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Pulseaudio no sound card detected. Dummy output only

- by Zach Smith

I'm using 12.10 Quantal with Openbox and a .xinitrc script at login instead of a display manager. Its a relatively fresh install and I noticed when I opened pavucontrol the only output was a dummy one. I check around and it appears that my soundcard is physically installed but Pulseaudio isn't detecting it. I'm really unsure what I should do but any help getting my audio back would be appreciated. Edit: further info if its at all useful: dante@dante-ubuntu:~$ uname -a && aplay -l && cat /proc/asound/version && head -n 1 /proc/asound/card*/codec#* Linux dante-ubuntu 3.5.0-17-generic #28-Ubuntu SMP Tue Oct 9 19:31:23 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux aplay: device_list:252: no soundcards found... Advanced Linux Sound Architecture Driver Version 1.0.25. == /proc/asound/card0/codec#0 <== Codec: ATI R6xx HDMI == /proc/asound/card1/codec#0 <== Codec: IDT 92HD81B1X5

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No sound 12.04 (Dummy output)

- by Edgar Adrian Alvarez

Its been weeks with no sound. I feel like Ive tried everything but somethings just dont seem right. I am a new user and so far i love Ubuntu but this sound issue is making me unsure. Im NOT muted. Ive tried multiple jacks, front and back. in alsamixer it says choose sound card and I have only thr 'hda Intel' option. In pulseaudio I only have 'dummy output'. When I have Youtube on I can see audio being detected in pavucontrol but nothing is coming out of the speakers. Im getting desperate, some one please walk me thru this. http://www.alsa-project.org/db/?f=c7377242d96ea884edebd807f4fe71f619b8d6af What more information should i provided?

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How to make Bluetooth headset appear as a sound device?

- by torbengb

I have an onboard VIA sound card but no speakers attached. Instead, I want to use my bluetooth headset as my only sound device, both input and output (mono). I plugged in the bt dongle and was impressed that it was ready to use within seconds. I then paired it to my Jabra BT500 headset. No problems so far. I then went to Sound Preferences but no bluetooth is listed, only the onboard VIA sound card. Question: How can I enable my headset, and make it the default all the time?

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How to route KVM virtual machine audio to Ubuntu 11.10 host using virt-manager?

- by iGadget

I've been using KVM in combination with Virt-Manager and Remmina at a fair success up until now. The issue I need to solve now is to get audio from a virtualized Windows XP and make it audible on the Ubuntu 11.10 host. Remmina / RDP works for 'simple' audio (system sounds and such), but when the source gets trickier (e.g. Flash audio), Remmina / RDP messes up. So I figured I'd just connect to the machine directly using Virt-Manager. Unfortunately, it seems that even though I have successfully configured the AC97 audio device on WinXP, it's unable to get it's output to the Ubuntu host. This is probably because Virt-Manager uses VNC (and AFAIK, VNC doesn't transport audio). Does anyone know if there is a solution to fix this? I've heard of Spice, but the installation required so much voodoo last time I checked, I figured I'd let that solution boil to maturity a little longer ;) But perhaps there are other options I haven't thought of yet (which don't require switching to VirtualBox / VMware)...

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No options for sound output

- by Chef Flambe

Newbie here. Trying out Ubuntu for kicks. I have a 10.04 Ubuntu system dual booting on my Windows 7 laptop, fully updated, an Asus A53S. I have an external Samsung monitor/tv hooked up via HDMI. I'm getting sound from the external monitor fine when using Win7 but can't get anything with Ubuntu. I've read threads here about similar stuff and they say go check my sound options but I don't have any additional options than the one listed - Internal Audio. What else can I check/do?

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What trick will give most reliable/compatible sound alarm in a browser window for most browsers

- by Dirk Paessler

I want to be able to play an alarm sound using Javascript in a browser window, preferably with the requirement for any browser plugins (Quicktime/Flash). I have been experimenting with the tag and the new Audio object in Javascript, but results are mixed: As you can see, there is no variant that works on all browsers. Do I miss a trick that is more cross-browser compatible? This is my code: // mp3 with Audio object var snd = new Audio("/sounds/beep.mp3");snd.play(); // wav with Audio object var snd = new Audio("/sounds/beep.wav");snd.play(); // mp3 with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.mp3" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); // wav with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.wav" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); }

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Why is a FLAC encoded from a decoded MP3 bigger than the MP3?

