Geo's sound post corner

[georgia]

PAUL WHITE explains why a great reverb doesn't always make a mix sound better.



Most studio musicians appreciate the importance of reverb in recorded music, but a large number of demos are still spoiled either through over-use or inappropriate use of this crucial effect. Pop music is rarely recorded in a natural acoustic environment, unless you have access to one of the top studios with a really good live room, so what tends to happen is that sounds are recorded in a fairly dry room, then treated with artificial reverberation to make them sound natural. The same is true of electronically generated sounds that have no natural ambience of their own -- they need some added reverberation to make them sound believable. Problems arise, however, when the type and amount of reverberation are wrong.

SPACE: THE FINAL FRONTIER

Take any solo'd track and add reverb to it, and the chances are that it'll sound bigger (in a spatial sense) and more impressive. That's because a solo'd track usually has plenty of space between the sounds. But when the whole mix is playing, there's a danger that reverb can fill all the important spaces that let the detail within the recording shine through. By its very nature, reverb occurs after the sound that caused it, so the effect of too much reverb is to 'smear' percussive events, reducing the contrast between beats and the spaces between those beats. Though the average signal level of a reverb-processed sound may be higher than the untreated signal, the chances are that it'll actually sound less loud with the reverb added, because one of the ways in which we perceive loudness is to subconsciously register the difference in level between peaks and the lower-level sounds that come between them. The less contrast there is, the less loud the peaks sound. (Incidentally, it is possible to make peaks sound even louder by extending their duration slightly, which is why gated reverb sounds so powerful. The high level of reverb stretches percussive sounds, but then it suddenly stops, leaving plenty of contrast with the following quieter sounds.)



"The effect of too much reverb is to 'smear' percussive events, reducing the contrast between beats and the spaces between those beats."



Gated reverb has become something of a cliché -- though you shouldn't let that put you off using it altogether -- but similar results can be obtained by using ambience settings. These are characterised by strong early reflections but very little dense reverb afterwards. Dedicated ambience settings tend to appear only on the better reverb devices, but, providing you have a unit where the relative level of early reflections can be adjusted, you can approximate the effect well enough by setting the ER level to maximum. Pick a bright reverb algorithm with well-defined early reflections, then set a short decay time so that there's minimal 'ring' after the initial sound. These strong early reflections will help strengthen and widen sounds without smearing them excessively; even difficult sounds such as bass drums and bass guitar can sound good with just a touch of ambience added.

KEEP YOUR DISTANCE

Perhaps the biggest dilemma with reverb is that vocals sound great with lots of it, but as soon as you add it, the vocal loses the 'up-front' quality that we expect from a contemporary pop mix. This problem is related to psychoacoustics -- the nearer we are to a sound, the greater the proportion of direct sound we hear. In most environments, nearby sounds tend to seem fairly dry, whereas those at a distance may be more reverberant. Even in a reverberant environment, the perceived level of reverb will be lower for those sounds that are closest, as a greater proportion of direct sound reaches our ears. Distance also tends to dull sound, so for a vocal to sound 'up-front' it needs to be bright and dry -- and these conditions aren't always flattering to a voice.

Fortunately, there are ways to use reverb without losing the required sense of intimacy. One of these is to use a fairly short decay time, again with a high early reflections content. The other way is to place a pre-delay of several tens of milliseconds before the reverb, to provide some separation between the initial dry sound and the ensuing reverberation. Combining these two techniques can work effectively, but you'll still need to keep the reverb level under control. Listen to a selection of good contemporary music mixes and you'll find that many use so little reverb that you don't actually notice it unless you specifically listen for it.



"Because low frequencies take up so much headroom, it's better to remove the low end before the signal gets to the reverb unit input -- which should result in a better signal-to-noise performance."



The quality of the reverb processor being used also makes a huge difference, so save your best processors for vocals and drums, where the differences show up most. With the better processors, adding a lot of reverb doesn't seem to bury the sound in the same way that lots of reverb from a cheap unit does, and you can often get away with a much higher reverb-to-dry ratio before the sound becomes unnaturally muffled. Good reverb quality is particularly evident at short decay times, where lesser units may start to ring or sound unduly coloured. If you have access to a really good reverb unit, you might find that a vocal line sounds more effective treated with a higher level of fairly short reverb than it does with a lower level of a longer reverb.

A potential problem with using a bright-sounding reverb on vocals is that any sibilance in the original performance will already have been exaggerated by any compression that's been used, and once you add a bright reverb it may reach an annoying level. De-essing vocals often results in a lispy quality to the sound, so a kinder solution may be to de-ess the input to the reverb unit and leave the dry vocals as they were. Figure 2 (below) shows how this might be done.

BRIGHT IDEAS

From what's been said so far, you can probably deduce that sparing amounts of bright reverb, or reverb rich in early reflections, will help give a sound presence, width and interest without pushing it to the back of the mix. However, if bright reverbs are chosen for everything in the mix, the contrast element once again gets lost. Classical instruments tend to benefit from more natural reverb treatments, and in most cases that means using quite a lot of HF damping and HF rolloff to simulate a concert hall type of environment. Even in rooms with very hard surfaces, such as stone cathedrals, the reverb can be much less bright than you might expect, mainly due to the air absorption of high frequencies in the large distances between walls.



"Classical instruments tend to benefit from more natural reverb treatments, and in most cases that means using quite a lot of HF damping and HF rolloff to simulate a concert hall type of environment."



Using lots of early reflections can also be a bad idea when processing classical instruments, because when a large ensemble plays in a real concert hall, the early reflections have a tendency to disappear. The reason is quite simple: the pattern of early reflections depends on the position of the performer relative to the nearby walls and other boundaries, but if each performer is in a different position on stage, each will generate a slightly different set of early reflections. When these are all added together, individual reflections tend to become masked by the increased complexity of the reflected sound.

SUMMARY

Reverberation is a very powerful effect, without which no studio would be complete, but there are dangers associated with its use. In terms of perspective, excessive use of reverb pushes sounds to the back of the mix, while adding more than the merest hint of reverb to bass sounds clutters up the low end alarmingly. There are occasions when long reverb settings work, but these generally require musical arrangements that leave a lot of space for them to work in.

Adding much in the way of reverb to sustained pad sounds seldom works, as the sustain of the pad hides the reverb, which means that you have to add a lot to make the effect noticeable. As a rule, smoother sounds benefit more from coarse treatments with widely spaced early reflections while percussive sounds need a higher density of reverb -- otherwise the early reflections sound like somebody ripping cloth! Once you've picked an appropriate reverb sound, you then have to decide whether there's enough space in the music to let you use it as an obvious effect, or whether you should add as little as you can to create a convincing sense of space.



MUD GUARD

Reverb units can tend to emphasise anything that's bassy or muddy in the material being processed, and since most of the energy in a typical pop music mix resides at the low end of the audio spectrum, perhaps this isn't surprising. The problem can be reduced by EQ'ing some of the low end out of the reverb, and the easiest way to do this is to feed the returns through a couple of mixer channels rather than aux returns, so that you can use the channel EQ to apply bass cut. However, this isn't actually the best way to do the job. Because low frequencies take up so much headroom, it's better to remove the low end before the signal gets to the reverb unit input -- which should result in a better signal-to-noise performance. Any type of equaliser patched before the reverb input will do the job, but the high-pass side-chain filters on a noise gate such as the Drawmer DS201 (set to Key Listen mode) are particularly good for this purpose because of their steep 12dB/octave slope. Using these, it's possible to almost surgically remove the low end without changing the mid and high frequencies in any obvious way. Figure 1 shows a suitable patch for accomplishing this.

cheers
geo

[georgia]

Wallace Clement Sabine (1868-1919)

Sabine's Reverberation Formula
Wallace Clement Sabine was a pioneer in architectural acoustics. A century ago he started experiments in the Fogg lecture room at Harvard, to investigate the impact of absorption on the reverberation time. It was on the 29th of October 1898 that he discovered the type of relation between these quantities. Sabine derived an expression for the duration T of the residual sound to decay below the audible intensity, starting from a 1,000,000 times higher initial intensity:

T = 0.161 V/A

where V is the room volume in cubic meters, and A is the total absorption in square meters. Sabine's reverberation formula has been applied successfully for many years to determine material absorption coefficients by means of reverberation rooms. Keeping in mind some conditions with regard to the sound field diffusion and the value of A, Sabine's formula is still widely accepted as a very useful estimation method for the reverberation time in rooms.

