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• Solutions for exporting a remote desktop app (display and audio)

  - by Richard

  I'm looking for a solution that will allow me to export a desktop app running on a server to a client machine. The server is ideally Linux, the desktop is Windows (+Mac for icing on the cake). The export should be encrypted and I need to support multiple clients from one server. I only want to export an individual app, not a whole desktop, and ideally am looking for open source solutions. The obvious, cheapest, simplest choice is to use X tunnelled over ssh (e.g using Xming on the desktop) but X doesn't support audio. What are the alternatives? Or is there a way to support audio using X or in parallel to X? Thanks

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• FreePBX: Asterisk in the Cloud (EC2) Audio Problems

  - by neezer

  Please pardon the newbie question, but I can't seem to figure this out. I followed the Voxilla's tut to the tee: http://voxilla.com/2009/10/15/voxill...p-by-step-1457 But in making calls, my softphones connect, yet no audio (in either direction). I know from poking around the forums that this is generally caused by two factors: NAT and audio codecs. I (being new to the arena), however, don't know which. I believe I have Asterisk and the clients restricted to just ulaw, and I also believe I have the correct ports open, and my externip set correctly (I think the Voxilla AMI does this automatically, since it's in the cloud). I'm a bit lost. I'd be happy to post whatever configuration files that might help, provided you tell me where they are on the filesystem. But like I said before, this is effectively a vanilla install of Voxilla's own FreePBX AMI. I'd appreciate any help or guidance here. Thanks!

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• FreePBX: Asterisk in the Cloud (EC2) Audio Problems

  - by neezer

  Please pardon the newbie question, but I can't seem to figure this out. I followed the Voxilla's tut to the tee: http://voxilla.com/2009/10/15/voxill...p-by-step-1457 But in making calls, my softphones connect, yet no audio (in either direction). I know from poking around the forums that this is generally caused by two factors: NAT and audio codecs. I (being new to the arena), however, don't know which. I believe I have Asterisk and the clients restricted to just ulaw, and I also believe I have the correct ports open, and my externip set correctly (I think the Voxilla AMI does this automatically, since it's in the cloud). I'm a bit lost. I'd be happy to post whatever configuration files that might help, provided you tell me where they are on the filesystem. But like I said before, this is effectively a vanilla install of Voxilla's own FreePBX AMI. I'd appreciate any help or guidance here. Thanks!

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• Audio input problem in Ubuntu 9.10

  - by Andrea Ambu

  My audio input is a mix of my mic output and my sound card output. I'd like it to be just my mic output. I was able to do so in Ubuntu 9.04 but the interface is 9.10 is totally changed and I tried every my creativity was able to think. It's really annoying when talking to other people over the internet because they keep hearing their voice back. I'm not sure I explained it in clear way so I'll give you an example: What I do: I put an mp3 on play or a video on youtube then open a recorder and start to talk on my mic. What happens: both my voice and audio from mp3/youtube get reordered, even if I put headphones volume to 0 (via hardware). What I'd like to happen: Only my voice should be recorded. I'm sure I'm missing some technical term, but that's the problem and I'd like to solve it in Ubuntu 9.10, any idea?

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• Audio problems with asus notebook with Bluetooth and usb devices in win 7

  - by QuickSilver

  My notebook is Asus P53E - core i5 Windows 7 installed Audio from PC speakers and headphone is distorted when i turn on bluetooth or some usb device plugged in. I belive this is a software issue. I tried updating my audio drivers but nothing help. Any help will be appreciated. Update: After a few days digging I found that this problem is causing by the asus sound enhancement application SonicFocus. The distortion stops while turning off sonic focus. Can anyone help me with a solution other than turning off SonicFocus

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• Screenflick Audio option in MacBook Pro

  - by John

  When after I shut down my MacBook Pro by holding the power button for a few sec, (which I found is bad for the computer, so I will not do anymore) I found that my speaker doesn't play until I plug in and out earphone into the machine. When my speaker is not working like this, and when I am on a random webcam chatting site like chatroulette.com, they can hear the music playing on my iTunes when I choose Screenflick Audio option in the Mic setting. But when the Speaker is working back again, they don't hear the music playing even when I do Screenflick Audio mode. How can I make it work? Also, how do you make the chatting partner hear my music playing on my computer while I talk to them (not via my speaker, since it's bad quality).

