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Due to the room resonances, the spectral components of the diffuse signal are not identical with those of the direct signals, particularly for small rooms. When mixing direct and diffuse signals, how are the readouts from the display to be interpreted?
With pink noise and a psophometric filter (»dBA«), the levels of both the direct and the diffuse components are almost identical, with an error of ±0.2dB or less.
It's all very well to say that, but getting the right data seems to be not as trivial as it looks prima facie. Can I verify that by my own with a do-it-yourself recipe?
Yes. Set the unit's parameters as follows: DryLEV ±0, 1stLEV OFF, 2ndLEV ±0, Density 0, BassGain ±0, RT60Hi LIN, RT60Lo LIN, and 2ndCUT NONE. Feed all relevant inputs simultaneously with pink noise of -20dB. First disable the diffuse signal, and measure the direct level with the psophometer. Then enable the diffuse signal again, but turn off the direct signal instead. Again with the psophometer, check the output level for every combination of room size andRT60.
Jul 2008, updated Feb 2010
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While installing new software, I experience problems with XSOFT and RS-232. How can I verify the interaction between my PC, an attached USB-to-RS-232 adapter, and the Yardstick, which is connected to the USB adapter via a null-modem cable?
When clicking the Download program's START button, a small dialog box will pop up and, at the same time, a BREAK signal will be output from the serial port. Do not confirm the dialog box yet. First check if the BREAK from the PC appears at the unit indeed. Check the voltage at pin 3 of your RS-232 port (at the PC or USB adapter end of the null-modem cable), or pin 2 (at the device end). While the BREAK is active, the voltage must be between +5 and +15 volts.
Whenever a Yardstick sees a BREAK while being turned on, it responds by sending another BREAK back to the PC. If you click the dialog box now, the PC verifies that back BREAK. As soon as a back BREAK is recognized, the PC enters the XSOFT protocol. You can verify the back BREAK at pin 3 of the device, or at pin 2 near the PC. Be aware that a back BREAK needs a functional cable link, i.e. both connectors plugged in. For connecting your multimeter, try to remove the hood from one of the connectors.
Jul 2008
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In contrast to the printed circuit board of competitors' products for 96 and 192 kHz operation, I cannot locate SRC chips (SRC=Sample Rate Converter) inside the Yardstick. To me, it seems that QUANTEC burdens the sample rate conversion job on the DSP, don't you?
No way – there's not any sample rate conversion at all. If, as it's the case for the 2492 and 2496, we explicitly allow 192 kHz operation, the entire algorithm operates at 192 kHz then, and all delay cells are quadrupled compared to 48 kHz. Or, from the other side: if a 2492 Yardstick with a 2492_QRS_22x4_SIMPLE plug-in is operated at just 48 kHz, a lackadaisical DSP has to wait 75% of its time for the next sample.
Jul 2008, updated Jan 2010
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To me it appears that its predecessors sound more mellow and with more bass than the 249x Yardsticks. Do I hear too much into things?
You are right, and I'll give you the low down.
Besides the basic RT60 setting, there are the RT60Lo and RT60Hi parameters for modifying the reverberation time gradually towards lower and higher frequencies. One should expect that both parameters modify nothing but the reverberation time, and there should be almost no audible effect when e.g. fed with stationary pink noise. This exactly is the behavior of the new 249x series.
With the older 2402 Yardstick and its DSP limitations, the result is a bit different. Besides the influence on reverberation time, there is always a frequency-dependent boost or cut in the reverberation level. If e.g. RT60Lo will be set to 2.5, the bass will be a bit louder, too, just as the bass potentiometer of a Baxandall filter network would be operated in parallel. In the same manner, the treble level will be reduced when setting RT60Hi to e.g. 0.25.
The frequency characteristic is so simple that it could be mimicked with an equalizer in the effect-send or return path of your console. In fact, it's identical to a Baxandall equalizer with 6dB/octave.
Now back to the 249x series: for high frequency components, you may reduce their levels with the good old Bandwidth filters (renamed to 2nd CUT recently). For low frequency components, there is a new parameter pair called Bass Gain and Bass Edge – available since Rev. 2.0. This allows to archive bass gains and bass edges in the presets.
Jul 2008, updated Feb 2010
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With headphones, an output correlation of 50% does a wonderful job. But when monitoring with loudspeakers, we will run into the problem that the left speaker also hits the right ear, and the right speaker also hits the left ear. Is there a way to counterbalance that?
Available since Rev. 2.0, there is a new parameter correlation, which unbalances, or detunes, the device's output matrix in order to optionally emphasize or deemphasize the lateral or the center signal. This multi-step parameter is available for each output pair separately. With an appropriately allotted emphasis for the lateral signal, crosstalk between speakers may be compensated, at least to a certain degree.
Jul 2008
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Why are there no analog inputs and outputs anymore? – The world is analog, and many recording studios still swear on analog audio technology.
AD/DA converters age at a much higher rate than outboard signal processors. This means that separate AD/DA converters typically are a more economical approach.
Digital audio inputs and outputs are an almost perfect interface: full 24 bits in and out – no noise, no distortion, and, if required, double or quadruple sample rates.
Our units are designed for an operating life of 15 to 20 years. Analog inputs and outputs, especially A/D and D/A converter chips, are changing from year to year. What today means the spearhead of technology has to be paid for with astronomical prizes – but may turn out to be outdated in just a few years. Here is an example: our 2493 builds on (currently still) high-performance and thus expensive converters. But what if a new trend will further manifest, which favors IIR instead of FIR anti-alias filters (because of the IIR's non-existent pre-echoes)? What about the 2493 with its time-tested linear-phase FIR filters – will these filters turn out to become a problem in a year or two? From our point of view, it doesn't make a lot of sense to burden an outboard unit with possibly doubtful converters, until its end of life.