Sabin as Unit of Sound Absorption
The unit of sound absorption is square meter, referring to the area of open window. This unit stems from the fact that sound energy travelling toward an open window in a room will not be reflected at all, but completely disappear in the open air outside. The effect would be the same if the open window would be replaced with 100 % absorbing material of the same dimensions.


Therefore, 1 square meter of 100 % absorbing material has an absorption of 1 square meter of open window. In honor of W.C. Sabine, the unit of absorption is also named sabin or metric sabin. However, these units are used not very often. One sabin is the absorption of one square foot of open window, and one metric sabin is the absorption of one square meter of open window.


Sabine Units of Common Materials (Given Per Square Foot or Per Unit)
Material
125Hz 250Hz 500Hz 1kHz 2kHz 4kHz

Carpet - heavy on concrete
0.02 0.06 0.14 0.37 0.60 0.65

Carpet - heavy on 40oz hair felt
0.08 0.24 0.57 0.69 0.71 0.73

Carpet - heavy with latex backin on foam or 40hz weave
0.08 0.27 0.39 0.34 0.48 0.63

Carpet- outdoor/indoor
0.01 0.05 0.10 0.20 0.45 0.65

Wood Floor
0.15 0.11 0.10 0.07 0.06 0.07

Concrete Floor
0.01 0.01 0.015 0.02 0.02 0.02

Linoleum, Asphalt-tile, or cork tile on concrete
0.02 0.03 0.03 0.03 0.03 0.02

Foam backed carpet on concrete
0.05 0.16 0.44 0.70 0.60 0.40

Heavy carpet + heavy foam underlay on concrete
0.15 0.25 0.50 0.60 0.70 0.80

Vinyl flooring
0.03 0.04 0.05 0.04 0.05 0.05

Gypsum board: 1/2" on 2 x 4s, 16" on centers (plasterboard)
0.29 0.10 0.05 0.04 0.07 0.09

9mm Plasterboard over 20mm air gap
0.30 0.2 0.15 0.05 0.05 0.05

Plaster, gypsum or lime smooth finish on tile or brick
0.013 0.015 0.02 0.03 0.04 0.05

Plaster: gypsum or lime, smooth finish on lath
0.14 0.10 0.06 0.05 0.04 0.03

Concrete Block, course
0.36 0.44 0.31 0.29 0.39 0.25

Concrete Block, painted
0.10 0.05 0.06 0.07 0.09 0.08

Plaster on brick
0.013 0.015 0.02 0.03 0.04 0.05

Brickwork
0.05 0.04 0.02 0.04 0.05 0.05

Owens-Corning Frescor, painted, 5/8" thick Mounting 7 Drop Ceiling
0.69 0.86 0.68 0.87 0.90 0.81

Drapes cotton 14oz/sq yd
- draped to 7/8 area
0.03 0.12 0.15 0.27 0.37 0.42
- draped to 3/4 area
0.04 0.23 0.40 0.57 0.53 0.40
- draped to 1/2 area
0.07 0.37 0.49 0.81 0.65 0.54

Cotton drapes draped to half area. 15oz/sq yd
0.07 0.37 0.49 0.81 0.65 0.54

Drapes medium velor, 18oz sq yd draped to 1/2 area
0.14 0.35 0.55 0.72 0.70 0.65

Acoustical tile, average 1/2" thick
0.07 0.21 0.66 0.75 0.62 0.49

Acoustical tile, average 3/4" thick
0.09 0.28 0.78 0.84 0.73 0.64

50mm Acoustic Foam
0.08 0.25 0.60 0.90 0.95 0.90

100mm Acoustic Foam
0.20 0.70 0.99 0.99 0.99 0.99

50mm Mineral Wool (Med Density)
0.20 0.45 0.70 0.80 0.80 0.80

Plate glass
0.18 0.06 0.04 0.03 0.02 0.02

6mm glass
0.10 0.06 0.04 0.03 0.02 0.02

Window glass
0.35 0.25 0.18 0.12 0.07 0.04

Breeze block
0.25 0.40 0.60 0.50 0.75 0.50

LF panel absorber
0.28 0.22 0.17 0.09 0.10 0.11

Perforated Helmholz absorber, 4-inch depth, mineral wool damping, 0.79% perforation.
0.40 0.840 0.40 0.160 0.140 0.12

Perforated Helmholz absorber,8-inch depth, mineral wool damping, 0.79% perforation.
0.98 0.88 0.52 0.21 0.16 0.14

Broad-band absorber, 1-inch fibreglass slab at mouth of 7-inch deep cavity
0.67 0.98 0.98 0.93 0.98 0.96

Padded seat (unoccupied)
0.10 0.20 0.25 0.30 0.40 0.30

College students informally dressed seated in tablet arm chairs (per person)
2.5 2.9 5.0 5.2 5.0

Audience seated, depending on spacing and upholstry of seats (per person)
2.5 3.5 4.0 4.5 5.0 4.5

Audience seated, depending on spacing and upholstry of seats (per person)
4.0 5.0 5.5 6.5 7.0 7.0


thought this might be fun to learn about...

cheers
geo

[joenovice]

Georgia Rocks!

Seriously, I've read so much of this stuff and I can't say enough "thanks." You've compiled so much information you should consider writing a book. Georiga's Audio Post Guide and Desk Reference.

I'd buy it. Aside from other websites, I've only seen a few books related to the topic.

[georgia]

here's some info about Soundmaster and VVTR.. Also Soundmaster now has a VVTR mode, a VVTR loose mode and a VVTR 1 frame mode. iF you have a video file that the frame edges are not perfect, contact soundmaster and they should be able to hook you up with the new settings...

Just a quick note to summarise the whole situation with respect to
VirtualVTR Pro, Kona3 and Soundmaster on Film Mix Stages,
particularly running 23.98 @ 29.97 configurations.

We have worked with Soundmaster and AJA over the last few weeks to
put into place a setup which covers pretty much all the bases you
will meet on a mix stage, ensuring frame accuracy and tight lock,
even when cross locking 23.98 material to an otherwise 29.97 world.

AJA Have created a beta version of a new Kona3 driver with a special
function we can call to precisely determine the frame on the air,
even when operating in 3:2 pull up or with cross locked timecode.
This driver along with the latest VirtualVTR Pro seems to work very
well.

The new Kona3 driver should be available on request to AJA, the one
we are using is:
Kona3_2006-12-13@10_57_14

NOTE: This is designed as a comprehensive explanation of the
technical considerations surrounding the use of VirtualVTR Pro on a
mix stage, with AJA Kona3, and Soundmaster ION, playing a variety of
video formats with the option of 23.98 to 29.97 cross locking. Its a
technical guide for ENGINEERS, not a user guide. Please do not
reproduce this information (some is proprietary) for the time being.

--------------------------------------------------------------------------------------------------
- QuickTime, VirtualVTR Pro and Kona operate in either a 'scheduled'
or 'unscheduled' video playback mode depending on a number of factors
which are determined as the movie opens. Scheduled mode prepares 30
frames in advance and gives them a target 'delivery' time to air to
be handled by the Kona driver, whilst Unscheduled mode has QuickTime
sending out 1 frame at a time through the software/hardware pipeline.

