Fully programmable 28-bit audio processor for enhanced
ATV sound—default audio processing flow loaded on reset
Implements Analog Devices, Inc. and third-party branded
audio algorithms
Adjustable digital delay line for audio/video
Synchronization for up to 200 ms stereo delay
High performance 24-bit ADC and DAC
94 dB DNR performance on DAC channels
95 dB DNR performance on ADC channels
Headphone output with integrated amplifiers
High performance pulse-width modulation (PWM) digital
outputs
Multichannel digital baseband I/O
4 stereo synchronous digital I
One 6-channel sample rate converter (SRC) and one stereo
SRC supporting input sample rates from 5 kHz to 50 kHz
One stereo synchronous digital I
S/PDIF output with S/PDIF input mux capability
2
Fast I
C control
Operates from 3.3 V (analog), 1.8 V (digital core), and 3.3 V
(digital interface)
Available in 80-lead LQFP
APPLICATIONS
General-purpose consumer audio post processing
Home audio
DVD recorders
Home theater in a box systems and DVD receivers
Audio processing subsystems for DTV-ready TVs
Analog broadcast capability for iDTVs
2
S input channels
2
S output
ADAV4601
GENERAL DESCRIPTION
The ADAV4601 is an enhanced audio processor targeting
advanced TV applications with full support for digital and
analog baseband audio.
The audio processor, by default, loads a dedicated TV audio flow
that incorporates full matrix switching (any input to any output),
automatic volume control that compensates for volume changes
during advertisements or when switching channels, dynamic
bass, a multiband equalizer, and up to 200 ms of stereo delay
memory for audio-video synchronization.
Alternatively, Analog Devices offers an award-winning graphical
programming tool (SigmaStudio™) that allows custom flows to be
quickly developed and evaluated. This allows the creation of
customer-specific audio flows, including the use of ADI library of
third-party algorithms.
The analog I/O integrates Analog Devices proprietary continuoustime, multibit Σ- architecture to bring a higher level of
performance to ATV systems, required by third-party algorithm
providers to meet system branding certification. The analog input
is provided by 95 dB dynamic range (DNR) ADCs, and analog
output is provided by 94 dB DNR DACs.
The main speaker outputs can be supplied as a digitally modulated
PWM stream to support digital amplifiers.
The ADAV4601 includes multichannel digital inputs and outputs.
In addition, digital input channels can be routed through integrated
sample rate converters (SRC), which are capable of supporting any
arbitrary sample rate from 5 kHz to 50 kHz.
Rev. B
Information furnished by Analog Devices is believed to be accurate and reliable. However, no
responsibility is assumed by Anal og Devices for its use, nor for any infringements of p atents or other
rights of third parties that may result from its use. Specifications subject to chan ge without notice. No
license is granted by implication or otherwise under any patent or patent rights of Analog Devices.
Trademarks and registered trademarks are the property of their respective owners.
AVDD = 3.3 V, DVDD = 1.8 V, ODVDD = 3.3 V, operating temperature = −40°C to +85°C, master clock 24.576 MHz, measurement
bandwidth = 20 Hz to 20 kHz, ADC input signal = DAC output signal = 1 kHz, unless otherwise noted.
PERFORMANCE PARAMETERS
Table 1.
Parameter Min Typ Max Unit Test Conditions/Comments
REFERENCE SECTION
Absolute Voltage V
V
Temperature Coefficient 100 ppm/°C
REF
ADC SECTION
Number of Channels 2 One stereo channel
Full-Scale Input Level 100 μA rms
Resolution 24 Bits
Dynamic Range (Stereo Channel)
A-Weighted 95 dB −60 dBFS with respect to full-scale analog input
Total Harmonic Distortion + Noise
(Stereo Channel)
Gain Mismatch 0.2 dB Left- and right-channel gain mismatch
Crosstalk (Left-to-Right, Right-to-Left) −110 dB
Gain Error −1 dB Input signal is 100 μA rms
Current Setting Resistor (R
Power Supply Rejection −87 dB 1 kHz, 300 mV p-p signal at AVDD
ADC DIGITAL DECIMATOR FILTER
CHARACTERISTICS
Pass Band 22.5 kHz
Pass-Band Ripple ±0.0002 dB
Stop Band 26.5 kHz
Stop-Band Attenuation 100 dB
Group Delay 1040 μs
PWM SECTION
Frequency 384 kHz Guaranteed by design
Modulation Index 0.976 Guaranteed by design
Dynamic Range
A-Weighted 98 dB −60 dBFS with respect to full-scale code input
Total Harmonic Distortion + Noise −80 dB −3 dBFS with respect to full-scale code input
DAC SECTION
Number of Auxiliary Output Channels 6 Three stereo channels
Resolution 24 Bits
Full-Scale Analog Output 1 V rms
Dynamic Range
A-Weighted 94 dB −60 dBFS with respect to full-scale code input
Total Harmonic Distortion + Noise −86 dB −3 dBFS with respect to full-scale code input
Crosstalk (Left-to-Right, Right-to-Left) −102 dB
Interchannel Gain Mismatch 0.1 dB Left- and right-channel gain mismatch
Gain Error 0.525 dB 1 V rms output
DC Bias 1.53 V
Power Supply Rejection −90 dB 1 kHz, 300 mV p-p signal at AVDD
Output Impedance 235 Ω
1.53 V
REF
−90 dB −3 dBFS with respect to full-scale analog input
) 20 kΩ
ISET
External resistor to set current input range of ADC
for nominal 2.0 V rms input signal
At 48 kHz, guaranteed by design
Rev. B | Page 5 of 60
ADAV4601
Parameter Min Typ Max Unit Test Conditions/Comments
DAC DIGITAL INTERPOLATION FILTER
CHARACTERISTICS
Pass Band 21.769 kHz
Pass-Band Ripple ±0.01 dB
Transition Band 23.95 kHz
Stop Band 26.122 kHz
Stop-Band Attenuation 75 dB
Group Delay 580 μs
HEADPHONE AMPLIFIER Measured at headphone output with 32 Ω load
Number of Channels 2 One stereo channel
Full-Scale Output Power 31 mW rms 1 V rms output
Dynamic Range
A-Weighted 93 dB −60 dBFS with respect to full-scale code input
Total Harmonic Distortion + Noise −83 dB −3 dBFS with respect to full-scale code input
Interchannel Gain Mismatch 0.1 dB
DC Bias 1.53 V
Power Supply Rejection −85 dB 1 kHz, 300 mV p-p signal at AVDD
SRC
Number of Channels 8 Two channels (SRC1), six channels (SRC2)
Dynamic Range
Pass Band 21.678 kHz
Pass-Band Ripple 0.005 dB
Stop Band 26.232 kHz
Stop-Band Attenuation 110 dB
Group Delay 876 μs
DIGITAL INPUT/OUTPUT
Input Voltage High (VIH) 2.0 ODVDD V
Input Voltage Low (VIL) 0.8 V
Input Leakage
I
(SDIN0, SDIN1, SDIN2, SDIN3,
IH
LRCLK0, LRCLK1, LRCLK2, BCLK0,
BCLK1, BCLK2, SPDIF_OUT, SPDIF_IN)
IIH (RESET)
IIL (SDO0, SCL, SDA) −40 μA VIL = 0 V, equivalent to a 90 kΩ pull-down resistor
Output Voltage High (VOH) 2.4 V IOH = 0.4 mA
Output Voltage Low (VOL) 0.4 V IOL = −2 mA
Output Voltage High (VOH) (MCLK_OUT) 1.4 V IOH = 0.4 mA
Output Voltage Low (VOL) (MCLK_OUT) 0.4 V IOL = −3.2 mA
Input Capacitance 10 pF
SUPPLIES
Analog Supplies (AVDD) 3.0 3.3 3.6 V
Digital Supplies (DVDD) 1.65 1.8 2.0 V
Interface Supply (ODVDD) 3.0 3.3 3.6 V
Supply Currents
Analog Current 115 mA
Digital Current 160 mA
Interface Current 2 mA
At 48 kHz, guaranteed by design
At 48 kHz, guaranteed by design
40 μA V
13.5 μA V
= ODVDD, equivalent to a 90 kΩ pull-up resistor
IH
= ODVDD, equivalent to a 266 kΩ pull-up resistor
IH
MCLK = 24 MHz, ADCs and DACs active, headphone
outputs active and driving a 16 Ω load
Rev. B | Page 6 of 60
ADAV4601
Parameter Min Typ Max Unit Test Conditions/Comments
Power Dissipation 0.674 W
Standby Currents
Analog Current 7 mA
Digital Current 3 mA
Interface Current 1.6 mA
TEMPERATURE RANGE
Operating Temperature −40 +85 °C
Storage Temperature −65 +150 °C
TIMING SPECIFICATIONS
Table 2.
Parameter Description Min Max Unit Comments
MASTER CLOCK AND RESET
f
MCLKI frequency 3.072 24.576 MHz
MCLKI
tMP MCLKI period 40 325 ns
t
MCLKI high 10 ns
MCH
t
MCLKI low 10 ns
MCL
t
RESET
MASTER CLOCK OUTPUT
tCK MCLK_OUT period 8 162 ns
t
JIT
tCH MCLK_OUT high 45 55 %
tCL MCLK_OUT low 45 55 %
I2C PORT
f
SCL clock frequency 400 kHz
SCL
t
SCL high 600 ns
SCLH
t
SCL low 1.3 μs
SCLL
Start Condition
t
Setup time 600 ns Relevant for repeated start condition
SCS
t
Hold time 600 ns After this period, the first clock is generated
SCH
tDS Data setup time 100 ns
t
SCL rise time 300 ns
SCR
t
SCL fall time 300 ns
SCF
t
SDA rise time 300 ns
SDR
t
SDA fall time 300 ns
SDF
Stop Condition
t
Setup time 0 ns
SCS
SERIAL PORTS
Slave Mode
t
BCLK high 40 ns
SBH
t
BCLK low 40 ns
SBL
f
BCLK frequency 64 × fS
SBF
t
LRCLK setup 10 ns To BCLK rising edge
SLS
t
LRCLK hold 10 ns From BCLK rising edge
SLH
t
SDIN setup 10 ns To BCLK rising edge
SDS
t
SDIN hold 10 ns From BCLK rising edge
SDH
t
SDO delay 50 ns From BCLK falling edge
SDD
Master Mode
t
LRCLK delay 25 ns From BCLK falling edge
MLD
t
SDO delay 15 ns From BCLK falling edge
MDD
t
SDIN setup 10 ns From BCLK rising edge
MDS
t
SDIN hold 10 ns From BCLK rising edge
MDH
RESET low
200 ns
Period jitter 800 ps
Rev. B | Page 7 of 60
ADC, DAC, and headphone outputs floating,
low, MCLK = 24 MHz
RESET
ADAV4601
V
V
TIMING DIAGRAMS
t
= 1/
f
MP
MCLKI
MCLKI
RESET
t
RESET
Figure 2. Master Clock and Reset Timing
DVDD
t
MCH
t
MCL
07070-004
t
JIT
GND
t
CH
t
CK
t
CL
07070-035
Figure 3. Master Clock Output Timing
t
LRCLK1
BCLK1
SDINx
SDO0
LRCLK1
BCLK1
SDINx
SDO0
SLH
t
SLS
t
t
SDH
SDS
t
SDD
Figure 4. Serial Port Slave Mode Timing
t
MLD
t
t
MDH
MDS
t
MDD
Figure 5. Serial Port Master Mode Timing
DVDD
AVDD
ODVDD
07070-002
0.18V
1.0s MAX
0.33V
1.0s MAX
1.65V
3.0V
Figure 7. Power-Up Sequence Timing
DVDD
AVDD
ODVDD
07070-003
1.65V
0.18V
1.0s MAX
3.0V
0.33V
1.0s MAX
Figure 8. Power-Down Sequence Timing
1.8
0V
3.3V
0V
7070-033
1.8
0V
3.3V
0V
07070-034
100µAI
OL
TO OUTPUT
PIN
50pF
100µAI
Figure 6. Load Circuit for Digital Output Timing Specifications
ODVDD
OH
07070-032
Rev. B | Page 8 of 60
ADAV4601
ABSOLUTE MAXIMUM RATINGS
Table 3.
Parameter Rating
DVDD to DGND 0 V to 2.2 V
ODVDD to DGND 0 V to 4 V
AVDD to AGND 0 V to 4 V
AGND to DGND −0.3 V to +0.3 V
Digital Inputs DGND − 0.3 V to ODVDD + 0.3 V
Analog Inputs AGND − 0.3 V to AVDD + 0.3 V
Reference Voltage Indefinite short circuit to ground
Soldering (10 sec) 300°C
Stresses above those listed under Absolute Maximum Ratings
may cause permanent damage to the device. This is a stress
rating only; functional operation of the device at these or any
other conditions above those indicated in the operational
section of this specification is not implied. Exposure to absolute
maximum rating conditions for extended periods may affect
device reliability.
THERMAL RESISTANCE
θJA is specified for the worst-case conditions, that is, a device
soldered in a circuit board for surface-mount packages.
Thermal resistance is based on JEDEC 2S2P PCB.
Table 4.
Package Type θJA θ
80-Lead LQFP 38.1 7.6 °C/W
Unit
JC
THERMAL CONDITIONS
To ensure correct operation of the device, the case temperature
(T
) must be kept below 121°C to keep the junction temperature
ADC Ground.
ADC Supply (3.3 V).
No Connection to This Pin Allowed.
Digital Ground.
Digital Supply (1.8 V).
Active-Low Mute Request Input Signal.
I2C Data.
I2C Clock.
External Input to S/PDIF Mux/Left/Right Clock for SRC2 (Default).
External Input to S/PDIF Mux/Bit Clock for SRC2 (Default).
Digital Ground.
Digital Supply (1.8 V).
Serial Data Input 0/SRC Data Input.
Serial Data Input 1/SRC Data Input.
