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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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• Dedicated Servers: Is one better then two for LAMP pseudo HA setup? [closed]

  - by bikedorkseattle

  Possible Duplicate: How to find web hosting that meets my requirements? I know there are zillions of commentary about hosting out there, but I haven't read much about this. Our current well known host is having too many problems, the hardware we are on it subpar, and I'm ready to leave. A day of downtime can cost as much as our monthly hosting bill. A month of bad performance is just killing us right now, user and google wise. I'm wondering about running two dedicated boxes for LAMP, one running as the primary Nginx/Apache (proxy pass), and the other as the MySQL box. Running a single box scares the bejesus out of me because who knows how long it will take anyone to fix a raid card or whatever. The idea is to set this up using some sort of failover system using pacemaker and heartbeat. If one server goes down the other can take over for the other running both web and db. There are some good articles over at Linode about this. I have a few DBs that are 1GB+ and would like to load them into memory. Because of this, I'm shying away from a Linode HA setup because for the price I could do it with two dedicated like I described. Am I mad or an idiot? What are people out there doing for pseodu high availability good performance setups under $400/month? I'm a webmaster; I do a lot of things none of it that well :)

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• How do I produce "enjoyably" random, as opposed to pseudo-random?

  - by Hilton Campbell

  I'm making a game which presents a number of different kinds of puzzles in sequence. I choose each puzzle with a pseudorandom number. For each puzzle, there are a number of variations. I choose the variation with another pseudorandom number. And so on. The thing is, while this produces near-true randomness, this isn't what the player really wants. The player typically wants what they perceive to be and identify as random, but only if it doesn't tend to repeat puzzles. So, not really random. Just unpredictable. Giving it some thought, I can imagine hacky ways of doing it. For example, temporarily eliminating the most recent N choices from the set of possibilities when selecting a new choice. Or assigning every choice an equal probability, reducing a choice's probability to zero on selection, and then increasing all probabilities slowly with each selection. I assume there's an established way of doing this, but I just don't know the terminology so I can't find it. Anyone know? Or has anyone solved this in a pleasing way?

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• Does Google penalize pseudo-duplicate pages for different locations?

  - by mikewowb

  My compony's site's home page was not specificly optimized to any location. Now, I am planning to optimize it to Boston, and create ten or so other landing pages for other locations we serve. If we made these new pages by copying the original Boston one and changing the location's name (s/Boston/Montreal/), would Google consider them as duplicate pages and penalize us? What is the best practice for this?

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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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• 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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• 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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• Segmentation fault while feeding in an mpeg file through ffmpeg

  - by angel6

  Hi, I've set up FFserver as the streaming server. I'm trying to feed in an mpeg file. But it comes up with a segmentation fault. Does anyone know how to fix this? The following is the command-line output I get $ ./ffmpeg -i test1.mpg http://localhost:8090/feed1.ffm FFmpeg version SVN-r22945, Copyright (c) 2000-2010 the FFmpeg developers built on Apr 22 2010 19:18:45 with gcc 4.4.1 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-pthreads --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libx264 --enable-libxvid --enable-x11grab libavutil 50.14. 0 / 50.14. 0 libavcodec 52.66. 0 / 52.66. 0 libavformat 52.61. 0 / 52.61. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg @ 0xab0c420]max_analyze_duration reached Input #0, mpeg, from 'test1.mpg': Duration: 00:00:20.96, start: 0.768300, bitrate: 269 kb/s Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 160x120 [PAR 1:1 DAR 4:3], 104857 kb/s, 30 fps, 30 tbr, 90k tbn, 30 tbc Stream #0.1[0x1c0]: Audio: mp2, 32000 Hz, 2 channels, s16, 64 kb/s Output #0, ffm, to 'http://localhost:8090/feed1.ffm': Metadata: encoder : Lavf52.61.0 Stream #0.0: Audio: mp2, 22050 Hz, 1 channels, s16, 48 kb/s Stream #0.1: Video: mpeg1video, yuv420p, 160x128, q=2-31, 40 kb/s, 1000k tbn, 50 tbc Stream #0.2: Audio: libmp3lame, 22050 Hz, 1 channels, s16, 64 kb/s Stream #0.3: Video: msmpeg4, yuv420p, 352x240, q=2-31, 256 kb/s, 1000k tbn, 15 tbc Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Stream #0.1 -> #0.2 Stream #0.0 -> #0.3 Press [q] to stop encoding Segmentation fault

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• How to detect when video is buffering?

  - by Leon

  Hi guys, my question today deals with Flash AS3 video buffering. (Streaming or Progressive) I want to be able to detect when the video is being buffered, so I can display some sort of animation letting the user know to wait just a little longer. Currently my video will start up, hold on frame 1 for 3-4 secs then play. Kinda giving the impression that the video is paused or broken :( Update Thanks to iandisme I believe I'm faced in the right direction now. NetStatusEvent from livedocs. It seems to me that the key status to be working in is "NetStream.Buffer.Empty" so I added some code in there to see if this would trigger my animation or a trace statement. No luck yet, however when the Buffer is full it will trigger my code :/ Maybe my video is always somewhere between Buffer.Empty and Buffer.Full that's why it won't trigger any code when I test case for Buffer.Empty? Current Code public function netStatusHandler(event:NetStatusEvent):void { // handles net status events switch (event.info.code) { case "NetStream.Buffer.Empty": trace("¤¤¤ Buffering!"); //<- never traces addChild(bufferLoop); //<- doesn't execute break; case "NetStream.Buffer.Full": trace("¤¤¤ FULL!"); //<- trace works here removeChild(bufferLoop); //<- so does any other code break; case "NetStream.Buffer.Flush": trace("¤¤¤ FLUSH!"); //Not sure if this is important break } }

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• MPMoviePlayerContentPreloadDidFinishNotification seems more reliable than MPMoviePlayerLoadStateDidChangeNotification

  - by user567889

  I am streaming small movies (1-3MB) off my website into my app. I have a slicehost webserver, I think it's a "500MB slice". Not sure off the top of my head how this translates to bandwidth, but I can figure that out later. My experience with MPMoviePlayerLoadStateDidChangeNotification is not very good. I get much more reliable results with the old MPMoviePlayerContentPreloadDidFinishNotification If I get a MPMoviePlayerContentPreloadDidFinishNotification, the movie will play without stuttering, but if I use MPMoviePlayerLoadStateDidChangeNotification, the movie frequently stalls. I'm not sure which load state to check for: enum { MPMovieLoadStateUnknown = 0, MPMovieLoadStatePlayable = 1 << 0, MPMovieLoadStatePlaythroughOK = 1 << 1, MPMovieLoadStateStalled = 1 << 2, }; MPMovieLoadStatePlaythroughOK seems to be what I want (based on the description in the documentation): MPMovieLoadStatePlaythroughOK Enough data has been buffered for playback to continue uninterrupted. Available in iOS 3.2 and later. but that load state NEVER gets set to this in my app. Am I missing something? Is there a better way to do this?

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