DESCRIPTION
The WM9713L is a highly integrated input/output device designed for mobile computing and communications.
The chip is architected for dual CODEC operation, supporting Hi-Fi stereo Codec functions via the AC link interface, and additionally supporting voice Codec functions via a PCM type Synchronous Serial Port (SSP). A third, auxiliary DAC is provided which may be used to support generation of supervisory tones, or ring-tones at different sample rates to the main codec.
The device can connect directly to a 4-wire or 5-wire touchpanel, mono or stereo microphones, stereo headphones and a stereo speaker, reducing total component count in the system. Cap-less connections to the headphones, speakers, and earpiece may be used, saving cost and board area. Additionally, multiple analog input and output pins are provided for seamless integration with analog connected wireless communication devices.
All device functions are accessed and controlled through a single AC-Link interface compliant with the AC’97 standard. The 24.576 MHz masterclock can be input directly or generated internally from a 13MHz (or other frequency) clock by an on-chip PLL. The PLL supports a wide range of input clock from 2.048MHz to 78.6MHz.
The WM9713L operates at supply voltages from 1.8V to 3.6V. Each section of the chip can be powered down under software control to save power. The device is available in a small leadless 7x7mm QFN package, ideal for use in hand-held portable systems.

FEATURES
*AC’97 Rev 2.2 compatible stereo codec
-DAC SNR 94dB, THD –85dB
-ADC SNR 87dB, THD –86dB
-Variable Rate Audio, supports all WinCE sample rates
-Tone Control, Bass Boost and 3D Enhancement
*On-chip 45mW headphone driver
*On-chip 400mW mono or stereo speaker drivers
*Stereo, mono or differential microphone input
-Automatic Level Control (ALC)
-Mic insert and mic button press detection
*Auxiliary mono DAC (ring tone or DC level generation)
*Seamless interface to wireless chipset
*Resistive touchpanel interface
-Supports 4-wire and 5-wire panels
-12-bit resolution, INL ±2 LSBs (<0.5 pixels)
-X, Y and touch-pressure (Z) measurement
-Pen-down detection supported in Sleep Mode
*Additional PCM/I2S interface to support voice CODEC
*PLL derived audio clocks.
*Supports input clock ranging from 2.048MHz to 78.6MHz
*1.8V to 3.6V supplies (digital down to 1.62V, speaker up to 4.2V)
*7x7mm 48-lead QFN package

APPLICATIONS
*Personal Digital Assistants (PDA) with or without phone
*Smartphones
*Handheld and Tablet Computers

WM9713LGEFL/V, WM9713LGEFL/RV

DESCRIPTION
The WM9715L is a highly integrated input / output device designed for mobile computing and communications. The device can connect directly to a 4-wire or 5-wire touchpanel, mono or stereo microphones, stereo headphones and a mono speaker, reducing total component count in the system. Additionally, phone input and output pins are provided for seamless integration with wireless communication devices.
The WM9715L also offers up to four auxiliary ADC inputs for analogue measurements such as temperature or light. To monitor the battery voltage in portable systems, the WM9715L has two uncommitted comparator inputs.
All device functions are accessed and controlled through a single AC-Link interface compatible with the AC’97 standard (rev 2.2). Additionally, the WM9715L can generate interrupts to indicate pen down, pen up, availability of touchpanel data, low battery, and dead battery.
The WM9715L operates at supply voltages from 1.8 to 3.6 Volts. Each section of the chip can be powered down under software control to save power. The device is available in a small leadless 7x7mm QFN package, ideal for use in handheld portable systems.

FEATURES
*AC’97 Rev 2.2 compatible stereo codec
-DAC SNR 90dB, THD –86dB
-ADC SNR 88dB, THD –88dB
-Variable Rate Audio, supports all WinCE sample rates
-Tone Control, Bass Boost and 3D Enhancement
*On-chip 45mW headphone driver
*On-chip 400mW mono speaker driver
*Stereo, mono or differential microphone input
-Automatic Level Control (ALC)
*Auxiliary mono DAC (ring tone or DC level generation)
*Seamless interface to wireless chipset
*Resistive touchpanel interface
-Supports 4-wire and 5-wire panels
-12-bit resolution, INL ±3 LSBs (<0.5 pixels)
-X, Y and touch-pressure (Z) measurement
-Pen-down detection supported in Sleep Mode
*2 comparator inputs for battery monitoring
*Up to 4 auxiliary ADC inputs
*1.8V to 3.6V supplies
*7x7mm QFN

