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  • Recording Audio through M-Audio Keystudio

    - by interstar
    Hi, I'm trying to get my M-Audio Keystudio (which has an audio input as well as the keyboard) to record audio to Audacity. I'm in Ubuntu 10.10. When I look at the Sound Preferences I can select "M-Audio RunTime DFU Analog Stereo" as my input device. However, when I try to record in Audacity, Audacity remains frozen. The program seems to be running and recording, but the recording cursor won't advance. If I reset the audio input to the internal sound card, recording works normally. Any ideas what to look for?

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  • Is encoding needed in this decryption?

    - by Lijo
    I have a Encryption – Decryption scenario as shown below. //[Clear text ID string as input] -- [(ASCII GetByte) + Encoding] -- [Encrption as byte array] -- [Database column is in VarBinary] -- [Pass byte[] as VarBinary parameter to SP for comparison] //[ID stored as VarBinary in Database] -- [Read as byte array] -- [(Decrypt as byte array) + Encoding + (ASCII Get String)] -- Show as string in the UI My question is in the decryption scenario. After decryption I get a byte array. I am doing an encoding (IBM037) after that. Is it correct? Is there something wrong in the flow shown above? private static byte[] GetEncryptedID(string id) { Interface_Request input = new Interface_Request(); input.RequestText = Encodeto64(id); input.RequestType = Encryption; ProgramInterface inputRequest = new ProgramInterface(); inputRequest.Test_Trial_Request = input; using (KTestService operation = new KTestService()) { return ((operation.KTrialOperation(inputRequest)).Test_Trial_Response.ResponseText); } } private static string GetDecryptedID(byte[] id) { Interface_Request input = new Interface_Request(); input.RequestText = id; input.RequestType = Decryption; ProgramInterface request = new ProgramInterface(); request.Test_Trial_Request = input; using (KTestService operationD = new KTestService()) { ProgramInterface1 response = operationD.KI014Operation(request); byte[] decryptedValue = response.ICSF_AES_Response.ResponseText; Encoding sourceByteFormat = Encoding.GetEncoding("IBM037"); Encoding destinationByteFormat = Encoding.ASCII; //Convert from one byte format to other (IBM to ASCII) byte[] ibmEncodedBytes = Encoding.Convert(sourceByteFormat, destinationByteFormat,decryptedValue); return System.Text.ASCIIEncoding.ASCII.GetString(ibmEncodedBytes); } } private static byte[] EncodeTo64(string toEncode) { byte[] dataInBytes = System.Text.ASCIIEncoding.ASCII.GetBytes(toEncode); Encoding destinationByteFormat = Encoding.GetEncoding("IBM037"); Encoding sourceByteFormat = Encoding.ASCII; //Convert from one byte format to other (ASCII to IBM) byte[] asciiBytes = Encoding.Convert(sourceByteFormat, destinationByteFormat, dataInBytes); return asciiBytes; }

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  • No audio with headphones, but audio works with integrated speakers

    - by Pedro
    My speakers work correctly, but when I plug in my headphones, they don't work. I am running Ubuntu 10.04. My audio card is Realtek ALC259 My laptop model is a HP G62t a10em In another thread someone fixed a similar issue (headphones work, speakers not) folowing this: sudo vi /etc/modprobe.d/alsa-base.conf (or some other editor instead of Vi) Append the following at the end of the file: alias snd-card-0 snd-hda-intel options snd-hda-intel model=auto Reboot but it doesnt work for me. Before making and changes to alsa, this was the output: alsamixer gives me this: Things I did: followed this HowTo but now no hardware seems to be present (before, there were 2 items listed): Now, alsamixer gives me this: alsamixer: relocation error: alsamixer: symbol snd_mixer_get_hctl, version ALSA_0.9 not defined in file libasound.so.2 with link time reference I guess there was and error in the alsa-driver install so I began reinstalling it. cd alsa-driver* //this works fine// sudo ./configure --with-cards=hda-intel --with-kernel=/usr/src/linux-headers-$(uname -r) //this works fine// sudo make //this doesn't work. see ouput error below// sudo make install Final lines of sudo make: hpetimer.c: In function ‘snd_hpet_open’: hpetimer.c:41: warning: implicit declaration of function ‘hpet_register’ hpetimer.c:44: warning: implicit declaration of function ‘hpet_control’ hpetimer.c:44: error: expected expression before ‘unsigned’ hpetimer.c: In function ‘snd_hpet_close’: hpetimer.c:51: warning: implicit declaration of function ‘hpet_unregister’ hpetimer.c:52: error: invalid use of undefined type ‘struct hpet_task’ hpetimer.c: In function ‘hpetimer_init’: hpetimer.c:88: error: ‘EINVAL’ undeclared (first use in this function) hpetimer.c:99: error: invalid use of undefined type ‘struct hpet_task’ hpetimer.c:100: error: invalid use of undefined type ‘struct hpet_task’ hpetimer.c: At top level: hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: error: extra brace group at end of initializer hpetimer.c:121: error: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) make[1]: *** [hpetimer.o] Error 1 make[1]: Leaving directory `/usr/src/alsa/alsa-driver-1.0.9/acore' make: *** [compile] Error 1 And then sudo make install gives me: rm -f /lib/modules/0.0.0/misc/snd*.*o /lib/modules/0.0.0/misc/persist.o /lib/modules/0.0.0/misc/isapnp.o make[1]: Entering directory `/usr/src/alsa/alsa-driver-1.0.9/acore' mkdir -p /lib/modules/0.0.0/misc cp snd-hpet.o snd-page-alloc.o snd-pcm.o snd-timer.o snd.o /lib/modules/0.0.0/misc cp: cannot stat `snd-hpet.o': No such file or directory cp: cannot stat `snd-page-alloc.o': No such file or directory cp: cannot stat `snd-pcm.o': No such file or directory cp: cannot stat `snd-timer.o': No such file or directory cp: cannot stat `snd.o': No such file or directory make[1]: *** [_modinst__] Error 1 make[1]: Leaving directory `/usr/src/alsa/alsa-driver-1.0.9/acore' make: *** [install-modules] Error 1 [SOLUTION] After screwing it all up, someone mentioned why not trying using the packages in Synaptic - so I did. I have reinstalled the following packages and rebooter: -alsa-hda-realtek-ignore-sku-dkms -alsa-modules-2.6.32-25-generic -alsa-source -alsa-utils -linux-backports-modules-alsa-lucid-generic -linux-backports-modules-alsa-lucid-generic-pae -linux-sound-base -(i think i listed them all) After rebooting, the audio worked, both in speakers and headphones. I have no idea which is the package that made my audio work, but it certainly was one of them. [/SOLUTION]

