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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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  • Audio 2 dj soundcard configuration

    - by Jaroslav
    I have an http://www.native-instruments.com/#/en/products/dj/audio-2-dj/ The problem in settings it only sees one outpout, 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 to get mixer applet for "Built-in Audio Analog Stereo"

    - by gerrit
    In pavucontrol, I can choose between RV620 HDMI Audio [Radeon HD 3400 Series] and Built-in Audio. When the former is enabled, videos on (among others) Youtube play way too fast, but this answer solved my problem (though I don't know why). However, when I use Built-in Audio instead of RV620 HDMI Audio [Radeon HD 3400 Series], the mixer in my applet appears to be disabled; the icon is replaced by a blank and changing the volume has no effect, as the applet apparently only relays to RV620 HDMI Audio [Radeon HD 3400 Series]. How do I get an applet to control the volume for Built-in Audio instead?

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  • Audio Streaming Latency

    - by killianmcc
    I'm writing a UDP local area network video chat system and have got the video and audio streams working. However I'm experiencing a little latency (about half a second) in the audio and was wondering what codecs would provide the least latency. I'm using NAudio (http://naudio.codeplex.com/) which provides me access to the following codecs for streaming; Speex Narrow Band (VBR) Speex Wide Band (16kHz)(VBR) Speex Ultra Wide Band (32kHz)(VBR) DSP Group TrueSpeech (8.5kbps) GSM 6.10 (13kbps) Microsoft ADPCM (32.8kbps) G.711 a-law (64kbps) G.722 16kHz (64kbps) G.711 mu-law (64kbps) PCM 8kHz 16 bit uncompressed (128kbps) I've tried them out and I'm not noticing much difference. Is there any others that I should download and try to reduce latency? I'm only going to be sending voice over the connection but I'm not really worried about quality or background noises too much. UPDATE I'm sending the audio in blocks like so; waveIn = new WaveIn(); waveIn.BufferMilliseconds = 50; waveIn.DeviceNumber = inputDeviceNumber; waveIn.WaveFormat = codec.RecordFormat; waveIn.DataAvailable += waveIn_DataAvailable; void waveIn_DataAvailable(object sender, WaveInEventArgs e) { if (connected) { byte[] encoded = codec.Encode(e.Buffer, 0, e.BytesRecorded); udpSender.Send(encoded, encoded.Length); } }

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  • Java algorithm for normalizing audio

    - by Marty Pitt
    I'm trying to normalize an audio file of speech. Specifically, where an audio file contains peaks in volume, I'm trying to level it out, so the quiet sections are louder, and the peaks are quieter. I know very little about audio manipulation, beyond what I've learnt from working on this task. Also, my math is embarrassingly weak. I've done some research, and the Xuggle site provides a sample which shows reducing the volume using the following code: (full version here) @Override public void onAudioSamples(IAudioSamplesEvent event) { // get the raw audio byes and adjust it's value ShortBuffer buffer = event.getAudioSamples().getByteBuffer().asShortBuffer(); for (int i = 0; i < buffer.limit(); ++i) buffer.put(i, (short)(buffer.get(i) * mVolume)); super.onAudioSamples(event); } Here, they modify the bytes in getAudioSamples() by a constant of mVolume. Building on this approach, I've attempted a normalisation modifies the bytes in getAudioSamples() to a normalised value, considering the max/min in the file. (See below for details). I have a simple filter to leave "silence" alone (ie., anything below a value). I'm finding that the output file is very noisy (ie., the quality is seriously degraded). I assume that the error is either in my normalisation algorithim, or the way I manipulate the bytes. However, I'm unsure of where to go next. Here's an abridged version of what I'm currently doing. Step 1: Find peaks in file: Reads the full audio file, and finds this highest and lowest values of buffer.get() for all AudioSamples @Override public void onAudioSamples(IAudioSamplesEvent event) { IAudioSamples audioSamples = event.getAudioSamples(); ShortBuffer buffer = audioSamples.getByteBuffer().asShortBuffer(); short min = Short.MAX_VALUE; short max = Short.MIN_VALUE; for (int i = 0; i < buffer.limit(); ++i) { short value = buffer.get(i); min = (short) Math.min(min, value); max = (short) Math.max(max, value); } // assign of min/max ommitted for brevity. super.onAudioSamples(event); } Step 2: Normalize all values: In a loop similar to step1, replace the buffer with normalized values, calling: buffer.put(i, normalize(buffer.get(i)); public short normalize(short value) { if (isBackgroundNoise(value)) return value; short rawMin = // min from step1 short rawMax = // max from step1 short targetRangeMin = 1000; short targetRangeMax = 8000; int abs = Math.abs(value); double a = (abs - rawMin) * (targetRangeMax - targetRangeMin); double b = (rawMax - rawMin); double result = targetRangeMin + ( a/b ); // Copy the sign of value to result. result = Math.copySign(result,value); return (short) result; } Questions: Is this a valid approach for attempting to normalize an audio file? Is my math in normalize() valid? Why would this cause the file to become noisy, where a similar approach in the demo code doesn't?