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• Windows audio service fails to automatically start after VirtualBox install

  - by humble_coder

  I'm having a completely nonsensical issue in Windows XP SP3. Basically my "Windows Audio" service no longer starts automatically. Despite being set to "Automatic" I have to manually go in and start it. This issue didn't start until the most recent update of VirtualBox, but I can't find anything on the forums related to this specific issue. I've tried reinstalling the RealTek drivers as well, in the event that that had something to do with it. Any assistance is most appreciated! EDIT 1: It is the host's audio that won't start. The update of Virtualbox was merely the "marker" of when these events started occurring. Given it's the only variable/change I'm assuming a correlation.

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• No Audio Output Device is installed

  - by mabho

  Hi, this is an intermitent problem in my Sony Vaio model PCG-5K1L. I keep on getting a "No Audio Output Device is installed" when hovering my loudspeaker icon in Windows Vista. I have tried System Device Manager Sound Realtek High Definition Album Update Driver Software. The update process went through, but nothing happens. Still Vista does not seem to recognize my audio software. The strange part is that out of nothing my sound card can resume working to stop again hours later... If someone has any clues to solve this, please, help. Thanks a lot.

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• Redirect audio from laptop to desktop over LAN

  - by Ram Rachum

  I want to be able to play a song on my laptop and have it sound through my desktop's (infinitely better) speakers. If you're familiar with Input Director: I want something that is to audio what Input Director is to mouse/keyboard. I want something that automatically redirects all audio from the laptop to the desktop in real time, and I want that solution to require, like Input Director, minimum maintenance. Beyond the initial setup, I don't want to have to babysit the program that does this. I want something that launches automatically with Windows and just works, and also allows me to cancel it whenever I want. And also doesn't go crazy when the laptop is turned on in a different network where the desktop computer isn't available. Any suggestions for such a program? (I use Windows XP on both computers.)

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• How can I select an audio output device in directshow

  - by Vibhore Tanwer

  I was wondering how I can select the output device for audio in directshow. I am able to get available audio output devices in directshow. But how can I make one of these to be audio output device. Its always going for the default audio device. I want to be able to output audio on my choice of device. I have been struggling through google but couldn't find anything useful. All I could get was this link but it doesn't really solve my problem. Any help will be really helpful for me.

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• Read the audio input level peak in Cocoa

  - by Kenneth Ballenegger

  I'm trying to make an audio-sensitive animation, and for that purpose, I'm looking for a way to look up the current audio level. I'm looking for the peak within a set amount of time. (Think the red bar that stays on for a second or so, on an audio meter.) I've searched around for for something like this, and the only thing I could find was how to read a movie's audio levels, and how Quartz Compositions have access to this thru their iTunes Visualizer protocol. I'm looking for a way to read this from the microphone, although I'm also interested if you know how to read this from an audio file. Thanks!

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• Chrome/Webkit audio tag bug?

  - by Ronald

  I'm trying to get HTML5's audio tag to work in Chrome. The following code works flawlessly in Firefox, any ideas why it isn't working in Webkit? <html> <head> <script type="text/javascript"> function init(){ audio = new Audio("chat.ogg"); audio.play(); } </script> </head> <body onload="init()"> </body> I should also note that I tried this with an mp3 as well. Regardless of what format, whenever .play() is called on audio, Chrome responds with "undefined".

