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  • Change the User Interface Language in Ubuntu

    - by Matthew Guay
    Would you like to use your Ubuntu computer in another language?  Here’s how you can easily change your interface language in Ubuntu. Ubuntu’s default install only includes a couple languages, but it makes it easy to find and add a new interface language to your computer.  To get started, open the System menu, select Administration, and then click Language Support. Ubuntu may ask if you want to update or add components to your current default language when you first open the dialog.  Click Install to go ahead and install the additional components, or you can click Remind Me Later to wait as these will be installed automatically when you add a new language. Now we’re ready to find and add an interface language to Ubuntu.  Click Install / Remove Languages to add the language you want. Find the language you want in the list, and click the check box to install it.  Ubuntu will show you all the components it will install for the language; this often includes spellchecking files for OpenOffice as well.  Once you’ve made your selection, click Apply Changes to install your new language.  Make sure you’re connected to the internet, as Ubuntu will have to download the additional components you’ve selected. Enter your system password when prompted, and then Ubuntu will download the needed languages files and install them.   Back in the main Language & Text dialog, we’re now ready to set our new language as default.  Find your new language in the list, and then click and drag it to the top of the list. Notice that Thai is the first language listed, and English is the second.  This will make Thai the default language for menus and windows in this account.  The tooltip reminds us that this setting does not effect system settings like currency or date formats. To change these, select the Text Tab and pick your new language from the drop-down menu.  You can preview the changes in the bottom Example box. The changes we just made will only affect this user account; the login screen and startup will not be affected.  If you wish to change the language in the startup and login screens also, click Apply System-Wide in both dialogs.  Other user accounts will still retain their original language settings; if you wish to change them, you must do it from those accounts. Once you have your new language settings all set, you’ll need to log out of your account and log back in to see your new interface language.  When you re-login, Ubuntu may ask you if you want to update your user folders’ names to your new language.  For example, here Ubuntu is asking if we want to change our folders to their Thai equivalents.  If you wish to do so, click Update or its equivalents in your language. Now your interface will be almost completely translated into your new language.  As you can see here, applications with generic names are translated to Thai but ones with specific names like Shutter keep their original name. Even the help dialogs are translated, which makes it easy for users around to world to get started with Ubuntu.  Once again, you may notice some things that are still in English, but almost everything is translated. Adding a new interface language doesn’t add the new language to your keyboard, so you’ll still need to set that up.  Check out our article on adding languages to your keyboard to get this setup. If you wish to revert to your original language or switch to another new language, simply repeat the above steps, this time dragging your original or new language to the top instead of the one you chose previously. Conclusion Ubuntu has a large number of supported interface languages to make it user-friendly to people around the globe.  And since you can set the language for each user account, it’s easy for multi-lingual individuals to share the same computer. Or, if you’re using Windows, check out our article on how you can Change the User Interface Language in Vista or Windows 7, too! Similar Articles Productive Geek Tips Restart the Ubuntu Gnome User Interface QuicklyChange the User Interface Language in Vista or Windows 7Create a Samba User on UbuntuInstall Samba Server on UbuntuSee Which Groups Your Linux User Belongs To TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips VMware Workstation 7 Acronis Online Backup DVDFab 6 Revo Uninstaller Pro FetchMp3 Can Download Videos & Convert Them to Mp3 Use Flixtime To Create Video Slideshows Creating a Password Reset Disk in Windows Bypass Waiting Time On Customer Service Calls With Lucyphone MELTUP – "The Beginning Of US Currency Crisis And Hyperinflation" Enable or Disable the Task Manager Using TaskMgrED

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  • How can I separate the user interface from the business logic while still maintaining efficiency?

