rme totalmix - features_operation

FireWire Audio by RME - Technical Background

»Technical Information Index

Overview on current FireWire audio solutions

RME is the only manufacturer not to use a third party FireWire audio technology. All other manufacturers use dedicatedall-in-one FireWire audio chips. As a result, their access to bugfixes and improvements is more than limited. Here is a smalloverview of currently used chips:

BridgeCo: used by M-Audio, Terratec, ESI, Edirol, Presonus, Apogee, Tascam.

Philips AV LLC: used by Motu and Hercules.

Upcoming: a new chip from Texas Instruments, co-developed or supported by Echo. Will be used in upcoming Echo andMackie FireWire devices.

Upcoming: a new chip from TC, called Dice II. A few years ago TC have bought a company that produced non-audioFireWire chips. Now this chip, announced to be available more than a year ago, is said to finally arrive. If so, it will be usedin upcoming TC FireWire audio interfaces for sure.

Upcoming: a chip from Oxford Semiconductors called OXFW970. Not as feature-packed as the other chips, but expectedto be the cheapest one of all. It might cause another flood of very cheap FW audio interfaces in 2005.

Differences between RME and all others

The picture shows the difference between our solution and all others. A dedicated FireWire audio chip always includeseverything except for the physical interface (the upcoming TI chip will even include the PHY). The FireWire audio controlleritself, including a special RISC processor, and the so called Link Layer Controller are all combined in one chip. The RISCprocessor is also (mis-) used to provide some kind of audio DSP functionality, but is very limited in doing so. An alternativeis to use an externally connected DSP, which then allows for all kinds of effects and mix processing. The maindisadvantage of an external DSP is the limited transfer rate (bandwidth) between DSP and FW core, which limits thenumbers of processed channels significantly. For example, the matrix mixer of the HDSP MADI, calculating 8192 volumelevels simultaneously, is only possible because the calculation takes place directly within the FPGA, as it can use thefastest RAM (internal cells), and thus uses an otherwise not to be reached memory bandwidth.

RME have developed theFireWire audio core completelyon their own. Additionally, as thissolution is within a FPGA, it canbe changed, improved or bugfixedat any time. And our DSPfunctions, like TotalMix and levelmetering, are performed by anextra, dedicated audio DSP,which also resides within theFPGA.

The Link Layer Controller andPhysical chip that we use areboth from Texas Instruments.That should guarantee amaximum compatibility, as TI'schipsets are widely spread andused, even in Apple computers.The LLC is connected to theFPGA, giving us unlimited controlon the available functions of bothchips. Otherwise theconfiguration of the Link Layercontroller can usually only bechanged by the chipmanufacturer, by providing a newfirmware.

Tech Info FireWire Audio by RME - Technical Background http://www.rme-audio.com/english/techinfo/fwaudio_rme.htm

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Typical Pro audio features

As you might imagine, the custom FireWire chips have been designed by developers who have little knowledge of thesuperior features and functions of RME products. So these chips miss a lot that can not be added later, even not by afirmware update. Let's have a look at some typical Pro Audio features:

Lock to external signals in realtime

Support for varipitch/varispeed

Offset-free record/playback

Support for multiple units

Support for sample rates up to 192 kHz

Change of the buffer size in real-time

RME unique FireWire features

RME developed their FireWire audio controller with all the features in mind that RME PCI interface cards have. TheFireface simply should work as if it were a PCI card. And we were very successful with this approach. The following is a listof unique Fireface features that are not found in current FireWire interfaces from other manufactures:

Complete realtime sample rate lock

Complete realtime sample rate control, evenduring playback/record

Complete start/stop control. No changedoffset after reboot.

Extreme varispeed / external sync support,in all modes (DS / QS).

Reduced protocol overhead, by not sendingCIP (Common Isochronous Packet format)headers (AVC, mLan)

Working FireWire 800 solution

Working FW800 fix for Windows XP SP2

Latency change on the fly, while ASIO andGSIF are running

Hardware-based data packet check anddrop correction

Data Packet Check and Drop Correction

An extraordinary feature that deserves a detailed explanation. There are two reasons for clicks and drop outs:

Not enough time left for the audio engine to process the data. This is easily solved by increasing thebuffer size (latency)

Loss of data due to overloads on and flawed performance of the PCI bus

Note that all FW interfaces in IBM-compatible computers are connected to the PCI bus! There is no PC chipset withintegrated FW available. Macs have FW integrated.

