The TAS2505 is a low power digital input speaker amp with support for 24-bit digital I2S data mono
playback.
In addition to driving a speaker amp upto 4-Ω, the device also features a mono headphone driver and a
programmable digital-signal processing block. The digital audio data format is programmable to work with
popular audio standard protocols (I2S, left/right-justified) in master, slave, DSP and TDM modes. The
programmable digital-signal processing block can support Bass boost, treble, or EQ functions. An on-chip
PLL provides the high-speed clock needed by the digital signal-processing block. The volume level can be
controlled by register control. The audio functions are controlled using the I2C™ serial bus or SPI bus. The
device includes an on-board LDO that runs off the speaker power supply to handle all internal device
analog and digital power needs. The included POR as power-on-resetcircuit reliably resets the device into
its default state so no external reset is required at normal usage; however, the device does have a reset
pin for more complex system initialization needs. The device also includes two analog inputs for mixing
and muxing in both speaker and headphone analog paths.
Description
Figure 1-1. Simplified Block Diagram
The device can cover operations from 8kHz mono playback to mono 96kHz DAC playback, making it ideal
for portable battery-powered audio and telephony applications. The playback path offers signal processing
blocks for filtering and effects, flexible mixing of analog input signals as well as programmable volume
controls. The voltage supply range for theTAS2505 for analog is 1.5V–1.95V, and for digital it is
1.65V–1.95V. To ease system-level design, a low-dropout regulator (LDO) is integrated to generate the
appropriate analog supply from input voltages ranging from 2.7V to 5.5V. Digital I/O voltages are
supported in the range of 1.1V–3.6V. The required internal clock of the TAS2505 can be derived from
multiple sources, including the MCLK, BCLK or GPIO/DOUT pins or the output of the internal PLL, where
the input to the PLL again can be derived from the MCLK, BCLK or GPIO/DOUT pins. Although using the
internal, fractional PLL ensures the availability of a suitable clock signal, it is not recommended for the
lowest power settings. The PLL is highly programmable and can accept available input clocks in the range
of 512kHz to 50MHz.
The device is available in the 4mm × 4mm, 24-pin QFN package.
Only a small number of digital pins are dedicated to a single function; whenever possible, the digital pins
have a default function, and also can be reprogrammed to cover alternative functions for various
applications.
The fixed-function pins are RST LDO_SEL and the SPI_SEL pin, which are HW control pins. Depending
on the state of SPI_SEL, the two control-bus pins SCL/SSZ and SDA/MOSI are configured for either I2C or
SPI protocol.
Other digital IO pins can be configured for various functions via register control. An overview of available
functionality is given in Section 2.1.3.
2.1.2 Analog Pins
Analog functions can also be configured to a large degree. For minimum power consumption, analog
blocks are powered down by default. The blocks can be powered up with fine granularity according to the
application needs.
Chapter 2
SLAU472–February 2013
TAS2505 Application
2.1.3 Multifunction Pins
Table 2-1 shows the possible allocation of pins for specific functions. The PLL input, for example, can be
programmed to be any of 4 pins (MCLK, BCLK, DIN, GPIO).
: The MCLK pin can drive the PLL and Codec Clock inputs simultaneously.
(2)S(2)
: The BCLK pin can drive the PLL and Codec Clock and audio interface bit clock inputs simultaneously.
(3)S(3)
: The GPIO/DOUT pin can drive the PLL and Codec Clock inputs simultaneously.
(4)
D: Default Function
(5)
E: The pin is exclusively used for this function, no other function can be implemented with the same pin. (If GPIO/DOUT has
been allocated for General Purpose Output, it cannot be used as the INT1 output at the same time.)
To configure the settings seen in Table 2-1, please see the letter-number combination in Table 2-2 for the
appropriate registers to modify. In Table 2-2, the letter/number combination represents the row and the
column number from Table 2-1 in bold type.
Please be aware that more settings may be necessary to obtain a full interface definition matching the
application requirement (see Page 0, Register 25 to 33).
The TAS2505 features a mono audio DAC. It supports a wide range of analog interfaces to support
different headsets such as 16-Ω to 200-Ω impedance and analog line outputs. TheTAS2505 can drive a
speaker upto 4-Ω impedance.
