Search Results

Search found 10738 results on 430 pages for 'streaming video'.

Page 126/430 | < Previous Page | 122 123 124 125 126 127 128 129 130 131 132 133  | Next Page >

  • Does the size of the monitor Matter?

    - by Arsheep
    I have a old computer, and I want to buy a big LCD. The best I've found so far is Viewsonic's 24" LCD TFT monitor. So will it run without any problems, or do I need to upgrade the video cards or something as well? The computer is not too old: it has P4 board and celeron processor, with 128 graphics memory. And in display properties, it says that the maxium that I can use is 1280 x 1024 resolution. I am noob hardware-wise, so need help on this stuff. Thanks

    Read the article

  • Does the size of the monitor Matter?

    - by Arsheep
    I have a old computer, and I want to buy a big LCD. The best I've found so far is Viewsonic's 24" LCD TFT monitor. So will it run without any problems, or do I need to upgrade the video cards or something as well? The computer is not too old: it has P4 board and celeron processor, with 128 graphics memory. And in display properties, it says that the maxium that I can use is 1280 x 1024 resolution. I am noob hardware-wise, so need help on this stuff. Thanks

    Read the article

  • How can I stream my laptop's desktop to my iPad?

    - by Bane
    I recently got a wireless PC controller, and now I enjoy games playing from my bed. However, I find it hard to correctly place my laptop so it's comfortable, and it would be great if there was a way to view my screen through my iPad, with minimal lag; so I could simply leave my laptop behind me or on the desk and view the game from the iPad. Is this feasible/realistic? Additional info: Both the iPad and the laptop are on the same WiFi network

    Read the article

  • FFMPEG Splitting MP4 with Same Quality

    - by Pragmatic
    I have one large MP4 file. I am attempting to split it into smaller files. ffmpeg -i largefile.mp4 -sameq -ss 00:00:00 -t 00:50:00 smallfile.mp4 I thought using -sameq would keep the same quality settings. However, I must not understand what that does. I'm looking to keep the same quality (audio/video) and compression with the split files. However, this setting makes the split files much larger. What flag(s) do I need to set to keep the same quality and attributes in the split files while maintaining the same quality to size ratio? For instance if my original file is about 12 GB and is 1920x1080 with a bitrate of 10617kbps and a framerate of 23 frames/sec and 6 channel audio with 317kbps, I would like the split files to be the same only a third of this size (if i split it into three pieces).

    Read the article

  • Nvidia driver on Windows 7 causing black screen

    - by inKit
    I have just installed Windows 7 on a desktop machine and for the first time ever have had a really tough time doing so, its normally a nice smooth install. This time I found that the monitor would simply go black after completing the installation. I tried reinstalling about 3 times and this did not help. After much searching I discovered that it was the nvidia drivers that were playing up with win 7, so i booted into safe mode, disabled the device, then rebooted to complete the installation. Windows 7 now works fine as long as the nvidia 9600 gt video card is disabled. The moment I enable it, the system requires a reboot and the screen will go black before even getting to the log in screen. I have tried downloading the latest driver and installing it manually, I have also tried uninstalling the device and allowing windows 7 to install it itself. Nothing seems to work. any clues?

    Read the article

  • Why are my favorite websites becoming slower, over months?

    - by Wolfpack'08
    I spend a lot of my time at sites for watching online videos: youtube, gorillavid, thedailyshow.com etc. I used to watch the videos in full screen mode, and then that became very laggy. So, I started watching them with full-browser zooming. Then that became laggy. Recently, I've had to actually zoom out; otherwise, the video will lag so much that my PC locks. Could this be a symptom of my processor, RAM, or motherboard going bad? Has it, perhaps, anything to do with softwares like Chrome or the playeres the sites are using being updated?

    Read the article

  • Nvidia driver on Windows 7 causing black screen

    - by inKit
    I have just installed Windows 7 on a desktop machine and for the first time ever have had a really tough time doing so, its normally a nice smooth install. This time I found that the monitor would simply go black after completing the installation. I tried reinstalling about 3 times and this did not help. After much searching I discovered that it was the nvidia drivers that were playing up with win 7, so i booted into safe mode, disabled the device, then rebooted to complete the installation. Windows 7 now works fine as long as the nvidia 9600 gt video card is disabled. The moment I enable it, the system requires a reboot and the screen will go black before even getting to the log in screen. I have tried downloading the latest driver and installing it manually, I have also tried uninstalling the device and allowing windows 7 to install it itself. Nothing seems to work. any clues?

