
-20
-30
-40
-50
-60
-70
-0.5
-1
-1.5
-2
-80
-2.5
-90
-100
20
50 100 200 500 1k 2k
5k 10k 20k
-3
20
50 100 200 500 1k
FREQUENCY (Hz)
Figure 13.
2k
5k 10k 20k
FREQUENCY (Hz)
Figure 14.
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LM49350, LM49350RLEVAL
SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
www.ti.com
TYPICAL PERFORMANCE CHARACTERISTICS (continued)
Mono Voice ADC Frequency Response
fS = 8kHz, OSR = 128, CIN = 1μF, MIC gain = 6dB
Mono Voice ADC HPF Frequency Response
fS = 48kHz, OSR = 128, CIN = 1μF, MIC gain = 6dB
(Top-No HPF)
(From Left to Right:
HPF_Mode = '000', '001', '010', '011', '100')
+1
+0
+0.5
-10
-20
MAGNITUDE (dB)
MAGNITUDE (dB)
-0
-0.5
-1
-1.5
-2
-30
-40
-50
-60
-70
-80
-2.5
-90
-3
20
50
100 200
500
1k
2k
-100
20
4k
50 100 200 500 1k 2k
FREQUENCY (Hz)
Figure 15.
Figure 16.
Mono Voice ADC HPF Frequency Response
fS = 8kHz, OSR = 128, CIN = 1μF, MIC gain = 6dB
(Top-No HPF)
(From Left to Right:
HPF_Mode = '000', '001', '010', '011', '100')
ADC Output THD+N vs Frequency
Differential Line Input, Aux Gain = 0dB
VIN = 200mVRMS, fS = 48kHz
+0
10
5
-10
2
1
-30
0.5
THD+N (%)
MAGNITUDE (dB)
-20
-40
-50
-60
-90
0.002
0.001
20
-100
20
0.2
0.1
0.05
0.02
0.01
0.005
-70
-80
50
100 200
500
1k
2k
4k
50 100 200 500 1k 2k
Figure 17.
Figure 18.
ADC Output THD+N vs Frequency
Differential MIC Input, MIC Gain = 6dB
VIN = 100mVRMS, fS = 48kHz
ADC Output THD+N vs VIN
Differential Line Input, Aux Gain = 0dB
VIN = 1kHz, fS = 48kHz
10
10
5
5
2
1
2
1
THD+N (%)
0.5
THD+N (%)
5k 10k 20k
FREQUENCY (Hz)
FREQUENCY (Hz)
0.2
0.1
0.05
0.002
0.001
20
0.5
0.2
0.1
0.02
0.01
0.005
0.05
0.02
50 100 200 500 1k 2k
5k 10k 20k
FREQUENCY (Hz)
0.01
1m
5m
2m
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10m
500m
20m 100m
50m 200m
1
2
INPUT VOLTAGE (VRMS)
Figure 19.
18
5k 10k 20k
FREQUENCY (Hz)
Figure 20.
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
TYPICAL PERFORMANCE CHARACTERISTICS (continued)
ADC Output THD+N vs VIN
Differential MIC Input, MIC Gain = 6dB
VIN = 1kHz, fS = 48kHz
Loudspeaker THD+N vs Frequency
Differential Aux Input, Aux Gain = 0dB
VDD = 3.3V, POUT = 400mW, RL = 8Ω
10
10
5
2
2
1
1
THD+N (%)
THD+N (%)
5
0.5
0.2
0.5
0.2
0.1
0.1
0.05
0.05
0.02
0.02
0.01
1m 2m
0.01
20
5m 10m 20m 50m100m 200m 500m 1
50 100 200 500 1k 2k
Figure 21.
Figure 22.
Loudspeaker THD+N vs Frequency
Differential Aux Input, Aux Gain = 0dB
VDD = 5V, POUT = 400mW, RL = 8Ω
Loudspeaker THD+N vs Frequency
Differential Aux Input, Aux Gain = 0dB
LS_VDD = 3.3V, POUT = 500mW, RL = 4Ω
10
10
5
5
2
2
THD + N (%)
THD+N (%)
1
0.5
0.2
0.1
1
0.5
0.2
0.1
0.05
0.05
0.02
0.02
0.01
20
50 100 200 500 1k 2k
0.01
20
5k 10k 20k
50 100 200 500 1k 2k
FREQUENCY (Hz)
5k 10k 20k
FREQUENCY (Hz)
Figure 23.
Figure 24.
Loudspeaker THD+N vs Output Power
Differential Aux Input, Aux Gain = 0dB
VDD = 3.3V, VIN = 1kHz, RL = 8Ω
Loudspeaker THD+N vs Output Power
Differential Aux Input, Aux Gain = 0dB
VDD = 4.2V, VIN = 1kHz, RL = 8Ω
10
10
5
5
2
2
1
THD+N (%)
1
THD+N (%)
5k 10k 20k
FREQUENCY (Hz)
INPUT VOLTAGE (VRMS)
0.5
0.2
0.5
0.2
0.1
0.1
0.05
0.05
0.02
0.02
0.01
10m 20m
50m 100m 200m 500m
1
2
0.01
10m 20m
OUTPUT POWER (W)
Figure 25.
50m 100m 200m 500m
1
2
OUTPUT POWER (W)
Figure 26.
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www.ti.com
TYPICAL PERFORMANCE CHARACTERISTICS (continued)
Loudspeaker THD+N vs Output Power
Differential Aux Input, Aux Gain = 0dB
VDD = 5V, VIN = 1kHz, RL = 8Ω
Loudspeaker THD+N vs Output Power
Differential Aux Input, Aux Gain = 0dB
LS_VDD = 3.3V, RL = 4Ω, f = 1kHz
10
5
5
2
2
1
1
THD + N (%)
THD+N (%)
10
0.5
0.2
0.1
0.5
0.2
0.1
0.05
0.05
0.02
0.02
0.01
10m 20m
50m 100m 200m 500m
1
0.01
10m 20m 50m 100m 200m 500m 1
2
OUTPUT POWER (W)
OUTPUT POWER (W)
Figure 27.
Figure 28.
Loudspeaker THD+N vs Output Power
Differential Aux Input, Aux Gain = 0dB
LS_VDD = 4.2V, RL = 4Ω, f = 1kHz
Loudspeaker THD+N vs Output Power
Differential Aux Input, Aux Gain = 0dB
LS_VDD = 5V, RL = 4Ω, f = 1kHz
10
5
5
2
2
1
1
THD + N (%)
THD + N (%)
10
0.5
0.2
0.1
0.5
0.2
0.1
0.05
0.05
0.02
0.02
0.01
10m 20m 50m 100m 200m 500m 1
0.01
10m 20m 50m 100m 200m 500m 1
2 3
OUTPUT POWER (W)
+0
OUTPUT POWER (W)
Figure 30.
Loudspeaker PSRR vs Frequency
LS_VDD = 3.3V, Aux Gain = 0dB
Differential Aux Input to Ground
VRIPPLE = 200mVPP
Loudspeaker PSRR vs Frequency
LS_VDD = 4.2V, Aux Gain = 0dB
Differential Aux Input to Ground
VRIPPLE = 200mVPP
+0
-10
-20
-20
-30
-30
-40
PSRR (dB)
-40
PSRR (dB)
2 3
Figure 29.
-10
20
2 3
-50
-60
-70
-50
-60
-70
-80
-80
-90
-90
-100
-100
-110
-110
-120
20 50 100 200 500 1k 2k 5k 10k 20k 50k100k
-120
20 50 100 200 500 1k 2k 5k 10k 20k 50k100k
FREQUENCY (Hz)
FREQUENCY (Hz)
Figure 31.
Figure 32.
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
TYPICAL PERFORMANCE CHARACTERISTICS (continued)
+0
Loudspeaker PSRR vs Frequency
LS_VDD = 5V, Aux Gain = 0dB
Differential Aux Input to Ground
VRIPPLE = 200mVPP
Headphone THD+N vs Frequency
Stereo Aux Input, Aux Gain = 0dB
VDD = 3.3V, POUT = 7.5mW, RL = 32Ω
10
5
-10
2
1
-40
0.5
THD+N (%)
PSRR (dB)
-20
-30
-50
-60
-70
-80
0.2
0.1
0.05
0.02
0.01
0.005
-90
-100
-110
0.002
0.001
20
-120
20 50 100 200 500 1k 2k 5k 10k 20k 50k100k
50 100 200 500 1k 2k
FREQUENCY (Hz)
Figure 33.
Figure 34.