- by Ryan Thompson

To be more precise than in the title, suppose I have a MP3 file that is 320 kbps. If I decompress it, then logically, all the data except for roughly 320 kilobits out of each second of audio should be redundant data, able to be compressed away. So, when I encode the decompressed file to FLAC, or any other lossless codec, why is it so much larger? On a related note, is it theoretically possible to losslessly recover the source mp3 audio from a decompressed wav? (I know the mp3 itself is lossy. I'm asking if it's possible to re-encode without any further loss.) EDIT: Let me clarify the related question, and the rationale behind it. Suppose I have a wav that was decompressed from an MP3 file (and assume I don't have the mp3 itself for some reason). If I don't want to lose any more quality, I can re-encode it with FLAC or any other lossless encoder and get a larger file just to maintain the same quality. Or, I can re-encode it to mp3 again and get the same size as the original but lose more data. Obviously, neither of these cases is ideal. I can either have the original size or the original quality, but not both (I mean the quality of the original mp3, not the original lossless source). My question is: Can we get both? Is it theoretically possible to recover the lossy compressed data from the lossy decompressed data, without losing even more? If it is possible, I could imagine a lossless compression algorithm that compresses the audio with FLAC. Then it also scans the audio for any signs of previous lossy compression, and if detected, recompresses it losslessly to the original lossy file. Then it keeps whichever file is smaller.

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Terrible noises from subwoofer of ACER Aspire 6930 with Realtek sound chip

- by OneWorld

After approximately 5-15 min of listening to music my subwoofer begins to make terrible noises. He's just "coughing". That began after 6 months I had this computer. Now I found out, that I can temporarily fix this problem by "restarting" the audio stream of the application that plays music. For example reloading last.fm page (reloads the flash file). Another way to reset the audio playback is switching the speaker configuration shown below in the screenshot. According to many posts on the internet like http://www.tomshardware.co.uk/forum/52918-20-acer-aspire-6935g-speaker-problem ACER support isn't any help Exchanging hardware doesn't fix the problem Even the later models have this problem Turning off the volume of the subwoofer is not an option to me. I still have warranty (I bought an extension of one year). I already tried about 15 versions of the Realtek driver with no success. I am not sure but MAYBE the problem did not occur on the original windows vista that was shipped with this computer. However, I removed the original windows for good reasons (english). What do you suggest me? Did anyone fix this problem? Maybe by writing a script which resets the audio streams every 5 minutes? Shall I take the effort to deal with the acer support until they give me another model? (I won't have a computer than for a longer time, will spend money on telephone hotlines (1,30 EUR / min)......) Here are additional infos, if they are any help: Windows 7 64 Bit (Original was Windows Vista Home Premium 32 Bit) All specs Audio driver version:

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How do I activate the F_LINE input in a transplanted HP chassis?

- by admin

I have an HP Pavilion Media Center PC chassis, vintage 2003 or so and I replaced the motherboard in it with a newer (vintage 2009) HP motherboard, M2N68-LA (Narra 5). I have scoured the internet trying to find pinouts for the motherboard to no avail. My question concerns the front panel audio, specifically Line In. The old chassis was built for AC97 but the new mobo is build for the newer HD audio standard. I figured out by comparison & experimentally how to connect the Mic & Headphone jacks to the HD audio header of the mobo by adding a manual switch to set the SENSE lines. Now all works fine for Mic & headphone. The old chassis also has a front panel Line In jack that the newer HP chassis does not have. However, the new mobo has a 4 pin white connector labeled F_LINE that I believe is a line input. Under Windows 7 I see the two Line Inputs in the mixer but I can't get one of them to become active. The 4 pin F_LINE connector uses the two middle pins for ground, and presumably the other two for left and right audio inputs. There are no pins for sensing on that connector. Can anyone tell me how to use that F_LINE input for the front panel, or how to activate it?

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Sound doesn't work anymore after replacing RAM

- by thejh

Hello, today, I replaced one old RAM module with two newer, bigger ones, but now, the sound doesn't seem to work anymore. Already ran alsaconf and it didn't help. Output of lspci for the audio device: 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) Subsystem: Giga-byte Technology Device a002 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 21 Region 0: Memory at f5100000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Masking: 00000000 Pending: 00000000 Capabilities: [6c] HyperTransport: MSI Mapping Enable+ Fixed+ Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel The audio device is onboard and has six configurable outputs, two or so are also capable of being an input (if I remember it correctly), but I don't know how to control it under linux. Does somebody know how/whether replacing the RAM could be related to my problem and/or how to fix it?

