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  • Realtek HD Audio 5.1 on Windows 7

    - by Darth
    I have problem with Realtek 5.1 driver for Windows 7 x64. I've installed newest drivers for Realtek HD Audio, but 5.1 still doesn't work, the only thing that works is front stereo. However, when I click on single speaker in sound settings test, every one of them work. Sorry that the image isn't in english, but I hope you can understand the point.

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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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  • Use lineout on Mac for audio processing

    - by eltufto
    Does anyone know of a way to set the linein on a Mac to route from the lineout? I want to essentially work with the music played on my Mac. An alternative solution I suppose would be to set the Linein as the output from iTunes/other music player. For context: My specific use-case is audio processing in Processing with the minim library, calling getLineIn. Trying to create some visuals that respond to the music being played.

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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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  • 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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  • 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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  • 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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  • 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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  • 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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  • iPhone Audio Queue Service sample units

    - by pion
    I am looking at Audio Queue Services document specifically on the following code: // Writing an audio queue buffer to disk AudioFileWritePackets ( // 1 pAqData->mAudioFile, // 2 false, // 3 inBuffer->mAudioDataByteSize, // 4 inPacketDesc, // 5 pAqData->mCurrentPacket, // 6 &inNumPackets, // 7 inBuffer->mAudioData // 8 ); inBuffer-mAudioDataByteSize is the number of bytes of audio data being written. inBuffer-mAudioData is the new audio data to write to the audio file. Assuming the sample rate is 44100. AudioStreamBasicDescription mDataFormat; mDataFormat.mSampleRate = 44100.0f; mDataFormat.mBitsPerChannel = 16; ... NSInteger numberSamples = inBuffer->mAudioDataByteSize / 2; SInt16 *audioSample = (SInt16 *)inBuffer->mAudioData; I use core-plot to plot the above where x axis is number of sample [1 .. numberSamples] and the y axis is audioSample[0] .. audioSample[numberSamples]. I can see the chart in "real-time" where the y axis goes up and down depending the loudness of my voice. Beginner questions: What does the audioSample represent? What am I looking at here? What is the unit of audioSample? What do I need to do if I just want to plot the range between 50 - 100 Hz? Thanks in advance for your help.

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