•Analog inputs AINR and AINL, which can be used to pass-through or mix analog signals to output
stages
•Analog outputs class-D speaker driver and headphone/lineout driver providing output capability for the
DAC, AINR, AINL, or a mix of the three
2.3.1 Analog Inputs AINL and AINR
AINL (pin 3 or C2) and AINR (pin 4 or B2) are inputs to Mixer P and Mixer M along with the DAC output.
Also AINL and AINR can be configured inputs to HP driver. Page1 / register 12 provides control signals for
determining the signals routed through Mixer P, Mixer M and HP driver. Input of Mixer P can be
attenuated by Page1 / register 24, input of Mixer M can be attenuated by Page1 / register 25 and input of
HP driver can be attenuated by Page1 / register 22. Also AINL and AINR can be configured to a monaural
differential input with use Mixer P and Mixer M by Page1 / register 12 setting. All the options can be
viewed in the functional block diagram, Figure 2-6.
2.4Audio DAC and Audio Analog Outputs
The mono audio DAC consists of a digital audio processing block, a digital interpolation filter, a digital
delta-sigma modulator, and an analog reconstruction filter. The high oversampling ratio (normally DOSR is
between 32 and 128) exhibits good dynamic range by ensuring that the quantization noise generated
within the delta-sigma modulator stays outside of the audio frequency band. Audio analog outputs include
mono headphone and lineout and mono class-D speaker outputs. Because the TAS2505 contains a mono
DAC, it inputs the mono data from the left channel, the right channel, or a mix of the left and right
channels as [(L + R) ÷ 2], selected by page 0, register 63, bits D5–D4. See Figure 1-1 for the signal flow.
2.4.1 DAC
The TAS2505 mono audio DAC supports data rates from 8 kHz to 192 kHz. The audio channel of the
mono DAC consists of a signal-processing engine with fixed processing blocks, a digital interpolation filter,
multibit digital delta-sigma modulator, and an analog reconstruction filter. The DAC is designed to provide
enhanced performance at low sampling rates through increased oversampling and image filtering, thereby
keeping quantization noise generated within the delta-sigma modulator and observed in the signal images
strongly suppressed within the audio band to beyond 20 kHz. To handle multiple input rates and optimize
power dissipation and performance, the TAS2505 allows the system designer to program the
oversampling rates over a wide range from 1 to 1024 by configuring page 0, register 13 and page 0 /
register 14. The system designer can choose higher oversampling ratios for lower input data rates and
lower oversampling ratios for higher input data rates.
The TAS2505 DAC channel includes a built-in digital interpolation filter to generate oversampled data for
the delta-sigma modulator. The interpolation filter can be chosen from three different types, depending on
required frequency response, group delay, and sampling rate.
DAC power up is controlled by writing to page 0, register 63, bit D7 for the mono channel. The monochannel DAC clipping flag is provided as a read-only bit on page 0 / register 39, bit D7.
The DAC path of the TAS2505 features many options for signal conditioning and signal routing:
•Digital volume control with a range of -63.5 to +24dB
•Mute function
In addition to the standard set of DAC features the TAS2505 also offers the following special features:
•Digital auto mute
•Adaptive filter mode
2.4.1.1DAC Processing Blocks
The TAS2505 implements signal-processing capabilities and interpolation filtering via processing blocks.
These fixed processing blocks give users the choice of how much and what type of signal processing they
may use and which interpolation filter is applied.
The choices among these processing blocks allows the system designer to balance power conservation
and signal-processing flexibility. Table 2-3 gives an overview of all available processing blocks of the DAC
channel and their properties. The resource-class column gives an approximate indication of power
consumption for the digital (DVDD) supply; however, based on the out-of-band noise spectrum, the analog
power consumption of the drivers (AVDD) may differ.
The signal-processing blocks available are:
•First-order IIR
•Scalable number of biquad filters
The processing blocks are tuned for common cases and can achieve high image rejection or low group
delay in combination with various signal-processing effects such as audio effects and frequency shaping.
The available first-order IIR and biquad filters have fully user-programmable coefficients.
2.4.1.2DAC Processing Blocks – Signal Chain Details
2.4.1.2.1 Three Biquads, Filter A
Figure 2-1. Signal Chain for PRB_P2
2.4.1.2.2 Six Biquads, First-Order IIR, Filter A or B
Figure 2-2. Signal Chain for PRB_P1 and PRB_P3
Audio DAC and Audio Analog Outputs
2.4.1.3DAC User-Programmable Filters
Depending on the selected processing block, different types and orders of digital filtering are available. Up
to six biquad sections are available for specific processing blocks.
