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Part Name(s) : AN17828A
Panasonic
Panasonic Corporation
Description : AN17828A Audio IC

Reference Datasheet PDF : AN17821A

Part Name(s) : EV-2
Cirrus-Logic
Cirrus Logic
Description : Digital Audio Networking Processor

Introduction
The EV-2 provides a means of evaluating the CM-1 or CM-2 CobraNet™ Modules and the Cirrus Logic CobraNet Silicon Series of devices. In addition to evaluating the CM-1 or CM-2 (hereafter collectively referred to as the CM except where differences between the CM-1 and CM-2 exist), the user may also use the EV-2 as a development platform and as an example interface for CMs, the Cobranet Silicon Series, and other CobraNet related projects. The EV-2 connects to the CM via the modules host interface. An 8051-type microcontroller interfaces to the CMs host port, and a simple Audio router on the EV-2 allows multiple Audio inputs and outputs to connect to the CMs serial Audio interface. The EV-2 software provides a simple interface for Audio routing on the EV-2, as well as development support.

Features*:
• Analog Audio I/O: Two channels of analog Audio input converted to high quality, 24-bit, 48 kHz or 96 kHz digital Audio. Two channels of 24-bit, 48 kHz or 96 kHz digital Audio converted to high quality, analog Audio output. Refer to Appendix B for Audio I/O specifications.
• Digital Audio I/O: One stream of AES3 input and one stream of AES3 output. An AES3 stream is two channels of digital Audio. The AES3 input stream is sample rate converted.
• 8051-type microcontroller: 64kB on-chip Flash Program Memory, 1kB internal SRAM, 32kB external SRAM and in-system programmability.
• Field programmability: The supplied EV-2 software provides a means to reprogram EV-2 microcontroller firmware for field upgrades or user development.
• RS232 Interfaces: Two RS232 interfaces, one direct to the CM and another to the microcontroller.
• Routing flexibility: Route from any Audio source to any Audio sink using the supplied EV-2 software. Route to and from the CM as well as within the EV-2.
• Sine wave generation: A sine wave test tone may be used as an alternate Audio source. Minimal frequency and gain control is provided.
• Hex switches: Four hex formatted switches may be used for network identification of the CobraNet module and/or user development.
• Command line interface: The 8051, via its RS232 serial interface, can be used to configure the CM using a command line interface.Cobranet HMI variables can be viewed and modified using this interface. Refer to Appendix E for a description of the Command line interface.
• LED display: Three LED indicators are provided and may be used for user development.
• Power supply: Uses standard computer ATX power supply (not included).

ST-Microelectronics
STMicroelectronics
Description : SIX-CHANNEL DOLBY AC3/MPEG2 Audio DECODER

INTRODUCTION
The ST18-AU1 is a single-chip multi-function Audio processor for Dolby AC-3, MPEG-1/MPEG-2 Layer-I/II Audio encoded bitstreams, and DVD Linear PCM. It is capable of decoding up to 5.1 channels of input Dolby AC-3 or MPEG-2 multi-channel encoded Audio, and down mixing to 2 channels of PCM output Audio. Maximum input data rates for Dolby AC-3 bitstream and MPEG-2 Audio bitstream are 448 Kbits/s and 912 Kbits/s respectively. It also supports up to 8 channel linear PCM input with by-pass, down-sampling, and down-mixing function.