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How to boost playback volume in real time on media recorded with a very low volume.

- by L Marksman

I have never heard a satisfactory answer to this often misunderstood question, let me explain. Lets say I have a sound card and earphones/speakers that can play back audio loud enough in most cases. This is great but the problem is that you always find people who do not know how to record audio, from Youtube video's to music. So now you end up with a audio playback that only uses 10% or less of the capacity of your sound hardware, in vista/win 7 you will see this frequently in the mixer with the volume pushed up to max but the green sound level only goes up a millimeter or two. I am looking for (preferably free) software or a method to boost the sound level of any audio from any source in real time to use more of my hardware capacity similar to what VLC media player can do. Oh and please, do not tell me it is impossible. I am not trying to boost the volume past what my hardware is capable of, I am just trying to use my hardware's full capacity. Also please do not tell met to buy new hardware, I know I can use hardware amplification, I don't want to (like many others) spend money on a simple little problem like this. Thanks!

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Parsing SQLIO Output to Excel Charts using Regex in PowerShell

- by Jonathan Kehayias

Today Joe Webb ( Blog | Twitter ) blogged about The Power of Regex in Powershell, and in his post he shows how to parse the SQL Server Error Log for events of interest. At the end of his blog post Joe asked about other places where Regular Expressions have been useful in PowerShell so I thought I’d blog my script for parsing SQLIO output using Regex in PowerShell, to populate an Excel worksheet and build charts based on the results automatically. If you’ve never used SQLIO, Brent Ozar ( Blog | Twitter...(read more)

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Parsing SQLIO Output to Excel Charts using Regex in PowerShell

- by Jonathan Kehayias

Today Joe Webb ( Blog | Twitter ) blogged about The Power of Regex in Powershell, and in his post he shows how to parse the SQL Server Error Log for events of interest. At the end of his blog post Joe asked about other places where Regular Expressions have been useful in PowerShell so I thought I’d blog my script for parsing SQLIO output using Regex in PowerShell, to populate an Excel worksheet and build charts based on the results automatically. If you’ve never used SQLIO, Brent Ozar ( Blog...(read more)

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Playing NSF music in FMOD.net

- by Tesserex

So, as the title says, I want to be able to play NSF files using FMOD, because my project already uses FMOD and I'd rather not replace it. This will involve figuring out how existing players and emulators work and porting it. I haven't yet found an existing player that uses FMOD. My starting point is the MyNes source from http://sourceforge.net/projects/mynes/. There are two big steps between here and what I'm looking for. MyNes plays from a ROM, not NSF. So, I have to rip out the APU and get it to play NSF files. The MyNes APU uses SlimDX, so I have to convert that to FMOD.NET. I am really stuck about how to go about either of these, because I'm not that familiar with audio formats and it's hard finding resources online. So here are a few questions: From what I can tell from the NSF spec at http://kevtris.org/nes/nsfspec.txt, it's just contains the relevant memory section of the ROM, plus the header. If anyone can verify or correct this that would be great. The emulator APU uses data from the rest of the emulator to play, including things like cycle counts. I'm not sure what replaces this in a standalone player. Can't I just load all the music data at once into a stream and play it? Joining #1 and #2, does the header data from the NSF substitute for some of the ROM data in the emulator code? Using FMOD, will I be following the usercreatedsound example for loading a stream? And does this format count as PCM? Specifically MyNes says PCM8. Any tips on loading / playing the stream in FMOD are appreciated. As an aside, I don't really understand the loading / playing sections of the spec I linked at all. It seems to apply to 6502 systems / emulators only and not to my situation. I know it's a long shot for anyone here to have enough experience in this area to help, but anything you can provide is definitely appreciated. A link to an existing .NET library that does this would be even better, but I don't believe one exists.

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Playing NSF music in FMOD.net

