WAVESTREAM 8_16_25_32_40 Ku-band BUC中文手册

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Waves Smack Attack Transient Shaper用户指南说明书

ContentsIntroduction (2)Components (3)Quick Start Guide (4)Interface (5)Controls (6)Sensitivity and Level Controls (6)Audio Graph (7)Shape and Duration Controls ................................................................................................................. 9 Attack Shapes ................................................................................................................................................................ 9 Sustain Shapes ......................................................... . (10)Output Controls (11)WaveSystem Toolbar (3)IntroductionThank you for choosing Waves! In order to get the most out of your new Waves plugin, please take a moment to read this user guide. To install software and manage your licenses, you need to have a free Waves account. Sign up at . With a Waves account you can keep track of your products, renew your Waves Update Plan, participate in bonus programs, and keep up to date with important information.We suggest that you become familiar with the Waves Support pages: /support. There are technical articles about installation, troubleshooting, specifications, and more. Plus, you’ll find company contact information and Waves Support news.The Waves Smack Attack Transient Shaper adds smack, sting, spike, and bite to your drums and other transient-rich instruments. Use it to harden or soften attacks and to boost or reduce sustains. You can shape your transients in a creative way or use Smack Attack as a surgical tool. Smack Attack has zero latency and it processes in real time, so it transforms your transients live or in the studio.About Smack AttackTransients are short-duration, impulsive changes in a sound. They naturally occur in drums and other percussion instruments, and in most instruments whose sound begins with a strong attack. Smack Attack is designed to detect, shape, and manipulate these transient sounds. It identifies transients based on their fast change in amplitude. Detection is not based on an absolute peak level, so Smack Attack can discern transients within low- or high-level signals. Smack Attack acts on two parts of a transient: attack and sustain. It manipulates attack transients by making them sharper or smoother, and sustain transients by making them more or less prolonged. The amount of processing is proportional to the transient rate of change: higher rate-of-change means greater action. Separate Sensitivity controls for attack and sustain let you set independently which transients are processed: perhaps only the loudest transients, or all of them, or anything in between. Processing controls—Level Change, Shape, and Duration—are also adjustedseparately for attack and sustain. This gives you more control when shaping the overall impact of the sound.ComponentsWaveShell technology enables us to split Waves processors into smaller plugins, which we call components . Having a choice of components for a particular processor gives you the flexibility to choose the configuration best suited to your material.Smack Attack Transient Shaper has two components:•Smack Attack Mono: mono-in to mono-out•Smack Attack Stereo: stereo-in to stereo-outWaveSystem ToolbarUse the bar at the top of the plugin to save and load presets, compare settings, undo and redo steps, and resize the plugin. To learn more, click the icon at the upper-right corner of the window and open the WaveSystem Guide.You can mix the processed signal with the input sound to blend the transient effect, and you can adjust output level by up to ±24 dB to get a clean, spiky sound. You can also create a loud, overdriven output signal and then limit dynamically or saturate without dynamic processing. Each of these output methods yields distinct results.Smack Attack is designed primarily for individual instruments or mix groups where you want to boost, attenuate, or manipulate transients—or to build creative effects. On top of that, you can insert Smack Attack on the master buss to control transients in the full mix.Quick Start GuideFor the most part, getting the sound you want is a matter of experimenting with the controls and learning how each affects the transients and the overall sound. We do, however, suggest the following steps as a starting point.Part One: Attack1.Adjust the Attack Level up or down. Note that the orange line, the gain change graph, changes as you adjustAttack. Turn the Attack Level to its extreme settings to understand how it affects the transient sound.2.Experiment with the Shape and Duration of the attack. The narrower the shape, the more “surgical” the attack willsound. A wider shape can warm up the attack sound. Duration determines how long it takes attack gain to return to unity.3.Set the Sensitivity to the level where attack processing will begin. The green lines in the Audio Graph illustratethe Attack Sensitivity threshold setting. Transients within the lines will be processed, those outside will not.Part Two: Sustaine Sustain Level to control the increase or decrease the level of the sustain (release). The resulting gainchange is shown as the orange line in the Audio Graph, along with the gain change from Attack processing.Higher Sustain settings usually yield more air.5.Listen to the sustain, using different Shapes and Durations, to understand how these controls affect the sound ofthe transient and of the surrounding sound. There are three Sustain shapes, each with different a behavior.6.Set the Sustain Sensitivity. This determines the threshold for Sustain processing.Part Three: Output7.Mix sets the mix between the processed transient sound and the dry signal.e the Output control to set the output level of the plugin. The Output control precedes the Guard setting in thesignal flow, so it will influence the behavior of the three Guard modes.9.Guard provides three methods of containing output level, each with different results. Try them all!Audio Graph: Provides visual feedback of transientprocessing.Yellow = Sustain sensitivity;Orange = Gain changeAttack L evel: Sets boost or reduction of processedtransient.Attack Sensitivity: Sets the threshold for attackdetection.Sustain Level: Sets the gain of sustain processingSustain Sensitivity: Sets threshold for sustaindetection.Attack Shape: Sets the shape of the processedtransient. The first two tools are best suited for drums; the third shape is intended for otherinstruments as well as drums.Attack Duration: Sets the length of attackprocessing.Sustain Shape: Defines the curve of the sustainarea. Three different shapes are available.Sustain Duration: Sets length of sustain gainprocessing.Mix: Parallel mix: Dry/WetOutput: Add or reduce the output signal.Guard: Three modes: 1) Off, 2) Clip with -0.1 dBfs ceiling, 3) Limit with -0.1 dBfs ceiling. When theGuard is off, a red light indicates overs.Sensitivity and Level ControlsAttack Level (left)Controls the amount of level increase or decrease applied to any transient that exceeds the Attack Sensitivity setting. Increasing this value results in exaggerated, punchy transient attacks. Reducing the value softens the transient attack and usually provides a warmer transient sound.Range: -100 to +100Value is indicated in the middle of the control.Sustain Level (right)Controls the amount of level increase or decrease applied to the sustain of a transient. Higher settings tend to increase the “air” introduced by the transient.Range: -100 to +100Value is indicated in the middle of the control.Attack SensitivitySets the threshold range for Attack processing. The attack of any sound that has been identified as a transient and falls within this sensitivity range will be processed. Smack Attack processes only transients, not the other elements of the sound, so Attack Sensitivity does not apply to non-transient sounds.Range: 0–100, Value is displayed when the control is touched.Sustain SensitivitySets the threshold at which Sustain processing will begin. Higher Sensitivity settings usually yield a warmer sound with more “air.” Sustain Sensitivity and Sustain Level are highly interactive, so it’s important to experiment with both controls when designing the sustain sound.Range: 0–100, Value is displayed when the control is touched.The Attack and Sustain controls greatly influence each other. Change one control setting and you will likely influence how the other controls affect the sound. Experiment.Audio GraphThis display shows all essential gain information.Input audio is shown as a solid-blue peak waveform. This makes it easy to seeThere are two pairs of lines that indicate the Sensitivity settings for Attack andSustain. These are the thresholds of transients processing. The green linesshow the level at which attack processing will begin. When Attack Sensitivit y isset very low, the lines are far apart, indicating that only the loudest transients willbe processed. With higher Sensitivity settings, these lines are closer together,indicating that smaller transients can be being processed.Sustain Sensitivity shows the threshold of where Sustain processing will betriggered. Its function is identical to the Attack Sensitivity indicator.The orange trace represents the gain change introduced by the Smack Attack processor.When Attack and Sustain are both set to 0, there is no gain change, so the orangeGain Change line is flat.change indicator, as seen on the left. NegativeAttack Level is displayed as downward gainchange.processing. Increasing Sustain Level, as seen onthe left, or reducing it, tends to move the entiregain change structure up or down the graph.Shape and Duration ControlsThe shape and duration of the attack and sustain can greatly influence the impact and texture of a transient. Sharper tools yield effects “hug” the transient, enabling punchier sounds and opening up possibilities in hard-edged creative processing. Wider tools produce a wider attack that warms up the transient.Attack ShapesThe attack shape s allow shaping the “edginess” of the transient attack. These shapes are not completely symmetrical: the rise is always faster than fall.Needle A very sharp rise and fall of the attack processing. The attack remains very tight. ThisNail A very sharp shape, similar to the Needle but with a slower, sloped fall. Although thisis a very quick shape, is somewhat softer than the Needle. It’s well suited for drums.Blunt This is a softer shape. Its rise is somewhat milder, and its fall is appreciably slower.This is intended for all instruments, not just drums, but it can also deliver interesting andcreative drum effects.The needle and nail shapes have an instantaneous rise when the attack is sensed. The blunt shape has a slower rise.Attack DurationThis sets the length of attack processing, measured from initial rise.Longer durations make for warmer transients; shorter durations produce a spikier sound.Range: 0.5 ms–500 msValue is displayed when the control is touched.Sustain ShapesThe Sustain shapes help you control the way that transient processing finishes. Each provides a different shape for the way that the processed Sustain signal falls.Linear Non-Linear The sustain decays at a linear rate. This can provide a continuous, even, return to unity. Intended for drums.Depending on the Sustain Level setting, the non-linear tool can provide a rapidor gentle curve. Intended primarily for drums.Soft Linear This shape has a softer peak and a linear fall. It moves the attack sound towardfor other instruments.The Linear and non-linear shapes are intended for use with drums, but you can achieve interesting creative effects when using them with other instruments. Results can be unpredictable, but you may find just the sound you’re looking for.Sustain DurationSustain Duration is the time it takes the gain to return to unity after the attack gain begins to fall.Range: 30 ms–1000 msValue is displayed when the control is touched.Output ControlsThe output section provides dry/wet mix and output level controls. There is also a selector for determining how over-leveloutput signals are treated.MixSets the percentage of dry (unprocessed) vs. wet (processed) output.Range: 0–100%Value is displayed when the control is touched.OutputControls the output level of the plugin.Range: ±24 dBValue is displayed in the center of the control.The Output controls can greatly affect the final sound, especially for Attacks. Boosting the level of attacks can boost the output signal levels above 0 dBfs.GuardThe Guard section lets you choose how to treat high-level output signals. There are three modes:Limit Dynamic range is controlled with a fast limiter. L evel is limited to -0.1 dBfs.Clip Off Signal above 0dBfs is clipped without changing its dynamic range. This saturates the over-level spikes, which changes their sound character. Level is limited to -0.1 dBfs.When Guard is off, there is no limiting or clipping of the output signal. A red light indicates when the output is over level. It’s unlikely that a signal will clip internally, even when the over-level light is illuminated.It’s important to consider the rest of the processing chain when setting the output level. If Smack Attack is part of a series of plugins, and the Guard is off, use the Output Level control to set a level that i s appropriate for the other processors and the D/A converter.Signal flow in the Output section is: Mix➝Output Level➝Guard.Output Level IndicatorIn the Limit or Clip modes, the meter displays gain reduction. Meter range: 0 to -40 dBWhen Guard is off, the meter displays output level above 0 dBfs. Meter range: 0 to 40 dB。

