The limiter plug-in is comprised of four cascaded processes in the order below. Pre-processing gain control section, Programme enhancement and overshoot control section, Reconstruction metering and compensation section. Dithering and noise shaping section.
The primary purpose of the plug-in is to control peak levels and increase the volume, density and presence of musical programme without the excessive loss of transient and dynamic information that normally results from conventional peak limiting applications. In order to achieve this the application employs gain scaling and compression in the pre-process section and peak overshoot control in the enhancement section. These processes are used in conjunction to provide a means of allowing the sound of transient and dynamic information within your programme to exist in the output from the plug-in, despite very tight control of maximum peak sample values. The normal enhance control setting for this action to fully occur is 100% and above in normal mode, (or any enhance setting with Safe mode selected). A variable control is provided to allow the enhance process to be adjusted or removed completely (i.e. at 0% setting), in which case the pre-process section may be used as a conventional programme levelling application if desired. A Safe mode is provided which uses the enhance processing to permanently control peak levels. In this mode the enhance control varies the perceived loudness boost of the programme by modifying the processing law.
The Pre-process section is a musical programme levelling function. Its primary purpose is to control programme level over a wide range in order to provide optimum conditions for the following enhancement stage. When the enhancement is disabled in normal mode (at 0% with Safe mode deselected) the pre-process section can be used as a conventional levelling section in its own right. For clarity of operation the plug-in gain structure is arranged with the threshold permanently set to 0dB ref input normal level. Dynamic gain reduction is achieved by increasing the input gain sufficiently for the internal signal to surpass the 0dBr reference level. A total of 18dB gain boost is available for this purpose and the orange section of the input meter illustrates the level range within which gain reduction occurs when the plug-in is active. The final output modulation level is set by the output level control, which can be adjusted to compensate for dynamic conditions produced by the programme and limiter settings or produce a lower level mastered output if required. Programme gain is accurately controlled by look-ahead detection (to allow action before peaks are encountered) coupled with a logarithmic side chain employing multiple interdependent timing functions. Timing controls are provided to modify its action depending on programme type and production style. In order to reduce excessive shortterm gain modulation a selectable auto gain function is included which compensates for wider input level variations by imposing a longer-term time constant that underlies the peak timing. A variable progressive soft knee function provides varying degrees of soft limiting (for lower settings), right up to large-scale gain management active over the final 10dB of programme dynamic range.

Enhancement section.

The purpose of the enhancement process is to provide sample value limiting and overall programme loudness improvement. The process follows the pre-process section in the signal path and is controlled by a separate fader from 0% (no action) to 125% (maximum action). In normal mode the range from 0% to 100% fades in the effect to full level, at which complete sample value limiting occurs. Settings from 100% to 125% further modify the process to progressively increase loudness and programme density at the expense of increasing potential distortion artefacts. Safe mode is provided to allow absolute peak level control without excessive enhancer action, even when using slow attack settings. In Safe mode the enhance process is set to run permanently and the enhance slider modifies the action of the process (rather than the proportion of the effect). Setting ranges from 0% to 100% control the degree of programme loudness boost generated by the enhancer. The control region from 100% to 125% works identically to
normal mode. It should be noted that in safe mode signals at all levels are being processed permanently, therefore some minor changes to the programme dynamics can occur even for a minimum setting of 0%. The enhance process improves the perceived loudness and presence of the programme by modifying the dynamic and harmonic content of the signal. Since the method used is different from the pre-processing section, it can further enhance the perceived volume of a previously processed signal whilst suppressing all signal overloads. As the limiting action does not involve conventional sample value clipping, harsh distortions are avoided and programme detail and dynamic information is largely retained. Also, since the plug-in has internal headroom, transient levels greater than notional maximum modulation can pass from the limiter section into the enhancement stage. This means that percussive overshoots, that would normally be lost in a conventional limiter (or would give rise to overloads), may be included within the sonic results of the plug-in, producing both richer dynamic sonic detail and a useful reduction in the perceived artefacts of the limiting process, all without giving rise to any sample value overload. It is this property that allows slower attack times to be operated in the pre-process section without creating output overloads, which would otherwise result in a need to reduce output levels. The enhancement section can be used effectively on its own to produce programme enrichment and peak value limiting, by using minimal gain reduction and slow timing settings in the pre-process section. Or it can be used to enhance highly processed content from the pre-process section to achieve even greater perceived loudness. Because the enhancement process adds harmonic distortion during dynamic within the programme, under some conditions side effects may occur depending on the content of the programme material. Generally speaking side effects should be minimal when in Safe mode and for boost settings up to 100% when mastering in the presence of most commonly occurring complex and dense composite material. However some extra care may be needed in the case of single solo instrument tracks where there may be a predominance of sustained lower and middle frequency content. Settings above 100% are most useful when the programme type is intended to be very loud or where extra distortion may actually prove advantageous within the style of the production.

