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  • Bpm audio detection Library

    - by user58815
    I'm looking for a library that simplify tempo/bpm audio detection. Something similar to this http://adionsoft.net/bpm/ , but to use on *NIX machines. Any language, but preference goes to php, perl, python.

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  • Audio processing libraries for Ruby?

    - by J. Pablo Fernández
    Any recommendation on libraries to do audio processing in Ruby. I need to do the following two tasks: Find silences, for which I'm happy to just be able to iterate over each sample in the wave. Cut and paste pieces of wav files to form a new wav file. Convert wav to mp3, which I will probably leave to lame anyway. I'm looking for the equivalent of NAudio, a C# library.

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  • Audible Audio (.aa) file spec?

    - by Adam
    Does anyone know of a good resource on the Audible Audio (.aa) file spec? I'm trying to write a program that can use them, if no one knows of a resource, any tips on reverse engineering the spec my self? I opened it up in a Hex editor and poked around, looks like an MP3 but with a ton more header info.

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  • Printer redirection on server 2003

    - by user137841
    On windows server 2003 when one user connects to the server via RDP the default printer of the server for her profile does not change to the redirected printer of the session. This only happens with the one user all the other users default printers defaults to their session printer automatically. I tried the following solution but there was no \Terminal Server\Printer Redirection in gpedit.msc http://technet.microsoft.com/en-us/library/cc731963(v=ws.10).aspx Computer Configuration\Policies\Administrative Templates\Windows Components\Terminal Services\Terminal Server\Printer Redirection Is there a different place to check the Printer Redirection?

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  • Does Windows Home Server support folder redirection?

    - by Shawn Miller
    Windows provides the ability to redirect specific user folders to server locations, using a group policy extension called Folder Redirection. The Wikipedia article and Microsoft's marketing page for Windows Home Server suggests a tons of great features, but I don't see anything about specifically about Folder Redirection. I currently run a domain controller in the home so that I can push the folder redirection group policy down to all PCs. Windows Home Server looks like a better fit, but I'd hate to give up on the ease of folder redirection to automatically save all documents, music, pictures to the file server. Any thoughts?

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  • same flash file (.swf) downloaded multiple times on a page

    - by Gunjan
    I have a page that has a table with each row corresponding to an audio file. The last cell in each row embeds a simple flash audio player. The problem is that the flash file for the player is being downloaded for each row separately and as soon as rows go beyond 40-50 it crashes the browser. I tried using different players (1pixelout, flash-mp3-player) and the problem is still there, so its not a player specific issue. Is there any way to cache the player so that it is only downloaded once?

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  • Is Objective C fast enough for DSP/audio programming

    - by morgancodes
    I've been making some progress with audio programming for iPhone. Now I'm doing some performance tuning, trying to see if I can squeeze more out of this little machine. Running Shark, I see that a significant part of my cpu power (16%) is getting eaten up by objc_msgSend. I understand I can speed this up somewhat by storing pointers to functions (IMP) rather than calling them using [object message] notation. But if I'm going to go through all this trouble, I wonder if I might just be better off using C++. Any thoughts on this?

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  • C# Audio - How to time stretch (different tempo, same pitch)

    - by heath
    I'm trying to make a winform app in C# (VS2008) that can load an mp3 (other formats would be nice, but mp3 at a minimum) and be able to adjust the playback speed (tempo) without affecting pitch. I really don't need any other audio effects. I tried using DirectShow but that doesn't seem to offer time stretch capabilities. I was able to incorporate irrklang but that does not seem to have the time stretch capability either. So now I've moved on to SoundTouch. That certainly has the capabilities but I'm very unclear on how to implement in C#. After a few days of this, about all I've accomplished is using DLLImport on the SoundTouch DLL and am able to successfully retrieve a version number. At this point, I'm not even sure if I can do what I'm trying to do with SoundTouch. Can anyone offer some guidance either on how to implement SoundTouch or a different library with the capabilities that I'm looking for? Thank you.

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  • Ruby/Rails Audio Conversion Plugins?

    - by coneybeare
    I am looking for a good gem/plugin to convert user-uploaded audio files to different formats. One format in particular that I am interested in is converting to Apple .caf with ima4 compression for inclusion in an iPhone app. I have been using afconvert on my mac for this so far, but I need to do it on my linux box, server-side. Ideally, I would be able to work into paperclip. As an additional solution, ffmpeg could work, but I have not seen any .caf options for it. Anybody know of one?