    - by Uri
    Let's say that I want to show a form that represents 10 different objects on a combobox. For example, I want the user to pick one hamburguer from 10 different ones that contain tomatoes. Since I want to separate UI and logic, I'd have to pass the form a string representation of the hamburguers in order to display them on the combobox. Otherwise, the UI would have to dig into the objects fields. Then the user would pick a hamburguer from the combobox, and submit it back to the controller. Now the controller would have to find again said hamburguer based on the string representation used by the form (maybe an ID?). Isn't that incredibly inefficient? You already had the objects you wanted to pick one from. If you submited to the form the whole objects, and then returned a specific object, you wouldn't have to refind it later on since the form already returned a reference to that object. Moreover, if I'm wrong and you actually should send the whole object to the form, how can I isolate UI from logic?

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  • What Interface Toolkit is being recommended for Ubuntu on Nexus7/Mobile Devices?

    - by Baggers
    I understand this is a may be a very premature question given that the current build is for testing Ubuntu Core, but I have just bought a Nexus7 to join in with this Ubuntu on mobile adventure and can't help wanting to start writing some apps! I haven't really dabbled with either GTK or QT for touch apps yet and, having seen that Ubuntu TV is using Nux, I wondered what people on AskUbuntu-land would recommend. Hope someone out there know this! Cheers

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  • Java best practice Interface - subclasses and constants

    - by Taiko
    In the case where a couple of classes implements an interface, and those classes have a couple of constants in common (but no functions), were should I put this constant ? I've had this problem a couple of times. I have this interface : DataFromSensors that I use to hide the implementations of several sub classes like DataFromHeartRateMonitor DataFromGps etc... For some reason, those classes uses the same constants. And there's nowere else in the code were it is used. My question is, were should I put those constants ? Not in the interface, because it has nothing to do with my API Not in a static Constants class, because I'm trying to avoid those Not in a common abstract class, that would stand between the interface and the subclasses, because I have no functions in common, only a couple of constants (TIMEOUT_DURATION, UUID, those kind of things) I've read best practice for constants and interface to define constants but they don't really answer my question. Thanks !

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  • Is Objective C fast enough for DSP/audio programming

    - by morgancodes
    I've been making some progress with audio programming for iPhone. Now I'm doing some performance tuning, trying to see if I can squeeze more out of this little machine. Running Shark, I see that a significant part of my cpu power (16%) is getting eaten up by objc_msgSend. I understand I can speed this up somewhat by storing pointers to functions (IMP) rather than calling them using [object message] notation. But if I'm going to go through all this trouble, I wonder if I might just be better off using C++. Any thoughts on this?

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  • C# Audio - How to time stretch (different tempo, same pitch)

    - by heath
    I'm trying to make a winform app in C# (VS2008) that can load an mp3 (other formats would be nice, but mp3 at a minimum) and be able to adjust the playback speed (tempo) without affecting pitch. I really don't need any other audio effects. I tried using DirectShow but that doesn't seem to offer time stretch capabilities. I was able to incorporate irrklang but that does not seem to have the time stretch capability either. So now I've moved on to SoundTouch. That certainly has the capabilities but I'm very unclear on how to implement in C#. After a few days of this, about all I've accomplished is using DLLImport on the SoundTouch DLL and am able to successfully retrieve a version number. At this point, I'm not even sure if I can do what I'm trying to do with SoundTouch. Can anyone offer some guidance either on how to implement SoundTouch or a different library with the capabilities that I'm looking for? Thank you.

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  • Ruby/Rails Audio Conversion Plugins?

    - by coneybeare
    I am looking for a good gem/plugin to convert user-uploaded audio files to different formats. One format in particular that I am interested in is converting to Apple .caf with ima4 compression for inclusion in an iPhone app. I have been using afconvert on my mac for this so far, but I need to do it on my linux box, server-side. Ideally, I would be able to work into paperclip. As an additional solution, ffmpeg could work, but I have not seen any .caf options for it. Anybody know of one?

