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Archived from groups: rec.video.desktop (More info?)

Hello

I've encountered this unusual audio situation. From my various media
catalogs (analog and digital) I've performed routine audio adjustments
within Adobe Audition. Mono to Stereo, Normalize, Hiss removal -prior
to finalizing the project within Encore for a finished DVD.

This current project I'm taking BetaSP and Betacam tapes and running
them from my Sony UVW1200 player into the Canopus ADVC 100. From the
Canopus the firewire feeds into the computer for a Capture within Abobe
Premiere.

I'm painting an outline to show the usual routine. Once I edit the
material I export to NTSC DV high quality 4 MB VBR 2pass (mpeg2). After
a little fiddling with Audition to audio track is imported into Encore.

I've sampled almost every tool within Audition to adjust my current
media, the Betacam materials. Let me attempt to explain: In edit view
(mono track) instead of the usual peaks and valleys appearing in the
wave form I have a thick green band (in all appearances it seems to be
filling in the valleys) with tiny spikes appearing very close to the
edges of the band.

Another way of stating this is that the audio (voice) fills the wave
form pushing away those deep valleys and high peaks. In these lectures
if the speaker were to talk normal the audio would be good. However,
any emotional intonation, a sharp chuckle, creates a spike.

It's like a medical graph with normal breathing...highs and lows.
With this material it's more like flat lining. A wide green band with
tiny spikes. Is there a solution to correct this? How can I run a
capture and collect an audio track with valleys and peaks? Would I need
a mixer between the Sony player and the Canopus to correct the signal?
Is there something basic I'm missing? Please let me know you
thoughts.

Chuck M
 
Archived from groups: rec.video.desktop (More info?)

>From the time line it sounds fairly normal. I would say definitely -not
abnormal.
The intrusive spikes -that should be easily corrected.

First I make a copy of the original audio. Later I compare my
adjustments with the original. The fiddling is with normalize (97%)
Hard limiting, Dynamic Processing. And in other audio copies I expanded
to using filters to see if I could cultivate any improvements.

If I compare these tests with the original I could say that it sounds
'almost' normal but one can tell something is off. Thus the quest to
get the audio as normal as perfect.

The 15 tapes I've transferred so far have this same characteristic
anomaly.
A high tide affect in the audition edit track. - leaving no room for
undulating waves of highs and lows. But I must say I can't say its
distorted.

And your comment: I suspect that the UVW1200 may be feeding an audio
signal outside the normal audio range (below 100 Hz or above 15 kHz).

If this is so this can be corrected with an audio mixer setup between
the UVW 1200 and the Canopus? This I don't know as I'm struggling to
advance with this project and seek to correct this aberration.
 
Archived from groups: rec.video.desktop (More info?)

<parvardigar@yahoo.com> wrote in message
news:1125415367.367676.68220@g14g2000cwa.googlegroups.com...
> Hello
>
> I've encountered this unusual audio situation. From my various media
> catalogs (analog and digital) I've performed routine audio adjustments
> within Adobe Audition. Mono to Stereo, Normalize, Hiss removal -prior
> to finalizing the project within Encore for a finished DVD.
>
> This current project I'm taking BetaSP and Betacam tapes and running
> them from my Sony UVW1200 player into the Canopus ADVC 100. From the
> Canopus the firewire feeds into the computer for a Capture within Abobe
> Premiere.
>
> I'm painting an outline to show the usual routine. Once I edit the
> material I export to NTSC DV high quality 4 MB VBR 2pass (mpeg2). After
> a little fiddling with Audition to audio track is imported into Encore.
>
> I've sampled almost every tool within Audition to adjust my current
> media, the Betacam materials. Let me attempt to explain: In edit view
> (mono track) instead of the usual peaks and valleys appearing in the
> wave form I have a thick green band (in all appearances it seems to be
> filling in the valleys) with tiny spikes appearing very close to the
> edges of the band.
>
> Another way of stating this is that the audio (voice) fills the wave
> form pushing away those deep valleys and high peaks. In these lectures
> if the speaker were to talk normal the audio would be good. However,
> any emotional intonation, a sharp chuckle, creates a spike.
>
> It's like a medical graph with normal breathing...highs and lows.
> With this material it's more like flat lining. A wide green band with
> tiny spikes. Is there a solution to correct this? How can I run a
> capture and collect an audio track with valleys and peaks? Would I need
> a mixer between the Sony player and the Canopus to correct the signal?
> Is there something basic I'm missing? Please let me know you
> thoughts.
>
> Chuck M

