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  • HDMI audio disappears when the TV is turned off and on

    - by Devator
    I have a new HTPC build (it's running Windows 7 Ultimate x64) and it has a ATI Radeon 6450 as graphics card. When I boot the PC, everything works fine. However, when the TV is turned off and then back on I notice there's no sound anymore. There are lots of other people who have this issue aswell. I've found this workaround however it doesn't seem to work for me. I have the newest drivers (12.4). Someone said it was fixed in 9.1 but I still have the issue. Any help is appreciated.

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  • Distortion problem with Creative audio equalizer

    - by e-t172
    Hi, I have a problem with the Creative Console EQ, I don't know if it's fixable or not (is the EQ software or hardware on these cards?). Basically, I have enormous distortion with certain sounds in the 30 - 125 hz range. When this happens I get some sort of "frrzzzz" (sorry, I'm french and don't really know the correct english word for that) on top of the original sound. I have a Sound Blaster Audigy SE. I'm using the Daniel_K drivers, on Windows 7 Profesionnal x64. All the effects are disabled except EQ. Steps to reproduce Put the card in 24bit/96khz mode. The problem is also present with 16bit/48khz but seems to be less audible. In the Creative Console, use the following EQ: (full size) Play this sound at a reasonably high volume. You should hear distortion on the two "booms". Especially the second one. Disable Creative EQ. Play the sound in an application with an integrated EQ (e.g. foobar2000, ffdshow) using the same EQ parameters. There is no distortion. Conclusion: the Creative EQ is broken. Is anyone having the same problem? I'm also interested in the results with other Creative cards or even other brands soundcards with a similar EQ feature.

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  • Kubuntu solution for analysing/recording streamed flash audio?

    - by marcusw
    I have a Kubuntu system which I stream amateur radio sound to via a flash interface. I want to be able to record the sound that the flash player is making at the press of a button. I also need the capability to do (hopefully real-time) spectrum analysis on the sound. I need a program (a firefox add-on would be ideal) that can do this for me.

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  • Making audio CDs en mass - Linux based solutions?

    - by The Journeyman geek
    My mom's sings and gives away cds to people. Invariably it falls to me to have to burn cds for her, and burning 50-100 cds on a single drive is a pain. I DO have a handful of cd burners and a slightly geriatric old PIII 450. This is what i want to be able to do - either point an application at a folder of WAV or MP3s, say how many copies i need on CLI (since then i can SSH into the system and use it headless) feed 2 or more CD burners cds until its done, OR pop in a single CD into a master drive and have its contents duplicated to 2 or more burners. I'd rather have it running on linux, be command line based, and be as little work as possible - almost automatic short of telling it how many copies i want would be ideal. I'm sure i'll have people wondering about legality - My mom sings her own music, and its classical, and older than copyright law, so, that's a non issue. I just want a way to make this chore a little easier, short of telling my mom to do it herself.

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  • Converting from samplerate/cutoff frequency to pi-radians/sample in a discrete time sampled IIR filter system.

    - by Fake Name
    I am working on doing some digital filter work using Python and Numpy/Scipy. I'm using scipy.signal.iirdesign to generate my filter coefficents, but it requires the filter passband coefficents in a format I am not familiar with wp, ws : float Passband and stopband edge frequencies, normalized from 0 to 1 (1 corresponds to pi radians / sample). For example: Lowpass: wp = 0.2, ws = 0.3 Highpass: wp = 0.3, ws = 0.2 (from here) I'm not familiar with digital filters (I'm coming from a hardware design background). In an analog context, I would determine the desired slope and the 3db down point, and calculate component values from that. In this context, how do I take a known sample rate, a desired corner frequency, and a desired rolloff, and calculate the wp, ws values from that? (This might be more appropriate for math.stackexchange. I'm not sure)

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  • periodically unable to play media

    - by avorum
    So I don't know if this is right place to ask this at all but I've gotten good help here before so I thought I'd ask. For the last year or so periodically my computer would start refusing to play media. In browser players would say they were playing but they weren't. No audio and the video wasn't moving forward although it would show the first frame of the video to be shown. iTunes would act similarly, thinking it was playing without actually playing any music. This persists across browsers, various application categories, etcetera. It can often be fixed by rebooting but it is only a short term solution. Does anyone know of anything that might cause this erratic behavior? I'm using Windows 7 64bit. If additional information would help please ask. Alternatively, if this isn't the right site for this I would greatly appreciate some direction to a site better suited to my question. Thanks in advance for any help.

