[QLab] M-Audio Firewire 410

[Panos Couros]

Hi all

Im new here and posting this to get some further advice to a problem that I have already reported to figure53 support.

The problem that I am facing is that Qlab is outputting very corrupt and distorted audio through my M-Audio 410 Firewire interface. Similar to bit loss distortion. Both interface and audio are in good condition and work perfectly outside of Qlab.

I am testing from both a G4 powerbook (1.67Ghz + 1GB RAM) - as well as an Intel MacBook (1.83Ghz Core duo + 512 RAM). Both are running the latest system software.

I have bought a few days worth of license for testing (both video and audio - we are running visuals as well). The video runs smoothly, but the audio is totally unworkable.

Has anyone come across similar issues? Could it be slow processor? Lack of RAM or is it the M-Audio Interface that is most likely the culprit?

Any assistance will be most appreciated

Thanks

Panos

____________________________________

Sydney - Australia

http://panos.theatreink.com.au

[Paul Gotch]

On Mon, Jun 29, 2009 at 11:48:10PM +1000, Panos Couros wrote:
> Has anyone come across similar issues? Could it be slow processor? Lack
> of RAM or is it the M-Audio Interface that is most likely the culprit?

These M-Audio interfaces and their drivers have been a persistent source
of trouble. Some people have managed to get them working acceptably but
it's voodoo and no one really has something which works all the time.

Unfortunately there are a lot of them about because they are cheap :/

Look in the list archives for tales of woe.

Interfaces which have very high quality drivers and support (and a
price tag to match) include:
- Metric Halo
- RME

Interfaces which are of acceptable quality (some people have had
problems but they are cheaper) include:
- MOTU

-p
--
Paul Gotch
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[Jeremy Lee]

Panos,

The FW410 is the #1 culprit of audio problems on this list.  Look to the archives for discussions.

Generally accepted interfaces that work well are made by RME, MOTU and Metric Halo.  Some people have success with the M-Audio and Presonus gear, but it seems more spotty.

And I would say that a Macbook with 512M ram is woefully inadequate.  Just OS 10.5.x takes more ram than that to work without going deep into virtual memory.  And the PowerBook may not work very well at all with QLab v2- it worked well with v1, but the new version needs more resources than may be available on the PB.

Best,

Jeremy

________________________________________________________
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-- 

Jeremy Lee

    Sound Designer, NYC - USA 829

        http://www.jjlee.com

[Jon Ares]

On Mon, Jun 29, 2009 at 8:35 AM, Jeremy Lee<jerem...@jjlee.com> wrote:
> The FW410 is the #1 culprit of audio problems on this list.  Look to the
> archives for discussions.
> Generally accepted interfaces that work well are made by RME, MOTU and
> Metric Halo.  Some people have success with the M-Audio and Presonus gear,
> but it seems more spotty.

In my experience, the Edirol FA-101 that I use has never given me a
lick of trouble, and is very affordable... it's been heavily used for
the last month+ on several shows. Not using it for anything fancy -
just getting audio out of the Mac Mini it's attached to.

--
Jon Ares
www.arescreative.com

[Matt Otto]

I use a 410 with my QLab system and I do not like it either but I have yet to run into this problem. It sounds more like to me that your computer is very much under powered to run video and audio on the same machine. I recommend heading to crucial.com and maxing out the RAM on your intel macbook.

Matt Otto
(609) 707-2978

Off List: motto...@gmail.com
http://www.mattottosound.com

[Dave Tosti-Lane]

I'm sure Chris already mentioned this to you, but are you running any compressed format audio? Mp3 conversion has to happen on the fly when Qlab is playing one, and that can bring the system to its knees pretty quickly. I've had students build shows and complain of stuttering and glitching, only to discover that the problem was one mp3 file amongst their playback sequences. Converted that file to aiff or wave, and playback of the same sequence is fine.

Perhaps this might be especially problematic with such thin RAM resources.

