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       

FEATURES
DLow Power High Quality Audio DAC
DStereo Audio DAC Support Rates up to
48 ksps
DHigh Quality 97-dBA Stereo Audio Playback
Performance
DLow Power: 11-mW Stereo Audio Playback at
48 ksps
DOn-Chip 325-mW, 8-W Speaker Driver
DStereo Headphone Amplifier With Capless
Output Option
DIntegrated PLL for Flexible Audio Clock
Generation
DProgrammable Digital Audio
Bass/Treble/EQ/De-Emphasis
DMicrophone and AUX Inputs Available for
Analog Sidetone Mixing
DMicrophone Bias and Pre−Amp
DSPI and I2S Serial Interfaces
DFull Power-Down Control
D32-Pin 5y5 mm QFN Package
APPLICATIONS
DMP3 Players
DDigital Still Cameras
DDigital Video Camcorders
DESCRIPTION
The TLV320DAC26 is a high-performance audio DAC with
16/20/24/32-bit 97-dBA stereo playback.The audio output
drivers on the ’DAC26 are highly flexible, having
software-programmable low or high-power drive modes to
optimize system power dissipation. The outputs can be
configured to supply up to 330 mW into a bridge terminated
8- load, can support stereo 16- headphone amplifiers
in ac-coupled or capless output configurations, and can
supply a stereo line-level output
A programmable digital audio effects processor enables
bass, treble, midrange, or equalization playback
processing. The digital audio data format is programmable
to work with popular audio standard protocols (I2S, DSP,
Left/Right Justified) in master or slave mode, and also
includes an on-chip programmable PLL for flexible clock
generation capability. Highly configurable software power
control is provided, enabling stereo audio playback at 48
ksps at 11 mW with a 3.3-V analog supply level.
The ’DAC26 is available in a 32 lead QFN.
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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.
Copyright 2004, Texas Instruments Incorporated
SPI is a trademark of Motorola.
I2S is a trademark of Phillips Electronics.
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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.
PACKAGE/ORDERING INFORMATION
PRODUCT PACKAGE PACKAGE
DESIGNATOR OPERATING
TEMPERATURE RANGE ORDERING NUMBER TRANSPORT MEDIA,
QUANTITY
TLV320DAC26
QFN-32
RHB
−40°C to 85°C
TLV320DAC26IRHB Tubes, 74
TLV320DAC26
QFN-32
RHB
−40
°
C to 85
°
C
TLV320DAC26IRHBR Tape and Reel, 3000
PIN ASSIGNMENTS
QFN(TOP VIEW)
31 30 29 28 27
910
1
2
3
4
5
6
7
8
24
23
22
21
20
19
18
17
DRVDD
VGND
DRVSS
HPL
AVDD
NC
NC
NC
DVSS
IOVDD
MCLK
SCLK
MISO
MOSI
SS
NC
32 26
11 12 13 14 15
MICBIAS
MICIN
AUX
NC
AVSS
DVDD
BCLK
NC
DIN
PWD
LRCK
RESET
16
HPR
25
NC
NC
NC
DAC26
Terminal Functions
QFN
PIN NAME DESCRIPTION QFN
PIN NAME DESCRIPTION
1 DVSS Digital core and IO ground 17 NC No connect
2 IOVDD IO supply 18 NC No connect
3 MCLK Master clock 19 NC No connect
4 SCLK SPI serial clock input 20 AVDD Analog power supply
5 MISO SPI serial data output 21 HPL Left channel audio output
6 MOSI SPI serial data input 22 DRVSS Speaker ground
7 SS SPI slave select input 23 VGND Virtual ground for audio output
8 NC No connect 24 DRVDD Speaker /PLL supply
9 MICBIAS Microphone bias voltage 25 HPR Right channel audio output
10 MICIN Microphone input 26 RESET Device reset
11 AUX Auxiliary input 27 LRCK Audio DAC word-clock
12 NC No connect 28 PWD Hardware powerdown
13 NC No connect 29 DIN Audio data input
14 NC No connect 30 NC No connect
15 AVSS Analog ground 31 BCLK Audio bit−clock
16 NC No connect 32 DVDD Digital core supply
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ABSOLUTE MAXIMUM RATINGS
over o p e r ating free-air temperature range unless otherwise noted(1)(2)
UNITS
AVDD to AVSS −0.3 V to 3.9 V
DRVDD to DRVSS −0.3 V to 3.9 V
IOVDD to DVSS −0.3 V to 3.9 V
DVDD to DVSS −0.3 V to 2.5 V
AVDD to DRVDD −0.1 V to 0.1 V
AVSS to DRVSS to DVSS −0.1 V to 0.1 V
Analog inputs to AVSS −0.3 V to AVDD + 0.3 V
Digital input voltage to DVSS −0.3 V to IOVDD + 0.3 V
Operating temperature range −40°C to 85°C
Storage temperature range −65°C to 105°C
Junction temperature (TJ Max) 105°C
QFN package
Power dissipation (TJ Max − TA)/θJA
QFN package
θJA Thermal impedance 123°C/W
Lead temperature
Soldering vapor phase (60 sec) 215°C
Lead temperature
Infrared (15 sec) 220°C
(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.
(2) If the ’DAC26 is used to drive high power levels to an 8- load for extended intervals at ambient temperatures above 70°C, multiple vias should be
used t o electrically and thermally connect the thermal pad on the QFN package to an internal heat-dissipating ground plane on the user’s PCB.
ELECTRICAL CHARACTERISTICS
At +25°C, AVDD,DRVDD,IOVDD = 3.3 V, DVDD = 1.8 V, Int. Vref = 2.5 V, Fs (Audio) = 48 kHz, unless otherwise noted
PARAMETER TEST CONDITIONS MIN TYP MAX UNITS
MICROPHONE INPUT
Input resistance 20 k
Input capacitance 10 pF
MICROPHONE BIAS
Voltage D4 = 0 control register 05H/Page2 2.5 V
Voltage
D4 = 1 control register 05H/Page2 2.0 V
Sourcing current 4.7 mA
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ELECTRICAL CHARACTERISTICS
At +25°C, AVDD,DRVDD,IOVDD = 3.3 V, DVDD = 1.8 V, Int. Vref = 2.5 V, Fs (Audio) = 48 kHz, unless otherwise noted (continued)
PARAMETER TEST CONDITIONS MIN TYP MAX UNITS
DAC INTERPOLATION FIL TER
Pass band 20 0.45 Fs Hz
Pass band ripple ±0.06 dB
Transition band 0.45 Fs 0.5501 Fs Hz
Stop band 0.5501 Fs 7.455 Fs Hz
Stop band attenuation 65 dB
Filter group delay 21/Fs sec
De−emphasis error ±0.1 dB
DAC LINE OUTPUT 1-kHz sine wave input, 48 ksps, output drivers
in low power mode, load = 10 k, 10 pF
Full scale output voltage (0 dB) By design, D10−D9 = 00 in control register
06H/Page2 corresponding to 2-VPP output
swing
0.707 Vrms
Output common mode By design, D10−D9 = 00 in control register
06H/Page2 corresponding to 2-VPP output
swing
1.35 V
SNR Measured as idle channel noise, A-weighted 85 97 dBA
THD 0-dB FS input, 0-dB gain −95 dB
PSRR 1 kHz, 100 mVpp on AVDD(2) VGND powered
down 56 dB
DAC HEADPHONE OUTPUT 1-kHz sine wave input, 48 ksps, output drivers
in high power mode, load = 16 , 10 pF
Full scale output voltage (0 dB) By design, D10−D9 = 00 in control register
06H/Page2 corresponding to 2-VPP output
swing
0.707 Vrms
SNR Measured as idle channel noise, A-weighted 85 97 dBA
THD −1 dB FS input, 0-dB gain −91 −55 dB
PSRR 1 kHz, 100 mVpp on AVDD(1) VGND powered
down 54 dB
Mute attenuation 121 dB
Maximum output power D10−D9 = 00 in control register 06H/Page2 30 mW
Digital volume control gain −63.5 0 dB
Digital volume control step size 0.5 dB
Channel separation Between HPL and HPR 80 dB
DAC SPEAKER OUTPUT Output driver in high power mode,
load = 8 ,, connected between HPR and HPL
pins. D10−D9 = 10 in control register
06H/Page2 corresponding to 2.402-VPP output
swing
Output power 0 dB input to DAC 325 mW
SNR Measured as idle channel noise, A-weighted 102 dBA
THD −1 dB FS input, 0-dB gain −86 dB
THD
−6 dB FS input, 0-dB gain −88 dB
(1) DAC PSRR measurement is calculated as:
PSRR +20 log10ǒVSIGsup
VHPRńLǓ
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ELECTRICAL CHARACTERISTICS
At +25°C, AVDD,DRVDD,IOVDD = 3.3 V, DVDD = 1.8 V, Int. Vref = 2.5 V, Fs (Audio) = 48 kHz, unless otherwise noted (continued)
PARAMETER TEST CONDITIONS MIN TYP MAX UNITS
VOLTAGE REFERENCE
Voltage range
VREF output programmed as 2.5 V 2.3 2.5 2.7
V
Voltage range
VREF output programmed as 1.25 V 1.15 1.25 1.35
V
Voltage range External VREF. By design, not tested in
production. 1.2 2.55 V
Reference drift Internal VREF = 1.25 V 29 ppm/°C
Current drain Extra current drawn when the internal
reference is turned on. 650 µA
DIGITAL INPUT / OUTPUT(1)
Internal clock frequency 8.8 MHz
Logic family CMOS
Logic level: VIH IIH = +5 µA 0.7xIOVDD V
VIL IIL = +5 µA −0.3 0.3xIOVDD V
VOH IOH = 2 TTL loads 0.8xIOVDD V
VOL IOL = 2 TTL loads 0.1xIOVDD V
Capacitive load 10 pF
POWER SUPPLY REQUIREMENTS
Power supply voltage
AVDD(2) 2.7 3.6 V
DRVDD(2) 2.7 3.6 V
IOVDD 1.1 3.6 V
DVDD 1.525 1.95 V
IAVDD
2.2
Stereo audio playback IDRVDD
power mode, VGND off, PLL
0mA
Stereo audio playback
IDVDD
off 2.4
mA
IAVDD
0.1
PLL IDRVDD Additional power consumed
1.3 mA
PLL
IDVDD
0.9
mA
IAVDD
0.3
VGND IDRVDD Additional power consumed
0.9 mA
VGND
IDVDD
0
mA
Hardware power down All currents 2µA
(1) Internal oscillator is designed to give nominally 8-MHz clock frequency. However , due to process variations, this frequency can vary from device
to device. All calculations for delays and wait times in the data sheet assume an 8-MHz oscillator clock.
