![Millennia HV-3D 4通道话放]()
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价格面议
- 产品简介：QQ:148-0362713 TEL:186-006-07968 项目经理：博乐
供应销售Millennia全线产品！
它拥有出色的设计，音质*佳，技术规范与机械做工都是**之作。值得一提的是，HV-3D为您提供了HV-3B前置放大器所带有的多重通道，价格*加实惠公道。
在电路设计方面，HV-3D与Millennia公司推出的HV-3B系列高压话筒前置放大器产品有诸多相似之处。HV-3D拥有标准的高解析度增益切换（36步长，每步长1.5 dB）、每个通道可供选用的B&K（DPA，微分功率分析）130V话筒输入、直流串接输入（这将使缎带麦克风与其它动圈话筒的效果*为出色）以及被动式相加选择等等。
在现今的*音频设备市场上，不少话筒前置放大器产品都采用了真空电子管、集成电路功能模块以及音频变压器的设计，这类产品往往想通过“色染”来达到所谓的真正的效果器的处理效果。然而，真正严格意义上的录音、混音、母带处理以及监听是需要客观真实声音的再现的。这一设计理念在HV-3D身上有十分出色的体现，它带有+23 dBu的本地化动态余量，不需要额外的衰减器或切换点（而这正是传统话放设备所必不可少的装置），减少了可引起音质下降的源泉。*为可贵的是，HV-3D所具有的+32 dBu的输出动态余量，这就使*高的动态稳定性以及线形化得到了保证。
从输入到输出，Millennia HV-3D的信号路径都与较小化的放大器拓朴结构相平衡，其出色的电路设计，多样化的选择，匹配度*好的元件配置确保了在各种动态偏移或频率延展的情况下，HV-3D都将带给您通透的音质，*水准**无庸置疑。此外，虽然HV-3D所配备的标准音频线长度*过1000英尺，但实践证明，在使用中未发现任何信号损失或质量下降。性能**的HV-3D四通道话筒前置放大器在2000年曾获得了TEC大奖的提名。
HV-3D-4真实的还原你声音的一切，并且具有非常宽泛的动态范围，*小的音染，如此高的品质在业内受到非常多的人的**，MillenniaMedia是前置放大的可能选择，非常适合一些高挡麦克风前置放大，在用HV-3 D 录管弦乐队采样，它的声音透明、纯朴而**。如果只能有一台前置放大，我将选择Millennia。对于打击乐，Millennia HV-3同样是许多采样音色库的可以选择。并且有很多的音色文件的制作都是依靠这款*级的话放来录制，成为大家使用的流行的音色文件的。
Millennia产品是鉴定声学录音和现场应用的可以选择—古典音乐的识别标准，使用它的有全世界*级艺术家（帕瓦罗蒂、三大男高音、席利迪翁等)，许多好莱坞大片（星球大战1和2，泰坦尼克等数百部），还有对声学音乐有要求的工程师和制作人，以及着名的音频厂商，全世界主要的广播网络。世界上较**的交响乐队和歌剧院大厅也安装了Millennia产品。Millennia产品始终是*音频的可以选择设备。HV-3麦克风前置放大器是MillenniaMedia产品中较为**的一个，迄今为止全世界已经安装的HV-3 *过14000通道。
● 分立元件晶体管后级
● 标准高分辨率增益开关(36步进)
● 通道增益匹配到 .04 dB
● 四通道或八通道两种规格
● 屏蔽的环型变压器供电
● 轻松再现逼真的音乐动态
主要特点
● 分立元件晶体管后级
● 标准高分辨率增益开关(36步进)
● 通道增益匹配到 .04 dB
● 四通道或八通道两种规格
● 屏蔽的环型变压器供电
● 轻松再现逼真的音乐动态
技术指标
● 较小增益：8.5 dB
● 较大增益 (每步1.5 dB, 36步进)：60.5 dB (up to 78 dB on rqst)
● 频率响应 +0 / -3 dB：4 Hz －300 kHz
● 立体声分离度 ： 25 V/ms
● 较大输入电平 20 Hz - 40 kHz：+23 dBu*
● 较大输出电平20 Hz - 40 kHz：+32 dBu 通道串扰 35 dB 增益, 1 kHz, +27 dBu
● 所有输出：优于 -120 dB (> 20 Bits)
● 幻像输入阻抗 @ 20 Hz ：6,700 Ohms
● 幻像输入阻抗@ 20 kHz ：6,800 Ohms
● 输出阻抗： 24.3 ohms (x2) B&K (DPA)
● 话筒供电：+130 VDC, 较大,直接供电（非幻象）
● 幻像供电：+48 VDC, +/- 2 VDC
● 较大功耗：45 W,
● 电源 ：100 －240 VAC, 50/60 Hz...
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北京麦田中旺数码科技有限公司
2022-06-07
![Buzz Audio QSP-20话放]()
-
价格面议
- 产品简介：QQ:148-0362713 TEL: 项目经理：博乐
供应销售Buzz Audio全线产品！
We wanted to build a new mic preamplifier that exhibited the qualities of our Class A range but at reduced price to the end user. By packing four preamps into one chassis with one power supply, the per channel cost is significantly reduced. But you won't find any cut corne here, each channel uses two of our Class A amplifie at the input and a Lundahl* audio traformer at the output. The excellent sound produced by this combination has been well proven in our 500 Series products. If the QSP-20 is the only preamp you may own, we guarantee it will sound great - no matter what you throw at it.
Like most of our microphone preamplifie, the Buzz Audio QSP-20 does not use a traformer at the input. Itead we couple direct to the Class A (BE41) amplifier modules with hi-quality WIMA capacito. This approach presents a much less reactive load to the microphone and allows it breathe better. In other words, the QSP is like an exteion to the microphone and this mea you can clearly hear the differences between all microphone types and brands. The HI Z - LO Z switch allows you change the load the microphone sees, and this can produce subtle tonal changes depending on the type of microphone.
Another advantage of the direct coupled input is noise - or lack of it. You will discover the QSP has an extremely low noise floor making it useful for low output microphones such as ribbo. The low noise plus the high output headroom results in a preamp with 92dB dynamic range at full gain of 70dB.
In addition to the four mic inputs, you also have four high impedance itrument inputs on the front panel. These inputs can be used for keyboards and itruments with unbalanced pickups. Both the mic and line inputs benefit from the sound of the Lundahl output traformer which exhibits an incredible frequency respoe right out to 250kHz, whilst adding a subtle amount of harmonic distortion at the lower frequencies. Traients are preserved and the bottom end sounds full making the QSP-20 the perfect choice for all music styles.
Applicatio:
?Modern music recording
?Acoustic music recording
?Mobile recording set ups
?Drums!
