基于VS1053芯片的录音装置设计

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VS1053_MP3模块用户手册.pdf

项目工作电压工作电流 Voh Vol
说明
3.3V(串电阻后，可与 5V 系统连接)
MP3、OGG、WMA、WAV、MIDI、AAC、FLAC（需要加载 patch）
WAV(PCM/IMA ADPCM)、OGG（需要加载 patch）
1 路 3.5mm 耳机接口、1 路 3.5mm LINE IN 接口、IIS 接口、供电及控制接口
用户手册
2
VS1053 MP3 模块用户手册
板载 3.5mm 耳机插口，可直接插入耳机欣赏高品质音乐；
高性能音频编解码模块
板载咪头（MIC），无需外部麦克风，即可实现录音；板载 IIS 输出，可以接外部 DAC，获得更高音质；
板载电源指示灯，上电状态一目了然；
P1 排针为模块的供电与通信接口，采用 1*10P 排针，各引脚详细描述如表 2.2.1 所示：
序号
名称
说明
1
GND 地
2
5V
5V 供电口，只可以供电
3
3.3V 3.3V 供电口，当使用 5V 供电的时候，这里可以输出 3.3V 电压给外部使用
4
XCS 片选输入（低有效）
5
XDCS 数据片选/字节同步
用户手册
1
2.使用说明
VS1053 MP3 模块用户手册
高性能音频编解码模块
2.1 模块简介
VS1053 MP3 模块一款高性能音频编解码模块，该模块接口丰富、功能完善，仅需提供电源（3.3V/5.0V），即可通过单片机（8/16/32 位单片机均可）控制模块实现音乐播放，或者录音等功能 -VS1053 MP3 模块不但外观漂亮，而且功能齐全、接口丰富，模块尺寸为 34mm*52.6mm，并带有安装孔位，非常小巧，并且利于安装，可方便应用于各

Nucleus系统的移动终端录音功能设计

N ucl e us系统的移动终端录音功能设计3■西安邮电学院田磊　摘　要针对Nucleus系统的移动设备硬件平台,分析了自适应多速率(AMR)编解码算法的工作原理,提出了基于AMR语音压缩算法的语音录音功能的设计方案,重点研究了手机语音多媒体软件设计。

通过交叉编译环境,对软件进行了调试,运行良好。

　关键词录音　多媒体手机　软件设计　AMR　Nucleus引　言随着移动通信的不断发展,多媒体技术在手机中的应用也越来越广。

手机的核心技术主要在硬件和软件两个方面。

硬件是核心层,包括基带模块和射频模块两部分。

软件分为底层软件和应用层软件两大部分。

1　手机平台体系结构研究(1)硬件系统结构硬件平台主要包括射频模块、基带模块、电源管理模块和外围设备。

文中采用了数字基带、模拟基带和电源管理模块集于单芯片的基带解决方案,这种解决方案具有较高的集成度,不仅增强了系统的稳定性,降低了功耗,同时有效地降低了设计过程中的复杂度。

(2)软件系统结构软件系统设计主要包括嵌入式实时操作系统Nucleus 3基金项目:西安邮电学院青年教师基金项目(10120430)。

的移植、应用程序的编写以及交叉编译出相应的可执行文件。

Nucleus系统提供TCP/IP网络、图像界面Grafix、文件系统File等模块。

整个手机软件主要由GSM协议栈、嵌入式Nucleus 操作系统、设备驱动和人机界面组成。

2　自适应多速率语音编码算法分析2.1　编码原理AMR语音编码根据其实现的功能可分为L PC分析、基音搜索、代数码本搜索三部分。

其中L PC分析完成的主要功能是获得10阶L PC滤波器的10个系数,并对转化成的线谱参数L SF进行量化;基音搜索用以获得基音延迟和基音增益这两个参数;代数码本搜索则是为了获得索引和代数码本增益及码本增益的量化。

AMR编码器的信号流程如图1所示。

在编码之前,使用高通滤波函数和幅度调整函数进行预处理。

在定点运算时,将输入样本值除以2来减少溢出结　语本文设计的视频检测运动控制卡,可以在三维空间内准确定位目标物体,同时采用动态跟踪的方式对目标物体进行夹取。

VS1053b音频解码器编码器

VS1053b是单片Ogg Vorbis/MP3/AAC/WMA/MIDI音频解码器，及IMA ADPCM编码器和用户加载的Ogg Vorbis编码器。

它包含了一个高性能、有专利的低功耗DSP处理器内核VS_DSP4、工作数据存储器、供用户应用程序和任何固化解码器一起运行的16KiB 指令RAM及0.5KiB多的数据RAM、串行的控制和输入数据接口、最多8个可用的通用I/O引脚、一个UART、并有一个优质的可变采样率立体声ADC（“咪”、“线路”、“线路+咪”或“线路*2”）和立体声DAC、和跟随的一个耳机功放及一个公共电压缓冲器。

特性●Ogg Vorbis解码；MPEG1&2音频阶层III(CBR+VBR+ABR)；阶层I和II可选；MPEG4/2AAC‐LC(+PNS)，HE‐AAC V2(级别3)(SBR+PS)；WMA4.0/4.1/7/8/9所有特性注1（profiles）(5‐384kbps)；WAV(PCM+IMA ADPCM)；通用MIDI1/SP‐MIDI格式0的文件●用软件插件进行Ogg Vorbis编码（2007第四季可用）●“咪/线路”的输入信号可实现IMA ADPCM编码(立体声)●支持MP3和WAV的数据流●EarSpeaker空间效果注2处理●低音和高音控制●只用一个单独的12..13MHz时钟运作●也可以使用一个24..26MHz时钟运作●内建PLL时钟乘法器●低功耗运作●芯片内建高质量和通道间无相位误差的立体声DAC●过零交叉注3（Zero‐cross）侦测和平滑的音量调整●立体声耳机驱动器可以驱动一个30Ω的负载●安静的电源通断功能●可扩展外部DAC的I2S接口●分离的模拟、数字、IO供电电源●供用户代码和数据使用的片内RAM●用于控制和数据的串行接口●可以作为从模式的辅助处理器使用●特殊应用可使用SPI FLASH存储器引导●可用于调试的UART接口●可用软件增加新功能和提供最多8个GPIO●符合RoHS无铅标准的封装（绿色环保）。