With the Fireface, the latter is more often showing up than before, as OHCI chips are more sensitive to PCI busperformance problems than our Hammerfall PCI cards.


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RME offer an easy way to differ between clicks causedby CPU overload (which is solved by increasing thelatency/buffer size), and the loss of complete FireWiredata packets. The FireWire communication between PCand Fireface includes watermarking of the packets. TheFireface is able to detect whenever and how manypackets got lost. The number of errors detected is thendisplayed in the Settings dialog right below 'Video'. Nopackets lost - no display. The display will be resetwhenever a new playback starts.

This display allows to easily check the performance ofany DAW with regards to transmission errors, that arenot an RME problem. They are the same for any othermanufacturer, caused by the combination ofmotherboard/FireWire controller. There is no way tochange this performance aspect by the driver.

On another note, the Error display can give confidencein working with otherwise often unpredictable hardware.Start audio and fool around with windows, hard drivesand network. If no error shows up in the Settings dialog,the DAW is fully compatible and reliable, at least on theaudio hardware side. If clicks and drop outs occurwithout any error displayed, the DAW might be notpowerful enough to handle realtime audio, but that's adifferent story.

So far part one. Part two is even more interesting. Whenever a data packet gets lost (see above), the current sampleposition shifts by the amount of lost samples. This way first the additional safety buffer is eaten up, then the position getsso critical that the application has no time left to process the buffer. Even at minimal CPU load the audible effect is likehaving 100% CPU load - complete distortion. Next the position wraps around which makes latency rise dramatically.

When suffering from such effects, the best advice is to get a new computer. But of course RME tries to get the most evenout of such a situation. There are not much choices to resolve this problem:

Stop the hardware. It depends on the stability and robustness of the audio application wether it survivessuch a state.

Stop the hardware and restart it shortly thereafter (stop for about one second). Better than before, butstill might cause trouble with the audio application.

Stop and restart the hardware as quickly as possible.

That's exactly what RME do. The Fireface stops audio for about 3 ms (!), this way resetting the sample position on everycase of lost packages/drop out. It works fantastic, is an outstanding unique feature of RME hardware, and once againshows the tremendous possibilities when not relying on a fixed chip solution.

The quick restart corrects even the position of record data. Record a 5 minute track with (for example) Cubase or Nuendoand intentionally cause a PCI overload. As a result, the recorded wavefile will include a segment of 3 ms muted audio. Allthe following audio samples are still at the correct position! A drop out of 3 ms even gives the chance to make the drop outinaudible. Adding some samples at the beginning and the end of the mute segment, just 'drawing' them, or filling the mutesegment with a copied part makes the short drop out completely inaudible.

Conclusion and Summary

RME came late with their first FireWire solution, the Fireface 800. Nevertheless RME immediately became the leader inFireWire audio. We have Pro features today, where others can only wait on newer chips, that might fix something, mightadd something else, but most probably will not give the superior solution that RME already provides. RME again becamethe reference, and looking at the competitor’s solutions we will stay to be the reference for quite some time!

Copyright © Matthias Carstens, 2004.

All entries in this Tech Infopaper have been thoroughly checked, however no guarantee for correctness can be given. RME cannot beheld responsible for any misleading or incorrect information provided throughout this document. Lending or copying any part of thecomplete document or its contents is only possible with the written permission from RME.

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Copyright © 2004 RME. All rights reserved. RME is a registered trademark.This website contains names and marks of other companies.


3 of 3 9/11/2008 4:26 PM

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TotalMixPart 1: The Hammerfall DSP Mixer - Hardware and Technology


The Hammerfall DSP series includes several DSP functions - without having a DSP on board. Since our first card (theDigi24), RME goes for FPGAs (Field Programmable Gate Arrays.) Instead of working with hard-wired chips (like multimediacodecs), which offer functions implemented by the manufacturer, we use our own circuits, that are completely designed byus. This is possible with the FPGA, which is, in contrast to an ASIC (chip manufactured according to a customer'sdemands), even re-programmable. In the past, it was necessary to change an EPROM on our cards for this. TheHammerfall DSP's hardware configuration inside the FPGA can even be updated by a software / driver update.

