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  • Website does not automatically fit to iphone screen

    - by Ploetzeneder
    Hello, The following code does not fit onto the iphone screen; how do I have to define the viewport? <html> <body> <center> <div id="karteu" style="background: url('../customer/Karten/karte1.jpg') no-repeat left center;width:714px;height:540px;" > </div> </body> </html> Normally the site should be zoomed, so i first should see the website in small, and then be able to zoom that i see it in the original size, but in my case it does not, when i call the site, the zoom is, that the image has this original size already, and that i have to scroll, but i dont want to scroll,...i want to use the normal safari mobile zoom and then scroll The solution at the bottom does not zoom anything. I want to see the overview of the image at the beginning. Then i want to be able to zoom with the normal safari zoom functions,..

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  • Linux application that bundles multiple incoming audio and video streams into one container file?

    - by StackedCrooked
    I've been assigned to implement a video on-demand service for a local university. Different aspects of the lectures (video, audio, screen cast, white board) will be recorded. During a lecture all these data streams arrive at one Linux server. This server should transcode and bundle all these streams into one container (Matroska) file. My options seem to be: Write a GStreamer application do something with FFMPEG do something with VLC ...? Has anyone done something similar in the past? Can you recommend something? Edit For those interested, here are a few of my findings: Matroska is not a good format for streaming (it's possible, but it's not its primary intent) For Flash streaming you can use MPEG4 If you want to combine different videos into one video where each subvideo occupies a rectangular portion of the total screen, then this GStreamer script is useful (I found it on this blog post). Desktop capture works fine with VLC

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  • VST plugin : using FFT on audio input buffer with arbitrary size, how ?

    - by Led
    I'm getting interested in programming a VST plugin, and I have a basic knowledge of audio dsp's and FFT's. I'd like to use VST.Net, and I'm wondering how to implement an FFT-based effect. The process-code looks like public override void Process(VstAudioBuffer[] inChannels, VstAudioBuffer[] outChannels) If I'm correct, normally the FFT would be applied on the input, some processing would be done on the FFT'd data, and then an inverse-FFT would create the processed soundbuffer. But since the FFT works on a specified buffersize that will most probably be different then the (arbitrary) amount of input/output-samples, how would you handle this ?

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  • Successfully concatenating multiple videos

    - by wiseguydigital
    My mission is to create videos out of old web slideshows. To start with I have jpegs and audio files that worked as Flash slideshows in an old system, structured such as this: Audio structure my_audio_1.mp3 (this file is a 3 second mp3 of silence) my_audio_2.mp3 my_audio_3.mp3 my_audio_4 etc... roughly 30 mp3s per slideshow Image structure my_image_1.jpg (this acts as the opening slide) my_image_2.jpg my_image_3.jpg my_image_4. etc... roughly 30 images per slideshow. As there are almost 100 slideshows that must be converted to video, I have created a web-based interface using PHP to automate the process, that sits on a local system and attempts to combine the files using shell_exec(). The process uses the following workflow: Loop through each slide and make an avi or mpeg. So for instance my_mini_video_2.avi would be a video that consists of my_image_2.jpg and has a soundtrack of my_audio_2.mp3. This slide would last the length of my_audio_2.mp3. Join / stitch / concat all of the mini videos to create the final video (Using a combination of cat and either mencoder or ffmpeg (I have also tried avimerge but to no avail). Transcode the new 'master' video to various formats such as flv etc. I thought this would be simple and have been close on many occasions but it still won't work. I can't get past stage 2 as I can't get a perfect 'master' video. I have now experimented with Mencoder, FFMpeg and seem to have been through every combination I can think of. The problem is that the audio and visuals never sync, no matter what I try. Also, I have even tried created audio-less mini videos, joining the MP3s into one long MP3 using both cat and mp3wrap and then assigning the new long MP3 as the audio track, but this always produces either a very short file or a badly slowed down file and makes the female voiceover sound like a male boxer!!! There appears to be no problems at all with the original files. Does anybody have any experience in producing a video successfully from the same kind of starting point? Or any ideas on what I may be doing wrong? As an example: If I create silent mini-videos, and stitch them together into 'temp-master.mpg' and then join the MP3s together into single MP3 called 'temp-master-audio.mp3', the audio file's duration is 09:10 and the video file's duration is 08:35. They should be the same and the audio will seem sloooow. I haven't posted code as I have written lots and lots of combinations.

