Hi, Just thought I'd add this one for anyone with specific questions I can help with.

I've posted some misc stuff I've written and/or gathered about tech and post... As I trip over stuff I'll post it.

cheers
geo

Then let me help you and stick it to the top... I have read some of your posts over at the DUC and I think this spot is well deserved!

Thanks!! I appercaite the praise from a pro! Hope to help all....

Since this is GEAR SLUTZ and I'm one. I'll list some of my toys:

Dub Stage:
Euphonix System 5 Hybrid - 2 seat console with 300 Channels for mix.
Neve and Euphonix Converters for Analog and Digital IOs.
FC727 For Protools TDM IO to the System5
SSL converter for Protoold HD IO to System5
Neve Digital IOs for connects to two TC Electronic System 6000's
The 2 6000's have 16 digital IO each and lot's-o-software.
DK600M metering with remote screen
DSM7.1 Down mixer
DA98HR for printmastering
Dolby Digital DMU
DTS T2 Tower
DTS Threatre playback system
GENELEC 5.1 system
misc old analog devices like Lexicon verbs, Keypex Gates, Gainbrains Limiters, LA2A comps, etc.
Protools MIX+ with 4 farm cards and bazillions of plugins
Protools HD with 64 IO using SSL converters
Protools LE Digi002 (3)
Protools LE digi003 (1)
VVTR pro on quad intel using AJA Kona Card for HD output video to HD projector and VUtec microperf screen
SONY HD playback deck, Beta-SP, DVCpro50, and others

Editing rooms:
Mostly Protools Mbox, Digi002s etc w/dvtoolkit running 7.x Protools on MAC quad intels with Genelec monitors.

Foley Pit/ADR
PRotools DIGI003 w/DVtoolkit, Focusrite Mic Pres, Misc gear
Genelec 1031A 5.1 system
DC30+ and multiple screens for monitoring video

Editing Picutre on G5s and Quad intels with FCP 5 HD, Avid Mojo and using PC based SONY XPRI and HDcam for uncompressed Full 1920x1080p HD editing


cheers
geo

here's a powerpoint presentation on the history of audio in film I did a year ago or so for a quick presentation...



ok, never mind... for some reason I can't get the .ppt or the zip file to up load. if anyone wants a copy feel free to email me at [email protected] and i'll mail the power point presentation to them.

cheers
geo

some film crew links for people looking for post gigs...

WWW.FILMSTAFF.COM http://www.filmstaff.com/index.asp?camp=101

WWW.MANDY.COM

WWW.CRAIGSLIST.COM

WWW.BLS.GOV/OCO/CG/CGS038.HTM

http://www.panix.com/~jviii/filmlctr/jv3film1.html
http://www.afilm.dk/
www.film-connection.com
http://www.productionhub.com/
http://www.hollywoodnetwork.com/
http://www.usfilmproduction.com/
The Independent Filmmakers Internet Resource Guide
http://www.media-match.com/index.php?affid=9999

http://www.assistantdirectors.com/

http://www.filmmaker.co.za/

http://www.filmcrewonline.com/

http://www.crewnet.com/

http://www.caryn.com/biz/ibuzz/crew-prod.html

http://www.ukscreen.com/crew

http://www.nmfilm.com/locals/faqs/

http://www.eej.com/film_crew.htm

http://www.crewmarket.com/

cheers
geo

A drive array is a collection of hard disk drives that are grouped together. When you talk about Raid, there is often a distinction between physical drives and arrays and logical drives and arrays. Physical arrays can be divided or grouped together to form one or more logical arrays. These logical arrays can be divided into logical drives that the operating system sees. The logical drives are treated like single hard drives and can be partitioned and formatted accordingly. The Raid controller is what manages how the data is stored and accessed across the both the physical and logical arrays. It ensures that the operating system only sees the logical drives and does not need to worry about managing the underlying schema. As far as the system is concerned, it's dealing with regular hard drives. A Raid controller's functions can be implemented in hardware or software. Hardware implementations are better for Raid levels that require large amounts of calculations. The single individual Raid levels don't address every application requirement that exist. So, to get more functionality, someone thought of the idea of combining Raid levels. The main benefit of using multiple Raid levels is the increased performance. Usually combining Raid levels means using a hardware Raid controller. The increased level of complexity of these levels means that software solutions are no practical. Raid 0 has the best performance out of the single levels and it is the one most commonly being
combined. Not all combinations of Raid levels exist. The most common combinations are Raid 0+1 and 1+0. The difference between 0+1 and 1+0 might seem subtle… the difference lies in the amount of fault tolerance. Both these levels require at least 4 hard drives to implement so this can get a bit expensive.. ok lets hit the details of Raid levels…

Raid 0
This is the simplest level of Raid… and it just involves striping. Data redundancy is not even present in this level, so it is not recommended for applications where data is critical. This level offers the highest level of performance out of any single Raid level. At least 2 hard drives are required, preferably identical, and the maximum depends on the Raid controller. None of the space is wasted as long as the hard drives used are identical. it's relatively low cost and high performance gain. This level is good for most people that don't need any data redundancy. It works with SCSI and IDE/ATA implementations. Finally, it's important to note that if any of the hard drives in the array fails, you lose everything.

Raid 1
This level is usually implemented as mirroring. Two identical copies of data are stored on two drives. When one drive fails, the other drive still has the data to keep the system going. Rebuilding a lost drive is very simple since you still have the second copy. This adds data redundancy to the system and provides some safety from failures. Some implementations add an extra Raid controller to increase the fault tolerance even more. It’s ideal for applications that use critical data. Even though the performance benefits are not great, it really helps with preserving data. It is also relative simple and has a low cost of implemention. Most Raid
controllers nowadays implement some form of Raid 1.

Raid 2
This level uses bit level striping with Hamming code ECC. The technique used here is somewhat similar to striping with parity but not really. The data is split at the bit level and spread over a number of data and ECC disks. When data is written to the array, the Hamming codes are calculated and written to the ECC disks. When the data is read from the array, Hamming codes are used to check whether errors have occurred since the data was written to the array. Single bit errors can be detected and corrected immediately. This is the only level that really deviates from traditional Raid ideas. Remember, this level is very complicated and expensive Raid controller hardware is needed.

Raid 3
Raid 3 uses byte level striping with dedicated parity. In other words, data is striped across the array at the byte level with one dedicated parity drive holding the redundancy information. The idea behind this level is that striping the data increasing performance and using dedicated parity takes care of redundancy. 3 hard drives are required. 2 for striping, and 1 as the dedicated parity drive. Although the performance is good, the added parity does slow down writes. The parity information has to be written to the parity drive whenever a write occurs. This increased computation calls for a hardware controller, so software
implementations are not practical. Raid 3 is good for applications that deal with large files since the stripe size is small.

Raid 4
This level is very similar to Raid 3. The only difference is that it uses block level striping instead of byte level striping. The advantage in that is that you can change the stripe size to suit application needs. This level is often seen as a mix between Raid 3 and Raid 5, having the dedicated parity of Raid 3 and the block level striping of Raid 5. Again, you'll probably need a hardware Raid controller for this level. Also, the dedicated parity drive continues to slow down performance in this level as well.

Raid 5
Raid 5 uses block level striping and distributed parity. This level tries to remove the bottleneck of the dedicated parity drive. With the use of a distributed parity algorithm, this level writes the data and parity data across all the drives. Basically, the blocks of data are used to create the parity blocks which are then stored across the array. This removes the bottleneck of writing to just one parity drive. However, the parity information still has to be calculated and written whenever a write occurs, so the slowdown involved with that still applies. The fault tolerance is maintained by separating the parity information for a block from the actual data block. This way when one drive goes, all the data on that drive can be rebuilt from the data on the other drives. Recovery is more complicated than usual because of the distributed nature of the parity. Just as in Raid 4, the stripe size can be changed to suit the needs of the application. Also, using a hardware controller is probably the more practical solution. Raid 5 is one of the most popular Raid levels being used today. It appears to be the best combination of performance, redundancy, and storage efficiency.

Raid 0+1
This combination uses Raid 0 for it's high performance and Raid 1 for it's high fault tolerance. Let's say you have 8 hard drives. You can split them into 2 arrays of 4 drives each, and apply Raid 0 to each array. Now you have 2 striped arrays. Then you would apply Raid 1 to the 2 striped arrays and have one array mirrored on the other. If a hard drive in one striped array fails, the entire array is lost. The other striped array is left, but contains no fault tolerance if any of the drives in it fail.

Raid 1+ 0
Raid 1+0 applies Raid 1 first then Raid 0 to the drives. To apply Raid 1, you split the 8 drives into 4 sets of 2 drives each. Now each set is mirrored and has duplicate information. To apply Raid 0, you then stripe across the 4 sets. In essence, you have a striped array across a number of mirrored sets. This combination has better fault tolerance than Raid 0+1. As long as one drive in a mirrored set is active, the array can still function. So theoretically you can have up to half the drives fail before you lose everything, as opposed too nly two drives in Raid 0+1.

In conclusion
Ok now that you know the different Raid levels and configurations, why would you even bother? Well it really all depends on your application and the Raid level you use. However, in general using Raid provides data redundancy, fault tolerance, increased capacity, and increased performance. Data redundancy protects the data from hard drive failures. This benefit is good for companies or individuals that have critical or important data to protect, or just anyone that's paranoid about losing their gigabytes of data. Fault tolerance goes hand in hand with redundancy in providing a better over-all storage system. The only Raid level that does not have any form of redundancy or fault tolerance is Raid 0. Raid also provides
increased capacity by combining multiple drives. The efficiency of how the total drive storage is used depends on the Raid level. Usually, levels involving mirroring need twice as much storage to mirror the data. And lastly, the reason most people go to Raid is for the increase in performance. Depending on the Raid level used, the performance increase is different. For applications that need raw speed, Raid is definitely the way to go.

Here is a simple view of Raid:
Mirroring gives you Redundancy …therefore Data security goes up. Write performance goes down due to duplicated writes ( the amount varies by implementation). and read performance goes up, since there are two spindles with duplicated data that can be accessed by the system. In fact, in some implementations, the data that is closest to the read head of a given spindle is chosen for read making the seek and latency time drop dramatically ( note: again this depends on how your system is implemented and how you configure caching algorithms. The main thing to remember here is that the Raid controller writes the same data blocks to each mirrored drive. Each drive or array has the same information in it To set up mirroring the number of drives will have to be in the power of 2 for obvious reasons. The drawback here is that both drives are tied up during the writing process which limits parallelism and can hurt performance. A good Raid controller will only read from one of the drives since the data on both are the same. While the other is used to read, the free drive can be used for other requests. This increases parallelism, which is pretty much the concept behind the performance increase of Raid.

Stripping
Spreading that single file across a bunch-o-drives. Security of data drops ( more spindles & drive mechanics to break) but this gives you almost unlimited size of a “single” logical disk. Add two 60 gig disks get 1 120 gig disk. . Striping improves the performance of the array by distributing the data across all the drives. The main principle behind striping is parallelism. Imagine you have a large file on a single hard drive. If you want to read the file, you have to wait for the hard drive to read the file from beginning to end. Now, if you break the file up into multiple pieces and distribute it across multiple hard drives, you have all these drives reading a part of the file at the same time. You only have to wait as long as it takes to read each piece since the drives are working in parallel. The same is true if you were writing a large file to a disk. Transfer performance is greatly increased. The more hard drives you have, the greater the increase in performance. The stripe size is a largely debated topic. There is no ideal stripe size but certain sizes work best with certain applications. The performance effects of increasing or decreasing stripe size are apparent. Using a small stripe size will enable files to be broken up more and distributed across the drives. The transfe performance will increase due to the increased parallelism. However, this also increases the randomness of the position of each piece of the file. As you probably guessed already, using a large stripe size will do the opposite of decreasing the size. The data will be less distributed and transfer performance is decreased. The randomness is decreased as well. The best way to find out the right stripe size for your particular application is to experiment. Start out with a medium stripe size and try decreasing or increasing the siz and recording the difference in over-all performance. Remember, if you want to move or transfer a file somewhere, the controller accesses both drives simultaneously, which is where the performance gain kick in. It only takes half the time to transfer the file. If you increase the number of hard drives, the file will be transferred in 1/Nth the time it takes to transfer from 1 hard drive .

Mirroring and stripping
Add them both together data redundancy is up, security of data is better, read performance goes up, much faster ( depending on configuration again), write performance suffers depending on implementation

Sorry for being so long winded…. It just seemed that there is some confusion with regard to Raid capabilities and benefits.

Since metadata like Dialnorm is becoming more and more important in post, here's some data from Dolby to spark some thinking...


Dolby ®
Metadata Guide

Dolby, Pro Logic, and the double-D symbol are registered trademarks of Dolby Laboratories.
Surround EX is a trademark of Dolby Laboratories. Issue 3
© 2005 Dolby Laboratories, Inc. All rights reserved. S05/14660/16797
Dolby Laboratories, Inc.

