TBProAudio Bundle v2025.10_WIN-CE.VR

套装效果器插件TBProAudio Bundle 2025
VST插件格式：
VST/VST3/AAX

TBProAudio Bundle 2025专业音频插件包，集成动态均衡器插件、响度匹配插件、通道条插件及多通道计量工具，提升混音母带效率与音质。
此安装包为完整版的TBProAudio插件，插件的简介并不完整，详细内容请访问官方网站查看。

翻译：
TBProAudio 开发的音频插件可帮助音乐制作人更快、更好地完成工作。我们采用最新的算法和标准（如 EBU R128），使我们的客户能够提供最佳品质。

包括
ABLM2 – AB 响度匹配插件
AMM2 – 自动混音器插件
CS-5501V2 – 扩展通道条插件
dEQ6V4 – 动态均衡器插件
dpMeter5 – 多通道测量仪插件
dpMeterXT3 – 多通道测量仪插件
DSEQ3 – 动态频谱均衡器插件
DynaRide2 – 高级增益插件
Euphonia3 – 频谱匹配插件
FinalLoud3- 响度匹配插件
GainRider3 – 人声增益均衡器插件
gEQ12V4 – 12 波段图形均衡器插件
GSatPlus – 饱和器插件
Impress3 – 宽频立体声压缩器插件
ISOL8-混音监控工具
LAxLimit4-宽带限幅器插件
mvMeter2 – 多变量表插件
SLM2V2 – 智能响度最大化插件
ST1V2 – 空间工具
sTiltV2-均衡器插件
TBPA Move-滤波器插件
TBPA Volume – 音量增益插件

1.ABLM2
响度匹配
ABLM 是一种增益控制，可对特效链进行电平匹配和采样精确的 AB 对比。

大多数 DAW 都能轻松启用/禁用（AB）单个插件甚至整个特效链，但在大多数情况下，两种情况下的响度会有明显差异。为了客观地比较插件链的影响，输入信号的响度应与输出信号的响度一致。

在响度相同的情况下比较插件效果，可以避免 “响度越大越好 “的误区。

功能特点
简单的图形用户界面
匹配前置和后置特效响度
将前置效果响度与参考电平相匹配
3 种响度匹配模式：手动、自动和增益阶段
特效链驱动
过载保护
多达 256 对发送器/接收器，包括自动通道选择
自动延迟测量/补偿
6 种测量模式 有效值总和、有效值平均值、EBU R 128 SL、EBU R 128 ML、VU、最大峰值
可调有效值测量窗口
旁路 ABLM
4 个快照插槽
对 CPU 非常友好
详细信息：https://www.tbproaudio.de/products/ablm

2.CS-5501V2
通道条
CS-5501 是一款通道条插件，具有双均衡器、门/扩展器、去噪压缩器、限制器和饱和度，以及灵活的模块路由和超采样功能。CS-5501 是非常成功的 CS-3301 的继承者。它保持了 CS-3301 的流畅音质，并在所有跟踪和混音情况下提供了更大的灵活性。

功能特点
2x 7 段均衡器
2 倍噪声门/扩展器
2x 压缩器，VCA/FET/OPTP 设计
2 倍修整器
2x 限幅器
饱和
模拟声音的热噪声
均衡器容差模拟
HQ 过度采样
信号过冲保护
可链接的输入/输出推子，包括相位反转
模块参数链接
扩展计量、输入/输出/增益降低、峰值、有效值、EBU 和 VU
示波器
频谱分析仪
感知响度匹配（由 AB-LM Lite 支持）
灵活的 AB-LM Lite 路由选项
模块路由
灵活的剪辑模块位置
为许多效果模块提供广泛的侧链支持
多种舒适功能，如独奏和信号监控
预置管理
参数随机化
可选颜色主题
大型且易于使用的图形用户界面
撤消/重做
自由缩放图形用户界面
64 位内部处理
详细信息：https://www.tbproaudio.de/products/cs-5501

3.DSEQ3
动态均衡器
DSEQ 是一种在频域中工作的动态处理器。DSEQ 通过自动调整频段，可在运行中消除数字声音的粗糙度。
DSEQ 可用于所有混音和母带情况：