Digital audio ports are much easier to keep up with converter advances. So we recommend: use your digital I/O Yardstick with external AD/DA converters, and replace them every few years with a then state-of-the-art model. Then the cost/performance ratio will be your decision, not ours...
Aug 2008
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Is it realistic to consider rebuilding an existing unit, e.g. from AES3 to Analog?
All 249x units are designed with modular subassemblies. So it may look straightforward to exchange one or more modules. The main problem lies in the then obsolete and useless front panel and rear hood, which are costly to be recycled or disposed. Conclusion: never say never...
Sep 2008
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A very important and huge advantage with the M7 is that its diffuse field can perfectly blend with prerecorded ambiences. For example, I have a live recording which has a lot of live Early Reflections, but because of too much audience, the decay is shorter than it should be. With the M7, I can match the Size very precisely to the recording's space, and I can add diffuse field only to make the reverb tail longer, not to mention that its reverb tail really lives, not static as the recorded real reverb.
When I used the 2402, this was my main problem with it. As soon as I wanted to add longer tail to a prerecorded material, the sound field started to be a little more crowded and too spatial.
Due to the somewhat restricted DSP power in 1997, the 2402 had one room size only: 100,000 cubic meters, which is extraordinary large. Such a room will never match smaller rooms like clubs.
Thirty years ago, one would have solved your problem with a plate. A plate has an incredible density, right away from the start. There are almost no early reflections. In this respect, a plate closely resembles a small reverberation chamber. And possibly resembles the M7 diffuse field generator output.
The corresponding QUANTEC approach is easy and straightforward:
Pick a 249x Yardstick room which has been set one size step smaller than the room you want to correct. All early reflections emerging from this smaller room will be masked by the pronounced live initial reflections. While the live echoes have exhausted, the Yardstick has "warmed up", which means it is dense enough and ready to step in. You may fine-tune the switchover timing with appropriate postdelay.
As the sizes of the initial-reflection real room, and the late reverb Yardstick room, are of the same magnitude, their resonances will blend nicely. Now you can add Yardstick reverb tails at will, without being bothered by much-too-late early reflections from a totally wrong room size, way too large to be melted.
Oct 2008, updated Jan 2010
Category:
All,
Tonmeister
Expert: Wolf
Question asked by: Arvisura
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I have serious problems with the user interface. The big switches and rotary wheel are ok, but to use only those is quite time consuming. The mini switches are really not very practical if you happen to have big hands, plus you need to be really on top of the unit to see what you are doing.
You seem to have overlooked Parameter Follow Me mode. When enabled, the left display follows every mini key pressure immediately, showing the current parameter name and value with larger characters. There is also a large status bar for the selected parameter. You can turn the wheel to modify the parameter.
Confirm that Parameter Follow Me (System Setup / Preset) has been turned ON. With the big switches and the rotary wheel, step down the menu Edit Scratch A until you control an arbitrary parameter's value from the rotary wheel. Instead of allocating a new parameter by going up, sideways, then down again, just hit the appropriate mini switch.
Oct 2008
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If I install a new plug-in on an already used bank, what happens to the presets of the previous installation?
It depends on the type of plug-in. If you simply install a newer version of a functionally identical plug-in, your presets will be maintained. If the parameters of the new version have been modified or expanded, all previous parameter versions will be recognized by the new software, and appropriately converted on-the-fly. New, previously non-existent parameters will be initialized to uncommitted values, which try to minimize changes of the previous presets' sonic characteristic.
If you install an older version of the same plug-in, and both preset structures are identical, your presets will be maintained. If caveats found, the older software assumes an unknown, non-related effect, and the preset area will be erased. If the over-installation has been done inadvertently, the old presets can be salvaged as long as the over-installed plug-in has not yet been manually selected, confirmed, and launched. It's only an actively running plug-in, which irrevocably erases or converts the presets.
Example: Whenever a DLY plug-in will be installed on a bank which previously hosted a QRS plug-in, all presets will be lost. Functionally different plug-ins do all structure their preset areas at their own discretion, and are blissfully ignorant of any foreign heritage.
Incidentally, the installation tools' default setting always targets at an unused bank.
Oct 2008
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How can I recognize that a plug-in will definitely not damage my presets?
Installing only newer versions of otherwise functionally identical plug-ins is a safe bet. Functionally identical means that the identification strings of both plug-ins match in every aspect but the version number. Exceptions from this rule are feasible, but will be documented explicitly.
A plug-in can be identified from the 3rd and 4th lines of the plug-in boot loader. This same text will be re-used in the filename of the installation tool. The filename always starts with the device type, followed by the full name of the plug-in, and ends with a version number. Blanks, '.', or other special characters will be replaced by '_'. The boot loader may show the following text:
The corresponding installation tool will then be called:
2498_QRS_88x1_COMPLX_20.EXE
Oct 2008
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What's the use of all that SIMPLE, MEDIUM, and COMPLX stuff, anyhow?
For any DSP algorithm, extensive feature lists or higher sample rates can be traded against each other. The QRS algorithm's Yardstick 1.x versions have been designed under the assumption to be fully operational even at 192kHz. On the other hand this means that, at 48kHz, the DSP is active only one quarter, and otherwise waiting in idle the rest of the time.
For the user, such an approach is straightforward: any sample rate from 38 to 216kHz is acceptable. The unit digests everything, and the operating surface remains consistent.
For an ambitious algorithm designer, such an approach may be frustrating. How much more can be realized with the otherwise uselessly wasted double or quadruple DSP power. Before continuing, the designer must come to a momentous decision: If I double complexity, my algorithm cannot be operated at 192kHz anymore; but at least will reach 96kHz. If I need four times the DSP power, 48kHz is the end of the line. Losing a few customers when resorting from 192kHz to 96kHz may not really hurt your company, but losing some 10 to 20% when sacrificing 96kHz will do. Moreover, such a move will alienate all the rest.
