TI PCM1774

 PCM1774
SLAS551 – JULY 2007
16-Bit, Low-Power Stereo Audio DAC With Analog Mixing, Line and Headphone Outputs
FEATURES
•
•
•
•
•
•
•
•
Analog Front End:
– Stereo Single-Ended Input
– Microphone Amplifier (12 dB, 20 dB)
Analog Back End:
– Stereo/Mono Line Output With Volume
– Stereo/Mono Headphone Amplifier With
Volume
Analog Performance:
– Dynamic Range: 93 dB
– 40-mW + 40-mW Headphone Output at
RL = 16 Ω
Power Supply Voltage
– 1.71 V to 3.6 V for Digital I/O Section
– 1.71 V to 3.6 V for Digital Core Section
– 2.4 V to 3.6 V for Analog Section
– 2.4 V to 3.6 V for Power Amplifier Section
Low Power Dissipation:
– 6.4 mW in Playback, 1.8 V/2.4 V, 44.1 kHz
– 3.3 μW in Power Down
Sampling Frequency: 5 kHz to 50 kHz
Operation From a Single Clock Input Without
PLL
System Clock:
– Common-Audio Clock (256 fS/384 fS), 12/24,
13/26, 13.5/27, 19.2/38.4, 19.68/39.36 MHz
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–
–
–
–
–
–
–
•
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•
2 (I2C™) or 3 (SPI) Wire Serial Control
Programmable Function by Register Control:
Digital Attenuation: 0 dB to –62 dB
Digital Gain of DAC: 0, 6, 12, 18 dB
Power Up/Down Control for Each Module
6-dB to –70-dB Gain for Analog Outputs
0/12/20 dB for Microphone Input
0-dB to –21-dB Gain for Analog Mixing
Three-Band Tone Control and 3D Sound
Analog Mixing Control
Pop-Noise Reduction Circuit
Short Protection Circuit
Package: 4-mm × 4-mm QFN Package
Operation Temperature Range: –40°C to 85°C
APPLICATIONS
•
•
•
Portable Audio Player, Cellular Phone
Video Camcorder, Digital Still Camera
PMP/DMB/PND
DESCRIPTION
The PCM1774 is a low-power stereo DAC designed
for portable digital audio applications. The device
integrates headphone amplifier, line amplifier, line
input, boost amplifier, programmable gain control,
analog mixing, and sound effects. It is available in a
small-footprint, 4-mm × 4-mm QFN package. The
PCM1774 supports right-justified, left-justified, I2S,
and DSP formats, providing easy interfacing to audio
DSP and decoder/encoder chips. Sampling rates up
to 50 kHz are supported. The user-programmable
functions are accessible through a two- or three-wire
serial control port.
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.
I2C is a trademark of Philips Electronics.
PRODUCTION DATA information is current as of publication date.
Products conform to specifications per the terms of the Texas
Instruments standard warranty. Production processing does not
necessarily include testing of all parameters.
Copyright © 2007, Texas Instruments Incorporated
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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.
ABSOLUTE MAXIMUM RATINGS
over operating free-air temperature range (unless otherwise noted) (1)
Supply voltage
VDD, VIO, VCC, VPA
Ground voltage differences: DGND, AGND, PGND
Input voltage
MAX
UNIT
–0.3 to 4
V
±0.1
V
–0.3 to 4
V
±10
mA
Ambient temperature under bias
–40 to 110
°C
Storage temperature
–55 to 150
°C
Junction temperature
150
°C
Lead temperature (soldering)
260
°C, 5 s
Package temperature (reflow, peak)
260
°C
Input current (any pin except supplies)
(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.
RECOMMENDED OPERATING CONDITIONS
over operating free-air temperature range (unless otherwise noted)
VSS
MIN
NOM
MAX
Analog supply voltage, VCC, VPA
2.4
3.3
3.6
V
Digital supply voltage, VDD, VIO
1.71
3.3
3.6
V
Digital input logic family
Digital input clock frequency
Analog output load resistance
CMOS
SCKI system clock
LRCK sampling clock
3.072
18.432
MHz
8
48
kHz
LOL and LOR
10
HPOL and HPOR
16
Analog output load capacitance
Digital output load capacitance
TA
2
UNIT
Operating free-air temperature
–40
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kΩ
Ω
30
pF
10
pF
85
°C
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ELECTRICAL CHARACTERISTICS
All specifications at TA = 25°C, VDD = VIO = VCC = VPA = 3.3 V, fS = 48 kHz, system clock = 256 fS, and 16-bit data (unless
otherwise noted).
PARAMETER
TEST CONDITIONS
MIN
TYP
MAX
UNIT
Audio Data Characteristics
DATA FORMAT
Resolution
16
Bits
2
I S, left-,
right-justified,
DSP
Audio data interface format
Audio data bit length
16
Bits
MSB first, 2s
complement
Audio data format
Sampling frequency (fS)
5
System clock
50
VDD < 2 V
27
VDD > 2 V
40
kHz
MHz
Digital Input/Output
CMOS
compatible
Logic family
VIH
VIL
IIH
IIL
VOH
VOL
0.7 VIO
Input logic level
0.3 VIO
VIN = 3.3 V
Input logic current
10
VIN = 0 V
IOH = –2 mA
Output logic level
–10
0.75 VIO
IOL = 2 mA
0.25 VIO
Vdc
μA
Vdc
Digital Input to Line Output Through DAC (LOL and LOR)
RL = 10 kΩ, volume = 0 dB, analog mixing = disabled
DYNAMIC PERFORMANCE
SNR
Full-scale output voltage
0 dB
Dynamic range
EIAJ, A-weighted
Signal-to-noise ratio
EIAJ, A-weighted
2.828
86
Channel separation
THD+N
Total harmonic distortion + noise
0 dB
Load resistance
VPP
1
Vrms
93
dB
93
dB
91
dB
0.008%
10
kΩ
Line Input to Line Output Through Mixing Path (LOL and LOR)
RL = 10 kΩ, volume = 0 dB, analog mixing = enabled
DYNAMIC PERFORMANCE
SNR
Full-scale input and output voltage
0 dB
Signal-to-noise ratio
EIAJ, A-weighted
2.828
84
VPP
1
Vrms
93
dB
Digital Input to Headphone Output Through DAC (HPOL and HPOR)
RL = 16 Ω or 32 Ω, ALC = OFF, volume = 0 dB, speaker = powered down, analog mixing = disabled, not capless mode
DYNAMIC PERFORMANCE
Full-scale output voltage
0 dB
SNR
Signal-to-noise ratio
EIAJ, A-weighted
THD+N
Total harmonic distortion + noise
2.828
84
Vrms
93
dB
30 mW, RL = 32 Ω,
volume = 0 dB
0.1%
40 mW, RL = 16 Ω,
volume = –1 dB
0.03%
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VPP
1
3
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ELECTRICAL CHARACTERISTICS (continued)
All specifications at TA = 25°C, VDD = VIO = VCC = VPA = 3.3 V, fS = 48 kHz, system clock = 256 fS, and 16-bit data (unless
otherwise noted).
PARAMETER
TEST CONDITIONS
Load resistance
PSRR
Power-supply rejection ratio
MIN
TYP
MAX
UNIT
Ω
16
200 Hz, 140 mVPP
–40
1 kHz, 140 mVPP
–45
20 kHz, 140 mVPP
–32
dB
Line Input to Headphone Output Through Mixing Path (HPOL and HPOR)
RL = 16 Ω or 32 Ω, ALC = OFF, volume = 0 dB, speaker = powered down, analog mixing = enabled, not capless mode
DYNAMIC PERFORMANCE
SNR
Full-scale output voltage
0 dB
Signal-to-noise ratio
EIAJ, A-weighted
Load resistance
2.828
84
VPP
1
Vrms
93
dB
Ω
16
Filter Characteristics
INTERPOLATION FILTER FOR DAC
Pass band
0.454 fS
Stop band
0.546 fS
±0.04
Pass-band ripple
Stop-band attenuation
–50
dB
dB
Group delay
19/fs
s
De-emphasis error
±0.1
dB
±0.2
dB
ANALOG FILTER
Frequency response
f = 20 kHz
Power Supply and Supply Current
VIO
1.71
3.3
3.6
VDD
1.71
3.3
3.6
2.4
3.3
3.6
2.4
3.3
3.6
4.5
10
mA
1
10
μA
14.8
33
mW
3.3
33
μW
VCC
Voltage range
VPA
Supply current
Power dissipation
BPZ input, all active, no load
All inputs are held static
BPZ input
All inputs are held static
Vdc
Temperature Condition
Operation temperature
θJA
4
–40
Thermal resistance
85
40
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°C
°C/W
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PIN ASSIGNMENTS
15
14
13
12
LRCK
PGND
VPA
HPOR/LOR
HPOL/LOL
PCM1774RGP
(TOP VIEW)
11
17
9
SCKI
VCC
18
8
DGND
AIN1R
19
7
VDD
AIN1L
20
6
VIO
1
2
3
4
5
DIN
AGND
MC/SCL
BCK
MD/SDA
10
MS/ADR
16
MODE
VCOM
Table 1. TERMINAL FUNCTIONS
TERMINAL
I/O
DESCRIPTION
NAME
NO.
AGND
17
–
Ground for analog
AIN1L
20
I
Analog input 1 for L-channel
AIN1R
19
I
Analog input 1 for R-channel
BCK
10
I/O
Serial bit clock
DGND
8
–
Digital ground
DIN
5
I
Serial audio data input
HPOL/LOL
15
O
Headphone/lineout for R-channel
HPOR/LOR
14
O
Headphone/lineout for L-channel
LRCK
11
I/O
Left and right channel clock
MC/SCL
4
I
Mode control clock for three-wire/two-wire interface
MD/SDA
3
I/O
Mode control data for three-wire/two-wire interface
MODE
1
I
Two- or three-wire interface selection (LOW: SPI, HIGH: I2C)
MS/ADR
2
I
Mode control select for three-wire/two-wire interface
PGND
12
–
Ground for speaker power amplifier
SCKI
9
I
System clock
VCC
18
–
Analog power supply
VCOM
16
–
Analog common voltage
VDD
7
–
Power supply for digital core
VIO
6
–
Power supply for digital I/O
VPA
13
–
Power supply for power amplifier
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5
6
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VCOM
AIN1R
AIN1L
VCOM
Power On
Reset
COM
PG5
PG6
VIO
0 to -62dB / Mute
0, 6, 12, 18 dB
VDD
DGND
Analog Input R-ch
0/12/20dB 0 to –21dB
PG2
LRCK
DIN
VPA
VCC
DS
DAC
Digital
Filter
PGND
DS
DAC
AGND
DAR
DAL
SW4
SW5
SW6
SW3
SW2
SW1
MXR
MXL
ROUT
MONO
LOUT
MODE
Serial Interface (SPI/I2C)
MD/SDA MC/SCL MS/ADR
Digital
Filter
ATP
DGP
Audio Interface
BCK
Analog Input L-ch
Clock
Manager
0/12/20dB 0 to –21dB
PG1
Power Up/Down
Manager
SCKI
6 to –70dB
HPR
6 to –70dB
HPL
HPOR/
LOR
HPOL/
LOL
Module of Possible Power Up/Down
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FUNCTIONAL BLOCK DIAGRAM
PCM1774
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TYPICAL PERFORMANCE CURVES
All specifications at TA = 25°C, VDD = VIO = VCC = VPA = 3.3 V, fS = 48 kHz, system clock = 256 fS, and 16-bit data, unless
otherwise noted.
