DESCRIPTION
The WM8720 is a high performance stereo DAC designed for audio applications such as CD, DVD, home theatre systems, set top boxes and digital TV. The WM8720 supports data input word lengths from 16 to 24-bits and sampling rates up to 96kHz. The WM8720 consists of a serial interface port, digital interpolation filter, multi-bit sigma delta modulator and stereo DAC in a small 20-pin SSOP package. The WM8720 also includes a digitally controllable mute and attenuator function on each channel.
The WM8720 supports a variety of connection schemes for audio DAC control. The SPI-compatible serial control port provides access to a wide range of features including onchip mute, attenuation and phase reversal. A hardware controllable interface is also available.
The programmable data input port supports a variety of glueless interfaces to popular DSPs, audio decoders and S/PDIF and AES/EBU receivers.

FEATURES
*Performance
- 102dB SNR (‘A’ weighted @48kHz),
- THD: -95dB @ 0dB FS
*5V or 3.3V supply operation
*Sampling frequency: 8kHz to 96kHz
*Input data word: 16 to 24-bit
*Hardware or SPI compatible serial port control modes
- Hardware mode: system clock, reset, mute, de-emphasis
- Serial control mode: mute, de-emphasis, digital attenuation (256 steps), zero mute, phase reversal, power down

APPLICATIONS
*CD, DVD audio
*Home theatre systems
*Set top boxes
*Digital TV

WM8720EDS, WM8720EDS/R, WM8720SEDS, WM8720SEDS/R

Trackback :: http://datasheetblog.com/trackback/2866

댓글을 달아 주세요 Comment

Description
The MC13028A is a third generation C–QUAM stereo decoder targeted for use in low voltage, low cost AM/FM E.T.R. radio applications. Advanced features include a signal quality detector that analyzes signal strength, signal to noise ratio, and stereo pilot tone before switching to the stereo mode. A “blend function” much like FM stereo has been added to improve the transition from mono to stereo. The audio output level is adjustable to allow easy interface with a variety of AM/FM tuner chips. The external components have been minimized to keep the total system cost low.

Features
*Adjustable Audio Output Level
*Stereo Blend Function
*Stereo Threshold Adjustment
*Operation from 2.2 V to 12 V Supply
*Precision Pilot Tone Detector
*Forced Mono Function
*Single Pinout VCO
*IF Amplifier with IF AGC Circuit
*VCO Shutdown Mode at Weak Signal Condition

MC13028AD, MC13028AP

Trackback :: http://datasheetblog.com/trackback/2737

댓글을 달아 주세요 Comment

DESCRIPTION
The TPA6141A2 (also known as TPA6141) is a Class-G DirectPath™ stereo headphone amplifier with selectable gain. Class-G technology maximizes battery life by adjusting the voltage supplies of the headphone amplifier based on the audio signal level. At low level audio signals, the internal supply voltage is reduced to minimize power dissipation. DirectPathTM technology eliminates external DC-blocking capacitors.
The device operates from a 2.5 V to 5.5 V supply voltage. Class-G operation keeps total supply current below 5.0 mA while delivering 500 mW per channel into 32 Ω. Shutdown mode reduces the supply current to less than 3 mA and is activated through the EN pin.
The device has built-in pop suppression circuitry to completely eliminate disturbing pop noise during turn-on and turn-off. The amplifier outputs have short-circuit and thermal-overload protection along with ±8 kV HBM ESD protection, simplifying end equipment compliance to the IEC 61000-4-2 ESD standard.
The TPA6141A2 (TPA6141) is available in a 0,4 mm pitch, 16-bump 1,6 mm × 1,6 mm WCSP (YFF) package.