SpectralWave 40 80 V-Node S 多服务平台说明书

2Mbit/s (Max. 128ports)
Cross connect capacity
VC-4
64 x 64
VC-3
192 x ቤተ መጻሕፍቲ ባይዱ92
VC-12
4032 x 4032
Specification for Layer-2 switched service IEEE 802.1Q Port/Tag VLAN IEEE 802.1D Spanning Tree Protocol IEEE 802.1p Prioritization IEEE 802.3x Flow Control Jumbo Frame
Bandwidth upgrade up to STM-16 For future traffic growth, the V-Node S can be economically upgraded to dual 2-Fiber STM-16 ring system by just replacing optical interfaces.
The E-Line type of connection service using L2SW can be easily expanded to Ethernet LAN (E-LAN) service type of multipoint-to-multipoint connections. Security among the connections is guaranteed by VLAN method.
Integrated management with NEC’s other products The INC-100MS offers total management of the NEC’s photonic transport network composed of the SpectralWave C-Node, U-Node, the SpectralWave DWDM systems and the other SMS series SDH products.

WAVE文件头信息以及PCM数据的读取（C++版本）

WAVE文件头信息以及PCM数据的读取（C++版本）WA VE文件头信息以及PCM数据的读取/*************文件的处理方法**************/#include "math.h"#includeusing namespace std;#define N 200 //观察前面的200点的PCM数据float *Read_Data(FILE *fp_speech,int front_info);//读取PCM 数据的关键函数int main(){FILE *fp;fp=fopen("E:\\aaa.wav","rb");//为读，打开一个wav文件if((fp=fopen(""E:\\aaa.wav"","rb"))==NULL) //若打开文件失败，退出{printf("can't open this file\n");exit(0);}/*********RIFF WA VE Chunk的输出*********//****************************************RIFF WAVE Chunk==================================| |所占字节数| 具体内容|==================================| ID | 4 Bytes | 'RIFF' |----------------------------------| Size | 4 Bytes | |----------------------------------| Type | 4 Bytes | 'WA VE' |******************************************/char id_RIFF[4],type_RIFF[4];int size_RIFF;cout<<endl<<"riff "<<endl;<="" chunk的输出:="" p="" ve="" wa="">//读取ID=RIFF 4字节fseek(fp,0L,0);fread(id_RIFF,sizeof(int),1,fp);cout<<"RIFF标识："<<id_riff[0]<<id_riff[1]<<id_riff[2]<<id_riff[3]<<endl;< p=""> //读取文件大小4字节fseek(fp,4L,0);fread(&size_RIFF,sizeof(int),1,fp);cout<<"文件大小："<<size_riff<<endl;< p="">//读取W AVE标识4字节fseek(fp,8L,0);fread(type_RIFF,sizeof(int),1,fp);cout<<"WA VE标识："<<type_riff[0]<<type_riff[1]<<type_riff[2]<<type_riff[3]<<endl ;< p="">/*******Format Chunk的输出*******//****************************************Format Chunk====================================== ============================ ==| | 字节数| 具体内容|====================================== ============================ ==| ID | 4 Bytes | 'fmt ' |--------------------------------------------------------------------| Size | 4 Bytes | 数值为16或18，18则最后有附加信息|-------------------------------------------------------------------- ----| FormatTag | 2 Bytes | 编码方式，一般为0x0001 | |-------------------------------------------------------------------- || Channels | 2 Bytes | 声道数目，1--单声道；2--双声道| |-------------------------------------------------------------------- || SamplesPerSec | 4 Bytes | 采样频率| |-------------------------------------------------------------------- || AvgBytesPerSec| 4 Bytes | 每秒所需字节数| |===> W AVE_FORMA T -------------------------------------------------------------------- || BlockAlign | 2 Bytes | 数据块对齐单位(每个采样需要的字节数) | |-------------------------------------------------------------------- || BitsPerSample | 2 Bytes | 每个采样需要的bit数| |-------------------------------------------------------------------- || | 2 Bytes | 附加信息（可选，通过Size来判断有无）| |-------------------------------------------------------------------- ----******************************************/char id_FORMA T[4];int size_FORMA T,SamplesPerSec,AvgBytesPerSec;short int FormatTag,Channels,BlockAlign,BitsPerSample,addinfo;cout<<endl<<"format "<<endl;<="" chunk的输出:="" p=""> //读取famt标识4字节fseek(fp,12L,0);fread(id_FORMA T,sizeof(int),1,fp);cout<<"fmt标识："<<id_format[0]<<id_format[1]<<id_format[2]<<endl;< p=""> //读取size 4字节//fseek(fp,16L,0);//fread(&id_FORMA T,sizeof(int),1,fp);//cout<<"size大小："<<<endl;<="">//编码方式2字节fseek(fp,20L,0);fread(&FormatTag,sizeof(short),1,fp);cout<<"编码方式："<<formattag<<endl;< p="">//声道数2字节fseek(fp,22L,0);fread(&Channels,sizeof(short),1,fp);cout<<"声道数目："<<channels<<endl;< p="">//采样频率4字节fseek(fp,24L,0);fread(&SamplesPerSec,sizeof(int),1,fp);cout<<"采样频率："<<samplespersec<<endl;< p="">//每秒所需字节数4字节fseek(fp,28L,0);fread(&AvgBytesPerSec,sizeof(int),1,fp);cout<<"每秒所需字节数："<<avgbytespersec<<endl;< p=""> //数据块对齐单位2字节fseek(fp,32L,0);fread(&BlockAlign,sizeof(short),1,fp);cout<<"数据块对齐单位："<<blockalign<<endl;< p="">//每个采样需要的bit数2字节fseek(fp,34L,0);fread(&BitsPerSample,sizeof(short),1,fp);cout<<"每个采样需要的bit数："<<bitspersample<<endl;< p="">//附加信息2字节fseek(fp,36L,0);fread(&addinfo,sizeof(short),1,fp);cout<<"附加信息："<<addinfo<<endl;< p="">/*******Fact Chunk的输出*******//****************************************Fact Chunk==================================| |所占字节数| 具体内容|==================================| ID | 4 Bytes | 'fact' |----------------------------------| Size | 4 Bytes | 数值为4 |----------------------------------| data | 4 Bytes | |----------------------------------******************************************/char index;char id_FACT[4];int size_FACT,data_FACT;/*******Data Chunk的输出*******//****************************************Data Chunk==================================| |所占字节数| 具体内容|==================================| ID | 4 Bytes | 'data' |----------------------------------| Size | 4 Bytes | |----------------------------------| data | | |----------------------------------******************************************/char id_DATA[4];int size_DATA;float *PCMdata;//检验标识的第一个字符，判断是否存在Fact Chunk信息段，如果没有就直接输出Data 段fseek(fp,38L,0);fread(&index,sizeof(char),1,fp);if (index=='f')cout<<endl<<"fact "<<endl;<="" chunk的输出:="" p=""> //FACT标识4字节fseek(fp,38L,0);fread(&id_FACT,sizeof(int),1,fp);cout<<"FACT标识："<<id_fact[0]<<id_fact[1]<<id_fact[2]<<id_fact[3]<<endl;<p="">//size大小4字节fseek(fp,42L,0);fread(&size_FACT,sizeof(int),1,fp);cout<<"size大小："<<size_fact<<endl;< p="">//FACT段data 2字节fseek(fp,46L,0);fread(&data_FACT,sizeof(int),1,fp);cout<<"FACT段data："<<data_fact<<endl;< p="">/************Data Chunk的输出***************/cout<<endl<<"data "<<endl;<="" chunk的输出:="" p=""> //DATA标识4字节fseek(fp,50L,0);fread(&id_DATA,sizeof(int),1,fp);cout<<"DATA标识："<<id_data<<endl;< p="">//PCM数据长度4字节fseek(fp,54L,0);fread(&size_DATA,sizeof(int),1,fp);cout<<"PCM数据长度："<<size_data<<endl;< p="">/*******以下为重要的PCM数据输出*******/fseek(fp,46L,0);PCMdata=Read_Data(fp,46);cout<<"PCM数据："<<endl;< p="">for (int i=0;i<n;i++)< p="">{cout<<pcmdata[i]<<" ";<="" p="">if((i+1)%5==0)cout<<endl;< p="">}}if (index=='d') //为真，则为只存在data数据段cout<<endl<<"data "<<endl;<="" chunk的输出:="" p=""> //DATA标识4字节fseek(fp,38L,0);fread(&id_DATA,sizeof(int),1,fp);cout<<"DATA标识："<<id_data[0]<<<id_data[2]<<id_data[3]<<endl;<=""> //PCM数据长度4字节fseek(fp,42L,0);fread(&size_DATA,sizeof(int),1,fp);cout<<"PCM数据长度："<<size_data<<endl;< p="">/*******以下为重要的PCM数据输出*******/fseek(fp,46L,0);PCMdata=Read_Data(fp,46);cout<<"PCM数据："<<endl;< p="">for (int i=0;i<n;i++)< p="">{cout<<pcmdata[i]<<" ";<="" p="">if((i+1)%5==0)cout<<endl;< p="">}}fclose(fp);return 0;}float *Read_Data(FILE *fp_speech,int front_info){short data; //为什么用short？参看文献中的相应wav文件的pcm数据存放格式。

