TEXAS INSTRUMENTS TLV320DAC3120 Technical data

TLV320DAC3120
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Low-Power Mono Audio DAC With Embedded miniDSP
and Mono Class-D Speaker Amplifier
Check for Samples: TLV320DAC3120

1 INTRODUCTION

1.1 Features 1.2 Applications

• Mono Audio DAC With 95-dB SNR
• Instruction-Programmable Embedded miniDSP
• Supports 8-kHz to 192-kHz Sample Rates
• Mono Class-D BTL Speaker Driver (2.5 W Into 4 or 1.6 W Into 8 )
• Mono Headphone/Lineout Driver
• Two Single-Ended Inputs With Output Mixing and Level Control
• Microphone Bias
• Built-in Digital Audio Processing Blocks With User-Programmable Biquad, FIR Filters, and DRC
• Programmable Digital Audio Processor for Bass Boost/Treble/EQ With up to Six Biquads for Playback
• Pin Control or Register Control for Digital Playback Volume-Control Settings
• Integrated PLL Used for Programmable Digital Audio Processor
• I2S, Left-Justified, Right-Justified, DSP, and TDM Audio Interfaces
• I2C Control With Register Auto-Increment
• Full Power-Down Control
• Power Supplies: – Analog: 2.7 V–3.6 V – Digital Core: 1.65 V–1.95 V – Digital I/O: 1.1 V–3.6 V – Class-D: 2.7 V–5.5 V (SPKVDD AVDD)
• 5-mm × 5-mm 32-QFN Package
Portable Audio Devices
eBook
Portable Navigation Devices

1.3 Description

The TLV320DAC3120 is a low-power, highly integrated, high-performance mono DAC with 24-bit mono playback.
The device integrates several analog features, such as a microphone bias, headphone drivers, and a mono speaker driver capable of driving a 4-Ω load. The TLV320DAC3120 has a fully programmable miniDSP for digital audio processing. The digital audio data format is programmable to work with popular audio standard protocols (I2S, left/right-justified) in master, slave, DSP, and TDM modes. Bass boost, treble, or EQ can be supported by the programmable digital-signal processing block. An on-chip PLL provides the high-speed clock needed by the digital signal-processing block. The volume level can be controlled by either a pin control or by register control. The audio functions are controlled using the I2C serial bus.
The TLV320DAC3120 is available in a 32-pin QFN package.
SLAS659–NOVEMBER 2009
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.
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 testingof all parameters.
Copyright © 2009, Texas Instruments Incorporated
Digital
Audio
Processing
and
Serial
Interface
SDIN
BCLK
WCLK
MCLK PLL
HPVDDHPVSS
SPKVDDSPKVSS
AVDDAVSS
SPKVSS
SPKVDD
VOL/MICDET
SCL
SDA
GPIOGPIO1
DAC
MIXER P1/R35
RESET
DVDDDVSS
IOVDD
IOVSS
AIN2
AIN1
2 V/2.5 V/AVDD
MICBIAS
Note: Normally, MCLK is PLL input; however, BCLK or GPIO1 can also be PLL input.
Audio Output Stage Power Management
De-Pop
and
Soft Start
RC CLK
P1/R33–R34
P1/R46
I C
2
Mono DAC
SPKP SPKM
Class-D Speaker
Driver
6 dB to 24 dB
(6-dB steps)
Analog
Attenuation
0 dB to –78 dB
and Mute
(0.5-dB steps)
P1/R42
P1/R38
SPKP
SPKM
Class A/B
Headphone/Lineout
Driver
0 dB to 9 dB (1-dB steps)
Analog
Attenuation
HPOUT
P1/R36
P1/R40
P1/R30–R31
L Data R Data (L+R)/2 Data
P0/R63
7-Bit ADC P0/R116
Left and Right
Volume-Control Register
P0/R117
Digital Vol
24 dB to
Mute
miniDSP
P0/R64
S
0 dB to –78 dB
and Mute
(0.5-dB steps)
B0360-01
TLV320DAC3120
SLAS659–NOVEMBER 2009
This integrated circuit can be damaged by ESD. Texas Instruments recommends that all integrated circuits be handled with appropriate precautions. Failure to observe proper handling and installation procedures can cause damage.
ESD damage can range from subtle performance degradation to complete device failure. Precision integrated circuits may be more susceptible to damage because very small parametric changes could cause the device not to meet its published specifications.
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This data manual is designed using PDF document-viewing features that allow quick access to information. For example, performing a global search on "page 0 / register 27" produces all references to this page and register in a list. This makes is easy to traverse the list and find all information related to a page and register. Note that the search string must be of the indicated format. Also, this document includes document hyperlinks to allow the user to quickly find a document reference. To come back to the original page, click the green left arrow near the PDF page number at the bottom of the file. The hot-key for this function is alt-left arrow on the keyboard. Another way to find information quickly is to use the PDF bookmarks.
Figure 1-1. Functional Block Diagram
NOTE
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P0048-12
AVSSSPKVSS
IOVSS
SPKVDD
24123222321420519618717
8
16
25
15
26
14
27
13
28
12
29
1130
1031
932
NC
SPKM
AIN2
SPKP
AIN1
SPKVDD
MICBIAS
SPKVSS
VOL/MICDET
SPKM
SCL
DVSS
SDA
AVDD
RHB Package
(Top View)
SPKP
IOVDD
HPOUT
DVDD
HPVDD
NC
HPVSS
DIN
NC
WCLK
RESET
BCLK
GPIO1
MCLK
TLV320DAC3120
TLV320DAC3120
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2 PACKAGE AND SIGNAL DESCRIPTIONS

2.1 Package/Ordering Information

PRODUCT PACKAGE TEMPERATURE ORDERING NUMBER QUANTITY
PACKAGE
DESIGNATOR
TLV320DAC3120 QFN-32 RHB –40°C to 85°C

2.2 Device Information

OPERATING TRANSPORT MEDIA,
RANGE
TLV320DAC3120IRHBT Tape and reel, 250 TLV320DAC3120IRHBR Tape and reel, 3000
Table 2-1. TERMINAL FUNCTIONS
TERMINAL
NAME NO.
AIN1 13 I Analog input #1 routed to output mixer AIN2 14 I Analog input #2 routed to output mixer AVDD 17 Analog power supply AVSS 16 Analog ground BCLK 7 I/O Audio serial bit clock DIN 5 I Audio serial data input DVDD 3 Digital power – digital core DVSS 18 Digital ground GPIO1 32 I/O General-purpose input/output and multifunction pin HPOUT 27 O Headphone/lineout driver output HPVDD 28 Headphone/line driver and PLL power HPVSS 29 Headphone/line driver and PLL ground IOVDD 2 Interface power IOVSS 1 Interface ground MCLK 8 I Exterrnal master clock
Copyright © 2009, Texas Instruments Incorporated PACKAGE AND SIGNAL DESCRIPTIONS 3
I/O DESCRIPTION
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TERMINAL FUNCTIONS (continued)
TERMINAL
NAME NO.
MICBIAS 12 O Micophone bias voltage NC 4, 15, 30 No connection RESET 31 I Device reset SCL 10 I/O I2C control-bus clock input SDA 9 I/O I2C control-bus data input SPKM 19, 23 O Class-D speaker driver inverting output SPKP 22, 26 O Class-D speaker driver noninverting output SPKVDD 21, 24 Class-D speaker driver power supply SPKVSS 20, 25 Class-D speaker driver power-supply ground VOL/MICDET 11 I Volume control or microphone/headphone/headset detection WCLK 6 I/O Audio serial word clock
I/O DESCRIPTION

