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  • How to change remote desktop audio quality?

    - by Vili
    When listening to some music on my remote computer, the audio quality is low. I've found this: HKLM\SOFTWARE\Microsoft\Windows NT\CurrentVersion\Drivers32\Terminal Server\RDP There are two interesting keys: EnableMP3Codec - with 0x00001 MaxBandwidth - this is 0x000056b9 Does anybody know anything about these keys? How should they work? What if i set EMP3C to 0? What if i change MaxBandwidth to higher number? Should i change these on my local computer and/or on the remote computer?

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  • Speech to text software (audio transcribing) for mac

    - by GiH
    What is the best speech to text software for mac? I have an hours worth of audio that I need to transcribe, and I'd really like to not have to do it manually :-). I prefer free options and I like open source so if there is a project I'd like to know. All answers are welcome though.

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  • Good Audio Splitter

    - by Jeremy White
    I need to get audio from my computer's headphone jack and push the output to 2 sets of speakers. I have tried using a cheap splitter from Fry's, but one set of speakers ends up acting as a microphone (!?) for the other set of speakers. What's the best way to split headphone output and get best quality with no interference on each set of speakers? I would, of course, also be interested in why the cheap splitter causes one set of speakers to start acting as a microphone.

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  • Looking for a SECURE Audio Player for Windows

    - by Avery Payne
    I know there are dozens of audio players for windows, but which one has: the best security record (least reported vulnerabilities) the least number of security exploits the best security implementation, along with the reasoning behind the implementaiton (i.e. how it is handled). As Windows users start to see more and more reports of media-based attacks against Windows Media Player (& co.) it would be useful to have a player that has some proactive security approaches to handling exploits.

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  • Writing ~ 630 MBs of MP3 to a Regular Audio CD

    - by Maxim Z.
    I have around 630 MBs of an MP3 podcast that I want to burn to a CD so that a friend can listen to it while driving. I understand that MP3 is very different from the normal audio CD format, but would it be possible to fit this amount onto a typical 700mb CD in the correct (non-MP3) format? I don't really care if some quality is lost. Thanks in advance!

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  • Looking for audio-sample-managing/librarian software

    - by Fuxi
    hi all, i'm looking for a good software for managing audio samples (.wav) it should be capable of: quickly browsing through directory/file-structure (eg. cursorkeys) + autoplay waveform-vizualisation + possibility to set cursor position (for playing) possibility to create own favourite-folder-structure + quickly sort in by sample-type any recommendations? thanks

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  • finding audio driver

    - by Jason S
    I have a soundcard in my new PC at work that does not have the right driver installed, and I'm not sure what the manufacturer is. (it's a generic PC running Windows XP) Is there a way to figure out what device it is w/o having to open the thing up or pester my IT department? When I look at it in Device Manager it says IDT High Definition Audio CODEC and when I look around in the Properties dialog boxes there's a Device Instance ID of HDAUDIO\FUNC_01&VEN_111D&DEV_76C1&SUBSYS_12974001&REV_1001\4&3612FC2B&0&0001

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  • Increase Audio volume on Avi files.

    - by Arch Angel
    Is there a simple way to adjust audio volume in avi files, like using MP3gain so i can set them all the same it is a problem having to keep adjusting the volume. I have tried VirtualDub but it doesnt handle my avi files as they are encoded in XVID

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  • Audio problems with Windows 7 running through Bootcamp

    - by cust0s
    I've been using (and enjoy using) Windows 7 through bootcamp on my mac for some time now. The only problem I've been having is with the audio, I seem to get a low hissing noise which is more prevalent when wearing headphones. Whilst it's not stopping me from using Windows 7 it's something I'd like fixed. I've tried downloading the latest drivers from Realteks website, but to no avail. Any suggestions?

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  • Sony Vegas: minimal audio glitches

    - by Fuxi
    When doing video-editing with Sony Vegas, I noticed that it kind of produces minimal audio glitches very often. With glitch, I mean a copied fragment of approximately 1/2 second, as if I copied it and directly pasted it again. I'm assuming it must be something with puffers. Does anyone know of this problem? I have a pretty fast Windows 7 system with fast SATA drives.

