[jason]

I've seen questions in this group over the years about
removing reverb in recordings. The answers were not very
encouraging. Now, in the latest Audition release, Adobe
has added a "de-reverb" effect. It isn't a panacea by any
means from the demos they've posted. The result sounds a
bit "flangy" if you crank up the aggressiveness slider,
but it seems to improve recordings made under less than
ideal conditions.

[Ty Ford[_2_]]

On Thursday, October 25, 2018 at 9:33:04 PM UTC-4, Jason wrote:
I've seen questions in this group over the years about
removing reverb in recordings. The answers were not very
encouraging. Now, in the latest Audition release, Adobe
has added a "de-reverb" effect. It isn't a panacea by any
means from the demos they've posted. The result sounds a
bit "flangy" if you crank up the aggressiveness slider,
but it seems to improve recordings made under less than
ideal conditions.

I have a izotope RX 6 advanced de-verb plugin and find that it works on some phone recordings that somehow get early reflections. Not all, but some. I have produced about 180 half-hour radio interview shows that are the resultant edit of phone conversations. DE-verb came to market and I tried it once. Wow! It did help on some files. I had never thought about "reverb" on a phone call, but after that, I could hear the problem with new files and 99% of the time de-verb tightened up the file. You may have to jiggle with the settings to get the best out of it, but those adjustments are pretty easy.

The only issue is that processing the file in Pro Tools creates latency and you have to allow some extra time on the back end of the file (set the cursor past the end of the file) of it'll chop off the last moments of audio.

[polymod]

"Ty Ford" wrote in message
...

On Thursday, October 25, 2018 at 9:33:04 PM UTC-4, Jason wrote:
I've seen questions in this group over the years about
removing reverb in recordings. The answers were not very
encouraging. Now, in the latest Audition release, Adobe
has added a "de-reverb" effect. It isn't a panacea by any
means from the demos they've posted. The result sounds a
bit "flangy" if you crank up the aggressiveness slider,
but it seems to improve recordings made under less than
ideal conditions.

I have a izotope RX 6 advanced de-verb plugin and find that it works on some
phone recordings that somehow get early reflections. Not all, but some. I
have produced about 180 half-hour radio interview shows that are the
resultant edit of phone conversations. DE-verb came to market and I tried it
once. Wow! It did help on some files. I had never thought about "reverb" on
a phone call, but after that, I could hear the problem with new files and
99% of the time de-verb tightened up the file. You may have to jiggle with
the settings to get the best out of it, but those adjustments are pretty
easy.

The only issue is that processing the file in Pro Tools creates latency and
you have to allow some extra time on the back end of the file (set the
cursor past the end of the file) of it'll chop off the last moments of
audio.



I don't use Pro Tools, but I think there may be ways to avoid the latency
issues with Ozone products...

https://www.izotope.com/en/support/k...pensation.html

Poly



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[Mike Rivers[_2_]]

On 10/27/2018 12:23 PM, polymod wrote:
I don't use Pro Tools, but I think there may be ways to avoid the
latency issues with Ozone products...

Latency compensation is just that - compensation. With it enabled, the
recorded track plays back at the right time because it's been adjusted.
But if you're monitoring through a plug-in in real time, there will be
delay. This is true with any digital recording process. It's way we have
"zero latency" input monitoring, and why anything that isn't a direct
HARDWARE connection between input and output doesn't really have zero
latency.

--

For a good time, call http://mikeriversaudio.wordpress.com

[geoff]

On 28/10/2018 6:05 AM, Mike Rivers wrote:
On 10/27/2018 12:23 PM, polymod wrote:
I don't use Pro Tools, but I think there may be ways to avoid the
latency issues with Ozone products...

Latency compensation is just that - compensation. With it enabled, the
recorded track plays back at the right time because it's been adjusted.
But if you're monitoring through a plug-in in real time, there will be
delay. This is true with any digital recording process. It's way we have
"zero latency" input monitoring, and why anything that isn't a direct
HARDWARE connection between input and output doesn't really have zero
latency.


