Callers say my voice sounds muffled when using bluetooth.

I have a motorola bluetooth h710/h715 and as soon as I got the iphone 4, everyone is telling my I sound muffled when using it, compared to my 3gs. Anyone else have this issue?

Can a few of you help me with a test? I have a Jawbone 2 & a Jawbone Icon. Both exhibit a strange behavior with my iPhone 4. If either of these headsets are on, paired & connected AND a call is answered or initiated, things are fine.
When it all goes downhill is in this scenario:
My headset is OFF. I receive or make a call and mid-call, I decide to turn on the headset (which I keep off to save battery). The Jawbone connects and automatically switches the iPhone from PHONE to JAWBONE. This is all normal right? Okay here's the problem. The call quality has static, sounds muffled and the caller tells me I sound like I'm in a tin can. If I then switch back to the phone (through the iPhone page of Sources) and then back again, to the Jawbone, most times--quality is back to normal. Obviously this is not something we all want to be doing. Can anyone do this test with their headset .. any brand and even some of you with Icons? It would be good to know if the culprit is the Bluetooth Headset or the iPhone. Remember, it's important that you make a call with your Bluetooth headset in the OFF position and then turn it on mid-call to use it.

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Can a few of you help me with a test? I have a Jawbone 2 & a Jawbone Icon. Both exhibit a strange behavior with my iPhone 4. If either of these headsets are on, paired & connected AND a call is answered or initiated, things are fine.
When it all goes downhill is in this scenario:
My headset is OFF. I receive or make a call and mid-call, I decide to turn on the headset (which I keep off to save battery). The Jawbone connects and automatically switches the iPhone from PHONE to JAWBONE. This is all normal right? Okay here's the problem. The call quality has static, sounds muffled and the caller tells me I sound like I'm in a tin can. If I then switch back to the phone (through the iPhone page of Sources) and then back again, to the Jawbone, most times--quality is back to normal. Obviously this is not something we all want to be doing. Can anyone do this test with their headset .. any brand and even some of you with Icons? It would be good to know if the culprit is the Bluetooth Headset or the iPhone. Remember, it's important that you make a call with your Bluetooth headset in the OFF position and then turn it on mid-call to use it.

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I am not too sure if this is affected by BT software stack or RF design...
iPhone 3G with FW 2.x provides Cisco VPN as well as Email support for corporate usage. However, it's weird that business man wouldn't want to use BT headset and talk to people... We are not asking about new feature support, like A2DP, but reliable, stable BT connection ONLY!!
I've read that lots folks spent a lot of money on purchasing many different BT headsets and tried to get it work with iPhone 3G. I myself tried: Plantronics 510/815/855, Sony HBH-DS970/980, Nokia BH-903. None of them can work fine with my iPhone 3G ever! And iPhone 3G even cause BH-903 and DS-980 hang up after pairing or finishing call!!
That's why I don't believe it's just a "SPECIAL CASE". How could a product FOR BUSINESS purpose get this kind of basic function problem?
I hope we will find this issue could be resolved completely with new F/W, like Safari crash problem before. Other than this BT IOT problem, iPhone 3G is really the phone/companion could provide complete solution fulfill my personal and business needs.

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Hello there, Baxter530.
The following Knowledge Base article provides pretty extensive steps to troubleshoot Bluetooth on your iPhone:
iOS: How to troubleshoot Bluetooth connections
http://support.apple.com/kb/TS4562
Thanks for reaching out to Apple Support Communities.
Cheers,
Pedro.

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Hi, eklinck.
Thank you for visiting Apple Support Communities.
Sometimes closing all applications open in multitasking and power cycling the device can resolve the issue.
iOS: Force an app to close
http://support.apple.com/kb/ht5137
However, if you are still experiencing the issue I have included the best troubleshooting article.
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http://support.apple.com/kb/TS3581
Cheers,
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I'm running XP Media Center 2002.
I just installed Live! Cam Voice. Started using it and is was fine out of the box. I updated the drivers and i also decided to update the drivers for my SoundBlaster Audigy 2 ZS.
The problem only seemed to occur when using the Motion Detection. Shortly after using it, I'd get a Microsoft error prompting the WebCam Center to shut down.
First I uninstalled the software and reinstalled it using the original drivers on the supplied disk. I still had the problem. So i rolled back the drivers on my sound card. That seems to have fixed the problem... Thought you'd want to know.
To be clear the Cam driver I'm using is .0.8.206. I believe this is the one that came w/ the install disk. Hope this helps anyone else who's experienced similar problems.....

