26th March 2007
Joined: Dec 2006
Location: NY NY
Note: THE FOLLOWING IS A PERSONAL OPINION, NOT A FACT, the only fact about is that it will probably piss off at least 200 people....
I would not recommend/nor would I be happy mixing for FILM and THEATRICAL utilizing a bass management enabled A-Chain. I feel that it may not provide you with a proper understanding of what the mix will sound like in a theatrical B-Chain system.
I see no reason why you shouldn't mix on a bass managed system for BROADCAST or GAMES etc. Since that will end up being the B-chain anyway. So i'd go for it with your gear, what the heck!
BUT, I know people who do this and love their systems and delivery great mixes...
I mix on a hugeass Genelec system, my Foley Pit has a 5.1 genelec system, and all my editing systems have genelecs...
so I'm not going to start to throw ANY stones here. Each to their own.
PS: this post obviously helped exactly 0%.. sorry.
PPS: no i'm not a Genelec spokesperson.....
26th March 2007
Joined: Mar 2007
that makes sense. since it is direct to dvd, i'm using it. i love genelecs also. these weren't my choice- i think these dynaudios lack some low end goodness but are great otherwise. however with bass management, everything sounds great. not a lot different, but just nicer, warmer.
4th April 2007
Joined: Dec 2006
Location: NY NY
SOme mixed ranting on DIALnorm and DOLBY encoding issues
collection of a couple rants of mine in the past:
HDnet normally wants a digital video tape with LtRt and DOlby-E. The DOlby-E stream must be laid on the tape so that the audio ON TAPE is in sync with the video. So, depending on the frame rate of the video tape you will need to off-set the audio so that after passing thu the encoder the audio is in sync ON TAPE. The encoding process places an amount of delay on the output signal due to the encoding time. Additionally, during decoding of the DOlby-E stream the audio will be once again delayed. Normally you can delivery in sync ON TAPE or in sync AFTER the decoder or even out of sync by not doing any delay time adjustments. It depends on the delivery spec. Again, HDnet normally wants it in sync ON TAPE because they engest it to a server without decoding so all the media on the server is video+audio sync'd.
The same is true for an AC-3 delivery. Also note, that at 29.97 there is about 1 frame of offset for the encode process and 1 frame for the decode process.
DOlby-E is editable since is is a FRAME based delivery medium.. DOlby-AC3 is a linear digital stream. You cannot edit the AC3 stream ( ok I actually have a couple times in an emergency, but its a crap shoot as to getting the punch in and out clean enough to work sonically and meet QC requirements. ( Don't try this at home folks we're trained professionals... )
DOlby E is used to broadcast production transfer and can be decoded and re-encoded any number of times without loss or problem. DOlby-AC3 cannot be ecoded and reencoded without significant loss of quality. DOlby AC3 and DOlby-E can support broadcast delivery up to the cable head or brodcast local center. From there AC3 is the delivery methodology for the final audio to the consumer. ( Like your cable box )
If you are trying to FIT a mix into a broadcast require spec, lowering the mix is not the right answer. It will take the lower end SFX and ambiences and drop them 3 db as well. You need to remix to meed the DIALNORM reqirement. The biggest problem in all this is that Discovery and almost all the other major broadcasters are using DIALNORM backwards. But until they get it right we are forced to mix to a set DIALNORM setting.. The LM100 DIALNORM methodology is as follows: Whatever channel is selected for monitoring, that channel is monitored using the LM100 and the custom software within the LM100 that reads the Dialogue level. As it does this it averages over the duration of the program. It also has the ability to differenciate between Actual Dialogue and non-dialogue events on the selected channel. Once you have the DIALNORM (actual) you can then encode the overall audio as DOlby-E or DOlby-AC3 with metadata. Part of the metadata is in fact the DIALNORM setting. The DIALNORM metadatasetting should accuratly match the ACTUAL DIALNORM average. If this is done, then no matter what the level of the mix really is, the level between all the mixes on a program channel will be raised or lowered to meet the broadcast DIALNORM setting. What is happening in reality is the broadcasters are forcing you to MIX to a specific DIALNORM setting instead of using DIALNORM properly. So if you just lower your mix total level you will probably lose a lot of the quiet SFX and Ambience areas and this may be not what you or your client wants. You would be better off remixing the spot or program to the correct the DIALNORM setting.
We do this alot with outside facilities for just this reason. In fact one of the things we are doing more and more of is DOlby-E and DOlby-AC3 mastering for smaller houses for this very reason.
Seems i'm at DOlby more and more fixing this stuff before it goes to the Broadcast company. go figure.
this is not about the mixers, sound designers, re-recordings issues, editing, avid editors or the like. THe reason for the LM100 is to provide the customer with an acceptable product that does not make them hit the channel changer and go to a competitor. LM100 specs, or actually, Dialog norm specs provide for a means to control the level of a broadcast stream in such a was as to allow a listener to sit down, turn on the TV, set the level, and enjoy the TV shows. If the specs are met for Dialnorm, the listener at home does not have to raise the volume for a quiet show, and then dive across the room for the remote, while ducking incomming tweeters, when the commericals or another show starts.
Although dialognorm is being misunderstood by the broadcasters, and misused, it still works the way its being stated in the broadcast specs. Dialnorm can be checked in 2 ways, there is a float window of 10 seconds (i think.. it might be up to 30 secs i'll have to recheck that)... and it can check a show from beginning to end and offer an average. If you take the average and set you dialognorm meta data to MATCH the actual dialnorm from your show. AND, if everyone else on the broadcast steam in question does the same thing, the broadcaster and set top box manufactures can set a level ( lets say -27 ) that works: and no matter how loud or quiet you mixed your project. The level is adjusted up or down to MEET the dialnorm directed by the broadcast stream, set top box, or dvd player. Thus all the programs are the same level while you listen at home.
One of the problems we are trying to overcome at present is the direction the Broadcasters are taking, forcing a delivery of dialnorm to be -27. This works, but its not the way its supposed to be. DOlby is setting up a ad-hoc standards committee including broadcasters, engineers, post facilities and the like to work on this and other level based delivery issues.
what the broadcasters SHOULD be doing is forcing their QC department to simply check to assure Dialnorm meta data = dialnorm acutal... AND, of course, specifiying in the delivery specs that, we as content providers, match our Dialnorm Meta data to Actual Dialnorm on our deliveries. The problem is that it means the broadcaster has to run everything through once to check QC instead of just spot checking some random area in the mix.
Whats supposed to happen is the following:
You don't have to set a spec of -27 diagnorm. WHAT YOU DO NEED TO DO, is to set a spec that says the diagnorm meta data MUST BE EXACTLY what the actual Diagnorm really is. If the metadata and the acutal DN do not match kick it back in QC.
If in fact this were accomplished with:
a commercial at a true DN -20 and metadata DN set to -20
a show at true DN -29 and metadata DN set to -29
a commerical at true DN -19 and metadata DN set to -19
When each is played back with the proper setting the levels will be consistant with each other.
What screws this up are the folks not using DN correctly, by checking the level of their show / commerical and then setting DN to -31 when the true DN is -12. ( or whatever ) The whole concept of Dialogue Norm is to tell the system that the show is +/- and automatically adjust it to resolve the differencial to a standard level. At least thats what its supposed to do.
1. the mixer/client sets the appropriate metadata for the audio program being created by checking DIALNORM. these are set to match.
2. The resulting audio program, together with metadata, is encoded as a DOlby E stream and sent to the broadcaster.
3. the DOlby E stream is decoded, checked, and adjusted as a matched program/metadata pair, reencoded as DOlby E, leaves the studio and goes to Master Control, where bunches of DOlby E streams are decoded back to their individual audio/metadata sets.
4. The audio program/metadata pair that is selected is sent to the transmission DOlby Digital encoder, which encodes the incoming audio program according to the metadata stream associated with it.
5. The metadata and the broadcast encoder, as well as the home decoder work togther to deliver the audio so that my decoder controls the levels for the commercial, or show. each using the metadata carried within the individual segment. OR in the case of the broadcast company decoding the final signal and sending it out to my crappy TV, the metadata once again keeps all the various items within the same program level at the broadcast head end.
Oh, my decoder says
"this is supposed to be diagnorm of -27 but its diagnorm of -10...oops droppin' down a bit... ahh thats better...
oh... look this one is diagnorm -30 oops lets kick it up a little... ahhh.. much better.
of this one is JUST right at diagnorm -27 cool... I can take a beer break."
bottom line is the DOlby-E stream should be mixed to what the client or mixer wants.. Then check the DIALNORM, then match the Metadata. Then encode properly in DOlby-E the mix in DOlby-E is NOT modified at all. The Metadata defines the method in which the mix is heard at the end of the pipeline. So the DOlby-E stream with meta data is delivered to the Broadcast head, transfered or re-transmitted to the local cable or other broadcaster and decoded with metadata. the metadata is supposed to be passed the the final DOlby-AC3 stream during the re-encode and the mixes are adusted by the broadcasters DIALNORM setting so whe you receive the mix at home and you decode it at home OR its decoded and sent analog, the mixes are all adjusted dynamically to meet the broadcasters DIALNORM setting. If your DIALNORM is low or high the mix is adjusted so the listening community just hears a nice standard level across Ads and Programs.
8th April 2007
Joined: Dec 2006
Location: NY NY
LEQ-A LEQ-M LM-100 differences
LEQ-M and LEQ-A are measurements, using different weighting curves, of Loudness for long term program averaging. LEQ-A uses A-weighted and LEQ-M uses the CCIR weighting.
LEQ-A has a slight lifting bump in the 100hz to 1kz range and LEQ-M has a flatter lift to around 10k with a bump between 5k and 10k
LM-100 uses a patented methodology selecting the Dialogue from the overall mix and determining average loundness based on either the overall program lenght ( infinite setting ) or based on a sliding windowed duration. These settings will create different end results in measurements. Where LEQ-A and LEQ-M looks at the OVERALL program loudness, LM-100 averages only dialogue.
If you mix a threatrical TRAILER LEQ-M tells you the overall average level of the trailer, music, sfx and dialogue. This is important as there are specific specs for mixing threatrical trailers.
LM-100'a ave number is used for broadcast specs for mixing dialogue.
17th April 2007
Joined: Sep 2006
you are my hero,
this is the third time I've read this thread now and I still get something new out of it every time!
Are any broadcasters using dialnorm correctly? Sounds like a bit of a pain having to mix to a dialog spec.
22nd April 2007
Joined: Dec 2006
Location: NY NY
excerpt from a post website.... good info
This is a great primer on post written by Bruce C. Nazarian M.P.S.E.
here's the website Audio Post FAQ's
* Production Dialogue Editing - In order for the production audio recorded on the set or on location to be properly mixed, a Dialogue Editor needs to properly prepare it. This means locating the proper take from the recorded production audio, checking sync (so it actually works with the picture properly), and eliminate extraneous noise so the Mixer has clean dialogue to use during the Mix.
* ADR [Automated Dialogue Replacement] - In cases where the production audio is too noisy, or otherwise unusable (bad line reading, airplane fly-by, etc.) the Dialogue Editor will "cue" the line for ADR. This means replacing that line or lines of dialogue using the Automated process of Dialogue Replacement. This process takes place on the ADR Stage, a specialized recording studio where the actor can record lines in sync with the picture.
* Once a replacement line of dialogue has been recorded, the Dialogue or ADR Editor will check the sync carefully, editing the take if necessary to precisely match it to the picture, and prepare it for the Mixing Stage. This process is also known as "looping".
* Sound Effects Editing and Design - Ever wonder how they made the sound of Darth Vader's helmet breath, or the Empire's Tie Fighters, or that great train wreck sequence from "The Fugitive"? - Sound Effects Editors and Sound Designers are how. The process of adding sound effects (backgrounds like: air, rivers, birds, traffic, and hard effects like: gunshots, door slams, body falls, etc.) has been the domain of sound effects editors for years. Although originally edited using 35mm magnetic film, recent years have seen the development of many different Digital Sound Editing systems. More and more projects are using digital technology because of the efficiency and quality it can bring to sound effects. Sound Designers use digital and analogue technology to create sound effects that have never been heard before, or to artistically create specific "mood" sounds to complement the director's vision of the visuals.
* Foley - Taking its name from Jack Foley, the Hollywood sound effects person generally regarded as the "father" of these effects, Foley effects are sounds that are created by recording human movement in sync with the picture. Different from the environmental backgrounds and hard effects that comprise edited sound effects, Foley effects are sounds like footsteps, prop movement, cloth rustling, etc. The players involved in this process are the Foley Mixer, who records the sounds, and the Foley Walkers who create those sounds. After the Foley Effects are recorded, the Foley Editor will make any slight timing adjustments necessary to ensure that they are exactly in sync with the final picture.
* Music Composition - Music for film/TV falls into three general categories: Score, Source and Songs. The Composer is the individual hired with the responsibility to prepare the dramatic underscore. Source music is that music we hear coming from an on screen or off screen device of some kind; some examples are radio source music, phonograph records, TV show themes, when seen on a TV set in the shot, and many other similar variations. Source music may be original, or licensed from a number of libraries that specialize in the creation of "generic" music. Songs may occupy either function, depending on the dramatic intent of the director. Using "Pulp Fiction" as an example, Director Quentin Tarantino hired a Music Supervisor (Karyn Rachtman, FYI) to "score" the picture using period music of the 1970's almost exclusively. Most contemporary films use a combination of score and source music.
* Music Editing - The Music Editor assists the Composer in the preparation of the dramatic underscore. Frequently working also with the Music Supervisor the Music Editor will take timings for the Composer, (usually during a spotting session )in order to notate the specific locations in the film where underscore or source music will punctuate the narrative. Once the underscore is recorded, and the source music gathered, the Music Editor will usually be the person who edits or supervises the final synchronization of all music elements prior to the mix.
* Mixing (also called Dubbing) - The Mixers have the responsibility of balancing the various elements, i.e., - the Dialogue (and ADR), Music, Sound Effects, and Foley Effects, in the final mix. The Dialogue Mixer, (also called the Lead Mixer or Gaffing Mixer) commands the mixing stage; his partners in the mix are the Effects Mixer and the Music Mixer. On large features, it is not uncommon to have an additional mixer handling just the Foley effects. On huge pictures with tight deadlines, it is possible that several teams of mixers are working simultaneously on numerous stages in order to complete the mix by the release date.
Where does post-production begin ?
If you haven't shot your film yet, it begins before you shoot - by selecting the finest production dialogue mixer you can afford. The little bit extra paid to a great production mixer can save you tenfold later in post-production.
What does the production sound mixer do ?
The production mix team are the individuals charged with recording your live dialogue, in sync with the camera team. The Production Sound Mixer is your most important ally at this stage in the movie's production. Although you will be anxious to complete as many setups as possible during each shooting day, a little extra time guven to the sound mixer to allow him to capture scene ambience (called room tone) will pay off hamdsome dividends later during our dialogue editing. The production mixer will have with him a boom operator, who handles the boom mics, and usually a cable person, who will be in charge of wrangling the audio cables needed to mike the set appropriately. Usually they will record on a Nagra recorder, but digital recordings on Portable Time code DAT machines are becoming more common.
We are shooting our film on location...what now ?
Generally, each day after the completion of the shoot, the production audio rolls will be sent to an audio post house for transfer to "dailies" form. If the film is being edited filmstyle, using 35mm mag audio and film dupes (as opposed to electronically, using an Avid or Lightworks edit system), the production select takes will be transferred to 35mm mag film. This sprocket-based medium will allow the film editor or assistant to sync that day's select film takes with the audio track that corresponds to it.
If the production is being edited electronically, using a computer-based edit system, the options are a bit different. Frequently, a video post house will be engaged during shooting to telecine the selected and printed film takes. In addition, they will transfer the production audio from Nagra or DAT and generally synchronize the dailies onto some form of videotape, for later digitizing into the Avid or Lightworks editing system. Syncing dailies at the video house eliminated the need for the assistant film editor to do it, and allows the assistant to load the editing system instead. An important task to accomplish during the digitizing is for the assistant to correctly log in the dailies time code that is recorded on the Nagra or DAT location tracks. This will allow the EDL (edit decision list) that is created later on to accurately reflect the original time code that was shot with that scene, and allows the audio post house to electronically automate the re-loading of the production dailies, should they need to be replaced.
