ADAU1x61：音频编解码器

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罗德 USB 音频接口 AI-1 用户手册说明书

USER GUIDEOVERVIEWThe AI-1USB Audio Interface adds studio-quality Inputand Output capabilities to your PC or Mac, turning your recording software into a full recording setup.With its high quality Neutrik XLR-1/4” Combo Jack combined input and discrete Class-A preamp, the AI-1 allows you to connect a microphone or instrument orline level audio signal to your Mac/Windows computerfor recording. The AI-1provides the necessary A/D conversion at up to 24 bit/96kHz.The AI-1also allows playback and monitoring of audio either direct from the microphone or via playback fromthe recording software. This is done via a high-quality discrete headphone amplifier and outputs via either the ¼’’ headphone output on the front of the unit or the balanced ¼’’ outputs on the rear of the unit. Features:• Ultra low-noise Class-A discrete preamplifier• Sampling rate of up to 96kHz/24-Bit• Premium discrete headphone amplifier• XLR-1/4” Combo Jack Instrument / line level input• 2 x Balanced 1/4” speaker outputs• Rugged body with Satin Black Finish• USB-C/3 connectivitySYSTEM REQUIREMENTSAppleMacOS 10.10 or later with USB 2.0 or 3.0 USB portWindowsWindows 7 OS or later with USB 2.0 or 3.0 USB port The AI-1 is USB bus powered so there is no need for an external power supply.Some laptops may limit the USB output power when running on battery. This could mean the AI-1 does not receive enough power to run optimally. If this is the case, connect your laptop to the power supply when using the AI-1.TIPS FOR BEST RESULTSPhantom power needs to be activated for most condenser microphones, but is not necessary for dynamic microphones or instruments. It is recommended that you deselect phantom power before connecting devices that do not need it.When setting levels, adjust the input gain so the signal level LED flashes yellow during the loudest parts of the performance. This will give the best performance without distortion or noise.The high-power headphone amplifier in the AI-1 is capable of producing extremely high volume levels in some headphones. Start with the volume turned right down, and turn up until the level is comfortable.Please turn headphone level down before unplugging headphones to mix via speakers.CONNECTING YOUR AI-1If your computer doesn’t automatically switch its default input/output to the RODE AI-1 when connected, you can set this up with the following steps:Mac OS Go to System Preferences > SoundConfirm that ‘RODE AI-1’ is selected as the input and output device.Windows Go to Start > Control Panel > Hardware and Sound > Sound > Manage Audio Devices (alternatively right click the speaker icon in the bottom right of the taskbar and select Playback)Confirm that ‘RODE AI-1’ is selected as the default device in both the Playback and Recording tabs.The AI-1 is a class compliant device and therefore does not need any drivers installed.For future firmware updates, please visit /ai1HARDWARE FEATURES1. XLR-1/4” Combo Input Balanced input via 3-pin XLR-1/4” Neutrik combo socket. Microphones, instruments(guitars) and line level instruments (synthesizers) can beconnected via this input.2. Signal LED Indicates signal input level. No LED indicatesbelow -60dB. Green LED indicates an input level of atleast -40dB, Yellow indicates an input level above -12dB and Red indicates an input level of above -3dB.3. Phantom Power LED Indicates when phantom power48V is applied to XLR-1/4” Combo Jack input. Gain level knob must be pushed to turn on/off.4. Gain Level Knob Gain knob for adjusting input gain.Press knob to turn on/off phantom power.5. Playback/Monitor Level Knob Gain knob for adjustingheadphone / speaker volume. Press knob to turn Direct Monitoring on/off.6. Direct Monitor LED Indicates when Direct Monitoringis on/off. When Direct Monitoring is ON, audio from the microphone will be routed directly to the headphones.When Direct Monitoring is OFF, you will hear only audio from the computer.7. USB LED Indicates when the AI-1 is connected via theUSB cable.8. Headphone Jack Output Connector for headphones,¼’’ TRS jack required.Front1. USB-C Port USB-C type connector. Connect to yourlaptop or computer with the USB cable supplied.2. Speaker outputs 2 x ¼” impedance balanced outputs. NoteSpeaker output is muted when headphones are connected. Unplug headphones to enable monitoring via the speaker output. To avoid feedback loops when monitoring on speakers, please disable any connected microphone. When ready to record with a mic, either turn down speakers or switch to headphone monitoring.BackSPECIFICATIONSComputer Connectivity USBForm Factor Desktop Simultaneous I/O 1 x 2Number of Preamps 1Phantom Power YesBit Depth24-bitSample Rates44.1 kHz48 kHz88.2 kHz96 kHzAnalog Inputs 1 x Neutrik XLR-1/4” combo Analog Outputs 2 x 1/4” (impedance balanced)1 x 1/4” (headphones) Direct Monitor YesUSB 1 x USB Type CBus Powered YesOS Requirements Mac OS 10.10 or laterWindows 7 or laterPower USB bus-poweredDepth100mm total (with knobs)chassis: 88mmWidth124mmHeight38mmWeight1lb 3.7oz560gDynamic Range 104dBAEquivalent Input Noise @ Maximum Gain (Source Impedance 150 ohms, 20Hz-20kHz, A-weighted)–128dBAMax Input +6dBuTHD <0.008% (-30dBu input, 30dB gain)Frequency Response (Measured after ADC)20Hz – 20kHz better than ±1dBGain Range 0dB – >45dBInput Impedance 1.3K OhmsADVANCED SPECIFICATIONSMaximum Output Level -6dBu Frequency Response 20Hz – 20kHz better than ±1dB Dynamic Range 112dBA Max output power at 1% THD 32Ohms – >210mW 300Ohms – >390mW Dynamic Range 99dBAFrequency Response (Measured after ADC)20Hz – 20kHz better than ±1dBGain Range 0dB – >45dBMax Input +12dBuTHD <0.04% (-10 dBu input, 0dB gain)Input Impedance 900K Ohms。