Any decent software the plugin reports its latency to the host
application and the playback is adjusted to suit. Obviously can't work
for 'live' though.

geoff

[Les Cargill[_4_]]

Mike Rivers wrote:
On 10/27/2018 12:23 PM, polymod wrote:
I don't use Pro Tools, but I think there may be ways to avoid the
latency issues with Ozone products...

Latency compensation is just that - compensation. With it enabled, the
recorded track plays back at the right time because it's been adjusted.
But if you're monitoring through a plug-in in real time, there will be
delay. This is true with any digital recording process. It's way we have
"zero latency" input monitoring, and why anything that isn't a direct
HARDWARE connection between input and output doesn't really have zero
latency.


A digital mixer cannot have actual zero latency. It can have
sub-millisecond latency. Well below anything like the Haas limit. I
used to use the Fostex VF16 as a gig mixer and nobody noticed anything.

"Zero latency" for things like the Focusrite Scarlett series just means
"minimum possible latency" - it has a very minimal digital mixer built
in.

--
Les Cargill

[Mike Rivers[_2_]]

On 10/28/2018 1:35 PM, Les Cargill wrote:
A digital mixer cannot have actual zero latency. It can have
sub-millisecond latency. Well below anything like the Haas limit. I
used to use the Fostex VF16 as a gig mixer and nobody noticed anything.

For sound through the air, small amounts of latency are no problem at
all. Think about playing an electric guitar and standing 5 feet from the
amplifier. Sound travels about 1 foot in a millisecond, so, assuming
there's essentially no latency between when you pick a string and when
the sound gets to the loudspeaker - a reasonable assumption given the
speed of electricity - that sound won't reach your ear until 5
milliseconds after you've picked the string.

Where latency is a problem is when there are two paths for a sound to
get to your ear. When you're speaking or singing, the sound of your
voice gets to your ear through two relatively short paths, pretty close
to equal length - one from inside your throat directly up to your
eardrum and the other through the air from your mouth to your ear. But
when you put headphones on, the situation changes. You still have the
"internal" path, but the external path is replaced by what feeds the
headphones. When that's on the order of 1.5 to 3.5 millisonds and pretty
close to the same SPL at your eardrum as the sound through your throat,
you've created a comb filter that puts several notches right in the
speech range and your voice sounds un-natural.

The engineer in the control room or the rest of the band will hear your
voice just fine, it's only the singer who's affected, and it's only when
he's singing. When he hears the playback (if he can perform well,
hearing this odd version of his voice) it'll sound fine to him.

Many people tell me "I've never hear that" and I believe the reason is
that they have the headphone volume enough higher than the internal
volume so that the notches aren't deep enough to create much havoc. If
the singer starts out loud and then asks to be turned up, he'll never
hear it. But it's particularly annoying to spoken word artists who only
want enough volume in the headphones to know that things are working -
which is why direct analog monitoring is usually the setup when there's
someone in the vocal booth.

You can simulate this easily in a DAW. Record a voice track, then copy
it to another track and shift one track by 1 or 2 milliseconds. Set them
to equal volume and listen to one, then both summed.

"Zero latency" for things like the Focusrite Scarlett series just means
"minimum possible latency" - it has a very minimal digital mixer built
in.

I call that "Zero latency for large values of zero."


--

For a good time, call http://mikeriversaudio.wordpress.com

[Ralph Barone[_3_]]

Les Cargill wrote:
Mike Rivers wrote:
On 10/27/2018 12:23 PM, polymod wrote:
I don't use Pro Tools, but I think there may be ways to avoid the
latency issues with Ozone products...

Latency compensation is just that - compensation. With it enabled, the
recorded track plays back at the right time because it's been adjusted.
But if you're monitoring through a plug-in in real time, there will be
delay. This is true with any digital recording process. It's way we have
"zero latency" input monitoring, and why anything that isn't a direct
HARDWARE connection between input and output doesn't really have zero
latency.


A digital mixer cannot have actual zero latency. It can have
sub-millisecond latency. Well below anything like the Haas limit. I
used to use the Fostex VF16 as a gig mixer and nobody noticed anything.

"Zero latency" for things like the Focusrite Scarlett series just means
"minimum possible latency" - it has a very minimal digital mixer built
in.


And if you want to get uselessly pedantic, even analog has latency.