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• fmod VS2008 unresolved externals in dependant project

  - by Tom J Nowell

  Im currently trying to use the latest stable fmod ex in my project. I have a main executable in a project called engine4, and a project named DX9Platform in the solution as well which ti depends on. All the fmod code is in this DX9Platform project, which generates a lib file. DX9Platform includes fmodex_vc.lib and builds fine. However buildign Engien4 results in unresolved external symbol messages referencing files that use fmod in the DX9Platform project I have tried adding fmodex_vc.lib to the Engine4 project, with no success, how do I fix this? Heres the linker output: 3>------ Build started: Project: Engine4, Configuration: Release Direct3D9 Win32 ------ 3>Linking... 3>DX9PlatformLib.lib(CFmodSound.obj) : error LNK2001: unresolved external symbol _FMOD_System_Create 3>DX9PlatformLib.lib(CFmodSound.obj) : error LNK2001: unresolved external symbol "public: enum FMOD_RESULT __thiscall FMOD::System::createSound(char const *,unsigned int,struct FMOD_CREATESOUNDEXINFO *,class FMOD::Sound * *)" (?createSound@System@FMOD@@QAE?AW4FMOD_RESULT@@PBDIPAUFMOD_CREATESOUNDEXINFO@@PAPAVSound@2@@Z) 3>DX9PlatformLib.lib(CFmodSound.obj) : error LNK2001: unresolved external symbol "public: enum FMOD_RESULT __thiscall FMOD::System::getVersion(unsigned int *)" (?getVersion@System@FMOD@@QAE?AW4FMOD_RESULT@@PAI@Z) 3>DX9PlatformLib.lib(CFmodSound.obj) : error LNK2001: unresolved external symbol "public: enum FMOD_RESULT __thiscall FMOD::System::init(int,unsigned int,void *)" (?init@System@FMOD@@QAE?AW4FMOD_RESULT@@HIPAX@Z) 3>DX9PlatformLib.lib(CFModAudioObject.obj) : error LNK2001: unresolved external symbol "public: enum FMOD_RESULT __thiscall FMOD::System::playSound(enum FMOD_CHANNELINDEX,class FMOD::Sound *,bool,class FMOD::Channel * *)" (?playSound@System@FMOD@@QAE?AW4FMOD_RESULT@@W4FMOD_CHANNELINDEX@@PAVSound@2@_NPAPAVChannel@2@@Z) 3>DX9PlatformLib.lib(CFModAudioObject.obj) : error LNK2001: unresolved external symbol "public: enum FMOD_RESULT __thiscall FMOD::Channel::getPaused(bool *)" (?getPaused@Channel@FMOD@@QAE?AW4FMOD_RESULT@@PA_N@Z) 3>DX9PlatformLib.lib(CFModAudioObject.obj) : error LNK2001: unresolved external symbol "public: enum FMOD_RESULT __thiscall FMOD::Channel::setPaused(bool)" (?setPaused@Channel@FMOD@@QAE?AW4FMOD_RESULT@@_N@Z) 3>DX9PlatformLib.lib(CFModAudioObject.obj) : error LNK2001: unresolved external symbol "public: virtual class IAudioObject * __thiscall CFModAudioObject::LoadFile(char const *)" (?LoadFile@CFModAudioObject@@UAEPAVIAudioObject@@PBD@Z) 3>D:\media\desktop\engine4\Engine4\Output\Release Direct3D9\Engine4.exe : fatal error LNK1120: 8 unresolved externals 3>Build log was saved at "file://d:\media\desktop\engine4\Engine4\Engine4\intermediate\Release Direct3D9\BuildLog.htm" 3>Engine4 - 9 error(s), 0 warning(s) ========== Build: 1 succeeded, 1 failed, 0 up-to-date, 1 skipped ==========

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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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• Can the Flash CS4 [embed] tag be made to export assets to frame 2 rather than frame 1?