Attack timing.

The addition of an attack timing control is a significant departure from conventional limiter applications and requires some explanation for best results. Because the level detection within the plug-in looks ahead of the gain control, peaks in the programme are acted upon in advance of the gain reducing process. Therefore at the fastest setting of attack control programme peaks are controlled within a very small margin (less than +0.25dB with respect to continuous sine input conditions). The attack control provides a means of increasing the attack time to achieve a favourable improvement in the sonic qualities of the peak reduction process by allowing peak programme transient events to escape hard gain reduction. Since the plug-in has internal headroom these overshoot peaks are retained and not clipped. Peak overshoots resulting from the combination of the programme material and the action of the pre-process stage are then passed to the Enhancement section where their sonic signatures can be added to the final programme sound. Providing Safe mode is selected or the enhancement control is set to 100% or more in normal mode, no output sample value overloads will occur from the plug-in, regardless of attack or release time settings. Therefore a combination of slower attack times and the enhancement process is a very powerful way to include transients in the output programme that would normally be removed by conventional limiting processes. It can create a sonic quality and impression of dynamic range that belies the degree to which the programme is actually being limited. As an example of the difference this can make, if slower attack settings are used without the enhancer and the output gain is reduced to accommodate the overshoots (avoiding over loads), using the same attack settings with the Enhancer can provide up to (and beyond) a 3dB increase in average level due to being able to legally increase the output gain setting and around another 2dB of perceived loudness due to the enhance action itself. This can result in a perceived loudness increase of 5dB to 6dB! In general very fast attack times will more readily remove extremely fine detail and short-term events, but will produce greater harmonic disturbance in operation. Slower attack times will progressively allow finer detail to escape the harsh sound of fast limiting and longer term events will tend to assume a more rounded peak profile. Such settings are usually kinder to the musical programme.

Please note that the only way slower attack settings can be used without a potential need for a significant reduction of the output level control is when the Enhancer is set at or above 100% or with Safe mode selected.
Note. Using Pre-process section without the Enhance section.
It should be noted that since the pre-process section is a programme gain controller rather than a simple sample clipper, programme peaks can cause a small increase in maximum output sample value, even at the fastest attack time settings. If Safe mode isnt selected or the enhance section is not active at or above 100% setting in normal mode, these peaks will appear at the output of the plug-in. Increasing the attack times will further increase peak overshoots, so if tight level control is required without the enhancer it is best to leave the attack at minimum setting. Since the plug-in has internal level headroom, the output level control can be safely used to compensate for any artistically intended overshoot without fear of causing internal signal clipping.

Release timing.

The release control has a very wide range to accommodate the maximum possible range of programme and production style. The ability to set very fast release times is provided specifically to allow for the modelling of short-term peaks over restricted gain reduction ranges up to a maximum of around 4dB. Such settings will result in high levels of distortion for larger gain reduction ranges and are therefore unsuitable for overall level control situations. The auto gain function can be used very effectively to compensate for large level changes whilst still allowing fast peak modelling for shorter peak events, as attack and release controls are permanently functional. So it is a good idea to keep the auto gain active under most circumstances. Generally speaking faster release times produce the greatest perceived loudness, since gain recovery happens quickly after peak events have passed and average programme levels are affected only during the shortest possible periods. However, since the gain recovery can begin to occur between the waveform peaks of lower frequencies in the programme there is a trade off to be made between the speed of release and the generation of distortion. Such distortion may be desirable under many conditions, particularly in loud popular music productions where some low frequency harmonics may add warmth and presence to the programme. Adjusting release timing over a wide range provides a method to tune these effects to suit the production style. Longer release times are far more forgiving of gain changes and allow greater overall compression, but will result in a quieter sounding output programme. If the auto gain function is not selected moderate release time settings (above around 0.2 seconds) may produce audible gain pumping due to longer and more noticeable recovery periods. If such settings are needed the aim becomes one of fitting the release timing to the natural rhythm of events in the programme. Under these conditions better results may be achieved by increasing the input gain somewhat thereby compressing further and applying the Soft Knee function in order to compress gently over an increased portion of the dynamic range. In this way the transition in and out of compression will become gentler and less obvious.

