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  • How to link processing power of old computers together?

    - by redIago
    Hey all, I'm sitting on 8 old computers of varied sorts that are more or less useless at this point for any other purpose really. Is there a way I could link their hardware or processing power or whatever together over wifi and use one as like a central computer? Like it would be cool to distribute the processing of some video game or encryption generating program over the collective computers. Any way to do all this? Thanks.

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  • How can I stream audio signals from various devices/computers to my home server?

    - by Breakthrough
    I currently have a headless home server set up (running Ubuntu 12.04 server edition) running a simple Apache HTTP server. The server is near an audio receiver, which controls a set of indoor and outdoor speakers in my home. Recently, my father purchased a Bluetooth adapter, which our various laptops and cellphones can connect to, outputting the music to the speakers. I was hoping to find a solution that worked over Wi-Fi, namely because it won't cost anything (I already have a server with an audio card), and it doesn't depend on Bluetooth. Is there any cross-platform (preferably free and open-source) solution that I can use which will allow me to stream audio to my home server, over my home network, from a wide variety of devices (laptops running Windows/Linux or cellphones running Android/BB/iOS)? I need something that works at least with Windows and Android. Also, just to clairfy, I want something that simply allows devices to connect to my server and output an audio signal without any action on the server end (since it's a server hidden away near my receiver). Any subsequent connection attempt should be dropped, so only one device can be in control of the stereo at once.

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  • Help with dynamic range compression function (audio)

    - by MusiGenesis
    I am writing a C# function for doing dynamic range compression (an audio effect that basically squashes transient peaks and amplifies everything else to produce an overall louder sound). I have written a function that does this (I think): public static void Compress(ref short[] input, double thresholdDb, double ratio) { double maxDb = thresholdDb - (thresholdDb / ratio); double maxGain = Math.Pow(10, -maxDb / 20.0); for (int i = 0; i < input.Length; i += 2) { // convert sample values to ABS gain and store original signs int signL = input[i] < 0 ? -1 : 1; double valL = (double)input[i] / 32768.0; if (valL < 0.0) { valL = -valL; } int signR = input[i + 1] < 0 ? -1 : 1; double valR = (double)input[i + 1] / 32768.0; if (valR < 0.0) { valR = -valR; } // calculate mono value and compress double val = (valL + valR) * 0.5; double posDb = -Math.Log10(val) * 20.0; if (posDb < thresholdDb) { posDb = thresholdDb - ((thresholdDb - posDb) / ratio); } // measure L and R sample values relative to mono value double multL = valL / val; double multR = valR / val; // convert compressed db value to gain and amplify val = Math.Pow(10, -posDb / 20.0); val = val / maxGain; // re-calculate L and R gain values relative to compressed/amplified // mono value valL = val * multL; valR = val * multR; double lim = 1.5; // determined by experimentation, with the goal // being that the lines below should never (or rarely) be hit if (valL > lim) { valL = lim; } if (valR > lim) { valR = lim; } double maxval = 32000.0 / lim; // convert gain values back to sample values input[i] = (short)(valL * maxval); input[i] *= (short)signL; input[i + 1] = (short)(valR * maxval); input[i + 1] *= (short)signR; } } and I am calling it with threshold values between 10.0 db and 30.0 db and ratios between 1.5 and 4.0. This function definitely produces a louder overall sound, but with an unacceptable level of distortion, even at low threshold values and low ratios. Can anybody see anything wrong with this function? Am I handling the stereo aspect correctly (the function assumes stereo input)? As I (dimly) understand things, I don't want to compress the two channels separately, so my code is attempting to compress a "virtual" mono sample value and then apply the same degree of compression to the L and R sample value separately. Not sure I'm doing it right, however. I think part of the problem may the "hard knee" of my function, which kicks in the compression abruptly when the threshold is crossed. I think I may need to use a "soft knee" like this: Can anybody suggest a modification to my function to produce the soft knee curve?

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  • Why does 3.5 mm audio out work through headphones but not through external speakers?