And this goes on all during the filming?
Yes. Dailies transfers will continue until there are no more dailies coming in, and shooting has wrapped. During this time the editor may also need reprints of previously transferred takes, or prints of previously unprinted takes. They are processed in the same manner.
We are done shooting...now what happens?
Now the real fun begins. The editor has been syncing dailies all during shooting, choosing which scenes should begin to form the final cut. During the next several weeks, the process of editing will continue as the decisions are narrowed down to final choices. It is at this time that the final form of the film begins to take shape. Although the film editor may have been assembling the "editor's cut" during the shooting period, the first formal edit period is generally referred to as the director's cut, and it is when the first full assembly of the film is refined.
Do I need Audio Post during editing?
Well, yes. During the editing you may still need reprints of selected takes or outtakes. The audio post facility will duplicate these for you. But the real job is starting to come into view: the locked cut.
What is the locked cut ?
In short, the final version of the finished film. Although it may receive a small edit here or there in the next few weeks, the film is essentially "locked" into this form.
What happens once the cut is locked ?
Audio Post begins now in earnest. Once the cut has been locked, the film can be spotted for the placement of sound effects and music. The Supervising Sound Editor, the Director and possibly the Film Editor and Composer will gather at one or more spotting sessions to determine the film's audio post needs. "Spotting for music" is the process of viewing the locked cut and deciding where the music score will be, and where the source music will be needed. "Spotting for sound" is the process of determining:
* if and where any dialogue problems may exist, so that ADR can be cued to be recorded
* where sound effects are needed and what kind
* what Foley effects will be needed in the film, and where
* If Sound design (the creation of special sound effects), will also be needed.
What actually happens after 'spotting'?
The real job of audio post has now begun. In the next weeks or months, the sound editors will locate and synchronize all of the sound effects needed in the film. If necessary, they will create Field Recordings of new sound effects needed for the film. The Foley supervisor will cue all of the Foley effects that will be needed; they will be recorded by the Foley Mixer and the Foley Walkers; the ADR supervisor will cue all of the Automated Dialogue Replacement lines that need to be recorded during the ADR sessions, and the Music Editor will begin providing for the needs of the Composer and/or music supervisor. The Dialogue editor(s) will begin preparing the production audio for final mixing, and the ADR editors can commence editing in the ADR lines, once they have been recorded.
What happens after spotting ?
Typically, the next few weeks or months are occupied with sound editing of all types. The Director will be checking on the various aspects of the sound job as time progresses, to be sure that his vision is being realized. Usually, there is provision for one or more "effects reviews" where the effects are listen to and approved. The same goes for Foley, Dialogue, ADR, Sound Design and Music. When everything is completed and approved, the next step is Mixing (also called 'dubbing' or 're-recording').
What happens during the mix ?
During the mix, the edited production dialogue and ADR, sound effects, Foley and Musical elements that will comprise the soundtrack are assembled in their edited form, and balanced by a number of mixers to become the final soundtrack. In New York, single-mixer sessions are more commonplace than in Hollywood, where two-mixer and three-mixer teams are the norm.
The mixers traditionally divide the chores between themselves: the Lead Mixer usually handles dialogue and ADR, and may also handle music in a two-man team. In that case, the Effects mixer will handle sound effects and Foley. In three-man teams, they usually split Dialogue, Effects and Music; sometimes the music mixer handles Foley, sometimes the effects mixer covers it.
To keep the mix from becoming overwhelming, each mixer is actually creating a small set of individual sub-mixes, called STEMS. These mix stems (dialogue, effects, Foley, music, adds, extras, etc) are easier to manipulate and update during the mix.
When mixing is done, what then ?
After the mix is completed and approved, films generally require a last step called Printmastering, that combines the various stems into a final composite soundtrack. When this is completed, an optical or digital sound track can be created for a feature film release print.
It is also usual at this time to run an 'M & E' (which stands for Music and Effects) track. This is essentially the film's soundtrack with the English language dialogue removed. This allows foreign language versions of the project to be dubbed easily, while preserving the original music, sound effects and Foley. During the M & E, effects or Foley that are married to the production dialogue tracks are removed along with the dialogue. To "fully-fill" an M & E for a quality foreign release, those effects and Foley must be replaced.
Television movies usually do not require print masters, unless they have been created using SURROUND SOUND techniques. In most cases, the final stems are combined during a process called LAYBACK, at which time the soundtrack is united with a final edited master videotape for ultimate delivery.
22nd April 2007
Joined: Dec 2006
Location: NY NY
part two of primer written by Bruce C. Nazarian M.P.S.E.
What about optical soundtracks ?
Optical soundtracks (we mentioned them earlier). Almost all of the release formats, including the digital ones have provision for some kind of optical soundtrack, even if only as a backup. The optical soundtrack refers to the two-channel soundtrack that is carried on the optical track of the film release print.
How do I get an optical soundtrack ?
Once your surround sound format has been selected (see the paragraph below for more), you need to order an optical soundtrack negative for the film. In the case of LCRS mixes, a traditional two-channe; Printmaster track is created, and this is sent to an optical sound house for the creation of the optical negative. The optical sound house will record the soundtrack onto 35mm film using a special camera, and some will also develop their own soundtrack masters. Once the optical negative is shot and developed, it can be incorporated into your answer printing process, and a composite answer print containing your complete soundtrack can be printed or "shot" at your film lab. This usually happens during the first or second trial answer print phase.
: THX - Dolby - Ultra*Stereo - DTS - SDDS?
This is a BIG question. This one point alone causes much confusion amongst filmmakers. Please take a moment and read this paragraph carefully. If you need more information after that, please contact either Gnome Productions or Magnolia Studios and we will help you out.
First, about THX.
THX [tm] is not something that you DO to your soundtrack, it is just a set of sound reproduction or mixing conditions that optimize the sound of your film's soundtrack in exhibition. Simply put, the THX standards that many dubbing stages and movie theaters adhere to are a way of being certain that "what you mix is what you get", so to speak. You may choose to mix in a stage that is THX certified, and you may not. If you do, your soundtrack should sound reasonably the same in THX theaters all around the world. It is this standardization that THX brings to the filmmaking community.
You may want to visit the THX Web Site for further information. They can be found at http://www.thx.com/thx/thxmain.html
To make sense out of the rest of the names, we need to know about Film (and Television) Surround Sound
Film sound tracks (and some television ones) go beyond just Left-Right Stereo; there is a Center Channel for the dialogue, and at least one "Surround Sound" channel. The Surround channel is used to project the sound out into the theater, to "surround" the audience. This is to enhance the illusion of being "in the picture". This four-channel format is called LCRS (for the Left, Center, Right and Surround channels that the soundtrack contains). Although the technical means behind this process is beyond the scope of this discussion, suffice it to say that it works well enough to have become a standard format for release prints for many years.
You've probably already figured out that you cannot reproduce a four-channel soundtrack from a medium that only plays back two tracks. You are very right. In order to reproduce the LCRS soundtrack from a traditional film optical soundtrack (more on opticals later) you need a way to encode the channels....the Matrix
The Surround Sound Matrix Encoder (or, how to put FOUR into the space where TWO should go!)
The solution is to use an encoding device that can fold the four channels of audio down into the two channels available on the film's optical soundtrack. When the audio tracks have been processed this way, they are labeled Lt/Rt [Left Total/Right Total] in order to distinguish them from ordinary Left/Right Stereo soundtracks. The Surround Sound Matrix Encoder is a necessary piece of hardware that the audio post house must have available during your film's mix, in order to create the surround soundtrack.
The Licensing of Surround Sound formats
Now we're really getting into the heart of the matter. Dolby Labs, Ultra-Stereo Labs, DTS (Digital Theater Systems) and Sony [SDDS] all have technologies available for the encoding of film surround soundtracks into film release prints. Although these processes vary somewhat as to their method, they essentially accomplish similar things. Additionally, some of these vendors offer Digital Encoding formats (Dolby Digital, DTS and SDDS currently, and Ultra-Stereo soon to come).
The Differences in Surround Sound formats
In the most basic form, Theatrical Surround Sound consists of LCRS: Left, Center, Right, and mono Surround. A soundtrack can be encoded into this format by using a Dolby or Ultra-Stereo encoding matrix during the film's Printmastering session. DTS also has a process called DTS Stereo that can create a typical LCRS film soundtrack (check with DTS directly for more on their specific processes...).
Surround Sound formats beyond L-C-R-S:
Some of the surround sound encoding processes can create different, more complex soundtrack formats; Dolby SR/D and DTS, for example, can create six-track soundtracks for release, and Sony's SDDS is an eight-track format. In the case of six tracks, you get Left, Center, Right, Left Surround, Right Surround and a Sub-woofer channel (for enhanced low-frequency response). The split surrounds (as they are called) make it possible to move sounds around in the surround speakers, or to use stereo background sounds for even more impressive film soundtracks (Jurassic Park comes to mind, here). And if you heard Jurassic Park in a good THX theater with a DTS Digital soundtrack, you know what the sub-woofers are there for! That T-Rex really gave the sub woofers a run for their money, as well as Jeff Goldblum...Six-track sound reproduction has been with us for a while, since 70mm film releases have had the ability to deliver a six-track soundtrack that was magnetically encoded on the release print. This, unfortunately, was very expensive to produce, and problematic to control quality.
Sony's SDDS (Sony Dynamic Digital Sound) uses an eight-track delivery configuration that adds two speakers in between the Left/Center and Center/Right positions in the front speaker wall. Known variously as InterLeft, InterRight or LeftCenter and RightCenter, these channels allow for additional separation of music, effects and dialogue in the front speaker wall, while preserving the split surround format.
The Differences in Digital Sound delivery methods
The three digital systems (Dolby, DTS and SDDS) use proprietary methods to deliver the digital audio to the theater; two of these methods (Dolby, SDDS) encode the digital soundtrack onto the release print. DTS uses a different method, that of encoding a "timing stripe" onto the release print, and synchronizing a digital audio playback from an accompanying CD-ROM that carries the encoded soundtrack. In either case, the digital audio is reproduced in the theater with the same fidelity it was recorded at during the encoding process. This system neatly bypasses the traditional limitations of optical soundtracks: noise, bandwidth limitations, and headroom (transient peak) limits. Soundtracks sound cleaner, clearer and louder as a result. Please don't take this as a condemnation of optical soundtracks. A well-mixed movie can (and they still do) sound great with a well-produced optical soundtrack.
To summarize this difficult topic:
* THX specifies a set of standards that affect how sound is recorded and reproduced in a movie theater.
You get the benefits of the THX standard whenever you mix in a THX-certified mixing stage.
There is NO additional fee required.
You may display the THX logo in your film's credits if you sign a simple one-page form.
* Dolby Surround is a 4-channel optical surround format; this format is encoded in the optical soundtrack
You must license this format from Dolby Labs; There IS a license fee for this service
* Ultra-Stereo is a 4-channel optical surround format; this format is encoded in the optical soundtrack
You must license this format from Ultra-Stereo Labs; There IS a license fee for this service
* DTS is a 6-channel digitally-encoded surround format; this format is encoded on an external CD-ROM, but the timing and other information in encoded on the film release print;
You must license this format from Digital Theater Systems (DTS); There IS a license fee for this service
* Dolby Digital is a 6-channel digitally-encoded surround format; it is encoded on the film release print;
You must license this format from Dolby Labs; There IS a license fee for this service
* SDDS is an 8-channel digitally-encoded surround format; it is encoded on the film release print;
You must license this format from Sony Corporation - SDDS division; There IS a license fee for this service
I have got a video project - What's this DVD, AC-3?
relax - take a breath and we'll walk you through this...It's actually pretty simple;
Surround sound program on video materials are now released in a number of analog AND digital forms...
* Straight Left-Right Stereo program is still utilized a lot for Television, and Industrial formats...
* VHS Home video releases can be encoded in Dolby Surround (L,C,R,S), just like feature films;
* Laserdisc releases have also been using digitally encoded L,C,R,S surround formats, just like VHS
* NEW DIGITAL VIDEO RELEASE FORMATS have allowed for new DIGITAL SOUND FORMATS
o AC-3 - is a digitally-encoded surround sound format that is capable of reproducing six tracks of sound
+ Ac-3 actually refers to Dolby's Audio Compression 3 format used to compress the data
o DVD releases are also utilizing AC-3 digital sound format as well as traditional Surround Sound
All of these formats can easily be handled or prepared by a knowledgeable sound house. Please contact us if you have specific questions that you would like answered...no obligation, of course...
My mix sounded great on the mixing stage - but my print isn't in sync!
Well, we didn't say this would be EASY, just that we could help take some of the mystery out of it for you...You should IMMEDIATELY contact your post sound house and tell them what you've experienced. The Sound Supervisor on your show should be willing to take some time and help you sort this out. In the meantime, here's a few things that you can check on:
Some likely possibilities:
(1) If the Final Mix Printmaster has been transferred or copied, be sure the copy was done correctly. We have had experiences where a perfectly fine Printmaster was thrown out of sync because a copy was made first, and the optical shot from the copy;
(2) If the soundtrack DRIFTS from being in sync to gradually being more and more OUT of sync during the reel, suspect this possibility: If the Printmaster is on Multitrack tape, the SMPTE code on the tape could cause the optical soundtrack to drift in speed; If you mixed to VIDEO TAPE, a slight difference between 29.97 frame code and 30.00 frame code could throw you out of sync by many frames over 1000 film feet. If the soundtrack was shot on Mag, a mistake in running the film chain at video speed could cause the Mag to be "offspeed", just like the Multrack tape example above;
(3) If the Mag Printmaster was in sync when you reviewed the final mix, check to be sure the film lab didn't accidentally "misprint" the soundtrack by moving the optical negative a perf or two, or a frame or two when they married it to the picture. This can easily happen IF THE HEAD POP or TAIL POP is not EXACTLY CORRECT on your final Printmaster.
(4) If you printmastered in 2000-foot film reels, and FOR ANY REASON these reels were then separated and rejoined later, this poses a prime opportunity for sync to slip. If the beginning of a 2000 foot reel is in sync, and the last 1000 feet is suddenly (and consistently) out of sync until the end of the reel, suspect this phenomenon immediately.
(5) If one or two shots suddenly are out of sync but were IN sync when you mixed, ask yourself this: did you mix from an Avid or Lightworks (or other electronic edit system) output? If so, it's possible the film negative was not cut to the exact same shot length as the electronic output; Have you verified the length of all optical effects? If you have inserted optical effects, they may not have been counted exactly right, and you may have gained (or lost) a perf or frame or two in the effect; either way, your soundtrack will lose sync right then and there, and STAY out of sync for the rest of the reel (unless another optical effect error magically puts it back in sync again!)
(6) Finally, when all else fails, it is remotely possible that the optical negative might be offspeed. A quick call to the optical sound house will help them verify this for you.
My foreign distributor says I need an "Emenee" to make a sale ?
Actually, it's an "M and E" or "M&E". This element comprises the "MUSIC and EFFECTS" elements of your original soundtrack, with ALL of the English language dialogue and Walla removed to allow for foreign language dubbing. In most contemporary post sound packages, an "M&E" is allowed for in the original bid. This process requires preparation during the original sound editing, as well as some additional Foley coverage that might NOT be needed for a straight domestic release. If you NEED an M&E, be sure that you tell your post sound house that UP FRONT. It WILL add some dollars to your post bid, but you WILL want it, if you are to have any possibility of a foreign release or sale at all. Preparing this element NOW will buy you plenty of "peace of mind" later on. The M&E can be on Mag, on DA-88, on DAT, or on almost any format that can be synchronized. It DOES NOT need to be converted to an Optical soundtrack form at this time...only later, when a new foreign Printmaster is created after the foreign language has been added to it.