FPGAs have become more powerful during the years. The newest generation used for the Hammerfall DSP (Xilinx Spartan2) allows for computations in an FPGA (provided the designer is clever), that could only be performed by DSPs before.Even better: Again we have been able, thanks to optimal adaptions, to surpass the performance of pre-configured devices(Motorola, Sharc etc.) in the functions needed! Our mixer, for instance, processes up to 1456 channels internally with100MHz and uses far more than 500MByte/s memory bandwidth - with this alone, a Motorola DSP is completely busy. OurFPGA on the other side, calculates the peak and RMS levels of 52 channels at the same time! Only as a side note, wemay point towards the also included 3 ADAT receivers and transmitters and a completely digital SPDIF receiver andtransmitter...

In the following, the basic technology behind the Hammerfall DSP mixer is being described, including its realization inhardware with bits and bytes. Information regarding the software user interface, which presents all features easilyoverseeable and manageable, can be found in the Tech Info TotalMix: Software, features, operation.

What for...

...do I need a mixer in an I/O-card? There are several scenarios:

setting up delay-free submixes (headphone mixes)unlimited routing of inputs and outputs (free utilization, patchbay function)distributing signals to several outputs at a timesimultaneous playback of different programs over only one stereo channelmixing of the input signal to the playback signal (complete ASIO Direct Monitoring)

The last point was most important for us at the beginning. Our present hardware (DIGI96 and Hammerfall series) were onlyworking in replace mode because of the missing mixer, i. e. on the track just being routed input to output, you could onlyhear the input signal, but not the playback signal. This 'tape recorder mode' was a big step forward at the time, but itrequires an additional external mixing console. With increasing computing power and improved software, more and moreusers want to use the 'mixer' provided with the software though. In order to hear the input signal together with the playbacksignal during monitoring, there are only two ways to achieve this: either the latency / buffer size of the card is so small(ideally under 6ms) that the software can completely calculate the data, or the card mixes input and playback signals on itsown, so that monitoring can be latency-free (because the signal goes directly from input to output inside the hardware.)

In principle, a mixer being able to mix every input to any playback track once would have been sufficient - perfect ASIO 2.0Direct Monitoring. After some thinking, however, this was way too little for us. If at all, it should be possible to mix any inputchannel into any number of playback channels - more flexibility has never been a drawback.

From this point, it was not far to the next demand: The playback tracks should be entirely mixable and routable in the mixer.This would allow for creating several completely independent submixes and freely routing the playback channels. Finally,when summing several tracks onto one, there will soon be too high a level: digital overload. Therefore, the levels of thephysical outputs must be reducible. And because the playback tracks always pass the mixer, there must be a transparent'unity gain' setting, guaranteeing bit-accurate playback. You can see already: even without effects and EQ, a mixer canbecome quite demanding...

Internal

The essential criteria for the power of a digital mixer are the number of channels that must be processed simultaneously,and the internal resolution. Example Digiface: 26 inputs plus 26 playback tracks can be mixed to 28 physical outputs in anyorder. This means (26 + 26) x 28 = 1456 combinations simultaneously possible. The mixer must be able to calculate 1456channels!

As long as monitoring would have been the only point for this concept, one would have got away with 16 bit resolution.

TotalMix: The Hammerfall DSP Mixer - Hardware and Technology http://www.rme-audio.com/english/techinfo/hdsp_tmhard.htm

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Including the playback tracks, this is no longer possible. Because they are always routed through the mixer, it does notonly have to have higher resolution than 24 bit in order to avoid rounding errors for gain changes, but it also has to supplyadditional headroom for gain rises and the summing of several channels.