VS1053B编解码器的VDR分布式声音采集卡设计

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ＴＣＰ／ＩＰ协议栈的嵌入式芯片，降低了系统的功耗和软件设计的复杂性，增强了系统的灵活性和可靠性。

VS1053 PDF

Public DocumentVS1053b Ogg Vorbis EncoderVSMPG“VLSI Solution Audio Decoder”Project Code:Project Name:VSMPGRevision HistoryRev.Date Author Description1.60c2010-05-10HH Enhanced this document,removed XP/Vista demo.1.60b2010-05-04HH Corrected formatting error in this document.1.602010-03-30HH Enhanced sound quality,added stereo VU meter. 1.502009-07-02HH Added byte alignment,mono channel selection.1.402009-03-24HH SCI CLOCKF bugﬁx.1.312009-02-26HH Added Chapter5.2.1.302009-01-15HH Added sample counter.1.212008-11-21HH Corrected contents for SCI CLOCKF.1.202008-11-20HH Many new features,see Chapter6.1.102008-05-22HH Improved quality,many new proﬁles.1.012007-12-21HH Added XP/Vista demo binary.1.002007-12-03HH Initial version.Table of Contents1Introduction4 2Using the Ogg Vorbis Encoder52.1Limitations and Requirements (5)2.2Ogg Vorbis Encoder Proﬁles (5)2.2.1Proﬁle Groups (5)2.2.2Proﬁle Quality Settings (6)2.3Running the VS1053b Ogg Vorbis Encoder (8)2.3.1VS1053b Ogg Vorbis Encoder Registers (8)2.3.2Loading and Starting the Code (9)2.3.3Reading Ogg Vorbis Data (10)2.3.4Reading Additional Data while Recording Ogg Vorbis (11)2.3.5Finishing Ogg Vorbis Recording (11)2.4Recording Levels and Automatic Gain Control(AGC) (12)2.4.1Reading the Recording Level (12)2.4.2Setting AGC (13)2.4.3Building a Useful VU Meter (14)2.4.4Converting from Linear to Decibel Scale (16)2.5Samplerate Considerations (17)2.6Post-Processing the Recording with VorbisGain (17)3The Ogg Vorbis Format183.1Introduction to Ogg Vorbis (18)3.1.1Variable Bit-Rate (19)3.1.2Transcoding (19)3.2Compressing into Ogg Vorbis with a PC (20)3.2.1Preparation (20)3.2.2Compressing CD-Quality Music (20)3.2.3Compressing Speech (21)3.3Post-Processing with VorbisGain (22)4How to Load a Plugin23 5Building a Streaming System245.1Sample Counter (25)5.2Adjusting VS1053b Playback Samplerate (26)6Latest Version Changes27 7Contact Information291.INTRODUCTION 1IntroductionThe VS1053b is a unique device in the VLSI Solution’s VS10XX audio codec family in that it allows for the user to encodeﬁles into the highly eﬃcient Ogg Vorbis format.This makes it possible for the user to build a device that can record high-quality stereo sound while retaining moderateﬁle sizes.This document is an instruction manual on how to use the VS1053b Ogg Vorbis encoder application as well as an introduction to the Ogg Vorbis format itself.Chapter2describes how to load and run diﬀerent Ogg Vorbis proﬁles on a VS1053b.It also discusses recording levels and recording level meters.Chapter3brieﬂy introduces the Ogg Vorbis format.It also has some suggestions on how to encode data to Ogg Vorbis format as eﬃciently as possible on a PC.If you are unfamiliar with the Ogg Vorbis format,please read this chapterﬁrst.Chapter4tells how to load a plugin code to a VS10XX chip.Chapter5describes how a streaming VS1053b system can be built.The document version history is provided in Chapter6.Finally,Chapter7contains VLSI Solution’s contact information.2Using the Ogg Vorbis EncoderThe VS1053b Ogg Vorbis Encoder application is provided as a plugin in the same package as this document,downloadable athttp://www.vlsi.ﬁ/en/support/software/vs10xxapplications.html.Before loading the application into your VS1053b,select one of the proﬁles shown below.2.1Limitations and Requirements•Maximum SPI(SCI)clock speed is3.5Mbit/s.If a higher speed is used,there may be occasional data read errors.2.2Ogg Vorbis Encoder ProﬁlesThere are45diﬀerent Ogg Vorbis proﬁles available.They are divided intoﬁve groups, depending on their samplerate and number of channels.2.2.1Proﬁle GroupsTheﬁve proﬁle groups are presented below.As a comparison,bitrateﬁgures for IMA ADPCM and16-PCM(WAV)are also shown.Ogg Vorbis ProﬁlesProﬁle File name1SRate Ch BRat2W A V3IMA4Time name Hz kbit/s kbit/s kbit/s h/GB Voice venc08k1qXX.plg800011512832149 Wideband Voice venc16k1qXX.plg160001282566579 Wideband Stereo Voice venc16k2qXX.plg1600024951213045HiFi Voice venc44k1qXX.plg4410018770617926 Stereo Music venc44k2qXX.plg4410021351411358161Replace XX with a quality value between00and10.See Chapter2.2.2for details.2Estimate for quality value05.Actual bitrate depends on content being recorded.3Comparison bitrate for a16-bit WAVﬁle with the same speciﬁcations.4Comparison bitrate for a4-bit IMA ADPCMﬁle with the same speciﬁcations.2.2.2Proﬁle Quality SettingsEach of the low-sample-rate proﬁle groups has eleven quality proﬁles,numbered from00 to10.The high-sample-rate proﬁle groups have six quality proﬁles from00to05.This chapter presents theﬁve proﬁle groups and gives estimates of typical bitrates that can be obtained with the proﬁle.In general,quality setting05is designed to be a typical value that gives good quality for the application.Qualities00and01are for emergency use only when there just isn’t storage space for better quality settings.High qualities,starting from07or08, are intended for transparent sound quality,the intent being that compression cannot be heard anymore even under stringent circumstances.The“Voice”ProﬁlesThe“Voice”proﬁles are intended for speech applications.VoiceProﬁle number0001020304050607080910Typical kbit/s68101113152024273033The“Wideband Voice”Proﬁles“Wideband Voice”is intended to be used when high speech quality is required.Wideband VoiceProﬁle number0001020304050607080910Typical kbit/s610141823283745525863The“Wideband Stereo Voice”Proﬁles“Wideband Stereo Voice”is intended to be used when high speech quality with directional information is required.Wideband Stereo VoiceProﬁle number0001020304050607080910Typical kbit/s917254441496684101119136The“HiFi Voice”ProﬁlesWhen extremely high quality speech is required,use the“HiFi Voice”proﬁles.HiFi VoiceProﬁle number000102030405Typical kbit/s364959717987The“Music”ProﬁlesThe“Music”proﬁles are intended for HiFi music,and are capable of oﬀering very high-quality stereo sound.MusicProﬁle number000102030405Typical kbit/s446380991171352.3Running the VS1053b Ogg Vorbis Encoder2.3.1VS1053b Ogg Vorbis Encoder RegistersRegister Bits DescriptionSCI MODE14Select MIC/LINE112Set to1when told in the instructions1VU meter stereo mode activationSCI AICTRL015:0Maximum signal level,set to0SCI AICTRL115:0Recording gain(1024=1×)or0for automatic gain controlSCI AICTRL215:0Maximum autogain ampliﬁcation(1024=1×,65535=64×)SCI AICTRL30W:Finish recording,set to01R:Recordingﬁnished,set to02R:There is at least one byte to read,set to03W:Input channel select(only mono proﬁles),0=left,1=right7:4W:Max samples in frame=n×4096,0=no limit15:8R:The next data byte if available,set to0Before activating Ogg Vorbis recording,the user must initalize registers SCI AICTRL0 -SCI AICTRL3.SCI AICTRL1and SCI AICTRL2can be altered during recording.SCI AICTRL0records the maximum absolute value of the signal.The maximum value of this linear register is0x7FFF in mono mode,or0x7F7F in stereo mode.For more information on how to use this register,see Chapter2.4.1.SCI AICTRL0is updated once for