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  • Downsampling and applying a lowpass filter to digital audio

    - by twk
    I've got a 44Khz audio stream from a CD, represented as an array of 16 bit PCM samples. I'd like to cut it down to an 11KHz stream. How do I do that? From my days of engineering class many years ago, I know that the stream won't be able to describe anything over 5500Hz accurately anymore, so I assume I want to cut everything above that out too. Any ideas? Thanks. Update: There is some code on this page that converts from 48KHz to 8KHz using a simple algorithm and a coefficient array that looks like { 1, 4, 12, 12, 4, 1 }. I think that is what I need, but I need it for a factor of 4x rather than 6x. Any idea how those constants are calculated? Also, I end up converting the 16 byte samples to floats anyway, so I can do the downsampling with floats rather than shorts, if that helps the quality at all.

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  • Audio -- How much performance improvement can I expect from from reducing function calls by using bu

    - by morgancodes
    I'm working on an audio-intensive app for the iPhone. I'm currently calling a number of different functions for each sample I need to calculate. For example, I have an envelope class. When I calculate a sample, I do something like: sampleValue = oscilator->tic() * envelope->tic(); But I could also do something like: for(int i = 0; i < bufferLength; i++){ buffer[i] = oscilatorBuffer[i] * evelopeBuffer[i]; } I know the second will be more efficient, but don't know by how much. Are function calls expensive enough that I'd be crazy not to use buffers if I care event a tiny bit about performance?

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  • Java playback of 24 bit audio is incorrect

    - by Paul Hampson
    I am using the javax sound API to implement a simple console playback program based on http://www.jsresources.org/examples/AudioPlayer.html. Having tested it using a 24 bit ramp file (each sample is the last sample plus 1 over the full 24 bit range) it is evident that something odd is happening during playback. The recorded output is not the contents of the file (I have a digital loopback to verify this). It seems to be misinterpreting the samples in some way that causes the left channel to look like it is having some gain applied to it and the right channel looks like it is being attenuated. I have looked into whether the PAN and BALANCE controls need setting but these aren't available and I have checked the windows xp sound system settings. Any other form of playback of this ramp file is fine. If I do the same test with a 16bit file it performs correctly with no corruption of the stream. So does anyone have any idea why the Java Sound API is modifying my audio stream?

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  • Iphone progressive download audio player

    - by joynes
    Hi! Im trying to implement a progressive download audio player for the iphone, ie using http and fixed size mp3-files. I found the AudioStreamer project but it seems very complicated and works best with endless streams. I need to be able to find out the total length of audiofiles and I also need to be able to seek in the files. I found a hacked deviation from AudioStreamer but it doesnt seem to work very well for me. http://www.saygoodnight.com/?p=14 Im wondering if there is a more simple way to achieve my goals or if there are some better working samples out there? I found the bass library but not much documentation about it. /Br Johannes

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  • Streaming audio - where to start?

    - by Adam Davis
    I need to develop an embedded audio streaming server. Requirements: Voice quality or better Intended for low power wifi transmission Broad support in existing software and devices (ie, windows media player, quicktime, vlc, iPhone, Android, etc). Royalty/patent free, or cheap to license Preferences: Low overhead TCP/IP based streaming protocol Voice grade codec (easy to implement in software, no DSP, 32bit CPU if needed) Would be nice if it supported HTML5 browsers, but is there any codec (such as raw) that is supported by the latest browsers that is lower overhead than MP3? Therefore: What are the relevant streaming protocols I should be looking at? What are the relevant codecs I should be looking at? What transport streams should I be looking at? What am I missing, or where else should I be looking for this type of need?

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  • Android - Audio recorder FileNotFound

    - by david
    Hi, I'm trying to record audio this.recorder = new android.media.MediaRecorder(); this.recorder.setAudioSource(android.media.MediaRecorder.AudioSource.MIC); this.recorder.setOutputFormat(android.media.MediaRecorder.OutputFormat.DEFAULT); this.recorder.setAudioEncoder(android.media.MediaRecorder.AudioEncoder.DEFAULT); this.recorder.setOutputFile("pruebaAudioRecorder.mp4"); this.recorder.prepare(); this.recorder.start(); but when i call prepare method throws the FileNotFound exception. Should I create the file before prepare method? something like new File(...) If so, which should be the file path? thx a lot.

