Full Duplex Lost with Wave Files on SB Live! Value (W

Following a clean install upgrade from WinME to WinXP SP2 on a Dell Dimension 400 with an OEM-installed SB Li've! Value (WDM) PCI card onboard, full duplex wave audio (recording one wave file while playing back one or more other wave files) ups and disappears ? ME had it, XP doesn?t. The record side of software mixers I use on the XP OS (Quartz AudioMaster, NCH Swift Sound MixPad/WavePad) fail to acquire or record the wave signal(s) being played back by the mixer side of the application. Stand alone recorders (Roxio Easy Audio Capture, HotKey Sound Recorder) also fail to acquire or record the wave signal(s) being played back by the aforementioned mixers. But if I switch the record input to microphone during the record pass (on those apps that permit such midstream switching), the mic records without incident.
Creative?s Auto Update shows all drivers for the sound card are up to date. Applying Creative?s Knowledge Base Solution ID 645 passes all DXDIAG tests with no problems (text file results available if anyone cares to see them) but the manual test using twin Sound Recorders fails as the second copy (record-designated) Sound Recorder fails to acquire or record a signal from the first copy (playback-designated) Sound Recorder (no error codes display at any time). Clean booting WinXP and running the same record passes doesn?t rectify the problem.
Anyone know what?s going on here? And how to fix it's I?m stumped.
TIA . . .

For benefit of anyone who hits this thread in a search:
FWIW - Following 5-weeks of troubleshooting this problem with Microsoft Tech Support (including a parallel install of the XP SP2 OS), we reached the following conclusion: Since the OEM sound card (Creative Model CT4780) was known to operate in full-duplex mode without incident under my system's prior ME OS but the same card fails to consistently operate in full-duplex mode while running under a virgin parallel install of the XP OS, the source of my full-duplex problem must lie in the files and drivers resident on the XP OS CDROM and dropped to my system by the OS loader/installer. There really isn't any other plausible explanation. Absent its drivers, a sound card's nati've hardware state is either going to be full-duplex or it isn't - it would never be full-duplex sometimes and half-duplex (or simplex) the rest.
The problem was resolved when a retail, nearly identical, sound card (Creative Model SB040) and the drivers and files from its CDROM, were installed on the same system.
Perhaps, somewhere, drivers exist that will permit the CT4780 to operate consistently in full-duplex mode on an XP SP2 OS - but neither Microsoft, nor I, had any success in locating them.

Similar Messages

Having trouble with wav files and sample rates

Hi ,I am having trouble with wav files and sample rates .I have been sent multiple projects on wav as the main instrumental ; I wish to record in 48.000kHz .Now comes the problem.When I try to change the project to 48k It seems to pitch up the track.I can't have them send the logic/project file as most have outboard synths,different plug ins etc.This particular case the producer has recorded the synth task in 41.000 kHz .My successful outcome would be to be able t create a project file in 48 kHz .And NOT pitch up whne I add the instrumenta wav file .Any help would be gratefully recieved,this is my first post so any mistakes I may have made go easy

You'll have to convert the actual synth audio file file that the producer gave you to 48kHz. You can do this in the audio Bin in Logic.

Emailed voicemails with .wav files no longer play

My office has a VoIP system that I have set up so it sends me an email (Gmail not Outlook) with the voice mail message attached as a .wav file. On Sunday my Droid Razr M stopped working and Verizon made me do a hard reset to restore factory settings. Since then I can't open the voice mail messages. When I try to play the message I get this error message: "Info No app can open this attachment for viewing."
What got reset/removed during the factory reset and how can I restore the ability to play voice mails?

its the updates to google play music causing the problem
go to settings > apps > all and remove updates to google play music and it will start working

Having trouble with wav files

We convert voicemails to wav and push them to each persons Exchange account. All iPhone users can listen to the wav file with the exception of one person. When they click on the wav file nothing happens. They are using a 3Gs. Any thoughts on what to look for?

I too can't get wav files from my exchange server to download to my 3GS but I used to Be able to. I also cannot download them to my iPad either. I am assuming then that this must be something with my server but I don't think anything changed on it. Any thoughts?

