09为汽车应用实现完美音频处理方案,支持Bongiovi DPS专利技术

FAW音响方案FAW音响方案是一项旨在提升汽车音响体验的创新性解决方案。

本文将介绍该方案的特点、优势及应用，探讨其对驾驶者和乘客的影响，以及对汽车行业带来的新机遇。

一、方案特点FAW音响方案是一种整合了先进音频技术的汽车音响系统。

该方案具有以下特点：1. 多声道环绕音效：FAW音响方案采用多声道环绕音效技术，能够提供更加真实立体的音频体验。

无论是听音乐、观看电影还是打电话，用户都能感受到身临其境的沉浸式音效。

2. 自动环境适应：FAW音响方案通过传感器和智能算法，能够自动感知车内环境的各种噪音和共振情况，进而动态调整音响参数，提供更清晰、准确的声音。

3. 个性化设置：FAW音响方案还支持个性化设置，用户可以根据自己的喜好调节音效参数，如音量、音质、低音效果等，以获得最佳的音乐享受。

二、方案优势FAW音响方案相比传统汽车音响系统，具有许多显著的优势：1. 高品质音效：FAW音响方案采用高品质音频芯片和扬声器，能够呈现更加逼真、细腻的音效，使用户尽情享受高品质音乐。

2. 智能控制：FAW音响方案配备智能控制系统，能够根据车速、加速度等信息动态调整音效参数，确保音响效果始终稳定，提供卓越的用户体验。

3. 节能环保：FAW音响方案采用先进的节能技术，能够在确保音质的同时，降低能耗，减少对汽车电瓶的负荷，对环境更加友好。

三、方案应用FAW音响方案适用于各类汽车，包括私家车、商务车以及豪华车等。

无论是出行还是商务会议，FAW音响方案都能够提供卓越的音乐享受和通话体验。

此外，该方案还可用于专业音响设备，如影院音响系统、会议室音响系统等领域。

四、对驾驶者和乘客的影响FAW音响方案对驾驶者和乘客有着积极的影响：1. 提升驾驶乐趣：驾驶者可以享受到高品质的驾驶音乐，营造舒适愉悦的驾驶环境，提高驾驶乐趣和专注力，从而增强行车安全性。