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  • Static noise in headphones

    - by John Murdoch
    I have a Asus P6T based system. I was using the on-board sound (plugging in Logitech X-230 2.1 analog speakers in the green "front speakers" 3.5mm analog output, then plugging in my headphones in that). I was quite happy with the sound quality (didn't hear any static noise if volume was turned down to my normal listening level). Then about a week ago I started having terrible static noise from the left channel, and no normal audio on that left channel. Right channel had more static noise than usual but did have a bit of sound. I tried using the AC'97 in front of my case but that seemed to have no signal. I decided my on-board sound card has gone bad and bought an internal sound card to replace it (Startech 7.1Ch PCI). This fixed the "no sound from left channel problem", but I had much more audible static noise. I decided the card was low quality and/or it had interference from all the other things happening inside the computer case, and bought a Sweex SC016 external USB sound card. But even with that I have static noise in headphones. Positioning the USB sound card differently doesn't seem to help. Trying the other analog outputs (e.g., surround) doesn't help. The static noise in all cases is proportional to the volume. I have tried different headphones, but the situation is situation though perhaps the flavour of the static noise changes slightly. So what are my options? a) Get another, more expensive, external USB sound card hoping the quality will improve? b) Get another, more expensive, internal sound card (PCIe 1x perhaps) hoping the quality will improve? c) Get a dedicated DAC box? d) Get some Hi-Fi earphones? Suggestions? tl;dr - Two different sound cards both still have static noise in headphones.

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  • How can I remotely display images on my computer?

    - by Jakob
    What I Have: A laptop booted with Ubuntu and a stationary computer dual-booted with Ubuntu and Vista, both connected through a wireless ad-hoc network. What I Want: I want a way to display images in fullscreen on my stationary, using my laptop as a "remote control". I want to be able to choose another picture at any time and have my stationary computer remain in fullscreen mode at all times. Preferably, I should also be able to display just an empty (black) screen. How can I arrange for this? What I Have Tried: I have tried simply SSH:ing into my stationary computer and opening the image files using an image viewer, but all of the ones that I have tried (Eye of Gnome, Mirage, Gwenview, and others) open a new window for every new image. I don't know how to force them into using a single instance. I have tried using the VLC remote control command line interface, but apart from seeming somewhat unreliable (exiting with segmentation faults at one point), it also displays some images with a green border and forces me to pause playback in order for the image to remain on screen. Bonus Question: In my final setup, I also need to play music through my stationary computer's speaker and have the ability to switch to another track at any point, like with the images. Preferably, I would like to control the images and the audio through the same interface. How can I best achieve this?

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  • Rendering a frame is producing noise from speakers in Windows and Linux

    - by Robber
    When any hardware accelerated application is rendering a frame (or many of them) a very short noise is coming from my speakers. This can be a game, a WebGL application or XBMC. When the application/game is rendering many frames per second (like most of them do) the noise is a continuous buzzing that gets higher pitched with higher framerates. This applies to Linux and Windows, so I'd assume it's a hardware problem. The current hardware in the PC is: CPU: Core2Quad Q9550 GPU: Radeon HD 5770 RAM: 2x2GB DDR2 Motherboard: Asus P5QLD PRO PSU: be quiet! Pure Power 530W Screen and speakers: Old 720p LCD TV connected via VGA and aux cable Muting the TV stops the noise, muting Windows doesn't. I tried replacing the PSU first (used a Tagan 700W PSU before) because I thought it was a power problem. It wasn't. I tried replacing the motherboard (used a ASUS P5B SE before) next because I thought it was a sound card problem. It wasn't. I tried the GPU in a different PC because I thought it was a broken graphics card. It worked perfectly fine in the other PC. I thought it might be interference, but moving the audio cable around changes absolutely nothing. I tried using an HDMI cable instead and that did work, but is not an option since my TV has only one HDMI input and I need that for my PS3.

