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  • how to upload a audio file using REST webservice in Google App Engine for Java

    - by sathya
    Am using google app engine with eclipse IDE and trying to upload a audio file. I used the File Upload in Google App Engine For Java and can able to upload the file successfully. Now am planning to use REST web service for it. I had analyzed in developers.google but i failed. Can anyone suggest me how to implement REST Web services in google app engine using Eclipse. The code google provided is shown below, // file Upload.java public class Upload extends HttpServlet { private BlobstoreService blobstoreService = BlobstoreServiceFactory.getBlobstoreService(); public void doPost(HttpServletRequest req, HttpServletResponse res) throws ServletException, IOException { Map<String, BlobKey> blobs = blobstoreService.getUploadedBlobs(req); BlobKey blobKey = blobs.get("myFile"); if (blobKey == null) { res.sendRedirect("/"); } else { res.sendRedirect("/serve?blob-key=" + blobKey.getKeyString()); }}} // file Serve.java public class Serve extends HttpServlet { private BlobstoreService blobstoreService = BlobstoreServiceFactory.getBlobstoreService(); public void doGet(HttpServletRequest req, HttpServletResponse res) throws IOException { BlobKey blobKey = new BlobKey(req.getParameter("blob-key")); blobstoreService.serve(blobKey, res); }} // file index.jsp <%@ page import="com.google.appengine.api.blobstore.BlobstoreServiceFactory" %> <%@ page import="com.google.appengine.api.blobstore.BlobstoreService" %> <% BlobstoreService blobstoreService = BlobstoreServiceFactory.getBlobstoreService(); %> <form action="<%= blobstoreService.createUploadUrl("/upload") %>" method="post" enctype="multipart/form-data"> <input type="file" name="myFile"> <input type="submit" value="Submit"> </form> // web.xml <servlet> <servlet-name>Upload</servlet-name> <servlet-class>Upload</servlet-class> </servlet> <servlet> <servlet-name>Serve</servlet-name> <servlet-class>Serve</servlet-class> </servlet> <servlet-mapping> <servlet-name>Upload</servlet-name> <url-pattern>/upload</url-pattern> </servlet-mapping> <servlet-mapping> <servlet-name>Serve</servlet-name> <url-pattern>/serve</url-pattern> </servlet-mapping> Now how to provide a rest web service for the above code. Kindly suggest me an idea.

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  • Audio doesn't work on Windows XP guest (WS 7.0)

    - by Mads
    Hi, I can't get audio to work with on a Windows XP guest running on VMware Workstation 7.0 and Ubuntu 9.10 host. Windows fails to produce any audio output and the Windows device manager says the Multimedia Audio Controller is not working properly. Audio is working fine in the host OS. When I open Multimedia Audio Controller properties it says: Device status: The drivers for this device are not installed (Code 28) If I try to reinstall the driver I get the following error message: "Cannot Install this Hardware There was a problem installing this hardware: Multimedia Audio Controller An Error occurred during the installation of the device Driver is not intended for this platform" Has anyone else experienced this problem?

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  • Bsplayer - load audio tracks from external files

    - by torran
    I have a movie file: Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : [email protected] Format settings, CABAC : Yes Format settings, ReFrames : 5 frames Muxing mode : Container [email protected] Codec ID : V_MPEG4/ISO/AVC Duration : 54mn 13s Bit rate : 3 380 Kbps Nominal bit rate : 3 459 Kbps Width : 1 280 pixels Height : 720 pixels Display aspect ratio : 16:9 Frame rate : 23.976 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.153 Stream size : 1.28 GiB (88%) Writing library : x264 core 88 r1471 1144615 Audio ID : 2 Format : AC-3 Format/Info : Audio Coding 3 Codec ID : A_AC3 Duration : 54mn 16s Bit rate mode : Constant Bit rate : 384 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Stream size : 149 MiB (10%) and additional audio files in same folder: .mp3 and .ac3. How can I load them with bsplayer? Right click-audio-audio streams is empty. If i open the movie with media players classic I can switch audio files.

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  • How do I manage the technical debate over WCF vs. Web API?