- In unscheduled mode, in some cases, there can be a situation where
the current time reported by quicktime is slightly offset from the
actual frame which is on air (typically less than a frame, but can be
as much as a frame). Also, in some cases the QuickTime perception of
time may be working on a frame edge offset from the real frame edge.
This can lead to rounding errors, once again affecting overall sync.
Whilst it is still possible to get good results in unscheduled mode,
there is a greater possibility for occasional sync mis-locks, and so
wherever possible, Scheduled mode is preferred. In Unscheduled mode
you will need to determine and enter the 'sync correction' parameter
in VirtualVTR for the codec in use, compensating for codec, video
hardware and then projector latencies.

- In Scheduled mode, QuickTime prepares a series of frames in
advance, and tells Kona when each frame belongs on the air. This has
the added benefit of having a cache of frames ready in the driver and
can make for smoother playback with challenging media formats like
DVCProHD, particularly where the CPU is working hard. In theory if
we ask QuickTime for the current time, it will tell us, and in theory
the frame targeted for that time *should* be on the air. However, in
practice the slewing process used by Soundmaster will 'slip' the sync
between 'proposed time' and 'actual time' for the delivery of the
frames (not least because variations in speed during slew will put
the QuickTime timeline frame edge out of sync with the genlock edge).
Consequently it is not safe to 'take Quicktime's word for it' when
reporting timecode back to the Soundmaster. Fortunately AJA have
provided a mechanism which allows us to bypass or correct QuickTime's
own perception of time, and replace it with a literal report of which
frame has actually 'just left the building'. This offers superbly
accurate sync. In Scheduled mode the 'sync correction' parameter in
VirtualVTR is theoretically zero, but you may need to enter a value
to compensate for projector latencies.

There are a number of factors which determine whether QuickTime
selects Scheduled or Unscheduled mode when opening a movie.
Essentially QuickTime builds a 'codec calling chain' as it opens a
file, figuring out what steps are required to get from the Movie file
(in a compressed format typically, at a certain size) across to the
Video Output (which might expect uncompressed frames, at a specific
size). QuickTime asks the Video card driver what format of video it
expects. If mismatches exist between source and destination, then
QuickTime has to put processes into the calling pipeline to correct
for these. Typically, if QuickTime includes conversion processes, it
will choose Unscheduled mode of playback. If there are no
mismatches, it will choose Scheduled mode. So, then simplest way to
ensure scheduled mode is to make sure your Sound movie matches the
basic format requirements of the Video card driver (eg. uncompressed
8-bit, full frame size).

As we are aware in practice, it is rare to see a movie in
uncompressed format arrive on a mix stage, and fortunately, there are
still lots of configurations where things dont match 1:1 and we can
still get scheduled playback. For example - The Kona driver natively
expects 8 or 10 bit uncompressed Video at 720x576, or 720x486 (SD) or
1920x1080, 1280x720 (HD). However, it *also* supports a range of
other codecs and sizes *natively* which means that QuickTime doesn't
have to install a pipeline processor, and scheduled mode can work.
The other parameters Kona will accept include

FRAME SIZES:
- 720x486, 720x576, 1920x1080, 1280x960 (standard sizes)
- 720x480 frame size (particularly for DV-NTSC video frames which use
this size - it will center the picture, not stretch)
- 1440/1280x1080 frame size (particularly for DVCProHD 1080 video
frames which natively this size - it will stretch the picture)
- 960x720 frame size (particularly for DVCProHD 720 video frames
which natively this size - it will stretch the picture)
- 640 x 480 (not sure whether it stretches or centers)
- 320 x 240 and some other offline RT sizes (stretches)
CODECS:
- 8 and 10 but 2vuy or v210 (standard uncompressed codecs)
- DV25, DV50, DVCProHD
- MotionJPEG, PhotoJPEG
- Avid Meridien
- IMX (??? not certain of this)
- Some other uncompressed codecs with color components in different
arrangements.

If you are using a frame size which doesn't match, or a codec which
doesn't match, then its likely playback will be unscheduled.
VirtualVTR Pro indicated Scheduled or Unscheduled playback using a
Green or Red 'Disk' LED which shows the number of cached frames (this
is always zero in the case of unscheduled playback).

- VirtualVTR Pro superimpose Graphics buffer.
VirtualVTR Pro also has the option of inserting a video frame buffer
between QuickTime and the Kona, where graphics can be superimposed
(or it can be used as a conversion compatibility layer). Like any
other pipeline processor, this causes the chain to be built in
unscheduled mode. However for some more exotic codecs, like H264,
HDV, DNxHD and MPEG2, it may be necessary to use the Graphics buffer
as a translation layer, and live with unscheduled mode. These codecs
are interframe and typically long GOP which makes it almost
impossible to deliver them directly to the Kona card without
processing. VirtualVTR Pro can allow you to play these formats direct
to air, but you will have to use the Graphics overlay buffer. The
other option is to convert the format to DVCProHD or medium Q
PhotoJPEG before opening in VVTR Pro (this technique *would* allow
Scheduled mode, but takes time before you can work).

- Making Scheduled Mode Sync accurate
As described, Scheduled mode is good, but we must get data from the
Kona card not from QuickTime if we are to report time accurately. In
regular framerate=timecode modes, we can simply ask the Kona every
timecode frame, which video frame is on the air. This gives very
accurate results. However there are times when Kona cannot return
this information, particularly during slewing (during which time QT
operates in unscheduled mode), also, when cross locking (playing
23.98 picture with a 29.97 timecode - Soundmaster locked to NTSC),
the information from Kona would not correspond directly with the
timecode and Soundmaster reference, so we use a variation technique,
called XCheck (in VVTR Pro prefs).

Essentially, the prefs you need are ' Query VOut for Time' - to make
VirtualVTR Pro use the Kona representation of Time during Scheduled
playback. In Unscheduled playback this is ignored. Also, the
'XCheck' additional mode uses a more advanced call, only available in
Kona3.3+ which makes the Query Vout mode work properly when cross
locked. It should be safe to leave both these options turned ON at
all times and it will use them when available, or defer to other
techniques when not. There is one exception to this rule - if your
synchroniser expects to watch the 9-pin 'Servo Lock' status flag
(Soundmaster does not, I believe) then you will find that the Servo
Lock flag never comes on in a situation where Query VOut is selected,
but scheduled mode is not used (since we are waiting for the
Scheduled mode to settle *before* we enable servo Lock).

- 24, 30 and Pull Down considerations
VirtualVTR Pro has a 0.1% pull down option which will automatically
conform a movie when opened from 24 or 30 FPS to 23.98 or 29.97 FPS.
Note that it does NOT process the sound, so after pull down you must
ignore the embedded audio in VirtualVTR Pro since it will be out of
sync. If you prefer other pre-processing QT conform tools exist
(cinema tools ?). Normally the VirtualVTR Pro conform will be non
destructive - ie. if you close the movie it will rever to 24/30 FPS.
However if you TIMESTAMP the movie whilst it is open and pulled down,
this will permanently conform the movie file to the pulled down rate.
A very useful tool in VirtualVTR Pro is the 'Get Info' in the bin -
select this when a movie is closed, and you will see the Movie and
Media Timescale and the frame duration values - these are the
fundamental QuickTime time units which the movie is built with. Most
common will be 2500/2500/100 for a PAL movie, 2997/2997/100 for an
NTSC movie, and 23976/23976/1000 for a 'perfect' 23.98 movie. You
might see the first parameter set to 600 which is a common error
caused by QuickTime player - however if you have the VVTR Pref
'Correct Movie Timescale' set, this will be corrected as the movie
opens, to match the second parameter. The first parameter is used to
locate VVTR frame by frame, so its quite important that it
corresponds to the actual frames. You might see 24/24/1 for a 24 fps
movie, 25/25/1 for PAL or even things with 80 as the duration
(instead of 1, 100, or 1000). This is where you should take care.
Values like 2398/2398/80 are NOT accurate enough because of rounding
errors (whereas 23976/23976/800 is), and the 80 duration movie (which
might be something like an NTSC export from a 24 FPS movie), is
likely to have a long term sync drift. In particular when exporting
from a Film Composer (24 FPS) it is VITAL that the user does NOT try
to 'create' a 29.97 or a 23.98 movie- this will most likely cause the
situation described due to inaccuracies. Instead the FilmComposer
should output a true 24 or 30 FPS movie, and VVTR Pro will pull it
down during playback.