INTERPOLATION FILTER, STOP BAND
INTERPOLATION FILTER, PASS BAND
0.2
0
-20
0.1
Amplitude - dB
Amplitude - dB
-40
-60
0
-80
-0.1
-100
-120
0
-0.2
1
2
f - Frequency - xfs
3
4
0
0.1
0.2
0.3
f - Frequency - xfs
0.4
Figure 1.
Figure 2.
THREE-BAND TONE CONTROL (BASS, MIDRANGE,
TREBLE)
THREE-BAND TONE CONTROL (BASS)
Figure 3.
Figure 4.
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0.5
7
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TYPICAL PERFORMANCE CURVES (continued)
All specifications at TA = 25°C, VDD = VIO = VCC = VPA = 3.3 V, fS = 48 kHz, system clock = 256 fS, and 16-bit data, unless
otherwise noted.
THREE-BAND TONE CONTROL (MIDRANGE)
THREE-BAND TONE CONTROL (TREBLE)
Figure 5.
Figure 6.
THD+N/SNR vs POWER SUPPLY
DAC TO HEADPHONE OUTPUT, 16-Ω
THD+N/SNR vs POWER SUPPLY
DAC TO LINE OUTPUT, 10-kΩ
93
0.02
92
SNR
91
0.01
90
0
2
2.5
3
3.5
4
fIN = 1 kHz
94
0.011
SNR
0.01
93
0.009
92
THD+N
91
0.008
90
0.007
2
Power Supply - V
Figure 7.
8
2.5
3
Power Supply - V
Figure 8.
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3.5
4
SNR - Signal To Noise Ratio
0.03
THD+N - Total Harmanic Distortion + Noise - %
94
THD+N
SNR - Signal To Noise Ratio
THD+N - Total Harmanic Distortion + Noise - %
fIN = 1 kHz
0.04
95
0.012
95
0.05
PCM1774
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SLAS551 – JULY 2007
TYPICAL PERFORMANCE CURVES (continued)
All specifications at TA = 25°C, VDD = VIO = VCC = VPA = 3.3 V, fS = 48 kHz, system clock = 256 fS, and 16-bit data, unless
otherwise noted.
OUTPUT POWER vs POWER SUPPLY
(HEADPHONE, 16-Ω)
THD+N vs OUTPUT POWER
(HEADPHONE, 16-Ω, VOLUME = 6 dB)
100
120
THD+N - Total Harmanic Distortion + Noise - %
fIN= 1 kHz
100
PO - Output Power - mW
Vol = 6 dB
80
60
Vol = 0 dB
40
20
2.5
3
Power Supply - V
3.5
4
3.6 V
2.7 V
1
0.1
20
40
60
80
PO - Output Power - mW
100
120
Figure 9.
Figure 10.
THD+N vs OUTPUT POWER
(HEADPHONE, 16-Ω, VOLUME = 0 dB)
OUTPUT SPECTRUM (DAC TO HEADPHONE OUTPUT,
16-Ω)
1
0
fIN = 1kHz
fIN= 1 kHz / –60 dB
-20
2.4 V
-40
2.7 V
Amplitude - dB
THD+N - Total Harmanic Distortion + Noise - %
3.3 V
2.4 V
10
0.01
0
0
2
fIN = 1 kHz
0.1
3.3 V
3.6 V
-60
-80
-100
-120
0.01
-140
0
20
40
PO - Output Power - mW
60
80
0
Figure 11.
5
10
f - Frequency - kHz
15
20
Figure 12.
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DETAILED DESCRIPTION
Analog Input
The AIN1L and AIN1R pins can be used as microphone or line inputs with selectable 0-, 12-, or 20-dB boost and
1-Vrms input. All of these analog inputs have high input impedance (20 kΩ), which is not changed by gain
settings. One pair of inputs is selected by register 87 (AIL0 and AIR0).
Gain Settings for Analog Input
The gain of the analog signals can be adjusted from 0 dB to –21 dB in 1-dB steps following the 0-, 12-, or 20-dB
boost amplifier. The gain level can be set for each channel by registers 89 (GMR[2:0], GML [2:0].
D/A Converter
The DAC includes a multilevel delta-sigma modulator and an interpolation filter. These can be used to obtain
high PSRR, low jitter sensitivity, and low out-of-band noise quickly and easily. The interpolation filter includes
digital attenuator, digital soft mute, three-band tone control (bass, midrange and treble), and 3-D sound
controlled by registers 92 to 95. The de-emphasis filter (32, 44.1 and 48 kHz) is controlled by registers 68 to 70
(ATL[5:0], ATR[5:0], PMUL, PMUR, DEM[1:0]). Oversampling rate control can reduce out-of-band noise when
operating at low sampling rates by using register 70 (OVER).
Common Voltage
The VCOM pin is normally biased to 0.5 VCC, and it provides the common voltage to internal circuitry. It is
recommended that a 4.7-μF capacitor be connected between this pin and AGND to provide clean voltage and
avoid pop noise. The PCM1774 may have a little pop noise on each analog output if a capacitor smaller than
4.7 μF is used.
Line Output
The HPOL/LOL and HPOR/LOR pins can drive a 10-kΩ load and be configured by register 74 (HPS[1:0]) as a
monaural single-ended, monaural differential, or stereo single-line output with 1-Vrms output. These outputs
include an analog volume amplifier that can be set from 6 dB to –70 dB and mute in steps of 0.5-, 1-, 2- or 4-dB.
Each output is controlled by registers 64 and 65 (HLV[5:0], HRV[5:0], HMUL, HMUR). No dc blocking capacitor
is required when connecting an external speaker amplifier with monaural differential input. The center voltage is
0.5 VCC with zero data input.
Headphone Output
The HPOL/LOL and HPOR/LOR pins can be configured as a stereo, monaural, or monaural differential
headphone output by register 74 (HPS[1:0]). These pins have more than 30 or 40 mWrms output power into a
32- or 16-Ω load, either through a dc blocking capacitor or without a capacitor. These outputs include an analog
volume amplifier that can be set from 6 dB to –70 dB in steps of 0.5, 1, 2, or 4 dB. Each is controlled by
registers 64 and 65 (HLV[5:0], HRV[5:0], HMUL, HMUR). The center voltage is 0.5 VCC with zero data input.
Analog Mixing and Bypass
Mixing amplifiers (MXL, MXR) mix inputs from the AIN pins. The analog inputs are selected by register 87 (AIR0,
AIL0) and can bypass the DAC and connect the mixed signal to the headphone or speaker outputs by register
88 (MXR[2:0], MXL[2:0]). The gain of the analog inputs is controlled by register 89 (GMR[2:0], GML[2:0]). These
functions are suitable for FM radio, headset, and other analog sources without an ADC.
Digital Gain Control
A portable application with small speakers may be require a high sound level when playing back audio data
recorded at low level. Digital gain control (DGC) can be used to amplify the digital input data by 0, 6, 12 or 18 dB
by setting register 70 (SPX[1:0]).
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3-D Sound
A 3-D sound effect is provided by mixing L-channel and R-channel data with a band-pass filter with two
parameters, mixing ratio and band pass filter characteristic, that can be controlled by register 95 (3DP[3:0],
3FLO).
Three-Band Tone Control
Tone control has bass, midrange, and treble controls that can be adjusted from 12 dB to –12 dB in 1-dB steps
by registers 92 to 94 (LGA[4:0], MGA[4:0] and HGA[4:0]). Register 92 (LPAE) attenuates the digital input signal
automatically to prevent clipping of the output signal at settings above 0 dB for bass control. LPAE has no effect
on midrange and treble controls.
Digital Monaural Mixing
The audio data can be converted from stereo digital data to mixed monaural digital data. The conversion occurs
in the internal audio interface section and is controlled by register 96 (MXEN).
Zero-Cross Detection
Zero-cross detection minimizes audible zipper noise while changing analog volume and digital attenuation. This
function applies to the digital input or digital output as defined by register 86 (ZCRS).
Short Protection
The short-circuit protection on each headphone output prevents damage to the device while an output is shorted
to VPA, an output is shorted to PGND, or any two outputs are shorted together. When the short circuit is detected
on the outputs, the PCM1774 powers down the shorted amplifier immediately. The short-protection status can
be monitored by reading register 77 (STHC, STHL, SCHR) through the I2C interface. Short-circuit protection
operates in any enabled headphone amplifier.
Pop-Noise Reduction Circuit
The pop-noise reduction circuit prevents audible noise when turning the power supply on/off and powering the
device up/down in portable applications. It is recommended to establish the register settings in the sequence
that is shown in Table 3 and Table 4. No particular external parts are required.
Power Up/Down for Each Module
Using register 72 (PMXL, PMXR), register 73 (PBIS, PDAR, PDAL, PHPR, PHPL) and register 90 (PCOM),
unused modules can be powered down to minimize power consumption (7 mW during playback only).
Digital Audio Interface
The PCM1774 can receive I2S, right-justified, left-justified, and DSP formats in both master and slave modes.
These options can be selected in register 70 (PFM[1:0]), register 81 (RFM[1:0]) and register 84 (MSTR).
Digital Interface
All digital I/O pins can interface at various power supply voltages. VIO pin can be connected to a 1.71-V to 3.6-V
power supply.
Power Supply
The VCC pin and the VPA pin can be connected to 2.4 V to 3.6 V. The same voltage must be applied to both pins.