    - by Saeed Neamati
    I'm managing a team of like 15 developers now, and we are stuck at a point on choosing the technology, where the team is broken into two completely opposite teams, debating over usage of WCF vs. Web API. Team A which supports usage of Web API, brings forward these reasons: Web API is just the modern way of writing services (Wikipedia) WCF is an overhead for HTTP. It's a solution for TCP, and Net Pipes, and other protocols WCF models are not POCO, because of [DataContract] & [DataMember] and those attributes SOAP is not as readable and handy as JSON SOAP is an overhead for network compared to JSON (transport over HTTP) No method overloading Team B which supports the usage of WCF, says: WCF supports multiple protocols (via configuration) WCF supports distributed transactions Many good examples and success stories exist for WCF (while Web API is still young) Duplex is excellent for two-way communication This debate is continuing, and I don't know what to do now. Personally, I think that we should use a tool only for its right place of usage. In other words, we'd better use Web API, if we want to expose a service over HTTP, but use WCF when it comes to TCP and Duplex. By searching the Internet, we can't get to a solid result. Many posts exist for supporting WCF, but on the contrary we also find people complaint about it. I know that the nature of this question might sound arguable, but we need some good hints to decide. We're stuck at a point where choosing a technology by chance might make us regret it later. We want to choose with open eyes. Our usage would be mostly for web, and we would expose our services over HTTP. In some cases (say 5 to 10 percent) we might need distributed transactions though. What should I do now? How do I manage this debate in a constructive way?

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  • How do you setup the Audio plugin for Flowplayer?

    - by codeninja
    I'm having a bit of trouble getting the Audio player to work. Basically I want to initiate an mp3 player doing something like this <a href="path-to-my-audio.mp3" id="player" ></a> and then use the $f() call to initate the player. I've followed the instructions here (http://flowplayer.org/plugins/streaming/audio.html) This doesnt seem to be work and I'm not sure what's wrong because I'm able to play videos in this way. Thanks for your help!

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  • Core Audio on iPhone - any way to change the microphone gain (either for speakerphone mic or headpho

    - by Halle
    After much searching the answer seems to be no, but I thought I'd ask here before giving up. For a project I'm working on that includes recording sound, the input levels sound a little quiet both when the route is external mic + speaker and when it's headphone mic + headphones. Does anyone know definitively whether it is possible to programmatically change mic gain levels on the iPhone in any part of Core Audio? If not, is it possible that I'm not really in "speakerphone" mode (with the external mic at least) but only think I am? Here is my audio session init code: OSStatus error = AudioSessionInitialize(NULL, NULL, audioQueueHelperInterruptionListener, r); [...some error checking of the OSStatus...] UInt32 category = kAudioSessionCategory_PlayAndRecord; // need to play out the speaker at full volume too so it is necessary to change default route below error = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category); if (error) printf("couldn't set audio category!"); UInt32 doChangeDefaultRoute = 1; error = AudioSessionSetProperty (kAudioSessionProperty_OverrideCategoryDefaultToSpeaker, sizeof (doChangeDefaultRoute), &doChangeDefaultRoute); if (error) printf("couldn't change default route!"); error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioRouteChange, audioQueueHelperPropListener, r); if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", (int)error); UInt32 inputAvailable = 0; UInt32 size = sizeof(inputAvailable); error = AudioSessionGetProperty(kAudioSessionProperty_AudioInputAvailable, &size, &inputAvailable); if (error) printf("ERROR GETTING INPUT AVAILABILITY! %d\n", (int)error); error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioInputAvailable, audioQueueHelperPropListener, r); if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", (int)error); error = AudioSessionSetActive(true); if (error) printf("AudioSessionSetActive (true) failed"); Thanks very much for any pointers.

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  • Html5 Audio plays only once in my Javascript code.

    - by Poul
    I have a dashboard web-app that I want to play an alert sound if its having problems connecting. The site's ajax code will poll for data and throttle down its refresh rate if it can't connect. Once the server comes back up, the site will continue working. In the mean time I would like a sound to play each time it can't connect (so I know to check the server). Here is that code. This code works. var error_audio = new Audio("audio/"+settings.refresh.error_audio); error_audio.load(); //this gets called when there is a connection error. function onConnectionError() { error_audio.play(); } However the 2nd time through the function the audio doesn't play. Digging around in Chrome's debugger the 'played' attribute in the audio element gets set to true. Setting it to false has no results. Any ideas?

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  • How accurately (in terms of time) does Windows play audio?

    - by MusiGenesis
    Let's say I play a stereo WAV file with 317,520,000 samples, which is theoretically 1 hour long. Assuming no interruptions of the playback, will the file finish playing in exactly one hour, or is there some occasional tiny variation in the playback speed such that it would be slightly more or slightly less (by some number of milliseconds) than one hour? I am trying to synchronize animation with audio, and I am using a System.Diagnostics.Stopwatch to keep the frames matching the audio. But if the playback speed of WAV audio in Windows can vary slightly over time, then the audio will drift out of sync with the Stopwatch-driven animation. Which leads to a second question: it appears that a Stopwatch - while highly granular and accurate for short durations - runs slightly fast. On my laptop, a Stopwatch run for exactly 24 hours (as measured by the computer's system time and a real stopwatch) shows an elapsed time of 24 hours plus about 5 seconds (not milliseconds). Is this a known problem with Stopwatch? (A related question would be "am I crazy?", but you can try it for yourself.) Given its usage as a diagnostics tool, I can see where a discrepancy like this would only show up when measuring long durations, for which most people would use something other than a Stopwatch. If I'm really lucky, then both Stopwatch and audio playback are driven by the same underlying mechanism, and thus will stay in sync with each other for days on end. Any chance this is true?