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  • Reproduce PIPE functionality in IronPython

    - by Muppet Geoff
    Hi, I am hoping some genious out there can help me out with this... I am using sox to merge and resample a group of WAV files, and pipe the output directly to the input of NeroAACEnc for encoding to AAC format. I originally ran the process in a script, which included: sox.exe d:\audio\1.wav d:\audio\2.wav d:\audio\3.wav -c 1 -r 22050 -t wav - | neroAacEnc.exe -q 0.5 -if - -of test.m4a This worked as expected. The '-' in the comand line translates as 'Pipe/redirect input/output (stdin/stdout)' - So Sox pipes to stdout, and NeroAACEnc reads from stdin, the | joins them together. I then migrated the whole solution to Python, and the equivalent command became: from subprocess import call, Popen, PIPE runwav = Popen(['sox.exe', 'd:\audio\1.wav', 'd:\audio\2.wav', 'd:\audio\3.wav', '-c', '1', '-r', '22050', '-t', 'wav', '-'], shell=False, stdout=PIPE) runm4b = call(['neroAacEnc.exe', '-q', '0.5', '-if', '-', '-of', 'test.m4a'], shell=False, stdin=runwav.stdout) This also worked like a charm, exactly as expected. Slightly more convoluted, but hey :) Well now I have to move it to IronPython, and the Subprocess module isn't available (the partial implementation that is, doesn't have Popen/PIPE support - plus it seems silly to add a custom library when there is probably a native alternative). I should mention here, that I opted for IronPython over C#, because I am comfortable with Python now - however, there is a chance of moving it again later to C# native, and I am using IronPython to ease myself into it :) I have no C# or .net experience. So far I have the following equivalent, that sets up the 2 processes: from System.Diagnostics import Process wav = Process() wav.StartInfo.UseShellExecute = False wav.StartInfo.RedirectStandardOutput = True wav.StartInfo.FileName = 'sox.exe' wav.StartInfo.Arguments = 'd:\audio\1.wav d:\audio\2.wav d:\audio\3.wav -c 1 -r 22050 -t wav -' wav.Start() m4b = Process() m4b.StartInfo.UseShellExecute = False m4b.StartInfo.RedirectStandardInput = True m4b.StartInfo.FileName = 'neroAacEnc.exe' m4b.StartInfo.Arguments = '-q 0.5 -if - -of test.m4a' m4b.Start() I know that these 2 processes start (I can see Nero and Sox in the task manager) but what I can't figure out (for the life of me) is how to string the two output/input streams together, as with the previous two solutions. I have searched and searched, so I thought I'd ask! If anyone knows either: How to join the two streams with the same net result as the Python and Commandline versions; or A better way to acheive what I am trying to do. I would be extremely grateful! Many thanks in advance, Geoff P.S. A code sample based off the above would be awesome :) or a specific code example of a similar process that I can easily translate... this has broked my brayne.

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  • How to use infinit live streams with JAVE library? (Java, ffmpeg)

    - by Ole Jak
    So I want to use JAVE to save mp3 radio stream to my File system. I have this code for file saving but what shall I do to save a stream (stop on timer for ex) File source = new File("source.wav"); File target = new File("target.mp3"); AudioAttributes audio = new AudioAttributes(); audio.setCodec("libmp3lame"); audio.setBitRate(new Integer(128000)); audio.setChannels(new Integer(2)); audio.setSamplingRate(new Integer(44100)); EncodingAttributes attrs = new EncodingAttributes(); attrs.setFormat("mp3"); attrs.setAudioAttributes(audio); Encoder encoder = new Encoder(); encoder.encode(source, target, attrs);

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  • XNA - Mouse coordinates to word space transformation