(2) It is recommended that AVDD and DRVDD be set to the same voltage for the best performance. It is also recommended that these supplies be
separated on the user’s PCB.
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FUNCTIONAL BLOCK DIAGRAM
Control
Interface
MCLK
HPL
M
U
X
DAC
High Power
Low Power
DAC
High Power
Low Power
2.0 V / 2.5 V
+
0 to -63.5 dB
(0.5 dB steps)
Digital
Bass,
Midrange,
Treble,
EQ
Processing
Digital
Filters
Programmable
PLL
Digital Audio
Serial
Interface
SPI Serial
Interface and
Data
Processing
OSC
DRVDD DRVSS AVDD AVSS DVDD DVSS IOVDD
HPR
VGND
MICBIAS
MICIN
AUX
PWD
LRCK
BCLK
DIN
SS
SCLK
MOSI
MISO
RESET
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SPI TIMING DIAGRAM
ttd
ta
tsck
ten1 ten2
tw(swk)
tw(swk) tr
tf
tvth2 tdis
th1
tsu
SS
SCLK
MISO
MOSI
MSB OUT BIT . . . 1 LSB OUT
MSB OUT BIT . . . 1 LSB OUT
TYPICAL TIMING REQUIREMENTS
All specifications at 25°C, DVDD = 1.8 V (1)
PARAMETER
IOVDD = 1.1 V IOVDD = 3.3 V
UNITS
PARAMETER
MIN MAX MIN MAX
UNITS
tw(sck) SCLK pulse width 27 18 ns
ten1 Enable lead time 18 15 ns
ten2 Enable lag time 18 15 ns
tdSequential transfer delay time 18 15 ns
taSlave MISO access time 18 15 ns
tdis Slave MISO disable time 18 15 ns
tsu MOSI data setup time 6 6 ns
th1 MOSI data hold time 6 6 ns
th2 MISO data hold time 4 4 ns
tvMISO data valid time 22 13 ns
trRise time 6 4 ns
tfFall time 6 4 ns
(1) These parameters are based on characterization and are not tested in production.
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AUDIO INTERFACE TIMING DIAGRAMS
LRCK/ADWS
BCLK
DIN
td(WS)
ts(DI) th(DI)
Figure 1. I2S/LJF/RJF Timing in Master Mode
TYPICAL TIMING REQUIREMENTS (FIGURE 1)
All specifications at 25°C, DVDD = 1.8 V (1)
PARAMETER
IOVDD = 1.1 V IOVDD = 3.3 V
UNITS
PARAMETER
MIN MAX MIN MAX
UNITS
td(WS) LRCK delay time 25 15 ns
ts(DI) DIN setup time 6 6 ns
th(DI) DIN hold time 6 6 ns
trRise time 10 6 ns
tfFall time 10 6 ns
(1) These parameters are based on characterization and are not tested in production.
LRCK/ADWS
BCLK
DIN
td(WS)
ts(DI) th(DI)
td(WS)
Figure 2. DSP Timing in Master Mode
TYPICAL TIMING REQUIREMENTS (FIGURE 2)
All specifications at 25°C, DVDD = 1.8 V(1)
PARAMETER
IOVDD = 1.1 V IOVDD = 3.3 V
UNITS
PARAMETER
MIN MAX MIN MAX
UNITS
td(WS) LRCK delay time 25 15 ns
ts(DI) DIN setup time 6 6 ns
th(DI) DIN hold time 6 6 ns
trRise time 10 6 ns
tfFall time 10 6 ns
(1) These parameters are based on characterization and are not tested in production.
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LRCK/ADWS
BCLK
DIN
tL(BCLK)
ts(DI) th(DI)
th(WS) tS(WS)
tH(BCLK)
tP(BCLK)
Figure 3. I2S/LJF/RJF Timing in Slave Mode
TYPICAL TIMING REQUIREMENTS (FIGURE 3)
All specifications at 25°C, DVDD = 1.8 V (1)
PARAMETER
IOVDD = 1.1 V IOVDD = 3.3 V
UNITS
PARAMETER
MIN MAX MIN MAX
UNITS
tH(BCLK) BCLK high period time 35 35 ns
tL(BCLK) BCLK low period time 35 35 ns
ts(WS) LRCK setup time 6 6 ns
th(WS) LRCK hold time 6 6 ns
ts(DI) DIN setup time 6 6 ns
th(DI) DIN hold time 6 6 ns
trRise time 5 4 ns
tfFall time 5 4 ns
(1) These parameters are based on characterization and are not tested in production.
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LRCK/ADWS
BCLK
DIN
tL(BCLK)
th(DI)
tS(WS)
tH(BCLK) th(WS)
tP(BCLK)
th(WS) tS(WS)
Figure 4. DSP Timing in Slave Mode
TYPICAL TIMING REQUIREMENTS (FIGURE 4)
All specifications at 25°C, DVDD = 1.8 V (1)
PARAMETER
IOVDD = 1.1 V IOVDD = 3.3 V
UNITS
PARAMETER
MIN MAX MIN MAX
UNITS
tH(BCLK) BCLK high period 35 35 ns
tL(BCLK) BCLK low period 35 35 ns
ts(WS) LRCK setup time 6 6 ns
th(WS) LRCK hold time 6 6 ns
ts(DI) DIN setup time 6 6 ns
th(DI) DIN hold time 6 6 ns
trRise time 5 4 ns
tfFall time 5 4 ns
(1) These parameters are based on characterization and are not tested in production.
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TYPICAL CHARACTERISTICS
−160
−140
−120
−100
−80
−60
−40
−20
0
dB
0 5000 10000 15000 20000
Hz
Figure 5. DAC FFT Plot (TA = 25°C, 48 ksps, 0 dB, 1 kHz Input, AVDD = 3.3 V, RL = 10 k)
−150
−130
−110
−90
−70
−50
−30
−10
0
dB
0 5000 10000 15000 20000
Hz
Figure 6. DAC FFT Plot (TA = 25°C, 48 ksps, −1 dB, 1 kHz Input, AVDD = DRVDD = 3.3 V, DVDD = 1.8 V, RL
= 16 )
−94
−92
−90
−88
515 2535
THD − Total Harmonic Distortion − dB
Output Power − mW
Figure 7. High Power Output Driver THD vs Output Power
(TA =25°C, AVDD, DRVDD = 3.3 V, RL = 16 W)
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OVERVIEW
The TLV320DAC26 is a highly integrated stereo audio DAC for portable computing, communication, and entertainment
applications. The ’DAC26 has a register-based architecture where all functions are controlled through the registers and
onboard state machines.
The ’DAC26 consists of the following blocks (refer to the block diagram):
DAudio DAC
DAuxiliary Inputs for analog pass through functionality
Audio data is transferred between the host DSP/µP via a standard 4-wire interface and supports a variety of modes (i.e.,
I2S, DSP, etc).
Control of the ’DAC26 and its functions is accomplished by writing to different registers in the ’DAC26. A simple command
protocol is used to address the 16-bit registers. Registers control the operation of the audio DAC. The control and auxiliary
functions are accessed via a SPI bus.
A typical application of the ’DAC26 is shown in Figure 8.
MCLK
PWD
DOUT
LRCK
DIN
BCLK
MISO
MOSI
SS
SCLK
2.2 k W
Audio
Auxiliary Input AUX
MICBIAS
MICIN
HPR
VGND
HPL
8W
Speaker
I2S Interface
SPI Interface
Master Clock Input
ADC Word Select
Serial Output to CPU/DSP
DAC Word Select
Serial Input From CPU/DSP
Serial Clock Input
Serial Output to SPI Master
Serial Input From SPI Master
SPI Slave Select Input
SPI Serial Clock Input
Figure 8. Typical Circuit Configuration
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OPERATION−AUDIO DAC
Audio Analog I/O
The ’DAC26 has one mono audio input (MICIN) typically used for microphone recording, and an auxiliary input (AUX) that
can be used as a second microphone or line input. The ’DAC26 has an analog pass through mode where by the input from
the Mic and AUX input can be routed to any one of the analog output drivers. The dual audio output drivers have
programmable power level and can be configured to drive up to 325 mW into an 8- speaker, or to drive 16- stereo
headphones at over 30-mW per channel, or to provide a stereo line-level output. The power level of the output drivers is
controlled using bit D12 in control register REG−05H/Page2. The ’DAC26 also has a virtual ground (VGND) output driver,
which can optionally be used to connect the return terminal of headphones, to eliminate the ac-coupling capacitors needed
at the headphone output. The VGND amplifier is controlled by bit D8 of REG−05H/Page2. A special circuit has also been
included in the ’DAC26 to insert a short keyclick sound into the stereo audio output, even when the audio DAC is powered
down. The keyclick sound is used to provide feedback to the user when a particular button is pressed or item is selected.
