ESS SABRE ES9028PRO Owner's manual

CONFIDENTIAL ADVANCE INFORMATION
Analog Reinvented
ES9028PRO
32-Bit HyperStream®II
8-Channel Audio DAC
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
FEATURES
BENEFITS
Patented 32-bit HyperStream®II DAC
oUp to 135dB DNR (mono mode)
o–120dB THD+N
oIndustry’s highest performance 32-bit audio DAC with unprecedented
dynamic range and ultra-low distortion
oSupports both synchronous and asynchronous sampling modes
Patented Time Domain Jitter Eliminator
oUnmatched audio clarity free from input clock jitter
64-bit accumulator and 32-bit processing
oDistortion-free signal processing
Versatile Digital Input
oSupports SPDIF, PCM (I2S, LJ, RJ 16-32-bit), DoP or DSD input
oSupports up to 768kHz PCM, DSD256 via DoP and native DSD1024
oSupports up to 1.536MHz external oversampling filter
oSupports master and slave timing modes
Integrated DSP functions
oClick-free, soft mute and output volume changes
oProgrammable Zero detection
oDe-emphasis for 32kHz, 44.1kHz, and 48kHz sampling
Customizable output configuration
oMono, stereo, or 8-channel output with either current-mode or voltage-mode
operation (current-mode gives lower THD)
User Programmable Oversampling Filter
o7 ready-to-use preset filters with linear/minimum phase and low-delay options
oSupports custom coefficients for unique sound signature
oSupports external oversampling filter
Clock Gearing
oReduces operating frequency for lower power consumption
Gain Calibration ADC
oEnables uniform output level across all chips
THD compensation
oMinimizes distortion from external PCB components and layout
Full 8-to-8 channel mapping
oAllows channels to be remapped for optimized PCB routing
APPLICATIONS
Digital-Audio Workstations Professional Audio Equipment
Blu-ray / SACD players A/V Receivers
The SABRE®ES9028PRO HyperStream®II 8-channel audio DAC is a high performance 32-bit solution designed for
Audiophile and Studio equipment applications such as SACD players, Blu-ray players, digital preamplifier, A/V receivers,
studio consoles and digital audio workstations.
Part
Number
Description
Package
DNR
(dB)
THD+N
(dB)
32-bit
DAC
I2S / DoP / DSD
/ SPDIF Input
Jitter
Reduction
ES9028PRO
SABRE®PRO 32-bit 8-ch
HypersStream II Audio DAC
64
eTQFP
135 (mono)
129 (8-Ch)
–120
Yes
Yes
Yes
Using the critically acclaimed ESS’ patented HyperStream®II architecture and Time Domain Jitter Eliminator, the
ES9028PRO 32-Bit Audio DAC delivers an unprecedented DNR of up to 135dB in mono mode and THD+N of –120dB, the
industry’s highest performance level that will satisfy the needs of the most demanding audio applications.
The ES9028PRO handles up to 32-bit 768kHz PCM, DSD256 via DoP and native DSD1024 data in master or slave timing
modes. Custom sound signature is supported via a fully programmable FIR filter with 7 presets. Residual distortion from
suboptimal PCB components and layout can be minimized using ES9028PRO’s unique THD compensation circuit, while
chip-to-chip gain variation is minimized via a built-in auto gain calibration circuit.
The SABRE®ES9028PRO sets the standard for HD audio performance, SABRE SOUND®, in a cost-effective, easy-to-use
64-eTQFP package for today’s most demanding digital-audio applications.

CONFIDENTIAL INFORMATION v3.6
July 6, 2020
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
2
FUNCTIONAL BLOCK DIAGRAM
TYPICAL APPLICATION BLOCK DIAGRAM
FL
FR
C
SL
SR
LFE
BL
BR
Audio
Processor ES9028PRO
64-eTQFP
Blu-Ray Player
SACD/CD Player
Home Theater Receiver
Studio Console
Digital Audio Workstation

July 6, 2020
CONFIDENTIAL INFORMATION v3.6
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
3
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
PIN LAYOUT