Headphone THD+N vs Frequency
Stereo Aux Input, Aux Gain = 0dB
VDD = 5V, POUT = 7.5mW, RL = 32Ω
Headphone THD+N vs Frequency
Differential Aux Input, Aux Gain = 0dB
A_VDD = 3.3V, POUT = 7.5mW, RL = 16Ω
10
5
2
1
2
0.5
1
THD + N (%)
THD+N (%)
10
5
0.2
0.1
0.05
0.02
0.01
0.005
0.002
0.001
20
0.5
0.2
0.1
0.05
0.02
50 100 200 500 1k 2k
0.01
20
5k 10k 20k
50 100 200 500 1k 2k
FREQUENCY (Hz)
5k 10k 20k
FREQUENCY (Hz)
Figure 35.
Figure 36.
Headphone THD+N vs Output Power
Stereo Aux Input, Aux Gain = 0dB
VDD = 3.3V, VIN = 1kHz, RL = 32Ω
Headphone THD+N vs Output Power
Stereo Aux Input, Aux Gain = 0dB
VDD = 5V, VIN = 1kHz, RL = 32Ω
10
10
5
5
2
2
1
THD+N (%)
1
THD+N (%)
5k 10k 20k
FREQUENCY (Hz)
0.5
0.2
0.5
0.2
0.1
0.1
0.05
0.05
0.02
0.02
0.01
1m
2m
5m
10m 20m
50m 100m
0.01
1m
2m
5m
10m 20m
OUTPUT POWER (W)
OUTPUT POWER (W)
Figure 37.
Figure 38.
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TYPICAL PERFORMANCE CHARACTERISTICS (continued)
Headphone THD+N vs Output Power
A_VDD = 3.3V, Stereo Aux Input, Aux Gain = 0dB
RL = 16Ω, f = 1kHz
10
Headphone PSRR vs Frequency
Differential Aux Input to Ground, Aux Gain = 0dB
VRIPPLE = 200mVPP
+0
-10
5
-30
1
-40
PSRR (dB)
THD + N (%)
-20
2
0.5
0.2
0.1
-50
-60
-70
-80
-90
0.05
-100
0.02
0.01
1m
-110
2m
5m
10m
20m 50m
-120
100m
20 50 100 200 500 1k 2k
5k 10k 20k 50k 100k
FREQUENCY (Hz)
OUTPUT POWER (W)
Figure 39.
Figure 40.
Headphone Crosstalk vs Frequency
Stereo Aux Inputs, Aux Gain = 0dB, RL = 32Ω
Earpiece THD+N vs Frequency
Differential Aux Input, Aux Gain = 0dB
A_VDD = 3.3V, POUT = 20mW, RL = 32Ω
+0
10
-10
5
2
-30
THD + N (%)
CROSSTALK (dB)
-20
-40
-50
-60
-70
0.5
0.2
0.1
0.05
-80
0.02
-90
-100
20
1
50 100 200 500 1k 2k
0.01
20
5k 10k 20k
50 100 200 500 1k 2k
FREQUENCY (Hz)
5k 10k 20k
FREQUENCY (Hz)
Figure 41.
Figure 42.
Earpiece THD+N vs Output Power
Differential Aux Input, Aux Gain = 0dB
A_VDD45 = 3.3V, RL = 32Ω, f = 1kHz
Earpiece PSRR vs Frequency
Differential Aux Input to Ground, Aux Gain = 0dB
VRIPPLE = 200mVPP
10
+0
-10
5
-30
1
-40
PSRR (dB)
THD + N (%)
-20
2
0.5
0.2
0.1
-50
-60
-70
-80
-90
0.05
-100
0.02
0.01
1m
-110
2m
5m
10m
20m 50m
100m
-120
20
50 100 2005001k 2k
OUTPUT POWER (W)
FREQUENCY (HZ)
Figure 43.
22
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5k 10k 20k 50k 100k
Figure 44.
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
TYPICAL PERFORMANCE CHARACTERISTICS (continued)
AUXOUT THD+N vs Frequency
Differential Aux Input, Aux Gain = 0dB
VDD = 5V, VOUT = 1VRMS, RL = 5kΩ
AUXOUT THD+N vs Output Voltage
Differential Aux Input, Aux Gain = 0dB
VIN = 1kHz, RL = 5kΩ
10
5
10
2
1
2
1
0.5
5
THD+N (%)
THD+N (%)
0.5
0.2
0.1
0.05
0.02
0.01
0.005
0.002
0.001
20
0.2
0.1
0.05
0.02
0.01
0.005
50 100 200 500 1k 2k
0.002
0.001
10m 20m
5k 10k 20k
FREQUENCY (Hz)
50m 100m 200m 500m
1
2
OUTPUT VOLTAGE (VRMS)
Figure 45.
Figure 46.
Figure 47.
AUXOUT PSRR vs Frequency
Differential Aux Input to Ground, Aux Gain = 0dB
VRIPPLE = 200mVPP
+0
-10
-20
-30
PSRR (dB)
-40
-50
-60
-70
-80
-90
-100
-110
-120
20 50 100 200 500 1k 2k 5k 10k 20k 50k100k
FREQUENCY (Hz)
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SYSTEM CONTROL
Method 1. I2C Compatible Interface
I2C SIGNALS
In I2C mode the LM49350 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
LM49350 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 48. 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 49. 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 LM49350 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.
24
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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 50. 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
MSB
ack from slave
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 51. Example I2C Write Cycle
When a READ function is to be accomplished, a WRITE function must precede the READ function, as shown in
the Figure 52 waveform.
ack from slave
ack from slave repeated start
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 52. Example I2C Read Cycle
SDA
10
8
7
6
1
8
2
7
SCL
5
1
3
4
9
Figure 53. I2C Timing Diagram
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I2C TIMING PARAMETERS (1)
Symbol
(1)
26
Limit
Parameter
1
Hold Time (repeated) START Condition
2
3
Min
Max
Units
0.6
µs
Clock Low Time
1.3
µs
Clock High Time
600
ns
4
Setup Time for a Repeated START Condition
600
5
Data Hold Time (Output direction, delay generated by LM49350)
300
900
ns
5
Data Hold Time (Input direction, delay generated by the Master)
0
900
ns
6
Data Setup Time
7
Rise Time of SDA and SCL
20+0.1Cb
300
ns
8
Fall Time of SDA and SCL
15+0.1Cb
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
CB
Capacitive Load for Each Bus Line
10
ns
100
ns
200
pF
NOTE: Data specified by design
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
Device Register Map
Table 1. Device Register Map (1)
Address
Register
7
6
5
4
3
2
1
0
BASIC SETUP
0x00h
PMC
CHIP
PORT2
PORT1
MCLK
OSC
PLL2
PLL1
CHIP
SETUP
ACTIVE
CLK OVR
CLK OVR
OVR
ENB
ENB
ENB
ENABLE
0x01h
PMC
CLOCKS
0x02h
PMC
CLK_DIV
PMC_CLK_SEL
PMC_CLK_DIV(R)
PLLs
0x03h
PLL2_CLK_SEL
PLL1_CLK_SEL
0x04h
PLL1 M
0x05h
PLL1 N
PLL1 M
0x06h
PLL1
N_MOD
0x07h
PLL1 P1
PLL1 P1 [7:0]
0x08h
PLL1 P2
PLL1 P2[7:0]
0x09h
PLL2 M
0x0Ah
PLL2 N
0x0Bh
PLL2
N_MOD
0x0Ch
PLL2 P
0x10h
CLASSD
AUXL_LS
AUXR_LS
MICL_LS
MICR_LS
DACL_LS
DACR_LS
0x11h
HEAD
PHONESL
AUXL_HPL
AUXR_HPL
MICL_HPL
MICR_HPL
DACL_HPL
DACR_HPL
AUXL_HPR
AUXR_
HPR
MICL_HPR
MICR_HPR
DACL_
HPR
DACR_
HPR
AUXL_AX
AUXR_AX
MICL_AX
MICR_AX
DACL_AX
DACR_AX
CP_
FORCE
AUX-6dB
LS-6dB
HP-6dB
EPMODE
MICL_
ADCR
MICR_
ADCL
DACL_
ADCR
DACR_
ADCL
PLL1 N
PLL2 P2[8]
PLL1 P1[8]
PLL1 N_MOD
PLL2 M
PLL2 N
PLL2 P[8]
PLL2 N_MOD
PLL2 P[7:0]
ANALOG MIXER
0x12h
HEAD
PHONESR
0x13h
AUX_OUT
0x14h
OUTPUT
OPTIONS
0x15h
ADC
AUXL_
ADCR
AUXR_
ADCL
0x16h
MICL_LVL
MUTE
SE/DIFF
MIC_L_LEVEL
0x17h
MICR_LVL
MUTE
SE/DIFF
MIC_R_LEVEL
0x18h
AUXL_LVL
0x19h
AUXR_LVL
DIFF_MODE
0x20h
ADC BASIC
DSPONLY
0x21h
ADC
CLOCK
0x22h
ADC_DSP
FROM
LINEL
AUX_L_LEVEL
FROM
LINER
AUX_R_LEVEL
ADC
ADC_CLK_SEL
MUTE_R
MUTE_L
ADC_OSR
MONO
ADC_CLK_DIV (T)
ADC_TRIM
DAC
(1)
0x30h
DAC_BASI
C
0x31h
DAC_CLOC
K
0x32h
DAC_DSP
DSPONLY
DAC_CLK_SEL
MUTE_R
MUTE_L
DAC_OSR
DAC_CLK_DIV (S)
DAC_TRIM
Unless otherwise specified, the default values of the I2C registers is 0x00h.