Programme limiting procedures.
There are many approaches to limiting within current productions trends, but most approaches fall into two categories; loudness maximisation and general gain control. A very wide range of control is provided in the plug-in to make both these situations possible with ease. The key to successful limiting is to understand that we are much more sensitive to the rate of change of gain than we are to absolute level. Therefore successful limiting has a tendency to fall into an appropriate mixture of two simultaneous but conceptually separate actions. Fast control over small level ranges - because they are too quick for us to notice and too small to produce damaging harmonic distortions. Slow control over larger level ranges - because the gain changes are slow enough to escape obvious notice and the rate of change of level is slow enough to avoid intrusive modulation effects and distortion.
5.5.1. Loudness maximisation.
The aim of this procedure is to achieve an overall average increase in the level of the programme by reducing the size of short-term peaks and applying extra gain to move the programme up into the extra range freed up by the removal of the peaks. Signal before limiting.

Signal after limiting.

To achieve this it is customary to select relatively fast attack and release times whilst judiciously increasing the input levels so that only the offending programme peaks are subject to reduction by the limiter and the average modulation level is increased. The Sonnox Limiter can produce significantly superior results in loudness maximisation because it can fully limit the signal even when using slower attack times. This leads to much lower distortion and less removal of dynamic programme information. The timing controls on the pre-process section of Limiter can be used freely to make subtle modification to this process in order to achieve the best possible results.
5.5.2. General gain management.
The aim of this procedure is usually to preserve the short-term dynamics of the programme as far as possible, whilst ensuring that no levels surpass maximum peak modulation. This most often entails responding to the peak level of the programme as quickly as possible and re-scaling the gain in the longer term, such that musical dynamics are only minimally affected in the short term. The Limiter can excel in general gain management because of the wide range of control in the pre-process section and the ability of the enhance stage to control the level of short term peaks, which means that musically kinder attack and release times can be used without risk of transient over loads. Moderate attack times and slower release times usually perform best for this function. The Auto Gain function is particularly useful in this case as it provides a method to achieve long term gain control whilst allowing a degree of fast gain riding over a reduced range. In this case moderately fast recovery times can be arranged to control short-term events with the auto gain managing long-term level changes. Higher Soft Knee settings can also considerably improve perceived quality as it acts like a variable ratio compressor that starts at lower levels. This allows the limiter to pre-view the signals at moderate levels and reduce the rate of change of gain in the loudest peak regions.

Reconstruction metering section.
An important fact which is often overlooked is that in any discrete time sampling system it is possible to create sample values that may not be decoded and reconstructed correctly. Whilst it is true that if left unchanged a signal properly converted into the digital domain by a perfect ADC will always produce sample values that can be legally decoded in a perfect DAC, further processing of those samples can result in a decoded signal that is illegally high and therefore may not be faithfully reproduced, even if no sample value limiting occurs. The following is an illustration of one example of this situation.
With a signal frequency that is exactly of the sample rate and in phase (i.e. 12KHz at 48K or 11.25KHz at 44.1K) it is possible to generate full level output from a DAC from samples that barely reach 70% of full value within the digital domain. If the level of this signal were increased towards maximum allowable sample value (close to +/-1) the area representing the reconstruction provided by the DAC filter would produce signals that are considerably in excess of maximum modulation. Since the vast majority of metering within workstation environments responds to sample value only, the above example would show a level of around 3dB below clipping. However any further increase in the level of the signal would result in a potentially illegal output level from the system converter. As this error would not be reported on metering within the workstation, in this particular case a possible 3dB overload can result if the signal is increased to a maximum reading on the workstation meters. This phenomenon is sometimes termed as inter-sample peaking. Although the above example is somewhat extreme and specific, there is plenty of potential for this to occur within the mixing environment. Combining a number of processed contributing tracks and limiting the result to the maximum possible modulation level in order to satisfy current industry trends, using only peak value metering, provides a recipe for such hidden errors. Since the very purpose of a limiting application is most often to increase average modulation levels, a reconstruction meter with both manual and automatic correction processing has been included in the Limiter plug-in, in order to provide the user with a method to avoid or repair such errors.

Meter operation.

When the Recon Meter is selected the meter is switched from conventional peak sample value mode into reconstruction mode. In this mode peak reconstruction levels will be displayed on the meter. Levels in the red overload range of the meter represent the presence of potential reconstruction errors, as illustrated below using the previous example.

