Audigy 2 ZS Output Sound bleeding into input(m

Hi,
I'm having trouble with the Audigy 2 ZS in Windows x64 with
SBA2_WEB64_LB_2_03_0004.exe &
AudCon_PCApp_LB__20_23.exe
installed.
The problem is when sound is being send to the Audigy output port (head phones) it's also being sent to my mic (input port). I can physically disconnect the mic from the Audigy 2 ZS card and I still get a duplicate output into the input port. Even if I mute the mic in the sound control panel.
Example, If I play a MP3 in winamp and start recording using Windows sound recorder, it will record the output prefectly without anything connected to the back of the card.
The real issue is the feedback people are getting when trying to talk to me using Vent or Teamspeak. Anything they say, it echos right back to them.
CMSS on or off doesn't make a difference.
If anyone has any ideas, I'd like to hear them? I'm thinking it might be a bad card but didn't want to send it back until I tried everything.
-Wojek

The problem was with the Sound recording device being "select"ed on the Wave instead of the Microphone. Even though the Microphone was working, it was still also sending the output or the Wave along with it.
-Wojek

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compile this file first....
import java.applet.*;
import java.awt.*;
import java.awt.event.*;
import java.awt.event.ActionEvent;
import java.awt.event.ActionListener;
import javax.swing.*;
import java.io.IOException;
import java.io.File;
import java.net.URL;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.TargetDataLine;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.AudioFileFormat;
public class VoicePasswordRecorder extends Thread {
 private TargetDataLine m_line;
 private AudioFileFormat.Type m_targetType;
 private AudioInputStream m_audioInputStream;
 private File m_outputFile;
 public VoicePasswordRecorder(TargetDataLine line,
 AudioFileFormat.Type targetType,
 File file)
 m_line = line;
 m_audioInputStream = new AudioInputStream(line);
 m_targetType = targetType;
 m_outputFile = file;
 /** Starts the recording.
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 (ii) The thread is started,
 public void start()
 /* Starting the TargetDataLine. It tells the line that
 data needs to be read from it. If this method isn't
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 /* Starting the thread. This call results in the
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 data is actually read from the line.
 super.start();
 /** Stops the recording.
 Note that stopping the thread is not necessary. Once
 no more data can be read from the TargetDataLine, no more data
 can be read from the AudioInputStream. The method 'AudioSystem.write()'
 (called in 'run()' returns. Returning from 'AudioSystem.write()'
 is followed by returning from 'run()', and then the thread
 stops automatically.
 public void stopRecording()
 m_line.stop();
 m_line.close();
 /** Main working method.
 'AudioSystem.write()' is called.
 Internally, it works like this: AudioSystem.write()
 contains a loop that is trying to read from the passed
 AudioInputStream. Since there is a special AudioInputStream
 that gets data from a TargetDataLine, reading from the
 AudioInputStream leads to reading from the TargetDataLine. The
 data read this way is then written to the passed File. Before
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 to the desired audio file type. Reading continues until no
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 it happens if no more data can be read from the TargetDataLine.
 This happens if the TargetDataLine is stopped. Then,
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 public void run()
 try
 AudioSystem.write(
 m_audioInputStream,
 m_targetType,
 m_outputFile);
 catch (IOException e)
 e.printStackTrace();
 public static void main(String[] args)
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 TargetDataLine targetDataLine = null;
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 and writing the data to a file.
 VoicePasswordRecorder recorder = new VoicePasswordRecorder(
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 targetType,
 outputFile);
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 catch (IOException e)
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 out("\tjava VoicePasswordRecorder -h");
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import java.awt.*;
import java.awt.event.ActionEvent;
import java.awt.event.ActionListener;
import java.io.*;
import javax.sound.sampled.*;
import javax.swing.*;
public class AppletTest extends JApplet
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 stoprecording = null;
 recorder = null;
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 getContentPane().setLayout(gridbag);
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 gridbag.setConstraints(panel, c);
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 stoprecording.setEnabled(true);
 startrecording.addActionListener(this);
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 gridbag.setConstraints(stoprecording, c);
 getContentPane().add(stoprecording, c);
 int width = 300;
 int height = 200;
 setSize(width, height);
 setVisible(true);
 boolean looping = false;
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 if(ae.getActionCommand().equals("Start"))
 createRecorder();
 recorder.start();
 System.out.println("Start Button Clicked");
 } else
 if(ae.getActionCommand().equals("Stop"))
 recorder.stopRecording();
 System.out.println("Stop Button Clicked");
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 String strFilename = "C:\test.wav";
 File outputFile = new File(strFilename);
 AudioFormat audioFormat = new AudioFormat(javax.sound.sampled.AudioFormat.Encoding.PCM_SIGNED, 44100F, 16, 2, 4, 44100F, false);
 javax.sound.sampled.DataLine.Info info = new javax.sound.sampled.DataLine.Info(javax.sound.sampled.TargetDataLine.class, audioFormat);
 TargetDataLine targetDataLine = null;
 try
 targetDataLine = (TargetDataLine)AudioSystem.getLine(info);
 targetDataLine.open(audioFormat);
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 comp.append("unable to get a recording line\n");
 e.printStackTrace();
 System.exit(1);
 javax.sound.sampled.AudioFileFormat.Type targetType = javax.sound.sampled.AudioFileFormat.Type.WAVE;
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 e.printStackTrace();
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 private JButton startrecording;
 private JButton stoprecording;
 private SimpleAudioRecorder recorder;
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<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=windows-1252">
<meta http-equiv="Content-Language" content="en-us">
<title>Home Page</title>
<meta name="GENERATOR" content="Microsoft FrontPage 5.0">
<meta name="ProgId" content="FrontPage.Editor.Document">
<meta name="Microsoft Theme" content="artsy 011, default">
<meta name="Microsoft Border" content="tl, default">
</head>
<body background="arttilea.jpg" bgcolor="#000000" text="#FFFFCC" link="#FF9900" vlink="#999900" alink="#669933">

