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Mastering Software & Plug-ins

The<b> Algorithmix Easy reNOVAtor:</b> 
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<b>What is it?</b>
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Denoising techniques work well for stationary or quasi-stationary broadband noise. Short clicks coming from old records or switching noise can be successfully removed and replaced by declicker plug-in. However, neither a denoiser nor a declicker can successfully cope with longer complex audio disturbances, especially if they overlap material we want to recover. This fact has motivated us to develop a special tool, reNOVAtor and later its brother, easyreNOVAtor, especially optimized for time-effective workflow.
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EasyreNOVAtor is intended to rescue high-resolution live or location recordings from extraneous sounds and unwanted audio disturbances. Often, from the performance point of view, brilliant live recordings can not be accepted for CD release, because of disturbing noises like coughs, squeaky chairs, feedback squeals, horn of a passing truck, a bell from the neighboring tower, or ring tones from a cellular phone. Such disturbances are especially critical if they occur in quiet passages when recording classical music. Similarly, location recordings film, broadcast or TV often include annoying incidents which disqualify them to be used as originally intended. In these cases any kind of traditional equalization or editing methods is very time consuming and causes audible discontinuities in desired signal and ambience. Often after hours of hard work even the best post-pro engineer will capitulate.
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<b>How does it work?</b>
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EasyreNOVAtor helps when all other editing tricks and processing methods fail. It allows localization, identification and very precise removal of unwanted audio events mostly without audibly affecting the audio material we want to keep. The removed sound is replaced by a signal re-synthesized from the surrounding material. Unlike time editing methods, easyreNOVAtor does not make deep gaps in the sound track when eradicating a disturbing sound event. Rather, it’s an exactly tailored hole in the spectral representation of the processed signal that can be removed and replaced.
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<b>easyreNOVAtor or reNOVAtor?</b>
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easyreNOVAtor is optimized for very time-effective workflow and therefore equipped with well prepared preset-like controls. It’s ideal for post-pro broadcast, TV and film engineers working under extreme time pressure. When you deal with extremely tenacious disturbances or you just want to broaden your abilities and increase performance in cleaning audio material do upgrade to reNOVAtor that features extended manual parameter setup and some additional advanced functions.

Algorithmix Easy reNOVAtor
A modified version of reNOVAtor equipped with well prepared preset-like controls, especially suitable for very time-effective workflow

<strong>Algorithmix K-Stereo</strong> is a patented psychoacoustical process (US Pat.7076071) that extracts the ambience inherent in ordinary recordings, and is capable of spreading that uncorrelated ambience around the soundstage, and enlarging the size of that soundstage, both deeper and wider.<br>
In addition, K-Stereo enhances the depth and imaging of the instruments and vocals without adding any artificial reverberation. It does not have a sound of its own; it just enhances the existing ambience and early reflections. K-Stereo is also capable of making a natural mono to stereo conversion.
<p>The plug-in is intended to be compatible with the new VST version 3.0 and is the first in the new Algorithmix CHROMIUM SERIES, a set of reference mastering plug-ins.</p>
<p><strong>The World’s first Ambience Recovery Processor</strong><br>
  Welcome to a brand-new category of audio product: the world’s first Ambience Recovery Processor. You are about to experience a sound enhancement plug-in that can add the last polish to an excellent audio recording, or improve a good recording. It’s based on a psychoacoustical process developed and patented by US mastering engineer Bob Katz.</p>
<p><strong>Unobtrusive and Natural Ambience Enhancement</strong><br>
  For the first time, the mastering engineer can easily enhance, expand, and equalize the ambience in a recording. K-Stereo works unobtrusively and naturally. It can help a “small” rock&roll recording become bigger, make a mono recording sound stereophonic. It can improve the definition, depth, width and space of any recording. K-Stereo is primarily a mastering tool, but it is also useful during post-production, and occasionally during mixing. K-Stereo provides control over reverb returns or ambience mikes after the program has been mixed, and it enhances the shape, spread, tonality and depth of that reverb. After mastering, the resulting reverberation envelops the direct sound but the mix becomes clearer and the definition improves due to the psychoacoustic nature of the process.</p>
<p><strong>Merging Different Ambience Types</strong><br>
  K-Stereo helps to glue a recording together without need for dynamic compression. Recordings that were made with multiple stereo microphone techniques sound more realistic and cohesive, as K-Stereo removes or reduces the artificial “edges” between the different spaces that were mixed together.</p>
<p><strong>Ambience Enhancement Without Adding Any Artificial Reverb</strong><br>
  K-Stereo can help give a demo the polish of a better mix, or reduce the weaknesses of a poor mix. There’s no substitute for a great mix; no substitute for use of stereo and surround microphone techniques in a natural acoustic space. However, an experienced mastering engineer will discover that K-Stereo can put the final polish on nearly any mix.</p>
<p>This is accomplished without adding artificial reverb, without the muddy effects and artifacts that are associated with adding reverberation to an existing mix. Plus you won’t have to spend an hour playing with reverb settings to avoid the “room within a room” effect, or attempting to match the color of the original recorded reverb. The color and feel of the original reverberation is preserved. K-Stereo improves the spread and diffusion of the inherent reverb and reveals the critical early reflections that define the position and location of the instruments. It can’t turn a $600 reverb into a $6000 model, but it can give the $600 reverb some of the qualities of the more expensive model.</p>
<p><strong>Adding and Modifying Real Space to Vocals and Instruments</strong><br>
  You can instantly create a deeper, wider soundstage, move instruments and vocalists that are too far up front away from the listener. Instruments and vocalists take on a spatial, kinetic, quality, improving the illusion of being in a real space, without coloration, pitch modulation, “phasiness” or artifacts often noted with other processes. You can restore or enhance the front-to-back depth in stereo recordings that otherwise sound flat and one-dimensional. Recordings become integrated and organic; “in your face” recordings can be made to sound just right. The effect can range from subtle and intimate to extremely spacious with the touch of a knob, without losing center-panned information (as would occur with competing techniques that reduce the M to S ratio).</p>
<p><strong>Mono-to-Stereo Conversion</strong><br>
  K-Stereo also performs natural mono-to-stereo conversion, by bringing out the ambience in the original mono source and spreading it to the sides in a stereophonic fashion, and with an extremely mono-compatible result.</p>
<p><strong>Ambience Enhancement Keeping the Character of a Mix Unchanged</strong><br>
  K-Stereo does not have a sound of its own; it just enhances the existing ambience and early reflections. Initially, the effect appears subtle. Even when you turn the effect up too loud, it doesn’t sound bad. You’ll hear a natural, diffuse ambient field which is extracted (derived) from the ambience in the source recording. The recording’s clarity, spatiality, depth and soundstage is enhanced without creating phasing or comb-filtering effects, without matrixing or altering mid/side ratio (unless you choose to do so), without changing the “mix”, without fancy steering, and with no effect on the tonal balance of the direct sound. The tonal balance of the ambience can be altered so as to produce a more pleasant sonic effect, which is a different and more subtle form of equalization.</p>
<p><strong>Preserving Ambience in Coded Recordings</strong><br>
Normally, conversion from long wordlength to 16-bit CD format as well as coding (data compression) to mp3 or other coded formats for the Internet tends to narrow the space and the depth. But through the use of the K-Stereo process and a good dithering algorithm (today included in every serious DAW), amazingly, a 16-bit master can have much of the depth of the 24-bit original, and mp3s as well! A 24-bit and/or high sample rate master destined for a high resolution medium will also improve if the original mix was small or “depth-deprived”.</p>
<p>And finally a secret: K-Stereo makes the sound louder, but rarely causes clipping. (shhhh).</p>