Two methods are provided to correct for this. Since the output level fader precedes the metering, errors may be corrected manually by simply reducing the output level setting by the same amount as the maximum error level reported on the meter.

Auto Comp.

Under normal circumstances errors are interspersed throughout the programme often restricted to certain specific events. The Auto Comp function is provided to address the situation where it may be undesirable to reduce the level of the whole programme to avoid transient errors. When Auto Comp is selected the level of the output is automatically controlled to repair reconstruction errors by the minimum amount required and only for the duration of the error. In this way the loudness of parts of the programme unaffected by the errors remains as high as possible. A combination of Auto Comp and manual output level reduction can be used to strike a compromise, if the action of the error correction becomes intrusive in the presence of very large and intermittent error conditions.
Dither and Noise Shaping.
The finite mathematical precision provided by digital audio systems and the effects of dither have been a source of confusion in the audio community for some years. Such discussion may lead to possible misconceptions, which could prevent the user from achieving maximum performance from systems in use. Therefore the dithering options provided in the Limiter warrant some prior explanation.

Conventional Dither.

In both 24bit and 16bit output word length selections, high pass TPDF (triangular probability density function) dithering is applied to the output of the plug-in. Since any signal related error caused by finite word length limitation is turned into constant random noise with no relation to the signal itself, such dithering provides complete removal of harmonic distortion due to precision limits, which are an inescapable result of any numerical signal representation. Dithering also suppresses any possibility that the programme will suffer loss of harmonic signal resolution due to word length restriction. The following plots illustrate this in action.
The first graph shows the damaged spectral result of passing low-level 1KHz sine wave through a 16 bit undithered truncation.
The next plot shows the exact same signal and truncation to 16 bits but with the HP TPDF dither applied.
It can be seen that all harmonic errors have been removed. Also since the FFT analysis method provides an enhanced view of the signal below the noise floor, it can also be seen that there is effectively no low level floor below which a signal will fail to pass. To illustrate this fact the following plot shows a 1KHz signal at 120dBr passing through a dithered 16bit system. This corresponds to a signal 24dB below the level of the least significant bit, the effective channel SNR is added in blue for illustration purposes.

This shows that dither turns a quantised numerical signal conduit into the equivalent of a naturally continuous (unquantised) system, which exhibits a finite signal to noise ratio with no practical limit to harmonic signal resolution. In other words the inescapable presence of quantisation in numerical systems does not forcibly lead to discontinuity or resolution loss in the signal. Misunderstandings of this fact underpin many of the most damaging misconceptions surrounding digital audio systems. It can also be deduced from the above plots that any undithered digital representation of an audio signal is effectively illegal.

Noise Shaping dither.

If for some reason SNR figures of 93dB at 16bits (or 143dB at 24bits) prove insufficient, noise shaping can provide an apparent increase in SNR, but there are some potentially hidden costs. Noise shaped dithering is a mechanism that aims to reduce the perceived loudness of the noise of a dithered signal by either forcing the spectrum of the noise out of the audible range or placing it into frequency ranges to which we are less sensitive. In this way the noise at very low levels may be reduced and even lost entirely if they are at the limit of our hearing within ambient noise conditions. The following plot illustrates this process.
The red line shows the original 16bit dithered output with the 120dBr signal passing. The blue line shows the effect of noise shaping (type 1 at 100%) on the same signal transfer. It can be seen that the noise has been substantially reduced in the regions up to around 8KHz where we are most sensitive, at the expense of extra noise energy in the higher ranges above 10KHz at which we are less sensitive. Such processing can have a dramatic effect on the perceived intrusion of low-level noise. However as is always the case, one cannot get something for nothing and it can be seen from the above plot that the total noise power across the whole range must remain constant to satisfy the dithering requirement. This means the noise level necessarily increases in some ranges of the spectrum. A level increase anywhere in the spectrum must be accommodated by an increase in total peak noise level. The following plot illustrates this in action with the previous test conditions.
The first plot shows a sample value plot of the conventional TPDF dithered signal. The second shows the increase in values caused by the application of the type1 noise shaping at 100%. Focussing the dither energy into a more restricted range than would naturally occur causes the level to increase.