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  • Audio problems with Windows 7 running through Bootcamp

    - by cust0s
    I've been using (and enjoy using) Windows 7 through bootcamp on my mac for some time now. The only problem I've been having is with the audio, I seem to get a low hissing noise which is more prevalent when wearing headphones. Whilst it's not stopping me from using Windows 7 it's something I'd like fixed. I've tried downloading the latest drivers from Realteks website, but to no avail. Any suggestions?

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  • n68 s-ucc front panel audio

    - by user264522
    I have a motherboard asrock n68 s-ucc, and i need a hand identifying which pins on the board to connect my front audio header connectors to. On the connectors I have: double connector: GND and MIC IN double connector: LINE OUT FL and LINE OUT RL double connector: LINE OUT FR and LINE OUT RR simple connector: MIC POWER On the motherboard I have: GND PRESENCE# MIC_RET OUT_RET MIC2_L MIC2_R OUT2_R J_SENSE OUT2_L manual of motherboard and my front panel Thanks so much to all

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  • Dell 4600 Driver for Audio Card Windows 7

    - by Littlet-ENG
    I just installed windows 7 on dell 4600 desktop. the Audio card is a Creative Labs Sound Blaster Audigy 2. I've look all over, and I can't find a driver for windows 7. I've also looked at Vista Drivers. It seems Dell is not supporting Windows 7 with this computer. I've got a work around, but would like a driver if possible. Any help would be appreciated.

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  • windows live playback left and right audio channel

    - by user1254761
    I have a multichannel (4x stereo) audiocard (m-audio delta1010lt) and want to playback /playthru some of the channels live. But I am only able to playback/playthru the left channel on each stereo-input (CH1, CH3, CH5, CH7). For CH2,CH4,CH6,CH8 I see the Windows Volume-Indicator going up and down in the Windows Record-Audiosettings but I don't hear any playback sound. Is there a way to playback/playthru all input channels?

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  • Split audio into tracks?

    - by Mark
    I've recorded some music from an internet radio stream. I want to split it into separate audio tracks, preferably automatically (where ever there's a pause, or using some other clever algo). Anyone know of some free software that can do this?

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  • Audio organizing via CLI

    - by Radek Šimko
    I'm looking for some software for my OpenSUSE, which with I would be able to organize my audio files. I've found one, which may be good, but it's unable to run without X server (in CLI). http://musicbrainz.org/doc/MusicBrainz_Picard I'm not looking for ID3 renamers. There're maybe hundreds of them... I'm looking for software, which has its own database, or is able to communicate with some database, like CDDB, Gracenote, last.fm etc.

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  • Serving a video and audio upload and streaming intense site

    - by Pollux Khafra
    I'm about to launch a new site that allows user to both upload/stream audio and video and I don't know anything about the server side of things. My original plan was to just use a dedicated server through Hostgator but from what I'm reading, Cloud hosting or Load balanced clustered is the best way to go for what Im trying to do. All the articles seem to have an agenda to sell you on an affiliate web host so how do I really need to do this?

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  • Linux: grab audio from a video clip

    - by liori
    Hello, I'd like to take audio track from a video clip in FLV container and save it to something playable by portable music players. Are there any easy to use tools for that? I know how to do that using console tools (mplayer+lame/oggenc), but I'd like to get something clickable, preferably for GNOME. Thanks!

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  • Using R in Processing through rJava/JRI?

