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Old 15th July 2008
happy to help.

Old 4th August 2008
X-Curve History by Tomlinson Holman

A History of the X Curve

The X curve celebrates nearly a quarter century of helping interchange in the industry.

In the history of multichannel sound, the standardization of the electroacoustic frequency response for monitoring film stands as one of the most significant developments. It was standardizing the monitor frequency response at the ear of listeners that provided for better interchangeability of program material, from studio-to-studio, studio-to-theater, and film-to-film. Work started on formal standardization of the monitor frequency response for large rooms for film in 1975 on both the national and international levels. The work resulted in the standards ANSI-SMPTE 202 in the U.S., the first edition of which was officially published in 1984, and ISO 2969 on the international level. Actually, the standardized response was in use for some years before the formal standards were adopted.

The X Curve: The measured electro-acoustic frequency response presented to the ears of listeners in a dubbing stage or motion picture theater. The curve is to be measured under specific conditions, and is to be adjusted for room volume as specified in the standards referenced in the text.

The background behind this work began with Texas acousticians C. P. and C. R. Boner, who established in the 1960s that a "house curve" was a needed concept. They showed that a flat electroacoustic frequency response in a large room sounds too bright on well-balanced program material. This was subsequently found to be correct by other researchers, such as Robert Schulein and Henrik Staffeldt, as well. While Boner's practice was for speech reinforcement systems that did not require theater-to-theater uniformity in the same way that film does, nonetheless the concept of a house curve traces back to them. This development paralleled the introduction of 1/3 octave room equalization, since there would be little point in establishing a house curve if sound systems could not be adjusted to it.

Ioan Allen of Dolby Laboratories realized that the idea of a house curve was a valuable one after applying Dolby A-type noise reduction to optical soundtracks and extending the bandwidth of the track. While we think of Dolby A as principally noise reduction of between 10 and 15 dB depending on frequency when used in a tape context, in the case of the application to optical soundtracks, most of the advantage in dynamic range was taken to extend the bandwidth. The ordinary mono Academy-type soundtrack had sufficiently low noise for its time only by imposition of a strong high-frequency roll-off that made the effective bandwidth of soundtrack reproduced in theaters about 4 kHz. If such a track was reproduced with a wide-range monitor, the noise was excessive. By extending the high-frequency bandwidth of the monitor, and applying Dolby A NR to tame the noise, a very useful extension of bandwidth from about 4 to 12 kHz was achieved, while lowering the noise a fairly small amount.

Then came the question of the best frequency response for the monitor. In an English dubbing stage, Allen did an experiment with a nearfield, flat hi-fi loudspeaker vs. the farfield film monitor loudspeaker, a VitaVox. He adjusted the frequency response by equalizing the film monitor until the balance was similar, although the monitor loudspeakers of the day only extended to about 8 kHz before giving up the ghost. The electroacoustic response curve Allen found measured with a microphone was flat to 2 kHz, then down 1 dB per one-third octave, to -6 dB at 8 kHz, and falling beyond. This was named the X curve, for eXtended response, whereas the older Academy curve got dubbed the N curve, for Normal response (although one wouldn't consider it normal today).

When extended-range compression drivers and constant directivity horns became available around 1980, the question became, "How should the X curve be applied to this new development?" The new systems had a full octave of high-frequency bandwidth over older systems, but delivered nearly the same output response across a range of angles, rather than concentrating the response on axis as frequency went up as the older driver-horn combinations did.

One theory floated in the middle '70s was that the need for a house curve was based on an artifact of the method of measurement rather than a real need for sound to be rolled-off at high frequencies in large spaces. This was because the quasi-steady-state pink noise stimulus measured by a real-time analyzer in a room is time blind, lumping the direct sound, reflections, and reverberation together indistinguishably. If the different soundfields had different responses, the pink noise stimulus plus RTA could not sort out the differences and would basically average all the responses. Since the microphone is in the farfield of the loudspeaker where reverberation is dominant, then the response with a collapsing directivity horn vs. frequency could be expected to be rolled-off at high frequencies, since the contribution of all the off-axis angles would dominate over the direct sound. Nevertheless, in this condition, the direct sound could be flat, and we might respond to the flat direct sound and ignore the later-arriving response as listeners.

If we then were to change to a constant directivity horn, with its output more constant over all angles within its coverage, and the system is tuned to a "house curve," then it might be expected to sound duller than the older horns, at least on axis at a distance. That's because, under these conditions, both the direct sound and the reverberant sound would be rolled-off and on the same curve. So one of the first experiments I did on this combination was to play conventionally mixed program material over constant directivity horns equalized to the X curve to see if the sound was too dull. It was not; in fact, with the bandwidth extension from 8 to 16 kHz, it actually sounded somewhat brighter, but this was due to the extended compression driver response instead of having to do with the equalization curve.

So what's going on here? This was later explained by Dr. Brian C. J. Moore, author of numerous refereed journal articles on psychoacoustics and the book, An Introduction to the Psychology of Hearing. The rolled-off house curve has a good basis in psychoacoustics, because a soundfield originating at a distance is "expected" to be more rolled-off than one originating nearby. It is a little like optical illusions in vision that show, despite occupying the same area on the retina, pictures look bigger on a larger screen, even when a small screen is closer and takes up the same horizontal and vertical angles. As it turns out, both spectrum and level are affected by the perception of the size of space you are in, and "getting it to match" perfectly from large to small room in physical sound pressure level and response does not result in sounding the same.

With the additional octave of high-frequency extended range of more modern drivers and horns came the need to calibrate the X curve to the highest audible frequencies. Later editions of the SMPTE and ISO standards show the roll-off to 8 kHz as originally standardized, but added rolloff from the extended curve in the bands above 8 kHz. Some users don't employ this additional roll-off, staying on the original X curve to 16 kHz, but in an experiment I did at USC, I found that following the letter of the standard was an improvement in high-frequency balance and interchangeability of program material. This was done in a very sensitive experiment, reported earlier in Surround Professional, that involved playing trailers in a large theater exactly as they sounded in the dubbing stage, with the agreement from the people who had supervised their mixes that they sounded correct, and this involved eight trailers mixed in a variety of studios. Both level and response standards had to be perfect to accomplish this, and just a 1 dB error over several octaves that crept in during setup was heard, and had to be corrected.

Another development of the X curve is how it should vary with room volume. Although a variation in the response with room volume was written into the original standard, further work shows that the response should be "hinged" at 2 kHz, and turned up at high frequencies in smaller rooms. Curves that extend the range out to higher frequencies before breaking away from flat do not seem to interchange as well.

Today, the major factors affecting interchangeability no longer have to do with the target curve, since the X curve is very well accepted, but rather have to do with how the curve is to be measured and adjusted electroacoustically. The standard calls for such needed items to make good measurements of quasi steady-state noise as spatial averaging, temporal averaging, and the proper use of measurement microphones. The largest variations among different practitioners are in the use of microphones. The problem is that the soundfield seen by a microphone in a large room is a mixture of direct sound, early reflections, and reverberation. Standard measurement 1/2-inch microphones demonstrate very different high-frequency response when measured anechoically on axis and with a diffuse field. Differences are on the order of 6 dB in the top octave between the two, and response in rooms is highly affected by the differences between these two. Only by the use of small, low-diffraction microphones, such as 1/4-inch or smaller diaphragm mics, are the differences kept small.

The best usage of measurement microphones today is to calibrate small ones for grazing incidence across the diaphragm rather than perpendicular to the soundfield, because, this way, the microphone will demonstrate the most similar response for the direct sound (across the diaphragm) and reverberation (a diffuse field). One of the primary ways in which problems show up in this area is in the difference exhibited between sound originating from a more-or-less point sound screen channel vs. a surround array: 1/2-inch microphones make serious errors between these two because the soundfields generated under the two conditions are so different.

The X curve now has nearly a quarter century of use and has absolutely acted to help interchange in the industry. Combined with level standards, and de facto industry standards such as speaker directivities, the whole film industry has benefited without a doubt. Problems linger in applying the standards uniformly due to different methods of measurement. Also, when heard over a modern flat loudspeaker in a small room, program material balanced on an X curve monitor sounds overly bright. That's because the original experiment that set the curve was made many years ago, without the frequency range available from today's components. This is not too important because, so long as everyone agrees to use the same curve, then the response sounds the same to the mixer on the dubbing stage as to the audience member in any auditorium. Interchangeability of X curve material with home video can be handled with a simple re-equalization. The ATSC television standard recognizes the differences, sending a flag that tells receiving equipment whether the program material was balanced on an X curve monitor, or on a flat monitor in a small room, and home equipment can take appropriate action to re-equalize the program accordingly.

Heres a PDF that goes into more detail...

Attached Files
File Type: pdf the_mythical_x_curve.pdf (528.5 KB, 851 views)
Old 8th August 2008
Some notes on CONVERSION of Audio for PAL / NTSC / FILM

If the PAL version was a frame-to-frame transfer, and is thus running faster than the original film (25fps vs 24, or even PAL video at 25 and HD video at 23.98), the key is to first get your stems back up to film speed, which would involve an SRC using 47952 as the source sample rate. THEN, when it's back at film speed (48k), you could do the NTSC>PAL SRC, the 4.1% thing. Maybe? Try just the center channel using the fastest setting as a test, and if it works you can do the whole thing at "Best" quality. Of course, the pitch will be faster (about 1/4 step) as well, in sync with the faster speed of the film. To get it in sync but at the original pitch you have to do a whole time-compression/pitch-shift thing which is a whole different process.

25 to 23.976, Divide those two numbers, and you get 1.0427093760427093760427093760427. When truncated and expressed as percentage, it becomes 104.27094% which is the number that you feed time conversion software such as, Prosoniq Timefactory or Nuendo timestretch

NTSC to PAL would be 23.976/25 = 0.95904 = 95.904%
NTSC to PAL SR = 48000(25/23.976) = 50050 rounded.
PAL to NTSC SR = 48000(23.976/25) = 46034 rounded.

A time-stretch version is recommended, vs SRC conversion.
MPEX2 in Nuendo is apparently a good tool.

Old 25th November 2008
Some Misc troubleshooting for APPLE and PROTOOLS

Some Troubleshooting tips for APPLE based PROTOOLS system that I've collected:

Pro Tools Tech Support Folder , tools and utilities to assist in diagnosing and troubleshooting is here: Digidesign | Support | Tech Support Folder - Utilities and Troubleshooting Tools

Other house keeping items:

Delete Pro Tools preferences.
Go to Users > “your user name” > Library > Preferences
Delete 'com.digidesign.protoolsLE.plist', 'DAE Prefs' (folder), 'DigiSetup.OSX' and 'Pro Tools preferences'.
Empty trash, then restart the computer.

You can also use the Pro Tools Preference and Database Helper to trash preferences and databases automatically, as well as providing a few other useful tools. (This application is provided by a 3rd party and is not tested or supported by Digidesign)

Repair Permissions
-Quit Pro Tools and launch Apple's "Disk Utility" application, located in:

MacHD>Applications & Utilities.

-Select your boot drive (the whole drive, not the volume underneath the drive)
-Go to the 'First Aid' tab and select "Repair Disk Permissions"

Apple recommends doing this any time you install new software, update your OS or reinstall any software.

Databases and Volumes

This step can be useful when receiving random 'assertion' or 'neoaccess' errors, especially when recording or saving.

-Delete the "Digidesign Databases" folders on the first level of all mounted hard drives.
-Delete the "Volumes" folder:
Pro Tools versions 7.3 and earlier it's located in MacHD > Library > Application Support > Digidesign > Databases.
Pro Tools 7.4 and higher will find it in MacHD > Library > Application Support > Digidesign > Databases > Unicode.

-Empty trash, then restart the computer.


First verify you are using a supported Mac and your system meets all of the minimum requirements. You can find compatibility information (computers, hard drives, operating systems, requirements, etc.) in the Support section of the website:
Digidesign | Support

Choose from the product list for compatibility information, and click on the links in the compatibility section for each product for additional information.

- Verify that you have the Minimum or Suggested amount of Ram loaded in your system
For current Pro Tools systems, that is the following:
1 GB (1024 MB) or more highly recommended. The best user experience has been reported using 2 GB or more.

- Verify that your version of the Mac OS is supported with your version of Pro Tools:
Pro Tools LE Version Compatibility Grid for Mac OS X
More Information:
Mac OS X 10.4 Requirements with Pro Tools 6 & 7
Mac OS X 10.2 & 10.3 Requirements with Pro Tools 6

- Make sure all drives are formatted Mac OS Extended (Journaled) with OS X’s Disk Utility. Pro Tools can not use UFS or HFS volumes. If the drive was originally formatted in OS 9 or with any other application, backup the drive and reinitialize it with Disk Utility. More information:
Hard Drive Requirements - Pro Tools LE for Mac OS X

IMPORTANT! You MUST use a secondary hard drive (not your main OS drive) for recording and playback of audio in Pro Tools. Recording or playback from the OS drive is known to be problematic and the cause of many different error types. If you are using your system drive and encountering errors, the first thing you should do is get a compatible drive.

Pro Tools supports recording and playback to a secondary drive that meets the following requirements:

7200 rpm or faster
9ms seek time or faster

Firewire drives must have the Oxford 911 (FW400 port), Oxford 912 (FW400 & FW800 ports) or Oxford 924 (FW800 ports) Bridge chip. USB drives are not supported and are known to be problematic. Here are some useful links to determine if the drive you have, or are considering, has the proper chipset:

Chipsets used in Maxtor external drives
Chipsets used in Seagate external drives

Pro Tools FireWire Drive Requirements on Mac OS X
Pro Tools EIDE/ATA Hard Drive Requirements on Mac OS X
Pro Tools SATA Hard Drive Requirements on Mac OS X
Pro Tools SCSI Hard Drive Requirements on Mac OS X

- If you are running a supported ATTO card make sure you install the ATTO Configuration Tool, this is a different application from Express. It can be found on the Pro Tools CD installer or in the downloads section. This application installs necessary extensions for all supported ATTO cards. More information:
Qualified SCSI HBA Cards — Pro Tools Systems for Mac OS X

General Setup

Pro Tools Technical Support, Registration and Setup Videos

These videos have information on the following topics:

Product Registration
Technical Support - Searching Answerbase and getting answers to your questions
Pro Tools System Compatibility - How to determine if your system is compatible
Mbox 2 family - Information on what's in the box
Authorizing Ignition Pack 2

- In System Preferences > Display, set your monitor resolution to a minimum of 1024 X 768.