Help with wav files

To whom this may concern, can you please help me with this problem!
I am trying to pass a wav file without the header just the data into a Vector in Java.
I need to convert this Matlab (adaptive decorrelation filter) file into Java.
function [v1, v2, a, b, c] = adf(y1, y2, mu1, mu2, delay, Na, Nb);
%[v1, v2, a, b, c] = adf(y1, y2, mu1, mu2, delay, Na, Nb) implements the
%adaptive decorrelation filter. The Lx1 vectors y1 and y2 are microphone
%signals, mu1 and mu2 are ADF stepsizes, delay is any constant delay that
%the microphone signals might have incurred, and Na and Nb are the ADF
%filter lengths. The function returns the separated signals in v1 and v2,
%and the ADF coefficients in a and b.
a = zeros(Na, 1); %ADF coefficients
b = zeros(Nb, 1); %ADF coefficients
L = length(y1); %lenght of Microphone signals
vvec1 = zeros(Nb, 1);
vvec2 = zeros(Na, 1);
yvec1 = zeros(Nb, 1);
yvec2 = zeros(Na, 1);
v1 = zeros(L, 1);
v2 = zeros(L, 1);
for i = 1+delay: L
yvec1 = [y1(i); yvec1(1: Nb-1)];
yvec2 = [y2(i); yvec2(1: Na-1)];
v1(i) = y1(i-delay) - yvec2'*a;
v2(i) = y2(i-delay) - yvec1'*b;
vvec1 = [v1(i); vvec1(1: Nb-1)];
vvec2 = [v2(i); vvec2(1: Na-1)];
a = a+mu1*vvec2*v1(i);
b = b+mu2*vvec1*v2(i);
end
c =[1;zeros(Na+Nb-2, 1)] - conv(a,b); %impulse response of C(Z) = 1 - A(Z)B(Z)

To whom this may concern, can you please help me with
this problem!
I am trying to pass a wav file without the header just
the data into a Vector in Java.Piece of cake until you reach conv. -:) Convolute right.

Problem with .wav files not being able to play on CD players

What am I doing wrong? I took some .wav recordings, loaded them into iTunes, then transferred to a CD-R. The files from the CD-R would play on both Mac and PC computers, but not on any CD players. When loading the files, they automatically went into iTunes. Should I have somehow put them into Garage Band - would that even make a difference? I thought maybe my discs were bad/wrong, so purchased Music CD-R and have the same problem.
Please, can someone help? Thanks!

Although WAV files have the same audio coding as an audio CD, a disc full of WAV files does not act like an audio CD; it lacks the indexing and ToC needed in order for a "regular" CD player to play it. As Diane notes, put your WAV files in an iTunes playlist and burn it as an *audio CD.*

Help with wav file, stop and play buttons

Hey everyone,
I'm new to actionscript, and flash for that matter. I created
a new flash file, and a new layer. On that layer I put 2 buttons,
btnPlay and btnStop.
I found some code on this forum, and tried it, but it isn't
working. The code is below. I get no errors, but when I test the
movie, it doesn't play any music. I selected this layer, selected
frame 1, and put this code in the actions window.
mySound = new Sound();
btnPlay.onRelease = function () {
mySound.attachSound("adagio");
mySound.start(0);
btnStop.onRelease = function() {
mySound.stop("adagio");
I should also note, I have another layer with a movie clip on
it, and another layer with a picture on it.
Any help would be appreciated,
Thanks,
Jesse

As another note--I right clicked on the wav file in my
library and selected linkage and export to first frame as well as
export for actionscript (and named the linkage "adagio"). Might
help with a solution.

Full duplex bluetooth with Iphone?

Both my wife and I have a 3G Iphone and our Jabra BT530 headsets seem to not work as full duplex. Is this an issue with Iphone? The Jabra BT530 claims to work in full duplex mode. When I test the system and talk while the other person talks, they can't hear me; it totally cuts out they say. Tried this on two Iphones.