Algorithmix K-Stereo Ambience Processor (VST 3.0)
High-Resolution VST processor for natural and unobtrusive enhancement, expansion, and equalization of the ambience, space, imaging, depth and width in a recording without adding artificial reverberation or causing unwanted coloration or artifacts

<b>Algorithmix LP Split Comp</b> is a creative dynamic tool with unparalleled transparency and sound purity combining the best of both the analog and digital worlds.
<p>The split function allows dynamic processing only in the band of interest. Unlike many other split or multiband compressors the frequency crossover is made with low-noise linear-phase filters. They allow perfect band recombination after processing. The character of the PlugIn can be continuously adjusted between being more compressor or rather limiter. It provides a lot of psychoacoustically optimized features like automatic dynamic characteristics, soft knee, and parameterized soft clipper. To reduce the problem of alias frequencies being often a hidden problem in digital non-linear processors, a reference-quality oversampling has been implemented.</p>
<p>LP SplitComp is the second one in the new Algorithmix CHROMIUM SERIES, a set of reference mastering PlugIns. The current release is implemented as a stereo version, however it is intended to make it VST 3.0 compatible as soon as the major audio editing systems have switched to the new VST version. The compatibility to VST 3.0 will induce a multichannel version of the LP SplitComp.</p>
<p><b>Most fundamental but least understood</b><br>
  Dynamic processors, particularly compressors belong to the most fundamental tools in every recording, mixing or mastering studio. Despite the popularity and broadening of compressors they are probably the least understood and most misused processors in daily routine of the studio engineer. One reason might be that sonic changes of a good compressor are very subtle and hardly perceptible for an unexperienced listener. Another reason is the variety of parameters which differ from one unit to the other and even for the same parameter name exhibit a deviant behavior. The parameters are mostly interacting in some way and need a lot of experiments and understanding to learn how to use them in a purposeful manner.</p>
<p>A digitally implemented, well-designed mastering compressor has not necessarily less parameters, but they are precisely defined and de-coupled. In addition, due to novel, extraordinary possibilities of the digital signal processing, supported by psychoacoustic findings, digital compressors open a new quality window for mastering engineers.</p>
<p><b>Digital architecture with analog sound</b><br>
  The LP SplitComp arose from many years of experience of the Algorithmix audio algorithm architects, countless experiments, and critical sonic analysis of analog compressor legends. A serious, often unrecognized problem of nonlinear digital processors is a quite high amount of alias frequencies. They can ruin the best intentions spent for a good functionality. Therefore the audio path in our compressor has been equipped with a reference-quality oversampling to reduce unwanted alias frequencies.</p>
<p><b>Parallel Compressing</b><br>
  Parallel compressing is known already for many years. It is accomplished by mixing the compressed signal with the direct signal in a mixing console. In digital technology it is extremely important to provide an exact delay alignment in the direct channel to avoid any unwanted signal cancellation because of phase differences. Therefore we implemented a properly time-aligned direct signal path. The Selective mode the LP SplitComp provides a new unique derivative of parallel compression.</p>
<p><b>Split it!</b><br>
  Of course, the LP SplitComp can cut a good figure as a Full-Band compressor, but its true distinctiveness arises in the Split-Band mode. The signal is divided in two complementary parts: the first one is compressed and the second one is just passed through without dynamic changes. Due to the high-resolution linear-phase technology used for the implementation of the split filter, the reconstruction of the compressed and complementary band is performed completely without phase distortion contrary to some other split or multiband compressors. The concept of the LP SplitComp allow for very precise frequency-conscious dynamic processing. Especially if you need to “repair” specific spectral parts in dense mixes you will appreciate the surgical capabilities of this tool. Because of a noise-free design you can use this compressor repeatedly on the same material, each time concentrating on a particular dynamic task or problem.</p>
<p><b>Looking into the future</b><br>
  Because of using look-ahead delay in the signal path, the overs can be caught very precisely. Together with a respectively chosen attack time, numerous methods of transient treatment are possible. Using the Com/Lim feature you can make the processor more acting as a limiter or compressor. You can even use two instances of the LP SplitComp, the first one as a compressor and the second one as a limiter.</p>
<p><b>Side Chain</b><br>
  The side chain of the compressor includes a range of filters to make signal detection frequency dependent. The filters can be adjusted identical to the split filter bank to perfectly master frequency-selective tasks like de-essing or proximity effect removal. The static characteristic allows adjusting different grades of knee rounding (soft-knee) in order to make compression smoother, less obtrusive. A sophisticated release computer provides program-dependent reaction to any complex audio material.</p>
<p><b>A lot of helpful features</b><br>
At the end of the signal path a soft clipper takes care that the output signal does not exceed the level adjusted in the Ceiling field. In addition it allows introducing some controlled distortions. To make the understanding and operation of the LP SplitComp as easy as possible we have implemented some additional features. The signal Monitoring is very comprehensive but yet intuitive. You can click on different nodes in the block diagram and thus listen to the relevant audio signals within the compressor architecture.</p>
<p>An additional feature, offered in a compressor only by Algorithmix, is the Differ function. It allows listening to the difference between input and output signal. So you can listen to the changes you’ve done to the signal in the PlugIn. This difference is possible solely because the whole path in the LP SplitComp is linear-phase.</p>
<p>You can also open a compact auxiliary Metering panel showing all level meters and indicators simultaneously and place it to any position of the screen.</p>
<p><b>You will love it</b><br>
  We believe, we have created a reference-quality compressor for mastering engineers that opens new ways in efficient, transparent and surgical dynamic processing. This compressor, however, is also intended as a creative tool for any other kind of computer audio processing in recording and mixing as well as in post-production, broadcast and TV studios.</p>

