Geo's sound post corner

[georgia]

ok, well...umm.......

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....

(OPINION ON)

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!

(OPINION OFF)



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.


Cheers
geo


PS: this post obviously helped exactly 0%.. sorry.

PPS: no i'm not a Genelec spokesperson.....

[hef]

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.

[georgia]

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.
The system
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.

cheers
geo

[georgia]

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.

cheers
geo

[legareal]

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.

[georgia]

This is a great primer on post written by Bruce C. Nazarian M.P.S.E.

here's the website Audio Post FAQ's

enjoy

* 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.




cheers
geo

[georgia]

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.

What about: 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.


LCRS

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.


cheers
geo

[starcrash13]

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.

[flyboogie]

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.

[georgia]

When I create my stems, I create the following on my console:

Dialogue 5.1
Music 5.1
SFX 5.1 ( includes Foley and SFX )
AMB 5.1


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 )

Wink
cheers
geo

[starcrash13]

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.

[georgia]

oops... sorry... yes. we do other things in there as well..
Broadcast 5.1 mixes, Music stuff and the like...



my bad!

Cheers
geo

[georgia]

“X” CURVE

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.



cheers
geo

[georgia]

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.

AMI

Alternate Mark Inversion. A T1 line coding method. This is the older of the two commonly available. See Line Coding, T1. See Also B8ZS.

ANI

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.


Asynchronous Data

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.

B8ZS

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

B Channel

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.

Bell Labs

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.

Bellcore

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.


BERT

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.

Bit Rate

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.

Blocking

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.

BRI

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.

BRITE

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.




Business Office

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.

CCIS

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.

Channel

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.


Choke Exchange

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

Circuit

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.

Circuit Switching

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

CLASS

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.

CLEC

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.





CLID

Calling Line Identification. This is the ISDN and SS7 equivalent of Caller ID; I.E. the number of the calling party. See also ANI.

CO

Central Office. The Telco facility where your local telephone circuit leads to. Contains Switches and Trunks as well as the local telephone circuits.

Codec

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.

Combination Trunk

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

Concentration

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.

CPE

Customer Premise Equipment- Customer owned equipment located at his/her facility. In the USA and Canada the NT1 is part of the CPE.


CSD

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.

CSU

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.

CSU/DSU

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.

CSV

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.

Custom ISDN

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.

D Channel

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.


DCE

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.

Dedicated Circuit

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.

DID

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.

DID Number

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.



DID Trunk

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.