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  • Filter design for audio signal.

    - by beanyblue
    What I am trying to do is simple. I have a few .wav files. I want to remove noise and filter out specific frequencies. I don't have matlab and I intend to write my own code for all the filters. Right now, I have a way to read the .wav file and dump out the structure into a text file. My questions are the following: Can I directly apply the digital filters on this sampled data?{ ie, can I directly do a convolution between my input samples and h(n) for the filter function that i choose?). How do I choose the number of coefficients for the Window function? I have octave, so if someone can point me to anything that gives me some idea on how to process the .wav file using octave, that would be great too. I want to be able to filter out the frequency and then listen to the sound again. Is this possible with octave? I'm just a beginner with these kinds of things, so please bear with me if my questions are too naive. Any help will be great.

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  • How can I get it the Free Music Archive audio player or is there a better alternative?

    - by Dennis Hodapp
    I'm looking at free streaming audio players for web browsers that I can use in a project. I really like the audio player used on http://freemusicarchive.org/. Are they using an open source audio player and can I get a hold of it? Or is it closed source? Also if there are any open-source audio players that anybody knows about I'd love to know about them (preferable to have one with no flash). Last thing...is HTML5 going to be able to replace audio streaming players?

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  • SQL University: Parallelism Week - Part 3, Settings and Options

    - by Adam Machanic
    Congratulations! You've made it back for the the third and final installment of Parallelism Week here at SQL University . So far we've covered the fundamentals of multitasking vs. parallel processing and delved into how parallel query plans actually work . Today we'll take a look at the settings and options that influence intra-query parallelism and discuss how best to set things up in various situations. Instance-Level Configuration Your database server probably has more than one logical processor....(read more)

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  • Windows not remembering default audio device?

    - by Lynda
    I prefer the audio output on my computer to use the standard audio jack output due to volume issues. But I am using a monitor with HDMI. I have chosen to set the default audio device to be "Speakers" But every time I reboot the default audio device is the HDMI Output again. I am running Windows 7 64bit. Why does it not remember the default device? (I do shutdown and boot up properly without errors.)

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  • No audio device detected

    - by Paul
    My computer has no audio. It says NO AUDIO DEVICE. I already installed Realtek AC97 and Realtek High Definition Audio Driver and I also pasted stream.dll to the Windows and system32 folders and I restarted my computer but it still says NO AUDIO DEVICE. Please help me. Thanks

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  • ASUS N45SF - play subwoofer with audio connected

    - by Jaroslav Bucko
    I have notebook ASUS N45SF. It comes with dedicated subwoofer, which is connected to separate audio jack. When I connect any audio device to audio jack, internal speakers remain silent, but subwoofer too. I want to let subwoofer play even with audio device connected to jack. Are there any drivers or settings in OS, which would eneble this behaviour? I have Win7/Ubuntu dualboot so OS doesnt matter. Thanks

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  • what is the best mid/high-end class audio/music creation audio sound card?