The specific sound of the keyclick can be adjusted by varying several register bits that control its frequency, duration, and
amplitude.
Audio Digital Interface
Digital audio data samples are transmitted between the ’DAC26 and the audio processor via the serial bus (BCLK, LRCK,
DIN) that can be configured to transfer digital data in four different formats: right justified, left justified, I2S, and DSP. The
four modes are MSB-first and operate with variable word length of 16, 20, 24, or 32 bits. The digital audio serial bus of the
’DAC26 can operate in master or slave mode, depending on its register settings. The word-select signal (LRCK) and bit
clock signal (BCLK) are configured as outputs when the bus is in master mode. They are configured as inputs when the
bus is in slave mode. The LRCK is representative of the audio DAC sampling rate and is synchronized with DIN.
DDAC SAMPLING RATE
The Audio Control 1 register (Register 00H, Page2) determines the sampling rates of the audio DAC, which is scaled
down from a reference rate (Fsref). When the audio DAC is powered up, it is configured by default as an I2S slave with
the DAC operating at Fsref.
DWORD SELECT SIGNALS
The word select signal (LRCK) indicates the channel being transmitted:
LRCK = 0: left channel for I2S mode
LRCK = 1: right channel for I2S mode
For other modes see the timing diagrams below.
Bitclock (BCLK) Signal
In addition to flexibility as master or slave mode, the BCLK can also be configured in two transfer modes—256−S and
Continuous Transfer Modes. These modes are set using bit D12/REG−06h/Page2.
D256−S TRANSFER MODE
In the 256−S mode, the BCLK rate always equals 256 times the maximum of the LRCK frequencies. In
the 256−S mode, the DAC sampling rate equal to Fsref (as selected by bit D5−D0/REG−00h/Page2) and left−justified
mode is not supported.
DCONTINUOUS TRANSFER MODE
In the continuous transfer mode, the BCLK rate always equals two times the word length of the maximum of the LRCK
frequencies.
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DRIGHT-JUSTIFIED MODE
In right-justified mode, the LSB of the left channel is valid on the rising edge of the BCLK preceding the falling edge of
LRCK. Similarly, the LSB of the right channel is valid on the rising edge of the BCLK preceding the rising edge of LRCK.
BCLK
LRCK
DIN/ n n−1 1 00 n n−1 1 0
LSBMSB
n−2 2 2n−2
1/fs
Left Channel Right Channel
Figure 9. Timing Diagram for Right-Justified Mode
DLEFT-JUSTIFIED MODE
In left−justified mode, the MSB of the right channel is valid on the rising edge of the BCLK, following the falling edge of
ADWS or LRCK. Similarly the MSB of the left channel is valid on the rising edge of the BCLK following the rising edge of
ADWS or LRCK.
BCLK
LRCK
DIN n n−1 1 0 n n−1 1 0
LSBMSB
n n−1n−2 2 n−2 2
1/fs
Left Channel Right Channel
Figure 10. Timing Diagram for Left-Justified Mode
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DI2S MODE
In I 2S mode, the MSB of the left channel is valid on the second rising edge of the BCLK after the falling edge of ADWS or
LRCK. Similarly the MSB of the right channel is valid on the second rising edge of the BCLK after the rising edge of
LRCK.
BCLK
LRCK
DIN n n−1 1 0 n n−1 1 0
LSBMSB
n
1 clock before MSB
n−2 2 n−2 2
1/fs
Left Channel Right Channel
Figure 11. Timing Diagram for I2S Mode
DDSP MODE
In DSP mode, the falling edge of LRCK starts the data transfer with the left channel data first and immediately followed
by the right channel data. Each data bit is valid on the falling edge of BCLK.
BCLK
LRCK
DIN/ n n−1 1 0 n n−1 1 0
LSBMSB
n n−11 0
MSB LSB
n−2 2 n−2 2 n−2
MSBLSB
1/fs
Left Channel Right Channel
Figure 12. Timing Diagram for DSP Mode
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AUDIO DATA CONVERTERS
The ’DAC26 has a stereo audio DAC. The DAC can operate with a maximum sampling rate of 53 kHz and support all audio
standard rates of 8 kHz, 11.025 kHz, 12 kHz, 16 kHz, 22.05 kHz, 24 kHz, 32 kHz, 44.1 kHz, and 48 kHz. By utilizing the
flexible clock generation capability and internal programmable interpolation, a wide variety of sampling rates up to 53 kHz
can be obtained from many possible MCLK inputs.
When the DAC is operating, the ’DAC26 requires an applied audio MCLK input. The user should also set
bit D13/REG−06H/Page2 to indicate which Fsref rate is being used.
Typical audio DACs can suffer from poor out-of-band noise performance when operated at low sampling rates, such as
8 kHz or 11.025 kHz. The ’DAC26 includes programmable interpolation circuitry to provide improved audio performance
at such low sampling rates, by first upsampling low-rate data to a higher rate, filtering to reduce audible images, and then
passing the data to the internal DAC, which is actually operating at the Fsref rate. This programmable interpolation is
determined using bit D5−D3/REG−00H/Page2.
For example, if playback of 11.025-kHz data is required, the ’DAC26 can be configured such that Fsref = 44.1 kHz. Then
using bit D5−D3/REG−00H/Page2, the DAC sampling rate (Fs) can be set to Fsref/4, or Fs = 11.025 kHz. In operation, the
11.025-kHz digital input data is received by the ’DAC26, upsampled to 44.1 kHz, and filtered for images. It is then provided
to the audio DAC operating at 44.1 kHz for playback. In reality, the audio DAC further upsamples the 44.1 kHz data by a
ratio of 128x and performs extensive interpolation filtering and processing on this data before conversion to a stereo analog
output signal.
PLL
The ’DAC26 has an on-chip PLL to generate the needed internal DAC operational clocks from a wide variety of clocks
available in the system. The PLL supports an MCLK varying from 2 MHz to 50 MHz and is register programmable to enable
generation of required sampling rates with fine precision.
DAC sampling rates are given by
DAC_FS = Fsref/N1
where, Fsref must fall between 39 kHz and 53 kHz, and N1, N2 =1, 1.5, 2, 3, 4, 5, 5.5, 6 are register programmable.
The PLL can be enabled or disabled using register programming.
DWhen PLL is disabled
Fsref +MCLK
128 Q
Q = 2, 317
In this mode, the MCLK can operate up to 50 MHz, and Fsref should fall within 39 kHz to 53 kHz.
DWhen PLL is enabled
Fsref +MCLK K
2048 P
P = 1, 2, 3, , 8
K = J.D
J = 1, 2, 3, .,64
D = 0, 1, 2, , 9999
P, J, and D are register programmable, where J is an integer part of K before the decimal point, and D is a four-digit fractional
part of K after the decimal point, including lagging zeros.
Examples: If K = 8.5, Then J = 8, D = 5000
If K = 7.12, Then J = 7, D = 1200
If K = 7.012, Then J = 7, D = 120
The PLL is programmed through Registers 1BH and 1CH of Page2.
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DWhen PLL is enabled and D = 0, the following condition must be satisfied
80 MHz vMCLK K
Pv110 MHz
2MHzvMCLK
Pv20 MHz
4vJv55
DWhen PLL is enabled and D 0, the following condition must be satisfied
80 MHz vMCLK K
Pv110 MHz
10 MHz vMCLK
Pv20 MHz
4vJv11
Example 1:
For MCLK = 12 MHz and Fsref = 44.1 kHz
P = 1, K = 7.5264 J = 7, D = 5264
Example 2:
For MCLK = 12 MHz and Fsref = 48.0 kHz
P = 1, K = 8.192 J = 8, D = 1920
STEREO AUDIO DAC
Each channel of the stereo audio DAC consists of a digital audio processing block, a digital interpolation filter, digital
delta-sigma modulator, and an analog reconstruction filter. The DAC is designed to provide enhanced performance at low
sample rates through increased oversampling and image filtering, thereby keeping quantization noise generated within the
delta-sigma modulator and signal images strongly suppressed within the audio band to beyond 20 kHz. This is realized
by keeping the upsampled rate constant at 128 x Fsref and changing the oversampling ratio as the input sample rate is
changed. For Fsref of 48 kHz, the digital delta-sigma modulator always operates at a rate of 6.144 MHz. This ensures that
quantization noise generated within the delta-sigma modulator stays low within the frequency band below 20 kHz at all
sample rates. Similarly, for Fsref rate of 44.1 kHz, the digital delta-sigma modulator always operates at a rate of 5.6448
MHz.
