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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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  • UTF-8 encoding problem with flash mysql and php

    - by alibhp
    Hi, As you may know, I am programming an on-line game using FLASH. I am connecting my FLASH 8 movie with MySQL database through PHP. I am doing very good in that, and I have everything working fine. The problems come when I am trying to insert (Using the INSERT SQL func) data to the database that are non-english. In other words, UTF-8 data. I red a lot of articls about that stuff and found and apply the fallowing: 1. In PHP4, you need to tell the PHP to use UTF-8 when using the xml_parser_crater() func, however, in PHP5 that is done automatically. Even though I told PHP5 to use the UTF-8 when calling the func. Adding the header to the XML sent to PHP from flash. Force the FLASH to use UTF-8 encoding in the preference options. Set the encoding in MySQL to UTF-8 (utf8_unicode_ci with InnoDB engine). I can read and insert the other language data correctly in the phpadmin as well. I did all that in my coding, and still I can't insert such data. one more strange thing is that, when I use the same link, that the FLASH using, with the XML, that the FLASH creating, on the browser (google chrome), I got the data inserted right in the database!!!!! I am about to get crazy about that stuff, What am I missing? what cause the problem? Thank you in advance.

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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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  • What is the best service/tool to put short audio clips on a website so users can click and listen im

    - by Edward Tanguay
    I'm making a foreign language flashcard website in which I want to have 100s of short 3-10 second audio files available for users to click and listen. So I am looking for a tool/service such as YouTube or Screenr.com but for audio which e.g.: allows me to easily upload multiple kinds of audio files: mp3, wav, etc. easy to manage them online (delete, replace) has a simple, small player (e.g. flash) that integrates nicely into any site

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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 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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  • 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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  • How to make TXMLDocument (with the MSXML Implementation) always include the encoding attribute?

    - by Fabricio Araujo
    I have legacy code (I didn't write it) that always included the encoding attribute, but recompiling it to D2010, TXMLDocument doesn't include the enconding anymore. Because the XML data have accented characters both on tags and data, TXMLDocument.LoadFromFile simply throws EDOMParseErros saying that an invalid character is found on the file. Relevant code: Doc := TXMLDocument.Create(nil); try Doc.Active := True; Doc.Encoding := XMLEncoding; RootNode := Doc.CreateElement('Test', ''); Doc.DocumentElement := RootNode; <snip> //Result := Doc.XMl.Text; Doc.SaveToXML(Result); // Both lines gives the same result On older versions of Delphi, the following line is generated: <?xml version="1.0" encoding="ISO-8859-1"?> On D2010, this is generated: <?xml version="1.0"?> If I change manually the line, all works like always worked in the last years. UPDATE: XMLEncoding is a constant and is defined as follow XMLEncoding = 'ISO-8859-1';

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  • How can I avoid encoding mixups of strings in a C/C++ API?

    - by Frerich Raabe
    I'm working on implementing different APIs in C and C++ and wondered what techniques are available for avoiding that clients get the encoding wrong when receiving strings from the framework or passing them back. For instance, imagine a simple plugin API in C++ which customers can implement to influence translations. It might feature a function like this: const char *getTranslatedWord( const char *englishWord ); Now, let's say that I'd like to enforce that all strings are passed as UTF-8. Of course I'd document this requirement, but I'd like the compiler to enforce the right encoding, maybe by using dedicated types. For instance, something like this: class Word { public: static Word fromUtf8( const char *data ) { return Word( data ); } const char *toUtf8() { return m_data; } private: Word( const char *data ) : m_data( data ) { } const char *m_data; }; I could now use this specialized type in the API: Word getTranslatedWord( const Word &englishWord ); Unfortunately, it's easy to make this very inefficient. The Word class lacks proper copy constructors, assignment operators etc.. and I'd like to avoid unnecessary copying of data as much as possible. Also, I see the danger that Word gets extended with more and more utility functions (like length or fromLatin1 or substr etc.) and I'd rather not write Yet Another String Class. I just want a little container which avoids accidental encoding mixups. I wonder whether anybody else has some experience with this and can share some useful techniques. EDIT: In my particular case, the API is used on Windows and Linux using MSVC 6 - MSVC 10 on Windows and gcc 3 & 4 on Linux.

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  • How can i make changes to this file Encoding?

    - by SuperUserMan
    I have these 3 files 21/08/2014 07:15 PM 122 Tw2AWK.csv 21/08/2014 07:15 PM 125 Tw2Notepad.csv 21/08/2014 07:15 PM 119 Tw2REPL.csv C:\myfilesfile Tw2AWK.csv TwREPL.csv Tw2Notepad.csv Tw2AWK.csv; UTF-8 Unicode text, with CRLF line terminators Tw2REPL.csv; UTF-8 Unicode text Tw2Notepad.csv; UTF-8 Unicode (with BOM) text, with CRLF line terminators HEX of these files is as follows C:\myfilesxxd -p Tw2REPL.csv 0a222344656c686947616e675261706520776173206120736d616c6c2069 6e636964656e7420746f2023536d616c6c5261706973744a6169746c6579 20646e61696e6469612e636f6d2f696e6469612f7265706f72742d69e280 a6207069632e747769747465722e636f6d2f6762565070776637744f22 C:\myfilesxxd -p Tw2AWK.csv 0d0a222344656c686947616e675261706520776173206120736d616c6c20 696e636964656e7420746f2023536d616c6c5261706973744a6169746c65 7920646e61696e6469612e636f6d2f696e6469612f7265706f72742d69e2 80a6207069632e747769747465722e636f6d2f6762565070776637744f22 0d0a C:\myfilesxxd -p Tw2Notepad.csv efbbbf0d0a222344656c686947616e675261706520776173206120736d61 6c6c20696e636964656e7420746f2023536d616c6c5261706973744a6169 746c657920646e61696e6469612e636f6d2f696e6469612f7265706f7274 2d69e280a6207069632e747769747465722e636f6d2f6762565070776637 744f220d0a I want Tw2REPL.csv to look like Tw2Notepad.csv How can I do it? NOTE: I have do this all via command line (batch) . I can use any 3rd party standalone exe's though. I am on Windows XP Please help, its very important for me

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  • Determining default character set of platform in Java

    - by Anand
    I am programming in Java I have the code as: byte[] b = test.getBytes(); In the api it is specified that if we do not specify character encoding it takes the default platform character encoding. What is meant by "default platform character encoding" ? Does it mean the Java encoding or the OS encoding ? If it means OS encoding the how can i check the default character encoding of Windows and Linux ? Is there anyway we can get the default character encoding using command line ?

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  • iphone audio streaming

    - by mobapps99
    Hi , i'm developing an application which uses audio streaming. For streaming audio from internet i'm using the AudioStreamer class. The audio streamer has four state isPlaying, isPaused ,isWaiting, and isIdle . My problem is that when the audio streamer is in the state "isWaiting" and at that time if i get a phone call Audio queue fails giving the error "Audio queue start failed." Any has solution for this? help....

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