 Please enter login
 details below:
<form>
<table border=0 cellpadding=3 cellspacing=3>
<tr>
 <td align=right width="74">
 Username
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 <td width="335">
 <input type=text
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 </td>
</tr>
<tr>
 <td align=right width="74">
 Password
 </td>
 <td width="335">
 
 <input
 type=password name=password size="20">
 </td>
</tr>
<tr>
 <td align=center width="74">
 Voice
 Password Check
 </td>
 <td width="335">
 
<APPLET CODE="AppletTest.class" ALIGN="MIDDLE" WIDTH=272 HEIGHT=38 NAME="AppletTest.class"></APPLET> 
 </td>
</tr>
<tr>
 <td colspan=2 align=center>
 <input type=submit value="Enter Website" name="Submit">
 </td>
</tr>
</table>
</form>
 This page was last updated on 25/03/04.
</body>
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Regards,
Mayur Patel

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i am posting my code
import java.io.FileReader;
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OSA example.vi ‏42 KB

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Mac
OS X Yosemite: Version 10.10.2
iMac (27-inch, Late 2013)
Processor : 3.2 GHz Intel Core i5
Hard-drive : 1 TB SATA Disk
Ram: 8 GB 1600 MHz DDR3
Graphics: NVIDIA GeForce GT 755M 1024 MB
Built-in Output:
Default System Output Device:       Yes
Manufacturer:                                  Apple Inc.
Output Channels:                             2
Current SampleRate:                      48000
Transport:                                        Built-in
Yeti Stereo Microphone:
Default Input Device:                      Yes
Default Output Device:                    Yes
Input Channels:                               2
Manufacturer:                                 Blue Microphones
Output Channels:                            2
Current SampleRate:                      48000
Transport:

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Last edited by agahnim (2013-07-04 00:45:35)

I fixed it (for now, at least!)
I ran the below command:
$ cat /proc/asound/cards
0 [U0x46d0x804 ]: USB-Audio - USB Device 0x46d:0x804
USB Device 0x46d:0x804 at usb-0000:00:1a.0-1.2, high speed
1 [MID ]: HDA-Intel - HDA Intel MID
HDA Intel MID at 0xf9ff8000 irq 47
2 [NVidia ]: HDA-Intel - HDA NVidia
HDA NVidia at 0xfbd7c000 irq 17
Basically, my USB webcam, which also acts as a microphone, was being treated as my default sound card. I changed this by creating a file in /etc/modprobe.d with the following:
options snd-hda-intel index=0
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$ cat /proc/asound/cards
0 [MID ]: HDA-Intel - HDA Intel MID
HDA Intel MID at 0xf9ff8000 irq 47
1 [NVidia ]: HDA-Intel - HDA NVidia
HDA NVidia at 0xfbd7c000 irq 17
2 [U0x46d0x804 ]: USB-Audio - USB Device 0x46d:0x804
USB Device 0x46d:0x804 at usb-0000:00:1a.0-1.2, high speed
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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

Change Output Sound Device

Is there anyway to change the output sound device in iTunes without going through System Preference?
I have an iMac with built-in Speakers, and a 5.1 Griffin Firewave over Firewire. I want my music to come through my external speakers but I don't need my system sounds to come through those, right now I can only change it through System preference and haven't found a way to divide them up.
This is do-able in the Apple DVD Player... any thoughts for this function in iTunes? Is there some place I can put this in a suggestion box if it's not already a function?