24 SDIN2 Serial Data Input 2/SRC Data Input.
Rev. B | Page 10 of 60
ADAV4601
Pin No. Mnemonic Description
25 SPDIF_IN0/SDIN3 External Input to S/PDIF Mux/SRC Data Input/Serial Data Input 3 (Default).
26 SPDIF_IN1/LRCLK0
27 SPDIF_IN2/BCLK0
28 ODGND Digital Ground.
29 ODVDD Digital Interface Supply (3.3 V).
30 MCLK_OUT Master Clock Output.
31 DVDD Digital Supply (1.8 V).
32 DGND Digital Ground.
33 MCLKI/XIN Master Clock/Crystal Input.
34 XOUT Crystal Output.
35 SPDIF_IN4/BCLK1 External Input to S/PDIF Mux/Bit Clock for Serial Data I/O (Default).
36 SPDIF_IN3/LRCLK1 External Input to S/PDIF Mux/Left/Right Clock for Serial Data I/O (Default).
37 SDO0/AD0 Serial Data Output. This pin acts as the I2C address select on reset. It has an internal pull-down resistor.
38 SPDIF_OUT/SDO1 Output of S/PDIF Mux/Serial Data Output.
39 PWM_READY PWM Ready Flag.
40 DVDD Digital Supply (1.8 V).
41 DGND Digital Ground.
42 PWM1A
43 PWM1B
44 PWM2A
45 PWM2B
46 PWM3A
47 PWM3B
48 PWM4A
49 PWM4B
50
No Connection to This Pin Allowed.
Left Auxiliary Output 3.
Right Auxiliary Output 3.
Left Auxiliary Output 4.
Right Auxiliary Output 4.
No Connection to This Pin Allowed.
Left Auxiliary Output 1.
Right Auxiliary Output 1.
No Connection to This Pin Allowed.
Left Auxiliary Input 1.
Right Auxiliary Input 1.
Rev. B | Page 11 of 60
ADAV4601
TYPICAL PERFORMANCE CHARACTERISTICS
0
–20
–40
–60
–80
–100
MAGNITUDE (dB)
–120
–140
–160
–180
07
192384576
FREQUENCY (kHz)
68
07070-007
Figure 10. DAC Composite Filter Response (48 kHz)
0
–30
–60
–90
–120
–150
–180
MAGNITUDE (dB)
–210
–240
–270
–300
03
12825684
FREQUENCY (kHz)
07070-010
Figure 13. ADC Composite Filter Response (48 kHz)
0
–20
–40
–60
–80
–100
MAGNITUDE (dB)
–120
–140
–160
09
244872
FREQUENCY (kHz)
6
07070-008
Figure 11. DAC Pass-Band Filter Response (48 kHz)
0.6
0.4
0.2
0
MAGNITUDE (d B)
–0.2
–0.4
0
–30
–60
–90
MAGNITUDE (dB)
–120
–150
–180
09
Figure 14. ADC Pass-Band Filter Response (48 kHz)
0.04
0.03
0.02
0.01
0
–0.01
MAGNITUDE (d B)
–0.02
–0.03
2448726
FREQUENCY (kHz)
07070-011
–0.6
02
816 4
FREQUENCY (kHz)
07070-009
Figure 12. DAC Pass-Band Ripple (48 kHz)
–0.04
02
Figure 15. ADC Pass-Band Ripple (48 kHz)
Rev. B | Page 12 of 60
816 4
FREQUENCY (kHz)
07070-012
ADAV4601
0
0
–20
–40
–60
–80
–100
MAGNITUDE (d BV)
–120
–140
–160
020000
400080001200016000
FREQUENCY (Hz)
Figure 16. DAC Dynamic Range
0
–20
–40
–60
–80
–100
MAGNITUDE (dBV)
–120
–140
–160
020000
400080001200016000
FREQUENCY (Hz)
Figure 17. DAC Total Harmonic Distortion + Noise
0
–20
–20
–40
–60
–80
–100
MAGNITUDE (dBV)
–120
–140
–160
020000
07070-013
400080001200016000
FREQUENCY (Hz)
07070-016
Figure 19. ADC Total Harmonic Distortion + Noise
0
–20
–40
–60
–80
GAIN (dB)
–100
–120
–140
010.90.80.70.60.50. 40.30.20. 1
7070-014
NORMALIZE D FREQUENCY
.0
07070-017
Figure 20. Sample Rate Converter Transfer Function
–40
–60
–80
–100
MAGNITUDE (dBV)
–120
–140
–160
020000
400080001200016000
FREQUENCY (Hz)
07070-015
Figure 18. ADC Dynamic Range
Rev. B | Page 13 of 60
ADAV4601
TERMINOLOGY
Dynamic Range
The ratio of a full-scale input signal to the integrated input noise
in the pass band (20 Hz to 20 kHz), expressed in decibels (dB).
Dynamic range is measured with a −60 dB input signal and is equal
to (S/[THD+N]) + 60 dB. Note that spurious harmonics are below
the noise with a −60 dB input; therefore, the noise level establishes
the dynamic range. The dynamic range is specified with and
without an A-weight filter applied.
Pass Band
The region of the frequency spectrum unaffected by the
attenuation of the filter of the digital decimator.
Pass-Band Ripple
The peak-to-peak variation in amplitude response from equal
amplitude input signal frequencies within the pass band,
expressed in decibels.
Stop Band
The region of the frequency spectrum attenuated by the filter
of the digital decimator to the degree specified by stop-band
attenuation.
Gain Error
With a near full-scale input, the ratio of the actual output to the
expected output, expressed in dB.
Interchannel Gain Mismatch
With identical near full-scale inputs, the ratio of the outputs of
the two stereo channels, expressed in decibels.
Crosstalk
Ratio of response on one channel with a grounded input to a
full-scale 1 kHz sine wave input on the other channel, expressed
in decibels.
Power Supply Rejection
With no analog input, the signal present at the output when a
300 mV p-p signal is applied to power supply pins, expressed in
decibels of full scale.
Group Delay
Intuitively, the time interval required for an input pulse to appear at
the output of the converter, expressed in milliseconds (ms); more
precisely, the derivative of radian phase with respect to radian
frequency at a given frequency.
Rev. B | Page 14 of 60
ADAV4601
PIN FUNCTIONS
DETAILED PIN DESCRIPTIONS
Tabl e 5 shows the pin numbers, mnemonics, and descriptions
for the ADAV4601. The input pins have a logic threshold
compatible with 3.3 V input levels.
SDIN0, SDIN1, SDIN2, and SDIN3/SPDIF_IN0
Serial data inputs. These input pins provide the digital audio
data to the signal processing core. Any of the inputs can be
routed to either of the SRCs for conversion; this input is then
not available as a synchronous input to the audio processor but
only as an input through the selected SRC. The serial format for
the synchronous data is selected by Bits[3:2] of the Serial Port
Control Register 1. If the SRCs are required, the serial format is
selected by Bits[12:9] of the same register. The synchronous inputs
are capable of using any pair of serial clocks, LRCLK0/BCLK0,
LRCLK1/BCLK1, or LRCLK2/BCLK2. By default, they use
LRCLK1 and BCLK1. See Figure 26 for more details regarding
the configuration of the synchronous inputs.
SDIN3 is a shared pin with SPDIF_IN0. If SDIN3 is not in use, this
pin can be used to connect an S/PDIF signal from an external
source, such as an MPEG decoder, to the ADAV4601 on-chip
S/PDIF output multiplexer. If SPDIF_OUT is selected from one
of the SPDIF_IN (external) signals, the signal is simply passed
through from input to output.
LRCLK0/SPDIF_IN1, BCLK0/SPDIF_IN2,
LRCLK1/SPDIF_IN3, BCLK1/SPDIF_IN4,
LRCLK2/SPDIF_IN5, and BCLK2/SPDIF_IN6
By default, LRCLK1 and BCLK1 are associated with the
synchronous inputs, LRCLK0 and BCLK0 are associated with
SRC1, and LRCLK2 and BCLK2 are associated with SRC2.
However, the SRCs and synchronous inputs can use any of
the serial clocks (see Figure 26). LRCLK0, BCLK0, LRCLK1,
BCLK1, LRCLK2, and BCLK2 are shared pins with SPDIF_IN1,
SPDIF_IN2, SPDIF_IN3, SPDIF_IN4, SPDIF_IN5, and
SPDIF_IN6, respectively. If LRCLK0/LRCLK1/ LRCLK2 or
BCLK0/BCLK1/BCLK2 are not in use, these pins can be used
to connect an S/PDIF signal from an external source, such as an
MPEG decoder, to the ADAV4601 on-chip S/PDIF output
multiplexer. If SPDIF_OUT is selected from one of the SPDIF_IN
(external) signals, the signal is simply passed through from input
to output.
SDO0/AD0
Serial data output. This pin can output two channels of digital
audio using a variety of standard 2-channel formats. The clocks
for SDO0 are always the same as those used by the synchronous
inputs; therefore, LRCLK1 and BCLK1 are used by default,
although SDO0 is capable of using any pair of serial clocks,
LRCLK0/BCLK0, LRCLK1/BCLK1, or LRCLK2/BCLK2.
The Serial Port Control Register 1 selects the serial format for the
synchronous output. On reset, the SDO0 pin duplicates as the
2
I
C® address select pin. In this mode, the logical state of the pin
is polled for four MCLKI cycles following reset. The address
select bit is set as the majority poll of the logic level of the pin after
the four MCLKI cycles.
SPDIF_OUT/SDO1
The ADAV4601 contains an S/PDIF multiplexer functionality that
allows the SPDIF_OUT signal to be chosen from an internally
generated S/PDIF signal or from the S/PDIF signal of an external
source, which is connected via one of the SPDIF_IN pins. This pin
can also be configured as an additional serial output (SDO1) as an
alternate function.
MCLKI/XIN
Master clock input. The ADAV4601 uses a PLL to generate the
appropriate internal clock for the audio processing core. A clock
signal of a suitable frequency can be connected directly to this pin,
or a crystal can be connected between MCLKI/XIN and XOUT
together with the appropriate capacitors to DGND to generate
a suitable clock signal.
XOUT
This pin is used in conjunction with MCLKI/XIN to generate a
clock signal for the ADAV4601.
MCLK_OUT
This pin can be used to output MCLKI or one of the internal
system clocks. Note that the output level of this pin is referenced to
DVDD (1.8 V) and not ODVDD (3.3 V) like all the other digital
inputs and outputs.
SDA
Serial data input for the I2C control port. SDA features a glitch
elimination filter that removes spurious pulses that are less than
50 ns wide.
Rev. B | Page 15 of 60
ADAV4601
SCL
Serial clock for the I2C control port. SCL features a glitch
elimination filter that removes spurious pulses that are
less than 50 ns wide.
MUTE
Mute input request. This active-low input pin controls the
muting of the output ports (both analog and digital) from the
ADAV4601. When low, it asserts mute on the outputs that are
enabled in the audio flow.
RESET
Active-low reset signal. After
are powered down. The blocks can be individually powered up
with software. When the part is powered up, it takes approximately
3072 internal clocks to initialize the internal circuitry. The internal
system clock is equal to MCLKI until the PLL is powered and
enabled, after which the internal system clock becomes 2560 × f
(122.88 MHz). When the PLL is powered up and enabled after
reset, it takes approximately 3 ms to lock. When the audio
processor is enabled, it takes approximately 32,768 internal system
clocks to initialize and load the default flow to the audio processor
memory. The audio processor is not available during this time.
RESET
goes high, the circuit blocks
AUXIN1L AND AUXIN1R
Analog inputs to the on-chip ADCs.
AUXOUT1L, AUXOUT1R, AUXOUT3L, AUXOUT3R,
AUXOUT4L, and AUXOUT4R
Auxiliary DAC analog outputs. These pins can be programmed
to supply the outputs of the internal audio processing for line
out or record use.
HPOUT1L and HPOUT1R
Analog outputs from the headphone amplifiers.
PLL_LF
PLL loop filter connection. A 100 nF capacitor and a 2 kΩ resistor
in parallel with a 1 nF capacitor tied to AVDD are required for
the PLL loop filter to operate correctly.
VREF
Voltage reference for DACs and ADCs. This pin is driven by an
internal 1.5 V reference voltage.
S
FILTA and FILTD
Decoupling nodes for the ADC and DAC. Decoupling
capacitors should be connected between these nodes and
AGND, typically 47 µF in parallel with 0.1 µF and 10 µF in
parallel with 0.1 µF, respectively.
PWM1A, PWM1B, PWM2A, PWM2B, PWM3A, PWM3B,
PWM4A, and PWM4B
Differential pulse-width modulation outputs are suitable for
driving Class-D amplifiers.
PWM_READY
This pin is set high when PWM is enabled and stable.
AVDD
Analog power supply pins. These pins should be connected
to 3.3 V. Each AVDD pin should be decoupled with a 0.1 µF
capacitor to AGND, as close to the pin as possible. In addition,
the ADC supply (Pin 4) and the DAC supplies (Pin 68 and Pin 71)
should share a 10 µF capacitor to ground. The PLL supply (Pin 53)
should have an additional 1 nF and 10 µF capacitor to ground,
and the headphone supply (Pin 59) should have an additional
10 µF capacitor to ground.
DVDD
Digital power supply pins. These pins should be connected to a 1.8 V
digital supply. For optimal performance, each DVDD/DGND
pair requires a 0.1 µF decoupling capacitor as close to the pin as
possible. In addition, these 0.1 µF decoupling capacitors are in
parallel with a single 10 µF capacitor.
ODVDD
Digital interface power supply pin. Connect this pin to a 3.3 V
digital supply. Decouple this pin with 10 µF and 0.1 µF capacitors to
DGND, as close to the pin as possible.
DGND
Digital ground.
AGND
Analog ground.
ODGND
Ground for the digital interface power supply.
ISET
ADC current setting resistor. See the ADC Inputs section for
more details.
Rev. B | Page 16 of 60
ADAV4601
FUNCTIONAL DESCRIPTIONS
POWER-UP SEQUENCE
The following sequence provides an overview of how to
initialize the IC:
1. Apply power to the ADAV4601.
2. Enable PLL via an I
3. Power up via an I
2
C write and wait 15 ms for PLL to lock.
2
C write to the global power-up bit in the
initialization control register (0x0000).