The coefficients of the available filters are arranged as sequentially-indexed coefficients in two banks. If
adaptive filtering is chosen, the coefficient banks can be switched in real time.
When the DAC is running, the user-programmable filter coefficients are locked and cannot be accessed
for either read or write.
However, the TAS2505 offers an adaptive filter mode as well. Setting page 8, register 1, bit D2 = 1 turns
on double buffering of the coefficients. In this mode, filter coefficients can be updated through the host and
activated without stopping and restarting the DAC. This enables advanced adaptive filtering applications.
In the double-buffering scheme, all coefficients are stored in two buffers (buffers A and B). When the DAC
is running and adaptive filtering mode is turned on, setting page 44, register 1, bit D0 = 1 switches the
coefficient buffers at the next start of a sampling period. This bit is set back to 0 after the switch occurs. At
the same time, page 44, register 1, bit D1 toggles.
The flag in page 44, register 1, bit D1 indicates which of the two buffers is actually in use.
Page 44, register 1, bit D1 = 0: buffer A is in use by the DAC engine; bit D1 = 1: buffer B is in use.
While the device is running, coefficient updates are always made to the buffer not in use by the DAC,
regardless of the buffer to which the coefficients have been written.
No0NoneC1, buffer AC1, buffer A
No0NoneC1, buffer BC1, buffer B
Yes0Buffer AC1, buffer AC1, buffer B
Yes0Buffer AC1, buffer BC1, buffer B
Yes1Buffer BC1, buffer AC1, buffer A
Yes1Buffer BC1, buffer BC1, buffer A
Page 44, Reg 1, Bit D1I2C Writes toWill Updates
The user-programmable coefficients C1 to C70 for the DAC processing blocks are defined on pages 44 to
46 for buffer A and pages 62 to 64 for buffer B.
The coefficients of these filters are each 24-bit, 2s-complement format, occupying three consecutive 8-bit
registers in the register space. Specifically, the filter coefficients are in 1.23 (one dot 23) format with a
range from –1.0 (0x800000) to 0.99999988079071044921875 (0x7FFFFF) .
2.4.1.3.1 First-Order IIR Section
The IIR is of first order and its transfer function is given by
The frequency response for the first-order IIR section with default coefficients is flat.
As part of the PowerTune strategy, the analog properties of the DAC are adjusted. As a consequence, the
full-scale signal swing achieved at the headphone and line outputs must be adjusted. Please see Table 2-
9 for the proper gain compensation values across the different combinations.
Table 2-9. DAC Gain vs. PowerTune Modes
DAC PowerTune ModePowerTune ModeHeadphone Gain
Control
Page 1,Register 3, Bits (D4-CM = 0.75V, Gain forCM = 0.9V, Gain for
D2)375mV
000PTM_P3, PTM_P400
001PTM_P244
010PTM_P11414
2.4.2.2DAC Digital-Volume Control
The DAC has a digital volume-control block which implements programmable gain. Each channel has an
independent volume control that can be varied from 24 dB to –63.5 dB in 0.5-dB steps. The mono-channel
DAC volume can be controlled by writing to page 0, register 65, bits D7–D0. DAC muting and setting up a
master gain control to control the mono channel is done by writing to page 0, register 64, bits D3. The
gain is implemented with a soft-stepping algorithm, which only changes the actual volume by 0.125 dB per
input sample, either up or down, until the desired volume is reached. The rate of soft-stepping can be
slowed to one step per two input samples by writing to page 0, register 63, bits D1–D0. Note that the
default source for volume-control level settings is controlled by register writes to page 0, register 65.
During soft-stepping, the host does not receive a signal when the DAC has been completely muted. This
may be important if the host must mute the DAC before making a significant change, such as changing
sample rates. In order to help with this situation, the device provides a flag back to the host via a readonly register, page 0, register 38, bit D4 for the mono channel. This information alerts the host when the
part has completed the soft-stepping, and the actual volume has reached the desired volume level. The
soft-stepping feature can be disabled by writing to page 0, register 63, bits D1–D0.