From the effective level values highlighted in the plots it can be seen that the application of noise shaping has increased the effective noise level from around 93dBr to 80dBr, an increase of roughly 13dB. From this it can be understood that the design of suitable noise shaping frequency curve is a trade off between the perceived loudness of the noise under certain conditions and the increase in overall level of the dither signal, much of this trade off relies on what we can hear (psycho-acoustics). Significant research has been carried out over the years into various approaches to this issue and several accepted curves are in use around the industry. The Oxford Limiter includes 4 noise-shaping curves. Types 1 and 3 are fifth order and types 2 and 4 are third order designs representing a varied set of trade-offs to suit most programme types, as illustrated below.
Whilst it is understood that the selection of noise shaping type is largely a matter of user preference, generally speaking types 1 and 2 produce the most dramatic reduction in overall noise loudness, with type 1 being the most effective of all. Types 3 and 4 provide gentler responses, which under some circumstances may produce less intrusive sounding spectrums, at the expense of higher audible residual noise. Type 3 also provides greater noise attenuation in the range between 10KHz and 16KHz at the expense of higher noise levels in the mid ranges.
Noise shaping Depth control.
From the previous section it can be seen that noise shaping can potentially cause unwanted effects in equipment and processes down line, particularly if the programme is to be further modified, such as in mastering situations. Some unwanted effects may include: Marked increase in noise levels if the file is not transferred intact bit for bit, (i.e. if further processing is done). Premature meter readings in silence. Premature peak level over loads (as increased dither levels add to peak signal value). Unwanted low-level behaviour in dynamics processing. Disturbance caused to data reduction encoding processes such as MP3, WMA etc. Increased audibility (unmasking) of various errors that may occur in play-out systems.

PowerCore (Macintosh)

Double click the.dmg installer file or icon for your product installation to begin. Follow the onscreen instructions. When you install your plugins, they will be placed into the /Library/Audio/Plug-Ins/VST/PowerCore/Sony folder, and you have the option of registering them for use as Audio Units compatible plugins. The new mono-only versions will be installed into a sub-directory of the above path called Mono. The new replacement backward-compatibility versions will be placed into a sub-directory of the above called Backward-Compatibility. Please note: In the past there have been unreliability problems with the process of wrapping Oxford plug-ins for Audio Units compatibility. To improve this situation, the downloaded installer.dmg volume now contains the latest TCAU Patcher installer supplied by TC Electronic. Use this if you wish to be sure that your plugins are wrapped correctly for Audio Units compatibility.

PowerCore (Windows)

If any older versions of the plugins are installed, move them to a safe storage location outside of the VSTPlugins directory so that hosts will not find them. If you do not do this, the loading of old sessions may fail to find the new plugins correctly and there may be conflicts that lead to crashes. Double click the installer icon for your product to begin, and follow the onscreen instructions. When the plugins are installed, the setup program will attempt to detect your shared VSTPlugins directory. However, you may also select another location if desired. The default installation location is \VstPlugins\PowerCore\Sony for all the new combined auto-mono-stereo versions, and \VstPlugins\PowerCore\Sony\Mono for the new mono-only versions, and \VSTPlugins\PowerCore\Sony\Backward-Compatibility for the new replacement versions of the old mono-only or stereo-only plug-ins (applies only to the Oxford EQ and Oxford Limiter plugins.)

VST Native (Windows)

Double click the installer icon for your product to begin, and follow the onscreen instructions. When the plugins are installed, the setup program will attempt to detect your shared VSTPlugins directory. However, you may also select another location if desired. The default installation location is \VstPlugins\Native\Sony.

Audio Units (Macintosh)

Double click the installer icon for your product to begin. Follow the onscreen prompts. You will need your authorised iLok plugged into a free USB port on your machine at all times when using the plug-in.

