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• 802.11n Audio Interference

  - by colithium

  Symptoms My audio is riddled with pops/crackles/glitches. Sometimes I swear the glitches sound exactly like the facebook messenger sound (best comparison I can give). Cause Using DPC Latency Checker, it reports the latency to be an abysmal 17,500µs (0.0175s). The first thing I did was disable my 802.11n wireless adapter. This immediately dropped the latency to a nice 250µs. When I re-enabled the adapter, it jumped right back up. I'm 99% certain that this is the cause of my audio glitches. Solution What can I do about it besides using wired Ethernet or buying a whole new adapter? My adapter is a Dell Wireless 1505 Draft 802.11n WLAN Mini-Card. To be honest, I've had nothing but trouble with the 802.11n standard and am contemplating just going back to g.

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• Change Audio title from English to Sinhalese using ffmpeg

  - by user330461

  I insert an extra Sound track in my video file and it works well. ffmpeg -i news.mov -i news.wav -map 0:0 -map 0:1 -map 1:0 -pass 1 -vcodec libx264 -preset fast -b 512k -minrate 512k -maxrate 512k -bufsize 512k -threads 0 -f mp4 -an -y /dev/null && ffmpeg -i news.mov -i news.wav -map 0:0 -map 0:1 -map 1:0 -pass 2 -acodec libfaac -ab 128k -ac 2 -vcodec libx264 -preset fast -b 512k -minrate 512k -maxrate 512k -bufsize 512k -threads 0 -f mp4 news.mp4 The default audio track come with the label "English" and I would like to give it a label "Sinhalese" The Second Audio track come up without a label as "track#1" and I would like to give that a label of "Tamil". How do I do that ?

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• How to generate a 8 bit per sample wav audio file in VLC

  - by Ahmed safan

  I'm using the following vlc command line to extract first 5 minutes of audio from video file "-I dummy -vvv --no-sout-video --sout-audio --no-sout-rtp-sap --no-sout-standard-sap --ttl=1 --sout-transcode-threads=5 --sout-transcode-high-priority --sout-keep --sout #transcode{acodec=s16l,channels=1,samplerate=8000,ab=64}:std{mux=wav,access=file,dst="c:\dest.wav"} "c:\originalvideo.mpg" --start-time=0 --stop-time=300 vlc://quit"; if ab=64 =64 k bits per second and samples per second=8 k samples then bits per sample=64/8=8 bits per sample but the problem is that the output file always has samples of 16 bits per sample. I know that sample can contain bits from 8 , 16, 24 to 32 bits per sample. i want to get 8 bits per sample file how can this be done ?

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• Can't find generic USB audio driver for a Samson COU1 USB microphone

  - by marcipollo

  I am unable to use a Samson USB CO1U microphone on a PC running XP, SP3. When I plug it into the USB port, Windows generates the sound indicating that it has found new hardware, and the green LED on the mic lights. But, it does not work, and the device manager reports that it cannot find a driver after searching. The same mic works on a Vista machine. Samson has no driver on their Web site, and insists that the generic audio driver in Windows should work. (http://www.samsontech.com/PRODUCTS/productpage.cfm?prodID=1810). I cannot find a generic USB audio driver at Microsoft.com. Can anyone help? Larry

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• No HDMI audio - Windows 8 - ASUS H81M-PLUS

  - by Paul Wright

  I have an issue with HDMI audio on Windows 8 using an ASUS H81M-PLUS motherboard (without an external GFX card). There are many forum posts advising you to go into playback devices and setting HDMI to be default - I have done this. To eliminate what works and what doesn't work: I have not been able to get sound from my HDTV using HDMI. I have used this HDMI cable with my PS3, so this cable should be fine. I am able to use the HDMI cable in extended mode, so that I have two monitors (including the TV), just no audio. This HDMI cable goes straight from the motherboard to the TV. Below I have included 'Device manager', and 'Playback Devices' (Sound). Device Manager Playback Devices, showing disabled and disconnected devices I am at a loss. I have uninstalled all drivers, and then rebooted and made windows look for the correct ones, made sure the HDMI device was default. Thanks, Paul

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• Can't find generic USB audio driver for a Samson COU1 USB microphone

  - by user10321

  I am unable to use a Samson USB CO1U microphone on a PC running XP, SP3. When I plug it into the USB port, Windows generates the sound indicating that it has found new hardware, and the green LED on the mic lights. But, it does not work, and the device manager reports that it cannot find a driver after searching. The same mic works on a Vista machine. Samson has no driver on their Web site, and insists that the generic audio driver in Windows should work. (http://www.samsontech.com/PRODUCTS/productpage.cfm?prodID=1810). I cannot find a generic USB audio driver at Microsoft.com. Can anyone help? Larry

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• Tool to bulk speed up/convert an audio file

  - by User1

  I want to listen to certain podcasts on my phone but I have two common problems: The audio is in some weird format (some don't play on my phone). The audio is slow. I want to use something like sox or avconv to bulk convert the files. Since this is just voice and going on a cell phone, small low-quality files would be best for me. I had some good success using avconv: avconv -i weird.wma normal.ogg Unforunately, this command creates an enormous ogg file and I can't get it play faster. Ideally, this particular file would play at 170% of the original speed.

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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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• 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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• 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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