A Guide to Dolby Metadata

Metadata provides unprecedented capability for content producers to deliver the highest quality audio to consumers in a range of listening environments. It also provides choices that allow consumers to adjust their settings to best suit their listening environments. In this document, we first discuss the concept of metadata:
• Metadata overview
We then discuss the three factors controlled by metadata that most directly affect the consumer’s experience:
• Dialogue level
• Dynamic range control (DRC)
• Downmixing
Finally, we define each of the adjustable parameters, and provide sample combinations:
• Individual parameters
• Metadata combinations
1 Metadata Overview
Dolby® Digital and Dolby E are both data-rate reduction technologies that use metadata. Metadata is carried in the Dolby Digital or Dolby E bitstream, describing the encoded audio and conveying information that precisely controls downstream encoders and decoders. In normal operation, the encoded audio and metadata are carried together as a data stream on two regular digital audio channels (AES3, AES/EBU, or S/PDIF). Metadata can also be carried as a serial data stream between Dolby E and/or Dolby Digital equipment. Metadata allows content providers unprecedented control over how original program material is reproduced in the home. Dolby Digital is a transmission bitstream (sometimes called an emission bitstream) intendedfor delivery to the consumer at home through a medium such as DTV or
by one metadata stream. The consumer’s Dolby Digital decoder reproduces the program audio according to the metadata parameters set by the program creator, and according to settings for speaker configuration, bass management, and dynamic range that are chosen by the consumer to match his specific home theater equipment and environmental conditions.

Dolby E is a distribution bitstream capable of carrying up to eight channels of encoded audio and metadata. The number of programs ranges from one single program (Program Config: 5.1) to eight individual programs on a single Dolby E stream (Program Config: 8 × 1). Each program is discrete, with its own metadata in the Dolby E stream. Some metadata parameters in a Dolby E stream automatically configure a Dolby Digital encoder at the point of transmission, while others affect only the consumer’s Dolby Digital decoder operation. Dolby E is a professional technology used for broadcast applications, such as program origination and distribution; the Dolby E bitstream carries the entire metadata parameter set. Dolby Digital, used for consumer applications, such as transmission to the home or for DVD authoring, employs a subset of the full metadata parameter set called Dolby Digital metadata; the Dolby Digital bitstream carries only
those parameters necessary for proper decoding by the consumer. Metadata is first inserted during program creation or mastering, and is carried through transmission in a broadcast application or directly onto a DVD. The metadata provides control over how the encoded bitstream is treated at each step on the way to the consumer’s decoder.

Here’s an example of how it works:
In a broadcast truck parked outside a football stadium, the program mixer chooses the appropriate metadata for the audio program being created. The resulting audio program, together with metadata, is encoded as Dolby E and sent to the television station via fiber, microwave, or other transmission link. At the receiving end of this transmission, the Dolby E stream is decoded back to baseband audio and metadata.
The audio program and the metadata are monitored, altered, or re-created as other elements of thprogram are added in preparation for broadcast. This new audio program/metadata pair, reencoded as Dolby E, leaves the postproduction studio and passes through the television station to Master Control, where many incoming Dolby E streams are once again decoded back to their individual baseband digital audio/metadata programs. The audio program/metadata pair that is selected to air is sent to the transmission Dolby Digital encoder, which encodes the incoming audio program according to the metadata stream associated with it, thereby simplifying the transmission process. Finally, the Dolby Digital signal is decoded in the consumer’s home, with metadata providing the information for that decoding process. Through the use of metadata, the mixer in the truck has been able to control the home decoder for the sporting event, while segments such as news breaks, commercials, and station IDs are similarly decoded, each using metadata carried within each individual segment.

This control, however, requires the producer to set the metadata parameters correctly, since they affect important aspects of the audio—and can seriously compromise the final product if set improperly. Although most metadata parameters are transparent to consumers, certain parameters affect the output of a home decoder, such as downmixing for a specific speaker configuration, or when the consumer chooses
Dynamic Range Control to avoid disturbing family and neighbors.

TheDolbyEbitstream containsboththe5.1-and two-channel programs’ encodedaudio,andeach program'smetadata. TheDolbyDigital bitstream containsasingleprogram’s encodedaudioand
correspondingmetadata.

Metadata Flow from Production to Consumer
In the simplest terms, there are two functional classifications of metadata: Professional: These parameters are carried only in the Dolby E bitstream. They are used to automatically configure a downstream Dolby Digital encoder, allowing maximum control by the content producer over how the encoded bitstream istreated at each step on the way to the consumer’s decoder. Consumer: These parameters are carried in both the Dolby E and the Dolby Digital bitstream. The consumer’s Dolby Digital decoder uses these parameters to create the best possible audio program possible on each consumer’s playback system. Consumer parameters include the DRC values, which are ultimately enabled by the end user’s selection, as discussed in Section 3, Dynamic Range Control.

Both types of metadata can be examined, modified, or passed through during encoding. A/D Converter Type

Special Parameters
There are other professional parameters included in the Dolby E bitstream that are not under direct user control, such as Timecode and Pitch Shift.

Timecode
Dolby E bitstreams carry timecode information in hours:minutes:seconds:frames format.

Pitch Shift
The Pitch Shift parameter can be generated automatically by a Dolby E decoder to control the Dolby Model 585 Time Scaling Processor. If the input to the Dolby E decoder is not at normal play speed (as with varispeed or program play), then the Pitch Shift Code parameter indicates the amount of audio pitch shifting required to restore the original program pitch.

Dialogue Level
Dialogue Level (also known as dialogue normalization or dialnorm) is perhaps the single most important metadata parameter. The Dialogue Level setting represents the long-term A-weighted average level of dialogue within a presentation, Leq(A). This level can be quantified with the Dolby Model LM100 Broadcast Loudness Meter. When received at the consumer’s Dolby Digital decoder, this parameter setting determines a level shift in the decoder that sets, or normalizes, the average audio output of the decoder to a preset level. This aids in matching audio volume between program sources. In broadcast transmission, the proper setting of Dialogue Level ensures that the consumer receives a standard listening level, so switching channels or watching a television program through the commercial breaks doesn’t require adjusting the volume. Using the same standard for all content, whether conveyed by broadcast television, DVD, or other media, enables the consumer to switch between sources and programs while maintaining a comfortable and consistent listening level. The proper setting of the Dialogue Level parameter also enables the Dynamic Range Control profiles chosen by the content producer to work as intended in less-than- optimal listening environments, and is essential in any content production, whether it is for transmission in a broadcast stream or for direct distribution to consumers, as with DVDs.
Note: Programs without dialogue, such as an all-music program, still require a careful setting of the Dialogue Level parameter. When setting the parameter for such content, it is useful to compare the program to the level of other programs. The goal is to allow the consumer to switch to your program without having to adjust the volume control.

The Scale
The scale used in the Dialogue Level setting ranges in 1 dB steps from –1 to –31 dB. Contrary to what you might assume at first, a setting of –31 represents no level shift in the consumer’s decoder, and –1 represents the maximum level shift. Here’s why: Dolby Digital consumer decoders normalize the average output level—that is, the output level averaged over time using the equivalent loudness method, Leq(A)—
to –31 dBFS (31 dB below 0 dB full-scale digital output) by applying a shift in level based on the Dialogue Level parameter setting. Note: The –31 dBFS Leq(A) should not be confused with the station reference level (often –18 or –20 dBFS). It is common to have different Leq(A) values for program material that has the same reference level. An average loudness level of –31 dBFS Leq(A) is quite compatible with facilities running at a
variety of reference levels. When a decoder receives an input signal with a Dialogue Level setting of –31, it applies no level shift to the signal because this indicates to the decoder that the signal already matches the target level and therefore requires no shift. In contrast, a louder program requires a shift to match the –31 dB standard. When the Dialogue Level parameter setting is –21, the decoder applies a 10 dB level shift to the signal. When the setting is –11, it applies a 20 dB level shift, and so on.
A Simple Rule:
31 + (dialogue level value) = Shift applied
Example:
31 + (–21) = 10 dB

The most important point to remember is that in setting the Dialogue Level parameter, you are providing your listener with an essential service. For your listeners, setting this level properly means:
• The volume level is consistent with other programs.
• The DRC profiles you make available to them work as you intend.
Once dialogue level is set, you can set up DRC profiles to further benefit the consumer.

Dynamic Range Control
Different home listening environments present a wide range of requirements for dynamic range. Rather than simply compressing the audio program at the transmission source to work well in the poorest listening environments, Dolby Digital encoders calculate and send Dynamic Range Control (DRC) metadata with the signal. This metadata can then be applied to the signal by the decoder to reduce the signal’s
dynamic range. Through the proper setting of DRC profiles during the mastering process, the content producer can provide the best possible presentation of program content in virtually any listening environment, regardless of the quality of the equipment, number of channels, or ambient noise level in the consumer’s home.Many Dolby Digital decoders offer the consumer the option of defeating the Dynamic
Range Control metadata, but some do not. Decoders with six discrete channel outputs (full 5.1-channel capability) typically offer this option. Decoders with stereo, mono, or RF-remodulated outputs, such as those found on DVD players and set-top boxes, often do not. In these cases, the decoder automatically applies the most appropriateDRC metadata for the decoder’s operating mode. The Dolby Digital stream carries metadata for the two possible operating modes in the decoder. The operating modes are known as Line mode and RF mode due to the type of output they are typically associated with. Line mode is typically used on decoders with six- or two-channel line-level outputs and RF mode is used on decoders that have an RF-remodulated output. Full-featured decoders allow the consumer to select whether to use DRC and if so, which operating mode to use. The consumer sees options such as Off, Light Compression, and Heavy Compression instead of None, Line mode, and RF mode. Advanced decoders may also allow custom scaling of the DRC metadata. All that needs to be done during metadata authoring, or encoding, is selection of the dynamic range control profiles for Line mode and RF mode. The profiles are described in the following sections.
Note: While the use of DRC modes during decoding is a consumer-selectable feature, the Dialogue Levelparameter setting is not. Therefore, setting the Dialogue Level parameter properly is essential before previewing a DRC profile.

Line Mode
Line mode offers these features:
• Low-level signal boost compression scaling is allowed.
• High-level signal cut compression scaling is allowed when not downmixing.
• The normalized dialogue level is reproduced from the decoder at a constant
loudness level of –31 dBFS Leq(A), assuming the Dialogue Level parameter
is set correctly.
Line-level or power-amplified outputs from two-channel set-top decoders, two- channel digital televisions, 5.1-channel digital televisions, Dolby Digital A/V surround decoders, and outboard Dolby Digital adapters use Line mode.
Consumer control of the dynamic range is limited when downmixing. Products with stereo or mono outputs do not usually allow consumer scaling of Line mode. This is because these devices are usually downmixing (for example, when receiving a 5.1-channel signal). However, in these products, the consumer may have a choice between Line mode and RF mode.

RF Mode
In RF mode, high- and low-level compression scaling is not allowed. When RF mode is active, that compression profile is always fully applied. RF mode is designed for products (such as set-top boxes) that generate a downmixed signal for connection to the RF/antenna input of a television set; however, it is also useful in situations where heavy DRC is required—for example, when small PC speakers are used for DVD playback. In RF mode, the overall program level is raised 11 dB, this results in dialogue being reproduced at a level of –20 dBFS Leq(A), while the peaks are limited to prevent signal overload in the D/A converter. By limiting headroom, severe overmodulation of television receivers is prevented. The 11 dB gain provides an average loudness level that compares well with existing analog television broadcasts. In some situations it may be necessary to further constrain signal peaks above the average dialogue level so that there is less than 20 dB headroom. The selection of a suitable RF mode profile achieves this.

Dynamic Range Control Profiles Six preset DRC profiles are available to content producers: Film Light, Film
Standard, Music Light, Music Standard, Speech, and None.

• Film Light
Max Boost: 6 dB (below –53 dB)
Boost Range: –53 to –41 dB (2:1 ratio)
Null Band Width: 20 dB (–41 to –21 dB)
Early Cut Range: –26 to –11 dB (2:1 ratio)
Cut Range: –11 to +4 dB (20:1 ratio)

• Film Standard
Max Boost: 6 dB (below –43 dB)
Boost Range: –43 to –31 dB (2:1 ratio)
Null Band Width: 5 dB (–31 to –26 dB)
Early Cut Range: –26 to –16 dB (2:1 ratio)
Cut Range: –16 to +4 dB (20:1 ratio)

• Music Light (No early cut range)
Max Boost: 12 dB (below –65 dB)
Boost Range: –65 to –41 dB (2:1 ratio)
Null Band Width: 20 dB (–41 to –21 dB)
Cut Range: –21 to +9 dB (2:1 ratio)
Dolby Laboratories, Inc. Metadata Guide

• Music Standard
Max Boost: 12 dB (below –55 dB)
Boost Range: –55 to –31 dB (2:1 ratio)
Null Band Width: 5 dB (–31 to –26 dB)
Early Cut Range: –26 to –16 dB (2:1 ratio)
Cut Range: –16 to +4 dB (20:1 ratio)

• Speech
Max Boost: 15 dB (below –50 dB)
Boost Range: –50 to –31 dB (5:1 ratio)
Null Band Width: 5 dB (–31 to –26 dB)
Early Cut Range: –26 to –16 dB (2:1 ratio)
Cut Range: –16 to +4 dB (20:1 ratio)

None
No DRC profile selected. The dialogue level parameter (dialnorm) is still applied. These choices are available to the content producer for both Line mode and RF mode. The content producer chooses which of these profiles to assign to each mode; when the consumer or decoder selects a DRC mode, the profile chosen by the producer is applied. In addition to the DRC profile, metadata can limit signal peaks to prevent clipping during downmixing. This metadata, known as overload protection, is inserted by the encoder only if necessary. For example, consider a 5.1-channel program with signals at digital full-scale on all channels being played through a stereo, downmixed line- level output. Without some form of attenuation or limiting, the output signal would obviously clip. Correct setting of the Dialogue Level and DRC profiles normally prevents clipping and unnecessary application of automatic overload protection. Note: DRC profile settingsare dependent on an accurate dialogue level setting. Improper setting of the dialogue level parameter may result in excessive and audible application of overload-protection limiting.