    - by Chris
    Hello, I have a computershop myself, and I repair computers. But one of the things I really don't know (yet) is the performace od audio cards for music creation with midi. I have searched and searched and came up with some good reviews, but after browsing for a couple of hours I could't see the trees trough the forrest :-D (it's a dutch expression) At one moment I thought the M-Audio - Delta 1010LT would be a good PCIe card, later on I read that this card was released years ago. (but that could be false information) Also any personal expierence would be great, but not necessairy. I have searched a few cards, and I hope someone can help me make a choice for a friend of mine. He's buget is between $100 and $350 I know there are audio cards from $ 500 - $1850,- this is just too expensive. The following specs are crucial: ASIO Midi Mic in minimal 5.1, 7.1 recommended it's not for airplay, but just to compose music at home. using Ableton and midi keyboard. 1. M-Audio - Delta 1010LT: 8 x 8 analog I/O 2 mic preamps or line inputs S/PDIF digital I/O (coaxial) with 2-channel PCM SCMS copy protection control digital I/O supports surround-encoded AC-3 and DTS pass-through 1 x 1 MIDI I/O directly drive up to 7.1 surround (bass management software included) software controlled 36-bit internal DSP digital mixing/routing +4dbu/-10dBV operation individually switched in software word clock I/O for sample accurate device synchronization 2. RME HDSP 9632: * Stereo Analog Ein- und Ausgang, symmetrisch*, 24-Bit/192kHz, > 110 dB SNR * Optionale Erweiterungsboards mit je 4 symmetrischen Ein- und Ausgängen * Alle analogen I/Os voll 192 kHz-fähig, also keine Reduzierung der Kanalzahl * 1 x ADAT Digital In/Out, 96 kHz-fähig (S/MUX) * 1 x SPDIF Digital In/Out, 192 kHz-fähig * 1 x Breakout Kabel für koaxialen SPDIF-Betrieb* * Also bis zu 16 Ein-und Ausgänge gleichzeitig nutzbar! * 1 x Stereo Kopfhörerausgang, parallel zum analogen Ausgang, aber eigene Pegelanpassung * 1 x MIDI I/O für 16 Kanäle Hi-Speed MIDI über Breakout Kabel * DIGICheck, RMEs einzigartiges Meter- und Analysetool mit Spectral Analyser, Professionelle Level Meter 2/8/16-Kanalig, Vector Audio Scope und diversen weiteren Analysefunktionen * HDSP Meter Bridge: Frei skalierbare Levelmeter mit Peak- und RMS Berechnung in Hardware * TotalMix: 512-Kanal Mischer mit 40 Bit interner Auflösung 3. EMU 1212M (1212 M) PCIe: * Top kwaliteit convertors 24-bit/192kHz convertors. * Hardware gestuurde effecten. * DSP zero-latency hardware mixen en monitoring. * Analoge en digitale I/O plus MIDI. * EMU Production Tools Software Bundle - Cakewalk SONAR , Steinberg Cubase LE, Ableton Live E-MU Edition **EMU 1212M PCI-e inputs/outputs:** * 2 balanced jack inputs. * 2 balanced jack outputs. * 24-bit/192kHz ADAT I/O. * 24-bit/192kHz Coaxiale S/PDif I/O switchable to AES/EBU. * MIDI I/O. 4. M-Audio Audiophile 192: - Up to 24-bit/192kHz audio - 2 balanced analog inputs (1/4” TRS) - 2 balanced analog outputs (1/4” TRS) - S/PDIF digital I/O (coaxial RCA connectors) with 2-channel PCM - SCMS copy protection control - Digital I/O supports surround-encoded AC-3 and DTS pass-through - Direct hardware input monitoring via separate balanced 1/4” TRS monitor outputs - Software routing of inputs and outputs - Digital I/O can be routed to/from external effects - 16-channel MIDI I/O - ASIO, WDM, GSIF 2 and Core Audio driver support for compatibility with most applications - 64-bit driver support for Windows - PCI 2.2 compatibility - Apple G5 compatible - Incompatible exceptions - Includes Ableton Live Lite music production software, so you can make music right away - Works with other Delta cards Technical Specifcations: - Compatibility - ASIO - WDM - GSIF 2 - Core Audio

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  • Convert audio file to FLAC with ffmpeg?

    - by elpsk
    can I convert one of this format to compatible 16000.0 Sample Rate FLAC file? kAudioFormatLinearPCM = 'lpcm', kAudioFormatAppleIMA4 = 'ima4', kAudioFormatMPEG4AAC = 'aac ', kAudioFormatMACE3 = 'MAC3', kAudioFormatMACE6 = 'MAC6', kAudioFormatULaw = 'ulaw', kAudioFormatALaw = 'alaw', kAudioFormatMPEGLayer1 = '.mp1', kAudioFormatMPEGLayer2 = '.mp2', kAudioFormatMPEGLayer3 = '.mp3', kAudioFormatAppleLossless = 'alac' I tried using ffmpeg ffmpeg -i audio.xxx -acodec flac audio.flac but result is FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard Mac OSX universal build for ffmpegX configuration: --enable-memalign-hack --enable-mp3lame --enable-gpl --disable-vhook --disable-ffplay --disable-ffserver --enable-a52 --enable-xvid --enable-faac --enable-faad --enable-amr_nb --enable-amr_wb --enable-pthreads --enable-x264 libavutil version: 49.0.0 libavcodec version: 51.9.0 libavformat version: 50.4.0 built on Apr 15 2006 04:58:19, gcc: 4.0.1 (Apple Computer, Inc. build 5250) Input #0, wsaud, from 'audio.alac': Duration: 00:00:03.8, start: 0.000000, bitrate: 199 kb/s Stream #0.0: Audio: adpcm_ima_ws, 24931 Hz, stereo, 199 kb/s Unable for find a suitable output format for 'audio.flac' I also installed flac codec for mac, but nothing... I tried also use convtoflac.sh (from http://legroom.net/software/convtoflac) but result is similar. Any idea to convert in flac?