Wavedream NET 网络增强传输使用指南说明书

Wavedream NET Network Enhanced TransportOwner’s ManualWAVEDREAM NETOWNER’S MANUALTable of ContentsAbout this document (2)Package Contents (3)Safety Precautions (3)Functional Description (4)Powering UP (5)CD playback (5)Server playback (6)Setting internal storage (8)Setting Roon (8)Remote control function (12)Specification Sheet (13)ROCKNA AUDIO LIMITED WARRANTY (15)About this documentThis manual is aimed at the end user of the Rockna Wavedream NET. It contains an overview of the device’s internal architecture, specific handling details, functional description, safety precautions and product warranty details. This document is not meant for service or repair operations, as these must be carried out only by qualified personnel.More information regarding Rockna Audio and Audiobyte products can be found on-line at the websites owned and operated by the company:https://https://Rockna Electronics S.R.L.Address: Strada Plopului, nr. 5, SuceavaZIP Code: 720145, RomaniaPhone:+40 770 125 694 – General inquiresEmail:************************–Technicalsupport************************–GeneralinquirieThank you for choosing the Rockna Wavedream transport. The Wavedream NET was designed to give you countless hours of musical enjoyment.Package Contents-Wavedream NET-Supply cable-IR remote control-User manual-(Warranty Card)Safety Precautions1) This device is meant for indoor use only.2) Protect device from excessive heat, humidity and liquid filled objects, such as vases.3) Clean only with dry cloth.4) Do not remove product cover while the device is plugged in the mains outlet.5) Use earth grounded outlet if available.6) Do not move the device while operational.7) Lightning or static electricity can affect normal operation of the device. Make sure that it is unplugged during a thunderstorm.8) Make sure the unit is unplugged if it is not to be used for a long period of time.Functional DescriptionFront view:Legend:1.Power switch2.Menu key3.Disc/Server selection key4.Play/next (cd playback)5.Disc/stop (cd playback)6.CD DRIVE tray7.Display8.IR sensor (do not cover)Back view:The conenctors on the back panel are self-explanatory.Powering UPPlug in power cord and WDNET will start in CD mode (no need to use power button on the first time). To use the remote, make sure “CD” key was pressed. This setting is sticky unless other device is selected.Press “L” on remote to switch between CD mode and server mode or D/S key (3) from the front panel. Assuming the unit is left connected to mains, you can shut down/power up the unit normally after first boot. Note: When pressing power button for shut down, a lag of a few seconds is normal before unit display goes off. This is to allow server to properly shut down all services. Do not press the power button multiple times.CD playbackYou can use WDNET as regular cd transport. The unit accepts regular CD’s (redbook) as well wav (up to 384k sample rate) files if burned on a DVD disc (in UDF format).Besides looking as a regular CD transport, NET is actually a true memory player. This means you don’t hear the disc as it spins, but you hear the sound of a solid-state memory. As a funny exercise, try to press eject (disc) button while in play. The disc will be ejected but the music will continue to play for tens of seconds.Basic Operation:1.Press DISC (5) to open drawer in order to insert a cd2.Press DISC again or push gently the drawer.3.Press PLAY (4) for music.4.If unit is in play mode, pressing PLAY again will skip to next track.5.If unit is in play mode, pressing DISC key will stop playback.6.If playback is already stopped, pressing DISC will eject the media.e remote for regular playback functions like direct access, skip,repeat etc.Note : CD playack is completely separated from the server. Thereforeactive outputs are I2S 1&2, AES/EBU, SPDIF and BNC. The USB ports are not used in CD mode.Server playbackTo properly use WDNET server, make sure you connect the unit to your local network via LAN cable. The web interface of WDNET can be accessed on your browser at address wdnetXXXXXXX.local , where XXXXXXXX is your serial number (can be found on the back of the unit).General interfaceVia web interface all relevant settings are available : switching on and off installed services, ripping and network drive settings. For MPD and Airplay, a different output audio device can be set, when a external USB dac is attached. For Roon, this can be made through the Roon settings menu.Network drive settingsRipping tabNote: for ripping, a external usb optical drive is required (internal drive from WDNET is not connected to server).Setting internal storageIf your WDNET comes with internal storage, or is upgraded accordingly, the drive “WDNETXXXXXXXX/SATA” (X is your serial no.) will show as a network share in your network. This is the drive where you can copy your music.Setting up RoonWDNET is a full Roon server. A Roon remote app installation is required on your smartphone or tablet. Additionally, the server can be controlled from another Roon instance installed on your Mac or PC.First, you have to connect to the Roon core installed on WDNET. It will look like below:After connecting to the core, we have to select the folders where music is stored, you should press “add folder”:To set correctly the local storage, use a path like below:After this step, Roon will start to index your local music:Now, let’s set up the audio device. Press “Manage audio devices”. A screen like below will show up.Press “Enable” on WAVEDREAM NET (ROCKNA, ALSA). On playback tab from the next screen and change DSD playback strategy to “DSD over PCM 1.0 (DoP). Press “Save settings”.Now you can give a name to WDNET in the Roon network. You can type “WDNET” for example in the empty field.In the end, all you have to do is to select the defined audio zone and Roon is ready to go.Remote control functionSpecification SheetINPUTS•DISC PLAYBACK 32 bit – 384 KHz PCM, 1 bit – 11.28 MHz DSD;•ETHERNET 32 bit-384 KHz PCM, 1 bit – 5.6 MHz DSD;•USB 1 32 bit – 384 KHz PCM, 1 bit – 11.28 MHz DSD;•USB 2 32 bit – 384 KHz PCM, 1 bit – 11.28 MHz DSD. OUTPUTS•HD-Link 32 bit – 384 KHz PCM, 1 bit – 11.28 MHz DSD;•AES/EBU 24 bit – 384 KHz PCM, 1 bit – 5.6 MHz DoP;•S/PDIF 24 bit – 192 KHz PCM, 1 bit – 5.6 MHz DoP;•BNC 24 bit – 192 KHz PCM, 1 bit – 5.6 MHz DoP;•VGA 1920×1080 max. resolution.General•Disc playback: CD audio, DSD & WAV files from Blu-Ray and DVD disc;•NETWORK playback: all usual files, PCM & DSD;•LOCAL STORAGE PLAYBACK: ALL USUAL FILES, PCM & DSD•PCM 44.1K-384K, DSD64-DSD256;•Local storage (ssd only): 1,2,4 or 8TB.System updateIn NET webpage you can check the right bottom corner – you can find the current software version. Press check for updates link to see if your unit is up to date.ROCKNA AUDIO LIMITED WARRANTYThree (3) YearsWARRANTY COVERAGE:ROCKNA AUDIO warranty obligation is limited to the terms set forth below.WHO IS COVERED:Rockna Audio warrants the product to the original purchaser or the person receiving the product as a gift against defects in materials and workmanship as based on the date of original purchase from an Authorized Dealer. The original sales receipt showing the product name and the purchase date from an authorized retailer is considered such proof. WHAT IS COVERED:The Rockna Audio warranty covers new products if a defect arises and a valid claim is received by Rockna Audio within the Warranty Period. At its option, Rockna Audio will either (1)repair the product at no charge, using new or refurbished replacement parts, or (2)exchange the product with a product that is new or which has been manufactured from new, or serviceable used parts and is at least functionally equivalent or most comparable to the original product in Rockna Audio current inventory, or (3)refund the original purchase price of the product. Rockna Audio warrants replacement products or parts provided under this warranty against defects in materials and workmanship from the date of the replacement or repair for one(1) year or for the remaining portion of the original product’s warranty, whichever provides longer coverage for you. When a product or part is exchanged, any replacement item becomes your property and the replaced item becomes Rockna Audio’s property. When a refund is given, your product becomes Rockna Audio’s property. Note: Any product sold and identified as refurbished or renewed carries a one(1) year limited warranty. Replacement product can only be sent if all warranty requirements are met. Failure to follow all requirements can result in delay.WHAT IS NOT COVERED - EXCLUSIONS AND LIMITATIONS:This Limited Warranty applies only to the new products manufactured by or for Rockna Audio that can be identified by the trade-mark, trade name, or logo affixed to it. This Limited Warranty does not apply to any non-Rockna Audio hardware product or any firmware, even if packaged or sold with the product. Non-Rockna Audio manufacturers, suppliers, or publishers may provide a separate warranty for their own products packaged with the bundled product. Rockna Audio is not liable for any damage to or loss of any programs, data, or other information stored on any media contained within the product, or any non-Rockna Audio product or part not covered by this warranty. Recovery or reinstallation of programs, data or other information is not covered under this Limited Warranty.This warranty does not apply (a)to damage caused by accident, abuse, misuse, misapplication, or non-Rockna Audio product, (b)to damage caused by service performed by anyone other than Rockna Audio or Rockna Audio Authorized Service Location, (c)to a product or a part that has been modified without the written permission of Rockna Audio, or (d)if any Rockna Audio serial number has been removed or defaced, or (e) product,accessories or consumables sold “AS IS” without warranty of any kind by including refurbished Rockna Audio product sold “AS IS” by some retailers.This Limited Warranty does not cover:• Shipping charges to return defective product to Rockna Audio.• Labor charges for installation or setup of the product, adjustment of customer controls on the product, and installation or repair of systems outside of the product.• Product repair and/or part replacement because of improper installation, connections to improper voltage supply, abuse, neglect, misuse, accident, unauthorized repair or other cause not within the control of Rockna Audio. • Damage or claims for products not being available for use, or for lost data or lost firmware.• Damage occurring to product during shipping.• A product that requires modification or adaptation to enable it to operate in any country other than the country for which it was designed, manufactured, approved and/or authorized, or repair of products damaged by these modifications.• A product used for commercial or institutional purposes(including but not limited to rental purposes).• Product lost in shipment and no signature verification receipt can be provided.• Failure to operate as per Owner’s Manual.REPAIR OR REPLACEMENT AS PROVIDED UNDER THIS WARRANTY IS THE EXCLUSIVE REMEDY FOR THE CONSUMER. ROCKNA AUDIO SHALL NOT BE LIABLE FOR ANY INCIDENTAL OR CONSEQUENTIAL DAMAGES FOR BREACH OF ANY EXPRESS OR IMPLIED WARRANTY ON THIS PRODUCT. EXCEPT TO THE EXTENT PROHIBITED BY APPLICABLE LAW, ANY IMPLIED WARRANTY OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE ON THIS PRODUCT IS LIMITED IN DURATION TO THE DURATION OF THIS WARRANTY.Some states do not allow the exclusions or limitation of incidental or consequential damages, or allow limitations on how long an implied warranty lasts, so the above limitations or exclusions may not apply to you.Page | 17©Rockna Electronics S.R.L., Suceava, Romania。