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• Optical SPDIF audio from motherboard not working with receiver

  - by simon b

  Hi, I hope someone can help; I can't get my SPDIF optical out working through my receiver and all the responses I can see on the web assume you have a sound card, while I settled for the (seemingly high end) sound on my motherboard (Asus P7P55D-E PRO), which appears to limit some of my options. My set-up is a "new out of the box" one and is: *Windows 7 PC (using PowerDVD10 for DVDs/Blurays and Windows media player for music) *Asus P7P55D-E PRO motherboard - has 8-channel audio TRS jacks and SPDIF optical and coaxial out *An old Yamaha receiver, whose only multi-channel input options are optical in and 6 channel RCA in. However, it still can handle DTS and DD *Boston Acoustic Soundware XS 5.1 speakers I've currently got the SPDIF optical out from the motherboard connected to the in on my receiver, have SPDIF enabled in the sound menu and the light is glowing red down the fibre. But I'm getting no sound at all. What I want is to be able to play DVDs/BluRays in 5.1 but also to be able to play music in multi-channel mode (even though I know this will be "fake" multichannel; it's more about where I sit in the room and my requirement to use the sub because the Boston is a satellite/sub set-up) My questions are: *Will optical work at all for multi-channel? THe latest posts I can see suggest it does but some people seem to say optical only outputs stereo. Whom to believe? *Even if it does work, I've read that I have to disable AC-3 decoding, or make various other changes, which don't seem to be possible without the menu options that a sound-card brings. Is the motherboard-only option just too inflexible? *Although my SPDIF device is enabled in the sound menu, it insists under "Jack information" that it is a "rear panel RCA jack", when of course it is not (both TOSLINK and rCA jacks do exist). Has the PC just forgotten that it has an optical? *I think I could relatively easily connect the 8-channel 3.5mm TRS jacks to my receiver 6-ch input jacks by way of TRS/RCA cables, but would that not stop me from being able to play music from media-player in multi-channel mode, as I'm not sure the motherboard can cope *Or do I need to bite the bullet and buy a sound-card? And if so, how can I be sure the one I get doesn't have the same problem? Any thoughts gratefully received, Cheers, simon

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• Good Audio Splitter

  - by Jeremy White

  I need to get audio from my computer's headphone jack and push the output to 2 sets of speakers. I have tried using a cheap splitter from Fry's, but one set of speakers ends up acting as a microphone (!?) for the other set of speakers. What's the best way to split headphone output and get best quality with no interference on each set of speakers? I would, of course, also be interested in why the cheap splitter causes one set of speakers to start acting as a microphone.

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• Traktor Audio 2 DJ soundcard configuration

  - by Jaroslav

  I have a Traktor Audio 2 DJ USB sound card (the first version of what it's now called simply Traktor Audio 2) The problem in settings it only sees one output, when there should be two (I need that for Mixxx etc.) Also I want to be able set the sample rate to one of these: 44.1, 48, 88.2, 96 kHz or at least check which one is set. Additionally if possible setting the latency would be an advantage. Some info: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: TraktorAudio2 [Traktor Audio 2], device 0: Traktor Audio 2 [Traktor Audio 2] Subdevices: 1/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 $ cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfdcfc000 irq 45 1 [TraktorAudio2 ]: snd-usb-caiaq - Traktor Audio 2 Native Instruments Traktor Audio 2 (usb-0000:00:1d.7-8)

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• moving audio over a local network using GStreamer

  - by James Turner

  I need to move realtime audio between two Linux machines, which are both running custom software (of mine) which builds on top of Gstreamer. (The software already has other communication between the machines, over a separate TCP-based protocol - I mention this in case having reliable out-of-band data makes a difference to the solution). The audio input will be a microphone / line-in on the sending machine, and normal audio output as the sink on the destination; alsasrc and alsasink are the most likely, though for testing I have been using the audiotestsrc instead of a real microphone. GStreamer offers a multitude of ways to move data round over networks - RTP, RTSP, GDP payloading, UDP and TCP servers, clients and sockets, and so on. There's also many examples on the web of streaming both audio and video - but none of them seem to work for me, in practice; either the destination pipeline fails to negotiate caps, or I hear a single packet and then the pipeline stalls, or the destination pipeline bails out immediately with no data available. In all cases, I'm testing on the command-line just gst-launch. No compression of the audio data is required - raw audio, or trivial WAV, uLaw or aLaw encoding is fine; what's more important is low-ish latency.