Features:
?The vivid sound of discrete Class A amplifie in the signal path
?The lush tone of Lundahl audio output traforme
?Up to 70dB of gain
?+28dBu output headroom
?Culmination of 20 yea design experience
?Robust chassis cotruction
?On board linear power supplies - no wall warts
?Easy to use control layouts
?Competitive pricing...
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北京麦田中旺数码科技有限公司
2022-06-07
![AMS Neve DPA话放]()
-
价格面议
- 产品简介：Q tel: 项目经理：博乐
供应销售AMS Neve全线产品！
1073话放诞生于1970年5月10日。经过时间的考验，1073之声成为了*级制作人和录音师较为渴望得到的声音之一。较初，1073被用作于Wessex A88这款经典调音台的话放，在20世纪70年代，很多良好的调音台都采用了1073作为它们的话放，比如NEVE的8014调音台。
1073拥有了NEVE经典之声的精华，曾经启发了很多话放设计师，成为众多**话筒放大器的模仿对象。比如Great River NV系列话放。
1073是人声，吉他以及各种声学乐器的较佳选择。
1073是百万元*级调音台NEVE 88R的话放组件(88R是世界*级的模拟调音台。普通模拟台与**数字台区别不大，但*级模拟调音台的各方面指标要远远*出*级数字调音台)。它只有1U的机身高度。在如此小的空间之内，就可以获得*级设备的声音.
1073具有低指标的本底噪声和失真。但它较大的特点却是*强的过载承受能力。较大输出级别达到了+26dBu（600欧姆），这已经达到了模拟设备的承受极限，后级模拟设备一般很难承受这么高的输入电平。充分发挥了模拟设备动态大的优势（远大于数字设备）。另外1073DPD的模数转换电平可以在机体的*部进行选择，具有+18dBu和+26dBu两个档位，对于PP值出现频率比较低的音乐信号，往往可以考虑采用+18dBu的档位，这样可以更多的降低量化噪声的含量。
1073是变压器耦合的晶体管放大器。它采用了NEVE本公司生产的10468 变压器，这款放大器经测试，其性能指标要**同等价位的话筒放大器。1073采用了较低效率却较高质量的A类放大电路,这种电路设计可以较大限度的让放大器工作在它的线性区域，在可听频段保持很低的声染色，很平直的频率响应曲线，并且不可能出现交越失真。
1073有着两*放大电路，*级为晶体管的放大，*二级为可微调+ -10dBu的变压器放大。
1073有着很强的放大能力，较多可达80dB（步长为5dB），达到了市面可见的*致。
1073有两个输入阻抗档位可以选择（一般的话放是固定输入阻抗的）。有300欧姆和1200欧姆两个档位。300欧姆这个额外的低输入阻抗设计是为了很好的配合时下流行的*低输出阻抗话筒.
1073DPA详细规格：
输入阻抗：
话筒 300Ω 或1200Ω
线路 20kΩ
频率响应：
60dB 增益 into 600 20Hz_20KHz 0.5dB
60dB 增益 into 600 20Hz_40KHz 3dB
等效输入噪声：话筒 60dB增益 <-125dB
总谐波失真：+20dBu into 600 <0.07%
技术指标：
· 输入阻抗：
o 话筒 300Ω 或1200Ω
o 线路 20kΩ
· 频率响应：
o 60dB 增益 into 600 20Hz_20KHz 0.5dB
o 60dB 增益 into 600 20Hz_40KHz 3dB
· 等效输入噪声：话筒 60dB增益 <-125dB
· 总谐波失真：+20dBu into 600 <0.07%...
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北京麦田中旺数码科技有限公司
2022-06-07
![AMS NEVE DPD话放]()
-
价格面议
- 产品简介：QQ:148-0362713 TEL:186-006-07968 项目经理：博乐 供应销售AMS Neve全线产品！
Neve1073DPD结合了我们的两个世界着名A类Neve1073话放单元和灵活的数字输出功能192kHz的麦克风前置放大器，适合现代录音环境-全部容纳在一个一U机架设备中。1073DPD这款NEVE话放是1073的升级版，它们的内部构造和电路制作艺和经典的1073话放一样。
1073DPD是由两个Neve 1073麦克风前级加上一个*级类比/数码转换器组成 的,提供了多个数码格式的高素质音响。在着名的Neve A 级输出点上加上一系 列的数码功能选项包括高达192kHz采样频率的PCM。*特别的是Neve DSD 输出功能。类比iert点被组装在类比/数码转换器之前.**的类比电路设计加 上**数码音响品质使1073DPD理所当然的成为数码编辑工作台的录音前级。
1073具有低指标的本底噪声和失真。但它的特点却是*强的过载承受能力。输出级别达到了+26dBu（600欧姆），这已经达到了模拟设备的承受极限，后级模拟设备一般很难承受这么高的输入电平。充分发挥了模拟设备动态大的优势（远大于数字设备）。另外1073DPD的模数转换电平可以在机体的*部进行选择，具有+18dBu和+26dBu两个档位，对于PP值出现频率比较低的音乐信号，往往可以考虑采用+18dBu的档位，这样可以更多的降低量化噪声的含量。
技术参数：
1073 DPD
频率响应: 48kHz +/-1.25dB 106dB
规格:
1073DPA & DPD 为1U机身, 19” 长的标准可入机架式机身设计.
高度 . 宽度 重量
mm (inches) /mm (inches)/ kg (lbs)...
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北京麦田中旺数码科技有限公司
2022-06-07
![AMS Neve DPX 立体声话放+EQ]()
-
价格面议
- 产品简介：QQ:148-0362713 TEL:186-006-07968 项目经理：博乐
供应销售AMS Neve全线产品！
1970年推出的经典，不仅1073很快成为良好的生产商和艺术家的可以选择麦克风前置放大器。发现了一些过去40年的记录，较着名的1073大，有力的声音补充任何音乐流派–从摇滚到说唱乐的经典。质量是传奇。设计和建造，在英国，现在不1073dpx生产与配套的零部件来保证声音的原始确切的规格是真的。1073dpx具有1073类设计的麦克风前置放大器的两个通道，每3频段均衡器（固定高频加二切换带切割/升压能力）和高通滤波器。这是1073dpx MIC前置和情商的结合，增添了温馨和深度记录，带来了微妙的气氛，保持空间定位，捕捉***的图像–体现，认为许多工程师和生产者，真正的没有声音的本质。在经典的1073个功能模块的建设，1073dpx介绍集成的DI输入，幻象电源的额外功能，可选的插入功能，可选择耳机监听和I/O连接器的每个通道的麦克风和乐器的简单和直接的联系。机架式2U单元包括一个外部的多电压电源。
特征
?少见marinair?变压器用于每个输入和输出阶段。
?方便连接线、双麦克风，前面的单元输入，可切换麦克风/线路输入连接在后面的单位
?切换+48V幻象电源，高/低阻抗，-20dB垫和地面电梯的输入阶段
?插入发送和返回回路特性允许外部设备容易被连接的1073dpx音频链，如单声道/立体声限制器/压缩机；?EQ前插入发送/返回线水平将可从麦克风前置放大器的输出级阶段回归
?后EQ插入发送/返回一个线路电平输出发送可从后均衡阶段和之前的主要输出阶段的回报
?独立通道输出电平控制
?7级LED信号电平表为每个通道可以显示麦克风/线路/ DI输入阶段，或输出阶段的水平对+24DB
?耳机输出可调音量控制，可以听1频道，2频道或两者相结合的渠道
?外部多电压电源...
