Output Sound Level Balance Shifts Unvoluntarily....

Not sure if this would be the correct forum, but...
Occasionally i have noticed that in the system preferences, SOUND, that the balance will shift towards the right or left occasionally and involuntarily.
I listen to music through itunes via airport express, but otherwise i have the headphone jack connected via cable to my receiver for video(audio) etc.
Can anyone tell me why this might happen and how i can prevent it?
thanks
stefanos

This is an ongoing bug in OS X. There are a couple of utilities available to correct this (until Apple do):
http://www.versiontracker.com/dyn/moreinfo/macosx/28723
Matt

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Output sound level from gramophone and from computer should be the same

Dear friends,
Why the output sound level isn't equal if we listen the LP directly from the gramophone (AUX1) or from the computer during the recording the track on the computer?
We are using The E-MU 0404 USB 2.0 Audio/MIDI Interface:
http://ixbtlabs.com/articles2/proaudio/emu-0404-usb.html
http://www.emu.com/products/product.asp?product=10447&nav=features
and we have connected E-MU with the (gramohone) speakers through the headphone output on the E-MU and turn the headphone level to the maximum, but the sound level is still lower from the same speakers.
If we compare the track after the recording with the LP on the gramphone is the level of the sound the same/equal.
So we need to have all three sound levels equal:
-from the LP form the gramophone
-from the computer during the recording
-from our track on the hard disk after the recording
Thank you for your answer.

function(){return A.apply(null,[this].concat($A(arguments)))}
Connecting loudspeakers to the headphone output on the E-mu sounds like completely the wrong thing to do altogether - you should
only connect headphones to a headphone output, and nothing else at all.
We will buy the cable as you mentioned and connect it in the line level (S/PDIF).
function(){return A.apply(null,[this].concat($A(arguments)))}
Almost certainly you will need to adjust the monitoring levels in Patchmix to get this the way you want it
Then we will try to adjust the monitoring levels in Patch (I will report if and how did we manage it).
Our equiptment is:
·       turntable Stabi Kuzma audio komponents ( http://www.kuzma.si/AmplioCMS2/public/EnterPublic.cms2?langId=60&request_locale=en_US )
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Any automation for output sound level conflict?

I just upgraded my home computer to a shiny new(er) G5 iMac, running 10.5.5. I don't know which of those are the necessary conditions for a change in output sound level behavior that I'm trying to tame. I have generally kept my cordless headphones permanently plugged in, rendering the old machine (G3 iMac, 10.4.11) silent to the outer (quiet) world of my house, including when it does a scheduled startup in the morning with some people still asleep. This worked well until the new machine arrived. Now the machine is still silent (outside of the headphones) all the time except when it powers up -- the startup chord now plays through the speakers regardless of a plug in the output jack. The problem comes with the output sound level. For headphones, the overall sound level needs to be at maximum. But the overall sound level seems to be the only volume control on the startup tones, and it is then at wherever it was set when the machine was last shut down. So if I forget to run the sound level down to 1 or zero before shutting down, the startup blasts everyone out of bed! So: I now have a sticky note on the iMac reminding me to turn down the sound at shutdown, which will usually work. My question: is there a more elegant way? Is there some way to automate the turning down at logout time, say? I'm inexperienced at scripting, but happy to learn. Any and all suggestions appreciated!

There were no responses to my problem, so I will post a resolution. I dug a little, and found there are several add-ons out there for this problem.
One is http://www5e.biglobe.ne.jp/~arcana/software.en.html, which has the elegant interface of an added system preference pane. However, it is in a beta release, and temporary administrator authentication does not work. As far as I can tell, in order to get it to work, you have to login and run with administrator privileges, never a good idea. Maybe someday this app will work.
But I hit paydirt with an app called Psst, http://www.satsumac.com/Psst.php. This does just what I wanted. The interface isn't as elegant; you just run the app and make your settings, then quit. It leaves a daemon process running at every startup that does just what you want: it saves the global audio volume and resets the volume to the Psst setting at shutdown. Then at the following startup, the startup chime sounds at whatever the Psst setting was, then the audio volume is set back to the saved value. You can just forget it's there, having once set startup volume level.
Too bad that Apple didn't build this in the latest Mac OS, but Psst seems to fill the gap.

Sound Level

I have connected my Apple TV to seperate speakers. (Speakers from Bang & Olufson with buildt-in amplifiers, Beosound 4000). Im using my Iphone as remote control. Dont have TV connected, am only interested in sound.
But the output sound level from the Apple TV is way to high. Cant turn the volume down enough. It goes from OFF og loud, just with one step on the volume bar on the remote.
Is there a fix to this problem ?
And please, a fix that does not involve buying other speakers.