On the other hand, the entire customer base will be preserved if various complexities of an algorithm will be offered. Sometimes it doesn't make sense to stuff all bells and whistles into one plug-in. Quite often it may be realistic to trade-in some of the less important features for a result, which can be operated at double rate, too. Many a high-end customer may even favorite a feature-reduced version. Motto: why should I need Baxandall filters for my $3000 preamp, anyway...
Exactly this is the idea of SIMPLE, MEDIUM, and COMPLX. For a SIMPLE plug-in, all unnecessary ballast will be avoided; this is sort of a minimum version, but fully operational up to 216kHz. The COMPLX plug-in will be stuffed with all bells and whistles, even if just of occasional use; it's the flagship. And, right in between, a MEDIUM plug-in may be positioned, which provides some potential to play with, but compromises only marginally when it comes to sonic quality.
Quite often, the following relations can be assumed: COMPLX:48kHz, MEDIUM:48..96kHz, SIMPLE:48..192kHz; but don't take that for granted. To make it clear, all plug-ins have their maximum sample rate labeled as x1, x2, and x4.
The most universal and most commonplace plug-in is definitely the SIMPLE version. If the required design target can be achieved with it, any future plug-in should be limited to this singular version. This reduces maintenance efforts for the software staff, and minimizes gray zones and learning curves for the end user.
Oct 2008
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Why do QUANTEC Room Simulations sound more impressive and more realistic than real rooms?
First a formal objection on the part of the Reverberation Czar! - It goes without saying that natural rooms still sound better than our simulation models, first and foremost because of their outstanding spatiality. However, this only holds true as long as the listener is physically positioned inside the room, and thus gets flooded by the omnidirectional sound waves.
As soon as one tries to capture the room impression with microphones, all 3D magic will vanish.
Every sound transducer the acoustic event has to come about on its way to our ears is a bottleneck, which strips off a certain amount of the original room response's spatiality and liveliness. This is by no means a devaluation of our state-of-the-art microphones. But fact is that the microphone bundles all the omnidirectionally-arriving signals on its diaphragm, and then tunnels the precious payload through an electrical mono channel. Even if you would totally shut-off one ear while at the church, that does, on no account, sound monaural.
As the omnidirectional output of our room acoustic models is fed straight into the audio signal path, instead of being tunneled through a microphone's diaphragm, our models clearly sound more live than a natural room captured with microphones. What is still unavoidable, even for us, is the bundling through the diaphragm of the playback transducer, currently the final acoustic difference remaining between a QUANTEC Room Simulation model and a person's physical presence inside a real room.
From the number of sound transducers involved, three quality classes may be derived:
- Class 0 – no sound transducer:
The listener is physically positioned inside a real room; flooded by the omnidirectional sound waves.
- Class 1 – one sound transducer:
The listener harks attentively to the QUANTEC Room Simulation model through loudspeakers or headphones.
- Class 2 – two sound transducers:
The listener follows a recording where room acoustics have been captured by microphones. Only those components of the natural room acoustics not yet stripped-off by the multiplication of recording and playback transducers will remain.
While QUANTEC Room Simulation with its parameterized models belongs to class 1, the nowadays extremely popular convolution-type reverberation is clearly at class 2 candidate. Reason is that the convolution chain still comprises of two sound transducers: first the microphones to capture the room's “fingerprint”, then later the playback transducers to monitor the convolution result.
Strange: if the room fingerprints would be deducted (i.e. “stolen”) from our Room Simulation models, a convolution-type reverberator would ascend to class 1. Still unachievable for any size of fingerprint library would be our parameterized models' unrivaled ease of ad-hoc characterization.
Nov 2008
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Out of curiosity, I've installed an older 1.x software on a unit already converted to 2.x. Now all my 2.x plug-ins have vanished, including my valuable preset collection. Too bad...
Keep cool – that's what has happened: The Yardstick's bank 0 of a 2.x installation is being used by the SWITCH boot loader, while in version 1.x, the (one and only) effect software has been settled there. If, after an existing 2.x installation, a 1.x software is being installed, it will overwrite the 2.x boot loader. Because of the missing boot loader, the „bridge“ to the 2.x plug-ins and their preset libraries has gone. But they still exist, safely captured further down the Flash memory. It's just the „access“ which needs to be fixed.
How can the access be re-enabled?
Download the current SWITCH code from our website, and re-install it to the boot sector again. Additional benefit: the 1.x software, installed by you out of curiosity, will not be overwritten initially, but will be offered by SWITCH to be relocated to a bank further down the Flash memory. Before copying, SWITCH will compile a list of all banks, regardless of occupied or free. Now it's up to you to enter a preferred target bank, wait a few minutes while the installation is being relocated, and then proceed with installing SWITCH until finished. With the next reboot, not only all your 2.x plug-ins are being advertised again, without any change. As an additional bonus, your stray 1.x installation will be offered for booting, too.
Dec 2008
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When doing some rework on older sessions, I sometimes wish to access earlier versions like 1.6, 1.7, or 1.8 - including the original presets of that time. Would it be possible to smuggle various 1.x versions into the unit, by repeatedly overwriting the SWITCH boot loader. Then relocate each one to another bank, in order to have all of them at my disposal when reopening a session?
That's okay.
Dec 2008
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I've just finished overwriting my former 2.1 installation with version 2.3. Now I found that only parts of my preset collection can be used. Amidst my presets there are lots of voids yawning - I can read EMPTY all over the place. Looks like I've lost parts of my presets due to compatibility problems...