Do I need to know about the academy rolloff ?
Well, although it is a holdover from film sound's infancy, we need to be aware of it, since it does have some relevance in certain circumstances. The academy rolloff is a specific frequency response curve that is used in dubbing stages to simulate the effect that the old-time optical soundtrack would have on the frequency of the final soundtrack. With advances in technology in today's film industry, its use is diminishing, although it has been used on mono theatrical trailers to this day.
23rd April 2007
Lives for gear
Joined: Nov 2006
Location: Marin County, CA, USA
my Foley Pit has a 5.1 genelec system
You mix foley in 5.1? I don't get it.
It's all mixed in mono here. Edited in mono and re-recorded to LCR premixes with the verbs in the L and R.
24th April 2007
Joined: Apr 2007
Location: Puerto Rico
Wow, even though it's gonna take me months to really let all this info sink in, it's so worth it.
Thanks Georgia, keep up the good work, we are reading.
24th April 2007
Joined: Dec 2006
Location: NY NY
When I create my stems, I create the following on my console:
SFX 5.1 ( includes Foley and SFX )
And once in a while i'll put something in the rears from Foley... or center or where ever via the 5.1 stem. ( obviously I never use the .1 for Foley ) But I have used all the other channels...
..and, yes.. 90% of the time it's center channel, 9% of the time its L R and the 1% of the time in the rears...
but i'm kind of a perfectionist so go figure.... ( and a little anal at times )
24th April 2007
Lives for gear
Joined: Nov 2006
Location: Marin County, CA, USA
Right, but you mentioned a 5.1 system "in your foley pit" which had me confused.
Besides, I wasn't referring to actual dubbing mixing, but rather foley mixing (while recording). Seemed odd to me to have a 5.1 system in the control room of your foley stage. Maybe you use the same room for things other than shooting foley?
Oh, and a 5.1 stem for dialog? You must be re-mixing Children of Men.
Sorry, now I'm just winding you up.
25th April 2007
Joined: Dec 2006
Location: NY NY
oops... sorry... yes. we do other things in there as well..
Broadcast 5.1 mixes, Music stuff and the like...
4th May 2007
Joined: Dec 2006
Location: NY NY
Pink noise is played through each speaker or channel of a sound sound system. Then, equalization adjustments are made to each channel until the measurement of the pink noiseconforms to curve called the X curve. In the beginning, theatre loudspeakers suffered from both a limited frequency range as well as a poor frequency response. By implementing equalization, you can solve some of these issues. A room the size of a threatre will create acoustic anolomies. When equalized with pink noise for a flat response in a theatre, speakers deliver too much treble. The resulting sound is way too bright and a bit painful. The X curve was an attempt to normalize the sound in a large room.
Taken for an article I had lying around…my compliments to the author….
It was also noted that larger theatres would exhibit a somewhat steeper high frequency roll off, and that smaller theatres would exhibit a slightly reduced roll off of the high frequencies. This finding was officially noted in 1990. Beyond that, there have been few additional guidelines to aid technicians in the interpretation of these measurements and the equalization of cinema systems. Several years ago, the measurement system evolved with the use of four microphones placed around the auditorium to pickup the sound. While some have steadfastly defended this approach, in the final analysis it is no better than a single microphone pickup. Different, yes. But whether one uses a single microphone or four, by including all the reverberation, the resulting measurements are equally unreliable. While some have been critical of the way cinema sound systems are measured and equalized, I think the real disappointment is that as the loudspeakers have evolved, the methods employed to measure their behavior in theatres have not evolved far enough or quickly enough. Ioan Allen’s work of a quarter century ago was important and should not be understated. It represented a valuable component in Dolby’s efforts to introduce Dolby Stereo as well as improve cinema sound. It later became the basis for the SMPTE 202-M as well as the ISO-2969 motion picture audio standards. It also opened the door for many other improvements in all aspects of movie sound and paved the way for the introduction of wideband three-way loudspeakers as well as sound systems with a nine octave response, first introduced to movie theatres by my company in 1979.
In fairness, since the original work on the X curve was done with older theatre speakers having significant frequency response and frequency range limitations, it was impossible to glean further insights into what the shape of the curve might be with full-range highoutput loudspeakers in theatres of different sizes. Such speakers were unavailable at the time. That has now changed and a lot has been learned. Indeed, both Ioan and I have separately presented papers with similar findings on the varying shapes of the X curve.
The Real X Curve also shows that real-time measurements of the frequencies below 100 Hz, are also room dependent. While some theatres will exhibit a slightly rolled off bass region, many will show quite an elevated measurement in these frequencies. From this we see that the practice of automatically and artificially rolling off these lower frequencies, contributes to the lack of bass in many motion picture sound systems.
During the International Theatre Equipment Association technical seminars in 1999, Ioan Allen presented his own findings on the characteristics of real-time analyzer measurements of pink noise in theatres of different sizes. His presentation also included so-called waterfall charts showing how the shape of the pink noise measurement actually evolves as reverberation accumulates over time and results in response curves of varying shapes. The bass build up below 100 Hz is also seen in this graph that he has kindly provided for this article. He pointed out that the X curve itself “is a myth.” That is to say the high frequency roll off seen when measuring pink noise with real-time analyzers does not
indicate a roll off in the frequency response of the sound system. He reminded us that the roll off seen in such measurements is a result of the accumulated reverberation being included in the measurement. Now that the varying shapes of the X curve are more clearly understood, are we now fully prepared to equalize cinema sound systems to perfection? Well, not quite. We have a problem. Before we can properly equalize a sound system with pink noise, we need to know what the shape of the curve should be for the particular theatre we are in, when the response we actually hear with program material is flat. Determining that requires the use of screen speakers with a flat on-axis frequency response. Since most high frequency horns used in cinemas are the constant directivity type, with their own characteristic rolled off high frequency response, finding the correct place for the knee of the curve for a particular room is unlikely. Perhaps less difficult is knowing how the lowest frequencies should measure. The best way to handle the frequencies below 100 Hz is to adopt a what you see is what you get policy and do not equalize.
Another equally frustrating problem is the inability of the pink noise / real-time analyzer approach to accurately convey what is going on in the frequency range from about 100 to 400 Hz. For the sake of simplicity, my own Real X Curve chart does not show how these frequencies can sometimes measure at reduced amplitudes, rather than flat, in good sounding systems. In my experience, however, the actual shape of the frequency response depicted by an analyzer in these frequencies is not consistent from theatre to theatre, even though the sound systems involved may have the same tone. Furthermore, the way speakers behave in these frequencies can be influenced by the room.
How they should measure with pink noise is also room dependent. Sound systems tuned so that the analyzer shows a flat response between 100 and 400 Hz will often sound bloated, boomy or “honky,” while others will sound fine. There seems to be as many solutions to the challenges of tuning motion picture sound systems as there are technicians and authors who choose to write about them. The proof of the success of any technique is in the listening, however, not in the rhetoric. Those really interested in learning what works best merely need to stick their heads in the different rooms, setup different ways and hear for themselves. Fortunately, the differences are very evident, making judgments easy.
A new measurement system is needed. Whenever it arrives, the inventors will surely find themselves standing on the shoulders of Ioan Allen. Until we have a reliable method for measuring what something sounds like, it turns out that his original approach to the equalization of those older theatre speakers of the 1970s, remains the best solution to tuning a sound system. By comparing the sound heard from theatre speakers to a known high quality source, one can hear the difference and make adjustments accordingly. Since there are still no such measurement methods, we will need to rely on our ears for listening.
31st May 2007
Joined: Dec 2006
Location: NY NY
ISDN Primer...part 1
For those interested in ISDN and just getting onto delivering ADR/Dialogue/Mix/whatever via Telcom in realtime. Or as I like to say ISOCRONUSLY....
Integrated Services Digital Network (ISDN) is an international standard that defines a worldwide, completely digital switched telephone network. ISDN is designed to carry large amounts of information and has a number of potential uses, such as high-speed modem communications and desktop videoconferencing. For broadcast and professional audio, ISDN offers unique opportunities for the transmission of high-quality audio.
ISDN configurations: The form of ISDN of most interest to broadcasters and audio professionals is Basic Rate Interface, or BRI. (In Europe, this service is called S0.) On a single pair of ordinary phone wires, BRI offers two "bearer" channels at a 64kbps transmission rate and one "data" channel at 16kbps. This configuration is often referred to as 2B+D. When ISDN BRI is installed in your facility, each line is brought in on only one pair of wires.
ISDN is full duplex and calls are dialed and routed just like analog calls. The Telos Zephyr uses the two "B" channels for bidirectional audio (transmitted as digital data), ancillary RS-232 data, and inter-unit signaling. The "D" channel is reserved exclusively for telephone network signaling.
There is also ISDN Primary Rate Interface (PRI), called S2M in Europe. In the Western Hemisphere, PRI offers 23 "B" channels and one "D" channel. In Europe and Asia, this service offers 30 "B" channels and one "D" channel. ( While the Zephyr does not support PRI directly, special equipment or a PBX switch can break a PRI into multiple BRI's.)
ISDN Basic Rate Interface (BRI) is often called "2B+D", referring to its three duplex channels.
ISDN availability and cost: From Germany and Portugal to Hong Kong and Singapore, ISDN lines are used extensively. The opportunities for connectivity between countries is increasing, and international calls can be accomplished effortlessly. In the US, the Regional Bell Operating Companies are gradually making ISDN available in more localities. Currently, there is better than 85% coverage, and by the end of 1996, ISDN should be available in nearly all of the US. Most long-distance ISDN connections within the US are 64kbps, with a few connections limited to the old 56kbps standard. Zephyr can operate at both rates with the same superior performance. In most countries, the monthly maintenance fees and per call charges for each "B" channel approximate the costs of an ordinary analog business line. Generally, ISDN costs significantly less than dedicated lines. And an ISDN "B" channel can be used as a standard analog phone line, using a special ISDN telephone (or a POTS terminal adaptor and a regular phone), when full bandwidth is not required.
ISDN versus other services: Switched 56, as the name implies, has only one channel at 56kbps and is often available in US locations where ISDN service has not yet been implemented. An ISDN number can call a Switched 56 number and vice versa. When used with an external CSU/DSU, the Telos Zephyr works flawlessly with Switched 56. In addition, Zephyr, using its V.35/X.21 data port and an external CSU/DSU, can also transmit high-quality audio over fractional T-1, DDS (dedicated digital service), or any similar service.
For point-to-point audio delivery, ISDN has advantages over satellite. ISDN eliminates the inflexibility of reserved satellite time. ISDN is fully two-way, and startup hardware costs are significantly lower. Overall, ISDN has significant advantages for most occasional and point-to-point feeds and offers economical and reliable backup to your satellite system. While satellite is still a viable choice for full-time, one-way, point-to-multipoint transmission, ISDN BRI and PRI can be a more flexible and economical option for moderate-sized networks. With ISDN cost declining in many locations, larger networks may find it a very appealing alternative to satellite distribution.
Like a modem is used with analog data communications, a terminal adapter must be used to interface the data from your codec to the telephone network's fully digital ISDN connections. Our Telos Zephyr makes connection to ISDN easy by incorporating both a codec and a terminal adapter into a single integrated unit. In the past, you needed to connect external terminal adapters to your codecs. Unlike other manufacturers who buy ISDN terminal adapters to build into their codecs, we developed our own. This enables us to enhance the feature set for broadcast users and create a unified, easy-to-use set of controls for both the codec and terminal adapter functions.
Alternate Mark Inversion. A T1 line coding method. This is the older of the two commonly available. See Line Coding, T1. See Also B8ZS.
Automatic Number Identification- A system, originally designed for use by Interexchange carriers (IEC’s) which transmits the "billed party number" along with a call. Note that the billed party number is not necessarily the number of the line placing the call. ANI predates SS7 and can operate in with analog as well as digital trunks. See also CLID.
A form of serial data communication which is not clocked. To keep the bit stream synchronized start and stop bits are used. RS-232 computer data is commonly asynchronous data. In contrast to synchronous data.
Bipolar 8 (with) Zero Substitution. A T1 line coding method. This is the more modern line coding method of the two commonly available. See Line Coding, T1. See also AMI
Bearer Channel. One of the multiple user channels on an ISDN circuit. Used to carry user's data; i.e. coded audio data in the case of Zephyr or ZephyrExpress.
Behind the PBX
This is our own definition and refers to when one privately owned phone system is tied to another privately owned phone system. It is a limited Tandem application. See Tandem Switch and Tandem Tie Trunk Switching below.
The basic research facility which was AT&T's primary research facility. Bell Labs was spun off with Lucent Technologies. Many very important discoveries have been made at Bell Labs including the transistor, communications theory, and radio astronomy.
BELL COmmunications REsearch. See Telcordia. The research and development organization owned by the RBOCs. Bellcore represents the RBOCs in developing standards for Telco equipment and in testing equipment compliance to those standards. Bellcore also offers educational and training programs open to all interested parties.
1) Bit Error Rate Test- A test for digital lines which involves looping a data path and sending a test pattern. Data returning is compared to the sent data to check for errors. Depending on the "Test Pattern" used, BERTs may or may not uncover problems. A line which only has occasional problems will need a BERT of sufficient time duration to catch that intermittent problem. A five minute BERT of an ISDN BRI circuit will only catch severe problems.
2) A Bit Error Rate Tester. The test equipment used to perform a Bit Error Rate Test.
Billing Telephone Number
The main phone number which all calls on a PRI are billed to. This information is required when configuring a PRI PBX.
The capacity of a digital channel. ISDN calls are set up at a given bit rate, either 64Kbps or 56Kbps. The bit rate cannot be changed during a call. See Kbps.
When a circuit switched call cannot be completed. The percentage of blocked calls to the number of calls attempted forms the basis of a statistic called "grade of service". While it is economically infeasible to build a network which would have no blocking, the phone company are held accountable by the utility commissions to keep blocking below tariffed levels. The concept of blocking cannot be applied to packet networks, only circuit switched networks.
ISDN Basic Rate Interface. The common form of ISDN with 2 Bearer Channels and one D Channel. All three channels are on a single copper pair and encoded with type 2B1Q coding.
Basic Rate Interface Transmission Extension. A technology where ordinary T-1 trunks (or any other digital carrier system) are used to extend ISDN service. See also Repeater.
The part of the phone company where you call if they mess up your bill, to report problems, and to order service. Not necessarily technically literate.
Called Party Address
This is the destination phone number of a call delivered to a switch. For instance this could be the CLID of a call delivered to a PBX using DID or two-way trunks. See also DID.
Common Channel Interoffice Signaling. A signaling system where network information such as address and routing information are handled externally to the actual communications (voice) path. SS7 (Signaling System 7) is the internationally standardized CCIS system. Deployment of CCIS increased efficiency since no communications (voice) channels are used merely to report an "all trunks busy" or "far end busy" conditions. It also decreased toll fraud substantially since it removed the potential for access to the signaling information that was inherent to in-band signaling schemes. CCIS also enables CLASS features as well as sophisticated re-routing features for "intelligent network" applications. See also In Band Signaling.
An actual path you can talk or send data over. This is what you are paying the phone company for. For instance, ISDN BRI lines can be ordered with 1 or 2 active channels and these channels can be configured for voice calls (CSV), data calls (CSD) or both (alternate CSD CSV). A channel does not necessarily have it’s own unique telephone number. See ISDN.
A telephone exchange which is assigned to Radio and TV stations, Promoters, and other users which will be receiving large numbers of simultaneous calls. The idea is to group all of these users on a single exchange so when all routes into that exchange are in use "normal" users (on other exchanges) will not experience blocking of incoming or outgoing calls. Trunks from other local exchanges into the choke exchange are deliberately limited to just a few paths so callers will get an "all trunks busy" instead of completely blocking their local exchange. However, when one of the choke exchange users experiences a large number of calls (as when your station runs a contest) the other choke exchange users will be blocked because all trunks into the choke exchange will be busy. See Blocking and Concentration
A physical path through which electrical signals can pass. It consists of a network of conductors and other components, separated by insulators. Technically this term cannot be applied to fiber optic or other "non-metallic" paths.