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Table 1. Device Register Map(1) (continued)
Address
Register
7
6
5
4
3
2
1
0
DIGITAL MIXER
0x40h
IPLVL1
PORT2_RX_R_LVL
PORT2_RX_L_LVL
PORT1_RX_R_LVL
PORT1_RX_L_LVL
0x41h
IPLVL2
INTERP_L_LVL
INTERP_R_LVL
ADC_R_LVL
ADC_L_LVL
0x42h
OPPORT1
R_SEL
L_SEL
0x43h
OPPORT2
0x44h
OPDAC
0x45h
OPDECI
MONO
SWAP
SWAP
MONO
SWAP
ADCR
PORT2R
R_SEL
PORT1R
MXRCLK_SEL
L_SEL
ADCL
PORT2L
R_SEL
PORT1L
L_SEL
AUDIO PORT 1
0x50h
BASIC
0x51h
CLK_GEN1
STEREO_SY STEREO_S
NC_
YNC_
MODE
PHASE
CLK_PH
SYNC_MS
CLK_MS
CLK_SEL
TX_ENB
RX_ENB
STEREO
HALF_CYCLE_DIVDER
SYNTH_DE
NOM
0x52h
CLK_GEN2
SYNTH_NOM
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
SYNC_WIDTH(MONO MODE)
SYNC_RATE
TX_WIDTH
RX_WIDTH
TX_EXTRA_BITS
AUDIO PORT 2
STEREO_SY STEREO_S
NC_
YNC_
MODE
PHASE
0x60h
BASIC
0x61h
CLK_GEN1
0x62h
CLK_GEN2
0x63h
SYNC_
GEN
0x64h
DATA_
WIDTH
0x65h
RX_MODE
A/ULAW
COMPAND
0x66h
TX_MODE
A/ULAW
COMPAND
CLK_PH
SYNC_MS
CLK_MS
CLK_SEL
TX_ENB
RX_ENB
STEREO
HALF_CYCLE_DIVDER
SYNTH_
DENOM
SYNTH_NOM
SYNC_WIDTH(MONO MODE)
SYNC_RATE
TX_WIDTH
RX_WIDTH
TX_EXTRA_BITS
MSB_POSITION
RX_MODE
MSB_POSITION
TX_MODE
EFFECTS ENGINE
0x70h
ADC FX
0x71h
DAC FX
ADC
ADC
ADC
ADC
ADC
SCLP ENB
EQ ENB
PK ENB
ALC ENB
HPF_ENB
DAC
DAC
DAC
DAC
DAC
SCLP ENB
3D ENB
EQ ENB
PK ENB
ALC ENB
ADC EFFECTS
0x80h
0x81h
0x82h
0x83h
0x84h
0x85h
28
HPF
HPF MODE
ADC
SOURCE
SOURCE
STEREO
ALC 1
OVR
SEL
LINK
ADC
LIMITER
NG_ENB
ALC 2
ADC
NOISE_FLOOR
ALC_TARGET_LEVEL
ALC 3
ADC
ATTACK_RATE
ALC 4
ADC
ALC 5
SAMPLE_RATE
PK_DECAY_RATE
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SNAS359D – SEPTEMBER 2008 – REVISED JUNE 2012
Table 1. Device Register Map(1) (continued)
Address
0x86h
0x87h
0x88h
0x89h
0x8Ah
Register
7
6
5
4
3
2
ADC
ADC
MAX_LEVEL
ALC 7
ADC
MIN_LEVEL
ALC 8
ADC L
ADC_L_LEVEL
LEVEL
ADC R
ADC_R_LEVEL
LEVEL
EQ BAND 1
0x8Ch
EQ BAND 2
0x8Dh
EQ BAND 3
0x8Eh
EQ BAND 4
0x8Fh
0
HOLDTIME
ALC 6
0x8Bh
1
LEVEL
FREQ
Q
LEVEL
FREQ
Q
LEVEL
FREQ
Q
LEVEL
FREQ
EQ BAND 5
LEVEL
FREQ
0x90h
SOFTCLIP
1
SOFT
KNEE
0x91h
SOFTCLIP
2
RATIO
0x92h
SOFTCLIP
3
LEVEL
0x98h
LVLMONL
ADC LEFT LEVEL MONITOR
0x99h
LVLMONR
ADC RIGHT LEVEL MONITOR
0x9Ah
FXCLIP
THRESHOLD
ADC EFFECT MONITORS
0x9Bh
ALCMONL
0x9Ch
ALCMONR
SCLP_R
SCLP_L
EQ_R
EQ_L
CLIP
CLIP
CLIP
CLIP
SCLP_R
SCLP_L
DISTORT
DISTORT
SCLP_L
SCLP_R
DISTORT
DISTORT
GAIN_R
CLIP
GAIN_L
CLIP
ADC_R
ADC_L
CLIP
CLIP
ADC LEFT ALC MONITOR
ADC RIGHT ALC MONITOR
DAC EFFECTS
0xA0h
0xA1h
0xA2h
0xA3h
0xA4h
0xA5h
0xA6h
0xA7h
DAC
STEREO
ALC 1
LINK
DAC
LIMITER
NG_ENB
ALC 2
DAC
NOISE_FLOOR
AGC_TARGET_LEVEL
ALC 3
DAC
ATTACK_RATE
ALC 4
DAC
ALC 5
SAMPLE_RATE
PK_DECAY_RATE
DECAY_RATE/RELEASE_RATE
DAC
HOLDTIME
ALC 6
DAC
MAX_LEVEL
ALC 7
DAC
MIN_LEVEL
ALC 8
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Table 1. Device Register Map(1) (continued)
Address
0xA8h
0xA9h
Register
7
6
5
4
3
DAC L
DAC R
0xABh
EQ BAND 1
0xACh
EQ BAND 2
0xADh
EQ BAND 3
0xAEh
EQ BAND 4
0xAFh
0
DAC_R_LEVEL
LEVEL
DAC_3D
1
DAC_L_LEVEL
LEVEL
0xAAh
2
ATTEN
FILTER_TYPE
EFFECT_
MODE
EFFECT_LEVEL
LEVEL
FREQ
Q
LEVEL
FREQ
Q
LEVEL
FREQ
Q
LEVEL
FREQ
EQ BAND 5
LEVEL
FREQ
0xB0h
SOFTCLIP
1
SOFT
KNEE
0xB1h
SOFTCLIP
2
RATIO
0xB2h
SOFTCLIP
3
LEVEL
THRESHOLD
DAC EFFECT MONITORS
0xB8h
LVLMONL
DAC LEFT LEVEL MONITOR
0xB9h
LVLMONR
DAC RIGHT LEVEL MONITOR
0xBAh
FXCLIP
0xBBh
ALCMONL
0xBCh
ALCMONR
0xE0h
GPIO
SCLP_R
SCLP_L
EQ_R
EQ_L
3D_R
3D_L
CLIP
CLIP
CLIP
CLIP
CLIP
CLIP
SCLP_R
SCLP_L
DISTORT
DISTORT
SCLP_L
SCLP_R
DISTORT
DISTORT
TEMP
SHORT
GAIN_R
CLIP
GAIN_L
CLIP
DAC LEFT ALC MONITOR
DAC RIGHT ALC MONITOR
GPIO
GPIO_RX
GPIO_TX
GPIO_MODE
SPREAD SPECTRUM
SS_
DISABLE
0xF1h
SS
0xF8h
ADC_C0_
LSB
ADC_C0_LSB
0xF9h
ADC_C0_
MSB
ADC_C0_MSB
0xFAh
ADC_C1_
LSB
ADC_C1_LSB
0xFBh
ADC_C1_
MSB
ADC_C1_MSB
0xFCh
ADC_C2_
LSB
ADC_C2_LSB
0xFDh
ADC_C2_M
SB
ADC_C2_MSB
0xFEh
AUX_LINE_
OUT
RSVD
RSVD
ADC COMPENSATION FILTER
30
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AUX_LINE_
OUT
RSVD
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Basic PMC Setup Register
This register is used to control the LM49350's Basic Power Management Setup:
Table 2. PMC_SETUP (0x00h)
Bits
0
Field
CHIP_ENABLE
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.
CHIP _ENABLE
Chip Status
0
Turn Chip Off
1
Turn Chip On
This enables the primary PLL
1
PLL1_ENB
PLL1_ENABLE
PLL1 Status
0
PLL1 Off
1
PLL1 On
This enables the secondary PLL
2
PLL2_ENB
PLL2_ENABLE
PLL2 Status
0
PLL2 Off
1
PLL2 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 clock source 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 is 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.