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  • Downsampling and applying a lowpass filter to digital audio

    - by twk
    I've got a 44Khz audio stream from a CD, represented as an array of 16 bit PCM samples. I'd like to cut it down to an 11KHz stream. How do I do that? From my days of engineering class many years ago, I know that the stream won't be able to describe anything over 5500Hz accurately anymore, so I assume I want to cut everything above that out too. Any ideas? Thanks. Update: There is some code on this page that converts from 48KHz to 8KHz using a simple algorithm and a coefficient array that looks like { 1, 4, 12, 12, 4, 1 }. I think that is what I need, but I need it for a factor of 4x rather than 6x. Any idea how those constants are calculated? Also, I end up converting the 16 byte samples to floats anyway, so I can do the downsampling with floats rather than shorts, if that helps the quality at all.

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  • Java playback of 24 bit audio is incorrect

    - by Paul Hampson
    I am using the javax sound API to implement a simple console playback program based on http://www.jsresources.org/examples/AudioPlayer.html. Having tested it using a 24 bit ramp file (each sample is the last sample plus 1 over the full 24 bit range) it is evident that something odd is happening during playback. The recorded output is not the contents of the file (I have a digital loopback to verify this). It seems to be misinterpreting the samples in some way that causes the left channel to look like it is having some gain applied to it and the right channel looks like it is being attenuated. I have looked into whether the PAN and BALANCE controls need setting but these aren't available and I have checked the windows xp sound system settings. Any other form of playback of this ramp file is fine. If I do the same test with a 16bit file it performs correctly with no corruption of the stream. So does anyone have any idea why the Java Sound API is modifying my audio stream?

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  • Audio -- How much performance improvement can I expect from from reducing function calls by using bu

    - by morgancodes
    I'm working on an audio-intensive app for the iPhone. I'm currently calling a number of different functions for each sample I need to calculate. For example, I have an envelope class. When I calculate a sample, I do something like: sampleValue = oscilator->tic() * envelope->tic(); But I could also do something like: for(int i = 0; i < bufferLength; i++){ buffer[i] = oscilatorBuffer[i] * evelopeBuffer[i]; } I know the second will be more efficient, but don't know by how much. Are function calls expensive enough that I'd be crazy not to use buffers if I care event a tiny bit about performance?

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  • Iphone progressive download audio player

    - by joynes
    Hi! Im trying to implement a progressive download audio player for the iphone, ie using http and fixed size mp3-files. I found the AudioStreamer project but it seems very complicated and works best with endless streams. I need to be able to find out the total length of audiofiles and I also need to be able to seek in the files. I found a hacked deviation from AudioStreamer but it doesnt seem to work very well for me. http://www.saygoodnight.com/?p=14 Im wondering if there is a more simple way to achieve my goals or if there are some better working samples out there? I found the bass library but not much documentation about it. /Br Johannes

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  • Streaming audio - where to start?

    - by Adam Davis
    I need to develop an embedded audio streaming server. Requirements: Voice quality or better Intended for low power wifi transmission Broad support in existing software and devices (ie, windows media player, quicktime, vlc, iPhone, Android, etc). Royalty/patent free, or cheap to license Preferences: Low overhead TCP/IP based streaming protocol Voice grade codec (easy to implement in software, no DSP, 32bit CPU if needed) Would be nice if it supported HTML5 browsers, but is there any codec (such as raw) that is supported by the latest browsers that is lower overhead than MP3? Therefore: What are the relevant streaming protocols I should be looking at? What are the relevant codecs I should be looking at? What transport streams should I be looking at? What am I missing, or where else should I be looking for this type of need?

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  • Android - Audio recorder FileNotFound

    - by david
    Hi, I'm trying to record audio this.recorder = new android.media.MediaRecorder(); this.recorder.setAudioSource(android.media.MediaRecorder.AudioSource.MIC); this.recorder.setOutputFormat(android.media.MediaRecorder.OutputFormat.DEFAULT); this.recorder.setAudioEncoder(android.media.MediaRecorder.AudioEncoder.DEFAULT); this.recorder.setOutputFile("pruebaAudioRecorder.mp4"); this.recorder.prepare(); this.recorder.start(); but when i call prepare method throws the FileNotFound exception. Should I create the file before prepare method? something like new File(...) If so, which should be the file path? thx a lot.