Well, glad to see i am not the only one.
I posted about this issue back in October, and the forum admin removed my post!
I have experimented with this issue many times, and it's always the same result.
If you use your iphone and a BT headset, and the other person is talking, you can't talk over them - because it's only half duplex.
Worse is if you use your iphone with a BT headset and call someone who is driving in their car. The noise from their car occupies the channel, and the only way they can hear you, is if you yell into the headset!
I use my phone for several hours of talktime everyday, and therefore am dependant upon a BT headset (otherwise my arm would fall off).
Because the iphone is essentially un-useable with a headset, I can not use it as a phone!!!
I was praying that Apple would fix the issue with V3.0, but SOL!
I am very surprised that more people do not complain about this. Can we be the only ones to notice the problem? Surely not!

Audition 5.5 - Help with .wav files for phone system - making a hold message PC & Mcc

Hello everyone,
I am really frustrated trying to create a .wav file for our phone system and I was hoping someone might have some suggestions as I am new to using Adobe Audition. Our phone system will only accept .wav files for our hold messages but the tricky part is that is can only be a maxiimum streaming bit rate of 64 kbps. Everything we've created under 256 kbps becomes in-audible and I'm really not sure what to do as the current messages sound fantastic. Our current hold message is 35 seconds long and we have another one we are about to do which will be 2 minutes long (uuuugh).
- I've tried first converting the .wav file to an .mp3, lowering the bit rate to 64 kbps, then converting back to .wav but the bit rate goes up to almost 1500 kbps after re-converting.
- I've tried making the track mono, reducing the sample rate to 22k but the size ends up being somewhere between 512 and 768. If I lower the sample rate much further and convert to 8bit, the audio sounds terrible.
- I've also tried using Adobe Media Encoder, Audacity, Hand Brake, OS X Automator and a few other programs all with the same results.
- I've tried re-recording the audio on a handheld .mp3 recorder hoping the initial quality would be less (originally recorded on a PC through Audition with a external mic).
Unfortuantely, the person who used to do these conversions is no longer with the company, never documented the process and all I know is that they used to do this on a Mac. Also, the software that uplaods the files is a Linux based which is out of my area of expertise so I'm not able to tinker with it to make any headway.
** Any sort of suggestion would be greatly aprpeciated as I really don't know much about audio. I've been working on this for 2 weeks now and this is really driving me crazy. **