Nope. Those “knocked out teeth” are all those former factory presets, which have been haphazardly interspersed with the Local user preset archive up to version 2.1 – just like a rope of pearls. All older factory presets, and many new ones, have been relocated to library directories like Dialog-Lib and Music-Lib. All users may refill those residual EMPTY voids with their own presets now.
Dec 2008
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Can I access older user presets located in a foreign bank?
Yes – that's generally possible. Older presets inherited from previous software versions will be converted to the new preset format on the fly. If the new preset format contains parameters not yet available in the older version, similar parameters will be converted (if possible). If all else fails, new parameters will be set to their default values. The idea is to make the updated preset sound as similar as possible to the one from the older software. In spite of all automatism, we strongly recommend to check the updated presets thoroughly.
Jan 2009
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Can I access factory presets located in a foreign bank?
Yes, but not before they have been copied to the foreign Local archive.
Jan 2009
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Bullshit! - My Yardstick does not fit between the rails of my rack. Aren't those QUANTEC people in a position to adhere to the 19” specs?
According to IEC60297, the clearance between the 19” rack rails is required to be 450 mm minimum. Our Yardsticks external width is 448.2 mm maximum – so where's the problem?
19" Specs.
Mar 2009
Category:
All,
Maintenance
Expert: Wolf
Question asked by: Stagetec ;-)
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Is erasing of a bank mandatory before installing a new plug-in?
No. But if a bank has not explicitly been cleared, the previous Setup and all local User Presets remain in that bank. When updating to a newer version of the same plug-in, it often makes sense not to erase that bank. Launched for the first time, a newer plug-in tries to adopt the remaining data, then start to build on that heritage. Before, for whatever reason, installing an older version, a bank should always be cleared in advance!
Aug 2008
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What about controlling my Yardstick from multiple positions?
It's possible to operate a Yardstick from both front panel and web browser window in parallel. It may haptically be useful to select a parameter from the browser, but modify it by turning the front panel wheel. Or, to select a Preset archive from the browser, but stepping through the Presets from the wheel.
The only critical function is the bargraph display. Bargraphs only work on front panel and one browser concurrently. Opening multiple browser windows to control the same Yardstick from different PCs would, at the current stage of development, disarrange the bargraphs.
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For jump cuts in movie post production, I regularly need a frame-accurate switchover of the room acoustics. With the Yardstick I run into two problems:
1. While turning Preset selection, it takes me 2-3 seconds until the Preset has been loaded and will be audible ("Selecting, Initializing...").
2. If old and new Presets are located near opposite edges of the library, while still underway to the target, I'd risk some Presets being loaded temporarily.
Is there a better practice?
The simplest way would be loading old and new room situation to Scratch A and B. When the jump cut occurs, simply hit the A/B button.
How long would that take?
When switching between small rooms, there's only the gap for the intentional fading down and up again. The larger the room size, the longer the transition will take, which, by surprise, will still sound organically. To name the numbers: 1E0=>5ms, 1E1=>15ms, 1E2=>40ms up to 1E6=>almost 1sec (always take the room size you are switching to).
And what about repeated jump cuts in rapid succession?
Enter Setup and look for the menu item Preset Load (Operations Guide 3.1 - p. 100), then select Enter. While in Enter mode, a Preset will no longer automatically load as soon as the wheel slows to a standstill, but will wait for a confirmation from the ENTER key. This allows for switchover at the push of a button, with the abovementioned initialization delays. A proven trick is to copy the required room sequence to adjacent Preset slots in the user area, and then retrieve them sequentially.
Dec 2009
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Competitive products always provide plenty of different algorithms: Room, Hall, Plate, Space - exactly what I need in daily routine. As far as I understand, a Yardstick has a single algorithm only, which seems a bit narrow-gauged to me, isn't it?
No way. You're right that a Yardstick does not provide more than that single QRS algorithm, but that's a powerful, all-inclusive solution. Its universalism is hidden within its parameters. Just by well-directed tuning of the various parameters, you can end up inside a bread bin, or a giant cathedral with multiple side naves and transepts. Other than competitive devices, which recommend an optimized reverb for a specific type of sound (e.g. Voice, Guitar, Drums), the QRS algorithm is being defined by the room the musicians are playing in, independent of the construction of their instruments.
If I'd squeeze the entire mix through reverb, or a room for your sake, I'd totally loose transparency and comprehensibility.
That's true for reverb, but not for QUANTEC Room Simulation. The QRS algorithm applies diffuse components individually for each instrument, according to its momentary style of playing - fully automatically, without any intervention. Just the same as with a real room that never needs to be tailored to a specific instrument. The trick lies simply in an exact simulation of the physical laws effective inside a room. In particular it's the attack behavior, i.e. the gradual effervescence of the room. The longer a tone will be maintained, the further it will climb up the attack slope, and the louder it will tilt over into the reverb tail. With this "launch pad" trick, long tones lead to strong, short tones to gentle reverberation - all of its own volition. All-important: each one individually, i.e. without any interdependency of the envelopes. (»Hallelujah diagrams« - see links below).
Let's construct an extreme example: if we concurrently feed a room with a flute and a spoken voice, the low-bandwidth, sostenuto flute tones will play around the room much louder, haunting and longer observable, while the short, wide-bandwidth transients of the spoken consonants just tip the room. To clarify the consequence of this approach even further, let's simply exchange the roles for a moment. Sixteenth-note sequences from the flute just tip the room, while a sustained "Aum" will considerably pump up the room response. This is one of the secrets why transparency will be so strikingly maintained with the QUANTEC approach.
To get back to your question: it's the QRS algorithm's ability to do all things right with just a single parameter setting. With multiple dedicated algorithms, you'd need multiple reverb units, one for each instrument class. More than once it will take you hours of fiddling about with the various units, until all the individual components will end up coordinated in an acceptable manner.