Digital Audio Processing
The DAC channel consists of optional filters for de-emphasis and bass, treble, midrange level adjustment, or speaker
equalization. The de-emphasis function is only available for sample rates of 32 kHz, 44.1 kHz, and 48 kHz. The transfer
function consists of a pole with time constant of 50 µs and a zero with time constant of 15 µs. Frequency response plots
are given in the Audio DAC Filter Frequency Responses section of this data sheet. The de-emphasis filter can be enabled
or bypassed depending on bit D0 of register 05H/Page2.
The DAC digital effects processing block also includes a fourth order digital IIR filter with programmable coefficients (one
set per channel). The filter is implemented as cascade of two biquad sections with frequency response given by:
ǒN0 )2 N1 z*1)N2 z*2
32768 *2 D1 z*1*D2 z*2Ǔǒ N3 )2 N4 z*1)N5 z*2
32768 *2 D4 z*1*D5 z*2Ǔ
The N and D coefficients are fully programmable, and the entire filter can be enabled or bypassed depending on bit D1 of
register 05H/Page2. The coef ficients for this filter implement a variety of sound effects, with bass-boost or treble boost being
the most commonly used in portable audio applications. The default N and D coefficients in the part are given by:
N0 = N3 = 27619 D1 = D4 = 32131
N1 = N4 = −27034 D2 = D5 = −31506
N2 = N5 = 26461
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and implement a shelving filter with 0 dB gain from dc to approximately 150 Hz, at which point it rolls off to a 3-dB attenuation
for higher frequency signals, thus giving a 3-dB boost to signals below 150 Hz. The N and D coefficients are represented
by 16-bit twos complement numbers with values ranging from –32768 to +32767. Frequency response plots are given in
the Audio DAC Filter Frequency Responses section of this data sheet.
Interpolation Filter
The interpolation filter upsamples the output of the digital audio processing block by the required oversampling ratio. It
provides a linear phase output with a group delay of 21/Fs.
In addition, a digital interpolation filter provides enhanced image filtering and reduces signal images caused by the
upsampling process that are below 20 kHz. For example, upsampling an 8-kHz signal produces signal images at multiples
of 8 kHz (i.e., 8 kHz, 16 kHz, 24 kHz, etc). The images at 8 kHz and 16 kHz are below 20 kHz and still audible to the listener;
therefore, they must be filtered heavily to maintain good output quality. The interpolation filter is designed to maintain at
least 65-dB rejection of images that land below 7.455 Fs. In order to utilize the programmable interpolation capability, the
Fsref should be programmed to a higher rate (restricted to be in the range of 39 kHz to 53 kHz when the PLL is in use),
and the actual Fs is set using the dividers in bit D5−D3/REG−00H/Page2. For example, if Fs = 8 kHz is required, then Fsref
can be set to 48 kHz, and the DAC Fs set to Fsref/6. This ensures that all images of the 8-kHz data are sufficiently attenuated
well beyond the ~20-kHz audible frequency range.
Delta-Sigma DAC
The audio digital-to-analog converter incorporates a third order multibit delta-sigma modulator followed by an analog
reconstruction filter. The DAC provides high-resolution, low-noise performance, using oversampling and noise shaping
techniques. The analog reconstruction filter design consists of a 6 tap analog FIR filter followed by a continuous time RC
filter. The analog FIR operates at a rate of 128 x Fsref (6.144 MHz when Fsref = 48 kHz, 5.6448 MHz when Fsref = 44.1 kHz).
Note that the DAC analog performance may be degraded by excessive clock jitter on the MCLK input. Therefore, care must
be taken to keep jitter on this clock to a minimum.
DAC Digital Volume Control
The DAC has a digital volume control block, which implements programmable gain. The volume level can be varied from
0 dB to –63.5 dB in 0.5 dB steps. In addition, there is an independent mute bit for each channel. The volume level of both
channels can also be changed simultaneously by the master volume control. The gain is implemented with a soft-stepping
algorithm, which only changes the actual volume by one step 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 through bit D1 of control register
04H/Page2.
Because of soft-stepping, the host does not know when the DAC has been actually muted. This may be important if the
host wishes to mute the DAC before making a significant change, such as changing sample rates. In order to help with this
situation, the ’DAC26 provides a flag back to the host via a read-only register bit (D2−D3 of control register 04H/Page2)
that 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 programming D14=1 in register 1DH in Page02. If soft-stepping is
enabled, the MCLK signal to the device should not be changed until the DAC power-down flag is set. When this flag is set,
the internal soft-stepping process and power-down sequence is complete, and the MCLK can be stopped if desired.
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The ’DAC26 also includes functionality to detect when the user switches are on or off the de-emphasis or digital audio
processing functions, to first (1) soft-mute the DAC volume control, (2) change the operation of the digital effects
processing, and (3) soft-unmute the part. This avoids any possible pop/clicks in the audio output due to instantaneous
changes in the filtering. A similar algorithm is used when first powering up or down the DAC. The circuit begins operation
at power up with the volume control muted, then soft-steps it up to the desired volume level. At power down, the logic first
soft-steps the volume down to a mute level, then powers down the circuitry.
DAC Power Down
The DAC power-down flag ( D6 of REG05H/Page2) along with D10 of REG05H/Page2 denotes the power-down status
of the DAC according to Table 1. Table 1. DAC Powerdown Status
[D10,D6] POWERUP / DOWN STATE OF DAC
[0,0] DAC is in stable power-up state
[0,1] DAC is in the process of powering up. The length of this state is determined by PLL and output driver
power-up delays controlled by register programming.
[1,0] DAC is in the process of powering down. The length of this state is determined by soft-stepping of volume
control block and DAC pop reduction sequencing controlled by register programming.
[1,1] DAC is in a stable power-down state.
AUDIO OUTPUT DRIVERS
The ’DAC26 features audio output drivers which can be configured in either low power mode or high power mode depending
on the load and output power required. By default, at reset the output drivers are configured in low power mode. In this mode,
the output drivers can drive a full-scale line-level signal into loads of 10 k minimum or drive moderate amplitude signals
into loads of 16 minimum.
The output drivers can also be configured in high power mode by setting bit D12 of Reg05H/Page2 to 1. In this mode, each
output driver can deliver up to 30 mW per channel into a headphone speaker load of 16 . The headphones can be
connected in a single-ended configuration using ac-coupling capacitors, or the capacitors can be removed and virtual
ground (VGND) powered for a capless output connection. The typical headphone jack configuration for these two modes
is shown in Figure 15. Note that the VGND amplifier must be powered if the capless configuration is used.
In the case of an ac-coupled output, the value of the capacitors is typically chosen based on the amount of low-frequency
cut that can be tolerated. The capacitor in series with the load impedance forms a high-pass filter with −3 dB cutof f frequency
of 1/(2πRC) in Hz, where R is the impedance of the headphones. Use of an overly small capacitor reduces low-frequency
components in the signal output and leads to low-quality audio. When driving 16- headphones, capacitors of 220-µF (a
commonly used value) result in a high-pass filter cutoff frequency of 45 Hz, although reducing these capacitors to 50 µF
results in a cutof f frequency of 199 Hz, which is generally considered noticeable when playing music. The cutoff frequency
is reduced to half of the above values if 32- headphones are used instead of 16 .
The ’DAC26 programmable digital effects block can be used to help reduce the size of capacitors needed by implementing
a low frequency boost function to help compensate for the high-pass filter introduced by the ac-coupling capacitors. For
example, by using 50-µF capacitors and setting the ’DAC26 programmable filter coefficients as shown below, the frequency
response can be improved as shown in Figure 14.
Filter coefficients (use the same for both channels):
N0 = 32767, N1 = −32346, N2 = 31925, N3 = 32767, N4 = 0, N5 = 0
D0 = 32738, D1 = −32708 D4 = 0, D5 = 0
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−10
−12
−16
−20 0 100 200 300 400 500 600
Gain − dB
−6
−4
f − Frequency − Hz
0
700 800 900 1 k
−18
−14
−8
−2
Figure 13. Uncompensated Response For 16-W Load and 50-mF Decoupling Capacitor
−10
−15
−20 0 100 200 300 400 500 600
Gain − dB
−5
0
f − Frequency − Hz700 800 900 1 k
Figure 14. Frequency Response For 16-W Load and 50-mF Decoupling Capacitor After Gain
Compensation Using a Suggested Set of Coefficients for Audio Effects Filter
Using the capless output configuration eliminates the need for these capacitors and removes the accompanying high-pass
filter entirely. However, this configuration does have one drawback – if the RETURN terminal of the headphone jack (which
is wired to the ’DAC26 VGND pin) is ever connected to a ground, that is shorted to the ’DAC26 ground pin, then the VGND
amplifier enters short-circuit protection, and the audio output does not function properly.
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’DAC26
HPR
HPL
VGND Headphone Jack
’DAC26
HPR
HPL
VGND Headphone Jack
Figure 15. Headphone Configurations, AC-Coupled (left) and Capless (right)
The audio output drivers in high power mode can also be configured to drive a mono differential signal into a speaker load
of 8- minimum. The speaker load should be connected dif ferentially between the HPR and HPL outputs. Several options
are possible for playback of DAC data in this case. If a stereo digital signal is available, this signal can be sent in normal
stereo fashion to the audio DAC. The programmable digital effects filters can then be used to invert one channel, so that
the signal applied across the speaker load is (LEFT + RIGHT), or effectively a mono-mix of the two channels. A simple
example of how to implement this inversion using the programmable filters is to set the coefficients as follows:
Left−channel coefficients: N0=32767, N1=0, N2=0, N3=32767, N4=0, N5=0
D1=0, D2=0, D4=0, D5=0
Right−channel coefficients: N0=−32767, N1=0, N2=0, N3=32767, N4=0, N5=0
D1=0, D2=0, D4=0, D5=0
This provides no spectral shaping; it only inverts the right channel relative to the left channel, such that the signals at HPL
and HPR are (LEFT) and (−RIGHT), with the signal across the speaker then being LEFT+ RIGHT. In a general case when
spectral shaping is also desired, the inversion can be accomplished simply by setting N0, N1, and N2 coefficients of one
channel to the negative of the values set for the other channel. Note that the programmable filtering must be enabled by
setting bit D1/REG−05H/Page2 to 1.
