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• Mobile opera have background sound support?

  - by Mark

  I make browser/html/js games. One of my biggest pains in the arse is the lack of background sound support in mobile safari. This lack of support makes high value games pretty much impossible. Does anyone know if opera mini supports html5 audio, or any mobile browser for that matter. If not, what are some alternatives methods.

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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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• AVFoundation: Video to OpenGL texture working - How to play and sync audio?

  - by j00hi

  I've managed to load a video-track of a movie frame by frame into a OpenGL texture with AVFoundation. I followed the steps described in the answer here: iOS4: how do I use video file as an OpenGL texture? and took some code from the GLVideoFrame sample from WWDC2010 which can be downloaded here: http://bit.ly/cEf0rM How do I play the audio-track of the movie synchronously to the video. I think it would not be a good idea to play it in a separate player, but to use the audio-track of the same AVAsset. AVAssetTrack* audioTrack = [[asset tracksWithMediaType:AVMediaTypeAudio] objectAtIndex:0]; I retrieve a videoframe and it's timestamp in the CADisplayLink-callback via CMSampleBufferRef sampleBuffer = [self.readerOutput copyNextSampleBuffer]; CMTime timestamp = CMSampleBufferGetPresentationTimeStamp( sampleBuffer ); where readerOutput is of type AVAssetReaderTrackOutput* How to get the corresponding audio-samples? And how to play them? Edit: I've looked around a bit and I think, best would be to use AudioQueue from the AudioToolbox.framework using the approach described here: AVAssetReader and Audio Queue streaming problem There is also an audio-player in the AVFoundation: AVAudioPlayer. But I don't know exactly how I should pass data to it's initWithData-initializer which expects NSData. Furthermore I don't think it's the best choice for my case because a new AVAudioPlayer-instance would have to be created for every new chunk of audio samples, as I understand it. Any other suggestions? What's the best way to play the raw audio samples which i get from the AVAssetReaderTrackOutput?

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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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• playing only part of a sound using FMOD

  - by carneades

  I'm trying to play only part of a sound using FMOD, say frames 50000-100000 of a 200000 frame file. I have found a couple of ways to seek forward (i.e. to start playback at frame 50000) but I have not found a way to make sure the sound stops playing at 100000. Is there any way FMOD can natively do this without having to add lbsndfile or the like into the picture? I should also mention that I am using the streaming option. I have to assume that these sounds are arbitrarily large and cannot be comfortably/quickly loaded into memory.

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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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• Java and gstreamer-java initialisation error

  - by Mark

  I am building a small app which will play streaming audio from the internet in java (mainly internet radio stations). I have decided to use the gstreamer-java library for the sound, which uses JNA. I would like to include a check in the code, to see whether the gstreamer library has been initialised. When I have left the "Gst.init()" code out (to mimic when the library has not been initialised correctly), the application throws out the following messages: (process:21888): GLib-GObject-CRITICAL **: /build/buildd/glib2.0-2.22.3/gobject/gtype.c:2458: initialization assertion failed, use IA__g_type_init() prior to this function (process:21888): GLib-CRITICAL **: g_once_init_leave: assertion `initialization_value != 0' failed The app calls the gstreamer-java library. The error messages appear but the thread continues to run, hogging the CPU. Is there any way to catch the error or to add a check to prevent it from happening? An alternative would be to put the "Gst.init()" in the main class, but I am not sure if this would always guarantee the gstreamer library is initialised.

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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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• 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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• issue getting dynamic Config parameter in Grails taglib

  - by Mick Knutson

  I have a dynamic config parameter I want to get like: String srcProperty = "${attrs ['src']}.audio" + ((attrs['locale'])? "_${attrs['locale']}" : '') assert srcProperty == "prompt.welcomeMessageOverrideGreeting.audio" where my config has: prompt{ welcomeMessageOverrideGreeting { audio = "/en/someFileName.wav" txt = "Text alternative for /en/someFileName.wav" audio_es = "/es/promptFileName.wav" txt_es = "Texto alternativo para /es/someFileName.wav" } } While this works fine: String audio = "${config.prompt.welcomeMessageOverrideGreeting.audio}" and: assert "${config.prompt.welcomeMessageOverrideGreeting.audio}" == "/en/someFileName.wav" I can not get this to work: String audio = config.getProperty("prompt.welcomeMessageOverrideGreeting.audio")

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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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• The fastest way to encode image+audio for Youtube from command line?