  - by Tim Knauf

  We're working on a Flash CS4 project where the main .fla file has ballooned in size and 'Publish' is taking forever. I suspect a large amount of the size (and at least some of the compile time) is due to the quantity of audio symbols in the library. I would love to remove this unnecessary bloat from the .fla file. I've experimented with removing an audio symbol from the library and using the [embed] metadata tag instead, like so: [Embed(source="audio/music/EndOfLevelDitty.mp3")] public var EndOfLevelDitty:Class The resulting published file works perfectly, but there is a problem. Our game uses a preloader on the first frame of the timeline, so all other classes need to be exported in frame 2 (as set in Publish Settings ActionScript 3.0 Settings). So a size report normally begins like this: Frame # Frame Bytes Total Bytes Scene ------- ----------- ----------- ---------------- 1 284515 284515 Scene 1 2 5485305 5769820 (AS 3.0 Classes Export Frame) However, if I use an [embed] tag on a small sound, my size report is now: Frame # Frame Bytes Total Bytes Scene ------- ----------- ----------- ---------------- 1 363320 363320 Scene 1 2 5407240 5770560 (AS 3.0 Classes Export Frame) As you can see, the embedded sound has been exported into frame 1 rather than frame 2. If I were to embed all sounds in this manner, the size of frame 1 would grow to be huge, and users would be looking at a white screen for ages before the preloader frame even loaded. So my question is this: can I use an [embed] tag but have the embedded asset export in frame 2 instead of frame 1? Project constraints: Our team composition means we can't change to pure Flex at this stage. The compiled .swf needs to be 'all in one', so we can't split the preloader into a separate file, and we can't access external resources. Edit: I'd also settle for having the audio in an embedded library SWC, but there seems to be no way to make that embed in frame 2 either; it always ends up in frame 1.

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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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• Selectively replacing words outside of tags using regular expressions in PHP?

  - by Gary Willoughby

  I have a paragraph of text and i want to replace some words using PHP (preg_replace). Here's a sample piece of text: This lesson addresses rhyming [one]words and ways[/one] that students may learn to identify these words. Topics include learning that rhyming words sound alike and these sounds all come from the [two]ending of the words[/two]. This can be accomplished by having students repeat sets of rhyming words so that they can say them and hear them. The students will also participate in a variety of rhyming word activities. In order to demonstrate mastery the students will listen to [three]sets of words[/three] and tell the teacher if the words rhyme or not. If you notice there are many occurances of the word 'words'. I want to replace all the occurances that don't occur inside any of the tags with the word 'birds'. So it looks like this: This lesson addresses rhyming [one]words and ways[/one] that students may learn to identify these birds. Topics include learning that rhyming birds sound alike and these sounds all come from the [two]ending of the words[/two]. This can be accomplished by having students repeat sets of rhyming birds so that they can say them and hear them. The students will also participate in a variety of rhyming word activities. In order to demonstrate mastery the students will listen to [three]sets of words[/three] and tell the teacher if the birds rhyme or not. Would you use regular expressions to accomplish this? Can a regular expression accomplish this?

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• Python script, runs well, but not perfectly, debugging help.

  - by S1syphus

  What it does (sort of)... or is meant to, the script reads from a csv file that contains information on sound files and create a play list exactly 60 minutes long. An example csv, contains: their title, duration (in seconds), minium total time to be played (in minutes) An example is: Soundfoo,120,10 Soundbar,30,6 Sounddev,60,20 Soundrandom,15,8 The script works out the minimum instances of plays, take 'Soundfoo' for example, the length of each sample is 120 seconds and the minimum time to be played is 10 minutes, so basic maths 10*60/120 gives the number of instances the song is to be played, in this case 5. It is meant to take minimum number of instances and spread out equally from each other; so there will never be a period where for example Soundbar is played twice in a row. Then if the minium instances of each song has been used, and there is still time with in the 60 min, how is it possible to tell it to go back and fill the time by selecting each sound and including it till the 60 min is filled while remaining sparsely populated. Heres the issue(s)! The script fails to calculate the actual time require to play all the sounds in a file and the total time of the playlist, the thing is tho it doesn't get it wrong all the time maybe 3/5 times, even if I run it on the same csv file it will give me different answers. Here is the file I shall run the script on e for sake of ease to see the issue: Sound1,60,10 Sound2,60,10 Sound3,60,10 Sound4,60,10 Sound5,60,10 Sound6,60,10 I'll do it three times and post the results: 1 Required playtime in minutes: 60 Actual time in minutes to play all required ads: 62 Total playtime in minutes: 62.0 2 Required playtime in minutes: 60 Actual time in minutes to play all required ads: 71 Total playtime in minutes: 71.0 3 Required playtime in minutes: 60 Actual time in minutes to play all required ads: 60 Total playtime in minutes: 60.0 Relevant Code: pastebin.com/demkBXk6 And finally... in context: http://pastebin.com/demkBXk6 If you made it down to here, thanks for staying and reading, kudos.