Extremely high resolution does not make sense and causes a high consumption of resources within the FPGA. We havetherefore decided to have a basic fixed-point arithmetic with the following values:

The multiplier has 40 bit resolution. This comes from a maximum audio resolution of 24 bit, that can bealtered in gain with a resolution of 16 bit (65563 steps) with a fader (24 + 16 = 40.)

The adder has 36 bit resolution. From these, 7 bit are used for the necessary headroom (up to 52channels with 6 dB gain can be mixed to one channel.) The audio signal has again 24 bit resolution, theother 5 bit are used for the LSB to decrease rounding errors.

The output signal is truncated to 24 bit without dither.

Faders: 16 bit resolution, equals 65536 steps, which are provided following a special function for correctperception. Gain change input and playback channels: +6 dB down to maximum attenuation. Outputs: 0dB down to maximum attenuation. Because of the screen display, the number of values that can be setwith the mouse is dependant on the number of pixels. This is 137. In fine-mode (shift key), this value ismultiplied by 8 for 1096 different gain values.

Pan: pan law 6 dB. This means, the signal is attenuated by 6 dB in center setting. From center to leftand right, there are 50 steps on either side. Linearly processed and displayed, from L 1.00 via 0.50 (C)to R 1.00.

Total Overload

As mentioned before, the mixer has two fundamental properties: It can not be overloaded and offers bit-accuracy at unitygain, i. e. it passes 24 bit CRC tests.

In order to avoid internal overloads when summing (of 52 channels with +6 dB gain each in this case), the 24 bit resolutionof the input signals is not enough. Mixing of signals means adding, i. e. the new signal generally has a higher level than thesingle input signals. The worst case is completely correlated, in other words identical signals: already for two signals, theoutput level is twice as high (+6 dB.) With every further addition this factor is decreasing though.

The overload safety of TotalMix can be demonstrated easily even without truck loads of external hardware. The followingexamples were made using the software HpW Works (to generate and analyze the test signal), a Digiface and a DIGI96/8.The oscilloscope used is a Tektronix TDS 210, connected to the analog output of the Digiface.

First, we digitally generated an interesting test signal (gotta have some fun...), consisting of several sines with 0dBFSaltogether, highest possible digital level in other words. In an application, the recorded signal was then sent to all 26 tracksof the Hammerfall DSP and still played back unaltered. The inputs and outputs of the Digiface used in this test wereconnected together, applying the played back signal also to all 26 inputs.


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By means of TotalMix, all playback and input channels were now routed to the left channel of ADAT 1 (1+2.) This way, 52channels sum up into one. For analysis, the output signal ADAT1 was routed through a DIGI96/8. As can be seen in thisscreenshot, output channel 1 is of course totally overloaded, but after attenuation of the output level with the fader by ca.40 dB, the signal presents itself unaltered.

This test is easy to follow up, but in fact not completely correct, because the signals at the inputs are delayed by twosamples, which is the reason for the addition not being ideal. By the way, the addition of identical signals follows simplemathematical laws: log(10) of the number of channels, multiplied by 20 (same procedure as other dB-calculations in theaudio field.) This leads to some 34.3 dB, plus 6 dB max. gain, equals 40.3 dB. The overload headroom of TotalMix is 7 bit,translating to 7 x 6 dB = 42 dB, and is thus covering the theoretical maximum. In real life, the level rise is much less,because the added signals have less in common (correlation.)

Signal Quality

Besides, the signal quality is of course very important when making gain changes. The source signal should ideally not bechanged, no matter how strong the level may be changed. This is only possible within limits though. For decreasing levels,the lowest components of the signal get lost automatically because of the limited resolution at the ports. In this case, wespeak about 24 bit, meaning levels below -144 dBFS.

As can be seen in the figure below, with 24 bit it is well possible to create and display levels below -144 dBFS. The 2ndharmonic of the test signal e. g. is at -160 dBFS. By attenuation by 10 dB it can be shown without effort that TotalMixleaves the original signal unaltered (no visible distortions, no change of the level relations.) Even the 2nd harmonic showscleanly at -170 dBFS.


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Attenuating by 30 dB lets the signal dive under the measurement limit of the FFT, i. e. the noisefloor. This noisefloor - inreality quantization distortions - is generated by truncation at 24 bit, in this case caused by outputting the test signal via anADAT port.