- by Tesserex

So, as the title says, I want to be able to play NSF files using FMOD, because my project already uses FMOD and I'd rather not replace it. This will involve figuring out how existing players and emulators work and porting it. I haven't yet found an existing player that uses FMOD. My starting point is the MyNes source from http://sourceforge.net/projects/mynes/. There are two big steps between here and what I'm looking for. MyNes plays from a ROM, not NSF. So, I have to rip out the APU and get it to play NSF files. The MyNes APU uses SlimDX, so I have to convert that to FMOD.NET. I am really stuck about how to go about either of these, because I'm not that familiar with audio formats and it's hard finding resources online. So here are a few questions: From what I can tell from the NSF spec at http://kevtris.org/nes/nsfspec.txt, it's just contains the relevant memory section of the ROM, plus the header. If anyone can verify or correct this that would be great. The emulator APU uses data from the rest of the emulator to play, including things like cycle counts. I'm not sure what replaces this in a standalone player. Can't I just load all the music data at once into a stream and play it? Joining #1 and #2, does the header data from the NSF substitute for some of the ROM data in the emulator code? Using FMOD, will I be following the usercreatedsound example for loading a stream? And does this format count as PCM? Specifically MyNes says PCM8. Any tips on loading / playing the stream in FMOD are appreciated. As an aside, I don't really understand the loading / playing sections of the spec I linked at all. It seems to apply to 6502 systems / emulators only and not to my situation. I know it's a long shot for anyone here to have enough experience in this area to help, but anything you can provide is definitely appreciated. A link to an existing .NET library that does this would be even better, but I don't believe one exists.

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Oracle User Productivity Kit Translation

- by ultan o'broin

Oracle's customers just love the User Productivity Kit (UPK). I hear only great things about it from our international customers at the Oracle Usability Advisory Board meetings too. The UPK is the perfect solution for enterprise applications training needs (I previously reviewed a fine book about UPK btw). One question I am often asked is how source content created using the UPK can be translated into another language. I spoke with Peter Maravelias, Principal Product Strategy Manager for UPK about this recently. UPK is already optimized for easy source-target translation already. There is even a solution for re-recording demos. Here's what you can do to get your source content into another language: Use UPK's ability to automatically translate events and actions. UPK comes with XML templates that allow you to accomplish this in 21 languages with a simple publishing action switch. These templates even deal with the tricky business of using gender-based translations. Spanish localization template sample Japanese localization template sample Use the Import and Export localization features to export additional custom content in a format like XLIFF, easily handled by translation tools. You could also export and import in Word format. Re-record the sound (audio) files that go with the recordings, one per screen. UPK's granular approach to the sound files means that timing isn't an option. Retiming demos isn't required. A tip here with sound files and XLFF-exported custom content is to facilitate translation context by avoiding explicit references to actions going on in the screen recordings. A text based storyboard with screenshots accompanying the sound files should also be provided to the translators. Provide a glossary of terms too. Use the re-record option in UPK to record any demo from a translated application. This will allow all the translated UI labels to be automatically captured. You may be required to resize any action events here due to text expansion issues. Of course, you will need translated data in the translated application too, so plan for this in advance. However, source-target language skills aren't required for the re-recording. The UPK Player itself, of course, is also available from Oracle along with content and doc in 21 languages. The Developer and Setup is also translated in a smaller number of languages. Check the Oracle UPK website for latest details. UPK is a super solution for global enterprise applications training deployments allowing source content to be translated into multiple languages easily. See this post on the UPK blog for more insight too!

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Overwrite previous output in Bash instead of appending it

- by NES

For a bash timer i use this code: #!/bin/bash sek=60 echo "60 Seconds Wait!" echo -n "One Moment please " while [ $sek -ge 1 ] do echo -n "$sek " sleep 1 sek=$[$sek-1] done echo echo "ready!" That gives me something like that One Moment please: 60 59 58 57 56 55 ... Is there a possibility to replace the last value of second by the most recent so that the output doesn't generate a large trail but the seconds countdown like a real time at one position? (Hope you understand what i mean :))

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No output devices in sound settings - therefore, no sound

- by Kev Quirk

I've just performed a fresh install of Ubuntu 13.10, and I've noticed that sound isn't working. When I go to sound settings, I can see that there is absolutely no sound devices detected. However, I do have my speakers installed and turned on, plus my machine has an internal speaker as well. I've seen other posts where people mention that they have "Dummy device" listed, this isn't the problem here, the output device section is completely blank. Any help is appreciated. Thanks, Kev

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Error splicing file: Input/output error with a USB SD/HC card reader [closed]

- by PirateRussell

I recently got a new Droid Bionic, and it has the SD/HC card. Today, I got a new USB card reader that reads the HC format. When I plug it into my Linux Mint 11 (katya), Gnome 32-bit computer, I get this error every I try to copy or move any file off of the card onto my desktop: Error splicing file: Input/output error I don't have the problem on a Windows Vista computer. Any ideas??? Thanks in advance...