3 ELECTRICAL SPECIFICATIONS

3.1 Absolute Maximum Ratings

over operating free-air temperature range (unless otherwise noted)
AVDD to AVSS –0.3 to 3.9 V DVDD to DVSS –0.3 to 2.5 V HPVDD to HPVSS –0.3 to 3.9 V SPKVDD to SPKVSS –0.3 to 6 V IOVDD to IOVSS –0.3 to 3.9 V Digital input voltage IOVSS – 0.3 to IOVDD + 0.3 V Analog input voltage AVSS – 0.3 to AVDD + 0.3 V Operating temperature range –40 to 85 °C Storage temperature range –55 to 150 °C Junction temperature (TJMax) 105 °C
QFN package
(1) Stresses beyond those listed under Absolute Maximum Ratings may cause permanent damage to the device. These are stress ratings
only, and functional operation of the device at these or any other conditions beyond those indicated under Recommended Operating Conditions is not implied. Exposure to absolute-maximum-rated conditions for extended periods may affect device reliability.
Power dissipation (TJMax – TA)/R R
Thermal impedance (with thermal pad soldered to board) 35 °C/W
θJA
(1)
VALUE UNIT
θJA
W
Table 3-1. System Thermal Characteristics
Power Rating at 25°C Derating Factor Power Rating at 70°C Power Rating at 85°C
2.3 W 28.57 mW/°C 1 W 0.6 W
(1) This data was taken using 2-oz. (0.071-mm thick) trace and copper pad that is soldered to a JEDEC high-K, standard 4-layer
3-in. × 3-in. (7.62-cm × 7.62-cm) PCB.
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3.2 Recommended Operating Conditions

over operating free-air temperature range (unless otherwise noted)
MIN NOM MAX UNIT
(1)
AVDD
Referenced to AVSS DVDD Referenced to DVSS HPVDD Power-supply voltage range Referenced to HPVSS SPKVDD
(1)
Referenced to SPKVSS IOVDD Referenced to IOVSS
Speaker impedance Load applied across class-D output pins (BTL) 4 Headphone impedance AC-coupled to R
V
I
MCLK
(3)
Analog audio full-scale input voltage
Mono line output load impedance
AVDD = 3.3 V, single-ended 0.707 V
AC-coupled to R
Master clock frequency IOVDD = 3.3 V 50 MHz SCL SCL clock frequency 400 kHz T
A
Operating free-air temperature –40 85 °C
(1) To minimize battery-current leakage, the SPKVDD and SPKVDD voltage levels should not be below the AVDD voltage level. (2) All grounds on board are tied together, so they should not differ in voltage by more than 0.2 V maximum for any combination of ground
signals. By use of a wide trace or ground plane, ensure a low-impedance connection between HPVSS and DVSS.
(3) The maximum input frequency should be 50 MHz for any digital pin used as a general-purpose clock.
(2) (2)
(2)
(2)
(2)
L
L
2.7 3.3 3.6
1.65 1.8 1.95
2.7 3.3 3.6
2.7 5.5
1.1 3.3 3.6
16
10 k
V
RMS

3.3 Electrical Characteristics

At 25°C, AVDD = HPVDD = IOVDD = 3.3 V, SPKVDD = 3.6 V, DVDD = 1.8 V, fS(audio) = 48 kHz, CODEC_CLKIN = 256 × fS, PLL = Off, VOL/MICDET pin disabled (unless otherwise noted)
PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
INTERNAL OSCILLATOR—RC_CLK
Oscillator frequency 8.2 MHz
VOLUME CONTROL PIN (ADC); VOL/MICDET pin enabled
Input voltage range 0 / register 116, bit D7 = 1 and page 0 / register 67, 0 V
Input capacitance 2 pF Volume control steps 128 Steps
MICROPHONE BIAS
Voltage output V
Voltage regulation mV
VOL/MICDET pin configured as volume control (page bit D7 = 0)
Page 1 / register 46, bits D1–D0 = 10 2.25 2.5 2.75 Page 1 / register 46, bits D1–D0 = 01 2 At 4-mA load current, page 1 / register 46, bits D1–D0
= 10 (MICBIAS = 2.5 V) At 4-mA load current, page 1 / register 46, bits D1–D0
= 01 (MICBIAS = 2 V)
0.5 ×
AVDD
5
7
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Electrical Characteristics (continued)
At 25°C, AVDD = HPVDD = IOVDD = 3.3 V, SPKVDD = 3.6 V, DVDD = 1.8 V, fS(audio) = 48 kHz, CODEC_CLKIN = 256 × fS, PLL = Off, VOL/MICDET pin disabled (unless otherwise noted)
PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
DAC HEADPHONE OUTPUT, AC-COUPLED LOAD = 16 (SINGLE-ENDED), DRIVER GAIN = 0 dB, PARASITIC CAPACITANCE = 30 pF
Full-scale output voltage
(0 dB) SNR Signal-to-noise ratio Measured as idle-channel noise, A-weighted THD Total harmonic distortion 0-dBFS input –85 –65 dB
THD+N 0-dBFS input –82 –60 dB
Total harmonic distortion +
noise
Mute attenuation 87 dB PSRR Power-supply rejection ratio
P
O
Maximum output power mW
DAC LINEOUT (HP Driver in Lineout Mode)
SNR Signal-to-noise ratio Measured as idle-channel noise, A-weighted 95 dB THD Total harmonic distortion 0-dBFS input, 0-dB gain –86 dB
THD+N 0-dBFS input, 0-dB gain –83 dB
Total harmonic distortion +
noise
DAC DIGITAL INTERPOLATION FILTER CHARACTERISTICS
See Section 5.6.1.4 for DAC interpolation filter characteristics.
DAC OUTPUT TO CLASS-D SPEAKER OUTPUT; LOAD = 4 Ω (DIFFERENTIAL), 50 pF
Output voltage Vrms
Output, common-mode 1.8 V
SNR Signal-to-noise ratio 6 dB, measured as idle-channel noise, A-weighted 88 dB
THD Total harmonic distortion –65 dB
THD+N –63 dB
Total harmonic distortion + SPKVDD = 3.6 V, BTL measurement, DAC input = –6
noise dBFS, CM = 1.8 V, class-D gain = 6 dB PSRR Power-supply rejection ratio –44 dB
Mute attenuation 110 dB
P
O
Maximum output power 1.5 W
(1) Ratio of output level with 1-kHz full-scale sine-wave input, to the output level with the inputs short-circuited, measured A-weighted over a
20-Hz to 20-kHz bandwidth using an audio analyzer.
(2) All performance measurements done with 20-kHz low-pass filter and, where noted, A-weighted filter. Failure to use such a filter may
result in higher THD+N and lower SNR and dynamic range readings than shown in the Electrical Characteristics. The low-pass filter removes out-of-band noise, which, although not audible, may affect dynamic specification values.
(3) DAC to headphone-out PSRR measurement is calculated as PSRR = 20 X log(V
Output common-mode setting = 1.65 V 0.707 Vrms
(1) (2)
(3)
Ripple on HPVDD (3.3 V) = 200 mVp-p at 1 kHz –62 dB
80 95 dB
RL= 32 , THD+N –60 dB 20 RL= 16 , THD+N –60 dB 60
SPKVDD = 3.6 V, BTL measurement, CM = 1.8 V, DAC input = 0 dBFS, class-D gain = 6 dB, THD 2.3 –16.5 dB
SPKVDD = 3.6 V, BTL measurement, CM = 1.8 V, DAC input = –2 dBFS, class-D gain = 6 dB, THD 2.1 –20 dB
SPKVDD = 3.6 V, BTL measurement, DAC input = mute, CM = 1.8 V, class-D gain = 6 dB
SPKVDD = 3.6 V, BTL measurement, class-D gain = (with respect to full-scale output value of 2.3 Vrms)
SPKVDD = 3.6 V, BTL measurement, DAC input = –6 dBFS, CM = 1.8 V, class-D gain = 6 dB
SPKVDD = 3.6 V, BTL measurement, ripple on SPKVDD = 200 mVp-p at 1 kHz
SPKVDD = 3.6 V, BTL measurement, CM = 1.8 V, class-D gain = 18 dB, THD = 10%
1
SPKVDD = 4.3 V, BTL measurement, CM = 1.8 V, class-D gain = 18 dB, THD = 10%
SPKVDD = 5.5 V, BTL measurement, CM = 1.8 V, class-D gain = 18 dB, THD = 10%
/ V
HPL
HPVDD
).
2.5
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Electrical Characteristics (continued)
At 25°C, AVDD = HPVDD = IOVDD = 3.3 V, SPKVDD = 3.6 V, DVDD = 1.8 V, fS(audio) = 48 kHz, CODEC_CLKIN = 256 × fS, PLL = Off, VOL/MICDET pin disabled (unless otherwise noted)
PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
DAC OUTPUT TO CLASS-D SPEAKER OUTPUT; LOAD = 8 (DIFFERENTIAL), 50 pF
SPKVDD = 3.6 V, BTL measurement, CM = 1.8 V, DAC input = 0 dBFS, class-D gain = 6 dB, THD 2.2
Output voltage Vrms
Output, common-mode 1.8 V
SNR Signal-to-noise ratio 6 dB, measured as idle-channel noise, A-weighted 87 dB
THD Total harmonic distortion –67 dB
THD+N –66 dB
Total harmonic distortion + SPKVDD = 3.6 V, BTL measurement, DAC input = –6
noise dBFS, CM = 1.8 V, class-D gain = 6 dB PSRR Power-supply rejection ratio
Mute attenuation 110 dB
P
O
Maximum output power 1 W
Output-stage leakage current SPKVDD = 4.3 V, device is powered down
for direct battery connection (power-up-reset condition)
DAC POWER CONSUMPTION
DAC power consumption based per selected processing block, see Section 5.4
(1) DAC to speaker-out PSRR is a differential measurement calculated as PSRR = 20 × log(V
–16.5 dB SPKVDD = 3.6 V, BTL measurement, CM = 1.8 V,
DAC input = –2 dBFS, class-D gain = 6 dB, THD 2.1 –20 dB
SPKVDD = 3.6 V, BTL measurement, DAC input = mute, CM = 1.8 V, class-D gain = 6 dB
SPKVDD = 3.6 V, BTL measurement, class-D gain = (with respect to full-scale output value of 2.3 Vrms)
SPKVDD = 3.6 V, BTL measurement, DAC input = –6 dBFS, CM = 1.8 V, class-D gain = 6 dB
SPKVDD = 3.6 V, BTL measurement, ripple on
(1)
SPKVDD = 200 mVp-p at 1 kHz
SPKVDD = 3.6 V, BTL measurement, CM = 1.8 V, class-D gain = 18 dB, THD = 10%
SPKVDD = 4.3 V, BTL measurement, CM = 1.8 V, class-D gain = 18 dB, THD = 10%
SPKVDD = 5.5 V, BTL measurement, CM = 1.8 V, class-D gain = 18 dB, THD = 10%
SPK(P + M)
/ V
–44 dB
0.7
1.6
80 nA
).
SPKVDD
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WCLK
BCLK
DIN
t (WS)
d
t (DI)
S
t (DI)
h
t
r
t
f
TLV320DAC3120
SLAS659–NOVEMBER 2009
Electrical Characteristics (continued)
At 25°C, AVDD = HPVDD = IOVDD = 3.3 V, SPKVDD = 3.6 V, DVDD = 1.8 V, fS(audio) = 48 kHz, CODEC_CLKIN = 256 × fS, PLL = Off, VOL/MICDET pin disabled (unless otherwise noted)
PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
DIGITAL INPUT/OUTPUT
Logic family CMOS
V
IH
IIH= 5 μA, IOVDD 1.6 V IIH= 5 μA, IOVDD < 1.6 V IOVDD
V
IL
V
OH
V
OL
Logic level
IIL= 5 μA, IOVDD 1.6 V –0.3 IIL= 5 μA, IOVDD < 1.6 V 0 IOH= 2 TTL loads V
IOL= 2 TTL loads V
Capacitive load 10 pF
0.7 ×
IOVDD
0.8 ×
IOVDD