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  • Extract Audio from a Video File with Pazera Free Audio Extractor

    - by DigitalGeekery
    Have you ever wanted to extract some or all of the audio from a video file?  Today we’ll take a look at Pazera Free Audio Extractor. A simple audio converter that specializes in that very task. Download the Pazera Free Audio Extractor. (See download link below) You’ll need to unzip the download folder, but there is no need to install the application. Simply double-click the AudioExtractor.exe file to run the application. To add your video files to the queue to be converted, click on the Add files  button at the top left. You can add multiple files to the queue and convert them all at one time. Browse for your video file, and click Open.   Your video will be added to the Queue for processing.   Under Output directory you can choose to output to a folder of your choice. Outputting to the same folder as the input folder is the default.   Pazera Free Audio Extractor includes pre-configured profiles that will simplify the process of choosing conversion settings. To load a profile, choose one from the Profile drop down list and then click the Load button. You can choose to output to MP3, AAC, AC3, WMA, FLAC, OGG or WAV file format.   You will see the profile update the Audio settings in the panels at the lower left of the application. If you wish, you may also select your own custom settings. Advanced Settings The Advanced settings can be used if you want to extract only a portion of the the audio, such as a clip of dialog or a song from a movie. To extract only a portion of the audio, set the start time by selecting the Start time offset check box, then entering the time in the video clip where the audio begins. To set the end time, begin by selecting the Duration check box. Now, you can either select the Duration radio button and enter the amount of time for which you would like to extract the audio, or you can select the End time offset radio button and enter the time in the video clip where the audio ends. When you are ready to convert, click the CONVERT button on the menu at the top of the screen.   An output box will open and display the conversion progress. When finished, click Close.   Now you are ready to enjoy your audio clip. Pazera Free Audio Extractor is a basic audio tool that is easy enough for everyone to use. It runs on Windows only and supports most common video formats including AVI, FLV, MP4, MPG, MOV, 3GP, and WMV. Download Free Audio Extractor 1.3 Similar Articles Productive Geek Tips Eufony Free Audio Player – Resource Gentle Audio PlayerConvert .3GP and .3G2 Files to AVI / MPEG for FreeTurn Off Auto-Play of Audio and Video CDs and DVDs in UbuntuHow to Make/Edit a movie with Windows Movie Maker in Windows VistaEasily Change Audio File Formats with XRECODE TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 Use Printflush to Solve Printing Problems Icelandic Volcano Webcams Open Multiple Links At One Go NachoFoto Searches Images in Real-time Office 2010 Product Guides Google Maps Place marks – Pizza, Guns or Strip Clubs

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  • C# Check if character exists in encoding

    - by Alvin Wong
    I am writing a program that a part renders a bitmap font in CP437. In a function that renders the text with I want to be able to check whether a char is available in CP437 before the encoding conversion, like: public static void DrawCharacter(this Graphics g, char c) { if (char_exist_in_encoding(Encoding.GetEncoding(437), c) { byte[] src = Encoding.Unicode.GetBytes(c.ToString()); byte[] dest = Encoding.Convert(Encoding.Unicode, Encoding.GetEncoding(437), src); DrawCharacter(g, dest[0]); // Call the void(this Graphics, byte) overload } } Without the check, any characters outside CP437 will result in a '?' (63, 0x3F). I want to hide any invalid characters completely. Is there an implementation of char_exist_in_encoding other than the following stupid approach? public static bool char_exist_in_encoding(Encoding e, char c) { if (c == '?') return true; byte[] src = Encoding.Unicode.GetBytes(c.ToString()); byte[] dest = Encoding.Convert(Encoding.Unicode, Encoding.GetEncoding(437), src); if (dest[0] == 0x3F) return false; return true; } Perhaps not very relevant, but the bitmap is created like this: Bitmap b = new Bitmap(256 * 8, 16); Graphics g = Graphics.FromImage(b); g.TextRenderingHint = System.Drawing.Text.TextRenderingHint.SingleBitPerPixelGridFit; Font f = new Font("Whatever 8x16 bitmap font", 16, GraphicsUnit.Pixel); for (byte i = 0; i < 255; i++) { byte[] arr = Encoding.Convert(Encoding.GetEncoding(437), Encoding.Unicode, new byte[] { i }); char c = Encoding.Unicode.GetChars(arr)[0]; g.DrawString(c.ToString(), f, Brushes.Black, i * 8 - 3, 0); // Don't know why it needs a 3px offset } b.Save(@"D:\chars.png");