CONFIDENTIAL INFORMATION v3.6
July 6, 2020
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
4
PIN DESCRIPTIONS
Pin
Name
Pin Type
Reset State
Pin Description
1
VDD_L
Power
Power
Analog Power (+1.2V) for Left channels
2
DAC1
AO
Driven to ground via RDAC
Differential Positive Analog Output 1
3
DAC1B
AO
Driven to ground via RDAC
Differential Negative Analog Output 1
4
DAC3B
AO
Driven to ground via RDAC
Differential Negative Analog Output 3
5
DAC3
AO
Driven to ground via RDAC
Differential Positive Analog Output 3
6
AGND_L
Ground
Ground
Analog Ground for Left channels
7
AVCC_L
Power
Power
Low-Noise Analog Power (+3.3V) for Left channels
8
RESETB
I
Tri-stated
Global Reset Input, Active Low
9
DGND
Ground
Ground
Digital Ground
10
AVCC_L
Power
Power
Low-Noise Analog Power (+3.3V) for Left channels
11
AGND_L
Ground
Ground
Analog Ground for Left channels
12
DAC5
AO
Driven to ground via RDAC
Differential Positive Analog Output 5
13
DAC5B
AO
Driven to ground via RDAC
Differential Negative Analog Output 5
14
DAC7B
AO
Driven to ground via RDAC
Differential Negative Analog Output 7
15
DAC7
AO
Driven to ground via RDAC
Differential Positive Analog Output 7
16
VDD_L
Power
Power
Analog Power (+1.2V) for Left channels
17
AVCC_L
Power
Power
Low-Noise Analog Power (+3.3V) for Left channels
18
AGND_L
Ground
Ground
Analog Ground for Left channels
19
DGND
Ground
Ground
Digital Ground
20
DVDD
Power
Power
Digital Power (+1.2V) for core logic
21
SDA
I/O
Tri-stated
I2C Serial Data Input / Output
22
SCL
I
Tri-stated
I2C Serial Clock Input
23
XOUT
AO
Floating
Crystal oscillator output
24
XIN
AI
Floating
Crystal oscillator input (Note: can also just be a clock input)
25
VCCA
Power
Power
Power (+3.3V) for oscillator / Gain Calibration
26
GPIO4
I/O
Tri-stated
GPIO 4
27
GPIO3
I/O
Tri-stated
GPIO 3
28
GPIO2
I/O
Tri-stated
GPIO 2
29
DVDD
Power
Power
Digital Power (+1.2V) for core of chip
30
DGND
Ground
Ground
Digital Ground
31
AGND_R
Ground
Ground
Analog Ground for Right channels
32
AVCC_R
Power
Power
Low-Noise Analog Power (+3.3V) for Right channels
33
VDD_R
Power
Power
Analog Power (+1.2V) for Right channels
34
DAC8
AO
Driven to ground via RDAC
Differential Positive Analog Output 8
35
DAC8B
AO
Driven to ground via RDAC
Differential Negative Analog Output 8
36
DAC6B
AO
Driven to ground via RDAC
Differential Negative Analog Output 6
37
DAC6
AO
Driven to ground via RDAC
Differential Positive Analog Output 6
38
AGND_R
Ground
Ground
Analog Ground for Right channels
39
AVCC_R
Power
Power
Low-Noise Analog Power (+3.3V) for Right channels

July 6, 2020
CONFIDENTIAL INFORMATION v3.6
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
5
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
PIN DESCRIPTIONS (continued)
Pin
Name
Pin Type
Reset State
Pin Description
40
GPIO1
I/O
Tri-stated
GPIO 1
41
ADDR
I
Tri-stated
Chip Address Select
42
AVCC_R
Power
Power
Low-Noise Analog Power (+3.3V) for Right channels
43
AGND_R
Ground
Ground
Analog Ground for Right channels
44
DAC4
AO
Driven to ground via RDAC
Differential Positive Analog Output 4
45
DAC4B
AO
Driven to ground via RDAC
Differential Negative Analog Output 4
46
DAC2B
AO
Driven to ground via RDAC
Differential Negative Analog Output 2
47
DAC2
AO
Driven to ground via RDAC
Differential Positive Analog Output 2
48
VDD_R
Power
Power
Analog Power (+1.2V) for Right channels
49
AVCC_R
Power
Power
Low-Noise Analog Power (+3.3V) for Right channels
50
AGND_R
Ground
Ground
Analog Ground for Right channels
51
DGND
Ground
Ground
Digital Ground
52
DATA8
I/O
Tri-stated
DSD Data8 or SPDIF Input 9
53
DATA7
I/O
Tri-stated
DSD Data7 or SPDIF Input 8
54
DATA6
I/O
Tri-stated
DSD Data6 or SPDIF Input 7
55
DATA5
I/O
Tri-stated
DSD Data5 or PCM Data CH7 / CH8 or SPDIF Input 6
56
DATA4
I/O
Tri-stated
DSD Data4 or PCM Data CH5 / CH6 or SPDIF Input 5
57
DATA3
I/O
Tri-stated
DSD Data3 or PCM Data CH3 / CH4 or SPDIF Input 4
58
DATA2
I/O
Tri-stated
DSD Data2 or PCM Data CH1 / CH2 or SPDIF Input 3
59
DATA1
I/O
Tri-stated
DSD Data1 or PCM Frame Clock or SPDIF Input 2
60
DATA_CLK
I/O
Tri-stated
PCM Bit Clock or DSD Bit Clock or SPDIF Input 1
61
DVDD
Power
Power
Digital Power (+1.2V) for core of chip
62
AVDD
Power
Power
Digital Power (+1.8V / +3.3V) for top pad ring of chip
63
AGND_L
Ground
Ground
Analog Ground for Left channels
64
AVCC_L
Power
Power
Low-Noise Analog Power (+3.3V) for Left channels
Exposed
Pad
Can be left open, connected to digital or analog ground.
Internally connected to substrate via a conductive epoxy
Notes:
VDD_L, VDD_R and DVDD are internally connected.
I = Digital Input I/O = Input / Output
AI = Analog Input AO = Analog Output
All unused digital inputs should be connected to ground directly, or via a pull-down resistor of 4.7kΩto 47kΩ
5V Tolerant Pins (3.3V AVDD Supply Only)
The following pins are 5V tolerant:
•RESETB
•SDA and SCL
•GPIO1-4
•ADDR
•DATA1-8
•DATA_CLK