3.4 Timing Characteristics

3.4.1 I2S/LJF/RJF Timing in Master Mode

0.3 ×
IOVDD
0.1 ×
IOVDD
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V
V
All specifications at 25°C, DVDD = 1.8 V
Note: All timing specifications are measured at characterization but not tested at final test.
PARAMETER IOVDD = 1.1 V IOVDD = 3.3 V UNIT
MIN MAX MIN MAX
td(WS) WCLK delay 45 20 ns ts(DI) DIN setup 8 6 ns th(DI) DIN hold 8 6 ns t
r
t
f
Rise time 25 10 ns Fall time 25 10 ns
Figure 3-1. I2S/LJF/RJF Timing in Master Mode
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WCLK
BCLK
DIN
t (WS)
h
t (BCLK)
H
t (DI)
S
t (BCLK)
L
t (DI)
h
t (WS)
S
t
r
t
f
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SLAS659–NOVEMBER 2009

3.4.2 I2S/LJF/RJF Timing in Slave Mode

All specifications at 25°C, DVDD = 1.8 V
Note: All timing specifications are measured at characterization but not tested at final test.
PARAMETER UNIT
tH(BCLK) BCLK high period 35 35 ns tL(BCLK) BCLK low period 35 35 ns ts(WS) WCLK setup 8 6 ns th(WS) WCLK hold 8 6 ns ts(DI) DIN setup 8 6 ns th(DI) DIN hold 8 6 ns t
r
t
f
Rise time 4 4 ns Fall time 4 4 ns
Figure 3-2. I2S/LJF/RJF Timing in Slave Mode
IOVDD = 1.1 V IOVDD = 3.3 V
MIN MAX MIN MAX
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WCLK
BCLK
DIN
t (WS)
d
t (WS)
d
t (DI)
S
t (DI)
h
t
f
t
r
TLV320DAC3120
SLAS659–NOVEMBER 2009

3.4.3 DSP Timing in Master Mode

All specifications at 25°C, DVDD = 1.8 V
Note: All timing specifications are measured at characterization but not tested at final test.
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PARAMETER UNIT
IOVDD = 1.1 V IOVDD = 3.3 V
MIN MAX MIN MAX
td(WS) WCLK delay 45 20 ns ts(DI) DIN setup 8 8 ns th(DI) DIN hold 8 8 ns t
r
t
f
Rise time 25 10 ns Fall time 25 10 ns
Figure 3-3. DSP Timing in Master Mode
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WCLK
BCLK
DIN
t (WS)
h
t (WS)
h
t (BCLK)
L
t
r
t
f
t (DI)
S
t (BCLK)
H
t (DI)
h
t (WS)
S
t (WS)
S
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3.4.4 DSP Timing in Slave Mode

All specifications at 25°C, DVDD = 1.8 V Note: All timing specifications are measured at characterization but not tested at final test.
SLAS659–NOVEMBER 2009
PARAMETER UNIT
IOVDD = 1.1 V IOVDD = 3.3 V
MIN MAX MIN MAX
tH(BCLK) BCLK high period 35 35 ns tL(BCLK) BCLK low period 35 35 ns ts(WS) WCLK setup 8 8 ns th(WS) WCLK hold 8 8 ns ts(DI) DIN setup 8 8 ns th(DI) DIN hold 8 8 ns t
r
t
f
Rise time 4 4 ns Fall time 4 4 ns
Figure 3-4. DSP Timing in Slave Mode
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STO
STA STA
STO
SDA
SCL
t
BUF
t
LOW
t
SU;STA
t
HIGH
t
HD;STA
t
r
t
HD;STA
t
HD;DAT
t
SU;DAT
t
SU;STO
t
f
T0295-02
TLV320DAC3120
SLAS659–NOVEMBER 2009