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  • MODx character encoding

    - by Piet
    Ahhh character encodings. Don’t you just love them? Having character issues in MODx? Then probably the MODx manager character encoding, the character encoding of the site itself, your database’s character encoding, or the encoding MODx/php uses to talk to MySQL isn’t correct. The Website Encoding Your MODx site’s character encoding can be configured in the manager under Tools/Configuration/Site/Character encoding. This is the encoding your website’s visitors will get. The Manager’s Encoding The manager’s encoding can be changed by setting $modx_manager_charset at manager/includes/lang/<language>.inc.php like this (for example): $modx_manager_charset = 'iso-8859-1'; To find out what language you’re using (and thus was file you need to change), check Tools/Configuration/Site/Language (1 line above the character encoding setting). This needs to be the same encoding as your site. You can’t have your manager in utf8 and your site in iso-8859-1. Your Database’s Encoding The charset MODx/php uses to talk to your database can be set by changing $database_connection_charset in manager/includes/config.inc.php. This needs to be the same as your database’s charset. Make sure you use the correct corresponding charset, for iso-8859-1 you need to use ‘latin1′. Utf8 is just utf8. Example: $database_connection_charset = 'latin1'; Now, if you check Reports/System info, the ‘Database Charset’ might say something else. This is because the mysql variable ‘character_set_database’ is displayed here, which contains the character set used by the default database and not the one for the current database/connection. However, if you’d change this to display ‘character_set_connection’, it could still say something else because the ’set character set’ statement used by MODx doesn’t change this value either. The ’set names’ statement does, but since it turns out my MODx install works now as expected I’ll just leave it at this before I get a headache. If I saved you a potential headache or you think I’m totally wrong or overlooked something, let me know in the comments. btw: I want to be able to use a real editor with MODx. Somehow.

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  • guvcview recording video and audio out of synchronisation in Ubuntu 10.10