Here are a couple of questions. How does the audio sound, when played back
from the time-line? What kind of 'fiddling' are you doing in Audition. Do
all of the BetaSp/BetaCam transfers display the same 'problem'? If yes, and
if the audio sounds very distorted, I suspect that the audio output of the
UVW1200 is set such to provide a signal substantially above the -10 db that
the DV100 expects to see. If the audio played from the time-line before
fiddling is not distorted and sounds like you expect it should, knowing the
originals, then I suspect that the UVW1200 may be feeding an audio signal
outside the normal audio range (below 100 Hz or above 15 kHz).

Steve King
 
Archived from groups: rec.video.desktop (More info?)

parvardigar wrote ...
> I've sampled almost every tool within Audition to adjust my current
> media, the Betacam materials. Let me attempt to explain: In edit view
> (mono track) instead of the usual peaks and valleys appearing in the
> wave form I have a thick green band (in all appearances it seems to be
> filling in the valleys) with tiny spikes appearing very close to the
> edges of the band.
>
> Another way of stating this is that the audio (voice) fills the wave
> form pushing away those deep valleys and high peaks. In these lectures
> if the speaker were to talk normal the audio would be good. However,
> any emotional intonation, a sharp chuckle, creates a spike.
>
> It's like a medical graph with normal breathing...highs and lows.
> With this material it's more like flat lining. A wide green band with
> tiny spikes. Is there a solution to correct this? How can I run a
> capture and collect an audio track with valleys and peaks? Would I need
> a mixer between the Sony player and the Canopus to correct the signal?
> Is there something basic I'm missing? Please let me know you
> thoughts.

Sorry, I'm not following your description. Almost sounds like noisy
hash with a bit of program audio thrown in just for fun. :-(
Suggest posting a few representative seconds of your wav file
in a binaries newsgroup (like news:alt.binaries.sounds.tv or such)
 
Archived from groups: rec.video.desktop (More info?)

sample of audio in
http://www.meherbabalibrary.com/Shack/

as sample

I'll check back tomorrow off to doctor with mother.
 
Archived from groups: rec.video.desktop (More info?)

<parvardigar@yahoo.com> wrote in message
news:1125418593.497552.280810@g47g2000cwa.googlegroups.com...
> >From the time line it sounds fairly normal. I would say definitely -not
> abnormal.
> The intrusive spikes -that should be easily corrected.
>
> First I make a copy of the original audio. Later I compare my
> adjustments with the original. The fiddling is with normalize (97%)
> Hard limiting, Dynamic Processing. And in other audio copies I expanded
> to using filters to see if I could cultivate any improvements.
>
> If I compare these tests with the original I could say that it sounds
> 'almost' normal but one can tell something is off. Thus the quest to
> get the audio as normal as perfect.
>
> The 15 tapes I've transferred so far have this same characteristic
> anomaly.
> A high tide affect in the audition edit track. - leaving no room for
> undulating waves of highs and lows. But I must say I can't say its
> distorted.
>
> And your comment: I suspect that the UVW1200 may be feeding an audio
> signal outside the normal audio range (below 100 Hz or above 15 kHz).
>
> If this is so this can be corrected with an audio mixer setup between
> the UVW 1200 and the Canopus? This I don't know as I'm struggling to
> advance with this project and seek to correct this aberration.