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北京麦田中旺数码科技有限公司
2022-06-07
![AMS Neve 8801 通道条]()
-
价格面议
- 产品简介：QQ:148-0362713 TEL: 项目经理：博乐
供应销售AMS Neve全线产品！
在尼夫8801通道就像是拥有了一个完整的通道从传说中的不88r控制台–广泛认为是其中较好的模拟音乐游戏机开发过。包含在一个单一的机架单元，8801结合了经典的尼夫麦克风前置放大器和均衡达到了惊人的88R式动态处理器电路。手工建造，这一突破通道提供了一个前端的所有质量和多功能性，已经成为了永远的代名词。
8801的信号路径由我们的质量非常高尼夫麦克风前置放大器，高通和低通滤波器，四阶高音乐均衡，压缩机，门，动态侧链和插入点。信号路径的命令可以很容易地修改为生产者要求。
过滤段和/或均衡部分甚至可以被路由到动态侧链定制任何输入信号的动态响应。一个可选的Neve设计的模拟数字转换器可以容纳所有的标准采样率高达192kHz，以及直接DSD转换。听力测试中，许多工程师和生产者**考虑这些不设计转换器声音独立转换器成本*过整个8801单元。
*的可与（MAC和PC兼容）允许8801将与工作站的无缝集成，提供各种外部设备设置瞬时记忆。创造力的重要工具，真正的设计声音质量的基于计算机的系统的灵活性。很简单，8801是给任何工作室较终尼夫模拟声音的较方便的方法。
集锦
?麦克风前置放大器具有高和低通滤波器
?4波段均衡与高度的音乐，的声音
?88式压缩机和门
?插入发送和返回
?EQ或过滤器侧链
?USB端口直接连接到PC或MAC
?可选尼夫模拟数字转换器...
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北京麦田中旺数码科技有限公司
2022-06-07
![AMS Neve 8803 EQ]()
-
价格面议
- 产品简介：Q TEL: 项目经理：博乐
供应销售AMS Neve全线产品！
均衡电路来自于经典的旗舰88模拟控制台
全手工制作在伯恩利，英格兰
+ / - 20dB输入调整
高通滤波器的25Hz 300Hz 12dB每八度
低通滤波器3kHz到15kHz 12dB每八度
EQ高频1.5～18KHz，可选择的Q值
EQ中高0.8khz到9kHz，Q范围：0.3到7连续变量
EQ中低120Hz到2kHz，Q范围：0.3到7连续变量
EQ低频440Hz 33Hz，可选择的Q值
均衡器旁路开关
通过USB端口MAC和PC控制软件设置记忆模式
XLR平衡话筒及线路输入组合接口，+4dBu，-10dBv工作
XLR平衡输出
外部100v-250v电源...
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北京麦田中旺数码科技有限公司
2022-06-07
![Vertigo Sound VSC-2 压缩器]()
-
价格面议
- 产品简介：Q TEL: 项目经理：博乐
供应销售Vertigo sound全线产品！ 中国少见代理**！
The Mercedes of VCA compresso from German manufacturer Vertigo Sound! The VSC-2 Quad Discrete Compressor is so named because it uses 4 VCA’s built by hand using only discrete components. It’s design is a nod toward the best VCA based compresso of the 70s and 80s.
The Vertigo VCA is in fact named 1979. Some of the best features of these classic Compresso have been complimented in the VSC. This classic based sound is combined with a modern mastering grade signal path. Some unique and innovative features lie beneath the bonnet, such as the ratio which increases with level in ‘Soft’ mode.
VSC-2 Unique Features
?One for all! True Mono and True Stereo
?Dual - Mono, no shared contols
?In Stereo Mode Channel A controls become master
?The Stereo signal is phase coherent under all settings
?No phase shift or L-R problems can occur due to the innovative SC design
?The Signal path is of Mastering quality
?Real stereo mode! Channel A is stereo master
?There is a minimum of circuitry with a hard relay bypass
?Settings from Soft Knee (Soft=TipToe) to Hard Knee (Brick)
?The Ratio selection switch compeates to provide equal loudness, allowing immediate comparison
?Side chain High Pass Filte with two switchable frequencies are provided onboard
?Metering is analogue with carefully chosen ballistics, and beautifully lit!
?Handsome parameter control via customized scale ranges
?Internal components are of the ultimate quality, modified where necessary
?A Toroid traformer eures no PSU noise
?The ‘1979’ VCA is made of individual components, no Integrated Circuits
?Encased in a ceramic material for thermal stability
?This 1979 unit adds character to the sound, similar to some Vintage units;
?The harder it compresses the more it colou
?This VCA delive Real Analogue punch, so difficult to achieve with modern software plug-i
?A Jeen traformer and a state of the art line-out driver complete the picture by providing perfectly balanced In and Outs
The heart of the VSC-2 – the discrete VCA 1979
99% of all newer desig use an Integrated Circuit for this job which makes production easy and cost exteive.
Here's the big difference!
The Vertigo Sound 1979 is a fully discrete handbuilt V-oltage C-ontrolled A-mplifier.
This VCA colou the signal the more you compress and 'make up' the output.
This basic effect plus a very well thought out circuitry and choice of components gives you a fresh, punchy precise sound without sounding dull or nasty even if the unit is pushed hard.
The precision of this technique makes the Vertigo ideally suited for Mastering and Mix Buss compression.
Mono Mode is the exact opposite. Each channel has entirely separate control and metering. Crosstallk is less than minus 100dB, like two separate units!
This provides two world class CompressoLimite for Tracking and Mixing.
Technical Specificatio
?Stereo / Dual Mono switchable
?Jeen Balanced In. That 1646 or Burr Brown Balanced Out
?Dynamic Range: 115 db
?Frequency respoe: 10 Hz ... 70 kHz (- 3 db)
?Max. Output Level: + 25 dBu / 600 Ohm balanced floating
?Signal to Noise Ratio at + 6 dBu = 97 db (20..20 kHz, unweighted, RMS)
?Noise: – 91 dBu (20 Hz – 22 khz – unweighted, RMS) @ 0 db Unity Gain
?Crosstalk between channels: > 100 db at any frequency...