– 消除人声
– 消除鼓、吉他、人声等录音中的共鸣
– 消除数字噪音
– 平衡混音
– 消除频率区域屏蔽
– 支持粉红噪声混音/母带制作

功能特点
智能 AI 功能、
智能 AI GUI 页面
自适应斜率
自适应阈值
平滑线性/自然相位动态滤波器
全频率范围处理，10Hz – 22kHz
7 种不同的质量模式：生态环保、生态、正常、高和超、超 2/3
左/右和中/侧处理
检测器信号的全局斜率（如粉红噪声混音/母带处理）
自定义阈值曲线
增益降低限制
压缩控制
自动计算最佳斜率参数
自动计算最佳阈值参数
12 个独立的预滤波器频段，模拟/数字风格
支持侧链
参数 A/B
撤消/重做
多种信号监测模式（预滤波器、delta、侧链）
多种频谱分析模式（立体声、左、右、中、侧）
感知响度匹配（由 AB-LM Lite 支持）
扩展输入/输出表，峰值/有效值/EBU/VU
窄带扫频模式
所有处理模式的采样旁路切换功能
最小采样率高达 192 kHz
“真实 “超采样，高达 4 倍
离线渲染模式
增益降低光谱仪
均衡器小抄
连续计算频谱斜率
精确的参数输入
易于使用的图形用户界面
免费的图形用户界面缩放和大小调整
多种预设，包括智能 AI 设置
预置管理，包括自定义设置
完全支持 DAW 自动化
64 位内部处理
非常高效的 CPU 使用设计
智能静音处理
详细信息：https://www.tbproaudio.de/products/dseq

4.GainRider3
增益骑乘
Gainrider 是一款（人声）增益骑乘工具，可完全控制增益骑乘过程。它提供更多的骑乘算法，并减少插件延迟。

功能特点
3 种骑乘模式：响度调平、抑制、响度调平到侧链
详细控制增益变化（速度、预延时、最小、最大、空闲）。
灵活的预延时，包括零延时模式。
基于 EBUR128 ML 的响度测量
大型显示屏，可直观监测插件参数的影响。
灵活的预滤波器
侧链（VST3/RTAS/AAX/AU）或通道 3+4 (VST2)。
读/写自动化数据
A/B 控制。
详细信息：https://www.tbproaudio.de/products/gainrider

5.gEQ12V4
gEQ12 是一款 12 波段立体声/MS 图示均衡器，具有精确的频谱分析器和最小/线性相位模式。

当今的混音和母带制作过程需要一个功能齐全但易于使用的均衡器，它能准确地塑造音轨、母线或主音轨的音色。

gEQ12 提供了所有必要的工具，包括感知响度匹配和一些独特的功能。

gEQ12 的图形用户界面可完全调整大小和扩展。

功能特点
12 个独立滤波器频段，扩展频率范围 10Hz – 48kHz
16 种滤波器类型，模拟/数字设计，零延迟 IIR
滤波器斜率从 6 到 96dB
滤波器独奏
每个频段独立的立体声/左/右/中/侧处理
7 种处理模式：零延迟、最小/线性相位
最低采样率可达 192 kHz
“真正的 “超采样，高达 4 倍
带 32k FFT 的高清频谱分析仪
前/后均衡频谱、最大/平均/实时频谱、立体声/左/右/中/侧频谱
后置滤波器预测频谱
平滑频谱、ISO R40 平滑频谱和原始数据频谱
参考频谱、delta 参考频谱
加载波形/aiff 文件作为参考
均衡器小抄
频谱图
频率跟踪器
扩展输入/输出表，峰值/有效值/EBU/VU
感知响度匹配（由 AB-LM Lite 支持）
窄带扫描模式
频谱冻结模式
所有处理模式的采样精确 A/B 切换
精确的参数输入
撤消/重做
大而易用的图形用户界面
自由缩放图形用户界面
交互式均衡器显示
预置管理
完全支持 DAW 自动化，滤波器频率变化流畅
64 位内部处理
详细信息：https://www.tbproaudio.de/products/geq12

6.GSatPlus
饱和度
GSat+ 源自 TBProAudio CS5501V2 通道条插件上的饱和度模块，但增加了 Gearspace.com 团队要求和建议的额外功能。它有几种不同的饱和模式、高可调性、超采样、老式 VU 表和跨平台软件兼容性–还有一个按钮可让您直接访问 Gearspace.com，以备您需要任何技巧、窍门或灵感，或想与其他用户交换预置。

功能特点
具有三种不同 “基于电子管 “特性模式的饱和度
剪辑保护
VU 表
信号监控
过度采样
插件旁路
预置管理
大型且易于使用的可自由扩展图形用户界面
64 位内部处理
详细信息：https://www.tbproaudio.de/products/gsatplus

7.mvMeter2
多变量仪表
mvMeter2 是 mvMeter 的后续产品，增加了单/双仪表显示、所有仪表模式的可调参考电平、可调仪表延迟和预设管理功能。