Algorithmix LP SplitComp
High-resolution linear-phase split compressor for VST-compatible audio workstations

Precise emulation of legendary analog equalizers, extended by progressive, new models: standard / vintage / modern / experimental. A sonic paradise for every recording and mixing engineer.<BR> <BR>A few years ago we did not even think about spending our time to develop perhaps the 157th parametric equalizer on the market. Why one more? Is it possible to be noticed among all the other already established products? 
<P>It was you, our customers using our <I><B>LinearPhase PEQ Red</B></I> and<I> <B>Orange</B></I> who asked Algorithmix to develop a classical recording/mixing equalizer but with its unparalleled purity and transparency. The biggest problem we had in the beginning was the equalizer style, so we closely analyzed many of them. To satisfy our customers as much as possible, we decided to implement not only one, but an entire collection of traditional (minimal-phase) equalizers. Most of them emulate analog archetypes; some are based on new ideas. 
<P>We hope you enjoy the dozen shades of Blue and find that favorite sound you previously could get only from very expensive outboard gear. 
<P><B>Shelf and Cut Filter Library</B> 
<P>In <I>Classic PEQ Blue</I>, not only are various bell filters used but also different kinds of shelving filters. Nine of the serial equalizers use a new shelving filter generation characterized by cut-off frequency defined in the middle of the transition part. We found these filter definitions more intuitive than the classical ‘– 3 dB below maximum’. The old definition is only used in the Classic Asymmetrical type to conform to its original predecessor. All 2nd order shelving filters have a Q adjustment to emulate vintage characteristics with their specific bumps at higher slopes. Also, the parallel equalizers are equipped with respective shelving filters shapes being typical for the old parallel PEQs and interacting with other bands as did their analog predecessors. 
<P>Every complex PEQ is equipped with cut filters. In <I>Classic PEQ Blue</I>, each of the 12 equalizers can assign 1st and 2nd order cut filters. The 2nd order filters have a Q adjustment to create so-called resonant filters characteristics and very steep brickwall filters after cascading more of them (Butterworth, Bessel, Elliptic). 
<P><B>The best of the analog and digital world</B><BR><I>Classic PEQ Blue</I> is a creative equalizing tool combining the best of both the analog and digital worlds. We modeled the most legendary analog equalizers including two parallel ones and added a few experimental characteristics only possible in digital domain. To avoid bell filter asymmetry at high frequencies, typical for many digital equalizers, we have applied reference-quality upsampling techniques, automatically switchable if the sampling frequency of the input signal is 44.1 or 48 kHz. By using proprietary filter algorithms, we have achieved a huge dynamic range, as well as extremely low noise and distortion level and thus unparalleled sound purity--impossible with any analog circuitry. The whole equalizer collection works with sample rates up to 384 kHz and therefore is perfectly suitable for DSD post-processing. Several instances can be opened simultaneously. Complete setups can be easily exchanged between them. The true frequency response display is zoomable and in the DirectX version the whole PlugIn can be enlarged to the full screen.</P>

Algorithmix LPEQ Classic Blue
12-in-1 - The most complete collection of classic equalizers ever in one DirectX / VST plug-in: ten serial plus two parallel

Every mastering engineer knows that equalization of final complex mixes or orchestral recordings with stereo microphones is very critical and limited. 
<P>Any correction of a single instrument or vocal part influences other instruments and can negatively change their sound characteristics. This is because all classic equalizers change the phase of the fundamentals and their harmonics and the phase shift is frequency dependent. 
<P>The LinearPhase PEQ Orange does not shift the phase, it only boosts or cuts the amplitude in a given frequency range. Consequently you can apply much more boost without changing the sound character of your recording. Due to our proprietary low-noise filter design in the time domain, the distortions (THD+N) are extremely low ensuring crystal clear, neutral, and uncolored sound. 
<P>The <B>LinearPhase PEQ Orange</B> DirectX/VST PlugIn is unique in the world of audio components. Almost all parametric equalizers being used are implemented with filters that accomplish phase shift, e.g. the original signal is remixed with its phase-shifted version. Because the amount of the phase shift is frequency dependent some cancellation or amplification at certain desired frequencies takes place. It works, but with a major disadvantage - the different signal components are spread all over time, so that the time relationship between harmonics in the processed signal is heavily affected. The result is that a nice sharp bass drum becomes slurred and muddy and vocal tracks become strident or brittle. Algorithmix is proud to offer you a true remedy against difficult equalizing tasks, the <B>LinearPhase PEQ Orange</B> PlugIn, a linear-phase equalizer which can be handled exactly like its classic predecessors, but cannot be "heard".<BR><BR>The <B>LinearPhase PEQ Orange</B> successfully copes with sonic problems that cannot be solved with any classical equalizer without side effects. You can boost lower frequency regions by even 10dB without mud or slush and with no loss of transients at all. You can remove the sibilance from a vocal with a sharp notch without affecting the whole mix like occurs when using analog or IIR-based digital EQ. 
<P><B>LinearPhase PEQ Orange</B> better preserve the time relationships of harmonics in the original waveform. This translates to smoother top end, sweeter and bigger midrange, and a clearer more distinct bottom end as compared with phase-shifted EQ. The resultant sound seems less processed, more like it occurred naturally in the air in front of the microphone. The mixes become clear and transparent, the instruments more defined and realistic.<BR><BR>The <B>LinearPhase PEQ Orange</B> works in the time domain. It sounds more analytical than its brother, the <B>LinearPhase PEQ Red,</B> and therefore is specially recommended for difficult mastering and re-mastering tasks on dense mixes. It performs flawlessly with up to 384 kHz sampling frequency and allows extended frequency setup up to 80 kHz, thus being ideally applicable for high-resolution DSD post-processing, including its ultrasonic region. </P>