I once transferred from a deck that was outputting an oscillation frequency
above the hearing range --- up about 40kHz IIRC. You could see that if you
switch to Spectral view. There's a toggle on the tool bar. However, if you
are hard limiting and 'dynamic processing', depending on your settings, you
could easily get a wave-form that looks like grass with all blades about the
same height. Richard is right. Put something up that we can hear. Email
me a short clip if you like. Take out the spam block stuff in my email
address. I won't be able to listen until late tonight.

Steve King
 
Archived from groups: rec.video.desktop (More info?)

<parvardigar@yahoo.com> wrote in message
news:1125430189.469619.88180@g43g2000cwa.googlegroups.com...
> sample of audio in
> http://www.meherbabalibrary.com/Shack/
>
> as sample
>
> I'll check back tomorrow off to doctor with mother.


The audio in your sample was either originally recorded using lots of
limiting, which I doubt, or you have used considerable hard limiting as well
as gating to reduce the background noise in between speeches.

If I had this audio, as it is, I would use Audition's parametric equalizer
to roll off the bass. I set the Low shelf cutoff at 18 Hz and rolled off 20
db on the scale at the far left of the freq depiction screen. Sounds like a
lot, but I think it sounds more realistic. Then, I boosted 5.6 db at 2800
Hz using a Q of 1 to increase the presence a little. I also set the master
gain of the parametric equalizer at -2 db to compensate for the mid-range
boost.

Steve King
 
Archived from groups: rec.video.desktop (More info?)

parvardigar wrote ...
> sample of audio in
> http://www.meherbabalibrary.com/Shack/

Looks like ordinary clipping at some point in your chain.
The peaks are chopped of right at -6dBfs.

The voice and the microphone have also teamed up to
create a nasaly-spike that could be tamed with some EQ.

But something is causing that significant clipping and I
would go through the acquisition chain to see if it can
be eliminated. Are you sure the levels are all correct
at the different points in the chain?

If that is the way it is actually recorded on the tape, perhaps
Audition/s "Clip Restoration" "filter" could help(?)
 
Archived from groups: rec.video.desktop (More info?)

Hello

I am infinitely thankful this invaluable assistance in unraveling the
properties of this audio clip and especially the insightful
instructions. In haste, earlier I inadvertently supplied an
'altered' audio track. My apologies. The analysis on the tweaked
audio track could be misleading.

For accuracy in determining the exact nature of this problem please
find a sample of the original audio track at
www.meherbabalibrary.com/Shack. (Origsmp.wav) This is the original
clip. Mono. No Hard Limiting. It is with the actual levels and the
'apparent' clipping. This original clip comes from the master that
originated from lectures recorded on BetaSP tapes in a shoot 10-20
years ago. Since those events decades ago these tapes have been in
storage. No alterations.

I can only assume that the video camera during the shoot may have been
calibrated in a fashion that created this frequency. All of the master
tapes have this same overall audio characteristic. This full collection
is over 100 tapes. Thus if I can collect the precise editing formula
that formula can be applied to the full inventory of tapes.

Later this morning I'll be able to work with Steve King's procedure
using the parametric equalizer tips on the original clip.

Thank you for this priceless assistance.

Chuck M
 
Archived from groups: rec.video.desktop (More info?)

Re: http://www.meherbabalibrary.com/Shack/Origsmp.wav

The sample of the original still looks like plain, ordinary clipping
to me. It is not terribly bad. You could even use the sound track
as-is (depending on how you want to use it and what your resources/
budget is fo the production). I'd certainly try the "filter" in Audition
that is provided for repairing this kind of clipping.

You could ask over in the audio newsgroup: news:rec.audio.pro,
but you would likely get the same answer.
 
Archived from groups: rec.video.desktop (More info?)