3.4.5 I2C Interface Timing

All specifications at 25°C, DVDD = 1.8 V
Note: All timing specifications are measured at characterization but not tested at final test.
PARAMETER Standard-Mode Fast-Mode UNITS
MIN TYP MAX MIN TYP MAX
f
SCL
t
HD;STA
t
LOW
t
HIGH
t
SU;STA
t
HD;DAT
t
SU;DAT
t
r
t
f
t
SU;STO
t
BUF
C
b
SCL clock frequency 0 100 0 400 kHz Hold time (repeated) START condition. 4 0.8 μs
After this period, the first clock pulse is generated.
LOW period of the SCL clock 4.7 1.3 μs HIGH period of the SCL clock 4 0.6 μs Setup time for a repeated START 4.7 0.8 μs
condition Data hold time: For I2C bus devices 0 3.45 0 0.9 μs Data setup time 250 100 ns SDA and SCL rise time 1000 20 + 0.1 C SDA and SCL fall time 300 20 + 0.1 C
b b
Set-up time for STOP condition 4 0.8 μs Bus free time between a STOP and 4.7 1.3 μs
START condition Capacitive load for each bus line 400 400 pF
Figure 3-5. I2C Interface Timing
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300 ns 300 ns
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f − Frequency − kHz
−160
−140
−120
−100
−80
−60
−40
−20
0
0 5 10 15 20
Amplitude − dBFS
G001
AVDD = HPVDD = 3.3 V IOVDD = SPKVDD = 3.3 V DVDD = 1.8 V
f − Frequency − kHz
−160
−140
−120
−100
−80
−60
−40
−20
0
0 5 10 15 20
Amplitude − dBFS
G002
AVDD = HPVDD = 3.3 V IOVDD = SPKVDD = 3.3 V DVDD = 1.8 V
−100
−90
−80
−70
−60
−50
−40
−30
−20
−10
0
0.00 0.02 0.04 0.06 0.08 0.10 0.12 0.14
G003
THD+N − Total Harmonic Distortion + Noise − dB
PO − Output Power − W
HPVDD = 3.3 V CM = 1.65 V
HPVDD = 3.6 V CM = 1.8 V
HPVDD = 3 V CM = 1.5 V
HPVDD = 2.7 V CM = 1.35 V
IOVDD = 3.3 V DVDD = 1.8 V Driver Gain = 9 dB RL = 16
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4 TYPICAL PERFORMANCE

4.1 DAC Performance

AMPLITUDE AMPLITUDE
vs vs
FREQUENCY FREQUENCY
SLAS659–NOVEMBER 2009
Figure 4-1. FFT - DAC to Line Output Figure 4-2. FFT - DAC to Headphone Output
Copyright © 2009, Texas Instruments Incorporated TYPICAL PERFORMANCE 13
TOTAL HARMONIC DISTORTION + NOISE
vs
OUTPUT POWER
Figure 4-3. Headphone Output Power
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−70
−60
−50
−40
−30
−20
−10
0
0.0 0.5 1.0 1.5 2.0 2.5 3.0 3.5 4.0
G004
THD+N − Total Harmonic Distortion + Noise − dB
PO − Output Power − W
Driver Gain
= 6 dB
Driver Gain
= 12 dB
AVDD = HPVDD = 3.3 V IOVDD = 3.3 V SPKVDD = 5.5 V DVDD = 1.8 V RL = 4
Driver Gain
= 18 dB
Driver Gain
= 24 dB
−70
−60
−50
−40
−30
−20
−10
0
0.0 0.5 1.0 1.5 2.0 2.5 3.0 3.5 4.0
G005
THD+N − Total Harmonic Distortion + Noise − dB
PO − Output Power − W
AVDD = 3.3 V HPVDD = 3.3 V IOVDD = 3.3 V DVDD = 1.8 V Driver Gain = 18 dB RL = 4
SPKVDD = 5.5 V
SPKVDD = 4.3 V
SPKVDD = 3.3 V
SPKVDD = 3.6 V
−70
−60
−50
−40
−30
−20
−10
0
0.0 0.5 1.0 1.5 2.0 2.5
G006
THD+N − Total Harmonic Distortion + Noise − dB
PO − Output Power − W
Driver Gain
= 6 dB
Driver Gain
= 12 dB
AVDD = HPVDD = 3.3 V IOVDD = 3.3 V SPKVDD = 5.5 V DVDD = 1.8 V RL = 8
Driver Gain
= 24 dB
Driver Gain
= 18 dB
−70
−60
−50
−40
−30
−20
−10
0
0.0 0.5 1.0 1.5 2.0 2.5 3.0
G007
THD+N − Total Harmonic Distortion + Noise − dB
PO − Output Power − W
AVDD = 3.3 V HPVDD = 3.3 V IOVDD = 3.3 V DVDD = 1.8 V Driver Gain = 18 dB RL = 8
SPKVDD = 5.5 V
SPKVDD = 4.3 V
SPKVDD = 3.3 V
SPKVDD = 3.6 V
TLV320DAC3120
SLAS659–NOVEMBER 2009

4.2 Class-D Speaker Driver Performance

TOTAL HARMONIC DISTORTION + NOISE TOTAL HARMONIC DISTORTION + NOISE
vs vs
OUTPUT POWER OUTPUT POWER
Figure 4-4. Max Class-D Speaker-Driver Output Figure 4-5. Class-D Speaker-Driver Output Power
Power (RL= 4 Ω) (RL= 4 Ω)
TOTAL HARMONIC DISTORTION + NOISE TOTAL HARMONIC DISTORTION + NOISE
vs vs
OUTPUT POWER OUTPUT POWER
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Figure 4-6. Max Class-D Speaker-Driver Output Figure 4-7. Class-D Speaker-Driver Output Power
14 TYPICAL PERFORMANCE Copyright © 2009, Texas Instruments Incorporated
Power (RL= 8 Ω) (RL= 8 Ω)
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f − Frequency − kHz
−160
−140
−120
−100
−80
−60
−40
−20
0
0 5 10 15 20
Amplitude − dBFS
G008
AVDD = HPVDD = 3.3 V IOVDD = SPKVDD = 3.3 V DVDD = 1.8 V
f − Frequency − kHz
−160
−140
−120
−100
−80
−60
−40
−20
0
0 5 10 15 20
Amplitude − dBFS
G009
AVDD = HPVDD = 3.3 V IOVDD = SPKVDD = 3.3 V DVDD = 1.8 V
I − Current − mA
0.0
0.5
1.0
1.5
2.0
2.5
3.0
3.5
0.0 0.5 1.0 1.5 2.0 2.5 3.0 3.5 4.0
V − Voltage − V
G010
Micbias = 2 V
Micbias = 2.5 V
Micbias = AVDD (3.3 V)
TLV320DAC3120
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4.3 Analog Bypass Performance

AMPLITUDE AMPLITUDE
vs vs
FREQUENCY FREQUENCY
Figure 4-8. FFT - Line In Bypass to Line Output
SLAS659–NOVEMBER 2009
Figure 4-9. FFT - Line In Bypass to Headphone
Output

4.4 MICBIAS Performance

VOLTAGE
vs
CURRENT
Figure 4-10. Micbias
Copyright © 2009, Texas Instruments Incorporated TYPICAL PERFORMANCE 15
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HPVDD
HPVSSSPKVDD SPKVSS
AVDD
AVSSSPKVSS
VOL/MICDET
MICBIAS
TLV320DAC3120
SPKVDD
22 Fm0.1 Fm0.1 Fm 22 Fm
SVDD
AVDD
AVSS
0.1 Fm
10 Fm
0.1 Fm
10 Fm
+3.3VA
1 Fm
DVDD
DVSS IOVDD IOVSS
0.1 Fm 10 Fm
+1.8VD
IOVDD
0.1 Fm 10 Fm
Host Processor
SDIN
BCLK
WCLK
MCLK
SCL
SDA
RESET
GPIO1
S0400-06
AIN1
AIN2
To External
MIC Circuitry
Analog In
SPKM SPKM
SPKP SPKP
HPR
HPL
8- or Speaker
W 4-W
Stereo Headphone Out
R1
34.8 kW
P1
25 kW
R2
9.76 kW
TLV320DAC3120
SLAS659–NOVEMBER 2009

5 APPLICATION INFORMATION

5.1 Typical Circuit Configuration

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Figure 5-1. Typical Circuit Configuration
16 APPLICATION INFORMATION Copyright © 2009, Texas Instruments Incorporated
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5.2 Overview

The TLV320DAC3120 is a highly integrated mono audio DAC for portable computing, communication, and entertainment applications. A register-based architecture eases integration with microprocessor-based systems through standard serial-interface buses. This device contains a two-wire I2C bus interface, which allows full register access. All peripheral functions are controlled through these registers and the onboard state machines.
The TLV320DAC3120 consists of the following blocks:
miniDSP digital signal-processing block
Audio DAC
Dynamic range compressor (DRC)
Mono headphone/lineout amplifier
Class-D mono amplifier capable of driving 4-Ω or 8-speakers
Pin-controlled or register-controlled volume level
Power-down de-pop and power-up soft start
Analog inputs
I2C control interface
Power-down control block Following a toggle of the RESET pin or a software reset, the device operates in the default mode. The I2C
interface is used to write to the control registers to configure the device. The I2C address assigned to the TLV320DAC3120 is 001 1000. This device always operates in an I2C
slave mode. All registers are 8-bit, and all writable registers have readback capability. The device auto-increments to support sequential addressing and can be used with I2C fast mode. Once the device is reset, all appropriate registers are updated by the host processor to configure the device as needed by the user.