AXIS T61 Audio and I O Interface Series 用户手册说明书

AXIS T61Audio and I/O Interface Series AXIS T6101Audio and I/O InterfaceAXIS T6112Audio and I/O Interface 用户手册目录关于本手册 (3)产品概述 (4)解决方案概述 (5)其他设置 (6)关于产品 (6)关于摄像机的内置帮助 (6)音频 (6)事件 (7)故障排查 (9)技术问题、线索和解决方案 (9)规格 (10)LED指示灯 (10)连接器 (10)关于本手册关于本手册本用户手册描述了几种产品。

这意味着您可能会找到不适用于您产品的说明。

产品概述产品概述AXIS T6101和AXIS T61121LED指示灯2I/O连接器3音频输出4音频输入5网络连接器（PoE输入）6网络连接器（PoE输出）7PoE交换机（2类和3类）AXIS T61121麦克风2外壳3侧盖4安装支架其他设置其他设置关于产品当您将产品连接到具有更新固件版本的支持Axis网络摄像机时，摄像机网页中将显示音频和I/O的设置。

您可通过摄像机网页进行本手册中描述的设置。

关于摄像机的内置帮助您可通过摄像机网页访问内置帮助。

该帮助提供了产品的功能及其设置的更详细信息。

音频向录像添加音频打开音频：1.转到设置>音频，然后打开允许音频。

2.转到输入>类型，然后选择您的音频源。

编辑用于录制的流配置文件：3.转到设置>流，然后单击流配置文件。

4.选择流配置文件，然后单击音频。

5.选中此复选框，然后选择包含。

6.单击保存。

7.单击关闭。

允许双向音频通信注当您在摄像机的用户界面中设置了双向音频通信后，请使用视频管理系统来利用该功能。

该示例解释如何通过产品进行音频通信。

将麦克风和扬声器连接到产品：1.将麦克风连接至音频输入接头。

2.将扬声器连接至音频输出接头。

其他设置允许在摄像机网页中使用双向音频：1.转到设置>流，并包含音频。

altair-g1.1 牵牛星用户指南说明书

本出版物的任何部分，包括但不仅限于图片，文字，代码与交互功能，未经声韵音响或其授权人的书面许可，不得复制。

本手册仅作提供信息之用，而不应被视为一种承诺。

声韵音响有权对各种细节进行变更,不必另行通知。

声韵音响对可能出现在本手册中的错误不承担责任。

AURALiC, inspire the music, Lightning Streaming, Purer-Power, ORFEO 以及它们的图标是声韵音响的注册商标。

这些商标或商业外观不得以任何可能引起消费者混淆的方式，或任何贬低、诽谤声韵音响的方式用于与本网站以及声韵音响无关的任何产品或服务上。

本网站上的非声韵音响持有的商标，是其商标权利人所独有的财产，这些权利人可能与本网站有相应的关系，也可能没有关系，或由声韵音响所赞助。

未经声韵音响或相关商标所有人的书面许可，本网站上的任何内容都不应被解释为以默许或其他方式授予许可而使用本网站上出现的商标的权利。

目录符合标准说明 (4)关于保修 (5)包装清单 (5)放置和线缆连接方式 (6)放置 (6)检查您的交流电电压 (6)线缆连接方式 (7)前面板 (8)使用您的 ALTAIR G1.1 (9)启动和休眠 (9)欢迎界面 (10)主菜单 (11)正在播放 (12)列表 (13)输入 (13)系统 (14)流媒体播放器 (16)处理器 (17)音乐图书馆 (18)Lightning 网页控制界面 (21)为i OS 开发的L ightning DS (22)使用其他软件 (22)使用A LTAIR G1.1 作为U SB 解码器 (23)将 ALTAIR G1.1 作为流媒体播放器 (23)网络设置 (23)通过以太网连接到您的网络 (24)通过 Wi-Fi 连接到您的网络 (26)开始使用 Lightning DS 控制软件 (30)智能红外遥控 (32)维修服务 (33)规格 (35)AURALiC 产品登记表 (36)为了降低触电风险，请不要私自拆开产品外壳。

教室音频系统安装及操作手册说明书

Classroom Audio System教室音频系统 Installation and Operating Manual安装及操作手册V 1.3重要的安全说明重要的安全说明1. 在安装和使用设备前请先仔细阅读本安全操作规程。

2. 请保存好您的安全操作指南便于以后作参考用。

3. 请遵守所有设备操作指南中的“警告”事项。

4. 须遵守各项操作指南中的规章原则。

5. 清洁设备：清洁设备之前，请先关掉电源，从插座中拔出设备插头，将各连接的系统单元拆卸出来，清洁时请用干燥的软布擦拭。

6. 未经生产厂家同意，不要使用任何不匹配的附件配置，这都有可能引起危险事故。

7. 勿将设备置于潮湿或靠近热源的地方，以免发生危险。

8. 设备不应遭受水滴或水溅，不应放置诸如花瓶一类装满液体的物品。

9. 电源插头作为断接装置，应便于操作。

10. 设备应可靠连接到带保护接地的电网电源输出插座上。

11. 勿将设备放置在不稳固的台面上；在运输过程中避免设备遭受强烈振动而引起损坏，建议在运输前选用合适的包装或使用原包装。

12. 请勿阻塞设备上的通风开口，并保持室内的空气通畅，便于设备的维护。

13. 供电电压：AC 100 V-240 V 50 Hz/ 60 Hz14. 设备连接所需要的延长电缆线请绕道穿行，勿有重物挤压，这样能有效维护系统的正常工作。

15. 每套系统中所连接的接收器不得超过规定数量，否则可能会导致整个系统中设备的异常工作，如有特殊要求请与距离您最近的深圳台电售后服务中心取得联系。

16. 确保设备不被任意拆开机壳，也不允许任何硬质导体或液态物质残留在机壳内。

17. 设备有需要维护时，不要自行拆卸，请及时与距离您最近的深圳台电售后服务中心取得联系。

18. 所有TAIDEN产品将提供一定期限（详见保修卡）免费保修，但人为损坏除外，例如：A. 设备因人为作用被摔坏；B. 因操作员操作不当而导致设备受损；C. 自行拆卸后而导致部分设备零件受损或丢失。