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北京麦田中旺数码科技有限公司
2022-06-07
![VERTIGO SOUND VSM-2 MIX SATELLITE MK2 母带处理器]()
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价格面议
- 产品简介：QQ:148-0362713 TEL:186-006-07968 项目经理：博乐 中国少见代理**
The Vertigo Sound VSM-2 is a creative fully analog Mix and Mastering tool. The VSM offe MS, Blending and two sophisticated coloration generato in 2U size.
The VSM-2 is also available in a “basic veion” without harmonic generato. This represents a 'All In One'-solution for Mastering studios with no more need to invest in mastering cooles, switching devices, monitor controlle, MS En & de-code etc.
Like all Vertigo Sound products, the VSM-2 is manufactured to the highest industry standards to guarantee the utmost performance and maximum durability. The VSM-2 Mix Satellite (MK2) now comes with new potentiomete. The noise floor is 4dB lower than the already very low noise Mk1 veion. All controls are stepped excluding 2nd/3rd Drive (Full Veion), Mix, level & shape.
FEATURES
VSM-2 Basic:
?All analog, latency free processing
?Two fully balanced ierts on XLR offering MS En & DE-coding
?Level compeated MS En & DE-coding
?Iert 1 provides Dry-Wet Mix control 'Parallel Compression' for LR and MS
?Iert 2 provides 'M&S Stereo Width' control
?Level compeated AFL Solo for M and S-Signal
?All controls can be easily recalled via stepped controls, pointer knobs & fine scales
?Hardwire bypass for all and each processing stage
?Excellent headroom with up to +30dbu at the I & Outs
?Level matching of the inputs and outputs
?Fast input and output metering plus various clip leds
?Color coded switch controls for easy intuitive operation
?Superior crafted & anodized front plate in 'sun orange' & blue metallic silk screen print
VSM-2 'Full Veion' additional features:
?Two independent stereo coloration circuits
?Stereo & M or S Generation of pure 2nd & 3rd harmonics
?Each coloration generator is independently switchable to M, S or LR
?Excessive harmonic filtering possibilities for each harmonic generator
?RMS LED level control for equal level of harmonic generato
?Unique 'Track' mode for processing individual itruments
?Soft & Hard clip switching capability
The VSM-2 a.k.a. Mix Satellite offe plenty of hugely useful functio for mixing and mastering in one unit. Of coue in daily studio life, even in the wot case scenario, you won't need all the functio of the VSM-2 at once. Carefully spot the problems you want to fix and choose the appropriate processing technique of the VSM-2.
The Mix Satellite allows you to try different processing strategies without any loss of time!
You can switch between conventional Stereo and MS processing within seconds or you can blend the Mix with the unprocessed signal by dialing only one knob.The Solo M&S Function allows you to listen closely to M or S and to optimize your processing setting!
Technical progress and purity of analogue and digital equipment effectively removed distortion as a stylistic device from the audio enginee' tool repertoire of today.The VSM-2 Harmonic generato allow you to colour, saturate, enhance, compress, limit or distort your program material! Now you are able to put real analog life into 'dead sounds' out of the box. All paramete to create your own kind of musical coloration are on the hand with the Mix Satellite VSM-2!
SPECIFICATIO
?All I & Outs electronically balanced
?Dynamic Range: 128 db
?Frequency respoe: 10 Hz ... 100 kHz (+- 1 db)
?Max. Output Level: + 30 dBu
?THD at +28dbu <= 0,003%
?Signal to Noise Ratio at + 6 dBu = 104 db (20..20 kHz, unweighted, RMS)
?Noise: – 98 dBu (20 Hz – 22 khz – unweighted, RMS) @ 0 db Unity Gain
?Power coumption: max 30 Watts...
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北京麦田中旺数码科技有限公司
2022-06-07
![VERTIGO SOUND VSP-2 双通道话放]()
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价格面议
- 产品简介：QQ:148-0362713 TEL:186-006-07968 项目经理：博乐 中国少见代理**
The VSP-2 is Vertigo Sounds 3rd big impact design and is setting a new benchmark in microphone amplification.
The heart of this unit is Vertigo Sound ** discrete Twin Op Amp circuit called the OPA (operational amplifier) 1976.
These 'in house' engineered & built discrete amplifier stage is combined with Jee world famous JT-16 Mic traforme and provide a smooth warm and defined sound from the lowest bass to the very top end of the sonic spectrum.The JT-16 traformer can handle very high input levels and provides a very linear phase behaviour because of low capacitance problems due to it`s low ratio of 1:2. Further the low ratio of the traformer forces the discrete OP Amp to provide most of the amplification in the VSP-2. This ' full throttle','pedal to the floor' circuitry enables a unique, rich, full and characterful sound.
Controls:
Gain control: 10-position Grayhill Rotaryswitch with 6db step increments. The full range of the unit is 12-72db
Fine Gain control: 41 click Alps 0-6db fine gain trim
120Hz/off/80hz: Switch that engages lowcut input filter 12db/oct at 80hz or 120 HZ.
Signal & Clip LED: This LED either indicates Green for a reasonable signal level presence at the output or red for close to Overload. There is no inbetween orange state to indicate clipping clearly.
LED green: - 6dBu to 19dBu at the output of the VSP-2 >>>> reasonable output level
LED red: +19dBu at the output of the VSP-2 >>>> close to clipping level
Note: The maximum output of the VSP-2 is +24dbU, the red clipping led indicates the maximum capable input level of most common converte and analog gear.
ITRUMENT IN: This is an unbalanced input for guitar, bass, samples, etc. This unbalanced input goes directly to the TWIN OPA input without using any circuitry inbetween. (Real DI Input) Simply throw the toggle switch and you're off. The output is muted when switching to avoid loud clicks in your monito.
48V : 48V switches ramped phantom power.
-20dB: Pads the input signal 20dB down. The pad does not affect the input impedance of the VSP-2! The pad circuitry is switched out of the Input stage completley when disabled.
Please use the Pad for the gain Settings 12dB-18db-24dB and 30dB only!!! The Pad does not effect the sound of your signal but settings like 54dB with Pad engaged e.G. will woen your signal to noise ratio.
PHASE: Revees the polarity of the pre amp's output.
The heart of the VSP-2 – the discrete OP AMP 1976
99% of all newer desig use an I-ntegrated C-ircuit for this job which makes production easy and cost exteive. Here's the big difference!
The Vertigo Sound 1976 is a fully discrete operational amplifier.
This very well thought out classic circuitry together with the Jeen JT16 traormer and Panasonic Coupling Caps plus the specific choice of components of the OP Amp gives you a fresh, punchy precise colourful sound without sounding dull or nasty at anytime....