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  • Change the audio output device in Firefox

    - by Zanami Zani
    I'm trying to play music through Ventrilo and currently I use Virtual Audio Cable. The way it works is that in foobar2000 (a music playing program) I set the output device in preferences to Virtual Audio Cable. Then in Ventrilo I log in to another name and set the input device to Virtual Audio Cable. This routes the music through the Virtual Audio Cable and allows me to play the music through Ventrilo. However, I would also like to change the output device for Firefox (or any other browser) or "Plugin Container for Firefix" to Virtual Audio Cable so that I could play music from Pandora or YouTube on to Ventrilo. Unfortunately I could not find an option for this anywhere.

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  • Record the audio stream from HDMI monitor

    - by Nick
    I am trying to record sound playing on my comupter with Audacity but am running into some troubles. I have the stereo mix set to be the default audio recorder but it doesn't pick up the audio that is being played through my HDMI monitors speakers: Playback Recording When I plug in headphones the stereo mix will pick up the audio stream and I can record but not when playing through the HDMI. I have installed the latest audio drivers and have tried all the different record options to no avail. How can I capture the Audio stream going through the HDMI?

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  • Dynamic audio score/music

    - by Joel Martinez
    I'm interested in developing a game who's background music changes with the mood and scenario of the game's action. Of course many existing games do this (halo for example), but I was interested in any resources/papers/articles talking about the techniques to develop a system like this. I have some ideas, and I understand that this will be equally challenging to implement at the code level as it will be to come up or acquire music that fits this model. Any links or, answers with ideas in them would he appreciated. Edit: this is the kind of info I'm looking for :) http://halo.bungie.org/misc/gdc.2002.music/

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  • Play audio over network with Windows 7?

    - by Josh
    I have a unique situation where I'd like to stream audio (ALL audio, not just mp3s, etc) from my laptop to another computer over the network. I live in a studio apartment and my laptop is my main computer but I'd like it's audio to play on my htpc with a nice stereo system. Since it's a studio, both computers are in the same room so I don't want 2 sets of speakers. I want my computer to directly play back through the stereo. I used to do this with pulseaudio but my job now requires that I run Windows full time. I'm aware of Shoutcast and other similar streaming solutions but I don't want any transcoding done. It's a waste of CPU and not to mention my laptop fans, and I don't mind the network bandwidth that uncompressed audio requires. Is there a way to run Shoutcast without encoding? Also, I know that Windows Remote Desktop can play audio over the network pretty easily. Is this part of .Net that I could just code a simple app that streams the audio without RD'ing in? I also don't want to run it over a physical wire. :)

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  • Audio card with built-in ground isolator?

    - by Dave Jarvis
    What audio cards would you recommend that eliminate hum, and hard-drive & mouse movement signal interference? Hardware components: Motherboard. Asus P5Q SE Audio. Realtek ALC 1200, 8-Channel High-Definition Audio CODEC (on board) Harddrive. WD Caviar 320 GB Mouse. Logitech Marbleman USB Mixer. Mackie d.4 Pro Amplifier. Sonance Sonamp 260 All components are plugged into the same Monster Power HDP 910 powerbar (does not help eliminate noise). I have no other components plugged in. The computer uses a Monster iCable 1000 to go from mini (on board audio) to RCA (mixer). I have moved the cable as far from other cables as possible. A ground loop isolator between the mixer and on board audio eliminates all noise. I would rather not use a ground loop isolator; an internal audio card that is Linux-compatible (Kubuntu) would be ideal. Suggestions?

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  • Downmix ALL SYSTEM audio to mono - Windows 7

    - by Mike K.
    I'm deaf in one ear and want to use my headphones when playing a game and talking with my friends on Skype/TS/Mumble/etc while also sometimes listening to music. I need ALL my system audio to be downmixed to mono so that my ONE hearing ear gets ALL audio channels instead of split stereo audio. No, none of the other similar questions on superuser have a solution. My headphone properties does not have a 'Mono' option, I don't have a 'Headphone Virtualization' option, and my Realtek HD audio driver software doesn't have these options either (driver was updated 11/14/2012). Don't even talk about setting the balance of one side of the headphones to 0. You're not paying attention if you suggest that. JACK and Virtual Audio Cable didn't work. It's possible I configured them wrong, but I followed the steps I found in related questions and still got split stereo out. TL;DR I need a viable, working, software solution (I say software because I have a USB headset) for forcing ALL system audio to mono so that I can hear literally everything through the one earpiece. Thanks!

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  • Windows Media Based video audio converter?