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• Audio output from Silverlight

  - by leecarter

  I'm looking to develop a Silverlight application which will take a stream of data (not an audio stream as such) from a web server. The data stream would then be manipulated to give audio of a certain format (G.711 a-Law for example) which would then be converted into PCM so that additional effects can be applied (such as boosting the volume). I'm OK up to this point. I've got my data, converted the G.711 into PCM but my problem is being able to output this PCM audio to the sound card. I basing a solution on some C# code intended for a .Net application but in Silverlight there is a problem with trying to take a copy of a delegate (function pointer) which will be the topic of a separate question once I've produced a simple code sample. So, the question is... How can I output the PCM audio that I have held in a data structure (currently an array) in my Silverlight to the user? (Please don't say write the byte values to a text box) If it were a MP3 or WMA file I would play it using a MediaElement but I don't want to have to make it into a file as this would put a crimp on applying dynamic effects to the audio. I've seen a few posts from people saying low level audio support is poor/non-existant in Silverlight so I'm open to any suggestions/ideas people may have.

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• Pumpktris: The Tetris-Enabled Jack-o’-Lantern [Video]

  - by Jason Fitzpatrick

  You can carve a pumpkin, you might even go high-tech and wire it up with a few LEDs, but can you play Tetris on it? Check out this fully functional Tetris clone built into a jack-o’-lantern. The build comes to us courtesy of tinker Nathan at HaHaBird, who writes: One of my habits is to write down all the crazy, fleeting ideas I have, then go back to review later rather than judging right off the bat, or even worse, forgetting them. Earlier in the month I was looking through that idea notepad and found “Make Tetris Pumpkins” from sometime last year. My original plan had been to make forms to shape pumpkins into Tetris pieces as they grew, then stack them together for Halloween. Since Halloween was only a few weeks away and it was too late to start growing pumpkins, I thought “Why not make a pumpkin you can play Tetris on instead?” Watch the Pumpktris in action via the video above or hit up the link below to see exactly how he went about building it. Pumpktris [via Geek News Central] 6 Start Menu Replacements for Windows 8 What Is the Purpose of the “Do Not Cover This Hole” Hole on Hard Drives? How To Log Into The Desktop, Add a Start Menu, and Disable Hot Corners in Windows 8

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• Jack of all trades, master of none [closed]

  - by Rope

  I've got a question similar to this one: Is looking for code examples constantly a sign of a bad developer? though not entirely. I got off college 2 years ago and I'm currently struggling with a University study. Most likely I'll have to drop out and start working within the next couple of months. Now here's the pickle. I have no speciality what so ever. When I got out of college I had worked with C, C++ and Java. I had had an internship at NEC-Philips and got familiar with C# (.NET) and I taught myself how it worked. After college I started working with PHP, HTML,SQL, MySQL Javascript and Jquery. I'm currently teaching myself Ruby on Rails and thus Ruby. At my university I also got familiar with MATLAB. As you can see I've got a broad scope of languages and frameworks I'm familiar with, but none I know inside-out. So I guess this kinda applies to me: "Jack of all trades, master of none.". I've been looking for jobs and I've noticed that most of them require some years of experience with a certain language and some specifications that apply to that language. My question is: How do I pick a speciality? And how do I know if I'll actually enjoy it? As I've worked with loads of languages how would I be able to tell this is right for me? I don't like being tied down to a specific role and I quite like being a generalist. But in order to make more money I would need a specialisation. How would I pick something that goes against my nature? Thanks in advance, Rope.