- Playing 24 FPS SD movies Progressive on a projector.
As we all know, there is no 24 FPS standard definition video mode.
However you will all have come across 24 FPS SD Quicktime movies,
from Avid Film Composer, or Final Cut Pro (typically Avid Meriden
format or DV format). On a film Mix stage there is a great desire to
preserve the 'true to film' 24 FPS progressive scan material and
project it in this format without corrupting the integrity with the
traditional 3:2 pull up and interlacing of the picture. Fortunately,
AJA have again come to the rescue, and Kona3 has a 24 FPS SD native
display mode, and a hardware upconvertor which allows you to play a
24 FPS Avid (or DV) Movie in fully Scheduled mode, with Kona
upconverting to 1080 24p (or 23.98p) on the HD-SDI output. This
offers your customer a 24p Progressive screening of their QuickTime
movies, which is as close to mixing against film as you are going to
get these days. This can all be done, even on a stage which uses NTSC
sync and 29.97 FPS Timecode to the Soundmaster (you will require a
coherent NTSC and TriLevel sync generator).

VirtualVTR Pro and Kona offer the most flexible playback platform for
Mix Stages, supporting most common desktop video formats natively
with no conversion.

Clearly there is alot of information in this document, which we have
chosen to share with you so you are fully educated. However in
practice users of VirtualVTR Pro do NOT need to know all this:

You may want to define a fixed set of video formats which you are
willing to accept from your customers (just like you do with the
linear decks you have available), and then define a set of prefs for
your operators to match each incoming format. You can always expand
your list of accepted formats over time. In your specification you
should clearly define the video frame size and the codec.

Just like your users don't understand the technical details of HDCam
tape, they do know which machine to put it into, and which buttons
to press. So it is with VirtualVTR Pro - define some standards for
your customers and have a fixed user process for dealing with each
standard. In these times where there are hundreds of format
variations it's vital to nail it down to some extent. We have done
everything possible to ensure that the widest range of movie formats
will play, without conversion, and with accurate sync, but there are
limits to what is possible and this is where you must create some
boundaries. In most cases the last resort is to convert (and perhaps
scale) the movie with QuickTime player into one of the supported
formats, and in general this will allow an almost unlimited range of
incoming formats.

Our system now works with "pro" and "not so pro" video files.


cheers
geo

[georgia]

Determining the Harmonic Frequencies

speed = frequency • wavelength
frequency = speed/wavelength
f2 = v / 2

speed = frequency • wavelength
frequency = speed/wavelength
f3 = v / 3

The frequency of the third harmonic is three times the frequency of the first harmonic. The frequency of the nth harmonic is n times the frequency of the first harmonic.

fn = n • f1

The inverse of this pattern exists for the wavelength values of the harmonics. The wavelength of the second harmonic is one-half (1/2) the wavelength of the first harmonic.

The wavelength of the third harmonic is one-third (1/3) the wavelength of the first harmonic.

And the wavelength of the nth harmonic is one-nth (1/n) the wavelength of the first harmonic.

n = (1/n) • 1

cheers
geo

[georgia]

I feel your pain. I had a very similar problem recently. I backed up a session then put it in the Trash and Emptied and then realized I had audio files in that session folder that were from a different session. Since I had used Save Session Copy In to do the backup, those files weren't copied. So they were gone. Or so I thought.
First and most important thing: Do NOT use the hard disk for anything!! Don't even launch Pro Tools with the thing connected. Pro Tools may very well update the .ddb file on that drive which could overwrite important data. Just leave it unconnected until you are ready to attempt recovery, which I will describe next.