    Read the article

  • 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.

    Read the article

  • HTTP Headers for Unknown Content-Length

    - by jocull
    I am currently trying to stream content out to the web after a trans-coding process. This usually works fine by writing binary out to my web stream, but some browsers (specifically IE7, IE8) do not like not having the Content-Length defined in the HTTP header. I believe that "valid" headers are supposed to have this set. What is the proper way to stream content to the web when you have an unknown Content-Length? The trans-coding process can take awhile, so I want to start streaming it out as it completes.

    Read the article

  • Finding out estimated duration of a stream using Core Audio

    - by Reflog
    I am streaming a MP3 over network using custom feeding code, not AVAudioPlayer (which only works with URLs) using APIs like AudioFileStreamOpen and etc. Is there any way to estimate a length of the stream? I know that I can get a 'elapsed' property using: if(AudioQueueGetCurrentTime(queue.audioQueue, NULL, &t, &b) < 0) return 0; return t.mSampleTime / dataFormat.mSampleRate; But what about total duration to create a progress bar? Is that possible?

    Read the article

  • FMS NetConnection.Connect.Close happening when starts and even in the middle of video in Flash with

    - by Sunil Kumar
    Hi I have developed a Flash Video player in Flash CS3 with Action Script 2.0 to play video from Adobe Flash Media Server 3.5. To play video from FMS 3.5, first I have to verify my swf file on FMS 3.5 server console so that it can be ensure that RTMP video URL only be play in verified SWF file. Right now I am facing problem of "NetConnection.Connect.Close" when I try to connect my NetConnection Object to FMS 3.5 to stream video from that server. So now I am getting this message "NetConnection.Connect.Close" from FMS 3.5. When this is happening in my office area at the same time when I am checking the the same video url from out side the office (With help of my friends who is in another office) area it is working fine. My friends naver faced even a single issue with NetConnection.Connect.Close. But in my office when I got message NetConnection.Connect.Close, I can play another streaming video very well like mtv.com jaman.com rajshri.com etc. Some time FMS works fine and video starts playing but in the middle of the video same thing happen "NetConnection.Connect.Close" There is no issue of Bandwidth in my office. I do't know why this is happening. Please see the message when I am getting "NetConnection.Connect.Close" message. NetConn == data: NetConn == objectEncoding: 0 NetConn == description: Connection succeeded. NetConn == code: NetConnection.Connect.Success NetConn == level: status NetConn == level: status NetConn == code: NetConnection.Connect.Closed Please help Thanks & regards Sunil Kumar

    Read the article

  • How do I close a database connection in a WCF service?

    - by Dan
    I have been unable to find any documentation on properly closing database connections in WCF service operations. I have a service that returns a streamed response through the following method. public virtual Message GetData() { string sqlString = BuildSqlString(); SqlConnection conn = Utils.GetConnection(); SqlCommand cmd = new SqlCommand(sqlString, conn); XmlReader xr = cmd.ExecuteXmlReader(); Message msg = Message.CreateMessage( OperationContext.Current.IncomingMessageVersion, GetResponseAction(), xr); return msg; } I cannot close the connection within the method or the streaming of the response message will be terminated. Since control returns to the WCF system after the completion of that method, I don't know how I can close that connection afterwards. Any suggestions or pointers to additional documentation would be appreciated. Dan

    Read the article

  • Seeking through a streamed MP3 file with HTML5 <audio> tag

    - by Kyle Slattery
    Hopefully someone can help me out with this. I'm playing around with a node.js server that streams audio to a client, and I want to create an HTML5 player. Right now, I'm streaming the code from node using chunked encoding, and if you go directly to the URL, it works great. What I'd like to do is embed this using the HTML5 <audio> tag, like so: <audio src="http://server/stream?file=123"> where /stream is the endpoint for the node server to stream the MP3. The HTML5 player loads fine in Safari and Chrome, but it doesn't allow me to seek, and Safari even says it's a "Live Broadcast". In the headers of /stream, I include the file size and file type, and the response gets ended properly. Any thoughts on how I could get around this? I certainly could just send the whole file at once, but then the player would wait until the whole thing is downloaded--I'd rather stream it.