In conclusion, or 'Summing up', we can say: TotalMix causes practically no signal degradation thanks to simple fixed pointarithmetic. Signal-to-noise ratio (SNR) and total harmonic distortion (THD) of the original signal are left practicallyunchanged.

But what happens, if the user does not see the overload display and the signal is clipped in spite of the headroom at theoutput? With a bad design, this may lead to calculation errors that can show as folding down of the original waveform inthe anti-phase direction for the whole duration of the clipping. Even shortest impulses, which could be clipped inaudiblyotherwise, cause dreadful and loud distortions.

Again TotalMix works exemplarily and limits the signal as clean as a precision brickwall limiter, without causing furtherdisturbances. The left picture shows a full scale sine of several inputs mixed on one output. The right picture showsTotalMix limiting peaks in music, performing like a digital peak-clipper.

Software

All further functions necessary for easy and practical use are not determined by the hardware, but the software, whichcontrols the hardware functions. This includes:

storing of all settingsspecial functions by hot keys: fader to 0, pairwise configuration, fine-mode etc.mute and solo functionsdifferent display modes


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More information on this topic can be found in the Tech Info TotalMix: Software, features, operation.

Glossary

16 bit: 65,536 level steps or 96 dB theoretical dynamic range

24 bit: 16,777,200 level steps or 144 dB theoretical dynamic range

ADAT optical: TOSLINK interface using the Alesis protocol. Allows for up to 8 channels at 24 bitresolution and 48 kHz sample rate

Audio word length: Resolution of the digital signal in bit

Bit: Smallest digital unit. With respect to resolution, 1 bit translates to 6 dB of dynamic range

dB: Decibel, logarithmic value for easier display of relations

dBFS: dB Full Scale. Logarithmic level value relative to digital maximum level

DR: Dynamic Range. Relation of maximum level to noise level with a stimulating signal at -60dB

FFT-Analysis: Fast Fourier Transform. Spectral decomposition of any given signal

LSB: Least Significant Bit

Sample rate: Number of samples / probes being taken of the audio signal within 1 second

SNR: Signal-to-noise ratio. Relation of maximum level and noise level

THD: Total Harmonic Distortion. sum of 2nd to 10th harmonic in relation to 1st harmonic

Truncation: Cutting off the less significant bits of an audio signal


All entries in this Tech Infopaper have been thoroughly checked, however no guarantee for correctness can be given. RME cannot beheld responsible for any misleading or incorrect information provided throughout this manual. Lending or copying any part or thecomplete document or its contents is only possible with the written permission from RME.




5 of 5 9/11/2008 3:47 PM

TotalMixPart 2: The Hammerfall DSP Mixer - Software, Features, Operation


Having introduced the hardware design in the Tech Info TotalMix: Hardware and Technology, this Tech Info describes thesoftware aspect. We have tried to render all functions of the hardware mixer in a simple and effective way via a practicallyoriented user interface. See for yourself...

The Concept

As explained in detail in Part 1, TotalMix hardly knows any limitations. When mixing all channels simultaneously with eachother while being able to mix one channel several times differently onto other channels, bus and inline structurescustomarily found on analogue consoles won't suffice. The mixer surface, however, should of course appear in a knownway in order to shorten the time needed to get used to it. Conventional designs with batteries of subgroup and routingbuttons? With analogy to analogue consoles, channel routing would be doubled, the master section would then have 28subgroup faders. The usual appearance of a mixing console would still be missing, because TotalMix doesn't have eitherEQ nor aux sends - what remains are faders, pan-pots and the many many buttons...

After dismissing numerous concepts, we ended up exactly where the company Steinberg has been with Cubase for a longtime. Actually, the well-known Cubase mixer is the (almost) perfect answer for unlimited routing possibilities. And becausesomething (almost) perfect can hardly be done better, TotalMix looks like a simplified variant of the Cubase mixer (noeffects, no inserts etc.) - anyway, at first sight. Beneath the surface, we have chosen different ways though, we'll comeback to this later. Another important condition was the maximum size: the whole mixer should fit into a 1024 screen, whichhas determined the appearance significantly.