    - by tommy-o-dell
    Hi guys, Is it possible to run R in Processing through rJava/JRI? If I deployed a Processing app on the web, would the client need R on their system? I'm looking to create an interactive information dashboard that I can deploy on the web. It seems that Processing is probably my best bet for the interactive/web part of things. Unfortunately, it doesn't look like there are many math/stats functions built-in. And there aren't any libraries for plotting data either. I've been using R and gpplot2 for a few months and am thrilled (amazed) at how easily it manipulates and plots data. So I'm wondering now if can get the best of both worlds and run R through a Processing applet. From the JRI website: JRI is a Java/R Interface, which allows to run R inside Java applications as a single thread. Basically it loads R dynamic library into Java and provides a Java API to R functionality. It supports both simple calls to R functions and a full running REPL. In a sense JRI is the inverse of rJava and both can be combined (i.e. you can run R code inside JRI that calls back to the JVM via rJava). The JGR project makes the full use of both JRI and rJava to provide a full Java GUI for R. JRI uses native code, but it supports all platforms where Sun's Java (or compatible) is available, including Windows, Mac OS X, Sun and Linux (both 32-bit and 64-bit). Thanks for the advice :)

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  • How to write C++ audio processing applications?

    - by cesko82
    Hi everyone, I'm an Electronics and Telecommunications student, next to my graduation. I'm gonna work on a project that involves my knowledge about DSP, music and audio in general. I allready know all the basic mathematic instruments and all the stuff I need to manage it, such as FFT, circular convolution ecc ecc. I want to learn C++ programming basically for one reason: it's very important in the professional world!!! And I think it's one of the most used to write applications working with audio, especially when it's about real time processing. Ok, after this small introduction I would like to know first, which are the most used libraries to work with audio processing in c++?? I was longer looking on the web but i couldn't find a lo of working stuff. (I work under linux with eclipse CDT enviroment). Then I would like to know if there are good sources to learn how to write some working code, such as for example how to write a simple low pass filter. Basically now i will not write real time applications, I would like to start from the processing of a WAV file, or even better an MP3 file, so basically on vectors of samples. Let's say that basically for now I would like to extract the waveform from an audio file, and save it to a thumbnail or to a PNG image. Ok, for now I think it's all I would need. Any ideas, advices, libraries, books, interesting sources about that? Thanks a lot in advance for any kind of answer. Giovanni.

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  • Serverside image processing

    - by spol
    I am designing a web application that does server side image processing in real time. Processing tasks include applying different effects like grayscale, blur, oil paint, pencil sketch etc on images in various formats. I want to build it using java/servlets which I am already familiar with. I found 3 options, 1) Use pure java imaging libraries like java.awt or http://www.jhlabs.com/ip/index.html 2) Use command line tools like Gimp/ImageMagick 3) Use c,c++ image libraries that have java bindings. I don't know which of the above options is good keeping the performance in mind. It looks like option 2) and 3) are good performance wise, but I want to be sure before I rule out 1). I have also heard gimp cannot be run using command line unless gtk or xwindows is already installed on the server. Will there be any such problems with 2) or 3) while running them server side? Also please suggest any good image processing libraries for this purpose.

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  • Parallel processing slower than sequential?

