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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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• 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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• 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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• 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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• Toggling audio on click?

  - by angela

  please look at this fiddle http://jsfiddle.net/rabelais/yLdkj/1/ The above fiddle shows three bars that on hover play audios. How do I change this so the music plays and pauses on click instead. Also if one audio is playing and another is clicked how can the already playing song pause? $("#one").mouseenter(function () { $('#sound-1').get(0).play(); }); $("#one").mouseleave(function () { $('#sound-1').get(0).pause(); }); $("#two").mouseenter(function () { $('#sound-2').get(0).play(); }); $("#two").mouseleave(function () { $('#sound-2').get(0).pause(); }); $("#three").mouseenter(function () { $('#sound-3').get(0).play(); }); $("#three").mouseleave(function () { $('#sound-3').get(0).pause(); });

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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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• audio stream sampling rate in linux

  - by farhan

  Im trying read and store samples from an audio microphone in linux using C/C++. Using PCM ioctls i setup the device to have a certain sampling rate say 10Khz using the SOUND_PCM_WRITE_RATE ioctl etc. The device gets setup correctly and im able to read back from the device after setup using the "read". int got = read(itsFd, b.getDataPtr(), b.sizeBytes()); The problem i have is that after setting the appropriate sampling rate i have a thread that continuously reads from /dev/dsp1 and stores these samples, but the number of samples that i get for 1 second of recording are way off the sampling rate and always orders of magnitude more than the set sampling rate. Any ideas where to begin on figuring out what might be the problem?

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• audio error in vmware running mac os x

  - by PenguinSource

  simple synchronous loading of an audio file (.mp3) in a cocos2d app makes my vmware disconnect the sound. the error is display bottom right, saying 'error in creating sound stream; sound is disconnected' i read that it might be cause of my vmware's version (mine is 8) but I'm looking for a fix, not to downgrade to another version. before i get that error, the sound on the system works just fine (youtube, etc) the exact code im calling is.. [CDSoundEngine setMixerSampleRate: CD_SAMPLE_RATE_MID]; [[CDAudioManager sharedManager] setResignBehavior: kAMRBStopPlay autoHandle:Yes]; soundEngine = [SimpleAudioEngine sharedEngine]; [soundEngine preloadBackgroundMusic:@"somemp3.mp3"]; [soundEngine playBackgroundMusic:@"somemp3.mp3"]; maybe the bit rate is too high .. ? thanks

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• How to calculate the audio file duration in core audio?

  - by mystify

  I have this info variable which is of this type: struct AudioStreamBasicDescription { Float64 mSampleRate; UInt32 mFormatID; UInt32 mFormatFlags; UInt32 mBytesPerPacket; UInt32 mFramesPerPacket; UInt32 mBytesPerFrame; UInt32 mChannelsPerFrame; UInt32 mBitsPerChannel; UInt32 mReserved; }; How could I calculate the total duration of the audio file, in seconds?

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• Seeking not working in HTML5 audio tag

  - by lord_wilmore

  I have a lighttpd server running locally. If I load a static file on the server (through an html5 audio tag), it plays and seeks fine. However, seeking doesn't work when running a dev server (web.py/CherryPy) or if I return the bytes via a defined action url instead of as a static file. It won't load the duration either. According to the "HTTP byte range requests" section in this Opera Page it's something to do with support for byte range requests/partial content responses. The content is treated as streaming instead. What I don't understand is: If the browser has the whole file downloaded surely it can display the duration, and surely it can seek. What I need to do on the web server to enable byte range requests (for non-static urls). Any advice would be most gratefully received.

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• Advice for building a browser-based audio mixer up to 32 tracks

  - by Jonathan P.

  As a personal hobby I am looking to build an online audio mixer where I can upload individual instrument tracks, control individual volumes of each track, and export the mixed down version. I've been trying (and have come pretty close) with javascript. I really would like to stay away from flash if possible, but I'm really looking for suggestions for technologies to try. If anyone has any suggestions on languages that are good at stuff like this or libraries that I am missing, please let me know! I have a test environment that I have been using: http://driverstestpractice.com/sandbox Currently all tracks on the site are set to the click track in order to test the track sync (which as you can tell is a little off)! Thanks!