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• loading child swf as3

  - by RichW

  Hi, I've been given an fla to make some changes too. Basically its a fairly long timeline animation with sound. So far I've successfully added a few button functions for sound etc.. but one has got me stumped. One of the buttons needs to load a child swf. I'm using the code below but I'm recieving an error - 'Error #1009: Cannot access a property or method of a null object reference'. I believe this may be refferring to an object that isn't set yet but I have no idea which one it is: Code: var mcExt:MovieClip = new MovieClip(); var ldr:Loader = new Loader(); ldr.contentLoaderInfo.addEventListener(Event.COMPLETE, swfLoaded); ldr.load(new URLRequest("Downloads.swf")); function swfLoaded(e:Event):void { mcExt = MovieClip(ldr.contentLoaderInfo.content); ldr.contentLoaderInfo.removeEventListener(Event.COMPLETE, swfLoaded); mcExt.x = 50; mcExt.y = 50; addChild(mcExt); } Any help on what is going wrong would be greatly appreciated! Thanks

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• iphone-AVAudio Player Crashes

  - by user2779450

  I have an app that uses an avaudio player for two things. One of them is to play an explosion sound when a uiimageview collision is detected, and the other is to play a lazer sound when a button is pressed. I declared the audioplayer in the .h class, and I call it each time the button is clicked by doing this: NSURL *url = [NSURL fileURLWithPath:[[NSBundle mainBundle] pathForResource:@"/lazer" ofType:@"mp3"]]; NSError *error; audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error]; if (error) { NSLog(@"Error in audioPlayer: %@", [error localizedDescription]); } else { [audioPlayer prepareToPlay]; } [audioPlayer play]; This works fine, but after many uses of the game, the audio will stop play when i hit the button, and when a collision is detected, the game crashes. Here is my crash log: 2013-09-18 18:09:19.618 BattleShip[506:907] 18:09:19.617 shm_open failed: "AppleAudioQueue.41.2619" (23) flags=0x2 errno=24 (lldb) Suggestions? Could there be something to do with repeatedly creating an audio player? Alternatives maybe?

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• Application lifecycle and onCreate method in the the android sdk

  - by Leif Andersen

  I slapped together a simple test application that has a button, and makes a noise when the user clicks on it. Here are it's method: @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); Button b = (Button)findViewById(R.id.easy); b.setOnClickListener(this); } public void onClick(View v) { MediaPlayer mp = MediaPlayer.create(this, R.raw.easy); mp.start(); while(true) { if (!mp.isPlaying()) { mp.release(); break; } } } My question is, why is onCreate acting like it's in a while loop? I can click on the button whenever, and it makes the sound. I might think it was just a property of listeners, but the Button object wasn't a member variable. I thought that Android would just go through onCreate onse, and proceed onto the next lifecycle method. Also, I know that my current way of seeing if the sound is playing is crap...I'll get to that later. :) Thank you.

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• Setting Position of source and listener has no effect

  - by Ben E

  Hi Guys, First time i've worked with OpenAL, and for the life of my i can't figure out why setting the position of the source doesn't have any effect on the sound. The sounds are in stero format, i've made sure i set the listener position, the sound is not realtive to the listener and OpenAL isn't giving out any error. Can anyone shed some light? Create Audio device ALenum result; mDevice = alcOpenDevice(NULL); if((result = alGetError()) != AL_NO_ERROR) { std::cerr << "Failed to create Device. " << GetALError(result) << std::endl; return; } mContext = alcCreateContext(mDevice, NULL); if((result = alGetError()) != AL_NO_ERROR) { std::cerr << "Failed to create Context. " << GetALError(result) << std::endl; return; } alcMakeContextCurrent(mContext); SoundListener::SetListenerPosition(0.0f, 0.0f, 0.0f); SoundListener::SetListenerOrientation(0.0f, 0.0f, -1.0f); The two listener functions call alListener3f(AL_POSITION, x, y, z); Real vec[6] = {x, y, z, 0.0f, 1.0f, 0.0f}; alListenerfv(AL_ORIENTATION, vec); I set the sources position to 1,0,0 which should be to the right of the listener but it has no effect alSource3f(mSourceHandle, AL_POSITION, x, y, z); Any guidance would be much appreciated