Are you connected via analogue cables or optical?
'one step' on which remote, the B&O remote?
I would not expect AppleTV to output on analogue at a higher level than any standard line level device - you're not connected to a high sensitivity input on the B&O such as a phono or mic input are you?
Far better to have the AppleTV output it's music at a standard level and to control the volume on teh B&O unit in my opinion.
AppleTV only recently had volume control for music using the white remote up/down arrows - seems to work ok but I had my reservations about it reducing sound quality in the past.

Sound level at 100% after restart

The output sound level on my Mac after restart is at 100%. It is extremely annoying because I use headphones a lot and a pair of amplified speakers (when not using my headphones). Just some minutes ago it almost destroyed my ears. I heat it!
Does anybody out there can please help me to solve this problem?
Thanks in advance.

Zap the coreaudio process in Activity monitor, like this:
1. Open Activity Monitor.app (Applications > Utilities > Activity Monitor.app)
2. Select ‘All Processes’ from the drop down menu next to the search bar (called ‘Filter:’) and type in ‘coreaudio’.
3. Select the process name and click the ‘Quit Process’ button at the top. Chose ‘Quit’ or ‘Force Quit’ from the resulting dialogue box.
The coreaudio process will automatically restart itself (if you look closely at the PID number in Activity Monitor you’ll notice it changes after you hit ‘Quit’) and your sound problems should be solved.
If that doesn't work, return soundprefs to default like this:
4. Open Terminal.app (Applications/Utilities/Terminal.app) and copy/paste this command into the Terminal window
rm /Library/Preferences/com.apple.soundpref.plist
then press ‘return’ on your keyboard.
5. Restart your mac and test.

Help with timing, input from Daq, output sound

Hi
I am a student member of OSA, working on a laser listener project to be used in examples for high schools students. It is a pretty old and simple experiment but something I think students would be into. {any suggestions for other experiments anyone might have I would love to hear}
I read a voltage from a Daq off a reciever circuit, that signal is noisey so I filter it for the human voice range, 60Hz - 2000Hz. Then that filtered signal goes to the play waveform express VI. It works but the snag is I keep getting a "beeping" in the output sound, I believe this is from the loop cycling. I have thought of something like a master/slave loops, storing the data in an array then waiting a sec or two and playing the sound from this data so I dont have to wait on the Daq to acquire new data. Any help or suggestions are greatly apprciated.
This is a rough version sorry about the mess. I think it should also be noted that if the "Time Duration" is larger that 0.02 then that makes the number of samples larger than what the Daq can handle.
Thank you very much in advance for all of the help and your time.
Jason
Attachments:
OSA example.vi ‏42 KB

Hi Jason,
I took a look at what is happening in the play waveform express VI and the issue may be related to starting/stopping the sound card every time the loop iterates, similar to what I suggested with the DAQmx VIs in my previous post. To look into the code behind an express VI, you should copy the express VI to another section of your code or to another VI completely, because once you show the block diagram for it, you will not be able to use the express VI configuration dialog anymore for that instance of the VI. Once you copy the play waveform express VI, right click on the copy and select "Open Front Panel." Then, navigate to the block diagram and keep opening the subVIs until you find the "Simple Write" VI (see below for a screenshot of this VI). Here, you will see that there is a "Sound Output configure" VI as well as a "Sound Output Clear" VI. Since these are within the while loop of your top level VI, the "beeping" in your output may be caused by the constant configuring and stopping of the sound card with these VIs. What I suggest is that you use the code in the express VI as an example to code your own sound output vi that is configured once outside the while loop and stopped once after the while loop. Hope this helps!

HT5517 How can I adjust sound level when streaming radio from my Apple TV through Airplay speakers?

I have a 2nd Gen Apple TV with an airport express set up as airplay speakers. I can stream the new radio through using the Airplay speakers pretty well. The problem is that once I've got it going, I cannot adjust the sound level without laboriously backing out of the radio menu, going back to the top level menu then drilling down into the Airplay settings to raise or lower the speaker's sound level.
Is there an easier way to do this?

go to system preferences > Sound > Output. then change the output from "built-in" to HDMI.
Maher

My pc sound level is very low on playback from dvd and most websites even though its at 100 percent

the playback level on dvd and most video streams is very low.(youtube seems ok)This is my second pavillion g7 w/ windows 7 .The first had a defective fan but the sound level was ok and there was a graphic eq section for sound control with other output options.I've yet to find that area on this pc in the month I,ve had it even though their identical computers.I,m at a loss for the cause of the problem.Headphones work fine but are annoying.