4. A default flow is automatically loaded on power-up. If a
user-defined flow is loaded, see the Loading a Custom
Audio Processing Flow section for additional information.
5. Depending on the I/O blocks required, other steps may
need to be taken; for example, headphone outputs may
need to be tristated. See the ADC Inputs, DAC Voltage
Outputs, PWM Outputs, Headphone Output and S/PDIF
Input/Output sections that describe the I/O blocks in detail.
6. Un mute.
MASTER CLOCK OSCILLATOR
Internally, the ADAV4601 operates synchronously to the master
MCLKI input. All internal system clocks are generated from this
single clock input using an internal PLL. This MCLKI input
can also be generated by an external crystal oscillator connected
to the MCLKI/XIN pin or by using a simple crystal oscillator
connected across MCLKI/XIN and XOUT. By default, the master
clock frequency is 24.576 MHz; however, by using the internal
dividers, an MCLKI of 12.288 MHz, 6.144 MHz, and 3.072 MHz
are also supported.
EXTERNAL CL OCK/
CRYSTAL
I2C
OSC
DIVIDER
REGISTER
Figure 21. Master Clock
Figure 22 shows the external circuit recommended for proper
operation when using a crystal oscillator. Due to the effect of
stray capacitance, consideration must be given to the value of
C1 and C2 when calculating the desired C
C+
=
LOAD
MASTER CLO CK FREQUENCY
[24.576MHz, 12.288MHz,
6.144MHz, 3.072MHz]
3.072MHz
DIVIDER W ORD
[÷8, ÷4, ÷2, ÷1]
for the crystal.
LOAD
CCCC
++
)2)(1(
pgpg
21
CCCC
+++
21
pgpg
21
C
S
PLL
REFERENCE
CLOCK
7070-018
where:
C
and C
pg1
C
is the PCB stray capacitance.
S
A good rule of thumb is to approximate
between 5 pF and 10 pF and
are the pin to ground capacitances.
pg2
C
pg1
C
to be between 2 pF and 3 pF.
S
C1
C2
Figure 22. Circuit for Crystal Resonator
XIN
XOUT
and C
07070-100
pg2
to be
I2C INTERFACE
The ADAV4601 supports a 2-wire serial (I2C compatible)
microprocessor bus driving multiple peripherals. The ADAV4601
is controlled by an external I
controller. The ADAV4601 is in slave mode on the I
during self-boot. While the ADAV4601 is self-booting, it becomes
the master, and the EEPROM, which contains the ROMs to be
booted, is the slave. When the self-boot process is complete, the
ADAV4601 reverts to slave mode on the I
should access the I
2
C bus while the ADAV4601 is self-booting
(refer to the Application Layer section and the Loading a
Custom Audio Processing Flow section).
Initially, all devices on the I
the devices monitor the SDA and SCL lines for a start condition
and the proper address. The I
establishing a start condition, defined by a high-to-low transition
on SDA while SCL remains high. This indicates that an address/
data stream follows. All devices on the bus respond to the start
condition and read the next byte (7bit address plus the R/
MSB first. The device that recognizes the transmitted address
responds by pulling the data line low during the ninth clock
pulse. This ninth bit is known as an acknowledge bit.
All other devices on the bus revert to an idle condition. The R/
bit determines the direction of the data. A Logic Level 0 on the
LSB of the first byte means the master writes information to the
peripheral. A Logic Level 1 on the LSB of the first byte means the
master reads information from the peripheral. A data transfer takes
place until a stop condition is encountered. A stop condition occurs
when SDA transitions from low to high while SCL is held high.
The ADAV4601 determines its I
the SDO0 pin after reset. Internally, the SDO0 pin is sampled by
four MCLKI edges to determine the state of the pin (high or
low). Because the pin has an internal pull-down resistor default,
the address of the ADAV4601 is 0x34 (write) and 0x35 (read).
An alternate address, 0x36 (write) and 0x37 (read), is available
by tying the SDO0 pin to ODVDD via a 10 kΩ resistor. The I
interface supports a clock frequency of up to 400 kHz.
2
C master device, such as a micro-
2
C bus. No other devices
2
C bus are in an idle state, wherein
2
C master initiates a data transfer by
2
C bus, except
W
2
C device address by sampling
W
bit)
2
C
Rev. B | Page 17 of 60
ADAV4601
Table 6. Single Word I2C Write1
S Chip address,
W
R/
= 0
1
S = start bit, P = stop bit, and AS = acknowledge by slave.
Table 7. Burst Mode I2C Write1
S Chip
address,
W
R/
= 0
1
S = start bit, P = stop bit, and AS = acknowledge by slave.
Table 8. Single Word I2C Read1
S Chip address,
W
R/
= 0
1
S = start bit, P = stop bit, AM = acknowledge by master, and AS = acknowledge by slave.
Table 9. Burst Mode I2C Read1
S Chip address,
W
= 0
R/
1
S = start bit, P = stop bit, AM = acknowledge by master, and AS = acknowledge by slave.
SCL
AS Subaddress high AS Subaddress low AS Data Byte 1 AS Data Byte 2 … AS Data Byte N P
AS Subaddress
high
AS Subaddress
AS Subaddress
AS Subaddress
high
high
low
AS Data-Word 1,
AS Subaddress
low
AS Subaddress
low
Byte 1
AS Data-Word 1,
AS S Chip address,
R/W = 1
AS S Chip address,
R/W = 1
Byte 2
AS Data-Word 2,
AS Data-Word 2,
Byte 1
AS Data Byte 1 AM Data
Byte 2
AS Data-Word 1
Byte 1
AM Data-Word 1
Byte 2
Byte 2
… AM Data
AS … P
P
Byte N
AM … P
SDA0
START BY
MASTER
SCL
(CONTINUED)
SDA
(CONTINUED)
0
FRAME 1
FRAME 2
1
1
CHIP ADDRESS BYTE
SUBADDRESS BYTE 2
ADR
0
SEL
R/W0
ACK BY
ADAV4601
ACK BY
ADAV4601
Figure 23. I
2
C Write Format
FRAME 2
SUBADDRESS BYTE 1
FRAME 3
DATA BYTE 1
ACK BY
ADAV4601
ACK BY
ADAV4601
STOP BY
MASTER
07070-101
Rev. B | Page 18 of 60
ADAV4601
SCL
SDA
START BY
MASTER
SCL
(CONTINUED)
SDA
(CONTINUED)
SCL
(CONTINUED)
SDA
(CONTINUED)
00
1
CHIP ADDRESS BYTE
SUBADDRESS BYTE 2
READ DATA BYTE 1
0
FRAME 1
FRAME 3
FRAME 5
1
ADR
0
SEL
R/W
ACK BY
ADAV4601
ACK BY
ADAV4601
MASTER
Figure 24. I
ACK BY
REPEATED
START BY
MASTER
2
C Read Format
FRAME 2
SUBADDRESS BYTE 1
00
FRAME 6
READ DATA BYTE 1
I2C READ AND WRITE OPERATIONS ADC INPUTS
Tabl e 6 shows the timing of a single word write operation.
Every ninth clock, the ADAV4601 issues an acknowledge by
pulling SDA low.
Tabl e 7 shows the timing of the burst mode write sequence.
Tabl e 7 shows an example where the target destination registers
are two bytes. The ADAV4601 auto-increments its subaddress
register counter every two bytes until a stop condition occurs.
The timing of a single word read operation is shown in Tab l e 8 .
W
Note that the first R/
bit is still 0, indicating a write operation.
This is because the subaddress must be written to set up the
internal address. After the ADAV4601 acknowledges the receipt
of the subaddress, the master must issue a repeated start command
followed by the chip address byte with the R/
W
set to 1 (read).
The ADAV4601 responds with the read result on SDA. The
master then responds every ninth clock with an acknowledge
pulse to the ADAV4601.
Tabl e 9 shows the timing of the burst mode read sequence.
Tabl e 9 shows an example where the target read registers are two
bytes. The ADAV4601 increments its subaddress register every two
bytes because the requested subaddress corresponds to a register or
memory area with word lengths of two bytes. Other address
ranges may have a variety of word lengths ranging from one to six
bytes; the ADAV4601 always decodes the subaddress and sets
the auto-increment circuit so that the address increments after the
appropriate number of bytes.
The ADAV4601 has two ADC inputs. By default, this is configured
as a single stereo input; however, because the audio processor is
programmable, these inputs can be reconfigured.
The ADC inputs are shown in Figure 25. The analog inputs are
current inputs (100 µA rms FS) with a 1.5 V dc bias voltage. Any
input voltage can be accommodated by choosing a suitable
combination of input resistor (R
using the formulas
R
= V
IN
FS rms
R
= 2R
ISET
Resistor matching (typically 1%) between RIN and R
important to ensure a full-scale signal on the ADC without
clipping. A 10 µF dc blocking capacitor is also required at the input.
After reset, the ADCs are in a power-down state. The ADCs can
be powered up using the global power-up in the initialization
control register (0x0000). In power critical applications, it is
possible to use the analog power management register (0x0005)
to power-up or power-down individual ADCs.
ANALOG INPUT
100µA rms
FULL SCALE
AUXIN1L
ANALOG INPUT
100µA rms
FULL SCALE
AUXIN1R
10µF
10µF
ADAV4601
1
FRAME 4
CHIP ADDRESS BYTE
1
0
0
) and ISET resistor (R
IN
/100 µA rms
IN/VIN
20kΩ
+
DC BIAS
1.5V
20kΩ
+
DC BIAS
R
ISET
20kΩ
Figure 25. Analog Input Section
1.5V
ACK BY
ADR
SEL
ACK BY
MASTER
R/W
ACK BY
ADAV4601
STOP BY
MASTER
24-BIT
ADC
24-BIT
ADC
ISET
is
ISET
07070-102
)
07070-103
Rev. B | Page 19 of 60
ADAV4601
I2S DIGITAL AUDIO INPUTS
The ADAV4601 has four I2S digital audio inputs that are, by
default, synchronous to the master clock. Also available are two
SRCs capable of supporting any nonsynchronous input with a
sample rate between 5 kHz and 50 kHz. Any of the serial digital
inputs can be redirected through the SRC. Figure 26 shows a
block diagram of the input serial port.
SDIN0
SDIN1
SDIN2
SDIN3
LRCLK0
BCLK0
LRCLK1
BCLK1
LRCLK2
SDIN0
SDIN1
SDIN2
SDIN3
LRCLK0
BCLK0
LRCLK1
BCLK1
LRCLK2
BCLK2
SDIN0
SDIN1
SDIN2
SDIN3
LRCLK0
BCLK0
LRCLK1
BCLK1
LRCLK2
BCLK2
BCLK2
Figure 26. Digital Input Section
Synchronous Inputs and Outputs
The synchronous digital inputs and outputs can use any of the
BCLK or LRCLK inputs as a clock and framing signal. By default,
BCLK1 and LRCLK1 are the serial clocks used for the synchronous
inputs. The synchronous port for the ADAV4601 is in slave mode
by default, which means the user must supply the appropriate
serial clocks, BCLK and LRCLK. The synchronous port can also
be set to master mode, which means that the appropriate serial
clocks, BCLK and LRCLK, can be generated internally from the
MCLK; therefore, the user does not need to provide them. The
serial data inputs are capable of accepting all of the popular
audio transmission standards (see the Serial Data Interface
section for more details).
SRC1
SRC2
SRC2B
SRC2C
SRC2A
SRC2B
SRC2C
AUDIO
PROCESSOR
07070-020
Asynchronous Inputs
The ADAV4601 has two SRCs, SRC1 and SRC2, that can be
used for converting digital data, which is not synchronous to
the master clock. Each SRC can accept input sample rates in the
range of 5 kHz to 50 kHz. Data that has been converted by the
SRC is input to the part and is then synchronous to the internal
audio processor.
The SRC1 is a 2-channel (single-stereo) sample rate converter
that is capable of using any of the three serial clocks available.
The SRC1 can accept data from any of the serial data inputs
(SDIN0, SDIN1, SDIN2, and SDIN3). When selected as an input
to the SRC, this SDIN line is assumed to contain asynchronous
data and is then masked as an input to the audio processor to
ensure that asynchronous data is not processed as synchronous
data. By default, SRC1 uses the LRCLK0 and BCLK0 as the
clock and framing signals.
The SRC2 is a 6-channel (3-stereo) sample rate converter that is
capable of using any of the three serial clocks available. The SRC2
can accept data from any of the serial data inputs (SDIN0, SDIN1,
SDIN2, and SDIN3). When selected as an input to the SRC, this
SDIN line is assumed to contain asynchronous data and is then
masked internally as an input to the audio processor to ensure
that asynchronous data is not processed as synchronous data. By
default, SRC2 uses the LRCLK2 and BCLK2 as the clock and
framing signals.
The first output (SRC2A) from SRC2 is always available to the
audio processor. The other two outputs are muxed with two of
the serial inputs before being available to the audio processor.
SRC2B is muxed with SDIN2, and SRC2C is muxed with SDIN3.
By default, these muxes are configured so that the synchronous
inputs are available to the audio processor. The SRC2B and
SRC2C channels can be made available to the audio processor
simply by enabling them by register write.
When using the ADAV4601 in an asynchronous digital-in-todigital-out configuration, the input digital data is input to the
audio processor core from one of the SRCs, using the assigned
BCLK/LRCLK as a framing signal. The digital output is synchronous
to the BCLK/LRCLK, which is assigned to the synchronous port;
the default clock in this case is BCLK1 and LRCLK1.
Serial Data Interface
LRCLK is the framing signal for the left- and right-channel
inputs, with a frequency equal to the sampling frequency (f
).
S
BCLK is the bit clock for the digital interface, with a frequency
of 64 × f
(32 BCLK periods for each of the left and right channels).
S
The serial data interface supports all the popular audio interface
standards, such as I
The interface mode is software selectable, and its default is I
2
S, left-justified (LJ), and right-justified (RJ).
2
S.
The data sample width is also software selectable from 16 bits,
20 bits, or 24 bits. The default is 24 bits.