漫步者原子豆编解码

漫步者原子豆编解码【最新版】目录1.漫步者原子豆编解码器介绍2.编解码器的原理与特点3.应用范围与实际效果4.总结正文漫步者原子豆编解码器是一款性能出色的音频处理设备，适用于音乐制作、影视后期制作等领域。

它能够实现音频信号的高质量编解码，提供高品质的音频体验。

编解码器的原理是通过对音频信号进行数字信号处理，将模拟音频信号转换成数字信号，以便于存储和传输。

在播放时，再将数字信号转换回模拟信号。

这种处理方式能够有效地降低音频信号在传输过程中的损耗，保证音频质量。

漫步者原子豆编解码器采用了先进的数字信号处理技术，确保了音频信号的高保真度。

漫步者原子豆编解码器具有如下特点：1.高采样率：编解码器支持高达 32 位的采样率，使得音频信号的细节表现更加丰富，提高了音频质量。

2.高比特深度：比特深度是指音频信号的数字量化精度，漫步者原子豆编解码器支持高达 24 位的比特深度，使得音频信号的表现更加细腻，降低了数字失真。

3.多种格式支持：编解码器支持多种音频格式，包括 MP3、WAV、FLAC 等，满足用户不同的需求。

4.便捷的操作：漫步者原子豆编解码器采用一体化的设计，用户只需连接音频设备即可使用，无需复杂的设置。

漫步者原子豆编解码器在音乐制作、影视后期制作等领域有广泛的应用。

在音乐制作中，编解码器能够提供高品质的音频信号，使得音乐作品的表现更加细腻丰富。

在影视后期制作中，编解码器能够确保音频信号的高质量，提升观众的观影体验。

总的来说，漫步者原子豆编解码器是一款性能出色的音频处理设备，能够实现音频信号的高质量编解码，提供高品质的音频体验。

BGM产品介绍

BGM产品公共广播系统一. X1400数字音频广播系统1.系统主机X14010产品概述X1400数字音频广播主机是整个系统的核心，对整个系统进行管理和触发信号的响应。

产品特性◆通过RJ45网络接口连接到系统局域网,避免了繁琐的接线工作。

◆系统主机软件可以为系统提供各种系统管理服务。

◆用户可以通过PC客户端软件对系统主机进行配置和监控。

◆主机也可脱离PC客户端的控制管理，自动运行。

◆一路RS485接口与集中式控制输入器连接,实现消防联动。

◆能同时通过网络传送4路内置立体声音频信号。

2. 数字音频广播系统控制台X14013产品概述X14013是一款嵌入式的数字广播系统控制单元，具有远程实时控制，监视数字寻址广播系统运行状态功能。

控制台采用人性化的外观设计，具有操作简单，易于操作一体式设计特点。

产品特性◆一体机的构架，外观轻巧。

◆采用触摸屏操作方式◆轻松点击触摸屏进行远程呼叫功能。

◆人性化的操作界面，操作简单方便。

◆具有用于单独TCP/IP网口,独立CobraNet网口。

◆本地话筒接口配BGM界面式话筒,◆配有鼠标和键盘的PS2接口.◆系统可以配置4台控制台。

3. 数字音频四通道编码器X14015产品概述X14015是一款基于COBRANET协议和嵌入式设计的四通道音频信号转换的编码器，可同时将四路音频数字编码，进行网络传输的功能。

编码器采用台式立体结构，箱体采用黑漆色的经典设计气息。

具有操作简单，功能强大的设计特点。

产品特性◆红外学习串口和4路独立的红外控制输出接口，实现外部音源控制功能◆一路RS485接口连接电话广播处理器，实现远程电话广播。

◆四路立体声音频输入连接本地外部音源◆四个控制输入接口用于连接消防触发信号如果只需要单一通道您还可以选择数字音频单通道编码器X14014。

4. 数字音频解码器X14017产品概述X1407是一款基于COBRANET协议和嵌入式设计的四通道音频信号转换的编码器，4路独立解码通道对网络数字信号进行音频解码和输出，可实现不同分区播放不同音乐功能。