The VDD pin and the VIO pin can be connected to 1.71 V to 3.6 V. A different voltage can be applied to each of
these pins (for example, VDD = 1.8 V, VIO = 3.3 V).
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DESCRIPTION OF OPERATION
System Clock Input
The PCM1774 can accept clocks of various frequencies without a PLL. They are used for clocking the digital
filters and automatic level control and delta-sigma modulators and are classified as common-audio and
application-specific clocks. Table 2 shows frequencies of the common-audio clock and application-specific clock.
Figure 13 shows the timing requirements for system clock inputs. The sampling rate and frequency of the
system clocks are determined by the settings of register 86 (MSR[2:0]) and register 85 (NPR[5:0]). Note that the
sampling rate of the application-specific clock has a little sampling error. The details are shown in Table 9.
Table 2. System Clock Frequencies
CLOCK
FREQUENCIES
Common-audio clock
11.2896, 12.288, 16.9344, 18.432 MHz
Application-specific clock
12, 13, 13.5, 24, 26, 27, 19.2, 19.68, 38.4, 39.36 MHz
tw(SCKH)
0.7 VIO
SCKI
0.3 VIO
tw(SCKL)
T0005-12
PARAMETERS
SYMBOL
MIN
UNITS
System-clock pulse duration, high
tw(SCKH)
7
ns
System-clock pulse duration, low
tw(SCKL)
7
ns
Figure 13. System Clock Timing
Power-On Reset and System Reset
The power-on-reset circuit outputs a reset signal, typically at VDD = 1.2 V, and this circuit does not depend on
the voltage of other power supplies (VCC, VPA, and VIO). Internal circuits are cleared to default status, then all
analog and digital outputs have no signal. The PCM1774 does not require any power supply sequencing. Set
Register data after turning all power supplies on.
System reset is enabled by setting register 85 (SRST = 1). After the reset sequence, the register data is reset to
SRST = 0 automatically. All circuits are cleared to their default status at once by the system reset. Note that the
PCM1774 has audible pop noise on the analog outputs when enabling SRST.
Power On/Off Sequence
To reduce audible pop noise, a sequence of register settings is required after turning all power supplies on when
powering up, or before turning the power supplies off when powering down. If some modules are not required for
a particular application or operation, they should be placed in the power-down state after performing the
power-on sequence. The recommended power-on and power-off sequences are shown in Table 3 and Table 4,
respectively.
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Table 3. Recommended Power-On Sequence
(1)
(2)
(3)
STEP
REGISTER
SETTINGS
1
–
2
4027h
Headphone amplifier L-ch volume (–6 dB) (2)
3
4127h
Headphone amplifier R-ch volume (–6 dB) (2)
6
4427h
Digital attenuator L-ch (–24 dB) (2)
7
4527h
Digital attenuator R-ch (–24 dB)
8
4620h
DAC audio interface format (left-justified) (3)
12
49E0h
DAC (DAL, DAR) and analog bias power up
13
5601h
Zero-cross detection enable
14
4803h
Analog mixer (MXL, MXR) power up
15
5811h
Analog mixer input (SW2, SW5) select
16
49ECh
Headphone amplifier (HPL, HPR, HPC) power up
18
4A01h
VCOM power up
19
5230h
Analog front end (D2S, MCB, PG1, 2, 5, 6) power up
20
5711h
Analog input (MUX3, MUX4) select. Analog input (MUX1, MUX2) select
NOTE
Turn on all power supplies (1)
(2)
VDD should be turned on prior to or simultaneously with, the other power supplies. It is recommended to set register data with the
system clock input after turning all power supplies on.
Any level is acceptable for volume or attenuation. Level should be resumed by register data recorded when system powers off.
Audio interface format should be set to match the DSP or decoder being used.
Table 4. Recommended Power-Off Sequence
(1)
(2)
(3)
(4)
STEP
REGISTER
SETTINGS
1
447Fh
DAC L-ch digital soft-mute enable (1)
2
457Fh
DAC R-ch digital soft-mute enable (1)
4
5811h
Analog mixer input (SW2, SW5) select
5
49ECh
Headphone amplifier (HPL, HPR, HPC) power up (2)
6
5200h
Analog front end (D2S, MCB, PG1, 2, 5, 6) power down
7
5A00h
PG1, PG2 gain control (0 dB)
8
4A00h
VCOM power down
9
–
Wait time (750 ms)
10
49E0h
Headphone amplifier (HPL, HPR, HPC) power down, speaker amplifier (SPL, SPR) power down
11
4800h
Analog mixer (MXL, MXR) power down
12
4900h
DAC (DAL, DAR) and analog bias power down
13
–
NOTE
(3)
Turn off all power supplies. (4)
Any level is acceptable for volume or attenuation.
The headphone amplifier must be operating during the power-off sequence.
PCM1774 requires time for VCOM to reach the ground level from the common level. The wait time allowed depends on the settings of
register 125 PTM[1:0], RES[4:0]. The default setting is 750 ms for VCOM = 4.7 μF.
Power supply sequencing is not required. It is recommended to turn off all power supplies after setting the registers with the system
clock input.
Power-Supply Current
The current consumption of the PCM1774 depends on power up/down status of each circuit module. In order to
reduce the power consumption, disabling each module is recommended when it is not used in an application or
operation. Table 5 shows the current consumption in some states.
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Table 5. Power Consumption Table
OPERATION MODE
CONDITION
VOL
[V]
VIO
VDD
VCC
VPA
TOTAL
Zero Data
fS = 44.1 kHz
RL = 0 Ω
1.8
0.000
0.000
–
–
0.000
2.8
0.000
0.000
–
–
0.000
3.3
0.000
0.000
–
–
0.000
2.4
–
–
0.001
0.000
0.002
2.8
–
–
0.001
0.000
0.003
3.3
–
–
0.001
0.000
0.003
1.8
0
0.84
–
–
1.5
2.8
0.03
1.47
–
–
4.2
3.3
0.04
1.84
–
–
6.2
2.4
–
–
1.68
0.38
4.9
2.8
–
–
1.81
0.41
6.2
3.3
–
–
1.96
0.46
8.0
1.8
0
0.84
–
–
1.5
2.8
0.03
1.47
–
–
4.2
3.3
0.04
1.84
–
–
6.2
2.4
–
–
1.38
0.38
4.2
2.8
–
–
1.50
0.41
5.3
3.3
–
–
1.64
0.46
6.9
1.8
0
0.84
–
–
1.5
2.8
0.03
1.47
–
–
4.2
3.3
0.04
1.84
–
–
6.2
2.4
–
–
1.38
0.38
4.2
2.8
–
–
1.50
0.41
5.3
3.3
–
–
1.65
0.46
7.0
1.8
0
1.29
–
–
2.3
2.8
0.03
2.26
–
–
6.4
3.3
0.04
2.82
–
–
9.4
2.4
–
–
1.38
0.38
4.2
2.8
–
–
1.50
0.42
5.4
3.3
–
–
1.64
0.46
6.9
1.8
0
0.84
–
–
1.5
2.8
0.03
1.47
–
–
4.2
3.3
0.04
1.84
–
–
6.2
2.4
–
–
1.68
0.38
4.9
2.8
–
–
1.81
0.41
6.2
3.3
–
–
1.96
0.46
8.0
1.8
0
0.84
–
–
1.5
2.8
0.03
1.47
–
–
4.2
3.3
0.04
1.84
–
–
6.2
2.4
–
–
1.53
0.38
4.6
2.8
–
–
1.66
0.41
5.8
3.3
–
–
1.81
0.46
7.5
All Power Down
Zero Data
fS = 44.1 kHz
RL = 0 Ω
All Active
Zero Data
fS = 44.1 kHz
RL = 10 Ω
Line Output
Zero Data
fS = 44.1 kHz
RL = 16 Ω
Headphone Output
Zero Data
fS = 44.1 kHz
RL = 16 Ω
Headphone Output with Sound
Effect
Zero Data
fS = 44.1 kHz
RL = 16 Ω
Headphone Output with Stereo
Analog Mixing
Zero Data
fS = 44.1 kHz
RL = 16 Ω
Headphone Output with Mono
Analog Mixing
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Table 5. Power Consumption Table (continued)
OPERATION MODE
Headphone Output with Stereo
Analog Mixing
Headphone Output with Mono
Analog Mixing
(1)
CONDITION
VOL
[V]
VIO
VDD
VCC
VPA
TOTAL
Zero Data
fS = 44.1 kHz
RL = 16 Ω
No Digital Input (1)
1.8
0
0
–
–
0.0
2.8
0
0
–
–
0.0
3.3
0
0
–
–
0.0
2.4
–
–
0.68
0.38
2.5
2.8
–
–
0.69
0.41
3.1
3.3
–
–
0.71
0.46
3.9
1.8
0
0
–
–
0.0
2.8
0
0
–
–
0.0
3.3
0
0
–
–
0.0
2.4
–
–
0.52
0.38
2.2
2.8
–
–
0.54
0.42
2.7
3.3
–
–
0.55
0.46
3.3
Zero Data
fS = 44.1 kHz
RL = 16 Ω
No Digital Input (1)
POWER SUPPLY CURRENT [mA]
PD [mW]
All digital inputs are held static.
Audio Serial Interface
The audio serial interface for the PCM1774 comprises LRCK, BCK, DIN, and DOUT. Sampling rate (fS), left and
right channel are present on LRCK. DIN receives the serial data for the DAC interpolation filter, and DOUT
transmits the serial data from the ADC decimation filter. BCK clocks the transfer of serial audio data on DIN and
DOUT in its high-to-low transition. BCK and LRCK should be synchronized with audio system clock. Ideally, it is
recommended that they be derived from it.
The PCM1774 requires LRCK to be synchronized with the system clock. The PCM1774 does not require a
specific phase relationship between LRCK and the system clock.
The PCM1774 has both master mode and slave mode interface formats, which can be selected by register 84
(MSTR). In master mode, the PCM1774 generates LRCK and BCK from the system clock.
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Audio Data Formats and Timing
The PCM1774 supports I2S, right-justified, left-justified, and DSP formats. The data formats are shown in
Figure 16 and are selected using registers 70 and 81 (RFM[1:0], PFM[1:0]). All formats require binary
2s-complement, MSB-first audio data. The default format is I2S. Figure 14 shows a detailed timing diagram.