音频编码（PCM,ADPCM，WAVE文件）

⾳频编码（PCM,ADPCM，WAVE⽂件）ADPCM⽂件解码详解⼀、搞了⼏天终于搞定这个ADPCM解码了，之前找了很多的资料，⼤致描述的都是千篇⼀律，但是基本上都没有说到细节上，让我也⾛了不少弯路，其实主要在细节，⽹上给的算法是正确的，但是直接运⽤根本就不⾏，噪⾳很⼤。

这⼀点让我⼀直很头疼，最后还是看了英⽂资料，才得到解答，还是⽼外的原始资料好。

⼆、给个英⽂参考⽹址吧这两个讲的很详细，请仔细阅读！通过阅读我发现细节在与adpcm格式的wav⽂件的block的特点，每⼀个block包含header和data两部分，Typedef struct{short sample0; //block中第⼀个采样值（未压缩）BYTE index; //上⼀个block最后⼀个index，第⼀个block的index=0;BYTE reserved; //尚未使⽤}MonoBlockHeader关键是我们要抓住每⼀个block的header⾥⾯的信息，即sample0，运算的时候注意运⽤！三、还是给个代码吧,多的也不说了！1、adpcm.c⽂件代码#include"adpcm.h"/* Intel ADPCM step variation table */staticintindexTable[16]={-1,-1,-1,-1,2,4,6,8,-1,-1,-1,-1,2,4,6,8,};staticintstepsizeTable[89]={7,8,9,10,11,12,13,14,16,17,19,21,23,25,28,31,34,37,41,45,50,55,60,66,73,80,88,97,107,118,130,143,157,173,190,209,230,253,279,307,337,371,408,449,494,544,598,658,724,796,876,963,1060,1166,1282,1411,1552,1707,1878,2066,2272,2499,2749,3024,3327,3660,4026,4428,4871,5358,5894,6484,7132,7845,8630,9493,10442,11487,12635,13899,15289,16818,18500,20350,22385,24623,27086,29794,32767};voidadpcm_decoder(char*inbuff,char*outbuff,intlen_of_in,structadpcm_state *state ){int i=0,j=0;chartmp_data;structadpcm_state *tmp_state =state;longstep;/* Quantizer step size */signedlongpredsample;/* Output of ADPCM predictor */signedlongdiffq;/* Dequantized predicted difference */intindex;/* Index into step size table */intSamp;unsignedcharSampH,SampL;unsignedcharinCode;/* Restore previous values of predicted sample and quantizer stepsize index*/predsample =state->valprev;index =state->index;for(i=0;i<len_of_in*2;i++){tmp_data=inbuff[i/2];if(i%2)inCode=(tmp_data&0xf0)>>4;elseinCode=tmp_data &0x0f;step =stepsizeTable[index];/* Inverse quantize the ADPCM code into a predicted differenceusing the quantizer step size*/diffq =step >>3;if(inCode &4)diffq +=step;if(inCode &2)diffq +=step >>1;if(inCode &1)diffq +=step >>2;/* Fixed predictor computes new predicted sample by adding theold predicted sample to predicted difference*/if(inCode &8)predsample -=diffq;elsepredsample +=diffq;/* Check for overflow of the new predicted sample*/if(predsample >32767)predsample =32767;elseif(predsample <-32768)predsample =-32768;/* Find new quantizer stepsize index by adding the old indexto a table lookup using the ADPCM code*/index +=indexTable[inCode];/* Check for overflow of the new quantizer step size index*/if(index <0)index =0;if(index >88)index =88;/* Return the new ADPCM code */Samp=predsample;if(Samp>=0){SampH=Samp/256;SampL=Samp-256*SampH;}else{Samp=32768+Samp;SampH=Samp/256;SampL=Samp-256*SampH;SampH+=0x80;}outbuff[j++]=SampL;outbuff[j++]=SampH;}/* Save the predicted sample and quantizer step size index fornext iteration*/state->valprev =(short)predsample;state->index =(char)index;}2、adpcm.h⽂件代码#ifndefADPCM_H#defineADPCM_H#ifdef__cplusplusextern"C"{#endifstruct adpcm_state {short valprev; /* Previous output value */char index; /* Index into stepsize table */};voidadpcm_decoder(char*inbuff,char*outbuff,intlen_of_in,structadpcm_state *state ); #ifdef__cplusplus} /* extern "C" */#endif#endif/* ADPCM_H*/3、main.c⽂件代码#include"stdio.h"#include "stdlib.h"#include "adpcm.h"#defineCFG_BlkSize 256charch[CFG_BlkSize]; //⽤来存储wav⽂件的头信息charsavedata[CFG_BlkSize*4];unsignedcharRiffHeader[]={'R','I','F','F',// Chunk ID (RIFF)0x70,0x70,0x70,0x70,// Chunk payload size (calculate after rec!)'W','A','V','E',// RIFF resource format type'f','m','t',' ',// Chunk ID (fmt )0x10,0x00,0x00,0x00,// Chunk payload size (0x14 = 20 bytes)0x01,0x00, // Format Tag ()0x01,0x00, // Channels (1)0x40,0x1f,0x00,0x00,// Sample Rate, = 16.0kHz0x80,0x3e,0x00,0x00,// Byte rate 32.0K0x02,0x00, // BlockAlign == NumChannels * BitsPerSample/80x10,0x00 // BitsPerSample};unsignedcharRIFFHeader504[]={'d','a','t','a',// Chunk ID (data)0x70,0x70,0x70,0x70 // Chunk payload size (calculate after rec!)};/****************************************************************函数名称： main功能描述：输⼊参数： none输出参数： none****************************************************************/voidmain(void){FILE *fpi,*fpo;unsignedlongiLen,temp;structadpcm_state ADPCMstate;unsignedlongi =0;unsignedlongj;fpi=fopen("f:\\lk\\test.adpcm","rb"); //为读，打开⼀个wav⽂件if((fpi=fopen("f:\\lk\\test.adpcm","rb"))==NULL) //若打开⽂件失败，退出{printf("can't open this file\n");printf("\nread error!\n");printf("\n%d\n",i);exit(0);}fseek(fpi,0,SEEK_END);iLen=ftell(fpi);printf("\n======================================================\n");printf("\n========================%d========================\n",iLen);printf("\n======================================================\n");if((iLen-44)%CFG_BlkSize)iLen =(iLen-44)/CFG_BlkSize+1;elseiLen =(iLen-44)/CFG_BlkSize;fpo=fopen("f:\\lk\\new.pcm","rb+"); //为写，打开⼀个wav⽂件if((fpo=fopen("f:\\lk\\new.pcm","rb+"))==NULL) //若打开⽂件失败，退出{printf("can't open this file\n");printf("\nwrite error!\n");exit(0);}fseek(fpo,0,SEEK_SET);fwrite(RiffHeader,sizeof(RiffHeader),1,fpo); //写⽂件rifffwrite(RIFFHeader504,sizeof(RIFFHeader504),1,fpo); //写 data块头while(i<iLen){fseek(fpi,48+i*CFG_BlkSize,SEEK_SET);fread(ch,1,CFG_BlkSize,fpi);printf("\n======================================================\n");for(j=0;j<100;j++)printf("| %d |",ch[j]);printf("\n======================================================\n");////////////////////////添加读取BlockHeader部分开始////////////////////////////////if(i ==0){ADPCMstate.index =0; //第⼀个block的index为 0 当前的BlockSize为 256 即采样点数为（256-4）*2+1 = 505}else{ADPCMstate.index =ch[2];}ADPCMstate.valprev =(short)ch[0]+((short)(ch[1]))*256; //每⼀个block⾥⾯帧头有⼀个未压缩的数据存储时先低后⾼savedata[0]=ch[0]; //存储第⼀个没有被压缩的数据savedata[1]=ch[1]; //存储第⼀个没有被压缩的数据////////////////////////添加读取BlockHeader部分结束////////////////////////////////adpcm_decoder(&ch[4],&savedata[2],CFG_BlkSize-4,&ADPCMstate);//解码出来了（256-4）*4 个字节temp =(CFG_BlkSize-4)*4+2;fseek(fpo,44+i*temp,SEEK_SET); //开始写声⾳数据fwrite(savedata,temp,1,fpo);i++;}temp *=i;RiffHeader[4]=(unsignedchar)((40+temp)&0x000000ff);RiffHeader[5]=(unsignedchar)(((40+temp)&0x0000ff00)>>8);RiffHeader[6]=(unsignedchar)(((40+temp)&0x00ff0000)>>16);RiffHeader[7]=(unsignedchar)(((40+temp)&0xff000000)>>24);fseek(fpo,4,SEEK_SET);fwrite(&RiffHeader[4],4,1,fpo);RiffHeader[40]=(unsignedchar)(temp&0x000000ff);RiffHeader[41]=(unsignedchar)((temp&0x0000ff00)>>8);RiffHeader[42]=(unsignedchar)((temp&0x00ff0000)>>16);RiffHeader[43]=(unsignedchar)((temp&0xff000000)>>24);fseek(fpo,40,SEEK_SET);fwrite(&RiffHeader[40],4,1,fpo);fclose(fpi);fclose(fpo);printf("\n==========================OK!=========================\n");}四、以上是给出的代码，绝对管⽤，读者在实验时候请在vc++6.0环境下建⽴⼯程，实验时候请在f:\\lk\\下放置⼀个adpcm格式的⽂件和⼀个空的pcm格式⽂件，当然了这个pcm和adpcm其实都是wav格式的，试验者可以随意命名格式，我为了区分才这样命名后缀的，希望我总结的能够读者带来帮助，谢谢您的阅读！PCM数据格式1. ⾳频简介经常见到这样的描述: 44100HZ 16bit stereo 或者 22050HZ 8bit mono 等等.44100HZ 16bit stereo: 每秒钟有 44100 次采样, 采样数据⽤ 16 位(2字节)记录, 双声道(⽴体声);22050HZ 8bit mono: 每秒钟有 22050 次采样, 采样数据⽤ 8 位(1字节)记录, 单声道;当然也可以有 16bit 的单声道或 8bit 的⽴体声, 等等。