You will need three programs (well, two actually, but a text editor makes life simpler):
1. Terminal (found in your Applications:Utilities folder)
2. HexEdit (freeware)
3. TextEdit (in your Apps folder) or, my favorite, TextEdit (by Haxial)
You will also need an extra hard disk or two, and a Pro Tools system capable of playing the same audio files you are trying to recover (i.e.: if you're trying to get back 96kHz files, you need a 002 or HD rig.) For simplicity's sake we're going to refer to your precious hard disk as the Source Disk and the extra hard disks as your Destination Disk(s).
Here's what you're going to do
1. Read raw data off of the hard disk, 1GB at a time, to a second hard disk.
2. Attach audio file wrapper data to each raw data file. Do this THREE times if you are trying to recover 24-bit audio data. Do it TWICE if you're recovering 16-bit data. I'll explain.
3. Import the new audio files into Pro Tools.
4. Manually comb through the audio files in Pro Tools to find the data you want. This part of the process may take quite a while and test your patience, but if the data is really important, you'll do it.
Some caveats (in no particular order)
1. This is going to take time. Quite a bit of time. If you need this done quickly, and you're getting paid, it might be better to send it to a data recovery company and pay big $$. That's up to you.
2. This is going to take up a lot of disk space. The bigger the hard disk you are trying to recover, the more space you'll eventually need. As a rule of thumb, figure on 4x the amount of disk space from the original drive. If you don't have this much extra space, you can do it in chunks, but some of the time-saving techniques won't be as useful.
3. You need to know what type of audio files you're looking for. File format, sample rate and bit depth. If you are trying to recover a whole bunch of different file types or you don't know the types, you're in for a REALLY LONG HAUL. You'll basically have to repeat this entire procedure for each file type. If you're willing to do that you must have some really important audio files to recover. Good luck.
4. Again (worth repeating), do NOT write to your affected disk drive until you have recovered your audio or given up. This is really important. Since you emptied the trash, the computer doesn't know where those audio files are and could write over them without warning if you save something to that disk.
How to do it
1. Connect your Source and Destination Disks to your computer.
2. Boot up.
3. Launch Terminal.
4. In Terminal type su. Enter your password. You're now in superuser mode. Be careful.
Now you need to find out some info via unix:
5. In Terminal type df. This will show the mount point of your drives. On far right is the name of the disk; e.g. /Volumes/MyDiskName. On far left is "unix device mount point", e.g. /dev/disk1s9. 1 is the disk number (e.g. 1st disk found since booting). 9 is the partition number.
6. Find your disk in this list by the name and note down the mount point. This is how you will tell unix where to read raw data from.
Here goes the main recovery effort. I suggest you read ahead before you actually type this stuff.
7. In Terminal type dd if=/dev/rdisk1s9 of=/Volumes/MyDestinationDisk/01 count=2m
Okay, what's all this about?
dd This is a unix command that allows raw data reading/writing.
if= Tells dd command that this is the Input File.
/dev/rdisk1s9 This is your mount point that you found in step 5. The numbers will probably be different on your system. The r is added in to tell dd that you want to do a raw disk read. The r is very important!
of= Tells dd command that this is the Output File.
/Volumes/MyDestinationDisk/01 Here's where you need to put in your Destination Disk name. Keep the /Volumes/ part since this is the same on all OS X systems and then type the name of the hard disk volume you want to write to. Add a slash after the name and then type a file name for the new raw data file you're going to create. I use numbers, like 01 since this command will be executed many times. Each time I increase this file number by 1 to keep things organized. (Unix scripters will see opportunity for automation here but doing it manually gives the same results.)
count=2m This tells dd how much data you want to read. 2m=1GB. (Unix deals in 512 Kilobyte chunks.)
8. If this all looks fine and dandy to you, press Enter. You won't see much going on but if you look at your drive access lights you'll see that reads and writes are occuring.
Now you may be wondering how unix knows which 1GB of data to read off of your drive? Simple. It just reads the first 1GB of data. So how do you get it to read the 2nd GB? or the 3rd? Or the 49th? Easy. Just add an additional command at the end of the dd line that says skip=2m. This tells dd to start reading raw data 1GB from the beginning of the disk. You'd use this to create your second raw data file. Your third file would need skip=4m added to it. The fourth will need skip=6m. Etc. A handy equation for this is of=N skip=(N*2-2)m. I.e.: Your Output File number is N and the number in the skip part is N*2 - 2.
So your second raw data recovery will look like:
dd if=/dev/rdisk1s9 of=/Volumes/MyDestinationDisk/02 count=2m skip=2m
Your third raw data recovery will look like:
dd if=/dev/rdisk1s9 of=/Volumes/MyDestinationDisk/03 count=2m skip=4m
Your fourth raw data recovery will look like:
dd if=/dev/rdisk1s9 of=/Volumes/MyDestinationDisk/04 count=2m skip=6m
And so on. Until you've recovered all your data or run out of disk space on Destination Disk. If you have to stop in the middle, make note of where you left off and DISCONNECT your Source Disk.
How to speed up the raw data recovery process.
You may have noticed that each 1GB of data takes a long time to recover. I don't suggest you use your computer for any other tasks during this process so you probably want to do this stuff late at night or whenever the computer is not in use. But you don't want to be there babysitting the thing all night long. There's a solution. You can type commands into unix one after another if you seperate them with a semicolon. That's what I use TextEdit for. It's much easier to copy and paste a whole bunch of those commands into TextEdit, then scroll through and change the Output File names (the numbers) and the skip commands. Then you can copy and paste out of TextEdit back into Terminal. You'll end up with something like this:
dd if=/dev/rdisk1s9 of=/Volumes/MyDestinationDisk/01 count=2m; dd if=/dev/rdisk1s9 of=/Volumes/MyDestinationDisk/02 count=2m skip=2m; dd if=/dev/rdisk1s9 of=/Volumes/MyDestinationDisk/03 count=2m skip=4m; dd if=/dev/rdisk1s9 of=/Volumes/MyDestinationDisk/04 count=2m skip=6m; etc.
I find that resizing the TextEdit window so that one dd command fits perfectly on a line also helps in making the edits correctly.
Be careful! This is a powerful technique for getting things done in a big batch, but if you make a small typing error you can do disastrous things to your hard disks. Or you can accidentally write each new raw data file over the previous one, which means you won't really know what you recovered and you'll have to start over. If you know how to write shell scripts in unix, I'm sure you'll be automating this whole process. I found it more satisfying to be able to see each and every command written out before I let them loose on my system.
Also, don't forget that you need to do the superuser mode change each time you launch Terminal. And you should recheck the df command to make sure the mount points didn't change (they can every time the system boots.)
You're one-third of the way there.
The next step is to add the audio file wrappers so you can import these new raw data files into Pro Tools, but first you need to create the wrapper files. There are two files to make a wrapper, one that comes first, then your raw data, then a second wrapper. I'll call these wrappers header and footer. You'll use the same footer for each raw data file, but you'll need to create three headers for each raw data file. Why? Pro Tools records 24-bit data. 24-bit data is broken up into three 8-bit chunks. Since you recovered raw data, you don't know what the right ordering of the three chunks. The only way to make sure you can get all audio data back is to create three versions of each raw data file, each version being a different ordering of the 8-bit chunks. So you'll need to create three header files, the second one being 1 byte longer than the first, and the third being 2 bytes longer than the first. When you attach the raw data to each of these, the byte ordering will start in each of the three possible places allowing you to find all the audio data that may be in the raw files. Sounds a little complicated but it's really not.
Creating header and footer files
1. Disconnect your precious Source Disk!!!
2. Launch Pro Tools.
3. Create a session in the same file format, sample rate and bit depth as the data you're trying to recover. If you don't know, you'll have to create headers and footers for each file type and do everything from here on over and over until you find your data. If this is your situation, think long and hard about how important those files are. Unless you have original recordings of spacealiens I suggest you forget trying to recover and re-record stuff. If you DID have original recordings of spacealiens and you didn't make a backup copy immediately then I suggest you sell your computer and get a job in a less intellectually demanding field. That said, mistakes happen. I made such a mistake. Luckily, I remembered the file type, sample rate, bit depth and even the number of channels of what I was looking for. So I pressed on.
4. Make a selection on your timeline that will create an audio file 1GB in size.
5. Press Option-Shift-3 to create a blank audio file with your selection.
6. You'll need to do this several times to zero in on the exact time needed to create a 1GB file. I'm not sure how exact you have to be but I got it to the exact file size through trial and error. Come to think of it, it's probably better to have a little bigger file than exactly 1GB because the headers and footers will take a little space. Oh well. My recovery worked well. I got back over 99% of my lost data (about 4 hours worth) and that was good enough for me.
7. Quit Pro Tools once you have your 1GB audio file. Name that file something like 1GBAudioFile.
8. Now launch HexEdit.
9. Open 1GBAudioFile in HexEdit. A sea of numbers will fill the window. Don't flinch.
10. HexEdit shows the raw hex data of the open file on the left side and the "human readable" version on the right. Sometimes you can make out words on the right side, but for audio data it just looks like garbage. Scroll down through the data until you see a large amount of zeroes that goes on forever. This is the audio data. Since you created a blank audio file, it's just zeroes in there, so it's easy to see where it begins and ends.
11. Find the beginning of the audio data.
12. Copy all the data from just before the audio data all the way to the beginning of the file.
13. Create a new file in HexEdit and copy this data into it.
14. Name this file "HeaderA".
15. In HeaderA file, copy and paste a single 00 to the end of the file.
16. Name this file "HeaderB".
17. In HeaderB file, copy and paste a single 00 to the end of the file.
18. Name this file "HeaderC". You now have three header files, each one byte longer than the previous one. Now for the footer.
19. Find the end of the audio data in the original 1GBAudioFile. It's the where the zeros all end and you get a few non-zero numbers showing up. There will be more zeros after this (where the waveform drawing data is stored) so make sure you scroll in far enough to get the end of actual audio data. This could take a while since you're looking at A LOT of data.
20. Copy all the data from the end of the audio data to the end of the file.
21. Create a new file and copy this data into it.
22. Name this file "Footer".
23. Quit HexEdit.
24. Eat something, take a break.
25. Copy the three Header and one Footer files into the same folder with your raw audio data files.
26. Launch Terminal.
27. Type cd and then drag the folder containing your raw audio files into the Terminal window. It will autofill the pathname for you. Hit Enter. Now you are in the same directory (folder) as your raw audio files so commands will look for files in this directory.
28. Type cat HeaderA 01 Footer > 01A; cat HeaderB 01 Footer > 01B; cat HeaderC 01 Footer > 01C This is three commands in a row (note the semicolons.) Each one says concatenate file HeaderA then file 01 then file Footer in that order into a new file called 01A. Then do the same for B, then C. So you now have three copies of your raw data file with the audio file wrappers on them. They are now ready to be imported into Pro Tools.
29. Repeat step 28 for every raw data file you recovered. You can queue up a lot of these commands with semicolons just like you did with the dd command earlier, as long as you have the disk space. As you can now see, you'll have three more 1GB files on your drive for each 1GB raw data file, making 4GB of data. This is where the 4x space requirement comes from. If you're doing an 80GB drive, you'll eventually need 320GB of space to check every bit of space on the disk for audio (no pun intended.) Breaking it up into smaller chunks, like doing 10GB of a time might be necessary if you don't have lots of free disk space. And remember, don't use your original precious hard disk!! Don't even think about writing to it until you are completely done recovering data from it.
30. If you are trying to recover 16-bit data, then you only need to create two copies of each raw data file. There are only two bytes of 8 bits each in this case.
The moment of truth
You have three 1GB files for each 1GB of space on your original disk. Now you need to put them in Pro Tools and start listening to them and looking at them to identify audio data and save it.
1. Launch Pro Tools.
2. Create a new session at the same sample rate, bit depth, format as your recovered data.
3. Import files 01A, 01B and 01C to Audio Tracks.
4. Let the waveforms redraw. If they don't automatically redraw, select them in the Region Bin and force the redraw.
5. You are now looking at the the first 1GB of data from your drive, presented with three different byte orderings. You will probably see large regions of what looks like solid full code noise (looks like a brick when zoomed out). Interspersed with these bricks you'll hopefully see what looks like regular audio.
6. Turn down your speakers!!! You're going to be hearing some unpleasant noises coming from your system. Do not use headphones.
7. Solo one of the tracks and start playing the parts that look like actual audio data. If it sounds right, then go ahead and select and delete the regions on the other two tracks that are in the same place as the good audio data. You're deleting what is the same data as the good stuff, just shifted one or two bits so what should be the bottom 8 bits of your audio is now the top 8 bits or something like that. Go ahead and listen to some of it. Depending on the original material, there may be some interesting stuff in there especially if you like noise and distortion.
8. Repeat step 7 for each of the three tracks until you've gotten rid of all the stuff that you know is junk. You'll be left with mostly real audio and probably some unknown sections that don't appear to have audio on them in any of the tracks. Listen to the unknown stuff and determine if it's of any value. Most likely it's junk.
9. Repeat it all with the each and every file you recovered and created.
10. You may have noticed while going through the audio that some stuff will get cut off at the end of a file and then pick up at the beginning of the next one. This is helpful so you can edit stuff back together. You'll also notice that you've lost all meaningful timing relationships between tracks that were recorded at the same time, or edited together later. You'll have to reassemble your session manually. This may sound nightmarish, but if this data is really that important to recover and you've come this far, you can probably recreate the session even better than you did originally. Good luck.