If soft-stepping is enabled, the CODEC_CLKIN signal should be kept active until the DAC power-up flag is
cleared. When this flag is cleared, the internal DAC soft-stepping process is complete, and
CODEC_CLKIN can be stopped if desired. (The analog volume control can be ramped down using an
internal oscillator.)
Audio DAC and Audio Analog Outputs
output swing at500mV
RMS
0dB full scale input0dB full scale input
output swing at
RMS
2.4.3 Interrupts
Some specific events in the TAS2505, which may require host-processor intervention, can be used to
trigger interrupts to the host processor. This avoids polling the status-flag registers continuously. The
TAS2505 has two defined interrupts, INT1 and INT2, that can be configured by programming page 0,
register 48 and page 0, register 49. A user can configure interrupts INT1 and INT2 to be triggered by one
or many events, such as:
•Overcurrent condition in headphone drivers/speaker drivers
•Data overflow in the DAC processing blocks and filters
Each of these INT1 and INT2 interrupts can be routed to output pin GPIO. These interrupt signals can
either be configured as a single pulse or a series of pulses by programming page 0, register 48, bit D0
and page 0, register 49, bit D0. If the user configures the interrupts as a series of pulses, the events
trigger the start of pulses that stop when the flag registers in page 0, register 42 and page 0, register 44
are read by the user to determine the cause of the interrupt.
The digital filter coefficients must be programmed through the control interface. All digital filtering for the
DAC signal path must be loaded into the RAM before the DAC is powered on. (Note that default
ALLPASS filter coefficients for programmable biquads are located in boot ROM. The boot ROM
automatically loads the default values into the RAM following a hardware reset (toggling the RST pin) or
after a software reset. After resetting the device, loading boot ROM coefficients into the digital filters
requires 100 μs of programming time. During this time, reading or writing to page 8 through page 15 for
updating DAC filter coefficient values is not permitted. (The DAC should not be powered up until after all
of the DAC configurations have been done by the system microprocessor.)
2.4.5 Updating DAC Digital Filter Coefficients During PLAY
When it is required to update the DAC digital filter coefficients during play, care must be taken to avoid
click and pop noise or even a possible oscillation noise. These artifacts can occur if the DAC coefficients
are updated without following the proper update sequence. The correct sequence is shown in Figure 2-5.
The values for times listed in Figure 2-5 are conservative and should be used for software purposes.
There is also an adaptive mode, in which DAC coefficients can be updated while the DAC is on. For
details, see Section 2.4.1.3.
Audio DAC and Audio Analog Outputs
Figure 2-5. Example Flow For Updating DAC Digital Filter Coefficients During Play
The TAS2505 has four digital mixing blocks. Each mixer can provide either mixing or multiplexing of the
digital audio data. The first mixer/multiplexer can be used to select input data for the mono DAC from left
channel, right channel, or (left channel + right channel) / 2 mixing. This digital routing can be configured by
writing to page 0, register 63, bits D5–D4.
2.4.7 Analog Audio Routing
The TAS2505 has the capability to route the DAC output to either the headphone or the speaker output. If
desirable, both output drivers can be operated at the same time while playing at different volume levels.
The TAS2505 provides various digital routing capabilities, allowing digital mixing or even channel
swapping in the digital domain. All analog outputs other than the selected ones can be powered down for
optimal power consumption.
2.4.7.1Analog Output Volume Control
The output volume control can be used to fine-tune the level of the mixer amplifier signal supplied to the
headphone driver or the speaker driver. This architecture supports separate and concurrent volume levels
for each of the four output drivers. This volume control can also be used as part of the output pop-noise
reduction scheme. This feature is available even if the DAC is powered down.
2.4.7.2Headphone Analog Output Volume Control
For the headphone output, the analog volume control has a range from 0 dB to –78 dB in 0.5-dB steps for
most of the useful range plus mute, as shown in Table 2-10. This volume control includes soft-stepping
logic.
Changing the analog volume for the headphone is controlled by writing to page 1, register 22, bits D6–D0.
Routing the signal from the output of the analog volume control to the input of the headphone power
amplifier via Mixer P and Mixer M is done by writing to page 1, register 12, bit D2.
The analog volume-control soft-stepping time is based on the setting in page 0, register 63, bits D1–D0.