Another QUANTEC premium: all the instruments are always playing together within the same room, and don't live their parallel lives in different rooms.
Dec 2009
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Competitive products always provide dozens of adjustable Initial Reflections; but only two with the Yardstick. Is QUANTEC running out of reflections?
Less is often more, especially in this situation. QUANTEC has recognized very early, that, within a real room, each initial reflection is being emitted from the same structure that also generates the reverberation tail. In its simplest case, a room consists of six boundaries; left, right, front, rear, top, and bottom. If we'd allow a sound engineer to add arbitrary early reflections, which, because of the freedom of choice, can no longer be consistently co-created by the late-reflection structures, would do more harm than good to a coherent sonic image. Physics simply do not allow separately created echo patterns to seamlessly merge with the subsequent room response - disruptions and side actions are inevitable. In this context, we were told of a singer who once put this phenomenon into words. »With this device, I had the impression for the first time that there was no more annoyances between my voice and the room.«
What exactly, from the QUANTEC viewpoint, does the established competition do wrong?
To preserve complexity, i.e. to minimize DSP power, most competitors split room response into two partial algorithms
- A multi-tap delay line with a fistful of taps, which, independent of the contents it is fed with, repetitively reiterates the same distinctive early reflection pattern. This approach follows the idea of a raytracing model, where a few initial echoes (»early reflections«) are derived from the geometric circumstances of that room. The temporal and panoramic distribution of the delay line echoes is told to characterize the desired room.
- Separated from the direct signal by suitable Predelay, follows a continuum with as much density as possible. Allegedly, any room behavior according to size, individual reflections, and room resonances can be completely ignored. What irritates us most: the one and only design target for this continuum is to increase its density beyond limits. This does mean nothing else than, as far as DSP resources can deliver, to end in musically dead "White Noise" - for any type and any size of room.
QUANTEC's conclusion: an evermore reiterated pattern is an annoyance, sooner or later. A similar effect is known from cheap loudspeakers, which continuously repeat their impulse response with each and every transient coming in. Second: an "acoustic harassing fire" in contradiction to the room would completely destroy the interaction between tone duration and reverberation level, which is so stunning with the QRS algorithm. Net result: the QRS algorithm would still supply clean reverberation; but its legendary transparency will be lost and gone.
But why are there still two adjustable initial reflections, after all?
What has been outlined above is basically required for musically inspiring rooms. Movie and radio drama dialogs are a different story. What's needed there over and over again are standard acoustics, painted with a pretty wide brush. Most situations are "nonmusical" rooms with pronounced boundary surfaces in the near field, e.g. a hallway, a car, or a staircase. For such near field simulations, the reverberation tail is generally second-rank, and the emphasis of a simulation is shifting towards early reflections, indeed. There, and only there, one single, representative early reflection for left and right each does make sense.
Dec 2009
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Compared to the competition, QUANTEC is showing off with a highly idiosyncratic approach for yonks. Such an esoterically prancing idea looks pretty untrustworthy to me, as it suggests that QUANTEC follows other physical laws than the rest of the universe. Seems like pure marketing wishy-washy, isn't it?
It goes without saying that QUANTEC is bound to operate within the limits of physical laws, as anybody else. Major difference and unique selling proposition is our way to approach common room-acoustic phenomena via mechanisms and models defined by the air fill of the room. In other words: we focus on the transportation medium of the sonic energy, and include those frequency-selective resonators within a room. This is contrary to the competition, which follows delay line patterns deducted by a raytracing perspective. Translated to a wind instrument, the competition concentrates on the wooden or brass instrument body, while we turn our attention to the vibrating air column within the instrument.
Still frivolous. As resonance behavior in the frequency domain may be represented by impulse response in the time domain, and vice versa, simply by Fourier transforming one into the other. Plain-talking: both approaches lead to absolutely identical results; they are two faces of the same coin.
Your statement is valid for unlimited resources only, which means abundant DSP power. Whenever one is bound to economize DSP power, because one has a meager 1% compared to that of a real room, one should thoroughly evaluate the important details to devote the scarce resources to, and what uncalled-for knick-knack would go unheard anyway. Point is rather: what can be omitted without harm, and which one of the contrary approaches may lead to music-esthetically pleasing results faster and easier.
Well, based on a resonance approach, QUANTEC claim that they're in a position to extract more useful results from limited resources than the competition. Which means they fish out those better 1%, simulate only that, then spoof the auditory system music-esthetically more pleasing than the competition.
Exactly. Approaching room acoustics from the frequency domain, instead of geometry, shifts the importance of various physical phenomena essential for the simulation of a certain target room acoustic behavior. Main target for QUANTEC are musically ambitious room acoustics. More than 30 years ago, QUANTEC was the first to recognize that the silver bullet for simulation of musical rooms lies in modeling the engaging and subsiding of resonators. Those who index a room's boundary surfaces, then raytrace geometry, then deduct tapped delay lines from that data, will inevitably dissipate their energies and DSP power on musically meaningless sidelines.
Dec 2009
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My reverb unit offers 4 algorithms - a Yardstick just one. How come?
My Chinese guitar from the DIY store has 4 pickups - an Ovation just one. How come?
Dec 2009
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With regard to price: why are those Yardsticks considerably more expensive than the competition?
Just have a quick listen - and you'll put this question aside.
Dec 2009
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Rating: 4.0 of 5. 1 vote(s).
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Most competitive units feature a Modulation parameter. Has QUANTEC diligently missed the need for modulation?
Yes, as modulation is of no significance for Room Simulation.
Without random modulation, all reverberation tails would be equal.