Hi all

Thanks

Panos

____________________________________

Sydney - Australia

http://panos.theatreink.com.au

[Mark Valenzuela]

> The problem that I am facing is that Qlab is outputting very corrupt
> and distorted audio through my M-Audio 410 Firewire interface.
> Similar to bit loss distortion. Both interface and audio are in good
> condition and work perfectly outside of Qlab.
>

I have uses a lot of these interfaces as they are the most common one
around theatres in Philly. I have experienced what you're talking
about, and do think it's a problem with the interface, although when
it has happened to me, sound from any application will be distorted
and not just sound from QLab. A restart of the device, or a power
cycle of the computer seems to do the trick usually. However, this has
never happened to me during a random time. It always happens either
when the computer and interface are being turned on and first making
the connection, or after the computer has been woken up from sleep.
It's happened to me on my laptop when the power cable has been pulled
accidentally and it's been forced to use battery power for a minute or
two.

The 410 is VERY finicky. DO NOT let your machine sleep when the 410 is
plugged in. It also has problems when it's not getting the power it
thinks it wants. You may try using the power adapter if you have been
powering it just through the FireWire cable. The instructions say not
to, but I have done this before with no issue, nor have I heard about
anyone having an issue doing this. Make sure you have the proper
driver for your OS, and you might do an uninstall/reinstall of that
driver just in case.

And you DEFINITELY need more RAM.

Good Luck!

Mark

[panoracle]

Thanks for your comments everyone. Ive temporarily got around the
problem by using the 2 x USB interfaces that I already have, plus the
audio from my laptop. Not ideal by any stretch, but as a solution to
my problem it works for the time being.

I am running 24 bit sound for all. I'm finding the MacBook working
very smoothly, running 6 channels of audio plus video.

I have the opportunity to buy a very affordable interface that will do
the trick for the upcoming show Im designing. It is a Terratec Phase
26 and its drivers are just Core Audio. Does anyone here have any
experience with this interface?

The 410 just did not work regardless of everything I tried (including
all of the suggestions you all offered), so I guess that I'm facing
buying a new trusted interface.

cheers

Panos

[John Leonard]

Panos,

Is your source material 24 bit, I wonder? I came across this recently, when
all the source material for the show was 16 bit (CD tracks), but the
designer insisted on converting everything to 24 bit, thus increasing the
file size, but achieving nothing else.

Just wondering if this is a common procedure and why it might have come
about.

John

On 01/07/2009 15:02, "panoracle" <pa...@iinet.net.au> wrote:

> I am running 24 bit sound for all

[panoracle]

HI John,

Yes almost all of my source material is 24 bit, with the exception of
a few moments of extant music that Im referencing. I'm creating mainly
in Protools, with a bit of Logic - using software instruments and
field recordings (recorded on my Fostex FR2Le in 96bit).

I cant think why someone would upscale like that ?

cheers

Panos

> Is your source material 24 bit, I wonder?

[Andy Dolph]

On Wed, Jul 1, 2009 at 11:10 AM, panoracle <pa...@iinet.net.au> wrote:

I cant think why someone would upscale like that ?

because more bits gives a hire s/n ratio - who cares if there's no data in those bits - that's just a detail..... ;)

</cynicism> 

Andy 

[Michael Keniger]

The only reason to 'upscale' like that is when you are using SFX v5.6, in which audio files at 16bit sound horrible when fading or running at any level below "-15" in the mixer (weird audio artifacts and noise/distortion not present in the oiginal files).  The same audio files converted to 24-bit will sound much better though the windows audio engine.

 

>
> I cant think why someone would upscale like that ?
>
> cheers
>
> Panos
>

---

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[Steven Devino]

Glad you close it with </cynicism> otherwise this could be a long thread.....

Steve Devino

http://graniterockslive.com

603-867-6843

Authorized Dealer For

Metric Halo

APB Dynasonics Consoles

Countryman and Associates

[John Leonard]

Totally bizarre...

John

On 01/07/2009 16:47, "Michael Keniger" <reg...@hotmail.com> wrote:

[Jeremy Lee]

John,

This was a necessity in the days of SFX.  16 bit audio through the SFX engine sounded horrific if you ran it through the mixer at less than -20 (think ambiances, etc).  24 bit audio at underscore levels only sounded semi-bad.  The extra bits forced the SFX audio engine to do the math at 24 bits instead of 16, and there was a lot less truncation/ aliasing noise that way.