特点
具有经典模拟 VU 和 PPM 表的功能
多种测量模式： 峰值、有效值、EBU R128、VU 和 PPM
多通道测量：立体声、左、右、中、侧
单表和双表显示
所有仪表模式的参考电平均可调整
可调仪表延迟
可调仪表保持
可调 OL 电平
预设管理
可调预增益，包括增益匹配
实例标签
大而精确的实时仪表
64 位内部处理
5 种不同的仪表主题
详细信息：https://www.tbproaudio.de/products/mvmeter2

8.SLM2V2
宽带限制器
SLM2 是一款智能立体声响度最大化器，包括 “模拟香料 “和 16 倍超采样。其设计目标是在保持透明度的同时，最大限度地提高音频信号的响度。

SLM2 DSP 在规定的时间内分析输入立体声信号的每个采样，并计算出最佳音量增幅。这保证了最大可能的透明度。

通过下面的可选削波级，可以进一步提高整体音量。通过启用过采样，可以减少信号混叠。
SLM2 适用于所有限制情况，如轨道/总线和主限制。

特点
高透明度砖墙式限幅器
可切换输入信号的饱和度
完全控制攻击和释放
可切换前瞻性
2 种限制器模式
2 种上限模式
附加驱动和剪切级
立体声链接
超采样高达 16 倍
易于使用的图形用户界面
预置管理器
大而精确的现场仪表
支持所有采样率
详细信息：https://www.tbproaudio.de/products/slm2

9.ST1V2
立体成像控制
ST1 是一款简单的工具，可让您控制立体声音频信号的宽度和全景。它包括单声道转立体声功能、立体声场增强器、包含频谱平移的丰富平移旋钮和低音聚焦功能。相位和相关性仪表可直观地监测平移位置和立体声宽度。

功能特点
相位表
相关表
可调式单声道到立体声转换器
可变立体声宽度
自动平移旋钮，带有可变平移律和多种舒适功能
频谱平移
低音聚焦
预置管理
多种实用预置
大型且易于使用的图形用户界面
自由的图形用户界面缩放
64 位内部处理
详细信息：https://www.tbproaudio.de/products/st1

10.AMM2
AMM 提供多种高级混音模式，并对每个通道进行大量控制。
AMM 可用于将一轮讨论、多个发言人的对话，甚至多达 32 个麦克风的音频会议快速混音为立体声信号。
AMM 支持多种通道配置，从单声道到 32 个立体声通道，具体取决于 DAW 的功能。

11.dEQ6V4
dEQ6 通过压缩器/扩展器控制每个频段的滤波器增益。根据节目素材的不同，它可以降低或提高特定频率范围内的增益。压缩器/扩展器的检测器信号由预置滤波器处理，该滤波器由动态滤波器的中心频率和 Q 因子控制。这样可以确保压缩器/扩展器只被指定频率范围内的内容激活。

12.dpMeter5
dpMeter 是一款精确的数字音频多通道仪表，包括 RMS、EBU R 128、对话框门控和 TruePeak（采样间）测量。它是广受欢迎的 dpMeter4 的后续产品，增加了图形用户界面大小调整和预设管理功能。

13.dpMeterXT3
dpMeterXT 是一个响度测量插件和应用程序，完全符合 EBU R128-2014、ITU-R BS.1770-4、ATSC A/85 和其他几个特定区域的响度标准。
dpMeterXT 支持从 1.0 到 9.1/10.0 的多通道配置，具体取决于您的系统配置或 DAW 功能。

14.DynaRide2
DynaRide 是一款高级增益附加插件，具有多个检测器程序、侧链支持和可切换的预延迟。最重要的是，DynaRide 可以通过将骑行过程限制为特定的音频信号（如语音、人声和低音）来增强输出信号。

15.Euphonia3
Euphonia 是一款最佳音频频谱平衡插件，可根据一些标准的平均频谱曲线自动均衡任何音频输入。

在音乐制作的最终混音后立即使用 Euphonia，可以消除低/高端缺失、混音错误、一般频谱平衡问题，甚至还能补偿糟糕的听音环境。

16.FinalLoud3
FinalLoud 能快速解决音频工程中一个非常常见的难题：为任何音频素材达到一定的目标响度水平，同时保持给定的真实峰值上限。

因此，FinalLoud 将增益控制、高质量的真峰值限制器和精确的响度表结合在一个易于使用的界面中。增益控制可将响度水平提升至目标水平，与此同时，限制器可确保达到真实峰值上限。

17.Impress3
Impress 是一款宽带立体声压缩器，包括侧链、
Impress2 是一款宽带立体声压缩器，包括侧链、各种预滤波器选项和最先进的超采样。