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  • Youtube has no voice but the music continues just fine?

    - by Prix
    PC CONFIG: gigabyte EP45C UD3R with the realtek HD onboard 4gb dual channel Qcore 2.83ghz When i watch to videos on youtube now the voice some times is in static and some times so low that you can hear it while the sound continues just fine... For example if can hear to things like guitar or a train etc but the voice of whoever is speaking is gone or very low or pure static when watching the videos. I know some videos have a really great quality and some are HD 1080p so this was something not expected to happen. I can aswell play videos on my WMP11 just fine i have ccc-p installed also tried k-lite, both on the latest stable avaiable. I havent tried anything else related to flash but something is either wrong with my drivers or youtube. I have installed the latest drivers to make sure they are up-to-date but this didnt help either. What i have tried so far: removed the audio drivers and re-installed remove any codec pack i had and re-installed k-lite, test, didnt worked remove any codec pack i had and re-installed cccp, test, didnt worked checked the control panel sound configurations, tried chaging to phone stereo, to 5.1 which is what my headphone is. checked the realtek manager, tried changing the sound channels from 2CH to 6CH to reflect my headphone, didnt work. rebooted after every change of the above tries. tried chrome, firefox and internet explorer with the same results didnt w

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  • Google Chrome on Linux not using ALSA for sound

    - by DarkMoon
    On my laptop, I've got an .asoundrc file that outputs sound to my USB headset. This works fine for SMplayer and Firefox. However, Google Chrome (at least, Flash-based and HTML5-based videos and HTML5-based audio in Chrome) plays through the laptop speakers instead. I've tried running Chrome from a command-line, hoping there would be some helpful output, but no such luck. I've tried looking through Google for whether Chrome even uses ALSA, or if it uses something else, but I have been unsuccessful in this. This question seems to be the same issue, but no suggestion was made. Anyone have any ideas? I'm running Gentoo with a 3.10.17 kernel, 1.0.27 ALSA utils, 2.6.5 FVWM, and 36.0.1985.143 Chrome. If you need more info, please let me know. EDIT: I've configured the USB headset as the default ALSA device. Volume levels for both headset and onboard are set and un-muted using alsamixer. My .asoundrc file is as follows. ctl.!default { type hw card Headset } pcm.dmixer { type dmix ipc_key 1024 slave { pcm { type hw card Headset } period_size 1024 buffer_size 4096 } bindings { 0 0 1 1 } } pcm.!default { type plug slave.pcm dmixer }

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  • Windows 8 Communication Sound Setting not working

    - by blackmastiff
    I've been having a problem on my new laptop recently which is familiar but baffling the usual fixes. I'm running Windows 8 with an onboard Realtek soundcard. It's similar to the one on my older computer running Windows 7. The problem is, when I'm in Skype or Mumble, Windows changes the sound output to lower everything else automatically. I've disabled the communications sound change option on the communications tab within sound devices and checked all the applications settings to insure that they are not responsible. They aren't, and I noticed something else. When I'm in the sound properties dialog, and I switch to the microphone tab, the same audio output reduction occurs. This seems to say to me that the microphone must be responsible in some way, but seeing as I uninstalled all the drivers and installed windows drivers instead, I'm confused as to why this would be occurring. Any thoughts? EDIT: I just tried disabling the built in microphone and the sound no longer get changed. More confused now? As soon as I turn it back on, the sound gets dropped again. Incidentally, the fix for this on windows 7 was this question: Windows 7 lowers applications' volume automatically I've got my computer set that way and it doesn't work.