- In System Preferences > Classic > Start/Stop tab, Uncheck Start Classic when you log in.

- In System Preferences > Date & Time, verify that the date is set correctly and that you are not using 24 hour time.

- If you are using a 2005 or newer Powerbook G4 with Sudden Motion Sensor, please disable SMS according to the Apple information located here:

Sudden Motion Sensor and video editing performance
Sudden Motion Sensor: Advanced Tips

Energy Saver

-Open System Preferences (Located in Apple Menu, Dock, or Applications Folder)
-Click on Energy Saver (in the Hardware section)
-Set the "Sleep Sliders" to Never (Computer Sleep) and Never (Display Sleep)
-Make sure the box next to "Put the hard disk(s) to sleep when possible" is unchecked
-Click on the options tab at the top
-If you have a "Processor Performance" drop down menu select "Highest"


-Click on the AirPort Icon in the Menu Bar in the upper right corner of the screen (Left of the time)
-Select "Turn AirPort Off" from the menu
-If you don't see the AirPort Icon in the Menu Bar then:
-Open System Preferences (Located in Apple Menu, Dock, or Applications Folder)
-Click on "Network"
-Click on the "Show" drop down menu and select "AirPort" (if there is no AirPort option, then AirPort is not installed on your computer)
-Under the "AirPort" tab, towards the bottom of the window, check the box next to "Show AirPort status in menu bar"
-Follow the first two steps after that is done


-Open System Preferences (Located in Apple Menu, Dock, or Applications Folder)
-Click on "Bluetooth" under the Hardware section
-Click on the "Settings" tab at the top of the screen
-Make sure "Bluetooth Power: Off" if not click the "Turn Bluetooth Off" Button

Firewire Networking

- Open System Prefs (Located in Apple Menu, Dock, or Applications Folder)
- Click on 'Network'
- In the drop-down box next to 'Show', select 'Network Port Configurations'
- Uncheck the box next to 'Built-in Firewire'
- Click on 'Apply Now' button in the lower right hand corner of the window.

Pace Drivers

You want to make sure you have the most current version of these drivers.

-Visit the PACE website at:

Welcome to PACE Anti-Piracy

-Below where it says "END USERS" on the right side of the page, there is a 'Download Drivers' drop down menu
-Click on the menu and select 'Mac OS X Extensions'
-This should start downloading 'macextsx.dmg'
-Once this file is downloaded, double click on 'macextsx.dmg'
-This should bring up a temporary disk called 'InterLok Extensions Installer'
-Open the 'InterLok Extensions Installer' disk, and double-click 'InterLok Extensions Install'
-Proceed through the installation
-Restart your computer when installation has finished

Troubleshooting tips

Please download and install the Pro Tools Tech Support Folder , which has tools and utilities to assist in diagnosing and troubleshooting potential problems you may encounter.

Delete Pro Tools preferences.
-Go to Users > “your user name” > Library > Preferences
-Delete 'com.digidesign.protoolsLE.plist', 'DAE Prefs' (folder), 'DigiSetup.OSX' and 'Pro Tools preferences'.
-Empty trash, then restart the computer.

You can also use the Pro Tools Preference and Database Helper to trash preferences and databases automatically, as well as providing a few other useful tools. (This application is provided by a 3rd party and is not tested or supported by Digidesign)

Repair Permissions
-Quit Pro Tools and launch Apple's "Disk Utility" application, located in:

MacHD>Applications & Utilities.

-Select your boot drive (the whole drive, not the volume underneath the drive)
-Go to the 'First Aid' tab and select "Repair Disk Permissions"

Apple recommends doing this any time you install new software, update your OS or reinstall any software.

Databases and Volumes

This step can be useful when receiving random 'assertion' or 'neoaccess' errors, especially when recording or saving.

-Delete the "Digidesign Databases" folders on the first level of all mounted hard drives.
-Delete the "Volumes" folder:
Pro Tools versions 7.3 and earlier it's located in MacHD > Library > Application Support > Digidesign > Databases.
Pro Tools 7.4 and higher will find it in MacHD > Library > Application Support > Digidesign > Databases > Unicode.

-Empty trash, then restart the computer.

Uninstall and reinstall Pro Tools
-Run the Pro Tools installer from your CD-ROM or web download.
-Click on the Custom Install menu and choose "Uninstall".
-Click continue and then choose the option for "Clean" uninstall
-Click OK when it reports that the uninstall was successful and then reinstall Pro Tools.

New User Account

Try creating a new user with admin privileges in System Preferences > Accounts.
-Click the Lock to authenticate
-Enter password
-Click the "+" (plus sign) under the list of users
-Type In Pro Tools for the Name
-Enter a password and verify (optional)
-Check the box 'Allow user to administer this computer'
-Click 'Create Account'

Then login to this new account and run Pro Tools

-Go to the Apple menu and go to 'Log Out (Username)'
-Login in to Pro Tools
-If there is no icon in the Dock, navigate to MacHD>Applications>Digidesign>Pro Tools
-Double-click on 'Pro Tools LE'

Disable Virus Protection
Disable any Virus Protection Software. Check for evidence of anti-virus software in Library > Startup items. Move these items out of this folder and restart the CPU (do NOT remove the DigidesignLoader or PACESupport folders).

Other Troubleshooting
- Remove any unnecessary USB or Firewire devices or any other extraneous hardware and then restart the computer.

- In the case of possible hardware issues with your Mac the next step is to test the Mac hardware. To do this, boot off of the Apple Hardware Test CD or Software Restore DVD (depending on Mac model) that shipped with your Mac. If you are booting off of the Software Restore DVD, hold the option key while the Mac starts up, then choose Apple Hardware Test. Once Apple Hardware Test loads, press Control+L. This puts Apple Hardware Test in loop mode. Click Extended Test, and let it go (preferably overnight). When you return, the hardware test will have found an error or it will be continuing. If it finds an error, it will give you an error code and you will be able to see which iteration of the loop it found the error on. If it has not found an error, it will still be looping and it will indicate which iteration of the loop is currently in progress. Either way, the only way to restart your Mac is to hold in the power key to shut it down first.

Login Items

Certain items that always start up when you log in to your computer can conflict with Pro Tools operations. The way to check what is starting up every time you log in is:

-Open System Prefs (Located in Apple Menu, Dock, or Applications Folder)
-Click "Accounts"
-Select your account from the list on the left hand side of the window
-Click on the "Login Items" tab on the right hand side of the window
-Go through the list and select each item and click the "-" (minus) button below the list to remove the item


Certain items that always start up when turn on your computer can conflict with Pro Tools operations. The way to check what is starting up every time you start your computer is to navigate to:


Any Digidesign Files/Folders located in this folder should be okay for use with Pro Tools. Common files/folders to have are:

Digidesign Loader
Digidesign Mbox 2
PACE Support

Any other files/folders should be backed up and removed for optimal Pro Tools operation


FileVault is an Apple utility to help protect your files. When FileVault is turned "ON" this can conflict with Pro Tools and the installation of Pro Tools. Make sure FileVault is disabled following these steps:

-Open System Prefs (Located in Apple Menu, Dock, or Applications Folder)
-Click "Security"
-In this window look where is says:

FileVault protection is (on/off) for this account

-If FileVault is on click the button to turn Off FileVault

Spotlight Indexing

If Spotlight Indexing is running in the background, this can cause errors in Pro Tools. To disable Spotlight Indexing follow these steps:

To turn this off there are two methods, both require administrator permissions:


Start up the Terminal (/Applications/Utilities/)
Type the following:

cd /etc

sudo pico hostconfig (enter your password or press return)

An editor will open with the following entry: SPOTLIGHT=-YES- Replace 'YES' with 'NO'

press ctrl-x, then 'Y' and then press return Close the Terminal and restart the computer.

Use Spotless:

You may also use Spotless, but it is a paid option ($12.95). Download Spotless at Spotless information page. Installation and usage instructions are included with the download.

Only the indexing will be disabled, you will still be able to search with Spotlight after using this utility.


If you are having performance issues with Pro Tools, you might also try disabling the Dashboard for Tiger.

How to disable Dashboard

1) Open Terminal and type:

defaults write mcx-disabled -boolean YES

2) once that is done you need to "restart" the dock by typing:

killall Dock
(make sure you get the capital 'D')

How to enable Dashboard

1)Open terminal and type:
defaults write mcx-disabled -boolean NO

2)once that is done you need to "restart" the dock by typing:

killall Dock
(make sure you get the capital 'D')

Plug-In Compatibility

Make sure that all Plug-Ins are compatible with Pro Tools 7.

-Open MacHD>Library>Application Support>Digidesign>Plug-Ins
-Check the Version of each Plug-In

You can check the version by selecting the plug-in and go to the "File" menu and choose 'Get Info', or hit Command-I and look at the version information under the 'General' Section

-Digidesign Plug-Ins should be version 7.0 (Dynamics III version 6.9) or higher
-Check with third party plug-in manufacturer's to confirm that the plug-in version you have is compatible with Pro Tools 7, and here:

Pro Tools 7 Plug-In Compatibility

Linked from the Pro Tools Plug-Ins section of the website:
Pro Tools Plug-Ins

General NeoAccess or Assertion errors

If you receive one of these errors, usually after recording or when trying to save a session, it can usually be resolved by following the steps for trashing preference and database files listed above.

How to Remove Expired (Demo) Plug-Ins & Software:

- See Answerbase 19348

If you're experiencing noise from the Mbox 2, there are several things to try:

- Make sure you're using balanced cables on all balanced inputs and outputs. Remember that the Mbox 2 outputs are unbalanced, and should use unbalanced connections.
- Try using a power conditioner for all your electronic gear (Furman, Monster).
- Run your outputs directly to shielded monitors and not through a mixer to see if the noise goes away.
- Some noise complaints arise from noisey power supply units inside computers - try installing and running on another computer (or if you're using a laptop, unplug the power supply and run off of batteries to see if noise goes away).
- Unplug and turn off all unnecessary gear and run only the computer, the Mbox 2, and monitors.
- Listen through headphones.
- Make sure you're not running through a USB hub or other device - the Mbox 2 always needs to be plugged directly into the computer.
- Try all USB ports on the computer to see if the noise lessens.
- Trace your cable path for long cable runs, and isolate your cables from possible interference (other gear, florescent lighting, CRT monitors, etc).

Lastly, if you feel a repair is necessary, please contact Digidesign Tech Support to set up a Return Authorization:
Digidesign | About Us | Contact Digidesign | USA & Canada Contact Information

Digi 002/002R users experiencing any combination of the following symptoms and whose serial number ends in the letter A-F should contact Digidesign Tech Support by phone for information on getting your unit serviced:

The unit keeps on clicking when powering up, or when launching Pro Tools.
"Unable to locate Digidesign Hardware" error message when launching Pro Tools.
Dots or blank scribble strips (002 only)
Firewire light in back is blinking or turned off completely
Sample Rate light is off or blinking
Unit won't power up, or only powers up intermittently
Only the Mute light is lit when powering up and you are unable to disable Mute
Old 27th November 2008
Gear Maniac
jimlongo's Avatar

Originally Posted by georgia View Post
Some Troubleshooting tips for APPLE based PROTOOLS system that I've collected:
Some great tips there Georgia, thank you.
You might want to add Applejack to your arsenal of computer maintenance items.

Let's you fix your hard drive, fsck, permissions, delete caches and virtual memory, and check for bad preferences from single user startup mode. That and periodic in the terminal is all I use nowadays.
Old 2nd December 2008
wages for industry personnel...

here's some starting points for wages in the industry.... hourly and weekly rates are listed.

