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• Alternate play / pause button for WordPress wpaudio soundmanager plugin

  - by j-man86

  Hello! I am using the wpaudio plugin to convert mp3 links into a javascript/flash audio player. My problem is that I use this plugin in two areas on my site: one on a black background, and one on a white background. I need to use an alternate set of play/pause buttons for each page (white buttons for the black background and vice versa). I am at a total loss on how to do this. I need to some how incorporate a "if page is..." statement into the wpaudio.js but I don't know how to do this with jQuery. Can anyone help? Thanks so much!

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• Manipulating multi-track ogg files programatically

  - by Chad Birch

  I'm planning to create a program for manipulating multi-track OGG files, but I don't have any experience with the relevant libraries, so I'm looking for recommendations about which language/library to use for this. I don't really have any preference for the language, I'll happily code it in C, C#, Python, whatever makes things the easiest (or even possible). Perhaps it's even a possibility to automate Audacity somehow? In terms of requirements, I'm not looking for anything particularly fancy. It will probably be a command-line program, I don't need to be able to play the audio, draw image representations of the waveforms, etc. The program will basically be used as a converter, but I need to do some processing before outputting. That is, I need the ability to programatically remove some tracks, set panning per-track, change track volumes, etc. Nothing too complex, just some basic processing, and then output the result in either MP3 or a format easily converted to MP3, such as WAV. Any suggestions or general information would be appreciated, thanks.

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• AudioQueueOfflineRender returning empty data

  - by hyn

  I'm having problems using AudioQueueOfflineRender to decode AAC data. When I examine the buffer after the call, it is always filled with empty data. I made sure the input buffer is valid and packet descriptions are provided. I searched and found that a few others have had the same problem: http://lists.apple.com/archives/Coreaudio-api/2008/Jul/msg00119.html Also, the inTimestamp argument doesn't make sense to me. Why should the renderer care where in the audio the beginning of the buffer corresponds to? The function throws an error if I pass in NULL, so I pass in the timestamp anyway.

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• How to extract semi-precise frequencies from a WAV file using Fourier Transforms

  - by Seisatsu

  Let us say that I have a WAV file. In this file, is a series of sine tones at precise 1 second intervals. I want to use the FFTW library to extract these tones in sequence. Is this particularly hard to do? How would I go about this? Also, what is the bast way to write tones of this kind into a WAV file? I assume I would only need a simple audio library for the output. My language of choice is C

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• Write wave files to memory in Java

  - by Cliff

  I'm trying to figure out why my servlet code creates wave files with improper headers. I use: AudioSystem.write( new AudioInputStream( new ByteArrayInputStream(memoryBytes), new AudioFormat(22000, 16, 1, true,false), memoryBytes.length ), AudioFileFormat.Type.WAVE, servletOutputStream ); taking a byte array from memory containing raw PCM samples and a servlet output stream that gets returned to the client. In the result I get a normal wave file but with zeros in the chunk size fields. Is the API broken? I would think that the size could be filled in using the size passed in the audio input stream. But now, after typing this out I'm thinking its not making this info available to the outer write() method on AudioSystem. It seems like the AudioSystem.write call needs a size parameter unless it is able to pull the size from the stream... which wouldn't work with an arbitrary sized stream. Does anyone know how to make this example work?

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• Why doesnt R.raw.'songname' not work on android devices?

  - by James Rattray

  I have some media (Audio tracks) on an app... With file path 'R.raw.test' I use some code to get it into a mediaplayer... MediaPlayer.create(Textbox.this, R.raw.fly); And it works PERFECTLY on the Android Emulator... (Plays track on click of button) Can someone explain why, when I put it on my Archos (5 IT) it doesnt work at all? -As soon as the button is clicked, it crashes... Do you have to do something to file paths or what? Please help... Thanks alot... James

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• Create mp3 previews from wav and aiff files

  - by August Lilleaas

  I would like to create a program that makes mp3s of the first 30 seconds of an aiff or wav file. I would also like to be able to choose location and length, such as the audio between 2:12 and 2:42. Are there any tools that lets me do this? Shelling out is OK. The application will run on a linux server, so it would have to be a tool that works on linux. I don't mind doing it in two steps - i.e. a tool that first creates the cutout of the aiff/wav, then pass it to a mp3 encoder.

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• Detecting when Bluetooth is disabled on iOS5

  - by Non Umemoto

  I'm developing blog speaker app. I wanna pause the audio when bluetooth is disabled like iPod app. I thought it's not possible without using private api after reading this. Check if Bluetooth is Enabled? But, my customer told me that Rhapsody and DI Radio apps both support it. Then I found iOS5 has Core Bluetooth framework. https://developer.apple.com/library/ios/documentation/CoreBluetooth/Reference/CoreBluetooth_Framework/CoreBluetooth_Framework.pdf CBCentralManagerStatePoweredOff status seems like the one. But, the description says this api only supports Bluetooth 4.0 low energy devices. Did anyone try doing the same thing? I want to support current popular bluetooth headsets, or bluetooth enabled steering wheel on the car. I don't know if it's worth trying when it only supports some brand new bluetooth.