Mono, Stereo and Mono->Stereo Versions (PowerCore).
The Sonnox Oxford Limiter version for TC PowerCore automatically detects whether Stereo or Mono operation is called for by requesting the number of input and output audio channels from the host. With one in one out, mono operation is selected. With two in two out, stereo operation is selected. With one in and two out, stereo operation is selected and the single input is automatically duplicated to both DSP channels. The Oxford Limiter comes as 6 DLLs (Windows) or 6 Bundles (Mac). The two main ones to use are Oxford Limiter and Oxford Limiter Direct. These will work in both Stereo and Mono mode. The other four are provided to maintain backward compatibility so that projects saved with the old mono-only or stereo-only versions will continue to work. On Windows XP, the stereo-only and mono-only DLLs are contained in directory \VSTPlugins\PowerCore\Sony\Backward-Compatibility, and the new stereo or mono versions are contained in \VSTPlugins\PowerCore\Sony (the new location for Oxford plugins for PowerCore.)
Integrated Native-PowerCore Versions (VST-only)
The VST PowerCore plugins for Mac/Intel-Mac (and WinXP soon) have a major new feature the Native DSP code is now included and integrated seamlessly.
Project Portability, Upgrade-ability and Fail-Safe Working
This allows for a whole new set of freedoms when it comes to working with Powercore. For example: Imagine working in your studio using a Powercore-accelerated system. Some of your Oxford Powercore plugins are in Powercore mode, and some in Native mode, to allow you to balance your resources. You save your project and copy it to your laptop (which has no Powercore hardware), and the project runs error free on your laptop because your Oxford Plugins automatically and quietly revert to native processing mode if they do not find any Powercore hardware. So you can continue to work on your mix as you travel on the train or bus or while on tour. When you return to your studio, you can copy your project back in, and your plugins will automatically revert back to Powercore, if they were originally set to Powercore mode. Effectively, this means your Powercore-accelerated system will run with less CPU burden than your non-powercore-accelerated one, but your project is completely portable between the two. This allows an unprecedented level of portability when using Oxford Plugins. You no longer have to change the Oxford Powercore plugins to Native plugins or vice versa. You simply click on the switch and it is done, with the settings preserved. Now imagine being a Native plugin user who is running out of CPU power. What you want is to use a hardware accelerator to free your CPU of some of its burden, but without the hardware accelerator taking over your system and the way you work, without having to swap out Native plugins and replace them with Powercore versions. With the new combination Native-PowerCore plugins, this is exactly what you can do. You can work in Native mode until you run out of CPU. Then you can begin switching some of your plugins over to using hardware acceleration (with Powercore cards), and they will run in Powercore mode if the hardware is available, and in Native mode if not. If there is any problem with the hardware that would otherwise cause you to be unable to run the plugins, they simply revert to native mode. Thus the inbuilt native mode acts as a safe-guard against hardware failure, or as a safe-guard against running out of hardware resources. Finally, forward thinking individuals can purchase or upgrade to the Powercore plugins as an insurance against running out of CPU, long before they even have any Powercore hardware. If or when the individual decides they need hardware acceleration, they can purchase it, and their old projects will continue to work on the new hardware.

How it works.

The new Powercore plugins have some new menu options and indicators to allow you complete control over the way this new feature is handled. Firstly, on the top right of the plugin in the title bar, there is a new icon that shows the current processing mode that the plugin is actually using, either Powercore or VST Native, like this:
Clicking on this icon brings up a new menu, one that allows you to set which mode you want the plugin to run in, your desired mode of operation: either Use PowerCore hardware if it is available, or Use Native DSP. If you select Native mode, this instance of this plugin will run in Native mode from this moment forward. If you select Powercore mode, this instance of this plugin will use powercore hardware if it is available, or it will automatically switch to native mode if there is no Powercore hardware available, or if there is a problem while loading onto Powercore (such as there are no more hardware resources left, or there is a problem with the fire-wire connection). Your desired mode of operation is saved with the project so that when you load your project back in, the desired mode returns back to what it was, on a plugin by plugin basis. When you load presets using the on-board preset manager, this desired mode data in the preset is ignored, so your desired mode stays as you set it when trying out different presets. However, when you load presets with the host, or load a project, this mode is acted on so that a project load restores your plugins exactly as you left them. When using a desired mode of Use powercore if available, it is quite normal for the icon to display VST, rather than Powercore. This shows the actual mode is currently Native mode, for one reason or another.
Switching between Native and Powercore mode can be done at any time, even while playing. However, since switching while playing will cause a temporary discontinuity in the sound, it is wiser to switch while stopped. By default, the automatic switch over from Powercore to Native (for example because the plugin failed to find any powercore hardware) occurs without error messages being displayed. The menu that pops up when you click on the above icons also has an option that allows you to know about any messages that occur. This is useful, for example, if you are wondering why your plugins are reverting to Native mode when you are expecting them to stay in Powercore mode. In this mode, all error messages will be displayed except for the case where there is no hardware, or there are no powercore drivers (in which case it is obvious why the plugin is running in native mode.) Lastly, there is now a new preference in the main menu under the Sonnox button that allows you to specify what your default desired mode is for this plugin type. If you prefer your plugin to come up in Native mode when you insert a new plugin, then set this option. If you prefer it to come up in powercore-if-available mode, then leave this preference not set.