It is a great pity you are using Leopard. Up until Tiger there was a great little free app called Detour that let you assign separate audio channels for different apps. You could Google "Detour" and see if there is another utility that will work for Leopard?
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Does anyone know of a mini sterio with surround sound and optical input in Mac Mini form factor?

Does anyone know of a mini sterio with surround sound and optical input in Mac Mini form factor?
I use a Mini as the family media center and would like to have 5.1 or 7.1 sound for films but don't want a huge piece of eletronic equipment (I have one of those, but it isn't hooked up because it doesn't fit on the bookshelf). Years ago I used a PC and still use the Creative Audigy speaker system (Jimmy rigged to the Mini) that I ran from the PC. The Audigy PCI sound card obviously doesn't work on the Mini, so I just have two chanel plus sub. That has been fine since I switched to the Mini in 2007, but am updating to a new Mini in the summer and would like to upgrade my sound as well.
I also wouldn't be opposed to a USB soundcard that can drive the speakers. In fact, my wife would prefer smaller and less visible, ergo USB card hidden in the back.
Suggestions?

Hello John, I figured it was a signal issue. All the speakers fire and the sound is great. I was running the speakers thru a Dell PC with a Creative X-Fi Elite Pro, THX Certified, 7.1 soundcard using a fiber optic from the Mac Pro to the Z906's, it worked great until the Dell died!!! When I bought the soundcard from Creative I also bought the GigaWorks S750, THX Certified, 700 watt, 7.1 speaker system, used the speakers for 11 years and then the woofer/amp said "I QUIT" Creative quit making the speaker system and the sound card. The satellite speakers from the Creative GigaWorks speaker system still sound great (rated at 78 watt each) and are a little better speaker than the one's that came with the Z906 and the wattage from the 906 is sufficient to drive the satellites without any distortion. Thank you for addressing my issue, you confirmed what I suspected all along, just needed to hear it from someone with the same setup.
One last question, I purchased a Diamond Multimedia USB Soundcard, can't use the fiber optic(not supported by Mac) but the green, tan and black RCA's plug in and produces adequetly. When you plug your 906's into the Mac Pro using the fiber optic how do you set your speaker configuration? When I plug into the mac pro with fiber optic the 'Audio Midi Setup' does not seem to see the 5.1 configuration. Any thoughts there?
Carl

Garage Band, I am using an MBOX with a digi design audio core as a mic interface to recod into Garage Band. Now when I Change the preferences to use the MBOX I can not move the recorder slider and there is no sound going into garage band? Any solutions

Garage Band, I am using an MBOX with a digi design audio core as a mic interface to recod into Garage Band. Now when I Change the preferences to use the MBOX I can not move the recorder slider and there is no sound going into garage band? Any solutions

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BTW
Ive tried all the system sound prefs, and garage band drivers. Back and forth.
cheers
ds

Audigy 2 ZS outputs buzz instead of mu

My Audigy 2 ZS was working fine, but today suddenly it went awry. I was running foobar2000 (music player) and StarCraft, when suddenly all my StarCraft sound turned into a steady buzz. My foobar music still played properly though. I exited StarCraft and ran AIM and IM'd myself, but I got a buzz instead of the AIM noise. All these buzzes were coming out the speakers, and there was no buzz when no sound was being outputted, so I assumed it was a sound card problem.
I've since removed the Audigy 2 ZS from my system and have no problem with the sound using integrated sound. The latest drivers for the Audigy 2 ZS were installed in my system as well.
Does anyone know what went wrong with my sound card today?

The same problem keeps occuring over time, but removing the card and then re-inserting it into the PCI slot solves it.. temporarily.
Does anyone know why this would happen?

Audigy 2 ZS - popping sound in mce

Hi,
I recently installed windows 7 (32 bit). I set up the speakers in windows media center to use the coax digital output (and 5. speakers) and when I click the button to test the speakers I just hear a clicking/popping sound.
If I set the speakers up for the single RCA analog plug and 5., the test button makes a sound ok but only for 2 channels.
I've installed the most recent driver I could find for audigy 2 zs, which is v 2.8.
Can anyone suggest how to get the speakers working in mce?
Thanks,
Phil

&Re: Audigy 2 ZS - popping sound in mceO No, you're not belittling me at all. There are so many settings applictions that I don't know where to start or even how to find them all. Good tip re the windows/creative control applets.
Setting all the # speakers and bit rates the same in the windows and creative apps didn't seem to help so I tried disabling the spdif output device and now I get all channels in mce. Yay. Thanks for your help.
Seems like the speakers device works but the spdif device does not. My sound card has a 2.5mm digital output port which I connect to the coax digial input on my denon amp with a mono 2.5mm to single rca cable. Is the spdif device supposed to work with this kind of connection?
Also, am I likely to lose anything by using the speaker device instead of the spdif device?
Thanks,
Phil

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