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  • Getting Audio from a Zone

    - by bleonard
    Now that I have Firefox and Java Web Start running from a zone, the last piece of the puzzle was audio (essential because most Flash content is accompanied by sound).  In the global zone there's a nice little utility called audiotest for testing your sound: bleonard@solaris:~$ audiotest Sound subsystem and version: SunOS Audio 4.0 (0x00040003) Platform: SunOS 5.11 snv_151a i86pc *** Scanning sound adapter #1 *** /dev/sound/audio810:0dsp (audio engine 0): audio810#0 - Performing audio playback test... <left> ................OK <right> ...............OK <stereo> ..............OK <measured sample rate 47727.00 Hz (-0.57%)> *** All tests completed OK *** Of course, before you can try audiotest in a zone, it must be installed: root@myzone:~# pkg install audio-utilities Packages to install: 1 Create boot environment: No DOWNLOAD PKGS FILES XFER (MB) Completed 1/1 6/6 0.4/0.4 PHASE ACTIONS Install Phase 20/20 PHASE ITEMS Package State Update Phase 1/1 Image State Update Phase 2/2 However, we'll need to do more than just install audiotest: root@myzone:~# audiotest /dev/mixer: No such file or directory The device file is missing from /dev. The audio devices also need to be added to the zone. For this we modify the zone configuration as follows: bleonard@solaris:~$ sudo zonecfg -z myzone Password: zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/audio* zonecfg:myzone:device> end zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/sound/* zonecfg:myzone:device> end zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/mixer* zonecfg:myzone:device> end zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/sndstat zonecfg:myzone:device> end zonecfg:myzone> verify zonecfg:myzone> exit Then reboot the zone: bleonard@solaris:~$ sudo zoneadm -z myzone reboot After which, audiotest should work: root@myzone:~# audiotest Sound subsystem and version: SunOS Audio 4.0 (0x00040003) Platform: SunOS 5.11 snv_151a i86pc *** Scanning sound adapter #1 *** /dev/sound/audio810:0dsp (audio engine 0): audio810#0 - Performing audio playback test... <left> ................OK <right> ...............OK <stereo> ..............OK <measured sample rate 48208.00 Hz (0.43%)> *** All tests completed OK *** You can also examine /dev/sndstat for additional information: root@myzone:~# cat /dev/sndstat SunOS Audio Framework Audio Devices: 0: audio810#0 Intel AC'97, ICH (DUPLEX) Mixers: 0: audio810#0 Intel AC'97, ICH AC'97 codec: SigmaTel STAC9700 However, when testing the sound from Firefox (from a user account other than root), such as this recent Flash presentation on Solaris availability, you may still be disappointed. This is simply a permissions problem, as the devices only have read and write permissions for root: root@myzone:~# ls -l /dev/audio* crw------- 1 root root 99, 3 Jul 1 10:21 /dev/audio crw------- 1 root root 99, 4 Jul 1 10:21 /dev/audioctl To address this: root@myzone:~# chmod 777 /dev/audio* root@myzone:~# chmod 777 /dev/sound/* And you should be all set.

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  • How to use an Audio Unit on the iPhone

    - by CodeToaster
    I'm looking for a way to change the pitch of recorded audio as it is saved to disk, or played back (in real time). I understand Audio Units can be used for this. The iPhone offers limited support for Audio Units (for example it's not possible to create/use custom audio units, as far as I can tell), but several out-of-the-box audio units are available, one of which is AUPitch. How exactly would I use an audio unit (specifically AUPitch)? Do you hook it into an audio queue somehow? Is it possible to chain audio units together (for example, to simultaneously add an echo effect and a change in pitch)? EDIT: After inspecting the iPhone SDK headers (I think AudioUnit.h, I'm not in front of a Mac at the moment), I noticed that AUPitch is commented out. So it doesn't look like AUPitch is available on the iPhone after all. weep weep Apple seems to have better organized their iPhone SDK documentation at developer.apple.com of late - now its more difficult to find references to AUPitch, etc. That said, I'm still interested in quality answers on using Audio Units (in general) on the iPhone.