CONFIDENTIAL INFORMATION v3.6
July 6, 2020
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
6
System Clock and Audio Inputs
Sampling Rate Notations
Mode
FSR
raw sample rate at
audio interface
fs
sample rate for filter
specification
DSD
DATA_CLK
FSR / 64
DoP
Frame Clock Rate
FSR / 4
Serial (PCM) Normal Mode
Frame Clock Rate
FSR
Serial (PCM) OSF Bypass Mode
Frame Clock Rate
FSR / 8
SPDIF
SPDIF Audio Rate
FSR
System Clock (XIN) and Audio Master Clock (MCLK)
The system clock (XIN) can be generated with a crystal using the built-in oscillator or supplied externally.
oThe maximum XIN frequency is 100MHz as specified in ANALOG PERFORMANCE and XIN Timing.
oThe audio master clock (MCLK) is divided down from XIN via clock_gear in Register 0: System Registers.
oThe minimum MCLK frequency for a given raw sample rate FSR is specified in ANALOG PERFORMANCE.
oThe minimum MCLK frequency for a given I2C clock is specified in the table under I2C Timing Table.
PCM Pin Connections
Pin Name
Description
DATA1
Frame clock
DATA5~2
8-channel PCM serial data
DATA_CLK
Bit clock for PCM audio format
Note: DATA_CLK frequency must be (2 x serial_length) x FSR.
serial_length can be set in Register 2: Serial Data Configuration and Automute Enable
SPDIF Pin Connections
Pin Name
Description
GPIO4~1
SPDIF input 13~10
DATA8~1
SPDIF input 9~2
DATA_CLK
SPDIF input 1
An SPDIF source multiplexer allows for up to 13 SPDIF sources to be connected to the data and GPIO pins selectable via
Register 11: SPDIF Mux and GPIO Inversion. SPDIF input mode can be manually selected by input_select in Register 1:
Input selection or automatically selected if auto_select in Register 1: Input selection is set to a mode allowing automatic
SPDIF selection.
DSD Pin Connections
Pin Name
Description
DATA8~1
8-channel DSD data input
DATA_CLK
Bit clock for DSD data input
Note: DATA_CLK frequency must be FSR.

July 6, 2020
CONFIDENTIAL INFORMATION v3.6
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
7
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
Master Mode
The DAC can become an audio timing master via master_mode in Register 10: Master Mode and Sync Configuration.
oThe ‘input_select’ bits in Register 1: Input selection must be set correctly to select either DSD or serial master mode.
The Bit Clock frequency can be configured using one of the following two methods:
oSet the desired master_div in Register 10: Master Mode and Sync Configuration, or
oUse NCO mode to set FSR using Register 42-45: Programmable NCO. When in NCO mode the master_div setting will
be ignored.
An available GPIO pin can be configured to output MCLK using Register 8: GPIO1-2 Configuration and Register 9: GPIO3-4
Configuration.
DATA_CLK
DATA1
DATA5~2SD[3:0] (Serial PCM Data)
DATA_CLK
DATA1
DATA5~2
GPIOx
MCLK (Master Clock)
SLAVE PCM MODE MASTER PCM MODE
BCLK (Bit Clock)
LRCK (Frame Clock)
BCLK (Bit Clock)
LRCK (Frame Clock)
SD[3:0] (Serial PCM Data)
DATA_CLK
DATA8~1
DATA_CLK
DATA8~1
GPIOx
MCLK (Master Clock)
SLAVE DSD MODE MASTER DSD MODE
DSD DATA_CLK
DSD DATA[7:0]
DSD DATA_CLK
DSD DATA[7:0]