APPLICATIONS
*Personal Digital Assistants (PDA)
*Smartphones
*Handheld and Tablet Computers

WM9715LGEFL/V, WM9715LGEFL/RV, WM9715LGEFL, WM9715LGEFLV

GENERAL DESCRIPTION
The CS42L51 is a highly integrated, 24-bit, 96 kHz, low power stereo CODEC. Based on multi-bit, delta-sigma modulation, it allows infinite sample rate adjustment between 4 kHz and 96 kHz. Both the ADC and DAC offer many features suitable for low power, portable system applications.
The ADC input path allows independent channel control of a number of features. An input multiplexer selects between line-level or microphone level inputs for each channel. The microphone input path includes a selectable programmable-gain pre-amplifier stage and a low noise MIC bias voltage supply. A PGA is available for line or microphone inputs and provides analog gain with soft ramp and zero cross transitions. The ADC also features a digital volume attenuator with soft ramp transitions. A programmable ALC and Noise Gate monitor the input signals and adjust the volume levels appropriately.
The DAC output path includes a digital signal processing engine. Tone Control provides bass and treble adjustment of four selectable corner frequencies. The Mixer allows independent volume control for both the ADC mix and the PCM mix, as well as a master digital volume control for the analog output. All volume level changes may be configured to occur on soft ramp and zero cross transitions. The DAC also includes de-emphasis, limiting functions and a beep generator delivering tones selectable across a range of two full octaves.
The stereo headphone amplifier is powered from a separate positive supply and the integrated charge pump provides a negative supply. This allows a ground-centered analog output with a wide signal swing and eliminates external DC-blocking capacitors.
In addition to its many features, the CS42L51 operates from a low-voltage analog and digital core, making this CODEC ideal for portable systems that require extremely low power consumption in a minimal amount of space.
The CS42L51 is available in a 32-pin QFN package in both Commercial (-10 to +70° C) and Automotive grades (-40 to +85° C). The CDB42L51 Customer Demonstration board is also available for device evaluation and implementation suggestions. Please see “Ordering Information” on page 85 for complete details.

FEATURES
*24-bit Converters
*4 kHz to 96 kHz Sample Rate
*Multi-bit Delta Sigma Architecture
*Low Power Operation
– Stereo Playback: 12.93 mW @ 1.8 V
– Stereo Record and Playback: 20.18 mW @ 1.8 V
*Variable Power Supplies
– 1.8 V to 2.5 V Digital & Analog
– 1.8 V to 3.3 V Interface Logic
*Power Down Management
– ADC, DAC, CODEC, MIC Pre-Amplifier, PGA
*Software Mode (I²C® & SPI™ Control)
*Hardware Mode (Stand-Alone Control)
*Digital Routing/Mixes:
– Analog Out = ADC + Digital In
– Digital Out = ADC + Digital In
– Internal Digital Loopback
– Mono Mixes
*Flexible Clocking Options
– Master or Slave Operation
– High-Impedance Digital Output Option (for easy MUXing between CODEC and Other Data Sources)
– Quarter-Speed Mode - (i.e. Allows 8 kHz Fs while maintaining a flat noise floor up to 16 kHz)

APPLICATIONS
*HDD & Flash-Based Portable Audio Players
*MD Players/Recorders
*PDAs
*Personal Media Players
*Portable Game Consoles
*Digital Voice Recorders
*Digital Camcorders
*Digital Cameras
*Smart Phones