Algorithmix LPEQ Orange Phase Linear EQ
10-band precise linear-phase parametric equalizer

Every mastering engineer knows that equalization of final complex mixes or orchestral recordings with stereo microphones is very critical and limited. 
<p>Any correction of a single instrument or vocal part influences other instruments and can negatively change their sound characteristics. This is because all classic equalizers change the phase of the fundamentals and harmonics and the phase shift is frequency dependent.
</p><p>The <span style="font-weight: bold;">Algorithmix LinearPhase PEQ Red</span> does not shift the phase. It only boosts or cuts the amplitude in a given frequency range. Consequently, you can apply much more boost without changing the sound character of your recording. Due our proprietary low-noise filter design in the frequency domain, the distortions (THD+N) are extremely low, ensuring crystal clear sound quality.
</p><p>The <b>LinearPhase PEQ Red</b> DirectX/VST PlugIn is unique in the world of audio components. Almost all parametric equalizers being used are implemented with filters that accomplish phase shift, e.g.: the original signal is remixed with its phase-shifted version. Because the amount of the phase shift is frequency dependent some cancellation or amplification at certain desired frequencies takes place. It works, but with a major disadvantage--the different signal components are spread all over time, so that the time relationship between harmonics in the processed signal is heavily affected. The result is that a nice sharp bass drum becomes slurred and muddy, and vocal tracks become strident or brittle. Algorithmix is proud to offer you a true remedy against difficult equalizing tasks--the <b>LinearPhase PEQ Red</b> PlugIn, a linear-phase equalizer which can be handled exactly like its classic predecessors, but cannot be "heard".
</p><p>Experts say that most of digital equalizers do not sound like their analog predecessors, especially when working with low sampling frequencies: 44.1 or 48kHz. Spectrum modifications in the high-frequency region sound as improperly balanced. The reason is so called frequency warping which makes the bells asymmetric and high shelving and high-cut filters much steeper as theoretically set up. To make the <b>LinearPhase PEQ Red</b> analog sounding, special corrections have been applied to the original digital filters making them look exactly like their analog references. 
</p><p>The <b>LinearPhase PEQ Red</b> features the <i>Continuous Slope Filter™</i> technology, proprietary to Algorithmix® and the only one of its kind, worldwide. Unusual flexible shelving and cut filters allow adjustment of any slope in the range from 0 to 24dB/octave. 
</p><p>The <b>LinearPhase PEQ Red</b> better preserves the time relationships of harmonics in the original waveform. This translates to smoother top end, sweeter and bigger midrange, and a clearer more distinct bottom end as compared with phase-shift EQ. The resultant sound seems less processed, more like it occurred naturally in the air in front of the microphone. The mixes become clear and transparent, the instruments more defined and "realistic". You can boost lower frequency regions by even 10dB without mud or slush and with no loss of transients at all. You can remove the sibilance from a vocal with a sharp notch without affecting the whole mix like occurs when using analog or IIR-based digital EQ. 
</p><p>The <b>LinearPhase PEQ Red</b> works in the frequency domain. It sounds a bit softer than his brother, <b>LinearPhase PEQ Orange</b>. It perfectly performs with up to 384kHz and allows extended frequency setup up to 80kHz. These features make it perfectly applicable for high-resolution DSD post-processing, inclusive the ultrasonic region.</p>

Algorithmix LPEQ Red Phase Linear EQ
10-band precise linear-phase parametric equalizer

<B>Algorithmix NoiseFree™ Pro</B> automatically removes or reduces any kind of constant background noise like hiss, surface noise, hum, or camera sound.
<P>Due to the Algorithmix® proprietary, psycho-acoustically optimized noise reduction algorithm, <B>NoiseFree™ Pro</B> efficaciously preserves timbre, ambience, and low-level details of the original signal, as well as highly reduces the appearance of artifacts.<BR>On one hand, it can be used for high-resolution re-mastering tasks. On the other hand, it is a real weapon in cleaning up critical speech recordings for forensic purposes.</P>
<P>One quite common problem in the daily business of an audio engineer is a noisy signal. The standard methods of noise extraction by means of filters do not work if spectral components of noise overlap the desired signal. In such situations the only solution is to use sophisticated signal processing algorithms to suppress unwanted noise in a much more intelligent way. </P>
<P>The <B>NoiseFree™ Pro</B> DirectX PlugIn effectively removes broadband noise from pre-recorded audio material. It has been designed to perfectly cover a wide range of applications: from the high-quality de-noising of valuable music treasures, to cleaning location recordings for film from environmental noise, even the treatment of critical forensic material recorded at a very poor signal-to-noise ratio and/or in reverberate environment. Typical tasks for the <B>NoiseFree™ Pro</B> include removal of tape-hiss, surface noise of old records and wax cylinders, broadcast noise, microphone and preamp noise, as well as enhancement of conversations and interviews that lack intelligibility.</P>
<P>Unlike other systems, the <B>NoiseFree™ Pro</B> DirectX PlugIn works virtually without artifacts when using the correct settings for all parameters.</P>
<P>The de-noising process can work based on either the built-in white noise profile or a recorded noise profile. The unique Learn algorithm allows recording a noise sample from virtually any part of audio material, even if there is no noise-only part available. A sophisticated 5-band Noise Profile EQ is provided for modifying the shape of noise profiles and smoothing them for optimal performance. The parameters Ambience recovery and Decorrelation allow fine tuning the <B>NoiseFree™ Pro</B> to differentiate between noise and sharp signal transients. The Chase function--mainly designed for speech de-noising--automatically detects any fluctuations in the background noise and adjusts the Noise Profile according to these changes.</P>
<P><B>NoiseFree™ Pro</B> works impeccably with sampling rates up to 192 kHz. Thus it is perfectly suitable for hi-res DSD post-production. Since the CPU requirement for the <B>NoiseFree™ Pro</B> DirectX PlugIn is quite low, you can change and optimize all parameters while listening to the audio material in real-time.</P>

Algorithmix NoiseFree Pro
DirectX plug-in that effectively removes broadband noise while still excellently preserving the audio quality of the original signal