其设计目标是涵盖当今混音和母带处理中的各种压缩应用，重点是低混叠。因此，Impress 可用来塑造微妙的小军鼓音效，在乐器总线上粘合不同的音轨，甚至完成去噪或配音等复杂任务。

18.ISOL8
Isol8 是一款先进的混音监控工具。它将频率范围分为 5 个频段。这 5 个频段可以单独独奏或静音。这将帮助你在混音和母带处理过程中专注于特定的频率范围。

此外，你还可以将 ISLO8 用作灵活的多频段分离器，与复杂的插件链配合使用。分频后的信号最终可以混合在一起。Isol8 最初设计用于主音轨，但也可用于单个音轨总线。

功能特点
5 个可调频段
每个频段单独的独奏/静音功能
Linkwitz-Riley 分频滤波器设计
24/48dB/Oct 滤波器斜率
多种滤波器通道模式（立体声/左/右/中/侧）
多种监听模式（立体声/左/右/中/侧）
就地或居中监听
交换左/右声道
可调输出电平
响度调节功能
侧链监听
键盘控制
多通道分割
易于使用的大型图形用户界面
免费 GUI 缩放
64 位内部处理
GPU 和非 GPU 版本

19.LAxLimit4
LAxLimit 是一款先进的前瞻性宽带链接立体声限幅器，包括 ISP（采样间峰值）检测和超采样。
LAxLimit 可对瞬态和自适应释放曲线进行更多控制。它现在增加了扩展计量（RMS、EBU 和 Dialog 门控）。
设计的目标是涵盖当今混音和母带处理中的各种限制应用，并重点关注低混叠问题。

设计
LAxLimit 专为母带处理、数字编辑、多媒体以及任何需要以最高质量和最低混叠限制数字信号的应用而设计。LAxLimit 采用先进的前瞻性算法，可确保超快响应和无过射。

为了满足当今电视、广播和音乐制作的要求，LAxLimit 提供基于 ITU-1770 规范的真峰值限制 (ISP)，并在此基础上提供高达 8 倍的 “真实 “过采样。与真实峰值限制相结合，可达到最高的制作标准。

LAxLimit 包括一个先进的自适应释放控制系统，可大量减少固定释放时间造成的伪音。为了提高限制灵敏度，两个立体声通道可以自由连接/断开。

LAxLimit 可作为处理链中的最后一个插件使用（砖墙限制）。

特点
最先进的低混叠前瞻性限幅器设计
可自由调节的限制器特性
详细的瞬态控制
可自由调节的自适应释放曲线
电平检测模式：基于 ITU BS1770.4 的峰值和 ISP
有效值、EBU R128 和 Dialog 门控响度测量
“真实 “超采样，最高 8 倍
立体声链接
内置剪辑器
驱动模式
统一增益和三角监测模式
预置管理
中/侧处理模式
易于使用的图形用户界面
免费图形用户界面缩放
大而精确的实时仪表
所有采样率
产品需要使用许可证密钥激活
为 LAxLimit 客户提供免费更新
提供演示版

20.sTiltV2
sTilt 是一款线性/自然相位和 IIR 滤波器，可围绕给定的中心频率倾斜音频频谱。

特点
无失真处理引擎
斜率可调，从 -6dB/Oct 到 +6dB/Oct
中心频率可调
5 种音质模式：低、环保、中、高、最高
3 种滤波器模式：线性相位、自然相位和零延迟
单元自动增益
可切换夹子保护
采样精确 A/B 比较
立体声/左/右/中/侧通道选择
灵活的通道监控
输出增益
信号混合
免费 GUI 缩放
GPU 和非 GPU 版本

注：此插件会在音频路径中增加大量延迟，通常由 DAW（PDC）补偿。

21.TBPA Move
节奏滤波器
TBPAMove 由两个独立的滤波器和一个高分辨率示波器组成。
滤波器的每个参数（谐振、截止、增益和平移）均可由 DAW 自动控制和调制器控制。DAW 自动控制可以自定义（延迟、平滑和与默认值混合）。调制类型可以是侧链、LFO、步进音序器或多级包络发生器。

TBPAMove 是一款 DAW 插件，用于创建有节奏的滤波器、增益和平移效果。
TBPAMove 由两个独立的滤波器和一个高分辨率示波器组成。
滤波器的每个参数（谐振、截止、增益和平移）均可由 DAW 自动控制和调制器控制。
DAW 自动控制可以自定义（延迟、平滑和混合默认设置）。
调制类型可以是侧链、LFO、步进音序器或多级包络发生器。