I have the same problem. I only get 6-7 fps max. I've never seen anything above 0 fps which is a disgrace. My computer is and Athlon64 X2 4800 with 2GB RAM so it's nothing to do with spec. This behaviour was the same in driver version .0.8 as well as the latest one... .0.02 or something.
Also, since the last driver update, the whitebalance and focus when on automatic is dreadful. I'm not impressed.

Sound leak when using unsigned

I've generated a simple test case which should make this pretty straightforward. I'm using Linux Ubuntu 11.04 with pulseaudio, if it makes any difference.
Basically, I generate a very simple tonal sound for the musical note A3 using Java's java.sound.sampled APIs, and Clip to open and play it.
That's all well and good, but to avoid leaking sound resources, I have to listen to the Clip (or Line) for a STOP event, and then close it. That's fine too. If I play two sounds at the same time and do this, it works fine, too...
unless the sound encoding is unsigned. For some reason, switching it from signed to unsigned (and translating the tonal wave appropriately) makes me only able to close one of the Lines, and the other one goes rogue.
Here's my code:
http://pastebin.com/1mxBJ61t
or
import java.util.Timer;
import java.util.TimerTask;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.Clip;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineEvent;
import javax.sound.sampled.LineListener;
import javax.sound.sampled.LineEvent.Type;
public class Test
     public static void main(String[] args)
          int volume = 64; //0-256
          double seconds = 2.0;
          double tone = 220.0; //Musical note A3 = 220.0
          final int sps = 80000; //sample rate (samples per second)
          final boolean signed = false;
          final byte[] toneData = makeTone(sps,tone,(int) (sps * seconds),volume,signed);
          //play the first tone
          try
               play(toneData,sps,signed);
          catch (Exception e)
               e.printStackTrace();
          //halfway through, play the second tone
          new Timer().schedule(new TimerTask()
                    @Override
                    public void run()
                         try
                              play(toneData,sps,signed);
                         catch (Exception e)
                              e.printStackTrace();
               },(long) (seconds * 500));
          //since the program doesn't know to terminate itself, we'll give it an explicit terminate
          new Timer().schedule(new TimerTask()
                    @Override
                    public void run()
                         System.out.println("Terminating");
                         System.exit(0);
               },(long) (seconds * 2000));
     //generates a simple tonal wave
     public static byte[] makeTone(int sps, double freq, int duration, int volume, boolean signed)
          byte[] r = new byte[duration];
          for (int x = 0; x < duration; x++)
               double wave = Math.sin((double) x / (sps / freq) * 2 * Math.PI);
               //taper off the end so it doesn't cut prematurely and 'blip'
               double taper = duration - x < 100 ? (duration - x) / 100. : 1;
               byte b = (byte) ((volume * wave * taper) + (signed ? 0 : 128));
               r[x] = b;
          return r;
     public static void play(byte[] data, int sps, boolean signed) throws Exception
          AudioFormat afmt = new AudioFormat(sps,8,1,signed,false);
          DataLine.Info info = new DataLine.Info(Clip.class,afmt);
          Clip c = (Clip) AudioSystem.getLine(info);
          c.open(afmt,data,0,data.length);
          c.start();
          //listen for the clip to stop, and then close it
          c.addLineListener(new LineListener()
                    @Override
                    public void update(LineEvent event)
                         System.out.println("Sound event: " + event.getType());
                         if (event.getType() == Type.STOP) event.getLine().close();
     }A few modifications you can make to see what I'm talking about:
Line 21, change signed to true, and everything works great.
Line 92, assuming signed is false, comment out this line, where I close the stream, and re-run. You will observe 2 Stop events. Enable it again and you will observe only 1 Stop event.
Am I doing something wrong, or is Java's sound system just that deplorable?
Thanks in advance,
-IsmAvatar

