Search Results

Search found 5167 results on 207 pages for 'audio compression'.

Page 25/207 | < Previous Page | 21 22 23 24 25 26 27 28 29 30 31 32  | Next Page >

  • How can I maximally compress .gz files in Nautilus?

    - by Takkat
    When selecting Compress... from the right click context menu in Nautilus I am able to quickly compress files to .gz format. However by default Nautilus does not use maximum compression. Can I make Nautilus to use maximum compression like gzip -9? Using gconftool or gconf-editor to set the compression_level for File Roller to maximum seems right but infortunately has not the desired effect and will not lead to maximum compressed files. As this is the expected way of how to set compression levels a bug report has been filed upstream. Any ideas for a workaround are welcome.

    Read the article

  • Need help with some IIS7 web.config compression settings.

    - by Pure.Krome
    Hi folks, I'm trying to configure my IIS7 compression settings in my web.config file. I'm trying to enable HTTP 1.0 requests to be gzip. MSDN has all the info about it here. Is it possible to have this config info in my own website's web.config file? Or do i need to set it at an application level? Currently, I have that code in my web.config... <system.webServer> <urlCompression doDynamicCompression="true" dynamicCompressionBeforeCache="true" /> <httpCompression cacheControlHeader="max-age=86400" noCompressionForHttp10="False" noCompressionForProxies="False" sendCacheHeaders="true" /> ... other stuff snipped ... </system.webServer> It's not working :( HTTP 1.1 requests are getting compressed, just not 1.0. That MSDN page above says that it can be used in :- Machine.config ApplicationHost.config Root application Web.config Application Web.config Directory Web.config So, can we set these settings on a per-website-basis, programatically in a web.config file? (this is an Application Web.config file...) What have i done wrong? cheers :) EDIT: I was asked how i know HTTP1.0 is not getting compressed. I'm using the Failed Request Tracing Rules, which reports back:- DYNAMIC_COMPRESSION_START DYNAMIC_COMPRESSION_NOT_SUCESS Reason: 3 Reason: NO_COMPRESSION_10 DYNAMIC_COMPRESSION_END

    Read the article

  • How can I stop Ubuntu from playing audio from 2 interfaces at the same time?

    - by Solignis
    Hi there, I just loaded Ubuntu 10.10 Maverick on my home machine. The machine is running a Core2Duo E6750 on an MSI motherboard with an Nvidia GTX260-OC Graphics card. The problem I am having as stated in the title is for some reason Ubuntu is playing audio through my headphone coming out from the computer and it is also playing the audio at the exact same time through the HDMI connection coming out of the graphics card, it has a plug to allow this. What is going on, I have never seen this before. Most importantly of all can it be fixed so that I can sepertate the 2 interfaces, the one is a standard PC audio IO and the HDMI one is connected through the mobo's internal SPDIF. More information can be provided if required.

    Read the article

  • Default audio device gives an error on WINDOWS 7 (x64) when triing to run VLC from CMD (VideoLAN, VL

    - by Ole Jak
    I use WINDOWS 7 (x64) (Russian) I want to stream life audio from my default audio capture device (microphone) When I set up VLM settings using visual enviroment instruments - VLM settings it all works fine. But when I export created settings/configuration *.vlm file and try to inport it into VLM it gives me nothing I opened that .vlm there is some text... so now I try to run VLC with default settings like this: vlc -i dshow:// --dshow-adev= :sout=#transcode{acodec=mp3,ab=128,channels=2,samplerate=44100}:std{access=http,mux=raw,dst=127.0.0.1:8084} but it dies giving me errors...=( So what shall I do to do live MP3 streaming from my default audio input device using VLC in non UI mode?