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• Compare two audio files of beat/tempo and rating in iphone

  - by Senthil Kumar

  Hello, I want to develop iPhone application should have the ability to count the number of phrases that are received when user sing on mic. This application should also have the ability to decipher whether the users phrases are in or out of cadence with a preset beat.When user sing on mic Instrumental music only play. So I have to merge the User Recorded voice with Instrumental music this is one Audio file.Already i have on original Song file.I have to compare both and give the Rating to users. [Note: Instrumental music is without vocal of Original Song file] Can you please help me?. Thanks Vadivelu

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• Capturing Mac OS X System Audio output with Python

  - by richbs

  Hello, I've been trying to "hijack" the Mac OS X system audio using PyAudio and save to a wav in python. That is, I do not want to record from an input device such as a microphone. I want to grab the sound output from any or all applications. I have followed the tutorials on the PyAudio site but these do not appear to cover my use case and when I try to read from the output stream I unsurprisingly get the paCanNotReadFromAnOutputOnlyStream exception. Fair enough! Is there a way to do what I am proposing with the PyAudio or other FOSS Python Library?

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• Analysis Services with excel as front end - is it possible to get the nicer UI that powerpivot provi

  - by AJM

  I have been looking into PowerPivot and concluded that for "self service BI" and ahoc buidling of cubes it has its uses. In particular I like the enhanced UI that you get from using PowerPivot rather than just using a PivotTable hooked up to an analysis services datasource. However it seems that hooking up PowerPivot to an existing analysis services cube is not a solution for "organisational BI". It is not always desireable to suck millions of rows into excel at once and the interface between PowerPivot and analysis services is very poor in my book. Hence the question is can an existing analysis services solution get the enhanced ui features that power pivot brings, withoout using powerpivot as the design tool? If powerpivot is aimed ad self service/personal BI then it seems bizare that the UI for this is better than for bigger/more costly analysis services solutions.

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• Specify a custom dictionary for FxCop and Visual Studio source analysis

  - by Marko Apfel

  Renaming the default custom dictionary from CustomDictionary.xml to an other name – for instance FxCop.CustomDictionary.xml needs some additional changes to work in involved applications. Visual Studio Team System code analysis For Visual Studio Team System code analysis this file should be added as a link to all projects and setted to be the Build Action CodeAnalysisDirectory. Build target In a build target the command line tool FxCopCmd should be called with the /dictionary parameter: <Target Name="FxCop"> <Exec Command="&quot;$(ProjectDir)..\..\build\FxCop\FxCopCmd.exe&quot; /file:&quot;$(TargetPath)&quot; /project:&quot;$(ProjectDir)..\EsriDE.SfgPraxair.FxCop&quot; /directory:&quot;$(ProjectDir)..\..\lib\Esri.ArcGIS&quot; /directory:&quot;$(ProjectDir)..\..\lib\Microsoft&quot; /dictionary:&quot;$(ProjectDir)..\FxCop.CustomDictionary.xml&quot; /out:&quot;$(OutDir)..\$(ProjectName).FxCopReport.xml&quot; /console /forceoutput /ignoregeneratedcode"> </Exec> <Message Text="FxCop finished." /> </Target> FxCop-GUI (standalone application) In FxCop-GUI is no option to specify an own file name – but you could add a hint in the FxCop project file. Open your this file and look for the line: <CustomDictionaries SearchFxCopDir="True" SearchUserProfile="True" SearchProjectDir="True" /> Then change it to: <CustomDictionaries SearchFxCopDir="True" SearchUserProfile="True" SearchProjectDir="True"> <CustomDictionary Path="FxCop.CustomDictionary.xml"/> </CustomDictionaries> Ready :-)

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• Great Example of a Simple Cost-Benefit Analysis

  - by BuckWoody

  I saw a post the other day that you should definitely go check out. It’s a cost/benefit decision, and although the author gives it a quick treatment and doesn’t take all points in the decision into account, you should focus on the process he follows. It’s a quick and simple example of the kind of thought process we should have as data professionals when we pick a server, a process, or application and even platform software. The key is to include more than just the price of a piece of software or hardware. You need to think about the “other” costs in the decision, and then make the right one. Sometimes the cheapest option is the cheapest, and other times, well, it isn’t. I’ve seen this played out not only in the decision to go with a certain selection, but in the options or editions it comes in. You have to put all of the decision points in the analysis to come up with the right answer, and you have to be able to explain your logic to your team and your company. This is the way you become a data professional, not just a DBA. You can check out the post here – it deals with Azure, but the point is the process, not Azure itself: http://blogs.msdn.com/eugeniop/archive/2010/03/19/windows-azure-guidance-a-simplistic-economic-analysis-of-a-expense-migration.aspx Share this post: email it! | bookmark it! | digg it! | reddit! | kick it! | live it!