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  • Unable to Turn On Media Streaming in Windows Media Player 12 on Windows 7

    - by Chau Chee Yang
    I have 2 PC installed with Windows 7 and Media Player 12. I would like to use Play To feature on both PC connected via LAN. Both PC (A and B) run media player in standard user account. I able to turn on media streaming option in PC A (with privilege access prompt) without any problem. However, PC B also prompt privilege access but no response after enter administrator password. Both PC follow same configuration steps. I may use "play to" PC A (in standard user account) from other PC without any problem. But I can't "play to" PC B in standard user account. I can only run media player in administrator account for "play to" to function. I have tried uninstall and reinstall media player via "Programs and Features" in control panel on PC B. However, it doesn't work too. Does anyone has similar experience as me failing to turn on media streaming that running Windows media player in standard user account?

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  • Architecture behind live streaming [on hold]

    - by l19
    I'm a Comp Sci undergraduate student, and I'm currently trying to understand the architecture behind streaming. I hear several terms and I'm not quite sure how they are related (e.g. streaming, broadcasting, ingesting, etc.) Is there a blog post or book that explains: How it all works in a high-level view (the workflow) The architecture (i.e. I capture content using my camera and want to display it real-time to an audience. I imagine that the content will be transferred to a server, but how does that server transmit the information to several users simultaneously?) Thanks!

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  • Is an Intranet music streaming server legal?

    - by Jon Smock
    We are a large organization with thousands of users, and we're peaking on our Internet usage. Many of those users are streaming music while they work. We're wondering if providing a music streaming server internally would help on bandwidth. How legal is that? Here are two scenarios: 1) We purchase a body of music legally and stream it internally (I assume this is illegal) 2) We pull music feeds from free, legal, online sources and "rebroadcast" internally (I assume this is legal) We want to save bandwidth and help our users, but we want to do it in an ethical and legal way.

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  • MWS2K8R2: Enabling Media Sharing using Streaming Media Services Role

    - by TheLizardKing
    So I have a Microsoft Windows Server 2008 R2 that stores a large collection of media (mostly mp3s) and I want to be able to deliver these files using a server/client setup with Windows Media Player being the client. I downloaded and installed Streaming Media Services Role. I even setup a publishing point with on-demand access. My issue is I can connect using WMP12 but it only connects as more of a stream and not a shared library. I can pause/play/skip as if it's a powerful radio station which is ok in my book but what I'd really like to do is allow me to control my music remotely, search and play for artists, maybe create playlists (not required but nice) and even connect it to an xbox. Is Streaming Media Services Role not what I should be using for this? Would installing WMP and sharing using that mechanism be a better option? Any Ideas?

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  • Darwin Streaming Server Interval Role

    - by Asmv9
    I want to use the Interval Role in Darwin streaming server,more specifically i want to call the QTSS_SetIntervalRoleTimer() when the server starts streaming a video file.But i have problem in doing this as from what it seems,the method returns QTSS_Err when i do that. I believe that the problem is due to the fact that the callback is made in a a module-created thread. Is there a specific place where this callback must be done?(if i put the callback in the register role of my module it works,but i dont want this,because i dont want the timer starting when the server starts).Any help will be useful,thank u in advance.

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  • Dowload size of Streaming Videos

    - by Excalibur2000
    I would like to know that if a website advertises a streaming download as say 100MB, would my download to my computer be 100MB ? Would there be streaming control packets that a service provider would charge for over and above the 100MB content ? Assume the latest RealPlayer viewer. The rub for me is that I have downloaded MIT lectures and according to my file manager the file sizes have matched up to the download sizes on YouTube. However my ISP seems to think that the streams were larger and charged me for more than the file size of the download. I am left wondering where the data came from.

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  • bursty streaming video

    - by broiyan
    What is the cause and solution for the bursty streaming media problem? Example: when streaming from youtube, audio (and video) will pause and start intermittently. When it starts it will be bursty, that is, it will play several seconds of sound in just a fraction of a second. Normal sounds are rendered unrecognizable. Then it may pause and after a few seconds, resume with another burst. The video seems to burst along with the audio. This was observed on Ubuntu 12.04 with Google Chrome.