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北京麦田中旺数码科技有限公司
2022-06-07
![Lexicon PCM96 Surround - Digital 混响器]()
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价格面议
- 产品简介：QQ:148-0362713 TEL:186-006-07968 项目经理：博乐
供应销售Lexicon 全线产品！
Lexicon PCM96 Surround - Digital I/O
在pcm96建设成功，新的词汇pcm96环绕提供了更多的预设，更多的配置选项，以及更多的输入和输出。给你的pcm96环绕混响效果的行业标准，具有*大的灵活性。使用pcm96环绕作为一个插件和你的地位，或者把它连接到你的混合板。无论哪种方式，您都有多种配置选项可供选择，而不必移动任何电缆。
的pcm96环绕可以与6 XLR立体声AES输入和输出，或两个DB25六通道模拟量I/O和1个DB25六通道数字I/O两版本还具有MIDI，字时钟，以太网，和火线?。
特点：
?可在两种模式：?六通道XLR / AES输入和输出
?两DB25六通道模拟量I/O和1个DB25六通道数字I / O
新并行立体声和环绕配置
50?传奇Lexicon混响，延迟，和调制的影响
词汇的新的环绕室算法，用于产生一个巨大的空间阵列相关的音乐和职位的影响
?新球场的算法9无缝自动化和控制通过以太网或火线
?*过2200工厂预置，包括从语音词汇的大量藏书被公认的经典作品
?硬件插件在MAC?VST音频单元?，能力，或安排软件
32位浮点处理
?高达96kHz的取样率
?外部BNC字时钟
?MIDI，出来，并通过...
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北京麦田中旺数码科技有限公司
2022-06-07
![Bricasti Design Model 7 M7混响器]()
-
价格面议
- 产品简介：QQ:148-0362713 TEL:186-006-07968 项目经理：博乐
供应销售Bricasti 全线产品！
一直以来，好的单机版硬件混响效果器，表现总是比较好的插件优异，然而，这个领域的选择总因着数位设备很高的研发费用，而难免受限于少数几个厂牌与型号。
Lexicon 224's, 224XL's以及480L's至今仍很**，这些经典机种其绵密与具有包覆感的声音特性，至今仍无法取代！新的机种像是TC M4000以及System 6000或是Lexicon 960，受人喜爱也相对的安静，但残响总让人感觉是浮在讯号的而并不是变成讯号的一部分。对于各种不同的应用场合来说，这些机器通常表现很优异，但对于某些特定场合来说就暴露了其缺点，特别是对于直**录的鼓以及人声。
Bricasti是一个新的公司，这名字对于某些人来说也一定很陌生，不过他的血统却很，这个公司较早是起源于负责设计Lexicon经典效果器的团队，着名的480L也是出自他们之手，他们现今的设计仍然强烈保留住当年的个性，但具备数位的输入以及输出端子，以及*友善的人机介面。
Bricasti M7因其源于经典Lexicon的血统，在这一台流线型的1U机身当中，具备令人怀念的丰富且深沉的残响，它编辑的功能很广泛，但具有逻辑且操作简单，而工厂的预设值涵盖了许多组非常实用的效果。当年的Lexicon 480L设计师很自豪的说：“Bricasti M7不只是来简单模拟或者替代Lexicon 480L，它有自己的声音”
我们将这机器与Lexico - PCM91以及经典的224(Lexicon较**的残响之一)比较，虽然某些虽然有些原厂预设像是PCM91的重组，但Bricasti的声音却*为丰富，具备许多现代残响的包覆感。很明显的，残响的尾巴*平滑自然，表示它的处理能力*上一层楼。当我们用 M7和Lex 224对比，它的表现*为耀眼，许多基本的预设值都不容易辨识，但两者都具备丰富且深沉的声音特性，不过Bricasti*安静，要说到编辑的能力那就根本无从比较，M7在各方面都胜出，从各种数位输入到广大的内建预设以及编辑所有参数的能力都是。
很少有制造商会对新的机种自我宣扬，但Bricasti M7确实如此。它是一台良好的机器，并且很快就会在发烧音乐；后期制作以及广告录音室等许多不同场合中被广泛使用，如果你打算购买一台**的残响，并且只打算买一台，那么Bricasti M7**是可以选择，因为它具有**的音质和相对友善的价格。
预设效果特性介绍
?100组少见的预设效果
?革命性的新残响演算法：True Stereo Reverb Process
?经典与**的预设效果; Halls, Plates, Rooms, Chambe, Ambient Spaces
?每组效果具备12组预设可调参数
?100组使用者设定效果
?4组面板可自行设定的“Favorites”的较常使用效果，可快
?比较以及切换
硬体设计特性
?双独立电源供应
?专为类比电路设计的独立环形变压器
?专位数位电路设计的高表现切换式变压器
?数位与类比模组分离
?抗腐蚀不锈钢外壳
?面板、旋钮以及按键采电镀铝壳处理
?Postive feel 2 db拨段式类比输入控制
?MIDI I/O
?美国手工制造
Bricasti M7 是混响器的尖峰之作，你能说哪个话筒较好么，你能说哪一个话放每个大棚都有么，你能说哪个压缩就一定适合所有人么，都没有，但你能说哪个混响创造的效果是*的么，当然，那就是Bricasti。M7是公认的目前较好的混响器，他的优良的性价比让许多欧美、港台的大腕*和老师都趋之若鹜，很多老师原来工作台里面的TC Reverb 6000、Lexicon 960L等都在这短短几年之中替换成了Bricasti M7，而仅2万多就能获得*级的混响品质是以前想都不敢想的事情。
M7是由原来480L的团队开发，由Brian开创的*级美国**，他的设计理念就是永远不妥协，只要较好的声音，所以从他的电路板设计可以看到，每个混响类型都是独立的混响芯片，ADDA也采用了行业里较**的转换芯片，电源采用了双供电模式，杜绝干扰，这种设计试问还有哪个商业混响器能做到，很多人都说MaxxBCL里的ADDA好，或者某个设备的ADDA能强过声卡，但是你们用过M7的ADDA吗，你们知道M7的ADDA品质吗。真正用过的人就知道M7的数模转换品质是真正不亚于目前主流的ADDA。
混响的好坏店主就知道一点，不怎么样的混响器经常获得不了自己想要的声音，因为你要仔细调整每一个细节参数，而*级的混响器会让你觉得无论怎么调都是好听的，只是好听的方式不同，所以如果你可以有单独购买混响器的钱，M7就是你较好的选择，不要觉得它贵，因为它的品质已经远远*过了它的*，它的ADDA也起码值一万块了，你说它还贵么。...
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北京麦田中旺数码科技有限公司
2022-06-07
![Antelope Audio Zen Tour 声卡 音频接口]()
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价格面议
- 产品简介：Q TEL: 项目经理：博乐
The New Approach to Analog Circuitry
Pushing the Antelope Audio’s Zen Studio to the next level, the MP32 is a front-end recording device with 32-channel mic preamp that fits within two rack spaces. Each of the 32 Class A preamps have phantom power, and four channels can be utilized as Hi-Z itrument inputs. The MP32 has incredible headroom and an extremely open sound, which can create magnificent results when teamed with the up to 65 db of gain each channel can get.