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• Crash in audio resampler with some audio rates - FFMPEG PHP ( Solved! )

  - by Olaf Erlandsen

  i have a problem with this command( FFMPEG PHP ): Command: ffmpeg -i 62f76f050494f0ed6a5997967c00c0c0.wmv -ss 0 -t 99 -y -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 -f flv 62f76f050494f0ed6a5997967c00c0c0.flv Output: FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [asf @ 0xe81670]max_analyze_duration reached Input #0, asf, from '/var/www/resources/tmp/62f76f050494f0ed6a5997967c00c0c0.wmv': Metadata: WMFSDKVersion : 12.0.7601.17514 WMFSDKNeeded : 0.0.0.0000 IsVBR : 0 Duration: 00:00:50.87, bitrate: 2467 kb/s Stream #0.0: Audio: wmapro, 44100 Hz, stereo, flt, 256 kb/s Stream #0.1: Video: vc1, yuv420p, 950x460 [PAR 1:1 DAR 95:46], 25 fps, 25 tbr, 1k tbn, 25 tbc Output #0, flv, to '/var/www/resources/media/62f76f050494f0ed6a5997967c00c0c0.flv': Metadata: encoder : Lavf52.64.2 Stream #0.0: Video: flv, yuv420p, 950x460 [PAR 1:1 DAR 95:46], q=2-31, 200 kb/s, 1k tbn, 29.97 tbc Stream #0.1: Audio: libmp3lame, 11025 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 72 fps= 0 q=5.0 size= 0kB time=10.91 bitrate= 0.0kbits/s Multiple frames in a packet from stream 0 Warning, using s16 intermediate sample format for resampling frame= 141 fps=139 q=5.0 size= 103kB time=8.15 bitrate= 103.2kbits/s frame= 220 fps=144 q=5.0 size= 875kB time=10.92 bitrate= 656.6kbits/s frame= 290 fps=143 q=5.0 size= 1525kB time=13.74 bitrate= 909.1kbits/s frame= 356 fps=141 q=5.0 size= 2153kB time=15.99 bitrate=1103.1kbits/s frame= 427 fps=141 q=5.0 size= 2847kB time=18.70 bitrate=1247.0kbits/s frame= 497 fps=141 q=5.0 size= 3771kB time=21.16 bitrate=1460.0kbits/s frame= 575 fps=142 q=5.0 size= 4695kB time=24.61 bitrate=1563.0kbits/s frame= 639 fps=141 q=5.0 size= 5301kB time=26.80 bitrate=1620.2kbits/s frame= 703 fps=139 q=5.0 size= 5829kB time=29.36 bitrate=1626.2kbits/s frame= 774 fps=139 q=5.0 size= 6659kB time=32.39 bitrate=1684.0kbits/s frame= 842 fps=139 q=5.0 size= 7915kB time=35.27 bitrate=1838.6kbits/s frame= 911 fps=139 q=5.0 size= 9011kB time=37.98 bitrate=1943.4kbits/s frame= 975 fps=138 q=5.0 size= 9788kB time=40.59 bitrate=1975.3kbits/s frame= 1041 fps=138 q=5.0 size= 10904kB time=43.83 bitrate=2037.9kbits/s frame= 1115 fps=138 q=5.0 size= 11795kB time=46.24 bitrate=2089.8kbits/s frame= 1183 fps=138 q=5.0 size= 12678kB time=48.74 bitrate=2130.7kbits/s frame= 1247 fps=137 q=5.0 size= 13964kB time=51.36 bitrate=2227.5kbits/s frame= 1271 fps=136 q=5.0 Lsize= 15865kB time=58.86 bitrate=2208.1kbits/s video:15366kB audio:462kB global headers:0kB muxing overhead 0.238956% Problem: Warning, using s16 intermediate sample format for resampling I've also tried changing the parameter From -ar 44100 to -ar 11025 Thanks! Solution: Read this link: http://en.wikipedia.org/wiki/MP3#Bit_rate

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• Get to Know a Candidate (18 of 25): Jack Fellure&ndash;Prohibition Party