6
7
(1)
PORT2_CLK_OVR
CHIP_ACTIVE
PORT2_CLK_OVR
Comment
0
I/O control is automatic
1
PORT_CLK input forced on
This bit is used to read back 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 over ride 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 LM49350's Basic Power Management Setup:
Table 3. 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)
01
Internal 300kHz Oscillator
10
DAC SOURCE CLOCK
11
ADC SOURCE CLOCK
PMC Clock Divide Register
This register is used to control the LM49350's Power Management Circuits Clocks:
Table 4. PMC_SETUP (0x02h) (Default data value is 0x50h)
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) to get a ≈300kHz PMC clock
from a 12MHz or 12.288MHz MCLK.
Program this divider with the division you want, 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
LM49350 Clock Network
Refer to Figure 54
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 four different OSR settings (128, 125, 64,
32). The ADC can operate at three different OSR settings (128, 125, 64). 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
one of the LM49350's dual PLLs can be used to generate the desired clock frequency. Otherwise, if a
12.288MHz is available, then the PLL can be bypassed to reduce power consumption. The DAC clock divider (S
divider) or ADC clock divider (T divider) can also be used to generate the correct clock. If an 18.432 MHz clock is
available, the S or T 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, PLL1's output, or PLL2's output.
For instances where a PLL must be used, the PLL input clock can come from three sources. The clock input to
PLL1 or PLL2 can come from the MCLK input, the PORT1_CLK input, or the PORT2_CLK input.
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The LM49350'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 (R 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.
Table 5. 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
192
—
—
24.576 MHz
12.288 MHz
Table 6. ADC Clock Requirements
ADC Sample Rate
(kHz)
Clock Required at B
(OSR = 128)
Clock Required at B
(OSR= 125)
Clock Required at B
(OSR = 64)
8
2.048 MHz
2 MHz
1.024 MHz
11.025
2.8224 MHz
2.75625 MHz
1.4112 MHz
12
3.072 MHz
3 MHz
1.536 MHz
16
4.096 MHz
4 MHz
2.048 MHz
22.05
5.6448 MHz
5.5125 MHz
2.8224 MHz
24
6.144 MHz
6 MHz
3.072 MHz
32
8.192 MHz
8 MHz
4.096 MHz
44.1
11.2896 MHz
11.025 MHz
5.6448 MHz
48
12.288 MHz
12 MHz
6.144 MHz
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-300 kHz
INTERNAL
OSCILLATOR
%R
C
PMC
A
PLL2
%S
Stereo DAC
1 ± 25 MHz
0 _> 50 MHz
PLL1
MCLK
MIXER
0 _ 50 MHz
%T
Stereo ADC
B
AUDIO PORT 1
PORT1_CLK
0 _ 50 MHz
R, S, T = Half Cycle 1, 1.5, 2, 2.5 _> 128
AUDIO PORT 2
PORT2_CLK
Figure 54. Internal Clock Network
PLL Setup Registers
PLL_P2
1 ± 25 MHz
% P1
9
P2A = 0,1 + 0/2 _> 256
P= 0.1 + 0/2 _> 64
0.5 - 50 MHz
Phase Comparator
and Charge Pump
I
%M
9
PLL_P2
140 to
210 MHz
1 ± 25 MHz
VCO
% P2
P2B = 0,1 + 0/2 _> 256
0.7 < 5 MHz
7
%N
6'M
8
N=0, 1 + 0/32 _> 250
8
PLL_M
5
PLL_N PLL_N_MOD
Figure 55. PLL1 Loop
PLL_P
P= 0.1 + 0/2 _> 64
0.5 - 50 MHz
%M
Phase Comparator
and Charge Pump
I
9
VCO
1 ± 25 MHz
140 to 240 MHz
%P
P1 = 0,1 + 0/2 _> 256
0.7 < 5 MHz
7
%N
6'M
8
_
N=0, 1 + 0/32 > 250
PLL_M
8
5
PLL_N PLL_N_MOD
Figure 56. PLL2 Loop
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The LM49350 contains two PLLs for flexible operation of its dual audio ports. PLL1 has a P1 and P2 output
divider thereby allowing PLL1 to generate two distinct clock outputs. The equations for PLL1's generated output
clocks are as follows:
fOUT1 = (fIN . N1 / M1 . P1)
fOUT2 = (fIN . N1 / M1 . P2)
(1)
(2)
N1 = PLL1_N + PLL1_N_MOD
M1 = (PLL1_M + 1) / 2
P1 = (PLL1_P1 + 1) / 2
P2 = (PLL1_P2 + 1) / 2
(3)
(4)
(5)
(6)
where:
The equations for PLL2's generated output clock are as follows:
fOUT3 = (fIN.N2 / M2.P)
(7)
where:
N2 = PLL2_N + PLL2_N_MOD
M2 = (PLL2_M + 1) / 2
P = (PLL2_P + 1) / 2
(8)
(9)
(10)
The VCO frequency and comparison frequencies are as follows:
fVCO = fOUT.P
fCOMP = fIN/M
(11)
(12)
Keep fVCO between 140MHz to 240MHz and keep fCOMP between 700kHz to 5MHz.
Table 7. PLL Settings for Common System Clock Frequencies
fIN (MHz)
M
N
N_MOD
P
fOUT (MHz)
Error (Hz)
12
2.5
32
0
12.5
12288000
0
13
15.5
175
26
12
12287970
–30
14.4
12.5
128
0
12
12288000
0
16.2
13.5
128
0
12.5
12288000
0
16.8
3.5
32
0
12.5
12288000
0
19.2
12.5
96
0
12
12288000
0
19.68
20.5
160
0
12.5
12288000
0
19.8
16.5
128
0
12.5
12288000
0
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
13.5
15.5
213
28
16.5
11289589
–11
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
19.68
20.5
147
0
12.5
11289600
0
19.8
27.5
196
0
12.5
11289600
0
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
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Table 7. PLL Settings for Common System Clock Frequencies (continued)
fIN (MHz)
M
N
N_MOD
P
fOUT (MHz)
Error (Hz)
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
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
Table 8. PLL_CLOCK_SOURCE (0x03h)
Bits
Field
1:0
PLL1_CLK_SEL
Description
This selects the source of the input clock to PLL1
PLL1_CLK_SEL
PLL1 Input Clock Source
00
MCLK
01
PORT1_CLK
10
PORT2_CLK
11
RESERVED
Table 9. PLL1_M (0x04h)
36
Bits
Field
6:0
PLL1_M
Description
This programs the PLL1 M divider to divide from 1 to 64.
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PLL1_M
PLL1 Input Divider Vaue
000000
1
000001
1
000010
1.5
000011
2
000100
2.5
000101
3
—
—
1111101
63
1111110
63.5
1111111
64
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Table 10. PLL1_N (0x05h)
Bits
Field
7:0
PLL1_N
Description
This programs the PLL1 N divider to divide from 1 to 250.
PLL1_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 11. PLL1_N_MOD (0x06h)
Bits
Field
4:0
PLL1_N_MOD
Description
This programs the sigma-delta modulator in PLL1
PLL1_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
PLL1_P1[8]
This sets the MSB of the 1st P Divider on PLL1 which is part of a standard half-cycle divider
control.
6
PLL1_P2[8]
This sets the MSB of the 2nd P Divider on PLL1 which is part of a standard half-cycle divider
control.
Table 12. PLL1_P1 (0x07h)
Bits
Field
7:0
PLL1_P1[7:0]
Description
This programs the 8 LSBs of the PLL1's P1 Divider. These LSBs combine with PLL1_P1[8] which
allows the P1 divider to divide by up to 256
PLL1_P1
P1 Divider Value
000000000
1
000000001
1
000000010
1.5
000000011
2
000000100
2.5
000000101
3
—
—
111111101
255
111111110
255.5
111111111
256
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Table 13. PLL1_P2 (0x08h)
Bits
Field
7:0
PLL1_P2[7:0]
Description
This programs 8 LSBs of PLL1's P2 Divider. These LSBs combine with PLL1_P2[8] which allows
the P2 divider to divide by up to 256
PLL1_P2
P2 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. PLL2_M (0x09h)
Bits
Field
6:0
PLL2_M
Description
This programs the PLL2 M divider to divide from 1 to 64.
PLL2_M
PLL2 Input Divider Value
0000000
1
0000001
1
0000010
1.5
0000011
2
0000100
2.5
0000101
3
—
—
1111101
63
0000010
63.5
1111111
64
Table 15. PLL2_N (0x0Ah)
38
Bits
Field
7:0
PLL2_N
Description
This programs PLL2's N divider to divide from 10 to 250.