To enable the output drivers to deliver higher output power, the DAC output swing should be set to its highest level by setting
bit D10−D9/REG−06H/Page2 to 11. It is possible to increase power even further by disabling the built-in short-circuit
protection by programming bit D8 of Reg1DH/Page2 to 1. In this case care must be taken so a short-circuit at the output
does not occur. Figure 16 shows a typical jack configuration using a capless output configuration. In this configuration, the
’DAC26 drives the loudspeaker whenever headphones are not inserted in the jack and drives the headphones whenever
it is inserted in the jack.
’DAC26
HPR
HPL
VGND Headphone Jack
Loud Speaker
Figure 16. Speaker Connection
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−100
−90
−80
−70
−60
−50
−40
−30
−20
−10
0
0 50 100 150 200 250 300 350
2 V PP
THD − Total Harmonic Distortion − dB
PO − Output Power − mW
2.402 VPP
Figure 17. THD vs Output Power Delivered to an an 8-W Load (255C, AVDD = DRVDD = 3.3 V, DVDD = 1.8
V, DAC Output Swing Set to 2 V and 2.4V, and Short-Circuit Protection Disabled)
−100
−90
−80
−70
−60
−50
−40
−30
−20
−10
0
2.7 2.8 2.9 3 3.1 3.2 3.3 3.4 3.5 3.6
AVDD, DRVDD − V
THD − Total Harmonic Distortion − dB
Figure 18. THD vs AVDD, DRVDD Supply Voltage (255C When Driving a −1 dB, 1-kHz Sinewave From the
DAC Into an 8-W Load, with DAC Output Swing Set to 2.4 V, and Short-Circuit Protection Disabled)
The ’DAC26 incorporates a programmable short-circuit detection/protection function with different modes of operation.
During the insertion or removal of a headphone plug from the jack, the output pins of the drivers may be accidentally shorted,
causing the part to potentially draw a huge current, which may cause the power supply voltages to dip. Bits D8−D7 of
REG−1DH/Page2 control how the short-circuit detection/protection operates in the ’DAC26. One option is to fully disable
short-circuit protection, which also enables the audio output drivers to deliver more power to a low-impedance load (such
as an 8- speaker). However, care must be taken to prevent any short-circuit from occurring while the part is in this mode.
A second programmable configuration enables current-limiting in the audio output drivers, so that excessive currents
cannot be provided if the outputs are shorted. It also enables the internal short-circuit detection function, which can detect
excess current being drawn from the drivers and set a short-circuit detect flag (Page2, REG−1DH, bit D6). This flag can
be read by the user to power down the drivers if desired. This flag is cleared only if the short-circuit condition is removed.
If the user does not monitor this flag and powers down the drivers when a short-circuit occurs, the current-limiting prevents
excessive currents from being drawn, but power dissipation is higher due to this limited current flowing through the short.
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In a third programmable configuration, the ’DAC26 can be programmed to monitor and automatically power down the audio
output drivers upon detection of a short-circuit condition (Page2, REG−1DH, bit D7), in addition to setting the short-circuit
flag in Page2, REG−1DH, bit−D6. When the device has detected a short and resulted in this condition, the short-circuit flag
is cleared when all the routings to the speaker driver are disabled (i.e., DAC, Analog Mixer , and Keyclick blocks are powered
down by user).
AUDIO OUTPUT DRIVER POWER-ON POP REDUCTION SCHEME
The ’DAC26 implements a pop reduction scheme to reduce audible artifacts during power up and power down of the audio
output drivers. This scheme can be controlled by programming bits D2 and D1 of REG1EH/Page2. By default, the driver
pop reduction scheme is enabled and can be disabled by programming bit D2 of Reg1EH/Page2 to 1. When this scheme
is enabled and the virtual ground connection is not used (VGND amplifier is powered down), the audio output driver slowly
charges up any external ac-coupling capacitors to reduce audible artifacts. Bit D1 of REG1EH/Page2 provides control of
the charging time for the ac-coupling capacitor as either 0.8 sec or 4 sec. When the virtual ground amplifier is powered up
and used, the external ac-coupling capacitor is eliminated, and the power up time becomes 1 ms. This scheme takes ef fect
whenever the audio output drivers are powered up due to enabling any of the DAC, the Analog Mixer, or the Keyclick
Generator.
Pop Reduction for DAC Routing
Whenever the audio DAC is powered on or off, a slight change in the output d c o ffset voltage may occur and can be heard
as a weak pop in the output. In order to reduce this artifact, the ’DAC26 implements a DAC pop reduction scheme, which
is programmable using bits D5−D2 in REG−1DH/Page2. Bit D5 enables the scheme, which implements a slow transition
between the starting dc level and the final dc level. For best results, program bits D4−D2 in REG1DH/Page2 to 100.
AUDIO MIXING
Analog Mixer
The analog mixer can be used to route the analog input selected (MICIN or AUX) through an analog volume control and
then mix it with the audio DAC output. The analog mixer feature is available only if single-ended MICIN or AUX is selected
as the input. This feature is available even if the DAC is powered down. The analog volume control in this path has a gain
range from 12 dB to –34.5 dB in 0.5-dB steps plus mute and includes soft-stepping logic. The internal oscillator is used
for soft-stepping whenever the DAC is powered down.
KEYCLICK
A special circuit has been included for inserting a square−wave signal into the analog output signal path based on register
control. This functionality is intended for generating keyclick sounds for user feedback. Register 04H/Page2 contains bits
that control the amplitude, frequency, and duration of the square-wave signal. The frequency of the signal can be varied
from 62.5 Hz to 8 kHz and its duration can be programmed from 2 periods to 32 periods. Whenever this register is written,
the square-wave is generated and coupled into the audio output, going to both audio outputs. The keyclick enable bit D15
of control register 04H/Page2 is reset after the duration of keyclick is played out. This capability is available even when
the DAC is powered down.
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SPI DIGITAL INTERFACE
All ’DAC26 control registers are programmed through a standard SPI bus. The SPI allows full-duplex, synchronous, serial
communication between a host processor (the master) and peripheral devices (slaves). The SPI master generates the
synchronizing clock and initiates transmissions. The SPI slave devices depend on a master to start and synchronize
transmissions.
A transmission begins when initiated by a master SPI. The byte from the master SPI begins shifting in on the slave SPIDIN
(MOSI) pin under the control of the master serial clock. As the byte shifts in on the SPIDIN pin, a byte shifts out on the
SPIDOUT (MISO) pin to the master shift register.
The idle state of the serial clock for the ’DAC26 is low, which corresponds to a clock polarity setting of 0 (typical
microprocessor SPI control bit CPOL = 0). The ’DAC26 interface is designed so that with a clock phase bit setting of 1
(typical microprocessor SPI control bit CPHA = 1), the master begins driving its MOSI pin and the slave begins driving its
SPIDOUT pin on the first serial clock edge. The SS pin can remain low between transmissions; however, the ’DAC26 only
interprets command words which are transmitted after the falling edge of SS.
Hardware Reset
The device requires a low-to-high pulse on RESET after power up for correct operation. A hardware reset pulse initializes
all the internal registers, counters, and logic.
Hardware Power Down
By default the PWD pin is configured as a hardware power-down (active low) signal. The device powers down all the internal
circuitry to save power. All the register contents are maintained. Some counters maintain their value.
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’DAC26 COMMUNICATION PROTOCOL
Register Programming
The ’DAC26 is entirely controlled by registers. An SPI master controlls the reading and writing of these registers by the
use of a 16-bit command, which is sent prior to the data for that register . The command is constructed as shown in Figure 19.
The command word begins with a R/W bit, which specifies the direction of data flow on the SPI serial bus. The following
four bits specify the page of memory this command is directed to, as shown in Table 2. The next six bits specify the register
address on that page of memory to which the data is directed. The last five bits are reserved for future use and should be
written only with zeros.
Table 2. Page Addressing
PG3 PG2 PG1 PG0 PAGE ADDRESSED
0 0 0 0 0
0 0 0 1 1
0 0 1 0 2
0 0 1 1 Reserved
0 1 0 0 Reserved
0 1 0 1 Reserved
0 1 1 0 Reserved
0 1 1 1 Reserved
1 0 0 0 Reserved
1 0 0 1 Reserved
1 0 1 0 Reserved
1 0 1 1 Reserved
1 1 0 0 Reserved
1 1 0 1 Reserved
1 1 1 0 Reserved
1 1 1 1 Reserved
To read all the first page of memory, for example, the host processor must send the command 0x8000 to the ’DAC26 – this
specifies a read operation beginning at page 0, address 0. The processor can then start clocking data out of the ’DAC26.