Rev. B | Page 20 of 60
ADAV4601
K
I2S Mode Right-Justified (RJ) Mode
In I2S mode, the data is left-justified, MSB first, with the MSB
placed in the second BCLK period following the transition of
the LRCLK. A high-to-low transition of the LRCLK signifies the
beginning of the left channel data transfer, and a low-to-high
transition on the LRCLK signifies the beginning of the right
channel data transfer (see Figure 27).
Left-Justified (LJ) Mode
In LJ mode, the data is left-justified, MSB first, with the MSB
placed in the first BCLK period following the transition of the
LRCLK. A high-to-low transition of the LRCLK signifies the
beginning of the right-channel data transfer, and a low-to-high
transition on the LRCLK signifies the beginning of the leftchannel data transfer (see Figure 28).
LRCLK
BCLK
SDO0
MSB
LEFT CHANNEL
LSB
Figure 27. I
In RJ mode, the data is right-justified, LSB last, with the LSB
placed in the last BCLK period preceding the transition of LRCLK.
A high-to-low transition of the LRCLK signifies the beginning
of the right-channel data transfer, and a low-to-high transition
on the LRCLK signifies the beginning of the left-channel data
transfer (see Figure 29).
RIGHT CHANNEL
LSB
1 /F
S
2
S Mode
MSB
07070-021
LRCLK
BCLK
SDO0
MSB
LEFT CHANNEL
LSB
1 /F
MSB
S
RIGHT CHANNE L
LSB
07070-022
Figure 28. Left-Justified Mode
LRCL
BCLK
SDO0
LEFT CHANNEL
MSBLSB
Figure 29. Right-Justified Mode
1 /F
S
RIGHT CHANNEL
MSB
LSB
07070-023
Rev. B | Page 21 of 60
ADAV4601
DAC VOLTAGE OUTPUTS
The ADAV4601 has six DAC outputs, configured as 3-stereo
auxiliary DAC outputs. However, because the flow is customizable,
it is programmable. The output level is 1 V rms full scale. The DAC
outputs should have a 10 nF capacitor to ground for filtering out
high frequency noise. Following the filtering capacitor, a 10 µF
is required for dc blocking.
After reset, the DACs are in a power-down state. They can power
up quickly using the global power-up in the initialization control
register (0x0000). A popless and clickless power-up and powerdown are also possible.
In power critical applications, it is possible to use the Analog
Power Management 1 register (0x0005) to power up or power
down individual DACs.
such as power-up and power-down. During PWM power-up
and power-down, this pin remains low to signify that the
outputs are not in a valid state. This functionality helps to
eliminate pop/click and other unwanted noise on the outputs.
To accommodate different power stages, the point at which the
PWM_READY signal goes high is programmable. It can go
high when the PWM outputs begin their ramp-up scheme
(PWM_READY early), or it can be programmed to go high
when this ramp-up scheme is complete (PWM_READY late).
This is shown in Figure 33, and it is configured in the PWM
control register (0x001F).
Each set of PWM outputs comprises complementary outputs.
The modulation frequency is 384 kHz, and the full-scale duty
cycle has a ratio of 97:3.
10µF
+
DAC
DAC
DAC
10nF
10nF
10µF
10nF
10nF
10µF
10nF
10nF
10µF
+
10µF
+
10µF
+
+
+
AUXOUT1L
AUXOUT1R
AUXOUT3L
AUXOUT3R
AUXOUT4L
AUXOUT4R
07070-104
Figure 30. DAC Output Section
PWM OUTPUTS
In the ADAV4601, the main outputs are available as four PWM
output channels, which are suitable for driving Class-D amplifiers.
After reset, the PWM channels are in a power-down state. Writing
to the miscellaneous control register (0x000A) enables the PWM
channels. To help ensure popless and clickless power-up and
power-down, there is an enable/disable pattern that is specially
constructed to bring the PWM channels from a zero condition
to a 50/50 duty-cycle square wave (effectively, a zero signal into
the PWM block). This takes 365 ms to complete and can be seen in
Figure 33.
Designed for use in conjunction with this ramp-up scheme, the
ADAV4601 features a status pin, PWM_READY, that indicates
when the PWM outputs are in a state that can cause pops/clicks,
365ms206µs
PWM1A
+
PWM
MODULATOR
PWM
MODULATOR
PWM
MODULATOR
PWM
MODULATOR
–
+
–
+
–
+
–
PWM1A
PWM1B
PWM2A
PWM2B
PWM3A
PWM3B
PWM4A
PWM4B
PWM_READY
07070-025
Figure 31. PWM Output Section
HEADPHONE OUTPUT
There is a dedicated stereo headphone amplifier output that is
capable of driving 32 loads at 1 V rms.
After reset, the headphone output is tristated. The tristate is
disabled using the headphone control register (0x000B). Using
the same register, the gain of the headphone amplifier can be set in
+1.5 dB steps from +1.5 dB to −45 dB. The headphone output
should have a 10 µF capacitor for dc blocking.
AUXOUT4L
10µF
PA
DAC
Figure 32. Headphone Output Section
+
10µF
+
AUXOUT4R
HPOUT1L
HPOUT1R
07070-106
PWM1B
PWM READY
PWM READY EARLY PWM READY LATE
DIFFERENTIAL PWM
IN PHASE
DIFFERENT IAL PWM
OUT OF PHASE—
VALID AUDIO
07070-105
Figure 33. PWM Early
Rev. B | Page 22 of 60
ADAV4601
I2S DIGITAL AUDIO OUTPUTS
One I2S output, SDO0, uses the same serial clocks as the serial
inputs, which are BCLK1 and LRCLK1 by default. If an additional
digital output is required, an additional pin can be reconfigured
as a serial digital output, as shown in Figure 34.
L
R
L
R
I2S OUTPUT
INTERFACE
BCLK1
LRCLK1
Figure 34. I
S/PDIF
OUTPUT
2
S Digital Outputs
SDO0
SPDIF_OUT (SDO1)
7070-027
S/PDIF INPUT/OUTPUT
The S/PDIF output (SPDIF_OUT/SDO1) uses a multiplexer
to select an output from the audio processor or to pass through
the unprocessed SPDIF_IN signals, as shown in Figure 35. On
the ADAV4601, the S/PDIF inputs, SPDIF_IN0/SPDIF_IN1/
SPDIF_IN2/SPDIF_IN3/SPDIF_IN4/SPDIF_IN5/SPDIF_IN6,
are available on the SDIN3, LRCLK0, BCLK0, LRCLK1, BCLK1,
LRCLK2, and BCLK2 pins, respectively. It is possible to have all
seven S/PDIF inputs connected to different S/PDIF signals at one
time. A consequence of this setup is that none of the LRCLKs and
BCLKs are available for use with the digital inputs SDIN0, SDIN1,
SDIN2, and SDIN3. If there is only one S/PDIF input in use, using
the SDIN3 pin as the dedicated S/PDIF input is recommended; this
enables BCLK0/LRCLK0, BCLK1/LRCLK1, and BCLK2/LRCLK2
to be used as the clock and framing signals for the synchronous
and asynchronous port. If SDIN3 is used as an S/PDIF input, it
should not be used internally as an input to the audio processor
because it contains invalid data. Similarly, if BCLK or LRCLK is
used as the S/PDIF input, they can no longer be used as the lock
and framing signals for SDIN0, SDIN1, SDIN2, and SDIN3. The
S/PDIF encoder supports only consumer formats that conform to
IEC-600958.
By default, the
immediately ramp to an unmuted state. However, it is also
possible to have the unmute operation controlled by a control
register bit. In this scenario, even if the
the device does not unmute until a bit in the control register is set.
This can be used when the user wants to keep the outputs
muted, even after the pin has gone high again, for example, in
the case of a fault condition. This allows the system controller
total control over the unmute operation.
AUDIO PROCESSOR
The internal audio processor runs at 2560 × fS; at 48 kHz, this is
122.88 MHz. Internally, the word size is 28 bits, which allows
24 dB of headroom for internal processing. Designed specifically
with audio processing in mind, it can implement complex audio
algorithms efficiently.
By default, the ADAV4601 loads a default audio flow, as shown
in Figure 48. However, because the audio processor is fully
programmable, a custom audio flow can be quickly developed
and loaded to the audio processor.
The audio flow is contained in program RAM and parameter
RAM. Program RAM contains the instructions to be processed by
the audio processor, and parameter RAM contains the coefficients
that control the flow, such as volume control, filter coefficients,
and enable bits.
GRAPHICAL PROGRAMMING ENVIRONMENT
Custom flows for the ADAV4601 are created in a powerful dragand-drop graphical programming application called SigmaStudio.
No knowledge of assembly code is required to program the
ADAV4601. Featuring a comprehensive library of audio processing
blocks (such as filters, delays, dynamics processors, and third-party
algorithms), sigma studio allows a quick and simple creation of
custom flows. For debugging purposes, run-time control of the
audio flow allows the user to fully configure and test the created
flow.
The ADAV4601 mute input can be used to mute any of the
MUTE
analog or digital outputs. When the
selected outputs ramp to a muted condition. Unmuting is
handled in one of two ways and depends on the register setting.
pin goes low, the
Rev. B | Page 23 of 60
07070-109
Figure 36. SigmaStudio Window
ADAV4601
SIGMASTUDIO PIN ASSIGNMENT
Inputs and outputs are defined as numbers in SigmaStudio.
Each number corresponds to a physical input or output on the
ADAV4601. Table 10 and Table 11 show these relationships.
0 SDOL0
1 SDOR0
2 to 7 No connect
8 PWM1/AUXOUT3L
9 PWM2/AUXOUT3R
10 AUXOUT4L/Headphone 1L
11 AUXOUT4R/Headphone 1R
12 AUXOUT1L
13 AUXOUT1R
14 PWM3
15 PWM4
16 to 19 No connect
20 SPDIF OUTL
21 SPDIF OUTR
APPLICATION LAYER
Unique to the ADAV46xx family is the embedded application
layer, which allows the user to define a custom set of registers to
control the audio flow, greatly simplifying the interface between
the audio processor and the system controller. This allows the
ADAV4601 to appear as a simple fixed function register-based
device to the system controller.
When a custom flow is created, a user-customized register map
can be defined for controlling the flow. Each register is 16 bits,
but controls can use only one bit or all 16 bits. Users have full
control over which parameters they use and the degree of
control they have over those parameters during run time. The
combination of the graphical programming environment and
the powerful application layer allows the user to quickly develop
a custom audio flow and still maintain the usability of a simple
register-based device.
LOADING A CUSTOM AUDIO PROCESSING FLOW
The ADAV4601 can load a custom audio flow from an external
2
C ROM. The boot process is initiated by a simple control register
I
write. The EEPROM device address and the EEPROM start address
for the audio flow ROMs can all be programmed.
For the duration of the boot sequence, the ADAV4601 becomes the
master on the I
2
C bus. Transfer of the ROMs from the EEPROM to
the ADAV4601 takes a maximum of 1.06 sec, assuming that the full
audio processor memory is required, during which time no other
devices should access the I
the ADAV4601 automatically reverts to slave mode, and the I
2
C bus. When the transfer is complete,
2
C bus
master can resume sending commands.
ADDRESS
ON
DEFAULT
AUDIO
PROCESSOR
MEMORY
CODE
I2C PORT
CUSTOM
CODE
LOAD
ON
COMMAND
7070-029
AUDIO
PROCESSOR
DATA
LOAD
RESET
BOOT-UP
ROM
EXTERNAL
BOOT-UP ROM
47260 BYTES (MAX )
Figure 37. External EEPROM Booting
NUMERIC FORMATS
It is common in DSP systems to use a standardized method
of specifying numeric formats. Fractional number systems are
specified by an A.B format, where A is the number of bits to the
left of the decimal point and B is the number of bits to the right
of the decimal point.
The ADAV4601 uses the same numeric format for both the
coefficient values (stored in the parameter RAM) and the signal
data values.
The serial port accepts up to 24 bits on the input and is signextended to the full 28 bits of the DSP core. This allows internal
gains of up to 24 dB without internal clipping.
A digital clipper circuit is used between the output of the DSP
core and the DACs or serial port outputs (see Figure 38). This
clips the top four bits of the signal to produce a 24-bit output
with a range of +1.0 (minus 1 LSB) to −1.0. Figure 38 shows
the maximum signal levels at each point in the data flow in
both binary and decibel levels.
4-BIT SIGN EXTENSION
DATA IN
SERIAL
PORT
1.23
(0dB)
Figure 38. Numeric Precision and Clipping Structure
1.23
(0dB)
PROCESSING
(5.23 FORMAT)
5.23
(24dB)
SIGNAL
5.23
(24dB)
DIGITAL
CLIPPER
1.23
(0dB)
07070-110
ROMS AND REGISTERS
The ADAV4601 contains four ROMS: program, instruction,
parameter, and LUT. A default set of ROMs is stored on chip
and is loaded on power-up. A set of ROMs defining a custom
flow can be stored externally on an EEPROM and can be loaded
after power-up.
Program ROM
Program ROM is 42-bits wide and occupies Address 0x1400 to
Address 0x1FFF. This is where the audio flow generated in
SigmaStudio is stored.
Instruction ROM
Instruction ROM is 33-bits wide and occupies Address 0x3000
to Address 0x327F. This is where the application layer register
map is stored.
Parameter ROM
Parameter ROM is 28-bits wide and occupies Address 0x1000 to
Address 0x13FF. Default parameters for default flow and custom
flow are stored here.
LUT ROM
LUT ROM is 28-bits wide and occupies Address 0x4000 to
Address 0x57FF. This contains the parameters for both flows
combined.
Rev. B | Page 25 of 60
SAFE LOADING TO PARAMETER RAM AND
TARGET/SLEW RAM
Up to five safe load registers can be loaded with parameter
RAM address data. The data is transferred to the requested
address when the RAM is idle. It is recommended to use this
method for dynamic updates during run time. For example, a
complete update of one biquad section can occur in one audio
frame. This method is not available for writing to the program
RAM or control registers.