2. 提供沉浸式体验：乘客能够感受到身临其境的音效，无论是欣赏音乐还是观看电影，都能够获得更加沉浸式的体验，提高旅行舒适度。

Integration INSTANT GRATIFICATIONIFBMW-C, IFBMW-S100% DEDICATED TO REPLACING ORIGINAL EQUIPMENT Plug&PlayTo pursue our involvement in new markets and to respond to the various problems which arise when improving original car audio systems, we are increasing our efforts in the integration market.Our leitmotiv: to guarantee affordable audiophile sound whilst simplifying the installation process.Two years after the launch of our first dedicated kit, the development of our Plug&Play line has resulted in two new speaker driver kits which are no longer specific to only one vehicle, but to various models of a world-renowned brand, BMW®.This new range is composed of a 2-way component kit, IFBMW-S (a pair of woofers + a pair of tweeters) and a coaxial kit, IFBMW-C.Our philosophy for these products remains the same, namely the compatibility or the optimum adaptability into the original locations of the manufacturer’s most popular models: 3 Serie, 1 Serie and X1. These Plug&Play kits have been designed for optimisation in terms of mechanics (installation in the original location) as well as in terms of acoustics.Simplicity and pleasure!By guaranteeing a maximum installation time of one hour, and by keeping the factory head unit, the IFBMW-C and IFBMW-S provide the perfect solution to those car owners who wish to discover the pleasure of listening to the best audiophile sound on the market. Performance and features:∙ A luminium/magnesium inverted dome tweeter with phase plug.∙ W oofer equipped with a Polyglass cone and a plug for better dispersion.∙ A ll the integrated crossovers facilitate fitting the speaker drivers into the original locations.Everything is straightforward for the user because the car manufacturer’s standards are used for the connectors. These kits, 100% made in France, are fully coherent with our ready-to-play logic. All the necessary components to carry out a sleek and discreet installation are supplied (speaker drivers + installation manual and video + specific tools for installation).We manufacture world-renowned loudspeakers and the continuous innovation in our R&D department keeps our production workshops busy.100% DEDICATED BMW ®(5 Serie / 3 Serie / 1 Serie / X1) EASY TO INSTALLInstallation 100% dedicated to BMW (5 Serie / 3 Serie / 1 Serie / X1) in less than an hour.IFBMW-C demonstration video: /en/ifbmw-c IFBMW-S demonstration video : /en/ifbmw-sIFBMW-S2-way coaxial kit Car compatibility:- BMW 1 Serie (E81/82/87/88/F20*)- BMW 3 Serie (E90/91/F30/F31*)- BMW 5 Serie (F10*)- BMW X1 (E84)IFBMW-C2-way component kit Car compatibility:- BMW 1 Serie (E81/82/87/88/F20*)- BMW 3 Serie (E90/91/F30/F31*)- BMW 5 Serie (F10*)- BMW X1 (E84)* Technical wire adaptation to fit in those cars models. Non-exhaustive list of vehicles (update on ).Enjoy the same acoustic quality in your car as in your living room...IFBMW-CIFBMW-SFOCAL ® is a trademark of FOCAL-JMLAB ®© Focal-JMlab 2013 - SCEB-131018/2 - Focal ® is a trademark of Focal-JMlab ®.Due to constant technological advances, Focal-JMlab ®reserves its right to modify specifications without notice. Non-binding pictures-Photos L ’Atelier Sylvain Madelon. BMW ® is a trademark of BMW.Technical specifications。

汽车影音系统所用到的芯片随着科技的不断发展，汽车影音系统已经成为现代汽车不可或缺的一个功能。

而汽车影音系统的核心是芯片，它承担着处理音视频信号的重要任务。

下面我们来看看汽车影音系统所用到的一些常见芯片。

1. 音频解码芯片音频解码芯片是汽车影音系统中非常重要的一种芯片。

它负责将数字音频信号解码成模拟音频信号，以供音箱放大器进行放大和播放。

常见的音频解码芯片有Cirrus Logic的CS8416、TI的PCM5102A 等。

这些芯片具有高性能和低功耗的特点，能够提供高质量的音频输出。

2. 视频解码芯片视频解码芯片是汽车影音系统中处理视频信号的关键芯片。

它可以将数字视频信号解码成模拟视频信号，并通过显示屏显示出来。

常见的视频解码芯片有NXP的TDA8954、Realtek的RTD2660等。

这些芯片支持多种视频格式的解码，能够实现高清视频的播放效果。

3. 处理器芯片处理器芯片是汽车影音系统中的大脑，它负责控制整个系统的运行。

处理器芯片可以处理音频、视频和其他操作，还可以支持多种连接方式，如蓝牙、Wi-Fi等。

常见的处理器芯片有Qualcomm的Snapdragon系列、NXP的i.MX系列等。

这些芯片具有强大的计算能力和丰富的接口，能够提供流畅的用户体验。

4. 存储芯片存储芯片是汽车影音系统中存储音视频文件和系统软件的关键芯片。

它可以提供大容量的存储空间，以存储用户喜欢的音乐、电影等媒体文件。

常见的存储芯片有Micron的NAND Flash、Samsung的eMMC等。

这些芯片具有高速读写和稳定可靠的特点，能够满足汽车影音系统的存储需求。

5. 音频功放芯片音频功放芯片是汽车影音系统中负责放大音频信号的关键芯片。

它可以将解码后的音频信号放大到适合车内音箱的电平，以提供清晰、逼真的音效。

常见的音频功放芯片有TI的TPA3116、ST的TDA7850等。

这些芯片具有高效能和低失真的特点，能够提供高品质的音频输出。

打造专属的“德赛西威”作者：罗葵来源：《音响改装技术》2014年第01期“千呼万唤始出来，犹抱琵琶半遮面”！随着一汽大众全新一代高尔夫开始热销，德赛西威全新Golf A7精品导航产品也揭开面纱。