    - by SIJAR
    I finally got Guvcview, a great software for Logitech webcam and it does all the stuff that one wants out of it. But I'm not satisfy with the video recording, video and audio out of synchronisation also video seems to be in slow motion. Please help so that I can tweak in and get a good video recording with the webcam. Below is the log of Guvcview ------------------------------------------------------------------------------- guvcview 1.4.1 video_device: /dev/video0 vid_sleep: 0 cap_meth: 1 resolution: 640 x 480 windowsize: 1024 x 715 vert pane: 578 spin behavior: 0 mode: mjpg fps: 1/25 Display Fps: 0 bpp: 0 hwaccel: 1 avi_format: 4 sound: 1 sound Device: 4 sound samp rate: 0 sound Channels: 0 Sound delay: 0 nanosec Sound Format: 85 Pan Step: 2 degrees Tilt Step: 2 degrees Video Filter Flags: 0 image inc: 0 profile(default):/home/sijar/default.gpfl starting portaudio... bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started language catalog= dir:/usr/share/locale type:UTF-8 lang:en_US.utf8 cat:guvcview.mo mjpg: setting format to 1196444237 capture method = 1 video device: /dev/video0 libv4lconvert: warning more framesizes then I can handle! libv4lconvert: warning more framesizes then I can handle! /dev/video0 - device 1 libv4lconvert: warning more framesizes then I can handle! libv4lconvert: warning more framesizes then I can handle! Init. UVC Camera (046d:0825) (location: usb-0000:00:1d.7-5) { pixelformat = 'YUYV', description = 'YUV 4:2:2 (YUYV)' } { discrete: width = 640, height = 480 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 160, height = 120 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 176, height = 144 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 320, height = 176 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 320, height = 240 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 352, height = 288 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 432, height = 240 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 544, height = 288 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 640, height = 360 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, ... repeats a couple of times ... vid:046d pid:0825 driver:uvcvideo Adding control for Pan (relative) UVCIOC_CTRL_ADD - Error: Operation not permitted checking format: 1196444237 VIDIOC_G_COMP:: Invalid argument compression control not supported fps is set to 1/25 drawing controls control[0]: 0x980900 Brightness, 0:255:1, default 128 control[0]: 0x980901 Contrast, 0:255:1, default 32 control[0]: 0x980902 Saturation, 0:255:1, default 32 control[0]: 0x98090c White Balance Temperature, Auto, 0:1:1, default 1 control[0]: 0x980913 Gain, 0:255:1, default 0 control[0]: 0x980918 Power Line Frequency, 0:2:1, default 2 control[0]: 0x98091a White Balance Temperature, 0:10000:10, default 4000 control[0]: 0x98091b Sharpness, 0:255:1, default 24 control[0]: 0x98091c Backlight Compensation, 0:1:1, default 1 control[0]: 0x9a0901 Exposure, Auto, 0:3:1, default 3 control[0]: 0x9a0902 Exposure (Absolute), 1:10000:1, default 166 control[0]: 0x9a0903 Exposure, Auto Priority, 0:1:1, default 0 resolutions of format(2) = 19 frame rates of 1º resolution=6 Def. Res: 0 numb. fps:6 --------------------------------------- device #0 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 (hw:0,0) Host API = ALSA Max inputs = 2, Max outputs = 2 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #1 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - MIC ADC (hw:0,1) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #2 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - MIC2 ADC (hw:0,2) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #3 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - ADC2 (hw:0,3) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #4 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - IEC958 (hw:0,4) Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #5 Name = USB Device 0x46d:0x825: USB Audio (hw:1,0) Host API = ALSA Max inputs = 1, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #6 Name = front Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.012 Def. high input latency = -1.000 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #7 Name = iec958 Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #8 Name = spdif Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #9 Name = pulse Host API = ALSA Max inputs = 32, Max outputs = 32 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #10 Name = dmix Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.043 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #11 [ Default Input, Default Output ] Name = default Host API = ALSA Max inputs = 32, Max outputs = 32 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 ---------------------------------------------- SampleRate:0 Channels:0 Video driver: x11 A window manager is available VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_CTRL for user class controls control(0x0098091a) "White Balance Temperature" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_EXT_CTRLS on single controls for class: 0x009a0000 control(0x009a0902) "Exposure (Absolute)" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_CTRL for user class controls control(0x0098091a) "White Balance Temperature" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_EXT_CTRLS on single controls for class: 0x009a0000 control(0x009a0902) "Exposure (Absolute)" failed to set (error -1) Cap Video toggled: 1 (/home/sijar/Videos/Webcam) 25371756K bytes free on a total of 39908968K (used: 36 %) treshold=51200K using audio codec: 0x0055 Audio frame size is 1152 samples for selected codec IO thread started...OK [libx264 @ 0x8cbd8b0]using cpu capabilities: MMX2 SSE2 Cache64 [libx264 @ 0x8cbd8b0]profile Baseline, level 3.0 [libx264 @ 0x8cbd8b0]non-strictly-monotonic PTS shift sound by -9 ms shift sound by -9 ms shift sound by -9 ms AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... AUDIO: droping audio data (/home/sijar/Videos/Webcam) 25371748K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... Cap Video toggled: 0 Shuting Down IO Thread AUDIO: droping audio data stop= 4426644744000 start=4416533023000 VIDEO: 146 frames in 10111.000000 ms = 14.439719 fps Stoping audio stream Closing audio stream... close avi Last message repeated 145 times [libx264 @ 0x8cbd8b0]frame I:2 Avg QP:14.10 size: 24492 [libx264 @ 0x8cbd8b0]frame P:103 Avg QP:16.06 size: 20715 [libx264 @ 0x8cbd8b0]mb I I16..4: 48.4% 0.0% 51.6% [libx264 @ 0x8cbd8b0]mb P I16..4: 57.5% 0.0% 0.0% P16..4: 40.2% 0.0% 0.0% 0.0% 0.0% skip: 2.3% [libx264 @ 0x8cbd8b0]final ratefactor: 62.05 [libx264 @ 0x8cbd8b0]coded y,uvDC,uvAC intra: 79.7% 92.2% 68.4% inter: 62.4% 87.5% 48.0% [libx264 @ 0x8cbd8b0]i16 v,h,dc,p: 23% 17% 41% 19% [libx264 @ 0x8cbd8b0]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 30% 24% 26% 2% 5% 3% 3% 3% 4% [libx264 @ 0x8cbd8b0]i8c dc,h,v,p: 53% 20% 23% 4% [libx264 @ 0x8cbd8b0]ref P L0: 63.0% 37.0% [libx264 @ 0x8cbd8b0]kb/s:-0.00 total frames encoded: 0 total audio frames encoded: 0 IO thread finished...OK IO Thread finished enabling controls Cap Video toggled: 1 (/home/sijar/Videos/Webcam) 25379744K bytes free on a total of 39908968K (used: 36 %) treshold=51200K using audio codec: 0x0055 Audio frame size is 1152 samples for selected codec IO thread started...OK [libx264 @ 0x8cfba20]using cpu capabilities: MMX2 SSE2 Cache64 [libx264 @ 0x8cfba20]profile Baseline, level 3.0 [libx264 @ 0x8cfba20]non-strictly-monotonic PTS shift sound by -236 ms shift sound by -236 ms shift sound by -236 ms (/home/sijar/Videos/Webcam) 25377044K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25373408K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... (/home/sijar/Videos/Webcam) 25370696K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... (/home/sijar/Videos/Webcam) 25367680K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25364052K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25360312K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25356628K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25352908K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25349316K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25345552K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25341828K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25338092K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25334412K bytes free on a total of 39908968K (used: 36 %) treshold=51200K Cap Video toggled: 0 Shuting Down IO Thread stop= 4708817235000 start=4578624714000 VIDEO: 1604 frames in 130192.000000 ms = 12.320265 fps Stoping audio stream Closing audio stream... close avi Last message repeated 1603 times [libx264 @ 0x8cfba20]frame I:16 Avg QP:14.78 size: 42627 [libx264 @ 0x8cfba20]frame P:1547 Avg QP:16.44 size: 28599 [libx264 @ 0x8cfba20]mb I I16..4: 21.6% 0.0% 78.4% [libx264 @ 0x8cfba20]mb P I16..4: 28.1% 0.0% 0.0% P16..4: 70.5% 0.0% 0.0% 0.0% 0.0% skip: 1.4% [libx264 @ 0x8cfba20]final ratefactor: 88.17 [libx264 @ 0x8cfba20]coded y,uvDC,uvAC intra: 74.4% 95.8% 83.2% inter: 75.2% 94.6% 69.2% [libx264 @ 0x8cfba20]i16 v,h,dc,p: 27% 17% 40% 16% [libx264 @ 0x8cfba20]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 25% 21% 3% 6% 4% 5% 4% 7% [libx264 @ 0x8cfba20]i8c dc,h,v,p: 61% 18% 18% 4% [libx264 @ 0x8cfba20]ref P L0: 64.0% 36.0% [libx264 @ 0x8cfba20]kb/s:-0.00 total frames encoded: 0 total audio frames encoded: 0 IO thread finished...OK IO Thread finished enabling controls Shuting Down Thread Thread terminated... cleaning Thread allocations: 100% SDL Quit Video Thread finished write /home/sijar/.guvcviewrc OK free audio mutex closed v4l2 strutures free controls free controls - vidState cleaned allocations - 100% Closing portaudio ...OK Closing GTK... OK