There are ten safe load registers operating in pairs of five, where
five of them store addresses and five of them store data. To safe
load a register, move its address into a safe load address register
and move its data into the corresponding safe load data register. If it
is a parameter RAM, set Bit 4 in Register 0x0200 to 1 to initiate the
safe load. If it is a target/slew RAM, set Bit 5 in Register 0x0200
to 1 to initiate the safe load.
The safe load data registers are located from Address 0x2040 to
Address 0x2044 and are five-bytes wide.
The safe load address registers are located from Address 0x2045 to
Address 0x2049 and are two-bytes wide.
The last five instructions of the program RAM are used for the safe
load process; therefore, the program length should be limited to
2555 cycles (2560 − 5). It is guaranteed that the safe load occurs
within one LRCLK period (21 µs at f
= 48 kHz) of the initiate
S
safe transfer bit being set. Safe load only updates those safe load
registers that have been loaded with new data since the last safe
load operation. For example, if only two parameters or target
RAM locations are updated, it is only necessary to load two of
the safe load registers; the other safe load registers are ignored
because they contain old data.
READ/WRITE DATA FORMATS
The read/write formats of the control port are designed to be
byte oriented. This allows easy programming of common microcontroller chips. To fit into a byte-oriented format, 0s are appended
to the data fields before the MSB to extend the data-word to
eight bits. For example, 28-bit words written to the parameter
RAM are appended with four leading 0s to equal 32 bits (4 bytes);
40-bit words written to the program RAM are not appended
with 0s because they are already a full five bytes. These zeropadded data fields are appended to a 3-byte field consisting of a
7-bit chip address, a read/write bit, and a 16-bit RAM/register
address. The control port knows how many data bytes to expect
based on the address given in the first three bytes.
The total number of bytes for a single location write command
can vary from five bytes (for a control register write) to eight bytes
(for a program RAM write). Burst mode can be used to fill
contiguous register or RAM locations. A burst mode write begins
by writing the address and data of the first RAM or register
location to be written to. Rather than ending the control port
transaction (by issuing a stop command in I
be done in a single-address write, the next data-word can be
written immediately without specifying its address.
2
C mode), as would
ADAV4601
The ADAV4601 control port auto-increments the address of each
write even across the boundaries of the different RAMs and registers.
TARGET/SLEW RAM
The target/slew RAM is a bank of 64 RAM locations, each of
which can be set to autoramp from one value to a desired final
value in one of four modes.
When a program is loaded into the program RAM using one or
more locations in the slew RAM to access the internal coefficient
data, the target/slew RAM is used by the DSP. Typically, these
coefficients are used for volume controls or smooth cross-fading
effects, but they can also be used to update any value in the
parameter RAM. Each of the 64 locations in the slew RAM is
linked to a corresponding location in the target RAM. When a new
value is written to the target RAM using the control port, the
corresponding slew RAM location begins to ramp toward the
target. The value is updated once per audio frame (LRCLK period).
The target RAM is 34 bits wide. The lower 28 bits contain the target
data in 5.23 format for the linear and exponential (constant decibels
and RC) ramp types. For constant time ramping, the lower 28 bits
contain 16 bits in 2.14 format and 12 bits to set the current step.
The upper six bits are used to determine the type and speed of
the ramp envelope in all modes. The format of the data write for
linear and exponential formats is shown in Tabl e 1 2 . Tab le 13 shows
the data write format for the constant time ramping.
In normal operation, write data to the target/slew RAM using
the safe load registers as described in the Safe Loading to
Parameter RAM and Target/Slew RAM section. A mute slew
RAM bit is included in the audio core control register to
simultaneously set all the slew RAM target values to 0. This is
useful for implementing a global multichannel mute. When this
bit is de-asserted, all slew RAM values return to their original
premuted states.
Table 12. Linear, Constant Decibels, and RC Ramp Data Write
Byte 0 Byte 1 Bytes[2:4]
000000,
Curve_Type[1:0]
Time_Const[3:0],
Data[27:24]
Table 13. Constant Time Ramp Data Write
Byte 0 Byte 1 Bytes[2:4]
000000,
Curve_Type[1:0]
Update_Step[0],
#_of_Steps[2:0], Data[15:12]
There are four types of ramping curves: linear, constant decibels,
RC, and constant time.
• The linear ramping curve—The value slews to the target
value using a fixed step size.
• The constant decibels ramping curve—The value slews to
the target value using the current value to calculate the step
size. The resulting curve has a constant rise and decay when
measured in decibels.
Data[23:0]
Data[11:0],
Reserved[11:0]
• The RC ramping curve—The value slews to the target value
using the difference between the target and current values
to calculate the step size, resulting in a simple RC response.
• The constant time ramping curve—The value slews to the
target value in a fixed number of steps in a linear fashion. The
control port mute has no effect on this type of ramping curve.
Table 14. Target/Slew RAM Ramp Type Settings
Settings Ramp Type
00 Linear
01 Constant decibels
10 RC
11 Constant time
The following sections detail how the control port writes to the
target/slew RAM to control the time constant and ramp type
parameters.
Ramp Types[1:3]—Linear, Constant Decibels, and RC
(34-Bit Write)
The target word for the first three ramp types is broken into
three parts. The 34-bit command is written with six leading 0s
to extend the data write to five bytes. The parts of the target
RAM write are
• Ramp type (two bits)
• Time constant (four bits)
• 0000 = fastest
…
• 1111 = slowest
• Data (28 bits): 5.23 format
Ramp Type 4—Constant Time (34-Bit Write)
The target word for the constant time ramp type is written in
five parts, with the 34-bit command written with six leading 0s
to extend the data write to five bytes. The parts of the constant
time target RAM write are
• Ramp type (two bits).
• Update step (one bit). Set to 1 when a new target is loaded
to trigger a step value update. The value is automatically
reset after the step value is updated.
• Number of steps (three bits). The number of steps needed
to slew to the target value is set by these three bits, with the
number of steps equal to 2
• Reserved (12 bits). When writing to the RAM, set all of
these bits to 0.
Rev. B | Page 26 of 60
ADAV4601
Target/Slew RAM Initialization
On reset, the target/slew RAM initializes to preset values. The
target RAM initializes to a linear ramp type with a time constant of 5
and the data set to 1.0. The slew RAM initializes to 1.0. These
defaults result in a full-scale (1.0 to 0.0) ramp time of 21.3 ms.
Linear Update
A linear update is the addition or subtraction of a constant value,
referred to as a step. The following equation describes this step
size as
13
2
=
Step
()
52
−×
t
CONST
10
20
The result of the equation is normalized to a 5.23 data format.
This produces a time constant range from 6.75 ms to 213.4 ms
(−60 dB relative to 0 dB full scale). An example of this kind of
update is shown in Figure 39 and Figure 40. All slew RAM figure
examples, except the half-scale constant time ramp plot (Figure 45),
show an increasing or decreasing ramp between −80 dB and 0 dB
(full scale). All figures except the constant time plots (Figure 44,
Figure 45, and Figure 46) use a time constant of 0x7 (0x0 being
the fastest and 0xF being the slowest).
1.0
0.8
0.6
0.4
0.2
0
–0.2
OUTPUT LEVEL (V)
–0.4
–0.6
–0.8
–1.0
TIME (ms)
Figure 39. Slew RAM—Linear Update Increasing Ramp
1.0
0.8
0.6
0.4
0.2
0
–0.2
OUTPUT LEVEL (V)
–0.4
–0.6
–0.8
–1.0
2010030
TIME (ms)
Figure 40. Slew RAM—Linear Update Decreasing Ramp
35251552010030
07070-111
3525155
7070-112
Constant Decibels and RC Updates (Exponential)
An exponential update is accomplished by shifts and additions
with a range from 6.1 ms to 1.27 sec (−60 dB relative to 0 dB full
scale). When the ramp type is set to 01 (constant decibels), each
step size is set to the current value in the slew data. When the
ramp type bits are set to 10 (RC), the step size is equal to the
difference between the values in the target RAM and the slew
RAM (see Figure 41, Figure 42, and Figure 43).
A constant time update is calculated by adding a step value that
is determined after each target is loaded. The equation for this
step size is
Step = (Target Data − Slew Data)/(Number of Steps)
Figure 44 shows a plot of the target/slew RAM operating in
constant time mode. For this example, 128 steps are used to reach
the target value. This type of ramping takes a fixed amount of time
for a given number of steps, regardless of the difference in the
initial state and the target value. Figure 45 shows a plot of a
constant time ramp from −80 dB to −6 dB (half scale) using
128 steps; therefore, the ramp takes the same amount of time
as the previous ramp from −80 dB to 0 dB. A constant time
decreasing ramp plot is shown in Figure 46.
1.0
0.8
0.6
0.4
0.2
0
–0.2
OUTPUT LEVEL (V)
–0.4
–0.6
–0.8
–1.0
TIME (ms)
3525155201003
0
7070-116
Figure 44. Slew RAM—Constant Time Update Increasing Ramp, Full Scale
1.0
0.8
0.6
0.4
0.2
0
–0.2
OUTPUT LEVEL (V)
–0.4
–0.6
–0.8
–1.0
TIME (ms)
352515520100
30
07070-117
Figure 45. Slew RAM—Constant Time Update Increasing Ramp, Half Scale
1.0
0.8
0.6
0.4
0.2
0
–0.2
OUTPUT LEVEL (V)
–0.4
–0.6
–0.8
–1.0
TIME (ms)
3525155201003
0
07070-118
Figure 46. Slew RAM—Constant Time Update Decreasing Ramp, Full Scale
LAYOUT RECOMMENDATIONS
Parts Placement
The priority for decoupling is VREF, FILTA, FILTD, PLL_LF, and
finally the supplies. For effective decoupling in all cases, make sure
the decoupling capacitor sees the respective ground pin before the
ground plane.
The 1 nF and 100 nF bypass capacitors for the PLL loop filter
should be placed as close as possible to the ADAV4601. All 10 µF
and 0.1 µF bypass capacitors, which are recommended for every
analog, digital, and power/ground pair, should also be placed as
close as possible to the ADAV4601 with priority given to the 0.1 µF
capacitor.
The ADC input voltage-to-current resistors and the ADC
current set resistor should be placed as close as possible to the
respective pins.
Crystal Oscillator Circuit
All traces in the crystal oscillator circuit (see Figure 22) should be
kept as short as possible to minimize stray capacitance. In addition,
avoid long board traces connected to any of these components
because such traces may affect crystal startup and operation.
PWM Outputs
All PWM output differential pairs should be matched in length,
that is, PWM1A = PWM1B, PWM2A = PWM2B.
Grounding
A split ground plane should be used in the layout of the ADAV4601
with the analog and digital grounds connected underneath the
ADAV4601 using a single link. This layout is to avoid possible
ground loop currents in the analog and digital ground planes.
Components in the analog signal path should be placed away
from the digital signals. No signal traces should cross the gap
between the planes.
Bits[15:12] SDO0 output Source for the SDO0 output channel. 0000
0x0 = reserved 0x1 = ADC1 0x2 = reserved 0x3 = SDIN0 0x4 = SDIN1 0x5 = SDIN2/SRC2 Channel B 0x6 = SDIN3/SRC2 Channel C 0x7 = SRC1 0x8 = SRC2 Channel A 0x9 = reserved 0xA = reserved 0xB = reserved 0xC = delayed main input 0xD = main input after loudness 0xE = subchannel
Bits[11:8] SPDIF output Source for the SPDIF output. 0000
0x0 = reserved 0x1 = ADC1 0x2 = reserved 0x3 = SDIN0 0x4 = SDIN1 0x5 = SDIN2/SRC2 Channel B 0x6 = SDIN3/SRC2 Channel C 0x7 = SRC1 0x8 = SRC2 Channel A 0x9 = reserved 0xA = reserved 0xB = reserved 0xC = delayed main input 0xD = main input after loudness 0xE = subchannel
Bits[7:4] AUXOUT1 output Source for the AUXOUT1 output. 0000
0x0 = reserved 0x1 = ADC1 0x2 = reserved 0x3 = SDIN0 0x4 = SDIN1 0x5 = SDIN2/SRC2 Channel B 0x6 = SDIN3/SRC2 Channel C 0x7 = SRC1 0x8 = SRC2 Channel A 0x9 = reserved 0xA = reserved 0xB = reserved 0xC = delayed main input 0xD = main input after loudness 0xE = subchannel
Bits[3:0] Reserved Always write as 0 if writing to this register. 0000
Rev. B | Page 32 of 60
ADAV4601
Address 0x0102 Main and Headphone 1/AUXOUT4 Input Trim Register (Default: 0x0E0E)
Table 17.
Bit No. Bit Name Description Default
Bits[15:14] Reserved Always write as 0 if writing to this register. 00
Bits[13:8] Main input trim
0x00 = +14 dB 0x01 = +13 dB … 0x07 = +7 dB … 0x0E = 0 dB … 0x15 = −7 dB 0x1B = −13 dB 0x1C = −14 dB
Bits[7:6] Reserved Always write as 0 if writing to this register. 00
Bits[5:0] Headphone 1/AUXOUT4 input trim
0x00 = +14 dB 0x01 = +13 dB … 0x07 = +7 dB … 0x0E = 0 dB … 0x15 = −7 dB … 0x1B = −13 dB 0x1C = −14 dB
These register bits are used to gain or attenuate the input to the main channel
processing path from −14 dB to +14 dB in +1 dB steps.
These register bits are used to gain or attenuate the input to the Headphone 1/
AUXOUT4 channel processing path from −14 dB to +14 dB in +1 dB steps.
Address 0x0103 SDO0/AUXOUT3 and SPDIF Input Trim Register (Default: 0x0E0E)
001110
001110
Table 18.
Bit No. Bit Name Description Default
Bits[15:14] Reserved Always write as 0 if writing to this register. 00
Bits[13:8] SDO0/AUXOUT3 input trim
0x00 = +14 dB 0x01 = +13 dB … 0x07 = +7 dB … 0x0E = 0 dB … 0x15 = −7 dB 0x1B = −13 dB 0x1C = −14 dB
Bits[7:6] Reserved Always write as 0 if writing to this register. 00
These register bits are used to gain or attenuate the input to the SDO0 and the
AUXOUT3 processing path from −14 dB to +14 dB in +1 dB steps.