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  • How can I determine what codec is being used?

    - by jldugger
    This forum comment and this superuser answer suggest that the audio compression contributes to loss of quality. I've noticed that music played over my BT setup sometimes pitch bends in ways I don't remember the original doing, and I'm wondering if SBC has something to do with it. I'm using Ubuntu 10.10 on a Mac Pro, connecting to a pair of Sony DR-BT50's. Is there a way to inspect which Bluetooth codec pulseaudio is using, what codecs both ends of the bluetooth link support?

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  • How do audio based games such as Audiosurf and Beat Hazard work?

    - by The Communist Duck
    Note: I am not asking how to make a clone of one of these. I am asking about how they work. I'm sure everyone's seen the games where you use your own music files (or provided ones) and the games produce levels based on them, such as Audiosurf and Beat Hazard. Here is a video of Audiosurf in action, to show what I mean. If you provide a heavy metal song, you would get a completely different set of obstacles, enemies, and game experience from something like Vivaldi. What does interest me is how these games work. I do not know much about audio (well, data-side), but how do they process the song to understand when it is settling down or when it's speeding up? I guess they could just feed the pitch values (assuming those sorts of things exist in audio files) to form a level, but it wouldn't fully explain it. I'm either looking for an explanation, some links to articles about this sort of thing (I'm sure there's a term or terms for it), or even an open-source implementation of this kind of thing ;-) EDIT: After some searching and a little help, I found out about FFT (Fast Fourier Transform). This maybe a step in the right direction, but it is something that does not make any sense to me..or fits with my physics knowledge of waves.

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  • Flash Media Server Streaming: Content Protection

    - by dbemerlin
    Hi, i have to implement flash streaming for the relaunch of our video-on-demand system but either because i haven't worked with flash-related systems before or because i'm too stupid i cannot get the system to work as it has to. We need: Per file & user access control with checks on a WebService every minute if the lease time ran out mid-stream: cancelling the stream rtmp streaming dynamic bandwidth checking Video Playback with Flowplayer (existing license) I've got the streaming and bandwidth check working, i just can't seem to get the access control working. I have no idea how i know which file is played back or how i can play back a file depending on a key the user has entered. Server-Side Code (main.asc): application.onAppStart = function() { trace("Starting application"); this.payload = new Array(); for (var i=0; i < 1200; i++) { this.payload[i] = Math.random(); //16K approx } } application.onConnect = function( p_client, p_autoSenseBW ) { p_client.writeAccess = ""; trace("client at : " + p_client.uri); trace("client from : " + p_client.referrer); trace("client page: " + p_client.pageUrl); // try to get something from the query string: works var i = 0; for (i = 0; i < p_client.uri.length; ++i) { if (p_client.uri[i] == '?') { ++i; break; } } var loadVars = new LoadVars(); loadVars.decode(p_client.uri.substr(i)); trace(loadVars.toString()); trace(loadVars['foo']); // And accept the connection this.acceptConnection(p_client); trace("accepted!"); //this.rejectConnection(p_client); // A connection from Flash 8 & 9 FLV Playback component based client // requires the following code. if (p_autoSenseBW) { p_client.checkBandwidth(); } else { p_client.call("onBWDone"); } trace("Done connecting"); } application.onDisconnect = function(client) { trace("client disconnecting!"); } Client.prototype.getStreamLength = function(p_streamName) { trace("getStreamLength:" + p_streamName); return Stream.length(p_streamName); } Client.prototype.checkBandwidth = function() { application.calculateClientBw(this); } application.calculateClientBw = function(p_client) { /* lots of lines copied from an adobe sample, appear to work */ } Client-Side Code: <head> <script type="text/javascript" src="flowplayer-3.1.4.min.js"></script> </head> <body> <a class="rtmp" href="rtmp://xx.xx.xx.xx/vod_project/test_flv.flv" style="display: block; width: 520px; height: 330px" id="player"> </a> <script> $f( "player", "flowplayer-3.1.5.swf", { clip: { provider: 'rtmp', autoPlay: false, url: 'test_flv' }, plugins: { rtmp: { url: 'flowplayer.rtmp-3.1.3.swf', netConnectionUrl: 'rtmp://xx.xx.xx.xx/vod_project?foo=bar' } } } ); </script> </body> My first Idea was to get a key from the Query String, ask the web service about which file and user that key is for and play the file but i can't seem to find out how to play a file from server side. My second idea was to let flowplayer play a file, pass the key as query string and if the filename and key don't match then reject the connection but i can't seem to find out which file it's currently playing. The only remaining idea i have is: create a list of all files the user is allowed to open and set allowReadAccess or however it was called to allow those files, but that would be clumsy due to the current infrastructure. Any hints? Thanks.