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  • Client-side framework for web-app with good audio support

    - by Poita_
    I'm trying to create a client-side web app that generates music procedurally using some user-input parameters, so I'm looking for a framework (e.g. Flash, Silverlight etc.) that has the capability to play audio at a specified pitch. Whether it is playing a WAV/MP3 file, using MIDI output, or just playing beeps doesn't really matter -- I just need something that will enable me to generate arbitrary music client-side. I've done a bit of searching and it appears that Flash might have the ability to change pitch with the help of a third-part plugin, but I couldn't find anything similar for Silverlight. I can go a try all them out manually if need be, but I thought I'd ask here first just in case anyone had tried something like this before. Thanks in advance

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  • iPhone game audio and background music

    - by Boon
    Have a few questions related to adding sounds to my game, specifically intro music (for splash), background music (loop) and button event sounds. Hope you can share your knowledge on this. 1) Should I use compressed sounds or uncompressed sounds? Or perhaps a combination of the two? Are there any limitations on the iPhone hardware that I should be aware of -- for example, the ability to play multiple compressed sounds? 2) What's the best audio format for my purpose? 3) For background music, I am thinking of using AVAudioPlayer. For button event sounds, I am thinking of using AudioServicesPlaySystemSound, what do you think? 4) Any other issues I should be aware of? Thank you!

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  • directx audio video error message in debugmode

    - by clamp
    I have a c#/winforms application that uses directx to play some video and audio. whenever i start my application in debugmode i get this annoying message. i can click "continue" and everything seems to work fine. but i still want to get rid of this message. it does not show up in releasemode. Managed Debugging Assistant 'LoaderLock' has detected a problem in 'C:\pathtoexe.exe'. Additional Information: DLL 'C:\WINDOWS\assembly\GAC\Microsoft.DirectX.AudioVideoPlayback\1.0.2902.0__31bf3856ad364e35\Microsoft.DirectX.AudioVideoPlayback.dll' is attempting managed execution inside OS Loader lock. Do not attempt to run managed code inside a DllMain or image initialization function since doing so can cause the application to hang.

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  • Naudio - putting audio stream into values [-1,1]

    - by denonth
    Hi all I need to put my audio stream into values of [-1,1]. Can someone tell me a good approach. I was reading byte array and float array from stream but I don't know what to do next. Here is my code: float[] bytes=new float[stream.Length]; float biggest= 0; for (int i = 0; i < stream.Length; i++) { bytes[i] = (byte)stream.ReadByte(); if (bytes[i] > biggest) { biggest=bytes[i]; } } and I don't know how to put values into stream. Because byte is only positive values. And I need to have from [-1,1] for (int i = 0; i < bytes.Count(); i++) { bytes[i] = (byte)(bytes[i] * (1 / biggest)); }

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  • audio power on AudioQueue

    - by Tomoyuki
    Hi everyone. I'm now creating an Application using speech recognition.To check the Audio Power coming in through the microphone, I wrote a method as follows. -(void)checkPower(AudioqueRef)queue{ UInt32 expectedSize= sizeof(AudioQueueLevelMeterState); AudioQueueGetProperty(queue, kAudioQueueProperty_CurrentLevelMeter, audioLevels, expectedSize); NSLog(@"average:%f peak:%f",audioLevels.mAveragePower,audioLevels.mPeakPower); } I found that sometimes mAveragePower was larger than mPeakPower, and when mAveragePower was 1.0, in other words, averagePower is regarded as max, mPeakPower was lower than 1.0. I think that generally this result is inpossible. please Let me know if you have any information about sound power on CoreAudio. thanks.

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  • High level audio crossfading library for python

    - by tcoopman
    I am looking for a high level audio library that supports crossfading for python (and that works in linux). In fact crossfading a song and saving it is about the only thing I need. I tried pyechonest but I find it really slow. Working with multiple songs at the same time is hard on memory too (I tried to crossfade about 10 songs in one, but I got out of memory errors and my script was using 1.4Gb of memory). So now I'm looking for something else that works with python. I have no idea if there exists anything like that, if not, are there good command line tools for this, I could write a wrapper for the tool.

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