  - by Brian Lanham

  DISCLAIMER: This is not a post about “Romney” or “Obama”. This is not a post for whom I am voting.  Information sourced for Wikipedia.  NOTE:  I apologize for getting this entry out of order. Fellure (born October 3, 1931) is an American perennial political candidate and retired engineer.  Fellure has formally campaigned for President of the United States in every presidential election since 1988 as a member of the Republican Party. He asserts on his campaign website that his platform based on the 1611 Authorized King James Bible has never changed. As a candidate, he calls for the elimination of the liquor industry, abortion and pornography, and advocates the teaching of the Bible in public schools and criminalization of homosexuality. He has blamed the ills of society on those he has characterized as "atheists, Marxists, liberals, queers, liars, draft dodgers, flag burners, dope addicts, sex perverts and anti-Christians." After another run in 2008, Fellure initially ran for the Republican Party's 2012 presidential nomination. He then decided to seek the nomination of the Prohibition Party at the party's national convention in Cullman, Alabama The Prohibition Party (PRO) is a political party in the United States best known for its historic opposition to the sale or consumption of alcoholic beverages. It is the oldest existing third party in the US. The party was an integral part of the temperance movement. While never one of the leading parties in the United States, it was once an important force in the politics of the United States during the late 19th century and the early years of the 20th century. It has declined dramatically since the repeal of Prohibition in 1933. The party earned only 643 votes in the 2008 presidential election. The Prohibition Party advocates a variety of socially conservative causes, including "stronger and more vigorous enforcement of laws against the sale of alcoholic beverages and tobacco products, against gambling, illegal drugs, pornography, and commercialized vice." Fellure has Ballot Access in: LA Learn more about Jack Fellure and Prohibition Party on Wikipedia.

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• No audio in my ubuntu system

  - by hap497

  Hi, I am running ubuntu 9.10. But there is no sound in my environment. When I go to System-Preference, there is no 'sound' entry there. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: I82801AAICH [Intel 82801AA-ICH], device 0: Intel ICH [Intel 82801AA-ICH] Subdevices: 1/1 Subdevice #0: subdevice #0 $ lsmod Module Size Used by usb_storage 52576 3 binfmt_misc 8356 1 vboxvfs 34620 0 vboxvideo 1884 1 drm 159584 2 vboxvideo agpgart 34988 1 drm snd_intel8x0 30168 2 snd_ac97_codec 101216 1 snd_intel8x0 ac97_bus 1532 1 snd_ac97_codec snd_pcm_oss 37920 0 snd_mixer_oss 16028 1 snd_pcm_oss snd_pcm 75296 3 snd_intel8x0,snd_ac97_codec,snd_pcm_oss snd_seq_dummy 2656 0 snd_seq_oss 28576 0 iptable_filter 3100 0 snd_seq_midi 6432 0 ip_tables 11692 1 iptable_filter x_tables 16544 1 ip_tables snd_rawmidi 22208 1 snd_seq_midi snd_seq_midi_event 6940 2 snd_seq_oss,snd_seq_midi ppdev 6688 0 snd_seq 50224 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_mid i_event snd_timer 22276 2 snd_pcm,snd_seq snd_seq_device 6920 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi ,snd_seq psmouse 56500 0 serio_raw 5280 0 snd 59204 14 snd_intel8x0,snd_ac97_codec,snd_pcm_oss,snd_mixer_ oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_ti mer,snd_seq_device i2c_piix4 9932 0 parport_pc 31940 0 soundcore 7264 1 snd snd_page_alloc 9156 2 snd_intel8x0,snd_pcm vboxguest 143836 7 vboxvfs lp 8964 0 parport 35340 3 ppdev,parport_pc,lp pcnet32 32644 0 mii 5212 1 pcnet32 floppy 54916 0 ~:987:2$ lspci 00:00.0 Host bridge: Intel Corporation 440FX - 82441FX PMC [Natoma] (rev 02) 00:01.0 ISA bridge: Intel Corporation 82371SB PIIX3 ISA [Natoma/Triton II] 00:01.1 IDE interface: Intel Corporation 82371AB/EB/MB PIIX4 IDE (rev 01) 00:02.0 VGA compatible controller: InnoTek Systemberatung GmbH VirtualBox Graphics Adapter 00:03.0 Ethernet controller: Advanced Micro Devices [AMD] 79c970 [PCnet32 LANCE] (rev 40) 00:04.0 System peripheral: InnoTek Systemberatung GmbH VirtualBox Guest Service 00:05.0 Multimedia audio controller: Intel Corporation 82801AA AC'97 Audio Controller (rev 01) 00:06.0 USB Controller: Apple Computer Inc. KeyLargo/Intrepid USB 00:07.0 Bridge: Intel Corporation 82371AB/EB/MB PIIX4 ACPI (rev 0 00:0b.0 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB2 EHCI Controller ~:988:3$