5.2.1 Device Initialization

5.2.1.1 Reset
The TLV320DAC3120 internal logic must be initialized to a known condition for proper device function. To initialize the device to its default operating condition, the hardware reset pin (RESET) must be pulled low for at least 10 ns. For this initialization to work, both the IOVDD and DVDD supplies must be powered up. It is recommended that while the DVDD supply is being powered up, the RESET pin be pulled low.
The device can also be reset via software reset. Writing a 1 into page 0 / register 1, bit D0 resets the device.
5.2.1.2 Device Start-Up Lockout Times
After the TLV320DAC3120 is initialized through hardware reset at power-up or software reset, the internal memories are initialized to default values. This initialization takes place within 1 ms after pulling the RESET signal high. During this initialization phase, no register-read or register-write operation should be performed on the DAC coefficient buffers. Also, no block within the codec should be powered up during the initialization phase.
5.2.1.3 PLL Start-Up
Whenever the PLL is powered up, a start-up delay of approximately of 10 ms occurs after the power-up command of the PLL and before the clocks are available to the codec. This delay is to ensure stable operation of the PLL and clock-divider logic.
Copyright © 2009, Texas Instruments Incorporated APPLICATION INFORMATION 17
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SLAS659–NOVEMBER 2009
5.2.1.4 Power-Stage Reset
The power-stage-only reset is used to reset the device after an overcurrent latching shutdown has occurred. Using this reset re-enables the output stage without resetting all of the registers in the device. Each of the two power stages has its own dedicated reset bit. The headphone power-stage reset is performed by setting page 1 / register 31, bit D7 for HPOUT. The speaker power-stage reset is performed by setting page 1 / register 32, bit D7 for SPKP and SPKM.
5.2.1.5 Software Power Down
By default, all circuit blocks are powered down following a reset condition. Hardware power up of each circuit block can be controlled by writing to the appropriate control register. This approach allows the lowest power-supply current for the functionality required. However, when a block is powered down, all of the register settings are maintained as long as power is still being applied to the device.

5.2.2 Audio Analog I/O

The TLV320DAC3120 features a mono audio DAC. It supports a wide range of analog interfaces to support different headsets and analog outputs. The TLV320DAC3120 interfaces to output drivers (8-, 16-, 32-).

5.3 miniDSP

The TLV320DAC3120 features a miniDSP core which is tightly coupled to the DAC. The fully programmable algorithms for the miniDSP must be loaded into the device after power up. The miniDSP has direct access to the digital stereo audio stream, offering the possibility for advanced, very low-group-delay DSP algorithms. The miniDSP has 1024 programmable instructions, 896 data memory locations, and 512 programmable coefficients (in the adaptive mode, each bank has 256 programmable coefficients).
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5.3.1 Software

Software development for the TLV320DAC3120 is supported through TI's comprehensive PurePath™ Studio software development environment, a powerful, easy-to-use tool designed specifically to simplify software development on Texas Instruments miniDSP audio platforms. The graphical development environment consists of a library of common audio functions that can be dragged and dropped into an audio signal flow and graphically connected together. The DSP code can then be assembled from the graphical signal flow with the click of a mouse. See the TLV320DAC3120 product folder on www.ti.com to learn more about PurePath Studio and the latest status on available, ready-to-use DSP algorithms.

5.4 Digital Processing Low-Power Modes

The TLV320DAC3120 device can be tuned to minimize power dissipation, to maximize performance, or to an operating point between the two extremes to best fit the application. The choice of processing blocks, PRB_P4 to PRB_P22 for mono playback and PRB_R4 to PRB_R18 for mono recording, also influences the power consumption. In fact, the numerous processing blocks have been implemented to offer a choice among configurations having a different balance of power-optimization and signal-processing capabilities.
18 APPLICATION INFORMATION Copyright © 2009, Texas Instruments Incorporated
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SLAS659–NOVEMBER 2009

5.4.1 DAC Playback on Headphones, Mono, 48 kHz, DVDD = 1.8 V, AVDD = 3.3 V, HPVDD = 3.3 V

DOSR = 128, Processing Block = PRB_P12 (Interpolation Filter B)
Power consumption = 15.4 mW
Table 5-1. PRB_P12 Alternative Processing Blocks, 15.4 mW
Processing Block Filter Estimated Power Change (mW)
PRB_P4 A 0.57 PRB_P5 A 1.48
PRB_P6 A 1.08 PRB_P13 B 0.56 PRB_P14 B 0.27 PRB_P15 B 0.89 PRB_P16 B 0.31
DOSR = 64, Processing Block = PRB_P12 (Interpolation Filter B)
Power consumption = 15.54 mW
Table 5-2. PRB_P12 Alternative Processing Blocks, 15.54 mW
Processing Block Filter Estimated Power Change (mW)
PRB_P4 A 0.37
PRB_P5 A 1.23
PRB_P6 A 1.15 PRB_P13 B 0.43 PRB_P14 B 0.13 PRB_P15 B 0.85 PRB_P16 B 0.21

5.4.2 DAC Playback on Headphones, Mono, 8 kHz, DVDD = 1.8 V, AVDD = 3.3 V, HPVDD = 3.3 V

DOSR = 768, Processing Block = PRB_P12 (Interpolation Filter B)
Power consumption = 14.49 mW
Table 5-3. PRB_P12 Alternative Processing Blocks, 14.49 mW
Processing Block Filter Estimated Power Change (mW)
PRB_P4 A –0.04 PRB_P5 A 0.2
PRB_P6 A –0.01 PRB_P13 B 0.1 PRB_P14 B 0.05 PRB_P15 B –0.03 PRB_P16 B 0.07
Copyright © 2009, Texas Instruments Incorporated APPLICATION INFORMATION 19
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SLAS659–NOVEMBER 2009
DOSR = 384, Processing Block = PRB_P12 (Interpolation Filter B)
Power consumption = 14.42 mW
Table 5-4. PRB_P12 Alternative Processing Blocks, 14.42 mW
Processing Block Filter Estimated Power Change (mW)
PRB_P4 A 0.16
PRB_P5 A 0.3
PRB_P6 A 0.2 PRB_P13 B 0.15 PRB_P14 B 0.07 PRB_P15 B 0.18 PRB_P16 B 0.09

5.5 Analog Signals

The TLV320DAC3120 analog signals consist of:
Microphone bias (MICBIAS)
Analog inputs AIN1 and AIN2, which can be used to pass-through or mix analog signals to output stages
Analog outputs class-D speaker driver and headphone/lineout driver providing output capability for the DAC, AIN1, AIN2, or a mix of the three
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5.5.1 MICBIAS

The TLV320DAC3120 includes a microphone bias circuit which can source up to 4 mA of current, and is programmable to a 2-V, 2.5-V, or AVDD level. The level can be controlled by writing to page 1 / register 46, bits D1–D0. This functionality is shown in Table 5-5.
Table 5-5. MICBIAS Settings
D1 D0 FUNCTIONALITY
0 0 MICBIAS output is powered down. 0 1 MICBIAS output is powered to 2 V. 1 0 MICBIAS output is powered to 2.5 V. 1 1 MICBIAS output is powered to AVDD.
During normal operation, MICBIAS can be set to 2.5 V for better performance. However, depending on the model of microphone that is selected, optimal performance might be obtained at another setting, so the performance at a given setting should be verified.
The lowest current consumption occurs when MICBIAS is powered down. The next-lowest current consumption occurs when MICBIAS is set at AVDD.