I would put good money on the assumption that this is completely unnecessary in QLab.  Is that correct?  Everything is likely pumped through a 32bit floating point mixer regardless of the source material.

Jeremy

On Jul 1, 2009, at 10:13 AM, John Leonard wrote:

Panos,

Is your source material 24 bit, I wonder? I came across this recently, when
all the source material for the show was 16 bit (CD tracks), but the
designer insisted on converting everything to 24 bit, thus increasing the
file size, but achieving nothing else.

Just wondering if this is a common procedure and why it might have come
about.

John

On 01/07/2009 15:02, "panoracle" <pa...@iinet.net.au> wrote:

I am running 24 bit sound for all

[John Leonard]

This explains a lot. I still think it’s bizarre. I trust it’s changed in Version 6? (Which I will have to use later this year.)

Thanks for the explanation. I still think it’s bizarre: was this a known work-around? Stated in the manual? One of those ‘cognoscenti only’ things?


Regards,

John

On 01/07/2009 18:19, "Jeremy Lee" <jerem...@jjlee.com> wrote:

[Drew Dalzell]

I'm not sure that it was ever in the manual.  I know that I made a point about talking about it whenever I taught an SFX class.

LCS does a similar thing, when you run files through AudioMove and convert them you bump all of your files to 32 bit, and I believe 48k, no matter what the original bit depth and sample rate.  Audiobox also forced everything to 48k, didn't it?  I've always hated doing a 44.1 to 48k conversion unless I really needed to, especially when the source material is 44.1 off of sound effects CD's.

I know I run most of my ProTools session at 24 bit, since I've got a mixture of my own sound effects recordings at 24 bit, and library sound effects at 16 bit and I'd rather have the whole session at the higher bit rate, and I then export everything for playback at 24bit, which is an old habit from SFX not coping well with mixed bit rates in one workspace.  All of these things are now gone as of 6.x in SFX.  It can handle mixed bit rates, MP3 files, whatever you want to throw at it, all in one show, heck even within one multieffect cue if you want.

Drew

[Michael Keniger]

V6 makes use of ASIO soundcard drivers (v5.6 did not), and does not suffer from the same bit-depth problem.  hooray.

---

This explains a lot. I still think it’s bizarre. I trust it’s changed in Version 6? (Which I will have to use later this year.)

Regards,

John

---

Beyond Hotmail — see what else you can do with Windows Live. Find out more.

[John Leonard]

OK – well I’m a wiser person now. Thanks again. Yes, the Audiobox ran at 48k, but, as Charlie will no doubt remind us, 44.1 is a domestic sample rate. Professional s/r is supposed to be 48k.

John

On 01/07/2009 20:06, "Drew Dalzell" <drew.d...@drewdalzell.com> wrote:

[Andy Dolph]

Yes - but how many of us have commercial SFX libraries at 48k?

Andy

[Richard B. Ingraham]

While others are free to disagree I think it’s being a touch over stated.  Maybe I’m just more cloth eared than my esteemed colleagues but I never heard large amounts of distortion of the audio at levels like -15 to -20.  Maybe that’s because I always normalized my audio files to something close to what I would expect to play them at in the theatre, so I wouldn’t have to run levels at -50 or the like for soft sound cues. 

 

Yes it’s true you would get truncation noise in SFX 5.x at lower levels, but I seem to remember having to run things at a lot lower levels than -20 in order to perceive any quality difference and I don’t think I used 24 bit files on any shows I ever designed.  I almost always use 16 bit, 44.1 KHz files and they sound just fine to my ears.  Just my opinion.

 

This issue in SFX 5.x was due to the fact that the audio engine was part of the operating system and a large part of it was out of the control of Stage Research.  It’s also why some versions of Direct X (Direct Sound to be precise) or the Windows OS itself would change the quality of SFX’s playback.  This was the main reason why Stage Research spent the better part of 2 years writing their own audio engine so we wouldn’t be dependent on the OS to do any of the heavy lifting for us.