Downmixing
Downmixing is a function of Dolby Digital that allows a multichannel program to be reproduced over fewer speaker channels than the number for which the program is optimally intended. Simply put, downmixing allows consumers to enjoy a DVD or digital television broadcast without requiring a full-blown home theater setup.
As with stereo mixing where the mix is monitored in mono on occasion to maintain compatibility, multichannel audio mixing requires the engineer to reference the mix to fewer speaker channels to ensure compatibility in downmixing situations. In this way, Dolby Digital, using the metadata parameters thatcontrol downmixing, is an “equal opportunity technology,” in that every consumer who receives the
Dolby Digital data stream can enjoy the best audio reproduction possible, regardless of the playback system.
It is important to consider the output signals from each piece of equipment that can receive a Dolby Digital program in the home. Table 2 shows the output types from different equipment.

Set-top boxes, used to receive terrestrial, cable, or satellite digital television, typically offer an analog mono signal modulated on the RF/Antenna output, a line-level analog stereo signal, and an optical or coaxial digital output. DVD players offer an analog stereo and a digital output, and some offer a six-channel analog output (for a 5.1- channel presentation). Portable DVD players offer analog stereo, headphone, and
digital outputs. DVD players in computers and game consoles offer a digital output as well as analog stereo, headphone, and possibly six-channel analog outputs. 5.1- channel amplifiers, decoders, and receivers have six-channel analog outputs and possibly six speaker-level outputs.

In all of these cases, a Dolby Digital decoder creates the analog audio output signal. In the case of the set-top box or DVD player, the analog stereo output is a downmixed version of the Dolby Digital data stream. The digital output delivers the Dolby Digital data stream to either a downstream decoder or an integrated amplifier with Dolby Digital decoding.

In each of these devices, the analog stereo output is one of two different stereo downmixes. One type is a stereo-compatible Dolby Surround downmix, of the multichannel source program that is suitable for Dolby Surround Pro Logic® decoding. This kind of downmix is also called Pro Logic or Left total/Right total
(Lt/Rt). The other type is a simple stereo representation (called Left only/Right only, or Lo/Ro) suitable for playback on a stereo hi-fi or on headphones, and from which a mono signal is derived for use on an RF/Antenna output. The difference between the downmixes is how the Surround channels are handled. The Lt/Rt downmix sums the Surround channels and adds them, in-phase to the Left channel and out-of-phase to
the Right channel. This allows a Dolby Surround Pro Logic decoder to reconstruct the L/C/R/S channels for a Pro Logic home theater. The Lo/Ro downmix adds the Left and Right Surround channels discretely to the Left and Right speaker channels, respectively. This preserves the stereo separation for stereo-only monitoring and produces a mono-compatible signal. In all downmixes, the LFE channel is not
included.
On most home equipment, the consumer can use the product’s user interface to choose the appropriate stereo output for his playback system. The mono signal feeding the RF/Antenna output is usually derived from the Lo/Ro downmix. There are separate metadata parameters that govern the Lo/Ro and Lt/Rt downmixes. Certain metadata parameters allow the engineer to select how the stereo downmix is constructed and which stereo analog signal is preferred, but Lt/Rt is the default selection in all consumer decoders. See Section 5, Parameter Definitions, for more information on individual parameters.
During downmixing, as we have seen, the adjustment of Dynamic Range Control parameters is limited. Broadly speaking, the stereo outputs use the Line mode compression profile while the mono signal uses RF mode compression. As with dynamic range control, downmixing is ultimately dependent upon each consumer’s unique listening environment.
While the engineer must optimize the multichannel mix for reproduction in an ideal listening environment, it is also important to preview the mix in downmixing conditions to ensure compatibility with different playback systems when selecting the downmixing metadata parameters. These previews can be achieved in real time using the DP570 Multichannel Audio Tool.

5 Parameter Definitions
This section explains both professional and consumer metadata parameters in greater detail.
Metadata parameters include:
• Universal parameters
• Extended Bitstream Information (Extended BSI) parameters
Extended BSI parameters are active only when both the producer chooses to use them and the consumer’s decoder is capable of reading them. All decoders can successfully decode a metadata stream without Extended BSI parameters, and Extended BSI parameters translate seamlessly to decoders that read only universal parameters. Note: Universal parameters include both professional and consumer metadata.

5.1 Universal Parameters
All universal parameters are supported by Dolby E encoders and decoders; all except Program Configuration and Program Description Text are supported by all Dolby Digital encoders and decoders.

Program Configuration
This parameter determines how the audio channels are grouped within a Dolby E bitstream. Up to eight channels can be grouped together in individual programs, where each program contains its own metadata. The default setting is 5.1 + 2. Program Description Text This parameter is a 32-character ASCII text field that allows the metadata author to enter a description of the audio program. For example, this field may contain the name of the program (Movie Channel Promo), a description of the program source (Football Main Feed), or the program language (Danish).

Dialogue Level
The Dialogue Level parameter is discussed in Section 2, Dialogue Level.

Channel Mode
This parameter (also known as Audio Coding mode) indicates the active channels within the encoded bitstream and affects both the encoder and consumer decoder. This parameter instructs the encoder which inputs to use for this particular program; it tells the decoder what channels are present in this program so the decoder can deliver the audio to the correct speakers. The setting is described as X/Y, where X is the number of front channels (Left, Center, Right) and Y the number of rear (Surround) channels. The availability of certain channel modes depends on the Dolby Digital encoder data rate and whether the LFE channel is present. For example, you can’t have a mono stream with an LFE channel (1.1!) or a 3/2 stream at 96 kbps. Appropriate data rates are shown in the definition of each setting.

Note: The presence of the LFE channel is indicated through a different metadata parameter (see LFE Channel).

Channel
Mode
Setting Definition and Data Rate 1+1 Dual mono (not valid for DTV broadcast or DVD production)
1/0 Mono From 56 kbps, usually 96 kbps
2/0 Stereo From 96 kbps, usually 192 kbps
3/0 From 256 kbps
2/1 From 256 kbps
3/1 From 320 kbps
2/2 From 320 kbps
3/2 From 384 kbps, often 448 kbps

LFE Channel
The status of the LFE Channel parameter indicates to a Dolby Digital encoder whether an LFE Channel is present within the bitstream. Channel mode determines whether the LFE Channel parameter can be set. You must have at least three channels to be able to add an LFE channel.
LFE Channel Setting Enabled Disabled

Bitstream Mode
This parameter describes the audio service contained within the Dolby Digital bitstream. A complete audio program may consist of a main audio service (a complete mix of all the program audio), an associated audio service comprising a complete mix, or one main service combined with an associated service. To form a complete audio program, it may be (but rarely is) necessary to decode both a main service and an associated service using a maximum total bit rate of 512 kbps. Refer to the Guide to the Use of the ATSC Digital Television Standard, Document A/54 (see www.atsc.org) for further information. Although a detailed description of each option follows, in practice, most programming uses the default setting, Complete Main. An example of an exception to this rule is a special karaoke DVD, or an emergency service within digital television.


Bitstream Mode Setting Definition
Complete Main
(CM)
CM flags the bitstream as the main audio service for the program and indicates that all elements are present to form a complete audio program. Currently, this is the most common setting. The CM service may contain
from one (mono) to six (5.1) channels. Main M&E (ME) The bitstream is the main audio service for the program, minus a dialogue channel. The dialogue channel, if any, is intended to be carried by an associated dialogue service. Different dialogue services can be associated with a single ME service to support multiple languages.

Assc. Visual Imp.
(VI)
This is typically a single-channel program intended to provide a narrative description of the picture content service. The VI service may also be a complete mix of all program channels, comprising up to six channels. Assc. Hear Imp. (HI) This is typically a single-channel program intended to convey audio that has been processed for increased intelligibility and decoded along with the main audio service. The HI service may also be a complete mix of all program channels, comprising up to six channels.

Assc. Dialogue (D) This is typically a single-channel program intended to provide a dialogue
channel for an ME service. If the ME service contains more than two channels, the D service is limited to only one channel; if the ME service is two channels, the D service can be a stereo pair. The appropriate channels of each service are mixed together (requires special decoders).

Assc. Commentary (C)
This is typically a single-channel program intended to convey additional commentary that can be optionally decoded along with the main audio service. This service differs from a dialogue service because it contains an optional, rather than a required, dialogue channel. The C service may also be a complete mix of all program channels, comprising up to six channels.

Assc. Emergency (E)
This is a single-channel service that is given priority in reproduction. When the E service appears in the bitstream, it is given priority in the decoder and the main service is muted.

Assc. Voice Over
(VO) This is a single-channel service intended to be decoded and mixed to the Center channel (requiresspecial decoders).

Main Sv Karaoke (K)
The bitstream is a special service for karaoke playback. In this case, the Left and Right channels contain music, the Center channel has a guide melody, and the Left and Right Surround channels carry optional backing vocals.
Line Mode Compression Profile
Line mode is discussed in Section 3, Dynamic Range Control.
RF Mode Compression Profile
RF mode is discussed in Section 3, Dynamic Range Control.
Dolby Laboratories, Inc. Metadata Guide

RF Overmodulation Protection
This parameter is designed to protect against overmodulation when a decoded Dolby Digital bitstream is RFmodulated. When enabled, the Dolby Digital encoder includes
pre-emphasis in its calculations for RF Mode compression. The parameter has no effect when decoding using Line mode compression. Except in rare cases, this parameter should be disabled.

RF Overmodulation Protection Setting Enabled Disabled

Center Downmix Level
When the encoded audio has three front channels (L, C, R), but the consumer has only two front speakers (left and right), this parameter indicates the nominal downmix level for the Center channel with respect to the Left and Right channels. Dolby Digital decoders use this parameter during downmixing in Lo/Ro mode when Extended BSI parameters are not active.

Center Downmix Level
Setting Definition
0.707 (–3 dB) default The Center channel is attenuated 3 dB and
sent to the Left and Right channels.
0.596 (–4.5 dB) The Center channel is attenuated 4.5 dB and
sent to the Left and Right channels.
0.500 (–6 dB) The Center channel is attenuated 6 dB and
sent to the Left and Right channels.

RF Overmodulation Protection Setting Enabled Disabled

Center Downmix Level
When the encoded audio has three front channels (L, C, R), but the consumer has only two front speakers (left and right), this parameter indicates the nominal downmix level for the Center channel with respect to the Left and Right channels. Dolby Digital decoders use this parameter during downmixing in Lo/Ro mode when Extended BSI parameters are not active.

Center Downmix Level
Setting Definition
0.707 (–3 dB) default The Center channel is attenuated 3 dB and
sent to the Left and Right channels.
0.596 (–4.5 dB) The Center channel is attenuated 4.5 dB and
sent to the Left and Right channels.
0.500 (–6 dB) The Center channel is attenuated 6 dB and
sent to the Left and Right channels.

Surround Downmix Level
When the encoded audio has one or more Surround channels, but the consumer does not have surround speakers, this parameter indicates the nominal downmix level for the Surround channel(s) with respect to the Left and Right front channels. Dolby Digital decoders use this parameter during downmixing in Lo/Ro mode when Extended BSI parameters are not active.

Surround Downmix Level Setting Definition
0.707 (–3 dB) default The Left and Right Surround channels are each attenuated 3 dB and sent to the Left and Right front channels, respectively. 0.5 (–6 dB) Same as above, but the signal is attenuated 6 dB.
0 (–999 dB) The Surround channel(s) are discarded.

Dolby Surround Mode
This parameter indicates to a Dolby Digital decoding product that also contains a Dolby Pro Logic decoder (for example a 5.1-channel amplifier), whether or not the two-channel encoded bitstream contains a Dolby Surround (Lt/Rt) program that requires Pro Logic decoding. Decoders can use this flag to automatically switch on Pro Logic decoding as required.

Dolby Surround Mode
Setting Definition
Not Dolby Surround The bitstream contains information that was not
encoded in Dolby Surround. Dolby Surround The bitstream contains information that was encoded in Dolby Surround. After Dolby Digital decoding, the bitstream is decoded using Pro Logic.
Not Indicated There is no indication either way.

Audio Production Information
This parameter indicates whether the mixing level and room type values are valid. If Yes, then a receiver or amplifier could use these values as described below. If No, then the values in these fields are invalid. In practice, only high-end consumer equipment implements these features.

Audio Production Information Setting Definition Yes Mixing Level and Room Type parameters are valid.
No Mixing Level and Room Type parameters are invalid and should be ignored.

Mixing Level
The Mixing Level parameter describes the peak sound pressure level (SPL) used during the final mixing session at the studio or on the dubbing stage. The parameter allows an amplifier to set its volume control such that the SPL in the replay environment matches that of the mixing room. This control operates in dialogue level control, and is best thought of as the final volume setting on the consumer’s equipment. This value can be determined by measuring the SPL of pink noise at studio reference level and then adding the amount of digital headroom above that level. For example, if 85 dB equates to a reference level of –20 dBFS; the mixing level is 85 + 20, or 105 dB.