Native versus Powercore

Generally speaking the Native version of the DSP is identical to the Powercore DSP in terms of the algorithm and order of processing. However, in plugins involving dynamic gain changes such as the Oxford Dynamics, Oxford TransMod and Oxford Limiter, the gain calculations are done in the logarithmic domain. The Native DSP uses a true logarithm/anti-logarithm whereas, by necessity, the Powercore DSP versions have to use a very close approximation. This means that for the Dynamics, TransMod and Limiter, the sound of the Powercore DSP is not 100% exactly the same as the sound of the Native DSP. Neither one sounds better nor worse; it is just that if you try to cancel them out by putting one in anti-phase, they will not quite cancel all the way. Native mode is best for recording since the inherent delay is smaller. For mixing, when the number of inserts can climb to being very large, this is when using hardware acceleration such as Powercore can really help to relieve the burden on the CPU so that you can still listen in real time. When changing from one mode to another it is important that you are aware that the delay also changes, and this will upset the delay compensation that your host has arrived at. Although the plugin does re-export the new delay, and tell the host that the delay has changed, you may need to save your project and reload it in order for the host to set up the delay compensation correctly. Indeed, in Cubase/Nuendo, this has always been true for the Oxford Powercore Plugins regardless of this new feature. This is because the delay is dependent on the sample rate and block-size, and so cannot be determined correctly by the plugin until the plugin is fully loaded, by which time the host has already set up the delay compensation. One final contrast is that some of the plugins have slight functionality differences when in Native mode. For example, the Limiter and Dynamics plugins have the option of not applying dither when in Native mode. This means that if your plugin is set to the no dither option and you switch to powercore mode, it will have to revert to using 24 bit dithering, because when using Powercore it is imperative that the signal is properly dithered to preserve sonic integrity. Other examples of differences occur in the preferences menus. For example, when in Native mode, the No Latency option has no meaning, and is either greyed out or replaced with a preference that is Native specific such as Enable 24 bit dithering.

Presets

With regards to presets, the installer installs the factory-supplied presets into two locations firstly, as VST presets into the default place for the on-board presets manager to see (/library/Application Support/Sonnox/Oxford Plugins/Presets/Powercore), and secondly, as Audio Units presets into the default place for Logic to see (/library/Audio/Presets/Sonnox (AU Poco)). This allows you to use either of these mechanisms for loading factory presets. There is one important purposeful difference between loading a preset with the on-board preset manager, and using the host mechanism: Loading with the on-board preset manager will leave the Native-Mode/PowerCore-Mode switch as is, where as loading with the Host mechanism will force the mode switch to be as stored in the host preset data. The reason for this is so that loading a project in any Host will return the project exactly as you left it, especially

www.barryrudolph.com Cakewalk Rapture Radial The Radial Reamping pack is their J48 direct box and X-Amp active reamplifier packaged together in a lightweight, hard plastic Zebracase. The J48 is an active direct box powered by 48volt phantom power that is well suited for high impedance sources like guitars Reampin9 Pack =
Rapture is the second (after Dimension Pro) in Cakewalk's new line of virtual synths that will run on any DAW/sequencer system platform in common use. Rapture's installer CDRom includes versions in Audio Units (AU), RTAS, D xi (both 32 and 64-bit) and VSTi formats. Rapture is a wavetable synth with a virtual front panel that enables endless tweaking of any of the sounds in its vast library of over 600 sounds. The 600 includes Basses, Leads, Pads, Arpeggios, Textures, Electronic Percussion and Sequences. I installed Rapture's RTAS version into my ProTools rig and went to work. I liked the modern looking interface, the comprehensive MIDI Control Matrix and the XN pad for assigning multiple Rapture parameters to X and Y-axis -this is cool for live performance. This synth has a lot of electronica/dance/rave/techno oriented sounds, but the interface is so complete, you can take any of the sounds and change them radically to fit any musical style/genre. Features you'll find important for programming and operation include: pristine sound quality due to the non-aliasing resampling engine; six fully programmable, stereo sound-generating elements with mixer; multi-waveform oscillators with ring modulation; two 16-mode filters per oscillator with LoFi and Drive effects; over 40 envelope generators, LFOs, and step generators; MIDI Modulation Matrix; and global FX and dynamic Step Generator. Basically you can mix together up to six different elements that comprise a program. The Global FX Stage has three multi-effect units with nine delays and three different choruses, three parametric EOs, and a dynamic Step Generator for volume with independent UR control. More details can be found at www.cakewalk.com or you can call 617-423-9004. Rapture sells for $199 MSRP. I
and basses. It has a -15dB pad, ground lift, feed-through for sending to an amp, polarity reverse, 80Hz high-pass rumble filter and a unique merge mode for mixing two signals (like a stereo keyboard) down to mono. X-Amp is a reamping device -an interface that allows low impedance pre-recorded line level signals from your tape deck or DAW to be sent to the high impedance input of a guitar amp and/or pedal effect. Features include: XLR balanced line level input with LED peak indicator, level control, all Class-A circuitry, both direct and transformer-isolated guitar amplifier outputs, dual ground lifts, and polarity reverse for correcting the inherent phase differences between guitar amps. Reamping is a technique that reduces the urgency and stress of getting the 'ultimate guitar sound' during a session. We used the reamping kit while working with a band in a small DAW based writing room. There were no amps in the room, but with the reamping kit we were able to record the guitar direct with the ,)48 while simultaneously sending the signal to a Line 6 POD-XT for a quick guide sound. Later, with the performance perfected and recorded, we went to another room full of a great assortment of guitar amps and vintage outboard gear. Using the X-Amp, we connected the audio from the direct guitar track to an amp and effects and tweaked to our hearts content, creating the coolest guitar sounds without worrying about slowing down the creative process. We love this newfound ability so much that the Reamping Kit has now become a mainstay in our recording/production workflow. Both the J48 and X-Amp are housed in 14-gauge, crush-proof I-beam framed boxes and come together in the attache-style Zebracase. Call Radial Engineering at 604-942-1001 or visit www.radialeng.com.
Sony Oxford Limiter Plug-ln
For ProTools users, ~ony Oxford has a truly reliable tool to maximize stereo mix levels right up to the legal limit. Legal meaning no overloads that CO pressing plants could reject your master CO for replication. The Oxford Limiter is a specially designed program limiter that uses logarithmic side chain processing, adaptive timing functionality

