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Search found 5304 results on 213 pages for 'audio streaming'.

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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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• Video/audio streaming does not stop even if UIWebView is closed - iPad

  - by lostInTransit

  Hi I see this issue only on the iPad. The same things works as expected on the iPhone. I am opening the URL from my application in a UIWebView. If the URL is a normal web page, it works fine as expected. But if the URL is that of a remote video/audio file, the UIWebView opens the default player which is again good. Now when I dismiss the UIWebView (by clicking on the Done button on the player), the streaming doesn't stop and the audio/video keeps playing in the background (I cannot see it but it does keep playing in the background, can hear it). The UIViewController in which the webview was created is also dealloced (I put in a log statement in the dealloc method) but the streaming doesn't stop. Can someone please help me out on why this could be happening? And how can I stop the audio/video streaming when the UIWebView is closed? Thanks.

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• Using openssl encryption for Apple's HTTP Live Streaming

  - by Rob

  Has anyone had any luck getting encrypted streaming to work with Apple's HTTP Live Streaming using openssl? It seems I'm almost there but my video doesn't play but I don't get any errors in Safari either (like "Video is unplayable" or "You don't have permission to play this video" when I got the key wrong). #bash script: keyFile="key.txt" openssl rand 16 > $keyFile hexKey=$(cat key.txt | hexdump -e '"%x"') hexIV='0' openssl aes-128-cbc -e -in $fileName -out $encryptedFileName -p -nosalt -iv ${hexIV} -K ${hexKey} #my playlist file: #EXTM3U #EXT-X-TARGETDURATION:000020 #EXT-X-MEDIA-SEQUENCE:0 #EXT-X-KEY:METHOD=AES-128,URI="key.txt" #EXTINF:20, no desc test.ts.enc #EXT-X-ENDLIST I was using these docs as a guide: http://tools.ietf.org/html/draft-pantos-http-live-streaming

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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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• 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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• How create a peer-to-peer Video Streaming WEB app ??

  - by xRobot

  Hi guys! I was wondering if it's possibile to create a web-client that can connect two users and then it sends their webcam stream to each other. This should happen without using Red5 or any other flash/media server. What protocol can I use? What api etc...

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• How can I listen to FM Radio streaming without connecting to the internet?

  - by Vicheanak

  Many phones has the functionality to listen to FM radio without connecting to the internet. Just wondering that how can I do this on a computer? Please give me some advices, thanks a lot.

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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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