No sound events in xfce4

I have installed xfce. My problem is that I can't enable sound events. There is an option in the settings to enable it but it has no effect at all. I have installed libcanberra has suggested. I do not use pulseaudio but plain alsa (and I would prefer not to use pulseaudio). This is not a permission or driver problem since applications are able to use the sound just fine. I do not know what to do from here. Has anyone an idea?
Last edited by olive (2013-01-29 06:59:52)

Have you figured out something about this problem? I seem to have the same problem here: https://bbs.archlinux.org/viewtopic.php?id=160059
I also use alsa.
Last edited by kellerman (2013-04-05 18:47:45)

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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
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               e.printStackTrace();
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               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

Aren't Sound Blasters what they used to

Hello everyone,
I have been a Creative Labs user for several years, especially before 2000 (Soundblaster 2.0, SB6 Vibra ISA (2x), SB Pro .0), and was absolutely pleased with them, especially with DOS games. Those were the happy times...
Anyway, in more recent times, after years of listening to trustworthy but not really performing AC'97 integrated soundcards, I decided to purchase an "audio accelerator" in the form of an Audigy 4 (the one with the EMU 0300 chip and no firewire/remote control, if that means anything).
Anyway, it worked fine for a couple of weeks or so, I got none of the SCP problems posted here, however that warning in the manual about the "VIA hard disk corruption problems" troubled me...after some research I saw it was mainly about older VIA chipsets (I had an A64 3200+ with KT800T VIA chipset so I relaxed and said "this cannot happen to me, seems like a problem solved long ago". How wrong I was....
One day, out of the blue, while playing Battlefield 942 I got a horrible BSOD and had to reboot. After than, and after a VIA drivers update, I noticed that the sound stumbled and HARD DISK SAVING stumbled too, which was much more dangerous. I reverted to an older version of VIA drivers that didn't cause that problem, but one day I got another BSOD coupled with disk corruption, and, apparently, what later revealed to be a SATA CONTROLLER PHYSICAL DAMAGE!!!
To make the long story short, I had to replace my motherboard, and after checking these forums I discovered that disk corruption, PCI Bus Mastering and SCP problems have been plaguing the whole soundblaster product line ever since the Li've! series and were never really solved, plus it's not clear if they are purely software or hardware, or both.
In any case, this card caused me a lot of expenses, and after reading the horror stories here, I am really afraid to plug it into any of my PCs, in fact I'll try to sell it hoping that the manufacturer's reputation doesn't make it impossible.
So really, what's the deal here? Is it just me, or Sound Blasters just aren't what they used to be? How can a whole class of hardware problems persist for so long, even trhough the X-Fi series? If I had seen those forums BEFORE handing out 70 Euros for that d4mn thing, I would have probably kept my money and stick to AC'97 with rock-solid drivers and hardware!Message Edited by CreativeSkeptic on 03-5-200704:20 PM

Well, I don't know how the X-Fi series scores in any of those fields, as I can only speak for my experiences with the Audigy 4.
Apart from the times where it obviously glitched and/or interacted badly with the rest of the hardware, it played quite smoothly, and MIDI playback took noticeably less CPU time than with the software synth (at least according to task manager, and obviously not accounting for PCI bus utilisation), which was one of the reasons I initially bought this card.
Exactly what makes MIDI playback different than playing sound effects in games? Since both are made from samples read from memory and sent to the sound card in different streams (up to 32 or 64 depending on your model), they should be exactly the same (well OK, midi is perhaps sort of more "continuous", but a game laden with sound effects may generate 0s of sound events per second).
By admitting that such "hardware accelerated" playback is beyond the capabilities of the current PCI bus is ridiculous, and obviously defeats the purpose of an accelerator sound card. Then why should one ever buy one, if in the end it's going to perform worse than software mixing and synthesis?
The numbers also don't convince me: the PCI bus has a 33 MHz clock speed and a 32-bit data bus, for a nominal data rate of 33 MB/sec. Even in ths case you're using 92 KHz, 24 bit resolution soundfonts for 64 channels at once, That means a "mere" 36.684 MB/sec at once. OK, using stereo soundfont would actually double that, and reach a whooping 73+ MB/sec, which is probably impractical.
For more "normal" game sound effects and soundfonts, admitting they use 48 KHz, 6 bit samples in stereo, that's a measly 2.28 MB/sec for using all 64 channels. Since virtually no midi really uses 64 channels, as don't most games, that leaves a margin of 0: compared to the PCI bus capabilities. To make a comparison, a TV tuner card I have, when capturing a PAL signal, produces something like a continuous stream of 33 MB/sec of raw data that can go on for hours, yet I never had any problems RECORDING that, with real-time MPEG compression (in software, it was a cheap TV tuner) while recording the sound with the integrated soundcard.
If the Audigy or the X-Fi or whatever cannot handle similar or inferior data rates, then something is basically flawed. Sure, there is still subdivision of the data stream into PCI packets, a maximum allowed bus mastering time and timing-sensitivity in the case of MIDI/sound effects, but still, glitchy playback and/or multi-second pausing indicates a much worse error than "missing" control of the PCI bus for a few cycles.
Regarding the AWE32 comments...does this imply than all or most of the current Sound Blaster lineup is technically inferior to an ISA card released 0 years ago? Then the "hardware acceleration" is but an illusion, and we should all perhaps stick to integrated or "normal" soundcards?

How do I enable all sound files wav,MP3 on windows 7 64-bit change system sounds

How do I enable all sound files wav,MP3 on windows 7 64-bit change system sounds it will only give me the option to choose wav?

Windows 7 uses .wav (wave) files for the sound events. The default folder that Windows 7 uses to store the sound files in is located at C:\Windows\Media.
You cannot use a MP3 file but there are lots of (WAV to MP3 converters) available on internet which can help you to convert MP3 to wav serving your purpose of changing system sound.
//Click on Kudos and Accept as Solution if my reply was helpful and answered your question//
I am an HP employee!!

Sound pressure spectra with SVT

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I have a project which I have been having a bit of trouble implementing. I'm performing several measurements at 48kHz/24Bit of some sound events. My requirement are as follows for sound pressure spectra analysis.
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4. Peform analysis using SVL Zoom Power Spectrum.vi.
Few Issues:
I dont know how to implement the Anti-aliasing filter.
Is the method I have used ok with measuring sound spectra? Or should I perform the FFT then convert to dB?
I have tried to implement the same analysis using Matlab and although I get the same FFT profile, my dB levels are different where the 16kHz noise is almost the same level as the 8kHz noise? Not sure what is happening there, but I want to concentrate on using LV for now.
I dont have any guidance available or experience with this sort of analysis. I have trialed my program out on a sample wav file which has two separate distinct noises events, one at 8kHz and another at 16kHz. My FFT is showing that the 16kHz is a much higher sound level (95dB) than the 8kHz (80dB) but when I listen to the wav via a pair of studio monitor speakers, the 8kHz is much more audible. Wouldn't the implementation of A-weighting filter tend to make the 8kHz noise more prevalent in the FFT?
Any suggestions on this topic are appreciated.
Thanks!

Hi David,
If you want to enable the enhanced alias filter, you can use a DAQmx channel property node. I would recommend placing a DAQmx Channel property node on the block diagram and then going to Analog Input » General Properties » Advanced » Enhanced Alias Rejection Enable. See if that works and let me know.
Jake H | Applications Engineer | National Instruments
Attachments:
Enhanced Alias Rejection.png ‏3 KB

No sound events in xfce4

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