every Vorbis block,i.e.10...170times per second with the current proﬁles.SCI AICTRL1controls linear recording gain.1024is equal to digital gain1(recom-mended for best quality),2048is equal to digital gain2,and so on.If the user wants to use automatic gain control(AGC),SCI AICTRL1should be set to0.Speech applica-tions are often better oﬀusing some AGC,as this helps to get relatively uniform speech loudness in recordings.SCI AICTRL2controls the maximum AGC gain if SCI AICTRL1is set to0.This limits ampliﬁcation of noise when there is no signal.For more information on recording levels, see Chapter2.4.SCI AICTRL3oﬀers run-time controls,channel selection for mono input,the maximum number of samples that are allowed in one Ogg frame with4096sample granularity,and the next unread byte,if any.Normally0(no limit)is a good value for granularity,but streaming applications often have maximum delay considerations,thus making limiting necessary.Note that limiting frame size isn’t entirely accurate,so always select a maxi-mum frame size slightly lower than your absolute upper delay limit.Example:your samplerate is44100Hz and you want frame output at least every500ms. Then you’ll need samples at22100sample intervals.Round this downwards to the closest 4096multiple,and you’ll get16384samples,so you’ll write0x40to SCI AICTRL3.The bitrate penalty of this example is≈0.5kbit/s.For this to work your reads must be byte aligned as explained in Chapter2.3.3.2.3.2Loading and Starting the CodeTo load and start the VS1053b Ogg Vorbis Encoder,do the following steps:1.If you have been doing ADPCM recording,clear bit SM ADPCM from registerSCI MODE(0).2.Set the VS1053b clock to the highest value just below55.3MHz.Example:if theexternal clock is12.288MHz,this can be done by setting the clock to4.5X,or55.296MHz by writing0xC000to register SCI CLOCKF(3).If you use anotherinput clock,adjust SCI CLOCKF accordingly.Examples:Value for12MHz is0xC3e8,12.288MHz is0xC000,13MHz is0xA4E2.3.Set SCI BASS(2)to0.4.Disable any potential user application by setting SCI AIADDR(10)to0.5.Disable all interrupts except the SCI interrupt by writing0x2to VS1053’s in-ternal register VS1053INT ENABLE.This is done byﬁrst writing0xC01A to SCI WRAMADDR(7),then0x2to SCI WRAM(6).6.Load the plugin proﬁle you intend to run(Chapter4).Note that you will have toredo this each time you want to activate recording.7.Set bit SM ADPCM(12)in register SCI MODE(0)to1.At the same time,youcan also select LINE input instead of MIC input by setting bit SM LINE1(14)to1.If you want to use the VU meter in stereo mode,set also bit SM LAYER12(1)to1.Do not set SM RESET(2)at the same time!8.Set recording level control registers SCI AICTRL1(13)and SCI AICTRL2(14).Typical good values for conservative AGC are0and4096,respectively.For a HiFi application,recommended values are1024and0,respectively.9.If you want to use a VU meter,write0x8000to SCI AICTRL0(12)if using a monoproﬁle or0x8080is using a stereo proﬁle.10.Set a proper value(often0)to SCI AICTRL3(15).11.Activate the encoder by writing0x34to register SCI AIADDR(10).12.Wait until DREQ pin is high before reading any data.2.3.3Reading Ogg Vorbis DataAfter Ogg Vorbis recording has been activated,registers SCI HDAT0and SCI HDAT1 have new functions.The Ogg Vorbis bitstream buﬀer size is409616-bit words,or8KiB.Theﬁll status of the buﬀer can be read from SCI HDAT1.If SCI HDAT1is greater than0,you can read that many16-bit words from SCI HDAT0.If data is not read fast enough from SCI HDAT0,the buﬀer overﬂows and returns to empty state.A data overﬂow will result in an incorrectﬁle that may be undecodable. However,because of the large size of the bitstream buﬀer and the relatively slow bitrates of Ogg Vorbis,this situation should be easy to avoid.Ogg frames are byte aligned.Because of the16-bit SCI interface this may be a problem for streaming applications where it is important to forward any Ogg frame as soon as possible:the latest frame may pend upon its last byte.If transferring data with a low delay is important,you can read whether there is one extra data byte by reading SCI AICTRL3twice in a row.The result of the latter read will contain the correct data: if bit2is1,then bits15:8contain the odd byte.After reading the odd byte successfully, discard bits15:8of the next read from SCI HDAT0because they contain the same data. (NOTE:if you don’t stream or don’t need the shortest possible delay,you can ignore this paragraph!)If you are having trouble with receiving data,notice that all Ogg Vorbisﬁles always begin with the following4bytes:0x4f0x670x670x53(the string“OggS”).If you get0x67 0x4f0x530x67(“gOSg”)instead,you are storing the least and most signiﬁcant byte in the16-bit data words incorrectly.If you get0x000x000x670x53,you have read data from SCI HDAT0too soon after starting the application.2.3.4Reading Additional Data while Recording Ogg VorbisYou can get extra side information while recording Ogg Vorbis data to see whether VS1053b is still working.The following VS1053b X memory addresses may be read for extra data:X Memory Address Description0x816LSb’s of recording time(seconds)0x916MSb’s of recording time(seconds)0xC16LSb’s of average bitrate(bits/s)0xD16MSb’s of average bitrate(bits/s)0x180016LSb’s of sample counter0x180116MSb’s of sample counterTo read the average bitrate,do the following.First write0xC to SCI WRAMADDR(7). Then read from SCI WRAM(6).This is16least signiﬁcant bits of the bitrate.Then read the16more signiﬁcant bits by reading SCI WRAM again.You can read the recording time as a sanity check that VS1053b is working:if the register contents don’t change every second,you’ll have to take protective measures.2.3.5Finishing Ogg Vorbis RecordingTo create fully compliant Ogg Vorbisﬁles,Ogg Vorbis bitstreams need to be shut down properly.The following algorithm can be used to implement this:1.Set bit0of SCI AICTRL3(15)to1.2.Continue reading data through SCI HDAT0and SCI HDAT1as usual,but checkSCI AICTRL3’s bit1from time to time.When this bit turns to1,the Ogg Vorbis encoder hasﬁnished writing to the buﬀer.3.Write the remaining words from the bitstream buﬀer as normal using SCI HDAT0and SCI HDAT1.4.Read SCI AICTRL3twice(changed from version1.4!)and check whether bit2is1in the latter read.If it is,then don’t write to theﬁle the last byte(bits7:0)of the last word that have come through SCI HDAT0.5.Reset VS1053b to normal state using software reset.Remember to clear registerSCI MODE(0)bit SM ADPCM(12)if you don’t wish to start ADPCM recording.Remember to also set bit SM LAYER12(1)to an appropriate value.6.If you want to restart recording,you have to completely reload and restart therecording application.2.4Recording Levels and Automatic Gain Control(AGC) The VS1053b Ogg Vorbis Encoder oﬀers signal level monitoring through SCI AICTRL0. It is recommended that devices that oﬀer recording would show a signal level in a decibel scale.For the decibel scale,see Chapter2.4.4.This can be done by showing and clearing SCI AICTRL0contents at regular intervals.A good VU meter should be implemented in such a way that it visually advices the user to avoid using the last6dB of the available dynamic range.2.4.1Reading the Recording LevelThe recording level meter has two settings:mono and stereo.You can use mono mode with both mono and stereo streams,but you should use the stereo mode only with stereo proﬁles.The new meter is compatible with versions pre1.60if SCI MODE bit SM LAYER12is clearedﬁrst.Recording Level Meter:Mono ModeActivate mono mode by clearing SCI MODE(0)register bit SM LAYER12(1)to0.To read the level,repeat the following loop:•Write0x8000to SCI AICTRL0.