<parvardigar@yahoo.com> wrote in message
news:1125500266.471284.257470@z14g2000cwz.googlegroups.com...
> Hello
>
> I am infinitely thankful this invaluable assistance in unraveling the
> properties of this audio clip and especially the insightful
> instructions. In haste, earlier I inadvertently supplied an
> 'altered' audio track. My apologies. The analysis on the tweaked
> audio track could be misleading.
>
> For accuracy in determining the exact nature of this problem please
> find a sample of the original audio track at
> www.meherbabalibrary.com/Shack. (Origsmp.wav) This is the original
> clip. Mono. No Hard Limiting. It is with the actual levels and the
> 'apparent' clipping. This original clip comes from the master that
> originated from lectures recorded on BetaSP tapes in a shoot 10-20
> years ago. Since those events decades ago these tapes have been in
> storage. No alterations.
>
> I can only assume that the video camera during the shoot may have been
> calibrated in a fashion that created this frequency. All of the master
> tapes have this same overall audio characteristic. This full collection
> is over 100 tapes. Thus if I can collect the precise editing formula
> that formula can be applied to the full inventory of tapes.
>
> Later this morning I'll be able to work with Steve King's procedure
> using the parametric equalizer tips on the original clip.
>
> Thank you for this priceless assistance.
>
> Chuck M


Either the audio was recorded as we see it in the wave-form "origsmp.wav"
or, as I suspect, the signal going to the DAC-100 from the Beta deck was so
hot that the clipping took place in the DAC. Otherwise, why would the peaks
all be at essentially 0 dBfs. Try another session of digitizing and turn
down the output of the Beta deck audio. Get a digitized recording where the
peaks are below 0 dBfs, then you can look at the waveform and have some
confidence that you are looking at an accurate picture of what's on tape.
Another way to do this would be to run just the audio through your sound
card line in. Adjust the computer's internal mixer to set levels properly.
I think you have to go through this or a similar process before you go on to
the next step of trying to sweeten the audio with EQ.

One of my reasons for thinking that your problem is caused by the digitizing
audio chain is that I have trouble believing that 100 BetaSP tapes were all
recorded by hammering the camera's limiter in the same way.

Steve King

Steve
 
Archived from groups: rec.video.desktop (More info?)

"Richard Crowley" <richard.7.crowley@intel.com> wrote in message
news:df4l57$chh$1@news01.intel.com...
> Re: http://www.meherbabalibrary.com/Shack/Origsmp.wav
>
> The sample of the original still looks like plain, ordinary clipping
> to me. It is not terribly bad. You could even use the sound track
> as-is (depending on how you want to use it and what your resources/
> budget is fo the production). I'd certainly try the "filter" in Audition
> that is provided for repairing this kind of clipping.
>
> You could ask over in the audio newsgroup: news:rec.audio.pro,
> but you would likely get the same answer.

I think hne needs to find out if the clipping occurred in the camera, when
the original video taping was done, or later in the transfer to computer
process. I suspect the latter.

Steve King
 
Archived from groups: rec.video.desktop (More info?)

You say: Betacam tapes and running them from my Sony UVW1200 player
into the Canopus ADVC 100.

You have a small problem here...the Beta UVW1200 audio output is set to
+4dbu out. I recommend that you go to a local Guitar Center, ask a
salesman for a Beherenger mixer, the cheapest one they have, and tell
him you need something that will accept +4(at 600 ohms balanced) in and
give you -10 out ( at 47k ohms unbalanced). Someone should be able to
set you up with something reasonably priced. Than you run the out from
your mixer into the ADVC100 RCA in.
 
Archived from groups: rec.video.desktop (More info?)

pilgrimprophet wrote ...
> You say: Betacam tapes and running them from my Sony UVW
> 1200 player into the Canopus ADVC 100.
>
> You have a small problem here...the Beta UVW1200 audio output is set to
> +4dbu out. I recommend that you go to a local Guitar Center, ask a
> salesman for a Beherenger mixer, the cheapest one they have, and tell
> him you need something that will accept +4(at 600 ohms balanced) in and
> give you -10 out ( at 47k ohms unbalanced). Someone should be able to
> set you up with something reasonably priced. Than you run the out from
> your mixer into the ADVC100 RCA in.