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  • Using sound forge 6.0 what will be the need to upgrade to latest version

    - by Jayapal Chandran
    I had been using sound forge 6.0 not recently but long back. I edit mp3 files for my purpose and some more filters like flange, pan, fade in out, recording, line in recording, extracting sound from video files (mpg, avi(divx), etc...), increasing the default volume, editing treble and bass effects, and etc... I am not going to use it professionally. I use it just like that. Now when i checked i could see Sound Forge Audio Studio 10 is the latest version for my purpose. Others are too high i think. Besides, i had been using Gold Wave version 4 very extensively just to edit sound files mostly mp3. and here is the reason for me to change to sound forge. It is when we edit mp3 files it deflashes(making it raw i think) before editing. after editing if i save it asks for the format to save and i will choose mp3. At this point it again applies the compression process which makes the sound file lossy. When i did the same with sound forge it did not deflash. It just edited the file as mp3. May be i dont know whether gold wave has the same option. So, please suggest. oh i had asked a question already like this... here it is goldwave vs sound forge in editing mp3 files

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  • Is acousting fingerprinting too broad for one audio file only?

    - by IBG
    So we were looking for some topics related to audio analysis and found acoustic fingerprinting. As it is, it seems like the most famous application for it is for identification of music. Enter our manager, who requested us to research and possible find an algorithm or existing code that we can use for this very simple approach (like it's easy, source codes don't show up like mushrooms): Always-on app for listening Compare the audio patterns to a single audio file (assume sound is a simple beep) If beep is detected, send notification to server With a flow this simple, do you think acousting fingerprinting is a broad approach to use? Should we stop and take another approach? Where to best start? We haven't started anything yet (on the development side) on this regard, so I want to get other opinion if this is pursuit is worth it or moot.

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  • Thinkpad speaker turns mute - Linux Codec issue?

    - by Curlew
    At some point a few days ago the speakers on my Lenovo Thinkpad T410 (Model number: 2537A11) suddenly stopped working randomly. This error happens every time I watch a video or listen to a music file. The sound just abruptly stops. At the moment, I can't produce a single sound no matter what I do. I am using Debian GNU/Linux on this laptop and there doesn't appear to be anything else wrong (the fan is working, no abnormal heat (staying around ~40°C), no other obvious errors or problems). Here is the output of a nice program someone pointed me to: martin@martin:~/Downloads$ sudo python run.py --monitor Using temporary directory: /dev/shm/hda-analyzer You may remove this directory when finished or if you like to download the most recent copy of hda-analyzer tool. Downloading file hda_analyzer.py Downloading file hda_guilib.py Downloading file hda_codec.py Downloading file hda_proc.py Downloading file hda_graph.py Downloading file hda_mixer.py Downloaded all files, executing hda_analyzer.py Watching 1 cards ====================================== Sound is working normally and then it stops and the following lines appear: Diff for codec 0/0 (0x14f15069): --- +++ @@ -164,17 +164,17 @@ Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000010: OUT Pin Default 0x901701f0: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT - Power: setting=D0, actual=D0 + Power: setting=D3, actual=D3 Connection: 2 0x10* 0x11 Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE And now there is also an error in the dmesg output hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. I changed the bdl_pos_adj to various numbers (-1, 0, 64, 1024) and either there is no change at all or dmesg reports that the adjustment is too big. I wonder if this bdl_pos_adj is the real reason for the error. Here is my hardware information provided by alsa-info.sh website. Okay, i did some serious testing and even installed Windows and now i officially conclude that this is a hard-ware related issue with my Laptop speakers. Reason: The error occurs in my installed Debian Linux, an Ubuntu Live distribution and Windows XP No error-message appears in all of the OS. The sound just keeps running and i can't hear a thing. I tested different setups, including OSS, ALSA and the pulseaudio server on top If i use my new usb-headphones i can hear sound all the time without any sudden silences. So obviously, although hard to believe, my laptop speakers are not okay (never heard of similar cases). I'll award the bounty to anyone who can point me to good tutorials or the procedure how to exchange my T410 speakers (i still have warranty. The laptop was bought in Germany, but now i am in Denmark). Or to someone who can explain me the output from hda-analyzer (big log above).

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  • Is acoustic fingerprinting too broad for one audio file only?