    - by Gabriel Butcher
    I have a pretty annoying problem. I would like to create a drawing program, using winform + XNA combo. The most important part would be to transform the mouse position into the XNA drawn grid - I was able to make it for the translations, but it only work if I don't zoom in - when I do, the coordinates simply went horrible wrong. And I have no idea what I doing wrong. I tried to transform with scaling matrix, transform with inverse scaling matrix, multiplying with zoom, but none seems to work. In the beginning (with zoom value = 1) the grid starts from (0,0,0) going to (Width, Height, 0). I was able to get coordinates based on this grid as long as the zoom value didn't changed at all. I using a custom shader, with orthographic projection matrix, identity view matrix, and the transformed world matrix. Here is the two main method: internal void Update(RenderData data) { KeyboardState keyS = Keyboard.GetState(); MouseState mouS = Mouse.GetState(); if (ButtonState.Pressed == mouS.RightButton) { camTarget.X -= (float)(mouS.X - oldMstate.X) / 2; camTarget.Y += (float)(mouS.Y - oldMstate.Y) / 2; } if (ButtonState.Pressed == mouS.MiddleButton || keyS.IsKeyDown(Keys.Space)) { zVal += (float)(mouS.Y - oldMstate.Y) / 10; zoom = (float)Math.Pow(2, zVal); } oldKState = keyS; oldMstate = mouS; world = Matrix.CreateTranslation(new Vector3(-camTarget.X, -camTarget.Y, 0)) * Matrix.CreateScale(zoom / 2); } internal PointF MousePos { get { Vector2 mousePos = new Vector2(Mouse.GetState().X, Mouse.GetState().Y); Matrix trans = Matrix.CreateTranslation(new Vector3(camTarget.X - (Width / 2), -camTarget.Y + (Height / 2), 0)); mousePos = Vector2.Transform(mousePos, trans); return new PointF(mousePos.X, mousePos.Y); } } The second method should return the coordinates of the mouse cursor based on the grid (where the (0,0) point of the grid is the top-left corner.). But is just don't work. I deleted the zoom transformation from the matrix trans, as I didnt was able to get any useful result (most of the time, the coordinates was horrible wrong, mostly many thousand when the grid's size is 500x500). Any idea, or suggestion? I trying to solve this simple problem for two days now :\

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  • Trouble installing ubuntu server on virtualbox (osx)

    - by audio.zoom
    Hello all, I'm trying to install lucid lynx 10.04.2 server on a virtualbox on snow leopard. I have 2 server iso files freshly downloaded one i386 and one 64bit. When I try to start the virtual machine with either one set to be the cd drive I'm getting the same error: Failed to open a session for the virtual machine Ub. Failed to load VMMR0.r0 (VERR_SUPLIB_OWNER_NOT_ROOT). Unknown error creating VM (VERR_SUPLIB_OWNER_NOT_ROOT). Couldn't find anything on it on google so I'm trying to see if anyone else has dealt with this issue. Thanks much in advance! edit: just downloaded the 32bit desktop edition to same avail edit2: ran Disk Utility' replair permissions then restarted. New error VERR_SUPLIB_WORLD_WRITABLE (instead of VERR_SUPLIB_OWNER_NOT_ROOT)

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  • Does any economically-feasible publicly available software compare audio files to determine if they are dupes?

    - by drachenstern
    In the vein of this question http://unix.stackexchange.com/questions/3037/is-there-an-easy-way-to-replace-duplicate-files-with-hardlinks is there any software that will automatically parse a library of my songs and find the ones that really are duplicates that one can be eliminated? Here's an example: My brother used to be a huge fan of remixing CDs. He would take all of his favorite tracks and put them on one. Then he would use my computer to read them in. So now I have like 6 copies of Californication on my HDD, and they're all a few bytes difference overall. I have hundreds of songs in my library like this. I want to trim them down to having uniques. They don't all have correct ID3 tags, so figuring out that Untitled(74).mp3 is the same as californication.mp3 is the same as whowrotethis.mp3 is tricky. I do NOT want to consider a concert album and a studio album rip to be the same (if I just did artist/title matching I would end up with this scenario, which doesn't work for me). I use Windows (pick your platform) and will be getting an OSX box later in the year. I'll run Linux if that's what it takes to get it organized. I have unprotected AAC and mp3 files. Bonus points for messing with WAV or MIDI and bonus points for converting from those into MP3 (I can always use Audacity and LAME to convert later if I know they match or to convert ahead of time if that will make things easier). Are there any suggestions, or do I need to goto Programmers or SO and build a list of requirements for comparing these things and write the software myself?

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  • How does Amarok rip Audio CDs (in Ubuntu Lucid)?