特点：
2 个独立滤波器，带谐振、截止和增益平移控制
各种滤波器类型（如低通、高通、峰值和搁架）
串行和并行处理
8 个独立调制器
侧链调制器
LFO 调制器
步进音序器调制器
多级包络发生器
多种随机化选项
参数变化的 AHR 包络延时
平滑参数变化，减少混叠现象
VST3 采样精确的自动化数据处理
带有音频和参数包络显示的大型示波器
图形用户界面比例
64 位内部处理

22.TBPA Volume
TBPA Volume 允许您独立于混音台的音量推子控制音频信号的音量。这对于调整插件链中不同插件之间的增益/音量，或用于音量调制/自动化非常有用。

功能：
免点击 64 位内部处理
主音量推子、左/中和右/侧推子
左/右、中/侧处理
可调最小/最大音量值
将最小值视为负无穷大
将最大音量锁定为 0 dB
音量调制显示
智能电源按钮
平滑的参数插值
低混叠
支持精确采样自动化 (VST3/CLAP/AU)
预设管理
自由的 GUI 缩放
————————————————————————————————————
原文：
TBProAudio develops audio tools which help music producers to do their job quicker and better. We are using the latest algorithms and standards (e.g. EBU R128) to enable our customers to deliver the best quality.

Included:
ABLM2 – AB Loudness Match
AMM2 – Auto mixer
CS-5501V2 – Extended Channel Strip
dEQ6V4 – Dynamic equalizer
dpMeter5 – Multi Channel Meter
dpMeterXT3 – Multi Channel Meter
DSEQ3 – Dynamic Spectral Equalizer
DynaRide2 – Gain rider
Euphonia3 – Spectrum matching
FinalLoud3- Loudness matching
GainRider3 – Vocal Gain Riding Leveller
gEQ12V4 – 12-Band Graphic Equalizer
GSatPlus – Saturator
Impress3 – Wideband Stereo Compressor
mvMeter2 – Multivariable Meter
SLM2V2 – Smart Loudness Maximizer
ST1V2 – Spatial Tool
TBPA Volume – Volume Gain
ISOL8-Mix monitoring tool
LAxLimit4-Wideband Limiter
sTiltV2-Equalizer

1.ABLM2
Loudness matching
ABLM is a gain control which enables level matched and sample accurate AB comparison of FX chains.

Most DAWs can easily enable/disable (AB) single plug-in or even whole FX chains, but in most case there is an audible loudness difference between both cases. In order to compare the impact of a plugin chain objectively the loudness of the incoming signal should match with the loudness of outgoing signal.

Comparing the plugin effect at equal loudness levels avoids the “louder is better” – pitfall.

Features
simple GUI
match pre and post-FX loudness
match pre-FX loudness with reference level
3 loudness match modes: manual, auto and gain stage
FX chain drive
over protection
up to 256 sender/receiver pairs including auto channel select
automatic latency measurement/compensation
6 measurement modes: RMS Sum, RMS Avg, EBU R 128 SL, EBU R 128 ML, VU, Max Peak
adjustable RMS measurement window
bypass ABLM
4 snapshot slots
very CPU friendly

2.CS-5501V2
Channel strip
CS-5501 is a channel strip plugin with double EQ, gate/expander, deesser compressor, limiter and saturation, flexible module routing and oversampling. CS-5501 is the sucessor of the very sucessfull CS-3301. It keeps the smooth sound of CS-3301 and provides much more flexibility in all tracking and mixing situations.

Features
2x 7 band EQ
2x noise gate/expander
2x compressor, VCA/FET/OPTP design
2x dresser
2x limiter
saturation
thermal noise for analogue sound emulation
EQ tolerance simulation
HQ over sampling
signal overshot protection
link-able input/output fader including phase inversion
module parameter link
extended metering, input/output/gain reduction, Peak, RMS, EBU and VU
oscilloscope
spectrum analyzer
perceptual loudness matching (powered by AB-LM Lite)
flexible AB-LM Lite routing options
module routing
flexible clip module position
extensive side-chain support for many effect modules
many comfort functions like soloing and signal monitoring
preset management
parameter randomizer
selectable color theme
large and easy to use GUI
undo/redo
free GUI scaling
64-bit internal processing

3.DSEQ3
Dynamic equalizer
DSEQ is a dynamic processor working in the frequency domain.DSEQ removes digital harshness on the fly thanks to self-adjusting frequency bands.
DSEQ can be used in all mix and master situations:

– de-essing vocals
– taming resonances in e.g. drum, guitar, vocal recordings
– removing digital harshness
– balancing the mix
– de-masking frequency regions
– support of pink noise mixing/mastering