    - by Rickster
    I have a computer that has a 3.5 mm audio jack on the front of it. The computer itself has no speakers, so this is the only way to hear sound. If I plug headphones into it, the audio properly plays through the headphones, and if I plug in external speakers it used to play through them as well. Just today I turned on my computer and the audio no longer plays through the speakers, but if I plug in the headphones instead it works. The speakers aren't broken, as both the speakers and headphones work in my iPod and play music. I thought that 3.5 mm jacks could not send data back to the computer, and the computer had no way of differentiating between different devices plugged into the jack. If this is true, how is it that the computer plays audio through the headphones but not through speakers plugged into the same 3.5 mm jack, and both devices are functional? Or is my knowledge on 3.5 mm jacks incorrect? I don't believe drivers are important, as the same driver runs the 3.5 mm jack for all devices, but if necessary I can provide additional information. Any ideas would be appreciated. Thanks!

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  • Creating a music catalog in C# and extracting first 30 seconds as soon as the first words are sung

    - by Rad
    I already read a question: Separation of singing voice from music. I don’t need this complex audio processing. I only need some detection mechanism that would detect that there is some voice/vocal playing while the music is playing (or not playing) I need to extract first 30 seconds when a vocalist starts singing along with full band music. See question 2 below. I want to create a music catalog using ASP.NET MVC 2 and Silverlight clients and C#.NET 4.0 programming language that would be front store. On the backend I would also like to create a desktop WPF/Windows application to create the music catalog from already existing music files, most of which have metadata in them ID3v1, ID3v2.3, ID3v2.4, iTunes MP4, WMA, Vorbis Comments and APE Tags etc. I would possibly like to create a web service that would allow catalog contributors to upload a zipped album and trigger metadata extraction of music data and extraction of music segments as described below. I would be happy if I achieve no. 1 below. Let's say I have 1000ths of songs in mp3 (or other formats) grouped in subfolders using some classification (Genre, Artists, Albums, Composers or other groupings). I want to create tables in DB that would organize songs so they can be searched based on different criteria (year, length, above classification or by song title, description etc) like what iTune store allows to their customers. I want to extract metadata from various formats (I will try to get songs in mp3 format, but there may be other popular formats) and allow music Catalog manager person to add missing data from either desktop or web applications. He or other contributors can upload zipped music via an HTML or Silverlight upload or WPF. Can anybody suggest open source libraries, articles, code snippets that can do that in an automatic way using .NET and possibly SQL Server DB? My main questions are these. This is an audio processing challenge. I want to extract 2 segments of music (questions 1 and 2): 1. How to extract a music segment: 1-2 seconds before a vocal starts singing and up to 30 seconds from that point in time and 2. Much more challenging is to find repeating segments (One would usually find or recognize the names of the songs and songs are usually known by these refrains. How would I go about creating a list of songs that go great together like what Genius from iTune does? Is there any characteristics of music that can be used to match songs? The goal is for people quickly scan and recognize songs i.e. associate melody, words with a title/album so they can make intelligent decisions like buying a song, create similar mood songs. Thanks, Rad

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  • ANSI SQL Hierarchical Data Processing Basics

    The SQL-92 standard unknowingly and without planning introduced the capability to perform full hierarchical data processing with its introduction of the LEFT Outer Join operation. This natural hierarchical processing capability will be explained in this article.

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  • Google I/O 2010 - Batch data processing with App Engine

    Google I/O 2010 - Batch data processing with App Engine Google I/O 2010 - Batch data processing with App Engine App Engine 201 Mike Aizatsky In this session, attendees will learn how to write map() functions, how to do simple reduce() operations, how to run these over large datasets, and how App Engine is used to accomplish such parallelism. For all I/O 2010 sessions, please go to code.google.com From: GoogleDevelopers Views: 6 0 ratings Time: 38:45 More in Science & Technology

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  • 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.

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  • 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?

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  • 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 :)

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  • 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.

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  • Hardware advice for bitmap / openGL image processing server?

    - by pdizz
    I am trying to work out a build for a processing server to handle bitmap processing as well as openGL rendering for chroma-keying images and Photoshop automation. My searches here and on Google have turned up surprisingly few results, and seeing that there aren't tags for bitmap or image processing I take it this is a specialized application. The bitmap processing is very cpu-intensive while the chroma-keying and Photoshop stuff is gpu-intensive. I doubt this is a case of over-optimization as our company batches thousands of images a day (currently on individual workstations) and any saving in processing time and workstation down-time would be beneficial. Does anyone have any experience with this type of processing server? Any special considerations that would go into a build like this or am I over-thinking it?