    - by Hanno Fietz
    I'm in the process of moving my CD collection into my Amarok library. Mostly, it works great. Sometimes however, the process just hangs forever. The problem seems to occur at random (i. e. often, but not always at the same disk/track) and the consequences range from none (successful after cancel/retry) to Amarok's internal db becoming completely messed up. I would like to investigate and file a proper bug report or find a fix / workaround, but I don't understand how Amarok does the ripping. When all is working, there's a lame process encoding to a temporary file, which appears in my collection once it's finished. When the process hangs, that lame command is still there, but waiting forever for data on stdin, which seems to come from a third process. That seems to be kio_audiocd, but I don't know whether that's correct and what it's supposed to do. What's going on?

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  • Slowdown upon router/modem setup change

    - by Ollie Saunders
    I’ve been using a Belkin FSD7632-4 modem router to connect to my TalkTalk provided ADSL internet connection for some time and been pretty happy with it. Recently, however, the connection has been failing and I decided to get a ASUS RT-N16 instead, which is also a much more capable router generally. The ASUS RT-N16 doesn’t come with a modem built-in so I purchased as Zoom modem as well. I’ve set them both up and am using them to post this message. But I’m a bit miffed to find that I get a significantly and consistently slower downstream rate from the new configuration than with the old Belkin. Belkin modem router: downstream: 3.45 mbps upstream: 0.73 mbps ASUS router + Zoom modem: downstream: 2.71 mbps upstream: 0.66 mbps Any ideas why this is? The really weird thing about this is that the Zoom supports ADSL2 and ADSL2+ but I don’t think the old Belkin does. At first I thought it might be due to the Zoom modem being limited to PPPoE instead of PPPoA, which my ISP supports, but then I tried using PPPoE with the Belkin and that still gave a high speed. I’m using VC-Mux encapsulation with both. VPI of 0 and VCI of 38. I pulled this data off the Zoom: Mode: ADSL2 Line Coding: Trellis On Status: No Defect Link Power State: L0 Downstream Upstream SNR Margin (dB): 12.3 11.8 Attenuation (dB): 43.0 24.9 Output Power (dBm): 12.9 0.0 Attainable Rate (Kbps): 3936 844 Rate (Kbps): 3194 840 MSGc (number of bytes in overhead channel message): 59 10 B (number of bytes in Mux Data Frame): 99 14 M (number of Mux Data Frames in FEC Data Frame): 2 16 T (Mux Data Frames over sync bytes): 1 8 R (number of check bytes in FEC Data Frame): 8 8 S (ratio of FEC over PMD Data Frame length): 1.9833 9.0594 L (number of bits in PMD Data Frame): 839 219 D (interleaver depth): 32 2 Delay (msec): 15 4 Super Frames: 15808 14078 Super Frame Errors: 0 4294967232 RS Words: 513778 111753 RS Correctable Errors: 126 4294967238 RS Uncorrectable Errors: 0 N/A HEC Errors: 0 4294967279 OCD Errors: 0 0 LCD Errors: 0 0 Total Cells: 1920175 237597 Data Cells: 205993 392 Bit Errors: 0 0 Total ES: 0 0 Total SES: 0 0 Total UAS: 34 0

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  • how to stream audio and video files, but use any media player on Windows (without using Windows file

    - by RamyenHead
    I want to access and play media files on machine S (Windows XP) from machine C (Windows XP). Using Windows File Sharing ("share this folder" stuff), if it works, I would share the folder containing media files on machine S, and I would be able to play media files, sitting in front of C, using any media player I want. Windows somehow ensures that the remote files behave like local files. But Windows file sharing won't work for me, is there any alternative? If two machines were both Linux, I would install an SSH server on S and use Nautilus from C to access and play media files. The reason why I can't use Windows file sharing is, my campus use two different subnets, I have S and C on different subnets and it seems that the firewall governing the whole network in campus doesn't allow file sharing between different subnets. I tried changing Windows Firewall settings on S to allow C in, it still wouldn't work, so it must be the other firewall.

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  • How to split audio into multiple channels from optical S/PDIF or 1/8"?

