Only Noise (high frequence pulses) on Firewire Interface Motu 828mkII

hello,
i drive Logic 7.2.3 and Wave Burner 1.2 a long time, since yesterday i get only noises and high frequence pulses on my interface.
I Tunes and Quick Time works well whith Motu/Firewire.
Only Logic and Wave Burner does!!
I tried different settings in Logic or in Audio-Midi-Configuration.
Additionaly i get sync problem messages in Logic, also in WB.. the SPL is not moving right!!
Has anybody an idea??
Thanks
Patch

There are lots of posts on this topic - the search function would have turned them up. Try this:
http://discussions.apple.com/thread.jspa?messageID=1477845&#1477845

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Daq 6211 high frequency ouput

HI guys.
Ive got my hands on a USB 6211 daq device. Im trying to generate a high frequency pulse output from one of the 2 counters. The frequency is in the range of 5 to 20 kHz. Can someone tell me if thats possible with this USB 6211 device?. Ive been searching but cant seem to find the answers. Are the counters on this device just for reading HIgh Frequency inputs or can they ouput them also. Any help would be much appreciated.
Solved!
Go to Solution.

That manual has helped a lot. I should have looked at properly in the first place. Thanks .
I have one more question if anybody can help. I'm using the daq 6211 to create a pulse train to control a stepper. The stepper driver takes in a pulse which in turn will turn the stepper and the frequency of these pulses determines the speed.
My problem is that, using the counter examples in labview, I only have an option to create a continous pulse train. I would like top be able to dynamically adjust the frequency while the vi is running and Ideally i want to be able to ramp up the frequency so that the stepper can get to a high speed. If i try to adjust the controls with the example vi's there is no change in frequency unless i stop it and start again with a different frequency.
Has anyone come accross this themselves or am i missing something obvious??.

Bandpass filter amplifies only noise at high frequencies and saturates the opamp output

I have designed a fourth order bandpass filter using the opamps TL081C. It is designed to operate at a center frequency of 40 kHz with a bandwidth of 2 kHz and a gain of -12.5 . I am using an SCB-68 board for data transmission and reception, one channel each for transmitter and receiver. I have grounded the adjacent channels to both the transmitter and the receiver in order to avoid crosstalk. However, at the output, I get a signal which looks like an amplitude modulated signal and is, most probably, noise. The amplitude of this signal is always around 9V, no matter what is the amplitude of the input (even if the input is 0V, the result is the same). If I scale down the whole BPF to operate at a center frequency of around 20 kHz, everything works fine. Therefore, I think the problem lies not in the design of the BPF but the use of the board at higher frequencies. Please reply as soon as possible.

You have done something wrong. You should read the manual for both the SCB-68 and your DAQ card once more. In such cases I often use MAX, and a battery as source. Then a DAQ input is not connected (floating) you will see measurements like you are doing now. Some high level voltage often modulated with mains frequency.
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High Frequency Noise from Headphones Out Jack

I have isolated the following issue to audio that is monitored via the headphones output on the back of my iMac:
I've been encountering a very high-pitched (and at times a cycling through high frequencies) noise when working with headphones to monitor audio. It's not the headphones and the problem is not limited to any single application. The noise is sometimes barely audible, and it comes and goes - but repeatedly and often. Sometimes the frequency of the noise is so high that I can imagine that those with poor or compromised hearing wouldn't even notice it. But the highest frequencies are still detectable - kind of like a faint ringing in the ears or a ringing television set that has its volume turned all the way down.
I've done everything I can imagine to troubleshoot the problem including asking a Mac genius, but nobody seems to have encountered this before. The noise can be triggered off (not masked) by any other incoming audio, but eventually it reappears. I will be editing a file, the noise will appear, and I can literally silence it for the moment by pressing a volume key or playing back a portion of audio.
Anyone else experiencing this issue? Is there a fix? I do a lot of audio engineering and this problem is very distracting. I haven't purchased any outboard gear to monitor with yet, so I don't know if I will hear this noise through an external audio interface or not. Thankfully, this noise isn't being recorded to disk. It appears to be a little monster that is haunting only the playback.