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PLL2_N
Comment
00000000 to 00001010
10
00001011
11
00001100
12
00001101
13
00001110
14
00001111
15
—
—
11111000
248
11111001
249
11111010 to 11111111
250
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Table 16. PLL2_N_MOD (0x0Bh)
Bits
Field
4:0
PLL2_N_MOD
Description
This programs the sigma-delta modulator in PLL2
PLL2_N_MOD
5
PLL2_P[8]
Fractional Part of N
00000
0
00001
1/32
00010
2/32
00011
3/32
00100
4/32
00101
5/32
—
—
11101
29/32
11110
30/32
11111
31/32
This is the MSB of the P Divider on PLL2.
Table 17. PLL2_P (0x0Ch)
Bits
Field
Description
7:0
PLL2_P[7:0]
This programs the 8 LSBs of PLL2's P Divider. These LSBs combine with PLL2_P[8] which allows
the P divider to divide by up to 256
PLL2_P
P Divides by
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 LM49350's Analog Mixer:
Table 18. CLASS_D_OUTPUT (0x10h)
Bits
Field
Description
0
DACR_LS
The right DAC output is added to the loudspeaker output.
1
DACL_LS
The left DAC output is added to the loudspeaker output.
2
MICR_LS
The right MIC input is added to the loudspeaker output. Setting this bit enables MIC BIAS.
3
MICL_LS
The left MIC input is added to the loudspeaker output. Setting this bit enables MIC BIAS.
4
AUXR_LS
The right AUX input is added to the loudspeaker output.
5
AUXL_LS
The left AUX input is added to the loudspeaker output.
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CLASS D LOUDSPEAKER AMPLIFIER
The LM49350 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 LM49350 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 LM49350 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 LM49350 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 LM49350 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
LM49350 and in the transducer load. The amount of power dissipation in the LM49350'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 LM49350 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 LM49350’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
LM49350
Class D
Speaker
10 pF
LSFerrite Bead
10 pF
LSGND
Figure 57. EMI/RFI Filter for the Class D Amplifier
Table 19. 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
MICR_HPL
The right MIC input is added to the left headphone output. Setting this bit enables MIC BIAS.
3
MICL_HPL
The left MIC input is added to the left headphone output. Setting this bit enables MIC BIAS.
4
AUXR_HPL
The right AUX input is added to the left headphone output.
5
AUXL_HPL
The left AUX input is added to the left headphone output.
Table 20. RIGHT HEADPHONE_OUTPUT (0x12h)
Bits
Field
0
DACR_HPR
The right DAC output is added to the right headphone output.
Description
1
DACL_HPR
The left DAC output is added to the right headphone output.
2
MICR_HPR
The right MIC input is added to the right headphone output. Setting this bit enables the MIC BIAS
output.
3
MICL_HPR
The left MIC input is added to the right headphone output. Setting this bit enables the MIC BIAS
output.
4
AUXR_HPR
The right AUX input is added to the right headphone output.
5
AUXL_HPR
The left AUX input is added to the right headphone output.
HEADPHONE AMPLIFIER FUNCTION
The LM49350 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.
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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 Schematic Diagram.
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 Schematic Diagram.
Table 21. AUX_OUTPUT (0x13h)
Bits
Field
0
DACR_AUX
The right DAC output is added to the AUX output.
Description
1
DACL_AUX
The left DAC output is added to the AUX output.
2
MICR_AUX
The right MIC input is added to the AUX output. Setting this bit enables the MIC BIAS output.
3
MICL_AUX
The left MIC input is added to the AUX output. Setting this bit enables the MIC BIAS output.
4
AUXR_AUX
The right AUX input is added to the AUX output.
5
AUXL_AUX
The left AUX input is added to the AUX output.
AUXILIARY OUTPUT AMPLIFIER
The LM49350’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 22. OUTPUT_OPTIONS (0x14h)
42
Bits
Field
0
EPMODE
1
HP_NEG_6dB
If set, both HPL and HPR are attenuated by 6dB. This is useful when adding stereo signals
that need more headroom due to being highly correlated.
2
LS_NEG_6dB
If set the class D output is attenuated by 6dB. This is useful when adding stereo signals that
need more headroom due to being highly correlated.
3
AUX_NEG_6dB
4
CP_FORCE
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Description
If set the HPR output is driven with the negative input of the HPL output stage.
If set the AUX output is attenuated by 6dB. This is useful when adding stereo signals that
need more headroom due to being highly correlated.
If set, a -LS_VDD rail will be created on HP_VSS, even if the HP output stage is not required.
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Table 23. ADC_INPUT (0x15h)
Bits
Field
0
DACR_ADCR
The right DAC output is added to the ADC right input.
Description
1
DACL_ADCL
The left DAC output is added to the ADC left input.
2
MICR_ADCR
The right MIC input is added to the ADC right input. Setting this bit enables MIC BIAS.
3
MICL_ADCL
The left MIC input is added to the ADC left input. Setting this bit enables MIC BIAS.
4
AUXR_ADCR
The right AUX input is added to the ADC right input.
5
AUXL_ADCL
The left AUX input is added to the ADC left input.
Table 24. MIC_L_INPUT (0x16h)
Bits
Field
3:0
MIC_L_LEVEL
Description
This sets the gain of the left microphone preamp.
MIC_L_LEVEL
Gain
0000
6dB
0001
8dB
0010
10dB
0011
12dB
0100
14dB
0101
16dB
0110
18dB
0111
20dB
1000
22dB
1001
24dB
1010
26dB
1011
28dB
1100
30dB
1101
32dB
1110
34dB
1111
36dB
4
SE_DIFF
If set, the MIC_L negative input is ignored.
5
MUTE
If set, the left microphone preamp is muted.
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Table 25. MIC_R_INPUT (0x17h)
44
Bits
Field
3:0
MIC_R_LEVEL
4
SE_DIFF
5
MUTE
Description
This sets the gain of the right microphone preamp.
MIC_R_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_R negative input is ignored.
If set, the right microphone preamp is muted.
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Table 26. AUX_L_INPUT (0x18h)
Bits
Field
5:0
AUX_L_LEVEL
6
FROM_LINE_L
Description
This programs the left AUX input level. All gain changes are performed at zero crossings.
AUX_L_LEVEL
Level
AUX_L_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
12dB
001001
–33dB
101001
12dB
001010
–31.5dB
101010
12dB
001011
–30dB
101011
12dB
001100
–28.5dB
101100
12dB
001101
–27dB
101101
12dB
001110
–25.5dB
101110
12dB
001111
–24dB
101111
12dB
010000
–22.5dB
110000
12dB
010001
–21dB
110001
12dB
010010
–19.5dB
110010
12dB
010011
–18dB
110011
12dB
010100
–16.5dB
110100
12dB
010101
–15dB
110101
12dB
010110
–13.5dB
110110
12dB
010111
–12dB
110111
12dB
011000
–10.5dB
111000
12dB
011000
–9dB
111001
12dB
011001
–7.5dB
111010
12dB
011010
–6dB
111011
12dB
011100
–4.5dB
111100
12dB
011101
–3dB
111101
12dB
011110
–1.5dB
111110
12dB
011111
0dB
111111
12dB
If set, the LEFT_MIC/LINE differential input is routed to the AUX_L input amplifier for line level volume control.
This bit overrides the DIFF_MODE (bit 7 of 0x19h) setting.
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Table 27. AUX_R_INPUT (0x19h)
Bits
5:0
6
7
46
Field
Description
AUX_R_LEVEL This programs the right AUX input level. All gain changes are performed at zero crossings.
AUX_R_LEVEL
Level
AUX_R_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
12dB
001001
–33dB
101001
12dB
001010
–31.5dB
101010
12dB
001011
–30dB
101011
12dB
001100
–28.5dB
101100
12dB
001101
–27dB
101101
12dB
001110
–25.5dB
101110
12dB
001111
–24dB
101111
12dB
010000
–22.5dB
110000
12dB
010001
–21dB
110001
12dB
010010
–19.5dB
110010
12dB
010011
–18dB
110011
12dB
010100
–16.5dB
110100
12dB
010101
–15dB
110101
12dB
010110
–13.5dB
110110
12dB
010111
–12dB
110111
12dB
011000
–10.5dB
111000
12dB
011000
–9dB
111001
12dB
011001
–7.5dB
111010
12dB
011010
–6dB
111011
12dB
011100
–4.5dB
111100
12dB
011101
–3dB
111101
12dB
011110
–1.5dB
111110
12dB
011111
0dB
111111
12dB
FROM_LINE_R If set, the RIGHT_MIC/LINE differential input is routed to the AUX_R input amplifier for line level volume
control. This bit overrides the DIFF_MODE (bit 7) setting.
DIFF_MODE
If set, the stereo single-ended inputs AUX_L and AUX_R convert to a mono differential input pair MONO_IN+
and MONO_IN-.
(MONO_IN+) - (MONO_IN-) is routed to the AUX_L input amplifier.
(MONO_IN-) - (MONO_IN+) is routed to the AUX_R input amplifier.
