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  • Flash player, HD videos and games are choppy

    - by Aimad Majdou
    I have a problem with flash player. HD videos from Youtube or Vimeo and flash games do not play smoothly. I'm using Flash player 11, Windows 7 Sp1, and my graphic card is Intel GMA 4500. Device Manager shows me that all drivers are installed on my computer, so i don't have any problems with drivers. When I run Google chrome, Resource Monitor shows me 15% ~ 40 % of CPU Usage and 40% used Physical Memory, but when I watch a video on Youtube or play a Flash game, the Resource Monitor shows 70% - 90% CPU Usage. Also, when I run some HD Video (Frame width : 1920, Frame height : 1080) on my computer, Device Manager shows me 80% ~ 100% of CPU Usage. before I Reinstall Windows 7, HD Videos and flash games were play smoothly I hadn't any problem with them !! I hope all these informations are enough to answer my question.

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  • Empty track in Windows Media Player 12 that can't be deleted

    - by David Brown
    Right after I installed a fresh copy of Windows 7, I synced with my DropBox account that contains all of my music and added the directory to Windows Media Player 12. I now have a strange track that really isn't a track at all. It's grouped under "Unknown Artist" and has absolutely no text. The only reason I know it's there is because it highlights on mouse-over. Double-clicking on it does nothing. When the song before it ends, Windows Media Player stops playing altogether until I choose a different song (it should continue to play the next song on its own). When I try to delete this mysterious track, nothing happens. I've cleared my library and re-imported everything, but this empty track keeps appearing. I have also checked my Music directory and there is no empty MP3. What is going on here?

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  • In search of a good audio player for Ubuntu 9.10

    - by Joe Casadonte
    If this should be marked Community Wiki, please let me know. I'm switching from XP to Ubuntu, and I have been very disappointed with the selection of media players available. I'm primarily interested in an audio player, but integrated video and library management is OK, too. My criteria: Must be able to play audio CDs (I'm shocked how many apps this does away with, right away) Must be able to play MP3 & WAV; OGG, SHN, FLAC are all bonuses Repeat and Shuffle modes are a must FreeDB / GraceNote through a proxy is a must (if it can read a PAC file, that would be awesome) It needs to be really small, e.g. skinnable or an applet Ability to execute a playlist is a plus Gapless MP3 playback a plus I'm running Gnome, but I'm not totally adverse to a KDE app. Command-line only is also a viable option. Some that I've tried: RhythmBox - probably the best of the lot that I've tried; I don't like its mini mode (doesn't show the song being played) and I can't figure out how to get it to hit FreeDB/GraceNote through a proxy Songbird - can't play CDs, playlist management is atrocious Banshee Jajuk Maybe a couple of more. Thanks! UPDATE I tried out VLC, Amarok and Songbord (again). VLC I eventually got to work (I had some kind of bad configuration). It seemed way more involved than I was looking for out of a music player, and in general more geared to video than audio. I couldn't fathom its library management, which I think it has; maybe it doesn't, and that's why I couldn't figure it out. Amaork looked very promising but the library management was not to my liking, and the way it handled a playlist with both MP3 and WAV is inexplicable at best. I did like some aspects of the UI, but not enough to keep it. Songbird is very finicky, but I like the library management. Sort of. It kept telling me my Watch folder was invalid, even thought it clearly was accessible. Playlist management is bizarre, and the message that it was deleting source files whenever I deleted a playlist had me too worried to keep using it. Had it been able to play CDs, maybe I would have persevered. Audacious, while a bit odd at times, does seem to do what I want. If it had a library manager, I wouldn't have bothered trying any of the others. Thanks for the help, everyone!