5.5.2 Analog Inputs AIN1 and AIN2

AIN1 (pin 13) and AIN2 (pin 14) are inputs to the output mixer along with the DAC output. Page 1 / register 36 provides control signals for determining the signals routed through the output mixer. The output of the output mixer then can be attenuated or amplified through the class-D and/or headphone/lineout drivers.
20 APPLICATION INFORMATION Copyright © 2009, Texas Instruments Incorporated
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5.6 Audio DAC and Audio Analog Outputs

The mono audio DAC consists of a digital audio processing block, a digital interpolation filter, a digital delta-sigma modulator, and an analog reconstruction filter. The high oversampling ratio (normally DOSR is between 32 and 128) exhibits good dynamic range by ensuring that the quantization noise generated within the delta-sigma modulator stays outside of the audio frequency band. Audio analog outputs include mono headphone/lineout and mono class-D speaker outputs. Because the TLV320DAC3120 contains a mono DAC, it inputs the mono data from the left channel, the right channel, or a mix of the left and right channels as [(L + R) ÷ 2], selected by page 0 / register 63, bits D5–D4. See Figure 1-1 for the signal flow.

5.6.1 DAC

The TLV320DAC3120 mono audio DAC supports data rates from 8 kHz to 192 kHz. The audio channel of the mono DAC consists of a signal-processing engine with fixed processing blocks, a programmable miniDSP, a digital interpolation filter, multibit digital delta-sigma modulator, and an analog reconstruction filter. The DAC is designed to provide enhanced performance at low sampling rates through increased oversampling and image filtering, thereby keeping quantization noise generated within the delta-sigma modulator and observed in the signal images strongly suppressed within the audio band to beyond 20 kHz. To handle multiple input rates and optimize power dissipation and performance, the TLV320DAC3120 allows the system designer to program the oversampling rates over a wide range from 1 to 1024 by configuring page 0 / register 13 and page 0 / register 14. The system designer can choose higher oversampling ratios for lower input data rates and lower oversampling ratios for higher input data rates.
The TLV320DAC3120 DAC channel includes a built-in digital interpolation filter to generate oversampled data for the delta-sigma modulator. The interpolation filter can be chosen from three different types, depending on required frequency response, group delay, and sampling rate.
DAC power up is controlled by writing to page 0 / register 63, bit D7 for the mono channel. The mono-channel DAC clipping flag is provided as a read-only bit on page 0 / register 39, bit D7.
5.6.1.1 DAC Processing Blocks
The TLV320DAC3120 implements signal-processing capabilities and interpolation filtering via processing blocks. These fixed processing blocks give users the choice of how much and what type of signal processing they may use and which interpolation filter is applied.
The choices among these processing blocks allows the system designer to balance power conservation and signal-processing flexibility. Table 5-6 gives an overview of all available processing blocks of the DAC channel and their properties. The resource-class column gives an approximate indication of power consumption for the digital (DVDD) supply; however, based on the out-of-band noise spectrum, the analog power consumption of the drivers (HPVDD) may differ.
The signal-processing blocks available are:
First-order IIR
Scalable number of biquad filters
The processing blocks are tuned for common cases and can achieve high image rejection or low group delay in combination with various signal-processing effects such as audio effects and frequency shaping. The available first-order IIR and biquad filters have fully user-programmable coefficients.
Copyright © 2009, Texas Instruments Incorporated APPLICATION INFORMATION 21
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Interp. Filter A
BiQuad
C
BiQuad
B
BiQuad
A
to
Modulator
Digital
Volume
Ctrl
from
Interface
´
Interp.
Filter
A,B
DRCHPF
BiQuad
F
BiQuad
E
BiQuad
D
BiQuad
C
BiQuad
B
BiQuad
A
IIR
to
Modulator
Digital
Volume
Ctrl
from
Interface
´
Interp.
Filter
A,B
BiQuad
F
BiQuad
E
BiQuad
D
BiQuad
C
BiQuad
B
BiQuad
A
IIR
to
Modulator
Digital
Volume
Ctrl
from
Interface
´
TLV320DAC3120
SLAS659–NOVEMBER 2009
Table 5-6. Overview – DAC Predefined Processing Blocks
Processing First-Order Number of Resource
Block No. IIR Available Biquads Class
PRB_P4 A Mono No 3 No 4 PRB_P5 A Mono Yes 6 Yes 6
PRB_P6 A Mono Yes 6 No 6 PRB_P12 B Mono Yes 0 No 3 PRB_P13 B Mono No 4 Yes 4 PRB_P14 B Mono No 4 No 4 PRB_P15 B Mono Yes 6 Yes 6 PRB_P16 B Mono Yes 6 No 4 PRB_P20 C Mono Yes 0 No 2 PRB_P21 C Mono Yes 4 Yes 3 PRB_P22 C Mono Yes 4 No 2
Interpolation Filter Channel DRC
5.6.1.2 DAC Processing Blocks – Signal Chain Details
5.6.1.2.1 Three Biquads, Filter A
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Figure 5-2. Signal Chain for PRB_P4
5.6.1.2.2 Six Biquads, First-Order IIR, DRC, Filter A or B
Figure 5-3. Signal Chain for PRB_P5 and PRB_P15
5.6.1.2.3 Six Biquads, First-Order IIR, Filter A or B
Figure 5-4. Signal Chain for PRB_P6 and PRB_P16
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22 APPLICATION INFORMATION Copyright © 2009, Texas Instruments Incorporated
Interp.
Filter
B,C
IIR
to
Modulator
Digital
Volume
Ctrl
from
Interface
´
Interp.
FilterB
DRCHPF
to
Modulator
Digital
Volume
Ctrl
BiQuad
D
BiQuad
C
BiQuad
B
BiQuad
A
from
Interface
´
Interp.
FilterB
BiQuad
D
BiQuad
C
BiQuad
B
BiQuad
A
to
Modulator
Digital
Volume
Ctrl
from
Interface
´
Interp.
FilterC
DRCHPF
IIR
to
Modulator
Digital
Volume
Ctrl
BiQuad
D
BiQuad
C
BiQuad
B
BiQuad
A
from
Interface
´
Interp.
FilterC
BiQuad
D
BiQuad
C
BiQuad
B
BiQuad
A
IIR
to
modulator
Digital
Volume
Ctrl
from
Interface
´
TLV320DAC3120
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5.6.1.2.4 IIR, Filter B or C
Figure 5-5. Signal Chain for PRB_P12 and PRB_P20
5.6.1.2.5 Four Biquads, DRC, Filter B
Figure 5-6. Signal Chain for PRB_P13
5.6.1.2.6 Four Biquads, Filter B
SLAS659–NOVEMBER 2009
Figure 5-7. Signal Chain for PRB_P14
5.6.1.2.7 Four Biquads, First-Order IIR, DRC, Filter C
Figure 5-8. Signal Chain for PRB_P21
5.6.1.2.8 Four Biquads, First-Order IIR, Filter C
Figure 5-9. Signal Chain for PRB_P22
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Copyright © 2009, Texas Instruments Incorporated APPLICATION INFORMATION 23
LargestPositiveNumber:
=0.111 1111 1111 111 =0.999969482421875=1.0 – 1LSB
1
LargestNegativeNumber:
=1.0000 0000 0000 000 =0x8000= –1.0(bydefinition)
1 1 1
1 1 1
S.xxxx xxxx xxxx xxx.. x x x
SignBit
2 Bit
–1
2 Bit
–4
2 Bit
–15
Fraction
Point
TLV320DAC3120
SLAS659–NOVEMBER 2009
5.6.1.3 DAC User-Programmable Filters
Depending on the selected processing block, different types and orders of digital filtering are available. Up to six biquad sections are available for specific processing blocks.
The coefficients of the available filters are arranged as sequentially-indexed coefficients in two banks. If adaptive filtering is chosen, the coefficient banks can be switched in real time.
When the DAC is running, the user-programmable filter coefficients are locked and cannot be accessed for either read or write.
However, the TLV320DAC3120 offers an adaptive filter mode as well. Setting page 8 / register 1, bit D2 = 1 turns on double buffering of the coefficients. In this mode, filter coefficients can be updated through the host and activated without stopping and restarting the DAC. This enables advanced adaptive filtering applications.
In the double-buffering scheme, all coefficients are stored in two buffers (buffers A and B). When the DAC is running and adaptive filtering mode is turned on, setting page 8 / register 1, bit D0 = 1 switches the coefficient buffers at the next start of a sampling period. This bit is set back to 0 after the switch occurs. At the same time, page 8 / register 1, bit D1 toggles.
The flag in page 8 / register 1, bit D1 indicates which of the two buffers is actually in use. Page 8 / register 1, bit D1 = 0: buffer A is in use by the DAC engine; bit D1 = 1: buffer B is in use. While the device is running, coefficient updates are always made to the buffer not in use by the DAC,
regardless of the buffer to which the coefficients have been written.
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Table 5-7. Adaptive-Mode Filter-Coefficient Buffer Switching
DAC Powered Up Page 8, Reg 1, Bit D1 I2C Writes to Updates
No 0 None Page 8, Reg 2–3, buffer A Page 8, Reg 2–3, buffer A No 0 None Page 12, Reg 2–3, buffer Page 12, Reg 2–3, buffer B
Yes 0 Buffer A Page 8, Reg 2–3, buffer A Page 12, Reg 2–3, buffer B Yes 0 Buffer A Page 12, Reg 2–3, buffer Page 12, Reg 2–3, buffer B
Yes 1 Buffer B Page 8, Reg 2–3, buffer A Page 8, Reg 2–3, buffer A Yes 1 Buffer B Page 12, Reg 2–3, buffer Page 8, Reg 2–3, buffer A
Coefficient Buffer in
Use
B
B
B
The user-programmable coefficients for the DAC processing blocks are defined on pages 8 and 9 for buffer A and pages 12 and 13 for buffer B.
The coefficients of these filters are each 16-bit, 2s-complement format, occupying two consecutive 8-bit registers in the register space. Specifically, the filter coefficients are in 1.15 (one dot 15) format with a range from –1.0 (0x8000) to 0.999969482421875 (0x7FFF) as shown in Figure 5-10.
24 APPLICATION INFORMATION Copyright © 2009, Texas Instruments Incorporated
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Figure 5-10.
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1
0 1
15 1
1
N N z
H(z)
2 D z
-
-
+
=
-
1 2
0 1 2
15 1 2
1 2
N 2 N z N z
H(z)
2 2 D z D z
- -
- -
+ ´ +
=
- ´ -
TLV320DAC3120
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SLAS659–NOVEMBER 2009
5.6.1.3.1 First-Order IIR Section
The IIR is of first order and its transfer function is given by
The frequency response for the first-order IIR section with default coefficients is flat.
Table 5-8. DAC IIR Filter Coefficients
Filter Filter Coefficient Default (Reset) Values
First-order IIR N0 Page 9 / registers 2–3 0x7FFF (decimal 1.0 – LSB value)
N1 Page 9 / registers 4–5 0x0000 D1 Page 9 / registers 6–7 0x0000
DAC Coefficient,
Mono Channel
5.6.1.3.2 Biquad Section
The transfer function of each of the biquad filters is given by
Table 5-9. DAC Biquad Filter Coefficients
(1)
(2)
Filter Coefficient Mono DAC Channel Default (Reset) Values
Biquad A N0 Page 8 / registers 2–3 0x7FFF (decimal 1.0 – LSB value)
N1 Page 8 / registers 4–5 0x0000 N2 Page 8 / registers 6–7 0x0000 D1 Page 8 / registers 8–9 0x0000 D2 Page 8 / registers 10–11 0x0000
Biquad B N0 Page 8 / registers 12–13 0x7FFF (decimal 1.0 – LSB value)
N1 Page 8 / registers 14–15 0x0000 N2 Page 8 / registers 16–17 0x0000 D1 Page 8 / registers 18–19 0x0000 D2 Page 8 / registers 20–21 0x0000
Biquad C N0 Page 8 / registers 22–23 0x7FFF (decimal 1.0 – LSB value)
N1 Page 8 / registers 24–25 0x0000 N2 Page 8 / registers 26–27 0x0000 D1 Page 8 / registers 28–29 0x0000 D2 Page 8 / registers 30–31 0x0000
Biquad D N0 Page 8 / registers 32–33 0x7FFF (decimal 1.0 – LSB value)
N1 Page 8 / registers 34–35 0x0000 N2 Page 8 / registers 36–37 0x0000 D1 Page 8 / registers 38–39 0x0000 D2 Page 8 / registers 40–41 0x0000
Biquad E N0 Page 8 / registers 42–43 0x7FFF (decimal 1.0 – LSB value)
N1 Page 8 / registers 44–45 0x0000 N2 Page 8 / registers 46–47 0x0000 D1 Page 8 / registers 48–49 0x0000 D2 Page 8 / registers 50–51 0x0000
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0
–10
–20
–30
–40
–50
–60
–70
–80
–90
1
3 4
5 6
7
2
Magnitude – dB
FrequencyNormalizedtof
S
DACChannelResponseforInterpolationFilter A (RedLineCorrespondsto –65dB)
TLV320DAC3120
SLAS659–NOVEMBER 2009
Table 5-9. DAC Biquad Filter Coefficients (continued)
Filter Coefficient Mono DAC Channel Default (Reset) Values
Biquad F N0 Page 8 / registers 52–53 0x7FFF (decimal 1.0 – LSB value)
N1 Page 8 / registers 54–55 0x0000 N2 Page 8 / registers 56–57 0x0000 D1 Page 8 / registers 58–59 0x0000 D2 Page 8 / registers 60–61 0x0000
5.6.1.4 DAC Interpolation Filter Characteristics
5.6.1.4.1 Interpolation Filter A
Filter A is designed for an fSup to 48 ksps with a flat pass band of 0 kHz–20 kHz.
Table 5-10. Specification for DAC Interpolation Filter A
Parameter Condition Value (Typical) Unit
Filter-gain pass band 0 … 0.45 f Filter-gain stop band 0.55 fS… 7.455 f Filter group delay 21/f
S
S
±0.015 dB –65 dB
S
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s
5.6.1.4.2 Interpolation Filter B
Filter B is specifically designed for an fSup to 96 ksps. Thus, the flat pass-band region easily covers the required audio band of 0 kHz–20 kHz.
Filter-gain pass band 0 … 0.45 f Filter-gain stop band 0.55 fS… 3.45 f Filter group delay 18/f
26 APPLICATION INFORMATION Copyright © 2009, Texas Instruments Incorporated
Parameter Condition Value (Typical) Unit
Figure 5-11. Frequency Response of DAC Interpolation Filter A
Table 5-11. Specification for DAC Interpolation Filter B
±0.015 dB –58 dB
S
S
S
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s
DACChannelResponseforInterpolationFilterB (RedLineCorrespondsto –58dB)
0.5 1 1.5 2 2.5 3 3.5
FrequencyNormalizedtof
S
0
–10
–20
–30
–40
–50
–60
–70
–80
Magnitude – dB
0
–10
–20
–30
–40
–50
–60
–70
Magnitude – dB
0 0.2 0.4 0.6 0.8 1 1.2 1.4
FrequencyNormalizedtof
S
DACChannelResponseforInterpolationFilterC (RedLineCorrespondsto –43dB)
TLV320DAC3120
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Figure 5-12. Frequency Response of Channel Interpolation Filter B
5.6.1.4.3 Interpolation Filter C
Filter C is specifically designed for the 192-ksps mode. The pass band extends up to 0.4 × f (corresponds to 80 kHz), more than sufficient for audio applications.
S
Figure 5-13. Frequency Response of DAC Interpolation Filter C
Table 5-12. Specification for DAC Interpolation Filter C
Parameter Condition Value (Typical) Unit
Filter-gain pass band 0 … 0.35 f Filter-gain stop band 0.6 fS… 1.4 f Filter group delay 13/f