CONFIDENTIAL INFORMATION v3.6
July 6, 2020
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
8
Function Description
Soft Mute
When Mute is asserted the output signal will ramp to the - level. When Mute is reset the attenuation level will ramp back up
to the previous level set by the volume control register. Asserting Mute will not change the value of the volume control
register. The ramp rate is set by Register 6: De-emphasis Filter & Volume Ramp Rate according to the following relationship:
Automute (PCM and SPDIF modes only)
Automute is disabled by default and can be enabled by setting automute_time to a non-zero value. Automute is triggered
when the following conditions are met:
Mode
Detection Condition
Time
PCM
SPDIF
Data is lower than automute_level
for the specified time
Automute_time can be set using Register 4: Automute Time.
Automute_level can be set using Register 5: Automute Level.
The automute status can be read using automute_status in Register 64: Chip ID and Status or via a GPIO pin
programmed as Automute Status using Register 8: GPIO1-2 Configuration or Register 9: GPIO3-4 Configuration.
The triggered automute behavior can be configured using Register 2: Serial Data Configuration and Automute Enable to one
of the followings:
•No action
•Soft Mute
•Ramp all channels to ground to reduce power consumption
•Soft Mute then ramp all channels to ground
The ramp-to-ground rate can be configured to
using Register 14: Soft-Start Configuration.
Volume Control
Each channel has an independently controlled digital attenuation circuit which can be set to attenuate from 0dB to –127dB in
0.5dB steps. When a new volume level is set, the digital attenuation circuit will ramp softly to the new level. To ensure silent
digital volume transitions each 0.5dB step can take as many as 64 intermediate steps depending on the volume_rate setting
in Register 6: De-emphasis Filter & Volume Ramp Rate.
Master Trim
The master trim sets the 0dB reference level for the digital volume control of each DAC. The master trim is programmable
via Register 24-27: Master Trim.The master trim registers store a 32bit signed number and should never exceed the full
scale signed value 32’h7FFFFFFF.
18dB Channel Gain (PCM mode only)
A +18dB gain can be applied on a per-channel based using Register 62: +18dB Channel Gain, in addition to volume control
and master trim. Note that the output will be clipped if the +18dB gain results in larger than full scale output.

July 6, 2020
CONFIDENTIAL INFORMATION v3.6
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
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ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
De-emphasis
The de-emphasis feature is included for audio data that has utilized the 50/15s pre-emphasis for noise reduction. There are
three de-emphasis filters, one for 32kHz, one for 44.1kHz, and one for 48kHz selectable via deemph_sel and bypassed via
deemph_bypass in Register 6: De-emphasis Filter & Volume Ramp Rate.
The de-emphasis filter can automatically be applied when an SPDIF stream sets the de-emphasis flag. It will auto detect the
sample rate (32k, 44.1k, 48k) in either consumer or professional formats and then apply the correct de-emphasis filter. The
automatic enabling of the de-emphasis filter can be enabled via auto_deemph in Register 6: De-emphasis Filter & Volume
Ramp Rate.
Preset Oversampling FIR Filters
Seven pre-programmed digital filters are selectable for SPDIF and PCM serial mode via filter_shape in Register 7: Filter
Bandwidth and System Mute. See ANALOG PERFORMANCE, PCM FILTER FREQUENCY RESPONSE and PCM FILTER
IMPULSE RESPONSE for more information.
Custom Oversampling FIR Filter
The FIR filter can also be programmed as a two-staged interpolation filter with custom coefficients to achieve unique sound
signature. Custom coefficients can be generated using MATLAB and then downloaded using a custom C code.
Example Source Code for Loading a Filter
// only accept 128, 64 or 16 coefficients
// Note: The coefficients must be quantized to 32 bits for this method!
// Note: Stage 1 consists of 128 or 64 values (0-127 being the coefficients)
// Note: Stage 2 consists of 16 values (0-13 being the coefficients, 14-15 are zeros)
// Note: Stage 2 is symmetric about coefficient 13. See the example filters for more information.
byte fir_badr = 32;
byte coeff_stage = (byte)(coeffs.Count == 64 ? 0: 1);
for (int i = 0; i < coeffs.Count; i++)
{
// stage 1 contains 128 or 64 coefficients, while stage 2 contains 16 coefficients
registers.WriteRegister(fir_badr, (byte)((coeff_stage << 7) + i));
// write the coefficient data
registers.WriteRegister(fir_badr+1, (byte)(coeffs[i] & 0xff));
registers.WriteRegister(fir_badr+2, (byte)((coeffs[i] >> 8) & 0xff));
registers.WriteRegister(fir_badr+3, (byte)((coeffs[i] >> 16) & 0xff));
registers.WriteRegister(fir_badr+5, 0x02); // set the write enable bit
}
// disable the write enable bit when we’re done
registers.WriteRegister(fir_badr+5, (byte)(setEvenBit ? 0x04 : 0x00));
Oversampling Filter (OSF) Bypass
The oversampling FIR filter can be bypassed using bypass_osf in Register 37: Programmable FIR Configuration, sourcing
data directly into the IIR filter. The audio input should be oversampled at 8 x fs rate when OSF is bypassed to have the same
IIR filter bandwidth as PCM audio sampled at fs rate. For example, a signal with 44.1kHz sample rate can be oversampled
externally to 8 x 44.1kHz = 352.8kHz and then applied to the serial decoder in either I2S, LJ, or RJ format. The maximum
sample rate that can be applied is 1.536MHz (8 x 192kHz).
IIR Filter
Four filters with cutoffs at 47kHz, 50kHz, 60kHz, and 70kHz scaled by fs/44100 are selectable via iir_bw in Register 7: Filter
Bandwidth and System Mute. See ANALOG PERFORMANCE and IIR FILTER RESPONSE for more information.