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  • Client-side framework for web-app with good audio support

    - by Poita_
    I'm trying to create a client-side web app that generates music procedurally using some user-input parameters, so I'm looking for a framework (e.g. Flash, Silverlight etc.) that has the capability to play audio at a specified pitch. Whether it is playing a WAV/MP3 file, using MIDI output, or just playing beeps doesn't really matter -- I just need something that will enable me to generate arbitrary music client-side. I've done a bit of searching and it appears that Flash might have the ability to change pitch with the help of a third-part plugin, but I couldn't find anything similar for Silverlight. I can go a try all them out manually if need be, but I thought I'd ask here first just in case anyone had tried something like this before. Thanks in advance

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  • iPhone game audio and background music

    - by Boon
    Have a few questions related to adding sounds to my game, specifically intro music (for splash), background music (loop) and button event sounds. Hope you can share your knowledge on this. 1) Should I use compressed sounds or uncompressed sounds? Or perhaps a combination of the two? Are there any limitations on the iPhone hardware that I should be aware of -- for example, the ability to play multiple compressed sounds? 2) What's the best audio format for my purpose? 3) For background music, I am thinking of using AVAudioPlayer. For button event sounds, I am thinking of using AudioServicesPlaySystemSound, what do you think? 4) Any other issues I should be aware of? Thank you!

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  • directx audio video error message in debugmode

    - by clamp
    I have a c#/winforms application that uses directx to play some video and audio. whenever i start my application in debugmode i get this annoying message. i can click "continue" and everything seems to work fine. but i still want to get rid of this message. it does not show up in releasemode. Managed Debugging Assistant 'LoaderLock' has detected a problem in 'C:\pathtoexe.exe'. Additional Information: DLL 'C:\WINDOWS\assembly\GAC\Microsoft.DirectX.AudioVideoPlayback\1.0.2902.0__31bf3856ad364e35\Microsoft.DirectX.AudioVideoPlayback.dll' is attempting managed execution inside OS Loader lock. Do not attempt to run managed code inside a DllMain or image initialization function since doing so can cause the application to hang.

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  • Toggling audio on click?

    - by angela
    please look at this fiddle http://jsfiddle.net/rabelais/yLdkj/1/ The above fiddle shows three bars that on hover play audios. How do I change this so the music plays and pauses on click instead. Also if one audio is playing and another is clicked how can the already playing song pause? $("#one").mouseenter(function () { $('#sound-1').get(0).play(); }); $("#one").mouseleave(function () { $('#sound-1').get(0).pause(); }); $("#two").mouseenter(function () { $('#sound-2').get(0).play(); }); $("#two").mouseleave(function () { $('#sound-2').get(0).pause(); }); $("#three").mouseenter(function () { $('#sound-3').get(0).play(); }); $("#three").mouseleave(function () { $('#sound-3').get(0).pause(); });

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  • audio stream sampling rate in linux

    - by farhan
    Im trying read and store samples from an audio microphone in linux using C/C++. Using PCM ioctls i setup the device to have a certain sampling rate say 10Khz using the SOUND_PCM_WRITE_RATE ioctl etc. The device gets setup correctly and im able to read back from the device after setup using the "read". int got = read(itsFd, b.getDataPtr(), b.sizeBytes()); The problem i have is that after setting the appropriate sampling rate i have a thread that continuously reads from /dev/dsp1 and stores these samples, but the number of samples that i get for 1 second of recording are way off the sampling rate and always orders of magnitude more than the set sampling rate. Any ideas where to begin on figuring out what might be the problem?