Motion Picture Editor
" On Call " 5-day week $2,653.16

Trailer Editor
Weekly Guarantee 45 hrs 1.5x after 40

Supervising Sound Editor
"On Call" 5-day week

Sound Editor
Weekly guarantee 48.6 hrs
1.5x after 48.6

Head Music Editor
"On Call" 5-day week

Music Editor
Weekly guarantee 48.6 hrs
1.5x after 48.6

Assistant Editor
Weekly guarantee 45 hrs
1.5x after 40

Apprentice Editor
Weekly guarantee 40 hrs
1.5x after 40

Head Librarian
"On Call" 5-day week

Supervising Librarian
Weekly guarantee 43.2 hrs
1.5x after 40

Weekly guarantee 43.2 hrs
1.5x after 40

Entry Level
Music/ Re-recording Mixer
Daily guarantee 9 hrs
1.5x after 9 hrs
$59.43/ hr
$534.87/ day

Music/ Re-recording Mixer
Weekly guarantee 48.6 hrs
1.5x after 48.6 hrs
$50.410/ hr
$2,449.92/ wk
$43.840/ hr
$2,130.61/ wk

Supervising Engineer
Daily guarantee 9 hrs
1.5x after 9 hrs
$59.43/ hr
$534.87/ day

Supervising Engineer
Weekly guarantee 48.6 hrs
1.5x after 48.6 hrs
$50.410/ hr
$2,449.92/ wk
$43.840/ hr
$2,130.61/ wk

Daily guarantee 9 hrs
1.5x after 9 and/or 40 hrs
$46.15/ hr
$415.35/ day
$40.22/ hr
$361.98/ day

Weekly guarantee 48.6 hrs
1.5x after 40 hrs
$40.533/ hr
$2,144.18/ wk
$35.439/ hr
$1,874.73/ wk

Service Recorder
Daily guarantee 9 hrs
1.5x after 9 and/or 40 hrs
$40.53/ hr
$364.77/ day
$35.45/ hr
$319.05/ day

Service Recorder
Weekly guarantee 48.6 hrs
1.5x after 40 hrs
$36.340/ hr
$1,922.39/ wk
$31.877/ hr
$1,686.29/ wk

Utility Sound Technician
Daily guarantee 9 hrs
1.5x after 9 and/or 40 hrs
$40.53/ hr
$364.77/ day
$35.45/ hr
$319.05/ day

Utility Sound Technician
Weekly guarantee 48.6 hrs
1.5x after 40 hrs
$36.340/ hr
$1,922.39/ wk
$31.877/ hr
$1,686.29/ wk

Microphone Boom Oper.
Daily guarantee 9 hrs
1.5x after 9 and/or 40 hrs
$40.53/ hr
$364.77/ day
$35.45/ hr
$319.05/ day

Microphone Boom Oper.
Weekly guarantee 48.6 hrs
1.5x after 40 hrs
$36.340/ hr
$1,922.39/ wk
$31.877/ hr
$1,686.29/ wk

Daily guarantee 9 hrs
1.5x after 9 and/or 40 hrs
$38.84/ hr
$349.56/ day
$34.01/ hr
$306.09/ day

Weekly guarantee 48.6 hrs
1.5x after 40 hrs
$34.942/ hr
$1,848.42/ wk
$30.690/ hr

Sound Service Person
Daily guarantee 9 hrs
1.5x after 9 and/or 40 hrs
$16.58/ hr
$149.22/ day

Sound Service Person
Daily guarantee 9 hrs
1.5x after 9 and/or 40 hrs
$16.58/ hr
$149.22/ day

Old 2nd December 2008
Lives for gear

What is the source for your last post on salaries?
Old 2nd December 2008
Motion Picture Editors Guild... 2008 - 2009 data...

Motion Picture Editors Guild - Wages

Old 2nd December 2008
Lives for gear
dr.sound's Avatar

Originally Posted by georgia View Post
Motion Picture Editors Guild... 2008 - 2009 data...

Motion Picture Editors Guild - Wages

To be very clear, most (all people) should go under "Independent Post Production Agreement" NOT "The Majors". Also, If you're in LA, use the West Coast, New York, The East Coast agreement. These wages are base pay. Some people get "over scale" depending on their pull, experience, etc. Many places are not paying over scale these days.
Old 3rd December 2008
you are most assuredly correct Doc! this was intended as a conceptual set of numbers for a starting point. obviously some people can and will charge more, some need to charge less.... And every project is different...

Old 7th January 2009
Dolby Surround LtRt info

DOLBY surround. LtRt Left total, Right total vs LoRo L only R only.. it started as a consumer version of Dolby's multichannel film format.
LtRt is a L C R S ( left center right and mono surround ) mix that is encoded down to a , and i'm going to use this very loosely, "stereo" mix. The "stereo" mix is really the four tracks encoded to 2 tracks. Then upon playback the LtRt can be heard in just plain mono or stereo or decoded back to L C R S. LtRt or Dolby Surround was the first of Dolby's multichannel film format. ( 1980 something ) The encoded file fits nicely on VHS or other 2 track device and then you get surround off of these 2 track systems.
The encoding process relies heavily on phase to encode and decode the positional information within the mix.

you can use a DOLBY SEU 4 or Dolby DP563 hardware devices to encode LtRt, or you can create an LtRt optical track starting with a Dolby DMU or LtRt files... You can use the DOLBY surround tools in protools to create LtRt ( both encode and decode )...

remember that an LtRt track is just analog audio vs an AC3 or Dolby-E encode which is a digital bit stream. You can create an AC3 encode using only L C R S very similar to an LtRt for DVDs with A-Pack or other AC3 encoder, but again it will create a digital bit stream, not an analog audio file.

to mix... just mix in L C R S... paying close attention to any coupling or phase issues with similar materials in multiple speakers... music, ambience etc. then downmix to LtRt and LoRo and mono checking each for positing of material. When mixing you can use the Dolby Surround tools to create 2 stereo aux tracks with the encoder on them Master( L,R ) slave( C,S ) and them place the decoder (master, slave) after the Encoder so that you can monitor thru the encoder/decoder and simply hit a button to hear whats going on in LCRS, LR, mono... the details will become obvious when you look at the software plugins...


PS: i think today the whole this is called DOLBY ANALOG by Dolby... sorry I haven't had my morning coffee so this reply is a little disjointed..


some additional from Neil Wilkes across the pond...

There is Dolby ProLogic and Dolby ProLogic II, which are utterly different systems despite the incremental naming protocol - DPL has dual mono rear channels & feeds the same information to both Ls & Rs and DPL II uses proper separate Ls/Rs where available. In addition, all DPL II en/decoders are fully backwards compatible with the earlier systems.
SRS Circle Surround is also a Matrix Lt/Rt technology.

Lt/Rt is basically a generic term taken to mean any multichannel mix that is matrixed down into a stereo compliant stream & will play back on a stereo system, but if the stream is fed through the correct decoder you will get the surround mix back out of it.

There are Pros & Cons to this.
Pros -
1 - you get to supply a single stream that will play back in stereo if no surround setup is present.
2 - It is widely used in TV/Broadcast.
3 - You can even take the Lt/Rt stream & further reduce to AC3 (Dolby Digital) as long as you set the metadata flag (in a soft encoder) or push the switch/button (in a hardware encoder) for "Dolby Surround Encoded"
Cons -
1 - It is a compromise for both types. If you start with a fully discrete 5.1 mix, then it will *not* sound the same when decoded again. Neither will the stereo mix be as good as one that was especially mixed for stereo.
2 - It's a matrix system - not discrete - and as Georgia points out there are all manner of possible pitfalls involved.
3 - It makes delivering an M&E mix much more complex as you would need to completely reset the encoder resulting in a vastly different sounding mix on a dubbed language version.

When I have to deliver an Lt/Rt stream I use a VST plugin en/decoder so that switching between the various modes is extremely easy (source/encoded/decoded).

A good place to start research on this is at the Dolby Labs technical library where you will find a lot of helpful material.
Old 31st March 2009
Some mixing info from Danijel... cool stuff.

Standard Mixing Levels for Movie Theater, DVD, TV, Radio and Games
This post should serve as a little guide to the resources available on-line on the topic of audio levels in different media. It has been compiled due to the big frequency of questions on the topic, and thanks to the big amount of answers in this forum!
Since audio in media is an ever-changing field, this post will be updated as I stumble upon new and interesting infos or links. If you have insight into data that you think should be included or corrected, please PM me, or post it here.
For further, specific questions on mixing levels, you can post in this thread, or start a new one.

Movie theater

There are no guidelines in terms of average loudness, peak or any other level measurement. You achieve proper levels by properly calibrating your listening environment, so that it resembles the environment of the theater.

To calibrate your room, read this:
DUC: Room Calibration for Film and TV Post
(or, in a nutshell)
Then mix by ear. "If it sounds good, it is good" - JoeMeek.

Here's a useful discussion:
FILM & Broadcast - Levels

However, there is a maximum loudness level for theatrical trailers and commercials which is measured with the Dolby Model 737 Soundtrack Loudness Meter.
Trailer loudness should not exceed 85 dB Leq(m), as regulated by TASA.
Commercial loudness should not exceed 82 dB Leq(m), as regulated by SAWA.


Here, same rules apply as with the theatrical mix, except that the monitoring is different (near-field, no X-curve), the room is smaller, it is calibrated lower, AND there is the dialnorm parameter if your sound is AC3 encoded.

Read about dialnorm here:
Geo's sound post corner (section about Dialogue Level)
and here:
Home Theater Hi-Fi: Dialogue Normalization

You have to determine your target dialnorm BEFORE you start mixing, so you can adjust your listening level accordingly. Most DVD's are mixed for dialnorm -27dB (because that setting is the most compatible with the theatrical mix), but some use the full dynamic range (-31dB).

TV (everything BUT commercials)

Every broadcaster has it's own specs. You have to get the specs of your target TV channel.

Detailed Specs

They can be very detailed, like the Discovery specs or the PBS specs (section 3). They will tell you exactly what is your max peak level, average dialogue level, average overall level, what measurement instrument is to be used etc. Meter that the networks usually specify is Dolby LM100.

Here are two threads about mixing against LM100:
Mixing with the Dolby LM100
Anyone have experience mixing while adhering to specs monitored by the Dolby LM 100?

This is great! A post by Mark Edmondson, Audio Post Production Supervisor at Discovery:
Dolby LM100 and Discovery deliverables - Digi User Conference

Basic Specs

The other extreme is on the minimalistic side, like the RTL or BBC specs which give you only the maximum peak level, and the reference level. This is what it's like in most of Europe, AFAIK (if you have some bogus specs to share, please post some links).

- REFERENCE LEVEL - it is used for equipment alignment, and doesn't have a direct relation to actual mixing levels.
In EBU countries it is -18dBFS and corresponds to electrical level of 0dBu (per EBU R68).
In SMPTE countries it is -20dBFS and corresponds to electrical level of +4dBu (per SMPTE RP155).
Sometimes refered to as: Zero level, Line-up level, 0VU.
Broadcast Audio Operating Levels for Sound Engineers
Reference Levels on Common Metering Scales
The Ins and Outs Of (Sound on Sound)

- MAXIMUM PEAK LEVEL - this is where you set your brickwall limiter on the master buss, or otherwise not go over it (although in some of the specs, short peaks of 3 to 5 dB over this value are allowed - go figure!).

What can make the confusion here is that the average dialogue level is not exactly specified.

In a perfect world, you would calibrate your listening environment to the ITU-R BS.775-1 standard (-20dBFS pink @79dB SPL/C/slow), [or EBU 3276 and EBU 3276-S if you are in Europe] and then mix by ear. In that case you would get average dialogue levels at around -27dBFS RMS.

However, this way, your mix could turn out too quiet, as there's a loudness war in broadcasting, probably in part due to the loudness of commercials and the loudness war in music. (e.g. PBS has upped their dialnorm from -27dBFS to -24dBFS in 2007).
Average dialogue loudness that works for me (dramatic program, regional stations in the Balkan peninsula) is -22dBFS RMS. To achieve that, I calibrate my monitoring to 74dB, and thus reduce the headroom by 5dB when compared to the ITU's 79dB reference.

However, your best bet is to talk to someone who regularly delivers for the given broadcaster or in a given market, and ask him about his average dialogue level, or how his listening is calibrated. Chances are someone at this forum will be able to help, too.

Further Reading

More about broadcast delivery specs:
Geo's sound post corner

A great intro to broadcast audio:
Audio for Digital Television

All this and much more:
CAS Seminars - 'What Happened to My Mix?' - The Work Flow From Production Through Post Production - Cinema Audio Society

Dialnorm was to be implemented in broadcast too (as Dolby imagined), but it isn't, so far:
DTV Audio: Understanding Dialnorm

Food for thought on setting up variable monitoring level:
Bob Katz - Level Practices

TV commercials

Again, you have to get the specs of your target TV channel, but you will most likely only use the max peak value they provide. Below that, you can compress as much as you wish - it's a loudness war, similar to the one in popular music production.
There are efforts in regulating this problem:
US: H.R. 6209: Commercial Advertisement Loudness Mitigation Act (
UK: UK commercials for TV - perceived loudness issue - Digi User Conference


I can't say much about radio levels, so perhaps someone who is experienced with radio could chime in.

Here's a BBC technical specification, but I don't know how much it applies to different radio stations:
BBC Radio Resources // Programme Delivery // Glossary

Less is more (straight from the horse's mouth) - Bob Orban talks about what goes on with your mix in the radio station:
Radio Ready: The Truth


Absence of standards:
Video Game Reference Level/Standards
THX: Establishing a Reference Playback Level for Video Games

A thread at SDO with some advice and some official information from Sony and Microsoft:
Niveau Sonore en jeux vidéo :: Sound Designers.Org (Babelfish English translation)
(Note: the Xbox360 document is in English)

Old 12th April 2009
Some info on TV broadcast

Tim Carol wrote this...


There is little sense in having an emission coding system such as Dolby Digital (AC-3) that can carry 5.1 channels of audio if you can't get 5.1 channels to the encoder. In 1996, all commonly used VTRs had only four audio channels; servers could in theory do more but were not generally configured that way, and digital plants rarely had more than two AES pairs available for routing. Once the Dolby Digital (AC-3) system was in place as part of the ATSC standard, Craig Todd, Louis Fielder, Kent Terry and others at Dolby began to investigate ways to efficiently distribute the multiple channels of audio-they foresaw issues that were still a few years away from becoming a really big problem.

So what is Dolby E? Contrary to some rumors, it is not high-rate Dolby Digital (AC-3). This approach was considered, but there were too many benefits to be had for starting over with a different set of goals. What I mean by that is the goal of the Dolby Digital (AC-3) system is to deliver up to 5.1 channels of audio to consumers using the fewest bits possible while still preserving excellent audio quality. As we will see, this is not the goal of Dolby E. To meet this goal, the Dolby Digital (AC-3) encoder is rather complex and takes about 187 milliseconds from the time it receives audio until the time it produces a Dolby Digital (AC-3) output. This is analogous to the video encoding process-high-quality but low bit-rate means the encoder is going to need processing time. Although this encoding latency is small (far less than video encoding latency) and is taken into account in the multiplexer, this amount of delay is difficult to deal with in production and distribution.

Also, while the audio quality of Dolby Digital (AC-3) is very good, it would not be appropriate to use it for multiple encode/decode cycles. This might enable coding artifacts to become audible. I say might because the artifacts are unpredictable-sometimes you might hear them with certain material, sometimes you might not. Again, high-rate Dolby Digital (AC-3) minimizes the chance of this occurring, but it could.