CONFIDENTIAL INFORMATION v3.6
July 6, 2020
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
10
Time Domain Jitter Eliminator and DPLL
By default, the DAC works in Jitter Eliminator mode allowing the audio interface timing to be asynchronous to MCLK. A DPLL
constantly updates the FSR/MCLK ratio to calculate the true 32-bit timing of the incoming audio samples allowing the ESS
patented Time Domain Jitter Eliminator to remove any distortion caused by jitter.
•The DPLL acquisition speed can be set by lock_speed in Register 10: Master Mode and Sync Configuration.
•The PCM/SPDIF DPLL bandwidth can be set via dpll_bw_serial in Register 12: Jitter Eliminator / DPLL Bandwidth
•The DSD DPLL bandwidth can be set via dpll_bw_dsd in Register 12: Jitter Eliminator / DPLL Bandwidth
For best performance, the DPLL bandwidth should be set to the minimum setting that will keep the DPLL reliably in lock.
Sample Rate Calculation
The raw sample rate (FSR) can be calculated from Register 66-69 : DPLL Number using the following formula:
Synchronous Mode (PCM mode only)
The DPLL can be bypassed if the incoming PCM audio is synchronous to MCLK with the relationship MCLK=128FSR. This
can be enabled via 128fs_mode in Register 10: Master Mode and Sync Configuration.
DAC Full-Scale Gain Calibration
DAC gain calibration enables uniform output level across multiple chips by compensating for chip-to-chip gain variations.
The DAC full-scale gain-calibration system works by comparing an internal resistor to an external precision resistor of known
value. The two resistors are set up as a voltage divider that is connected between power and ground. The value of the
internal resistor changes with semiconductor process variations so by measuring the divider’s voltage output, using an ADC,
the process variation from nominal can be measured and this is used to correct the DAC gain. As all the DAC channels are
on the same monolithic chip, the channel-to-channel gain variation is very small and does not need to be trimmed.
There are two ADCs and either of the ADC inputs can be used to drive the auto-calibration circuit. The circuit uses the ADC
value, as decimated by the internal programmable decimation filters, to scale the master_trim value. Master_trim can be
programmed as normal but will be scaled by the ADC value when in automatic-calibration mode. In this mode, master_trim
can be set once by enabling automatic calibration, and the DAC output levels will be consistent across all DAC devices.
•Full-scale gain-calibration is enabled using calib_en in Register 63: Auto Calibration.
•calib_sel in Register 63: Auto Calibration selects which ADC to use
•calib_latch in Register 63: Auto Calibration determines whether to use the new ADC correction value or ignore it.
•ADC values update at the ADC_CLK rate which is also programmable in Register 46: ADC Configuration.
The ADC decimation filters may also be programmed to a lower bandwidth to help smooth out any voltage transients on the
divider output.

July 6, 2020
CONFIDENTIAL INFORMATION v3.6
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
11
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
THD Compensation
THD Compensation can be used to minimize distortion from external PCB components and layout through the generation of
inverse second and third harmonic components matching the target system distortion profile.
THD compensation can be enabled via thd_enb in Register 13: Jitter Eliminator / DPLL Configuration & THD Bypass.
The coefficient for manipulating second harmonic distortion is stored in Register 28-29: THD Compensation C2.
The coefficient for manipulating third harmonic distortion is stored in Register 30-31: THD Compensation C3.
All channels use the same compensation coefficients.
Full Channel Mapping, Mono Mode and Stereo Mode
Channel mapping allows output channels to be remapped to arbitrary input channels for optimized PCB routing
Clearing stereo_mode in Register 15: GPIO Input Selection & Volume Configuration allows the data for each DAC to be
sourced from any input channel using Registers 38-41: DAC Channel Mapping.
•Mono mode can be implemented by channel mapping all output DAC sources to the same input channel
Setting stereo_mode in Register 15: GPIO Input Selection & Volume Configuration will source DAC channels 1/3/5/7 and
channels 2/4/6/8 from input channel 1 and 2.
In SPDIF mode, DAC channels 1/3/5/7 and channels 2/4/6/8 are sourced from SPDIF input left and right channels.
Power Supplies
To minimize THD+N, AVCC_L and AVCC_R must be powered by low-noise +3.3V supplies. Although AVCC_L and
AVCC_R could be powered from a single low-noise supply, crosstalk would be compromised and so separate +3.3V supplies
are highly recommended. The ES9311Q dual ultra-low noise regulator is designed to power AVCC_L and AVCC_R and
minimize THD+N and crosstalk on all SABRE PRO DACs.