CS42L51-CNZ, CS42L51-CNZR, CS42L51-DNZ, CS42L51-DNZR

DESCRIPTION
The TLV320AIC33 is a low power stereo audio codec with stereo headphone amplifier, as well as multiple inputs and outputs programmable in single-ended or fully differential configurations. Extensive register- based power control is included, enabling stereo 48-kHz DAC playback as low as 14 mW from a 3.3-V analog supply, making it ideal for portable battery-powered audio and telephony applications.
The record path of the TLV320AIC33 contains integrated microphone bias, digitally controlled stereo microphone preamplifier, and automatic gain control (aAnGalCog), iwnpituhtsm. Tx/hmeupxlacyabpaacbkiliptyatahminocnlgudtehse mmixu/lmtipulex capability from the stereo DAC and selected inputs, through programmable volume controls, to the various outputs.
The TLV320AIC33 contains four high-power output drivers as well as three fully differential output drivers. The high-power output drivers are capable of tdorivfoinugr achvaanrnieetyls ooff losaindglceo-ennfidgeudrat1io6n-Ws, hinecaluddpihnognueps using ac-coupling capacitors, or stereo 16-W headphones in a capacitorless output configuration. In addition, pairs of drivers can be used to drive 8-W speakers in a BTL configuration at 500 mW per channel.
The stereo audio DAC supports sampling rates from 8 kHz to 96 kHz and includes programmable digital filtering in the DAC path for 3D, bass, treble, midrange effects, speaker equalization, and de-emphasis for 32-kHz, 44.1-kHz, and 48-kHz rates. The stereo audio ADC supports sampling rates from 8 kHz to 96 kHz and is preceded by programmable gain amplifiers providing up to +59.5-dB analog gain for low-level microphone inputs.
The serial control bus supports SPI or I2C protocols, while the serial audio data bus is programmable for I2S, left/right-justified, DSP, or TDM modes. A highly programmable PLL is included for flexible clock generation and support for all standard audio rates from a wide range of available MCLKs, varying from 512 kHz to 50 MHz, with special attention paid to the most popular cases of 12-MHz, 13-MHz, 16-MHz, 19.2-MHz, and 19.68-MHz system clocks.
The TLV320AIC33 operates from an analog supply of 2.7 V–3.6 V, a digital core supply of 1.525 V–1.95 V, and a digital I/O supply of 1.1 V–3.6 V. The device is available in 5 X 5-mm, 80-ball MIcroStar Junior™ BGA and 7 X 7-mm, 48-lead QFN.

FEATURES
*Stereo Audio DAC
– 100-dBA Signal-to-Noise Ratio
– 16/20/24/32-Bit Data
– Supports Rates From 8 kHz to 96 kHz
– 3D/Bass/Treble/EQ/De-emphasis Effects
*Stereo Audio ADC
– 92-dBA Signal-to-Noise Ratio
– Supports Rates From 8 kHz to 96 kHz
*Ten Audio Input Pins
– Programmable in Single-Ended or Fully Differential Configurations
– 3-State Capability for Floating Input Configurations
*Seven Audio Output Drivers
– Stereo 8-W, 500-mW/Channel Speaker Drive Capability
– Stereo Fully Differential or Single-Ended Headphone Drivers
– Fully Differential Stereo Line Outputs
– Fully Differential Mono Output
*Low Power: 14-mW Stereo 48-kHz Playback With 3.3-V Analog Supply
*Programmable Input/Output Analog Gains
*Automatic Gain Control (AGC) for Record
*Programmable Microphone Bias Level
*Programmable PLL for Flexible Clock Generation
*Control Bus Selectable SPI or I2C
*Audio Serial Data Bus Supports I2S, Left/Right-Justified, DSP, and TDM Modes
*Alternate Serial PCM/I2S Data Bus for Easy Connection to Bluetooth™ Module
*Digital Microphone Input Support
*Extensive Modular Power Control
*Power Supplies:
– Analog: 2.7 V–3.6 V.
– Digital Core: 1.525 V–1.95 V

TLV320AIC33IZQE, TLV320AIC33IZQER, TLV320AIC33IGQE, TLV320AIC33IGQER, TLV320AIC33IRGZT, TLV320AIC33IRGZR

Description
 The CD22354A and CD22357A are monolithic silicongate, double-poly CMOS integrated circuits containing the band-limiting filters and the companding A/D and D/A conversion circuits that conform to the AT&T D3/D4 specifications and CCITT recommendations.
The CD22354A provides the AT&T m-law and the CD22357A provides the CCITT A-law companding characteristic.
The primary applications for the CD22354A and CD22357A are in telephone systems.
These circuits perform the analog and digital conversions between the subscriber loop and the PCM highway in a digital switching system.
The functional block diagram is shown below.
With flexible features, including synchronous and asynchronous operations and variable data rates, the CD22354A and CD22357A are ideally suited for PABX, central office switching system, digital telephones as well as other applications that require accurate A/D and D/A conversions and minimal conversion time.