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• AVAudioPlayer currentTime problem

  - by StrAbZ

  Hi, I'm trying to use the audioplayer with a slider in order to seek into a track (nothing complicated). But I have a weird behavior... for some value of currentTime (between 0 and trackDuration), the player stop playing the track, and goes into "audioPlayerDidFinishPlaying:successfully:" with successfully to NO. And it did not go into "audioPlayerDecodeErrorDidOccur:error:" It's like it can't read the time i'm giving to it. For exemple the duration of the track is: 295.784424 seconds i set the currentTime to 55.0s (ie: 54.963878 or 54.963900 or 54.987755, etc... when printed as %f). The "crashes" always happen when the currentTime is 54.987755... and I really don't understand why... So if you have any idea... ^^

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• Unable to call WMP's controls.play() function in VisualBasic

  - by A.J.

  I have the following code: http://pastebin.com/EgjbzqA2 which is basically just a stripped down version of http://www.dreamincode.net/forums/topic/57357-mymusic-player/. I want the program to play one file repeatedly, however, this function doesn't work for some reason. The program plays each file once and then stops. Private Sub Player3_PlayStateChange(ByVal NewState As Integer) Handles Player3.PlayStateChange Static Dim PlayAllowed As Boolean = True Select Case CType(NewState, WMPLib.WMPPlayState) Case WMPLib.WMPPlayState.wmppsReady If PlayAllowed Then Player3.controls.play() End If Case WMPLib.WMPPlayState.wmppsMediaEnded ' Start protection (without it next wouldn't play PlayAllowed = False ' Play track Player3.controls.play() ' End Protection PlayAllowed = True updatePlayer() End Select End Sub

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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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• Architecture of chatroulette

  - by user317163

  Could somebody explain to me the architecture behind chatroulette? I was thinking about a similar project that would only implement Audio support (for starters). Is the best way to set this up a flash server? If so, how should I go about getting into flash, will I need flex 4? I have some beginner experience with c++, c# and java but I have never developed anything for the web. I was also wondering how the randomizer matches up the participants. How would you code something like this. Im obviously pretty clueless here and I'd greatly appreciate some advice regarding this problem -- I don't expect copy and paste solutions. It would just be nice to hear how you guys would tackle this problem. Thank you very much

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• the commands ls and get of ftp are not working in vmware

  - by mnish

  Hi, Iam using vmware player version 3.1 to boot a minix 3 os image. After booting the minix os I want to get some files from a server using ftp. the ftp connection to the server works but when i use the commands "ls" or "get" nothing happens except it says "200 PORT command successful" and it hanges in there. The only thing i can do after typing ls+enter or get+enter is to exit the ftp by using ctrl+c. If anyone knows a solution to this? please help. Thank you

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• Play Shoutcast MP3 radio stream with Python?

  - by Zachary Brown

  I have managed to create an online radio station using Shoutcast and Sam Broadcaster. Now, I am wanting to build my own player for that radio station. I am not sure where to begin, I have googled, but no luck. I am using Python 2.6 on Microsoft Windows. I have managed to capture the stream and save it as an MP# on the hard disk, just not sure what to do with it next. I tried playback of the file, but it always pulls up errors. This is the code I have so far: import urllib target = open("broadcast.mp3") conn = urllib.urlopen("http://78.159.104.175:80") while True: target.write(con.read(5200)) Any help would be greatly appreciated!

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• How to use AVAudioPlayer to play M4P

  - by SimpleCode

  I want to use AVAudioPlayer to play M4p files from iTunes generator. I could use uiWebView to play it and it worked, but I want to put an image of my choice on the background of the player instead of the "quick time" logo. Can you help me as how to do this? Here is an example of the m4p file I want to play with AVAudioPlayer. http://a801.phobos.apple.com/us/r1000/059/Music/b9/0d/be/mzi.zqnndbil.aac.p.m4p Seriously, big thanks in advance!!

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• About data size filled in the buffer

  - by Bohan Lu

  I need low-latency audio in my project, and I know Android 2.3 supports OpenSL ES. I have read documents and sample code and I decide to use Android simple buffer queue to do the play and record. I now try to write a simple application to do the test. However, I have some questions about recording. If I set the recorder stop when it is recording, how do I know the exact number of bytes filled in the last buffer if it is not filled up ? In 1.1 version, the callback function has some parameters about buffer and its filled data, but there is no such parameters in version 1.0.1. Is there any way to get this information ? Any suggestion would be greatly appreciated !

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• Making of a "Babbelbox" where you can speak to for partys

  - by Spidfire

  Ive got a project to make for a party, its called in holland a "Babbelbox". its a computer with a webcam and microphone that can be used to make a kind of video log of everyone who wants to say something about the party. But the problem is that i dont know where to start. ive made a kind of video show system in c but i cant save any data to a good format so it wont jam my harddisk in one hour full. Requirements: Record video + audio Recoding has to start after pressing a button Good compression over the recorded videos (would be even better if it can to be read by final cut pro or premiere pro) Light wight programm would be nice but i could scale up the computer power

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• How to turn off Video Acceleration programatically.