    - by Josh M.
    I have a motherboard which has an optical S/PDIF output or 1/8". I'd like to "split" that signal into the appropriate channels so that I can then connect that to the wires behind my car's headunit which, in turn, run to the amp. The factory Bose amp just takes a single connector with a million wires running out of it, so that's why I would need to separate the signal into separate channels. On the other end there are four RCA connectors: front left, front right, rear left, rear right. The sub-woofer signal does not require an additional connection. Edit: Revised to include S/PDIF or 1/8".

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  • Hardware Mediaplayer display

    - by Eric Audio
    I'm looking for a keyboard or just a little display to attach on my keyboard or something like that, what will show me the music tracks i'm playing in windowsmedia player, itunes, etc. I did some research and the only thing I found are gaming keyboards, but i'm not shure if these show my music tracks. So my question: Does somebody knows a keyboard who show the music tracks or just a little display? Bye, Eric

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  • Quality wise, is Windows Media Audio 10 Professional equivalent to WMA?

    - by Louis
    I noticed that for encoding CD rips, Zune is still using WMA 9.2 instead of WMA 10 Pro. On a given file using the highest quality VBR settings looks like this: VBR Quality 98, 44 kHz, stereo 1-pass VBR On the same file if I use WMA 10 Pro, with the same settings, the resulting file is about 20% smaller. Using my ears, I'm unable to tell the difference, but I'm wondering if this was the goal of WMA 10 Pro (to be as good as WMA at a lower bitrate). Is the quality of a WMA 10 Pro file equal to that of a WMA 9.2 file encoded with the same settings?

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  • Show floor-plans online like a map

    - by Quora Feans
    Given a floor-plan, which is too big for any screen, even if it is a 17" one, how can I show it online like a map? It would need further functionality that a browser alone does not have (just zoom in/out the entire image won't do the trick). The image will be breaked down into smaller jpgs, so the user will not have to download the whole floorplan at once.It will need some zoom in/zoom out button, and some way or bookmarking position (x,y). open-source solutions prefered.

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  • How can I get Windows 7 to switch audio from a monitor (with built-in speakers) to headphones when t

    - by tnorthcutt
    I have an HP dv5t laptop running Windows 7 64 bit with an Acer H235H monitor connected to it via an HDMI cable. The monitor has built-in speakers, which are a huge improvement over the laptop's speakers. However, when I want to use headphones, right now, I have to connect them to the laptop, then right-click the sound icon in the task bar, select Playback Devices, right click the monitor, and disable it. Is there any way to get Windows 7 to automatically switch the output to the headphones when they're plugged in? That's the behavior that happens without the monitor attached (i.e. it will switch from the laptop speakers to headphones when headphones are plugged in). I have the same issue with a Sony Vaio laptop running Windows 7 64-bit and an identical monitor, for reference.

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  • How do I make an on-screen HUD in libgdx?

    - by Devin Carless
    I'm new to libgdx, and I am finding I am getting stumped by the simplest of things. It seems to want me to do things a specific way, but the documentation won't tell me what that is. I want to make a very simple 2d game in which the player controls a spaceship. The mouse wheel will zoom in and out, and information and controls are displayed on the screen. But I can't seem to make the mouse wheel NOT zoom the UI. I've tried futzing with the projection matrices in between Here's my (current) code: public class PlayStage extends Stage { ... public void draw() { // tell the camera to update its matrices. camera.update(); // tell the SpriteBatch to render in the // coordinate system specified by the camera. spriteBatch.setProjectionMatrix(camera.combined); spriteBatch.begin(); aButton.draw(spriteBatch, 1F); playerShip.draw(spriteBatch, 1F); spriteBatch.end(); } } camera.zoom is set by scrolled(int amount). I've tried about a dozen variations on the theme of changing the camera's projection matrix after the button is drawn but before the ship is, but no matter what I do, the same things happen to both the button and the ship. So: What is the usual libgdx way of implementing an on-screen UI that isn't transformed by the camera's projection matrix/zoom?