蓝牙音频开发包Winbond W681360编解码器板用户手册说明书

Bluetooth Audio Development Pack Winbond W681360 Codec BoardUser GuidePart Number ACC-005The information contained in this document is subject to change without notice. EZURiO Ltd makes no warranty of any kind with regard to this material including, but not limited to, the implied warranties of merchant ability and fitness for a particular purpose. EZURiO Ltd shall not be liable for errors contained herein or for incidental or consequential damages in connection with the furnishing, performance, or use of this material.© Copyright 2006 EZURiO Limited. All rights reserved.No part of this document may be photocopied, reproduced, or translated to another language without the prior written consent of EZURiO.Other product or company names used in this publication are for identification purposes only and may be trademarks of their respective owners.Bluetooth® Development KitWinbond Audio Codec BoardPart Number: ACC-0051.General DescriptionThe EZURiO Winbond Codec Evaluation Board plugs into the EZURiO Developers kit and allows you to rapidly test and evaluate Bluetooth audio applications using the EZURiO Bluetooth Intelligent Serial Module to implement the wireless link.The ACC-005 evaluation board is based on the Winbond W681360 codec - a 3V, single channel, 13 bit linear voice-band codec, which is pin compatible to the Motorola MC145483. The codec is used to digitise incoming audio from the microphone into PCM data and convert the PCM digital audio output of the Bluetooth chip into an analogue signal for the headphones. The codec board has a microphone input and headphone output which are compatible with standard PC headsets.The W681630 codec has several features such as power down mode and high pass filter disable (to allow frequencies down to DC to be used). The ACC-005 codec evaluation board provides options to allow these features to be tested.The W681360 incorporates a feature that allows the volume of the codec output to be digitally controlled via 3 bits of the PCM data stream. The BISM II provides an AT command (ATS589) that allows you to control the volume of the codec.This document provides you with information to prototype and evaluate your own audio application. Once you have tried out your application, you will be able to design your own audio solution based around the Winbond codec and the EZURiO Bluetooth Intelligent Serial module.Bluetooth is a trademark owned by Bluetooth SIG, Inc., USA and licensed to EZURIO Ltd.2.OverviewThe codec board is powered by an on-board 3.3V regulator to reduce noise to a minimum. The PCM control signals for the codec go directly to the Bluetooth module on the motherboard via the 10-way connector, as do the 3 push button switches. This allows the switches to be used with an external program that implements the upper portion of headset or Handsfree profile.The microphone input, designed to interface to PC compatible headsets, has a fixed gain of 16 set by external components to the codec (the amplifier itself is part of the codec). Part of the microphone signal is mixed into the headphone output signal via VR2. This feature is known as “sidetone” and allows the user to hear their own voice when speaking. It is commonly used in telephony applications to give the user the necessary audio feedback that their ears expect.The audio output gain is by default fixed at 1. By fitting VR1, the audio gain can be made adjustable.The 120mW stereo output amplifier U3 ensures that the codec board can drive standard 32Ωstereo headphones while keeping total harmonic distortion down to 0.1%.Component PlacementNote that not allcomponents are fitted –non-fitted components areshown without pads. Referto Section 7 for details ofcomponent fitment andspecification.3.Codec Board Quick Start Guide3.1 Getting StartedThe codec board is supplied with a right angle, 10 way connector that can be used to connect it to the main developers kit. If required, this should be soldered to the main board. Alternatively other connectors or ribbon cables can be used.3.2 Equipment Required (not supplied)•Headsets (with microphone) (Standard PC headsets are fine)•EZURiO Wireless Developers Kit•BISM II Bluetooth module (Firmware release V9_20_22 onwards supports audio volume control)Normally two sets of development kit are required to test both ends of an audio link. If an application is being developed with an existing endpoint, such as a mobile phone or headset, only one set may be needed.3.3 Motherboard Jumper SettingsBefore using the codec board, there is a jumper setting on the motherboard that needs to be checked. This is CB1, next to the USB adaptor, which must be removed. If fitted it will short out the PCM output from the codec and prevent it operating. CB1 is only relevant for the WLAN 802.11 data module.3.4 Procedure:1)Plug the BISM II into the socket on the Dev Kit, connect to a PC serial port and power up.See the dev kit manual for different power supply options.2)Check that AT commands are working using EZURiO terminal. (Refer to blu2i Quick StartGuide if needed)3)Run the “ATI3” command to find out the firmware release number. If it is less thanV9_20_22, contact EZURiO to get a firmware upgrade for the BISM II. (Note: older versions of firmware will work, but audio output will be at half the full volume and the ats589=7 command will not be recognised)4)Power down, plug the codec board into the dev kit and power up. Check that ATcommands are working.Configure the Slave unit as follows:AT&F* Restore system defaultsATZ Reset the unit= 4 Makeconnectable and discoverableATS512ATS0=1 Answer after 1 ringATS531=1 Keep AT command mode going after a connection isestablishedATS589=7 Set Max. Volume level (requires firmware V9_20_22)AT&W Save the above settingsATZ Reset the unit.5) Find out the Bluetooth address of the Slave Unit by typing ATI4<return>6) Configure the Master Unit as follows:AT&F* Restore System DefaultsATZ Reset the unitATS531= 1 Keep the AT commands going after a connection isestablishedATS589=7 Set volume to maximumAT&W Save to flashATZ Reset the unit.ATD008098nnnnnn Connect to the slave (substitute your slave’s Bluetoothaddress that you found in step 5 for nnnnnn)AT+BTA1 Establish an audio link – displays AUDIO ON on both sides.(Alternatively AT+BTA7 can be used and the units willnegotiate the best link type.)An Audio link is now established between the two units.AT=BTA0 will turn off the audio link (but still leave the units connected).To change volume use ATS589. ATS589=0 gives minimum, ATS589=7 gives maximum. 