(unless overriden by the respective FROM_LINE bits).
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ADC Control Registers
This register is used to control the LM49350's ADC:
Table 28. ADC Basic (0x20h)
Bits
Field
0
MONO
Description
This sets mono or stereo operation of the ADC.
MONO
1
2
OSR
ADC Operation
0
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
64
128
MUTE_L
If set, a digital mute is applied to the Left (or mono) ADC output.
3
MUTE_R
If set, a digital mute is applied to the Right ADC output.
6.4
ADC_CLK_SEL
This selects the source of the ADC clock domain, ADC_SOURCE_CLK.
ADC_CLK_SEL
7
ADC_DSP_ONLY
Source
000
MCLK
001
PORT1_RX_CLK
010
PORT2_RX_CLK
011
PLL1_OUTPUT2
100
PLL2_OUTPUT
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 asyncronous 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.
Table 29. 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
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Table 30. ADC TRIM (0x22h)
Bits
Field
Description
7:0
ADC_TRIM
If set, the ADC is compensated with recommended compensation filter coefficients. The recommended
ADC compensation filter coefficients are programmed as follows:
Register 0xF8h set to 0x00h
Register 0xF9h set to 0x01h
Register 0xFAh set to 0x96h
Register 0xFBh set to 0xFBh
Register 0xFCh set to 0x30h
Register 0xFDh set to 0x62h
DAC Control Registers
This register is used to control the LM49350's DAC:
Table 31. 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
11
32
2
MUTE_L
This digitally mutes the Left DAC output.
3
MUTE_R
This digitally mutes the Right DAC output.
6:4
DAC_CLK_SEL
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
PLL1_OUTPUT1
100
PLL2_OUTPUT
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 32. DAC_CLK_DIV (0x31h)
48
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 0x00.
Program this divider with the division you want, multiplied by 2, and subtract 1.
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DAC_CLK_DIV
Divides by
00000000
1
00000001
1
00000010
1.5
00000011
2
—
—
11111101
127
11111110
127.5
11111111
128
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Digital Mixer Control Registers
DIGITAL MIXER
The LM49350’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 without having to enable the analog of the DAC and ADC inorder 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
LM49350 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.
Figure 58. Digital Mixer
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The LM49350 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 LM49350's digital mixer:
Table 33. Input Levels 1 (0x40h)
Bits
Field
1:0
PORT1_RX_L
_LVL
3:2
5:4
7:6
50
PORT1_RX_R
_LVL
PORT2_RX_L
_LVL
PORT2_RX_R
_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
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.
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PORT2_RX_R_LVL
Level
00
0dB
01
–6dB
10
–12dB
11
–18dB
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Table 34. 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
Level
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
Level
00
0dB
01
–6dB
10
–12dB
11
–18dB
Table 35. Audio Port 1 Input (0x42h)
Bits
Field
1:0
L_SEL
3:2
R_SEL
Description
This selects which input is fed to the Left TX Channel of Audio Port 1.
L_SEL
Selected Input
00
None
01
ADC_L
10
PORT2_RX_L
11
DAC_INTERP_L
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. The swap bit can be used to control which
microphone is being used for audio port transmit. For example, if LEFT_MIC is used as a primary handset
microphone and RIGHT_MIC is used a headset microphone, the SWAP bit allows the audio port to select
one of the microphones at a time for audio port transmit via the ADC.
5
MONO
If set, the right channel is ignored and the left channel becomes (left+right)/2.
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Table 36. Audio Port 2 Input (0x43h)
Bits
Field
1:0
L_SEL
Description
This selects which input is fed to Audio Port 2's Left TX Channel.
L_SEL
3:2
R_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. The swap bit can be used to control which
microphone is being used for audio port transmit. For example, if LEFT_MIC is used as a primary handset
microphone and RIGHT_MIC is used a headset microphone, the SWAP bit allows the audio port to select
one of the microphones at a time for audio port transmit via the ADC.
5
MONO
If set, the right channel is ignored and the left channel becomes (left+right)/2.
Table 37. 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
Table 38. Decimator Input Select (0x45h)
Bits
Field
1:0
L_SEL
3:2
52
R_SEL
Description
This selects which input is fed to the left ADC's decimator input.
L_SEL
Selected Input
00
None
01
PORT1_RX_L
10
PORT2_RX_L
11
DAC_INTERP_L
This selects which input is fed to the right ADC's decimator input.
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R_SEL
Selected Input
00
None
01
PORT1_RX_R
10
PORT2_RX_R
11
DAC_INTERP_R
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Table 38. Decimator Input Select (0x45h) (continued)
Bits
Field
Description
5:4
MXR_CLK_SEL
This selects sets the source of the Digital Mixer Clock. The 'Auto' setting will automatically select the source
with the highest clock frequency. Whenever the DAC interpolator (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
2
1
0
2
1
Right Word
Figure 59. I2S Serial Data Format (24 bit example)
I2S_CLK
I2S_SYNC
I2S_SDO/
I2S_SDI
23
22
21
20
2
1
0
23
22
21
20
Left Word
23
Right Word
Figure 60. Left Justified Data Format (24 bit example)
I2S_CLK
I2S_SYNC
I2S_SDO/
I2S_SDI
0
23
22
21
3
Left Word
2
1
0
23
22
21
3
0
Right Word
Figure 61. Right Justified Data Format (24 bit example)
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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
11
10
9
Short frame sync mode
Long frame sync mode
Figure 62. PCM Serial Data Format (16 bit example)
The following registers are used to control the LM49350'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 39. 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
7
54
STEREO_SYNC_PHASE
SYNC_INVERT
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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.
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
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Table 40. 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 input of
V
this divider should be around 12MHz. The default of this divider is 0x00, i.e. bypassed.
Program this divider with the division you want, 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 41. 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
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 42. 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
56
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.
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SYNC_WIDTH
Width of SYNC (in bits)
000
1
001
2
010
4
011
7
100
8
101
11
110
15
111
16
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Table 43. DATA_WIDTHS (0x54h/64h)
Bits
Field
2:0
RX_WIDTH
5:3
7:6
TX_WIDTH
TX_EXTRA_BITS
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
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
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Table 44. RX_MODE (0x55h/65h)
Bits
Field
0
RX_MODE
5:1
MSB_POSITION
Description
This sets the RX data input justification with respect to the SYNC signal.
RX_MODE
Description
0
MSB Justified
1
LSB Justified
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
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Compand Mode
0
μLaw
1
A-Law
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Table 45. TX_MODE (0x56h/x66h)
Bits
Field
0
TX_MODE
5:1
MSB_POSITION
Description
This sets the TX data output justification with respect to the SYNC signal.
TX_MODE
Description
0
MSB Justified
1
LSB Justified
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
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Digital Effects Engine
DIGITAL SIGNAL PROCESSOR (DSP)
The LM49350 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 LM49350
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. The data paths on each DSP engine are 24 bits
wide for ultimate flexibility. Both DSP engines feature digital volume control, automatic level control (ALC), digital
soft clip compression, and a 5-band parametric EQ. The ADC DSP engine adds a dedicated high-pass filter to
reduce wind noise or pop noise during uplink. The DAC DSP engine adds a digital 3D algorithm that allows for
stereo widening of the original audio signal. The effects chain of each DSP engine is shown by the diagrams
below.
Figure 63. ADC DSP Effects Chain
Figure 64. 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 3D stereo widening 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 46. ADC EFFECTS (0x70h)
60
Bits
Field
Description
0
ADC_HPF_ENB
This enables the ADC's High Pass Filter.
1
ADC_ALC_ENB
This enables the ADC's Auto 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.
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Table 47. DAC EFFECTS (0x71h)
Bits
Field
0
DAC_ALC_ENB
Description
This enables the DAC's Auto Level Control.
1
DAC_PK_ENB
This enables the DAC's Peak Detector.
2
DAC_EQ_ENB
This enables the DAC's 5-band Parametric EQ.
3
DAC_3D_ENB
This enables the DAC's Stereo Widening Circuit.
4
ADC_SCLP_ENB
This enables the DAC's Soft Clip Feature.
Table 48. HPF MODE (0x80h)
Bits
Field
2:0
HPF_MODE
Description
This configures the ADC's High Pass Filter. To calculate the –3dB cutoff frequency, multiply the
coefficient by the sample rate (Hz): fC = XN.fS(Hz)
HPF_MODE
Coefficient
Filter Characteristics
fC = 220Hz for:
000
X0 = 0.0275
8kHz Voice
001
X1 = 0.01833
12kHz Voice
010
X2 = 0.01375
16kHz Voice
011
X3 = 0.009166
24kHz Voice
100
X4 = 0.006875
32kHz Voice
101
X5 = 0.003125
32kHz Audio
110
X6 = 0.0020833
48kHz Audio
111
X7 = 0.0015625
fC = 100Hz for:
fC =150Hz for:
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 LM49350. 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 auxiliary input amplifier gain or microphone preamplifier
gain. The dual ALCs can be used to regulate the level of the analog (Stereo Auxiliary, mono differential, Stereo
MIC/LINE) and digital (Port1 Data In, Port2 Data In) audio inputs. The ALC regulated output can be routed to any
of the LM49350’s amplifier outputs for playback. The ALC regulated output can also be routed to Audio Port1 or
Audio Port2 for digital data transmission via I2S or PCM.