•Wait for at least1/50s.Note:Omitting this step will not cause a malfunction, but constant reading of side information causes load on the VS1053and may cause unexpected crackles in sound.•Check whether SCI AICTRL0&0x8000is0.If not,wait a little more and read again.•Use the value as explained in this Chapter.Repeat the loop.Recording Level Meter:Stereo ModeActivate stereo mode by setting SCI MODE(0)register bit SM LAYER12(1)to1.To read the left and right channel levels,repeat the following loop:•Write0x8080to SCI AICTRL0.•Wait for at least1/50s.Note:Omitting this step will not cause a malfunction, but constant reading of side information causes load on the VS1053and may cause unexpected crackles in sound.•Check whether SCI AICTRL0&0x8080is0.If not,wait a little more and read again.Note:wait until both bits have cleared!•Left channel value is SCI AICTRL0&0x7F00.•Right channel value is(SCI AICTRL0&0x7F)×256.•Use the values as explained in this Chapter.Repeat the loop.2.4.2Setting AGCWhen the highest dynamic range and soundﬁdelity is required,AGC should be turned oﬀand recording gain should be set to1(SCI AICTRL1=1024).A good example of this would be music recording.This is particularly recommended for the“HiFi Voice”and“Music”proﬁles,although there might be cases where recording level control would be needed even with these cases.However,in some cases it is required that the audio dynamic range is compressed.An example of such a case is when a device should retain a uniform recording level of a discussion of several people or of one person moving closer and further from the recording device.In such a case,it is a good idea to use AGC.The AGC unit adjusts signal power in such a way that the maximum sample value from a sine wave would become as close to16300as possible.If the signal is too strong, recording level is decreased,and vice versa.The maximum recording level can be set with register SCI AICTRL2.When AGC is used,conservetive maximum gain values often give the best sounding results.Example:12dB(SCI AICTRL1=0,SCI AICTRL2=4096).In some cases more extreme values may help to make quiet speech more intelligible,but such values may also add excessive background noise and make sound quality less pleasing.2.4.3Building a Useful VU MeterIn an encoder application,if the recording level is too low,extraneous background noise may be introduced to the sound.Conversely,if the recording level is so high that the highest values cannot be represented numerically,signal clipping occurs,and this may cause severe distortion to sound.In a recording device,it is useful to have a VU meter that shows the signal level so that both too low signal levels and clipping is avoided.This is very important so that the user has a chance to either adjust the recording or input signallevel.−40 dB = 161−30 dB = 512−20 dB = 1625−12 dB = 4096−6 dB = 81920 dB = 16384+5 dB = 29193OVERLOAD = 29193OVERLOADFigure2.1:Example Colour VUMeter.OVERLOADFigure2.2:Example Monochrome VU Meter.Figures2.1and 2.2show example VU meters.The0dB reference point has been set to signal level16384,which is one half of the maximum amplitude,leaving a6dB headroom for the device.The lowest signal level shown is a choice that can vary a lot depending on the application. While-40dB is a high-ﬁdelity favourite,a VU meter will work just as well with a lower limit of-20dB.If display space is scarce,low limit could even be set to-12dB.Between-6and+6dB the VU meter precision should preferably be1dB,and at most 2dB.Below-12dB step size can be several decibels.The important thing in a VU meter is to visually show the user that it is not recommended to regularly use the highest6dB of the recording scale,and that an overload situation (≥+5dB,or over linear value29193)is an error condition.To help this the OVERLOAD symbol should be kept lit or blinking for at least2seconds each time an overload situation occurs.Many of these cues are often ignored in digital VU meters,and partly as a result of this even professional recordings are all too often made at recording levels that destroy signal integrity.Figure2.3:Real Record Display with Monochrome VU Meter.Figure2.3shows a real implementation of a recording display that uses a Monochrome VU rmation shown on the top line includes recording time,free disc space, overload warning and a bitrate counter.All this data except from the free disc space number have been obtained from the VS1053b Ogg Vorbis Encoder.Recording gain and AGC can be set at the center of the screen if a proﬁle is used that support these functions.However,they are greyed out in this picture because they are not available in the Stereo Music proﬁle that has been used.The VU Meter is at the bottom.The solid line is the current recording level as read from SCI AICTRL0and converted to Decibel scale as shown in Chapter2.4.4.The greyed line is the top level of the last3seconds.If the greyed line≥+5dB,the orange OVERLOAD message blinks twice a second for three seconds.2.4.4Converting from Linear to Decibel ScaleTo convert from linear to dB scale on architectures where multiplications and logarithms are expensive operations,the following code can be used:const unsigned short linToDBTab[5]={36781,41285,46341,52016,58386}; /*Converts a linear16-bit value between0..65535to decibels.Reference level:32768=96dB(largest VS1053b number is32767=95dB). Bugs:-For the input of0,0dB is returned,because minus infinity cannotbe represented with integers.-Assumes a ratio of2is6dB,when it actually is approx.6.02dB.*/unsigned short LinToDB(unsigned short n){int res=96,i;if(!n)/*No signal should return minus infinity*/ return0;while(n<32768U){/*Amplify weak signals*/res-=6;n<<=1;}for(i=0;i<5;i++)/*Find exact scale*/if(n>linToDBTab[i])res++;return res;}2.5Samplerate ConsiderationsThe Ogg Vorbis encoder cannot always work at exactly the right samplerate.To be exact, the samplerate is accurate only when the input clock is12.288MHz and the nominal samplerate of a proﬁle is either8000Hz or16000Hz.In all other cases the samplerate can be calculated as follows.First let:•f i=input clock divided by4000(same value that you write to SC FREQ(bits10:0) of register SCI CLOCKF,or3072if you have written0to these bits to indicate12.288MHz clock).•f n=nominal samplerate of the proﬁle(8000,16000or44100).•r=3if f n=8000,otherwise r=2.•Now let u=f n×r.