    - by acidzombie24
    This may seem like an odd question. Right now ONLY windows media player, VLC and media player classic opens and plays my audio video correctly. Virtualdub plays it back with the wrong framerate and losses the audio, Avidemux 2.5 seems to be able to dump the audio/video but the video (like all other apps) is either a bad framerate or is wrong (glitches and bad framerate or bad dump). Nothing recognizes the audio file and when playing the video Avidemux (and most other things) die. FFMPEG cant seem to split the video or audio (using copy -an and etc) and this is getting me very angry. VLC dumps the video incorrectly when i try dumping it with that too. What can i use to convert the video? its streaming so it starts at 26mins in and ends at 28 (this is where apps have the problem. They dont know this and fudge everything or crash). I manage to dump the audio with Avidemux but virtualdub and ffmpeg says unreconized codec. Even if i cant convert it (it seems compressed enough) i want to at least attach it back into an AVI.

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  • How to speed up drawing of scaled image? Audio playback chokes during window resize.

    - by Paperflyer
    I am writing an audio player for OSX. One view is a custom view that displays a waveform. The waveform is stored as a instance variable of type NSImage with an NSBitmapImageRep. The view also displays a progress indicator (a thick red line). Therefore, it is updated/redrawn every 30 milliseconds. Since it takes a rather long time to recalculate the image, I do that in a background thread after every window resize and update the displayed image once the new image is ready. In the meantime, the original image is scaled to fit the view like this: // The drawing rectangle is slightly smaller than the view, defined by // the two margins. NSRect drawingRect; drawingRect.origin = NSMakePoint(sideEdgeMarginWidth, topEdgeMarginHeight); drawingRect.size = NSMakeSize([self bounds].size.width-2*sideEdgeMarginWidth, [self bounds].size.height-2*topEdgeMarginHeight); [waveform drawInRect:drawingRect fromRect:NSZeroRect operation:NSCompositeSourceOver fraction:1]; The view makes up the biggest part of the window. During live resize, audio starts choking. Selecting the "big" graphic card on my Macbook Pro makes it less bad, but not by much. CPU utilization is somewhere around 20-40% during live resizes. Instruments suggests that rescaling/redrawing of the image is the problem. Once I stop resizing the window, CPU utilization goes down and audio stops glitching. I already tried to disable image interpolation to speed up the drawing like this: [[NSGraphicsContext currentContext] setImageInterpolation:NSImageInterpolationNone]; That helps, but audio still chokes during live resizes. Do you have an idea how to improve this? The main thing is to prevent the audio from choking.

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  • SQL Peer-to-Peer Dynamic Structured Data Processing Collaboration

    Unstructured and XML semi-structured data is now used more than structured data. But fixed structured data still keeps businesses running day in and day out, which requires consistent predictable highly principled processing for correct results. For this reason, it would be very useful to have a general purpose SQL peer-to-peer collaboration capability that can utilize highly principled hierarchical data processing and its flexible and advanced structured processing to support dynamically structured data and its dynamic structured processing. This flexible dynamic structured processing can change the structure of the data as necessary for the required processing while preserving the relational and hierarchical data principles. This processing will perform freely across remote unrelated peer locations anytime and transparently process unpredictable and unknown structured data and data type changes automatically for immediate processing using automatic metadata maintenance.

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  • Play audio file data - Spring MVC

    - by Vijay Veeraraghavan
    In my web-application, I have various audio clips uploaded by the users in the database stored in the BLOB column. The audio files are low bit rate WAV files. The clips are secured, one can see only those clips he has uploaded. Instead of user downloading the clip and playing it in his player, I need it be steamed online in the web page itself. In the jsp I use the <audio> tag with the source mapping to the controller mappping url. <td> <audio controls><source src="recfile/${au.id}" type="audio/mpeg" /></audio> </td> Where, the recfile is the request mapping and the au.id is the audio id. In the controller I process the request like below @RequestMapping(value = "/recfile/{id}", method = RequestMethod.GET, produces = { MediaType.APPLICATION_OCTET_STREAM_VALUE }) public HttpEntity<byte[]> downloadRecipientFile(@PathVariable("id") int id, ModelMap model, HttpServletResponse response) throws IOException, ServletException { LOGGER.debug("[GroupListController downloadRecipientFile]"); VoiceAudioLibrary dGroup = audioClipService.findAudioClip(id); if (dGroup == null || dGroup.getAudioData() == null || dGroup.getAudioData().length <= 0) { throw new ServletException("No clip found/clip has not data, id=" + id); } HttpHeaders header = new HttpHeaders(); I tried this too //header.setContentType(new MediaType("audio", "mp3")); header.setContentType(new MediaType("audio", "vnd.wave"); header.setContentLength(dGroup.getAudioData().length); return new HttpEntity<byte[]>(dGroup.getAudioData(), header); } When the jsp loads, the controller get the request, it serves back the audio data fetched from the database, the jsp too shows the player with the controls. But when I play it nothing happens. Why is it? Am I missing anything in the configuration? Am I doing it right?