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  • NST-475LX-BK NAS Enclosure Is It Any Good?

    - by peter
    Hi All, Has anyone got one of these NST-475LX-BK? It is this, Vantec NexStar LX Ultra Gigabit NAS Hard Drive Enclosure. http://www.vantecusa.com/front/product/view_detail/403#rev What are your experiences with it? What type of file system does it format the drive to? It has an eSata cable so it must support NTFS right? Or does it use ext3? How good is the media streaming? What type of files can it stream? MPEG 2? Thanks.

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  • Is it possible for a faulty processor to cause audio static/noise?

    - by Tom
    I have a Core 2 Extreme processor I received from a friend and have set up an XBMC box using it. However, I constantly get audio static whenever playing any music or videos. Here is a video of the sound: http://www.youtube.com/watch?v=SqKQkxYRVA4 I have tried replacing everything short of the case and the processor, including cables, audio interfaces, operating systems, ram, etc, leading me to think it might be either the case shorting out the motherboards I have tried or a faulty processor. Is it possible for a faulty processor to cause audio static/noise? Any feedback would be appreciated. Edit - Here's a list of things I have tried: Reinstalling OS Installing/upgrading/repairing PulseAudio/Alsa Installing alternate OSes, straight Ubuntu, Lubuntu, Xubuntu, Arch, Mint, Windows 7 Switching audio from the external card to internal Optical, audio out through HDMI, audio out through headphones Different ports on receiver (my main desktop sounds fine on the same sound system) Different optical cables Unplugging everything unnecessary from the motherboard (1 HD, 1 Stick of Ram, 1 Keyboard) Swapping out ram Swapping out the motherboard Replacing the Graphics Card (was replaced due to fan being noisy, not specifically for this problem) Different harddrives Swapping power supply Disabling onboard audio Switching Power Cable Plugging in through surge protector Plugging into different outlet on separate circuit

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  • Linux: set up media server to stream video via the Internet?

    - by Hassan
    How do I set up a media server in Linux which streams video over the internet? Is it easy to do this? I want a server that will actually encode video in real time to allow it to stream over sometimes slow or unreliable networks. Basically, I want a server that works on the internet. I have a directory with a bunch of video files, and want to make this accessible to myself remotely. For other situations, I found great and useful software (such as the PS3 media server). I'd like to find something equally as useful for streaming video over the internet.

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  • [SOLVED} How do I restore my audio after uninstalling Ventrilo?

    - by Marcx
    Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. Update: Here's a sample of how I hear others people voices.. Audio Sample After some tests, also the desktop has the same problem. (I tried TeamSpeak3) Here are some details on my laptop and desktop Laptop Dell Studio 1555 Core 2 Duo P8600 2.4Ghz 4Gb Ram Dual Channel Ati HD 4570 512Mb dedicated (up to 2048) IDT High Definition Audio Desktop Motherboard Asus P5KPL-AM Dual Core CPU E5200 2.50Ghz 2x2GB PC6400 Dual Channel Ati Radeon HD 4650 512MB VIA High Definition Audio Both computers have Windows 7 Professional 64Bit. So how do I restore my audio? SOLVED The problem was in router firmware, there was a bug that recognized VoIP traffic as a DOS attack and the router grambled every packet... I've installed the newest firmware and everything is fine :)

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  • Problem with WCF Streaming