A system where a dedicated channel is allocated to the users of that call for the duration of that call. That channel is allocated for the duration of the call regardless if information is being transmitted at any given moment. Bandwidth through the channel is fixed, at no time may this bandwidth be exceeded. If this bandwidth is not used it is wasted. While inherently inefficient, the dependable and reliable nature of circuit switching makes it ideally suited to real-time voice and audio/video conferencing applications. This is in stark contrast to systems where statistical multiplexing is used. See Statistical Multiplexing
Custom Local Area Signaling Services. A variety of enhanced features (usually on analog lines) that take advantage of the ability of modern SS7 technology’s ability to transmit information about the calling party. CLASS includes such features as Caller ID, Automatic Callback, Call Trace (initiated by subscriber), Selective Call Rejection, etc.
Competitive Local Exchange Carrier. Your local telephone service provider who is one of the new-generation providers rather than a RBOC or Independent. A CLEC is really just an Independent, albeit one formed after the divestiture of AT&T. See LEC.
Calling Line Identification. This is the ISDN and SS7 equivalent of Caller ID; I.E. the number of the calling party. See also ANI.
Central Office. The Telco facility where your local telephone circuit leads to. Contains Switches and Trunks as well as the local telephone circuits.
COder/DECoder. A device which takes digitized audio and "codes" it in order to reduce the transmission bit rate and which can also simultaneously "decode" such coded audio. Strictly speaking, a codec does not include an ISDN terminal adaptor and related equipment.
A trunk (channel) which can both make and receive calls. This generally refers to analog ground start or loop start trunks, although the term can be applied to ISDN BRI or PRI channels as well. Each combination trunk normally has a telephone number, although they are frequently part of a hunt group and only one number may be published for that group. Also called a Both Way Trunk. This is not the same as a Two-way DID trunk. See DID Trunk, Hunt Group and Trunk
The basic premise is to share facilities wherever possible. For instance, while there may be thousands of customers served by a given Central Office, there will be substantially less than that number of calls which can be handled simultaneously. And even fewer long distance calls can be made simultaneously. The art of Traffic Engineering is to have enough capability that calls are rarely blocked, but not any more than that. See also Choke Exchange and Blocking.
Customer Premise Equipment- Customer owned equipment located at his/her facility. In the USA and Canada the NT1 is part of the CPE.
Circuit Switched Data- A dial-up data communications channel which, once established, looks like a transparent data pipe. Also, the type of ISDN service required to utilize this capability of an ISDN circuit. In contrast to CSV.
Channel Service Unit. The NCTE used in the USA & Canada to terminate a T1 line. Typically the CSU must be provided by the end user. See NCTE.
A device which incorporates the functions of a CSU (Channel Service Unit) and a DSU (Data Service Unit) and interfaces between a Switched-56 (or "Dedicated Digital Service") line and a user's data equipment such as the Zephyr.
Circuit Switched Voice- A dial-up communications circuit for voice grade communication. Also, the type of ISDN service required to use this capability of an ISDN circuit. In contrast to CSD.
An ISDN protocol which pre-dates National ISDN-1. In most cases National ISDN-1 is also available. The Northern Telcom DMS-100 switch supports "Custom DMS ISDN". The AT&T/Lucent 5ESS switch supports "Custom Point-to-Point" (PTP) and Custom Point-to-MultiPoint (PMP). The ISDN protocol has no relation to where one may call. The Telos Zephyr and TWOx12 do not support PMP.
Data Channel or Delta Channel (depending on who you ask). The channel which handles ISDN network related data between the user's equipment and the Telco switch. Used to carry data to set up calls and receive calls. Some Telco's also allow users to use the D channel to access the packet data network, with appropriate terminal equipment.
Data Communication Equipment. When using serial communications such RS-232, V.35, or X.21, the DCE is the device sending/receiving from the Telco line. ie: a modem or CSU/DSU. In contrast to DTE.
A permanent channel between two locations. As opposed to a Switched Circuit
Dedicated Digital Service
A "Hardwired" or "Nailed Up" digital circuit which is permanently connected between 2 points. Typically 56Kbps or 64Kbps. Dedicated digital lines are frequently cheaper than ISDN for full time service. Also called Digital Data System, or DDS.
Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without going through an attendant or auto-attendant.
DID Extension or DID Station
A specific phone within a PBX which can be called from the public telephone network without going through an attendant or auto-attendant.
A phone number used to route calls from the telephone network to a specific phone in a PBX (the DID extension). DID requires special DID trunks or ISDN PRI "two-way DID" trunks. Blocks of DID numbers (typically 10 or 20) are purchased from the LEC or CLEC for use on the PBX. The number of DID numbers usually substantially exceeds the number of trunks in the system.
A Direct Inward Dialing Trunk. A trunk (channel) which can only receive calls. A group of telephone numbers (DID numbers) are associated with a given trunk group, however there is no one-to-one correspondence between the individual channels and these numbers. The PBX uses the DID number given it by the phone company to route the channel to the correct DID extension within the PBX extension. This allows some or all PBX stations to receive calls directly without going through an attendant (or auto attendant) Note that there are almost always more DID numbers than there are DID trunks. See DID Number and DID Extension.
Your seven digit telephone number (without the area code), as found in the telephone directory. Zephyr generally does not need you to enter the directory number as part of installation.
Directory 1&2 (Zephyr)
The Utility menu on the Zephyr where the 7 digit Directory Numbers can be entered during set up. The Zephyr generally does not require these numbers.
Dialed Number Identification Service- A service, typically offered by a long distance company on 800 lines, that provides the number dialed by the caller. This allows a caller to receive specific treatment depending on the number dialed.
Digital Subscriber Line. Typically refers to an ISDN line or a T-1 line, although the term is also frequently used to mean the next generation beyond ISDN. Sometimes xDSL is used to indicate that the writer is referring to any of a number of emerging DSL technologies.
Data Service Unit. See CSU/DSU.
Data Terminal Equipment- When using serial communications such RS-232, V.35, or X.21, the DTE is the device sending/receiving from a modem or CSU/DSU. In contrast to DCE.
Dual Tone Multi Frequency. The standard tone-pairs used on telephone terminals for dialing using in-band signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals support only 12 of them (0-9, * and #). These are also sometimes referred to as "Touch Tones" (actually a copyrighted trade name held by AT&T). Note that while digital data terminals have the same symbols, ISDN uses "common channel signaling" (over the D channel) and therefore does not necessarily generate any tones at all. However many terminals still generate the tones since they will still be used on occasion to access services (such as voicemail or automated attendant) at the far end using in-band tones.
A common type of digital telephone trunk widely deployed outside the US and Canada. Has 30 available 64Kbps channels (called DSø ’s) plus a B channel and a sync/control channel.
Extended Superframe. A type of Line format supported on T1 circuits. The Telco determines the line format and line encoding of your line. See Line Format
The pan European ISDN protocol standardized by ETSI. This protocol is used throughout Europe and has been adopted in many other countries outside the USA & Canada. See MSN
31st May 2007
Joined: Dec 2006
Location: NY NY
ISDN primer ... part 2
Another name for a Central Office. See CO
A circuit path using separate pairs for send and receive. This term is also used when referring to digital channels that inherently have discrete send and receive paths, regardless of the number of pairs (or other media) used.
A unit of data which is defined by the specific communications protocol used. See Line Format, T1.
On a POTS line an incoming call is signaled by periodically applying an AC ring voltage to the line. Since there is a semi random period before the ring, and pauses between rings, it is possible to seize a line which is "about to ring" (and answer a call) when attempting to place an outgoing call. When this scenario happens it is called glare. Glare is much less likely if Ground Start trunks are used. See Ground Start Trunk.
Grade Of Service
This is simply the ratio of calls blocked to total calls in a decimal form. So a grade of service of P.08 would represent 8% blocking. Telephone tariffs regulate the acceptable average grade of service which must be provided on public networks.
Ground Start Trunk
A type of telephone trunk where the request to make an outgoing call (i.e. request for dial tone) is made by briefly grounding the Tip conductor. Many PBX systems use ground start trunks as they are less prone to glare than Loop start trunks. See Loop Start Trunk. See also Glare
High Density Bipolar 3. An E1 line coding method. This is the more modern line coding method of the two commonly available. See Line Coding, T1. See also AMI and B8ZS
A group of telephone channels configured so that if the first is busy (engaged) the call goes to the next channel, if that channel is busy it goes to the next channel, etc. Hunt groups may hunt from the highest to the lowest, the lowest to the highest, or on some other arbitrary pattern. But the order of hunting will usually be fixed, beginning with one channel and working through ("hunting") until an unused channel is found. The term may have originated back in the old manual switchboard days when the operator literally hunted for an unused jack to plug a cord into. This arrangement is very common in business scenarios where a single incoming number (the Listed Directory Number) is given to the public, but multiple incoming channels are supported. See LDN
A device which converts from a two-wire signal such as POTS lines to a four-wire system (separate send and receive paths) such as used in the pro-audio world. While this task is theoretically quite simple, the fact the impedance of most phone lines varies widely across frequency complicates matters. The Telos 10 telephone system was the first practical DSP (Digital Signal Processing) based hybrid and applied the then brand-new technology to this problem
InterExchange Carrier. "Long Distance" carrier. Handles Interlata and interstate calls. Also referred to as IXC.
Incumbent Local Exchange Carrier. A local Exchange Carrier which entered the marketplace before the enactment of the 1996 Telecom act;. i.e. a telephone company which is neither an Indie nor an RBOC. See LEC and CLEC.
In Band Signaling
A signaling system where network information such as address and routing information are handled over the communications (voice) path itself. Usually the information is represented in the form of tones, however DC loop current signaling also qualifies as In Band Signaling. See also CCIS.
Any of the phone companies in existence at the time of divestiture that were not affiliated with the Bell System. See RBOC, LEC, and CLEC
A vendor of telecommunications Customer Premise Equipment (CPE) other than a Bell Operating Company (RBOC) or AT&T. This term was originated by AT&T and was meant to be derisive towards the fledgling industry when the courts said it was OK for end users to buy equipment from someone other than the Bell System. This industry flourished, in spite of AT&T’s disdain, and ironically the RBOCs were not allowed to sell CPE under the terms of the break up of AT&T. With the current state of deregulation, the RBOCs are slowly re-entering this business.
A method of combining and later separating 2 data streams that does not involve the telephone network. Inverse multiplexing is the most common way of achieving this function in Codecs operating at 128kbps or less over ISDN.
IOC Capability Packages
ISDN Ordering Code system. This system was devised by the National ISDN User’s Forum and Bellcore to simplify ordering new ISDN lines in the USA and Canada. Using a single code specifies all line specifications. The Telos Zephyr, ZephyrExpress, TWO, and TWOx12 support IOC package "S". Search for document NIUF 428-94 at NIUF’s web page for more information.
Integrated Services Digital Network- A relatively new and highly flexible type of telephone service which allows dialing on digital lines with multiple bi-directional "Bearer" channels each with a capacity of 56 or 64 Kbps and a single bi-directional "D channel". See BRI and PRI.
The "language" used for communication between the Telco’s switch and the customer’s Terminal Adapter. Each ISDN circuit has one protocol, and the protocol has no effect on where or whom one may call. See ETS 300, National ISDN, and Custom ISDN.
A term used in Europe for ISDN BRI. See BRI.
A term used in Europe for ISDN PRI. See PRI.
Incoming Service Grouping. See Hunt Group
IntereXchange Carrier- See IEC.
KiloBits Per Second. Measure of digital channel capacity.
Key Telephone System
A system that allows multiple telephones to share multiple pre-determined telephone lines. The system provides indicators to allow the users to understand the status of each line available on a given phone. It is up to the user to provide the intelligence to select an unused line, or answer a ringing line, for example. See also PBX.
Local Access and Transport Area. The area within which calls are routed by your Local Exchange Carrier (LEC). Under the divestiture of the Bell System calls going outside of this area must be handled by an Interexchange carrier (IEC). With the latest round of de-regulation the usual IEC companies are being allowed to compete in the IntraLATA long distance market and LECs are beginning to be permitted to handle InterLATA calls.
Listed Directory Number. When a number of Telco channels share the same hunt group it is customary to give out only one phone number for the group, although generally each channel will have its own number. The number given out is the "Listed Directory Number" since that is the number that would be listed in the Telephone Directory and given to customers. See also Hunt Group.
Local Exchange Carrier. Your local telephone service provider which is either an RBOC or an Independent. In other words, a traditional phone company. In contrast to CLEC or IEC.
An electrical connection between a telephone service provider’s switch (LEC or CLEC) and a telephone terminal or Key system. An electrical connection between a telephone service provider’s switch and another switch is called a trunk. Note that some type of physical lines offer more than one channel. I.E. a BRI circuit has 2 channels, called B channels. This term is a confusing one. See Channel. See also Station Line
The circuit in the Telco switch to which your line is connected. On an ISDN circuit the line card performs a role analogous to the NT1 in adapting and equalizing the circuit.
Line Coding, T1
The clock signal for T1 is derived at the far end from the data bits themselves. Therefore T1 lines have certain restrictions as to the data allowed. No more than 15 zeros shall be sent in a row; and average density of 12.5% ones must be maintained. The CSU is responsible to ensure that these requirements are met. The line encoding method, AMI or B8ZS determines exactly how these requirements are met while still allowing recovery of the original data at the far end. Your Telco will determine the method used on a specific circuit. B8ZS is preferred. E1 circuits have similar restrictions. HDB3 is preferred for E1 circuits.
Line Format, T1
Modern T1 circuits usually use either Superframe (sometimes called SF or D4) or Extended Superframe (sometimes called ESF) line formatting. The type of framing used is determined by your Telco. ESF is preferred. See ESF and SF
If your local Telco is a former Bell Operating Company then any call outside of your LATA or any Interstate call is considered long distance and is handled by an IEC.
If your local phone company is an independent then only interstate and inter-phone company calls are considered long distance. The above is true regardless of whether you are referring to a dedicated line or a dial up call.
However, under the current state of deregulation, toll calls within a LATA may now be covered by the IEC, and in some cases RBOCs are being permitted to handle InterLATA calls.
The telephone circuit from the CO to the customer's premises. Generally refers to a copper cable circuit.
Process of actually measuring the loss on a prospective ISDN line to see if it can be used for ISDN service . The actual loss on the line determines whether ISDN service can be offered without a repeater. Generally ISDN is available up to 18,000 feet from the serving Central Office. It may not be available within this range, or may be available further from the CO. Only a loop qualification can tell for sure. Not all Telcos will extend ISDN lines with Repeaters.
Loop Start Line
A plain old telephone line. The telephone terminal signals the "off hook" condition by allowing DC current to flow. See Ground Start Trunk. See also Glare.
Loop Start Trunk
A plain old telephone line connected to a PBX switch. See Loop Start Line. The PBX signals the "off hook" condition by allowing DC current to flow. Ground Start Trunks are generally preferred for use on PBXs to prevent glare. See Ground Start Trunk. See also Glare.
Company which now makes the former AT&T 5ESS switch, as well as various other pieces of Telco gear and semiconductors. Lucent was split off from AT&T in 1996 and owns Bell Labs. As of approximately 1999 Lucent sold their telephone set manufacturing business (and the right to use the AT&T name on telephone sets) to V-Tech. V-Tech is using the AT&T name and line as their high-end line.
Multiple Subscriber Number. This is a telephone number associated with an ETS 300 BRI line. Providers of ETS 300 often give you three MSNs with a BRI, although additional MSNs can be purchased. An ISDN terminal will "ring" (provide an alerting signal) only when calls are made to the MSN (or MSNs) entered in that terminal. If a terminal has no MSNs entered it will "ring" whenever there is a call to any of the MSN’s on that BRI. See ETS 300 and Directory Number.