The End




cheers
geo

[georgia]

HD Technical Requirements for High Definition Programming

2. AUDIO SPECIFICATIONS

Audio program material shall be produced using current industry standards and accepted norms. The audio portion of the master and source audio and videotapes must be produced so that no noise, static, dropouts or extraneous distortion is recorded in the audio.

Program audio must reflect reference tone level. Audio levels must be consistent throughout the program.

2.1 Stereo (LPCM) Programs


2.1.1 Phasing

Stereo audio must be fully mono compatible, i.e. the audio channels must be in the proper phase. NOTE: Full Mono compatibility means that when the left and right stereo channels are actively combined to mono there is no discernible change in audio level or fidelity.

Full mix and M & E audio tracks should be phase coherent (synchronized) and level matched to prevent difficulty editing between these tracks, as necessary.

2.1.2 Sound to Video Synchronization (Lip-synchronization)

The relative timing of sound to video should not exhibit any perceptible error. Sound should not lead or lag the vision by more than 10ms. This synchronization must be achieved at the last point at which the program supplier, or their facility provider, has control of the signal.

2.1.3 Headroom

Transmission limiters clip at +8 dB. For broadcast stereo tracks, transient audio peaks must not exceed +8 dB above reference tone when measured on an audio meter using the "True-peak" ballistic set (0 ms rise, 200 ms fall). For 5.1 surround mixes, audio peaks may rise as high as +17 dBm (-3 dBfs). When mastering to a digital format and/or using an Absolute Scale or Peak meter, where "0" is at the top of the scale and reference tone is at -20 dBfs, broadcast stereo tracks should peak at no more than -12 dBfs.

2.1.4 Audio compression:

Program audio should have good dynamic range, within the parameters listed above, but not be overly dynamic. While some compression may be needed to control the dynamic range of the program audio, excessive audio compression of the final mix should be avoided as this reduces the perception of audio quality by the listener.

2.2 Surround Programs

2.2.1 Formats

5.1, 5.0 or LCRS mixes are permitted. Surround English Fullmix (regardless of configuration (5.1, 5.0, etc) shall be expressed as Dolby 'E' on ch3/4 of HDCAM master.

2.2.2 Documentation

An Audio Program Data Sheet shall be delivered with the master tape. (See accompanying example)

2.2.3 20bit Dolby E (6 channel)

Valid metadata in the Dolby 'E' stream for all contribution/transmission parameters is mandatory.

Timecode shall be present in the bit stream, reflecting picture master.

The Dolby E stream shall be formatted such that the program is in sync following Dolby 'E' decoding using a DP572 or equivalent.

Note: One frame of audio delay is incurred for both Dolby E encoding and decoding. Program audio that is advanced two frames relative to picture prior to Dolby E encoding will therefore be advanced one frame as it is recorded to the HDCAM master. Following normal playback, the Dolby E decode cycle will delay one additional frame, bringing the program back into sync.

Maximum permissible audio peaks in a 5.1 or 5.0 soundtrack shall be -3dBFS (+17dBm)

Although the Max dynamic range (max. peaks) for 5.1 channel mixes is considerably higher than for Stereo-only LPCM mixes, it is understood that many 5.1 mixes will have a dynamics structure which more closely resembles a -10dBFS stereo mix in order to facilitate the simple creation of an Lt/Rt fold-down mix.

Regardless of the gain structure of a 5.x channel surround mix, it is crucial that the supplied DIALNORM value accurately reflects the Leq (A) of program dialogue.

2.2.4 Stereo English Full-mix (LPCM, conventional stereo digital)

This shall be recorded on channels 1 and 2 of the HDCAM master tape and may be used for screening and/or Standard Definition Transmission.

This mix shall be derived from the 5.x channel surround mix. i.e. "Fold-down" of the 5.1 or 5.0 mix to LCRS or Stereo (L/R).

This stereo mix should be expressed as Dolby Surround (Lt/Rt) whenever possible, or Lo/Ro if Dolby Surround encoding is not available. Tape labeling and slate information shall reflect the nature of channels 1 and 2 (either Lt/Rt or Lo/Ro). In either case (Dolby Stereo or not), the LPCM stereo Full-mix shall obey the conventional specifications for audio delivery (e.g. Max peaks to 8dB over ref.).

2.3 Channel Allocations

All HDCAM masters should have the following audio channel allocations:

(A) SURROUND PROGRAM

Channel 1 - Program left (Lt or Lo)
Channel 2 - Program right (Rt or Ro)
Channel 3 - Dolby E
Channel 4 - Dolby E
Address Track - SMPTE drop frame time code


(B) STEREO PROGRAM

Channel 1 - Program left (Lt or Lo)
Channel 2 - Program right (Rt or Ro)
Channel 3 - M&E left
Channel 4 - M&E right

*If the stereo program is Dolby Surround encoded (Lt/Rt), then any stereo M&E mix

(where applicable) shall also be expressed as Lt/Rt.

Address Track - SMPTE drop frame time code

Dolby E Mastering Information

Date Program Start Time
Program Title Episode# or Sub Title
Producer Director
Post Sound Facility Mix Engineer

Dolby E Formatting
Sampling Frequency ? 48 kHz (mandatory)
Bit Resolution ? 16-bit ? 20-bit ? 24-bit
Time Code Format ? 23.976 ? 25/50 ? 29.97/59.94 DF
Tape Format ? HDCAM
Program Configuration ? 5.1 + 2 ? 5.1 ? 4
Sync (frame offset) ? -1 ? 0 ? +1

Audio Service Configuration Bitstream Information
Audio Coding Mode ? 3/2 ? 3/1 Audio Production Information ? YES ? NO
Bitstream Mode ? Complete Main ? Main M&E Original Bitstream ? YES ? NO
LFE Filter ? Enabled ? Disabled Copyright ? YES ? NO
Mix Room Type ? Large ? Small
Mix Level