    Read the article

  • Can I reprogram a microphone input to be used as an audio output? (on XP)

    - by qftme
    I have a five year old Sony Vaio laptop (vgn-fw31m) that has had impact damage to the audio-output mini-jack for about the last year or so. In a recent discussion with my brother, we wondered whether it would be possible to write a program that would enable windows to use the microphone mini-jack input as the audio-output? As I currently use this laptop for work I am not keen to risk pulling it apart in order to replace the components comprising the audio-out. I therefore 'hope' that a programming solution exists. I would really appreciate any advice on this and eagerly await your response. Kind regards, qftme :)

    Read the article

  • Order by text and then by number

    - by Chaim Chaikin
    I have data like: Audio 1 File 10 Audio 2 Audio 3 File 11 Audio 13 Audio 22 File 20 Test 22 Audio 10 File 1 File 2 I need it order first by the text (i.e. Audio, File, Test) and then by number (1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22 etc.) The problem is that sorting it returns something like this: Audio 1 Audio 10 Audio 13 Audio 2 Audio 22 Audio 3 File 1 File 10 File 11 File 2 File 20 Test 22 While the result I want is: Audio 1 Audio 2 Audio 3 Audio 10 Audio 13 Audio 22 File 1 File 2 File 10 File 11 File 20 Test 22 If they were just numbers (i.e. without the audio, file, test) then I could just sort numerically. However, how can I sort here first by text and then by number.

    Read the article

  • Does Compressed Sensing bring anything new to data Compression?

    - by anon
    Compressed sensing is great for situations where capturing data is expensive (either in energy or time), and thus less samples can now be taken. However, in situations like image compression, given that the data is already on the computer -- does compressed sensing offer anything? For example, would it offer better data compression? Would it result in better image search?... (Note: If you don't know what the field of Compressed Sensing is, please do not respond.)

    Read the article

  • KDE: How can I select audio output device for mplayer?

    - by grimripper
    I recently installed Kubuntu 13.10 64-bit, and I'm having a problem with selecting audio output device. In Phonon, when I select audio device preference order and press Apply, Amarok and Dragon will immediately switch to the preferred device. VLC and SMplayer are not affected. VLC has its own setting for selecting the output device, but SMplayer remains a problem. It always plays audio on internal audio, and I can't change output to HDMI. How can I select HDMI for SMplayer's audio output device? I don't know if it matters, but when I select HDMI audio in Phonon and click Test, the test sound plays on the internal audio output as well. In the hardware settings tab, the front left and front right test buttons play audio on HDMI. Also, volume up/down buttons affect HDMI volume when SMplayer is focused. This would make sense if I could get SMplayer to play audio over HDMI, but it would be better if the volume keys affected SMplayer's own volume, or the "mplayer2: audio stream" which appears in volume control while mplayer is playing. EDIT: I've recompiled mplayer with alsa support, and can now select the audio output device from SMplayer's settings. Didn't affect the issue with Phonon of course, but it's a suitable workaround.

    Read the article

  • AVAudioRecorder - Continue recording to file after user stops recording by leaving the application a

    - by Tegeril
    Can this be done? And if not, how far down towards Core Audio do I need to go (what method of recording should I be using instead)? I've noticed the behavior of AVAudioRecorder is to overwrite a file if it finds one at the path provided when you request that it record again, so I know that's not going to work. I'm also curious about file format restriction with this idea. Can you effectively resume an AAC or IMA4 encoding (the length of the files I want to record make WAV and probably even Apple Lossless prohibitive)? Thanks.

    Read the article

  • Sharp HealthCare Reduces Storage Requirements by 50% with Oracle Advanced Compression

    - by [email protected]
    Sharp HealthCare is an award-winning integrated regional health care delivery system based in San Diego, California, with 2,600 physicians and more than 14,000 employees. Sharp HealthCare's data warehouse forms a vital part of the information system's infrastructure and is used to separate business intelligence reporting from time-critical health care transactional systems. Faced with tremendous data growth, Sharp HealthCare decided to replace their existing Microsoft products with a solution based on Oracle Database 11g and to implement Oracle Advanced Compression. Join us to hear directly from the primary DBA for the Data Warehouse Application Team, Kim Nguyen, how the new environment significantly reduced Sharp HealthCare's storage requirements and improved query performance.