FEATURES
*TI Class-G Technology Significantly Prolongs Battery Life and Music Playback Time
–0.6 mA / Ch Quiescent Current
–50% to 80% Lower Quiescent Current Than Ground-Referenced Class-AB Headphone Amplifiers
*DirectPathTM Technology Eliminates Large Output DC-Blocking Capacitors
–Outputs Biased at 0 V
–Improves Low Frequency Audio Fidelity
*Active Click and Pop Suppression
*Fully Differential Inputs Reduce System Noise
–Also Configurable as Single-Ended Inputs
*SGND Pin Eliminates Ground Loop Noise
*Wide Power Supply Range: 2.5 V to 5.5 V
*100 dB Power Supply Noise Rejection
*Gain Settings: 0 dB and 6dB
*Short-Circuit Current Limiter
*Thermal-Overload Protection
*±8 kV HBM ESD Protected Outputs
*0,4 mm Pitch, 1,6 mm × 1,6 mm WCSP Package

APPLICATIONS
*Cellular Phones / Music Phones
*Portable Media / MP3 Players
*Portable CD / DVD Players

TPA6141A2YFFR, TPA6141A2YFFT

Trackback :: http://datasheetblog.com/trackback/2523

댓글을 달아 주세요 Comment

DESCRIPTION
The PCM1796 is a monolithic CMOS integrated circuit that includes stereo digital-to-analog converters and support circuitry in a small 28-lead SSOP package. The data converters use TI’s advanced segment DAC architecture to achieve excellent dynamic performance and improved tolerance to clock jitter. The PCM1796 provides balanced current outputs, allowing the user to optimize analog performance externally. The PCM1796 accepts PCM and DSD audio data formats, providing easy interfacing to audio DSP and decoder chips. The PCM1796 also
interfaces with external digital filter devices (DF1704, DF1706, PMD200). Sampling rates up to 200 kHz are supported. A full set of user-programmable functions is accessible through an SPI or I2C serial control port, which supports register write and readback functions. The PCM1796 also supports the time division multiplexed command and audio (TDMCA) data format.

FEATURES
*24-Bit Resolution
*Analog Performance:
- Dynamic Range: 123 dB
- THD+N: 0.0005%
*Differential Current Output: 4 mA p-p
*8× Oversampling Digital Filter:
- Stop-Band Attenuation: –98 dB
- Pass-Band Ripple: ±0.0002 dB
*Sampling Frequency: 10 kHz to 200 kHz
*System Clock: 128, 192, 256, 384, 512, or 768 fS With Autodetect
*Accepts 16-, 20-, and 24-Bit Audio Data
*PCM Data Formats: Standard, I2S, and Left-Justified
*DSD Format Interface Available
*Interface Available for Optional External Digital Filter or DSP
*TDMCA or Serial Port (SPI/I2C)
*User-Programmable Mode Controls:
- Digital Attenuation: 0 dB to –120 dB, 0.5 dB/Step
- Digital De-Emphasis
- Digital Filter Rolloff: Sharp or Slow
- Soft Mute
- Zero Flag for Each Output
*Compatible With PCM1792 (Pins and Mode Controls)
*Dual Supply Operation:
- 5-V Analog, 3.3-V Digital
*5-V Tolerant Digital Inputs
*Small 28-Lead SSOP Package

APPLICATIONS
*A/V Receivers
*SACD Players
*DVD Players
*HDTV Receivers
*Car Audio Systems
*Digital Multitrack Recorders
*Other Applications Requiring 24-Bit Audio

PCM1796DB, PCM1796DBR

Trackback :: http://datasheetblog.com/trackback/2019

댓글을 달아 주세요 Comment

DESCRIPTION
 The UTC PA0202 is a monolithic integrated circuit that stereo bridged audio power amplifiers capable of producing 2 W into 3Ω with a 5V supply voltage or 800mW into 3Ω with a 3.3V supply
voltage.
The UTC PA0202 simplifies design and frees up board space for other features.
Both of the depop circuitry and the thermal shutdown protection circuitry are integrated in UTC PA0202, that reduce clicks and pops noise during power up or shutdown mode operation.
A MUX control terminal (HP/LINE) allows selection between the two sets of stereo input signals.
To simplify the audio system design, UTC PA0202 combines a stereo bridge-tied loads (BTL)mode for speaker drive and a stereo single-end (SE)mode for headphone drive into a single chip, where both modes are easily switched by the SE/BTL input control pin signal.