    - by IBG
    We were looking for some topics related to audio analysis and found acoustic fingerprinting. As it is, it seems like the most famous application for it is for identification of music. Enter our manager, who requested us to research and possible find an algorithm or existing code that we can use for this very simple approach (like it's easy, source codes don't show up like mushrooms): Always-on app for listening Compare the audio patterns to a single audio file (assume sound is a simple beep) If beep is detected, send notification to server With a flow this simple, do you think acoustic fingerprinting is a broad approach to use? Should we stop and take another approach? Where to best start? We haven't started anything yet (on the development side) on this regard, so I want to get other opinion if this is pursuit is worth it or moot.

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  • Opus : le nouveau codec audio open-source est standardisé, il ferait mieux que six codecs propriétaires réunis dixit Mozilla

    Opus : le nouveau codec audio open-source est standardisé Il couvre les usages de six codecs propriétaires et le ferait mieux dixit Mozilla Une victoire historique. Pour Mozilla, la standardisation du codec audio open-source Opus est un évènement de cette envergure. La Fondation y voit « le début de la fin des formats propriétaires [dans l'audio] ». Ce projet de standardisation a été mené à bien grâce à une collaboration entre le monde open-source (dont est issue la Fondation) et des entreprises privées dont Microsoft (au travers de Skype) ou Google. Cette standardisation devrait permettre à Opus de mieux s'imposer que ses prédécesseurs (comme Vorbis)...

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  • Dancer.js : une API audio open source de haut niveau en JavaScript pour lier animations visuelles et musique

    Créez de belles animations visuelles sur vos musiques préférées grâce à ce framework javascript Vous avez forcément, à un moment ou à un autre, utilisé cette fonctionnalité sur votre lecteur de musique préféré. Je parle de ces animations graphiques à base de lignes colorées, de bulles qui éclatent et bien d'autres formes au rythme de votre chanson favorite. Et bien il est maintenant possible d'intégrer de telles animations dans votre site web grâce à Dancer.js ! Cette API est utilisable avec l'API Audio Data de Mozilla ainsi qu'avec l'API Web Audio de Webkit et flash fallback. Dancer.js utilise les fréquences audio en temps réel pour les lier à des effe...

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  • Recovering damaged external hard disk by installing internally

    - by nfarshchi
    I had a 1TB Western Digital (My book series) 3.5" USB3. One day, the SATA to USB3 converter board was damaged and has not worked since. I decided to open the cover and use the HDD as an internal HDD. When I attached the HDD to my PC and booted up in Windows, it asked me which type of ????? I want to use "MBR or GBR" (I dont remember the exact question) I chose MBR and Windows gave me a 1TB empty Hard drive. I tried to recover with recover my files and some other recovery programs but no success. Some one told me that you should choosed GBR instead of MBR . How can I do that now? Another guy told me that the SATA to USB3 converter board is coded to save data on HDD and you can not use them internally without losing data, and I should find another SATA to USB3 board (exact same). It is impossible to find because they are not produced any more. Please help me to find a solution to bring back my data. UPDATE I have 1TB WD "Mybook" USB 3. the board that convert sata to usb3 was damaged. so when the HDD was in the box computer did not recognize it. I opened the box and remove HDD to use it internal. after connecting to my PC windows showed me one massage that I had two choice MBR or GPT I choosed MBR one and windows gave me 1TB empty new volume. I tried many recovery software to recover my data but no success. I brought it to one expert recovery company and they told me the converter board (SATA to USB3) make some encryption on data and with out that board you cannot recover any thing. so I bought another empty WD box and put the HDD inside but even after that also there is no file. I tried to recover again in this state but no success. so I have some unanswered question. does this converted boards make any password or encryption? if yes how can I solve it? does using many recovery programs affected my data? any suggestion or solution for bring back my data? I had use recovery programs such as : recover my files , EaseUS data recovery, easy recovery, test disk, Ontrack easy recovery . Note: when I was using test disk it asked me to choose which partition table I want to use. as it was I choose NTFS, does this made any change on data?

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  • Supported Audio Formats of Qt4 Phonon?