Only audio inputs that are considered signals (rather than noise) are sent to the ALC’s peak detector block. The
peak detector compares the level of the audio input versus the ALC target level (TARGET_LEVEL). Signals
lower than the target level will be amplified and signals higher than the target level will be attenuated. Any audio
input that is lower than the level specified by the noise floor level (NOISE_FLOOR) will be considered as noise
and will be gated from the ALC’s peak detector in order to avoid noise pumping. So it is important to set
NOISE_FLOOR to correlate with the signal to noise ratio of the corresponding audio path. In some instances (ie.
Conference calls), it may be desirable to mute audio input signals that consist solely of background noise from
the audio output. This is accomplished by enabling the ALC’s noise gate (NG_ENB). When the noise gate is
enabled, signals lower than the noise floor level will be muted from the audio output.
If the audio input signal is below the target level, the ALC will increase the gain of the corresponding volume
control until the signal reaches the target level. The rate at which the ALC performs gain increases is known as
decay rate (DECAY RATE). But before each ALC gain increase the ALC must wait a predetermined amount of
time (HOLD TIME). If the audio input signal is above the target level, the ALC will decrease the gain of the
corresponding volume control until the signal reaches the target level. The rate at which the ALC performs
attenuation is known as attack rate (ATTACK RATE). The ALC’s peak detector tracks increases in audio input
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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 ALC SAMPLE RATE
setting (for DAC ALC) or the ALC MODE setting (for ADC ALC).
(1) Decay hold time, (2) Slow Decay, (3) Quick Attack
(2)
(3)
(1)
target level
peak detection
and ADC output
attack
decay
12 dB
microphone gain
12 dB
14 dB
signal below target
10 dB
signal above target
Figure 65. ALC Example
Table 49. ADC_ALC_1 (0x81h)
Bits
Field
2:0
SAMPLE_RATE
Description
This programs the timers on the ALC with the closest sample rate of the ADC.
SAMPLE_RATE
62
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_SEL
If SOURCE_OVR is set then this manually overrides the selection of the input amplifier that is used to
alter the gain for ALC operation.
0 = Both ALCs control AUX gain
1 = Both ALCs control MIC gain
6
SOURCE_OVR
If set, the output of the ALC is not set automatically but is controlled by the SOURCE_SEL bit. If
cleared each ALC controls the input gain of the amplifier (AUX or MIC) that is set to that ADC channel
(or MIC if both are selected).
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Table 50. ADC_ALC_2 (0x82h)
Bits
Field
Description
3:0
NOISE_FLOOR
This sets the anticipated noise floor. Signals lower than the noise floor specified will be gated from the
ALC to avoid noise pumping.
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 51. 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.
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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 52. 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
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 53. 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
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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
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
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 54. 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
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 55. 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. If the volume
control is less than 6 bits the relevant LSBs are used as the limit and the MSBs are ignored.
Table 56. ADC_ALC_8 (0x88h)
Bits
Field
Description
5:0
MIN_LEVEL
This sets the minimum allowed gain of the volume control to the output amplifier. 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 57. ADC_L_LEVEL (0x89h) (Default data value is 0x33h)
68
Bits
Field
5:0
ADC_L_LEVEL
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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
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Table 58. ADC_R_LEVEL (0x8Ah) (Default data value is 0x33h)
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
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Table 59. 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.
6:2
70
LEVEL
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FREQ
Frequency (Hz)
00
60
01
80
10
100
11
120
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 60. EQ_BAND_2 (0x8Ch)
Bits
Field
1:0
FREQ
6:2
7
LEVEL
Q
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
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 61. 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
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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 62. 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 63. 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
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FREQ
Frequency (Hz)
00
7k
01
9k
10
11k
11
20k
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 64. SOFTCLIP1 (0x90h)
Bits
Field
3:0
THRESHOLD
4
SOFT_KNEE
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
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 65. SOFTCLIP2 (0x91h)
76
Bits
Field
4:0
RATIO
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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 66. SOFTCLIP3 (0x92h)
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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DAC Effects Registers
Table 67. DAC_ALC_1 (0xA0h)
Bits
Field
2:0
SAMPLE_ RATE
Description
This programs the timers on the ALC with the closest DAC sample rate.
SAMPLE_ RATE
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 68. DAC_ALC_2 (0xA1h)
Bits
Field
3:0
NOISE_FLOOR
4
78
NG_ENB
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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 69. DAC_ALC_3 (0xA2h)
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 70. DAC_ALC_4 (0xA3h)
80
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.
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ATTACK_RATE
Time between gain steps(us)
00000
21
00001
42
00010
83
00011
167
00100
250
00101
333
00110
417
00111
542
01000
729
01001
958
01010
1250
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 71. 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
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
Time
000
1.3ms
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 72. 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
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 73. 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 74. DAC_ALC_8 (0xA7h)
82
Bits
Field
5:0
MIN_LEVEL
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Description
This sets the minimum allowed gain to the digital level control when the ALC is used.
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Table 75. DAC_L_LEVEL (0xA8h) (Default data value is 0x33h)
Bits
Field
5:0
DAC_L_LEVEL
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
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Table 76. DAC_R_LEVEL (0xA9h) (Default data value is 0x33h)
84
Bits
Field
5:0
DAC_R_LEVEL
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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
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Table 77. DAC_3D (0xAAh)
Bits
Field
0
EFFECT_MODE
Description
This sets the digital 3D stereo enhancement mode.
EFFECT_MODE
2:1
EFFECT_LEVEL
Loudspeaker
1
Headphone
This sets the applied level of 3D effect.
EFFECT_LEVEL
6:3
7
FILTER_TYPE
ATTENUATE
Type
0
Level
00
25%
01
37.50%
10
50%
11
75%
This sets the 3D effect filter response.
FILTER_TYPE
Response
0000
200Hz HPF
0001
300Hz HPF
0010
600Hz HPF
0011
900Hz HPF
0100
200Hz-500Hz BPF
0101
200Hz-1kHz BPF
0110
200Hz-1.6kHz BPF
0111
200Hz-2.5kHz BPF
1000
300Hz-1kHz BPF
1001
300Hz-1.6kHz BPF
1010
300Hz-2.5kHz BPF
1011
600Hz-1kHz BPF
1100
600Hz-1.6kHz BPF
1101
600Hz-2.5kHz BPF
1110
900Hz-1.6kHz BPF
1111
900Hz-2.5kHz BPF
If set, the inputs are reduced by 6dB before 3D effects are applied in order to avoid
clipping.
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Table 78. EQ_BAND_1 (0xABh)
Bits
Field
1:0
FREQ
6:2
86
LEVEL
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Description
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
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. EQ_BAND_2 (0xACh)
Bits
Field
1:0
FREQ
6:2
7
LEVEL
Q
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
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 80. EQ_BAND_3 (0xADh)
Bits
Field
1:0
FREQ
6:2
7
88
LEVEL
Q
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Description
This sets the Mid peak filter's center frequency. The cut-off frequencies shown are based
on a 48kHz sample rate. Using lower sample rates will scale down the cut-off frequencies
proportionately.
FREQ
Frequency (Hz)
00
600
01
800
10
1k
11
1.2k
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 81. 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 82. 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
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FREQ
Frequency (Hz)
00
7k
01
9k
10
11k
11
20k
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 83. SOFTCLIP1 (0xB0h)
Bits
Field
3:0
TRESHOLD
4
SOFT_KNEE
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
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 84. SOFTCLIP2 (0xB1h)
92
Bits
Field
4:0
RATIO
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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 85. SOFTCLIP3 (0xB2h)
Table 40:
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 86. GPIO (0xE0h)
Bits
Field
3:0
GPIO_MODE
4
Description
This sets the mode of the GPIO pin.
GPIO_MODE
GPIO Function
0000
OFF (input disabled)
0001
GPIO_RX
0010
GPIO_TX
0011
HP_ENB (out)
0100
HP_ENB (out)
0101
LS_ENB (out)
0110
LS_ENB (out)
0111
SHORT_CCT or THERMAL (out)
1000
SHORT_CCT or THERMAL or CLIP (out)
1001
CLIP (out)
1010
ADC_NG_ACTIVE (out)
1011
ADC_NG_ACTIVE (out)
1100
MIC_MUTE (in)
1101
MIC_MUTE (in)
1110
CHIP_ENB (in)
1111
CHIP_ENB (in)
GPIO_TX
If set, the GPIO pin will transmit a logic high whenever GPIO_MODE is set to '0010'.