Features
smart AI function,
smart AI GUI page
adaptive slope
adaptive threshold
smooth linear/natural phase dynamic filter
full frequency range processing, 10Hz – 22kHz
7 different quality modes: eco eco, eco, normal, high and ultra, ultra 2/3
left/right and mid/side processing
global slope for detector signal (e.g. pink noise mixing/mastering)
custom threshold curve
limit of gain reduction
compression control
automatic optimal slope parameter calculation
automatic optimal threshold parameter calculation
12 independent pre-filter bands, analogue/digital style
side-chain support
parameter A/B
undo/redo
multiple signal monitoring modes (pre-filter, delta, side chain)
multiple spectrum analyzing modes (stereo, left, right, mid, side)
perceptual loudness matching (powered by AB-LM Lite)
extended input/output meter, Peak/RMS/EBU/VU
narrow-band sweeping mode
sample-exact bypass toggle for all processing modes
sample rates min. up to 192 kHz
“real” over sampling, up to 4x
offline render modes
gain reduction spectrograph
EQ cheat sheets
continuous calculation of the spectrum slope
precise parameter input
easy to use GUI
free GUI scaling and resizing
many presets including smart AI setups
preset management including custom setups
full DAW automation support
64-bit internal processing
very efficient CPU usage design
smart silence processing

4.GainRider3
Gain riding
Gainrider is a (vocal-) gain riding tool which gives fully control over the gain riding process. It offers more ride algorithms and reduces plugin delay.

Features
3 ride modes: loudness leveling, ducking, loudness leveling to side-chain
detailed control of gain change (speed, pre-delay, min, max, idle).
flexible pre-delay including zero delay mode.
loudness measurement based on EBUR128 ML
large display to monitor visually impact of plugin parameters.
flexible pre-filter.
side chaining (VST3/RTAS/AAX/AU) or channel 3+4 (VST2).
read/Write automation data.
A/B control.

5.gEQ12V4
gEQ12 is a 12 band stereo/MS graphic equalizer with accurate spectrum analyzer and minimum/linear phase modes.

Today’s mixing and mastering processes require a full featured but easy to use equalizer which can accurately shape the tone of a track , bus or master track.

gEQ12 provides all necessary tools including perceptual loudness matching in one package paired with some unique features.

The GUI of gEQ12 is fully resizeable and scalable.

Features
12 independent filter bands, extended frequency range 10Hz – 48kHz
16 filter types, analogue/digital design, zero delay IIR
filter slope from 6 to 96dB
filter solo
independent stereo/left/right/mid/side processing per band
7 processing modes: zero latency, minimum/linear phase
sample rates min. up to 192 kHz
“real” over sampling, up to 4x
hi-res spectrum analyzer with 32k FFT
pre/post-EQ spectrum, max/average/live spectrum, stereo/left/right/mid/side
post filter predictive spectrum
smoothed, ISO R40 smoothed and raw data spectrum
reference spectrum, delta reference spectrum
Load wave/aiff files as reference
EQ cheat sheets
spectrogram
frequency tracker
extended input/output meter, Peak/RMS/EBU/VU
perceptual loudness matching (powered by AB-LM Lite)
narrow-band sweeping mode
spectrum freeze mode
sample exact A/B toggle for all processing modes
precise parameter input
undo/redo
large and easy to use GUI
free GUI scaling and resizing
interactive EQ display
preset management
full DAW automation support, smooth filter frequency change
64-bit internal processing

6.GSatPlus
Saturation
GSat+ is derived from the saturation module on the TBProAudio CS5501V2 channel strip plug-in, but adds extra features requested and suggested by the Gearspace.com team. It has several different saturation modes, high tweakability, oversampling, old-school VU meters and cross-platform software compatibility – as well as a button to take you directly to Gearspace.com in case you need any tips, tricks or inspiration or want to swap presets with other users.

Features
Saturation with three different “tube-based” character modes
Clip protection
VU meters
Signal monitoring
Over sampling
Plug-in bypass
Preset management
Large and easy to use freely-scalable GUI
64-bit internal processing

7.mvMeter2
Multivariable meter
mvMeter is a multivariable meter including RMS, EBUR128, VU and PPM measurement. mvMeter2 is the successor of mvMeter and adds single/dual meter display, adjustable reference level for all meter modes, adjustable meter delay and preset management.

Features
behavior of classic analog VU and PPM meters
multiple measurement modes: PEAK, RMS, EBU R128, VU and PPM
multi channel metering: stereo, left, right, mid, side
single and dual meter display
adjustable reference level for all meter modes
adjustable meter delay
adjustable meter hold
adjustable OL level
preset management
adjustable pre-gain including gain matching
instance label
large and accurate live meters
64-bit internal processing
5 different meter themes

8.SLM2V2
Wideband limiter
SLM2 is a smart stereo loudness maximizer including “analog spice” and 16x over-sampling. The goal of the design was to maximize the loudness of the audio signal while maintaining transparency.