CONFIDENTIAL INFORMATION v3.6
July 6, 2020
ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
12
Audio Interface Formats
Several digital audio transport formats are supported to allow direct connection to common audio processors. Auto detection
circuitry is enabled by default to detect the input format. The input mode can be explicitly set using Register 1: Input
selection. The following diagrams outline the supported formats (using stereo 2-channel inputs as an example).
PCM LJ, RJ and I2S Formats
The following number of BCLK edges are present per frame (left plus right):
o16-bit mode: 32 BCLKs
o24-bit mode: 48 BCLKs
o32-bit mode: 64 BCLKs
31 30 29 21031 30 29 210
23 22 21 210 23 22 21 210
SD
32-bit MSB LSB MSB LSB
MSB MSBLSB LSB
19 18 17 210 19 18 17 210
MSB MSBLSB LSB
15 14 13 210 15 14 13 210
MSB MSBLSB LSB
23 22
MSB
MSB
19 18
MSB
15 14
MSB
31 30
SD
24-bit
SD
20bit
SD
16bit
BCLK
LRCLK LEFT RIGHT
LEFT JUSTIFIED FORMAT
31 30 29 21031 30 29 210
23 22 21 210 23 22 21 210
SD
32-bit MSB LSB MSB LSB
MSB MSBLSB LSB
19 18 17 210 19 18 17 210
MSB MSBLSB LSB
15 14 13 210 15 14 13 210
MSB MSBLSB LSB
MSB
31 30
SD
24-bit
SD
20bit
SD
16bit
BCLK
LRCLK LEFT RIGHT
RIGHT JUSTIFIED FORMAT
31 30 29 21031 30 29 210
23 22 21 210 23 22 21 210
SD
32-bit MSB LSB MSB LSB
MSB MSBLSB LSB
19 18 17 210 19 18 17 2 1 0
MSB MSBLSB LSB
15 14 13 210 15 14 13 210
MSB MSBLSB LSB
23 22
MSB
MSB
19 18
MSB
15 14
MSB
31 30
SD
24-bit
SD
20bit
SD
16bit
BCLK
LRCLK LEFT RIGHT
I2S FORMAT

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ESS TECHNOLOGY, INC. 237 South Hillview Drive, Milpitas, CA 95035, USA. Tel (408) 643-8800 • Fax (408) 643-8801
DoP (DSD over PCM) Audio Format
The DoP format packs DSD data into PCM frames. The incoming data is identified as DoP if the DSD Markers 0x05 and
0xFA alternating each frame clock cycle are present as illustrated below.
Frame Cycle
1 Left
1 Right
2 Left
2 Right
3 Left
3 Right
DSD Marker
0x05
0x05
0xFA
0xFA
0x05
0x05
Note: DoP requires 24-bit or 32-bit PCM mode and cannot be handled by 16-bit PCM mode.
•24-bit mode: DoP data consists of 8-bit marker in the MSB followed by 16-bit DSD data
•32-bit mode: DoP data consists of 8-bit marker in the MSB followed by 16-bit DSD data and 8-bit padding
Native DSD Format
DATA
WS
BCK
8-bit DSD Marker
16 Bits of DSD Audio8-bit DSD Marker 8 Bits of 0 Padding 8 Bits of 0 Padding16 Bits of DSD Audio
0 31 30 2 1 0 31 30 2 1 0 31 30
................ ................
Left Channel Right Channel
D.. D0 D1 D2 D3 D4
DSD1
DSD2
DCLK
D.. D0 D1 D2 D3 D4
DSD1
DSD2
DCLK
D.. D0 D1 D2 D3 D4
DSD NORMAL MODE
DSD PHASE MODE

CONFIDENTIAL INFORMATION v3.6
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ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
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14
Serial Control Interface
The registers inside the chip are programmed via an I2C interface. The diagram below shows the timing for this interface.
The chip address can be set to 2 different settings via the “ADDR” pin.
ADDR
CHIP ADDRESS
0
0x90
1
0x92
Notes:
1. The “ADDR” pin is used to create the CHIP ADDRESS. (0x90, 0x92)
2. The first byte after the CHIP ADDRESS (“ADDRESS”) is the register address.
3. The second byte after the CHIP ADDRESS (“DATA”) is the data to be programmed into the register.
4. Multi-byte reads are NOT supported and will cause the I2C decoder to become unresponsive until a reset occurs.
I2C Timing Table
Start Start StartStop
Parameter
Symbol
MCLK
Constraint
Standard-Mode
Fast-Mode
Unit
MIN
MAX
MIN
MAX
SCL Clock Frequency
fSCL
< MCLK/20
0
100
0
400
kHz
START condition hold time
tHD,STA
4.0
-
0.6
-
s
LOW period of SCL
tLOW
>10/MCLK
4.7
-
1.3
-
s
HIGH period of SCL (>10/MCLK)
tHIGH
>10/MCLK
4.0
-
0.6
-
s
START condition setup time (repeat)
tSU,STA
4.7
-
0.6
-
s
SDA hold time from SCL falling
tHD,DAT
0
-
0
-
s
SDA setup time from SCL rising
tSU,DAT
250
-
100
-
ns
Rise time of SDA and SCL
tr
-
1000
300
ns
Fall time of SDA and SCL
tf
-
300
300
ns
STOP condition setup time
tSU,STO
4
-
0.6
-
s
Bus free time between transmissions
tBUF
4.7
-
1.3
-
s
Capacitive load for each bus line
Cb
-
400
-
400
pF