LRCK
50% of VIO
tw(BCH)
t(LB)
tw(BCL)
BCK
50% of VIO
tw(BCY)
t(BL)
DIN
50% of VIO
t(DS)
t(DH)
PARAMETERS
t(BCY)
BCK pulse cycle time (I2S, left- and right-justified formats)
MIN
MAX
UNITS
1/(64 fS) (1)
1/(256 fS) (1)
BCK pulse cycle time (DSP format)
tw(BCH)
BCK high-level time
35
ns
tw(BCL)
BCK low-level time
35
ns
t(BL)
BCK rising edge to LRCK edge
10
ns
t(LB)
LRCK edge to BCK rising edge
10
ns
t(DS)
DIN set up time
10
ns
t(DH)
DIN hold time
10
tr
Rising time of all signals
10
ns
tf
Falling time of all signals
10
ns
(1)
16
fS is the sampling frequency.
Figure 14. Audio Interface Timing (Slave Mode)
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t(SCY)
50% of VIO
SCKI
t(DL)
50% of VIO
LRCK
(Output)
tw(BCL)
tw(BCH)
t(DB)
t(DB)
50% of VIO
BCK
(Output)
t(BCY)
50% of VIO
DIN
t(DS)
t(DH)
PARAMETERS
MIN
MAX
UNIT
1/(256 fS) (1)
t(SCY)
SCKI pulse cycle time
t(DL)
LRCK edge from SCKI rising edge
t(DB)
BCK edge from SCKI rising edge
t(BCY)
BCK pulse cycle time
tw(BCH)
BCK high level time
146
ns
tw(BCL)
BCK low level time
146
ns
t(DS)
DATA setup time
10
ns
t(DH)
DATA hold time
10
ns
(1)
0
40
ns
0
40
ns
1/(64 fS) (1)
fS is up to 48 kHz. fS is the sampling frequency.
Figure 15. Audio Interface Timing (Master Mode)
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(a) Right-Justified Data Format; L-Channel = HIGH, R-Channel = LOW, LRPC = 0
1/fS
LRCK
L-Channel
R-Channel
BCK
(= 32 fS, 48 fS, or 64 fS)
16-Bit Right-Justified
DIN
14 15 16
1
2
3
14 15 16
MSB
(b)
I2S
1
LSB
2
3
14 15 16
MSB
LSB
Data Format; L-Channel = LOW, R-Channel = HIGH, LRPC = 0
1/fS
LRCK
L-Channel
R-Channel
BCK
(= 32 fS, 48 fS or 64 fS)
DIN
1
2
3
14 15 16
MSB
1
LSB
2
3
14 15 16
MSB
1
2
LSB
(c) Left-Justified Data Format; L-Channel = HIGH, R-Channel = LOW, LRPC = 0
1/fS
LRCK
L-Channel
R-Channel
BCK
(= 32 fS, 48 fS, or 64 fS)
DIN
1
2
3
MSB
14 15 16
1
LSB
2
3
14 15 16
MSB
1
2
1
2
1
2
LSB
(d) Burst BCK Interface Format in Master Mode; L-Channel = HIGH, R-Channel = LOW, LRPC = 0
1/fS
LRCK
L-Channel
R-Channel
BCK
( 64 fS )
DIN
1
2
3
MSB
14 15 16
1
LSB
2
3
14 15 16
MSB
LSB
(e) DSP Format, LRPC = 0
1/fS
LRCK
BCK
(= 32 fS, 48 fS, 64 fS , 128 fS or 256 fS)
DIN
1
2
3
MSB
14 15 16
LSB
1
2
3
MSB
14 15 16
LSB
T0009-07
NOTE: All audio interface formats support BCK = 64 fS in master mode (register 69, MSTR = 1). The fS of BCK at setting
multi-sampling rate (register 85 and 86, NPR[5:0] and MSR[2:0]) is shown in Table 9 and Table 10.
Figure 16. Audio Data Formats
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THREE-WIRE INTERFACE (SPI, MODE (PIN 28) = LOW)
All write operations for the serial control port use 16-bit data words. Figure 17 shows the control data word
format. The most-significant bit must be 0. There are seven bits, labeled IDX[6:0], that set the register address
for the write operation. The least-significant eight bits, D[7:0], contain the data to be written to the register
specified by IDX[6:0].
Figure 18 shows the functional timing diagram for writing to the serial control port. To write the data into the
mode register, the data is clocked into an internal shift register on the rising edge of the MC clock. The serial
data should change on the falling edge of the MC clock, and MS should be LOW during write mode. The rising
edge of MS should be aligned with the falling edge of the last MC clock pulse in the 16-bit frame. MC can run
continuously between transactions while MS is in the LOW state.
LSB
MSB
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
D7
D6
D5
Register Index (or Address)
D4
D3
D2
D1
D0
Register Data
R0001-01
Figure 17. Control Data Word Format for MD
(1) Single Write Operation
16 Bits
MS
MC
MD
MSB
LSB
MSB
(2) Continuous Write Operation
8 Bits x N Frames
MS
MC
MD
MSB
LSB MSB
Register Index
8 Bits
LSB MSB
Register (N) Data
LSB
Register (N+1) Data
MSB
LSB
Register (N+2) Data
N Frames
T0012-03
Figure 18. Register Write Operation
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Three-Wire Interface (SPI) Timing Requirements
Figure 19 shows a detailed timing diagram for the serial control interface. These timing parameters are critical
for proper control port operation.
tw(MHH)
MS
50% of VIO
t(MLS)
tw(MCL)
tw(MCH)
t(MLH)
MC
50% of VIO
t(MCY)
LSB
MD
50% of VIO
t(MDS)
t(MDH)
T0013-08
PARAMETERS
MIN
TYP
MAX
UNIT
500 (1)
ns
MC low level time
50
ns
MC high level time
50
ns
tw(MHH)
MS high level time
See (1)
ns
t(MLS)
MS falling edge to MC rising edge
50
ns
t(MLH)
MS hold time
20
ns
t(MDH)
MD hold time
15
ns
t(MDS)
MD setup time
20
ns
t(MCY)
MC pulse cycle time
tw(MCL)
tw(MCH)
(1)
3/(128 fS) s (min), where fS is sampling rate.
Figure 19. SPI Interface Timing
TWO-WIRE INTERFACE [I2C, MODE (PIN 28) = HIGH]
The PCM1774 supports the I2C serial bus and the data transmission protocol for the I2C standard as a slave
device. This protocol is explained in I2C specification 2.0.
In I2C mode, the control terminals are changed as follows.
TERMINAL NAME
PROPERTY
DESCRIPTION
MS/ADR
Input
I2C address
MD/SDA
Input/output
I2C data
MC/SCL
Input
I2C clock
SLAVE ADDRESS
MSB
1
LSB
0
0
0
1
1
ADR
R/W
The PCM1774 has its own 7-bit slave address. The first six bits (MSBs) of the slave address are factory preset
to 100011. The last bit of the address byte is the device select bit, which can be user-defined by the ADR
terminal. A maximum of two PCM1774 can be connected on the same bus at one time. The PCM1774 responds
when it receives its own slave address.
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Packet Protocol
The master device must control packet protocol, which consists of start condition, slave address with read/write
bit, data (if write) or acknowledgment (if read), and stop condition. The PCM1774 supports only slave receiver
and slave transmitter.
SDA
SCL
1−7
8
9
1−8
9
1−8
9
Slave Address
R/W
ACK
DATA
ACK
DATA
ACK
St
Sp
R/W: Read Operation if 1; Otherwise, Write Operation
ACK: Acknowledgement of a Byte if 0
DATA: 8 Bits (Byte)
Start
Condition
Stop
Condition
Write Operation
Transmitter
M
M
M
S
M
S
M
S
M
Data Type
St
Slave Address
R/W
ACK
DATA
ACK
DATA
ACK
Sp
Read Operation
Transmitter
M
M
M
S
S
M
S
M
M
Data Type
St
Slave Address
R/W
ACK
DATA
ACK
DATA
NACK
Sp
M: Master Device
St: Start Condition
S: Slave Device
Sp: Stop Condition
T0049-03
Figure 20. Basic I2C Framework
WRITE OPERATION
The master can write any PCM1774 registers in a single access. The master sends a PCM1774 slave address
with a write bit, a register address, and data. When undefined registers are accessed, the PCM1774 does not
send any acknowledgment. Figure 21 shows a diagram of the write operation.
Transmitter
M
M
M
S
M
S
M
S
M
Data Type
St
Slave Address
W
ACK
Reg Address
ACK
Write Data
ACK
Sp
M: Master Device S: Slave Device
St: Start Condition W: Write ACK: Acknowledge Sp: Stop Condition
R0002-01
Figure 21. Framework for Write Operation
READ OPERATION
The master can read PCM1774 register. The value of the register address is stored in an indirect index register
in advance. The master sends a PCM1774 slave address with a read bit after storing the register address. Then
the PCM1774 transfers the data which the index register specifies. Figure 22 shows a diagram of the read
operation.
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Transmitter
M
M
M
S
M
S
M
M
M
S
Data Type
St
Slave Address
W
ACK
Reg Address
ACK
Sr
Slave Address
R
ACK
S
M
M
Read Data NACK Sp
M: Master Device S: Slave Device St: Start Condition
Sr: Repeated Start Condition ACK: Acknowledge Sp: Stop Condition NACK: Not Acknowledge
W: Write R: Read
R0002-02
NOTE: The slave address after the repeated start condition must be the same as the previous slave address.
Figure 22. Read Operation
Timing Diagram
PARAMETERS
CONDITIONS
MIN
MAX
UNIT
100
kHz
fSCL
SCL clock frequency
Standard
t(BUF)
Bus free time between a STOP and START condition
Standard
4.7
μs
t(LOW)
Low period of the SCL clock
Standard
4.7
μs
t(HI)
High period of the SCL clock
Standard
4
μs
t(RS-SU)
Setup time for START condition
Standard
4.7
μs
t(S-HD)
Hold time for START condition
Standard
4
μs
t(D-SU)
Data setup time
Standard
250
t(D-HD)
Data hold time
Standard
0
900
ns
t(SCL-R)
Rise time of SCL signal
Standard
20 + 0.1 CB
1000
ns
t(SCL-R1)
Rise time of SCL signal after a repeated START condition and
after an acknowledge bit
Standard
20 + 0.1 CB
1000
ns
t(SCL-F)
Fall time of SCL signal
Standard
20 + 0.1 CB
1000
ns
t(SDA-R)
Rise time of SDA signal
Standard
20 + 0.1 CB
1000
ns
t(SDA-F)
Fall time of SDA signal
Standard
20 + 0.1 CB
1000
t(P-SU)
Setup time for STOP condition
Standard
4
CB
Capacitive load for SDA and SCL line
t(SP)
Pulse duration of suppressed spike
2
Figure 23. I C Interface Timing
22
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ns
μs
400
pF
25
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USER-PROGRAMMABLE MODE CONTROLS
Register Map
The mode control register map is shown in Table 6. Each register includes an index (or address) indicated by
the IDX[6:0] bits.