Clicks and crackles are inseparable companions of old vinyl and 78 rpm records. Before re-mastering to CD they have to be removed and the remaining gaps have to be reconstructed properly.
<P>The <B>Algorithmix ScratchFree™ Pro</B> DirectX Plug-In copes with any kind of transient noises, efficaciously preserving the audio quality of the original signal.</P>
<P>The <B>ScratchFree™ Pro</B> DirectX Plug-In effectively removes clicks and crackles from old vinyl and shellac records and cleans up audio recordings tainted by switching noise, static discharge, digital cross-talk, or thyristor buzz.</P>
<P>Unlike other systems, the <B>ScratchFree™ Pro</B> DirectX Plug-In works virtually without artifacts when using correct settings for all parameters. It includes an excellent de-crackling algorithm preserving timbre, ambience, and low-level details in the original signal.</P>
<P>Since the CPU requirement for the <B>ScratchFree™ Pro</B> DirectX Plug-In is very low, you can change and optimize all parameters while listening to the audio in real-time.</P>
<P>The de-scratching algorithm consists of two main parts: the de-clicking module and the de-crackling module. While the de-clicking module is used to remove severe clicks from old shellac and vinyl records or switching noise originating from improper setups of digital audio equipment, the de-crackling module removes any remaining small clicks and crackles.</P>
<P>The signal scope helps you to find ideal settings for the DeCrackler. The advanced parameters Width, Smooth, and DePlop are useful to minimize the appearance of artifacts. The unique Differ feature allows intuitive parameter settings. You can switch between the output signal and the input/output difference, i.e., the part of input signal taken out by the de-scratching algorithm. Normally, this differential signal should not contain any audible parts of the original audio material that you want to preserve.</P>
<P><B>ScratchFree™ Pro</B> provides pre-defined application profiles (Type). They preset the internal parameters and external advanced parameters to help you in typical restoration situations: digital spikes, shellac and vinyl. In addition to its main application, the removal of clicks and crackles, the <B>ScratchFree™ Pro</B> DirectX Plug-In effectively diminishes any kind of distortion caused by signal overload (Clipping).</P>

Algorithmix ScratchFree Pro
DirectX plug-in that effectively removes clicks, crackles and any other transient noises with virtually no artifacts and no degradation of the original signal

<B>Absolutely unbelievable!</B>
<P><B>Algorithmix reNOVAtor™</B> helps when all other audio tools and editing tricks fail.<BR><BR>It can eliminate disturbing audio events while perfectly preserving the sound quality of your original material, because every spectral modification is executed in linear-phase domain.</P>
<P>This is a extremely time-saving tool for mastering or post-production engineers struggling with disturbing noises in live recorded music, interviews or film sound.<BR>If squeaky chairs, coughs, performance noises, falling keys or ringing call phones make your recording unacceptable, take <B>reNOVAtor™</B>!</P>
<P>Imagine that during a music festival you recorded a unique live performance in a concert hall or church. After checking the recorded material in your studio before production and mastering you decide the recording was successful in general, except for a few annoying disturbances during some quiet passages: somebody's cough, a squeaky chair, the horn of a passing truck, a bell from the neighboring clock tower. In addition, despite the exceptional artistic interpretation there were a few significant errors: a loud scratch in the part of violin soloist and one too early tone in the brass section.</P>
<P>All this makes your recording unacceptable and, of course, the concert cannot be repeated. As an experienced tonemeister, you know very well that all traditional techniques and tricks fail when you try to remove the disturbances mentioned above. In such a situation any kind of traditional equalization or sophisticated editing methods is usually time consuming and causes discontinuities or at least audible changes in level and timbre of the desired signal and ambience. You will say “no way” and close the project.</P>
<P>That was yesterday. Today, we at Algorithmix® are proud to unveil the <B>reNOVAtor™</B>, our unique solution. In such hopeless situations, <B>reNOVAtor™</B> successfully rescues problematic live recordings from audio disturbances and unwanted noises. You will be amazed by the high sound quality of the repaired section and how quickly it works. </P>
<P>The <B>reNOVAtor™</B> PlugIn allows localization, identification and very precise removal of unwanted audio events without affecting the audio material you want to keep. The removed sound is replaced by a signal re-synthesized from the surrounding material. The <B>reNOVAtor™</B> does not make deep gaps in your sound track when eradicating a disturbing sound event. Rather, it’s an exactly tailored hole in the spectral representation of the processed signal that can be removed and replaced. The interpolation may even be restricted to certain gain ranges within the selected area, which is very useful if only a certain part of the signal needs to be treated (e.g. one specific harmonic). The <B>reNOVAtor™</B> window is fully resizable for increased accuracy and optimal compatibility with all screen resolutions.</P>
<P>Working with the <B>reNOVAtor™</B> is easy and intuitive. The <B>reNOVAtor™</B> loads the requested part of audio material you've chosen and analyzes it. The result is displayed as a 3D spectrogram with time on the horizontal axis, frequency on the vertical axis and amplitude of the spectral components color-coded. The color assignment follows the order of the rainbow: red and yellow for low energy; green and blue for middle energy; and finally purple and white for high energy. After getting some experience, this 3D spectrogram representation allows a good feeling for localization and identification of sudden unwanted acoustical events. The spectral area of interest can be precisely marked with a resizable rectangular window. A Play button allows you to hear different parts of the processed signal.</P>

Algorithmix reNOVAtor
Time-saving, high-resolution audio repair processor freeing problematic live recordings from unwanted tenacious audio disturbances