    Read the article

  • Optimum encoding standard for flowplayer to play mp4

    - by renjucool
    I'm using flow player 3.1.1 for streaming videos to my browser.The videos are uploaded by the users and they may upload different formats. What will be solution to stream the videos as mp4 , what ever be the format they upload. I'm currently using ffmpeg commands. ffmpeg -i "InputFile.mp4" -sameq -vcodec libx264 -r 35 -acodec libfaac -y "OutputFile.mp4" But video files of more size(say 100mb) are taking a minute more for laoding in to the flowplayer and buffering. I think the problem with my encoding. Welcome your valuable Suggestions!!!

    Read the article

  • Why does use of H264 in sender/receiver pipelines introduce just HUGE delay?

    - by Serguey Zefirov
    When I try to create pipeline that uses H264 to transmit video, I get some enormous delay, up to 10 seconds to transmit video from my machine to... my machine! This is unacceptable for my goals and I'd like to consult StackOverflow over what I (or someone else) do wrong. I took pipelines from gstrtpbin documentation page and slightly modified them to use Speex: This is sender pipeline: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 Receiver pipeline: !/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H263-1998" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false Those pipelines, a combination of H263 and Speex, work fine enough. I snap my fingers near camera and micropohne and then I see movement and hear sound at the same time. Then I changed pipelines to use H264 along the video path. The sender becomes: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! x264enc bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 And receiver becomes: #!/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false This is what happen under Ubuntu 10.04. I didn't noticed such huge delays on Ubuntu 9.04 - the delays there was in range 2-3 seconds, AFAIR.

    Read the article

  • vista bandwith reservation

    - by user185646
    I would like to write my own version of Microsoft Live labs pivot.http://www.getpivot.com/ For this i will use realtime texture streaming technology like John Carmack did for doom4. But i would like to use Windows vista SetFileBandwidthReservation api to have the best throughput possible. For example // reserve bandwidth of 200 bytes/sec result = SetFileBandwidthReservation( hFile, 1000, 200, FALSE, &transferSize, &outstandingRequests ); What i dont understand is the lpTransferSize and lpNumOutstandingRequests return parameters. How should i next read the file for this to be the most worth it. Should i do exactly lpNumOutstandingRequests number of request of size lpTransferSize. Or can i do one synchronous request bigger than lpTransferSize.

    Read the article

  • Watermarking Flash Videos (server-side)

    - by Roberto Aloi
    Hi all, I have a bunch of flash videos that I need to watermark with user related information, to make illegal re-distribution of these files harder. I'm wondering how can this be done server-side. If done client-side, it will be quite easy for the user to intercept the videos before they are watermarked. Since the watermark should contain user-specific information I can't really watermark the videos before encoding them (unless I have an encoded video per user - not feasible). I'm expecting this to affect the streaming performances a lot, though. Any idea how this can be done (possibly in an efficient way)?

    Read the article

  • Remote stream multiple files in SOLR

    - by Mark
    I want to use SOLR's remote-streaming facility to extract and index the content of files. This works fine if I pass stream.file=xxx as a parameter to the http GET method. However, I have a lot of these, and want to batch them up (i.e. not have to have a GET per file). Is there a way I can do this in SOLR? e.g. I'd like to be able to POST some xml like this: <add> <doc stream_file="filename"> <field name="id">123</field> </doc> <doc>...

    Read the article

  • Seeking not working in HTML5 audio tag

    - by lord_wilmore
    I have a lighttpd server running locally. If I load a static file on the server (through an html5 audio tag), it plays and seeks fine. However, seeking doesn't work when running a dev server (web.py/CherryPy) or if I return the bytes via a defined action url instead of as a static file. It won't load the duration either. According to the "HTTP byte range requests" section in this Opera Page it's something to do with support for byte range requests/partial content responses. The content is treated as streaming instead. What I don't understand is: If the browser has the whole file downloaded surely it can display the duration, and surely it can seek. What I need to do on the web server to enable byte range requests (for non-static urls). Any advice would be most gratefully received.