True for reverberation, but totally ineffectual for Room Simulation. Here's our competitors' problem: in front of the actual reverberation tail is a cluster of so called initial reflections, which are told to characterize a room according to a raytracing concept. Such a cluster has one major problem - with each and every transient fed into the system, the cluster reiterates the same response over and over again. This kind of "fricative", which sticks out of any plosive like p, t, k, will be fatiguing, and will turn out to become boaring or annoying before long. Quite similar to a cheap loudspeaker, which cannot cope with fast transients, and thus will replay over and over again its critical onset pattern: kprwt, kprwt, kprwt.
Our competitors circumvent that problem by randomly moving the initial reflection delay line taps slightly. The goal is that the fricative will mutate quickly, thus prevent the hearing from associating subsequent fricatives with preceding fricatives. Basically, the problem seems to be solved, but unluckily this trick has serious adverse effects due to the inevitable Doppler effect. Moving the taps would result in frequency fluctuations, which detune the incoming signal. The reverb commemorates Chorus effect - it whines, and is no longer "piano proof".
But such a chorus reverb may sound ridiculously beautiful, e.g. with a guitar solo.
As a creative means of design, chorus reverb certainly has its value. But in a real room, frequency modulation won't happen, so there is no such parameter in the context of Room Simulation.
Just a minute! - There's a paper by David Griesinger (»Lexicon«) where he demonstrates that, due to airflow in large rooms, slight frequency deviations can definitely be detected.
Extremely thrilling reading for HVAC engineers. But where's the music-esthetical benefit? - Show me one conductor or choirmaster, who pilgrimages to the caretaker to ask him to crank up heating or ventilation - in order to make the room sound more pleasing.
It's sort of weird, isn't it?
And what's QUANTEC's trick to suppress the undesired regularities within the algorithm?
No trick at all. With Room Simulation, this problem simply doesn't occur, as we don't add separate initial reflections. As documented elsewhere, loudness and character of each reverberation tail depends tone by tone on the duration of the stimulus. This will be mixed with the previous history, i.e. longer tones played recently, whose tails remain louder in the mix. Based on individual tone durations, it's quite possible that originally staggered tones will decay with synchronous tails (»Hallelujah diagrams« - see links below).
The individual composition of each reverb tail according to tone duration and previous history leads to continuously changing interferences between one another. Such interference results in frequency-dependent accentuation and cancelation, which is beyond association for human hearing, thus sounds very lifelike and musically pleasing. But without the cumbersome Doppler effect now, which, if noticed too late, could lead to aggravating surprise during subsequent processing steps.
Dec 2009
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How do I use a Yardstick in the context of a 5.1 or 7.1 production? - Part 1 - Original Recording
Similar to a mixing console, a Yardstick doesn't process encoded material in the sense of x.1. It builds upon the unencoded raw version of the audible material. The effects send path operates out of the console as usual; the effects return will, together with all the direct signals, be encoded to x.1 in a subsequent processing step.
There's some danger that through the effects send bundle, certain instruments may enter the device twice, with arbitrary phasing. Example: if the violin microphones overhear the cello section playing behind the violins, the cello sound would come in over two effects send channels. If I'd mix both strings channels to feed the effects send path, I would risk unacceptable comb filter effects on the cello reverberation.
Refrain from mixing in advance; benefit from using all 8 effect send paths. Fortunately, the 2498 Yardstick offers an unparalleled property: all 8 paths into the simulated room are completely phase-insensitive. In other words: if one would exchange polarity on one of the inputs, there would be no change in sound at all.
Conclusion: those cancellations you're afraid of would need some care only if the "small surround" 2496 Yardstick with its 2 inputs would be used. If you'd use the "large surround" 2498 Yardstick with its 8 input channels instead, none of such cancellations would develop at all.
Note: in a surround production, if one would move along a secondary stereo downmix version either, all cancellation problems should have been solved there already - independent of a Yardstick. When working on a 5.1 surround production, such a controlled free-rider downmix is an ideal effects send for the "small surround" 2496 Yardstick with its 2 inputs and 6 outputs.
Dec 2009
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How do I use a Yardstick in the context of a 5.1 or 7.1 production? - Part 2 - Post Production
Completed x.1 productions, which should be re-opened and augmented with Room Simulation, need to be unassembled (»decoded«) before they can be fed into a simulated room. Fortunately, the 2498 Yardstick offers an unparalleled property: all 8 paths into the simulated room are completely phase-insensitive. In other words: if one would exchange polarity on one of the inputs, there would be no change in sound at all.
To exploit this feature, assign each single channel of your 5.1 or 7.1 production to one of the 8 virtual driving speakers. Not just the 5 or 7 satellite channels, but all 6 or 8. Regardless of the incoming phasing on the 6 or 8 channels - especially picky would be the crossover frequency between subwoofer and satellites - nothing like cancellations would develop at all.
Dec 2009
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How do I use a Yardstick in the context of a 5.1 or 7.1 production? - Part 3 - Stereo-to-Surround
Here we take advantage of another inherent convenience of the QRS algorithm: an added simulated room will routinely blend with an already recorded room or reverberation. Contrary to competitors' reverb units, such a double reverberation will not degrade transparency at all, or marginally only.
For this type of projects, both 2496 and 2498 Yardsticks are applicable likewise. To integrate otherwise idle DSP resources of the 2498's #3 to #8 input channels, please use it in the 2->8 mode, optimized for this purpose (instead of 8->8).
Dec 2009
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In web forums one can routinely read that 'the QUANTEC' is an outstanding primary reverb. Actually, what's a primary reverb?
Many music productions deploy different reverb units and reverb plug-ins, tweaked to specific instrument classes in an optimum way. This requirement for optimum tweaking is the result from a limited range of artifact-free operation of low to medium-cost reverb algorithms, which may be useful for vocals only, or solo instruments only, or percussion only, etc. Argumentum e contrario, one has to be prepared for inacceptable side effects for every wrong instrument class. Because of this persistent "requirement for chasing the instruments", automation is indispensable here.