Ultimate User Friendly Controls
Via the Mac and PC compatible control panel, MP32 use can control the mic gain levels, change input types and even recall presets. Thanks to an extremely quick set up experience, it’s easy to get started with the MP32 and immediately start tweaking each of the 32 mic pres. Every aspect of control is literally a mouse click away with the MP32, whether it be changing gain structures or creating new preamp settings, and the system also features a remote control for added ease .
Match Made in Analog and Digital Heaven...
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北京麦田中旺数码科技有限公司
2022-06-07
![Antelope Audio Orion 32 HD 音频接口]()
-
价格面议
- 产品简介：QQ:148-0362713 TEL:186-00607968 项目经理：博乐
Known for their incredible digital output, Antelope Audio has once again created another tool, the MP8d, which will propel your recording studio to the next level. The MP8d is an eight-channel mic pre, which comes packed with impeccable A/D conveion and preamps based on the brand’s ever-popular Zen Studio and MP32. Offering precisely accurate sounds with natural warmth and smoothness to recordings, the MP8d was designed to create the purest quality sound, no matter how hard one pushes signal levels.
Analog Veatility
The MP8d mic pres are combo XLT connecto, which accept both mic and line level signals. Hi-Z inputs are associated with channels 1 and 2, and are accessible via dedicated T connecto on the front panel. Ierts for connecting your favorite effects outboard gear are available on the rear panel. A high-end headphone amp is accessible from the front panel to provide the convenience of integrated monitoring. The analog connectivity is accomplished with a D-Sub 25 connector directly outputting the mic signals at line level, making MP8d a perfect companion device for Antelope's Orion32 and Zen Studio audio interfaces.
Digital Efficiency
A plethora of digital connecto offer a wide variety of optio, providing extreme flexibility and enhanced productivity. MP8d provides S/PDIF, AES/EBU, TOSLINK, ADAT and MADI connectio expand the product compatibility and allow for both easy connection to other outboard gear or DAW, as well as cascading two or more MP8d units, expanding the channel count.
Flexible Management
MP8d relies on the highly sophisticated yet easy to use software control concept developed for Orion32 and Zen Studio. The intuitive user interface with color-coded drag and drop signal routing, multiple mixe and powerful DSP engine with on-board effects, makes the device extremely flexible and suited to various recording applicatio. Programmable presets allow for easy and fast recall of preferred configuratio, while ergonomic aluminum knobs on the front panel provide precise mic gain adjustment in real-time.
Inputs
?Analog Inputs ?2 x HI-Z Inputs on T
?8 x MIC/LINE on XLR Combo
?2 x Ierts on T(Inputs 1,2 )
?Digital Inputs ?1 x Fiber Optic MADI
?Word Clock ?1 x 10M Atomic Clock input @ 75 Ohms 3Vpp on BNC 44.1 - 192kHz
?1 x Word Clock Input @ 75 Ohms 3Vpp on BNC 44.1 - 192kHz
?2 x Word Clock Outputs @ 75 Ohms 3Vpp on BNC 44.1 - 192kHz square wave signal
Outputs
?Analog Outputs ?8 x Lines on DB25 (8 channels)
?1 x Stereo Headphone out on T
?Digital Outputs ?1 x Fiber Optic MADI
?1 x S/PDIF
?2 x ADAT
?4 x AES/EBU on DB9
?1 x Toslink
?USB ?USB 2.0, Type B...
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北京麦田中旺数码科技有限公司
2022-06-07
![Antelope Audio Orion 32+音频接口]()
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价格面议
- 产品简介：供应销售Antelope全线产品！
世界** 1U 机架式 32-通道 AD/DA 转换器兼音频主时钟。这款新的设备支持 MADI 以及 USB 接口，时钟采用了 Antelope 的 64-位 Acoustically Focused Clocking (AFC) 技术。Orion 32 通过其定制的内建 USB 芯片支持 32-通道 192 kHz I/O 数字音频流，这为任意支持 USB 的数字音频工作站或计算机提供了非常简单的连接方式。通过其 Fiber Optic MADI I/O 接口，转换器可提供 96kHz 32 通道的音频，可以与任意适当的 MADI 设备进行连接。
Orion 32 还支持提供 16 I/O 通道的 ADAT 协议，*进一步的提供了与大量音频设备的兼容性。经过 8 D-SUB 25 I/O 接口的模拟信号会经过多通道转换器输入和输出。
主要特点：
?Antelope Audio **的 AD/DA 转换技术
?通过内建的 USB 芯片支持高达 192kHz 32 通道 I/O
?32 通道 I/O 通过 iber Optic MADI I/O 连接
?8 D-SUB 25 接口用于 AD 和 DA
?16 通道在 4 ADAT 输入和输出接口
?Antelope 的 64-位 Acoustically Focused Clocking 与 Atomic 输入
?Antelope 的 Oven Controlled Oscillator 时钟稳定技术
?4 个时钟蔬菜以及 1 个字始终输入
?5 个用于**简便的恢复喜好设置的预设
?1U 机架大小，15 瓦功耗，发热*低
?可用于 Windows 和 OS X 的方便的桌面应用
羚羊的卡是较近的大热门，音质属于Aurora档次之上，自带usb接口，又有32路模拟输入输出通道，自带ADAT、SPDIF，而且有较热门的MADI接口，这卡的时钟又*为*级，所以目前国外很多大师都换了这块卡作为自己的ADDA，较新到货的*级音频接口，一定不要**哦...
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北京麦田中旺数码科技有限公司
2022-06-07
![Antelope Audio Discrete 8 话放 音频接口]()
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价格面议
- 产品简介：供应销售Antelope全线产品！
对于那些拒绝接受妥协的音频质量有1000万个铷原子时基准发生器基于真实原子技术。10M是专门设计来吸引较识别音响和音响。
在实现较终的工具
模拟声音
铷原子钟是一个**的伴侣的等时线音频主时钟如OCX或ocx-v.当连接到等时线1000万，这些设备的自动切换从炉基晶体以优越的铷原子核。等时线1000万是实现模拟声音的良好工具。认为，1000万大概是较响钟”的生产。
一个坚固的机械化做工
原子钟是由一个非常坚固的机械结构的区别，使其耐机械冲击。在使事情简单易用的羚羊的传统，只是连接并打开电源；状态指示灯，一眼可见的，所有的信息，**表现在易于阅读的频率显示。
这是一款，为吸引具有较*级鉴赏能力的音响发烧友和音效的设计。
如果您的目标，是想把录音室的能力发挥到*致，那么您肯定意识到，100,000次跳动之后，误差比石英钟还要小的计时装置，能带给您的震撼！
特点：
·瑞士制铷核运算器，**度达到10亿分之0.03。
·*精密原子振荡器，稳定度为每1000年1秒的误差。
·铷核运算器型号为FEI5660 Compatible Stanford P10 Equivalent系列。
·带自动切换装置的双后备电源供应系统。
·8个10MHz的BNC输出端口
·与所有能**10MHz信号的设备兼容。
·提供*为简便的安装操作，只需要正确地连接线路，并提供电源
·坚固稳定的外形设计与构造
·可与同系列下的OCX，OCX-V和Trinity型号**搭配...