 

I think if anyone takes the time (and while I personally didn’t spend a great deal of time doing any comparisons, colleagues I trust have and reported back) to compare the sound of SFX 6 to 5.x you’ll be delightfully surprised.

 

My only guess is that Direct Sound didn’t do floating point math to do it’s mixing calculations, so when you ran sounds at low levels it was essentially just throwing data away.  SFX 6 is an entirely different animal and I have heard no complaints about the audio quality of the mixing engine to date.

 

Hope that helps.

 

Richard Ingraham

[Christopher Ashworth]

On Jul 1, 2009, at 12:19 PM, Jeremy Lee wrote:
>
> I would put good money on the assumption that this is completely
> unnecessary in QLab. Is that correct? Everything is likely pumped
> through a 32bit floating point mixer regardless of the source
> material.

Correct for both version 1 and 2.

-C

[Jake Davis]

On Jul 1, 2009, at 12:06 PM, Drew Dalzell wrote:
>
> LCS does a similar thing, when you run files through AudioMove and
> convert them you bump all of your files to 32 bit, and I believe
> 48k, no matter what the original bit depth and sample rate.

While true that a file must be 48K to play in LCS, there is no bit-
depth requirement. AudioMove will create files ranging from 8-bit
fixed to 32bit float.

That being said, the native format for playback is AIFF 48K/32bit-
float. You can send it 48K/16bit-fixed .WAV files, but it uses a bit
more processing power. Large shows playing back multiple streams
(16+) in sync (especially if chasing timecode) would do well to use
the native format.

On the flip side, it can read the metadata off BWF files and land
those files in a deck relative to the timecode in your DAW. Very
handy indeed (as long as you are expecting that behavior) :-o

FYI

Cheers,
-Jake-

[Panos Couros]

...so, I guess no-one has used a Terratec Phase
26 audio interface??

cheers,

Panos

[Matt Carpenter]

I took a look at the Terratec interface, and from a professional audio standpoint I see three problems;

1. RCA outs.  
2. USB interface.  There is a reason most quality audio interfaces use firewire.  It is higher bandwidth and more reliable.  FW was originally a replacement for SCSI with much attention given to reliability, whereas USB was originally a replacement for the PS2 port for peripherals like your keyboard and mouse.  
3. Terratec’s nuts & bolts are consumer grade AV input devices.  They are not a professional audio company.
   

This may or may not be a good interface, but for those reasons, I would stay away from it.

Just my $0.02
Cheers
Matt


--
Matt Carpenter
Director of Sound
PCPA Theaterfest
so...@pcpa.org
805-928-7731 ex 3140

[Panos Couros]

Thanks for that Matt. You have confirmed my suspicions.

My problem at this point is that Im confined to USB due to the
available computer equipment that I have for this production, so Im
trying to make the best of the situation. Professional independent
theatre in Sydney Australia is a far cry from the wonderfully
resourced industry you seem to have over there, and elsewhere in the
world. Still we keep at it and continue to make silk purses from sows
ears.

Given the above, do you have any other multichannel USB audio
interface recommendations (prefer with balanced TRS outputs)? The
Macbook Im using doesn't have firewire. Im upgrading RAM today to 2GB
just in case (although it works fine at 512 - Im not chancing it).

Would the low RAM be a reason why when I preset a time and fade curve
in Qlab, the timing never seems to be accurate in terms of actual
sound output?

so many questions??

thanks

Panos

> This may or may not be a good interface, but for those reasons, I would stay away from it.

[Mark Valenzuela]

>
> The
> Macbook Im using doesn't have firewire.

Please forgive me if this is a silly question, but are you SURE you
don't have firewire? There has only been one Macbook that doesn't have
firewire, and that was the first generation Unibody one that came out
earlier this year. According to what you've told us, you've got the
first generation Macbook that came out a couple of years ago, which
most certainly does have a firewire port.