2.4.7.3Class-D Speaker Analog Output Volume Control
For the speaker outputs, the analog volume control has a range from 0 dB to –78 dB in 0.5-dB steps for
most of the useful range plus mute, as seen in Table 2-10. The implementation includes soft-stepping
logic.
Routing the DAC output signal to the analog volume control via Mixer P and Mixer M is done by writing to
page 1, register 12, bits D3. Changing the analog volume for the speaker is controlled by writing to page 1
/ register 46, bits D6–D0.
The analog volume-control soft-stepping time is based on the setting in page 0, register 63, bits D1–D0.
2.4.8 Analog Outputs
Various analog routings are supported for playback. All the options can be viewed in the functional block
diagram, Figure 2-6.
Note: If only use analog input from AINL or, and AINR to
HPOUT as P1/R12/D1=1 or, and P1/R12/D0=1, need to
set P1/R24/D7=1 as HP Out mixer forcedly powered-up.
Mixer P
P1/R12/D1
P1/R12/D0
P1/R12/D2
P1/R12/D7
P1/R12/D6
P1/R12/D5
P1/R12/D4
P1/R12/D3
P1/R22
HP Volume Control
Audio DAC and Audio Analog Outputs
2.4.8.1Headphone Drivers
The TAS2505 features a mono headphone driver (HPOUT) that can deliver up to 28 mW channel, at 1.8-V
supply voltage, into a 16-Ω load. The headphones are used in a single-ended configuration where an accoupling (dc-blocking) capacitor is connected between the device output pins and the headphones. The
headphone driver also supports 32-Ω and 10-kΩ loads without changing any control register settings.
The headphone driver can be configured to reduce the power consumption in the half drive ability mode
by writing 1 to page 1, register 10, bits D2 = 1, also in this mode the headphone driver can support
lineout-drive as well.
The common-mode voltage is set to ≤ AVDD/2.
The headphone driver can be powered on by writing to page 1, register 9, bit D5. The HPOUT output
driver gain can be controlled by writing to page 1 / register 16 bits D5–D0, and it can be muted by writing
to page 1, register 16, bit D6.
The TAS2505 has a short-circuit protection feature for the headphone drivers, which is always enabled to
provide protection. The output condition of the headphone driver during short circuit can be programmed
by writing to page 1, register 11, bit D1. If D1 = 0 when a short circuit is detected, the device limits the
maximum current to the load. If D1 = 1 when a short circuit is detected, the device powers down the
output driver. The default condition for headphones is the current-limiting mode. For a short circuit on the
channel, the output is disabled and a status flag is provided as read-only bits on page 0 / register 45,
bit D5. If shutdown mode is enabled, then as soon as the short circuit is detected, page 0, register 9,
bit D5 (for HPOUT) clears automatically. Next, the device requires a reset to re-enable the output stage.
Resetting can be done in two ways. First, the device master reset can be used, which requires either
toggling the RST pin or using the software reset. If master reset is used, it resets all of the registers.
Second, a dedicated headphone power-stage reset can also be used to re-enable the output stage, and
that keeps all of the other device settings. The headphone power stage reset is done by setting page 1,
register 9, bit D5 for HPOUT. If the fault condition has been removed, then the device returns to normal
operation. If the fault is still present, then another shutdown occurs. Repeated resetting (more than three
times) is not recommended, as this could lead to overheating.
The TAS2505 has an integrated class-D mono speaker driver (SPKP/SPKM) capable of driving an 8-Ω or
4-Ω differential load. The speaker driver can be powered directly from the battery supply (2.7 V to 5.5 V)
on the SPKVDD pins; however, the voltage (including spike voltage) must be limited below the absolutemaximum voltage of 6 V.
The speaker driver is capable of supplying 800 mW per channel with a 3.6-V power supply. Through the
use of digital mixing, the device can connect one or both digital audio playback data channels to either
speaker driver; this also allows digital channel swapping if needed.
The class-D speaker driver can be powered on by writing to page 1, register 45, bit D1. The class-D
output-driver gain can be controlled by writing to page 1, register 48, bits D6–D4, and it can be muted by
writing to page 1, register 48, bit D6 - D4 = 000.
The TAS2505 has a short-circuit protection feature for the speaker drivers that is always enabled to
provide protection. If the output is shorted, the output stage shuts down on the overcurrent condition.