Elements of the Surface

The visible design of the mixer is mainly determined by the architecture of the HDSP system:

Upper row: hardware inputs. The level shown is that of the input signal, i. e. Fader independent. Perfader and routing window, any input channel can be routed and mixed to any hardware output (thirdrow.)

Middle row: playback channels (playback tracks of the software.) Per fader and routing window, anyplayback channel can be routed and mixed to any hardware output (third row.)

Lower row: hardware outputs. Because they refer to the output of a subgroup, the level can only beattenuated here (in order to avoid overloads), routing is not possible. This row has two additionalchannels, the analog outputs.

The two additional outputs in the lower row cause some free space above on the right side, used for a Quick AccessPanel.

Every single channel has various elements:Input and playback channels each have a mute and solo button.

Below each there is the panpot, realized as indicator bar (L/R) in order to save space.

In the window below this, the present level is displayed in RMS or Peak, being updated about every half asecond. Overs are indicated here by an additional red dot.

Then comes the fader with a level meter. The meter shows both peak values (zero attack, 1 sample is enough fordisplaying full scale) by means of a yellow line and mathematically correct RMS values by means of a green bar.The RMS display has a relatively slow time constant, so that it shows the average loudness quite well.

Below the fader, the current gain and panorama values are shown.

The white area shows the channel name, the black area shows the current routing target.

TotalMix: The Hammerfall DSP Mixer - Software, Features, Operation http://www.rme-audio.com/english/techinfo/hdsp_tmsoft.htm

1 of 5 9/11/2008 3:48 PM

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Having introduced the hardware design in the Tech Info TotalMix: Hardware and Technology, this Tech Info describes the software aspect. We have tried to render all functions of the hardware mixer in a simple and effective way via a practically oriented user interface. See for yourself...

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Tour de TotalMix

Join us for a Tour de TotalMix, where we will explain all functions of the surface step bystep. Starting up TotalMix, the last settings are recalled automatically. When executing theapplication for the first time, a default file is loaded, sending all playback tracks 1:1 to thecorresponding hardware outputs with 0 dB gain. The faders in the upper row are set tomaximum attenuation (called m.a. in the following), so there is no monitoring of the inputchannels.

We will now create a small submix for the analogue headphone output. Please start amultitrack playback and connect your headphones to the analogue output. In playbackchannel 1 (labeled 'Out 1'), click onto the routing window below the label. A list pops up,showing a checkmark in front of '1+2'. Click onto 'Analog'. The list disappears, the routingwindow no longer shows '1+2', but 'Analog'. Now move the fader with the mouse. As soonas the fader value is unequal m.a., the present state is being stored and routing isactivated. Move the fader button to around 0 dB. The present gain value is displayedbelow the fader in green letters. In the lower row, on channels 27 and 28 (AN.L. andAN.R.), you can also see the level of what you are hearing in the phones. The level meterof the hardware output shows the outgoing level. Click into the area above the fader anddrag the mouse in order to set the panorama, in this case the routing between channels 27and 28. The present pan value is also being displayed below the fader.

Please carry out the same steps for 'Out 2' now, in order to route it to the analogue outputas well.

Often signals are stereo, i. e. a pair of two channels. It is therefore helpful to be able tomake the routing settings for two channels at once. Press the Ctrl-key and click into therouting window of 'Out 3' with the key pressed. The routing list pops up with a checkmarkat '3+4'. Click onto 'Analog'. Now, channel 4 has already been set to 'Analog' as well.

When you want to set the fader to exactly 0 dB, this can be difficult, depending on themouse configuration. Move the fader close to the 0 position and now press the Shift-key.This activates the fine-mode, which stretches the mouse movements by a factor of 8. Inthis mode, a gain setting accurate to 0.1 dB is no problem at all.