4.Bluetooth SCO Links – A Primer4.1 Normal SCOBluetooth uses a Synchronous Connection-Orientated link (SCO) for audio. All this means is that for an audio link, the bandwidth needed to maintain the data rates required by the audio link is pre-allocated between the master and slave. This ensures audio data is always transmitted at the required data rate, and takes priority over the transmission of digital data.The Bluetooth specification for SCO is such that there is no re-transmission if data is corrupted or lost. This explains the crackling and popping that occurs when you get to the limits of radio range.The actual data rate over the air is 64 kbits/sec. There are 1600 timeslots available per second and when a master transmits a SCO packet in one timeslot, the slave replies with its SCO packet in the next. The SCO packet size is fixed at 240 bits (30 bytes). This means when a SCO link is established using the HV3 packet type, two out of every 6 timeslots are used up by the SCO link. This means there is enough bandwidth to have up to three SCO links active between a master and slave at the same time. In this scenario, there are no spare timeslots for other data.There are 3 main types of SCO packets, HV1, HV2 and HV3 (High Quality Voice). As mentioned earlier, the HV3 packet type has a 1 to 1 mapping between incoming audio data and the data transmitted over the air. There is no error correction possible with HV3.With HV1, each bit is transmitted 3 times and a simple voting algorithm is used at the other end to correct for any bit errors. This means that only 10 bytes of actual audio data can be transmitted in a SCO packet. To maintain the 64 kbits/sec data rate, all 6 timeslots have to be used for the SCO link, leaving no bandwidth available for data.With HV2, an FEC algorithm is used to correct for 1 bit errors. This increases the data packet size by 50%. This means that only 20 bytes of actual audio data can be transmitted in a SCO packet. To maintain the 64 kbits/sec data rate, 4 out of every 6 timeslots are used for the SCO link.AT+BTA1 enables HV3AT+BTA2 enables HV2AT+BTA4 enables HV1AT+BTA7 allows the link manager to negotiate which packet type to use, the default is HV14.2 Enhanced SCOEnhanced SCO or eSCO was implemented as part of the 1.2 Bluetooth Core Specification Release. The main driving factor was to improve audio quality. This has been achieved by: 1)including a CRC as part of the audio data packet to allow error detection and a re-transmission request. 2)allowing higher data rates by using packets that span more than 1 timeslot 3) allowing asymmetric links to allow high quality audio to be streamed in one direction.eSCO offers significantly better audio quality, but has to be configured at both ends of the link before a unit is enabled to accept incoming connections or enquiries.To try out eSCO, add the ATS584=1 command to the commands listed in the quick start section immediately after the AT&F* and ATZ commands.Both ends of the link must be configured for eSCO for the audio link to be established. If one end is set to eSCO and the other to SCO, you will get an “AUDIO FAIL” when the AT+BTA1 command is issued.The following are the packet types associated with the AT+BTA commands for eSCO.AT+BTA1 – EV3 packet. Up to 30 bytes + CRC. Uses up 1 timeslotAT+BTA2 – EV4 packet. Up to 120 bytes + CRC + 2/3 FEC. Up to 3 timeslotsAT+BTA4 – EV5 packet. Up to 180 bytes + CRC. Up to 3 timeslots. Currently Unsupported4.3 SCO / eSCO Transport DelaysThe following delays have been measured between incoming audio and audio output at the other end of a Bluetooth link.Normal SCO: AT+BTA1 7.84 ms AT+BTA2 9.24 ms AT+BTA4 10.8 msEnhanced SCO AT+BTA1 12.1 ms AT+BTA2 33.4 ms AT+BTA4 41.2 msAs can be seen, the additional error correction of eSCO comes with a transport delay penalty. This is because a buffer is needed to ensure that there is still data to output while waiting for a corrupted data packet to be re-transmitted.For AT+BTA1 and normal SCO, the data is transmitted once every 6 timeslots so the transport delay is expected to be 6/1600 = 3.75ms. When doing loop-round testing with the codec, i.e. with no transport delay, it was found that from input to output, the codec added ~1ms of delay at 1kHz and 1.5ms at lower frequencies.4.4 PCM TimingThe codec samples at 8 kHz. The default mode of operation of the codec is 16 bit Receive Gain Adjust Mode. In this mode, in every 8 kHz cycle, 16 bits of data is clocked into the codec. The first 13 bits are PCM audio data, the last 3 bits are volume data. Of the last three bits, 000 equates to maximum volume (ATS589=7), 111 equates to minimum volume (Ats589=0).At maximum volume, the output signal matches the amplitude of the input signal at the other end of the Bluetooth link. It is more appropriate to think of this feature as being an attenuation control.The clock rate used for sampling is 250kHz (4µs). 16 clock cycles takes 64µs. 8kHz equates to 125µs.The same timing is used for all packet types in both SCO and eSCO modes.5.Frequency Response5.1 Codec Frequency ResponseThe codec frequency response can be measured by connecting PCM_IN from the codec to PCM_OUT to the codec (PCM_OUT from J1, the 10 way connector has to be disconnected). A 1kΩ pull down resistor is needed on PCM_OUT to ensure maximum volume setting.The following graph shows the measured frequency response. For this test, R32, the side-tone resistor was removed to prevent audio feedback.A 1V peak to peak sine wave was injected into the microphone circuit and its amplitude measured at TP5, A0, the input to the codec. The output from the codec was measured on TP6, PA0+.The chart below shows the codec frequency response with the High Pass Filter Enable (HB – Pin 16) pin set high and set low.As can be seen from the chart, the codec frequency response is flat between 300 and 3,300 Hz. With the high pass filter on, the 3dB points are at 150Hz and 3,600 Hz respectively. With the high pass filter off, the 3dB point goes down to approximately 15Hz.5.2 Bluetooth Link Frequency ResponseThe Codec 13bit linear data is coded within the Bluetooth chip using CVSD (Continuous Variable Slope Decode) encoding for transport over the Bluetooth link. CVSD is essentially a form of Adaptive Differential PCM (ADPCM) and is well suited for voice transmission. It is forgiving of individual bit corruption as each bit only implements an up or a down shift relative to the previous level (corruption of the MSB of a 13 bit sample would create a much larger error term than is possible with ADPCM). A draw back of ADPCM is that it cannot track large delta changes in signal quickly enough. For voice, this does not present a problem.The chart below shows the frequency response of the Bluetooth link at different levels of input sine wave.As can be seen, the frequency response can only be considered to be flat when the input voltage level is less than a 0.3V peak to peak sine wave.6.Circuit DescriptionThis section describes the individual parts of the circuit and give design information aboutthe components, to allow you to adapt the circuitry of the codec board for your own implementation.6.1 Audio AmplifierThe Winbond codec is capable of driving a 32Ω load directly if the gain of the output amplifier is reduced by a factor of 4. This is done by Setting R1 to 39kΩ.Of the stereo headsets tested, it was found that 32Ω was a common impedance for each earpiece. For a stereo headset where two speakers are being driven in parallel this would be equivalent to driving a 16Ω load. This is out of the codec’s specification so a small headphone amplifier, U3, has been used on the evaluation board. This is not required if the impedance of the earpiece is equal or greater than 32Ω.The large 100 μF decoupling capacitors have been used so that the codec could be tested in its “high pass filter mode disabled” configuration. If you do not require a frequency response to go down below 300 Hz, then these capacitors can be reduced to small values. The main design consideration is the impedance should not be significant compared to the impedance of the headphone selected at frequencies of interest.E.g. if using a 32Ω headphone and expecting a 3dB point at 300 Hz, then the decoupling capacitor impedance could be 32Ω at 300Hz i.e. 10 μF. This requires a much smaller footprint than the 100μF used in the reference design.6.2 Driving the Headset Directly from the CodecThis will achieve the most cost effective design but care must be taken to ensure that the 32Ω specification of load is met by selecting an appropriate headset.Remove R10, R13 and R12. Fit R11, R9, R38 as zero ohm links. Fit 39kΩ in place of R1 to reduce the gain by 4.In-house testing showed