Golf A7原车娱乐系统的升级，从年初开始的市场调研到项目团队一天天的研发，过程可谓“历经艰辛”。

它选用独特的产品构架和全新的改装思维，使得娱乐系统升级后Golf A7既大方又个性。

外观&功能特色：Golf A7的主机显示屏是8寸红外高清触摸屏，看上去高雅又不失霸气，而且它的触控手感不错。

它的机芯很薄，不仅散发着浓郁的科技感，也为原车升级节省空间。

同时升级的主机盖板与原车配色保持一致，加上选用相同连接器，这就让“改装”无迹可寻。

功能方面，自然要和Golf A7“门当户对”。

整个主机界面风格简约得体，以方面体验者使用为原则。

在导航、手机与车机的互动互联方面更加人性化，注意到更多的细节，比如导航界面右侧有K码设置，方便位置信息的分享以及车友之间的联系；同时导航语音提示音量可以自行调节。

德赛西威专属的SiVi LINK技术也支持多种手机端口，操作顺畅；有Line out功放输出接口，支持数字电视、行车记录仪、空气净化模块接入，并有随动倒车轨迹。

在必不可少的音乐方面，它运用全新音乐e栈，和收音界面完美结合，支持无损音源Ape播放等。

图片（外观&功能特色）装车全程回放镜头一：原车CD和空调控制器给原车CD和空调控制器来个大特写！银灰色的色调，整个中控台的布局看上去协调、沉稳！镜头二：取空调、CD机盖板升级新的系统，当然要将原有的取出来。