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  • 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.

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  • 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.

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  • "Winamp style" spectrum analyzer

    - by cvb
    I have a program that plots the spectrum analysis (Amp/Freq) of a signal, which is preety much the DFT converted to polar. However, this is not exactly the sort of graph that, say, winamp (right at the top-left corner), or effectively any other audio software plots. I am not really sure what is this sort of graph called (if it has a distinct name at all), so I am not sure what to look for. I am preety positive about the frequency axis being base two exponential, the amplitude axis puzzles me though. Any pointers?

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  • 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.

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  • 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.

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  • 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.

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  • ALERT: Error Processing US Wage Attachment Elements In Payroll Run After RUP Patches

    - by LuciaC
    Customers who have run the Upgrade Wage Attachments process after applying the 2012 RUP are reporting errors similar to those listed below when either running a quickpay or processing a payroll for employee(s) with involuntary deductions. Error: HR_51118_HRPROC_ERR_ON_ASG ASGNO 1115 APP-PAY-51118: Error was encountered when processing assignment 1115 HR_51119_HRPROC_ERR_OCC_ON_ET ETNAME: Garnishment 3 APP-PAY-51119: Error was encountered when processing Element Type Garnishment 3 HR_6881_HRPROC_ORA_ERR SQLERRMC ORA-01403: No data found SQL_NO 520 TABLE_NAME pay_input_values_f APP-PAY-06881:Error ORA-01403: no data found has occured in table pay_input_values_f at location 520 This issue was logged in Bug 14679161 - QUICK PAY ERROR AFTER RUP (2012) AND WAGE ATTACHMENT UPGRADE APP-PAY-06881. The following one off patches have been released to My Oracle Support to resolve this issue*: 11i -  Patch 14679161 12.0 - Patch 14849394:R12.PAY.A 12.1 - Patch 14849394:R12.PAY.B * IMPORTANT:  Depending on when/if customers have run the Wage Attachment upgrade process will determine the appropriate action to take. Any customer who is encountering the above error and/or has run the Wage Attachment upgrade process AFTER applying the 2012 RUP (applicable to their release level) should log a Service Request with Oracle Support to receive assistance on the necessary steps to take to resolve the problem BEFORE applying the above patch. Any customer who has not yet run the Wage Attachment Upgrade process (either before or after applying the 2012 RUP), should follow the action plan documented in the patch readme. For those customers who have already run the Wage Attachment Upgrade process BEFORE applying the 2012 RUP, should apply the patch (applicable to your release) listed above. Be sure to run any post install processes, such as the data install utility and HR global driver.  See the patch readme for full details. Please consult Note 404478.1: Americas (US, CA, MX) HCM High Priority Alert for the latest Alert status.

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  • Impact of Server Failure on Coherence Request Processing

    - by jpurdy
    Requests against a given cache server may be temporarily blocked for several seconds following the failure of other cluster members. This may cause issues for applications that can not tolerate multi-second response times even during failover processing (ignoring for the moment that in practice there are a variety of issues that make such absolute guarantees challenging even when there are no server failures). In general, Coherence is designed around the principle that failures in one member should not affect the rest of the cluster if at all possible. However, it's obvious that if that failed member was managing a piece of state that another member depends on, the second member will need to wait until a new member assumes responsibility for managing that state. This transfer of responsibility is (as of Coherence 3.7) performed by the primary service thread for each cache service. The finest possible granularity for transferring responsibility is a single partition. So the question becomes how to minimize the time spent processing each partition. Here are some optimizations that may reduce this period: Reduce the size of each partition (by increasing the partition count) Increase the number of JVMs across the cluster (increasing the total number of primary service threads) Increase the number of CPUs across the cluster (making sure that each JVM has a CPU core when needed) Re-evaluate the set of configured indexes (as these will need to be rebuilt when a partition moves) Make sure that the backing map is as fast as possible (in most cases this means running on-heap) Make sure that the cluster is running on hardware with fast CPU cores (since the partition processing is single-threaded) As always, proper testing is required to make sure that configuration changes have the desired effect (and also to quantify that effect).

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  • 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

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  • 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!

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  • 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?

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