that with a 32Ω load and with R1 set to 39kΩ, that there was some distortion at zero cross-over but that it was not easily perceptible.Even though the output signal level had been reduced by a factor of 4, on the headsets tested, the volume levels sounded loud enough for most applications. It is important to check this with the target headset for your application.6.3 Microphone CircuitThe microphone circuit is designed for an electret microphone (which is commonly used in PC applications). Typically this would be powered by 5V via a 2.2kΩ series resistor. In the reference design, it is powered by 3.3V to ensure a clean supply regardless of the power supply used to power the Dev kit. This reduces the sensitivity of the microphone - you should test your application with the microphone and voltage you intend to use in order to determine your component values.The gain of the microphone is set by R22 and R24, with gain being equal to R22/R24. The current values are 62K and 3.9K, giving a gain of approximately 16. When changing to a different gain, R27 and R25 should be set to the new values as well. This ensures that the load seen by common mode noise on the microphone is identical and prevents it from being amplified.R31 is a no fit resistor. It’s purpose is to facilitate test modes where a user wants to loop audio output directly back to the audio input to conduct an over the air audio test.6.4 SidetoneWhen we talk, we hear our own voice, which is part of normal speech perception. If our ears are covered by headphones, we do not hear our voice, which is perceived as abnormal. (Try covering your ears while talking to notice the difference).To compensate for the loss in feedback to the ear when it is covered with a headphone, most telephony systems inject some of the microphone signal back into the audio output path so that the person perceives their own speech as normal. This feature is commonly referred to as sidetone.Variable resistor VR2 allows you to control the amount of sidetone that is fed back to the audio output so that the user perceives their speech as normal.If the headset design does not totally cover the ear, then the sideband circuitry can be omitted.6.5 Power DownFor battery powered audio applications, the power down feature of the codec allows you to turn it off and save power when it is not being used. This feature can be tested by fitting R7 with a 0Ωlink and controlling the PUI input of the codec via MPIO_5.For AT commands, MPIO_5 translates to GPIO 7.The put GPIO 7 into output mode, use “ats610=$040”To turn the codec on, use “ats627=1”To turn the codec off, use “ats627=0”6.6 Alternative PCM_CLKSome applications require that the PCM Clock is driven by external circuitry. This requires the PCM Interface provided by the BISM to be put in Slave mode and a clock is supplied by the external circuitry on MPIO_7.Contact Ezurio for further details if this is a requirement.6.7 SwitchesThe switches S1, S2 and S3 have no defined function. They are there to assist you to prototype your audio application. e.g. If your application requires a button to be pressed for the user to answer an incoming connection, you can prototype that function using one of the switches provided.ATS620 allows you to read the status of the GPIO ports.No switches pressed: ATS620? => $0028S1 pressed (GPIO 9) ATS620? => $0128S2 pressed (GPIO 7) ATS620? => $0068S3 pressed (GPIO 8) ATS620? => $00A86.8 High Pass Filter EnableThe W681360 can have its High Pass filter enabled or disabled, depending on the state of the HB pin (Pin 16). This is pulled high or low by R3 or R4 (Default). See section 5.1 for more details.6.9 GPIO to MPIO MappingAT commands use GPIO numbers to represent I/O lines. These GPIO numbers map to physical signals drawn on the schematics as MPIO lines. Some of the GPIO/MPIO lines are used when providing a full RS232 interface.The following tables gives the mapping between GPIO, MPIO and RS232 signals.DCD MPIO_3RI MPIO_2DTR MPIO_9DSR MPIO_8GPIO_1 MPIO_0GPIO_2 MPIO_1GPIO_3 MPIO_9GPIO_4 MPIO_10GPIO_5 MPIO_11GPIO_6 MPIO_4GPIO_7 MPIO_5GPIO_8 MPIO_6GPIO_9 MPIO_7Note: For the BISM PA (Class 1 design), MPIO_0 and MPIO_1 are used to control the RF switch so are not available to the AT Command Set.7. Bill of MaterialsNot all components are fitted, as some provide alternative functionality or implement non-standard options.Refer to the previous sections and the schematic for information on the component function. Components marked in blue are not fitted.Reference Part ToleranceDescription Manufacture r Part No / FootprintC1,C7100nF20%Ceramic Capacitor0805 C2,C3,C6 10uF '+80/-20% Tantalum Capacitor TANA C4,C5,C810nF20%Ceramic Capacitor0805C9,C10 100uF 20% Electrolytic Capacitor Panasonic EEE0JA101SP C11,C12,C17,C18 2.2uF '+80/-20% Ceramic Capacitor 0805 C13 22uF '+80/-20% Ceramic Capacitor 1210 C14 100nF '+80/-20% Ceramic Capacitor 0805 C15,C19 100pF 20% Ceramic Capacitor 0805 C161.0uF'+80/-20%Ceramic Capacitor0805D1,D2,D3,D4,D5,D6,D7,D8 BAT54S Dual Schottky Diode BAT54S Zetex BAT54S J1 10 Way 0.1" R/A PCB Socket Harwin M20-7891046 J2,J3 3.5mm 3way Audio Jack Skt Schurter 4832.232L110uHThin Film Inductor1210 R1,R2,R5,R35,R36,R37 10K 1% Thick Film Resistor 0805 R3,R7,R8,R9,R11,R34,R38 0R Not Fitted 5% Thick Film Resistor 0805 R4,R6,R10,R12,R13,R330R5%Thick Film Resistor0805 R14,R28,R29,R30 1K 5% Thick Film Resistor 0805 R152K2 Not Fitted 5% Thick Film Resistor 0805 R16,R17,R18,R19,R24,R25 3.9K 1% Thick Film Resistor 0805 R26,R20 1.5K 5% Thick Film Resistor 0805 R23,R21 200K 5% Thick Film Resistor 0805 R27,R22 62K1% Thick Film Resistor 0805 R31 62K Not Fitted 1% Thick Film Resistor 0805 R32 75K5% Thick Film Resistor0805 S1,S2,S3OMRON/B3S-1000Push Button Switch SPNO SMD Omron B3S-1000U1 AME8800AEFT 3.3V Low Drop Out Regulator300mA AME AME8800AEFT U2 W681360RG W681360RG CODEC Winbond W681360RG U3 LM4908MM Dual Headphone Amplifier Nat. Semi. LM4909MMVR1 20K Not Fitted 20% 20K Trimmer Vishay TS53YL 20K 20% TR VR250K20%50K TrimmerVishayTS53YL 50K 20% TR8. References1. Winbond W681360 Data Sheet – /PDF/Sheet/W681360.pdf2. ACC-005 Schematic – ERBLU49-002A1-029.DisclaimersEZURIO’S WIRELESS PRODUCTS ARE NOT AUTHORISED FOR USE AS CRITICAL COMPONENTS IN LIFE SUPPORT DEVICES OR SYSTEMS WITHOUT THE EXPRESS WRITTEN APPROVAL OF THE MANAGING DIRECTOR OF EZURIO LTD.The definitions used herein are:a) Life support devices or systems are devices which (1) are intended for surgical implant into the body, or (2) support or sustain life and whose failure to perform when properly used in accordance with the instructions for use provided in the labelling can reasonably be expected to result in a significant injury to the user.b) A critical component is any component of a life support device or system whose failure to perform can be reasonably expected to cause the failure of the life support device or system, or to affect its safety or effectiveness.EZURiO does not assume responsibility for use of any of the circuitry described, no circuit patent licenses are implied and EZURiO reserves the right at any time to change without notice said circuitry and specifications.9.1 Data Sheet StatusThis data sheet contains preliminary data for use with Engineering Samples. Supplementary data will be published at a later date. EZURiO Ltd reserve the right to change the specification without prior notice in order to improve the design and supply the best possible product.Please check with EZURiO Ltd for the most recent data before initiating orcompleting a design. Designers should check the production status of any engineering firmware used during development before it is deployed.。