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• Data Mappers, Models and Images

  - by James

  Hi, I've seen and read plenty of blog posts and forum topics talking about and giving examples of Data Mapper / Model implementations in PHP, but I've not seen any that also deal with saving files/images. I'm currently working on a Zend Framework based project and I'm doing some image manipulation in the model (which is being passed a file path), and then I'm leaving it to the mapper to save that file to the appropriate location - is this common practise? But then, how do you deal with creating say 3 different size images from the one passed in? At the moment I have a "setImage($path_to_tmp_name)" which checks the image type, resizes and then saves back to the original filename. A call to "getImagePath()" then returns the current file path which the data mapper can use and then change with a call to "setImagePath($path)" once it's saved it to the appropriate location, say "/content/my_images". Does this sound practical to you? Also, how would you deal with getting the URL to that image? Do you see that as being something that the model should be providing? It seems to me like that model should worry about where the images are being stored or ultimately how they're accessed through a browser and so I'm inclined to put that in the ini file and just pass the URL prefix to the view through the controller. Does that sound reasonable? I'm using GD for image manipulation - not that that's of any relevance. UPDATE: I've been wondering if the image resizing should be done in the model at all. The model could require that it's provided a "main" image and a "thumb" image, both of certain dimensions. I've thought about creating a "getImageSpecs()" function in the model that would return something that defines the required sizes, then a separate image manipulation class could carry out the resizing and (perhaps in the controller?) and just pass the final paths in to the model using something like "setImagePaths($images)". Any thoughts much appreciated :) James.

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• Convert a Relative URL to an Absolute URL in Actionscript / Flex

  - by Bear

  I am working with Flex, and I need to take a relative URL source property and convert it to an absolute URL before loading it. The specific case I am working with involves tweaking SoundEffect's load method. I need to determine if a file will be loaded from the local file system or over the network from looking at the source property, and the easiest way I've found to do this is to generate the absolute URL. I'm having trouble generating the absolute URL for sound effect in particular. Here were my initial thoughts, which haven't worked. Look for the DisplayObject that the Sound Effect targets, and use its loaderInfo property. The target is null when the SoundEffect loads, so this doesn't work. Look at FlexGlobals.topLevelApplication, at the url or loaderInfo properties. Neither of these are set, however. Look at the FlexGlobals.topLevelApplication.systemManager.loaderInfo. This was also not set. The SoundEffect.as code basically boils down to var url:String = "mySound.mp3"; /*>> I'd like to convert the URL to absolute form here and tweak it as necessary <<*/ var req:URLRequest = new URLRequest(url); var loader:Loader = new Loader(); loader.load(req); Does anyone know how to do this? Any help clarifying the rules of how relative urls are resolved for URLRequests in ActionScript would also be much appreciated. edit I would also be perfectly satisfied with some way to tell whether the url will be loaded from the local file system or over the network. Looking at an absolute URL it would just be easy to look at the prefix, like file:// or http://.

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• Need events to execute on timer events, metronome precision.

  - by user295734

  I setup a timer to call an event in my application. The problme is the event execution is being skewed by other windows operations. Ex. openning and window, loading a webpage. I need the event to excute exactly on time, every time. When i first set up the app, used a sound file, like a metronome to listen to the even firing, in a steady state, its firing right on, but as soon do something in the windows environment, the sound fires slower, then sort of sppeds up a bit to catch up. So i added a logging method to the event to ctahc the timer ticks. From that data, it appears that the timer is not being affected by the windows app, but my application event calls are being affected. I figured this out by checking the datetime.now in the event, and if i set it to 250 milliseconds, which is 4 clicks per second. You get data something like below. (sec):(ms) 1:000 1:250 1:500 1:750 2:000 2:250 2:500 2:750 3:000 3:250 3:500 3:750 (lets say i execute some windows event)(time will skew) 4:122 4:388 4:600 4:876 (stop doing what i was doing in windows) (going to shorten the data for simplicit, my list was 30sec long) 5:124 5:268 5:500 5:750 (you would se the time go back the same milliseconds it was at the begining) 6:000 6:250 6:500 6:750 7:000 7:250 7:500 7:750 So i'm thinking the timer continues to fire on the same millisecond every time, but its the event that is being skewed to fire off time by other windows operations. Its not a huge skew, but for what i need to accomplish, its unacceptable. Is there anyhting i can do in .NET, hoping to use XAML/WPF application, thats will correct the skewing of events? thx.