  - by Pavel Vlasov

  I have an mp3 and image and I want to make a simple clip to upload onto Youtube. Is there a fast solution? If video formats are so bad designed, then maybe it is possible to use a prerendered video-only clip? This works good except it takes as much time as the audio lasts: ffmpeg -loop_input -r ntsc -i "%IMAGE%" -i "%AUDIO%" -r 1 -acodec copy -shortest -re -force_fps "%VIDEO%" This takes a second but results in a black screen video that is successfully played by a desktop video player but not acceptable by Youtube: ffmpeg -i "%IMAGE%" -i "%AUDIO%" -acodec copy "%VIDEO%" Windows 7. Preserving audio quality is preferred over video quality.

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• How do I merge MP4 files without audio going out of sync?

  - by djangofan

  Is there a tool I can use that can merge MP4 files without throwing the audio out of sync? I generated some MP4 files from a DVD using AVIDemux but whatever tool I try to use always ends up throwing the audio out of sync with the video. The further you get into the video the further off-sync the audio is. By themselves the MP4/AAC videos have perfect audio-video sync. later tonight i might try http://www.headbands.com/gspot/ to examine the file before and after to see if anything changed in the media format.

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• How do I add another audio stream to an MP4 file?

  - by RandomEngy

  I've got an MP4 video file and I want to add another AAC audio track to it. I've tried YAMB and MeGUI (frontends for MP4Box) and it plays correctly in Zoom Player, but it picks the wrong track in WMP and plays both at once in Quicktime. I think this might have to do with designating the default audio track somehow. Does anyone know how to specify the default audio track with YAMB/MeGUI or know of another way of adding a track to an MP4 file?

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• How to export or view audio file references in a PDF?

  - by redshift

  I have a an interactive PDF file that is over 90+ pages long. Each page is a map with city names that contains a Spanish pronunciation of that city in a .wav file. I'd say there are about 10-15 audio files for each map which comes out to 1000+ audio files. Is there a way to extract/export a list of the sound file names associated with each map? I tried to save the PDF to an HTML file, but it only exported images and text, and because the audio files were embedded in the PDF, the file names did not carry over to the HTML file. Any other ideas? I need to see what audio file goes with what map/page.

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• how to prevent MPMoviePlayer controls from hiding

  - by huevos de oro

  I am trying to implement a custom MPMoviePlayer to play mp3 audio. I have got it working in portrait mode along with an overlay window over the native controls - thanks to other stackoverflow posts. The current issue is the song progress control shows up when the media window opens (blue bar taking up the first 40 odd pixels), but seems to disappear when the song starts leaving a white bar. It will then re-appear when touching the area, so functionally works fine. I would like to find a way to ensure the controls always stay visible but have not found an appropriate property in the reference. Ideally I would like to have my custom control to replace the default, more because I would like to change the position that the look and feel. This being said, I understand it is not possible as the current position in the song from a MPMoviePlayer cannot be accessed.

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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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• How do I play back a WAV in ActionScript?

  - by Jeremy White

  Please see the class I have created at http://textsnip.com/51013f for parsing a WAVE file in ActionScript 3.0. This class is correctly pulling apart info from the file header & fmt chunks, isolating the data chunk, and creating a new ByteArray to store the data chunk. It takes in an uncompressed WAVE file with a format tag of 1. The WAVE file is embedded into my SWF with the following Flex embed tag: [Embed(source="some_sound.wav", mimeType="application/octet-stream")] public var sound_class:Class; public var wave:WaveFile = new WaveFile(new sound_class()); After the data chunk is separated, the class attempts to make a Sound object that can stream the samples from the data chunk. I'm having issues with the streaming process, probably because I'm not good at math and don't really know what's happening with the bits/bytes, etc. Here are the two documents I'm using as a reference for the WAVE file format: http://www.lightlink.com/tjweber/StripWav/Canon.html https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ Right now, the file IS playing back! In real time, even! But...the sound is really distorted. What's going on?

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