Table 6. Mode Control Register Map
REGISTER
IDX[6:0]
(B14–B8)
Register 64
40h
Volume for HPA (L-ch)
RSV
HMUL
HLV5
HLV4
HLV3
HLV2
HLV1
HLV0
Register 65
41h
Volume for HPA (R-ch)
RSV
HMUR
HRV5
HRV4
HRV3
HRV2
HRV1
HRV0
Register 68
44h
DAC digital attenuation and soft mute (L-ch)
RSV
PMUL
ATL5
ATL4
ATL3
ATL2
ATL1
ATL0
Register 69
45h
DAC digital attenuation and soft mute (R-ch)
RSV
PMUR
ATR5
ATR4
ATR3
ATR2
ATR1
ATR0
Register 70
46h
DAC over sampling, de-emphasis, audio interface, DGC
DEM1
DEM0
PFM1
PFM0
SPX1
SPX0
RSV
OVER
Register 72
48h
Analog mixer power up/down
RSV
RSV
RSV
RSV
RSV
RSV
PMXR
PMXL
Register 73
49h
DAC and HPA power up/down
PBIS
PDAR
PDAL
RSV
PHPR
PHPL
RSV
RSV
Register 74
4Ah
Analog output configuration select
RSV
RSV
RSV
RSV
HPS1
HPS0
RSV
PCOM
Register 75
4Bh
HPA insertion detection, short protection
RSV
RSV
RSV
RSV
SDHR
SDHL
RSV
RSV
Register 77
4Dh
Shut down status read back
RSV
RSV
RSV
RSV
STHR
STHL
RSV
RSV
Register 82
52h
PG1, 2, 5, 6, power up/down
RSV
RSV
PAIR
PAIL
RSV
RSV
RSV
RSV
Register 84
54h
Master mode
RSV
RSV
RSV
RSV
RSV
MSTR
RSV
BIT0
Register 85
55h
System reset, sampling rate control, data swap
SRST
LRPC
NPR5
NPR4
NPR3
NPR2
NPR1
NPR0
Register 86
56h
BCK configuration, sampling rate control, zero-cross
MBST
MSR2
MSR1
MSR0
RSV
RSV
RSV
ZCRS
Register 87
57h
Analog input select (MUX1, 2, 3, 4)
RSV
RSV
AIR1
AIR0
RSV
RSV
AIL1
AIL0
Register 88
58h
Analog mixing switch (SW1, 2, 3, 4, 5, 6)
RSV
MXR2
MXR1
MXR0
RSV
MXL2
MXL1
MXL0
Register 89
59h
Analog to analog path (PG5, 6) gain
RSV
GMR2
GMR1
GMR0
RSV
GML2
GML1
GML0
Register 90
5Ah
Microphone boost
RSV
RSV
RSV
RSV
RSV
RSV
G20R
G20L
Register 92
5Ch
Bass boost gain level
LPAE
RSV
RSV
LGA4
LGA3
LGA2
LGA1
LGA0
Register 93
5Dh
Middle boost gain level
RSV
RSV
RSV
MGA4
MGA3
MGA2
MGA1
MGA0
Register 94
5Eh
Treble boost gain level
RSV
RSV
RSV
HGA4
HGA3
HGA2
HGA1
HGA0
Register 95
5Fh
Sound effect source select, 3D sound
RSV
3DEN
RSV
3FL0
3DP3
3DP2
3DP1
3DP0
Register 96
60h
digital monaural mixing
RSV
RSV
RSV
RSV
RSV
RSV
RSV
MXEN
Register 124
7Ch
PG1/PG2 additional Gain
RSV
RSV
RSV
RSV
RSV
RSV
G12R
G12L
Register 125
7Dh
Power up/down time control
RSV
PTM1
PTM0
RES4
RES3
RES2
RES1
RES0
HPA: Headphone amplifier
PGx: Analog input buffer
DESCRIPTION
B7
B6
B5
B4
B3
B2
B1
B0
DAC: D/A converter
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Register Definitions
Registers 64 and 65
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
Register 64
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
HMUL
HLV5
HLV4
HLV3
HLV2
HLV1
HLV0
Register 65
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
HMUR
HRV5
HRV4
HRV3
HRV2
HRV1
HRV0
IDX[6:0]: 100 0000b (40h): Register 64
IDX[6:0]: 100 0001b (41h): Register 65
HMUL: Analog Mute Control for HPL (Line or Headphone L-Channel)
HMUR: Analog Mute Control for HPR (Line or Headphone R-Channel)
Default value: 1
HPOL/LOL and HPOR/LOR can be independently muted to zero level when HMUL and HMUR = 1. These
settings take precedence over analog volume level settings.
HMUL, HMUR = 0
Mute disabled
HMUL, HMUR = 1
Mute enabled (default)
HLV[5:0]: Analog Volume for HPL (Headphone L-Channel)
HRV[5:0]: Analog Volume for HPR (Headphone R-Channel)
Default value: 00 0000.
HPOL/LOL and HPOR/LOR can be independently controlled between 6 dB and –70 dB, with step size
depending on the gain level as shown in Table 7. Outputs may have zipper noise while changing levels. This
noise can be reduced by selecting zero-cross detection (register 86, ZCRS).
Table 7. Headphone Gain Level Setting
HLV[5:0],
HRV[5:0]
24
STEP
GAIN LEVEL
SETTING
HLV[5:0],
HRV[5:0]
STEP
GAIN LEVEL
SETTING
HLV[5:0],
HRV[5:0]
STEP
GAIN LEVEL
SETTING
11 1111
3F
6 dB
10 1001
29
–5 dB
01 0011
13
11 1110
3E
5.5 dB
10 1000
28
–5.5 dB
01 0010
12
11 1101
3D
5 dB
10 0111
27
–6 dB
01 0001
11
11 1100
3C
4.5 dB
10 0110
26
–6.5 dB
01 0000
10
–24 dB
11 1011
3B
4 dB
10 0101
25
–7 dB
00 1111
0F
–26 dB
11 1010
3A
3.5 dB
10 0100
24
–7.5 dB
00 1110
0E
–28 dB
11 1001
39
3 dB
10 0011
23
–8 dB
00 1101
0D
–30 dB
11 1000
38
2.5 dB
10 0010
22
–8.5 dB
00 1100
0C
11 0111
37
2 dB
10 0001
21
–9 dB
00 1011
0B
11 0110
36
1.5 dB
10 0000
20
–9.5 dB
00 1010
0A
–36 dB
11 0101
35
1 dB
01 1111
1F
–10 dB
00 1001
09
–38 dB
11 0100
34
0.5 dB
01 1110
1E
–10.5 dB
00 1000
08
–40 dB
11 0011
33
0 dB
01 1101
1D
–11 dB
00 0111
07
–42 dB
11 0010
32
–0.5 dB
01 1100
1C
–12 dB
00 0110
06
–46 dB
11 0001
31
–1 dB
01 1011
1B
–13 dB
00 0101
05
–50 dB
11 0000
30
–1.5 dB
01 1010
1A
–14 dB
00 0100
04
10 1111
2F
–2 dB
01 1001
19
–15 dB
00 0011
03
10 1110
2E
–2.5 dB
01 1000
18
–16 dB
00 0010
02
–62 dB
10 1101
2D
–3 dB
01 0111
17
–17 dB
00 0001
01
–66 dB
10 1100
2C
–3.5 dB
01 0110
16
–18 dB
00 0000
00
–70 dB
10 1011
2B
–4 dB
01 0101
15
–19 dB
10 1010
2A
–4.5 dB
01 0100
14
–20 dB
0.5 dB
0.5 dB
1 dB
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–21 dB
1 dB
–22 dB
–23 dB
–32 dB
2 dB
–34 dB
–54 dB
4 dB
–58 dB
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SLAS551 – JULY 2007
Registers 68 and 69
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
Register 68
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
PMUL
ATL5
ATL4
ATL3
ATL2
ATL1
ATL0
Register 69
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
PMUR
ATR5
ATR4
ATR3
ATR2
ATR1
ATR0
IDX[6:0]: 100 0100b (44h): Register 68
IDX[6:0]: 100 0101b (45h): Register 69
PMUL: Digital Soft Mute Control for DAL (DAC L-Channel)
PMUR: Digital Soft Mute Control for DAR (DAC R-Channel)
Default value: 0
The digital inputs of the DAC can be independently muted by setting PMUL and PMUR = 1. The digital data is
changed from the current attenuation level to mute level by a 1-dB step for every 8/fS time period. When PMUL
and PMUR are set to 0, the digital data is changed from the mute level to the current attenuation level by a 1-dB
step for every 8/fS time period. In the PCM1774, audible zipper noise can be reduced by selecting zero-cross
detection (register 86, ZCRS).
PMUL, PMUR = 0
Mute disabled (default)
PMUL, PMUR = 1
Mute enabled
ATL[5:0]: Digital Attenuation Setting for DAL (DAC L-Channel)
ATR[5:0]: Digital Attenuation Setting for DAR (DAC R-Channel)
Default value: 11 1111b
The digital inputs of the DAC can be independently attenuated. The attenuation of the digital input is changed by
a 1-dB step for every 8/fS time period. Audible zipper noise in the PCM1774 can be reduced by selecting
zero-cross detection (register 86, ZCRS).