OK, It is very easy to reproduce the problem, just follow this steps:
(1) Insert any headphones to iMac.
(2) Do NOT play any iTunes or any application with sound.
(3) Now, set volume with the volume control keys on your keyboard, set it to the MAX.
(4) You can keep pressing on the increase volume key and every time you hit the key you should hear a sound.
Just as when the sound stop, you will hear a high pitch. Keep repeat step 4 and wait a while, then occasionally you will hear a high pitch sound that will last to longer than 30 seconds. The high pitch will be even easier to reproduce if you alternate before decreasing volume and increase volume by 1 level, each time, wait and you will hear the high pitch too.
This high pitch sound can be heard on external speaker plug into the headphone jacks too.
BTW, I turned OFF WIFI, Bluetooth, removed all USB devices except keyboard and mouse, adjust monitor brightness, they have no effect on the high pitch, it is still there!
I am going to call the Apple Care tomorrow. Will keep you guys posted.
Message was edited by: chienlim

Intel iMac 20' users - any problems with noise on your firewire interfaces?

Someone please tell me that they have a quiet intel iMac 20'/Firewire interface set up, I'm having problems with a lot of digital noise from my Saffire LE & badly am hoping it's 'fixable' or just a faulty unit but more so, praying it's not noise from the iMacs Power supply that will interfere with any firewire interface I get???
I've read many posts about G5's suffering from noisy psu's which was one factor in my choosing an intel iMac & I chose the Saffire on it's great reviews for sound quality, there's no way I can work with the noise I'm getting & I'm pretty much panicking that if it IS the iMac & firewire thats the problem I'll be screwed as i only have 6 pin firewire as a useable option (really don't want to use usb2)
Incidentally, the 'budget' set up I had previously (PC & soundblaster audigy) didnt suffer from any noise like this (at least not that was even audible with the mixer levels at conversational level!) which i guess rules out any blame of this on 'mains hum/interference'?
The noise is there with the saffire's monitor output knob at any setting & with no inputs connected, it;s even worse if i power the saffire with it's external psu instead of off buss power. I'm not using a firewire hub & I've tried switching firewire leads & removing the firewire hard drive i have in the other port in case this was affecting it.
Would really appreciate hearing from anyone with similar troubles or even better anyone thats NOT having this problem with a 20' intel iMac & firewire interface so at least i know there's light (& quiet!) at the end of the tunnel
Adam
Intel iMac 20' Mac OS X (10.4.7)

funnily enough I'm going the other way, I have a rack of external modules/>sampler/fx etc & a crate of looms I've just unplugged as they've not been >turned on in so long!!!
I hear you there!
The noise is there without any inputs at all to the interface, its defiantely >digital though, its like a toned version of the sound of an old telephone >modem.
What I mean, is try disconnecting everything, drives and all. You might even disconnect the interface and crank up the iMacs volume to see if it is there. If it is, you can try the UPS and see if that takes care of it. Then add them in one at a time. All it takes is one cheap power brick in the mix to cause all kinds of trouble (if your using powered monitors, that brick too). Noisy components can actually send noise back into the source power.
I know it can be really frustrating, especially if you have noise coming from more that one source. The key is to strip everything until it is quiet and then add them back in to see where the worst is. Then come up with a plan to try to get rid of the noise.

Regza 40" LCD TV emits high frequency noise

A few months ago I was lucky to win this television, a 106 cm Regza HD LCD.
I was very pleased because I could never have afforded it myself, and I am very happy with its performance generally.
What I find however is that it emits a high frequency noise that triggers tinitus to my hearing. (Tinitus is a high pitched squealing noise heard by people who have damaged their ears, sometimes called ringing in the ears).
I guess my hearing is already sensitive through listening to lots of live music over the years, but I think that this noise is excessive.
I figure that a monitor this size is using a fair amount of power and that maybe the power supply could be overworking itself.
The monitor emits this noise at a consistent level even before the sound from the tv show begins.
I am interested to find out what suggestion forum users may have for reducing this background noise?
The monitor is still under warranty as it is only 6 months old.
I wanted to be prepared to discuss it with Toshiba warranty people if forum users can suggest first of all what could be causing the noise, and secondly, possible ways of addressing the issue.
Thanks petekey4

Hello Pete
As you can see this is Toshiba computer forum and I don't believe you will find many people here who want to discuss about Toshiba LCD TV.
Anyway, if the TV is new and with valid warranty the best option for you is to contact service people and let them check your new TV.