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  • How to create a virtual audio device and stream audio input with it

    - by Steven Rosato
    Is it possible to create another audio device and redirect only wanted input streams to it? Here's my concrete problem: I am broadcasting a game via XFire and it uses the Windows audio device to capture any audio I receive. As I am broadcasting, other users who watch the video stream are communicating with me over Skype, and they hear themselves back within the video stream and it is entirely logical since I am broadcasting the audio I hear. What I want to do is create another audio device within Windows and redirect (pipe) ONLY the audio input from that game and not the input reveived from Skype. I would then tell XFire to use that newly created "virtual" audio device to broadcast and therefore my partners won't hear themselves back. Is there any software that can do that or can it be achieved natively with Windows? (I am under Windows 7).

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  • How to create a virtual audio device and stream audio input with it

    - by Steven Rosato
    Is it possible to create another audio device and redirect only wanted input streams to it? Here's my concrete problem: I am broadcasting a game via XFire and it uses the Windows audio device to capture any audio I receive. As I am broadcasting, other users who watch the video stream are communicating with me over Skype, and they hear themselves back within the video stream and it is entirely logical since I am broadcasting the audio I hear. What I want to do is create another audio device within Windows and redirect (pipe) ONLY the audio input from that game and not the input reveived from Skype. I would then tell XFire to use that newly created "virtual" audio device to broadcast and therefore my partners won't hear themselves back. Is there any software that can do that or can it be achieved natively with Windows? (I am under Windows 7).

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  • what is the best way to stream a audio file to website users/listners

    - by Naveen Chamikara Gamage
    I'm developing a music site which will stream audio files stored in a server to users, audio files will be played through flash player placed in a webpage.. As I heard I need to use a streaming media server for streaming audio files ( like 2mb to 3mb in size).. Do I need to use one? I found some streaming media server softwares like http://www.icecast.org - but as in their documentation, It is used for streaming radio stations and live streaming purposes, but I just need to stream audio files faster and in low size (low bandwidth) with good quality.. I heard I need to encode the audio files first and then send them to listeners and in their end audio files need to be decoded again. Is that true? How can I do that? if I need to use a special web server, where should I host my files? Any good hosting providers? if I host audio files in a normal web server, they will use HTTP or TCP to deliver my audio files to users/ listners but I found that HTTP and TCP are not good ways to use for multi media purposes like streaming audio and video files, and they are used for delivering HTML and stuff. I found I should use RSTP or UDP for streaming audio files.. What should I use? I know that .MP3 files has much better quality than the other formats but it also gives huge size to the audio files.. which format should I use for audio files? Most of the best quality audio files are more than 7mb so I'm planning to convert them my self using a software so I could get low size files with some level of good quality. If I'm converting my audio files what is the good BITRATE I should use for my files? Any known best softwares for converting audio files while keeping quality in a good level? Note** - I know that I will not need complex requirements at the beginning of the site but I wanted to what are the best ways like they are using for soundcloud.com

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  • Combine Multiple Audio Files into a single higher-quality audio File

    - by namenlos
    BACKGROUND My team gave a demo to a large audience - we recorded the audio of the demo in multiple locations in the room (3) the audio was recorded using cheap laptop microphones I was not involved in the recording of the audio or the demo Both audio files suck in some form the first one is of a recording near the speaker - which clearly gets his voice but the the audience is audience is muffled - also this one is slightly noisy The second recording was done in the middle of the audience - it gets the audience questions clearly but actually gets the speaker rather sometimes well and sometimes poorly (not all the speakers spoke loudly enough to be heard) MY QUESTION Is there any techinque or software which can be used to merge these audio files in such a way that the best qualities of each are preserved. I am NOT asking now to simply merge them together in one track - I've already done that in Audacity and it is certainly better - what I am looking for could be considered closer to how HDR images are created - multiple exposures combined into an enhanced new version which is not simply an average of the inputs. NOTE Am not an "Audio" guy - just a normal user

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  • What encoding is used by javax.xml.transform.Transformer?