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北京麦田中旺数码科技有限公司
2022-06-07
![Antelope Orion Studio Rev 2017 MKII 音频接口]()
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价格面议
- 产品简介：供应销售Antelope全线产品！
在音频和视频格式的浮躁的世界，新羚羊音频三一将主时钟是从多年来的人群，与其他厂商只能渴望的那种特征。用*的功能和控制，新的三位一体是较好的和较通用的主时钟可在任何地方。还有，在三一如音频到384khz，调速只提供*特的功能，九个独立的发电机，和一个三显示！当然，这一切都是在一个直观的菜单的用户界面，就像你会期望从羚羊。
特征：
?*四代声学聚焦时钟采用64位DSP**到一个新的水平，在详细的聚焦声
对于*低抖动?恒温晶体振荡器或原子输入
?工厂校准优于+ / - 0.001 ppm的精度
?财富的巨大的灵活性，后面板连接
?综合音频齿轮拳击任何，任何
?原子输入极限性能
?三独立音频同步发电机上384khz，和调速控制+ / - 200每分
?三独立同步高清发电机，涵盖16种格式
?三独立和同时SD发电机，同时提供PAL和NTSC制式。
?全欧洲和美国拉UPS /下拉的SD和HD视频
?USB连接PC和MAC可以远程控制，加载/保存预置和固件升级
?双冗余电源，以增加可靠性的自动切换
炉控制
在等时线三位一体的晶体振荡器是在一个温度控制的烤箱，通常用于军事和卫星通信。这使振荡器在恒定的温度使值100倍比同类产品*稳定。
原子钟
三位一体的等时可以锁定到地球上较**的时钟：原子钟。工作室能真正与原子钟的功率达到**的稳定性，只有同步时钟能够锁定这一**的技术。
全新的64位设计
三位一体是许多年的时钟知道从羚羊的设计师之大成，是一种*为一体化设计导致*严格和*集中的声音质量，全新的设计。*四代声学聚焦时钟与64位DDS起着非常重要的角色，为什么让你的声音甚至三位一体转换器比以往*好。...
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北京麦田中旺数码科技有限公司
2022-06-07
![Apogee Symphony IO 8x8音频接口]()
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价格面议
- 产品简介：供应销售APOGEE全线产品！
Symphony I/O集合了Apogee 25年的设计精华是Apogee目前出品的音质较棒的多通道音频接口。模块化的设计，可根据你实际的需求选择扩展模块，有多达5种模块可选，每台设备有2个扩展接口。
Symphony I/O模块化的设计让人眼*亮，尽管他们不是*个这么干的人，但他们这次的设计和规模都远*以往，看得出这是一家有高瞻远瞩的公司对未来N年数字音频发展情况做出的判断，Apogee在下一盘很大的棋，这盘棋庞大到目前Apogee都无法掌控，他们的整个系列产品线和发展规划可以一直延伸到2011甚至是2012年。
Symphony I/O主要特性：
?* Apogee目前出品的音质较棒的多通道音频接口，24bit/192kHz，改进的C777主时钟
?* 在音频接口模式下就是 一个USB 2.0音频接口
?* 可连接 Symphony PCIe 卡或Symphony Mobile Express/34卡，通过CoreAudio联合使用
?* 可直接连接Pro Tools HD PCIe卡作为Pro Tools HD的音频接口直接使用
?* 独立模式下可作为一个AD/DA转换器使用
?* 支持64位内核
?* 4个Symphony I/O可连接到一个Symphony 64或Pro Tools HD2使用
?* UV22HR抖动处理
?* 可连接X-Symphony的X系列和Rosetta系列转换器
?* 所有输入/输出接口的通路都是灵活可调的
?* 有两个扩展位置，有5种不同的扩展输入/输出接口可以选择，包括带光纤接口的8路模拟输入/输出接口模块和8通道的话放模块
?* 每台设备较多可扩展到32通道模拟和数字输入/输出接口
?* 模块化的设计，根据你实际的需求选择扩展模块
?* 可支持PCI、USB和网线连接
?* 较*值的Hi-End级别音频接口
?* Matro 2控制软件
?* 可通过Pro Tools HD或Pro Tools的控台（比如ICON）来控制话放
?* Sbus、MADI连接、Eucon控制协议
Symphony I/O可扩展的5种扩展卡包括：
?* 8模拟输入/输出接口+8数字输入/输出接口（8月发售）
?* 8模拟输入/输出接口+8AES数字输入/输出接口（8月发售）
?* 8话放输入/输出接口（10月发售）
?* 16模拟输入+16数字输出接口（10月发售）
?* 16模拟输出+16数字输入接口（10月发售）...
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北京麦田中旺数码科技有限公司
2022-06-07
![Lavry AD122-96 MX音频接口]()
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价格面议
- 产品简介：供应销售Lavry全线产品！
QQ:148-0362713 TEL:186-006-07968 项目经理：博乐
Lavry AD122-96 MX
The Lavry Gold AD122-96 MX allows clarity even when recording at full scale. The audibly superior results obtained with the MX can be a real competitive edge when recording hot.
This beautiful piece of technology boasts extremely close inter-channel matching that extends to the lowest signal levels preserves the spatial placement and depth of the stereo image. Ultra-stable internal and external clocking circuitry virtually eliminates the non-musical distortion effects caused by jitter in conveion. With unique self-calibration circuitry, conveion accuracy is maintained over a wide range of operating temperatures, and the effects of circuitry aging are minimized.
Building on the magnificent track-record of the LavryGold AD122-96 MkIII, the AD122-96 MX retai the traparency of the MkIII with the additional benefit of even more graceful handling of clipping. The MkIII has a reputation for handling clipping with less hahness than other brands of converte, and now, the MX takes this quality to a new level.
For those who utilize clipping as a method for attaining the maximum audible level on their recordings, the AD122-96 MX provides a more musical result without the loss of quality typically associated with pushing the clipping envelope. Below the highest peak levels, the MX is as linear as the MkIII; so the best qualities of the MkIII are retained for the majority of the music signal. When recording at the level where the highest peaks are clipped, the MX offe increased musicality when compared to most other AD converte.