E1编解码器

E1专线音视频编解码器用户手册二○○四年十一月目录一、产品简介.............................................................................................. ４功能特点.............................................................................................................................................. ４二、产品结构.............................................................................................. ４内部布置：.......................................................................................................................................... ４外形尺寸:............................................................................................................................................ ４三、技术指标.............................................................................................. ５四、接口说明.............................................................................................. ６1. 前面板 ......................................................................................................................................... ６2. 后面板 ......................................................................................................................................... ６3. 接口指示说明： ....................................................................................................................... ６五、接线说明.............................................................................................. ７1. 视频接线..................................................................................................................................... ７2. 音频接线..................................................................................................................................... ７3. E1接线 ........................................................................................................................................ ７4. 控制接线..................................................................................................................................... ８六、透明串口定义 .................................................................................... ９串口定义： ....................................................................................................................................... ９拨码开关： ................................................................................................................................... １０七、典型应用.......................................................................................... １１八、产品装箱清单 ................................................................................ １２第２页序言简介本音视频编解码器是为适应基于电信E1（2M口）传输通道而设计的，采用强大的MPEG2压缩方式，具有强大的即时图像捕捉和图像压缩功能。

海康DS解码器说明

16 路 1080P@30fps 及以下分 32 路 1080P@30fps 及以下或 64 路 1080P@30fps 及
辨率
分辨率
以下分辨率
画面分割数
1/4/6/8/9/12/16
1/4/6/8/9/12/16/25 1/4/6/8/9/12/16/25/36
外
网络接口
1 个 RJ45 10M/100M/
SXGA:1280*1024@60Hz,
720P:1280*720@60/50Hz,
XVGA:1280*960@60Hz, UXGA:1600*1200@60Hz,
1080p:1920*1080@60/50Hz,
WXGA:1280*800@60Hz/1366*768@60Hz,
输
WXGA+:1440*900@60Hz,
XVGA:1280*960@60Hz, UXGA:1600*1200@60Hz,1080p:1920*1080@60/50Hz,
输
WXGA:1280*800@60Hz/1366*768@60Hz,WXGA+:1440*900@60Hz,
入
WSXGA+:1680*1050@60Hz
参 DVI-I 输入分辨率 XGA:1024*768@60Hz/800*600@60Hz, SXGA:1280*1024@60Hz,720P:1280*720@60/50Hz,
DS-6912UD 支持 12 路 1200W，或 24 路 800W，或 36 路 500W，或 60 路 300W，或 96 路 1080P@30fps 及以下分辨率同时实时解码；
DS-6916UD 支持 16 路 1200W，或 32 路 800W，或 48 路 500W，或 80 路 300W，或 128 路 1080P@30fps/3Mbps 及以下分辨率同时实时解码；

华硕主板

数字家庭主板M2NDH-支持AMD®SocketAM2Athlon64FX/Athlo64X2/Athlon64/Sempron -AMDLive!™Ready-强大扩充能力：1xPCI-Ex16、2xPCI-E、3xPCI-华硕WiFi-APSolo-华硕DHRemote™-华硕MP3-In™-华硕Q-Connector-高保真音频中央处理器支持AMD®SocketAM2Athlon64FX/Athlo64X2/Athlon64/Sempron 支持AMDCool'n'Quiet™技术AMD64架构，同时兼容32位和64位计算AMDLive!™Ready芯片组NVIDIAnForce®430MCP前端总线2000/1600MT/s内存双通道内存架构4x240-pinDIMM内存插槽，支持最大容量高达8GB的DDR2800/667/533ECC和non-ECC、un-buffered内存扩充插槽1xPCI-Expressx16插槽2xPCI-Expressx1插槽3xPCI2.2插槽存储装置/RAID-1xUltraDMA133/100/66/33-4xSerialATA3.0Gb/s-NVIDIAMediaShield™RAID通过SerialA TA设备支持RAID0、1、0+1、5和JBOD网络功能NVIDIAnForce®430内建GigabitMAC，支持externalAttansicPHY无线局域网：54MbpsIEEE802.11b/g（华硕WiFi-APSolo）音频功能ADI6声道高保真音频CODEC背板S/PDIF数字音频输出USB高达8个USB2.0/1.1接口M2N-VMDH-AMDSocketAM2-NVIDIAGeForce6100/nForce430-双通道DDR2800/667/533-1xPCIExpressx16+1xPCIExpressx1+2xPCI-双VGA：DVI-D和D-Sub-8声道高保真音频-2x1394a接口中央处理器支持AMD®SocketAM2Athlon64X2/Athlon64FX/Athlon64/Sempro nAMDCool'n'Quiet™技术AMD64架构，兼容32位和64位计算AMDLive!™Ready芯片组NVIDIAGeForce6100/nForce430前端总线2000/1600MT/s 内存双通道内存架构4x240-pinDIMM插槽，支持最大容量为8GB的DDR2800/667/533non-ECC，un-buffered内存显卡集成GeForce6100GPU高清晰视频处理，最高分辨率可达1920x1440（@75Hz）支持RGB显示；UXGA1600x1200（@60Hz）支持DVI-D显示支持双VGA输出：DVI-D和RGB注意：DVI-D不能用来输出RGB信号至CRT。