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  • <audio> elements not working on WordPress

    - by dannystewart
    Hello all, I have a small WordPress site. I do a lot of audio work and I'm trying to post HTML5 audio clips in blog entries on WordPress. For some reason it isn't working. It might have something to do with the style I'm using on my WordPress site but I haven't been able to nail it down. I know my audio tags are valid, as they work elsewhere. Here's an example audio tag: <audio src="http://files.dannystewart.com/dom2008.mp3"></audio> And here's a page demonstrating it not working: http://www.dannystewart.com/html5-audio-test/ I'm quite sure this is something very simple that I've just missed, but any pointers would be appreciated. Thanks!

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  • How do I set up live audio streams to a DLNA compliant device?

    - by Takkat
    Is there a way to stream the live output of the soundcard from our 12.04.1 LTS amd64 desktop to a DLNA-compliant external device in our network? Selecting media content in shared directories using Rygel, miniDLNA, and uShare is always fine - but so far we completely failed to get a live audio stream to a client via DLNA. Pulseaudio claims to have a DLNA/UPnP media server that together with Rygel is supposed to do just this. But we were unable to get it running. We followed the steps outlined in live.gnome.org, this answer here, and also in another similar guide. As soon as we select the local audio device, or our GST-Launch stream in the DLNA client Rygel displays the following message and the client states it reached the end of the playlist: (rygel:7380): Rygel-WARNING **: rygel-http-request.vala:97: Invalid seek request This is how we configured GST-Launch in rygel.conf: [GstLaunch] enabled=true launch-items=mypulseaudiosink mypulseaudiosink-title=Audio on @HOSTNAME@ mypulseaudiosink-mime=audio/x-wav mypulseaudiosink-launch=pulsesrc device=<device> ! wavpackenc For <device> we tried with the default sink name, this name appended with .monitor, and in addition with upnp-sink and upnp.monitor that was created when we selected DLNA media server from paprefs. We also tried to encode using lamemp3enc with no luck. These are our pulseaudio modules: http://paste.ubuntu.com/1202913/ These are our sinks: http://paste.ubuntu.com/1202916/ Did we miss any other additional configuration needed to get this running? Are there any other alternatives for sending the audio of our soundcard as live stream to a DLNA client?

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  • Concatenation of a 2 second silence audio with a normal audio not working

    - by user1665130
    I have a code for concatenation of files using ffmpeg.Here silence.wav is a mute audio file with 2 seconds length. I need to prepend this mut audio file to REC00096_Jun-06-2014 16.47.28.wav. I tried the folowing code. ffmpeg -i D:\vishnu\silence.wav -i D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav \-filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' \-map '[out]' output.wav Following is the error i am getting. D:\vishnu>ffmpeg -i silence.wav -i "D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav" -filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' -map '[out]' outp ut.wav ffmpeg version N-59036-g5d8e4f6 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 12 2013 22:01:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 58.100 / 52. 58.100 libavcodec 55. 45.101 / 55. 45.101 libavformat 55. 22.100 / 55. 22.100 libavdevice 55. 5.102 / 55. 5.102 libavfilter 3. 92.100 / 3. 92.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, wav, from 'silence.wav': Metadata: encoder : Lavf55.22.100 Duration: 00:00:02.02, bitrate: 4234 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 5.1, s16, 4 233 kb/s Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav': Duration: 00:00:08.04, bitrate: 384 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, mono, s16, 384 kb/s [wav @ 036f5e40] Invalid stream specifier: '[out]'. Last message repeated 1 times Stream map ''[out]'' matches no streams. D:\vishnu>

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  • HTML5 <audio> Safari live broadcast vs not