Another drawback of using Dolby Digital (AC-3) for distribution is that although its data is packetized into frames, these frames do not regularly line up with video frames (see Fig. 1).

You might be thinking, "PCM audio is packetized into AES frames that do not line up exactly with video frames either so what is the problem?" Good point, but Dolby Digital (AC-3) frames carry bit-rate reduced (i.e. compressed) audio, not baseband audio. Although a video edit would cause little problem for baseband audio, cutting a mid-Dolby Digital (AC-3) frame will cause major problems. After decoding, the results will be audio mutes if you are lucky, clicks and pops if you are not. Dolby Digital (AC-3) is simply not intended to be used this way.

This did not stop some early adopters, however, and at least one major DBS provider used Dolby Digital (AC-3) recorded on one of the AES pairs of a Digital Betacam recorder to carry the 5.1 channel audio of movies. Did it work? Absolutely, and even when the Digital Betacam tapes were not long enough to hold an entire film and the Dolby Digital (AC-3) stream had to be switched mid-movie, it hardly ever caused a glitch. They were lucky! It can be done, but it is not easy and is not recommended. Clearly, it was time for a new system designed specifically for the task.


Some of the original goals for this new system were that it had to be video frame-bound so that it could be easily edited or switched, had to be able to handle multiple encode/decode cycles (at least 10) while causing no audible degradation, had to carry eight channels of audio and metadata, had to fit into a standard size AES pair of channels and had to do its encoding and decoding in less than a video frame.

Dolby E satisfies this tall order

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. The system will accept up to eight channels of baseband PCM audio and metadata and fit them onto a single 20-bit, 48kHz AES pair (i.e., 1.92 Mbps), or it will fit six channels plus metadata into a 16-bit, 48kHz AES pair (i.e., 1.536 Mbps). After decode, PCM audio and metadata are output to feed a Dolby Digital (AC-3) encoder.

Fig. 2

In Fig. 2 you can see how Dolby E frames match video frames. Although only NTSC and PAL rates are shown, the system will also work with 23.976 and 24 fps material.

Notice the small gap between the Dolby E frames. This is called the Guard Band and is a measurable gap in the bitstream. Switching here allows a seamless splice of the audio signals. In fact, upon decode, the audio is actually crossfaded at the switch point, which is a remarkable feat for compressed audio.

As the goal for the Dolby E bit-rate reduction algorithm was to enable multiple encode/decode cycles or concatenations, the audio quality is maintained for a minimum of 10 generations. This does not mean that at 11 generations the audio falls apart, but rather that absence of any artifacts is no longer guaranteed. I was one of the listening test participants at Dolby during the development of the "E" algorithm. I spent two rather unpleasant afternoons listening to all kinds of audio samples both pre- and post-ten generation Dolby E. I consider myself a critical listener, but those were some of the hardest listening tests I have ever participated in. This was not like comparing apples and oranges, more like comparing two perfect apples-one was ever so slightly different than the other. In a word: Maddening!


Dolby E, like Dolby Digital (AC-3), is carried on a standard AES pair. It can be recorded, routed and switched just like standard PCM audio in an AES pair (see Fig. 3).

Fig. 3

However, there are some strict requirements. The path for the AES pair must be bit-for-bit accurate. This means that there can be no level changes, sample rate converters, channel swaps or error concealment in the path. Remember that although the Dolby E data is in an AES pair, it is not audio until it is eventually decoded. Any processing that causes changes in the data will destroy the information. These "Gotchas" are hidden everywhere, especially sample rate converters, so be prepared to really evaluate your facility. An invaluable resource is the Dolby E partner program, run by Jeff Nelson at Dolby. You can find a bunch of very useful information at Manufacturers and individual products that have been tested to pass Dolby E are listed here.

When baseband audio is not possible, Dolby E has become the de facto standard. Since it began shipping in September 1999, more than 1,000 encoders and decoders have been sold, and the system is the source for virtually all 5.1 channel Dolby Digital (AC-3) broadcasts here and abroad.

One last point. The question I was probably asked most often was: "Why is it called Dolby E?" Simple: "E" comes after "D." So, as Steve Lyman likes to say "Dolby E is for Distribution and Dolby D (i.e. AC-3) is for Emission."

Attached Images
Geo's sound post corner-tn_t.1849_i.01_f-tc-06_25_03-fig1.jpg Geo's sound post corner-tn_t.1849_i.02_f-tc-06_25_03-fig2.jpg Geo's sound post corner-tn_t.1849_i.03_f-tc-06_25_03-fig3.jpg 
Old 12th April 2009
Sync in audio / video systems

Another writ up from Tim Carol I thought interesting.

Let's explore some long-held beliefs about the perceptibility of A/V sync. Most film editors can easily detect sync errors of plus or minus one-half of a film frame. As the frame rate in the U.S. is 24fps and in Europe it is 25fps, this equates to approximately +/- 20 msec. Other figures abound, such as plus or minus one video frame (+/- 33/40 msec), and a curiously tipped figure of +5/-15 msec. This last one comes from a specification set by Dolby Laboratories for Dolby Digital (AC-3) decoder performance. The requirement is that audio cannot lead video by more than 5 msec and it cannot lag by more than 15 msec. Why, you might ask, is it tighter in one direction than the other?

It is a fact that light travels much faster than sound, and we are all used to seeing this proven, even if we don't notice it. Here is an example: a basketball hitting the court in a large sports venue will look and sound relatively correct to the first few rows. With the tickets I usually end up with (much, much farther back), the sound lags behind the sight of the ball hitting the floor. If it were possible to get any farther back from the court, the sound would lag even more, but no one seems to complain because it all seems correct. Imagine for a moment if the timing was reversed. As you are watching the game, the sound of the ball hitting the floor arrives before the ball hits. This would be a very unnatural sight and would likely seem wrong even if you were in the first few rows. Human perception of A/V sync is far more sensitive to the unnatural occurrence of sound before action.

The International Telecommunications Union (ITU) released a specification called BT.1359-1 in 1998. It was based on research that showed the reliable detection of A/V sync errors was between 45 msec audio leads video to 125 msec audio lags video. Remember, this is just the detectability region; the acceptability region is an even wider +90 to -185 msec. This study used "normal" people (i.e. no film editors), which helps to explain why the range is so wide. These are worst-case numbers and the goal should always be +/- 0 msec.


A/V sync errors within the TV plant are not new to digital television; they have plagued NTSC for decades. Some basic guidelines to keep in mind are that audio operations, digital or analog, are generally very low latency. Dynamic range compression, equalization, mixing, etc. can all be accomplished in only a couple of milliseconds in the digital domain, falling to microseconds in the analog domain. Rarely, if ever, are compensating video delays required as the latency is so low.

Video processing, on the other hand, takes far more time to accomplish. This is due to both the high bandwidth and the frame-based structure of video signals. Similar to audio, any time a video signal is digitized, operations on that signal will take longer. Unlike audio, however, most video effects do not have the ability to be accomplished in the analog domain, so some video processing delay is inevitable and compensating audio delays are required. It is imperative that compensation be provided for any video device that has delay in excess of a few milliseconds; otherwise A/V sync will become variable and will change as the signal path is modified by routing or patching. To help with situations like this, the ITU made an additional, very logical recommendation called ITU-R BT.1377, which suggests that video and audio equipment be labeled to indicate processing delay - if it is variable delay, indicate the range - and that any delays are noted in milliseconds to avoid discrepancies due to differing frame rates.

One very common device to watch out for is the video frame synchronizer. Due to its very nature, a frame synchronizer typically produces between one and two variable frames of delay. Getting video and audio from a remote production to the final point of emission (i.e., the local television station) typically requires the signals pass through multiple frame synchronizers

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. If the audio is not suitably delayed to match, the results will be predictably ugly. Suitable audio delays are available, and some will even track the variable video delay and guarantee perfect sync.

Another group of devices to watch out for are digital video effects (DVE) systems, which can add many, many frames of delay. Since the delay of a DVE is generally a fixed value, a commonly available fixed value audio delay can also be used. Keeping these systems permanently inline, if possible, will also help to reduce variable A/V sync.


With the in-plant situations described above it is easy to measure A/V sync with an oscilloscope and a known good reference signal, such as the standard "beep flash," and adjust the system for minimal offset. What about transmitting the signals to consumers?

In NTSC, if the audio and video are synchronized at the input terminals of the transmitter, the program will be reproduced properly when received. ATSC is not so simple. The audio and video signals are encoded separately and then multiplexed together into a transport stream that is sent off to the modulator and RF sections of the transmitter. The encoding process takes time and it is different for audio and video. The multiplexer must know this timing exactly so that it can generate the proper Presentation Time Stamp (PTS) values, which are used by the decoder to - you guessed it - present the audio and video in sync. If the ATSC encoder is self-contained, meaning the Dolby Digital (AC-3) and MPEG video encoders are built-in, A/V sync calibration is very simple and has likely been preset. If the Dolby Digital (AC-3) encoder is external, as is presently the case if you wish to transmit 5.1 channel audio, then the situation is slightly more complex.


In either case, the most accurate way to make the measurement is to analyze the transport stream. I know it is tempting to use a consumer (or professional) receiver and look at the audio and video outputs to check sync. Unfortunately, this is very unreliable and probably caused much of the initial A/V sync trouble in DTV.

Interra Digital Video Technology (information technologies pune india songs at has a software application called SyncCheck that will take a transport stream, demultiplex the audio and video streams, decode them and display them as shown in Fig. 1.

The decoded audio waveform is shown for selected channel (all 5.1 are available), and an actual frame-by-frame display of the video can also be seen. As SyncCheck is decoding the transport stream using reference software decoders, very precise measurement is possible without nonstandard or incorrect DTV decoders giving false indications of A/V sync.

You might be wondering how to get the transport stream into SyncCheck, and I was wondering the same thing. It turns out that a relatively inexpensive (<$400) HDTV receiver is available on a PCI card and it turns a PC into a DTV receiver. Importantly, the card and its included software allow the received transport stream to be stored to the computer's hard drive. SyncCheck can import this transport stream, demultiplex it, decode it and display it by simply supplying the "beep flash" test signal to your encoder (in sync of course), then analyzing the transport stream and making any required adjustments. This is a very cost-effective way to ensure that your station's HDTV emission encoder and multiplexer is set correctly.

Once you have aligned the transport stream for perfect A/V sync, you can then look at the outputs of DTV receivers and know with confidence that the signal you are sending is correct.

Attached Images
Geo's sound post corner-tn_t.1853_i.01_10-23-02-fig1.jpg 
Old 20th April 2009
Dolby Metadata settings.

The attached picture is a table of possible metadata setting as a starting point for Dolby Metadata. the document is the DOLBY METADATA GUIDE.


Attached Thumbnails
Geo's sound post corner-metadta-settting.jpg  
Attached Files
File Type: pdf 18_Metadata.Guide.pdf (130.3 KB, 816 views)
Old 5th May 2009
Stereo downmixes (or fold-downs)

Left total/Right total (Lt/Rt)
Lt/Rt is a downmix suitable for decoding with a Dolby
Lt = L + -3dBC - -3dB(Ls + Rs)
Rt = R + -3dBC + -3dB(Ls + Rs)
(Ls,Rs 90-degree phase shifted)

Left only/Right only (Lo/Ro)
Lo/Ro is a downmix suitable for stereo playback
Lo = L + -3dBC + attLs
Ro = R + -3dBC + attRs
(att=-3dB or -6dB or -9dB or 0)

5.1 downmix

5.1 to LR Ls to L drop 6 db (.5) , Rs to R drop 6 db (.5), C to L & R drop 3db (.707) , lose the sub.
Lt = FL + s*SL + c*C
Rt = FR + s*SR + c*C
where s (=surround mix) is usually something between 0.5 and 1
and c (=center mix) is usually 0.7. This would be a "normal" stereo downmix.
Lt = FL + s*(SL+SR) + c*C
Rt = FR - s*(SL+SR) + c*C // 180° phase shift SL+SR
with s=0.5 and c=0.7.

Old 30th May 2009
Gear Maniac
wyndrock's Avatar
Thumbs up Thank you so much Georgia!


Pardon my slipping into the conversation..

I'm in school, starting my second year. I'll be following an "editing/sound" path through my BFA program. I've mixed audio for music, in local studios for several years, on Logic Pro systems, decided to cross into the visual arts, just recently, at 38 years of age. I honestly thought that I was a smart person about audio, less than one year ago. I thought I knew it all. I was recently sat on my butt, behind the tool-shed and shown that I knew absolutely nothing. It was an awakening blow to my ego! ...I no longer have one, by the way.

I am floored, daily, by the differences in film/tv sound and audio for music. These examples, above, are completely overwhelming for me at this point in my journey. I feel almost withered under the bombardment of all of this information. Also, it is more than I have capacity to understand at this point in my journey. Most of the basic measurements and standards that you reference here, we haven't even touched on yet, in our program. This thread is a scary place for one that is new to film and television sound. But, it is exciting, because I know that eventually, I'll be rocking and rolling along with you guys in the future.

I'm very interested in where you went to school, and how, at such a young age, you are so informed of all of this technical information. Am I eventually going to learn all of this, before I leave school, or will this be the education that happens after graduation, that I hear so much about? Kind of like "Glad you went to film school, son, now let me teach you how we make films". I imagine this will be happening to me once I get a real job. I will be seeking work on the east coast, most likely in NC, once I graduate.

Georgia, what I really want to say here is; I don't know your age or your path/background, but you are everything I wish I could be. Totally impressed. I feel like a tyke, looking up at mama using her hula hoop in the back yard, not understanding quite how it works, but knowing that I want to try it.

I look forward to reading your further posts in other areas on this board. Thank you for your contribution to my education. May you receive all the best in life.

Roger Harless, Jr.
[email protected]
[email protected]
Old 30th May 2009
LOL... young age! I love you man, marry me!
I just turned 53 this May.