Processing Extended Bitstream Information
Dialog Normalization Preferred Stereo Downmix Mode ? Not Indicated
RF Overmod Protection ? Enabled ? Disabled ? Lt/Rt Preferred
Digital De-emphasis ? Enabled ? Disabled ? Lo/Ro Preferred
DC Filter ? Enabled ? Disabled Lt/Rt Center Downmix Level
Bandwidth Lowpass ? Enabled ? Disabled Lt/Rt Surround Downmix Level
LFE Lowpass Filter ? Enabled ? Disabled Lo/Ro Center Downmix Level
Digital De-emphasis ? Enabled ? Disabled Lo/Ro Surround Downmix Level

Dynamic Range Control
Line Mode ? None ? Speech ? Film Std. ? Film Light ? Music Std. ? Music Light
RF Mode ? None ? Speech ? Film Std. ? Film Light ? Music Std. ? Music Light

Downmix Processing
Dolby Surround Mode ? Not Indicated ? Dolby Surround ? Not Dolby Surround
Center Downmix Level ? -3 dB ? -4.5 dB ? -6 dB
Surround Downmix Level ? -3 dB ? -6 dB ? -999 dB
Surround 3 dB Attenuation ? Enabled ? Disabled
90-Degree Phase-Shift ? Enabled ? Disabled

Track Format Notes
Channel 1 Front Left
Channel 2 Front Right
Channel 3 Center
Channel 4 LFE (where applicable)
Channel 5 Surround Left
Channel 6 Surround Right
Channel 7
Channel 8

MUSIC AND EFFECTS TRACKS

TECHNICAL FORMAT

2.4 Accompanying Audio Multi-track Format (if required)

Accepted format is DA-88.

2.4.1 TRACK ALLOCATION


8 Track Digital Audio (DA-98 or DA-88)

Track 1 - English Fullmix Left (Lt if available)

Track 2 - English Fullmix Right (Rt if available)

Track 3 - undipped BG/FX Left (Lt if available)

Track 4 - undipped BG/FX Right (Rt if available)

Track 5 - undipped Music Left (Lt if available)

Track 6 - undipped Music Right (Rt if available)

Track 7 - Narration/VO dialogue

Track 8 - On-camera/Actuality dialogue

29.97 SMPTE Time Code on the Time Code Track to be synchronous with picture master(s).

2.4.2 Mix reference

Reference on all Masters shall be -20dbFS (or equivalent) and peak program level shall be restricted to 8db above reference (or -12dbFS)

2.4.3 Timecode

On DA-88 Master, timecode shall match picture Masters (i.e. 01:00:00:00 program start, drop-frame)

2.4.4 Sample Rates

On DA-88 Master, sampling rate shall be 48kHz (16bit) and noise shaping (where applicable) shall not be used on Mix Stems (tracks 3 through 8). If noise shaping is employed on stereo full mix, this shall be noted on tape labels.

2.4.5 Audio Compression and Limiting

Mix Stems shall NOT be dynamically buss-limited (i.e. stems are not restricted to the 12db over ref. peak limit). Stems summed at unity gain shall result in an unlimited version of the stereo full mix.

2.4.6 Reference Signals

Test tones for all Multi-track Masters shall be 1kHz tone @ -20dbFS.



cheers
geo

[Frank S.]

Georgia - Who are those tech specs from? Most of it is very similar to the clients I deal with, but there are some differences. I want to make sure I have the latest deliverable requirements!

Thanks...
Frank

[jacobfarron]

This is No. HD-05.2 Discovery.

[georgia]

Every company had different delivery specs... Lots of the same with many overlapping standards, but there's always something that i'll nail you if yo don't read the specs before QC and shipping....

cheers
geo

PS: I posted these, becuase they are a good example of industry delivery requirements for newbies to look at...

[georgia]

here's some links and some blu-ray information. I needed to do my homework since I realized how little I actually knew about Blu-ray...

Blu-ray Disc is a next-generation, optical disc format that enables the ultimate high-def entertainment experience. Blu-ray Disc provides these key features and advantages:
Maximum picture resolution. Blu-ray Disc delivers full 1080p* video resolution to provide pristine picture quality.
Largest capacity available anywhere (25 GB single layer/50 GB dual layer). Blu-ray Disc offers up to 5X the capacity of today’s DVDs.
Best audio possible. Blu-ray Disc provides as many as 7.1 channels of native, uncompressed surround sound for crystal-clear audio entertainment.
Enhanced interactivity. Enjoy such capabilities as seamless menu navigation, exciting, new bonus features, and network/Internet connectivity.
Broadest industry support from brands you trust. More than 90% of major Hollywood studios, virtually all leading consumer electronics companies, four of the top computer brands, the world’s two largest music companies, PLAYSTATION® 3 and the leading gaming companies, all support Blu-ray Disc.
The largest selection of high-def playback devices.Blu-ray Disc is supported by many of the leading consumer electronics and computing manufacturers. That means you can maximize the use of your HDTV and your home entertainment system with the widest selection of high-def playback devices—including players, recorders, computers, aftermarket drives and the PLAYSTATION® 3 game console.
Backward compatibility**. Blu-ray Disc players enable you to continue to view and enjoy your existing DVD libraries.
Disc robustness. Breakthroughs in hard-coating technologies enable Blu-ray Disc to offer the strongest resistance to scratches and fingerprints.

Public Specifications
http://www.blu-raydisc.com/assets/do...0307-13404.pdf
http://www.blu-raydisc.com/assets/do...sual-12838.pdf
http://www.blu-raydisc.com/assets/do...0305-12955.pdf
http://www.blu-raydisc.com/assets/do...rmat-12834.pdf
http://www.blu-raydisc.com/assets/do...gies-12835.pdf
Dolby Authoring and Mastering Solutions for High-Definition Disc Media, Blu-ray DVD, HD DVD, and DTV
Blu-ray.com - Blu-ray Movies, Players, Recorders, Media and Software

codecs for Blu-ray

Linear PCM (LPCM) - up to 8 channels of uncompressed audio. (mandatory)
Dolby Digital (DD) - format used for DVDs, 5.1-channel surround sound. (mandatory)
Dolby Digital Plus (DD+) - extension of Dolby Digital, 7.1-channel surround sound. (optional)
Dolby TrueHD - lossless encoding of up to 8 channels of audio. (optional)
DTS Digital Surround - format used for DVDs, 5.1-channel surround sound. (mandatory)
DTS-HD High Resolution Audio - extension of DTS, 7.1-channel surround sound. (optional)
DTS-HD Master Audio - lossless encoding of up to 8 channels of audio. (optional)


Blu-ray Disc for Movie Distribution

Introduction
Most people know about Blu-ray Disc's basic features: It can store 25 GB (single layer) or 50 GB (dual layer) on a single-sided disc - about 5 to 10 times the capacity of DVD. As a result, Blu-ray Disc supports the highest quality HD video available in the industry (up to 1920 x 1080 at 40 Mbit/sec). Large capacity means no compromise on video quality. Furthermore, a Blu-ray Disc has the same familiar size and look as DVD, allowing for compatibility with existing discs.

Compatibility across full family
Blu-ray Disc Rewritable (BD-RE) and related video specifications were first defined in 2003. The Blu-ray Disc ROM format for movie distribution is fully based on this specification when it was defined in 2004. As a result, users can play home-recorded discs on all of their Blu-ray Disc equipment; there are no playback compatibility issues as with rewritable DVD formats. The Video Distribution format was widely expanded to offer content producers a full range of additional features unavailable in the home recording format.

Video highlights
The BD-ROM format for movie distribution supports three highly advanced video codecs, including MPEG-2, so an author can choose the most suitable one for a particular application. All codecs are industry standards, meaning easy integration with existing authoring tools, and choice from wide range of encoding solutions. All consumer video resolutions are available:
- 1920 x 1080 HD (50i, 60i and 24p)
- 1280 x 720 HD (50p, 60p and 24p)
- 720 x 576/480 SD (50i or 60i)

Audio highlights
The BD-ROM format for movie distribution supports various advanced audio codecs, so an author can choose the most suitable for a particular application. The high capacity and data rate of Blu-ray Disc allow for extreme high quality audio in up to 8 channels to accompany High Definition video. Final audio specifications include DTS (core format), Dolby Digital AC-3 and LPCM (up to 96/24) . Optionally, the format might support DTS++ and LPCM 192/24 7.1.