Grandstream Wave 电话客户端用户手册说明书

潮流网络技术有限公司技术支持潮流网络技术有限公司为客户提供全方位的技术支持。

您可以与本地代理商或服务提供商联系，也可以与公司总部直接联系。

地址：深圳市南山区科技园北区新西路16号彩虹科技大厦4楼邮编：518057网址：客服电话：*************客服传真：*************技术支持论坛：/forums网上问题提交系统：/support/submit-a-ticket商标声明和其他潮流网络商标均为潮流网络技术有限公司的商标。

本文档提及的其他所有商标或注册商标，由各自的所有人拥有。

注意由于产品版本升级或其他原因，本文档内容会不定期进行更新。

除非另有约定，本文档仅作为使用指导，本文档中的所有陈述、信息和建议不构成任何明示或暗示的担保。

目录更新日志 (1)软件版本1.0.1.9(Android TM)/1.1.9(iOS TM) (1)软件版本1.0.0.17(Android TM)/1.0.15(iOS TM) (1)软件版本1.0.0.6 (1)欢迎使用 (2)概述 (2)软件特性 (2)基本操作 (4)设备要求 (4)下载及安装 (4)基本手势 (5)使用Wave (6)登录帐号 (6)忘记密码 (6)拨打电话 (7)重拨 (7)使用通话记录 (8)使用联系人 (8)接听电话 (9)通话 (9)通话保持与恢复 (11)静音 (11)声音通道切换 (12)未接来电 (12)呼叫转移 (13)盲转 (13)指定转 (14)会议室 (14)即时会议 (15)预约会议 (16)音频会议 (17)视频会议 (17)视频会议成员 (18)结束会议 (19)链接入会 (20)语音邮件 (20)联系人 (21)搜索联系人 (21)查看联系人 (22)常用联系人 (22)通话记录 (24)设置 (25)帐号信息 (25)关于 (26)常见问题解答 (27)图表目录图表1手势说明图 (5)图表2登录界面 (6)图表2忘记密码界面 (6)图表3拨号盘界面 (7)图表4使用通话记录拨号 (8)图表5使用联系人拨号 (8)图表6来电振铃界面 (9)图表7单路通话 (10)图表8语音通话保持界面 (11)图表9通话静音界面 (11)图表10通话时切换声音通道界面 (12)图表11未接来电提醒界面 (12)图表12呼叫转移-盲转界面 (13)图表13呼叫转移-指定转界面 (14)图表14会议室界面 (15)图表15即时会议设置界面 (15)图表16预约会议设置界面 (16)图表17音频会议界面 (17)图表18视频会议界面 (18)图表19会议参会者界面 (19)图表20主持人退出会议界面 (19)图表21链接入会界面 (20)图表22语音邮件收听界面 (21)图表23联系人主界面 (21)图表24联系人搜索界面 (22)图表25联系人详情界面 (22)图表26联系人详情中设置常用联系人界面 (23)图表27联系人详情中设置常用联系人界面 (23)图表28通话记录 (24)图表29通话记录详情 (24)图表30设置界面 (25)图表31帐号信息界面 (25)用户手册图表32关于界面 (26)用户手册更新日志更新日志本节主要介绍Wave App最新的版本更改及新功能的增加。

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.。

DirectSound 与Waveout的区别

DirectSound 与Waveout的区别Q Directsound 与 Waveout 有何不同？A Waveout 是在32位的Windows上的一种老旧且过时，用来播放数字音讯的应用程序接口（ApplicationProgramming Interface，简称API）。

旧的Windows操作系统（如Win9x WinNT4)在 Wav eout 的完成度很高（因为 waveout 是针对这些操作系统设计的），如果你想获得最好的效能，你应该在这些操作系统上使用 Waveout 输出。

然而 Waveout 的功能有所局限，它无法支持「混和多重音讯流」的功能。

这显示在Win2kXP下的 Waveout，只是为了旧的软件的兼容性所提供的，也因此Win2kXP下 Waveout 的完成度很糟，它没有使用任何的硬件加速功能，所有的混音动作都是用软件来执行（因此当CPU的使用率很高时，常常会发生类似CD跳针的断音现象）。

Directsound 是种较新、较现代化的声音播放 API，都已经内建在最近的32位Windows操作系统中。

Directsound 支持混和多重音讯流、独立的音量控制、硬件加速层及硬件仿真层（如果某些功能硬件无法支持，可以用软件来仿真，因此程序设计师无须担心他们的新 l33t码无法在旧的声霸卡16上运作）。

一般来说，只要你的操作系统安装了适当的声卡驱动程序及最新的 DirectX，Direstsound 都应该可以运作的很好（除了WinNT4以外）。

在Win2kXP下，Directsound 比 waveout 更好，因为在这些操作系统里，Directsound 的完成度相对的比 waveout 来的更高（比 waveout 占用较少的CPU资源，自由度较高，且不会有 Waveout 常见的小毛病）。

Directsound 原本是被设计来让游戏利用系统的硬件加速功能，而无须直接接触低阶的硬件函数（就如同 DirectX 其它的组件）。

waveout系列API实现pcm音频播放

waveout系列API实现pcm⾳频播放最近做⼀个播放组件，也算是折腾1周了，收获还算不少。

回想下整个编码过程中磕磕碰碰⾛了不少弯路，最⼤的杯具就是，太相信⽹上现有代码例⼦。

国内⽹上关于waveout的⽂章不少，但基本就那⼏篇转载，其中的问题也没有⼈指出。

为了⽅便⼤家⽤到时少被误导，在此留下我的笔记（如果被我误导了，我先道歉－，－）代码不多，直接上关键部分（本⼈认为多余代码贴上去百害⽽⽆⼀利）：⼀、初始化设备bool WinAudioPlay::DevOpen(){if (!m_bPalyStata){WAVEFORMATEX wfx;ZeroMemory(&wfx,sizeof(WAVEFORMATEX));wfx.wFormatTag = WAVE_FORMAT_PCM;wfx.nChannels = 2;wfx.nSamplesPerSec = 44100L;wfx.wBitsPerSample = 16;wfx.cbSize = 0;wfx.nBlockAlign = wfx.wBitsPerSample * wfx.nChannels / 8;wfx.nAvgBytesPerSec = wfx.nChannels * wfx.nSamplesPerSec * wfx.wBitsPerSample / 8;if(::waveOutOpen (0,0,&wfx,0,0,WAVE_FORMAT_QUERY)){Plug::PlugMessageBox(L"wave设备初始化失败～");return false;}if (::waveOutOpen(&m_hWaveOut, WAVE_MAPPER, &wfx, (DWORD)&WinAudioPlay::waveOutProc, (DWORD)this, CALLBACK_FUNCTION)) {Plug::PlugMessageBox(L"wave设备初始化失败～");return false;}m_iBlockNum = 0;m_bPalyStata = true;m_spbrTgthr.reset(new boost::barrier(2));m_sptrdWaveOutTgthr.reset(new boost::thread(boost::bind(&WinAudioPlay::ThrdWaveOutTogether,this)));}return true;}⼆、接收pcm格式数据，并加载到声卡缓冲区bool __stdcall WinAudioPlay::play_audio( const void* buffer, int len ){if (!m_bPalyStata)return false;if (BLOCK_MAX <= m_iBlockNum || len <= 0){return true; //超过缓冲最⼤包，不继续播放}LPWAVEHDR pWaveHeader = new WAVEHDR;memset(pWaveHeader, 0, sizeof(WAVEHDR));pWaveHeader->dwLoops = 1;pWaveHeader->dwBufferLength = len;pWaveHeader->lpData = new char[len];if (!pWaveHeader->lpData){delete pWaveHeader;return false;}memcpy(pWaveHeader->lpData, buffer, len);if (::waveOutPrepareHeader(m_hWaveOut, pWaveHeader, sizeof(WAVEHDR))){delete pWaveHeader->lpData;delete pWaveHeader;return false;}if (::waveOutWrite(m_hWaveOut, pWaveHeader, sizeof(WAVEHDR))){delete pWaveHeader->lpData;delete pWaveHeader;return false;}m_iBlockNum++;return true;}三、回调函数void CALLBACK WinAudioPlay::waveOutProc( HWAVEOUT hwo, UINT uMsg, DWORD dwInstance, DWORD dwParam1, DWORD dwParam2 ) {WinAudioPlay* pThis=(WinAudioPlay*)dwInstance;if(WOM_DONE == uMsg) //播放完成{while(NULL != pThis->m_lpWaveHdrFromCallbackProc){boost::this_thread::interruptible_wait(1);}pThis->m_lpWaveHdrFromCallbackProc = (LPWAVEHDR)dwParam1;pThis->m_spbrTgthr->wait();}return ;}四、线程同步播放void WinAudioPlay::ThrdWaveOutTogether(){while(!m_b_exit){m_spbrTgthr->wait();if (NULL != m_lpWaveHdrFromCallbackProc){::waveOutUnprepareHeader(m_hWaveOut, m_lpWaveHdrFromCallbackProc, sizeof(WAVEHDR));delete[] m_lpWaveHdrFromCallbackProc->lpData;delete m_lpWaveHdrFromCallbackProc;m_lpWaveHdrFromCallbackProc = NULL;(m_iBlockNum > 0)?(m_iBlockNum--):(m_iBlockNum = 0);}}if (m_hWaveOut != NULL){::waveOutReset(m_hWaveOut);::waveOutClose(m_hWaveOut);}}五、关闭线程，释放资源WinAudioPlay::~WinAudioPlay(){m_bPalyStata = false;while(0 != m_iBlockNum){Sleep(1);}m_b_exit = true;m_spbrTgthr->wait();m_sptrdWaveOutTgthr->join();}我相信聪明的你，借助msdn能够很快理解上⾯意思，我就不多打字了（⽔平有限，怕误⼈⼦弟～）但是需要注意以下⼏点：⼀、waveOutProc回调函数中绝对不能调⽤waveOut系列函数（可⽤线程同步实现通知在另⼀个线程调⽤）⼆、调⽤waveOutReset函数时，函数执⾏完毕才返回，期间不可以调⽤hWaveOut三、注意释放资源and so on...其实还有很多，我就不多写了，以上3点中前两点处理不好，会发⽣线程死锁。