Mixing Level Setting
80 to 111 dB in 1 dB increments

Room Type
The Room Type parameter describes the equalization used during the final mixing session at the studio or on the dubbing stage. A Large room is a dubbing stage with the industry standard X-curve equalization; a Small room has flat equalization. This parameter allows an amplifier to be set to the same equalization as that heard in the final mixing environment.

Room Type Setting
Not Indicated
Large
Small

Copyright Bit
This parameter indicates whether the encoded Dolby Digital bitstream is copyright protected. It has no effect on Dolby Digital decoders and its purpose is purely to provide information.
Copyright Bit Setting Yes No

Original Bitstream
This parameter indicates whether the encoded Dolby Digital bitstream is the master version or a copy. It has no effect on Dolby Digital decoders and its purpose is purely to provide information.
Original Bitstream Setting Yes No

Note: The parameters DC Filter, Lowpass Filter, LFE Lowpass Filter, Surround 3 dB Attenuation, and Surround Phase Shift appear after the Extended BSI parameters on Dolby E and Dolby Digital equipment menus.

DC Filter
This parameter determines whether a DC-blocking 3 Hz highpass filter is applied to the main input channels of a Dolby Digital encoder prior to encoding. This parameter is not carried to the consumer decoder. It is used to remove DC offsets in the program audio and would only be switched off in exceptional circumstances.

DC Filter Setting Enabled Disabled

Lowpass Filter
This parameter determines whether a lowpass filter is applied to the main input channels of a Dolby Digital encoder prior to encoding. This filter removes high- frequency signals that are not encoded. At the suitable data rates, this filter operates above 20 kHz. In all cases it prevents aliasing on decoding and is normally switched on. This parameter is not passed to the consumer decoder.

Lowpass Filter Setting Enabled Disabled

LFE Lowpass Filter
This parameter determines whether a 120 Hz eighth-order lowpass filter is applied to the LFE channel input of a Dolby Digital encoder prior to encoding. It is ignored if the LFE channel is disabled. This parameter is not sent to the consumer decoder. The filter removes frequencies above 120 Hz that would cause aliasing when decoded. This filter should only be switched off if the audio to be encoded is known to have no signal above 120 Hz.
LFE Lowpass Filter Setting Enabled Disabled

Surround 3 dB Attenuation
The Surround 3 dB Attenuation parameter determines whether the Surround channel(s) are attenuated 3 dB before encoding. The attenuation actually takes place inside the Dolby Digital encoder. It balances the signal levels between theatrical Dolby Laboratories, Inc. Metadata Guide mixing rooms (dubbing stages) and consumer mixing rooms (DVD or TV studios). Consumer mixing rooms are calibrated so that all five main channels are at the same sound pressure level (SPL). To maintain compatibility with older film formats,
theatrical mixing rooms calibrate the SPL of the Surround channels 3 dB lower than the front channels. The consequence is that signal levels on tape are 3 dB louder. Therefore, to convert from a theatrical calibration to a consumer mix, it is necessary to reduce the Surround levels by 3 dB by enabling this parameter.
Surround 3 dB Attenuation Setting Enabled Disabled

Surround Phase Shift
This parameter causes the Dolby Digital encoder to apply a 90-degree phase shift tothe Surround channels. This allows a Dolby Digital decoder to create an Lt/Rt downmix simply. For most material, the phase shift has a minimal impact when the Dolby Digital program is decoded to 5.1 channels, but it provides an Lt/Rt output that can be decoded with Pro Logic to L, C, R, S, if desired. However, for some phase- critical material (such as music) this phase shift is audible when listening in a 5.1- channel format. Likewise, some material downmixes to a satisfactory Lt/Rt signal without needing this phase shift. It is therefore important to balance the needs of the 5.1 mix and the Lt/Rt downmix for each program. The default setting is Enabled.
Surround Phase Shift Setting Enabled Disabled

5.2 Extended Bitstream Information Parameters
In response to requests from content producers, Dolby Laboratories modified the definitions of several metadata parameters from their original definition as described in ATSC document A/52. The revised definitions allow more information to be carried about the audio program and also allow more choices for stereo downmixing. When the metadata parameters carried in Dolby Digital were first described, they were generically called Bitstream Information, or BSI. We refer to the additional parameter definitions as Extended BSI. Because the revised definitions affect metadata parameters that were not used by the consumer decoders, all decoders will be compatible with the revised bitstream. Newer decoders that are programmed to detect and decode the new parameters will be able to implement the new features Extended BSI provides.

Products that allow emulation of the effects of metadata, such as the DP570, normally have a feature that allows emulation of a new (or compliant) decoder or a legacy decoder.

Preferred Stereo Downmix Mode
This parameter allows the producer to select either the Lt/Rt or the Lo/Ro downmix in a consumer decoder that has stereo outputs. Consumer receivers are able to override this selection, but this parameter provides the opportunity for a 5.1-channel soundtrack to play in Lo/Ro mode without user intervention. This is especially useful on music material.

Preferred Stereo Downmix Mode
Setting
Not Indicated
Lt/Rt Preferred
Lo/Ro Preferred

Lt/Rt Center Downmix Level
This parameter indicates the level shift applied to the Center channel when adding to the left and right outputs as a result of downmixing to an Lt/Rt output. Its operation is similar to the center downmix level in the universal metadata.

Lt/Rt Center Downmix Level Setting
1.414 (+3.0 dB)
1.189 (+1.5 dB)
1.000 (0.0 dB)
0.841 (–1.5 dB)
0.707 (–3.0 dB)
0.595 (–4.5 dB)
0.500 (–6.0 dB)
0.000 (–999 dB)

Dolby Laboratories, Inc. Metadata Guide
23
Lt/Rt Surround Downmix Level
This parameter indicates the level shift applied to the Surround channels when downmixing to an Lt/Rt output. Its operation is similar to the surround downmix level in the universal metadata.

Lt/Rt Surround Downmix Level Setting
0.841 (–1.5 dB)
0.707 (–3.0 dB)
0.595 (–4.5 dB)
0.500 (–6.0 dB)
0.000 (–999 dB)

Lo/Ro Center Downmix Level
This parameter indicates the level shift applied to the Center channel when adding to the left and right outputs as a result of downmixing to an Lo/Ro output. When Extended BSI parameters are active, this parameter replaces the Center Downmix Level parameter in the universal parameters.

Lo/Ro Center Downmix Level Setting
1.414 (+3.0 dB)
1.189 (+1.5 dB)
1.000 (0.0 dB)
0.841 (–1.5 dB)
0.707 (–3.0 dB)
0.595 (–4.5 dB)
0.500 (–6.0 dB)
0.000 (–999 dB)

Lo/Ro Surround Downmix Level
This parameter indicates the level shift applied to the Surround channels when downmixing to an Lo/Ro output. When Extended BSI parameters are active, this parameter replaces the Surround Downmix Level parameter in the universal parameters.

Lo/Ro Surround Downmix Level Setting
0.841 (–1.5 dB)
0.707 (–3.0 dB)
0.595 (–4.5 dB)
0.500 (–6.0 dB)
0.000 (–999 dB)

Surround EX Mode
This parameter is used to identify the encoded audio as material encoded in Surround
EX TM
. This parameter is only used if the encoded audio has two Surround channels. An amplifier or receiver with Dolby Digital Surround EX decoding can use this parameter as a flag to switch the decoding on or off automatically. The behavior is similar to that of the Dolby Surround Mode parameter.
Surround EX Mode
Not Indicated
Not Surround EX
Dolby Surround EX

A/D Converter Type
This parameter allows audio that has passed through a particular A/D conversion stage to be marked as such, so that a decoder may apply the complementary D/A process.
A/D Converter Type Setting Standard HDCD

Hi, some random data on pullups and pulldowns... I don't remember where I found this, but, I found it interesting...

When some engineer had the bright idea of operating the NTSC television system at 59.94 fields per second rather than a nice simple 60, life became more complicated for the sound guys. Telecine is the culprit!! 24 pictures in one second on film do not fit elegantly into 30 available video frames in one second. To complicate things further the video system does not use real seconds. Consequently the film on a Telecine machine is being played at 23.9 frames per second NOT 24, as it was shot. As we discovered when we introduced Time Code to the 1/4" analog tracks, the production mixer had to use 30 FPS time code during recording so that the post house could slow the tape down 0.1% by synchronizing it with a 29.94 time code. That made picture and sound synchronous, if not on speed. Good enough for TV! The term PULL DOWN was born.

Now we have those cool little DAT machines. They operate with rotary scanning heads and internal clock frequencies and other things reminiscent of Video Recorders. Scary `eh what?

Because of the way DAT machines work, they are so stable that no resolving is necessary to achieve synch. That is, the DAT machine will provide a constant speed playback exceeding that of an analog 1/4" tape with crystal sync. Sync requires that the picture also have the same accuracy in speed or that both picture and sound have the same inaccuracy in speed.

In traditional film production the DAT tape needs only to be played back and transferred to mag film without any resolving, providing that the mag film recorder is running from a crystal controlled motor and the camera was, likewise, "Crystal Sync".

Professional DAT machines provide analog and AES/EBU digital outputs. They also usually operate at one of two sampling frequencies 48KHz and 44.1 KHz. Most dialog recording for motion Pictures and Television is done at 48 KHz. The music industry uses 44.1 KHz. to match their CDs.

Which sampling rate to use should be discussed with the production (particularly the post-production) people before beginning the shoot.

In MOST situations the Time Code DAT machine will behave in much the same way as the Time Code 1/4" analog machine did as long as only the analog output is considered. That is, record with 30FPS TC on the set and the Telecine guys will "PULL DOWN" by synchronizing the playback with 29.94XX Time Code. Picture and Sound match...everyone is happy!!

Some engineer will say "We have all this neat digital sound on the tape, why convert it to analog so soon"? He has a point...

Digital audio signals are at a specific sample rate or frequency. If they vary from their specified frequency nothing comes out. Changing the sampling rate from the nominal 48 or 44.1 should only be attempted by professional drivers under controlled conditions. We equipment manufacturers provide adequate horsepower to get you into trouble....you have to know how to stay safe!

The D2 DIGITAL video recorder has a 48KHz. audio track. It must have 48 KHz. or it accepts nothing. If the production mixer records his track with a 48KHz. sampling rate and the Telecine PULLS it DOWN to 47.952 KHz. there is no audio to record onto the videotape. Conventional practice is to convert the digital to analog, transfer the analog to the D2 and then reconvert the analog back to digital. It works.

A more daring solution would be for the production mixer to "PULL UP" on location, and record at 48.048 KHz. (with a 30FPS time code). Then the "PULL DOWN" in telecine would produce a digital audio signal of 48 KHz. making the D2 machine very happy. Remember that this makes the DAT tape totally useless for normal playback.

Many productions are being edited on digital workstations (AVID, LIGHTWORKS etc.) They accept the picture from videotape that is already "PULLED DOWN" by the Telecine process. The sound needs to be "PULLED DOWN" as well to match the picture. These machines are all digital so they are happiest with a digital audio source. A tape recorded at 48 KHz. and then "PULLED DOWN" to 47.952 is useless to them. They have to go through the digital to analog and analog to digital conversion to get the 48 KHz. they need. Had the production mixer "PULLED UP" and recorded at 48.048 the editor could directly down load the digital audio "PULLED DOWN" to 48 KHz. Remember, the DAT cassette will not provide a normal digital signal at nominal speed when recorded "PULLED UP". For a production where the final product is a videotape, this should work. If it is a film production which will be released in film the technique needs more thought...

The picture is "PULLED DOWN" by 0.1% to edit as video. The Key Code cut list for the negative cutters effectively provides the "PULL UP" to bring the film back to 24FPS. If the sound has been recorded with a "PULL UP" to accommodate the editors, it will be impossible to provide a digital signal at nominal speed to lay back to the film. At this point an analog signal will be all that is required and all is well. If, however, the post guys want to stay in digital (I refuse to use the term "DIGITAL DOMAIN") it would have been prudent to have made a nominal speed (48 KHz.) digital dub during the "PULL DOWN" for the editing process.

Music video is another "can of worms", probably on the set as well as on the sound cart. The music production company gives you a DAT cassette for playback with Time Code at 29.97. You know that the film, shot at 24FPS will be "PULLED DOWN" to 23.9 in telecine, but not the sound (it's already made with 29.97 time code) What to do? "PULL UP" the DAT on playback so the sound is 0.1% faster than original. The actors will move 0.1% faster. In telecine the "PULL DOWN" on the film will make the action correct and properly match the sound. The "PULL UP" on playback also makes the 29.97 Time Code into 30 for the electronic slate. Be sure to use a cable or radio link to the electronic slate as "jamming" the generator will not be accurate.

Perhaps it would be worth putting up with the flicker and the eight field color phase of PAL to be able to work at 25FPS for everything.

Maybe simply changing NTSC to a true 60FPS and shooting film at 30FPS with 3 perf pull down would do it ... anyone ever thought of that.....?