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  • Cocos2d-x CCFollow Zooming issue

    - by blakey87
    Hi I am currently building a cocos2d-x game which incorporates pinch zoom using CCLayerPanZoom class which can be found here The problem is basically when using CCFollow and zooming and out, it does'nt zoom on the actually followed node, so the camera appears to zoom towards the bottom left corner of the screen, when I would rather it zoom centrally on the followed node. If I could resolve this I would pretty darn happy. I converted a fix from the cocos2d objective C version in the CCfollow class to cocos2d-x which improved the issue,but if you look at the post in latter link you will see the guy is having the exact same problem, he gave up on fixing it sadly. I think its close but I don't really know what going on, hopefully someone out there has already faced and fixed this problem. My converted code is below. CCPoint p1 = ccpMult(m_obHalfScreenSize, m_pTarget->getScale() ); CCPoint p2 = ccpMult(m_pobFollowedNode->getPosition(), m_pTarget->getScale() ); CCPoint offect = ccpMult(ccpSub(p1, m_obHalfScreenSize), 0.5f); CCPoint tempPos = ccpAdd(ccpSub(p1, p2), offect); m_pTarget->setPosition(ccp(clampf(tempPos.x,m_fLeftBoundary,m_fRightBoundary), clampf(tempPos.y,m_fBottomBoundary,m_fTopBoundary))); I have attached before and after to hopefully make things more clear.

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  • 2D map/plane with nodes overlayed that supports panning, scaling and clicking on nodes

    - by garlicman
    I'm trying my hand at Android development and seem to be running into an invisible ceiling in trying to get what I want accomplished. Basically I'm trying to create an app that renders a 2D surface map that I can (pinch) zoom and pan. I'll have to place nodes on the surface of the map that will scale/zoom and pan in relation to the surface. I started out with a 2D ImageView approach and got as far as pinch zoom, pan and laying nodes as relative ImageViews, but all the methods I tried to get X,Y,W,H for the 2D surface were always off for some reason. Additionally, I was never able to scale the node ImageViews correctly, and as a result never got far enough to try and work out their X,Y scaled offset. So I decided to get back to 3D rendering. Conceptually pan/zoom is camera manipulation, so I don't have to mess with how to scale the 2D map or the nodes. But I need a starting point or sample to get me going that's close to what I'm trying to achieve. A sample on a translucent spinning cube isn't helping as much as I need it to. Any tips? Links, insults and sympathy are all welcome!

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  • Creating user UI using Flixel

    - by Jamie Read
    I am new to game development but familiar with programming languages. I have started using Flixel and have a working Breakout game with score and lives. What I am trying to do is add a Start Screen before actually loading the game. I have a create function that adds all the game elements to the stage: override public function create():void // all game elements { How can I add this pre-load Start Screen? I'm not sure if I have to add in the code to this create function or somewhere else and what code to actually add. Eventually I would also like to add saving, loading, options and upgrades too. So any advice with that would be great. Here is my main game.as: package { import org.flixel.*; public class Game extends FlxGame { private const resolution:FlxPoint = new FlxPoint(640, 480); private const zoom:uint = 2; private const fps:uint = 60; public function Game() { super(resolution.x / zoom, resolution.y / zoom, PlayState, zoom); FlxG.flashFramerate = fps; } } } Thanks.

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  • Square Reader Modified to Record Off Old Reel-to-Reel Tape [Video]

    - by Jason Fitzpatrick
    The Square Reader is a tiny magnetic credit card reader that has taken the mobile payment industry by storm. This clever hack dumps the credit card reading in favor of snagging the audio from old music reels. Evan Long was curious about whether the through-the-headphones interface of the Square Reader could be used to read audio data off old magnetic recordings. With a very small modification (he had to bend a metal tab inside the reader to allow the audio tape to slide through more easily) he was able to listen to and record audio off old reels. Watch the video above to see it in action or hit up the link below to read more about his project. iPod Meets Reel [via Make] HTG Explains: What Is Windows RT and What Does It Mean To Me? HTG Explains: How Windows 8′s Secure Boot Feature Works & What It Means for Linux Hack Your Kindle for Easy Font Customization

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  • which default.list should i modify for default applications and what are the differences between the 2

    - by damien
    I would like to add miro to the default application GUI in system settings/default applications.I added ;miro.desktopnext to all rhythmbox.desktop entries eventually discovering if it was not added to audio/x-vorbis+ogg=rhythmbox.desktop as audio/x-vorbis+ogg=rhythmbox.desktop;miro.desktop it would not appear in the system settings/default applications drop down list for audio. I can find default.list in either /etc/gnome/defaults.list or /usr/share/applications/defaults.list modifying either gives me the same results.What is the difference and which is the correct list to modify?

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