<B>Absolutely unbelievable!</B>
<P><B>Algorithmix reNOVAtor™</B> helps when all other audio tools and editing tricks fail.<BR><BR>It can eliminate disturbing audio events while perfectly preserving the sound quality of your original material, because every spectral modification is executed in linear-phase domain.</P>
<P>This is a extremely time-saving tool for mastering or post-production engineers struggling with disturbing noises in live recorded music, interviews or film sound.<BR>If squeaky chairs, coughs, performance noises, falling keys or ringing call phones make your recording unacceptable, take <B>reNOVAtor™</B>!</P>
<P>Imagine that during a music festival you recorded a unique live performance in a concert hall or church. After checking the recorded material in your studio before production and mastering you decide the recording was successful in general, except for a few annoying disturbances during some quiet passages: somebody's cough, a squeaky chair, the horn of a passing truck, a bell from the neighboring clock tower. In addition, despite the exceptional artistic interpretation there were a few significant errors: a loud scratch in the part of violin soloist and one too early tone in the brass section.</P>
<P>All this makes your recording unacceptable and, of course, the concert cannot be repeated. As an experienced tonemeister, you know very well that all traditional techniques and tricks fail when you try to remove the disturbances mentioned above. In such a situation any kind of traditional equalization or sophisticated editing methods is usually time consuming and causes discontinuities or at least audible changes in level and timbre of the desired signal and ambience. You will say “no way” and close the project.</P>
<P>That was yesterday. Today, we at Algorithmix® are proud to unveil the <B>reNOVAtor™</B>, our unique solution. In such hopeless situations, <B>reNOVAtor™</B> successfully rescues problematic live recordings from audio disturbances and unwanted noises. You will be amazed by the high sound quality of the repaired section and how quickly it works. </P>
<P>The <B>reNOVAtor™</B> PlugIn allows localization, identification and very precise removal of unwanted audio events without affecting the audio material you want to keep. The removed sound is replaced by a signal re-synthesized from the surrounding material. The <B>reNOVAtor™</B> does not make deep gaps in your sound track when eradicating a disturbing sound event. Rather, it’s an exactly tailored hole in the spectral representation of the processed signal that can be removed and replaced. The interpolation may even be restricted to certain gain ranges within the selected area, which is very useful if only a certain part of the signal needs to be treated (e.g. one specific harmonic). The <B>reNOVAtor™</B> window is fully resizable for increased accuracy and optimal compatibility with all screen resolutions.</P>
<P>Working with the <B>reNOVAtor™</B> is easy and intuitive. The <B>reNOVAtor™</B> loads the requested part of audio material you've chosen and analyzes it. The result is displayed as a 3D spectrogram with time on the horizontal axis, frequency on the vertical axis and amplitude of the spectral components color-coded. The color assignment follows the order of the rainbow: red and yellow for low energy; green and blue for middle energy; and finally purple and white for high energy. After getting some experience, this 3D spectrogram representation allows a good feeling for localization and identification of sudden unwanted acoustical events. The spectral area of interest can be precisely marked with a resizable rectangular window. A Play button allows you to hear different parts of the processed signal.</P>

Algorithmix reNOVAtor Pro Tools on PPC MAC (G4,G5), Windows
Time-saving, high-resolution audio repair processor freeing problematic live recordings from unwanted tenacious audio disturbances

<span style="font-weight: bold;">Audio Ease BarbaBatch V4</span> is a batch sound file conversion program for Mac OSX. It includes version 3.9 for Mac OS9. All information on this page refers to the OSX version. Please click here for info on the OS9 version.<br><br>For over a decade BarbaBatch has been winning awards and acclaim for its top quality conversions.<br><br>Its unrivaled samplerate conversion algorithm, its support for high end files like BWF, Sonic Solutions and 32 bit float files, its 192kHz sampling rate support and Redbook CD image extraction has made it popular among mastering engineers.<br><br>A Wide range of telephony formats together with dynamic compression and samplerate conversion that make low quality audio sound not as low as you'd expect, has made it exell in telephony audio.<br><br>Smooth and fast batch capabilities and extensive logging, allowing tens of thousands of files to be converted to multiple output formats in one run make it the production choice for 6 out of 10 top selling computer games.<br><br>Automated CD ripping of faded snippets to many streaming sound formats make it ideal for delivery of web based audio.<br><br>"Best-Sounding, easiest to use batch audio file format conversion package"<br>Interactivity Magazine<br><br>"The Best Conversion Tool Money Can Buy" MacWorld - Nominated for MacWorld Eddy<br><br>"Excellent sound quality, fast processing,extensive file format support." - Key Buy - Keyboard Magazine<br><br>"Best-Sounding, easiest to use batch audio file format conversion package" Interactivity Magazine<br><br>"Quick and Easy to use, BarbaBatch produces great sounding output, and it's the most efficient batch-processor"<br>Editors Choice award - Electronic Musician Magazine<br><br>"Superior batch processing" - Top Tool award - Interactivity Magazine<br><br>"Screaming-fast conversions with awesome-sounding results" - MacAddict Review<br><br>Hyper award - New Media Magazine<br><br><span style="font-weight: bold;">Supported File Types</span><br><br>AIFF<br>Dawn AIFF<br>Sonic Solutions AIFF<br>AIFC (uncompressed)<br>AIFC IMA 4:1<br><br>Audio CD Tracks<br><br>IInteger/Float - Quicktime Movie<br>IMA 4:1 - Quicktime Movie<br>QDesign Music Basic (v1 v2) - Quicktime Movie<br>Qualcomm Purevoice - Quicktime Movie<br>u-law - Quicktime Movie<br>a-law - Quicktime Movie<br>MACE 3:1- Quicktime Movie<br>MACE 6:1- Quicktime Movie<br>-<br>MPEG 1 Layer I<br>MPEG 1 Layer II<br>MP3 (LAME codec)<br>MP4 AAC<br>-<br>Sound Designer I<br>Sound Designer II<br>-<br>Wave<br>Wave 32 bit Float (Samplitude)<br>Wave u-law<br>Wave a-law<br>Microsoft ADPCM<br>-<br>Broadcast Wave - MPEG II (compressed bwf)<br>Broadcast Wave - PCM (uncompressed bwf)<br>-<br>VOC<br>VOC 16 bit<br>-<br>NeXT/Sun linear (.snd)<br>NeXT/Sun u-law (.au)<br>NeXT/Sun a-law (.au)<br>-<br>System 7 Sound<br><br>VOX-<br>Dialogic Vox ADPCM (no header)<br>Dialogic Vox PCM (no header)<br>Dialogic Vox a-law (no header)<br>Dialogic Vox u-law (no header)<br>-<br>Pika (adpcm)<br>.vap (single segment Dialogic ADPCM Annotated Voice)<br>-<br>CCITT G.711 (8 bits a-law, 8 kHz)<br>CCITT G.711 (8 bits u-law, 8 kHz)<br>CCITT G.721 40 Kbps (4 bits ADPCM, 8 kHz)<br>CCITT G.723 16 Kbps (2 bits ADPCM, 8 kHz)<br>CCITT G.723 24 Kbps (3 bits ADPCM, 8 kHz)<br>CCITT G.723 40 Kbps (5 bits ADPCM, 8 kHz)<br>CCITT G.726 24 Kbps (3 bits ADPCM, 8 kHz)<br>CCITT G.726 32 Kbps (4 bits ADPCM, 8 kHz)<br>CCITT G.726 40 Kbps (5 bits ADPCM, 8 kHz)<br>CCITT G.729 8 Kbps (CS-ACELP, 8 kHz)<br>-<br>Amiga IFF/8SVX<br>AVR<br>Paris linear (Ensoniq)<br>Dyaxis MacMix (Studer)<br><br>Redbook Audio CD Images (IMAGE.DAT or DDP)