Simply put .... film runs at 24 frames per second through a film camera. (That is 90 feet per minute for 35mm) Film runs at 23.976 frames per second through an NTSC Telecine transfer machine. That is 0.1% slow. To make the sound fit the slowed down picture the sound must also be slowed down by 0.1%. Pretty simple eh? It works with no problem in the analog world. The sound is just a bit slow and slightly lower pitched. No big deal. (unless you are a musician with "Golden Ears") However, in the digital world we should keep our sampling rates constant if we intend to use the AES/EBU digital signal for anything practical. That means that the DAT played back at 0.1% slow will have a digital sampling rate also 0.1% slow. Not Good! The simple solution is .... RECORD the material at 0.1% FAST (48.048KHz. instead of 48.00KHz). When the Telecine transfer process slows it down by 0.1% to match the picture the DAT plays back with a 48.00KHz sampling rate... the AES/EBU standard. In fact the cassette recorded at 48.048 will play back at 48.00 on a standard DAT machine ... great for downloading into a Digital Editing system. No bit rate conversion ... higher quality ... simpler.

What about Time Code? If you use 30 Non Drop code at 48.048 it becomes 29.97 Non Drop code when played back at 48.00. If you use 30 Drop Frame code at 48.048 it becomes 29.97 Drop Frame when played at 48.00.

Sync playbacks for Music Videos are basically the opposite, but, usually using 44.1KHz sampling rates. Some people suggest that the music suppliers should produce a special cassette at 44.056 with 29.97 time code. This, played at 44.1 will produce a playback 0.1% fast with 30 fps time code. It will work but a second DAT at 44.1 will be needed for Telecine or other transfers. It is much simpler to have the music suppliers simply supply a standard 44.1KHz 29.97fps (Drop or Non Drop) time code cassette. They need do nothing special or different; then play it back on the set at 44.144KHz. This speeds it up the necessary 0.1% and provides the 30fps time code for the slate. No one, other than the playback mixer has to do anything special and he is, after all, the "specialist" anyway.


cheers
geo

Tons of essential info on here. Big up!

Introduction to film sound OUTLINE AND NOTES FOR LECTURE


Film is by nature a visual experience
Even so you cannot afford to underestimate the importance of film sound.
A Meaningful sound track is often as complicated as the image on the screen.

The Sound Mix
Sound Mix is comprised of three areas
• the human voice
• sound effects
• music

These three area (STEMS) must be mixed and balanced so as to produce the necessary emphases which in turn create desired effects and emotions.

Film sound is comprised of conventions and innovations.

We expect an acceleration of music during car chases and creaky doors in horror films. Effects of sound are often subtle and noted only by subconscious. foster an awareness of film sound as well as picture in creating the entire story

The Acoustic World
 sound film reveals our acoustic environment
 the speech of things
 whisperings of nature
 muttering of the sea to the din of a city
 roar of machinery to gentle patter of rain on a windowpane
 The meaning of a floorboard creaking in a deserted room
 a bullet whistling past our ear
 the explosion or the car crash that overwhelms our ears
 the deathwatch beetle ticking in old furniture
 the forest spring tinkling over the stones

IMHO
Sound in film allows the story to have a voice!

Sound & Silence
Preparing for sound design in a film can teach us to analyze day to day noise with our ear and read the sound of life's symphony.
Our ears hear the different voices, and sounds in the general din of day to day and distinguish teach person or thing as a unique character
The director can lead up with both eye and ear to where he/she want us to be both physically and emotionally in a scene or the entire film
The noise of a machine has a different coloring for us if we see the whirling machinery at the same time. The sound of a wave is different if we see its movement.
Just as the shade and value of a color changes according to what other colors are next to it in the film picture, the timbre of a sound changes in accordance with the gesture of the visible source of the sound seen together with the sound itself
 sound film offers acoustic and optical impressions that are equivalently linked together into a single picture.

IMHO
Picture & Sound are EQUAL partners that must work together to create a story.

In a close-up in which the surroundings are not visible, a sound that seeps into the shot sometimes impresses us as mysterious, simply because we cannot see its source.
It produces the tension arising from curiosity and expectation. Sometimes the audience does not know what the sound is they hear, but the character in the film can hear it, turn his face toward the sound, and see its source before the audience does. This handling of picture and sound provides rich opportunities for effects of tension and surprise.

IMHO
Silence is a very important tool, often overlooked.

Silence, too, is an acoustic effect,
but only where sounds can be heard.
silence is one of the most specific dramatic effects of the sound film.
No other art can reproduce silence, not paintings or sculpture or written word.
Even on the stage silence appears only rarely as a dramatic effect and then only for short moments.


Radio plays cannot make us feel the depths of silence at all, because when no sounds come from our radio, the whole performance has ceased, as we cannot see any silent continuation of the action. The sole material of the radio play being sound, the result of the cessation of sound is not silence but just nothing.

Things that we see as being different from each other, appear even more different when they emit sounds.
There are thousands of different sounds and voices, but the substance of silence appears one and the same for all.
That is at first hearing. Sound differentiates visible things, silence brings them closer to each other and makes them less dissimilar.

Examples of Silence

We see our protagonist in a room in an apartment during the middle of the day in NYC, we hear the morning breeze blow the sound of a bird chirp over to us from the neighboring street, from the top of a high building we hear the a construction worker on a ledge, we can hear the wail of a police siren a mile away.
In this scene, you are hearing, silence. The silence in this case being the complete lack of sounds in the apartment or on the street right next to our protagonist.. the din of the city vanishes, and we focus on the silence…the lack of expected sound.

Examples of Silence
• a fierce battle occurring and then, silence, decimated by a huge explosion. That silent moment offers us the introspective and personal moment of our solder just before the impact.
the silence when we hear the distant sound or the slightest rustle near us.
Silence is the buzzing of a fly on the windowpane that fills the whole room with sound
The ticking of a clock that smashes time into fragments with sledgehammer blows.
Silence is greatest when we hear very distant sounds in a very large space, but not the sounds closest to us.
We accept seen space as real only when it contains sounds as well, for these give it the dimension of depth.
In a film, silence can be extremely vivid and varied,
A silent glance can speak volumes; its soundlessness, surrounded by life’s noise, makes it even more expressive
A single silent figure may explain the reason for the silence, make us feel its weight, its menace, its tension.

Pre Production

The Sound Crew

Supervising Sound Editor
Sometimes the Sound Designer
The manager , the sound boss
If the Sup Sound Editor is Not the Sound designer then generally we separate
• Project and personnel management: Sup Sound Editor
• Artist decisions: Sound Designer

Script Review and Story Board Review

Script review with DP, Director
• What shots, locations, number of characters
• sound problems, planned resolution,
• sound requirements for vision of director

Story board review
• Framing Issues
• Type of mics to “get the shot”
• Sound or MOS

IMHO
The Picture Forms the Sound

IMHO
It is crucial for the Director, the DP and the Sound Designer to be in perfect sync.

IMHO
Sound, when done correctly, must match the picture. There is a broad expanse of “match” but the picture must provide the platform for the sound and the sound must be appropriate for the picture, if these 2 items do not match then overall message will be lost.


Shooting format
Type of project
• film, video, camera type, dolly shots, cranes, helicopter
Frame rate & format
• ( 24, 25, 29.97 ) 24p HD SD 16mm 35 mm
Final output
• Film, DVD, broadcast, video, internet
Sound Package Selection
• Single or Dual System
• Type of Recorder
• Number of channels
• Type and Quantity of Mics
• Production Sound Crew size

Budgeting & Scheduling for Sound
Selected system cost
Crew size and cost ( including hidden costs )
• Food
• Transportation
• Insurance for equipment
• Tapes
• backups
Shooting Schedules
• Time allocated per day
• Plan for sound tests and setups
• Plan for sound setup and picture setup
• Wild tracks, Room tone

Selecting your system

To Sync or Not to Sync or be sunk…
If you sync you save time later
If you don’t you save some time now

Single or Dual system Sound Recording

Single System
Dual System

IMHO
Pay close attention to sound, just like you do to the lighting, and framing. Use a dual system when and where ever possible. Utilize a LAV mic ( wireless ) for each main character in a scene and use at least 1 stick with a shotgun mic….

Single system
Direct to camera

Dual systems
Cassette
DAT
Time code DAT
Nagra
Digital disk based recorders
DA88, DA98, ADAT


Note: Record sound on a second media with TC (DAT, Nagra, DEVA, PD-6 etc). To keep TC on audio identical to TC on video you have to drive the video camera with TC and Tri-level sync. None of the Beta style Camcorders have continuous TC. There are unpredictable TC jumps when turning power off. These cameras need TC and Genlock (=Tri level sync) from one source or there is serious risk of “green flashes”. There are 2 external portable generators on the market that do this in 2003. They do away with all remote sync + TC wires. The 2nd audio track on the camera is saved for audio.


cassette recorder
Normally, one of the stereo channels is reserved for recording the 60hz. sync signal from either the camera (if it is set up for cable sync) from a crystal sync generator.
This leaves the other stereo channel for the microphone signal. If the cassette recorder has been so modified, it can also be used to resolve the field audiotapes to either magnetic film or videocassette. Otherwise, a sound lab can be used to accomplish these transfers.

DAT recorder for production audio
DAT recorder can be used as long as the field audio cassettes are resolved correctly
DAT recorder is very similar to a videocassette recorder in the way that it scans the tape during record and playback.
A servo system is required to keep the spinning head and linear tape transport locked together. DAT recorders must have their own internal crystal oscillator in order for this servo system to operate.
Portable DAT recorders like the Sony TC-D7, D8, D10ProII, the new PCM-M1 and the Tascam DA-P1, however, do not allow for any resolving other than their innate ability to play back in real time against their own internal clock.
They cannot be slaved to some other external master source such as a videocassette recorder, for instance (see "...recording sound for...video").
If the DAT recorder will be used only for transferring the field audio cassettes to magnetic film and if the mag film recorder is also referenced to crystal, there is usually no problem.
However, if you are having your film transferred to video on a scanner, the film will be transferred at 23.976fps (not 24fps).
Your DAT source material will now be running faster than your picture track! After just a couple of minutes the sync drift will be quite noticeable to the average viewer.

Time code DAT, TC recorders
• The more sophisticated DAT recorders do accept external sync references.
• These models are commonly used by sound labs for resolving DAT cassettes to magnetic film or videocassette.
• If you will be editing your film project on a non-linear system, you can also convert the data rate of the audio files after you import them into the computer.
• ****** the speed of the audio "wave" files downward by the necessary 1/10 of 1 percent (-.1%).
• programs and utilities are available for the Mac/Avid system and most optical houses can do this for you.

Slate
• What is "slating" Slating is the process of providing positive identification marks for the start of a lip-sync take on both the picture film and the sound track tape. These markings are very important in the editing stage to greatly simplify the process of locating the exact sync position between picture and sound. There are various methods for creating these markings although the original "clapboard" technique is the simplest and is still quite common.
• Electronic slating devices are now often used especially in documentary film production where the clapboard would prove impractical. These include everything from the low-cost blinking LED versions to devices with digital readouts for take numbers to the more complex and more costly time code systems.

Smart Slate
o Derives time code from the audio recording system and feeds it to the TC reader on the slate… This gives a sound editor the ability to match audio with picture via time code data for dailies and editing.
o Slating on the set
 CLEAN
 NEAT
 READABLE
 IN FRAME
 IN FOCUS
 HOLD For 5 seconds
 Call out shot and take
 SLAP the damn thing hard and fast
 If you miss – TAIL SLATE ( upside down )
• A good slate person can even write the tail slate while the shot is in progress on the slate upside down so when the flipped slate is presented for the tail slate the writing is right side up.

IMHO
I highly recommend the use of a smart slate. And a smarter Slate person, who has the neatest handwriting on the set. You need to be completely Anal about documentation throughout the shoot and Slating is part of it!

Time Code
• Time code was originally developed for video tape editing in an effort to make that process more accurate. Over the years it has been adapted for film production
• The main difference between time code and 60hz. sync has to do with time code's ability to stamp each and every frame with it's own special time marking.
• Since every frame has it's own unique code it is possible to not only locate a specific point in the film or audio track but, it is also possible to sync up to any specific point between the two mediums (also known as chasing).
• Time code, while a very useful tool for the advanced film producer, is not a requirement for double-system sync sound film production. You can make a film with perfectly synced audio by using the same old 60hz. system people have been using for years. Just make sure you use a slate to mark your scenes!
60 Hz sync
• The term 60hz. sync is a method of producing lip-synchronized motion pictures.
• In the early days of double-system sound recording, both the motion picture camera and the magnetic film recorder driven by "synchronous" AC motors.
• In North America we use an electrical frequency of 60hz. and a voltage of around 120vac.
• A synchronous motor is designed in such a way that it's rotational speed is in direct relation to the frequency of the AC current powering it rather than the voltage level of this current.
• even though the voltage might vary up or down, a synchronous motor would always run at a constant speed regardless of these fluctuations. Since both the film camera and film recorder were equipped with synchronous motors, they would both always run at the same speed (i.e. 24fps @60hz.).
• Thus, early picture and sound synchronization was achieved by the ability of the synchronous motor to maintain an accurate speed. as long as you remained in the studio and had a source of AC current to plug everything into this worked.

When it became necessary to take the camera out of the studio and away from any source of AC current a new method had to be devised.

Early attempts at so called "portable" double-system sync had the camera and recorder tied together by a cable.