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  • How to programmatically generate an audio podcast file with chapters and text track?

    - by adib
    Hi Anybody know how to programmatically generate audio podcast files with bookmarks that can be used in iTunes / iPod / iPhone / iPod touch? Specifically text bookmarks (bookmarks with titles) that the listener can skip to a specific point in time in the audio file. Also how to add the text transcription of the podcast's content. Even better if you have an example Cocoa code or library to write the audio file. Thanks.

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  • how do i merge two audio files and one video file in to a video file using c# ?

    - by wingdings
    i wrote a program in c# using directshow , that captures all devices' audios , and video from single device (webcam or external camera) , now that my requirement is to merge selected audio files with one video file and i can not get it done in c#. so i need a program or libraries that merges one(or several) audio file(s) and one video file and save it as an avi VIDEO file ,, both audio file and video files are in avi format.

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  • How do web servers enforce the same-origin policy?

    - by BBnyc
    I'm diving deeper into developing RESTful APIs and have so far worked with a few different frameworks to achieve this. Of course I've run into the same-origin policy, and now I'm wondering how web servers (rather than web browsers) enforce it. From what I understand, some enforcing seems to happen on the browser's end (e.g., honoring a Access-Control-Allow-Origin header received from a server). But what about the server? For example, let's say a web server is hosting a Javascript web app that accesses an API, also hosted on that server. I assume that server would enforce the same-origin policy --- so that only the javascript that is hosted on that server would be allowed to access the API. This would prevent someone else from writing a javascript client for that API and hosting it on another site, right? So how would a web server be able to stop a malicious client that would try to make AJAX requests to its api endpoints while claiming to be running javascript that originated from that same web server? What's the way most popular servers (Apache, nginx) protect against this kind of attack? Or is my understanding of this somehow off the mark? Or is the cross-origin policy only enforced on the client end?

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  • How to speed up drawing of scaled image? Audio playback chokes during window resize.

    - by Paperflyer
    I am writing an audio player for OSX. One view is a custom view that displays a waveform. The waveform is stored as a instance variable of type NSImage with an NSBitmapImageRep. The view also displays a progress indicator (a thick red line). Therefore, it is updated/redrawn every 30 milliseconds. Since it takes a rather long time to recalculate the image, I do that in a background thread after every window resize and update the displayed image once the new image is ready. In the meantime, the original image is scaled to fit the view like this: // The drawing rectangle is slightly smaller than the view, defined by // the two margins. NSRect drawingRect; drawingRect.origin = NSMakePoint(sideEdgeMarginWidth, topEdgeMarginHeight); drawingRect.size = NSMakeSize([self bounds].size.width-2*sideEdgeMarginWidth, [self bounds].size.height-2*topEdgeMarginHeight); [waveform drawInRect:drawingRect fromRect:NSZeroRect operation:NSCompositeSourceOver fraction:1]; The view makes up the biggest part of the window. During live resize, audio starts choking. Selecting the "big" graphic card on my Macbook Pro makes it less bad, but not by much. CPU utilization is somewhere around 20-40% during live resizes. Instruments suggests that rescaling/redrawing of the image is the problem. Once I stop resizing the window, CPU utilization goes down and audio stops glitching. I already tried to disable image interpolation to speed up the drawing like this: [[NSGraphicsContext currentContext] setImageInterpolation:NSImageInterpolationNone]; That helps, but audio still chokes during live resizes. Do you have an idea how to improve this? The main thing is to prevent the audio from choking.

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  • Question about the evolution of interaction paradigm between web server program and content provider program?

    - by smwikipedia
    Hi experts, In my opinion, web server is responsible to deliver content to client. If it is static content like pictures and static html document, web server just deliver them as bitstream directly. If it is some dynamic content that is generated during processing client's request, the web server will not generate the conetnt itself but call some external proram to genearte the content. AFAIK, this kind of dynamice content generation technologies include the following: CGI ISAPI ... And from here, I noticed that: ...In IIS 7, modules replace ISAPI filters... Is there any others? Could anyone help me complete the above list and elabrate on or show some links to their evolution? I think it would be very helpful to understand application such as IIS, TomCat, and Apache. I once wrote a small CGI program, and though it serves as a content generator, it is still nothing but a normal standalone program. I call it normal because the CGI program has a main() entry point. But with the recenetly technology like ASP.NET, I am not writing complete program, but only some class library. Why does such radical change happens? Many thanks.