Table 8. Digital Attenuation Setting
ATL[5:0],
ATR[5:0]
ATTENUATION LEVEL
SETTING
ATL[5:0],
ATR[5:0]
ATTENUATION LEVEL
SETTING
ATL[5:0],
ATR[5:0]
ATTENUATION LEVEL
SETTING
11 1111
3F
0 dB (default)
10 1001
29
–22 dB
01 0011
13
–44 dB
11 1110
3E
–1 dB
10 1000
28
–23 dB
01 0010
12
–45 dB
11 1101
3D
–2 dB
10 0111
27
–24 dB
01 0001
11
–46 dB
11 1100
3C
–3 dB
10 0110
26
–25 dB
01 0000
10
–47 dB
11 1011
3B
–4 dB
10 0101
25
–26 dB
00 1111
0F
–48 dB
11 1010
3A
–5 dB
10 0100
24
–27 dB
00 1110
0E
–49 dB
11 1001
39
–6 dB
10 0011
23
–28 dB
00 1101
0D
–50 dB
11 1000
38
–7 dB
10 0010
22
–29 dB
00 1100
0C
–51 dB
11 0111
37
–8 dB
10 0001
21
–30 dB
00 1011
0B
–52 dB
11 0110
36
–9 dB
10 0000
20
–31 dB
00 1010
0A
–53 dB
11 0101
35
–10 dB
01 1111
1F
–32 dB
00 1001
09
–54 dB
11 0100
34
–11 dB
01 1110
1E
–33 dB
00 1000
08
–55 dB
11 0011
33
–12 dB
01 1101
1D
–34 dB
00 0111
07
–56 dB
11 0010
32
–13 dB
01 1100
1C
–35 dB
00 0110
06
–57 dB
11 0001
31
–14 dB
01 1011
1B
–36 dB
00 0101
05
–58 dB
11 0000
30
–15 dB
01 1010
1A
–37 dB
00 0100
04
–59 dB
10 1111
2F
–16 dB
01 1001
19
–38 dB
00 0011
03
–60 dB
10 1110
2E
–17 dB
01 1000
18
–39 dB
00 0010
02
–61 dB
10 1101
2D
–18 dB
01 0111
17
–40 dB
00 0001
01
–62 dB
10 1100
2C
–19 dB
01 0110
16
–41 dB
00 0000
00
Mute
10 1011
2B
–20 dB
01 0101
15
–42 dB
10 1010
2A
–21 dB
01 0100
14
–43 dB
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Register 70
Register 70
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
DEM1
DEM0
PFM1
PFM0
SPX1
SPX0
RSV
OVER
IDX[6:0]: 100 0110b (46h): Register 70
DEM[1:0]: De-Emphasis Filter Selection
Default value: 00
A digital de-emphasis filter is in front of the interpolation filter. One of three de-emphasis filters can be selected
corresponding to the sampling rate, 32 kHz, 44.1 kHz, or 48 kHz.
DEM[1:0]
De-Emphasis Filter Selection
00
OFF (default)
01
32 kHz
10
44.1 kHz
11
48 kHz
PFM[1:0]: Audio Interface Selection for DAC (Digital Input)
Default value: 00
The audio interface for the DAC digital input has I2S, right-justified, left-justified, and DSP formats.
PFM[1:0]
Audio Interface Selection for DAC Digital Input
00
I2S format (default)
01
Right-justified format
10
Left-justified format
11
DSP format
SPX[1:0]: Digital Gain Control for DAC Input
Default value: 00
These bits are used to gain up the digital input data.
SPX[1:0]
Digital Gain Control for DAC input
00
0 dB (default)
01
6 dB
10
12 dB
11
18 dB
OVER: Oversampling Control for Delta-Sigma DAC
Default value: 0
This bit is used to control the oversampling rate of delta-sigma DAC. When the PCM1774 operates at low
sampling rates (less than 24 kHz) and the SCKI frequency is less than 12.5 MHz, OVER = 1 is recommended.
OVER = 0
128 fS (default)
OVER = 1
192 fS, 256 fS, 384 fS
26
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Register 72
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
RSV
RSV
RSV
PMXR
PMXL
Register 72
IDX[6:0]: 100 1000b (48h) Register 72
PMXR: Power Up/Down for MXR (Mixer R-Channel)
PMXL: Power Up/Down for MXL (Mixer L-Channel)
Default value: 0
These bits are used to control power up/down for the analog mixer.
PMXL, PMXR = 0
Power down (default)
PMXL, PMXR = 1
Power up
Register 73
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
PBIS
PDAR
PDAL
RSV
PHPR
PHPL
RSV
RSV
Register 73
IDX[6:0]: 100 1001b (49h): Register 73
PBIS: Power Up/Down Control for Bias
Default value: 0
This bit is used to control power up/down for the analog bias circuit.
PBIS = 0
Power down (default)
PBIS = 1
Power up
PDAR: Power Up/Down Control for DAR (DAC and R-Channel Digital Filter)
PDAL: Power Up/Down Control for DAL (DAC and L-Channel Digital Filter)
Default value: 0
These bits are used to control power up/down for the DAC and interpolation filter.
PDAR, PDAL = 0
Power down (default)
PDAR, PDAL = 1
Power up
PHPR: Power Up/Down Control for HPR (Line or R-Channel Headphone Output)
PHPL: Power Up/Down Control for HPL (Line or L-Channel Headphone Output)
Default value: 0
These bits are used to control power up/down for the headphone amplifier.
PHPR, PHPL = 0
Power down (default)
PHPR, PHPL = 1
Power up
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Register 74
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
RSV
HPS1
HPS0
RSV
PCOM
Register 74
IDX[6:0]: 100 1010b (4Ah): Register 74
HPS[1:0]: Line or Headphone Output Configuration
Default value: 00
HPOL/LOL and HPOR/LOR can be configured selected as follows.
HPS[1:0]
Line or Headphone Output Configuration
00
Stereo output (default)
01
Single monaural output
10
Differential monaural output
11
Reserved
PCOM: Power Up/Down Control for VCOM
Default value: 0
This bit is used to control power up/down for VCOM.
PCOM = 0
Power down (default)
PCOM = 1
Power up
Register 75
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
RSV
SDHR
SDHL
RSV
RSV
Register 75
IDX[6:0]: 100 1011b (4Bh): Register 75
SDHR: Short Protection Status for HPR (R-Channel Headphone)
SDHL: Short Protection Status for HPL (L-Channel Headphone)
Default value: 0
Short-circuit protection can be disabled if this function is not needed in an application.
SDHR, SDSL = 0
Enabled (default)
SDHR, SDHL = 1
Disabled
Register 77
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
RSV
STHR
STHL
RSV
RSV
Register 77
IDX[6:0]: 100 1101b (4Dh): Register 77
STHR: Short Protection Status for HPR (R-Channel Headphone)
STHL: Short Protection Status for HPL (L-Channel Headphone)
These bits can be used to read short protection status through the I2C interface.
STHR, STHL = 0
Detect short circuit
STHR, STHL = 1
Not detect short circuit
28
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Register 82
Register 82
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
PAIR
PAIL
RSV
RSV
RSV
RSV
IDX[6:0]: 101 0010b (52h): Register 82
PAIR: Power Up/Down for PG2 and PG6 (Gain Amplifier for R-Channel Analog Input)
PAIL: Power Up/Down for PG1 and PG5 (Gain Amplifier for L-Channel Analog Input)
Default value: 0
These bits are used to control power up/down for PG2 and PG6 (gain amplifier for analog input).
PAIR, PAIL = 0
Power down (default)
PAIR, PAIL = 1
Power up
Registers 84–86
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
Register 84
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
RSV
RSV
MSTR
RSV
BIT0
Register 85
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
SRST
LRPC
NPR5
NPR4
NPR3
NPR2
NPR1
NPR0
Register 86
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
MBST
MSR2
MSR1
MSR0
RSV
RSV
RSV
ZCRS
IDX[6:0]: 101 0100b (54h): Register 84
IDX[6:0]: 101 0101b (55h): Register 85
IDX[6:0]: 101 0110b (56h): Register 86
MSTR: Master or Slave Selection for Audio Interface
Default value: 0
This bit is used to select either master or slave mode for the audio interface. In master mode, the PCM1774
generates LRCK and BCK from the system clock. In slave mode, it receives LRCK and BCK from another
device.
MSTR = 0
Slave interface (default)
MSTR = 1
Master interface
BIT0: Bit Length Selection for Audio Interface
Default value: 1
This bit is used to select the data bit length for DAC input.
BIT0 = 0
Reserved
BIT0 = 1
16 bits (default)
SRST: System Reset
Default value: 0
This bit is used to enable system reset. All circuits are reset by setting SRST = 1. After completing the reset
sequence, SRST is set to 0 automatically.
SRST = 0
Reset disabled (default)
SRST = 1
Reset enabled
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LRPC: LRCK Polarity Control
Default value: 0
This bit is used to reverse L-channel and R-channel audio data.
LRPC = 0
Normal (default)
LRPC = 1
Reverse
NPR[5:0]: System Clock Rate Selection
Default value: 000000
MSR[2:0]: System Clock Dividing Rate Selection in Master Mode (Register 70)
Default value: 000
These bits are used to select the system clock rate and the dividing rate of the input system clock. See Table 9
for the details.
30
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Table 9. System Clock Frequency for Common-Audio Clock
SYSTEM CLOCK
SCK (MHz)
ADC SAMPLING RATE
ADC fS (kHz)
6.144
8.192
12.288
18.432
5.6448
11.2896
(1)
DAC SAMPLING RATE
DAC fS (kHz)
REGISTER SETTINGS (1)
MSR[2:0]
NPR[5:0]
BIT CLOCK
BCK (fS)
24 (SCK/256)
010
00 0000
64
16 (SCK/384)
011
00 0000
64
12 (SCK/512)
100
00 0000
64
8 (SCK/768)
101
00 0000
64
6 (SCK/1024)
110
00 0000
64
4 (SCK/1536)
111
00 0000
64
32 (SCK/256)
010
00 0000
64
16 (SCK/512)
100
00 0000
64
8 (SCK/1024)
110
00 0000
64
48 (SCK/256)
010
00 0000
64
32 (SCK/384)
011
00 0000
64
24 (SCK/512)
100
00 0000
64
16 (SCK/768)
101
00 0000
64
12 (SCK/1024)
110
00 0000
64
8 (SCK/1536)
111
00 0000
64
48 (SCK/384)
011
00 0000
64
24 (SCK/768)
101
00 0000
64
12 (SCK/1536)
111
00 0000
64
22.05 (SCK/256)
010
00 0000
64
14.7 (SCK/384)
011
00 0000
64
11.025 (SCK/512)
100
00 0000
64
7.35 (SCK/768)
101
00 0000
64
5.5125 (SCK/1024)
110
00 0000
64
3.675 (SCK/1536)
111
00 0000
64
44.1 (SCK/256)
010
00 0000
64
29.4 (SCK/384)
011
00 0000
64
22.05 (SCK/512)
100
00 0000
64
14.7 (SCK/768)
101
00 0000
64
11.025 (SCK/1024)
110
00 0000
64
7.35 (SCK/1536)
111
00 0000
64
Other settings are reserved.