The ’DAC26 automatically increments its address pointer to the end of the page; if the host processor continues clocking
data out past the end of a page, the ’DAC26 sends back the value 0xFFFF.
Likewise, writing to page 1 of memory consists of the processor writing the command 0x0800, which specifies a write
operation, with PG0 set to 1, and all the ADDR bits set to 0. This results in the address pointer pointing at the first location
in memory on Page 1. See the section on the ’DAC26 memory map for details of register locations
BIT 15
MSB BIT 14 BIT 13 BIT 12 BIT 11 BIT 10 BIT 9 BIT 8 BIT 7 BIT 6 BIT 5 BIT 4 BIT 3 BIT 2 BIT 1 BIT 0
LSB
R/W* PG3 PG2 PG1 PG0 ADDR5 ADDR4 ADDR3 ADDR2 ADDR1 ADDR0 0 0 0 0 0
Figure 19. ’DAC26 Command Word
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COMMAND WORD DATA DATA
SS
SCLK
MOSI
Figure 20. Write Operation for ’DAC26 SPI Interface
COMMAND WORD
DATA DATA
MOSI
SCLK
MISO
SS
Figure 21. Read Operation for ’DAC26 SPI Interface
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’DAC26 MEMORY MAP
The ’DAC26 has several 16-bit registers which allow control of the device as well as providing a location for results from
the ’DAC26 to be stored until read by the host microprocessor . These registers are separated into three pages of memory
in the ’DAC26: a data page (Page 0) and control pages (Page 1 and Page 2). The memory map is shown in Table 3.
Table 3. Memory Map
Page 0: Reserved Page 1: Auxiliary Control Registers Page 2: Audio Control Registers
ADDR REGISTER ADDR REGISTER ADDR REGISTER
00 Reserved 00 Reserved 00 Audio Control 1
01 Reserved 01 Reserved 01 Reserved
02 Reserved 02 Reserved 02 DAC Gain
03 Reserved 03 Reserved 03 Analog Sidetone
04 Reserved 04 Reset 04 Audio Control 2
05 Reserved 05 Reserved 05 DAC Power Control
06 Reserved 06 Reserved 06 Audio Control 3
07 Reserved 07 Reserved 07 Digital Audio Effects Filter Coef ficients
08 Reserved 08 Reserved 08 Digital Audio Effects Filter Coef ficients
09 Reserved 09 Reserved 09 Digital Audio Effects Filter Coef ficients
0A Reserved 0A Reserved 0A Digital Audio Ef fects Filter Coefficients
0B Reserved 0B Reserved 0B Digital Audio Ef fects Filter Coefficients
0C Reserved 0C Reserved 0C Digital Audio Effects Filter Coef ficients
0D Reserved 0D Reserved 0D Digital Audio Effects Filter Coef ficients
0E Reserved 0E Reserved 0E Digital Audio Ef fects Filter Coefficients
0F Reserved 0F Reserved 0F Digital Audio Ef fects Filter Coefficients
10 Reserved 10 Reserved 10 Digital Audio Effects Filter Coef ficients
11 Reserved 11 Reserved 11 Digital Audio Ef fects Filter Coefficients
12 Reserved 12 Reserved 12 Digital Audio Effects Filter Coef ficients
13 Reserved 13 Reserved 13 Digital Audio Effects Filter Coef ficients
14 Reserved 14 Reserved 14 Digital Audio Effects Filter Coef ficients
15 Reserved 15 Reserved 15 Digital Audio Effects Filter Coef ficients
16 Reserved 16 Reserved 16 Digital Audio Effects Filter Coef ficients
17 Reserved 17 Reserved 17 Digital Audio Effects Filter Coef ficients
18 Reserved 18 Reserved 18 Digital Audio Effects Filter Coef ficients
19 Reserved 19 Reserved 19 Digital Audio Effects Filter Coef ficients
1A Reserved 1A Reserved 1A Digital Audio Ef fects Filter Coefficients
1B Reserved 1B Reserved 1B PLL Programmability
1C Reserved 1C Reserved 1C PLL Programmability
1D Reserved 1D Reserved 1D Audio Control 4
1E Reserved 1E Reserved 1E Audio Control 5
1F Reserved 1F Reserved 1F Reserved
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’DAC26 CONTROL REGISTERS
This section describes each of the registers shown in the memory map of Table 3. The registers are grouped according
to the function they control. In the ’DAC26, bits in control registers can refer to slightly different functions depending on
whether you are reading the register or writing to it.
’DAC26 Data Registers (Page 0)
The data registers in Page 0 are reserved.
PAGE 1 CONTROL REGISTER MAP
REGISTER 00H: Reserved
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15−D0 Reserved R FFFFH Reserved
REGISTER 01H: Reserved
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15−D0 Reserved R FFFFH Reserved
REGISTER 02H: Reserved
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15−D0 Reserved R FFFFH Reserved
REGISTER 03H: Reference Control
Register 03H is reserved.
REGISTER 04H: Reset Control
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15−D0 RSALL R/W FFFFH Reset All. W riting the code 0xBB00, as shown below, to this register causes the ’DAC26 to reset all
its registers to their default, power−up values.
1011101100000000 => Reset all registers
Others => Do not write other sequences to this register.
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PAGE 2 CONTROL REGISTER MAP
REGISTER 00H: Audio Control 1
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15−D14 R/W 00 Reserved
D13−D12 AIN R/W 00 Analog Input Mux
00 => Analog Input = Single-ended input MIC
01 => Analog Input = Single-ended input AUX
10 => Analog Input = Differential input MICIN and AUX
11 => Analog Input = Differential input MICIN and AUX
D11−D10 WLEN R/W 00 DAC W ord Length
00 => Word length = 16 bit
01 => Word length = 20 bit
10 => Word length = 24 bit
11 => Word length = 32 bit
D9−D8 DATFM R/W 00 Digital Data Format
00 => I2S mode
01 => DSP mode
10 => Right justified
11 => Left justified
Note: Right justified mode is NOT valid only when the DAC sampling ratio is Fsref/11.5 or
Fsref/5.5
D7−D6 Reserved R/W 00 Reserved
Note: Only write a 0 to this bit
D5−D3 DACFS R/W 000 DAC Sampling Rate
000 => DAC FS = Fsref/1
001 => DAC FS = Fsref/(1.5)
010 => DAC FS = Fsref/2
011 => DAC FS = Fsref/3
100 => DAC FS = Fsref/4
101 => DAC FS = Fsref/5
110 => DAC FS = Fsref/(5.5)
111 => DAC FS = Fsref/6
Note: Fsref can be set between 39 kHz and 53 kHz
D2−D0 Reserved R/W 000 Reserved
REGISTER 01H: Reserved
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15−D0 R FFFFH Reserved
REGISTER 02H: DAC Gain Control
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15 DALMU R/W 1 DAC Left Channel Muted
1 => DAC left channel muted
0 => DAC left channel not muted
D14−D8 DALVL R/W 1111111 DAC Left Channel Volume Control
0000000 => DAC left channel volume control = 0 dB
0000001 => DAC left channel volume control = −0.5 dB
0000010 => DAC left channel volume control = −1.0 dB
−−−−−
1111110 => DAC left channel volume control = −63.0 dB
1111111 => DAC left channel volume control = −63.5 dB
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REGISTER 02H: DAC Gain Control (continued)
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D7 DARMU R/W 1 DAC Right Channel Muted
1 => DAC right channel muted
0 => DAC right channel not muted
D6−D0 DARVL R/W 1111111 DAC Right Channel Volume Control
0000000 => DAC right channel volume control = 0 dB
0000001 => DAC right channel volume control = −0.5 dB
0000010 => DAC right channel volume control = −1.0 dB
−−−−−
1111110 => DAC right channel volume control = −63.0 dB
1111111 => DAC right channel volume control = −63.5 dB
REGISTER 03H: Analog Sidetone Control
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15 ASTMU R/W 1 Analog Sidetone Mute Control
1 => Analog sidetone muted
0 => Analog sidetone not muted
D14−D8 ASTG R/W 1000101 Analog Sidetone Gain Setting
0000000 => Analog sidetone gain setting = −34.5 dB
0000001 => Analog sidetone gain setting = −34 dB
0000010 => Analog sidetone gain setting = −33.5 dB
−−−−−
1000101 => Analog sidetone gain setting = 0 dB
100011 0 = > Analog sidetone gain setting = 0.5 dB
−−−−−
1011100 => Analog sidetone gain setting = 11.5 dB
1011101 => Analog sidetone gain setting = 12 dB
1011110 => Analog sidetone gain setting = 12 dB
1011111 => Analog sidetone gain setting = 12 dB
−−−−−
11xxxxx => Analog sidetone gain setting = 12 dB
D7−D1 Reserved R/W 1 Reserved
Note: Only write a 1 to this bit
D0 ASTGF R 0 Analog Sidetone PGA Flag ( Read Only )
0 => Gain applied /= PGA register setting
1 => PGA applied = PGA register setting.
Note: Analog sidetone gain is implemented at zero crossings of the signal.
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REGISTER 04H: Audio Control 2
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15 KCLEN R/W 0 Keyclick Enable
0 => Keyclick disabled
1 => Keyclick enabled
Note: This bit is automatically cleared after giving out the keyclick signal length equal to the
programmed value.