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  • High level audio crossfading library for python

    - by tcoopman
    I am looking for a high level audio library that supports crossfading for python (and that works in linux). In fact crossfading a song and saving it is about the only thing I need. I tried pyechonest but I find it really slow. Working with multiple songs at the same time is hard on memory too (I tried to crossfade about 10 songs in one, but I got out of memory errors and my script was using 1.4Gb of memory). So now I'm looking for something else that works with python. I have no idea if there exists anything like that, if not, are there good command line tools for this, I could write a wrapper for the tool.

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  • audio power on AudioQueue

    - by Tomoyuki
    Hi everyone. I'm now creating an Application using speech recognition.To check the Audio Power coming in through the microphone, I wrote a method as follows. -(void)checkPower(AudioqueRef)queue{ UInt32 expectedSize= sizeof(AudioQueueLevelMeterState); AudioQueueGetProperty(queue, kAudioQueueProperty_CurrentLevelMeter, audioLevels, expectedSize); NSLog(@"average:%f peak:%f",audioLevels.mAveragePower,audioLevels.mPeakPower); } I found that sometimes mAveragePower was larger than mPeakPower, and when mAveragePower was 1.0, in other words, averagePower is regarded as max, mPeakPower was lower than 1.0. I think that generally this result is inpossible. please Let me know if you have any information about sound power on CoreAudio. thanks.

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  • Naudio - putting audio stream into values [-1,1]

    - by denonth
    Hi all I need to put my audio stream into values of [-1,1]. Can someone tell me a good approach. I was reading byte array and float array from stream but I don't know what to do next. Here is my code: float[] bytes=new float[stream.Length]; float biggest= 0; for (int i = 0; i < stream.Length; i++) { bytes[i] = (byte)stream.ReadByte(); if (bytes[i] > biggest) { biggest=bytes[i]; } } and I don't know how to put values into stream. Because byte is only positive values. And I need to have from [-1,1] for (int i = 0; i < bytes.Count(); i++) { bytes[i] = (byte)(bytes[i] * (1 / biggest)); }

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  • audio error in vmware running mac os x

    - by PenguinSource
    simple synchronous loading of an audio file (.mp3) in a cocos2d app makes my vmware disconnect the sound. the error is display bottom right, saying 'error in creating sound stream; sound is disconnected' i read that it might be cause of my vmware's version (mine is 8) but I'm looking for a fix, not to downgrade to another version. before i get that error, the sound on the system works just fine (youtube, etc) the exact code im calling is.. [CDSoundEngine setMixerSampleRate: CD_SAMPLE_RATE_MID]; [[CDAudioManager sharedManager] setResignBehavior: kAMRBStopPlay autoHandle:Yes]; soundEngine = [SimpleAudioEngine sharedEngine]; [soundEngine preloadBackgroundMusic:@"somemp3.mp3"]; [soundEngine playBackgroundMusic:@"somemp3.mp3"]; maybe the bit rate is too high .. ? thanks

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  • How to calculate the audio file duration in core audio?

    - by mystify
    I have this info variable which is of this type: struct AudioStreamBasicDescription { Float64 mSampleRate; UInt32 mFormatID; UInt32 mFormatFlags; UInt32 mBytesPerPacket; UInt32 mFramesPerPacket; UInt32 mBytesPerFrame; UInt32 mChannelsPerFrame; UInt32 mBitsPerChannel; UInt32 mReserved; }; How could I calculate the total duration of the audio file, in seconds?

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  • Seeking not working in HTML5 audio tag

    - by lord_wilmore
    I have a lighttpd server running locally. If I load a static file on the server (through an html5 audio tag), it plays and seeks fine. However, seeking doesn't work when running a dev server (web.py/CherryPy) or if I return the bytes via a defined action url instead of as a static file. It won't load the duration either. According to the "HTTP byte range requests" section in this Opera Page it's something to do with support for byte range requests/partial content responses. The content is treated as streaming instead. What I don't understand is: If the browser has the whole file downloaded surely it can display the duration, and surely it can seek. What I need to do on the web server to enable byte range requests (for non-static urls). Any advice would be most gratefully received.

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