    Read the article

  • iPhone SDK SDL_openAudio with Multitasking Support

    - by brokedid
    Hello, I'm playing audio from a Online Live RTPS Stream with ffmpeg(because Apple doesn't support rtsp live streaming). Now I would play my Stream in the background. I started a thread in the background and registered the music for Background support. When the Application is entering in Background the NSThread is paused, and then Resuming after returning from background. If I start playing a Music (MP3-Stream) in the Application which use official Apple Frameworks then when the App is entering Background both Streams are played. What can I do to fix this?

    Read the article

  • java virtual machine - how does it allocate resources?

    - by Will
    I am testing the performance of a data streaming system that supports continuous queries. This is how it works: - There is a polling service which sends data to my system. - As data passes into the system, each query evaluates based on a window of the stream at the current time. - The window slides as data passes in. My problem is this, when I add more queries to the system, I should expect the throughput to decrease because it can't cope the data rate. However, I actually observe an increase in throughput. I can't understand why this is the case and I am guessing that it's something to do with the way the JVM allocates CPU, memory etc. Can anyone shed any light to my problem?

    Read the article

  • Does the Lenovo t60p vga port support an s-video signal?

    - by Matthijs Wessels
    I just bought a new television. The problem is it turns out it doesn't have a VGA port. It does have: s-video, component, hdmi and scart. My Lenovo t60p only has vga. If have search frantically for a solution and even though it seems I have sooo many options they are all dead ends. Or I keep ending up having to buy a 100 euro box to convert the signal. However, I found that some video cards support s-video through the vga port. It says look it up in your video cards documentation. I have a Lenovo t60p laptop with a ATI MOBILITY FireGl v5250. But I can't seem to get my hands on any documentation where this is supposed to be documented. I found this website: http://forum.notebookreview.com/showthread.php?t=179529&highlight=s-video There this guy says he thinks it's in the t60 but dropped in the t61, but suggests to the guy with the t60 that it won't work. I can't really conclude anything from that. Furthermore, I am not looking for the best of the best quality. So when I found this: *http://www.amazon.com/VideoSecu-Computor-Presentation-Converter-VGA2TV/dp/B000X3FAJU/ref=pd_cp_e_3_img I woudl be quite happy with this. Except that I don't think I can order it because I don't live in the US. Can anybody give me a definite answer, to whether the vga port of my lenovo t60p ati firegl v5250 supports s-video? So that I can just by a vga to s-video cable to achieve my goal.

    Read the article

  • Les rumeurs sur le service de streaming musical par abonnement de YouTube se précisent, YouTube Music Key serait facturé à 9,99 dollars par mois

    Les rumeurs sur le service de streaming musical par abonnement de YouTube se précisent, YouTube Music Key serait facturé à 9,99 dollars par mois Depuis quelques mois des rumeurs circulaient sur YouTube et des tests potentiels d'un nouveau service qui facturerait la consommation de musique et clip vidéo sans publicité et octroierait aux abonnés la possibilité de télécharger des chansons dans leurs dispositifs mobiles. Nos confrères d'Android Police ont mené leur petite enquête sur le sujet et...

    Read the article

  • Android custom media controller using vidtry

    - by Mathias Lin
    I want to use a custom media controller in my Android app and therefore looking at the vidtry code (http://github.com/commonsguy/vidtry), especially Player.java: The sample works fine as it comes. But I want the sample to play the fixed video automatically on app startup (so I don't want to enter a URL). I added: @Override public void onStart() { super.onStart(); address.setText("/sdcard/mydata/category/1/video_agkkr6me.mp4"); go.setEnabled(true); onGo.onClick(go); } Strange thing here is that if I run the app, the audio of the video plays but the image doesn't show. Everything else works fine (progress bar, etc.). I can't figure out the difference between the manual click on the go-button and the programmatic one. I looked at the code and didn't see any difference that might occur between manual and programmatic click. I checked if any elements (esp. surface) might be hidden, but it's not. I even tried a surface.setVisibility(View.INVISIBLE); surface.setVisibility(View.VISIBLE); in case some issue with the redrawing, but no difference. The video image does show when I manually hit the go button, but just not on start up automatically.

    Read the article

< Previous Page | 122 123 124 125 126 127 128 129 130 131 132 133  | Next Page >