i use professional studio monitors to monitor the audio
the issue , is if i turn up the audio it will clip on premieres main output. so i cant turn it up any more.
the specific moment i took the screenshot the audio volume was at -18db, but even if it was up at 0db it would not even reach half volume on my computers main output
this is the case regardless of the nature of the audio, be it stock music, dialogue, or sound effects maybe there is a hidden master output that im unaware of

Labview Synthesiser: Problem outputting sound via MacBook soundcard

Hi all!
First time posting here, hopefully i'm in the right section of the forum. Anyway, i'm a part time student and currently studying a module on Labview in university. As a mini project i decided to build a synth using Labview. I've it mostly built but i'm having a bit of trouble outputting sound to the soundcard. I'm also having a bit of trouble getting the waveforms to play for a longer time period.
I was sort of copying the setup of one of the example VIs (generating sound vi i think) and another vi i found online but i can't seem to get mine to work using my synth design. I've two problems, one is that the waveform only plays for a very short time but the main problem is that i'm getting an error (error 4803) saying the soundcard cannot accomodate the specified configuration but as far as i can see my setup is more or less the same as the generating sound vi (which works on my fine macbook). Obviously i'm missing something so i decided to come on here and ask for help.
I'm guessing the datatype connected to my sound output configure vi could be causing a problem since it has a red dot on the input terminal. Any suggestions on how i should fix this?
I've my vi attached. Any help would be appreciated!
Cheers!
Edit: I've already fixed the error 4803 problem. Had to change the input to the sound output configure sub vi. Now i just have to figure out how to get the sound to play for longer. Any ideas anyone?
Solved!
Go to Solution.
Attachments:
LabVIEWSynth.vi ‏94 KB

OK. You have several problems.
The cluster order in your Format cluster is Rate, Bits, Channels while the order in the "sound format" cluster on Sound Output Configure.vi is sample rate (S/s), number of channels, bits per sample. LabVIEW connects clusters according to the cluster order. How to avoid this: Pop up on the sound format conpane terminal on the Sound Output Configure.vi icon on the block diagram and choose Create Control. That will put a control with the same names and cluster order on the front panel. You can edit the names if you wish as long as you do not change the cluster order. The alternate is to Unbundle the data from your cluster control and then bundle it to the input cluster. I show this in the modification of your VI attached.
The VI does not respond to the Stop button until the event structure executes, which only happens when a key is pressed. Fix: Add an event case for the Stop Value Changed event. I show this in the modification of your VI attached.
The VI does not recognize changes in Octave, Amplitude, Osc Select, or Filter Frequency until the second keypress after changing any of these controls. Why? Dataflow. Those controls are probably read within microseconds after an iteration of the for loop starts. They are not read again until the next iteration regardless of when or how many times they are changed. The loop will not iterate until the Event structure completes, which only happens when a key is pressed. The Fix: Event cases for Value Changes on those controls. Note that this does not work because now there is no defined frequency. So, you also need some shift registers. Because of the problems mentioned, I did not put this into the modified VI.
Next, the event structure freezes the front panel until the code inside has completed. This becomes quite apparent when you set the duration to 2 seconds and press several keys quickly. The fix to this and the problem in the paragraph above is a parallel loop archtitecture, such as the Producer/Consumer Design Pattern.
Not a problem but a different way of doing something: Use Scale by Power of 2 from the Numeric palette in place of the case structure connected to Octave. I show this in the modification of your VI attached.
Now to your question about tone duration: The duration of a signal generated by the Sine Waveform.vi and the others is determined by the sampling frequency and the number of samples. You are a student so you can do the math. You need to adjust the number of samples because the sampling frequency is fixed.
The modified VI works fine on my iMac.
Lynn
Attachments:
LabVIEWSynth.2.vi ‏89 KB

Playing with loopback: sound level doubled.

I've already bought Mainstage (version 2.2) and the loopback plugin is great! I've just one issue: when I record using loopback there are 2 signals in the output so the volume is higher, but when the loopback is playing there's only 1 output and so the sound level is lower.
This is my situation:
In one channel strip there's EVP88
In another auxiliary channel strip there's the loopback plugin assigned to the Bus1.
In the first channel strip I put a send to the bus1 in which there's the loopback. The "send level" is 0.0. The channel strip levels are all on 0 dB.
The problem is that doing in this way the level of the evp88 is doubled when it pass through the loopback.
Because in the Output I have the evp88 (at -6dB for example), then the sound goes through the bus1 in which there's the loopback channel strip who sends again the sound to the output (again at -6dB).
So the result is that when I play something during recording on loopback the sound I hear is for example at 0dB (because in the output there are 2 signals). When i stop playing and the loopback is playing the recorder track, the sound is less than 0dB (Because there's only the signal from the loopback).
How can i solve this issue?
Thanks guys