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• Verizon SongID - How is it programmed?

  - by CheeseConQueso

  For anyone not familiar with Verizon's SongID program, it is a free application downloadable through Verizon's VCast network. It listens to a song for 10 seconds at any point during the song and then sends this data to some all-knowing algorithmic beast that chews it up and sends you back all the ID3 tags (artist, album, song, etc...) The first two parts and last part are straightforward, but what goes on during the processing after the recorded sound is sent? I figure it must take the sound file (what format?), parse it (how? with what?) for some key identifiers (what are these? regular attributes of wave functions? phase/shift/amplitude/etc), and check it against a database. Everything I find online about how this works is something generic like what I typed above. From audiotag.info This service is based on a sophisticated audio recognition algorithm combining advanced audio fingerprinting technology and a large songs' database. When you upload an audio file, it is being analyzed by an audio engine. During the analysis its audio “fingerprint” is extracted and identified by comparing it to the music database. At the completion of this recognition process, information about songs with their matching probabilities are displayed on screen.

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• How to add video into a webpage for mobile web browsers.

  - by payling

  Our company is making a mobile version of our website. We have several product videos we want to show on the mobile version. When I try to use <a href="video.wmv">video</a> I get sound playing but a black screen on my htc incredible android os phone. I'm thinking that the video is playing but in a different browser window. I need it to display all in one window without having to switch to a different window. I tried the html embed tags and get no video or sound at all, from what I've read these tags are not very realiable cross browser. I also just tried the html5 video tags below. I get an icon identifying that it's a video file but it doesn't play. <video src="video.wmv" controls="controls"> your browser does not support the video tag </video> Is there a special format the video file needs to be in? Should I be using the href or embed tags, what other options do I have? If it helps to know, I'm using the mobile doctype on my webpages. Thanks

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• Trimming bit of the beginning off a recorder waveform

  - by Lowgain

  I've got a flash 10.1 app that lets me record microphone input to a wav without a media server, which I am saving to an Amazon S3 bucket. I have another process running on a server which gets wavs from this bucket, converts to mp3 using LAME and puts them into another bucket. This all works fine, but in converting wav mp3, about 0.1sec or so of silence is added to my sound. In the application this are being used in, perfect sync is critical, so I need to trim off that little bit. If I have to trim it off the original waveform that is okay, I don't expect anything important to happen in that first fraction of a second. What is the best way to go about this? I am using Adobe's WavWriter to convert by ByteArray into a proper waveform. Is there a way I can easily trim off the first few samples from my ByteArray without invalidating the structure? Alternatively, is there a good server-side tool I can use to trim the wav before running it through LAME, or an argument I can give LAME? Or, could I even trim that sound off the mp3 after it has been converted? Thanks!

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• Where'd my sounds go?

  - by Dane Man

  In my Row class I have the initWithCoder method and everything restored in that method works fine, but only within the method. After the method is called I loose an array of sounds that is in my Row class. The sounds class also has the initWithCoder method, and the sound plays fine but only in the Row class initWithCoder method. After decoding the Row object, the sound array disappears completely and is unable to be called. Here's my source for the initWithCoder: - (id) initWithCoder:(NSCoder *)coder { ... soundArray = [coder decodeObjectForKey:@"soundArray"]; NSLog(@"%d",[soundArray count]); return self; } the log shows the count as 8 like it should (this is while unarchiving). Then the row object I create gets assigned. And the resulting row object no longer has a soundArray. [NSKeyedArchiver archiveRootObject:row toFile:@"DefaultRow"]; ... row = [NSKeyedUnarchiver unarchiveObjectWithFile:@"DefaultRow"]; So now whenever I call the soundArray it crashes. //ERROR IS HERE NSLog(@"%d",[[row soundArray] count]); Help please (soundArray is an NSMutableArray).

