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• Synced audio ouput on multiple machines? VLC? hardware solutions?

  - by zimmer62

  I'm wondering if there is any software or hardware solutions to synced audio or audio and video across multiple computers or devices on a network. I've seen Sonos, and it might be a good solution, but it's also a very expensive solution. I'd like to be able to play something with realtime audio output on one PC, but hear it on speakers throughout the house, being it the home theater receiver, or another computer in another room. I saw a solution using the apple iport express, but the latency was unacceptable for anything other than just music. I'd like to avoid running audio wires with baluns to a bunch of amplifiers scattered all over the place when I have cat5 run everywhere. Is anyone familiar with using this kind of process for whole home audio? The latency is a big deal for me, if I've got video attached to the sound (e.g. watching a hockey game)

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• What is the best way to watch internet movies (Hulu, etc.) on my TV?

  - by Jeffrey

  I have a TV in a room that can only get wireless access to my home network. Without having to drag one of my PCs into the room, what is the best (cheapest) way to watch online movies?

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• Tools to track website’s video source [closed]

  - by Stan

  I've been using Fiddler2 or Orbit grab video plugin to track this site and try to find the video source link. But both don't work. Can anyone suggest a tool that can retrieve the video for offline viewing. Thanks.

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• How to encode video to use HTML 5 video tag? [closed]

  - by exquisitor

  I'm trying to use <video> tag feature. I've encoded my file with ffmpeg: ffmpeg -i 1.vob -f ogg -vcodec libtheora -b 800k -g 300 -acodec libvorbis -ab 128k out.ogv I've placed this text in index.html: <video src="out.ogv" controls></video> I've loaded this files to nginx document root and opened index.html in Firefox 3.5. I see the first frame, see controls, but can't play video. This video plays good even from my server. How should I encode video to view it in browser? UPD: If I start playing video from the middle, everything works fine.

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• Drawing a rectangle on a video in C#

  - by Haxed

  Hi I want to draw a rectangle on a video stream(web cam video or loaded saved video) that I have streaming on a picture box. This is a C# application and I am using EmguCV 2.1.0.0. I have been successful in displaying the video stream on the picturebox in the form. Can I use Emgucv to draw on the video or should I use something else ? Can I use Dshownet or something like that ? Thanks for taking the time to read this. Many Thanks

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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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• 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?

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• 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.

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• wowza vs Flash Media Server (FMS / FMIS) - ease of integration with ASP.Net

  - by alchemical

  We're creating a web site offering one to many video chat and trying to decide on which of these streaming servers to go with. Looking at around 256kbps live streams, hoping to achieve at least 1000 simultaneous streams on one 8-core server. Wowza is cheaper (1k vs 5k for FMS), and appears to be used successfully by many sites (StreamLive, Justin.TV, etc.). However, some people have expressed that it may be more difficult to work with. I.e. fine-tuning it, less documentation, integration with ASP.Net code, etc. Wondering if anyone with real-world experience with either of these servers can advise regarding how easy or difficult to use and integrate they are for a site like this. Also wondering if there is any performance difference (lag, etc.).

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• 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

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• iPhone Live Video Stream Media Player

  - by happyhammer83

  I'm hoping to make an app that streams live video that has a view placed on top with labels and a button on it. From my research and testing of the http video streaming feature (available since iPhone 3.0 OS), it seems that you create a webview that points to the index html that contains the converted video stream, and this displays as a quicktime video in the app. This means that I don't have control over the Media Player that is opened. Does anyone know how you can control this? I know that the Apple's MoviePlayer sample code shows you how to place views on top of a MediaPlayer video, but how can this be done with a http live stream? Thanks in advance.

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• 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.

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• 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!!!

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• 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.

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• 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)?

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• 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>...

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• How to split movie and play parts to look as a whole?

  - by luksow

  I'm writing software which is demonstraiting video on demand service. One of the feature is something similiar to IIS Smooth Streaming - I want to adjust quality to the bandwith of the client. My idea is, to split single movie into many, let's say - 2 seconds parts, in different qualities and then send it to the client and play them. The point is that for example first part can be in very high quality, and second in really poor (if the bandwith seems to be poor). The question is - do you know any software that allows me to cut movies precisly? For example ffmpeg splits movies in a way that join is visible and really annoying (seconds are the measure of precision). I use qt + phonon as a player if it matters. Or maybe you know any better way to provide such feature, without splitting movie into parts?