The SLM2 DSP analyses every sample of the incoming stereo signal within a defined time frame and calculates the optimal volume increase. This guarantees maximum possible transparency.

The overall loudness can be further increased by the following, optional clipper stage. Signal aliasing can be reduced by enabling over-sampling.
SLM2 is used in all limiting situation like track/bus and master limiting.

Features
highly transparent brick-wall-limiter
switchable saturation of input signal
complete control over attack and release
switchable look ahead
2 limiter modes
2 ceiling modes
additional drive and clip stage
stereo link
over-sampling up to 16x
easy to use GUI
preset manager
large and accurate live meters
supports all sample rates

9.ST1V2
Stereo imaging control
ST1 is a simple tool that lets you control the width and the panorama of a stereo audio signal. It includes a mono-to-stereo function, a stereo field enhancer, a rich panning knob that includes spectral panning, and a bass-focus feature. The phase and correlation meter visually monitors the panning position and stereo width.

Features
Phase meter
Correlation meter
Adjustable mono to stereo converter
Variable stereo width
Automatable pan knob with variable pan law and many comfort functions
Spectral panning
Bass focus
Preset management
Many useful presets
Large and easy to use GUI
free GUI scaling
64-bit internal processing

10.AMM2
AMM offers several advanced mix modes and extensive controls for each channel.

AMM can be used to mix quickly a discussion round, a dialog with several speakers or even an audio conference with up to 32 microphones to a stereo signal.
AMM supports multiple channel configurations, from mono up to 32 stereo channels, depending on the DAW capabilities.

11.dEQ6V4
dEQ6 controls the filter gain of each band by a compressor/expander. This reduces or increases the gain in a specific frequency range, depending on the program material. The detector signal for the compressor/expander is processed by a pre filter which is controlled by the centre frequency and Q-factor of the dynamic filter. This ensures that the compressor/expander is only activated by content in the specified frequency range.

12.dpMeter5
dpMeter is a precise digital audio multi channel meter including RMS, EBU R 128, Dialog gated and TruePeak (intersample) measurement. It is the successor of the very popular dpMeter4 and adds GUI resizing and preset management.

13.dpMeterXT3
Multi Channel Meter
dpMeterXT is a loudness metering plugin and application fully compliant with the EBU R128-2014, ITU-R BS.1770-4, ATSC A/85 and several other regionspecific loudness standards.

dpMeterXT supports multiple channel configurations, from 1.0 up to 9.1/10.0, depending on your system configuration or DAW capabilities.

14.DynaRide2
DynaRide is an advanced gain rider plug-in with multiple detector programs, side chain support and switchable pre-delay. On top of it DynaRide can enhance the output signal by limiting the riding process to specific audio signals like speech, vocal and bass.

15.Euphonia3
Euphonia is an optimal audio spectrum balancing plugin which equalizes automatically any audio input based on a few standard average spectrum curves.

Using Euphonia right after the final mix of your music production could eliminate missing low/high end, mixing mistakes, general spectrum balancing problems or even compensate poor listening environment.

16.FinalLoud3
FinalLoud solves quickly a very common challenge in audio engineering: reaching a certain target loudness level for any audio material while maintaining given True Peak ceiling.

Therefore FinalLoud combines a gain control, a high quality True Peak limiter and an accurate loudness meter under an easy-to-use interface. The gain control pushes the loudness level to the target level and in parallel the limiter ensures the True Peak ceiling.

17.Impress3
Impress is a wideband stereo compressor including side-chaining,
various prefilter options and state-of-the-art over sampling.

The goal of the design was to cover a broad range of compression applications in today’s mixing and mastering situations with strong focus on low aliasing. So, Impress could be used to shape subtle a snare sound, glue different tracks on instrument bus and even do complex tasks like deessing or voice over’s.

18.ISOL8
Isol8 is an advanced mix monitoring tool. It divides the frequency range into 5 bands. These 5 bands can be soloed or muted individually. This will help you to concentrate on certain frequency ranges during the mixing and mastering process.

On top you can use ISLO8 as a flexible multi-band splitter with following complex plugin chains. The split signal can finally be mixed together. Isol8 is originally designed to be used on the master track, but it can also be used on individual audio track busses.