Or just use an attenuating pad. Like an RDL STP-1 "Stick-On
Universal Attenuator" ($44 at www.markertek.com) or four 20-
cent resistors).
The ADVC-100 is certainly consumer line level (-10dBv)
 
Archived from groups: rec.video.desktop (More info?)

"Richard Crowley" <richard.7.crowley@intel.com> wrote in message
news:df59e9$mav$1@news01.intel.com...
> pilgrimprophet wrote ...
>> You say: Betacam tapes and running them from my Sony UVW
>> 1200 player into the Canopus ADVC 100.
>>
>> You have a small problem here...the Beta UVW1200 audio output is set to
>> +4dbu out. I recommend that you go to a local Guitar Center, ask a
>> salesman for a Beherenger mixer, the cheapest one they have, and tell
>> him you need something that will accept +4(at 600 ohms balanced) in and
>> give you -10 out ( at 47k ohms unbalanced). Someone should be able to
>> set you up with something reasonably priced. Than you run the out from
>> your mixer into the ADVC100 RCA in.
>
> Or just use an attenuating pad. Like an RDL STP-1 "Stick-On
> Universal Attenuator" ($44 at www.markertek.com) or four 20-
> cent resistors).
> The ADVC-100 is certainly consumer line level (-10dBv)

An attenuator would be my first choice, either home-made with those four 20
cent resistors or one of the Shure 10/20/30 dB pads.

Steve King
 
Archived from groups: rec.video.desktop (More info?)

Here's the disadvantage. Computer side there's no issues arriving
at solutions, however, with external audio appliances (referenced in
the thread) I'm inexperienced. I do know that my current setup is
basic. From the Beta player the audio feeds out through left/right XLR
cables and connected to the XLR cables are the RCA adaptors. The XLR
cables plug into the ADVC 100 RCA jacks. From the ADVC 100 the firewire
connects to the computer.

Here's the confusion. I've looked over the comments on Shure
10/20/30 dB pads, attenuating pads, the RDL STP-1 "Stick-On Universal
Attenuator", and finally researching the Behringer found this
inexpensive unit that had two XLR inputs, the UB802.

Excuse my stumbling -the beta player must interface with the ADVC 100
as from this unit the analog signals translate into DV via its firewire
port. Thus I need a device that supports input from the beta players
XLR out audio cable. Wouldn't I need the mixer for the XLR
connections? The information on the RDL STP didn't reveal any XLR
support.


Thanks
 
Archived from groups: rec.video.desktop (More info?)

parvardigar wrote ...
> Here's the disadvantage. Computer side there's no issues arriving
> at solutions, however, with external audio appliances (referenced in
> the thread) I'm inexperienced. I do know that my current setup is
> basic. From the Beta player the audio feeds out through left/right XLR
> cables and connected to the XLR cables are the RCA adaptors. The XLR
> cables plug into the ADVC 100 RCA jacks. From the ADVC 100 the
> firewire
> connects to the computer.
>
> Here's the confusion. I've looked over the comments on Shure
> 10/20/30 dB pads, attenuating pads, the RDL STP-1 "Stick-On Universal
> Attenuator", and finally researching the Behringer found this
> inexpensive unit that had two XLR inputs, the UB802.
>
> Excuse my stumbling -the beta player must interface with the ADVC 100
> as from this unit the analog signals translate into DV via its
> firewire
> port. Thus I need a device that supports input from the beta players
> XLR out audio cable. Wouldn't I need the mixer for the XLR
> connections? The information on the RDL STP didn't reveal any XLR
> support.

The XLR and RCA are just different types of mechanical
connectors. The output level (voltage) from the Beta player
is too "hot" (high voltage) for what the ADVC-100 is
expecting. You need to reduce ("attenuate" or "pad") the
signal level so that it doesn't exceed the capacity o f the
ADVC-100. The likely reason the peaks are clipped is that
the ADVC-100 is simply running out of gas and can't cope
with the high-level signals from the Beta player.