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  • Configure audio on HP ENVY 4 ultrabook

    - by phodu_insaan
    I want to configure audio for ubuntu 12.04 on my laptop. Currently the audio just does not play. If i try and plug in headphones then somewhere midway to being fully plugged in the audio plays on the headphones, I plug in further and the sound disappears. How do I get this to work? lspci | grep audio Audio device: Intel Corporation Panther Point High Definition Audio Controller My laptop is one the beats edition HP laptops, and the driver for win7 was an IDT HD audio driver. --EDIT-- The output for cat /proc/asound/card0/codec* | grep Codec is Codec: IDT 92HD91BXX Codec: Intel PantherPoint HDMI I need to get both the IDT card to work and the HDMI card to work with my TV. --EDIT-- --EDIT 2-- I have added blacklist snd-usb-audio to the end of the file /etc/modprobe.d/alsa-base.conf Now the sound plays from my laptop speakers only when I plug in a headphones/external speaker. Otherwise no sound. :( Please help getting everything working as it should. --EDIT 2-- Thanks

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  • Accessing VMware Player across the network

    - by Tiffany Walker
    My network: 192.168.2.2 (computer) 255.255.255.0 (mask) 192.168.2.254 (gateway) The VMplayer is: 192.168.233.129 (ip) 255.255.255.0 (mask) 192.168.233.255(bcast) It's obvious the player can't be access from other systems because of the wrong network settings. However, I am not sure as to how to change that. Obviously I have to edit the ifcg-eth0 but I know I also need to edit it on the host system to make it accessably. Is this done from inside VMware Player? Or do I need to edit the network devices? The host box is Windows. Inside VMware is Linux.

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  • VMWare Player Run as Service

    - by Marshal
    I have a VMWare Player running a VM at all times. However I want the server to run that Virtual Machine on boot (without logging in). This will allow me basically to restart the Server and when it loads to the login screen it's already running VMWare. Any ideas? I have tried reading up on it however websites suggest VMWare Server. This is no longer backed by VMWare anymore. So I am looking to either make VMWare Player a service or if anyone else has any other ideas? Running Win 7 on my server btw Thanks for looking.

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  • VMWare Player 5.0 Upgrade - Bridged Networking No Longer Works

    - by Syndog
    I've been using VMWare Player 4 on my Windows 7 PC, running my VMs under a bridged networking configuration without any problem. I settled on bridged networking two reasons... I often use VMs to host web applications for test purposes. I never could get NAT to work for me anyway, but no love lost, considering reason #1. Now that I've upgraded to VMWare Player 5, the situation has completely reversed. In other words, bridged networking no longer works at all, and NAT does. Has anyone else come across this? Any ideas on getting bridged networking to work again? This happens with both Windows and Linux guest OSes. VMWare Tools has been fully updated. I've tried the usual vmnetcfg.exe utility fix, to no avail.

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  • About AMR audio file playing issue on different devices

    - by user352537
    I have got a quite strange problem here. I am developing an IM software and need to play audio files recorded by another client on Android. The same audio file I've got can be played with AVAudioPlayer on 3GS(IOS 4.2.1) device and simulator 4.2. But when I tried by play it on iPhone4(iOS 4.3.3), the function "play" always return NO. I also tried with two iPhone devices, the audio files recorded by iPhone client can be played on both 3GS and iPhone4. So I asked the Android developers about the record parameters they've used. They said that the "AudioEncoder" used by them was "DEFAULT". There are also some other parameters as following: **private AudioEncoder() {} public static final int DEFAULT = 0; /** AMR (Narrowband) audio codec */ public static final int AMR_NB = 1; /** @hide AMR (Wideband) audio codec */ public static final int AMR_WB = 2; /** @hide AAC audio codec */ public static final int AAC = 3; /** @hide enhanced AAC audio codec */ public static final int AAC_PLUS = 4; /** @hide enhanced AAC plus audio codec */ public static final int EAAC_PLUS = 5;** Does anybody know what's the matter?

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  • iPhone SDK Smaller CAF files: lower recording quality with Audio Queues?