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朴ＬＬ
有大十１０ｄ的信噪０Ｂ
比（Ｎ）ＳＲ，而在立体声回放模式下的功耗小足５Ｗ。ｏｒ
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富士通微电子推出两款新型大规模集成电路（１Ｌ）Ｓ，可
ＡＤＵ１６Ａ７１音频编解码器中增加了ＳｇＤＰ数字音频支持全高清视频（２ｉｍａＳ１０点 × １８９００行）Ｈ２４格式下的编、的．６处理器，可为终端用户定制音频体验。ＡＤＩ公司的Ｓｇ解码，ｉ— 这两款产品扩充了其在Ｈ２４Ｃ．６ＯＤＥＳ产品的阵ＣＬｌｍａｔｄｏ图形化设计工具能够与Ａ公司Ｓｕｄｘ算容。ＭＢ６５ＳｕｉＤｌｏｎＭａ８Ｈ５芯片将率先推出，该芯片在低功耗方面具有法包、用户开发的算法和／第三方音频算法轻松的结合行业领先水平，或在进行全高清编码时，包含内置存储器的功
便携式媒体播放器、便
携式导航设备、数码相机及其它移动音频与电话应用。ＡＡ３１具ＤＵ１６
耳机中，以创造一个无线网络的音乐环境。蘩弱翻髓圈
Ｂｏａｄｃｏｍｗｗｗ．ｏａｄｃｏｍｒｂｒｃｏｍ
ＭＢ６５／６全高清Ｈ２４ＣＯＤＣ８Ｈ５５：。６Ｅ
Ｐ接Ｃ或录像机。整弱露嘲ＢＣＭ２４整合芯片使手机制造商能够对他们的较低样，ＣＩＮ可以方便的连接Ｐ０９
端产品的基本功能进行扩展，同时还为他们的客户提供有
Ｆｊｓｗｗｗｆｉｕｃｒｕｉｕｔ．ｊｓ，ｏｕｔｎ
世界寝子元器件２０２ｇｅ＠ＤＣＯｒ０８１ｃＣＣｎ
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（行扫描）逐的处理速度，从而进一步提升图像画质。两款新产品均使用富士通实验室具有自主知识产权的
成了Ｂｕｔｏｈ２１ｌｏｔＶ．＋增强数据率（Ｄ）ｅＥＲ基带、射频和软件以画质算法，实现了高画质并利用高压缩技术降低了视频处
ＡｎａｌｏｇＤｅｖｉｃｅｓＷＷＷ．ａｎａｌｏｇ，ｎｃｏｒ
和其它便携式设备，能获得更长的连续记录和播放高清视
频的时间，时还能保持超高画质。ＭＢ６６ＣＥ芯８Ｈ５ＯＤＣ
片延续了目前ＭＢ６１８Ｈ５的超高画质，并可支持每秒６Ｏ帧
决方案。ＡＡ１６与ＤＵ３１ＡＡ１６４ｂ音频编ＤＵ７１２ｉｔ解码器适合无线手机、
部的适配器，并町使音频流进入汽车和家庭内部的立体声调频接收机中。ＢＭ２４Ｃ０９还支持立体声蓝牙音频流（调频无线电或数字音乐文件），同时可将这些音频流分发至多个体声
总汇
吸引力的音乐体验，以继
数搴家电元器件
ＡＤｘ１音频编解码器ＡＵ６：
续推动音乐手机品牌的快
速增长。ＢＣＭ２４被赋０９予Ｓｒｄｏ话音处理ｍａｔｉＡｕ
和增强的蓝牙射频功能，可以改善话音的质量并延
Ａ推出一对用于高性能便携式音频电子产品的编解长单耳机的连接距离。在ＤＩ
码器（编码器／解码器） —— ＡＡ３１ＡＤＵ７１能在不ＢＭ２４ＤＵ１６与Ａ１６，ＣＯｇ中集成了调频发
影响音频质量的情况卜延长
控制。模拟输入级支持立体声线路输入、立体声模拟麦克风
ＭＢ６５８Ｈ５在全高清编码时，包括内置存储器的功耗，
接口，以及为麦克风偏置电压供电的电路。模拟输入级还支其总功耗仪为５０ｍＷ，０从而使得使用该芯片的数字摄像机
持新的高性能、低噪声数字麦克风标准。霹弱霞圈
与集成。
耗总共为５０ｍＷ。此外，０即将推出的ＭＢ６５８Ｈ６芯片可支
ＡＡ３１ＡＤＤＵ１６与ＡＵ１６音频编解码器包括立体声音持处理全高清视频（７１簿秒６Ｏ帧（逐行扫描）（０）可进步）６ｐ，频ＡＤＣ与ＤＣ，Ａ支持８Ｈ一６ｋｚｚ９Ｈ的采样速率和数字音量提高图像画质。ｋ
及一个高性能的调频立体声无线电收发器。款针对移动设理的工作量。该为增强产品的外围连接性，款产－设置了很两口ｌ］
Ｕ总线上的主机接口，是一备的・代主控制器接口（Ｃ）ＨＩ解决方案扩充了Ｂｏｄｏｒａｃｍ多接口。可直接连接到外部ＣＰ公司成功的无线整合芯片产品线，增加了调频发射功能、增组１宽的并行接口；一组Ｔ６位Ｓ接口可用作视频接口。另强的话音和立体声音频处理以及嵌入式软件功能。外，使用串行接口可以大大减少主机接口上的脚数量；同