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Beat detection and FFT

- by Quincy

So I am working on a platformer game which includes music with beat detection. I am currently using a simple if the energy that is stored in the history buffer is smaller then the current energy there is a beat. The problem with this is that ofcourse if you use songs like rock songs where you have a pretty steady amplitude this isn't going to work. So I looked further and found algorithms splitting the sound into multiple bands using FFT. I then found this : http://en.literateprograms.org/Cooley-Tukey_FFT_algorithm_(C) The only problem I'm having is that I am quite new to audio and I have no idea how to use that to split the signal up into multiple signals. So my question is : How do you use a FFT to split a signal into multiple bands ? Also for the guys interested, this is my algorithm in c# : // C = threshold, N = size of history buffer / 1024 public void PlaceBeatMarkers(float C, int N) { List<float> instantEnergyList = new List<float>(); short[] samples = soundData.Samples; float timePerSample = 1 / (float)soundData.SampleRate; int sampleIndex = 0; int nextSamples = 1024; // Calculate instant energy for every 1024 samples. while (sampleIndex + nextSamples < samples.Length) { float instantEnergy = 0; for (int i = 0; i < nextSamples; i++) { instantEnergy += Math.Abs((float)samples[sampleIndex + i]); } instantEnergy /= nextSamples; instantEnergyList.Add(instantEnergy); if(sampleIndex + nextSamples >= samples.Length) nextSamples = samples.Length - sampleIndex - 1; sampleIndex += nextSamples; } int index = N; int numInBuffer = index; float historyBuffer = 0; //Fill the history buffer with n * instant energy for (int i = 0; i < index; i++) { historyBuffer += instantEnergyList[i]; } // If instantEnergy / samples in buffer < instantEnergy for the next sample then add beatmarker. while (index + 1 < instantEnergyList.Count) { if(instantEnergyList[index + 1] > (historyBuffer / numInBuffer) * C) beatMarkers.Add((index + 1) * 1024 * timePerSample); historyBuffer -= instantEnergyList[index - numInBuffer]; historyBuffer += instantEnergyList[index + 1]; index++; } }

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How to grep the output of youtube-dl?

- by mohtaw

The normal output for youtube-dl is the following [download] Downloading video #3 of 33 [youtube] WbWb0u8bJrU: Downloading webpage [youtube] WbWb0u8bJrU: Downloading video info webpage [youtube] WbWb0u8bJrU: Extracting video information [download] Resuming download at byte 107919109 [download] Destination: Lec 6.mp4 [download] 86.2% of 137.18MiB at 48.80KiB/s ETA 06:37 I need to show the first and last monitor the downloading I use the command youtube-dl -cit -f 18 URL | grep -e ETA -e "Downloading video #" It's not working only the first line is working while the last line is not, and I see the download is running as the file size grows

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"Ghost" output from locate?

- by Hailwood

I deleted some files, but they seem to still exist. Can anyone please explain the output of this: m@work:~$ locate cfx.css | xargs rm m@work:~$ locate cfx.css /var/www/wfox/hbr.co.nz/cfx/a/c/cfx.css /var/www/wfox/modules/gallery/cfx/a/c/cfx.css /var/www/wfox/phoenix/fp.co.nz/cfx/a/c/cfx.css /var/www/wfox/tmp.co.nz/cfx/a/c/cfx.css m@work:~$ cat /var/www/wfox/hbr.co.nz/cfx/a/c/cfx.css cat: /var/www/wfox/hbr.co.nz/cfx/a/c/cfx.css: No such file or directory

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I can't capture the output screen of a game tutorial

- by user12543

Hello, I'm trying to do a screen capture with gtk-recordmydesktop and other capture tools but for some reason, it will not capture an output screen while its running. It captures one screen but doesn't show the rest. I can capture my screen running if I use my flip camcorder but recordmydesktop will not work. Is there a way to capture live video playing on the desktop without using an outside camera? Brian

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Dumping .NET classes to debug output

Class to convert .NET classes into readable debug output with less effort...Did you know that DotNetSlackers also publishes .net articles written by top known .net Authors? We already have over 80 articles in several categories including Silverlight. Take a look: here.

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Why df and du show different output

- by Nischay

When I execute command df -h /tmp it says disk utilization is 100% but when it tried du -sh /tmp it says disk utilization is 2%. I want to know why these commands shows different output and these two commands work and what is the solution of this problem. /var is installed on it own file system .I am using Ubuntu 12.04 server edition on my vps account.Due to this problem utilization of /tmp 100% according to df some programs complain about free space in /tmp.

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