    - by H4mm3rHead
    Hi, I was looking at this thread: http://stackoverflow.com/questions/1935040/how-to-handle-large-file-uploads-via-wcf I need to have a web service hosted at my provider where i need to upload and download files to. We are talking videos from 1Mb to 100Mb hence the streaming approach. I cant get it to work, i declared an Interface: [ServiceContract] public interface IFileTransferService { [OperationContract] void UploadFile(Stream stream); } and all is fine, i implement it like this: public string FileName = "test"; public void UploadFile(Stream stream) { try { FileStream outStream = File.Open(FileName, FileMode.Create, FileAccess.Write); const int bufferLength = 4096; byte[] buffer = new byte[bufferLength]; int count = 0; while((count = stream.Read(buffer, 0, bufferLength)) > 0) { //progress outStream.Write(buffer, 0, count); } outStream.Close(); stream.Close(); //saved } catch(Exception ex) { throw new Exception("error: "+ex.Message); } } Still no problem, its published to my webserver out on the interweb. So far so good. Now i make a reference to it and will pass it a FileStream, but the argument is now a byte[] - why is that and how do i get it the proper way for streaming? Edit My binding look like this: <bindings> <basicHttpBinding> <binding name="StreamingFileTransferServicesBinding" transferMode="StreamedRequest" maxBufferSize="65536" maxReceivedMessageSize="204003200" /> </basicHttpBinding> </bindings> I can consume it without problems, and get no errors - other than my input parameter has changed from a stream to a byte[]

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  • XML streaming with XProc.

    - by Pierre
    Hi all, I'm playing with xproc, the XML pipeline language and http://xmlcalabash.com/. I'd like to find an example for streaming large xml documents. for example, given the following huge xml document: <Books> <Book> <title>Book-1</title> </Book> <Book> <title>Book-2</title> </Book> <Book> <title>Book-3</title> </Book> <!-- many many.... --> <Book> <title>Book-N</title> </Book> </Books> How should I proceed to loop (streaming) over x-N documents like <Books> <Book> <title>Book-x</title> </Book> </Books> and treat each document with a xslt ? is it possible with xproc ?

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  • How to write C++ audio processing applications?

    - by cesko82
    Hi everyone, I'm an Electronics and Telecommunications student, next to my graduation. I'm gonna work on a project that involves my knowledge about DSP, music and audio in general. I allready know all the basic mathematic instruments and all the stuff I need to manage it, such as FFT, circular convolution ecc ecc. I want to learn C++ programming basically for one reason: it's very important in the professional world!!! And I think it's one of the most used to write applications working with audio, especially when it's about real time processing. Ok, after this small introduction I would like to know first, which are the most used libraries to work with audio processing in c++?? I was longer looking on the web but i couldn't find a lo of working stuff. (I work under linux with eclipse CDT enviroment). Then I would like to know if there are good sources to learn how to write some working code, such as for example how to write a simple low pass filter. Basically now i will not write real time applications, I would like to start from the processing of a WAV file, or even better an MP3 file, so basically on vectors of samples. Let's say that basically for now I would like to extract the waveform from an audio file, and save it to a thumbnail or to a PNG image. Ok, for now I think it's all I would need. Any ideas, advices, libraries, books, interesting sources about that? Thanks a lot in advance for any kind of answer. Giovanni.

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  • WCF and streaming requests and responses

    - by Cheeso
    Is it correct that in WCF, I cannot have a service write to a stream that is received by the client? My understanding is that streaming is supported in WCF for requests, responses, or both. Is it true that in all cases, the receiver of the stream must invoke Read ? I would like to support a scenario where the receiver of the stream can Write on it. Is this supported? Let me show it this way. The simplest example of Streaming in WCF is the service returning a FileStream to a client. This is a streamed response. The server code is like this: [ServiceContract] public interface IStreamService { [OperationContract] Stream GetData(string fileName); } public class StreamService : IStreamService { public Stream GetData(string filename) { FileStream fs = new FileStream(filename, FileMode.Open) return fs; } } And the client code is like this: StreamDemo.StreamServiceClient client = new WcfStreamDemoClient.StreamDemo.StreamServiceClient(); Stream str = client.GetData(@"c:\path\to\myfile.dat"); do { b = str.ReadByte(); //read next byte from stream ... } while (b != -1); (example taken from http://blog.joachim.at/?p=33) Clear, right? The server returns the Stream to the client, and the client invokes Read on it. Is it possible for the client to provide a Stream, and the server to invoke Write on it? In other words, rather than a pull model - where the client pulls data from the server - it is a push model, where the client provides the "sink" stream and the server writes into it. Is this possible in WCF, and if so, how? What are the config settings required for the binding, interface, etc? The analogy is the Response.OutputStream from an ASP.NET request. In ASPNET, any page can invoke Write on the output stream, and the content is received by the client. Can I do something similar in WCF? Thanks.