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Table 10. System Clock Frequency for Application-Specific Clock
SYSTEM CLOCK
SCK (MHz)
13.5
27
12
24
19.2
32
ADC SAMPLING RATE
ADC fS (kHz)
DAC SAMPLING RATE
DAC fS (kHz)
REGISTER SETTINGS
MSR[2:0]
NPR[5:0]
BIT CLOCK
BCK (fS)
48.214 (SCK/280)
010
00 0010
70
44.407 (SCK/304)
010
00 0001
76
32.142 (SCK/420)
010
10 0010
70
24.107 (SCK/560)
100
00 0010
70
22.203 (SCK/608)
100
00 0001
76
16.071 (SCK/840)
100
10 0010
70
12.053 (SCK/1120)
110
00 0010
70
8.035 (SCK/1680)
110
10 0010
70
48.214 (SCK/560)
010
01 0010
70
44.407 (SCK/608)
010
01 0001
76
32.142 (SCK/840)
010
11 0010
70
24.107 (SCK/1120)
100
01 0010
70
22.203 (SCK/1216)
100
01 0001
76
16.071 (SCK/1680)
100
11 0010
70
12.053 (SCK/2240)
110
01 0010
70
8.035 (SCK/3360)
110
11 0010
70
48.387 (SCK/248)
010
00 0100
62
44.117 (SCK/272)
010
00 0011
68
32.258 (SCK/372)
010
10 0100
62
24.193 (SCK/496)
100
00 0100
62
22.058 (SCK/544)
100
00 0011
68
16.129 (SCK/744)
100
10 0100
62
12.096 (SCK/992)
110
00 0100
62
8.064 (SCK/1488)
110
10 0100
62
48.387 (SCK/496)
010
01 0100
62
44.117 (SCK/544)
010
01 0011
68
32.258 (SCK/744)
010
11 0100
62
24.193 (SCK/992)
100
01 0100
62
22.058 (SCK/1088)
100
01 0011
68
16.129 (SCK/1488)
100
11 0100
62
12.096 (SCK/1984)
110
01 0100
62
8.064 (SCK/2976)
110
11 0100
62
48.484 (SCK/396)
011
00 0110
66
44.444 (SCK/432)
011
00 0101
72
32.323 (SCK/594)
011
10 0110
66
24.242 (SCK/792)
101
00 0110
66
22.222 (SCK/864)
101
00 0101
72
16.161 (SCK/1188)
101
10 0110
66
12.121 (SCK/1584)
111
00 0110
66
8.080 (SCK/2376)
111
10 0110
66
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Table 10. System Clock Frequency for Application-Specific Clock (continued)
SYSTEM CLOCK
SCK (MHz)
38.4
13
26
19.68
39.36
ADC SAMPLING RATE
ADC fS (kHz)
DAC SAMPLING RATE
DAC fS (kHz)
MSR[2:0]
NPR[5:0]
BIT CLOCK
BCK (fS)
48.484 (SCK/792)
011
01 0110
66
44.444 (SCK/864)
011
01 0101
72
32.323 (SCK/1188)
011
11 0110
66
24.242 (SCK/1584)
101
01 0110
66
22.222 (SCK/1728)
101
01 0101
72
16.161 (SCK/2376)
101
11 0110
66
12.121 (SCK/3168)
111
01 0110
66
8.080 (SCK/4752)
111
11 0110
66
47.794 (SCK/272)
010
00 1000
68
43.918 (SCK/296)
010
00 0111
74
31.862 (SCK/408)
010
10 1000
68
23.897 (SCK/544)
100
00 1000
68
21.959 (SCK/592)
100
00 0111
74
15.931 (SCK/816)
100
10 1000
68
11.948 (SCK/1088)
110
00 1000
68
7.965 (SCK/1632)
110
10 1000
68
47.794 (SCK/544)
010
01 1000
68
43.918 (SCK/592)
010
01 0111
74
31.862 (SCK/816)
010
11 1000
68
23.897 (SCK/1088)
100
01 1000
68
21.959 (SCK/1184)
100
01 0111
74
15.931 (SCK/1632)
100
11 1000
68
11.948 (SCK/2176)
110
01 1000
68
7.965 (SCK/3264)
110
11 1000
68
48.235 (SCK/408)
011
00 1010
68
44.324 (SCK/444)
011
00 1001
74
32.156 (SCK/612)
011
10 1010
68
24.117 (SCK/816)
101
00 1010
68
22.162 (SCK/888)
101
00 1001
74
16.078 (SCK/1224)
101
10 1010
68
12.058 (SCK/1632)
111
00 1010
68
8.039 (SCK/2448)
111
10 1010
68
48.235 (SCK/816)
011
01 1010
68
44.324 (SCK/888)
011
01 1001
74
32.156 (SCK/1224)
011
11 1010
68
24.117 (SCK/1632)
101
01 1010
68
22.162 (SCK/1776)
101
01 1001
74
16.078 (SCK/2448)
101
11 1010
68
12.058 (SCK/3264)
111
01 1010
68
8.039 (SCK/4896)
111
11 1010
68
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REGISTER SETTINGS
33
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SLAS551 – JULY 2007
MBST: BCK Output Configuration in Master Mode
Default value: 0
This bit is used to control the BCK output configuration in master mode. In master mode, this bit sets the BCK
output configuration to normal mode or burst mode. In normal mode (MBST = 0), the BCK clock runs
continuously. In burst mode (MBST = 1), the BCK clock runs intermittently, and the number of clock cycles per
LRCK period is reduced to equal the number of bits of audio data being transmitted. Operating in burst mode
reduces the power consumption of VIO (I/O cell power supply). This is effective in master mode (register 69
MSTR = 1).
MBST = 0
Normal mode (default)
MBST = 1
Burst mode
ZCRS: Zero-Cross for Digital Attenuation/Mute and Analog Gain Setting
Default value: 0
This bit is used to enable the zero-cross detector, which reduces zipper noise while the digital soft mute, digital
attenuation analog gain setting, or analog volume setting is being changed. If no zero-cross data is input for a
512/fS period (10.6 ms at a 48-kHz sampling rate), then a time-out occurs and the PCM1774 starts changing the
attenuation, gain, or volume level. The zero-cross detector cannot be used with continuous-zero and dc data.
ZCRS = 0
Zero-cross disabled (default)
ZCRS = 1
Zero-cross enabled
Register 87
Register 87
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
AIR1
AIR0
RSV
RSV
AIL1
AIL0
IDX[6:0]: 101 0111b (57h): Register 87
AIL0: AIN1L Selector (MUX1)
Default value: 0
This bit is used to select the analog input, AIN1L.
AIL0
AIN L-channel Select
0
Disconnect (default)
1
AIN1L
AIR0: AIN1R Selector (MUX2)
Default value: 0
This bit is used to select one of the three stereo analog inputs, AIN1R.
AIR0
AIN R-channel Select
0
Disconnect (default)
1
AIN1R
34
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Register 88
Register 88
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
MXR2
MXR1
MXR0
RSV
MXL2
MXL1
MXL0
IDX[6:0]: 101 1000b (58h): Register 88
MXR2: Mixing SW6 to MXR (R-Channel Mixing Amplifier) From L-Channel Analog Input
Default value: 0
This bit is used to mix the analog source into MXR (R-ch mixing amplifier) from the L-ch analog input.
MXR2 = 0
Disable (default)
MXR2 = 1
Enable
MXR1: Mixing SW4 to MXR (R-Channel Mixing Amplifier) From R-Channel Analog Input
Default value: 0
This bit is used to mix the analog source into MXR (R-ch mixing amplifier) from the R-ch analog input.
MXR1 = 0
Disable (default)
MXR1 = 1
Enable
MXR0: Mixing SW5 to MXR (R-Channel Mixing Amplifier) From R-Channel DAC
Default value: 0
This bit is used to mix the analog source into MXR (R-ch mixing amplifier) from the R-ch DAC.
MXR0 = 0
Disable (default)
MXR0 = 1
Enable
MXL2: Mixing SW3 to MXL (L-Channel Mixing Amplifier) From R-Channel Analog Input
Default value: 0
This bit is used to mix the analog source into MXL (L-ch mixing amplifier) from the R-ch analog input.
MXL2 = 0
Disable (default)
MXL2 = 1
Enable
MXL1: Mixing SW1 to MXL (L-Channel Mixing Amplifier) From L-Channel Analog Input
Default value: 0
This bit is used to mix the analog source into MXL (L-ch mixing amplifier) from the L-ch analog input.
MXL1 = 0
Disable (default)
MXL1 = 1
Enable
MXL0: Mixing SW2 to MXL (L-Channel Mixing Amplifier) From L-Channel DAC
Default value: 0
This bit is used to mix the analog source into MXL (L-ch mixing amplifier) from the L-ch DAC.
MXL0 = 0
Disable (default)
MXL0 = 1
Enable
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SLAS551 – JULY 2007
Register 89
Register 89
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
GMR2
GMR1
GMR0
RSV
GML2
GML1
GML0
IDX[6:0]: 101 1001b (59h): Register 89
GMR[2:0]: Gain Level Control for PG6 (Gain Amplifier for Analog Input or R-Channel Bypass)
GML[2:0]: Gain Level Control for PG5 (Gain Amplifier for Analog Input or L-Channel Bypass)
Default value: 000
These bits are used for setting the gain level of the analog source to the mixing amplifier. It is recommended to
set the gain level to avoid saturation in the analog mixer.
GMR[2:0]
GML[2:0]
Gain Level Control for PG6
Gain Level Control for PG5
000
–21 dB (default)
001
–18 dB
010
–15 dB
011
–12 dB
100
–9 dB
101
–6 dB
110
–3 dB
111
0 dB
Register 90
Register 90
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
RSV
RSV
RSV
G20R
G20L
IDX[6:0]: 101 1010b (5Ah): Register 90
G20R: 20-dB Boost for PG2 (Gain Amplifier for AIN1R)
Default value: 0
This bit is used to boost the microphone signal when the analog input is small.
G12R
(REGISTER 124)
G20R
(REGISTER 90)
PG2 GAIN
0
0
0 dB (default)
0
1
20 dB
1
0
12 dB
1
1
Reserved
G20L: 20-dB Boost for PG1 (Gain Amplifier for AIN1L)
Default value: 0
This bit is used to boost the microphone signal when the analog input is small.