FEATURES
* Improves depop circuitry to eliminate turn-on and turn-off transients in output
* Output power:
- 2W(typ.)@5V into 3Ω with 0.2% THD+N max (1kHz)
- 800mW(typ.)@3.3V into 3Ω with 0.2% THD+N max (1kHz)
* Fully specified for use with 3-Ω Loads
* Stereo switchable bridged/single-ended power amplifiers
* Input MUX select terminal
* Thermal-shutdown protection
* Shutdown mode available

PA0202-N24-R
PA0202-N24-T
PA0202L-N24-R
PA0202L-N24-T

Trackback :: http://datasheetblog.com/trackback/1654

댓글을 달아 주세요 Comment

GENERAL DESCRIPTION
This product is SD Digital-To-Analog Converter for digital audio System (CDP). The product contains Serial-to- Parallel Converter and Compensation Filter, Digital Volume Attenuator by the MICOM Interface, De-Emphasis Filter, FIR filter, Sinc Filter, digital sigma-delta modulator, analog postfilter, AIF (Anti-Image-Filter). The normal input and output channels provides 90dB SNR (Signal to Noise Ratio) over in band (20kHz).
The product employs the 1bit 4th-order sigma-delta architecture with 16bit resolution, over sampling of 64X. And analog postfilter with low clock sensitivity and linear phase, filters the shaping-nosie and outputs analog voltage with high resolution. An on-chip reference voltage is included to allow single supply operations.

FEATURES
* 16bit SD Digital-To-Analog Converter
* On-Chip Analog Postfilter
* Filtered Line-Level Outputs, Linear Phase Filtering
* On-Chip Voltage Reference
* 90dB SNR
* Sampling Rate 44.1kHz
* Input Rate 1Fs or 2Fs by Normal Mode/Double Mode Selection
* Zero Input Detection Mute
* On-Chip Compensation Filter
* Input Volume Attenuator by the MICOM Interface
* On-Chip De-Emphasis Filter
* On-Chip 4 times oversampling Digital Filter
* Low Clock Jitter Sensitivity
* Single 3.3V~2.5V Power Supply

APPLICATIONS
CD Player, Portable CD Player, CD-ROM, Video-CD, Mini-Disk, DVD etc

TAG DAC, Stereo

Trackback :: http://datasheetblog.com/trackback/1435

댓글을 달아 주세요 Comment

FEATURES
* DIGITALLY-CONTROLLED ANALOG VOLUME CONTROL:
Two Independent Audio Channels
Serial Control Interface
Zero Crossing Detection
Mute Function
* WIDE GAIN AND ATTENUATION RANGE:
+31.5dB to -95.5dB with 0.5dB Steps
* LOW NOISE AND DISTORTION:
120dB Dynamic Range
0.0004% THD+N at 1kHz
* LOW INTERCHANNEL CROSSTALK:
-126dBFS
* NOISE-FREE LEVEL TRANSITIONS
* POWER SUPPLIES: 15V Analog, +5V Digital
* AVAILABLE IN DIP-16 AND SOL-16 PACKAGES
* PIN AND SOFTWARE COMPATIBLE WITH THE PGA2311 AND CIRRUS LOGIC CS3310

APPLICATIONS
* AUDIO AMPLIFIERS
* MIXING CONSOLES
* MULTI-TRACK RECORDERS
* BROADCAST STUDIO EQUIPMENT
* MUSICAL INSTRUMENTS
* EFFECTS PROCESSORS
* A/V RECEIVERS
* CAR AUDIO SYSTEMS

DESCRIPTION
The PGA2310 is a high-performance, stereo audio volume control designed for professional and high-end consumer audio systems. The ability to operate from ±15V analog power supplies enables the PGA2310 to process input signals with large voltage swings, thereby preserving the dynamic range available in the overall signal path. Using high performance operational amplifier stages internal to the PGA2310 yields low noise and distortion, while providing the capability to drive 600Ω loads directly without buffering. The three-wire serial control interface
allows for connection to a wide variety of host controllers, in addition to support for daisy-chaining of multiple PGA2310 devices.