Rev. B | Page 33 of 60
001110
ADAV4601
Bit No. Bit Name Description Default
Bits[5:0] SPDIF input trim
0x00 = +14 dB 0x01 = +13 dB … 0x07 = +7 dB … 0x0E = 0 dB … 0x15 = −7 dB … 0x1B = −13 dB 0x1C = −14 dB
These register bits are used to gain or attenuate the input to the SPDIF processing
path from −14 dB to +14 dB in +1 dB steps.
Address 0x0104 AUXOUT1 Input Trim Register (Default: 0x0E0E)
Table 19.
Bit No. Bit Name Description Default
Bits[15:14] Reserved Always write as 0 if writing to this register. 00
Bits[13:8] AUXOUT1 input trim
0x00 = +14 dB 0x01 = +13 dB … 0x07 = +7 dB … 0x0E = 0 dB … 0x15 = −7 dB 0x1B = −13 dB 0x1C = −14 dB
Bits[7:0] Reserved Always write as 0 if writing to this register. 00000000
These register bits are used to gain or attenuate the input to the AUXOUT1
channel processing path from −14 dB to +14 dB in +1 dB steps.
Address 0x0105 Main Delay Register (Default: 0x0000)
001110
001110
Table 20.
Bit No. Bit Name Description Default
Bits[15:13] Reserved Always write as 0 if writing to this register. 000
Bits[12:0] Main delay These register bits are used to specify the lip synchronization delay for the main channel. 0000000000000
0x0000 = 20.83 μs (1 sample delay at 48 kHz) 0x0001 = 20.83 μs (1 sample delay) 0x0002 = 41.66 μs (2 sample delay)
… 0x1C20 = 150 ms (7200 sample delay)
Rev. B | Page 34 of 60
ADAV4601
Address 0x0106 Automatic Volume Control (Default: 0x350C)
Table 21.
Bit No. Bit Name Description Default
Bits[15:12] AVC maximum gain
0x0 = 15 dB 0x1 = 14 dB 0x2 = 13 dB 0x3 = 12 dB 0x4 = 11 dB 0x5 = 10 dB ... 0xF = 0 dB
Bits[11:8] AVC decay time
Used to control the maximum gain in the range of 0 dB to 15 dB. This is the maximum gain
that can be applied to the input signal to reach the desired output level.
Used to control the decay time in the range of 20 ms to 12 sec. The decay time
corresponds to the time required for the output to reach the desired level.
Used to control the maximum attenuation in the range of −18 dB to −3 dB in
+1 dB steps. This is the maximum attenuation that can be applied to the input
signal to reach the desired output level.
Used to control the required output level of the AVC block in the range of −3 dBFS to
−18 dBFS in −1 dB steps. If the input signal is greater than the output level that has
been set by these bits, the gain is automatically reduced.
0011
0101
0000
1100
Rev. B | Page 35 of 60
ADAV4601
Address 0x0107 Main Sever Band EQ Control Register (Default: 0x0018)
Table 22.
Bit No. Bit Name Description Default
Bits[15:11] Reserved Always write as 0 if writing to this register. 00000
Bits[10:8] Main EQ band These register bits control the frequency band of the equalizer. 000
0x0 = 120 Hz 0x1 = 200 Hz 0x2 = 500 Hz 0x3 = 1200 Hz 0x4 = 3000 Hz 0x5 = 7500 Hz 0x6 = 12,000 Hz
Bits[7:6] Reserved Always write as 0 if writing to this register. 00
Bits[5:0] Main EQ gain
0x00 = +12 dB 0x01 = +11.5 dB … 0x0A = +7 dB … 0x18 = 0 dB … 0x26 = −7 dB … 0x2F = −11.5 dB 0x30 = −12 dB
These register bits are used to control the required gain of the equalizer. The gain of the equalizer
is changed in 0.5 dB steps.
Address 0x0108 Main Channel Loudness Register (Default: 0x0000)
011000
Table 23.
Bit No. Bit Name Description Default
Bits[15:7] Reserved Always write as 0 if writing to this register. 000000000
Bits[6:5] Cutoff frequency These register bits are used to control the cutoff frequency of the loudness. 00
00b – 10 Hz 01b – 30 Hz 10b – 50 Hz 11b – 70 Hz
Bits[4:0] Loudness level These register bits are used to control the required level of loudness. 00000
0x00 = 0 dB 0x01 = +1 dB … 0x0E = +14 dB 0x0F = +15 dB
The low crossover frequency is the cutoff frequency of the crossover low-pass
filter. This means that only the frequencies under this frequency are sent to the
woofer output. The frequency changes in 10 Hz steps.
The high crossover frequency is the cutoff frequency of the crossover high-pass
filter. This means that only the frequencies above this frequency are sent to the
tweeter output.
Address 0x010A Crossover Trim Register (Default: 0x0E0E)
000101
000101
Table 25.
Bit No. Bit Name Description Default
Bits[15:14] Reserved Always write as 0 if writing to this register. 00
Bits[13:8] Tweeter trim
0x00 = +14 dB 0x01 = +13 dB
… 0x07 = +7 dB
… 0x0E = 0 dB
… 0x15 = −7 dB
… 0x1B = −13 dB 0x1C = −14 dB
Bits[7:6] Reserved Always write as 0 if writing to this register. 00
These register bits are used to gain or attenuate the input to the tweeter outputs from
−14 dB to +14 dB in +1 dB steps.
Rev. B | Page 37 of 60
001110
ADAV4601
Bit No. Bit Name Description Default
Bits[5:0] Woofer trim
0x00 = +14 dB 0x01 = +13 dB
… 0x07 = +7 dB
… 0x0E = 0 dB
… 0x15 = −7 dB
… 0x1B = −13 dB 0x1C = −14 dB
These register bits are used to gain or attenuate the input to the woofer outputs from
−14 dB to +14 dB in +1 dB steps.
Address 0x010B ADI Bass Control Register (Default: 0x0062)
Table 26.
Bit No. Bit Name Description Default
Bits[15:9] Reserved Always write as 0 if writing to this register. 0000000
Bits[8:4] Boost value
0x00 = 0 dB 0x01 = 1 dB 0x02 = 2 dB 0x03 = 3 dB 0x04 = 4 dB 0x05 = 5 dB 0x06 = 6 dB 0x07 = 7 dB … 0x1E = 30 dB 0x1F = 31 dB
Bits[3:0] Boost frequency
The boost ranges from 0 dB to 31 dB, and it controls the maximum dynamic gain applied to
the algorithm.
The boost frequency ranges from 20 Hz to 320 Hz, and it designates the center frequency
for the boosting filter. The frequency is increased in 20 Hz steps.
001110
00110
0010
Rev. B | Page 38 of 60
ADAV4601
Address 0x010C and Address 0x010D Tweeter Left Balance Control Registers (Default: 0x0080, 0x0000)
These two registers (0x010C and 0x010D) control the balance for the left tweeter output.
The balance control words are 28-bit words in twos complement and a 5.23 format. This means there are five bits to the left of the decimal
point of which the most significant bit is the sign bit.
2
To simplify updating the tweeter left balance control, the I
the balance control register can be updated in a single I
2
be updated using the following I
C write format:
<dev addr><010C><32-bit data transfer>
Note that the tweeter left balance control is a 32-bit parameter; therefore, both Register 0x010C and Register 0x010D must be written to.
Writing anything less than the 32 bits to these registers does not update the parameter.
Table 27.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:0] Tweeter left balance control register[27:0] 0x010C Bits[11:0] = tweeter left balance control register[27:16] 000010000000
Bits[15:0] Tweeter left balance control register[27:0] 0x010D Bits[15:0] = tweeter left balance control register[15:0] 0000000000000000
Address 0x010E and Address 0x010F Tweeter Right Balance Control Registers (Default: 0x0080, 0x0000)
These two registers (0x010E and 0x010F) control the balance for the right tweeter output.
The balance control words are 28-bit words in twos complement and a 5.23 format. This means there are five bits to the left of the decimal
point of which the most significant bit is the sign bit.
C address pointer auto-increments when writing and reading. This means that
2
C block write. Therefore, it is recommended that the tweeter left balance control
Table 28.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:0] Tweeter right balance control register[27:0] 0x010E Bits[11:0] = tweeter right balance control register[27:16] 000010000000
Bits[15:0] Tweeter right balance control register[27:0] 0x010F Bits[15:0] = tweeter right balance control register[15:0] 0000000000000000
Address 0x0110 and Address 0x0111 Woofer Left Balance Control Registers (Default: 0x0080, 0x0000)
These two registers control the balance for the left woofer output.
The balance control words are 28-bit words in twos complement and a 5.23 format. This means there are five bits to the left of the decimal
point of which the most significant bit is the sign bit.
Table 29.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:0] Woofer left balance control register[27:0] 0x0110 Bits[11:0] = woofer left balance control register[27:16] 000010000000
Bits[15:0] Woofer left balance control register[27:0] 0x0111 Bits[15:0] = woofer left balance control register[15:0] 0000000000000000
Rev. B | Page 39 of 60
ADAV4601
Address 0x0112 and Address 0x0113 Woofer Right Balance Control Registers (Default: 0x0080, 0x0000)
These two registers control the balance for the right tweeter output.
The balance control words are 28-bit words in twos complement and a 5.23 format. This means there are five bits to the left of the decimal
point of which the most significant bit is the sign bit.
Table 30.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:0] Woofer right balance control register[27:0] 0x0112 Bits[11:0] = woofer right balance control register[27:16] 000010000000
Bits[15:0] Woofer right balance control register[27:0] 0x0113 Bits[15:0] = woofer right balance control register[15:0] 0000000000000000
Address 0x0114 and Address 0x0115 Main Volume Control Registers (Default: 0x0080, 0x0000)
These two registers control the volume for the main channel output.
The volume control words are 28-bit words in twos complement and a 5.23 format. This means there are five bits to the left of the decimal
point of which the most significant bit is the sign bit.
Table 31.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:0] Woofer right balance control register[27:0] 0x0114 Bits[11:0] = main volume Bits[27:16] 000010000000
Bits[15:0] Woofer right balance control register[27:0] 0x0115 Bits[15:0] = main volume Bits[15:0] 0000000000000000
Address 0x0116 Tweeter Peak Limiter Control Register (Default: 0x0F00)
This register controls the peak limiter for the main output tweeter.
Table 32.
Bit No. Bit Name Description Default
Bits[15:13] Reserved Always write as 0 if writing to this register. 000
Bits[12:8] Post gain These register bits control the post gain in the range of +15 dB to −15 dB in +1 dB steps. 01111
0x00 = +15 dB 0x01 = +14 dB
… 0x08 = +7 dB
… 0x0F = +0 dB
… 0x16 = −7 dB
… 0x1D = −14 dB 0x1E = −15 dB
Bits[7:5] Hold time
0x0 = 0 ms 0x1 = 10 ms 0x2 = 20 ms 0x3 = 30 ms 0x4 = 40 ms 0x5 = 50 ms 0x6 = 60 ms 0x7 = 70 ms
These register bits control the hold time for the limiter, which is the time in ms that the limiter
holds the attenuated level after the current input to the limiter function falls below the limiter
threshold.
These register bits control the decay time for the limiter in the range of 10 dB/s to 320 dB/s in
10 dB/s steps.
Address 0x0117 Woofer Peak Limiter Control Register (Default: 0x0F00)
00000
Table 33.
Bit No. Bit Name Description Default
Bits[15:13] Reserved Always write as 0 if writing to this register. 000
Bits[12:8] Post gain These register bits control the post gain in the range of +15 dB to −15 dB in +1 dB steps. 01111
0x00 = +15 dB 0x01 = +14 dB
… 0x08 = +7 dB
… 0x0F = 0 dB
… 0x16 = −7 dB
… 0x1D = −14 dB 0x1E = −15 dB
Rev. B | Page 41 of 60
ADAV4601
Bit No. Bit Name Description Default
Bits[7:5] Hold time
0x0 = 0 ms 0x1 = 10 ms 0x2 = 20 ms 0x3 = 30 ms 0x4 = 40 ms 0x5 = 50 ms 0x6 = 60 ms 0x7 = 70 ms
Bits[4:0] Decay time
These register bits control the hold time for the limiter, which is the time in ms that the limiter
holds the attenuated level after the current input to the limiter function falls below the limiter
threshold.
These register bits control the decay time for the limiter in the range of 10 dB/s to 320 dB/s in
10 dB/s steps.
000
0000
Rev. B | Page 42 of 60
ADAV4601
Address 0x0118 Headphone 1/AUXOUT4 Seven Band EQ Control Register (Default: 0x0018)
Table 34.
Bit No. Bit Name Description Default
Bits[15:11] Reserved Always write as 0 if writing to this register. 00000
Bits[10:8]
0x00 = +12 dB 0x01 = +11.5 dB … 0x0A = +7 dB … 0x18 = 0 dB … 0x26 = −7 dB … 0x2F = −11.5 dB 0x30 = −12 dB
Headphone 1/
AUXOUT4 EQ band
Headphone 1/
AUXOUT4 EQ gain
Address 0x011A and Address 0x011B Headphone 1/AUXOUT4 Left Balance Control Registers (Default: 0x0080, 0x0000)
These two registers control the balance for the Left Headphone 1 and AUXOUT 4 output.
The balance control words are 28-bit words in twos complement and a 5.23 format. This means there are five bits to the left of the decimal
point of which the most significant bit is the sign bit.
These register bits control the frequency band of the equalizer. 000
These register bits are used to control the required gain of the equalizer. The gain of the
equalizer is changed in 0.5 dB steps.
001110
Table 35.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:0]
Bits[15:0]
Woofer left balance
control register[27:0]
Woofer left balance
control register[27:0]
0x011A Bits[11:0] = Headphone 1/AUXOUT 4 left balance control register[27:16] 000010000000
0x011B Bits[15:0] = Headphone 1/AUXOUT 4 left balance control register[15:0] 0000000000000000
Address 0x011C and Address 0x011D Headphone 1/AUXOUT4 Right Balance Control Registers (Default: 0x0080, 0x0000)
These two registers control the balance for the Right Headphone 1 and AUXOUT 4 output.