FEATURES
■ Single chip multi-function Audio decoder able to decompress DOLBY AC-3, MPEG-1 and MPEG-2 Audio streams.
■ Maximum 5.1 channel DOLBY AC-3 decoding to 2 channel mixed down output with DOLBY surround compatible or karaoke capable option.
■ Variable bit rate MPEG-1 layer II Audio decoding, and MPEG-2 multi-channel Audio decoding for karaoke capable application.
■ Input data rates
    ■ up to 448 Kbits/s for AC-3 decoder
    ■ up to 912 Kbits/s for MPEG-1 or MPEG-2 Audio decoder
■ Supports up to 8 channel DVD linear PCM input at max rate of 6.144 Mbits/s down-mixing and/ or sub-sampling to 2 to 6 channels.
■ Accepts MPEG-1 or DVD/MPEG-2 PES input packets.
■ Programmable D950 core
■ System time clock provides A/V synchronization and PTS packet extraction.
■ Automatic error concealment on CRC or synchronization error.
■ 6 channel PCM Audio output at 16/18/20/24 bit. Sampling rate of 32/44.1/48/96 kHz.
■ Two on-chip PLLs providing full circuit operation with only one external 27 MHz clock.
■ I2C interface for host control
■ Multi-format i2S serial data input port and decoded Audio PCM output port.
■ IEC-958 (S/PDIF) formatter and transmitter for DOLBY AC-3, MPEG Audio bit stream, or Audio PCM.
■ Dedicated hardware for emulation and test, IEEE 1149.1 (JTAG).
■ 3.3V power supply, I/O’s 5V tolerant, 0.35µM HCMOS6 technology.
■ 160 pin PQFP package

APPLICATIONS
■ Digital video disc (DVD) player
■ Digital TV (DBS/DVB) receiver
■ PC multimedia
■ Consumer digital Audio

Part Name(s) : SPCA751A-P101
ETC
Unspecified
Description : single chip signal processor optimized for MPEG Audio decoding and voice recording

[Sunplus Technology Co., Ltd.]

Description
The SPCA751A is a single chip signal processor optimized for MPEG Audio decoding and voice recording. It is developed to achieve a better performance/cost ratio for MPEG Audio players.
The SPCA751A is especially designed for standalone Audio players, the system controller can easily carry out the MPEG Audio decoding process by the use of a general serial IO/control interface for MPEG bit stream in/out and playback control.
Decoded Audio PCM data are output to external DAC through a programmable normal/I2S DAC interface, such that most of common Audio DACs can be cooperated with SPCA751A to meet different customers requirements.

Features
• Single chip MPEG Audio decoder
   - Conforming to MPEG1/MPEG2 Audio layer 2/3
   - Extension to MPEG lower sampling rates
• Digital sound control
   - Digital volume control
   - Stereo/Mono channel select
   - Digital sound equalizer
• Internal auto-generate Audio clock
   - Sampling frequency from 8 kHz up to 48 kHz
• Programmable Audio DAC interface
   - Support both normal and I2S Audio DAC formats
   - Audio clock polarity programmable
   - Internal auto-generated oversampling clock for DAC
   - Accept external Audio clock for sampling rate control
• Serial data IO and control interface
   - Easy for the host processor to command
• Low power dissipation
• PLL embedded
   - Require only 16.934MHz crystal, resistors, and capacitors to supply the system clock
• Built-in Digital Recording option
   - Embedded 10-bit 8 kHz Audio ADC
   - SACM_S480 recording with 4.8 kbit/sec
   - SACM_S3200 recording with 32 kbit/sec
• Device Parameter
   - Supply voltage : 3.0 ~ 3.6 volts
   - IO interface : 5 volts tolerance, TTL compatible
   - Package : 44-pin LQFP
   - Power consumption: less than 150 mW @ 3.6 volts

Description : Video ICs 5:2 Crosspoint TRX Audio I/O OSD 16bTTL

GENERAL DESCRIPTION
The ADV7625 is a high performance, five-input, dual-output, High-Definition Multimedia Interface (HDMI®) transceiver with crosspoint and splitter capabilities. The ADV7625 supports 3 GHz video and features two independent HDMI receivers, two independent HDMI transmitters, two Audio output ports, two Audio input ports, and a pixel port input. The ADV7625 supports all HDCP repeater functions through fully tested Analog Devices, Inc., repeater software libraries and drivers.