  - by Stefan

  Im using the Windows Media Player OCX in a program runned on hundreds of computers (dedicated). I have found out that when video acceleration is turned on to "full", on some computers it will cause the video to fail to play correct, with green squares between movies and so on. Turn the acceleration to "None" and everything is fine. This program is runned on ~800 computers that will autoupdate my program. So I want to add to the startup to my program that it turns off the video acceleration. The question is, how do I turn off video Acceleration Programatically? All computers are running XP and at least the second service pack. It would take me ages to manually logg in to all those computers and change that setting so thats why I want the program to be able to do it automagically for me.

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• How to read tags out of m4a files in .NET?

  - by dkackman

  I've got some heavily modified code that ultimately came from the Windows Media SDK that works great for reading tags out of MP3 and WMV files. Somewhere along the line, Windows Media Player added support for .m4a files (was it in Windows 7?) but the Windows Media API doesn't seem to reflect that addition (or at least IWMMetadataEditor2::OpenEx pukes on an .m4a file). What would be some good C# code or links on how to dig meta data tags out of m4a files? (Google has come up dry on the C# front.) UPDATE AtomicParsley did indeed end being the best approach. Since that code is a command line tool however I ended up having to create a managed wrapper around some of its functionality in order to use in-process. It is posted on google code if anyone else needs such a thing.

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• How to receive a datastream from a device on your computer, in C#

  - by WebDevHobo

  I plan to build a small audio-recorder app in C#. My laptop has a built in Microphone that's always active, so I want to use that as an early-stage test. I would simply start recording, save the file as a .wav or even use the LAME dll to make it into an MP3. The problem is, I don't know how to contact that microphone. Do I use a library that can detect a device, or do I just catch a stream of bytes from the port that the device is on? I don't have any experience with receiving data from connected devices. I suppose that I'll need to enter all the data into a byte array and then Serialize that into a WAV file, but I'm not sure. Can I get some pointers on this subject?

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• For the iPad or iPhone, how do you control the system Volume? For example, have a button that mutes

  - by SolidSnake4444

  I would like to make a button in my iPad app (probably will be similar to iPhone apps) that when I push this button, all audio is muted, even when you exit the app. I don't see anyway that you can control the volume, although I'm sure other apps have that I have seen in the app store for the iPhone. I also read some places that doing this would reject you from the app store. How could I go about lowering, or highering the volume of the iPad from an app that works even when the app closes? Thank you!

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• How to remove the file which has opened handles?

  - by SKINDER

  PROBLEM HISTORY: Now I use Windows Media Player SDK 9 to play AVI files in my desktop application. It works well on Windows XP but when I try to run it on Windows 7 I caught an error - I can not remove AVI file immediately after playback. The problem is that there are opened file handles exist. On Windows XP I have 2 opened file handles during the playing file and they are closed after closing of playback window but on Windows 7 I have already 4 opened handles during the playing file and 2 of them remain after the closing of playback window. They are become free only after closing the application. QUESTION: How can I solve this problem? How to remove the file which has opened handles? May be exists something like "force deletion"? Thanks.

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• "Winamp style" spectrum analyzer

  - by cvb

  I have a program that plots the spectrum analysis (Amp/Freq) of a signal, which is preety much the DFT converted to polar. However, this is not exactly the sort of graph that, say, winamp (right at the top-left corner), or effectively any other audio software plots. I am not really sure what is this sort of graph called (if it has a distinct name at all), so I am not sure what to look for. I am preety positive about the frequency axis being base two exponential, the amplitude axis puzzles me though. Any pointers?

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• Java stop MIDI playback

  - by user456268

  Hi I have java application which plays midi messages from sequence. I'm doing this using jfugue library. the problem is when I'm tryingto stop playback with stop button (which call sequencer.stop() and sequencer.close()) the last played note is sound all of rest time, and I can't stop it. So I'm asking about solution about stopping all audio and MIDI too! sound playback from java application. Notice: If you want propose just mute volume, you need to know that I want end-use will be able to press play button again and hear the sound again, so muting volumr will be not a solution, or explain please. Thank you!

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• Python frequency detection

  - by Tsuki

  Ok what im trying to do is a kind of audio processing software that can detect a prevalent frequency an if the frequency is played for long enough (few ms) i know i got a positive match. i know i would need to use FFT or something simiral but in this field of math i suck, i did search the internet but didn not find a code that could do only this. the goal im trying to accieve is to make myself a custom protocol to send data trough sound, need very low bitrate per sec but im also very limited on the transmiting end so the recieving software will need to be able custom (cant use an actual hardware/software modem) also i want this to be software only (no additional hardware except soundcard) thanks alot for the help.

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