9ov7 wrote:
Hello everyone,
I am really frustrated trying to create a .wav file for our phone system and I was hoping someone might have some suggestions as I am new to using Adobe Audition. Our phone system will only accept .wav files for our hold messages but the tricky part is that is can only be a maxiimum streaming bit rate of 64 kbps. Everything we've created under 256 kbps becomes in-audible and I'm really not sure what to do as the current messages sound fantastic. Our current hold message is 35 seconds long and we have another one we are about to do which will be 2 minutes long (uuuugh).
- I've tried first converting the .wav file to an .mp3, lowering the bit rate to 64 kbps, then converting back to .wav but the bit rate goes up to almost 1500 kbps after re-converting.
Eugh! Converting to MP3 is absolutely not the thing to do!
- I've tried making the track mono, reducing the sample rate to 22k but the size ends up being somewhere between 512 and 768. If I lower the sample rate much further and convert to 8bit, the audio sounds terrible.
- I've tried re-recording the audio on a handheld .mp3 recorder hoping the initial quality would be less (originally recorded on a PC through Audition with a external mic).
Okay, in reverse order: I know it sounds daft, but actually you want to start with the best recording you can possibly have. You are going to have to reduce its bit rate, but if you reduce a crap signal's bit rate, it inevitably ends up worse than reducing a good signal's one; when you think about that for a moment, it's going to make sense (I hope...).
As for what you need to do - well it's going to have to be a mono file anyway (you haven't got a stereo telephone system), and that will immediately halve the streaming rate, but the other thing you need to bear in mind is that telephones have a limitited bandwidth, and that your sample rate, even at 22k, is going to give you more bandwidth than the system has. If you can reduce it to 8k, it will work fine. This will still give you an upper frequency limit of 4kHz, and that's more than adequate for telephone speech, believe it or not...
The other thing, and this is where I think you may be running into trouble, is that 8 bit streaming is pretty much the norm here. All it does is to increase the noise floor, so that the signal to noise ratio is about 48dB instead of the 96dB it would be with a 16-bit signal. But, to get this to work to your advantage, you have to optimise the audio you are using in the first place. Chances are that you need to compress your recording so that it has a relatively limited dynamic range before reducing it to 8-bit, and then you'll be fine. This is better anyway for telephone systems - there's far less chance of bits of the message being missed because they're too quiet. Also, you don't want massive great peaks in the recording - these just reduce the maximum overall level for the majority of the message sound. So, you need to apply a limiter to get rid of them, leaving the majority of the speech sounding just as good, only louder. All of this you can easily do in Audition - don't need any other software at all.
So, my guess is that the previous maker of your messages knew exactly what he was doing. Yes, he could have documented the basic steps, but he couldn't have been absolutely specific about it because no two recordings are the same. I'll attempt to explain the basic steps in English, but how well you will be able to follow this, I'm not sure...
The basic process I'd go through with the recordings is firstly to normalize them to 0dB, then look at how far down most of the 'green' parts are - this is the bulk of the signal, and where all of the meaning is. Then, you apply a limiter to this so that those green parts don't have peaks sticking out of them. Then, if you need to, normalize again so that the green part moves to the upper part of the waveform space (towards where 0dB is). Then, if you need to, apply compression to the signal so that the level of it sounds about the same all through the message.
Now, when you reduce your signal to mono, 8k, and 8-bit, pretty much everything about the original signal will still be there. That should stream 8-bit words at 8k, making 64k streaming for a single mono channel. This will work fine, but only if you optimise the signal first - otherwise no chance!

Audio bug with WAV files: use mp3, not wav for syncing long audio files

Just a tip:
I've noticed another audio bug in Presenter (including 7.05) but there is a workaround. I often export the audio as a single, long clip from pre-recorded lecture videos, and sync it with the powerpoint slides in Presenter. I've found it easiest to import the long audio file into the first slide, then sync using the Edit Audio window. I find that much easier to use than the Sync audio window.
However, If you use WAV format for the imported audio file, sometimes Presenter will simply refuse to play back anything beyond about 5 minutes into the timeline in the Edit Audio window. Nothing fixes it - restarting, re-importing, etc. However the problem goes away if you save the audio as mp3.
Ellen
http://thedesignspace.net

Why did you post this in the LiveType forum?
I would recommending posting your question in the Final Cut Pro forum.
Have you tried trashing your preferences?
Anyways... here is a thread with a similar problem:
http://discussions.apple.com/thread.jspa?threadID=656515

HT1409 Album Art Does Not Work With WAV Files

I seem to have discovered that album artwork does not "stick" to files in WAV format. Is this true? If so, it should be mentioned in this article.
Thanks!

Hi there,
I have started a thread about this in the Finder and Dock category:
http://discussions.apple.com/thread.jspa?threadID=2464851&tstart=0
It looks as if we're not the only ones.

Upgrade from LR2 to LR4 - Have Photoshop CS5 Extended - LR adjustments now lost with RAW files

I have:
Windows 7 64bit
Lots of ram
Lots of disk space
Lots of everything
I have CS5 Photoshop and LR2 - and all works perfectly. I import my RAW files from my Canon 5d2 - and I can edit and adjust - then if I need to export to PS to make further adjustments before saving back to LR. Nice and easy like it should be.
I've installed LR4 as an upgrade to LR2.
Now when I edit my files in LR - and then want to tweak a bit more in PS - when I 'edit in> photoshop' I get my original file - not my edited file, which is very frustrating.
I want to be able to operate like LR2 - make some edits - then off to PS - then back before any final edits,
Why has it changed? And when will we get it back to editing the file with the adjustments built in?