36
G12L
(REGISTER 124)
G20L
(REGISTER 90)
PG1 GAIN
0
0
0 dB (default)
0
1
20 dB
1
0
12 dB
1
1
Reserved
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Register 92
Register 92
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
LPAE
RSV
RSV
LGA4
LGA3
LGA2
LGA1
LGA0
IDX[6:0]: 101 1100b (5Ch): Register 92
LPAE: Gain Adjustment for Bass Boost Gain Control
Default value: 0
A gain setting for bass boost may cause digital data saturation, depending on the input data level. Where this
could occur, LPAE can be used to set the same attenuation level as the bass boost gain level for the digital
input data.
LPAE = 0
Disable (default)
LPAE = 1
Enable
LGA[4:0]: Bass Boost Gain Control
Default value: 0 0000
These bits are used to set the bass boost gain level for the digital data. The detailed characteristics are shown in
the Typical Performance Curves.
LGA[4:0]
TONE CONTROL GAIN (BASS)
LGA[4:0]
0 0000
0 dB (default)
0 1111
0 dB
0 0011
12 dB
1 0000
–1 dB
0 0100
11 dB
1 0001
–2 dB
0 0101
10 dB
1 0010
–3 dB
0 0110
9 dB
1 0011
–4 dB
0 0111
8 dB
1 0100
–5 dB
0 1000
7 dB
1 0101
–6 dB
0 1001
6 dB
1 0110
–7 dB
0 1010
5 dB
1 0111
–8 dB
0 1011
4 dB
1 1000
–9 dB
0 1100
3 dB
1 1001
–10 dB
0 1101
2 dB
1 1010
–11 dB
0 1110
1 dB
1 1011
–12 dB
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TONE CONTROL GAIN (BASS)
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SLAS551 – JULY 2007
Register 93
Register 93
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
MGA4
MGA3
MGA2
MGA1
MGA0
IDX[6:0]: 101 1101b (5Dh): Register 93
MGA[4:0]: Middle Boost Gain Control
Default value: 0 0000
These bits are used to set the midrange boost gain level for the digital data. The detailed characteristics are
shown in the Typical Performance Curves.
MGA[4:0]
TONE CONTROL GAIN (MIDRANGE)
MGA[4:0]
0 0000
0 dB (default)
0 1111
TONE CONTROL GAIN (MIDRANGE)
0 dB
0 0011
12 dB
1 0000
–1 dB
0 0100
11 dB
1 0001
–2 dB
0 0101
10 dB
1 0010
–3 dB
0 0110
9 dB
1 0011
–4 dB
0 0111
8 dB
1 0100
–5 dB
0 1000
7 dB
1 0101
–6 dB
0 1001
6 dB
1 0110
–7 dB
0 1010
5 dB
1 0111
–8 dB
0 1011
4 dB
1 1000
–9 dB
0 1100
3 dB
1 1001
–10 dB
0 1101
2 dB
1 1010
–11 dB
0 1110
1 dB
1 1011
–12 dB
Register 94
Register 94
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
HGA4
HGA3
HGA2
HGA1
HGA0
IDX[6:0]: 101 1110b (5Eh): Register 94
HGA[4:0]: Treble Boost Gain Control (fC = 5 kHz)
Default value: 0 0000
These bits are used to set the treble boost gain level for the digital data. The detailed characteristics are shown
in the Typical Performance Curves.
38
HGA[4:0]
TONE CONTROL GAIN (TREBLE)
HGA[4:0]
0 0000
0 dB (default)
0 1111
0 dB
0 0011
12 dB
1 0000
–1 dB
0 0100
11 dB
1 0001
–2 dB
0 0101
10 dB
1 0010
–3 dB
0 0110
9 dB
1 0011
–4 dB
0 0111
8 dB
1 0100
–5 dB
0 1000
7 dB
1 0101
–6 dB
0 1001
6 dB
1 0110
–7 dB
0 1010
5 dB
1 0111
–8 dB
0 1011
4 dB
1 1000
–9 dB
0 1100
3 dB
1 1001
–10 dB
0 1101
2 dB
1 1010
–11 dB
0 1110
1 dB
1 1011
–12 dB
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TONE CONTROL GAIN (TREBLE)
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SLAS551 – JULY 2007
Register 95
Register 95
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
3DEN
RSV
3FL0
3DP3
3DP2
3DP1
3DP0
IDX[6:0]: 101 1111b (5Fh): Register 95
3DEN: 3-D Sound Effect Enable
Default value: 0
This bit is used for enabling the 3-D sound effect filter. This filter has two independently controlled parameters.
3DEN = 0
Disable (default)
3DEN = 1
Enable
3FL0: Filter Selection for 3-D Sound
Default value: 0
This bit is used for selecting from two types of filter, narrow and wide. These filters have a different 3-D
performance effect.
3FL0 = 0
Narrow (default)
3FL0 = 1
Wide
3DP[3:0]: Efficiency for 3-D Sound Effects
Default value: 0000
These bits are used for adjusting the 3-D sound efficiency. Higher percentages have greater efficiency.
3DP[3:0]
3D Sound Effect Efficiency
0000
0% (default)
0001
10%
0010
20%
0011
30%
0100
40%
0101
50%
0110
60%
0111
70%
1000
80%
1001
90%
1010
100%
1011
:
1111
Reserved
:
Reserved
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Register 96
Register 96
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
RSV
RSV
RSV
RSV
MXEN
IDX[6:0]: 110 0000b (60h): Register 96
MXEN: Digital Monaural Mixing
Default value: 0
This bit is used to enable or disable monaural mixing in the section that combines L-ch data and R-ch data.
MXEN = 0
Stereo (default)
MXEN = 1
Monaural Mixing
Register 124
Register 124
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
RSV
RSV
RSV
RSV
RSV
G12R
G12L
IDX[6:0]: 111 1100b (7Ch): Register 124
G12R: 12 dB Boost for PG2 (Gain Amp for AIN1R and AIN2R)
G12L: 12 dB Boost for PG1 (Gain Amp for AIN1L and AIN2L)
Default value: 0
This bit is used to boost a small analog signal, microphone input. See Register 90 information for the detail
settings.
40
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Register 125
Register 125
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
0
IDX6
IDX5
IDX4
IDX3
IDX2
IDX1
IDX0
RSV
PTM1
PTM0
RES4
RES3
RES2
RES1
RES0
IDX[6:0]: 111 1101b (7Dh): Register 125
PTM[1:0]: Power-Up/Down Time Control
Default value: 00
RES[4:0]: Resister Value Control
Default value: 1 1100
These bits are used to optimize audible pop noise and ramp up time for headphone output at device power
on/off.
Table 11. Power Up/Down Time Control
RES[1:0]
VCOM REGISTER VALUE
1 0000
60 kΩ
1 1000
24 kΩ
1 1100
12 kΩ
1 1110
6 kΩ
Others
Reserved
VCOM CAPACITOR
[μF]
10
4.7
2.2
1
RES[4:0]
PTM[1:0]
POWER-UP TIME
[ms]
1 1110
00
450
750
1 1100
11
900
1500
1 1000
Do not set.
–
–
1 0000
Do not set.
–
–
1 1110
01
250
400
1 1100
00
450
750
1 1000
11
900
1500
1 0000
Do not set.
–
–
1 1110
10
100
300
1 1100
01
250
400
1 1000
00
450
750
1 0000
11
900
1500
1 1110
Do not set.
–
–
1 1100
10
100
300
1 1000
01
250
400
1 0000
00
450
750
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POWER-DOWN
TIME [ms]
NOTE
Default
41
PCM1774
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SLAS551 – JULY 2007
CONNECTION DIAGRAMS
To Regulator
9
10
11
SCKI
VIO
BCK
VDD
LRCK
DIN
5
DGND
VCC
2
3
4
MODE
C1
C2
VPA
MC/SCL
1
Pull Down or Up
AGND
MS/ADR
MD/SDA
PCM1774
PGND
6
7
C4
C5
8
18
C6
17
13
12
C7
20
AIN1L
19
AIN1R
VCOM
16
C8
15
HPOL/LOL
C3
14
HPOR/LOR
Stereo
Headphone
C9
Figure 24. Connection Diagram
Table 12. Recommended External Parts
C1–C2
1–4.7 μF
C4
0.1 μF
C5, C6, C7
C8, C9
HPOL
+
1 μF
C3
1–4.7 μF
10–220 μF
HPOL
+
CL
CL
16 W
4.7 W
16 W
16 W
16 W
HPOR
HPOR
+
+
CR
CR
4.7 W
CL, CR – mF
fC – Hz
CL, CR – mF
fC – Hz
10
995
10
770
47
212
47
163
100
100
100
77
220
45
220
35
S0223-01
Figure 25. High-Pass Filter for Headphone Output
42
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PACKAGE OPTION ADDENDUM
www.ti.com
5-Oct-2007
PACKAGING INFORMATION
Orderable Device
Status (1)
Package
Type
Package
Drawing
Pins Package Eco Plan (2)
Qty
PCM1774RGPR
ACTIVE
QFN
RGP
20
3000 Green (RoHS &
no Sb/Br)
CU NIPDAU
Level-2-260C-1 YEAR
PCM1774RGPRG4
ACTIVE
QFN
RGP
20
3000 Green (RoHS &
no Sb/Br)
CU NIPDAU
Level-2-260C-1 YEAR
PCM1774RGPT
ACTIVE
QFN
RGP
20
250
Green (RoHS &
no Sb/Br)
CU NIPDAU
Level-2-260C-1 YEAR
PCM1774RGPTG4
ACTIVE
QFN
RGP
20
250
Green (RoHS &
no Sb/Br)
CU NIPDAU
Level-2-260C-1 YEAR
Lead/Ball Finish
MSL Peak Temp (3)
(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
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Addendum-Page 1
PACKAGE MATERIALS INFORMATION
www.ti.com
4-Oct-2007
TAPE AND REEL BOX INFORMATION
Device
Package Pins
Site
Reel
Diameter
(mm)
Reel
Width
(mm)
A0 (mm)
B0 (mm)
K0 (mm)
P1
(mm)
W
Pin1
(mm) Quadrant
PCM1774RGPR
RGP
20
SITE 41
330
12
4.3
4.3
1.5
8
12
Q2
PCM1774RGPT
RGP
20
SITE 41
180
12
4.3
4.3
1.5
8
12
Q2
Pack Materials-Page 1
PACKAGE MATERIALS INFORMATION
www.ti.com
4-Oct-2007
Device
Package
Pins
Site
Length (mm)
Width (mm)
PCM1774RGPR
RGP
20
SITE 41
346.0
346.0
29.0
PCM1774RGPT
RGP
20
SITE 41
190.0
212.7
31.75
Pack Materials-Page 2
Height (mm)
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