WAVE文件的头文件定义

WAVE文件的头文件定义typedef struct _TWavHeader{long rId;long rLen;long wId;long fId;long fLen;WORD wFormatT ag;WORD nChannels;long nSamplesPerSec;long nAvgBytesPerSec;WORD nBlockAlign;WORD wBitsPerSample;long dId;long wSampleLength;}TWavHeader;TWavHeader wh;wh.rId = 0x46464952;wh.rLen = 36;wh.wId = 0x45564157;wh.fId = 0x20746d66;wh.fLen = 16;wh.wFormatTag = 1;wh.nChannels = wChannels;wh.nSamplesPerSec = lRate;wh.nAvgBytesPerSec = wChannels * lRate * (wResolu tion / 8);wh.nBlockAlign = wChannels * (wResolution / 8);wh.wBitsPerSample = wResolution;wh.dId = 0x61746164;wh.wSampleLength = 0;int nHandle = FileCreate(strFileName);FileSeek(nHandle, 0, 0);FileWrite(nHandle, &wh, sizeof(wh));FileClose(nHandle);===================================== wav文件包括头和数据两部分，其结构如下：（从文件头开始依次排列）1）首先是字符串“RIFF”，占4个字节。

2）波形块的大小：DWORD，占4字节。

Bitstream3X控制器使用说明-乐声音频

BitStream 3X MIDI控制器快速入门成都乐声科技王昕<一> 连接将Bitstream 3X 加入到电脑音乐系统中：以下是Bitstream 3X以及其他MIDI设备接入电脑的示意图：a.Bitstream 3X可以通过随机附送的USB线接入电脑。

b.也可通过MIDI电缆或者SYNC24电缆将其他MIDI设备接入系统中来。

（例如：MIDI主键盘，SYNC24 设备…）<二> 驱动安装与硬件重置Bitstream 3X也属于“即插即用”设备，因为其兼容USB-MIDI协议，在Windows XP、OS X和Linux中，Bitstream 3X 不需要安装任何特殊驱动，插上识别一次之后就可以使用。

a.首次插入Bitstream 3X请注意，当Bitstream 3X插入电脑后，电脑将自动安装所需要的常规驱动，以后连入Bitstream 3X直接将被电脑识别。

b.当电脑识别Bitstream 3X后，在Bitstream 3X上也将做出被识别的反映，显示屏右上方将显示USB图形。

（如下图显示）c.类同与其他的USB AUDIO DEVICE，当电脑识别Bitstream 3X后，它所附带的MIDI外围也将在系统以及SONAR等音序软件中列出，对应识别列表如下图显示：（注：Bitstream 3X接入电脑后如果不习惯电脑识别的USB AUDIO DEVICE名称，Wave Idea 公司还在Bitstream 3X所附送的光盘中带有安装信息文件---Waveidea_usb_driver.zip。

将该安装信息文件装入系统后，在Windows XP设备管理器中就会出现Wave Idea Bitstream 3X的名字。

）<三> 硬件对应软件操作快速入门Bitstream 3X提供的控制功能包括：8个推子、1个Crosse Fader、32个旋钮和14个按钮统统可以自定义参数，并且允许将定义的情况编组，可以方便地记忆和唤出场景。

NI ELVIS II系列产品规格说明说明书

Modes ..............................................Analog edge triggering, analog edge triggering with hysteresis, and analog window triggering
Arbitrary Waveform Generator/Analog Output
Number of channels.........................2
DAC resolution................................16 bits
DNL.................................................±1 LSB
Maximum working voltage for analog inputs (signal + common mode) ................ ±11 V of AIGND
CMRR (DC to 60 Hz) ..................... 90 dB
Source..............................................AI<0..15>, ScopeCH0, ScopeCH1
Small signal bandwidth (–3 dB)......1.2 MHz
Input FIFO size................................4095 samples
Scanlist memory ..............................4095 entries
Data B signal stream, programmed I/O

GOLDENWAVE GD-02 高性能解码耳放一体机使用说明书

XLR-4 芯平衡耳机插孔
2. 后面板
RCA 前级信号输出 -L，-R
AES 数字信号输入
保险丝
RCA 线路信号输出 -L，-R
同轴数字信号输入
XLR 前级信号输出 -R，-L
光纤数字信号输入
AC 电源插座
XLR 线路信号输出 -R，-L
USB 信号输入 (B 型 ) 电源开关
6
连接 GD-02
1. 环境要求 (1) 请将设备放在一个平稳坚固的底面上。 (2) 确保设备环境通风良好，产生的热量可以及时散去。
(2) 通过 USB 接口连接 iPhone：需要一条 Lightning 至 USB 转换器 ( 选购 )*。
使用随机附送的 USB 线缆，方形接口接入 GD-02，USB 接口接入转换器，转换器 Lightning 接口接入 iPhone。
附送的 USB 线缆
Lightning 至 USB 转换器
17
PC 设置
(9) 在弹出的界面点击“高级”选项卡，在默认格式选择“24 位， 44100Hz( 录音室音质 )”或者“32 位，44100Hz( 录音室音质 )”，再点击确定即可。
8
连接 GD-02
(4) 通过 AES、光纤以及同轴接口连接数字信号源： GD-02 配备了 AES、光纤以及同轴接口，通过对应类型的线缆，可以连接不同类型的数字设备。例如：CD 机、数字播放器以及便携播放器等等。
CD 机
数字播放器
便携播放器
AES
同轴接口
光纤接口
9
连接 GD-02
3. 连接其他音频设备可以将本机连接音源和其他音频设备之间，本机包含 2 种输出信号类型 ,
同时输出信号。 (1) 前级信号输出 (PREAMP OUTPUT)，包含 1 组 XLR-3 平衡接口以及 1

wave文件格式

WAVE文件作为多媒体中使用的声波文件格式之一，它是以RIFF格式为标准的。

RIFF是英文Resource Interchange File Format的缩写，每个WAVE文件的头四个字节便是“RIFF”。

WAVE文件由文件头和数据体两大部分组成。

其中文件头又分为RIFF／WAV文件标识段和声音数据格式说明段两部分。

WAVE文件各部分内容及格式见附表。

常见的声音文件主要有两种，分别对应于单声道（11.025KHz采样率、8Bit 的采样值）和双声道（44.1KHz采样率、16Bit的采样值）。

采样率是指：声音信号在“模→数”转换过程中单位时间内采样的次数。

采样值是指每一次采样周期内声音模拟信号的积分值。

对于单声道声音文件，采样数据为八位的短整数（short int 00H-FFH）；而对于双声道立体声声音文件，每次采样数据为一个16位的整数（int），高八位和低八位分别代表左右两个声道。

WAVE文件数据块包含以脉冲编码调制（PCM）格式表示的样本。

WAVE文件是由样本组织而成的。

在单声道WAVE文件中，声道0代表左声道，声道1代表右声道。

在多声道WAVE文件中，样本是交替出现的。

WAVE文件格式说明表偏移地址字节数数据类型内容文件头00H 4 char "RIFF"标志04H 4 long int 文件长度08H 4 char "WAVE"标志0CH 4 char "fmt"标志10H 4 过渡字节（不定）14H 2 int 格式类别（10H为PCM形式的声音数据)16H 2 int 通道数，单声道为1，双声道为218H 2 int 采样率（每秒样本数），表示每个通道的播放速度，1CH 4 long int 波形音频数据传送速率，其值为通道数×每秒数据位数×每样本的数据位数／8。