With this strategy, the musicians are playing in separate rooms. It's the job of the primary reverb, a device with outstanding quality, to re-collect them and put them back into a single room. Problem is that a primary reverb generally compromises transparency of the cumulative sound image, and thus will often be used reluctantly.
A QUANTEC Room Simulator meets highest standards at this critical position. Most of its Music Library presets flawlessly process all instrument classes simultaneously, i.e. the entire mix. Side effects such as coloration, clatter, intransparency, or mushy sound are almost nonexistent with a QUANTEC-based primary "reverb".
Another peculiarity of primary reverberation is that it is an always-on effect, which means that its settings are modified infrequently. To put it into perspective: automation is seldom required; similar to monitoring speakers. This may explain why, despite unavailable automation, even the very first 1982 Room Simulator generation is still in daily operation worldwide.
Jan 2010
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2496 or 2498 have more outputs than I need - what to do with surplus outputs?
Simply ignore them and leave unconnected. Please avoid mixing surplus outputs to used ones. Due to the 50% correlation output matrix, any cross connection would cancel out 50% of the reflections, which would result in both thinned out and 3dB lower level reverberation. On the other hand, this behavior may come in handy when mixing L and R channels, as the resulting mono signal has lower diffuse energy.
Jan 2010
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2498 has more inputs than I need - what to do with surplus inputs?
Contrary to outputs, you should drive all input pairs in parallel from your less than 8-channel input signal. Each input channel pair contributes 25% towards total number of a room's normal modes and reflections. If one or more inputs remain idle, the resulting reverberation will by unnecessarily thin and a bit lower in volume.
Moreover: each input pair of a QRS COMPLEX plug-in behaves like a QRS SIMPLE plug-in. So only a close concert of all inputs will end up in that wonderful COMPLEX sound
When driven from one stereo pair only, it's recommended to feed the 2498 through its XLR input. Such operation will not only forward the input signal pair to all 8 virtual speakers, but compensates for any level fluctuations, too.
Jan 2010
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What exactly is the low-frequency edge for Room Simulation?
It should be noted that a room can only produce reverberation on frequencies (»spectral lines«) where there are room resonances (»modes«) available. For any gaps in between, there's never any reverberation. The low-frequency edge is determined by the lowest 'standing wave' that can manifest itself within a given room. This depends very much on the size of a room. For large rooms, the edge is a few Hz, while for boxes and cabinets, room resonances can manifest only above a few hundred Hz. For DC, which would correspond to a constant air pressure, all QUANTEC Room Simulations deliver a zero, which means that DC components are being suppressed.
For all those users who'd like to have a hell of a party with their supernova, there is an additional goodie hidden in the Effect Setup menu. With the Subsonic switch, the DC zero will be converted to a pole. In reality, this would mean that a given air pressure in a room would continuously increase over the course of a few minutes, until the room would burst and collapse. In simulation, a DC offset would increase and increase until full level is crossed, and output overflow occurs. In other words: the Subsonic setting would annul the law of energy conservation. Useful for such kind of effects is the AES/EBU port's ability to transfer unlimited DC, which is generally not the case for analog AD/DA.
The chief attraction is that 0Hz has changed to a pole (»resonance«) now instead of a zero (»cancellation«), which means that there is no lower frequency edge any more. Even below 1Hz there is still a lot of cannonade and boom, which resembles no less than a respectable earthquake. Made-to-measure for those addicts of D-BOX motion code and Action Seats.
Jan 2010
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More than once one can read an indication that for the 2498, the phasing of the 8 input channels is told to be irrelevant. Already difficult to understand in theory, I cannot imagine such a pipe dream to work in practice?
Acoustic quantum mechanics.
Come again?
Heisenberg uncertainty principle.
Weren't we just in audio engineering yet?
For simplicity I'll call it "QUANTEC physics". I'm talking about phenomena to become relevant whenever the frequency resolution of an acoustic phenomenon will become more filigree than the frequency resolution of the sense of hearing. Room simulation is such a situation, especially for large rooms, where the distances of room modes ("room resonances", "spectral lines", "standing waves") are often closer than 0.1Hz to one another. Considering that, from a narrow band viewpoint, both phase and amplitude behavior radically change from line to line (40dB are not uncommon), one can only speak about wild, completely irregular random fluctuations. To get hold of meaningful figures at all, a statistical physical definition of the room is an option. Simply use third octave band noise instead of sine waves, and your frequency response will be flat and clean.
Unlike a crystal-stabilized sine wave oscillator, the bell-shaped spectral width of a human voice or a musical instrument will stimulate an entire bundle of adjacent room modes simultaneously. The exact hits will change permanently along the way, as the bowed or blown instrument's pitch will fluctuate continuously.
Short digression: imagine you have two uncorrelated noise generators identical in construction. If you mix both outputs, the output noise signal increases by 3dB. What do you think you hear if you'd activate the 180° key on generator A or B (or both)? - No change at all - all variants sound absolutely the same!
Exactly herein lies the trick for multiple phase-insensitive effect send inputs. Instead of mixing the 8 signals additively before feeding them into the reverberation chamber, they're being forwarded disjoint. Not before the room itself, the various input signals stimulate their individual and not exactly predictable room modes. Which signal eventually hits which room mode at which location remains unknown, and can only be quantified statistically. And once again: unaffected by imaginary (180° or whatever) keys at the inputs.
Jan 2010
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All Yardstick I/Os are AES/EBU. Does that mean that I cannot operate a Yardstick in my S/PDIF studio?
Not at all.