The first US "standardized" multi-platform ISDN protocol. The first version is National ISDN-1. As of mid 1996 National ISDN-2 has been implemented in some areas and is fully backward compatible with National ISDN-1.
Network Channel Terminating Equipment. NCTE is a general term that can be applied to a CSU or NT1 or other equipment terminating a digital line at the customer’s premises. In many countries the NCTE is provided by the Telco. The USA is not one of those countries.
National ISDN User’s Forum. A user’s group formed under the National Institute of Technology (NIST) in the USA. The NIUF is a neutral forum where the switch manufacturers and Telcos can get input from users and CPE manufacturers regarding the implementation of National ISDN. Among NIUF’s successful projects have been the IOC ordering codes. Their web page is at National Institute of Standards and Technology
. See National ISDN.
Northern Telecom (previously Northern Electric). The Canadian company which was once the manufacturing arm of Bell Canada. Manufacturer of the DMS-100 switch as well as many other pieces of telecom equipment.
"Network Termination Type 1". The termination at the customer premises of an ISDN BRI circuit. The NT1 performs the role of line termination of the "U" interface and Code/Decodes from the line's 2B1Q coding scheme. The customer end of the NT1 interfaces using the "S" or "T" interface. The NT1 is frequently part of the "Terminal Adapter" and is built-in to Zephyr, ZephyrExpress, Telos TWO and TWOx12 systems sold in the USA & Canada.
Private Branch Exchange. A privately owned switch. Basically a PBX is a private "business" telephone system which also interfaces to the telephone network. Many PBX's can now offer ISDN BRI service, usually over the S Interface. A few vendors are now offering BRI over the U interface as well. Be wary of these ISDN protocols since they have not been as well tested. They may or may not work with a given piece of CPE.
Primary Interexchange Carrier. This is your default "1+" carrier used for interLATA calls. In some areas you may have two PICs, one for interLATA calls, and one for intraLATA long distance calls (in which case it stands for Primary Intraexchange Carrier). In some areas intraLATA long distance calls are still handled by your RBOC, in others you now have a choice.
AT&T "Custom Point to Point" Custom ISDN Protocol. Not supported by the Zephyr or ZephyrExpress. See Custom ISDN and ISDN Protocol.
Point Of Presence. The local facility where your IEC maintains a switch. This is where your long distance calls get routed so that your IEC can handle them.
This is a pretty general term. Newton’s Telecom Dictionary 10th edition defines a port as "An entrance to or an exit from a network". Many phone equipment vendors refer to ports as the physical interface between a Switch and a Line or Trunk. Product literature often refers to the number of ports on a phone system. In this context it refers to the number of phone or lines (or sometimes the combination) the system supports.
Plain Old Telephone. A black, rotary-dial desk phone. Usually a Western Electric model 500 set. Outdated term.
Plain Old Telephone Service. Regular old-fashioned analog loop start phone service.
Primary Rate Interface. A form of ISDN with 23 "B Channels" and one "D channel." All 24 channels are on a single cable. Functionally related to T1 telephone circuits. In Europe PRI has 30 "B Channels" and one "D Channel".
The act of configuring an ISDN line. Also refers to the complete line configuration information.
Power provided on pins 7 and 8 of the "S" interface cable. This power is used so that an NT-1 can provide power to a terminal (usually a phone). In some cases it is used to allow a terminal to power an NT1. The U.S. versions of the Zephyr, ZephyrExpress, and Telos TWO supply PS2 power in the "S" jack. This power arrangement is also used in the Telos TWOx12 and 2101 to power Desktop Directors.
AT&T "Custom Point to Point" Custom ISDN Protocol. Point-to-Point lines have only one incoming phone number which must be dialed twice to connect to both lines. See Custom ISDN and ISDN Protocol.
Regional Bell Operating Company. One of the regional companies formed when AT&T got out of the local telephone business. Each RBOC (or "baby bell") owns a number of the former "Bell Operating Companies". The Bell Operating Companies are the traditional local phone companies (pre-1984), except where one’s service is from an "Independent" (non bell) telephone company or a CLEC. Due to their former association with the Bell System RBOCs are regulated by the FCC differently than are independent Telcos. In many cases the Bell Operating Company structure is no longer used. For instance, here in Ohio we now deal directly with the RBOC, Ameritech, while the old Bell Operating Company, Ohio Bell Telephone, no longer exists. Another trend is mergers among the RBOCs (and in some cases the independents as well). See CLEC and LEC.
Receive Data. Data coming from the network, or DCE towards the DTE. Also, a light on a modem or CSU/DSU which lights to indicate presence of this signal.
Regional Bell Operating Company
A device intended to extend ISDN telephone service to site further from the central office than could normally be served. i.e.: beyond 18,000 feet. ISDN repeater technologies include "BRITE", "Virtual ISDN", "Lightspan", and "Totalreach". Some Telcos do not use repeaters. Compatibility between a given piece of CPE and a repeater is less certain than if that CPE were directly connected to the switch.
Robbed Bit Signaling
A signaling scheme that "borrows" bits on each T1 channel for use as signaling channels. On SF T1's there are two bits, the A bit and the B bit in each direction. On ESF T1's there is also a C and D bit in each direction, although they are rarely used. Using these bits, various older analog trunk interfaces can be emulated over a T1. For instance, address signaling using 10 pulse per second (rotary style) digit groups over these bits. Since robbed bit signaling interferes with the least significant bit, only 7 bits can be used for sensitive data applications, leaving only a 56kbps channel for data applications.
See Hunt Group
The electrical interface between the NT-1 and the Terminal Adapter or other ISDN equipment. ISDN equipment with built-in NT1’s do not necessarily provide access to the S interface (the Zephyr, ZephyrExpress and Telos TWO do). Multiple devices can share an NT-1 by connecting on the S interface. Also known as the S passive bus.
European term for ISDN BRI. See BRI and ISDN 2.
European term for ISDN PRI. See PRI and ISDN 30.
Unlike telegraphy, teletypewriter and POTS lines, most digital lines (such as ISDN) use a voltage rather than current mode of operation. Sealing Current allows a controlled amount of current to be passed through a telecom circuit for purposes of "healing" damage caused by corrosion. Bellcore specifies sealing current on the ISDN U interface.
Superframe. A type of Line format supported on T1 circuits. The Telco determines the line format and line encoding of your line. ESF is the preferred Line Format on T1 circuits. See Line Format.
See Statistical Multiplexing
31st May 2007
Joined: Dec 2006
Location: NY NY
ISDN primer... Part 3
A Subscriber Loop Carrier Circuit system manufactured by AT&T (now Lucent). SLC-96 has its own version of T1 framing between it and the CO. This interface is the Bellcore TR-008 or the newer GR-303 interface which are specialized versions of T1 intended to allow transparent transport of analog features such as Caller ID, Call Waiting, etc.
Subscriber Line Interface Circuit, see Line Card.
The equipment used with the AT&T (Lucent) SLCC Subscriber Loop Carrier Circuit, a system used to multiplex a number of subscriber loops onto a single circuit to reduce fixed costs.
Service Profile IDentifier- On the "National ISDN", "AT&T Custom PMP" and "Custom DMS" ISDN protocols, the Telco switch must receive correct SPID(s) from the CPE before it will allow access to ISDN service. Intended to allow multiple configurations on ISDN lines shared among different types of CPE equipment. While your SPID may include your area code and telephone number, the SPID is distinct from the telephone number. For the National ISDN , Custom PMP, and DMS custom ISDN protocols the Zephyr requires the user to program SPIDs into it. Custom PTP and ETS 300 protocols do not require a SPID.
A telephone circuit from a PBX to a telephone on that PBX. Since this is a telephone-to-switch connection it is considered to be a "line". See Line and Trunk.
A method of improving effective bandwidth of a Telco channel. Statistical Multiplexing takes advantage that there are typically many pauses in a conversation. By taking advantage of this fact, and not sending the pauses, improvements in efficiency can be made. Also referred to as silence suppression. See Circuit Switched.
The customer of a Telecommunications company. This term dates back to when a local Telephone Company was formed at the specific request of a group of customers who agreed in advance to "subscribe" to the service.
Telephone company switching device which "makes the connection" when you place a call. Modern switches are specialized computers. ISDN service is provided from a "Digital" switch, most commonly an AT&T model "5ESS", Northern Telcom model "DMS-100", or Siemens model "EWSD". The switch, and related software running on it, will determine which ISDN protocol(s) will be available to customers connected to it. See also PBX.
(ZephyrExpress)- The utility menu item where the ISDN Protocol is selected. See Telco Setting and National ISDN.
A channel which is not permanent in nature, but is connected through a switching device of some kind. The switching device allows a switched circuit to access many other switched circuits (the usual "dial up" type of telephone channels). Once the connection is made however, the complete capacity of the channel is available for use. As opposed to a Dedicated Circuit.
A type of digital telephone service developed in the mid-1980's which allows dialing on a single 56Kbps line. Each Switched-56 circuit has 1 or 2 copper wire-pairs associated with it. Switched-56 is being rapidly replaced with ISDN, which is cheaper and more flexible. See also CSU/DSU.
A form of serial data which uses a clock signal to synchronize the bit stream. SInce, unlike Asynchronous Data, no start and stop bits are use, data throughput is higher than with asynchronous data. ISDN uses Synchronous data.
A proprietary Telos 2.048mbps channelized link. This link uses the DSX-1 electrical protocol and has 30 channels at 64kbps each.
A common type of digital telephone carrier widely deployed within the US, Canada, and Japan. Has 24 64Kbps channels (called DSÆ ’s). The most common framing scheme for T1 "robs" bits for signaling leaving 56kbps per channel available.
The electronic interface between an ISDN device and the NT1. The terminal adaptor handles the dialing functions and interfaces to the user's data equipment as well as to the NT1 on the "S" or "T" interface.
A switch which is between two others. It connects two trunks together. Long distance calls on a LEC line go through a long distance tandem that passes them through to the long distance provider’s switch.
Tandem Tie Trunk Switching
When a PBX switch allows a Tie Line call to dial out of the switch. For example, if switch "A" in Arkansas has a tie line to switch "B" in Boise, Boise could use the tie line to make calls from switch "A".
Transmit Data. Data coming from the DTE towards the DCE or network. Also, a light on a modem or CSU/DSU which lights to indicate presence of this signal.
Telephone Company. Your local telephone service provider. In the 21st century you generally have a choice of Telcos if you are a business in a major metropolitan area in the USA. Competition is coming to the Telecom industry around the world.
(Zephyr, TWO, TWOx12)- The menu selection where the ISDN protocol is selected. Choices are Natl I-1 (for National ISDN and DMS-100 Custom Functional ISDN), AT&T Cust or PTP (for AT&T Custom Point-to-Point), and ETS300 (for Euro-ISDN). See also Switch Type.
Formerly BellCore. The research and development organization owned by the telephone companies. Telcordia represents the phone companies in developing standards for Telco equipment and in testing equipment compliance to those standards. Telcordia also offers educational and training programs open to all interested parties. BellCore was sold to SAIC in 1997. Telcordia is responsive to both RBOCs and independent Telcos. Their web site is Telcordia Technologies - OSS, network engineering and consulting telecommunications solutions
A Trunk between two PBXs. Note, a tie line is a dedicated circuit, not a switched circuit. See Trunk
A telco "work order" used to track Customer Repairs within the Telco. If you call someone "inside" the telco's repair department they will need this number to proceed. It will also be needed whenever you call to check on the status of a repair. Always ask for this number when initiating a repair request.
A communications path between two switching systems. Note that many trunks may be on a single Circuit (if that circuit has multiple Channels). The trunks most users will deal with are between the Telco switch and a PBX. However, a Tie Trunk can connect two PBXs.
A number of telephone channels which are functionally related. Most common is the Hunt Group. Other common types include Incoming Trunk Groups and Outgoing Trunk Groups. See also Combination Trunks and Two-way DID Trunks.
Two-way DID Trunk
An ISDN PRI (or T1) line equipped for direct inward dialing. Most trunks are related to a given phone number, either alone or as part of a hunt group. In the case of a "normal" (ie analog) DID Trunk a group of phone numbers are associated with that DID trunk (or group of trunks) and incoming calls include the DID Number, so the PBX can route that call to the correct DID Extension. This is exactly how ISDN PRI functions, with the DID information coming in over the D Channel. There is a big difference between a normal DID Trunk and a Two-way DID trunk over ISDN PRI. For one thing, ISDN PRI is digital. More importantly, you cannot dial out over a true DID trunk and you can dial out over a PRI. See DID
A circuit path where only a single pair of wires is used. A Hybrid is used to convert from two wire to Four Wire circuits. No hybrid is perfect, and those used by the phone company can be pretty bad. But the Telos TWO family is approaching perfection!
The interface between the ISDN BRI line and the NT1. This can be considered the ISDN "phone jack" in the USA & Canada and is frequently in the form of a RJ-11 or RJ-45 type telephone jack.
A serial data interface for synchronous data. V.35 uses balanced signal and data lines. The Zephyr models 9201, 9200, 9101, and 9100 support V.35 using part #9812 cable.
The particular protocol (i.e National ISDN-1 or ETS 300) running on a specific switch. Not all variants are valid for a specific switch. The switch brand and model plus the variant defines the ISDN protocol. See ISDN Protocol.
An alternative to repeaters which uses a local Telco Switch to act as a repeater and which then sends the signal onto another switch which supports ISDN. See also Repeater.
A serial data interface for synchronous data popular in Europe. X.21 uses balanced data and unbalanced signal lines. The Zephyr models 9201, 9200, 9101, and 9100 support X.21 using part #9822 cable.
27th August 2007
Joined: Dec 2006
Location: NY NY
Some additional Definitions...
A & B CUTTING: A method of assembling original material in two separate rolls, allowing optical
effects to be made by double printing (A and B Printing).
Aaton Code In-camera keykode/timecode reader.
Action Safe Area The area of a television picture that is visible on consumer television sets.
ADR (Automatic Dialogue Replacement) Recording new dialogue or re-recording dialogue where
the production sound is unusable or obscured.
Ambient Sounds/Effects Sounds recorded as part of the dialogue track.
Analog An electrical signal that is continuously variable.
Answer Print The first print (combining picture and sound, if a sound picture), in release form, offered
by the laboratory to the producer for acceptance. It is usually studied carefully to determine whether
changes are required prior to printing the balance of the order.
Application Method of developing an optical sound track area on a composite film print.
Application Splash Occurs when the chemical used to develop the soundtrack area on film spills
over onto the picture area, damaging the silver in the print stock. Often appears as a purplish-black
area on the screen.
Artifacts Refers to video blemishes, noise, trails, etc. Any physical interruption of the video image is
called an artifact and is usually introduced electronically.
Aspect Ratio The ratio of the picture width to picture height. The standard U.S. television aspect ratio
is 4:3-four units wide to three units high (1.1:33). Other ratios include 1.66:1, 1.85:1, and 2.35:1. If
these alternate ratios are preserved in the film-to-tape transfer, you have an option to put a solid black
bar at the top and bottom of the TV screen.
Auto Assembly Automatic combining of edits on videotape conforming to a prepared edit decision
list (EDL) with little or no human involvement.
B Negative Film term referring to takes not originally slated to be printed from dailies but later called
to be printed. Has carried over into videotape and refers to non-circled takes that are later transferred
as alternative takes.
Betacam SP This is a composite analog 1/2" videotape. There are two channels of analog audio and
two channels of discrete or AFM channels. The analog audio channels are normal audio tracks. The
AFM channels are actually recorded in the video portion of the tape. Therefore, they can be laid down
simultaneously with laying down picture. If, however, you lay down audio on channels 3 and 4 after
picture has been laid down, you will record over the picture. Conversely, if you insert picture after recording
on audio channel 3 or 4, you will erase the audio in that portion.
BLOW-UP PRINTING: Optical printing resulting in a picture image size other than that of the original
Camera Report The form filled in for every camera roll exposed to explain what is on the roll and any
special printing or transfer instructions.
Check Print First film print used to check color corrections.
Chrominance The color portion of a video signal. Also called chroma.