播放软件利用此值可以估计缓冲区的大小。

常用工具软件中文录音编辑处理器 WAVECN 2.0.0.5

常用工具软件中文录音编辑处理器 WA VECN 2.0.0.5WaveCN 是一个32位的音频编辑软件，可以通过本软件对音频数据进行编辑修改和修饰特殊效果。

另外，WaveCN 通过安装AudioExtractor 插件（已经包含在安装包中）后，还支持从视频文件中分离音频。

一般来说，只要这个视频文件能够通过Windows Media Player 进行播放，则WaveCN 就可以将其音频部分分离出来。

因此，WaveCN 软件具有以下特性：● 录制音频，支持电平监控、支持后台录音、热键控制、定时录音、声控录音等。

● 支持多种音频文件格式打开、保存。

● 音频数据编辑（包含有剪切、复制、粘贴等多种常用编辑操作）。

● 较多的音频处理效果。

● 方便易用的多文档处理界面。

● 可以通过插件扩充功能。

● 高保真的采样率转换。

● 支持多重剪贴板，以及剪贴板预览。

● 支持标记列表● 支持声控录音和直接录，并保存为WAV 文件。

● 支持可编排的定式录音。

用户通过WaveCN 软件，在编辑音频文件过程中，可以保存为多种格式。

如下列列举了该软件所支持的文件，并且全部均支持读取或者保存操作，如表11-2所示。

在该软件的窗口中，用户可以看到所包含有菜单、工具栏（又称“快捷按钮”）、插件功能按钮栏、文件选择栏、多用途工具栏、波形观察和标尺、功率计、状态显示条和状态栏等，如图11-13所示。

图11-13 WaveCN 窗口菜单栏工具栏插件功能按钮文件选择多用途工具栏波形观察和标功率计状态显示条和状态栏●菜单栏该软件的菜单栏与其他软件的菜单栏操作相同，都是用于文件操作。

但是，在菜单中执行【文件】|【新建】命令与工具栏中单击【新建】按钮，区别在于前者会弹出一个【新建文件】对话框。

该对话框主要用于设置新文件的属性，而后者则直接安装软件选项中的默认设置给出一个空白的文件。

在菜单栏中，执行【文件】|【文件信息】命令，可以修改文件的属性（即频率、采样位数、声道数值，如44.1KHz，16位，立体声）以及文件中可以携带的信息。

音频插件——wave3.0教程

音频插件（Wave3_0）效果各种简介（教程+下载）点击此处鼠标右键目标另存为下载第一种效果：AudioTrack(音频轨）它是针对通常我们见到的普通的音轨的，综合了4段EQ均衡、压缩、噪声门三种效果器。

如果你用过T－racks(母带处理软件），你会发觉它与AudioTrack 的功能非常相像。

只不过T-racks的界面要漂亮得多了。

下图是AudioTrack(音频轨）的界面：使用它之后，你的整个混音作品就可以站立在坚实的基础之上了。

在均衡方面，具体操作和参数设置可以参照Ultrafunk--EQ效果器。

在压缩方面可以参照Ultrafunk的压缩效果器。

在Gate（噪声门）的几个参数中，只有Floor(基底）是我们不熟悉的，不熟悉怎么办，试试就知道了。

1、Rel为Release(释放）的缩写。

2、将鼠标放在按钮上，鼠标会变化成一个双向的箭头，照此方向拖动，可以直接调整按钮相应的值。

3、如果你觉得某一步操作错误，可以点击左上方的UNDO按钮，将这一步取消。

4、如果你在Output(输出）的设置上调得过大，右面的No Clip(没有削波）会变成Out Clip（溢出，削波），同时在下面显示出已超过多少。

如果你将此时的设置处理成波形，则会看到被放大的电平信号和被削去的“刺儿头”。

5。

你可以在Setup A中选择一种预置方案，点击Setup A之后再显示出来的Setup B中再调入一种方案，然后进行两种方案的对比。

再点一下A->B 或是B->A 都会让两种方案统一。

第二种效果器：C1 Compressor（C1 压缩器）如下图：左面为压缩／扩展电平表。

中间的按钮分别是：Low Ref,低反射，点击之后改变为峰值反射；Makeup,电平弥补；Threshold,阀值；Ratio,压缩比率；Attack,起音时间；Release,释放时间；第三种效果器C1 Gate(噪声门）如下图：除了有些按钮的功能与C1 compressor不同之外，基本界面都很相似。

WAVE文件格式解析

WAVE⽂件格式解析WAVE ⽂件作为Windows多媒体中使⽤的声⾳波形⽂件格式之⼀，它是以RIFF（Resource Interchange File Format）格式为标准的。

这⾥不针对RIFF⽂件格式做介绍，不太了解的可以参考“”⼀⽂。

WAVE⽂件构成每个WAVE⽂件的头四个字节便是“RIFF”。

WAVE ⽂件由⽂件头和数据体两⼤部分组成。

其中⽂件头⼜分为 RIFF／WAV ⽂件标识段和声⾳数据格式说明段两部分。

相对于RIFF⽂件，只是将“RIFF”chunk的form id替换为“WAVE”。

下表是⼀个典型的WAVE⽂件各部分构成及其长度字段。

注意所有数据采⽤windows默认的⼩端存储。

（FOURCC是⼀个特殊的四字节码，判断时按照字符顺序判断即可。

）域长度内容说明chunkID4Chunk ID: "RIFF"，FOURCC四字节码chunksize4Chunk size: 4+nWAVEID4WAVE ID: "WAVE"，FOURCC四字节码WAVE chunks n Wave chunks包含格式信息和⾳频采样数据，分为“format” chunk和“data”chunk两部分。

Format chunkFormat chunk⽤于说明data chunk中PCM数据的格式。

主要三种不同的format chunk格式（不同的格式码）。

如下表：域长度内容说明ckID4Chunk ID: "fmt "，FOURCC四字节码，注意最后⼀个填充是空格。

cksize4Chunk size: 16 or 18 or 40 wFormatTag2Format code，格式码nChannels2Number of interleaved channels，采样声道数（交织存储）nSamplesPerSec4Sampling rate (blocks per second)，⾳频采样率nAvgBytesPerSec4Data rate，⾳频码率nBlockAlign2Data block size (bytes)，⾳频数据块⼤⼩（单位字节）wBitsPerSample2Bits per sample，量化位数（⽐如8bits、16bits、32bits）cbSize2Size of the extension (0 or 22)，扩展字段长度wValidBitsPerSample2Number of valid bits，有效的位长度dwChannelMask4Speaker position mask，声道描述掩码，⽐如左声道、右声道等SubFormat16GUID, including the data format code，数据格式码标准中定义的wFormatTag（Format code）可取值范围如下表：Format Code PreProcessor Symbol Data 0x0001WAVE_FORMAT_PCM PCM0x0003WAVE_FORMAT_IEEE_FLOAT IEEE float，[-1.0f,1.0f] 0x0006WAVE_FORMAT_ALAW8-bit ITU-T G.711 A-law 0x0007WAVE_FORMAT_MULAW8-bit ITU-T G.711 µ-law 0xFFFE WAVE_FORMAT_EXTENSIBLEDetermined by SubFormatPCM格式当wFormatTag为0x0001时，表⽰WAVE⽂件中存储的是PCM格式的⾳频数据。

VB DirectSoundCapture详细说明

VB DirectSoundCapture详细说明'###############################################'# #'# DirectX-DirectSound练习程序#'# 搜集：realline #'# 时间：2001-05-21 #'# #'###############################################'在使用DirectX 之前必须先“引用”DirectX 库函数。

'dtx为DirectX 的对象简码Option ExplicitDim m_dx As New DirectX7 '声明一个DirectX7 对象，并同时生成一个新对象；Dim m_ds As DirectSound '声明一个DirectSound 对象；Dim m_dsBuffer As DirectSoundBuffer '声明一个DirectSoundBuffer 对象；Dim m_dscap As DirectSoundCapture '声明一个DirectSoundCapture 对象；'DirectSoundCapture 独立于DirectSound,是DirectX 中的一个组件，它允许捕获wave 声音Dim m_dscapBuffer As DirectSoundCaptureBuffer '声明一个DirectSoundCaptureBuffer 对象；'DirectSoundCaptureBuffer 的具体属性方法：'信息类'GetCaps '获得DSCBCAPS 数据结构；'GetCurrentPosition '获取文件播放指针位置，需要使用DSCURSORS 数据来获取' Type DSCURSORS' lPlay As Long '当前播放指针位置' lWrite As Long '当前数据读取指针的位置，一般都是先读取在播放，因此该指针位置比lPlay靠后' End Type'GetFormat '获得WAVEFORMATEX 数据结构；'GetStatus '获得CONST_DSCBSTATUSFLAGS 数据结构；'Enum CONST_DSCBSTATUSFLAGS' DSCBSTATUS_CAPTURING = 1 '获得当前捕获的缓存数据' DSCBSTATUS_LOOPING = 2 '获得当前捕获的和循环捕获的缓存数据'End Enum'缓存控制类'ReadBuffer '读取捕获的声音缓存'object.ReadBuffer(start As Long,size As Long,buffer As Any,flags AsCONST_DSCBLOCKFLAGS)'start '数据读入的起始位置，字节型。

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★★★★（7020207）整理制作目录1、引言 (3)1.1、保修 (3)1.2、技术支持 (3)1.3、设备维修服务 (3)1.4、注意事项 (3)2、安全 (5)2.1、避免水和连接器上的水分 (5)2.2、保证设备通风 (5)2.3、使用正确的电源 (5)2.4、防止雷击和电源浪涌 (6)2.5、禁止往设备内插如异物 (6)3、系统描述 (7)3.1、空间功率合成 (7)3.2、接口与外形尺寸 (8)3.2.1、“IF IN”连接器（J1） (8)3.2.2、“RF OUT”连接器（J2） (8)3.2.3、DC和监控（J3） (8)3.2.4、LED指示灯 (11)3.2.5、安装面尺寸 (11)3.2.6、风扇 (14)3.2.7、监控电缆 (14)4、连接和配置功放 (15)4.1、设备开箱 (15)4.2、配件，可能附带以下配件以协助单元的安装： (15)4.3、检查设备的接口 (15)4.4、检查配套电缆 (15)4.5、将本装置安装 (16)4.6、启动发射开关 (16)5、技术参数： (17)6、维护与保养 (20)6.1、一般性维护 (20)6.2、风扇组件的更换 (20)7、规范与标准 (21)7.1、标准规定认证 (21)7.2、EC符合性声明 (21)8、故障诊断 (22)8.1、两个LED灯不亮，风扇转 (22)8.1.1、首先检查功放TX为启用状态 (22)8.1.2、然后确认IF输入和RF输出 (22)8.2、两个LED灯不亮，风扇也不转 (22)8.2.1、首先确认供电航空头有直流电 (23)8.2.2、然后检查DC和M＆C航空头连接是否充分。