High frequency Labview-Simulink Interface

Hi,
I am building a Simulink and Labview model to work with each other using SIT. However, before I even begin, I would like to know whether this Labview-Simulink interface will or will not be able to run at a relatively high frequency - possibly around 80 Hz? Also, if there is any tips or hints on how to make the program run faster, such as run Simulink and Labview on different computers which are connected via LAN, etc., they are highly appreciated. And if this is of any help, I am using a P4 2.0 GHz with 256 MB of RAM computer.
Thanks and have a nice day!

Lan,
A large part of the speed will depend on the size of the model. The CPU would need to be divided between simulating the model and doing the updates, but since you are looking for a high speed it is probably a simple model. I ran a very simple model and tested to see how often I could receive values from the NI Sink in the model as well as write to a parameter of the model. In this simple setup, I recieved updates from the NI Sink at a rate faster then 1000 Hz (less then 1 ms per update on average), and writing to the parameter happened a bit slower at about every 25 ms. Since it is windows these numbers are only averages and individual iterations will vary.
If you want to provide the model with a stimulus as it is running from an external input (like a DAQ
card) then you will want to use a real-time system. This functionality is also part of SIT, where you use Real-time Workshop to build a dll and example VIs for running the model on a LabVIEW Real-Time target. Building a DLL in this way you can still use the same user interface to view the parameters and values in the model, but the inports and outports of the model will be connected to hardware. In this type of simulation the rates can be significantly faster.
Carl L
National Instruments
www.ni.com/ask

Disabling lower frequencies/bitrates of Firewire audio interface possible?

I have a Phonic Firefly 302 Firewire audio interface. With Audio/Midi setup I set all channels to 24kbit/96khz or 192khz. Problem: after every replugging of the Firewire cable, the unit defaults to 16bit/44khz again.
Is there a way to lock this setting?
Is there a way to fool OSX that other bitrates are not available? Like deleting the 44khz, 48khz, 88khz and 16 bit options from some .kext or plist?

For my Firewire interface, I just opened up Audio/MIDI Setup like you, set my interface as the default in System Settings, then set the Input and Output format for my chosen settings, 44.1/24.
Every time I fire up the interface it remembers my settings. Strange that yours doesn't.

Clipping using Firewire Interface Presonus Firebox

Hello, I'm new to this forum and recently just getting acquainted with using GB 2 and my new Presonus Firebox interface, which I acctually bought from reading the recommendations on this forum. So thanks for the suggestions!
My issue is I'm recording my electric bass and acoustic electric guitar (one at a time) and both instruments are clipping during playback. I can't even hear or monitor my playing during recording.
I've attempted adjusting the levels on both the bass and acoustic guitar AND on the Firebox, from low-mid and high AND also in the System Preferences input volume control. Whether I record with low or mid volumes I 'm clipping.
I'm using inputs in the front, lines 1 & 2 for mic/instrument. I've even tried using the rear inputs 3 & 4 but I don't even get a reading. I must be doing something wrong.
So I wonder if this has anything to do with the kind of cables I'm using for recording. Or my computer's speed and memory. I'm using an iMac G4 from a few years back.
Previously, I was using my old Powerbook G4 for recording, also from a few years back, and I did NOT get any clipping when I recorded my acoustic electric guitar, but I did have to raise the levels on the guitar AND on the Firebox - almost to the max! - to get a decent volume recorded.
Please help! I'm still getting acquainted with all this techy stuff. Thanks in advance
imac Mac OS X (10.3.9)