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• 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.

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• 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?

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• 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?

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• 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...

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• A PHP script to stream internet radio?

  - by Honus Wagner

  I've been searching and searching and I haven't yet come up with a solution to host my own streaming audio player. I'm looking for a way to host an internet radio player that connects to whatever streams I enter in and plays them. I'm not looking to play my MP3s or anything like that. I'm looking to play content from 181.fm or 1Club.fm, for example. I'd even settle for ShoutCast-only streams. I've been to www.wavestreaming.com but it didnt work for me. I'm guessing its because in the very first box where you enter your website url, it leads in for you: http//www. then you fill in the rest. My site is https:// and does not contain a www. in the URL. I'm guessing that has something to do with it. Any links, suggestions for search topics, or even a brief technical overview of what I should be looking into would be greatly appreciated. Thanks for your time.

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• Play .ts video file on Android?

  - by user359519

  I am pretty new at streaming video, so please bear with me. :) I am trying to port an m3u8 stream over from iPhone to Android. Looking in the m3u8 feed, I found some .ts files. From what I can tell, .ts files are, themselves, wrappers that contain the video stream (Elementary Stream). Is it possible to play a .ts file in Android? (The docs only list 3gp and mp4 as supported formats.) Is there a way to extract the Elementary Stream and just process the video feed? If that is in 3gp or mp4, I should be ok. Will Stagefright handle .ts? Is Stagefright even available? I read that there are/were some problems with it. (As a further caveat, I am not getting much help from my server guys. They are pushing for a Flash player solution, including a proprietary player. They will not provide me with a 3gp or an mp4 feed, but I'm hoping I can find that in the .ts file.) I'm open to other suggestions. Thanks for your patience with this newbie. :)

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• From ASPX to WCF

  - by Barguast

  I'm hoping someone can advise me on how to solve my networking scenario. Both the client and server are to be C# / .NET based. I basically want to invoke some kind of web service from my client in order to retrieve both binary data (e.g. files) and serialised objects and lists of objects (e.g. database query results). At the moment, I'm using ASPX pages, using the query string to provide parameters and I get back either the binary data, or the binary data of the serialised messages. This affords me a lot of flexbility, and I can choose how to transmit the data, perform simulatanous requests, cancel ongoing requests, etc. Since I can control the serialised format, I can also deserialise lists of objects as they are received which is crucial. My problem isn't a problem as such, but this feels a little hack-ish and I can't help but wonder if there are better ways to go about it. I'm considering moving on to WCF or perhaps another technology to see if it helps. However, I need to know if it helps with my scenarios above that is; Can a WCF method return a list of objects, and can the client receive the items of this list as they arrive as opposed to getting the entire list on completion (i.e. streaming). Does anyone know of any examples of this? Am I likely to get any performance benefits from this? I don't know how well ASPX pages are tuned for this, as it surely isn't their primary purpose. Are there any other approaches I should consider? Thanks for your time spent reading this. I hope you can help.

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• moving audio over a local network using GStreamer

  - by James Turner

  I need to move realtime audio between two Linux machines, which are both running custom software (of mine) which builds on top of Gstreamer. (The software already has other communication between the machines, over a separate TCP-based protocol - I mention this in case having reliable out-of-band data makes a difference to the solution). The audio input will be a microphone / line-in on the sending machine, and normal audio output as the sink on the destination; alsasrc and alsasink are the most likely, though for testing I have been using the audiotestsrc instead of a real microphone. GStreamer offers a multitude of ways to move data round over networks - RTP, RTSP, GDP payloading, UDP and TCP servers, clients and sockets, and so on. There's also many examples on the web of streaming both audio and video - but none of them seem to work for me, in practice; either the destination pipeline fails to negotiate caps, or I hear a single packet and then the pipeline stalls, or the destination pipeline bails out immediately with no data available. In all cases, I'm testing on the command-line just gst-launch. No compression of the audio data is required - raw audio, or trivial WAV, uLaw or aLaw encoding is fine; what's more important is low-ish latency.

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