•Let d= f i×2000+u2u•Now our real samplerate f s=2000×f id×rExample:We are running at an input clock of12.288MHz and we are running the Stereo Music proﬁle that has a nominal samplerate of44100Hz.First f i=122880004000=3072.Because f n=44100,then r=2. Thus u=44100×2=88200.This leads to d= 3072×2000+88200288200 =70And,ﬁnally,our real samplerate f s=2000×307270×2≈43885.7(Hz)As can be seen,in this example the nominal and real samplerate have a diﬀerence of 0.5%.While this is normally not signiﬁcant,it is good to know and important in the case of streaming.2.6Post-Processing the Recording with VorbisGainIt is recommended to post-process your own recording later on a PC machine using VorbisGain to get equal loudness with other recordings.For details,see Chapter3.3.3The Ogg Vorbis FormatThis chapter gives a brief introduction to the Ogg Vorbis format and presents recommen-dations for basic parameters when transcoding MP3ﬁles or compressing CD content into Ogg Vorbisﬁles on a PC.The intent is to give VLSI Solution’s customers information on how to create high-quality Ogg Vorbis content.Before using the information in this document,you should get the free Ogg Vorbis encoder “oggenc”,minimum version1.0.2(2005),or a version with aoTuV tunings.While the instructions will work with older versions,sound quality will not be as good.Download oggenc at /.It is also strongly encouraged to get VorbisGain,a program that tagsﬁles with loudness information(Chapter3.3).For this example,vorbisgain v0.36has been used.Download VorbisGain at /vorbisgain.html.3.1Introduction to Ogg VorbisOgg Vorbis is a license-free audio codec that allows for high-quality sound around a large gamut of bit-rates,ranging from roughly10to500kbit/s.With a given bit-rate,Vorbis is capable of surpassing the quality of MP2,MP3,WMA and AAC LC,particularly in low-bit-rate applications.An Ogg Vorbisﬁle is a Vorbis compressedﬁle that has been encapsulated inside an Ogg stream.The defaultﬁle extension for Ogg Vorbisﬁles is“.ogg”.An Ogg Vorbisﬁle consists of two major parts:header and actual data.Every Ogg Vorbisﬁle begins with a header which contains compression tables that the decoder needs to correctly decodeﬁrst.After the header has been fully received,Vorbis audio may be decoded.The size of the header information is around4KiB for most encoders. (However,for VLSI Solution’s VS1053b encoder,header size is only roughly1.5KiB.)After the header has been decoded,full random access is available in an Ogg Vorbisﬁle. As opposed to many other formats,an Ogg Vorbisﬁle contains its own time code,so absolute position in aﬁle can be displayed after random access operations(fast forward /rewind).Ogg Vorbis is not particularly well suited for encoding very short,low bit-rate samples.E.g.a16kbit/s audio sample that lasts for one second would take2KiB for the actual audio data,so the default4KiB header triples theﬁle size.However,for normal appli-cations where audio lasts for more than10seconds,this header overhead is negligible.3.1.1Variable Bit-RateFor a long time almost all MP3ﬁles had a constant bit-rate,typically128kbit/s.While constant bit-rate(CBR)makes it easy to calculate a relation betweenﬁle size/position and song length,and while it makes it easier to streamﬁles,CBR is not optimal for quality.When there is a diﬃcult passage in music,there may not be enough bits available to encode audioﬂawlessly.Conversely,in passages that are easier to encode there may be bits left unused because the encoder didn’t need all the bits available in the bitstream.Where CBR tries to keep the bit-rate constant regardless of content,Variable Bit-Rate (VBR)tries to keep quality constant.This will have the eﬀect of changing the bit-rate of theﬁle depending on how diﬃcult a particular portion of music is to encode.Because sound quality is always kept constant,VBR has a bit advantage to CBR.Thus,VBR allows either for smallerﬁles with the same perceived sound quality or equally sizedﬁles with better perceived sound quality.Ogg Vorbis is by nature a variable bit-rate format and works best when this feature is not limited.Depending on the material the bit-rate can have signiﬁcantﬂuctuations: e.g.songs that have been compressed with a quality setting with a nominal bit-rate of 96kbit/s can easily have average bit-rates between80to105kbit/s,or even more with extreme material.3.1.2TranscodingTranscoding is the process of converting from one lossy format to another,like from MP3 to Ogg Vorbis.It is important to understand that although Ogg Vorbis is a better format than MP3, transcoding never enhances sound quality.What has been lost in the initial MP3com-pression stage,will stay lost when converting to Ogg Vorbis,and there will always be some additional quality loss.While the extra quality loss can be small enough that it may be impossible to tell,the end result still is never better than the originalﬁle was. If possible,it it recommended that users should avoid transcoding to avoid generation loss.If transcoding is done(e.g.to get smallerﬁles for a portable device),it is a good idea to keep the originalﬁles.For best results,compress Ogg Vorbisﬁles from uncompressed sources,like CDs.3.2Compressing into Ogg Vorbis with a PCThe whole process of getting Ogg Vorbisﬁles looks roughly like this:•Preparation:Decode an MP3/WMA/AAC etcﬁle or preferably rip a CD to WAV format and make sure that you have“oggenc”(Chapter3.2.1).•Compression to Ogg Vorbis(Chapters3.2.2and3.2.3).•Optional post-processing with VorbisGain(Chapter3.3).3.2.1PreparationIf you intend to transcode(=recompress)MP3ﬁles,youﬁrst have to convert them to 16-bit RIFF WAV format(usually known for a“.wav”ﬁle name extension).However, before you do that,please have a look at Chapter3.1.2for details on transcoding.If you encode CDs,you have to make sure they are read initially into WAV format.It is recommended to use CD sources for compression if possible.When you have audio in WAV format,you can use oggenc to compressﬁles,either in stereo or mono.Before continuing to the next stage,make sure you have the free Ogg Vorbis encoder “oggenc”,minimum version1.0.2(2005).3.2.2Compressing CD-Quality MusicIf we assume the inputﬁle is calledﬁle.wav and the output is to be calledﬁle.ogg,the following parameters may be used.It is assumed that the input audio is sampled at the CD44.1kHz sample rate.While these commands will also work for other samplerates,quality andﬁle sizes may diﬀer.For each quality setting,the table shows the command needed to convert theﬁle,a typical bit-rate for that quality,how many hours of music could be stored to a1gibibyte memory card,and a short description of the mode.For comparison purposes,also an uncompressed CD bitstream is shown in the table.CD music compressionCommand Typ.Hours/Descriptionkbit/s1GiBoggenc-q-1ﬁle.wav4553Smallestﬁles size,better than96kbit/s MP3 oggenc-q0ﬁle.wav6437Ok quality,equals112kbit/s MP3 oggenc-q2ﬁle.wav9625Better than most128kbit/s MP3ﬁles oggenc-q4ﬁle.wav12819Often CD transparentoggenc-q6ﬁle.wav19212CD quality almost alwaysCD1411 1.7Uncompressed CDGraphical front-ends usually contain one or several CBR bit-rate slider(s)and a quality slider that usually goes from-1to10.CBR sliders should be deactivated,and the quality slider should be set to the“-q”value presented in the table.。