FEATURES
   5-input, 2-output crosspoint HDMI transceiver
   HDMI support
      3 GHz video support (up to 4k × 2k)
      Audio return channel (ARC)
      3D TV support
      Content type bits
      CEC 1.4-compatible
      Extended colorimetry
   Character- and icon-based on-screen display (OSD)
      3D OSD overlay on all mandatory 3D formats
      Support for OSD overlay on 3 GHz video formats
   High-bandwidth Digital Content Protection (HDCP 1.4)
   HDCP repeater support: up to 127 KSVs supported
   300 MHz maximum TMDS clock frequency (up to 4k × 2k)
   48-/36-/30-bit Deep Color input modes supported
   Ultralow jitter digital PLL (100% deskew)
   TTL pixel port input
      Allows digital video input to facilitate analog video support
      Interlaced-to-progressive converter
   2 independent HDMI receivers for 5 input ports
      3 GHz support on all inputs
      Adaptive equalizer for cable lengths up to 30 meters
      Flexible internal EDID RAM supports dual EDIDs
      Replication of either dual EDID on any input port
      5 V detect inputs
      Hot Plug assert control outputs
   2 independent HDMI transmitters
      3 GHz support on all outputs
      EDID data extraction
      Hot Plug detect (HPD) inputs
      Audio return channel (ARC) receiver per transmitter
      3 GHz color space converter (CSC) per transmitter
   Audio
      HDMI-compatible Audio interface
      2 independent 8-channel Audio extraction ports
      2 independent 8-channel Audio insertion ports
      S/PDIF (IEC 60958-compatible) digital Audio input/output
      Super Audio CD® (SACD) with DSD input/output interface
      High bit rate (HBR) Audio
      Dolby® TrueHD
      DTS-HD Master Audio
      Full Audio input and output support
   General
      Interrupt controller
      Standard identification (STDI) circuit
      Software libraries, driver, and application available

APPLICATIONS
   AVR
   Soundbar with HDMI repeater support
   Matrix switch
   Other repeater applications

Part Name(s) : ALC250
ETC2
Unspecified
Description : TWO CHANNEL AC’97 2.3 Audio CODEC with EQUALIZER

[Realtek Semiconductor Corp.]

General Description
The ALC250 is a 20-bit DAC and 18-bit ADC full duplex AC97 2.3 compatible stereo Audio CODEC designed for PC multimedia systems, including host/soft Audio and AMR/CNR based designs. The ALC250 incorporates proprietary converter technology to achieve a high SNR, greater than 100 dB, sensing logics for device reporting and Universal Audio Jack® to improve user interface. The ALC250 AC97 CODEC supports multiple CODEC extensions with independent variable sampling rates and built-in 3D effects. The ALC250 CODEC provides two pairs of stereo outputs with independent volume controls, a mono output, and multiple stereo and mono inputs, along with flexible mixing, gain and mute functions to provide a complete integrated Audio solution for PCs. The circuitry of the ALC250 CODEC operates from a +3.3V digital power and +5V analog power supply with EAPD (External Amplifier Power Down) control for use in notebook and PC applications. The ALC250 integrates a 50mW/20Ω headset Audio amplifier into the CODEC, saving BOM costs. The ALC250 also supports the SPDIF out function, which is compliant to AC97 2.3, which can offer easy connection of PCs to consumer electronic products, such as AC3 decoder/speaker and mini disk devices. The ALC250 CODEC supports host/soft Audio from Intel ICHx chipsets as well as Audio controller based VIA/SIS/ALI/AMD/nVIDIA/ ATI chipset.