看到没？黄色的取盖条形工具正缓缓移动，终于一会就将整个空调盖完好取出来了。

空调旋钮没有了“外衣”，看上去很“纯净”！同样的方法，再将CD机的外壳取出！镜头三：原CD机“出位”这会子，原车的CD机要被取出来了。

黄色的条形工具也被取机钥匙代替，看上去瘦弱的钥匙扣，居然有那么强悍的力量，能让CD机“退位”。

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.。

安霸A7LA70方案1. 概述安霸A7LA70方案是安霸公司旗下的一款芯片方案。

安霸公司是全球领先的汽车电子和自动驾驶解决方案提供商之一，在影像处理领域拥有广泛的经验和技术积累。

A7LA70方案是基于安霸公司自主研发的处理器架构设计的，具有较高的性能和灵活的应用能力。

2. 主要特性安霸A7LA70方案具有以下主要特性：2.1 高性能图像处理A7LA70方案采用了一系列先进的图像处理算法和技术，能够实现高效的图像处理和分析。

其内置的图像处理器可以快速处理各种图像传感器采集到的图像数据，并提供高质量、低时延的图像处理结果。

通过使用紧凑的指令集和优化的硬件架构设计，A7LA70方案具备了出色的计算性能和良好的能效比。

2.2 多功能应用支持A7LA70方案可以广泛应用于汽车电子领域，包括车载摄像头、智能驾驶辅助系统、行车记录仪等。

其灵活的架构设计和强大的算法库使得A7LA70方案能够满足不同应用场景的需求，具备高度的可配置性和可扩展性。

2.3 基于深度学习的图像识别支持A7LA70方案集成了深度学习算法加速器，可以快速进行图像识别和目标检测任务。

通过引入先进的深度学习模型和训练技术，A7LA70方案可以实现高度准确的图像识别，支持各种复杂场景下的目标检测和分类。

3. 应用案例3.1 车载摄像头A7LA70方案可以用于车载摄像头系统，提供高质量的图像处理和视频编码能力。

车载摄像头通过安装在车辆前后、侧面等位置，可以实时监测车辆周围的环境，并提供安全驾驶辅助功能，如倒车影像、车道偏离预警等。

3.2 智能驾驶辅助系统基于A7LA70方案的智能驾驶辅助系统可以通过摄像头采集道路信息和车辆状态，并根据实时分析结果提供相应的驾驶建议。

该系统可以识别道路标志、行人、车辆等目标，并进行跟踪和检测，为驾驶员提供安全的驾驶体验。

3.3 行车记录仪A7LA70方案可用于行车记录仪，通过车载摄像头采集车辆行驶过程中的图像和视频数据，并进行实时的图像处理和压缩编码，最终生成高质量的行车记录视频。

SABINE Navigator系列音频处理器（NAV240，NAV360，NAV480）在当今生活中，小到我们每个人都拥有的手机，大到航天飞机，无不渗透的数字化的先进技术。

所以说，数字化将成为现代高科技的标志，也是各个领域追求的发展方向。

在当今音视频这个弱电系统之一的专业领域，追求数字化也是它的发展方向。

我们先来说音响领域的数字化。

以往，在音响领域当中，人们大部分使用的都是模拟设备，与数字设备相比较，模拟设备存在很多弊端。

首先，模拟设备受到电子器件本身的限制，整个设备的技术参数就会受到一定的影响；其次，声音信号在模拟设备传输的过程中会有很大的损耗和干扰；再次，模拟设备不利于储存信息。

由于一系列的原因，音响设备的数字化将成为必然。

这是因为数字设备比模拟设备能更准确更迅速的处理、传输和存储信息。

目前，我们有一些音响设备已经实现了数字化，比如，数字调音台，数字效果器等，他们不仅拥有了以往模拟设备的功能，而且还能实现很多模拟设备不能完成的功能，并且这些设备拥有更高的技术指标和性能。

由此看来，数字技术在音响领域中应用，可以得到更广泛的发展。

NAVIGATOR系列数字音频处理器的强大功能Sabine是一家拥有多年专业音频处理研发经验的厂家，旗下的Navigator系列音频处理器拥有强大的DSP处理能力、良好的产品扩展性和直观简便的人机对话软件。

这个系列的产品按照输入/输出通道分为四个型号：NAV240（2进4出）、NAV360（3进6出）、NAV480/NAV4802（4进8出）、NAV8802（8进8出）。

它适用于剧场、会议中心、商业大厦、流动演出等各种大中小型扩声场所。

Sabine数字处理器是一个硬件与软件相结合的音频系统，硬件用来对声音信号进行转换和处理，软件是对声音系统进行设计和控制。

硬件与软件的结合，使Sabine数字处理器具有了相当强大的功能。

我们知道，以前，一套音响系统的设计，首先要起草设计多种方案，然后要根据用户预算决定相对较为合适的方案，之后要订购设备和购买线材，接着在机架上进行装配、连接和调试，一整套音频系统才算完成。

1、网络音频处理器Audiaflex 系列Audiaflex是音频工程师领域中数字产品的新标准，与前一代Audia产品相比，Flex保持了DSP 处理功能强大，软件简单好用的特点，并在输入输出接口方面改用了插卡式结构，配置更加灵活。

机箱一共可容纳12块插卡，每块插卡为2路输入或2路输出，另有回声消除卡、电话耦合卡、网络电话卡和功率放大器卡。

用户可根据需要自由选择插卡的数量和类型，只要不超过插卡总数限制，任何组合，Flex都能支持。

Flex机箱有带或不带CobraNet网络音频信号接口的两款设备，满足用户联网使用或单机使用的不同需求。

使用时，用户先在电脑上用随机提供的软件进行设计，从软件上丰富的产品库中选择需要的模块，用搭积木的形式完成系统设计，然后将设计图下载到Flex主机中，主机就可以进行复杂而功能强大的音频处理。