    Read the article

  • Oracle saves with Oracle Database 11g and Advanced Compression

    - by jenny.gelhausen
    Oracle Corporation runs a centralized eBusiness Suite system on Oracle Database 11g for all its employees around the globe. This clustered Global Single Instance (GSI) has scaled seamlessly with many acquisitions over the years, doubling the number of employees since 2001 and supporting around 100,000 employees today, 24 hours a day, 7 days a week around the world. In this podcast, you'll hear from Raji Mani, IT Director for Oracle's PDIT Group, on how Oracle Database 11g and Advanced Compression is helping to save big on storage costs. var gaJsHost = (("https:" == document.location.protocol) ? "https://ssl." : "http://www."); document.write(unescape("%3Cscript src='" + gaJsHost + "google-analytics.com/ga.js' type='text/javascript'%3E%3C/script%3E")); try { var pageTracker = _gat._getTracker("UA-13185312-1"); pageTracker._trackPageview(); } catch(err) {}

    Read the article

  • How do I get the compression on specific dynamic body

    - by Mike JM
    Sorry, I could not find any tag that would suit my question. Let me first show you the image and then write what I want to do: I'm using box2D. As you can see there are three dynamic bodies connected to each other (think of it as a table from front view).The LEG1 and LEG2 are connected to the static body. (it's the ground body). Another dynamic body is falling onto the table. I need to get the compression in the LEG1 and LEG2 separately. Joints have GetReactionForce() function which returns a b2Vec, which in turn has Length() and LengthSqd functions. This will give the total sum of the forces in any taken joint. But what I need is forces in individual bodies that are connected with joints. Once you connect several bodies with a single joint it again will show the sum of forces which is not useful.Here's the case iI'm talking about:

    Read the article

  • Hybrid Columnar Compression

    - by user12620172
    You heard me in the past talk about the HCC feature for Oracle databases. Hybrid Columnar Compression is a fantastic, built-in, free feature of Oracle 11Gr2. One used to need an Exadata to make use of it. However, last October, Oracle opened it up and now allows it to work on ANY Oracle DB server running 11Gr2, as long as the storage behind it is a ZFSSA for DNFS, or an Axiom for FC. If you're not sure why this is so cool or what HCC can do for your Oracle database, please check out this presentation. In it, Art will explain HCC, show you what it does, and give you a great idea why it's such a game-changer for those holding lots of historical DB data. Did I mention it's free? Click here: http://hcc.zanghosting.com/hcc-demo-swf.html

    Read the article

  • How to get audio driver for compaq c700 ?

    - by Leena
    Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

    Read the article

  • How to get audio driver for compaq c700 ?

    - by Leena
    Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

    Read the article

  • How to get audio driver for compaq c700 ?

    - by Leena
    Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

    Read the article

  • How to enable gzip HTTP compression on Windows Azure dynamic content

    - by Steven
    Hi all, I've been trying unsuccessfully to enable gzip HTTP compression on my Windows Azure hosted WCF Restful service which returns JSON only from GET and POST requests. I have tried so many things that I would have a hard time listing all of them, and I now realise I have been working with conflicting information (regarding old version of azure etc) so think it best to start with a clean slate! I am working with Visual Studio 2008, using the February 2010 tools for Visual Studio. So, according to the following link, HTTP compression has now been enabled .. http://msdn.microsoft.com/en-us/library/ff436045.aspx ... and I've used the advice at the following page (the URL compression advice only), but I get no compression. http://blog.smarx.com/posts/iis-compression-in-windows-azure <urlCompression doStaticCompression="true" doDynamicCompression="false" dynamicCompressionBeforeCache="true" /> It doesn't help that I don't know what the difference is between urlCompression and httpCompression. I've tried to find out but to no avail! Could the fact that the tools for Visual Studio were released before the version of Azure which supports compression be a problem? I read somewhere that with the latest tools, you can choose which version of Azure OS you want to use when you publish ... but I don't know if that's true, and if it is, I can't find where to choose. Could I be using a pre-http enabled version? I've also tried blowery http compression module, but no results. Does any one have any up-to-date advice on how to achieve this? i.e. advice that relates to the current version of the Azure OS. Cheers! Steven

    Read the article

  • Streaming Audio over UDP to Android

    - by Mr. Pig
    Is it possible to have Android (perhaps via MediaPlayer or a different existing class) accept media streams over UDP? I've successfully had MediaPlayer connect to an HTTP stream (as well as static files hosted on an HTTP server) but I'm wondering how one would go about accepting a stream from a UDP source. I've seen this and suppose a solution similar to that (where I download the stream via an independent UDP socket and then move the data to a MemoryBuffer that I then pass to MediaPlayer) is an option but I'm curious if a method already exists in the SDK, and if it does not, what other options do I have? Thanks

    Read the article

  • Audio recording and playback in Silverlight

    - by Ramesh
    I have a Silverlight 4 application that records user's voice through the mic. Now, as soon as the recording is completed, I need to play the recorded voice back to the user before posting it to the server. Is it at all possible to play it back to the user without getting into format conversions etc? Any ideas are welcome. Thanks!