Advanced users will probably have only cross-read this section, because this implementation is both intuitive and common.But now it's getting interesting. Please set 'Out 4' to a gain of around -20 dB and the pan close to center. Now click ontothe routing window. You'll now see two checkmarks, one at '3+4', the other one at 'Analog'. Click onto 'SPDIF'. The windowdisappears, fader and panpot jump to their initial values, the signal can now be routed to the SPDIF output. You cancontinue, until all entries have got a checkmark, i. e. you can send the signal to all outputs simultaneously. This is one ofseveral differences to the Cubase mixer, which does not allow for multiple selections.

You will certainly have noticed that the headphone mix has not changed, while you were routing the channel to otheroutputs and setting different gain values. With all analogue and most digital mixing desks, the fader setting would affect thelevel for every routed bus - not so for TotalMix. TotalMix allows for setting all fader values individually - a feature owners ofmany expensive digital consoles would like to see on their desks. Therefore the faders and the panpots jump to theappropriate setting as soon as another routing is chosen.

Back to 'Out 4'. How to get rid of the checkmark, that is to say the routing? Just by moving the fader to m.a. This settingdeactivates the routing...why route if there is no level?...Click onto '3+4' in the routing window, pull the fader down, openthe routing window again - the checkmark is gone.

Click here for a larger picture of Submix mode


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Such a wide range of possibilities make it difficult to maintain the overview. Because practically all hardware outputs canbe used for different submixes, as shown. And when opening the routing windows you might see an army of checkmarks,but you don't get an overview, i.e., how the signals come together and where. This problem is removed by the (trulyingenious) view mode 'Submix'. In this mode, all (!) routing windows jump to the routing pair just being selected. As can beseen in the picture, you can then see immediately, which channels, which fader and pan settings make a submix (here'Analog'.)

Click here for larger picture of Solo mode

Mute works pre-fader, thus mutes all active routings of the channel. As soon as any Mute button is pressed, the MasterMute button lights up in the quick access area. It can switch all selected mutes off and on again. You can comfortablymake mute groups to activate and deactivate this way. The same holds true for the Solo and the Master Solo buttons. Solois working as a solo-in-place. As soon as one Solo button is pressed, all other Mute buttons are activated and light up. ButTotalMix would not be an Intelligent Audio Solution, if it didn't behave as you'd expect from a mixing console. If you, forinstance, mute 'Out 1' to 'Out 4' and press Solo for 'Out 5', of course all Mute buttons will light up. If you deactivate Solo,the Mute buttons for 'Out 1' to 'Out 4' light up as before. And if you chose Solo for a channel of this Mute group, mute willbe deactivated, but immediately activated again, if Solo is released. Exactly as it should always be (but hardly anywhereis....)


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Hotkeys

TotalMix knows only a few, but very effective key combinations, that make work (read: setting the mixer up) considerablyeasier and faster. The Shift-key for the fine-mode for faders and panpots has already been mentioned. But the Ctrl-key cando far more than changing the routing pairwise:

Clicking anywhere into the fader area with the Ctrl-key pressed, sets the fader to 0 dB, -6 dB for thehardware outputs.

Clicking anywhere into the pan area with the Ctrl-key pressed, sets the panorama to <C> meaning'Center'.

The faders can also be moved pairwise, corresponding to the stereo-routing settings. This can be achieved by pressingthe Alt-key and is especially comfortable when setting the SPDIF and analogue output level. At the same time, TotalMixalso supports combinations of these keys. If you press Ctrl and Alt at the same time, clicking with the mouse makes thefaders jump to 0 dB pairwise, and they can be set pairwise by Shift-Alt in fine-mode.

Also very useful: the faders have two mouse areas. The first area is the fader button, which can be grabbed at any placewithout changing the position. This avoids unwanted changes when clicking onto it. The second area is the whole fadersetting area. Clicking into this area makes the fader jump to the mouse at once. If you want to set several faders to m.a. forinstance, it is sufficient to click onto the lower end of the fader path. Which happens pairwise with the Alt-key pressed.

The Quick Access Area...

...really justifies its name. You can find some options here, which make using TotalMix child's play and almost revolutionizemonitoring. We have already described the Master Mute and Solo buttons, they allow for group-based working with thesefunctions.

In the View section the single rows can be made visible or invisible. If theinputs are not needed for a pristine playback mix, the whole upper row fallsout of the picture after a click on the input button. If the hardware outputsdon't interest you either, the surface can thus be reduced to the playbackchannels to save space. All combinations are possible.