对Flex的控制可以通过电脑软件、Biamp遥控面板或者AMX、Crestron 等流行的中控系统完成。

特点：●最多24路输入输出借口，每块板卡提供2路信号，另有回声消除卡、电话耦合卡、网络电话卡和功率放大器卡可选，机箱有带或不带CobraNet网络音频信号接口两种可选。

●软件可显示完整的音频系统设计（多台主机可在一张设计图上反映）●通过电脑进行系统设计和控制●可通过RS-232串行口遥控●Biamp提供音量控制模块和程序控制模块备选●设计软件中自带检测工具●可通过以太网进行遥控●具有多级密码保护功能●系统设计灵活，功能强大●通过CE和UL/C-UL等认证体系软件中的处理模块库●调音台：标准调音台、自动调音台、矩阵调音、调音台联动模块和房间组合模块●均衡器：图形均衡器、参量均衡器、反馈抑制器●滤波器：高通滤波器、低通滤波器、谐振滤波器和相位校正滤波器●分频器：2分频、3分频和4分频●动态处理：自动电平控制、压限器、优先器、环境噪声补偿德国功能●矩阵：2x4到56x56●延时：0到2000毫秒●控制模块：音量、相位开关、哑音、程序预设好逻辑控制●电平表：信号检测、峰值表和均方值表●信号发生器：纯音振荡、扫描、粉噪和白噪●检测工具：幅度畸变、相位畸变技术指标：●频率响应（20Hz~20kHz@+4dBu）：+0/-0.4dB●THD+N（20Hz~20kHz@+4dBu）：线路电平<0.006% ；话筒电平<0.04% ●等效输入噪声（20Hz~20kHz，66dB gain,150ohm）：-125dBu●动态范围（20Hz~20kHz，0dB）：>107dB●最大增益（输入通道）：66dB●串音（@1kHz）：线路电平<-80dB ; 话筒电平<-75dB●输出阻抗（平衡式）：200ohm●输入阻抗（话筒或线路平衡式）：8k ohm●最大输出电平（平衡式）：+24dBu●最大输入电平（话筒或线路）：+24dBu●幻象供电：+48VDC(7mA/input)●输入增益：0dB~+66dB●采样频率：48kHz●A/D-D/A转换器：24Bit●功耗（100~240V AC 50/60Hz）：<110watts●尺寸（高度x宽度x深度）：2U(89mm)x19inches(483mm)x(283mm)●重量：13.62Ibs(6.2kg)DVD*3、无线麦克风、电脑小会议室1小会议室2小会议室3小会议室4大会议室宴会厅2进2出插卡2路输入卡、IP-2∙每个卡提供了两个麦克风/线路输入通道∙平衡式输入的插件阻隔带连接器∙0 〜66分贝增益范围的麦克风或线路来源2路输出卡、OP-2e∙每个卡提供了两个麦克风/线路输出通道∙平衡输出插件阻隔带连接器∙-100 〜12分贝的电平音量控制∙可选的最佳界面/性能2路回声消除卡、AEC-2HD∙双通道的回声消除并降低背景的噪音∙每个输入配置了专用的宽频回声处理芯片∙包括Biamp的专利技术TrueSoundTM AEC 运算法则提供真正20Hz至20kHz宽频回声消除∙在DSP上，不使用AudiaFLEX资源2路电话卡、TI-2∙发起电话呼叫，包括：双音多频音拨号，快速拨号，重拨，闪断∙来电自动应答∙线路回声消除∙连续在线状态监测和故障2路网络电话卡、V oIP-2∙网络电话- 双通道互联网协议语音卡∙支持SIP协议2通道功率卡、PA-2∙直流电4,6,8欧姆扬声器∙软件配置5至30瓦特双通道10至60瓦(桥接)∙升级AudiaFLEX可将功率提升至120w，高达12的PA 扩展盒EXPI、EXPO、EXPA-4、EXPO-4、EXPI/O-2AudiaEXPO-4是Audia®的输出扩展接口，是音频工程领域中数字产品的新标准。