I just blame everything on my parents.... I got a great set of genes. Both on the Art and the technology side. I've been doing this as a third career in my life for about 15 years now, give or take. I'm completely self taught in this field, but I have an BS in computer science and an MBA...not they help much in this business...

But, I can say without a doubt, my experience and knowledge comes from studying and working with many many people who know way more than I do. The guys on this forum are really the stars here. I just have a nack for the technical and the art and fond a nitch in production and post-production. Just take it one step and a time and your do fine!

Old 30th May 2009
tri-level bi-levle sync

SD and HD are 2 different standards. They require two different types of sync.

SD bi-sync and HD tri-sync are different signals.

SD Bi-sync supports computer video, composite video, s-video, and component video, by using 2 voltage levels ( high and low )... systems using bi-syn are triggered by the differential voltage at the leading ( negative) edge of the signal. Bi-sync basically runs at 0 votage ( black) then drops to a negative voltage and then back to zero ( 2 states or BI-sync)

Tri-sync provides a more exacting sync for the 3 component signals. HD has sync info on all three channels ( Y, Pb, and Pr). Tri sync starts at 0 volts ( video black ) then goes negitive, then a positive voltage, then back to 0 volts ( 3 states, Tri-Sync). This also fixed the introduction of a voltage differential by bi-sync signals into the actual video signal. the tri-sync signal is triggered by the on the negative transition, then on the positive transition. thus 0,-300mv,+300mv,0 3 sync points... This all happens in the Horizontal blanking interval. There is something called the White reference level that is set at 700mv. the 700mv drops to 0 at the start of the blanking period. the signal drops to -300mv, the to +300mv, then to 0 the back to reference white 700mv. This is the HD Analog horizontal timing data where HD tri-level sync is used. ( I don't remember what the white level is for bi-sync.. sorry pulling most of this out of my head... ) There are also a couple of issues revolving around the Progressive vs interlaced HD frame timing that comes into play here.

bi-sync is on the Y-signal on SD video. tri-sync is on all 3 channels for HD. ( check out SMPTE 274M ad SMPTE 296M )

so the two are incompatable for sync .. there are a lot of options out there for devices that support both bi-sync and tri-sync.

the sync IO accepts SD BiSync or "traditional" video black, the HDSync IO accepts Tri-sync signals. As far as gear it all depends on what you are syncing... if its protools to an HD video deck and you are laying back audio, you can use a tri-sync box that offers bi-sync. The bisync can be routed to the protools rig and the tri-sync can be used for the HD deck. If you are dealing with laying back an HD video signal from a video editing system with an AJA Kona3 card to an HD video deck directly connected, you don't need
a tri-sync gen, because the AJA kone3 offers this to the deck via the video signal (read the Kona3 card for details, i don't remember exactly how they do it, but they do) if you are using a euphonix system 5 HDMI IO and a HD video deck you need a tri-sync generator. So it all depends on what you are trying to sync.

I hope this helps a bit with some of the tech background.
Old 21st June 2009
Audio Cable Design and other selection information.

here's some research on how electricity really works... things like the skin effect, and fun stuff like that. This battle over cable designs within the audio realm makes me laugh out loud sometimes...

Skindepth, Litz wire, Braided conductors, and resistance
Transmission Line Theory

Here's some more trivia for your sunday reading....

What Makes a Good Audio Cable?
Criteria for what supposedly made one cable perform better or worse than another is remarkably inconsistent. One manufacturer's claims countered and negated the claims made by a different manufacturer. None of the manufacturers offer documented, measurable evidence that it was producing a superior cable. Instead, we find claims of allegedly superior components or materials used in cable construction. For example, a few leading manufacturers claimed that the most important factor for a cable was low capacitance, using the justification that cable capacitance shunts upper frequencies to ground. In order to lower the capacitance, these companies increased conductor spacing to simultaneously achieve a goal of increased inductance. This approach had drastic side effects, however. Merely decreasing capacitance without taking other realities of signal transmission into consideration increased the noise pickup and introduced a blocking filter. Both of these effects would obviously degrade sonic performance rather than improving it.

Another cable manufacturer advertised that its cable "employs two polymer shafts to dampen conductor resistance", but offered no evidence to prove it. Still another audiophile company claimed that because its cable was flat, "with no twist, it has no inductance". In general, inductance can indeed be reduced by making conductors larger or bringing them closer together. However, physics shows that, in reality, no cable can be built without some level of inductance, so this claim is without scientific merit.

Cylindrical Cable Conductors and Skin Effect
Most of the popular loudspeaker and musical instrument cables on the market employ cylindrical (a.k.a. round-diameter) cables as conductors. Unfortunately, cylindrical cable designs have a number of serious drawbacks, including current bunching, skin effect phenomenon, and frequency effects that lower the performance of the cable.

It's a common misconception to think about electrical transmission in cables in terms of direct current (DC) alone. Even experienced electrical engineers frequently ignore the ramifications of frequency on cable performance. In the case of DC, current is indeed uniformly distributed across the entire cross-section of the wire conductor, and the resistance is a simple function of the cross-sectional area. Adding the frequency of an electrical signal to the equation complicates the situation, however. As frequency increases, the resistance of a conductor also increases due to skin effect.

Skin effect describes a condition in which, due to the magnetic fields produced by current following through a conductor, the current tends to concentrate near the conductor surface. As the frequency increases, more of the current is concentrated closer to the surface. This effectively decreases the cross-section through which the current flows, and therefore increases the effective resistance. The current can be assumed to concentrate in an annulus at the wire surface at a thickness equal to the skin depth. For copper wire the skin depth vs. frequency is as follows:

60 Hz = 8.5 mm, 1kHz =2.09 mm, 10 kHz =0.66 mm, 100 kHz =0.21 mm.

Note that the skin depth becomes very small as the frequency increases. Consequently, the center area of the wire is to a large extent bypassed by the signal as the frequency increases. In other words, most of the conductor material effectively goes to waste since little of it is used to transmit the signal. The result is a loss of cable performance that can be measured as well as heard.

Current Bunching
Current bunching (also called proximity effect) occurs in the majority of cables on the market that follow the conventional cylindrical two-conductor design (i.e., two cylindrical conductors placed side-by-side and separated by a dielectric).

When a pair of these cylindrical conductors supplies current to a load, the return current (flowing away from the load) tends to flow as closely as possible to the supply current (flowing toward the load). As the frequency increases, the return current decreases its distance from the supply current in an attempt to minimize the loop area. Current flow will therefore not be uniform at high frequencies, but will tend to bunch-in. The current bunching phenomenon causes the resistance of the wires to increase as frequency increases, since less and less of the wire is being used to transmit current. The resistance of the wire is related to its cross-sectional area, and as the frequency increases, the effective cross-sectional area of the wires decreases. In order to convey the widest frequency audio signal to a loudspeaker, you want to use as much of the conductor cross-section as possible, so excessive current bunching is extremely inefficient.

Disadvantages of Rectangular Conductors
As a means of bypassing the skin effect and current bunching problems associated with cylindrical conductor designs, some cable manufacturers have developed rectangular conductors as an alternative. These designs typically use a one-piece, solid core conductor. Computer simulation showing the magnitude (volts/meter) of the electric field between two solid rectangular conductors. The conductors have a cross section area equivalent to a 10 gauge conductor. The spacing between the two conductors is 2mm with a voltage of +1 volt applied to the top conductor and -1 volt applied to the bottom conductor.
Computer simulation showing the magnitude (volts/meter) of the electric field between two hollow oval conductors. The conductors have a cross section area equivalent to a 10 gage conductor. The spacing between the two conductors is 2mm with a voltage of +1 volt applied to the top conductor and -1 volt applied to the bottom conductor.

A solid rectangular conductor of this type is undesirable because the sharp corners produce high electric field values that over time can break down the dielectric, causing a failure of the cable. In general, cables with solid conductors are prone to shape distortions and kinking due to their poor flexibility. This becomes an especially important issue with rectangular cable designs. The sharp corners from rectangular conductors tend to chafe the cable dielectric if the cable is repeatedly flexed or put under stress, and this chafing can lead to a short that could conceivably damage your loudspeakers.

Characteristic Impedance Complexity
Another parameter that is critical in cable design is characteristic impedance. But because of its complexity, this important factor is often misunderstood.

The characteristic impedance of a cable is given by Z = [(R + jwL)/(G + jwC)]1/2 where R is the series resistance, L is the series inductance, G is the shunt conductance, C is the shunt capacitance, and w is the angular frequency (w = 2pief).

Note that this is not a simple number for a cable, but one which changes with frequency. It is also important to note that R, L, G, and C also change with frequency, making the impedance of a cable even more frequency dependent.
milli-ohm/loop 100 ft . Z is a complex number, and it is common practice in the cable industry to simplify the situation by assuming a loss less transmission line and, in turn, assuming that R and G are zero. While this may be a valid approximation at high frequencies, it is not valid at low audio frequencies if you plan to construct an accurate model of a cable.

For example, stating that a speaker cable has a constant, characteristic impedance of 10 ohms across the entire frequency range of 20 to 20,000 Hz is a drastic oversimplification that, in the end, is simply untrue. The same type of statement is also inaccurate when applied to loudspeakers, as the table below shows. A speaker only has a constant impedance of 8 ohms at a single fixed frequency. To state otherwise is to ignore the complexity of impedance changes as signal frequency changes.

Frequency Blurring
To minimize frequency blurring, it is important that the speaker cable parameters do not change with frequency. Ideally, the resistance and inductance would remain constant as the frequency of the signal changes.

The faintest sound wave a normal human ear can hear is 10(-12) Wm(-2). At the other extreme of the spectrum, the threshold of pain is 1 Wm(-2). This is a very impressive auditory range. The ear, together with the brain, constantly performs amazing feats of sound processing that our fastest and most powerful computers cannot even approach.

As long ago as 1935 Wilska 2 succeeded in measuring the magnitude of movement of the eardrum at the threshold of audio sensitivity across various frequencies. At 3,000 Hz, it takes a minimal amount of eardrum displacement (somewhat less than 10-9 cm or about 0.01 times the diameter of an atom of hydrogen) to produce a minimal perceptible sound. This is an amazingly small number! The extremely small amount of acoustic pressure necessary to produce the threshold sensation of sound brings up an interesting question. Does the limiting factor in hearing minimal level sounds lie in the anatomy and physiology of hearing or in the physical properties of air as a transmitting medium? We know that air molecules are in constant random motion, a motion related to temperature. This phenomenon is known as Brownian movement and produces a spectrum of thermal-acoustic noise.

In 1933, Sivian and White3 experimentally evaluated the pressure magnitudes of these thermal sounds between 1kHz and 6 kHz. They observed that throughout the measured spectrum the root-mean-square thermal noise pressure was about 86 decibels below one dyne per square centimeter. The minimum root-mean-square pressure that can produce audible sensation between 1 kHz and 6 kHz in a human being with average hearing is about 76 decibels below one dyne per square centimeter, but in some people with exceptionally acute hearing may approach 85 decibels. These figures indicate that the acuity of persons possessing a high sensitivity of hearing closely approaches the thermal noise level, and a particularly good auditory system actually does approach this level. Furthermore, it is not likely that animals possess greater acuity of hearing in this spectrum, as their hearing would also be limited by thermal noise. What this means is that the human audio system is extremely sensitive.

1 Henry W. Ott, Noise Reduction Techniques in Electronics System (New York, NY John Wiley and Sons, 1988, p. 150)

2 Wilska, A.: Eine methode zur Bestimmung der Horschwellenamplituden des Trommelfells bei verschiedenen Frequenzen, Skandinav. Arch. Physiol., 72:161, 1935.

3 Sivian, L.J., and White, S.D.: On minimum audible sound fields, J. Acous. Soc. Am., 4:288, 1933
Old 21st June 2009
More on cable selection and design


How big should the conductors be?
The required size (or gauge) of the conductors depends on three factors: (1) the load impedance; (2) the length of cable required; and (3) the amount of power loss that can be tolerated. Each of these involves relationships between voltage (volts), resistance (ohms), current (amperes) and power (watts). These relationships are defined with Ohm's Law. The job of a speaker cable is to move a substantial amount of electrical current from the output of a power amplifier to a speaker system. Current flow is measure in amperes. Unlike instrument and microphone cables, which typically carry currents of only a few milliamperes (thousandths of an ampere), the current required to drive a speaker is much higher; for instance, an 8-ohm speaker driven with a 100-watt amplifier will pull about 3-1/2 amperes of current. By comparison, a 600-ohm input driven by a line-level output only pulls about 2 milliamps. The amplifier's output voltage, divided by the load impedance (in ohms), determines the amount of current "pulled" by the load. Resistance limits current flow, and decreasing it increases current flow. If the amplifier's output voltage remains constant, it will deliver twice as much current to an 8-ohm load as it will to a 16-ohm load, and four times as much to a 4-ohm load. Halving the load impedance doubles the load current. For instance, two 8-ohm speakers in parallel will draw twice the current of one speaker because the parallel connection reduces the load impedance to 4 ohms.
(For simplicity's sake we are using the terms resistance and impedance interchangeably; in practice, a speaker whose nominal impedance is 8 ohms may have a voice coil DC resistance of about 5 ohms and an AC impedance curve that ranges from 5 ohms to 100 ohms, depending on the frequency, type of enclosure, and the acoustical loading of its environment.)

How does current draw affect the conductor requirements of the speaker cable?
A simple fact to remember: Current needs copper, voltage needs insulation. To make an analogy, if electrons were water, voltage would be the "pressure" in the system, while current would be the amount of water flowing. You have water pressure even with the faucet closed and no water flowing; similarly, you have voltage regardless of whether you have current flowing. Current flow is literally electrons moving between two points at differing electrical potentials, so the more electrons you need to move, the larger the conductors (our "electron pipe") must be. In the AWG (American Wire Gauge) system, conductor area doubles with each reduction of three in AWG; a 13 AWG conductor has twice the copper of a 16 AWG conductor, a 10 AWG twice the copper of a 13 AWG, and so on.