(Current limiting is not an available option for the higher-current speaker driver output stage.) In case of a
short circuit, the output is disabled and a status flag is provided as a read-only bit on page 0, register 46,
bit D7.
If shutdown occurs due to an overcurrent condition, then the device requires a reset to re-enable the
output stage. Resetting can be done in two ways. First, the device master reset can be used, which
requires either toggling the RST pin or using the software reset. If master reset is used, it resets all of the
registers. Second, a dedicated speaker power-stage reset can be used that keeps all of the other device
settings. The speaker power-stage reset is done by setting page 1, register 45, bit D1 for SPKP and
SPKM. If the fault condition has been removed, then the device returns to normal operation. If the fault is
still present, then another shutdown occurs. Repeated resetting (more than three times) is not
recommended, as this could lead to overheating.
To minimize battery current leakage, the SPKVDD voltage level should not be less than the AVDD
voltage level.
The TAS2505 has a thermal protection (OTP) feature for the speaker driver which is always enabled to
provide protection. If the device is overheated, then the output stops switching. When the device cools
down, the output resumes switching. An overtemperature status flag is provided as a read-only bit on
page 0, register 45, bit D7. The OTP feature is for self-protection of the device. If die temperature can be
controlled at the system/board level, then overtemperature does not occur.
Audio DAC and Audio Analog Outputs
2.4.9 Audio Output-Stage Power Configurations
After the device has been configured (following a RST) and the circuitry has been powered up, the audio
output stage can be powered up and powered down by register control.
These functions soft-start automatically. By using these register controls, it is possible to turn all four
stages on at the same time without turning two of them off.
See Table 2-11 for register control of audio output stage power configurations.
Table 2-11. Audio Output Stage Power Configurations
Audio Output PinsDesired FunctionPage 1 / Register, Bit Value
The TAS2505 has a built-in LDO which can generate the analog supply (AVDD) also the digital supply
(DVDD) from input voltage range of 2.7 V to 5.5 V with high PSRR. If combined power supply current is
50 mA or less, then this LDO can deliver power to both analog and digital power supplies. If the only
speaker power supply is present and LDO Select pin is enabled, the LDO can power up without requiring
other supplies. This LDO requires a minimum dropout voltage of 300 mV and can support load currents up
to 50 mA. For stability reasons the LDO requires a minimum decoupling capacitor of 1 µF (±50%) on the
analog supply (AVDD) pin and the digital supply (DVDD) pin. If use this LDO output voltage for the digital
supply (DVDD) pin, the analog supply (AVDD) pin connected to the digital supply (DVDD) externally is
required.
The LDO is by default powered down for low sleep mode currents and can be enabled driving the
LDO_SEL pin to SPKVDD (Speaker power supply). When the LDO is disabled the AVDD pin is tri-stated
and the device AVDD needs to be powered using external supply. In that case the DVDD pin is also tristated and the device DVDD needs to be powered using external supply. The output voltage of this LDO
can be adjusted to a few different values as given in the Table 2-12. A Circuit Configuration with Internal
LDO is shown in Section 1.3
2) POR_RTB resets I C, I S, Signal Processing and all registers.
22
3) SW_
registers but not digital interfaces
RSTB from register bit resets only the Signal Processing
and all
.
POR_RSTB
After
15
SCLs
1.8V
Default
IOVDD
DVDD
POR_RSTB
time
time
time
SCLK
www.ti.com
2.4.11 POR
TAS2505 has a POR (Power On Reset) function as shown Figure 2-7. This function insures that all
registers are automatically set to defaults when a proper power up sequence is executed. The function
consume approximately 35uA from the DVDD so if needed this can be disabled by page 1, register 1, bit
D3 = 1.
The following paragraphs are intended to guide a user through the steps necessary to configure the
TAS2505.
Step 1
The system clock source (master clock) and the targeted DAC sampling frequency must be identified.
Depending on the targeted performance, the decimation filter type (A or B) and DOSR value can be
determined:
•Filter A should be used for 48-kHz high-performance operation; DOSR must be a multiple of 8.
•Filter B should be used for up to 96-kHz operations; DOSR must be a multiple of 4.