    - by Simon Reeves
    Please can you answer a couple of questions based on the code below (excludes the try/catch blocks), which transforms input XML and XSL files into an output XSL-FO file: File xslFile = new File("inXslFile.xsl"); File xmlFile = new File("sourceXmlFile.xml"); TransformerFactory tFactory = TransformerFactory.newInstance(); Transformer transformer = tFactory.newTransformer(new StreamSource(xslFile)); FileOutputStream fos = new FileOutputStream(new File("outFoFile.fo"); transformer.transform(new StreamSource(xmlFile), new StreamResult(fos)); inXslFile is encoded using UTF-8 - however there are no tags in file which states this. sourceXmlFile is UTF-8 encoded and there may be a metatag at start of file indicating this. am currently using Java 6 with intention of upgrading in the future. What encoding is used when reading the xslFile? What encoding is used when reading the xmlFile? What encoding will be applied to the FO outfile? How can I obtain the info (properties) for 1 - 3? Is there a method call? How can the properties in 4 be altered - using configuration and dynamically? if known - Where is there info (web site) on this that I can read - I have looked without much success.

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  • Two audio streams - headphones and speakers

    - by Sylvester
    What I want (this is probably hard for most to answer, as this is a very unique setup) is to have two different streams (this means audio splitter is not an option, as it will still only be one stream) of audio - one through the headphones and one through the main speakers. I can do the audio rerouting using virtual audio cables, however the problem is this: i cannot get both headphones AND speakers to play even just one stream, let alone two seperate ones. using "split front and back audio into seperate streams is not an option, as the actual MB F_PANEL is faulty (nothing to do with the case front panel, just so you know. that works fine) So, first things first. I need it to recognise the headphones as a seperate audio device so that Virtual Audio Cables will detect it and allow me to route the necessary audio to the headphones only. I also need to be able have sound play through speakers and headphones together what i want to achieve overall, is this: have the ENTIRE computer's sounds picked up by VAC, and stream them to Line1. then have Line1 stream to the headphones. that way whatever's being streamed is heard through the headphones, while the entire system sounds (including those not streamed) are played through speakers.

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  • What is really happening when we change encoding in a string?

    - by Jim Thio
    http://php.net/manual/en/function.mb-convert-encoding.php Say I do: $encoded = mb_convert_encoding ($original); That looks like simple enough. WHat I am imagining is the following $original has a pointer to the way the string is actually encoded. Something like char * kind of thing. And then there are things like what the character actually encoded. It's probably somewhere along UTF-64 kind of thing where each glyph is indeed a character. Now when we do $encoded = mb_convert_encoding ($original); several thing can happen: the original internal representation doesn't change however it is REINTERPRETED so that the code that show up differs the original string that it represent doesn't change however the ENCODING change. Which one is right?

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  • Why are the analoge stereo input and output of my M-Audio 24/96 soundcard not available to me in Ubu

    - by user37968
    I have installed Lucid on an old Mac PowerPC G4 desktop with a M-Audio Audiophile 24/96 soundcard. The only inputs and outputs I can select in the audio preferences are digital ones for the digital input and output. "lspci -v" shows the card as so: 0001:10:13.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) Subsystem: VIA Technologies Inc. Device d634 Flags: bus master, medium devsel, latency 16, IRQ 53 I/O ports at 0440 [size=32] I/O ports at 04b0 [size=16] I/O ports at 04a0 [size=16] I/O ports at 0400 [size=64] Capabilities: <access denied> Kernel driver in use: ICE1712 Kernel modules: snd-ice1712 "cat /proc/asound/cards" as so: 0 [Tumbler ]: PMac Tumbler - PowerMac Tumbler PowerMac Tumbler (Dev 21) Sub-frame 0 1 [M2496 ]: ICE1712 - M Audio Audiophile 24/96 M Audio Audiophile 24/96 at 0x440, irq 53 "aplay -L" shows these as listed: pulse Playback/recording through the PulseAudio sound server front:CARD=Tumbler,DEV=0 PowerMac Tumbler, PowerMac Tumbler Front speakers front:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi Front speakers surround40:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.0 Surround output to Front and Rear speakers surround41:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.1 Surround output to Front, Center, Rear and Subwoofer speakers iec958:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi IEC958 (S/PDIF) Digital Audio Output I believe it is a problem with detecting the analogue input/output. Sometimes I can get sound from the device but it is a sheet of white noise and tinkering makes it go away again I don't know if that is a separate problem or if it is linked to not being able to see the analogue input/outputs in the sound preferences. Any help would be greatly appreciated As for the white noise I have installed the Envy24 control panel and spend lots of time playing with the settings but when I can get the white noise I can never get it to an quality where I can actually hear what is being played. The internal speaker plays audio fine and plugging in a NI Audio 4DJ via usb also plays sound, although with some static but I believe that is due to an underpowered usb2 pci expansion card not being able to get enough electricity to the device. Alternatively I have seen other people with problems with this device so it may be a bug in the driver but that is another matter. I would like to get the M-Audio card working so I can begin to enjoy my music once again. As a note, I do not currently have any audio equipment capable of sending or receiving audio via the digital inputs and output so I can not check if they are working. The sound preferences show a wide range of digital in and out options with various surround sound options but no analogue ins and outs.