(23)8.3、“LOL”黄灯亮 (23)8.3.1、首先确认调制器10MHz (23)8.3.2、然后检查连接 (23)8.4、“SUM FLT”红灯亮 (23)8.4.1、首先检查风扇和通风 (23)8.4.2、通过监控查看温度 (23)1、引言本用户手册提供的信息和指示的WAVESTREAM设备的安装和操作。

本用户手册旨在用于培训的现场技术人员，系统工程师，负责卫星网络。

本用户手册是WAVESTREAM的8 - ，16 - ，25 - 和40瓦Ku波段系列火柴盒块式上变频器（BUC），以下简称为“单元”L-波段（950-1450MHz或950-1700MHz）输入和Ku波段输出（14.0-14.5GHz或13.75-14.5GHz）。

单元密封符合室外防雨标准，N-型输入，WR-75波导的输出，电源输入为24VDC或48VDC可选，32针圆形监控航空接头。

1.1、保修WAVESTREAM保修期定义在随附的报价单和采购订单的条款及销售条件。

在保修期内的设备需要返回到WAVESTREAM维修的，请联系您在WAVESTREAM客户服务的代表。

在信件表明设备的型号/序列号。

保修不适用于任何使用不当或不正确的维护和保养导致的故障。

尽管销售的条款及条件规定的保修，WAVESTREAM没有义务提供以下服务：» 非WAVESTREAM的人员自己维修该设备。

» 非标配套设备的使用所造成的损坏或故障。

» 非WAVESTREAM书面许可而修改服务，或集成到其他产品。

1.2、技术支持在本用户手册的范围之外的设备操作技术信息或支持您需要，请通过电子邮件support@联系技术支持。

1.3、设备维修服务千万不要尝试自己维修设备。

任何故障联系WAVESTREAM或经销商。

应该注意的是，只有外部风扇用户可维修的，请参见第6章 - 维护的更换风扇。

警告：打开或取下盖子的单位，可能会接触到危险电压，高功率射频辐射，以及保修失效。

1.4、注意事项要使该设备正常运行，必须是严格和明确地遵循本手册中的说明。

进行操作之前用户必须阅读和理解本手册的内容。

不完整阅读理解并遵守本手册中的所有内容之前，操作该设备可能会造成的设备、部件损坏或任何人员伤害，WAVESTREAM不承担任何相关责任。

WAVESTREAM不转让任何根据其专利，商标，版权或普通法权或类似权利的其他许可证。

WAVESTREAM还有权对任何产品或零件进行任何更改，恕不另行通知。

注意：本设备使用时会产生高频辐射。

这可能会带来安全隐患，第2章-安全中列出的预防措施。

2、安全对设备包含微电子和电气元件。

按照本节中的所有安全预防措施。

使用和操作前请仔细阅读并遵守所有的安全与对设备的操作说明。

保留这些说明，以备将来参考。

本节包括：使用单位设备的安全注意事项。

警告：有触电的危险：不要打开设备。

设备内部有高电压电。

设备中没有户可维修的部件（除外部风扇）。

请勿尝试自行维修本产品。

高功率射频的危险：除非在RF输出波导法兰被正确地连接到该系统的其余部分或高功率负载，否则不要进行加电操作，该设备发射高功率射频能量对人体是有害的。

操作设备时，人不要在波导口前面或附近。

静电危险：电子设备很容易被ESD（静电释放）损坏，使用该设备时应采取必要标准防静电措施。

操作时，请遵守以下注意事项：2.1、避免水和连接器上的水分» IF输入，电源，M＆C连口应紧密连接和密封，用防水措施处理接口，如防水胶带。

» WR-75的波导管的连接，以防止污染物进入界面，安装时正确的O形垫圈。

2.2、保证设备通风» 给设备的两侧保留空间来提供良好的通风环境，以确保设备稳定运行。

» 在必要时可使用强制空气对流通风。

» 不要堵塞通风口。

» 为了避免过热和确保的通风孔不堵塞，将本设备四周至少2英寸空间，并保证空气流动。

如果设备被放置在一个封闭的空间，如建筑物或雷达天线罩，请确保提供适当的通风和内部工作温度不超过最高额定温度60℃。

2.3、使用正确的电源» 设备已通过军用航空头或通过中频电缆供电。

为了防止设备损坏，确保正确的电压范围和足够的输入功率。

设备外壳附有的部件号可查到电压，例如：M B B - K U S 02 5 - D S0 0其中，C=24VDC / D =48VDC/ J =48VDC或K = 24VDC （中频供电）2.4、防止雷击和电源浪涌» 为了防止电压浪涌和内置静电积累，安装设备时的接地符合国家电气和无线电设备接地标准。

» 为了确保设备连续的和不受来自主电源线路的异常干扰，使用不间断电源（UPS）供电。

2.5、禁止往设备内插如异物» 触摸单元内部零件对您或设备是危险的。

切勿将任何物体穿透槽或开口，因为这可能会导致触电危险，短路，火灾或静电放电（ESD）。

如果一个异物掉进设备，马上拔掉电源插头，否则可能会发生严重损坏。

3、系统描述WAVESTREAM系列使用AdvantageTM的空间功率合成技术提供RF功率，其体积小，重量轻。

小的外形尺寸，可以单元直接安装到大多数VSAT天线的双工器发射口。

本节包括：» 简要介绍WAVESTREAM的独特空间功率合成技术。

» 接口描述包括引脚和配合连接器的信息» 更多信息请访问WAVESTREAM网站：3.1、空间功率合成“PowerStream TM Amplifier TM由一个阵列的固态MMIC放大器安装在卡堆放在波导外壳，见下图。

这些卡中的每一个上有小的输入天线接收的输入信号通过输入波导和小的输出天线，相应发射放大后的输出信号转换成输出波导。

PowerStreamTM Deck AmplifierTM使用WAVESTREAM的专利空间功率合成WAVESTREAM的Deck Amplifier TM架构提供了传统的固态功放使用微带或波导相结合的显着优势。

所有单独的的MMIC放大器输出在空间上结合成一个连贯的输出光束在一个步骤中，使放大器的高效。

甲板AmplifierTM可以被显示为85％至90％的结合效率，以相结合的装置的数量无关，与二进制MMIC放大器元件的数目增加，而传统升中派位组合器的效率降低。

WAVESTREAM的架构允许MMIC的被选择的数量不被约束到二进制（幂）结合以达到最佳的工作功率。

3.2、接口与外形尺寸设备有以下主要接口：3.2.1、“IF IN”连接器（J1）» “IF IN”连接器是一个N 型连接器。

» 本接口采用IF 与10MHz 参考信号复用。

» 标称阻抗为50欧姆。

» 在某些型号上，VDC 也复用在此接口。

注意事项无内部偏置-T 的单位：对于单位，内部偏置-T：如果单位不具有内部偏置-T 选项，然后不认沽DC N 型连接器（J1）。

此外，如果调制解调器有一个直流电源 - 禁用它。

如果本机内部偏置-T 选项，电源应只适用于键入N CONNCETOR（J1）。

不要把DC 直流和M＆C 连接器（J3）。

3.2.2、“RF OUT”连接器（J2）» “RF OUT”连接器是一个WR75波导法兰。

» 为了防水密封波导法兰接口，安装时O 形橡胶环应放置在圆形凹槽中。

» 波导口内部可见到一个防水膜，请勿刺破这层膜。

3.2.3、DC 和监控（J3）» DC 和M＆C 连接器是一个32针的卡口式圆形连接器。

它的配合连接器电缆部分MS3126F18-32S。

DC及监控接口引脚说明：3.2.4、LED指示灯单元有2个LED状态指示灯。

注：LED可以通过监控禁用。

本节中描述假定LED没有被禁用。

» “LOL”指示灯：黄色LEDLED亮起时，变频器失锁，IFL没有10 MHz参考信号或其电平过低。

单元失锁的10MHz参考，变频器失效，但是，放大器仍处于启用状态。

» “SUM FLT”指示灯：红色LED此LED亮起时，表示该单位是温度过高或有一个关键的内部放大器故障。

温度 - 单元的温度已超出其内部限位。

内部放大器的功率模块自动切断，因为内部温度达太高会使放大器损坏。

放大器故障 - 一个关键的内部电路发生故障或输入电压/电流超出了其允使用范围。

3.2.5、安装面尺寸单元的设计，可以直接安装到天线双工器WR75输出法兰发射口，但也可通过其侧面，正面和背面的螺丝孔固定。

注意：这些尺寸图给出的是单元的IF输入，DC和M＆C接口与输出波导口在两个外表面上。

但也可以订购接口都在同一面上，其配置外形图，请咨询厂家。

3.2.6、风扇» 单元冷却风扇，户外防水的处理风扇。

» 风扇应定期清洁，维护周期将取决于该地区环境的灰尘和其他污染物。

如有需要，也可以现场可更换的一个WAVESTREAM的风扇套件。

» 当环境温度低于-10℃，风扇会关闭。

3.2.7、监控电缆如果M＆C电缆大于6英尺，建议多芯屏蔽电缆可用于降低噪声拾取。