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  • Looping HTML5 audio on the iPhone

    - by Peeps
    I'm trying to make a HTML5 webapp that simply plays a sound over and over and over again, on my iPhone. I don't know any Obj-C to do it natively. What I have works fine, but the sound only plays once: <!DOCTYPE html> <html> <head> <title>noisemaker!</title> <meta http-equiv="content-type" content="text/html; charset=utf-8" /> <meta name="viewport" content="maximum-scale=1, minimum-scale=1, width=device-width, user-scalable=no" /> <meta name="apple-mobile-web-app-capable" content="yes" /> </head> <body> <audio src="noise.mp3" autoplay controls loop></audio> </body> </html> Is there a way to either bypass the QuickTime audio screen and loop it in the webpage, or get the QuickTime audio screen to loop the sound?

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  • Spring Integration 1.0 RC2: Streaming file content?

    - by gdm
    I've been trying to find information on this, but due to the immaturity of the Spring Integration framework I haven't had much luck. Here is my desired work flow: New files are placed in an 'Incoming' directory Files are picked up using a file:inbound-channel-adapter The file content is streamed, N lines at a time, to a 'Stage 1' channel, which parses the line into an intermediary (shared) representation. This parsed line is routed to multiple 'Stage 2' channels. Each 'Stage 2' channel does its own processing on the N available lines to convert them to a final representation. This channel must have a queue which ensures no Stage 2 channel is overwhelmed in the event that one channel processes significantly slower than the others. The final representation of the N lines is written to a file. There will be as many output files as there were routing destinations in step 4. *'N' above stands for any reasonable number of lines to read at a time, from [1, whatever I can fit into memory reasonably], but is guaranteed to always be less than the number of lines in the full file. How can I accomplish streaming (steps 3, 4, 5) in Spring Integration? It's fairly easy to do without streaming the files, but my files are large enough that I cannot read the entire file into memory. As a side note, I have a working implementation of this work flow without Spring Integration, but since we're using Spring Integration in other places in our project, I'd like to try it here to see how it performs and how the resulting code compares for length and clarity.

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  • Virtual audio driver (microphone)

    - by Dalamber
    Hello guys, I want to develop a virtual microphone driver. Please, do not say anything about DirectShow - that's not "the way". I need a solution that will work with any software including Skype and MSN. And DirectShow doesn't fit these requirements. I found AVStream Filter-Centric Simulated Capture Driver (avssamp.sys) in Windows 7 WDK. What I need is an audio part of it. By default it reads avssamp.wav and plays it. But this driver is registered as WDM streaming capture device. And I want it in Audio Capture Device. There are some posts in the web but they are all the same: http://www.tech-archive.net/Archive/Development/microsoft.public.development.device.drivers/2005-05/msg00124.html http://www.winvistatips.com/problem-installing-avssamp-audio-capture-sources-category-t184898.html I think registering this filter-driver as audio capture device will make Skype recognize it as a microphone and thefore I will be able to push any PCM file as if it goes from mic. If someone already faced this problem before, please help. Thanks in advance.

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  • android spectrum analysis of streaming input

    - by TheBeeKeeper
    for a school project I am trying to make an android application that, once started, will perform a spectrum analysis of live audio received from the microphone or a bluetooth headset. I know I should be using FFT, and have been looking at moonblink's open source audio analyzer ( http://code.google.com/p/moonblink/wiki/Audalyzer ) but am not familiar with android development, and his code is turning out to be too difficult for me to work with. So I suppose my questions are, are there any easier java based, or open source android apps that do spectrum analysis I can reference? Or is there any helpful information that can be given, such as; steps that need be taken to get the microphone input, put it into an fft algorithm, then display a graph of frequency and pitch over time from its output? Any help would be appreciated, thanks.

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  • How do I restore my audio after uninstalling Ventrilo?

    - by Marcx
    Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. So how do I restore my audio?

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  • How do I use different audio devices for different apps in Windows 8?

    - by Eclipse
    Besides switching the default audio device, how can I send the audio from one app (say x-box music) to one audio device, and another (say the video app) to another audio device? Edit: Looking further, I found this: http://channel9.msdn.com/Events/BUILD/BUILD2011/APP-408T At 16:16, he demonstrates exactly what I'm wanting to do, but when I go to the devices charm, I get a message: "You don't have any devices that can receive content from Music".

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