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  • Why are the analoge stereo input and output of my M-Audio 24/96 soundcard not available to me in Ubuntu Lucid

    - by MIDoubleKO
    I have installed Lucid on an old Mac PowerPC G4 desktop with a M-Audio Audiophile 24/96 soundcard. The only inputs and outputs I can select in the audio preferences are digital ones for the digital input and output. "lspci -v" shows the card as so: 0001:10:13.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) Subsystem: VIA Technologies Inc. Device d634 Flags: bus master, medium devsel, latency 16, IRQ 53 I/O ports at 0440 [size=32] I/O ports at 04b0 [size=16] I/O ports at 04a0 [size=16] I/O ports at 0400 [size=64] Capabilities: <access denied> Kernel driver in use: ICE1712 Kernel modules: snd-ice1712 "cat /proc/asound/cards" as so: 0 [Tumbler ]: PMac Tumbler - PowerMac Tumbler PowerMac Tumbler (Dev 21) Sub-frame 0 1 [M2496 ]: ICE1712 - M Audio Audiophile 24/96 M Audio Audiophile 24/96 at 0x440, irq 53 "aplay -L" shows these as listed: pulse Playback/recording through the PulseAudio sound server front:CARD=Tumbler,DEV=0 PowerMac Tumbler, PowerMac Tumbler Front speakers front:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi Front speakers surround40:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.0 Surround output to Front and Rear speakers surround41:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.1 Surround output to Front, Center, Rear and Subwoofer speakers iec958:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi IEC958 (S/PDIF) Digital Audio Output I believe it is a problem with detecting the analogue input/output. Sometimes I can get sound from the device but it is a sheet of white noise and tinkering makes it go away again I don't know if that is a separate problem or if it is linked to not being able to see the analogue input/outputs in the sound preferences. Any help would be greatly appreciated As for the white noise I have installed the Envy24 control panel and spend lots of time playing with the settings but when I can get the white noise I can never get it to an quality where I can actually hear what is being played. The internal speaker plays audio fine and plugging in a NI Audio 4DJ via usb also plays sound, although with some static but I believe that is due to an underpowered usb2 pci expansion card not being able to get enough electricity to the device. Alternatively I have seen other people with problems with this device so it may be a bug in the driver but that is another matter. I would like to get the M-Audio card working so I can begin to enjoy my music once again. As a note, I do not currently have any audio equipment capable of sending or receiving audio via the digital inputs and output so I can not check if they are working. The sound preferences show a wide range of digital in and out options with various surround sound options but no analogue ins and outs.

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  • Audio Static/Interference regardless of audio interface?

    - by Tom
    I currently am running a media center/server on a Lubuntu machine. The machine specs: Core 2 Duo Extreme EVGA SLI 680i MotherBoard 2 GB DDR2 Ram 3 Hard Drives no raid - WD Caviar Black, Green, and Samsung Spinpoint Galaxy GTX 220 1GB External USB Creative XI-FI Extreme Card 550W Power Supply This machine is hooked up through an optical cable to an ONKYO HTR340 Receiver through the XIFI card. Whenever I play any audio regardless if it is through XBMC, the default audio player, a flash video, etc, I get a horrible static sound that randomly gets louder. Here is a video of the sound: http://www.youtube.com/watch?v=SqKQkxYRVA4 This static comes in randomly, sometimes going away for short periods, but eventually always comes back. So far I have tried everything I could think of: Reinstalling OS Installing/upgrading/repairing PulseAudio/Alsa Installing alternate OSes, straight Ubuntu, Lubuntu, Xubuntu, Arch, Mint, Windows 7 Switching audio from the external card to internal Optical, audio out through HDMI, audio out through headphones Different ports on receiver (my main desktop sounds fine on the same sound system) Different optical cables Unplugging everything unnecessary from the motherboard (1 HD, 1 Stick of Ram, 1 Keyboard) Swapping out ram Swapping out the motherboard Replacing the Graphics Card (was replaced due to fan being noisy, not specifically for this problem) Different harddrives Swapping power supply Disabling onboard audio Pretty much everything short of swapping the CPU. I haven't been able to narrow down the problem and it is getting frustrating. Is it possible that the CPU is faulty and might cause a problem such as this, or that the PC case is shorting out the motherboard? Any kind of suggestions will be appreciated.

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  • No digital audio output with Asus Xonar DG

    - by Lunatik
    I've purchased an Asus Xonar DG as replacement for faulty onboard audio in a Medion 8822 as it has an optical output which is all I really need to feed my HTPC. I uninstalled the previous drivers/devices, switched the PC off, inserted the Asus card, powered up, disabled the onboard audio in the BIOS, then installed the driver that came on the CD (same version as on Asus' website as of today) and everything went perfectly - no errors. I set the audio devices up in Windows and in the Asus utility (SPDIF enabled, 6-ch audio) as I would expect to see them work, but the only thing is I have no digital audio from test tones within Windows/the Asus utility, PCM audio or Dolby Digital from DVD. Analogue audio is fine. I've uninstalled things and reinstalled a couple of times now, as well as trying almost all combinations of analogue/digital outputs but can't get it sorted. Does anyone have any tips on how to get this working? This card has just been released so there isn't much out there to go on. Notes: The light on the toslink port is lit. OS is Vista 32-bit SP2 and all up to date, pretty much a fresh install with almost no 3rd party applications installed This page seems to suggest that a digital output device in Windows is not needed with Xonar cards as it was with the previous Realtek so I have it set to Analog. The only other output device is S/PDIF pass-thru