But power amp outputs are rated in watts. How are amperes related to watts?
Ohm's Law says that current (amperes) times voltage (volts) equals power (watts), so if the voltage is unchanged, the power is directly proportional to the current, which is determined by the impedance of the load. (This is why most power amplifiers will deliver approximately double their 8-ohm rated output when the load impedance is reduced to 4 ohms.) In short, a 4-ohm load should require conductors with twice the copper of an 8-ohm load, assuming the length of the run to the speaker is the same, while a 2-ohm load requires four times the copper of an 8-ohm load. Explaining this point leads to the following oft-asked question:

How long can a speaker cable be before it affects performance?
The ugly truth: Any length of speaker cable degrades performance and efficiency. Like the effects of shunt capacitance in instrument cables and series inductance in microphone cables, the signal degradation caused by speaker cabling is always present to some degree, and is worsened by increasing the length of the cable. The most obvious ill effect of speaker cables is the amount of amplifier power wasted.

Why do cables waste power?
Copper is a very good conductor of electricity, but it isn't perfect. It has a certain amount of resistance, determined primarily on its cross-sectional area (but also by its purity and temperature). This wiring resistance is "seen" by the amplifier output as part of the load; if a cable with a resistance of one ohm is connected to an 8-ohm speaker, the load seen by the amplifier is 9 ohms. Since increasing the load impedance decreases current flow, decreasing power delivery, we have lost some of the amplifier's power capability merely by adding the series resistance of the cable to the load. Furthermore, since the cable is seen as part of the load, part of the power which is delivered to the load is dissipated in the cable itself as heat. (This is the way electrical space-heaters work!) Since Ohm's Law allows us to calculate the current flow created by a given voltage across a given load impedance, it can also give us the voltage drop across the load, or part of the load, for a given current. This can be conveniently expressed as a percentage of the total power.

How can the power loss be minimized?
There are three ways to decrease the power lost in speaker cabling:
First, minimize the resistance of the cabling. Use larger conductors, avoid unnecessary connectors, and make sure that mechanical connections are clean and tight and solder joints are smooth and bright.

Second, minimize the length of the cabling. The resistance of the cable is proportional to its length, so less cable means less resistance to expend those watts. Place the power amplifier as close as practical to the speaker. (Chances are excellent that the signal loss in the line-level connection to the amplifier input will be negligible.) Don't use a 50-foot cable for a 20-foot run.

Third, maximize the load impedance. As the load impedance increases it becomes a larger percentage of the total load, which proportionately reduces the amount lost by wiring resistance. Avoid "daisy-chaining" speakers, because the parallel connection reduces the total load impedance, thus increasing the percentage lost. The ideal situation (for reasons beyond mere power loss is to run a separate pair of conductors to each speaker form the amplifier.

Is the actual performance of the amplifier degraded by long speaker cables?
There is a definite impact on the amplifier damping factor caused by cabling resistance/impedance. Damping, the ability of the amplifier to control the movement of the speaker, is especially noticeable in percussive low-frequency program material like kick drum, bass guitar and tympani. Clean, "tight" bass is a sign of good damping at work. Boomy, mushy bass is the result of poor damping; the speaker is being set into motion but the amplifier can't stop it fast enough to accurately track the waveform. Ultimately, poor damping can result in actual oscillation and speaker destruction.

Damping factor is expressed as the quotient of load impedance divided by the amplifier's actual source impedance. Ultra-low source impedances on the order of 40 milliohms (that's less than one-twentieth of an ohm) are common in modern direct-coupled solid-state amplifiers, so damping factors with an 8-ohm load are generally specified in the range of 100-200. However, those specifications are taken on a test bench, with a non-inductive dummy load attached directly to the output terminals. In the real world, the speaker sees the cabling resistance as part of the source impedance, increasing it. This lowers the damping factor drastically, even when considering only the DC resistance of the cable. If the reactive components that constitute the AC impedance of the cable are considered, the loss of damping is even greater.

Although tube amplifiers generally fall far short of sold-state types in damping performance, their sound can still be improved by the use of larger speaker cables. Damping even comes into play in the performance of mixing consoles with remote DC power supplies; reducing the length of the cable linking the power supply to the console can noticeably improve the low-frequency performance of the electronics.

What other cable problems affect performance?
The twin gremlins covered in "Understanding the Microphone Cable," namely series inductance and skin effect, are also factors in speaker cables. Series inductance and the resulting inductive reactance adds to the DC resistance, increasing the AC impedance of the cable. An inductor can be thought of as a resistor whose resistance increases as frequency increases. Thus, series inductance has a low-pass filter characteristic, progressively attenuating high frequencies. The inductance of a round conductor is largely independent of its diameter or gauge, and is not directly proportional to its length, either.
Skin effect is a phenomenon that causes current flow in a round conductor to be concentrated more to the surface of the conductor at higher frequencies, almost as if it were a hollow tube. This increases the apparent resistance of the conductor at high frequencies, and also brings significant phase shift.

Taken together, these ugly realities introduce various dynamic and time-related forms of signal distortion which are very difficult to quantify with simple sine-wave measurements. When complex waveforms have their harmonic structures altered, the sense of immediacy and realism is reduced. The ear/brain combination is incredibly sensitive to the effects of this type of phase distortion, but generally needs direct, A/B comparisons in real time to recognize them.

How can these problems be addressed?
The number of strange designs for speaker cable is amazing. Among them are coaxial, with two insulated spiral "shields" serving as conductors; quad, using two conductors for "positive" and two for "negative"; zip-cord with ultra-fine "rope lay" conductors and transparent jacket; multi-conductor, allegedly using large conductors for lows, medium conductors for mids, and tiny conductors for highs; 4 AWG welding cable; braided flat cable constructed of many individually insulated conductors; and many others. Most of these address the inductance question by using multiple conductors and the skin effect problem by keeping them relatively small. Many of these "esoteric" cables are extraordinarily expensive; all of them probably offer some improvement in performance over ordinary twisted-pair type cables, especially in critical monitoring applications and high-quality music systems. In most cases, the cost of such cable and its termination, combined with the extremely fragile construction common to them, severely limits their practical use, especially in portable situations. In short, they cost too much, they're too hard to work with, and they just aren't made for rough treatment. But, sonically, they all bear listening to with an open mind; the differences can be surprisingly apparent.

Is capacitance a problem in speaker cables?
The extremely low impedance nature of speaker circuits makes cable capacitance a very minor factor in overall performance. In the early days of solid state amplifiers, highly capacitive loads (such as large electrostatic speaker systems) caused blown output transistors and other problems, but so did heat, short circuits, highly inductive loads and underdesigned power supplies.

Because of this, the dielectric properties of the insulation used are nowhere near as critical as that used for high-impedance instrument cables. The most important consideration for insulation for speaker cables is probably heat resistance, especially because the physical size constraints imposed by popular connectors like the ubiquitous 1/4" phone plug severely limit the diameter of the cable. This requires insulation and jacketing to be thin, but tough, while withstanding the heat required to bring a relatively large amount of copper up to soldering temperature. Polyethylene tends to melt too easily, while thermoset materials like rubber and neoprene are expensive and unpredictable with regard to wall thickness PVC is cheap and can be mixed in a variety of ways to enhance its shrink-resistance and flexibility, making it a good choice for most applications. Some varieties of TPR (thermoplastic rubber) are also finding use.

Why don't speaker cables require shielding?
Actually, there are a few circumstances that may require the shielding of speaker cables. In areas with extreme strong radio frequency interference (RFI) problems, the speaker cables can act as antennae for unwanted signal reception which can enter the system through the output transistors. When circumstances require that speaker-level and microphone-level signals be in close proximity for long distances, such as cue feeds to recording studios, it is a good idea to use shielded speaker cabling (generally foil-shielded, twisted-pair or twisted-triple cable) as "insurance" against possible crosstalk form the cue system entering the microphone lines. In large installations, pulling the speaker cabling in metallic conduit provides excellent shielding from both RFI and EMI (electromagnetic interference). But, for the most part, the extremely low impedance and high level of speaker signals minimizes the significance of local interference.

Why can't I use a shielded instrument cable for hooking an amplifier to a speaker, assuming it has the right plugs?
You can, in desperation, use an instrument cable for hooking up an amplifier to a speaker. However, the small gauge (generally 20 AWG at most) center conductor offers substantial resistance to current flow, and in extreme circumstances could heat up until it melts its insulation and short-circuits to the shield, or melts and goes open-circuit, which can destroy some tube amplifiers. Long runs of coaxial-type cable will have large amounts of capacitance, possibly enough to upset the protection circuitry of some amplifiers, causing untimely shut-downs. And of course there is enormous power loss and damping degradation because of the high impedance of the cable.

¥ Ballou, Greg, ed., Handbook for Sound Engineers: The New Audio Cyclopedia, Howard W. Sams and Co., Indianapolis, 1987.
¥ Cable Shield Performance and Selection Guide, Belden Electronic Wire and Cable, 1983.
¥ Colloms, Martin, "Crystals: Linear and Large," Hi-Fi News and Record Review, November 1984.
¥ Cooke, Nelson M. and Herbert F. R. Adams, Basic Mathematics for Electronics, McGraw-Hill, Inc., New York, 1970.
¥ Davis, Gary and Ralph Jones, Sound Reinforcement Handbook, Hal Leonard Publishing Corp., Milwaukee, 1970.
¥ Electronic Wire and Cable Catalog E-100, American Insulated Wire Corp., 1984.
¥ Fause, Ken, "Shielding, Grounding and Safety," Recording Engineer/Producer, circa 1980.
¥ Ford, Hugh, "Audio Cables," Studio Sound, Novemer 1980.
¥ Guide to Wire and Cable Construction, American Insulated Wire Corp., 1981.
¥ Grundy, Albert, "Grounding and Shielding Revisited," dB, October 1980.
¥ Jung, Walt and Dick Marsh, "Pooge-2: A Mod Symphony for Your Hafler DH200 or Other Power Amplifiers," The Audio Amateur, 4/1981.
¥ Maynard, Harry, "Speaker Cables," Radio-Electronics, December 1978,
¥ Miller, Paul, "Audio Cable: The Neglected Component," dB, December 1978.
¥ Morgen, Bruce, "Shield The Cable!," Electronic Procucts, August 15, 1983.
¥ Morrison, Ralph, Grounding and Shielding Techniques in Instrumentation, John Wiley and Sons, New York, 1977.
¥ Ott, Henry W., Noise Reduciton in Electronic Systems, John Wiley and Sons, New York, 1976.
¥ Ruck, Bill, "Current Thoughts on Wire," The Audio Amateur, 4/82.

Old 21st June 2009
more on cable seletion...

This is some of ESP's comments that I find interesting and agree with from my past work in Communications, RF and engineering.


All well designed interconnects will sound the same. This is a contentious claim, but is regrettably true - regrettable for those who have paid vast sums of money for theirs, at least. I will now explain this claim more fully.

The range (and the associated claims) of interconnects is enormous. We have cables available that are directional - the signal passes with less intrusion, impedance or modification in one direction versus the other. I find this curious, since an audio signal is AC, which means that electrons simply rush back and forth in sympathy with the applied signal. A directional device is a semiconductor, and will act as a rectifier, so if these claims are even a tiny bit correct, I certainly don't want any of them between my preamp and amp, because I don't want my audio rectified by a directional cable.

Oxygen free copper (or OFC) supposedly means that there is no oxygen and therefore no copper oxide (which is a rectifier) in the cable, forming a myriad of micro-diodes that affect sound quality. The use of OFC cable is therefore supposed to improve the sound.

Try as I might (and many others before me), I have never been able to measure any distortion in any wire or cable. Even a length of solder (an alloy of tin and lead) introduces no distortion, despite the resin flux in the centre (and I do realise that this has nothing to do with anything - I just thought I'd include it :-). How about fencing wire - no, no distortion there either. The concept of degradation caused by micro-diodes in metallic contacts has been bandied about for years, without a shred of evidence to support the claim that it is audible.

At most, a signal lead will have to carry a peak current of perhaps 200uA with a voltage of maybe 2V or so. With any lead, this current, combined with the lead's resistance, will never allow enough signal difference between conductors to allow the copper oxide rectifiers (assuming they exist at all) to conduct, so rectification cannot (and does not) happen.

What about frequency response? I have equipment that happily goes to several MHz, and at low power, no appreciable attenuation can be measured. Again, characteristic impedance has rated a mention, and just as with speaker cables it is utterly unimportant at audio frequencies. Preamps normally have a very low (typically about 100 Ohms) output impedance, and power amps will normally have an input impedance of 10k Ohms or more. Any cable is therefore mismatched, since it is not sensible (nor is it desirable) to match the impedance of the preamp, cable and power amp for audio frequencies.

Note: There is one application for interconnects where the sound can change radically. This is when connecting between a turntable and associated phono cartridge and your preamp. Use of the lowest possible capacitance you can find is very important, because the inductance of the cartridge coupled with the capacitance of the cable can cause a resonant circuit within the audio band.

Should you end up with just the right (or wrong) capacitance, you may find that an otherwise respected cartridge sounds dreadful, with grossly accentuated high frequency performance. The only way to minimise this is to ensure that the interconnects have very low capacitance, and they must be shielded to prevent hum and noise from being picked up.

At radio frequencies, Litz wire is often used to eliminate the skin effect. This occurs because of the tendency for RF to try to escape from the wire, so it concentrates on the outside (or skin) of the wire. The effect actually occurs as soon as the frequency is above DC, but becomes noticeable only at higher frequencies. Litz wire will not affect your hi-fi, unless you can hear signals above 100kHz or so (assuming of course that you can find music with harmonics that go that high, and a recording medium that will deliver them to you). Even then, the difference will be minimal.

In areas where there is significant electromagnetic pollution (interference), the use of esoteric cables may have an effect, since they will (if carefully designed) provide excellent shielding at very high radio frequencies. This does not affect the audio per se, but prevents unwanted signals from getting into the inputs or outputs of amps and preamps.