5.6.2 DAC Digital-Volume Control

The DAC has a digital volume-control block which implements programmable gain. Each channel has an independent volume control that can be varied from 24 dB to –63.5 dB in 0.5-dB steps. The mono-channel DAC volume can be controlled by writing to page 0 / register 65, bits D7–D0. DAC muting and setting up a master gain control to control the mono channel is done by writing to page 0 / register 64, bits D3 and D1. The gain is implemented with a soft-stepping algorithm, which only changes the actual volume by
Copyright © 2009, Texas Instruments Incorporated APPLICATION INFORMATION 27
S
S
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±0.03 dB –43 dB
S
s
TLV320DAC3120
SLAS659–NOVEMBER 2009
0.125 dB per input sample, either up or down, until the desired volume is reached. The rate of soft-stepping can be slowed to one step per two input samples by writing to page 0 / register 63, bits D1–D0. Note that the default source for volume-control level settings is controlled by register writes to page 0 / register 65. Use of the VOL/MICDET pin to control the DAC volume is ignored until the volume-control source selected has been changed to pin control (page 0 / register 116, bit D7 = 1). This functionality is shown in Figure 1-1.
During soft-stepping, the host does not receive a signal when the DAC has been completely muted. This may be important if the host must mute the DAC before making a significant change, such as changing sample rates. In order to help with this situation, the device provides a flag back to the host via a read-only register, page 0 / register 38, bit D4 for the mono channel. This information alerts the host when the part has completed the soft-stepping, and the actual volume has reached the desired volume level. The soft-stepping feature can be disabled by writing to page 0 / register 63, bits D1–D0.
If soft-stepping is enabled, the CODEC_CLKIN signal should be kept active until the DAC power-up flag is cleared. When this flag is cleared, the internal DAC soft-stepping process is complete, and CODEC_CLKIN can be stopped if desired. (The analog volume control can be ramped down using an internal oscillator.)