    - by Nick
    In my iPhone app I have voice recording functionality the utilizes Audio Queue voice recording functions of the SDK. I'm saving directly to CAF format and using the following settings for the AudioStreamBasicDescription reference: audioFormat.mFormatID = kAudioFormatLinearPCM; I can see that there are other format ids I could use like: kAudioFormatLinearPCM kAudioFormatAppleLossless kAudioFormatAppleIMA4 kAudioFormatiLBC kAudioFormatULaw kAudioFormatALaw My knowledge of sound formats is very limited so my question is... which of these should I use to create the lowest compressed audio recording files? Plus, are there other settings I should apply to lower the quality and filesize even further?

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  • How to program a real-time accurate audio sequencer on the iphone?

    - by Walchy
    Hi... I want to program a simple audio sequencer on the iphone but I can't get accurate timing. The last days I tried all possible audio techniques on the iphone, starting from AudioServicesPlaySystemSound and AVAudioPlayer and OpenAL to AudioQueues. In my last attempt I tried the CocosDenshion sound engine which uses openAL and allows to load sounds into multiple buffers and then play them whenever needed. Here is the basic code: init: int channelGroups[1]; channelGroups[0] = 8; soundEngine = [[CDSoundEngine alloc] init:channelGroups channelGroupTotal:1]; int i=0; for(NSString *soundName in [NSArray arrayWithObjects:@"base1", @"snare1", @"hihat1", @"dit", @"snare", nil]) { [soundEngine loadBuffer:i fileName:soundName fileType:@"wav"]; i++; } [NSTimer scheduledTimerWithTimeInterval:0.14 target:self selector:@selector(drumLoop:) userInfo:nil repeats:YES]; In the initialisation I create the sound engine, load some sounds to different buffers and then establish the sequencer loop with NSTimer. audio loop: - (void)drumLoop:(NSTimer *)timer { for(int track=0; track<4; track++) { unsigned char note=pattern[track][step]; if(note) [soundEngine playSound:note-1 channelGroupId:0 pitch:1.0f pan:.5 gain:1.0 loop:NO]; } if(++step>=16) step=0; } Thats it and it works as it should BUT the timing is shaky and instable. As soon as something else happens (i.g. drawing in a view) it goes out of sync. As I understand the sound engine and openAL the buffers are loaded (in the init code) and then are ready to start immediately with alSourcePlay(source); - so the problem may be with NSTimer? Now there are dozens of sound sequencer apps in the appstore and they have accurate timing. I.g. "idrum" has a perfect stable beat even in 180 bpm when zooming and drawing is done. So there must be a solution. Does anybody has any idea? Thanks for any help in advance! Best regards, Walchy

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  • Best practice for C++ audio capture API under Linux?

    - by braddock
    I need to create a C++ application with a simple audio recording from microphone functionality. I can't say that there aren't enough audio APIs to do this! Pulse, ALSA, /dev/dsp, OpenAL, etc. My question is what is the current "Best practice" API? Pulse seems supported by most modern distros, but seems almost devoid of documentation. Will OpenAL be supported across different distros, or is it too obscure? Have I missed any? Is there not a simple answer? thanks!

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  • Why does my audio keep reverting to stereo?

    - by GraemeF
    I have an ASUS P7P55D LE motherboard with onboard sound and I am running Windows 7 64-bit, and I am using the SPDIF output from my motherboard to my receiver. On my receiver I can see which audio channels are in use (i.e. stereo or 5.1) In the audio interface properties I can test DTS Audio and Dolby Digital outputs and both work fine, but when I try to play a game with 5.1 sound (I've tried Left 4 Dead and Dragon Age Origins) it reverts to stereo. I was getting this behaviour with the default Microsoft drivers so I installed the latest from the ASUS website but there is no difference. I notice that in the Advanced tab the Default Format only allows me to choose from various 2 channel formats so maybe that is something to do with it? How can I get 5.1 output all of the time?

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  • Windows 7 Audio problem..