Cable sync was accomplished by deriving a signal from the camera to indicate the exact speed at which it was running from moment to moment. Therefore, when the camera was running at exactly 24fps, a small electrical generator fitted to the camera's drive motor would produce a signal of exactly 60hz. This sync signal was then sent by a cable to the battery operated tape recorder where it was recorded on a separate channel of the tape as a sync track. Cameras now had to only have a motor that ran relatively constant (i.e. constant speed motors) since any speed fluctuations between the camera and the recorder would be reproducible when the sync track on the tape was played back, later.

Pilotone

The term Pilotone was originally a trade name for this process and is now more or less synonymous with the phrase 60hz. Sync

With 60hz. sync every frame looks like every other. If it were not for the slate markings created at the head (or sometimes the tail) of every scene it would be almost impossible to match up picture takes with audio takes (known as synching up the dailies) during the editing process.

Resolvers

Resolving is the process used to obtain lip-synchronization between two different mediums. A resolver is an electronic device that matches the speed of the sound recorder it is connected to (usually called the slave) to some other reference (known as the master). The most common use for the resolver is transferring audio recorded in the field to an editable medium such as magnetic film or video tape.

Without the resolver it would be impossible to provide for an accurate, frame-for-frame relation between two dissimilar devices since their running speeds would be completely independent of each other. The resolver can overcome this problem by comparing the frequency of the 60hz. signal from the sync track recorded on the tape to the master frequency it has been set to.

If the resolver detects a difference between these two frequencies it will create an error signal equal to this difference. The error signal is returned to the slave recorder causing a speed correction to take place until the two sync frequencies match each other exactly.

A brief History

We’ve been mixing in 5.1 and surround for a year or two now. Mostly in someone else’s studio. After a rather long period of research, soul search, and for whatever reason, search me, we decided to enter the 21st century. Our studio, Samurai Music went over to the “other” side. We are now a 5.1 Surround Sound Facility… Of course the first thing we did, err..well, actually the second thing we did, the first being, the emptying of our checking account…., was to order a solid and capable 5.1 system.
Since we already owned a Protools 24|Mixplus system we selected Kind-of-Loud’s Smart Pan Pro, the AXIS surround panner and, of course, my favorite… Genelec Monitors and Subwoofers.
So lets break this down a bit. Genelec 1030A, 5 of them, plus a single 1092A (about the size of a big trash compactor). Note: we also have 1031A’s but I personally prefer the 1030A’s for surround sound mixing.
Protools 5.0, Smart Pan Pro, Tweetie, Woofie, and Realverb 5.1. Axis Hardware/Software Panner.
After pulling everything out of boxes and installing our newly beloved 5.1 system, installing the software, mucking about with a new DVD player and generally adding another 2 miles of cable to the studio…. We sat back and listened… it just sounded so.. well, lets put it this way… I never want to listen to stereo again…ever. Wow. Until you’ve heard a good 5.1 system you haven’t lived, and I’m not talking about the ones in most movie theaters (aka: blown tweeters, ripped subs, channels out, levels wrong etc etc), I’m talking about a well tuned 5.1 system in a good listening room. Breathtaking. Anyway, after a couple of days of listing, watching DVDs and Surround CDs, we looked at the system, shrugged, and opened the hood.
Most of what we found made complete sense. The one thing that rather baffled some of our staff was bass management. After digging through manuals, the Internet, books, etc, I came up with the following simplified review of 5.1 bass management.. Since all this research was sitting here, I figured, why not share it. So for you surround sound dummies, like me, here’s my attempt to de-mystify 5.1 bass management.

5.1 Bass Management – an overview

Bass management revolves around the use of Sub-frequencies from the 5 channels (Left, Right, Center, Left Surround (Ls), and Right Surround (Rs)), as well as the use of the Low Frequency Effects Channel (LFE) Together this set of audio data represents the .1 channel in the 5.1. Low frequencies, nominally around 80 to 100 Hz are filtered from the 5 main channels, routed and summed to the sub-woofer. It’s just like a normal stereo system with the additional of a sub. The difference is in the addition of LFE. The LFE channel offers a unique way to add additional “oomph” to any mix, either in music or with effects such as the standard Hollywood explosion, tornado, sinking ship, train wreck, etc. Simply add a bit of low frequency via the LFE at just the right moment and bingo the room shakes like the A-train in a New York Subway. Managing the added low frequency content is made relatively easy with the addition of the LFE channel. The LFE channel provides an independent bass path that can be utilized at will without affecting, or being affected by, the normal bass coming from the 5 main channels. With the ability to utilize both the traditional subwoofer signal and the addition of an LFE audio path, 5.1 bass management plays an important, although sometimes overlooked, role in 5.1 mixing.
We selected the Kind-of-Loud system in order to utilize its bass management abilities, as well as, its natural ability to easily manipulate a sound field in an array of ways. Additionally, our new Genelec sub-woofer provides its own rudimentary bass management. With the internal Genelec capabilities, and the addition of our Smart Pan Pro, Woofie, and Tweetie plug-ins we can manipulate the bass in more ways than we though possible. In fact we’ve found a number of ways to completely decimate a two thousand dollar sub and our neighbor’s goodwill in less that the time it takes to generate a 20Hz 120db sine wave…

Genelec Monitoring System’s Bass Management

Lets start with the Genelec speaker system. Our system includes 5 matched Genelec full range monitors and 1 sub-woofer. The sub woofer provides 4 inputs and 3 outputs. I/O (Input /Output) 1 is the Left Front channel (L), I/O 2 is the Center Channel (C), I/O 3 in the Right Front Channel (R), and Input 4 is the sub input (S). The reason for this is that Genelec has included a low frequency filter that has a center frequency at 85Hz as part of the subwoofer system. All signals from the L, C, and R can be routed through the subwoofer. When the L, C, and R speakers are routed through the sub all signals 85 Hz and below are routed to the subwoofer. Additional low frequency signals, like LFE, can be independently routed to the sub via the SUB input. If you wish to bypass the internal filter, Genelec kindly provides a simple 1/4” jack and a wiring diagram for a bypass/ mute switch. I can hear my system without a sub, or with the flick of a switch, provide a justifiable homicide plea for my downstairs neighbor.

Bass Management Over Simplified

On top of the Genelec’s bass management, Kind-of-Loud’s software provides even more bass management capabilities. Ok, we need to deal with low frequencies from general audio tracks and specific special effects. Let’s take a look at the basic block diagram. A single audio signal, lets say a typical Hollywood explosion, is placed on an audio track. We route the track to the 5 main channels via a 5.1 panning system, or simply routing on any 8-buss console. In this case we’ll use the Smart Pan Pro and its associated Software.
The input signal is routed to the panning software plug-in where we can send the basic explosion audio to any, or all of our 5 main channels (L, R, C, Ls, Rs). Additionally, via the LFE output we can bus the same signal to our LFE channel.
So far we have full range audio heading for up to 5 independent channels:
 The L and R channels are routed to the Smart Pan Pro (SPP) LR outputs
 The C Channel to the SPP CS outputs
 The Ls and Rs channels to the SPP LsRs outputs
You might be thinking I forgot the Sub channel. I didn’t, it gets sorted out later down the audio pipeline.
Finally, via the LFE gain control, we can send the same signal via an independent route to the subwoofer.
Let’s follow each signal path separately.
L / R signal path
The full range audio is routed to the bass management, where it is filtered and the low frequency sent to the Subwoofer, while the remaining audio is sent on to the L and R front speakers. The low frequency audio is then sent to the bass management module where independent gain adjustment can be made to the overall bass levels. Following the input gain, the low frequency audio is sent to the Bass Extension on to the Bass Redirection.
Center / Sub signal path
The full range Center channel audio is routed to the bass management, where it is filtered and the low frequency sent to the Subwoofer, while the remaining audio is sent on to the Center speaker. Again, like the L and R signal path, the low frequency audio is then sent to the bass management module where independent gain adjustment can be made to the overall bass levels. Following the input gain, the low frequency audio is again, sent to the Bass Extension on to Bass Redirection.


Ls / Rs signal path
The full range audio signals of the Left surround channel and the Right Surround Channel are routed to the bass management, where the low frequency filtered signal sent to the Subwoofer, while the remaining audio is sent on to the Ls and Rs (rear) channels. Again, like the L and R signal path, the low frequency audio is then sent to the bass management module where independent gain adjustment can be made to the overall bass levels. Following the input gain, the low frequency audio is again, sent to the Bass Extension on to Bass Redirection.
Bass Extension
Bass Extension allows the low frequencies to be sent, not only to the sub, but back to all 5 main channels (L,R,C,Ls,Rs) channels. This allows for the bass to emanate from all 5 speakers (L,R,C, Ls, and Rs). The Bass extension can be disabled. In this case the low frequencies are routed to the Bass Redirection module only.
Bass Redirection
Bass Redirection provides Mutes for all 5-channel low frequency signals. Un-muted low frequency signals are then summed and sent to the subwoofer via the Sub input.
LFE Channel
Finally, the Low Frequency Effects channel provides a completely independent path to the subwoofer from our initial explosion. The LFE send (in this case the LFE gain on the SPP Plug-in) routes the LFE signal directly the bass management where its’ gain can be separately adjusted, muted and routed to the subwoofer. In this manner you can decide exactly when, what and how much low frequency is utilized for exactly what purpose.
Between the monitor system’s bass management and the software bass management we have the ability to add as much or as little bass as necessary to meet the needs of the moment. But, a word of caution… bass frequencies can reach the sub from four specific ways in this system.
1. LFE signal
2. Bass Redirection
3. Bass Extension
4. Genelec’s internal filter
I should take a minute to mention a couple of issues regarding phase relationships with the Sub/LFE channel. Make sure you align your subwoofer correctly, since the sub and your 5 other speakers are now rather distant from one another. Remember, low frequencies are omni-directional so you have a reasonable amount of “wiggle room” for sub placement. If you have a subwoofer system similar to Genelec make sure that you either bypass the filter in the Genelec or mute the bass re-direction from the software. If you don’t you will run the risk of sending low frequencies to the sub from different sources, potentially out of phase.

In closing, It is entirely possible to significantly test the structural integrity of not only the subwoofer, but of your studio and your ears. Be careful, it’s amazing what a bunch-o-in-phase-bass signals can accomplish. It’s also amazing what a well-placed Low Frequency Effect can accomplish as well.
So, Get ready to Rumble!

cheers
geo

Big wow for a Post forum and thanks Georgia for condensing all the info about Dolby E .Maybe i missed it on the bass management post but wich crossover freq. do you choose for 5.1television for the Lfe 80 100 120 or to taste ...?

Eric

holy freaking awesome. Thank you!!!

I normally do 80hz as a default.... but if a given delivery spec calls for a specific cross over frew then thats the one I use ( DOLBY, DTS, and various specs I have in house call for different crossover points )

cheers
geo

Here's a great document about post audio work flow and sync.


cheers
geo

Remember this is an overview and not ALL the technical issues are addressed here. This is from a lecture I give to indie film makers....

cheers
geo


Transfer in Post

Jumping ahead for a moment

resolving process: you must be careful to select the correct master sync frequency! If you shoot film at 24fps and have it transferred to video, the film will actually be transferred at 23.976 fps. Therefore, to maintain the correct sync relationship, the field audio must be transferred against a master clock frequency of 59.94hz.

This is easily accomplished with a resolver by simply connecting a video reference to the external sync input. The resolver will substitute the external video reference at 59.94hz for the internal reference of 60hz.

The field audio will now be resolved to video at the same rate as the film to video transfer.

Video Transfer in Post

It will not matter to the sync sound recorder whether you shoot at 24 or 30fps for video transfer.
The crystal oscillator will remain at 60hz either way.

If you are shooting at 30fps and will have your film transferred to video on a scanner it will be transferred at 29.97fps.
Therefore, to maintain the correct sync relationship, the field audio must be transferred against a master clock frequency of 59.94hz.

Again This is easily accomplished with a resolver by simply connecting a video reference to the external sync input.

The resolver will substitute the external video reference at 59.94hz for the internal reference of 60hz. The field audio will now be resolved to video at the same rate as the film is being transferred to video.
Recording sync Sound with Time code for film

• Camera rolls at 24 Fr/sec
• Sound Time code is 30 Fr/sec Non Drop
• DAT sample rate is 48K
• Exception: some long form TV shows use 30Fr/sec drop frame
• Have all involved communicate: Editor –Post production supervisor (or whoever is technically competent) should send written specs to camera crew, sound department, transfer (telecine) house, picture editors, post sound editors, final mix stage.
• Slate must show camera (Fps) and sound TC speeds. Label and all reports items Producers: Have everybody communicate
• It’s in the details
• Stick to the plan

Labels for Tape

• sound-boxes should include:
 Camera frame rate
 Sound TC speed Drop or ND
 Sample rate if DAT
 Reference level
 (-20 digital = 0VU )(1khz tone @ -20 = 1.43 v)
 Production Title, Production company name with phone number
 Date, Roll #, "DO NOT SUM" or "SPLIT TRACKS" or "SUM TRACKS" or "MONO" Optional: TC start and TC end (helps post staff when lost in a mass of material) -- (Maxell 124 labels are big enough for all that on a DAT!)
 Playback labels: music reel box should be labeled too; the on set playback operator and telecine want to know:
 Source of Time code [48Tr, DA-88, 2Tr
 Studio Master, DAW…]
 Original TC speed [29.97 typically]
 Original sample rate [47.952K]
 Play at 48K Suggested speed and/or sample rate changes on PB while playing back for filming (or taping).
 Producers Stay Awake



Producers and Time Code ( why they need each other )

It’s only the producer’s money that will be wasted if no one pays attention to time code from production through post. Even though it’s feasible to keep high quality audio from production through editing to release intact, it is not likely. Someone will inevitably screw up, and that will cost money. Watch this process like a hawk and stay on top of it.