The balance control words are 28-bit words in twos complement and a 5.23 format. This means there are five bits to the left of the decimal
point of which the most significant bit is the sign bit.
Table 36.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:0]
Bits[15:0]
Woofer left balance
control register[27:0]
Woofer left balance
control register[27:0]
0x011C Bits[11:0] = Headphone 1/AUXOUT 4 right balance
control register[27:16]
0x011D Bits[15:0] = Headphone 1/AUXOUT 4 right balance
control register[15:0]
Rev. B | Page 43 of 60
000010000000
0000000000000000
ADAV4601
Address 0x011E and Address 0x011F Headphone 1/AUXOUT4 Volume Control Registers (Default: 0x0080, 0x0000)
These two registers control the volume for the headphone and AUXOUT4 outputs.
The volume control words are 28-bit words in twos complement and a 5.23 format. This means there are five bits to the left of the decimal
point of which the most significant bit is the sign bit.
Table 37.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:0] Headphone 1 volume control[27:0]
Bits[15:7] Reserved Always write as 0 if writing to this register. 000000000
Bits[6:5] Cutoff frequency These register bits are used to control the cutoff frequency of the loudness. 00
00b = 10 Hz 01b = 30 Hz 10b = 50 Hz 11b = 70 Hz
Bits[4:0] Loudness level
0x00 = 0 dB 0x01 = 1 dB … 0x0E = 14 dB 0x0F = 15 dB
These register bits are used to control the required level of loudness. It can be
changed in 1 dB steps.
Address 0x0121 Mute Control Register (Default: 0x0000)
000010000000
0000000000000000
00000
Table 39.
Bit No. Bit Name Description Default
Bits[15:8] Reserved Always write as 0 if writing to this register. 00000000
Bit[7] Mute tweeter output Mutes the tweeter output. 0
0b = mutes 1b = unmutes
Bit[6] Mute woofer output Mutes the woofer output. 0
0b = mutes 1b = unmutes
Bit[5] Mute AUXOUT3 output Mutes the AUXOUT3 output. 0
0b = mutes 1b = unmutes
Bit[4] Mute HP1/AUXOUT4 output Mutes the Headphone 1 and AUXOUT4 output. 0
0b = mutes 1b = unmutes
Bit[3] Mute SDO0 output Mutes the SDO0 output. 0
0b = mutes 1b = unmutes
Bit[2] Mute SPDIF output Mutes the SPDIF output. 0
0b = mutes 1b = unmutes
Rev. B | Page 44 of 60
ADAV4601
Bit No. Bit Name Description Default
Bit[1] Mute AUXOUT1 output Mutes the AUXOUT1 output. 0
0b = mutes 1b = unmutes
Bit[0] Reserved Always write as 0 if writing to this register. 0
Address 0x0122 Audio Flow Control Register (Default: 0x8001)
Table 40.
Bit No. Bit Name Description Default
Bit[15] Reserved Always write a 1 if writing to this register. 1
Bit[14] Enable AVC When set to 1, it enables the AVC function. 0
0b = disabled 1b = enabled
Bit[13] Enable main delay When set to 1, it enables the lip synchronization delay. 0
0b = disabled 1b = enabled
Bit[12] Enable main EQ When set to 1, it enables the seven band equalizer. 0
0b = disabled 1b = enabled
Bit[11] Enable ADI bass When set to 1, it enables the ADI bass. 0
0b = disabled 1b = enabled
Bit[10] Enable main loudness When set to 1, it enables the loudness. 0
0b = disabled 1b = enabled
Bit[9] Enable main beeper
0b = beeper and channel 1b = beeper only
Bit[8] Enable main limiter When set to 1, it enables the tweeter and woofer peak limiters. 0
0b = disabled 1b = enabled
Bit[7] Enable speaker EQ When set to 1, it enables the eight band speaker equalizer for the tweeter output. 0
0b = disabled 1b = enabled
Bit[6] Enable crossover bypass When set to 1, it enables the crossover low-pass and high-pass filters for the main channel. 0
0b = disabled 1b = enabled
Bit[5] Tweeter output control
0b = tweeter only 1b = tweeter and woofer
Bit[4] Enable subchannel LPF Used to control the LPF on the subchannel. 0
0b = enabled 1b = disabled
Bit[3] Enable HP1 EQ When set to 1, it enables the seven band equalizer for the Headphone 1 channel. 0
0b = disabled 1b = enabled
When set to 1, it leaves only the beeper on the main channel. By default, this register bit
is set to 0, which means the main channel input is added to the beeper.
When set to 1, the tweeter and woofer outputs are added together and output on
the tweeter output.
0
0
Rev. B | Page 45 of 60
ADAV4601
T
f
Bit No. Bit Name Description Default
Bit[2] Enable HP1 loudness When set to 1, it enables the loudness for the Headphone 1 channel. 0
0b = disabled 1b = enabled
Bit[1] Enable HP1 beeper
0b = beeper and channel 1b = beeper only
Bit[0] Enable spatializer When set to 1, it enables the ADI spatializer. 1
0b = disabled 1b = enabled
When set to 1, it leaves only the beeper on the Headphone 1 channel. By default, this bit
is 0, which means the Headphone 1 channel input is added to the beeper.
Address 0x0123 Main Beeper Control Register (Default: 0x0005)
This register controls the main beeper block.
Table 41.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:8] Volume gain These register bits control the volume of the beeper in the range of −38 dB to −10 dB in +2 dB steps. 0000
0x0 = off 0x1 = −38 dB 0x2 = −36 dB
... 0xE = −12 dB 0xF = −10 dB
Bits[7:6] Reserved Always write as 0 if writing to this register. 00
Bits[5:0] Frequency
This register is used to control the low-pass filter cutoff frequency for the subwoofer channel.
0
000101
Table 42.
Bit No. Bit Name Description Default
Bits[15:4] Reserved Always write as 0 if writing to this register.000000000000
Bits[3:0] LPF sub cutoff
hese register bits control the cutoff frequency of the low-pass filter of the subwoofer
channel. The range of values is 60 Hz to 340 Hz in 20 Hz steps.
0x0 = of
This register is used to set the delay for the SRC channel.
Table 43.
Bit No. Bit Name Description Default
Bits[15:12] Reserved Always write as 0 if writing to this register. 0000
Bits[11:0] SRC delay The range of values is 20.83 μs to 42 ms in 20.83 μs (1 sample delay) steps. 000000000000
0x000 = 20.83 μs (1 sample delay) 0x001 = 41.66 μs (2 sample delay)
... 0x7E1 = 42 ms (2017 sample delay)
Address 0x0127 SRC Control Register (Default: 0x0030)
This register is used to enable the DSP mute circuit for SRC1 and SRC2. If SRC is enabled and detects an error, it will mute the output of
the SRC. It also bypasses the de-emphasis filters of the SRC.
Table 44.
Bit No. Bit Name Description Default
Bits[15:6] Reserved Always write as 0 if writing to this register. 0000000000
Bit[5] SRC1 de-emphasis filter bypass This bypasses the SRC1 de-emphasis filter. 0
0b = bypass 1b = enabled
Bit[4] SRC2 de-emphasis filter bypass This bypasses the SRC2 de-emphasis filter. 0
0b = bypass 1b = enabled
Bit[3] SRC1 DSP mute circuit bypass If an error is detected, it mutes the output of SRC1. 0
0b = enabled 1b = disabled
Bit[2] SRC2 Channel A DSP mute circuit bypass If an error is detected, it mutes the output of SRC2. 0
0b = enabled 1b = disabled
Bit[1] SRC2 Channel B DSP mute circuit bypass If an error is detected, it mutes the output of SRC2. 0
0b = enabled 1b = disabled
Bit[0] SRC2 Channel C DSP mute circuit bypass If an error is detected, it mutes the output of SRC2. 0
0b = enabled 1b = disabled
Rev. B | Page 47 of 60
ADAV4601
MAIN CONTROL REGISTERS
DETAILED REGISTER DESCRIPTIONS
Address 0x0000 Initialization Control Register (Default: 0x0080)
Table 45.
Bit No. Bit Name Description Default
Bits[15:13] Reserved Always write as 0 if writing to this register. 000
Bit[12] Slew mute
0b = unmuted 1b = muted
Bit[11] Reserved Always write as 0 if writing to this register. 0
Bits[10:9] MCLKI frequency select Used to select the MCLKI pin frequency. 00
00b = 512 × frequency sample (FS) (24.576 MHz) 01b = 256 × FS (12.288 MHz) 10b = 128 × FS (6.144 MHz) 11b = 64 × FS (3.072 MHz)
Bits[8:7] SRC2 Channel A input select Used to select the source for SRC2 Channel A. 01
00b = SDIN0 01b = SDIN1 10b = SDIN2 11b = SDIN3
Bits[6:5] SRC1 input select Used to select the source for SRC1. 00
00b = SDIN0 01b = SDIN1 10b = SDIN2 11b = SDIN3
Bit[4] SRC2 Channel A enable Used to enable SRC2 Channel A. 0
0b = disabled 1b = enabled
Bit[3] SRC1 enable Used to enable SRC1 Channel A. 0
0b = disabled 1b = enabled
Bit[2] GSB enable When set to 1, the ADAV4601 enters standby mode. 0
0b = disable standby or not in standby 1b = enable standby or not in standby
Bit[1] Audio processor enable Set to 1 to enable the audio processor. 0
0b = disabled 1b = enabled
Bit[0] GPU
0b = use power management register 1b = global power-up
When set to 1, all slew parameters ramp to zero. It is recommended to set this
bit to 1 prior to downloading a new program to reduce any risk of pops or clicks
on the output.
Globally powers up all parts of the device. If read back, it indicates the status of
the global power-up.
0
0
Rev. B | Page 48 of 60
ADAV4601
Address 0x0004 Serial Port Control 1 Register (Default: 0x0000)
Table 46.
Bit No. Bit Name Description Default
Bits[15:13] Reserved Always write as 0 if writing to this register. 000
Bits[12:11] SRC2 serial mode Used to select the format of the data for SRC2. 00
00b = I2S 01b = left-justified 10b = right-justified 11b = not applicable
Bits[10:9] SRC2 word width Used to specify the word width of the data. 00
00b = 24 bits 01b = 20 bits 10b = 16 bits 11b = not applicable
Bits[8:7] SRC1 serial mode Used to select the format of the data for SRC1. 00
00b = I2S 01b = left-justified 10b = right-justified 11b = not applicable
Bits[6:5] SRC1 word width Used to specify the word width of the data. 00
00b = 24 bits 01b = 20 bits 10b = 16 bits 11b = not applicable
Bit[4] Sync master slave
0b = slave 1b = master
Bits[3:2] Sync serial mode Used to select the format of the data for the inputs used by the synchronous serial input block. 00
00b = I2S 01b = left-justified 10b = right-justified 11b = not applicable
Bits[1:0] Sync word width Used to specify the word width of the data for the synchronous digital inputs. 00
00b = 24 bits 01b = 20 bits 10b = 16 bits 11b = not applicable
Used to set master or slave mode for the synchronous input. In slave mode, LRCLK1 and BCLK1
are provided by another source. In master mode, the ADAV4601 provides LRCLK1 and BCLK1.
Address 0x0005 Analog Power Management 1 Register (Default: 0x8000)
0
Table 47.
Bit No. Bit Name Description Default
Bit[15] DAC standby disable Set to 1 after reset, which means all DACs are in normal mode but are still powered down. 1
0b = DACs in low power mode 1b = DACs in normal mode
Bit[14] Reserved Always write as 0 if writing to this register. 0
Bit[13] REF BUF Provides the voltage reference for the analog core. 0
1b = block powered up 0b = block powered down
Bit[12] Reserved Always write as 0 if writing to this register. 0
Bit[11] Reserved Always write as 0 if writing to this register. 0
Rev. B | Page 49 of 60
ADAV4601
Bit No. Bit Name Description Default
Bit[10] ADC1 right Powers on the ADC1 right channel. 0
1b = block powered up 0b = block powered down
Bit[9] ADC1 left Powers on the ADC1 left channel. 0
1b = block powered up 0b = block powered down
Bit[8:7] Reserved Always write as 0 if writing to this register. 00
Bit[6] AUXDAC3 right Powers on the AUXDAC3 right channel. 0
1b = block powered up 0b = block powered down
Bit[5] AUXDAC3 left Powers on the AUXDAC3 left channel. 0
1b = block powered up 0b = block powered down
Bit[4] Reserved Always write as 0 if writing to this register. 0
Bit[3] Reserved Always write as 0 if writing to this register. 0
Bit[2] Reserved Always write as 0 if writing to this register. 0
Bit[1] AUXDAC1 right Powers on the AUXDAC1 right channel. 0
1b = block powered up 0b = block powered down
Bit[0] AUXDAC1 left Powers on the AUXDAC1 left channel. 0
1b = block powered up 0b = block powered down
Address 0x0006 Analog Power Management 2 Register (Default: 0x0000)
Table 48.
Bit No. Bit Name Description Default
Bits[15:10] Reserved Always write as 0 if writing to this register. 000000
Bit[9] PLL Powers on the PLL. 0
1b = block powered up 0b = block powered down
Bits[8:6] Reserved Always write as 0 if writing to this register. 000
Bit[5] Reserved Always write as 0 if writing to this register. 0
Bit[4] Reserved Always write as 0 if writing to this register. 0
Bit[3] HP1 DAC right Powers on the HP1 DAC right channel. 0
1b = block powered up 0b = block powered down
Bit[2] HP1 DAC left Powers on the HP1 DAC left channel. 0
1b = block powered up 0b = block powered down
Bit[1] HP1 AMP right Powers on the HP1 AMP right channel. 0
1b = block powered up 0b = block powered down
Bit[0] HP1 AMP left Powers on the HP1 AMP left channel. 0
1b = block powered up 0b = block powered down
Rev. B | Page 50 of 60
ADAV4601
Address 0x0007 Digital Power Management Register (Default: 0x0000)
Table 49.