Hi
Within the Loopback GUI, there is an option for Monitoring
CCT

Sound level options for voice analysis

I have two questions.
I am analysing voice signals using GRAAS microphone, NI9234, cDAQ 9178 & labview 2011. I am trying to produce a two dimensional graph of the signal with SPL (DB)(y-axis) versus frequency (x-axis). I noticed that the SVT sound level.vi from the sound & vibration toolkit outputs into 4 different values;
1. exponential averaging sound level (dB),
2. Leq sound level (dB),
3. running leq sound level (dB), and
4. peak sound level (dB).
Which of this four should I use.
Is it right that I should use linear weightage for human voice analysis?
Thanks in advance

If you've put the sound level to max & still not so comprehensible audio, it could be a hardware fault. A long shot, this haps to my friend check the small hole for the speaker if there is grit/dirt blocking it.
Knowledge not shared is knowledge wasted!
If you find it helpfull, it's not hard to click the STAR..

How do change the sound level input at the premiere cs6

Hello:
I would like to know how to change the sound level input al the premiere cs6?.
Sincerely, Gabriel Conde

The input sound levels will be determined by the media files you use. The output volume can be adjusted in several ways.
https://helpx.adobe.com/x-productkb/global/offline-help.html

HP Pavillion 15-p046na sound level on playback

Hi,
I've recently bought an HP Pavillion 15 laptop with AMD/Realtek HD audio hardware and Beats Audio.
Windows 8.1operating system. It's brand new out of the box with no updates made.
When I playback mp3 music via Windows Media Player, the volume is getting boosted during quiet parts. This is especially noticeable on tracks which fade out, the sound level gets progressively higher as the song fades out, like it's trying to maintain a constant playback volume. This creates a lot of background noise on music recorded from vinyl, for example.
I've checked it isn't the recording, by transferring the same mp3 file to another device and playing it there - the song plays back exactly as recorded, no problem.
I can't find any options in the audio devices (or beats audio) which might control this behaviour. Is there an equaliser somewhere in the system?
p.s. it behaves the same even with Beats Audio disabled.
p.p.s it behaves the same with internal speakers, headphones and external speakers.
This question was solved.
View Solution.

Hello, @MusicManJohn
Welcome to the HP Forums.
It appears that you're having issues with audio playback. I will try to assist you on this.
As you mention not dong any updates yet, start with those. Complete all Windows updates, then use the HP Support Assistant to make sure all drivers are updated.
From there, read over this document:
No Sound from Speakers (Windows 8)
You have checked a few things from it already, just go over to see if there is anything else.
You can also try these steps (not sure if they will work for this issue, but have for other audio issues)
Open Windows Control Panel > Sound > Under Playback tab, double click on Speakers to open Properties > Go to Advanced tab > From the dropdown, select the highest quality stereo output. By default its on the second lowest setting which is DVD quality > Apply and close.
If that doesn't help, try this:
Move from the Playback tab to the Recording tab. Double click on Stereo Mix to open the properties and repeat the steps done earlier. Go to Advanced tab, select highest quality from the drop down > Apply and close.
Please let me know how this goes.
Thank you for posting on the HP Forums.
I worked on behalf of HP.

I Need to Even out the Sound Level from Tune to tune

This is the problem I'm facing ....
After recording a number of tunes ... and then burning them in iTunes ... I'll find that the ouput level changes from tune to tune.
For example ... one tune's overall sound level is too low compared to the next one.
When I'm mixing I of course try to have the master output as high as possible without clipping.
I have been using Fission to normalize each tune individually ... but
Is there a way to even out the levels from tune to tune. Of course i have "Sound Check" marked in iTunes preferences.
Here is a strange but true story ...
Someone took one of my disks with this problem .... went to a Sony store where they copied it ...
and the sound levels are more even I swear. Is there some sort of software in cd copiers that can do this?
Thanks ......

Here are a couple of articles that give a useful overview of Mastering:
Preparing Your Music For Mastering
Mastering: How The Pros Do It
Other than that, the Normalizing you're doing is the easiest way to even out the relative levels.
WH

Usage of sound level measurement

sir
i am using TES 1352A sound level meter to acquire sound signals.i have installed the sound and vibration toolkit. i am acquring the signal through NI USB 9234 module connected to the sound level meter.Now my problem is how to convert the voltage output from DAQ into decibles. if possible help me with an VI.
thank u

I have used this instrument once. The output is the sound level. It is scaled to a factor x Db per volt output
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

Output Sound Level Balance Shifts Unvoluntarily....

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