Features
● Built- in 7 Bands of Digital Hardware Equalizer for Optimizing Speaker Response
● Single chip with high S/N ratio (>100 dB)
● Meets performance requirements for Audio on PC99/2001 systems
● Meets Microsoft WHQL/WLP 2.0 Audio requirements
● 20-bit DAC and 18-bit ADC resolution
● Compliant with AC’97 2.3 specifications
   -LINE/HP-OUT, MIC-IN and LINE-IN sensing
   -14.318MHz-Æ24.576MHz PLL saves crystal
   -12.288MHz BITCLK input can be consumed
   -Integrated PCBEEP generator to save buzzer
   -Interrupt capability
   -Page registers and Analog Plug&Play
● Support of S/PDIF out is fully compliant with AC’97 rev2.3 specifications
● Three analog line-level stereo inputs with 5-bit volume control: LINE_IN, CD, AUX
● High quality differential CD input
● Two analog line-level mono input: PCBEEP, PHONE-IN
● Supports double sampling rate (96KHz) of DVD Audio playback
● Two software selectable MIC inputs
● +6/12/20/30dB boost preamplifier for MIC input
● Stereo output with 6-bit volume control
● Mono output with 5-bit volume control
● Headphone output with 50mW/20Ω amplifier
● 3D Stereo Enhancement
● Multiple CODEC extension capability
● External Amplifier Power Down (EAPD) capability
● Power management and enhanced power saving features
● Stereo MIC record for AEC/BF application
● DC Voltage volume control
● Auxiliary power to support Power Off CD
● Adjustable VREFOUT control
● EQ operation can be controlled by 2 pins of serial bus
● 2 Universal Audio Jack (UAJ)® for front panel
● Support 32K/44.1K/48K/96KHz of S/PDIF output
● Support 32K/44.1K/48KHz of S/PDIF input
● Power support: Digital: 3.3V; Analog: 3.3V/5V
● Standard 48-Pin LQFP Package
● EAX™ 1.0&2.0 compatible
● Direct Sound 3D™ compatible
● A3D™ compatible
● I3DL2 compatible
● HRTF 3D Positional Audio
● Sensaura™ 3D Enhancement (optional)
● 10 Bands of Software Equalizer
● Voice Cancellation and Key Shifting in Kara OK mode
● AVRack® Media Player
● Configuration Panel to improve Experience of User

Description : 24-Bit, 192 kHz Stereo Audio CODEC

General Description
The CS4270 is a high-performance, integrated Audio CODEC. The CS4270 performs stereo analog-to-digital (A/D) and digital-to-analog (D/A) conversion of up to 24-bit serial values at sample rates up to 216 kHz. Standard 50/15 µs de-emphasis is available for sampling rates of 44.1 kHz for compatibility with digital Audio programs mastered using the 50/15 µs pre-emphasis technique.
Integrated level translators allow easy interfacing be tween the CS4270 and other devices operating over a wide range of logic levels.
Independently addressable high-pass filters are available for the right and left channel of the A/D. This allows the A/D to be used in a wide variety of applications where one Audio channel and one DC measurement channel is desired.

D/A Features
♦ High Performance
   – 105 dB Dynamic Range
   – -95 dB THD+N
♦ Selectable Serial Audio Interface Formats
   – Left-Justified up to 24-bit
   – I²S up to 24-bit
   – Right-Justified 16-, and 24-Bit
♦ Control Output for External Muting
♦ On-Chip Digital De-Emphasis
♦ Popguard® Technology
♦ Multi-bit ∆Σ Conversion
♦ Digital Volume Control
♦ Single-Ended Output

A/D Features
♦ High Performance
   – 105 dB Dynamic Range
   – -95 dB THD+N
♦ Multi-bit ∆Σ Conversion
♦ High-Pass Filter to Remove DC Offsets
♦ Selectable Serial Audio Interface Formats
   – Left-Justified up to 24-bit
   – I²S up to 24-bit
♦ Single-Ended Input

System Features
♦ Direct Interface with Logic Levels 1.8 V to 5 V
♦ Internal Digital Loopback
♦ Stand-Alone or Control Port Functionality
♦ Single-Ended Analog Architecture
♦ Supports all Audio Sample Rates from 4 kHz to 216 kHz
♦ 3.3 V or 5 V Core Supply

Stand-Alone Mode Feature Set
♦ System Features
   – Serial Audio Port Master or Slave Operation
   – Single-, Double-, or Quad-Speed Operation
♦ D/A Features
   – Auto-Mute on Static Samples
   – 44.1 kHz 50/15 µs De-emphasis Available
   – Selectable Serial Audio Interface Formats
      • Left-Justified up to 24-bit
      • I²S up to 24-bit
♦ A/D Features
   – High-Pass Filter
   – Selectable Serial Audio Interface Formats
      • Left-Justified up to 24-bit
      • I²S up to 24-bit

Software Mode Feature Set
♦ System Features
   – Serial Audio Port Master or Slave Operation
   – Internal Digital Loopback Available
♦ D/A Features
   – Selectable Auto-mute
   – 44.1-kHz De-emphasis Filters
   – Configurable Muting Controls
   – Volume Control
   – Selectable Serial Audio Interface Formats
      • Left-Justified up to 24-bit
      • I²S up to 24-bit
      • Right-Justified 16, and 24-bit
♦ A/D Features
   – Selectable High-Pass Filter or DC Offset Calibration
   – Selectable Serial Audio Interface Formats
      • Left-Justified up to 24-bit
      • I²S up to 24-bit