Features:
5 adjustable frequency bands
solo/mute function for each band individually
Linkwitz-Riley crossover filter design
24/48dB/Oct filter slope
multiple filter channel modes (Stereo/Left/Right/Mid/Side)
multiple monitor modes (Stereo/Left/Right/Mid/Side)
in-place or centered monitoring
swap left/right channel
adjustable output level
loudness dim function
side-chain monitoring
keyboard control
multi channel split
large and easy to use GUI
free GUI scaling
64-bit internal processing
GPU and non GPU version

19.LAxLimit4
LAxLimit is an advanced look ahead, wideband linked-stereo limiter including ISP (inter sample peak) detection and oversampling.
LAxLimit offers more control over transients and adaptive release curves. It adds now extended metering (RMS, EBU and Dialog gated).
The goal of the design was to cover a broad range of limiting applications in today’s mixing and mastering situations with strong focus on low aliasing.

Design
LAxLimit is specifically designed for mastering, digital editing, multimedia, and any application that requires limiting of the digital signal with top notch quality and lowest aliasing. LAxLimit guarantees ultra fast and overshot free response by using advanced look ahead algorithms.

In order to fulfill today’s TV, broadcast and music production requirements LAxLimit offers True-Peak limiting (ISP) based on ITU-1770 specification and on top of it up to 8x “real” oversampling. Combined with True-Peak limiting reaches even highest production standards.

LAxLimit includes an advanced adaptive release control system which reduces massively artifacts caused by fixed release times. In order to increase limiting sensitivity both stereo channels can be freely linked/unlinked.

LAxLimit is designed to be used as last plugin in the processing chain (brick-wall limiting).

Features:
state of the art low aliasing look ahead limiter design
freely adjustable limiter character
detailed transient control
freely adjustable adaptive release curves
level detection modes: peak and ISP based on ITU BS1770.4
RMS, EBU R128 and Dialog gated loudness measurement
“real” oversampling, up to 8x
stereo link
built-in clipper
drive mode
unity gain and delta monitor modes
preset management
mid/side processing mode
easy to use GUI
free GUI scaling
large and accurate live meters
all sample rates
product needs to be activated with license key
free update for LAxLimit customers
demo version available

20.sTiltV2
sTilt is a linear/natural phase and IIR filter which tilts the audio spectrum around a given center frequency.

Features:
distortion free processing engine
adjustable slope from -6dB/Oct to +6dB/Oct
adjustable center frequency
5 quality modes: low, eco, medium, high, max
3 filter modes: linear phase, natural phase and zero delay
unit auto-gain
switchable clip protection
sample exact A/B comparison
stereo/left/right/mid/side channel selection
flexible channel monitoring
output gain
signal mixing
free GUI scaling
GPU and non GPU version

Note: This plugin adds significant latency to the audio path, which is usually compensated by DAW (PDC).

21.TBPA Move
Rhythmic filter
TBPAMove consists of two independent filters and a high-resolution oscilloscope.
Each parameter of the filter (resonance, cutoff, gain and pan) can be controlled by DAW automation and a modulator. The DAW automation can be customized (delayed, smoothed and mixed with default). The modulation types can be side-chain, LFO, step sequencer or multi-stage envelope generator.

TBPAMove is a DAW plugin for creating rhythmic filter, gain and pan effects.
TBPAMove consists of two independent filters and a high-resolution oscilloscope.
Each parameter of the filter (resonance, cutoff, gain and pan) can be controlled by DAW automation and a modulator.
The DAW automation can be customized (delayed, smoothed and mixed with default setting).
The modulation types can be side-chain, LFO, step sequencer or multi-stage envelope generator.

Features:
2 independent filter with resonance, cut-off, gain pan controls
various filter types (e.g. low pass, high pass, peak and shelf)
serial and parallel processing
8 independent modulators
side-chain modulator
LFO modulator
step sequencer modulator
multi-stage envelope generator
multiple randomization options
AHR envelopedelay of parameter changes
smoothing of parameter changes with low aliasing
VST3 sample accurate automation data processing
large oscilloscope with audio and parameter envelope display
GUI scale
64-bit internal processing

22.TBPA Volume
TBPA Volume allows you to control the volume of the audio signal independently from the volume fader of the mix console. This is useful for adjusting the gain/volume between different plug-ins within a plug-in chain or for volume modulation/automation.

Features：
click-free 64-bit internal processing
master volume fader, left/mid and right/side fader
left/right, mid/side processing
adjustable min/max volume values
treat min value as negative infinity
lock max volume to 0 dB
volume modulation display
smart power button
smooth parameter interpolation
low aliasing
support of sample accurate automation (VST3/CLAP/AU)
preset management
free GUI scaling

VST619音频资源网收集整理！