    - by Peter Parente
    I'm attempting to embed an HTML5 audio element pointing to MP3 or OGG data served by a PHP file . When I view the page in Safari, the controls appear, but the UI says "Live Broadcast." When I click play, the audio starts as expected. Once it ends, however, I can't start it playing again by clicking play. Even using the JS API on the audio element and setting currentTime to 0 fails with an index error exception. I suspected the headers from the PHP script were the problem, particularly missing a content length. But that's not the case. The response headers include a proper Content- Length to indicate the audio has finite size. Furthermore, everything works as expected in Firefox 3.5+. I can click play on the audio element multiple times to hear the sound replay. If I remove the PHP script from the equation and serve up a static copy of the MP3 file, everything works fine in Safari. Does this mean Safari is treating audio src URLs with query parameters differently than URLs that don't have them? Anyone have any luck getting this to work? My simple example page is: <!DOCTYPE html> <html> <head></head> <body> <audio controls autobuffer> <source src="say.php?text=this%20is%20a%20test&format=.ogg" /> <source src="say.php?text=this%20is%20a%20test&format=.mp3" /> </audio> </body> </html> HTTP Headers from PHP script: HTTP/1.x 200 OK Date: Sun, 03 Jan 2010 15:39:34 GMT Server: Apache X-Powered-By: PHP/5.2.10 Content-Length: 8993 Keep-Alive: timeout=2, max=98 Connection: Keep-Alive Content-Type: audio/mpeg HTTP Headers from direct file access: HTTP/1.x 200 OK Date: Sun, 03 Jan 2010 20:06:59 GMT Server: Apache Last-Modified: Sun, 03 Jan 2010 03:20:02 GMT Etag: "a404b-c3f-47c3a14937c80" Accept-Ranges: bytes Content-Length: 8993 Keep-Alive: timeout=2, max=100 Connection: Keep-Alive Content-Type: audio/mpeg I tried hard-coding the Accept-Ranges header into the script too, but no luck.

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  • Stream video from a computer to a tablet

    - by NullOrEmpty
    I would like to watch presentations and college videos I have on my computer, from my tablet. There are almost a hundred of GB in videos, so I would like to watch them in streaming from my computer. My wife has also tons of videos so I have decided to do a kind of streaming local service to watch the videos from our tablets. I have found an article about how to do it with Internet Information Server, but the article is relaying in an application that is not free (Expression Encoder). Since this is for home fun, I am not willing to pay, so I would like to ask for some free encoder that can do the trick. I have no idea about streaming. Actually, I tried to hang the files directly on IIS, and the browser tried first to download some of them and some other (mp4?) got played badly, so I could not get a smooth video experience. What is missing? What is the deal with H.264 ? Can I roll with VC-1 and play videos in my Android tablet with streaming with an acceptable quality through my WLAN home network? Any better solution? Thanks.

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  • Windows Audio Issue

    - by Nikki
    This one is driving me nuts. Hoping someone can shed some light. I'm running windows 7 using onboard audio. It's been fine for over 2 years but lately there's a problem every time I play audio. I hear a small soft burst of static and the volume turns itself down from 50% to 23%. Once at 23%, it plays fine. No related events logged in viewer. No reported problems with the device. Different headphones, same problem. I played around with audio settings for hours but the problem persists. EDIT: ok more info: Motherboard: ECS G31T-M LGA775 System info displays this: Name High Definition Audio Device Manufacturer Microsoft Status OK PNP Device ID HDAUDIO\FUNC_01&VEN_1106&DEV_E721&SUBSYS_10192683&REV_1001\4&3D4E739&0&0001 Driver c:\windows\system32\drivers\hdaudio.sys (6.1.7600.16385, 297.00 KB (304,128 bytes), 14/07/2009 9:51 AM) I'll keep adding info as I find it. The question I want resolved is; Is it faulty hardware? If so, I can buy a sound card. I can't imagine software is responsible since I haven't installed anything new for weeks. Virus scans are clear as well. The static burst is irritating to say the least. Tried 2 different headphones and separate speakers. Same problem. I know it's not an easy problem but I was hoping someone had encountered the same thing.

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  • Simulating audio playback on headless linux server

    - by afro
    Hi people, We have a headless linux server (Debian 5) we use for runnin integration tests of our web-page code. Among these tests are ones implemented using Selenium, which practically simulates a user browsing our pages and clicking on things. One of these tests is failing now, because it involves starting a flash-based audio player and checking to see whether the progress bar gets displayed properly. The reason this test fails is that there is no way to play the audio, and no sound card on the machine, which has simple webserver hardware. So, my question would be: Is there a simple way of giving a program the impression that its audio output is being processed, and playback is taking place? I don't have to record the playback, or redirect it or anything like that, just a dummy soundcard, like the dummy X-server we aer using, which actually does not need to display stuff. I have tried using JACK, but it's too complicated, and the documentation does not even answer this very simple question. I also installed alsa on the server; it 'pretends' to run, but when a program tries to play audio, just spews error and debug information having to do with the non-existence of a soundcard. It would be really awesome if one of you has a simple answer to this question. Cheers, Ulas