    - by Marcx
    hi, I've a dell studio 1555 bought on september with Windows 7 64bit Professional on it. Audio devide works proprerly, while listening/watching audio/video contents (from disk or internet) Using VoIP Program Ventrilo Audio from others people works good, and I hear their voices clearly... Using VoIP Program such as Teamspeak 2/3, MSN, Skype I hear a really disturbed voice, and it's impossible to comprehend somethings... Anyway I can say that the first week that I've bought and installed Win7 Skype worked good, this problem happened after I installed Ventrilo... I tried to remove it (ventrilo) but nothing happened:( Please help me :D Thanks Marcx

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  • Integrated Graphics and Audio as Media Center?

    - by Will
    I'm considering setting up a PC as a Media Center. Mainly to watch movies (ideally HD quality) and listening to music, but also to perform tasks like e-mail, web browsing, ... I quite like the looks and the price of this barebone: http://www.asus.de/Barebone_PC/S_Series_7L/S2P8H61E However it comes with integrated graphics and audio and only has one free PCI-Express slot. Which would mean in the worst case, where both integrated graphics and audio turn out to be insufficient, I could only upgrade one. So is integrated graphics and audio sufficient for a media center solution? Cheers, Will

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  • Audio broken on clean install of ThinkPad T60

    - by Ben Alpert
    I just reinstalled a clean version of XP SP3 on a ThinkPad T60. After downloading all of the missing drivers from the Lenovo site, everything seems to work except for the audio. I installed the audio driver from http://www-307.ibm.com/pc/support/site.wss/MIGR-62928.html (version 5.10.1.4326) so SoundMAX shows up in the control panel and everything in the Sound preferences looks okay, but no sound comes out of the speakers when I try to play music or hear the interface sound effects. There's also no sound if I plug in headphones and listen. Some forums suggested installing hotfix Q888111 but it's already included in SP3 so I can't install it. I also tried reinstalling the audio driver mentioned above in case something went wrong the first time, but it still doesn't work. Does anyone know how to make the sound work?

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  • Enable multiple audio output on Windows 7

    - by patrick
    For Windows 7, 64 bit: I have a digital SPDIF output to my stereo, which controls speakers in other rooms. I also have a set of speakers connected to the regular audio jack at the computer. This allows me to send music to the kitchen while my child plays games on the computer. Works great. Except when I'm playing games and still want to listen to music. ;-D I know I can manually switch WMP to play through the speakers instead of SPDIF, but I was wondering if there's any way to enable simultaneous audio out in Windows 7? Virtual Audio Card is a non-starter because I'm running 64 bits and the VAC driver isn't signed.

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  • HDMI Audio drops out when display enters powersave

    - by Jared Tritsch
    I have a Windows 8 machine with an AMD APU attached to my Home Theater system through HDMI (HDMI routes through a Home Theater AMP, then into the TV). Here's my problem, Whenever the display is interrupted, usually by the TV being turned off or into powersave mode, the audio device lists as "Disconnected" in windows audio devices and I cant get it to re-recognize that the HDMI audio is, in fact, plugged in. The only solution I have found so far is to restart the machine, which will then recognize the device without any problems, until the next time the TV turns off and the problem once again resurfaces. Has anyone else seen this phenomenon? I have no idea if its the GPU, the HDMI interface, the AMP, or even the TV itself, as there really isn't much a way to tell...

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  • Logitech bluetooth audio receiver quality

    - by lietus
    I bought a Logitech Bluetooth Audio receiver, and have run into a problem: When playing audio, it contains short hisses/clicks on higher tones or vocal+heavy instrumental music (Sound quality appears to be unclean). I'm not sure what could be the reason, but this only happens when I play from certain devices. My laptops (a Dell XPS Win7, and an Asus Eee seashell Win7) and my phone (Samsung Player 5) both have the problem. However, when I tried using a Samsung S2 phone, the audio was crystal clear. Seems that this could be something with Bluetooth transiting device.. Has anyone encountered a similar problem?