S/PDIF Channel Status bits are being recognized properly. As most sound cards deliver wrong sampling frequency indication beyond x1, there's an optional fallback mode based on an internal sampling frequency counter.
For adapter cables and plugs, there's an S/PDIF section in the manual with exact schematics.
Jan 2010
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Contrary to a much hyped competitive unit, my choir stands amidst the room here, not in front of it. Why does QUANTEC perform better than the other guys?
Business secret!
Buy our no-hype product and enjoy it.
Jan 2010
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What's the stunning transparency of QUANTEC Room Simulation based upon?
Psychoacoustics - the cocktail party effect in this case.
Whom?
The cocktail party effect, discovered in 1953, describes the ability of the human auditory system to focus its listening attention on a single talker, among a mixture of conversations and background noises, ignoring other conversations.
How is this related to the acoustic transparency of a room?
Within such selective perception, the hearing reaches noise suppression from 9 to 15 dB, i.e., the acoustic source, on which humans concentrate, subjectively seems to be many times louder than ambient noise of the same level. This even holds, if the ambient noise is louder than the information by a similar amount (»negative S/N«). On the other hand, a microphone recording would capture mainly background noise, as any selective perception does not apply.
Within a room, a side effect of the cocktail party effect would result in a greatly reduced perception of spatiality: the information sounds dry and with minimum reverberation. A microphone positioned at the same place would capture blurred, slushy, and over-reverberating information instead.
Once again, QUANTEC seems to be ahead of the pack...
That's by no means our merit! - It was surprise to us, too, that even within the scope of the cocktail party effect, perfect Room Simulation behaves just like a real room, and provides the human hearing with the necessary clues. If those clues are unavailable, as it's the case with our competitors' reverberation devices, the human auditory system reacts just like the microphone: the sound is unintelligible, slushy, and with far too much reverberation.
Feb 2010
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Whenever I try to adjust a parameter from the mouse wheel, the slider jumps right to the top or bottom stop.
Open your mouse driver configuration. Within various mouse wheel properties, you'll find something like advance one screen page per mouse wheel step. This option needs to be disabled.
Feb 2010
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Whenever I'm recording the impulse response of any QUANTEC Room Simulation, for putting it through my convolution plug-in, its spatiality collapses altogether. To name a number, the 60 to 150 feet depth of a sacred building has completely vanished into thin air. No idea of what's going wrong. Maybe some copy protection QUANTEC has hidden intelligently?
It's neither you doing something wrong, nor there's a copy protection. It simply doesn't work.
QUANTEC really cannot make a claim for themselves, that proven mathematical and physical methods like convolution and Fourier transforms ironically fail, as soon as they're applied to their Room Simulation algorithm.
Nobody did insist on that. It goes without saying that convolution works; you do hear flawless reverberation, don't you? It's just it's spacial depth that you're missing; which has gone flat somewhere in the course of your manipulation.
Flawless indeed. But, where exactly, the spatiality has fallen by the wayside?
Right from the start - when feeding the unit.
In plain language: first I feed the left input with a click, and record the impulse response on both outputs. Then click on the right, and again record both outputs. What's wrong with it?
For now, you've recorded no more than two, let's call them "labyrinths": one for 100% left, and one for 100% right. Now hurry to proceed with sampling center, slight left, slight right, and all the rest of it.
Just wait! - I've two labyrinths - one for left, and one for right. If I'd feed the unit with a center signal, i.e. mono into both labyrinths at once, the output always delivers the sum of both labyrinths. Generally, this should hold for any panpot settings, right?
In the context of Room Simulation, we don't deal with two delayline-based labyrinths, but with hundreds of thousands of resonators distributed throughout the room. From the perfect coordination of all those resonators - and only from there - results that stunning transparency.
Feeding the unit with a sine wave test signal and very fine frequency steps (<<1Hz) would stimulate a non-specified subset of those resonators, to a more or less extent. Depending on the phase and amplitude conditions at the two inputs, quite a few resonators may not even respond at all. Moreover, one cannot estimate if a specific resonator will respond to the left, right, or only to a specific phase or amplitude relationship between the two input channels. In other words, as jagged and bumpy as the amplitude and phase behavior of a single labyrinth, is crosstalk between the two labyrinths. The operative point here is that those hundreds of thousands resonators jump wildly with even the slightest frequency drift, while your two labyrinths stubbornly deliver their vector sum, regardless of the frequency - definitely a bit of a yawn.
I must admit that my approach would indeed force a majority of those resonators into lockstep. This may paralyze time-of-arrival stereophony, but what puzzles me is that intensity stereophony does collapse likewise. Did I overlook another important detail?
Due to the complex crosstalks within a room, one may realistically imagine one resonator at one specific room position, which would respond to either left or right channel, but not both. With mono, genuine Room Simulation may deliver a resonance gap here, while your convolution clone may still deliver the sum, e.g. a peak - as it does with any other resonator. Be aware that you haven't captured those singularities while taking the fingerprints. Moreover, just 1 Hz higher, both approaches might match again, and 2 Hz higher, some completely unexpected behavior could occur.
In short: while having sampled the two room fingerprints, you've completely disregarded the "crosstalk domain".
Finally, it looks to me that there's no feasible way to counterfeit the QUANTEC Room Simulation algorithm by means of a convolution plug-in?
Sure enough, there is one dedicated configuration where a convolution clone would be 100% identical.
Would you mind to tell me?
The idea is to just lever out the uncapturable "Crosstalk Domain". Take care of feeding your Yardstick always with a mono signal. Just feed both left and right input with the click, record both outputs' IRs, and then off to convolution!
With this trick, both the Yardstick and its convoluted IR clone do really sound exactly the same?
Absolutely - both are as flat as pancakes now.
Feb 2010
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