5
GPIO_RX
This bit reports what logic level is present on the GPIO pin.
6
SHORT_CCT
7
THERMAL_EVENT
If set, the GPIO records that a short circuit event has occurred on the class D outputs.
If set records that a temperature event has occurred on the die.
Clear on Write (1).
Table 87. DEBUG1 (0xF0h)
Bits
Field
1:0
DAC_DITHER
_LVL
3:2
94
DAC_DITHER
_MODE
4
Not Used
5
SOFT_RESET
7:6
RSVD
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Description
This sets the amount of DAC dither. Lower levels of the dither may improve the noise floor of
the DAC.
DAC_DITHER
_LVL
Level
00
Very Small
01
Small
10
Medium (Default)
11
Large
This sets the DAC dither mode.
DAC_DITHER
_MODE
Mode
00
AUTOMATIC
01
ON
10
OFF
If set, the LM49350 enters RESET mode. To bring the LM49350 back out of RESET mode,
then set this bit back to zero.
Reserved
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Table 88. Spread Spectrum (0xF1h)
Bits
Field
1:0
RSVD
Description
2
SS_DISABLE
Reserved
If this bit is set, Spread Spectrum mode will be disabled from the Class D amplifier.
Table 89. ADC Compensation Filter C0 LSBs (0xF8h)
Bits
Field
7:0
ADC_CO_LSB
Description
Bits
Field
7:0
ADC_CO_MSB
Bits 7:0 of C0[15:0]
Table 90. ADC Compensation Filter C0 MSBs (0xF9h)
Description
Bits 15:0 of C0[15:0]
Table 91. ADC Compensation Filter C1 LSBs (0xFAh)
Bits
Field
7:0
ADC_C1_LSB
Description
Bits 7:0 of C1[15:0]
Table 92. ADC Compensation Filter C1 MSBs (0xFBh)
Bits
Field
7:0
ADC_C1_MSB
Description
Bits 15:0 of C1[15:0]
Table 93. ADC Compensation Filter C2 LSBs (0xFCh)
Bits
Field
Description
7:0
ADC_C2_LSB
Bits 7:0 of C2[15:0]
Table 94. ADC Compensation Filter C2 MSBs (0xFDh)
Bits
Field
7:0
ADC_C2_MSB
Bits
Field
Description
Bits 15:0 of C2[15:0]
Table 95. AUX_LINEOUT (0xFE)
4:0
RSVD
5
AUX_LINE_OUT
Description
Reserved
If set, the earpiece amplifier operates in a low current drive mode for line out applications in
order to reduce power consumption.
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Schematic Diagram
Figure 66. Demo Board Schematic
96
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Demonstration Board Layout
Figure 67. Top Silkscreen Layer
Figure 68. Top Layer
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Figure 69. Inner Layer 1
Figure 70. Inner Layer 2
98
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Figure 71. Bottom Silkscreen Layer
Figure 72. Bottom Layer
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APPLICATION NOTE FOR LM49350
POWER CONNECTIONS
Recommended target application circuit must provide same voltage level for A_VDD and LSVDD to get
performance on Electrical Specifications on LM49350 datasheet.
VDD
D_VDD
I/O_VDD A_VDD
LS_VDD
LEFT_MIC+
HP_VSS
LEFT_MICVREF
0.5 - 50
MHz
CP+
MIC_BIAS
CP-
RIGHT_MIC+
RIGHT_MIC-
MCLK
2
I C
Baseband
Controller
LM49350
2
I S / PCM
(PORT1)
GPIO
HPL
HPR
LS+
LS8Ö
AUX_OUT+
2
Bluetooth
Transceiver
I S / PCM
(PORT2)
32Ö
AUX_OUT-
DGND LSGND AGND AUX_L AUX_R
Synthesized
FM Radio / TV Tuner
Figure 73. Recommended Power Connection
100
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MICROPHONE BIAS CONFIGURATIONS
Schematic Considerations for MEMs Microphones
The internal microphone bias of the LM49350 is provided through a two stage amplifier. Adding a capacitor larger
than 100pF directly to this pin can cause instability. In many cases, when using MEMs microphones, a larger
bypass capacitor is required on the MIC_BIAS pin. To avoid oscillations and to keep the device stable, it is
recommended to add a resistor (RB) greater than 10Ω in series with the capacitor (CB). Another option is to bias
the MEMs microphone from the 1.8V supply used for D_VDD/IO_VDD.
VDD
LS_VDD
LEFT_MIC+
LEFT_MICRB
MIC_BIAS
RIGHT_MIC+
CB
RIGHT_MIC-
Figure 74. Schematic for MEMs Microphones
Schematic Considerations for ECM Microphones
When using ECM microphones, refer to the configurations shown in Figure 73 or Figure 74 to bias the
microphones. In many cases, an RC filter is required to provide a more stable microphone bias (see Figure 74).
In this case, a 10Ω resistor (RB ) in series with CB is recommended.
VDD
VDD
A_VDD
A_VDD
LS_VDD
LS_VDD
LEFT_MIC+
LEFT_MIC-
LEFT_MIC-
LEFT_MIC+
MIC_BIAS
RIGHT_MIC+
MIC_BIAS
RB
RIGHT_MIC-
CB
RIGHT_MIC+
RIGHT_MIC-
Figure 75. Schematic Option for ECM Microphones
Figure 76. Schematic Option for ECM Microphones
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PCB LAYOUT CONSIDERATIONS
Microphone Inputs
When routing the differential microphone inputs the electrical length of the two traces should be well matched.
The differential input pair can be routed in parallel on the same plane or the traces can overlap on two adjacent
planes. It is important to surround these traces with a ground plane or trace to isolate the microphone inputs from
the noise coupling from the class D amplifier.
Class D Loudspeaker
To minimize trace resistance and therefore maintain the highest possible output power, the power (LS_VDD) and
class D output (LS-, LS+) traces should be as wide as possible. It is also essential to keep these same traces as
short and well shielded as possible to decrease the amount of EMI radiation.
Capacitors
All supply bypass capacitors (for A_VDD, D_VDD. I/O VDD, and LS_VDD), and charge pump capacitors should be
as close to the device as possible. Careful consideration should be taken with the ground connection of the
analog supply (A_VDD) bypass cap, for proper performance it should be referenced to a low noise ground plane.
The charge pump capacitors and traces connecting the capacitor to the device should be kept away from the
input and output traces to avoid noise coupling issues.
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REVISION HISTORY
Rev
Date
1.0
09/03/08
Description
Initial released.
1.01
09/04/08
Text edits.
1.02
09/22/08
Text edits.
1.03
10/24/08
Text edits.
1.04
12/15/08
Text edits and replaced the top silkscreen layer.
1.05
05/27/09
Added the EMI/RFI section and the corresponding graphic.
1.06
05/29/09
Text edits.
1.07
04/09/10
Text edits.
1.08
04/15/10
Text edits.
1.09
09/17/10
Added the Application section required for Leadcore (chipset partner).
1.10
03/23/11
Input minor text edits.
1.11
04/05/11
Added sections 29.2 and 29.3 including their corresponding graphics, then generated a
CONFIDENTIAL version for LEADCORE.
1.12
04/12/11
Edited Figure 32 and input text edits.
1.13
04/13/11
Input text edits.
1.14
08/24/11
Added table: RX_MODE (0x55h/65h).
1.15
03/16/12
Added the one more Timing Char table (DVDD = I/OVDD = 1.8V with the 2 diagrams (Timing
I2S Master and Timing for I2S Slave).
1.16
06/29/12
Edited Figures 2, 3, 4, and 5 (Typical Application circuit diagrams).
Copyright © 2008–2012, Texas Instruments Incorporated
Product Folder Links: LM49350 LM49350RLEVAL
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103
PACKAGE OPTION ADDENDUM
www.ti.com
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)
LM49350RL/NOPB
ACTIVE
DSBGA
YPG
36
250
Green (RoHS
& no Sb/Br)
SNAG
Level-1-260C-UNLIM
-40 to 85
GJ8
LM49350RLX/NOPB
ACTIVE
DSBGA
YPG
36
1000
Green (RoHS
& no Sb/Br)
SNAG
Level-1-260C-UNLIM
-40 to 85
GJ8
(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
www.ti.com
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)
LM49350RL/NOPB
DSBGA
YPG
36
250
178.0
12.4
LM49350RLX/NOPB
DSBGA
YPG
36
1000
178.0
12.4
Pack Materials-Page 1
B0
(mm)
K0
(mm)
P1
(mm)
W
Pin1
(mm) Quadrant
3.63
3.63
0.76
8.0
12.0
Q1
3.63
3.63
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)
LM49350RL/NOPB
DSBGA
YPG
LM49350RLX/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 = 3.49 mm, Min = 3.43 mm
E: Max = 3.49 mm, Min = 3.43 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.
www.ti.com
12/12
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