5.6.3 Volume-Control Pin

The range of voltages used by the 7-bit SAR ADC is shown in the Electrical Characteristics table. The volume-control pin is not enabled by default, but it can be enabled by writing 1 to page 0 /
register 116, bit D7. The default DAC volume control uses software control of the volume, which occurs if page 0 / register 116, bit D7 = 0. Soft-stepping the volume level is set up by writing to page 0 / register 63, bits D1–D0.
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When the volume-pin function is used, a 7-bit Vol ADC reads the voltage on the VOL/MICDET pin and updates the digital volume control. (It overwrites the current value of the volume control.) The new volume setting which has been applied due to a change of voltage on the volume control pin can be read on page 0 / register 117, bits D6–D0. The 7-bit Vol ADC clock source can be selected on page 0 / register 116, bit D6. The update rate can be programmed on page 0 / register 116, bits D2–D0 for this 7-bit SAR ADC.
The VOL/MICDET pin gain mapping is shown in Table 5-13.
Table 5-13. VOL/MICDET Pin Gain Mapping
VOL/MICDET PIN SAR OUTPUT DIGITAL GAIN APPLIED
0 18 dB 1 17.5 dB 2 17 dB
: : 35 0.5 dB 36 0.0 dB 37 –0.5 dB
: : 89 –26.5 dB 90 –27 dB 91 –28 dB
: :
125 –62 dB 126 –63 dB 127 Mute
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1
0 1
HPF
15 1
1
N N z
H (z)
2 D z
-
-
+
=
-
1
0 1
LPF
15 1
1
N N z
H (z)
2 D z
-
-
+
=
-
TLV320DAC3120
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SLAS659–NOVEMBER 2009
The VOL/MICDET pin connection and functionality are shown in Figure 1-1. As shown in Table 5-13, the VOL/MICDET pin has a range of volume control from 18 dB down to –63 dB,
and mute. However, if less maximum gain is required, then a smaller range of voltage should be applied to the VOL/MICDET pin. This can be done by increasing the value of R2 relative to the value of (P1 + R1), so that more voltage is available at the bottom of P1. The circuit should also be designed such that for the values of R1, R2, and P1 chosen, the maximum voltage (top of the potentiometer) does not exceed AVDD/2 (see Figure 5-1). The recommended values for R1, R2, and P1 for several maximum gains are shown in Table 5-14. Note that In typical applications, R1 should not be 0 , as the VOL/MICDET pin should not exceed AVDD/2 for proper ADC operation.
Table 5-14. VOL/MICDET Pin Gain Scaling
R1 P1 R2 DIGITAL GAIN RANGE
(k) (k) (k) (dB)
25 25 0 0 V to 1.65 V 18 dB to –63 dB 33 25 7.68 0.386 V to 1.642 V 3 dB to –63 dB
34.8 25 9.76 0.463 V to 1.649 V 0 dB to –63 dB
ADC VOLTAGE
for AVDD = 3.3 V
(V)

5.6.4 Dynamic Range Compression

Typical music signals are characterized by crest factors, the ratio of peak signal power to average signal power, of 12 dB or more. To avoid audible distortions due to clipping of peak signals, the gain of the DAC channel must be adjusted so as not to cause hard clipping of peak signals. As a result, during nominal periods, the applied gain is low, causing the perception that the signal is not loud enough. To overcome this problem, DRC in the TLV320DAC3120 continuously monitors the output of the DAC digital volume control to detect its power level relative to 0 dBFS. When the power level is low, DRC increases the input signal gain to make it sound louder. At the same time, if a peaking signal is detected, it autonomously reduces the applied gain to avoid hard clipping. This results in sounds more pleasing to the ear as well as sounding louder during nominal periods.
The DRC functionality in the TLV320DAC3120 is implemented by a combination of processing blocks in the DAC channel as described in Section 5.6.1.2.
DRC can be disabled by writing to page 0 / register 68, bits D6–D5. DRC typically works on the filtered version of the input signal. The input signals have no audio information
at dc and extremely low frequencies; however, they can significantly influence the energy estimation function in DRC. Also, most of the information about signal energy is concentrated in the low-frequency region of the input signal.
To estimate the energy of the input signal, the signal is first fed to the DRC high-pass filter and then to the DRC low-pass filter. These filters are implemented as first-order IIR filters given by
(3)
(4)
The coefficients for these filters are 16 bits wide in 2s-complement format and are user-programmable through register write as given in Table 5-15.
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Table 5-15. DRC HPF and LPF Coefficients
Coefficient Location
HPF N0 C71 page 9 / registers 14 to 15 HPF N1 C72 page 9 / registers 16 to 17 HPF D1 C73 page 9 / registers 18 to 19 LPF N0 C74 page 9 / registers 20 to 21 LPF N1 C75 page 9 / registers 22 to 23 LPF D1 C76 page 9 / registers 24 to 25
The default values of these coefficients implement a high-pass filter with a cutoff at 0.00166 × DAC_fS, and a low-pass filter with a cutoff at 0.00033 × DAC_fS.
The output of the DRC high-pass filter is fed to the processing block selected for the DAC channel. The absolute value of the DRC-LPF filter is used for energy estimation within the DRC.
The gain in the DAC digital volume control is controlled by page 0 / registers 65 and 66. When the DRC is enabled, the applied gain is a function of the digital volume-control register setting and the output of the DRC.
The DRC parameters are described in sections that follow.
5.6.4.1 DRC Threshold
The DRC threshold represents the level of the DAC playback signal at which the gain compression becomes active. The output of the digital volume control in the DAC is compared with the set threshold. The threshold value is programmable by writing to page 0 / register 68, bits D4–D2. The threshold value can be adjusted between –3 dBFS and –24 dBFS in steps of 3 dB. Keeping the DRC threshold value too high may not leave enough time for the DRC block to detect peaking signals, and can cause excessive distortion at the outputs. Keeping the DRC threshold value too low can limit the perceived loudness of the output signal.
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The recommended DRC threshold value is –24 dB. When the output signal exceeds the set DRC threshold, the interrupt flag bits at page 0 / register 44,
bits D3–D2 are updated. These flag bits are sticky in nature, and are reset only after they are read back by the user. The non-sticky versions of the interrupt flags are also available at page 0 / register 46, bits D3–D2.
5.6.4.2 DRC Hysteresis
DRC hysteresis is programmable by writing to page 0 / register 68, bits D1–D0. These bits can be programmed to represent values between 0 dB and 3 dB in steps of 1 dB. It is a programmable window around the programmed DRC threshold that must be exceeded for disabled DRC to become enabled, or enabled DRC to become disabled. For example, if the DRC threshold is set to –12 dBFS and the DRC hysteresis is set to 3 dB, then if the gain compression in DRC is inactive, the output of the DAC digital volume control must exceed –9 dBFS before gain compression due to the DRC is activated. Similarly, when the gain compression in the DRC is active, the output of the DAC digital volume control must fall below –15 dBFS for gain compression in the DRC to be deactivated. The DRC hysteresis feature prevents the rapid activation and de-activation of gain compression in DRC in cases when the output of the DAC digital volume control rapidly fluctuates in a narrow region around the programmed DRC threshold. By programming the DRC hysteresis as 0 dB, the hysteresis action is disabled.
The recommended value of DRC hysteresis is 3 dB.
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