Features
* Meets or Exceeds All AT&T D3/D4 Specifications and CCITT Recommendations
* Complete CODEC and Filtering Systems: No External Components for Sample-and-Hold and Auto-Zero Functions.
Receive Output Filter with (SIN X)/X Correction and Additional 8kHz Suppression
* Variable Data Clocks - From 64kHz . . . . . . . . . . . . . . . . . . . . . 2.1MHz
* Receiver Includes Power-Up Click Filter
* TTL or CMOS-Compatible Logic
* ESD Protection on All Inputs and Outputs

Applications
* PABX
* Central Office Switching Systems
* Accurate A/D and D/A Conversions
* Digital Telephones
* Cellular Telephone Switching Systems
* Voice Scramblers - Descramblers
* T1 Conference Bridges
* Voice Storage and Retrieval Systems
* Sound Based Security Systems
* Computerized Voice Analysis
* Mobile Radio Telephone Systems
* Microwave Telephone Networks
* Fiber-Optic Telephone Networks

CD22354AE
CD22357AE
CD22357A

Description
 VIA Technologies’ VT1616™ 20-bit ∑Δ audio codec conforms to the AC’97 2.2 and S/PDIF Output specifications.
The VT1616 integrates Sample Rate Converters on all channels andcan be adjusted in 1Hz increments.
There is a provision in hardware for down-mixing the 6 channels into stereo when only two endpoints are available.
The analog mixer circuitry integrates a stereo enhancement to provide a pleasing 3D surround
sound effect for stereo media.
This codec is designed with aggressive power management to achieve low power consumption.
When usedwith a 3.3V analog supply, power consumption is further reduced.
The primary applications for this part are desktop and portable personal computers multimedia subsystems. However, it is suitable for any system requiring 6-channel audio output for home theater systems at competitive prices.

Features
• AC’97 2.2 compliant codec
• 20-bit, stereo ADC and6-channel stereo DACs
• 1 Hz resolution VSR on all channels
• IntegratedIEC958 line driver for S/PDIF output
• S/PDIF compressed digital or LPCM audio out
• Hardware downmix option to 2 channels
• ADC DC removal for removing recording white noise
• 4-bit 3D stereo expansion for simulatedsurround
• 4 stereo, 2 mono analog line-level inputs
• Secondline-level output with volume control
• External Audio Amplifier Control
• Low Power consumption mode
• Exceeds Microsoft® WHQL logo requirements
• 3.3V digital, 3.3 or 5V analog power supply
• 48-pin LQFP small footprint package

Overview
The two Channel Codec Filter PEB 2265 IOM-2 – SICOFI-2 is the logic continuation of a well-established family of Siemens Codec-Filter-ICs.
The IOM-2 – SICOFI-2 is a fully integrated PCM codec and filter fabricated in low power 1 mm CMOS technology for applications in digital communication systems.
Based on an advanced digital filter concept, the PEB 2265 provides excellent transmission performance and high flexibility.
The new filter concept (second generation) lends to a maximum of independence between the different filter blocks.
Each filter block can be seen like an one to one representative of the corresponding network element.
Only very few external components are needed, to complete the functionality of the IOM-2 – SICOFI-2.
The internal level accuracy is based on a very accurate bandgap reference.
The frequency behavior is mainly determined by digital filters, which do not have any fluctuations.
As a result of the new ADC and DAC concepts linearity is only limited by second order parasitic effects.
Although the device works with only one single 5-V supply there is a very good dynamic range available.

Features
• Single chip CODEC and FILTER to handle two CO- or PABX-channels
• Specification according to relevant CCITT, EIA and LSSGR recommendations
• Digital signal processing technique
• Programmable interface optimized to current feed SLICs and transformer solutions
• Four pin serial IOM-2 Interface
• Single power supply 5 V
• Advanced low power 1mm analog CMOS technology
• Standard 64-pin P-MQFP-64 package
• High performance Analog to Digital Conversion
• High performance Digital to Analog Conversion
• Programmable digital filters to adapt the transmission behavior especially for
– AC impedance matching
– transhybrid balancing
– frequency response
– gain
• Advanced test capabilities
– all digital pins can be tested within a boundary scan scheme (IEEE 1149.1)
– five digital loops
– four analog loops
– two programmable tone generators per channel
• Comprehensive development platform available
– software for automatic filter coefficient calculation – QSICOS
– Hardware development board – STUT 2465

Description
VS1103b is a single-chip MIDI/ADPCM/WAV audio decoder and ADPCM encoder that can handle upto three simultaneous audio streams.
It can also act as a Midi synthesizer.
VS1103b contains a high-performance, proprietary low-power DSP processor core VS DSP4, working data memory, 5 KiB instruction RAM and 0.5 KiB data RAM for user applications, serial
control and input data interfaces, 4 general purpose I/O pins, an UART, as well as a high-quality variable-sample-rate mono ADC and stereo DAC, followed by an earphone amplifier and a common buffer.
VS1103b receives its input bitstreams through serial input buses, which it listens to as a system
slave.
The input streams are decoded and passed through digital volume controls to an 18-bit oversampling, multi-bit, sigma-delta DAC.
Decoding is controlled via a serial control bus.
In addition to basic decoding, it is possible to add application specific features, like DSP effects, to user RAM memory.