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• Cross-Platform Language + GUI Toolkit for Prototyping Multimedia Applications

  - by msutherl

  I'm looking for a language + GUI toolkit for rapidly prototyping utility applications for multimedia installations. I've been working with Max/MSP/Jitter for many years, but I'd like to add a text-based language to my 'arsenal' for tasks apart from 'content production'. (When it comes to actual media synthesis, my choices are clear [SuperCollider + MSP for audio, Jitter + Quartz + openFrameworks for video]). I'm looking for something that maintains some of the advantages of Max, but is lower-level, faster, more cross-platfrom (Linux support), and text-based. Integration with powerful sound/video libraries is not a requirement. Some requirements: Cross-platform (at least OSX and Linux, Windows is a plus) Fast and easy cross-platform GUIs with no platform-specific modification GUI code separated from backend code as much as possible Good for interfacing with external serial devices (micro-controllers) Good network support (UDP/TCP) Good libraries for multi-media (video, sound, OSC) are a plus Asynchronous synchronous UNIX integration is a plus The options that come to mind: AS3/Flex (not a fan of AS3 or the idea of running in the Flash Player) openFrameworks (C++ framework, perhaps a bit too low level [looking for fast development time] and biased toward video work) Java w/ Processing libraries (like openFrameworks, just slower) Python + Qt (is Qt appropriate for rapid prototyping?) Python + Another GUI toolkit SuperCollider + Swing (yucky GUI development) Java w/ SWT Any other options? What do you recommend?

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• Adding exclusive filter for <static initializer> in findbugs

  - by MilanAleksic

  Hi all, I want my findbugs report not show the following error: DM_NUMBER_CTOR: Method invokes inefficient Number constructor; use static valueOf instead The problem is that this happens in groovy-generated code files, so I can't control the source code - that is why I want to exclude it and add it to my exclude filter. I do not want to add explicitly class (since I make API that many tools will use, I want my filter to be generic). I would not like to completely remove this bug from the report by type, I would really like to only exclude this bug from appearing if it happenned in "static initializer" methods. Any idea? I tried the filter below but no luck, maybe somebody has better idea? <Match> <Method name="~.*static initializer.*" /> <Bug pattern="DM_NUMBER_CTOR" /> </Match> Here is the "stacktrace" of FindBugs in that case: In class net.milanaleksic.cuc.tools.sound.SoundPlayerTool In method net.milanaleksic.cuc.tools.sound.SoundPlayerTool.() Called method new Long(long) Should call Long.valueOf(long) instead In SoundPlayerTool.groovy

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• Unsupported smapling rate in flex/actionscript

  - by Rajeev

  In action script i need Loading configuration file /opt/flex/frameworks/flex-config.xml t3.mxml(10): Error: unsupported sampling rate (24000Hz) [Embed(source="music.mp3")] t3.mxml(10): Error: Unable to transcode music.mp3. [Embed(source="music.mp3")] The code is <?xml version="1.0"?> <!-- embed/EmbedSound.mxml --> <mx:Application xmlns:mx="http://www.adobe.com/2006/mxml"> <mx:Script> <![CDATA[ import flash.media.*; [Embed(source="sample.mp3")] [Bindable] public var sndCls:Class; public var snd:Sound = new sndCls() as Sound; public var sndChannel:SoundChannel; public function playSound():void { sndChannel=snd.play(); } public function stopSound():void { sndChannel.stop(); } ]]> </mx:Script> <mx:HBox> <mx:Button label="play" click="playSound();"/> <mx:Button label="stop" click="stopSound();"/> </mx:HBox> </mx:Application>

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• How do I keep from running out of memory on graphics for an Android app?

  - by user279112

  I've been working on an Android app in Eclipse, and so far, my program hasn't really grown past midget size. However I've already run into an issue with an Out of Memory error. You see, I've been using graphics comprised solely of bitmaps and PNGs in this program, and recently, when I tried to add a little bit more functionality to the program (mainly including a few more bitmaps and causing an extra sprite to be created), it started crashing in the graphics thread's constructor - sprite's constructor. When I tracked the problem down, it turned out to be an Out of Memory error that is seemingly caused by adding too many picture files to the program and creating Drawables out of them. This would be a problem, as I really don't have that many picture resources worked into that program...maybe 20 or so. I haven't even started to include sound yet. These images aren't all that fancy. My questions are this: 1) Are programs for the Android phone really that limited on how much memory they can employ, or is it probably something other than the 20-30 resource pictures causing that error? 2) If the memory for Android apps is so awful it can't even handle 20-30 picture resources being loaded into Drawables that exist at the same time, then how in the world are you supposed to make decent graphics and sound for that thing? Thanks.

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• Swt file dialog too much files selected?