D14−D12 KCLAC R/W 100 Keyclick Amplitude Control
000 => Lowest amplitude
100 => Medium amplitude
11 1 => Highest amplitude
D11 Reserved R/W 0 Reserved
Note: Only write a 0 to this bit
D10−D8 KCLFRQ R/W 100 Keyclick Frequency
000 => 62.5 Hz
001 => 125 Hz
010 => 250 Hz
011 => 500 Hz
100 => 1 kHz
101 => 2 kHz
110 => 4 kHz
111 => 8 kHz
D7−D4 KCLLN R/W 0001 Keyclick Length
0000 => 2 periods key click
0001 => 4 periods key click
0010 => 6 periods key click
−−−−−
1110 => 30 periods key click
1111 => 32 periods key click
D3 DLGAF R 0 DAC Left Channel PGA Flag ( Read Only )
0 => Gain applied /= PGA register setting
1 => Gain applied = PGA register setting
Note: This flag indicates when the soft-stepping for DAC left channel is completed
D2 DRGAF R 0 DAC Right Channel PGA Flag ( Read Only )
0 => Gain applied /= PGA register setting
1 => Gain applied = PGA register setting
Note: This flag indicates when the soft-stepping for DAC right channel is completed
D1 DASTC R/W 0 DAC Channel PGA Soft-Stepping Control
0 => 0.5dB change every LRCK
1 => 0.5dB change every 2 LRCK
D0 Reserved R/W 0 Reserved
Note: Only write a 0 to this bit
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REGISTER 05H: DAC Power Control
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15 PWDNC R/W 1 DAC Power-Down Control
0 => DAC powered up
1 => DAC powered down
D14 Reserved R/W 0 Reserved (During read the value of this bit is 0. Write only 0 into this location.)
Note: Only set this bit to the reset value 0 or 1
D13 ASTPWD R/W 1 Analog Sidetone Power-down Control
0 => Analog sidetone powered up
1 => Analog sidetone powered down
D12 DAODRC R/W 0 Audio Output Driver Control
0 => Output driver in low power mode
1 => Output driver in high power mode
D11 ASTPWF R 1 Analog Sidetone Power-Down Flag
0 => Analog sidetone powered down is not complete.
1 => Analog sidetone powered down is complete.
D10 DAPWDN R/W 1 DAC Power-Down Control
0 => Power up the DAC
1 => Power down the DAC
D9 Reserved R/W 1 Reserved
Note: Only set this bit to the reset value 0 or 1
D8 VGPWDN R/W 1 Driver Virtual Ground Power Down
0 => Power up the VGND amp
1 => Power down the VGND amp
D7 Reserved R/W 1 Reserved
Note: Only set this bit to the reset value 0 or 1
D6 DAPWDF R 1 DAC Power-Down Flag (See DAC Power down section of this data sheet)
0 => DAC power down is not complete.
1 => DAC power down is complete.
D5 Reserved R/W 0 Reserved
Note: Only set this bit to the reset value 0 or 1
D4 VBIAS R/W 0 VBIAS Voltage
0 => VBIAS output = 2.5 V
1 => VBIAS output = 2.0 V
D3−D2 Reserved R/W 0 Reserved
Note: Only set this bit to the reset value 0 or 1
D1 EFFCTL R/W 0 Digital Audio Effects Filter Control
0 => Disable digital audio effects filter
1 => Enable digital audio ef fects filter
D0 DEEMPF R/W 0 De−Emphasis Filter Enable
0 => Disable de-emphasis filter
1 => Enable de-emphasis filter
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REGISTER 06H: Audio Control 3
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15−D14 DMSVOL R/W 00 DAC Channel Master Volume Control
00 => Left channel and right channel have independent volume controls
01 => Left channel volume control is the programmed value of the right channel volume control.
10 => Right channel volume control is the programmed value of the left channel volume control.
11 => same as 00
D13 REFFS R/W 0 Reference Sampling Rate. This setting controls the coefficients in the de-emphasis filter. If an
Fsref above 48 kHz is being used, then it is recommended to set this to the 48-kHz setting,
otherwise either setting can be used.
0 => Fsref = 48.0 kHz
1 => Fsref = 44.1 kHz
D12 DAXFM R/W 0 Master Transfer Mode
0 => Continuous data transfer mode
1 => 256−s data transfer mode
D11 SLVMS R/W 0 DAC Master Slave Control
0 => ’DAC26 is slave
1 => ’DAC26 is master
D10−D9 DAPK2PK R/W 00 DAC Max Output Signal Swing and Common Mode Voltage
00 => DAC max output signal swing = 2.0 V, VCM = 1.35 V
01 => DAC max output signal swing = 2.192 V (only recommended for analog supply of 3.0 V
and digital supply of 1.65 V and above), VCM = 1.48 V
10 => DAC max output signal swing = 2.402 V (only recommended for analog supply of 3.3 V
and digital supply of 1.8 V and above), VCM = 1.62 V
11 => DAC max output signal swing = 2.633 V (only recommended for analog supply of 3.6 V
and digital supply of 1.95 V), VCM = 1.78 V
D8 Reserved R/W 0 Reserved
Note: Always write the reset value to this bit
D7 DALOVF R 0 DAC Left Channel Overflow Flag ( Read Only )
0 => DAC left channel data is within saturation limits.
1 => DAC left channel data has exceeded saturation limits.
Note : This flag is reset only on register read.
D6 DAROVF R 0 DAC Right Channel Overflow Flag ( Read Only )
0 => DAC right channel data is within saturation limits.
1 => DAC right channel data has exceeded saturation limits.
Note : This flag is reset only on register read.
D5−D0 Reserved R/W 0 Reserved
Note: Always write the reset value to this bit
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REGISTER 07H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_N0 R/W 27619 Left channel digital audio effects filter coefficient N0
REGISTER 08H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_N1 R/W −27034 Left channel digital audio effects filter coefficient N1
REGISTER 09H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_N2 R/W 26461 Left channel digital audio effects filter coefficient N2
REGISTER 0AH: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_N3 R/W 27619 Left channel digital audio effects filter coefficient N3
REGISTER 0BH: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_N4 R/W −27034 Left channel digital audio effects filter coefficient N4
REGISTER 0CH: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_N5 R/W 26461 Left channel digital audio effects filter coefficient N5
REGISTER 0DH: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_D1 R/W 32131 Left channel digital audio effects filter coefficient D1
REGISTER 0EH: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_D2 R/W −31506 Left channel digital audio effects filter coefficient D2
REGISTER 0FH: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_D4 R/W 32131 Left channel digital audio effects filter coefficient D4
REGISTER 10H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 L_D5 R/W −31506 Left channel digital audio effects filter coefficient D5
REGISTER 11H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_N0 R/W 27619 Right channel digital audio effects filter coef ficient N0
REGISTER 12H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_N1 R/W −27034 Right channel digital audio effects filter coef ficient N1
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REGISTER 13H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_N2 R/W 26461 Right channel digital audio effects filter coef ficient N2
REGISTER 14H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_N3 R/W 27619 Right channel digital audio effects filter coef ficient N3
REGISTER 15H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_N4 R/W −27034 Right channel digital audio effects filter coef ficient N4
REGISTER 16H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_N5 R/W 26461 Right channel digital audio effects filter coef ficient N5
REGISTER 17H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_D1 R/W 32131 Right channel digital audio effects filter coef ficient D1
REGISTER 18H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_D2 R/W −31506 Right channel digital audio effects filter coef ficient D2
REGISTER 19H: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_D4 R/W 32131 Right channel digital audio effects filter coef ficient D4
REGISTER 1AH: Digital Audio Effects Filter Coefficients
BIT NAME READ/
WRITE RESET VALUE
(IN DECIMAL) FUNCTION
D15−D0 R_D5 R/W −31506 Right channel digital audio effects filter coef ficient D5
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REGISTER 1BH: PLL Programmability
BIT NAME READ/
WRITE RESET VALUE FUNCTION
D15 PLLSEL R/W 0 PLL Enable
0 => Disable PLL
1 => Enable PLL
D14−D11 QVAL R/W 0010 Q value. Valid only if PLL is disabled.
0000 => 16
0001 => 17
0010 => 2
0011 => 3
−−−−−
1100 => 12
1101 => 13
11 10 => 14
1111 => 15
D10−D8 PVAL R/W 000 P value. Valid when PLL is enabled
000 => 8
001 => 1
010 => 2
011 => 3
100 => 4
101 => 5
110 => 6
11 1 => 7
D7−D2 JVAL R/W 000001 J value. Valid only if PLL is enabled.
000000 => Not valid
000001 => 1
000010 => 2
−−−−−
111110 => 62
111111 => 63
D1−D0 Reserved R 00 Reserved (write only 00)
REGISTER 1CH: PLL Programmability
BIT NAME READ/
WRITE RESET VALUE FUNCTION
D15−D2 DVAL R/W 0 (in decimal) D value. Used when PLL is enabled.
D value is valid from 0000 to 9999 in decimal.
Programmed value greater than 9999 is treated as 9999.
00000000000000 => 0 decimal
00000000000001 => 1 decimal
D1−D0 Reserved R 00 Reserved (write only 00)
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REGISTER 1DH: Audio Control 4
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15 R/W 0 Reserved
D14 DASTPD R/W 0 DAC PGA Soft-Stepping Control
0 => Soft-stepping enabled
1 => Soft-stepping disabled
D13 ASSTPD R/W 0 Analog Sidetone Soft-Stepping Control
0 => Soft-stepping enabled
1 => Soft-stepping disabled
D12−D9 Reserved R/W 0 Reserved
Note: Only write a 0 to this bit
D8 SHCKT_DIS R/W 0 Disable Short Circuit Detection
0 => Short circuit detection enabled
1 => Short circuit detection disabled
D7 SHCKT_PD R/W 0 Power Down Drivers if Short Circuit Detected
0 => No auto power down of drivers on short circuit.