Cable capacitance can have a dramatic effect on sound quality, and more so if you have long interconnects. Generally speaking, most preamps will have no problem with small amounts of capacitance (less than 1nF is desirable and achievable). With high output impedance equipment (such as valve preamps), cable capacitance becomes more of an issue.

For example, 1nF of cable capacitance with a preamp with an output impedance of 1k will be -3dB at 160kHz, which should be acceptable to most. Should the preamp have an output impedance of 10k, the -3dB frequency is now only 16kHz - this is unacceptable.

I tested a couple of cable samples, and (normalised to a 1 metre length) this is what I found

Single Core Twin - One Lead Twin- Both Leads Twin - Between Leads
Capacitance 77pF 191pF 377pF 92pF
Inductance 0.7uH 1.2uH 0.6uH NT
Resistance 0.12 Ohm 0.38 Ohm 0.25 Ohm NT

NT - Not Tested

These cables are representative of medium quality general purpose shielded (co-axial) cables, of the type that you might use for making interconnects. The resistance and inductance may be considered negligible at audio frequencies, leaving capacitance as the dominant influence. The single core cable is obviously better in this respect, with only 77pF per metre. Even with a 10k output impedance, this will be 3dB down at 207kHz for a 1 metre length.

Even the highest inductance I measured (1.2uH) will introduce an additional 0.75 Ohm impedance at 100kHz - this may be completely ignored, as it is insignificant.

The only other thing that is important is that the cables are properly terminated so they don't become noisy, and that the shield is of good quality and provides complete protection from external interfering signals. Terminations will normally be either soldered or crimped, and either is fine as long as it is well made. For the constructor, soldering is usually better, since proper crimping tools are expensive.

The use of silver wire is a complete waste, since the only benefit of silver is its lower resistance. Since this will make a few micro-ohms difference for a typical 1m length, the difference in signal amplitude is immeasurably small with typical pre and power amp impedances. On the down side, silver tarnishes easily (especially in areas where there is hydrogen sulphide pollution in the atmosphere), and this can become an insulator if thick enough. I have heard of some audiophiles who don't like the sound of silver wire, and others who claim that solid conductors sound better than stranded. Make of this what you will :-D

The use of gold plated connectors is common, and provides one significant benefit - gold does not tarnish readily, and the connections are less likely to become noisy. Gold is also a better conductor that the nickel plating normally used on "standard" interconnects. The difference is negligible in sonic terms.

There is no reason at all to pay exorbitant amounts of hard earned cash for the "Audiophile" interconnects. These manufacturers are ripping people off, making outlandish claims as to how much better these cables will make your system sound - rubbish! Buy some good quality audio coaxial cable and connectors from your local electronics parts retailer, and make your own interconnects. Not only will you save a bundle, but they can be made to the exact length you want.

Using the cheap shielded figure-8 cable (which generally has terrible shields) is not recommended, because crosstalk is noticeably increased, especially at high frequencies. That notwithstanding, for a signal from an FM tuner even these cheapies will be fine (provided they manage to stay together - most of them fall to bits when used more than a few times), since the crosstalk in the tuner is already worse than the cable. With typical preamp and tuner combinations, you might get some interference using these cheap and nasty interconnects, but the frequency response exceeds anything that we can hear, and distortion is not measurable.

hope this stuff helps debunk the snake oil salespeople!

here's some good selection info:

Old 1st September 2009
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appreciate all the info.take care
Old 15th October 2009
you're welcome...

Old 21st November 2009
More on encoding for DVD and AC3 part 1

Some more info on DVD encoding ( AC3 ) i've dug up.


The primary references for the information contained in this guide are two guides on Dolby's web site. The first is Standards and Practices for Authoring Dolby Digital and Dolby E Bitstreams, which has the best information on Dynamic Range Compression. The other is Dolby Digital Professional Encoding Guidelines which gives an excellent explanation of the dialogue Normalization parameter. You will need Adobe Acrobat Reader to view these .pdf documents.

Philosophy of Dolby Digital

Dolby Labs has been doing high-quality audio with cutting-edge techniques for a long time, using their past experience as a guide. As such, there is often confusion about their methods and philosophy to those of us who are not privy to that information. Of prime example is the current problem: Why is Dolby Digital so much quieter compared to my original sound?

Most audio destined for DVDs is audio originally recorded for use in the movie theater. The movie industry has a huge advantage when producing audio for the theater -- the theater has large speakers and amplifiers, and a quiet, near-ideal listening environment. Huge dynamic ranges are possible, where the slightest whisper of dialogue is audible, yet gunshots and explosions can be earth-shattering. Dolby's dilemma was: "How do we bring this audio, with its huge dynamic range, into the home?" This is a major problem -- most homes don't have the speakers and amplifiers necessary to shake the living room. Further, background noise in the home can easily drown out those subtleties in the soundtrack.

Dolby's answer is to allow the decoder to modify the sound to compensate for these problems. Low-volume sounds are boosted automatically so they can be heard, whereas high-volume sounds are quieted down so that speakers aren't blown and other persons in the home are not disturbed. Further, Dolby Digital allows for different program material to be equalized, so that volume does not have to be adjusted when switching between inherently quiet programs and inherently loud programs.

Decoder Specifics

The methods I'm about to present here for encoding Dolby Digital are generic and do not apply specifically to any one encoder. All Dolby-certified encoders (and some non-certified ones) will have the appropriate parameters available to follow this procedure. I have personally tested the Sonic Foundry 5.1 Plug-In Pack for ACID Pro, as well as Sonic Foundry Soft Encode. These methods should also work for BeSweet, Vegas Video + DVD, Scenarist, and other software-based encoders.

Basic Parameters

Every Dolby Digital encoder has some basic parameters that need to be set.

The first is the channel combination, presented as (Number of front channels)/(Number of rear channels), with an optional ".1" added to represent a low-frequency-effects (LFE) channel if present. i.e. 2/0 represents normal left and right stereo sound. 3/2.1 represents a standard "5.1" setup, of Front Left, Front Right, Front Center, Rear Left, Rear Right, and LFE. This parameter should obviously be set to the number of channels of program material you will be encoding.

The other major basic parameter is the bitrate. Obviously, higher bitrates allow for less compression. Typical bitrates used are 192 kbps for 2/0 program material, and 448 kbps for 5.1 program material.

Referencing Volume to a Known Level - Dialogue Normalization

To meet the Dolby Digital requirement that different programs should have approximately the same listening level (thus the consumer does not have to adjust volume level between programs), Dolby Digital incorporates a parameter called dialogue Normalization. This metadata parameter tells the decoder how far away from the reference level the average sound pressure level of the material's dialogue is.

The movie industry masters their soundtracks in a specific way. The maximum rated sound level (where all amplifiers are putting out their rated power) is 0 dB. Sounds below that level are rated in terms of how many decibels (dB) they are down from that maximum level. As such, these values are negative. The movie industry typically masters the "normal" listening level of dialogue (where people are speaking in a normal voice) at -31 dBFS. In other words, a speaking voice is at an average of -31 dB when referenced to the 0 dB maximum sound level, hence the term decibels of full scale (dBFS).

Since movie content is the largest class of programs to go on DVD, Dolby chose -31 dBFS as the reference level for audio on DVD, where 0 dB represents the maximum encodable digital sound level (full scale).

The dialogue normalization parameter needs to be set to the LAeq level of your program material's dialogue. LAeq stands for the long-term A-weighted sound pressure level. Loosely, this is the average volume level of your source material's dialogue. Us lowly consumers really don't have a tool that can measure this parameter, but we can get close. Sonic Foundry's Sound Forge has a "Normalization" feature that can measure the RMS level of a .wav file (or the portion thereof containing dialogue). CoolEdit may also have a feature like this. To use it in Sound Forge, open your .wav file containing the movie audio. Select a section containing dialogue (no sound effects or music). Go to "Process"/"Normalize". Select the "Average RMS Power (Loudness)" radio button. Then click the "Scan Levels" button. The displayed "RMS" level is very close (within 1-2 dB) to the LAeq level.

That RMS level is the number that the dialogue normalization parameter should be set to. In other words, if the RMS level in Sound Forge shows as -17.6 dB, set the dialogue normalization parameter in your Dolby Digital encoder to -18 dBFS.

The decoder will perform an attenuation of (31 + dialnorm) dB to the program material when played back. So, in this case, the decoder will attenuate by (31 + -18) = 13 dB. This will bring the average sound level of the material to (-17.6 - 13) = -30.6 dBFS. The program is now played back at approximately -31 dBFS, the reference level.

-31 dBFS is a lower average volume level than what is typical from other sources. It will be noticeable that you will have to turn the volume up on your system when playing a DVD versus playing broadcast, tape, or other non-Dolby Digital program material.

Allowing Comfortable Listening - Dynamic Range Compression

Meeting the other end of the requirement, that the consumer should be able to listen to quiet and loud sections of the program without having to adjust volume levels, requires a decrease in the dynamic range of the program. A movie, with whispers at -50 dbFS and explosions at -5 dBFS can't be comfortably listened to in the average home. The whisper is drowned out by extraneous background noise, and if the explosion is played at a tolerable level that doesn't wake up the neighbors, regular dialogue at -31 dBFS requires straining to adequately hear.

Dolby solves this problem by compressing the dynamic range of the program material. Quiet sounds are automatically boosted in volume so that they're audible, and loud sounds are automatically cut down in volume to tolerable levels.

There are several dynamic range compression profiles available that are custom tailored to the particular flavor of program material. However, all of them share the same basic features. All of the compression profiles can be thought of as an input-output "black box", where certain input volume levels are mapped to certain output volume levels. Observe this graph, which is a graph of one of the Dolby Digital compression profiles (Film Light).

The blue line is the "unity gain" line, also referred to as the "no compression" line. This line represents that the dynamic range compressor feature of the decoder is essentially turned off, and no boost or cut of the program material is done.

The purple line is the compression profile for "Film Light". It is divided into 5 different sections, as are the other Dolby Digital compression profiles:

Unity Gain = Volume neither boosted nor cut
Variable Boost = Beginning of increasing volume of soft sounds
Constant Boost = Increase volume by a fixed amount for very soft sounds
Early Cut = Beginning of attenuating volume for loud sounds
Cut = Very loud sounds almost clamped to a maximum volume level

The application of a compression profile like this allows the soft sounds to be heard while preventing speaker overdrive and disturbances by the loud sounds.

The Dolby Digital encoder offers 5 different compression profiles that can be specified depending on the nature of the program material being encoded. This graph illustrates all of the available profiles. The profiles range from no compression ("None"), to fairly light compression ("Music Light") all the way to extremely aggressive compression ("Speech").

For the exact dB numbers where each range of dynamic range compression is located on the graph, see the Dolby documents cited in the references at the beginning of this guide.

Many authors, when compressing audio to Dolby Digital, are turned off by the idea of dynamic range compression. You have this well-mixed audio with nice dynamic range and are then going to kill it by compressing that dynamic range. This is a valid concern, but should be answered by looking at what the listening environment is going to be. If you are authoring a DVD only for yourself, and you have a home theater room that can deliver theater-like sound, perhaps a compression profile of "None" is suitable for you. However, this profile may not sound good in a more mundane living room. Some experimentation may be in order to determine what compression profile will sound best for you. Most Hollywood DVDs use "Film Light" or "Film Standard".

The following point, however, cannot be stressed enough: In order for the Dynamic Range Compression to work as designed, the Dialogue Normalization parameter MUST be properly set first!

All of the dynamic range compression profiles assume that the average volume level of the program material's dialogue being fed to the dynamic range compressor is -31 dBFS. If that is not the case, boost or cut will be applied to the material when it shouldn't be!

A prime example is the situation where an average volume .wav file (with an LAeq of -16 dBFS, for example) is fed to Soft Encode using Soft Encode's default dialogue Normalization and Dynamic Range Compression settings. The default dialogue Normalization setting is -27 dB, and the default Dynamic Range Compression is set to "Film Standard". Because of the misadjusted dialnorm parameter, only (31 + -27) = 4 dB of attenuation is applied to the audio, so the average volume level is (-16 - 4) = -20 dBFS instead of the expected -31 dBFS. This places the audio on the DRC graph at the incorrect position, and now most of the audio is being played back in the Early Cut and Cut ranges. This causes the audio to sound flat and dull, with a possible audible "pumping" of the volume up and down as the decoder changes between Early Cut and Cut based on average volume level. Here is a representative graph. If dialnorm had been properly set to -16 dB, the audio would be centered at -31 dBFS, and would sound like it is supposed to.

Old 21st November 2009
DVD encoding of AC3 part 2

ine Mode and RF Mode

The Dolby Digital compressors have the ability to further alter the compression profile to compensate for different transport mediums. Most of the time, audio is transported between devices in "Line Mode", where a line-level is used. There is also "RF Mode", meant for broadcasting of Dolby Digital and devices that send audio via RF cables to a TV set. RF mode sound from the decoder uses a higher average volume level (-20 dBFS vice -31 dBFS) in order to correlate volume level well with other, non-Dolby broadcast audio, and also can use a more aggressive Dynamic Range Compression to prevent overmodulating the signal. There is an option in most Dolby Digital encoders to turn on that overmodulation limiter (in Soft Encode, it is labeled as "RF Overmodulation Protection"). Since we are primarily interested in authoring for DVD which will operate in Line Mode, we do NOT want to insert the additional Dynamic Range Compression that RF Overmodulation Protection will add. Therefore, for DVD authoring, the RF Overmodulation Protection option should be turned off.

Here are 6 graphs that will help in understanding what's happening to your mixes...

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Geo's sound post corner-gbmusic.gif  
Old 22nd November 2009
Film / Video post speed/frame rate issues...