    - by zebediah49
    EDIT: For anyone who stumbles upon this in the future: Imagemagick uses a MP library. It's faster to use available cores if they're around, but if you have parallel jobs, it's unhelpful. Do one of the following: do your jobs serially (with Imagemagick in parallel mode) set MAGICK_THREAD_LIMIT=1 for your invocation of the imagemagick binary in question. By making Imagemagick use only one thread, it slows down by 20-30% in my test cases, but meant I could run one job per core without issues, for a significant net increase in performance. Original question: While converting some images using ImageMagick, I noticed a somewhat strange effect. Using xargs was significantly slower than a standard for loop. Since xargs limited to a single process should act like a for loop, I tested that, and found it to be about the same. Thus, we have this demonstration. Quad core (AMD Athalon X4, 2.6GHz) Working entirely on a tempfs (16g ram total; no swap) No other major loads Results: /media/ramdisk/img$ time for f in *.bmp; do echo $f ${f%bmp}png; done | xargs -n 2 -P 1 convert -auto-level real 0m3.784s user 0m2.240s sys 0m0.230s /media/ramdisk/img$ time for f in *.bmp; do echo $f ${f%bmp}png; done | xargs -n 2 -P 2 convert -auto-level real 0m9.097s user 0m28.020s sys 0m0.910s /media/ramdisk/img$ time for f in *.bmp; do echo $f ${f%bmp}png; done | xargs -n 2 -P 10 convert -auto-level real 0m9.844s user 0m33.200s sys 0m1.270s Can anyone think of a reason why running two instances of this program takes more than twice as long in real time, and more than ten times as long in processor time to complete the same task? After that initial hit, more processes do not seem to have as significant of an effect. I thought it might have to do with disk seeking, so I did that test entirely in ram. Could it have something to do with how Convert works, and having more than one copy at once means it cannot use processor cache as efficiently or something? EDIT: When done with 1000x 769KB files, performance is as expected. Interesting. /media/ramdisk/img$ time for f in *.bmp; do echo $f ${f%bmp}png; done | xargs -n 2 -P 1 convert -auto-level real 3m37.679s user 5m6.980s sys 0m6.340s /media/ramdisk/img$ time for f in *.bmp; do echo $f ${f%bmp}png; done | xargs -n 2 -P 1 convert -auto-level real 3m37.152s user 5m6.140s sys 0m6.530s /media/ramdisk/img$ time for f in *.bmp; do echo $f ${f%bmp}png; done | xargs -n 2 -P 2 convert -auto-level real 2m7.578s user 5m35.410s sys 0m6.050s /media/ramdisk/img$ time for f in *.bmp; do echo $f ${f%bmp}png; done | xargs -n 2 -P 4 convert -auto-level real 1m36.959s user 5m48.900s sys 0m6.350s /media/ramdisk/img$ time for f in *.bmp; do echo $f ${f%bmp}png; done | xargs -n 2 -P 10 convert -auto-level real 1m36.392s user 5m54.840s sys 0m5.650s

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  • MPlayer refuses to generate mono wav file

    - by JCCyC
    I want to downsample an existing audio file to 8KHz mono. This command line downsamples it to stereo: mplayer -quiet -vo null -vc dummy -af volume=0,resample=8000:0:1 -ao pcm:waveheader:file="/tmp/blah1.wav" ~/from_my_cellphone.3ga It generates a file that the file utility identifies as stereo: $ file /tmp/blah1.wav /tmp/blah1.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz Now, if I read the documentation correctly, I should add pan=1:0.5:0.5 so I get a file that's half the size: mplayer -quiet -vo null -vc dummy -af volume=0,resample=8000:0:1:pan=1:0.5:0.5 -ao pcm:waveheader:file="/tmp/blah2.wav" ~/from_my_cellphone.3ga But it doesn't! blah2.wav is identical to blah1.wav! What am I doing wrong?

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  • Reading audio with Extended Audio File Services (ExtAudioFileRead)

    - by Paperflyer
    I am working on understanding Core Audio, or rather: Extended Audio File Services Here, I want to use ExtAudioFileRead() to read some audio data from a file. This works fine as long as I use one single huge buffer to store my audio data (that is, one AudioBuffer). As soon as I use more than one AudioBuffer, ExtAudioFileRead() returns the error code -50 ("error in parameter list"). As far as I can figure out, this means that one of the arguments of ExtAudioFileRead() is wrong. Probably the audioBufferList. I can not use one huge buffer because then, dataByteSize would overflow its UInt32-integer range with huge files. Here is the code to create the audioBufferList: AudioBufferList *audioBufferList; audioBufferList = malloc(sizeof(AudioBufferList) + (numBuffers-1)*sizeof(AudioBuffer)); audioBufferList->mNumberBuffers = numBuffers; for (int bufferIdx = 0; bufferIdx<numBuffers; bufferIdx++ ) { audioBufferList->mBuffers[bufferIdx].mNumberChannels = numChannels; audioBufferList->mBuffers[bufferIdx].mDataByteSize = dataByteSize; audioBufferList->mBuffers[bufferIdx].mData = malloc(dataByteSize); } UInt32 numFrames = fileLengthInFrames; error = ExtAudioFileRead(extAudioFileRef, &numFrames, audioBufferList); Do you know what I am doing wrong here?

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