Thanks for replying again.
sabre150 wrote:
IsmAvatar wrote:
I dropped it down to 8000, but it still does the same thing for me. II didn't say it would - just that 80,000 seemed a very high sample rate. Unless you are playing sound to bats using hardware much better than I am then anything higher than a sample rate of 44 KHz is a waste of time. At my age the highest frequency I can hear is about 8 KHz so any sample rate much above 16 KHz is too high.
Makes sense.
have noticed varying behaviors.
I'm fairly new to this sound stuff, and at this point I'm just trying to get it to work. Get what to work? You have not really described, or a least I have not understood, exactly what you are trying to do. You have a clip of sound you are trying to play more than once but you are trying to "avoid leaking sound resources". Can you explain what you mean by this and why you think it necessary; a reference to some Java sound documentation would maybe help us understand.
Well, here's the sound documentation that I'm basing my code off of: http://download.oracle.com/javase/1.5.0/docs/guide/sound/programmer_guide/chapter4.html (and chapter 3)
Basically, my program has a play button that will play a sound that the user has loaded in (complex encodings or formats are not a concern currently). I need to be able to figure out how to handle the user clicking on the button multiple times. I would at least expect an average system to be able to play the sound twice concurrently (overlapping). Maybe I just have the wrong concept when I talk about "sound resources", but it seems like I run into problems down the road, like on my computer, after I've played the sound 30 times, it starts erroring, or if I try to play the sound twice, and then once more after they're done, it causes the application to hang. To me, this seems like it's looking for available lines, and just plum running out, or choosing a line it thinks is available, and hitting a wall. Idunno.
I have a .wav file that uses PCI Unsigned 8000 Hz that was exhibiting this problem. I don't think the '.wav' file was not exhibiting the problem I think your code is/was .
Fair enough. It's easier to fix the code then to expect to have certain types of wavs.
Ultimately it leads to the application hanging entirely,
I would be interested in seeing an SSCCE ( http://pscode.org/sscce.html ) that exhibits this since I have never met it.
I think it would be wise to deal with one problem at a time for now. The program that exhibits it is large and would be time-consuming to simplify. I figure the problems are interrelated, so if we can fix the current problem, the apparent freezing issue should resolve itself as well. If not, then I can make an SSCCE for that.
and I have to forcibly quit the JVM. Being able to close all sounds completely curtails that "feature", but it's apparently not an option for unsigned encodings.Since the difference between the two is just a toggle of one bit I don't really believe that this is a signed/unsigned problem. I believe there is something more fundamental to explain the problem. I expected as much. It's always nice to be able to fix one's own code/thinking than to hit a roadblock with the limitations of an API.
>>
Plus, your exception doesn't sound that desirable either.I didn't say it was desirable - I was just saying what I observed. My exception seems logical since the line I was trying to open was already open so a LineUnavailableException makes sense.I suppose so, yes. And I'm guessing your system is more than capable of playing two lines concurrently, if done correctly.
I suppose the problem that you're seeing (and probably mine, too) is that it's trying to recycle the Line (just a guess - I could be completely off). For whatever reason, mine lets me play the recycled line twice, but yours doesn't. Both ways cause issues.
I suppose this tells me that if I want to play another sound concurrently, I need another line, not a recycled one. Does that seem about right? If so, how might I go about doing that?
If not, perhaps I'm a little thick and need some hints.
In short, the problem is I would like to be able to play two sounds concurrently (seems like a reasonable enough request), but either I can't using my current code code (as you've seen), or it causes other issues (my side). So I'd like to know how to fix the code to enable this.
Greatly appreciated.
-IsmAvatar

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Hi,
I have been using vent for about 6 months on a bootcamp blackbook from 2006 with realtek audio drivers without any hazzle at all.
Now I bought a new macbook pro which have Cirrus Logic HD audio drivers. If I pull up normal XP sound recorder program I hear my voice in normal speed when testing microphone. IF I start ventrilo and enable outgoing voice communications and do a test recording on XP sound recorder program my voice speed is decreased. Luckily that program has an increaser of 100% in effects settings and that turns my speech recorded back to normal.
This also happens when I play Lord of the Rings Online without ventrilo on - My theory is that when there is two applications wanting to use the mic it decreases like it splits up as in Vent + (normal windows use) or LOTRO + (normal windows use).
This is very annoying as it makes voice chat impossible since its in ultra slow.
Please if anyone have any suggestions as to how to turn something off to make it only register 1 use of mic or any info u need let me know.
Tested with a headset with microphone and same deal (non-usb headset)

Hello every1,
i have the same Issue in ventrilo. My voice is so slow when i talk. but when i installed Skype my voice sounds normal and i think its better than Ventrilo.
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Hi,
Try a clean installation of the flash plugin/activeX as follows.
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Regards,
DP-K
****Click the White thumb to say thanks****
****Please mark Accept As Solution if it solves your problem****
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Microsoft MVP - Windows Experience

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First words are lost when using Bluetooth voice de...

This happens with any Nokia Nxx or Exx when used with bluetooth HF device, like Nokia HF-300. Is there any solution to this problem? In practice you can hardly use any navigation with bluetooth voice. The delay itself is not a problem but the phone should wait 0.x seconds after opening the line to the HF device. It would be nice if you could configure this kind of delay to the phone.

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Message Edited by confiture on 18-May-2009 08:56 PM

Callers say my voice sounds muffled when using bluetooth.

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