Audio Ease BarbaBatch V4
Award-winning sound file conversion program for Mac

Avid Aphex Big Bottom Pro/Aural Exciter Bundle
TDM Plug-In

DINR, Digidesign's award-winning Intelligent Noise Reduction plug-in, effectively reduces unwanted noise — including tape hiss, guitar-amp buzz, and air conditioner rumble — for cleaner, more professional sounding audio.
<p>
Since DINR analyzes and subtracts noise entirely within the digital realm, the results can be virtually free of side effects — such as distortion, dynamic modulation (pumping and breathing), and the undesired fluctuations in frequency response associated with conventional noise reduction systems.
<p>

<b>FEATURES </b><p>

Professional-quality broadband noise reduction <br>
Broadband Noise Reduction compatible with Pro Tools systems running on Mac OS or Windows <br>
High fidelity, real-time processing with TDM and file-based processing with AudioSuite <br>
Unparalleled cost-effectiveness <br>
Automation features including dynamic TDM plug-in automation <br><br>

<b>SYSTEM REQUIREMENTS </b><br><br>

Digidesign-approved Pro Tools system running 4.1.1 software (or higher)

<p>
Maximum Supported Sample Rate:<br> 96 kHz<p>

Avid DINR (TDM + AS)
Avid/Digidesign Intelligent Noise Reduction Plug-In (TDM + AS)

<FONT size=2>DINR, Digidesign's award-winning Intelligent Noise Reduction plug-in, effectively reduces unwanted noise - including tape hiss, guitar-amp buzz, and air conditioner rumble - for cleaner, more professional sounding audio. Now, it's available as an Audio Suite plug-in for LE systems!<BR><BR>Since DINR analyzes and subtracts noise entirely within the digital realm, the results can be virtually free of side effects - such as distortion, dynamic modulation (pumping and breathing), and the undesired fluctuations in frequency response associated with conventional noise reduction systems.<BR><BR><BR><B>Features:</B><BR></FONT>
<UL>
<LI><FONT size=2>Professional-quality broadband noise reduction<BR></FONT></LI>
<LI><FONT size=2>High-fidelity, file-based processing with AudioSuite<BR></FONT></LI>
<LI><FONT size=2>Listen to the noise being removed with DINR AudioSuite "Audition Noise" function<BR></FONT></LI>
<LI><FONT size=2>Post-processing removes undesirable artifacts with DINR AudioSuite "Artifact Removal" function<BR></FONT></LI>
<LI><FONT size=2>Unparalleled cost-effectiveness</FONT></LI></UL><BR>

Avid DINR LE (AS)
DINR Plug-In For Pro Tools LE!

Developed by the creator of the parametric EQ, George Massenburg, Massenburg DesignWorks® Hi-Res Parametric EQ 3.0 emulates the filter curves of Massenburg’s legendary GML 8200 reference-standard equalizer. Designed for Pro Tools|HD® systems, version 3.0 offers new RTAS® support, giving you the flexibility to use the EQ with Pro Tools LE® and Pro Tools M-Powered™ systems too. With its unprecedented clarity, unmatched smoothness, and excellent high-frequency response, MDW® Hi-Res Parametric EQ will help you achieve optimal definition in your mixes. <BR><BR><B>Features</B> 
<UL>
<LI>Wide frequency selection, from 10 Hz to 41 kHz, with variable Q and choice of three- or five-band EQ 
<LI>Double-precision 48-bit processing for unprecedented clarity 
<LI>Emulates GML 8200 constant shape reciprocal filter curves — the industry-standard reference 
<LI>IsoPeak® functionality lets you quickly solo a target frequency band 
<LI>A/B snapshots allow settings comparisons 
<LI>Supports both TDM and RTAS plug-in formats </LI></UL><B>Digidesign MDW® Hi-Res Parametric EQ</B> emulates the constant shape reciprocal filter curves of Massenburg’s legendary GML 8200 equalizer — the industry-standard reference. The plug-in offers a choice of three or five bands of equalization (both versions support the ultra-wide 10 Hz to 41 kHz frequency range) to target certain frequencies, or conserve DSP resources when only three bands of EQ are needed. <BR><BR><B>Professional Features<BR></B>MDW Hi-Res Parametric EQ offers a number of features to help you achieve better mixes. It allows you to view the frequency grid in 6, 12, or 24 dB resolution for more editing precision. IsoPeak® lets you quickly isolate and sweep through the frequency spectrum to find a targeted frequency. You can also toggle between two EQ band settings using A/B snapshots, copy one setting to the other, and invert the phase of a track. And when used with an ICON worksurface or Digidesign® control surface, the EQ organizes the bands into ergonomic groups on the channel strips. <BR><BR><B>High-Resolution Filtering</B><BR>MDW Hi-Res Parametric EQ 3.0 lives up to its high-resolution name. The plug-in delivers double-precision 48-bit processing for optimal clarity with natural-sounding results. It also provides high-resolution 96 kHz sample-rate processing for audio recorded at 48 kHz or 96 kHz, and 192 kHz sample-rate processing for 192 kHz audio, allowing for increased headroom, fewer artifacts, and more predictable filter curves. And with a variable Qs (adjustable bandwidths) from 25.6 (1/18 of an octave) to 0.1 (6-2/3 octaves), MDW Hi-Res Parametric EQ 3.0 delivers unmatched precision. <BR><BR><B>Excellent DSP Efficiency<BR></B>MDW Hi-Res Parametric EQ 3.0 allows you to tailor the number of frequency bands for better DSP processing efficiency. Use the three-band version to conserve resources, allowing you to instantiate the EQ plug-in up to eight times per DSP chip at 48 kHz and 96 kHz on Pro Tools|HD® Accel systems. With the five-band version, you can instantiate the EQ plug-in up to four times per DSP chip at 48 kHz and 96 kHz on Pro Tools|HD Accel systems. <BR><BR><B>System Requirements</B> 
<UL>
<LI>Digidesign-qualified Windows- or Mac-based Pro Tools|HD®, Pro Tools LE®, or Pro Tools M-Powered™ system running Pro Tools® 7.3 or higher software 
<LI>iLok USB Smart Key (sold separately), Internet access, and free iLok.com account for authorization </LI></UL>