You clearly already have something with female XLR
connectors to plug into the Beta player, and male RCA
connectors to plug into the ADVC-100. You could do
something as simple as plug in one of those Shure
attenuators between your cable and the Beta player.
The RDL unit assumes you are going to connect some
cables with whatever you need at the other end.

If it were me, I would add the two 20-cent resistors
inside the XLR connector shells to knock the +4dB
signal from the Beta down to the -10dB levels that
the ADVC-100 is expecting.
 
Archived from groups: rec.video.desktop (More info?)

"Richard Crowley" <rcrowley@xpr7t.net> wrote in message
news:11he7p5m76etma4@corp.supernews.com...
> parvardigar wrote ...
>> Here's the disadvantage. Computer side there's no issues arriving
>> at solutions, however, with external audio appliances (referenced in
>> the thread) I'm inexperienced. I do know that my current setup is
>> basic. From the Beta player the audio feeds out through left/right XLR
>> cables and connected to the XLR cables are the RCA adaptors. The XLR
>> cables plug into the ADVC 100 RCA jacks. From the ADVC 100 the firewire
>> connects to the computer.
>>
>> Here's the confusion. I've looked over the comments on Shure
>> 10/20/30 dB pads, attenuating pads, the RDL STP-1 "Stick-On Universal
>> Attenuator", and finally researching the Behringer found this
>> inexpensive unit that had two XLR inputs, the UB802.
>>
>> Excuse my stumbling -the beta player must interface with the ADVC 100
>> as from this unit the analog signals translate into DV via its firewire
>> port. Thus I need a device that supports input from the beta players
>> XLR out audio cable. Wouldn't I need the mixer for the XLR
>> connections? The information on the RDL STP didn't reveal any XLR
>> support.
>
> The XLR and RCA are just different types of mechanical
> connectors. The output level (voltage) from the Beta player
> is too "hot" (high voltage) for what the ADVC-100 is
> expecting. You need to reduce ("attenuate" or "pad") the
> signal level so that it doesn't exceed the capacity o f the
> ADVC-100. The likely reason the peaks are clipped is that
> the ADVC-100 is simply running out of gas and can't cope
> with the high-level signals from the Beta player.
>
> You clearly already have something with female XLR
> connectors to plug into the Beta player, and male RCA
> connectors to plug into the ADVC-100. You could do
> something as simple as plug in one of those Shure
> attenuators between your cable and the Beta player.
> The RDL unit assumes you are going to connect some
> cables with whatever you need at the other end.
>
> If it were me, I would add the two 20-cent resistors
> inside the XLR connector shells to knock the +4dB
> signal from the Beta down to the -10dB levels that
> the ADVC-100 is expecting.

Here is an attenuator you can build into each XLR connector. Solder a 1
kOhm resistor across pins 2 and 3 of the XLR. Remove the wire from pin two
of the XLR. Solder a 10 kOhm resistor to pin 2. Solder the wire removed
from pin 2 to the other end of the 10 k resistor. Do the same for the
second XLR connector. Now you have a 20 dB attenuator in both left and
right channels between the BetaSP deck and the ADVC 100. That should take
care of your clipping, which as Richard Crowley said, we believe is now
occuring inside the ADVC 100.

Steve King
 
Archived from groups: rec.video.desktop (More info?)

Concluding comments:

Gentlemen, Thank you. We are magnificently successful.

The non profit group allowed for the investment in the Shure A15AS
15/20/25 Attentuator.
The perfect solution.

Even though the audio is smaller and the byte size larger than the
previous two samples if you wish to examine 'audio test delete'
please do so in the usual location /Shack.

We have a well behaved, normal audio track. I'm very appreciative for
every ones assistance.

Thank you
Chuck M
 
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