Bit No. Bit Name Description Default
Bits[15:8] Reserved Always write as 0 if writing to this register. 00000000
Bit[7] PWM Powers on the PWM channels. 0
1b = block powered up 0b = block powered down
Bit[6] S/PDIF TX Powers on the S/PDIF transmitter. 0
1b = block powered up 0b = block powered down
Bit[5] Reserved Always write as 0 if writing to this register. 0
Bit[4] SRC2 Powers on SRC2. 0
1b = block powered up 0b = block powered down
Bit[3] SRC1 Powers on SRC1. 0
1b = block powered up 0b = block powered down
Bit[2] Reserved Always write as 0 if writing to this register. 0
Bit[1] ADC/DAC engine Powers on the ADC/DAC engine. 0
1b = block powered up 0b = block powered down
Bit[0] Audio processor Powers on the audio processor core. 0
1b = block powered up 0b = block powered down
Address 0x0009 SPDIF Transmitter Control Register (Default: 0x0000)
Table 50.
Bit No. Bit Name Description Default
Bits[15:8] Reserved Always write as 0 if writing to this register. 00000000
Bits[14:12] SPDIF output select Selects the source for the SPDIF output. 000
000b = output internally generated SPDIF 001b = output SPDIF_IN2 010b = output SPDIF_IN1 011b = output SPDIF_IN0 100b = output SPDIF_IN3 101b = output SPDIF_IN4 110b = output SPDIF_IN5 111b = output SPDIF_IN6
Bit[11] SPDIF disable Enables or disables the SPDIF transmitter. 0
0b = enabled 1b = disabled
Bit[10] PRE edge Sets the edge to be used for the preamble. 0
0b = rising edge 1b = falling edge
Bit[9] Validity polarity Used to indicate to the receiver if the data in the transmitted stream is valid audio data. 0
0b = valid data sent 1b = invalid data sent
Bit[8] Copy flag Used to indicate to the receiver if the data is copyright material. 0
0b = copyright 1b = not copyright
Bits[7:0] Channel status
Used to specify the type of equipment in use; not applicable when the SPDIF Mux Bits[14:12]
are set to anything other than 000b.
00000000
Rev. B | Page 51 of 60
ADAV4601
Address 0x000A Misc Control Register (Default: 0x0800)
Table 51.
Bit No. Bit Name Description Default
Bit[15] PWM ready flag (read-only)
0b = PWM ready pin low 1b = PWM ready pin high
Bit[14] Enable selected PWM channels When enabled, all PWM channels selected by Bits[10:7] can be used. 0
0b = all PWM channels disabled 1b = selected PWM channels enabled
Bit[13] MCLK_OUT CLK type select Used to configure the MCLK_OUT pin. 0
0b = crystal frequency on MCLK_OUT 1b = internally generated clocks on MCLK_OUT
Bit[12] PWM enable/disable patterns Enables the enable/disable patterns for the PWM block. 0
0b = enable/disable pattern not used 1b = use enable/disable pattern
Bit[11] DAC mod offset
0b = not locked 1b = locked
Bit[4] MCLK_OUT enable Enables the clock chosen by Bit[13] and Bits[3:1] to be output on the MCLK_OUT pin. 0
0b = MCLK_OUT function disabled 1b = MCLK_OUT function enabled
Indicates the current status of the PWM ready pin. When PWM ready is low, the
PWM is not enabled. When PWM ready is high, the PWM is enabled and stable.
Adds dc offset to the DAC Σ-Δ modulator to eliminate idle tones. It is recommended
that this bit be disabled before the ADC/DAC engine is powered up in
Control Register 0x0007[1].
The PWM outputs are disabled by default, which means that the outputs are
at GND. When this bit is set to 1 and Bit[14] of this register is set to 1, then the
PWM Enable 4 channel is enabled.
The PWM outputs are disabled by default, which means that the outputs are
at GND. When this bit is set to 1 and Bit[14] of this register is set to 1, then the
PWM Enable 3 channel is enabled.
The PWM outputs are disabled by default, which means that the outputs are
at GND. When this bit is set to 1 and Bit[14] of this register is set to 1, then the
PWM Enable 2 channel is enabled.
The PWM outputs are disabled by default, which means that the outputs are
at GND. When this bit is set to 1 and Bit[14] of this register is set to 1, then the
PWM Enable 1 channel is enabled.
Set to 1 when the sample rate converter (SRC) locks to the incoming data,
indicating the data is valid.
Set to 1 when the sample rate converter (SRC) locks to the incoming data,
indicating the data is valid.
Address 0x000B Headphone Control Register (Default: 0x0000)
Table 52.
Bit No. Bit Name Description Default
Bits[15:8] Reserved Always write as 0 if writing to this register. 00000000
Bit[7] HP1 mute When set to 1, mutes the headphone output immediately without ramping. 0
0b = unmuted 1b = mute
Bit[6] HP1 short-circuit protect Enables the short-circuit protection for the headphone amplifier. 0
0b = disabled 1b = enabled
Bit[5] HP1 tristate Disables tristating of the headphone amplifier. 0
0b = enabled 1b = disabled
Bits[4:0] Headphone 1 gain/attenuation Used to apply analog attenuation to the headphone amplifier. 00000
00000b = 0 dB 00001b = −1.5 dB 00010b = −3 dB … 11101b = −43.5 dB 11110b = −45 dB 11111b = +1.5 dB
Address 0x000C Serial Port Control 2 Register (Default: 0x8004)
It should be noted that SDIN3, LRCLK0, BCLK0, LRCLK1, BCLK1, LRCLK2, and BCLK2 can also be used as SPDIF_IN0, SPDIF_IN1,
SPDIF_IN2, SPDIF_IN3, SPDIF_IN4, SPDIF_IN5, and SPDIF_IN6.
Selects the frequency of the internally generated clock to be output on the MCLK_OUT pin. 000
Table 53.
Bit No. Bit Name Description Default
Bits[15:14] SCR2 clock select Used to select the serial clocks used for the input to SRC2. 10
00b = uses LRCLK0 and BCLK0 01b = uses LRCLK1 and BCLK1 10b = uses LRCLK2 and BCLK2 11b = reserved
Bits[13:12] SRC1 clock select Used to select the serial clocks used for the input to SRC1. 00
00b = uses LRCLK0 and BCLK0 01b = uses LRCLK1 and BCLK1 10b = uses LRCLK2 and BCLK2 11b = reserved
Bit[11] Reserved Always write as 0 if writing to this register. 0
Rev. B | Page 53 of 60
ADAV4601
Bit No. Bit Name Description Default
Bit[10] Digout Enable 1
0b = PWM1A and PWM1B in normal operation 1b = PWM1A and PWM1B used as SDO2 and SDO3
Bit[9] Digout Enable 2 Used to change the function of SPDIF output to serial digital output SDO1. 0
0b = SPDIF output normal operation 1b = SPDIF output used as SDO1
Bits[8:7] BCLK frequency (master) Used to set the BCLK frequency when the synchronous serial port is in master mode. 00
00b = 64 × frequency sample, FS (3.072 MHz) 01b = 128 × FS (6.144 MHz) 10b = 256 × FS (12.288 MHz) 11b = reserved
Bits[6:5] Reserved Always write as 0 if writing to this register. 00
Bit[4] Dither enable
0b = disabled 1b = enabled
Bits[3:2] Synchronous port clock select Used to select the serial clocks used for the synchronous digital inputs. 0
00b = uses LRCLK0 and BCLK0 01b = uses LRCLK1 and BCLK1 10b = uses LRCLK2 and BCLK2 11b = reserved
Bit[1]
0b = disabled 1b = enabled
Bit[0] Reserved Always write as 0 if writing to this register. 0
8-channel time division
multiplexing enable
Used to change the function of the PWM1A and PWM1B pins to additional serial
digital outputs, SDO2 and SDO3.
When set to 1, it performs dithering on the digital output when the word width
is set to 20 bits or 16 bits. This reduces the effect of truncation noise.
When set to 1, time division multiplexing mode is enabled. 0
Address 0x0018 Audio Mute Control 1 Register (Default: 0x7F00)
0
0
Table 54.
Bit No. Bit Name Description Default
Bits[15:8] PWM output latency
0x00 = 1.066 ms 0x01 = 2.133 ms … 0x5F = 101.33 ms … 0xFE = 270.93 ms 0xFF = 272 ms
Bits[7:6] Reserved Always write as 0 if writing to this register. 00
Bit[5] PWM zero enable Used to specify the final condition of the PWM channels after a mute. 0
0b = PWM not zeroed after audio mute 1b = PWM zeroed after audio mute
Bit[4] Mute clear select
0b = mute pin rising edge clears mute bit 1b = mute clear gated by clear mute bit
Bit[3] Audio mute Used to control the software mute. 0
0b = unmute 1b = mute
Set the delay from the 50/50 duty-cycle square wave to zero on GND when the output
is muted and Bit[5] is set to 1.
Mute clear select bit. When the mute pin is used to mute the device, the part can be
unmuted in two ways, depending on the condition of this bit.
Rev. B | Page 54 of 60
01011111
0
ADAV4601
Bit No. Bit Name Description Default
Bit[2] Clear mute
0b = no change 1b = clear pin mute
Bit[1] Mute status (read-only) Displays the status of the ADAV4601 mute. 0
0b = currently unmuting or unmuted 1b = currently muting or muted
Bit[0] Mute flag (read-only) Set to 1 when the ADAV4601 is in mute. 0
0b = unmuted 1b = muted
Set to 1 to unmute the ADAV4601 when the external pin has been used to mute the
part and the mute clear select bit is set. Having performed the required action, this bit
automatically resets to 0.
Address 0x0019 PWM Status Register (Default: 0x0000)
Table 55.
Bit No. Bit Name Description Default
Bits[15:4] Reserved Always write as 0 if writing to this register. 000000000000
Bit[3] PWM4 status Set when the PWM4 outputs have gone from zero to 50/50 duty cycle. 0
0b = PWM4 disabled 1b = PWM4 enabled
Bits[2] PWM3 status Set when the PWM3 outputs have gone from zero to 50/50 duty cycle. 0
0b = PWM3 disabled 1b = PWM3 enabled
Bits[1] PWM2 status Set when the PWM2 outputs have gone from zero to 50/50 duty cycle. 0
0b = PWM2 disabled 1b = PWM2 enabled
Bit[0] PWM1 status Set when the PWM1 outputs have gone from zero to 50/50 duty cycle. 0
0b = PWM1 disabled 1b = PWM1 enabled
Address 0x001F PWM Control Register (Default: 0x1070)
0
Table 56.
Bit No. Bit Name Description Default
Bits[15:14] PWM ready configure 00b = PWM ready forced low 00
01b = PWM ready forced high 10b = PWM ready late 11b = PWM ready early
Bit[13] Reserved Always write as 0 if writing to this register. 0
Bit[12] Reserved Always write a 1 if writing to this register 1
Bits[11:7] Reserved Always write as 0 if writing to this register. 00000
Bits[6:4] Reserved Always write a 1 if writing to this register. 111
Bits[3:0] Reserved Always write as 0 if writing to this register. 0000
Bits[15:7] Reserved Always write as 0 if writing to this register. 000000000
Bit[6] SRC2 Channel C enable Used to enable Channel C of SRC2. 0
0b = disabled 1b = enabled
Bits[5:4] SRC2 Channel C input select Used to select the serial data input for SRC2 Channel C. 11
00b = SDIN0 01b = SDIN1 10b = SDIN2 11b = SDIN3
Bit[3] Reserved Always write as 0 if writing to this register. 0
Bit[2] SRC2C Channel B enable Used to enable Channel B of SRC2. 0
0b = disabled 1b = enabled
Bits[1:0] SRC2C Channel B input select Used to select the serial data input for SRC2 Channel B. 10
00b = SDIN0 01b = SDIN1 10b = SDIN2 11b = SDIN3
Address 0x008E SPDIF Transmitter Control 2 Register (Default: 0x002D)
Table 58.
Bit No. Bit Name Description Default
Bits[15:8] Reserved Always write as 0 if writing to this register. 00000000
Bits[7:4] Channel status sampling frequency
0x0 = 44.1 kHz 0x2 = 48 kHz 0x3 = 32 kHz
Bit[3] SPDIF TX word length field size Selects the maximum SPDIF transmitter word length. 1
0b = 20 bits maximum 1b = 24 bits maximum
Bits[2:0] Transmitter word length
Used to set the channel status sampling rate in the SPDIF transmitted
stream; should not change the sample rate but only the status bits.
Used to select how many of the bits set by Bit[3] carry valid data.
If 24-bit maximum
0010
101
Rev. B | Page 56 of 60
ADAV4601
Address 0x0200 EEPROM Self Boot Control Register (Default: 0x0000)
Table 59.
Bit No. Bit Name Description Default
Bits[15] Self-boot enable It initiates a self-boot from the external EEPROM. 0
0b = normal operation 1b = initiates self-boot
Bits[14:6] Reserved Always write as 0 if writing to this register. 000000000
Bit[5] Safe load target/slew RAM Initiates a safe load to the target/slew RAM; cleared when safe load completed. 0
0b = finished 1b = safe load request
Bit[4] Safe load parameter RAM Initiates a safe load to the parameter RAM; cleared when safe load completed. 0
0b = finished 1b = safe load request
Bits[3:0] Reserved Always write as 0 if writing to this register. 0000
Model Temperature Range Package Description Package Option
ADAV4601BSTZ
EVAL-ADAV4601EBZ
1
Z = RoHS Compliant Part. The ADAV4601 is a Pb-free, environmentally friendly product. It is manufactured using the most up-to-date materials and processes. The
coating on the leads of each device is 100% pure Sn electroplate. The device is suitable for Pb-free applications and can withstand surface-mount soldering at up to
255°C (±5°C). In addition, it is backward compatible with conventional SnPb soldering processes. This means the electroplated Sn coating can be soldered with Sn/Pb
solder pastes at conventional reflow temperatures of 220°C to 235°C.
Purchase of licensed I2C components of Analog Devices or one of its sublicensed Associated Companies conveys a license for the purchaser under the Philips I2C Patent
Rights to use these components in an I2C system, provided that the system conforms to the I2C Standard Specification as defined by Philips.