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  • No digital audio output with Asus Xonar DG

    - by Lunatik
    I've purchased an Asus Xonar DG as replacement for faulty onboard audio in a Medion 8822 as it has an optical output which is all I really need to feed my HTPC. I uninstalled the previous drivers/devices, switched the PC off, inserted the Asus card, powered up, disabled the onboard audio in the BIOS, then installed the driver that came on the CD (same version as on Asus' website as of today) and everything went perfectly - no errors. I set the audio devices up in Windows and in the Asus utility (SPDIF enabled, 6-ch audio) as I would expect to see them work, but the only thing is I have no digital audio from test tones within Windows/the Asus utility, PCM audio or Dolby Digital from DVD. Analogue audio is fine. I've uninstalled things and reinstalled a couple of times now, as well as trying almost all combinations of analogue/digital outputs but can't get it sorted. Does anyone have any tips on how to get this working? This card has just been released so there isn't much out there to go on. Notes: The light on the toslink port is lit. OS is Vista 32-bit SP2 and all up to date, pretty much a fresh install with almost no 3rd party applications installed This page seems to suggest that a digital output device in Windows is not needed with Xonar cards as it was with the previous Realtek so I have it set to Analog. The only other output device is S/PDIF pass-thru

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  • Digital audio input on Macbook?

    - by Ken
    I have: a Macbook (not Pro), don't know the exact model but it's a Core 2 Duo 2.0GHz and probably what Wikipedia calls the "Late 2006" or "Mid 2007" model a DVD player, region-free, that has "Coax and TosLink optical digital audio outputs" I want to make an MP3 of the audio track of some DVDs (for learning a new language), and I can't use the Macbook's built-in DVD drive because it's a different region (ugh!). I'm sure I can connect the DVD player to the Macbook with an analog audio cable. However, if it's possible I'd prefer to keep the signal digital. I'm not even positive if my old Macbook has digital audio in, and if so what I need to connect to it. (I've done plenty of home audio geeking, but always in analog!) Will a "Toslink cable" plus a "Toslink Female to Mini-Plug Male Adapter" (found on Amazon) let me connect my things together? It looks like the pieces will fit but I'd like to hear someone confidently knowledgeable on the matter before I buy something. Thanks!

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  • Configure TV Capture card to not use external audio jack for TV audio output

    - by Adam D.
    I had this working with MythTV on Ubuntu 9.1. Then a power surge killed the motherboard. After replacing the motherboard, ram and cpu, the card does not produce any audio except through the output jack on the back of the card. I do not want to use a cable to go from the back of the card to the audio in on the built in sound card of the new mother board. FYI, the old motherboard did not have an on-board sound card. There was a separate audio card installed. There's some configuration that has to be done to have it work the same way again. I just have no idea where to start. This is regarding wintv hauppauge mythtv linux ubuntu 9.10 audio

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  • Extracting and re-combining audio from a video

    - by waitinforatrain
    I'm fed up with Vegas, Premiere, Pinnacle, SoundForge and all of the other editors I've tried. I'm trying to modify the audio of an AVI file, but all of the editors I've tried insist on re-encoding the video when I save the changes, and either reduce the quality or take hours. Is there a way to just extract the audio stream from a video with ffmpeg, edit it in an audio editor and then recombine the saved changes with the video?

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  • Switch encoding of terminal with a command

    - by Tomas Lycken
    One of the servers I quite often ssh to uses western encoding instead of utf-8 (and there's no way I can change that). I've started writing a bash script to connect to this server, so I won't have to type out the entire address every time, but I would like to improve this script so it also changes the encoding of the terminal window correctly. The change I need to do can be performed using the mouse by navigating to "Terminal"-"Set Character Encoding..."-"Western (ISO-8859-1)". Is there a terminal command that does the same thing, for the current terminal window/screen? To clarify: I'm not interested in ways of switching the locale of the system on the remote site - that system is administered by someone else, and I have no idea what stuff might depend on the latin-1 encoding there. What I want to do is to let this terminal window on my side switch character encoding to the above mentioned, in the same way I can do with my mouse and the menus.

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  • USB Audio device not selectable in settings after Camtasia Studio

    - by DJDavid98
    I tried to use Camtasia Studio 7 Recorder today, and when I selected the "Record Microphone and System audio" option, the recorder crashed, and then my sound settings icon disappeared. If I tried to open the Camtasia Recorder's Options or tried to select the option again, the program crashed. After holding the power button to force restart (so My settings wouldn't save, theoretically) the audio device was still not showing up in the sound settings, although it still is displayed under the "Hardware" tab, and is Functional. Or at least that's what windows thinks. I tried uninstalling Camtasia, and no sucess either. The USB audio device is labeled "C-Media USB Headphone Set". OS: Windows XP SP3 hu-HU Is there any way I could make it work again, like registry editing or something?

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