大华医疗行业安防系统解决方案

大华医疗行业安防系统解决方案浙江大华技术股份有限公司行业事业部/文教卫事业部目录第一章. 行业背景 (5)第二章. 需求分析 (7)第三章. 设计原则及依据 (9)3.1 设计原则 (9)3.2 设计依据 (10)第四章. 总体设计 (12)4.1 方案框图 (12)4.1.1 医疗单位方案框图 (12)4.1.2 远程管理中心方案框图 (13)4.2 大华医疗安防整体解决方案子系统设计 (13)第五章. 详细设计 (16)5.1 监控中心功能与设计 (16)5.1.1 设计需求 (16)5.1.2 系统设计 (16)5.1.3 系统特点 (18)5.2 视频监控子系统功能与设计 (18)5.2.1 设计需求 (19)5.2.2 系统设计 (19)5.2.3 系统特点 (25)5.3 智能停车场子系统功能与设计 (25)5.3.1 设计需求 (25)5.3.2 系统设计 (25)5.3.3 系统特点 (28)5.4 门禁子系统功能与设计 (29)5.4.1 设计需求 (29)5.4.2 系统设计 (29)5.4.3 系统特点 (32)5.5 报警管理子系统功能与设计 (32)5.5.2 系统设计 (33)5.5.3 系统特点 (34)5.6 远程会诊子系统功能与设计 (35)5.6.1 设计需求 (35)5.6.2 系统设计 (35)5.6.3 系统特点 (36)5.7 远程探视子系统功能与设计 (36)5.7.1 设计需求 (36)5.7.2 系统设计 (37)5.7.3 系统特点 (38)5.8 智能分析子系统功能与设计 (38)5.8.1 设计需求 (38)5.8.2 系统设计 (39)5.8.3 系统特点 (42)5.9 大屏显示子系统功能与设计 (42)5.9.1 设计需求 (42)5.9.2 系统设计 (43)5.9.3 系统特点 (44)5.10 电子巡更子系统功能与设计 (44)5.10.1 设计需求 (44)5.10.2 系统设计 (45)5.10.3 系统特点 (46)5.11 传输网络系统功能与设计 (46)5.11.1 设计需求 (46)5.11.2 系统需求 (47)5.11.3 系统特点 (48)5.12 存储系统功能与设计 (48)5.12.1 设计需求 (48)5.12.2 系统设计 (49)5.13 远程管理中心功能与设计 (50)5.13.1 设计需求 (50)5.13.2 系统设计 (50)5.13.3 系统特点 (51)第六章. 大华医疗安防整体解决方案亮点 (52)6.1 智能视频分析应用 (52)6.2 智能运维 (54)6.3 可视化医疗废弃物管理 (55)6.4 解决车难管、车难停问题 (56)6.5 协助医疗单位提升业务能力 (57)6.6 提升医疗单位品牌价值 (57)第七章. 产品推荐 (58)7.1 前端设备 (58)7.2 存储设备 (67)7.3 智能分析服务器 (71)7.4 软件平台 (72)7.5 大屏显示 (73)7.6 出入口管理系统 (74)第八章. 部分案例 (82)第一章.行业背景随着社会经济的高速发展，我国逐年加大对医疗体制改革和建设的投入，医疗行业发展迅猛，医疗单位数量和规模也在不断扩大，中国医疗事业正迎来蓬勃发展的机遇期。