I can't even hear or monitor my playing during recording.
You should be able to listen to (monitor) the tracks. Double click on the instrument icon on the track and from the monitor pulldown menu select 'on'. Monitoring is off by default.
As far as levels, don't worry about the main level. It can clip in the recording stage. Worry about the individual track levels. You may not be able to get them into the yellow but at least get them as high in the green as possible.
You have a number of gains to play with. My suggestion is to leave the system gain on its default setting. Play with the gains on your guitars (if you have any) and the Presonus. There is usually a track gain and a main gain. Put the main on about 1/2 and then adjust the track gain. If this creates too much background noise do the opposite. Every interface is different so you may need to play with it.
Note that the harder you strum the more volume goes to your mic so in setting up the levels make sure you are strumming as hard as you would in the loudest part of the piece. This will give you your top level.
If you are clipping slightly and occasionally it might be all right. I have had several recording go slightly into the red with no digital clipping. There may be a little wiggle room in Garageband's meters.
The main meter is a different story. All tracks combine together, some frequencies are identical which may boost that frequency more so you end in the red a lot. Most of us use some sort of compression to control this. You want the signal high but don't want it to clip.
The best plugin for beginners is the C3 Multi Band Compressor 1.2. It comes with a mess of presets. The mastering presets are particularly useful for your master track. You can download it here. Follow the directions to install it.
Here is a goofy thing. Garageband defaults to hiding the Master track. This is plain silliness because you need to mess with the Master track to get your final product done. In Garageband press ⌘ +b to see the master track. Double click on the master track and you will see pulldown menus that you can select various plugins. This is where you would use the C3 plugin mentioned above. Plugins used in the master track pulldown menus apply to all tracks.
This is a great page to learn more about garageband basics:
Apple - Support - GarageBand
Kurt

How do I connect and detect a high frquency pulse (0-3500hz) with a variable pulse voltage (0v low,5-10v high), on a SC-2345 connected to a PCI-6224?

How do I connect and detect a high frquency pulse (0-3500hz) with a variable pulse voltage (0v low,5-10v high), on a SC-2345 connected to a PCI-6224? Labview doesn't generate a waveform if I try to graph the signal, and my analog input doesn'y recognize that the signal is even pulsing, only that it has a voltage. Some advice would be much appreciated.
Thanks in advance,
Garrett

Hi Garrett,
It sounds like you want to do Frequency Measurements of a signal that is somewhat digital.
There are two ways you can really do this:
Wire your signal to the counter input and use the Digital Frequency examples (Help > Find Examples > Browse > Hardware Input and Output > DAQmx > Counter Measurements > Digital Frequency)
Continue taking analog measurements and use some sort of logic to determine where you are getting rising and falling edges.
I would recommend doing method #1, but you must determine which one is the best for your system.
If you want to connect to the counter pins then you will find the
terminals located on the side of your SC-2345. If you don't need
to condition your signals then simply wire everything into the proper
pins for your counter (below).
Default NI-DAQmx Counter Terminals (Connector 0)
Terminal
Counter Context (Default)
Motion Encoder Context
Signal Name
37
CTR 0 SRC
CTR 0 A
PFI 8
3
CTR 0 GATE
CTR 0 Z
PFI 9
45
CTR 0 AUX
CTR 0 B
PFI 10
2
CTR 0 OUT
PFI 12
42
CTR 1 SRC
CTR 1 A
PFI 3
41
CTR 1 GATE
CTR 1 Z
PFI 4
46
CTR 1 AUX
CTR 1 B
PFI 11
40
CTR 1 OUT
PFI 13
Otis
Training and Certification
Product Support Engineer
National Instruments

Help Counting high frequency voltage spikes

Equipment: NI USB-6229 (250 kS/sec Analog In., 16 bit DAC, 32 bit Counters and internal clocks <= 80 Mhz)
LabView 14
Problem:
I have an experimental application where I need to count voltage spikes (integer #) caused by electrons hitting a sensor. These spikes can be as frequent as 500,000 Counts/sec. The spikes are not going to be the same voltage everytime, but they will be visible above the noise so I need to allow the user to select a Threshold Voltage that triggers a real count rather than noise spikes.
Attempts:
To count such a high frequency, I deduce that I need to use a Counter Input to read fast enough, HOWEVER, I wasn't able to find a way to set a threshold voltage for a Counter Input because I believe they expect a TTL signal anyways, which I won't have. To set the threshold, I realize that Analog Input reads can be triggered at a selected level which is great but the Analog Input Sampling Rate is only 250kS/sec which won't catch every count in my project.
I have a program that uses the Count Edges channel and it is accurate within 3% of the expected # of counts. I was just testing it with a function generator and the program doesn't count unless the signal's voltage is above 2-3 V which won't work for my application. I will post what I have. Does anybody know of a way to trigger off only at selected Voltage levels using counters, or know of a way to filter through the noise to get real spikes?
Thanks!
Solved!
Go to Solution.
Attachments:
ElectronCountsTest.vi ‏29 KB