IMHO
Picture editors usually can’t be bothered with location TC nor post sound. You, the producer, have to make sure that location time code from the flexfile from the telecine is loaded into the Editing Station. This has to be compiled with the picture cut list EDL and given to post sound editors. If you don’t pay attention through these steps, you will pay for it in time and money later.

IMHO
Make sure that your audio is recorded well in production, give audio focus during a shoot. It will save you money later.


• Location TC has to get to the Avid intact.
• While assistant editors digitize the picture, they have to note the in/out location TC numbers for all takes.
• a sound editor has to use parts of non-selected takes for “fixes.” If they are smart assistants, they take care to note sound rolls as well as camera rolls.
• Really smart assistants scan the paper sound reports into graphic files on the computer and they won’t get lost as easily and accompany the project throughout.


Producers have to stay on top of the 30 and 29.97 issue. Also producers have to stay on top of the Drop and Non-Drop frame issue. If mistakes are made there is always a way to resolve it, but this will just cost more money.


IMHO
The later the corrections are made in the production process, the more it costs. Time code makes good sound cheap, fast and easy. All decisions have to be made ahead of time.

IMHO
ASSUME NOTHING and pay attention to your surroundings!


How to keep sound and picture in sync

• 24 frame/sec film shoot
o Shoot film at 24 Fr/Sec
o Roll sound with 30 Fr/Sec (non-drop, or drop) Time code and 48K sample rate
o Pre-roll only sound for 10 sec or more.
o Have the time code slate show numbers for 2 sec. to camera
o Slate appropriately
TOD (Time of Day) vs. REC-RUN Time Code

Time of Day Time Code

If the sound recordist records time of day TC (TOD), he/she should refresh the sync on the slate by jamming it at least every 4 hours.


Record Run Time Code

If the sound recordist uses RECORD-RUN TC there is never a problem with too short a pre-roll as Time Code on DAT is continuous.
TC has to be transmitted to the slate.
Someone has to watch that the numbers on the slate are rolling and are correct

• advantage is that you can use inaccurate TC generators in your Recorder.
• Some recorders only have a relatively inaccurate TC generator. Record run Time code eliminates the need to pre roll (the 10 sec. minimum for sound only).
• Error possibility: Since the time code visible on the slate does not move until it is refreshed by the generator from the DAT telecine operators need to know not to take the first visible frame of time code on the slate to punch into your telecine controller.
• They must wait until they have moving code and pick any of those frames.
• Editors like the continuous TC on the DAT as it lets them find takes easier in post sound
• When I record with DAT, I normally use REC RUN and keep the ID Write in manual mode. I can then write an ID just prior to pre-slating. This way, it looks to post as if I never stopped when I roll for the actual take, which is only a new start for the recorder but a continuous TC.
• In real time it might be thirty minutes after pre-slating though. The time code is continuous and the ID# does not advance.
• This allows me to slate, log the take number and ID# long before I call "speed". Al I need to do for me to call speed is to press the record button and hear the confidence monitor playback.
• I transmit the TC to slate and don't have to worry about excessive pre rolls that waste tape and there is plenty of pre-rolled TC for telecine as all TC is continuous.

Advantages of RECORD RUN Time code

• No drift, no wrong code due to wrong switch settings. When transmitting to a slate that has no TC generator (or at least a disabled one), the numbers on the slate can only be the ones being recorded.
• Instant speed, no pre-roll required. When using REC RUN, the pre-roll is built into the previous take.
• Time code is uninterrupted from the beginning of the tape to the end.
• No worrying about resetting or re-jamming time code after a battery change or power loss in recorder or video camera
• No need to re-jam slate every 4 hours to compensate for drift.
• When using 1-hour tapes, 24 consecutive tapes can have unique, non repeated time code. Makes it easier for post to find takes. Makes final post bookkeeping nice and neat, saves them from having to read labels on tapes.
• When the slate numbers are rolling, the cameraman can assume you have speed
• If the slate numbers are not rolling, the camera-assistant (Slate person) or maybe even the operator can assume you don't have speed.
• Assuming they pay attention is giving away a lot of your responsibility. You might be screwing up totally and no one notices. You better have a good monitor!
Film Dailies

This is a straightforward mechanical resolving situation;

picture and sound are lined up manually in a synchronizer and kept mechanically parallel as usual for the last 60 or so years. No pre-roll necessary.

No time code is needed here, unless DAW editing systems are used.

IMHO
If you are using a DAT be very careful about setting Ids on the DAT correctly and Logging them

As much as I love transfer technicians, they tend to be underpaid and over worked….need I say more?
They often work with machines that mute in fast forward or have a scan with unusable audio.

29.97nd / 29.97drop NTSC videotape Dailies

Transfer picture to NTSC tape running at a standard 29.97 Fr/Sec: Picture is transferred using 3/2 pull down to expose 30 frames (60 fields) of video in the same second that 24 film frames were exposed originally. The 3 and 2 refer to the process where one film frame is transferred to 2 video fields and the next film frame is transferred to 3 video fields. This process adds the 12 additional fields (6 frames) needed to make 30 out of 24. This is how ARRI shows it schematically on their website www.arri.com/infodown/cam/ti/p-1008.pdf The 4 film frames are called ABCD the corresponding video frames A1,A2,B1,B2,B3, etc.

So far so good. Now you have 30 frame video from 24 frame film.

Now this is slowed by 0.1% to compensate for color video's real speed of 29.97 Fr/Sec.

Sound follows this slow-down (“pull-down”) of 0.1% to 29.97.

In telecine transfer, the colorist parks the picture on an easy to read time code number. The number is then punched in the telecine computer and all is automatic from then on. It is here where the sound playback machine (¼ inch or DAT) needs the 10 or so seconds to come up to perfect video speed. Videotape dailies get a new TC starting with 1:00:00 at tape roll 1. An EDL is kept to track original location TC, film negative footage (keycode), and the new telecine TC. An EDL (edit decision list) is a database file on a computer disk that accompanies the video tape from then on and is imported in the editing computer.

How To Stay In Sync With HD Cameras

23-frame time code

Just as we were getting used to having to contend with pull-ups and downs when dealing with a 24-frame film shoot, we get another headache. HD actual time code rate is 23.976 fps. It is also referred to as 23.98 fps. Since there is no 23.976 frame rate selection, the problem actually lends itself to a simplified solution .

Because most video workstations and edit bays perform in the NTSC 29.97fps format, the HD picture must be down-converted to NTSC. There is no problem converting 23.976 fps to 29.97 fps. These two frame rates are closely related to each other, and the conversion works perfectly. Since 23.976 is a workable solution, it also means there is no .1% slow down (that we have all come to know and hate) from shooting with 24-frame film. No speed change for picture. So what about the time code for sound?

If you are running a DAT recorder and want to have the same time code on audio as you have running in the camera in a “Free Run” (time of day) situation, you can cross jam 23.976 to 29.97 fps from camera to audio.

In this scenario, the camera acts as the master time code.

To do this, you will have to use a Synch Box, or a time code generator/reader.

Whichever one you choose to use, set it to 29.97 fps and jam sync it from the 23.976 fps of the camera. The camera will have a BNC connector for time code output. feed the 29.97 fps into your DAT recorder from the Synch box.

Have your DAT recorder set to 29.97 fps.

If you are still running a time code slate, make sure that it is set to 29.97 fps as well when you jam it from the DAT recorder.

Now that sound is running 29.97 fps and camera at 23.976, what happens at the end of the day? As stated earlier, the picture must be down converted if it is to be edited in video (NTSC) on a non-linear system like the AVID. During the down conversion process, the 23.976 fps is converted to 29.97 fps and a window burn-in of 23.976 fps is created for reference. Now the 29.97 picture time code is frame accurate with the production audio time code and is easily synched for editing.

When using this method in the field, there are a few things to watch out for.
• When some HD cameras changes batteries or is powered down in any way, you must jam sync again!
• Changing the batteries creates a time code skip and suddenly you will have a possible 6 frame offset between sound and picture.
• There must be excellent communication between camera and sound on this. Most HD cameras will not lose frames when put on standby.

Without the TriSync, there is the possibility of a 1 frame offset, no more than that.

TRI-Level Sync


sync reference is used in a camera to accurately define each frame. (frame sync)

NTSC analog video, has the sync signal consisting of a Blanking Pulse (sync pulse + color burst + back porch) followed by the video image data. This sync information is repeated every scan line.

Tri-Level Sync systems
Generally used in Multi camera setting only.

• Ambient Tri-level "Lockit" boxes (ACL 202CT) delivers time code with a built-in Tri-level sync generator.
• Denecke was released, their Tri-level Syncbox model SB-T.

Provides seamless Camera ( picture switching for HD & SD systems ) High Definition camcorders cannot Genlock to a standard NTSC or PAL video sync signal. HD uses Tri-level sync. The Tri-level sync signal consists of a three-level sync pulse (zero volt (0V) Blank, -0.3 V pulse, +0.3 V pulse) followed by the video image data. The signal is repeated for every scan line while it creates the entire HD frame.


But there are other reasons to employ genlocking, apart from its more traditional use in live switching multi-camera productions.

If you just jam, or cross-jam, time code in respect to itself only, you can get up to a one-frame difference between the camcorders in a multi-camera set up. Or in other words, this means that the time code numbers might be out of step with each other, by up to one frame, in any multi-camera set up.

Using time code that is linked to camera sync in a fixed relationship, eliminates this random time code offset problem. A simple jamming operation doesn't establish a precise relationship between the time code and camcorder sync. It is random within a single frame.

When jamming time code with a Tri-level "Lockit" box, you are jamming the time code in respect to a reference source of Tri-level sync. In other words, the advancing time code numbers are timed to run in a precise relationship with the sync stream.

This means that as the camcorder begins to built a new frame, the time code advances to the next frame number, at that precise moment in time. As every new frame starts, the next time code number appears at precisely the same moment.

When you jam these new Tri-level "Lockit" boxes in respect to each other, you are not only getting the time code numbers to roll in perfect step with each other, but you are also getting the Tri-level sync stream in perfect step with each other.

IMHO
There are many issues to pay attention to regarding sync sound for an HD production. Communication is the most important part, but with everybody using the same game plan all should be good. Working with 23.976 fps there is no pull-down with 29.97 fps, - no .1% pull-down problems.


cheers
geo

Wow! My brian hurts. heh

Thx for all the info Geo.thumbsup

links for metering and scales

http://www.ebu.ch/en/technical/trev/trev_293-emmett.pdf
http://www.ebu.ch/CMSimages/en/tec_d...tcm6-10506.pdf
Digital Domain - Level Practices (Part 1)
THE INS AND OUTS OF
dkaudio - Google Search
Shure - Products
Peak Programme Meter

the pictures are various PPM vs VU scales...

cheers
geo

It's been said several times already, but thanks for all this info. This is great stuff!

Television standards are most definitely not.

If you are going to mix for delivery to Broadcast you must absolutly beyond a shadow of a doubt get a written copy of the delivery specs for the SPECIFIC broadcaster you are delivering to.

Every broadcaster has DIFFERENT SPECS! I have a 3" binder FULL of specs from all sorts of broadcast facilities, cable companies, individual corporations that provide content to broadcasters, etc.... I would share this, BUT, they are company confidential. In most cases You can ask for them when you are delivering program material, but its notoriously hard to get the written tech specs out of the hands of the broadcasters.. Why? it's beyond me! I have no clue as to why they are go greedy about their specific specs, but it the way things are. But, anyway, get a copy of the spec for your given delivery.

The specs will generally call for
Peak limits, LM100 Dialnorm settings, Track positions, general levels, what goes on what track... etc etc.


Also don't forget who your listening audience is. It ranges from MONO TV sets with the speaker in the BACK of the set facing the wall all the way up to people like me who listen to TV on huge genelec 5.1 systems!

So besides meeting all the delivery physical specs, and boy there are some crazy specs out there... you need to still mix a "good" program.



ok.. here are some delivery spec examples:

"-20 db full scale where 0VU corresponds to -20 dbfs AES signals"
"The peak level cannot exceed +10VU Reference SMPTE RP155"
"Line mode compression - Film Standard"
"Mix level 105db"
"Lt/Rt 0.707 (-3db)"
"Tape Format HDCAM or SRW"
"Channel 1 - Left Channel 2 - Right Channel 7&8 - Dolby E"
"Dialognorm -27"
"Nominal Operating Level -20 db (LAeq)

or this....