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• Optical Audio out stuck on on a MacBook

  - by Clinton Blackmore

  Apple have made an interesting headphone port for the MacBook (and some other Intel Mac models). It works like a standard jack: nothing plugged in - audio comes out of built-in speakers headphones/external speakers plugged in - plays through headphones/external speakers but you can also use a special adapter (which trips a tiny microswitch) to get an optical audio out signal (which you can presumably plug into a nice surround-sound system). This is all well and good except when, like auto-tracking, it doesn't work, and you are left with nothing to adjust. Users report that they get no sound when they have nothing plugged in and that a red light emanates from the headphone port. If you go to System Preferences - Sound - Output, it will say (IIRC) "Optical Out" instead of "Internal Speakers". The only solution I'm aware of is to try to reset the switch by inserting and removing a set of headphones or a toothpick, perhaps wiggling it inside of the port, and hoping that you luck out and get it. Are there other ways to fix this problem? Does anyone know where the microswitch is or have a good technique to reset it?

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• I can play "test" sounds, but no other audio works

  - by Callum

  I'm running Windows XP, and last night my PC was infected by a frustrating virus (one of those viruses that won't let you open virus checkers, etc). I finally killed it 2 hours later, but it involved some heavy duty anti-dote. One side effect is my audio is now gone. Except it's not entirely gone, because when I open the Realtek HD Audio Manager in the task bar, I can play all the "test" sounds. The speakers, the sound card, etc, are therefore working fine. But things like YouTube or Windows Media Player, there's no sound. I'm guessing there's a setting that needs to be reconfigured somewhere.. but where? Maybe relevant: One thing I did do last night was "play" with the system registry. Any help would be greatly appreciated. Thanks. SOLVED! The two hour battle with my computer virus resulted in my computer permanently thinking it was in Safe Mode, regardless of how it booted up. I was able to "fix" this by following the post by hsandler in this thread: http://www.petri.co.il/forums/showthread.php?t=23032&page=2 I then rebooted.. and let me tell you, the Windows Startup music has never sounded so sweet. Thanks to all, especially James, whose advice gave me a major clue as to what the problem was.

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• How to slow down audio files?

  - by verve

  I need a program (with an easy learning curve) that lets me slow down mp3 (at the very least this format) music and audiobook files. The software needs to be able to slow down the audio at the chosen speeds without altering the pitch and accuracy of the words being pronounced. Perhaps like the language software "Byki Deluxe's" "SlowSound" feature? I'm learning a foreign language (German) and I find the speeds at which the books are being read too fast. I need to hear the pronunciation of each word much more clearly to learn how to pronounce the words myself. Is there such a product out there? Now, I know you can slow down stuff in VLC but it sounds really artificial. I need something that slows down audio files without altering the accuracy of the words being pronounced. It doesn't have to be freeware; ease of use and quality is more important to me. Win 7 64-bit. IE 8. Edit: Are there any software-for-pay like Audacity? Only the beta works in Win 7. Also, I'd prefer to be able to slow down a file live and not have to create a new file to use the feature.

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