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REGISTER SETTINGS
Note: Multi-byte registers use little-endian byte ordering scheme with the least significant byte stored at the lowest register
address and most significant byte stored at the highest register address.
Register 0: System Registers
Bits
[7:4]
[3:2]
[1]
[0]
Mnemonic
osc_drv
clk_gear
reserved
soft_reset
Default
4’b0000
2’b00
1’b0
1’b0
Bit
Mnemonic
Description
[7:4]
osc_drv
Oscillator drive specifies the bias current to the oscillator pad.
•4’b1111: shut down the oscillator
•4’b1110: ¼ bias
•4’b1100: ½ bias
•4’b1000: ¾ bias
•4’b0000: full bias (default)
[3:2]
clk_gear
Configures a clock divider network that can reduce the power
consumption of the chip by reducing the clock frequency supplied to
both the digital core and analog stages.
•2'b00: MCLK = XIN (default)
•2'b01: MCLK = XIN / 2
•2'b10: MCLK = XIN / 4
•2'b11: MCLK = XIN / 8
[1]
reserved
[0]
soft_reset
Software configurable hardware reset with the ability to reset the design
to its initial power-on configuration.
•1’b1: resets the SABRE DAC to its power-on defaults
•1’b0: normal operation (default)
Note: This register will always read as “1’b0” as the power-on default
for this register is “1’b0”. A reset can be verified by checking the status
of other modified registers.

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ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
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16
Register 1: Input selection
Bits
[7]
[6]
[5]
[4]
[3:2]
[1:0]
Mnemonic
user_bits
spdif_ig_data
spdif_ig_valid
reserved
auto_select
input_select
Default
1’b0
1’b0
1’b0
1’b0
2’b11
2’b00
Bit
Mnemonic
Description
[7]
user_bits
Both SPDIF channel status bits and SPDIF user bits are available for readback
via the I2C interface. To reduce register count, the channel status bits and user
bits occupy the same register space. Setting user_bits will present the SPDIF
user bits on the read-only register interface instead of the default channel status
bits.
•1’b1: presents the SPDIF user bits on the read-only register interface
•1’b0: presents the SPDIF channel status bits on the read-only register
interface (default)
[6]
spdif_ig_data
Configures the SPDIF decoder to ignore the ‘data’ flag in the channel status bits.
•1’b1: ignore the data flag in the channel status bits and continue to
process the decoded SPDIF data
•1’b0: mute the SPDIF data when the data flag is set (default)
Note: Enabling the SPDIF output when data is present could cause undesirable
noise if the SPDIF data is compressed audio or a non-standard format.
[5]
spdif_ig_valid
Configures the SPDIF decoder to ignore the ‘valid’ flag in the SPDIF stream.
•1’b1: ignore the valid flag and continue to process the decoded SPDIF
data
•1’b0: mute the SPDIF data when the valid flag is invalid (default)
[4]
reserved
[3:2]
auto_select
Allows the SABRE DAC to automatically select between either serial, SPDIF or
DSD input formats.
•2’b11: automatically select between DSD, SPDIF or serial data (default)
•2’b10: automatically select between SPDIF or serial data
•2’b01: automatically select between DSD or serial data
•2’b00: disable automatic input decoder and instead use the information
provided by register 1[1:0]
[1:0]
input_select
Configures the SABRE DAC to use a particular input decoder if auto_select is
disabled.
•2’b11: DSD
•2’b10: reserved
•2’b01: SPDIF
•2’b00: serial (default)
Note: Register 1[3:2] must be set to 2’b00 for input_select to function.