Before you start a sound project find out the:
1. frame rate
2. project speed
3. post production life cycle
A. how the project was shot
B. how the project was captured
C. how the project was edited
D. how the project was exported for audio edit
E. format the client wants you to work

The first question to ask when dealing which projects: is the picture frame rate in sync with 48kHz. If so there will be no need to do a sample rate conversion or digitize via analog sources to change the sample rate of the incoming audio signal. Otherwise check out these various project paths....

1. Feature film – double system at 24fps and 48kHz audio recording for 24fps postproduction.
2. Film-based television providing sync dailies on DigiBeta (23.976 and 48kHz) for 23.976
3. Film or HD Production at 23.976 with single or double system audio recording for 23.976
4. Feature film and film-based television production at 24fps with hard disk recording at 48.048kHz for 23.976 postproduction.

1. Feature Film Double System

Most, if not all, feature film production intended for theatrical shoots film at 24fps while recording audio digitally at 48kHz. the film is now running at 23.976fps during the telecine process in order to create a known 2:3 pulldown cadence to the 29.97fps video rate. Once digitized into a 24p project, the frames are “stamped” as 24fps in order to play back in sync with audio captured directly via AES/EBU or Broadcast WAV files recorded at 48kHz. Because the audio was captured digitally – either synced to work clock or imported as 48kHz – it expects to be in sync with the picture as it was originally captured – 24fps. The native sample rate of a 24p project is 48kHz and all other rates are resolved to that during capture. When playing back at 48kHz, the audio plays back .1% faster creating a true 24fps playback from 23.976 sync sources. When capturing digitally at 48kHz, no samples are converted. It is a digital clone.

2. Film-Based Television with Sync Dailies
The transfer facility has already resolved the original shooting rate of 24fps to 23.976 and has sample-rate-converted the digital audio sources to be in sync in the digital source tapes. the audio must be sample rate converted when going from 24fps to 23.976 on the video. The path looks like this: Picture: 24 -> 23.976 to 29.97 video creating 2:3 pulldown Audio: 48kHz -> 47.952 slow down (.1%) sample corrected -> 48kHz to 29.97 video. If the editors are cutting in a 30i project (29.97 NTSC video), the audio sample rate is unchanged when capturing – it is a digital clone.

If it’s decided that postproduction will work in a 24p project, the digitized samples are slowed to bring everything back to a true 24fps = 48kHz environment.
In this case, the postproduction should be done in a 23.976 project type, since it assumes that the 48kHz audio sample rate is in sync with picture playing back at 23.976fps from the DigiBeta captured sources. It has the same result than that of a film-to-tape transfer to tape. But since there is no need to speed up to true 24fps in this project, audio samples remain untouched at 48Khz throughout the postproduction process, through the audio mix and back to the NTSC broadcast master. Using this project type for this workflow will only go through one sample rate conversion during the film to tape transfer.

3. Film or HD Production at 23.976
shooting rate is 23.976fps because of the audio consideration when down converting to NTSC. No one wanted to deal with a sample rate conversion in the audio when working in a fully digital environment. In a double system environment, the DAT or hard disc recorder records at 48kHz. So shooting at 23.976fps eliminates the need to do a sample rate conversion. The resulting NTSC down convert is now the same as in the previous example where 23.976 video with 2:3 pulldown is in a Digital tape with sync 48kHz audio.
If working double system, the DAT or BWF files from the hard disk recorder, the 48kHz recording will come straight in with no sample rate conversion or speed change to sync with the 23.976 picture.

4. Feature Film with 48.048kHz Audio Recording
audio workflow at 23.976 with the film running at 24fps. This workflow is only for picture capture frame rate of true 24fps and a NTSC postproduction workflow. DAT, and more common to this workflow, hard disk recorders, can record at 48.048 kHz – which is really just 48kHz with a .1% speed up as part of the capture.

Film/24p Settings
editing systems with 23.976 project types support a 48.048kHz BWF import workflow. If no sample rate conversion is chosen, the imported files are stamped as 48kHz, thus slowing them down by .1%; the same amount that the film is slowed down during the film to tape transfer. This way no sample rate conversion is performed, and a digital audio pipeline is maintained for the postproduction process.

Capture, Edit, Digital Cut
Capture: The project type determines the native capture rate of the project, either 23.976 or 24p. It also determines the native audio sample rate of that project that will not have a sample rate conversion or analog process involved when capturing, playing, or digital cut.

Edit: In the Film/24p settings you will see the “Edit Play Rate” as either 23.976 or 24. This control sets the play rate of the timeline. It does not affect any of the digital cut output settings. This control lets you set a default state of frame rate for outputs that are made directly to tape, such as a crash record.

Digital Cut: Here you can output the timeline as 23.976, 24, or 29.97. The important thing to remember is that this is the playback speed of the Avid timeline, not the source tape destination. The NTSC frame rate of 29.97 cannot be changed. What is changing is the frame rate of the picture within the NTSC signal.

1. 23.976. This creates a continuous 2:3 cadence from beginning to end of a sequence and is the expected frame rate of a broadcast NTSC master from 24 frame sources.

2. 24: This is used for feature film production to create a true “film projected” speed from an Avid timeline on NTSC video. It is also the output type to use when using picture reference in a Digidesign Pro Tools system using OMF media from a 24p project type. Note that this is not a continuous 2:3 cadence. Adjustments are made over 1000 frames with the pulldown cadence. No frames are dropped, just the field ordering with the 2:3 cadence.

3. 29.97: Timeline will play back 25% faster to create a 1:1 film frame to video frame relationship. This can be considered a 2:2:2:2 pulldown cadence. This
output is useful for animation workflow or low cost kinescope transfers where a 2:3 pulldown cannot be properly handled

Convert 60i to 24P
Use this option for standard interlaced NTSC shot at 1/60th sec shutter speed, where you wish to edit at 24P for the purpose of transfer to film or to author a 24P DVD. If this option is selected, all film effects (widescreen, grain, red boost) will be disabled. These effects can be added after editing.

Convert 3:2 Pulldown to 24P
Use this option for NTSC which was shot in 24P normal mode with a standard 3:2 pulldown, or with video that originated on 24 frames/sec film, where you wish to edit at 24P for the purpose of transfer to film or to author a 24P DVD. If this option is selected, all film effects (widescreen, grain, red boost) will be disabled. These effects can be added after editing.

Convert 2:3:3:2 pulldown to 24P
Use this option for NTSC video that was shot in 24P with a 2:3:3:2 pulldown, or 24P-NTSC archival material created with a 2:3:3:2 pulldown. Convert 2:3:3:2 Pulldown to 24P is the only option that works without recompression of the video data.

Output 23.976 (23.98 )
Use this option to output 23.976 frames/sec Quicktime with 48000 Hz audio, instead of 24.000 frames/sec Quicktime and 48048 Audio. This option works best with editing programs that can set the timeline to exactly 23.976 frames/sec. If this option is not used, then the Quicktime's playback rate is 24.000 fps and the audio playback rate is set to 48048 Hz to keep perfect sync, and the 24.000 frames/sec timeline must be set up for 48048 Hz audio.

So find out exactly what path the production team used and find out how i was edited and finally what speed/frame rate they want you to work in and to deliver to.
Old 22nd November 2009
Notes on shooting formats... FYI

720p is one of the HDTV formats. It means the image has 720 lines of "vertical resolution", i.e. 720 pixels from top to bottom. The p stands for progressive, which means that each frame is a single full-resolution image, unlike some formats that use interlacing.

720p has a horizontal resolution of 1280, so the entire image is 1280x720 pixels (approx. 920,000 pixels in total).

Frame Rate
720p can have any of five different frame rates (frames per second, or fps): 720p24, 720p25, 720p30, 720p50 or 720p60. Traditionally, PAL countries use 25fps (50 fields), NTSC countries use 30fps (60 fields). 24fps is the frame rate for film, making this an ideal format to use for film conversions.

When used with 50 or 60 fps, 720p has the best frame rate of the first-generation high-definition formats. This is one advantage it has over 1080i and 1080p. However, future versions of the 1080 formats are envisioned to use higher frame rates, which would give them the edge.

720p is generally compatible with most televisions and computer monitors. Although some image processing is usually required, this applies to other HDTV formats as well.
720p has initially been preferred by sports broadcasters, as it tends to work better with fast-moving images than 1080i and 1080p.

1080p is one of the HDTV formats. It means the image has 1080 lines of "vertical resolution", i.e. 1080 pixels from top to bottom. The p stands for progressive, which means that each frame is a single full-resolution image, unlike 1080i in which each frame consists of two interlaced fields.

1080p video usually has 1920 lines of horizontal resolution, making a total image size of 1920x1080 pixels (2,073,600 total). The aspect ratio is 16x9.

Frame Rate
1080p can be specified as 1080p24, 1080p25 or 1080p30 — the additional number refers to the number of frames per second (fps). Traditionally, PAL countries use 25fps, NTSC countries use 30fps. 24fps is the frame rate for film, making this an ideal format to use for film conversions.

In the future it is envisioned that 1080p50 and 1080p60 will become the production standards, combining the benefits of progressive scan with a higher number of total images. Currently, bandwidth considerations make these formats impractical, but this could change with more efficient codecs.

1080i is one of the HDTV formats. It means the image has 1080 lines of "vertical resolution", i.e. 1080 pixels from top to bottom. The i stands for interlaced (see below).

1080i video usually has 1920 lines of horizontal resolution, making a total image size of 1920x1080 pixels and an aspect ratio of 16x9.

There are some variations; for example, HDV has a resolution of 1440x1080 but maintains a widescreen aspect ratio by using rectangular pixels.

1080i is similar to 1080p, except that it uses the interlaced format rather than progressive. This means that each frame consists of two fields each showing only half the pixels. One field shows the odd lines, the other field shows the even lines.

Frame Rate
1080i can be specified as 1080i25 or 1080i30 — the additional number refers to the number of frames per second (fps). Traditionally, PAL countries use 25fps, NTSC countries use 30fps.

DTV-HDTV Comparison Chart

Digital television, or DTV, is the new industry standard for broadcasting picture and sound using digital signals, allowing for dramatic improvements in both picture and sound quality vs. conventional NTSC analog programming. DTV programming can be delivered in either of two basic formats: standard analog definition, (SDTV) and High Definition (HDTV).

480 is the number of lines. The "p" refers to progressive, a type of video scanning where all the lines that make up a video picture, or frame, are transmitted simultaneously. There are several progressive digital television formats.

720p Compatible

Displays 720p signals as 720p, without any conversion. 720 is the number of lines. The "p" refers to progressive, a type of video scanning where all the lines that make up a video picture, or frame, are transmitted simultaneously. There are several progressive digital television formats.

1080i Compatible

Displays 1080i signals as 1080i, without any conversion. 1080 is the number of lines. The "i" refers to interlaced, a type of video scanning where the odd- and even-numbered lines of a video picture, or frame, are transmitted consecutively as two separate interleaved fields. Analog NTSC video uses interlaced scanning, as do several digital television formats.

Additionally, 1080p is the shorthand name for a category of video modes. The number 1080 represents 1,080 lines of vertical resolution,[1] while the letter p stands for progressive scan or non-interlaced. 1080p is considered an HDTV video mode. The term usually assumes a widescreen aspect ratio of 16:9, implying a horizontal (display) resolution of 1920 dots across and a frame resolution of 1920 × 1080 or about 2.07 million pixels. The frame rate in hertz can be either implied by the context or specified after the letter p (such as 1080p30, meaning 30 frames per second).

DTV Format Details

HDTV is the highest form of digital television, delivering up to 1,080 interlaced scan lines. HDTV produces images that far surpass any you've ever seen in a home environment! SDTV, or Standard Definition, also represents a dramatic improvement over today's TV, with the added benefit of allowing stations to broadcast multiple programs within the same bandwidth as an HDTV signal.

1080i has a higher resolution and will have better quality but since it is interlaced video it will be more unstable.

The 720p has lower resolution but since it is progressive the image will be more stable and possible less distortion.

I think your preference will depend on how your eyes view the images – the 1080i is too jumpy or unstable for some people and they prefer the 720p resolution.
Progressive scan works in the same manner as your computer monitor. It writes one full frame of video from left to right across the screen every 1/60 of a second. Since the entire image is drawn at one time--as opposed to an interlaced image where the even lines are drawn first, followed by the odd lines--a progressively scanned video image looks more stable than an interlaced one. Progressive scan also introduces fewer motion artifacts, such as jagged diagonal lines and movement in fine detail, into the picture.

Many current progressive players fall back into a very watchable but very soft video mode when they aren't sure whether the source is film or not. In the worst case, the entire film will look excessively soft, which means they're getting exactly no benefit at all from their progressive player.

Just was poking around nd dug this stuff up. Thought it might be nice to share.
Old 22nd November 2009
Lives for gear
danijel's Avatar
Georgia, while you're in the outburst mode.......

Do you have any info on the Dolby DMU to share? I noticed there is NOTHING about it on the internet!
Old 22nd November 2009
Dolby DMU Stuff

Hmmm... lets see. The one I use is all AES Digital IO. Just place it in record chain plug in timecode and 48Khz black sync and Bobs' your uncle. the unit takes 6 channel discreet and records an AC3 digital stream for the 35mm digital audio and an LtRt for the optical track, to an Optical Disk drive. The DMU can operate at most frame rates, 24, 23.976, 29.97, and 30, it really doesn't care. Everything is counted in feet and frames on the DMU. You normally record to the DMU one reel at a time with tones in front of each reel. I also record voice with Project name, reel number, other data as necessary, this get tracked to the MO just before the tones and 2-pop. So don't froget to setup tones and 2-pops ( as well as tail pops ) for each reel.

Everything is real time except for the copy operation. Normally after the printmaster session, a copy of the Optical disk is made on location and you are done!...

more to follow!

Attached Thumbnails
Geo's sound post corner-dmu-back-panel.gif   Geo's sound post corner-img_0752.jpg   Geo's sound post corner-img_0750.jpg  
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