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  • Linux based audio prodcuction tutorials

    - by thelinuxer
    I have been searching for a while for Linux based audio production tutorials. All I can find is tool based tutorials. For example I found tutorials on how to use jack, ardour, lmms ..etc. What I need is tutorials that teaches professional audio production with opensource/free tools, like those already available for protools and likes. If any one can guide me to any videos/articles available it would be highly appreciated. Thanks.

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  • Firefox pour Android introduit la « navigation en tant qu'invité » et le support de l'API Web Audio

    Firefox pour Android introduit la « navigation en tant qu'invité » et le support de l'API Web AudioA la suite de la sortie de Firefox 25, Mozilla a également publié une mise à jour de son navigateur pour les possesseurs de terminaux sous Android.Firefox pour Android hérite de quelques fonctionnalités de version desktop, notamment la prise en charge de l'API Web Audio, une spécification du W3C pour les effets audio avancés à partir de HTML5. Cette nouvelle API permettra, par exemple, aux ingénieurs...

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  • Get rid of 0.5s latency when playing audio over Bluetooth with A2DP

    - by brillout.com
    As described in the title I experience a half a second delay when playing audio over Bluetooth with A2DP. This makes watching movies not possible as the sound is not synchronised with the video. I'm not sure if the delay is caused by the Bluetooth connection, the A2PD protocol, or the A2DP implementation on my Ubuntu 12.04. Anyways, is this a normal lag? Is there a way to play audio over Bluetooth without any latency?

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  • New HP dm4 - No audio on ubuntu 11.10 64bits

    - by Haze1
    I just got a new laptop, HP dm4, and I'm having problems getting the audio to work properly on it. http://www.alsa-project.org/db/?f=7b697a35465a9f7236fb94deb9ff97fa65e55489 I tried to edit /etc/modprobe.d/alsa-base.conf and added: option snd-hda-intel model=ref this caused the audio to work, but it's muffled. I'm wondering if anybody knows what would be the correct options to get this POS to work. Thanks in advance

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  • No audio on an HP dm4

    - by Haze1
    I just got a new laptop, HP dm4, and I'm having problems getting the audio to work properly on it. http://www.alsa-project.org/db/?f=7b697a35465a9f7236fb94deb9ff97fa65e55489 I tried to edit /etc/modprobe.d/alsa-base.conf and added: option snd-hda-intel model=ref this caused the audio to work, but it's muffled. I'm wondering if anybody knows what would be the correct options to get this POS to work.

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  • Is it possible to play multiple audio streams from one "jukebox" to multiple Airport Express devices?

    - by Alex Reynolds
    I have set up a Mac mini as a jukebox that streams audio to an Airport Express in another room in the house, using the AirPlay/AirTunes feature in iTunes. I control this with the iOS Remote app, and this works great. At the present time, it looks like the Mac mini's copy of iTunes gets taken over by the Remote app, while streaming. If I set up a second Airport Express in room B, is there a way to set it up (as well as the jukebox) so that it can receive and play its own unique music stream ("stream B"), separate from what's going on at the Mac mini, or in room A, which is playing stream A? To accomplish this, I would be happy to buy a copy of Rogue Amoeba's AirFoil if it will allow sending multiple, separate audio streams from one computer to the multiple wireless bridges, while using the Remote app (or a Rogue Amoeba equivalent for iOS). However, it is unclear to me from their site documentation, whether that is possible or not. I'd prefer to give the points to an answer that solves this problem. If you don't know if it can be done, or do not think it can be done, please allow others to answer. I appreciate your help. Thanks for your advice.

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  • Setup for a live (low-latency) audio video broadcast over Wi-Fi?

    - by Majal Mirasol
    The Upgrade We are capturing audio (from mixer) and video (from a camera) from a main auditorium and passing it to separate rooms within the building. We used to have done this via manual audio/video cables and wires. We wanted to "upgrade" the system and wirelessly broadcast the stream via Wi-Fi. The Problem In our current setup (Wirecast running on A10 on a Wireless-N network), we have the problem of delay. Our streams are delayed from a minute up to five minutes on the clients (laptop/iPad/Android). This had not been a problem from the previous wired connections. Since the wireless network is local, we thought that a delay of less than a second should be achievable. Our Question And so it goes. Anybody there who has any experience for a setup that has both low latency and at the same time user-friendly to clients streaming in the program? Any recommendations would be highly appreciated. (Our current setup in on Windows 7, but setup on a dedicated Linux box is preferred, if achievable.)

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