VS1053B中文资料应用

4.2 推荐工作值 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
4.3 模拟电路特性 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
• EarSpeaker 空间效果注 2 处理
• 低音和高音控制 • 只用一个单独的 12..13MHz 时钟运作 • 也可以使用一个 24..26MHz 时钟运作 • 内建 PLL 时钟乘法器 • 低功耗运作 • 芯片内建高质量和通道间无相位误差的立体声 DAC • 过零交叉注 3（Zero‐cross）侦测和平滑的音量调整 • 立体声耳机驱动器可以驱动一个 30Ω 的负载 • 安静的电源通断功能 • 可扩展外部 DAC 的 I2S 接口 • 分离的模拟、数字、IO 供电电源 • 供用户代码和数据使用的片内 RAM • 用于控制和数据的串行接口 • 可以作为从模式的辅助处理器使用 • 特殊应用可使用 SPI FLASH 存储器引导 • 可用于调试的 UART 接口 • 可用软件增加新功能和提供最多 8 个 GPIO • 符合 RoHS 无铅标准的封装（绿色）
7.7.1 两个 SCI 写操作 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24 7.7.2 两个 SDI 字节 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24 7.7.3 在两个 SDI 字节之间的 SCI 操作 . . . . . . . . . . . . . . . . . . . . . . 25

MP3音乐播放器芯片VS1053B芯片使用教程

Macro_Set_CLK_Low();
if(ucMaskCode & ucSendData) {
Macro_Set_SI_High(); } else {
Macro_Set_SI_Low(); }
Macro_Set_CLK_High(); ucMaskCode >>= 1;
} }
/******************************************************************************
MP3 音乐播放器芯片 VS1053B 芯片使用教程
MP3，MP4，数码相机曾经是奢侈品。科技发展了，这些东西也就成了普通手机的附属品了。但是作为手艺人学习这些芯片了解这些高性能芯片是很有必要的。学习了解它既拓展了自己的眼界，也可以提升自己对嵌入式系统的认识。原子战舰板上就板载了一颗高性能 MP3 音乐播放器芯片 VS1053B 芯片。通过原子源码对一些参数的修改，也能欣赏美妙的音乐。这是一款真正的数字产品。还可以作为录音机使用。
VS1053B，LCD，RDA5802 收音机芯片，ADXL345 重力加速度传感器芯片等等这些芯片，它们有一个共同点：寄存器，寄存器地址，数据帧，命令字，I2C, SPI 协议等等。学会这些外设芯片的驱动程序编写方法，才能真正发挥 ARM 处理器的强大功能。下面是单片机程序的主要部分预览： /* * MP3 模块测试程序 *
* 用途：MP3 模块测试程序
*
vs1003 的硬件测试程序，主控芯片为 STC12LE5A60S2
*
其他的微处理器(带 SPI 接口的)只需稍加修改即可适用
*
对于不带硬 SPI 接口的微处理器可以用 IO 进行 SPI 的时序模拟

一种基于LED可见光通信的无线音乐播放系统[实用新型专利]

专利名称：一种基于LED可见光通信的无线音乐播放系统专利类型：实用新型专利
发明人：肖纯贤,张淑敏,鲁金铭,薛晨桥,吴雯,上官质彬,郑胤申请号：CN201720347924.8
申请日：20170328
公开号：CN206564600U
公开日：
20171017
专利内容由知识产权出版社提供
摘要：本实用新型公开了一种基于LED可见光通信的无线音乐播放系统，包括音乐播放器和安装发送信号模块的壳体，所述音乐播放器中设置有接受电路：且所述接受电路连接在所述音乐播放器中的播放喇叭：所述壳体中安装有STM32微控制模块和VS1053音频解码模块；且所述STM32微控制模块和VS1053音频解码模块相互电连接；所述VS1053音频解码模块连接驱动电路。

该基于白光LED无线通信的音乐播放器系统，并研究其在红外遥控下的播放程序设计。

经过调试，该系统能流畅播放SD 卡中多种格式的音乐文件，同时正确显示歌曲名称、音量大小等至LCD屏。

音乐信息直线传输距离达到5米，LED照明灯光无明显闪烁，红外遥控响应速度快，满足实际照明和音乐播放的使用要求。

申请人：南开大学
地址：300000 天津市南开区卫津路94号
国籍：CN
更多信息请下载全文后查看。

基于视觉识别的生活垃圾分类装置设计

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机驱动器Ａ、ＢꎻＰ２.４、Ｐ２.５用于接收光电传感器及限位开
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图４控制系统框架图
分类储运机构如图３所示ꎬ主要是为了实现４类生活
垃圾的分类存储ꎮ 该机构由中间轴、支撑座、驱动电机、垃
圾桶、固定板、万向球、同步带轮等组成ꎮ 中间轴上固定有
支撑座、固定板和大同步带轮ꎬ４类垃圾桶放置于支撑座
上方并穿过固定板上的孔来限制径向运动ꎬ用以支撑分类
别的新型生活垃圾分类装置ꎮ 进行分类装置的机械结构设计ꎬ包括投放识别机构、分类储运机
构等部分ꎻ进行控制系统的设计ꎬ完成关键元器件的选型及各单元电路设计ꎻ基于ＭｏｂｉｌｅＮｅｔ可
分离卷积神经网络建立垃圾识别模型ꎬ构建数据集训练样本ꎻ完成分类装置的实验测试ꎮ 测试
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带轮传动实现垃圾桶的转动ꎬ此外ꎬ采用ＬＸＷ５－１１Ｎ１限
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VS1053B编解码器的VDR分布式声音采集卡设计