  - by InsertNickHere

  Hi there, the swt file dialog will give me an empty result array if I select too much files (approx. 2500files). The listing shows you how I use this dialog. If i select too many sound files, the syso will show 0. Debugging tells me, that the files array is empty in this case. Is there any way to get this work? FileDialog fileDialog = new FileDialog(mainView.getShell(), SWT.MULTI); fileDialog.setText("Choose sound files"); fileDialog.setFilterExtensions(new String[] { new String("*.wav") }); Vector<String> result = new Vector<String>(); fileDialog.open(); String[] files = fileDialog.getFileNames(); for (int i = 0, n = files.length; i < n; i++) { if( !files[i].contains(".wav")) { System.out.println(files[i]); } StringBuffer stringBuffer = new StringBuffer(); stringBuffer.append(fileDialog.getFilterPath()); if (stringBuffer.charAt(stringBuffer.length() - 1) != File.separatorChar) { stringBuffer.append(File.separatorChar); } stringBuffer.append(files[i]); stringBuffer.append(""); String finalName = stringBuffer.toString(); if( !finalName.contains(".wav")) { System.out.println(finalName); } result.add(finalName); } System.out.println(result.size()) ;

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• Syncing two AS3 NetStreams

  - by Lowgain

  I'm writing an app that requires an audio stream to be recording while a backing track is played. I have this working, but there is an inconsistent gap in between playback and record starting. I don't know if I can do anything to make the sync perfect every time, so I've been trying to track what time each stream starts so I can calculate the delay and trim it server-side. This also has proved to be a challenge as no events seem to be sent when a connection starts (as far as I know). I've tried using various properties like the streams' buffer sizes, etc. I'm thinking now that as my recorded audio is only mono, I may be able to put some kind of 'control signal' on the second stereo track which I could use to determine exactly when a sound starts recording (or stick the whole backing track in that channel so I can sync them that way). This leaves me with the new problem of properly injecting this sound into the NetStream. If anyone has any idea whether or not any of these ideas will work, how to execute them, or some alternatives, that would be extremely helpful! Been working on this issue for awhile

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• why create CLSID_CaptureGraphBuilder2 instance always failed in a machine

  - by Yigang Wu

  It's a real strange issue, the machine information below is from DXDiag. There is no error reported, but create CLSID_CaptureGraphBuilder2 instance always failed in the machine. It's okay to create CLSID_FilterGraph. Before create CLSID_CaptureGraphBuilder2, I have called CoInitialize and created CLSID_FilterGraph. Only this machine has the error, what dll related with this interface or any function needed to call before to make it work? Thanks in advance. System Information Time of this report: 4/24/2010, 09:46:58 Machine name: TURION Operating System: Windows XP Home Edition (5.1, Build 2600) Service Pack 3 (2600.xpsp_sp3_qfe.100216-1510) Language: Japanese (Regional Setting: Japanese) System Manufacturer: To Be Filled By O.E.M. System Model: MS-7145 BIOS: Default System BIOS Processor: AMD Turion(tm) 64 Mobile Technology MT-30, MMX, 3DNow, ~1.6GHz Memory: 768MB RAM Page File: 376MB used, 1401MB available Windows Dir: C:\WINDOWS DirectX Version: DirectX 9.0c (4.09.0000.0904) DX Setup Parameters: Not found DxDiag Version: 5.03.2600.5512 32bit Unicode DxDiag Notes DirectX Files Tab: No problems found. Display Tab 1: No problems found. Sound Tab 1: No problems found. Sound Tab 2: No problems found. Music Tab: No problems found. Input Tab: No problems found. Network Tab: No problems found.

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• implementation musical instrument using audio unit

  - by Develop.Kim

  post same question at apple developer forum ,too hi first sorry that my english is poor.. i want develop iphone application that playing musical instrument like 'ocarina' but don't need blow mic features. so first i tried to find that how implementation 'virtual musical instrument ' in iphone development. the during the decide implementation using 'Audio Unit' to report this article (link) so i want two kind of questions. i recognize that the 'musical instrument' can be divided into three sound that 'attack', 'sustain' , 'release'. 'decay' maybe included (link) . How implementation when audio unit base 'AUInstrumentBase' each sound ? i download sample 'SinSynth' (link) . i want play note this instrument unit for analyze source and study. Is there way to using AULab? expected the way using MIDI input . but i don't have MIDI. in addition, i wonder that i would think it right the way. to ask the advice... thank for reading poor english my article.

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