PGA2310PA
PGA2310PAG4
PGA2310UA

Trackback :: http://datasheetblog.com/trackback/1407

댓글을 달아 주세요 Comment

DESCRIPTION
The LX1705 is a fully integrated stereo class-D CMOS audio amplifier. optimized for highly efficient operation and minimum system cost.
The stereo BTL (Bridge-tied-load) configuration uses 3-level PWM modulation. This allows eliminating the LC filter to reduce the system cost and simplify the system design. The LX1705 outputs 8W into each of two channels with better than 90% efficiency. The entire signal path from input to output is differential to reject any sources of common-mode noise or distortion.
The part features on–board H-bridge output stages with low RDSON. External bootstrap capacitors are all that is required to provide the gate drive to the all-NFET output stage since on-board bootstrap diodes are provided.
The LX1705 also features Mute and Standby modes, POP-free turn-on and turn-off, under-voltage lockout for both input supplies, and multi-level overtemperature protection.
The LX1705 is offered in a small thermally efficient footprint, low profile surface mountable 32-pin Micro Lead Quad Package (MLPQ) in 5mm x 5 mm.

KEY FEATURES
* Filter Free Operation
* 6W +6W Output Power @ 8Ω load: THD+N < 1%
* High Efficiency > 90%
* Full Audio Bandwidth: 20Hz to 20kHz
* Low Distortion < 0.25% @ 30% Max Power, 1kHz
* High Signal-to-Noise Ratio: 90dB
* Wide Supply Voltage Range 5.0V ~ 15V
* 5mA Per Channel Typical Quiescent Current
* Turn ON/OFF POP Free
* Standby / Mute Feature
* Built-in Under Voltage Lockout
* Thermal Protection

APPLICATIONS
* LCD TV
* Car Navigation
* MP3 Docking Stations
* Portable Sound System

LX1705ILQ

Trackback :: http://datasheetblog.com/trackback/1053

댓글을 달아 주세요 Comment

Features
* Two full-bridge channels, 100 W each
* 106 dB Dynamic Range - both channels
* 0.015% THD+N at 1 W
* Power Supply Rejection (PSR) feedback allowing amplifier to operate from low cost linear unregulated power supplies
* Spread Spectrum Modulation - Reduces Modulation Energy
* Passes CISPR and FCC requirements for radiated and power line conducted emissions
* Independent peak signal limiting per channel
* Thermal and over-current protection
* > 85% amplifier efficiency
* Works with GUI to configure the board
* Demonstrates recommended layout and grounding arrangements

Description
The CRD44600-PH-FB PWM Amplifier demonstrates the CS44600, Cirrus’ multi-channel pure digital PWM controller. This reference design implements a twochannel amplifier which delivers 100 W per full-bridge channel into 8 Ω loads using a single +50 V supply (at 1% THD+N). A 155 W unregulated linear power supply is used to power the CRD44600-PH-FB.
As shown below, the CS44600 IC takes two stereo digital audio PCM inputs and converts them to PWM outputs. This 64-pin LQFP PWM controller provides an integrated sample rate converter for 32 kHz-192 kHz input sample rate support, volume up/down, speaker load compensation, peak limiting to prevent amplifier clipping, power supply ripple compensation, and AM frequency interference elimination.
This reference design uses the the Philips TDA8939, an integrated power stage back end for digital amplifiers (two TDA8939 parts configured as full-bridges are used for this two-channel design). Current limiting and thermal protection are provided by the TDA8939.
The inductor/capacitor 2nd order low pass filter (LPF) removes high frequency components from the output signal effectively converting it from digital to analog.


CRD44600-PH-FB

Trackback :: http://datasheetblog.com/trackback/1023

댓글을 달아 주세요 Comment

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
TAG CODEC, Stereo

Trackback :: http://datasheetblog.com/trackback/944

댓글을 달아 주세요 Comment