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• C#: Streaming an Audio file from a Server to a Client

  - by Andreas Grech

  I am currently writing an application that will allow a user to install some form of an application (maybe a Windows Service) that will open a port on it's PC and given a particular destination on the hard disk, will then be able to stream mp3 files. I will then have another application that will connect to the server (being the user's pc) and be able to browse the hosted data by connecting to that PC (remotely ofcourse) given the port, and stream mp3 files from the server to the application I have found some tutorials online but most of them are about File Servers in C# and they download allow you to download a whole file. What I want is to stream an mp3 file so that it starts playing when a certain number of bytes are download (ie, whilst it is being buffered) How do I go about in accomplishing such a task? What I need to know specifically is how to write this application (that I will turn into a Windows Service later on) that will listen on a specified port a stream files, so that I can then access the files by something of the sort: http://<serverip>:65000/acdc/wholelottarosie.mp3 and hopefully be able to stream that file in a WPF MediaPlayer. [Update] I was following this tutorial about building a file server and sending the file from the server to the client. Is what I have to do something of the sort? [Update] Currently reading this post: Play Audio from a Stream using C# and I think it looks very promising as to how I can play streamed files; but I still don't know how I can actually stream the files from the server.

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• How to play simultaneous multiply audio sources in Silverlight

  - by Shurup

  I want to play simultaneous multiply audio sources in Silverlight. So I've created a prototype in Silverlight 4 that should play a two mp3 files containing the same ticks sound with an intervall 1 second. So these files must be sounded as one sound if they will be played together with any whole second offsets (0 and 1, 0 and 2, 1 and 1 seconds, etc.) I my prototype I use two MediaElement (me and me2) objects. DateTime startTime; private void Play_Clicked(object sender, RoutedEventArgs e) { me.SetSource(new FileStream(file1), FileMode.Open))); me2.SetSource(new FileStream(file2), FileMode.Open))); var timer = new DispatcherTimer { Interval = TimeSpan.FromMilliseconds(1) }; timer.Tick += RefreshData; timer.Start(); } First file should be played at 00:00 sec. and the second in 00:02 second. void RefreshData(object sender, EventArgs e) { if(me.CurrentState != MediaElementState.Playing) { startTime = DateTime.Now; me.Play(); return; } var elapsed = DateTime.Now - startTime; if(me2.CurrentState != MediaElementState.Playing && elapsed >= TimeSpan.FromSeconds(2)) { me2.Play(); ((DispatcherTimer)sender).Stop(); } } The tracks played every time different and not simultaneous as they should (as one sound).

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• Procesing 16bit sample audio

  - by user2431088

  Right now i have an audio file (2 Channels, 44.1kHz Sample Rate, 16bit Sample size, WAV) I would like to pass it into this method but i am not sure of any way to convert the WAV file to a byte array. /// <summary> /// Process 16 bit sample /// </summary> /// <param name="wave"></param> public void Process(ref byte[] wave) { _waveLeft = new double[wave.Length / 4]; _waveRight = new double[wave.Length / 4]; if (_isTest == false) { // Split out channels from sample int h = 0; for (int i = 0; i < wave.Length; i += 4) { _waveLeft[h] = (double)BitConverter.ToInt16(wave, i); _waveRight[h] = (double)BitConverter.ToInt16(wave, i + 2); h++; } } else { // Generate artificial sample for testing _signalGenerator = new SignalGenerator(); _signalGenerator.SetWaveform("Sine"); _signalGenerator.SetSamplingRate(44100); _signalGenerator.SetSamples(16384); _signalGenerator.SetFrequency(5000); _signalGenerator.SetAmplitude(32768); _waveLeft = _signalGenerator.GenerateSignal(); _waveRight = _signalGenerator.GenerateSignal(); } // Generate frequency domain data in decibels _fftLeft = FourierTransform.FFTDb(ref _waveLeft); _fftRight = FourierTransform.FFTDb(ref _waveRight); }

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