In all cases, DOSR is limited in its range by the following condition:
2.8 MHz < DOSR x DAC_fS < 6.2 MHz(3)
Based on the identified filter type and the required signal processing capabilities, the appropriate
processing block can be determined from the list of available processing blocks (PRB_P1, PRB_P2 and
PRB_P3).
Based on the available master clock, the chosen DOSR and the targeted sampling rate, the clock divider
values NDAC and MDAC can be determined. If necessary, the internal PLL can add a large degree of
flexibility.
In summary, CODEC_CLKIN (derived directly from the system clock source or from the internal PLL)
divided by MDAC, NDAC, and DOSR must be equal to the DAC sampling rate DAC_fS. The
CODEC_CLKIN clock signal is shared with the DAC clock generation block.
CODEC_CLKIN = NDAC × MDAC × DOSR × DAC_fS(4)
NDAC and MDAC can be chosen independently in the range of 1 to 128. In general, NDAC should be as
large as possible as long as the following condition can still be met:
MDAC × DOSR / 32 ≥ RC(5)
RC is a function of the chosen processing block and is listed in Table 2-3.
The common-mode voltage setting of the device is determined by the available analog power supply. This
common-mode (input common-mode) value is common across the ADC, DAC and analog bypass path.
The output common-mode setting is determined by the available analog power supplies (AVdd and ) and
the desired output-signal swing.
At this point, the following device specific parameters are known:
PRB_Rx, DOSR, NDAC, MDAC, input and output common-mode values.
If the PLL is used, the PLL parameters P, J, D and R are determined as well.
Step 2
Setting up the device via register programming:
The following list gives a sequence of items that must be executed in the time between powering the
device up and reading data from the device:
1. Define starting point:
(a) Power up applicable external power supplies
(b) Set register page to 0
(c) Initiate SW reset
2. Program Clock Settings
(a) Program PLL clock dividers P, J, D and R (if PLL is necessary)
(b) Power up PLL (if PLL is necessary)
(c) Program and power up NDAC
(d) Program and power up MDAC
(e) Program OSR value
(f) Program I2S word length if required (16, 20, 24, or 32 bits)
(g) Program the processing block to be used
(h) Micellaneous page 0 controls
At this point, at the latest, the analog power supply must be applied to the device
3. Program Analog Blocks
(a) Set register page to 1
(b) Disable coarse AVDD generation
(c) Enable Master Analog Power Control
(d) Program common-mode voltage
(e) Program headphone-specific de-pop settings (if a headphone driver is used)
(f) Program routing of DAC output to the output amplifier (headphone and lineout or speaker)
(g) Unmute and set gain of output drivers
(h) Power up output drivers
4. Apply waiting time determined by the de-pop settings and the soft-stepping settings of the driver gain
or poll page 1, register 63
5. Power up DAC
(a) Set register page to 0
(b) Power up DAC channels and set digital gain
(c) Unmute digital volume control
Detailed examples can be found from Section 4.0.7 to Section 4.0.12.
2.5PowerTune
PowerTune
The TAS2505features PowerTune, a mechanism to balance power-versus-performance tradeoffs at the
time of device configuration. The device can be tuned to minimize power dissipation, to maximize
performance, or to an operating point between the two extremes to best fit the application.
2.5.1 PowerTune Modes
2.5.1.1DAC - Programming PTM_P1 to PTM_P4
On the playback side, the performance is determined by a combination of register settings and the audio
data word length applied. For the highest performance setting (PTM_P4), an audio-data word length of 20
bits is required, while for the modes PTM_P1 to PTM_P3 a word length of 16 bits is sufficient.
PTM_P1PTM_P2PTM_P3PTM_P4
Pg 1, Reg 3, D(4:2)0x20x10x00x0
Audio Data word length16 bits16 bits16 bits20 or more bits
Pg 0, Reg 27, D(5:4)0x000x000x000x1, 0x2, 0x3
2.5.1.2Processing Blocks
The choice of processing blocks, PRB_P1 to PRB_P3 for playback, also influences the power
consumption. In fact, the numerous processing blocks have been implemented to offer a choice between
power-optimization and configurations with more signal-processing resources.
The tables in this section give recommendations for various DAC PowerTune modes. Typical performance
and power-consumption numbers for line-out signals are listed.
All measurements were taken with the PLL turned off, no signal is present, and the DAC modulator is fully
running. PowerTune modes which are not supported are marked with an ‘X’.