Features
• Mixes three audio sources
– General MIDI 1+ / SP-MIDI
– WAV (PCM + IMA ADPCM)
– Microphone or line input
• Encodes IMA ADPCM from microphone, line input or mixed output
• Input streams can use different sample rates
• EarSpeaker Spatial Processing
• Bass and treble controls
• Operates with a single 12. . . 13 MHz clock
• Internal PLL clock multiplier
• Low-power operation
• High-quality on-chip stereo DAC with no phase error between channels
• Stereo earphone driver capable of driving a 30­ load
• Separate operating voltages for analog, digital and I/O
• 5.5 KiB On-chip RAM for user code / data
• Serial control and data interfaces
• Can be used as a slave co-processor
• SPI flash boot for special applications
• UART for debugging purposes
• New functions may be added with software and 4 GPIO pins

General Description
The AS3514 is a low power stereo audio codec and is designed for Portable Digital Audio Applications. It allows playback in CD quality and recording in FM-stereo quality. It has a variety of audio inputs and outputs to directly connect electret microphones, 16Ω headset, 4Ω speaker and auxiliary signal sources via a 10-channel mixer. It only consumes 22mW in playback mode.
Further the device offers advanced power management functions. All necessary ICs and peripherals in a flash based Digital Audio Player are supplied by the AS3514. The power
management block generates 9 different supply voltages out of the battery supply. CPU, NAND flash, SRAM, memory cards, LCD back-light, USB RX/TX can be powered. The different
supply voltages are programmable via the serial control interface. It also contains a charger and is designed for battery supplies from 1V to 5V.
The AS3514 has an on-chip, phase locked loop (PLL) controlled, clock generator. It generates 44.1kHz, 48kHz and other sample rates defined in MP3, AAC, WMA, OGG VORBIS etc. No additional external crystal or PLL is needed. Further the AS3514 has an independent 32kHz real time clock (RTC) on chip which allows a complete power down of the system CPU.

Key Features
Multi-bit Sigma Delta Converters
– DAC: 18bit with 94dB SNR (‘A’ weighted) , 48kHz
– ADC: 14bit with 82dB SNR (‘A’ weighted), 16kHz
2 Microphone Inputs
– 3 gain pre-setting (28dB/34dB/40dB) with AGC
– 32 gain steps @1.5dB and MUTE
– supply for electret microphone
– microphone detection
– remote control by switch
2 Line Inputs
– volume control via serial interface
– 32 steps @1.5dB and MUTE
– stereo or 2x mono or mono differential
Line Outputs
– volume control via serial interface
– 32 steps @1.5dB and MUTE
– 1Vp @10kΩ
Audio Mixer
– 10 channel input/output mixer with AGC
– mixes line inputs and microphones with DAC
– left and right channels independent
High Efficiency Headphone Amplifier
– volume control via serial interface
– 32 steps @1.5dB and MUTE
– 2x40mW @16Ω driver capability
– headphone and over-current detection
– phantom ground eliminates large capacitors
High Power Speaker Amplifier
– volume control via serial interface
– 32 steps @1.5dB and MUTE
– 2x500mW @8Ω driver capability
– over-current detection
Power Management
– step up for system supply (3.0V – 3.6V)
– step down for CPU core (0.85V – 1.8V, 250mA)
– step up for backlight (15V, 38.5mA)
– LDO for digital supply (2.9V, 200mA)
– LDO for analogue supply (2.9V, 200mA)
– LDO for peripherals (1.7V-3.3V, 200mA)
– LDO for peripherals (3.1V-3.3V, 200mA)
– LDO for RTC (1.0V-2.5V, 2mA)
– LDO for USB 1.1 transceiver (3.26V, 10mA)
– battery supervision
– 10sec emergency shut-down
Battery Charger
– automatic trickle charge (50mA)
– prog. constant current charging (100-400mA)
– prog. constant voltage charging (3.9V-4.25V)
Real Time Clock
– ultra low power 32kHz oscillator
– 32bit RTC sec counter
– selectable alarm (seconds or minutes)
General Purpose ADC
– 10bit resolution
– 16 inputs analogue multiplexer
Interfaces
– I²S digital audio interface
– 2 wire serial control interface
– watchdog via serial interface
– power good pin
– 128bit unique ID (OTP)
– 17 different interrupts
Package CTBGA64 [7.0x7.0x1.1mm] 0.8mm pitch