Coding Ink stamping or burning numbers into the edges of work print and work track to mark sync
points. Done with a coding machine.
Color Bars Test pattern used to determine if a video signal is calibrated correctly.
Color Correction The altering of the color balance by modifying the picture color, tint, hue, etc., on
either film or videotape. Also referred to as color balance from scene to scene.
Colorist A telecine operator who corrects the color and light balance while transferring film to
videotape or videotape to videotape.
Composite Audio A fully mixed audio track with dialogue, music and effects married together. May
be stereo or mono.
Composite Film Print A 16mm or 35mm film print that contains a sound track
on the film element.
Component Video A system of signal recording and processing that maintains the original video
elements video elements separately rather than combined (encoded) into a single, composite signal.
Composite Video A video signal in which the luminance and chrominance elements have been
combined. NTSC, PAL, and SECAM are examples of composite signals. Basically a form of analog
video compression allowing for economical broadcasting.
Conforming A variation of a layback. Instead of recording the audio directly to the videotape master,
you record to a second audiotape machine, allowing you to create a multitrack audio element that can
hold several different languages. Also involves matching or syncing a sound track to match an existing
Dl A component digital videotape in 19mm cassette format. There are four channels of audio. This
format displays no generation loss on multilayering work. A single Dl videotape machine can play back
or record in either NTSC or PAL. The longer the tape run time, the thinner the stock.
D2 A component digital videotape in 19mm cassette format. There are four channels of audio. Picture
and audio quality are superior to 1" and Betacam SP.
D3 A ½" composite video format. There are four channels of audio.
D4 Considered bad luck in Japan so a D4 videotape format does not exist.
D5 A ½" component video format. As with Dl, the signal is noncompressed. It also has a provision for
HDTV recording by use of about 5:1 compression. It can also play back D3 tapes and provide
DA88 Audiotape The audio is recorded on hi-8mm metal particle tape stock and provides up to eight
channels of audio recording. The standard sampling rate is 48hz when referencing to video. DA88
tapes must be preformatted at a 48khz sampling rate referenced to video. At the film shoot, the DA88
will be connected to a 60hz reference. The tape will thus be pulled up when recording the production
audio track. This method allows you to get a pull-down effect when working with the tapes in telecine,
videotape editing, and sound editorial.
Dailies Picture and sound work prints of a day’s shooting; usually an untimed one-light print, made
without regard to color balance. Produced so that the action can be checked and the best takes
selected; usually shown before the next day’s shooting begins.
DAT (Digital Audiotape) This is a two channel digital audiotape format with a separate channel for
recording timecode. Because it is digital instead of analog, the sound quality is considered superior to
Daylight Develop Rush film processing through a lab within a few hours instead of overnight.
DCT A digital tape recorder using the DCT method to compress the signal before recording it to tape.
A widely used method of compression.
Decibel (dB) A unit of measurement indicating ratios of currents, voltages or power and used to
represent audio transmission levels, gains and losses. A decibel describes the smallest perceptible
change in audio level. The human ear can perceive 1 dB changes in loudness in the aural range.
DEFT A device for converting NTSC video signals into PAL video signals. This
is a high-quality
standards conversion generally accepted by countries around the world.
Digital Betacam This is a ½" digital metal tape format. There are four channels of audio. Some
models will play back both analog and digital Betacam cassettes.
Digitize Process of loading video and audio into an off-line editing system. Quality of digitized
material (number of frames captured) depends on the amount of storage space on the system.
Director's Cut Rough-cut created by the director once the editor's cut is completed. Usually followed
by the producer's cut and picture lock.
Dissolve An optical or camera effect in which one scene gradually fades out at the same time that a
second scene fades in. There is an apparent double exposure during the center portion of a dissolve
sequence where the two scenes overlap.
Drift When an element does not keep a steady speed during playback. This is usually caused when
there is no timecode to lock to or when the record machine power source was faulty, causing the
recording to vary in speed. This term is also used when speaking about color-correction settings on a
telecine that appear to have changed over time due to light-tube burn.
DFTC (Drop-Frame Timecode) SMPTE timecode created to match run time, or clock time, exactly.
Two frames of timecode are dropped every minute, except every tenth minute. Because it gives exact
run time, broadcasters require masters to be delivered with DFTC.
Drop-Out Temporary signal loss on a video- or audiotape. Shows up randomly as white spots or thin
horizontal lines on video and silence on audio.
Dubbing (Audio) Combining of all sound tracks (dialogue, music and effects) onto a single master
source. Also known as mixing.
thus preserving your original negative.
Duplication (Video Dub) Making videotape copies.
DVD (Digital Video Disk) Cutting edge technology for recording on a five-inch CD using compression
for picture and sound quality superior to VHS.
Edge Numbers Sequential numbers printed along the edge of a strip of film by the manufacturer to
designate identification, thus allowing frames to be easily identified in an edit list. Human and machinereadable.
Keykode is the trademark name for Kodak edge numbers. The combination of letters and
numbers identifies specific information about a particular roll of film, such as place of manufacture.
Editing To arrange the various shots, scenes, and sequences, or the elements of the sound track, in
the order desired to create the finished film.
Editor's Off-Line or Work Cassette Small-format videotape created from videotape master for use
in off-line editing. Timecode matches the master tape and could include visible windows containing the
Keykode information, audio timecode information, etc.
EDL (Edit Decision List) List of edits created during off- line or film editorial.
Effects When working with picture, this refers to visual effects. In audio this refers to sound effects.
Fades When picture or audio slowly disappears. In film these are created as opticals. In on-line they
are done electronically.
Field One-half of a complete picture (or frame), containing all the odd or even scanning lines of the
pictured. In television, one of two complete sequences of raster lines forming an image.
Film Perforation Also called "perf." Holes punched at regular intervals for the length of film, intended
to be engaged by pins, pegs, and sprockets as the film is transported through the camera, projector, or
Film Processing Procedure during which exposed film is developed, fixed, and washed to produce
either a negative or a positive image.
Film Splice Place where two pieces of film are joined by either glue or tape.
Flash Frames In a film element these are white frames between frames with image on them. In
video, these are mistimings in the EDL or editing that leave empty frames between cuts.
Flatbed Can be a Kem, Steinbeck, or other brand of film editing system for viewing picture and track
FLEx File Disk 3.5" floppy computer disk that contains all of the telecine information gathered during
a telecine transfer. Can include Keykode numbers, tape timecode, camera-roll identifiers, sound-roll
timecode, and comments.
Foley Background sounds added during audio sweetening to heighten realism, e.g., footsteps, door
closing, bird calls, heavy breathing, short gasps, etc.
Frame The individual picture image on a strip of motion picture film. A compete video image made up
of two or three video fields.
Frames per Second (FPS) The speed that film or videotape is running.
Gray Scale (Chip Chart) A chart with various shades of gray, which is photographed during
production and used by the film processing laboratory to color correct film.
Half-inch four-track Analog audiotape with three channels for sound recording and one channel
designated for the timecode channel.
Hazeltine Machine sometimes used to color-correct film prints.
HDTV (High-Definition Television) TV signal with extra lines and bandwidths broadcast with a
higher resolution than currently used.
Heads-Out When the beginning of the material is left on the outside of the reel
(as opposed to tailsout).
IN (Internegative) A duplicating film stock that turns into negative when printed from a positive print.
Used to make opticals, and titles, and as a source for making interpositive (IP) prints.
Inserts Additional footage often shot during post production to create an effect or a cutaway shot or
Interlock A system that electronically links a projector with a sound recorder; used during postproduction
to view the edited film and sound track, to check timing, pacing, synchronization, etc.
IP (Interpositive) A positive print made from a negative or internegative on special film stock.
IPS (Inches per Second) Refers to the speed at which audio reel-to-reel ½" formats are recorded.
Either 7½ or 15 IPS is standard.
Jam (Jam Sync) Process of synchronizing a secondary time code generator with a preselected
master time code, i.e., synchronizing the smart slate and the audio time code to the same clock.
Keykode Number Kodak’s machine-readable key numbers, Includes 10-digit key number,
manufacture identification code, film code and offset in perforations.
Keykode Reader Device attached to a telecine or part of a bench logger which reads bar code from
motion picture film and provides electronic output to a decoder. The edge numbers are logged
automatically without human error in about 10 percent of the time it would take for manual entry.
27th August 2007
Joined: Dec 2006
Location: NY NY
Lab Roll A roll of motion picture film made up of more than one camera roll spliced together. Labs
create these rolls for film that will go through telecine so the operator is not constantly changing reels.
These rolls are usually built in either 1000 foot or 2000 foot lengths for 35mm film and 1200 foot
lengths for 16mm film.
Layback Transferring the finished audio track back to the master video tape.
Laydown Recording sound from an audio source or video element to another audio element. During
this process, timecode can be added or altered, channel configurations rearranged, or audio levels
Leader Opaque or clear film attached to the head and tail of film rolls used for threading a motion
Letterbox wide-image is screen, at the top and the bottom of the screen. Traditionally, this is filled in with black bars.
Locked Cut/Locked Picture Final version of a show after all the changes have been incorporated.
Locon A motion picture print made on low-contrast stock.
LokBox Synchronizing mechanism that locks film and videotape to run backwards or forwards
together. Used for negative cutting.
Luminance The measured value of brightness; reflected light measure on motion picture screens as
footlamberts or candelas per square meter. The brightness or contrast of the video signal.
Matte The black bars found at the top and bottom of the picture when a wide screen format is
projected on a television set. An opaque outline that limits the exposed area of a picture, either as a
cut-out object in front of the camera or as a silhouette on another strip of film. Also used in 35mm
projectors to show the correct aspect ratio of a film.
Mixing . Combining of all sound tracks (dialogue, music and effects) onto a single master source.
Also known as audio dubbing
MOS Term for picture without sound. Acronym used to represent the German slang "mit out
Moviola A trademarked name for a machine with a small rear-projection screen and the capacity to
play back several sound tracks. Used in editing and for reviewing portions of the film during production.
Also used to synchronize or interlock picture and sound track in editing. Newer devices called “flat-bed
viewers” are replacing the upright Moviolas.
Nagra Professional ¼" audiotape recorder.
Negative The term “negative” is used to designate any of the following (in either black-and-white or
color): (1) The raw stock specifically designed for negative images. (2) the negative image. (3)
Negative raw stock that has been exposed but has not been processed. (4) Processed film bearing a
Negative Assembly Film is spliced to create lab rolls, or negative is spliced to create a cut picture.
Also referred to as negative cutting.
Negative Dirt Dirt on the film negative element. Can appear white. In some cases it will appear as
sparkles across the screen caused by negative dust. Because the film emulsion is very soft, dirt can
become embedded into the film stock and can only be removed by being washed by the laboratory.
Negative Scratch A scratch in the camera negative which usually appears white on the base side. If
it has penetrated through the yellow, cyan, or magenta layers, it may appear to have a slight tint of
Noise Reduction Electronic reduction of observable grain in the picture. Noise-reduction devices can
minimize discernible grain structure of film, but caution should be observed when
Unwanted side effects can include strobing and trailing images and reduction in picture resolution.
Non-Drop-Frame Timecode A type of SMPTE time code that continuously counts a full 30 frames
per second. As a result, non-drop, frame-time code does not match real time.
NTSC (National Television Standards Committee) Committee that established the color
transmission system used in the U.S. and some other countries. Also used to indicate the system itself
consisting of 525 lines of information, scanned at a rate of approximately 30 frames per second.
Off-Line The process of creatively assembling the elements of a production, to communicate the
appropriate message or story, and/or calculating the order, timing and pace with user-friendly
equipment such as film, 3/4” videotape or non-linear computer editing systems.
On-Line Final assembly or editing utilizing master tape sources. Usually done on high-quality
computer editing system with computer-generated effects.
Opticals Refers to film effects, film titles, and film dissolves and fades. Term has carried over into
videotape and is sometimes used to indicate video special effects.
Paintbox Digital graphics generator made by Quantel. Paint, pen, and airbrush are available
electronically to the graphic artist. A generic term used to describe electronic picture fixes to individual
video frames to mask dirt and scratches.
PAL (Phase Alternating Line) Color television system developed in Germany, and used by many
European and other countries. PAL consists of 625 lines scanned at a rate of 25 frames per second.
Positive Dirt Often built in during printing, this appears black on the screen.
Positive Scratch A scratch in a film print element. Usually appears black on the screen.
Printmaster A stereo mix master audio element consisting of two or four channels of audio.
Production Sound Audio recorded during principle photography on location.
QC (Quality Control) The act of scrutinizing audio, video or film elements for technical specifications
Quarter-Inch (1/4") Audiotape This is a two-track analog audio recording magnetic tape.
Raw Stock Unexposed film or audio stock.
processing of numerous duplicate subject made for general theater distribution. A composite theatrical print in 35mm, printed from an
Reversal Process photographic process which an by secondary
development of the silver latent image development and bleach. the case camera,
the first developer changes the latent image to a negative silver image. This is destroyed by a bleach
and the remaining silver halides are converted to a positive image by a second developer. The
bleached silver and any traces of halides may now be removed with hypo.
RGB (Red, Green, Blue) Red green & blue, the primary color components of the additive color
system used in color television.
Rough cut Assembly of edited shots prior to picture lock.
Saturation Term used to describe color brilliance or purity. When color film images are projected at
the proper brightness and without interference from stray light, colors that appear bright, deep, rich,
and undiluted are said to be “saturated.”
Script Notes A copy of the shooting script prepared by the script supervisor noting camera angles,
what lines were recorded by which camera, shooting order, shot lengths and circled takes.
l6mm Mag Magnetic 16mm audiotape that can hold up to two separate channels of audio. Contains
sprocket holes, so there is no need for timecode for editing.
l6mm Optical Track Mono only. Read by a light on a projector or telecine. Contains sprocket holes
so there is no need for timecode for editing.
Smart Slate Production clapper that includes a lighted readout of the timecode being recorded onto
the production sound audiotape.
SMPTE (Society of Motion Picture and Television Engineers) The committee of engineers that
sets the rules for use of timecode and other technical procedures in the United States and various
Sound Report Form filled in for each audiotape recorded that describes what is on the tape and any
technical instructions for playback.
Splice Joining of two film or audio pieces. Usually done with splicing tape but also can be “hot
spliced” with cement so the finished end-to-end product can function as a single piece of film when
passing through a camera, film processing machine, or projector.
Sync (Synchronous) Align sound and image precisely for editing, projection, and printing. In dailies,
when you hear the clapper close and see it at the same moment, it is considered "in sync".
Sync Sound Sound that is recorded with the intention of being married to a picture at an exact point.
Tails-Out When the end of the material is left on the outside of the reel.
A for scanning motion picture converting videotape.
Temp Dub Temporary music and effects added to a rough-cut version of a project for network or
35MM Mag Magnetic audiotape that can hold up to six tracks of audio. Contains sprocket holes, so
there is no need for timecode for editing.
35MM Optical Track Two-track (mono or stereo) audio format. Contains sprocket holes, so there is
no need for timecode for editing.
Three-Quarter-Inch (¾") Videotape Magnetic tape stock for playback and record. Contains two
channels of audio and a separate timecode channel. Most commonly used for off-line editing and
3:2 Pulldown The formula used to convert 24 frames per second of film to 30 frames per second of
Timecode The numbering system adopted by SMPTE that assigns a number to each video frame
indicating hours, minutes, seconds, and frames.
Timecode Generator An electronic device that outputs timecode.
Transfer A general term for recording from one source to another element.
VHS Tape Video home system ½" videotape.
analog plus two channels of high fidelity (hi-fi) audio.
Visible Timecode Timecode burned into a video picture so it can be seen when viewing the picture.
VITC (Vertical Interval Timecode) Time code that is recorded in the vertical blanking interval about
the active picture area. Can be read from videotape in the “still” mode.
Wetgate Print A print created using a chemical process that coats the negative, hopefully filling in
digs and scratches or imperfections that occur in the negative to help restore the image.