    Read the article

  • Android - Getting audio to play through earpiece

    - by Donal Rafferty
    I currently have code that reads a recording in from the devices mic using the AudioRecord class and then playing it back out using the AudioTrack class. My problem is that when I play it out it plays vis the speaker phone. I want it to play out via the ear piece on the device. Here is my code: public class LoopProg extends Activity { boolean isRecording; //currently not used AudioManager am; int count = 0; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); am = (AudioManager) getSystemService(Context.AUDIO_SERVICE); am.setMicrophoneMute(true); while(count <= 1000000){ Record record = new Record(); record.run(); count ++; Log.d("COUNT", "Count is : " + count); } } public class Record extends Thread { static final int bufferSize = 200000; final short[] buffer = new short[bufferSize]; short[] readBuffer = new short[bufferSize]; public void run() { isRecording = true; android.os.Process.setThreadPriority (android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); int buffersize = AudioRecord.getMinBufferSize(11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT); AudioRecord arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize); AudioTrack atrack = new AudioTrack(AudioManager.STREAM_MUSIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize, AudioTrack.MODE_STREAM); am.setRouting(AudioManager.MODE_NORMAL,1, AudioManager.STREAM_MUSIC); int ok = am.getRouting(AudioManager.ROUTE_EARPIECE); Log.d("ROUTING", "getRouting = " + ok); setVolumeControlStream(AudioManager.STREAM_VOICE_CALL); //am.setSpeakerphoneOn(true); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); am.setSpeakerphoneOn(false); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); atrack.setPlaybackRate(11025); byte[] buffer = new byte[buffersize]; arec.startRecording(); atrack.play(); while(isRecording) { arec.read(buffer, 0, buffersize); atrack.write(buffer, 0, buffer.length); } arec.stop(); atrack.stop(); isRecording = false; } } } As you can see if the code I have tried using the AudioManager class and its methods including the deprecated setRouting method and nothing works, the setSpeatPoneOn method seems to have no effect at all, neither does the routing method. Has anyone got any ideas on how to get it to play via the earpiece instead of the spaker phone?

    Read the article

  • Core-audio - constructing an AudioBufferList struct (Q about c struct definition)

    - by mustISignUp
    The definition of AudioBufferList looks weird to me… i guess my C is not so good struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[kVariableLengthArray]; }; typedef struct AudioBufferList AudioBufferList; Why AudioBuffer mBuffers[kVariableLengthArray]; and not AudioBuffer *mBuffers; ? kVariableLengthArray appears to be == 1. Eh? I think i have it working but would appreciate it if anyone could set me straight.

    Read the article

  • Finding out estimated duration of a stream using Core Audio

    - by Reflog
    I am streaming a MP3 over network using custom feeding code, not AVAudioPlayer (which only works with URLs) using APIs like AudioFileStreamOpen and etc. Is there any way to estimate a length of the stream? I know that I can get a 'elapsed' property using: if(AudioQueueGetCurrentTime(queue.audioQueue, NULL, &t, &b) < 0) return 0; return t.mSampleTime / dataFormat.mSampleRate; But what about total duration to create a progress bar? Is that possible?

    Read the article

  • Core-audio - constructing an AudioBufferList

    - by mustISignUp
    The definition of AudioBufferList looks weird to me… i guess my C is not so good struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[kVariableLengthArray]; }; typedef struct AudioBufferList AudioBufferList; Why AudioBuffer mBuffers[kVariableLengthArray]; and not AudioBuffer *mBuffers; ? kVariableLengthArray appears to be == 1. Eh? I think i have it working but would appreciate it if anyone could set me straight.

    Read the article

< Previous Page | 21 22 23 24 25 26 27 28 29 30 31 32  | Next Page >