1 => Auto power down drivers on short circuit.
D6 SHCKT_FLAG R 0 Short Circuit Detected Flag
0 => Short circuit not detected
1 => Short circuit detected
D5 DAC_POP_RED R 0 DAC POP Reduction Enable
0 => Disable POP reduction
1 => Enable POP reduction
D4 DAC_POP_RED_
SET1 R/W 0 DAC POP Reduction Setting 1
0 => Fast setting
1 => Slow setting
D3−D2 DAC_POP_RED_
SET2 R/W 00 DAC POP Reduction Setting 2
00 => Long setting
11 => Short setting
D1−D0 Reserved R XX Reserved
REGISTER 1EH: Audio Control 5
BIT NAME READ/
WRITE RESET
VALUE FUNCTION
D15−D3 Reserved R/W 0 Reserved
Note: Only write a 0 to this bit
D2 DRV_POP_DIS R/W 0 Audio Output Driver POP Reduction Enable
0 => Enabled
1 => Disabled
D1 DRV_POP_LEN R/W 0 Audio Output Driver POP Reduction Duration
0 => Output driver ramps to final voltage in approximately 0.8 sec, if VGND is
powered down (1 msec otherwise).
1 => Output driver ramps to final voltage in approximately 4 sec, if VGND is powered
down (1 msec otherwise).
D0 Reserved R 0 Reserved. Always write a 0 to this bit.
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LAYOUT
The following layout suggestions should provide optimum performance from the ’DAC26. However, many portable
applications have conflicting requirements concerning power, cost, size, and weight. In general, most portable devices
have fairly clean power and grounds because most of the internal components are very low power. This situation means
less bypassing for the converter power and less concern regarding grounding. Still, each situation is unique and the
following suggestions should be reviewed carefully.
For optimum performance, care must be taken with the physical layout of the ’DAC26 circuitry. Power to the ’DAC26 must
be clean and well bypassed. A 0.1-µF ceramic bypass capacitor must be placed as close to the device as possible. A 1-µF
to 10-µF capacitor may also be needed if the impedance between the ’DAC26 supply pins and the system power supply
is high.
The ground pins must be connected to a clean ground point. In many cases, this is the analog ground. Avoid connections
which are too near the grounding point of a microcontroller or digital signal processor . If needed, run a ground trace directly
from the converter to the power supply entry or battery connection point. The ideal layout includes an analog ground plane
dedicated to the converter and associated analog circuitry.
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DAC CHANNEL DIGITAL FILTER
DAC Channel Digital Filter Frequency Response
0
−20
−40
−60
−80
−100
−120
−140
−160 0 0.5 1 1.5 2 2.5 3 3.5
Magnitude − dB
Frequency − Hz
Frequency Response of Full DAC Channel Digital Filterat Fs = 48 kHz
x 105
DAC Channel Digital Filter Pass-Band Frequency Response
0.04
0.02
0
−0.02
−0.04
−0.06
−0.08
−0.1
−0.12
−0.14
0.5 1 1.5 2
Magnitude − dB
Frequency − Hz
Frequency Response of Full DAC Channel Digital Filter at Fs = 48 kHz
x 104
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DEFAULT DIGITAL AUDIO EFFECTS FILTER RESPONSE AT 48 ksps
0
−0.5
−1
−1.5
−2
−2.5
−3
100101102103104
Magnitude − dB
Frequency − Hz
Frequency Response of 4th Order Effects Filter With Default Coefficients Set
DE-EMPHASIS FILTER FREQUENCY RESPONSE
De-Emphasis Filter Response at 32 ksps
Gain − dB
Frequency − Hz
Digital De-Emphasis Frequency Response at Fs = 32 kHz
0
−1
−2
−3
−4
−5
−6
−7
−8
−9
−10 0 2000 4000 6000 8000 10000 12000 14000 16000
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De-Emphasis Error at 32 ksps
0.3
0.25
0.2
0.15
0.1
0.05
0
−0.05
−0.10 2000 4000 6000 8000 10000 12000 14000 16000
Gain − dB
Frequency − Hz
De-Emphasis Error With Respect to Ideal Frequency Response For Fs = 33 kHz
De-Emphasis Filter Frequency Response at 44.1 ksps
0
−1
−2
−3
−4
−5
−6
−7
−8
−9
−100 0.5 1 1.5 2
Gain − dB
Frequency − Hz
Digital De-Emphasis Frequency Response For Fs = 44.1 kHz
x 104
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De-Emphasis Error at 44.1 ksps
Gain − dB
Frequency − Hz
De-Emphasis Error With Respect to Ideal Frequency Response For Fs = 44.1 kHz
0.3
0.25
0.2
0.15
0.1
0.05
0
−0.05
−0.10 0.5 1 1.5 2 2.5 x 104
De-Emphasis Frequency Response at 48 ksps
0
−1
−2
−3
−4
−5
−6
−7
−8
−9
−10 0 0.5 1 1.5 2 2.5
Gain − dB
Frequency − Hz
Digital De-Emphasis Frequency Response at Fs = 48 kHz
x 104
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De-Emphasis Error at 48 ksps
0.3
0.25
0.2
0.15
0.1
0.05
0
−0.05
−0.1 0 0.5 1 1.5 2 2.5
Gain − dB
Frequency − Hz
De-Emphasis Error With Respect to Ideal Frequency Response For Fs = 48 kHz
x 104
PLL PROGRAMMING
The on-chip PLL in the ’DAC26 can be used to generate sampling clocks from a wide range of MCLK’s available in a system.
The PLL works by generating oversampled clocks with respect to Fsref (44.1 kHz or 48 kHz). Frequency division generates
all other internal clocks. The table below gives a sample programming for PLL registers for some standard MCLK’s when
PLL is required. Whenever the MCLK is of the form of N x 128 x Fsref (N=2,3,17), PLL is not required.
Fsref = 44.1 kHz
MCLK (MHz) P J D ACHIEVED FSREF % ERROR
2.8224 1 32 0 44100.00 0.0000
5.6448 1 16 0 44100.00 0.0000
12 1 7 5264 44100.00 0.0000
13 1 6 9474 44099.71 0.0007
16 1 5 6448 44100.00 0.0000
19.2 1 4 7040 44100.00 0.0000
19.68 1 4 5893 44100.30 −0.0007
48 4 7 5264 44100.00 0.0000
Fsref = 48 kHz
MCLK (MHz) P J D ACHIEVED FSREF % ERROR
2.048 1 48 0 48000.00 0.0000
3.072 1 32 0 48000.00 0.0000
4.096 1 24 0 48000.00 0.0000
6.144 1 16 0 48000.00 0.0000
8.192 1 12 0 48000.00 0.0000
12 1 8 1920 48000.00 0.0000
13 1 7 5618 47999.71 0.0006
16 1 6 1440 48000.00 0.0000
19.2 1 5 1200 48000.00 0.0000
19.68 1 4 9951 47999.79 0.0004
48 4 8 1920 48000.00 0.0000
PACKAGING INFORMATION
Orderable Device Status (1) Package
Type Package
Drawing Pins Package
Qty Eco Plan (2) Lead/Ball Finish MSL Peak Temp (3)
TLV320DA26IRHBG4 ACTIVE QFN RHB 32 73 Green (RoHS &
no Sb/Br) CU NIPDAU Level-2-260C-1 YEAR
TLV320DAC26IRHB ACTIVE QFN RHB 32 73 Green (RoHS &
no Sb/Br) CU NIPDAU Level-2-260C-1 YEAR
TLV320DAC26IRHBG4 ACTIVE QFN RHB 32 73 Green (RoHS &
no Sb/Br) CU NIPDAU Level-2-260C-1 YEAR
TLV320DAC26IRHBR ACTIVE QFN RHB 32 3000 Green (RoHS &
no Sb/Br) CU NIPDAU Level-2-260C-1 YEAR
TLV320DAC26IRHBRG4 ACTIVE QFN RHB 32 3000 Green (RoHS &
no Sb/Br) CU NIPDAU Level-2-260C-1 YEAR
(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.
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.
PACKAGE OPTION ADDENDUM
www.ti.com 6-Dec-2006
Addendum-Page 1
TAPE AND REEL INFORMATION
*All dimensions are nominal
Device Package
Type Package
Drawing Pins SPQ Reel
Diameter
(mm)
Reel
Width
W1 (mm)
A0
(mm) B0
(mm) K0
(mm) P1
(mm) W
(mm) Pin1
Quadrant
TLV320DAC26IRHBR QFN RHB 32 3000 330.0 12.4 5.3 5.3 1.5 8.0 12.0 Q2
PACKAGE MATERIALS INFORMATION
www.ti.com 16-Feb-2012
Pack Materials-Page 1
*All dimensions are nominal
Device Package Type Package Drawing Pins SPQ Length (mm) Width (mm) Height (mm)
TLV320DAC26IRHBR QFN RHB 32 3000 338.1 338.1 20.6
PACKAGE MATERIALS INFORMATION
www.ti.com 16-Feb-2012
Pack Materials-Page 2
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