"Stereo audio must be fully mono compatable"
"The audio mix should also be well balanced and equalized, with dialogue and narration clearly able to be heard"
"Program audio must reflect tone level"
"audio levels are evaluated using three different measurements, audio signal peak, average loudness, and dialogue loudness"
"Peak audio levels are evaluated using a digital true-peak meter with zero rise response"
"Assuming a reference level of -20 dBFS, peak audiomay not rise above -10 dBFS at any point in the program"
" (place broadcaster name here) also requires that program dialogue levels be analyzed using a Dolby LM100 broadcast loudness meter."
"assuming a reference level of -20 dBFS (+4dBu) VU levels should consistantly fall between -26dBFS and -28dBFS as an average for the entire program."


or this


"The audio dynamic range of commercial materials must be suitable for television broadcast."
"Related to test level ( nominal -20dbFS), instantaneous peaks must not exceed -2dbFS"
"Average peaks should not exceed -8dbFS."
"Average Dialogue should not exceed -27dbFS."
"Dialogue norm = -25dB"

So you can see that there are some basic "Sort-of-standards" but when you get into the nity gritty of each delivery spec..... ,BTW, these just scratch the surface of the actual spec documents... you will see extensive in consistancies and varying requirements between actual hard delivery requirements.

We did a series of NAPA parts commericials for Superbowl last year ( not tihs past one ) and we had to create 7 different physical tapes with 7 different mixes to meet the delivery requirements for all the deliveries...

cheers
geo

Here a signal flow diagram showing DOLBY DIGITAL signal path for a typical broadcaster.



cheers
geo

Georgia,

Thank you for your time and generosity, this information is great!
Are you accepting marriage proposals online?

All the best,

Makoto

cheers
geo

Some audio / video guidelines for shooting HD and dealing with audio in the new HD world....

enjoy....

Audio/Video Guidelines for Separate Sound Recording

Record video at 23.976 fps (not 24 fps)

This is to allow a field down converter to be used. Video playing at 24 fps cannot be downconverted. If video is recorded at 24 fps, the Post facility will "pull it down" to 23.976 fps in the downconversion process. Use Internal Timecode in Record Run mode, and the "Backspace Edit" function of the camcorder if possible.This will provide continuous, unbroken timecode on the videotape. If broken timecode occurs and insufficient preroll is available, then the post production facility will either have problemsin the audio synchronizing and editorial processes, or have to make clones of the masters that have unbroken timecode. Please consult with your camera technician for details of the backspace edit mode.

IF "Time of Day" timecode is used, then always allow sufficient preroll (minimum 5 Seconds at each start) Provide at least thirty seconds of bars at the beginning of each tape. Do not clip or crush the video signal- use the "zebra" function in the camera view finder.

External waveform monitor and picture monitor are recommended. These will allow you to view exposure levels directly, as wellas giving the camera operator a better indication of focus.Watch out for bright lights or other sources that go into white clip.Be aware of the detail in the dark portions of the image, make sure that those areas are not crushed.Try to leave sufficient room in the signal for tape to tape color correction.Consult your camera technician for details and setup of the "zebra" function.

Use either a Gray Card Plus or an 11 step gray scale chart at every lighting change
Shoot the card in the main subject area, zoom to at least 1/3 full screen image of card This is not strictly necessary, however the cinematographer may find it valuable when looking at the exposure in the waveform monitor. These may also prove valuable to the editor and colorist if there are questions regarding the exposure level during the editorial or tape to tape color correction process.

Please note any special filters or exposures in the camera report
These will be noted in the downconversion/audio synchronization log file

Allow sufficient preroll when possible (minimum 5 seconds)
In the event of broken timecode, this will allow sufficient preroll for audio synchronization.

Allow sufficient postroll (2-5 seconds)
This is to allow sufficient postroll to exit the audio synchronization event cleanly.

Make sure the smart slate is in view and legible at every clap. If tail slates are needed, make sure that they are upside down and legible.

Note MOS events in the camera report
These will be noted in the downconversion/audio synchronization log file

All video is downconverted digitally at unity settings. No color corrections are applied during the conversion process.

Time Code Guidelines for Separate Sound Recording
Always use a smart slate.
Resync the smart slate frequently (at least once every few hours). If the smart slate drifts with respect to the audio timecode, the post production facility encounters time consuming problems during the audio synchronization process. Instances of timecode drift will be noted in the downconversion/audio synchronization log file

Set Smart Slate Time Code frequency at 29.97 fps, Non Drop Frame Timecode
Slate and DAT run together at 29.97 fps; the VTR runs at 23.976 fps Some productions use 29.97 timecode converted from the 23.976 code, or vice versa. Your camera rental facility should have details on equipment to provide this capability.

Record digital audio at 48Khz with 29.97 fps NDF timecode.
Always allow 5 seconds of preroll for DAT audio.
If insufficient preroll occurs, the audio cannot be synchronized during the downconversion/audio sync session, and will have to be transferred as "wild track". This causes time consuming delays both in the audio synchronization and off-line edit process. Insufficient preroll will be noted in the downconversion/audio synchronization log file

Provide at least thirty seconds of tone at -18db at the start of each tape.
This level may vary among machine types. The tone should be recorded at the standard reference level of the DAT.

For digital audio, never allow the level to reach 0db (Maximum, full scale level)
If the audio reaches or exceeds this level, there will be distortion.
Instances of this will be noted in the downconversion/audio synchronization log file

Video and Audio Guidelines for Sound on Tape HD Recordings

Record video at 23.976 fps (not 24 fps)
This is to allow a field down converter to be used. 24 fps video cannot be downconverted.
If video is recorded at 24 fps, the Post facility will "pull it down" to 23.976 fps in the downconversion process.

Use Internal Timecode in Record Run mode, and the "Backspace Edit" function of the camcorder if possible.
This will provide continuous, unbroken timecode on the videotape. If broken timecode occurs and insufficient preroll is available, then the post production facility will either have problems in the audio synchronizing and editorial processes, or have to make clones of the masters that have unbroken timecode.
Please consult with your camera technician for details of this mode.

IF "Time of Day" timecode is used, then always allow sufficient preroll (minimum 5 Seconds at each start) Provide at least thirty seconds of bars at the beginning of each tape.

Do not clip or crush the video signal- use the "zebra" function in the camera view finder.
External waveform monitor and picture monitor are recommended. These will allow you to view exposure levels directly, as well as giving the camera operator a better indication of focus. Watch out for bright lights or other sources that go into white clip. Be aware of the detail in the dark portions of the image, make sure that those areas are not crushed. Try to leave sufficient room in the signal for tape to tape color correction.
Consult your camera technician for details and setup of the "zebra" function.

Use either a Gray Card Plus or an 11 step gray scale chart at every lighting change
Shoot the card in the main subject area, zoom to at least 1/3 full screen image of card
This is not strictly necessary, however the cinematographer may find it valuable when looking at the exposure in the waveform monitor. These may also prove valuable to the editor and colorist if there are questions regarding the exposure level during the editorial or tape to tape color correction process.

Please note any special filters or exposures in the camera report
These will be noted in the downconversion/audio synchronization log file

Allow sufficient preroll when possible (5 seconds minimum)
In the event of broken timecode, this will allow sufficient preroll for off line synchronization.

Allow sufficient postroll (2-5 seconds)
This is to allow sufficient postroll to exit the off line synchronization event cleanly.

All video is downconverted digitally at unity settings. No color corrections are applied during the conversion process.
Smart slate not required when recording the sound on the HD videotape.
Provide at least thirty seconds of tone at -18db (standard reference level) at the start of each tape.
Record tone during the video color bars.

For digital audio, never allow the level to reach Maximum, full scale level
If the audio reaches or exceeds this level, there will be distortion. Instances of this will be noted in the downconversion/audio synchronization log file If the audio reaches or exceeds this level, there will be distortion. Consult your camera technician for details and setup of the audio recording.

Off Line Preparation, Separate Audio System

Down Conversion is made from each HD 24p tape to an SD 30 fps tape
Any format Standard Definition tape is available, Digi Beta or Beta SP is recommended
Both or either 24 frame and 30 frame windows may be inserted. Client specifies the location. Depending of the type of conversion, there may some limitations with window placement. In general, it is safe to put the windows below center screen, and together, if both are required. The aspect ratio can be either letterbox, center cut or 16x9 anamorphic. Client specifies conversion aspect ratio. If timecode is broken and insufficient preroll is available, a new HD clone with unbroken timecode may have to be made at this time.
In general, TPG recommends that all HD material be downconverted to an intermediate format, then that intermediate tape and the field audio tape be synchronized, and a "Circle Take Offline Master" (CTOM) be created from them.

Audio is synchronized and transferred to each standard def tape from the DAT.
The DAT is pulled down to 29.97 fps at this time (only if required).
Usually, only circle takes are synchronized and transferred. Client specifies which audio to transfer.
All circle takes are documented in a Flex file or an ALE file for off-line batch digitizing.
An audio timecode window may be added to the SD tape at this time. Client specifies window location. There are some limitations to the window location. In general, it is safe to put the audio window anywhere below center screen.

Client is delivered Standard Def tapes with sync audio and a Flex or ALE file.
A one for one relationship between the HD tapes and the SD tapes is maintained.

It is recommended that the client use and Avid Film composer for the off-line system
Avid Project must be set for "Film Mode". All cuts must be digitized as for "A Frame" film transfers. If not, Project and subsequent EDL integrity will be compromised If an Avid is not used, TPG cannot explicitly guarantee the accuracy of any 30 to 24 frame EDL conversion

Off Line Preparation, Sound on HD Tape

Down Conversion is made from each HD 24p tape to an SD 30 fps tape
Any format Standard Definition tape is available, Digi Beta or Beta SP is recommended
Both or either 24 frame and 30 frame windows may be inserted. The 30 fps video timecode is always derived directly from the 24 fps timecode, and is "A" framed.
The HD video is pulled down to 23.976 fps at this time (if required) All HD material is downconverted.

DAT Backup Tapes
DAT tapes, made from the 24p HD original, with 30 fps timecode derived from the 24 fps timecode may be made at this time for separate Audio Post Production.

No Audio Synchronization is required
Since audio synchronization does not occur, no Flex or ALE file is generated for off line use. Media may be subclipped by the Avid assist as required.

Client is delivered Standard Def tapes with audio. No Flex or ALE files are supplied.
A one for one relationship between the HD tapes and the SD tapes is maintained.
Since audio synchronization does not occur, no Flex or ALE file is generated for off line use. Media may be subclipped by the
Avid assist as required.

It is recommended that the client use and Avid Film composer for the off-line system
Project must be set for "Film Mode". All cuts must be digitized as for "A Frame" film transfers. If not, Project and subsequent EDL integrity will be compromised If an Avid is not used, TPG cannot explicitly guarantee the accuracy of any 30 to 24 frame EDL conversion

On Line Preparation

Client will deliver the Off-Line Master video tape.
Any Standard Definition format may be delivered, Digi Beta or Beta SP is recommended
Off-Line Master must have the off-line audio on Channels 1 and 2. This audio will serve as a guide track to ensure edit accuracy.

Client will deliver the final Avid cut sequence file on format of choice (CD, DVD, Zip, etc.) or emailed.
TPG will generate a 24 fps EDL from the off line project
If an Avid is not used, TPG cannot explicitly guarantee the accuracy of any 30 to 24 frame EDL conversion

Offline master will be upconverted to HD 24p.

This is to allow on-line video conform of EDL, with an audio guide track for reference.

On Line Assembly, Color Correction and Titling

On Line assembly from converted EDL to either Sony HDCam or Panasonic AJD 3700
A "textless" master is created for subsequent color correction Audio conform is optional, depending on the needs of the client and the type of project.
"As Assembled" EDL is created for Tape to Tape color correction session
"Video Only" EDL is generated in On-Line bay for color correction

Color Correction from textless master to either Sony HDCam or Panasonic AJD 3700
A "Color Corrected Textless" master tape is created from the Textless Master

HD Titling session to create the Final Master from the Color Corrected Textless Master
The Titling session must always follow the Color Correction session, since titles do not need correction.
Textless material should be added to the final master at this time.

Try to avoid moving titles in 24p
Moving Titles can have very undesirable artifacts. If the client requires moving titles, the titles must be evaluated on a case by case
basis. The general rule is, the slower the title movement, the better it will look. This is also true of rolls and crawls.

QC and Restoration

100% QC of Final Master
All problems are noted, including any technical, video or audio problems
QC list is generated for Client, and can either be delivered via hardcopy or via email.

Any restoration should occur on the Color Corrected, Textless Master
Since this is video originated material, dirt will not be a problem, so restoration would consist of boom or rig removal, logo or product removal, and other, similar type processes.
Restoration should always occur on the Final, Corrected Master to avoid confusion and duplicate work.

Cheers
geo

wow, this is amazing.., one thing leads to another and here I am rolling in answers to questions I didn't know existed.

thanks for your help at DUC, for pointing me here, gearslutz and for all this amazing info. I'll have to get some ink for the printer, there's too much to read from the sceen. The paper will not leave my font door though.., all credit due of course

cheers

no worries mate!


cheers
geo

you recommended on that other thread not to mix with bass management. although i didn't mention that i'm working on nearfields- which have freq response to 40hz (dynaudio air 15s). the dynaudio's have a bass management setting and i can make the crossover 80z. would you still recommend that? not being full range monitors isn't it wise to mix with bass management and utilitze the subwoofer?

keep in mind, this is a direct to dvd project- and i believe that's what dolby and dts encode for, right?

btw, your thread here is amazing. THANK YOU.