I've already installed ACR 6.7 - and that's helped a little - and I now get the edited file - but not the whole edited file..
So if I crop - that edit carries over..
If I adjust exposutre - that edit carries over...
If I change one of the 'new' settings like 'blacks' - that doesn't carry over...
See below - the image on the right has some 'black adjustment' - and then when edited in PS - you can see a very distinct difference:

.wav file not recognized in Premiere 8

I am trying to edit a video for which I am using a .wav audio file for the audio. I have done this before with no problem. Now Premiere tells me it does not recognize the .wav media file, and may be missing a codec.
When I go back to other projects I have done this way before, I find that I can't open the audio portion of ANY of them. On the project, the audio line shows, but is flat-lined.
Have I lost something. FYI, I am combining audio from concerts (recorded on Pro-tools and bounced to a .wav file) with video of the same concerts recorded on two video cameras. Then, I combine and synchronize to create a finished video with good sound and picture.
Thanks for any suggestions!
Thad Reynolds

Thad,
Welcome to the forum.
With PCM/WAV, the Audio is not compressed, so this should not be a CODEC issue. One thing to note is that there is also LPCM/WAV, and for some reason, a few of the Adobe products have issues with the LPCM, where the PCM version causes no issues.
I'd just run one of the WAV files through G-Spot (PC-only), or MediaInfo (PC, or Mac), and see what it says about the files.
I also assume that you are on the Mac, considering that you mention ProTools. Is that correct?
Were the older Projects, also with WAV files, created on the Mac, or perhaps another computer?
In very general terms, PrE should work perfectly with PCM/WAV files at 48KHz 16-bit. These are uncompressed files, and are the native format for Audio in PrE.
Good luck,
Hunt

Using usb wav files to itunes library mp3?

Ugh, 2 hrs. later I can finally ask a question,
Ok quick reference - Using a USB(16gb) with wav files I would like to add to my library and convert to mp3 (like importing a CD). I already have my preferences set to MP3 (192), like importing a CD it can automatically convert to mp3. Now using a USB mem. I would like to know how to do the same without having to go through each album to convert to MP3.
I know I cannot be alone on this, I have hundreds of albums to go through and editing and deleting each album will take months. Can't this **** program do an auto-convert like ripping a CD?
I hope my question is clear enough to understand.

O M F G ! I am such a NOOB, I didn't know it was a link to a file download.
Are you for real Clark? Ok i'm goin' for it honey!
we'll see what happens.
well so far, what music I already have the info was reversed on all tracks. Album is now Artist and or Genre "ClusterFu*k" of info all out of order. Album art is whatever Hal wants it to be, and the all new 'import' feature ... drum roll please.
POOP! I feel a bubblin' comin'.
Oh well I'm early enough into the redo my itunes 5 times a year, that I can... YES!! START OVER AGAIN!! weeeee!.....
Oh yeah, don't forget to delete the new file -WA WAAAa WA, WAAAAAAA!
Message was edited by: Mudd Butt

I back my music and photos from my windows laptop before it broke on a external hard drive . I just got a new mac book can I use the hard drive to put my wave files and JPEG files on to my mac book

My old lap top broke I save all my music what is wave and my pitches is jpeg on external hard drive
the old laptop was a window base laptop
so I got a new mac coming and I been told my hard drive won't won't on the mac and I lose all my photos and music
any help
thanks

If your external drive has the right ports, you should be able to plug it up to your Mac and copy over your photos and wav files. You'll probably (I don't know this for sure as it's been a long, long time since I've dealt with wav files) need to convert the wav files to MP3 or some other format (but I would try opening them directly from iTunes, first - wait a second... yes, iTunes will open wav files).
There are only a fixed number of ways that Windows drive can be formatted and a Mac can read (though not necessarily write) them all. So you should have no trouble with that.
Good luck,
Clinton
MacBook Pro (15-inch Late 2011), OS Mavericks 10.9.4, 16GB Crucial RAM, Crucial M500 960GB SSD, 27” Apple Thunderbolt Display

Full Duplex Lost with Wave Files on SB Live! Value (W

Similar Messages

Maybe you are looking for