    - by Nikwin
    I am making a music player in PyQt4, and I am using Phonon to play the music itself. This application is aimed primarily at Windows, but I plan on also supporting Mac and Linux versions. What I want to know is which audio formats are supported by Phonon so that I can ensure that the user only enters those files.

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  • FMOD on non-playing audio

    - by coldrising
    Hey, is there any way to get the audio spectrum of a section of a song using FMOD if it is not playing? Can I render a full song waveform using FMOD (+opengl/openframeworks/etc.) before the song is playing?

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  • AVAudioPlayer won't play audio file after AVAudioRecorder

    - by Kevin
    I create a .caf audio file using AVAudioRecorder and if I try and play it back using AVAudioPlay I get no sound on the iPhone (if played in simulator works fine). If I close my application and reopen the file plays fine. Also I am not able to adjust the phone volume after recording unless I close and reopen my application. Any ideas?

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  • List input and output audio devices in Applet

    - by Jhonny Everson
    I am running a signed applet that needs to provide the ability for the user to select the input and output audio devices ( similar to what skype provides). I borrowed the following code from other thread: import javax.sound.sampled.*; public class SoundAudit { public static void main(String[] args) { try { System.out.println("OS: "+System.getProperty("os.name")+" "+ System.getProperty("os.version")+"/"+ System.getProperty("os.arch")+"\nJava: "+ System.getProperty("java.version")+" ("+ System.getProperty("java.vendor")+")\n"); for (Mixer.Info thisMixerInfo : AudioSystem.getMixerInfo()) { System.out.println("Mixer: "+thisMixerInfo.getDescription()+ " ["+thisMixerInfo.getName()+"]"); Mixer thisMixer = AudioSystem.getMixer(thisMixerInfo); for (Line.Info thisLineInfo:thisMixer.getSourceLineInfo()) { if (thisLineInfo.getLineClass().getName().equals( "javax.sound.sampled.Port")) { Line thisLine = thisMixer.getLine(thisLineInfo); thisLine.open(); System.out.println(" Source Port: " +thisLineInfo.toString()); for (Control thisControl : thisLine.getControls()) { System.out.println(AnalyzeControl(thisControl));} thisLine.close();}} for (Line.Info thisLineInfo:thisMixer.getTargetLineInfo()) { if (thisLineInfo.getLineClass().getName().equals( "javax.sound.sampled.Port")) { Line thisLine = thisMixer.getLine(thisLineInfo); thisLine.open(); System.out.println(" Target Port: " +thisLineInfo.toString()); for (Control thisControl : thisLine.getControls()) { System.out.println(AnalyzeControl(thisControl));} thisLine.close();}}} } catch (Exception e) {e.printStackTrace();}} public static String AnalyzeControl(Control thisControl) { String type = thisControl.getType().toString(); if (thisControl instanceof BooleanControl) { return " Control: "+type+" (boolean)"; } if (thisControl instanceof CompoundControl) { System.out.println(" Control: "+type+ " (compound - values below)"); String toReturn = ""; for (Control children: ((CompoundControl)thisControl).getMemberControls()) { toReturn+=" "+AnalyzeControl(children)+"\n";} return toReturn.substring(0, toReturn.length()-1);} if (thisControl instanceof EnumControl) { return " Control:"+type+" (enum: "+thisControl.toString()+")";} if (thisControl instanceof FloatControl) { return " Control: "+type+" (float: from "+ ((FloatControl) thisControl).getMinimum()+" to "+ ((FloatControl) thisControl).getMaximum()+")";} return " Control: unknown type";} } But what I get: Mixer: Software mixer and synthesizer [Java Sound Audio Engine] Mixer: No details available [Microphone (Pink Front)] I was expecting the get the real list of my devices (My preferences panels shows 3 output devices and 1 Microphone). I am running on Mac OS X 10.6.7. Is there other way to get that info from Java?

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  • Play any audio for given time

    - by Dipen
    I want to play any file for 6 seconds. Also suppose the audio is bigger then 6 sec the application will play only for 6 sec.and if it is less then 6 sec then play continuously. So is there any inbuilt option from any framework?

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