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  • Streaming flash video does not work on my Mac OS X

    - by dehmann
    Flash videos do not work properly on my Mac. On this Vimeo video, for example, it shows only the beginning frame, and audio stutters like crazy, playing audio for a quarter second or so, then silence, then playing again, etc. I have Flash version 10,0,42,34 on Mac OS 10.5.8. It's a PowerBook G4 (PPC). I tried it in Firefox 3.5.5 and Safari 4.0.3. I tried reinstalling Flash, restarting the computer, and using a fresh user profile in Firefox (so that no extensions are interfering with the site), loading the video fully before playing, but nothing helps. I noticed that youtube videos work better, once loaded enough, although the picture does halt briefly once every 10 or so seconds, even when it's fully loaded.

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  • video streaming infrastructure advice

    - by Alchemical
    We would like to set-up a live video-chat web site and are looking for basic recomendations for software and hardware set-up. Most streams will be broadcast live from a single person with a web cam, etc., and viewed by typically 1-10 people, although there could be up to 100+ viewers on the high side. Audio and video do not have to be super-high quality, but do need to be "good enough". The main point is to convey the basic info in the video (and audio). If occasionally the frame-rate drops low and then goes back to normal fairly soon, we could live with that. Budget is an issue, so we are in general looking for a lower cost solution that will give us most of what we need in temers of performance and quality. We are looking at Peer1 for co-lo. The rest of our web site will be .Net / Windows platform. We are open to looking at any platform for the best streaming solution, although our technical expertise is currently more on the Windows side.

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  • Setting Up a Virtual Sound Device To Stream Output via TCP/IP

    - by Martindale
    I'm interested in installing a custom driver / device that will open a socket (or listen) and stream audio via TCP/IP in both Windows and Linux. I would like to be able to specify this device as my "Output" for specific applications so that I can route my audio through a completely unique machine (for example, in complex Synergy setups where my headphones might be connected to one machine, but audio is being generated by another.) In Linux, I expect this to be very easy, but I anticipate having to install a custom driver in Windows.

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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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  • 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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  • Writing an audio player in C#

    - by Malki
    Hi, I have a pretty cool idea for a very special media player. I like to think about this project as a mini-startup, since I don't yet know if my idea is practical. Anyways, before implementing my idea, I first need to be able to implement a simple audio player. My preferred language for this project is C#, simply because it's so easy to use, but any other object oriented language would be fine too I guess. I started out with no knowledge whatsoever about audio. My main goals right now are: Being able to play audio files - as many formats as possible (sort of a VLC type player, but only audio for now). Being able to analyze audio files - as in, reading frequency, amplitude, volume, and other information about the audio. I think maybe a good idea here is to be able to analyze one file format (PCM?), and then temporarily converting any file I want to analyze to that format. This is in order to later implement a mechanism that compares songs and identifies similar songs to recommend to the user (this feature isn't part of my idea, but I figured since it exists in many players nowadays, I need to have it too if I want be able to compete with them). BTW - I currently don't have any knowledge about audio/wavelengths/frequencies and such, so I'd appreciate it if someone could point me in the right direction about this analyzation feature. Maybe in the future I'd expand to playing video files as well, but for now I'm concentrating on audio. After searching the Internet for a while, I've come across LAME. Problem is, it's not C#, and I'm not sure how to use it. I know there is something called "Interoperability", that is supposed to let me work with native DLL files through C#. Any information about that would be helpful as well. Any help would be much appreciated. Thanks, Malki :)

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  • Read Song Title/Artist from a live audio stream with Silverlight 4?

    - by Brent Pabst
    I have a SL4 project that is successfully streaming a great sounding WMA audio stream from a remote location. All of the MediaElement actions are straight forward. What I want to do is read the attributes that are passed as text along with the Audio stream. For instance the encoder of the stream embeds the title of the stream, the title of the song playing and the name of the artist for the current song. How would I pick this out using Silverlight 4 and then display it in a Label to the user? It sure would be easier than writing a bunch of web services to do the same thing. Windows Media Player and WinAmp all get the information I am just not seeing it in the MediaElement object collection.

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