Application
Portable Digital Audio Player and Recorder
PDA, Smartphone

GENERAL DESCRIPTION

The AK4645 is a stereo CODEC with a built-in Microphone-Amplifier and Headphone-Amplifier. The AK4645 features analog mixing circuits and PLL that allows easy interfacing in mobile phone and portable A/V player designs. The AK4645 is available in a 32pin QFN, utilizing less board space than competitive offerings.

FEATURES
1. Recording Function
• 4 Stereo Input Selector
• Stereo Mic Input (Full-differential or Single-ended)
• Stereo Line Input
• MIC Amplifier (+32dB/+26dB/+20dB or 0dB)
• Digital ALC (Automatic Level Control)
(+36dB ∼ −54dB, 0.375dB Step, Mute)
• ADC Performance: S/(N+D): 83dB, DR, S/N: 86dB (MIC-Amp=+20dB)
S/(N+D): 88dB, DR, S/N: 95dB (MIC-Amp=0dB)
• Wind-noise Reduction Filter
• Stereo Separation Emphasis
• Programmable EQ
2. Playback Function
• Digital De-emphasis Filter (tc=50/15μs, fs=32kHz, 44.1kHz, 48kHz)
• Bass Boost
• Soft Mute
• Digital Volume (+12dB ∼ −115.0dB, 0.5dB Step, Mute)
• Digital ALC (Automatic Level Control)
(+36dB ∼ −54dB, 0.375dB Step, Mute)
• Stereo Separation Emphasis
• Programmable EQ
• Stereo Line Output
- Performance: S/(N+D): 88dB, S/N: 92dB
• Stereo Headphone-Amp
- S/(N+D): 70dB@7.5mW, S/N: 90dB
- Output Power: 70mW@16Ω (HVDD=5V), 62mW@16Ω (HVDD=3.3V)
- Pop Noise Free at Power ON/OFF
• Analog Mixing: 4 Stereo Input
3. Power Management
4. Master Clock:
(1) PLL Mode
• Frequencies:
- MCKI pin: 11.2896MHz, 12MHz, 12.288MHz, 13MHz, 13.5MHz, 19.2MHz, 24MHz, 26MHz, 27MHz
- LRCK pin: 1fs
- BICK pin: 32fs or 64fs
(2) External Clock Mode
• Frequencies: 256fs, 512fs or 1024fs (MCKI pin)
5. Output Master Clock Frequencies: 32fs/64fs/128fs/256fs
6. Sampling Rate:
• PLL Slave Mode (LRCK pin): 7.35kHz ∼ 48kHz
• PLL Slave Mode (BICK pin): 7.35kHz ∼ 48kHz
• PLL Slave Mode (MCKI pin):
8kHz, 11.025kHz, 12kHz, 16kHz, 22.05kHz, 24kHz, 32kHz, 44.1kHz, 48kHz
• PLL Master Mode:
8kHz, 11.025kHz, 12kHz, 16kHz, 22.05kHz, 24kHz, 32kHz, 44.1kHz, 48kHz
• EXT Master/Slave Mode:
7.35kHz ∼ 48kHz (256fs), 7.35kHz ∼ 26kHz (512fs), 7.35kHz ∼ 13kHz (1024fs)
7. μP I/F: 3-wire serial, I2C Bus (Ver 1.0, 400kHz High Speed Mode)
8. Master/Slave mode
9. Audio Interface Format: MSB First, 2’s complement
• ADC : 16bit MSB justified, I2S, DSP Mode
• DAC : 16bit MSB justified, 16bit LSB justified, 16-24bit I2S, DSP Mode
10. Ta = −30 ∼ 85°C
11. Power Supply:
• AVDD, DVDD: 2.6 ∼ 3.6V (typ. 3.3V)
• HVDD: 2.6 ∼ 5.25V (typ. 3.3V/5.0V)
• TVDD (Digital I/O): 1.6 ∼ 3.6V (typ. 3.3V)
12. Package: 32pin QFN (5mm x 5mm,0.5mm pitch)
13. Register Compatible with AK4644


AKD4645