Submix sets all routing windows to the same selection as described before.Deactivating Submix automatically recalls the previous view.

By the way, the mixer can also be made smaller horizontally, and, scrolled.TotalMix can be made substantially smaller and space-saving on thedesktop/screen, if you have to have to monitor or set only a few channels.

The Presets are one of the mightiest and most useful features of TotalMix.Behind the 8 buttons, 8 files are hidden. These contain the complete mixerstate. Just try it: all faders and other settings follow the changing of preset(s)in real-time, just by a single mouse click. The Save button allows for storingthe present settings in the present preset. You can change back and forthbetween a signal distribution, complete input monitoring and varioussubmixes without any problem.

Also here, RME's love for details can be seen. If any parameter is being altered after loading a preset (e.g. moving a fader), the preset display flashes in order to announce that something was changed, stillshowing, which state the present mix is based on.

If no preset button is lit, another preset had been loaded via the File menu and 'Open file'. Mixer settingscan of course be saved the usual way, and with long file names.

Up to three Hammerfall DSP systems can be used simultaneously. The Card buttons switch between thesystems. Systems, because card 1 can be a Digiface, but card 2 can also be a Multiface.

It remains to mention that the number of ADAT channels is reduced to half automatically when choosingdouble speed operation (88.2 or 96 kHz.) The display is adjusted accordingly, fader settings remain storedvia a complex algorithm.

The Level Meters

Having set a new standard with the level meters of DIGICheck , we could of course not compromise for Hammerfall DSP.


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However, this time the level meters should be realized in hardware, in order to be capable of using them independent ofthe software in use. Besides, the CPU load for calculating the RMS values for 52 channels with a decent display would nothave been unproblematic.

The level meters integrated in TotalMix - considering their size - cannot be compared with a pure meter bridge (availablelater.) Nevertheless, we have integrated all our know-how and many useful functions already here.

Peak and RMS is displayed for every channel. 'Level Meter Setup' makes various options available:

Display range 40 or 60 dBPeak Hold time adjustableNumerical display selectable either Peak or RMSNumber of consecutive samples for Overload display (1 to 15)RMS display absolute or relative to 0 dBFS

The latter is a point often overlooked, but nonetheless important. RMS shows 3 dB less for sine signals. This ismathematically correct, but not very reasonable for a level meter. Therefore, we had corrected DIGICheck's RMS displayby 3 dB, a full scale sine signal shows both 0 dBFS Peak and RMS. This setting also yields directly readable signal-to-noise values, while other applications (like WaveLab) will show a value 3 dB better than actual (because the referenceis not 0 dB, but -3 dB.)

The value displayed in the text field is independent of the setting40/60 dB, it represents the full 24 bit range of the RMSmeasurement, thus making possible a SNR measurement 'RMSunweighted', which you would otherwise need extremelyexpensive measurement devices for. An ADI-8 DS connected tothe Digiface will therefore show around -113 dB on all 8channels.

We are aware of the fact that this level display might causesome alarm, constantly bringing the reduced dynamic range ofyour equipment, maybe of the whole studio, in front of youreyes. Nice to have everything 24 bit - but still noise and humeverywhere in the range around -90 dB or worse... sorry, butthis is hard reality. The up-side about it is that TotalMix allowsfor constantly monitoring the signal quality without effort. Thus itcan be a valuable tool for sound optimization and error removalin the studio.

Conclusion

With a file size of only 200 kB (despite all bitmaps, graphic elements and elaborate functions), TotalMix sets its mark alsoregarding the need of resources. We are absolutely sure: TotalMix will make you enthusiastic, our competitors unsure, andset a new standard even in the area of typical DSP-based I/O-cards - a standard others will have to compete with. And thisis only version 1.0...


All entries in this Tech Infopaper have been thoroughly checked, however no guarantee for correctness can be given. RME cannot beheld responsible for any misleading or incorrect information provided throughout this manual. Lending or copying any part or thecomplete document or its contents is only possible with the written permission from RME.




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rme totalmix - features_operation

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