Thanks for the reply johnsold. I didn't think to use a comparator but that is good to know that I have that option. I was and still am hoping for some kind of trick to do this programmatically. One other idea that I was playing with is offsetting two or three different Reads on the same signal to read the signal at different times. If this is a possibility, it may be able to double or triple my sample rate to 500k or 750kS / sec. Anyone else have any ideas on this solution?

High-frequencies getting filtered

Hi all, not sure where to ask this, I hope someone can give
me some insight.
When I go to a website that allows me to stream audio and
video from my webcam - it pops up the flash dialogue box to ask my
permission....that's fine - then I choose to route my soundcard's
input to the stream instead of my webcam microphone and it works -
but for some reason, it rolls off the high frequencies, and it
sounds muddy and not clean and tight. With music, all the hi-hats
are gone, and all the sizzle is missing, not great!
I have been trying hard to find any info about HOW flash
grabs the audio, or where I can change the settings as to how it
interfaces with the incoming audio, but I can't find anything
anywhere. The only settings seem to be: line-in, and level - that's
it. No other audio controls of any sort...
...in testing - I tried listening to other people's streams
on the same site, and they have clear audio! I can directly A/B
between mine and others and mine is DEFINITELY filtered down....
I have tested my soundcard input into other recording
software, like ableton live and it is FINE. No filtering is
happening at the soundcard level, and recording into other
applications is pristine and perfect.
So, when I go thru the Flash interface, it is filtered - and
since I'm trying to stream live audio productions, it is
frustrating to say the least.
my setup - xp home, all updated, latest flash version 9....my
soundcard is an external motu-2408-mkII with ASIO drivers and
regular drivers
So, I guess my question is....since my sound input seems to
work fine, but only going through Flash does it get filtered - HOW
do I access the Flash drivers that affect this, Is there ANY info
about how flash interfaces with my audio anywhere....
Any advice will be greatly appreciated !! Thanks all, best
regards,
trevor

You might want to try out "Flash Media Encoder" for streaming
Live input from various devices. It has nice UI for setting Mp3
bitrates and can be used to publish "audio only" stream
also.

Which firewire interface for Final Cut Pro on MacPro?

I currently own a Presonus Firepod, and it seems to work pretty well, but may sell this when I upgrade from a MacMini to a MacPro. I use my Mac for Logic Pro currently.
I anticipate running Final Cut Pro in the near future, so I'm wondering about the ability to read/generate SMPTE time code that the MOTU traveller supplies?
There will be no external mixer, so this interface will connect directly to speakers & headphones. (would be nice to be able to mute speakers and control headphone mix using just the firewire interface).
I don't anticipate needing more than 2 Mic inputs - and my current mike is a Rode NT1. (As the NT1 is not a top-of-the-line mic, unsure if I should place a lot of emphasis on worrying about the pre-ams?)
I would like decent mic preamps, and ability to graphically monitor input and output levels (is this graphic monitoring necessary, or is just the single overload indicator sufficient?).
Which interfaces in $600-$1500 range would work best with Final Cut Pro - and Logic Pro?