VS1053B编解码器的VDR分布式声音采集卡设计
刘毅;程明
【期刊名称】《单片机与嵌入式系统应用》
【年(卷),期】2013(13)11
【摘要】为了满足未来航行数据记录仪声音采集卡分布式、低成本、低功耗和高可靠性的要求,以STM32F103为核心控制器,结合VS1053B音频编解码芯片和
W5300以太网通信芯片设计了一种采用OggVorbis声音编码算法的分布式声音采集卡,给出了声音采集卡的软硬件设计.在硬件设计上,采用了独立的可加载OggVorbis声音编码算法以及内置TCP/IP协议栈的嵌入式芯片,降低了系统的功耗和软件设计的复杂性,增强了系统的灵活性和可靠性.
【总页数】4页(P50-53)
【作者】刘毅;程明
【作者单位】北京海兰信数据科技股份有限公司,北京100084;北京海兰信数据科技股份有限公司,北京100084
【正文语种】中文
【中图分类】TP368.1
【相关文献】
1.基于S3C2410型微处理器和UDA1341型立体声音频编解码器的嵌入式音频系统设计 [J], 高建华;王殊
2.基于LabVIEW的USB数据采集卡声音测量系统设计 [J], 刘晓峰;王毓顺;许中运;
于永芳
3.泛华恒兴发布专为声音和振动应用设计的高精度数据采集卡 [J],
4.泛华恒兴发布专为声音和振动应用设计的高精度数据采集卡 [J],
5.泛华恒兴发布专为声音和振动应用设计的高精度数据采集卡 [J],
因版权原因，仅展示原文概要，查看原文内容请购买。

VS1053_MP3模块用户手册.pdf

项目接口特性解码格式编码格式对外接口板载录音其他特性模块尺寸
项目 DAC 分辨率总谐波失真（THD）动态范围（A-加权）信噪比通道隔离度（串扰）咪头（MIC）放大增益咪头（MIC）总谐波失真咪头（MIC）信噪比 LINE IN 信号幅度 LINE IN 总谐波失真 LINE IN 信噪比 LINE IN 阻抗
用户手册 4
VS1053 MP3 模块用户手册
高性能音频编解码模块
主机必须在判断 DREQ 有效（高电平有效）后，才可以发送数据，每次可以发送 32 字节。这里我们重点介绍一下 SCI。SCI 串行总线命令接口包含了一个指令字节、一个地址字节
和一个 16 位的数据字。读写操作可以读写单个寄存器，在 SCK 的上升沿读出数据位，所以主机必须在下降沿刷新数据。SCI 的字节数据总是高位在前低位在后的。第一个字节指令字节，只有 2 个指令，也就是读和写，读指令为：0X03，写指令为：0X02。
图 2.3.3.2 SCI 写时序图 2.3.3.2 中，其时序和图 2.3.3.1 基本类似，都是先发指令，再发地址。不过写时序中，我们的指令是写指令（0X02），并且数据是通过 SI 写入 VS1053 的， SO 则一直维持低电平。另外，在图 2.3.3.1 和图 2.3.3.2 中，DREQ 信号上都产生了一个短暂的低脉冲，也就是执行时间。这个不难理解，我们在写入和读出 VS1053 的数据之后，它需要一些时间来处理内部的事情，这段时间，是不允许外部打断的，所以，我们在 SCI 操作之前，最好判断一下 DREQ 是否为高电平，如果不是，则等待 DREQ 变为高。了解了 VS1053 的 SPI 读写，我们再来看看 VS1053 的 SCI 寄存器。 2.3.2 SCI 寄存器 VS1053 总共有 16 个 SCI 寄存器，通过这些寄存器实现对 VS1053 的各种控制，VS1053 的所有 SCI 寄存器如表 2.3.2.1 所示：

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（ＳＴＭ３２ＦｌＯ３ＲＢＴ６）控制运行，各个部分协调运作。
硬件部分介绍：本次设计所做的是耩于ＳＴＭ３２的录音器，解码芯片ＶＳ１０５３的作用不言而喻，因为此次录音文件是保存在ＳＤ卡上的，自然少不了对ｓＤ卡的介绍。ＳＴＭ３２Ｆ１０３系列微处理器是首款基于ＡＲＭｖ７一Ｍ体系结构的３２位标准ＲＩＳＣ（精简指令集）处理器，提供很高的代码效率，在通常８位和ｌ６位系统的存储空间上发挥了ＡＲＭ内核的高性能。该系列微处理器工作频率为７２ｔ￣ｉｚ，内置高达１２８Ｋ字节的Ｆｌａｓｈ存储器。其内部结构图如图ｌ所示：
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科学技术
基于ＶＳ１０５３芯片的录音装置设计
周璐璐
（重庆电子工程职业学院，重庆４０４１Ｏ０）
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ＳＴＭ３２ＭＣＵ系列包含五个产品线：１．基本型系￣ＳＴＭ３２ＦｌＯＩ：３６ＭＨｚ最高主频，具有高达ｌ￣ＩＢ的片一Ｅ闪存。２．ＵＳＢ基本型系列ＳＴＭ３２Ｆ１０２：４８ＭＨｚ最高主频，具有全速ＵＳＢ模块。３．增强型系列ＳＴＭ３２Ｆｌ０３：７２ＭＨｚ最高主频，具有高达１ＭＢ的片上闪存，集成电机控制、ＵＳＢ和ＣＡＮ模块。４．互联型系列ＳＴＭ３２Ｆ１Ｏ５／１０７：７２Ｍ｝ｔｚ最高主频，具有以太网ＭＡＣ。ＣＡＮ以及ＵＳＢ２．０ＯＴＧ功能。本系统采用ＳＴＭ３２Ｆｌ０３ＲＢＴ６作为控制核心，该芯片具有６通道的１６位ＰｗＭ输出，ＲＴＣ时钟，两个高速ＳＰＩ，两个ＩＩｃ，三个串口，自带ＵＳＢ和ＣＮ接【Ａ＿＝Ｉ。具有１２８Ｋｆｌａｓｈ￥￣２０Ｋ的ＳＲＡＭ，对本系统来说足够了，同时以后还有很大的升级空间。本系统采用ＶＳ１０５３解码芯片，ＶＳ１０５３是荷兰ＶＬＳＩ公司出品的一款高性能的解码芯片。它不仅可以实现对ＭＰ３／ＯＧＧ／ｗＭＡ／ＦＬＡｃ／ｗＡＶ／ＡＡｃ／ＭＩＤＩ等音频格式的解码，同时还可以支持ＡＤＰＣＭ／０ＧＧ等格式的编码，性