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  • Ubuntu 11.04: No Audio Output

    - by Jason George
    I installed a fresh copy of Ubuntu 11.04 earlier this week and I'm having trouble getting my audio online. It was working fine in 10.04 and all the resources I can find on troubleshooting seem to be fairly dated so I'm not sure if they apply. CMI8788 [Oxygen HD Audio] Analog Stereo Duplex Playing a WMA file shows 0.00db output when I mouse over the sound controller in the status bar. Obviously, no output from my speakers. I tried adjusting the profile, thinking I might have the wrong one. That seems to have made things worse. Where mouse over originally said something along the lines of "Oxygen HD Audio," it now reads "Dummy Output." Selecting "Test Speakers" in sound preferences crashes the dialog. Any pointers would be great.

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  • FFmpeg add multiple audio files to video at specific points

    - by Arran
    I have two audio files, each about 3 minutes long. I want to take the first 10 seconds of each file and add them each to a video file at specific points - 0 seconds and 10 seconds. So the resulting video should be 20 seconds long. I've got this far: ffmpeg -i video.mov -ss 0 -t 20 -itsoffset 0 -i audio1.mp3 -itsoffset 10 -i audio2.mp3 -acodec copy -vcodec copy out.mov ...but the resulting video has 20 seconds of the first audio file only, the second audio file doesn't start at 10 seconds like it should. Any help would be appreciated, thanks!

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  • Flash Player Plugin on Firefox will not update on Ubuntu 11.10 32 bit

    - by George
    I am running on Ubuntu 11.10 32 bit, and my flash player plug-in for Firefox is outdated. I must update my plug-in to flash player 11, but I am not able. I have already installed Adobe Flash Player via Ubuntu Software center (As it prompted me to after trying to download from the Adobe site) Without the plug-in updated, videos can not be played from directly on a Facebook page, and I believe that it distorts the video player on Youtube. (Image of player can be seen at imgur.com/gYbFx)

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  • Problem in installing player/gazebo

    - by usama yaseen
    I am using ubuntu 10.04 (64-bit), i have sucessfully installed all the dependancies of player and gazebo, then i installed player 3.01 and then gazebo 0.9,Now the when i run this gazebo worlds/pioneer2dx.world I get a gazebo interface, everything is ok uptill now, but when i run this cd gazebo/player player gazebo.cfg player: error while loading shared libraries: libplayerdrivers.so.3.0: cannot open shared object file: No such file or directory I guess the problem is somewhere in the environmental variables, can anyone please help me to solve this?

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  • synchronizing audio over a network

    - by sharkin
    I'm in startup of designing a client/server audio system which can stream audio arbitrarily over a network. One central server pumps out an audio stream and x number of clients receives the audio data and plays it. So far no magic needed and I have even got this scenario to work with VLC media player out of the box. However, the tricky part seems to be synchronizing the audio playback so that all clients are in audible synch (actual latency can be allowed as long as it is perceived to be in sync by a human listener). My question is if there's any known method or algorithm to use for these types of synchronization problems (video is probably solved the same way). My own initial thoughts centers around synchronizing clocks between physical machines and thereby creating a virtual "main timer" and somehow aligning audio data packets against it. Some products already solving the problem: http://www.sonos.com http://netchorus.com/ Any pointers are most welcome. Thanks. PS: This related question seem to have died long ago.

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  • Audio Detection in Matlab

    - by insane-36
    I am writing a matlab code that would be able to read the audio file and then compare it to the another audio and recognize if those audio are the voice of the same person. In both type of the audio, would have the same word utterance and the audio is about 1 minutes long. I have come to know that the approach of sliding windows using hamming window would work best on this approach but have a very little idea on this. The simple code to read an audio file and then display a portion of 10s is as below : [x,fs, nbits]= wavread('01-AudioTrack 01.wav'); subplot(211) plot(x) title('Entire Wave') smallRange = 1:100000; subplot(212) plot(smallRange,x(smallRange)) How do I make Hamming window each of 10ms in this case and what approaches should I take to deal with this problem ?

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