For the same level of detail and accuracy, the MX model retai the -127dB noise floor 20-20kHz and 0.00005% distortion for signals below -40dBFS as the AD122-96 MkIII. Extremely close inter-channel matching that extends to the lowest signal levels preserves the spatial placement and depth of the stereo image. Ultra-stable internal and external clocking circuitry virtually eliminates the non-musical distortioneffects caused by jitter in conveion.
With unique self-calibration circuitry, conveion accuracy is maintained over a wide range of operating temperatures, and the effects of circuitry aging are minimized over yea of use. In addition to the improved handling of clipped signals, Lavry’s exclusive selectable Soft Saturation also facilitates “hotter” recording levels without undesirable artifacts.
Soft Saturation is a digital emulation of analog tape saturation which leaves the signal completely linear below the threshold, with selectable 3 or 6dB of additional level. For signals above the threshold, the process has no attack and release time cotants which could cause undesirable “pumping” or “breathing” by interacting with compression or limiting applied either before or after conveion. The MX model offe the flexibility of the Soft Saturation modes or linear operation, with the ability to handle high levels of clipping without the loss in quality common to other converte.
Key Features
?Graceful handling of clipping allows more apparent level without loss of musicality
?24 bit conveion at 96kHz, 88.2kHz, 48kHz and 44.1kHz
?Unequaled detail and performance
?True -127dB noise floor
?0.00005% Total Harmonic Distortion + Noise
?Lavry’s exclusive αcoustic βit Correction? for wordlength reduction with dither and noise shaping
?Switchable digital soft-knee limiter, +3 or +6dB gain
?Precision reference meter bridge
?Fully programmable digital test tones for system alignment
?AES and word clock external synchronization 40-51kHz, 88.2kHz, 96kHz
?DC removal
?Absolute Polarity Inveion
?Multi-converter synchronization
PERFORMANCE
?Noise Floor at 44.1-48KHz sampling: ?-130dB ±1dB from full scale A weighted
?-126dB ±0.5dB from full scale unweighted
?Noise Floor at 96KHz sampling: ?-127dB ±1dB from full scale A weighted
?-123dB ±1dB from full scale unweighted.
?Output Bits: ?16 to 24 bits, user selectable.
?Total Harmonic Distortion and Noise (Specified for 22-24 Bits output, balanced inputs): ?0.00005% for signals lower than -40dBF
?0.0004% for signals lower than -20dBF
?0.0008% for signals lower than -6dBF
?Channel Separation: ?Greater than 120dB (0-20KHz)
ANALOG INPUT
?The analog signal can be amplified by 0-12dB utilizing the front panel screw adjust trim pots. Full-scale signal with 0dB gain is 24dBu, (full-scale signal with 12dB gain is 12dBu).
?Input impedance over 0-100 kHz: greater than 20k ohm (pin 2 to pin 3).
?XLR connector: AES/EBU levels into 110 Ohms for digital audio or TTL level into 110 Ohms word clock
DIGITAL OUTPUT
?XLR AES/EBU output traformer coupled. Compatible with most RCAPDIF inputs using simple wired adapte or adapter cables.
SYNCHRONIZATION
?BNC connector: ?TTL level word clock
POWER
?90-264 VAC @ 47-63Hz , Fuses 0.** Slow 250V Fuse Size 0.25inch x 1.25 inch.
?The power supply automatically adjusts to AC inputs in the range of 90 to 264 volts at line frequencies between 47 and 63 Hz. It is not necessary to change any settings.
PHYSICAL...
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北京麦田中旺数码科技有限公司
2022-06-07
![Lavry AD122 96-MKIII音频接口]()
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价格面议
- 产品简介：供应销售Lavry全线产品！
QQ:148-0362713 TEL:186-006-07968 项目经理：博乐
Lavry AD122 96-MKIII
The Lavry AD122-96MKIII converts analog signals to a true 24 bit digital audio data stream. Coistently used for many of today's top recordings, the AD122-96MKIII features -127dB dynamic range (unweighted), providing optimum musical results.
Features:
?96kHz, 88.2kHz, 48kHz and 44.1kHz conveion frequencies
?Unequalied detail and performance
?True -127dB noise floor
?0.00005% total harmonic distortion + noise
?Conveion to 24 bits
?Built-in acoustic bit correction re-dithering to 16-20 bit formats
?Switchable digital soft-knee limiter, +3 or +6dB gain
?Precision reference meter bridge
?Fully programmable digital test tones for system alignment
?AES and word clock external synchronization 40-51kHz, 88.2kHz, 96kHz
?DC removal
?Absolute polarity inveion
? Multi-converter synchronization
Achieving 20 or more bits of resolution places a heavy demand on the linearity and noise of analog signals prior to the A/D conveion.
Linearity
Linearity problems (typically associated with larger signals) should be avoided to prevent overall signal distortio. Proper circuit design places non linear devices iide a negative feedback path, thus allowing the trafer function to depend on a few linear component (such as resistor ratio). Closer examination reveals that the effectiveness of negative feedback is limited by the open loop gain of the amplifier (the available gain prior to the feedback action). Cotructing a 10dB gain stage out of, say, a 50dB open loop amplifier provides 40dB of feedback; thus circuit nonlinearities are corrected for by a factor of 100. A 1% nonlinearity reduces to a .01% level. The same stage, when set to a gain of 20dB, yields only 50-20=30dB of 'linearizing power'; thus the same 1% circuit nonlinearity is corrected to 1/31.6 = .03%. The 'obvious' solution of increasing circuit open loop gain is riddled with problems ranging from impulse respoe settling time to loss of bandwidth.
Noise
The quantization (A/D conveion) of analog noise causes loss of fine signal detail. The quantized noise introduces time varying signal dependent distortio. Such sample by sample deviation can not be compared to the seemingly 'cotant' analog background noise. The interaction between the noise and the quantization process effects the sonic detail at much higher amplitude than the noise level itself. The source material may have its own noise level, but the A/D manufacturer is in no position to decide which part of the material is signal and which part is noise. (Is it undesirable noise or is it the intentional recording of the wind blowing through the trees?) Thus, any signal must be reproduced accurately. All we can do, and all we should do amounts to avoiding additional A/D noise. A/D noise coists of two parts: analog noise (prior to quantization), and quantization noise. The quantization noise level is 'fixed' by the converter itself. Model AD122-96 MKIII quantization noise relative full scale is about 122dBFS. Keeping complete system noise level at 122dBFS requires careful treatment of the analog path. Each circuit component (especially resisto and semiconducto) generates its own noise. The complex electron motion within the various materials adds up to a cumulative noise. Intentional amplification of signals yields undesirable amplification of noise. Once again, each dB of added gain takes away a dB from the available dynamic range.
AD122-96 MKIII Input Range...
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北京麦田中旺数码科技有限公司
2022-06-07