 

Part Name(s) : GS9023A GS9023ACFY
Gennum
Gennum -> Semtech
Description : GENLINX™ IIGS9023A Embedded Audio CODEC

BRIEF DESCRIPTION
The GS9023A is a highly integrated, single chip solution for the multiplexing/demultiplexing of digital Audio channels into and out of digital video signals. The GS9023A supports the multiplexing/demultiplexing of 20 or 24-bit synchronous Audio data with a 48kHz sample rate.
Audio signals with different sample rates may be sample rate converted to 48kHz before and after the GS9023A using Audio sample rate converters.

KEY FEATURES
• single chip embedded Audio solution
• operates as an embedded Audio multiplexer or demultiplexer
• full support for 48kHz synchronous 20/24bit Audio
• 4 channels of Audio per GS9023A
• cascadable architecture supports additional Audio channels
• multiplexes and demultiplexes arbitrary ANC data packets
• support for 143, 177, 270, 360 and 540 Mb/s video standards
• full processing of Audio parity, channel status and user data
• multiplexes and demultiplexes Audio control packets
• EDH generation and insertion when in Multiplex Mode
• 3.3V core with 3.3V or 5VI/O (requires 5V supply)
• complies with SMPTE 272M A, B, and C

APPLICATIONS
SDI Embedded Audio

Part Name(s) : GS9023B GS9023BCVE3
Gennum
Gennum -> Semtech
Description : GENLINX® II GS9023B Embedded Audio CODEC

Brief Description
The GS9023B is a highly integrated, single chip solution for the multiplexing/demultiplexing of digital Audio channels into and out of digital video signals. The GS9023B supports the multiplexing/demultiplexing of 20 or 24-bit synchronous Audio data with a 48kHz sample rate. Audio signals with different sample rates may be sample rate converted to 48kHz before and after the GS9023B using Audio sample rate converters.

Key Features
• single chip embedded Audio solution
• operates as an embedded Audio multiplexer or demultiplexer
• full support for 48kHz synchronous 20/24 bit Audio
• 4 channels of Audio per GS9023B
• cascadable architecture supports additional Audio channels
• multiplexes and demultiplexes arbitrary ANC data packets
• support for 143, 177, 270, 360 and 540 Mb/s video standards
• full processing of Audio parity, channel status and user data
• multiplexes and demultiplexes Audio control packets
• EDH generation and insertion when in Multiplex Mode
• 3.3V core with 3.3V or 5V I/O (requires 5V supply)
• complies with SMPTE 272M A, B, and C

Applications
SDI Embedded Audio

Part Name(s) : GS9023 GS9023-CFY
Gennum
Gennum -> Semtech
Description : GENLINX™II GS9023 Embedded Audio CODEC

DESCRIPTION
The GS9023 is a highly integrated, single chip solution for the multiplexing/demultiplexing of digital Audio channels into and out of digital video signals. The GS9023 supports the multiplexing/demultiplexing of 20 or 24 bit synchronous Audio data with a 48kHz sample rate. Audio signals with different sample rates may be sample rate converted to 48kHz before and after the GS9023 using Audio sample rate converters.

FEATURES
• single chip embedded Audio solution
• operates as an embedded Audio multiplexer or demultiplexer
• full support for 48kHz synchronous 20/24 bit Audio
• 4 channels of Audio per GS9023
• cascadable architecture supports additional Audio channels
• multiplexes and demultiplexes arbitrary ANC data packets
• support for 143, 177, 270, 360 and 540 Mb/s video standards
• full processing of Audio parity, channel status and user data
• multiplexes and demultiplexes Audio control packets
• EDH generation and insertion when in Multiplex Mode
• 3.3V core with 3.3V or 5V I/O (requires 5V supply)
• complies with SMPTE 272M A, B, and C

APPLICATIONS
SDI Embedded Audio

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