With firewire, your options for good interfaces will increase. It's
understandable if money is tight, but with these types of equipment
you definitely get what you pay for.

And yes, increasing the Ram in the machine is absolutely necessary.

Mark

[Panos Couros]

not a silly question - Im just not totally sure about the integrity of
my firewire port (although it probably just is the dreaded 410
mentioned earlier) so I had ruled it out and was investigating the USB
route. However. thanks for pointing that out. I probably still have a
reliable firewire option (just no available other Firewire interfaces
to test my port on) and that is the direction I continue working
towards. USB is currently working so I was investigating that further.
My email above is a little misleading, apologies for that.

cheers
Panos

[Matt Carpenter]

As dreaded as the 410 is, I have had many sucessful show runs on one. In fact my 410 has caused me less trouble than my MOTU Ultralite, costing easily twice the price of the 410. Work by a few strict rules: do not "hot plug" it, do not sleep the computer with it, bus power it (hey it works fine for me) - not external power, power up the computer first, then the 410 after the computer is fully booted, get rid of other drivers like digidesign core audio and MOTU and make sure you have the most recent version of the 410 drivers.

Honestly, my only two real problems with the 410 are its' lack of balanced outputs, and bad S/N ratio.

If you are currently having terrible problems with the unit, though, I wouldn't trust it. I find that they work better with some macs than others.

Good luck,
Matt

________________________________________
From: qlab-b...@lists.figure53.com [qlab-b...@lists.figure53.com] On Behalf Of Panos Couros [pa...@iinet.net.au]
Sent: Wednesday, July 01, 2009 6:24 PM
To: ql...@lists.figure53.com

Subject: Re: [QLab] M-Audio Firewire 410

not a silly question - Im just not totally sure about the integrity of

[John Leonard]

Increasingly, the major libraries are making their material available as 48/24 files on DVD or hard-drive, rather than 44.1/16. All film and TV audio is expected to be delivered at 48k and their market is much bigger than ours, I would think. All my own recordings are at either48/24 or 96/24 and recently I’ve done some surround work at 192/24 just for experimental purposes.

John

On 01/07/2009 21:40, "Andy Dolph" <acd...@gmail.com> wrote:

[Andy Dolph]

Do you find you ca hear much difference between 48, 96 & 192?

Sent from my iPhone

________________________________________________________

[John Leonard]

Sometimes, yes, but it depends entirely on the equipment that I’m using and what I’m listening to. There is some seriously nice high-end kit out there at the moment, like the ULN-8.

John

On 02/07/2009 16:09, "Andy Dolph" <acd...@gmail.com> wrote:

Do you find you can hear much difference between 48, 96 & 192?

[Jeremy Lee]

It was fairly well known if you followed the discussion groups.  But it certainly wasn't in the manual.

[Jeremy Lee]

I believe that LCS can actually deal with anything, as long as it's
48k. You can set up AudioMove to convert to 16 or 24 bit AIFF or WAV
files, and WildTracks deals with them very well.

On Jul 1, 2009, at 3:06 PM, Drew Dalzell wrote:

> I'm not sure that it was ever in the manual. I know that I made a
> point about talking about it whenever I taught an SFX class.
>
> LCS does a similar thing, when you run files through AudioMove and
> convert them you bump all of your files to 32 bit, and I believe
> 48k, no matter what the original bit depth and sample rate

--

Jeremy Lee
Sound Designer, NYC - USA 829
http://www.jjlee.com

[Jeremy Lee]

And can also be a pain in the butt! Nothing like creating a piece of
music in DP with a bar for nothing, bouncing it to WAV, importing to
LCS just to have it plop the file in the timeline where the *bounce*
started! Bit me on a couple of occasions, and was helpful in a couple
as well...

On Jul 1, 2009, at 5:59 PM, Jake Davis wrote:

> On the flip side, it can read the metadata off BWF files and land
> those files in a deck relative to the timecode in your DAW. Very
> handy indeed (as long as you are expecting that behavior) :-o

--

Jeremy Lee
Sound Designer, NYC - USA 829
http://www.jjlee.com