I saw the Traveller ads in Videography mentioning
time code, so I presumed it had an advantage over
other interfaces.
I found this in some ad copy:
"""The Travler provides on-board SMPTE time code synchronization features that allow you to slave your Traveler system to SMPTE time code without a synchronizer. The Traveler provides a DSP-driven phase-lock engine with sophisticated filtering that provides fast lockup times and sub-frame accuracy. The included MOTU SMPTE Console software provides a complete set of tools to generate SMPTE for striping, regenerating or slaving other devices to the computer."""
So it likely uses an audio port for timecode I/O. However, unless you have external equipment that you want to slave to Logic Pro, (or have Logic slave to) e.g. a timecode DAT, then this really is not necessary. FCP would not see the Travaller as a "controllable" device.
For Final Cut Pro use, I'd probably shoot in HDV.
When shooting a green screen near my computer (or
maybe even on-site), I wondered if there would be an
advantage to recording the audio thru an interface
like the traveller.
The advantage is that you could capture additional mics and audio tracks with your mac while the cameras shoot. Just slate your takes and sync up later.
Do I actually need an interface that provides time
code, or is that only needed for movie work?
I have SMPTE I/O on my Unitor8, directly accessable in Logic Pro. In 8 years I've probably used it twice to capture audio from some modular digital multitracks. So, unless you have some specific need.....
Maybe it is simpler just to connect to the mic
directly to the camera's input?
Indeed. You should always have camera audio available. Anything else is supplemental. Alternatively, you could use the Travaller or RME, etc. as a "live to camera" mixer in a studio situation. Plug your mics into the interface, create a submix and send it to the camera while simultaneously capturing each mic to an individual track in Logic Pro. There are many possibilities.
G5 QuadPPC2.5, Ram:12.5g, Magma PCIe-PCI expansion Mac OS X (10.4.8) 2X Hammerfall DSP Digiface+Multiface, Digi 002r, 4X UAD-1

How to change the display in the waveform graph to enable higher frequency displays.

I am working with waveforms of very high frequencies upto 2MHz. When the frequency of the waveform is increased above 500Hz, the display in the waveform graph goes blank.
Please help me.
Thanks

Dear Labviewguru,
Thanks a lot for your response.
The problem had been technical, and it has been solved.
Actually, I didn't specify the number of samples while generating the waveform, so only frequencies upto half the default value (1000) were being displayed.
Regards,
Roopa

Using iPod with high frequencies

Hi there - I'm having treatment for tinnitus (high-pitched sound in the ear). There is a treatment where you listen to music interlaced with high frequency sounds which is supposed to stimulate the brain & the cilia inside the ear. They have CDs available but recommend not to use iPod. My question is, if you transferred the CDs using Apple Lossless would the high frequencies come through? Bear in mind some of these frequencies are up to around 14 KHz and there are all sorts of subtleties in the signal. One question is transferring it to the PC using Apple Lossless, the other question is playing it back on the 5G iPod. I'm also thinking of getting a Nano just for this - would it play it back OK? Thanks!
MacBook Pro 15", 60 GB 5G iPod Mac OS X (10.4.4)
14 iBook, 768 RAM, 60GB Mac OS X (10.4.4)

This is only my personal opinion, which can be wrong.
First, human can only listen to 20KHz most, thats why the CD's frequency response only cover in the range of 20-20KHz.
For iPod - the apple mentioned the following
iPod is designed to provide the best-quality sound. iPod has a powerful 60 mW amplifier so it can deliver audio as loud as you want. It also has a 20 Hz to 20 kHz frequency response, which means you can hear distortion-free music at the lowest or highest pitches. Even the iPod headphones are designed to give you the best-quality sound. The earbud-style headphones rely on Neodymium transducers, a rare earth magnet that significantly enhances frequency response and overall sound quality. Most other headphones use aluminum, cobalt, or ceramic drivers; at the same size, the Neodymium driver is five times as powerful. Because the strength of the driver increases the accuracy of the sound, the iPod headphones provide a high-energy listening experience while minimizing distortion.
To ensure CD-quality sound, be sure to use a compression setting of 160 kbit/s (the default setting in iTunes) or higher when you "rip" tracks from your CD into MP3 format, or 128 kbit/s in MP4 format (AAC).
Therefore, if you rip the music from a CD, it does not matter whether it is MP3, AAC or Apple Lossless, it will no reduce the frequency response, however, there is noticeable difference of sound quality from different format, of course with Apple Lossless format being the best, but much bigger size.

Only Noise (high frequence pulses) on Firewire Interface Motu 828mkII

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