faye Specialty Snare Drum
The Taye Specialty Snare Drums are made and finished by hand and are remarkably beautiful. The Solid Maple Shell Snare drum is lathed from one piece of hand-selected North American Sugar Maple log -there are no seams, no staves, and no plies. If you opt for a wood hoop drum, you'll get 14ply Sugar Maple hoops, hand finished in 100 percent tung nut oil and articulated hoop claws with resonance control spacers to allow quick head changes. Available sizes: 14 x 7, 14 x6,14x5and 14x4inches. Expect to pay around $1,500 for this beauty. Less expensive at $800 is the 10-ply Sugar Maple Shell Snare drum made from 100 percent North American Sugar Maple. Taye uses a proprietary shell making process that combines high pressure and controlled heat, resulting in a very rigid 7.5mm thick shell. The tone is pure with a fat bottom-end and definitive attack. Wood Hoop models are also fitted with 14-ply Sugar Maple hoops and articulated hoop claws. Available sizes: 14 x 7, 14 x 6 and 14 x 5-inches. Al.so priced at $800 are the aluminum alloy snare drums cold hammered into their final form -a 14 X 6-inch drum. The shell is 6mm thick with precision machined bearing edges and fitted with 14-ply Sugar Maple hoops and articulated hoop claws with resonance control spacers. For more information, please visit www.tayedrums.com, e-mail info@tayedrums.com or phone 909-628-9589. 18

and look-ahead

techniques. I installed this wonder in my ProTools HO3 Accel rig and immediately i put it to work on my mix of a new parade theme for a chain of amusement parks. Using this plug-in is somewhat like using any compressorl limiter with Gain Reduction meter, Input Gain, Attack, Release, Variable Soft Knee controls, but you should study the manual beforehand to get the most out of it and your mix. I used it on the master stereo buss and adjusted it until I had more loudness, with minimal side effects -dulling the high frequencies and the dynamic "pumping and breathing" of the track. The Oxford Limiter is a powerful tool and has very exacting metering that can be trusted to tell when you are overloading the mix buss and the TOM buss or exceeding the maximum allowable output level for 16-bit COs. In fact, the meters indicate inter-sample overloads and then the Auto feature allows for dynamic correction of reconstruction overloads in real-time. The Enhance function allows volume and punch to be applied beyond what is available from conventional limiters without over-loading. Finally, there are full dither facilities available including variable depth noise shaping for better mastering output quality in either 24- or 16-bit final master files. The Oxford Limiter Plug-in for ProTools TOM (PTH-LIMG2) has become a mainstay -a requirement on all my mixes and sells for $530 MSRP while the LE version (PTL-LIMG2) goes for $350. More informa, tion is available at www.sonyplugins.com. ~

MUSIC CONNECTION JULY 3, 2006- JULY 16, 2006