Avid MDW Hi-Res Parametric EQ 3.0 - TDM & RTAS
Optimum High-Res Digital EQ Plug-in Featuring Double Precision Processing

If you mix in Pro Tools and want your final audio tracks to have that professionally mastered sheen, add <span style="font-weight: bold;">Avid Maxim</span>
to your list of "must have" plug-ins. More than just a world-class peak
limiter, Maxim optimizes the overall level of the audio input while
preserving the integrity of the original sound. Although Maxim is
ideally suited to the stereo master of a mix, it's flexible enough to
be used as a dynamics processor on any channel in the Pro Tools mixing
environment. <p>
Maxim also offers built-in dithering, on-line help, and a full-color
Histogram, making it an indispensable plug-in for mixing and mastering
in any Pro Tools environment. </p><p>
<b>Features:</b> <br></p>
<ul><li>

High-quality optimization of audio levels </li><li>

Advanced, look-ahead peak limiting</li><li> 
 
Full-color Histogram </li><li>

Professional sound quality </li></ul>
<p> <br>

<b>System Requirements: </b>

Digidesign approved Pro Tools system running 4.1.1 software (or higher).</p>

Avid Maxim
Peak Limiting and Sound Level Maximizing Plug-In for Pro Tools ( TDM + RTAS + AS)

Tightly integrated with Digidesign Pro Tools software, the Neyrinck SoundCode for Dolby Digital plug-in suite provides mastering-quality workflow tools that enable you to encode and decode Dolby Digital (AC-3) audio directly within Pro Tools software, without the need for dedicated Dolby hardware. Whether you are mixing in surround, delivering surround audio for DVD video, decoding audio from a DVD video within Pro Tools, creating a DVD reference disk, or producing a surround-compatible audio CD, SoundCode provides a range of powerful features to help you accomplish your goals, all within the industry-standard Pro Tools environment. <BR><BR><B>Faster-Than-Real-Time* Encoding </B><BR>The SoundCode Encoder offers a fast way to encode audio as AC-3 and WAV files from a Pro Tools session, ready for use with DVD audio or video authoring applications, CD burning applications, or any other application that uses the Dolby Digital standard. Because the SoundCode Encoder operates as an AudioSuite™ plug-in, you can encode Dolby Digital faster than real time* to accelerate your workflow. SoundCode encodes to all AC-3 file types, including AC-3 with time code. <BR><BR>SoundCode also encodes to 16- or 24-bit stereo interleaved WAV files so you can lay back the Dolby Digital audio to digital tape or create an audio disc for playback on a consumer stereo system that has a Dolby Digital decoder. With the SoundCode Encoder, you can adjust preprocessing settings and add extensive metadata to ensure greater consistency in the presentation of audio on a variety of playback systems. The SoundCode Encoder also gives you the option to import the encoded Dolby Digital audio back into the Pro Tools timeline, where you can decode and monitor it using the real-time SoundCode Decoder. <BR><BR><B>Punch In/Out Capabilities </B><BR>SoundCode includes a time-saving Punch In/Out encoding feature that allows you to punch in fixes to an existing AC-3 file. If you've already encoded the audio for a two-hour program but need to replace a single line of dialog, SoundCode lets you encode just that line, without having to re-encode the entire two-hour soundtrack. SoundCode then automatically “punches in” that line to the correct location within the previously encoded file.

Avid Neyrinck SoundCode for Dolby Digital
Dolby Digital Encoding Plug-in for Pro Tools - RTAS

Sonic NoNOISE has long been recognized as the world's premier solution for restoring archival audio recordings. With advanced processes to isolate and eliminate audio artifacts such as hiss, scratches, hum, mechanical and impulsive noises, NoNOISE can be used to restore old recordings, remove unwanted noises from field and location recordings, and repair audio materials that have suffered damage. <P>

The NoNOISE suite of products encompasses a number of specific processing technologies:<P>

Manual Declicking provides tools for isolating and removing individual clicks within soundfiles. <BR>
Production Declicking detects and eliminates clicks automatically in a multi-pass process, and can be used to remove thousands of clicks within a recording. <BR>
Broadband Denoising removes broadband noise, or hiss, which is one of the most common forms of audio degradation, by analyzing noise content dynamically adapting the denoising operation to the characteristics of the material. 
<BR>
Decrackle removes another common type of impulse noise called crackle, in which small impulses crowd against one another, producing a nearly continuous noise, like bacon frying in a pan. 
<BR>
High-Resolution Filters — include Presence, Notch, High and Low Shelf, High and Low Pass, DC-Removal, Emphasis, De-emphasis, RIAA and more. 
<P>"Sonic NoNoise is an extremely powerful arsenal of audio restoration techniques that no digital audio engineer should be without," said Rolf Hartley, General Manager of Professional Products at Sonic. "Sonic is extremely pleased to make our world-renowned NoNOISE suite available to the vast installed Digidesign user community."

Avid Sonic Solutions NoNoise Bundle
Powerful Noise Reduction/Audio Restoration Plug-in Suite

Bomb Factory Pultec Bundle
Bomb Factory Pultec Bundle

<b>Automatic dehissing that <i>actually</i> works is now a reality!</b><br><br><b>The Problem <br></b>In the past, digital dehissers that used no fingerprinting or noise reduction profiling were prone to side-effects known variously as 'twittering' and 'glugging'. On earlier CEDAR processes, it was possible to avoid these unwanted sounds by careful and judicious use of the controls, but it has never been possible - until now - to remove broadband noise correctly with almost no user intervention.<br><br><b>The solution</b><br>"The first so-called 'automatic' dehisser that works! And it really does..." <br><br>Auto Dehiss embodies a more advanced algorithm than any previous dehisser. It offers significantly improved performance and has a unique "Auto" mode that enables the software to determine the broadband noise content, removing this without the introduction of unwanted side-effects or artifacts. We have even retained a manual mode that allows you to control all parameters and fine-tune the noise reduction.

CEDAR Audio Auto-Dehiss 3
Automatic Hiss removal module for Cambridge Host System

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