Wild Picture or audio shot without a sync relationship to specific picture.
Work Print Any picture or sound track print, usually a positive, intended for use in the editing process
to establish through a series of trail cuttings the finished version of a film. The purpose is to preserve
the original intact (and undamaged) until the cutting points have been
XFR Shorthand slang for “transfer.”
31st August 2007
Joined: Feb 2007
Location: Los Angeles
Thanks geo!!! This is great....
5th September 2007
Joined: Dec 2006
Location: NY NY
TASA trailer standards
I read an interesting post on the DIGI site and thought i'd post this for reference.
The TASA Standard
Recommendations from the TASA Ad Hoc Committee
for regulating motion picture trailer audio volume.
With the advent of stereo and multi-channel digital trailers, loud
trailer sound became the number one complaint in movie theatres.
Trailers were far louder than the features they preceded, and theatres
compensated by turning down the volume of the trailers and thus features.
To solve this problem, the TASA AD HOC COMMITTEE created a
STANDARD for trailer volume to rectify the situation. It should be
noted that the TASA Committee believes the complexity of the problem
is such that any “Solution” must be tried and proven in the field; the
procedures may be imperfect at first due to the complexity of the
problem. The actual “number” or “upper volume limit” may be adjusted
periodically as experience in the field demands.
Since the original implementation of the TASA Standard in 1999,
the measurement procedures described herein have been adopted as an
The standard is divided into five sections and an annex:
1) MEASURING TECHNIQUES. This section defines in
engineering terms the techniques used to quantify trailer volume
into useful units of measurement.
2) THE NUMERICAL UPPER LIMIT. This is the actual
“number” or upper volume limit that trailers should not exceed
using the measuring techniques set forth in section 1.
3) THE TASA CERTIFICATE. This section outlines the
modus operandi for independent audio engineering firms to
“certify” trailers that satisfy the TASA Standard.
4) INDEPENDENT AUDIO ENGINEERING COMPANY
QUALIFICATIONS. This section details the qualifications
that independent audio engineering companies must have in
order to issue TASA Certificates.
5) POST RELEASE BLIND PRINT CHECKING. This
section outlines the recommended procedures for blind field
checks to insure compliance by all parties participating in the
INFORMATIVE ANNEX: DUB STAGE AND OPTICAL
CAMERA RECOMMENDED PRACTICES. This section
details the recommended procedures to be followed at the dub
stages and at the optical camera.
PART ONE: MEASURING TECHNIQUES
Recommended Practice: Method of Measurement for Equivalent
Perceptual Loudness of Motion Picture Soundtracks.
“The Method of Measurement for Equivalent Perceptual Loudness of
Motion Picture Soundtracks” is a system for producing a NUMBER that
relates to the perceptual loudness of motion picture soundtracks. The
number produced can be used to quantify and regulate the maximum
audio volume of motion picture trailers.
DFFS: Distribution Format Sound System.
Frequency weighting equalizer: a device having a frequency response
that makes the system correspond with perceived loudness, roughly
accounting for the frequency response of human hearing.
Frequency response: the amplitude response of a system as a function
of input frequency, usually rated in decibels over a frequency range.
Pink noise: random, stochastic signal having a continuous spectrum with
equal energy per equal logarithmic intervals of frequency, and with
Gaussian probability distribution of instantaneous amplitude.
Sound Equivalent Level: the average amplitude of sound measured
over an interval of time, calculated according to the equation in section
Trailers: previews of coming attractions presented prior to a feature
film in a motion picture theatre.
Relative and Absolute Sound Pressure Levels for Motion Picture Multichannel
Sound Systems – SMPTE Recommended Practice – RP200 –
A12.004 – 1690
“Are Movies too Loud?” – Ioan Allen, SMPTE Journal (March 22nd
British Standard BS 5550 7.4.2:2000 Specification for maximum
recording levels for commercials and trailers.
ISO 21727 Cinematography – Method of measurement of perceived
loudness of motion-picture audio material.
1.4 METHOD OF MEASUREMENT
The method of measurement shall be as described by the block diagram
given below, with an Input Calibration section, a Frequency Weighting
Equalizer to better correspond to human hearing response than unweighted
measurements, a Sound Equivalent Level section for assessing
the cumulative effect of the energy of the sound level over the time
interval of the program, and a Meter Indicator. The elements of the block
diagram are further defined in sections 1.4.1 – 1.4.4. The measurement
interval, and the accuracy and precision of measurements are given in
sections 1.4.5 – 1.4.6.
Other methods of measurement are permissible so long as the results are
equivalent to those specified herein within the required accuracy of
Block Diagram of Method of Measurement
1.4.1 INPUT CALIBRATION
The input calibration section shall scale an input voltage, defined for each
of the Distribution Format Sound Systems (DFSS), to a voltage that
corresponds to a reference Sound Pressure Level. Each DFSS shall
provide calibration information and test materials, such as test films or
discs, to Independent Audio Engineering Companies in order to make
possible independent verification of soundtrack level on any trailer.
For example, using SMPTE Standards, 20 dBFS on a digital
medium represents the reference level. A DFSS may produce
4 dBu at 20 dBFS, and may be designed so that such an
electrical reference level produces 85 dBC Sound Pressure Level
re: 20 Pa for each channel.
For multi-channel sound, each of the source channels shall be electrically
summed in the correct proportion to the Sound Pressure Level calibration
of the individual channels. For example, if the surround level is
calibrated at 82 dBC for each channel rather than 85 dB given in the
example above, the contribution of each surround channel to the sum
shall be 3 dB less than a screen channel.
To prevent differences between electrical addition (vector) and acoustical
addition in the reverberant field of a room (scalar), each of the channels
shall employ a separate detector circuit, and the output of each of the
detector circuits shall be added.
1.4.2 FREQUENCY RESPONSE AND TOLERANCE OF
FREQUENCY WEIGHTING EQUALIZER
The frequency weighting equalizer is based on an International
Telecommunications Union recommended filter for the assessment of
background noise in audio programs. This filter (more accurately,
equalizer) has also been found to be useful for the purpose of assessing
the human response to the loudness of soundtracks. The frequency
response of the equalizer, and the tolerance on the response, is given in
the following table:
Frequency in Hz Response in dB Tolerance in dB
31 -35.5 + 2.0
63 -29.5 + 1.4
100 -25.4 + 1.0
200 -19.4 + 0.85
400 -13.4 + 0.7
800 -7.5 + 0.55
1000 -5.6 + 0.5
2000 0.0 + 0.5
3150 3.4 + 0.5
4000 4.9 + 0.5
5000 6.1 + 0.5
6300 6.6 + 0.0
7100 6.4 + 0.2
8000 5.8 + 0.4
9000 4.5 + 0.6
10000 2.5 + 0.8
12500 -5.6 + 1.2
14000 -10.9 + 1.4
16000 -17.3 + 1.65
20000 -27.8 + 2.0
31500 -48.3 + 2.8, - ∞
Note that for the purposes of insertion gain, the frequency 2.0 kHz is used
for the 0 dB reference level. For the purposes of tolerance, the insertion
gain is to be adjusted at the reference frequency of 6.3 kHz to 6.6 dB,
since this is the center frequency of the boost in the equalizer. If a 1 kHz
reference frequency is used, levels shall be offset by 5.6 dB, as shown in
1.4.3 SOUND EQUIVALENT LEVEL
1.4.4 METER INDICATION
The meter indication shall be the result of the frequency weighting
equalizer and the sound equivalent level circuit, with scaling to represent
the acoustical Sound Pressure Level which the program material would
produce when playing the test material specified in the next paragraph at
the Standard Fader Setting over a sound system calibrated to the standard
of the DFSS in use.
The frequency response of theatrical sound systems, the B-chain
response, specified in SMPTE 202, is deliberately not to be accounted for
in this method of measurement. The X curve response is not to be a part
of the frequency weighting equalizer.
Note: Acoustical rather than electrical summation of the channels,
and the fact that the X curve is not accounted for in the electrical
measurement described herein, will probably make the electrical
based measurement described in this method of measurement
different from the reading of a Sound Level Meter - even if one
were to be equipped with the frequency equalizer specified herein
and made to measure Leq M. In addition, variations from room to
room, including seat location selection, would make Sound Level
Meter measurements unreliable.
1.4.5 MEASUREMENT INTERVAL
The length of the measurement in time shall correspond + 3 seconds to
the length of the audio program material. In practice, the start button of
the measuring device shall be pushed within 3 seconds of the first audio
heard by the audience. The stop button shall be pushed within 3 seconds
of the final audio heard by the audience. The measurement does not stop
during any silences within the body of the trailer. The measurement does
not start with the academy leader, the green card or any other visual or
footage reference; the only section to be measured for this standard is the
section between the 1st audio heard, plus or minus 3 seconds, and the
final audio heard, plus or minus 3 seconds.
Any material measured using these procedures with a duration of 30
seconds or less shall be measured to within + 1 second of the length of
the program. In other words, the start button of the measuring device shall
be pushed within 1 second of the first audio, and the stop button shall be
pushed within 1 second of the final audio.
1.4.6 ACCURACY AND PRECISION OF MEASUREMENT
Independent Audio Engineering Companies that measure trailers to check
for TASA compliance (See Parts 3 and 4), shall maintain the accuracy of
the measurement procedure to within + 0.3 dB. This tolerance on
accuracy shall include a summation of all sources of error, including, but
not limited to: input calibration error, insertion gain, error in the
frequency weighting filter or elsewhere, calculation of the sound
equivalent level, and meter indication. Independent Audio Engineering
Companies shall not be responsible for the accuracy errors that occur due
to calibration error on the part of the Distribution Format Sound Systems,
which shall be maintained by the manufacturers of the various sound
systems, including variations due to hardware and software upgrades.
The precision considered in a pass/fail response shall be to the nearest 1
dB. The Independent Audio Engineering Companies may maintain
internal records to greater precision.
PART TWO: THE NUMERICAL UPPER
2.1 A FLEXIBLE NUMBER
Any volume standard for trailers must take into account feature volume as
well as playback habits of theatres in order to be effective. Establishing a
standard will have an element of trial and error in the field, and may not
be achievable in a single precipitous drop in trailer volume. It may be
prudent to take several conservative drops in volume over the course of
several months in order to reach a desirable level without accidentally
“overshooting” and making the trailers too low. To this end, the TASA
Ad Hoc Committee meets periodically to determine whether or not the
standard requires revision.
PART THREE: THE TASA CERTIFICATE
3.1 “COMPLIANCE” REQUIRES A TASA CERTIFICATE
The upper volume limit established by TASA can be adhered to on the
dub stage and on the optical track negative, (or other theatrical release
medium), but full TASA compliance requires that an “Independent Audio
Engineering Company” (an “I.A.E.”) measure either a composite print or
a soundtrack-only print of the trailer, using the measuring techniques and
upper limit guidelines established in Parts 1 and 2 of the TASA Standard.
If the mix passes the TASA Standard, the Independent Audio Engineering
Company will issue a “TASA Certificate of Compliance”. Only when the
I.A.E. has issued the TASA Certificate will the trailer have satisfied the
3.2 I.A.E. CERTIFICATION
Independent Audio Engineering Companies who wish to issue TASA
Certificates must apply for Certification by the TASA Committee.
Applications should be submitted in writing to the chairperson of the
TASA Committee for consideration by the full committee.
3.3 I.A.E. DECERTIFICATION
Independent Audio Engineering Companies can be decertified at any time
by the TASA Committee for engineering or administrative failures.
3.4 METER CALIBRATION GRACE
If a trailer fails to pass the TASA Standard at an I.A.E. and the certifying
engineer discovers that unintentional Leq M meter mis-calibration on the
dub stage was to blame, the certifying engineer may still issue a TASA
Certificate. No further certificates may be issued for trailers mixed on
that dub stage until the meter is recalibrated.
3.5 TASA CERTIFICATE INFORMATION
The TASA Certificate must include: the name of the I.A.E., the name of
the trailer certified, the name of the studio, the date mixed, the dub stage,
the Leq M number measured, the name and signature of the certifying
engineer, and any other pertinent information.
PART FOUR: INDEPENDENT AUDIO
In order to be certified by the TASA Committee, any Independent Audio
Engineering Company must meet the following standards:
4.1 The trailer volume measuring work must be performed by, or under
the direct management and responsibility of either:
a) a registered Professional Engineer with a current license from
the State of California, or
b) a recognized audio expert, with recognition consisting of Fellow
grade membership in the Audio Engineering Society, the British
Kinematograph Sound and Television Society, the Institute of
Electrical and Electronics Engineers, or the Society of Motion
Picture and Television Engineers, or
c) any engineer who demonstrates to the TASA Technical
Committee proficiency in the measuring techniques described in
4.2 The I.A.E. shall maintain primary standards for rms ac voltage and
thus equivalent Sound Pressure Level reference traceable to the National
Institute for Standards and Technology. The I.A.E. must maintain the
necessary test equipment to qualify the frequency response of the
equalization network, and tone burst generators as required to qualify the
time response of the sound equivalent level measurement to within the
requirements of the section “Accuracy and Precision of Measurement”.
The I.A.E. shall be responsible for the calibration of its own equipment,
and for maintaining the accuracy of measurement described in the
4.3 The I.A.E. shall not be owned in whole or in part by any entity that
produces, distributes, or exhibits motion pictures or associated trailers
intended for theatrical exhibition. An exception to this rule can be made
if the Independent Audio Engineering Company excludes itself from
measuring the related company’s product; i.e. if the company is owned by
a parent company that produces films on occasion, the I.A.E. must
exclude itself from certifying the parent company’s product. The I.A.E.
in these circumstances must make its equipment and personnel available
to be supervised (at the I.A.E.’s expense) by an engineer approved by the
4.4 The I.A.E. shall not be owned in whole or in part by any entity that
manufactures or distributes a Distribution Format Sound System.
4.5 The I.A.E. shall not be owned in whole or in part by any entity that
edits, mixes, or otherwise makes trailers for theatrical feature films. The
same exception can be made for this rule as the one for 4.3: The I.A.E.
cannot certify product that it or its related company has worked on unless
supervised by an engineer approved by the TASA Committee.
4.6 The I.A.E. shall not be owned in whole or in part by any entity that
produces soundtrack negatives or prints for trailers for theatrical feature
PART FIVE: POST RELEASE BLIND PRINT
5.1 The TASA Committee recommends that any studio or entity
adopting the TASA STANDARD will in so adopting, be giving the
certifying I.A.E. implicit and unconditional permission to pull one print of
each trailer at random (from the lab or depot of the studio’s choice) for a
blind volume check. The I.A.E. will double check the audio volume
using the engineering standards set forth in section 1. This will serve to
insure full compliance by all parties.
5.2 An I.A.E. performing a post release check on a trailer may issue a
notice of TASA noncompliance if the trailer print fails the Standard.
5.2.1 An I.A.E. may also issue a notice of noncompliance if a studio
does not make a randomly pulled blind check print available.
5.3 The I.A.E. will forward a copy of the notice of noncompliance to
the chair of the TASA Committee, to the studio that released the trailer,
and to any other entities that adopt the TASA Standard.
INFORMATIVE ANNEX: DUB STAGE AND
OPTICAL CAMERA RECOMMENDED
6.1 Trailers shall be mixed at volume levels that are comfortable to the
ear on the dub stage. Care should be taken on dialog levels in particular,
such that they do not exceed normal feature film levels.
6.2 Upon completion of a trailer mix, the mixer shall measure the
Leq M . If the resultant number exceeds the recommended upper limit in
effect at the time of the mix, the mixer shall remix the trailer to lower the
volume until it meets the upper limit or falls below. If the Leq M number
is below the upper limit, the mixer shall not raise the volume of the mix
solely to meet the upper limit since this would likely result in painfully
loud dialog levels.
6.3 Upon completion of the mix, the mixer shall note the Leq M number
into a log to be kept on the dub stage. In addition, the mixer shall fill out
a dub stage report which lists the trailer mixed, the dub stage, and the
Leq M number measured on the stage. This dub stage report shall be
forwarded with a trailer soundtrack-only print, composite print, or other
digital audio delivery medium to the I.A.E. for final TASA certification.