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  • iPhone, JQTouch and HTML5 audio tags

    - by Moo
    I am having an issue with JQTouch (latest beta) and html5 audio tags on 'sub pages' - the audio tag works before any page transitions are done, and cease to work afterward. For example: http://richardprice.dyndns.ws/test.html and http://richardprice.dyndns.ws/test2.html are identical other than I swap the "current" class between the two divs - all the audio tags play the same mp3. On test.html the audio tag on the initial page works, but when you switch to Page 2 the audio tag on that page does not (and sometimes results in a browser crash). Switch back to Page 1 and the audio tag on that page has ceased to work. test2.html is the same test but with the initial pages reversed, and the same thing happens - Page 2 (now the initial page) plays the audio, Page 1 does not, and switching back to Page 2 results in the audio no longer working. Thoughts?

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  • <audio> element autobuffers no matter what

    - by pthulin
    I'm trying to make a web based media player using the HTML5 audio element implemented in Firefox 3.5 and Chrome. Reading Mozillas documentation, omitting the autobuffer attribute should result in the audio src not being requested: if specified, the audio will automatically begin being downloaded, even if not set to automatically play. This continues until the media cache is full, or the entire audio file has been downloaded, whichever comes first However, on the server side I notice the files are being requested anyway. My sample page is very simple: <html> <body> <audio src="1.ogg"></audio> <audio src="2.ogg"></audio> </body> </html>

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  • Modify audio pitch of recorded clip (m4v)

    - by devcube
    I'm writing an app in which I'm trying to change the pitch of the audio when I'm recording a movie (.m4v). Or by modifying the audio pitch of the movie afterwards. I want the end result to be a movie (.m4v) that has the original length (i.e. same visual as original) but with modified sound pitch, e.g. a "chipmunk voice". A realtime conversion is to prefer if possible. I've read alot about changing audio pitch in iOS but most examples focus on playback, i.e. playing the sound with a different pitch. In my app I'm recording a movie (.m4v / AVFileTypeQuickTimeMovie) and saving it using standard AVAssetWriter. When saving the movie I have access to the following elements where I've tried to manipulate the audio (e.g. modify the pitch): audio buffer (CMSampleBufferRef) audio input writer (AVAssetWriterAudioInput) audio input writer options (e.g. AVNumberOfChannelsKey, AVSampleRateKey, AVChannelLayoutKey) asset writer (AVAssetWriter) I've tried to hook into the above objects to modify the audio pitch, but without success. I've also tried with Dirac as described here: Real Time Pitch Change In iPhone Using Dirac And OpenAL with AL_PITCH as described here: Piping output from OpenAL into a buffer And the "BASS" library from un4seen: Change Pitch/Tempo In Realtime I haven't found success with any of the above libs, most likely because I don't really know how to use them, and where to hook them into the audio saving code. There seems to be alot of librarys that have similar effects but focuses on playback or custom recording code. I want to manipulate the audio stream I've already got (AVAssetWriterAudioInput) or modify the saved movie clip (.m4v). I want the video to be unmodifed visually, i.e. played at the same speed. But I want the audio to go faster (like a chipmunk) or slower (like a ... monster? :)). Do you have any suggestions how I can modify the pitch in either real time (when recording the movie) or afterwards by converting the entire movie (.m4v file)? Should I look further into Dirac, OpenAL, SoundTouch, BASS or some other library? I want to be able to share the movie to others with modified audio, that's the reason I can't rely on modifying the pitch for playback only. Any help is appreciated, thanks!

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  • Pre-load audio files at the client-side for later use

    - by awj
    I'm building an online test which implements audio (mp3) using the native audio player (i.e. non Flash-based). The test shows one question at a time and loads each subsequent question asynchronously. Some questions have an accompanying audio file, others don't, and the audio files can be several MB in size. So what I'm hoping to do is to preload the audio files client-side at the start of the test and then move these into place when the relevant question comes up. So far I've tried loading an audio file into a QuickTime player, then when that question comes up I use jQuery's clone(true) method to copy this into a part of the page which is displayed. However, when I do this the QuickTime player has to reload the audio file from source. Same is true for Windows Media Player. Does anyone have any suggestions as to how I can preload the audio client-side and then call it forward when needed?

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  • Will An External Audio Interface Reduce CPU Load?

    - by Yar
    I am considering buying a very-low-latency audio interface like this one. One question is if it will reduce CPU load (I'm at about 60%+ and my Macbook has 2.4ghz and 4gigs ram) during intensive audio processing. If the answer is "yes," how will it compare to a different, cheaper firewire audio interface? My thought is that offloading the processing is always the same gain, regardless.

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