Exceed DVD feature set
The Blu-ray Disc movie distribution format was designed to offer all of the features and the familiar user interface model of DVD-Video. However, content producers have a wide array of new and extended features to be included in a Blu-ray Disc title. For this, two profiles are available:

"HDMV" mode
Offers all features of DVD-Video and more. The authoring process is in line with DVD-Video creation.

"BD-J" mode
Offers unparalleled flexibility and features, because it is based on the Java runtime environment. It allows for extensive interactive applications, and offers Internet connectivity.


"HDMV" mode

Introduction
"HDMV" mode was designed to offer exciting new features, while keeping the authoring process as simple as possible. It streamlines the production of both Blu-ray Disc as well as DVD-Video titles, as the production process incorporates many identical phases. It offers improved navigational and menu features, improved graphics and animation, improved subtitling support and new features like browsable slideshows.

"Out-of-mux" reading
Unlike DVD-Video, the Blu-ray Disc format allows for data to be read from a different location on the disc, while uninterruptedly decoding and playing back video. This allows the system to call up menus, overlay graphics, pictures, button sounds, etc. at user request without stopping playback. Some examples of possibilities will be explained later.

Graphic planes
Two individual, full HD resolution (1920x1080) graphics planes are available, on top of the HD video plane. One plane is assigned to video-related, frame accurate graphics (like subtitles), and the other plane is assigned to interactive graphical elements, such as buttons or menus. For both planes, various wipes, fades and scroll effects are available, for example to present a menu.

Button graphics
Menu buttons can have three different states: Normal, Active and Selected. They support 256 color full-resolution graphics and animation, thereby greatly surpassing the capabilities of DVD-Video. Buttons can be called and removed during video playback, there is no need to return to a "menu screen".

Button sounds
Button sounds can be loaded into memory of the Blu-ray Disc player. When a user highlights or selects a menu option, the sound can be played (such as a voice-over explaining the highlighted menu choice, or button clicks). These button sounds can even be mixed with the running audio from the movie or menu.

Multi-page menus
In DVD-Video, playback was interrupted each time a new menu screen is called. Due to Blu-ray Disc's ability to read data from the disc without interrupting the current audio/video stream, a menu can consist of several pages. Users will be able to browse through the menu pages or select different menu paths, while the audio and video remain playing in the background.

User-browsable slideshows
In DVD-Video, user browsable slideshows were not possible with uninterrupted audio. As a result of Blu-ray Disc's ability to read data from the disc without interrupting the current audio/video stream, users can browse through various still pictures while the audio remains playing. This applies not only to forward and backward selecting: A user can make different selections on what picture to view (or select from a screen presented with thumbnail images) while the audio remains playing.

Subtitles
In DVD-Video, subtitles were stored in the audio/video stream, and therefore they had limitations on the number of languages and display styles. Again, it is due to Blu-ray Disc's ability to read data from the disc without interrupting the current audio/video stream, that subtitles can be stored independently on the disc. A user may select different font styles, sizes and colors for the subtitles, or location on screen, depending on the disc's offerings. Subtitles can be animated, scrolled or faded in and out.


"BD-J" mode

Introduction
"BD-J" mode was designed to offer the content provider almost unlimited functionality when creating interactive titles. It is based on Java 2 Micro Edition, so programmers will quickly be familiar with the programming environment for BD-J. Every Blu-ray Disc player will be equipped with a Java interpreter, so that it is capable of running discs authored in BD-J mode.

Graphical User Interface
In BD-J mode, the author has complete freedom in designing the user interface. The interface is controllable by using standard navigational buttons on the remote. It can display up to 32-bit dynamically generated graphics (millions of colors), and it supports the display of pictures in standard file formats like JPEG, PNG, etc.

Playback control
The BD-J application can act as the sole interface to the disc's contents (thus replacing the player's on-screen controls as with discs authored in HDMV mode). The BD-J environment offers all of the playback features of HDMV mode, including the selection of subtitle, trick play modes, angles, etc. Video can even be scaled dynamically, so that it can be played in a small size in the corner of a menu, and resume full screen when a selection is made.

Storage
A Blu-ray Disc player might contain a small amount of non-volatile system storage (flash memory). This system storage can be used to store game scores, bookmarks, favorites from a disc, training course results, etc. As a manufacturer's option, a Blu-ray Disc player may also be equipped with Local Storage (hard disk, to allow large amounts of data like audio/video to be stored).

Internet connection
The BD-J system supports basic Internet protocols like TCP/IP and HTTP. The player may connect to the disc publisher's web site to unlock certain content on the disc (after certain conditions, like payment, are met), or dynamically display certain info (like theater playing schedules for a movie) on the screen. The disc's program may be extended with JPEG pictures or audio fragments downloaded from the Internet, or it can even stream full new audio/visual content to Local Storage.

Conclusion
The Blu-ray Disc format for Movie Distribution offers two flexible profiles for the creation of titles. It was designed to allow for the streamlined development of Blu-ray Disc (HD) and DVD-Video (SD) titles at the same time, if needed. Basic menus and navigation can be identical. However, it also offers many new functions that will benefit both the author (by offering flexible ways of creating disc content), as well as end users (by offering exciting new functionality compared to DVD-Video)

Blu-ray Disc for Video
What is the quality of Blu-ray Disc video?
Blu-ray Disc offers HDTV video quality that far surpasses any other medium or broadcast format available today. With High Definition video with a resolution of up to 1920x1080 and up to a 54 Mbit/sec bandwidth (roughly double that of a normal HDTV broadcast), no other format can match Blu-ray Disc's video quality. Furthermore, due to the overwhelming capacity of a Blu-ray Disc, no tight compression algorithms that may alter the picture quality are required, as with other formats that offer less recording space. Depending on the application, Blu-ray Disc also supports other video formats, including standard definition TV.

How much video will fit on a Blu-ray Disc?
As with DVD, this depends on the decisions on the usage of video bandwidth, the number of audio tracks and other criteria made by the author of the disc. Furthermore, the choice of the used codec also influences playback time. On average, a single-layer disc can hold a High Definition feature of 135 minutes using MPEG-2, with additional room for 2 hours of bonus material in standard definition quality. A double-layer disc even extends these numbers up to 3 hours in HD quality and 9 hours of SD bonus material. Using any of the advanced codecs, these numbers can even be significantly increased.

Do I need a new (HD) TV to use Blu-ray Disc?
No. Pre-recorded Blu-ray Disc titles will play on any standard definition TV set, even if the video was encoded in High Definition. Likewise, a Blu-ray Disc recorder can also record standard definition video, for example from regular TV broadcasts or camcorders. A Blu-ray Disc can store around 10 hours of broadcast quality standard definition video on a single-layer disc, or around 20 hours on a dual-layer disc.

How does Blu-ray Disc region coding work?

Contrary to DVD, the Blu-ray Disc region coding system divides the world into only 3 regions, called regions A, B and C. The usage of region coding on a Blu-ray Disc movie title is a publisher's option. A Blu-ray Disc player will play any movie title that does not have region coding applied, plus all titles of its corresponding region.

Region A:
- North America
- Central America
- South America
- Korea
- Japan
- South East Asia

Region B:
- Europe
- Middle East
- Africa
- Australia
- New Zealand

Region C:
- Russia
- India
- China
- Rest of World


Blu-ray.com - Blu-ray Recorders
Blu-ray.com - Blu-ray Drives
Blu-ray.com - Blu-ray Media

Blu-ray Disc

geeze, Now I feel even dumber... I dug thru all this over the weekend and i'm going to have to do more studying and research... I've got a Blu-ray project coming in a month....

cheers
geo

[Lori]

This information is very helpful. Thank you.