The reason for the dub stage report is to help the I.A.E. check the
calibration on the dub stages. If every mix that gets checked by an I.A.E.
comes with a report, then every mix will help confirm that the Leq M
meters are in calibration. This will prevent out- of-calibration meters
from being discovered only when a mix fails the TASA Standard.
6.4 Under no circumstance should studios bracket tracks at the optical
camera. All sound decisions must be made on the dub stage with the
guidance of the sound mixer.
The TASA Ad Hoc Committee is comprised of marketing post production
representatives from all of the studios, as well as engineers from the
digital sound companies, mixers, and independent audio experts. The
TASA Committee does not endorse any one company’s product or
services. All TASA volume recommendations are recommendations
only. The TASA Committee believes the method of measurement
described herein is the best available at the time of this writing. If and
when better volume assessment techniques are developed, the TASA
Committee may amend the TASA Standard. Any outside agencies that
endorse or adopt the TASA Standard may make these recommendations
mandatory at their own discretion.
6th September 2007
Joined: Dec 2006
Location: NY NY
SOme Dolby Stuff
Dolby Stereo 1975 Cinema use. FL FR with C and RearMono matrixed
Dolby SR 1986 Cinema use. FL FR with C and RearMono matrixed
Dolby Surround 1986 First Home use. Analog. FL FR with C and RearMono matrixed
Dolby Pro Logic > 1986 Improved Dolby Surround. Upmix Stereo to Surround 4.0. FL FR C RearMono(x2)
Dolby Pro Logic II
2000 Improved Dolby Pro Logic. Upmix Mono(in Matrix mode) or Stereo to Surround 5.1 in either Matrix, Movie, Music, or Game mode.
FL FR C SL SR SUB
Dolby Pro Logic IIx
Upmix Stereo or Surround 5.1 to 6.1 or 7.1 in either Movie, Music, or Game mode.
FL FR C SL SR SUB and RearMono(x2)
Jazz, Hall & Stadium
Extra Decoders implemented by manufacturers.
These add effects like echo and delay to change sound.
Dolby Digital Adaptive Transform Coder 3 (AC3) 1992 Film
1995 Laser Disc
Discrete channel encoder/decoder. Pro Logic Decoder can be used for downmixed stereo inputs.
FL FR C SL SR SUB
Dolby Digital EX/Dolby Digital Surround EX 1999 6.1 or 7.1 (5.1 with Center Rear matrixed onto SL & SR)
FL FR C SL SR SUB RearMono(x1 or 2x)
Dolby Digital Plus
High bitrate. Currently uses 7.1 channels with support for more.
Dolby TrueHD Lossless encoder for High Definition Video Sound.
Eight full-range channels of 24-bit/96 kHz. Higher Bitrate than DD-Plus.
Dolby Digital Live On the Fly 5.1 encoder for Games.
24th September 2007
Joined: Jan 2007
My head is about to explode from all this information! And I love it. And I was just perusing it. That is amazing. Thank you so much.
24th October 2007
Joined: Dec 2006
Location: NY NY
Update to my facility...
I've been asked a few time about uhe systems I use at my home studio. So, I wanted to take a minute and update those that got this far in my posts. I have replaced my trusted Neve Capricorn Digital Console with a new Euphonix System5 Hybrid system. Around the new system, I have upgraded my facility. So now I have the following configuration:
Euphonix System5 Hybrid 300 Channel 2seat film/broadcast console.
Tied to the console are 2 Protool HD rigs utilizing the SSL Delta-Link MADI IOs, 1 Protool TDM system utilizing a EUphonix FC727 IO, and 1 Logic8 system Utilizing a HAmmerfall MADI IO card.
THe Protools HD rigs have 64 IO each, the TDM rig has 48 IO and the Logic system has 56 IO.
The Euphonix desk, besides 300 channels of full on digital console, offers what amounts to an ICON control surface for the protool rigs and a full Logic 8 control surface. All three formats can be on the desk surface together in real time. All our sync is handled via a SOUNDMASTER ION ATOM system.
We also have an LM100 and a DK600M for metering, both digitally tied to the desk.
We have 2 full on TC Electronic system 6000 rigs tied to the desk for 8 Engine 32 IO effects/verbs/etc. Additionally, we utilize a DNS2000 and the 6000 backdrop for dialogue editing. I have both a nearfiled Genelec system and a 5.1 Theatre layout Genelec monitoring system in my main room. On my desk we have two producer desks where we placed a pair of TFT displays and a KVM Switch set. On the KVM switch we have access at either desk independently, LM100 remote display, EMIX Euphonix console ops, Minimac for web surfing, VVTR access for Finalcut and Video playback in the room, Soundmaster ION sync control, and Logic8 duplicated screen from the music editing station. We have 2 flatscreens for each protool rig mounted on the Euphonix at either side of the room for protool editing. Finally, we have a DTS T-2 tower /DA98HR and a DOLBY DMU system digitally tied to the desk for printmasteing. We al ou sound design upstairs on Digi003s,Digi002s and Mbox systems using DVtoolkit optins and good ol' Soundminer network version for sound effects pulls.
1st December 2007
Joined: Aug 2005
I know it's been said, but wanted you to know that your effort is very appreciated here.
Thank you very much Georgia!!
Great info. Very generous from you!!
3rd December 2007
Joined: Dec 2006
Location: NY NY
thanks... I appreciate the note. Just trying to share and learn everyday!
11th December 2007
Lives for gear
Joined: Nov 2007
I'm soaking this up like a sponge....
Thanks tons for this!
26th December 2007
Joined: May 2006
I mostly read and don't post much. But after reading this sticky post I must thank you for all you have organized here. I teach Sound Design at Duquesne University and will recommend to my students that they follow this forum. I have avoided the recommendation due to the somewhat 'salty' hostname (we're a Catholic University) but your post here and the forum you host is well worth it. Thank you for all of your expertise and willingness to share it.
29th December 2007
Joined: Dec 2006
Location: NY NY
Thanks Dom. Happy to help. If I can be of any service or you have a question feel free to call the studio!
Happy New Years Everyone!!!!
PS: just got my hair done so i updated my avatar...
30th December 2007
Joined: Dec 2006
Location: NY NY
Here's a piece on Room Simulations for music and film
Typical production techniques involve only one or two inputs to the room simulation
system, thereby limiting the precision of source positioning to be only a matter of send
level differences and power panning. This “one source - one listener” model is not very
satisfying when producing for mono or stereo, but even worse when the reproduction
system is multichannel.
Multichannel recording and reproduction is an opportunity for the production engineer to
discriminate deliberately between scenes or instruments heard from a distance, and
sources directly engaging the listener.
For film work, engaging audio has a very pronounced effect for stimulating the viewer
emotionally, and may therefore significantly add to the illusion presented by the picture.
In the search of more authenticity in artificial room generation, long term studies of
natural early reflection patterns have led us to propose new production and algorithm
techniques. Using ray tracing in conjunction with careful adjustments by ear, we have
achieved simulation models with higher naturalness and flexibility, which is the basis of
true source positioning.
The paper will discuss two aspects of precise room simulation for multi source,
multichannel environments to cover distant and engaged listening:
l Present different production techniques
l Describe an algorithm structure to achieve the objectives
I. SINGLE SOURCE REVERB
By having only one or two inputs in a room simulator, the rendering is based upon
multiple sources sharing the same early reflection pattern, and therefore it is not really
In the real world, all actors or instruments are not piled up on top of each other.
I. I Music Production
In many studios, one good reverb is used to render the basic environment of a particular
mix. One aux send, set at different levels on the different channels, is used to obtain depth
and some complexity in the sound image.
To obtain a sound image of a higher complexity and depth, several auxes and reverbs
have normally been used. Tuning of the levels, pans and reverb parameters in such a setup
may be very time-consuming.
For effect purposes, anything goes, but if the goal is a representation of a natural room or
a consistent rendering of a virtual room, it may be hard to achieve using conventional
I .2 Film and Post Production
For applications where picture is added to the sound, several psychological studies have
proven audio to be better at generating entertainment pleasure and emotions than visual
inputs. When it comes to counting neurological synapses to the brain, vision has long
been known to be our dominant input source. However, a study by Karl Ktipfmtiller 
has suggested, that stimulation of even our conscious mind is almost equally well
achieved from visual compared to auditive inputs.
Sense No of Synapses Conscious input, bps
Eye IO. 000.000 40
Ear 100.000 30
Skin I. 000.000 5
Smell 100.000 1
Taste 1.000 I
Stimulation of conscious mind 
Realism in audio is just as important when it is accompanied by picture.
In multichannel work for film, several reverbs configured as mono in - mono out are often
used on discrete sources. By doing so, the direct sound and the diffused field are easy to
position in the surround environment. The technique is therefore especially effective for
point source distance simulation.
As an alternative, several stereo reverbs are used on the same sources to achieve a number
of de-correlated outputs routed to different reproduction channels.
With both approaches, adjustments can be very time-consuming, and a truly engaging
listening experience is difficult to achieve.
2. MULTIPLE SOURCE ROOM SIMULATION
To obtain the most natural sounding and precise room simulation, an artificial reverb
system should be based upon positioning of multiple sources in a virtual room. Each
source should have individual early reflection properties with regards to timing, direction,
filtering and level.
We have found this to be true for both stereo and multichannel presentations.
If the target format is 5.1, at least two directional configurations should exist in the room
simulator, namely for home (110 degree surround speakers) and theatre (side array
surround speakers) reproduction.
The room simulator should also be flexible enough to easily adopt to new multichannel
formats, e.g. the Dolby EX scheme.
By changing the production technique slightly, multiple sends from e.g. the Auxes, Group
busses or Direct outs of the mixing console can be used to define several discreet
positions as inputs to the room simulation system.
From a production point of view, multiple source room simulation can be configured two
ways, as described below. Any large scale console build for stereo production can adapt
to both routing schemes.
2. I The Additive Approach
The conventional approach to reverb is additive. Dry signals are fed to the reverb system,
and wet-only signals are returned and added at the mixer.
With a multiple input room simulator, this configuration works much better than with an
single source reverb, because at least each source can be approximated to fit the nearest
position rendered. However, normal power panning still needs to be applied in the mixer.
An even more precise rendering can be achieved using the integrated approach described
2.2 The Integrated Positioning Approach
The sources in a mix needing the most precise positioning and room simulation, should be
treated this way:
The source is completely positioned and rendered into a precise position by passing the
dry signal through the simulation system, from which a composite output from a number
of source positions are available.
XY positioning to any target format, stereo or multichannel, will be rendered as a best fit.
The positioning parameters (replacing conventional power panning) can be controlled
from a screen, a joystick or discrete X and Y controls.
With all positioning done in the room simulator, consoles made for stereo production may
thereby overcome some of their limitations.
3. ALGORITHM STRUCTURE
This part of our paper describes a generic algorithm currently in use for Multichannel
Room Simulator development. It is not a description of any particular present or future
product, but rather a presentation of the framework and way-of-thinking that has produced
our latest Room Simulation products and is expected to produce more in the future.
3. I Design conditions
The overall system requirements can be stated as follows:
l The system must be able to produce a natural-sounding simulation of a number of
sources in acoustic environments ranging from “phone-booth” to “canyon”
l The system should not be limited to simulating natural acoustics: Often quite
unnatural reverb effects are desired, e.g. for pop music or science fiction film effects.
l The system should be able to render the simulation via a number of different
reproduction setups, e.g. 5.1,7.1, stereo etc.
l The system should be modular so that new rooms, new source positions in existing
rooms, new source types or new target reproduction setups can be added with minimal
change to existing elements.
l The system should be easily tuneable: In our experience, no semi-automatic physical
modeling scheme, however elaborate, is likely to produce subjective results as good as
those obtained by skilled people tuning a user-friendly, interactive development
prototype by ear.
Fortunately there are a few factors that make the job easier for us:
There are no strict requirements for simulation accuracy: Certainly not physical
accuracy (the sound field around the listener’s head), and not even perceptual accuracy
(the listener’s mental image of the simulated event and environment). The listener has
no way of A/E3 switching between the simulation and the real thing, so only
credibility and predictability counts: The simulation must not in any way sound
artificial, unless intended to, and the perceived room geometries and source positions
should be relatively, but not absolutely, accurate.
Moore’s Law is with us. The continual exponential growth in memory and calculation
capacity available within a given budget frame has two effects: It constantly expands
the practical limits for algorithm complexity, and it makes it increasingly feasible to
trade in a bit of code overhead for improved modularity, tuneability, etc.
There are physical modeling systems readily available, which may provide a starting
point for the simulation.
3.2 Block diagram
The overall block diagram of the Room Simulator is shown in fig. 1. As often seen, the
system is divided into two main paths: An early reflections synthesis system consisting of
a so-called Early Pattern Generator (EPG) for each source and a common Direction
Rendering Unit (DRU) that renders the early reflections through the chosen reproduction
setup. And a Reverb system producing the late, diffuse part of the sound field. Note that -
contrary to what is normally the case - there is no direct signal path. The dry source
signals are merely Oth order reflections produced by their respective EPGs. In the
following, a more detailed description of the individual blocks is given.
3.3 Early Pattern Generators
Each EPG takes one dry source input and produces a large set of early reflections,
including the direct signal, sorted and processed in the following “dimensions”
The Level and Delay dimensions are easily implemented with high precision, the other 3
dimensions are each quantized into a number of predefined steps, for instance 12 different
directions. Normally, the direct signal will not be subjected to Diffusion or Color. The
quantization and step definition of the Direction dimension must be the same for all
sources, because it is implemented in the common Direction Rendering Unit. Physical
modeling programs such as Odeon [l] may provide an initial setting of the EPG.
3.4 Direction Rendering Unit
The purpose of this unit is to render a number of inputs to an equal number of different,
predefined subjective directions-of-arrival at the listening position via the chosen
reproduction setup, typically a 5-channel speaker system. Thus, the DRU may be a
simple, general panning matrix, a VBAP  system or an HRTF- or Ambisonics-based
3.5 Reverb Feed Matrix
The reverb feed matrix determines each source’s contribution to each Reverberator input
channel. Besides gain and delay controls, some filtering may also be beneficial here.
To ensure maximum de-correlation between output channels, each has its own
independent reverb “tail” generator. Controllable parameters include:
Reverberation time as a function of frequency T,(f)
We take particular pride in the fact that our “tail” can achieve such smoothness in both
time and frequency, and that modulation may be omitted entirely. This eliminates the risk
of pitch distortion and even the slightest Doppler effect, which tends to destroy focus of
the individual sources in a multichannel room simulator.
Again, an initial setting of T,(f) may be obtained from Odeon.
3.7 Speaker Control
This block is by default just a direct connection from input to output. But it may also be
used to check the stereo- and mono compatibility of the final simulation result by
applying a down-mixing to these formats. Also it provides delay- and gain compensation
for non-uniform loudspeaker setups, which may also - as a rough approximation - be used
the other way around to emulate non-uniform or misplaced setups and thus check the
simulation’s robustness to such imperfections.
The system described above is evidently a very open system under continual
development. At the time of writing these words, our test system is running in real time
on a multiprocessor SGI server with an 18-window graphical user interface providing
interactive access to approximately 2000 low- and higher-level parameters. However, this
is not the time or place to go into more details. When this paper is presented at the 107*
AES Convention in about 4 months, we will have more real life experience with the
If integrated positioning is used with multi-source room simulation, our experiments have
already shown how much there is to gain in terms of realism and working speed. But even
with the less radical additive approach, virtual rooms may be rendered more convincingly
with multi-source simulators.
For applications where picture is added to the sound, the most stimulating source will be
one, where audio and video are treated with equal attention to quality and detail.
The new possibilities available from multi-source room processors may be exploited to
generate a real quality improvement at the end listener, especially when his reproduction
system is multichannel.
More convincing sound generates more convincing picture.
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