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Register 2: Serial Data Configuration and Automute Enable
Bits
[7:6]
[5:4]
[3:2]
[1:0]
Mnemonic
automute_config
serial_bits
serial_length
serial_mode
Default
2’b00
2’b11
2’b11
2’b00
Bit
Mnemonic
Description
[7:6]
automute
config
Configures the automute state machine, which allows the SABRE DAC to
perform different power saving and sound optimizations.
•2’b11: perform a mute and then ramp all channels to ground when an
automute condition is asserted
•2’b10: ramp all channels to ground when an automute condition is
asserted
•2’b01: perform a mute when an automute condition is asserted
•2’b00: normal operation (default)
Note: Ramping DAC outputs to ground can reduce the power consumption of
the SABRE DAC in some situations.
Note: This process can be sped up by using the automute_time, volume_rate
and soft_start_time registers.
[5:4]
serial_bits
Selects how many bits consist of a data word in the serial data stream.
•2’b11: 32-bit data words (default)
•2’b10: 32-bit data words
•2’b01: 24-bit data words
•2’b00: 16-bit data words
[3:2]
serial_length
Selects how many DATA_CLK pulses exist per data word.
•2’b11: 32-bit data words (default)
•2’b10: 32-bit data words
•2’b01: 24-bit data words
•2’b00: 16-bit data words
[1:0]
serial_mode
Configures the type of serial data.
•2’b11 or 2’b10: right-justified mode
•2’b01: left-justified mode
•2’b00: I2S mode (default)

CONFIDENTIAL INFORMATION v3.6
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ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
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Register 3: Reserved
Bits
[7:0]
Mnemonic
reserved
Default
8’b00000000
Register 4: Automute Time
Bits
[7:0]
Mnemonic
automute_time
Default
8’d0
Bit
Mnemonic
Description
[7:0]
automute_time
Configures the amount of time the audio data must remain below the
automute_level before an automute condition is flagged. Defaults to
0 which disables automute.
Register 5: Automute Level
Bits
[7]
[6:0]
Mnemonic
reserved
automute_level
Default
1’b0
7’d104
Bit
Mnemonic
Description
[7]
reserved
[6:0]
automute_level
Configures the threshold which the audio must be below before an
automute condition is flagged. The level is measured in decibels (dB)
and defaults to -104dB.
Note: This register works in tandem with automute_time to create the
automute condition.

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Register 6: De-emphasis Filter & Volume Ramp Rate
Bits
[7]
[6]
[5:4]
[3]
[2:0]
Mnemonic
auto_deemph
deemph_bypass
deemph_sel
reserved
volume_rate
Default
1’b0
1’b1
2’b00
1’b1
2’b010
Bit
Mnemonic
Description
[7]
auto_deemph
Automatically engages the de-emphasis filters when SPDIF data is
provides and the SPDIF channel status bits contains valid de-
emphasis settings.
•1’b1: enables automatic de-emphasis
•1’b0: disables automatic de-emphasis (default)
[6]
deemph_bypass
Enables or disables the built-in de-emphasis filters.
•1'b1 disabled de-emphasis filters (default)
•1'b0 enables de-emphasis filters
[5:4]
deemph_sel
Selects which de-emphasis filter is used.
•2’b11: reserved
•2’b10: 48kHz
•2’b01: 44.1kHz
•2’b00: 32kHz (default)
[3]
reserved
Must be set to 1’b1 (default) for normal operation
[2:0]
volume_rate
Selects a volume ramp rate to use when transitioning between
different volume levels. The volume ramp rate is measured in decibels
per second (dB/s). Volume rate is in the range 0-7.

CONFIDENTIAL INFORMATION v3.6
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ES9028PRO 32-Bit HyperStream®II 8-Channel Audio DAC
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20
Register 7: Filter Bandwidth and System Mute
Bits
[7:5]
[4:3]
[2:1]
[0]
Mnemonic
filter_shape
reserved
iir_bw
mute
Default
3’b010
2’b00
2’b00
1’b0
Bit
Mnemonic
Description
[7:5]
filter_shape
Selects the type of filter to use during the 8x FIR interpolation phase.
•3’b111: brickwall filter
•3’b110: hybrid, fast roll-off, minimum phase filter
•3’b100: apodizing, fast roll-off, linear phase filter
•3’b101: reserved
•3’b011: slow roll-off, minimum phase filter
•3’b010: fast roll-off, minimum phase filter (default)
•3’b001: slow roll-off, linear phase filter
•3’b000: fast roll-off, linear phase filter
Note: The FIR filter is only applied to PCM data, DSD bypasses this phase
[4:3]
reserved
[2:1]
iir_bw
Selects the type of filter to use during the 8x IIR interpolation phase.
•2’b11: 1.5873fs (70k @ 44.1kHz)
•2’b10: 1.3605fs (60k @ 44.1kHz)
•2’b01: 1.1338fs (50k @ 44.1kHz)
•2’b00: 1.0757fs (47.44k @ 44.1kHz) (default)
Note: 47.44k filter should only be used for PCM data. Recommended
settings for DSD data are 50k, 60k or 70k.
[0]
mute
Mutes all 8 channels of the SABRE DAC.
•1’b1: mute all eight channels
•1’b0: normal operation (default)
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